Sip call between old gateway and new srst

[mcusine]

Dear Community,

I am trying to implement a new Catalyst with SRST without direct output to the PSTN.
I want the extensions to register with it and for outgoing calls to go to an old SIP gateway and incoming calls to reach the old gateway as before and be sent to the new srst.
How could I implement it in the simplest way with dial peers?
i attach a simple topology picture.
Thanks and regards,

[Maren Mahoney]

I don't see a topology picture associated with your post. But the short answer to your question is to look at the configuration guides for PSTN access for SRST hosts and simply change the target of the dial-peers (that would normally be pointing to the PSTN) and have those dial-peers point to your SIP gateway. If the incoming calls during an SRST event would also ingress via the old SIP gateway, the inbound "PSTN" dial-peers would also point to the old SIP gateway.

Essentially, look for a configuration example for SIP PSTN support for SRST (Google is your friend with that) and change out the PSTN SIP target in the example for your old SIP gateway.

Give it a shot and post a draft here and we can help you tidy it up.

Maren