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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