Re: [asterisk-users] Any way of creating a file to write to from the dialplan, or must I use AGI?

[John Kiniston]

Could it be SELinux blocking you?

If you change the path to /tmp does it work?

On Fri, Nov 4, 2016 at 3:14 PM, Jonathan H  wrote:

> That's just what I'm using, John.
>
> But I'm getting (eg)
>
> [Nov  4 21:46:16] ERROR[1676][C-0003]: func_env.c:449 file2format:
> Cannot open '/home/logs/anonymous.txt': No such file or directory
> [Nov  4 21:46:16] ERROR[1676][C-0003]: func_env.c:949 file_write:
> File '/home/logs/anonymous.txt' not in line format
>
> Asterisk is running as root (yeah, I know!), and has permissions on
> that directory. Hmmm
>
> On 4 November 2016 at 21:50, John Kiniston  wrote:
> > I'm able to use the FILE function to create files just fine.
> >
> > Set(FILE(${CALLFILE},,,al,u)=Extension: s)
> >
> > On Fri, Nov 4, 2016 at 2:26 PM, Jonathan H 
> wrote:
> >>
> >> Seems I can write to an existing file, but is there really no way of
> >> creating a new file to log some data to, without reverting to AGI?
> >> (will be different for each caller ID)
> >>
> >> --
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> >
> >
> >
> >
> > --
> > A human being should be able to change a diaper, plan an invasion,
> butcher a
> > hog, conn a ship, design a building, write a sonnet, balance accounts,
> build
> > a wall, set a bone, comfort the dying, take orders, give orders,
> cooperate,
> > act alone, solve equations, analyze a new problem, pitch manure, program
> a
> > computer, cook a tasty meal, fight efficiently, die gallantly.
> > Specialization is for insects.
> > ---Heinlein
> >
> > --
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> > Check out the new Asterisk community forum at:
> > https://community.asterisk.org/
> >
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> org/
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>
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-- 
A human being should be able to change a diaper, plan an invasion, butcher
a hog, conn a ship, design a building, write a sonnet, balance accounts,
build a wall, set a bone, comfort the dying, take orders, give orders,
cooperate, act alone, solve equations, analyze a new problem, pitch manure,
program a computer, cook a tasty meal, fight efficiently, die gallantly.
Specialization is for insects.
---Heinlein
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[asterisk-users] AMI version in Asterisk 14

[Telium Technical Support]

I noticed that Asterisk 14 has changed the output format for some commands
(eg: "Output: ").  However, the AMI reports version 2.8.0 which is the same
as Asterisk 13

 

Is that considered a bug?  Since the AMI output format has changed,
shouldn't the AMI version be incremented?  (Makes is hard for developers to
maintain compatability)

 

Hans

 

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Re: [asterisk-users] Any way of creating a file to write to from the dialplan, or must I use AGI?

[Jonathan H]

That's just what I'm using, John.

But I'm getting (eg)

[Nov  4 21:46:16] ERROR[1676][C-0003]: func_env.c:449 file2format:
Cannot open '/home/logs/anonymous.txt': No such file or directory
[Nov  4 21:46:16] ERROR[1676][C-0003]: func_env.c:949 file_write:
File '/home/logs/anonymous.txt' not in line format

Asterisk is running as root (yeah, I know!), and has permissions on
that directory. Hmmm

On 4 November 2016 at 21:50, John Kiniston  wrote:
> I'm able to use the FILE function to create files just fine.
>
> Set(FILE(${CALLFILE},,,al,u)=Extension: s)
>
> On Fri, Nov 4, 2016 at 2:26 PM, Jonathan H  wrote:
>>
>> Seems I can write to an existing file, but is there really no way of
>> creating a new file to log some data to, without reverting to AGI?
>> (will be different for each caller ID)
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> Check out the new Asterisk community forum at:
>> https://community.asterisk.org/
>>
>> New to Asterisk? Start here:
>>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
>
> --
> A human being should be able to change a diaper, plan an invasion, butcher a
> hog, conn a ship, design a building, write a sonnet, balance accounts, build
> a wall, set a bone, comfort the dying, take orders, give orders, cooperate,
> act alone, solve equations, analyze a new problem, pitch manure, program a
> computer, cook a tasty meal, fight efficiently, die gallantly.
> Specialization is for insects.
> ---Heinlein
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users

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Re: [asterisk-users] Any way of creating a file to write to from the dialplan, or must I use AGI?

[John Kiniston]

I'm able to use the FILE function to create files just fine.

Set(FILE(${CALLFILE},,,al,u)=Extension: s)

On Fri, Nov 4, 2016 at 2:26 PM, Jonathan H  wrote:

> Seems I can write to an existing file, but is there really no way of
> creating a new file to log some data to, without reverting to AGI?
> (will be different for each caller ID)
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at: https://community.asterisk.
> org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 
A human being should be able to change a diaper, plan an invasion, butcher
a hog, conn a ship, design a building, write a sonnet, balance accounts,
build a wall, set a bone, comfort the dying, take orders, give orders,
cooperate, act alone, solve equations, analyze a new problem, pitch manure,
program a computer, cook a tasty meal, fight efficiently, die gallantly.
Specialization is for insects.
---Heinlein
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Re: [asterisk-users] Any way of creating a file to write to from the dialplan, or must I use AGI?

[Jonathan H]

Yes, that would also work (thanks!).

It just seems a bit hacky - STAT...GotoIf... System..,Return...FILE

Has there been any previous discussion as to why FILE can't/won't
create a file and write to it in one shot?

If so, what was the outcome? Should I suggest it?

Thanks!

On 4 November 2016 at 21:32, John Covici  wrote:
> Won't the system command do it?
>
> On Fri, 04 Nov 2016 17:26:13 -0400,
> Jonathan H wrote:
>>
>> Seems I can write to an existing file, but is there really no way of
>> creating a new file to log some data to, without reverting to AGI?
>> (will be different for each caller ID)
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> Check out the new Asterisk community forum at: 
>> https://community.asterisk.org/
>>
>> New to Asterisk? Start here:
>>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>
> --
> Your life is like a penny.  You're going to lose it.  The question is:
> How do
> you spend it?
>
>  John Covici
>  cov...@ccs.covici.com
>
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>
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>
> New to Asterisk? Start here:
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>
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>http://lists.digium.com/mailman/listinfo/asterisk-users

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[asterisk-users] Suddenly getting lots of "Unable to send packet: Address Family mismatch between source/destination" but ONLY on 1 of 2 VPSs in same datacentre.

[Jonathan H]

Two VPSs. Identical setups with the exception of the extension.

Same version of everything, Asterisk 14.1, Ubuntu 16.10, same firewall
rules and so on -  box 2 was cloned from box 1.

Both VPSs run in the same datacentre.

Suddenly, after weeks of OK, I'm getting lots of this on ONE box only:

[Nov  4 21:23:04] NOTICE[1468]: res_hep.c:466 hep_queue_cb: Unable to
send packet: Address Family mismatch between source/destination
[Nov  4 21:23:04] NOTICE[1468]: res_hep.c:466 hep_queue_cb: Unable to
send packet: Address Family mismatch between source/destination

Googling this brings up something about IPv6, but my VPS is not IPv6
enabled, and nor is my ITSP, who can't think of why this is happening.

It doesn't appear to be adversely affecting call quality, but it's
making the console a bit busy. I know I can hide it, but I'd be
interested to know if there's anything I can do to diagnose WHY this
is happening.

Thanks

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Re: [asterisk-users] Any way of creating a file to write to from the dialplan, or must I use AGI?

[John Covici]

Won't the system command do it?

On Fri, 04 Nov 2016 17:26:13 -0400,
Jonathan H wrote:
> 
> Seems I can write to an existing file, but is there really no way of
> creating a new file to log some data to, without reverting to AGI?
> (will be different for each caller ID)
> 
> -- 
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> 
> Check out the new Asterisk community forum at: https://community.asterisk.org/
> 
> New to Asterisk? Start here:
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> 
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> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

 John Covici
 cov...@ccs.covici.com

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[asterisk-users] Any way of creating a file to write to from the dialplan, or must I use AGI?

[Jonathan H]

Seems I can write to an existing file, but is there really no way of
creating a new file to log some data to, without reverting to AGI?
(will be different for each caller ID)

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Re: [asterisk-users] Force hangup not working on stuck channel

[Matt Fredrickson]

Also, it looks like in
https://issues.asterisk.org/jira/browse/ASTERISK-21762 there might be
a workaround (see the last comment at the bottom).

Matthew Fredrickson

On Fri, Nov 4, 2016 at 2:01 PM, Matt Fredrickson  wrote:
> On Thu, Nov 3, 2016 at 11:16 AM, Carlos Chavez  
> wrote:
>> I am unable to force a hangup on a channel that has been stuck for over two
>> days:
>>
>> IAX2/from-CD-11006   oficina  27701 Up  Dial
>> IAX2/to-CD/2883   3467130007  46:24:59 Sotelo  Sotelo
>> IAX2/to-CD-20713
>>
>> I have tried "hangup request IAX2/from-CD-11006" several times but no joy.
>> I also see the following in the CLI:
>>
>> [Nov  3 10:05:54] WARNING[2879]: chan_iax2.c:4936 handle_call_token: Too
>> much delay in IAX2 calltoken timestamp from address X.X.X.X
>>
>> This is an IAX2 trunk between two Asterisk 1.8 servers (I know it is old but
>> new client so haven't had time yet to upgrade to 13).  Because this channels
>> is stuck
>>  all other calls between servers are not working.  The only way I have found
>> to resolve the problem is to stop and restart Asterisk.  This is obviously a
>> great inconvinience so is there a way for force iax to unload even if there
>> are channels in use?  Or any other way to kill these stubborn channels?
>
> If doing a soft hangup on them doesn't work, the only other way I know
> to do it is to restart Asterisk.  Sorry about the bad news :-(
>
> There's a part of me that's curious as to why the channel is stuck,
> but there's another part of me that says "1.8... run away quickly".
>
> You could try to replicate it with a modern branch (i.e. 13/14) and
> see if it still exists.  At the very least that'd leave you the option
> of posting a bug on the issue tracker about it.  Also, a lot of bugs
> have been fixed since 1.8, so it's quite possible that this issue is
> resolved as well.
>
> --
> Matthew Fredrickson
> Digium, Inc. | Engineering Manager
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA



-- 
Matthew Fredrickson
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

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Re: [asterisk-users] Force hangup not working on stuck channel

[Matt Fredrickson]

On Thu, Nov 3, 2016 at 11:16 AM, Carlos Chavez  wrote:
> I am unable to force a hangup on a channel that has been stuck for over two
> days:
>
> IAX2/from-CD-11006   oficina  27701 Up  Dial
> IAX2/to-CD/2883   3467130007  46:24:59 Sotelo  Sotelo
> IAX2/to-CD-20713
>
> I have tried "hangup request IAX2/from-CD-11006" several times but no joy.
> I also see the following in the CLI:
>
> [Nov  3 10:05:54] WARNING[2879]: chan_iax2.c:4936 handle_call_token: Too
> much delay in IAX2 calltoken timestamp from address X.X.X.X
>
> This is an IAX2 trunk between two Asterisk 1.8 servers (I know it is old but
> new client so haven't had time yet to upgrade to 13).  Because this channels
> is stuck
>  all other calls between servers are not working.  The only way I have found
> to resolve the problem is to stop and restart Asterisk.  This is obviously a
> great inconvinience so is there a way for force iax to unload even if there
> are channels in use?  Or any other way to kill these stubborn channels?

If doing a soft hangup on them doesn't work, the only other way I know
to do it is to restart Asterisk.  Sorry about the bad news :-(

There's a part of me that's curious as to why the channel is stuck,
but there's another part of me that says "1.8... run away quickly".

You could try to replicate it with a modern branch (i.e. 13/14) and
see if it still exists.  At the very least that'd leave you the option
of posting a bug on the issue tracker about it.  Also, a lot of bugs
have been fixed since 1.8, so it's quite possible that this issue is
resolved as well.

-- 
Matthew Fredrickson
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

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Re: [asterisk-users] pjsip transports from database.

[Bryant Zimmerman]

 
 On Friday, November 4, 2016 10:20 AM - Joshua Colp wrote:
 
 >>On Fri, Nov 4, 2016, at 10:26 AM, Bryant Zimmerman wrote:
>> Hey all
>>
>> I am trying to configure all my pjsip transports form a database table.
>> The issue I am running into is that pjsip is auto binding to 
0.0.0.0:5060
>> before it reads my list of transports from the database. This means 
that
>> my
>> entries for port 5060 are already bound and the settings in the 
database
>> are not loaded.
>>
>> When loading the transport form the .conf file it works as expected and
>> does not do an auto binding, but uses what is in the .conf
>>
>> Is there a way to have asterisk pjsip hold the default binding override
>> until after it has checked the database when sourcery .conf configures 
a
>> transport location other then pjsip.conf?

>>PJSIP has no auto binding or default binding. It will only bind to what
>>Is configured. Do you have it in both .conf and in realtime? Do you also
>>have chan_sip loaded?

Joshua
  
 You were correct. There was an old chan_sip.so in the bin folder that was 
being auto loaded. It was binding to 0.0.0.0:5060 causing the transports 
from the database for pjsip to fail. I forced down asterisk and deleted the 
chan_sip.so from the bin folder and the issue resolved. Looks like I need 
to go through and clean up old garbage from an earlier build so I don't get 
caught in the future. I also added a noload for chan_sip.so just incase one 
ever gets dropped back in the folder. 
  
 Much thanks for the direction here I spent a lot of time trying to figure 
out where the binding was coming from.
  
 Bryant

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Re: [asterisk-users] pjsip transports from database.

[Joshua Colp]

On Fri, Nov 4, 2016, at 10:26 AM, Bryant Zimmerman wrote:
> Hey all
>   
>  I am trying to configure all my pjsip transports form a database table. 
> The issue I am running into is that pjsip is auto binding to 0.0.0.0:5060 
> before it reads my list of transports from the database. This means that
> my 
> entries for port 5060 are already bound and the settings in the database 
> are not loaded.
>   
>  When loading the transport form the .conf file it works as expected and 
> does not do an auto binding, but uses what is in the .conf
>   
>  Is there a way to have asterisk pjsip hold the default binding override 
> until after it has checked the database when sourcery .conf configures a 
> transport location other then pjsip.conf?

PJSIP has no auto binding or default binding. It will only bind to what
is configured. Do you have it in both .conf and in realtime? Do you also
have chan_sip loaded?

-- 
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org

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[asterisk-users] pjsip transports from database.

[Bryant Zimmerman]

Hey all
  
 I am trying to configure all my pjsip transports form a database table. 
The issue I am running into is that pjsip is auto binding to 0.0.0.0:5060 
before it reads my list of transports from the database. This means that my 
entries for port 5060 are already bound and the settings in the database 
are not loaded.
  
 When loading the transport form the .conf file it works as expected and 
does not do an auto binding, but uses what is in the .conf
  
 Is there a way to have asterisk pjsip hold the default binding override 
until after it has checked the database when sourcery .conf configures a 
transport location other then pjsip.conf?

Thanks

Bryant

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Re: [asterisk-users] Is it possible that variables returned from AGI take a moment to "stick"?

[virendra bhati]

I don't think so any such method to return variable from AGI. But simple
solution is set variable in AGI and then you can get back after AGI call in
dialplan and these variable will be available until call finished.


---
 Virendra Bhati
+91-9718500594
+91-9250078532
Sr. Asterisk Developer
E-mail-: virbh...@gmail.com
Skype id:- virbhati2
New Delhi(India)


On Fri, Oct 21, 2016 at 6:06 PM, Jonathan H  wrote:

> I thought dialplan flow was that (normal!) agi was called, it did its
> thing (which include returning some dialplan variables/lists), and
> then when agi finished it returned to the dialplan which then reliably
> carried the product of agi.
>
> But I'm calling agi, scanning a path in python, and then finding that
> unless I call a 1 second wait in the dialplan AFTER the agi, sometimes
> the variable is empty, even though agi debug shows it was sent.
>
> Any tests I can do, or is this to be expected?
>
> --
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>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
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Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users