[asterisk-users] Any way of limiting incoming caller connection time without making 2 active calls for each incoming call?

2017-04-16 Thread Jonathan H 
The following setup prevents callers from going over 59 minutes:

--

[setup]
exten => setup,1,Answer()
same => n,Set(LIMIT_PLAYAUDIO_CALLER=yes)
same => 
n,Set(LIMIT_WARNING_FILE=/var/lib/asterisk/sounds/en_GB_TNS/time_limit_reached)
same => n,Dial(Local/s@root/n,3,L(354:6))
same => n,Hangup()

[root]

exten => s,1,Verbose(Call to: ${CALLERID(name)} from: ${CALLERID(num)})
same => n,etc etc

--

Thing is, each call shows as 2 calls in the console. Not a big
problem, but also now I want to offer a way of overriding that time
limit.

Is there a different or better way of doing this while still hanging
up after 59 minutes, even when the original incoming call has been
transferred elsewhere?

Thanks.

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Re: [asterisk-users] tcpbind and source IP address

2017-04-16 Thread Kseniya Blashchuk 
Hi!

Unfortunately pjsip is broken in Ubuntu Asterisk installed from repo. Yes I
also thought to try with pjsip, just to know if it's also affected. I'll
try to make a test next days.

On Sun, Apr 16, 2017, 8:18 PM Ludovic Gasc  wrote:

> Hi Kseniya,
>
> You might test with chan_pjsip: We have less production experience with
> chan_pjsip than chan_sip, however, for now, we are more and more confident
> in this new stack while we're digging in documentation and we're testing on
> production.
>
> However, I've no idea if you'll have the same issue with pjsip, but more
> chances of support on the issues tracker of Asterisk to have help.
>
> Regards.
>
>
> --
> Ludovic Gasc (GMLudo)
> Lead Developer Architect at ALLOcloud
> https://be.linkedin.com/in/ludovicgasc
>
> 2017-03-13 14:41 GMT+01:00 Kseniya Blashchuk :
>
>> Ok, thank you for the assistance!
>>
>> пн, 13 мар. 2017 г. в 16:38, Joshua Colp :
>>
>>> On Mon, Mar 13, 2017, at 10:32 AM, Kseniya Blashchuk wrote:
>>> > Tested with latest Asterisk 14.3.0 on Ubuntu 16 kernel 4.4.0-66-generic
>>> > and
>>> > Centos 7 kernel 3.10.0-514.10.2.el7.x86_64. Absolutely the same
>>> behavior.
>>> > Joshua, maybe you can advice what can be done further?
>>>
>>> You can file an issue but chan_sip is a community supported module, so
>>> there is no guarantee of when it would be looked at and resolved.
>>> Ultimately though someone has to spend the time to replicate what is
>>> going on, look into the code, and understand what is going on.
>>>
>>> --
>>> 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
>>>
>>> --
>>> _
>>> -- 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
>>>
>>
>> --
>> _
>> -- 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
>>
>
> --
> _
> -- 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] tcpbind and source IP address

2017-04-16 Thread Ludovic Gasc 
Hi Kseniya,

You might test with chan_pjsip: We have less production experience with
chan_pjsip than chan_sip, however, for now, we are more and more confident
in this new stack while we're digging in documentation and we're testing on
production.

However, I've no idea if you'll have the same issue with pjsip, but more
chances of support on the issues tracker of Asterisk to have help.

Regards.


--
Ludovic Gasc (GMLudo)
Lead Developer Architect at ALLOcloud
https://be.linkedin.com/in/ludovicgasc

2017-03-13 14:41 GMT+01:00 Kseniya Blashchuk :

> Ok, thank you for the assistance!
>
> пн, 13 мар. 2017 г. в 16:38, Joshua Colp :
>
>> On Mon, Mar 13, 2017, at 10:32 AM, Kseniya Blashchuk wrote:
>> > Tested with latest Asterisk 14.3.0 on Ubuntu 16 kernel 4.4.0-66-generic
>> > and
>> > Centos 7 kernel 3.10.0-514.10.2.el7.x86_64. Absolutely the same
>> behavior.
>> > Joshua, maybe you can advice what can be done further?
>>
>> You can file an issue but chan_sip is a community supported module, so
>> there is no guarantee of when it would be looked at and resolved.
>> Ultimately though someone has to spend the time to replicate what is
>> going on, look into the code, and understand what is going on.
>>
>> --
>> 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
>>
>> --
>> _
>> -- 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
>>
>
> --
> _
> -- 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] Asterisk/FFA version upgrade recommendation

2017-04-16 Thread Ludovic Gasc 
Hi,

We use with success on our production the builtin fax in Asterisk 13, based
on spandsp.
We have several lawyers that use this feature each day with no major issues
to my knowledge.
However, we have enabled T38 on the entire chain and we have a carrier that
handles T38 pretty well.

Before that, we had more frequent failures.

Yours.
--
Ludovic Gasc (GMLudo)
Lead Developer Architect at ALLOcloud
https://be.linkedin.com/in/ludovicgasc

2017-03-12 18:23 GMT+01:00 Mike Diehl :

> Hi all,
>
> I'm needing to upgrade Asterisk from 10.x to whatever the recommended
> version
> is that will allow me to continue to use Fax For Asterisk.
>
> I don't have many upgrade windows, I'd like to get the most bang for my
> buck,
> but I can't afford to be a beta tester on this server.
>
> The FFA site says that it's supported by Asterisk version 12 and lower, but
> version 12 doesn't seem to be supported.  Perhaps my information is
> outdated?
>
> Anyway, I can't go with the spandsp route because my system listens for AMA
> events that spandsp doesn't seem to produce and I can't emulate easily.
>
> Any recommendations would be very welcome.
>
> --
> Mike Diehl
>
>
>
>
> --
> _
> -- 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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