December 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

forcing ptime settings
by Matthew Fong 29 Dec '09

29 Dec '09

I use the SIP Termination service from ezcall inc (grnvoip.com) and I'm having some trouble playing .wav files into the media stream using FreeSWITCH. The audio either comes out really slow, or really fast. So a 60 second .wav file is either finished playing in 90 seconds (really slow) or finishes playing in 20 seconds (really fast). I believe this is caused by different ptime values that are being setup in the session. In the FreeSWITCH console I often received this error [WARNING] mod_sofia.c:808 We were told to use ptime 30 but what they meant to say was 20 I tried forcing the codec and ptime using absolute_codec_string='PCMU@30i' and it seemed to fix the really slow playback problem. but now I'm getting a [WARNING] mod_sofia.c:808 We were told to use ptime 30 but what they meant to say was 10 error and in some sessions the audio is playing back too fast (at 3x the speed). Is there a way I can force ptime to be 30 and avoid FreeSWITCH "fixing" the ptime values? Are there any other work arounds? --matt

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Local call uses public context?
by Lars Zeb 28 Dec '09

28 Dec '09

I am trying to setup a second FS box from scratch using v16048. What can cause a local call (81002, or 9996) to use context public? It's a standard vanilla install. http://pastebin.freeswitch.org/11629 Thanks, Lars

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twitter.com/freeswitch (its not ours)
by Brian West 28 Dec '09

28 Dec '09

Dear FreeSWITCHers, Someone has registered the freeswitch name and is squatting on twitter with it. They haven't used it in over a year and I would like to have this for our project as its clearly confusing. If you own this account please contact me off list. Thanks, Brian

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sound rpms
by Joseph L. Casale 28 Dec '09

28 Dec '09

So the spec from trunk says "Soundfiles are moving into a separate spec" but I can't find this spec anywhere in svn? Anyone know where it is? Thanks! jlc

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What's problem in SVN ?
by Dome Charoenyost 28 Dec '09

28 Dec '09

Dear sir, What's problem in SVN ? Not thing update after 23/12/2009 (16055) BG Dome C.

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[CRIT] mod_event_socket.c:337 Lost 8456 events!
by Nicolas Brenner 28 Dec '09

28 Dec '09

I just got into the fs cli and when I ran a 'show calls' I got the following message: 2009-12-24 09:58:20.058365 [CRIT] mod_event_socket.c:337 Lost 8456 events! What does this mean? does it mean the event_socket did not report 8456 events? Why could this happen? The answer to this is pretty critical to me, as I make and monitor calls through the socket. Thanks for your help! Nicolas

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Billing solutions information
by Lon Baker 28 Dec '09

28 Dec '09

Hey everyone, I am researching billing solutions for Freeswitch and want to consolidate the information with what others have found, then add it to the Wiki. There are seems to be a number of billing solutions by commercial providers, claiming they can integrate with Freeswitch, but nothing concrete explaining how far they go. Do they handle processing credit cards, prepaid, postpaid, reporting, lcr, etc? Mod_nibblebill handles the basics of updating a database table. The A2Billing teams says they are planning on adding support for Freeswitch in a few months. ASTPP.org says they support Freeswitch, but the site hasn't been updated since 2008. If you know about any solutions, links to solutions or any information can you send it to me? I will organize it and add it to the wiki. Thanks! Lon

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Adding H263 Video to Existing Call Fails First Time
by Jerry Richards 28 Dec '09

28 Dec '09

After establishing an audio call between two Bria softphones, and then starting video at the caller phone, FS replies to the re-INVITE with a 200 OK with only the PCMU codec. This looks incorrect. The audio call previously negotiated to the speex/16000 codec, and the re-INVITE from the caller added the H263-1998 codec. If I re-attempt to start video at the caller, then it is successful. I put a Freeswitch log 11596 into the pastebin that contains the complete scenario: establishing audio call, first failed start video attempt, and second successful start video attempt. Best Regards, Jerry

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SNOM shared lines with TLS problems?
by Yehavi Bourvine 28 Dec '09

28 Dec '09

Hello, Is there anyone who is using SNOM with TLS encryption and shared lines and it works? We have 1.0.5pre9 connected to SNOM-820 with shared lines between 2-3 SNOM phones. The TLS is defined by adding transport=tls to the registrar field (proxy is left blank). We noticed the following behaviour: - With non-shared line UDP and TLS both work ok. - With shared lines UDP works ok. - with shared line TLS works as long as only one phone is registered. - After the second TLS shared line registers we get busy for this extension. From the SNOM trace there is no incoming call attempt at all from FreeSwitch. Anyone has this setup working and can share some tips? Thanks, __Yehavi:

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No gsmopen in trunk?
by Max Bridgewater 28 Dec '09

28 Dec '09

Hi, I just did a fresh checkout of Freeswitch from trunk but it seems mod_gsmopen is not there. Am I doing something weird or it has been removed from trunk. Please advise! Thanks, Max.

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