[asterisk-users] tdm400p fxs module busy
Dear All
The setup is te110p with an 8 channels PRI to make and receive all calls.
SIP phones throughout the company.
TDM400p with 4 FXS modules to send/receive faxes and make credit card
transactions.
I have an analogue phone on the tdm400p for testing.
I can receive calls to the exten. There is a dialing tone.
However, when I try to make a call I get a busy signal.
Asterisk stated busy then hungup zap/32-1
why wont asterisk supply a resource from the te110p pri card for use by the
tdm400p FXS (fxo signalling)?
configs below:
[EMAIL PROTECTED] etc]# more zaptel.conf
# Autogenerated by /usr/sbin/genzaptelconf -- do not hand edit
# Zaptel Configuration File
#
# This file is parsed by the Zaptel Configurator, ztcfg
#
# It must be in the module loading order
# Span 1: WCT1/0 Digium Wildcard TE110P T1/E1 Card 0 HDB3/CCS RED
span = 1,0,0,ccs,hdb3,crc4
# termtype: te
bchan=1-8
dchan=16
# Span 2: WCTDM/0 Wildcard TDM400P REV H Board 1
fxoks=32
fxoks=33
fxoks=34
fxoks=35
# Global data
loadzone= uk
defaultzone = uk
[EMAIL PROTECTED] asterisk]# more zapata.conf
[trunkgroups]
[channels]
language=en
internationalprefix = 00
nationalprefix = 0
context=from-pstn
switchtype=euroisdn
pridialplan=local
priindication=outofband
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
group=1
callgroup=0
pickupgroup=0
immediate=no
echotraining=yes
echocancel=yes
echocancelwhenbridged=no
facilityenable=yes
musiconhold=default
overlapdial=yes
immediate=no
txgain=0.0
rxgain=0.0
signalling = pri_cpe
channel = 1-8
faxdetect=both
;faxdetect=incoming
;faxdetect=outgoing
;faxdetect=no
signalling = fxo_ks
echocancel=yes
pulsedial=yes
channel=32-35
[EMAIL PROTECTED] asterisk]# more extensions.conf
[general]
static=yes
writeprotect=yes
;
[globals]
OUTBOUND = Zap/g1
FAX1 = Zap/32
FAX2 = Zap/33
STREAMLINE1 = Zap/34
STREAMLINE2 = Zap/35
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
;
[macro-fax]
exten = s,1,Dial(${ARG1},20,t)
exten = s,3,Hangup
;
[default]
;setupdial out
include = from-pstn
;
;test dialplan
exten = _9xxx,1,SayDigits(${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,${PRESSOFFICE})
;
;record new voice files
Exten = 501,1,Wait(2)
Exten = 501,n,Record(/tmp/asterisk-recording:gsm)
Exten = 501,n,Wait(2)
Exten = 501,n,Playback(/tmp/asterisk-recording)
Exten = 501,n,wait(2)
Exten = 501,n,Hangup
;
;goto voicemail
exten=*98,1,VoiceMailMain([EMAIL PROTECTED])
;
[dialphone]
exten = 90,1,Macro(fax,${FAX1})
;
[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 = _9.,1,Set(CALLERID(number)=00)
exten = _9.,2,Dial(${OUTBOUND}/${EXTEN:1})
exten = _9.,3,Congestion()
exten = _9.,102,Congestion()
;
exten = 999,1,Dial,(${OUTBOUND}/999)
exten = ,1,Dial,(${OUTBOUND}/999)
;
exten = 90,1,Dial(Zap/32,15)
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[asterisk-users] TDM400 one way calls
Dear All
I have a problem with a TDM400 card with 4 x FXS modules.
The card carries extensions only and there are no incoming lines.
I can make a call to the extension on this card with no problems.
However, when I try and call out I just get a busy signal.
I also get an error message (as shown at the bottom). Is this a problem?
Configs below:
[EMAIL PROTECTED] etc]# more zaptel.conf
fxoks=1-4
loadzone=uk
defaultzone=uk
[EMAIL PROTECTED] asterisk]# more zapata.conf
[trunkgroups]
;define trunks here
[channels]
usecallerid=yes
hidecallerid=no
callwaiting=no
threewaycalling=yes
transfer=yes
echocancel=yes
echotraining=yes
immediate=no
;define channels
context=dialphone
signalling=fxo_ks
cidsignalling=v23 ; Added for UK CLI detection
cidstart=polarity
usecallerid=yes
channel = 1-4
[EMAIL PROTECTED] asterisk]# more extensions.conf
[general]
static=yes
writeprotect=yes
;
[globals]
FAX1 = Zap/1
FAX2 = Zap/2
STREAMLINE1 = Zap/3
STREAMLINE2 = Zap/4
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
;
;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})
;
[dialphone]
exten = 601,1,Macro(oneline,${FAX1})
;
asterisk*CLI reload chan_zap.so
-- Reloading module 'chan_zap.so' (Zapata Telephony)
== Parsing '/etc/asterisk/zapata.conf': Found
[Jun 21 10:24:26] WARNING[29786]: chan_zap.c:11072 process_zap: Ignoring
signalling
-- Reconfigured channel 1, FXO Kewlstart signalling
-- Reconfigured channel 2, FXO Kewlstart signalling
-- Reconfigured channel 3, FXO Kewlstart signalling
-- Reconfigured channel 4, FXO Kewlstart signalling
== Parsing '/etc/asterisk/users.conf': Found___
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[asterisk-users] TDM400 one way calls
Dear All
I have a problem with a TDM400 card with 4 x FXS modules.
The card carries extensions only and there are no incoming lines.
I can make a call to the extension on this card with no problems.
However, when I try and call a different extension I just get a busy signal.
I also get an error message (as shown at the bottom). Is this a problem?
Configs below:
[EMAIL PROTECTED] etc]# more zaptel.conf
fxoks=1-4
loadzone=uk
defaultzone=uk
[EMAIL PROTECTED] asterisk]# more zapata.conf
[trunkgroups]
;define trunks here
[channels]
usecallerid=yes
hidecallerid=no
callwaiting=no
threewaycalling=yes
transfer=yes
echocancel=yes
echotraining=yes
immediate=no
;define channels
context=dialphone
signalling=fxo_ks
cidsignalling=v23 ; Added for UK CLI detection
cidstart=polarity
usecallerid=yes
channel = 1-4
[EMAIL PROTECTED] asterisk]# more extensions.conf
[general]
static=yes
writeprotect=yes
;
[globals]
FAX1 = Zap/1
FAX2 = Zap/2
STREAMLINE1 = Zap/3
STREAMLINE2 = Zap/4
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${ARG2})
exten = s,103,Hangup
;
;
[default]
;setupdial out
;
;test dialplan
exten = _9xxx,1,SayDigits(${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,${PRESSOFFICE})
;
[dialphone]
exten = 601,1,Macro(oneline,${FAX1})
;
asterisk*CLI reload chan_zap.so
-- Reloading module 'chan_zap.so' (Zapata Telephony)
== Parsing '/etc/asterisk/zapata.conf': Found
[Jun 21 10:24:26] WARNING[29786]: chan_zap.c:11072 process_zap: Ignoring
signalling
-- Reconfigured channel 1, FXO Kewlstart signalling
-- Reconfigured channel 2, FXO Kewlstart signalling
-- Reconfigured channel 3, FXO Kewlstart signalling
-- Reconfigured channel 4, FXO Kewlstart signalling
== Parsing '/etc/asterisk/users.conf': Found___
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[asterisk-users] TDM400p and te110p configuration.
Dear users. My current setup uses a euroISDN E1 with 8 cahnnels for incoming and outgoing calls from a digium te110p. Currently all phones use SIP. However, I need to add some faxes lines and some POS credit card machines. These will require POTS lines with a fixed DDI. I have purchased the tdm400p and 4 FXS modules. My problem is with the zaptel.conf and zapata.conf. I am a little confused as how to separate the specific requirements for each card. How do I create a span for the tdm400p? I would imagine they require their own context and specific group? Also the channel numbers become a bit of a problem. Do they become sequential carrying on from each card I would imagine I need to modprobe the correct drivers for this card as well. Will there be any conflict? Here is my current zaptel and zapata confs with the te110p requirements ONLY. zaptel: loadzone = uk defaultzone = uk span = 1,0,0,ccs,hdb3,crc4 bchan = 1-15,17-31 dchan = 16 zapata: [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 Please would someone start me off in the right direction for adding these additional FXS devices.___ --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] TDM400p and te110p configuration.
I purchased FXS modules so that I could terminate the machines or faxes (eg just like a standard phone) the outgoing/incoming channel will be be provided by my E1. I hope I have the right modules for the job? - Original Message - From: Tzafrir Cohen [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Thursday, June 14, 2007 11:22 AM Subject: Re: [asterisk-users] TDM400p and te110p configuration. On Thu, Jun 14, 2007 at 09:45:01AM +0100, Matt Scott wrote: Dear users. My current setup uses a euroISDN E1 with 8 cahnnels for incoming and outgoing calls from a digium te110p. Currently all phones use SIP. However, I need to add some faxes lines and some POS credit card machines. These will require POTS lines with a fixed DDI. I have purchased the tdm400p and 4 FXS modules. *FXO* modules, right? My problem is with the zaptel.conf and zapata.conf. I am a little confused as how to separate the specific requirements for each card. How do I create a span for the tdm400p? You don't . Just 'fxsks' lines which look like bchan/dchan lines in zaptel.conf. In zapata.conf they are the same channels (with fxs_ks signalling). I would imagine they require their own context and specific group? Right. Also the channel numbers become a bit of a problem. Do they become sequential carrying on from each card cat /proc/zaptel/* I would imagine I need to modprobe the correct drivers for this card as well. Will there be any conflict? What type of conflict? The number of a channel is set when you load a driver (technically: when you register its spans to zaptel). If you want to make sure that the current channels of the E1 card keep their numbers, you should load the analog card second. On Debian systems you can guarantee that by e.g. putting the module names in the proper order in /etc/modules . Here is my current zaptel and zapata confs with the te110p requirements ONLY. zaptel: loadzone = uk defaultzone = uk span = 1,0,0,ccs,hdb3,crc4 bchan = 1-15,17-31 dchan = 16 # Something of the sort of: fxsks = 32-35 zapata: [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 ; something of the sort of: ; context = from-pots ; group = 2 cidsignalling = v23 cidstart = polarity channel = 32-35 Alternatively use genzaptelconf, but be sure to set lc_country to uk. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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 ___ --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
[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
[Asterisk-Users] Problem with FXO taking a call
Hi all.
I am unable to answer calls coming into asterisk
over PSTN. (UK)I want to have a call answered by my TDM400P/FXO module and
forwarded to a sip phone.
When I make a call from the PSTN to the BT line
installed on my FXO module the sip phone rings however, when i pick up
thecall using the sip phone, the incoming call is not answered/routed by
asterisk. As a result the sip phone is left hangingand the incoming call
remains unanswered.my zapata.conf now looks like
this.-; Configuration
file;[channels]language=ukgroup=1context=from-pstnusecallerid=nocidstart=polaritysignalling=fxs_kschannel
= 4-debug
info-*CLI
== Starting post polarity CID detection on channel 4 --
Starting simple switch on 'Zap/4-1' -- Executing
NoOp("Zap/4-1", "--- calling on 01189xxx (s) ---") innew
stack -- Executing Dial("Zap/4-1", "SIP/1001|20") in new
stack -- Called 1001 --
SIP/1001-5c18 is ringing -- SIP/1001-5c18 answered
Zap/4-1May 24 11:12:35 WARNING[32757]: chan_zap.c:3646
zt_handle_event:Ring/Off-hook in strange state 6 on channel 4 ==
Spawn extension (from-pstn, s, 2) exited non-zero on
'Zap/4-1' -- Hungup
'Zap/4-1'---
All the other variations of my configuration works
well, it is just this part.
Any help much
appreciated.
___
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[Asterisk-Users] tdm400p fxo not working
Dear all. I have a tdm400p with an FXO module in slot 4 and an FXS module in slot 1. I have not configured the FXS port in an attempt to keep things simple. The problem is that when I call the POTS number (assigned by phone company) asterisk is seeing the call but then not doing anything with it. The verbose output from asterisk is as follows: -- *CLI == Starting post polarity CID detection on channel 4 -- Starting simple switch on 'Zap/4-1'May 19 15:10:29 NOTICE[30934]: chan_zap.c:5542 ss_thread: Got event 17 (Polarity Reversal)...May 19 15:10:31 WARNING[30934]: chan_zap.c:5582 ss_thread: CID timed out waiting for ring. Exiting simple switch -- Hungup 'Zap/4-1' --- From the caller end it just rings constantly. I have the following configurations: zaptel.conf fxsks=4loadzone=ukdefaultzone=uk zapata.conf ; Zapata telephony interface; Configuration file;[channels]language=ukgroup=1context=from-pstnsignalling=fxs_kschannel = 4 extensions.conf [from-pstn]exten = s,1,Dial(SIP/1001,20)exten = s,2,Hangup The SIP elements of my system are working well, I just need to get this incoming call on a POTS line working. I have tried to keep things as simple as possible. Does anyone know why my call is not being handed to my sip phone? What is CID timed out waiting for ring? Is this something to do with caller ID? I have tried it with a 'wait' command in the extensions.conf as well but no joy. Kind regards ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] multiple sip accounts from same sip registrar
Hi Peter. I think I probably put my email rather badly. However you did manage to spot my problem and solve it for which I am very grateful!! The bottom line is you cannot have different context for the same sip provider, and it works as you state in your reply. Thanks again. Matt - Original Message - From: Peter Bowyer [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, May 18, 2005 8:25 AM Subject: Re: [Asterisk-Users] multiple sip accounts from same sip registrar On 17/05/05, Matt Scott [EMAIL PROTECTED] wrote: Dear all, I have an asterisk sip issue which I don't believe is unique. I use a registrar (sipgate.co.uk) where I have 3 different accounts. These accounts provide me with three seperate local phone numbers which allow me to allocate them to seperate users. By using just one of these accounts I can set asterisk up to send and receive calls no problem. However, when I start to introduce an additional account I start to run into problems. if I do a 'sip show peers' with a good config I think it may outline the problem sip show peers Name/username HostDyn Nat ACL Mask Port Status 1005/1005 (Unspecified)D 255.255.255.255 0 Unmonitored 1004/1004 (Unspecified)D 255.255.255.255 0 Unmonitored 1003/1003 (Unspecified)D 255.255.255.255 0 Unmonitored 1002/1002 10.0.0.52D 255.255.255.255 5060 Unmonitored 1001/1001 10.0.0.51D 255.255.255.255 5060 Unmonitored sipgate1/321 217.10.79.219N 255.255.255.255 5060 OK (52 ms) I'm not sure what you think the problem is, you haven't told us... but anyway, I haven't succeeded in sending sipgate inbound calls through separate contexts, but I deal with them all in a single context - the calls will arrive at an extension matching the individual sipgate username in the register command. Works for me and several others Peter -- Peter Bowyer Email: [EMAIL PROTECTED] Tel: +44 1296 768003 VoIP: sip:[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] multiple sip accounts from same sip registrar
Dear all, I have an asterisk sip issue which I don't believe is unique. I use a registrar (sipgate.co.uk) where I have 3 different accounts. These accounts provide me with three seperate local phone numbers which allow me to allocate them to seperate users. By using just one of these accounts I can set asterisk up to send and receive calls no problem. However, when I start to introduce an additional account I start to run into problems. if I do a 'sip show peers' with a good config I think it may outline the problem sip show peersName/username Host Dyn Nat ACL Mask Port Status 1005/1005 (Unspecified) D 255.255.255.255 0 Unmonitored1004/1004 (Unspecified) D 255.255.255.255 0 Unmonitored1003/1003 (Unspecified) D 255.255.255.255 0 Unmonitored1002/1002 10.0.0.52 D 255.255.255.255 5060 Unmonitored1001/1001 10.0.0.51 D 255.255.255.255 5060 Unmonitoredsipgate1/321 217.10.79.219 N 255.255.255.255 5060 OK (52 ms) I think it maybe a host specific ip address which must be in a table somewhere in asterisk. I have tried setting it up as a peer and dynamic but still no joy. Is there a limitation to this within asterisk. I have provided a sip.conf below (adjusted), will I need to implement a SER box (more things to learn which is all good provided it sorts my problem) [general]port = 5060bindaddr = 0.0.0.0disallow=allallow=ulawallow=alawallow=gsm;register = [EMAIL PROTECTED]/*** register = ***:[EMAIL PROTECTED]/**[sipgate1]type=friendcontext=from-sipgate1fromuser=** username=authuser=* secret=**host=sipgate.co.ukfromdomain=sipgate.co.uknat=yesdtmfmode=infoqualify=yesinsecure=verycanreinvite=no;[sipgate2]type=friendcontext=from-sipgate2fromuser=* username=** authuser=*** secret=* host=sipgate.co.ukfromdomain=sipgate.co.uknat=yesdtmfmode=infoqualify=yesinsecure=verycanreinvite=no;[1001]type=friendusername=1001secret=*host=dynamicdtmfmode=rfc2833context=from-sipphones;mailbox=1001allow=alawallow=ulaw kindest regards ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: multiple sip accounts from same sip registrar
- Original Message - From: Matt Scott To: asterisk-users@lists.digium.com Sent: Tuesday, May 17, 2005 5:59 PM Subject: multiple sip accounts from same sip registrar Dear all, I have an asterisk sip issue which I don't believe is unique. I use a registrar (sipgate.co.uk) where I have 3 different accounts. These accounts provide me with three seperate local phone numbers which allow me to allocate them to seperate users. By using just one of these accounts I can set asterisk up to send and receive calls no problem. However, when I start to introduce an additional account I start to run into problems. if I do a 'sip show peers' with a good config I think it may outline the problem sip show peersName/username Host Dyn Nat ACL Mask Port Status 1005/1005 (Unspecified) D 255.255.255.255 0 Unmonitored1004/1004 (Unspecified) D 255.255.255.255 0 Unmonitored1003/1003 (Unspecified) D 255.255.255.255 0 Unmonitored1002/1002 10.0.0.52 D 255.255.255.255 5060 Unmonitored1001/1001 10.0.0.51 D 255.255.255.255 5060 Unmonitoredsipgate1/321 217.10.79.219 N 255.255.255.255 5060 OK (52 ms) I think it maybe a host specific ip address which must be in a table somewhere in asterisk. I have tried setting it up as a peer and dynamic but still no joy. Is there a limitation to this within asterisk. I have provided a sip.conf below (adjusted), will I need to implement a SER box (more things to learn which is all good provided it sorts my problem) [general]port = 5060bindaddr = 0.0.0.0disallow=allallow=ulawallow=alawallow=gsm;register = [EMAIL PROTECTED]/*** register = ***:[EMAIL PROTECTED]/**[sipgate1]type=friendcontext=from-sipgate1fromuser=** username=authuser=* secret=**host=sipgate.co.ukfromdomain=sipgate.co.uknat=yesdtmfmode=infoqualify=yesinsecure=verycanreinvite=no;[sipgate2]type=friendcontext=from-sipgate2fromuser=* username=** authuser=*** secret=* host=sipgate.co.ukfromdomain=sipgate.co.uknat=yesdtmfmode=infoqualify=yesinsecure=verycanreinvite=no;[1001]type=friendusername=1001secret=*host=dynamicdtmfmode=rfc2833context=from-sipphones;mailbox=1001allow=alawallow=ulaw kindest regards ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
