Sponsor: VoiceMeUp - Corporate & Wholesale VoIP Services

VoIP Mailing List Archives
Mailing list archives for the VoIP community
 SearchSearch 

[Freeswitch-users] mod_fifo uuid_transfer into mod_conference audio issue


 
Post new topic   Reply to topic    VoIP Mailing List Archives Forum Index -> freeSWITCH Users
View previous topic :: View next topic  
Author Message
chris at maxpowersoft.com
Guest





PostPosted: Wed Apr 15, 2009 9:02 am    Post subject: [Freeswitch-users] mod_fifo uuid_transfer into mod_conferenc Reply with 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
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
Back to top
brian at freeswitch.org
Guest





PostPosted: Wed Apr 15, 2009 9:22 am    Post subject: [Freeswitch-users] mod_fifo uuid_transfer into mod_conferenc Reply with quote

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



Brian West
brian@freeswitch.org (brian@freeswitch.org)



-- Meet us at ClueCon! http://www.cluecon.com
Back to top
Display posts from previous:   
Post new topic   Reply to topic    VoIP Mailing List Archives Forum Index -> freeSWITCH Users All times are GMT - 5 Hours
Page 1 of 1

 
Jump to:  
You cannot post new topics in this forum
You cannot reply to topics in this forum
You cannot edit your posts in this forum
You cannot delete your posts in this forum
You cannot vote in polls in this forum


Powered by phpBB © 2001, 2005 phpBB Group

VoiceMeUp - Corporate & Wholesale VoIP Services