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custom field in sip message

Evgeny Andreev
Beginner
Beginner

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

4 Replies 4

Nipun Singh Raghav
Cisco Employee
Cisco Employee
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.

Nipun Singh Raghav
"We cannot solve our problems with the same thinking we used when we created them"

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?

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