- FreeSWITCH-users - FreeSWITCH Mailing Lists

Codec negotiation
by James H Thompson 02 Apr '07

02 Apr '07

I wondering how FreeSwitch does codec negotiation in the following call flow: SIP Endpoint A ----> FreeSwitch ----> SIP Endpoint B For example, if the SIP Endpoint A offers Codecs: G.711u, G.711a, G.729 when FreeSwitch sets up the Call to SIP Endpoint B, how does it determine what list of Codecs to offer to SIP Endpoint B? What I'm looking for is for Freeswitch to always offer the same list of Codecs to SIP Endpoint B as it received from SIP Endpoint A. In this application, having the RTP packets bypass FreeSwitch and go directly between SIP Endpoint A and SIP Endpoint B is not an option since the two SIP Endpoints cannot talk to each other directly. SIP Endpoint A is in public IP space, and SIP Endpoint B is in private IP space.

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Windows/Mac/Linux GUI Coders out there?
by Brian West 02 Apr '07

02 Apr '07

Anyone good with any of these platforms please message me off list please. Thanks, Brian

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session.streamFile problem
by Tamas Cseke 28 Mar '07

28 Mar '07

Hello, I have a problem with session.streamFile js method it seems nothing happen. after i canceled the call, i got a warning "Bad Write". http://pastebin.freeswitch.org/461 Please take a look. Thanks, Tamas

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IRC dev exchange
by julian howard 27 Mar '07

27 Mar '07

What's a good IRC chat client? ____________________________________________________________________________________ Food fight? Enjoy some healthy debate in the Yahoo! Answers Food & Drink Q&A. http://answers.yahoo.com/dir/?link=list&sid=396545367

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new sofia-sip test branch.
by Michael Jerris 26 Mar '07

26 Mar '07

I have had some people asking recently if we will be upgrading to the new sofia-sip release 1.12.5. We will not due to some major issues we had identified with that release, however I have just created a new branch that will track the sofia-sip darcs repo and our svn trunk (the issues with their previous release have now been resolved). Feel free to try it out and provide some feedback, so that we can get this into our trunk. Thanks in advance. Mike http://svn.freeswitch.org/svn/freeswitch/branches/mikej/sofiasip-upgrade

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PRI cards
by Tom Chandler 26 Mar '07

26 Mar '07

Wanting to move to FreeSwitch from another switch, I was wondering if FreeSwitch has support for the Digium based PRI cards. I checked the archives, but did not find information. Thank You Tom Chandler

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spidermonkey- session done->script terminate
by Tamas Cseke 24 Mar '07

24 Mar '07

Hello, I need help using session object, please help me. "Note: While the JavaScript application will continue to execute after a hangup is detected the session object will be gone and any attempt to access it will cause the JavaScript to terminate. Save off any variables you might need for post hangup processing and use your local variables instead." How can i avoid the script termination, is it possible to somehow safely check if the session hungup? exactly i'd like to call an user, if don't answer call another, i can't use simply mod_bridgecall because i know the destinations just in runtime, i try use fetchURLHash to determine the next user, when one fails. var queue = argv; var hash = new Array(); while (true) { // fetch first agent fetchURLHash("http://localhost:8000/lockagent/" + queue + "/"+session.uuid, "hash"); var agent = hash["agent"]; //call agent //... new_session = new Session(session, "sofia/devil/" + agent + "%192.168.1.93", //dest "xml",//dialplan "default",//context "",//cid_name ""), //cid_num "",//network_addr "",//ani "",//anii "",//rdnis "",//username 10);//to // i need help in this condition // on timeout, new_session is done, and if i try access its properties, script terminate if (session.ready() && new_session.ready()) { bridge(session, new_session); fetchURLHash("http://localhost:8000/unlockagent/" + agent + "/" + session.uuid, "hash"); break; } else { // should try again, but script terminated console_log("notice","timeout, try again...\n"); //wait a bit } fetchURLHash("http://localhost:8000/unlockagent/" + agent + "/" + session.uuid, "hash"); } Thanks any help, Tamas

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mod_xml_curl
by Jonas Gauffin 24 Mar '07

24 Mar '07

Hello Thanks to Anthony for helping me out with mod_xml_curl. I've updated the wiki a bit about it, but it would be nice if one of the developers could validate the information i´ve entered. //Jonas

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Verizon Patent Bomb Goes Off
by michael papet 23 Mar '07

23 Mar '07

here's the best-so-far analysis of verizon's enforcement of painfully obvious patents against Vonage. http://ipurbia.com/2007/03/verizon-patent-analysis.html Anyone have any ideas how this would affect freeswitch?

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Re: [Freeswitch-users] Bridging dingaling to sofia
by Kieran O'Loughlin 22 Mar '07

22 Mar '07

Hey Jason, Thanks for the feedback. However my setup is a little different than you suggested. I described it in detail in an earlier email, but here's the brief synopsis. I have freeswitch installed on a machine with an Internet IP (no NAT it's right on the public internet). Freeswitch is using dingaling to connect with gtalk. I'm bridging any calls that come from gtalk to my sip provider (blueface.ie). I know calls to my blueface number work, I've been using them for a year. When I call using googletalk the sip bridge to blueface is not making it through. If I reconfigure to bridge to freeworlddialup then it does go through. My guess is that blueface is rejecting incoming calls from my freeswitch server, probably on the grounds that it doesn't have a valid registration from there and it doesn't know the ip as a valid sip provider. Thanks, Kieran.

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