[asterisk-users] Conference calls wont traverse my trunk

2013-06-27 Thread DadoMaker
My conference call wont go thru my SIP trunk. I may be missing a dialplan configuration setting as my PCM phone to phone calls go over the (GSM) tunk. The server with the conference: exten = 5777,1,GoTo(conf-confDemo,join,1) [conf-confDemo] exten = join,1,ConfBridge(confDemo/S/1) The server

[asterisk-users] recommendations for RJ-11 surge supressors?

2013-06-27 Thread Eric Cooper
I'd like to protect my expensive Digium FXO cards from spikes on my three incoming PSTN lines. Does anyone have any recommendations? -- Eric Cooper e c c @ c m u . e d u -- _ -- Bandwidth and Colocation Provided by

Re: [asterisk-users] recommendations for RJ-11 surge supressors?

2013-06-27 Thread A J Stiles
On Thursday 27 June 2013, Eric Cooper wrote: I'd like to protect my expensive Digium FXO cards from spikes on my three incoming PSTN lines. Does anyone have any recommendations? Does your telco not fit surge suppressors to the NTE as a matter of standard practice? Perhaps we are spoiled in

Re: [asterisk-users] Conference calls wont traverse my trunk

2013-06-27 Thread Steve Totaro
On Thu, Jun 27, 2013 at 9:53 AM, DadoMaker dadoma...@gmail.com wrote: My conference call wont go thru my SIP trunk. I may be missing a dialplan configuration setting as my PCM phone to phone calls go over the (GSM) tunk. The server with the conference: exten =

Re: [asterisk-users] recommendations for RJ-11 surge supressors?

2013-06-27 Thread Andrew Latham
On Thu, Jun 27, 2013 at 10:34 AM, A J Stiles asterisk_l...@earthshod.co.uk wrote: On Thursday 27 June 2013, Eric Cooper wrote: I'd like to protect my expensive Digium FXO cards from spikes on my three incoming PSTN lines. Does anyone have any recommendations? Does your telco not fit surge

Re: [asterisk-users] Conference calls wont traverse my trunk

2013-06-27 Thread DadoMaker
The cogerence works but doesnt go over my trunk. Its bypassing and the codec is PCM of phone. But in phone to phone call, the rtp traverses the trunk and I capture gsm packets to verify. The sip debug for conf call setup and leave: *CLI == Using SIP RTP CoS mark 5 -- Executing

Re: [asterisk-users] recommendations for RJ-11 surge supressors?

2013-06-27 Thread Dave Fullerton
On 06/27/2013 10:37 AM, Andrew Latham wrote: On Thu, Jun 27, 2013 at 10:34 AM, A J Stiles asterisk_l...@earthshod.co.uk wrote: On Thursday 27 June 2013, Eric Cooper wrote: I'd like to protect my expensive Digium FXO cards from spikes on my three incoming PSTN lines. Does anyone have any

Re: [asterisk-users] recommendations for RJ-11 surge supressors?

2013-06-27 Thread John Novack
A J Stiles wrote: On Thursday 27 June 2013, Eric Cooper wrote: I'd like to protect my expensive Digium FXO cards from spikes on my three incoming PSTN lines. Does anyone have any recommendations? Does your telco not fit surge suppressors to the NTE as a matter of standard practice? Perhaps

Re: [asterisk-users] Use Allworx Phones With Vanila Asterisk PBX? (Jr Richardson)

2013-06-27 Thread Mc GRATH Ricardo
Hi It seems these device it works similar way as a Cisco phone, so no way to look how to configure the phone because is based on template files downloaded from main system, files are downloading by TFTP option #66 (new software configurations file etc.) when device boot up. If user need to

Re: [asterisk-users] Conference calls wont traverse my trunk

2013-06-27 Thread DadoMaker
Found a syntax err in my dialplan on the far side Asterisk config. Thanks, Dado On Thu, Jun 27, 2013 at 10:41 AM, DadoMaker dadoma...@gmail.com wrote: The cogerence works but doesnt go over my trunk. Its bypassing and the codec is PCM of phone. But in phone to phone call, the rtp traverses