I think I have it configured properly. ztcfg -vv shows it as channel 1 and zttool shows it as OK. But I can't dial out. When I try, it shows it arrive in teh right stack, but then issues the following errors: channel.c:1676 ast_request: No channel type registered for '{PSTN-1}' app_dial.c:554 dial_exec: Unable to create channel of type '{PSTN-1}' = = Everyone is busy at this time My config files are below: sip.conf [general] port = 5060 bindaddr = 0.0.0.0 disallow=all allow=gsm allow=ulaw allow=alaw allow=G723.1 context=from-sip [2000] type=friend username=2000 secret=xxxxxxxx host=dynamic mailbox=2000 extensions.conf [general] static=yes writeprotect=yes [globals] PSTN-1=Zap/1 [from-sip] exten => 2000,1,Dial(SIP/2000,20) exten => 2000,2,Voicemail(u2000) exten => 2000,102,Voicemail(b2000) exten => 2000,103,Hangup exten => 2999,1,VoicemailMain(${CALLERIDNUM}) include => to-pstn [to-pstn] exten => _1NXXNXXXXXX,1,Dial(${PSTN-1}/${EXTEN}) exten => _NXXNXXXXXX,1,Dial({PSTN-1}/1${EXTEN}) Can anyone help me out here? Thanks, Scott
On Thu, 2004-05-20 at 09:44, Pats1776 wrote:> channel.c:1676 ast_request: No channel type registered for '{PSTN-1}' > app_dial.c:554 dial_exec: Unable to create channel of type '{PSTN-1}' > = = Everyone is busy at this time > [to-pstn] > exten => _1NXXNXXXXXX,1,Dial(${PSTN-1}/${EXTEN}) > exten => _NXXNXXXXXX,1,Dial({PSTN-1}/1${EXTEN})You're missing a $ on the second dial line on {PSTN-1} -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 "In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss."
Post your zapata.conf and zaptel.conf Nik> -----Original Message----- > From: asterisk-users-admin@lists.digium.com > [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Pats1776 > Sent: Thursday, May 20, 2004 9:45 AM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] x100p card + dailing out > > > I think I have it configured properly. ztcfg -vv shows it as > channel 1 and zttool shows it as OK. But I can't dial out. > > When I try, it shows it arrive in teh right stack, but then > issues the following errors: > > channel.c:1676 ast_request: No channel type registered for > '{PSTN-1}' app_dial.c:554 dial_exec: Unable to create channel > of type '{PSTN-1}' = = Everyone is busy at this time > > My config files are below: > > sip.conf > > [general] > > port = 5060 > bindaddr = 0.0.0.0 > disallow=all > allow=gsm > allow=ulaw > allow=alaw > allow=G723.1 > context=from-sip > > [2000] > type=friend > username=2000 > secret=xxxxxxxx > host=dynamic > mailbox=2000 > > > > extensions.conf > > [general] > static=yes > writeprotect=yes > > [globals] > PSTN-1=Zap/1 > > [from-sip] > exten => 2000,1,Dial(SIP/2000,20) > exten => 2000,2,Voicemail(u2000) > exten => 2000,102,Voicemail(b2000) > exten => 2000,103,Hangup > > exten => 2999,1,VoicemailMain(${CALLERIDNUM}) > > include => to-pstn > > [to-pstn] > exten => _1NXXNXXXXXX,1,Dial(${PSTN-1}/${EXTEN}) > exten => _NXXNXXXXXX,1,Dial({PSTN-1}/1${EXTEN}) > > > Can anyone help me out here? > > Thanks, > > Scott > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > >
You might want to try removing the hyphen. It could be misinterpreting it? Might want to try simplifying things a bit too for testing purposes. Take out the PSTN-1 and put in the ZAP/1 directly into your dial plan to verify that * can access the ZAP channel correctly. -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Pats1776 Sent: Thursday, May 20, 2004 10:07 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] x100p card + dailing out Thanks for the syntax error fix, but I'm still having the same problem. Funny thing was, I never caught that syntax error because so far I was only trying with the preceding '1'. I can't seem to find this error relating to the x100p cards via google, the asterisk mailing list archives, or the wiki. Any other ideas? Scott ----- Original Message ----- From: "Eric Wieling" <eric@fnords.org> To: <asterisk-users@lists.digium.com> Sent: Thursday, May 20, 2004 10:59 AM Subject: Re: [Asterisk-Users] x100p card + dailing out> On Thu, 2004-05-20 at 09:44, Pats1776 wrote: > > channel.c:1676 ast_request: No channel type registered for > > '{PSTN-1}' app_dial.c:554 dial_exec: Unable to create channel of > > type '{PSTN-1}' = = Everyone is busy at this time [to-pstn] > > exten => _1NXXNXXXXXX,1,Dial(${PSTN-1}/${EXTEN}) > > exten => _NXXNXXXXXX,1,Dial({PSTN-1}/1${EXTEN}) > > You're missing a $ on the second dial line on {PSTN-1} > > -- > Eric Wieling * BTEL Consulting * 504-899-1387 x2111 "In a > related story, the IRS has recently ruled that the cost of Windows > upgrades can NOT be deducted as a gambling loss." > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users_______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Yes. That is what the two lines look like. It has been the same error since those were changed to get rid of the PSTN-1 variable. Scott ----- Original Message ----- From: "Leo Ann Boon" <leo@innovax.com.sg> To: <asterisk-users@lists.digium.com> Sent: Thursday, May 20, 2004 7:24 PM Subject: Re: [Asterisk-Users] x100p card + dailing out> > >I removed the PSTN-1 variable reference and started referencing it asZap/1> >and also ZAP/1, without any difference - same errors. > > > >I believe the hyphen you were talking about was the one in PSTN-1. > > > > > > > > > Sanity check, make sure the lines look like this: > > [to-pstn] > exten => _1NXXNXXXXXX,1,Dial(Zap/1/${EXTEN}) > exten => _NXXNXXXXXX,1,Dial(Zap/1/1${EXTEN}) > > Then, make a call and post the messages from Asterisk. > > Cheers > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users