What's wrong with Asterisk?

Hi, I have an Asterisk registered to a voip provider. The proccess is simple. I call my mobile using the Asterisk, it will play the corresponding voice and when I enter the DTMF, my AgiServer will catch the data.

However, there are some flaws that I don’t know the real reason why it happens. Sometimes, when I trigger the originate call my Java application (implemented with AsteriskJava), it will not call my mobile but when I check the logs on Asterisk, the voice is playing and waiting for my DTMF. It happens sometimes but sometimes it calls my mobile. Is it on the bandwidth?

I am guessing the problem is NOT with asterisk, but either with your upstream provider OR your cellphone/cell carrier.

A couple things which would be helpful are a sip trace and an rtp debug for the time when you claim asterisk is ‘broken’.

I am guessing you will see that your carrier returned a 200 OK and the rtp stream is flowing… why it never terminated at your physical cellphone, well that may remain a mystery.

[One other thing you could try is recording the outgoing call to ‘hear’ what asterisk is hearing while waiting for DTMF input]

Hello, thanks for the input. I’ll try that out. How can I trace the sip btw? :confused:

at the asterisk CLI

CLI*>sip set debug

To Disable

CLI*>sip set debug off

If you have lots of sip traffic you can filter the SIP trace by using an IP filter

CLI*>sip set debug ip

Thanks. I tried that before but don’t know the use of it. :smiley: How about the one you mention?