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:
[root@asterisk 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
[root@asterisk 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 = CUSTSERVE1&CUSTSERVE1
;
FSEXT1 = SIP/400 ; Angela
;FSEXT2 = SIP/403 ; Nigel
FSEXT3 = SIP/410 ; Matt
;
;ELLIS = SIP/411
;QUEENS = SIP/412
;FSSHOPS = ELLIS&QUEENS
;
QUAD = SIP/450
;
LONDONSOLE1 = SIP/421 ; Zoe
;LONDONSOLE2 = SIP/422 ; Laura
;LONDONSOLE = LONDONSOLE1&LONDONSOLE2
;
;PRESS1 = SIP/431 ; Lucy
;Press2 = SIP/432 ; Gemma
;PRESSOFFICE = PRESS1&PRESS2
;
[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(888800)
;
;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(${CALLERIDNUM}@${CONTEXT})
;
[from-pstn]
;this is linked to zapata.conf and defines where the ddi points
exten => 888800,1,Dial(SIP/401&SIP/402,15)
exten => 888800,2,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 => _9xxxxxxxxxxx,1,Dial,(${OUTBOUND}/${EXTEN:1})
exten => _9xxxxxxxxxxx.,2,Congestion()
exten => _9xxxxxxxxxxx,102,Congestion()
;
exten => 999,1,Dial,(${OUTBOUND}/999)
exten => 9999,1,Dial,(${OUTBOUND}/999)
;
[root@asterisk 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
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On 5/22/07, Matt Scott <matt@mgsnet.co.uk> 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?
In your dial lines you have an extrac comma (,)
exten => _9xxxxxxxxxxx,1,Dial,(${OUTBOUND}/${EXTEN:1})
should be
exten => _9xxxxxxxxxxx,1,Dial(${OUTBOUND}/${EXTEN:1})
or
exten => _9xxxxxxxxxxx,1,Dial,${OUTBOUND}/${EXTEN:1}
________________________________
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] 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:
[root@asterisk 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
[root@asterisk 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 = CUSTSERVE1&CUSTSERVE1
;
FSEXT1 = SIP/400 ; Angela
;FSEXT2 = SIP/403 ; Nigel
FSEXT3 = SIP/410 ; Matt
;
;ELLIS = SIP/411
;QUEENS = SIP/412
;FSSHOPS = ELLIS&QUEENS
;
QUAD = SIP/450
;
LONDONSOLE1 = SIP/421 ; Zoe
;LONDONSOLE2 = SIP/422 ; Laura
;LONDONSOLE = LONDONSOLE1&LONDONSOLE2
;
;PRESS1 = SIP/431 ; Lucy
;Press2 = SIP/432 ; Gemma
;PRESSOFFICE = PRESS1&PRESS2
;
[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(888800)
;
;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(${CALLERIDNUM}@${CONTEXT
<mailto:$%7bCALLERIDNUM%7d@$%7bCONTEXT> })
;
[from-pstn]
;this is linked to zapata.conf and defines where the ddi points
exten => 888800,1,Dial(SIP/401&SIP/402,15)
exten => 888800,2,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 => _9xxxxxxxxxxx,1,Dial,(${OUTBOUND}/${EXTEN:1})
exten => _9xxxxxxxxxxx.,2,Congestion()
exten => _9xxxxxxxxxxx,102,Congestion()
;
exten => 999,1,Dial,(${OUTBOUND}/999)
exten => 9999,1,Dial,(${OUTBOUND}/999)
;
[root@asterisk 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
______________________________________________________________________
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On 5/22/07, Morgan Gilroy <morgan@telappliant.com> wrote:> In your dial lines you have an extrac comma (,) > > exten => _9xxxxxxxxxxx,1,Dial,(${OUTBOUND}/${EXTEN:1}) > > should be > > exten => _9xxxxxxxxxxx,1,Dial(${OUTBOUND}/${EXTEN:1}) > > or > > exten => _9xxxxxxxxxxx,1,Dial,${OUTBOUND}/${EXTEN:1}Good catch Morgan!