cancel
Showing results for 
Search instead for 
Did you mean: 
cancel
2216
Views
5
Helpful
6
Replies

CME 12.0 Caller-ID issue

Michael Durham
Level 4
Level 4

A while back I was having some trouble with outbound caller-id and someone came up with a SIP profile that corrected my issue.  And now I have another issue on the same router when it comes to caller-id on an outbound call.  Hopefully someone can come up with an idea to solve this one too.

Currently we have 3 phone numbers for three different divisions within our small company.  Two of them are working exactly as needed. But the third on is callfwd-all to 386.911.1111 and the company number is 386.400.0000.

When someone calls our number 3864000000, the call is forwarded to 386.911.1111 and is working.

The problem is, the original caller's phone number is showing up on the forwarded to phone.  We want the company's number to show up on the forwarded-to phone, NOT the callers.

ie: customer 352.000.0000 calls our office at 386.400.000 and that call is forwarded to an employee's cell phone at 386.911.1111.  Right now the incoming caller-id shown on the employee's cell phone is 352.000.0000 but we want it to show 386.400.0000 on their cell phone.  If the customer's name comes through, that is fine.  If the call is missed, we have other CDRs to get the missed call number so that is NOT a concern.  Also, we do not want to affect outbound calls made on the other phone numbers that we have when calls are placed from the office phones.

I was told that we could do what we want via a loopback-dn but I cannot seem to get that to work and my post in the Cisco forum has gone no where.

Below is the SIP output modified to protect our privacy but maybe someone could help me change the outbound caller-id for the forwarded calls on the 386.400.000 number only.

Here what we did for the other caller-id issue.

voice class sip-profiles 1
request INVITE sip-header From modify "Michael T. Durham" "CertSys"
dial-peer voice 200 voip
voice-class sip profiles 1

 

678280: Jan 27 11:40:03.476: //189840/23A67EB/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 133.99.949.69:5010;branch=z9hG4bKe91fab07ab0499aec495.0,SIP/2.0/UDP 127.0.0.8;branch=z9hGVe-25uhCDFfXmZj4MWgc6lQXHlhZdDFfcNrQdHUfXHFrcb4FsS4eo6RTaYMfUKgZP6u34D4TND25z6M6d6Ubcvk.VvXHl*
From: "MICHAEL DURHAM" <sip:13520000000@133.99.956.128:9119>;tag=as107dc6
To: <sip:13864000000@sip.itsp.com:5010>
Date: Wed, 27 Jan 2021 16:40:03 GMT
Call-ID: 3187b0307f8ce889c79804633@133.99.956.128
CSeq: 102 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-15.7.3.M3
Session-ID: 00000000000000000000000000000000;remote=119789f195b978a511ebb104
Content-Length: 0


678281: Jan 27 11:40:03.476: //189841/23A73B7F0/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:13869111111@sip.phonesexpress.com:5010 SIP/2.0
Via: SIP/2.0/UDP 213.96.213.58:5060;branch=z9hG4bK2DA7B1718
Remote-Party-ID: "MICHAEL DURHAM" <sip:13520000000@213.96.213.58>;party=calling;screen=no;privacy=off
From: "MICHAEL DURHAM" <sip:13520000000@sip.phonesexpress.com>;tag=A62F8954-2678
To: <sip:13869111111@sip.phonesexpress.com>
Date: Wed, 27 Jan 2021 16:40:03 GMT
Call-ID: 23A7D861-5FF511EB-A7F4CE82-5D8E17CB@213.96.213.58
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 0592336-16098427-28175578-15693291
User-Agent: Cisco-SIPGateway/IOS-15.7.3.M3
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1611765603
Contact: <sip:13520000000@213.96.213.58:5060>
Diversion: <sip:13864000000@213.96.213.58>;privacy=off;reason=unconditional;counter=1;screen=no
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 68
Session-ID: 119789f195b95779ebb104;remote=000000000000000000000000000000
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 247

v=0
o=CiscoSystemsSIP-GW-UserAgent 9052 9084 IN IP4 213.96.213.58
s=SIP Call
c=IN IP4 213.96.213.58
t=0 0
m=audio 17220 RTP/AVP 0 101
c=IN IP4 213.96.213.58
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

678282: Jan 27 11:40:03.520: //189841/23A73A7F0/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 213.96.213.58:5060;branch=z9hG4b718;rport=52383
From: "MICHAEL DURHAM" <sip:13520000000@sip.phonesexpress.com>;tag=A64-2678
To: <sip:13869111111@sip.phonesexpress.com>;tag=9c1e0102a8b6ae55706.9a0b
Call-ID: 23A7D861-5F1EB-A72-5D8E17CB@213.96.213.58
CSeq: 101 INVITE
Proxy-Authenticate: Digest realm="itsp.com", nonce="YBGYj2ARsoIy1ppoF5GjDwuX", qop="auth"
Server: Anv Edge Proxy 3.5
Content-Length: 0


678283: Jan 27 11:40:03.520: //189841/23A0/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:13869111111@sip.phonesexpress.com:5010 SIP/2.0
Via: SIP/2.0/UDP 213.96.213.58:5060;branch=z9hG4bB1718
From: "MICHAEL DURHAM" <sip:13520000000@sip.phonesexpress.com>;tag=A628
To: <sip:13869111111@sip.phonesexpress.com>;tag=9c1e0102a8b68706.9a0b
Date: Wed, 27 Jan 2021 16:40:03 GMT
Call-ID: 231-5FEB-A7F4CE82-5DCB@213.96.213.58
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Session-ID: 119789f1bb104;remote=5752f17423af566a71dc6
Content-Length: 0


678284: Jan 27 11:40:03.520: //189841/23A0/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:13869111111@sip.phonesexpress.com:5010 SIP/2.0
Via: SIP/2.0/UDP 213.96.213.58:5060;branch=z9hG4bK2D62F
Remote-Party-ID: "MICHAEL DURHAM" <sip:13520000000@213.96.213.58>;party=calling;screen=no;privacy=off
From: "MICHAEL DURHAM" <sip:13520000000@sip.phonesexpress.com>;tag=A62F2678
To: <sip:13869111111@sip.phonesexpress.com>
Date: Wed, 27 Jan 2021 16:40:03 GMT
Call-ID: 231-5FB-A7F4CE82-5D8E@213.96.213.58
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 0598-1607-288-156291
User-Agent: Cisco-SIPGateway/IOS-15.7.3.M3
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 102 INVITE
Timestamp: 1611765603
Contact: <sip:13520000000@213.96.213.58:5060>
Diversion: <sip:13864000000@213.96.213.58>;privacy=off;reason=unconditional;counter=1;screen=no
Expires: 180
Allow-Events: telephone-event
Proxy-Authorization: Digest username="CISCO-ITSP",realm="itsp.com",uri="sip:13869111111@sip.phonesexpress.com:5010",response="f83143c143e962686bd3b744",nonce="YBGYj2ARl2OYIy1ppoF5GjDwuX",cnonce="D4F0",qop=auth,algorithm=md5,nc=00000001
Max-Forwards: 68
Session-ID: 119789f195b954bb104;remote=000000000000000000000000
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 247

v=0
o=CiscoSystemsSIP-GW-UserAgent 9052 IN IP4 213.96.213.58
s=SIP Call
c=IN IP4 213.96.213.58
t=0 0
m=audio 17220 RTP/AVP 0 101
c=IN IP4 213.96.213.58
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

678285: Jan 27 11:40:03.564: //189841/23A70/SIP/Msg/ccsipDisplayMsg:
Received:

TDC_CME_Router#SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 213.96.213.58:5060;branch=z9hG4b62F;rport=52383
From: "MICHAEL DURHAM" <sip:13520000000@sip.phonesexpress.com>;tag=A654-2678
To: <sip:13869111111@sip.phonesexpress.com>
Call-ID: 23A61-5FB-A7F2-5D8E17CB@213.96.213.58
CSeq: 102 INVITE
Server: Anv Edge Proxy 3.5
Content-Length: 0


678286: Jan 27 11:40:04.312: //189841/23A73BA0/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 213.96.213.58:5060;rport=52383;branch=z9hG4bK2DA7C162F
Record-Route: <sip:133.99.949.69:5010;lr=on;nat=yes>
From: "MICHAEL DURHAM" <sip:13520000000@sip.phonesexpress.com>;tag=A64-2678
To: <sip:13869111111@sip.phonesexpress.com>;tag=as43b
Call-ID: 2361-5FFB-A782-5D8CB@213.96.213.58
CSeq: 102 INVITE
User-Agent: itsp Server v10.3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:127.0.0.8;itsphash=enc-aKu.c68DctC6lfXH2KC68-ct3JXH8J>
Content-Length: 0


678287: Jan 27 11:40:04.312: //189840/23A6B/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 133.99.949.69:5010;branch=z9hG4bK1fab07ab0499aec495.0,SIP/2.0/UDP 127.0.0.8;branch=z9KVe-25uhCDFrQdHUfXHFrc6lQXHlwaYj6FfcNrQdHUfXHFrcb4FsS4eo6RTaYMfUKgZP6u34D4TND25z6M6d6Ubcvk.Vvk5sHlQXHl*
From: "MICHAEL DURHAM" <sip:13520000000@133.99.956.128:9119>;tag=as10c6
To: <sip:13864000000@sip.itsp.com:5010>;tag=A62F-B2B
Date: Wed, 27 Jan 2021 16:40:03 GMT
Call-ID: 3187b06e889c79804633@133.99.956.128

TDC_CME_Router#CSeq: 102 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Remote-Party-ID: <sip:13869111111@213.96.213.58>;party=called;screen=no;privacy=off
Contact: <sip:13864000000@213.96.213.58:5060>
Record-Route: <sip:133.99.949.69:5010;lr=on;nat=yes>
Server: Cisco-SIPGateway/IOS-15.7.3.M3
Session-ID: 245eea9a457a3184;remote=119789f195b954b5b77978a511ebb104
Content-Length: 0


678289: Jan 27 11:40:13.516: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
CANCEL sip:13864000000@213.96.213.58:5060 SIP/2.0
Via: SIP/2.0/UDP 133.99.949.69:5010;branch=z907b5b1fab07ab0499aec495.0
From: "MICHAEL DURHAM" <sip:13520000000@133.99.956.128:9119>;tag=as10
To: <sip:CISCO-ITSP-INTERNAL-13864000000@sip.itsp.com:5010>
Call-ID: 3187b0307f8c3804633@133.99.956.128
CSeq: 102 CANCEL
Max-Forwards: 69
Content-Length: 0


678290: Jan 27 11:40:13.516: //189840/23A7EB/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 133.99.949.69:5010;branch=z9hG4bKa2a2.f1e97ab0499aec495.0
From: "MICHAEL DURHAM" <sip:13520000000@133.99.956.128:9119>;tag=as10e27dc6
To: <sip:13864000000@sip.itsp.com:5010>
Date: Wed, 27 Jan 2021 16:40:13 GMT
Call-ID: 3187b86e889c79804633@133.99.956.128
CSeq: 102 CANCEL
Session-ID: 245eea9a4578a3184;remote=119789f195b954b5b7b104
Content-Length: 0

6 Replies 6

Scott Leport
Level 7
Level 7

Hi there,

 

I think I was the last guy to reply in your other post. Any reason why you would want this behaviour? Wouldn't the called party lose the ability to call the originator back if they missed the call?

 

Anyway, have you seen this article? Does this help?

https://community.cisco.com/t5/collaboration-voice-and-video/configure-and-troubleshoot-call-forward-to-the-pstn-using-sip/ta-p/3118287

 

I will look at that link.  It doesn't matter why we want to do this and I have multiple sources to find the original caller-id if we need it.  Plus, we can just ask them for it.

The link did not help.  We still need a solution.....

Can you try calling-number local under Telephoney service/ Voice register Global and see if it makes any change.



Response Signature


I tried that today and it did not work.  Still shows original caller-id

Michael Durham
Level 4
Level 4

I finally figured out a solution to this issue and it does NOT include loopback-dn's to be configured.  

First, create you ephone-dn as normal with your call-forwarding number

ephone-dn 5 octal
number 13865551234 no-reg both
label DSA 
description DSA
name DSA
call-forward all 13865559876
call-forward busy 5000
call-forward noan 5000 timeout 25
corlist incoming Keyring-National

Then create a translation rule and profile

voice translation-rule 8
rule 1 /.........../ /13865551234/
voice translation-profile DSA_CLID
translate calling 8

Finally, create a very specific dial-peer as follows

dial-peer voice 205 voip
translation-profile outgoing DSA_CLID
preference 1
destination-pattern 13865559876
session protocol sipv2
session target sip-server
voice-class codec 1
voice-class sip profiles 1
dtmf-relay rtp-nte

That is it.  Very clean and simple