[asterisk-users] H323 <-> SIP

[tlott at gmx.net]

Hi

The communcation between an alcatel telephone switchbox and a sip phone (using asterisk h.323 implementation) isnt working fully bidirectional.

The user at the alcatel telephone switchbox can hear the user who is speaking on the sip phone but not the other way around.

Could that be a miss-configuration or a incompatibility between asterisk h.323 and pwlib/openh323?

The only allowed codec is alaw and the alcatel telephone switchbox is configured as gatekeeper.

Im using asterisk 1.2.12.1, pwlib 1.11.0 and openh323 1.19.0.1

Greetings
Tobi
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