On Fri, 6 Nov 2009 14:43:50 +0000, Veselin K wrote: >Thank you Michael, >Any advise on how to design my setup to avoid transcoding? > >Maybe: > >Incoming: PSTN -alaw-> Asterisk -alaw-> SIP Phone >Outgoing: SIP Phone -alaw-> Asterisk -alaw-> IAX2 Provider > >Am I understanding this correctly? >As long as the phone uses the same codec as the PSTN/IAX2 providers, >then Asterisk should not need to transcode?
Right. Keep it simple. Allow only alaw in your configs and use SIP phones set to prefer alaw. Michael >Regards, >Veselin K > >On Fri, Nov 06, 2009 at 06:43:51AM -0600, Michael Graves wrote: >> On Wed, 4 Nov 2009 16:44:02 +0000, [email protected] wrote: >> >> >Hello, >> >does this sound as a good combination, mini-itx board with Atom >> >dual core 1.6ghz 2G ram and a sangoma USB? >> > >> >For a setup with PSTN for incoming and IAX2(alaw/gsm) for outgoing calls. >> > >> >- Would you say its a good choice from a hardware perspective? >> >- Roughly how many concurrent calls would one of these be able to handle? >> >> Probably as much as your bandwidth can handle. Check ont the voip wiki >> (http://www.voip-info.org) and use the search term "dimensioning." >> You'll find lots of older references to systems running at 400 MHz - 1 >> GHz passing many calls as long as they don't transcode between codecs. >> >> I myself have a little FIT-PC2 that I'm starting to use for Asterisk. >> It's basically a netbook, like the hardware you describe, but tiny and >> very low power. Ideal for a small office or home office. >> >> Michael >> -- >> Michael Graves >> mgraves<at>mstvp.com >> http://www.mgraves.org >> o713-861-4005 >> c713-201-1262 >> sip:[email protected] >> skype mjgraves >> Twitter mjgraves >> >> >> >> >> _______________________________________________ >> -- 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 > >_______________________________________________ >-- 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 > -- Michael Graves mgraves<at>mstvp.com http://www.mgraves.org o713-861-4005 c713-201-1262 sip:[email protected] skype mjgraves Twitter mjgraves _______________________________________________ -- 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
