Testing the new lists.digium.com server. Apologies for the email noise.
Digium IT
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Check out the new Asterisk community forum at:
the
customizations in there, but now that I am using Realtime voicemail from MySQL,
my voicemail.conf file has to be an empty file.
So does anyone know how it would be possible for me to customize the content of
the email, other than hacking the source?
Cheers,
Brad Hughes
.
Asterisk connected as a SIP client, under 2ms.
The higher latency with the Cisco's don't seem to effect performance at all.
--
Brad
To: asterisk-users@lists.digium.com
From: seandar...@gmail.com
Date: Sat, 23 Oct 2010 12:31:58 -0400
Subject: [asterisk-users] Why such high latency on internal lan
I have been using SiSky Enterprise Edition to integrate Skype with asterisk.
You can even call saved skype users from your asterisk system, by creating
speed dials in SiSky. Unfortunately it is not a free product but it is very
reasonable.
Thank you,
Brad Finberg
- Original Message
to implement
an A-M and N-Z but the phone labels the directories as External Directory
I also realized changing the Titleabc/Title or the Promptabc/Prompt
does nothing...
Is there a way to change the External Directory to CompanyName A-M etc...?
Thanks,
Brad
-BEGIN PGP SIGNATURE-
Version: GnuPG
there had a similar issue and
found the fix?
Asterisk server is 1.4.26
Cisco 7961G is running SIP version 8.5-2S
Thanks.
-Brad
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To UNSUBSCRIBE
/0206-08522f28
-- Packet2Packet bridging SIP/0206-08522f28 and SIP/0203-08529f68
== Spawn extension (inside_sip_phones, 0203, 2) exited non-zero on
'SIP/0206-08522f28'
asterisk*CLI
Thanks.
-Brad
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk
Thank you,
Brad Finberg___
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in the extension you want to transfer too.
Thank you,
Brad Finberg
- Original Message -
From: Michael as...@nettrust.co.nz
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Cc:
Date: Tuesday, July 14 2009 11:07 PM
Subject: [asterisk-users] call transfer
.
This causes much confusion.
Can anyone please tell me how to configure Asterisk properly for working
behind a SIP Proxy?
Below you will find our configuration.
Thanks,
Brad
Here is the channel for our SIP provider:
[my_provider]
type=peer
host=my.provider.com
username=100-phone
secret=mysecret
Hello,
If anybody has any idea's to where I should start looking to fix the below
subscription problem. If there is another mailing list I should post this to
please let me know.
Thank you,
Brad Finberg
- Original Message -
From: Brad Finberg b...@finberghouse.com
To: Asterisk
Anybody have any idea's
Thank you,
Brad Finberg
- Original Message -
From: Brad Finberg b...@finberghouse.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Cc:
Date: Thursday, April 9 2009 9:45 AM
Subject: [asterisk-users] notifyringing
state Ringing for Notify User 103
Thank you,
Brad Finberg___
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does anyone have a sample dialplan for vici dial that does not include any pri
stuff.
I am running exclusively SIP for everything and trying to edit the sample
dialplan and removing anything to do with a pri card is becoming a nightmare!
Thank you!
we just did a brand new installation of asterisk 1.4 on ubuntu with a sagnoma
t-1 card
everything went smooth (other than fighting a little outbound call issue that
we are sure is a tdm network to sagnoma issue)
inbound calls are fine
dialplan is silly basic with outbound channels set to
I am trying to lauch a first outbound call.
I am connected to my telco via a peer which is a little different from what I
consider the norm.
extinsions.conf
[To_Bandwidth]
ignorepat = 9
exten = 9,1,Dial(Sip/g2/)
exten = 9,2,Congestion
sip.conf
[To_Bandwidth]
canreinvite=yes
context=from-pstn
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf
Of Brad
Sent: Friday, August 15, 2008 12:09
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Basic outbound calling issue
I am trying to lauch a first outbound call
-09ef2cc0 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
== Auto fallthrough, channel 'SIP/100-09f2ee18' status is 'CONGESTION'
--- On Fri, 8/15/08, Brad [EMAIL PROTECTED] wrote:
From: Brad [EMAIL PROTECTED]
Subject: Re: [asterisk-users] Basic outbound calling issue
== Everyone is busy/congested at this time (1:0/1/0)
== Auto fallthrough, channel 'SIP/100-b7c03ce8' status is 'CONGESTION'
--- On Fri, 8/15/08, Brad [EMAIL PROTECTED] wrote:
From: Brad [EMAIL PROTECTED]
Subject: Re: [asterisk-users] Basic outbound calling issue : a lot closer
To: asterisk
I figure it out, asterisk is using the wrong ip address.
I have bind address set to the correct ip address. How to I force asterisk to
use the correct ip address?
--- On Fri, 8/15/08, Brad [EMAIL PROTECTED] wrote:
From: Brad [EMAIL PROTECTED]
Subject: Re: [asterisk-users] Basic outbound
, 8/8/08, Brad [EMAIL PROTECTED] wrote:
From: Brad [EMAIL PROTECTED]
Subject: [asterisk-users] VICIDial error
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Date: Friday, August 8, 2008, 6:02 PM
Warning: Cannot modify header information - headers
of concept
To: [EMAIL PROTECTED]
Date: Friday, August 8, 2008, 4:18 PM
Hey Brad,
The simplest way I thought to implement it for a client who
needed
multiple calls to be placed based on time was to code a
deamon that
would query the db every given interval, check if there
were any calls
Warning: Cannot modify header information - headers already sent by (output
started at /home/telecom/public_html/vicidial/admin.php:1175) in
/home/telecom/public_html/vicidial/admin.php on line 1187
Warning: Cannot modify header information - headers already sent by (output
started at
in advance,
Brad
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returning newbie.
Trying to register ekiga for the first time to my asterisk server only.
[204]
user=204
context=internal
type=friend
secret=xxx
insecure=very
canreinvite=no
host=dynamic
disallow=all
allow=ulaw
allow=alaw
nat=no
Can anyone tell me what I am missing?
I am not
:17 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] ekiga register problems
On Mon, May 28, 2007 at 10:45:25PM -0400, Brad Sumrall wrote:
returning newbie.
Trying to register ekiga for the first time to my asterisk server only.
[204]
user=204
context=internal
type
a text message through asterisk!
Brad
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On Fri, May 11, 2007 at 11:06:35AM -0400, Mike wrote:
Hi,
I have a question of using 2 SIP providers. Let's say I have provider A and
provider B, and I would like my calls to go to A, and then B if A wasn`t
available
What would be really cool, but require special code in the chan_sip
I installed the asterisk GUI, Asterisk web manager, it loads fine, but if
I go to the AGI section, I get a permission denied
Obviously apache cannot access the /etc/asterisk directory.
I added apache as group, but still the same problem.
Suggestion any one?
a Cisco phone, I
would imagine polycom has a similar features)
Let me know if this helps!!!
Polycom is very picky!
Brad
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jim Freeze
Sent: Wednesday, April 25, 2007 4:37 PM
To: Asterisk Users Mailing List - Non
Pix usually uses NAT,
A quick fix is to simply forward the ports in your NAT statements.
If the pix is new, call Cisco and cheat like I do so often!
Brad
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ed Nuñez
Sent: Wednesday, April 25, 2007 9:31 AM
the job done quickly!
Outside of that, this mailing list is a great place for support, we all work
together!
Brad
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Hans Witvliet
Sent: Wednesday, April 25, 2007 5:24 PM
To: asterisk-users@lists.digium.com
Subject
php is rusty.
Maybe you can hook a brother up with the proper code to grab caller id and
query mysql?
To answer your question, Yes, you are on the right track!
Brad
[EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of adriano ghezzi
Sent
ports are open with telnet:port number both ways, telnet is your
friend.
If it works, close the holes up and consult your firewall docs
Brad
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Hadar Pedhazur
Sent: Wednesday, April 25, 2007 6:13 PM
To: Asterisk
Personally, I look for specialty applications. Work smart not hard!
I myself am looking for outstanding marketers for a fire hot industry /
telecom application. I have all of the correct duckies in a row, just need
to send it to the market the correct way.
[EMAIL PROTECTED]
_
Hard reset the phone first!
Provision and see if it is fixed.
No?
Upgrade software (watch out for provisioning changes).
Still rebooting?
Downgrade software.
Still rebooting?
You now have a new door stop!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
test
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wayne Jensen
Sent: Wednesday, April 25, 2007 7:41 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Random Asterisk deaths
Asterisk 1.2.13 (newest available
I am very confident the 7960G has a sip load. I know for sure the regular
7960 does and the G just means gigabit interface. The 7970 was the only one
that didn't because of all the color interface/touch screen, and Cisco was
still pushing call manager big time, so skinny was the only load
.
Sincerely,
Brad
[EMAIL PROTECTED]
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, when I try to do a factory reset (holding down #, power cycling) it
never asks for the reset key sequence and never said it detected the key
sequence.
Any advice would be appreciated.
Thanks,
Brad
--
This message has been scanned for viruses and
dangerous content by MailScanner
Indeed this might be the failing point... Unfortunately, because I have no
Cisco CCO account anymore, I have no access to firmware... I will try to find
a copy of an old firmware for these phones. If I can find one, I hope it
fixes my problem.
Thanks,
Brad
Apologies in advance
this new gateway!!!
i.e. establish a VPN connection to an outside router from an internal router
and drive the call through there.
Brad
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of A. Levy
Sent: Tuesday, March 27, 2007 6:54 AM
To: asterisk-users@lists.digium.com
Hello everyone
My name is Brad, I am an old Asterisk Vet of the very early days just coming
back to join the group.
Ok, for starters, I feel like the monkey with the light bulb looking at
extensions.conf and sip.conf.
It has been some time.
A friend ask me to set up a asterisk server
On Sat, Mar 24, 2007 at 12:13:25PM -0700, Martin Joseph wrote:
On 2007-03-23 14:37:18 -0700, Tom Lynn [EMAIL PROTECTED] said:
Now I know where they've been spending my remaining balance...
I still use Sellvoip as my primary terminator, and have found the call
quality to be superior
I am geet this error, I assume because I have zero digium hardware
installed. This is to be an entirely web based PBX.
Can anyone point me to an easy 123 for installing zaptel in dummy form?
I need music on hold for a VPS server.
Brad
-Original Message-
From: [EMAIL PROTECTED]
[mailto
On Wed, Mar 14, 2007 at 09:37:45AM -0500, Steve Totaro wrote:
Another interesting (from an American's perspective anyways) is that
inbound calls on cell phones are free. Even if you buy a SIM with a
little pre-paid time and use up the time, you can still receive inbound
calls for free for a
Another idea that has just come to me regarding bluetooth and a PBX is
like this.
Many people would like to use headsets with their IP phones. Some
support wired headsets, but bluetooth headsets can be a good choice
for a headset -- no wires, many people often have one, and there is
a rich
On Fri, Mar 16, 2007 at 11:32:31AM -0700, Tom Lynn wrote:
Has anyone been successful in getting a refund from SellVoip when you've
cancelled service?
You were able to cancel service with Sellvoip? That's impressive, that
implies they actually responded to a request you made to cancel
service.
On Fri, Mar 16, 2007 at 04:16:21PM -0700, Tom Lynn wrote:
At this point, I'm simply contacting the State of Washington Attorney
General's office. They're ignoring my e-mails and I'm done monkeying
around.
It makes no sense. The put together a good system on the tech end,
Asterisk based,
On Tue, Mar 06, 2007 at 11:14:15PM -0500, Steve Totaro wrote:
Mine goes to chan_bluetooth. Somewhat of a pain getting it going but I
am totally floored with how cool it is!
Thanks,
Steve Totaro
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Do I get 1.4.1 by just doing an svn up in a checkout of the 1.4 branch, or
do I have to switch to a new tag or branch for what I have checked out?
I did an svn up and there are new files, but nothing in the change
files about it being 1.4.1.Many packages with various minor
versions tend to
On Sun, Mar 04, 2007 at 08:50:48AM -0600, Kevin P. Fleming wrote:
Brad Templeton wrote:
I did an svn up and there are new files, but nothing in the change
files about it being 1.4.1.Many packages with various minor
versions tend to have the master branch (like 1.4) mean The latest
On Sun, Mar 04, 2007 at 02:34:21PM -0600, Kevin P. Fleming wrote:
Brad Templeton wrote:
In many packages there is some file (usually the change log) which always
tells you what version of the program you have in your hands, in terms
of the program's current version number -- of course you
On Fri, Feb 23, 2007 at 09:11:18AM +0100, [EMAIL PROTECTED] wrote:
hi guy, i have a problem, i have an sellvoip account and i want
configure asterisk for outbound calls.
Alas, the best sellvoip configuration, I eventually had to conclude,
was not to use sellvoip. They have good quality
On Tue, Feb 20, 2007 at 12:08:15PM -0700, Natambu Obleton wrote:
Why not make it like DNS and have each provider have their lookups
deligated to a local server and then each ISP will run a caching
server that will use a serial number system to get updates.. just like
DNS.
I know there are
On Mon, Feb 19, 2007 at 09:02:56PM -0500, C F wrote:
I doubt it's CNAM since it has old an outdated listings.
On 2/19/07, Paul [EMAIL PROTECTED] wrote:
Does google really have the true CNAM database? When I enter my number,
I get a search result for my business listing at yellowpages.com
On Mon, Feb 19, 2007 at 03:01:50PM -0500, Paul wrote:
I think terms of service for most CNAM providers prohibits sharing the
data and limits the amount of time it can be cached for your own reuse.
I don't know why they manage to get this level of control over the cnam database
so that they
On Mon, Jan 29, 2007 at 12:05:28PM +0100, Benny Amorsen wrote:
PC When I set for Extern1/2 canreinvite=yes it works, but
PC Intern-2-Extern doesn't work because Asteisk gives out the
PC private IP-Adresses of Int1/2
Asterisk can't give out a public IP-address for Int1/2. Where
would it get
On Tue, Jan 30, 2007 at 12:00:17PM +0100, Patrick Cervicek wrote:
Brad Templeton schrieb:
On Mon, Jan 29, 2007 at 12:05:28PM +0100, Benny Amorsen wrote:
PC When I set for Extern1/2 canreinvite=yes it works, but
PC Intern-2-Extern doesn't work because Asteisk gives out the
PC private IP
On Tue, Jan 30, 2007 at 10:23:09PM +0100, Benny Amorsen wrote:
PC == Patrick Cervicek [EMAIL PROTECTED] writes:
PC But then all RTP Traffic of my internal phones will go over
PC Asterisk. I want RTP to go Peer-to-Peer. == Intern-2-Intern
PC and Extern-to-Extern should go P2P and
On Sat, Jan 27, 2007 at 01:55:31PM -0500, Jerry wrote:
On Fri, Jan 26, 2007 at 12:34:30PM +, Tim Panton wrote:
Unless you are monitoring calls, want full CDR etc,
then that's what you want anyway.
CDR are not affected by how the audio flows.
While technically true, I believe
On Thu, Jan 25, 2007 at 10:19:06PM -0800, Yuan LIU wrote:
Asterisk1 -- NAT1 --- { Internet } --- NAT2 -- Asterisk2
If Asterisk1 can talk to Asterisk2 at trunk level, I'll be happy.
While I'm not sure of what tricks * plays at all levels, you
can certainly make this work if you have control of
On Fri, Jan 26, 2007 at 12:34:30PM +, Tim Panton wrote:
For a remote phone, not on the same network as the Asterisk
box (in which event the NAT worries are different) you definitely
want to use the same protocol for the phone as for your
term/orig provider. Otherwise you will be forced
On Wed, Jan 24, 2007 at 11:09:21PM -0800, Yuan LIU wrote:
From: Brad Templeton [EMAIL PROTECTED]
On Mon, Jan 22, 2007 at 09:59:06AM +, Tim Panton wrote:
In the meanwhile, use IAX, which understands about NAT pretty well.
If you have multiple SIP phones on a LAN behind a NATing router
On Mon, Jan 22, 2007 at 09:59:06AM +, Tim Panton wrote:
In the meanwhile, use IAX, which understands about NAT pretty well.
If you have multiple SIP phones on a LAN behind a NATing router, just
put a small asterisk box on the LAN. It can manage your hairpin
calls internally, save you
On Tue, Jan 23, 2007 at 02:01:43PM -0600, Tom wrote:
Has anyone found a high quality wireless headset that works well with
Cisco 7960 IP phones on an asterisk system?
I tried the vxxi offering but the sound quality was pretty bad.
Since these are pricey, I don't want to sample blindly.
Some NAT problems you can solve, some you never will.
Many modern phones have NAT support in them, via STUN, or a static external IP
address. Most NATs also offer port forwarding, so you can open a hole for the
SIP port in the NAT so all outside can reach it.
(With port forwarding, you need a
On Sun, Jan 07, 2007 at 04:12:27PM +, Thomas Kenyon wrote:
Brad Templeton wrote:
For SIP phone calling * box, relay to other * box and out to SIP
phone, you definitely want SIP all the way.
Unless bandwidth between the * servers is a concern, then you're better
off keeping the link
On Fri, Jan 05, 2007 at 11:33:02AM +, Gordon Henderson wrote:
On Thu, 4 Jan 2007, Noah Miller wrote:
Hi Damon -
Can anyone comment on the overhead added when a SIP call comes into one
asterisk box, is routed to another with IAX instead of SIP, and is then
sent
to the UA from the
On Fri, Jan 05, 2007 at 05:37:22PM -0500, Allen Casteran wrote:
Mike wrote:
You're quite right, I typed before thinking. Upload is the problem
anyways, since it usually (in homes) uses much more limited bandwidth
than downloading does.
No answer to my question though: How do you people
On Fri, Jan 05, 2007 at 02:29:03PM -0800, CM Rahman wrote:
Anybody selling DID flat rate for less then $4 with sip or iax incoming? Let
me know
Thanks
vbuzzer charges $2 for flat rate DIDs, not quite sure how they do it.
However, I have had some clicks and pops of dropped packets with them
I was just trying astman with the latest svn trunk from Dec 31. It
connects, but if I attempt to make a call, asterisk segfaults, but
in pthread_kill in /lib/tls/libpthread.so not in the asterisk code.
Is this something others have seen? This is with glibc-2.3.4-2
I just upgraded to 2.3.6 (the
On Wed, Jan 03, 2007 at 11:15:19PM -0500, Jeronimo Romero wrote:
I'm going to be rolling out asterisk at a small office and one requested
feature was the ability to have a phone that can be configured so that
ringtones can be configured according to the callerid of the caller.
Does anyone
I decided to give the whole family IP phones for christmas,
all hooked into my asterisk server, so all the nephews can
have their own lines.
However, one of the phones I got was the SNOM 200. That's worked
fine for me on my own network, but I'm having bad luck getting
it to work behind NAT
On Sat, Dec 23, 2006 at 02:02:18AM -0800, Brad Templeton wrote:
I've tried all the various NAT settings on the SNOM 200 (with
the last firmware rev they made) but reports are that's broken.
The SDPs and Contact headers it sends out are always the natted
address, even if I tell it to use STUN
On Wed, Dec 20, 2006 at 02:34:36PM -0500, Phil Finkler wrote:
I've gotten this Polycom 501 pretty much licked, but I need to know if
there's a way in a dialplan to say if someone dials their own extension
it goes straight to voicemail and asks them for their password. I
thought I saw an
On Mon, Dec 11, 2006 at 06:48:18PM -0500, Andrew Joakimsen wrote:
You need to understand how NAT works, if you can chan2 and chan2 is behind a
NAT and suddenly someone else is invited to chan2's IP address port 5060
chan2's router willl say WTF I dont have an estabished connection on port
5060
Cisco devices (7912, ata-168, 7960 etc.) don't support STUN.
However, they do let you define a static external NAT IP
address, and parameters to send a keep-alive out through the
NAT on a regular basis.
However, I want to make these devices work in an environment
where they are behind a NAT
On Wed, Dec 06, 2006 at 08:13:00AM -0500, Paul wrote:
Also, I should have mentioned that many of these providers advertise
business plans on their website. How can anyone honestly advertise
phone, fax, email hosting, web hosting, etc. to the business community
without 24/7 support?
I like
On Tue, Dec 05, 2006 at 11:36:12AM -0500, Al Bochter wrote:
And if you get someone over at Vonage that knows that to do you can
connect without the FXO
It is like FWD you have to get the KEY from Vonage for this to work.
And more to the point there are so many VoIP providers out there,
most
I have not seen anybody on the web to have found this so I thought
I would check here. Anybody got this firmware? I've found
firmware for the 400, but it doesn't seem to load in the 410.
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On Fri, Dec 01, 2006 at 04:55:51PM -0500, John Novack wrote:
In most hybrid business systems one does NOT place a call on hold, but
begins a transfer, either a specific function button or intercom button
which automatically places the call on hold, gives a new dialtone and
another extension
On Fri, Dec 01, 2006 at 07:37:35PM -0700, Ken Williams wrote:
I was able to set a program to speed dial the park extension. Then a user
just hits TNFR followed by the line I've programmed to speed dial park.
If you get the HOLD button to do this, I'd love to hear how :).
Oh, that would
On Thu, Nov 30, 2006 at 12:03:24AM -0600, Lacy Moore - Aspendora wrote:
The question is what is the best interface? On our old system, we put the
caller on hold, went to another phone, pressed pickup and then entered the
extension where the call is on hold. I never liked that, especially if I
On Thu, Nov 30, 2006 at 02:50:21PM -0500, Tom Rymes wrote:
for example: In your example above where they can't figure out how to
transfer, why don't you edit features.conf and define the transfer
key as # or something. Then, when they have a call for Bill across
they way, they can do
On Wed, Nov 29, 2006 at 06:05:31PM -0500, Steve Sobol wrote:
On Mon, 27 Nov 2006, Brad Templeton wrote:
On Mon, Nov 27, 2006 at 11:20:27PM -0500, Andrew Joakimsen wrote:
Can you explain how ValetParking and twenty minutes worth of dialplan
creativitiy can't do the same EXACT thing you
On Wed, Nov 29, 2006 at 04:49:38PM -0700, Joseph wrote:
What I have is that each device is listening on different port ex.
[pstn-5665] ; incoming/outgoing calls on FXO port
type=friend
...
port=5066 ; port on Pstn line
...
[318] ; incoming/outgoing calls on FXS Sipura-2002
type=friend
On Mon, Nov 27, 2006 at 01:46:58AM -0800, Steve Langstaff wrote:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Brad Templeton
Sent: 25 November 2006 21:02
[snip]
...the UI I think most people want, which is, just put
the call
On Mon, Nov 27, 2006 at 04:05:34AM -0800, Steve Langstaff wrote:
What I describe is different. There are no shared lines, but if
you put a call on hold on one phone on a non-shared line you
can go to another -- any other in the pickup group, whether
it is registered to have the
On Mon, Nov 27, 2006 at 11:20:27PM -0500, Andrew Joakimsen wrote:
Can you explain how ValetParking and twenty minutes worth of dialplan
creativitiy can't do the same EXACT thing you are describing? Sometimes the
simplest answer is never the most obvious
Yeah. With valet parking (or any
On Sat, Nov 25, 2006 at 07:17:47AM -0500, marvin horst wrote:
The valet system gets us partway from what I read, but it still uses the
arbitrary number slots. It still requires the user know to transfer a
call to the valet.
no you can park to a specific number (lotname)
exten =
On Wed, Nov 22, 2006 at 04:51:26PM -0800, Ira wrote:
At 03:14 PM 11/22/2006, you wrote:
The missing piece of the puzzle: I'm extension 203. I want any call I park
to get parked at extension 2203. I want a call my boss parks to park at
2205, since he's ext. 205. In other words, I want calls
On Wed, Nov 22, 2006 at 11:29:01PM -0500, Michelle Dupuis wrote:
48VDC is a long time telco standard - and has become the Power over Ethernet
standard.
Keep in mind that 'electricity' isn't the measure - it's power. Power is
not synonymous with voltage.
More to the point, there is a
I'm having an issue with call park on my new Linksys 922. It has
soft menu keys for doing call transfer (which I always think is a good
idea because it's amazing how every phone has a different xfer interface
and people always get confused).
However, I can't get a good call park working on it.
A follow up on my message about my SNOM 200 phones now powering from
my 802.3af Netgear FS108p PoE box.
To follow up for those finding this thread on searches...
I purchased some PowerDSine 6001 units (very cheap on ebay) and they
power the SNOM 200 fine. Some Buffalo units also did this.
So
Ok, not exactly an Asterisk problem, but...
I picked up some SNOM 200 phones because SNOM's have been recommended for use
with Asterisk and they have line buttons that can subscribe to presence.
However, they don't appear to power up when connected to my Negear FS108P,
which is an 802.3af
On Fri, Nov 10, 2006 at 02:34:31PM +1100, Paul Hales wrote:
I had a 200, and it worked fine with POE.
The standard power connector was the RJ-11 style as mentioned below.
Weird item that one.
The successor to the 200, known as a 190 does NOT support poe, while the
320 does.
Yeah,
On Thu, Nov 02, 2006 at 11:23:08AM -0500, Guillermo Salas M. wrote:
On Wed, 2006-11-01 at 16:05 -0500, Vladimir Montealegre Estailes wrote:
Hello list partners
you know about a softphone made in java attachable in a web page?
GNU!
I'm using JIAXClient [1] to permit to any
On Wed, Nov 01, 2006 at 06:16:25PM +0100, Benny Amorsen wrote:
BT == Brad Templeton [EMAIL PROTECTED] writes:
BT The correct behaviour, as I see it is:
BT a) Native bridge when connecting two external channels --
BT everybody is on the real internet b) Native bridge when connecting
I've read a lot of the descriptions of handling NAT with Asterisk,
and the use of both the nat and canreinvite flags. I am very
familiar with Sip and NAT but have not seen an answer to the following
question.
My Asterisk server runs on a machine with two ethernets. One is
an external net,
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