January 2007 - FreeSWITCH-dev - FreeSWITCH Mailing Lists

Unintended call termination
by Per Møller 11 May '07

11 May '07

I am testing FS with a range of phones. My first problem is that when I make a call from a Snom 3xx phone (www.snom.com) the moment the call is answered - the call is terminated. If I instead make the call from one of my Linksys phones, the call works fine. Below is the output from the console from the moment the call is answered until the call was terminated. I call from locally attached phone 12 to locally attached phone 11 (in this case 2 Snom 360 phones). Maybe there is some indication to the problem here: 2007-01-23 07:19:12 [DEBUG] mod_sofia.c:4447 event_callback() event [nua_i_error] status [500][Responding to a Non-Existing Request] session: sofia/standard/12(a)192.168.11.222 2007-01-23 07:19:12 [DEBUG] switch_ivr.c:1961 audio_bridge_thread() read: sofia/standard/12(a)192.168.11.222 Bad Frame.... Bubye! Somebody have any idea how to fix this, or perhaps some hints on how to debug this further? // Per ------------------------------------------------------------ 2007-01-23 07:19:12 [DEBUG] mod_sofia.c:2562 sip_i_state() Channel sofia/standard/11 entering state [ready] 2007-01-23 07:19:12 [DEBUG] mod_sofia.c:2567 sip_i_state() Remote SDP: v=0 o=root 217248721 217248722 IN IP4 192.168.11.187 s=call c=IN IP4 192.168.11.187 t=0 0 m=audio 29442 RTP/AVP 8 101 a=rtpmap:8 pcma/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 2007-01-23 07:19:12 [DEBUG] mod_sofia.c:2201 negotiate_sdp() Codec Compare [pcma:8]/[PCMA:8] 2007-01-23 07:19:12 [INFO] mod_sofia.c:1379 tech_set_codec() Set Codec sofia/standard/11 pcma/8000 20 ms 2007-01-23 07:19:12 [INFO] mod_sofia.c:1427 activate_rtp() RTP [sofia/standard/11] 192.168.11.222:16408->192.168.11.187:29442 codec: 8 ms: 20 2007-01-23 07:19:12 [DEBUG] switch_rtp.c:500 switch_rtp_create() Starting timer [soft] 160 bytes per 20000ms 2007-01-23 07:19:12 [NOTICE] mod_sofia.c:2806 sip_i_state() Channel [sofia/standard/11] has been answered 2007-01-23 07:19:12 [INFO] mod_sofia.c:1427 activate_rtp() RTP [sofia/standard/12(a)192.168.11.222] 192.168.11.222:16410->192.168.11.144:25064 codec: 8 ms: 20 2007-01-23 07:19:12 [DEBUG] switch_rtp.c:500 switch_rtp_create() Starting timer [soft] 160 bytes per 20000ms 2007-01-23 07:19:12 [DEBUG] mod_sofia.c:1528 sofia_answer_channel() Local SDP sofia/standard/12(a)192.168.11.222: v=0 o=FreeSWITCH 164101169533152819133 164101169533152819133 IN IP4 192.168.11.222 s=FreeSWITCH c=IN IP4 192.168.11.222 t=0 0 a=sendrecv m=audio 16410 RTP/AVP 8 101 a=rtpmap:8 pcma/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 2007-01-23 07:19:12 [NOTICE] switch_ivr.c:2870 switch_ivr_originate() Channel [sofia/standard/12(a)192.168.11.222] has been answered 2007-01-23 07:19:12 [DEBUG] switch_ivr.c:2894 switch_ivr_originate() Originate Resulted in Success: [sofia/standard/11] 2007-01-23 07:19:12 [DEBUG] mod_sofia.c:1890 sofia_receive_message() De-activate timed RTP! 2007-01-23 07:19:12 [DEBUG] mod_sofia.c:2562 sip_i_state() Channel sofia/standard/12(a)192.168.11.222 entering state [completed] 2007-01-23 07:19:12 [DEBUG] mod_sofia.c:1890 sofia_receive_message() De-activate timed RTP! 2007-01-23 07:19:12 [DEBUG] switch_ivr.c:3389 switch_ivr_multi_threaded_bridge() sofia/standard/11 State Change CS_HOLD -> CS_LOOPBACK 2007-01-23 07:19:12 [DEBUG] switch_core.c:3115 switch_core_session_run() (sofia/standard/11) State LOOPBACK 2007-01-23 07:19:12 [DEBUG] mod_sofia.c:1302 sofia_on_loopback() SOFIA LOOPBACK 2007-01-23 07:19:12 [NOTICE] mod_sofia.c:4467 event_callback() Hangup sofia/standard/12(a)192.168.11.222 [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER] 2007-01-23 07:19:12 [INFO] switch_channel.c:969 switch_channel_perform_hangup() Kill sofia/standard/12(a)192.168.11.222 [1] 2007-01-23 07:19:12 [DEBUG] mod_sofia.c:2562 sip_i_state() Channel sofia/standard/12(a)192.168.11.222 entering state [ready] 2007-01-23 07:19:12 [DEBUG] mod_sofia.c:4447 event_callback() event [nua_i_error] status [500][Responding to a Non-Existing Request] session: sofia/standard/12(a)192.168.11.222 2007-01-23 07:19:12 [DEBUG] switch_ivr.c:1961 audio_bridge_thread() read: sofia/standard/12(a)192.168.11.222 Bad Frame.... Bubye! 2007-01-23 07:19:12 [INFO] switch_ivr.c:1968 audio_bridge_thread() Kill sofia/standard/11 [3] 2007-01-23 07:19:12 [DEBUG] mod_sofia.c:1896 sofia_receive_message() Re-activate timed RTP! 2007-01-23 07:19:12 [DEBUG] switch_ivr.c:1975 audio_bridge_thread() BRIDGE THREAD DONE [sofia/standard/12(a)192.168.11.222] 2007-01-23 07:19:12 [NOTICE] switch_ivr.c:2001 audio_bridge_on_loopback() Hangup sofia/standard/11 [CS_LOOPBACK] [NORMAL_CLEARING] 2007-01-23 07:19:12 [INFO] switch_channel.c:969 switch_channel_perform_hangup() Kill sofia/standard/11 [1] 2007-01-23 07:19:12 [DEBUG] switch_core.c:2970 switch_core_session_run() (sofia/standard/11) State HANGUP 2007-01-23 07:19:12 [INFO] switch_ivr.c:1968 audio_bridge_thread() Kill sofia/standard/12(a)192.168.11.222 [3] 2007-01-23 07:19:12 [DEBUG] mod_sofia.c:1896 sofia_receive_message() Re-activate timed RTP! 2007-01-23 07:19:12 [DEBUG] switch_ivr.c:1975 audio_bridge_thread() BRIDGE THREAD DONE [sofia/standard/11] 2007-01-23 07:19:12 [DEBUG] switch_core.c:2970 switch_core_session_run() (sofia/standard/12(a)192.168.11.222) State HANGUP 2007-01-23 07:19:12 [DEBUG] mod_sofia.c:1251 sofia_on_hangup() Channel sofia/standard/11 hanging up, cause: NORMAL_CLEARING 2007-01-23 07:19:12 [DEBUG] mod_sofia.c:1267 sofia_on_hangup() Sending BYE to sofia/standard/11 2007-01-23 07:19:12 [DEBUG] switch_core.c:2674 switch_core_standard_on_hangup() Standard HANGUP sofia/standard/11, cause: NORMAL_CLEARING 2007-01-23 07:19:12 [DEBUG] switch_core.c:3413 switch_core_session_thread() Session 14 (sofia/standard/11) Locked, Waiting on external entities 2007-01-23 07:19:12 [INFO] switch_core.c:3418 switch_core_session_thread() Session 14 (sofia/standard/11) Ended 2007-01-23 07:19:12 [NOTICE] switch_core.c:3294 switch_core_session_destroy() Close Channel sofia/standard/11 2007-01-23 07:19:12 [DEBUG] mod_sofia.c:1251 sofia_on_hangup() Channel sofia/standard/12(a)192.168.11.222 hanging up, cause: DESTINATION_OUT_OF_ORDER 2007-01-23 07:19:12 [DEBUG] mod_sofia.c:1267 sofia_on_hangup() Sending BYE to sofia/standard/12(a)192.168.11.222 2007-01-23 07:19:12 [DEBUG] switch_core.c:2674 switch_core_standard_on_hangup() Standard HANGUP sofia/standard/12(a)192.168.11.222,cause: DESTINATION_OUT_OF_ORDER 2007-01-23 07:19:12 [DEBUG] switch_core.c:3413 switch_core_session_thread() Session 13 (sofia/standard/12(a)192.168.11.222) Locked,Waiting on external entities 2007-01-23 07:19:12 [INFO] switch_core.c:3418 switch_core_session_thread() Session 13 (sofia/standard/12(a)192.168.11.222) Ended 2007-01-23 07:19:12 [NOTICE] switch_core.c:3294 switch_core_session_destroy() Close Channel sofia/standard/12(a)192.168.11.222

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Attention Employers: Developer Available For Hire
by Anthony Minessale 01 Mar '07

01 Mar '07

Hi, One of our voulenteer developer/contributors has just decided to make a carreer move. Since he is familiar with Asterisk/OpenPBX/FreeSWITCH and has been a good supporter, I thought any employers who need a good VoIP developer would like to learn about this oppurtunity. http://www.hyperdyne.org/justinu_resume.pdf Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com JABBER:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@66.250.68.194 iax:guest@66.250.68.194/888 googletalk:freeswitch@gmail.com pstn:712-432-7800

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OpenH323/OPAL module for FreeSWITCH
by Brian West 29 Jan '07

29 Jan '07

FreeSWITCHERS, We are now looking for someone interested in coding mod_opal/ openh323, If you're willing and able to step up to the plate and code these modules I'll do my part and try to find funding to pay for your time and efforts. Anyone interested in this please email me off list and we'll make arrangements and go over a game plan. You can also contact me on IRC via bkw_ on irc.freenode.net Thanks, Brian West FreeSWITCH.org PS: If you're interested in helping fund this please email me also!

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Re: [Freeswitch-dev] mod_conference is not able to play profile sounds
by Anthony Minessale 25 Jan '07

25 Jan '07

That was an oversight. I fixed it in rev 4056 Today. Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com JABBER:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- Original Message ---- From: Paweł Pierścionek <pawel.pierscionek(a)voiceworks.pl> To: Freeswitch-dev(a)lists.freeswitch.org Sent: Wednesday, January 24, 2007 9:16:20 PM Subject: [Freeswitch-dev] mod_conference is not able to play profile sounds Hi, User entering a conference should hear a set of sounds (enter-pin, bla bla). conference_local_play_file requires full path conference_play_file function requires only a file name and prepends it with sound-prefix :) So either the conference_local_play_file should prepend the filename or the calls to conference_local_play_file should be given full-path to sound files (and apparently are not). Urtho, _______________________________________________ Freeswitch-dev mailing list Freeswitch-dev(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org ____________________________________________________________________________________ Have a burning question? Go to www.Answers.yahoo.com and get answers from real people who know.

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mod_conference is not able to play profile sounds
by Paweł Pierścionek 25 Jan '07

25 Jan '07

Hi, User entering a conference should hear a set of sounds (enter-pin, bla bla). conference_local_play_file requires full path conference_play_file function requires only a file name and prepends it with sound-prefix :) So either the conference_local_play_file should prepend the filename or the calls to conference_local_play_file should be given full-path to sound files (and apparently are not). Urtho,

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ASR/Lumenvox
by Brian West 25 Jan '07

25 Jan '07

We are pleased to announce mod_lumenvox to the SVN tree today. If you have any questions or wish to purchase speech ports (all sales support the project) please contact Mikej or myself on IRC or via email. Here is our Lumenvox Pizza demo in Javascript. http://fisheye.freeswitch.org/browse/~raw,r=3702/FreeSWITCH/trunk/ scripts/pizza.js If you happen to have Speech Ports that you have purchased for use with other projects those are directly usable in FreeSWITCH also. Thanks, Brian West FreeSWITCH

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PRI Stack.
by Brian West 24 Jan '07

24 Jan '07

Attention: We are currently looking for someone with the skill an know how to write a PRI stack from scratch that include all the maintenance routines. If you are interested in accomplishing something like this I'll do what I can to find funding for you. The end result would need to be released under the MPL/BSD or MIT style license. Please contact me off list if you're interested. We already have someone looking for this and has some money up for someone to write it. Check out http://wiki.freeswitch.org/wiki/Bounty Brian West FreeSWITCH

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mod_xml_rpc.c:76 do_config() open of xml_rpc.conf failed
by Paweł Pierścionek 24 Jan '07

24 Jan '07

Hi, With the new split config thingy I cannot get xml_rpc.conf.xml to be included :( If I copy the content of this file to the main freeswitch.xml than everything is ok but if I use the template's #include xml_rpc.conf.xml then I get this error in the log: 2007-01-24 03:23:30 [ERR] mod_xml_rpc.c:76 do_config() open of xml_rpc.conf failed and xml_rpc port is being set to 0 (=random). Urtho,

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mod_unistim needs developers.
by Brian West 24 Jan '07

24 Jan '07

Attention Everyone. The Author of chan_unistim in Asterisk has given us an MPL license to his code to use in FreeSWITCH. Now I'm Calling on someone in the community to step up and code the driver based on his code. I'll be willing to send you a Nortel phone to test and port with. If you're willing to take up this task please message me off list or bkw_ on #freeswitch. The code is written it just needs to be ported to FreeSWITCH. Someone with a bit of time and skill and willingness to learn the FreeSWITCH API should be able to do this in short order. Thanks, Brian West

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mod_xml_rpc.c:76 do_config() open of xml_rpc.conf failed
by Paweł Pierścionek 24 Jan '07

24 Jan '07

Hi, With the new split config thingy I cannot get xml_rpc.conf.xml to be included :( If I copy the content of this file to the main freeswitch.xml than everything is ok but if I use the template's #include xml_rpc.conf.xml then I get this error in the log: 2007-01-24 03:23:30 [ERR] mod_xml_rpc.c:76 do_config() open of xml_rpc.conf failed and xml_rpc port is being set to 0 (=random). Urtho,

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