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> Subject: [asterisk-users] Fwd: Asterisk With Cisco Voice Router
>
> Hi,
>
> In our office, we're slowly migrating from a cisco call manager set up
> to asterisk. Problem is management doesn't want to buy any other
> hardware as they had already invested a lot in
Thank David and Neeraj for your input.
Neeraj, I posted the configs in my first post, but i've also attached
some extracts here. they haven't changed much.
David, You're absolutely right and i think the problem could be the
reverse dial-peer or DTMF configuration. I think I have the
corresponding
--
Message: 1
Date: Sat, 16 May 2009 14:46:27 +0300
From: Timothy Smith
Subject: [asterisk-users] Fwd: Asterisk With Cisco Voice Router
To: Asterisk Users Mailing List - Non-Commercial Discussion
Message-ID:
<416fc8170905160446r5815fd87m67e62506ad9ac...@mail.gm
On Sat, May 16, 2009 at 10:22 AM, Timothy Smith wrote:
> I have finally managed to get voice working. I both parties can hear
> each other. The problem was nating. Our network is fairly big and
> these machines are atleast 2 switches from each other. I just enabled
> it (nat=route or nat=yes) and
Steve Howes schrieb:
> Check about the sip.conf 'insecure' option. I have had to use it in
> the past for similar stuff. I think it was 'insecure=very' but that
> might be deprecated by now..
insecure=very should now be written as insecure=port,invite
Philipp Kempgen
--
AMOOMA GmbH - Ba
David,
Thanks a lot for your input. I will enable DSP farming. Like some
other techies, I just wanted to see it work before i consider others
things.
I have finally managed to get voice working. I both parties can hear
each other. The problem was nating. Our network is fairly big and
these machi
On Sat, May 16, 2009 at 7:46 AM, Timothy Smith wrote:
> I have added the 3845 router as my SIP gateway (on asterisk 1.6.0.9),
> and also a dialpeer to forward on the router to forward calls to my
> asterisk. It works properly but the problem is there is NO AUDIO! I
> have tried to change codec but
Thanks Steve for this tip.
I have insecure=very is not yet deprecated. I have added it but still no good.
I personally think the problem could be with the codecs. Any ideas?
I have attached some debug info.
Regards,
Tim
On Sat, May 16, 2009 at 3:25 PM, Steve Howes wrote:
>
> On 16 May 2009, a
On 16 May 2009, at 12:46, Timothy Smith wrote:
>
>
> Has anyone had the above set up working successfully? Attached are
> some confs.
>
> Thanks a lot for your assistance.
Check about the sip.conf 'insecure' option. I have had to use it in
the past for similar stuff. I think it was 'insecure
Hi,
In our office, we're slowly migrating from a cisco call manager set up
to asterisk. Problem is management doesn't want to buy any other
hardware as they had already invested a lot in cisco. The main cause
of this is asterisk's added features like unique FAX number for
everyone in the company
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