Summary:ASTERISK-06277: Tandem Asterisk fails
Reporter:Aryn Nakaoka (anakaoka)Labels:
Date Opened:2006-02-07 21:44:41.000-0600Date Closed:2011-06-07 14:02:47
Versions:Frequency of
Description:We currently have an Asterisk 1.0.11 Server connected to a PRI. The Main Sever also has SIP And IAX connections. The IAX connections connect to Asterisk 1.0.11 servers connected to Toshiba PBXs. The SIP Connections connect to Polycom IP Phones.

When we call out of the Toshiba PBX to a number that gets a Telco intercept message (because its busy or unavalible) - the intercept message is never heard on the Toshiba CTX.

However, if we make the same call with an IP SIP Phone off the main server (with the PRI) we will hear the intercept message.

I was reading from the other post about "ra" - we tried it in the dial string on the main PBX, we heard ringing on the Toshiba PBX but the call still did not complete. We then tried it on the Asterisk connected to the Toshiba and we could not even call out...


Currently we have a setup as such:

PSTN PRI -> Asterisk 1.0.11 #1 with TE110P (core box) -> IAX2 connection over Point to Point T1 -> Asterisk 1.0.11 #2 with TE110P -> Toshiba PBX

i have an Adtran 550 connected to the PRI and then connected to the Core Asterisk box, so we can see what is going on. when a
SIP connection is made, the "proceed message" starts to play the intercept message and you can hear it over the SIP connection. However, without a "ringing" message from the PRI, the Asterisk does not send the call over the IAX to the #2 Asterisk Server - i am looking for a way to do that.

Comments:By: Aryn Nakaoka (anakaoka) 2006-02-08 02:02:27.000-0600

After some testing with a 1.2.3 Asterisk , 1.2.3 Zaptel box, i think we found out the new drivers do avoid out issue.

we have PRI -> Asterisk 1.0.11 -> IAX -> Asterisk 1.2.3 -> PRI working just fine.

By: Tilghman Lesher (tilghman) 2006-02-08 09:44:15.000-0600

Resolved by updating to latest release