to it!
Regards
Garry Taylor
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anyone have this data?
Regards
Garry Taylor
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-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Ronald Wiplinger
Sent: Monday, 29 November 2004 6:35 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Dial plan for TDM22B
I finally got the TDM22B to
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Daniel Chester
Sent: Monday, 29 November 2004 2:28 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] D-LINK PoE switch, does it work
with cisco or do I need
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Nick Bachmann
Sent: Saturday, 27 November 2004 2:02 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Is Busydetect obsolete in the
latest CVS?
Garry
a warning
Ignoring signalling is singnalling=fxs_ks obsolete also?
Regards
Garry Taylor
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sense to
configure a telephone for fxo_ks? Or should it be configured for fxo_ls?
Regards
Garry Taylor
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Hey thanks Greg :)
Did I spell signalling wrong also? Doesn't look like it, but the error
message is odd, don't you think?
Regards
Garry Taylor
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Greg Blakely
Sent: Saturday, 27 November 2004 12:54 PM
]: chan_zap.c:3463 zt_handle_event: Didn't
finish Caller-ID spill. Cancelling.
-- Zap/2-1 is ringing
-- Zap/2-1 is ringing
And then I will have this PBX ready for my customer.
Regards
Garry Taylor
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf
Hi,
Has anyone seen this problem -
Nov 25 18:09:14 WARNING[12923]: chan_zap.c:3463 zt_handle_event: Didn't
finish Caller-ID spill. Cancelling.
I started to get this message after upgrading from 1.0.2 stable to the
latest CVS.
Hope someone can help me out here.
Regards
Garry Taylor
both number and name, and I am using
the same config on both PBX, except for the group statement and the channel
statement is 1 of course.
Regards
Garry Taylor
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I would like to see the answer to this also, I have experienced the same
problem but not had much time to look at it.
regards
Garry Taylor
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Matthew Boehm
Sent: Tuesday, 23 November 2004 5:46 AM
Cisco 79xx phones are NOT 802.3af compliant or even compatible. If you have
a mid-span 802.3af injector, this can work with the phone, provide you
follow the instructions at -
http://www.voip-info.org/tiki-index.php?page=Cisco%20POE
If you have an end-span injector, such as 3-com switch forget
forget to flame me if I am wrong!
Regards
Garry Taylor
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What about IPEYA? Do they have a commercial Asterisk license, or are they
breaking GPL also?
http://www.ipeya.com/SOHO_Portal.html
Regards
Garry Taylor
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Matt Riddell
Sent: Friday, 12 November 2004 5
about IPEYA
-Original Message-
From Garry Taylor
Sent: Friday, November 12, 2004 5:55 AM
What about IPEYA? Do they have a commercial Asterisk
license, or are
they breaking GPL also? http://www.ipeya.com/SOHO_Portal.html
Regards
Garry Taylor
Have you ever had
It is not grandstream, it comes from Integrated Networks China. Take a look
at this ---
http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItemcategory=23777item=5729980269
rd=1
:(
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Steve Totaro
Sent: Tuesday,
is local
to * and IAX phone is remote to asterisk. Anyone have any ideas? Could this
be a jitter problem, I am not using a jitter buffer in the iax.conf. If I
test with two IAX phones both running iLBC the voice quality is fine in both
directions.
Regards
Garry Taylor
On Sun, 31 Oct 2004, Garry Taylor wrote:
Hi All,
I was doing some testing between on extension running SIP
at G.711alaw
and an IAX extension runing iLBC (also GSM) and found that
the voice
from the IAX user has a lot of packet loss (very bad voice quality)
toward the SIP phone
Open UDP port 4569 in your NAT and point it to the IP address of your *.
Works great for me!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Eric Wieling
Sent: Thursday, 21 October 2004 10:25 PM
To: Asterisk Users Mailing List - Non-Commercial
Title: Message
ITG is
Internet Telephony Gateway, too bad that itdon't support
SIP
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Luis
CzopSent: Thursday, 21 October 2004 10:30 PMTo:
'[EMAIL PROTECTED]'Subject: [Asterisk-Users]
...
Thanks for all.
Garry Taylor wrote:
Open UDP port 4569 in your NAT and point it to the IP
address of your
*. Works great for me!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Eric Wieling
Sent: Thursday, 21 October 2004 10:25 PM
Hi,
How to set fwd iax as a peer? The config that I got from FWD does not show
how to do this?
Also how to get the status from unmonitored to what you have?
Appreaciate any help on this.
Regards
Garry Taylor
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED
Hi,
I am attempting to put together country tones for indications.conf and
zonedata.c, and hope someone can help me.
1. Are these two files the only ones that have the country tone/indications?
2. How to get my country tones included into zonedata.c, who would I send
them to for inclusion?
3. Can
Incoming calls without calling id go the the s extension.
extensions.conf
[incoming]
exten = s,1,Hangup
zapata.conf
context=incoming
channel = 1
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Alex van Es
Sent: Thursday, 21 October 2004 1:42 AM
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