We are Using The Asterisk® Open Source PBX asterisk 220.127.116.11 version we are also using konference(app_konference) 1.4 .
we are registering 3 extension on one linux based hardfone using pjsua 1.8 as the useragent…
the dial plan used for each hardfone is as follows.
as follows here is dial plan in extensions.conf
exten => 0011,1,Wait(0.05)
exten => 0011,2,Queue(0011)
exten => 0111,1,Dial(SIP/0111)
exten => 0211,1,Dial(SIP/0211)
exten => 0012,1,Wait(0.05)
exten => 0012,2,Queue(0012)
exten => 0112,1,Dial(SIP/0112)
exten => 0212,1,Dial(SIP/0212)
When i call from hardfone using pjsua from sip:0011 to sip:firstname.lastname@example.org
the call is confirmed on pjsua for phone1’s sip:0011 sip entity but asterisk generates new channel event of SIP/0111 or 0211 randomly
To start with does this issue lie with Asterisk or the sip usragent on the hardfone…?
If the problem is with asterisk & or the dial plan is there a resoultion…?
Does this issue
thanks in advance