[asterisk-users] Logging into queue homed off remote system

2009-05-29 Thread Ekelund, Bryan
Greetings all, I have an interesting problem I am trying to work around. I currently have 2 * servers running in separate offices, using IAX2 to trunk between them, and queues in our main office. I'll call them Office_A and Office_B. I use Polycom 501s with a primary and secondary server, the

[asterisk-users] Connecting multiple office with multiple servers

2009-07-21 Thread Ekelund, Bryan
Greetings all, I currently manage a two-server asterisk system that connects two of our offices. Running 1.4 on CentOS 5.2 on both sides. We use Polycom 501 phone and register the phones to both systems, and use SIP peering to interconnect the two systems. We had been using IAX, but found that

Re: [asterisk-users] Connecting multiple office with multiple servers

2009-07-21 Thread Ekelund, Bryan
: [asterisk-users] Connecting multiple office with multiple servers On Tue, 2009-07-21 at 09:46 -0400, Ekelund, Bryan wrote: Greetings all, I currently manage a two-server asterisk system that connects two of our offices. Running 1.4 on CentOS 5.2 on both sides. We use Polycom 501 phone and register

[asterisk-users] Voicemail Crash - ODBC Realtime

2009-09-21 Thread Ekelund, Bryan
Running 1.4.26.2 on CentOS 5.3 Using ODBC with mysql for voicemail storage. Everytime my dialplan tries to open a connection to save a message or retrieve a stored message, Asterisk dumps out and restarts with: Asterisk ended with exit status 127 Asterisk died with code 127. Automatically

Re: [asterisk-users] Voicemail Crash - ODBC Realtime

2009-09-21 Thread Ekelund, Bryan
:03 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Voicemail Crash - ODBC Realtime On Monday 21 September 2009 01:46:55 pm Ekelund, Bryan wrote: Running 1.4.26.2 on CentOS 5.3 Using ODBC with mysql for voicemail storage. Everytime my dialplan tries

Re: [asterisk-users] Voicemail Crash - ODBC Realtime

2009-09-21 Thread Ekelund, Bryan
Realtime Ekelund, Bryan escribió: Upon further review, it is not dumping out, just restarting on its own with the same error. No .dmp in /tmp Check that you are running asterisk with the -g option. Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center

[asterisk-users] SIP over TCP/TLS for 1.4 branch

2009-11-23 Thread Ekelund, Bryan
Any word on when (or if) SIP over TCP for 1.4 branch is making an appearance? Looking to possibly do an OCS integration, but would prefer to not upgrade to 1.6 or throw OpenSer/Kamailio in the mix. Bryan Ekelund WHI Solutions, Inc. bekel...@whisolutions.com STATEMENT OF CONFIDENTIALITY:

Re: [asterisk-users] SIP over TCP/TLS for 1.4 branch

2009-11-23 Thread Ekelund, Bryan
: [asterisk-users] SIP over TCP/TLS for 1.4 branch On Mon, Nov 23, 2009 at 4:05 PM, Ekelund, Bryan bekel...@whisolutions.com wrote: Any word on when (or if) SIP over TCP for 1.4 branch is making an appearance? Looking to possibly do an OCS integration, but would prefer to not upgrade to 1.6 or throw