First time configuring FreePBX

General Help
1

Hello all,

While I may be advanced in VoIP after having been using hardware solutions like the AllWorx 10x & 6x systems, I am confident in my abilities to use TrixBox. However this morning after downloading the 1.2 release and installing it on an Athlon 2.0ghz system, I had a few questions.

Currently I have a wild card 100X FXO card installed in the system. I have a direct connect to the main NID box on the house into the WildCard 100X card. However, when I was configuring the Outgoing Lines connection area and Trunks area, I could not get the sip telephone software on my laptop to go beyond the PBX.

For example I setup extension 150 and am able to dial to other extensions that I have created. However when hitting 9 + 123 + 456 + 7890 I get two error messages, “that all circuits are busy”, or “unable to connect your call…”

Brad

2

Do you have phone plugged into the card??
Can u pick it up and dail out??

You do have phone service right, and I do not think the x100p will power more than two rings over a short wire…it takes power for each ringer and some for wiring

and to do that you would have to take the inbound from teleco disconnet it from the house go to card and from card to home phone wiring.

3

Ok go remove the stuff after thr 9|.

now fire up the shell prompt and run asterisk -rvvvv
watch the screen
and make a call …what happpens

4

This is the output from the asterisk prompt:

[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>

5

The outbound routes are the area I’m having difficulty on. I created a new outbound route to utilize the trunk line that was setup in the trunks area. All of that is recognized just fine, however I think the issue might be in the fact that my actual routing isn’t setup correct? The only thing in the outbound routes by default is the “9|.”. I wasn’t sure what I needed else to setup in order for the PBX to take advantage of the trunk.

6

Now that I’m back at my regular PC, here is my settings for the following with FreePBX:

TRUNK

Outbound Caller ID: “My name” <1234567890> (real number is there)
Maximum Channels: 1
Dial Rules: 1304|NXXXXXX
Outbound Dial Prefix:
Zap Identifier: g0

OUTBOUND ROUTE

Route Name: 9_outside
Route Pass:
Emergency Dialing:
Dial Patterns: 9|.; NXXNXXXXXX; NXXXXXX
Trunk Sequence: ZAP/g0

7

bump

Anyone have ideas?? Is the Motorola Card X100P at fault here for being a crappy card and not working right, or is this a software issue?

8

Brad,

Did you setup Outbound routing? You need to setup an outbound route that utilizes the trunk you’ve created.

Alex

On 9/14/06, bawalker <[email protected] ([email protected])> wrote:[quote]Hello all,

While I may be advanced in VoIP after having been using hardware solutions like the AllWorx 10x & 6x systems, I am confident in my abilities to use TrixBox.  However this morning after downloading the 1.2 release and installing it on an Athlon 2.0ghz system, I had a few questions.

Currently I have a wild card 100X FXO card installed in the system.  I have a direct connect to the main NID box on the house into the WildCard 100X card.  However, when I was configuring the Outgoing Lines connection area and Trunks area, I could not get the sip telephone software on my laptop to go beyond the PBX.

For example I setup extension 150 and am able to dial to other extensions that I have created.  However when hitting 9 + 123 + 456 + 7890 I get two error messages, “that all circuits are busy”, or “unable to connect your call…”

Brad

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Alex Robar
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