Re: [Asterisk-Users] SIP app_queue

2003-08-03 Thread Mark Spencer
Sounds like it's not cancelling right... Post to the bug tracker and be sure to include SIP debug (and maybe some commentary about what's going on in each situation). Mark On Sat, 2 Aug 2003, Brian West wrote: I have figured out that its a problem in app_queue, could be the interaction

Re: [Asterisk-Users] SIP app_queue

2003-08-03 Thread Brian West
I opened one up but I had mistekenly selected feature instead of the correct category. I'm in the process of getting sip debug right now. I'm also going to submit one on roundrobin queue routing because its not doing anything but ringing the first person in the queue even when others are logged

[Asterisk-Users] SIP app_queue

2003-08-02 Thread Brian West
I noticed a few issues with app_queue just wanted to know if its sip related or ata186 related: Ext 111 and Ext 112 are dynamically loged into the queue via AddQueueMember. Call hits queue with fewestcalls routing. Rings ext 111 if 111 doesn't answer. It rings ext 112. If for some reason ext

Re: [Asterisk-Users] SIP app_queue

2003-08-02 Thread Brian West
I have figured out that its a problem in app_queue, could be the interaction between chan_sip and app_queue. Or the ATA is on crack. in chan_sip if I change case 501: /* Not Implemented */ if (owner) ast_queue_control(p-owner, AST_CONTROL_CONGESTION, 0); break; to: case 501: /* Not