One FXS port can only handle one call. A PRI T1 gateway can handle 23 call channels. A single T1 Data line with SIP can handle about 18 call channels running G711, 37 channels running g729
Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 ---------------------------------------- From: "Nick Khamis" <[email protected]> Sent: Thursday, November 03, 2011 10:09 PM To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[email protected]> Subject: Re: [asterisk-users] DID from Direct from Telco Fair enough, In regards to the the diagram discussed earlier: Telco Lines -> Gateway T1 -> SIP Proxy -> Media Servers -> Customer I understand that a T1 Gateway that has 480 channels, can handle up to 240 calls. That is more than enough for the "Gateway T1 -> SIP Proxy" part of the diagram. I just want to make terribly sure I understand the "Telco Lines -> Gateway T1". If the Gateway T1 plugs into only 1 FXS port, is that FXS port only capable of handling 2 channels, i.e., one call? Thanks in Advnace, Nick. On Thu, Nov 3, 2011 at 9:24 PM, James Sharp <[email protected]> wrote: > On 11/03/2011 09:16 PM, Nick Khamis wrote: >> >> Hello James, >> >> Thank you so much for your response. We just purchased an AudioCodes >> MP124 for this job. And setting >> up OpenSIPS as the proxy. As I mentioned earlier, Bell Canada is the >> Telco here in Toronto. As for other >> Telcos around the world, for example Bell South in the states, is it >> possible to have them route a block of >> Florida phone numbers to our FXS port here in Canada, or do we have to >> have a T1 gateway + SIP Proxy in Florida, >> routing the calls to our setup in Toronto and vice versa? > > Routing Florida numbers up to Canada would get you charged LD per minute > fees. You can go with a provider like Level 3 or Global Crossing and they > can hand you a T1 circuit that has DIDs from many different areas in the US. > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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