Douglas Garstang
2006-Jul-07 07:33 UTC
[asterisk-users] OT: Sipura SPA-3000 ATA Directing Calls toAsterisk
> -----Original Message----- > From: Brian Capouch [mailto:brianc@palaver.net] > Sent: Thursday, July 06, 2006 4:19 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] OT: Sipura SPA-3000 ATA Directing Calls > toAsterisk > > > 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<:s@192.168.1.1>) > > 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.Actually, it seems vastly more complicated than that. You need to create a user under the 'PSTN Tab'. This becomes the From: user. You then also need to configure a user under the 'Line 1' tab. However, this user doesn't actually get used, because under the 'User 1' tab you need to set 'Cfwd All Dest' to forward all calls to the extension on Asterisk you want to dial. If you leave this 'Cfwd All Dest' field empty, then you just get a dial tone, and it waits for a number to dial. HOWEVER, eventhough the user under the 'Line 1' tab never gets used, the 'Proxy' field still needs to have a valid host name, (foobar doesn't work) or it won't send calls to that Asterisk box. Why is it this difficult?
Dave Cotton
2006-Jul-07 08:18 UTC
[asterisk-users] OT: Sipura SPA-3000 ATA Directing Calls toAsterisk
On Fri, 2006-07-07 at 08:33 -0600, Douglas Garstang wrote:> > -----Original Message----- > > From: Brian Capouch [mailto:brianc@palaver.net] > > Sent: Thursday, July 06, 2006 4:19 PM > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Subject: Re: [asterisk-users] OT: Sipura SPA-3000 ATA Directing Calls > > toAsterisk > > > > > > 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<:s@192.168.1.1>) > > > > 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. > > Actually, it seems vastly more complicated than that. You need to create a user under the 'PSTN Tab'. This becomes the From: user. You then also need to configure a user under the 'Line 1' tab. However, this user doesn't actually get used, because under the 'User 1' tab you need to set 'Cfwd All Dest' to forward all calls to the extension on Asterisk you want to dial. If you leave this 'Cfwd All Dest' field empty, then you just get a dial tone, and it waits for a number to dial. HOWEVER, eventhough the user under the 'Line 1' tab never gets used, the 'Proxy' field still needs to have a valid host name, (foobar doesn't work) or it won't send calls to that Asterisk box. > > Why is it this difficult?Just as Brian said, I also use SPA3000 units (last count 6 of them) all the time to make and receive POTS calls and have not had any trouble at all. -- Dave Cotton <dcotton@linuxautrement.com>