February 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Errors compiling trunk on a fresh system
by Cavalera Claudio Luigi 05 Feb '09

05 Feb '09

Hello, I'm trying to compile a brand new fs on a clean system. Revision: 11630 After the usual ./bootstrap.sh ./configure --enable-core-odbc-support I was getting this at make http://pastebin.freeswitch.org/7011 so i cd into utils under libs/sofia-sip/utils issued a doxygen -u but still getting this http://pastebin.freeswitch.org/7012 any hint on what could be? Best Regards, 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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FS in ISP Mode
by Nik Middleton 04 Feb '09

04 Feb '09

Do apologise about the hijacking, Question: My ISP sends inbound calls via 5060, so it seems I need to renumber the ports, but that leaves my SIP end points who authenticate also needing 5060, can they be combined? Regards, -----Original Message----- From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Brian West Sent: 04 February 2009 22:42 To: freeswitch-users(a)lists.freeswitch.org Subject: Re: [Freeswitch-users] FS in ISP Mode Don't let the names of the profiles confuse you... they are just names. internal is on port 5060 has auth on... external is on 5080 and doesn't' have auth on and lets all calls into the public context without auth. Also when you post to the mailing list do not hijack a thread. Hijacking happens when you take an existing message, click reply, change the subject and start a new body. That will hijack the thread. In your case the thread was "Re: [Freeswitch-users] Q931 decoding Update". So next time please click new message and input the address freeswitch-users(a)lists.freeswitch.org . Thanks, Brian West FreeSWITCH.org On Feb 4, 2009, at 4:30 PM, Nik Middleton wrote: > Hi Guys, > > Excuse my ignorance, but I'm just starting with FS. > > I've loaded FS onto one of our servers in a datacenter. I'm > registering with our PSTN breakout provider just fine, but I'm a > little confused about internal/external. > > Given that we have no internal clients, as they're all external, > should I switch the ports over so that 5060 is the external port? > > I think I know the answer, but would like confirmation > > Regards > > > _______________________________________________ > 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 _______________________________________________ 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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Gateway settings
by Nik Middleton 04 Feb '09

04 Feb '09

Hi Guys, Need a little help here; I connect to my PSTN provider via the LAN, Question: As the provider authenticates on IP, how do I not send a password? In the .xml file if I remove the password entry it complains Secondly, the contact should be my local address, not the public one. What do I need to do here? Finally what does FS do to determine if the status is up, is there an asterisk equivalent of qualify going on here? Regards Name My Provider Scheme Digest Realm 172.16.1.2 Username myname Password yes >From <sip: myname @172.16.1.2;transport=udp> Contact <sip: myname @ExternalIP:5080;transport=udp> To sip: myname @172.168.1.2 Proxy myname:172.168.1.2 Context default Expires 600 Freq 600 Ping 0 PingFreq 0 State NOREG Status UP

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mod_limit
by Chav Paskov 04 Feb '09

04 Feb '09

Hi , is it possible to use mod_limit in case if the end point is not registered / gateway for example/. Regards Chav

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Recording background music and voice is out of sync
by Daniel Liang 04 Feb '09

04 Feb '09

What I did was the following: First, I sent the playback command: call-command: execute execute-app-name: playback execute-app-arg: <filename> Then I send uuid_record (Sorry, it was not Record command): api uuid_record <uuid> start <filename> 120 I also tried replacing the playback command with: api uuid_displace <uuid> start <filename> 0 mux But the end results are the same. The recorded user's voice is about 0.5 second behind the expected result. Thanks, Daniel -----Original Message----- From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Brian West Sent: February 3, 2009 6:36 PM To: freeswitch-users(a)lists.freeswitch.org Subject: Re: [Freeswitch-users] Recording background music and voice is outof sync Can you show us an example of how you're doing this? Playback and Record aren't async so you'll need to show us how you're doing this. Also don't hijack threads you hit replay on the one "Re: [Freeswitch- users] FreeSwitch setup as a "Dumb" SBC" as the subject, deleted the subject and started a new body. That hijacks the thread and that can cause your problem to go ignored in some cases if people aren't interested in the thread topic depending on how their reader threads the emails. Please click new message and type freeswitch- users(a)lists.freeswitch.org in and then input your subject and body to start a new thread. Thanks, Brian West FreeSWITCH.org On Feb 3, 2009, at 8:21 PM, Daniel Liang wrote: > Hi, > > I was trying to record a background music with a user's voice at the > same time. I did a playback and started recording. But the recorded > user's voice and the background music is about 0.5 second out of sync. > I also tried to use uuid_displace instead of playback, but I got the > same result. _______________________________________________ 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 <http://www.freeswitch.org/>

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SIP Authentication
by Ali Al-Rubaie 04 Feb '09

04 Feb '09

Hi, I have a problem in SIP registration (authentication) with FreeSWITCH server. The SIP messages are: recv 292 bytes from udp/[209.82.10.250]:3458 at 16:35:24.758862: ------------------------------------------------------------------------ REGISTER sip:209.82.10.235 SIP/2.0 Via: SIP/2.0/UDP 209.82.10.250:1059 From: sip:1001@209.82.10.235 To: sip:1001@209.82.10.235 Contact: sip:1001@209.82.10.250:1059 Call-ID: fc383368-e3e0-4c32-b87b-c3127d0cc2d8(a)192.168.10.23 CSeq: 597775306 REGISTER Content-Length: 0 Expires: 3600 ------------------------------------------------------------------------ send 582 bytes to udp/[209.82.10.250]:1059 at 16:35:24.763948: ------------------------------------------------------------------------ SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 209.82.10.250:1059 From: sip:1001@209.82.10.235 To: <sip:1001@209.82.10.235>;tag=7yam2F01ZH3vH Call-ID: fc383368-e3e0-4c32-b87b-c3127d0cc2d8(a)192.168.10.23 CSeq: 597775306 REGISTER User-Agent: FreeSWITCH-mod_sofia/1.0.2-hacked Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces WWW-Authenticate: Digest realm="209.82.10.235", nonce="40b63193-85c2-4ed9-874e-c03f81be313d", algorithm=MD5, qop="auth" Content-Length: 0 ------------------------------------------------------------------------ recv 466 bytes from udp/[209.82.10.250]:3458 at 16:35:24.772834: ------------------------------------------------------------------------ REGISTER sip:209.82.10.235 SIP/2.0 Via: SIP/2.0/UDP 209.82.10.250:1059 From: sip:1001@209.82.10.235 To: sip:1001@209.82.10.235 Contact: sip:1001@209.82.10.250:1059 Call-ID: fc383368-e3e0-4c32-b87b-c3127d0cc2d8(a)192.168.10.23 CSeq: 597775307 REGISTER Content-Length: 0 Expires: 3600 Authorization: Digest username="1001",realm="209.82.10.235",nonce="40b63193-85c2-4ed9-874e-c03f81be313d",response="eebe0ea43319e82cc5f6dba5877de706",uri="sip:209.82.10.235" ------------------------------------------------------------------------ send 458 bytes to udp/[209.82.10.250]:1059 at 16:35:24.774354: ------------------------------------------------------------------------ SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 209.82.10.250:1059 From: sip:1001@209.82.10.235 To: <sip:1001@209.82.10.235>;tag=873c4aH5vtSFD Call-ID: fc383368-e3e0-4c32-b87b-c3127d0cc2d8(a)192.168.10.23 CSeq: 597775307 REGISTER User-Agent: FreeSWITCH-mod_sofia/1.0.2-hacked Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Content-Length: 0 ------------------------------------------------------------------------ What I have noted is that the client does not send the values for "cnonce" and "nc" in the response. I'm not sure if this is the reason, however how this problem can be solved? Thanks, Ali

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gateway
by Jonas Gauffin 04 Feb '09

04 Feb '09

Hello I'm trying to make outbound calls through my gateway provider. My calls got rejected and I asked them why. Apparently I need to use 5060 as source port, since they validate both my IP and the port that the messages come from. Is this possible with freeswitch? If so, what config settings should I set? Regards, Jonas

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mod_voicemail: Limiting the number how often a menu is repeated
by Helmut Kuper 03 Feb '09

03 Feb '09

Hello, today I searched for a way to limit the number of menu repeatings in mod_voicemail to let's say 3 times and when it reached the limit voicemail should abort. But I couldn't find a hint. Any ideas? regards helmut

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mod_sofia "ReINVITE"
by Adam Long 03 Feb '09

03 Feb '09

In every one of my SIP sessions FreeSwitch appears to be inserting .. Contact: sip:mod_sofia@XXX.XXX.XXX.XXX:5060 Is this normal? I only ask as it is causing some of my end points to RE-INVITE back to this after the initial ( INVITE <---- > 100 Trying < --- > 200 OK ) call setup. If this is not normal or by design I can provide more details on configuration and dialplan. Thanks! Regards, -Adam

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Application language to support C or C++?
by lee jason 03 Feb '09

03 Feb '09

Hi All, I saw the applications using FreeSwitch library can be written in JavaScript, Perl, Python and Lua but I need to use Linux C or C++ for applications, Is FreeSwitch can supported it? Where can I get the sample codes? My Linux platform is base on Fedora. Thanks a lot. Jason

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