[SR-Users] Invite - negotiate to asterisk as peer not kamailio

[Scott, Matt]

Pstn->Kamailio->dispatcher->asterisk


When calls come in, they are sent to asterisk, and the peer is negotiated as Kamailio.
How can I have different peer settings, for example dtmf?

I'm thinking I want the call to come in, then when the invite is sent to asterisk, the invite should be from the pstn-peer and not Kamailio, make sense?
Then I can have separate peers, with their own dtmf/audio settings, but I see the invite as:

INVITE sip:+1866NXXNXXX@<kamailio<sip:+1866NXXNXXX@%3ckamailio> ip>:5060 SIP/2.0
Via: SIP/2.0/UDP <Kamailio ip>;branch=z9hG4bKc087.12db3c91.0
Via: SIP/2.0/UDP <pstn-peer ip>:5060;branch=z9hG4bK0cB824c55100a4f2b8b
From: <sip:+1NXXNXXXXXXXX@<pstn-peer ip>:5060;isup-oli=0>;tag=gK0c731df8
To: <sip:+1NXXNXXNXXX@<kamailio ip>:5060>
Call-ID: 1024240378_131584110@<pstn-peer ip>

Any help or suggestions are very welcome..


Matt Scott







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