- FreeSWITCH-users - FreeSWITCH Mailing Lists

mod_dingaling sending wrong port numbers?
by Tom C 19 Apr '13

19 Apr '13

I'm trying to track down why Google Voice isn't working for me. When it stopped working a couple months ago, I learned that I needed to disable the video codec in autoload_configs/dingaling.conf.xml. That fixed things for a while, but for the past month it has been failing again. In my debug log for an INCOMING call, I see this: 2013-04-19 14:30:37.154689 [DEBUG] mod_dingaling.c:1584 Send rtp Candidate mysubdomain.dyndns.com:17554 [27Pe5xE7PiNurvTC] 2013-04-19 14:30:37.154689 [DEBUG] mod_dingaling.c:1584 Send rtcp Candidate mysubdomain.dyndns.com:17555 [mnwhL1eq9SKrduUQ] 2013-04-19 14:30:37.154689 [DEBUG] mod_dingaling.c:1627 Accepted 0 of 0 rtp candidates. 2013-04-19 14:30:37.154689 [DEBUG] mod_dingaling.c:1629 Accepted 0 of 0 rtcp candidates. 2013-04-19 14:30:37.154689 [DEBUG] mod_dingaling.c:1632 Accepted 0 of 0 video_rtp candidates 2013-04-19 14:30:37.154689 [DEBUG] mod_dingaling.c:1635 Accepted 0 of 0 video_rctp candidates The domain is correct, but where is mod_dingaling getting those random port numbers? Those ports are not available on my network. Shouldn't mod_dingaling use the port range that is specified in autoload_configs/switch.conf.xml? How do I give mod_dingaling a range of ports to use? To put this in context, I'm trying to track down why Google Voice is no longer working for me. I got the "Accepted 0 of 0 candidates" errors a few months ago, but I removed the video codec from autoload_configs/dingaling.conf.xml, and the problem went away. Now those error messages are back, and I can't send or receive calls via Google Voice. Updated to the latest git on April 15th. (1.5.1b+git~20130415T194750Z~be248879ae (git be24887 2013-04-15 19:47:50Z)), and still the same problem as on the February build that I was running before.

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Jira Open Tickets and Patches (IMPORTANT INFORMATION)
by Ken Rice 19 Apr '13

19 Apr '13

For the last several weeks Ray and myself have embarked on a weekly meeting to go thru the outstanding tickets on Jira and work toward resolution on them. We have asked for reporters to provide updated information on many of them and a good number of those reporters have responded. To those that have updated their tickets as requested Thank You! To those ignoring the the requests for more information, if you have not responded to requests for more information within 14 days, your ticket will likely be closed unless you respond. This policy will be enforced on a ticket by ticket basis. With the core team making the final call. At some point in the future we will likely automate this process. Now, that being said, the goal here is not to just reduce the number of open tickets, but to make sure we are focusing efforts where we can actually do something about it. -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch

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Friday Free For All Activate
by Ken Rice 19 Apr '13

19 Apr '13

Join us for the Free For All, the Bridge is open! sip:888@conference.freeswitch.org Or use any of the DIDs K -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch

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Come see our own Ken Rice on the voice users conference today!
by Anthony Minessale 19 Apr '13

19 Apr '13

http://vuc.me #vuc on freenode Starts in 30 min. -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900

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FreeSWITCH + Raspberry Pi = SwitchPi today on the VUC Conference
by Ken Rice 19 Apr '13

19 Apr '13

Hey guys, time for a little self promotion. I will be speaking on the VUC Conference today, Topic will be running FreeSWITCH on the Raspberry PI for use as a hard phone. There will be some technical discussion as well as showing off the hardware. Join us at 12:00 Noon Eastern via http://vuc.me/live I suggest using the link to watch via YouTube. Also the 888 conference bridge will be linked for those wanting to listen. (sip:888@conference.freeswitch.org) -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch

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fusionpbx not staying logged in
by Jason Holden 19 Apr '13

19 Apr '13

I dug through the Wiki and hoping this isn't too off topic but does anyone know why with fusionpbx once installed and I log in it keeps logging out asking for the username and password once I click on an item?

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mod_fifo ongoing calls information
by Ashish gautam 19 Apr '13

19 Apr '13

Hi All, I want to get information about which members are talking to which caller in real time. How can I do that? Please throw some light. Thanks.

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State of Call Progress Analysis (CPA)
by Lynn Nielson 19 Apr '13

19 Apr '13

I am looking for comments and opinions on the current state of Call Progress Analysis (CPA) with FreeSwitch. I believe Sangoma has a solution with their Lyra product but that a current module for that product is still in development. At one time, work was being done with LumenVox, but that has stopped. There was a module (mod_woomera), part of Dialogic, but it is over two years old. And there is the VMD and AVMD modules that indicate a high processing load to run effectively. Mostly, it seems that any CPA used needs to run separate from FreeSwitch in a proxy or re-directed mode to be effective under any kind of heavy load. A few years ago, this topic was popular, but lately it has either been regulated out of existence or found to be in-effective. Is there anyone with experience using call progress analysis or opinions on the topic? Thanks, -- Lynn Nielson Green Seed Technologies

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G729 codec INCOMPATIBLE_DESTINATION
by Jimmy Chang 19 Apr '13

19 Apr '13

Hi, I used FreeSWITCH Version 1.3.17+git~20130312T033518Z~7d29a92f55 and client could talk with G729 without problem. Today I install a new FS and the version is FreeSWITCH Version 1.5.1b+git~20130417T010754Z~538d9beacd. I copied the entire conf directory in old FS to new FS. I got the problem. 2013-04-18 21:25:06.899049 [DEBUG] sofia.c:5694 Remote SDP: v=0 o=Dialogic_SIP_CCLIB 62147888 62147889 IN IP4 10.0.110.41 s=Dialogic_SIP_CCLIB c=IN IP4 10.0.110.41 t=0 0 m=audio 49152 RTP/AVP 18 101 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 2013-04-18 21:25:06.899049 [DEBUG] switch_core_media.c:2597 Audio Codec Compare [G729:18:8000:30:8000]/[G729:18:8000:20:8000] 2013-04-18 21:25:06.899049 [DEBUG] switch_core_media.c:2597 Audio Codec Compare [telephone-event:101:8000:30:0]/[G729:18:8000:20:8000] 2013-04-18 21:25:06.899049 [DEBUG] switch_core_media.c:2671 Substituting codec G729@30i@8000h 2013-04-18 21:25:06.899049 [DEBUG] switch_core_media.c:1617 No audio codec available 2013-04-18 21:25:06.899049 [DEBUG] switch_core_media.c:2752 Set 2833 dtmf send payload to 101 2013-04-18 21:25:06.899049 [NOTICE] sofia.c:5812 Hangup sofia/ivvr/9999110 [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION] Any advice? Thanks. Jimmy

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uuid_bridge causes DESTINATION_OUT_OF_ORDER with FS rev 1.2.7
by Siri MM 19 Apr '13

19 Apr '13

Hello All, I am working on migrating our call system from FreeSWITCH 1.0.6 to 1.2.7. The setup is as follows: * CallerDevice registered with Remote FreeSWITCH * CalleeDevice registered with Local FreeSWITCH * Local FreeSWITCH and Remote FreeSWITCH are in different subnets I have a C application which is connected to, and interacts with FreeSWITCH. On receiving a call from Remote FreeSWITCH on Local FreeSWITCH, the call is parked, and then a couple of Lua scripts are executed: 1. first one sends an INVITE to the callee device via freeswitch interface, which makes the callee device to ring 2. Once the call is accepted on the device, a second script executes, which calls the uuid_bridge applicaiton to bridge the parked call UUID and the callee UUID However, most of the times (it does work a few times), the call establishment fails with a DESTINATION_OUT_OF_ORDER error. This exact setup, when run with the 1.0.6 version, works without any glitches. When I peeped through the code, I see that, the call state of the callee leg, after going into HIBERNATE STATE, never recovers, because of which the call times out in the method uuid_bridge_on_soft_execute(), and throws this error. In case of the 1.0.6 version, this doesn't happen, and 'State Change CS_HIBERNATE -> CS_RESET' takes place, and the call continues. The uuid_bridge_on_hibernate() handler, for some reason, doesn't seem to be getting called. Would appreciate any inputs on how to resolve this.. I probably need to update my application to work with 1.2.7 version, but am not sure! Logs at : http://pastebin.freeswitch.org/20800 Thanks!

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