-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 cisco. The main cause
of this is asterisk's added features
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
On Sat, May 16, 2009 at 10:22 AM, Timothy Smith timotsm...@gmail.com 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
Date: Sat, 16 May 2009 14:46:27 +0300
From: Timothy Smith timotsm...@gmail.com
Subject: [asterisk-users] Fwd: Asterisk With Cisco Voice Router
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID
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
On 16 May 2009, at 12:46, Timothy Smith wrote:
blah
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
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 st...@geekinter.net wrote:
On Sat, May 16, 2009 at 7:46 AM, Timothy Smith timotsm...@gmail.com 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
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
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 -
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