Did you look at logger.conf?
From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of bilal ghayyad [EMAIL
PROTECTED]
Sent: Sunday, September 16, 2007 5:21 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] stop log/debug messages into
I am getting the following message when trying to lookup up a number via
Dundi:
Feb 1 13:39:24 NOTICE[20146]: pbx_dundi.c:1309 update_key: No such key
'office.pbx.bluegrass.net.pub' for creating RSA encrypted shared key for
'00:a0:c9:55:91:89'!
I have created keys on each box with astgenkey -n
Anyone in Winnipeg Canada?
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The Grandstream ATA (480 I think...) does this and usually costs less
than the Sipura. It has 1 FXS and 1 FXO.
-Jonathan
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Damon Estep
Sent: Thursday, February 02, 2006 8:28 AM
To:
I have been planning to do the same thing but never got around to it, I
actually did write a nice class to wrap the interface to the manager but
it isn't complete.
Would you be willing to share your work?
-Jonathan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
It's something like exten = 15,1,Dial(Console/DSP)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anthony
Azzopardi
Sent: Saturday, February 04, 2006 2:10 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] How
Do people not use the Grandstream ATA's because they are cheap or
because there is actually a problem with them?
They have a 2 line version for around $50 that I have used in various
locations. I have about 8 or so. They seem to do an excellent job.
-Jonathan
-Original Message-
From:
BOFH told me he uses it to listen to his co-workers
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith
Sent: Thursday, February 09, 2006 12:27 PM
To: asterisk-users@lists.digium.com
Subject: SOLVED: Re: [Asterisk-Users]
This feature also works on the IP301 phones. The obvious caveat is that
it is one-way only. Still nice for an all-page though.
-Jonathan
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith
Sent: Thursday, February 09,
A demonstration:
exten = _X.,1,Set(GROUP()=${CALLERID(num))
exten = _X.,n,Set(CDR(AccountCode)=${CALLERID(num))
exten = _X.,n,GotoIf($[${GROUP_COUNT(${CALLERID(num))} 2]?103)
exten = _X.,n,Macro(trunk,${EXTEN},residential)
exten = _X.,n,Hangup
exten =
Why don't you just give the secretary the boss' REAL extension and give a
different extension to the world that just rings the secretary?
-jonathan
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Ricardo Carvalho
Sent: Friday,
If it's using RBS then 56k is the right number.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of JR Richardson
Sent: Saturday, January 27, 2007 12:55 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] max tnt pri voice
This is an interesting idea, did you come up with anything?
Are your users logging into an AD domain? A script to interact with the
Asterisk server could be run after login which adds an extension mapping the
user to the phone. One set of extensions for the users (which is published) and
I would be willing to mirror it also….
_
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro
Sent: Wednesday, March 14, 2007 9:39 AM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] What happend to
Abdul, it doesnt sound like you
need to do anything to the Quintum. I would recommend making your dial plan execute
the AGI script of your choice no matter what number is dialed from the context
where the quantum users land.
-Jonathan
From: [EMAIL PROTECTED]
[mailto:[EMAIL
Ive only used a Quintum a few
times,sorry.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Abdul
Sent: Friday, August 25, 2006 6:49
AM
To:
Asterisk-Users@lists.digium.com
Subject: RE: [asterisk-users]
quintum Calling Card
Hello Jonathan,
I tried in
Title: RE: [asterisk-users] Polycom 501 config questions
Dumb question here: Why the
need to dial 9 for an outside line? If your extensions are less than 7 digits
long then you know anything "XXX." is an outside call
Maybe this isn't true everywhere, just
curious.
-Jonathan
Works that way for me. IN SPA-841 for example, both lines are on the
same user/pass and the device registers once but line one rings and if I
answer it then get another call, line two rings.
-Jonathan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of jj
Wiley is definitely right. It would be dangerous not to have a firewall
for security reasons.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wiley
Siler
Sent: Wednesday, August 10, 2005 2:27 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Is it not for a card with 4 FXO? I spent several hours the other day
trying to figure out what I had done wrong and I ahd forgotten to
connect the power cable.
I setup several of these before and couldn't figure out why this one
didn't work. It appears that's all it waqs.
Without the power
We had this problem about 8-10 months ago and the end cause was IRQ
scheduling problems with the card.
We put it in a slot with a fixed IRQ and changed something else (sorry,
I don't remember what it was) to fix the IRQ problem and the error
seconds went away.
-Jonathan
-Original
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Jonathan k. Creasy
Sent: Wednesday, August 10, 2005 3:31 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Is it mandatory to give power
supply to TDM400Pcard
I didn't necessarily mean a separate firewall device, but I wouldn't put
a machine out there without a firewall either between it and the net or
on it (iptables for example)
As far as If I know what I am doing goes, I have not read the source
of everything that *is* required in my environment so
Ok, I just unplugged my power connector to a card with 4 FXO modules and
they no longer work.
Plug it back in and it works.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rich
Adamson
Sent: Thursday, August 11, 2005 12:05 AM
To: Asterisk Users
There is pfSense (based on monowall) which I like also. www.pfsense.com
-Jonathan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alex
Sent: Wednesday, August 10, 2005 11:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
supplytoTDM400Pcard
On Thursday 11 August 2005 09:30, Jonathan k. Creasy wrote:
Ok, I just unplugged my power connector to a card with 4 FXO modules
and
they no longer work.
You're *sure* you've got FXO modules and not FXS ones? FXO plug into
regular
phone lines, FXS plug into telephones...
Unless
YeahI think that every install I have done the first thing that
happens is why is there a delay before the call connects? and the
answer is you have to hit dial or wait 10 seconds.
-Jonathan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tom Rymes
I use the 300 and 301 models. Haven't used them extensively though. The
most common phone for me is the Sipura SPA-841.
-Jonathan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jeremy
Melanson
Sent: Thursday, August 11, 2005 4:05 PM
To: Asterisk Users
This poll by the company which made amp might be useful to you.
http://www.coalescentsystems.ca/index.php?option=com_polltask=resultsi
d=4
Distribution for your Asterisk/AMP system(s)?
Red Hat Enterprise, White Box, CentOS
48 41.4%
Debian
19 16.4%
Novell/SUSE
, with no luck. My software
phones work fine. In my asterisk log I get a error message like Failed
to authenticate user. Did you have this problem?
--Shaun
Jonathan k. Creasy wrote:
I use the 300 and 301 models. Haven't used them extensively though.
The
most common phone for me is the Sipura SPA-841
but using existing phone
Jonathan k. Creasy wrote:
YeahI think that every install I have done the first thing that
happens is why is there a delay before the call connects? and the
answer is you have to hit dial or wait 10 seconds.
What all phones does that apply to? I'm fairly certain
but using existing phone
Jonathan k. Creasy wrote:
YeahI think that every install I have done the first thing that
happens is why is there a delay before the call connects? and the
answer is you have to hit dial or wait 10 seconds.
What all phones does that apply to? I'm fairly certain
an old PBX but using existing phone
Jonathan k. Creasy wrote:
YeahI think that every install I have done the first thing that
happens is why is there a delay before the call connects? and the
answer is you have to hit dial or wait 10 seconds.
What all phones does that apply to? I'm
Check out pfSense.
www.pfSense.com
It has SIPProXD on it. This software also has a huge list of truly
awesome features.
-Jonathan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bastian
Schern
Sent: Thursday, August 11, 2005 9:41 AM
To:
Jonathan k. Creasy [EMAIL PROTECTED] writes:
even when I have taken all other security measures I also lock down
a box with a firewall
You still should follow the lists. As long as your computer is
processing data received from the net, it may be vulnerable;)
Generic attacks against the IP stack
PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adam
Goryachev
Sent: Thursday, August 11, 2005 8:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Newbie Question: Building anAsterisk
systemtoreplace an old PBX but using existing phone
Jonathan k
Our vendor told us we can't buy the 841's anymoreanyone else have
this problem or have a vendor that is still selling them?
-Jonathan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alex
Ternero
Sent: Wednesday, August 10, 2005 9:44 PM
To: 'Asterisk
I think the foneBRIDGE is too expensive for what it does. IMHO
-jonathan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Cory
Andrews
Sent: Tuesday, August 16, 2005 4:06 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
translations? What about docs and support? What are the
chances the box is really just an mini * server?
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Jonathan k. Creasy
Sent: Tuesday, August 16, 2005 2:51 PM
To: Asterisk Users Mailing List
Yes, you could do that with Asterisk and Cepstral/Festival.
-Jonathan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Thursday, August 11, 2005 6:36 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] is this possible
What problem are you trying to solve with this? Just stepping out on a
limb but it sounds like you are trying to swat a fly with an F-16.
-Jonathan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Christoph
Eicke
Sent: Wednesday, August 17, 2005 4:34 AM
I'm testing one now. We got the 2x and the 4 port add-on license.
It seems to be a nice product. It's a basic VOIP PBX. We are using it to
provide a customer site with 2 SIP Trunks, 6 phones, voicemail, failover
to PSTN and 2 FXS ports.
We are evaluating whether we want to deploy asterisk to
No
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Wednesday, August 17, 2005 7:38 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] How many TDM22P Card can be used on the same
PC ?
Is it possible to use 24
PROTECTED] nombre de Jonathan k.
Creasy
Enviado el: jueves, 18 de agosto de 2005 11:59
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Asunto: RE: [Asterisk-Users] How many TDM22P Card can be used on the
same PC ?
No
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
Maybe you didn't intend this for the list?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tim Pushor
Sent: Thursday, August 18, 2005 3:17 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Set voicemail maximum
The power supply could definitely be the problem. You tried a difference
TDM04B right?
-Jonathan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Patrick
Fortin
Sent: Thursday, August 18, 2005 3:21 PM
To: asterisk-users@lists.digium.com
Subject:
Do you need a hangup in your dialplan?
-Jonathan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Howard
Leadmon
Sent: Friday, August 19, 2005 4:06 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] Ascend Pipeline
I have some netweb 302 phones which we used when we first started
evaluating Asterisk. These phones would do this all the time. It seemed
to have nothing to do with the network ping time. I never really did
check in to what was going on because it only happened with those phones
and we don't use
Is the php script available somewhere?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jeremy
Gault
Sent: Monday, August 22, 2005 10:34 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] All Page ??
I have (sort
w and W allow call recording when passes as options to DIAL, in this
case they are being passed as options to D().
-jonathan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John
Novack
Sent: Monday, August 22, 2005 3:51 PM
To: Asterisk Users Mailing List
I'm not having any problems with the SPA-841's at the moment. I have 15
of them in use right now. The other phones we use are the Polycom
IP30X's and they are really nice phones for that price range. I haven't
tried a really expensive phone so I may not know what I'm missing.
-Jonathan
I'm
I think he's talking about putting protection from the PSTN lines not
the incoming power.
-Jonathan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark
Phillips
Sent: Thursday, August 25, 2005 11:32 AM
To: Asterisk Users Mailing List - Non-Commercial
I have a problem where my Asterisk server stops answering new TCP
requests and begins to use 99.9% of the CPU on my box. The server is a
64bit Xeon with 2GB of ram.
I haven't been able to recreate the problem but it occurs sometimes when
there is a call coming from my provider (via IAX) to a
Well for one thing, on a PRI it is usually still transmitted with a bit
set that tells the system to hide it.
-Jonathan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alexander
Lopez
Sent: Thursday, March 16, 2006 9:49 PM
To: Asterisk Users Mailing
-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of C F
Sent: Thursday, March 16, 2006 10:01 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] OT: Unblocking bloced CID
On 3/16/06, Jonathan k. Creasy [EMAIL PROTECTED] wrote:
Well for one
I am having this problem also. I have 2 systems running 1.2.5. I had the
problem and one system was running 1.2.4 and the other was running a CVS
HEAD from October so I upgraded them both to 1.2.5 with no success.
-Jonathan
-Original Message-
From: [EMAIL PROTECTED]
We are doing this with the latest spandsp, iaxmodem and hylafax.
Seems to work very well for us so far.
-Jonathan
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Michael Gaudette
Sent: Tuesday, March 21, 2006 3:34 PM
To: 'Asterisk
Do you want to dial an outgoing line as well as the SIP line?
Dial(SIP/${OUTGOING}/${EXTEN}) ?
I can't say obviously without more info but it sounds to me like you are
looking for the wrong solution
-Jonathan
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
It's a toll free number. You can call it from anywhere and the costs of the
call go on the callee not the caller.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of C F
Sent: Wednesday, March 22, 2006 7:50 AM
To: Asterisk Users Mailing
I haven't read every message in this thread so I apologize if this is a
repeat. Have you considered using the cfg files and an ftp server to
configure the phones? I have found it to be very convenient as a way to
manage many phones spread out across several locations as well as
maintaining one or
This is not a dumb question.
Most of the other replies I have read mentioned various ways to connect
to the pstn. I wanted to mention why it makes sense to do that. Many of
the companies I have installed asterisk for didn't even have their
system on a network with a gateway. They have dedicated
I think there is a bug related to this. I haven't been able to track it
down or really recreate it with any certainty yet. When I do I'll post
something to Mantis. If you have any info to share with me about your
situation when this occurs let me know.
I have noticed that I can get it to occur
I have found this to be true also.
[whatever] has to match username=
It appears that it ignores the username field for IAX users.
-Jonathan
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Tomas Komarek
Sent: Monday, March 27,
You have to use H323 the last time I did
anything with their equipment. It has been almost a year but I think it went
fairly smoothly. Do you have a specific question?
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Stephen Arulraj
Sent: Friday, March 31, 2006
I agree we have this working also.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Boris Bakchiev
Sent: Friday, March 31, 2006 8:55 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users]
Title: Re: [Asterisk-Users] Blocked channels,according to our telco... leading
to CONGESTION status
Id say try it out and see what the
CPU load is. Its not that hard to drop it in your dialplan and give it a
try. Its much easier than figuring out all the possible variables in
your setup
I have had this working but not reliably. It seemed to work like this:
Phone A watched B and C.
Phone B watched A and C
and Phone C watched A and B.
I could see on Phone A (601) when phone B (501) was on the phone. Phone
C never saw the status of either and Phone B would show the status of C.
I apologize if this information is posted elsewhere. Unfortunately I
haven't found it yet if it is. I'm not familiar with the channel
counting features could you please explain? Also, how are you tagging
the phones to account codes?
You can limit calls using the set/check group commands.
Anyone know what has happened to the local calling guide?
http://members.dandy.net/~czg/search.html
-Jonathan
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Does anyone know the format for the TOS element in the Polycom
config?
-Jonathan
Jonathan Creasy
Network Engineer
BluegrassNet Development
www.bgnd.com www.bluegrass.net
o. 502-589-4638
c. 502-889-5567
h. 502-541-0566
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I can dial other extensions internally, and can get to voicemail, but
when I try an outside number, I hear dial tone, the digits dialed, yet
nothing happens when I press Send.
Nothing appears on the Asterisk CLI screen.
Did the upgrade modify the dialplan setting on your phone? This
I could be wrong but off the top of my head I think that it is in the
features section of the config file.
-Jonathan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anton
Krall
Sent: Tuesday, April 18, 2006 4:47 PM
To: 'Asterisk Users Mailing List -
Below is a snipped debug on our PRI. We are getting number
only for the CallerID but the telco says they are sending us Name and Number.
We are getting the Name in a second frame but Asterisk is not passing it to the
device it rings. The message below says Presenation allowed of network
Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] PRI caller ID
Pleaase read the archives or the wiki - you will shortly find you
need a wait in your dialplan
On Apr 19, 2006, at 8:10 AM, Jonathan k. Creasy wrote:
Below is a snipped debug on our PRI. We are getting
Just appending the area code variable is not always going to be correct.
You will need to lookup (google local calling guide) the proper NPA for
the NXX you are dialing. For example, in Louisville, Ky if you dial
948-1592 you will actually reach 812-948-1592 instead of 502-948-1592
even though
I am not by any means recommending this to anyone but I wanted to
publish this for reference.
I have an Asterisk system connected to a provider via IAX trunks. There
are 32 phones on our network and we have about 400 calls per day to/from
our system. The hardware running this is a Pentium Pro
I have a strange problem. I have a central server with my PRI on it.
There are three peripheral servers connected via IAX.
I have a 64bit system for my central server and the backup system is a
32bit system. If I have forwarding (sip redirect) turned on and
forwarding to an outside number (i.e.
, 2005 9:00 PM
To: asterisk-users@lists.digium.com
Subject: RE: [Asterisk-Users] No Audio from Console but
mpg123fromshellworksfine.
On 10/16/2005, Jonathan k. Creasy [EMAIL PROTECTED] wrote:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir
Cohen
You can do it with a Polycom (and probably a Cisco) by setting an Alert
var and it will handle the call using a defined class.
Search for paging.
-Jonathan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jason
Pyeron
Sent: Tuesday, October 18, 2005
I probably can't provide any better information for you, however, have
you looked through the Polycom configuration files. The button mappings
are there. I haven't spent much time with it so I can not attest to what
you can map them to do.
Hope this helps you a little.
-Jonathan
-Original
I dont get it.
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jerry Richmond
Sent: Thursday, October 20, 2005
9:46 AM
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users] user
name
I am geting e-mail but asterisk doesn't know my user name or
Anyone know what's causing this:
-- SIP read from x.x.x.x:56800:
ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.200.16;branch=z9hG4bKba5c00f818387D67
From: Xxx xxx sip:[EMAIL PROTECTED];tag=69375B3E-ACF6C78B
To: sip:[EMAIL PROTECTED];user=phone;tag=as57402fc2
CSeq: 1 ACK
Voicetronix OpenSwitch6
http://www.telephonyware.com/telephonyware/tw3.html
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Asterisk-Users@lists.digium.com
Anyone using the Genesys framework with an Asterisk PBX?
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To UNSUBSCRIBE or
Has anyone figured out how to make the shared line
appearance thing work with asterisk?
From wiki: http://www.voip-info.org/wiki/view/Polycom+Phones
Supports shared lines (but asterisk does
not) - Anyone having details on the specifications used for Shared Call /
Bridged Line
Can a Polycom IP601 with the addon modules be setup to work like an
attendant console showing the status of other lines?
How does that sort of thing work with Asterisk?
-Jonathan
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Any IRQ or duplex problem with your NIC? Any collisions or errors?
I have had similar results to others here in that conferences with
50-100 users are just fine even on fairly outdated hardware.
-Jonathan
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL
Is the ambient noise in the room high?
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Matt Riddell
Sent: Tuesday, November 08, 2005 8:53 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users]
Title: Extension Ring on Multiple Phones
EXTEN= 100,1,DIAL(SIP/ONESIP/TWOSIP/THREE)
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dave Morrow
Sent: Tuesday, November 08, 2005 1:51
PM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject:
I guess I should have read up further before I posted a response.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Mojo with Horan Company, LLC
Sent: Tuesday, November 08, 2005 2:51 PM
To: Asterisk Users Mailing List -
I thought there was a sip image for that phone?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John
Reynolds
Sent: Tuesday, November 08, 2005 4:28 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Re: Cisco
Anyone used a sip from from Planet Network?
VIP-153
http://www.planetnw.com/
http://www.planetstoresite.com/Merchant2/merchant.mvc?Screen=PRODStore_Code=PNIProduct_Code=VIP-152TCategory_Code=VOIP
___
--Bandwidth and Colocation
mailbox= in the sip.conf
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sixto Diaz
Sent: Tuesday, November 15, 2005
9:33 AM
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Message
waiting notification
Hi i want to notify a user that he has an
Subject: Re: [Asterisk-Users]
Message waiting notification
i want to ring the phone user or change the
tone is this posible with mailbox= ?
- Original Message -
From: Jonathan k.
Creasy
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent
Use notepad if you must edit them on a windows box.
Nano/Pico/Joe are pretty user friendly editors for the *nix environment.
-Jonathan
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Chuck Bunn
Sent: Tuesday, November 15, 2005
FXO ports are an interface between your system and a phone carrier. FXS
ports are an interface between your system and a phone station (or
handset).
You can send outbound calls on an FXO port as well as receive them.
To dial on the TDM card you would dial : DIAL(Zap/g1/${number}) or
I am trying to replace the overhead paging function of an old phone
system. There is a device with an RJ11 connection connected to two
screws on the phone system. The two screws are on a cartridge labeled as
the subject of this message.
I thought the other device was probably a station andthat I
I can't seem to compile IAXmodem.
sh build:
iaxmodem.c: In function `cleanup': iaxmodem.c iaxmodem-cfg.ttyIAX lib
README termpkg-ttydforfax.patch TODO
iaxmodem.c:90: error: too many arguments to function `iax_register'
iaxmodem.c: In function `main':
iaxmodem.c:705: error: `IAX_EVENT_CNG'
For which equipment?
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of xcel
Sent: Friday, November 18, 2005
11:53 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users]
Provisioning server
Can any one help me in setting up Provisioning sever
??
What context are your phones in? (context= in sip or iax config)
If your phones are in the local-users context, they will be able to dial
numbers found in local-users, extensions and local.
If your phones are in the long-users context, they will be able to dial
numbers in long-users, local,
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