Philip Prindeville wrote:
Patrick McHardy wrote:
Philip Prindeville wrote:
Patrick McHardy wrote:
What module options are you using for the SIP helper and how is call
setup in asterisk configured (directrtpsetup, canreinvite, ...)?
For the PSTN's switch:
CanReinvite : Yes
I vaguely recall some problem in the implementation of this feature,
something with missing bridging of the RTP streams that was still
necessary under some circumstances. Might be worth to try turning
it off.
What about the module options?
http://www.amazon.de/Applied-Cryptography-Protocols-Algorithms-Source/dp/0471117099/ref=sr_1_4?ie=UTF8&s=books-intl-de&qid=1228500191&sr=1-4
The SIP helper module options.
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