Re: [asterisk-users] Asterisk and CentOS 8

[Carlos Chavez]

    They only problem I have found so far is while trying to install 
Alembic for SQLAlchemy (for realtime configs).  Those are the only 
packages that I cannot get working properly.  Vanilla Asterisk works 
fine  with the only extra package needed being libedit-devel that is not 
included in any "official" repo.  You need to download the Fedora Core 
29 packages to in order to successfully compile Asterisk.  That being 
said, I would not recommend trying to put this in production any time soon.


On 10/17/2019 11:19 AM, George Joseph wrote:
At the current time, we do not recommend attempting to build Asterisk 
on CentOS 8.  Many packages Asterisk uses are not yet available and 
would require building from their sources.  The Asterisk packages are 
also not available in the EPEL 8 or CentOS 8 repositories yet for the 
same reason.


We'll update you when we think it's safe.


--
*George Joseph*
Digium - A Sangoma Company | Software Developer | Software Engineering
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
direct/fax: +1 256 428 6012
Check us out at: https://digium.com  · 
https://sangoma.com 



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Re: [asterisk-users] PJSIP Setup Outbound SIP Trunk

[Ahmed Chohan]

Thanks for reply.

After going through the all configurations, there was syntax error with the
dial plan for outbound call i.e. previously I was using
"Dial(PJSIP/trunk_proxy/${EXTEN})" and was unable to make outbound calls.
Later changed to "Dial( PJSIP/${EXTEN}@ trunk_proxy)" it worked as expected
i.e. no need to set auth/reg for the SIP trunk as not setting it up at SIP
Proxy end.

Date: Wed, 16 Oct 2019 13:27:30 -0500
> From: Kevin Harwell 
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> 
> Subject: Re: [asterisk-users] PJSIP Setup Outbound SIP Trunk
> Message-ID:
>  e2cuhg0xpdfpkida8zrkokvpv1s4ymqs9kgpop+a...@mail.gmail.com>
> Content-Type: text/plain; charset="utf-8"
>
> On Mon, Oct 14, 2019 at 11:56 AM Ahmed Chohan 
> wrote:
>
> > Hi,
> >
> > I've currently migrating from chan_sip to chan_pjsip, for now I'm able to
> > setup and configured extensions in PJSIP and incoming trunks but unable
> to
> > configure outbound trunk as getting unauth/unregistered trunk endpoint
> > message error message when making outbound calls. However, for inbound
> > calls I'm not facing any issues.
> >
> > I would like to know how can I configured outbound sip trunk bypassing
> > registration and auth?
> >
>
> Where are the messages coming from? Is Asterisk sending an outbound
> registration, but getting rejected? If so make sure your username/password
> credentials are correct.
>
>
> >
> > See below current configuration;
> >
> > [trunk_proxy]
> > type=endpoint
> > transport=transport-udp
> > context=fromsip
> > disallow=all
> > allow=ulaw
> > aors=trunk_proxy
> > force_rport=no
> > direct_media=yes
> > ice_support=no
> > trust_id_inbound=yes
> > outbound_auth=trunk_proxy
> >
> > [trunk_proxy]
> > type=aor
> > contact=sip:10.3.120.208:5060
> >
> > [trunk_proxy]
> > type=identify
> > endpoint=trunk_proxy
> > match=10.3.120.208
> >
> > [trunk_proxy]
> > type=auth
> > auth_type=userpass
> > password=
> > username=sip_proxy
> >
> > [trunk_proxy]
> > type=registration
> > outbound_auth=trunk_proxy
> > server_uri=sip:10.3.120.208:5060
> > client_uri=sip:10.3.120.208:5060
> > auth_rejection_permanent=no
> >
> > --
> > Regards,
> >
> > Ahmed Munir Chohan
> >
> > --
> > _
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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
>
>
>
> --
> Kevin Harwell
> Digium - A Sangoma Company | Senior Software Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
> Check us out at: https://digium.com & https://asterisk.org
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Regards,

Ahmed Munir Chohan
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[asterisk-users] Asterisk and CentOS 8

[George Joseph]

At the current time, we do not recommend attempting to build Asterisk on
CentOS 8.  Many packages Asterisk uses are not yet available and would
require building from their sources.  The Asterisk packages are also not
available in the EPEL 8 or CentOS 8 repositories yet for the same reason.

We'll update you when we think it's safe.


-- 
*George Joseph*
Digium - A Sangoma Company | Software Developer | Software Engineering
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
direct/fax: +1 256 428 6012
Check us out at: https://digium.com · https://sangoma.com
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Re: [asterisk-users] Delays on conferences

[Tony Mountifield]

In article ,
Marcelo Terres  wrote:
> 
> We are having a weird issue with conferences.
> 
> Let me explain:
> 
> A enters conference room.
> B enters conference room.
> 
> When B talks, A can listen it immediately. When A talks, took almost a
> second to B receives the audio.
> 
> Scenario:
> Asterisk 11 with meetme.
> CentOS 6/7, Dahdi 2.9/2.11
> 
> I know it is an old version, but we can't change it now. We are moving to
> Asterisk 16 next year, but currently that is our reality.
> 
> Any ideas of what could be causing this? Or any ideas of how to debug it?

What kind of channels are they? SIP? IAX? DAHDI?

What happens if you have 3 calls in the conference? In that case, when you
have a delay from one speaking, do both the others hear the delay, or just
one of them? That would determine whether the delay is on audio going to
Asterisk or from Asterisk.

Do you have internal_timing set in asterisk.conf?

What timing module are you using?

Does it always happen, or just sometimes?

Cheers
Tony
-- 
Tony Mountifield
Work: t...@softins.co.uk - http://www.softins.co.uk
Play: t...@mountifield.org - http://tony.mountifield.org

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[asterisk-users] Delays on conferences

[Marcelo Terres]

Hello.

We are having a weird issue with conferences.

Let me explain:

A enters conference room.
B enters conference room.

When B talks, A can listen it immediately. When A talks, took almost a
second to B receives the audio.

Scenario:
Asterisk 11 with meetme.
CentOS 6/7, Dahdi 2.9/2.11

I know it is an old version, but we can't change it now. We are moving to
Asterisk 16 next year, but currently that is our reality.

Any ideas of what could be causing this? Or any ideas of how to debug it?

Thanks.

Regards,

Marcelo H. Terres 
https://www.mundoopensource.com.br
https://twitter.com/mhterres
https://linkedin.com/in/marceloterres
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