Re: [asterisk-users] please help
Watch out the dialplan sequence
you have 1 then n, n , anf finally a 2. try changing the last 2 for n
if you are only dialing the number
0678922645
You have to remove the leading _ (underscore) and the ending point . ,
they are used only when dialing regular patterns. in your case wil
be 0678922645XX where is as many numbers as you want.
On Mon, May 30, 2011 at 11:54 PM, mahesh katta maheshka...@flexydial.comwrote:
Remove the _ in front of your dialplan,like
exten = 0678922645,1,--
On Mon, May 30, 2011 at 11:00 PM, salaheddine elharit
salah.elharit...@gmail.com wrote:
Hello list
i have configured astersik 1.4 with sip i have a question
when i put in dial plan.conf
exten = _0678922645.,1,Set(CALLERID(number)=520460587)
exten = _0678922645
.,n,MixMonitor(zap_g1_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))
exten = _0678922645
.,n,Dial(Zap/g1/${EXTEN},30,A(this-call-may-be-monitored-or-recorded))
exten = _067892264*5*,2,Hangup()
i can not call my number but when i delet the last number '5' i can call
without any issue
i want to put all the number please any hel to solve this issue
thanks and regards
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Best Regards,
Mahesh Katta
*BUZZ**WORKS* Business Services Private Limited
BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI
201, Crystal Tower, 75 Gundavali Cross Lane, Andheri Kurla Road, Andheri
(E) Mumbai 400069
GSM +91.97029.70779 | Phone +91.22.4229.2634 | Fax +91.22.4229.2634
Web http://www.buzzworks.com
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Camilo A. Echeverry J.
301 7553789
-
Y todo lo que hagáis, hacedlo de corazón, como para el Señor y no para los
hombres.
Colonences 3:23
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[asterisk-users] Asterisk + USSD
Hi. This might be off topic. Does somebody know some ASterisk+USSD Implementation ? Thanks. -- Camilo A. Echeverry J. 301 7553789 - Y todo lo que hagáis, hacedlo de corazón, como para el Señor y no para los hombres. Colonences 3:23 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [IAX] Everyone is busy/congested at this time (1:0/0/1)
add this line at the end of the IAX account definition and try again
requirecalltoken=no
On Wed, Apr 27, 2011 at 2:40 PM, John Alexis kasteris...@gmail.com wrote:
Unfortunatelly that doesn't change anything. I got exactly the same error
(Everyone is busy/congested at this time (1:0/0/1) ... ).
I did a dialplan reload before testing of course.
2011/4/25 Camilo Echeverry camiloecheve...@gmail.com
As I see in your iax.conf, IAX Peer belogs to special context, which
means 444 is allowed to make calls to extensions only on the same context
(Extension 111), can you call extension 111 ?
may be the other extensions are in the default context and you can
receive calls because extension 444 (dial IAX2/444) exists in that conext.
Try adding this in the [special] conext
;this will dial any 3 digit Extension to IAX
exten = _XXX,1,Dial(IAX2/${EXTEN})
exten = _XXX,n,hangup()
that may solve your problem
On Sat, Apr 23, 2011 at 3:38 AM, John Alexis kasteris...@gmail.comwrote:
Hi,
Sorry to insist, but I still not have any solution. Does anybody have an
idea ?
Thanks!
2011/4/20 John Alexis kasteris...@gmail.com
Hi,
I have a problem with IAX accounts...
I set up a few months ago an Asterisk server, with mysql support to load
iax accounts.
Settings seems fine because apparently the system works as expected.
Yesterday I tried to add another iax account in the iax.conf directly.
And I have a problem with this new account (named 444).
I can authenticate from 444 to the server, and I can receive calls from
other softphones (which parameters are loaded from the mysql database
iaxfriends).
BUT, i cannot call other softphones. I always got a message in the log
saying Everyone is busy/congested at this time (1:0/0/1).
So, i don't know where is the probleme : is it from iax accounts loaded
from the database, or the new account 444 ???
Below are the conf files and verbose output.
Thank you very much for your help :)
-
- iax.conf
-
[general]
bindport=4569
delayreject=yes
language=fr
autokill = yes
calltokenoptional = 0.0.0.0/0.0.0.0
minregexpire = 60
maxregexpire = 500
mohsuggest=default
careinvite=no
rtcachefriends=yes
[444]
type=friend
host=dynamic
context=special
secret=iop
-
- extconfig.conf:
-
[general]
[settings]
iaxusers = mysql,asterisk,iaxfriends
iaxpeers = mysql,asterisk,iaxfriends
voicemail = mysql,asterisk,voicemail
-
- Mysqldump from iaxfriends
-
INSERT INTO iaxfriends
(name,type,phonenumber,username,mailbox,secret,dbsecret,context,regcontext,host,ipaddr,port,defaultip,sourceaddress,mask,regexten,regseconds,accountcode,mohinterpret,mohsuggest,inkeys,outkey,language,callerid,cid_number,sendani,fullname,trunk,auth,maxauthreq,requirecalltoken,encryption,transfer,jitterbuffer,forcejitterbuffer,disallow,allow,codecpriority,qualify,qualifysmoothing,qualifyfreqok,qualifyfreqnotok,timezone,adsi,amaflags,setvar)
VALUES
('admin.my.domain','friend','100','admin@my.domain','123','','default','','dynamic','10.0.100.56','26564','','','','','0','','','','','','en','admin.my.domain','','','','','md5','','','','','','','all','gsm,ulaw,alaw','','','','','','','','','')
;
INSERT INTO iaxfriends
(name,type,phonenumber,username,mailbox,secret,dbsecret,context,regcontext,host,ipaddr,port,defaultip,sourceaddress,mask,regexten,regseconds,accountcode,mohinterpret,mohsuggest,inkeys,outkey,language,callerid,cid_number,sendani,fullname,trunk,auth,maxauthreq,requirecalltoken,encryption,transfer,jitterbuffer,forcejitterbuffer,disallow,allow,codecpriority,qualify,qualifysmoothing,qualifyfreqok,qualifyfreqnotok,timezone,adsi,amaflags,setvar)
VALUES ('alice.my.domain','friend','111','admin@my.domain
','','alice@my.domain','456','','default','','dynamic','10.0.100.221','42478','','','','','1303301760','','','','','','en','alice.my.domain','','','','','md5','','','','','','','all','gsm,ulaw,alaw','','','','','','','','','')
;
-
- extensions.conf:
-
[general]
[externe]
exten = 555,1,Dial(IAX2/111)
exten = 555,n,Hangup()
[special]
exten = 111,1,Dial(IAX2/111)
exten = 111,n,Hangup()
[default]
exten = 444,1,Dial(IAX2/444)
exten = 444,n,Hangup()
- Sip.conf (SIP server):
[general]
context=default
allowoverlap=no
udpbindaddr=0.0.0.0
tcpenable=no
tcpbindaddr=0.0.0.0
srvlookup=yes
-
- Logs server:
-
-- Accepting AUTHENTICATED call from 10.0.100.238:
requested format = gsm,
requested prefs = (),
actual format = ulaw,
host prefs = (),
priority = mine
-- Executing [111@special:1] Dial(IAX2/444-436, IAX2/111) in
new stack
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [111@special:2] Hangup(IAX2/444-436, ) in new
stack
== Spawn extension (special, 111, 2) exited non-zero on 'IAX2/444-436'
-- Hungup 'IAX2/444-436'
-- Accepting AUTHENTICATED call from
Re: [asterisk-users] [IAX] Everyone is busy/congested at this time (1:0/0/1)
As I see in your iax.conf, IAX Peer belogs to special context, which means
444 is allowed to make calls to extensions only on the same context
(Extension 111), can you call extension 111 ?
may be the other extensions are in the default context and you can
receive calls because extension 444 (dial IAX2/444) exists in that conext.
Try adding this in the [special] conext
;this will dial any 3 digit Extension to IAX
exten = _XXX,1,Dial(IAX2/${EXTEN})
exten = _XXX,n,hangup()
that may solve your problem
On Sat, Apr 23, 2011 at 3:38 AM, John Alexis kasteris...@gmail.com wrote:
Hi,
Sorry to insist, but I still not have any solution. Does anybody have an
idea ?
Thanks!
2011/4/20 John Alexis kasteris...@gmail.com
Hi,
I have a problem with IAX accounts...
I set up a few months ago an Asterisk server, with mysql support to load
iax accounts.
Settings seems fine because apparently the system works as expected.
Yesterday I tried to add another iax account in the iax.conf directly. And
I have a problem with this new account (named 444).
I can authenticate from 444 to the server, and I can receive calls from
other softphones (which parameters are loaded from the mysql database
iaxfriends).
BUT, i cannot call other softphones. I always got a message in the log
saying Everyone is busy/congested at this time (1:0/0/1).
So, i don't know where is the probleme : is it from iax accounts loaded
from the database, or the new account 444 ???
Below are the conf files and verbose output.
Thank you very much for your help :)
-
- iax.conf
-
[general]
bindport=4569
delayreject=yes
language=fr
autokill = yes
calltokenoptional = 0.0.0.0/0.0.0.0
minregexpire = 60
maxregexpire = 500
mohsuggest=default
careinvite=no
rtcachefriends=yes
[444]
type=friend
host=dynamic
context=special
secret=iop
-
- extconfig.conf:
-
[general]
[settings]
iaxusers = mysql,asterisk,iaxfriends
iaxpeers = mysql,asterisk,iaxfriends
voicemail = mysql,asterisk,voicemail
-
- Mysqldump from iaxfriends
-
INSERT INTO iaxfriends
(name,type,phonenumber,username,mailbox,secret,dbsecret,context,regcontext,host,ipaddr,port,defaultip,sourceaddress,mask,regexten,regseconds,accountcode,mohinterpret,mohsuggest,inkeys,outkey,language,callerid,cid_number,sendani,fullname,trunk,auth,maxauthreq,requirecalltoken,encryption,transfer,jitterbuffer,forcejitterbuffer,disallow,allow,codecpriority,qualify,qualifysmoothing,qualifyfreqok,qualifyfreqnotok,timezone,adsi,amaflags,setvar)
VALUES
('admin.my.domain','friend','100','admin@my.domain','123','','default','','dynamic','10.0.100.56','26564','','','','','0','','','','','','en','admin.my.domain','','','','','md5','','','','','','','all','gsm,ulaw,alaw','','','','','','','','','')
;
INSERT INTO iaxfriends
(name,type,phonenumber,username,mailbox,secret,dbsecret,context,regcontext,host,ipaddr,port,defaultip,sourceaddress,mask,regexten,regseconds,accountcode,mohinterpret,mohsuggest,inkeys,outkey,language,callerid,cid_number,sendani,fullname,trunk,auth,maxauthreq,requirecalltoken,encryption,transfer,jitterbuffer,forcejitterbuffer,disallow,allow,codecpriority,qualify,qualifysmoothing,qualifyfreqok,qualifyfreqnotok,timezone,adsi,amaflags,setvar)
VALUES ('alice.my.domain','friend','111','admin@my.domain
','','alice@my.domain','456','','default','','dynamic','10.0.100.221','42478','','','','','1303301760','','','','','','en','alice.my.domain','','','','','md5','','','','','','','all','gsm,ulaw,alaw','','','','','','','','','')
;
-
- extensions.conf:
-
[general]
[externe]
exten = 555,1,Dial(IAX2/111)
exten = 555,n,Hangup()
[special]
exten = 111,1,Dial(IAX2/111)
exten = 111,n,Hangup()
[default]
exten = 444,1,Dial(IAX2/444)
exten = 444,n,Hangup()
- Sip.conf (SIP server):
[general]
context=default
allowoverlap=no
udpbindaddr=0.0.0.0
tcpenable=no
tcpbindaddr=0.0.0.0
srvlookup=yes
-
- Logs server:
-
-- Accepting AUTHENTICATED call from 10.0.100.238:
requested format = gsm,
requested prefs = (),
actual format = ulaw,
host prefs = (),
priority = mine
-- Executing [111@special:1] Dial(IAX2/444-436, IAX2/111) in new
stack
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [111@special:2] Hangup(IAX2/444-436, ) in new stack
== Spawn extension (special, 111, 2) exited non-zero on 'IAX2/444-436'
-- Hungup 'IAX2/444-436'
-- Accepting AUTHENTICATED call from 10.0.100.50:
requested format = ulaw,
requested prefs = (),
actual format = gsm,
host prefs = (gsm|ulaw|alaw),
priority = mine
-- Executing [444@default:1] Dial(IAX2/alice.my.domain-8277,
IAX2/444) in new stack
-- Called 444
-- Call accepted by 10.0.100.238 (format gsm)
-- Format for call is gsm
-- IAX2/444-4734 is ringing
-- IAX2/444-4734
[asterisk-users] sterisk+SS7 Error: chan_dahdi.c: Unable to start PBX on DAHDI/288-1
Hi. Dont know if this is an Asterisk or Dahdi or LibSS7 Error. So Im writing to Asterisk List. If somebody knows where to search (dahdi lists or libSS7 lists) will be appreciated. Im getting this error after a certain time, My config is: Hardware: 3 Digium Quad E1 TE4XXP libss7 version: SVN-branch-1.0-r286 DAHDI Version: 2.4.0 Echo Canceller: Asterisk 1.6.2.14 CentOS release 5.5 (Final) Kernel 2.6.18-194.el5PAE The error is: * * */var/log/asterisk/messages:[Mar 12 07:47:17] WARNING[26421] chan_dahdi.c: Unable to start PBX on DAHDI/26-1* */var/log/asterisk/messages:[Mar 12 07:47:17] WARNING[26421] chan_dahdi.c: Unable to start PBX on CIC 26* Is a very simple PBX which receives the calls in SS7 and redirects them (via IAX2 trunk) to another Asterisk which is connected to an avaya PBX using the same hardware but with PRI singaling. -- Camilo A. Echeverry J. 301 7553789 - Y todo lo que hagáis, hacedlo de corazón, como para el Señor y no para los hombres. Colonences 3:23 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AGI GET DATA and WAIT FOR DIGIT don't work
Hi.
I'm trying to get digits form the user via agi
something like this: this only should print result=asciicode
but none of the functions even wait until timeout ..
they just pass .. (after a nanosecond)
the las print is always timeout.
Any clue ..?
my $callerid = $AGI{'callerid'} ;
if($callerid !~ /[0-9]{7,20}/){
#way numbre one
print EXEC PLAYBACK please_enter_your_number \\\n; my $result =
STDIN;
print WAIT FOR DIGIT 3000\n; my $result = STDIN;
# Way number two
# print GET DATA please_enter_your_number \-1\ \10\; my $result =
STDIN;
}
print STDERR $result;
--
Camilo Echeverry
Your life would be very empty if you had nothing to regret.
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Re: [asterisk-users] WAIT FOR DIGIT not working
Hi .
I had the same problem but downloaded a test script and wait for digit
worked.
the only visible difference is that I wat nos using strict,
so I am rewriting the AGI with
use strict;
Hope this help.
On 9/14/06, Joel Lansden [EMAIL PROTECTED] wrote:
Hello all,
I have been trying to solve this problem for days, with no luck.
When I run an AGI script from my extensions.conf, it seems no matter what
I do, the WAIT FOR DIGIT command will not work. The system just flies
past it without waiting a single millisecond, and of course my script
crashes because it doesn't have the input it needs. I have run 3 different
versions of Asterisk in the hopes of clearing this up, and presently am on
1.2.12.1.
My script is simple:
#!/usr/bin/perl
use POSIX;
$| = 1;
sub trim {
my @out = @_;
for (@out)
{
s/^\s+//;
s/\s+$//;
}
return wantarray ? @out : $out[0];
}
while(STDIN) {
chomp;
last unless length($_);
if (/^agi_(\w+)\:\s+(.*)$/) {
$AGI{$1} = $2;
}
}
print EXEC Ringing\n;
print EXEC Wait 1\n;
print EXEC Answer\n;
print EXEC Festival 'Please enter the extension you want to call'\n;
$target = ;
print WAIT FOR DIGIT 5000\n;
$target .= STDIN;
print WAIT FOR DIGIT 5000\n;
$target .= STDIN;
print WAIT FOR DIGIT 5000\n;
$target .= STDIN;
print STDERR Result was $target\n;
That's all there is to it, but it won't work.
Can anyone help?
Thanks!!!
~Joel
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Camilo Echeverry
Your life would be very empty if you had nothing to regret.
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[asterisk-users] Playback() does not work
Hi.I've installed Asterisk with a MD3200 modem,zaptel modules recognize the card,when i dial to asterisk, it answers but when I Playback(something) do not receive any audio, only a sound like audio static but I created in extensions.conf[demo]iclude= defaultand when in the console type the commandCLI dial sthe [default] context (included by [demo]) plays perfectly on the soundcard Notice that I only modified these files:zapte.confzapata.confextensions.confAny Idea ..?Am I missing something ..?--ThanksCamilo. ___ --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] Simple But important question (for me)
Hi.I'm 100% newbie (in asterisk)I need to know if i can use astersik for something like this:1- receive the call (obvious)2- get the Caller ID3- Send the CID to another application and get some info from a Database example: Your address is some address 4- Get that info and convert it into voice (by mixing various audio files)5- return it to the Caller (as audio)6- use keypress as menu options menu or confirmation responses (i know asterisk can do this) sorry is that sounds pretty obvious to you, but as I said I'm new on this.after this (if the answer is yes) i will read as much documentation as possible to do the rest by myself.-- -- Papita = papa pequeñaPapota = papa grandePaputa = Papa Gigante ..?-- ___ --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
