[Asterisk-Users] Several H323 bugfixes - working SIP <-> H.323 translator

[Bartosz Jozwiak]

> Hi all,
> 
>   in an effort to create a SIP <-> H.323 translator we've found and fixed
> several problems in H.323 channel. These inlcude:
> 
> for SIP->H.323 calls
> 
> - no ringback tone 
> - ringback not related to H.323 events
> - one-way audio with Cisco CallManager
> - incorrect Caller ID
> 
> for H.323->SIP calls
> 
> - not able to establish call with Cisco IOS 12.3(4)T
> - ringback not related to SIP events
> - no support for 183 Call Progress
> - incorrect Caller ID
> 
> 
>    Please find the patches against aterisk 0.7.2 release below.
> 
> 
> M.
> 

Did you put these files to bugs.digium.com ?