Any answer? =(
2014-02-06 Alexandr Usov <blessen...@gmail.com>: > 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/ > usernum...@kamailio.host.name > > > What we have. > > > My Kamailio config: > *http://pastebin.com/p7YxsFaw <http://pastebin.com/p7YxsFaw>* > > > 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 > > > ### Asterisk dialing local peer: > ... > exten => _X,n,Dial(SIP/${EXTEN}@sip.cloudpbx.com.ua,12,tT) > ... > > > ### Asterisk attended transfer from 1 to 9 exten: > > * http://pastebin.com/k9H4vMgx <http://pastebin.com/k9H4vMgx>* > > Problem #1: > Peer 9 receive clid as aster...@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 > > > > ### Asterisk blind transfer dump log: > > *http://pastebin.com/NXqXingR <http://pastebin.com/NXqXingR>* > > Problem #3: > BLINDTRANSFER=SIP/sip.cloudpbx.com.ua-000000ba > > Need #3: > BLINDTRANSFER=SIP/9...@sip.cloudpbx.com.ua-000000ba > > > Main problem it is missed peer number - SIP/*peer_number*@ > sip.cloudpbx.com.ua. And aster...@sip.cloudpbx.com.ua instead > 1...@sip.cloudpbx.com.ua on attended transfer. > > Asterisk sip.conf domain settings: > realm=sip.cloudpbx.com.ua > fromdomain=sip.cloudpbx.com.ua > ;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 > aster...@sip.cloudpbx.com.ua clid > > > Any help from all - wellcome! > P.S. domain sip.cloudpbx.com.ua not exist > > >
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