I have two asterisks with different ips and I made a connection between them by SIP trunk.
The problem is that when I forward a call, the call goes with the standard extension S, but I was wondering if there is a way for me to differentiate the call in another way.
Normally, if you have a tandem trunk between two Asterisks, you would use static addresses, and would specify the full SIP URL (pjsip) in the dial call:
PJSIP/endpoint/extension@endpoint-IP
or
PJSIP/endpoint/extension
to use the domain specified by endpoint.
You can actually do this with dynamic addresses - that is how ITSP outgoing calls are normally handled, but caller ID handling gets more complicated.
I think this format wont work using pjsip
https://wiki.asterisk.org/wiki/display/AST/Dialing+PJSIP+Channels
You are probably right. My fault.
We re long time chan_sip users, that 's why we think it should work too on pjsip, and me particularly don’t see a reason why this format was not added as valid one too
The answer is that it complicates the implementation greatly and can quickly make things extremely messy and brittle. To that end the specific simple implementation was done instead.
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