Integration - works.
Problem - dialing peer to peer via Kamailio OK but with missing VARs and extension number, on dialing/transferring.
Maybe you know other way to configure Asterisk dialplan for users, registered on kamailio and alowing dial as SIP/UserNumber insted SIP/UserNumber@kamailio.host.name
Asterisk's user (peer) - registered on kamailio:
[3]
host=192.168.144.212
qualify=yes
dtmfmode=rfc2833
canreinvite=no
context=local-routing
host=dynamic
type=friend
directmedia=no
nat=no
qualify=yes
disallow=all
allow=ulaw
allow=alaw
allow=g729
call-limit=2
limitonpeers=yes
callcounter=yes
callerid=Usov Mob <3>
### Asterisk queue members:
;syntax: member => interface,[,penalty][,membername][,state_interface][,ringinuse]
member=SIP/1@sip.cloudpbx.com.ua,1,1001,SIP/1,no
member=SIP/3@sip.cloudpbx.com.ua,1,1003,SIP/1,no
...Problem #1:
Peer 9 receive clid as asterisk@sip.cloudpbx.com.ua
Need #1:
1@sip.cloudpbx.com.ua.
Problem #2:
TRANSFERERNAME=SIP/sip.cloudpbx.com.ua-000000b5
Need #2:
TRANSFERERNAME=SIP/1@sip.cloudpbx.com.ua-000000b5
Problem #3:
BLINDTRANSFER=SIP/sip.cloudpbx.com.ua-000000ba
Need #3:
BLINDTRANSFER=SIP/9@sip.cloudpbx.com.ua-000000baMain problem it is missed peer number - SIP/peer_number@sip.cloudpbx.com.ua. And asterisk@sip.cloudpbx.com.ua instead 1@sip.cloudpbx.com.ua on attended transfer.
;domain=sip.cloudpbx.com.ua ;; temporary not uses becaouse not accepting GOIP sim ports registration on asterisk with IP-address of sip proxy instead domain sip.cloudpbx.com.ua; and it's not helps to change asterisk@sip.cloudpbx.com.ua clid
Any help from all - wellcome!P.S. domain sip.cloudpbx.com.ua not exist