Hello,
In a standard sip flow, the call goes like: sip user A --> kamailio -->
pstn --> landline user B. However, when user A has a bad internet access, the
audio is broken. So what I want is to let sip user A send a invite to kamailio
first, then kamailio send invite to user A and B's landline number through
pstn, then bridge the two call together.
I understand this can be achieved by using FREESWITCH originate and bridge
command. I've tried but there's no audio both ways, which really makes me feel
stupid of myself. So I'm wondering if this can be done with kamailio? If so,
how?
Thanks
Jesse
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