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SIP URI Command

Chuan Liu
Level 1
Level 1

Hi,

I have a couple of CME's connected via SIP trunks.  In CME_A, I need to match incoming calls against SIP uri 'host' instead of 'user-id' from CME_B. (Matching user-id works fine).  Under 'voice class uri sip' command in CME-A, what value does the host command take?

voice class uri From-CME-B sip

host ??????

I have tried the ip address of CME_B 200.162.180.9 with no success. The following debug shows the SIP invite information in CME_B:

202257: Mar 21 13:49:56.217: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:64800000000@202.162.176.51:5060 SIP/2.0
Via: SIP/2.0/UDP 202.162.180.9:5060;branch=z9hG4bK232CFDE
From: "6499705558" <sip:6499705558@202.162.180.9>;tag=200C5380-1D51
To: <sip:64800000000@202.162.176.51>
Date: Mon, 21 Mar 2011 00:49:56 GMT
Call-ID: F991D55B-528B11E0-863CF659-D3CCAD72@202.162.180.9
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE:  1800
Cisco-Guid: 4166684058-1384845792-2251814489-3553406322
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1300668596
Contact: <sip:6499705558@202.162.180.9:5060>
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 388

v=0
o=CiscoSystemsSIP-GW-UserAgent 7782 9672 IN IP4 202.162.180.9
s=SIP Call
c=IN IP4 202.162.180.9
t=0 0
m=audio 19382 RTP/AVP 8 0 18 125 100 101
c=IN IP4 202.162.180.9
a=rtp
tt-triton-cme#map:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:125 X-CCD/8000
a=rtpmap:100 X-NSE/8000
a=fmtp:100 192-194
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

Thanks.

1 Reply 1

Michael Solomonides
Cisco Employee
Cisco Employee

Hi Chuan,

This can take 1 regular expression and as many ipv4: ipv6: and dns: entries as required

voice class uri test sip

host 10.86.176.13[5-9]

host ipv4:1.23.4.5

host dns:google.com

I would wonder if you have an issue with the dial-peer rather than the voice class uri.