[asterisk-users] Voicemails do not email through asterisk

2009-05-06 Thread Damon Brown
Hello All, I am running Asterisk 1.4.23.1 on debian lenny and having issues with it sending out voicemail emails. Let me preface with the following: 1. I have tested with sendmail and ssmtp (with valid smtp server) 2. Googled quite a bit to only find the above 3. The mail.log/err/info shows

[Asterisk-Users] TE110P hybrid configuration for data and voice

2005-09-20 Thread Damon Brown
Hello hopefully someone can answer this :) We currently have an asterisk pbx connected to a FXO channel bank to 10 pots lines. Works great. But due to increasing costs and business load, we have ordered a dedicated T1. We plan on transfering the service to the T1 and cancelling the POTlines.

[Asterisk-Users] SIP call termination on PSTN lines

2005-08-03 Thread Damon Brown
or extensions I can tell it to hang up at the line?? Thanks in advance!! -- Damon Brown Damon Brown Consulting www.technicate.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Supervised transfer over SIP to outside POTS lines

2005-07-27 Thread Damon Brown
Hello all, I am trying to complete my dial plan and have come up with an interesting situation. My configuration is set up with 12 xlite SIP clients on SUSE linux workstation. They are calling out via 10 analog lines, TE110P-rhino 24 fxo. It all works and dials out great ... but ... this

Re: R: [Asterisk-Users] Zaptel error: Unable to create channel op type'Zap'

2005-07-27 Thread Damon Brown
I worked over two days setting up a card on fc 3, I had 2.6.9 installed. No action on the TE110P at all. I downloaded the stable zaptel/asterisk CVS and found an uncompiled wx110xp driver and recompiled. It starts up great now. There are some issues with fc3 FC3 has some issues with loading

[Asterisk-Users] Supervised transfer over SIP to outside POTS lines

2005-07-27 Thread Damon Brown
Hello all, I am trying to complete my dial plan and have come up with an interesting situation. My configuration is set up with 12 xlite SIP clients on SUSE linux workstation. They are calling out via 10 analog lines, TE110P-rhino 24 fxo. It all works and dials out great ... but ... this

[Asterisk-Users] [PLEASE RESPOND] Supervised transfer over SIP to outside POTS lines

2005-07-27 Thread Damon Brown
PLEASE RESPOND IF THERE'S A SOLUTION I am trying to complete my dial plan and have come up with an interesting situation. My configuration is set up with 12 xlite SIP clients on SUSE linux workstation. They are calling out via 10 analog lines, TE110P-rhino 24 fxo. It all works and dials out