similar to: TA register to Asterisk and getting down after notify msg, why?

Not sure that's an accurate subject line, but... I've started looking at Asterisk as a possibility for a small PBX here. I'm thinking of an ISDN (BRI) card for connection to the telco, with some analogue converters (Sipura) for existing phones too. I'm pretty sure that the functionality I want is all there (and then some!) but am not clear on one point. At present, I'm

I have a channel bank (TA750) and a PRI with 30 channels connected to a TE405P, in the channel bank I have a extension to a fax machine, but it doesn't work to send or receive fax. There are any advice ? Kind regards, Miguel

I am having a problem using the Adtran 750 FXO quad card with a Groundstart trunk line. I am able to receive calls on the trunk line, however dialing out is not working. The Adtran does not seem to be doing the signaling. Has anyone used the 750 FXO card in Groundstart mode? Any special configuration issues that I should be aware of? Syed Akbar Alico Systems Inc www.alicosystems.com Tel:

Hello, with regard to the description of testing asterisk + USB ISDN TA on OpenSlug: http://www.nslu2-linux.org/wiki/OpenSlug/Asterisk and NSLU2 running Debian http://peter.korsgaard.com/articles/debian-nslu2.php I've tested the same thing (asterisk as VoIP/PSTN gateway) on an NSLU2 running Debian (which uses the CPU in little endian mode). Results so far: - driver (mISND) compiles

Has anybody had any luck getting an Adtran Total Access 616 working via the Ethernet port/MGCP to an * box? The voice lines don't seem to be coming up and I wasn't sure if I had something missing. Here's my mgcp.conf file: [general] port = 2427 bindaddr = 0.0.0.0 [10.189.189.31] context=nicktest host=10.189.189.31 canreinvite=no line => aaln/1 line => aaln/2

Hello All, I'm having a weird problem when connecting up to a TA 750 from adtran. The problem I'm seeing is that the third wire on my 66 block is behaving as the tip (or ring) for every extension. Is this indicative of a bad BCU? The only extension that works properly is extension Zap 2. Every other extension is crossed with Zap 2. Very weird. Anyone see this before? Did I get a

I am unable to get my 9133i to register with my asterisk server. I am including config files below, this a simple test network so there's nothing secret in the config files. I have upgraded the phone to the latest software version (1.4.3) I'm not sure what the problem is. I can call the phone from a softphone, but the 9133i says "no service" on the screen and I can't dial

I've been working with an adtran ta 850 hooked to a t100p pretty much all day today, and I haven't gotten past configuring zaptel.conf and zapata.conf. For some reason, when I pick up analog phone hooked up to the first module of a quad fxs card in the second slot of the ta 850, asterisk thinks that all four of the fxs modules in that card are going off hook. If I pick up a phone hooked

This is a single aastra 9113i sip phone. asterisk 1.0.9 Why do I keep seeing this in the logs ? ------------------------------------------------------ Sep ?8 18:44:25 VERBOSE[18779]: Scheduling destruction of call '9157b7d5ef36e8ec556a68e446d4ad59@192.168.1.100' in 15000 ms Sep ?8 18:44:31 DEBUG[18779]: Setting NAT on RTP to 0 Sep ?8 18:44:31 VERBOSE[18779]: 11 headers, 2 lines Sep ?8

Asterisk 1.0.3 Sayson 480i running .78 release (problem may not be Sayson specific, it's just that's what's deployed) Problem: Asterisk rejects registrations every so often even though nothing has changed either with Sayson or Asterisk configuration (and previous registrations have succeeded) SIP trace of successful registration: =============================

Hi, I tried to cable #21 with a thomson cable modem mta: <-- SIP read from 192.168.153.100:5060: INVITE sip:%2321@195.38.96.5:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.153.100;branch=z9hG4bK1aa77a586 Max-Forwards: 70 Content-Length: 258 To: "#21" <sip:%2321@195.38.96.5:5060> From: sip:15800115@195.38.96.5:5060;tag=da42eb89613306c Call-ID:

Hi, I'm trying to get the match_auth_username=yes sip configuration working. It's mentioned as an experimental new feature of 1.6.2.x. (I'm using 1.6.2.8) The sip.conf example states: ; if available, match user entry using the ; 'username' field from the authentication line ; instead of the From: field. But still I've been unable to authenticate using username

I have an Avaya 4602 IP phone that was previously working with Asterisk. It was being used elsewhere for several months, and I recently set it up again to work with Asterisk. Everything works fine for several minutes -- I am able to receive and make calls as expected. However, after a few minutes, and every few minutes thereafter, I get the following message on the console: -- Got SIP