, but its not
clear how. Anyone got a 9971 working with 1.9.2-2SR1-9 want to share their
SEPxxx.cnf.xml?
I'd like to regress to SIP 1.8, as I think that may fix the problem, but so
far I haven't been able to locate, ahem, a copy.
Cheers
Patrick
On 4 May 2013 18:56, Patrick Lidstone patr...@lidstone.net
with more q's
Cheers
Patrick
On 5 May 2013 16:17, Patrick Lidstone patr...@lidstone.net wrote:
Getting closer...
As a heads-up, the files in the link do not include the locale information
(gd-sip.jar), but I have tracked down something suitable for that...
The phones now get all the files
I'm an asterisk hobbyist, and I've got my hands on some cisco 9971's
preloaded with SIP (I think they might only come in SIP flavour actually?).
I am quite excited about the possibilities with this kit - especially video
calls. Unlike the earlier Cisco phones (e.g. 79 series), these can't be
used
Stoyan Marinov wrote:
Checkout http://firewall.cx for cisco downloads
Looks promising - a later firmware load, so the file I was looking for was
not present, but still hopeful!
Many thanks for the tip,
Patrick
--
_
--
We're about ready to go ahead with a nice 6 line (maybe later
8) ISDN setup with [EMAIL PROTECTED] and the quad Junghanns card.
Before we do, could anyone confirm for me that BT's ISDN2e
lines do actually provide Asterisk with the DDI number? We
need to be able to route incoming calls
I want to be able to dial a 'pager' extension from an phone on my
asterisk server, and have it ring all other extensions *except* the
extension from which I am calling (because call waiting is enabled on
most extensions by default) - effectively giving me the ability to
page all other
I have a second-hand 7960 which I am attempting to upgrade to use a SIP
image.
The phone currently has a firmware release which doesn't seem to be listed
in Cisco docs - P003AM30. On reboot, it finds the tftp server and requests
the firmware image listed in OX79XX.txt correctly, displaying
I have a second-hand 7960 which I am attempting to upgrade to
use a SIP
image.
The phone currently has a firmware release which doesn't seem
to be listed
in Cisco docs - P003AM30. On reboot, it finds the tftp server
Here's how I performed the upgrade:
Downgrade from the
Make sure that you have done the following:
1.) Set up the phone to use DHCP to get an address *or* manually
configured an e-mail address using the settings on the phone.
2.) Set the DHCP server to give out the correct TFTP server address,
*or* configure Alternate TFTP Server = yes and
Hello,
I would like to implement a home GSM gateway using asterisk. What
would you recommend me as a low-cost hardware for creating a gsm
channel? I found 2n gsm gateway, that supports sip and chan_blue for
bluetooth connections. Any recommendations?
Basically, I want to end calls to
Message: 5
Date: Wed, 27 Apr 2005 12:04:30 +0100
From: Johan Akerstrom [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Good FXO for UK use.
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID:
[EMAIL PROTECTED]
Content-Type:
Date: Tue, 26 Apr 2005 12:36:26 +0100
From: Razza [EMAIL PROTECTED]
Subject: [Asterisk-Users] Good FXO for UK use.
To: asterisk-users@lists.digium.com
Message-ID:
!~!UENERkVCMDkAAQACABgAT1LIZaT+QEqJAR
K3kpBGu8KQMG2M/[EMAIL PROTECTED]
Hi there
Just a general question, has anybody experienced any problems
with any Digium telephony cards in the UK, specifically with
BT (British Telecom) lines. I just want to make sure there
are no compatibility issues before purchasing cards, (mainly
TDM400P's)
Any comments would
When a caller hangs up (e.g. after leaving a voicemail), my British Telecom
exchange sends a continuous tone for about 15s and then silence. I can't get
asterisk to recognise this tone as a hangup indication.
I have tried indications.conf with both country=uk and country=us.
My zapata.conf has
Kelly Griffin [EMAIL PROTECTED] wrote:
Depending on what you have the context of this setup for in your
zapata.conf, don't have the following statement in that
section of your
extensions.conf.
exten = s,1,Answer
or
whatever the CID of this FXO port is as in
exten =
Chaps,
I recently added an incoming VOIP account to my asterisk box. When the
PSTN number associated with this account is dialled, the call rings once
and then asterisk starts playing music on hold, even though all the
extensions continue to ring. Variations of answer() and ringing() don't
seem to
Please excuse me if this is a niaive question...
I have Cisco 7940 (but same applies to Snom's too), and it would be
convenient to have multiple extensions on the same phone registered
against the same asterisk instance. (E.g. one extension which is
associated with work, one extension which is
Does anyone know the procedure for adding a serial output to a cheap
caller
display unit. If I can find a way of doing this then I'm sure there
will be
away for linux to take the CallerID info, write it to a file, * to
open that
file an read the number from it.
Sorry I never got round to
First, is the lack of UK CLI on the x100P hardware or
software related?
Don't know. AIUI the primary difference between BT and Bell caller
id standards is that BT requires:
- Hardware to detect line reversal prior to first ring indicating
imminent arrival of the CID data
- And then to detect
My ISDN phone is able to display No. Suppressed and No. Unavailable,
depending on whether a caller wittheld their identity or is simply out
of area. I'd like to be able to make the same distinction with asterisk,
ideally within extensions.conf. I've had a look at the chan_capi source
code, but am
I am struggling getting asterisk to work on my firewall box.
The Linux box is a firewall running Mandrake 9.2 and
shorewall for security and NAT. Asterisk is compiled and
running on the firewall box with a modified sample
configuration. I am connecting to it using a Sipura on the
From: Iain Stevenson [EMAIL PROTECTED]
Anyone else seeing SIP registration requests rejected by FWD?
I don't seem
to be able to register any longer - even though my SIP config
remains the
same.
Iain
Yes, me too - for about the last week I'd guess. I'm guessing that it is
this
I'm testing * (CVS-09/16/03-02:07:49 with zaprtc 0.0.1) with Fritz!PCI
(chan_capi 0.3.0), and have a couple of funny things - I wonder if
anyone else has seen them:
- Now and then, * just exits. Until now I had lowish-level
verbosity on,
so all I saw was 'Executing last minute
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