Hi,
in desperate need for this help please, i have 5 trunk that are open and ready, but when i make a call it only route to one so when someone else making another call at the same time, they got a busy attendent and drop the call. not sure on what to do now, i know it is with the route of the trunk but how can be configured different?
any help please
David,
please see: http://freepbx.org/forum/freepbx/installation/so-you-have-a-problem-and-want-help
In addition to what it tells you, you are going to need to provide details about your trunks and how they are configured.
We are not mind readers so we don’t know how anything is setup which makes it impossible with the details you have provided to give you guidance.
Hi and thank you for replying, i am roocky, when it comes to PBX, what do you mean my details on the trunck cnfig. and how can i ger them so you would be able to help m. please and thank you
He means, “details about your trunks and how they are configured” - just what he said. Are we not speaking English here?
Let me guess, someone else set this system up for you and you don’t have the foggiest idea what we’re asking of you. If I’m right, then you need to pass the question along to that person so they can answer it. If, for some reason you cannot do that, then you need to spend some time reading about how your system operates so you can ask questions that someone will want to attempt to answer. If you are using the Elastix distro then Google and read “Elastix Without Tears”, or if PBX in a Flash then “PIAF Without Tears”
Sorry my friend, that my knowldge with the PBX not to the extreme, that is why i am asking for your help.
All the configuration that i have done is thru the GUI interface, and i did not use the command file to do the same. If you require me to provide you with the .conf file, i would with pleasure, would you like me to post any of the .conf file i would be more then happy too for someone would like to assist me or help me. just let me know what file and i would.
I never used Asterisk or PBX before i started this work when i just got hire in my current job, i have read many of how the system operates, so i could got the system up to where it is now, i have called the Digium support for assistant with their product (AEX800) and have that figure out when much where missing from the config files. The support person told me their is something wrong with the way the calls to be routed, amd that i have 5 truncks can be used, but when making a call it only uses #2 and at the same time making another call from different phone it also uses the same trunk2, for that reason i need your help that tech person could not give me any details since it is an open source.
Thank you again for replying and looking forward to hear from you and your valubale assistant.
Cheers
Thanks again.
David,
So that you understand it took you two additional posts from my reply to just tell us what card you were using, that is details. based on your intial post how in the world are we to know who’s card you are using, what type of interfaces it has (which means what files it uses) without you providing those details.
Did you read the link I provided? It clearly say’s to provide versions of things running, is it a hand built or distro based box, what direction did you follow, etc. without these details we just can’t help.
We don’t know which version of asterisk you are using? 1.2, 1.4, 1.6? Please be very specific. There is a big difference between 1.4.21-2 and 1.4.22 as asterisk changed things at that point and came out with DAHDI to replace Zaptel so unless these things are told to us we can’t help and will stop responding.
something for you to think about.
If I said to you my form of transport to work every morning is broken and I need help getting it fixed. Can you help me. Is there enough details there for you to start helping? Do I drive a car, truck, bike, etc. what type of fuel or is it a hybrid? maybe I take a public bus and and it broke? You jsut don’t know so how do you start to help me? You do it by asking for the details…
The quickest way to get help is to start with the details, the more the better.
Thank you,
Asterisk version is 1.4.21.2 and Zaptel 1.4.12.1 also Libpri 1.4.8.
I followed the direction on how to install PBX in flash that provided on the site, created trunks (ZAP), extensions, ring group and so on all thru the GUI interface. The card that i am using as previousley mentioned is Digium AEX800b.
i have 3 analog line connecting to the card, and one of them used to receive and send fax. the card has 8 ports and three of them are disable and five are enable which the three phone lines are connected to. i can make a one call out and i can receive one call in, but when someone else picks up the phone during that call automoticaly will receive “circuts are busy please try to call again”. I have checked the extensions.conf file and nothing for the ZAP or trunks to guide me with direction of calls.
what file output i can provide you with to examin if that will help?
Cheers
How did you configure the card? one trunk per line? multiple lines per? yet again you have not provided enough.
The answers are in the details and you have not pealed back enough layers of the onion yet to help.
Please post the config files for the card setup.
Sorry, my friend i am over welmed with the system, alost like you not sure what to do or provide.
I did configuered the card as stated in the guide as follow;
(Initial Setup):
-
Please run the following command to generate a quick zaptel.conf file and a zapata-channels.conf file: “genzaptelconf”
-
after running this, you will need to apply these configurations to the card/drivers. Run the following: “ztcfg -vvv”.
-
Now that you have applied those settings, you will need to make sure that your newly written zapata-channels.conf file is included in zapata.conf. “cd /etc/asterisk” then “vi zapata.conf”.
-
hit the “i” or “insert” key and add the following line in the file: “#include zapata-channels.conf”
that was the setup for it.
zapata-auto.conf file info,
; Autogenerated by /usr/local/sbin/genzaptelconf – do not hand edit
; Zaptel Channels Configurations (zapata.conf)
;
; This is not intended to be a complete zapata.conf. Rather, it is intended
; to be #include-d by /etc/zapata.conf that will include the global settings
;
callerid=asreceived
; Span 1: WCTDM/0 “Wildcard AEX800 Board 1” (MASTER)
#signalling=fxo_ks
#; Note: this is an extension. Create a ZAP extension in AMP for Channel 1
#context=from-internal
#group=1
#channel => 1
signalling=fxs_ks
; Note: this is a trunk. Create a ZAP trunk in AMP for Channel 2
context=from-zaptel
group=0
channel => 2
#signalling=fxo_ks
#; Note: this is an extension. Create a ZAP extension in AMP for Channel 3
#context=from-internal
#group=1
#channel => 3
#signalling=fxo_ks
#; Note: this is an extension. Create a ZAP extension in AMP for Channel 4
#context=from-internal
#group=1
#channel => 4
signalling=fxs_ks
; Note: this is a trunk. Create a ZAP trunk in AMP for Channel 5
context=from-zaptel
group=0
channel => 5
signalling=fxs_ks
; Note: this is a trunk. Create a ZAP trunk in AMP for Channel 6
context=from-zaptel
group=0
channel => 6
signalling=fxs_ks
; Note: this is a trunk. Create a ZAP trunk in AMP for Channel 7
context=from-zaptel
group=0
channel => 7
signalling=fxs_ks
; Note: this is a trunk. Create a ZAP trunk in AMP for Channel 8
context=from-zaptel
group=0
channel => 8
the extensions_custom.conf file as:
; This file contains example extensions_custom.conf entries.
; extensions_custom.conf should be used to include customizations
; to AMP’s Asterisk dialplan.
; Extensions in AMP have access to the ‘from-internal’ context.
; The context ‘from-internal-custom’ is included in ‘from-internal’ by default
[from-internal-custom]
exten => 1234,1,Playback(demo-congrats) ; extensions can dial 1234
exten => 1234,2,Hangup()
exten => h,1,Hangup()
include => custom-recordme ; extensions can also dial 5678
; custom-count2four,s,1 can be used as a custom target for
; a Digital Receptionist menu or a Ring Group
[custom-count2four]
exten => s,1,SayDigits(1234)
exten => s,2,Hangup
; custom-recordme,5678,1 can be used as a custom target for
; a Digital Receptionist menu or a Ring Group
[custom-recordme]
exten => 5678,1,Wait(2)
exten => 5678,2,Record(/tmp/asterisk-recording:gsm)
exten => 5678,3,Wait(2)
exten => 5678,4,Playback(/tmp/asterisk-recording)
exten => 5678,5,Wait(2)
exten => 5678,6,Hangup
and the File zapata-channels.conf file is as follow
; Autogenerated by /usr/local/sbin/genzaptelconf – do not hand edit
; Zaptel Channels Configurations (zapata.conf)
;
; This is not intended to be a complete zapata.conf. Rather, it is intended
; to be #include-d by /etc/zapata.conf that will include the global settings
;
; Span 1: WCTDM/0 “Wildcard AEX800 Board 1” (MASTER)
;;; line="1 WCTDM/0/0 FXOKS (In use)"
signalling=fxo_ks
callerid=“Channel 1” <6001>
mailbox=6001
group=5
context=from-internal
channel => 1
callerid=
mailbox=
group=
context=default
;;; line="2 WCTDM/0/1 FXSKS (In use) RED"
signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
channel => 2
context=default
;;; line="5 WCTDM/0/4 FXSKS (In use) RED"
signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
channel => 5
context=default
;;; line="6 WCTDM/0/5 FXSKS (In use) RED"
signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
channel => 6
context=default
;;; line="7 WCTDM/0/6 FXSKS (In use) RED"
signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
channel => 7
context=default
;;; line="8 WCTDM/0/7 FXSKS (In use) RED"
signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
channel => 8
context=default
i hope that is almost enough info if more, i will…
Also in the /etc/asterisk the zapata.conf is like this:
; Zapata telephony interface
;
; Configuration file
include zapata-channels.conf
[trunkgroups]
[channels]
language=en
context=from-zaptel
signalling=fxs_ks
rxwink=300 ; Atlas seems to use long (250ms) winks
;
; Whether or not to do distinctive ring detection on FXO lines
;
;usedistinctiveringdetection=yes
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=no
echotraining=800
rxgain=0.0
txgain=0.0
group=0
callgroup=1
pickupgroup=1
immediate=no
;faxdetect=both
faxdetect=incoming
;faxdetect=outgoing
;faxdetect=no
#include zapata-auto.conf
;Include AMP configs
#include zapata_additional.conf
~
~
~
~
~
ok a big part of the issue is those directions. The line they asked you to insert needs to be done further down.
The common place for it is right after line:#include zapata-auto.conf
So the end of the file would look like
#include zapata-auto.conf
# include zapata-channels.conf
;Include AMP configs
#include zapata_additional.confWere you have it placed there is not a context directive (that’s one of those lines in square brackets like [channels]).
In fact the file needs happen after the line [channels] as the commands are all a part of the channels context.
alright, i did replace that line place it at the right place as you recomended, and nothing change for receiving or dialing, still if one making a call and we receive a call that person will get the message.
on the other hand i remember during the installation to get rid of one way audio, to edit /etc/asterisk/sip_custom.conf and the suggestion to add the actual public IP address and private IP address example like this:
externip=180.12.12.12
localnet=192.168.1.0/255.255.255.0
on my set up, Asterisk server is on the same switch as the phones. that switch getting network and DHCP from a router, to make it provide this system with different net then the rest of my network. ( my internel office net is 192.168.2.xxx and Asterisk, Phones are getting the 192.168.4.xxx from the router)
my /etc/asterisk/sip_custom.conf look like this:
externip=192.168.4.199 =====> this is the asterisk server ip address
localnet=192.168.4.0/255.255.255.0
i thought that maybe is the issue, but i try to provide much info as possible.
Back to your respond, replacing #incude zapata-channels.conf
did not change anything for the issue that i am having.
either you are using bad sources or are not remembering correctly.
For sip nat issues the file should be sip_general_custom.conf
And that will only fix and address issues with SIP based calls that traverse your firewall. Please see: http://freepbx.org/configuration_files for the details.
Now when dealing with a network with multiple subnets that are protected via a firewall or not nat’d you should provide a localhost- line for each and every ntwork subnet that has phone gear on it Specially those that are not on the same subnet as phone system.
If you have phones that are external to the firewall and traverse the nat you also need to include nat=yes in that file.
But I can’t tell as you’ve provided no details again to know if you are using sip based phones, IAX based or all of them are analog and connected to the card.
all phones that i am using they are SIP phones, nothing with IAX or analog,
the only analog signal connected to the card is the actual phone lines from the phone company (3lines).
SIP phones only i have the Mitel 5224 phones and Cisco 7940G both are SIP protocol.
Please check that you have the localnet statements in the proper place as they are not if you edited the sip.conf file (it get’s overwritten and your changes will go away).
ok this what i have done,
; Zapata telephony interface
;
; Configuration file
[trunkgroups]
[channels]
language=en
context=from-zaptel
signalling=fxs_ks
rxwink=300 ; Atlas seems to use long (250ms) winks
;
; Whether or not to do distinctive ring detection on FXO lines
;
;usedistinctiveringdetection=yes
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=no
echotraining=800
rxgain=0.0
txgain=0.0
group=0
callgroup=1
pickupgroup=1
immediate=no
;faxdetect=both
faxdetect=incoming
;faxdetect=outgoing
;faxdetect=no
#include zapata-auto.conf
include zapata-channels.conf
;Include AMP configs
#include zapata_additional.conf
But now when i call the office it keep on ringing but not at the office and if i try to call out keep getting the error " all circuit are busy now"
From one to another, the office is out of phones…
– Executing [4011307@Jasco-out:1] Macro(“SIP/1013-03962ef0”, “user-callerid|SKIPTTL|”) in new stack
– Executing [s@macro-user-callerid:1] Set(“SIP/1013-03962ef0”, “AMPUSER=1013”) in new stack
– Executing [s@macro-user-callerid:2] GotoIf(“SIP/1013-03962ef0”, “0?report”) in new stack
– Executing [s@macro-user-callerid:3] ExecIf(“SIP/1013-03962ef0”, “1|Set|REALCALLERIDNUM=1013”) in new stack
– Executing [s@macro-user-callerid:4] Set(“SIP/1013-03962ef0”, “AMPUSER=1013”) in new stack
– Executing [s@macro-user-callerid:5] Set(“SIP/1013-03962ef0”, “AMPUSERCIDNAME=Eric Lumsden”) in new stack
– Executing [s@macro-user-callerid:6] GotoIf(“SIP/1013-03962ef0”, “0?report”) in new stack
– Executing [s@macro-user-callerid:7] Set(“SIP/1013-03962ef0”, “AMPUSERCID=1013”) in new stack
– Executing [s@macro-user-callerid:8] Set(“SIP/1013-03962ef0”, “CALLERID(all)=“Eric Lumsden” <1013>”) in new stack
– Executing [s@macro-user-callerid:9] Set(“SIP/1013-03962ef0”, “REALCALLERIDNUM=1013”) in new stack
– Executing [s@macro-user-callerid:10] ExecIf(“SIP/1013-03962ef0”, “0|Set|CHANNEL(language)=”) in new stack
– Executing [s@macro-user-callerid:11] GotoIf(“SIP/1013-03962ef0”, “1?continue”) in new stack
– Goto (macro-user-callerid,s,20)
– Executing [s@macro-user-callerid:20] NoOp(“SIP/1013-03962ef0”, “Using CallerID “Eric Lumsden” <1013>”) in new stack
– Executing [4011307@Jasco-out:2] Set(“SIP/1013-03962ef0”, “_NODEST=”) in new stack
– Executing [4011307@Jasco-out:3] Macro(“SIP/1013-03962ef0”, “record-enable|1013|OUT|”) in new stack
– Executing [s@macro-record-enable:1] GotoIf(“SIP/1013-03962ef0”, “1?check”) in new stack
– Goto (macro-record-enable,s,4)
– Executing [s@macro-record-enable:4] AGI(“SIP/1013-03962ef0”, “recordingcheck|20090210-122631|1234283191.4”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20090210-122631|1234283191.4: Outbound recording not enabled
– AGI Script recordingcheck completed, returning 0
– Executing [s@macro-record-enable:5] MacroExit(“SIP/1013-03962ef0”, “”) in new stack
– Executing [4011307@Jasco-out:4] Macro(“SIP/1013-03962ef0”, “dialout-trunk|3|4011307||”) in new stack
– Executing [s@macro-dialout-trunk:1] Set(“SIP/1013-03962ef0”, “DIAL_TRUNK=3”) in new stack
– Executing [s@macro-dialout-trunk:2] GosubIf(“SIP/1013-03962ef0”, “0?sub-pincheck|s|1”) in new stack
– Executing [s@macro-dialout-trunk:3] GotoIf(“SIP/1013-03962ef0”, “0?disabletrunk|1”) in new stack
– Executing [s@macro-dialout-trunk:4] Set(“SIP/1013-03962ef0”, “DIAL_NUMBER=4011307”) in new stack
– Executing [s@macro-dialout-trunk:5] Set(“SIP/1013-03962ef0”, “DIAL_TRUNK_OPTIONS=tr”) in new stack
– Executing [s@macro-dialout-trunk:6] Set(“SIP/1013-03962ef0”, “OUTBOUND_GROUP=OUT_3”) in new stack
– Executing [s@macro-dialout-trunk:7] GotoIf(“SIP/1013-03962ef0”, “1?nomax”) in new stack
– Goto (macro-dialout-trunk,s,9)
– Executing [s@macro-dialout-trunk:9] GotoIf(“SIP/1013-03962ef0”, “0?skipoutcid”) in new stack
– Executing [s@macro-dialout-trunk:10] Set(“SIP/1013-03962ef0”, “DIAL_TRUNK_OPTIONS=”) in new stack
– Executing [s@macro-dialout-trunk:11] Macro(“SIP/1013-03962ef0”, “outbound-callerid|3”) in new stack
– Executing [s@macro-outbound-callerid:1] ExecIf(“SIP/1013-03962ef0”, “0|SetCallerPres|”) in new stack
– Executing [s@macro-outbound-callerid:2] ExecIf(“SIP/1013-03962ef0”, “0|Set|REALCALLERIDNUM=1013”) in new stack
– Executing [s@macro-outbound-callerid:3] GotoIf(“SIP/1013-03962ef0”, “1?normcid”) in new stack
– Goto (macro-outbound-callerid,s,6)
– Executing [s@macro-outbound-callerid:6] Set(“SIP/1013-03962ef0”, “USEROUTCID=”) in new stack
– Executing [s@macro-outbound-callerid:7] Set(“SIP/1013-03962ef0”, “EMERGENCYCID=”) in new stack
– Executing [s@macro-outbound-callerid:8] Set(“SIP/1013-03962ef0”, “TRUNKOUTCID=”) in new stack
– Executing [s@macro-outbound-callerid:9] GotoIf(“SIP/1013-03962ef0”, “1?trunkcid”) in new stack
– Goto (macro-outbound-callerid,s,12)
– Executing [s@macro-outbound-callerid:12] ExecIf(“SIP/1013-03962ef0”, “0|Set|CALLERID(all)=”) in new stack
– Executing [s@macro-outbound-callerid:13] GotoIf(“SIP/1013-03962ef0”, “1?exit”) in new stack
– Goto (macro-outbound-callerid,s,11)
– Executing [s@macro-outbound-callerid:11] MacroExit(“SIP/1013-03962ef0”, “”) in new stack
– Executing [s@macro-dialout-trunk:12] ExecIf(“SIP/1013-03962ef0”, “0|AGI|fixlocalprefix”) in new stack
– Executing [s@macro-dialout-trunk:13] Set(“SIP/1013-03962ef0”, “OUTNUM=4011307”) in new stack
– Executing [s@macro-dialout-trunk:14] Set(“SIP/1013-03962ef0”, “custom=ZAP/2”) in new stack
– Executing [s@macro-dialout-trunk:15] ExecIf(“SIP/1013-03962ef0”, “0|Set|DIAL_TRUNK_OPTIONS=M(setmusic^)”) in new stack
– Executing [s@macro-dialout-trunk:16] Macro(“SIP/1013-03962ef0”, “dialout-trunk-predial-hook|”) in new stack
– Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit(“SIP/1013-03962ef0”, “”) in new stack
– Executing [s@macro-dialout-trunk:17] GotoIf(“SIP/1013-03962ef0”, “0?bypass|1”) in new stack
– Executing [s@macro-dialout-trunk:18] GotoIf(“SIP/1013-03962ef0”, “0?customtrunk”) in new stack
– Executing [s@macro-dialout-trunk:19] Dial(“SIP/1013-03962ef0”, “ZAP/2/4011307|300|”) in new stack
== Everyone is busy/congested at this time (1:0/0/1)
– Executing [s@macro-dialout-trunk:20] Goto(“SIP/1013-03962ef0”, “s-CHANUNAVAIL|1”) in new stack
– Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
– Executing [s-CHANUNAVAIL@macro-dialout-trunk:1] GotoIf(“SIP/1013-03962ef0”, “1?noreport”) in new stack
– Goto (macro-dialout-trunk,s-CHANUNAVAIL,3)
– Executing [s-CHANUNAVAIL@macro-dialout-trunk:3] NoOp(“SIP/1013-03962ef0”, “TRUNK Dial failed due to CHANUNAVAIL (hangupcause: 66) - failing through to other trunks”) in new stack
– Executing [4011307@Jasco-out:5] Macro(“SIP/1013-03962ef0”, “dialout-trunk|1|4011307||”) in new stack
– Executing [s@macro-dialout-trunk:1] Set(“SIP/1013-03962ef0”, “DIAL_TRUNK=1”) in new stack
– Executing [s@macro-dialout-trunk:2] GosubIf(“SIP/1013-03962ef0”, “0?sub-pincheck|s|1”) in new stack
– Executing [s@macro-dialout-trunk:3] GotoIf(“SIP/1013-03962ef0”, “0?disabletrunk|1”) in new stack
– Executing [s@macro-dialout-trunk:4] Set(“SIP/1013-03962ef0”, “DIAL_NUMBER=4011307”) in new stack
– Executing [s@macro-dialout-trunk:5] Set(“SIP/1013-03962ef0”, “DIAL_TRUNK_OPTIONS=tr”) in new stack
– Executing [s@macro-dialout-trunk:6] Set(“SIP/1013-03962ef0”, “OUTBOUND_GROUP=OUT_1”) in new stack
– Executing [s@macro-dialout-trunk:7] GotoIf(“SIP/1013-03962ef0”, “1?nomax”) in new stack
– Goto (macro-dialout-trunk,s,9)
– Executing [s@macro-dialout-trunk:9] GotoIf(“SIP/1013-03962ef0”, “0?skipoutcid”) in new stack
– Executing [s@macro-dialout-trunk:10] Set(“SIP/1013-03962ef0”, “DIAL_TRUNK_OPTIONS=”) in new stack
– Executing [s@macro-dialout-trunk:11] Macro(“SIP/1013-03962ef0”, “outbound-callerid|1”) in new stack
– Executing [s@macro-outbound-callerid:1] ExecIf(“SIP/1013-03962ef0”, “0|SetCallerPres|”) in new stack
– Executing [s@macro-outbound-callerid:2] ExecIf(“SIP/1013-03962ef0”, “0|Set|REALCALLERIDNUM=1013”) in new stack
– Executing [s@macro-outbound-callerid:3] GotoIf(“SIP/1013-03962ef0”, “1?normcid”) in new stack
– Goto (macro-outbound-callerid,s,6)
– Executing [s@macro-outbound-callerid:6] Set(“SIP/1013-03962ef0”, “USEROUTCID=”) in new stack
– Executing [s@macro-outbound-callerid:7] Set(“SIP/1013-03962ef0”, “EMERGENCYCID=”) in new stack
– Executing [s@macro-outbound-callerid:8] Set(“SIP/1013-03962ef0”, “TRUNKOUTCID=”) in new stack
– Executing [s@macro-outbound-callerid:9] GotoIf(“SIP/1013-03962ef0”, “1?trunkcid”) in new stack
– Goto (macro-outbound-callerid,s,12)
– Executing [s@macro-outbound-callerid:12] ExecIf(“SIP/1013-03962ef0”, “0|Set|CALLERID(all)=”) in new stack
– Executing [s@macro-outbound-callerid:13] GotoIf(“SIP/1013-03962ef0”, “1?exit”) in new stack
– Goto (macro-outbound-callerid,s,11)
– Executing [s@macro-outbound-callerid:11] MacroExit(“SIP/1013-03962ef0”, “”) in new stack
– Executing [s@macro-dialout-trunk:12] ExecIf(“SIP/1013-03962ef0”, “0|AGI|fixlocalprefix”) in new stack
– Executing [s@macro-dialout-trunk:13] Set(“SIP/1013-03962ef0”, “OUTNUM=4011307”) in new stack
– Executing [s@macro-dialout-trunk:14] Set(“SIP/1013-03962ef0”, “custom=ZAP/5”) in new stack
– Executing [s@macro-dialout-trunk:15] ExecIf(“SIP/1013-03962ef0”, “0|Set|DIAL_TRUNK_OPTIONS=M(setmusic^)”) in new stack
– Executing [s@macro-dialout-trunk:16] Macro(“SIP/1013-03962ef0”, “dialout-trunk-predial-hook|”) in new stack
– Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit(“SIP/1013-03962ef0”, “”) in new stack
– Executing [s@macro-dialout-trunk:17] GotoIf(“SIP/1013-03962ef0”, “0?bypass|1”) in new stack
– Executing [s@macro-dialout-trunk:18] GotoIf(“SIP/1013-03962ef0”, “0?customtrunk”) in new stack
– Executing [s@macro-dialout-trunk:19] Dial(“SIP/1013-03962ef0”, “ZAP/5/4011307|300|”) in new stack
== Everyone is busy/congested at this time (1:0/0/1)
– Executing [s@macro-dialout-trunk:20] Goto(“SIP/1013-03962ef0”, “s-CHANUNAVAIL|1”) in new stack
– Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
– Executing [s-CHANUNAVAIL@macro-dialout-trunk:1] GotoIf(“SIP/1013-03962ef0”, “1?noreport”) in new stack
– Goto (macro-dialout-trunk,s-CHANUNAVAIL,3)
– Executing [s-CHANUNAVAIL@macro-dialout-trunk:3] NoOp(“SIP/1013-03962ef0”, “TRUNK Dial failed due to CHANUNAVAIL (hangupcause: 66) - failing through to other trunks”) in new stack
– Executing [4011307@Jasco-out:6] Macro(“SIP/1013-03962ef0”, “dialout-trunk|2|4011307||”) in new stack
– Executing [s@macro-dialout-trunk:1] Set(“SIP/1013-03962ef0”, “DIAL_TRUNK=2”) in new stack
– Executing [s@macro-dialout-trunk:2] GosubIf(“SIP/1013-03962ef0”, “0?sub-pincheck|s|1”) in new stack
– Executing [s@macro-dialout-trunk:3] GotoIf(“SIP/1013-03962ef0”, “0?disabletrunk|1”) in new stack
– Executing [s@macro-dialout-trunk:4] Set(“SIP/1013-03962ef0”, “DIAL_NUMBER=4011307”) in new stack
– Executing [s@macro-dialout-trunk:5] Set(“SIP/1013-03962ef0”, “DIAL_TRUNK_OPTIONS=tr”) in new stack
– Executing [s@macro-dialout-trunk:6] Set(“SIP/1013-03962ef0”, “OUTBOUND_GROUP=OUT_2”) in new stack
– Executing [s@macro-dialout-trunk:7] GotoIf(“SIP/1013-03962ef0”, “1?nomax”) in new stack
– Goto (macro-dialout-trunk,s,9)
– Executing [s@macro-dialout-trunk:9] GotoIf(“SIP/1013-03962ef0”, “0?skipoutcid”) in new stack
– Executing [s@macro-dialout-trunk:10] Set(“SIP/1013-03962ef0”, “DIAL_TRUNK_OPTIONS=”) in new stack
– Executing [s@macro-dialout-trunk:11] Macro(“SIP/1013-03962ef0”, “outbound-callerid|2”) in new stack
– Executing [s@macro-outbound-callerid:1] ExecIf(“SIP/1013-03962ef0”, “0|SetCallerPres|”) in new stack
– Executing [s@macro-outbound-callerid:2] ExecIf(“SIP/1013-03962ef0”, “0|Set|REALCALLERIDNUM=1013”) in new stack
– Executing [s@macro-outbound-callerid:3] GotoIf(“SIP/1013-03962ef0”, “1?normcid”) in new stack
– Goto (macro-outbound-callerid,s,6)
– Executing [s@macro-outbound-callerid:6] Set(“SIP/1013-03962ef0”, “USEROUTCID=”) in new stack
– Executing [s@macro-outbound-callerid:7] Set(“SIP/1013-03962ef0”, “EMERGENCYCID=”) in new stack
– Executing [s@macro-outbound-callerid:8] Set(“SIP/1013-03962ef0”, “TRUNKOUTCID=”) in new stack
– Executing [s@macro-outbound-callerid:9] GotoIf(“SIP/1013-03962ef0”, “1?trunkcid”) in new stack
– Goto (macro-outbound-callerid,s,12)
– Executing [s@macro-outbound-callerid:12] ExecIf(“SIP/1013-03962ef0”, “0|Set|CALLERID(all)=”) in new stack
– Executing [s@macro-outbound-callerid:13] GotoIf(“SIP/1013-03962ef0”, “1?exit”) in new stack
– Goto (macro-outbound-callerid,s,11)
– Executing [s@macro-outbound-callerid:11] MacroExit(“SIP/1013-03962ef0”, “”) in new stack
– Executing [s@macro-dialout-trunk:12] ExecIf(“SIP/1013-03962ef0”, “0|AGI|fixlocalprefix”) in new stack
– Executing [s@macro-dialout-trunk:13] Set(“SIP/1013-03962ef0”, “OUTNUM=4011307”) in new stack
– Executing [s@macro-dialout-trunk:14] Set(“SIP/1013-03962ef0”, “custom=ZAP/6”) in new stack
– Executing [s@macro-dialout-trunk:15] ExecIf(“SIP/1013-03962ef0”, “0|Set|DIAL_TRUNK_OPTIONS=M(setmusic^)”) in new stack
– Executing [s@macro-dialout-trunk:16] Macro(“SIP/1013-03962ef0”, “dialout-trunk-predial-hook|”) in new stack
– Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit(“SIP/1013-03962ef0”, “”) in new stack
– Executing [s@macro-dialout-trunk:17] GotoIf(“SIP/1013-03962ef0”, “0?bypass|1”) in new stack
– Executing [s@macro-dialout-trunk:18] GotoIf(“SIP/1013-03962ef0”, “0?customtrunk”) in new stack
– Executing [s@macro-dialout-trunk:19] Dial(“SIP/1013-03962ef0”, “ZAP/6/4011307|300|”) in new stack
== Everyone is busy/congested at this time (1:0/0/1)
– Executing [s@macro-dialout-trunk:20] Goto(“SIP/1013-03962ef0”, “s-CHANUNAVAIL|1”) in new stack
– Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
– Executing [s-CHANUNAVAIL@macro-dialout-trunk:1] GotoIf(“SIP/1013-03962ef0”, “1?noreport”) in new stack
– Goto (macro-dialout-trunk,s-CHANUNAVAIL,3)
– Executing [s-CHANUNAVAIL@macro-dialout-trunk:3] NoOp(“SIP/1013-03962ef0”, “TRUNK Dial failed due to CHANUNAVAIL (hangupcause: 66) - failing through to other trunks”) in new stack
– Executing [4011307@Jasco-out:7] Macro(“SIP/1013-03962ef0”, “outisbusy|”) in new stack
– Executing [s@macro-outisbusy:1] Playback(“SIP/1013-03962ef0”, “all-circuits-busy-now|noanswer”) in new stack
– <SIP/1013-03962ef0> Playing ‘all-circuits-busy-now’ (language ‘en’)
– Executing [s@macro-outisbusy:2] Playback(“SIP/1013-03962ef0”, “pls-try-call-later|noanswer”) in new stack
– <SIP/1013-03962ef0> Playing ‘pls-try-call-later’ (language ‘en’)
– Executing [s@macro-outisbusy:3] Macro(“SIP/1013-03962ef0”, “hangupcall”) in new stack
– Executing [s@macro-hangupcall:1] ResetCDR(“SIP/1013-03962ef0”, “w”) in new stack
– Executing [s@macro-hangupcall:2] NoCDR(“SIP/1013-03962ef0”, “”) in new stack
– Executing [s@macro-hangupcall:3] GotoIf(“SIP/1013-03962ef0”, “1?skiprg”) in new stack
– Goto (macro-hangupcall,s,6)
– Executing [s@macro-hangupcall:6] GotoIf(“SIP/1013-03962ef0”, “1?skipblkvm”) in new stack
– Goto (macro-hangupcall,s,9)
– Executing [s@macro-hangupcall:9] GotoIf(“SIP/1013-03962ef0”, “1?theend”) in new stack
– Goto (macro-hangupcall,s,11)
– Executing [s@macro-hangupcall:11] Hangup(“SIP/1013-03962ef0”, “”) in new stack
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on ‘SIP/1013-03962ef0’ in macro ‘hangupcall’
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on ‘SIP/1013-03962ef0’ in macro ‘outisbusy’
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/1013-03962ef0
i got a suggestion from paid tech support but couldn’t provide me with anymore info since they do not support and open source, his suggestion as follow:
It seems that your dialing rules are set to always dial out of Zap/2-1 versus a group, i.e. g0 (group 0). This will need to be changed to dial out over all available channels. This should be a simple change in the GUI, or a small change in your dialplan
I also forgot to mention that change may take place one of the following locations;
Outbound Dialing Rules or Outgoing Dialing Rules (GUI)
Now that is all good, but i cannot find what is wrong with my dialplan, anyone know where to go or do in the PBX to have it all set up as thye mentioned.
CHeers