We had a situation where the 601 base went missing and the electrical
connection between the side cars and the 601 was broke. Might be worth a
look to see if the phone got damaged.
-Original Message-
From: Jerry Jones [EMAIL PROTECTED]
To: Asterisk Users Mailing List -
I believe asterisk requires the source/devel package. I'm afraid I'm not
going to be much help on that. We use Gentoo and it was a simple 'emerge
curl' and I was done.
-Original Message-
From: Damon Estep [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
-Original Message-
From: Damon Estep [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Date: Fri, 17 Nov 2006 13:32:53 -0700
Subject: [asterisk-users] wget from within asterisk?
What would be the simplest way to retrieve information form
a CNAM database that provides http
Make sure the curl library/package is installed, then re-compile asterisk.
We're using it on 1.2.
-Original Message-
From: Damon Estep [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Date: Fri, 17 Nov 2006 13:40:40 -0700
They both seem to work, but the Curl spits out warnings about being
deprecated. Ours are all configured using CURL.
-Original Message-
From: Damon Estep [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Date: Fri, 17 Nov
The Curl/CURL is an asterisk dialplan distinction.
-Original Message-
From: Damon Estep [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Date: Fri, 17 Nov 2006 15:06:49 -0700
Subject: RE: [asterisk-users] wget from within
I can confirm that the linksys routers cause ftp
problems. Is your FTP server set to use pasive mode?
-rb -Original
Message- From: "Curt Shaffer" [EMAIL PROTECTED] To:
"'Asterisk Users Mailing List - Non-Commercial Discussion'"
asterisk-users@lists.digium.com Date: Mon, 6 Nov 2006
ve. So I’m
guessing that its not. After you asked this I have googled passive FTP and
it seems to be on the money as to what is going on so I will try passive and
see if that helps. Thanks!
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Russ Beaupre Sent: Monday, Novemb
. Is there something in the
config that deals with NAT traversal with regards to how it is
provisioned?
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Russ Beaupre Sent: Monday, November 06, 2006 8:26
PM To: Asterisk Users Mailing List - Non-Commercial
Discussion
Su
Good suggestion. It now seems to roam between access points nicely, even
while a call is in progress.
What access pooints are you using?
-rb
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Kanuri, Seshu (Company IT) wrote:
Does anyone know how Polycom 500s will be able to update their time. My
setup for a time sync with Public domain Time servers is not successful.
Seshu
We had a user with a Sonic Wall Firewall who needed to set the snpt
server to the IP address of his
Andrei (MPI) wrote:
Hi Jared,
Thank you for your reply. That server is for asterisk only, things like
X-windows and samba were not even installed there. I limited what I
could from system point of view. Digium support has qualified the box as
clean for TDM400P operations.
It is not clicks and
Steven Wang wrote:
Hello
I try to set up voicemails for extension. When VoicemailMain gets called, it
prompts for mailbox and password. It seems not able to read from the phone.
So the authentication always fails.
I desparately need help to understand what is wrong. Here is a part of my
Matthew Boehm wrote:
Can someone else confirm that your phone does not recieve MWIs when using
SIP and RealTime?
I can confirm that. Polycoms and CVS-Head from a month or so ago. I
haven't tried it since.
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[EMAIL
Hi, everyone:
I was playing with the ODBC configuration to pull sip and voicemail
config info from a MSSQL2000 server. Everything works great except for
the message waiting indication on the Polycom phones (all three models).
If I move the sip registration info to sip.conf, the MWI starts
Wiley E. Siler wrote:
I have a solution that allows me to assign a soft key with no problems.
However, it seems like a waste the the hard button labeled Voice Mail is
not dialing right into voice mail. Is there a known way yo do this? I
have tried everything in the manual but it doesn't seem to
Wiley E. Siler wrote:
Hello All,
I have some Polycom IP 500 phones that I would like to have configured
for direct dialing to our voice mail system. So far I have been unable
to get the hard button labeled Voice Mail to connect to Asterisk without
first passing through the message center
Jorge Mendoza wrote:
Hi,
I'm testing a Polycom IP600.
With firmware version 1.1 the phone reboots at any time.
With firmware version 1.2, the first reboot was an endless reboot. Then
I moved the phone to another lan port, then it worked fine. Then I
installed again in the initial lan port and
Tomica Crnek wrote:
Hi everyone,
I have to test few models of SIP IP phones with Asterisk. I have seen on
voip-info.org that there are lot of phones that work ok with Asterisk.
But, I want to ask for suggestion - which models are the best for Asterisk?
We have a dozen or so Polycom IP 500's
Matthew John Darnell wrote:
Aloha,
Does anyone have a good source for Polycom SoundPoint® IP 600/500/300
phones?
http://www.reviewvideo.com/ Usually has them in stock. I've had good
luck ordering from them.
-rb
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Brian D'Arcy wrote:
Hello all,
Im having a nightmare of a time trying to get stable results with SIP
clients on Asterisk. I cant seem to find a configuration that works!
In our office, we run a Sonicwall Pro 200, which is a sip aware,
stateful firewall.
We've discovered that certain
It was my understanding that the SIP version is not available until May or June. My
IP600's work great , though...
-- Original Message --
From: Shad Mortazavi [EMAIL PROTECTED]
Reply-To: [EMAIL PROTECTED]
Date: Fri, 16 Apr 2004 07:07:45 -0400
Dear
Hi, all
We're using Asterisk CVS 3-19-04 with four polycom IP600s. The work
very well and we're quite happy with them. They register fine and all
four are able to place and receive calls, BUT two of them are behind NAT
routers and when they place a call on hold, the call is dropped within 5
router/voice server (using only IP voice trunks) we've been having some
issues (which are being worked on...Thanks Mark)
Russ Beaupre, P.E.
BoTech Communications Corp.
- Original Message -
From: Ray Burkholder
To: [EMAIL PROTECTED]
Sent: Wednesday, October 29, 2003 9:01 AM
Subject
I had a similar problem when using an AS5300 with SIP. I'm not familiar
with the 7940, but on the AS5300 there is a Voice-Activity-Detection
setting. Disabling the VAD on the AS5300 corrected the SIP music on hold
problem.
Russ Beaupre, P.E.
- Original Message -
From: Babak Pasdar
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