Hello, I have a set up with Asterisk-Kamailio as explained in http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb.
This set up has been working well for us for sometime now. We are now moving Asterisk to a stand alone server ( not installing it on the same box as asterisk ) but we are running into some issues. I have just made some changes to the Asterisk bind IP address and port and also set up Kamalio as a peer on asterisk.
When I make an extension to extension call the calls are failing from the FROMASTERISK route ( call is getting cancelled ). Can someone let me know what changes I need to make to correct the issue? Some pointer will help.
Thank you, Arun
hi: you can check by kamctl online, maybe some users are not there due to bind ..has expired.
2014-05-07 6:03 GMT+08:00 VOIP Tests kamailio.fs@gmail.com:
Hello, I have a set up with Asterisk-Kamailio as explained in http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb .
This set up has been working well for us for sometime now. We are now moving Asterisk to a stand alone server ( not installing it on the same box as asterisk ) but we are running into some issues. I have just made some changes to the Asterisk bind IP address and port and also set up Kamalio as a peer on asterisk.
When I make an extension to extension call the calls are failing from the FROMASTERISK route ( call is getting cancelled ). Can someone let me know what changes I need to make to correct the issue? Some pointer will help.
Thank you, Arun
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
I was able to resolve the issue, I had the wrong IP in the hosts file.
Regards, Arun
On Tue, May 6, 2014 at 10:18 PM, 5060 toasterisk@gmail.com wrote:
hi: you can check by kamctl online, maybe some users are not there due to bind ..has expired.
2014-05-07 6:03 GMT+08:00 VOIP Tests kamailio.fs@gmail.com:
Hello, I have a set up with Asterisk-Kamailio as explained in http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb .
This set up has been working well for us for sometime now. We are now moving Asterisk to a stand alone server ( not installing it on the same box as asterisk ) but we are running into some issues. I have just made some changes to the Asterisk bind IP address and port and also set up Kamalio as a peer on asterisk.
When I make an extension to extension call the calls are failing from the FROMASTERISK route ( call is getting cancelled ). Can someone let me know what changes I need to make to correct the issue? Some pointer will help.
Thank you, Arun
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
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
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5060 -
VOIP Tests