<html><head><meta http-equiv="Content-Type" content="text/html charset=us-ascii"></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space;">we shouldn't have to play workaround games... <div><br></div><div>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.</div><div><br></div><div>Mike</div><div><br><div><div>On Apr 10, 2014, at 2:21 PM, Steven Ayre <<a href="mailto:steveayre@gmail.com">steveayre@gmail.com</a>> wrote:</div><br class="Apple-interchange-newline"><blockquote type="cite">In this case looks like the bleg hangs up with NO_ANSWER fine<div><br></div><div>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.</div>
<div><br></div><div>hangup_after_bridge only applies to a 'successful' bridge which means 180/183/200 received</div><div><br></div><div>You could try this after the bridge:</div><div><br></div><div><action application="hangup" data="${bridge_hangup_cause}"></div>
<div><br></div><div><span></span><br><br>On Thursday, April 10, 2014, Brian West <<a href="mailto:brian@freeswitch.org">brian@freeswitch.org</a>> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
Not all Q.850 codes map into SIP, I know we are missing some that can not map correctly.<br>
<br>
--<br>
Brian West<br>
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iNUM: +883 5100 1420 9001<br>
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On Apr 10, 2014, at 6:19 AM, Vladimir Getmanshchuk <<a>vladget@gmail.com</a>> wrote:<br>
<br>
> Hello Everyone!<br>
><br>
> I have faced same issue as Ross at this thread: <a href="http://lists.freeswitch.org/pipermail/freeswitch-users/2011-January/066985.html" target="_blank">http://lists.freeswitch.org/pipermail/freeswitch-users/2011-January/066985.html</a><br>
><br>
> FS translates q.850 codes from b-leg to a-leg incorrect.<br>
> Looks like there is no solution to translate q850 in correct way?<br>
><br>
> Please advice!<br>
><br>
><br>
><br>
> Additional info:<br>
><br>
> # freeswitch -version<br>
> FreeSWITCH version: 1.2.11+git~20130711T181514Z~2eab94be78 (git 2eab94b 2013-07-11 18:15:14Z)<br>
><br>
><br>
><br>
> dialplan extension:<br>
> <extension name="default"><br>
> <condition field="destination_number" expression="^\+?222222222222"><br>
> <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/<a href="http://222222222222@2.2.2.2:5062/" target="_blank">222222222222@2.2.2.2:5062</a>"/><br>
> </condition><br>
> </extension><br>
><br>
><br>
> SIP trace:<br>
> #<br>
> U 2014/04/10 10:19:43.079198 <a href="http://2.2.2.2:5062/" target="_blank">2.2.2.2:5062</a> -> <a href="http://1.1.1.1:5080/" target="_blank">1.1.1.1:5080</a><br>
> SIP/2.0 480 Temporarily not available.<br>
> Via: SIP/2.0/UDP 1.1.1.1:5080;rport=5080;branch=z9hG4bKBXQm2F3cHgFcK.<br>
> From: "+111111111111" <<a>sip:+111111111111@1.1.1.1</a>>;tag=t8yp14HU53Xrj.<br>
> To: <<a href="http://sip:+222222222222@4.4.4.4:5060/" target="_blank">sip:+222222222222@4.4.4.4:5060</a>>;tag=gK0699dffd.<br>
> Call-ID: 5f60343b-3b3c-1232-d9ba-020820d8e22d.<br>
> CSeq: 58226505 INVITE.<br>
> Reason: Q.850;cause=19.<br>
> Content-Length: 0.<br>
> .<br>
><br>
> #<br>
> U 2014/04/10 10:19:43.084570 <a href="http://1.1.1.1:5060/" target="_blank">1.1.1.1:5060</a> -> <a href="http://2.2.2.2:5060/" target="_blank">2.2.2.2:5060</a><br>
> SIP/2.0 480 Temporarily Unavailable.<br>
> Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bK26dc.1612d515.0.<br>
> Via: SIP/2.0/UDP 3.3.3.3:5060;branch=z9hG4bK420c53ac;rport=5060.<br>
> Max-Forwards: 16.<br>
> From: "pbx234" <<a>sip:111111111111@3.3.3.3</a>>;tag=as333cf884.<br>
> To: <<a href="sip:0599#">sip:0599#</a><a href="http://222222222222@2.2.2.2:5060/" target="_blank">222222222222@2.2.2.2:5060</a>>;tag=veXaXcyc8118e.<br>
> Call-ID: <a href="http://6c86be36267a0f6d54a34060364a3490@3.3.3.3:5060/" target="_blank">6c86be36267a0f6d54a34060364a3490@3.3.3.3:5060</a>.<br>
> CSeq: 102 INVITE.<br>
> User-Agent: FS.<br>
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, NOTIFY.<br>
> Supported: timer, precondition, path, replaces.<br>
> Allow-Events: talk, hold, conference, refer.<br>
> Reason: Q.850;cause=16;text="NORMAL_CLEARING".<br>
> Content-Length: 0.<br>
> Remote-Party-ID: "Outbound Call" <<a>sip:222222222222@2.2.2.2</a>>;party=calling;privacy=off;screen=no.<br>
><br>
><br>
><br>
><br>
><br>
> 2014-04-10 10:50:03.470035 [DEBUG] sofia.c:4937 Remote Reason: 19<br>
> 2014-04-10 10:50:03.490051 [ALERT] switch_ivr.c:828 sofia/external/<a href="http://222222222222@2.2.2.2:5062/" target="_blank">222222222222@2.2.2.2:5062</a> receive message [SIGNAL_DATA]<br>
> 2014-04-10 10:50:03.490051 [DEBUG] sofia.c:5719 Channel sofia/external/<a href="http://222222222222@2.2.2.2:5062/" target="_blank">222222222222@2.2.2.2:5062</a> entering state [terminated][480]<br>
> 2014-04-10 10:50:03.490051 [NOTICE] sofia.c:6528 Hangup sofia/external/<a href="http://222222222222@2.2.2.2:5062/" target="_blank">222222222222@2.2.2.2:5062</a> [CS_EXCHANGE_MEDIA] [NO_ANSWER]<br>
> 2014-04-10 10:50:03.490051 [DEBUG] switch_channel.c:3130 Send signal sofia/external/<a href="http://222222222222@2.2.2.2:5062/" target="_blank">222222222222@2.2.2.2:5062</a>> _________________________________________________________________________<br>
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</blockquote></div>
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