[Asterisk-Users] need help
I am having an issue with the asterisk system not responding to dialed numbers during an active call. I'm not even sure where to look, zapata.conf? sip.conf? or the phone config? and worse I don't even know what Keywords to search for. Tim ___ 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] need help
Yes, I have a snom 190. I'm gonna check out the dtmf signalling now. thank you for the quick responces. Tim - Original Message - From: Ian Pattison [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Friday, April 29, 2005 9:44 AM Subject: Re: [Asterisk-Users] need help Is this a SIP phone? I had to upgrade the firmware on my SIP phones to alleviate this. It seems that the phone would actually disable it's own keypad after dialling. Ian [EMAIL PROTECTED] 29/04/2005 09:16 I am having an issue with the asterisk system not responding to dialed numbers during an active call. I'm not even sure where to look, zapata.conf? sip.conf? or the phone config? and worse I don't even know what Keywords to search for. Tim ___ 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 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] need help
Thank you, For the responces i had dtmfmode=inband when rcf2833 was the proper setting. I feel retarded that i missed that, but it happens. thanks again Tim Touhsaent - Original Message - From: [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Friday, April 29, 2005 9:38 AM Subject: Re: [Asterisk-Users] need help This is a DTMF issue, You must adjust this on the especific channel conf file. For example, ia sip phone cannot dial any number during an active call, you must see sip.conf and the config in your hardphone or softphone. Ismael. "Tim Touhsaent" [EMAIL PROTECTED] Enviado por: [EMAIL PROTECTED] 04/29/2005 03:16 PM Por favor, responda aAsterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Para "Asterisk Users Mailing List - Non-Commercial Discussion" asterisk-users@lists.digium.com cc Asunto [Asterisk-Users] need help I am having an issue with the asterisk system not responding to dialednumbers during an activecall. I'm not even sure where to look, zapata.conf? sip.conf? or the phoneconfig? and worse Idon't even know what Keywords to search for.Tim___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo 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] Active calls not responding to entries
I am having an issue with the phone system
recognizing keys in an active call. examples being when i call the extension of
VoiceMailMain() it does not recognize the numbers that i put in (eg. mailbox and
password) if i call from an internal line. However, if i call from an outside
line everything works properly. At this moment i have one snom 190 phone and 3
X-Lite softphones. Anouther example is If i call a bussiness with a voice menu I
am not able to navigate through the menu. I was wondering if this was an
asterisk issue or if it was in the phone's config. If it is an asterisk issue i
have attached my extensions and sip conf files. If not does anyone have an idea
of where to start, or had the problem before?
**
;extensions.conf by tim touhsaent
**
[general]
static=yeswriteprotect=yes
[globals]
; PHONE1 = snomphone190; PHONE2 =
softphone
ALLVOICEMAIL =
u109101102103104OUTGOING = Zap/g0ALL =
SIP/101SIP/102SIP/103SIP/104
; VMMAIN = main voice mail message; VMTIM
= vm/tim/message; VMPETER = vm/peter/message; VMRIGS =
Rigs's; VMDES = Des's
[default]
include = incominginclude =
internal
exten = i,1,Playback,wrongextexten =
i,2,Goto,s|1
exten = t,1,Playback,vm-goodbyeexten =
t,2,Hangup
[incoming]; wait one; exten =
s,1,Wait(0)
; Answer the phoneexten =
s,1,Answer;exten = s,2,MusicOnHold()exten =
s,2,Background(welcome,30,t); dial all phonesexten =
s,3,Dial(${ALL},30,t)exten = s,4,Goto,109|1; They must respond
within 10 seconds; else jump to priority t (terminate call)exten =
s,5,DigitTimeout,3exten = s,6,ResponseTimeout,10exten =
s,103,Dial($(ALL),30,t)exten = s,104,Goto,109|1
[outgoing]; outgoingexten =
_NXXNXX,1,Dial(${OUTGOING}/1${EXTEN});exten =
_NXXNXX.,2,Congestion()exten = _NXXNXX.,2,Goto,t|1
include = longdistance
[internal]; group email at 109exten =
109,1,VoiceMail(${ALLVOICEMAIL})exten = 109,2,Goto,t|1
; 1 - Tim's call planexten =
101,hint,SIP/101exten = 101,1,Dial(SIP/101,15,t);exten =
101,2,Playback,VMTIMexten = 101,2,VoiceMail(u101)exten =
101,3,Goto,t|1exten = 101,102,VoiceMail(u101)exten =
101,103,Goto,t|1
; 2 - Rigs's voicemailexten =
102,hint,SIP/102exten = 102,1,Dial(SIP/102,15,t);exten =
102,2,Playback,VMRIGSexten = 102,2,Voicemail(u102)exten =
102,3,Goto,t|1exten = 102,102,VoiceMail(u102)exten =
102,103,Goto,t|1
; 3 - Des's voicemailexten =
103,hint,SIP/103exten = 103,1,Dial(SIP/103,15,t);exten =
103,2,Playback,VMDESexten = 103,2,Voicemail(u103)exten =
103,3,Goto,t|1exten = 103,102,VoiceMail(u103)exten =
103,103,Goto,t|1
; 4 - Peter's voicemailexten =
104,hint,SIP/104exten = 104,1,Dial(SIP/104,15,t);exten =
104,2,Playback,VMDESexten = 104,2,Voicemail(u104)exten =
104,3,Goto,t|1exten = 104,102,VoiceMail(u104)exten =
104,103,Goto,t|1
; 8 - Voice mail administrationexten =
111,1,VoicemailMain([EMAIL PROTECTED])exten =
112,1,VoicemailMain([EMAIL PROTECTED])exten =
113,1,VoicemailMain([EMAIL PROTECTED])exten =
114,1,VoicemailMain([EMAIL PROTECTED]); exten
= 108,1,VoicemailMain([EMAIL PROTECTED])include
= outgoing
; t - terminate call
[longdistance]exten =
_1NXXNXX,1,Dial(${OUTGOING}/${EXTEN})exten =
911,1,Dial(${OUTGOING}/${EXTEN})exten =
411,1,Dial(${OUTGOING}/${EXTEN})
**
sip.conf
**
[general]port =
5060 ; port to bind
tobindaddr = 0.0.0.0
context = default
disallow=gsmallow=alawdisallow=ulaw
; register = 102::[EMAIL PROTECTED]:5060/102
[101]type=friend;secret=xxhost=dynamiccallerid=TIM101defaultip=10.0.100.132dtmfmode=inbandmailbox=101
[102]type=friendsecret=xxhost=dynamiccallerid=Rigs102defaultip=10.0.100.102dtmfmode=inbandmailbox=102
;auth=md5;secret=x;notransfer=1;host=dynamic;allow=gsm
[103]type=friendsecret=xhost=dynamiccallerid=Des103defaultip=10.0.100.108dtmfmode=inbandmailbox=103
[104]type=friend; secret=xxx
host=dynamiccallerid=Peter104defaultip=10.0.100.120dtmfmode=inbandmailbox=104
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Re: [Asterisk-Users] Zap Extensions unavailable after a call
make sure that you have a hangup command eg. . exten = s,2,Dial(SIP/101,30,t) exten = s,3,Hangup it would help if you put your extension.conf and zapata.conf file on the email so that someone can tell you more conclusivly what needs to be done - Original Message - From: Roberto Reiner Uhry [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Wednesday, April 20, 2005 1:29 PM Subject: [Asterisk-Users] Zap Extensions unavailable after a call Hi, I solved my last problem that was about receive calls. Now I have another one, that's after a end a phone the zap extension stay unavailable, until a restart on 1 minute. Does anybody know what could be it? Tkz, Reiner ___ 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
Re: [Asterisk-Users] Unable to create channel of type 'Zap'
To help you out i will post my config files... The only problem that i have
is in an active call i can't get my phones to send responses to voice menus
such as dialling the voicemailmain cmd.
I am using a TDM04b card with four ports instead of one my zapata.conf file
looks like:
[trunkgroups]
[channels]
;this block is standard features of the anolog lines
musiconhold=default
rxwink=300 ; seems to use long winks
usecallerid=yes
hidecallerid=no
callerid=Berkleigh Computer Systems
Callwaiting=yes
busydetect=no
callprogress=no
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
echotraining=800
rxgain=3.3 ; the gain options are volume adjustments (recievetransmit)
txgain=3.3
group=0
callgroup=1
pickupgroup=1
immediate=no
faxdetect=no
signalling=fxs_ks ;fxo cards use fxs signalling
callerid=asrecieved
context=default ; context in extensions.conf to use
channel=1-4 ; we have four availlable ports on our card
and my zaptel.conf has:
fxsks=1-4
defaultzone=us
loadzone=us
the only difference should be that you should have 1 instead of 1-4 in both
zapata and zaptel. as far as the extensions.conf i had issues with trying to
work with the demo file and just wrote my own, I found it easier, what you
prefer is your choice.
Tim Touhsaent
Berkleigh Computer Systems
Kutztown, PA
- Original Message -
From: Jaime Blanco [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Wednesday, April 20, 2005 2:33 PM
Subject: [Asterisk-Users] Unable to create channel of type 'Zap'
Hi,
I just installed the asterisk and the X100P card. I can receive calls
from
PSTN and it can ring on a Grandstream SIP Phone. From the SIP Phone I can
dial the demo extension on asterisk pbx. The issue is as soon as I try to
dial out 92714756 or another number I received the following message:
*CLI -- Executing Dial(SIP/1001-2b93, Zap/g2/2714756) in new
stack
Apr 20 02:27:40 NOTICE[245776]: app_dial.c:536 dial_exec: Unable to create
channel of type 'Zap'
== Everyone is busy at this time
-- Executing Congestion(SIP/1001-2b93, ) in new stack
== Spawn extension (from-sip, 92714756, 2) exited non-zero on
'SIP/1001-2b93'
Zapata.conf is:
[channels]
callwaiting=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
echocancel=yes
echocancelwhenbridged=no
rxgain=0.0
txgain=0.0
immediate=no
context=default
signalling=fxs_ks
channel=1
extensions.conf
;
; Static extension configuration file, used by ; the pbx_config module.
This
is where you configure all your ; inbound and outbound calls in Asterisk.
;
;
; The General category is for certain variables.
;
[general]
;
; If static is set to no, or omitted, then the pbx_config will rewrite ;
this file when extensions are modified. Remember that all comments ; made
in the file will be lost when that happens.
;
; XXX Not yet implemented XXX
;
static=yes
;
; if static=yes and writeprotect=no, you can save dialplan by ; CLI
command
'save dialplan' too ; writeprotect=no
; You can include other config files, use the #include command (without
the
';')
; Note that this is different from the include command that includes
contexts within ; other contexts. The #include command works in all
asterisk
configuration files.
;#include filename.conf
; The Globals category contains global variables that can be referenced
;
in the dialplan with ${VARIABLE} or ${ENV(VARIABLE)} for Environmental
variable ; ${${VARIABLE}} or ${text${VARIABLE}} or any hybrid ; [globals]
CONSOLE=Console/dsp ; Console interface for
demo
;CONSOLE=Zap/1
;CONSOLE=Phone/phone0
IAXINFO=guest ; IAXtel username/password
;IAXINFO=myuser:mypass
TRUNK=Zap/g2; Trunk interface
TRUNKMSD=1 ; MSD digits to strip
(usually 1 or 0)
;TRUNK=IAX2/user:[EMAIL PROTECTED]
;
; Any category other than General and Globals represent ; extension
contexts, which are collections of extensions.
;
; Extension names may be numbers, letters, or combinations ; thereof. If
an
extension name is prefixed by a '_'
; character, it is interpreted as a pattern rather than a ; literal. In
patterns, some characters have special meanings:
;
; X - any digit from 0-9
; Z - any digit from 1-9
; N - any digit from 2-9
; [1235-9] - any digit in the brackets (in this example,
1,2,3,5,6,7,8,9)
; . - wildcard, matches anything remaining (e.g. _9011. matches
; anything starting with 9011 excluding 9011 itself)
;
; For example the extension _NXX would match normal 7 digit dialings,
;
while _1NXXNXX would represent an area code plus phone number ;
preceeded by a one.
;
; Contexts contain several lines, one for each step of each ; extension,
which can take one of two forms as listed below
