t if it was the firewall we disabled the firewall on the Asterisk
server and moved the Asterisk server before the other firewalls we have.
What could the problem be? And even more important what could solve it
(and/or explain it)?
Kind regards,
-27 at 10:31 +0200, Mark Scholten wrote:
> Hello,
>
> We see some strange behavior with phone calls, we use Asterisk 1.8.3.3.
>
> SIP clients (all behind NAT at different locations, so not a single
> NAT solution is used):
> - x-lite
> - linksys pap2t
> - polycom
+0200, Mark Scholten wrote:
>What could the reason be audio in 1 direction is dropping? (Normally
>from the Asterisk server to the mentioned SIP clients.) No clear
>information is in the logs (it is like the call ended normally) and not
>all calls are having problem (most not, but i
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com] On Behalf Of James Lamanna
> Sent: Saturday, May 01, 2010 9:00 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] ATA shoo
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com] On Behalf Of John Novack
> Sent: Tuesday, May 04, 2010 12:21 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Bridging o
Hello,
I am working on getting the following to work and I couldn't find it in the
documentation I did read. Where should I look or does someone have an
example how I can do it?
Current situation:
Incoming call -> 3 SIP phones + 2 mobile phones ring -> if mobile phone goes
to voicemail the call i
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Sunday, August 01, 2010 3:58 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] # -key not to be 'transfer'
Hello list,
wh
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com] On Behalf Of Alejandro Imass
> Sent: Monday, August 02, 2010 9:00 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] FAX Optio
Hello,
I have a few questions regarding Asterisk 1.8.0. If you can answer a
question, please do so.
Is Asterisk 1.8.0 stable enough for production environments?
Is it possible (and if yes what is the best option) to use CDR MySQL with
Asterisk 1.8.0? With 1.6.x we use the add-on package for that,
> Good luck as with any new version there may be some bugs so if you bump up
against ones report them so they can be fixed.
> Also don't just drop it into production with out testing it on a box for a
bit. 1.8 has a lot of changes. Most appear to be for the better.
The only important difference I
Hello,
We did something like that in the past (but for 1 company, but it shouldn't
be really different). The easiest solution for us was to use a door opener
that could work with almost any "normall" phone connection and use a Linksys
pap2t or something similar.
With kind re
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