July 2014 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Dingaling In Component Mode "How To Call JingleTo SIP"
by Ali Jawad 11 Aug '14

11 Aug '14

Dear All This might be a duplicate post, I am sorry if it is. Normally I get direct feedback from the list. But this time I did not get any reply. So I was afraid it might not have been delivered. First I would like to state that dingaling is working in client mode with gtalk perfectly. I can call from gtalk to my sip extension "1000". I have setup component mode now I am using FS with Openfire, open does show the following: 1 fs.jabber2.mydomain.net LibDingaLing gateway However stats on the jabber server are Session Statistics: Packets Received/Sent: 1/1 So the component is REGISTERED, so far so good my problem now is how to initiate a call from XMPP to a SIP extension. Using client mode I would do the following: 1. Setup GTALK account and assign extension to it 2. Login with another gtalk account and dial the first gtalk account 3. The assigned extension rings I don't know how to do the same using component mode. Adding sip+1000@mydomain to the jabber client "Jabbin" does not popup an "add contact" on XTEN. Anything I missed here ? Thanks for any hints or suggestions The Freeswitch server logs are below: 2008-04-11 13:57:32 [NOTICE] libdingaling.c:1276 on_log() SEND: ------------------------------------------------------------------------ ------- <stream:stream xmlns:stream="http://etherx.jabber.org/streams" xmlns="jabber:component:accept" to="fs" version="1.0"></stream:stream> 2008-04-11 13:57:32 [INFO] libdingaling.c:1274 on_log() RECV: ------------------------------------------------------------------------ ------- <stream:stream xmlns:stream="http://etherx.jabber.org/streams" xmlns="jabber:component:accept" from="fs.jabber2.MyDomain" id="69fcaa7b"></stream:stream> 2008-04-11 13:57:32 [NOTICE] libdingaling.c:1276 on_log() SEND: ------------------------------------------------------------------------ ------- <handshake>72dac833190326bb82dddedae558b2b794df5e1b</handshake> 2008-04-11 13:57:32 [INFO] libdingaling.c:1274 on_log() RECV: ------------------------------------------------------------------------ ------- <handshake></handshake> 2008-04-11 13:57:32 [INFO] libdingaling.c:1274 on_log() RECV: ------------------------------------------------------------------------ ------- <iq type="get" id="970-504" from="component.jabber2.MyDomain" to="fs.jabber2.MyDomain"> <query xmlns="http://jabber.org/protocol/disco#info"></query> </iq> 2008-04-11 13:57:32 [NOTICE] libdingaling.c:1276 on_log() SEND: ------------------------------------------------------------------------ ------- <iq from="fs" to="component.jabber2.MyDomain" id="970-504" type="result"> <query xmlns="http://jabber.org/protocol/disco#info"> <identity category="gateway" name="LibDingaLing"></identity> <feature var="http://jabber.org/protocol/disco"></feature> <feature var="http://jabber.org/protocol/disco#info"></feature> <feature var="jabber:iq:version"></feature> <feature var="vcard-temp"></feature> <feature var="http://www.google.com/xmpp/protocol/voice/v1"></feature> <feature var="jabber:iq:last"></feature> </query> </iq>

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Call Rejection Drops Call
by Joel White 08 Aug '14

08 Aug '14

I am having an issue with calls coming from another SIP server into FS. Everything works fine with the call getting to its destination. And also works if the call is ignored and goes to voicemail. However if the call is rejected by a polycom endpoint the call is dropped completely and disconnects without going to voicemail. Here is a sample of the dialplan <action application="answer"/> <action application="bridge" data="sofia/internal/57331(a)192.168.100.25 "/>  <action application="voicemail" data="default ${domain_name} 57331"/> <action application="hangup"/> If someone could point me in the right direction

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New user question, does it have distrubtion which install like pbxinaflash
by John Joseph 07 Aug '14

07 Aug '14

Hi All, New user here. Would like to get advice on Does FreeSwitch has installation with GUI , like PBX in a flash for asterisk Thanks Joseph John

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Conference auto outcall rings with 10s delay?
by MrX 07 Aug '14

07 Aug '14

Hi All, Here is my problem: we are using the Freeswitch conference(mod_conference), when conference starts, we want it to automatically call group members in, so we use conference_set_auto_outcall like this: <action application="conference_set_auto_outcall" data="user/1000@$${domain}"/> <action application="conference_set_auto_outcall" data="user/1001@$${domain}"/> <action application="conference_set_auto_outcall" data="user/1002@$${domain}"/> <action application="conference_set_auto_outcall" data="user/1003@$${domain}"/> ...... but very often, there are some members which delay significantly(most about 10s, sometimes 3~4s) from conference call(3001) to it rings, while other members rings with no delay. And, if we make a second conference call instantly after last conference ends, the delay almost never happens. The longer the interval is, the bigger is the delay chance. When we tryed to use some packet capture tool to analyze, we found that SIP INVITE packet for the delayed peer was sent 10s after conference call, and there was no SIP packet at all during the 10s delay, So the delay is from serverside. The question is, may there be something wrong with our FreeSWITCH configuration, or the delay is as intended for some purpose? How can avoid the outcall delay and make all peers ring at the same time when conference starts? Thanks and Best Regard, David

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No Logfile / No console output
by Daniel Neubert 06 Aug '14

06 Aug '14

Hi there, I've a FreeSWITCH (Version 1.5.13b git e7ee405 2014-06-15 11:51:56Z 64bit) up and running in Ubuntu 14.04.1 LTS 64bit. The system works very well except one issue: There is no logfile / console output any longer. The console fs_cli works and responds to all commands. Setting log level returns +OK but nothing happens: freeswitch@10.211.11.2@internal> log level debug +OK The logfile freeswitch.log is empty (0 bytes). I've stopped FreeSWITCH several times and removed the file manually when FreeSWITCH was stopped. FS creates a new 0-byte file on restart but does not write to the file. Both (console and logfile) stopped working a few days ago without any apparent reason. Any clues? Regards Daniel

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Strange media behavior on WebRTC
by Oleg Stolyar 05 Aug '14

05 Aug '14

Hi guys, I am connecting to FreeSWITCH using WebRTC and after a user calls in, I play a beep and put them into a conference. Two strange things happen after that: 1. There is a noise in the user's leg that someone described like a wind in the tunnel. The noise lasts for several seconds, then the user can hear the hold music. The noise does not seem to happen if I don't play the beep before the conference. 2. About 3 seconds into the call, there is another very short beep - like a small portion of the original beep that the user hears. This absolutely does not happen with softphones - only with WebRTC. I tried it with Chrome and Opera - same thing is happening on both. I tried increasing the answer delay from 2 seconds to 3 and the noise now happens more rarely but the extra beep is still there almost every time. I am using JsSIP 3.0 but since it only handles signalling I don't think the problem is with the JS library. Has anyone run into this? Any advice? Here is the relevant excerpt from my dialplan. <condition field="destination_number" expression="^conf-(\S+)$"> <action application="set" data="answer_delay=3000"/> <action application="answer"/> <action application="playback" data="tone_stream://%(1000,0,600)" /> <action application="conference" data="$1@default"/> </condition>

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Limit Database
by Andre 05 Aug '14

05 Aug '14

HI, on Windows I'm trying to use Limit with Microsoft SQL DB but it's still using the sqlight database. I have mod_Db and Mod_Hash loaded and under db.config.xml I have my odbc setup like this <param name="odbc-dsn" value="sqldb64::" /> I know the odbc works because I have the core database setup the same <param name="odbc-dsn" value="sqldb64::" /> <param name="core-dbtype" value="MSSQL"/> Is there a step that I'm missing? Thanks

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Polycom hold problems?
by Yehavi Bourvine 04 Aug '14

04 Aug '14

Hi, After upgrading from Freeswitch 1.2.23 to 1.4.6 my users started to complain that they cannot use hold on incoming sessions on Polycom phones; when they try to return from hold the call is disconnected and I get a console error "CODEC renogotiation error". Our Polycom phones are set to offer SRTP, thus, on outgoing call they use SRTP but not on incoming. In the above scenario, when the user tries to return from hold, the phone sends SAVP with SRTP, although the original call was not encrypted, and this causes this error message. I had to disable SRTP on the Polycoms to make hold working again. BTW, this happens with Polycom 3.3.4 but not with 4.0.5 (but I cannot upgrade to it since it behaves erratically). Anyone else noticed this as well? Thanks, __Yehavi:

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Hong Kong/Singapore VoIP Carriers
by Eric Detterman 02 Aug '14

02 Aug '14

We're planning for a FS cluster in Singapore and Hong Kong and are looking for carriers to integrate with - preferably ones with higher quality & service tiers, even if it means paying a bit more. Does anyone have any recommendations for companies like this? So far Google hasnt been overly helpful.

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Latest master and latest v1.4 consume more CPU for conferences
by Oleg Stolyar 01 Aug '14

01 Aug '14

Hi guys, I have my FreeSWITCH running a bunch of conferences with two or three users using PCMU/8000 from all the user agents. I have the conference sample rate set to 12000 to get around a strange bug I reported a couple of months ago. In my old installation with master from May 22 2014, the servers consumed about 1.2%-1.3% CPU per two-person conference. In the latest master AND in the latest v1.4 it went up to about 1.6%-2.0% CPU per two-person conference. CPU usage by the FreeSWITCH process also jumps up and down more. Additionally (not sure if it's related) the latest master crashed less than 24 hours after it started taking live traffic. I ma trying to get the core dump and see if v1.4 also crashes but including the info here in case it gives someone an idea of what's going on with the CPU. Any ideas as to what's causing the increased CPU usage would be greatly appreciated.

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