- FreeSWITCH-users - FreeSWITCH Mailing Lists

g.729
by David Villasmil 26 Jun '08

26 Jun '08

Hello all, Is there a licensed g.729 yet? When will we be able to transcode from another codec to g.729?? I know we can pass-through, but transcode? cheers d

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Recent VAD enhancement?
by UV 25 Jun '08

25 Jun '08

I’m trying to send this and it doesn’t seem to post… here’s another attempt: After upgrading from latest stable version for windows (RC4) to latest version, I’ve noticed a significant loss of audio during a call. It almost seems like there’s an enhanced VAD that triggers the audio on and off – although the VAD feature is off. Why VAD? Because as you inject audio on the receiving side, you can then hear incoming audio from the transmitting side. Although this is not noticeable when using tone generation or shout, it’s mostly evident when playing MoH (Music on hold, not Medal of Honor). It sounds like about 90% of the audio is gone. We used identical configuration files for both tests. We’re still investigating this behavior and soon we’ll dive to network level and debug outputs. Anyone else encountered that as well? No virus found in this outgoing message. Checked by AVG. Version: 8.0.100 / Virus Database: 270.4.1/1513 - Release Date: 22/06/2008 07:52

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Ast/FS Rosetta Stone
by Michael Collins 25 Jun '08

25 Jun '08

FYI, I create a new wiki page: http://wiki.freeswitch.org/wiki/Rosetta_stone I thought maybe those who know about Asterisk and FreeSWITCH both could start dropping stuff in there. I think that people coming from Asterisk might have an easier time if they had some sort of frame of reference. Note that I put a disclaimer at the top - there isn't really a one-to-one Ast-to-FS translation for most of the stuff, but if you said to an experienced Asterisk user that "conf/dialplan/default.xml" is analogous to "/etc/asterisk/extensions.conf" then he/she would at least have a starting point. I'm hoping to add more stuff like the fact that there aren't really "priorities" in FS but that you can use scripting langs, ivr menus, etc. to mimic a lot of that kind of functionality. Anyway, your thoughts and suggestions are welcomed. I will do my best to add stuff as I am able... -MC

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Meeting time today. 5PM Central
by Brian West 25 Jun '08

25 Jun '08

Don't forget the wiki meeting is at 5pm today. Brian West sip:brian@freeswitch.org

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FS and TDM
by Anton 25 Jun '08

25 Jun '08

Hello! It's interesting to have any info on how good would be FS in the TDM switching, including the proper timing and timing propagation among systems? Also many TDM hardware cards allows in-card frame bridging (so no load on CPU than on the switch) - what with support of this? Or FreeSwitch not so supposed yet to be a core of a TDM switch? Regard, Anton.

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mod_shout.c:497 write_stream_thread() MP3 encode error -3!
by e schmidbauer 24 Jun '08

24 Jun '08

i am trying to stream a conference call via icecast2 i have an extension set up as follows: <action application="record_session" data="shout://source:password@localhost /lecture.mp3"/> <action application="conference" data="fs"/> when i dial the extension, i am able to connect to the conference and the lecture.mp3 stream appears for about 5 seconds and then it disappears. also i get an error message in the freeswitch console as follows: mod_shout.c:497 write_stream_thread() MP3 encode error -3! i've tried many variations of this extension with no success. i am able to play mp3s during the conference and also record the conference to an mp3 so i know that mod_shout is working. any help would be much appreciated. thanks.

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Update to dialplan/default.xml - x9992 = show_info extension
by Michael Collins 24 Jun '08

24 Jun '08

FYI, Brian recently committed a patch that adds a new extension to the dialplan: 9992, aka "show_info". It simply does an info app then hangs up. We added it so that you have a quick and easy means of doing simple testing. For example, if you have a SIP phone registered you can just dial 9992 and see a bunch of information dumped to the CLI. Or you can transfer a call to 9992 and quickly get an info dump that might help you debug a problem. If you haven't modified conf/dialplan/default.xml then you can update to the latest trunk, delete default.xml, then do a make samples and the newest default.xml dialplan file will be ready for use. If you are on IRC then be sure to let noobs know about 9992 because it will save you (and them) lots of hassle when debugging. -MC

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iSoftPhone strange behaviour
by Ivan C Myrvold 24 Jun '08

24 Jun '08

I have a softphone, iSoftPhone, I am trying out, but when registered to FreeSwitch, it behaves strangely. That is probably the softphone's fault, but I thought I would report it here, so I could provide the developer of this phone with some more intelligent response. I have iSoftPhone on the same subnet as FreeSwitch, and it is registered ok. I then call from iSoftPhone to another registered phone on the same subnet, and answers. The SIP messages are here: http://pastebin.freeswitch.org/4688 The call are up, and I have nromal audio. I see that iSoftPhone is sending regularly the message SIP/2.0 486 Busy Here to FreeSwitch. I have reported this to the developer, but never got any response back. After 1 minute, FreeSwitch sends a new INVITE sip:imyrvold@192.168.207.101:5060 SIP/2.0 to iSoftPhone, maybe because of the "486 Busy Here" message? I lose audio now, and I have to hangup the call. Can anyone comment on the above behaviour? Ivan

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iLBC question
by Alex Gusak 24 Jun '08

24 Jun '08

We have some problems with codec iLBC. There is a scheme for passing the call: +--------+ +------------+ ClientA----->| |-------->| | | SER | | FreeSWITCH | ClientB<-----| |<--------| | +--------+ +------------+ At FreeSWITCH allowed codecs iLBC@30i, PCMA. RTP runs through FreeSWITCH. When a clients set different codecs (eg A: iLBC, B: PCMA), or both customers PCMA codec installed both the client hear each other. If two clients installed codec iLBC -- clients do not hear each other, while RTP traffic FreeSWITCH passes through (in this case from one client comes payload type 97, and to another client FreeSWITCH sends payload type 102). If we include proxy_media = true, then clients hear each other (FreeSWITCH sends payload type, which came from the customer A). But then not working transcoding. Calls test at several softphones: zoiper, XLite, SJPhone - everywhere was a problem with iLBC. Where could be the problem? How in FreeSWITCH send to clientB iLBC payload type that incoming from clientA without enabled proxy_media? Thanks. -- Alex Gusak

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clustered freeswitch
by Vieri 24 Jun '08

24 Jun '08

Hi. I'm completely new to freeswitch and would like to migrate from asterisk. I have a couple of questions though: 1) I understand I won't be able to use my B410P Digium ISDN (mISDN) BRI card yet, right? 2) my TE120P Digium PRI card should work. 3) can freeswitch work in a master-master cluster? Example: FreeSWITCH-1 --- MySQL nbcluster tables --- FreeSWITCH-2 I suppose I can do this if I enable ODBC in FreeSWITCH. But will both FreeSWITCH boxes get "confused" if they use the "same" clustered database (active/active)? (by the way, does the built-in sqlite support clustered tables?) SIP softphones and ATA phones would register via DNS SRV (or custom method) to any one of the servers but only one (the idea is to load balance registrations). Say, extension 101 registers to FS-1 and 102 registers to FS-2. Could 101 contact 102 simply by looking it up the clustered database (eg. querying the table may yield something like sip:102@FS-2)? This would be similar to Asterisk 1.4+ Realtime SIP registrations. In Asterisk 1.2 I use IAX2 friend trunks between the 2 servers and DUNDi for the lookup. Would queues and ring groups also work in a "clustered" freeswitch system? In case of "queues", where is the caller position stored? Who's counting? FS-1 or FS-2? Could the counter be stored in the clustered SQL table so that both FreeSWITCH servers get it right in case calls come in on both servers? example: PSTN---PRI---FreeSWITCH-1 --- clustered SQL --- FreeSWITCH-2---PRI---PSTN Calls coming in from the PSTN reach both FS-1 and FS-2. I would like these calls to enter a "clustered" queue which bridges calls to operator extensions which can be registered on either server. The caller positions should be coherent across both servers. Has anyone done this already? Thanks for your feedback. Vieri

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