06-10-2016 05:44 PM - edited 03-13-2019 09:30 PM
I am trying to change the Caller ID of a particular dial-peer to simply reflect the company name.
Any calls that hit this dial peer need to have the company name in the voice class sip profile reflected there. I have tried several different ways and cannot get it to appear.
When I do a ccdebug sip I can see the name reflected in the SIP Invite there but once the call establishes it passes through the particular phone's name.
So if anyone can provide a simple voice-class sip profile that will manipulate the name for any number hitting that dial peer, it would be appreciated!
I have tried:
request INVITE sip-header Remote-Party-ID modify "\"(.*)\" " "\"Company Name\" "
and
request INVITE sip-header Remote-Party-ID modify "(<sip:).*(@.*>)" "\"Company Name\" \Company Number\2"
Both fail.
06-11-2016 04:02 AM
Josh,
It would help if you could provide, source invite and expected output, then we can help to build correct script.
You can try:
request INVITE sip-header Remote-Party-ID modify "Remote-Party-ID:(.*>).*" "Remote-Party-ID:Company Name"
but I don;t know if that's what you are looking for.
Leszek
06-11-2016 10:59 AM
Thanks for the response. All we really want to do si send the Coast Capital CNAM to our ITSP for anyone using this Dial Peer.
Here is a debug ccsip messages and it looks like it's doing what I want it to do prior to sending it to our ITSP:
INVITE sip:931234567890@1.1.1.1:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK2971064848-333874408
Max-Forwards: 26
Allow: INVITE,BYE,CANCEL,ACK,INFO,PRACK,OPTIONS,SUBSCRIBE,NOTIFY,REFER,REGISTER,UPDATE
From: "T.Aspect QA" <sip:27207@1.1.1.1>;tag=0_2971064848-333874409
To: <sip:931234567890@1.1.1.1>
Call-ID: 2971064848-333874407
CSeq: 1 INVITE
Min-SE: 90
Session-Expires: 1800;Refresher=uas
Supported: timer
Contact: "T.Aspect QA" <sip:27207@1.1.1.1:5060;transport=udp>
Content-Type: application/sdp
User-Agent: Mitel-3300-ICP 13.0.1.28
P-Asserted-Identity: "T.Aspect QA" <sip:27207@1.1.1.1:5060;transport=udp>
Content-Length: 233
v=0
o=- 8642 8642 IN IP4 10.223.88.38
s=-
c=IN IP4 10.223.88.38
t=0 0
m=audio 50188 RTP/AVP 18 0 8 121 101
a=rtpmap:121 L16/16000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
Jun 10 17:25:26.743: //3270011/D446ADBE826C/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK2971064848-333874408
From: "T.Aspect QA" <sip:27207@1.1.1.1>;tag=0_2971064848-333874409
To: <sip:931234567890@1.1.1.1>
Date: Sat, 11 Jun 2016 00:25:26 GMT
Call-ID: 2971064848-333874407
CSeq: 1 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-15.4.2.T1
Content-Length: 0
Jun 10 17:25:26.743: //3270012/D446ADBE826C/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:1234567890@1.1.1.1:5060 SIP/2.0
Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK648C2757
Remote-Party-ID: "Coast Capital" <sip:27207@1.1.1.1>;party=calling;screen=no;privacy=off
From: "Coast Capital" <sip:27207@1.1.1.1>;tag=D2786AC4-20B5
To: <sip:1234567890@1.1.1.1>
Date: Sat, 11 Jun 2016 00:25:26 GMT
Call-ID: D44749E6-2EA111E6-8272E984-3246E1A9@1.1.1.1
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 3561401790-0782307814-2188175748-0843506089
User-Agent: Cisco-SIPGateway/IOS-15.4.2.T1
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1465604726
Contact: <sip:27207@1.1.1.1:5060>
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 71
Session-Expires: 1800;refresher=uas
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 298
v=0
o=CiscoSystemsSIP-GW-UserAgent 9660 5496 IN IP4 17
ABC-VOIP-RTR1.1.1.1
s=SIP Call
c=IN IP4 1.1.1.1
t=0 0
m=audio 29494 RTP/AVP 0 8 101 19
c=IN IP4 1.1.1.1
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:19 CN/8000
a=ptime:20
Jun 10 17:25:26.747: //3270012/D446ADBE826C/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK648C2757
From: "Coast Capital" <sip:27207@1.1.1.1>;tag=D2786AC4-20B5
To: <sip:16047549487@1.1.1.1>
Call-ID: D44749E6-2EA111E6-8272E984-3246E1A9@1.1.1.1
CSeq: 101 INVITE
Timestamp: 1465604726
Content-Length: 0
Jun 10 17:25:28.219: //3270012/D446ADBE826C/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 183 Session Progress
Require: 100rel
Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK648C2757
RSeq: 1
To: <sip:11234567890@1.1.1.1>;tag=3674593528-221309
From: "Coast Capital" <sip:27207@1.1.1.1>;tag=D2786AC4-20B5
Call-ID: D44749E6-2EA111E6-8272E984-3246E1A9@1.1.1.1
CSeq: 101 INVITE
Allow: PUBLISH,MESSAGE,UPDATE,PRACK,SUBSCRIBE,REFER,INFO,NOTIFY,REGISTER,OPTIONS,BYE,INVITE,ACK,CANCEL
Contact: <sip:1234567890@1.1.1.1:5060>
Content-Type: application/sdp
Accept: application/sdp
Content-Length: 204
v=0
o=SBC2-VAN 1465600433540 1465600433540 IN IP4 1.1.1.1
s=sip call
c=IN IP4 1.1.1.1
t=0 0
m=audio 22928 RTP/AVP 0 8 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
Jun 10 17:25:28.219: //3270012/D446ADBE826C/SIP/Msg/ccsipDisplayMsg:
Sent:
PRACK sip:1234567890@1.1.1.1:5060 SIP/2.0
Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK648C387C
From: "T.Aspect QA" <sip:27207@1.1.1.1>;tag=D2786AC4-20B5
To: <sip:11234567890@1.1.1.1>;tag=3674593528-221309
Date: Sat, 11 Jun 2016 00:25:26 GMT
Call-ID: D44749E6-2EA111E6-8272E984-3246E1A9@1.1.1.1
CSeq: 102 PRACK
RAck: 1 101 INVITE
Allow-Events: telephone-event
Max-Forwards: 70
Content-Length: 0
Jun 10 17:25:28.219: //3270011/D446ADBE826C/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK2971064848-333874408
From: "T.Aspect QA" <sip:27207@1.1.1.1>;tag=0_2971064848-333874409
To: <sip:931234567890@1.1.1.1>;tag=D2787088-78
Date: Sat, 11 Jun 2016 00:25:26 GMT
Call-ID: 2971064848-333874407
CSeq: 1 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Remote-Party-ID: <sip:11234567890@1.1.1.1>;party=called;screen=no;privacy=off
Contact: <sip:931234567890@1.1.1.1:5060>
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-15.4.2.T1
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 244
v=0
o=CiscoSystemsSIP-GW-UserAgent 9626 1796 IN IP4 1.1.1.1
s=SIP Call
c=IN IP4 1.1.1.1
t=0 0
m=audio 29716 RTP/AVP 0 101
c=IN IP4 1.1.1.1
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
Jun 10 17:25:38.039: //3270012/D446ADBE826C/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Session-Expires: 1800;refresher=uas
Require: timer
Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK648C2757
To: <sip:11234567890@1.1.1.1>;tag=3674593528-221309
From: "Coast Capital" <sip:27207@1.1.1.1>;tag=D2786AC4-20B5
Call-ID: D44749E6-2EA111E6-8272E984-3246E1A9@1.1.1.1
CSeq: 101 INVITE
Allow: PUBLISH,MESSAGE,UPDATE,PRACK,SUBSCRIBE,REFER,INFO,NOTIFY,REGISTER,OPTIONS,BYE,INVITE,ACK,CANCEL
Contact: <sip:11234567890@1.1.1.1:5060>
Content-Type: application/sdp
Accept: application/sdp
Content-Length: 204
v=0
o=SBC2-VAN 1465600433540 1465600433540 IN IP4 1.1.1.1
s=sip call
c=IN IP4 1.1.1.1
t=0 0
m=audio 22928 RTP/AVP 0 8 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
Jun 10 17:25:38.043: //3270012/D446ADBE826C/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:11234567890@1.1.1.1:5060 SIP/2.0
Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK648C722A9
From: "T.Aspect QA" <sip:27207@1.1.1.1>;tag=D2786AC4-20B5
To: <sip:11234567890@1.1.1.1>;tag=3674593528-221309
Date: Sat, 11 Jun 2016 00:25:26 GMT
Call-ID: D44749E6-2EA111E6-8272E984-3246E1A9@1.1.1.1
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0
06-11-2016 12:28 PM
I'm glad all is working fine.
If the answer was helpful, please mark it as correct and rate it, so that we can consider this subject closed.
Leszek
06-13-2016 09:28 AM
I was hoping for someone to review the SIP traces to acknowledge whether or not we are doing it properly.
To me it looks good but at the other end it is still not showing Coast Capital for the Calling Party Name.....It shows the internal deskphone name.
06-14-2016 02:22 AM
All looks good.
Displaying Calling name on the other side depends on the SIP ITSP configuration, so you might want to check with them what's their configuration.
The only think you might want to test additionally would be to change screen=no, to screen=yes
other than that it looks good.
Leszek
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