Atxfer callback if busy

Asterisk Asterisk Support
1

Hello,

I have kamailio+asterisk. endpoint register to kamailio, calls goes to asterisk. А => [kamailio] => [asterisk] => [kamailio] => B.

Side A call to B. B atxfer to C and hangup(blind transfer). C doesn’t take call and hangup it. Next asterisk callback to B and bridge A and B back. Thats how it work on asterisk only.

But if endpoints registered to kamailio asterisk after C hangup send wrong INVITE, without number B:

INVITE sip:192.168.8.123 SIP/2.0
Via: SIP/2.0/UDP 192.168.8.98:5060;branch=z9hG4bK3baac36b

Is it any way to fix it? Maybe someone already fix it?

2

Please provide the complete SIP trace so that we can work out whether this is a Kamaillo misconfiguration, or an Asterisk bug.