August 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Different ptime causes call to fail
by Ben N 07 Aug '12

07 Aug '12

Hi All Wondering if someone could help me out with a ptime issue I'm having? Some background info, we have a SIP client that uses G729@20ms for Wi-Fi, and G729@50ms for 3G. The 3G call is the one I'm having trouble with. When a call is made from the SIP client to the PSTN via our carrier when a ptime of 50ms is used, the call fails. The carrier does support G729 with variable ptimes. I also think that the audio stream sent back from the carrier is G729 50ms, I have confirmed this by analyzing the RTP stream in Wireshark. In SIP terms, Freeswitch receives a 183 with SDP from the carrier which gets passed to the SIP client. Freeswitch then gets a 200 OK from the carrier, but Freeswitch does not send an ACK to the carrier for this. When 20ms is used, the ACK is sent. This could just be a by-product of a different problem, but I've yet to pin-point what that is. In my vars.xml, I have both global and outbound codecs set to "=G729@50i,G729", I've also tried them the other way around but I don't think this makes a difference. I have late negotiation set to false on my internal profile, and disable transcoding set to true so that whatever is dynamically used on the A-leg, gets used on the B-leg. I'm not using absolute codec strings for G729@50i, as I would think this make things fairly static, and then affects Wi-Fi calls that are trying to use 20ms. Out of curiosity, i set the absolute codec string to 50ms ( {absolute_codec_string=G729@50i} within bridge application) and the same thing happened. Oh and also I have my codec negotiation to "generous". In terms of the call failing, what happens is that the PSTN device rings, the SIP client goes to a ringing state, but when the PSTN device is picked up the SIP client does not progress to the next step. I have noticed that the signalling by this point is still stuck on sending an ACK to the carrier, so signalling probably isn't working anyway. Finally, I have mod_com_g729 loaded with a couple of licenses on the server. The idea is to not use a license for this kind of call, as both legs are supposed to be using G729@50i. I believe this is happening too when making a call, because I can't see and encoder/decoder being created in the FS console. Has anyone had G729 with ptime of 50ms working in passthru mode when using mod_com_g729? I My Freeswitch version is about a year old now, but I would like to avoid updating if possible. I can provide console/SIP logs if required! I really am stuck on this one, so thanks in advance for any help! If this turns out to be beyond the scope of this mailing list, then Freeswitch consultation would not be out of the question.

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Adhearsion <3 FreeSWITCH
by Ben Langfeld 07 Aug '12

07 Aug '12

One of the most often-heard questions from the Adhearsion community has long been: "Do you guys support FreeSWITCH?". I joined the Adhearsion Project with a great desire for it to support FreeSWITCH, and since then we've been working towards making that happen. At the last AdhearsionConf<http://adhearsion.com/conference>we announced the future that was to become Adhearsion 2. A large part of that vision was the ability to write your voice application once, and then to enjoy the portability across any supported telephony engine. With the release of Adhearsion 2.0 back in April, we made a huge leap in that direction, adding first-class support for Voxeo PRISM<http://voxeolabs.com/prism>. Today, we are very happy to finally announce initial support for FreeSWITCH in Adhearsion. I would also like to take a moment to share the philosophy behind the version numbers used by the Adhearsion team. We are big believers in SemVer<http://semver.org/>. To put it briefly: we accept the API promise and hold it sacred. Application developers should always feel confident upgrading versions of Adhearsion within a major number (eg. 2.0 to 2.1; even 2.0 to 2.9). We promise to make sure that Adhearsion remains perfectly backward compatible as long as the major version number remains the same. Anything else is a bug and we promise to fix it. So today marks the release of Adhearsion 2.1.0. Take a moment and see some of the other exciting improvements we have made in addition to FreeSWITCH support by reading the Changelog<https://github.com/adhearsion/adhearsion/blob/v2.1.0/CHANGELOG.md>. As always, we appreciate bug reports on Github Issues<https://github.com/adhearsion/adhearsion/issues>and will be happy to answer questions or provide assistance on the Adhearsion community mailing list <http://groups.google.com/group/adhearsion>. Regards, Ben Langfeld

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Head version not compiling on windows (libspandsp issue)
by Srini K 07 Aug '12

07 Aug '12

Hi, Today I updated the FreeSWITCH to head version and tried to compile it on Windows(Visual Studio 2010). I get the folloing errors on libspandspError 9 error LNK2019: unresolved external symbol _set_lab_gamut@32 referenced in function _read_tiff_image C:\FreeSWITCH\libs\spandsp\src\t4_tx.obj libspandsp Error 10 error LNK2019: unresolved external symbol _set_lab_illuminant@16 referenced in function _read_tiff_image C:\FreeSWITCH\libs\spandsp\src\t4_tx.obj libspandsp Error 11 error LNK2019: unresolved external symbol _lab_to_srgb@16 referenced in function _row_read C:\FreeSWITCH\libs\spandsp\src\t4_tx.obj libspandsp Error 12 error LNK1120: 3 unresolved externals C:\FreeSWITCH\Win32\Debug\libspandsp.dll libspandsp Any ideas? Thanks Srini

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webrtc connecting to FreeSwitch
by Jack 07 Aug '12

07 Aug '12

Has anyone been successful calling into freeswitch with webrtc through a webpage? If so , could you share your html? Thanks, Jack

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One way audio only outbound to only AT&T mobile numbers
by Jason A. Gould 07 Aug '12

07 Aug '12

FreeSwitch noob here. I currently have a mostly working setup here but I have one rather major problem. I can send and receive calls from most numbers perfectly, the only numbers that seem to be giving me problems are AT&T mobile numbers. Inbound AT&T calls work fine, but outbound AT&T calls only have inbound audio, audio from my switch to AT&T is lost. Naturally most of the mobile numbers I need to call are AT&T. According to my VoIP provider the problem is I'm sending out asynchronous RTP (my send and receive ports are different) and that the media gateway on the other end only allows synchronous RTP so my send and receive ports need to be the same. Unfortunately I can't seem to find any setting in FS to enable sync RTP. I found this under RTP issues:  <param name="rtp-timer-name" value="soft"/> But setting rtp-timer-name to none still uses different RTP ports for the outbound traffic.

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simulating a modem transaction...
by 8hector8 07 Aug '12

07 Aug '12

Hello everybody, Here is my issue. I need to simulate a modem transaction in C. It's a V.21 modem in 200 bits/s. This modem dial a number and waits for an "request to send" sent by the reciever. This is where I have to simulate a modem. The "request to send" is composed of a 2100 Hz tone for 2,4 secondes and 32 alternations of two tones : 1650 Hz and 1850 Hz, 15 ms each. In my idea, streamFile() function seams to be the good mean. The real issue is that when those tones are sent, the transmeter will send a "message" which is 16 digits in V.21 mode. 8 data bits, no parity, 2 stop bits. 200 bits/s means that each tone is 5 ms. 1180 Hz is a "1" and 980 is a "0". So, I need to get those 16 digits messages from tones back into numbers. So I need a way to listen the signals and analyse them and this is where I don't know how to do it. Maybe, I have to create a module into freeswitch and here also I need an help to do it. Thanks for your attention. Tony BECQ -- View this message in context: http://old.nabble.com/simulating-a-modem-transaction...-tp34260552p34260552… Sent from the Freeswitch-users mailing list archive at Nabble.com.

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Re: [Freeswitch-users] Run on Windows without install
by Tóth Róbert 07 Aug '12

07 Aug '12

Thanks, the solution was installing the redistributable... On 8/7/2012 11:48 AM, Peter Olsson wrote: > It doesn't do anything special at all as far as I know. > > It's enough to copy the files and just start freeswitch.exe (or FreeSwitchConsole.exe if you're in VS2010). > > If you want to register as a service you just use "-install" to install the service. And also, if you just copy the file to another computer, you need to make sure that vcredist for your compiler is installed. > > If this isn't working for you, please get back with info about what kind of problems that occur. > > /Peter > > > Från:freeswitch-users-bounces at lists.freeswitch.org <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> [mailto:freeswitch-users-bounces at lists.freeswitch.org <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>] För Tóth Róbert > Skickat: den 7 augusti 2012 15:02 > Till:freeswitch-users at lists.freeswitch.org <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> > Ämne: [Freeswitch-users] Run on Windows without install > > Hello > > I would like to use Freeswitch on Windows without installing by the instaler. > I build FS from git, but it won't run without a previous install. > > So i would like to know what does the installer do. > Or is there an installer what does the necessary settings, but not copy the files? > > thx > Robert

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Re: [Freeswitch-users] Run on Windows without install
by Peter Olsson 07 Aug '12

07 Aug '12

It doesn't do anything special at all as far as I know. It's enough to copy the files and just start freeswitch.exe (or FreeSwitchConsole.exe if you're in VS2010). If you want to register as a service you just use "-install" to install the service. And also, if you just copy the file to another computer, you need to make sure that vcredist for your compiler is installed. If this isn't working for you, please get back with info about what kind of problems that occur. /Peter Från: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] För Tóth Róbert Skickat: den 7 augusti 2012 15:02 Till: freeswitch-users(a)lists.freeswitch.org Ämne: [Freeswitch-users] Run on Windows without install Hello I would like to use Freeswitch on Windows without installing by the instaler. I build FS from git, but it won't run without a previous install. So i would like to know what does the installer do. Or is there an installer what does the necessary settings, but not copy the files? thx Robert !DSPAM:5021101932769323245192!

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Run on Windows without install
by Tóth Róbert 07 Aug '12

07 Aug '12

Hello I would like to use Freeswitch on Windows without installing by the instaler. I build FS from git, but it won't run without a previous install. So i would like to know what does the installer do. Or is there an installer what does the necessary settings, but not copy the files? thx Robert

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freeswitch_licence_server problem
by Ale 07 Aug '12

07 Aug '12

Hello, short: after installation fs show "Can't contact licence server." and freeswitch_licence_server in console show "Unrecognised resource G.729A/0". I've purchased a license, downloaded 194 installer, mod_com_g729.so is in lib64/freeswitch/mod dir, and license file without licenze.zip is in /etc/freeswitch. Freeswitch run as user freeswitch user on a centos. Where i unload mod_g729 and load the mod_com, server correctly starts. I also try to kill the server, and manually start it a root user, but nothing change. could someone give me any hints? Thanks alessandro

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