- FreeSWITCH-users - FreeSWITCH Mailing Lists

Freeswitch as a media server
by bmsword 09 Apr '09

09 Apr '09

hi,all Can freeswitch be integrated with another softswitch as a media server? if it can, how to configure? thanks! andy.

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freeswitch on a laptop listening on 127.0.0.1 and memory consumption
by Peter P GMX 09 Apr '09

09 Apr '09

I want to do the following: Due to missing Softclient with TLS/SRTP support on my Linux laptop (Zoiper is almost there, but not yet with SRTP) I want to install a local FS to listen on a local IP and then communicate via TLS/SRTP to my FS in the Office. As My Laptop has changing IPs (e.g. Ethernet-Cable, WLAN, UMTS) I want FS to listen on 127.0.0.1, connect my local VoIP client (Twinkle) to the local FS and communicate to the FS in my office through the public or LAN IP (maybe via STUN, public IP may change due to change of network connection). 1st Question: Is that possible or is another solution preferrable? 2nd Question: How can I change the amount of memory FS tries to reserve to an absolute minumum (I only have 1 call at a time). Currently it tries to reserve about 360M if I read that right. Best regards Peter

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Re: [Freeswitch-users] Two or more simultaneous calls not
by Traun Leyden 08 Apr '09

08 Apr '09

Hi Adam, I'm stumped .. I guess you could try the following: * Try with the trunk version of freeswitch. I don't think it will matter, but just in case * Try to simulate the same test with a Lua script. Do you see the same problem? If those don't turn up anything, then the next logical step would be to start adding printf() statements in the mod_python code and find out where it is getting stuck. In particular around the parts where it swaps the threadstate in and out. I might be able to create a patch for you, but try those other tests first. HTH, Traun > > Thanks for the response Traun. The version of Python is 2.4.3, and I > didn't > build it myself, I installed it with yum. > The version of Red Hat is 4.1.2-41. > "import threading" works fine, so I don't think it's a Python threading > issue. > The FreeSWITCH version I installed is the > freeswitch-1.0.3.tar.gz< > http://files.freeswitch.org/freeswitch-1.0.3.tar.gz> > located > at files.freeswitch.org. > I didn't make any major changes to the configuration; I enabled Python and > set-up the SIP profile, directory and dialplan. No other changes. > Any other help would be appreciated, since I really don't know where to > look. > > Thanks, > Adam

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speex can't find OggS header
by can_man＠gmx.de 08 Apr '09

08 Apr '09

Hello everyone, I am trying to get a Java Sip client working with speex/16000. FS sets the codec correctly and then starts sending packets to my client: 2009-04-08 21:46:34 [DEBUG] sofia_glue.c:2732 sofia_glue_negotiate_sdp() Audio Codec Compare [speex:100:16000:0]/[SPEEX:99:16000:20] 2009-04-08 21:46:34 [DEBUG] sofia_glue.c:1857 sofia_glue_tech_set_codec() Set Codec sofia/external5090/puli@97.101.59.118:5090 SPEEX/16000 20 ms 320 samples When the packets arrive jspeex can't decode them and I started to look at them manually to find out what the problem is. The payload of each RTP packet is 42 bytes and when looking for the "OggS" header I can't find it. Or is the ogg header not needed? Jspeex looks for it and as it can't find it, it stops decoding. Is FS sending one frame per packet? Thank you very much for your help. Best wishes, Phil Ps: Wireshark tells me the following for a sample package: Real-Time Transport Protocol Setup Method: SDP 10.. .... = Version: RFC 1889 Version (2) ..0. .... = Padding: False ...0 .... = Extension: False .... 0000 = Contributing source identifiers count: 0 0... .... = Marker: False Payload type: speex (100) Sequence number: 9365 Extended sequence number: 74901 Timestamp: 23680 Synchronization Source identifier: 0x004a235c (4858716) Payload: 2C6679456EE347FFD0A3D55B133771A9A100DB639F5B8ED2... Payload is 42 bytes. -- Psssst! Schon vom neuen GMX MultiMessenger gehört? Der kann`s mit allen: http://www.gmx.net/de/go/multimessenger01

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rtp/one way audio problem
by Gabriel Kuri 08 Apr '09

08 Apr '09

We're seeing occasional one way audio issues for international calls going out to one of several carriers. On roughly 2 out of 5 calls outbound, there is no audio on the the calling party's side, however the called party indicates they can hear the calling party perfectly well. NAT is not involved anywhere on our side. Sniffing the traffic shows the rtp stream tries to start, coming in from the carrier, but then stops, which is probably why there's no audio on the calling party's side. There is however, rtp going out from us to the carrier, which is probably why the called party hears the calling party OK. I enabled proxy_media=true and that seems to have fixed the problem (or the problem coincidentally stopped), so now I'm beginning to wonder if we're hitting any of the goofy Sonus bugs described here ... http://wiki.freeswitch.org/wiki/RTP_Issues I know my carrier uses a Cisco, but I also know their routes are transit, not direct routes, so I have no idea what other softswitches the rtp is going through before it finally hits the PSTN in whatever country is being called. Is there anyway to know if there's a Sonus in the media stream somewhere? Do the Cisco's do anything goofy that I need to be aware of as well? I can turn off proxy_media and run a pcapsipdump if that will help? As a side note, we have absolutely no problem terminating domestic calls via any of our domestic carriers/CLECs. This problem seems to be only plagued with International calling. Thanks, Gabe

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Using recordFile with Icecast - looses the end of the call
by Andy Ayers 08 Apr '09

08 Apr '09

Hi, I have mod_shout installed and I'm using session.recordFile to capture the audio in a call. When I specify a local file mp3 or wav the audio is captured fine. However, I'm using an icecast server to manage the audio for me and when I specify a remote mp3 location(shout://myserver.com/myaudio.mp3) the end of the call is missing off the resultant mp3 file. A wild shot in the dark I know but does anyone have any experience of this and how it might be resolved? Many thanks Andy

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Help about Conference and it's member_id
by to vietduc 08 Apr '09

08 Apr '09

Hi! I wonder how to get out the member_id of a member in a conference room? Currently, i found that it is increasing 1 each time, so the first one enter the conference room, his/her member_id would be 1, and the second is 2. Sadly, when i close a conference room (all members leave) and re-open it later, the member_id will be increased since the last time instead of begining at 1 as the first time (said, it may be 3, 4 and go on). Why is that (as i think a usual way is that if there is 3 members in a conference room, their member_id should be 1,2,3 or 0,1,2) and is there anyway to get conference's member_id of a member programatically? Thanks in advance! Duc To

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Re: [Freeswitch-users] Two or more simultaneous calls not
by Adam Wilt 08 Apr '09

08 Apr '09

Thanks for the response Traun. The version of Python is 2.4.3, and I didn't build it myself, I installed it with yum. The version of Red Hat is 4.1.2-41. "import threading" works fine, so I don't think it's a Python threading issue. The FreeSWITCH version I installed is the freeswitch-1.0.3.tar.gz<http://files.freeswitch.org/freeswitch-1.0.3.tar.gz> located at files.freeswitch.org. I didn't make any major changes to the configuration; I enabled Python and set-up the SIP profile, directory and dialplan. No other changes. Any other help would be appreciated, since I really don't know where to look. Thanks, Adam >Date: Wed, 8 Apr 2009 06:36:27 +0430 >From: Traun Leyden <tleyden(a)branchcut.com> >Subject: Re: [Freeswitch-users] Two or more simultaneous calls not > working >To: freeswitch-users(a)lists.freeswitch.org >Message-ID: > <f9f80d540904071906j74c53d54q2c4acab117be10f4(a)mail.gmail.com> >Content-Type: text/plain; charset="iso-8859-1" > >> >> >> Hello, >> >> I wrote an application using FreeSWITCH version 1.0.3, with mod_python and >> a >> 64 bit box on Red Hat. >> The app works fine when one person dials in, but when a second person dials >> in, the first call stops and waits until the second call is finished. It's >> really strange - if the first call is right in the middle of playing a >> prompt, it will just stop, and there will be dead air. As soon as the 2nd >> call hangs up, the prompt for the first call starts playing right where it >> left off. >> >> I previously had FreeSWITCH installed on a 32 bit CentOS box, and this was >> not happening. >> >> Does anybody have any idea what the cause of this could be? >> >> Thanks! > > >I'm running mod_python fine with 64-bit (Ubuntu 8) and fs svn 12793, and >have not seen that problem. Which version of python? Did you build it >yourself or was it from a package? If its from a package, please provide >the version of Red Hat you are using. > >One possible cause is that python was not compiled with multi-threading >support. I don't know how to check that however .. googled around and >didn't find anything. If you fire up >your python interpreter and type "import threading" do you get an error?

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Re: [Freeswitch-users] FreeSWITCH running on OpenWrt
by Fred 08 Apr '09

08 Apr '09

Carlos Talbot > Is there an interest in running FreeSWITCH on OpenWRT? I recently managed to compile and run a version for a MIPs based router, the Planex MZK-W04NU. Great news :-) I'm interested in running FS on any of this type of small hardware. Ideally, it should have a USB port so I can connect Sangoma's U100 connector to handle one or two POTS lines. Would the FS port you did handle this USB VoIP gateway? Thanks.

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how to set CF_VERBOSE_EVENTS
by Szymon Olko 08 Apr '09

08 Apr '09

I track channels via mod_socket_event, I saw in source there is such flag CF_VERBOSE_EVENTS to make all channel related events contain extended data. Is it possible to set it via 'originate' or 'conference xxx dial' commands. This would ease my system. In scenario when I call user which is not registered I get only CHANNEL_DESTROY event but I cannot connect it to my command. I can monitor BACKGROUND_JOB events but in some cases (like answering) it can take time to come. Basically now I need to wait for first event of those two types (CHANNEL_ORIGINATE , BACKGROUND_JOB), if I have extended data in all events I can ignore BACKGROUND_JOB. Regards Szymon

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