December 2008 - FreeSWITCH-users - FreeSWITCH Mailing Lists

FreeSWITCH port audio module
by Jason White 22 Dec '08

22 Dec '08

One of the valuable features of FreeSWITCH is that it can be used as a soft phone, as described on the wiki. In testing this, I discovered the following issues, any comments on which would be welcome. 1. FreeSWITCH>pa rescan -ERR no reply FreeSWITCH> After plugging in a USB head set and running this command, the new device wasn't enumerated by pa devlist. Restarting FreeSWITCH of course solved it, however. 2. More seriously, when using port audio (with a head set as the audio device, in case that's significant), I'm hearing digital distortion (clipped samples?) when the other party speaks slightly more loudly than usual. When calling via a Snom phone (rather than PortAudio) I have only experienced this distortion when using FreeSWITCH under G.722, but I'll have to do more testing to identify the exact combinations that produce it and those which don't. I haven't heard it with an 8khz call from a SIP phone, so it does appear to be a FreeSWITCH issue to some extent. I haven't been able to eliminate it by adjusting the Alsa settings of my audio device. 3. I've encountered errors while trying to access an Intel HDA sound card with FreeSWITCH, whereby PortAudio fails to open the audio device. Setting the sample rate in portaudio.conf.xml to 48 khz may have contributed to the solution, but there have been other changes to my system as well (including a FreeSWITCh upgrade to revision 10889. Some sound cards only suport 48 khz, apparently, so if others have problems, I would suggest adjusting the sample rate in the configuration as I did.

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2 Sipgate.de accounts - second one does not register
by can_man＠gmx.de 22 Dec '08

22 Dec '08

Hello, I have two sipgate.de accounts and would like to attache them both at the same time to freeswitch. My problem is that I am not sure how to dial out using the second account. For the first one I can do: <action application="bridge" data="sofia/gateway/sipgate.de/$1(a)sipgate.de"/> and everything works, but for the second one I always get: [INVALID_NUMBER_FORMAT] Whatever combination I try. E.g. "sofia/gateway/sipgate2/$1(a)sipgate.de" or "sofia/gateway/sipgate.de/$1@sipgate2" My sip profile looks like this: <include> <gateway name="sipgate.de"> <param name="realm" value="sipgate.de"/> <param name="proxy" value="sipgate.de"/> <param name="username" value="******"/> <param name="password" value="******"/> <param name="extension" value="10000"/> </gateway> </include> <include> <gateway name="sipgate2"> <param name="realm" value="sipgate.de"/> <param name="proxy" value="sipgate.de"/> <param name="username" value="******"/> <param name="password" value="******"/> <param name="extension" value="10000"/> </gateway> </include> I tried to change gateway name with realm, but no luck. Thank you very much for your help. Phil -- Psssst! Schon vom neuen GMX MultiMessenger gehört? Der kann`s mit allen: http://www.gmx.net/de/go/multimessenger

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Problem compiling socket2me
by Cavalera Claudio Luigi 22 Dec '08

22 Dec '08

Hello guys, I'm playing with fs fax capabilities following these guidelines: http://wiki.freeswitch.org/wiki/Examples_faxlib.jm I've compiled mod_fax with make mod_fax-install and that should have taken care also of spandsp. When I issue make in scripts/socket2me I get this error: socket2me.c:315: error: 'fax_state_t' has no member named 't30_state' Maybe something has changed in the api? Could you please help me track down the problem? Thanks, Claudio Internet Email Confidentiality Footer ----------------------------------------------------------------------------------------------------- La presente comunicazione, con le informazioni in essa contenute e ogni documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, comunicazione, divulgazione o simili basate sul contenuto di tali informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se avete ricevuto questa comunicazione per errore, vi preghiamo di darne immediata notizia al mittente e di distruggere il messaggio originale e ogni file allegato senza farne copia alcuna o riprodurne in alcun modo il contenuto. This e-mail and its attachments are intended for the addressee(s) only and are confidential and/or may contain legally privileged information. If you have received this message by mistake or are not one of the addressees above, you may take no action based on it, and you may not copy or show it to anyone; please reply to this e-mail and point out the error which has occurred. -----------------------------------------------------------------------------------------------------

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Setting codec/dtmf mode
by David Aldworth 22 Dec '08

22 Dec '08

I'm looking for the most effective way to make sure I'm always forcing inband dtmf and PCMU on the PSTN <-> FS side of inbound and outbound calls. FS is always in the middle of the media. The FS <-> SIP UA (customer) side will be rfc2833 and whatever the negotiated codec for that particular UA happens to be. I know I can set <param name="codec- prefs" value="PCMU"/> and <param name="inbound-codec-negotiation" value="greedy"/> in the internal sip profile but won't the external sip profile settings override this when UA dial out? (they hit the external profile first in this case) I'm basically fishing for suggestions on the best way to use start/ stop_dtmf for the inband detection and start/stop_dtmf_generate for sending the dtmf. In asterisk this would have been accomplished by setting up separate stanza's in sip.conf and setting the dtmfmode= and allow= line per the respective legs of the calls. So, calls coming to/from the PSTN would have dtmfmode=inband and allow=ulaw, meanwhile UA's connecting to asterisk would have dtmfmode=rfc2833 and allow=ulaw, gsm, etc. Why on earth would I be doing this? Well, in the interest of keeping the explanation short, we are limited to the common denominator of all our upstream PSTN carriers and they (or their equipment rather) always support this setup. Thanks for any advice. David

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"choppy" voice
by Jonas Gauffin 21 Dec '08

21 Dec '08

Hello I got problems with choppy voice. I just happens1 time of 10 or something like that. Incorrect call: http://pastebin.freeswitch.org/6479 working call: http://pastebin.freeswitch.org/6481 Any idea what is wrong? Thanks, Jonas

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Predictive Dialing
by Gopalakrishnan A.N 21 Dec '08

21 Dec '08

Hi, I would like to have predictive dialing. In asterisk we used manager api and for outbound we use originate. The originate command will dial a number where asterisk answer the call and then we predict the answering machine with the silence file. Inspite of that human voice is detected and transferred to agents. Like this how can we go ahead with Freeswitch? Any help would be appreciated. -- Thank you with regards, Gopal,

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sessions not ending up
by Gopalakrishnan A.N 21 Dec '08

21 Dec '08

Hi, I have configured the freeswitch, we are dialing through event socket, if i dial a call per day say around 200 to 300 calls, at the end of the day the sessions are not ending up in the freeswitch, i can able to see in the console till all the calls were hanged up, I am using .NET crm. Any suggestions would help us. -- Thank you with regards, Gopal,

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error creating IPv6 profile when corresponding IPv4 port in use
by Jason White 21 Dec '08

21 Dec '08

Here's the scenario (FreeSWITCH revision 10725). I have Asterisk listening on port 5060 under IPv4. This version of Asterisk does not support IPv6, and netstat -6 -a | grep sip suggests that, as expected, it isn't listening on port 5060 under IPv6. If I set up FreeSWITCH profiles to listen on port 5070, for example, under both IPv4 and IPv6 then it works fine. However, if I create an IPv4 profile to listen on port 5070, and an IPv6 profile to listen on port 5060, Sofia fails with a "port already in use" error, even though nothing should be using port 5060 under IPv6. >From the logs: tport_create(): 0x1b4d0e0 tport_bind_server(0x1b4d0e0) to */[2001:44b8:61::3b]:5060/sip tport_bind_server(0x1b4d0e0): calling tport_listen for udp tport_alloc_primary(0x1b4d0e0): new primary tport 0x1b4f010 tport_listen(0x1b4d0e0): bind(pf=10 udp/[2001:44b8:61::3b]:5060): Address already in use tport_destroy(0x1b4d0e0) nta: bind([2001:44b8:61::3b]:5060;transport=*): Address already in use nua: initializing SIP stack failed Obviously, I can easily use a different port, and hence this issue doesn't affect me. However, if it's a bug, I think it should still go into the bug list. I realize the problem might not be in Sofia or FreeSWITCH. Is Asterisk really taking over the IPv6 port as well? Environment: Debian Sid, kernel 2.6.26, x86_64 architecture. Note: once I learn enough about FreeSWITCH to configure it properly and set it up to meet my personal telephony needs, the Asterisk installation will be, shall we say, redundant.

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SQL Error
by fidibus83 19 Dec '08

19 Dec '08

Hello, Im a newbie in FS. I get an error from the freeswitch cli: 2008-12-18 10:09:14 [ERR] switch_core_db.c:100 switch_core_db_exec() SQL ERR [database disk image is malformed] I dont know what to do to remove this error! Can you help me? Regards

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[Java] Catching dingaling messages
by kriko 19 Dec '08

19 Dec '08

Hello! I was wondering if it would be possible to catch messages from dingaling. I saw it can print out messages into console when a user types in a message, but it doesn't understand it. I would like to catch that and do something (like initiate a call). I know you have to call you program via a dialplan, so I don't know how would it be really possible to invoke your java app when other events occur. Is it possible? Cheers, -- kriko

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