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chris at maxpowersoft.com Guest
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Posted: Wed Apr 15, 2009 9:02 am Post subject: [Freeswitch-users] mod_fifo uuid_transfer into mod_conferenc |
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FreeSWITCH Version 1.0.trunk (12933M)
Has anyone else run into this issue? Here is what I'm doing.
2 Callers call in (both can be internal; using a snom m3 and a snom 370
in this test).
Case 1:
When both callers dial in they are directly routed into a conference
room via a dialplan "myConference". Everything works perfectly in this
scenario.
Case 2:
Both users call into the FS server and are moved into the FIFO extension
5900. FIFO music is playing. I then from the console run a
uuid_transfer <uuid> -bleg myConference on both of these users.
When they enter into the conference room, neither user can talk nor hear
each other. If I route another caller into the room like Case 1, audio
is fine only for that unique caller.
Any ideas on this one? Extension 5900 has not been modified. I'm
wondering if there is something funny happening in mod_fifo. Or maybe
I'm just doing something incredibly silly.
myConference dialplan is really nothing more than: <action
application="conference" data="50000@default"/>
Love this application!
Regards,
Chris
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brian at freeswitch.org Guest
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Posted: Wed Apr 15, 2009 9:22 am Post subject: [Freeswitch-users] mod_fifo uuid_transfer into mod_conferenc |
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Chris, I am able to reproduce this issue so now you'll need to open a jira on this issue please follow the guidelines here http://wiki.freeswitch.org/wiki/Reporting_Bugs
On another note PLEASE do not hijack threads. You clicked reply.. cleared the body and the subject and sent the message. DO NOT DO THAT. Please click new message and input freeswitch-users@lists.freeswitch.org (freeswitch-users@lists.freeswitch.org)
Thanks,
Brian
On Apr 15, 2009, at 8:53 AM, Chris Danielson wrote:
Quote: | FreeSWITCH Version 1.0.trunk (12933M)
Has anyone else run into this issue? Here is what I'm doing.
2 Callers call in (both can be internal; using a snom m3 and a snom 370
in this test).
Case 1:
When both callers dial in they are directly routed into a conference
room via a dialplan "myConference". Everything works perfectly in this
scenario.
Case 2:
Both users call into the FS server and are moved into the FIFO extension
5900. FIFO music is playing. I then from the console run a
uuid_transfer <uuid> -bleg myConference on both of these users.
When they enter into the conference room, neither user can talk nor hear
each other. If I route another caller into the room like Case 1, audio
is fine only for that unique caller.
Any ideas on this one? Extension 5900 has not been modified. I'm
wondering if there is something funny happening in mod_fifo. Or maybe
I'm just doing something incredibly silly.
myConference dialplan is really nothing more than: <action
application="conference" data="50000@default"/>
Love this application!
Regards,
Chris
_______________________________________________
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org (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
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Brian West
brian@freeswitch.org (brian@freeswitch.org)
-- Meet us at ClueCon! http://www.cluecon.com |
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