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caller_entertained
2013 Jul 15
2
ignore 183 session progress in parallel call scenarios
Hi,
I am using asterisk 1.8.22 and have a problem when calling in parallel
several SIP endpoints and I am not sure how to resolve it. In this case
Asterisk will not bridge any audio to the caller before the 200 OK. Which
means any progress announcements, including remotely generated ringback,
are not passed back to the caller.
This behavior is completely correct, because there is no way to know