generally, you either need a 3pcc b2bua or you base your service on REFER.
The ctd script in examples supports the latter, as for example seen in
serweb.
-jiri
At 04:50 AM 11/24/2003, Stephen Miles wrote:
Hi all,
We have successfully managed to get Ser to act as a proxy between our backend SIP service
and clients out on the Internet using the nathelper modules and RTP proxying. However,
one of the features of our SIP service allows you to dial a number via a web interface.
This appears to work by sending an INVITE from the SIP service out to the IP phone using
the 'Diversion' and follow-me directives. The problem we appear to be having is
that the RTP information is setup not in the INVITE but rather in an OK status message
returned when we pick up the phone to dial out. From what I understand Ser does not look
inside Status messages but only in the initial INVITE, hence when we use the web interface
to dial out the SDP location is not being altered to point at our Ser proxy and therefore
the RTP is not being proxied as we want.
We have tried various things but the issue appears to be that Ser will not look inside OK
replies to INVITE's. We might be misunderstanding things here??? By the way all SIP
messages are going through the proxy as desired... The only issue we have is with the RTP
traffic.
Can anybody help us understand how to get around this problem (if in fact it is possible
to get around it using Ser)
Cheers,
Steve
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Jiri Kuthan
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