I am working with a VoiceGenie SIP VXML server which is very sensitive to SIP headers on INVITE - if the format is not just so, it cannot handle the calls.
I want to use openser to mask this problem by sending a consistent SIP INVITE that the VoiceGenie can handle.
Does anyone have any advice on how to do this?
This (simple) INVITE header works :
INVITE sip:7035470041@69.60.182.155:5060 SIP/2.0
none of these work :
INVITE sip:7035470041@69.60.182.155:5060;dtg=SIP SIP/2.0 INVITE sip:7035470041;npdi=yes;@69.60.182.155:5060;dtg=SIP SIP/2.0 INVITE sip:7035470041;cic=5119;@69.60.182.155:5060;dtg=SIP SIP/2.0 INVITE sip:7035470041;cic=5119;npdi=yes;@69.60.182.155:5060;dtg=SIP SIP/2.0
thanks,
gene
Hi Gene,
As I see, you actually want to change the first line of INVITE and not the headers..for changing the ruri you have the subst_uri() function (regexp based) in textops module: http://openser.org/docs/modules/1.0.x/textops.html#AEN141 make a regexp to get rid of the ruri params.
regards, bogdan
Gene Cohen wrote:
I am developing an application which requires the following call flow:
1. SIP Phone makes call which arrives at openser 2. Before processing the call openser connects the call to another address where the user hears a recorded message 3. When that call ends I want to connect the original SIP call as requested.
Has anyone done anything like this before?
Thanks gene
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Bogdan-Andrei Iancu -
Gene Cohen -
Juha Heinanen