Astrerisk 21,queue and contact

Asterisk Asterisk SIP
1

Hello to everybody
I have a strange situation, I have one contact that are connected from more one position, for exaple `pjsip show aor 246

pjsip show aor 246

      Aor:  <Aor..............................................>  <MaxContact>
    Contact:  <Aor/ContactUri............................> <Hash....> <Status> <RTT(ms)..>
==========================================================================================

      Aor:  246                                                  2
    Contact:  246/sip:246@192.168.0.146:5060               269e3f2fc0 Avail        12.240
    Contact:  246/sip:246@192.168.2.46:5060                6bc5fccb40 Avail         7.468


 ParameterName        : ParameterValue
 =================================================
 authenticate_qualify : false
 contact              : sip:246@192.168.0.146:5060
 contact              : sip:246@192.168.2.46:5060
 default_expiration   : 3600
 mailboxes            :
 max_contacts         : 2
 maximum_expiration   : 7200
 minimum_expiration   : 60
 outbound_proxy       :
 qualify_frequency    : 550
 qualify_timeout      : 3.000000
 remove_existing      : false
 remove_unavailable   : false
 support_path         : false
 voicemail_extension  :

now, i want insert eache number in different queue but the only method is

queue add member PJSIP/246 to 502

I want that the queue 502 respond only sip:246@192.168.2.46:5060
is there a way to achieve this?

2

You can probably come up with some clever dialplan logic using a Local channel… but how do you know, in an automated fashion, which one is “right”?

If hardcoded is sufficient then you can just do:

PJSIP/246/sip:246@192.168.2.46:5060
3

thanks @jcolp, but no result
when i digit

queue add member PJSIP/246/192.168.2.46:5060 to 502
Added interface 'PJSIP/246/192.168.2.46:5060' to queue '502'

but

queue show 502
502 has 0 calls (max 5) in 'ringall' strategy (2s holdtime, 18s talktime), W:0, C:2, A:9, SL:0.0%, SL2:0.0% within 0s
   Members:
      PJSIP/246/sip:246@192.168.2.46:5060 (ringinuse disabled) (dynamic) (Invalid) has taken no calls yet (login was 394 secs ago)

and the call don’t arrive

4

You would need to set the state interface as PJSIP/246 in the command.

5

yes, I was hoping for documentation, but nothing.
the workaround used in the past was to have two different numbers, I just wanted to avoid this.