You can enable this on a per-peer basis with:
sip peers:
canreinvite=yes
iax peers:
notransfer=no
Check the iax.conf.sample and sip.conf.sample files for usage.
Nitin Gupta wrote:
Hi I was wondering if its possible to make Dial command bridge two
channels and after bridging bypass asterisk, so that the voice doesn't
need to pass through my asterisk server.
For e.g., I have a user dialed in and he verifies himself and then
dials an international extension, after the call connects I don't want
the call to pass through asterisk server anymore. Is there any command
already there for any particular channel type?
Thanks,
Nitin
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