If you just save the page (from the browser) it will have the entire
configuration asa continuous html file. NewSipuraUtil is good too (more as a
backup and restore sort of facility). Bothways no passwords are passed.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
Have you compared the disconnect tone on both units - after all, you say one
of them works fine.
On mine (default since I didn't change this part) it is :
[EMAIL PROTECTED],[EMAIL PROTECTED];4(.25/.25/1+2) on the PSTN page
In case you're wondering, it is not working here either (not US).I use
suggest , should i record all my ivr file in g723
format all . increase my bandwidth!
/Salaque
On 5/14/06, AR Tarzi [EMAIL PROTECTED] wrote:
Unless reinviting works, wouldn't that add up to what he's experiencing
?
client - asterisk - service provider.. makes that 180k each connection
so 4 of them
Subject: Re: [Asterisk-Users] Confused !
how to use reinvite in my asterisk setup ?
thanks
Salaque
On 5/14/06, AR Tarzi [EMAIL PROTECTED] wrote:
I'm not an authority
but why don't you get some g729 codecs (10 or so) and use g729 all around.
Not allowing for ADSL overheads you can calculate your
Unless reinviting works, wouldn't that add up to what he's experiencing ?
client - asterisk - service provider.. makes that 180k each connection
so 4 of them would give 800k or so.
What I can't understand is: if only g723 is allowed, and Asterisk only
allows it as passthrough, how's the
For a linux newb who needs to wake up ? How does one do this ?
Copy/create a app_wakeme.c in the source directory then compile asterisk ?
How do I call it in dialplan ?
- Original Message -
From: Michael Iedema [EMAIL PROTECTED]
To: Asterisk Users asterisk-users@lists.digium.com
Sent:
There are analog phones with support for showing an envelope
(or message) on receiving VMWI. DECT phones from Siemens and Panasonic do that
too (although they are digital and wireless but they interface with analog
lines). In fact, the Siemens phones have a function to tell how many messages
message -- From: "AR Tarzi" [EMAIL PROTECTED]
SellVoIP appears to follow a US dialplan. A US
numberis dialled as 1NXXNXX whereas an international (to the US)
numberis dialled as 011X.
Frankly, I didn't ask whether international numbers like
I'm and [EMAIL PROTECTED] user - been so now for almost a year.
Lately, I've upgraded to the latest greatest.. (which is built on 1.2.5)
and am unable to install oh323.
I've already asked over at the ([EMAIL PROTECTED]) Sourceforge forum but no one seems to
think it worth answering.
The
I've read all the messages here.. seems everyone's forgotten the original
problem.
Yes, the polycoms are very slow at rebooting when configuring through its
http server, not only that, but it's also inconvenient because it reboots
for every change (each page) which means you have to pass
Not GSM/DECT but GSM/Wifi phones are available - This is not a
recommendation, I don't like what I've seen.
try www.imate.com (to start with) .. they have at least three types of GSM
phones that do Wifi .. They run windows so there are several sip softwares
and one IAX software that work with
Trading desk environments are always recorded. This is for conflict
resolution and there is no advice to clients. It is only used if the client
claims are contrary to the trader's - therefore where a loss is concerned.
Rather than test the legality, it is meant to resolve matters before they
That's because the duration is counted from the time of dialling. billsec is
what you want if it's to calculate the duration the call was active.
To change what shows you need to change call-log.php in
/var/www/html/admin/cdr/
Instead of duration extract billsec - you can still label it
When I receive a call from fwd, I'd like to insert a prefix
prior to the caller ID - 1) to be able to look it up in a database
ofidentified numbers and 2) for the receiver to be able to dial it
back.
So what I need is to identify the DID and based on that,
insert the prefix.
Any pointers
SellVoIP appears to follow a US dialplan. A US numberis
dialled as 1NXXNXX whereas an international (to the US) numberis
dialled as 011X.
Frankly, I didn't ask whether international numbers like
Barbados where the code remains as 1 butare international (to the US) need
the 011 or can be
Thank you. works like a charm. I'm using [EMAIL PROTECTED] so I had to massage
it into AMP's structure.
Your example is actually the reverse of what I needed to do, but that's not
the issue.
AMP uses a macro to dial (syntax almost exactly the same).
I feel this should be documented somewhere
I know this shouldn't be the place to ask this, but I've just
tried to upgrade my IP600 with bootrom 2.6.2 and SIP 1.5.2 and I'm getting
intotrouble here (I chose not to go to the higher software levels since
there's a warningabout using"secure" links.. I am not trying to
change anything
yikes,
25 and 110 will allow mail - but please without the whole digest attached.
And wouldn't your question be more useful with a better subject field? now
no one will see me addressing a question I know an answer to.
___
--Bandwidth and
could you please tell how it interfaces with Asterisk? Could I receive calls
into Asterisk? send calls out?
I've just downloaded it and am searching (unsuccessfully) for these on
Gizmo's site/software.
- Original Message -
From: Kerry Garrison [EMAIL PROTECTED]
To: 'Asterisk Users
Ignoring SS7, why exactly are you setting up several boxes ? there are quad
E1 cards no ?
This is way out of my league, but I just want to understand.
- Original Message -
From: Jean-Michel Hiver [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
I had a problem with DTMF with DISA.. I am using a Sipura SPA
3000 for the line. I set the FXO port impedance (on the PSTN line tab) to 900 as
advised by others and it worked.
Having said that, I'm sure you will be using some other FXO
adapter.. Just thought I'd tell.
- Original
On a polycom 600 which is working perfectly otherwise, I am
unable to use DTMF with IVR or such - not even to dialout of a Sipura setup
elsewhere. Other phones (analogue connected to ATA) are accepted.
I suspectthe phone is not using rfc2833 but I don't know
how to specify that it should
By observation (or better said, I don't know why)
clue:719705 @fromiaxfwd ??
Your peer and user settings should be taken from www.freeworlddialup.comwhich
dictate that your user context should be [iaxfwd] NOT your
userID (or fwd number)
They also have settings related to authentication and
In AAH create a DID using the number 719705 and direct it to
ring wherever you wish it to (extension.. extension group etc.)
- Original Message -
From:
Cristian Paun
To: asterisk-users@lists.digium.com
Sent: Tuesday, November 15, 2005
21:08
Subject:
- Original Message -
From: Angelito Manansala [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Sunday, November 13, 2005 14:31
Subject: [Asterisk-Users] How to check how many G729 codec license
installed
Guys, is the any CLI commands or info files where you can check how
I've used a Nokia 32 unattended (remote) for the past year or so.
David Uzzell [EMAIL PROTECTED] wrote in message
news:[EMAIL PROTECTED]
| Ok I have a question. Seen it come and go around the mailling list for a
| while but never really seen an answer that seems to sort it out.
|
| What is
Nokia's 32 is just one
example. A bit pricey but reliable.
- Original Message -
From:
Germán
Micale
Newsgroups:
gmane.comp.telephony.pbx.asterisk.user
Sent: Monday, January 24, 2005
22:20
Subject: RE: Mobile Callings
Thank you Andrew,The network is
I use Nokia 32s.
I don't know what a fritz
card is but they can act either as an FXO or as an FXS device. Beware though, if
you use them off a port that expects a telephone set (like an ATA or so) you'll
need a special cable to program the 32 properly - the cable is
pricey.Acting asanFXO,
Important.
1. Try the phone (set)
directly on the line.. - confirm you have dialtone
2. Make sure the phone is
picking up the line from pins 3 4 on the RJ11 ONLY .. i.e. if your line is
using a non-standard interface (and so does your phone) this is a possible
failure - not of the card,
al Message -
From:
Aaron Clauson
To: [EMAIL PROTECTED]
Sent: Monday, May 17, 2004 00:17
Subject: Re: [Asterisk-Users] X100P
Ireland Red Alarm (AR Tarzi)
Ahhh this could be my problem! I just checked whichwires on
the RJ11 cable had a voltage across them andit was
Could someone find the
time to tell me whether ALL functions in Asterisk are programmed using scripts
and contexts ?
What I need to find out
is whether the userCAN configure services for themselves.. Below, I chose
a sampling of a fresh question and answer (just as an example).
In this
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