Re: [asterisk-users] How to Clone Asterisk
On 2/2/07, Robert DeVries [EMAIL PROTECTED] wrote: I want to essentially transplant my existing Asterisk server to a new machine, and take the old sever out of service. Assuming I install Asterisk on the new machine, does anyone know what files I would have to copy over? What comes to mind are the *.conf files in /etc/asterisk, as well as the voicemail audio files. Anything else? Sometimes my installations goes to different directories. You should check first where are your files and what you make more (voicemail, monitor, etc) Conf: /etc/asterisk /etc/zaptel.conf Sounds: /usr/share/asterisk/sounds/ /var/lib/asterisk/sounds/ MOH: /usr/share/asterisk/mohmp3/ Logs: /var/log/asterisk/ AGIs: /var/lib/asterisk/agi-bin Database: /var/lib/postgresql -- Ralph Liebessohn ICQ: 74835911 Skype: liebessohn ___ --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] Fax from PAP2 through a zap channel to PSTN
On 2/1/07, Chung-lai Chan [EMAIL PROTECTED] wrote: Hello all, Can I send fax from PAP2 through a zap channel to PSTN? I have tried but it is not successful. Thank you for your help! Lai Try to remove echo cancellation (any type of cancellation) and VAD. I got good answer receiving fax as sip client behind a PAP2. -- Ralph Liebessohn ICQ: 74835911 Skype: liebessohn ___ --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: FW: [asterisk-users] Get dialed numbers in AGI
On 1/11/07, Mike D'Ambrogia [EMAIL PROTECTED] wrote: Ralph Kind of new to asterisk, and really new to AGI but it looks like you were trying to have the AGI script tell asterisk to read and lay the results into my_var and then regain control in the AGI script, is that correct? If so I don't think that will work since the dialplan variables are only exposed/visible when you start the AGI script, since you are still within the AGI script you'll probably never see my_var if called this way md Mike, that's my first agi. Easy to see, ahm? What I'm thinking to do is, play a sound, get the numbers dialed, see on a db what to do using the dialed numbers, get back to dialplan (go to a queue or something else). The way started to work is that you said, send dialed as parameter to agi and get it with $argv[1]. And I was trying to get it from stdin. Things I do not understand well and the documentations I've read are not so clear are the way agi work with some things like: - How can I fix this agi to work? AGI Rx exec read my_var|sound-file|5|||15 -- AGI Script Executing Application: (read) Options: (my_var|sound-file|5|||15) -- Accepting a maximum of 5 digits. -- Playing 'sound-file' (language 'en') -- User entered '85214' AGI Tx 200 result=0 AGI Rx get variable my_var AGI Tx 200 result=1 (85214) AGI Rx exec saydigits Resource id #1 // (this is the result of my_var) All the variables here was my_var, it worked for GET VARIABLE but didn't for SAYDIGITS and odbc connection. How can I SAYDIGITS of my_var or insert my_var value into a db? - What I need more to use WAIT FOR DIGIT? Because it didn't stop to wait for digits. - STDIN shoudn't get the result of READ or GET VARIABLE? Where these values go? -- Ralph Liebessohn ICQ: 74835911 Skype: liebessohn ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Get dialed numbers in AGI
On 1/11/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:
Steve Edwards wrote:
On Thu, 11 Jan 2007, Yuan LIU wrote:
AGI doesn't see the name var; all it sees is an array @ARGV (or
whatever in the respective language). As the documentation says,
values are passed like command line arguments.
But, in the interest of more maintainable code, you can pass the
option name and use getopt_long() in C or whatever language you use.
For example,
agi(block-ani,--ani=5,--dnis=8005551212,--debug,--verbose)
vs
agi(block-ani|5|8005551212|y|y)
AGI() does not support more than 1 parameter to AGIs.
# cat extensions.conf
exten = 8899,1,Read(my_var|sound-file|5|||15)
exten =
8899,2,AGI(/usr/share/asterisk/bin/firstagi.php|${my_var}|123|321|111|222)
# cat firstagi.php
$my_var=$argv[1];
fwrite(STDOUT,exec sayalpha $my_var \n);
fflush(STDOUT);
$my_var=$argv[2];
fwrite(STDOUT,exec sayalpha $my_var \n);
fflush(STDOUT);
$my_var=$argv[3];
fwrite(STDOUT,exec sayalpha $my_var \n);
fflush(STDOUT);
$my_var=$argv[4];
fwrite(STDOUT,exec sayalpha $my_var \n);
fflush(STDOUT);
$my_var=$argv[5];
fwrite(STDOUT,exec sayalpha $my_var \n);
fflush(STDOUT);
Results in console:
-- AGI Script Executing Application: (sayalpha) Options: (98765) // Result
of READ
-- AGI Script Executing Application: (sayalpha) Options: (123) // Other
parameters
-- AGI Script Executing Application: (sayalpha) Options: (321)
-- AGI Script Executing Application: (sayalpha) Options: (111)
-- AGI Script Executing Application: (sayalpha) Options: (222)
AGI receives more than 1 parameter.
--
Ralph Liebessohn
ICQ: 74835911
Skype: liebessohn
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Re: [asterisk-users] Get dialed numbers in AGI
On 1/11/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: What version of Asterisk? Perhaps it changed since the last time I had to deal with the issue. Perhaps it was fixed for 1.0, or maybe it was specific to asterisk-perl. Mike D'Ambrogia wrote: Not true for the php version, it will take multiple params into argv[0], argv[1], argv[2], etc Eric, I tried it on asterisk 1.2.13 and it worked with multiple params. -- Ralph Liebessohn ICQ: 74835911 Skype: liebessohn ___ --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 Compilation and Installation
On 1/11/07, bilal ghayyad [EMAIL PROTECTED] wrote: Hi List; I understand that I have to compile zaptel but what about asterisk? Is it enough to extract it? Well, how I will run asterisk (without compilation and installation)? Any advise? Regards Bilal Bilal, which distro you use? Using debian you could only # apt-get install asterisk and it will work. If you need I can send you a tutorial/script to install * on debian with cdr in postgres. -- Ralph Liebessohn ICQ: 74835911 Skype: liebessohn ___ --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 dialed numbers in AGI
On 1/10/07, Yuan LIU [EMAIL PROTECTED] wrote:
From: Ralph Liebessohn [EMAIL PROTECTED]
I did a quick test and it seems that everything passed to AGI is by value,
and there is no apparent relationship between variable named used in two
different AGI commands.
However, a small adaption of dial plan could accomplish what you wanted,
that is, to read the variable in dial plan, then pass its value to AGI.
Hope this helps.
Yuan Liu
Hello people,
next step.
With many other tests I may conclude that AGI is not saving my password, it
is giving it to asterisk temporaly and the next step executed by asterisk
doesn 't know the variable.
If I run SAYDIGITS after READ inside extensions it works fine. What I cannot
say running it on AGI.
The architecture of Yuan or Anton can work, but how pass the value from
dialplan to AGI or wich AGI librarie give me the function of READ in
dialplan?
I tried to pass value from dialplan using:
AGI(myagi.php|${var})
But AGI didn't see $var value. And using GET/SET VARIABLE and STREAM FILE
into AGI is passing through the commands, nothing is being done or waiting
to dial digits.
Is that the correct way to go?
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Skype: liebessohn
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Re: [asterisk-users] Get dialed numbers in AGI
On 1/11/07, Mike D'Ambrogia [EMAIL PROTECTED] wrote:
on the php AGI side the ${var} parameter passed in from your dialplan
will be exposed in PHP's $argv[1] array element
fwrite(STDOUT,exec saydigits $argv[1]\n);
as a side note $argv[0] contains the full path including filename to the
script
Hope this helps
Mike
Mike,
it didn't help.
I just SOLVED the problem! You're a genius.
Now I can get information from dialplan.
Do you know why the other ways didn't work?
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Skype: liebessohn
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Re: [asterisk-users] Get dialed numbers in AGI
On 1/11/07, Ralph Liebessohn [EMAIL PROTECTED] wrote: Mike, it didn't help. I just SOLVED the problem! You're a genius. Now I can get information from dialplan. Do you know why the other ways didn't work? -- Ralph Liebessohn ICQ: 74835911 Skype: liebessohn Errata. IT just SOLVED.. Not I just solved. -- Ralph Liebessohn ICQ: 74835911 Skype: liebessohn ___ --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] Get dialed numbers in AGI
Hi,
I'm trying to write a AGI in PHP to get the numbers dialed (with read()),
save it into a variable to insert it into a SQL server database. But I
cannot see results into the variable, it always return NULL.
Here is a piece of the AGI.
fwrite(STDOUT,exec Read my_var|/sound_to_play|5|||15 \n);
fflush(STDOUT);
$conn=odbc_connect('MSSQL', 'USER', 'PASS');
$query = odbc_exec($conn, INSERT INTO dialed(number) VALUES('$my_var'));
Even if I only show my_var value or try to use it inside asterisk, the value
is NULL.
There is another way to do it? Am I doing a mistake here?
I'm using Asterisk 1.2.13.
Thank you all.
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Skype: liebessohn
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Re: [asterisk-users] Get dialed numbers in AGI
On 1/10/07, Yuan LIU [EMAIL PROTECTED] wrote:
Then there must be an error somewhere. The variable READ() in Asterisk
should be usable. Should be able to use SayDigits() to play it back - or
no
value is read.
Yuan Liu
Hi Yuan and Anton,
Let's put here all AGI for test:
#!/usr/bin/php -q
?php
ob_implicit_flush(false);
error_reporting(0);
$stdin = fopen( 'php://stdin', 'r' );
if (!defined('STDIN'))
{
define('STDIN',fopen('php://stdin','r'));
}
if (!defined('STDOUT'))
{
define('STDOUT',fopen('php://stdout','r'));
}
if (!defined('STDERR'))
{
define('STERR',fopen('php://stderr','r'));
}
while(!feof($stdin))
{
$temp=trim(fgets(STDIN,4096));
if (($temp==) || ($temp=\n))
{
break;
}
$s=split(:,$temp);
$nome=str_subst(agi_,,$s[0]);
$agi[$nome]=trim($s[1]);
}
foreach($agi as $chave=$valor)
{
fwrite(STDERR,--$chave=$valor\n);
fflush(STDERR);
}
$my_var=123;
fflush(STDERR);
fwrite(STDERR,Just testing\\\n);
fflush(STDERR);
fwrite(STDOUT,exec read
my_var|//usr/share/asterisk/sounds/please-wait-connect-oncall-eng|5|||15
\n);
fwrite(STDOUT,exec saydigits ${my_var} \n);
fflush(STDOUT);
$conn=odbc_connect('MSSQL', 'asterisk', '123456');
$query = odbc_exec($conn, INSERT INTO usuario(nome) VALUES('$my_var'));
?
If I not startup $my_var=123; Saydigits receives a NULL as options. And so
nothing was inserted into db.
I tried to use WAIT FOR DIGIT but it makes no sense, asterisk passed through
it directly like Joel Lansden Joel AT digitalparadise DOT net reported on
9/14/06.
Is there another function or way to test it or I must try in another
asterisk box?
--
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ICQ: 74835911
Skype: liebessohn
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Re: [asterisk-users] Get dialed numbers in AGI
On 1/10/07, Lee Jenkins [EMAIL PROTECTED] wrote:
Ralph Liebessohn wrote:
Hi,
I'm trying to write a AGI in PHP to get the numbers dialed (with
read()), save it into a variable to insert it into a SQL server
database. But I cannot see results into the variable, it always return
NULL.
Here is a piece of the AGI.
fwrite(STDOUT,exec Read my_var|/sound_to_play|5|||15 \n);
fflush(STDOUT);
$conn=odbc_connect('MSSQL', 'USER', 'PASS');
$query = odbc_exec($conn, INSERT INTO dialed(number)
VALUES('$my_var'));
Even if I only show my_var value or try to use it inside asterisk, the
value is NULL.
There is another way to do it? Am I doing a mistake here?
I'm using Asterisk 1.2.13.
I'm not a php guy, but aren't we missing the part that retrieves the
value saved into my_var from the call to READ?
// In this part you run the read command and asterisk
// stores the value into the channel variable my_var
fwrite(STDOUT,exec Read my_var|/sound_to_play|5|||15 \n);
// In this part you are constructing your sql statement
// with a null value cause you didn't make a call to
// GET VARIABLE before constructing your sql.
$query = odbc_exec($conn, INSERT INTO dialed(number) VALUES('$my_var'));
--
Warm Regards,
Lee
Hi Lee,
thanks for the tip. I tried other methods trying to get the variable value,
but no success.
Doing a GET VARIABLE my_var after READ the get variable returns the value
I dialed, but doesn't give the exact value to it. I got Resource ID #1
instead.
Using:
fwrite(STDOUT,exec read
my_var|//usr/share/asterisk/sounds/please-wait-connect-oncall-eng|5|||15
\n);
fwrite(STDOUT,get variable my_var \n);
fflush(STDOUT);
$my_var=STDIN;
fwrite(STDOUT,exec saydigits $my_var \n);
I got it:
AGI Rx exec read
my_var|//usr/share/asterisk/sounds/please-wait-connect-oncall-eng|5|||15
-- AGI Script Executing Application: (read) Options:
(my_var|//usr/share/asterisk/sounds/please-wait-connect-oncall-eng|5|||15)
-- Accepting a maximum of 5 digits.
-- Playing '//usr/share/asterisk/sounds/please-wait-connect-oncall-eng'
(language 'en')
-- User entered '85214'
AGI Tx 200 result=0
AGI Rx get variable my_var
AGI Tx 200 result=1 (85214)
AGI Rx exec saydigits Resource id #1
-- AGI Script Executing Application: (saydigits) Options: (Resource)
AGI Tx 200 result=0
AGI Rx exec Resource id #1
-- AGI Script Executing Application: (Resource) Options: (id)
Jan 10 17:31:33 WARNING[4867]: res_agi.c:1147 handle_exec: Could not find
application (Resource)
AGI Tx 200 result=-2
I also tried:
$my_var=fwrite(STDOUT,get variable my_var \n);
But always I get 21 as value.
More tries?
--
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ICQ: 74835911
Skype: liebessohn
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Re: [asterisk-users] g729 registered
On 11/20/06, Ralph Liebessohn [EMAIL PROTECTED] wrote: On 11/20/06, Alex Robar [EMAIL PROTECTED] wrote: Hi Ralph, Have you setup your PAP2 to allow the 729 codec? I believe you actually have to tell it that it's allowed to use that codec before it will work. Cheers, Alex On 11/20/06, Ralph Liebessohn [EMAIL PROTECTED] wrote: Hi guys, I've registered some g729 licenses, during register process everything worked fine. But I'm not able to use this codec. I'm trying to use a linksys PAP2 to talk using g729 but I got this answer from asterisk: Should asterisk translate to another codec when trying to make a new call with iax? Why can't asterisk make a call using g729 and sip? Some configuration. Thanks. -- Ralph Liebessohn ICQ: 74835911 Skype: liebessohn -- Alex Robar [EMAIL PROTECTED] Hi Alex, I set on Audio configuration to enable g729a, g729a as preferred codec and use only preferred codec. Is only that right? With ulaw all calls work fine. -- Ralph Liebessohn ICQ: 74835911 Skype: liebessohn Hi folks, just to finish this thread. I was trying to call from a pap2 to a pap2, nobody said me tha linksys pap2 can make only one call per time using g729. I tried recording the call to asterisk, another servers and another pap2 and that works fine. Thank you. -- Ralph Liebessohn ICQ: 74835911 Skype: liebessohn ___ --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 config chanspy
On 10/17/06, Thirumal Saminathan [EMAIL PROTECTED] wrote: hi all, please any one help me ,how to configure chanspy application . and also send me if u have any sample configure file. -thiruHi,It could be very simple, like:exten = 123,1,ChanSpy(); Spy all channelsor more accuracy:exten =124,1,ChanSpy(SIP); Spy all sip channels if I can help you more, let me know!-- Ralph LiebessohnICQ: 74835911Skype: liebessohn ___ --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] Is 1.2.12.1 production ready
On 10/17/06, Mike Clark [EMAIL PROTECTED] wrote: We have several sites in this configuration with no nightly reboots. Allsites except one are problem free. One site still has dropped calls.None of the sites crashes and some of them have been up for a few weeks. Tom Vile wrote: fine for me here since it came out.We are running 15 extension all day long. On 10/16/06, *shadowym* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I am getting ready to image a production system.Right now I am planning on using Centos 4.4, Asterisk 1.2.12.1 http://1.2.12.1, Freepbx 2.1.3.I will be using a Sangoma A200D card. I read of some people having problems with Asterisk 1.2.12.1 http://1.2.12.1 crashing.Is this across the board or is there anyone out there with no problems.If you have 24/7 uptime and no nightly reboot crons I would definitely appreciate hearingabout it. Cheers ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com http://www.baldwintechsolutions.com Phone: 518-631-2855 x205 Fax: 518-631-2856Hi guys,I'm having a problem with chanspy.When I'm hearing the calls on third or forth change asterisk gives me: Asterisk ended with exit status 139 Asterisk exited on signal 11. And restart.I'm using postgres for CDR, asterisk 1.2.12.1, addons-1.2.4 and zaptel-1.2.If would could test it, it will be very nice.-- Ralph LiebessohnICQ: 74835911 Skype: liebessohn ___ --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] Monitor Current outgoing calls
On 10/10/06, George Masgras [EMAIL PROTECTED] wrote: Hello all!I'm currently using Asterisk in conjunction with a2billing andeverything seems to be working great so far. Now, all I'm missing issome sort of a GUI to monitor all calls going out through my trunks. I can always do 'sip show channels' or 'sip debug' from the console butI was wondering if there's anything that basically does the same thingbut in a nicer, easier way. a2billing comes with some very nice CDR and invoice reporting. is there anything that can do pretty much thesame thing but for calls that are currently in progress ?Thanks guys GeorgeTry Flash Operator Panel (FOP). -- Ralph LiebessohnICQ: 74835911Skype: liebessohn ___ --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: Mediatrix 1204 trix
On 9/20/06, C F [EMAIL PROTECTED] wrote: Erik is this for a Mediatrix 1204? If so where did you get thesesettings? In SNMP? or HTTP?From the Mediatrix documentation:Page 59 (87) These are footnotes to whereever the words registerserver are mentioned in the Manual: 1. The Mediatrix 1204 does not use the Registrar server.2. The Mediatrix 1204 does not use the Registrar server.Here is an old post about this: http://lists.digium.com/pipermail/asterisk-users/2004-February/028568.htmlOn 9/20/06, Erik [EMAIL PROTECTED] wrote: gateway sip mysipproviderno transport tcp bind interface WAN router domain mysipdomainrealm sip.mydomain.nlauthentication myusername password mypassworddefault-server mysipproviderserver 5060 loose-router registration-lifetime 300registrar mysipproviderserver use-default-serveruser myusername works for me (note that this is a modified Patton setup, so you might have to tweak the language a bit.) rgds, Erik C F wrote: Erik, I have tried it and it did NOT work, can you tell me where to enter that info? Have done it and it worked? On 9/19/06, Erik [EMAIL PROTECTED] wrote: mediatrix DOES support SIP Register, just enter authentication details and a registar server C F wrote: Keep in mind that the Mediatrix does not support register (AFAIK, anyhow). You have to create a static entry in sip.conf that has host set to the IP address of the Mediatrix On 9/18/06, Bill Michaelson [EMAIL PROTECTED] wrote: Thank you, C F and Florian. Now I must expose my ignorance about SIP and Mediatrix... I've adapted my sip.conf to essentially conform with what you've posted. So when I restart the Asterisk server, ethereal indicates that a NOTIFY goes to the Mediatrix (at 192.168.20.188), which responds with a 481, resulting in this message: -- Got SIP response 481 Subscription does not exist back from 192.168.20.188 My guess is that I'm missing a piece of the puzzle on the Mediatrix side of the configuration. Similarly, I've configured the Mediatrix via snmpset commands such that: telephonyAttributesAutomaticCallEnable[*] = 1 and telephonyAttributesAutomaticCallTargetAddress[*] = my desired extension(s) When I call the Mediatrix from POTS, it sends INVITE to Asterisk with the appropriate extension, but Asterisk responds with 404. I think I'm missing something involving REGISTER, but I'm foggy... would somebody clear the haze, please? In my floundering, I tried putting this into sip.conf: register = [EMAIL PROTECTED]/441 But the Mediatrix was unimpressed, rebuffing my entreaty with a: 405 Method Not Allowed I don't take rejection well, and so I'm loathe to speak with the Mediatrix again. I really need someone wiser to advise me... Message: 15 Date: Sat, 16 Sep 2006 21:59:34 -0400 From: C F [EMAIL PROTECTED] Subject: Re: [asterisk-users] Mediatrix 1204 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1; format=flowed I have the same setup as Florian, however I have dtmfmode set to rfc instead of inband On 9/16/06, Florian Overkamp [EMAIL PROTECTED] wrote: Bill Michaelson wrote: Would anyone be kind enough to post a sip.conf fragment as a sample for use with a Mediatrix 1204?Ours works with: [mtrix1] type=peer host=172.28.4.46 mask= 255.255.255.255 context=in-mtrix1 qualify=no canreinvite=no dtmfmode=inband disallow=all allow=ulaw Best regards, FlorianHi guys,I have a Mediatrix 1204 working in this way.In SIP.CONF:[general]context=default in Mediatrix:SIP Configuration Proxy Host: MyAsteriskIPPort: MySipPortTelephony/Advanced:Automatic Call Activation: EnabledAutomatic Call Target: sip:[EMAIL PROTECTED]:PortThis way when I call a number connected to Mediatrix it makes a call to 1122 in context [default]. But I didn't set any configuration to user like peer or something else to use with it.-- Ralph LiebessohnICQ: 74835911Skype: liebessohn ___ --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 'Hosting'
On 8/16/06, Douglas Garstang [EMAIL PROTECTED] wrote: Has anyone ever tried to run multiple instances of Asterisk on a single system, running each with a different username, and each in a separate base directory? Something like /home/pbx/business-1, home/pbx/business-2 etc? Did it work? I assume for every service that Asterisk runs, on each instance, you'd have to use a different port numbers, which may get confusing. Each businesses phones would have to be configred with different SIP ports then too. What about processes? I notice that Asterisk runs about 26 processes (or are they threads?) for a single instance.Doug.You can put Asterisk to hear in the same default port, but you must use another IP address, theoretically. -- Ralph LiebessohnICQ: 74835911Skype: liebessohn ___ --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] Circuit/channel Congestion
On 8/11/06, Ralph Liebessohn [EMAIL PROTECTED] wrote: On 7/24/06, Thomas Laurids Pedersen [EMAIL PROTECTED] wrote: I have the same card, but in my zaptel.conf I have the following linespan=1,1,0,hdb3,crc4as you can see from the status your line is down.BR Thomas Lincoln Zuljewic Silva Hello all. I have a Digium TE110P board and when I do a 'dial 4509' I got:Jul 24 11:44:33 NOTICE[8180]: app_dial.c:1049 dial_exec_full: Unable to create channel of type 'Zap' (cause 34 - Circuit/channel congestion) My zaptel.conf:span=1,0,0,cas,hdb3,crc4bchan=1-15,17-31dchan=16My zapata.conf:[channels]context=demopriindication=outofbandpridialplan=localprilocaldialplan=localoverlapdial=yes immediate=nocallprogress=yesbusydetect=noswitchtype=euroisdnsignalling=pri_netgroup=1callgroup=1pickupgroup=1channel = 1-15,17-31My /proc/zaptel/1Span 1: WCT1/0 Digium Wildcard TE110P T1/E1 Card 0 HDB3//CRC4 1 WCT1/0/1 Clear (In use) 2 WCT1/0/2 Clear (In use) 3 WCT1/0/3 Clear (In use) 4 WCT1/0/4 Clear (In use) 5 WCT1/0/5 Clear (In use) 6 WCT1/0/6 Clear (In use) 7 WCT1/0/7 Clear (In use) 8 WCT1/0/8 Clear (In use) 9 WCT1/0/9 Clear (In use)10 WCT1/0/10 Clear (In use)11 WCT1/0/11 Clear (In use)12 WCT1/0/12 Clear (In use) 13 WCT1/0/13 Clear (In use)14 WCT1/0/14 Clear (In use)15 WCT1/0/15 Clear (In use)16 WCT1/0/16 HDLCFCS (In use)17 WCT1/0/17 Clear (In use)18 WCT1/0/18 Clear (In use) 19 WCT1/0/19 Clear (In use)20 WCT1/0/20 Clear (In use)21 WCT1/0/21 Clear (In use)22 WCT1/0/22 Clear (In use)23 WCT1/0/23 Clear (In use)24 WCT1/0/24 Clear (In use) 25 WCT1/0/25 Clear (In use)26 WCT1/0/26 Clear (In use)27 WCT1/0/27 Clear (In use)28 WCT1/0/28 Clear (In use)29 WCT1/0/29 Clear (In use)30 WCT1/0/30 Clear (In use) 31 WCT1/0/31 Clear (In use)My pri show span 1:Primary D-channel: 16Status: Provisioned, Down, ActiveSwitchtype: EuroISDNType: NetworkWindow Length: 0/7Sentrej: 0SolicitFbit: 0 Retrans: 0Busy: 0Overlap Dial: -1T200 Timer: 1000T203 Timer: 1T305 Timer: 3T308 Timer: 4000T313 Timer: 4000N200 Counter: 3My zap show channels: Chan ExtensionContext Language MusicOnHold pseudodemo1demo2demo3demo4demo5demo6demo7demo 8demo 9demo 10demo 11demo 12demo 13demo 14demo 15demo 17demo 18demo 19demo 20demo 21demo 22demo 23demo 24demo 25demo 26demo 27demo 28demo 29demo 30demo 31demoThanks a lot!LincolnHi Thomas and everybody else,I am with a problem near Lincoln's. When I dial through ZAP (TE406P) the first call complete and as soon as it complete it hangup and all the calls after do not complete cause:NOTICE[4981]: app_dial.c:1029 dial_exec_full: Unable to create channel of type 'ZAP' (cause 34 - Circuit/channel congestion) == Everyone is busy/congested at this time (1:0/1/0)But this only occurs on my new machine (P4 Dual core with Intel d101ggc motherboard), on my old machine (AMD Sempron with pcchips motherboard) it works with the same installation. Do you guys know something to solve it? Have you ever heard something similar?-- Ralph LiebessohnICQ: 74835911Skype: liebessohn Hi guys,someguys told me to take off the echo cancel module from digium card and test it again. Running the card without the echo cancel module everything worked fine.I will make a few more tests but it appears that the problem is really the echo module. Thanks by the attention.-- Ralph LiebessohnICQ: 74835911Skype: liebessohn ___ --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] Circuit/channel Congestion
On 7/24/06, Thomas Laurids Pedersen [EMAIL PROTECTED] wrote: I have the same card, but in my zaptel.conf I have the following linespan=1,1,0,hdb3,crc4as you can see from the status your line is down.BR Thomas Lincoln Zuljewic Silva Hello all. I have a Digium TE110P board and when I do a 'dial 4509' I got:Jul 24 11:44:33 NOTICE[8180]: app_dial.c:1049 dial_exec_full: Unable tocreate channel of type 'Zap' (cause 34 - Circuit/channel congestion) My zaptel.conf:span=1,0,0,cas,hdb3,crc4bchan=1-15,17-31dchan=16My zapata.conf:[channels]context=demopriindication=outofbandpridialplan=localprilocaldialplan=localoverlapdial=yes immediate=nocallprogress=yesbusydetect=noswitchtype=euroisdnsignalling=pri_netgroup=1callgroup=1pickupgroup=1channel = 1-15,17-31My /proc/zaptel/1Span 1: WCT1/0 Digium Wildcard TE110P T1/E1 Card 0 HDB3//CRC4 1 WCT1/0/1 Clear (In use) 2 WCT1/0/2 Clear (In use) 3 WCT1/0/3 Clear (In use) 4 WCT1/0/4 Clear (In use) 5 WCT1/0/5 Clear (In use) 6 WCT1/0/6 Clear (In use) 7 WCT1/0/7 Clear (In use) 8 WCT1/0/8 Clear (In use) 9 WCT1/0/9 Clear (In use)10 WCT1/0/10 Clear (In use)11 WCT1/0/11 Clear (In use)12 WCT1/0/12 Clear (In use) 13 WCT1/0/13 Clear (In use)14 WCT1/0/14 Clear (In use)15 WCT1/0/15 Clear (In use)16 WCT1/0/16 HDLCFCS (In use)17 WCT1/0/17 Clear (In use)18 WCT1/0/18 Clear (In use) 19 WCT1/0/19 Clear (In use)20 WCT1/0/20 Clear (In use)21 WCT1/0/21 Clear (In use)22 WCT1/0/22 Clear (In use)23 WCT1/0/23 Clear (In use)24 WCT1/0/24 Clear (In use) 25 WCT1/0/25 Clear (In use)26 WCT1/0/26 Clear (In use)27 WCT1/0/27 Clear (In use)28 WCT1/0/28 Clear (In use)29 WCT1/0/29 Clear (In use)30 WCT1/0/30 Clear (In use) 31 WCT1/0/31 Clear (In use)My pri show span 1:Primary D-channel: 16Status: Provisioned, Down, ActiveSwitchtype: EuroISDNType: NetworkWindow Length: 0/7Sentrej: 0SolicitFbit: 0 Retrans: 0Busy: 0Overlap Dial: -1T200 Timer: 1000T203 Timer: 1T305 Timer: 3T308 Timer: 4000T313 Timer: 4000N200 Counter: 3My zap show channels: Chan ExtensionContext Language MusicOnHold pseudodemo1demo2demo3demo4demo5demo6demo7demo8demo 9demo 10demo 11demo 12demo 13demo 14demo 15demo 17demo 18demo 19demo 20demo 21demo 22demo 23demo 24demo 25demo 26demo 27demo 28demo 29demo 30demo 31demoThanks a lot!LincolnHi Thomas and everybody else,I am with a problem near Lincoln's. When I dial through ZAP (TE406P) the first call complete and as soon as it complete it hangup and all the calls after do not complete cause:NOTICE[4981]: app_dial.c:1029 dial_exec_full: Unable to create channel of type 'ZAP' (cause 34 - Circuit/channel congestion) == Everyone is busy/congested at this time (1:0/1/0)But this only occurs on my new machine (P4 Dual core with Intel d101ggc motherboard), on my old machine (AMD Sempron with pcchips motherboard) it works with the same installation. Do you guys know something to solve it? Have you ever heard something similar?-- Ralph LiebessohnICQ: 74835911Skype: liebessohn ___ --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] wildcard always busy
Hi guys,I am fighting to get a Wildcard TE405P working but it always start and put all channels in use. 14 TE4/0/1/14 Clear (In use) 15 TE4/0/1/15 Clear (In use) 16 TE4/0/1/16 HDLCFCS (In use) 17 TE4/0/1/17 Clear (In use)I've tried to downgrade zaptel and asterisk but it didn't solve the problem.Here is my zaptel.conf:span=1,0,1,ccs,hdb3,crc4,yellow #span=2,0,1,ccs,hdb3,crc4,yellow#span=3,0,1,ccs,hdb3,crc4,yellow#span=4,0,1,ccs,hdb3,crc4,yellowbchan=1-15,17-31dchan=16#bchan=32-46,48-62#dchan=47#bchan=63-77,79-93#dchan=78#bchan=94-108,110-124 #dchan=109loadzone = usloadzone = brdefaultzone = brAnd here my zapata.conf:[trunkgroups][channels]context=defaultswitchtype=nationalpridialplan=national rxwink=300 ; Atlas seems to use long (250ms) winksusecallerid=yeshidecallerid=nocallwaiting=yesusecallingpres=yescallwaitingcallerid=yesthreewaycalling=yestransfer=yescanpark=yes cancallforward=yescallreturn=yesechocancel=yesechocancelwhenbridged=yesechotraining=yesechotraining=400rxgain=0.0txgain=0.0group=1callgroup=1pickupgroup=1immediate=no ;group=1signalling=pri_cpechannel = 1-15,17-31;channel = 32-46,48-62,63-77,79-93;channel = 94-108,110-124I always got this error:chan_zap.c:8324 pri_dchannel: PRI got event: No more alarm (5) on Primary D-channel of span 1 And this when I try to make a call:chan_zap.c:2298 pri_find_dchan: No D-channels available! Using Primary channel 16 as D-channel anyway!Asterisk is unable to make and receive calls from E1. Is there something wrong in the configuration? How can I put it to work?-- Ralph LiebessohnICQ: 74835911Skype: liebessohn ___ --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] Stopping Queue after nobody picked up the call . .
Hi,When I am calling a queue and nobody pick the call the music on hold stop and start again.Does anybody know how to get it off and put the music on hold playing stopless until somebody pick the call? == Spawn extension (default, 12346, 1) exited non-zero on ' Local/[EMAIL PROTECTED],2' -- outgoing agentcall, to agent '1001', on 'Local/[EMAIL PROTECTED],1' -- Called Agent/1001 -- Executing Dial(Local/[EMAIL PROTECTED],2, SIP/12346) in new stack -- Called 12346 -- SIP/12346-0818ec88 is ringing -- Agent/1001 is ringing -- Nobody picked up in 15000 ms Here the MOH stops and start again == Spawn extension (default, 12346, 1) exited non-zero on 'Local/[EMAIL PROTECTED],2' -- outgoing agentcall, to agent '1001', on 'Local/[EMAIL PROTECTED] ,1' -- Called Agent/1001 -- Executing Dial(Local/[EMAIL PROTECTED],2, SIP/12346) in new stack -- Called 12346 -- SIP/12346-0818ec88 is ringing -- Agent/1001 is ringing -- Ralph LiebessohnICQ: 74835911Skype: liebessohn ___ --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] Ringing timer
On 7/26/06, Zenone [EMAIL PROTECTED] wrote: But my question was, is it possible to free the channel if it rings toolong?MichelUsing this thread, is there a way to make differents rings? When receiving a call from a internal user () rings different when a external agent calls (). -- Ralph LiebessohnICQ: 74835911Skype: liebessohn ___ --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] Test E1 channel
On 7/7/06, Moises Silva [EMAIL PROTECTED] wrote: One of the ends must be configured as pri_net and the other aspri_cpe. By the error I think the problem is with your configuration,does zttool says no alarms in spans?Post your configuration files zapata.conf and zaptel.confRegardsOn 7/7/06, Marco Mouta [EMAIL PROTECTED] wrote: by Ports i mean Spans :) On 7/7/06, Marco Mouta [EMAIL PROTECTED] wrote: Newbie guess, Don't you need to set one of the ports NT mode and the other one as TE mode? hope it helps Best regards, PS. give me some feed back if it solved.Hi folks,that was my first try.I had set all the first E1 channel as pri_net and all the second E1 channel as pri_cpe but I got this error. chan_zap.c: PRI Error: We think we-re the network, but they think they're the network, too.When I set everybody as pri_net this message stops.Today, I put the E1 channel to work, it was only set the channel to pri_cpe and dial ! I still without know why the previous tests didn´t work.Thanks everybody.-- Ralph Liebessohn ICQ: 74835911Skype: liebessohn ___ --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] Test E1 channel
Hi guys,I need to make a configuration to test a E1 channel, so, in the same context I created two extensions:exten = 555666,1,Dial(Zap/1/5556662)exten = 5556662,1,Dial(SIP/test) On the E1 card I linked with a cross cable the ports 1 and 2. The leds are signaling that the connection is ok.But when I call 555666 the calling don't goes to client SIP/test .If I call directly 5556662 rings on SIP/test. Do I have to config something else to receive calls on E1 channels?On monday the real E1 channel will be installed and I must test it.-- Ralph LiebessohnICQ: 74835911Skype: liebessohn ___ --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] Test E1 channel
On 7/7/06, James Hawks [EMAIL PROTECTED] wrote: When you dial directly you are bypassing the zap and just dialing an internal extension. So that is probably why dialing directly works. As far as the cross over cable between ports 1 and 2 I have never attempted something like that before. James HawksThe part of crossover is just to simulate a E1 channel. The another end of cable is another port of your E1 card. Some specifications of E1 and crossover E1 cables are here: http://www.hal-pc.org/~ascend/Max/max6000/gs/cables.htmhttp://www.alliancesystems.com/Products/CablesCategory.aspx?id=4 I didn't find the exactly site I got the specification.When I dial 5556662 into my E1 interface it should ring on channel 5556662 on the other end right? Using a crossover, the other end still being my asterisk. -- Ralph LiebessohnICQ: 74835911Skype: liebessohn ___ --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] Test E1 channel
On 7/7/06, Moises Silva [EMAIL PROTECTED] wrote: Oops, i missed the crossover cable part. I have used crossover cable,so it should work, butthe DNID must be complete. Wich signaling areyou using?RegardsHi Moises,I'm signalling=pri_net. I got this error too:app_dial.c: Unable to create channel of type 'Zap' (cause 0 - Unknown)-- Ralph LiebessohnICQ: 74835911Skype: liebessohn ___ --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] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!!
On 6/26/06, Josué Conti [EMAIL PROTECTED] wrote: OK Marco, irei efetuar os testes. Se você quiser, posso lhe ajudar no forum, estou a disposição. Assim que você criar as contas avise para podermos já ir colaborando. Saudações JosuéThe differences of licenses are here: https://www.nch.com.au/cgi-bin/register.exe?software=uplink The site only says that support is different.-- Ralph LiebessohnICQ: 74835911Skype: liebessohn ___ --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] teste E1 card
Hi,Can I, just for test, use a crossover cable linking 2 channels of my E1 card (TE406P) and dial from one channel to another?Is there any different way to do this?-- Ralph Liebessohn ICQ: 74835911Skype: liebessohn ___ --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] Fun with Echo -- Follow up
On 6/20/06, Andrew Kohlsmith [EMAIL PROTECTED] wrote: On Tuesday 20 June 2006 11:30, Brian Swan wrote: 3. Patience and lots of vi zconfig.h: Try each echo canceler, with and without the Aggressive option.What eventually worked for me was the MG2 with Aggressive cancelation.I hate to tell you this, but if you have turned on the aggressive suppressoryou aren't cancelling echo.You have turned your phone into a half-duplexcommunication medium.With the aggressive suppressor enabled, when zaptel detects you talking, it MUTES the received audio.Try it -- call up a friend and ask him to burp the alphabet.While he's doingthat, talk to him.You will stop hearing him whenever you talk.-A. OH God, 40 hours lost !-- Ralph LiebessohnICQ: 74835911Skype: liebessohn ___ --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] just softphone
On 5/18/06, Stefan Märkle [EMAIL PROTECTED] wrote: Try puting apermit=0.0.0.0/0.0.0.0In the sip.conf for your two phones.BTW: your extensions.conf looks silly, you'll only be able to call test3 from test3.Busy most of the time ;-) Stefan MärkleTry puting apermit=0.0.0.0/0.0.0.0in the sip.conf for your two phones.Fine, I tried it.But doesn't solve. So I just started from zero and installed all the system again and it starts to work almost normally. BTW: your extensions.conf looks silly, you'll only be able to call test3 from test3.Busy most of the time ;-)That's for the times when I feel alone and wants to talk to myself ! ! !It was just for tests. -- Ralph LiebessohnICQ: 74835911Skype: liebessohn ___ --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] just softphone
Hi,I'm trying to start with Asterisk, but I could not put 2 softphones to talk.The asterisk server rejects the connections always when I dial.May 17 07:49:22 NOTICE[1924]: Rejected connect attempt from 192.168.0.106What is necessary to put it to work?There is no need to configure external lines.extensions.conf [internal1]exten = 311000,1,Dial(SIP/teste1)[internal2] exten = 312000,1,Dial(SIP/teste2) [internal3]exten = 313000,1,Dial(SIP/teste3) [teste1]sip.conf[teste1]type=friendusername=teste1secret=123 qualify=yesnat=no host=dynamiccanreinvite=no context=internal[teste2]type=friendusername=teste2 secret=123 qualify=yesnat=nohost=dynamiccanreinvite=no context=internal2[teste3]type=friendusername=teste3secret=123 qualify=yesnat=nohost=dynamiccanreinvite=no context=internal3-- Ralph LiebessohnICQ: 74835911Skype: liebessohn ___ --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] just softphone
On 5/18/06, Benchev [EMAIL PROTECTED] wrote: I'm trying to start with Asterisk, but I could not put 2 softphones to talk. The asterisk server rejects the connections always when I dial. May 17 07:49:22 NOTICE[1924]: Rejected connect attempt from 192.168.0.106 What is necessary to put it to work? There is no need to configure external lines. extensions.conf [internal1] exten = 311000,1,Dial(SIP/teste1) [internal2] exten = 312000,1,Dial(SIP/teste2) [internal3] exten = 313000,1,Dial(SIP/teste3) [teste1] sip.conf [teste1] type=friend username=teste1 secret=123 qualify=yes nat=no host=dynamic canreinvite=no context=internal [teste2] type=friend username=teste2 secret=123 qualify=yes nat=no host=dynamic canreinvite=no context=internal2 [teste3] type=friend username=teste3 secret=123 qualify=yes nat=no host=dynamic canreinvite=no context=internal3Debug/verbose is too short, butprobably your peers cannot meet in a mutual context.Try:extensions.conf[default]include = internal[internal]exten = 311000,1,Dial(SIP/teste1)exten = 311000,2,Hangup ; Hangup is goodexten = 312000,1,Dial(SIP/teste2) exten = 312000,2,Hangupexten = 313000,1,Dial(SIP/teste3)exten = 313000,2,HangupPut context=internal or default in all your sip friends.Hope that would do.Benchev Benchev,thanks for the attention.But didn't solve the problem. I think it is something with access.I set debug and verbose to 10 and got this.extensions.conf ( I've changed internal by from-sip) [default]include = from-sipinclude = demo[from-sip]exten = 9222,1,Dial(SIP/9222,25)exten = 9222,2,Hangupexten = 9223,1,Dial(SIP/9223,25)exten = 9223,2,Hangup exten = 31200,1,Dial(SIP/312000,25)exten = 31200,2,Hangupsip.conf[general]context=defaultport=5060 bindaddr=0.0.0.0 ;srvlookup=yes[9222]type=friendcallerid = Nome - 9222 9222username=9222secret=9222host= dynamiccontext=from-sipdtmfmode=rfc2833nat=yescanreinvite=nocontext=internal [9223]type=friendcallerid = Nome - 9223 9223username=9223secret=9223host= dynamiccontext=from-sipdtmfmode=rfc2833nat=yescanreinvite=nocontext=internal [312000]type=friendusername=312000secret=312000host=dynamiccontext=from-sipdtmfmode=rfc2833nat=yescanreinvite=nocontext=internalWhen I try to connect From local machine the softphone only call itself: May 18 10:07:25 DEBUG[2218]: Allocating new SIP call for [EMAIL PROTECTED].1.73May 18 10:07:25 VERBOSE[2218]: -- Registered SIP '9222' at 192.168.1.73 port 5061 expires 1800May 18 10:07:25 VERBOSE[2218]: -- Saved useragent X-Lite release 1105d for peer 9222May 18 10:07:40 DEBUG[2218]: Auto destroying call ' [EMAIL PROTECTED]'When I call From network I got the error Call ended: unknown:May 18 10:10:14 VERBOSE[2213]: Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subcl ass: NEWMay 18 10:10:14 VERBOSE[2213]: Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 3ms SCall: 07747 DCall: 0 [192.168.0.106:4569 ]May 18 10:10:14 VERBOSE[2213]: Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 3ms SCall: 07747 DCall: 0 [192.168.0.106:4569 ] VERSION : 2May 18 10:10:14 VERBOSE[2213]: Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 3ms SCall: 07747 DCall: 0 [ 192.168.0.106:4569] VERSION : 2 CALLING NUMBER : 312000May 18 10:10:14 VERBOSE[2213]: Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subcl VERSION : 2 CALLING NUMBER : 312000 CALLING NAME : 312000May 18 10:10:14 VERBOSE[2213]: Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 3ms SCall: 07747 DCall: 0 [ 192.168.0.106:4569] VERSION : 2 CALLING NUMBER : 312000 CALLING NAME : 312000 FORMAT : 2May 18 10:10:14 VERBOSE[2213]: Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subcl ass: NEW Timestamp: 3ms SCall: 07747 DCall: 0 [192.168.0.106:4569] VERSION : 2 CALLING NUMBER : 312000 CALLING NAME : 312000 FORMAT : 2 CAPABILITY : 1550May 18 10:10:14 VERBOSE[2213]: Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 3ms SCall: 07747 DCall: 0 [ 192.168.0.106:4569] VERSION : 2 CALLING NUMBER : 312000 CALLING NAME : 312000 FORMAT : 2 CAPABILITY : 1550 USERNAME : 312000 Timestamp: 3ms SCall: 07747 DCall: 0 [ 192.168.0.106:4569]May 18 10:10:14 VERBOSE[2213]: Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW CALLING NAME : 312000 FORMAT : 2 CAPABILITY : 1550 USERNAME : 312000 CALLED NUMBER : 9222 DNID : 9222Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REJECT Timestamp: 00010ms SCall: 1 DCall: 07747 [ 192.168.0.106:4569] CAUSE : No authority foundMay 18 10:10:14 NOTICE[2213]: Rejected connect attempt from 192.168.0.106May 18 10:10:14 VERBOSE[2213]: Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subcl ass: NEW Timestamp: 3ms SCall: 07747 DCall: 0 [192.168.0.106:4569] VERSION : 2 CALLING NUMBER : 312000 CALLING NAME : 312000 FORMAT : 2 CAPABILITY : 1550 USERNAME :
