05-26-2012 03:07 PM - edited 03-21-2019 05:50 AM
I have an auto-attendant where option 1 redirects to an external mobile number.
When calling the auto-attendant internally, it works fine.
When calling in from an external number, the call just gets dropped.
The only differences in the debug are that when calling in from external and then getting redirected back out again - it appends a 44 (country code) before the external number.
The key differences when running a debug are:
The working debug:
006251: May 26 22:32:00.651: //3991/0CE21AB79292/SIP/Msg/ccsipDisplayMsg:
Sent:
PRACK sip:9999048@xxx.xxx.255.50:5060;nt_end_pt=YM0+~Kn9bz04c~Q021tl7_q55X3Q~UnErty5h9vicS~N*ugKomg1csSi+--PQ3DN3y1iGPatSwe*wW00vl70324f0q4~UHdWX0h~PbdPupzNcT4Qq3PlTdk!.f6g6fGI!k1CnUfzyQQ614~FUk1n5.uzvoiQ.1A;nt_server_host=xxx.xxx.255.50:5060 SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.193.197:5060;branch=z9hG4bKF73FFFFF89F
From: "PM" <sip:xxxxxxx8625@xxx.xxx.193.197>;tag=15665130-2474
To: <sip:077xxxxx656@11kers.siptrk.co.uk>;tag=1c1956414867
Date: Sat, 26 May 2012 21:32:00 GMT
Call-ID: FD0D64E-A6B111E1-9299D018-86C3778A@xxx.xxx.193.197
CSeq: 102 PRACK
RAck: 1 101 INVITE
Allow-Events: telephone-event
Max-Forwards: 70
Content-Length: 0
006252: May 26 22:32:00.707: //3991/0CE21AB79292/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
From: "PM"<sip:xxxxxxx8625@xxx.xxx.193.197>;tag=15665130-2474
To: <sip:077xxxxx656@11kers.siptrk.co.uk>;tag=1c1956414867
Call-ID: FD0D64E-A6B111E1-9299D018-86C3778A@xxx.xxx.193.197
CSeq: 102 PRACK
Via: SIP/2.0/UDP xxx.xxx.193.197:5060;rport=64680;branch=z9hG4bKF73FFFFF89F
Contact: <sip:9999048@xxx.xx.255.50:5060;nt_end_pt=YM0+~Kn9bz04c~Q021tl7_q55X3Q~UnErty5h9vicS~N*ugKomg1csSi+--PQ3DN3y1iGPatSwe*wW00vl70324f0q4~UHdWX0h~PbdPupzNcT4Qq3PlTdk!.f6g6fGI!k1CnUfzyQQ614~FUk1n5.uzvoiQ.1A;nt_server_host=xxx.xxx.255.50:5060>
User-Agent: Nortel SESM 14.0.9.7
Supported: em,timer,replaces,path,early-session,resource-priority
x-nt-party-id: -/
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
server: Audiocodes-Sip-Gateway-Media Gateway 3200/v.5.40A.042.004
Content-Length: 0
The broken debug:
006282: May 26 22:33:04.475: //3992/3443478F929A/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:xxxxxxx5536@xxx.xxx.255.50:5060;nt_end_pt=YM0+~K.d3f961iQ4Q~QJ1m0dPr75a161~KIi60~Ebd6naGg4azXk~NoShQj~Rn7_tiVk34Bq051WbA6*15OE1f!6mDV3At-ZBID6tbN~V!021w81QPuE_1Bm;nt_server_host=xxx.xxx.255.50:5060 SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.193.197:5060;branch=z9hG4bKF78D31
From: <sip:44xxxxxx8625@lon-1.e164.org.uk>;tag=15673FEC-F39
To: <sip:xxxxxxx5536@pstn1>;tag=63C29B8-4FD
Date: Sat, 26 May 2012 21:33:04 GMT
Call-ID: 764421b873384bec1747d6d6f73471dfe6b44fe83@xxx.xxx.255.50
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 0876824463-2796622305-2459619352-2260957066
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: 1338067984
Contact: <sip:xxxxxxx8625@xxx.xxx.193.197:5060>
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 294
v=0
o=CiscoSystemsSIP-GW-UserAgent 6766 1088 IN IP4 xxx.xxx.193.197
s=SIP Call
t=0 0
m=audio 18850 RTP/AVP 8 101 101
c=IN IP4 xxx.xxx.193.197
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
006283: May 26 22:33:04.483: //3996/3443478F929A/SIP/Msg/ccsipDisplayMsg:
Sent:
CANCEL sip:077xxxxx656@11kers.siptrk.co.uk:5060 SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.193.197:5060;branch=z9hG4bKF761805
From: <sip:xxxxxxxx536@xxx.xxx.193.197>;tag=15674A80-1870
To: <sip:077xxxxx656@11kers.siptrk.co.uk>
Date: Sat, 26 May 2012 21:33:04 GMT
Call-ID: 35DD7054-A6B111E1-92A8D018-86C3778A@82.16.193.197
CSeq: 101 CANCEL
Max-Forwards: 70
Timestamp: 1338067984
Reason: Q.850;cause=127
Content-Length: 0
006284: May 26 22:33:04.483: //3992/3443478F929A/SIP/Msg/ccsipDisplayMsg:
Sent:
BYE sip:01329315536@xxx.xxx.255.50:5060;nt_end_pt=YM0+~K.d3f961iQ4Q~QJ1m0dPr75a161~KIi60~Ebd6naGg4azXk~NoShQj~Rn7_tiVk34Bq051WbA6*15OE1f!6mDV3At-ZBID6tbN~V!021w81QPuE_1Bm;nt_server_host=xxx.xxx.255.50:5060 SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.193.197:5060;branch=z9hG4bKF79EB
From: <sip:44xxxxxx8625@lon-1.e164.org.uk>;tag=15673FEC-F39
To: <sip:xxxxxxxx536@pstn1>;tag=63C29B8-4FD
Date: Sat, 26 May 2012 21:33:04 GMT
Call-ID: 764421b873384bec1747d6d6f73471dfe6b44fe83@195.54.255.50
User-Agent: Cisco-SIPGateway/IOS-12.x
Max-Forwards: 70
Timestamp: 1338067984
CSeq: 102 BYE
Reason: Q.850;cause=127
Content-Length: 0
Solved! Go to Solution.
05-30-2012 05:04 PM
Hi Pete,
Can you please mark this thread as being answered, it helps people identified that a thread is finished and also helps with people doing searches for similar problems.
Cheers,
David.
05-27-2012 05:13 AM
Configure translation-profile and rules to adapt the number, eg strip country code.
All that is explained in the product documentation.
05-27-2012 09:32 AM
I'm confused as to what translation profile would need configuring - considering the AA to external number works when calling internally, but adds's a 44 when calling in externally?
05-27-2012 12:11 PM
Ok - a little more info here.
I've pointed the DDI to a single user extension.
If I CallFwdAll that extension to an external number, External calls are forwarded to a mobile as expected.
It's only when using the auto attendant - which is set for option 1 - 'Call other number' Parameter (mobile number)
I press option 1 and just get cut off.
Perhaps is there a call restriction in place which stops AA transferring calls externally?
I've tried using a different AA Script and have confirmed the 'Allow external call transfer' is checked.
05-28-2012 02:54 AM
Ok resolved - it turned out to be a codec issue after speaking to Cisco Support.
The re-routed call was using ulaw, whereas other calls were using alaw.
They set alaw onto all the dial peers and now everything works perfectly
05-30-2012 05:04 PM
Hi Pete,
Can you please mark this thread as being answered, it helps people identified that a thread is finished and also helps with people doing searches for similar problems.
Cheers,
David.
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