Like Carlos said, the SuperMicro are very good servers, however I don't have a VAR I could recommend on those, as I assume you don't want to put them together.I'll also recommend Tyan, we use some of their 1U gear and its been working flawlessly, but again no VAR to recommend as we build them
Then what would be a better solution?On 10/30/06, Pavel Jezek [EMAIL PROTECTED] wrote:
this is really ugly workaround, because using r option in dial youlose any other progress tones, including busy, congestion, and you willalways hear ring tone even in case of congestion...PJ
Michiel van Baak
Hi guys,
Im having a really strange problem, which Im
pretty sure has only appeared since my last upgrade (1.2.12.1) .
Its about NAT and Qualify. Im using
Asterisk to register with some external SIP providers. However, theyre
always marked as UNREACHABLE, when they werent before!
omar parihuana wrote:
Hi Folks,
I would like to recover all information about the calls, incoming
calls, call time, call history, etc in a Web Format, are there some
open source aplication for Asterisk that be easier for use. Pls
anything suggestion will be very appreciate.
In my experience DTMF works reliably only when sent over RTP using rfc2833. If you are using SIP, put this line under [general] section in sip.conf: dtmfmode = rfc2833. If you don't want to put this in [general], you can also put dtmfmode = rfc2833 in the declaration of each individual extension
And yes, you need to do the same for the phones or adapters you are using. They also have the various options for DTMF setup.
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To UNSUBSCRIBE or update options visit:
I have a SNOM 360 and a SNOM 300 on a lan. They
both are connecting to a server with a public IP outside of the lan (dedicated
server). When the 300 is on alone it works inboud and outbound. When they are
both plugged in then the 360 will call out and in and the 300 will only allow
inbound.
I concur with Conrad. Cisco phones were retrofitted for SIP, whereas
Snom phone are built around, and expressly for, the SIP standard. To be
in compliance with Cisco regs, you are also supposed to have a SIP User
license and a Smartnet contract for each phone if you abide by their
program. I
I dont know the name of the file, but you can do it customly in asterisk
Exten = X,1,Dial(SIP/1234ZAP/1/18005551212)
- Original Message -
From: Stephen Bosch [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday,
Brad Templeton wrote:
On Tue, Oct 31, 2006 at 07:40:35PM +0800, Leo Ann Boon wrote:
Have you tried setting the externalip and localnet parameters?
Localnet makes some sense, and is set (should be the default anyway, no?)
I don't think it's set by default. Anyone know how
On Tue, Oct 31, 2006 at 04:51:44PM -0500, C F wrote:
The correct behaviour, as I see it is:
a) Native bridge when connecting two external channels -- everybody is
on the real internet
It might not work if one of them is NATed. Therefore the correct way
to do this is to use
- Original Message -
From: Stephen Bosch [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, October 31, 2006 10:06 PM
Subject: Re: [asterisk-users] simultaneous ring - call groups or queues
orsomething else?
I've been doing a lot of reading over the last few weeks on Asterisk,
and will be implementing a test system this week to play with.
I've got two questions in regards to the ideal implementation for our
company. First, has anyone written any drivers to interface with
proprietary phones?
Also when I do sip show peers I get
sip show
peersName/username
Host Dyn Nat
ACL Port Status
sipmedia/XX
69.1.236.33
5060
Unmonitored10307/10307
65.8.212.215 D
N 60414 OK (147
ms)10305/10305
(Unspecified) D N
0 UNKNOWN
10306/10306
65.8.212.215 D
N 60414 OK (135
I knew I should've waited til tomorrow to send the e-mail so I could have a nights thought on the subject.
That being said, scratch the FXO/FXS thing, what I really picture is someway of passing proprietary information through the Asterisk PBX's on both ends to get remote locations on our
I've been losing patience with my current provider, a small company
called Sellvoip. Their termination is good, and they are
asterisk based, but they are understaffed and have no concept
of customer service. So I'm shopping.
I am interested in the opinions of others on the providers they
work
of course you can always use http://cacti.net/download_cacti.php
On 10/31/06, Andrew Joakimsen [EMAIL PROTECTED] wrote:
Check out voip-info.org, there are quite a few GUIS some even generate nice
graphs!
On 10/31/06, omar parihuana [EMAIL PROTECTED] wrote:
Hi Folks,
I would like to
I am running 1.4.0-beta2
Date: Tue, 31 Oct 2006 10:57:06 -0600 (CST)From: Anthony LaMantia [EMAIL PROTECTED]Subject: Re: [asterisk-users] Cisco 7960 Skinny calling SIP phone
To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com
Message-ID: [EMAIL
On Sun, 2006-10-29 at 22:41 +0100, Dominique Dartois [EMAIL PROTECTED]
wrote:
Hi,
I am using Asterisk 1.2.12.1 + the AEL2 patch.
If I use a variable instead of the extension itself, an
incoming call cannot
be connected.
${ID-FST1} =
I have a new client who has an existing Asterisk PABX and is looking
for us to install a TE110P for him, However he has a Dell SC420 and I
have never used one before.
I have had no problems with any other Dell servers which we use almost
exclusively.
Has anyone had any good/bad experiences
Andrew Joakimsen wrote:
Where are these DTMF tones going? From where? Be specific, post the
relevant
config file sections I can't read minds and I'd be surprised if 0.1% of
the people who read this can either
On 10/31/06, Jason Walker [EMAIL PROTECTED] wrote:
I have tried beta2,
On Mon, 2006-10-30 at 02:28 -0800, Pezhman Lali [EMAIL PROTECTED]
wrote:
Hi Dear
I want to use asterisk(1.2.7.1) as a router by caller
id.
I have only a DID number, I want to map this number to
some ip-phones , base on received
Then what would be a better solution?
Usually the IAX phone will play you a ring tone until the other end
answer. If you're phone doesn't do it, then it is a flaw in that
phone.
What phone is this ?
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http://www.digium.com/en/docs/misc/compatibility_notes.phpServer Compatibility The following list of servers are known
to be partially incompatible with Digium hardware. We do not recommend
using the following computers to set up an Asterisk server: Dell PowerEdge 1600Dell PowerEdge SC 420Dell
On Sun, Oct 29, 2006 at 04:47:48PM +0800, Will Roy wrote: I originally built my Asterisk server without installing the Zaptel package as it was going to be a purely SIP based system. However when I went to
setup conferencing using meetme I found out that app_meetme is dependant on the ztdummy for
You'll find the cost of a PRI varies dramatically from one telco to
another. I've heard numbers in one case where three analog pstn lines
cost the same as a PRI, another case where 16 analog pstn lines cost the
same as a PRI. And, having worked in the telecomm industry for many
years, there
Joe
While having done this before I am rather unhappy with current
offerings. A 4U or bigger HP DL5XX for example will run you upwards
of 10k. While they are some nice machines I am currently building my
own. You may want to contact Rhino about there new servers, I feel
that they are filling
My vote is definitely for Snom, I've worked with Cisco phones for
years, but the Snom is much better integrated, and the feature buttons
can be retooled for any environment, making custom installs very easy.
On 10/31/06, Conrad Wood [EMAIL PROTECTED] wrote:
On Tue, 2006-10-31 at 13:29 -0600,
- Original Message -
From: Ken Williams [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, November 01, 2006 2:10 AM
Subject: [asterisk-users] Newbie Questions
I've been doing a lot of reading over the last
Please see my previous email in regards to connecting the Comdial system to
asterisk. In regards to connecting Asterisk to the internet you do not need an
FXP. All you need is a NIC and to have a good connection to the internet. If you
can get Asterisk to talk to the Comdial system you
Your router might have a problem if there are several
devices behind NAT with the same port number. Either explicitly set the ports on
the phone (SIP,RTP, and risk that other ports like DNS, NTP, ... will have
the same problem)or buy another router that implements NAT/PAT
properly.
CS
You can put the Asterisk system in front (i.e., betweenthe PSTNand your Comdial system). This will let Asterisk choose whether the call should go out over the PSTN or the Internet using VoIP.
You would use the same for the second location, provided that is a complete Comdial system. You could
On 2006-10-31 17:29:47 -0800, Brad Templeton [EMAIL PROTECTED] said:
I've been losing patience with my current provider, a small company
called Sellvoip. Their termination is good, and they are
asterisk based, but they are understaffed and have no concept
of customer service. So I'm
Thank you for you responses re: my question on the architecture of Asterisk. Olle, your explanation was especially useful. I still feel like I'm missing a crucial part here. If Asterisk is an endpoint, and according to the example you gave in your response (if u1 hangs up then Asterisk decides to
On 10/31/06, Leo Ann Boon [EMAIL PROTECTED] wrote:
Rajkumar S wrote:
-- Executing Queue(SIP/1002-74e9, Auth-Enq|t) in new stack
-- Started music on hold, class 'default', on channel 'SIP/1002-74e9'
-- outgoing agentcall, to agent '1001', on
'Local/[EMAIL PROTECTED],1'
--
Hi Jamie -
Has anyone here reprogrammed their Polycom features keys using
sip/ipmid.cfg?
If so I would be really grateful if someone could send me an
example
Here's the keys line that I use for one of my clients:
keys key.scrolling.timeout=1
key.IP_500.37.function.prim=DialpadPound
On Wed, Nov 01, 2006 at 08:10:29AM +0800, Leo Ann Boon wrote:
Brad Templeton wrote:
The way I understand it, externalip and localnet work hand-in-hand. I do
agree with you that this is commonly used for Asterisk behind a NAT. I
believe these parameter just helps asterisk determine what
On Tue, Oct 31, 2006 at 08:03:56PM -0800, Martin Joseph wrote:
On 2006-10-31 17:29:47 -0800, Brad Templeton [EMAIL PROTECTED]
said:
I've been losing patience with my current provider, a small company
called Sellvoip. Their termination is good, and they are
asterisk based, but they are
Rajkumar S wrote:
On 10/31/06, Leo Ann Boon [EMAIL PROTECTED] wrote:
Someone correct me if I'm wrong: The Dial string is missing a '/n'
parameter for the Local channel. Without /n, Asterisk will do a native
transfer to SIP/1001 and lose the timeout value defined earlier.
What does '/n' refer
I have a Snom 360 phone that will not work on an Asterisk
server but it will on another server. This phone has been working for over 4
months or so. I can not figure it out. This is the only Snom phone that I
have so I can check it against another one. The PBX that fails, fails with any
I have a Snom 360 phone that will not work on an Asterisk
server but it will on another server. This phone has been working for
over 4 months or so. I can not figure it out. This is the only Snom
phone that I have so I can check it against another one. The PBX that
fails, fails with any
On 11/1/06, Leo Ann Boon [EMAIL PROTECTED] wrote:
Rajkumar S wrote:
On 10/31/06, Leo Ann Boon [EMAIL PROTECTED] wrote:
Someone correct me if I'm wrong: The Dial string is missing a '/n'
parameter for the Local channel. Without /n, Asterisk will do a native
transfer to SIP/1001 and lose the
Thank you Jessee,
Firmware seems to be recent(4.80A.025.004).
For 'noisy', I mean IP Phone -- * -- MP-114 side.
Audio quality of MP-114 -- PSTN -- Analog phone is
good.
I think it can be power ground or gain problem.
Any experience?
Thanks,
Jason
--- Jessee J Holmes [EMAIL PROTECTED] wrote:
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