Regarding All circuits are busy now recording

Asterisk Asterisk Support
1

Hi all,
I’m new to asterisk world. And trying to configure a asterisk server to terminate voip traffic to pstn network.
I have installed asterisk now and installed TDM800P 8port fxo module on it.
Then i configured simple outbound route and pstn trunk on it.
But when im trying to take a call to a mobile number from sip phone thru fxo card im getting all circuits are busy now recording. Below are the debug info. Please advice & this will be a grate help for me.

[root@localhost ~]# lsdahdi

Span 1: WCTDM/0 “Wildcard TDM800P Board 1” (MASTER)

1 FXS FXSKS (EC: MG2) RED
2 FXS FXSKS (EC: MG2) RED
3 FXS FXSKS (EC: MG2) RED
4 FXS FXSKS (EC: MG2) RED
5 FXS FXSKS (EC: MG2) RED
6 FXS FXSKS (EC: MG2) RED
7 FXS FXSKS (EC: MG2) RED
8 FXS FXSKS (EC: MG2) RED

[root@localhost ~]#
[root@localhost ~]# asterisk -vvvvvr
Asterisk 1.4.24, Copyright © 1999 - 2008 Digium, Inc. and others.
Created by Mark Spencer markster@digium.com
Asterisk comes with ABSOLUTELY NO WARRANTY; type ‘core 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 ‘core show license’ for details.

== Parsing ‘/etc/asterisk/asterisk.conf’: Found
Connected to Asterisk 1.4.24 currently running on localhost (pid = 2962)
Verbosity is at least 5
– Executing [0772328091@from-internal:1] Macro(“SIP/500-b7807e88”, “user-callerid|SKIPTTL|”) in new stack
– Executing [s@macro-user-callerid:1] Set(“SIP/500-b7807e88”, “AMPUSER=500”) in new stack
– Executing [s@macro-user-callerid:2] GotoIf(“SIP/500-b7807e88”, “0?report”) in new stack
– Executing [s@macro-user-callerid:3] ExecIf(“SIP/500-b7807e88”, “1|Set|REALCALLERIDNUM=500”) in new stack
– Executing [s@macro-user-callerid:4] Set(“SIP/500-b7807e88”, “AMPUSER=500”) in new stack
– Executing [s@macro-user-callerid:5] Set(“SIP/500-b7807e88”, “AMPUSERCIDNAME=man”) in new stack
– Executing [s@macro-user-callerid:6] GotoIf(“SIP/500-b7807e88”, “0?report”) in new stack
– Executing [s@macro-user-callerid:7] Set(“SIP/500-b7807e88”, “AMPUSERCID=500”) in new stack
– Executing [s@macro-user-callerid:8] Set(“SIP/500-b7807e88”, “CALLERID(all)=“man” <500>”) in new stack
– Executing [s@macro-user-callerid:9] Set(“SIP/500-b7807e88”, “REALCALLERIDNUM=500”) in new stack
– Executing [s@macro-user-callerid:10] GotoIf(“SIP/500-b7807e88”, “1?continue”) in new stack
– Goto (macro-user-callerid,s,19)
– Executing [s@macro-user-callerid:19] NoOp(“SIP/500-b7807e88”, “Using CallerID “man” <500>”) in new stack
– Executing [0772328091@from-internal:2] Set(“SIP/500-b7807e88”, “_NODEST=”) in new stack
– Executing [0772328091@from-internal:3] Macro(“SIP/500-b7807e88”, “record-enable|500|OUT|”) in new stack
– Executing [s@macro-record-enable:1] GotoIf(“SIP/500-b7807e88”, “1?check”) in new stack
– Goto (macro-record-enable,s,4)
– Executing [s@macro-record-enable:4] AGI(“SIP/500-b7807e88”, “recordingcheck|20100323-090425|1269315265.5”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20100323-090425|1269315265.5: Outbound recording not enabled
– AGI Script recordingcheck completed, returning 0
– Executing [s@macro-record-enable:5] MacroExit(“SIP/500-b7807e88”, “”) in new stack
– Executing [0772328091@from-internal:4] Macro(“SIP/500-b7807e88”, “dialout-trunk|1|0772328091||”) in new stack
– Executing [s@macro-dialout-trunk:1] Set(“SIP/500-b7807e88”, “DIAL_TRUNK=1”) in new stack
– Executing [s@macro-dialout-trunk:2] GosubIf(“SIP/500-b7807e88”, “0?sub-pincheck|s|1”) in new stack
– Executing [s@macro-dialout-trunk:3] GotoIf(“SIP/500-b7807e88”, “0?disabletrunk|1”) in new stack
– Executing [s@macro-dialout-trunk:4] Set(“SIP/500-b7807e88”, “DIAL_NUMBER=0772328091”) in new stack
– Executing [s@macro-dialout-trunk:5] Set(“SIP/500-b7807e88”, “DIAL_TRUNK_OPTIONS=tr”) in new stack
– Executing [s@macro-dialout-trunk:6] Set(“SIP/500-b7807e88”, “OUTBOUND_GROUP=OUT_1”) in new stack
– Executing [s@macro-dialout-trunk:7] GotoIf(“SIP/500-b7807e88”, “0?nomax”) in new stack
– Executing [s@macro-dialout-trunk:8] GotoIf(“SIP/500-b7807e88”, “0?chanfull”) in new stack
– Executing [s@macro-dialout-trunk:9] GotoIf(“SIP/500-b7807e88”, “0?skipoutcid”) in new stack
– Executing [s@macro-dialout-trunk:10] Set(“SIP/500-b7807e88”, “DIAL_TRUNK_OPTIONS=”) in new stack
– Executing [s@macro-dialout-trunk:11] Macro(“SIP/500-b7807e88”, “outbound-callerid|1”) in new stack
– Executing [s@macro-outbound-callerid:1] ExecIf(“SIP/500-b7807e88”, “0|SetCallerPres|”) in new stack
– Executing [s@macro-outbound-callerid:2] ExecIf(“SIP/500-b7807e88”, “0|Set|REALCALLERIDNUM=500”) in new stack
– Executing [s@macro-outbound-callerid:3] GotoIf(“SIP/500-b7807e88”, “1?normcid”) in new stack
– Goto (macro-outbound-callerid,s,6)
– Executing [s@macro-outbound-callerid:6] Set(“SIP/500-b7807e88”, “USEROUTCID=”) in new stack
– Executing [s@macro-outbound-callerid:7] Set(“SIP/500-b7807e88”, “EMERGENCYCID=”) in new stack
– Executing [s@macro-outbound-callerid:8] Set(“SIP/500-b7807e88”, “TRUNKOUTCID=0312231818”) in new stack
– Executing [s@macro-outbound-callerid:9] GotoIf(“SIP/500-b7807e88”, “1?trunkcid”) in new stack
– Goto (macro-outbound-callerid,s,12)
– Executing [s@macro-outbound-callerid:12] ExecIf(“SIP/500-b7807e88”, “1|Set|CALLERID(all)=0312231818”) in new stack
– Executing [s@macro-outbound-callerid:13] GotoIf(“SIP/500-b7807e88”, “1?exit”) in new stack
– Goto (macro-outbound-callerid,s,11)
– Executing [s@macro-outbound-callerid:11] MacroExit(“SIP/500-b7807e88”, “”) in new stack
– Executing [s@macro-dialout-trunk:12] ExecIf(“SIP/500-b7807e88”, “1|AGI|fixlocalprefix”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
> fixlocalprefix: Using pattern XXXXXXXXXX
== fixlocalprefix: Dialpattern XXXXXXXXXX matched. 0772328091 -> 0772328091
– AGI Script fixlocalprefix completed, returning 0
– Executing [s@macro-dialout-trunk:13] Set(“SIP/500-b7807e88”, “OUTNUM=0772328091”) in new stack
– Executing [s@macro-dialout-trunk:14] Set(“SIP/500-b7807e88”, “custom=DAHDI/g0”) in new stack
– Executing [s@macro-dialout-trunk:15] ExecIf(“SIP/500-b7807e88”, “0|Set|DIAL_TRUNK_OPTIONS=M(setmusic^)”) in new stack
– Executing [s@macro-dialout-trunk:16] Macro(“SIP/500-b7807e88”, “dialout-trunk-predial-hook|”) in new stack
– Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit(“SIP/500-b7807e88”, “”) in new stack
– Executing [s@macro-dialout-trunk:17] GotoIf(“SIP/500-b7807e88”, “0?bypass|1”) in new stack
– Executing [s@macro-dialout-trunk:18] GotoIf(“SIP/500-b7807e88”, “0?customtrunk”) in new stack
– Executing [s@macro-dialout-trunk:19] Dial(“SIP/500-b7807e88”, “DAHDI/g0/0772328091|300|”) in new stack
== Everyone is busy/congested at this time (1:0/0/1)
– Executing [s@macro-dialout-trunk:20] Goto(“SIP/500-b7807e88”, “s-CHANUNAVAIL|1”) in new stack
– Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
– Executing [s-CHANUNAVAIL@macro-dialout-trunk:1] GotoIf(“SIP/500-b7807e88”, “1?noreport”) in new stack
– Goto (macro-dialout-trunk,s-CHANUNAVAIL,3)
– Executing [s-CHANUNAVAIL@macro-dialout-trunk:3] NoOp(“SIP/500-b7807e88”, “TRUNK Dial failed due to CHANUNAVAIL (hangupcause: 0) - failing through to other trunks”) in new stack
– Executing [0772328091@from-internal:5] Macro(“SIP/500-b7807e88”, “outisbusy|”) in new stack
– Executing [s@macro-outisbusy:1] Playback(“SIP/500-b7807e88”, “all-circuits-busy-now|noanswer”) in new stack
– <SIP/500-b7807e88> Playing ‘all-circuits-busy-now’ (language ‘en’)
– Executing [s@macro-outisbusy:2] Playback(“SIP/500-b7807e88”, “pls-try-call-later|noanswer”) in new stack
– <SIP/500-b7807e88> Playing ‘pls-try-call-later’ (language ‘en’)
== Spawn extension (macro-outisbusy, s, 2) exited non-zero on ‘SIP/500-b7807e88’ in macro ‘outisbusy’
== Spawn extension (from-internal, 0772328091, 5) exited non-zero on ‘SIP/500-b7807e88’
– Executing [h@from-internal:1] Macro(“SIP/500-b7807e88”, “hangupcall”) in new stack
– Executing [s@macro-hangupcall:1] ResetCDR(“SIP/500-b7807e88”, “w”) in new stack
– Executing [s@macro-hangupcall:2] NoCDR(“SIP/500-b7807e88”, “”) in new stack
– Executing [s@macro-hangupcall:3] GotoIf(“SIP/500-b7807e88”, “1?skiprg”) in new stack
– Goto (macro-hangupcall,s,6)
– Executing [s@macro-hangupcall:6] GotoIf(“SIP/500-b7807e88”, “1?skipblkvm”) in new stack
– Goto (macro-hangupcall,s,9)
– Executing [s@macro-hangupcall:9] GotoIf(“SIP/500-b7807e88”, “1?theend”) in new stack
– Goto (macro-hangupcall,s,11)
– Executing [s@macro-hangupcall:11] Hangup(“SIP/500-b7807e88”, “”) in new stack
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on ‘SIP/500-b7807e88’ in macro ‘hangupcall’
== Spawn extension (from-internal, h, 1) exited non-zero on ‘SIP/500-b7807e88’

2
  1. are you sure you have FXO modules and not FXS ones? what is the code of the modules you have and/or what is their color?
  2. please post the DAHDI config file (lile /etc/dahdi/system.conf)
  3. please post the chan_dahdi config file (like /etc/asterisk/chan_dahdi.conf and every included file)

[quote]
– Executing [s@macro-dialout-trunk:19] Dial(“SIP/500-b7807e88”, “DAHDI/g0/0772328091|300|”) in new stack
== Everyone is busy/congested at this time (1:0/0/1)[/quote]

Basically you are trying to dial on DAHDI, group #0 (DAHDI/g0) and no lines could be picked up. The main reason, in my opinion, is that your DAHDI is wrongly configured.

HTH,
Ioan.

3

Hi indreias,

Thank you for your repply.

I ordered TDP800P with 8 FXO port card. It have two red plugable circuits.

I was unable to find the /etc/asterisk/chan_dahdi.conf on the mentioned location. There was only two below mentioned files.

[root@localhost ~]# ls /etc/asterisk/chan_dahdi
chan_dahdi_additional.conf chan_dahdi.conf.template

[root@localhost ~]# cat /etc/dahdi/system.conf

Autogenerated by /usr/sbin/dahdi_genconf on Tue Mar 23 09:58:29 2010 – do not hand edit

Dahdi Configuration File

This file is parsed by the Dahdi Configurator, dahdi_cfg

Span 1: WCTDM/0 “Wildcard TDM800P Board 1” (MASTER)

fxsks=1
echocanceller=mg2,1
fxsks=2
echocanceller=mg2,2
fxsks=3
echocanceller=mg2,3
fxsks=4
echocanceller=mg2,4
fxsks=5
echocanceller=mg2,5
fxsks=6
echocanceller=mg2,6
fxsks=7
echocanceller=mg2,7
fxsks=8
echocanceller=mg2,8

Global data

loadzone = us
defaultzone = us

Please help me to solve this issue.

Thank yo verry much.

4

red modules - OK - them are FXO

please post chan_dahdi_additional.conf

In order to find which file is including chan_dahdi_additional.conf please run:

and post the content of that file as well.

Ioan.

5

Please post you dialplan,too.Maybe there are some issues!

6

Hi indreias,

Thank you for your quick reply,

Hear are the requested information,

[root@localhost ~]# cat /etc/asterisk/chan_dahdi_additional.conf
;--------------------------------------------------------------------------------;
; Do NOT edit this file as it is auto-generated by FreePBX. All modifications to ;
; this file must be done via the web gui. There are alternative files to make ;
; custom modifications, details at: freepbx.org/configuration_files ;
;--------------------------------------------------------------------------------;
;

[root@localhost ~]#

[root@localhost ~]# grep chan_dahdi_additional.conf /etc/asterisk/*.conf
[root@localhost ~]#
[root@localhost ~]#

[root@localhost ~]# ls /etc/asterisk/
asterisk.conf iax.conf privacy.conf
cdr_mysql.conf iax_custom.conf queues_additional.conf
cdr_mysql.conf.bak iax_custom_post.conf queues.conf
chan_dahdi_additional.conf iax_general_additional.conf queues_custom.conf
chan_dahdi.conf.template iax_general_custom.conf queues_custom_general.conf
dahdi-channels.conf iax_registrations.conf queues_general_additional.conf
dahdi-channels.conf.bak iax_registrations_custom.conf queues_post_custom.conf
enum.conf indications.conf rtp.conf
extensions_additional.conf localprefixes.conf sip_additional.conf
extensions.conf logger.conf sip.conf
extensions_custom.conf manager_additional.conf sip_custom.conf
extensions_custom.conf.sample manager.conf sip_custom_post.conf
extensions_override_freepbx.conf manager.conf.bak sip_general_additional.conf
features_applicationmap_additional.conf manager_custom.conf sip_general_custom.conf
features_applicationmap_custom.conf meetme_additional.conf sip_nat.conf
features.conf meetme.conf sip_registrations.conf
features_featuremap_additional.conf modem.conf sip_registrations_custom.conf
features_featuremap_custom.conf modules.conf version
features_general_additional.conf musiconhold_additional.conf vm_email.inc
features_general_custom.conf musiconhold.conf vm_general.inc
freepbx_featurecodes.conf musiconhold_custom.conf voicemail.conf
freepbx_module_admin.conf no voicemail.conf.template
globals_custom.conf phone.conf zapata_additional.conf
iax_additional.conf phpagi.conf zapata.conf.template
[root@localhost ~]#

[root@localhost asterisk]# cat dahdi-channels.conf
; Autogenerated by /usr/sbin/dahdi_genconf on Tue Mar 23 09:58:29 2010 – do not hand edit
; Dahdi Channels Configurations (chan_dahdi.conf)
;
; This is not intended to be a complete chan_dahdi.conf. Rather, it is intended
; to be #include-d by /etc/asterisk/chan_dahdi.conf that will include the global settings
;

; Span 1: WCTDM/0 “Wildcard TDM800P Board 1” (MASTER)
;;; line="1 WCTDM/0/0"
signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
channel => 1
callerid=
group=
context=default

;;; line="2 WCTDM/0/1"
signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
channel => 2
callerid=
group=
context=default

;;; line="3 WCTDM/0/2"
signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
channel => 3
callerid=
group=
context=default

;;; line="4 WCTDM/0/3"
signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
channel => 4
callerid=
group=
context=default

;;; line="5 WCTDM/0/4"
signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
channel => 5
callerid=
group=
context=default

;;; line="6 WCTDM/0/5"
signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
channel => 6
callerid=
group=
context=default

;;; line="7 WCTDM/0/6"
signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
channel => 7
callerid=
group=
context=default

;;; line="8 WCTDM/0/7"
signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
channel => 8
callerid=
group=
context=default

7

Hi sunking,

I’m using simple dial plan.

all 10digit XXXXXXXXXX will route to fxo trunk.

8

You are not using a simple dialplan; there are large numbers of steps, including AGI steps, before the Dial applications is called.

You are actually getting a channel unavailable status, rather than a busy status. This is indicative of a misconfiguration.

I am not really familiar with Dahdi and circuit switched lines, but I rather guess that the RED at the end of the status lines means that there is a serious alarm condition on the line.

9

run the following command:

cat << EOF > /etc/asterisk/chan_dahdi.conf
[trunkgroups]
[channels]
#include dahdi-channels.conf
EOF

stop your Asterisk instance (I think you are using A FreePBX GUI so you should issue amportal stop)
start your Asterisk instance (…amportal start)

post here the results of:

dahdi_cfg -vv lsdahdi

HTH,
Ioan.

10

Thank you for ur repply. Hear are the requested information.

[root@localhost asterisk]# cat /etc/asterisk/chan_dahdi
chan_dahdi_additional.conf chan_dahdi.conf chan_dahdi.conf.template

[root@localhost asterisk]# cat /etc/asterisk/chan_dahdi.conf
[trunkgroups]
[channels]
#include dahdi-channels.conf
[root@localhost asterisk]#

[root@localhost asterisk]# amportal stop

STOPPING ASTERISK
Asterisk Stopped

STOPPING FOP SERVER
FOP Server Stopped
[root@localhost asterisk]# amportal start

SETTING FILE PERMISSIONS
Permissions OK

STARTING ASTERISK
Asterisk Started

STARTING FOP SERVER
FOP Server Started
[root@localhost asterisk]# dahdi_cfg -vv
DAHDI Tools Version - 2.1.0.2

DAHDI Version: 2.1.0.4
Echo Canceller(s): MG2
Configuration

Channel map:

Channel 01: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 01)
Channel 02: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 02)
Channel 03: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 03)
Channel 04: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 04)
Channel 05: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 05)
Channel 06: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 06)
Channel 07: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 07)
Channel 08: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 08)

8 channels to configure.

Setting echocan for channel 1 to mg2
Setting echocan for channel 2 to mg2
Setting echocan for channel 3 to mg2
Setting echocan for channel 4 to mg2
Setting echocan for channel 5 to mg2
Setting echocan for channel 6 to mg2
Setting echocan for channel 7 to mg2
Setting echocan for channel 8 to mg2
[root@localhost asterisk]# lsdahdi

Span 1: WCTDM/0 “Wildcard TDM800P Board 1” (MASTER)

1 FXS FXSKS (In use) (EC: MG2) RED
2 FXS FXSKS (In use) (EC: MG2) RED
3 FXS FXSKS (In use) (EC: MG2) RED
4 FXS FXSKS (In use) (EC: MG2) RED
5 FXS FXSKS (In use) (EC: MG2) RED
6 FXS FXSKS (In use) (EC: MG2) RED
7 FXS FXSKS (In use) (EC: MG2) RED
8 FXS FXSKS (In use) (EC: MG2) RED
[root@localhost asterisk]#

11

ok - we have to debug one more issue -> why lsdahdi show “FXS FXSKS” instead of “FXO FXSKS”

For this please provide the output from:

12

[quote=“indreias”]ok - we have to debug one more issue -> why lsdahdi show “FXS FXSKS” instead of “FXO FXSKS”

For this please provide the output from:

Hi indreias,

Now im @ office and i lost my connection to my home asterisk server. So as soon as i go home i will post that debug.
Any way thank you very much for your kind help.

13

Hi friend,

Hear are the requested details.

[root@localhost ~]# dmesg | grep AUTO
Port 1: Installed – AUTO FXO (FCC mode)
Port 2: Installed – AUTO FXO (FCC mode)
Port 3: Installed – AUTO FXO (FCC mode)
Port 4: Installed – AUTO FXO (FCC mode)
Port 5: Installed – AUTO FXO (FCC mode)
Port 6: Installed – AUTO FXO (FCC mode)
Port 7: Installed – AUTO FXO (FCC mode)
Port 8: Installed – AUTO FXO (FCC mode)

14

ok - so the ports are correctly identified as FXO ones…probably there is an error in lsdahdi output (as I do not see FXO there).

so - now we hope DAHDI is well configured but we see them all in RED. This is an indication that the line is not detected.
Did you put a PSTN line into one of the FXO ports? It is a trivial question but maybe you forget it.

Ioan.

15

hi frnd,

I think i found the problem. Those PST lines are connected through splitters. When i plug normal phone to the PSTN line through splitters it worked well. So i didn’t care about it. But unfortunately through splitter FXO won’t detect the pstn line. That’s terrible. When i removed the splitter and connect the line now its working fine.

Any way your last hint helped me to sort out this problem and i really want to thank you.

But i still have a small doubt. I don’t zapata.conf in my asterisk folder. So where should i mention the “answeronpolarityswitch=yes” & hanguponpolarityswitch=yes to support line reversal ?

Thank you.

16

glad to hear that you have now a working system.

for all zapata settings please use chan_dahdi.conf, under the [channels] context, just before you have the include command.

BR,
Ioan.