Hi,
I use in Production : Asterisk 1.2.9.1
We Use Asterisk as a SIP Transit Server to record centrally all the calls.
The call flow would be:
incoming calls : PSTN -> GW -SIP-> Asterisk(Record) -SIP-> Softswitch -> IP
Phone
outgoing calls : IP Phone -> Softswitch -SIP-> Asterisk(Record) -SIP-> GW
-> PSTN
Dial plan in Asterisk is quite simple:
[record]
exten => s,1,Set(CALLFILENAME=${TIMESTAMP}-${UNIQUEID})
exten => s,n,Set(CALLERID(name)=${CALLERID(name)})
exten => s,n,Set(CALLERID(number)=00${CALLERID(number)})
exten => s,n,MixMonitor(${CALLFILENAME}.WAV,b)
exten => s,n,Dial(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).
Any ideas why ?
How can I bypass this issue ?
Thanks,
Jean-Marc
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