n/media_index.c line 140.
Anyone has some insight, suggestions, or ways to better diagnose this?
Thanks in advance.
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Guido Falsi
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t for getting calls, to make a call your phone has to
authenticate each time he sends an invite. That's why it works without
being registered.
So, trying to bind authentication to originate calls to registrations is
conceptually wrong in
On 06/11/15 10:43, Luca Bertoncello wrote:
> Zitat von Guido Falsi :
>
>> So, trying to bind authentication to originate calls to registrations is
>> conceptually wrong in the SIP world. Maybe you can do that but that's
>> not the way the protocols have been engi
pictures,
recordings and other datatypes:
https://core.telegram.org/bots
Not sure if this is what you are looking for.
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Guido Falsi
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I think the way to achieve that is by using the Bridge application:
https://wiki.asterisk.org/wiki/display/AST/Bridge+Application
https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Application_Bridge
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27;ll have to hold up updating the FreeBSD Asterisk port.
Thanks in advance.
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Guido Falsi
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Check out the new Asterisk community forum at: https://c
On 11/22/16 11:46, Joshua Colp wrote:
> On Tue, Nov 22, 2016, at 05:55 AM, Guido Falsi wrote:
>> In my setup, which is FreeBSD, using pjsip 2.5.5 as sip backend I am
>> observing a regression when testing the latest Release Candidate.
>>
>> Any calls get refused and t
On 11/22/16 11:57, Guido Falsi wrote:
> On 11/22/16 11:46, Joshua Colp wrote:
>>> Is this a known issue being worked on? Should I file a bug report?
>>
>> Everything is tracked on JIRA, and there's no current issue open for
>> this so it's not being work
work depending on how close they are.
>
BTW, I had to disable it by defualt in the FreeBSD port, otherwise the
official binary packages would be broken on CPUs different from the ones
used by the FreeBSD project build cluster.
I think all linux distributions are/should do the same for defa
vps would be identical.
In fact there are chances the CPU will not be the same even by stopping
and then restarting the same VM. The Hypervisor could choose to start
the same VM on different iron depending on it's load balancing settings
and whim at the moment.
nexpired patents are usable on
> a royalty-free basis.[7] Thus, G.729 can be used free-of-charge.
> —>8—>8—
>
A better source is from the consortium which has been upholding the patents:
http://www.sipro.com/G729.html
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Guido Falsi
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me strings
Are you sure this should no be simply "de_DE"?
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Guido Falsi
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Check out the new Asterisk community forum at: https://community.ast
re not configuring where to listen. Simply
tell asterisk what IP to put in the SDP data. You don't need to suppose
or guess the content of the SIP/SDP packets, you can look at them by
enabling asterisk debug output.
you can enable it following this guide:
https://wiki.asterisk.org/wiki/display
On 11/01/2017 10:14, Antony Stone wrote:
>
> If you don't have a G.729 licence, don't offer G.729 to the peer.
AFAIK the g729 patents have expired or are granted royalty free for the
holder's declaration:
http://www.sipro.com/G729.
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