Retry-After header not being relayed

Asterisk Asterisk SIP
1

I’m using Asterisk 13.23.1 with chan_sip, and I’ve noticed that the Retry-After header does not get passed from the outgoing channel to the incoming channel in the SIP dialog - is there a reason for this? Only RFC 3261 - SIP: Session Initiation Protocol indicates that it can be used to distinguish whether a SIP 503 response is temporary or permanent, and therefore is presumably something that should be relayed to the originating channel. Also, given that it’s not happening automatically, is there a way of doing it in dialplan?

Thanks!

2

Asterisk is not a SIP proxy, nothing is “relayed” in the SIP sense. Each call leg is independent. Neither chan_sip or chan_pjsip provide such functionality to relay it back, and header access is only available on the incoming and outgoing INVITE.

3

OK, thanks for clarifying.

4

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