Advice on communication between Asterisk intances

hello

I’ve got a set up like this
ITSP1 === Ast1 === Whatever === Ast2 === ITSP2

Normal call path is: ITSP1 → Ast1 → Whatever.
For forwarded calls, the path becomes: ITSP1 → Ast1 → Whatever → Ast2 → ITSP2.

For some reasons, I need to repeat in signalling between Ast2 and ITSP2, some ITSP1 to Ast1 signalling parts.
I wonder what is best way to implement this.

My main questions are

  1. Should I use in-band (teaching Whatever to repeat signalling) or out-of-band communication.

  2. Which is the best SIP header to uniquely define each call between ITSP1 and Ast1 and be easily found
    by Ast2 to pulls the data it needs.

  3. I’ve read about Homer/HEP. Could it be used for this ?

Best regards

You’re going to need to elaborate on what this means. There are limited things you can communicate, though you can add custom headers to outgoing legs.

One of these reasons, is repeating caller’s location data.

Another reason may be repeating Stir Shaken Identity header (you need to generate a new Stir Skahen Identity header and send it along the one you got previously).

I was hoping that when a SIP was forwarding a call, it would keep the inbound Call-ID value in the outbound Call-ID one.

Looking at captures, it doesn’t.

Except if I overlooked an alternative, this seems to require in-band signalling between Asterisk instances.

You could add an extra header containing the callID value

Le 24/12/2025 à 12:13, oza4h07 via Asterisk Community a écrit :

[oza4h07] oza4h07 https://community.asterisk.org/u/oza4h07
December 24

I was hoping that when a SIP was forwarding a call, it would keep the
inbound Call-ID value in the outbound Call-ID one.

Looking at captures, it doesn’t.

Except if I overlooked an alternative, this seems to require in-band
signalling between Asterisk instances.


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    In Reply To

[oza4h07] oza4h07 https://community.asterisk.org/u/oza4h07
December 23

One of these reasons, is repeating caller’s location data. Another
reason may be repeating Stir Shaken Identity header (you need to
generate a new Stir Skahen Identity header and send it along the one
you got previously).

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Daniel

Asterisk is a back to back user agent, so the sessions on the two sides are different. Call-IDs are globally unique, so MUST NOT be the the same. (Well actually, one, or both, sides might not even be SIP,)

I recently discovered RFC7329/RFC7989 that introduce a Session-ID that may match what I was looking for.

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