Problems with Callsign

Asterisk Asterisk Support
1

Hello to the forum

My Server:
Debian lenny including all updates
Asterisk 1.6
5 Snom phones
Trunk Acoount so phones calls go over the internet
Asterisk is behind a firewall and is whit a Nat Rule on a public IP

The problem what i have is on all phones:

  1. Lift the receiver and then comes my dial sound is so far OK.
  2. I choose a phone number
  3. I get my dial sound until I’m mediated and then I hear nothing,again until the other partner whom I’ve called the receiver decreases.

So I dont know is the line busy or free.

What could this be and what you need from me about the configs for help?

Thanks in advance.

Habunus

2

“mediated” and “decreases” are the wrong English words. I’m not completely sure that I can guess what they should be.

I would start with sip show history output.

Version 1.6 is not precise enough; there are significant differences between 1.6.0, 1.6.1 and 1.6.2.

It sounds like your service provider is sending 183 Progress. In later 1.6.x’s, you need to call the Progress application, before Asterisk will allow in band progress signalling through.

Note that dial tone is generated by the phone before it starts talking to Aterisk.

3

I also don’t understand “callsign” in the subject. If it is not a mistranslation, it is going to be a trade name for something, but that is not explained in the main part of the question.