October 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Call hangup not working in perl session
by Samir Doshi 18 Oct '13

18 Oct '13

Hi Guys, I am using perl session to handle call flow. I need to hangup call at exact 1 minute. For that i have set below line, but somehow its not working. Call keeps running. $session->execute( "set", "execute_on_answer=sched_hangup +60" ); Can anybody please point out the issue or something I missing. Thanks in advance. ----- Thanks, Samir -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Call-hangup-not-working-in-pe… Sent from the freeswitch-users mailing list archive at Nabble.com.

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limit incoming concurrent fax/calls
by Matteo Cazzador 18 Oct '13

18 Oct '13

Hi, i've configured a fax server (Centos 6.2) using ICT Fax with Plivo Framework.I'm a novice about freeswitch. All goes well, but i've a request. I need to limit concurrent incoming calls (fax receive) max 50 contemporary incoming fax. I can use "limit" parameter (from freeswitch doumentation)? Is it correct to set it in /usr/local/freeswiitch/conf/dialplan/default.xml or public.xml? I want that limit globally, max 50 concurrent calls incoming non for single DID. Thank's a lot -- Rispetta l'ambiente: se non ti è necessario, non stampare questa mail. ****************************************** Ing. Matteo Cazzador Email: mcazzador(a)gmail.com ******************************************

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INCOMPATIBLE_DESTINATION bridging GSM to mod_skypopen
by Danny Iland 18 Oct '13

18 Oct '13

I'm using Freeswitch as a PBX for OpenBTS, to route calls from mobile phones to VoIP providers. When bridging a call from a cellular handset to a mod_skypopen interface, I get the following error as soon as the Skype call is answered. The handset then hangs up. It appears the problem lies in the 'audio codec compare' phase. 2013-10-16 17:54:21.932473 [DEBUG] switch_channel.c:3548 Send signal sofia/internal/IMSI[me]@[ip] [BREAK] 2013-10-16 17:54:21.932473 [NOTICE] mod_skypopen.c:2471 Channel [skypopen/skype101/echo123] has been answered 2013-10-16 17:54:21.932473 [DEBUG] switch_channel.c:3594 (skypopen/skype101/echo123) Callstate Change RINGING -> ACTIVE 2013-10-16 17:54:21.932473 [DEBUG] mod_skypopen.c:1232 [543dc3c|22dd4bf] [DEBUG_SKYPE 1232 ][skype101 ][UP,INPROGRS] MSG_ID=41 2013-10-16 17:54:21.932473 [DEBUG] mod_skypopen.c:2479 [543dc3c|22dd4bf] [DEBUG_SKYPE 2479 ][skype101 ][UP,INPROGRS] outbound_channel_answered! 2013-10-16 17:54:21.932473 [DEBUG] skypopen_protocol.c:1172 [543dc3c|22dd4bf] [DEBUG_SKYPE 1172 ][skype101 ][UP,INPROGRS] ACCEPTED here you send me 32776 2013-10-16 17:54:21.932473 [DEBUG] skypopen_protocol.c:1177 [543dc3c|22dd4bf] [DEBUG_SKYPE 1177 ][skype101 ][UP,INPROGRS] 4 SO_RCVBUF is 5120, size is 4 2013-10-16 17:54:21.932473 [DEBUG] skypopen_protocol.c:1181 [543dc3c|22dd4bf] [DEBUG_SKYPE 1181 ][skype101 ][UP,INPROGRS] 4 SO_SNDBUF is 5120, size is 4 2013-10-16 17:54:21.952511 [DEBUG] skypopen_protocol.c:207 [543dc3c|22dd4bf] [DEBUG_SKYPE 207 ][skype101 ][UP,INPROGRS] READING: |||ALTER CALL 32 SET_INPUT PORT="32776"||| 2013-10-16 17:54:21.952511 [DEBUG] sofia_glue.c:5226 Audio Codec Compare [GSM:3:8000:20:13200]/[PCMU:0:8000:20:64000] 2013-10-16 17:54:21.952511 [DEBUG] sofia_glue.c:5226 Audio Codec Compare [GSM:3:8000:20:13200]/[PCMA:8:8000:20:64000] 2013-10-16 17:54:21.952511 [DEBUG] sofia_glue.c:5370 No 2833 in SDP. Disable 2833 dtmf and switch to INFO 2013-10-16 17:54:21.952511 [DEBUG] switch_core_session.c:858 Send signal sofia/internal/IMSI[me]@[ip] [BREAK] 2013-10-16 17:54:21.952511 [NOTICE] switch_channel.c:3633 Hangup sofia/internal/IMSI[me]@[ip] [CS_EXECUTE] [INCOMPATIBLE_DESTINATION] Any advice on forcing a transcode or otherwise having the Audio Codec Compare result in a successful bridge is appreciated. Thanks, Danny Iland

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Re: [Freeswitch-users] Freesiwtch 1.2.14 Windows
by Jeff Leung 18 Oct '13

18 Oct '13

Did openssl download correctly? if not, nuke the tree ans start again. Denis Papes <shishko69(a)gmail.com> wrote: Download it again http://files.freeswitch.org/freeswitch-1.2.14.tar.bz2 and retry. On Thu, Oct 17, 2013 at 10:17 PM, Andre <andretodd(a)verizon.net> wrote: > Mine is not there. Should I just copy it there and recompile?**** > > ** ** > > ** ** > > *From:* freeswitch-users-bounces(a)lists.freeswitch.org [mailto: > freeswitch-users-bounces(a)lists.freeswitch.org] *On Behalf Of *Denis Papes > *Sent:* Thursday, October 17, 2013 4:08 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Freesiwtch 1.2.14 Windows**** > > ** ** > > I successfully compiled 64-bit version using Visual Studio 2010. File > nametab.h should be in same directory as xmltok.c, > <srcdir>\libs\xmlrpc-c\lib\expat\xmltok.**** > > ** ** > > On Thu, Oct 17, 2013 at 9:15 PM, Andre <andretodd(a)verizon.net> wrote:**** > > Hi, when I compile 64 bit I get this error**** > > **** > > Error 1 error C1083: Cannot open include file: 'nametab.h': No such file > or directory > \freeswitch-1.2.14\freeswitch-1.2.14\libs\xmlrpc-c\lib\expat\xmltok\xmltok.c > 10 1 xmltok**** > > **** > > Any idea what this means or how to fix it? Or does it even matter?**** > > Andre**** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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**** > > ** ** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > >

1 0

Survey - FreeSWITCH billing solutions
by Victor Chukalovskiy 17 Oct '13

17 Oct '13

Greetings! Everyone who runs some sort of billing for freeswitch, please provide your feedback here: http://www.surveymonkey.com/s/J8SL9TS Just 4 question. Less then a minute to complete! Survey runs for 2-3 days. I'll post final results in this thread for your benefit. Cheers, -Victor

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park_after_bridge
by Andre 17 Oct '13

17 Oct '13

Why does park after bridge only work in media bypass mode? s.SetVariable("park_after_bridge", "true");

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Re: [Freeswitch-users] mod_spandsp compile error
by Sadjad Seyed-Ahmadian 17 Oct '13

17 Oct '13

it is CentOS release 5.9 (Final) On 10/17/2013 07:13 AM, Sadjad Seyed-Ahmadian wrote: >Hi, >>I am try to install a FreeSWITCH stable version 1.2.13 on a fresh >CentOS. I installed all prereqired packages as they are in wiki, >configure goes well but while I try to do make I got this error on >mod_spandsp. >>>making all mod_spandsp >Creating mod_spandsp_la-mod_spandsp.lo >mkdir .libs >Compiling mod_spandsp.c ... >Creating mod_spandsp_la-udptl.lo >Compiling udptl.c ... >cc1: warnings being treated as errors >udptl.c: In function 'udptl_rx_packet': >udptl.c:174: warning: 'data' may be used uninitialized in this function >udptl.c:173: warning: 'msg' may be used uninitialized in this function >make[3]: *** [mod_spandsp_la-udptl.lo] Error 1 >make[2]: *** [mod_spandsp-all] Error 1 >make[1]: *** [mod_spandsp] Error 2 >make: *** [mod_spandsp] Error 2 >>>Would somebody please help me with that? >>Best Regards >Sadjad >Centos 6.4 doesn't do that. Which version of Centos are you using? I might be missing something, but those two warnings seem bogus. Steve

1 0

Voicemail beep
by Ali Pey 17 Oct '13

17 Oct '13

Hello, Is there a way to change the voicemail beep sound? The existing beep is too long and too loud. Thanks, Ali

2 6

Can a Jira summary/title be changed?
by Mario G 17 Oct '13

17 Oct '13

Anyone know if changing the summary title of your own Jira issue is possible? Issue titles may need to be changed because the real/root problem was not known at the time it was created and a correct summary may be more useful for others with the same problem. For instance, I have: FS-5375 FreeSwitch crash during paging/conference <- current FreeSwitch crash using G722 on OSX <- should be FS-5223 improve message: Error Creating SIP UA for profile xxx <- old internal profile TCP port stays in use after FS shutdown on OSX <- should be Thanks, Mario G

1 0

Is FIFO in memeroy or in DB?
by absdou24＠gmail.com 17 Oct '13

17 Oct '13

I hava two freeswitches ,these freeswitches share the same mysql, I make extension1 which stands Member join the fifo named 111 of FS1 to waiting for a Caller. Then a comming caller from FS2 hits the diaplan that matches fifo 111, I hope extension1 will answer the comming caller,However not. why?Is the data about FIFO in Memeroy,So FS2 can not get the extension1 status from the memory fron FS2. Please help me,Thanks a lot Keep in mind Quality First

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