Here is the output from a dial when starting asterisk with -vvvvv. The 1NXXNXXXXXX is actually the number not those characters FYI.
Thanks -- Executing Macro("SIP/103-a555", "dialout-trunk|1|1NXXNXXXXXX||") in new stack -- Executing GotoIf("SIP/103-a555", "1?3:2") in new stack -- Goto (macro-dialout-trunk,s,3) -- Executing Macro("SIP/103-a555", "user-callerid") in new stack -- Executing GotoIf("SIP/103-a555", "0?report") in new stack -- Executing GotoIf("SIP/103-a555", "0?start") in new stack -- Executing Set("SIP/103-a555", "REALCALLERIDNUM=103") in new stack -- Executing NoOp("SIP/103-a555", "REALCALLERIDNUM is 103") in new stack -- Executing Set("SIP/103-a555", "AMPUSER=103") in new stack -- Executing Set("SIP/103-a555", "AMPUSERCIDNAME=103") in new stack -- Executing GotoIf("SIP/103-a555", "0?report") in new stack -- Executing Set("SIP/103-a555", "CALLERID(all)=103 <103>") in new stack -- Executing NoOp("SIP/103-a555", "Using CallerID "103" <103>") in new stack -- Executing Macro("SIP/103-a555", "record-enable|103|OUT") in new stack -- Executing GotoIf("SIP/103-a555", "0 > 0?2:4") in new stack -- Goto (macro-record-enable,s,4) -- Executing AGI("SIP/103-a555", "recordingcheck|20060528-110627|1148832387.1") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck recordingcheck|20060528-110627|1148832387.1: Outbound recording not enabled -- AGI Script recordingcheck completed, returning 0 -- Executing NoOp("SIP/103-a555", "No recording needed") in new stack -- Executing Macro("SIP/103-a555", "outbound-callerid|1") in new stack -- Executing GotoIf("SIP/103-a555", "1?start") in new stack -- Goto (macro-outbound-callerid,s,3) -- Executing NoOp("SIP/103-a555", "REALCALLERIDNUM is 103") in new stack -- Executing Set("SIP/103-a555", "USEROUTCID=") in new stack -- Executing Set("SIP/103-a555", "EMERGENCYCID=") in new stack -- Executing Set("SIP/103-a555", "TRUNKOUTCID=") in new stack -- Executing GotoIf("SIP/103-a555", "1?trunkcid") in new stack -- Goto (macro-outbound-callerid,s,11) -- Executing GotoIf("SIP/103-a555", "1?usercid") in new stack -- Goto (macro-outbound-callerid,s,13) -- Executing GotoIf("SIP/103-a555", "1?report") in new stack -- Goto (macro-outbound-callerid,s,15) -- Executing NoOp("SIP/103-a555", "CallerID set to "103" <103>") in new stack -- Executing Set("SIP/103-a555", "GROUP()=OUT_1") in new stack -- Executing GotoIf("SIP/103-a555", "0?108") in new stack -- Executing Set("SIP/103-a555", "DIAL_NUMBER=1NXXNXXXXXX") in new stack -- Executing Set("SIP/103-a555", "DIAL_TRUNK=1") in new stack -- Executing AGI("SIP/103-a555", "fixlocalprefix") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix fixlocalprefix: Could not parse /etc/asterisk/localprefixes.conf -- AGI Script fixlocalprefix completed, returning 0 -- Executing Set("SIP/103-a555", "OUTNUM=1NXXNXXXXXX") in new stack -- Executing Set("SIP/103-a555", "custom=ZAP/g0") in new stack -- Executing GotoIf("SIP/103-a555", "0?16") in new stack -- Executing Dial("SIP/103-a555", "ZAP/g0/1NXXNXXXXXX|120|r") in new stack == Everyone is busy/congested at this time (1:0/0/1) -- Executing Goto("SIP/103-a555", "s-CHANUNAVAIL|1") in new stack -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1) -- Executing NoOp("SIP/103-a555", "Dial failed due to CHANUNAVAIL") in new stack -- Executing Macro("SIP/103-a555", "outisbusy|") in new stack -- Executing Playback("SIP/103-a555", "all-circuits-busy-now") in new stack -- Playing 'all-circuits-busy-now' (language 'en') -- Executing Playback("SIP/103-a555", "pls-try-call-later") in new stack -- Playing 'pls-try-call-later' (language 'en') == Spawn extension (macro-outisbusy, s, 2) exited non-zero on 'SIP/103-a555' in macro 'outisbusy' == Spawn extension (macro-outisbusy, s, 2) exited non-zero on 'SIP/103-a555' -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Sunday, May 28, 2006 5:47 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] TDM Connect to the Asterisk console with verbose turned on and try to dial. Post that output. Curt Shaffer wrote: > This is not [EMAIL PROTECTED] it is asterisk with FreePBX only. Yes the phone > line is > connected to the right port. No luck. Thanks. > > -----Original Message----- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of John Novack > Sent: Saturday, May 27, 2006 11:02 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] TDM > > > > Steve Totaro wrote: > > >> Is your machine seeing the card? /var/log/messages? Are you loading >> the zaptel drivers? modprobe zaptel, modprobe wctdm? >> >> > Would he get the ztcfg message if it were not? > Is the phone line plugged into the correct jack? > With only one module installed, the other three jacks lead to nowhere. > Also this seems to be [EMAIL PROTECTED] from the references, so perhaps > there is a context issue that the configuration files address. > AAH can really lead one down the garden path! > > John Novack > > >> Curt Shaffer wrote: >> >> >>> The TDM01B is 4 port capable but has only 1 FXO module. I'm running >>> asterisk 1.2.7 and zaptel 1.2.5. I cannot seem to get the TDM01B >>> working. When I do the zttool it shows 4/1/0. I can dial out from a >>> POTS phone up to the point that the cable plugs into the card. >>> >>> Here is my /etc/zaptel.conf >>> >>> loadzone=us >>> >>> fxsks=1 >>> >>> and here is my /etc/Zapata.conf >>> >>> [channels] >>> >>> language=en >>> >>> #include zapata_additional.conf >>> >>> context=from-zaptel >>> >>> signalling=fxs_ks >>> >>> faxdetect=incoming >>> >>> usecallerid=asreceived >>> >>> echocancel=yes >>> >>> callprogress=no >>> >>> busydetect=no >>> >>> echocancelwhenbridged=no >>> >>> echotraining=800 >>> >>> group=0 >>> >>> channel=>1 >>> >>> When I dial in Asterisk does not even show an initiation of the call. >>> When I dial out on that trunk I get all circuits busy. Ztcfg -vvv >>> shows the following >>> >>> ztcfg -vvv >>> >>> Zaptel Configuration >>> >>> ====================== >>> >>> Channel map: >>> >>> Channel 01: FXS Kewlstart (Default) (Slaves: 01) >>> >>> 1 channels configured. >>> >>> Any help would be appreciated. >>> >>> Curt >>> >> > > _______________________________________________ --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