- FreeSWITCH-users - FreeSWITCH Mailing Lists

Voicemails chopped at 10 seconds (randomly!)
by Peter Hartmann 27 May '14

27 May '14

Hey, I have a user that randomly gets voicemails chopped off at exactly 10 seconds. This is using a Sangoma A200 analog card....so it's not a NAT problem. Apparently this has nothing to do with silence detection as I've tried different settings for that. Cutoffs happen in mid word and the audio file length is exactly the same in each when it happens....not always. There's nothing unique about the way and which calls get to this users mailbox. IVR does this: "transfer 1201 XML default". Users mailbox is 1201 and the caller is redacted to 212xxxxxxx. Here's the FS Log for one of the calls that had a 10 second voicemail. http://pastebin.com/Fq5N9UCq Here's Local Extension dialplan snippet that the calls pass through in the event I'm doing something wrong there. http://pastebin.com/FK0zXs4A Admittedly this is a bit out of date: FreeSWITCH Version 1.5.8b+git~20140124T004436Z~bd7182c390~64bit (git bd7182c 2014-01-24 00:44:36Z 64bit) I'm perplexed! Thank very much in advance! Peter Hartmann Hartmann Computer Consulting http://blog.hartmanncomputer.com (212)203-8870 If I can't explain it to you in plain language, that means I don't understand it.

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Docs Call! And other Reminders!
by Ken Rice 27 May '14

27 May '14

Greetings FreeSWITCHers! Hope all of you in the US had a nice long weekend, but its time to get back to work. Calling all Documenters! Today, Tuesday the 27th of May 2PM CST (1700 GMT) on 888 the weekly Docs Team call will kick off! Join us and help continue the great work! Tomorrow, Wed the 28h of May, Join us as Eric Crist, Dir of IT at ClaimLynx joins us to talk about Faxing and FreeSWITCH on ClueCon Weekly! And Friday May 30th, mark your calendars for the FreeSWITCH Friday BugHunt starting about 2PM Central! Where are all these calls? sip:888@conference.freeswitch.org of course, and if you need PSTN or another method of access see http://wiki.freeswitch.org/wiki/Weekly_Conference_Call_Calling_Instructions -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH

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opus crashing freeswitch?
by Chris Martineau 27 May '14

27 May '14

Hi, Is anyone aware of an issue with implementing the opus codec on freeswitch? We have recently implemented opus and ever since the switch has crashed every hour or so. The logs show nothing around the time of the crash so I am thinking it is deep in the opus module. We are running the latest stable version 1.2.23. Any help you could offer would be appreciated. Regards Chris

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Custom variables for mod_conference
by Phil Dunks 27 May '14

27 May '14

Hi FS Guys It would be very useful for our application if we could set some arbitrary custom variables into a conference via the mod_conference api (e.g similar to uuid_setvar for setting channel variables). I have updated mod_conference to allow this, and it is working ok for me. I’m certainly not a C expert, but basically copied the pattern for other api commands. If I submit it as a patch please would you consider it for inclusion (and make any corrections that you feel necessary). Thanks Phil

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FIFO Caller ID
by Joshua Foshee 27 May '14

27 May '14

I am trying to append something to the caller ID name so that users know what queue a call is coming from. My phones do not support SIP UPDATE messages so the caller ID is not being updated. Can someone please define a little more for me the variable "fifo_export"? Also is there any other options to append to the caller name on a FIFO call? Thanks, Josh

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Fragmented IP protocol Issue with latest master and 1.4.4
by Oleg Stolyar 27 May '14

27 May '14

Hi guys, I upgraded to the master (and it's also happening in 1.4.4). I have some very long SIP addresses in my messages. It used to work fine on the master from August 2013. However, now when FS sends back the OK message, to invites with very long destination numbers, the originator softphone cannot receive it. This is only happening with UDP. TCP works fine. Here is a snapshot of wireshark for the master https://www.dropbox.com/s/0y0m4f4t4cgnwu0/Current%20Master.PNG And here is the snapshot of the same OK messages for the August 2013 FS that works https://www.dropbox.com/s/7wpsyw4r2skifpb/August%202013%20Version.PNG Any ideas how to fix this? Is this a known issue?

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(no subject)
by Jason Holden 27 May '14

27 May '14

Hi. Does anyone have any recommended dialplan or scripts that can be used to do the following. 1. update the IVR soundfile in real time 2. update the IVR options I tried using the Freeswitch record_sound_files.lua and see that for some reason it results in all system resources being used, the only fix it to restart fs. Thanks.

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calling multiple destinations
by Vincent Xia 27 May '14

27 May '14

i was trying to ring multiple endpoints (A calls B1 and B2 simultaneously) by doing <action application="bridge" data="user/${callee1},user/${callee2}"/> i was expecting one callee hung up once the other answers the call, but what i was experiencing is one hung up once the other rings, the FS console log says: [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/sip:callee1@1.2.3.4! [NOTICE] switch_ivr_originate.c:3448 Hangup sofia/callee2 [CS_CONSUME_MEDIA] [LOSE_RACE] so how can i make the calls happen as expected?

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Jira-6550 - Ansynch PTime not detected on transfered call
by Sean Devoy 27 May '14

27 May '14

Hi All, Just curious if anyone else has seen this, I opened a JIRA. My Configuration: * Voip ISP w/ preferred CODEC PCMU:20 * Phone A - Cisco SPA504G w/ pref Codem PCMU and PTime 20 * Phone A - Cisco SPA504G w/ pref Codem PCMU and PTime 30 My ISP calls go through an IVR. Calling through the IVR to Phone A is perfect, PCMU:20 <->PCMU:20. Calling in through IVR to Phone B is still great, FS detects the Ansych PTIME and adjusts to PCMU:20<->PCMU:20. But if the call goes through the IVR to Phone A (PCMU:20<->PCMU:20), and THEN is (attended) XFERed to Phone B, the result is a badly delayed (3 to 5 second) and degraded audio stream PCMU:20<->PCMU:30. Calling direct to Phone B, FS says "Asynch ptime not allowed, changing our end from 30 to 20." Freeswitch does not detect the same asynch ptime mismatch at the end of the attended xfer! I understand changing Phone B resolves this problem, but it is still a bug (I think). Sean

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ESL random hangup
by Grant Bagdasarian 27 May '14

27 May '14

Hello, Is it possible for a call, which is parked, and managed by an outbound ESL application, to be hung up automatically? I park the call, play something to it, then let it wait in parked a state. This problem does not happen all the time, its random. This is what I'm receiving: 2014-05-27 12:37:18.190395 [NOTICE] switch_core_state_machine.c:262 sofia/outbound/1000@1.1.1.1:5060 has executed the last dialplan instruction, hanging up. 2014-05-27 12:37:18.190395 [NOTICE] switch_core_state_machine.c:264 Hangup sofia/outbound/1000@1.1.1.1:5060 [CS_EXECUTE] [NORMAL_CLEARING] How can this happen? There is no dialplan when youre using ESL right? Except the socket application to handoff call control. Any tips? Regards, Grant

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