Nope, asterisk does the bridging. Asterisk can talk to SIP phones and H323 gateways/phones. It can also cross connect them.<br><br>Since I have SIP users plugged into asterisk, I have a dial plan that looks something like:
<br><br>exten => 100,1,Macro(local_sip_user,SIP/bill)<br>exten => 101,1,Macro(local_sip_user,SIP/bob)<br>exten => 102,1,Macro(local_sip_user,SIP/steve)<br>exten => _XXX,1,Macro(call_ccm,${EXTEN})<br>exten => _8XXX,1,Macro(call_ccm,${EXTEN:1})
<br><br>So, if you dial 100-102, you get a sip call, but if you dial 103, it would try to dial my CCM. If you dial 8100, it would call CCM anyway.<br><br>From the cisco side, I have some similar logic. That's pretty much it.
<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;">
So, asterisk does the bridging ... I asked on another list and the<br>answer was that asterisk could not do the job :O<br>The truth is that my setup should be fairly simple ... i do not need<br>any "cool" feature (voicemail and the like). I just need to call from
<br>one side to the other, for a reduced amount of users (so name mapping<br>could even be manual ... no problem).<br><br>Cesc<br><br>On 6/15/06, Gary Richardson <<a href="mailto:gary.richardson@gmail.com">gary.richardson@gmail.com
</a>> wrote:<br>> I'm bridging a Cisco Call Manager 3.2 system (h323 only) to an asterisk SIP<br>> setup. It works. There are issues, but that has more to do with Unity<br>> voicemail than the h323 implementations.
<br>><br>><br>> On 6/15/06, Cesc <<a href="mailto:cesc.santa@gmail.com">cesc.santa@gmail.com</a>> wrote:<br>> ><br>> 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>> _______________________________________________<br>> --Bandwidth and Colocation provided by <a href="http://Easynews.com">Easynews.com</a> --<br>><br>> Asterisk-Users mailing list<br>
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