February 2007 - freeswitch-dev@lists.freeswitch.org

[Freeswitch-dev] 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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[Freeswitch-dev] 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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[Freeswitch-dev] Minimum Requirements
by Shidan 28 Feb '07

28 Feb '07

Hi, just wondering what the minimum requirements for running Freeswitch in terms of MIPS and memory with default modules loaded and considering no transcoding and 10 concurrent calls. Cheers, Shidan

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[Freeswitch-dev] ANSI NI-2/Q.931
by janvb＠caselaboratories.com 26 Feb '07

26 Feb '07

hi, Can someone send me the Q.921 equivalent from ANSI-NI off list? Jan

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[Freeswitch-dev] SNMP Discussion.
by Brian West 24 Feb '07

24 Feb '07

FreeSWITCHer's, I'm going to fire up a discussion about SNMP and its usage in FreeSWITCH. What things would you want to perform or monitor via snmp in FreeSWITCH? I have a basic module that acts as a subagent now that can get the session count but we are looking at a way to let modules register handlers dynamically. Also if anyone wants to assist in writing the MIB please do so... my mind goes numb looking at those things :P Thanks, Brian West FreeSWITCH

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Re: [Freeswitch-dev] Ed's Comments
by Brian West 23 Feb '07

23 Feb '07

Ed, Great point. Now Lets find a few lead people to manage freeswitch- web and freeswitch-docs and provide guidance in these areas to direct all this awesome energy everyone has. I would like to thank Mitul Limbani, who has taken a lead part of the freeswitch-web efforts but he can't do everything alone. We invite everyone to get involved and we will do our part to make the resources available of those who know the information to assist in the process. Please contact me off list if you're interested in getting involved. #freeswitch-web - for web talk. #freeswitch-docs - for documentation efforts. Thanks, Brian West FreeSWITCH On Feb 22, 2007, at 12:29 PM, EdPimentl wrote: > Wow, this subject has gotten a lot of responses. > I like to see some of that energy and passion, translated into more > FS guides, docs, how-to, . > Much of FS is a black-box and we are all wanting to benefit from > the obviously awesome raw power of FS. > > My .02cents > > sincerely and best regards, > > Ed > > Mail: edpimentl[at]gmail.com > sip: sip:2150112@truphone.com > IM: edpimentl [AOL | Jabber | Yahoo | MSN ] > Voip: edpimentl [SKype | GoogleTalk ] > > Mobile Content Marketing/Management/Digital Delivery > http://mobilecentral.ws > > Mobile ( Relevant, Ambient, Location ) based Social Network > http://TagR.mobi (Alpha) > > Mobile Payment - P2P Payment > http://agilepay.ws > > Sponsor of P2PSIP open source [viasip_ng] project > Creating a standards base SKYPE (open) infrastructure > https://sourceforge.net/projects/viasip/ > Mailing List viasip_ng(a)freelists.org

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[Freeswitch-dev] pgp key
by Brian West 22 Feb '07

22 Feb '07

http://homepage.mac.com/brian.west/ or key 0x826C6BEE from keyserver.net /b

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[Freeswitch-dev] only one freeswitch at a time (using the same log directory, at least)
by David Nicol 22 Feb '07

22 Feb '07

at one point Monday I accidentally launched a thousand freeswitch processes and crashed my box. Well that was dumb. But here is a patch to switch.c that makes freeswitch refuse to start if the process listed in freeswitch.pid is still running, and also does flock locking on the logfile to serialize the time between performing that check and writing our own pid there. -- "I've never been elected dog-catcher and I've never had an employee." -- Bruce Sterling

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[Freeswitch-dev] FreeSWITCH Website
by Brian West 21 Feb '07

21 Feb '07

FreeSWICHer's Its crunch time. I need some help from the community in #freeswitch- web. We need to pull together and get ready to roll with our new website by next monday. Please anyone interested in helping me join #freeswitch-web ASAP so we can work together and get this done in time for eTel next week. Thanks, Brian West PS you're help is greatly appreciated...

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[Freeswitch-dev] Logos
by Brian West 21 Feb '07

21 Feb '07

For now can everyone send me their logos... one per file.. needs to fit in 400x300 or less for the website.. we'll be finishing up the display of them. So get them in jpeg format to me ASAP. We want to see if we can decide before next week on a basic logo. /b

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