Hi need some help please,
kamailio v-4.2.2
asterisk 13.1
;#### sip.conf
[general]
context=incoming calls
allowguest=yes
bindport=5080
bindaddr=192.168.1.77
rtcachefriends=yes
[kamailio]
type=friend
host=192.168.1.77
port=5060
context=kamailio-trunk
canreinvite=no
insecure=port,invite
dtmfmode=rfc2833
nat=force_rport,comedia
qualify=yes
i have followed this tutorial
http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb
My DDI provider sending call to ip address via DDNS.
DDI à DDNS à kamailio & asterisk in one server [192.168.1.77].
kamailio passing DDI to asterisk, but its sending to wrong context.
If any of the sip users is registered, the calls were sending to registered user context. If no sip user is online then calls has been send to kamailio-trunk. How can I get all DDI to kamailio-trunk? How can I solve this problem?
Many thanks
Sathees
On 19 Jan 2015, at 21:42, mail@sathees.co.uk mahan.sathees@gmail.com wrote:
kamailio passing DDI to asterisk, but its sending to wrong context.
If any of the sip users is registered, the calls were sending to registered user context. If no sip user is online then calls has been send to kamailio-trunk. How can I get all DDI to kamailio-trunk? How can I solve this problem?
THis is an Asterisk problem. Turn on SIP debugging in Asterisk and Asterisk will tell you how it matches an incoming request to your list of peers and users, and selects incoming context based on that. Based on that you can fix your sip.conf to always match the proper peer and thus get the calls to the context you want.
/Olle
You may setup your file extensions.conf is wrong.
Check it.
2015-01-20 8:45 GMT+01:00 Olle E. Johansson oej@edvina.net:
On 19 Jan 2015, at 21:42, mail@sathees.co.uk mahan.sathees@gmail.com wrote:
kamailio passing DDI to asterisk, but its sending to wrong context.
If any of the sip users is registered, the calls were sending to registered user context. If no sip user is online then calls has been send to kamailio-trunk. How can I get all DDI to kamailio-trunk? How can I solve this problem?
THis is an Asterisk problem. Turn on SIP debugging in Asterisk and Asterisk will tell you how it matches an incoming request to your list of peers and users, and selects incoming context based on that. Based on that you can fix your sip.conf to always match the proper peer and thus get the calls to the context you want.
/Olle
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-
Emilio Soler Muñoz -
mail＠sathees.co.uk -
Olle E. Johansson