Hi,
Trying to set up these two buttons on a snom 360. The message waiting
key seems to send a call to it's own number, which is obviously engaged
and where you are prompted to leave another message to yourself, and the
conference key seems to do nothing.
Anyone manged to overcome these
This is rather weird? What network do you receive this from? Neither
ITU q.931 nor ETSI EN 300 403-1 (EiroISDN definition) lists the Calling
Number IE among those that may be repeated.
q.931 (and q.931e) traces include both called number and calling number,
on all Uk variants. Due to the
Guys,
Anyone know if the default [EMAIL PROTECTED] install supports TDM400P cards at
all (Digium Fxo/Fxs port card) ??
Thx
Paul
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To
you'll have to get them off Cisco's site, there was a posting recently
stating where to obtain these without a cisco login.
Anyone care to remind us of where this was??? (did a quick search and didn't
see anything immediately).
Since we are outside the US, Cisco US refuse to 'sell' us a login
Got fed up going round in circles in the end. all for $8 worth of
access :(
Technically, Cisco wants you to pay for those images :)
Indeed, and I would if Cisco made it Technically possible! :)
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Add an 'l' on the end of the link... i.e. 081534.html
Then it'll work :)
Paul
-Original Message-
From: [EMAIL PROTECTED]
[mailto:asterisk-users-[EMAIL PROTECTED] On Behalf Of Vitalie
Apostu
Sent: 11 January 2005 16:12
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Ben,
Count me in... Novice though I am (and learning fast with the obligatory
vertical learning curve...)
Paul
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ben Merrills
Sent: 10 January 2005 13:58
To: Asterisk Users Mailing List -
I would disagree, purely because I'm getting the same message on an xp2100,
with just OS and asterisk running - and that's with approx 98% free time
Paul
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paradise Dove
Sent: 10 January 2005 15:18
To:
Title: Message
Phil,
http://www.voip-info.org/wiki-Asterisk
is a good place to start, and will point you to most resources.
Paul
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Phil Menico
Sent: 10 January 2005 15:20
To:
Asterisk-Users@lists.digium.com
Yup, just to confirm, same here...
Anyone know of a fix?
Paul
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Humberto
Aicardi
Sent: 06 January 2005 11:00
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RES: [Asterisk-Users]
http://www.thirdlane.com/screenshots.htm
(Asterisk PBX Manager from Thirdlane) looks like a
great program for eye candy configuration of
Asterisk.
However it costs lost of $, and Im currently only an experimenter
so to speak.
Anyone advice of a decent alternative that is similar??
http://www.thirdlane.com/screenshots.htm (Asterisk PBX Manager from
Thirdlane) looks like a great program for eye candy configuration of
Asterisk.
However it costs lost of $, and I'm currently only an experimenter so to
speak.
Anyone advice of a decent alternative that is similar?? Currently,
- apologies to those that dont like HTML mail!!)
In article [EMAIL PROTECTED],
Paul Brock [EMAIL PROTECTED] wrote:
Finally, Anyone know of a Digium hardware Reseller in the Uk at all??
www.telappliant.com
Cheers
Tony
--
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play
Hi John,
I've been using this service for a while now, works very well.
Had a couple of minor problems with IAX originally, but they were sorted
within a couple of hours, and they even called me to tell me they'd fixed
it.
If you use SIP with them though, you may find weirdness with incoming
Trying now to set up the final part of my * switch. I must admit I've had
great fun over the last week or so playing with it, and would like to thank
you guys for all the assistance (past and present, since I've been trawling
a lot of old posts!!!).
Scenario - using voiptalk.org to supply the
Using IAX as recommended by most - and therefore my IAX config goes
somewhere along these lines:
[general]
bindport=4569
bindaddr=192.168.1.150
language=en
bandwidth=low
[voiptalk]
type=peer
username=username
secret=password
host=iax.voiptalk.org
qualify=yes
[08450number]
type=friend
That actually tells why it doesn't work. :-) It can't find anything in
[from-sip] that matches the number you are trying to call.
You shouldn't put nat=yes in your sip.conf - * can see the Cisco's
directly, so no need to do that.
It seems you haven't included the [voiptalk] context in the
Discussion
Subject: Re: [Asterisk-Users] Troubleshooting Asterisk
On Fri, 17 Dec 2004 17:10:18 -
Paul Brock [EMAIL PROTECTED] wrote:
Dec 17 17:03:03 NOTICE[9786]: rtp.c:1193 ast_rtp_raw_write: RTP
Transmission
error to ipaddr:17664: Network is unreachable
==Spawn extension (local, 2001
Title: RE: [Asterisk-Users] Troubleshooting Asterisk
Great :-)
If you use context=from-sip in sip.conf, you should include the [voiptalk] context into your [from-sip] context. (in the extension.conf)
eg.
[from-sip]
include = 2001
include = 2002
include = voiptalk
This way the Cisco's
How are your Cisco's connected to *, and do they register them self on the
* if you reboot them?
sip show peers
Name/username HostDyn Nat ACL Mask
Port Status
2002/2002 192.168.1.152 N 255.255.255.255
5060 Always
2001/2001
Randy,
If this is the case, you might need this :
http://www.voip-info.org/wiki-Firmware+issues+on+7940+-+7960
Might be worth a go if you suspect it to be the problem...
Paul
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Randy MacKay
Sent: 16
Guys,
Ok nowhere near as complex as most of the
discussions on here ( ex telco engr for 18 years here). But thought Id
ask for some assistance.
Have just set up my first * Pbx having a play with
it and a couple of Cisco 7960 (configured as SIP) phones.
The phones are tftping
I asume you can't place calls from the Cisco's... you need a context in the
extension.conf for them. In sip.conf you tell them to use the [from-sip]
context - that context should be in the extension.conf eg.:
extension.conf:
[from-sip]
include = 2001
include = 2002
This allows the Cisco's to
Many thanks - have added this, but strangely enough it still doesn't work
phone-phone :(
/me continues to play
Could you post the output from the CLI (with verbose level at 4 or so) it
might give up some clues.
Certainly - and many thanks.
Not a problem - what info would be
No problem at all :-)
Just the output it makes as you try to call from one Cisco to the other.
Stupid question on my part, but how do you specify a level to debug at when
issuing the debug command???
Currently I'm running a debug, but I suspect it's at too high/low a level,
since I'm not
Gents,
Just a passing thought... is there any reason why the ability to search the
past posts on here isn't switched on?
Just wondered, since it makes much more sense to be able to search the old
archives if you have a problem, rather than ask the same question again and
again...
Paul
Just connect to asterisk with
asterisk -rv
It should produce some good output.
Hmm.. interesting.
If I try to connect phone - phone, I get no (yes I know.. NO) debug info at
all, which would suggest that the call isn't actually leaving the phone /
talking to the * switch
Or maybe
Randy,
Is it a new unit? The only reason I ask is that hitting the settings button
should let you straight in.
There is an Rs232 port on the bottom - however not oversure what it's used
for on the 7960's.
The reason I as wether it's new or not is that it might need firmware
resetting as per the
Gents,
New to the list, and also new to Asterisk.
I was wondering - other than the official Asterisk Handbook(Draft 2) that
I have stumbled across, is there any other literature that you more
experienced asterisk users would recommend.
Also, are there any more resources with regards to
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