Hi Jean-Marc,
I tried to use mixmonitor and seems that it works good. My problem is
about calls after a transfer: it seems that asterisk can completely
record a call in one file, only in case of blind transfer.
If I make an attended transfer I have 2 or more sound files which are
impossible to join.
Have you successfully recorded sound files of transfered calls in one file??
TIA
Giorgio Incantalupo
Jean-Marc Salsa wrote:
Indeed, perfect !
Thanks a lot ...
JM
On 2/17/07, *Trevor Peirce* <[EMAIL PROTECTED]
<mailto:[EMAIL PROTECTED]>> wrote:
Jean-Marc Salsa wrote:
>
> exten => s,n,Dial(SIP/[EMAIL PROTECTED],30,r
> <mailto: SIP/[EMAIL PROTECTED]
<mailto:SIP/[EMAIL PROTECTED]>,30,r>)
>
> Everything works perfectly, except when the softswitch, or the PSTN
> sends back RingBack Tone.
>
> I can see the RTP flow arriving to Asterisk,
> but, it seems that Asterisk doesn't forward it to the other party
> (next-hop).
Yes because you have the "r" in there, asterisk sends its own ringing.
If you want ringing to be heard from the PSTN, you need to leave that
option disabled.
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