Re: [asterisk-users] asterisk's internal database
i would like to read information from a file (txt) On Mon, Apr 28, 2014 at 9:29 PM, Rusty Newton wrote: > On Thu, Apr 24, 2014 at 6:34 AM, binary dreamer > wrote: > > hello everyone. > > > > I am running plain asterisk and I am using asterisk's internal database > for: > > -phonebook > > -blacklist numbers > > > > > > instead of having to update the database of new entry or delete an > entry, is > > it possible to have it in an external file such as txt? so every new > > entry/deletion will take place there. > > Are you wanting to swap out Asterisk's internal database with a > different data storage interface? If so, that isn't possible as far as > I know. > > If you are wanting to just read information from a file into Asterisk > variables.. there may be other ways to do what you want. > > -- > Rusty Newton > Digium, Inc. | Community Support Manager > 445 Jan Davis Drive NW - Huntsville, AL 35806 - US > direct: +1 256 428 6200 > > 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] IAX2 trunk on IPV6
Hi, I have installed asterisk-1.8.25.0 on an Ubuntu server which has both an ipv6 ip and ipv4 ip(real ip) assigned. And I have a client ubuntu with only ipv4 ip(local ip) installed asterisk-1.8.25.0 . I want to configure the client asterisk with the server asterisk as IAX2 peer and want to connect to the IPV6 ip. I bind the server with ipv6 and also sending the registration request from the client(peer) to the ipv6 address. But its not peering. following is the client's iax.conf register => peer1:peer1pass@[IPV6]:port [peer1] type=peer context=topeer username=peer1 secret=peer1pass trunk=yes host=XXX.XXX.XXX.XXX port= disallow=all allow=g729:40,g723:30 qualify=yes Also my confusion is what value will be in 'host' property. I assigned as host=[IPV6]...but it shows error. Can anyone help with this issue. Thanks in advance -- _ -- 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] Trunk issue
On 28-04-14 20:13, Haley,Scott A wrote: That seemed to fix it. Thanks to everyone. https://bugzilla.redhat.com/show_bug.cgi?id=1092150 HTH, Patrick -- _ -- 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 call forward for T1 incoming calls
On 04/28/2014 01:37 PM, Don Kelly wrote: On Fri, Apr 25, 2014 at 9:19 AM, Al lists wrote: Is there a way to divert incoming calls on DAHDI T1 channels so telco gets the diversion and send the call to new number and releasing the channel? Rusty Newton: I'm no PRI expert, but I do remember from my time working with the T1 interface cards that two B-channel transfers did something like this. Digium has documentation on that here: http://kb.digium.com/articles/Configuration/Two-B-Channel-Transfers If that doesn't help, and you have a Digium card; you might call Digium tech support to ask about it. Don Kelly adds: TBCT doesn't actually "divert," but it will "transfer" a call. The difference, which may not matter to you, is that one of the channels must be answered by Asterisk (unless something has changed). A call comes in, you decide you want to "divert" it--you place an outbound call on another channel. You must either answer the incoming call or wait for the outgoing call to answer before completing the TBCT. Another quirk to be aware of: You may wish to present the caller's number to the called party as "caller ID." Some carriers will not permit this. Also, the op referred to T1 channels. TBCT works on PRI, not "simple" T1. --Don Additionally, most providers charge for TBCT provisioning, so you would want to make certain that it is the correct way to get the functionality you are after. -- Daniel Taylor VP OperationsVocal Laboratories, Inc. dtay...@vocalabs.com http://www.vocalabs.com/(612)235-5711 -- _ -- 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 call forward for T1 incoming calls
On Fri, Apr 25, 2014 at 9:19 AM, Al lists wrote: > Is there a way to divert incoming calls on DAHDI T1 channels so telco > gets the diversion and send the call to new number and releasing the channel? Rusty Newton: I'm no PRI expert, but I do remember from my time working with the T1 interface cards that two B-channel transfers did something like this. Digium has documentation on that here: http://kb.digium.com/articles/Configuration/Two-B-Channel-Transfers If that doesn't help, and you have a Digium card; you might call Digium tech support to ask about it. Don Kelly adds: TBCT doesn't actually "divert," but it will "transfer" a call. The difference, which may not matter to you, is that one of the channels must be answered by Asterisk (unless something has changed). A call comes in, you decide you want to "divert" it--you place an outbound call on another channel. You must either answer the incoming call or wait for the outgoing call to answer before completing the TBCT. Another quirk to be aware of: You may wish to present the caller's number to the called party as "caller ID." Some carriers will not permit this. Also, the op referred to T1 channels. TBCT works on PRI, not "simple" T1. --Don -- _ -- 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's internal database
On Thu, Apr 24, 2014 at 6:34 AM, binary dreamer wrote: > hello everyone. > > I am running plain asterisk and I am using asterisk's internal database for: > -phonebook > -blacklist numbers > > > instead of having to update the database of new entry or delete an entry, is > it possible to have it in an external file such as txt? so every new > entry/deletion will take place there. Are you wanting to swap out Asterisk's internal database with a different data storage interface? If so, that isn't possible as far as I know. If you are wanting to just read information from a file into Asterisk variables.. there may be other ways to do what you want. -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 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
Re: [asterisk-users] Trunk issue
That seemed to fix it. Thanks to everyone. Thanks, Scott Haley 5-2244 If you are not the intended recipient of this message (including attachments), or if you have received this message in error, immediately notify us and delete it and any attachments. If you do not wish to receive any email messages from us, excluding administrative communications, please email this request to messa...@edwardjones.com along with the email address you wish to unsubscribe. For important additional information related to this email, visit www.edwardjones.com/US_email_disclosure. Edward D. Jones & Co., L.P. d/b/a Edward Jones, 12555 Manchester Road, St. Louis, MO 63131 © Edward Jones. All rights reserved. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Patrick Laimbock Sent: Monday, April 28, 2014 12:58 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Trunk issue On 28-04-14 19:49, Haley,Scott A wrote: > Now I am getting Permission denied. Have you checked if SELinux is blocking the app? Any blockage should show up as an 'AVC' in /var/log/audit/audit.log You can temporarily set SELinux to permissive with 'setenforce 0' and check if the problem goes away. HTH, Patrick -- _ -- 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 call forward for T1 incoming calls
On Fri, Apr 25, 2014 at 9:19 AM, Al lists wrote: > Is there a way to divert incoming calls on DAHDI T1 channels so telco gets > the diversion and send the call to new number and releasing the channel? I'm no PRI expert, but I do remember from my time working with the T1 interface cards that two B-channel transfers did something like this. Digium has documentation on that here: http://kb.digium.com/articles/Configuration/Two-B-Channel-Transfers If that doesn't help, and you have a Digium card; you might call Digium tech support to ask about it. -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 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
Re: [asterisk-users] Trunk issue
On 28-04-14 19:49, Haley,Scott A wrote: Now I am getting Permission denied. Have you checked if SELinux is blocking the app? Any blockage should show up as an 'AVC' in /var/log/audit/audit.log You can temporarily set SELinux to permissive with 'setenforce 0' and check if the problem goes away. HTH, Patrick -- _ -- 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] Trunk issue
Now I am getting Permission denied.
-- Executing [4000@phones:1] NoOp("SIP/7001-003a", "Starting TBS Dailer
App") in new stack
-- Executing [4000@phones:2] NoOp("SIP/7001-003a", "4000") in new stack
-- Executing [4000@phones:3] Gosub("SIP/7001-003a",
"tbs-utils,s,1,(4000)") in new stack
-- Executing [s@tbs-utils:1] NoOp("SIP/7001-003a", "Entering tbs-utils
for 4000") in new stack
-- Executing [s@tbs-utils:2] Set("SIP/7001-003a", "DIALGROUP1=") in new
stack
-- Executing [s@tbs-utils:3] Set("SIP/7001-003a", "DIALGROUP2=") in new
stack
-- Executing [s@tbs-utils:4] Set("SIP/7001-003a", "VM=") in new stack
-- Executing [s@tbs-utils:5] Set("SIP/7001-003a", "TIMER=") in new stack
-- Executing [s@tbs-utils:6] Set("SIP/7001-003a", "BRANCH=") in new
stack
-- Executing [s@tbs-utils:7] AGI("SIP/7001-003a", "tbsdial.agi") in new
stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/tbsdial.agi
tbsdial.agi: Failed to execute '/var/lib/asterisk/agi-bin/tbsdial.agi':
Permission denied
Thanks,
Scott Haley
5-2244
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Asghar Mohammad
Sent: Monday, April 28, 2014 12:42 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Trunk issue
if that is the case then check again Perl Asterisk AGI.
On Mon, Apr 28, 2014 at 7:33 PM, Haley,Scott A
mailto:scott.ha...@edwardjones.com>> wrote:
One more thing. I have this exact same script working on an Asterisk 1.8 box.
This is a new Asterisk 11.7 box.
Thanks,
Scott Haley
5-2244
-Original Message-
From:
asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com]
On Behalf Of Haley,Scott A
Sent: Monday, April 28, 2014 12:32 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Trunk issue
Here is the directory listing:
[root@nxdasterisk-3 agi-bin]# ls -al
total 12
drwxr-xr-x. 2 asterisk asterisk 4096 Apr 28 12:11 .
drwxr-xr-x. 12 asterisk asterisk 4096 Apr 28 12:26 ..
-rwxrwxr-x. 1 asterisk asterisk 590 Apr 28 11:55 tbsdial.agi
Thanks,
Scott Haley
5-2244
-Original Message-
From:
asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com]
On Behalf Of Eric Wieling
Sent: Monday, April 28, 2014 12:25 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Trunk issue
Odd. AGI scripts should hang waiting for input when run from the CLI. They
should not output anything. If the script is not set as executable you'd get
an error.
If you were not running it as the same user as asterisk runs as you should
still get a different error.
-Original Message-
From:
asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com]
On Behalf Of Haley,Scott A
Sent: Monday, April 28, 2014 1:13 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Trunk issue
It runs but hangs with the output of:
perl tbsdial.agi 81101
GET VARIABLE astexten
Right now, it is a simple perl script. Here is the entire script.
#!/usr/bin/perl
use Asterisk::AGI;
my $agi = new Asterisk::AGI;
my $dialgroup1 = "DIALGROUP1";
my $dialgroup2 = "DIALGROUP2";
my $vmvariable = "VM";
my $timer = "TIMER";
my $branch = "BRANCH";
my $input;
my $dg1value;
my $dg2value;
my $vmvalue;
my $branchvalue;
$input = $agi->get_variable("astexten");
#$agi->answer();
#$agi->stream_file("welcome");
$agi->set_variable($dialgroup1, "$dg1value"); $agi->set_variable($dialgroup2,
"$dg2value"); $agi->set_variable($vmvariable, "$vmvalue");
$agi->set_variable($timer, "$timervalue"); $agi->set_variable($branch,
"$branchvalue");
Thanks,
Scott Haley
5-2244
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or if you have received this message in error, immediately notify us and delete
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Louis, MO 63131 © Edward Jones. All rights reserved.
-Original Message-
From:
asterisk-users-boun...@lists.digium.com
Re: [asterisk-users] Trunk issue
if that is the case then check again Perl Asterisk AGI.
On Mon, Apr 28, 2014 at 7:33 PM, Haley,Scott A
wrote:
> One more thing. I have this exact same script working on an Asterisk 1.8
> box. This is a new Asterisk 11.7 box.
>
> Thanks,
> Scott Haley
> 5-2244
>
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] On Behalf Of Haley,Scott A
> Sent: Monday, April 28, 2014 12:32 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Trunk issue
>
> Here is the directory listing:
>
> [root@nxdasterisk-3 agi-bin]# ls -al
> total 12
> drwxr-xr-x. 2 asterisk asterisk 4096 Apr 28 12:11 .
> drwxr-xr-x. 12 asterisk asterisk 4096 Apr 28 12:26 ..
> -rwxrwxr-x. 1 asterisk asterisk 590 Apr 28 11:55 tbsdial.agi
>
> Thanks,
> Scott Haley
> 5-2244
>
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling
> Sent: Monday, April 28, 2014 12:25 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Trunk issue
>
>
> Odd. AGI scripts should hang waiting for input when run from the CLI.
> They should not output anything. If the script is not set as executable
> you'd get an error.
>
> If you were not running it as the same user as asterisk runs as you should
> still get a different error.
>
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] On Behalf Of Haley,Scott A
> Sent: Monday, April 28, 2014 1:13 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Trunk issue
>
> It runs but hangs with the output of:
> perl tbsdial.agi 81101
> GET VARIABLE astexten
>
>
> Right now, it is a simple perl script. Here is the entire script.
>
> #!/usr/bin/perl
>
>
> use Asterisk::AGI;
>
> my $agi = new Asterisk::AGI;
>
> my $dialgroup1 = "DIALGROUP1";
> my $dialgroup2 = "DIALGROUP2";
> my $vmvariable = "VM";
> my $timer = "TIMER";
> my $branch = "BRANCH";
> my $input;
> my $dg1value;
> my $dg2value;
> my $vmvalue;
> my $branchvalue;
>
>
>
> $input = $agi->get_variable("astexten");
>
> #$agi->answer();
> #$agi->stream_file("welcome");
>
>
>
>
>
>
> $agi->set_variable($dialgroup1, "$dg1value");
> $agi->set_variable($dialgroup2, "$dg2value");
> $agi->set_variable($vmvariable, "$vmvalue"); $agi->set_variable($timer,
> "$timervalue"); $agi->set_variable($branch, "$branchvalue");
>
> Thanks,
> Scott Haley
> 5-2244
>
>
>
>
>
> If you are not the intended recipient of this message (including
> attachments), or if you have received this message in error, immediately
> notify us and delete it and any attachments.
>
> If you do not wish to receive any email messages from us, excluding
> administrative communications, please email this request to
> messa...@edwardjones.com along with the email address you wish to
> unsubscribe.
>
> For important additional information related to this email, visit
> www.edwardjones.com/US_email_disclosure. Edward D. Jones & Co., L.P.
> d/b/a Edward Jones, 12555 Manchester Road, St. Louis, MO 63131 © Edward
> Jones. All rights reserved.
>
>
>
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling
> Sent: Monday, April 28, 2014 12:07 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Trunk issue
>
> Does the script generate an error when run outside of Asterisk? An AGI
> should simply wait for input when run outside of Asterisk.
>
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] On Behalf Of Haley,Scott A
> Sent: Monday, April 28, 2014 1:04 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Trunk issue
>
> I am trying to run an agi script and asterisk is not finding it. The
> output of the cli is as follows:
>
> -- Executing [s@tbs-utils:7] AGI("SIP/7002-001a", "tbsdial.agi") in
> new stack [Apr 28 12:00:05] WARNING[21812][C-000f]: res_agi.c:1681
> launch_script: Failed to execute '/var/lib/asterisk/agi-bin/tbsdial.agi':
> File does not exist.
>
> The file is in that directory and is owned by the user "asterisk". Why
> does it say the file does not exist?
> --
> _
> -- 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
Re: [asterisk-users] Trunk issue
One more thing. I have this exact same script working on an Asterisk 1.8 box.
This is a new Asterisk 11.7 box.
Thanks,
Scott Haley
5-2244
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Haley,Scott A
Sent: Monday, April 28, 2014 12:32 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Trunk issue
Here is the directory listing:
[root@nxdasterisk-3 agi-bin]# ls -al
total 12
drwxr-xr-x. 2 asterisk asterisk 4096 Apr 28 12:11 .
drwxr-xr-x. 12 asterisk asterisk 4096 Apr 28 12:26 ..
-rwxrwxr-x. 1 asterisk asterisk 590 Apr 28 11:55 tbsdial.agi
Thanks,
Scott Haley
5-2244
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling
Sent: Monday, April 28, 2014 12:25 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Trunk issue
Odd. AGI scripts should hang waiting for input when run from the CLI. They
should not output anything. If the script is not set as executable you'd get
an error.
If you were not running it as the same user as asterisk runs as you should
still get a different error.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Haley,Scott A
Sent: Monday, April 28, 2014 1:13 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Trunk issue
It runs but hangs with the output of:
perl tbsdial.agi 81101
GET VARIABLE astexten
Right now, it is a simple perl script. Here is the entire script.
#!/usr/bin/perl
use Asterisk::AGI;
my $agi = new Asterisk::AGI;
my $dialgroup1 = "DIALGROUP1";
my $dialgroup2 = "DIALGROUP2";
my $vmvariable = "VM";
my $timer = "TIMER";
my $branch = "BRANCH";
my $input;
my $dg1value;
my $dg2value;
my $vmvalue;
my $branchvalue;
$input = $agi->get_variable("astexten");
#$agi->answer();
#$agi->stream_file("welcome");
$agi->set_variable($dialgroup1, "$dg1value"); $agi->set_variable($dialgroup2,
"$dg2value"); $agi->set_variable($vmvariable, "$vmvalue");
$agi->set_variable($timer, "$timervalue"); $agi->set_variable($branch,
"$branchvalue");
Thanks,
Scott Haley
5-2244
If you are not the intended recipient of this message (including attachments),
or if you have received this message in error, immediately notify us and delete
it and any attachments.
If you do not wish to receive any email messages from us, excluding
administrative communications, please email this request to
messa...@edwardjones.com along with the email address you wish to unsubscribe.
For important additional information related to this email, visit
www.edwardjones.com/US_email_disclosure. Edward D. Jones & Co., L.P. d/b/a
Edward Jones, 12555 Manchester Road, St. Louis, MO 63131 © Edward Jones. All
rights reserved.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling
Sent: Monday, April 28, 2014 12:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Trunk issue
Does the script generate an error when run outside of Asterisk? An AGI should
simply wait for input when run outside of Asterisk.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Haley,Scott A
Sent: Monday, April 28, 2014 1:04 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Trunk issue
I am trying to run an agi script and asterisk is not finding it. The output of
the cli is as follows:
-- Executing [s@tbs-utils:7] AGI("SIP/7002-001a", "tbsdial.agi") in new
stack [Apr 28 12:00:05] WARNING[21812][C-000f]: res_agi.c:1681
launch_script: Failed to execute '/var/lib/asterisk/agi-bin/tbsdial.agi': File
does not exist.
The file is in that directory and is owned by the user "asterisk". Why does it
say the file does not exist?
--
_
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Re: [asterisk-users] Trunk issue
Here is the directory listing:
[root@nxdasterisk-3 agi-bin]# ls -al
total 12
drwxr-xr-x. 2 asterisk asterisk 4096 Apr 28 12:11 .
drwxr-xr-x. 12 asterisk asterisk 4096 Apr 28 12:26 ..
-rwxrwxr-x. 1 asterisk asterisk 590 Apr 28 11:55 tbsdial.agi
Thanks,
Scott Haley
5-2244
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling
Sent: Monday, April 28, 2014 12:25 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Trunk issue
Odd. AGI scripts should hang waiting for input when run from the CLI. They
should not output anything. If the script is not set as executable you'd get
an error.
If you were not running it as the same user as asterisk runs as you should
still get a different error.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Haley,Scott A
Sent: Monday, April 28, 2014 1:13 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Trunk issue
It runs but hangs with the output of:
perl tbsdial.agi 81101
GET VARIABLE astexten
Right now, it is a simple perl script. Here is the entire script.
#!/usr/bin/perl
use Asterisk::AGI;
my $agi = new Asterisk::AGI;
my $dialgroup1 = "DIALGROUP1";
my $dialgroup2 = "DIALGROUP2";
my $vmvariable = "VM";
my $timer = "TIMER";
my $branch = "BRANCH";
my $input;
my $dg1value;
my $dg2value;
my $vmvalue;
my $branchvalue;
$input = $agi->get_variable("astexten");
#$agi->answer();
#$agi->stream_file("welcome");
$agi->set_variable($dialgroup1, "$dg1value"); $agi->set_variable($dialgroup2,
"$dg2value"); $agi->set_variable($vmvariable, "$vmvalue");
$agi->set_variable($timer, "$timervalue"); $agi->set_variable($branch,
"$branchvalue");
Thanks,
Scott Haley
5-2244
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-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling
Sent: Monday, April 28, 2014 12:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Trunk issue
Does the script generate an error when run outside of Asterisk? An AGI should
simply wait for input when run outside of Asterisk.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Haley,Scott A
Sent: Monday, April 28, 2014 1:04 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Trunk issue
I am trying to run an agi script and asterisk is not finding it. The output of
the cli is as follows:
-- Executing [s@tbs-utils:7] AGI("SIP/7002-001a", "tbsdial.agi") in new
stack [Apr 28 12:00:05] WARNING[21812][C-000f]: res_agi.c:1681
launch_script: Failed to execute '/var/lib/asterisk/agi-bin/tbsdial.agi': File
does not exist.
The file is in that directory and is owned by the user "asterisk". Why does it
say the file does not exist?
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Re: [asterisk-users] Trunk issue
Odd. AGI scripts should hang waiting for input when run from the CLI. They
should not output anything. If the script is not set as executable you'd get
an error.
If you were not running it as the same user as asterisk runs as you should
still get a different error.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Haley,Scott A
Sent: Monday, April 28, 2014 1:13 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Trunk issue
It runs but hangs with the output of:
perl tbsdial.agi 81101
GET VARIABLE astexten
Right now, it is a simple perl script. Here is the entire script.
#!/usr/bin/perl
use Asterisk::AGI;
my $agi = new Asterisk::AGI;
my $dialgroup1 = "DIALGROUP1";
my $dialgroup2 = "DIALGROUP2";
my $vmvariable = "VM";
my $timer = "TIMER";
my $branch = "BRANCH";
my $input;
my $dg1value;
my $dg2value;
my $vmvalue;
my $branchvalue;
$input = $agi->get_variable("astexten");
#$agi->answer();
#$agi->stream_file("welcome");
$agi->set_variable($dialgroup1, "$dg1value"); $agi->set_variable($dialgroup2,
"$dg2value"); $agi->set_variable($vmvariable, "$vmvalue");
$agi->set_variable($timer, "$timervalue"); $agi->set_variable($branch,
"$branchvalue");
Thanks,
Scott Haley
5-2244
If you are not the intended recipient of this message (including attachments),
or if you have received this message in error, immediately notify us and delete
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administrative communications, please email this request to
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rights reserved.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling
Sent: Monday, April 28, 2014 12:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Trunk issue
Does the script generate an error when run outside of Asterisk? An AGI should
simply wait for input when run outside of Asterisk.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Haley,Scott A
Sent: Monday, April 28, 2014 1:04 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Trunk issue
I am trying to run an agi script and asterisk is not finding it. The output of
the cli is as follows:
-- Executing [s@tbs-utils:7] AGI("SIP/7002-001a", "tbsdial.agi") in new
stack [Apr 28 12:00:05] WARNING[21812][C-000f]: res_agi.c:1681
launch_script: Failed to execute '/var/lib/asterisk/agi-bin/tbsdial.agi': File
does not exist.
The file is in that directory and is owned by the user "asterisk". Why does it
say the file does not exist?
--
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Re: [asterisk-users] Trunk issue
file is executable?
can you show ls -l /var/lib/asterisk/agi-bin
On Mon, Apr 28, 2014 at 7:12 PM, Haley,Scott A
wrote:
> It runs but hangs with the output of:
> perl tbsdial.agi 81101
> GET VARIABLE astexten
>
>
> Right now, it is a simple perl script. Here is the entire script.
>
> #!/usr/bin/perl
>
>
> use Asterisk::AGI;
>
> my $agi = new Asterisk::AGI;
>
> my $dialgroup1 = "DIALGROUP1";
> my $dialgroup2 = "DIALGROUP2";
> my $vmvariable = "VM";
> my $timer = "TIMER";
> my $branch = "BRANCH";
> my $input;
> my $dg1value;
> my $dg2value;
> my $vmvalue;
> my $branchvalue;
>
>
>
> $input = $agi->get_variable("astexten");
>
> #$agi->answer();
> #$agi->stream_file("welcome");
>
>
>
>
>
>
> $agi->set_variable($dialgroup1, "$dg1value");
> $agi->set_variable($dialgroup2, "$dg2value");
> $agi->set_variable($vmvariable, "$vmvalue");
> $agi->set_variable($timer, "$timervalue");
> $agi->set_variable($branch, "$branchvalue");
>
> Thanks,
> Scott Haley
> 5-2244
>
>
>
>
>
> If you are not the intended recipient of this message (including
> attachments), or if you have received this message in error, immediately
> notify us and delete it and any attachments.
>
> If you do not wish to receive any email messages from us, excluding
> administrative communications, please email this request to
> messa...@edwardjones.com along with the email address you wish to
> unsubscribe.
>
> For important additional information related to this email, visit
> www.edwardjones.com/US_email_disclosure. Edward D. Jones & Co., L.P.
> d/b/a Edward Jones, 12555 Manchester Road, St. Louis, MO 63131 © Edward
> Jones. All rights reserved.
>
>
>
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling
> Sent: Monday, April 28, 2014 12:07 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Trunk issue
>
> Does the script generate an error when run outside of Asterisk? An AGI
> should simply wait for input when run outside of Asterisk.
>
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] On Behalf Of Haley,Scott A
> Sent: Monday, April 28, 2014 1:04 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Trunk issue
>
> I am trying to run an agi script and asterisk is not finding it. The
> output of the cli is as follows:
>
> -- Executing [s@tbs-utils:7] AGI("SIP/7002-001a", "tbsdial.agi") in
> new stack [Apr 28 12:00:05] WARNING[21812][C-000f]: res_agi.c:1681
> launch_script: Failed to execute '/var/lib/asterisk/agi-bin/tbsdial.agi':
> File does not exist.
>
> The file is in that directory and is owned by the user "asterisk". Why
> does it say the file does not exist?
> --
> _
> -- 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
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>http://lists.digium.com/mailman/listinfo/asterisk-users
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> _
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>http://www.asterisk.org/hello
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Re: [asterisk-users] Trunk issue
It runs but hangs with the output of:
perl tbsdial.agi 81101
GET VARIABLE astexten
Right now, it is a simple perl script. Here is the entire script.
#!/usr/bin/perl
use Asterisk::AGI;
my $agi = new Asterisk::AGI;
my $dialgroup1 = "DIALGROUP1";
my $dialgroup2 = "DIALGROUP2";
my $vmvariable = "VM";
my $timer = "TIMER";
my $branch = "BRANCH";
my $input;
my $dg1value;
my $dg2value;
my $vmvalue;
my $branchvalue;
$input = $agi->get_variable("astexten");
#$agi->answer();
#$agi->stream_file("welcome");
$agi->set_variable($dialgroup1, "$dg1value");
$agi->set_variable($dialgroup2, "$dg2value");
$agi->set_variable($vmvariable, "$vmvalue");
$agi->set_variable($timer, "$timervalue");
$agi->set_variable($branch, "$branchvalue");
Thanks,
Scott Haley
5-2244
If you are not the intended recipient of this message (including attachments),
or if you have received this message in error, immediately notify us and delete
it and any attachments.
If you do not wish to receive any email messages from us, excluding
administrative communications, please email this request to
messa...@edwardjones.com along with the email address you wish to unsubscribe.
For important additional information related to this email, visit
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Edward Jones, 12555 Manchester Road, St. Louis, MO 63131 © Edward Jones. All
rights reserved.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling
Sent: Monday, April 28, 2014 12:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Trunk issue
Does the script generate an error when run outside of Asterisk? An AGI should
simply wait for input when run outside of Asterisk.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Haley,Scott A
Sent: Monday, April 28, 2014 1:04 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Trunk issue
I am trying to run an agi script and asterisk is not finding it. The output of
the cli is as follows:
-- Executing [s@tbs-utils:7] AGI("SIP/7002-001a", "tbsdial.agi") in new
stack [Apr 28 12:00:05] WARNING[21812][C-000f]: res_agi.c:1681
launch_script: Failed to execute '/var/lib/asterisk/agi-bin/tbsdial.agi': File
does not exist.
The file is in that directory and is owned by the user "asterisk". Why does it
say the file does not exist?
--
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Re: [asterisk-users] Trunk issue
Does the script generate an error when run outside of Asterisk? An AGI should
simply wait for input when run outside of Asterisk.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Haley,Scott A
Sent: Monday, April 28, 2014 1:04 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Trunk issue
I am trying to run an agi script and asterisk is not finding it. The output of
the cli is as follows:
-- Executing [s@tbs-utils:7] AGI("SIP/7002-001a", "tbsdial.agi") in new
stack [Apr 28 12:00:05] WARNING[21812][C-000f]: res_agi.c:1681
launch_script: Failed to execute '/var/lib/asterisk/agi-bin/tbsdial.agi': File
does not exist.
The file is in that directory and is owned by the user "asterisk". Why does it
say the file does not exist?
--
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Re: [asterisk-users] Trunk issue
I am trying to run an agi script and asterisk is not finding it. The output of
the cli is as follows:
-- Executing [s@tbs-utils:7] AGI("SIP/7002-001a", "tbsdial.agi") in new
stack
[Apr 28 12:00:05] WARNING[21812][C-000f]: res_agi.c:1681 launch_script:
Failed to execute '/var/lib/asterisk/agi-bin/tbsdial.agi': File does not exist.
The file is in that directory and is owned by the user "asterisk". Why does it
say the file does not exist?
Thanks,
Scott Haley
5-2244
If you are not the intended recipient of this message (including attachments),
or if you have received this message in error, immediately notify us and delete
it and any attachments.
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administrative communications, please email this request to
messa...@edwardjones.com along with the email address you wish to unsubscribe.
For important additional information related to this email, visit
www.edwardjones.com/US_email_disclosure. Edward D. Jones & Co., L.P. d/b/a
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Re: [asterisk-users] Buzzing / Humming Noise
To me sounds like an electrical interference. I assume you're not using PoE. Try plugging the phone to someplace else and see if the noise still appears. Use wifi and battery powered cellphones with SIP clients to rule out - pbx - parts of the wired network To confirm it's an electrical interference, try to schedule an out-of-hours test when you switch off all PCs, fridge, whatever you may have in the office. I'm guessing the buzzing sound will not be present. Start switching PCs on one by one while listening, the buzzing sound will return at a certain power consumer device, or after having a certain number switched on. -- _ -- 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] Buzzing / Humming Noise
Hello, I'm afraid it has nothing to do with Asterisk, but bad ground loop on your electrical installations. Solve this, and you get rid of the buzzing. Le 28/04/2014 14:08, Shahid H a écrit : > Hi > > Lately I've been having problem with the handsets on the polycom > phones. The agents are complaining that the humming/buzzing can be > heard on random days. > > When the agents touch their computer the buzzing may increase louder > or reduced. Also when they grab the handset cable, the buzzing noise > is less loud and is loud again when I let it go. > > I have tried changing the handsets, problem not fixed. > > What is causing this? Is this networking issue? I have tried > restarting the PBX Servers, switches, etc. Still same problem. -- _ -- 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] Buzzing / Humming Noise
Hi Lately I've been having problem with the handsets on the polycom phones. The agents are complaining that the humming/buzzing can be heard on random days. When the agents touch their computer the buzzing may increase louder or reduced. Also when they grab the handset cable, the buzzing noise is less loud and is loud again when I let it go. I have tried changing the handsets, problem not fixed. What is causing this? Is this networking issue? I have tried restarting the PBX Servers, switches, etc. Still same problem. -- _ -- 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
