December 2008 - FreeSWITCH-users - FreeSWITCH Mailing Lists

libfreeswitch question
by Woody Dickson 02 Dec '08

02 Dec '08

Hi, I am just having a dumb question and hoping someone can help me. I am trying to run a c program with libfreeswitch embedded so I can use some external mechanism to keep track of freeswitch, but I am having problem while compiling: [root@localhost fs]# gcc switchnode.c -I/usr/local/freeswitch/include -L/usr/local/freeswitch/lib -lfreeswitch -lpthread switchnode.c: In function 'main': switchnode.c:11: warning: passing argument 1 of 'switch_core_init_and_modload' makes integer from pointer without a cast switchnode.c:11: warning: passing argument 3 of 'switch_core_init_and_modload' from incompatible pointer type /usr/local/freeswitch/lib/libfreeswitch.so: undefined reference to `clock_gettime' /usr/local/freeswitch/lib/libfreeswitch.so: undefined reference to `uuid_generate' /usr/local/freeswitch/lib/libfreeswitch.so: undefined reference to `crypt_r' collect2: ld returned 1 exit status [root@localhost fs]# Does anyone know which library is missing? Thanks, Woody

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Re: [Freeswitch-users] How to specify Path for sound files
by Jan Kubr 02 Dec '08

02 Dec '08

Hi Faisal, the path is either an absolute path or a path relative to the directory in the sound_prefix var in vars.xml. So this <action application="playback" data="/usr/local/freeswitch/sounds/en/us/callie/misc/8000/call_secured.wav"/> works fine on my box. You sure this one doesn't work for you? Jan > I tried to play a sound file using the dialplan given on the link > http://wiki.freeswitch.org/wiki/Playing_recording_external_media#Play_wav > > <action application="playback" data="/path/to/your.wav"/> > In place of /path/to/your.wave I used > "/en/us/callie/misc/8000/call_secured.wav" > "/usr/local/freeswitch/sounds/en/us/callie/misc/8000/call_secured.wav" > "/sounds/en/us/callie/misc/8000/call_secured.wav" > But none of these is useful bcoz when i call on 2009, which is > to b dialed to play the sound, same msg is > displayed "404 NOT FOUND" > Plz help me out.??? Faisal

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Re: [Freeswitch-users] Sound file as ringback
by Jan Kubr 01 Dec '08

01 Dec '08

Yes, setting the var to the full path works. Sorry, should have taken the "full path" in the wiki more seriously. MP3s are played only once, 8kHz WAVs work perfectly. Cheers, Jan > Can you try putting the full path to the file? Also what does the > console output look like? > > /b > > On Nov 30, 2008, at 12:30 PM, Jan Kubr wrote: > >> I have try different format of files (from 8KHz mono wavs to MP3s, all >> of which play fine via playback) and some caused the bridge to be >> finished immediately (with NO_USER_RESPONSE), some make it generate >> crazy beeping, but none is played while the phone is ringing on the >> other end.

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SIP INVITE timeout
by Gabriel Kuri 01 Dec '08

01 Dec '08

I have a phone that is registered to FS but is no longer available (Internet connection down, phone turned off, etc.). The registration still exists in the sip_registrations table (not expired yet), but the phone is not reachable on the network. According to my dialplan, if the bridge to the phone fails after 20 seconds, the call should be forwarded to a different box for handling (see dialplan below). <extension name="My Extension"> <condition field="destination_number" expression="1213XXXXXXX"> <action application="info"/> <action application="set" data="originate_timeout=20"/> <action application="set" data="hangup_after_bridge=true"/> <action application="bridge" data="${sofia_contact(default/1213XXXXXXX(a)mydomain.net"/> <action application="bridge" data="sofia/default/1213XXXXXXX(a)box.mydomain.net"/> </condition> </extension> If the phone is down and not responding to the INVITEs, it appears my carrier is canceling the SIP INVITE to FreeSWITCH after about 10 seconds. My timeout is 20 seconds. Is there anyway to deal with this situation, without going back to my carrier and asking them to increase their timeout on an INVITE? Call Progress: Carrier -> FS (INVITE) FS -> Carrier (100 Trying) <10 seconds pass while FS is attempting to contact the phone> Carrier -> FS (CANCEL) FS -> Carrier (200 OK) FS -> Carrier (487 Request Terminated) Carrier -> FS (ACK) Thanks ... Gabe

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Leg A terminated by Leg B on a uuid_bridge
by Simon Tang 01 Dec '08

01 Dec '08

Hello, I'm using event socket outbound, and have an issue where, after a bridge ends and is terminated by Leg B, Leg A is also terminated. Here's the call flow: 1. Call comes in (Leg A), session created, play welcome message. 2. From this session, originate and dial out using api originate 3. After the target answers (Leg B), bridge the 2 calls using api uuid_bridge 4. Leg B hangs up. 5. Leg A will be terminated. After step 4, Leg A is terminated. I do not want Leg A to hang up. I've tried setting "hangup_after_bridge=false" prior to the call, and that doesn't work. Having said that, I tried a similar test which does not end Leg A's call after Leg B hangs up, but I can't use this solution because, functionally, does not accomplish what I want it to do (i.e., I want to perform some actions on Leg B prior to the bridge, like send some DTMF tones, playback some messages, etc). I did not need to set the "hangup_after_bridge" variable (default should be false anyway). 1. Call comes in (Leg A), session created, play welcome message. 2. From this session, do a bridge by doing an execute bridge. 3. The target answers (Leg B) 4. Leg B hangs up. 5. Leg A will still be active. Any ideas would be appreciated. Thanks! Simon

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Re: [Freeswitch-users] Freeswitch-users Digest, Vol 29, Issue 189
by Traun Leyden 01 Dec '08

01 Dec '08

> > Message: 9 > Date: Sun, 30 Nov 2008 18:42:30 -0800 (PST) > From: Marc Orenberg <marc(a)kasteris.com> > Subject: [Freeswitch-users] Problem importing modules in mod_python > To: freeswitch-users(a)lists.freeswitch.org > Message-ID: <195670.44941.qm(a)web50805.mail.re2.yahoo.com> > Content-Type: text/plain; charset="iso-8859-1" > > In the latter versions of mod_python, I'm unable to import standard python > modules such as time and MySQLdb.? > For example, the following script works fine in version 1.0.1: > > ??? ??? import time > ??? ??? import os > ??? ??? from freeswitch import * > ??? ??? def handler(session, args): > ??? ??? ??? session.answer() > ??? ??? ??? session.execute("sleep", "2000") > ??? ??? ??? > session.streamFile("/usr/local/freeswitch/prompts/01Welcome.wav") > ??? ??? ??? return(session) > > But in freeswitch-1.0.latest.tar.gz, and svn versions 10556-10558, I get > the following error: > > ??? ??? 2008-11-30 21:13:09 [ERR] mod_python.c:129 eval_some_python() Error > reloading module > ??? ??? Traceback (most recent call last): > ??????? File "/usr/lib/python2.4/site-packages/scripts/test.py", line 1, in > ? > ??? ??? import time > ??? ??? ImportError: /usr/lib/python2.4/lib-dynload/timemodule.so: > undefined symbol: PyExc_ValueError > > Thanks for your help! > I have run into the same problem and put some documentation on the wiki: http://wiki.freeswitch.org/wiki/Mod_python#ImportError:_.2F...2Fdatetime.so… I think something changed in freeswitch in the way it is loading modules, or at least the way it is loading mod_python. This behavior appeared all of the sudden in recent freeswitch versions. HTH

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How to specify Path for sound files
by Faisal Maqsoodi 01 Dec '08

01 Dec '08

I tried to play a sound file using the dialplan given on the link http://wiki.freeswitch.org/wiki/Playing_recording_external_media#Play_wav <action application="playback" data="/path/to/your.wav"/> In place of /path/to/your.wave I used "/en/us/callie/misc/8000/call_secured.wav" "/usr/local/freeswitch/sounds/en/us/callie/misc/8000/call_secured.wav" "/sounds/en/us/callie/misc/8000/call_secured.wav" But none of these is useful bcoz when i call on 2009, which is to b dialed to play the sound, same msg is displayed "404 NOT FOUND" Plz help me out. Faisal

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Set variable for the outgoing leg
by regs＠kinetix.gr 01 Dec '08

01 Dec '08

All the variables that I set show up only in the a-leg CDR. How can I set a variable that can be used during the b-leg CDR generation?

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Multi FS behind same NAT, PRACK goes to wrong port
by x y 01 Dec '08

01 Dec '08

Hy! There are two different FS behind the same NAT, and there were Reigstration Failures about one or to times a day. The gateway status turned down, then I got 503 error codes. Then I set up the ext-ip to STUN, as the wiki requests it. Now I facing the next problem: Start the call, all goes right, INVITE goes to port 1352, then after 183 Session progress from port 1352, the PRACK package goes to 5060 instead of 1352, wich messes up the call procedure. Is there anyway to force PRACK to the port to the INVITE has been sent before? Cheers, Viktor Hirdetés (x) Váltson most olcsóbb kötelezőre a biztosítás-hu-val. www.biztositas.hu - a kötelező biztosítások kiindulópontja!

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Failed to create a udp socket on port 5060
by Faisal Maqsoodi 01 Dec '08

01 Dec '08

I m trying to play some basics on FreeSwitch plus twinkle softphone. Following error msg is displayed on the softphone: Failed to create a udp socket on port 5060. Address already in use & following on freeswitch window: X Error: BadWindow (invalid Window parameter) 3 Major opcode: 20 Minor opcode: 0 Resource id: 0x30001be Plz help me solving this. faisal

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