- FreeSWITCH-users - FreeSWITCH Mailing Lists

No ringback tone on bridge
by mbo 19 Nov '13

19 Nov '13

Hello, we receive calls from gateway A and bridge them using gateway B to the destination number . <extension name="test"> <action application="bridge" data="sofia/gateway/B/+49123456789" /> </extension> The problem is, that we do not hear a ringtone at all. If we put a "ring_ready" before the bridge, we can hear a ringtone immediately, but then the callers hear a ringtone even if the bridged destination is not ringing yet. <extension name="test"> <action application=ring_ready /> <action application="bridge" data="sofia/gateway/B/+49123456789" /> </extension> Is there a solution that the callers hears a ring tone as soon as the destination is ringing, but not before ? Thanks Markus

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Anyone really familiar with Confluence that whats to lend me a hand with a project?
by Ken Rice 19 Nov '13

19 Nov '13

Contact me off list or find me on IRC SwK on #FreeSWITCH via irc.freenode.net Thanks K -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org G+ ClueCon : http://fs0.us/cluecon-gplus FB ClueCon : http://fs0.us/cluecon-fb G+ FreeSwitch : http://fs0.us/freeswitch-gplus FB FreeSWITCH : http://fs0.us/freeswitch-fb Twitter : @FreeSWITCH_WIRE irc.freenode.net #freeswitch

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Channel variable to detect WebRTC is being offered
by Joegen Baclor 19 Nov '13

19 Nov '13

I am playing with FreeSWITCH 1.4 beta branch to create a WebRTC to SIP Bridge via ESL. Reading the code, it seems media_webrtc variable is only being set for outbound channel. What is the best way to detect that the inbound channel offer is a WebRTC stream? - joegen

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module install
by Joli Martinez 19 Nov '13

19 Nov '13

Hello, I have installed Freeswitch via yum on a CentOS box. Not sure how to add a module on Freeswitch. Could someone help me since this is installed via yum and not compiled. Thanks,

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Transferring calls - can Freeswitch instruct the up-stream provider to do the transfer?
by Clive Lansink 18 Nov '13

18 Nov '13

Hi everyone. I have an IVR application written in Python that utilises Freeswitch to handle all the SIP and VOIP stuff. However in this case, Freeswitch is really only the end point and is not the client's PBX system. The client would like an option that will cause the incoming call to be transferred to a specified destination. The destination is an outside number and not a Freeswitch extension. I am playing with this now. I believe I can use the API originate command to originate a call to the specified destination and use the bridge application to bridge them. But even if that works, it would mean the incoming call is still being routed right through Freeswitch and out again to the destination. Now is there a way I can tell Freeswitch to issue like a call transfer command to the up-stream provider so Freeswitch can then hang up and the call can be handled elsewhere? In the old days, and in fact I have done this in a much earlier version of this same IVR application, I would have sent a flash to the outside world, wait for dialtone then dial the destination number, wait for ring and then hang up. Can anyone advise how would I do something like that now? Cheers. Clive Lansink Email: Clive(a)Lansink.Co.NZ Phone: +64 9 520-4242 Mobile: +64 21 663-999 Fax: +64 21 789-150

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tuning/measuring - hostile networking
by Jon Hancock 18 Nov '13

18 Nov '13

Hi list, I’m working with FreeSwitch 1.5.x in BigBlueButton 0.81 We have a tough networking problem: Calls between China and New Zealand. We’re running FreeSwitch on a Hong Kong VPS. This is presumed to be the best middle ground to have a server handle the calls. The China domestic internet is messy and there’s a lot of filtering of traffic in/out of China. To compound the problem, traffic from New Zealand up to Hong Kong isn’t the greatest. For example, ping tests between a China user and the Hong Kong server are around 400ms and ping between New Zealand and the Hong Kong server are between 250-400ms. What I’d like to do is find methods to measure/log the latency, packet loss, etc and associate this with subjective call quality. This is our first step. As a second, I have an option of having the China side of the calls coming in over a VPN from China which could improve matters. I would need to understand how to have FreeSwitch use this VPN connect for one side of the call. Also, if someone knows some good settings for codecs, etc for such a hostile environment, that might be a good starting point. Any ideas appreciated! thanks, Jon -- Jon Hancock

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FreeSWITCH 1.2.15 Released!!!
by Ken Rice 17 Nov '13

17 Nov '13

Greetings FreeSWITCHers, The FreeSWITCH Development Team is proud to announce the release of FreeSWITCH 1.2.15! Where can you find it? http://files.freeswitch.org/freeswitch-1.2.15.tar.bz2 If you are tracking the FreeSWITCH Yum or Deb repos, packages are available via normal upgrade mechanisms. Whats new you might ask? Not a lot, this is a maintenance release and includes many bug fixes. And don¹t forget to follow us on Twitter, and your other favorite social media! -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org G+ ClueCon : http://fs0.us/cluecon-gplus FB ClueCon : http://fs0.us/cluecon-fb G+ FreeSwitch : http://fs0.us/freeswitch-gplus FB FreeSWITCH : http://fs0.us/freeswitch-fb Twitter : @FreeSWITCH_WIRE irc.freenode.net #freeswitch

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SRTP WITH PROXY MODE OF FREESWITCH ??
by Abdullah 17 Nov '13

17 Nov '13

HI EVERY ONE , IS THERE ANY WAY TO ENABLED STRP WITH PROXY MODE OF FREE-SWITCH.. I HAVE FALLOWED THE WIKI LINK BUT ITS WORK ONLY EARLY NEGOTIATION MODE .. PLEASE HELP ME WHO TO ENABLE STRP USING PROXY MODE , IF I ENABLE NO AUDIO IS HEARING.. Thanks -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/SRTP-WITH-PROXY-MODE-OF-FREES… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Set channel variable lcr
by Fernando Villarroel 16 Nov '13

16 Nov '13

Hi. How i can set a channel variable from lcr select (mod_lcr) Or how i can set some variable from ${lcr_auto_route} Any idea how i can do? Enviado desde mi iPhone

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REST API call from freeswitch
by Ashwin Jain 16 Nov '13

16 Nov '13

Hi, I wanted to do a REST API call for determining the outgoing number for bridge application. My REST API call response is { - phone: "+1415800000" } How can I parse my response and get the phone number in my dialplan? I am right now using mod_curl. -- Thanks and Regards, Ashwin Jain

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