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

[Olle E. Johansson]

17 jan 2013 kl. 18:07 skrev "Scott, Matt" <mscott at homeadvisor.com>:

> 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 have an Asterisk branch that separates peers beyond the proxy (pinetree-1.4) based on the via received headers.
I have it in production for Asterisk 1.4 in multiple places, but don't know the current state in relationship to
latest source code.

>  
> 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:

Remember that Asterisk first match incoming calls on users with the FROM header. You can separate based on the From header. But for PSTN, that doesn't really help...

If you have a limited set of profiles you can add port numbers to Asterisk peers, like have a peer on port 5060 and another on the kamailio IP on 5065. Have kamailio open both sockets and use force_socket to send from the proper one.

/O

>  
> INVITE sip:+1866NXXNXXX@<kamailio 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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