It simply makes no sense to me to go from GSM (digital) to FXO/FXS (analog) and back into the PBX (digital) again. That introduces more potential for all kinds of call quality trouble.
SIP <> GSM direclty is just a better idea, if it costs a bit more. Michael On Wed, 20 Feb 2008 16:46:31 +0800, Sam Tam wrote: >Well then what you need is 8x FXO card. >That will do the job >Contact me off list if you want to know more > >-----Original Message----- >From: [EMAIL PROTECTED] >[mailto:[EMAIL PROTECTED] On Behalf Of Kev S >Sent: Tuesday, January 29, 2008 9:53 AM >To: Asterisk Users Mailing List - Non-Commercial Discussion >Subject: Re: [asterisk-users] SIP <> GSM > >With that sort of set up, If for example i get a 8 channel GSM gateway >and the X100P can i make more than 1 concurrent call though the gateway >with the X100P or does it only support 1 call at a time? > >What im looking to do is get a multi channel GSM gateway, and have the >ability to make more than 1 call at once through it. > >Thanks > >-Kev > >Sam Tam wrote: >> Try cyber-telecom.net >> May be get a X100P with a CT-G1000 or G2000 >> >> Sam >> >> -----Original Message----- >> From: [EMAIL PROTECTED] >> [mailto:[EMAIL PROTECTED] On Behalf Of Michael >Graves >> Sent: Sunday, January 20, 2008 11:40 PM >> To: Asterisk Users Mailing List - Non-Commercial Discussion >> Subject: [asterisk-users] SIP <> GSM >> >> I'd like to add a device to my Asterisk server to leverage my cellular >> account. Does anyone on-list have experience with hardware gateways vs >> using cah_bluetooth and an old cell phone? >> >> I'm considering something like http://www.mobigater.com/index.php?p=5 >> >> Thanks, >> >> Michael >> -- >> Michael Graves >> mgraves<at>mstvp.com >> blog.mgraves.org >> o713-861-4005 >> c713-201-1262 >> sip:[EMAIL PROTECTED] >> skype mjgraves >> fwd 54245 >> >> >> >> _______________________________________________ >> -- 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 >> > > >-- >This message has been scanned for viruses and >dangerous content by Mail Call antivirus software, and is >believed to be clean. > > >_______________________________________________ >-- 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 blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves fwd 54245 _______________________________________________ -- 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
