Recording with call forwarding

Hello,
This is my problem :
I’m using asterisk 1.6.2.13 (with a manager application)
When a sip phone A is calling a sip phone B, the audio communication is registered using mixmonitor without any problems: the name of the file is A_B_datetime.wav
When the sip phone B forward all the incoming calls to a sip phone C using the message 302 (move temporarily), I’ve got my problem.
=> I can see in logs that I have 2 events calls to mixmonitor, one to create a file A_B_datetime.wav !!! and one to create a file A_C_datetime.wav

The first one contain all the audio communication, the second one is empty !!! And I would like to have only the second with the total com !
Can someone give me an explanation about this ?

Thanks by advance.
Sebastien.

Nobody can help me ?
I have two files recorded but the one with the audio com is not the good one.
Is it a bug or normal ?
Can someone test the same case and tell me if it’s the same result ?
Thanks by advance.

Sebastien

Hi

You posted a question on a friday evening, most peopel dont work over the weekend, unless they are a hobbyist

As to your prblem. Have you set the audiohook?

Ian

First, thanks for your reply.
I didn’t know this command (audiohook)
I will check this point with my colleagues and give you a response tomorrow (I’ve finish my job today).

Bye

So, I’ve tried with AudioHook but it’s not working
I’ve still have 2 files generated with MixMonitor :

  • the first one (between A and B) contains all the audio com
  • the second one ((between A and C) is empty
    I’ve read the doc about Audiohook and I’m not sure it’s what I want => if this command was ok I should have only A->B and not A->C and i want only A->C

Any more ideas ?
Thanks by advance

Edit : Audiohook allows to keep the same file when the sip phone B make a call transfer but in my case B is not doing a call transfer. A call B. B has informed Asterisk with a 302 move temporarily that call must be transfer to C so Asterisk redial to C automatically.

Hi

You need to look at your dialplan , and see where you are setting the file name, then maybe code it so that if a call is forwarded you change the file name to reflect this.

Its hard to give any more advice as you have posed no dialplanor verbose output of a call.

Thanks for your help.
I’ve done in a different way : checking all the resulting files