My environment is a standard Cisco Call Mangler VoIP solution that has
reached the capacity of the Unity VM system. The cost of another Unity
box is enough to prompt the decision makers to look for other solutions.
All the calls come into a Cisco 6509 and then to the CM.
The more I look at th
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On 5/26/05, Scott Herrick <[EMAIL PROTECTED]> wrote:
BUMP
It's CM 3.3.6
MAN that would be sweet if * could take the place of Unity!
Anybody?
:-)
I've got it working with Callmanager 3.3(5) and Asterisk (connected
with chan_oh
Shawn,
Thanks for taking the time to put this together.
I have the open h323 gatekeeper running on my * box and have
connectivity to the CM. I did it using Device > Trunk in CM. I was
looking at the Gateway config and did not understand how it would ever
know the IP address of the * and o
BUMP
It's CM 3.3.6
MAN that would be sweet if * could take the place of Unity!
Anybody?
:-)
[EMAIL PROTECTED] wrote:
Has anybody successfully (or I guess unsuccessfully for that matter)
implemented Cisco Call Manager and used an * box for voicemail? I
checked the wiki and google and I see
I have several of the IP-500's and will be getting the 501's in the future.
If anyone out there has a 501 in production, please check the Status
value in the "SIP SHOW PEERS" command. My polycoms are all over 95 ms
even on a LAN!
I'm hoping they did a little more than just add flash and http
James,
What kind of DHCP server are you using? I have seen more problems with
the DHCP server than with the phones/clients. If you're getting the
whole DHCP DORA exchange it might be a messed up lease value. Try a
different DHCP server and/or change the DHCP values one by one.
Good Luck
Patrick,
I would try to get * to talk SCCP(Skinny). The Cisco phones are either
going to talk SIP or SCCP but I'm not aware that they will talk both.
Take a look at http://chan-sccp.sourceforge.net/
Good Luck
Scott
Patrick Zwahlen wrote:
Hi,
I have the following config:
[7960] <--skinny-
CTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Scott Herrick
Sent: Saturday, April 30, 2005 8:49 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Polycom IP500 Forward problem codec issue
Polycom IP500 Forward problem codec issue
All,
I’m running the Polycom IP500 phones at several sites.
Polycom IP500 Forward problem codec issue
All,
I’m running the Polycom IP500 phones at several sites. My * server is
at a collocation site and I have complete control of the T1’s running to
the remote sites with the IP500 phones. Connectivity to the PSTN is
through a Cisco 2600 with a PRI car
KUJ,
Thanks for the help.
I'm still having some trouble and have a few questions.
1) When defining the bmp file name...
2) Where is the custom bmp file to be placed? The ftp folder that the phone is using to pull the configs or from a web server?
3)Did you have to change the microbrowser sect
Mark,
I have heard this problem. I'm not exactly sure what the cause is but
check for any duplex mismatches between the phone and the * box.
Hope this helps.
Scott H
Mark Benson wrote:
Yesterday I setup music on hold by downloading and installing mpg123 r
Now I have music on hold but it sounds t
I have several Polycom IP-500’s and a few of the Cisco 7960’s connected
to an Asterisk test box. When I add qualify=yes to the sip.conf and
then enter “sip show peers” on the console I get, on average, 85 ms for
the Polycom phones while the Cisco phones are half that. This is on a
LAN. Acr
All,
Has anyone got some example config files for the Polycom IP 500 SIP
phones?Specifically the sip.cfg, ipmid.cfg .cfg and any
others that are needed to get the phones registered with *. I have a
few of the Cisco 7960 phones working but the documentation and examples
for the polycoms is
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