the last thing i was trying to do was change the default password to same as voicemail. i also tried reversing these changes but doesnt work. this is my log. i should probably mention that im running trixbox 1.21. when i connect to the voicemail system remotely, i enter the username, then a
easy enough. thanks!Marco Mouta [EMAIL PROTECTED] wrote: just set your initial password to be equal to vm-account number, and Voicemail application will do that for you and will request users to setup a new password! On 10/9/06, stan ford [EMAIL PROTECTED] wrote:how does one force mandatory
/06, stan ford [EMAIL PROTECTED] wrote:how does one force mandatory password change on login? and a period of time to pass before mandating a password change?im using trixbox so if you have that info. that would be good, if you have it for
asterisk, im sure i could figure it out as well
how does one force mandatory password change on login? and a period of time to pass before mandating a password change?im using trixbox so if you have that info. that would be good, if you have it for asterisk, im sure i could figure it out as wellthx.
Yahoo! Messenger with Voice. Make
what about the interval of the registration? is 2 minutes too often? Dovid B [EMAIL PROTECTED] wrote: Timed out from what I have seen comes from either a poor internet connection or a problem with your ITSP.- Original Message - From: stan ford To: asterisk-users
anyone have experience with IntuitiveVoice's Asterisk system?
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i have this issue with failed registrations with my sip provider. it doesn't happen often, but it does happen. it also happens with 2 different vsp providers, so i dont think its them. this happens maybe 8 times a day, but then that doesn't sound too bad considering it registers itself every 2
stan ford [EMAIL PROTECTED] wrote:i have this issue with failed registrations with my sip provider. it doesn't happen often, but it does happen. it also happens with 2 different vsp providers, so i dont think its them. this happens maybe 8 times a day, but then that doesn't sound too bad
I have to setup a pbx system for a company, can someone suggest a configuration. Currently their phone bill is 1600 a monthCurrenlty 27 phone lines1/2of the calls are long distanceI'd like the savings of a voip network, but also the reliability of a pstn/pri.
How low will
I'm confused with something, maybe someone can explain to me. if your currently on a pri and are considering moving over to VOIP, that means you would have to purchase a t1 or fractional t1 for a your voip connections. but then, voip connections aren't as reliable as PRI. so then you would
it is
that complete IP networks in terms of telecom now look inevitable. And whether you do it yourself or it is done for you - it is the way things, many expect, are going to be in the next 5 or so years. stan ford [EMAIL PROTECTED] wrote:I'm confused with something, maybe someone can explain to me. if your
no, i have to retain my fax numbers hmm. thanks anyways.John Kington [EMAIL PROTECTED] wrote: At 12:15 AM 9/28/2006 -0700, you wrote:i can't for the life of me find a pay as you go termination and origination service.there's garfachi, but they don't offer DID's in anywhere else other than CA. Any
On fonalities web page, i see they offer pstn failback as a feature of their asterisk package. i've also heard before of failing back to a pri line if your t1 voip line fails. my question is. in order to have pstn or pri failback, dont you basically have to have all the equipment there on standby,
are failover pri's generally cheaper that their active counterparts?thanks alot.Shawn Kelley [EMAIL PROTECTED] wrote: Stan,I agree with the comment below, we switched from analog lines to a PRI andit's not always as reliable as some people think. We are in a somewhat rurallocation and we have
Lacy, can you confirm what i was saying about SIP Phones. if i fail from my voip connection to my pri, would i need to swap out my SIP phones with another type of digital phone?Lacy Moore - Aspendora [EMAIL PROTECTED] wrote: a couple things, if you guys could clear up for me.A) If i
i can't for the life of me find a pay as you go termination and origination service.there's garfachi, but they don't offer DID's in anywhere else other than CA. Any suggestions? Thanks.
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if you have to setup an office of 100 users now. would you rather setup a sip trunk,a t1-pri, or even a t1? and why?thx
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Look for support by whatever operating system you plan on running.
Henry Devito wrote:
Hi guys I know this has been asked on the list before, but my hard
drive crashed and I lost all of the past posts, I need to know what
motherboard works ok for asterisk, I have no problems with the Dual
and
with Asterisk given the appropriate
phones? (Cisco?)
Thanks,
Stan
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Where do I buy one?
Stan
Matt Riddell wrote:
James H. Thompson wrote:
Link to Sipura Press Release
http://www.sipura.com/Documents/SipuraPressRelease007.pdf
I've put it up on the news page in HTML (just in case it takes anyone
else as long as it takes me to open a PDF file!)
The URL
I looking at for equipment here? How do I possibly provide 250
phone ports with the Digium 4 port pci cards? Wouldn't I need a ton of
them?
Thanks,
Stan
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Hello,
I have been trying to setup * with Broadvoice.
I am using Gentoo Linux, and * 1.0.0, and now CVS.
My current config looks like:
(sip.conf)
[general]
port=5060
context=sip_incoming
tos=lowdelay
notifymimetype=text/plain
allow=gsm
allow=ulaw
allow=alaw
canreinvite=no
nat=no
register =
This line was reprovisioned, and the password was changed.
Stan
Stan Brinkerhoff wrote:
Hello,
I have been trying to setup * with Broadvoice.
I am using Gentoo Linux, and * 1.0.0, and now CVS.
My current config looks like:
(sip.conf)
[general]
port=5060
context=sip_incoming
tos=lowdelay
I am running the latest asterisk CVS.
[EMAIL PROTECTED] /etc/asterisk # locate demo-thanks
/var/lib/asterisk/sounds/demo-thanks.gsm
This directory has 150+ files.
I changed the [demo] section in extensions to [incoming] to play with
this included definition:
[incoming]
;
; We start with what to
ID prefixes.
Obviously nobody has any idea about dial plans better than me here! (and
I'm a newbie).
On Thu, 2004-08-12 at 15:51, Stan Dard wrote:
Hi
I've 'inherited' an existing Asterisk with a number of users, and some
pstn connections through its Zaptel card.
I've recently set up
Hi
I've 'inherited' an existing Asterisk with a number of users, and some
pstn connections through its Zaptel card.
I've recently set up another Asterisk which has no direct pstn access
and I've connected the 2 systems with IAX. The original system has an
extension number range 1xxx and the
On Mon, Aug 09, 2004 at 11:52:51AM +0100, Nick Barnes wrote:
The reason I ask is that I installed a BRI system (Single Fritz! AVM card
using chan_CAPI) last week which refused to work - turned out that British
Telecom had provisioned the line as a point-to-point and not
point-to-multipoint as
On Wed, Apr 07, 2004 at 12:37:43PM +0100, Jon Fautley wrote:
Morning Asterikians,
I've just got my nice shiny quadBRI card, and it seems to be working
very well - except for one little issue - CallerID.
The card is currently connected to an ISDN2e line in P2P mode, and an S0
adapter on
A minor gripe with our current system (* + CP7960s) is calls answered by
one handset showing as missed on others. RFC3326 seems to answer this
problem but I see no support for it in the cisco phones or * (obviously
less of a problem with the latter being OSS). Are cisco likely to add this?
Do
On Mon, Mar 15, 2004 at 11:01:46PM -0500, Greg Boehnlein wrote:
On Tue, 16 Mar 2004, Dean Collins wrote:
Cisco have the terminals around the other way, this is a well known
problem, do a search and you'll find what you need to do.
Cheers,
Dean
Alright.. since I'm the one that
Is anyone using a 3com 3CNJPSE to power a 7960G?
I have a couple of 7960Gs and 3CNJPSEs but no combination appears to
work. Both phones work fine with a cisco power cube. I get a 47.6V
reading across pins 5 and 8 on the patch cable coming from the 3CNJPSE.
The network still works through the
On Tue, Feb 24, 2004 at 06:17:27PM +, Chris Lee wrote:
I have been told that when I plug my X100p into the line I get a 36 Khom
loop condition and this may be affecting my ADSL connection (it keeps
dropping the line).
It may be to do with impedance differences here in the UK, But I know
On Wed, Feb 11, 2004 at 04:34:46PM -, Jason Ross wrote:
This is slightly off topic so sorry for the intrusion.
I've got a couple of 7940 phones I'd like to put on Smartnet but I'm
looking for what I need to order, what it roughly costs and finally a
reseller in the UK who is easy to deal
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