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

[Billy Huddleston]

I just tried this, and it's not working for me.. I can't call a 2600 or a
CCM...  What version of OpenH323 and PWLIB did you all use?


----- Original Message ----- 
From: "Marian Durkovic" <md at bts.sk>
To: <asterisk-users at lists.digium.com>
Sent: Thursday, March 18, 2004 10:35 AM
Subject: [Asterisk-Users] Several H323 bugfixes - working SIP <-> H.323
translator


> 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.
>
>
> --------------------------------------------------------------------------
> ----                                                                  ----
> ----   Marian Durkovic                       network  manager         ----
> ----                                                                  ----
> ----   Slovak Technical University           Tel: +421 2 524 51 301   ----
> ----   Computer Centre, Nam. Slobody 17      Fax: +421 2 524 94 351   ----
> ----   812 43 Bratislava, Slovak Republic    E-mail/sip: md at bts.sk    ----
> ----                                                                  ----
> --------------------------------------------------------------------------
>