Hi all,
I have a TE210P connected to two E1 and everything seems fine. I create
a script that originates a call from the first E1 to the second E1 and
then starts playing an announcement, the extension that answers the call
starts recording the announcement and place that in a directory.
The
OK
I have set the time and message
Luki writes:
Has anybody been able to get call waiting on the PSTN line?
As far as I recall, you will only hear a tone in the audio stream when
a second call comes in. The Sipura does not detect or handle it, but
if you flash the line on the FXS interface
Try something like this.
Note: I did not write these scripts. I would give credit to who did, but
unfortunately I do not remember where I got it.
Dan
[globals]
TRUNK1 = IAX2/user:[EMAIL PROTECTED]
TRUNK2 = IAX2/user:[EMAIL PROTECTED]
; Sets up the outgoing gateway according to availability
Remco Barende wrote:
Weird, I checked with KPJ before and he mentioned it is normal behaviour
for ISDN. My console is filled with messages like this :
== Primary D-Channel on span 1 up
== Primary D-Channel on span 1 down
Well, just wanted to share my experiences, over here in Belgium. We
On Mon, December 5, 2005 7:22, Remco Barende said:
Already HAVE Florz patch installed! :-(
What version of * and BRIstuff are you using?
Strange, sounds like the florz patch has not been effectively applied or
it's broken. I'm using an old version of bristuff :
Asterisk
With asterisk-1.0.9 version when I wanted to enable logging of CDR in MySQL I
needed to make softlink
ln -s /usr/src/asterisk-1.0.9/asterisk.h
/usr/src/asterisk-addons-1.0.9/asterisk.h
Now, on version 1.2.0 I don't have that file. Do I need it?
Thank you for your time.
--
Tomislav Parčina
Does anyone know of any chat lines/party line programs/agi/add ons for
asterisk to handle suck type of operations?
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See:Dec3 01:16:52 WARNING[20212]: chan_iax2.c:2747 create_addr: No such host:24
Are u sure exists 24 iax device ?Try with ip
2005/12/3, Cyrille Demaret
[EMAIL PROTECTED]:Hi,Same result with dial:-- Executing DeadAGI(SIP/205-0231, b) in new stack
-- Launched AGI Script
Try this
___
1st Machine sip.conf
[box2]
username=box1
type=friend
host=10.0.0.2
secret=*
in extensions.conf
exten = _XX,1,Dial(SIP/box2/${EXTEN})
__
2nd Machine sip.conf
[box1]
username=box2
type=friend
UK, London Based DID £1 per month
All number begin with 0208 0xx
Sam,
Please, if you are going to market London numbers, format them correctly!
The code for London is 020, therefore your numbers are 020 80xx .
[Blatent self-plug] If you or anyone wants to purchase numbers from the
On 4 Dec 2005, at 21:14, chawki hammoud wrote:Hi:Sorry,but i dont know what ethereal is,and for myasterisk version the iax is good on it because i madea lot of succesful iax connections with many voipproviders like "sixtel,voipjet..."Yep, I agree, your asterisk should be fine,but this provider is
Hi all
I had allread install asterisk server and two X-Lite softphones on two
different machines. whole processa of calling is going fine. But I
cann't able to hear ringing / any type of voice on both side.
The asterisk sever give following worning.
WARNING[1922]: res_musiconhold.c:205
Hi,
- Original Message -
From: amna saleem [EMAIL PROTECTED]
I have been using Asterisk-1.0.3 for quite some time now.My main aim
nowadays is to make iax-iax calls for which i am usin DIAX soft
phone.Theproblem is that sometimes the phone doesn`t register and at
others it gets
out of
Hello,
Since I've installed Asterisk 1.2 (from CVS) on a gentoo server, I've
got this warning when loading chan_iax2:
WARNING[10204]: chan_iax2.c:1254 try_firmware: Firmware file
'/var/lib/asterisk/firmware/iax/iaxy.bin' fails checksum
There is no problem after, IAX works well.
Is that a
Hi,
When sip device sends to Asterisk INVITE with no 'Contact' field, the
server should respond with all information it holds about client. When
reading database fields, 'fullcontact' is empty. So, whole procedure
ends with 'chan_sip.c:6393 register_verify: Failed to parse contact
info'.
Hi ! I'm having a problem with calls that come over ISDN PRI and go to DISA
app. Problem doesn't happen with calls from SIP phones to DISA or for a
calls over ISDN PRI to SIP phones. Asterisk is 1.2.0
This renders DISA completely unusable when call comes over PRI, since every
call gets hunged
Thanks all!
You've been very helpful!!!
- Original Message -
From: Tom Vile [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Sunday, December 04, 2005 10:36 PM
Subject: Re: [Asterisk-Users] New to [EMAIL PROTECTED]
Luki wrote:
Has anybody been able to get call waiting on the PSTN line?
As far as I recall, you will only hear a tone in the audio stream when
a second call comes in. The Sipura does not detect or handle it, but
if you flash the line on the FXS interface after hearing the tone, the
Sipura will
Hi all,
I am getting some error while i am trying to install oh323. I already
installed Pwlib and openh323 library, But i do not know from where the
following error is apearing.
asteriskaudio.cxx: In destructor `virtual
PAsteriskSoundChannel::~PAsteriskSoundChannel()':
asteriskaudio.cxx:167:
Luki wrote:
Has anybody been able to get call waiting on the PSTN line?
As far as I recall, you will only hear a tone in the audio stream when
a second call comes in. The Sipura does not detect or handle it, but
if you flash the line on the FXS interface after hearing the tone, the
Sipura will
cp wrote:
I have configured my SPA-3000 to direct all calls received over PSTN
interface to Asterisk. That is set up via a dial plan with S0 and works
fine. CID information is passed along as well. But I find it impossible
to setup call waiting on the PSTN line. I do subscribe to this service
I would like to know if it is possible to include a
variable in sip_nat.conf.
I have a file with my network configuration and I
want to parse it and to use LAN IP in sip_nat.conf.
Is there a way to parse a file and include variables
in a .conf file.
Amaury
Attention:Your mp3s arent higher than 128 bit/s2005/12/5, Vipul Patel [EMAIL PROTECTED]:
Hi all
I had allread install asterisk server and two X-Lite softphones on two
different machines. whole processa of calling is going fine. But I
cann't able to hear ringing / any type of voice on both side.
You can use shell script to generate sip_nat.conf file2005/12/5, Amaury BOSSE [EMAIL PROTECTED]:
I would like to know if it is possible to include a
variable in sip_nat.conf.
I have a file with my network configuration and I
want to parse it and to use LAN IP in sip_nat.conf.
Is
Amaury BOSSE a écrit :
I would like to know if it is possible to include a variable in
sip_nat.conf.
I have a file with my network configuration and I want to parse it and
to use LAN IP in sip_nat.conf.
Is there a way to parse a file and include variables in a .conf file.
Amaury
In
Hi
i have installed [EMAIL PROTECTED] and working as of with my extensions
and Sip provider
now iam looking to deploy prepaid application with a2billing
does any one successfully integrated
or any other docs make me to integrate
i dont see full docs even at [EMAIL PROTECTED] site
iam still
Hi,
Two things does your codec set in X-lite match what is set in the sip
file and have you rebooted since setting up music on hold. I should also
ask if ran a make and make install in the asterisk-addons directory,
this installs a mp3 player (among other things) in Asterisk 1.2?
Vipul
Thanks all for the replies.
I've narrowed it down to the phones dislike for my older 3COM switch. I
noticed on the weekend that when these missed calls occur, if I ping the
phone, the first few packets are dropped..almost like it's gone to
sleep..
David A. Morrow
Technical Systems
Title: Message
Hi,
Have
someone successfully configured the streaming MOH in Asterisk 1.2.0 using
streamplayer?
Regards
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Hi,
I'm using Asterisk
with a BRI Card (HFC Chipset) using the zaphfc driver.
I'm encountering the
following problem : when the first line is in use and a second incoming call
arrive, the console shows the following message :
Dec 5 14:40:52 WARNING[2323]:
chan_zap.c:7512 zt_pri_error:
Hi!
I have been using Asterisk-1.0.3 for quite some time now.My main aim
nowadays is to make iax-iax calls for which i am usin DIAX soft phone.The
problem is that sometimes the phone doesn`t register and at others it gets
out of the registration(after being registere for some time).Can anyone
This feature has worked for us since ver 1.0 (not cvs)
Alvaro Parres wrote:
Josheph:
I had have that problem, and it get solve when i take out the
incominglimit from my sip.cfg
Also if you send you sip.cfg and extensions.cfg will be easier to
help you
Tray it.
Alvaro Parres
Hi All
Apologise if this has been previously asked but I am fairly new to the list.
I have a VegaStream 400 and have succesfully connected the asterisk to the box
to make outgoing calls with no problems. I cannot for the life of me work out
how to recieve incoming calls. I have looked around
Dave Morrow wrote:
Thanks all for the replies.
I've narrowed it down to the phones dislike for my older 3COM switch. I
noticed on the weekend that when these missed calls occur, if I ping the
phone, the first few packets are dropped..almost like it's gone to
sleep..
Not likely to be
lokotes wrote:
When sip device sends to Asterisk INVITE with no 'Contact' field, the
server should respond with all information it holds about client. When
reading database fields, 'fullcontact' is empty. So, whole procedure
ends with 'chan_sip.c:6393 register_verify: Failed to parse contact
Hi
I got this warning message repeating itself in the log this morning
Dec 5 08:52:52 WARNING[25686]: format_wav.c:247 update_header: Unable to
find our position
Dec 5 08:52:52 WARNING[25686]: format_wav.c:247 update_header: Unable to
find our position
Dec 5 08:52:52 WARNING[25686]:
hi,
my asterisk 1.2 works very well with ISDN and SIP, but how can i transfer calls
from my phone (for example ISDN, MSN 400) to the another phone (ISDN, MSN 401)?
Or when the phone 401 rings, but my boss is not there, how can i take the
phonecall from 401 to 400? Do i need special options in my
Hello,
Would you please share your experience regarding h323 and oh323 in asterisk.
I am confused to choose one.
Thanks,
--
You don't have any choice, you already made it before you came
here.___
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Prely subjective, but I first installed h323 and it worked. Somewhere along the
line something happened and it no longer worked. Recompiling it etc seemed to
have no effect.
I then tried oh323 and it worked first time and has stayed working.
I probably did soemthing wrong, but oh323 seems to
Hi to all
i'm finding the procedures for install the ambient md 3200 chipset modem to
make tests, anybody have a link or the procedure to do that??
thanks to all
Vladimir
__
Visita http://www.tutopia.com y comienza a navegar m�s r�pido en Internet.
Tutopia es
Hi,
Push the '#' key followed by the extension for a blind transfer.
Thanks
Denny Schierz wrote:
hi,
my asterisk 1.2 works very well with ISDN and SIP, but how can i transfer calls
from my phone (for example ISDN, MSN 400) to the another phone (ISDN, MSN 401)?
Or when the phone 401 rings,
This worked perfectly.
Thanks,
Waldo
On Dec 5, 2005, at 4:32 AM, xcel wrote:
Try this
___
1st Machine sip.conf
[box2]
username=box1
type=friend
host=10.0.0.2
secret=*
in extensions.conf
exten = _XX,1,Dial(SIP/box2/${EXTEN})
username= did it.
Thanks,
Waldo
On Dec 5, 2005, at 2:14 AM, Luki wrote:
Any ideas on how to correctly set this up?
Try adding authuser= and/or username= to the configuration. Do a SIP
DEBUG and see what peer asterisk looks for when trying to authenticate
the INVITE. It probably can't find
Hi,
A while back I made the stupid mistake of deleting my log files 'full'
and 'messages' for asterisk. I recreated the files by 'touch' filename
and I have gone into the Asterisk CLI and tried both 'logger restart'
and 'logger rotate' but the logs still show nothing. I run 'logger show
I have an Asterisk system with Fedora Core 4.0, kernel
2.6.14-1.1637. It sometimes locks up with heavy load (e.g., lots
of HDLC messages). This requires a hard reboot.
I saw some other reports of hard lockups under load. I have
disabled as much as possible in the BIOS and as much as possible in
Chuck Bunn wrote:
drwxr-xr-x 4 asterisk asterisk 4096 Dec 5 08:22 .
-rw-r--r-- 1 root root 0 Dec 5 08:22 event_log
-rw-r--r-- 1 asterisk asterisk 1186 Nov 12 07:43 event_log.0
-rw-r--r-- 1 root root 0 Nov 18 06:37 full
-rw-r--r-- 1 root root
Let me simplify my problem. I have a single Aastra 9133i SIP phone and
latest Asterisk from SVN source running on Fedora Core 4. The phone
currently says No Service I would like to be able to dial 1234 from
the phone and get Asterisk to play back an audio message or say some
digits. I
One more thing. I upgraded the firmware of the 9133i to 1.3.
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On Mon, 2005-12-05 at 11:15 -0500, Robert La Ferla wrote:
Let me simplify my problem. I have a single Aastra 9133i SIP phone and
latest Asterisk from SVN source running on Fedora Core 4. The phone
currently says No Service I would like to be able to dial 1234 from
the phone and get
I tried to use DISA 1.2 with regular asterisk (not [EMAIL PROTECTED]), and had problems with it (losing the last digit or occasionally other digits), YMMV.
On 12/4/05, Richard Smith [EMAIL PROTECTED] wrote:
Hi all,
I was wondering whether the DISA function on the latest asterisk 1.2 stable
Pete Barnwell wrote:
I wasted a lot of time getting 9112is to work with almost identical
setup. The problem I eventually found was that the 9112is look for the
config file mymacaddress.cfg in upper case (eg 00085D035BC1.cfg) whereas
the documentation says they look for lower case, so they were
On Mon, 2005-12-05 at 11:27 -0500, Robert La Ferla wrote:
Pete Barnwell wrote:
I wasted a lot of time getting 9112is to work with almost identical
setup. The problem I eventually found was that the 9112is look for the
config file mymacaddress.cfg in upper case (eg 00085D035BC1.cfg) whereas
hi,
Quoting Chuck Bunn [EMAIL PROTECTED]:
Push the '#' key followed by the extension for a blind transfer.
absolut perfect, thanks :-) .Is there also a shortcut, to take a phone
call from
other phones to me?
cu denny
This
In our dialplan, we use centralized voicemail for SIP, IAX and cell phones.
This means, if a caller calls a user's DID, it tries his SIP/IAX extension,
then if he doesn't answer there, it tries his cell, then it goes to Comedian
Mail.
Everything works 100%, except when the user shuts his cell
Look into the findme feature, this will require the person receiving the callto push a buttonhit 1 to accept this call before a callgets transfered to a cell phone (or home phone for that matter), if nobody hits 1 it continuesin the dialplan, this will prevent calls from being transfered to cell
Dave Cotton wrote:
One thing is to do a factory reset to reinit everything, I did that with
my 9112i after upgrading the firmware.
I just did that. Now Asterisk is giving me the follow error: (0.99 is
my Asterisk server and 0.111 is the phone)
Dec 5 12:04:10 NOTICE[14222]:
Hello,
Hopefully someone can advise me on the last problem I have in my config.
Among my trunks I have an spa-3000 with the pstn connected to an
ata-186 that I am trying to bring into asterisk. All works perfectly
except apparently when I receive a malformed caller id from this
outside service
Neat macro
but not quite what Im looking for if I force call recipients to press 1 to
accept a call they will scream bloody murder. Good idea though.
-Original
Message-
From: Joe Pukepail
[mailto:[EMAIL PROTECTED]
Sent: Monday, December 05, 2005
10:20 AM
To: Asterisk Users
This is what I use. You pre-pend a '4' to the extension number (I used
that because that is how our old pbx worked). There is a number you can
use that will pickup any ringing extension but I forgot what that is. It
should be listed on the asterisk wiki for Pickup.
exten =
Thanks for your answer but I don't want to include a file, I only want to
include a variable.
Is it possible to execute linux commands like grep or top in a .conf file in
order to parse a file and get a variable?
-Message d'origine-
De : Administrator TOOTAI [mailto:[EMAIL PROTECTED]
On Monday 05 December 2005 12:09, Colin Anderson wrote:
Everything works 100%, except when the user shuts his cell phone off. When
that happens, and he doesn't pick up his SIP/IAX extension, it hits his
cell phone, and the cell carrier's default Unavailable message is played.
Asterisk detects
Hi,
To pick up another persons phone that is ringing dial '*8' followed by
their extension. To do an attended transfer dial '*2' followed by the
extension...
Hope that helps
Denny Schierz wrote:
hi,
Quoting Chuck Bunn [EMAIL PROTECTED]:
Push the '#' key followed by the extension for a
I solved it by registering the phone in the sip.conf.
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Hi,
I deleted the files and ran 'logger restart' - no dice, 'logger rotate'
- no dice, 'reload' - no dice, 'restart gracefully' - no dice. Logs are
not recreated???
Any other ideas
Thanks
Marco Supino wrote:
The user running asterisk doesnt have permission to write on the
files, delete
Hello,
Just a note to say the Asterisk Queues Tutorial at
http://www.orderlyq.com/asteriskqueues.html has been updated to take
account of changes in the 1.2.0 release. Anybody who has used our
tutorial to create their queues, or uses queues and is thinking of
upgrading, will probably find
We are using firmware version 6.3. Dont we need a service agreement to get the
latest drivers? We let ours expire since we werent having any problems. Isnt
it also true that once you upgrade the firmware there is no way to revert to an
earlier version? This is worrisome because we have heard
Thanks! It looks like you were right. We placed the phones and PBX on a
minimal, physically separate network and have had no problems. We were using a
3com unmanaged switch but have ordered an HP managed switch with VLANs and VoIP
QoS capabilities. We couldnt find anything about Shadow ping,
Turn off voicemail on his cell phone, give out his DID instead of his cell
#.
Send an SMS to his cellphone when new voicemail is left.
That's what we do now. Works fine.
As far as Dial()ing his cell goes, use 'r' (this is exactly what it's
designed
for) so that when the carrier is saying The
Hopefully, someone here has dealt with a Panasonic DBS in this way.
I have put an Asterisk server in front of our Panasonic DBS phone system.
The goal is to phase out our DBS, but during the transition, I still need to
have asterisk extensions access some features of our Panasonic.
The two
On Monday 05 December 2005 13:39, Colin Anderson wrote:
That appears to work *perfectly* but I don't get it. With the 'r' option
on, how can Asterisk determine that the user has answered the phone as
opposed to the carrier? Is it a signal that the carrier is sending?
Anyway, thanks. Works like
Anyone using any H.263+ video phones and want to relay their
experiences?
-Jonathan
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Hi,
Does anyone have any details about the Linksys one product that was just
announced? Does it use Asterisk?
Thanks
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I had a problem with DTMF with DISA.. I am using a Sipura SPA
3000 for the line. I set the FXO port impedance (on the PSTN line tab) to 900 as
advised by others and it worked.
Having said that, I'm sure you will be using some other FXO
adapter.. Just thought I'd tell.
- Original
No it does not user Asterisk. It is a proprietary system based around the
Call Manager products. Linksys sells the system to a service provider who
then offers the service to end users. Basically, LinksysOne is a means by
which service providers can offer a hosted PBX solution.
Kerry Garrison
You can find more information at http://www.linksysone.com/
Cory Andrews
Senior Partner
+++
VOIPSupply.com
454 Sonwil Drive
Buffalo, NY 14225
+++
voice - 716.630.1555 X22
email - [EMAIL PROTECTED]
fax - 716.630.1548
Chuck Bunn wrote:
Hi,
Does anyone have any details about
Any help on this pleaseHi, I am getting this error when compiling asterisk `ls *.c`: unrecognized optionh -DBUSYDETECT_MARTIN `ls *.c`Usage: /bin/sh [GNU long option] [option] ... /bin/sh [GNU long option] [option] script-file ...GNU long options: --debug --dump-po-strings
When transferring a call that came in on the Sipura and picked up by a
Polycom 501 (sip 1.52), then transferred to another polycom using the
transfer button on the polycom (havn't tried with the blind transfer
from the polycom phone), then as soon as the transfer is complete
(after pressing
Hi
We're using three line SIP phones (X-lite), very nice, with Asterisk 1.2
But we want to prevent either direct incoming calls or calls from other
extensions from ringing if the user is
in another incoming call (i.e incoming into the extension), making an outgoing
call or even checking their
Been working on testing asterisk 1.2 before upgrading our production
systems from 1.0.x and have found a few issues. The one I am
working on
now involves DTMF failure with the following setup:
*Linksys SPA-941* ---SIP--- *asteriskA(v.1.2)* IAX2 *asteriskB(v.1.2)* SIP- *Global
Try chan_oh323 and if it is not ok, try chan_h323
Both work well in different situations/equipments.
Isamar
On Mon, 5 Dec 2005, Innocent Evil wrote:
Hello,
Would you please share your experience regarding h323 and oh323 in asterisk.
I am confused to choose one.
Thanks,
--
You don't
Subject: RE: [Asterisk-Users] Linksys SPA-841 Missing Calls
Dave Morrow [EMAIL PROTECTED] wrote:
I've narrowed it down to the phones dislike for my older 3COM switch.
I noticed on the weekend that when these missed calls occur, if I ping
the phone, the first few packets are
I am still having a non-solved problem with Oh323/h323 and checking Digium
homepage after a long time, it looks like they need some dimes now to
support me in this case.
I have 46(2 T1) PSTN channels receiving calls through H323 protocol.
With oh323, after 40 channels in use, It crashes due
One other piece of information that I just stumbled on while doing a packet capture which may explain the whole thing:
The Cisco packet shows the RTP event as this:
RFC 2833 RTP Event
Event ID: DTMF Pound # (11)
End of Event: True
Reserved: False
Volume: 10
Event Duration: 1600
The Linksys
I'm curious if anything new has been determined on this phone? Is it
SIP compatible with Asterisk and, say, Broadvoice?
I'm a little wary that this may be vaporware. The phone doesn't seem to
be listed by the FCC. But, I would preorder one if it's Asterisk and
Broadvoice compatibile.
Hello,
i have succesfullu setup asterisk with Sangoma E1 card, evrything works well
but i don't know how to pass indications from telco switch to the user - when
users call bad number telco switch shuld talk unallocated number but its only
send PRI_CAUSE 1. How to pass voice indications thru
amaury BOSSE wrote:
Thanks for your answer but I don't want to include a file, I only want to
include a variable.
Is it possible to execute linux commands like grep or top in a .conf file in
order to parse a file and get a variable?
Look into the System() command:
I need to replace my switch. Does anyone have any recommendations for a
switch that is VoIP friendly? I want it to be a managed gigabyte switch.
There are lots of brands out there, but would prefer some recommendations
from the list.
-Charles
___
I like the chan_ooh323.
I like the idea of selfcontained H323 channel that doesn't rely external
libraries, often with specific versions that conflict with something
else.
OOH323 works right out of box and since we started using it to
interconnect Asterisk to Samsung OfficeServ 500 we had no
What is your port density requirement?
For 24 ports the LinkSys SRW2024 is awesome.
They street for less than $500 and have good QoS.
For a smaller switch, they make a 12 port variant.
Wiley
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of calvis
Sent:
I have a 24 port that is doing well for us. I will check out the LinkSys.
Thanks
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wiley Siler
Sent: Monday, December 05, 2005 2:18 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE:
So, we have
h323, oh323 and ooh323
I knew about h323 and oh323 but didn't know about ooh323.
What is URL of ooh323, I want to know more about them.
Thanks,
--
You don't have any choice, you already made it before you came here.
-Original Message-
From: [EMAIL PROTECTED]
Sent: Tue,
Wiley Siler wrote:
What is your port density requirement?
For 24 ports the LinkSys SRW2024 is awesome.
They street for less than $500 and have good QoS.
For a smaller switch, they make a 12 port variant.
Does the SRW2024 support port mirroring? I was shopping around, but
couldn't find any
On 12/5/05, calvis [EMAIL PROTECTED] wrote:
I need to replace my switch. Does anyone have any recommendations for a
switch that is VoIP friendly? I want it to be a managed gigabyte switch.
There are lots of brands out there, but would prefer some recommendations
from the list.
We use the
JP Carballo wrote:
amaury BOSSE wrote:
Thanks for your answer but I don't want to include a file, I only
want to include a variable.
Is it possible to execute linux commands like grep or top in a .conf
file in order to parse a file and get a variable?
Look into the System() command:
On 14:42, Mon 05 Dec 05, snacktime wrote:
On 12/5/05, calvis [EMAIL PROTECTED] wrote:
I need to replace my switch. Does anyone have any recommendations for a
switch that is VoIP friendly? I want it to be a managed gigabyte switch.
There are lots of brands out there, but would prefer
Cisco owns Linksys so they have some good features now.
64 VLANs, 8 port trunking groups, console port, 802.1p CoS support
Four Quality of Service egress queues per port let you prioritize
traffic via 802.1p.
http://www1.linksys.com/products/product.asp?grid=35scid=40prid=673
This can be
All my BT101's and GXP2000's are failing NTP update. My NTP server is on my
local LAN (and I've tried external ones), DNS is OK (and I've used IP address
instead of DNS name).
tcpdump on NTP server shows valid request, AND a valid response, yet the
phones still display 02-01-1900.
I have
Need a little help. Just set up an [EMAIL PROTECTED] box with 5 Polycom 501 phones. Everything works great except the messages button which when pressed results in asterisk responding Person at extension 102 is on the phone. Please leave a message after the tone. I have searched the web and
And about the protocol used to create this hierarchical network?
Should I use SIP (routing between SERs) or should I use IAX (routing
between Asterisks)?
About ENUM, Isnt the managing of the ENUM tree going to be very
complicated and heavy when we reach the millions of users?
Joao
Jan
Michiel van Baak wrote:
On 14:42, Mon 05 Dec 05, snacktime wrote:
On 12/5/05, calvis [EMAIL PROTECTED] wrote:
I need to replace my switch. Does anyone have any recommendations for a
switch that is VoIP friendly? I want it to be a managed gigabyte switch.
There are lots of brands out there,
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