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...-> Nortel CSG (Internal IP) <--> PSTN So far we have SIP to SIP working through Asterisk without any problems (using various sip clients). When I call from the PSTN to the CSG, here is what I see in the asterisk console: -- Executing Dial("SIP/10.10.100.40:5060", "SIP/mgamble") in new stack -- Called mgamble -- SIP/mgamble-7fdd is ringing -- SIP/mgamble-7fdd answered SIP/10.10.100.40:5060 -- Attempting native bridge of SIP/10.10.100.40:5060 and SIP/mgamble-7fdd The SIP/mgamble extention rings, however, when I pick up the phone I get no audio in et...

Hi I need a small PBX for use on the move. This means that outbound calls will need to be made over the cell phone network. Assuming a small hardware PBX with a spare mini-PCI slot or a USB slot then what hardware options do I have to get an outbound cellular channel? Options need to be rock solid, so no bluetooth to a cell phone kind of solutions need apply. Can any of the 3G usb

http://bugzilla.mindrot.org/show_bug.cgi?id=657 dtucker at zip.com.au changed: What |Removed |Added ---------------------------------------------------------------------------- Summary|Priv seperation causes |Priv seperation causes |segfault |setreuid error ------- Additional Comments From dtucker at

Hello Everyone, About a month ago, someone put a question to the list about which ADSI spec to purchase from Telcordia. I looked in the archives, and it appears that this question was never answered, so I'll put it to the list in a slightly different manner: Do I need to purchase the Telcordia specs in order to learn how to write my own ADSI scripts? If so, which one? I found the Black

I have ADSI working on my Aastra (Vista/Nortel) 350 phone and everything is working fine. However, I want the asterisk.adsi to load into the 'self-load' slot but can't figure out what the correct FDN for doing this is. Does anyone know the right FDN for the SL slot on these phones? Also, does anyone have any cool/interesting ADSI scripts they wouldn't mind sharing? I'm

I have a quick call routing question that I'm sure is simple, but for the life of me I can't figure out. For example, someone dials 94162384000 asterisk trys our first call route (our sip trunk) as per the extension rule below: exten => _9NXXXXXXXXX,1,Dial(SIP/${EXTEN:1}@10.10.100.40) However, this call fails because 94162384000 is one of our phone lines and our SIP gateway detects a