X100P Trunk/Outbound Route not working

Hello all,

Ever since writing my first post almost two months ago, I have purchased 2 of the Wildcard TDM400 cards. Currently only one has arrived and it has FXS ports which are working great. The other TDM400 which has the FXO ports hasn’t arrived yet. In the mean time I have been trying to get my Trixbox 1.2.3 system working with a Wildcard X100P for use with my trunk line which is a PSTN/POTS/Analog based telephone line.

However, the problem I have ran into is when I go to use my handset or softphone (xlite) to dial my cell phone 9+1+304+268+yyyy, I hear a very breif ring before the message “All circuits are busy, please try your call again later.”

I went back through to check on my Trunk settings and Outbound Routing settings. Here is what I have for my trunk settings:

name: ZAP/g0
CallerID: “”
Override: OFF
Max Channels: 1
Dial rules: 1+NXXNXXXXXX

Outbound Routes

Name: 9_Outside
Password: none
PIN: none
ER Dialing: none
Intracompany: none
Dial Patterns: 9|. 9|NXXXXXX
Trunk Sequence: 1 - ZAP/g0

These seem to be correct, but yet I can’t make an outgoing call or get incoming calls. When calling from a cell, it rings 4 times and then goes to my telephone company voicemail.

When watching what happens to asterisk when trying to make a call, I see this on the console:

[root@asterisk1 ~]# asterisk -rvvvv
== Parsing ‘/etc/asterisk/asterisk.conf’: Found
== Parsing ‘/etc/asterisk/extconfig.conf’: Found
Asterisk 1.2.12.1, Copyright © 1999 - 2006 Digium, Inc. and others.
Created by Mark Spencer [email protected]
Asterisk comes with ABSOLUTELY NO WARRANTY; type ‘show warranty’ for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type ‘show license’ for details.

Connected to Asterisk 1.2.12.1 currently running on asterisk1 (pid = 4039)
Verbosity was 1 and is now 4
– Executing Macro(“SIP/101-0856a4d8”, “dialout-trunk|1|8743715|”) in new stack
– Executing Set(“SIP/101-0856a4d8”, “DIAL_TRUNK=1”) in new stack
– Executing Set(“SIP/101-0856a4d8”, “DIAL_NUMBER=8743715”) in new stack
– Executing Set(“SIP/101-0856a4d8”, “ROUTE_PASSWD=”) in new stack
– Executing Set(“SIP/101-0856a4d8”, “DIAL_TRUNK_OPTIONS=tr”) in new stack
– Executing GotoIf(“SIP/101-0856a4d8”, “1?noauth”) in new stack
– Goto (macro-dialout-trunk,s,7)
– Executing Set(“SIP/101-0856a4d8”, “GROUP()=OUT_1”) in new stack
– Executing Macro(“SIP/101-0856a4d8”, “user-callerid”) in new stack
– Executing GotoIf(“SIP/101-0856a4d8”, “0?report”) in new stack
– Executing GotoIf(“SIP/101-0856a4d8”, “0?start”) in new stack
– Executing Set(“SIP/101-0856a4d8”, “REALCALLERIDNUM=101”) in new stack
– Executing NoOp(“SIP/101-0856a4d8”, “REALCALLERIDNUM is 101”) in new stack
– Executing Set(“SIP/101-0856a4d8”, “AMPUSER=101”) in new stack
– Executing Set(“SIP/101-0856a4d8”, “AMPUSERCIDNAME=upstairs”) in new stack
– Executing GotoIf(“SIP/101-0856a4d8”, “0?report”) in new stack
– Executing Set(“SIP/101-0856a4d8”, “CALLERID(all)=upstairs <101>”) in new stack
– Executing Set(“SIP/101-0856a4d8”, “REALCALLERIDNUM=101”) in new stack
– Executing NoOp(“SIP/101-0856a4d8”, “Using CallerID “upstairs” <101>”) in new stack
– Executing Macro(“SIP/101-0856a4d8”, “record-enable|101|OUT”) in new stack
– Executing GotoIf(“SIP/101-0856a4d8”, “0 > 0?2:4”) in new stack
– Goto (macro-record-enable,s,4)
– Executing DeadAGI(“SIP/101-0856a4d8”, “recordingcheck|20061109-133436|1163097276.26”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20061109-133436|1163097276.26: Outbound recording not enabled
– AGI Script recordingcheck completed, returning 0
– Executing NoOp(“SIP/101-0856a4d8”, “No recording needed”) in new stack
– Executing GotoIf(“SIP/101-0856a4d8”, “0?skipoutcid”) in new stack
– Executing Set(“SIP/101-0856a4d8”, “DIAL_TRUNK_OPTIONS=r”) in new stack
– Executing Macro(“SIP/101-0856a4d8”, “outbound-callerid|1”) in new stack
– Executing GotoIf(“SIP/101-0856a4d8”, “1?start”) in new stack
– Goto (macro-outbound-callerid,s,3)
– Executing NoOp(“SIP/101-0856a4d8”, “REALCALLERIDNUM is 101”) in new stack
– Executing GotoIf(“SIP/101-0856a4d8”, “1?normcid”) in new stack
– Goto (macro-outbound-callerid,s,9)
– Executing Set(“SIP/101-0856a4d8”, “USEROUTCID=”) in new stack
– Executing Set(“SIP/101-0856a4d8”, “EMERGENCYCID=”) in new stack
– Executing Set(“SIP/101-0856a4d8”, “TRUNKOUTCID=”) in new stack
– Executing GotoIf(“SIP/101-0856a4d8”, “1?trunkcid”) in new stack
– Goto (macro-outbound-callerid,s,16)
– Executing GotoIf(“SIP/101-0856a4d8”, “1?usercid”) in new stack
– Goto (macro-outbound-callerid,s,18)
– Executing GotoIf(“SIP/101-0856a4d8”, “1?report”) in new stack
– Goto (macro-outbound-callerid,s,23)
– Executing NoOp(“SIP/101-0856a4d8”, “CallerID set to “upstairs” <101>”) in new stack
– Executing GotoIf(“SIP/101-0856a4d8”, “1?nomax”) in new stack
– Goto (macro-dialout-trunk,s,15)
– Executing DeadAGI(“SIP/101-0856a4d8”, “fixlocalprefix”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
fixlocalprefix: Could not open /etc/asterisk/localprefixes.conf
– AGI Script fixlocalprefix completed, returning 0
– Executing Set(“SIP/101-0856a4d8”, “OUTNUM=8743715”) in new stack
– Executing Set(“SIP/101-0856a4d8”, “custom=ZAP/g0”) in new stack
– Executing GotoIf(“SIP/101-0856a4d8”, “0?customtrunk”) in new stack
– Executing Dial(“SIP/101-0856a4d8”, “ZAP/g0/8743715|120|r”) in new stack
== Everyone is busy/congested at this time (1:0/1/0)
– Executing Goto(“SIP/101-0856a4d8”, “s-CONGESTION|1”) in new stack
– Goto (macro-dialout-trunk,s-CONGESTION,1)
– Executing NoOp(“SIP/101-0856a4d8”, “Dial failed due to CONGESTION - failing through to other trunks”) in new stack
– Executing Macro(“SIP/101-0856a4d8”, “outisbusy|”) in new stack
– Executing Playback(“SIP/101-0856a4d8”, “all-circuits-busy-now|noanswer”) in new stack
– Playing ‘all-circuits-busy-now’ (language ‘en’)
– Executing Playback(“SIP/101-0856a4d8”, “pls-try-call-later|noanswer”) in new stack
– Playing ‘pls-try-call-later’ (language ‘en’)
– Executing Macro(“SIP/101-0856a4d8”, “hangupcall”) in new stack
– Executing ResetCDR(“SIP/101-0856a4d8”, “w”) in new stack
– Executing NoCDR(“SIP/101-0856a4d8”, “”) in new stack
– Executing GotoIf(“SIP/101-0856a4d8”, “1?theend”) in new stack
– Goto (macro-hangupcall,s,6)
– Executing Wait(“SIP/101-0856a4d8”, “5”) in new stack
== Spawn extension (macro-hangupcall, s, 6) exited non-zero on ‘SIP/101-0856a4d8’ in macro ‘hangupcall’
== Spawn extension (macro-hangupcall, s, 6) exited non-zero on ‘SIP/101-0856a4d8’ in macro ‘outisbusy’
== Spawn extension (macro-hangupcall, s, 6) exited non-zero on 'SIP/101-0856a4d8’
asterisk1*CLI>

Your trunk setting should look like this:

name: ZAP/g0
CallerID: “”
Override: OFF
Max Channels: 1
Dial rules:
Outbound Dial Prefix: 1

And you outbound route:

Outbound Routes

Name: 9_Outside
Password: none
PIN: none
ER Dialing: none
Intracompany: none
Dial Patterns: 9|.
Trunk Sequence: 1 - ZAP/g0

PROBLEM SOLVED. I discovered a cut in the cable after doing a inch by inch inspection. Something had partially sliced the cable so I replaced it, all works great now!!

I am having similar issues .

The cable cut you detected was it from your the modem to the asterisk box .

Or was it from the modem to your isp/vsp (i mean provider)

I will expect your reply.

Best Regards