Asterisk Voice HUB


I would like to ask if I can use Asterisk Server as a “bridge” between two ITSPs. In other words,

Service provide A send the voice traffic to Service Provider C through Service Provider B (acting as bridge/HUB). Service Provider B is connected to Service Provider A and C via SIP.

I am begginer in Asterisk and I would like ask you the following:

  1. Is it necessary to have a Dial-Plan in order to see how many calls and minutes are flowed via “SIP Bridge/HUB” between Service Provider A and C?
    2.How can be checked the voice CLI and ASR?
  2. Asterisk can send the logs to a SYSLOG server? Where this think can be configured (IP address of syslog)?
  3. How can be extracted de CDRs? Asterisk can create the csv files that contain the CDRs?

Finnaly someone can help me with a sample configuration for forwarding the traffic between two SIP networks based on the scenario described at the beginning of this message?

Thank you very much


  1. It is necessary to have a dialplan to do anything useful in Asterisk. I don’t understand the rest of the question, but suspect it has been machine translated.

  2. ${CALLERID(num)} for caller ID. Answer/Seize Ratio is not something that applies to individual calls. You would need to analyze CDRs or other logging.

  3. In the operating system’s place for configuring remote syslogging destinations.

  4. Asterisk can create CDRs, if you configure cdr.conf properly. They are not particularly useful for complex calls, for which you should use channel event logging.

I’d suggest one of:


paying for an Asterisk course;

hiring a consultant.

Thank you for answer. Meantime I found a very useful documentation.

Related to the rest of first question I wanted to say if the Dial-Plan is mandatory in order to see the traffic, which will pass the “Voice HUB”, on each destination in terms of minutes and number of calls.

Dial plans are mandatory if you want Asterisk to do anything with a call other than rejecting it. That has nothing to do with the call accounting.

On the call accounting, the more you deviate from Asterisk’s call accounting model, the more coding work you will have to do.