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cdnadmin
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This document was generated from CDN thread

Created by: christophe terpreau on 26-10-2011 09:56:07 AM
Hello,
i'm using atm a sip papt2 box to call using my sip provider. I want to use my CUCM 8.5 to place call through a sip trunk.
 
here is the log from the ATA box:
 

Via: SIP/2.0/UDP 192.168.10.230:5060;branch=z9hG4bK-4882e446
From: "0033183644040" <sip:[email protected]>;tag=c5b255163ae082co0
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 101 INVITE
Max-Forwards: 70
Contact: "0033183644040" <sip:[email protected]:5060>
Expires: 240
User-Agent: PnP PAP2T
Content-Length: 277
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces
Content-Type: application/sdp

 
and the one from cucm logs:
 

INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.200:5060;branch=z9hG4bK1445520ef7
From: <sip:[email protected]>;tag=22~e8453ebe-922b-4be2-8809-d5a36dd7f976-27199614
To: <sip:[email protected]>
Date: Wed, 26 Oct 2011 13:25:19 GMT
Call-ID: [email protected]
Supported: timer,resource-priority,replaces
Min-SE:  1800
User-Agent: Cisco-CUCM8.5
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Expires: 180
Allow-Events: presence, kpml
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Call-Info: <sip:192.168.10.200:5060>;method="NOTIFY;Event=telephone-event;Duration=500"
Cisco-Guid: 4055322368-0000065536-0000000021-3356141760
Session-Expires:  1800
P-Asserted-Identity: <sip:[email protected]>
Remote-Party-ID: <sip:[email protected]>;party=calling;screen=yes;privacy=off
Contact: <sip:[email protected]:5060>
Max-Forwards: 70
Content-Length: 0

 
Do you think it is possible to use a script to replace local ip like it is in ata box logs ?
 
Cheers
 

Subject: RE: Replace domain name
Replied by: Chris Bomba on 02-11-2011 09:43:41 AM
I am also trying to accomplish this.  I have an invite that is pointing to a domain instead of the IP of CUCM.  I need to replace the domain with the IP address of CUCM so the CallManager will try and route the call.

INVITE sip:7103;[email protected]:5060;maddr=10.73.109.14;transport=tcp;user=phone;x-nt-redirect=redirect-server SIP/2.0
From: "Yannick MICHEL"<sip:2159;[email protected];user=phone>;tag=1ff7a768-1b2490a-13c4-40030-1d794b-19618c4c-1d794b
To: <sip:7103;[email protected];user=phone>
Call-ID: 1fc30388-1b2490a-13c4-40030-1d794b-5b878fce-1d794b
CSeq: 1 INVITE
Via: SIP/2.0/TCP 10.73.178.1:5060;branch=z9hG4bK-1d794b-7321ce98-76f323ae
Max-Forwards: 70
Supported: 100rel,x-nortel-sipvc,replaces,timer
User-Agent: Nortel CS1000 SIP GW release_5.0 version_sse-5.50.12
P-Asserted-Identity: "Yannick MICHEL"<sip:2159;[email protected];user=phone>
Privacy: none
History-Info: <sip:7103;[email protected];user=phone>;index=1
x-nt-corr-id: 000000210e181b020b@00c08b0b086c-e4cf5411
Contact: <sip:2159;[email protected]:5060;maddr=10.73.178.1;transport=tcp;user=phone>
Min-SE: 0
Allow: INVITE,ACK,BYE,REGISTER,REFER,NOTIFY,CANCEL,PRACK,OPTIONS,INFO,SUBSCRIBE,UPDATE
Content-Type: multipart/mixed;boundary=unique-boundary-1
Content-Length: 791

[email protected]  needs to be replaced with 10.73.109.14???


Thanks,
Chris
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