On 10/10/07, Ex Vito <[EMAIL PROTECTED]> wrote: > Hi list, > > I'm evaluating a private telephony scenario of about 20 > locations - 300 phones, 50 FAX machines.
More than 1 PRI? > All other locations, small by themselves, would get SIP > phones managed by asterisk, since there is good IP > connectivity between all sites. Private network? How good? How saturated? Could be possible to just run ulaw if the quality is as good as your LAN > 1. On the locations where asterisk is installed, the > solution is "trivial"; either by connecting FAXes > to FXS ports on channelbanks or by managing > faxes with iaxmodem + Hylafax. Probably a > combination of both... Why channel banks? > 2. On the remaining locations "we have a problem" > b) T.38 is the answer to FoIP > > c) asterisk 1.2 does not support T.38 > > d) asterisk 1.4 only does T.38 passthrough, not good enough Use a VoIP provider with t.38 for your faxes... easy solution. > e) CallWeaver seems to support T.38 gatewaying, although I'd > rather move on with asterisk so as to leverage current experience > and knowledge and to keep installed base with the same software. I've been waiting for callwaver 1.2 final for a while. Tried some betas and T38 gateway didnt work even when we put a Sangoma card in the machine. Problem was on the SIP side. > [PSTN] <---PRI---> [asterisk] <---PRI---> [PRI-to-T38 GW] ... > ... <--SIP/T.38---> [T.38 ATA] <---FXS---> FAX machine Too many PRI... Try: PSTN <---PRI----> AS5300 <------SIP-----> Asterisk 1.2 PSTN <---PRI----> AS5300 <------SIP-----> Asterisk 1.4 <-----SIP----> T.38 ATA PSTN <---PRI----> AS5300 <------SIP-----> T.38 ATA > 4. Of course, I could use CallWeaver as a PRI-to-T.38 gateway... > But then again, how solid would it be ? With which ATAs ? > The CallWeaver website shows a very small amount of ATAs > confirmed to be 100% working in T.38. There's a reason why CallWeaver is beta. As much as I'd love to support their stuff. It's still in beta. > 5. Would I need to have a SIP proxy between the PRI-to-T.38 > gw and the T.38 ATAs or would they be able to talk to > each other directly ? (I'd say this would depend on the > specific equipment, but...) If that would be a requirement, > which way would you go, asterisk 1.4 ? Would SER forward > T.38 traffic ? SER is a SIP proxy. T.38 is irrelevant to it. I'd use 1.4, your setup seems pretty straightforward. You don't have a diverse population of SIP phones and locations to manage. _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
