November 2014 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Javascript scripting
by Robert Boardman 18 Nov '14

18 Nov '14

Hi, I cannot seem to get jsrun working on the windows win32 builds, the v8 modules dont seem to be there, is there something i'm missing? Thanks Robb

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perl esl
by Pedro Santos 18 Nov '14

18 Nov '14

Hi, i can not install the perlmod esl. There is no perl/ESL.so file. i am using centos 6 and the stable version from git of freeswitch. Regards

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How to get bridge call duration with lua
by jobin dcruz 17 Nov '14

17 Nov '14

Hi, I am using freeswitch bridge,every thing working with fine. <action application="set" data="call_timeout=10"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="ringback=${us-ring}"/> <action application="bridge" data="sofia/gateway/outside/${customer_number}"/> My problem is How to get bridge call duration with lua? Cannot update CDR table with bridge call details.How to update it? -- Jobin D'cruz (LAMP Developer) Email: jobindcruz(a)gmail.com | lampdeveloper.jobindcruz(a)gmail.com Php MySql Linux Apache Ajax FFmpeg FFmpeg-php Mplayer Zend Framework Codeigniter CakePhp Wordpress Drupal Joomla Html5 Css3 jQuery YUI MooTools Smarty Firefox Extension

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Freeswitch db for voicemail
by Alex Lake 17 Nov '14

17 Nov '14

Where does one configure the database that the voicemail system uses? Doesn't seem to be using the odbc/mysql database that I specified for core, but I'm not sure if voicemail is supposed to be part of core?

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fs_cli hangs
by akhil garg 17 Nov '14

17 Nov '14

Hi, I am using 1.5.4b freeswitch release. I am running freeswitch by commands "freeswtich" or "freeswtich -nc" running fs_cli hangs. it gives no output. I could see the socket is open by running command "netstat -anlp | grep 8021" "tcp 0 0 127.0.0.1:8021 0.0.0.0:* LISTEN 3084/freeswitch" Please help -- regards, akhil

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DTMF strange problems
by Kevin Mathy 17 Nov '14

17 Nov '14

Hi, I'm facing a "strange" problem with DTMF, on a FS server I'm using (Version 1.5.8b git fe2a4d6 2014-02-14 00:03:11Z 64bit). For some calls going through FS, when customers send DTMF tones (using RFC2833), I see those logs : 2014-11-14 15:47:48.516849 [DEBUG] switch_rtp.c:5683 RTP RECV DTMF 1:742 > 2014-11-14 15:47:48.516849 [DEBUG] switch_channel.c:489 RECV DTMF 1:742 > 2014-11-14 15:47:49.056857 [DEBUG] switch_rtp.c:5683 RTP RECV DTMF C:640 > 2014-11-14 15:47:49.056857 [DEBUG] switch_channel.c:489 RECV DTMF C:640 > 2014-11-14 15:47:49.356848 [DEBUG] switch_rtp.c:5683 RTP RECV DTMF 2:661 > 2014-11-14 15:47:49.356848 [DEBUG] switch_channel.c:489 RECV DTMF 2:661 > 2014-11-14 15:47:49.696846 [DEBUG] switch_rtp.c:5683 RTP RECV DTMF 3:665 > 2014-11-14 15:47:49.696846 [DEBUG] switch_channel.c:489 RECV DTMF 3:665 > 2014-11-14 15:47:50.256872 [DEBUG] switch_rtp.c:5683 RTP RECV DTMF 4:745 > 2014-11-14 15:47:50.256872 [DEBUG] switch_channel.c:489 RECV DTMF 4:745 > 2014-11-14 15:47:50.796855 [DEBUG] switch_rtp.c:5683 RTP RECV DTMF 5:741 > 2014-11-14 15:47:50.796855 [DEBUG] switch_channel.c:489 RECV DTMF 5:741 > 2014-11-14 15:47:51.236850 [DEBUG] switch_rtp.c:5683 RTP RECV DTMF 6:664 > 2014-11-14 15:47:51.236850 [DEBUG] switch_channel.c:489 RECV DTMF 6:664 > Here, the customer tried to send "123456". So, I don't know why this "C" character has been sent, and I'm sure the customer didn't send this, because there's no "C" touch on his phone's keypad... And this "C" results in a double "1" received on the IVR server (hosted on another server than FS). (Ex : 123456 => 1C23456 => 1123456) It doesn't happen everytime, but for around 80% of calls... So it's really a problem for us... I looked at PCAPs for concerned calls, and saw that DTMF tones are only sent with RFC2833, not with SIP INFO nor InBand. I first thought that digits may be received simultaneously with RFC2833 and InBand but definitely not. Do you think what could be the cause of that ? I don't use "start_dtmf" in my diaplans, I already saw in other ML topics that this may cause double digit problems. Is there a way to ask freeswitch to only handle 0-9*# characters ? Or any idea, I'll be glad to try anything that can help us ! Thanks for help, Kevin *Bien cordialement, Best Regards, **Kevin MATHY* | Ingénieur VoIP --

3 2

TLS setup failed
by Emrah 17 Nov '14

17 Nov '14

Hi list, I am able to use FS with SSLv23 with Blink Pro on Mac OS, but that’s about it. I get the following error if I connect with any other device (Bria iOS, Yealink phone, Join Softphone): TLS setup failed (error:00000001:lib(0):func(0):reason(1)) I came across this thread: http://freeswitch-users.2379917.n2.nabble.com/FS-with-SSL-TLS-issues-td7587… but it doesn’t seem to apply to my scenario. I am using a commercial certificate. My devices connect to a domain which has an SRV record which points to itself on the SSL port. SSL host is an A record and matches the CN in the certificate. Server cA check is even turned off on certain phones. The only error I get on my Yealink phone is this: Nov 14 11:04:03 SIP [524]: SDL <6+info > [000] SSL_connect (read done) Nov 14 11:04:03 SIP [524]: SDL <3+error > [000] SSL ERROR Nov 14 11:04:03 SIP [524]: SDL <3+error > [000] SSL_connect error I would appreciate to know how I could debug this further. Or if you have any clue at what may be going on. Thanks! Emrah

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Re: [Freeswitch-users] unable to do (raj)
by Kalyani Kulkarni 17 Nov '14

17 Nov '14

> > > originate sofia/internal/1000%192.168.1.94 &socket(192.168.1.94 9050 > async full) > try 192.168.1.94:9050 async full, the colon is missing in your syntax

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Get Billy Club now on iTunes
by Ken Rice 15 Nov '14

15 Nov '14

New Post on freeswitch.org from anthm check it out at http://ift.tt/14orplP Get Billy Club now on iTunes I posted last week about this from my phone but it didn’t work very well. CURSE YOU MOBILE BROWSERS! Anyway, My kid was in a movie 3 years ago when he was 12 and not even old enough to see it but it finally was released and is available on iTunes. He’s a mean baseball player who makes an ordinary kid grow up to be a killer. Check out Billy Club! Only $5 to rent and $10 to Own. http://ift.tt/14orplT

1 0

Hack for Zoiper speex codec payload problem
by Mateusz Bartczak 15 Nov '14

15 Nov '14

Hi All, I have problem with getting Zoiper with enabled codec speex/8000 working with FS. I saw some discussions about this problem before, but none of them ended with working solution. Maybe these days some of you can share some hints how to hack this Problem is that Zoiper sends SDP payload type 110 for speex/8000, but FS expects other type for that codec. I tried to rewrite SDP on OpenSIPS proxy that sits between end users and FS box, and then later rewrite replies with SDP payload generated by FS to 110 code, but calls always end-up with codec negotiation failure And yes, I know that zoiper is broken, please don't paste RFC here :) It gets very popular on mobile devices and I'm forced to provide support for it, even if its broken

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