hello everybody,
i want to configure a sip trunk between a cisco router and my system which
has asterisk. this is my scenario:
Freepbx-my system-cisco-routerFreepbx
my system acts like a router. in cisco, if i set just one codec in
dial-peers, every thing is ok and i can make a
Erick, are you doing advanced networking on the C or NAT reflection
for the traversal? Is your traversal connected? I would check for
firewall drops when trying to connect if the traversal is connected.
On Thu, Apr 2, 2015 at 6:50 PM, Erick Wellnitz ewellnitzv...@gmail.com wrote:
Not a
Correct, look at page 12 for the firewall section at
http://www.cisco.com/c/dam/en/us/td/docs/telepresence/infrastructure/vcs/config_guide/X8-5/Mobile-Remote-Access-via-VCS-Deployment-Guide-X8-5.pdf
Also, Nate mentioned newer port list, so double check your network
settings from this doc.
On
You need to use the newer docs. The port list changed for media.
From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Erick
Wellnitz
Sent: Thursday, April 02, 2015 6:50 PM
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] Expressway CE issue
Not a security or a
Hi Sam,
Do you need to transcode on the Cisco? (i.e. aren’t both endpoints supporting
the same set of codecs?)
You need to ensure, that if you use a codec list on one call-leg (dial-peer)
you support the same on the other call-leg (or outbound dial-peer) as well.
Trying transparent codec on
