Hi All – I’m trying to configure the following:
FreePBX 2.5.2.2
Asterisk 1.4.26.2
Digium Wildcard TE122 connected to E1 PRI
After several hours of playing around I’ve managed to get the PRI up and talking correctly. DID works fine and the test range of extensions I’ve configured are able to dial amongst themselves. The issue I’m experiencing is with dialing OUT from the system to the PSTN, which generates an ‘All circuits are busy now, please try your call again later’.
Console output produces the following with -vvvvvvvv:
— BEGIN CONSOLE OUTPUT —
– Executing [[email protected]:1] Macro(“SIP/500-0a033bf8”, “user-callerid|SKIPTTL|”) in new stack
– Executing [[email protected]:1] Set(“SIP/500-0a033bf8”, “AMPUSER=500”) in new stack
– Executing [[email protected]:2] GotoIf(“SIP/500-0a033bf8”, “0?report”) in new stack
– Executing [[email protected]:3] ExecIf(“SIP/500-0a033bf8”, “1|Set|REALCALLERIDNUM=500”) in new stack
– Executing [[email protected]:4] Set(“SIP/500-0a033bf8”, “AMPUSER=500”) in new stack
– Executing [[email protected]:5] Set(“SIP/500-0a033bf8”, “AMPUSERCIDNAME=Ext 500”) in new stack
– Executing [[email protected]:6] GotoIf(“SIP/500-0a033bf8”, “0?report”) in new stack
– Executing [[email protected]:7] Set(“SIP/500-0a033bf8”, “AMPUSERCID=500”) in new stack
– Executing [[email protected]:8] Set(“SIP/500-0a033bf8”, “CALLERID(all)=“Ext 500” <500>”) in new stack
– Executing [[email protected]:9] GotoIf(“SIP/500-0a033bf8”, “1?continue”) in new stack
– Goto (macro-user-callerid,s,18)
– Executing [[email protected]:18] NoOp(“SIP/500-0a033bf8”, “Using CallerID “Ext 500” <500>”) in new stack
– Executing [[email protected]:2] Set(“SIP/500-0a033bf8”, “_NODEST=”) in new stack
– Executing [[email protected]:3] Macro(“SIP/500-0a033bf8”, “record-enable|500|OUT|”) in new stack
– Executing [[email protected]:1] GotoIf(“SIP/500-0a033bf8”, “1?check”) in new stack
– Goto (macro-record-enable,s,4)
– Executing [[email protected]:4] AGI(“SIP/500-0a033bf8”, “recordingcheck|20090924-014924|1253720964.3”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20090924-014924|1253720964.3: Outbound recording not enabled
– AGI Script recordingcheck completed, returning 0
– Executing [[email protected]:5] MacroExit(“SIP/500-0a033bf8”, “”) in new stack
– Executing [[email protected]:4] Macro(“SIP/500-0a033bf8”, “dialout-trunk|1|92115500||”) in new stack
– Executing [[email protected]:1] Set(“SIP/500-0a033bf8”, “DIAL_TRUNK=1”) in new stack
– Executing [[email protected]:2] GosubIf(“SIP/500-0a033bf8”, “0?sub-pincheck|s|1”) in new stack
– Executing [[email protected]:3] GotoIf(“SIP/500-0a033bf8”, “0?disabletrunk|1”) in new stack
– Executing [[email protected]:4] Set(“SIP/500-0a033bf8”, “DIAL_NUMBER=92115500”) in new stack
– Executing [[email protected]:5] Set(“SIP/500-0a033bf8”, “DIAL_TRUNK_OPTIONS=tr”) in new stack
– Executing [[email protected]:6] Set(“SIP/500-0a033bf8”, “OUTBOUND_GROUP=OUT_1”) in new stack
– Executing [[email protected]:7] GotoIf(“SIP/500-0a033bf8”, “1?nomax”) in new stack
– Goto (macro-dialout-trunk,s,9)
– Executing [[email protected]:9] GotoIf(“SIP/500-0a033bf8”, “0?skipoutcid”) in new stack
– Executing [[email protected]:10] Set(“SIP/500-0a033bf8”, “DIAL_TRUNK_OPTIONS=”) in new stack
– Executing [[email protected]:11] Macro(“SIP/500-0a033bf8”, “outbound-callerid|1”) in new stack
– Executing [[email protected]:1] ExecIf(“SIP/500-0a033bf8”, “0|SetCallerPres|”) in new stack
– Executing [[email protected]:2] ExecIf(“SIP/500-0a033bf8”, “0|Set|REALCALLERIDNUM=500”) in new stack
– Executing [[email protected]:3] GotoIf(“SIP/500-0a033bf8”, “1?normcid”) in new stack
– Goto (macro-outbound-callerid,s,6)
– Executing [[email protected]:6] Set(“SIP/500-0a033bf8”, “USEROUTCID=”) in new stack
– Executing [[email protected]:7] Set(“SIP/500-0a033bf8”, “EMERGENCYCID=”) in new stack
– Executing [[email protected]id:8] Set(“SIP/500-0a033bf8”, “TRUNKOUTCID=0386623000”) in new stack
– Executing [[email protected]:9] GotoIf(“SIP/500-0a033bf8”, “1?trunkcid”) in new stack
– Goto (macro-outbound-callerid,s,12)
– Executing [[email protected]:12] ExecIf(“SIP/500-0a033bf8”, “1|Set|CALLERID(all)=0386623000”) in new stack
– Executing [[email protected]:13] ExecIf(“SIP/500-0a033bf8”, “0|Set|CALLERID(all)=”) in new stack
– Executing [[email protected]:14] ExecIf(“SIP/500-0a033bf8”, “0|SetCallerPres|prohib_passed_screen”) in new stack
– Executing [[email protected]:12] ExecIf(“SIP/500-0a033bf8”, “1|AGI|fixlocalprefix”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
> fixlocalprefix: Using pattern XXXXXXXXXX
> fixlocalprefix: Using pattern XXXXXXXX
== fixlocalprefix: Dialpattern XXXXXXXX matched. 92115500 -> 92115500
– AGI Script fixlocalprefix completed, returning 0
– Executing [[email protected]:13] Set(“SIP/500-0a033bf8”, “OUTNUM=92115500”) in new stack
– Executing [[email protected]:14] Set(“SIP/500-0a033bf8”, “custom=DAHDI/g0”) in new stack
– Executing [[email protected]:15] ExecIf(“SIP/500-0a033bf8”, “0|Set|DIAL_TRUNK_OPTIONS=M(setmusic^)”) in new stack
– Executing [[email protected]:16] Macro(“SIP/500-0a033bf8”, “dialout-trunk-predial-hook|”) in new stack
– Executing [[email protected]:1] MacroExit(“SIP/500-0a033bf8”, “”) in new stack
– Executing [[email protected]:17] GotoIf(“SIP/500-0a033bf8”, “0?bypass|1”) in new stack
– Executing [[email protected]:18] GotoIf(“SIP/500-0a033bf8”, “0?customtrunk”) in new stack
– Executing [[email protected]:19] Dial(“SIP/500-0a033bf8”, “DAHDI/g0/92115500|300|”) in new stack
== Everyone is busy/congested at this time (1:0/0/1)
– Executing [[email protected]:20] Goto(“SIP/500-0a033bf8”, “s-CHANUNAVAIL|1”) in new stack
– Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
– Executing [[email protected]:1] GotoIf(“SIP/500-0a033bf8”, “1?noreport”) in new stack
– Goto (macro-dialout-trunk,s-CHANUNAVAIL,3)
– Executing [[email protected]:3] NoOp(“SIP/500-0a033bf8”, “TRUNK Dial failed due to CHANUNAVAIL (hangupcause: 66) - failing through to other trunks”) in new stack
– Executing [[email protected]:5] Macro(“SIP/500-0a033bf8”, “outisbusy|”) in new stack
– Executing [[email protected]:1] Playback(“SIP/500-0a033bf8”, “all-circuits-busy-now|noanswer”) in new stack
– <SIP/500-0a033bf8> Playing ‘all-circuits-busy-now’ (language ‘en’)
– Executing [[email protected]:2] Playback(“SIP/500-0a033bf8”, “pls-try-call-later|noanswer”) in new stack
– <SIP/500-0a033bf8> Playing ‘pls-try-call-later’ (language ‘en’)
– Executing [[email protected]:3] Macro(“SIP/500-0a033bf8”, “hangupcall”) in new stack
– Executing [[email protected]:1] GotoIf(“SIP/500-0a033bf8”, “1?skiprg”) in new stack
– Goto (macro-hangupcall,s,4)
– Executing [[email protected]:4] GotoIf(“SIP/500-0a033bf8”, “1?skipblkvm”) in new stack
– Goto (macro-hangupcall,s,7)
– Executing [[email protected]:7] GotoIf(“SIP/500-0a033bf8”, “1?theend”) in new stack
– Goto (macro-hangupcall,s,9)
– Executing [[email protected]:9] Hangup(“SIP/500-0a033bf8”, “”) in new stack
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on ‘SIP/500-0a033bf8’ in macro ‘hangupcall’
== Spawn extension (macro-outisbusy, s, 3) exited non-zero on ‘SIP/500-0a033bf8’ in macro ‘outisbusy’
== Spawn extension (from-internal, 92115500, 5) exited non-zero on ‘SIP/500-0a033bf8’
– Executing [[email protected]:1] Macro(“SIP/500-0a033bf8”, “hangupcall”) in new stack
– Executing [[email protected]:1] GotoIf(“SIP/500-0a033bf8”, “1?skiprg”) in new stack
– Goto (macro-hangupcall,s,4)
– Executing [[email protected]:4] GotoIf(“SIP/500-0a033bf8”, “1?skipblkvm”) in new stack
– Goto (macro-hangupcall,s,7)
– Executing [[email protected]:7] GotoIf(“SIP/500-0a033bf8”, “1?theend”) in new stack
– Goto (macro-hangupcall,s,9)
– Executing [[email protected]:9] Hangup(“SIP/500-0a033bf8”, “”) in new stack
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on ‘SIP/500-0a033bf8’ in macro ‘hangupcall’
== Spawn extension (from-internal, s, 1) exited non-zero on ‘SIP/500-0a033bf8’
— END CONSOLE OUTPUT —
My configuration files are as follows:
— BEGIN /etc/zaptel.conf —
span=1,1,0,ccs,hdb3,crc4
bchan=1-15
bchan=17-31
dchan=16
loadzone=au
defaultzone=au
— END /etc/zaptel.conf —
— BEGIN /etc/asterisk/zapata.conf —
[channels]
language=en
usecallerid=yes
hidecallerid=no
callerid=asreceived
restrictcid=no
usecallingpres=yes
context=from-zaptel
switchtype=euroisdn
signalling=pri_cpe
immediate=no
overlapdial=yes
echocancel=yes
echocancelwhenbridged=yes
echotraining=800
channel=>1-15
channel=>17-31
group=0
— END /etc/asterisk/zapata.conf —
I have another FreePBX system which runs with a SIP trunk for external calling and works fine, but am really new to the Zaptel stuff. Any help/guidance you guys could offer would be greatly appreciated.
T.