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Pete Matheson
Beginner

External call in, CFwAll out - doesn't work

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


1 ACCEPTED SOLUTION

Accepted Solutions

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.

Cheers, David Trad. **When you rate a persons post, you are indicating a thank you or that it helped, but at the same time you are also helping to maintain the community spirit - You don't have to rate posts and you wont be looked down upon :) *

View solution in original post

5 REPLIES 5
paolo bevilacqua
Hall of Fame Master

Configure translation-profile and rules to adapt the number, eg strip country code.

All that is explained in the product documentation.

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?

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.

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

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.

Cheers, David Trad. **When you rate a persons post, you are indicating a thank you or that it helped, but at the same time you are also helping to maintain the community spirit - You don't have to rate posts and you wont be looked down upon :) *
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