Thanks. I just tested and got some more data but it didn't contain any variable containing MEDIA_TIMEOUT. Perhaps it's not really set anywhere? variable_hangup_cause and variable_originate_disposition contain NORMAL_CLEARING and SUCCESS respectively. I need a var which contains the real reason for the hangup of the bridge, which in this case is MEDIA_TIMEOUT as you can see from the logs.
El jue, 16-04-2009 a las 07:37 -0500, Anthony Minessale escribió: > turn on the debug option in mod_cdr_csv and you will get something > similar to the info app only at the end of the call > > > On Thu, Apr 16, 2009 at 3:19 AM, Mikael Aleksander Bjerkeland > <[email protected]> wrote: > El mié, 15-04-2009 a las 17:43 +0200, Mikael Bjerkeland > escribió: > > > Hi, > > > > I have two scenarios I'm having trouble figuring out and I'd > be happy > > if someone could tell me what I'm doing wrong. > > > > 1. leg_delay_start=N not working > > > > I am trying to delay the origination of the second leg in a > forked > > dial with the following: > > > > <action application="bridge" > > > > data="user/[email protected],[leg_delay_start=10]openzap/1/a/99355151"/> > > > > > > However the second leg is called at exactly the same time as > the first > > one. I am away from my testing environment right now, so I'm > sorry for > > not posting my logs. It appears to me that leg_delay_start > is broken > > on at least rev 13013. > > > > > > 2. I'd like to stop processing the dialplan after a bridge, > but not on > > specific hangup causes. If I get a MEDIA_TIMEOUT hangup > cause in the > > call I'd like to continue in the dialplan. Currently I have > the > > following: > > > > <action application="set" > data="hangup_after_bridge=true"/> > > <action application="set" > data="continue_on_fail=true"/> > > <action application="bridge" > > data="user/[email protected]"/> > > <!-- I will only get here if the first bridge is > rejected or > > TODO: I get a MEDIA_TIMEOUT on it --> > > <action application="bridge" > data="openzap/1/a/99355151"/> > > > > > > Any ideas on how to accomplish this? > > > I started testing this with the following dialplan: > > <extension name="mikael-nokia+fallback"> > <condition field="destination_number" expression="^503$"> > <action application="set" > data="hangup_after_bridge=false"/> > <action application="set" > data="continue_on_fail=true"/> > <action application="bridge" > > data="user/[email protected]"/> > <action application="info"/> > <action application="set" > data="followme_extension=99355151"/> > <action application="execute_extension" > data="post_call_followme_check"/> > <action application="hangup"/> > </condition> > </extension> > > <extension name="post_call_followme_check"> > <condition field="destination_number" > expression="^post_call_followme_check$"/> > <condition field="${originate_disposition}" > expression="^MEDIA_TIMEOUT|$${continue_on_fail_causes}$" > break="on-true"> > <action application="log" data="1 Follow me transferring > call > because of orig disposition: ${originate_disposition}"/> > <action application="transfer" > data="${followme_extension}"/> > </condition> > <condition> > <action application="log" data="1 Follow me call ended > normally > with orig disposition: ${originate_disposition}."/> > <action application="hangup"/> > </condition> > </extension> > > > ${originate_disposition} never has the value of MEDIA_TIMEOUT > since the > call was answered, which is absolutely correct, so what I am > searching > for now is how to get the actual hangup cause. The info app > doesn't show > MEDIA_TIMEOUT anywhere, but my logs show this: > > 2009-04-16 10:02:34 [DEBUG] switch_ivr_bridge.c:377 > audio_bridge_thread() > sofia/internal/sip:[email protected] > ending bridge by request from read function > 2009-04-16 10:02:34 [DEBUG] switch_ivr_bridge.c:456 > audio_bridge_thread() Send signal > sofia/internal/sip:[email protected] [BREAK] > 2009-04-16 10:02:34 [DEBUG] switch_ivr_bridge.c:452 > audio_bridge_thread() BRIDGE THREAD DONE > [sofia/internal/sip:[email protected]] > 2009-04-16 10:02:34 [DEBUG] switch_ivr_bridge.c:456 > audio_bridge_thread() Send signal > sofia/internal/[email protected] [BREAK] > 2009-04-16 10:02:34 [DEBUG] switch_core_state_machine.c:508 > switch_core_session_run() > (sofia/internal/sip:[email protected]) > State EXCHANGE_MEDIA going to sleep > 2009-04-16 10:02:34 [DEBUG] switch_core_state_machine.c:397 > switch_core_session_run() > (sofia/internal/sip:[email protected]) > Running State Change CS_HANGUP > EXECUTE sofia/internal/[email protected] info() > 2009-04-16 10:02:34 [DEBUG] switch_core_state_machine.c:448 > switch_core_session_run() > (sofia/internal/sip:[email protected]) > State HANGUP > 2009-04-16 10:02:34 [DEBUG] mod_sofia.c:315 sofia_on_hangup() > Channel > sofia/internal/sip:[email protected] hanging up, > cause: > MEDIA_TIMEOUT > 2009-04-16 10:02:34 [DEBUG] mod_sofia.c:370 sofia_on_hangup() > Sending > BYE to sofia/internal/sip:[email protected] > 2009-04-16 10:02:34 [DEBUG] switch_core_state_machine.c:46 > switch_core_standard_on_hangup() > sofia/internal/sip:[email protected] Standard HANGUP, > cause: > MEDIA_TIMEOUT > 2009-04-16 10:02:34 [DEBUG] switch_core_state_machine.c:448 > switch_core_session_run() > (sofia/internal/sip:[email protected]) > State HANGUP going to sleep > 2009-04-16 10:02:34 [INFO] mod_dptools.c:946 info_function() > CHANNEL_DATA: > Channel-State: [CS_EXECUTE] > Channel-State-Number: [4] > Channel-Name: [sofia/internal/[email protected]] > Unique-ID: [d505477c-2a5c-11de-9175-4ba93d212d75] > Call-Direction: [inbound] > Presence-Call-Direction: [inbound] > Answer-State: [answered] > Channel-Read-Codec-Name: [G722] > Channel-Read-Codec-Rate: [16000] > Channel-Write-Codec-Name: [G722] > Channel-Write-Codec-Rate: [16000] > Caller-Username: [mikael-ekiga] > Caller-Dialplan: [XML] > Caller-Caller-ID-Name: [Mikael Bjerkeland] > Caller-Caller-ID-Number: [mikael-ekiga] > Caller-Network-Addr: [10.0.255.251] > Caller-Destination-Number: [503] > Caller-Unique-ID: [d505477c-2a5c-11de-9175-4ba93d212d75] > Caller-Source: [mod_sofia] > Caller-Context: [customers] > Caller-Channel-Name: > [sofia/internal/[email protected]] > Caller-Profile-Index: [1] > Caller-Profile-Created-Time: [1239868906687578] > Caller-Channel-Created-Time: [1239868906687578] > Caller-Channel-Answered-Time: [1239868911327578] > Caller-Channel-Progress-Time: [1239868907307602] > Caller-Channel-Progress-Media-Time: [1239868911327578] > Caller-Channel-Hangup-Time: [0] > Caller-Channel-Transfer-Time: [0] > Caller-Screen-Bit: [true] > Caller-Privacy-Hide-Name: [false] > Caller-Privacy-Hide-Number: [false] > Other-Leg-Username: [mikael-ekiga] > Other-Leg-Dialplan: [XML] > Other-Leg-Caller-ID-Name: [Mikael Bjerkeland] > Other-Leg-Caller-ID-Number: [21651012] > Other-Leg-Network-Addr: [10.247.3.253] > Other-Leg-Destination-Number: [sip:[email protected]] > Other-Leg-Unique-ID: [d50bf8c4-2a5c-11de-9175-4ba93d212d75] > Other-Leg-Source: [mod_sofia] > Other-Leg-Context: [customers] > Other-Leg-Channel-Name: > [sofia/internal/sip:[email protected]] > Other-Leg-Screen-Bit: [true] > Other-Leg-Privacy-Hide-Name: [false] > Other-Leg-Privacy-Hide-Number: [false] > variable_sip_received_ip: [10.0.255.251] > variable_sip_received_port: [5065] > variable_sip_via_protocol: [udp] > variable_sip_authorized: [true] > variable_sip_mailbox: [4723695000] > variable_sip_auth_username: [mikael-ekiga] > variable_sip_auth_realm: [fs.voip.domain.com] > variable_mailbox: [4723695000] > variable_user_name: [mikael-ekiga] > variable_domain_name: [fs.voip.domain.com] > variable_effective_caller_id_number: [21651012] > variable_effective_caller_id_name: [Mikael Bjerkeland] > variable_caller_id_number: [21651012] > variable_caller_id_name: [Mikael Bjerkeland] > variable_line_open_for_external_calls: [true] > variable_room_number: [800] > variable_user_context: [customers] > variable_sip_from_user: [mikael-ekiga] > variable_sip_from_uri: [[email protected]] > variable_sip_from_host: [fs.voip.domain.com] > variable_sip_from_user_stripped: [mikael-ekiga] > variable_sip_from_tag: [942742a2-ca28-de11-854f-0015c583ee77] > variable_sofia_profile_name: [internal] > variable_sip_req_user: [503] > variable_sip_req_uri: [[email protected]] > variable_sip_req_host: [fs.voip.domain.com] > variable_sip_to_user: [503] > variable_sip_to_uri: [[email protected]] > variable_sip_to_host: [fs.voip.domain.com] > variable_sip_contact_user: [mikael-ekiga] > variable_sip_contact_port: [5065] > variable_sip_contact_uri: [[email protected]:5065] > variable_sip_contact_host: [10.0.255.251] > variable_channel_name: > [sofia/internal/[email protected]] > variable_sip_call_id: > [e82d42a2-ca28-de11-854f-0015c583e...@mikael-xpsm1530] > variable_sip_user_agent: [Ekiga/3.2.0] > variable_sip_via_host: [10.0.255.251] > variable_sip_via_port: [5065] > variable_sip_via_rport: [5065] > variable_max_forwards: [70] > variable_presence_id: [[email protected]] > variable_switch_r_sdp: [v=0 > o=- 1239868973 1239868973 IN IP4 10.0.255.251 > s=Opal SIP Session > c=IN IP4 10.0.255.251 > t=0 0 > m=audio 5090 RTP/AVP 9 8 117 0 116 101 120 > a=rtpmap:9 G722/8000/1 > a=rtpmap:8 PCMA/8000/1 > a=rtpmap:117 Speex/16000/1 > a=fmtp:117 sr=16000,mode=any > a=rtpmap:0 PCMU/8000/1 > a=rtpmap:116 Speex/8000/1 > a=fmtp:116 sr=8000,mode=any > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16,32,36 > a=rtpmap:120 NSE/8000 > a=fmtp:120 192-193 > m=video 5092 RTP/AVP 119 31 > a=rtpmap:119 theora/90000 > a=fmtp:119 > delivery-method="in_band";height=576;sampling="YCbCr-4:2:0";width=704 > a=rtpmap:31 h261/90000 > a=fmtp:31 CIF=1;QCIF=1 > ] > variable_ep_codec_string: > > [g...@8000h@0i,p...@8000h@0i,sp...@16000h@0i,sp...@16000h@0i,sp...@16000h@0i,p...@8000h@0i,h...@90000h@0i] > variable_hangup_after_bridge: [false] > variable_continue_on_fail: [true] > variable_dialed_user: [mikael-nokia] > variable_dialed_domain: [fs.voip.domain.com] > variable_switch_m_sdp: [v=0 > o=Nokia-SIPUA 603233522614072812 292890395656351010 IN IP4 > 10.247.3.253 > s=FreeSWITCH > c=IN IP4 10.247.3.253 > t=0 0 > m=audio 49152 RTP/AVP 8 101 13 > a=rtpmap:8 PCMA/8000/1 > a=rtpmap:101 telephone-event/8000/1 > a=fmtp:101 0-15 > a=rtpmap:13 CN/8000/1 > a=ptime:20 > a=maxptime:200 > m=video 0 RTP/AVP 99 > a=rtpmap:99 H264/90000 > ] > variable_remote_media_ip: [10.0.255.251] > variable_remote_media_port: [5090] > variable_read_codec: [G722] > variable_read_rate: [16000] > variable_write_codec: [G722] > variable_write_rate: [16000] > variable_video_possible: [true] > variable_remote_video_ip: [10.0.255.251] > variable_remote_video_port: [5092] > variable_sip_video_fmtp: [CIF=1;QCIF=1] > variable_sip_video_pt: [31] > variable_local_media_ip: [10.100.4.192] > variable_local_media_port: [56008] > variable_local_video_ip: [10.100.4.192] > variable_local_video_port: [59022] > variable_video_read_codec: [H261] > variable_video_read_rate: [90000] > variable_video_write_codec: [H261] > variable_video_write_rate: [90000] > variable_endpoint_disposition: [ANSWER] > variable_originate_disposition: [SUCCESS] > variable_bridge_channel: > [sofia/internal/sip:[email protected]] > variable_bridge_uuid: [d50bf8c4-2a5c-11de-9175-4ba93d212d75] > variable_signal_bond: [d50bf8c4-2a5c-11de-9175-4ba93d212d75] > variable_current_application: [info] > > > > How do I get the "raw" hangup cause first mentioned below? > > " > 2009-04-16 10:02:34 [DEBUG] mod_sofia.c:315 sofia_on_hangup() > Channel > sofia/internal/sip:[email protected] hanging up, > cause: > MEDIA_TIMEOUT > " > > As mentioned earlier the origination was in fact a success, > but since I > moved out of wi-fi coverage area I got a MEDIA_TIMEOUT which > should > trigger a transfer to my cell phone number. :-) > > > > > > Thanks, > > Mikael > > > _______________________________________________ > Freeswitch-users mailing list > [email protected] > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:[email protected] > GTALK/JABBER/PAYPAL:[email protected] > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:[email protected] > iax:[email protected]/888 > googletalk:[email protected] > pstn:213-799-1400 > 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