Satish Patel wrote:
snip /
I am using cross compile so i can't update GCC other wise it will effect on
my other packages anyway... tell me one thing i have host system kernel
version is 2.6.18 and i am compiling ARM embedded rootbuild with other
kernel version 2.6.22 so i need to compile my
this might be classifyed to some of you as ot but i wonder what this
will do for the asterisk community?
Begin forwarded message:
Date: October 5, 2008 5:17:36 PM GMT-08:00
Subject: Fonolo: Visually Navigate Dial IVR Phone Menus in Web/
Mobile Browser
Source: Tech[dot]Blog
Author: Abdul
Hi All,
I am getting these events in asterisk message log:
NOTICE[16647] chan_zap.c: Got event 4 (Alarm)...
NOTICE[16647] chan_zap.c: Alarm cleared on channel 1
after that asterisk exits silently until I restart it. Sometimes zapata
drivers also get in a state where I need to
2008/10/5 [EMAIL PROTECTED] [EMAIL PROTECTED]
Kevin P. Fleming wrote:
Olivier wrote:
2. R Hook-flash key is now available to transfer calls.
In s450IP web management server, its defaults settings are :
Application-type: dtmf-relay
Application-signal: 16
Is there anything to
2008/10/6 Andrew Joakimsen [EMAIL PROTECTED]
Yes, IMAP is IMAP... at least it is supposed to.
But not all IMAP servers use the same configuration. Not all IMAP
servers will use the same Master User IMAP setup, what works in
Dovecot might not work in UW or Exchange due to a prefix or some
Dear Sir,
I'm sending them Session Progress as you can see in the attached log
fle...Please let me know if they ahve any reason to not sending DTMF to me
Regards
On Fri, Oct 3, 2008 at 6:54 PM, Alex Balashov [EMAIL PROTECTED]wrote:
200 OK is a SIP response indicating the successful
Hi Mark,
made some other tests but the problem remains. I installed 1.4.22-rc5
but nothing changed. I opened an issue on mantis waiting for a fix.
Giorgio
___
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AstriCon 2008 -
2008/10/6 Alex Balashov [EMAIL PROTECTED]
I think the problem is that every [Dead]AGI call is still a distinct
invocation of the script, even if the interpreter stays loaded as an ELF
module or whatnot.
A good solution to this problem would be to use a FastAGI service,
wherein a daemon runs
Hi
triXbox.org can answer these questions. Google may also give a
balanced view. But yes, i can assure you, people are using Trixbox
from Fonality.
Steve
On 6 Oct 2008, at 10:24, broadband Voice wrote:
Anyone using Tribox from Fonality. I understand its open source and
free. Can I use
Anyone using Tribox from Fonality. I understand its open source and free.
Can I use it for a call center functionality? Thanks.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Giedrius Augys wrote:
What tools and programming (scripting) language do you use for FastAGI?
Whatever languages FastAGI APIs are available for. You are pretty much
limited to languages whose interpreter lends itself to invocation as a
standalone daemon, which may or may not exclude PHP and
Hi
...
and there is a new application called fax gateway (
http://bugs.digium.com/view.php?id=13405)
that can do gatewaying between T30 and T38 and vice versa.
Best regards
Daniel.
asterisk.
Actually it exists. 1.4 had passtrough mode and 1.6 can send and receive.
Regards,
Atis
--
On Mon, Oct 6, 2008 at 7:39 AM, Andrew Joakimsen [EMAIL PROTECTED] wrote:
I was going to write a blog once about the non-existent T.38 support
in asterisk hence my purchase of the above domain. It expires in 10
days. T.38 support in asterisk still does not exist but I don't have
any time. If
2008/10/3 Olivier [EMAIL PROTECTED]
Hi,
1. Here http://www.voip-info.org/wiki/view/Siemens+Gigaset+S450IP it is
mentioned MWI is now working.
In my testings with lastest 02123 firmware, MWI is blinking when missed
calls but not when a message in present in voicemail.
With SIP debug I can
Satish Patel wrote:
snip /
I think you may be right.
Can you not extract a set of kernel headers for 2.6.18 and point the zaptel
build to them when you are making it in your cross-compile environment? I
can't remember the switch off hand but I am sure there is a way to point
the make scripts
i am on Trixbox and it works as you want it to work.. if you need further help
i can offer some.. regards
AHD Tarek Sawah
Integrated Digital Systems
CCNA, MCSE, RHCE, VoIP
Syria: +963 944 618286
USA: +1 347 562 2308
Date: Mon, 6 Oct 2008 05:24:14 -0400From: [EMAIL PROTECTED]: [EMAIL
satish patel wrote:
snip /
I have set env on shell
KVERS=2.6.22.5
KSRC=/path/to/kernel-2.6.22.5/source
Maybe I misunderstood you then. I thought you said that your ARM system
was using a 2.6.18 kernel? If that is the case, then surely you need to
build your zaptel module against that
Hi all, I've done this a few times with other PBX's but swyx has stumped me!
I'm having some trouble getting Asterisk connected to a Swyx system using a
sangoma A104dx... currently the setup is:
BT - Swyx
The above setup works fine... what i'm trying to achieve is
BT SIP Trunks - Asterisk - Swyx
Hi,
I have followed this guide
http://www.voip-info.org/wiki/view/Setting+up+paging+with+a+sound+card
and have paging working ok, now I need to implement 'ringing'.
When someone calls I need the ringing to be send to overhead paging through
the sound card.
Any pointers?
Sincerely,
Robert
- Bill Michaelson [EMAIL PROTECTED] wrote:
The IP330 has a subminiature jack for headset/mic combos. Are there
quality headsets anyone would recommend for in-office use for heavy
users with these phones? Using any wiring path? I've tried a cell
phone earphone/mic, and it sounds OK,
Hi, according to discussion on asterisk IRC, where people said, that
macros will be depracated, I tried to migrate from macros to contexts
and Gosub
but if I try to use gosub in extensions.ael, ael compiler complains,
that I shouln't use Gosub app,
but I can't find ael keyword, that will be
- Andres wrote:
After looking at your iax.conf and extensions.conf I believe you are
under the misconception that if you 'register' to a provider, then you
can send and receive calls. The fact is that you 'register' to receive
calls, but you must define a trunk in order to Dial Out.
On Mon, 6 Oct 2008, Geraint Lee wrote:
Hi all, I've done this a few times with other PBX's but swyx has stumped me!
I'm having some trouble getting Asterisk connected to a Swyx system using a
sangoma A104dx... currently the setup is:
BT - Swyx
The above setup works fine... what i'm trying
Robert Augustyn wrote:
Hi,
I have followed this guide
http://www.voip-info.org/wiki/view/Setting+up+paging+with+a+sound+card
and have paging working ok, now I need to implement 'ringing'.
When someone calls I need the ringing to be send to overhead paging
through the sound card.
I have
brilliant idea - except it would be a sunday morning and another problem
the handsets that come with swyx aren't sip compatible :S
Cheers
Geraint
2008/10/6 Gordon Henderson
[EMAIL PROTECTED][EMAIL PROTECTED]
On Mon, 6 Oct 2008, Geraint Lee wrote:
Hi all, I've done this a few times
Hi, I'm having a problem conecting my asterisk 1.4.21 with zaptel 1.4.11
to a Panasonic PBX. I'm using dynamic features to send hook flash to the
zap channels to make a call transfer to the pbx without tying a channel.
When I call from asterisk to the Panasonic PBX I haven't any no problem,
Hey, all. We're rolling out VoIP, and I'm wondering about PoE
recommendations, as we're going to have to replace our current network
equipment. My first inclination would be to just plunk down the cash and
do a Cisco system, but I'm relatively certain that would get shot down by
finance. Any
Linksys SRW248P or something like that... something from linksys anyway are
quite capable of all you mentioned... maximum 24 port powered though iirc.
Geraint
2008/10/6 Ken D'Ambrosio [EMAIL PROTECTED]
Hey, all. We're rolling out VoIP, and I'm wondering about PoE
recommendations, as we're
We've had some bad experiences with Linksys in general (prior to going
VOIP) and avoided them. We're running now fully on the NetGear FS728TP
switch (24 port 10/100 POE, 4 port 1000 uplink, and 2 slots for fiber
modules).
Geraint Lee wrote:
Linksys SRW248P or something like that... something
We've used Linksys SRW224P units at quite a few places without issue. For a
little lower cost, we've also used Netgear FS726 series switches.
Personally, I prefer the Linksys ones - they have a serial port for
administration rather than relying on you doing it over the LAN (though they
have a
We're looking at using Global Crossing for our WAN infrastructure that's
spread across 9 states. We're hoping to gain some stability and one
point of contact for these sites, as our current infrastructure is
pathetic for VoIP.
I have a couple of questions.
1. Has anyone on this list
yes, thats the one i mean, 224p, the one i mentioned isn't capable of vlans
properly (which was strange, since it said it did)... i never had any
problems with them powering phones and cisco access points.
2008/10/6 Chris Bagnall [EMAIL PROTECTED]
We've used Linksys SRW224P units at quite a few
On Mon, 6 Oct 2008, Ken D'Ambrosio wrote:
Hey, all. We're rolling out VoIP, and I'm wondering about PoE
recommendations, as we're going to have to replace our current network
equipment. My first inclination would be to just plunk down the cash and
do a Cisco system, but I'm relatively
- Lucas Alvarez [EMAIL PROTECTED] wrote:
Hi, I'm having a problem conecting my asterisk 1.4.21 with zaptel
1.4.11
to a Panasonic PBX. I'm using dynamic features to send hook flash to
the
zap channels to make a call transfer to the pbx without tying a
channel.
When I call from
We've been EXTREMELY happy with the HP 5400ZL series chassis switch. Price per
port is about 1/3 that of Cisco when it comes to POE. Price is about $100 per
port and all ports are 1Gb with POE by default -- you can't get modules that
don't have 1Gb and POE. 10Gb uplinks are available with other
Your math is correct but the application is incorrect.
The OP requested a switch with solution with VLANs, PoE, and QoS? By that
they would be using the VLANS and QoS for separation of Data / Voice.
Gb uplinks are very useful in Data applications..
Alex
Kindly consider the environment
Obviously we don't need 1Gb connections for VOIP :)
Phones support pass through to the desktop and VLAN tagging.
The need for 1Gb ports comes from wanting to have 1Gb at the desktop.
Dave
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gordon Henderson
Doug,
That is interesting concept.
How do you add this to a ring group and does it stop when an extension is
picked up?
Thank you very much.
robert
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Doug Lytle
Sent: Monday, October 06, 2008 10:34 AM
Most phones support only 100M switching though Unless you run separate
cabling for VoIP and data but then you would not need the 1G uplink.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
David Gibbons
Sent: Monday, October 06, 2008 11:48 AM
On Mon, Oct 6, 2008 at 5:21 PM, Pavel Jezek [EMAIL PROTECTED] wrote:
Hi, according to discussion on asterisk IRC, where people said, that
macros will be depracated, I tried to migrate from macros to contexts
and Gosub
but if I try to use gosub in extensions.ael, ael compiler complains,
that I
Atis Lezdins wrote:
On Mon, Oct 6, 2008 at 5:21 PM, Pavel Jezek [EMAIL PROTECTED] wrote:
Hi, according to discussion on asterisk IRC, where people said, that
macros will be depracated, I tried to migrate from macros to contexts
and Gosub
but if I try to use gosub in extensions.ael, ael
Robert Augustyn wrote:
Doug,
That is interesting concept.
How do you add this to a ring group and does it stop when an extension is
picked up?
It depends on how you have your ring group setup, I personally only do
this with a single extension. And yes, the bullhorn sound stops when
the
Jay R. Ashworth wrote:
In our call center (250 agents), we use Panasonic KX-TCA60 headsets, which
I think we get for 7 or 8 bucks apiece, and sell to the agents at cost.
Thanks for that - they look good, and I found several recommendations for them
after I got yours and started looking for them.
Pavel Jezek wrote:
yes, I know, but I hear on IRC, that macros will be deprecated and
suggestion was to move to contexts,
personaly I would like also move away from macros, because macros have
some limitations, eg. variable number of arguments isn't possible with
classic macros,
macros
On Mon, 6 Oct 2008, Alex Balashov wrote:
Giedrius Augys wrote:
What tools and programming (scripting) language do you use for FastAGI?
Whatever languages FastAGI APIs are available for. You are pretty much
limited to languages whose interpreter lends itself to invocation as a
standalone
Two options worth considering:
1. Use a soft phone that supports call recording. The convenience of
recording directly to the PC might win some converts. X-Lite and
Ebeybeam do this nicely, amongst others.
2. Using the Polycom IP650 which has onboard call recording to a USB
device when the
That isn't real T.38 support, it's just Packet2Packet bridging that
works correctly. Still need to use a Cisco gateway to support sending
the faxes somewhere on the PSTN. But it does work and it is reliable,
I use it every day.
On Mon, Oct 6, 2008 at 7:32 AM, Atis Lezdins [EMAIL PROTECTED] wrote:
Right, it takes some doing to find a 1Gb switching phone though we ended up
going with a system based on the Cisco 7941G-GE. This model supports all of the
needed features including vlan tagging and 1Gb switching.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On
We're using D-Link DES-3028P switches (24 10/100 + 4 gbit). They also have the
DES-3052P which is a 48 port version of the switch. We're paying ~$500, I think
for the 24 port version from Graybar.
-Jon
- Original Message -
From: David Gibbons [EMAIL PROTECTED]
To: Asterisk Users
On Oct 6, 2008, at 12:56 PM, [EMAIL PROTECTED]
wrote:
We've been EXTREMELY happy with the HP 5400ZL series chassis switch.
Same here. We have 4 of them and they have worked very, very well. I
have 25 polycom phones at present doing PoE from them and everything
is working great. They are
Greetings list,
What are people using for nice pretty recording/playback interfaces on their
asterisk servers? I'm aware of ARI included with FreePBX, but are there any
others that aren't linked to a larger GUI?
I'm looking for something that'll integrate nicely with a non-GUI, non-AGI
The times they are a changing - or something like that.
while gb on phones is not the norm today, it s becoming more so on the
higher end flavors and will continue to do so
since the life span of your switches will be several years, thinking
ahead is a good thing
my only concern is having
Hi
1. What is the best way to convert wav (44000 Khz) to gsm format for
asterisk ? I;ve tried sox command but the outcome is not satisfying...The
built-in gsm files shipped with asterisk are simply superb ..How do i create
gsm files of similar quality ? Can anyone help me out ? if sox is the
Hey All -
Slight problem here - my install of 1.4.21.2 seems to be missing the
Queue application:
asterisk*CLI core show version
Asterisk 1.4.21.2 built by root @ asterisk.productiveconcepts.com on a
i686 running Linux on 2008-09-02 18:15:03 UTC
asterisk*CLI core show application Queue
Your
On Mon, Oct 6, 2008 at 8:04 PM, Andrew Joakimsen [EMAIL PROTECTED] wrote:
That isn't real T.38 support, it's just Packet2Packet bridging that
works correctly. Still need to use a Cisco gateway to support sending
the faxes somewhere on the PSTN. But it does work and it is reliable,
I use it
Ok then how do you make that an night_bell as your extension?
Thanks
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Doug Lytle
Sent: Monday, October 06, 2008 12:31 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
Ken D'Ambrosio wrote:
Hey, all. We're rolling out VoIP, and I'm wondering about PoE
recommendations, as we're going to have to replace our current network
equipment. My first inclination would be to just plunk down the cash and
do a Cisco system, but I'm relatively certain that would get
Josiah Bryan wrote:
Hey All -
Slight problem here - my install of 1.4.21.2 seems to be missing the
Queue application:
What does the CLI output say when you start asterisk and it gets to the
part where it tries to load app_queue.so?
Andres
http://www.neuroredes.com
asterisk*CLI core show
Hi Sirum,
Sriram wrote:
Hi
1. What is the best way to convert wav (44000 Khz) to gsm format for
asterisk ? I;ve tried sox command but the outcome is not satisfying...The
built-in gsm files shipped with asterisk are simply superb ..How do i create
gsm files of similar quality ? Can
- Bill Michaelson [EMAIL PROTECTED] wrote:
Further to this, I'm in the client office today and dealing directly
with the users who are reporters and editors for a periodical and
conduct many telephone interviews. They want to use their old
recording devices with the new phones, but are
- Singer Wang [EMAIL PROTECTED] wrote:
We've had some bad experiences with Linksys in general (prior to
going VOIP) and avoided them. We're running now fully on the NetGear
FS728TP switch (24 port 10/100 POE, 4 port 1000 uplink, and 2 slots for fiber
modules).
While I haven't worked with
Gotta love flukes - after stopping asterisk and restarting so I could
see the startup text, core show application Queue just worked . ???
Oh well. Thanks!
-josiah
Andres wrote:
Josiah Bryan wrote:
Hey All -
Slight problem here - my install of 1.4.21.2 seems to be missing the
Queue
I know I have asked about this before, but I thought that I would ask again
with some more detail and maybe someone will have an idea. This is my first
time to be setting up an asterisk server and I have a server running. I
sent Linksys PAP2T¹s to several remote users. Only one out of the four
Dear all, I know there are two actual versions of Asterisk: 1.4 and 1.6.
My scenario is: SIP server with 100-150 SIP users, voice mail and maybe IVR. I
will use GSM audio codec.
Maybe in the future I'll connect a E1 line to the PSTN.
What Asterisk version is better to me and why ???
Thank
If you happen to be looking for a SMALL poe switch for a home or lab:
Think twice before you buy a netgear FS1xxP. While they're great
because fanless, I've had 2 Netgear FS116p POE switches, and so far BOTH
have developed one or more 'dead' POE ports. The manufacturer has a
LIFETIME warranty,
On Oct 6, 2008, at 1:53 PM, Steve Anness wrote:
I know I have asked about this before, but I thought that I would
ask again with some more detail and maybe someone will have an
idea. This is my first time to be setting up an asterisk server and
I have a server running. I sent Linksys
Maybe it works in more recent versions? I don't know. Anyways this is
getting rather off-topic.
On Mon, Oct 6, 2008 at 2:23 PM, Atis Lezdins [EMAIL PROTECTED] wrote:
On Mon, Oct 6, 2008 at 8:04 PM, Andrew Joakimsen [EMAIL PROTECTED] wrote:
Hopefully it works. The one in CallWeaver doesn't.
I am using NAT so the ATAs are configured with a proxy server. Qualify is
set to yes. Here is what is happening. After they plug in the ATA on the
otherside, and things register and I can call and they can call. After
several minutes I try to call and then get the ³no-service² message. This
El lun, 06-10-2008 a las 10:57 +0100, Steven Howes escribió:
Hi
triXbox.org can answer these questions. Google may also give a
balanced view. But yes, i can assure you, people are using Trixbox
from Fonality.
Steve
On 6 Oct 2008, at 10:24, broadband Voice wrote:
Anyone using
On Mon, 6 Oct 2008, Alejandro Facultad wrote:
Dear all, I know there are two actual versions of Asterisk: 1.4 and 1.6.
There is also 1.2. It may not be supported but there are 1000's of people
out there (myself included) who are still using it.
My scenario is: SIP server with 100-150 SIP
In several places online, and in the Asterisk F.O.T. book, there is a
warning against using '_.' saying:
[it] should probably never be used.
However, the need often arises act on numeric extensions that begin with
*'s and #'s, and '+', and of course _X. does not match
I have tried exten =
Robert Augustyn wrote:
Ok then how do you make that an night_bell as your extension?
We have an after hours IVR, press 1 if you know the party that you're
trying to reach, press 2 for Dial By Directory and press 3 for the night
bell.
[incoming]
;
On Oct 6, 2008, at 3:52 PM, Gordon Henderson wrote:
The answer you are looking for is that you should be using a
supported,
stable version, and right now, 1.4 is the only one that fits. If I
were
starting today, I'd go with 1.4.
1.6.0 has just been released.
Personally I'd start with
I would suggst th same solution if you haven't started using Trixbox yet..
maybe you shoul give Elastix a try .. it has modules made spcialy for call
centers..
AHD Tarek Sawah
Integrated Digital Systems
CCNA, MCSE, RHCE, VoIP
Syria: +963 944 618286
USA: +1 347 562 2308
From: [EMAIL
__ Information from ESET Smart Security, version of virus signature
database 3497 (20081006) __
The message was checked by ESET Smart Security.
http://www.eset.com
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon
I would stick with 1.4 in production, how mad would you be if I gave you a cell
phone with new code and it didn’t work? Would you throw your cell phone at me
if it cut us off during phone calls from a bug? Some people are ok with trying
new stuff, others it costs money when they lose business
Hello, I'm trying to figure out how to implement 1.6.0 with some ldap
integration, but it's hard to figure out if I can do what I want.
Basically I want to do only some lookup of values from ldap, as
opposed to storing everything related to my sip users in ldap.
For instance, would there be a
El lun, 06-10-2008 a las 10:57 +0100, Steven Howes escribió:
Hi
triXbox.org can answer these questions. Google may also give a
balanced view. But yes, i can assure you, people are using Trixbox
from Fonality.
Steve
On 6 Oct 2008, at 10:24, broadband Voice wrote:
Anyone using
signature
database 3497 (20081006) __The message was checked by ESET Smart
Security.http://www.eset.com
_
See how Windows connects the people, information, and fun that are part of your
life.
http://clk.atdmt.com/MRT/go
I am using a Polycom 501 SIP phone behind NAT. Asterisk server is
public with no NAT... everything works on the Asterisk end just fine
EXCEPT that I can never check voice mail
After about 30 seconds the call drops with these messagess:
[Sep 30 23:47:48] WARNING[26819]: chan_sip.c:1950
Have you considered fiber?
Nick
On Sun, Oct 05, 2008 at 07:52:54PM -0700, Eric Fort wrote:
Here's a couple of distances I'm looking to cover (distances are +- 10%):
1 at 400M
1 at 600M
1 at 1800M
1 at 2400M
some of these links may already have pots circuits complete with
El lun, 06-10-2008 a las 13:23 -0700, Ron Stephan escribió:
And the documentation (not that trixbox is well documented ) was weak
IMHO.
Try reading:
http://www.elastixconnection.com/downloads/elastix_without_tears.pdf
Best regards,
--
Guillermo Salas M.
Telconet S.A.
Calle 15 y Avenida 24
Stephen, What exactly are you trying to accomplish? If you want basic call
in/out you're just about there. Changes need to be made in your
extensions.conf. Your phones, by default, are in the [default] context. In
other words when making a call it looks for extensions here. To allow
outbound
On Mon, 2008-10-06 at 18:25 +0200, Pavel Jezek wrote:
Atis Lezdins wrote:
On Mon, Oct 6, 2008 at 5:21 PM, Pavel Jezek [EMAIL PROTECTED] wrote:
Hi, according to discussion on asterisk IRC, where people said, that
macros will be depracated, I tried to migrate from macros to contexts
This message is usually caused by Asterisk not receiving an ACK after
about 30 seconds of attempts. There are countless misconfigured UAs and
proxies out there that don't handle ACK well, so it would be nice to be
able to turn this 'feature' off. What's annoying is that the explanation
has
2008/10/6 Brendan Martens [EMAIL PROTECTED]
Hello, I'm trying to figure out how to implement 1.6.0 with some ldap
integration, but it's hard to figure out if I can do what I want.
Basically I want to do only some lookup of values from ldap, as
opposed to storing everything related to my sip
Thanks for the reply. Hmmm
1. I would provide Asterisk its own LDAP directory and synchronize
it with entreprise directory as I think it should be simpler to
synchronize 2 LDAP directories than coordinate Asterisk and Active
Directory evolutions.
This may work, but my end goal is
The odd thing is on this particular phone it only happens when you
call voicemail.
It is certainly a bug in Asterisk, not the UA. Asterisk is trying to
send to 192.168.1.x which obviously is not possible. Something in the
NAT support is not working right.
On Mon, Oct 6, 2008 at 3:06 PM, SIP
I've used the smaller ones, I think 8pt with 4pt PoE stuck in drop
ceilings and such to power ORiNOCO APs and never had an issue.
As for the larger switches I've used Linksys SRW224P. I have a few
running for a few years without issues. They have GB uplink but the
individual ports are 100M.
On
I've used the smaller ones, I think 8pt with 4pt PoE stuck in drop
ceilings and such to power ORiNOCO APs and never had an issue.
That's a good data point. We too have an FS108p (like yours) and it has
been reliable so far. For us it's only been the FS116p's that have
failed. It seems
On Tue, Oct 7, 2008 at 2:22 AM, Andrew Joakimsen [EMAIL PROTECTED] wrote:
The odd thing is on this particular phone it only happens when you
call voicemail.
It is certainly a bug in Asterisk, not the UA. Asterisk is trying to
send to 192.168.1.x which obviously is not possible. Something in
On Oct 6, 2008, at 4:31 PM, Andrew Joakimsen wrote:
As for the larger switches I've used Linksys SRW224P. I have a few
running for a few years without issues. They have GB uplink but the
individual ports are 100M.
I recently purchased a few SRW208P switches. They work fine. If you
run
I was just looking to see if anyone knows about an open source app
using the xml interface. I just started dabbling with the xml
interface a little bit and it helps to look at what others are doing.
I am looking for a console type app for the operator. Very simple
operations like
Brendan Martens wrote:
On Oct 6, 2008, at 3:52 PM, Gordon Henderson wrote:
The answer you are looking for is that you should be using a
supported,
stable version, and right now, 1.4 is the only one that fits. If I
were
starting today, I'd go with 1.4.
1.6.0 has just been
hi all,
just wondering what's happening here:
i have a pap2 and an spa941. everytime i call my spa from my pap2 i can
see it being added dynamically on the regcontext:
[Oct 7 11:59:08] -- Saved useragent Linksys/SPA942-5.2.8 for peer
100100
[Oct 7 11:59:08] -- Added extension
Stephen, What exactly are you trying to accomplish? If you want basic
call
in/out you're just about there. Changes need to be made in your
extensions.conf. Your phones, by default, are in the [default] context.
In
other words when making a call it looks for extensions here. To allow
Steve Murphy wrote:
On Mon, 2008-10-06 at 18:25 +0200, Pavel Jezek wrote:
Atis Lezdins wrote:
On Mon, Oct 6, 2008 at 5:21 PM, Pavel Jezek [EMAIL PROTECTED] wrote:
Hi, according to discussion on asterisk IRC, where people said, that
macros will be depracated, I tried to
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