On Wed, 2004-07-21 at 11:48, Yiannis Costopoulos wrote:
Hi,
I know that this issue has been discused guite a lot, but I haven't managed
to get a definite answer. Is those two values supposed to be floats (e.g.
3.5) or integers with the percent symbol (e.g. 20%)?
It's on the Wiki:
On Fri, 2004-07-16 at 13:28, Eric Wieling wrote:
I'm having an oddball issue with a Polycom SoundPoint IP 500. As you
can see below Asterisk thinks there are 2 SIP channels active, but show
channels tells me there are no calls active. Anyone have any idea why
this is happening? The
On Thu, 2004-07-15 at 14:13, Joshua McClintock wrote:
I have an asterisk box setup with a T1 card (hooked to a pri from the
telco). Sometimes when people call a number that rings down that pri,
the first ring is really really long, like 3 normal rings put together.
My indications.conf is
On Wed, 2004-07-14 at 12:15, Areski wrote:
On Wed, 2004-07-14 at 16:47, Joseph wrote:
On Wed, 2004-07-14 at 10:01, Kanuri, Seshu wrote:
Hi All,
The CDR Tool in .PHP is working great. We have put this into production.
Here is the Link: http://67.109.153.236/asterisk-stat/cdr.php
On Tue, 2004-07-13 at 03:54, Holger Schurig wrote:
Also, you can use the callgroup feature in sip.conf
[111]
...
callgroup=1
callerid=Member 112345
[112]
...
callgroup=1
callerid=Member 212345
[113]
...
callgroup=1
callerid=Member 312345
then in your dialplan
On Fri, 2004-07-02 at 15:49, Bryan Brannigan wrote:
I would like to setup 2 Asterisk boxes. One would be located in our
office behind the firewall and hooked up to our analog lines. The other
would be located in a remote datacenter and used for our remote employees
to connect to. I would
On Fri, 2004-07-02 at 16:22, Bryan Brannigan wrote:
Depending on what you are planning to do in the datacenter you could
just put SIP phones/ATAs there rather than a full Asterisk server but
that would require some care in configuring your firewall.
Actually the users are will be remote
On Fri, 2004-07-02 at 18:17, Stephen J. Wilcox wrote:
A trace to the IP gives valuenet : http://www.valuenet.net/ a colo provider on
Level3.
I have a dislike for this kind of targeted spam on mailing lists, and are they
harvesting email addresses from their subscription .. I suggest nobody
On Fri, 2004-06-25 at 14:43, Mike Roberts wrote:
I'm having troubles...
So am I. Hang in there. Stay positive. Whatever you do, don't jump.
I am new to Asterisk and SIP. I was just given
this setup and it was running fine. And somehow it stopped. I thought it
was the DID(again) But it
On Thu, 2004-06-17 at 09:23, Troy Settle wrote:
A: Because we read the question in the previous message.
Q: Why should I post my reply above the quoted text?
You are assuming that everyone subscribed to the list is reading you
particular thread. If they're not, but are mostly just skimming
On Mon, 2004-06-14 at 23:03, twisted wrote:
[snip]
And now, for the Asterisk-Users dial plan:
[snip]
Before hanging up, there should be an extension reminding everyone that
top posting is super duper wrong and oh so annoying.
I used to top post, but now I understand why it's frowned upon. It
Hi Kyle,
On Fri, 2004-05-28 at 12:32, Kyle Hagan wrote:
If your interested please let me know. Im gonna be putting up a site for
downloading if there is enough interest.
I'd love to help test it out as well. Giddy up!
Kanwar
Systems Aligned Inc.
www.systemsaligned.com
Hi,
On Tue, 2004-05-18 at 06:49, Robinson Tim-W10277 wrote:
Some pointers to where in the source code we should
be looking would be great. We can then make the
tweaks and feed the changes back in to CVS if anyone
else is interested in this feature.
I believe that even if there isn't a lot
Hello all,
I've noticed several messages about the latest firmware on Snom's site,
2.05b, and today I see that another update is listed, 2.05c. However,
when I go to the download page (http://www.snom.com/support_dl_en.php),
the latest firmware version available for the Snom200 is 2.04g.
Are
Hi everyone,
On Thu, 2004-05-13 at 18:36, M3 Freak wrote:
The problem is when I test my setup with asterisk -vvvgc, asterisk
pukes and reports the following:
[chan_zap.so] = (Zapata Telephony)
== Parsing '/etc/asterisk/zapata.conf': Found
May 13 17:19:49 DEBUG[-1220444032]: chan_zap.c
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