On 11/21/05, Wilson Pickett [EMAIL PROTECTED] wrote:
Is there a syntax I can use to set a variable based on the evaluation
of an expression? I need something that will work in 1.0.9 and 1.2.
Isn't this what you're looking for:
set(VARIABLE=$[NULL${something}=NULL]})
I'm not quite sure I
scott wrote:
Hi
Thank you for your reply.
I have tried various definitions in the sipusers table but none seem to be
working :-(
I have attached mey structure and content export below for your attention.
You should have a look at this page :
Paul == Paul Liew [EMAIL PROTECTED] writes:
Paul You are correct - rxflash and flash in zapata does the
Paul equivalent, but I should also have said in my earlier post
Paul that you need to drop the max pulse time (for pulse
Paul dialling) to be less than the hook flash timing.
In article [EMAIL PROTECTED],
Ben Higley [EMAIL PROTECTED] wrote:
I have read on the wiki the many howto's to select data using the MYSQL
command. I would like to select multiple columns from a table using the
MYSQL command, however, it will only fetch one at a time.
You just need to provide
I managed to isolate the problem a bit more, maybe it will help to find a solution:The problem with the phones is not the initial registration, but the re-registration process.When I create a new extension the phone registers ok, but when the same phone tries to re-register it fails.
On
On 11/18/05 12:55 John Todd said the following:
affordable, which probably means $50 or less I suspect. This would be
a native Linux environment for all components. Again, while I have no
when, oh when, will folk like these support use downtrodden freebsd folk ?
:)
--
Regards,
Hello,
Here is my config :
Asterisk as registrar server :public ip:5050
Ser as outbound proxy server :public ip 5060
I wish ser to handle the packets between Nat box
(netfilter) and Asterisk However contact field in
the sip HF don't change from nat box to asterisk which
don't allow to keep
I've been playing around with AgentCallbackLogin, etc.
Now I get this message
---
Nov 22 17:31:45 WARNING[1889]: channel.c:784 channel_find_locked:
Avoided initial deadlock for '0x
8135b00', 10 retries!
---
whenever a user tries to dial into the system. Restarting asterisk and
even rebooting
The host column must contain 'dynamic' not your IP.
UPDATE sip_users SET
host = 'dynamic'
WHERE
name = '114';
*CLI sip show peers
Name/username
Host
Dyn Nat ACL Port Status
114/114
80.xxx.xxx.xxx
D
5060 Unmonitored
1 sip peers [1 online , 0 offline]
Just try it.
Interested in Open
I have a fairly simple menu structure, three options branch to submenus.
There is a long (several seconds) delay between pressing a key and getting
the next menu. This happens on 2 out of 3 of my menus for no apparent
reason. I am kind of at a loss as to what to look at. Any suggestions would
be
We've had no problems for a few weeks running Asterisk
CVS-D2005.10.28.07.54.
However, this morning, we're getting users complaining that they were
cut off - and I found these in the logs. This has happened 5 times this
morning, and there is an entry in the log at the appropriate time.
Is
On Mon, 2005-11-21 at 21:48 +, Mark Ackroyd wrote:
All,
I thought I'd post the answer to this, After I found what the problem was.
It was the cable from the TDM card to the phone socket. I used one that came
with an old modem and it worked a charm :-)
I've had that problem very often
Hi,
I have a problem with double ringback tone - outgoing connections to
PSTN. I do not use 'r' option in Dial function so I expect to hear
'real' sounds from pstn provider. But PAP2 generates extra ringback tone
itself! How to get rid of that?
Regards,
L
First of all, thank you for your answer, the only that does not claim to
not restart the box !
Asterisk is the last stable version via cvs, not cvs head
show version:
Asterisk CVS-v1-0-10/31/05-17:43:16 built by [EMAIL PROTECTED] on a i686
running Linux
So it was the last stable version on 31
I think I have heard in the past that someone mentioned to me there is a
codec that does not getting affected much because of packet loss.
Is there such thing?
Sam
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing
I also have never found anybody running an Asterisk system using app_icd.
Maybe app_queue is now after all flexible enough to be used in most cases.
Anybody else using different apps for Asterisk call centre applications?
l.
On Mon, 21 Nov 2005 20:30:33 +0100, Waldo Rubinstein [EMAIL
this is very welcome as i need to keep track of agent status using the SNOM BLF
Alexander Lopez wrote:
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Olle E Johansson
Sent: Monday, November 21, 2005 1:13 PM
To: Asterisk Users Mailing
On Monday 21 November 2005 23:49, Rainer Maier wrote:
Hi all,
I want to compile asterisk's newest version with mysql's newest version,
but I ran into a big problem.
At compile time for asterisk-addons-1.2.0 I get the following errors:
make
-- snip --
cc -fPIC -I../asterisk -D_GNU_SOURCE
Yes, something like below should do what you want.
Exten = _90.,1,Dial(ZAP/g1/7980${EXTEN:2})
_
From: Carlos Prieto [mailto:[EMAIL PROTECTED]
Sent: Tuesday, November 22, 2005 1:20 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Using Long Distance Operators
Hello,
Can we configure asterisk in order to send sip
requests to a outbound proxy
when asterisk get AOR of users agents with an private
ip ?
Asterisk AOR:[EMAIL PROTECTED] ip
|
|
sip proxy/nat box---user agent
192.168.0.0/24
Regards
Harry
Hi,
I've been using it on both P4 and AMD64 (32 and 64 bit).
Performance is about the same.
We also didn't have any special compile or usage problems.
David Lowes
Mark Quitoriano wrote:
anyone tried using asterisk on AMD64? how's the performance is better
than p4?
--
Regards,
Mark
On Fri, Nov 18, 2005 at 10:22:23AM -0600, Kevin P. Fleming wrote:
Leo Burd wrote:
Any ideas about what is going on?
Yes. You didn't read the warnings prominently displayed at the end of
'make install' about removing old modules from /usr/lib/asterisk/modules.
Does that include the 729
A simple sql command will do this.
-Original Message-
From: Kerry Garrison [mailto:[EMAIL PROTECTED]
Sent: Tuesday, November 22, 2005 1:10 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] Call waiting issue
Whenever I restart Asterisk, I
Hi all,
today I installed asterisk 1.2stable and than spandsp-0.0.2pre21 with
rxfax txfax. After I restart the asterisk and get the following
errors:
[app_rxfax.so] WARNING[6340]: loader.c:325 __load_resource: /usr/lib/
asterisk/modules/app_rxfax.so: undefined symbol:
Title: Message
There
is a problem with Avaya that DS1 cards are nor recognizing incoming FAX.
Using
unified messaging I must answer the call and if I hear a Faxpulse I have
to transferthe call to UM.
I want
Asterisk do this job. Recognize fax and send directly to UM.
In my
previous mail
Answering myself here.
It turned out that the machine already had kernelcapi installed and was doing
some weird things with the modules.
I removed it and reinstalled isdn-utils.
All is now well!
:)
--
Cheers,
Matt Riddell
___
Paul,
Thanx for your suggestions, but no luck ths far.On 11/22/05, [EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
Once when setting up a SIP based mobile phone
gateway, I had to use
(SIP/${EXTEN)@rupert) and set up an entry in sip.conf for rupert.
This lets you use passwords, etc.
Worth a
Hi,
What I do not understand is how dropped packets prevent the fax from
working. Faxes are designed to adopt to noise on the line by reducing
their connection speed. It seems like their is something else besides
packet loss going on here. Also why would the board work for receiving
faxes
I think you are thinking of iLBC:
http://www.voip-info.org/wiki-iLBC
Be aware that this codec is known to be pretty CPU intensive to accomplish its compression.
- PedroOn 11/22/05, Sam Tam [EMAIL PROTECTED] wrote:
I think I have heard in the past that someone mentioned to me there is acodec
Kerry Garrison wrote:
I have a fairly simple menu structure, three options branch to submenus.
There is a long (several seconds) delay between pressing a key and getting
the next menu. This happens on 2 out of 3 of my menus for no apparent
reason. I am kind of at a loss as to what to look at.
I noticed that asterisk.org now has asterisk and zaptel downloads for
version 1.0.10 but libpri, addons and sounds are still showing a 1.0.9
version number. Just wondering for those using the 1.0.x versions
of asterisk instead of the 1.2 versions - will libpri, addons and
sounds be updated to
And one more update that may help to find a solution to this problem.
If I run asterisk -rx reload the registration works fine until the next re-registration and then I have the same error again
Is there some solution for this problem exept runnning asterisk -rx reload all the time ?
On 11/22/05,
Dominik Simon wrote:
Hi all,
today I installed asterisk 1.2stable and than spandsp-0.0.2pre21 with
rxfax txfax. After I restart the asterisk and get the following errors:
-
But I only have spandsl-0.0.2 installed, and the libs are in /usr/
local/lib, see:
-rw-r--r-- 1 root root 946280
Due to some change I've been unable to identify, my Asterisk box is no
longer successfully passing caller ID to the called party with calls
placed through Voicepulse. This worked just fine until recently. Also,
identical code functions correctly (caller ID arrives) when the call is
sent via
Note - looks like the answer to this was posted out of *date* sequence on asterisk.org (it is below the 1.2.0 release notice):
direct from asterisk.org homepage:
Version
1.0.10 has been released of Asterisk and Zaptel. Libpri,
Asterisk-addons, and Asterisk-sounds contain no
Hi,
I previously posted a problem with my Zyxel P2000Wv2 wireless SIP phones
and agent logins. In order to solve this problem I am looking at SIP
debugging tools but I have limited experience with them. Some of the
visual tools will not work as they require a software SIP phone to use
and
On 11/22/05, Michael George [EMAIL PROTECTED] wrote:
On Fri, Nov 18, 2005 at 10:22:23AM -0600, Kevin P. Fleming wrote:
Leo Burd wrote:
Any ideas about what is going on?
Yes. You didn't read the warnings prominently displayed at the end of
'make install' about removing old modules from
Hello open(ser) asterisk users
Here is what i expect to do :
Asterisk: registrar with public ip port=5050
open(ser): outbound proxy with public ip port=5060
Asterisk don't support IM and presence so i want to
use SER because of it's a good proxy:
I want user agents behind nat send
Hi,
I do not know if you got a reply to your questions already but I found
that only the version 2 of this phone with the latest firmware works.
There was a bug in the fireware where only numerical characters could be
used to log in. Alpha numeric will not work unless the firmware is
We suffer with some bad CO lines in the Seattle Redmond
area. To compensate our gains have been tuned 10 rx and 2 tx. We have also
had to add a 3 second wait to outgoing calls because many times the front of
the number gets missed by the telco. Is there anything we can request from
the
Claim that emergancy health equipment does not function, that will put
them in action. Better yet tell them that 911 is not captured!
On 11/22/05, Justin Selleck [EMAIL PROTECTED] wrote:
We suffer with some bad CO lines in the Seattle Redmond area. To compensate
our gains have been tuned 10
Let me get this straight
All you are doing is registering the devices with SER (below you have
mentioned asterisk, and then you say they goto ser)
Once they are registered to ser you wish to send them to asterisk...is
this correct
If so, this does not seem to hard, NAT ius dealt with in ser,
I am the network administrator for a small school in Michigan. We are
currently using an older proprietary pbx system and are trying very hard to get
away from this one vender lock in. I have set up an asterisk server using the
version 1.2 of asterisk. Our current system uses mailboxes and
--- Justin Selleck [EMAIL PROTECTED] wrote:
We suffer with some bad CO lines in the Seattle
Redmond area. To
compensate our gains have been tuned 10 rx and 2 tx.
We have also had
to add a 3 second wait to outgoing calls because
many times the front of
the number gets missed by the
On Tue, 2005-11-22 at 09:57 -0500, Johnathan Falk wrote:
I am the network administrator for a small school in Michigan. We are
currently using an older proprietary pbx system and are trying very
hard to get away from this one vender lock in. I have set up an
asterisk server using the version
Hi,
I have a problem with our office PBX where outgoing FXO Zap channels
get bridged and
i cannot receive or make any phonecalls.
First I disabled flash function and we are using # sign to do
transfers between internal lines
but it still happends from time to time.
So is there a way to specify
I tried this dial command to get a sound to play to the caller on answer.
I have even tried to use the LIMIT_CONNECT_FILE option with no success.
As can be seen below the start_sound variable shows 'UNDEF'.
Are there some other settings I have missed out, eg. file location, type etc.
The sound
You lost me here. Was that a question or a
statement?
I might not be able to help, since my SER usage is
totally diffent,
but let me see if I got this right:
- You want the SER to forward REGISTER messages to
the Asterisk.
- The user agents use private IP addresses.
- You want the
Have an application where Cisco phones are being used in a noisy
environmentlooking for some type of external ringer or amplifier so
users can hear the phones ringing over the background noise. Anyone
familiar with such a device?
Thanks,
--
Cory J Andrews
Partner / Purchasing
On Tuesday 22 November 2005 08:54, Chuck Bunn wrote:
What I do not understand is how dropped packets prevent the fax from
working. Faxes are designed to adopt to noise on the line by reducing
their connection speed. It seems like their is something else besides
packet loss going on here. Also
Let me get this straight
All you are doing is registering the devices with
SER (below you have
mentioned asterisk, and then you say they goto ser)
No to asterisk.
Asterisk should handle INVITE, REGISTER via ser.
SER should handle IM/presence
Once they are registered to ser you wish to
Andrew Latham [EMAIL PROTECTED] wrote:
Claim that emergancy health equipment does not function, that will put
them in action. Better yet tell them that 911 is not captured!
I'm going to have to remember that one!
Doug
--
Doug Meredith ([EMAIL PROTECTED])
SystemGuard - Oracle remote support
Here's what I'm trying to do.. We have a small system, there are
only two of us. We both do sales and we both do support. We like
Queues better than music on hold with a bunch of dials happening in
the background to try our phones, then cells, etc. Problem is, we
don't like the idea of
Ok that's a big DUH on my part. And since most people like to have 1xx or
2xx for extensions, this is going to be a continuing problem. If you have a
large menu, you are going to quickly run out of digits. Otherwise, is there
some trick I can use to move between contexts to avoid this problem?
I am the network administrator for a small school in Michigan. We are
currently using an older proprietary pbx system and are trying very hard to get
away from this one vender lock in. I have set up an asterisk server using the
version 1.2 of asterisk. Our current system has a master
Hi Doug,
hi list,
I installed spandsp-0.0.2 were the libspandsp.so.0.0.1 are included,
now I installed die spandsp-0.0.3 an you see:
the same problem - and now there is the libspandsp.so.0.0.2:
[app_txfax.so]Nov 22 16:15:38 WARNING[29448]: loader.c:325
__load_resource:
Has anyone had any success installing AMP 1.10 on a Asterisk
1.2.0.
If so can anyone shed some light on how to install it?
I am looking for an install or someone sort of script to run
the installation and I can t see it.
Any assistance would be appreciated.
Thanks.
I fed up with X100P clone card and want to spurge for a
better solution. I do not need a router or firewall within this device
and really just need basic features. I am considering ATA adapters such as the
Sipura 3000, Cisco ATA, Grandstream 488
or a Digium Wildcard TDM400P with one FXO.
okay, so ALL your users are registering to asterisk...is that correct.
If so the problem is howto accept users from behind a NAT into asterisk,
or am I confusing things further.
If the above are true, where is SER in this, or are users hitting SER
and you are sending the REGISTER from ser
I'm not following, must be too tired. Are you saying that on startup I could
run a SQL command that toggles everyone's call waiting status?
-Kerry
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro
Sent: Tuesday, November 22, 2005 5:16 AM
To:
Hey Cory,
I havent come across a voip ring amplifier or visual indicator. Here
are some amplifiers and visual inidicators for an environment using
ATAs:
http://www.soundbytes.com/Merchant2/merchant.mvc?Screen=CTGYStore_Code=SBCategory_Code=PhoneRingAmplifier
Omar A. SabekOn 11/22/05, Cory
okay, so ALL your users are registering to
asterisk...is that correct.
Correct via ser as outbound sip proxy
If so the problem is howto accept users from behind
a NAT into asterisk,
or am I confusing things further.
the problem is in contact field.
when user agents send register we
Hi,
I am running Asterisk 1.2 and zaptel 1.2 with the latest Digium board
version.
Thanks
Andrew Kohlsmith wrote:
On Tuesday 22 November 2005 08:54, Chuck Bunn wrote:
What I do not understand is how dropped packets prevent the fax from
working. Faxes are designed to adopt to noise on
Which FXO gateway is better and has better sound quality.
AudioCodes?
Or
Mediatrix.
Thanks for your input
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
We are putting in an Asterisk VoIP solution and was wondering what the best
communications medium would be for this implementation. We are going to
need 20 telephone lines in/out of our business. We currently have a data
T1. Could we put another data T1 to use for Asterisk, or would it be
Can anyone point me to the changelog for 1.0.10?
Craig
- Original Message -
From: Pedro [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, November 22, 2005 10:04 PM
Subject: [Asterisk-Users] Asterisk 1.0.10
On 11/22/05, Cory Andrews [EMAIL PROTECTED] wrote:
Have an application where Cisco phones are being used in a noisy
environmentlooking for some type of external ringer or amplifier so
users can hear the phones ringing over the background noise. Anyone
familiar with such a device?
What
There is alot of documentation available if you looked on their website.
http://aussievoip.com.au/tiki-index.php?page=1.10.008-Installation
On 11/22/05, Goran Donev [EMAIL PROTECTED] wrote:
Has anyone had any success installing AMP 1.10 on a Asterisk 1.2.0.
If so can anyone shed some
Kevin Hanson wrote:
I thought I read on the list some time ago that the default for
'priorityjumping' is 'yes' so that upgrading to 1.2 won't break old
dialplans. Can anyone confirm or deny?
That is absolutely correct; unless the [general] section of your
extensions.conf contains
Hi,
Got here and you will see an example of an automated login.
http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20AgentCallbackLogin
Also the AgentCallbackLogin can be passed parameters automatically when
the extension is dialed.
exten =
Okay almost there :-)
So UA --- asterisk --- SER --- UA
is that it
harry gaillac wrote:
okay, so ALL your users are registering to
asterisk...is that correct.
Correct via ser as outbound sip proxy
If so the problem is howto accept users from behind
a NAT into asterisk,
or am
Comeo'n AGI guys..
Please say something.
Hi,
Using AUTOHANGUP, I can force a call duration within a time limit.
I would like to playback a message before 1 minute of autohangup.
How can I accomplish it?
Would anybody please give me right direction.
Thanks,
You don't have any
Hi Johnathan -
I am the network administrator for a small school in Michigan. We are
currently using an older proprietary pbx system and are trying very hard to
get away from this one vender lock in. I have set up an asterisk server
using the version 1.2 of asterisk. Our current system has
Johnathan,
I am also located in michigan. Maybee there is a way i can help with
this project. I currently use asterisk for alot of custom
applications. let me know i'll send you my phone number outside the list
Johnathan Falk wrote:
I am the network administrator for a small school in
Those came from astbill. I will make the changes and reupload, I have
gotten a few more changes as well.. Thanks :)
On Mon, 2005-11-21 at 22:03 -0800, Innocent Evil wrote:
Lots of country have wrong prefix.
Andorra,376 should be 1376
Angola,244should be 1244
Antarctica,6721
Doug/Peter/Others,
You do realize that you've all just violated your Terms of Service for
VoipJet right? Read: https://www.voipjet.com/tos.php
Now, go down to near the middle where it says:
NON-DISCLOSURE: ALL CUSTOMERS USING VOIPJET'S SERVICE ARE SPECIFICALLY
PROHIBITED FROM DISCLOSING TO
I personally prefer a single-box solution with a TDM400
(although I am one of the rare people who haven't had problems with the X100Ps I
have put in). My office uses an SPA-3000 on a phone line that has call forward
on busy to an iax.cc DID line, therefore I speak from experience with both
why don't you just build your cells into the queues and setup the queue
to ringall.
Jason Lixfeld wrote:
Here's what I'm trying to do.. We have a small system, there are
only two of us. We both do sales and we both do support. We like
Queues better than music on hold with a bunch of
Sometime this winter we want to move our company to asterisk from a very
old comdial executech phone system.
At this point I have a system setup at home that we've been using for
several months.
I've tried the grandstream bt101 but have had problems keeping it
working - some days the message
Andrew Kohlsmith wrote:
Faxes are designed to work around the noise and other signal problems inherent
in analog telephony. VOIP introduces an entirely different set of noise
factors that fax machines are frankly ill-equipped to deal with. Jitter and
dropped packets are the biggest of these
Just one thing,
Register the Uas to asterisk also as outbound proxy.
Asterisk will register to SER all the Uas.
We use this design:
Ua --Asterisk(NAT)-- Ser(public Ip)-- where do you want to go
It works perfectly.
Maybe I miss something?
Olivier
-Message d'origine-
De : [EMAIL
First, you have to configure your zapata.conf
sip.conf to support your hardware (see http://www.voip-info.org/wiki/index.php?page=Asteriskand
read
http://www.nufone.net/downloads/asteriskdocs/AsteriskTFOT.zip,this
last is a must-read one)
After that, you have to see if all incoming calls
We have tried both but given
up hope about them. So now we only use Quintum DX series. Amazing machine
Anders Svensson Bobas
Communication
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Goran Donev
Sent: den 22 november 2005 16:41
To:
Is there a way to install Asterisk from source and not stomp on your
already existing Asterisk installation? I don't see a configure
script and it looks like it's trying to find stuff in /etc/asterisk and
in /usr/lib/asterisk and probably other places.
- Jeremy Jones
I do it in reverse do all registration in SER since that is was it was
designed for, and then pass to asterisk, and in 1.2 asterisk it has a
slew of new features to help with SIP methods, having said that I havent
got round to testing any :-)
iqbal
Olivier Taylor wrote:
Just one thing,
Hello,
I have my SIP clients registered with names, and I want to implement the
voicemail in my Asterisk.
I have these lines to redirect the call to the voicemail:
exten = pereira,1,Answer
exten = pereira,2,Wait(1)
exten = pereira,3,VoiceMail(u${EXTEN})
exten = pereira,4,Playback(vm-goodbye)
I was looking for something off the shelf, this is a one off
application, and limited in scope I think they have about a dozen or so
handsets in a noisy area they need to beef up the ring volume or present
some visual indicator on an incoming call.
Cory J Andrews
Partner / Purchasing
--Original Message Text---
From: cp
Date: Tue, 22 Nov 2005 10:25:20 -0500
I fed up with X100P clone card and want to spurge for a better
solution. I do not need a router or firewall within this device and
really just need basic features. I am considering ATA adapters such as
the Sipura 3000,
Hello Jason,
if the system is so simple, why don't you connect the queue straight to a
couple of you terminals, i.e. not to Agent/101 but to SIP/214. This way
you have no login/logout.
Yours,
l.
On Tue, 22 Nov 2005 16:20:50 +0100, Jason Lixfeld
[EMAIL PROTECTED] wrote:
Here's what
I brought this up a while back and althought there are pieces that
interface * into Fax Telephony applications, there hasn't been something
that works with plain old analog modems.
Then I found this piece of code. From my initial tests it looks solid,
but I have no clue in how to interface this
On 11/22/05, Justin Selleck [EMAIL PROTECTED] wrote:
We suffer with some bad CO lines in the Seattle Redmond area. To compensate
our gains have been tuned 10 rx and 2 tx. We have also had to add a 3
second wait to outgoing calls because many times the front of the number
gets missed by
I have had the same issue with a PRI connected from asterisk to an
avaya system, it first worked fine, but then started doing this, what
is happening is that the D-channel is getting reset for some reason (I
have no clue why, but I was able to reproduce it between the avaya,
when CID Name was
May I recommend www.numberingplans.com as a resource for checking
international dial codes and indeed doing a reverse lookup to find out about
a number. We have used this as a resource in the past.
Regards
Neil
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
Also.. All that is required of the phone company is a minimum line
quality, anything else is at their pleasure. And if you want to push
them a little call up and enter the option to cancel your service.
That is the *fastest* way to get to people who can actually do
something for you as I found
On 11/22/05, Bharath Khambadkone [EMAIL PROTECTED] wrote:
Hello All,
I'm fairly new to asterisk. I have read about the problems about NAT, But
can't seem to find a solution.
My Asterisk is on a public domain, there is no NAT or firewall in front of
If no nat then why do you have nat=1 in
On 22/11/05, Matt [EMAIL PROTECTED] wrote:
Doug/Peter/Others,
You do realize that you've all just violated your Terms of Service for
VoipJet right? Read: https://www.voipjet.com/tos.php
Now, go down to near the middle where it says:
NON-DISCLOSURE: ALL CUSTOMERS USING VOIPJET'S SERVICE
On 11/22/05, Jason Lixfeld [EMAIL PROTECTED] wrote:
Here's what I'm trying to do.. We have a small system, there are
only two of us. We both do sales and we both do support. We like
Queues better than music on hold with a bunch of dials happening in
the background to try our phones, then
Has anyone tried the newest Polycom firmware? The release notes
indicate they have added support for a new BLA draft.
TIA,
Kevin
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My p2000w Works with asterisk.
Here is the sip.conf entry
[1006]
type= friend
subscribecontext = all-local
accountcode = 1006
amaflags= default
username= 1006
secret = whatever
host= dynamic
language= en
dtmfmode=
You could use a Digium TDM2422B, which has 8FXS and 8FXO, and leaves you
8 ports past that for future FXS or FXO expansion. That card, with a
normal Asterisk rackmount or tower server, and a mini patch panel and
amphenol cable I would think would do the trick.
For phones, I would suggest the
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