Kofax Communication Server 10.3.0 Guide
Reklamë
Reklamë
Kofax Communication Server
VoIP / FoIP Environment Guide
Version: 10.3.0
Date: 2019-12-13
© 2019 Kofax. All rights reserved.
Kofax is a trademark of Kofax, Inc., registered in the U.S. and/or other countries. All other trademarks are the property of their respective owners. No part of this publication may be reproduced, stored, or transmitted in any form without the prior written permission of Kofax.
Table of Contents
3
Kofax Communication Server VoIP / FoIP Environment Guide
4
Kofax Communication Server VoIP / FoIP Environment Guide
Concurrent Operation with KCS H.323 Integration for Voice on the Same Machine...........166
Check the ISDN Line Synchronization Up to the CISCO Gateway (BERT)........................ 166
5
Preface
This guide describes integration of Kofax Communication Server with Fax over IP and Voice over IP.
6
Related Documentation
This document refers to the following documentation:
• Voice Platform Technical Manual
• Fax over IP Technical Manual
For information about Cisco products, see the Cisco documentation.
• Cisco Fax Relay Trouble Shooting Guide: http://www.cisco.com/en/US/customer/tech/tk652/tk777/ technologies_tech_note09186a0080114565.shtml (free http://www.cisco.com registration required)
• Cisco IOS Documentation (use the search function at http://www.cisco.com to look up IOS configuration parameters or topics)
• Cisco Feature Navigator (http://www.cisco.com/go/fn)
7
Chapter 1
Overview
This chapter provides an overview of Fax over IP integration and supplementary services.
Fax over IP Integration
Third Party IP System
Alcatel PABX
Cisco Gatekeeper
Cisco Gateway
Cisco CM 4.x + Cisco GW
Cisco UCM 6.x + Cisco GW
Huawei SoftX3000-Softswitch
Mitel 3300 Voice
Patton GW R4.2
Patton GW R5.2
Protocol(s) Min.
FoIPv3 version
SIP
H.323+RAS (Term.) 3.02.03
Outbound
Modes
Inbound Modes
10 20 40 60 10 20 40 60
3.02.03 # # #
H.323+RAS (GW)
H.323, SIP
H.323
H.323, SIP
Dialogic DMG300x or DMG 400x SIP
SIP
SIP
H.323, SIP
H.323
SIP
3.03.01
3.02.03 # # #
3.02.03 #
3.02.03 #
3.11.06 # #
3.10.00 # #
3.09.04 #
3.02.03 # # #
3.02.03 #
3.03.02
#
#
#
#
#
#
#
#
#
#
#
#
#
#
Siemens HiPath 4000 V4
Siemens HiPath 4000 V6
Siemens OpenScape Voice V3.1
with Siemens RG87xx
Siemens OpenScape Voice V3.1
with Mediatrix 4404
H.323, SIP
H.323, SIP
SIP+Reg
SIP+Reg
3.02.03 # # #
3.13.03 # # #
3.02.03 # # #
3.02.03 # # #
# # #
# # #
# # #
# # #
Notes
5)
5)
5)
#14637,
5)
1), 5)
5)
2), 5)
5)
5)
5)
5)
5)
5)
3), 5)
5)
5)
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Kofax Communication Server VoIP / FoIP Environment Guide
Third Party IP System Protocol(s)
Siemens OpenScape Voice V4 with Mediatrix 4402 with Mediatrix 4124 with Mediatrix 1104 with AP1120 with RG8702
Siemens OpenScape Voice V6 with Mediatrix 4104 with Mediatrix 4404 with RG8702 with OSV Branch 50i
Siemens OpenScape Voice V7 with Mediatrix 4124 with Mediatrix 4404 with RG8702 with OSV Branch 50i
Siemens OpenScape Voice V8 with Mediatrix 4124 with Mediatrix 4404 with RG8702 with OSV Branch 50i
Swisscom SIP trunk
Avaya Aura CM 5.2.1
with SES for SIP
Avaya Aura CM 6.0.1
With Session Manager for SIP
Avaya Aura CM 6.2
With Session Manager for SIP
Avaya Aura CM 6.3
With Session Manager for SIP
Avaya Aura CM 7.0
With Session Manager for SIP
Avaya Aura CM 7.0.1
With Session Manager for SIP
SIP+Reg
SIP+Reg
SIP+Reg
SIP+Reg
SIP
SIP+H.323
SIP+H.323
SIP+H.323
SIP+H.323
SIP+H.323
SIP+H.323
Inbound/(Outbound Modes :
10 = Switch to T.38 w/o G.711 pass-through support
20 = Try T.38 with a fallback to G.711 pass-through
Min.
FoIPv3 version
Outbound
Modes
Inbound Modes
10 20 40 60 10 20 40 60
3.10.07 # # # # # #
Notes
5)
3.13.03 # # #
3.23.00
#
3.12.07 # # #
3.12.07 # # #
#
# # #
3.18.03 # # # # # # # #
3.24.08 # # # # # # # #
#
# # #
# # #
#
3.16.20 # # # # # # # #
3.22.05 # # # # # # # #
3.24.22 # # # # # # # #
3.26.11 # # # # # # # #
5)
4), 5)
4), 5)
4)
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Kofax Communication Server VoIP / FoIP Environment Guide
40 = Use G.711 pass-through unless T.38 is requested by remote side (default)
60 = Use G.711 pass-through and prevent switch to T.38
Symbol usage :
# Supported and most recommended option
# Supported
# Not supported
Notes :
1) Minimum software release on Siemens HiPath 4000 V4, HG3550 is HG35xx LW pzksti40.o4.019-15.
T.38 does not work reliable with older releases. See Hint 14385 or Siemens problem number
NA03559307.
2) Minimum software release on Siemens RG8702 Gateway is V1.3. T.38 did not work with older releases.
3) Inbound calls with V.34 may fail due to SPR70456.
4.) MWI does not work.
5.) The columns with the supported inbound/outbound mode does not consider G.711 pass-through mode.
This means that operation may be supported even if the column is marked as “not supported”.
Supplementary Services - Redirecting Number (Call Diversion)
Redirecting numbers can be delivered in various ways. We have tested four implementations for H.323
protocol and one SIP implementation:
H.450.3
Q.931
Cisco CM prop.
Cisco GW prop.
-
Tunneled Signaling Message +
+
+
VoIP GW 2600 12.3(11) GW 2600 12.3(7)
+ +
-
-
-
+
-
-
-
-
SIP diversion header + ?
?
-
+
?
-
-
-
GW 2821
-
-
+
+
+
-
CCM6 HiPath 4000
-
-
-
-
+
+
The first two implementations are standardized. Cisco CallManager and Cisco gateways have a proprietary implementation.
The Q.931 standard can deliver only a single number. The H.450.3 standard delivers two numbers: the first one (original called number) and the last one (diverting number).
On Cisco gateways, the support of various implementations may depend on the IOS version.
10
Chapter 2
Integration with Gateways
This chapter provides information about integration with gateways.
Cisco Gateways
This section provides information about integration with Cisco gateway.
FoIP Integration
This section describes a setup example of Kofax FoIP solution with a Cisco Router as VoIP/FoIP gateway.
The installation is done in two steps.
1.
Configuring the gateway. A VoIP application can be used to test if the gateway is working correctly.
2.
Installing FoIP instead of the VoIP application.
Prerequisites
• A voice and fax over IP gateway which supports H.323 or SIP and T.38.
If you use a Cisco router as gateway, you need a proper IOS software version installed on the router.
The third party products we use for testing can be found in the
section. Any problems arising with different products or versions are not supported by Kofax.
In this guide we assume that the LAN interface and the ISDN interfaces are properly configured. Refer to the Cisco documentation for details.
Tip
If you use a Cisco gateway and you want to know if it is compatible to TC/FoIP you can use the Cisco
Feature Navigator ( http://www.cisco.com/go/fn ). It is required to register an account at Cisco website before you can use it. The Feature Navigator lets you select a particular feature and shows you a complete list all Cisco products, which support this feature. Select the feature “T.38 Fax Relay for
VoIP H.323”. If your Cisco hardware appears in the list, your gateway is compatible to TC/FoIP.
• Windows computer with a static IP address (preferably the TCOSS server; with soundcard)
Configuring the Gateway
This section describes how to set up the gateway. An overview of this step is shown below:
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Kofax Communication Server VoIP / FoIP Environment Guide
1.
Boot the gateway and plug in the ISDN line to the ISDN BRI S/T 0-Port VIC 2B NT/TE. The OK-
LED must be activated. Make sure that the gateway has a LAN connection (10 or 100 Mbps) on the
FastEthernet 0/0 interface.
2.
Initially, a Cisco gateway is configured over the serial port. Connect the "CONSOLE" port of the gateway to a machine where a terminal application is running. Baud rate: 9600; Data 8 bit; Parity: none; stop: 1 bit; Flow control: none.
Hint: If the gateway already has an IP configuration it can also be accessed via telnet or SSH. If you do that, you have to enter “terminal monitor” to see traces in the telnet session.
If the gateway is booted and you press enter in the terminal, the standard prompt appears which ends with ">". Type "enable" and type the enable password. Then type "configure terminal" to enter the configuration mode. The configuration prompt appears which ends with "(config)#".
Cisco2620>
Cisco2620>enable
Password:
Cisco2620#configure terminal
Enter configuration commands, one per line. End with CNTL/Z.
Cisco2620(config)#
By typing "show run" in the enable mode you can display the configuration which is currently running.
It contains all configured interfaces, services, dial-peers and more. If you type "?" you get a list of commands available in the current context. Typing "exit" gets you back to the previous mode. In the configuration mode you have to precede the “show” command with “do” (“do show run”). Auto completion and “?” are not available in this mode.
The next two steps are only necessary if your gateway has no ISDN interface configured and no voice service defined yet. If you configure on a gateway where this is already set up correctly, you can continue with step 7.
3.
Configure the gateway’s LAN interface for H.323. This step is not necessary when you use SIP. Type
"interface FastEthernet 0/0" in the configuration mode. You will see the following prompt.
Cisco2620(config)#interface FastEthernet 0/0
Cisco2620(config-if)#
Let’s assume that the IP address of the gateway is 10.11.11.34. Type "h323-gateway voip interface" and "h323-gateway voip bind srcaddr 10.11.11.34". Then type "exit" to return to the configuration mode. If you type "show" something like this must appear in the configuration.
interface FastEthernet0/0
ip address 10.11.11.34 255.0.0.0
no ip mroute-cache
speed auto
half-duplex
h323-gateway voip interface
h323-gateway voip bind srcaddr 10.11.11.34
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Kofax Communication Server VoIP / FoIP Environment Guide
4.
Configure a VoIP/FoIP service on the gateway. As a prerequisite for the dial-peer created in the next step, a service is needed on the gateway. Type "voice service voip" in the configuration mode. The following prompt appears. You are now in the configuration menu for this service.
Cisco2620(config)#voice service voip
Cisco2620(conf-voi-serv)#
Configure the fax protocol by typing "fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback none". If you use H.323, type "h323". With the command "exit" you can return to the configuration mode. Check the service with show.
voice service voip
fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback none
h323
5.
Configure a dial-peer on the gateway. A dial peer is used to assign an MSN to an IP address.
If you want to associate the Windows machine (10.168.8.35) with the MSNs 8000 – 8999 (this means that the gateway makes a connection to 10.168.8.35 if a call to one of the MSNs 8000 - 8999 comes in from ISDN) you have to do the following. Type "dial-peer voice 8000 voip" to create a new dialpeer with the ID 8000
You are now in the configuration mode for this dial-peer. After configuring type "exit" to return to configuration mode. If you type "show" the dial-peer must appear in the configuration like this.
dial-peer voice 8000 voip
destination-pattern 8...
session target ipv4:10.168.8.35
dtmf-relay h245-alphanumeric
codec g711alaw
fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback none
no vad
Note The number specified at the destination-pattern (8…) in our example is used to match the dial-peer on an inbound call to this number but also on an outbound call due to the calling party number which the TCOSS server sets. “8…” like in this example matches exactly to inbound calls to number 8000 - 8999 and outbound calls from calling party numbers 8000 - 8999. In this example we use wildcards ‘.’ for exactly one arbitrary digit. You can also specify ‘T’ (interdigit timeout) for any number of arbitrary digits. For details please search for “Configuring Dial Plans, Dial Peers, and Digit Manipulation” on http://www.cisco.com/ and then look for “Destination Pattern” in this document.
If you use SIP you need to specify the protocol with “session protocol sipv2”and select “dtmf-relay sip-notify”.
dial-peer voice 8000 voip
destination-pattern 8...
session target ipv4:10.168.8.35
session protocol sipv2
dtmf-relay sip-notify
codec g711alaw
fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback none
no vad
6.
Save the configuration to the router by typing "write" in the enabled mode. This saves the changes to the startup configuration, i.e., that they are not lost at the next restart of the gateway.
Cisco2620#write
Building configuration...
[OK]
Cisco2620
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Kofax Communication Server VoIP / FoIP Environment Guide
Now that the gateway is configured, it is recommended to test the configuration with MyPhone. See
MyPhone (H.323 Telephone Software) .
Configuring FoIPv3
Open the KCS FoIP configuration tool and configure a call peer:
1.
Select the signaling protocol you use: H.323 or SIP.
2.
In the field Remote Address/Host, type the gateway IP address or host name.
3.
In the field Set Remote Address\Port, type the port number if you use a port other than the wellknown ports (H.323: 1720, SIP: 5060).
4.
Other fields are optional.
An example configuration is shown below:
Troubleshooting Hints
This section provides troubleshooting information.
Check ISDN Line
This works only with PtMP BRI lines, because common ISDN telephones do not support PtP lines directly.
Connect an ISDN telephone to the BRI interface. It must have power and a dial tone must be heard. Try to make a call to any other phone. Try to call the phone from another phone. The phone must be configured to accept the MSN that you dial.
Check ISDN Interface on the Gateway
If you unplug the ISDN line from the interface the gateway should display something like this on the terminal.
16:12:227633266687: %ISDN-6-LAYER2DOWN: Layer 2 for Interface BRI1/0, TEI 68 changed to
down
16:12:52: %ISDN-6-LAYER2DOWN: Layer 2 for Interface BR1/0, TEI 68 changed to down
16:12:52: %LINK-3-UPDOWN: Interface BRI1/0, changed state to down
This should appear, if you plug in the line. The OK-LED should be activated.
16:12:59: %ISDN-6-LAYER2UP: Layer 2 for Interface BR1/0, TEI 69 changed to up
16:12:59: %LINK-3-UPDOWN: Interface BRI1/0, changed state to up
If this is not the case, there might be something wrong with the ISDN interface on the gateway or the configuration of the interface.
When you call the interface from a telephone (i.e. dial 8000), the B1- or B2-LED should be activated on the VIC.
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Kofax Communication Server VoIP / FoIP Environment Guide
Disable Faststart on the Gateway
If you have mysterious problems with call establishment try to add the red config line to the gateways global voice service (see above how this is done).
voice service voip
...
h323
call start slow
Check Fax Relay and Signaling Activity on Gateway
You can enable debug traces on the gateway and then attempt to send a fax.
To enable T.38 trace, activate the enabled mode and enter "debug fax t30 all". With "debug fax ?" and
"debug fax t30 ?" you get a list of possible levels. Sending a single page fax from the IP to the ISDN side created the following trace.
Cisco2620#debug fax relay t30 all
Debugging fax relay t30
Cisco2620#
5d04h: %ISDN-6-CONNECT: Interface BRI1/0:1 is now connected to unknown
5d04h: 1/0/0 (24965) 447214020 fr-entered (10ms)
5d04h: 1/0/0 (24965) 447214700 fr-msg-det CSI
5d04h: 1/0/0 (24965) 447215390 fr-msg-det DIS
5d04h: 1/0/0 (24965) 447216710 fr-msg-tx TSI
5d04h: 1/0/0 (24965) 447217570 fr-msg-tx DCS
5d04h: 1/0/0 (24965) 447222490 fr-msg-det CFR
5d04h: 1/0/0 (24965) 447281510 fr-msg-tx EOP
5d04h: 1/0/0 (24965) 447283060 fr-msg-det MCF
5d04h: 1/0/0 (24965) 447284280 fr-msg-tx DCN
5d04h: %ISDN-6-DISCONNECT: Interface BRI1/0:1 disconnected from N/A N/A, call lasted
77 seconds
5d04h: %ISDN-6-DISCONNECT: Interface BRI1/0:1 disconnected from unknown , call lasted
77 seconds
5d04h: 1/0/0 (24965) 447286290 fr-end cause unknown 0x1
By entering "debug ?" you will get a list of all debug levels. The levels for h225 and h245 could also give relevant information for troubleshooting.
You can disable a debug level by entering the same command that enabled it with at preceding "no".
Example "no debug fax relay t30 all".
Note If you access the gateway via telnet, you have to enter “terminal monitor” to see the debug output on the telnet session.
Check If Correct Parameters (Codec, Etc.) Are Used for Calls
Make a call (outgoing or incoming) and type “show call active voice compact” while the call is active to find out if the parameters (codec, protocol, vad, etc.) from the associated dial peer are used.
This is the output for one active outgoing call (from FoIPv3 to an ISDN telephone). The call shows two lines, which represent the two calls legs. The first line is the call leg to FoIPv3, the second line is the call leg to the ISDN device with the number 0172.
Cisco2620#show call active voice compact
G<id> A/O FAX T<sec> Codec type Peer Address IP R<ip>:<udp>
Total call-legs: 2
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Kofax Communication Server VoIP / FoIP Environment Guide
G202 ANS T1 g711alaw VOIP P8000 10.168.8.35:8002
G202 ORG T2 g711alaw TELE P0172
Check Errors on the ISDN Interfaces of the Gateway
If received faxes sent over the gateway have missing pixel lines or the fax transmissions are even interrupted (mostly XL on sender side) or if interruptions and other distortions can be heard when making a call to FoIPv3 over the gateway with a telephone, then the ISDN line or the ISDN interface on the gateway might have a problem. These problems can be caused by defective hardware or clock synchronization problems (ISDN configuration of the interface).
By typing “show controller bri” for basic rate interfaces or “show controller E1” for primary rate lines, ISDN dependent line or interface problems on the gateway can be detected. Examine the error counters and the clockmode settings shown.
Check Active Calls for Packet Loss, Jitter and Network Delay on the Gateway
If you have irregular interruptions in sent or received faxes you should take a look at the LAN quality. The
Cisco gateways offer a simple method to check the jitter and network delay for received RTP data on active calls.
To do this connect to the gateway with telnet, enter the enabled mode and activate the debugging by typing “terminal monitor”. Send a multi-page test fax and then type “show voice call summary”. You will see a table with all open calls on the gateway. In the first column you can see the used interface, port and
B-channel.
Cisco2620#show voice call summary
PORT CODEC VAD VTSP STATE VPM STATE
============== ======== === ==================== ======================
1/0/0.1 - - -
1/0/0.2 14400 n S_FAX S_TSP_CONNECT
1/0/1.1 - - -
1/0/1.2 - - -
Pick your test call from the list and display the call’s DSP statistics by typing for example “show voice call
1/0/0.2”.
Cisco2620#show voice call 1/0/0.2
1/0/0 2
vtsp level 0 state = S_FAX callid 0x0372 B02 state S_TSP_CONNECT clld 8400 cllg 4318635348
Cisco2620# ***DSP FAX RELAY STATISTICS***
Max Jit Depth: 23, Max Nwk RxQ Depth 1, Jitter Overflow Pkt Drops: 0
Nwk RxQ Overflow: 0, Tx Pkts: 432, Tx Pkts Drops(Nwk Busy): 0
Rx Pkts: 11, Rx Pkts Loss: 0, Rx Invalid Pkts: 0, Rx Pkts Out Of Seq: 0
Recent Hi-Speed Modulation: 4TX Pages: 0
Max SendInQ Depth 2, Max RecvOutQ Depth 0
Max Hi-Speed Buf Usage 5SendInQ Overflow 0,RecvOutQ Overflow 0
Cisco2620#
Repeat this command about every 5 seconds while the call is active and keep watching the value Jitter
Overflow Pkt Drops. It is normal that this value increases a little at the beginning of a call. If it increases steadily until the end of the call, you probably have a serious problem on the LAN side.
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Kofax Communication Server VoIP / FoIP Environment Guide
Redirecting Number (Call Diversion) Using H.323
Cisco gateways 2620 and 2821 were used to test Call Diversion with H.450.3. The Cisco gateway 2821 with IOS version 12.4(10) delivers the redirecting number in an H.221 non-standard identifier (different to the non-standard identifier the Cisco CallManager uses). Also the Cisco gateway 2620 with IOS version 12.3(11) uses a non-standard identifier to transmit the redirecting number. With IOS version
12.3(7) the gateway uses H.450.3 to deliver the redirecting number. With H.450.3 it is possible to transmit two numbers when using multiple forwarding. The first redirecting number, to be termed original called number, and the last redirecting number, to be termed diverting number.
SIP VoIP IPV6 Integration
Cisco gateways with IOS version 12.4(22) or later, support SIP VoIP signaling using IPV6 protocol.
Given the gateway has at least one LAN interface IPV6 enabled and connected to the IPV6 network, SIP protocol can be enabled for both IPV4 and IPV6 in this way:
!
sip-ua protocol mode dual-stack
!
Then configure dial-peer(s) as usual, either using symbolic names or direct IPV6 addresses.
In the case IPV6 addresses are used, they must be enclosed in the “[]” brackets: dial-peer voice 8000 voip destination-pattern 8...
session protocol sipv2 session target ipv6:[fd96:eb5f:7508:5760:202:b3ff:feb8:fb29] dtmf-relay rtp-nte codec g711alaw fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback none no vad
Hints for Cisco Gateways
With Cisco gateway 28xx and IOS 12.4T(22) the T.38 FoIP does not work via IPV6 protocol.
The problem is that the UDP packets generated for T.38 stream by the gateway do not have the UDP checksum set (UDP checksum is always set to zero), which is mandatory with IPV6 protocol.
As a consequence, receiving FoIPV3 running on Windows 2008 server doesn’t receive any T.38 packets from the gateway as these packets are obviously being silently ignored by Windows IPV6 protocol stack due to missing UDP checksum.
Dialogic Media Gateway DMG4000 Series
Dialogic produces VoIP/FoIP media gateways which use a Windows Server 2008 operating system and a
Dialogic Diva PCI fax board for ISDN connectivity. The SIP integration is done via DIVA SipControl. The gateway can be managed like any other Windows computer; it includes nice features like remote desktop
17
Kofax Communication Server VoIP / FoIP Environment Guide connection and Wireshark network protocol analyzer. The main benefit of these gateways is that they support V.34 fax via T.38.
Hints for Dialogic Media Gateway
Integration Hints
• Activation of T.38 did not worked correctly if the T.38 was activated by the Gateway. The problem was that Gateway signaled T.38 but continued with G.711 packets. Therefore only mode 10 (=default) is support both for fax inbound and fax outbound.
• Inbound calls with V.34 may fail due to SPR70456 (which is fixed since 3.12.02)
Additional Hints Specific for DMG300x
Dialogic also delivered Media Gateway DMG300x. It is the predecessor of DMG400x, also supports
V.34 via T.38 but is now out of sales. All hints for DMG400x also apply to DMG300x, but here are some additional hints specific for DMG300x:
• The DMG3008BRI (Diva System 8.5.6 / SIPcontrol 2.0.2) requires a patch for prober support of outgoing fax calls with V.34/T.38 mode. This patch was provided as part of the Dialogic support case
C81394.
• The DivaSIPcontrol service of DMG3008BRI (Diva System 8.5.6 / SIPcontrol 2.0.2) may crash with an application error in sipcontrol.exe 2.04.49 at address 0x00033d05 after an internal call via Elmeg ISDN
BAPX to a none connected ISDN line. In such a case the outgoing call is terminated with error code IF
(I:NOANS;40;12487;9-103) and the DivaSIPcontrol service will be restarted. If the problem happens at least 3 times within one day, the service remains stopped!!! This problem also happens with sending with Brooktrout SR-140 to the Dialogic Gateway. It has never been recognized with DMG4008BRI.
Patton Gateway
This section provides information about integration with Patton gateway.
Configuring the Gateway
1.
Start the Patton gateway configuration utility.
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Kofax Communication Server VoIP / FoIP Environment Guide
2.
To enable fax transmission with T.38, configure a Voice Profile. In the Telephony menu select VoIP
Profiles . Type the name of the profile and click the add icon.
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Kofax Communication Server VoIP / FoIP Environment Guide
3.
Go to the Fax tab and add “t38-udp” to the Fax Transmission Methods as shown in the following figure.
Configuring FoIPv3
Since some Patton gateways send a corrupted CNG signal it can be necessary to set the Outbound T.38
Mode to Repeat sending CNG tone as G.711 audio until Gateway switches toT.38 mode .
Hints for Patton Gateway
It happened that after a firmware update the DHCP server was activated. If you do not use the DHCP server make sure that after an update the server is disabled. Click DHCP Server in the main menu. Make sure that “active” is not selected.
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Kofax Communication Server VoIP / FoIP Environment Guide
21
Chapter 3
Integration with Software PABX
This section provides information about integration with software PABX.
Cisco CallManager
This chapter describes integration with CISCO Call Manager 4 (CCM) and later. Note that since release
6.0 the CCM is being referred to as Cisco Unified Communication Manager (CUCM).
In this guide we use:
• Both names as synonyms regardless of the CCM/CUCM release
• For the common CCM/CUCM features which have been available since CCM 4 the installation description is based on screenshots taken on the CCM 4, only new features available on later versions
(CUCM 6 or 7) are described based on their respective screenshots.
Kofax Communication Server integrates with the CCM/CUCM by the means of H.323 or SIP signaling protocol.
KCS FoIP T.38 and Voice Integration Overview
Since version 4.2(3), the CCM has supported the T.38 Fax over IP protocol (also in case of calls over gateways which are connected to the CCM via the Media Gateway Control Protocol – MGCP). This enables KCS FoIP to integrate with the CCM instead of talking directly to particular gateway(s).
(*) See Prerequisites for details
The dial-plan and configuration for the KCS FoIP is maintained on the CCM. The CCM is the central routing point for all VoIP and FoIP calls. Calls from or to the Cisco FoIP gateways are established via the
CCM.
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Kofax Communication Server VoIP / FoIP Environment Guide
Consider the following CCM environment example:
Prerequisites
• CCM version 4.2(3), CUCM version 6.1(2) or 7.1 (available from http://www.cisco.com
).
Note T.38 fax relay over MGCP gateways is only supported by CCM 4.2(3). Not even CCM 5.0 (which was released earlier) supports it. MGCP support was not tested with CCM 6.0.
• T.38 fax relay must be enabled on all used gateways (see Configuring the Gateway for information on how to enable T.38 fax relay on a Cisco FoIP H.323 gateway, and learn how to enable it on MGCP CAcontrolled gateways from the Cisco IOS Documentation ).
• KCS 9.0 or higher
• KCS “FAX over IP Channel (T.38)” license
Gateway Integration Protocols
In theory, the KCS FoIP integration protocol is independent of the one used to interconnect the CISCO
Gateway with the CCM. But in the real life there are some limitations which are described in the following table:
Recommended Cisco Gateway Integration Type (H.323, MGCP, or SIP) for particular CCM version/KCS
FoIP integration combination are stated in the following table.
CCM/CUCM version
< 4.2(3)
4.2(3)
6.1(2)
7.1
KCS FoIP Integration Type
H.323
SIP
H.323
N/A
H.323, MGCP
H.323,MGCP, SIP
H.323,MGCP
N/A
SIP,MGCP(1), H.323
SIP,MGCP(1), H.323
gateway issues.
Functional Blocks on the CCM/CUCM Side
Given below is an overview of the required functional blocks on the CCM/CUCM side.
• Either H.323 inter-cluster trunk(s) in the case of H.323 integration (CCM 4 or later) or SIP trunk(s) in the case of SIP integration (CCM 6 or later)
• Route pattern(s)
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Kofax Communication Server VoIP / FoIP Environment Guide
• Route group and route/hunt list in the case of integrating several instances of the KCS FoIP component
(fault-tolerant installations)
• Message waiting on/off control numbers (in the case of MWI functionality)
• VoiceMail profiles (in the case of voice integration)
CUCM 6.x and 7.1 Remarks
Since release 6.0, CUCM supports T.38 along with SIP protocol.
The configuration interface of CUCM 6.0 is similar to the interface of CCM 4 but some menu items have changed. For example:
• Button to add new trunks etc. is now called “Add New” and is on the left.
• The button “Insert” is named “Save”.
• The menu item “Route Plan” is named “Call Routing”.
• The menu item “Route/Hunt List” is separated in two items.
• The menu item “Voice Mail” is in the menu bar.
Integration Without Encryption
This section describes about integration without encryption.
CCM/CUCM Configuration
Simply put, the CCM/CUCM configuration comprises following steps:
1.
Configure appropriate trunk type (mandatory).
•
H.323 Inter-Cluster Trunk Without Gatekeeper
•
H.323 Inter-Cluster Trunk With Gatekeeper
•
2.
In case of fault-tolerant installation
configure Route group and Route/Hunt List (optional).
3.
4.
Configure Message waiting control numbers (optional).
5.
(optional).
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Kofax Communication Server VoIP / FoIP Environment Guide
H.323 Inter-Cluster Trunk Without Gatekeeper
1.
Open a web browser (such as Internet Explorer) and type the following URL: http://
<CALLMANAGER>/ccmadmin. Note: Replace <CALLMANAGER> with the computer name of CCM.
2.
From the Device menu, select “Trunk”. Click “Add New”.
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3.
As the trunk type, select “Inter-Cluster Trunk (Non-Gatekeeper Controlled)”. As the device protocol, select “Inter-Cluster Trunk”.
4.
Click “Next” and fill the required fields like “Device Name” and “Device Pool”.
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5.
Enable “Redirecting Number IE Delivery – Inbound” and “Redirecting Number IE Delivery –
Outbound”.
6.
In the field “Server IP Address/Host Name”, type the IP address of the FoIP component and click “Save” to save this configuration of the trunk.
Also, if a second KCS FoIP component is installed (fault-tolerant scenario), do not add the second field “Server IP Address/Host Name” of the trunk, but create another inter-cluster trunk for it (repeat steps 1-6 and choose another name for it, for example “KCS2”).
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H.323 Inter-Cluster Trunk With Gatekeeper
1.
Open a web browser (such as Internet Explorer) and type the following URL: http://
<CALLMANAGER>/ccmadmin. Note: Replace <CALLMANAGER> with the computer name of Cisco
CallManager.
2.
From the Device menu, select Gatekeeper. Click “Add New”.
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3.
Type the IP address in the field “Host Name/IP Address” and click “Save”.
4.
From the Device menu, select Trunk. Click “Add New”.
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5.
Select “H.225 Trunk (Gatekeeper Controlled)” in “Trunk Type” and “H.225” in “Device Protocol” and click “Next”.
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6.
Click “Next” and specify all required information.
7.
In the field “Gatekeeper Name”, pick one of the configured gatekeeper entries. Set the “Terminal
Type” to “Gateway”. In the (optional) field “Technology Prefix”, enter the prefix of the numbers that
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Kofax Communication Server VoIP / FoIP Environment Guide the gatekeeper should route towards CCM through this trunk (such as type “91” to route all numbers starting with “91” to this trunk).
Note The parameter “Technology Prefix” matches the “Reg. Numbers setting” in the Call-Peers configuration of the KCS FoIP component.
SIP Trunk (CUCM 6.0 or Later)
1.
The first step is to create a SIP Trunk Secure Profile. Go to System | Security Profile | SIP Trunk
Security Profile.
2.
Enter a name and a description and set Outgoing Transport Type to “UDP”. Save the profile.
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3.
The next step is to configure a SIP trunk. Go to Device > Trunk and click Add New :
4.
Select “SIP Trunk” as Trunk Type and click Next .
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5.
Type a name and a description of the trunk and set Device Pool to Default. Transmit UTF-8 for calling Party Name is enabled to support 8-bit Unicode Transformation Format.
6.
Select Redirecting Diversion Header Delivery for inbound and outbound.
7.
Type the IP address of the KCS FoIP as Destination Address and select the SIP Trunk Security
Profile you created. Save and reset the trunk. A Routing Pattern for the SIP Trunk can now be created.
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Route Group and Route/Hunt List (Optional)
Route group and Route/Hunt List are only necessary in the case of fault-tolerant installations (more than one KCS FoIP).
1.
From the Call Routing menu, select Route/Hunt | Route Group (for fault-tolerant configuration only, if more than one voice server is connected to the CCM). Click “Add New".
2.
Enter the Route Group Name. Select the trunk(s) created in steps 1-6 as Available Devices and click
“Add to Route Group”. In the “Distribution Algorithm” select “Circular” if you want CallManager to perform load-balancing among available voice servers (preferred setting). If you (for any reason) needed CallManager to route all calls to the first (primary) voice server and use the second one only as the fall-back, choose “Top Down” algorithm.
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Kofax Communication Server VoIP / FoIP Environment Guide
Click “Add to Route Group” to add the selected device to the group:
Note Only the trunks which are not yet assigned to any other route group or routing pattern/hunt pilot will appear as Available Devices.
If you want to assign any existing trunk (such as “KCS1”) to a new route group which was already entered into the routing pattern/hunt pilot before (see Route Pattern ) you have to delete the corresponding route pattern from routing pattern/hunt pilot table.
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3.
After you have added all desired trunks into the group, click Save (you will see all chosen trunks under “Selected devices”):
4.
From the CallRouting menu, select Route/Hunt | Route List. Click Add New.
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5.
Enter the name of the Route List into the required filed “Route/Hunt List Name” and click Save:
6.
Click Add Route Group:
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Kofax Communication Server VoIP / FoIP Environment Guide
7.
Select the corresponding Route Group and click “Save”:
8.
You will be informed that the route group has been inserted into the Route/Hunt List. Click OK.
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9.
Click Reset in order to reset the Route/Hunt List:
10.
Click “Reset” in the Device Reset screen:
11.
It is also recommended to reset all trunks that have been entered into the Route/Hunt list.
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From the Device menu, select Trunk. Click Find to display all configured trunks. Open the desired trunk’s configuration by clicking it, then click Reset Trunk:
12.
Continue with Route Pattern .
Route Pattern
1.
From the Call Routing menu, select Route/Hunt | Route Pattern and click “Add New”.
2.
Type the required number (to be routed to KCS Trunk or Route List) in the field “Route Pattern” (for example, in our case “5900”), the Route Option must be set to “Route this pattern”.
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In the field “Gateway or Route List” select the inter-cluster trunk (for a single KCS server) or the route list (for multiple KCS servers). Click “Save”.
Messages Wait Integration (Optional)
CCM 4.0 or later supports a simple proprietary MWI signaling method which works in the following way:
There are two MWI control numbers defined in the CCM – the first one to turn the MWI lamps on
(“MWIOn” number) and the second to turn the MWI lamps off (“MWIOff” number).
In order to control the MWI lamp on particular CCM directory number (or extension such as 1234) the call originating entity (Cisco IP phone, incoming call via H.323 or SIP trunk) has to issue a call with following parameters:
• Calling party number equals the DN number to be controlled (such as 1234)
• Called party number equals to the proper MWI control number (MWIOn or MWIOff numbers)
Open a web browser and go to CallManager administration.
1.
From the menu, select Advanced Feature | Voice Mail | Message Waiting, then click “Add New”:
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2.
Add a unique Message Waiting Number for Message Waiting Indicator On (such as 6632 as in our case) and click Save. Repeat the same for Message Waiting Indicator Off (the number for MWI Off must be different):
3.
Click “Back to Find and List Message Waiting Numbers” to display the list of numbers:
Voicemail Profile (Optional)
Voicemail profile can be optionally configured on the CCM in the case of KCS voice integration in order to simplify handling of call diversions/call forwards for DN numbers towards KCS Server.
1.
Open a web browser and go to CallManager administration.
2.
From the menu, select Advanced Features | Voice Mail | VoiceMail Pilot, then click “Add a New Voice mail Pilot”:
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3.
Enter the KCS Voice access number into the field Voice mail Pilot Number and appropriate description and click Save button (refer to the Voice Platform Technical Manual for definition of Voice
Access Number).
4.
From the menu, select Advanced Features | Voice Mail | VoiceMail Profile, then click “Add New:
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5.
Enter the name of the profile, choose the Voice Mail Pilot created before and click Insert:
Having created the Voice Mail Profile, it is easy to activate the desired call diversion(s) on particular
DN number to KCS Voice Server simply by choosing the created profile and selecting desired call forward variants for example:
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Note The mandatory prerequisite for using CCM Voice mail profiles in this way is the Redirecting
Number function activated on the H.323 or SIP trunk towards KCS FoIP.
KCS FoIP Configuration
This section describes the KCS FoIP configuration.
KCS FoIP Configuration for H.323 Integration
Open the KCS FoIP configuration tool and set any call peer to the following values:
• Protocol must be set to “H.323”
• Remote Address\Host must be set to the CCM IP.
• Remote Address\Port must be set only if the port used by CCM inter-cluster trunk is different to the standard H.323 port 1720.
• The Authorization is not required
• Reg. Numbers are only required only in the case of H.323 Inter-cluster Trunk with Gatekeeper integration
Refer to the Fax over IP Technical Manual for details.
An example configuration is shown below:
Since FoIPv3 3.02.00 it is possible to define how T.38 mode is activated. These settings can be found under the T.38 settings. Unless you experience any trouble it is recommended to use the default settings as shown in the screen shot below:
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Kofax Communication Server VoIP / FoIP Environment Guide
Refer to the Fax over IP Technical Manual for details.
FoIPv3 Configuration for SIP Integration
Open the KCS FoIP configuration tool and set any call peer to the following values:
• Protocol must be set to “SIP”
• Remote Address\Host must be set the CCM IP.
• Remote Address\Port must be set only if the port used by CCM SIP Trunk is different to the standard
SIP port 5060.
• The Authorization and Reg. Numbers are not required.
Refer to the Fax over IP Technical Manual for details.
An example configuration is shown below:
Refer to the Fax over IP Technical Manual for details.
Integration with Encryption
Encryption is supported via SIP only as described in this chapter. It assumes that you already have a working integration via SIP without encryption. Note that media security is not supported via T.38 fax. If you need media security with fax then G.711 pass through mode is required.
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Kofax Communication Server VoIP / FoIP Environment Guide
CCM/CUCM Configuration
1.
Open “Cisco Unified CM Administration” and create a “SIP Trunk Security Profile”.
2.
Click Add New .
3.
Enter a name and a description. Change Port from default 5060 to 5061. Enter an X.509 Subject
Name (such as default FoIP: kic-electronic-documents-test-cert.kofax.com).
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4.
Set Device Security Mode to Encrypted. Set Incoming Transport Type and, Outgoing Transport Type to TLS.
5.
Switch to “Cisco Unified OS Administration” and log in (typically with a different user as used for
“Cisco Unified CM Administration”).
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6.
On the Security menu, click Certificate Management .
7.
Click Find .
8.
Select the CallManager.PEM
.
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9.
Click Download .
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10.
Save it as a PEM file.
11.
Open the saved PEM file with a text editor.
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12.
Adapt following parameters in ALL relevant SIP trunk configurations: a.
Enable SRTP mode: b.
In SIP Trunk Security Profile field, select the newly created secure profile. Set the Destination
Port to 5061.
KCS FoIP Configuration
1.
In the Voice tab, set “Media Security” to 2 or 3, MediaSecurityCryptoSuites to 2 and
MediaSecurityUnencryptedSrtcp tp 1:
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2.
In the Fax tab, set the “OutboundT38Mode” and “InboundT38Mode” to 60 in order to prevent T.38.
3.
In the SIP Signaling tab, set the “SipEnabledTransports”, “SipOutgoingTransport” and “Check
Certificate” as shown in the example screen shot below:
4.
Go to the “SSL Certificate” tab. Copy the whole content of the saved Callmanager.PEM certificate file and paste it in the “SSL Trusted CA Certificates” field.
In case you want to use an own SSL key, copy the SSL public key content to the field “SSL
Certificate” and your SSL private key to the field “SSL Private Key”. The “SSL Private Key” is encrypted when the configuration is saved. You can also use the self- signed KCS FoIP certificate which is provides as default FoIP configuration. Note that the self-signed KCS FoIP certificate does not provide reliable server authentication.
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Upload the FoIP Certificate to Cisco CallManager
1.
Open the KCS FoIP configuration, copy the complete content of the “SSL Certificate” field and paste it to a text file. Save it as PEM file (such as FoIP.PEM).
In case you want to use your certificate, copy the SSL public key content to the “SSL Certificate” field and your SSL private key to the “SSL Private Key” field. The “SSL Private Key” is encrypted when the configuration is saved.
2.
Open “Cisco Unified OS Administration” and log in. On the Security menu, click Certificate
Management.
3.
Click Upload Certificate .
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4.
Browse to the location, where the FoIP.PEM is stored, select “CallManager-trust” as certificate name and click Upload File.
Cisco Callmanager checks during upload for duplicate certificates.
CISCO Gateway Configuration Example
In the case of FoIP integration with encryption, also the Cisco gateway should be configured correspondingly to support SRTP and secure signaling.
This section provides an example as how to do it in the case the gateway is interconnected with CUCM through SIP Trunk.
The SIP gateway has to perform the certificate exchange with the CUCM (like the FoIP).
1.
Upload the CUCM certificate to Cisco gateway
Download the CUCM certificate from the CUCM in the same way as described for FoIP in the PEM format, open the terminal connection with the gateway and upload the certificate to the gateway utilizing following steps (the name “CCM-Cert” can be freely chosen): crypto pki trustpoint CCM-Cert enrollment terminal revocation-check none
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!--- Download the Cisco CallManager certificate, and paste
!--- the contents of the certificate, pem format.
crypto ca authenticate CCM-Cert
Enter the base 64 encoded CA certificate.
End with a blank line or the word "quit" on a line by itself
-----BEGIN CERTIFICATE-----
MIICIjCCAYugAwIBAgIIS4xQN3bIZUowDQYJKoZIhvcNAQEFBQAwFzEVMBMGA1UE
……
/DV5TbDUdre6Org1mn4uaMdrYzt1kQ==
-----END CERTIFICATE-----
Certificate has the following attributes:
Fingerprint MD5: 1EF154E3 70E40379 1C7003B9 B29E111B
Fingerprint SHA1: CAFA0F83 B04B2E65 71104B73 64BF6AEB ABE9EED9
% Do you accept this certificate? [yes/no]: yes
Trustpoint CA certificate accepted.
% Certificate successfully imported
2.
Generate self-signed certificate of the gateway (for example, set its CN=dev2821) crypto pki trustpoint dev2821
enrollment selfsigned
fqdn none
subject-name CN=dev2821
revocation-check none
rsakeypair dev2821
crypto ca enroll dev2821
% The fully-qualified domain name will not be included in the certificate
% Include the router serial number in the subject name? [yes/no]: no
% Include an IP address in the subject name? [no]: no
Generate Self Signed Router Certificate? [yes/no]: yes
Router Self Signed Certificate successfully created
!— View the certificate in PEM format, and copy the Self-signed CA certificate
!--- (output starting from “----BEGIN” to “CERTIFICATE----“) to a file named
such as dev2821.pem
crypto pki export dev2821 pem terminal
% Self-signed CA certificate:
-----BEGIN CERTIFICATE-----
MIIBhDCCAS6gAwIBAgIBATANBgkqhkiG9w0BAQQFADARMQ8wDQYDVQQDEwZTSVAt
….
s980Np7dLJU=
-----END CERTIFICATE-----
% General Purpose Certificate:
-----BEGIN CERTIFICATE-----
MIIBhDCCAS6gAwIBAgIBATANBgkqhkiG9w0BAQQFADARMQ8wDQYDVQQDEwZTSVAt
…
IBVdtA4CM/74qCjhtsu/jciaIe90BXs56wrj7ZC4m1sIMzDAHfsl7dJlB2IOw9Sk
s980Np7dLJU=
-----END CERTIFICATE-----
3.
Import the dev2821.pem certificate to the CUCM in the same way as the FoIP certificate gets imported
4.
Configure the SIP stack to use the self-signed certificate of the router in order to establish a SIP TLS connection from/to CUCM (the ip address is that of the CUCM)
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Note that the CN name (“dev2821” in the example) must be the same as the X.509 subject name configured in the corresponding CUCM’s “SIP Trunk Security Provider” of the SIP trunk connected to the gateway: sip-ua
crypto signaling remote-addr
10.20.168.90 255.255.255.255
trustpoint dev2821 strict-cipher
5.
Enable SIP TLS and SRTP support at dial-peer level: dial-peer voice 6000 voip
description 6000-6999 – CUCM 9.0
destination-pattern 6...
session protocol sipv2
session target ipv4:10.20.168.90
session transport tcp tls
srtp
dtmf-relay rtp-nte
codec g711ulaw
no fax-relay sg3-to-g3
no vad
6.
Apply SIP Security provider on the SIP trunk configured for the gateway with these settings:
7.
Enable SIP Trunk for the gateway for SRTP
Assigning Outbound CUCM Calls with Particular Inbound VoIP Dial-Peer
Each call on a Cisco gateway has to match particular inbound and outbound dial-peer.
While matching of the outbound dial-peers automatically occurs based on the dialed called party and the destination-pattern attribute of the dial-peer, matching of the incoming calls (such as from FoIP through
CUCM) can occur by the means of several methods utilizing calling or called party number of a call.
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The most straightforward and logical way to match an incoming FoIP call with an incoming dial-peer would be to match its calling party number with the destination-pattern attribute of the dial-peer: but the problem is that by default, FoIP calls doesn’t set any calling party number in the outgoing calls.
FoIP calls without calling party number can’t be matched with any concrete dial-peer (except for a possibility to match the inbound dial-peer based on the called party number, which doesn’t seem to make a real sense ), and therefore the call uses the global parameters configured in the voip service. In this way, the Cisco gateway can be globally setup either for secure SIP with SRTP, or without security, for all possibly connected FoIP instances and CUCM servers.
The simple workaround is to insert a prefix for the calling party number in the CUCM SIP trunk for the gateway to match particular incoming dial-peer (per destination-pattern attribute) and then remove it through number translation profile on the gateway in the inbound dial-peer just matched for the call.
Example :
Assume dial-peer for the CUCM with number range 6XXX.
Define the calling party prefix 6000 in the route pattern configuration for calls routed to the gateway’s SIP trunk:
Define translation profile (named such as CUCM90) to remove the prefix 6000 from the number: voice translation-rule 6000 rule 1 /^6000/ // voice translation-profile CUCM90 translate calling 6000
Apply this translation profile to the incoming numbers in the related dial-peer: dial-peer voice 6000 voip description 6000-6999 - CUCM 9.0
translation-profile incoming CUCM90 destination-pattern 6...
no voice-class sip srtp negotiate session protocol sipv2 session target ipv4:10.20.168.90
session transport tcp tls dtmf-relay rtp-nte srtp codec g711ulaw no fax-relay sg3-to-g3 no vad
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In this way, all incoming FoIP calls without or even with some particular calling party number are assigned with the inbound dial_peer 6000 where the prefix 6000 is removed from the calling party number (prior to routing the call further).
Hints
This section provides additional information about integration with Cisco CallManager.
T.38 Mode Does Not Work with MGCP Gateways
It has been recognized that in the case of SIP integration, CUCM rejects the activation of T.38 mode (with
SIP error code 488) in the following case:
1.
An MCGP gateway is used
2.
The feature “mgcp package-capability fm-package” is not enabled in the MGCP gateway
3.
FoIPv3 has enabled reception with RTP named telephone events (RTP-NTE) according to RFC
2833. RTP-NTE is always enabled in FoIPv3 3.06.00. Since FoIPv3 3.08.09 it can be enabled/ disabled in the SIP part of the configuration as shown in the screen shot below:
Note RTP-NTE is required for voice operation with SIP on order to received DTMF digits. It not required for Fax. The problem is that if RTP-NTE is enabled, is has also been added to the SDP session of the
INVITE message and this may prevent the activation of T.38 mode using CCM and an MGCP gateway.
Solution :
Either enable “mgcp package-capability fm-package” in the MGCP gateway or disable reception of DTMF digits via RFC 2833 (RTP-NTE) in FoIPv3.
More details can be found in the CISCO Support Community article: MGCP Gateway Design with CVP
(link: https://supportforums.cisco.com/docs/DOC-1172 ) and in the SPR00053967
Redirecting Number (Call Diversion)
In the case of the voice integration, it is always recommended to activate the inbound function
“Redirecting Number IE Delivery” on each H.323 trunk towards KCS and “Redirecting Diversion Header
Delivery” on each SIP trunk.
Example :
The redirecting number is delivered in the H.323 setup message (4636 in this example). The CCM 6 inserts this number in the Q.931 part of this message.
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In addition, it also delivers the redirecting number encapsulated in an H.221 non-standard identifier in the
H.225 section.
In case of multiple forwarding only the first number (original called number) is delivered.
Cisco Gatekeeper
This section describes how to use a Cisco gatekeeper with a Cisco gateway and FoIP.
Integration of a Cisco Gatekeeper
This chapter describes how to use a Cisco gatekeeper with a Cisco gateway and FoIP. In this example the gateway and the gatekeeper are running on the same physical device.
Configuration of the Cisco Gateway
The following configuration steps must be taken to use a Cisco gatekeeper with a Cisco gateway: h323-gateway voip interface h323-gateway voip id gk-zone ipaddr 172.20.148.35 1718 h323-gateway voip h323-id gw_35 h323-gateway voip tech-prefix 1# h323-gateway voip bind srcaddr 172.20.148.35
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These lines must be added to the interface configuration, such as interface GigabitEthernet0/0. In the second line gk-zone is the name of gatekeeper zone and 172.20.148.35 is the IP address of the gatekeeper. Line 3 defines the H.323-ID of the gateway. Line 4 defines the technology prefix which the gateway should register. The last line binds the interface with address 172.20.148.35 to the gatekeeper.
For incoming calls you need to define a dial peer with a number range: dial-peer voice 98000 voip
destination-pattern 98..
session target ras
codec g711alaw
fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback Cisco
no vad
As session target select ras.
In the configuration mode you need to type: gateway
Configuration of the Cisco Gatekeeper
To enter the gatekeeper configuration type gatekeeper in the configuration mode. Then define a gatekeeper zone and a default technology prefix. Type “no shutdown” to activate the gatekeeper.
gatekeeper
zone local gk-zone kofax.com 172.20.148.35
gw-type-prefix 1#* default-technology
no shutdown
In this example gk-zone is the name of the gatekeeper zone, Kofax.com is the domain and 172.20.148.35
is the IP address of the interface of the gatekeeper.
Configuration of FoIP
Open the configuration web page. Configure a call peer for the gatekeeper:
Select “H.323 with RAS” as protocol. Enter User ID and Password if required. Under Reg. Number specify the number with should be registered when FoIP is a terminal or the prefix if FoIP is a gateway.
Under H.323 Settings you can configure if FoIP registers as terminal or gateway. If FoIP works as gateway you can give the name of the gatekeeper zone.
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Configuration of Multiple Zones
On a gatekeeper you can configure multiple zones: gatekeeper
zone local gk-zone kofax.com 172.20.148.35
zone local myzone kofax.com
gw-type-prefix 1#* default-technology
gw-type-prefix 982* hopoff myzone
gw-type-prefix 981* hopoff gk-zone
gw-type-prefix 0* hopoff gk-zone
no shutdown
So that the gatekeeper relays a call to a different zone you need to configure the prefixes of the numbers that should be forwarded.
GNUGk 2.0.8
Kofax Communication Server Voice over IP can be integrated with GNUGK 2.0.8 using the H.323
protocol.
VoIP Integration via H.323
This integration corresponds with the most general use case. It uses open-source H.323 components that are available for free and is dedicated mainly for test and educational purposes.
Configure the GNUGk Gatekeeper
1.
Download the Windows executables (available as Windows ZIP file) and manual (PDF file) from http://www.gnugk.org/h323download.html
(for version 2.0.8 or later).
2.
Unzip all files into a directory of your choice (such as GNUGk).
3.
Create the configuration file “gatekeeper.ini” in the subdirectory GNUGk\bin . Another subdirectory
GNUGk\etc contains a couple of sample config files. You can start with the simplest one named
“gnugk.ini”. Copy it into GNUGk\bin and rename to “gatekeeper.ini”.
This config file contains following lines:
##
## A very simple configuration file.
## Everyone is allowed to connect to the status port.
##
[Gatekeeper::Main]
Fourtytwo=42
TimeToLive=300
Name=MyGnuGk
[GkStatus::Auth] rule=allow
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4.
Start gnugk.exe in the GNUGk\bin directory. The following screen appears:
Now the gatekeeper is running.
Configure the KCS Voice Server
1.
Setup the KCS voice server for H.323 integration in the auto-discovery gatekeeper mode using registration type gateway.
Choose any endpoint name, such as “KCS”, and prefix you would like to use to route calls towards
KCS server, such as “9” (that is, all numbers starting with “9” are routed to KCS server).
It is possible to install the voice server on the same computer along with the GNUGk gatekeeper although it is not recommended (see
Check for Open H.323 Listeners on the Local Interfaces ).
Note If you have several gatekeepers running in your network, it would be better not to use the auto-discovery method, but locate the GNUGk per IP address.
2.
Open the Telnet session to the server where GNUGk is running using port 7000
(open the command line box, and type telnet server_name 7000).
3.
Start KCS voice server and wait until TCECP starts.
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Kofax Communication Server VoIP / FoIP Environment Guide
4.
Type ‘r’ and <Enter> in the telnet session. The following screen that informs you that the endpoint with H323_ID=KCS has been registered as gateway, appears.
Now KCS voice server has already been registered on the gatekeeper and it is prepared to receive
(and originate) H.323 calls.
Configure the OpenPhone Application
What we need at this point is an H.323 compatible terminal to be able to register on the GNUGk and make an incoming call.
Although we can use one of the H.323 telephones available in the market (like Tiptel Innovaphone IP200), it is also possible to use the open-source software telephone application called “OpenPhone”:
1.
Download and install the software is described in
OpenPhone (H.323 Telephone Software)
.
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2.
In the gatekeeper options check “Use Gatekeeper” and “Discover Automatically”:
3.
Restart OpenPhone. The following screen that informs you on the successful gatekeeper registration, appears.
Now if you dial any number starting with digit 9, the call will be routed towards KCS server.
Huawei SoftX3000-Softswitch
Fax Integration via SIP Trunk
We can integrate via SIP trunk like a SIP Gateway. A call-peer with type SIP and the IP of the Huawei
Softswitch is required. See example below:
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Notes:
• The Softwitch does not support T.30 error correction mode. There all fax call uses the fall-back to none-
ECM.
• The Softwitch may fails if a single T.38 packet contains more than 64 data Bytes. This issue has been considered since FoIP 3.10.00. With FoIPv3 3.09.03 - 3.09.10 the configuration value
T38\NoneEcmTxInterval must be set to 36ms. Versions before 3.09.03 are not recommended.
• The Softswitch does not correctly handle a collision of Change-to-T38-mode Re-Invite transaction.
Such a problem may happen during loop tests from KCS FoIP to KCS FoIP if both the outgoing and the incoming side are using the T.38 activation mode 10 (start with G.711audio mode. Switch immediately to T.38 mode). Do avoid this problems, it is recommended to change the T.38 mode for outgoing calls to
20 (Send one CED tone as G.711 audio and then switch to T.38 mode)
Mitel 3300 ICP Voice
Fax Integration via SIP Trunk
The integration with Mitel 3300 ICP voice requires FoIP 3.09.04 or higher (correction of SPR56467). It is important to disable the session timer for the used SIP trunk because Mitel uses SIP UPDATE method which is optional and currently not supported by FoIP. The most important configuration parameters are shown below:
FoIP Configuration :
Use Protocol “SIP” and configure the IP (and optional port) used by Mitel. An example is shown below:
Mitel Configuration :
A sample configuration of the SIP trunk for KCS FoIP is shown below. It is important to set the session timer to “0”.
Parameter
SIP Peer Profile Label
Network Element
Registration User Name
Address Type
Outbound Proxy Server
Default CPN
Restriction
Trunk Service
Value
Kofax
2a72f460-1dd2-11b2-bb36-08000f1e9808
IP Address: 10.10.10.10
FALSE
10
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Parameter
Interconnect Restriction
Maximum Simultaneous Calls
Session Timer
Zone
SMDR Tag
NAT Keep alive
Enable Mitel Proprietary SDP
Use P-Asserted Identity Header
Use Restricted Character Set For Authentication
Disable Reliable Provisional Responses
Use Alternate Destination Domain
FQDN or IP Address
Ignore Incoming Loose Routing Indication
Suppress Use of SDP Inactive Media Streams
Enable Special Re-invite Collision Handling
Enable sending '+' for E.164 numbers
Force sending SDP in initial Invite message
Use To Address in From Header on Outgoing Calls
Force Answer - send SDP in initial Invite
Prevent the Use of IP Address 0.0.0.0 in SDP Messages
Use P-Preferred Identity Header
Route Call Using To Header
Private SIP Trunk
Public Calling Party Number Pass-through
Use Diverting Party Number as Calling Party Number
Build Contact Using Request URI Address
Renegotiate SDP To Enforce Symmetric Codec
Repeat SDP Answer If Duplicate Offer Is Received
Allow Peer To Use Multiple Active M-Lines
User Name
Password
Confirm Password
Authentication Option for Incoming Calls No Authentication
No
No
No
No
Yes
No
No
Yes
No
No
No
No
No
No
Yes
No
No
No
No
Yes
No
2
0
Value
1
2
0
FALSE
No
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Siemens HiPath 4000 V4.0
KCS FoIP can be integrated with Siemens HiPath 4000 using either H.323 or SIP protocol. See Fax over
for the exact versions required.
The most important configuration setting on HiPath is setting the protocol to native SIP or native H.323.
On FoIP side, HiPath has to be connected like a standard gateway; choose the protocol and type the IP address. See the examples below.
Example Integration via H.323
This section provides an example of integration using H.3.2.3.
HiPath 4000 – Native H.323 10 B Channels
ADD-BUEND:314,"NATIVE H323 KOFAX ",120,0,*,2,ON,0,0,NEUTRAL;
ADD-BFDAT:33,HG3550,BCHL60&BCHL120;
CHANGE-BFDAT:CONT,33,HG3550,1,1,;
CHANGE-BFDAT:OK,33,YES;
ADD-BCSU:IPGW,1,1,14,"Q2316-X10 ",1,"0",33,,,,10,,,0;
ADD-CGWB:1,14,NORMAL,10.10.40.114,255.255.255.0,,,,,,,,,,,,;
CHANGE-CGWB:CGW,1,14,GLOBIF,,,213,NO,0,10.10.40.1,,"100MBFD",0,0,30,0,4060,0.0.0.0;
CHANGE-CGWB:CGW,1,14,SERVIF,"TRM",;
CHANGE-CGWB:CGW,1,14,ASC,29100,29339,"48","184",YES,YES,YES,YES,PRIO1,G711A,NO,"30";
CHANGE-CGWB:CGW,1,14,ASC,,,,,,,,,PRIO2,G711U,NO,"30";
CHANGE-CGWB:CGW,1,14,ASC,,,,,,,,,PRIO3,NONE,NO,"30";
CHANGE-CGWB:CGW,1,14,ASC,,,,,,,,,PRIO4,NONE,NO,"20";
CHANGE-CGWB:CGW,1,14,ASC,,,,,,,,,PRIO5,NONE,NO,"20";
CHANGE-CGWB:CGW,1,14,ASC,,,,,,,,,PRIO6,NONE,NO,"20";
CHANGE-CGWB:CGW,1,14,ASC,,,,,,,,,PRIO7,NONE,NO,"20";
CHANGE-CGWB:CGW,1,14,DSP,"60";
CHANGE-
CGWB:CGW,1,14,GKDATA,,1719,"PRIMARYRASMANAGERID",,,1719,"SECONDARYRASMANAGERID",,120;
CHANGE-CGWB:CGW,1,14,MGNTDATA,,8000,,443;
CHANGE-CGWB:CGW,1,14,DMCDATA,0;
CHANGE-CGWB:CGW,1,14,WBMDATA,"HP4K-DEVEL",,ENGR;
CHANGE-CGWB:CGW,1,14,WBMDATA,"HP4K-SU",,SU;
CHANGE-CGWB:CGW,1,14,WBMDATA,"HP4K-ADMIN",,ADMIN;
CHANGE-CGWB:CGW,1,14,WBMDATA,"HP4K-READER",,READONLY;
CHANGE-CGWB:CGW,1,14,GWDATA,"PRIMARYRASMANAGERID",;
CHANGE-CGWB:CGW,1,14,H235DATA,NO,NO,"siemensGateway2003",,100,242-191-30-119-188-
83-173-161-43-0-70-36-218-74-169-221-78-102-174-170;
CHANGE-CGWB:CGW,1,14,LEGKDATA,314,888314,NO;
CHANGE-CGWB:CGW,1,14,SIPTRERH,NO,,,;
CHANGE-CGWB:CGW,1,14,SIPTRSSA,NO,0.0.0.0,5060,5061,120,0.0.0.0,5060,5061;
CHANGE-CGWB:CGW,1,14,DLSDATA,,10444,NO;
CHANGE-CGWB:CGW,1,14,JB,40,120,20,4,60,200,2;
ADD-GKREG:314,INTGW&HG3550V2&H323,,,0,0,1,,;
ADD-GKREG:114,EXTGW&HG3550V2&H323,10.10.40.115,888114,0,0,1,"NATIVE H323",TRADITIO;
ADD-COT:114,PRI&RCL&XFER&ANS&KNOR&CEOC&CEBC&CBBN&CBFN&FWDN&FNAN&BSHT&BLOC&LWNC&ATR
S&ROPT&NLCR&TSCS&ICZL&TRSC&DFNN&CFTR&CTAL&NLRD&AOCC&CDRD&MCIW&BCNE&AMFC&NTON,,;
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ADD-COP:114,SDL&BR64&TIM1&IDP4,,,,;
CHANGE-COSSU:COS,100,TA&TNOTCR&TTT,,,,,,,;
CHANGE-COSSU:COS,100,,NOCO&NOTIE,,,,,,;
CHANGE-COSSU:COS,100,,,TA&TNOTCR&BASIC&MULTRA,,,,,;
ADD-TDCSU:NEW,1-01-014-0,114,114,0,0,100,1,1,"TRUNKING ",0,"ECMAV2",8,,NONE,,,,G
DTR,N,TIE,NONE,N,0,,,,,,,31,MANY,,0,1,1,EMPTY,20,1,N,,,,,,16,8,1,10,,EC&G711&G729
AOPT,,314,DSC,Y,TRADITIO,HG3550IP,1&&10,Y,1,2,0,0,0,0,0,N,;
Note The Class Of Trunk (line ADD-COT) must not include the feature CFVA (CALL FORWARDING
VALIDATION PROCEDURE POSSIBLE; German: PRZL = ERREICHBARKEITSPRUEFUNG DES
UMLEITUNGSZIELES MOEGLICH). If CFVA is enabled then each activation / deactivation triggers a destination number validation check (special control call) which is not supported by KCS FoIP. As a consequence, this unsupported function may cause the following problems:
1.
The call-forwarding activation may fail
2.
The HiPath 4000 v4 may block any calls the used H.323 trunk for about 1 minute
Number Assignment
The chapter provides some configuration hints how to assign a range of internal telephone numbers for native H.323.
1.
If you have an access code with a fixed number of digits you can use an LCR digit pattern as shown below:
H500: AMO LDPLN STARTED
+----------------------------------------------------------+
| CD ---> |
+-------+-------------------------------------+------------+
| LDPNO | LDP | DIPLNUM |
+-------+-------------------------------------+------------+
…
| 34 | 73-XXXX | 0 |
• This configuration expects 4 digits (represented by X’s) after 73 which means that the number range is between 730000 and 739999. “34” is an internal LCR dialing plan number used the native
H.323 trunk. The actual used number is installation specific.
2.
Sample using closed numbering where numbers from 1120 to 1169 are routed to FoIPv3
ADD-RICHT:CD,114,114,,,0,ALL,"N H323
",,,314,,,,114,,,,,,1-1-114,YES,,1-1-114,NEUTRAL,NO,NO,;
ADD-LODR:7,,,,NPI,UNKNOWN,UNKNOWN;
ADD-LODR:7,,,,ECHO,1;
ADD-LODR:7,,,,END;
ADD-LODR:7,,,"7-CLOSED NUMBERING",;
ADD-LDAT:114,ALL,1,,314,7,1,,1,EMPTY,NONE,,4,,,,,,,,114-0,,,,;
ADD-WABE:1120&&1169,,,STN,N,,,,,114;
• ADD-WABE defines the destination for the extensions
• ADD-RICHT makes the relation of this destination to the trunk that has to be used.
3.
Sample using open numbering where numbers starting with 8 are routed to FoIPv3
ADD-WABE:8,,,TIE,N;
ADD-RICHT:LRTENEW,8,ALL,"N H323
",314,1-1-114,YES,,FIX,,,PP300,NO,,,,NO,NO,,1-1-114,NEUTRAL,NO,NO,NO,NO,NO,NO,NO;
ADD-LODR:8,,,,NPI,UNKNOWN,UNKNOWN;
ADD-LODR:8,,,,ECHO,1;
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ADD-LODR:8,,,,ECHO,2;
ADD-LODR:8,,,,END;
ADD-LODR:8,,,"8-OPEN NUMBERING",;
ADD-LDAT:8,ALL,1,,314,8,1,,1,EMPTY,NONE,,4,,,,,,,,114-0,,,,;
ADD-LDPLN:LCRPATT,0,"8"-"Z",,8,,,1;
• The 8 in ADD-WABE is the access code or prefix
• The “8”-“Z” in ADD-LDPLN is the dialing rule with time out. Here parameter BLOC must be set in the used COT (Class of trunk) is important.
HiPath 4000 – Codec Parameters
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HiPath 4000 – DSP Settings
KCS FoIP Configuration
The connection to HiPath 4000 has to be configured like a standard H.323 Gateway. No user authorization or number registration is required. A typical screen shot is shown below:
Example Integration via SIP
This section provides an example of integration using SIP.
HiPath 4000 – Native SIP (30 Channels)
ADD-BUEND:313,"NATIVE SIP TRUNK ",120 ,0 ,* ,2 ,ON ,0 ,0 ,NEUTRAL;
ADD-BFDAT:3,HG3550,BCHL60&BCHL120;
CHANGE-BFDAT:CONT,3,HG3550,2,3,;
CHANGE-BFDAT:OK,3,YES;
ADD-BCSU:IPGW,1,1,13,"Q2316-X ",1,"0",3,,,,60,,,0;
ADD-CGWB:1,13,NORMAL,10.10.40.18,255.255.255.0,,,,,,,,,,,,;
CHANGE-CGWB:CGW,1,13,GLOBIF,,,213,NO,0,10.10.40.1,,"100MBFD",30,0,0,0,4060,0.0.0.0;
CHANGE-CGWB:CGW,1,13,SERVIF,"TRM",;
CHANGE-CGWB:CGW,1,13,ASC,29100,29219,"184","104",YES,YES,YES,YES,PRIO1,G711A,NO,"30";
CHANGE-CGWB:CGW,1,13,ASC,,,,,,,,,PRIO2,G711U,NO,"30";
CHANGE-CGWB:CGW,1,13,ASC,,,,,,,,,PRIO3,NONE,NO,"30";
CHANGE-CGWB:CGW,1,13,ASC,,,,,,,,,PRIO4,NONE,NO,"20";
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CHANGE-CGWB:CGW,1,13,ASC,,,,,,,,,PRIO5,NONE,NO,"20";
CHANGE-CGWB:CGW,1,13,ASC,,,,,,,,,PRIO6,NONE,NO,"20";
CHANGE-CGWB:CGW,1,13,ASC,,,,,,,,,PRIO7,NONE,NO,"20";
CHANGE-CGWB:CGW,1,13,DSP,"60";
CHANGE-
CGWB:CGW,1,13,GKDATA,,1719,"PRIMARYRASMANAGERID",,,1719,"SECONDARYRASMANAGERID",,120;
CHANGE-CGWB:CGW,1,13,DMCDATA,0;
CHANGE-CGWB:CGW,1,13,WBMDATA,"HP4K-DEVEL",,ENGR;
CHANGE-CGWB:CGW,1,13,WBMDATA,"HP4K-SU",,SU;
CHANGE-CGWB:CGW,1,13,WBMDATA,"HP4K-ADMIN",,ADMIN;
CHANGE-CGWB:CGW,1,13,WBMDATA,"HP4K-READER",,READONLY;
CHANGE-CGWB:CGW,1,13,GWDATA,"PRIMARYRASMANAGERID",;
CHANGE-CGWB:CGW,1,13,H235DATA,NO,NO,"siemensGateway2003",,100,242-191-30-119-188-
83-173-161-43-0-70-36-218-74-169-221-78-102-174-170;
CHANGE-CGWB:CGW,1,13,LEGKDATA,10,888010,NO;
CHANGE-CGWB:CGW,1,13,SIPTRERH,NO,,,;
CHANGE-CGWB:CGW,1,13,SIPTRSSA,NO,0.0.0.0,5060,5061,120,0.0.0.0,5060,5061;
CHANGE-CGWB:CGW,1,13,DLSDATA,,10444,NO;
CHANGE-CGWB:CGW,1,13,JB,40,120,20,4,60,200,2;
ADD-GKREG:10,INTGW®GW&HG3550V2&SIP,,,0,0,1,,;
ADD-GKREG:114,EXTGW&HG3550V2&SIP,10.10.40.152,888158,0,0,1,,TRADITIO;
ADD-COT:113,PRI&RCL&ANS&CEBC&FWDN&BSHT&BLOC&ATRS&TSCS&TRSC&CTLS&HGTR&NTON,,;
ADD-COP:113,,TA,TA,,;
CHANGE-COSSU:COS,100,TA&TNOTCR&TTT,,,,,,,;
CHANGE-COSSU:COS,100,,NOCO&NOTIE,,,,,,;
CHANGE-COSSU:COS,100,,,TA&TNOTCR&BASIC&MULTRA,,,,,;
ADD-TDCSU:NEW,1-01-013-0,113,113,0,0,100,7,7,"N SIP
",113,"ECMAV2",8,,NONE,
,,,GDTR,N,TIE,NONE,N,0,,,,,,,10,MANY,1-1-113,0,10,1,EMPTY,113,10,N,,,,,,16,8,1,10
,,EC&G711&G729AOPT,,313,DSC,Y,TRADITIO,HG3550IP,1&&30,N,1,2,0,0,0,0,0,N,;
For codec and DSP configuration, see
.
Number Assignment
The chapter shows two example configurations how to assign an internal telephone number
1.
Sample using closed numbering where number from 1120 to 1169 are routed to FoIPv3
ADD-RICHT:CD,113,113,,,0,ALL,"N SIP
",,,314,,,,113,,,,,,1-1-113,YES,,1-1-113,NEUTRAL,NO,NO,;
ADD-LODR:7,,,,NPI,UNKNOWN,UNKNOWN;
ADD-LODR:7,,,,ECHO,1;
ADD-LODR:7,,,,END;
ADD-LODR:7,,,"7-CLOSED NUMBERING",;
ADD-LDAT:114,ALL,1,,314,7,1,,1,EMPTY,NONE,,4,,,,,,,,114-0,,,,;
ADD-WABE:1120&&1169,,,STN,N,,,,,113;
• ADD-WABE defines the destination for the extensions
• ADD-RICHT makes the relation of this destination to the trunk that has to be used.
2.
Sample using open numbering where number starting with 8 are routed to FoIPv3
ADD-WABE:8,,,TIE,N;
ADD-RICHT:LRTENEW,8,ALL,"N SIP
",314,1-1-113,YES,,FIX,,,PP300,NO,,,,NO,NO,,1-1-113,NEUTRAL,NO,NO,NO,NO,NO,NO,NO;
ADD-LODR:8,,,,NPI,UNKNOWN,UNKNOWN;
ADD-LODR:8,,,,ECHO,1;
ADD-LODR:8,,,,ECHO,2;
ADD-LODR:8,,,,END;
ADD-LODR:8,,,"8-OPEN NUMBERING",;
ADD-LDAT:8,ALL,1,,314,8,1,,1,EMPTY,NONE,,4,,,,,,,,113-0,,,,;
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ADD-LDPLN:LCRPATT,0,"8"-"Z",,8,,,1;
• The 8 in ADD-WABE is the access code or prefix
• The “8”-“Z” in ADD-LDPLN is the dialling rule with time out. Here the COT parameter BLOC is important.
HiPath 4000 – Gateway Properties
KCS FoIP Configuration
The connection to HiPath 4000 has to be configured like a standard SIP Gateway. No user authorization or number registration is required. A typical screen shot is shown below:
Siemens HiPath 4000 V6.0
KCS FoIP can be integrated with Siemens HiPath 4000 V6.0 using either H.323 or SIP protocol, both interconnection types were certified in the Siemens certification lab. See
for the exact versions required.
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The detailed test report of the certification including the configuration description for both KCS FoIP and
Siemens Hipath4000 V6.0 as well is available for Siemens technicians in the Siemens intranet.
Siemens OpenScape Voice V3.1 R2
KCS FoIP can be integrated with Siemens OpenScape Voice V3.1 R2 (formerly known as HiPath 8000)
using SIP protocol. See Fax over IP Integration for the exact versions required.
OpenScape Voice can integrate with FoIP in two ways, as a subscriber (BGL) or as an endpoint profile
(EPP). Configuration as a subscriber is simpler. Configuration as an endpoint is recommended.
On FoIP side, OpenScape Voice has to be connected like a standard gateway; choose the “SIP with registration” protocol, type the IP address and type a number in the Reg. Numbers field. See the examples below.
Configuration as Subscriber
1.
In OpenScape Voice, create a subscriber as you would do for a SIP phone. Kofax FoIP is able to register dynamically on the subscriber number you just created.
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Kofax Communication Server VoIP / FoIP Environment Guide
2.
In the feature profile, enable all appropriate features such as Call Transfer; Music on Hold, and Name
Delivery options.
3.
In the KCS FoIP configuration tool, configure a call-peer to the following values: a.
Set the Protocol to “SIP with registration” b.
Set the Remote Address\Host to HiPath 8000 IP. (Default port is 5060) c.
Set the Reg. Numbers to the subscriber number assigned for KCS FoIP.
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An example configuration is shown below:
Configuration as Endpoint
1.
In the OpenScape Voice:
First create an endpoint profile (EPP). Select the Call Transfer feature as well as Name delivery, etc.
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2.
Create an Endpoint and assign the EPP to it. In the tab Attributes nothing needs to be selected.
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3.
Go to the SIP tab and choose your registration type. Kofax supports both dynamic and static registration.
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4.
It is also important to add one or more aliases, from which at least one is the IP address or DNS record the endpoint is operating under.
The endpoint has now been set up. The next step is creating an empty destination (under
Destinations and Routes). Simply call it “toKofax” and click Save. We will fill in the details later on.
Now we can start creating a prefix access code.
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5.
Create a prefix access code to route all numbers starting with “222” to the KCS FoIP.
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6.
Create a Destination Code “222” which points to the Kofax destination we created earlier.
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7.
Open the destination and add a new route that points to the correct endpoint. Click Save to remember the configuration.
8.
In the KCS FoIP configuration tool, configure a call-peer to the following values: a.
Set Protocol to “SIP with registration” b.
Set Remote Address\Host to HiPath 8000 IP. (Default port is 5060) c.
Set Reg. Numbers to any non-empty number.
An example configuration is shown below:
Inbound Failover
If you want to support inbound failover (inbound calls may be handled by multiple instances of FoIP), you have to bundle multiple IP-trunks. Please contact your Siemens specialist for more details.
Siemens OpenScape Voice V4
KCS FoIP can be integrated with Siemens OSV V4.0 using either H.323 or SIP protocol, both interconnection types were certified in the Siemens certification lab. See
for the exact versions required.
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The detailed test report of the certification including the configuration description for both KCS FoIP and
Siemens OSV V4.0 as well is available for Siemens technicians in the Siemens intranet.
Siemens OpenScape Voice V6
KCS FoIP can be integrated with Siemens OSV V6.0 using SIP protocol, both interconnection types were
The detailed test report of the certification including the configuration description for both KCS FoIP and
Siemens OSV V6.0 as well is available for Siemens technicians in the Siemens intranet.
Siemens OpenScape Voice V7
KCS FoIP can be integrated with Siemens OSV V7.0 using SIP or SIP/TLS/SRTP protocol, all three interconnection types were certified in the Siemens certification lab. See
for the exact versions required.
The detailed test report of the certification including the configuration description for both KCS FoIP and
Siemens OSV V7.0 as well is available for Siemens technicians in the Siemens intranet.
Siemens OpenScape Voice V8
KCS FoIP can be integrated with Siemens OSV V8.0 using SIP or SIP/TLS/SRTP protocols, all three interconnection types were certified in the Siemens certification lab. See
for the exact versions required.
The detailed test report of the certification including the configuration description for both KCS FoIP and
Siemens OSV V8.0 as well is available for Siemens technicians in the Siemens intranet.
The summary of all Siemens certifications can be found here: http://partnerdialog.unify.com/portal/tecpartner/node/5&char=K&partnerid=270#
Avaya Aura Communication Manager 5.2.1
KCS FoIP 3.12.07 or later can be integrated with Avaya Aura Communication Manager (CM) using either
SIP or H.323 protocol. See
for the exact versions required.
When using the H.323 integration, KCS FoIP communicates directly with Avaya CM via H.323 trunks.
On the other hand, in the case of the SIP integration another component referred to as SIP Enablement
Services (SES) is necessary on the Avaya side. Then KCS FoIP talks via SIP with the SES, and SES communicates via dedicated SIP trunk (using TCP/tls transport only) with the Avaya CM.
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This chapter describes an example of a simple Avaya CM/SES installation with the following dialplan/ routing, including one ISDN BRI trunk towards another test PBX (Siemens Hicom300), one H.323 and SIP trunk towards KCS FoIP solution:
• There are a few telephones (stations) in the range 58xxx (such as 58410, 58411, 58412)
• Numbers starting with leading 0 are dialed to the ISDN BRI line (trunk group 1)
• Numbers starting with 7xxx are routed to the trunk group 13 (H.323 trunk)
• Numbers starting with 8xxx are routed to the trunk group 29 (SIP trunk)
• The routing is performed by the Automatic Route Selection (ARS) system which is triggered by the feature access code (FAC) 9
With the Avaya CM systems, it is a common practice to perform routing by the means of ARS or AAR features, and the usage for one of them is usually triggered by dialing the configured FAC code, typically
9.
Note that ARS/AAR dialing without the FAC option is available in the “system-parameters customeroptions”, page 3, but by default it is switched off and it can only be enabled by Avaya:
AAR is often used for internal dialing, typically used for routing calls between private "TIE Trunks", while
ARS is used for routing calls to the PSTN. Most businesses use the “9” as the FAC to access the ARS dial plan.
Uniform dialing plan
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This is why you can assign one "Dialed Access Code (DAC)" for AAR and one for ARS. Most use 9 for
ARS.
This FAC code can be then inserted automatically in the trunk groups, but with our example we assume to always explicitly dial the 9 in order to route the call to any of the configured trunks, and without 9 to dial any of the local extensions. For example:
• dial 901234567 to go out to the trunk 1 => ISDN BRI line
• dial 989123 to go out to the trunk 29 => SIP trunk towards KCS FoIP
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Avaya Aura CM General Configuration
Avaya CM is being configured using so called System Access Terminal (SAT) which can be used with the telnet client connected to the CM server.
1.
Start the telnet session with the CM server, authenticate with the system user and start the sat command: telnet 172.20.148.88 <Enter>
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2.
Choose w2ktt terminal type.
There are four general commands that can be used for the following purposes:
• Add : To add new objects such as trunk group, signaling group and more.
• Change : To change parameters of an existing object.
• Display : To show parameters of an existing object.
• List : To show a list of objects of the same class, such as trunks.
Usually, the configuration of particular object consists of several pages. In order to scroll through them press ESC-n for the next page and ESC-p for the previous page.
In order to cancel a command, press ESC-x . To execute (in the case of making changes), press
ESC-e .
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3.
Execute command “ display dialplan analysis ”.
Note In this example digit “9” is being used as Feature Access Code (FAC) and number range
“58XXX” are extensions (telephones).
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4.
Execute command “ display feature-access-codes ”.
Note The digit “9” triggers the Automatic Route Selection (ARS) feature which in the fact performs to call routing in the CM (alternatively, also the similar feature Automatic Alternate Routing (AAR) could be used).
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5.
Execute command “ display ars analysis ”.
This means the following:
• All dialed numbers starting with 0, min. length 4 and max. length18, are routed further through route pattern 1.
• All dialed numbers starting with 7, min/max. length 5/18 are routed further through route pattern
13.
• All dialed numbers starting with 8, min/max. length 5/18 are routed further through route pattern
29.
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6.
Execute command “ display ars digit-conversion 0 ”.
This screen gives a possibility to perform some digit conversions after the dialed number has triggered the ARS:
In this example, if we dial 958411, the number 58xxx is recognized as extension/station number, and the 9 is automatically dropped from the number.
Note The Avaya phones often automatically insert the FAC codes to the dialed number and the users do not need to dial them explicitly. In that case, this entry would handle the local extension range 58xxx correspondingly.
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7.
Execute command “ display route-pattern 1 ”.
See that the route pattern points to the trunk group number 1 (Grp No) and prior to dialing the number further into the trunk group 1, (the first) digit is removed at first.
Note Check route patterns 13 and 29, respectively in the same way (the only difference with our configuration is that those route patterns do not remove any digits from the number as it is not necessary for SIP or H.323 trunks).
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8.
Execute command “ display trunk-group 1 ” and inspect pages 1, 2 and 5.
Note The trunk type ISDN doesn’t need any signaling group to be assigned to.
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9.
Select digit handling overlap/overlap (in countries like Austria and Germany) to be able to use overlap sending and receiving over the trunk. Select enbloc/enbloc in countries that use a strict numbering plan not allowing any overlap dialing procedures.
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10.
Assign some hardware ports to the trunk group.
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11.
Execute the “ display node-names ip ” in order to see assigned symbolic host names for the
KCS FoIP instances.
Note The node “procr” has been created by the Avaya CM automatically for the own processor node.
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12.
Execute command “ display ip-network-region 1 ”.
This network region is using codec set 1 (which will be shown in the next screen).
Note This example is using only one network region, but in a more complex Avaya system several network regions may be defined, using different codec sets, but this is outside of scope of this guide)
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13.
Execute command “ display ip-codec-set 1 ”.
Note On the first page, either both or at least one of the G.711 codec must be configured (at least the one which is also configured in the KCS FoIP voice settings). The silence suppression must be disabled when using G.711 pass-through FoIP.
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14.
Proceed to the next page (ESC-n), and see that the T.38 standard is configured.
Note
• Set the Fax mode to “Pass-through” instead of “T.38 standard” if you want to use G.711 passthrough FoIP.
• The network region and codec settings performed above are relevant for both H.323 and SIP trunks configured in next chapters.
H.323 Integration
As for the H.323 integration, at first the signaling group (number 13 in our example) must be created and it must be assigned to the H.323 trunk (number 13 in our example).
Execute the command “display signaling-group 13”:
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As the far-end node name, enter the host name of the KCS FoIP host (at01d031 in our example).
As the near-node name, you should choose the node name of the local Avaya CM node (which is typically
“procr” for a small system).
Execute command “display trunk-group 13” and check pages 1 and 2:
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Note The trunk group is surprisingly isdn, but the carrier medium is set to H.323.
The signaling group must be set to the one created before (which has also the number 13 for simplicity), and also number of members should be great than 0 (number of members denotes the maximum parallel calls through this trunk).
Press ESC-n to see the next page:
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Note The digit handling (overlap/overlap) set in this configuration page.
Execute the command “display route-pattern 13” to see the route pattern pointing to this trunk group (13):
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Note This route-pattern is assigned with desired number range to be routed to this trunk in the ARS (see above).
SIP Integration
Unlike to the H.323 integration where the H.323 trunk in the CM points directly to the KCS FoIP server, for
SIP an additional system component is needed – SIP Enablement Services (SES) – where the KCS FoIP is connected. And on the CM side, a SIP trunk along with assigned SIP signaling group must be created for the communication between CM and the SES.
The SIP configuration consists of two parts:
1.
CM
2.
SES
Note The following screen shots show a co-resident configuration where CM and SES run on the same processor (S8300 in this case).
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CM Configuration
1.
Execute command “ display signaling-group 29 ”.
Note Both near and far end node names in the fact point to the same IP address (see node-names configuration) as this is a co-resident system running CM and SES on the same S8300 processor).
2.
Set the Group Type / Transport Method to SIP/tls.
Note tls is the only transport supported for the SIP trunk interconnecting SES with the CM.
3.
The Near-end Listen Port is by default set to the value 6001 and it is the port where CM listens for the SIP calls arriving from the SES.
Note This port will also be configured in SES.
4.
The Fear-end Listen Port is by default set to the value 5061 and it is the port where SES listens for the SIP calls arriving from the CM.
Note This port will also be configured in SES.
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5.
Execute command “ display trunk-group 29 ”.
The signaling group must be set to the one created before (which has also the number 29 for simplicity), and also number of members should be great than 0 (number of members denotes the maximum parallel calls through this trunk).
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6.
Execute the command “ display route-pattern 29 ” to see the route pattern pointing to this trunk group (29).
SES Configuration
Given the SES has already been enabled (basic Avaya CM/SES configuration, CM added to the SES and others), the following configuration tasks are necessary.
1.
Login to your CM via web interface and go to Administration->SIP Enablement Services .
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2.
Go to Communication Manager Servers->List and you will see the list of entered CM servers.
3.
Click Map .
Note You will see the “Contact” (sip:$(user)172.20.148.88:6001;tls) pointing to the SIP listener port
6001 that was configured in the signaling group 29 on the CM, using the transport tls.
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4.
Click Add Another Map or edit one of the existing maps to define the route pattern to be routed to this CM.
5.
Choose a name of this map entry and define the number to be routed to this CM by the means of regular expression.
In this example, the number 58xxx would be routed to this CM instance.
6.
Configure the number map which would be routed to the KCS FoIP. Go to Hosts->List . You will again see the list or at least one CM server as Host there.
Note One would expect to define IP addresses of other SIP servers like KCS FoIP here, but it is not the case. They are entered via contact fields for each configured CM Host.
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7.
Click Map and then add the following: a.
Contact : To define the IP address of the KCS FoIP server (incl. port number 5060 and transport=udp) and click Submit to enter it to the SES.
b.
Map : To define the number to be routed to the KCS FoIP by the means of regular expression.
Click Update to enter it to the SES. (In this example, all numbers of the type 89xxx are routed to this KCS FoIP.
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At the end, it looks like this.
Useful Troubleshooting/Tracing Options
On the CM, useful traces may be activated on each trunk and /or station using the trunk number (the TAC) or the station’s extension. In order to see the TAC numbers, execute the command “list trunk-group”:
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And see the TAC numbers of our trunks.
In order to trace the trunk number 29, execute the command “list trace tac 1407” and make appropriate
SIP call:
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As for the SIP calls, there is another useful utility called “traceSES” which can be started in the telnet session with the CM server.
Open the telnet session with your Avaya server, log on there, execute the command “traceSES” and press
“s” to start the capture. Make a SIP call and observe the trace:
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Avaya Aura Communication Manager 6.01
KCS FoIP 3.12.07 or later can be integrated with Avaya Aura Communication Manager (CM) 6.01 using
either SIP or H.323 protocol. See Fax over IP Integration for the exact versions required.
This version of Avaya CM (6.01) has been certified in the Avaya’s certification lab and there are two application notes written by Avaya which describe both types of the integrations – the H.323 and SIP, as well.
H.323: http://downloads.avaya.com/css/P8/documents/100141486
SIP: http://downloads.avaya.com/css/P8/documents/100140202
Avaya Aura Communication Manager 6.2
KCS FoIP 3.16.20 or later can be integrated with Avaya Aura Communication Manager (CM) 6.2 using
either SIP or H.323 protocol. See Fax over IP Integration for the exact versions required.
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This version of Avaya CM (6.2) has been certified in the Avaya’s certification lab and there are two application notes written by Avaya which describe both types of the integrations – the H.323 and SIP, as well.
H.323: http://www.devconnectprogram.com/fileMedia/download/8779508b-fff5-4082-948d-ba5f27a4f284
SIP: http://www.devconnectprogram.com/fileMedia/download/c6c66c4c-66e1-4cee-bff3-70adcf8a9b7f
Avaya Aura Communication Manager 6.3
KCS FoIP 3.22.05 or later can be integrated with Avaya Aura Communication Manager (CM) 6.3 using
This version of Avaya CM (6.3) has been certified in the Avaya’s certification lab and there are three application notes written by Avaya which describe both types of the integrations – the H.323, SIP, and SIP/
TLS/SRTP as well.
The highlight with Avaya Aura version 6.3 is that it finally supports T.38 with ECM, the drawback is that it works only up to the baud rate 9600.
H.323 (T.38 ECM and Pass-through) https://downloads.avaya.com/css/P8/documents/100182742
SIP (T.38 ECM and Pass-through) https://downloads.avaya.com/css/P8/documents/100182743
SIP/TLS/SRTP (Pass-through) https://downloads.avaya.com/css/P8/documents/100182744
Avaya Aura Communication Manager 7.0
KCS FoIP 3.22.24 or later can be integrated with Avaya Aura Communication Manager (CM) 7.0 using
This version of Avaya CM (7.0) has been certified in the Avaya’s certification lab and there are three application notes written by Avaya which describe both types of the integrations – the H.323, SIP, and SIP/
TLS/SRTP as well.
H.323 (T.38 ECM and Pass-through) https://www.devconnectprogram.com/fileMedia/download/ed25feca-41d2-4f98-83e7-3e67ac78c604
SIP (T.38 ECM and Pass-through)
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SIP/TLS/SRTP (Pass-through) https://www.devconnectprogram.com/fileMedia/download/abd8a96f-febe-429a-b531-b406442cb975
Avaya Aura Communication Manager 7.0.1
KCS FoIP 3.26.11 or later can be integrated with Avaya Aura Communication Manager (CM) 7.0.1 using
This version of Avaya CM (7.0.1) has been certified in the Avaya’s certification lab and there are three application notes written by Avaya which describe both types of the integrations – the H.323, SIP, and SIP/
TLS/SRTP as well.
H.323 (T.38 ECM and Pass-through) https://www.devconnectprogram.com/fileMedia/download/4a42fa87-c714-497c-82d0-eac8216cefb0
SIP (T.38 ECM and Pass-through) https://www.devconnectprogram.com/fileMedia/download/6a71638a-9dff-484f-bb15-26296c956700
SIP/TLS/SRTP (Pass-through) https://www.devconnectprogram.com/fileMedia/download/a353a515-02da-4a57-93f8-5a9bc762d6d0
Microsoft Lync Server 2013
KCS FoIP 3.17.04 or later can be integrated with Microsoft Lync Server 2013. The integration into earlier versions of Microsoft Lync Server is not supported. Compatibility with other devices can be found on
Microsoft Lync Server Interop Program page: http://technet.microsoft.com/en-us/lync/gg131938.aspx
Integration as SIP Trunk Without Encryption
This chapter describes the basic integration of KCS FoIP as SIP trunk without encryption. It is strongly recommended to perform this step also in cases where encryption is required in order to simplify troubleshooting.
FoIP Configuration
The FoIP Configuration is very simple.
1.
Configure the IP and port of the mediation server in the used call-peer.
2.
Set to T.38 mode in the fax section to 40 (Use G.711 pass-through unless T.38 is requested by remote side). The EnableV.34 check-box will be ignored because is not supported with G.711 passthrough mode.
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3.
Set the Sip transport for outgoing calls to TCP and enable reception of DTMF digits via RFC2833.
Here is a screen shot from a sample configuration:
Lync Server 2013 Configuration
Note that the Lync Server supports by user/pool/side specific configuration parameters. In order to keep things as simple as possible only the default identity “Global” is use within this chapter. An example with
screen shots can be found in chapter Example Configuration with Lync Server 2013 .
A high-level summary of the KCS FoIP integration (which is like an integration of a SIP Gateway) is listed below:
1.
Run the Lync Server Topology Builder.
a.
Take care that the TCP listener is enabled in the Mediation Server(s).
b.
Add KCS FoIP as SIP Gateway (using TCP transport).
c.
Publish the modified Topology.
d.
Close the Topology Builder .
2.
Run the Lync Server Deployment Wizard.
a.
Run Install or Update Lync Server System .
b.
Execute Step 2: Setup or Remove Lync Server Components .
c.
Execute Step 4: Start Services .
d.
Close the Deployment Wizard.
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3.
Configure a voice route to KCS FoIP and set the correct trunk configuration: a.
Run the Lync Server Control Panel .
b.
Open Voice Routing > Voice Policy > Global .
c.
Select the Associated PSTN usage (such as “Internal”).
d.
Create a new Route to KCS FoIP (associate starting digits with the gateway IP of KCS FoIP).
e.
Open Voice Routing > Trunk Configuration > Global .
f.
Set Encryption support level to Optional .
g.
Commit all changes.
h.
Open the Lync Server Management Shell .
i.
Disable RTCP time-outs and enable session timers in the trunk configuration.
Set-CsTrunkConfiguration -Identity Global -RTCPActiveCalls
$false
Set-CsTrunkConfiguration -Identity Global -RTCPCallsOnHold $false
Set-CsTrunkConfiguration -Identity Global -EnableSessionTimer $true
4.
If you want to use Fax over IP, media bypass must be enabled in the network configuration: a.
Run the Lync Server Control Panel.
b.
Open Voice Routing > Trunk Configuration > Global .
c.
Set Enable media bypass .
d.
Save and commit changes.
e.
Open Network Configuration > Global .
f.
Select Enable media > always bypass .
5.
Optionally, you can enable enterprise Voice in any Lync Client in order to support calls between KCS
FoIP and the Lync Client as described below: a.
Run the Lync Server Control Panel.
b.
Search and open the Lync user you want to enable from Users > User Search .
c.
Set Telephony to Enterprise Voice and assign a telephone number in the field Line URI.
d.
Commit changes.
Example Configuration with Lync Server 2013
The chapter shows an example configuration with Lync Server 2013, KCS FoIP, and PSTN Gateway and a Lync user.
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Configuration Description
The KCS FoIP configuration is the same as already shown in FoIP Configuration
.
The Lync Server Topology Builder defines the SIP domain and the connections to the Gateways.
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The Lync Server Control Panel User settings for “AU2“
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All other values are set to <Automatic>
Lync Server Control Panel, Voice Routing, Dial Plan, “Global”
The associated Normalization Rules are used for the following number conversions:
Dialed number
74??x
0x
Converted number
+990x
0x
Description
Normalize inbound numbers from Gateway
Normalization rule for outbound calls (it does not change the number but it must exist to allow outbound calls)
Lync Server Control Panel, Voice Routing, Voice Policy, “Global”
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Lync Server Control Panel, Voice Routing, Voice Policy, Route
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Lync Server Control Panel, Voice Routing, Voice Policy, Trunk Configuration, “Global”
Lync Server Control Panel, Network Configuration, Global, “Global”
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Lync Server Control Panel, Trunk Configuration
Use Cases
1.
Internal call from KCS FoIP to Lync User “AU2” (such as, Voice Player) a.
KCS FoIP calls the number +9902 b.
Lync user “AU2” gets the call
2.
Internal call from Lync User “AU2” to KCS FoIP (such as, Voice-mailbox access) a.
The Lync user “AU2” calls the number +9901 b.
KCS FoIP (+9901) get the calls
3.
Inbound call from telephone/fax via PSTN Gateway to Lync user “Tester03” a.
Telephone calls 734-74002 b.
The Mediation server gets the incoming call with extension 74002 and normalizes the number to
+9902 c.
Lync user “AU2” has the telephone number +9902 and it gets the call.
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4.
Inbound fax from fax via PSTN Gateway to KCS FoIP a.
The fax calls 734-7401 b.
The Mediation server gets the incoming call with extension 7401 and normalizes the number to
+9901.
c.
KCS FoIP (+9901) gets the inbound call.
5.
Outbound call from Lync User via PSTN Gateway to external telephone or fax.
a.
The Lync user “AU2” calls the number 0142 b.
The call is routed via PSTN Gateway to the telephone/fax at 0142
6.
Outbound call from KCS FoIP via PSTN Gateway to external telephone or fax.
a.
KCS FoIP calls the number 0142 b.
The call is routed via PSTN Gateway to the telephone/fax at 0142
Enable Encryption of SIP Messages
The description in this chapter assumes that you have a working integration as SIP trunk without security
as described in Integration as SIP Trunk Without Encryption
above.
Create SSL Key for FoIP Using the Windows Domain Controller
This section describes how to get an SSL key as PEM format using the Windows Domain controller certification authority. You can also use any other Certification Authority to generate an SSL key as described in the KCS Web Services (TWS) for KCS manual.
Part 1: Create a Web Service Certification Template with exportable keys
This part has to be done once on the domain controller. If you already have an appropriate template, you can re-use it. This description assumes that the template is called “Web Server 2”.
1.
Log in to the domain controller (as domain admin) and run mmc.exe to open the Windows
Management Console. Select File -> Add/Remove Snap-in.
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2.
Ensure that the snap-in for Certificate Templates is enabled.
3.
Activate “Certificate Templates”, select the template “Web Server” and start the action “Duplicate
Template”
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4.
Select a display name for the template and enable publish certificate in Active Directory. Optionally change the validity period.
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5.
Allow export of private keys and then click OK to save the template.
6.
Close the management console. It is not required to save the console settings.
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7.
Open Administrative Tools > Certification Authority > Certification Templates . Select New >
Certificate Template to Issue
8.
Select “Web Server 2“:
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Part 2: Generate a new key as PEM Format
1.
Open the Active Directory Certificate Server with url htts://{Domain-Controller}/certsrv:
2.
Select “Request a certificate”
3.
Select “advanced certificate request”
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4.
Select “Create and submit a request to this CA” and confirm the following operation:
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5.
Fill out the certificate request as shown below. Take care that template “Web Server 2” is used and that the Name matches both with Host/IP of KCS FoIP and the configured gateway name in the Lync
Topology Manager. Finish input with the “Submit” button:
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6.
You get the issued certificate. Click “Install this certificate”
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7.
Open Internet Options > Content > Certificates . Select the installed certificate and then click
Export:
8.
On the 2nd page of the export wizard, select to include the private key.
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9.
On the 3rd page select PKCS#12 format as shown below:
10.
Protect the key with a password.
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11.
Save the certificate is .pfx file
12.
Confirm to finish the export:
13.
Convert the generated file foip_ip.pfx with the openssl command-line openssl pkcs12 -in foip_ip.pfx -out foip_ip.pem -nodes
Into a PEM file foip_ip.pem. You will be prompted for the password used in step 10 above.
The resulting PEM file is required in the FoIP Configuration as described in the next chapter.
Configure the SSL key in KCS FoIP
This description assumes that you have created an SSL key as PEM format as described in the previous chapter. The keys and certificates insides this PEM files must be saved in the KCS FoIP configuration section as shown in the screen shots below:
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Note The “SSL Trusted CA Certificates” are only required if the certificate check is enabled in the SIP section of the FoIP configuration.
Enable SIP via TSL in the Lync Mediation Server
1.
Open the Lync Server 2013 Topology builder, download the current topology and ensure that at least one TLS listener port is enabled in the used Mediation server.
2.
Publish the topology if you made some changes.
Change Trunk Configuration from SIP/TCP to SIP/TSL
Using the Windows Domain Controller through
Configure the SSL key in KCSFoIP
. It is recommended that you verify that your installation is still working with TCP before you do the changes described in this section.
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KCS FoIP Configuration Changes
1.
Change the remote port in the used call-peer to the TLS listener of the used mediation server (see
Enable SIP via TSL in the Lync Mediation Server )
2.
Enable SIP via TSL in the SIP signaling.
Note
• The enabled transports may include SIP/UDP and SIP/TCP in addition to SIP/TSL.
• The outgoing sip transport must be set to SIP/TSL.
Lync Server Topology Changes
1.
Open the Lync Server 2013 Topology builder und download the current topology.
2.
Open the properties of the used Trunk and change the SIP transport protocol to TLS. Set the
“Listening port for IP/PSTN gateway” to the “Local sips port” configured in KCS FoIP. Set the
“Associated Mediation Server” port to mediation server TLS listener that is also used as remote port in the FoIP call-peer.
3.
Publish the modified topology.
Enable Media Encryption (SRTP)
The media encryption mode can be configured both in KCS FoIP (configuration value Voice -> Media
Security) and the Lync Server ( Lync Server Control Panel > Voice Routing > Trunk Configuration >
Encryption support level ).
It is recommended to set the Lync Trunk encryption level to "Optional" and the FoIP Media Security to
"Negotiate" as you can see in the screen shots below. Such a configuration will automatically use media security if SIP via TSL is used.
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All supported values are shown in the table below:
SIP
Transport
Lync Server Trunk encryption level
TCP
TSL
TSL
TSL
Not Supp.
#
-
#
#
Optional
#
-
#
#
Required
-
-
-
#
Disabled
#
-
#
#
FoIP Media Security
Negotiate
-
#
#
#
Always
-
-
-
#
Note The Media Bypass is ignored if media encryption is supported by one side of the call only. In that case, the media data is always transferred via mediation server which uses RTAudio compression that prevents reliable fax transmission!
Media
Encryption
No
No
No
Yes
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Troubleshooting and Hints
Here are some Lync Server specific hints. See also
Verify SIP Connection Between FoIP and Mediation Server
After FoIP is configured as gateway, the Lync Mediation starts sending SIP OPTIONS requests approximately once in 80 seconds. You can see the OPTIONS requests as SIP messages in the network trace.
If do not have any network trace you can also check the FoIP trace for GetServerState messages (from sip to fx7)
28/12:25:11.510 (3c2c/390c/08d8) Dump-Req: Message 'GetServerState' (203 byte) from 1(sip) ---> 3(fx7)
28/12:26:31.494 (3c2c/38a0/08da) Dump-Req: Message 'GetServerState' (203 byte) from 1(sip) --->
3(fx7) 28/12:27:51.489 (3c2c/1500/08dc) Dump-Req: Message 'GetServerState' (203 byte) from 1(sip) ---
> 3(fx7)
Call Failure with SIP Status 488
Microsoft Lync Server expects that the Gateway has the DTMF capability enabled. Otherwise, calls fail with Sip Status “488 Invalid incoming Gateway SDP: Gateway ParseSdpOffer Error: No DTMF support on
Gateway side”.
If you see this problem, ensure that the configuration option EnableRtpNte is enabled in the SIP section of the FoIP configuration.
Call Failure with SIP Status 404
Sip status 404 indicates that the dialed number is invalid. Here are some troubleshooting hints:
• Open the Lync Server Control Panel > Voice Routing > Test Voice Routing .
• Create a Test with the used number and verify the expected number translation, PSTN usage and route.
• Check associated trunks in the used route.
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Using the Lync Server Logging Tool
In some cases it is helpful to get more details about a failed call from the Lync Server logs as described below.
1.
Install the Microsoft Lync Server 2013 Debugging Tools (can be downloaded from http:// www.microsoft.com/en-us/download/details.aspx?id=35453 ).
2.
Open a power-shell as an Admin.
3.
Change to the debugging tools installation folder and run OCSLogger.exe. After a few seconds you should see the “Lync Server 2013 Logging Tool” window.
4.
Enable logging in the Mediation Server and SIP Stack as shown the following screen shots:
5.
Click “Start Logging” to start generation of log data.
6.
Click “Stop Logging” to stop generation of log data.
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7.
Click “View Log Files” and then “View” in order open the log data in a text editor.
8.
In many case you will get a good hint for the failure when searching for lines with “ms-diagnostics”.
See example below:
Some Lync Services Fail to Start
For some unknown reasons it may happen that the Lync Services do not start anymore due to any issue with the Lync Server certificates. In that case you will see the following symptoms:
• Important services like “Lync Server Mediation” and “Lync Server Front-End” fail to start up.
• When you open the Lync Server Control Panel you will get a security warning as shown in the screen shots below.
After Refresh you get the following certificate warning:
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This problem can be fixed by rerunning step 3 of the Lync Server – Deployment Wizard.
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Sip Trunk via TSL Does Not Work
If your TSL connection is not working it is recommend to search for “TlsFailure” in the Lync Server trace.
An example trace output when using an SSL key that is not trusted by the Lync server is shown in the screen shot below:
Configure a valid SSL key as described in
Configure the SSL key in KCS FoIP
.
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Chapter 4
Integration with SIP Providers
This section describes integration with SIP providers.
Swisscom SIP Trunk
The integration with Swisscom SIP trunks will be supported with FoIP for KCS 10.0 or higher. It has been certified using the following system overview:
Used 3rd party software/hardware versions:
• Genband Session Border Controller 8.3.5.0
• Aastra 415, Aastra 430, Aastra 470 Release 3.0 Software-Version 8428b1
FoIP Configuration
1.
Configure the SIP trunk the Genband Session Border Controller:
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2.
Change the fax inbound/outbound mode to 20 (for T.38 mode) or 60 (for pass-through mode): or
3.
Change the SIP parameters “Add media for T.38” and “Retry RequestT38” as shown in the screen shot below:
Note Option “Add media for T.38” will be set to “Yes”, if a Swisscom SIP trunk installation with KCS
FoIP 10.0.1 is updated. It is recommended to change this value to “No” in order to fix bug 672110.
4.
Set the own telephone number (caller–id) in the Fax over IP channel configuration as described in the TCOSS system manual chaper “Cost Center Parameter, Caller ID”. An example with number
0123456789 in configuration lines 276 to 279 is shown below:
'8*~=0123456789%TI~ , 276
'80~=0123456789%0~ , 277
'8I~=0123456789%I~ , 278
'8~=0123456789%~ , 279
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148
Chapter 5
Recommended Tools and Hints
This section describes recommended tools and hints.
Tools
This section describes recommended tools.
MyPhone (H.323 Telephone Software)
MyPhone is a freeware application (available from http://myphone.sourceforge.net
). It can be used to test a H.323 connection. This section describes how to test a direct connection to a gateway using H.323 with
MyPhone.
Prerequisites
• Make sure that MyPhone 0.2b12 or higher is running on the Windows machine.
• LAN connection between the gateway and the Windows machine must be allowed with the following ports and protocols in both directions (check firewall configuration)
Port
1720
1024 – 65535
1024 – 65535
Type
TCP
Dynamic TCP
Dynamic UDP
Purpose
H.323 call setup
H.245 (call parameters)
RTP (audio and fax stream data)
RTCP (control information)
MyPhone Configuration
If MyPhone issues an error on startup like "Could not open H.323 listener interface <local-ip>", another process may be running which listens on one of the ports listed in
(probably port 1720). You
).
Another reason for the problem can be that the local IP address of the machine has to be entered in
MyPhone’s network settings. This looks as follows if you have 10.168.8.35 as IP address.
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Be careful to have all video options deactivated in MyPhone. If MyPhone signals any video capabilities in a H.323 session, a call over our Cisco gateway will not work.
You should also take care that only the audio codecs that you have activated on the gateway are enabled in MyPhone. In our case this is only G.711 Alaw.
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Test Outgoing Connections (IP to ISDN)
Enter the number to call and gateway’s IP address in MyPhone’s Call to (host/IP/alias) field in the format
<number-to-call>@<gateway-IP-address> and click Call .
The telephone with the number 172 must be ringing now. If you pick up, the call is established and
MyPhone should look like this.
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Test Incoming Connections (ISDN to IP)
Dial a number configured in the range of the dial-peer on a PBX telephone. MyPhone indicates an incoming call on the user interface by displaying 10.11.11.34 is calling in the status field. When you click
Answer , the call is established.
OpenPhone (H.323 Telephone Software)
It is not recommended to install OpenPhone on the KCS Voice Server for the following two reasons.
• There are "compatibility problems" between the OpenH323 libraries used by OpenPhone and the ones, which are installed into c:\topcall\shared by KCS Voice Server setup. If you want to run
OpenPhone on the same machine make sure that the appropriate versions of the DLLs (pwlib.dll, ptlib.dll and openh323.dll) are in the same directory as openphone.exe.
• The second reason is the restriction that "only one H.323 listener per machine" is allowed. If it is required to run OpenPhone on the same machine you have to set its listener port to any other free port than 1720. You can make it listen on port 1721 by entering “127.0.0.1:1721” in the field “Local interfaces” in OpenPhone’s networking options.
1.
Download executables for OpenPhone 1.7.0 or later from http://openh323.sourceforge.net/
(openphone.exe, openh323.dll, pwlib.dll and ptlib.dll); copy them into dedicated directory like
OpenPhone.
2.
Start openphone.exe.
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3.
In the General Options , type your alias number, such as 192.
4.
In the Gatekeeper Options , select Use Gatekeeper and Discover Automatically .
Note If you have several gatekeepers running in your network, it would be better not to use automatic discovery, but type the IP address of the GNUGk directly as Static Host!
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5.
In Audio Codec Options , verify that G.711 u-law and A-law codecs are enabled.
6.
In the Video Options , disable everything what you can.
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7.
Restart OpenPhone and you should see the following screen informing you on the successful gatekeeper registration.
Now if you dial any number starting with digit 9, the call will be routed towards KCS Voice server.
SIP SoftPhone
Sometimes it can be useful to test the SIP connectivity between KCS server and the SIP gateway with a software phone.
For example, it can be done with the free “Phoner” as follows:
1.
Download Phoner Setup application from http://phoner.de/download_en.htm
2.
Start the Phoner setup and install it with default settings.
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3.
Start Phoner application and go to Options | Communication.
4.
In the SIP tab, set the API checkbox to SIP and type the IP address of the gateway in the field Proxy/Registrar. Click the Codecs button.
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5.
Disable all codecs except G.711 A-law (or G.711 mu-law)
6.
Type the phone number to be dialed and click the telephone icon.
7.
The destination phone should be ringing now. Pick up the phone; you should have the voice channels connected through.
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8.
Go to Window | Debug to see the debug output for the SIP session
Wireshark
Wireshark (formerly known as Ethereal) is an IP sniffer with support of extensive VoIP and FoIP protocol analysis. IP sniffers are programs that can capture the IP traffic in a network and analyze the packets according to the used protocols. This helps to find problems and their reasons.
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Wireshark can be used free of charge and is available from http://www.wireshark.org
. We use version
0.99.2 in this manual.
1.
To start capturing, select Capture | Interfaces from the menu. Pick your local interface used for FoIP by clicking the Prepare button on the right.
2.
The following window appears. You should again check if the correct local interface is selected.
If there is a lot of IP traffic you should limit the amount of data by setting a filter, which captures only the traffic between the gateway and the Windows machine where the VoIP/FoIP application is running. You can use logical operators like and/or/not in the filters. A filter rule to capture the whole traffic between our windows machine and the gateway would look like this.
host 10.168.8.35 and host 10.11.11.34
If you also have a CallManager or an H.323 gatekeeper involved, your capture filter must also include those IP addresses.
Refer to Wireshark help for more information on the filter syntax.
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3.
In the same dialog it is possible to write the captured traffic immediately into one or more files with limited size and cyclic overwriting as you know it from the KCS traces. Here is an example with max.
100 files with max. size 5 MB.
If you do not enter anything here your capture is only in the memory and can be saved later by selecting File – Save from the menu. You should in any case use .pcap as extension for files created with Wireshark.
4.
Click Start to start capturing. While this is active you will see the following live statistics.
5.
Click Stop to end capturing and see the captured traffic.
Note For information how to analyze and troubleshoot FoIP problems with Wireshark, refer to the
T.38 Fax over IP Trace Analysis Guide that comes with TC/SP.
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Extract Voice data from a network trace
This section explains how the extract voice data from a network trace.
1.
Mark any UDP Packet of the voice stream that should be extracted and then select function
Statistics -> RTP -> Stream Analysis . You will get a new window as shown in the screen shot below:
2.
Click Save payload and save the forward channel as .au file (such as C:\temp\voice.au
).
The saved file can be opened with Audacity (see Extract Voice data from a network trace ).
Audacity
Audacity ® is free, open source software for recording and editing sounds. It can be downloaded from the following address: http://audacity.sourceforge.net/
Hints
This section describes hints.
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Check for Open H.323 Listeners on the Local Interfaces
The Windows command line tool netstat you can display all open IP listeners on the system. If no inbound call (from ISDN to MyPhone) is possible or if MyPhone starts with an error message that it failed to open the listener, you should check all listeners open. Probably the H.323 signaling protocol is already used by any other application.
Here is an example of how the output of netstat should look after you started MyPhone.
C:\Documents and Settings\CHKA>netstat -a -b -n
Active Connections
Proto Local Address Foreign Address State PID
TCP 0.0.0.0:135 0.0.0.0:0 LISTENING 1500
RpcSs
[svchost.exe]
TCP 0.0.0.0:445 0.0.0.0:0 LISTENING 4
[System]
TCP 0.0.0.0:1025 0.0.0.0:0 LISTENING 1120
[lsass.exe]
TCP 10.168.8.35:1720 0.0.0.0:0 LISTENING 5008
[MyPhone.exe]
...
If here is a different application that uses port 1720 (or any of the required) you have to stop it.
Check the LAN Connections
Ping the gateway from the Windows machine. The gateway must respond.
C:\>ping 10.11.11.34
Pinging 10.11.11.34 with 32 bytes of data:
Reply from 10.11.11.34: bytes=32 time<10ms TTL=255
Reply from 10.11.11.34: bytes=32 time<10ms TTL=255
Reply from 10.11.11.34: bytes=32 time<10ms TTL=255
Reply from 10.11.11.34: bytes=32 time<10ms TTL=255
Ping statistics for 10.11.11.34:
Packets: Sent = 4, Received = 4, Lost = 0 (0% loss),
Approximate round trip times in milli-seconds:
Minimum = 0ms, Maximum = 0ms, Average = 0ms
Ping the Windows machine from the gateways standard prompt. The Windows machine must respond.
Cisco2620>ping 10.168.8.35
Type escape sequence to abort.
Sending 5, 100-byte ICMP Echos to 10.168.8.35, timeout is 2 seconds:
!!!!!
Success rate is 100 percent (5/5), round-trip min/avg/max = 1/1/4 ms
Cisco2620>
It must be guaranteed that no firewall between the gateway and the Windows machine denies connections on any of the required ports and protocols.
Do not forget to make sure that no other listeners (except OpenPhone) are active on the required ports on the Windows machine. You can use netstat for this purpose.
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Set Caller ID for Outgoing Calls
On incoming calls the dial-peer is chosen by the dialed number (called party number). For outgoing calls the dial-peer is selected by the calling party number, which FoIPv3 sets in H.323 information elements.
If the calling party number matches to a dial-peers destination pattern, then the configuration of that dialpeer is applied for the outgoing call. If no dial-peer matches, the global gateway configuration is applied
(defined in voice service VoIP etc.). If the global configuration is different from the KCS configuration, problems can occur, so it is highly recommended to set the calling party number for outgoing calls.
The calling party number can be set fixed for the channel (with a number conversion rule) or user dependent (with a ++CID line in the users template or by the cost center parameter and an entry in Arr99).
Refer to the TCOSS System Manual for details on that.
This is an example for setting the calling party number to 8000 for all outgoing calls in the TCOSS channel configuration.
‘8I~=I8000%~ ,254
‘8~=8000%~ ,255
To check if you are sending the calling party number to the gateway correctly, enable ISDN debug on the gateway by entering “debug isdn q931” (Cisco-specific) in the enabled mode and make an outgoing call.
Some traces like this appear where the calling and the called party number among other ISDN information is visible.
Cisco2620#debug isdn q931 debug isdn q931 is ON.
Cisco2620#
Cisco2620#
19:05:14: ISDN BR1/0 Q931: TX -> SETUP pd = 8 callref = 0x14
Bearer Capability i = 0x9090A3
Standard = CCITT
Transer Capability = 3.1kHz Audio
Transfer Mode = Circuit
Transfer Rate = 64 kbit/s
Channel ID i = 0x83
Display i = '8000'
Calling Party Number i = 0x80, '8000'
Plan:Unknown, Type:Unknown
Called Party Number i = 0x81, '08172'
Plan:ISDN, Type:Unknown
Sending Complete
Channel ID i = 0x89
Progress Ind i = 0x8188 - In-band info or appropriate now available
Date/Time i = 0x030B040B0A
Connected Number i = 0x018038313732
19:05:15: ISDN BR1/0 Q931: TX -> CONNECT_ACK pd = 8 callref = 0x14
To check if the right dial-peer is associated with your outgoing call, you can use “debug voip dialpeer all” (Cisco specific) and then make an outgoing call.
SIP Protocol Basics and Examples
SIP is an application-layer protocol that can establish, modify and terminate multimedia session such as telephony or fax calls. SIP messages are text encoded and use the UTF-8 character set. Much of the
SIP’s message and header syntax is identical to HTTP/1.1 protocol.
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SIP is being developed by the SIP Working Group within the Internet Engineering Task Force (IETF); the protocol is published as RFC 3261.
From the functional / call control point of view, SIP protocol could be compared with H.323 protocol stack, but it is simpler and uses fewer messages – requests and corresponding responses.
SIP protocol is transaction oriented, as in the fact each SIP request starts a separate transaction.
Most important SIP requests/transactions are:
• INVITE : initiates a call (equivalent to H.323/Setup) and changes call parameters (“RE-INVITE”)
(in the case of T.38 fax call, RE-INVITE is being used to switch from the voice mode into the T.38 mode, equivalent with H.323/RequestChangeMode)
• ACK : confirmation for final response to INVITE
• BYE : Disconnects the call
SIP responses are partly based on HTTP protocol responses; there are two main types of them:
• Provisional (1XX class) : Used by the servers to indicate call progress (like CallProceeding, Alerting with H.323) but do not terminate SIP transactions
For example, 100 stands for Trying (equivalent to H.323/CallProc) 180 stands for Ringing (equivalent to
H.323/Alerting)
• Final (2XX, 3XX. 4XX, 5XX, 6XX classes) : Used by the servers to terminate SIP transactions.
For example, 200 stands for OK (equivalent with H.323/Connect in the case of the INVITE transaction and with H.323 ReleaseComplete in the case of BYE transaction)
Example of an outgoing T.38 fax call from KCS towards gateway
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(10.18.136.68 is the KCS server, 10.11.11.35 the gateway):
1.
KCS starts the call establishment by sending the INVITE, and the gateway responds with 100 Trying
(“CallProc”), 180 Ringing (“Alerting”) and finally 200 OK (“Connect”).
This final response from gateway is acknowledged by sending ACK request and voice media channels are established.
Then KCS enforces the T.38 mode by sending the “(RE-)INVITE” (marked in blue below) and the gateway responds with 100 Trying and 200 OK. The T.38 communication can start now:
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2.
Once the T.38 communication has finished, KCS disconnects the call by sending BYE and the gateway responds by the 200 OK:
For further information on SIP, refer to the SIP documentation available on the Internet. For example:
• http://www.ietf.org/rfc/rfc3261.txt
(RFC that defines the SIP protocol).
Refer to the Overview of Operation ) for a good introduction.
• http://www.sipforum.org
In the white paper section, there are some good papers providing introduction on SIP protocol.
Concurrent Operation with KCS H.323 Integration for Voice on the Same
Machine
FoIPv3 can be installed on the same machine as the H.323 Integration for Voice (tce_h323) . This requires at least H.323 Engine for Voice (tce_h323) 1.02.06 included in TC/SP 7.59.06 and higher.
The necessary steps to install and configure both systems on one machine are described in the H.323
Voice Integration documentation under Concurrent Operation with T.38 Fax over IP (TC/FoIP) .
Check the ISDN Line Synchronization Up to the CISCO Gateway (BERT)
It is often necessary to verify the quality of the local PBX or PTT interconnection of the T.38 gateways due to different stability issues (like alarm conditions, slip errors, or too much of different analog fax errors like
XT, XV, XT etc. that may indicate poor line quality).
The easiest method how to verify the quality of the line (even remotely) is to perform so called Bit Error
Rate Testing (BERT). All what we need is an ISDN test equipment capable of doing BERT testing.
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The basic principle is simple: the ISDN tester (Test generator) sends a pseudo-random digital test pattern towards the Device under Test (DUT) that loops the received data transparently back (via the same Bchannel). The tester compares the sent and received data for a defined time and makes the test verdict in terms of error ratio (number of bit errors per 1000 bits, for example 1.10-6). This method is often being used with the Kofax lineservers and requires only one B channel between the ISDN tested and the DUT.
In order to test the line quality towards the T.38 gateway, it is possible to use the same ISDN tester as both Test generator and DUT as well like following:
1.
Connect the ISDN Tester to the available BRI/PRI line
2.
Make a call to the T.38 gateway through its ISDN line and use the special extension
3.
Configure the gateway so that upon receiving this special extension, the gateway establishes the 2nd call with the ISDN tester and answer the call on the tester
4.
Setup one of the calls into the LOOP mode on the tester and start the BERT test on another one
5.
In this way, the test patterns would proceed from the tester to the gateway, would be looped to the
2nd call towards the tester, there they would be looped back to the gateway and finally would be sent back to the tester like this:
Configuration Example for BERT on the Cisco 28xx Gateway
Assume there is a CISCO gateway 2821 connected to one E1 line and the ISDN Aurora tester connected to the public ISDN line with the number 8658927. The Cisco gateways is connected to the PBX reachable through the public number 8658929-79xxx, the escape digit for the public line in the PBX is 0. We define a special extension 098 so that if we dial the number 8658929-79098 from the tester the gateway would establish the 2nd call to the number 8658927 and connect both calls together.
The leading 0 in the special extension would force the gateway to route the 2nd call through the POTS dial-peer with the destination pattern 0T, but then, prior to dialing out, the gateway would perform the number translation due to translate-outgoing called command so that the number 08658927 would be dialed instead of 098:
Excerpt from the Cisco gateway configuration:
!
translation-rule 98
Rule 0 ^098 08658927
!
!
dial-peer voice 1 pots destination-pattern 0T progress_ind alert enable 8 progress_ind progress enable 8 progress_ind connect enable 8 translate-outgoing called 98 direct-inward-dial
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!
Bad Fax Quality Due to RTP-NTE
We have recognized a fax quality issue with CISCO UCM and CISCO Gateways in the following case:
1.
FoIP is connected as SIP trunk
2.
The SIP option EnableRtpNte (Support reception of DTMF digits via RFC 2833) is enabled.
3.
The initial connection will be created between FoIP and any CISCO softphone. The call is then transferred to a gateway.
In that case it may happen that the media data of the transferred call is routed via CISCO UCM which may cause a very bad fax quality.
If you have a similar use case, it is recommended to check the IP address of the actual used media connection in the Peer parameter of the FoIP trace as shown in the example below:
Inbound Call: CallerId=@4630, TSI= EC= (), Peer=10.20.30.40:19140
Outbound Call: Number=I4602 -> 4602, CallerId=, CSI=, EC= (), Peer=10.20.30.40:18414
Outgoing Secure SIP Call Fails with Error Code 12700
If an outgoing SIP call via TLS fails with error code 12700, there may be a TLS authentication issue.
Check the FoIP traces for “Could not connect TLS connection”. The example below shows an error case where the Certificate Check was enabled in the SIP configuration parameters but they remote server does not use a key that was issued by an computer using a certification configured in “SSL Trusted CA
Certificates”.
13/15:34:54.924 (2a08/2fd8/0007) {"SipTcpConnList" 0x1cdf47c} Could not connect TLS
connection, rhost=::ffff:172.20.150.4, rport=5067: Error 586157578: SSL verify error
20 unable to get local issuer certificate
Change to FoIP Configuration to either use a correct “SSL Trusted CA Certificate” or disable the certificate check.
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