No problem, I just want to try and trace peoples path into the community and make sure we have all the proper information at first glance. Even in a case of uncertainty, Jira is better because its a database of "issues" and we always have a "not a bug" resolution. Even in cases of mistaken configuration, it can be considered a valid issue and we can change defaults etc. Jiras are not expensive to create and easy to close, just a lot more organized. On Fri, Nov 16, 2012 at 3:24 AM, Chris Martineau < Chris.Martineau@semafone.com> wrote:
Hi, ****
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Sorry,****
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Don’t post that often and wasn’t sure if it was a configuration or known issue that somebody was already aware of and could have been a “you haven’t done this or set that type answer”?****
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I will produce the logs as requested and report as an issue on jira.****
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Regards****
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Chris****
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*From:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Anthony Minessale *Sent:* 16 November 2012 00:27 *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] uuid_media leaves behind zombie call****
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Issues should be reported on http://jira.freeswitch.org****
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You should be reproducing on git HEAD and including full logs as attachments.****
Also you need to enable the sofia sip traces.****
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console loglevel debug****
sofia loglevel all 9****
sofia global siptrace on****
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I'd like to figure out the steps you took to send this email so we can see were have continued to fail to make the policy clear,****
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On Thu, Nov 15, 2012 at 11:08 AM, Chris Martineau < Chris.Martineau@semafone.com> wrote:****
Hi,****
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I am trying to use uuid_media to switch in and out of the rtp stream and in essence it seems to work however I get some strange things happening.** **
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The first thing is that when the re-invites go out, the a and the b legs get the codec list in a different order i.e. 0 8 3 to the a leg and 8 0 3 to the b leg!****
If all the codecs are supported this forces a transcoding scenario as the a leg neg’s pcmu and the b leg pcma! As this is done via the event socket and not the dialplan I cannot see what I am supposed to set to correct this behaviour ( the calls all start in media bypass mode so no codec neg is done at call start).****
Any ideas why this should do this and how I can get round it? Setting the no transcoding variable has no effect on this.****
****
The second thing is that if you go into the media path and then going out of it again and then clear down it seems to leave the a leg channel hungup in the system. Show channels lists the channel just sitting in CS_HANGUP state. In the log it says Locked, waiting on external entities?****
Everything externally has cleared and wireshark shows a clean dialog with no resends of the BYE messages. Cleardown is from the a leg and FS has passed the BYE on to the b leg.****
If you try to kill it with uuid_kill it says that the uuid does not exist? The only way I can clear it is to restart FS which on closing it hangs around with a sofia error saying waiting for 1 session.****
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If I just go into the media path and cleardown without coming out again the bridge process seems to be broken as any BYE message does not get passed through, requiring both ends to be cleared separately. Also no UNBRIDGE event is fired in this scenario which I suppose you would expect if it wasn’t bridged, however rtp is flowing through as is signalling?****
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The same channel however is still left hanging.****
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I may be using this incorrectly or not preparing the environment however the info around this command is pretty basic and looks simple to use.****
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It is pretty critical for the application I am trying to build that this process works smoothly so any help you could offer would be greatly appreciated.****
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I have attached some debug logs.****
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Many thanks****
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Chris****
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_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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****
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-- 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****
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- 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