April 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Polycom register problem in private address
by zhaoxxqq 09 Apr '09

09 Apr '09

hi, I use FS server at public Address. I use polycom's IP550 at private address(192.168.0.120), Now there is a problem that the IP550 can not register to FS. But when I use account to eyebeam, the registering is OK. the attachment is my IP 550's config file, I think it must be NAT problem. Can anyone can help me solve it? 2009-04-09 zhaoxxqq

3 2

Status of Sangoma support in Windows?
by Peter Olsson 09 Apr '09

09 Apr '09

>From what I've found in the docs and lists, the support for Sangoma (PRI) cards is still not avaiable in the Windows port. Is this planned to be implemented in the future, or will it never be included in Win32? Just a curious thought, since I might need to use some PRI stuff in the future... Regards, Peter Olsson

2 2

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

09 Apr '09

Traun, thanks again for your help. I followed your advice and I made some progress! I tested with the latest trunk version and also with 1.0.2, and both exhibited the same behavior. I then tried writing a test script in Lua, and it worked fine. So this meant the problem was in the Python module (I was sure it was some FS config issue). So I started playing with a small test Python script, and I narrowed the problem down to when I'm using the "read" function. Here is my test script: from freeswitch import * def handler(session, args): #answer the call session.answer(); #play a file session.streamFile("long_prompt.mp3") # Test 1 - FAILED! digits = session.read(5, 10, "long_prompt.mp3", 3000, "#") # Test 2 - WORKED OK! #session.getDigits(1,"#",7000) #session.streamFile("long_prompt.mp3") # TEST 3 - WORKED OK! #digits = session.playAndGetDigits(5, 10, 1, 60, "#","long_prompt.mp3", "", "") #hangup session.hangup(); When I uncomment the code under test 1 and I make two simultaneous calls, the initial prompt plays for both calls just fine. But then the second prompt only plays on one of the channels and the other one just has dead air. When the first channel finishes playing the prompt, then the second channel starts playing it. Then I re-comment test 1 and uncomment either test 2 or test 3, Both prompts play just fine for both channels. So I think there may be a bug in the read() function somewhere. I took a look at it, but it's way over my head. Thanks again, Adam >Message: 8 >Date: Thu, 9 Apr 2009 02:36:08 +0430 >From: Traun Leyden <tleyden(a)branchcut.com> >Subject: Re: [Freeswitch-users] Two or more simultaneous calls not >To: freeswitch-users(a)lists.freeswitch.org >Message-ID: > <f9f80d540904081506p3f0647fei4cf843d0b1904bc0(a)mail.gmail.com> >Content-Type: text/plain; charset="iso-8859-1" >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

2 1

How to design my project ?
by Michele M 09 Apr '09

09 Apr '09

Hi there, I'm quite a newbie about freeswitch. I have an application (IVR) that needs to have endpoints SIP to register,answer the calls and transfer them to the right phones.(I( have my own SIP server).Moreover it needs also a ASR/TTS API' set to communicate with my ASR/TTS engine ( just for example let's assume it is Cepstral). I'd wouldn't want to have freeswitch running and communicate with it to accomplish that but just to use the libfreeswitch library embedded. As I don't know that much about freeswitch can it be done? or just I need to have freeswitch running as a must? Can somebody point me to the right place where to find example of using library embedded (best examples for what I'm trying to do) as I have not found that many? Thanks in advance Miki

3 3

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.

2 1

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

5 21

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

1 0

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

1 0

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

2 2

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

1 0