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mod_java.so load issue
by jonathan augenstine 22 Dec '08

22 Dec '08

I am installing Freeswitch on Fedora. I was building/installing the mod_java.so module and I encountered the following load issue: 2008-12-20 10:34:58 [CRIT] switch_loadable_module.c:839 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_java.so **/usr/local/freeswitch/mod/mod_java.so: invalid ELF header** Is this a build issue? I am assuming maybe there is a g++ option that is set incorrectly but my searches on Google and looking at gcc docs have not provided a solution. Jonathan

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If Bypass Media works why won't Proxy Media work?
by mszlazak＠aol.com 22 Dec '08

22 Dec '08

I find it strange that I can have to endpoints get audio went using bypass media mode but the audio fails to go across endpoints if I use proxy media mode. I'm trying to pass audio "internally" on the same machine between endpoints and have be advised that a reason the audio may fail to be passed is because there is some RTP timing and IP address/port issues. However, FS has no problem "connecting" ports if i change the mode to bypass media. This gives me the impression that something is wrong with FS proxy media mode. Any comments??

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Re: [Freeswitch-users] Phone lines ring busy after awhile
by Adam Wilt 22 Dec '08

22 Dec '08

Sorry, I'm using freeswitch-1.0.latest.tar.gz on Fedora. > > Message: 5 > Date: Mon, 22 Dec 2008 12:16:08 -0800 > From: "Michael Collins" <msc(a)freeswitch.org> > Subject: Re: [Freeswitch-users] Phone lines ring busy after awhile > To: freeswitch-users(a)lists.freeswitch.org > Message-ID: > <87f2f3b90812221216i42cbaeedx52cee1863e68667a(a)mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > what operating system and which revision of FreeSWITCH? > -MC > > On Mon, Dec 22, 2008 at 9:34 AM, Adam Wilt <wiltingtree(a)gmail.com> wrote: > > > Hello. I have an installation of FreeSwitch runnnig, and I wrote a > Python > > script to answer an inbound call and play an IVR. It works fine for > awhile, > > but eventually it stops answering the phone line and the phone rings > busy. > > When I look in the FreeSwitch logs I don't see anything unusual. But I > don't > > think it's a problem with the VOIP phone service because when I restart > > FreeSwitch the problem goes away. Any ideas? > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users(a)lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > >

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group call with BLF and pickup
by Seweryn Niemiec 22 Dec '08

22 Dec '08

Hi, I'm currently evaluating FreeSWITCH to see if I can migrate to it from Asterisk (which gives me now all functions I need, but has some problems with predictability). I need only one atypical functionality: 1. we have one main extension, which clients call. lets name it 5555 2. when someone calls 5555, then immediately rings only one phone (1001) and 15 seconds later 3 more phones (1002-1004). phone 1001 should be dialled once per call to 5555 3. all phones (1000-1010) have BLF monitoring 5555, so anyone can pick it up when someone calls it ad 3. it is acceptable that all phones monitor 1001 instead of 5555 to pick up connection to 5555 After reading and googling about possible implementation in FS for few hours I couldn't find anything useful. Could you tell if it is possible to implement in FS without artificial limbs (i dunno if it's best English word for what I mean :) like I had to do in Asterisk? greetings, Seweryn

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Phone lines ring busy after awhile
by Adam Wilt 22 Dec '08

22 Dec '08

Hello. I have an installation of FreeSwitch runnnig, and I wrote a Python script to answer an inbound call and play an IVR. It works fine for awhile, but eventually it stops answering the phone line and the phone rings busy. When I look in the FreeSwitch logs I don't see anything unusual. But I don't think it's a problem with the VOIP phone service because when I restart FreeSwitch the problem goes away. Any ideas?

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Mod Fax: Error, problems and questions...
by Dennis 22 Dec '08

22 Dec '08

hi, after we managed to setup fs with mod_fax and our socket outbound script, we have some questioons about an error and problems, when sending a fax: 1.) there is one error, we get always - no matter, if the fax was sent successfully or not. in the pastebin under http://pastebin.freeswitch.org/6338 you can see the error in the last line. this is the full log of a fax in fs console loglevel debug. 2.) fax works quite good. we couls send long faxes over a normal fax machine without any problem. but for fast testing we are using a softfax (fritz fax). here we have some more problems. mostly a fax with one page will pass through, but more pages will mostly fail. because we are new to the fax thing, we do not really know, what the messages tell us about failed faxes. here are the top 3 messages (unordered and always one at a time - nerver at once): variable_fax_result_text => Received a DCN while waiting for a DIS fax_result_text => The HDLC carrier did not stop in a timely manner fax_result_text => Unexpected message received could someone please tell us, where the problem might be? thanks dennis

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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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