Re: [asterisk-users] Asterisk 13.X with multiple IP addresses: Can I force a given chan_sip peer to a given IP address ?

[Dovid Bender]

Oliver,

Not per peer. What you can do is if you have a few spare IP's is set up
OpenSiPS and bounce the calls that way. The call will be Carrier ->
OpenSipS -> Asterisk. Asterisk will see the IP of OpenSipS and you can
decide there what to do with the call.



On Wed, Sep 6, 2017 at 5:39 AM, Olivier  wrote:

> Hello,
>
> I'm quite sure this question has already be asked previously but before
> diving into it with a lab setup, I would like to re-ask here the thereafter
> question.
>
> I've got a bunch of very old Asterisk boxes (lastest Asterisk version is
> 1.6.1.X), all belonging to the same network, I would like to centralize on
> a single Asterisk instance on a brand new box.
> This instance will be powered with lastest available LTS Asterisk
> (Asterisk 13.X, currently).
>
> I know this is possible with PJSIP transport setting but with chan_sip, is
> it possible to specify that to communicate with a given SIP peer, a
> specific IP address (and only this one) is used ?
>
> Best regards
>
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>
> Check out the new Asterisk community forum at: https://community.asterisk.
> org/
>
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Re: [asterisk-users] Asterisk 11.25.2

[Jerry Geis]

Marcelo,

I had to drop back to 11.25.1 yesterday. Had other things all day
yesterday. I have not had a chance to try anything yet with the suggestion
from Josh.

Jerry
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[asterisk-users] Asterisk 13.X with multiple IP addresses: Can I force a given chan_sip peer to a given IP address ?

[Olivier]

Hello,

I'm quite sure this question has already be asked previously but before
diving into it with a lab setup, I would like to re-ask here the thereafter
question.

I've got a bunch of very old Asterisk boxes (lastest Asterisk version is
1.6.1.X), all belonging to the same network, I would like to centralize on
a single Asterisk instance on a brand new box.
This instance will be powered with lastest available LTS Asterisk (Asterisk
13.X, currently).

I know this is possible with PJSIP transport setting but with chan_sip, is
it possible to specify that to communicate with a given SIP peer, a
specific IP address (and only this one) is used ?

Best regards
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[asterisk-users] current cpu recommendation for asterisk 13 + app_queue

[marek cervenka]

hi,

i know about architecture limits of app_queue

https://issues.asterisk.org/jira/browse/ASTERISK-25806


what CPUs are you actually using for asterisk + app_queue ? (my actual 
scenario 90simult calls, 50agents, call recording to SSD (mixmonitor),  
no transcoding, CDR/CEL via odbc to MariaDB)


customer offers
    Intel Xeon E5-2680v3 - 2,5GHz@9,6GT 30MB cache, 12core,HT, 
120W,LGA2011


i think for app_queue will be better
    Intel Xeon E5-2637v3 - 3,5GHz@9,6GT 15MB cache, 4core,HT, 
135W,LGA2011,tray



thanks

Marek



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Re: [asterisk-users] Asterisk 11.25.2

[Marcelo Terres]

Hello Jerry.

Does the Joshua's tips helped you to solve your issues or are you still
facing audios problems?

I am asking you because I need to update some servers but I can't have this
kind of problems.

Thanks.

Regards,

On 5 Sep 2017 2:02 pm, "Joshua Colp"  wrote:

> On Tue, Sep 5, 2017, at 09:56 AM, Jerry Geis wrote:
> > My setup using 11.25.1 was working. When I installed 11.25.2 I now get
> > "sort of" working.
> >
> > I am using NAT in the setup. When I have an internal phone and call out I
> > get audio both ways.
> > But when I call IN my phone rings but I have no audio.
> >
> > Is there a new setting I need to tweek ?
>
> You can try setting "strictrtp" to "no" in rtp.conf and seeing if that
> resolves the issue. If it does then getting a packet capture of the
> traffic could confirm why we are dropping the media. It may be that the
> source is changing without telling us, which the security fix protects
> against.
>
> --
> 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
>
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