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...8cde16223cc0716e325921a8a0654@10.0.0.254' of Request 102: Match Found Mar 31 06:56:26 DEBUG[2423] chan_sip.c: Stopping retransmission on '1c8d15f37596c5f47f4c0a2055895b92@10.0.0.254' of Request 102: Match Found Mar 31 06:56:35 DEBUG[2423] chan_sip.c: Auto destroying call '4FB313D2-C406-4843-A107-ECAF7CF3D55D@10.0.0.66' Mar 31 06:57:12 DEBUG[2423] chan_sip.c: Stopping retransmission on '7207867f25aa6f330b49adc367dcd6e3@10.0.0.254' of Request 102: Match Found Thanks

Anyone else taken a look at this? Looks exactly like Asterisk (the random source files I browsed all show Digium copyright) but with autoconf. https://developer.berlios.de/projects/solid-pbx/ From the summary page: Solid PBX is a multi-platform Soft Switch Class 5 software targeted for home/corporate/operator level markets

Ok here is one for you. I know we all do the this for 911: exten => _911,1,Dial(Zap/1/911) exten => _9911,1,Dial(Zap/1/911) And this probably is more then acceptable for most of us. However I have a system setup that uses SIP for most calls and 1 POTS line. We use a "least cost" routing that uses the POTS line for local calls AND SIP when appropiate. What I want to do is

Hello List, The main issue is server selection regarding PCI bus connectivity for Asterisk solutions. Most offerings on HP Proliants, something I've been looking into, include PCI-X and/or PCI-e expansion slots. I know PCI-e is totally different from PCI and PCI-X so, for now, that's not an issue. However, regarding PCI and PCI-X, and after googling for a while and checking wikipedia

This is a follow/find me script that I can't quite get to work, asterisk wont save forward/${calleridnum} to AstDB... any comments or thoughts on how to make this work or change it to work differently are appreciated. The voice prompts to go with all playback/background extensions are commented appropriately. I hope this code is of use to some of you and any help with a perfected

Hello. Has anybody knows how run two asterisk process in one hardware? (each one with its own configuration?) Thanks Juan Salas.

I've been working with Asterisk for a little while now, and have been looking recently at my next career opportunity. It seems from searching the various job sites that the predominant VOIP technology is not the applications-based open source approach we took, but Cisco, with a really heavy emphasis on the networking (ie network engineer) aspect. If you do a job search for (VOIP or