AW: SIP and H323

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> -----Ursprüngliche Nachricht-----
> Von: openh323gk-users-admin@xxxxxxxxxxxxxxxxxxxxx 
> [mailto:openh323gk-users-admin@xxxxxxxxxxxxxxxxxxxxx] Im 
> Auftrag von VoIP Newbie
> Gesendet: Montag, 27. Juni 2005 11:17
> An: openh323gk-users@xxxxxxxxxxxxxxxxxxxxx
> Betreff: Re:  SIP and H323
> 
> Can you make your h323/sip signaling gateway a public source 
> that benefit others?


Sorry, no, it's not available as public source. Even if it was, it wouldn't help you much, since it also requires a special hardware box (voice boards and so on).


> 
> Cheers.
> 
> On 6/27/05, Frank Fischer <frank.fischer@xxxxxxxxxxxxxxxx> wrote:
> > Hi Megan
> > 
> > We faced a similar problem some times ago.
> > As you did, we also found out, asterisk is not suitable as 
> a h323/sip 
> > gateway (poor voice quality, h323 implementation very 
> unreliable and 
> > lacking a lot of basic features).
> > We also evaluated yate but decided against it. It misses some very 
> > important functionality/features regarding codecs. Overall 
> yate looks 
> > like a project in it's startup phase - yet not ready for a 
> productive 
> > environment (but with the potential to grow). There are also some 
> > more, comercial products but none of them were able to fullfill our 
> > requirements (like h235 or any codec to any codec translation).
> > So we ended up in developing our own h323/sip signaling 
> gateway that 
> > interoperates with gnugk and ser.
> > 
> > Cheers
> > Frank
> > 
> > 
> > > -----Ursprüngliche Nachricht-----
> > > Von: openh323gk-users-admin@xxxxxxxxxxxxxxxxxxxxx
> > > [mailto:openh323gk-users-admin@xxxxxxxxxxxxxxxxxxxxx] Im 
> Auftrag von 
> > > Megan Willigs
> > > Gesendet: Montag, 27. Juni 2005 02:58
> > > An: openh323gk-users@xxxxxxxxxxxxxxxxxxxxx
> > > Betreff:  SIP and H323
> > >
> > > Hi everybody.
> > > I'm working in a VoIP enterprise, our network was developed over 
> > > H323.
> > > At the moment we have a few SIP phones, those are 
> accessing to our 
> > > H323 network through an Asterisk server in order to interconnect 
> > > both networks.
> > > We want to migrate from Asterisk to SER, and perform the 
> translation 
> > > (between SIP and H323) with Yate, but Yate doesn't work as the 
> > > documentation refers.  Anyone have experience configuring Yate to 
> > > perform the transation?
> > > (betwen SIP and H323). Is there another option to perform this 
> > > translation?.
> > >
> > > Thanks in advance.
> > > Megan Willigs
> > >
> > >
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> > > _______________________________________________________
> > >
> > > Posting: mailto:Openh323gk-users@xxxxxxxxxxxxxxxxxxxxx
> > > Archive: 
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> > > Homepage: http://www.gnugk.org/
> > >
> > >
> > 
> > 
> > 
> > 
> > 
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> > Posting: mailto:Openh323gk-users@xxxxxxxxxxxxxxxxxxxxx
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