[asterisk-users] Dial out issues.
Dear all.
I have what appears to be a configuration error but I cannot for the life of me
see what it is. (I am a newbie)
I have searched the wikki and google etc but still none the wiser. Any help
would be very gratefully received.
Problem:
Unable to make outgoing calls via E1 euroISDN Digium TE110p card, given
congestion signal as per config, unable to open zap channel. All incoming calls
work well.
Error Message:
[May 22 11:01:48] WARNING[9179]: channel.c:3024 ast_request: No channel type
registered for '(Zap'
[May 22 11:01:48] WARNING[9179]: app_dial.c:1090 dial_exec_full: Unable to
create channel of type '(Zap' (cause 66 - Channel not implemented)
Configs:
[EMAIL PROTECTED] asterisk]# cat sip.conf
[general]
bindport=5060
bindaddr=0.0.0.0
context=default
disallow=all
allow=gsm
allow=ilbc
allow=ulaw
allow=alaw
srvlookup=yes
;
[400]
type=friend
username=400
host=dynamic
secret=12345
regexten=400
dtmfmode=rfc2833
canreinvite=yes
nat=no
mailbox=400
;
[401]
type=friend
username=401
host=dynamic
secret=12345
regexten=401
dtmfmode=rfc2833
canreinvite=yes
nat=no
mailbox=401
;
[402]
type=friend
username=402
host=dynamic
secret=12345
regexten=402
dtmfmode=rfc2833
canreinvite=yes
nat=no
mailbox=402
;
[410]
type=friend
username=410
host=dynamic
secret=12345
regexten=410
dtmfmode=rfc2833
canreinvite=yes
nat=no
mailbox=410
;
[421]
type=friend
username=421
host=dynamic
secret=12345
regexten=421
dtmfmode=rfc2833
canreinvite=yes
nat=no
mailbox=421
;
[450]
type=friend
username=450
host=dynamic
secret=12345
regexten=450
dtmfmode=rfc2833
canreinvite=yes
nat=no
mailbox=450
;
[451]
type=friend
username=451
host=dynamic
secret=12345
regexten=451
dtmfmode=rfc2833
canreinvite=yes
nat=no
mailbox=451
;
[452]
type=friend
username=452
host=dynamic
secret=12345
regexten=452
dtmfmode=rfc2833
canreinvite=yes
nat=no
mailbox=452
[EMAIL PROTECTED] asterisk]# cat extensions.conf
[general]
static=yes
writeprotect=yes
;
[globals]
OUTBOUND = Zap/g1
CUSTSERVE1 = SIP/401 ;Teresa
CUSTSERVE2 = SIP/402 ; Louise
;CUSTSERVE3 = SIP/404 ; Helen
QUAD1 = SIP/451 ; Matt
QUAD2 = SIP/452 ; Johan
CUSTSERVE = CUSTSERVE1CUSTSERVE1
;
FSEXT1 = SIP/400 ; Angela
;FSEXT2 = SIP/403 ; Nigel
FSEXT3 = SIP/410 ; Matt
;
;ELLIS = SIP/411
;QUEENS = SIP/412
;FSSHOPS = ELLISQUEENS
;
QUAD = SIP/450
;
LONDONSOLE1 = SIP/421 ; Zoe
;LONDONSOLE2 = SIP/422 ; Laura
;LONDONSOLE = LONDONSOLE1LONDONSOLE2
;
;PRESS1 = SIP/431 ; Lucy
;Press2 = SIP/432 ; Gemma
;PRESSOFFICE = PRESS1PRESS2
;
[macro-oneline]
exten = s,1,Dial(${ARG1},20,t)
exten = s,2,Voicemail(u${MACRO_EXTEN})
exten = s,3,Hangup
exten = s,102,Voicemail(b${MACRO_EXTEN})
exten = s,103,Hangup
;
[macro-oneline1]
exten = s,1,Dial(${ARG1},20,t)
exten = s,2,Voicemail(u${ARG2})
exten = s,3,Hangup
exten = s,102,Voicemail(b${ARG2})
exten = s,103,Hangup
;
[default]
;setupdial out
include = from-pstn
;
;test dialplan
exten = _9xxx,1,SayDigits(${EXTEN:1})
;
;setup the phone extensions
exten = 400,1,Macro(oneline,${FSEXT1})
exten = 401,1,Macro(oneline,${CUSTSERVE1})
exten = 402,1,Macro(oneline,${CUSTSERVE2})
exten = 410,1,Macro(oneline,${FSEXT3})
exten = 421,1,Macro(oneline,${LONDONSOLE1})
exten = 450,1,Macro(oneline,${QUAD})
exten = 451,1,Macro(oneline,${QUAD1})
exten = 452,1,Macro(oneline,${QUAD2})
;
exten = 1000,1,Macro(oneline,${CUSTSERVE})
;exten = 2000,1,Macro(oneline,${FSSHOPS})
;exten = 3000,1,Macro(oneline,${PRESSOFFICE})
;
;setup the dial out via te110p
;exten = _X.,1,SetCIDNum(00)
;
;record new voice files
Exten = 501,1,Wait(2)
Exten = 501,n,Record(/tmp/asterisk-recording:gsm)
Exten = 501,n,Wait(2)
Exten = 501,n,Playback(/tmp/asterisk-recording)
Exten = 501,n,wait(2)
Exten = 501,n,Hangup
;
;goto voicemail
exten=*98,1,VoiceMailMain([EMAIL PROTECTED])
;
[from-pstn]
;this is linked to zapata.conf and defines where the ddi points
exten = 00,1,Dial(SIP/401SIP/402,15)
exten = 00,2,Voicemail(1000)
;
exten = 769611,1,Macro(oneline1,${FSEXT1})
exten = 769615,1,Macro(oneline1,${LONDONSOLE1})
;exten = 769616,1,Macro(oneline1,${LONDONSOLE2})
exten = 769636,1,Macro(oneline1,${FSEXT1},${401})
;exten = 769637,1,Macro(oneline1,${NIGEL})
;
exten = _9xxx,1,Dial,(${OUTBOUND}/${EXTEN:1})
exten = _9xxx.,2,Congestion()
exten = _9xxx,102,Congestion()
;
exten = 999,1,Dial,(${OUTBOUND}/999)
exten = ,1,Dial,(${OUTBOUND}/999)
;
[EMAIL PROTECTED] asterisk]# more zapata.conf
[channels]
language=en
usecallerid=yes
hidecallerid=no
callwaiting=no
callwaitingcallerid=yes
restrictcid=no
usecallingpres=no
threewaycalling=yes
callreturn=yes
transfer=yes
cancallforward=yes
echocancelwhenbridged=yes
echocancel=yes
musiconhold=default
rxgain=0.0
txgain=0.0
signalling=pri_cpe
switchtype=euroisdn
immediate=no
overlapdial=yes
pridialplan=unknown
prilocaldialplan=unknown
group=1
context = from-pstn
callerid=asreceived
channel = 1-8
Specs:
New IBM hardware, Intel 4 350mhz 512gig RAM
Digium E1 Card TE110P
Linux Fedcore4
asterisk 1.4
zaptel 1.4
libpri 1.4___
--Bandwidth and
Re: [asterisk-users] Dial out issues.
On 5/22/07, Matt Scott [EMAIL PROTECTED] wrote: Dear all. I have what appears to be a configuration error but I cannot for the life of me see what it is. (I am a newbie) I have searched the wikki and google etc but still none the wiser. Any help would be very gratefully received. Problem: Unable to make outgoing calls via E1 euroISDN Digium TE110p card, given congestion signal as per config, unable to open zap channel. All incoming calls work well. Error Message: [May 22 11:01:48] WARNING[9179]: channel.c:3024 ast_request: No channel type registered for '(Zap' [May 22 11:01:48] WARNING[9179]: app_dial.c:1090 dial_exec_full: Unable to create channel of type '(Zap' (cause 66 - Channel not implemented) It says channel not implemented. Did you compile asterisk before or after compiling and installing libpri? If you type module load chan_zap.so, what is the output? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Dial out issues.
In your dial lines you have an extrac comma (,)
exten = _9xxx,1,Dial,(${OUTBOUND}/${EXTEN:1})
should be
exten = _9xxx,1,Dial(${OUTBOUND}/${EXTEN:1})
or
exten = _9xxx,1,Dial,${OUTBOUND}/${EXTEN:1}
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt Scott
Sent: 22 May 2007 13:11
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Dial out issues.
Dear all.
I have what appears to be a configuration error but I cannot for the
life of me see what it is. (I am a newbie)
I have searched the wikki and google etc but still none the wiser. Any
help would be very gratefully received.
Problem:
Unable to make outgoing calls via E1 euroISDN Digium TE110p card, given
congestion signal as per config, unable to open zap channel. All
incoming calls work well.
Error Message:
[May 22 11:01:48] WARNING[9179]: channel.c:3024 ast_request: No channel
type registered for '(Zap'
[May 22 11:01:48] WARNING[9179]: app_dial.c:1090 dial_exec_full: Unable
to create channel of type '(Zap' (cause 66 - Channel not implemented)
Configs:
[EMAIL PROTECTED] asterisk]# cat sip.conf
[general]
bindport=5060
bindaddr=0.0.0.0
context=default
disallow=all
allow=gsm
allow=ilbc
allow=ulaw
allow=alaw
srvlookup=yes
;
[400]
type=friend
username=400
host=dynamic
secret=12345
regexten=400
dtmfmode=rfc2833
canreinvite=yes
nat=no
mailbox=400
;
[401]
type=friend
username=401
host=dynamic
secret=12345
regexten=401
dtmfmode=rfc2833
canreinvite=yes
nat=no
mailbox=401
;
[402]
type=friend
username=402
host=dynamic
secret=12345
regexten=402
dtmfmode=rfc2833
canreinvite=yes
nat=no
mailbox=402
;
[410]
type=friend
username=410
host=dynamic
secret=12345
regexten=410
dtmfmode=rfc2833
canreinvite=yes
nat=no
mailbox=410
;
[421]
type=friend
username=421
host=dynamic
secret=12345
regexten=421
dtmfmode=rfc2833
canreinvite=yes
nat=no
mailbox=421
;
[450]
type=friend
username=450
host=dynamic
secret=12345
regexten=450
dtmfmode=rfc2833
canreinvite=yes
nat=no
mailbox=450
;
[451]
type=friend
username=451
host=dynamic
secret=12345
regexten=451
dtmfmode=rfc2833
canreinvite=yes
nat=no
mailbox=451
;
[452]
type=friend
username=452
host=dynamic
secret=12345
regexten=452
dtmfmode=rfc2833
canreinvite=yes
nat=no
mailbox=452
[EMAIL PROTECTED] asterisk]# cat extensions.conf
[general]
static=yes
writeprotect=yes
;
[globals]
OUTBOUND = Zap/g1
CUSTSERVE1 = SIP/401 ;Teresa
CUSTSERVE2 = SIP/402 ; Louise
;CUSTSERVE3 = SIP/404 ; Helen
QUAD1 = SIP/451 ; Matt
QUAD2 = SIP/452 ; Johan
CUSTSERVE = CUSTSERVE1CUSTSERVE1
;
FSEXT1 = SIP/400 ; Angela
;FSEXT2 = SIP/403 ; Nigel
FSEXT3 = SIP/410 ; Matt
;
;ELLIS = SIP/411
;QUEENS = SIP/412
;FSSHOPS = ELLISQUEENS
;
QUAD = SIP/450
;
LONDONSOLE1 = SIP/421 ; Zoe
;LONDONSOLE2 = SIP/422 ; Laura
;LONDONSOLE = LONDONSOLE1LONDONSOLE2
;
;PRESS1 = SIP/431 ; Lucy
;Press2 = SIP/432 ; Gemma
;PRESSOFFICE = PRESS1PRESS2
;
[macro-oneline]
exten = s,1,Dial(${ARG1},20,t)
exten = s,2,Voicemail(u${MACRO_EXTEN})
exten = s,3,Hangup
exten = s,102,Voicemail(b${MACRO_EXTEN})
exten = s,103,Hangup
;
[macro-oneline1]
exten = s,1,Dial(${ARG1},20,t)
exten = s,2,Voicemail(u${ARG2})
exten = s,3,Hangup
exten = s,102,Voicemail(b${ARG2})
exten = s,103,Hangup
;
[default]
;setupdial out
include = from-pstn
;
;test dialplan
exten = _9xxx,1,SayDigits(${EXTEN:1})
;
;setup the phone extensions
exten = 400,1,Macro(oneline,${FSEXT1})
exten = 401,1,Macro(oneline,${CUSTSERVE1})
exten = 402,1,Macro(oneline,${CUSTSERVE2})
exten = 410,1,Macro(oneline,${FSEXT3})
exten = 421,1,Macro(oneline,${LONDONSOLE1})
exten = 450,1,Macro(oneline,${QUAD})
exten = 451,1,Macro(oneline,${QUAD1})
exten = 452,1,Macro(oneline,${QUAD2})
;
exten = 1000,1,Macro(oneline,${CUSTSERVE})
;exten = 2000,1,Macro(oneline,${FSSHOPS})
;exten = 3000,1,Macro(oneline,${PRESSOFFICE})
;
;setup the dial out via te110p
;exten = _X.,1,SetCIDNum(00)
;
;record new voice files
Exten = 501,1,Wait(2)
Exten = 501,n,Record(/tmp/asterisk-recording:gsm)
Exten = 501,n,Wait(2)
Exten = 501,n,Playback(/tmp/asterisk-recording)
Exten = 501,n,wait(2)
Exten = 501,n,Hangup
;
;goto voicemail
exten=*98,1,VoiceMailMain([EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] })
;
[from-pstn]
;this is linked to zapata.conf and defines where the ddi points
exten = 00,1,Dial(SIP/401SIP/402,15)
exten = 00,2,Voicemail(1000)
;
exten = 769611,1,Macro(oneline1,${FSEXT1})
exten = 769615,1,Macro(oneline1,${LONDONSOLE1})
;exten = 769616,1,Macro(oneline1,${LONDONSOLE2})
exten = 769636,1,Macro(oneline1,${FSEXT1},${401})
;exten = 769637,1,Macro(oneline1,${NIGEL})
;
exten = _9xxx,1,Dial,(${OUTBOUND}/${EXTEN:1})
exten = _9xxx.,2,Congestion()
exten = _9xxx,102,Congestion()
;
exten = 999,1,Dial,(${OUTBOUND}/999)
exten = ,1,Dial,(${OUTBOUND}/999)
;
[EMAIL PROTECTED] asterisk]# more zapata.conf
[channels]
language=en
usecallerid=yes
hidecallerid=no
callwaiting=no
callwaitingcallerid=yes
restrictcid=no
usecallingpres=no
threewaycalling=yes
callreturn
Re: [asterisk-users] Dial out issues.
On 5/22/07, Morgan Gilroy [EMAIL PROTECTED] wrote:
In your dial lines you have an extrac comma (,)
exten = _9xxx,1,Dial,(${OUTBOUND}/${EXTEN:1})
should be
exten = _9xxx,1,Dial(${OUTBOUND}/${EXTEN:1})
or
exten = _9xxx,1,Dial,${OUTBOUND}/${EXTEN:1}
Good catch Morgan!
___
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