Help...Inbound Calls Problem on Trixbox 1.1.1

Hi There,
I am new to Asterisk & Linux, Need your help in resolving the issue.

Trixbox 1.1 is configured as per documentation provided. I have 2 PRI links connected to TE205P card. One Incoming & One for outgoing. When ever i try to make call i am getting please try to call later. See below log from asterisk command line


-- Executing Macro("SIP/2138-ce60", "dialout-trunk|1|9083384347||") in new stack
-- Executing GotoIf("SIP/2138-ce60", "1?3:2") in new stack
-- Goto (macro-dialout-trunk,s,3)
-- Executing Macro("SIP/2138-ce60", "user-callerid") in new stack
-- Executing GotoIf("SIP/2138-ce60", "0?report") in new stack
-- Executing GotoIf("SIP/2138-ce60", "0?start") in new stack
-- Executing Set("SIP/2138-ce60", "REALCALLERIDNUM=2138") in new stack
-- Executing NoOp("SIP/2138-ce60", "REALCALLERIDNUM is 2138") in new stack
-- Executing Set("SIP/2138-ce60", "AMPUSER=2138") in new stack
-- Executing Set("SIP/2138-ce60", "AMPUSERCIDNAME=LDP Sekhar") in new stack
-- Executing GotoIf("SIP/2138-ce60", "0?report") in new stack
-- Executing Set("SIP/2138-ce60", "CALLERID(all)=LDP Sekhar <2138>") in new stack
-- Executing NoOp("SIP/2138-ce60", "Using CallerID "LDP Sekhar" <2138>") in new stack
-- Executing Macro("SIP/2138-ce60", "record-enable|2138|OUT") in new stack
-- Executing GotoIf("SIP/2138-ce60", "0 > 0?2:4") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing AGI("SIP/2138-ce60", "recordingcheck|20060715-190402|1153004642.2") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck

recordingcheck|20060715-190402|1153004642.2: Outbound recording not enabled
– AGI Script recordingcheck completed, returning 0
– Executing NoOp(“SIP/2138-ce60”, “No recording needed”) in new stack
– Executing Macro(“SIP/2138-ce60”, “outbound-callerid|1”) in new stack
– Executing GotoIf(“SIP/2138-ce60”, “1?start”) in new stack
– Goto (macro-outbound-callerid,s,3)
– Executing NoOp(“SIP/2138-ce60”, “REALCALLERIDNUM is 2138”) in new stack
– Executing Set(“SIP/2138-ce60”, “USEROUTCID=”) in new stack
– Executing Set(“SIP/2138-ce60”, “EMERGENCYCID=”) in new stack
– Executing Set(“SIP/2138-ce60”, “TRUNKOUTCID=”) in new stack
– Executing GotoIf(“SIP/2138-ce60”, “1?trunkcid”) in new stack
– Goto (macro-outbound-callerid,s,11)
– Executing GotoIf(“SIP/2138-ce60”, “1?usercid”) in new stack
– Goto (macro-outbound-callerid,s,13)
– Executing GotoIf(“SIP/2138-ce60”, “1?report”) in new stack
– Goto (macro-outbound-callerid,s,15)
– Executing NoOp(“SIP/2138-ce60”, “CallerID set to “LDP Sekhar” <2138>”) in new stack
– Executing Set(“SIP/2138-ce60”, “GROUP()=OUT_1”) in new stack
– Executing GotoIf(“SIP/2138-ce60”, “0?108”) in new stack
– Executing Set(“SIP/2138-ce60”, “DIAL_NUMBER=9083384347”) in new stack
– Executing Set(“SIP/2138-ce60”, “DIAL_TRUNK=1”) in new stack
– Executing AGI(“SIP/2138-ce60”, “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/2138-ce60”, “OUTNUM=9083384347”) in new stack
– Executing Set(“SIP/2138-ce60”, “custom=ZAP/g0”) in new stack
– Executing GotoIf(“SIP/2138-ce60”, “0?16”) in new stack
– Executing Dial(“SIP/2138-ce60”, “ZAP/g0/9083384347|120|r”) in new stack
== Everyone is busy/congested at this time (1:0/0/1)
– Executing Goto(“SIP/2138-ce60”, “s-CHANUNAVAIL|1”) in new stack
– Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
– Executing NoOp(“SIP/2138-ce60”, “Dial failed due to CHANUNAVAIL”) in new stack
– Executing Macro(“SIP/2138-ce60”, “outisbusy|”) in new stack
– Executing Playback(“SIP/2138-ce60”, “all-circuits-busy-now”) in new stack
– Playing ‘all-circuits-busy-now’ (language ‘en’)
– Executing Playback(“SIP/2138-ce60”, “pls-try-call-later”) in new stack
– Playing ‘pls-try-call-later’ (language ‘en’)
– Executing Macro(“SIP/2138-ce60”, “hangupcall”) in new stack
– Executing ResetCDR(“SIP/2138-ce60”, “w”) in new stack
– Executing NoCDR(“SIP/2138-ce60”, “”) in new stack
– Executing Wait(“SIP/2138-ce60”, “5”) in new stack

Please give me any suggestion. If some can post screen shots or configuration sample informaiton on TRIXBOX Trunks, Inbound & Outbound routes…that would be a great help.

thanks
Sekhar.

From the CLI command line what does

pri show span 1

and

pri show span 2

show? Do they both show as provisioned, up, and active?

Of interest is your zapata.conf file.

Hi Teran

I will soon submit the out of commands, meanwhil, how to configured zapata.conf on trixbox 1.1, i guess it will take care automatically. Shoud i just copy below configuration at /etc/zapata.conf?

[trunkgroups]

[channels]
group=1
context=default
signalling=pri_cpe
rxwink=200
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel = yes
echocancelwhenbridged = yes
echotraining = yes
; This is needed or some dtmf will be missed in voicemail
;echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
callgroup=1
pickupgroup=1
emdigitwait=700
immediate=no
usecallerid=yes
channel => 1-23
pridialplan=unknown

group=2
context=default
signalling=pri_cpe
rxwink=200
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel = yes
echocancelwhenbridged = yes
echotraining = yes
; This is needed or some dtmf will be missed in voicemail
;echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
callgroup=1
pickupgroup=1
emdigitwait=700
immediate=no
usecallerid=yes
channel => 24-47
pridialplan=unknown

Thanks.

Hi,

Here is the debug output. plese advice.


asterisk1CLI> pri show
debug span spans
asterisk1
CLI> pri show spans
asterisk1CLI> pri show span 1
No PRI running on span 1
asterisk1
CLI> pri show span 2
No PRI running on span 2
– Executing Macro(“SIP/2138-3ed8”, “dialout-trunk|1|19083384347||”) in new stack
– Executing GotoIf(“SIP/2138-3ed8”, “1?3:2”) in new stack
– Goto (macro-dialout-trunk,s,3)
– Executing Macro(“SIP/2138-3ed8”, “user-callerid”) in new stack
– Executing GotoIf(“SIP/2138-3ed8”, “0?report”) in new stack
– Executing GotoIf(“SIP/2138-3ed8”, “0?start”) in new stack
– Executing Set(“SIP/2138-3ed8”, “REALCALLERIDNUM=2138”) in new stack
– Executing NoOp(“SIP/2138-3ed8”, “REALCALLERIDNUM is 2138”) in new stack
– Executing Set(“SIP/2138-3ed8”, “AMPUSER=2138”) in new stack
– Executing Set(“SIP/2138-3ed8”, “AMPUSERCIDNAME=Sekhar Lakkoju”) in new stack
– Executing GotoIf(“SIP/2138-3ed8”, “0?report”) in new stack
– Executing Set(“SIP/2138-3ed8”, “CALLERID(all)=Sekhar Lakkoju <2138>”) in new stack
– Executing NoOp(“SIP/2138-3ed8”, “Using CallerID “Sekhar Lakkoju” <2138>”) in new stack
– Executing Macro(“SIP/2138-3ed8”, “record-enable|2138|OUT”) in new stack
– Executing GotoIf(“SIP/2138-3ed8”, “0 > 0?2:4”) in new stack
– Goto (macro-record-enable,s,4)
– Executing AGI(“SIP/2138-3ed8”, “recordingcheck|20060717-184524|1153176324.36”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20060717-184524|1153176324.36: Outbound recording not enabled
– AGI Script recordingcheck completed, returning 0
– Executing NoOp(“SIP/2138-3ed8”, “No recording needed”) in new stack
– Executing Macro(“SIP/2138-3ed8”, “outbound-callerid|1”) in new stack
– Executing GotoIf(“SIP/2138-3ed8”, “1?start”) in new stack
– Goto (macro-outbound-callerid,s,3)
– Executing NoOp(“SIP/2138-3ed8”, “REALCALLERIDNUM is 2138”) in new stack
– Executing Set(“SIP/2138-3ed8”, “USEROUTCID=”) in new stack
– Executing Set(“SIP/2138-3ed8”, “EMERGENCYCID=”) in new stack
– Executing Set(“SIP/2138-3ed8”, “TRUNKOUTCID=”) in new stack
– Executing GotoIf(“SIP/2138-3ed8”, “1?trunkcid”) in new stack
– Goto (macro-outbound-callerid,s,11)
– Executing GotoIf(“SIP/2138-3ed8”, “1?usercid”) in new stack
– Goto (macro-outbound-callerid,s,13)
– Executing GotoIf(“SIP/2138-3ed8”, “1?report”) in new stack
– Goto (macro-outbound-callerid,s,15)
– Executing NoOp(“SIP/2138-3ed8”, “CallerID set to “Sekhar Lakkoju” <2138>”) in new stack
– Executing Set(“SIP/2138-3ed8”, “GROUP()=OUT_1”) in new stack
– Executing GotoIf(“SIP/2138-3ed8”, “0?108”) in new stack
– Executing Set(“SIP/2138-3ed8”, “DIAL_NUMBER=19083384347”) in new stack
– Executing Set(“SIP/2138-3ed8”, “DIAL_TRUNK=1”) in new stack
– Executing AGI(“SIP/2138-3ed8”, “fixlocalprefix”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
– AGI Script fixlocalprefix completed, returning 0
– Executing Set(“SIP/2138-3ed8”, “OUTNUM=19083384347”) in new stack
– Executing Set(“SIP/2138-3ed8”, “custom=ZAP/g0”) in new stack
– Executing GotoIf(“SIP/2138-3ed8”, “0?16”) in new stack
– Executing Dial(“SIP/2138-3ed8”, “ZAP/g0/19083384347|120|r”) in new stack
== Everyone is busy/congested at this time (1:0/0/1)
– Executing Goto(“SIP/2138-3ed8”, “s-CHANUNAVAIL|1”) in new stack
– Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
– Executing NoOp(“SIP/2138-3ed8”, “Dial failed due to CHANUNAVAIL”) in new stack
– Executing Macro(“SIP/2138-3ed8”, “outisbusy|”) in new stack
– Executing Playback(“SIP/2138-3ed8”, “all-circuits-busy-now”) in new stack
– Playing ‘all-circuits-busy-now’ (language ‘en’)
– Executing Playback(“SIP/2138-3ed8”, “pls-try-call-later”) in new stack
– Playing ‘pls-try-call-later’ (language ‘en’)
– Executing Macro(“SIP/2138-3ed8”, “hangupcall”) in new stack
– Executing ResetCDR(“SIP/2138-3ed8”, “w”) in new stack
– Executing NoCDR(“SIP/2138-3ed8”, “”) in new stack
– Executing Wait(“SIP/2138-3ed8”, “5”) in new stack
– Executing Hangup(“SIP/2138-3ed8”, “”) in new stack

I am getting “All circuits are busy now. Please try your call later” when i dialout. Also, i dont see any debug log when i call 732-362-2138 from my cell phone. (no incoming calls at all).

Thanks
Sekhar.

what does the output of “ztcfg -vvv” look like ? and zaptel.conf and zapata.conf ?

Hi bacon buttie,

Here is the output,


[root@asterisk1 ~]# ztcfg -vvv

Zaptel Configuration

SPAN 1: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1)
SPAN 2: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1)

Channel map:

Channel 01: Clear channel (Default) (Slaves: 01)
Channel 02: Clear channel (Default) (Slaves: 02)
Channel 03: Clear channel (Default) (Slaves: 03)
Channel 04: Clear channel (Default) (Slaves: 04)
Channel 05: Clear channel (Default) (Slaves: 05)
Channel 06: Clear channel (Default) (Slaves: 06)
Channel 07: Clear channel (Default) (Slaves: 07)
Channel 08: Clear channel (Default) (Slaves: 08)
Channel 09: Clear channel (Default) (Slaves: 09)
Channel 10: Clear channel (Default) (Slaves: 10)
Channel 11: Clear channel (Default) (Slaves: 11)
Channel 12: Clear channel (Default) (Slaves: 12)
Channel 13: Clear channel (Default) (Slaves: 13)
Channel 14: Clear channel (Default) (Slaves: 14)
Channel 15: Clear channel (Default) (Slaves: 15)
Channel 16: Clear channel (Default) (Slaves: 16)
Channel 17: Clear channel (Default) (Slaves: 17)
Channel 18: Clear channel (Default) (Slaves: 18)
Channel 19: Clear channel (Default) (Slaves: 19)
Channel 20: Clear channel (Default) (Slaves: 20)
Channel 21: Clear channel (Default) (Slaves: 21)
Channel 22: Clear channel (Default) (Slaves: 22)
Channel 23: Clear channel (Default) (Slaves: 23)
Channel 24: D-channel (Default) (Slaves: 24)
Channel 25: Clear channel (Default) (Slaves: 25)
Channel 26: Clear channel (Default) (Slaves: 26)
Channel 27: Clear channel (Default) (Slaves: 27)
Channel 28: Clear channel (Default) (Slaves: 28)
Channel 29: Clear channel (Default) (Slaves: 29)
Channel 30: Clear channel (Default) (Slaves: 30)
Channel 31: Clear channel (Default) (Slaves: 31)
Channel 32: Clear channel (Default) (Slaves: 32)
Channel 33: Clear channel (Default) (Slaves: 33)
Channel 34: Clear channel (Default) (Slaves: 34)
Channel 35: Clear channel (Default) (Slaves: 35)
Channel 36: Clear channel (Default) (Slaves: 36)
Channel 37: Clear channel (Default) (Slaves: 37)
Channel 38: Clear channel (Default) (Slaves: 38)
Channel 39: Clear channel (Default) (Slaves: 39)
Channel 40: Clear channel (Default) (Slaves: 40)
Channel 41: Clear channel (Default) (Slaves: 41)
Channel 42: Clear channel (Default) (Slaves: 42)
Channel 43: Clear channel (Default) (Slaves: 43)
Channel 44: Clear channel (Default) (Slaves: 44)
Channel 45: Clear channel (Default) (Slaves: 45)
Channel 46: Clear channel (Default) (Slaves: 46)
Channel 47: Clear channel (Default) (Slaves: 47)
Channel 48: D-channel (Default) (Slaves: 48)

48 channels configured.


[root@asterisk1 etc]# cat zapata.conf
zapata.conf

[trunkgroups]

[channels]
group=1
context=default
signalling=pri_cpe
rxwink=200
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel = yes
echocancelwhenbridged = yes
echotraining = yes
; This is needed or some dtmf will be missed in voicemail
;echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
callgroup=1
pickupgroup=1
emdigitwait=700
immediate=no
usecallerid=yes
channel => 1-23
pridialplan=unknown

group=2
context=default
signalling=pri_cpe
rxwink=200
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel = yes
echocancelwhenbridged = yes
echotraining = yes
; This is needed or some dtmf will be missed in voicemail
;echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
callgroup=1
pickupgroup=1
emdigitwait=700
immediate=no
usecallerid=yes
channel => 24-48
pridialplan=unknown


[root@asterisk1 etc]# cat zaptel.conf
span=1,1,0,esf,b8zs
span=2,1,0,esf,b8zs
bchan=1-23
dchan=24
bchan=25-47
dchan=48
loadzone = us
defaultzone=us


you’re defining groups 1 and 2 in zapata.conf, and trying to send calls via a non-existent g0. somewhere in the GUI is an option to set the group identifier for a ZAP trunk, make your changes there.

Hi,

Thanks for the help,

I changed the GUI outbound trunk to ZAP/g1. How do assing ZAP/g2 for incoming? where should i change the configuration. also, my sip ext. are only 4 digit, i need get 10 number when i call any outside number.

i think you can create a second ZAP trunk with the g2 identifier, and set this as your incoming trunk (Incoming Settings ?). it’s been a while since i looked at AMP or FreePBX !

Hi Everyone,

First, Thanks to baconbuttie & Teran for inputs. I still need help.

This weekend i tried all possible options to my knowledge and unable to resolve the incoming calls issue on Trixbox 1.1. I want to use trixbox as it has lot of tools and web based management.

I did configured the system with CentOS 4.3 and both Incoming & Outgoing calls are working fine. For some reason (may be my vendor), i have to put NXXX (last 4 digits of DID) in my extensions.conf file. Everything went well on CentOS4.3 server. But, i need this on Trixbox. Can Any of the seniors/experts please help me in resolving this.

i have TE205P card. g1 (span 1) configured for Outbout and g2 (span 2) configured for inbound. I assume it is minor mistake i am doing some where on the Trixbox GUI.

please give me some clue.

if you had this working previously, can’t you apply the changes you made there to your FreePBX setup ? it’s not as if Asterisk is any different.

i think you probably need to add a bunch of DID entries in FreePBX and route them as needed.

to be quite honest, if you had this working without TrixBox you should consider going back to it, and add on the features you want. it’s not as if any of the components that make up TrixBox are exclusive and you’ll learn more about Asterisk and your system as a result.