[SR-Users] Having trouble Setup Kamailio as a load balancer for Asterisk servers using module dispatcher

[Usama_Shaikh]

Hi Henning,

Thanks a lot for your response.

i searched on the internet and tried it, and got some success in this…

(and what i did was to put some of the conf dialplans from general context
to my none-dial context) and i successfully got call when i login agent and
able to listen the
“conf-onlyperson.gsm” the only person audio… and also i was able to call
8101 to 8102 (but i dont know why not able to call vice versa)

but the issue now is (i suppose) now only at asterisk side because now again
i am not able to call which was working before please help me in this and if
you have idea why the behavior is intermittent so please point out the
stuff…Preformatted text

now my extensions.conf look like this is :



and now my sip.conf is



these are the logs in my asterisk -r while i am dialing and calling not
working





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