we shouldn't have to play workaround games... 

these are mapped in mod_sofia.c int he function hangup_cause_to_sip.  These values are based on RFC-4497 section 8.4.1.  This should be mapping NO_ANSWER to sip cause 480.  If that is not happening, its a bug.  If that is not the desired response, please review that rfc and if that value is incorrect, please open a jira.

Mike

On Apr 10, 2014, at 2:21 PM, Steven Ayre <steveayre@gmail.com> wrote:

In this case looks like the bleg hangs up with NO_ANSWER fine

The bridge app ends and returns to dialplan, you can see it logs it's reached the end of the dialplan. I think that's why it's hanging up with 16.

hangup_after_bridge only applies to a 'successful' bridge which means 180/183/200 received

You could try this after the bridge:

<action application="hangup" data="${bridge_hangup_cause}">



On Thursday, April 10, 2014, Brian West <brian@freeswitch.org> wrote:
Not all Q.850 codes map into SIP, I know we are missing some that can not map correctly.

--
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On Apr 10, 2014, at 6:19 AM, Vladimir Getmanshchuk <vladget@gmail.com> wrote:

> Hello Everyone!
>
> I have faced same issue as Ross at this thread: http://lists.freeswitch.org/pipermail/freeswitch-users/2011-January/066985.html
>
> FS translates q.850 codes from b-leg to a-leg incorrect.
> Looks like there is no solution to translate q850 in correct way?
>
> Please advice!
>
>
>
> Additional info:
>
> # freeswitch -version
> FreeSWITCH version: 1.2.11+git~20130711T181514Z~2eab94be78 (git 2eab94b 2013-07-11 18:15:14Z)
>
>
>
> dialplan extension:
>   <extension name="default">
>     <condition field="destination_number" expression="^\+?222222222222">
>       <action application="bridge" data="{max_forwards=70,progress_timeout=30,hangup_after_bridge=true,sip_cid_type=pid,zrtp_secure_media=false,hold_music=silence[absolute_codec_string=^^:G729@20i:PCMA@20i:PCMU@20i,origination_caller_id_number=+11111111111,origination_caller_id_name=+111111111111]sofia/external/222222222222@2.2.2.2:5062"/>
>     </condition>
>   </extension>
>
>
> SIP trace:
> #
> U 2014/04/10 10:19:43.079198 2.2.2.2:5062 -> 1.1.1.1:5080
> SIP/2.0 480 Temporarily not available.
> Via: SIP/2.0/UDP 1.1.1.1:5080;rport=5080;branch=z9hG4bKBXQm2F3cHgFcK.
> From: "+111111111111" <sip:+111111111111@1.1.1.1>;tag=t8yp14HU53Xrj.
> To: <sip:+222222222222@4.4.4.4:5060>;tag=gK0699dffd.
> Call-ID: 5f60343b-3b3c-1232-d9ba-020820d8e22d.
> CSeq: 58226505 INVITE.
> Reason: Q.850;cause=19.
> Content-Length: 0.
> .
>
> #
> U 2014/04/10 10:19:43.084570 1.1.1.1:5060 -> 2.2.2.2:5060
> SIP/2.0 480 Temporarily Unavailable.
> Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bK26dc.1612d515.0.
> Via: SIP/2.0/UDP 3.3.3.3:5060;branch=z9hG4bK420c53ac;rport=5060.
> Max-Forwards: 16.
> From: "pbx234" <sip:111111111111@3.3.3.3>;tag=as333cf884.
> To: <sip:0599#222222222222@2.2.2.2:5060>;tag=veXaXcyc8118e.
> Call-ID: 6c86be36267a0f6d54a34060364a3490@3.3.3.3:5060.
> CSeq: 102 INVITE.
> User-Agent: FS.
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, NOTIFY.
> Supported: timer, precondition, path, replaces.
> Allow-Events: talk, hold, conference, refer.
> Reason: Q.850;cause=16;text="NORMAL_CLEARING".
> Content-Length: 0.
> Remote-Party-ID: "Outbound Call" <sip:222222222222@2.2.2.2>;party=calling;privacy=off;screen=no.
>
>
>
>
>
> 2014-04-10 10:50:03.470035 [DEBUG] sofia.c:4937 Remote Reason: 19
> 2014-04-10 10:50:03.490051 [ALERT] switch_ivr.c:828 sofia/external/222222222222@2.2.2.2:5062 receive message [SIGNAL_DATA]
> 2014-04-10 10:50:03.490051 [DEBUG] sofia.c:5719 Channel sofia/external/222222222222@2.2.2.2:5062 entering state [terminated][480]
> 2014-04-10 10:50:03.490051 [NOTICE] sofia.c:6528 Hangup sofia/external/222222222222@2.2.2.2:5062 [CS_EXCHANGE_MEDIA] [NO_ANSWER]
> 2014-04-10 10:50:03.490051 [DEBUG] switch_channel.c:3130 Send signal sofia/external/222222222222@2.2.2.2:5062> _________________________________________________________________________
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