June 2016 - FreeSWITCH-users - FreeSWITCH Mailing Lists

tone detect tolerance?
by Yehavi Bourvine 25 Jun '16

25 Jun '16

Hi, Lately, probably after updating from 1.6.7 to 1.6.9, people are complaining that some faxes are received as voicemail. Indeed I don't see in my logs that the tone detect found the fax CNG tone. Hence, I have a couple of questions: - What is the tolerance of tone_detect function? - If I call it a few times with slightly different frequencies, will all of them trigger, or only the last one? Thanks! __Yehavi:

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Early Ring timeout
by Deepika Yadav 25 Jun '16

25 Jun '16

Hi, I am originating calls to mobile number from ESL APIs and want to capture the events of switch-off, busy phone, call rejection etc Corresponding Hangup causes in Freeswitch are -NO_USER_RESPONSE, USER_BUSY, CALL_REJECTED My diaplan is : <extension name="ivr_demo"> <condition field="destination_number" expression="^4441$"> <action application="playback" data="/usr/local/freeswitch/recordings/${folder}/trailer/trailer.wav"/> </condition> </extension> command is originate {folder=abc}sofia/gateway/MySIP/91XXXXXXXXXX 4441 The target phone recieves only around six rings and if phone is not answered, call automatically hangups by reporting hangup cause as NORMAL_CLEARING whereas I originate a conference call : originate {folder=abc}sofia/gateway/MySIP/91XXXXXXXXXX &conference(abc) the phone get around 12 rings and if call not anwered, hangup cause is reported as NO_USER_RESPONSE Is something wrong with the dialplan? Regards, Deepika

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BLF for Sip gateway?
by David Mitchell 24 Jun '16

24 Jun '16

I am setting up a small PBX with freeswitch / debian etc I am using Linksys SP3000 ATA to interface with the PSTN. In/Outbound calls work. I would like to set up a BLF to show if the PST N line is in use, I can configure it on the phone (Yealink T23G) and associate it with the PSTN extension (191) and if I dial 191 from another extension the lamp shows busy (and I am connected to the PSTN line...) However it doesn't show busy for inbound PSTN calls, nor for calls routed to the PSTN gateway via the dialplan (using action = bridge to <pstn number>@<SP200 ip address>:5061) - the calls do get connected to the PSTN. Is there a way to make this work - perhaps by setting the presence within the dialplan actions? David

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Question regarding UUID stacking in a FIFO
by David Ponzone 24 Jun '16

24 Jun '16

Guys, I am searching for a way to stack UUID into a FIFO. What I need is something like db insert/realm/key/${uuid} or hash insert/realm/key/${uuid} but that I need to keep several uuids with the same realm/key tuple. The ideal would be: I first insert realm/key/${uuid1} then insert real/ley/{uuid2} then if I do select/realm/key, it pops out ${uuid1) I can then manually remove it, and the next select will pop out ${uuid2} I don’t think it’s possible with hash/db, so does anyone a way/trick to achieve that ? (the goal is to intercept several simultaneous incoming calls in the RINGING state) Thank you David Ponzone Direction Technique email: david.ponzone(a)ipeva.fr <mailto:david.ponzone@ipeva.fr> tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr <blocked::http://www.ipeva.fr/> - www.ipeva-studio.com <blocked::http://www.ipeva-studio.com/> Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.

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Short ringback file duration leads to bridge cancel with LOSE_RACE
by Dmitry Sytchev 24 Jun '16

24 Jun '16

Hi all! When using ringback="http://something" or ringback="file.wav" and file duration is shorter than bridge timeout, FS doesn't repeat it from beginning. Instead, bridge cancels with LOSE_RACE despite there is only one outbound channel in bridge. Seems it doesn't depend on early media settings on bridge. Does anybody know if this is intended behavior or should I file a jira? -- Best regards, Dmitry Sytchev, IT Engineer

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Debian 8 package i386?
by Jason Moran 24 Jun '16

24 Jun '16

I have VMs of Deb8 for both 64bit and 32bit/i386. The instructions work great for 64bit, and FS installed as expected. However, on i386 the command # apt-get update && apt-get install -y freeswitch-meta-all returns E: Unable to locate package freeswitch-meta-all I looked in http://files.freeswitch.org/repo/deb/freeswitch-1.6/db/packages.db and see i386 packages for 1.4 (Wheezy) but for 1.6 I only see amd64. Is our old x86 server unsupported now?

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Audio skip issue with simultaneous ringing
by Ashwin Jain 24 Jun '16

24 Jun '16

Hi all, We are running Freeswitch version 1.4.26. We are facing a major problem related to audio skip with simultaneous ringing. In one flow we are ringing around 5 agents at the same time. Whenever agent 1 (any one of the agent) picks up the call, other legs (to other agents) gets a hangup event. Due to this, hangup event to all the other agent "CALL_ANSWER" event on the other side gets delayed by around 2-3 seconds and generally caller missed the first word from the agent. Event 1: Call Answered by agent 1 Event 2: Hangup for agent 2 (due to LOSE_RACE) Event 3: Hangup for agent 3 (due to LOSE_RACE) Event 4: Hangup for agent 4 (due to LOSE_RACE) Event 5: Hangup for agent 5 (due to LOSE_RACE) Event 6: Call answered to caller Time different between Event 1 and Event 6 is about 2-3 seconds. Any help would be really really helpful -- Thanks and Regards, Ashwin Jain

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conference play video
by Danny Gershman 23 Jun '16

23 Jun '16

I'm having an issue playing back video in a conference, running (1.6.8 tag and loaded up mod_av), which it only plays audio back (no visual). I'm dialed in using a softphone with VP8 codec. I run command: conference <conf> play http://example.org/play.mp4 Any pointers. Thanks, Danny Gershman

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Does Freeswtich need rtcp attribute in sdp to enable rtcp?
by Allan Kristensen 23 Jun '16

23 Jun '16

Hello, Does Freeswitch need the rtcp attribute in the sdp to enable rtcp? When connecting to asterisk (which does not include a rtcp attribute), Freeswitch does not start to listen on the rtcp port (rtp port +1). When connecting to other Freeswitch servers, which have this attribute in the sdp everything works fine. According to the RFC 3605 it should not be needed, unless it doesn't follow the parity rule (rtp port +1). ...Or is it just me? Best regards, Allan

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OpenBSC <-> Freeswitch ?!
by Anonim Stefan 22 Jun '16

22 Jun '16

Hi, Currently, is there such a module on Freeswitch to allow connection to OpenBSC. Basically what I want to do is what asterik does in [1]. *LCR = Linux Call Router [1] http://openbsc.osmocom.org/trac/wiki/OpenBSC_LCR Thanks, Stefan

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