[asterisk-users] Linksys 942 Call Transfer
Hi I have a customer who is using Linksys 942 phones. When they try to transfer a call the Asterisk CLI reports that both legs of the call must exist on the server. The call they are trying to transfer then drops. Does anyone know why this is and how to fix it? TIA Regards Jon ___ Want ideas for reducing your carbon footprint? Visit Yahoo! For Good http://uk.promotions.yahoo.com/forgood/environment.html ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Avaya IP Office DTMF Issue
Hi I have a client using a Avaya IP Office PBX that is taking a SIP trunk from me terminating on a * box. It all works perfectly apart from DTMF. Although you can hear the tones they don't seem to get recognised. I have tried DTMF mode auto, inband, out of band and rfc2833 but no luck. Any ideas? Regards Jon -- Jon Farmer Telford, Shropshire, UK ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IVR and MySQL
I write such apps all the time, i have written IVR apps that talk to SQL Server (Sage500), MySQL, our credit/debit card processing system. Regards Jon --- Thiago Maluf <[EMAIL PROTECTED]> wrote: > Hi Fabio, > of course that you can. > > One way to do it is working with app MYSQL(), where > you will put your sql as > argumment. > read more in > http://www.voip-info.org/wiki/view/Asterisk+cmd+MYSQL > > good luck, > Thiago Maluf Resende. > > 2007/8/14, Fabio Ardeola <[EMAIL PROTECTED]>: > > > > Hi > > > > Does somebody know if I can save the answers made > by > > the caller person on the IVR menu in a MySQL > Database? > > If yes, can I save the CallerID as well? > > > > Thanks, > > Fabio > > > > > > > > > > > Luggage? GPS? Comic books? > > Check out fitting gifts for grads at Yahoo! Search > > > http://search.yahoo.com/search?fr=oni_on_mail&p=graduation+gifts&cs=bz > > > > ___ > > --Bandwidth and Colocation Provided by > http://www.api-digital.com-- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > -- > > THIAGO MALUF RESENDE > Consultor Voip e Programador WEB (Voip Developer and > Web Developer) > Tel: +55 21 86042100 > e-mail: [EMAIL PROTECTED] > > ___ > --Bandwidth and Colocation Provided by > http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users ___ Yahoo! Answers - Got a question? Someone out there knows the answer. Try it now. http://uk.answers.yahoo.com/ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MySQL cmd % pattern matching
Try enclosing in single quotes. ie.
SELECT name from contacts where tel like '%${number}'
Jon Farmer
Telford, Shropshire, UK
- Original Message
From: Garth van Sittert <[EMAIL PROTECTED]>
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Monday, 4 December, 2006 12:38:07 PM
Subject: [asterisk-users] MySQL cmd % pattern matching
Hi All
Does anyone know how to use the MySQL cmd in Asterisk with LIKE and % in
the query?
I have:
exten => s,5,Set(query=SELECT name from contacts where tel like
%${number})
exten => s,6,MySQL(Connect connid hostname username password dbname)
exten => s,7,MySQL(Query resultid ${connid} ${query})
But there seems to be a problem with the % sign and I don't know how to
hash it out.
It works without the % sign.
Thanks
Kind Regards
Garth
___
--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
___
All new Yahoo! Mail "The new Interface is stunning in its simplicity and ease
of use." - PC Magazine
http://uk.docs.yahoo.com/nowyoucan.html
___
--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] MWI across multiple servers
I decided to write my own simple voicemail application via AGI and store all voicemails in MySQL. The nice thing was the user can retrieve via phone (local and remote), via email attachment and also via web download. You can listen to old and new messages and change your outgoing message too. Regards Jon Jon Farmer Telford, Shropshire, UK - Original Message From: "Porier, Jeremy M." <[EMAIL PROTECTED]> To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Wednesday, 6 December, 2006 4:20:04 PM Subject: [asterisk-users] MWI across multiple servers We are about to deploy six Asterisk servers across the state with SIP phones at each site registering to their "local" server. However, we are centralizing voicemail at our main campus to enable the transfer of voicemails between users regardless of site. It also simplifies our backup procedures for voicemail. Any tips for distributing MWI messages amongst those separate servers that phones are registering to? I suppose I could script something on the voicemail server to put a file in the inbox on the distributed servers but perhaps there is something more elegant I'm unaware of? If not, has anyone scripted this before and willing to share? Would be much appreciated. Thanks, Jeremy ___ --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 Send instant messages to your online friends http://uk.messenger.yahoo.com ___ --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] How to connect two asterisk server
Hi I would suggest a IAX2 trunk between the two servers. You will need to modify the dialplan to recognise which extensions are on each box and route accordingly. The fact your clients are SIP does not preclude you from using IAX2 to connect the servers. Regards Jon Jon Farmer Telford, Shropshire, UK - Original Message From: Thirumal Saminathan <[EMAIL PROTECTED]> To: Asterisk Users Mailing List - Non-Commercial Discussion ; [EMAIL PROTECTED] Sent: Thursday, 28 December, 2006 6:09:25 AM Subject: [asterisk-users] How to connect two asterisk server Hi all, I need to connect two asterisk server in same network and i'm using sip user as my clients.. plz anyone suggest me Regards, Thiru ___ --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 ___ All New Yahoo! Mail – Tired of [EMAIL PROTECTED]@! come-ons? Let our SpamGuard protect you. http://uk.docs.yahoo.com/nowyoucan.html___ --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] Cisco 7970 Unprovisioned
Are you setting the TFTP server address in the DHCP? Are you checking the TFTP log to see what files the phone is requesting and not finding? Regards Jon Jon Farmer Telford, Shropshire, UK - Original Message From: Token PBX <[EMAIL PROTECTED]> To: asterisk-users@lists.digium.com Sent: Saturday, 20 January, 2007 1:01:25 PM Subject: [asterisk-users] Cisco 7970 Unprovisioned Hi! I did manage to load phone with SIP image : SIP70.8-0-3S, made SEP-MAC.cnf.xml, but phone never read the configuration from it. On the screen it's written "Unprovisioned", and phone is not trying to register with asterisk. Please help!! MihaelaMJ ___ --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 ___ New Yahoo! Mail is the ultimate force in competitive emailing. Find out more at the Yahoo! Mail Championships. Plus: play games and win prizes. http://uk.rd.yahoo.com/evt=44106/*http://mail.yahoo.net/uk ___ --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] Dialplan programming vs. AGI vs. ???
This depends on your application. As you say you are able to do everything you require in dialplan at that is great. I have used AGI fairly extensively becuase the stuff I want to do can't be done in dialplan alone. For instance i have written a auto attendants that can be dynamically controlled by a non-techie user with real time and in call reconfiguration. Also i have written IVR apps that hook into our CRM and Accounting systems for fault reporting and credit card payments etc. What you need is the tool for the job like everything in life :-) Jon Farmer Telford, Shropshire, UK - Original Message From: Gordon Henderson <[EMAIL PROTECTED]> To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Wednesday, 31 January, 2007 12:20:11 AM Subject: [asterisk-users] Dialplan programming vs. AGI vs. ??? Just a general question on dialplan programming... I've implemented a fairly full-featured system using dialplan code only. I've not used any AGI for it, yet it ticks all the boxes I want it to tick (diverts, follow-me, voicemail, dnd, outdialing restrictions, simple auto-attendant, and numerous "star codes" to control it all) This is all aimed at the small/medium office PBX type application. But I'm curious as to the approach others use. Is doing dialplan coding in an AGI more efficient, or do people just do it that way because it's easier than learning dialplan code? Or are there some things that people think they can't do any other way? So I'm just after some ideas, really, possibly to work out if it's worth my while going down the AGI route for future projects, or not!?! Any feedback is most welcome! Cheers, Gordon ___ --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 ___ New Yahoo! Mail is the ultimate force in competitive emailing. Find out more at the Yahoo! Mail Championships. Plus: play games and win prizes. http://uk.rd.yahoo.com/evt=44106/*http://mail.yahoo.net/uk ___ --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] PHP AGI script callerid question
Have you tried
phpagi
http://phpagi.sourceforge.net/
Jon Farmer
Telford, Shropshire, UK
- Original Message
From: Michelle Dupuis <[EMAIL PROTECTED]>
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Friday, 26 January, 2007 5:52:27 PM
Subject: [asterisk-users] PHP AGI script callerid question
I am trying to set
callerid from a PHP script, using one of two functions as shown below (setid1
and setid2). The first function works great with regular names and
numbers, BUT, if I call the function with ("Test","UnknownNumber"), the cid
number gets set to "asterisk". Why is my passed number parameter not being
accepted in this case?
The second function
uses the new/recommended method of setting cid name and number, but it has NO
EFFECT. (i.e. the name and number remain at the asterisk default).
Why is this one not working?
Thanks,
MD
==
// Test #1
function setid1($name,$number) {
$newid
= "\"" . trim( substr( trim( $name ), 0, 15 )
)
. "\"<" .
trim( substr( str_replace( " ", "", $number ), 0, 24 )
)
.">";
obj->set_callerid( $newid );
}
// Test
#2
function
setid1($name,$number) {
$obj->cmd("Set(CALLERID(name)=\"$name\"");
$obj->cmd("Set(CALLERID(num)=\"$number\"");
}
___
--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
___
New Yahoo! Mail is the ultimate force in competitive emailing. Find out more at
the Yahoo! Mail Championships. Plus: play games and win prizes.
http://uk.rd.yahoo.com/evt=44106/*http://mail.yahoo.net/uk ___
--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] problem with asterisk AGI
Set a variable that you can then use GotoIf in the dialplan to branch to the required exten Jon Farmer Telford, Shropshire, UK - Original Message From: prasanth <[EMAIL PROTECTED]> To: asterisk-users@lists.digium.com Sent: Thursday, 8 February, 2007 10:06:07 AM Subject: [asterisk-users] problem with asterisk AGI I have a fairly complicated setup. Extensions (1,2 and 3). In 3 - I execute AGI in java which plays few wav files depending on external parameters. Can I have a dial plan inside my AGI? If not, how do I accomodate user who needs to reach extension 2 from my agi? I have tried stream file and get data but the two commands did not work at all. ___ --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 ___ What kind of emailer are you? Find out today - get a free analysis of your email personality. Take the quiz at the Yahoo! Mail Championship. http://uk.rd.yahoo.com/evt=44106/*http://mail.yahoo.net/uk ___ --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] AGI DTMF Problem
Hi I am writing a IVR app using phpagi and are coming up against a problem when trying to detect DTMF. If I use the get_data function I dont seem to be able to reliably detect 16 digits. If I try 10 digits then its fine but anything above that seems to have a problem. Any ideas anyone? Jon Farmer Telford, Shropshire, UK ___ All New Yahoo! Mail Tired of unwanted email come-ons? Let our SpamGuard protect you. http://uk.docs.yahoo.com/nowyoucan.html ___ --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] Remote-Party-ID and selective CLI withold
Hi One of my SIP providers need me to send the Remote-Party-ID with privacy=on to withhold CLI and privacy=off to show CLI. I want the option to withhold CLI selectable by my users. I have set sendprid=yes in the sip.conf but I cant find a way to toggle the privacy between on and off on a per call basis. Any ideas how to do this? Regards Jon __ Sent from Yahoo! Mail. A Smarter Email http://uk.docs.yahoo.com/nowyoucan.html ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Remote-Party-ID and selective CLI withold
Hi
Thanks for that, got it working selectable by user now.
I did know about the SetCallPres() but it had temporarily slipped my mind :-)
Regards
Jon
- Original Message
From: Phil Reynolds <[EMAIL PROTECTED]>
To: asterisk-users@lists.digium.com
Sent: Monday, 9 June, 2008 4:09:34 PM
Subject: Re: [asterisk-users] Remote-Party-ID and selective CLI withold
Quoting Jon Farmer <[EMAIL PROTECTED]>:
> Hi
>
> One of my SIP providers need me to send the Remote-Party-ID with
> privacy=on to withhold CLI and privacy=off to show CLI. I want the
> option to withhold CLI selectable by my users. I have set
> sendprid=yes in the sip.conf but I cant find a way to toggle the
> privacy between on and off on a per call basis. Any ideas how to do
> this?
exten => _141.,1,SetCallerPres(prohib)
exten => _141.,n,Goto(${EXTEN:3},1)
... provided of course that this appears in the same context... that
should do it.
--
Phil Reynolds
o mail: [EMAIL PROTECTED]
|L_ \ / Web: http://www.tinsleyviaduct.com/phil/
(_)- \/ Waltham 66, Emley Moor 69, Droitwich 79, Windows 95
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
__
Sent from Yahoo! Mail.
A Smarter Email http://uk.docs.yahoo.com/nowyoucan.html
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] injecting audio announcements into sip channel
--- Dinesh Nair <[EMAIL PROTECTED]> wrote: > take a look at the L() option to Dial(). The original poster said he need to play different messages at different call durations. In order to do that you would need to dynamically alter LIMIT_WARNING_FILE as the call progressed. Regards Jon Jon Farmer Telford, Shropshire, UK ___ Yahoo! Mail is the world's favourite email. Don't settle for less, sign up for your free account today http://uk.rd.yahoo.com/evt=44106/*http://uk.docs.yahoo.com/mail/winter07.html ___ --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] PHP/AGI Problem
Hi I am currently developing a IVR application using PHP/AGI. I am using the PHPAGI class at http://phpagi.sourceforge.net/ to handle the commuication with my *. The application basically asks a caller to enter in some information which is then processed and a answer is read back out to them. I want the application to loop back to the beginning after giving the answer so they can try another query. When they have finished they enter 00# and the application ends and * hangs up the call. All of this works fine. The problem is as follows. If the caller hangs up at any time during the application the following happens. 1. Asterisk console reports the call hung up. As follows == Spawn extension (default, 502, 3) exited non-zero on 'SIP/3753684-fabd' 2. However when I look in the server process list the PHP app is still running. ps -ax ... 8029 ?S 0:00 /usr/bin/php -q /usr/share/asterisk/agi-bin/test.php They only way to get rid of it is to killall -9 it. Any ideas how I can get asterisk to kill the script if the caller hangs up? Regards Jon Farmer ___ How much free photo storage do you get? Store your holiday snaps for FREE with Yahoo! Photos http://uk.photos.yahoo.com ___ 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] PHP/AGI Problem
--- Thierry Wehr <[EMAIL PROTECTED]> wrote: > Hello > > Did you tried a deadagi in place of agi > > A++ > > I am calling the PHP app via deadagi. I believe what might be happening is that the application is going into a internal loop waiting for DTMF to know what to do next. I am going to investigate if I very time I poll for DTMF I check the channel status and exit() the script if the line is anything but UP. Regards Jon Jon Farmer Skype: viperdude_uk ___ How much free photo storage do you get? Store your holiday snaps for FREE with Yahoo! Photos http://uk.photos.yahoo.com ___ 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] PHP/AGI Problem
--- Alex Barnes <[EMAIL PROTECTED]> wrote: > Although this isnt a substitute for a correctly > terminating script, > I would have thought that the PHP 'maximum script > execution time' > variable would kick-in > and kill the script eventually. Well I have already tried that I have the first line of the script saying set_time_limit(30); but it doesn't terminate the script. I have also tested detecting the channel_status and that doesn't seem to work either. This leads me to believe that the script is not running anymore but just held in limbo doing nothing except taking up resource. I would imagine that the scenario where the caller hangs up before the end of a AGI script is common so am I missing something or is the behaviour of my script normal? Regards Jon ___ How much free photo storage do you get? Store your holiday snaps for FREE with Yahoo! Photos http://uk.photos.yahoo.com ___ 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] MoH: mpg123 problems
--- yusuf <[EMAIL PROTECTED]> wrote: > Hey all, > I have read on voip-info.org that to configure MoH > asterisk requires the > use of mpg123. I have installed mpg123 and > restarted asterisk. But, > when i put a call on hold i get this error: > > May 25 14:13:03 WARNING[1872]: res_musiconhold.c:865 > > local_ast_moh_start: No class: default You need to configure your musiconhold.conf file see http://www.voip-info.org/wiki-Asterisk+config+musiconhold.conf for more details Regards Jon ___ How much free photo storage do you get? Store your holiday snaps for FREE with Yahoo! Photos http://uk.photos.yahoo.com ___ 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] PHP/AGI Problem
> Well, that is your problem. Don't use deadagi. > DeadAGI is for use if you > want to continue processing "after" the call hangs > up. That is why your > scripts are continuing to run. Use regular AGI. I get the same behaviour if I use deadagi or just agi Regards Jon ___ Yahoo! Messenger - NEW crystal clear PC to PC calling worldwide with voicemail http://uk.messenger.yahoo.com ___ 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] PHP/AGI Problem
--- Jay Milk <[EMAIL PROTECTED]> wrote: > Why are you using DeadAGI? Use AGI or EAGI instead, > unless you actually > want to run on a dead-channel. I used DeadAGI just to see if it had any different behaviour in relation to my proplem. I get the same results with DeadAGI EAGI and AGI. Regards Jon Jon Farmer Skype: viperdude_uk ___ Yahoo! Messenger - NEW crystal clear PC to PC calling worldwide with voicemail http://uk.messenger.yahoo.com ___ 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] PHP/AGI Problem
Matthew Boehm wrote: Well then you got something screwed up somewhere. I've got many PHP AGI's and as soon as the caller hangs up the script terminates. This is what I use: exten => _80059974XX,1,AGI(line_counter.php) Works like a champ. And yes, I'm using phpagi.php as well. Matthew, Can I ask, does your script loop ? My script keeps running until the user send the DTMF to hangup. Otherwise if the caller hangs up I want the script to terminate. The problem is it doesn't! Regards Jon ___ 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] PHP/AGI Problem
Moises Silva wrote:
could you post the script, the output of the script in the asterisk
console and which asterisk version are you working with?
See below
This is just a proof of concept script so its a bit basic...
#!/usr/bin/php -q
agi_exec("ANSWER");
while($y != 1){
$response = "";
$agi->agi_exec("STREAM FILE
/usr/share/asterisk/sounds/adsl/enter_ref #");
$res = getDTMF("#",7);
$agi->agi_exec("STREAM FILE
/usr/share/asterisk/sounds/adsl/you_entered #");
$response .= trim($res);
sayNumber($response);
switch($response) {
case "12345":
$agi->agi_exec("STREAM FILE
/usr/share/asterisk/sounds/adsl/adsl_username_is #");
$username = "JONFARMER";
$userarray = myStringSplit(strtolower($username),1);
$usercount = count($userarray);
for($i=0;$i<$usercount;$i++){
$agi->agi_exec("STREAM FILE
/usr/share/asterisk/sounds/letters/" . $userarray[$i] . " #");
}
$agi->agi_exec("STREAM FILE
/usr/share/asterisk/sounds/adsl/domain1t #");
break;
case "67891":
$agi->agi_exec("STREAM FILE
/usr/share/asterisk/sounds/adsl/adsl_username_is #");
$username = "test_user";
$userarray = myStringSplit(strtolower($username),1);
$usercount = count($userarray);
for($i=0;$i<$usercount;$i++){
$agi->agi_exec("STREAM FILE
/usr/share/asterisk/sounds/letters/" . $userarray[$i] . " #");
}
$agi->agi_exec("STREAM FILE
/usr/share/asterisk/sounds/adsl/domain2 #");
break;
case "00":
$agi->agi_exec("STREAM FILE thanks #");
exit();
break;
default:
$agi->agi_exec("STREAM FILE
/usr/share/asterisk/sounds/adsl/account_not_found #");
break;
}
}
$agi->agi_exec("STREAM FILE thanks #");
unlink("/tmp/*.wav");
function sayNumber($digit) {
global $agi;
$res = myStringSplit($digit,1);
$num = count($res);
for($x=0;$x<$num;$x++) {
$agi->agi_exec("STREAM FILE /usr/share/asterisk/sounds/digits/"
. $res[$x] ." #");
}
sleep(1);
}
function myStringSplit($string, $long){
$long--;
// Converting the string into an array
For( $i=0; isset($string{$i}); $i++){
$myarray[] = $string{$i};
}
// Making arrays from the array
$a=0;
$end = $long;
$iend = count($myarray) / ( $long + 1 );
For( $i=0; $i<$iend; $i++){
For( ; $a<=$end; $a++){
$array[$i] = $array[$i].$myarray[$a];
}
$end = $end + $long + 1;
}
return $array;
}
function getDTMF($term, $i) {
global $agi;
for($x=1;$x<$i;$x++) {
unset($res);
$res = $agi->agi_getdtmf(1,1,$term,$prompt=FALSE);
$agi->conlog("Res: " . $res[0]);
if($res[0] == "") {
break;
}
if($res[0] != "") {
$result1 .= $res[0];
//$agi->conlog("Result var: " . $result1);
}
$agi->conlog("Result var: " . $result1);
$agi->conlog("Channel: " . $agi->request["agi_channel"]);
$status = $agi->agi_channel_status($agi->request["agi_channel"]);
$agi->conlog("Status code: " . $status["status"]);
$agi->conlog("Description: " . $status["description"]);
if($status["description"] != "Line is up") {
exit();
}
}
return $result1;
}
?>
___
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] PHP/AGI Problem
Moises Silva wrote:
what version of asterisk you are using? i had some problems with agi
until i upgrade to asterisk 1.0.7
I am also using version 1.0.7. I installed it from the Xorcom CD ISO
if you run simple agi scripts works?
try using this 2, one is php, the other one is C, that will tell us if
the problem is the script you are using, php, or AGI itself...
OK both the PHP script and the C program worked as expected. The call
connected, the numbers were read out and the * server hungup. The AGI
script/program exited normally.
Now when I change the program to this
#include
main()
{
char line[80];
setlinebuf(stdout);
setlinebuf(stderr);
while (1)
{
fgets(line,80,stdin);
if ( strlen(line) <= 1 )
{
break;
}
}
while (1)
{
printf("SAY NUMBER 55 \"\"\n");
fgets(line,80,stdin);
printf("SAY NUMBER 66 \"\"\n");
fgets(line,80,stdin);
}
}
The numbers are read out continuously as expected but If the caller
hangs up the app does not exit. The console log say the caller hung up
but the app still shows in the process list. The same happened if I
changed the PHP script in the same way.
I am therefore assuming that AGI expects the script to detect the caller
hung up and to act accordingly. Therefore I guess my question is how do
I detect the caller hungup while a AGI script is running?
Regards
Jon
___
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] PHP/AGI Problem
Matthew Boehm wrote: You using phpagi v2? Some of your functions are built in I believe. No, I am not using phpagi v2, I will take a look, None of my AGI's loop but if someone hangs up in the middle of script execution the script dies. I'm almost 99% sure of that. All of my non-looping scripts exit ok. However what happens if a caller hangsup before the script has finished? Regards Jon ___ 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] PHP/AGI Problem
Benjamin West wrote: So if the user stays on the line, the php script never blocks or hangs, and the phone call terminates correctly, including the php script. However, if the user hangs up the phone, your php script never times out because it gets stuck in a state that doesn't count towards time out? OK can you explain what you mean by "script never blocks"? That doesn't make any sense does it? Lets say the state that doesn't count is accessing a database, or a system call. Why would your script get stuck in this state? What about executing a database query or executing a system() is dependent on the user being on the line in order to return to php execution? For example, your script is off querying a database, the user hangs up in the midst of it, and then your script should finish the query, possibly block on a write/read via AGI and then timeout as documented. In order for what you're saying to be true, it would have to go something like the following: script makes database query, user hangs up, script is stuck in database query only because the use hung up. Something about that scenario doesn't make sense. The script operates normally call from the commandline or if I take the loop out. Regards Jon e. ___ Yahoo! Messenger - NEW crystal clear PC to PC calling worldwide with voicemail http://uk.messenger.yahoo.com ___ 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] Little Php question
Ronald wrote: When starting the script I get a parse error (unexpected t_string) in line 15 which is the Exten line Can anybody help me out. (I have minimal php knowledge, so Im turning to you all) Change fputs($socket, "Exten: 12345678\r\n\); to fputs($socket, "Exten: 12345678\r\n"); Regards Jon ___ How much free photo storage do you get? Store your holiday snaps for FREE with Yahoo! Photos http://uk.photos.yahoo.com ___ 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] PHP/AGI Problem
Alex Barnes wrote:
Currently after:
$res = $agi->agi_getdtmf(1,1,$term,$prompt=FALSE);
You test for no DTMF and then simply return null.
Instead you could call the other piece of code you have:
$status = $agi->agi_channel_status($agi->request["agi_channel"]);
$agi->conlog("Status code: " . $status["status"]);
$agi->conlog("Description: " . $status["description"]);
if($status["description"] != "Line is up") {
exit();
}
Alex,
You are a star! I changed that bit of code based on what you said to this
function getDTMF($term, $i) {
global $agi;
for($x=1;$x<$i;$x++) {
unset($res);
$res = $agi->agi_getdtmf(1,1,$term,$prompt=FALSE);
if($res[0] == "") {
$status =
$agi->agi_channel_status($agi->request["agi_channel"]);
//$agi->conlog("Status code: " . $status["status"]);
//$agi->conlog("Description: " .
$status["description"]);
if($status["description"] != "Line is up") {
exit();
}
}
$agi->conlog("Res: " . $res[0]);
if($res[0] == "") {
break;
}
if($res[0] != "") {
$result1 .= $res[0];
//$agi->conlog("Result var: " . $result1);
}
$agi->conlog("Result var: " . $result1);
$agi->conlog("Channel: " . $agi->request["agi_channel"]);
$status =
$agi->agi_channel_status($agi->request["agi_channel"]);
$agi->conlog("Status code: " . $status["status"]);
$agi->conlog("Description: " . $status["description"]);
if($status["description"] != "Line is up") {
exit();
}
}
return $result1;
}
Now the script loops forever while the user is connected and exits if
the user hangs up.
Thanks to everyone who helped me out, much appreciated.
Jon
___
Yahoo! Messenger - NEW crystal clear PC to PC calling worldwide with voicemail http://uk.messenger.yahoo.com
___
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] PHP/AGI Problem
Michael Stearne wrote: Jon, What version of PHPAGI are you using? I am starting a PHPAGI app and want to know whether to use 1.12 or 2.0CVS. I am using 1.12 Regards Jon ___ How much free photo storage do you get? Store your holiday snaps for FREE with Yahoo! Photos http://uk.photos.yahoo.com ___ 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] PHP/AGI Problem
Benjamin West wrote: Michael, The version, in the context of Jon's problem, was irrelevant. Jon's problem was due to a small bug in his code, and not related to PHPAGI. Hi Benjamin, Actually I would say it was more to do with my lack of understanding with how Asterisk AGI worked and my relative inexperience with blocking/non-blocking streams. However with the help of this list I am pleased with I know have a fully working IVR app now and the first time I touch Asterisk was a week ago so I am chuffed :-) Regards Jon ___ Does your mail provider give you FREE antivirus protection? Get Yahoo! Mail http://uk.mail.yahoo.com ___ 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] Problem with SIP peer registration
Hi I am trying to get 2 incoming SIP accounts working from 2 different providers. One is sipgate.co.uk and the other is voipuser.org. If I load the Register command seperate they will both register phone and incoming works. If I try to load them both only sipgate registers. Anybody got any suggestions why? Regards Jon ___ Yahoo! Messenger - NEW crystal clear PC to PC calling worldwide with voicemail http://uk.messenger.yahoo.com ___ 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] Dial plan question
> I need to call a sip extension for 15 seconds, if > unanswered I then need to > call the same sip extension and an additional sip > extension for a further 15 > seconds, finally if the calls aren't answered I need > it to go to a generic > unavailable VM. > My question is if the first sip extension is busy, > and I don't have the "100 > + x" busy VM defined will it just carry on to the > next priority without > complaining or is there a more elegant way of > achieving this? > > > > Example of my dialplan: > > > > exten => 0870xxx,1,Wait(2) > > exten => 0870xxx,2,Answer() > > exten => 0870xxx,3,Playback(cust-greeting) > > exten => 0870xxx,4,SetCIDName(Tech) > > > exten => 0870xxx,5,Dial(SIP/4902,15,tr) > > exten => 0870xxx,6,Dial(SIP/4902&SIP/4903,15,tr) > > exten => 0870xxx,7,Voicemail(u7003) > > exten => 0870xxx,8,Hangup Hi Chris Yes that will work and is as you say a simple and fairly straightforward way of doing what you require. Regards Jon Jon Farmer Telford, Shropshire, UK ___ All new Yahoo! Mail "The new Interface is stunning in its simplicity and ease of use." - PC Magazine http://uk.docs.yahoo.com/nowyoucan.html ___ --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] Call queue drops call after 1 min
> I'm trying to setup a call queue, but it keeps > dropping calls that are > waiting for 1 min. Is there any way to make the > queue unlimited amount of > time waiting? or is there a maximum? Hi Make sure you are not setting the timeout parameter on the Queue command. Failing that can you post the CLI output when this happens and the relevant portions of your extension.conf and queues.conf Regards Jon Jon Farmer Telford, Shropshire, UK ___ Copy addresses and emails from any email account to Yahoo! Mail - quick, easy and free. http://uk.docs.yahoo.com/trueswitch2.html ___ --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] Sony Ericsson F250m, Sipura 3000 and Asterisk
Hi I have been asked if it possible to connect a SE F250M to Asterisk. I have never used one of these devices before but from what I have gathered they need a FXO interface. As the Asterisk box is hosted remotely would it possible to use a Sipura 3000 to provide the FXO interface and successfully use the F250M. If anyone has any pointers on this I would be grateful. Regards Jon -- Jon Farmer Telford, Shropshire, UK ___ --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] Sony Ericsson F250m, Sipura 3000 and Asterisk
Julian J. M. wrote: > I didn't test it with a Sipura, but a TDM400. You can check this page > for configuration codes for the F251M. > http://blog.julianmenendez.es/configuracion-fct-ericsson-f251m (In > Spanish). If the SPA-3000 supports detecting polarity reversals, > you'll need them. Thanks for that.. According to page 50 of this document http://www.sipura.com/Documents/SipuraSPAUserGuidev2.0.9.pdf it does support detecting polarity reversals so it looks promising. I would still be interested in hearing from anyone who actually has it working before purchasing the kit. Regards Jon -- Jon Farmer Telford, Shropshire, UK ___ --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] load average with MOH
Hi I experimented with using the native MOH player with Asterisk 1.2.x instead of using mpg123. However I discovered that with a queue playing MOH to 20 waiting callers I was getting a load average 1.00+ using the Asterisk native compared to 0.08 using mpg123. Is this normal? If so what is the incentive to use native MOH? Regards Jon -- Jon Farmer Telford, Shropshire, UK ___ --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] Check call duration on active call in CLI?
Hermann Wecke wrote: > voiplist wrote: >> Is there a command to check the call duration of an active call in >> the CLI? > > show channels verbose show channel shows among other things Elapsed Time: 0h2m47s Regards Jon -- Jon Farmer Telford, Shropshire, UK ___ --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] Asterisk and Siemens Legacy PBX
James Arscott wrote: > Hi, thanks to the original poster, I redid all the cabling and immediately > got the span to go OK between asterisk and the siemens legacy PBX. Only > problem now is working out how to handle the calls from the siemens > Worth pointing out at this stage I have no access to the siemens > configuration, so I could be shooting blind. > > I put span2 (which is connected to the siemens) into its own context > (inbound-from-siemens) and then tried to few simple attempts at Œreceiving¹ > the calls that the siemens is trying to make. However whatever I put all I > get via the asterisk console is : > > -- Extension '' in context 'inbound-from-siemens' from 'xx' does not > exist. Rejecting call on channel 0/31, span 2 > > That comes up each time a call is attempted from the siemens, the xx > shows as whichever direct dial number tried to dial out on the siemens, > which I initially was pleased to see, however I am now stumped at how I > should try to get asterisk to deal with these calls, am I barking up the > wrong tree ? No you are slowing barking up the right tree :-) The call is getting accepted by Asterisk in the context inbound-from-siemens. However it can't work out what to do with the call. You need to match the xx number with a extension number which is in the inbound-from-siemens context or another context included in it. For instance if the xxx number is 123456 you could use. [inbound-from-siemens] exten => 123456,1,Dial(SIP/101) to dial SIP phone 101 or if the numbers from the siemens follow a pattern ie they all start with 12 then you could use exten => _12,1,Dial(SIP/101) If you check the extensions.conf page at www.voip-info.org/wiki you will see loads of examples on how to construct a dialplan HTH Jon -- Jon Farmer Telford, Shropshire, UK ___ --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] Asterisk and Siemens Legacy PBX
James Arscott wrote: > I also tried just using s , this again did not work. I assumed the > ‘Extension ‘’ in context’ part of my debug meant that the siemens is not > sending, or asterisk can’t work out, what extension is being sent If > that makes sense It means that whatever context you have defined for the Zap span can't find a extension with the number the siemens is dialling. Look at the zap span config and see what context is defined and then make sure that context has the right extenensions defined. > Also to help me get my head around this, the ‘extension’ referred to > that should be being sent from the siemens, is this going to be the > number the siemens is dialing, if not, how do I get ‘access’ to that number? Yes its the number the siemens is dialling. > My goal is to just allow the siemens to make any call it wants via the > span 1 on the asterisk box, which is connected to a ‘real’ ISDN PRI. This is a everyday use for Asterisk :-) -- Jon Farmer Telford, Shropshire, UK ___ --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] PHP-AGI help
Yes you have a parse error in your PHP when I saved it locally and run it from the command line I got syntax error, unexpected '[', expecting ']' in test.php on line 33 Jon FarmerTelford, Shropshire, UK - Original Message From: Matthew Warren <[EMAIL PROTECTED]> To: asterisk-users@lists.digium.com Sent: Friday, 2 June, 2006 3:32:10 PM Subject: [Asterisk-Users] PHP-AGI help Can someone help me with this AGI script to send an email. It just isn't working. The file is being called in the dialplan and is saved as em.agi but it isn't sending the email. #!/usr/bin/php4 -q ___ --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
Re: [Asterisk-Users] How to retrieve voicemail
Victor Moreno wrote:
> Hi,
> voicemail are working ok, I receive message as attach via email.
> My question is :
> how can the user call asterisk and listen to his voicemessages ?
Set up a exten to voicemailmain passing the calling exten as the argument.
e.g.
exten => 121,1,VoiceMailMain(u${exten})
HTH
Jon
--
Jon Farmer
Telford, Shropshire, UK
___
--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] Problems with wcte11xp module
Hi I am in the process of commissioning a new * box for our sister company. Unlike us they want their incoming calls delivered on a ISDN 30 not SIP. I have got a TE110P for this project and have compiled the zaptel stuff. However when I modprobe wcte11xp it loads ok but all audio on SIP channels is lost. If I rmmod the driver then audio returns. What is going on? Any ideas? Regards Jon Jon Farmer Telford, Shropshire, UK ___ Yahoo! Messenger - NEW crystal clear PC to PC calling worldwide with voicemail http://uk.messenger.yahoo.com ___ --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] time update (7905)
--- Michiel van Baak <[EMAIL PROTECTED]> wrote: > On 09:16, Tue 28 Mar 06, Tomislav Vojvodic wrote: > > Hi everyone, > > > > I'm trying to update time on all Cisco 7905 phones > in my company.. is there > > some way to do it from asterisk? Don't have any 7905's but on our 7940's you set the DST settings in the SIPDefault.cnf. Then if you have told the phones to use NTP they update automatically. Regards Jon Jon Farmer Telford, Shropshire, UK ___ To help you stay safe and secure online, we've developed the all new Yahoo! Security Centre. http://uk.security.yahoo.com ___ --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] Problems with wcte11xp module
--- [EMAIL PROTECTED] wrote: > It means that you are loading the digium card up > with incorrect values. > > I had it happen to me recently. Aha I wonder that, are you referring to the span definition in the /etc/zaptel.conf? Regards Jon Jon Farmer Telford, Shropshire, UK ___ To help you stay safe and secure online, we've developed the all new Yahoo! Security Centre. http://uk.security.yahoo.com ___ --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] Problems with wcte11xp module
--- [EMAIL PROTECTED] wrote: > It means that you are loading the digium card up > with incorrect values. > Ok when I modprobe wcte11xp I get the following message ZT_CHANCONFIG failed on channel 26: No such device or address Any ideas? Regards Jon Jon Farmer Telford, Shropshire, UK ___ To help you stay safe and secure online, we've developed the all new Yahoo! Security Centre. http://uk.security.yahoo.com ___ --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 Plan Logic Problem
--- CC Jay <[EMAIL PROTECTED]> wrote: > Your (abridged) dialplan looks OK to me. > Nonetheless, you should: > 1) restart * then try again, and if that doesn't > work > 2) make sure you load the "correct" extensions.conf > (the one you think > you're loading) > Good luck! //lists.digium.com/mailman/listinfo/asterisk-users I sorted it, I needed to include the campon context before the mainmenu context in the default context. Regards Jon Jon Farmer Telford, Shropshire, UK ___ Yahoo! Photos NEW, now offering a quality print service from just 8p a photo http://uk.photos.yahoo.com ___ --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 Plan Logic Problem
Hi
I can't for the life of me work out why this is not
working. When in the campon contect if you hit a DTMF
key 2 you get moved to the exten => 2 defined in the
mainmenu context not the exten => 2 defined in the
campon context. What is wrong? The same happens if you
hit key 1.
[campon]
exten => _*1XXX,1,Answer
exten => _*1XXX,2,SetCallerID(${CALLERIDNUM})
exten => _*1XXX,3,SetVar(CALLEDEXTEN=${EXTEN:2,3})
exten => _*1XXX,4,ResponseTimeout(3)
exten => _*1XXX,5,Background(entagroup/campon)
exten => _*1XXX,6,SetVar(LOOPER=1)
exten => _*1XXX,7,Background(entagroup/silence)
exten => _*1XXX,8,NoOp()
exten => _*1XXX,9,GotoIf($[${LOOPER} < 10]?10:13)
exten => _*1XXX,10,Dial(Local/${CALLEDEXTEN},5,trm)
exten => _*1XXX,11,SetVar(LOOPER=$[${LOOPER} + 1])
exten => _*1XXX,12,Goto(9)
exten => _*1XXX,13,Goto(4)
exten => _*1XXX,14,Hangup
exten => 1,1,VoiceMail(b${CALLEDEXTEN})
exten => 1,2,Hangup
exten => 2,1,SetCallerID("Camped on ${CALLEDEXTEN}")
exten => 2,2,Goto(huntgroups,101,1)
exten => 2,3,Hangup
[mainmenu]
exten => s,1,Set(LOOPER=1)
exten => s,2,ResponseTimeout(6)
exten => s,3,Background(entagroup/mainmenu)
exten => s,4,Background(entagroup/silence)
exten => s,5,Set(LOOPER=$[${LOOPER} + 1])
exten => s,6,GotoIf($[${LOOPER} < 4]?mainmenu,s,2)
exten => s,7,Goto(huntgroups,0,1)
exten => t,1,GotoIf($[${LOOPER} < 4]?mainmenu,s,2)
exten => t,2,Hangup
exten => i,1,Goto(mainmenu,s,1)
exten => 1,1,Goto(sales,s,1)
exten => 2,1,Goto(finance,s,1)
exten => 0,1,Goto(huntgroups,0,1)
exten => #,1,Goto(mainmenu,s,1)
Jon Farmer
Telford, Shropshire, UK
___
Yahoo! Messenger - NEW crystal clear PC to PC calling worldwide with voicemail
http://uk.messenger.yahoo.com
___
--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] Cisco 7960 problems
Paul A Brown wrote: Hi All, Not sure if this is a phone problem or an Asterisk problem. Basically after a period of time (around 30 minutes but not too sure of the time) the phone no longer delivers any sounds. What I mean by that is. if I pick up the phone after a reset I get a dialtone. After around 30 minutes and I pick up phone I get no dial tone but I can still dial. I dialled the voicemail number, I can see on the asterisk console its asking for which vmail box and password but I hear nothing. Anyone heard anything like this before? What firmware are you using with the phone? SIP or SCCP? I have 2 7960's with 7914's attached using the latest chan_sccp and have not problems like describe. Regards Jon ___ --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] Cisco 7960 problems
Paul A Brown wrote: > Do you have a sccp config example I could look at > http://www.voip-info.org/wiki/view/SCCP-HOWTO2 Regards Jon ___ --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] Manager API Help
Darren Ellis wrote: > Hi All, > > Could someone send me a code frag on how to get a record from the > asterisk database into a PHP variable via the Manager API? > > I can issue calls, etc. from Manager. But I'm not comprehending how to > manipulate database variables. Google for phpagi, it is a class that implements the manager API. -- Jon Farmer Telford, Shropshire ___ --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] Call Forward and AGI
Hi i have a agi script that gets called when a user wants to dialout externally. it gets passed in the exten number and the number dialled and looks up in a db to see if they are allowed to dial the number. the problem is if someone forwards their phone to a external number the CALLERIDNUM is the CLID of the calling party not the extension forwarded thus the call is denied. Can anyone think of a way around this? -- Jon Farmer Telford, Shropshire ___ --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] Variables
Shaun wrote: > I have a call screening system setup, caller calls in runs a macro and sets > a far to track the recording that was taken of the callers name... then the > callee runs a macro also that plays him that recording (pulled from that var > that was set) This works fine until i use a queue in the middle of it > all... it appears that with queues that the file name stored in a var called > SCREEN_FILE is lost once the caller is taken out of the queue.. Is their a > uniq ID or somthing thats set to each call that i can use as the file name > so i can always play back that file that was recorded or is their a way to > to not loose the value of SCREEN_FILE once the caller is put into the queue? > I though about setting SCREEN_FILE as global but i think that will cause > problems with multiple calls and SCREEN_FILE being overwritten by other > callers and the screening macro running... > > If each call had a uniq session id i could easily just use that > > http://www.voip-info.org/wiki/view/Asterisk+variables See section about variable inheritance. Regards Jon -- Jon Farmer Telford, Shropshire ___ --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] Asterisk and SER hangup issue
Hi I have can get my phones to register with SER and dialout for PSTN via my Asterisk box over a SIP channel to my VoIP provider. If the phone requests hangup then the bridged channel on Asterisk gets destroyed however if the called party hangups the channel stays up and the phone connected. Anybody got any ideas? Regards Jon -- Jon Farmer Telford, Shropshire, UK ___ --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] billing realtime
JP Carballo wrote: > Yes, certainly, through deadagi. > I just have one question though, why reinvent the wheel? > There are prepaid systems that work with asterisk. > I have yet to find a prepaid system that allows multiple concurrent calls per account. Most seem to be based on a pin number also which I don't want. Anyone know of a system that allows concurrent calls? -- Jon Farmer Telford, Shropshire, UK ___ --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] billing realtime
Nick Hoffman wrote: > Hi Jon. If a customer has 10 minutes of call credit left and he makes 2 > concurrent calls, how do you know to cut off the 2 calls at the 5 minute > mark rather than cut off both calls after 10 minutes? That is the problem I am asking about :-) -- Jon Farmer Telford, Shropshire, UK ___ --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] Re: billing realtime
Tony Mountifield wrote: > The other situation to take account of is when the caller somehow adds > to his prepaid balance while he has one or more calls in progress, in > order to avoid being cut off during the call. Yes, this is a issue that needs to be considered. Also each call might be on a different cost per minute depending on the number called e.g. in the UK geographic calls are costed lower then mobile calls. The only solution I can think of at the moment is to write a daemon that uses the manager interface to hold all calls in memory and manages the current call credit available at the current time per account. If the credit expires for that account it hangs up all channels for that account. The only problem at the moment is I can't figure away to dynamically play a warning to the callers. -- Jon Farmer Telford, Shropshire, UK ___ --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] Re: billing realtime
Won't the called party hear the warning as well if you do that? Jon FarmerTelford, Shropshire, UK - Original Message From: Tony Mountifield <[EMAIL PROTECTED]> To: asterisk-users@lists.digium.com Sent: Wednesday, 26 April, 2006 3:08:18 PM Subject: [Asterisk-Users] Re: billing realtime Instead of hanging up the channel, transfer it (Action: Redirect) to an extension that does Playback(warning) followed by Hangup. You can send both caller and callee there if you use the ExtraChannel parameter to Redirect. Otherwise transferring one drops the other. ___ --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] Camp on?
I believe what you refer to is called "Ring Back When Free" at least thats how I know it in the UK. Regards Jon Jon FarmerTelford, Shropshire, UK - Original Message From: Patrick <[EMAIL PROTECTED]> To: asterisk-users@lists.digium.com Sent: Wednesday, 26 April, 2006 3:25:12 PM Subject: [Asterisk-Users] Camp on? Hi all, In .nl there is a feature provided by the incumbent that I would like to implement for an internal PBX setup. The incumbents feature does the following (adopted for internal PBX use, so no external/PSTN numbers are used): 1) pick up phone and dial an internal extension 2) if other side is busy, play a message "press 5 to get connected once the other side becomes available" 3) press 5 on phone 4) hangup 5) wait till phone starts ringing 6) pick up phone 7) other extension is automatically dialed again and you should hear it ring I believe this is called camp on. Found some examples on voip-info.org but they assume that you do not hangup the originating phone. Anyone have an idea how to implement this feature as described above? Thanks and regards, Patrick ___ --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] voicemail dialout
Hi How do i disable dialling out from voicemail? -- Jon Farmer Telford, Shropshire, UK ___ --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] voicemail dialout
Doug Lytle wrote: > It's enabled/disabled via the voicemail.conf I have commented out dialout=from-vm but the option is still given even though any number dialled results in unobtainable. So I dont want the option given. -- Jon Farmer Telford, Shropshire, UK ___ --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] voicemail dialout
Doug Lytle wrote: > You'll also need to do a stop/start of Asterisk. Done that also, no difference -- Jon Farmer Telford, Shropshire, UK ___ --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] voicemail dialout
Doug Lytle wrote: > Let's see that section of your voicemail.conf ; tz=central; Timezone from zonemessages above. Irrelevant if envelope=no. ; attach=yes; Attach the voicemail to the notification email *NOT* the pager email ; saycid=yes; Say the caller id information before the message. If not described, ; or set to no, it will be in the envelope ; cidinternalcontexts=intern; Internal Context for Name Playback instead of extension digits when saying caller id. ; sayduration=no; Turn on/off the duration information before the message. [ON by default] ; saydurationm=2; Specify the minimum duration to say. Default is 2 minutes ;dialout=from-vm; Context to dial out from [option 4 from the advanced menu] ; if not listed, dialing out will not be permitted sendvoicemail=no; Context to Send voicemail from [option 5 from the advanced menu] ; if not listed, sending messages from inside voicemail will not be ; permitted ; searchcontexts=yes; Current default behavior is to search only the default context ; if one is not specified. The older behavior was to search all contexts. ; This option restores the old behavior [DEFAULT=no] ; callback=fromvm ; Context to call back from ; if not listed, calling the sender back will not be permitted ; review=yes; Allow sender to review/rerecord their message before saving it [OFF by default operator=no ; Allow sender to hit 0 before/after/during leaving a voicemail to ; reach an operator [OFF by default] ; envelope=no ; Turn on/off envelope playback before message playback. [ON by default] ; This does NOT affect option 3,3 from the advanced options menu ; delete=yes; After notification, the voicemail is deleted from the server. [per-mailbox only] ; This is intended for use with users who wish to receive their voicemail ONLY by email. ; nextaftercmd=yes ; Skips to the next message after hitting 7 or 9 to delete/save current message. ; [global option only at this time] ; forcename=yes ; Forces a new user to record their name. A new user is ; determined by the password being the same as ; the mailbox number. The default is "no". ; forcegreetings=no ; This is the same as forcename, except for recording ; greetings. The default is "no". ; hidefromdir=yes ; Hide this mailbox from the directory produced by app_directory ; The default is "no". Jon Farmer Telford, Shropshire, UK ___ --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] voicemail dialout
Doug Lytle wrote:
> What do you see at the console when someone presses 4 from voice mail?
-- Executing VoiceMailMain("SIP/502-ac3f", "s502") in new stack
-- Playing 'vm-youhave' (language 'en')
-- Playing 'vm-no' (language 'en')
-- Playing 'vm-messages' (language 'en')
-- Playing 'vm-opts' (language 'en')
-- Playing 'vm-tomakecall' (language 'en')
-- Playing 'vm-starmain' (language 'en')
--
Jon Farmer
Telford, Shropshire, UK
___
--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] voicemail dialout
Doug Lytle wrote: > Jon, > > I don't know. I went into my voicemail.conf and put a semicolon in > front of that option, > > Re-attached to the Asterisk console and did a reload and the option was > no longer available from Advanced Options. > > I'm running 1.2.7.1 Do you mean the option was not offered to you or it did not work when offered. I am also on 1.2.7.1 -- Jon Farmer Telford, Shropshire, UK ___ --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] Dumping queue_log to MySQL
We use a Perl script called acd_logger.pl. I can't remember when I got it from now but if you want a copy then mail me directly.RegardsJon Jon FarmerTelford, Shropshire, UK- Original Message From: Kevin Savoy <[EMAIL PROTECTED]>To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Friday, 5 May, 2006 2:57:06 PMSubject: [Asterisk-Users] Dumping queue_log to MySQL Anyone have a working solution for this? I played with the demo that came with QueueMetrics to see how they were doing it and it was working for a bit but now somehow every night it stopped. Perl and Tail are still running on the server but the information is not dumping to the MySQL database. I don’t get any error messages anywhere telling me why it stops. As far as tail and perl are concerned everything is fine. We will be using this for a call center and need more reliability. Anyone got one working? Thanks _ Kevin Savoy Business Unit Telecom Analyst 2218 4th Ave W Williston, ND 58801 Ph: 701-774-4023 Fax: 701-774-2901 http://www.novo1.com Novo 1 is a service mark of Novo 1, Inc ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo 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
Re: [asterisk-users] Reception Console
Hi Yes I am interested. Regards Jon --- Paul Hales <[EMAIL PROTECTED]> wrote: > > We are currently writing a reception console for > Asterisk - if anyone is > interested in beta testing it, feel free to ask. > > Paul Hales > > -- > Paul Hales > Technical Manager > AsteriskIT > www.asteriskit.com.au > bus: 03 8320 8106 > mob: 0434 673 529 > > ___ > --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 > Jon Farmer Telford, Shropshire, UK ___ All new Yahoo! Mail "The new Interface is stunning in its simplicity and ease of use." - PC Magazine http://uk.docs.yahoo.com/nowyoucan.html ___ --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] Registration problem
Sergio R. D'Ippolito wrote: > Hi all, i have an * version: Asterisk SVN-branch-1.2-r45691, I need to > register a linksys 922 phone thru internet and when I make sip debug > command i see this debug information: > */SIP/2.0 401 Unauthorized/* > > /Via: SIP/2.0/UDP x.x.x.x:1025;branch=z9hG4bK-43bf8123;received=x.x.x.x/ > > /From: "SPA922" ;tag=685bbad1fae3325do0/ > > /To: "SPA922" ;tag=as4da6f6ce/ > > /Call-ID: [EMAIL PROTECTED]/ > > /CSeq: 5503 REGISTER/ > > /User-Agent: incore-PBX/ > > /Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY/ > > /WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="372b2479"/ Asterisk is asking the phone to resend the registration with WWW-Authenticate using MD5 hash. Make sure the phone supports this and retry. Or you could turn this option off in the sip.conf. Regards Jon -- Jon Farmer Telford, Shropshire, UK ___ --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] SIP v IAX2
Henry.L.Coleman wrote: > Its a bit like the VHS vs Beta war, both systems have their good and bad > points In the end, sales/marketing perception will always win regardless > of better technologies. That will be Skype then ;-) -- Jon Farmer Telford, Shropshire, UK ___ --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] rtptimeout on 1.8.4
Hi Since switching from 1.6.x to 1.8.4 I have noticed the following 1. When you do a 'core show channel ' the resulting information only shows data for "Frames In" , "Frames out" is always 0. 2. The rtptimeout option in the sip.conf no longer seems to work. I have this set to 60 seconds but have had channels which have not timeout when the rtp stops. If I subsequently do a "channel request hangup" then the CLI reports that there was a rtptimeout but will be hugely over the set amount. For instance on requesting the hangup on one such channel today it reported rtp timeout for 18000 seconds. Anybody got any ideas how to get 1.8.4 rtptimeout correctly? Regards Jon Jon Farmer Tel 07795 118140 -- _ -- 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] SIP Trace to troubleshoot one way of communications
On Aug 26, 2011 4:54 PM, "bilal ghayyad" wrote: > > Hi All; > > How can I get a SIP trace to troubleshoot a one way of communications? I need to see what is happenning in the packets to know the reason of the problem. > Install ngrep on the box. Then type something like. ngrep -tq -W byline port 5060 Replace with the IP of the UA you want to monitor. Regards Jon Sent from my iPad3. -- _ -- 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] Problem With Playing Busy Tone
Hi Since upgrading to 1.8.4.3 my callers no longer hear busy tone when I use playtones(). Here is the CLI output on such a case http://pastebin.com/TMBFhngh Any ideas anyone? Regards Jon -- _ -- 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] Asterisk 1.8.9.3 Problem With Logger
Hi I have recently upgraded a box to 1.8.9.3 and have noticed that randomly the logger will just stop working. It stops logging to the console and to the log files. Reloading logger actually freezes the console. It seems to happen when Asterisk tries to rotate it's log files but it may happen at other times too, The only way I have managed to get it back is to kill and restart asterisk. Any ideas what is going on. Regards Jon Jon Farmer Tel 07795 118140 -- _ -- 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] Asterisk 1.8.9.3 Problem With Logger
Is this a general issue or just affecting specific versions? Jon Farmer Tel 07795 118140 On 7 March 2012 16:33, Paul Belanger wrote: > On 12-03-07 04:29 AM, Jon Farmer wrote: >> >> Hi >> >> I have recently upgraded a box to 1.8.9.3 and have noticed that >> randomly the logger will just stop working. It stops logging to the >> console and to the log files. Reloading logger actually freezes the >> console. It seems to happen when Asterisk tries to rotate it's log >> files but it may happen at other times too, The only way I have >> managed to get it back is to kill and restart asterisk. Any ideas what >> is going on. >> > I'm pretty sure there is an existing issue in JIRA about this. Try > searching it first, if not open a new issue so we can triage it. It is > likely a deadlock so attach the required information for it. > > -- > Paul Belanger > Digium, Inc. | Software Developer > twitter: pabelanger | IRC: pabelanger (Freenode) > Check us out at: http://digium.com & http://asterisk.org > > -- > _ > -- 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 -- _ -- 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] Asterisk 1.8.9.3 Problem With Logger
Hi Just realised this is due to a FIFO blocking. Fixed that and all back to normal. Regards Jon Jon Farmer Tel 07795 118140 On 7 March 2012 16:33, Paul Belanger wrote: > On 12-03-07 04:29 AM, Jon Farmer wrote: >> >> Hi >> >> I have recently upgraded a box to 1.8.9.3 and have noticed that >> randomly the logger will just stop working. It stops logging to the >> console and to the log files. Reloading logger actually freezes the >> console. It seems to happen when Asterisk tries to rotate it's log >> files but it may happen at other times too, The only way I have >> managed to get it back is to kill and restart asterisk. Any ideas what >> is going on. >> > I'm pretty sure there is an existing issue in JIRA about this. Try > searching it first, if not open a new issue so we can triage it. It is > likely a deadlock so attach the required information for it. > > -- > Paul Belanger > Digium, Inc. | Software Developer > twitter: pabelanger | IRC: pabelanger (Freenode) > Check us out at: http://digium.com & http://asterisk.org > > -- > _ > -- 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 -- _ -- 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] Multiple Registrations
Hi I am researching if there is a practical number of SIP accounts that Asterisk can register against as a UA. I have an idea for a project but it would need to register multiple accounts from multiple providers to work. Regards Jon -- Jon Farmer Tel 07795 118140 -- _ -- 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] Question About Codecs
Hi I have a call into a MeetMe conference that when I do a "core show channel" returns NativeFormats: 0x4 (ulaw) WriteFormat: 0x1000 (g722) ReadFormat: 0x1000 (g722) Can someone explain what the differences between Native, Wite and Read are? Regards Jon -- Jon Farmer Tel 07795 118140 -- _ -- 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] RTP and Signalling Dropping
Hi I have a weird issue with a new 1.6.2.17.2 box. At random intervals it just stops responding to RTP and signalling (both SIP and IAX observed). All calls in progress lose audio both ways although the console shows the call legs still in progress. No signalling can be sent or is received. It is as though the server drops of the net for those protocols. I can still navigate the console. Killing an restarting Asterisk is the only way to bring it back. I can see nothing in the logs to indicate what is happening. The server is dual homed network one interface on a public address and the other interface on a private subnet that the phones sit on. It can do 100's even 1000's of calls before the issue happens and then BAM it drops off. The box is handling between 1500 - 3000 calls a day, mostly SIP and IAX with a small percentage of DADHI. Anyone any ideas what is going on or where to look next? Regards Jon -- Jon Farmer Tel 07795 118140 -- _ -- 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 Delimiter in 1.6
Hi I am currently using 1.2.x and 1.4.x behind OpenSER. One of the things I do on INVITES is to re-authenticate the user from OpenSER. Then when the INVITE gets passed to Asterisk I capture the AUTH to a variable in the dialplan and pass to an AGI script. I am now trying to set the same thing up in 1.6 However because the argument delimter in 1.6 has changed from pipe to comma this breaks as the AUTH line is also comma delimited. Thus the AGI sees the AUTH as extra arguments instead of a single argument. As the AUTH may contain varying number of arguments I need a new way for a my AGI to access this data. Does anyone have any ideas how I might go about this? Regards Jon -- Jon Farmer Tel 07795 118140 -- _ -- 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] AGI Delimiter in 1.6
On 16 September 2010 19:50, Danny Nicholas wrote: > Two suggestions; > #1. "escape" the , as \, > #2. quote the string so 1,2,3 is "1,2,3" I have thought about both of those ideas. Is it possible to escape the string in the dialplan? Applying quotes didn't seem to work, however I was pretty tired when I tried so it might just need a fresh set of eyes. Regards Jon -- Jon Farmer Tel 07795 118140 -- _ -- 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] AGI Delimiter in 1.6
> On 16 September 2010 19:50, Danny Nicholas wrote:
> If you make the string into a dialplan Variable, you can do pretty much
> anything with it. Let's say your dialplan is like this
>
> - exten => 1234,1,blah
> - exten => 1234,n,AGI(myagi.xx,"1234")
>
> Change line 2 to
> - exten => 1234,n,AGI(myagi.xx,${VARNAME})
>
> Then you just "do your magic" on ${VARNAME}
Yes, but the problem is I am trying to pass the whole AUTH line which
is key=value pairs seperated by commas. e.g. username=myusername,
domain=mydomain
This breaks when passing to an AGI in 1.6.
--
Jon Farmer
Tel 07795 118140
--
_
-- 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] Playing Audio To One Channel
Hi I have a call established and I want to play audio to just one channel on that call. Is this possible? If so, how? My google-fu has failed on this one. Regards Jon -- Jon Farmer Tel 07795 118140 -- _ -- 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] Playing Audio To One Channel
On 20 September 2010 14:23, Danny Nicholas wrote: > One option would be to play your audio through a conference; Asterisk seems > to have great controls over legs using that infrastructure. > That is not an option. I am using Asterisk as a media relay and want to play a message to the subscriber when call credit is low. However I don't want the other party to hear the message. Regards Jon -- Jon Farmer Tel 07795 118140 -- _ -- 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] Playing Audio To One Channel
On 20 September 2010 14:21, Jim Dickenson wrote: > One way to do it is to use ChanSpy and the whisper option. We use AMI to play > sound bits to one leg of the call. > > Something like > Hi I have tried your suggestion however I can't get it to work. When I send the originate via the manager interface the extensions get fired and doing a show channels shows the chanspy and playbacks working but I hear nothing. Any ideas? Regards Jon -- Jon Farmer Tel 07795 118140 -- _ -- 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] AGI Delimiter in 1.6
On 16 September 2010 22:23, Barry Miller wrote: > For an interim fix, setting res_agi=1.4 in the [compat] section of > asterisk.conf should work. See UPGRADE-1.6.txt . I have tried this but it still complains about the pipe not being a comma. Regards Jon -- Jon Farmer Tel 07795 118140 -- _ -- 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] AGI Delimiter in 1.6
Hi, I fixed it in the end by adding the sip headers I was interested in as extra "x" headers in the openser config. Then just capturing these in the asterisk dialplan as variables. Simples. Regards Jon On 21 Sep 2010 16:03, "Jonas Kellens" wrote: > On 09/21/2010 04:22 PM, Jon Farmer wrote: >> On 16 September 2010 22:23, Barry Miller wrote: >> >> >>> For an interim fix, setting res_agi=1.4 in the [compat] section of >>> asterisk.conf should work. See UPGRADE-1.6.txt . >>> >> I have tried this but it still complains about the pipe not being a comma. >> >> Regards >> >> Jon >> > > Hello, > > in asterisk 1.4 this works : > > exten => s,n,Queue(queuenametimeout,test.agi^VAR) > > in asterisk 1.6 this works : > > exten => s,n,Queue(queuenametimeout,"test.agi,VAR") > > So you need " ". > > > Jonas. > > -- > _ > -- 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 -- _ -- 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
