I'm bridging a Cisco Call Manager 3.2 system (h323 only) to an asterisk SIP setup. It works. There are issues, but that has more to do with Unity voicemail than the h323 implementations.<br><br><div><span class="gmail_quote">
On 6/15/06, <b class="gmail_sendername">Cesc</b> <<a href="mailto:cesc.santa@gmail.com">cesc.santa@gmail.com</a>> wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
Hi,<br><br>I am familiar with asterisk, though never actually tinkered with one<br>myself ... so i don't know the full extent of its capabilities.<br><br>I am facing a request to bridge a sip network and an h323 network.<br>
I would like to operate the sip with ser as the proxy and some<br>gatekeeper on the h323 side (not required though).<br>Actually, i have a few more points that may make it simpler<br>- i do not need codec negotiation: both sides are configured use
<br>the same (g711 alaw) by default.<br>- I have just a few "phones" on each side, so even "static routing"<br>can work, if that is of any help.<br>- it is not a production environment, for now. It is a demo/lab
<br><br>The question is ... can asterisk do the job?<br><br>Ideally, the bridge would be only signalling-wise (rtp to be direct<br>end-to-end). But, if someone had bad experience with this and would<br>recommend to use a B2BUA approach, please, tell me.
<br><br>I don't know if it makes a difference, but most of the calls would go<br>from the H323 side to the SIP side ... but i don't really want to<br>restrict SIP->H323.<br><br>Thanks a lot!<br><br>Cesc<br>_______________________________________________
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