[Asterisk-Users] ADSI Vista/Aastra 350
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 trying to learn as much as I can about ADSI. Thanks in advance, M. Gamble ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ADSI Programming
Is there any interest in starting an ADSI list somewhere so people can help each other out? I'm trying to get started with ADSI programming as well, and can't seem to find any ADSI information anywhere. If anyone is interested in starting an ADSI discussion list, contact me off list ([EMAIL PROTECTED]) and perhaps we can start one up. Regards, M. Gamble - Original Message - From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, September 10, 2003 12:52 PM Subject: [Asterisk-Users] ADSI Programming 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 Dolphin web site and downloaded their (windows) ADSI script IDE ($499, free but crippled demo), but found that the files that it generates are binaries that will only work with another piece of software that they sell for $299. Thanks, Tim McQueen ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call routing question
I have a quick call routing question that I'm sure is simple, but for the
life of me I can't figure out.
For example, someone dials 94162384000 asterisk trys our first call route
(our sip trunk) as per the extension rule below:
exten = _9NX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
However, this call fails because 94162384000 is one of our phone lines and
our SIP gateway detects a loop and returns a SIP 503 message. Is there a
way to have asterisk stip the '9' and try it as a local extension call as if
the user didn't dial 9? I try this (see below) and it failed:
exten = _9NX,2,Dial(${EXTEN:1})
Thanks in advance, I'm sure it's a simple problem and I'm just missing
something...
Regards,
M. Gamble
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[Asterisk-Users] RTP / SIP routing issues
Greetings. I am working on setting up an asterisk server (SIP only) and am running into a few issues getting RTP working correctly. Here is our setup: SIP Client (Public IP) --- Asterisk Server (Public IP/Private IP) -- 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 ether direction. Is there someway to better debug the 'native bridge'? Going the other way (from SIP to the PSTN) I can hear the audio from the SIP client over the PSTN, but I can't hear the PSTN audio comming back to the SIP client. Is anyone running a private SIP gateway behind asterisk like in this seniario? What needs to be done to get audio going both ways? Any hints? Thanks in advance, M. Gamble ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
