May 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

fs_cli
by Madovsky 01 May '10

01 May '10

Hi all, when I run from bash as "root" I get this error: /usr/local/freeswitch/bin/fs_cli -x "any_command" [ERROR] libs/esl/fs_cli.c:1181 main() Error Connecting [Socket Connection Error] Freeswitch is running without problems. I think I forgot to something.... svn17134M Thanks Franck

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How to get GMT date/time in channel variable "start_stamp", instead of local
by Vitalii Colosov 01 May '10

01 May '10

Hi, Right now channel variable "start_stamp" represents *local *date/time (to remind, it contains value in following format: 2010-05-01 16:31:58) Is there a similar variable (like start_stamp_gmt?) that will provide *GMT*formatted date/time? Or maybe there are some settings that will make FreeSWITCH provide all date / time formatted variables in GMT instead of local. Thank you, Vitalie

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event after hangup (Javascript)
by Alberto Escudero 01 May '10

01 May '10

Hi, Is there a way to trigger a customized event after the user hangup the call? Although i define the function on_hangup(hup_session, how) it seems i can not trigger and event inside of it although i can run logging. Hints are welcome. -aep -- Stopping junk mailers is good for the environment

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Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 46, Issue 347
by guru singh 01 May '10

01 May '10

> > From: "Dave Stevenson" <stevendt(a)primrosebank.net> > To: <freeswitch-users(a)lists.freeswitch.org> > Date: Fri, 30 Apr 2010 15:53:26 +0100 > Subject: Re: [Freeswitch-users] Picking up voicemail > Hi Brian, > > thanks for the pointer. > > The SPA-3012 is pretty strong on configuration options, but the documentation is very light on what they actually mean. Just so that I know, what is AVT actually doing ? > > regards > Dave DTMF Tx Method Select the method to transmit DTMF signals to the far end: InBand, AVT, INFO, Auto, InBand+INFO, or AVT+INFO. InBand sends DTMF using the audio path. AVT sends DTMF as eypents. INFO uses the SIP INFO method. Auto uses InBand or AVT based on the outcome of codec negotiation. The default is Auto. Shamelessly pasted from http://www.cisco.com/en/US/docs/voice_ip_comm/csbpvga/ata/administration/gu…. I was reading this manual when I saw your post. You should check it out if you haven't already. gs

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Error with media ptime
by leonardo alves 01 May '10

01 May '10

Hello, I am new to freeswitch and I have just installed the last version of freeswitch. I am doing some tests dialing with sip and when the call is answer I play an file. If I do 10 calls, in 5 or 6 of them I get this error in the console and the audio gets cut like with it was a bandwithd problem. 2010-04-27 22:20:56.236782 [WARNING] mod_sofia.c:999 We were told to use ptime 20 but what they meant to say was 30 This issue has so far been identified to happen on the following broken platforms/devices: Linksys/Sipura aka Cisco ShoreTel Sonus/L3 We will try to fix it but some of the devices on this list are so broken who knows what will happen. Does anyone knows if there is a way to fix this issue ? Thanks Leonardo

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libpri not working
by Tom 01 May '10

01 May '10

Voip gods, having problem with libpri on incoming call it gives me this [WARNING] ozmod_libpri.c:760 --Duplicate Ring on channel 1:16 (ignored) then it wait a few seconds then i get a fast busy. If i turn off libpri i get this [CRIT] ozmod_isdn.c:710 Received Release Complete with no matching channel 0 Iam running FreeSWITCH Version 1.0.head (svn-17188M) wanpipe-3.5.10.14 dahdi-2.3.0 Sangoma B601de card Tom Baldwin

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Aastra and SCA
by Mark Maly 01 May '10

01 May '10

Hi, I've tried to patiently figure this out by reading the wiki and this list. Unfortunately, I've been unable to get it right. I have 2 Aastra 6731is and a 51i and trying to get SCA working. I'm experiencing problems similar to the Cisco thread from last month - outgoing calls implement SCA well. Incoming calls ring all lines and appearances work, but when one phone is answered, the line appearances are removed from the remaining phones. I am not attempting to use any DNS. My configuration has all three phones plus FS on a local LAN. Nothing too fancy. Each line is configured for Broadsoft SCA and SCA bridging is enabled globally for the phones. Trying to update/replace an old phone at my church. Any help would be greatly appreciated. Thanks, Mark Mark.maly(a)molcs.org

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Picking up voicemail
by Dave Stevenson 01 May '10

01 May '10

Hi, I have FreeSWITCH running over an internal IP network, connected to the PSTN line with an SPA-3102. When an outside (PSTN) call comes in, a Group of phones are rung until the first one picks up or until the caller falls over to voice mail and can leave a message in the Group Mailbox (100). I can retrieve the messages from any local phone - so far, so good. I'd like to be able to call in through the same PSTN line and interrogate the group mailbox, or any mailbox come to that. Can someone point me in the right direction for how to set this up please ? i.e., once FreeSwitch has answered the call and the voicemail prompts have stated, can they be interrupted so that I can get to the voice mail extension. regards Dave

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Lag on extension behind spa3102
by guru singh 01 May '10

01 May '10

Hi, I am running FreeSWITCH Version 1.0.trunk (17134). I just got my spa3102 and have configured my phone as extension 1000. There's no PSTN or external VoIP yet. I am getting a lag whenever I dial out from the phone. For eg. when I dial the voicemail(4000) I hear "please enter your id....." not the part before that. There's a 10 sec lag when I call the demo IVR, there's also a lag when I dial other extensions(softphones). There's no lag whatsoever while dialing from softphones. Please recommend where I should be looking to resolve this before I move on to more adventurous stuff :) Thanks gs

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