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

Alexandr Usov blessendor at gmail.com
Fri Feb 7 15:05:41 CET 2014


Any answer? =(




2014-02-06 Alexandr Usov <blessendor at 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/
> 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
>
>
>
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