February 2016 - FreeSWITCH-users - FreeSWITCH Mailing Lists

WebRTC Core Dump for Video Conference.
by Veerabhadrarao Kankatala 16 Feb '16

16 Feb '16

Hello, I am using freeswitch-1.6.5 version for webRTC I am using Chrome to intiate call (WebRTC Client) to another Video device (Grand stream), and i am using Default Dialplan to make call both are registered in freeswitch WebRTC Client: 1001 Default user GrandStream device: 1009 Default user When i make audio calling it is working fine, but when i make Video call from WebRTC Client ----> GrandStream device, my freeswitch is crashing following is the log Please tell me how to resolve this issue and let me know still anything i am missing. -- Thanks & Regards Veerabhadrarao Kankatala Software Developer (C & Unix ) PANAMAX INFOTECH LIMITED Mobile: +91-8401231249 Messenger Id: Skype: veerabhadrarao.kankatala E-mail: veerabhadrarao.kankatala(a)panamaxil.com

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Problem with mod_spy
by Stefano Favaro 16 Feb '16

16 Feb '16

Hello, I have a problem with the mod_spy module. It seems that it just plays music and do not actually spy. I want to dial an extension with the number of the user I want to spy and wait for a call in or out for that user. Mod_spy application plays music but when a call is handled by the user I can't hear it continues to play the music on hold audio file. This is the dialplan: <extension name="eavesdrop"> <condition field="destination_number" expression="^88(.*)$|^\*0(.*)$"> <action application="answer"/> <action application="userspy" data="$1@myserver"/> </condition> </extension> If I dial 881000, for example, It means I want to spy on user 1000. I have in and out calls from user 1000 but I can't hear. userspy_show in fs_cli, I get : 1000@myserver : e9165eaf-e8d7-40ad-a8cd-d2963f363684 1 total spy I have FreeSwitch version FreeSWITCH Version 1.4.26-37~64bit (-37 64bit) SF.

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freeswitch 1.6 on Ubuntu 14.04 LTS
by Caragea Silviu 16 Feb '16

16 Feb '16

Hello, Is Ubuntu 14.04 supported on FS 1.6 ? Seems there are compilation issues for several modules Silviu

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extra header account code is not written to cdr if cancel is received a few ms after invite
by Panagiotis Skoulikaritis 16 Feb '16

16 Feb '16

Dear all I have an implementation FreeSWITCH as a sort of SBC, it is used to send the calls to the terminating carriers and do topology hiding, nothing fancy. Also I gather cdrs from the FreeSWITCH. In order to distinguish each customer on the FS cdrs I send an extra header containing the accountcode. I have noticed that if the call is canceled immediately on the same sec, the account code is not written on the cdr. To be more precise the cancel is send a few milliseconds after it has received the invite, and before the FreeSWITCH has sent the call to the terminating carrier (I'm using Homer Sipcapture to capture all the traces and I don't see an attempt being made at the terminating carrier) also I don't see a b-leg cdr. FreeSWITCH is writing both a-leg and b-leg cdrs in csv format. The dialplan that I use is simple <extension name="carrier-1"> <condition field="network_addr" expression="^(^xx\.xx\.xx\.xx|^yy\.yy\.yy\.yy)$"/> <condition field="destination_number" expression="^2#(\d+)$"> <action application="set" data="call_direction=outgoing"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/> <action application="set" data="sip_cid_in_1xx=false"/> <action application="set" data="inherit_codec=true"/> <action application="set" data="accountcode=${sip_h_X-ACCOUNTCODE}"/> <action application="unset" data="sip_h_X-ACCOUNTCODE"/> <action application="set" data="execute_on_answer=sched_hangup +21600 alloted_timeout"/> <action application="bridge" data="sofia/external/4567$1@zz.zz.zz.zz:5070"/> </condition> </extension> any idea how I can make sure that the account code will always be written ? Best Regards Panagiotis

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bridging out and custom signaling question
by Chris Mandra 16 Feb '16

16 Feb '16

Hi guys - I hope things are good where you are. I have a question: What is the easiest way to bridge a call out of freeswitch not using sip but using a custom signaling protocol that may be a restful protocol? Thank you ! chris

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Where to put Jitterbuffer?
by Rajil Saraswat 16 Feb '16

16 Feb '16

Hello, I have two FS instances connected to each other over VPN. The VPN sometimes has poor connectivity and I was wondering whether jitterbuffer can help to improve the call quality. I am using the OPUS codec between the FS instances since it is supposed to behave better in lower bandwidth. The ATA on both ends are Linksys SPA3102. My connections is as follows: ATA_A---->FS_A---->vpn---->FS_B---->ATA_B When user A is calling to B where should i specify jitter buffer amongst the following: 1. The bridge between Freeswitch instance FS_A and FS_B <action application="bridge" data="{jitterbuffer_msec=60}sofia/gateway/$1@FS_B_IP"/> 2. The bridge between Freeswitch instance FS_B and ATA_B <action application="bridge" data="{jitterbuffer_msec=60}sofia/internal/$1@ATA_B_IP"/> Thanks, Rajil

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multiple internal ip
by Antoine Durant 15 Feb '16

15 Feb '16

Hi, I would like to have more internal IP addresses to do some "Multiple Companies". The WAN IP is used (80.X.X.125) to mount all trunk (1 trunk by company) to the same provider sip. Is it possible to have an internal IP for each company: company-a.org => 192.168.1.1 company-b.org => 192.168.1.2 company-c.org => 192.168.1.3 company-X.org => 192.168.1.X How to use FreeSwitch this way? Thanks

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SWITCH_EVENT_CHANNEL_PROGRESS_MEDIA
by Andre DeMattia 15 Feb '16

15 Feb '16

Hi is there an application that will give me the same results as this event? switch_event_types_t.SWITCH_EVENT_CHANNEL_PROGRESS_MEDIA I see <https://wiki.freeswitch.org/wiki/Variable_execute_on_ring> execute_on_ring and execute_on_media. Wondering if either of these are the same as switch_event_types_t.SWITCH_EVENT_CHANNEL_PROGRESS_MEDIA Thanks Andre

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High availability on different networks
by Emrah 15 Feb '16

15 Feb '16

Hi list, I’m writing to gather your thoughts and suggestions on how to have a high availability FS setup on different networks. I am trying to achieve the following: - Load balance FreeSWITCH instances on 2 or more servers, possibly in different countries. - Shared user directory and dialplan, but I’m not sure if shared registrations would make sense. - If a server goes down, the phone should register on the alternative servers. Obviously we can’t keep calls up. I’m obviously not the first one out there doing this. I’m trying to learn from those who’ve come up with reliable solutions. I’ve tried sharing a registration table among multiple FS instances. But it was a beginners mistake. Even with the right path to reach the client, only the invites sent from the server used by the phone would be processed. If my phone registers on server A, then server A shares the info with server B, server B knows how to contact the phone but it won’t be able to. Supposedly because of NAT issues. I am aiming for fully independent FS instances that can back each other up and be used independently. I am guessing this would require some sort of SBC or external registrar server with a Kamailio or Repro. Anyway just trying to spark the conversation around this subject and hopefully we can come up with a formula that can help many with their FS deployments. My provider’s network just went all down in IPv4 and HA behind the same provider proved to be useless. Best, E

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Freeswitch doesnt transcode
by Rajil Saraswat 15 Feb '16

15 Feb '16

Hello, I have a remote sip phone (Linksys SPA3102) which only supports PCMU. When I call to this remote sip phone i get a 406 error that opus is not supported as shown by the sip trace below. However, if I force the codec to absolute like this {absolute_codec_string='PCMU,PCMA'}sofia/internal/303(a)192.168.1.5 the call works fine. Is there anyway I can make FreeSWITCH to automatically transcode without forcing the codec string in the dial plan? The codec preferences is set as global_codec_prefs=OPUS,PCMU,PCMA and outbound_codec_prefs=PCMU,PCMA,GSM ---------------------------siptrace-------------------------------- recv 333 bytes from udp/[192.168.1.5]:5060 at 08:02:16.368499: ------------------------------------------------------------------------ SIP/2.0 406 Not Acceptable Via: SIP/2.0/UDP 192.168.1.111;rport=5060;received=192.168.1.111;branch=z9hG4bKeS356tttajjej Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124 From: "202" <sip:202@192.168.1.111>;tag=DFX0FUvr2vNcm To: <sip:303@192.168.1.5>;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q CSeq: 87372504 INVITE Content-Length: 0 ------------------------------------------------------------------------ send 324 bytes to udp/[192.168.1.5]:5060 at 08:02:16.368591: ------------------------------------------------------------------------ ACK sip:303@192.168.1.5 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.111;rport;branch=z9hG4bKeS356tttajjej Max-Forwards: 68 From: "202" <sip:202@192.168.1.111>;tag=DFX0FUvr2vNcm To: <sip:303@192.168.1.5>;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124 CSeq: 87372504 ACK Content-Length: 0 ------------------------------------------------------------------------ 2016-02-14 08:02:16.356283 [DEBUG] sofia.c:6760 Channel sofia/internal/303(a)192.168.1.5 entering state [terminated][406] 2016-02-14 08:02:16.356283 [NOTICE] sofia.c:7779 Hangup sofia/internal/303(a)192.168.1.5 [CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED] Thanks Rajil

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