12 Oct
2017
12 Oct
'17
6:30 a.m.
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