Hi Steven, It really works with hangup_after_bridge=false, and <action application="hangup" data="${last_bridge_hangup_cause}"/> at dialplan. Thank you. But there are no any info at wiki about hangup_after_bridge change hangup cause to normal call clearing. Or do you think it obvious? On Thu, Apr 10, 2014 at 9:21 PM, Steven Ayre <steveayre@gmail.com> wrote:
In this case looks like the bleg hangs up with NO_ANSWER fine
The bridge app ends and returns to dialplan, you can see it logs it's reached the end of the dialplan. I think that's why it's hanging up with 16.
hangup_after_bridge only applies to a 'successful' bridge which means 180/183/200 received
You could try this after the bridge:
<action application="hangup" data="${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.
-- Brian West brian@freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com
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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.h...
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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-- Yours sincerely, Vladimir Getmanshchuk