custom field in sip message

[Evgeny Andreev]

Hello

sip app - SIP - cisco router voice bundle - SIP - genesys sip server

if i initiate call from sip app to genesys server with custom field in INVITE message (ClientCode: 123456), i couldn't see it on genesys server. On cisco router i see this field only in INVITE from sip app.

Voice call works fine

sip app - SIP - genesys sip server

in thist topology custom field is recieved on genesys server

dial-peers on router

dial-peer voice 121 voip
description **** MOBILE APP ***
destination-pattern 7800
session protocol sipv2
dtmf-relay cisco-rtp h245-alphanumeric h245-signal rtp-nte
codec g711alaw
no vad

dial-peer voice 100 voip
description ***GENESYS SIP SERVER***
huntstop
preference 3
destination-pattern 9[0,9].
session protocol sipv2
session target ipv4:172.28.240.56
dtmf-relay rtp-nte
codec g711alaw
no vad

Any ideas why cisco router cuts custom field??

Best regards

[R0g22]

Can you show the complete INVITE please ? You should be able to match the header against a sip profile and use a copy list to add it back when the INVITE is sent out. CUBE does a check and drops unsupported information so what you see if working as designed AFAIK.

[Evgeny Andreev]

Jun 22 13:23:24: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:9014@media1.as50464.net<mailto:sip%3A9014@media1.as50464.net> SIP/2.0
Record-Route: <sip:mo@172.26.0.16<mailto:sip%3Amo@172.26.0.16>;r2=on;lr=on;ftag=3RJdXhmP;vst=AAAAAE1VUlwfHhAUGQ5DWiEeUFRdVwUADxZBMDQ2NC5uZXQ-;nat=yes;rtp=SAVPF;did=337.89b2>
Record-Route: <sip:mo@172.26.0.16:443<http://sip:mo@172.26.0.16:443/>;transport=ws;r2=on;lr=on;ftag=3RJdXhmP;vst=AAAAAE1VUlwfHhAUGQ5DWiEeUFRdVwUADxZBMDQ2NC5uZXQ-;nat=yes;rtp=SAVPF;did=337.89b2>
Via: SIP/2.0/UDP 172.26.0.16;branch=z9hG4bK1bbd.fd560798bff7a5ccedfbaa4d1e090140.0
Via: SIP/2.0/UDP 195.191.76.184:5060;received=195.191.76.184;branch=UjhUioE-F0xb-03oAoHu297bzERk7Q2VdOqdEFrZ5U9vt;rport=16157
Contact: <sip:mkb_mobile2@195.191.76.<mailto:sip%3Amkb_mobile2@195.191.76.>184:16157;transport=udp;alias=195.191.76.184~16157~6>
From: <sip:mkb_mobile2@media1.as50464.net<mailto:sip%3Amkb_mobile2@media1.as50464.net>>;tag=3RJdXhmP
To: <sip:9014@media1.as50464.net<mailto:sip%3A9014@media1.as50464.net>>
CSeq: 20 INVITE
Call-ID: AwfUnFYIoy9p
Max-Forwards: 69
Supported: replaces, outbound
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE
Content-Type: application/sdp
X-Key: 4b2e96e85de7767202961fe05cd85f59d431ffa75e01956e0b3b6be009ACK sip:9014@172.28.241.251:5060<http://sip:9014@172.28.241.251:5060/> SIP/2.0
Call-ID: AwfUnFYIoy9p
CSeq: 20 ACK
Via: SIP/2.0/UDP 172.26.0.16;branch=z9hG4bK1bbd.c94f3aa074b6547bb498d124d768d9fb.0
Via: SIP/2.0/UDP 195.191.76.184:16157;received=195.191.76.184;rport=16157;branch=Mv5DjxB-4K2Q-YbHrBhLu6jaLs224XbEK9d7PyBxK27xt
From: <sip:mkb_mobile2@media1.as50464.net<mailto:sip%3Amkb_mobile2@media1.as50464.net>>;tag=3RJdXhmP
To: <sip:9014@media1.as50464.net<mailto:sip%3A9014@media1.as50464.net>>;tag=A706411C-1DDF
X-Key: 82116890e9969341396e9e77072a14f14ed89549e0d76f5bd42ea1f3a136097a9b95da85e11837bf9085ed62c4c7cac4c7d6e7ff41eb8a991a0df1500b9da028
Max-Forwards: 69
User-Agent: PPP PP>P1P0P9P;
DeviceUID: 5F0D6C2D-CB73-43EA-80E2-35D4158372D0
ClientCode: 886841

so i need to use sip-copy list?

[Mohammed al Baqari]

Configure voice class

voice class sip-hdr-passthrulist 10
passthru-hdr ClientCode

Under the inbound dial-peer put this command

voice-class sip pass-thru headers 10

[Anthony Holloway]

Just wondering, why do you have session protocol sipv2 on dial-peer 121, obviously trying to make this a SIP dial-peer, but then have H323 DTMF relay configured?