Hi All Kamailio Experts,
I have configured Kamailio (kamailio 3.1.5) as media server. All things
are working fine. Now i want to use Asterisk (Asterisk 1.6) for Outbound
Calls. For this purpose i have followed the web page :
http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb.
In this page, some points are not clear for me , as given below:
(1) In case you use *sipregs* you have to create a record for each
extension where to set the 'name' to value of 'name' from *sipusers*.
The rest is populated by Asterisk from registrations.
(2) Be sure you configure Asterisk *to not authenticate* SIP requests
coming from Kamailio.
I am not sure that my local users chat is working through kamailio or
asterisk, who is used for authorization.
Any specific Web page to correct the issue will highly appreciated
according to my scenario.
Kindly guide me. Thanks in advance.
--
Best Regards,
Vijay Thakur
(Assistant Manager - Networks)
Mobile : +91 8744018065
Mail : vijay.thakur(a)loopmethods.com
Loop IT Methods Private Limited
1st Floor, B-10, Sector-7, Noida, (U.P) India
Ph: +91 120 305 3481,82 (INDIA), +1 347 468 8631 (USA), +61 390 011 178 (AUS)
Fax: +91 971 728 330
Web:
www.loopmethods.com
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