Re: [asterisk-users] OT: Sipura SPA-3000 ATA Directing Calls to Asterisk
Douglas Garstang wrote: Somewhat off topic... I have a Sipura SPA-3000 ATA. Calls are coming in from the FXO port. I'm trying to get all calls forwarded to Asterisk. However, (and this is hard to believe), the docs say that 1-stage calling (I presume that means no PIN is required) is not possible with FXO-VOIP calls. I somehow managed to get it to work on another SPA-3000 once before ... although I don't know how to replicate it now. Has anyone done this? Can you provide any pointers? Thanks. Use the configuration tool on www.voxilla.com - it works fine. Make sure you choose to configure for Asterisk. Also, set the dftm mode to be commented out, then the tone generation works. ;dtmfmode=rfc2833 Also, I beleive CallerID does not work properly on some firmware versions. -- Chris Mason (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 UK 44.207.183.0271 Cell: 264-235-5670 Yahoo IM: [EMAIL PROTECTED] -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: Sipura SPA-3000 ATA Directing Calls to Asterisk
Brian Capouch wrote: > Thomas Kenyon wrote: >> >> For some reason when I do this, It only works if I have callerID >> switched off, otherwise I get authentication errors. >> > > Do you know of anyway to bulk-save the contents of all the config > screens on that unit? > > If so, I could scrub the passwords and send you the config for the one > I'm using. > > I just checked; I am getting the CallerID just fine when I bring calls > into my Asterisk box via the SPA3K. > > B. > That would be fantastic, thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: Sipura SPA-3000 ATA Directing Calls to Asterisk
On Thu, 06 Jul 2006 19:34:01 -0400, Brian Capouch <[EMAIL PROTECTED]> wrote: >Thomas Kenyon wrote: >> >> For some reason when I do this, It only works if I have callerID >> switched off, otherwise I get authentication errors. >> > >Do you know of anyway to bulk-save the contents of all the config >screens on that unit? Try NewSipuraUtil at http://www.dualarrow.com > >If so, I could scrub the passwords and send you the config for the one >I'm using. > >I just checked; I am getting the CallerID just fine when I bring calls >into my Asterisk box via the SPA3K. > >B. > >-- >This message has been scanned for viruses and >dangerous content by MailScanner, and is >believed to be clean. > >___ >--Bandwidth and Colocation provided by Easynews.com -- > >asterisk-users mailing list >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: Sipura SPA-3000 ATA Directing Calls to Asterisk
Thomas Kenyon wrote: For some reason when I do this, It only works if I have callerID switched off, otherwise I get authentication errors. Do you know of anyway to bulk-save the contents of all the config screens on that unit? If so, I could scrub the passwords and send you the config for the one I'm using. I just checked; I am getting the CallerID just fine when I bring calls into my Asterisk box via the SPA3K. B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: Sipura SPA-3000 ATA Directing Calls to Asterisk
Brian Capouch wrote: > Douglas Garstang wrote: >> Somewhat off topic... >> >> I have a Sipura SPA-3000 ATA. Calls are coming in from the FXO port. >> I'm trying to get all calls forwarded to Asterisk. However, (and this >> is hard to believe), the docs say that 1-stage calling (I presume >> that means no PIN is required) is not possible with FXO-VOIP calls. I >> somehow managed to get it to work on another SPA-3000 once before ... >> although I don't know how to replicate it now. Has anyone done this? >> Can you provide any pointers? Thanks. >> > I make and take calls on the FXO port of my SPA-3000 routinely. > > On the "PSTN Line" tab of the advanced screen, I have this in the > first "Dialplan" entry: > > (S0<:[EMAIL PROTECTED]>) > > Then below, I choose that dialplan (in my case, 1) for the value of > "PSTN Caller Default DP" > > Also, of course, you have to set the SIP server, username/pw, etc. I > have mine register, and because the FXS port is already on 5060 I use > 5061. For some reason when I do this, It only works if I have callerID switched off, otherwise I get authentication errors. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: Sipura SPA-3000 ATA Directing Calls to Asterisk
Douglas Garstang wrote: Somewhat off topic... I have a Sipura SPA-3000 ATA. Calls are coming in from the FXO port. I'm trying to get all calls forwarded to Asterisk. However, (and this is hard to believe), the docs say that 1-stage calling (I presume that means no PIN is required) is not possible with FXO-VOIP calls. I somehow managed to get it to work on another SPA-3000 once before ... although I don't know how to replicate it now. Has anyone done this? Can you provide any pointers? Thanks. I make and take calls on the FXO port of my SPA-3000 routinely. On the "PSTN Line" tab of the advanced screen, I have this in the first "Dialplan" entry: (S0<:[EMAIL PROTECTED]>) Then below, I choose that dialplan (in my case, 1) for the value of "PSTN Caller Default DP" Also, of course, you have to set the SIP server, username/pw, etc. I have mine register, and because the FXS port is already on 5060 I use 5061. HTH. B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] OT: Sipura SPA-3000 ATA Directing Calls to Asterisk
Somewhat off topic... I have a Sipura SPA-3000 ATA. Calls are coming in from the FXO port. I'm trying to get all calls forwarded to Asterisk. However, (and this is hard to believe), the docs say that 1-stage calling (I presume that means no PIN is required) is not possible with FXO-VOIP calls. I somehow managed to get it to work on another SPA-3000 once before ... although I don't know how to replicate it now. Has anyone done this? Can you provide any pointers? Thanks. Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users