12-06-2012 06:45 PM - edited 03-16-2019 02:36 PM
Hi All, I've seen some information on this topic but nothing that quite matches my situation yet.
CUCM 8.6 -> SIP TRUNK -> SME 8.6 -> SIP TRUNK -> 3925 -> PRI
What we are experiencing is that all outbound calls hear the say alerting ring back tone localized to the CUCM cluster regardless of the destination country that is called. At some point the CUCM/SME or gateway must be generating the alerting tone before the audio from the carrier is cutting through. I've tried several different things but can't seem to come up with the correct combination of settings to make it work. The situation is the same for two separate new CUCM cluster, one in the US and one in the UK. All calls outbound from the US hear US ringback and all calls outbound from the UK hear UK ringback. I'm sure it's something simple that I must be overlooking and any assistance is appreciated. Some relevant config shown below.
-Mike
sip
bind control source-interface Loopback0
bind media source-interface Loopback0
header-passing
error-passthru
registrar server
early-offer forced
midcall-signaling passthru
!
dial-peer voice 2001 pots
description *Outbound International*
destination-pattern 00[1-9]
progress_ind setup enable 3
progress_ind alert enable 8
progress_ind progress enable 8
progress_ind connect enable 8
port 0/0/0:15
forward-digits all
!
12-07-2012 12:45 AM
This should help.
voice-port 0/0/0:15
cptone GB
Rate if it helps.
12-07-2012 10:23 AM
Thanks, tried that. No change.
12-07-2012 04:26 AM
Can you remove all the progress_ind commands, and post "debug isdn q931" "debug ccsip message" for a call when you receive the "wrong ringback". Do not take other debugs, do not use attachments.
12-07-2012 10:30 AM
Hi Paolo, Here's the updated config on the dial peer. Very basic at this point.
===================
dial-peer voice 2001 pots
description *Outbound International*
destination-pattern 00[1-9]
port 0/0/0:15
forward-digits all
================
Call flow is: Phone in the UK set with UK Network Locale, dialing out through a local UK gateway via E1 PRI to a destination in the US. (external number mask of +49... is just for some other testing) Calling phone only hears UK ringback, and never the US ringback from the destination switch.
The SIP Trunk Profile from SME to the 3925 gateway is set as follows:
- Disable Early Media on 180 [Unchecked]
- SIP Rel1XX Options [Send PRACK if 1XX contains SDP]
The SIP trunk itself is set with MTP Required.
------Debug Output------------
1526379: Dec 7 18:22:29.427 GMT: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:+18052362881@10.32.254.32:5060 SIP/2.0
Via: SIP/2.0/TCP 10.32.3.9:5060;branch=z9hG4bK5410c788f0bda
From: "Mike Brown" <>;tag=686362~3c8ec1be-54ea-4c9f-b3ce-f353addaa6bf-43278881>
To: <>>
Date: Fri, 07 Dec 2012 18:22:29 GMT
Call-ID: dc36600-c2133e5-53cf7-903200a@10.32.3.9
Supported: 100rel,timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM8.6
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
Cisco-Guid: 0230909440-0000065536-0000000242-0151199754
Session-Expires: 1800
P-Asserted-Identity: "Mike Brown" <>>
Remote-Party-ID: "Mike Brown" <>;party=calling;screen=yes;privacy=off>
Contact: <>>
Max-Forwards: 69
Content-Type: application/sdp
Content-Length: 211
v=0
o=CiscoSystemsCCM-SIP 686362 1 IN IP4 10.32.3.9
s=SIP Call
c=IN IP4 10.32.254.31
t=0 0
m=audio 27264 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
1526380: Dec 7 18:22:29.431 GMT: ISDN Se0/0/0:15 Q931: Applying typeplan for sw-type 0x12 is 0x0 0x1, Calling num +496927223224
1526381: Dec 7 18:22:29.431 GMT: ISDN Se0/0/0:15 Q931: Sending SETUP callref = 0x0196 callID = 0x8117 switch = primary-net5 interface = User
1526382: Dec 7 18:22:29.431 GMT: ISDN Se0/0/0:15 Q931: TX -> SETUP pd = 8 callref = 0x0196
Sending Complete
Bearer Capability i = 0x8090A3
Standard = CCITT
Transfer Capability = Speech
Transfer Mode = Circuit
Transfer Rate = 64 kbit/s
Channel ID i = 0xA9839F
Exclusive, Channel 31
Display i = 'Mike Brown'
Calling Party Number i = 0x0181, '+496927223224'
Plan:ISDN, Type:Unknown
Called Party Number i = 0x81, '0018052362881'
Plan:ISDN, Type:Unknown
1526383: Dec 7 18:22:29.431 GMT: //589/0DC366000000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 10.32.3.9:5060;branch=z9hG4bK5410c788f0bda
From: "Mike Brown" <>;tag=686362~3c8ec1be-54ea-4c9f-b3ce-f353addaa6bf-43278881>
To: <>>
Date: Fri, 07 Dec 2012 18:22:29 GMT
Call-ID: dc36600-c2133e5-53cf7-903200a@10.32.3.9
CSeq: 101 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
1526384: Dec 7 18:22:29.603 GMT: ISDN Se0/0/0:15 Q931: RX <- CALL_PROC pd = 8 callref = 0x8196
Channel ID i = 0xA9839F
Exclusive, Channel 31
1526385: Dec 7 18:22:33.399 GMT: ISDN Se0/0/0:15 Q931: RX <- ALERTING pd = 8 callref = 0x8196
Progress Ind i = 0x8482 - Destination address is non-ISDN
1526386: Dec 7 18:22:33.411 GMT: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 183 Session Progress
Via: SIP/2.0/TCP 10.32.3.9:5060;branch=z9hG4bK5410c788f0bda
From: "Mike Brown" <>;tag=686362~3c8ec1be-54ea-4c9f-b3ce-f353addaa6bf-43278881>
To: <>;tag=47B9FB18-1227>
Date: Fri, 07 Dec 2012 18:22:29 GMT
Call-ID: dc36600-c2133e5-53cf7-903200a@10.32.3.9
CSeq: 101 INVITE
Require: 100rel
RSeq: 2762
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Remote-Party-ID: <0018052362881>;party=called;screen=no;privacy=off0018052362881>
Contact: <>>
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-12.x
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 247
v=0
o=CiscoSystemsSIP-GW-UserAgent 7284 8654 IN IP4 10.32.254.32
s=SIP Call
c=IN IP4 10.32.254.32
t=0 0
m=audio 23890 RTP/AVP 0 101
c=IN IP4 10.32.254.32
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
1526387: Dec 7 18:22:33.411 GMT: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
PRACK sip:+18052362881@10.32.254.32:5060;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.32.3.9:5060;branch=z9hG4bK5411275b5cc4a
From: "Mike Brown" <>;tag=686362~3c8ec1be-54ea-4c9f-b3ce-f353addaa6bf-43278881>
To: <>;tag=47B9FB18-1227>
Date: Fri, 07 Dec 2012 18:22:29 GMT
Call-ID: dc36600-c2133e5-53cf7-903200a@10.32.3.9
CSeq: 102 PRACK
RAck: 2762 101 INVITE
Allow-Events: presence, kpml
Max-Forwards: 70
Content-Length: 0
1526388: Dec 7 18:22:33.411 GMT: //589/0DC366000000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/TCP 10.32.3.9:5060;branch=z9hG4bK5411275b5cc4a
From: "Mike Brown" <>;tag=686362~3c8ec1be-54ea-4c9f-b3ce-f353addaa6bf-43278881>
To: <>;tag=47B9FB18-1227>
Date: Fri, 07 Dec 2012 18:22:33 GMT
Call-ID: dc36600-c2133e5-53cf7-903200a@10.32.3.9
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 PRACK
Content-Length: 0
1526389: Dec 7 18:22:39.627 GMT: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
CANCEL sip:+18052362881@10.32.254.32:5060 SIP/2.0
Via: SIP/2.0/TCP 10.32.3.9:5060;branch=z9hG4bK5410c788f0bda
From: "Mike Brown" <>;tag=686362~3c8ec1be-54ea-4c9f-b3ce-f353addaa6bf-43278881>
To: <>>
Date: Fri, 07 Dec 2012 18:22:29 GMT
Call-ID: dc36600-c2133e5-53cf7-903200a@10.32.3.9
CSeq: 101 CANCEL
Max-Forwards: 70
Content-Length: 0
1526390: Dec 7 18:22:39.627 GMT: //589/0DC366000000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/TCP 10.32.3.9:5060;branch=z9hG4bK5410c788f0bda
From: "Mike Brown" <>;tag=686362~3c8ec1be-54ea-4c9f-b3ce-f353addaa6bf-43278881>
To: <>>
Date: Fri, 07 Dec 2012 18:22:39 GMT
Call-ID: dc36600-c2133e5-53cf7-903200a@10.32.3.9
CSeq: 101 CANCEL
Content-Length: 0
1526391: Dec 7 18:22:39.635 GMT: //589/0DC366000000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 487 Request Cancelled
Via: SIP/2.0/TCP 10.32.3.9:5060;branch=z9hG4bK5410c788f0bda
From: "Mike Brown" <>;tag=686362~3c8ec1be-54ea-4c9f-b3ce-f353addaa6bf-43278881>
To: <>;tag=47B9FB18-1227>
Date: Fri, 07 Dec 2012 18:22:39 GMT
Call-ID: dc36600-c2133e5-53cf7-903200a@10.32.3.9
CSeq: 101 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Reason: Q.850;cause=16
Content-Length: 0
1526392: Dec 7 18:22:39.635 GMT: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:+18052362881@10.32.254.32:5060 SIP/2.0
Via: SIP/2.0/TCP 10.32.3.9:5060;branch=z9hG4bK5410c788f0bda
From: "Mike Brown" <>;tag=686362~3c8ec1be-54ea-4c9f-b3ce-f353addaa6bf-43278881>
To: <>;tag=47B9FB18-1227>
Date: Fri, 07 Dec 2012 18:22:29 GMT
Call-ID: dc36600-c2133e5-53cf7-903200a@10.32.3.9
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: presence, kpml
Content-Length: 0
1526393: Dec 7 18:22:39.647 GMT: ISDN Se0/0/0:15 Q931: TX -> DISCONNECT pd = 8 callref = 0x0196
Cause i = 0x8090 - Normal call clearing
1526394: Dec 7 18:22:39.835 GMT: ISDN Se0/0/0:15 Q931: RX <- RELEASE pd = 8 callref = 0x8196
1526395: Dec 7 18:22:39.835 GMT: ISDN Se0/0/0:15 Q931: TX -> RELEASE_COMP pd = 8 callref = 0x0196
12-07-2012 01:49 PM
The trace seems OK to me.
I think you need to take a capture of the media to confirm what it is. If (as I suspect) is UK tone, then telco is generating it, and you cannot do anything about beside complaining to them.
Discover and save your favorite ideas. Come back to expert answers, step-by-step guides, recent topics, and more.
New here? Get started with these tips. How to use Community New member guide