For lack of a better term, I have been tasked with creating a dial tree
crawler. The reason is that we have a soon to fail Octel system. The major
issue is that there is no way to port the dial tree recordings from the
Octel. So what I envision is creating a script that can somehow dial down
the
I wanted to see if there was anything reasonable in price out there yet that
performed an RF to IP bridge via asterisk. What I mean by this is callers
from PSTN can be patched to a UHF/VHF radio and vis-à-vis. I know there is
an option available for the Avaya systems but its a little out of the
@lists.digium.com
Subject: Re: [asterisk-users] RF to IP bridge
Curt Shaffer wrote:
I wanted to see if there was anything reasonable in price out there
yet that performed an RF to IP bridge via asterisk. What I mean by
this is callers from PSTN can be patched to a UHF/VHF radio and
vis-à-vis. I know
Half duplex is not an issue. Basically the idea is radio over IP. I don't
want to change the fact that we are using radios. For example, on an
enterprise level I'm going to be working with a crew to set this up for our
Avaya system. It is basically for emergency communications. Say the fire
chief
: [asterisk-users] RF to IP bridge
Quoting Curt Shaffer [EMAIL PROTECTED]:
I wanted to see if there was anything reasonable in price out there yet
that
performed an RF to IP bridge via asterisk. What I mean by this is callers
from PSTN can be patched to a UHF/VHF radio and vis-à-vis. I know
With all of the recent talk on the list about DUNDi, I have a question. From
the outset it appears that SER is often used for high availability solutions
and as a tool for almost clustering Asterisk boxes behind it. It appears to
me that DUNDi is providing a lot of this as well. Now I know DUNDi
I am experiencing a situation where I am getting intermittent choppy audio.
Here is the network layout:
Termination provider - IAX2 over the Internet - 20Mb fiber connection -
router - Asterisk
My ATA connection goes into the router between the fiber and the Asterisk
server on another
I have been working with a couple companies who are interested in
integrating Cisco VoIP (mostly call manager express) but utilizing Asterisk
for AA, VM, failover trunks etc. I have found some materials and guidance
out there but I think a list and/or wiki for general asterisk integration
with
Thanks a million! Just verified after putting it in my encrypted configs and
it works like a charm! :)
Curt
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dave Fullerton
Sent: Monday, March 12, 2007 12:15 PM
To: Asterisk Users Mailing List -
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steven Totaro
Sent: Monday, February 26, 2007 11:11 PM
To: asterisk-users@lists.digium.com
Subject: RE: [asterisk-users] To use asterisk or proprietary hardware,that
is the questio
From: shadowym [EMAIL
Found my answer for those who would like to know:
Profile Rule: [--key $A]http://your.addre.ss/$B/$MAC.cfg
GPP A: urtopsecretultrasecureaesencryptionkey
GPP B: OddBallDirectory123098
Hope that helps someone!
Curt
-Original Message-
From: Curt Shaffer [mailto:[EMAIL PROTECTED]
Sent
Just emailing the list to see if anyone out there has used Vitelity? If so
what has been your experience with service, support etc?
Thanks
Curt
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or
I put iptables on my asterisk box and an odd thing occurs. I allow 5060 and
1-2. As soon as I start iptables and make a call it literally takes
60-90 seconds before the call even starts to ring. As soon as I shut
iptables off, the call goes through immediately again. Its quite odd. The
: [asterisk-users] odd issue with IP tables
Post your IP tables configuration here if it isn't too big.
Ron
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Curt Shaffer
Sent: Saturday, November 18, 2006 5:05 PM
To: 'Asterisk Users Mailing
Nope the server is for Asterisk only. I have SSH on it for management,
FreePBX for configuration, SIP clients and IAX termination.
-Original Message-
From: Ron McLeod [mailto:[EMAIL PROTECTED]
Sent: Saturday, November 18, 2006 8:06 PM
To: 'Curt Shaffer'; 'Asterisk Users Mailing List
We have had good results mostly from this unit except for one issue that is
currently being looked into by Aastra. The issue is if a second call comes
in and the cordless answers then puts the call on hold audio drops one way
on the handset. Aastra was able to reproduce this and is working on it.
I am looking for an ATA that has had very reliable results
when passing FAX over IP. I was thinking of testing the Cisco (not Linksys) ATA
186 I1, ATA 186 I2, ATA 188 I1. This is what Im looking for:
FAX - PTSN - through Asterisk - ATA - Fax
Machine.
I have QoS from PSTN entry to
Has anyone out there implemented a system that does call routing via Cisco
gear but VM for everyone on the system via Asterisk? What have been your
successes and failures or issues?
Thanks
Curt
___
--Bandwidth and Colocation provided by Easynews.com
I was just going to test out the new Asterisk 1.4 GUI. I
downloaded it from source make;make install. I added my http.conf and modified
manager.conf. I restarted Asterisk and did a make checkconfig and it says
everything looks good. But I notice that the port 8088 is not listening when I
of integration?Cheers,DeanFrom: Curt Shaffer [mailto:cshaffer at gmail.com] Sent: Tuesday, November 07, 2006 11:08 PMTo: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: RE: [asterisk-users] Microsoft will enter VoIP market inearnestnextyear, says
Take a look at OVA.
mms://wm.microsoft.com/ms/exchange/2007/Phone_Based_User_Experience_With_Outlook_Voice_Access_300k.wmv
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alex Robar
Sent: Tuesday, November 07, 2006
9:13 PM
To: Asterisk
Users Mailing List -
I'm the friend mentioned here.
I am using the Aastra 480i CT. It is SIP to my PBX and IAX termination from
the PBX to my provider. My issue has a slight twist to it but the same
result. For instance his is always where as mine is frequent but not always.
After I got to finally see it first hand
is happening.
And I also wanted to add that I am running 1.4.0 firmware for this phone.
Thanks again!
-Original Message-
From: Curt Shaffer [mailto:[EMAIL PROTECTED]
Sent: Monday, November 06, 2006 6:58 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [asterisk-users
I am having an issue with doing FTP auto provisioning of
Polycom 501s when they are behind a NAT. If I put the phone on the same
subnet as the provision server it loads the configs and changes fine but as
soon as I put in behind a NAT it comes up with cannot contact boot server. I
have
autoprovision behind a NAT
I
can confirm that the linksys routers cause ftp problems. Is your FTP
server set to use pasive mode?
-rb
-Original Message-
From: Curt Shaffer [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List -
Non-Commercial Discussion'
asterisk-users
is working. It might be informative.
-Original Message-
From: Curt Shaffer [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List -
Non-Commercial Discussion'
asterisk-users@lists.digium.com
Date: Mon, 6 Nov 2006 20:17:07 -0600
Subject: RE: [asterisk-users] Polycom autoprovision behind a NAT
-Commercial Discussion
Subject: RE: [asterisk-users]
Polycom autoprovision behind a NAT
I'm not sure. We ended up putting in a d-link router
to get around the ftp problem. In most of our sites we have netscreen 5gt
routers and they work fine.
-Original Message-
From: Curt Shaffer
02, 2006 9:22 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] light web user interface
Curt Shaffer ([EMAIL PROTECTED]) wrote:
Basically I would like a page that would allow a user to log in and modify
their extension only. So for example, I log
Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users]
light web user interface
What attributes are you talking about ? Depending on what
they are it may be real simple to set something up.
- Original Message -
From: Curt Shaffer
To: 'Asterisk Users
Does anyone know of a really lightweight web interface that
allows users to log in and modify attributes of their extension only?
Thanks
Curt
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To
I just wanted to ask a general question to anyone that
serves as a service provider on the list out there. Are you using OpenSER and
Asterisk for your high availability and redundancy or DUNDI? Anyone have anything
to say as to which would be better for a service provider and why?
I have been struggling over central provisioning for quite
some time. I have eagerly watched each post with like problems but have yet to
find a reliable answer.
I have a Polycom 501 and I am trying to provision from an
FTP server, and just to take any routing out of the issue it is on
Do you mean .cfg and sip.cfg? Could you clarify for me please and I will try that. Thanks for the suggestion.
Curt
On 10/23/06, Andrew Joakimsen [EMAIL PROTECTED] wrote:
What if you just use the default configuration files?
On 10/23/06, Curt Shaffer
[EMAIL PROTECTED] wrote:
I have
It does boot with the defaults. Is this pointing at a corrupt config?
-Original Message-
From: Ivan Fetch [mailto:[EMAIL PROTECTED]
Sent: Monday, October 23, 2006 6:31 PM
To: Curt Shaffer
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Ivan Fetch
Sent: Monday, 23 October 2006 7:31 PM
To: Curt Shaffer
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom provision errors still! Arg
errors still! Arg!
No probs, maybe you should donate $5 to kerry's site to cover hosting
fees?
Cheers,
Dean
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Curt Shaffer
Sent: Monday, 23 October 2006 9:30 PM
To: 'Asterisk Users
I have an interesting issue. I have an Aastra 480i CT (the
one with the handset and the cordless). Here is the scenario:
Caller 1 calls in and the person on the handset answers the
call.
Caller 2 calls in and the person with the cordless answers
the call on the second line (because we
Can anyone point me to a good source for provisioning WRT54GP2 from a central server?
Thanks
Curt
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
Here is my extensions_custom.conf. The WakeUp context will
not work. If I change the context name to say, CRAP, it works like a charm. Can
anyone explain this?
[from-internal-custom]
exten = 1234,1,Playback(demo-congrats) ;
extensions can dial 1234
exten = 1234,2,Hangup()
exten =
]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle
Sent: Tuesday, September 05, 2006 4:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] config include issues
Curt Shaffer wrote:
Here is my
OK, so I had an issue where I needed to add a w when dialing
out my POTS line. But now when the calls go out my VoIP providers the w makes
the call fail. I am using freePBX and the only place I found to change this was
in the extensions.conf which makes it global. Am I missing something
This post spurred off of the comment of Michael Collins on
the Asterisk with PABX thread. I am going to post the relevant information
here:
I started w/ AAH,
then went back and learned the dialplan apps, scripting, etc. For some guys
like me, it's easier to start with a working (if
I remember the config edit from [EMAIL PROTECTED] but I
do not have it on my freePBX now. I dont mind using vi, Im very
comfortable in Linux. Thanks for the answers!
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael Collins
Sent: Monday, August 28, 2006 3:29
All,
I am looking at taking on a project for a hotel that is
using Teledex systems. I see that they have a SIP based phone and the information
says that there is some CMS server part that appears to be the brains behind
the device. My questions are; has anyone out there used this type
I posted earlier about an application not found error. I
have manually pointed the phone at the server but it just does not seem to ever
even hit it. I am going to do some network captures here soon after I walk away
from this computer for a while. But here is another question which I am
Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric
ManxPower Wieling
Sent: Tuesday, August 15, 2006 12:36 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Zap difficulties
Curt Shaffer wrote:
I am having a weird issue with my zap
I have a company that I am going to be moving away from a
legacy PBX to Asterisk. They use page zones pretty heavy and I would like to
keep that functionality. Basically when someone is not at their desk the
receptionist pages all of the phones, telling them there is a call. Does anyone
I was just talking with an unnamed provider and the guy told
me that they recommend their users not to use IAX because it is unstable at 50
concurrent calls and unusable at 100 or more calls. Now I have personally
worked on an asterisk box that was pushing more than 50 and there were no
OK, I may have done something stupid. I was trying to
upgrade my Polycom to the newest firmware I could find (1.6.7). I am also
trying to get provisioning working from a central server. I tired to reset with
holding 468* down and it kept the settings the phone had on the phone. From
what I
I am having a weird issue with my zap channel (Digium TDM01B).
Randomly it appears that the POTS line is not seeing all of the digits passed. We
have to dial a 1 and the area code to call most numbers here, and we get the
error that we need to dial a 1 and the area code when dialing this
Ok, now this may be my lack of understanding on the stats
readout of the IAX command but can someone explain the following:
I just had two calls going and did an iax2 show channels,
the lag for both was 0ms and the jitter was -0001ms. How is that possible?
Am I wrong that the lag is
I walked into a new potential * install yesterday. They are
running a Samsung Prostar DCS. Does anyone have any experience with these out
there that you could relay some things to look out for when integrating this
until the migration is complete? Or what would be the best way to integrate
And, someone correct me if I am wrong here, you want to make sure RTP is
getting quality as well. SIP is setting up, tearing down, and a few other
things but RTP is where the conversation is taking place.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
: [asterisk-users] Asterisk VOIP / Mikrotik
On Jul 28, 2006, at 10:55 AM, Curt Shaffer wrote:
And, someone correct me if I am wrong here, you want to make sure RTP
is getting quality as well. SIP is setting up, tearing down, and a few
other things but RTP is where the conversation is taking place
] Asterisk VOIP / Mikrotik
On Jul 28, 2006, at 12:12 PM, Curt Shaffer wrote:
Is it choppy internal or only over the trunk or both?
And as far as helping RTP, it should be as simple as adding the
ports to
your Queue. 1000-2000 by default I believe but you can check your
rtp.conf
file
I was just wondering if anyone out there has tried vitelity for
VoIP service If you did what is your story with how good/bad they are?
Thanks!
Curt
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing
I have a question. We are going to attempt mixing some SIP
and H323 solutions here. The H323 is possibly going to be phased out sooner or
later but this is the first step. I have set up an Asterisk server that is also
running GnuGK so we have one machine doing both SIP and acting as a
I have a requirement to set up an Asterisk server that will
handle H323. In the end this is used for video conferencing but it will be
transitioning other H323 devices to SIP at some point. My question is this:
Does anyone know of or have good documentation that explains how this
] On Behalf Of Curt Shaffer
Sent: Tuesday, 11 July 2006 10:47 AM
To: 'Development discussion of video media support in Asterisk'
Subject: RE: [Asterisk-video] Asterisk as an MCU
Thanks for the clarification. So if I want some functionality of an
MCU I
could use Asterisk as long as the clients were
Getting them here too.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mike Fedyk
Sent: Tuesday, June 27, 2006 5:49 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Mail loop?
Is anyone else getting messages from
Has anyone out there figured out how to emulate the Cisco
SRST functionality with *? If so would you mind letting me know the best
practices for this?
Thanks
Curt
___
--Bandwidth and Colocation provided by Easynews.com --
Does anyone out there have a recommendation for tools that
will monitor the quality of VoIP systems? I am looking for jitter and MOS
monitoring. I have a custom Nagios plugin that is alerting me if the jitter
jumps out of a 20ms but I am looking for a little more detail. I would not be
] Quality monitoring
Care to share your Nagios plugin?
Regards,
--
Anthony Rodgers
Business Systems Analyst
District of North Vancouver
Web: http://www.dnv.org
RSS Feed: http://www.dnv.org/rss.asp
On Jun 22, 2006, at 9:53 AM, Curt Shaffer wrote:
Does anyone out there have a recommendation for tools
I have been in the process of trying to troubleshoot a phone
system that is doing IAX trunking to a provider. The average RTA is 75ms with
spikes from time to time and jitter from time to time as well. My question is
this; How much can one trust this types of samples when going over the
I am trying to use an IAX softphone (ESCSoftphone) from my
Treo 700w. The qualify time is around 173ms. I have only tried setting
jitterbuffer=yes in the iax.conf config but the sound is ridiculous. The echo
is horrible and there is a screeching in the background on the receive end. Is
I have a situation when I dial out my Zaptel I am getting a
recording that I need to add a 1 or a 0 and the area code with this number. I
have tried appending this and the number going out the zap is 1NXXNXX so it
is going out with 1 and the area code. Someone has suggested that maybe
I am trying to perform a rollover when the
primary number is busy. This is coming from a POTS line. Apparently I need call
waiting on the POTS line as I get immediate busy from the FXS if I dont
have it. So my question is this. I have an Aastra 480I CT. The call forward
when busy here
I am using asterisk version 1.2.6 with Zaptel version 1.2.5.
I have a POTs line coming into a Digium TDM01B. It appears to not be getting
CID at all. If I hook up a POTS phone to the line CID comes through fine. Inbound
and outbound calls work fine but there is just no CID on inbound for
/9/06, Curt Shaffer [EMAIL PROTECTED] wrote:
I am using asterisk version 1.2.6 with Zaptel version 1.2.5. I have a POTs line coming into a Digium TDM01B. It appears to not be getting CID at all. If I hook up a POTS phone to the line CID comes through fine. Inbound and outbound calls work fine
at it and maybe see if there is a problem.
I'm guessing you want help with this.
On 6/9/06, Curt Shaffer [EMAIL PROTECTED] wrote:
I am using asterisk version 1.2.6 with Zaptel version 1.2.5. I have a POTs
line coming into a Digium TDM01B. It appears to not be getting CID at all.
If I hook up a POTS
usecallerid=yes
and caller id works. Not sure if that's the problem or not.
On 6/9/06, Curt Shaffer [EMAIL PROTECTED] wrote:
[channels]
language=en
#include zapata_additional.conf
context=from-zaptel
signalling=fxs_ks
faxdetect=incoming
usecallerid=asreceived
echocancel=yes
callprogress
I used a scenario like this before but I always ran into intermittent echo
issues that were just not worth the hassle for me so I switched to a sole IP
origination and termination service.
Just my personal experience!
HTH
Curt
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Curt Shaffer
Sent: Tuesday, June 06, 2006 10:37 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Vonage and FXO
I used a scenario like this before but I always ran into intermittent echo
issues
I too had the same problems. If you find out the best way for this let me
know!
Thanks
Curt
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Stephen Bosch
Sent: Sunday, June 04, 2006 4:06 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
,
then obviously something is not right in FreePBX. If it doesnt' then
that indicates your configuration files need tweaking.
Thanks,
Steve
Curt Shaffer wrote:
Here is the output from a dial when starting asterisk with -v. The
1NXXNXX is actually the number not those characters FYI.
Thanks
Does anyone out there have a sample config they can share
for the Polycom 501? Is it possible to do sub configs like you
can with the Aastra 9133i? It could be just me but the boot configs seem a bit
cryptic compared to the aastra. Also do any of you have any comparisons between
these and
PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro
Sent: Sunday, May 28, 2006 5:47 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] TDM
Connect to the Asterisk console with verbose turned on and try to dial.
Post that output.
Curt Shaffer
PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro
Sent: Sunday, May 28, 2006 5:47 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] TDM
Connect to the Asterisk console with verbose turned on and try to dial.
Post that output.
Curt Shaffer
The TDM01B is 4 port capable but has only 1 FXO module. Im
running asterisk 1.2.7 and zaptel 1.2.5. I cannot seem to get the TDM01B
working. When I do the zttool it shows 4/1/0. I can dial out from a POTS phone
up to the point that the cable plugs into the card.
Here is my
into the correct jack?
With only one module installed, the other three jacks lead to nowhere.
Also this seems to be [EMAIL PROTECTED] from the references, so perhaps
there is a context issue that the configuration files address.
AAH can really lead one down the garden path!
John Novack
Curt Shaffer
All,
I have been tasked with setting up video conferencing
utilizing asterisk. One of the requirements is a softset that has video
capabilities. Eyebeam looks promising but I was just wondering if anyone out
there knows of any freeware with comparable features of Eyebeam that they have
Sorry if this shows twice but it appears my first message
was quarantined because of my digital signature.
All,
I have been tasked with setting up video conferencing
utilizing asterisk. One of the requirements is a softset that has video
capabilities. Eyebeam looks promising but I
I have not but if you find one, please
pass it on because I have the same requirement.
Curt
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bruce Reeves
Sent: Wednesday, April 12, 2006
3:51 PM
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users]
to be published anywhere.
Curt Shaffer wrote:
I have not but if you find one, please pass it on because I have the
same requirement.
Curt
*From:* [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] *On Behalf
Just wondering everyones experience with Six Tel (http://www.iax.cc/show.php?go=local)?
They seem to have some really decent prices but I have heard some buyer
beware comments elsewhere.
Thanks
Curt
___
--Bandwidth and Colocation
I have this working. I have Asterisk connecting to my Vonage Linksys device
via Digium Wildcard X100P. No magic needed ;)
Curt
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bob McDowell
Sent: Wednesday, March 29, 2006 9:25 AM
To: [EMAIL PROTECTED];
Has anyone out there used eyeBeam v1.1 with Asterisk? If so what kind of results do
you have?
Thanks
Curt
smime.p7s
Description: S/MIME cryptographic signature
___
--Bandwidth and Colocation provided by Easynews.com --
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Daniel
Hazelbaker
Sent: Monday, March 27, 2006 1:32 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Receptionist Phones (was 3Com Phones)
Thanks for all the
Users Mailing List -
Non-Commercial Discussion'
Subject: RE : [Asterisk-Users] FXS
channel banks
How many phones lines ?
-Message d'origine-
De:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Curt Shaffer
Envoyé: vendredi 24 mars
2006 03:17
À: asterisk-users
I would not recommend the 3Com phones. I know to get most of them to even
work on 3Com systems you need to purchase licenses. For the prices you want
to pay you would definitely be better off going with something else.
The list price for the 3101 is $155
The list price for the 3102 is $240
The
Is anyone out there using FXS channel banks to connect
analog phones to Asterisk? If so do you have brand recommendations?
Thanks
Curt
smime.p7s
Description: S/MIME cryptographic signature
___
--Bandwidth and Colocation provided
I had the same kind of issue myself. The kernel was upgrading to 2.6.9-34
from 2.6.9-22 but for some reason it did not appear that way to the
compiler. I reinstalled Cent OS 4.2 and updated everything except for the
kernel and did a wget for the 2.6.9-22 source from the mirror and it worked
like a
This issue has been solved. What I found was that [EMAIL PROTECTED] was running a newer version of udev. Once I installed the newer version it came right up. The version
udev-039-10.10.EL4.3 works like a charm! Hope that helps someone out there.
Curt
On 3/8/06, Curt Shaffer [EMAIL PROTECTED
I have decided to move on from [EMAIL PROTECTED] and start
compiling asterisk myself now. I got a dedicated box and put my X100P in it. I
installed the server version of CentOS 4.2 (2.6.9-22.EL kernel) barebones. The
box is a dell GX270 workstation with 1GB of RAM. I got a fresh copy of
94 matches
Mail list logo
