[asterisk-users] 'NO ANSWER' with cdr duration of 0
I have a system running Asterisk 1.6.2.6 that generates about 80k calls/day. Calls are fired from Asterisk Manager (async originate -- 60 second timeout). I am capturing 100% of the originate responses (recorded in DB). About 5% of the calls result in a reason code of '3' (Remote End is Ringing). CDR Disposition of these calls is 'NO ANSWER', and billsec is 0. All good there. What's confusing me is that in *all* these cases duration is also recorded as 0. IIRC, duration should be time spent in the system, so should be roughly equal to my timeout (60 sec). Is this a bug, or am I not understanding correctly? I know there was a bug where all calls were recorded as 'NO ANSWER', I was plagued with that until updating to 1.6.2.6, this does not seem to be related. Any feedback is appreciated. Thanks, Josh McAllister -- _ -- 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] how to identify agi crash cause
Unfortunately, STDERR from AGI scripts does not make it to *’s log files. You can work around this by starting Asterisk under screen. Take a look at the AGI wiki under “CLI output” for more info. http://www.voip-info.org/wiki-Asterisk+AGI Be warned though, that when you re-attach to Asterisk’s screen make sure you detatch the screen, and do not Ctrl-C or exit the asterisk console as that will shutdown asterisk. Josh McAllister From: Danish Samad [mailto:[EMAIL PROTECTED] Sent: Thursday, June 08, 2006 11:37 AM Hi, Thanks for your reply. Dont the messages logged in /var/log/asterisk/messages contain error messages also dumped in tty9. If not how can I view the messages on the tty9 console, The problem is the server is hosted remotely and all I have is ssh access. Shutting down asterisk might not be an option, especially since I dont know how to reproduce the problem, it just happens sporadically. Regards, Danish On 6/8/06, Josh McAllister <[EMAIL PROTECTED]> wrote: STDERR from your agi will be shown on asterisk's tty. If you're using safe-asterisk to start, I believe this is redirected to tty9… Or, if you can afford to take asterisk down momentarily, you could just start asterisk without backgrounding it and you'll see what your script has to say there. Josh McAllister From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Danish Samad Sent: Thursday, June 08, 2006 8:25 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] how to identify agi crash cause Hi, I have a custom agi which at times does not exit gracefull and crashes in between. The logging options are set to the maximum but I dont see something conclusive in the asterisk log. I have noticed it crash after issuing the "SAY NUMBER" and "GET DATA" agi commands and the agi is spawned with no apparent reason after that. I tried running the application locally and debugged but could not reproduce the problem. I also tried enabling core file generation by specifying the following command in /etc/profile "ulimit -c unlimited > /dev/null 2>&1" but to no avail, I did not get any core file in /tmp or other locations. Can any one suggest a way to get a core dump of crashing agi's or some other way I can isolate the problem. Any help will be appreciated. Regards, Danish ___ --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 identify agi crash cause
STDERR from your agi will be shown on asterisk’s tty. If you’re using safe-asterisk to start, I believe this is redirected to tty9… Or, if you can afford to take asterisk down momentarily, you could just start asterisk without backgrounding it and you’ll see what your script has to say there. Josh McAllister From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Danish Samad Sent: Thursday, June 08, 2006 8:25 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] how to identify agi crash cause Hi, I have a custom agi which at times does not exit gracefull and crashes in between. The logging options are set to the maximum but I dont see something conclusive in the asterisk log. I have noticed it crash after issuing the "SAY NUMBER" and "GET DATA" agi commands and the agi is spawned with no apparent reason after that. I tried running the application locally and debugged but could not reproduce the problem. I also tried enabling core file generation by specifying the following command in /etc/profile "ulimit -c unlimited > /dev/null 2>&1" but to no avail, I did not get any core file in /tmp or other locations. Can any one suggest a way to get a core dump of crashing agi's or some other way I can isolate the problem. Any help will be appreciated. Regards, Danish ___ --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] Meetme and authentication
Title: Meetme and authentication Perhaps you’ve already figured this out, but I posted an example dialplan and small Perl AGI that would resolve this for you. As it happens this was posted the Friday before you sent this. Look for a posting from me on Friday, May 12th. Josh McAllister From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Herchi Silviu Sent: Tuesday, May 16, 2006 8:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Meetme and authentication Hi all, I have thoroughly read the available documentation and I can't seem to find a workaround for my setup… I'm trying to create a phone conference line that users would call using a unique phone number (no matter if they are moderators or just plain users). I use Asterisk 1.2.6 The available conferences are defined as follows: conf => 1000,user pin1, moderator pin1 conf => 1001,user pin2, moderator pin2 conf => 1002,user pin3, moderator pin3 … conf => 1009, user pin9, moderator pin9 The users are prompted whether they are a moderator or a user. When they choose, they are redirected to the conference they request: - using options aAPsX for moderators (moderator + marked + ask PIN + allow menu using *) - using options Psw for users (ask PIN + allow menu + wait for a marked user) My problem is that if a user chooses the "moderator" option, he can authenticate using any of the two PINs, and he can become an moderator for the conference by knowing only the user PIN… I think using two different phone numbers (one for users and one for moderators) is neither practical nor safe. Is there a way to authenticate users against only one of the password? For instance, math the password provided against only the moderator PIN, or only the user PIN. Thank you for your help, Silviu PS. Here is the dialplan : [ConfStart] exten => s,1,Answer exten => s,2,Set(TIMEOUT(response)=5) exten => s,3,Set(LANGUAGE()=conf) exten => s,4,Wait(1) exten => s,5,Background(welcome) ; "welcome, press * if you are a user of hold the line if you are a moderator" exten => *,1,MeetMe(|iMPsw|) ; for regular users exten => t,1,MeetMe(|aAiMPsX|) ; for moderators exten => i,1,GoTo(ConfStart,s,1) ___ --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] GET DATA and STREAM FILE comm ands, don´t work
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> [EMAIL PROTECTED]
> Sent: Monday, May 15, 2006 8:19 AM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] GET DATA and STREAM FILE commands,
> don´t work
>
...
>
> Now, below is my script in bash shell, scriptTest.bsh:
>
> #!/bin/bash
> #echo -e "Testing the working GET DATA and STREAM FILE\n" >&2
> echo -e "STREAM FILE demo-instruct \"\"\n"
> echo -e "GET DATA myprompt 4000 6\n"
> read getDigits
> echo -e "My Digits are: $getDigits\n" >&2 echo -e "HUNGUP\n"
The first thing asterisk does once it starts an AGI process is to send various
details about the caller to your script's STDIN. You need to read all that in
before issuing any commands.
Add the following to the top of your script:
declare -a array
while read -e ARG && [ "$ARG" ] ; do
array=(` echo $ARG | sed -e 's/://'`)
export ${array[0]}=${array[1]}
done
This will export various ENV variables containing the info asterisk is sending.
For more info look at the BASH resources on the Asterisk AGI page of
voip-info.org. (http://www.voip-info.org/wiki-Asterisk+AGI)
Josh McAllister
___
--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] MeetME Conferencing
Your welcome. It certainly could be done entirely in the dialplan using similar
logic, but this required a bit less mental horsepower. If your desire to avoid
AGI, is based on performance concerns, note that I have systems (Dell 2850
2xXEON 3.0) that terminate 8 PRIs and have had ALL channels loaded up with perl
AGI scripts and never skipped a beat. FWIW, these servers have 4G ram, and run
64bit RHES. Either way, glad I could get you closer to the end.
Josh McAllister
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Damon Estep
> Sent: Friday, May 12, 2006 7:06 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] MeetME Conferencing
>
> Josh,
>
> Thank you!
>
> I think the AGI could be bypassed by doing a realtime() to get the PIN
> from mySQL, also returning the variable that defines admin or user and
> jumping in the dialplan accordingly. Otherwise I would just end up having
> the AGI do the query because there is a need to store the users in the
> database to facilitate easy management.
>
> The admin menu and marked user options seem to be the key to making this
> work, so I will play with those.
>
>
>
>
>
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Josh McAllister
> Sent: Friday, May 12, 2006 2:08 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] MeetME Conferencing
>
> Ok, the script below (meetme.agi) will prompt for a valid pin up to 3
> times. If the pin matches one of the defined Admin pins, it will set the
> dialplan priority to 10 and exit, if User, sets to 20 and exits. Otherwise
> Hangs up.
>
> In the case of admin, these MeetMe options are used:
> a - Admin mode
> A - Marked mode
> c - Announce number of participants (optional of course)
> s - Present Admin menu by pressing '*'
> x - close conf when last marked user leaves.
>
> In the case of user:
> c s x are used as above, but we add:
> w - wait until marked user enters. (Plays MoH until then)
>
> The dialplan assumes you have a static pinless conference setup as conf
> #10.
>
> extensions.conf:
> exten => 5552323,1,Wait(1)
> exten => 5552323,2,Answer()
> exten => 5552323,3,AGI(meetme.agi)
> exten => 5552323,4,NoOp(Invalid Pin)
> exten => 5552323,5,Hangup()
>
> exten => 5552323,10,NoOp(Admin Pin)
> exten => 5552323,11,MeetMe(10,aAcsx)
> exten => 5552323,12,Hangup()
>
> exten => 5552323,20,NoOp(User Pin)
> exten => 5552323,21,MeetMe(10,cswx)
> exten => 5552323,22,Hangup()
>
>
>
> The script of course requires the Asterisk::AGI module.
>
> meetme.agi:
>
> #!/usr/bin/perl
> use Asterisk::AGI;
> my $AGI = new Asterisk::AGI;
> my $input = { %{$AGI->ReadParse()} };
>
> #our $DEBUG = 1;
>
> my @UserPins = ('1','2');
> my @AdminPins = ('9','8');
>
> my $mode = collectPin($AGI,5);
>
> $AGI->verbose("collectPin got '$mode'") if $DEBUG;
>
> if ($mode eq 'Admin') {
>$AGI->set_priority(10);
> } elsif ($mode eq 'User') {
>$AGI->set_priority(20);
> } else {
>$AGI->stream_file("goodbye",'""');
>$AGI->hangup;
> }
>
> exit;
>
> sub collectPin {
>my $AGI = shift;
>my $maxdigits = shift;
>
>my $tries = 0;
>
>#Three tries to select an existing pin.
>while ($tries < 3) {
> $AGI->stream_file("please-try-again",'""') if $tries > 0;
> $tries++;
> my $pin = $AGI->get_data('enter-conf-pin-number', "1",
> $maxdigits);
> $AGI->verbose("Got PIN $pin.") if $DEBUG;
> next unless $pin > 0;
>
> if ( grep(/^$pin$/, @AdminPins) ) {
> $AGI->stream_file("pin-number-accepted",'""');
> return 'Admin';
> } elsif ( grep(/^$pin$/, @UserPins) ) {
> $AGI->stream_file("pin-number-accepted",'""');
> return 'User';
> } else {
> $AGI->stream_file("conf-invalidpin",'""');
> }
>}
>
>return undef;
> }
>
>
> What can I say, I was bored.
>
> Enjoy,
>
> Josh McAllister
>
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Damon Estep
> Sent: T
RE: [Asterisk-Users] MeetME Conferencing
Ok, the script below (meetme.agi) will prompt for a valid pin up to 3 times. If
the pin matches one of the defined Admin pins, it will set the dialplan
priority to 10 and exit, if User, sets to 20 and exits. Otherwise Hangs up.
In the case of admin, these MeetMe options are used:
a - Admin mode
A - Marked mode
c - Announce number of participants (optional of course)
s - Present Admin menu by pressing '*'
x - close conf when last marked user leaves.
In the case of user:
c s x are used as above, but we add:
w - wait until marked user enters. (Plays MoH until then)
The dialplan assumes you have a static pinless conference setup as conf #10.
extensions.conf:
exten => 5552323,1,Wait(1)
exten => 5552323,2,Answer()
exten => 5552323,3,AGI(meetme.agi)
exten => 5552323,4,NoOp(Invalid Pin)
exten => 5552323,5,Hangup()
exten => 5552323,10,NoOp(Admin Pin)
exten => 5552323,11,MeetMe(10,aAcsx)
exten => 5552323,12,Hangup()
exten => 5552323,20,NoOp(User Pin)
exten => 5552323,21,MeetMe(10,cswx)
exten => 5552323,22,Hangup()
The script of course requires the Asterisk::AGI module.
meetme.agi:
#!/usr/bin/perl
use Asterisk::AGI;
my $AGI = new Asterisk::AGI;
my $input = { %{$AGI->ReadParse()} };
#our $DEBUG = 1;
my @UserPins = ('1','2');
my @AdminPins = ('9','8');
my $mode = collectPin($AGI,5);
$AGI->verbose("collectPin got '$mode'") if $DEBUG;
if ($mode eq 'Admin') {
$AGI->set_priority(10);
} elsif ($mode eq 'User') {
$AGI->set_priority(20);
} else {
$AGI->stream_file("goodbye",'""');
$AGI->hangup;
}
exit;
sub collectPin {
my $AGI = shift;
my $maxdigits = shift;
my $tries = 0;
#Three tries to select an existing pin.
while ($tries < 3) {
$AGI->stream_file("please-try-again",'""') if $tries > 0;
$tries++;
my $pin = $AGI->get_data('enter-conf-pin-number', "1", $maxdigits);
$AGI->verbose("Got PIN $pin.") if $DEBUG;
next unless $pin > 0;
if ( grep(/^$pin$/, @AdminPins) ) {
$AGI->stream_file("pin-number-accepted",'""');
return 'Admin';
} elsif ( grep(/^$pin$/, @UserPins) ) {
$AGI->stream_file("pin-number-accepted",'""');
return 'User';
} else {
$AGI->stream_file("conf-invalidpin",'""');
}
}
return undef;
}
What can I say, I was bored.
Enjoy,
Josh McAllister
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Damon Estep
Sent: Thursday, May 11, 2006 10:37 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] MeetME Conferencing
Static configs for the conference rooms are not an issue.
The main goal is to allow the moderator to determine when the conference
“starts” by having all participants hearing MOH until the moderator starts the
interactive call with a PIN known only to the moderator, and then allowing the
moderator (and only the moderator) to kick out all users from the keypad when
the call is over.
An additional benefit would be gained if authenticate() or realtime() app
commands could be used against a mysql database for the participant and
moderator pins so an app could be written easily to allow changing of the PINS
in the database.
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dan Austin
Sent: Thursday, May 11, 2006 10:29 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] MeetME Conferencing
I believe you can accomplish this with a well crafted dialplan.
If you did not have the restriction against out of tree modules, I would
recommend an app that strores the conference details in a database
and would allow just this kind of control.
Dan
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Damon Estep
Sent: Thursday, May 11, 2006 4:30 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] MeetME Conferencing
Not opposed to paying someone that can do it right ☺
As far as “coding” goes, you mean create the dialplan entries, not modify the
meetme source, correct?
Our application requires that this can be done in 1.2 release, not trunk and
not with an add-in that is not part of 1.2
If you have done it and would like to charge for you knowledge PM me, if you
are willing to post a sample free of charge do it here for the benefit of all.
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dean Collins
Sent: Thursday, May 11, 2006 5:18 P
RE: [Asterisk-Users] MySQL replication for voicemail
> > Hi Josh - > > > Another approach you may want to consider for data redundancy that > > does not rely on MySQL's finicky replication stuff is DRBD. Think of it > > as RAID-1 across Ethernet. I have used it in production on some VERY > > busy (> 1200 qps) MySQL servers for a couple years with no problems. > > I would very much welcome not having to use a MySQL's replication > (sorry to all you MySQL geeks)! I've read up a little on DRBD before, > but I've avoided it in this particular application because of a few > questions I had about it: > > 1) Does it work over slow-ish links? In this case, I'm going over WAN > links to offices around the country. This would be a function of the amount of data being changed, and the speed of your link. DRBD does support ASYNC mode, and there is a good chance this could work for you. > 2) Can it do two-way replication? No. This is primarily because of the limitations of the overlying filesystem. (DRBD sits below the FS). With newer fs's like GFS, this is now feasible, but DRBD is a little behind. Keep your eyes open for 0.8x to be released as it will support 2-way with GFS. If you want to maximize hardware resources, you can of course setup a partition replicating A -> B, and another replicating B -> A. > 3) Can it do N-way replication (i.e. multiple slaves)? We have > several offices offices that I'd like to have the replicated info. I know this was something that was in the works a while back. I haven't followed this issue that closely though, so I'm not sure if this has been worked out or not. I believe the proposed approach was to simply get rid of some arbitrary limitations DRBD imposed on letting you layer drbd on a drbd device. > > Thanks! > Noah > ___ > --Bandwidth and Colocation provided by Easynews.com -- To summarize, if you are currently doing MySQL replication with multiple masters, and multiple+N slaves, and you're trying to duplicate that... DRBD is probably not going to work for you. But I can say that what it does it does very well, and with little overhead. I've seen < 5% reduction in write performance with 15K SCSI drives, IE ~ 70MB/sec writes with DRBD vs ~72MB/sec without. ___ --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] MySQL replication for voicemail
Another approach you may want to consider for data redundancy that does not rely on MySQL's finicky replication stuff is DRBD. Think of it as RAID-1 across Ethernet. I have used it in production on some VERY busy (> 1200 qps) MySQL servers for a couple years with no problems. Another nice side effect is a relatively simple means of doing a total point in time "cold" backup of all you DB's without regard to the db engine(s) your using and minimal down time. If you can afford ~ 1 minute/day of downtime... stop MySQL, break DRBD replication, restart mysql, mount data on the rep. slave, backup data files, unmount, restart replication. DRBD tracks the deltas so it will catch up quickly, and that entire process can be automated. Anyways, if you're interested, check our http://drbd.org. Please make sure you fully understand what it's doing before using it in production though. A lack of understanding can lead to nasty things like replicating the wrong way. Note, that this can be used as a very simple means of providing warm standby * servers as well. Coupled with something like mon, you can provide for automatic failover as well. Josh McAllister -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Noah Miller Sent: Monday, May 08, 2006 10:17 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] MySQL replication for voicemail Hi Gary - > I have voicemessage ODBC storage working and MySQL replicating, > but I didn't setup the replication after-the-fact, therefore didn't need the > 'LOAD DATA FROM MASTER'. You may want to try a different method > (more manual) of synching your machines. Yes, I've been reading about some errors with LOAD DATA FROM MASTER that might not get fixed until after 5.1 is released. Yikes! I'll take your advice and move a mysqldump of the info over to the slaves, and manually set the replication points. > Does your voicemessages table contain specially large messages? > The errors you get don't suggest this, but the only change I had to make > was to increase the maximum packet size -- it was 1M by default and I > changed it to 16M. Aha. I didn't even know there was such a setting. I'm sure there will come a time when someone leaves a message larger than 1M. Thanks! Noah > On 5/8/06, Noah Miller <[EMAIL PROTECTED]> wrote: > > > > Does anybody have any ideas on what's causing this error? Why would > > MySQL not have enough memory? What does it mean, "points outside data > > file"? Is anybody else doing this successfully, or am I a lone freak? > > > ___ > --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 ___ --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] Can I recreate a Fax from a recorded file?
Sounds like a potential business opportunity. Someone could setup a fax proxy service that provides this sort of digital signing / archiving. The originator could simply dial a toll-free access number, receive a 2nd dialtone and then dial the destination. Meanwhile the proxy is recording the call, then decoding and allowing the archives to be viewed online along with all relevent call details. Hmm... Interesting. Josh McAllister -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Colin Anderson Sent: Thursday, May 04, 2006 1:12 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Can I recreate a Fax from a recorded file? >Why is this hard to fake at all? You send a different fax to your >system, and replace the Asterisk audio file with the one from the >altered fax. Additionally, the client has no realistic way of >verifying the correctness of your audio-to-fax translation tool; it >could just as easily output a TIFF file completely different from the >one that was actually faxed. That's interesting, I hadn't thought of it that way. I was thinking in terms of subtly modifying the original audio stream not outright replacing the recording and faking the datestamp! Given that, essentially recording the audio is the *same* as retaining the TIFF in terms of integrity vulnerability. How about this: (theoretical of course) 1. Fax comes in 2. Audio is recorded 3. A checksum of the audio is generated then relayed somehow to a seperate, secure system 4. In the event of a dispute, the checksum is retrieved, compared with the original audio file, then the original audio is "replayed" and the fax is regenerated. The 3. part I leave as an exercise for the reader. ___ --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] /var/spool/asterisk/outgoing/ prematurely hangingup
Just a shot in the dark... but have you tried Answer() before Playback()? Josh McAllister -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tom Engleward Sent: Monday, May 01, 2006 11:43 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] /var/spool/asterisk/outgoing/ prematurely hangingup I have a PSTN termination provider "foo" which will accept standard U.S. calls in the form 1<10 digit ph#>. I have an outbound route named "foo", with dial pattern "5|.", with the only entry in trunk sequence being "IAX2/foo". I have an X-lite local extension, on which I can dial 51<10 digit ph#>, and asterisk will call out over foo and the phone at <10 digit ph#> will ring. This rules out a lot of possible problems. extensions.conf includes this: [outgoingtest] exten => s,1,Playback(custom/testmsg) exten => s,2,Wait(1) exten => s,3,Hangup And yes, asterisk has been restarted since the last time any config files were modified. I have a test message at /var/lib/asterisk/sounds/custom/testmsg.gsm If I make the file "1.call" containing: Channel: IAX2/foo MaxRetries: 1 RetryTime: 5 WaitTime: 10 Context: outgoingtest Extension: 1<10 digit ph#> Priority: 1 and copy it to /var/spool/asterisk/outgoing/ then the phone doesn't ring, but this shows up on the asterisk console: -- Attempting call on IAX2/foo for 1<10 digit ph#>@outgoingtest:1 (Retry 1) -- Hungup 'IAX2/foo-7' -- Attempting call on IAX2/foo for 1<10 digit ph#>@outgoingtest:1 (Retry 2) -- Hungup 'IAX2/foo-8' The "foo-7" and "foo-8" on the console are different (numbers anywhere from 1 to 9) every time I try copying the file to outgoing. I tried using extension 51<10 digit ph#> instead of 1<10 digit ph#> in 1.call, but that didn't work either. Why is it failing? __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.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 ___ --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] CallerID Name problem
>From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lacy Moore - >Aspendora >Sent: Monday, May 01, 2006 3:06 PM >To: Asterisk Users Mailing List - Non-Commercial Discussion >Subject: Re: [Asterisk-Users] CallerID Name problem > >Do you get caller ID number? If so, WAITing is not going to help, since you >already get the info. If you >get caller ID number, then your telco is not >>sending the name. This is not necessarily true. I've always gotten cID number, but only recently when I added a wait(1) did I start getting channel vars populated with cID Name. Same as Eric, I was getting cID Name in the CDR records all along as well. Eric -- Go ahead and give it a shot... even if you are getting the cID number. This will likely fix your problem. Josh McAllister ___ --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
You keep eluding to the answer yourself. Asterisk Manager is the way to go. Check out http://search.cpan.org/~xantus/POE-Component-Client-Asterisk-Manager/. Relatively simple event based method for using Asterisk manager. What I would do is register a handler to track new calls, and calls ending. Every time you get a new call, add it to a hash with the customer_id as the key. Seperately register a callback that keeps re-calling itself at X second intervals. It would cycle through the hash of active calls decrementing remaining time for each, and then kick anyone with < 1 second remaining. I have a single script running 12 instances of POE::Component::Client::Asterisk::Manager (1 for each of 12 servers) under a single POE kernel to track > 2500 channels (comings and goings of MeetMe users) and it's had no problem keeping up. Just make sure that you avoid any long running loops as POE is not multi-threaded. For something like this, I think you'll find 1 instance of a single script much easier to track and debug than a whole bunch of instance of an AGI script. Josh McAllister -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jon Farmer Sent: Wednesday, April 26, 2006 7:27 AM To: [EMAIL PROTECTED] Cc: asterisk-users@lists.digium.com Subject: 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 ___ --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] Anyone played with app_amd?
I'm guessing this may be a question for dev list, but wanted to try my luck here first. I'm trying to compile app_amd (Answering Machine Detection) against 1.2.7.1 and am getting some errors. I should point out that I simply snarfed app_amd.c from http://svn.digium.com/view/asterisk/trunk/apps/app_amd.c?rev=14714 ...so if there are other includes and such that are required, that would likely be the problem, but I was hoping someone may have a workaround for compiling amd in the current release as I can not run developmental branches. Here are the errors I get: gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g3 -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=k8 -DZAPTEL_OPTIMIZATIONS -m64 -fomit-frame-pointer -fPIC -c -o app_amd.o app_amd.c app_amd.c: In function `amd_exec': app_amd.c:321: dereferencing pointer to incomplete type app_amd.c:321: dereferencing pointer to incomplete type app_amd.c:321: dereferencing pointer to incomplete type app_amd.c:323: dereferencing pointer to incomplete type app_amd.c:323: dereferencing pointer to incomplete type app_amd.c:323: dereferencing pointer to incomplete type app_amd.c:323: dereferencing pointer to incomplete type app_amd.c: In function `unload_module': app_amd.c:384: dereferencing pointer to incomplete type app_amd.c:384: dereferencing pointer to incomplete type make[1]: *** [app_amd.o] Error 1 make[1]: Leaving directory `/root/ast/ast-latest/asterisk-1.2.7.1/apps' make: *** [subdirs] Error 1 Thanks, Josh McAllister ___ --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 Forward and AGI
I'm sure there is more than 1 way to do this, but the first thing that comes to my mind is to set a channel variable with the exten # at the top of your extensions macro. Then use that channel var instead of CLID. Josh McAllister -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jon Farmer Sent: Wednesday, April 12, 2006 1:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [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 ___ --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] Anyone have a definitive list of Manager events per category?
Can anyone provide a complete list of events and to which category they are in? (ie. system,call,log,verbose,command,agent,user). I'm using * Manager in various ways with heavy call volume and would like to limit the events per connection as much as possible. Any help would be appreciated. Thanks, Josh McAllister ___ --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] Anyone have a definitive list of Manager eventsper category?
Title: [Asterisk-Users] Anyone have a definitive list of Manager eventsper category? My understanding is that is exactly what these categories do for you…. IE. If I were to create a user with read=call, that user would only get events in the call category. Am I wrong? If my assumption is correct, it would be of great benefit to know exactly which events are in which category. Josh From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wai Wu Sent: Tuesday, April 04, 2006 1:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Anyone have a definitive list of Manager eventsper category? I don't think you can selectively receive events. I am also write an app using heavy manager actions, and I put the filters on my app. So far, I have not seen traffic from these events do a dent to my application/network performance. From: [EMAIL PROTECTED] on behalf of Josh McAllister Sent: Tue 4/4/2006 2:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Anyone have a definitive list of Manager eventsper category? Can anyone provide a complete list of events and to which category they are in? (ie. system,call,log,verbose,command,agent,user). I'm using * Manager in various ways with heavy call volume and would like to limit the events per connection as much as possible. Any help would be appreciated. Thanks, Josh McAllister ___ --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] Soekris products available in the US?
Soekris is headquartered in Santa Cruz, CA. Buy direct from their website: http://www.soekris.com Josh McAllister > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Philip Trauring > Sent: Tuesday, March 29, 2005 6:43 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] Soekris products available in the US? > > Anyone know if Soekris products are available in the US? > > Thanks, > > Philip > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
