Hi there,
I am using Asterisk 13.17 with PJSIP, and I have a number of SIP accounts from my provider, two currently registered in my Asterisk. Let’s say 5414 and 5415.
What I am trying to achieve is to have diaplan hints for my accounts so I can indicate line state on my SIP phone’s keys.
I did my configuration with pjsip_wizard, which for instance creates two aor’s, like this:
The thing is:
on an incoming call, the call is matched to the correct PSTN number in a sense that it is distinguished correctly in the dialplan.
But the problem is, that the call, regardleds which PSTN number is called, is matched against the first aor (?) which is observable by the fact that regardless which number is called, it gets a call id hh5414/… .
And hence in every case the hint for hh5414 is triggered, even when the number 5415 is dialed.
I suspect the problem cause could be the way pjsip_wizard creates aor entries, which look like the following:
[hh5414]
type = aor
minimum_expiration = 60
default_expiration = 585
qualify_timeout = 3.000000
support_path = false
@pjsip_wizard = hh5414
max_contacts = 0
authenticate_qualify = false
contact = sip:tel.t-online.de
maximum_expiration = 7200
qualify_frequency = 60
remove_existing = false
[hh5415]
type = aor
minimum_expiration = 60
default_expiration = 585
qualify_timeout = 3.000000
support_path = false
@pjsip_wizard = hh5415
max_contacts = 0
authenticate_qualify = false
contact = sip:tel.t-online.de
maximum_expiration = 7200
qualify_frequency = 60
remove_existing = false
So to me it looks like both entries are the same for Asterisk and whenever a call comes in, whatever “the first one” is, is matched.
Any hints?