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G729 License Server
by Colin Mason 03 Sep '14

03 Sep '14

Hello, Is the latest "fs-201405280931-installer" compatible with FreeSWITCH 1.2? If so, is it possible to have multiple FreeSWITCH processes on a single hardware box use a single license server? Thanks, Colin

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fax baud rate 9600
by Marlok Tamas 03 Sep '14

03 Sep '14

Hi! I found this thread: http://lists.freeswitch.org/pipermail/freeswitch-users/2013-March/094032.ht… and I have the same problem (all faxes with transfer rate: 14400 are dropped) It seems that nobody ha an answer back then for the question (is there any way of forcing freeswitch to use transfer rate: 9600) The result: 2014-08-29 15:21:39.187130 [DEBUG] mod_spandsp_fax.c:512 ============================================================================== 2014-08-29 15:21:39.187130 [DEBUG] mod_spandsp_fax.c:524 Fax processing not successful - result (48) Disconnected after permitted retries. 2014-08-29 15:21:39.187130 [DEBUG] mod_spandsp_fax.c:529 Remote station id: 2014-08-29 15:21:39.187130 [DEBUG] mod_spandsp_fax.c:530 Local station id: <local_number> 2014-08-29 15:21:39.187130 [DEBUG] mod_spandsp_fax.c:531 Pages transferred: 0 2014-08-29 15:21:39.187130 [DEBUG] mod_spandsp_fax.c:533 Total fax pages: 0 2014-08-29 15:21:39.187130 [DEBUG] mod_spandsp_fax.c:534 Image resolution: 0x0 2014-08-29 15:21:39.187130 [DEBUG] mod_spandsp_fax.c:535 Transfer Rate: 14400 2014-08-29 15:21:39.187130 [DEBUG] mod_spandsp_fax.c:537 ECM status off 2014-08-29 15:21:39.187130 [DEBUG] mod_spandsp_fax.c:538 remote country: 2014-08-29 15:21:39.187130 [DEBUG] mod_spandsp_fax.c:539 remote vendor: 2014-08-29 15:21:39.187130 [DEBUG] mod_spandsp_fax.c:540 remote model: 2014-08-29 15:21:39.187130 [DEBUG] mod_spandsp_fax.c:542 ============================================================================== Thanks in advance! Cheers, tmarlok

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FreeSWITCH to Implement Video Transcoding - Sponsors Needed
by Anthony Minessale 02 Sep '14

02 Sep '14

The FreeSWITCH project intends to devote attention in the upcoming months to work towards improving video functionality. This includes working to get fully function video transcoding working inside FreeSWITCH. Currently FreeSWITCH video is limited to pass-through and with the new features available with WebRTC we have decided to focus on trying to improve the overall video capabilities in FreeSWITCH. We currently have an issue open with chrome that helps underscore the reason we need to implement native video codec functionality especially in Conferencing. https://code.google.com/p/chromium/issues/detail?id=263867 We anticipate that this effort will take as long as 6 or more months to complete and have a large cost in developer resources. We already have some sponsors and volunteers helping with the effort but we can use all the help we can get. If you are interested in helping out either by financially sponsoring the effort or helping to test and bug report, please contact us at consulting(a)freeswitch.org and let us know. -- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>* ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900

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Freeswitch Week in Review(Master Branch) August 24th-August 30th
by Kathleen King 02 Sep '14

02 Sep '14

Hello, again. This week in the FreeSWITCH master branch we had 20 commits. We had two new features with improvements to recording in mod_conference and added support for the dtmf format dtmf-relay in mod_sofia, and a bug fix in mod_rayo related to nested calls. The following bugs were squashed: 323a3d6 FS-6764 #resolve Avoid output of junk text by fs_cli. Thanks-to: Nathan Neulinger <nneul(a)neulinger.org> 2db039b FS-6720 - fix issue with SLA - had to do with answering in wrong state in mod_skinny Jira: https://jira.freeswitch.org/browse/FS-6720 dc88617 Lower log level of limit increases and releases 89b4b0c Fix nested join to calls and conferences in mod_rayo New features that were added: 0f1494a Improvements to conference recording in mod_conference 1dc4406 Added support for the dtmf format dtmf-relay in mod_sofia Jira: https://jira.freeswitch.org/browse/FS-6770 Improvements in cross platform build supports: 1c058e0 Check gcc support for -Wno-unused-result Feedback is welcome at kathleen.king(a)quentustech.com and the commits for this week are in the attached file along with the corresponding Jira links. -- Kathleen King Quentus Technologies, INC 1037 NE 65th St Suite 273 Seattle, WA 98115 Main: (253) 617-3210 Cell: (703) 859-3757 kathleen.king(a)quentustech.com

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Sip phone lose communication appears registered
by Fernando Flórez 02 Sep '14

02 Sep '14

Hello, Just setup an FS test server on amazon ec2. Everything works flawlessly but after a while sip phone arent reacheable even as they seem registered on FS and are able to do outbound calling. Any idea of what could be happening? If i restart the sip phones every starts working again for a litrle while. Thanks, Sent from my iPhone

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Freeswitch-1.4.7 source rpm
by Mujahid Ali 02 Sep '14

02 Sep '14

hi everyone, where i can get freeswitch-1.4.7-src.rpm at the following link http://files.freeswitch.org/ there is only taz.bz2 file freeswitch-1.4.7.tar.bz2 <http://files.freeswitch.org/freeswitch-1.4.7.tar.bz2> but not source rpm. is there is any 1.4.7 src rpm ? if yes than please give me the link

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Allowing calls with Mod nibblebill and zero balances
by Roland Selmer 02 Sep '14

02 Sep '14

I’ve got nibblebill working quite well and it’s a great little mod. However one use case I can’t get to work is allowing calls to happen when a users balance is 0 and I’d like to allow calls if they are ‘internal’ i.e. not using a gateway. Although I set nibble_rate=0, the dial plan doesn’t even get there as it sees a 0 balance and hits the hangup extension. Has anybody this use case working?

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Inband DTMF Woes
by dotnetdub 02 Sep '14

02 Sep '14

Hi List, We are using a carrier who insists on sending DTMF INBAND. I was using the start_dtmf application in my dialplan but this was generating a lot of talk off. I've switched to spandsp_start_dtmf application and this seems to work without any issue.! My next issue is that we have a customer who is using asterisk and an IVR that ends up in the voicemail application. Freeswitch seems to be stripping the 'BEEP' out of the RTP that the voicemail application sends. It does it a bit randomly - and I initally thought it was the spandsp start dtmf application. I've been playing with the thresholds and twists etc but it doesn't improve. I've captured the RTP and the leg from the asterisk PBX has the voicemail 'BEEP' but the leg to the carrier doesn't. I've done /event plain all when the server isn't busy to see if I can spot something happening when the BEEP is in the RTP but nothing comes up. It doesn't seem to be the span dsp dtmf stuff causing this. I'm at a bit of a loss. Anyone any ideas? Thanks, Brian

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perl startup scripts
by Preeti Trivedi 02 Sep '14

02 Sep '14

Hi, I am facing a problem with FS getting dumped while starting up. It worked fine when I had 2 startup-scripts in perl initially. I didn't face any problem in FS starting up. I added the 3rd script today and now FS is getting dumped while starting up few times. Few times, it is starting up without any problem.. The behaviour is random. Is there any parameter which I've to tune? any idea? FS version: FreeSWITCH Version 1.5.14b+git~20140804T185031Z~1960533260~64bit (git 1960533 2014-08-04 18:50:31Z 64bit) Output of 'scripts/backtrace-from-core' script: http://pastebin.freeswitch.org/23197 Regards, Preeti

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Mod_Shout - Padding mp3 stream with 1s of empty audio
by Keith Laaks 01 Sep '14

01 Sep '14

Hi, I updated an ubuntu 14.04 server (3.13.0-30-generic #55-Ubuntu SMP Fri Jul 4 21:40:53 UTC 2014 x86_64 x86_64 x86_64 GNU/Linux) used for conference calls to latest head this morning. I use mod_shout to stream audio from a radio station as the MOH. I noticed some gaps¹ every 10-20 seconds in the audio. Did not notice this before. Turning on debug in the console, I see the following message, corresponding with the above gaps' [DEBUG] mod_shout.c:952 Padding mp3 stream with 1s of empty audio. (http://<whatever url <http://<whatever> >) I have checked out the servers¹ internet connection and all looks good. Anybody else perhaps using mod_shout and also getting this? Best Regards, Keith

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