November 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Disastery Plan
by Nikita Tarasov 02 Dec '13

02 Dec '13

Hi guys. We have loadbalancer(opensips) and a huge amount of incoming calls going to ONE sip server, which is unscalable and priopretary. I need to create some kind of disastery plan for this server service is down or on high load. The main idea of this plan is, when disastery happends, to balance all incoming calls to freeswitch and keep them with music, while main server is down. On recovery calls should be forwarded to main server with the same destination number as they was. I have tryed to make it with mod_callcenter. It was done by placing calls in queue and when main server is on and re-registered on freeswitch, calls was forwarded. But without destination number. And I can't copy destination number to forwarded calls. Can you help me?

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freeswitch stopped working
by Joli Martinez 30 Nov '13

30 Nov '13

Hello, My freeswitch server does not want to start anymore. When I try to start it from the init.d script It say "freeswitch dead but subsys locked". When I go to "/usr/local/freeswitch/bin" and start it that way it does start but I am not able to dial anyone internal I get "switch_core_session.c:2616 Invalid Application transfer" The last thing I remember doing was last week I installed a python module but did not restart it until now. I also uncommened out the python module in modules.conf.xml and same result. Please help Thanks,

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(no subject)
by Vincent Xia 30 Nov '13

30 Nov '13

has anybody ever tested freeswitch 1.4 with webrtc support under windows platform? i tried to build and run fs 1.4 in windows 7 and windows server 2008 as the webrtc server but no success, i simply uncommented <param name="ws-binding" value=":5066"/> and had the jssip client connect to it, but nothing happened on the fs console, the log from chrome said it's "Connecting to WebSocket URI ws://fs_host_ip:5066 jssip-0.2.1.js:619", i did netstat -ano and saw freeswitch is already listening to 5066 and chrome is connecting to is TCP local_ip:59687 fs_host_ip:5066 ESTABLISHED 12316 (fs uses 5066, chrome uses 12316) so did i miss something to have freeswitch up for webrtc in windows? ps:the firewall is off and the same issue happens with tryit.jssip.net. Regards,

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Adjust Contact port in SUBSCRIBE request with TLS
by Jon Bruel 30 Nov '13

30 Nov '13

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g729
by Joli Martinez 30 Nov '13

30 Nov '13

I am looking to do some tests from my Freeswitch box to 2-3 sip clients over the WAN. This will only be a test and not for production. Is there a way to get g729 for free for testing purpose, or maybe a trial version. We will need the ability to make and receive calls on the sip clients from both internal and external callers. If the project goes into production we will buy the appropriate lic for it. Thanks,

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Friday Free For All
by Jeff Leung 29 Nov '13

29 Nov '13

Relaying this for Ken: 10:59 <@SwK> Hey Guys don't forget the friday free for all today 10:59 <@SwK> starting soon 11:00 <@SwK> http://fs0.us/call888 for how to join 11:02 <@SwK> ~twitter FridayFreeForAll! (F^3A) http://fs0.us/call888 All afternoon on the FreeSWITCH Bridge #OSS #SIP #WebRTC #VoIP

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Re: [Freeswitch-users] Apply Connection Charges on LegB
by Shahzad Bhatti 29 Nov '13

29 Nov '13

Thanks Avi and Andrew it works. Regards Shahzad Bhatti From: Avi Marcus <avi(a)avimarcus.net> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Thu, 28 Nov 2013 17:30:42 +0000 > Subject: Re: [Freeswitch-users] Apply Connection Charges on LegB > Ah, sorry! I read "apply connection charges" as "when the call connects" > and figured I understood. > > OK, so I presume you want to export:nolocal an execute_on_answer or > api_on_answer that calls something that charges the 5cents > http://wiki.freeswitch.org/wiki/Variable_execute_on_answer > http://wiki.freeswitch.org/wiki/Variable_api_on_answer > > It seems nibblebill supports this directly: > >> Adding/Deducting funds >> Inserting this in your dialplan: >> <action application="nibblebill" data="adjust 5.00"/> >> ... Use negative numbers to deduct from an account. > > ... you'd of course want to make sure this ends up in your logs somewhere > so you have an audit trail! > Seemingly this would be an execute_on_answer. > > Hope that helps! > -Avi > > On Thu, Nov 28, 2013 at 7:03 PM, Andrew Cassidy < > andrew(a)cassidywebservices.co.uk> wrote: > >> Avi, he wants to apply an additional charge to the call that is per call >> and not per minute, as well as the per minute rate. >> >>

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DTMF RTP issues- HELP
by Andrew 29 Nov '13

29 Nov '13

Hi, I found that DTMF, AVMD and Talk on/off will only work if RTP stream is running from both sides of the call. What I am doing is using the sampleclient program and making a call. If the sample client is not playing audio then none of these work. As soon as I play an audio file everything works. I need it so all this works on even if I am not streaming audio. RTP from the person I called should still work. Andrew

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FreeSWITCH not sending P-Asserted-Identity
by Vik Killa 29 Nov '13

29 Nov '13

I cannot get FS to send the P-Asserted-Identity header instead it always sends the P-Preferred-Identity header. According to http://wiki.freeswitch.org/wiki/Channel_Variables#sip_cid_type I must set the privacy flag in order for P-Asserted-Identity to be used, so here is my dialplan entry and the SIP INVITE that is sent: <extension name="outbound"> <condition field="destination_number" expression="^(\d{10})$"> <action application="lcr" data="$1"/> <action application="set" data="effective_caller_id_name=${outbound_caller_id_name}"/> <action application="set" data="effective_caller_id_number=${outbound_caller_id_number}"/> <action application="set" data="privacy=yes"/> <action application="privacy" data="yes"/> <action application="set" data="sip_h_Privacy=id"/> <action application="bridge" data="{sip_cid_type=pid,origination_privacy=hide_namehide_number}${lcr_auto_route}"/> </condition> </extension> ------------------------------------------------------------------------ INVITE sip:+17165555555@YYY.YYY.YYY.YYY:5060 SIP/2.0 Via: SIP/2.0/UDP 72.43.207.154;rport;branch=z9hG4bKK0pepF2H0atSp Max-Forwards: 68 From: "1000" <sip:7165551001@XX.XX.XX.XX>;tag=8U2rQ1pmXXaDQ To: <sip:+17165555555@YYY.YYY.YYY.YYY:5060> Call-ID: de66ec94-12f5-1230-cbb1-5e6ae73de10d CSeq: 27864844 INVITE Contact: <sip:mod_sofia@XX.XX.XX.XX:5060> User-Agent: FSwitch Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Privacy: id Content-Type: application/sdp Content-Disposition: session Content-Length: 209 X-FS-Support: update_display,send_info P-Preferred-Identity: "1000" <sip:7165551001@XX.XX.XX.XX> v=0 o=FreeSWITCH 1336383832 1336383833 IN IP4 XX.XX.XX.XX s=FreeSWITCH c=IN IP4 XX.XX.XX.XX t=0 0 m=audio 17984 RTP/AVP 0 9 8 3 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20

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Movember - Mens Health
by jay binks 28 Nov '13

28 Nov '13

So Ill keep it brief. Im trying to raise money to support Mens health. ( prostate cancer , depression , all these things ) anyways.. I know ill probably get flamed etc etc.. But its a good cause, and I wanted to ask if anyone can make a donation to help my team reach our goal. We have a team of 14 people and we are trying to raise $5k as a team. We are over 1/2 way there, and any support would be appreciated. Donations can be made at http://mobro.co/jaybinks and you can see the (aweful|amazing) moustache Ive grown this month. -- Sincerely Jay Binks

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