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Wikipedia Page on FreeSWITCH
by Brian West 11 May '07

11 May '07

http://en.wikipedia.org/wiki/FreeSWITCH I call on the community to go and add Notability to the Entry. Refer to this for guidance: http://en.wikipedia.org/wiki/Wikipedia:Notability Thanks, Brian West FreeSWITCH

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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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Re: [Freeswitch-dev] Automatically add user into Google Talk
by Anthony Minessale 08 May '07

08 May '07

You have wasted a great deal of time not being clear in your email. your picture with the arrows in it is very vague. test(a)gmail.com ---> FS ----> usera(a)gmail.com what does this mean? I am sure I understood your question the same way Peter did. According the the wording in your email and your graph, you are asking why it fails when freeswitch logged into a gmail account (test(a)gmail.com) tries to call a gmail user using a Googletalk client (usera(a)gmail.com) and usera does not have test on his buddy list. If that is what you are asking the reason is that the Googletalk client will not accept anything from another client (even another googletalk) unless it is authorized. Because FreeSWITCH is not a real jabber client, it will always send an authorization request along with an outbound call just in case. There is absoluetly no way for FreeSWITCH or Asterisk or anybody else to make Googletalk accept that request only the user sitting at the Googletalk application can. If you are talking about the other way around where useraa is trying to call test then you made it very unclear but that is what I think brian thought you meant when he replied. If that is the case, FreeSWITCH will auto-accept any authorization requests and return the gesture with an authorization request and another presense packet to seal the deal. You can always repeat the "+add" procedure in Googletalk for good measure. I suppose I can also ask the proverbial "have you updated your code?" because there are several changes on a daily basis that only improve the stability of the code. Consider this the answer to your question. If you have some sort of actual issue you can reliably reproduce that is not covered in this email then open a ticket on jira and work with the devlopers on irc. (but don't torture them!) 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: Farrukh Ahmed <f4fahmed(a)gmail.com> To: freeswitch-dev(a)lists.freeswitch.org Sent: Tuesday, May 8, 2007 11:05:03 AM Subject: Re: [Freeswitch-dev] Automatically add user into Google Talk Dear Peter, Hello, But i have seen that in Asterisk this will works. e.g If test(a)gmail.com connected to Asterisk and some one added him he will come online in his/her user list. Best Regards. -- Färrükh Ähmëd _______________________________________________ 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 __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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Re: [Freeswitch-dev] Automatically add user into Google Talk
by Farrukh Ahmed 08 May '07

08 May '07

Dear Peter, Hello, But i have seen that in Asterisk this will works. e.g If test(a)gmail.com connected to Asterisk and some one added him he will come online in his/her user list. Best Regards. -- Färrükh Ähmëd

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Re: [Freeswitch-dev] Automatically add user into Google Talk
by Farrukh Ahmed 07 May '07

07 May '07

Dear Brain, Hello, As far as i know and i have tested when call goes to any Gtalk user it will wait for the user to allow user which is connected to FS. e.g. test(a)gmail.com ---> FS ----> usera(a)gmail.com If usera(a)gmail.com will not allow test(a)gmail.com then he can not get the call. Best Regards. -- Färrükh Ähmëd

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Wiki Updates
by Brian West 07 May '07

07 May '07

FreeSWITCHers, Now that I am seeing many of you starting to build and use FreeSWITCH could you take some time to look over the wiki (wiki.freeswitch.org) and see if their is anything you can help clear up documentation on. Thanks, Brian West

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Automatically add user into Google Talk
by Farrukh Ahmed 07 May '07

07 May '07

Dear Developers, Hello, Is there any way that Free Switch will add the Google Talk user automatically into the buddy list. e.g If FS is connected with test(a)gmail.com and usera(a)gmail.com added test(a)gmail.com then it will be automatically added into the list of usera(a)gmail.com similarly for test(a)gmail.com Best Regards. -- Färrükh Ähmëd

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Request for spidermonkey patch testing.
by Michael Jerris 28 Apr '07

28 Apr '07

I have posted a new patch to the bug: http://jira.freeswitch.org/browse/MODLANG-3 That takes advantage of libetpan in our spidermonkey module to send smtp mail. Could a few people please test this out and provide some feedback as to if/how well it works. Thanks Mike

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Cluecon Lightning Talk Topics and Speakers
by Brian West 27 Apr '07

27 Apr '07

I have a couple of slots open at Cluecon for Lightning talks .. any one wishing to talk or has topic suggestions please let me know. http://www.cluecon.com Thanks, Brian

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SPA9000
by Brian West 20 Apr '07

20 Apr '07

Anyone happen to have access to one of these? /b

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