[SR-Users] My Kamailio + Asterisk != NOT realtime integration: TRANSFERERNAME and BLINDTRANSFER

[Alexandr Usov]

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 at 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 at sip.cloudpbx.com.ua,1,1001,SIP/1,no
member=SIP/3 at 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 asterisk at sip.cloudpbx.com.ua

Need #1:
1 at sip.cloudpbx.com.ua.

Problem #2:
TRANSFERERNAME=SIP/sip.cloudpbx.com.ua-000000b5

Need #2:
TRANSFERERNAME=SIP/1 at 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 at sip.cloudpbx.com.ua-000000ba


Main problem it is missed peer number - SIP/*peer_number*@
sip.cloudpbx.com.ua. And asterisk at sip.cloudpbx.com.ua instead
1 at 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
asterisk at sip.cloudpbx.com.ua clid


Any help from all - wellcome!
P.S. domain sip.cloudpbx.com.ua not exist
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sip-router.org/pipermail/sr-users/attachments/20140206/89ab0acc/attachment.html>