January 2008 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Snom Contacts and FreeSWITCH Interop
by Brian West 23 Jan '08

23 Jan '08

Does anyone have a way to contact a real person at Snom to get some feedback on these bugs? I have four bugs one of which they have actually confirmed (Ticket#2007123010000023) and said will be fixed but I have three more that I have reported but haven't received any response on so I re- reported them today... they were previously in the respond on Ticket#2007123010000023 but its been a while since I received anything: Ticket#2008012310000047 RFC 2833 doesn't work when SRTP is enabled. (Works great on Polycom) Ticket#2008012310000038 SDP bug related to RTP/SAVP vs RTP/AVP in sdp and not accepting SAVP when you should for SRTP. (Polycom works great on this one too offers RTP/SAVP and RTP/AVP) Ticket#2008012310000029 Phone lies about its true port number in registration and subscription when using TLS. Will say one thing when it really means another. (Again polycom does this correctly) Example the phone registers to FreeSWITCH using TLS you get this in the contact <sip:1000@10.0.1.241:2048;transport=tls;line=uwohhel1> <839>:netstat -na | grep 10.0.1.241 tcp 0 0 10.0.1.250:5061 10.0.1.241:2051 ESTABLISHED Notice they aren't the same port? This will never work and requires many hacks to make this work correctly and we shouldn't have too, it should correctly report its port on register and subscribe. I would like to get these issues resolved in the next firmware release. If someone from Snom can call me.. I'm in Central Time (CST) my cellphone is listed at the bottom of this email. Thanks, Brian West FreeSWITCH.org 918-424-9278

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Freeswitch crashes randomly while executing a javascript with odbc calls.
by B Karthik 22 Jan '08

22 Jan '08

Hi, I configured freeswitch to execute this script whenever a call is made from one number to another. function init() { use("ODBC"); var db = new ODBC("test", "testuser", "testpwd"); var path = "test.mp3"; if(db.connect()) { var tn = session.getVariable("destination_number"); var sql = "select path from sounds where tn = " +tn; db.exec(sql); if(db.nextRow()) { data = db.getData(); console_log("**********Playing back " + data["path"] + "**********"); path = data["path"]; } } else { console_log("Connection to database failed!"); } db = null; bridgeCall(path); } init(); This script executes fine when the call is initiated for the first time. For subsequent calls, the script may execute or freeswitch might crash with a segmentation fault. It usually crashes when the second call is made but it is not always the same. The last three lines output in the console before crashing is - 2008-01-22 07:22:45 [DEBUG] mod_spidermonkey.c:3202 js_api_use() Loading ODBC 2008-01-22 07:22:45 [DEBUG] switch_odbc.c:145 switch_odbc_handle_connect() Connecting test Segmentation fault (core dumped) ________________________________ OS: RHEL 4. Freeswitch Version: Latest svn checkout (22nd jan 2008 11 AM) Kindly help me in resolving this issue. Thanks in Advance, B Karthik

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Re: [Freeswitch-users] Freeswitch listens on assigned SIP port sometimes but not always (Anthony Minessale)
by B Karthik 22 Jan '08

22 Jan '08

Thanks a lot for answering this query. It was a mistake from my end. I changed the port in conf/default.xml to 5090 from 5060. This port (5090) was not always in listening state but only randomly available. The port mentioned in outbound.xml is always in listening state. So connecting to that port from a sip client works correctly always. I am however not sure if this is the right way. On Jan 19, 2008 5:11 AM, <freeswitch-users-request(a)lists.freeswitch.org> wrote: > Send Freeswitch-users mailing list submissions to > freeswitch-users(a)lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request(a)lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner(a)lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of Freeswitch-users digest..." > > Today's Topics: > > 1. Freeswitch listens on assigned SIP port sometimes but not > always (B Karthik) > 2. Re: Freeswitch listens on assigned SIP port sometimes but not > always (Brian West) > 3. Re: Freeswitch listens on assigned SIP port sometimes but not > always (Anthony Minessale) > 4. Voice Mail on Windows. (Ben Levy) > 5. Re: Voice Mail on Windows. (Brian West) > > > ---------- Forwarded message ---------- > From: "B Karthik" <carthick84(a)gmail.com> > To: freeswitch-users(a)lists.freeswitch.org > Date: Thu, 17 Jan 2008 17:32:47 +0530 > Subject: [Freeswitch-users] Freeswitch listens on assigned SIP port > sometimes but not always > Hi, > > I am a newbie to the wonderful freeswitch and linux world. I compiled and > installed freeswitch snapshot on Redhat Enterprise Linux Version 4. I > changed the SIP configuration to listen on a different port. Everything > worked fine initially. > > However when freeswitch is shutdown and restarted, it fails to listen on > the port designated for SIP (5090 in my case). The problem occurs randomly > i.e whenever freeswitch is shutdown through any of the various methods > (freeswitch -stop, shutdown command, kill). Surprisingly, it becomes fine > after a few hours. > > Please help me to resolve this issue. > > Thanks in advance, > > B Karthik > > > ---------- Forwarded message ---------- > From: Brian West <brian.west(a)mac.com> > To: freeswitch-users(a)lists.freeswitch.org > Date: Fri, 18 Jan 2008 12:48:40 -0600 > Subject: Re: [Freeswitch-users] Freeswitch listens on assigned SIP port > sometimes but not always > You might want to try running the latest SVN code. I don't think that > is a problem with FreeSWITCH but the OS not releasing the port fast > enough.... but trying the latest code and see if it still happens. > > /b > > > On Jan 17, 2008, at 6:02 AM, B Karthik wrote: > > > Hi, > > > > I am a newbie to the wonderful freeswitch and linux world. I > > compiled and installed freeswitch snapshot on Redhat Enterprise > > Linux Version 4. I changed the SIP configuration to listen on a > > different port. Everything worked fine initially. > > > > However when freeswitch is shutdown and restarted, it fails to > > listen on the port designated for SIP (5090 in my case). The problem > > occurs randomly i.e whenever freeswitch is shutdown through any of > > the various methods (freeswitch -stop, shutdown command, kill). > > Surprisingly, it becomes fine after a few hours. > > > > Please help me to resolve this issue. > > > > Thanks in advance, > > > > B Karthik > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users(a)lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > ---------- Forwarded message ---------- > From: Anthony Minessale <anthmct(a)yahoo.com> > To: freeswitch-users(a)lists.freeswitch.org > Date: Fri, 18 Jan 2008 13:51:12 -0800 (PST) > Subject: Re: [Freeswitch-users] Freeswitch listens on assigned SIP port > sometimes but not always > Also, make sure you are not running any other software > that may reserve the same port. > > I do a good amount of dev on a RHEL4 box and that > problem does not exist. > > > > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale@hotmail.com > GTALK/JABBER/PAYPAL: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: Brian West <brian.west(a)mac.com> > To: freeswitch-users(a)lists.freeswitch.org > Sent: Friday, January 18, 2008 12:48:40 PM > Subject: Re: [Freeswitch-users] Freeswitch listens on assigned SIP port > sometimes but not always > > You might want to try running the latest SVN code. I don't think that > is a problem with FreeSWITCH but the OS not releasing the port fast > enough.... but trying the latest code and see if it still happens. > > /b > > > On Jan 17, 2008, at 6:02 AM, B Karthik wrote: > > > Hi, > > > > I am a newbie to the wonderful freeswitch and linux world. I > > compiled and installed freeswitch snapshot on Redhat Enterprise > > Linux Version 4. I changed the SIP configuration to listen on a > > different port. Everything worked fine initially. > > > > However when freeswitch is shutdown and restarted, it fails to > > listen on the port designated for SIP (5090 in my case). The problem > > occurs randomly i.e whenever freeswitch is shutdown through any of > > the various methods (freeswitch -stop, shutdown command, kill). > > Surprisingly, it becomes fine after a few hours. > > > > Please help me to resolve this issue. > > > > Thanks in advance, > > > > B Karthik > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users(a)lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > ------------------------------ > Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it > now.<http://us.rd.yahoo.com/evt=51733/*http://mobile.yahoo.com/;_ylt=Ahu06i62sR8…> > > > ---------- Forwarded message ---------- > From: "Ben Levy" <blevy(a)apexvoice.com> > To: <freeswitch-users(a)lists.freeswitch.org> > Date: Fri, 18 Jan 2008 14:30:35 -0800 > Subject: [Freeswitch-users] Voice Mail on Windows. > > I am new to freeswitch and have been playing around with it over a week. > I have most things working but cant seem to figure out the voicemail module. > > > > > I use the following > > > > <anti-action application="voicemail" data="$${domain} $${domain} > ${dialed_ext}"/> > > > > I receive the following: > > > > 2008-01-18 13:58:17 [INFO] mod_dptools.c:1331 audio_bridge_function() > Originate Failed. Cause: NO_ANSWER > > 2008-01-18 13:58:17 [INFO] switch_core_session.c:689 > switch_core_session_thread() Session 6 (sofia/64.52.111.160/1001) Ended > > 2008-01-18 13:58:17 [NOTICE] switch_core_session.c:691 > switch_core_session_thread() Close Channel sofia/64.52.111.160/1001 > [CS_HANGUP] > > 2008-01-18 13:58:17 [NOTICE] switch_core_state_machine.c:134 > switch_core_standard_on_execute() Execute voicemail(64.52.111.160${dialed_ext}) > > 2008-01-18 13:58:17 [DEBUG] switch_core_state_machine.c:158 > switch_core_standard_on_execute() Expanded String voicemail(64.52.111.1601001) > > 2008-01-18 13:58:22 [DEBUG] switch_ivr_play_say.c:118 > switch_ivr_phrase_macro() No language specified - Using [en] > > 2008-01-18 13:58:34 [ERR] switch_ivr_play_say.c:143 > switch_ivr_phrase_macro() can't find language tag. > > 2008-01-18 13:58:34 [WARNING] switch_scheduler.c:84 task_thread_loop() > Task was executed late by 17 seconds 1 heartbeat (core) > > 2008-01-18 14:00:29 [DEBUG] sofia.c:114 sofia_event_callback() event > [nua_r_invite] status [487][Request Terminated] session: n/a > > 2008-01-18 14:00:29 [WARNING] switch_ivr_play_say.c:324 > switch_ivr_phrase_macro() macro [voicemail_play_greeting] did not match any > patterns > > 2008-01-18 14:00:34 [DEBUG] switch_ivr_play_say.c:118 > switch_ivr_phrase_macro() No language specified - Using [en] > > 2008-01-18 14:00:34 [DEBUG] sofia.c:114 sofia_event_callback() event > [nua_i_state] status [487][Request Terminated] session: n/a > > > > > > My sofia_profile and voicemail profile both take the domain as the name. > > > > Does anyone have any hints as to what the configuration problem is? > > > > Thanks in advance…. > > > > > > Regards, > > > > Ben > > > > > ---------- Forwarded message ---------- > From: Brian West <brian.west(a)mac.com> > To: freeswitch-users(a)lists.freeswitch.org > Date: Fri, 18 Jan 2008 17:40:07 -0600 > Subject: Re: [Freeswitch-users] Voice Mail on Windows. > Is the call being answered before the voicemail app is run? This also > concerns me "Task was executed late by 17 seconds 1 heartbeat (core)" > > > /b > > On Jan 18, 2008, at 4:30 PM, Ben Levy wrote: > > > I am new to freeswitch and have been playing around with it over a > > week. I have most things working but cant seem to figure out the > > voicemail module. > > > > I use the following > > > > <anti-action application="voicemail" data="$${domain} $${domain} $ > > {dialed_ext}"/> > > > > I receive the following: > > > > 2008-01-18 13:58:17 [INFO] mod_dptools.c:1331 > > audio_bridge_function() Originate Failed. Cause: NO_ANSWER > > 2008-01-18 13:58:17 [INFO] switch_core_session.c:689 > > switch_core_session_thread() Session 6 (sofia/64.52.111.160/1001) > > Ended > > 2008-01-18 13:58:17 [NOTICE] switch_core_session.c:691 > > switch_core_session_thread() Close Channel sofia/64.52.111.160/1001 > > [CS_HANGUP] > > 2008-01-18 13:58:17 [NOTICE] switch_core_state_machine.c:134 > > switch_core_standard_on_execute() Execute voicemail(64.52.111.160 $ > > {dialed_ext}) > > 2008-01-18 13:58:17 [DEBUG] switch_core_state_machine.c:158 > > switch_core_standard_on_execute() Expanded String > > voicemail(64.52.111.160 1001) > > 2008-01-18 13:58:22 [DEBUG] switch_ivr_play_say.c:118 > > switch_ivr_phrase_macro() No language specified - Using [en] > > 2008-01-18 13:58:34 [ERR] switch_ivr_play_say.c:143 > > switch_ivr_phrase_macro() can't find language tag. > > 2008-01-18 13:58:34 [WARNING] switch_scheduler.c:84 > > task_thread_loop() Task was executed late by 17 seconds 1 heartbeat > > (core) > > 2008-01-18 14:00:29 [DEBUG] sofia.c:114 sofia_event_callback() event > > [nua_r_invite] status [487][Request Terminated] session: n/a > > 2008-01-18 14:00:29 [WARNING] switch_ivr_play_say.c:324 > > switch_ivr_phrase_macro() macro [voicemail_play_greeting] did not > > match any patterns > > 2008-01-18 14:00:34 [DEBUG] switch_ivr_play_say.c:118 > > switch_ivr_phrase_macro() No language specified - Using [en] > > 2008-01-18 14:00:34 [DEBUG] sofia.c:114 sofia_event_callback() event > > [nua_i_state] status [487][Request Terminated] session: n/a > > > > > > My sofia_profile and voicemail profile both take the domain as the > > name. > > > > Does anyone have any hints as to what the configuration problem is? > > > > Thanks in advance…. > > > > > > Regards, > > > > Ben > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users(a)lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > .

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dialplan and move temporarily
by Jonas Gauffin 22 Jan '08

22 Jan '08

Hello Can I in some way specify that an extension have moved temporarily (status 302) in the dialplan? I'm using my own server to provide FS with dialplans, and some times the user have forwarded their extension(s) using my web interface. Regards, Jonas

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Establishing call over TLS fails
by Alois Komenda 22 Jan '08

22 Jan '08

Hello, I try to establish a call between two instances of eyeBeam via FreeSWITCH using TLS. Registering works fine but setting up the call fails. In the wireshark listings I can see that the called softphone aborts the TCP connection (it sends back a message with the RES-Flag set) after receiving the first package (client hello). In the log files of the called softphone it says that there is no target domain specified. Has anyone had similar problems with FS and eyeBeam? Or with any other software in general? Thanks in advance -- Alois Komenda Fraunhofer-Einrichtung für Systeme der Kommunikationstechnik ESK

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Open TollFree Termination Route available
by Ken Rice 21 Jan '08

21 Jan '08

Hey Guys... TollFreeGateway is proud to announce over a year has passed and we have been using FreeSwitch to route calls from a variety of sources out to the PSTN. Those Sources are are pretty much anyone with a sip client... Point your US Toll Free Traffic to sip.tollfreegateway.com (there is already an example of how to do this in the freeswitch default configs) Don¹t get stuck paying your ITSP for tollfreecalls... And for Tony, MikeJ, and Brian: Keep Up the great work we have all come to appreciate! K

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How to disable Call Waiting tone on FXS and extensions as well??
by Irshad Ahamed 21 Jan '08

21 Jan '08

Hi, Please let me know how we can disable/enable call waiting on FXS and extensions as well. My assumption is it is by default enabled. But I want to disable as well. Thanks in advance... BR, Irshad.

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Re: [Freeswitch-users] Video
by Anthony Minessale 20 Jan '08

20 Jan '08

There is a mod_h26x in tree that provides a passthru codec for h261 h263 and h264. The only thing we have implemented so far with video is the "echo" application that will also echo video in a separate thread. The core code for that can be found in switch_ivr_async.c at about line 63 and above. The next goal was probably to add another thread to bridged calls to pass video back and forth the same way the audio is done. >From there we needed a strategy on how to read and write video files. Asterisk uses 2 difft files but i would prefer to be able to read/write a standard multi-media format like mp4 etc. I think a core abstraction like we did for audio files may be unnecessary and it would probably be better to make a loadable module that implements functions to read and write the video files via the application interface. Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL: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: David Knell <dave(a)3c.co.uk> To: freeswitch-users(a)lists.freeswitch.org Sent: Sunday, January 20, 2008 7:52:45 AM Subject: [Freeswitch-users] Video Hi - Could someone write a few words on the state of video calling in FS? The no media mode works fine, but I'm interested in being able to take do things with the media streams. What I've done so far: - create an H.264 passthrough codec, derived from the G.729 one; - tamper with the video codec negotiation code to get that bit to work. What I'm not sure of is how much else is implemented or not and, before I hide in a dark corner with a large pot of coffee to work through what's done and what's not, I figured I might ask.. Cheers -- Dave -- David Knell, Director, 3C Limited T: 020 8114 8901 F: 020 8692 0677 M: 07773 800623 http://www.3c.co.uk _______________________________________________ Freeswitch-users mailing list Freeswitch-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ____________________________________________________________________________________ Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping

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Video
by David Knell 20 Jan '08

20 Jan '08

Hi - Could someone write a few words on the state of video calling in FS? The no media mode works fine, but I'm interested in being able to take do things with the media streams. What I've done so far: - create an H.264 passthrough codec, derived from the G.729 one; - tamper with the video codec negotiation code to get that bit to work. What I'm not sure of is how much else is implemented or not and, before I hide in a dark corner with a large pot of coffee to work through what's done and what's not, I figured I might ask.. Cheers -- Dave -- David Knell, Director, 3C Limited T: 020 8114 8901 F: 020 8692 0677 M: 07773 800623 http://www.3c.co.uk

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Voice Mail on Windows.
by Ben Levy 18 Jan '08

18 Jan '08

I am new to freeswitch and have been playing around with it over a week. I have most things working but cant seem to figure out the voicemail module. I use the following <anti-action application="voicemail" data="$${domain} $${domain} ${dialed_ext}"/> I receive the following: 2008-01-18 13:58:17 [INFO] mod_dptools.c:1331 audio_bridge_function() Originate Failed. Cause: NO_ANSWER 2008-01-18 13:58:17 [INFO] switch_core_session.c:689 switch_core_session_thread() Session 6 (sofia/64.52.111.160/1001) Ended 2008-01-18 13:58:17 [NOTICE] switch_core_session.c:691 switch_core_session_thread() Close Channel sofia/64.52.111.160/1001 [CS_HANGUP] 2008-01-18 13:58:17 [NOTICE] switch_core_state_machine.c:134 switch_core_standard_on_execute() Execute voicemail(64.52.111.160 ${dialed_ext}) 2008-01-18 13:58:17 [DEBUG] switch_core_state_machine.c:158 switch_core_standard_on_execute() Expanded String voicemail(64.52.111.160 1001) 2008-01-18 13:58:22 [DEBUG] switch_ivr_play_say.c:118 switch_ivr_phrase_macro() No language specified - Using [en] 2008-01-18 13:58:34 [ERR] switch_ivr_play_say.c:143 switch_ivr_phrase_macro() can't find language tag. 2008-01-18 13:58:34 [WARNING] switch_scheduler.c:84 task_thread_loop() Task was executed late by 17 seconds 1 heartbeat (core) 2008-01-18 14:00:29 [DEBUG] sofia.c:114 sofia_event_callback() event [nua_r_invite] status [487][Request Terminated] session: n/a 2008-01-18 14:00:29 [WARNING] switch_ivr_play_say.c:324 switch_ivr_phrase_macro() macro [voicemail_play_greeting] did not match any patterns 2008-01-18 14:00:34 [DEBUG] switch_ivr_play_say.c:118 switch_ivr_phrase_macro() No language specified - Using [en] 2008-01-18 14:00:34 [DEBUG] sofia.c:114 sofia_event_callback() event [nua_i_state] status [487][Request Terminated] session: n/a My sofia_profile and voicemail profile both take the domain as the name. Does anyone have any hints as to what the configuration problem is? Thanks in advance.. Regards, Ben

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