How could write log to a file using Sipp, and How could I know every call return status, I only want to kown every call return response status, for example 200 ..

I have installed asterisk on Ubuntu Natty, When I go into asterisk CLI & type in sip reload or any SIP related commands, it says SIP command not found. Anyone had a similar problem before...

I am using Asterisk PBX to call a softphone, i use thise command : "originate SIP/100 extension 4004" , in the dialplan, I have to get the CALLERID variable, but in this case, it's always empty! P...

I knew single line comments in asterisk dial plan that is ";" e.g ;exten => s,1,Playback(project/eligibility_points_msc) but what is used for multi line comments in dial plan?

Is anybody knows , how we can Originate an external number call using PHP AGI script ?

I have an Asterisk PBX under development, that I would like to link to a Skype account if possible. The idea is that people would call a particular Skype username, and be redirected to my SIP and t...

We are working on a web phone application that can make sip calls to other devices and make PSTN calls as well. We use Asterisk 1.8 as our sip server. The SIP calls from the web phone is working fi...

I'm writing a rtp video streamer for android that reads h264 coded data from an Android local socket and packetize it. The thing is that I did it but I keep getting black frames in the client side ...

I'm trying to use sox to convert asterisk voicemails to raw audio. The encoding information of the original wav is PCM S16 LE, so I thought I would just be able to do sox msg0000.wav msg0001.raw bu...

Asterisk has Asterisk Manager Interface (AMI) and Asterisk Gateway Interface (AGI), using which one can trigger PHP scripts at certain events from Asterisk. Using the same PHP scripts can als...

I am a beginner here, but I have been dealing with this unknown issue for a while now. Please view this simple program that I have coded and explain why it will not compile for me. The thing is tha...