Re: [asterisk-users] Sipura phone does not ring
Sorry for my delay in answer you. S0 is how pstn line is identified into spa3000. It means that an incoming call from S0 will be forwarded to [EMAIL PROTECTED] yor must configure in sip.conf an account for the sip pstn line and a context , in extensions.conf the same context with a pattern or number for dialing. example sip.conf [100] host=dynamic type=friend context=GW-PSTN-ZEL secret=x qualify=150 authuser=100 username=GW-PSTN-ZEL accountcode=ZEL-100 port=5061 disallow=all allow=ulaw extensions.conf [GW-PSTN-ZEL] exten=s,1,Answer exten=s,2,NoOp(${CALLERID}) exten=s,3,GotoIfTime(10:00-18:00,mon-fri,*,*?HLaboral,s,1) exten=s,4,GotoIfTime(09:00-10:00,mon-fri,*,*?DesvioFax,s,1) exten=s,5,Goto(Cerrado,s,1) I Have configured the spa 3000 with S0:[EMAIL PROTECTED] when spa 3000 receive a call, is forwarded to s extension in asterisk, but before, it is very important that sip user line be registered with asterisk for making calls. I hope it will help you 2006/11/29, Larry Alkoff [EMAIL PROTECTED]: Fran when you say specify the next hop do you mean the S0 line be an extension in sip.conf or a context in extensions.conf? Or should the line simply be tacked on to my [default] context? Larry Fran Oliveira wrote: I think it is wrong. You should specify the next hop with some like this S0:[EMAIL PROTECTED] 2006/11/23, Larry Alkoff [EMAIL PROTECTED]: Problem: SPA3000 phone does not ring for incoming PSTN call although I can dial out. I set up my Sipura with the Voxilla Wizard which is pretty good but leaves out some important details. The Voxilla Wizard for Supura SPA3000 gave me a setting for PSTN Tab - Dial Plans - Dial Plan 8 (S0:66610) Should I put extension [66610] in sip.conf with a context in extensions.conf that will contain dialing instructions? Can someone please tell me what the entries under [66610] and the associated context would look like? Or just tell me how to handle this - I'm been stuck for some time with this. The Wizard was nice enough to give detailed settings for sip.conf and extensions.conf but nothing about to handle Dial Plan 8 except You'll need to enter the extension you wish to forward all incoming PSTN calls to on your Asterisk server. I don't understand how to do that. Larry -- Larry Alkoff N2LA - Austin TX -- Larry Alkoff N2LA - Austin TX Using Thunderbird on Linux ___ --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] Sipura phone does not ring
Fran when you say specify the next hop do you mean the S0 line be an extension in sip.conf or a context in extensions.conf? Or should the line simply be tacked on to my [default] context? Larry Fran Oliveira wrote: I think it is wrong. You should specify the next hop with some like this S0:[EMAIL PROTECTED] 2006/11/23, Larry Alkoff [EMAIL PROTECTED]: Problem: SPA3000 phone does not ring for incoming PSTN call although I can dial out. I set up my Sipura with the Voxilla Wizard which is pretty good but leaves out some important details. The Voxilla Wizard for Supura SPA3000 gave me a setting for PSTN Tab - Dial Plans - Dial Plan 8 (S0:66610) Should I put extension [66610] in sip.conf with a context in extensions.conf that will contain dialing instructions? Can someone please tell me what the entries under [66610] and the associated context would look like? Or just tell me how to handle this - I'm been stuck for some time with this. The Wizard was nice enough to give detailed settings for sip.conf and extensions.conf but nothing about to handle Dial Plan 8 except You'll need to enter the extension you wish to forward all incoming PSTN calls to on your Asterisk server. I don't understand how to do that. Larry -- Larry Alkoff N2LA - Austin TX -- Larry Alkoff N2LA - Austin TX Using Thunderbird on Linux ___ --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] Sipura phone does not ring
I think it is wrong. You should specify the next hop with some like this S0:[EMAIL PROTECTED] 2006/11/23, Larry Alkoff [EMAIL PROTECTED]: Problem: SPA3000 phone does not ring for incoming PSTN call although I can dial out. I set up my Sipura with the Voxilla Wizard which is pretty good but leaves out some important details. The Voxilla Wizard for Supura SPA3000 gave me a setting for PSTN Tab - Dial Plans - Dial Plan 8 (S0:66610) Should I put extension [66610] in sip.conf with a context in extensions.conf that will contain dialing instructions? Can someone please tell me what the entries under [66610] and the associated context would look like? Or just tell me how to handle this - I'm been stuck for some time with this. The Wizard was nice enough to give detailed settings for sip.conf and extensions.conf but nothing about to handle Dial Plan 8 except You'll need to enter the extension you wish to forward all incoming PSTN calls to on your Asterisk server. I don't understand how to do that. Larry -- Larry Alkoff N2LA - Austin TX Using Thunderbird on Linux ___ --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
[asterisk-users] Sipura phone does not ring
Problem: SPA3000 phone does not ring for incoming PSTN call although I can dial out. I set up my Sipura with the Voxilla Wizard which is pretty good but leaves out some important details. The Voxilla Wizard for Supura SPA3000 gave me a setting for PSTN Tab - Dial Plans - Dial Plan 8 (S0:66610) Should I put extension [66610] in sip.conf with a context in extensions.conf that will contain dialing instructions? Can someone please tell me what the entries under [66610] and the associated context would look like? Or just tell me how to handle this - I'm been stuck for some time with this. The Wizard was nice enough to give detailed settings for sip.conf and extensions.conf but nothing about to handle Dial Plan 8 except You'll need to enter the extension you wish to forward all incoming PSTN calls to on your Asterisk server. I don't understand how to do that. Larry -- Larry Alkoff N2LA - Austin TX Using Thunderbird on Linux ___ --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