January 2016 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Longtime working user's phone now disconnecting
by Sean Devoy 14 Jan '16

14 Jan '16

Hi, I have an emotional user who now claims her phone fails at least once a week and everyone else's must be, but no one has ever reported it! My very favorite kind of complaint - entirely founded in emotion, myth and misunderstanding. In any case, I called her. I spent 4:11 listening to her complain, but not be able to sight a single instance I could research. Then <CLICK> the connection was lost. She claims she did not hang up, the line went dead. Perhaps there is an actual problem after all. I will add at least one of the errors she reported would lead me to believe there is a hardware (or power) issue on her desk phone (Cisco SPA540). I did look on the phone after the dropped call and the phone did NOT reboot and said it was still registered from 45 minutes earlier. Just FYI: The problems are reported to have started when COMCAST replaced their DSL and they got new external IP addresses. The phones are NATed. Can someone tell me what [BREAK], [NORMAL CLEARING], then "port ending bridge by request from write function" means? Just for background, this was an internal ext to ext dial in separate "domains" on my FS server. NO IVR involved. Here is a log excerpt: 12:32:12.178673 [DEBUG] sofia_glue.c:3429 sofia/external/220(a)redacted.com Setting audio receive payload in Re-INVITE to 0 12:36:23.318530 [DEBUG] switch_core_session.c:1016 Send signal sofia/external/sip:10@redacted_too:port [BREAK] 12:36:23.338532 [NOTICE] sofia.c:737 Hangup sofia/external/sip:10@redacted_too:port [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] 12:36:23.338532 [DEBUG] switch_ivr_bridge.c:569 sofia/external/sip:10@redacted_too:port ending bridge by request from write function 12:36:23.338532 [DEBUG] switch_channel.c:3187 Send signal sofia/external/sip:10@redacted_too:port [KILL] 12:36:23.338532 [DEBUG] switch_core_session.c:1351 Send signal sofia/external/sip:10@redacted_too:port [BREAK] 12:36:23.338532 [DEBUG] switch_ivr_bridge.c:645 BRIDGE THREAD DONE [sofia/external/220(a)redacted.com] 12:36:23.338532 [DEBUG] switch_ivr_bridge.c:675 Send signal sofia/external/sip:10@redacted_too:port [BREAK] 12:36:23.338532 [DEBUG] switch_ivr_bridge.c:645 BRIDGE THREAD DONE [sofia/external/sip:10@redacted_too:port] 12:36:23.338532 [DEBUG] switch_ivr_bridge.c:675 Send signal sofia/external/220(a)redacted.com [BREAK] Any ideas where I should look next or even what the problem is? NOTHING has changed on the server for months and months. The FS version is rather out of date, but has been stable for a long time. Thanks in advance. Sean

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How to know the number of in use licenses for sangoma D100 transcoding card
by Antonio Silva 14 Jan '16

14 Jan '16

hi, Is there a way to know the number of licenses in use by the sangoma card D100? I'm having some issues with the sangoma D100 and i wanted to know if it could be that i'm getting out of channels... Right now i use the command: "sangoma_codec sessions", which gives me the number of total sessions... but this gives strange values, in the card there are 40 licenses and the command in some periods returns more sessions than the available licenses. thanks for the help. -- Saludos / Regards / Cumprimentos, António silva

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Re: [Freeswitch-users] Freeswitch "Cluster"
by Pierre-Emmanuel Dutang 14 Jan '16

14 Jan '16

Hello, Yes it's kinda that, but I'm not sure to understand the explanation. What I'm looking for it's to be able to bridge 2 sessions from 2 differents freeswitch, if an application can manage a link between different freeswitch I m interested. Bests, On Wed, Jan 6, 2016 at 9:26 PM, Luis Daniel Lucio Quiroz < luis.daniel.lucio(a)gmail.com> wrote: > Is this what you are looking for? > > > https://okay.com.mx/en/entrepreneurs/balancing-clustering-and-high-availabi… > > Long text, please read it. > > -- > Luis Daniel Lucio Quiroz > CISSP, CISM, CISA > Linux, VoIP and much more fun > www.okay.com.mx > > Need LCR? Check out LCR for FusionPBX with FreeSWITCH > Need Billing? Check out Billing for FusionPBX with FreeSWITCH > > 2016-01-06 8:47 GMT-05:00 Pierre-Emmanuel Dutang <dutangp(a)gmail.com>: > >> Hello, >> >> I need to create a cluster of Freeswitch for a project. the project is >> simple, we call people through freeswitch and we want to bridge 2 sessions, >> pretty simple :) But to be more scalable I want to use multiple freeswitch, >> that mean a call can go through one freeswitch F1 and another call through >> freeswitch F2. What I want is to be able to bridge a call from F1 with a >> call from F2. >> >> The solution can be bridge both sessions from 2 different freeswitch or >> move a channel to be in the same freeswitch to be able to bridge both >> sessions. >> >> I have no idea if I can do that and how to do it. I just read some stuff >> about mods_cluster but nothing clear with explanation. >> >> Thanks for you help :) >> >> -- >> Pierre >> 06 52 08 00 52 >> >> <http://www.linkedin.com/in/pedutang> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting(a)freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.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 >> > > -- Pierre 06 52 08 00 52 <http://www.linkedin.com/in/pedutang>

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Intercept a call from a fifo
by Antonio Silva 14 Jan '16

14 Jan '16

Hi all, is it possible to intercept a channel from a fifo with the intercept command? scenario: a ----> fifo (with fifo calling ---> B) c -- dial to trigger the command intercept -b uuid of B It fails with "switch_ivr_bridge.c:1915 no uuid bridged to uuid of B" forget to say, it work if the call (fifo -> B) is answered.. Could it be possible now... in 2010 wasn't possible... http://freeswitch-users.2379917.n2.nabble.com/intercept-in-fifo-queue-td473… -- Saludos / Regards / Cumprimentos, António silva

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schedule Broadcast into a conference
by Matt Broad 14 Jan '16

14 Jan '16

Hi, I am wondering if there is an API command that will allow me to schedule a broadcast into a conference? I have tried the following: sched_broadcast 5 UUID playback::tmp/test.wav I have tried replacing the UUID with the conference UUID and no file plays (though the broadcast shows as being scheduled). I have also tried replacing UUID with one of the caller's UUID, the file does play but only to that one caller and not all callers. many thanks Matt

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Translate 415 response to 488 across bridge
by Spencer Thomason 13 Jan '16

13 Jan '16

Hello, We have a few UAs which reject a T38 ReINVITE with a 415 instead of a 488, ugh. If the UA sends a 415, freeswitch send a BYE with a Reason: Q.850;cause=79;text=“SERVICE_NOT_IMPLEMENTED” but if the UA sends a 488, the 488 is passed across the bridge and the T38 ReINVITE is successfully rejected and the call continues normally. Is there a way I can get Freeswitch to treat the 415 as a 488 and send a 488 across the bridge? If its not currently implemented, can anyone provide some hints and I’ll create a PR. Thanks! Spencer

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H264 not added in 200OK
by Frédéric LUART 13 Jan '16

13 Jan '16

Hello, I am trying to establish a video call from a linPhone to a Polycom device. Polycom is configured as an external gateway device on my Freeswitch. I have tried several tests and I have added the H264 configuration in file modules.conf.xml : <load module="mod_h26x"/> and also in vars.xml: <X-PRE-PROCESS cmd="set" data="global_codec_prefs=OPUS,G722,PCMU,PCMA,H264,VP8"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=OPUS,G722,PCMU,PCMA,H264,VP8"/> I am now able to establish the call from linphone to Polycom but only audio is established. I have looked at SIP trace with wireshark and the 200OK that is send back from Freeswitch to LinPhone has only VP8 in the SDP. Linphone -----> Invite (VP8, H264) -----> Freeswitch -----> Invite (VP8, H264) -----> Polycom <----- 200OK (VP8) <----- <----- 200OK (H264) <---- I have tried different configuration without success. Do you have an idea of what may be missing to have H264 in the 200OK instead of VP8 ? Or is there another solution ? Thanks in advance for your help Fred

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Help needed with DSS/BLF LED on phones
by Varsha Agarwal 13 Jan '16

13 Jan '16

Hi, I am trying to configure a programmable key on my SIP phone as DSS/BLF. The intention is to use this key to pickup calls from a pickup group. I also want the LED of the key to turn red when there is an active call in the pickup group. Right now, I can pickup the call fine with the pickup group, but LED is not working at all. Anybody knows what could be the problem. I tried 3 different phones and they all behave the same. Thanks, Varsha

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switch.conf.xml sessions-per-second issue
by Grant Bagdasarian 13 Jan '16

13 Jan '16

Hello, Does FS always reply with a 503 Maximum Calls In Progress regardless of the amount of calls attempts? So if the sessions-per-second is 10 and I generate 20 CPS, the remaining 10 will get a 503? Does this also apply to an higher number of attempts? Regards, Grant

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Freeswitch Version 1.6.5-11 crashes with Perl
by Ali Pey 13 Jan '16

13 Jan '16

Hello, I have Freeswitch Version 1.6.5-11 on a debian server and it crashes every time I try a perl call. I have tried running freeswitch as root and has the same problem. Has anyone used perl module with 1.6.5-11. I'd like to know if it's a bug or an installation problem on my side. The error I get is: freeswitch@internal> perl test.pl <https://contactmonkey.com/api/v1/tracker?cm_session=c9fa2dc2-5197-4125-b49c…> Socket interrupted, bye! Thanks, Ali Pey

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