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...-----Original Message----- From: Andrew Joakimsen [mailto:joakimsen@gmail.com] Sent: Wednesday, December 06, 2006 6:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] MWI across multiple servers How well would NFS work in this situation? On 12/6/06, Porier, Jeremy M. < jporier@ccu.edu> wrote: We are about to deploy six Asterisk servers across the state with SIP phones at each site registering to their "local" server. However, we are centralizing voicemail at our main campus to enable the transfer of voicemails between users regard...

Looks like we will be forced to make a move on our voicemail system as Nortel has declared Meridian Mail an end of life product. Frustrating thing is that it would seem their only reason for it is so they can force our hand to move to Call Pilot. Is there any documentation and feedback out there of how people have used Asterisk as a voicemail system for a legacy PBX? We've got about

hi all, I'm trying to send text messages to Snom 300 to show the credit remaining during the call... Sending a MESSAGE directly to the phone via udp i'm able to update the text on the display... but not during the conversation. I read about AOC, but i can't find any documentation about Asterisk + SIP + AOC Have you any experience, docs or workaround to suggest? Thx Ale

Hello! Is there any easy way to use the caller ID "display info" (CALLERID(name) in Asterisk) in dialplan just as we could use the number in: exten => _X./67803287, 1, <action> I have a SIP GSM device, and when a call comes in, it passes me the caller ID like so: -- Sip message Header: From: "67803287" <sip:gsm@192.168.10.1>;tag=... -- Asterisk variables:

I have two asterisk boxes (1.2.14) connected via SIP with Polycom 501s registered to each. I set callerid name and num before sending the call from one box to another but the phone registered to the receiving server only properly shows the caller name, not the number. The number on the phone always shows as the name of the sip registration of the "calling" server. Do I have to set a

We are having intermittent problems where the person we call reports static when we place an outgoing PSTN call. Only the person called hears static, to us the conversation sounds fine. Never happens on inbound calls. It doesn't matter what channel you call from (IAX, SIP, or Zap). We have a Sangoma A108D with hardware echo cancellation with 2 PRIs to Level3 and 2 PRIs to a Nortel Option

We have several Sangoma cards that we used during a transition time in our the replacement of our legacy voice system that we no longer need. Each of them saw about a month of service and are in good working order. We'd be happy to get 70% of retail for them. They are as follows: qty 2 A104D PCI w/ on board echo cancelation - $1500 each qty 1 A102D PCI w/ on board echo cancelation - $1050