search for: nengelberth

...less abandoned projects. Does anyone have any recommended solutions for a CentOS 6 / Asterisk 10 "vanilla" server? Not opposed to something commercial, provided it actually works and isn't a disaster to set up. Thank you, Noah Engelberth MetaLINK Technologies System Administration nengelberth at team-meta.net<mailto:nengelberth at team-meta.net> 419-636-0999 ext 100 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120227/edf4aaa8/attachment.htm> -------------- next part ----------...

I have an Asterisk 11.5 system, using SIP Realtime and operating as a ITSP. One of my customer's endpoints is a NetVanta 7100 PBX system that has a SIP trunk connection to my Asterisk box. The NV 7100 has a public IP on it that doesn't have any NAT between it and my Asterisk system. When the customer transfers a call from one handset to a voicemail box, the NV 7100 sends a RE-INVITE to

Hopefully someone can point me in the correct direction. I had a 1.4x system die on me yesterday, while I was prepping a new machine to replace it. Took the machine on site yesterday and spent the day and part of the evening getting things working. This morning, I finished up converting my dial plan, knowing there'd be calls of things that I missed. While testing, I've noted that all

The easiest way is to just run the Dial() command to forward the call to the hard fax without ever Answer()-ing the call. Without an Answer() on the call, Asterisk can't listen for fax detection (because the call hasn't been set up and there is no audio leg yet). Thank you, Noah Engelberth -----Original Message----- From: asterisk-users-bounces at lists.digium.com

I have a pair of Asterisk 11.5.1 servers operating as a load balanced cluster, with DNS SRV records set up to weight them 60/40 relative to each other (both at priority 0). The back-end is MySQL Realtime, and everything works pretty well with the Cisco SPA phones & ATAs that represent the majority of my endpoints. I recently tried to add an iPhone with the Bria softphone application, to

I have a system running Asterisk 11.2.1 that has had a couple calls between internal extensions get "stuck open". I didn't catch the verbose log for the first one, since I generally don't verbosely log to file, but the second one shows that the call that got stuck was dialed, but the caller hung up before the called device answered. This server is running a hotdesking