Hello Matt,
I have not seen how to add a site.
Could you help me (us) ?
Tks
Francois Bergeret,
France.
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Matt Riddell
(IT)
Envoyé : vendredi 6 octobre 2006 11:40
À : Asterisk Users Mailing List -
In article [EMAIL PROTECTED],
Ronald Wiplinger [EMAIL PROTECTED] wrote:
Is it exclusive? Either Realtime or priority n ???
If so, what is the better way?
I believe 'n' is just a shorthand way of writing previous line + 1,
and gets converted into an actual number as the dialplan is compiled.
Btw I just noticed there is no bootrom.ver
file in this zip folder.
http://www.freedomphones.net/polycom/files/SoundPointIP_BootROM_2_6_2.zip
Could this be why this version is failing?
Cheers,
Dean
From: Dean Collins
Sent: Sunday, 8 October 2006 1:30
AM
Hi all
I have installed asterisk
when any of the user device made on, it should contact Asterisk
and download the config
how can i asterisk does this job, does asterisk does
or i should have any other server to meet my requirement
Ram
___
On Sat, Oct 07, 2006 at 04:15:35PM -0500, Carlos Chavez wrote:
DOes anyone know if you can use an Astribank usb channelbank with a 64
bit linux distribution like CentOs? I saw a note that the driver is only
built when you have an i386 processor and a kernel= 2.6.10
Actually this has
aterisk does not do this, you need a provisioning server. google for pap2 and tftp.
-yair
On 10/8/06, ram
[EMAIL PROTECTED] wrote:
Hi all
I have installed asterisk
when any of the user device made on, it should contact Asterisk
and download the config
how can i asterisk does this job, does
On Fri, Oct 06, 2006 at 05:10:34PM -0500, Erick Perez wrote:
Example: if I setup system XYZ with asterisk, then load this magical
utility/procedure that counts how many writes the filesystem has done
to / or to /,/tmp,/var and after 24 hours the utility/procedure says:
10thousand
Hello asterisk-users,
I'm currently investigating a problem related to the Transfer app and
DTMF tones via SipInfo.
My setup depends on:
Asterisk 1.2.10
Zaptel 1.2.8
libpri 1.2.3
Elmeg IP 290 (snom190)
Wildcard TE400 (E1)
The following dialplan is given:
exten = 555, 1, Transfer(554);
exten
Alvaro Parres wrote:
Hi List:
I have the next diagram:
GSM G729
G729
IdeFisk -- Asterisk A -
[INTERNET] Asterisk B - PSTN ( Via Unicall / Zap )
The user at IdeFisk Login as
Hi
thanks for the quick reply
i will look on that
thanks
Ram
On 10/8/06, Yair Hakak [EMAIL PROTECTED] wrote:
aterisk does not do this, you need a provisioning server. google for pap2 and tftp.
-yair
On 10/8/06, ram [EMAIL PROTECTED] wrote:
Hi all
I have installed asterisk
when any of
On 08/10/2006, at 3:00 PM, Dean Collins wrote:
Whats the best ftp server to upload Polycom phone cfg’s from? I’m
finding it a bit hit and miss using BTF server.
I'm using vsftpd quite successfully on several Asterisk boxes with
Polycom IP501 phones. Though, I'm now considering switching
Hi,I'm very interesting in high availability sollutions for ASTERISK. And I know that big companies are using SOLARIS (and I think SUN CLUSTER) for their VoIP gateways. Solaris and Sun Cluster are both free, of course you must pay if you want support. But I have read in FAQ this:
Can i install
I have seen an Optus SHDSL box set up incorrectly before - and the tech
re-visited and set it up correctly within hours of being informed.
PaulH
On Fri, 2006-10-06 at 12:17 +1000, [EMAIL PROTECTED] wrote:
There was a bit of traffic on this list a while ago regarding OPTUS
multi line that
On Thu, Oct 05, 2006 at 10:38:32AM +1000, Devraj Mukherjee wrote:
Hello world,
My asterisk server doesnt seem to disconnect the call if someone
hangsup say while they are listening to the menu as a result of which
my phone is engaged forever. Any pointers on fixing this issue?
Which zaptel
As I recall, you nee to make sure you run this script with the DeadAGI
command, not just AGI. This will make sure that the dial command will
return to your script only after it is done.
--Brian
On Sat, Oct 07, 2006 at 10:45:10PM -0700, Ali wrote:
So what should I do?
On 10/7/06, Tzafrir
Hi All,I need your urgent help. i installed OH323 channel and it is working well with samll problem fake ring. I have my VoIP provider MCI and ATT when i am routing the call via OH323 from the SIP ATA like Linksys i am getting too much fake ring even some time real RBT is there and also i
Let me chime on on Astlinux and my personal experience. I have used Astlinux in the following installations:
1) boot from CF on VIA platform, store config settings on USB key drive
2) boot from CD on P3-800, store config settings on USB key drive
3) boot from CF on Soekris Net4801, store
Hi,I have created SIP extenstions and created Teliax Trunk using IAX2. I am making outgoing calls to USA successfully.When I am making a call to my DID number from outside, its telling that "The number you have dialed is not inservice". Here I am giving the output from Asterisk server console:
On Sun, 8 Oct 2006 10:07:28 +0200, Tzafrir Cohen wrote
On Sat, Oct 07, 2006 at 04:15:35PM -0500, Carlos Chavez wrote:
Alternatively, try rebuilding
http://updates.xorcom.com/trixbox/zaptel-1.2.8-2.xpp.r2223.src.rpm
I downloaded the file and did a rpmbuild --rebuild on the server but
I got it to work in the end - by removing the _ from the front of my
fixed allowed numbers (of course there wasn't any real pattern
matching there at all).
Thanks for the help
Mark
On 03/10/06, Marco Mouta [EMAIL PROTECTED] wrote:
If you really want _07. to be tested afterall the above
Hello,
I need an advice about USA Origination (only). Who you recommend to me
for a production environment like a customer care support?
I need:
1) Excellent call quality.
2) Stability.
3) Excellent support.
What will be the best increment schema? 60/15, 60/6, 6/6 ??
Thanks in advance.
R.
Hey everybody,
I've, within the last 3 weeks, moved over to a PRI from SBC/ATT. I've
received several complaints about dropped calls. Reviewing the archives
on PRI and dropped calls shows that I should set the resetinterval=never
in the zapata.conf and restart. This hasn't helped.
The
Another question,
Is anybody using the Tellabs 2572 EC with a PRI. When we moved from our
analog lines to a PRI, I thought it would be simple stuff moving the EC
to the PRI. Changed signaling, made sure that channel 24 wasn't being
ECd and everything came up. But, I was getting complaints of
Title: Message
We had
some problems with the FTP server on our Windows 2003 server - we worked with it
quite a bit, but just could not get the Pollycom's to access it. It did work
with backup up our Switchvox PBX.
We use
Bulletproof FTP server in the Windows world.
-Original
Title: Message
Yep, using sbs 2003 here.
Cheers,
Dean
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gary Eck
Sent: Sunday, 8 October 2006 1:17
PM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: RE: [asterisk-users] ftp
server
Title: Message
We
also have SBS2003 - we initially had ISA running on it, but decided it was not
worth the grief at our office.
In the
back of my mind, I was wondering if ISA was causing the
problem.
I had
used Bulletproof in the past, since it has bandwidth throttling - and it just
Title: Message
Unfortunately I disagree with you about
ISA, whilst it may cause problems through my lack of skillset from time to time
the functionality it introduces and protects my network cant be beat at any
price, being built into sbs 2003 is just a bonus.
Cheers,
Dean
On Sat, Oct 07, 2006 at 10:10:14PM -0500, Rich Adamson wrote:
Jay R. Ashworth wrote:
On Sat, Oct 07, 2006 at 10:31:16AM -0500, Rich Adamson wrote:
If you've messed up in connecting telephone lines to the wrong module,
the ringing voltage sent to a fxs module will destroy it. You would need
Title: Message
Yes,
we don't have the proper skillset, either. Even though we are a
reseller.
We
have standardized on Linux appliances called Snapgear . Of course, they
are not integrated into Active directory, so there are lots of things that are
harder to do.
However, since they are
Tony Mountifield wrote:
In article [EMAIL PROTECTED],
Ronald Wiplinger [EMAIL PROTECTED] wrote:
Is it exclusive? Either Realtime or priority n ???
If so, what is the better way?
I believe 'n' is just a shorthand way of writing previous line + 1,
and gets converted into an actual number as
Jay,
I understand. We can have discussion offline to discuss this further at [EMAIL
PROTECTED]
Cheers,
Ed
Sent from my BlackBerry® wireless handheld
-Original Message-
From: Jay R. Ashworth [EMAIL PROTECTED]
Date: Sun, 8 Oct 2006 14:18:07
To:asterisk-users@lists.digium.com
Couple suggestions:On Win32, try Serv-U FTP. It's very reliable and supports a variety of protocols like SFTP and the like. Commercial software.http://www.rhinosoft.comOn Linux, I've fallen in love with FreeNAS, and not just because it utilizes the m0n0wall GUI. You can boot it off a USB stick,
Hi Dean -
I'm using vsftpd quite successfully on several Asterisk boxes with
Polycom IP501 phones.
Just an addition: one requirement I had in deploying Polycom phones -
I wanted the user (and me) to be able to plug in a new phone and go
with no configuration needed on the Polycom end. The
Is there a way to check if a peer is registered with the other box and
forward the call there if a call comes in?
Yes, you can (if nothing else, I'm fuzzy this morning) try forwarding
the call and it will fail if the device is not registered because
Asterisk will report it not found with a SIP
Sean Kennedy wrote:
Hey list,
Short version:
I have a need to disable the hardware can on the tdm24xxp I have. I
figure it's something in zconfig.h in the zaptel directory, but I'll be
damned if I can figure it out.
Long version:
I have a tdm2403e card which is experiencing an odd problem;
I had a similar problem. Turned out I wired it backwards. The Tellabs
only does EC in one direction.
Doug Lytle wrote:
Another question,
Is anybody using the Tellabs 2572 EC with a PRI. When we moved from our
analog lines to a PRI, I thought it would be simple stuff moving the EC
to the
Eric ManxPower Wieling wrote:
I had a similar problem. Turned out I wired it backwards. The
Tellabs only does EC in one direction.
The Tellabs that I have do Sender side EC as well. But, I plan on being
on site next weekend and will try switching the wiring around.
Thanks for the
On Sun, Oct 08, 2006 at 11:50:53AM -0700, Robert Goodyear wrote:
On Linux, I've fallen in love with FreeNAS, and not just because it
utilizes the m0n0wall GUI. You can boot it off a USB stick, too,
which I find super-cool.
http://www.freenas.org
/me imagines some BSD fanatics readying
On Fri, Oct 06, 2006 at 12:31:24PM -0700, Martin Joseph wrote:
On Thu, Oct 05, 2006 at 11:41:32PM -0700, Sam Norris wrote:
Heat = #1 cause of disk failure. If they are roasting to the touch they
will fail in 2-3 months.
One word: smartd.
I didn't know it existed, and I'm amazed I didn't.
I assume that this
is from the echo canceller, but I am not sure.
A call is started
via SIP speakerphone.
When the handset is
picked up, there is a slight echo of your own voice after you speak.(duh, is
there any other kind of echo)
If the call is made
without the speaker phone, there is
I understand Jeremy and Kris point of view (BTW Kris, astlinux rocks!!)
However the main question was not aswered (or i didn't get it, did I ?)
If I use a Disk on Module that has 2million hours MTBF and a Read/Write lifecycle of 2million times, then, How many days/weeks/months/years will take to
On Fri, Oct 06, 2006 at 08:35:26AM -0800, Mojo with Horan Company, LLC wrote:
I'm pretty sure that when you AREN'T sending the DTMF inband, asterisk
detects it, and if the keys pressed don't lead to any recording/transfer
features, then it re-creates DTMF on the bridged channel. I mean to
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
[EMAIL PROTECTED] wrote:
Hello Matt,
I have not seen how to add a site.
Could you help me (us) ?
Tks
When you are in the site list:
Click the link:
http://asterisk.group.stumbleupon.com/sites/
It's titled, View/Add Sites
- --
Cheers,
Matt
On 09/10/2006, at 5:07 AM, Noah Miller wrote:
username and password is PlcmSpIp. vsftpd cannot handle capitalized
usernames, so if you want to use vsftpd, you have to manually
re-configure the username on each phone.
I use vsftpd and I'm using the default PlcmSpIp username just
fine. :)
On 08/10/2006, at 9:34 PM, Paul Hales wrote:
I have seen an Optus SHDSL box set up incorrectly before - and the
tech
re-visited and set it up correctly within hours of being informed.
Same with my Optus SHDSL box: The first tech misconfigured, so I kept
getting PRI restarts on my Sangoma
Whoops. I totally generalized under the realm of Non Windows didn't
I? Doh!
On Oct 8, 2006, at 2:27 PM, Tzafrir Cohen wrote:
On Sun, Oct 08, 2006 at 11:50:53AM -0700, Robert Goodyear wrote:
On Linux, I've fallen in love with FreeNAS, and not just because it
utilizes the m0n0wall GUI. You
On Fri, Oct 06, 2006 at 05:10:34PM -0500, Erick Perez wrote:
I understand Jeremy and Kris point of view (BTW Kris, astlinux rocks!!)
However the main question was not aswered (or i didn't get it, did I ?)
If I use a Disk on Module that has 2million hours MTBF and a Read/Write
lifecycle of
vsftpd.
i use seperate user id's for the phones (makes it
easier than to have all the configs in one dir). Works wonders. and for all you
windows lovers. Time to make the switch
- Original Message -
From:
Dean
Collins
To: asterisk-users@lists.digium.com
Sent:
Doug asked a while back if there was a way to share astdb between two
machines. If there was a way then the registry would be the same. While on
the topic I may be completly off however if two machines were to share the
same real time db as well as the same astdb (assuming we can share it
Tzafrir Cohen wrote:
H, I'm not sure that this is exactly the data you're after.
You're looking for the ammounts of writes for the disk block that gets
the most writes.
E.g: for a standard ext3 filesystem, the journal area would probably
have very frequent writes, whereas most of the
There's a product below on the market that checks whocalled.us to
determine if a telmarketer should get the Zapteller. Do you know if
that's something that could possibly be included into the blacklist or
in a macro.
http://venotec.com/product/tms/
signature.asc
Description: This is a
Are you able to track real time from a
windows machine the transactions occurring on your asterisk server if you have
vsftpf installed?
Cheers,
Dean
From:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
On Behalf Of Dovid B
Sent: Sunday, 8 October 2006 9:04
PM
To:
Tried looking around (ok maybe not well cause I am
a lil tired) but cant seem to find it. Can some one send me a link with the
diffrences between SIP and SIP-B ? Thanks.
Dovid
___
--Bandwidth and Colocation provided by Easynews.com --
Brian Capouch wrote:
Tony Mountifield wrote:
In article [EMAIL PROTECTED],
Ronald Wiplinger [EMAIL PROTECTED] wrote:
Is it exclusive? Either Realtime or priority n ???
If so, what is the better way?
I believe 'n' is just a shorthand way of writing previous line + 1,
and gets converted
http://www.freedomphones.net/polycom/files/PolyReboot.pl-script
can anyone give me an idea on how this reboot script works?
How do I load it up and cause it to occur etc? sorry for
being a newbie but searching on perl script information on google isnt
working.
Cheers,
Dean
Jeremy McNamara wrote:
Tzafrir Cohen wrote:
H, I'm not sure that this is exactly the data you're after.
You're looking for the ammounts of writes for the disk block that gets
the most writes.
E.g: for a standard ext3 filesystem, the journal area would probably
have very frequent writes,
Your server seems to be doing exactly what you are telling it to do:
-- Executing Playback(SIP/216.89.79.2-09e1d020, ss-noservice) in new stack-- Playing 'ss-noservice' (language 'en')
Read the extensions.conf directions on the wiki site:
On Mon, October 9, 2006 11:46 am, Dean Collins said:
Are you able to track real time from a windows machine the transactions
occurring on your asterisk server if you have vsftpf installed?
Yes... In an SSH session, tail -f /var/log/vsftpd.log will show you
everything you need.
Also, I have
On 09/10/2006, at 12:12 PM, Dean Collins wrote:
can anyone give me an idea on how this reboot script works?
I actually just use the SIP notify command on the Asterisk console to
remotely reboot my Polycom phones. It requires a pre-configured
sip_notify.conf file and the Polycom option to
I'm having trouble getting my TDM22B to answer a call. I have an analog line
plugged into each FXO modules (two analog lines) Neither answer or pickup the
call in Astrerisk when dialed from an external phone (eg cell phone).
I know the card is working modules zaptel wctdm are loaded. Here is
I have a need for a dialplan that call for the ability for people to dial *1XX and it send a callto extension 400 with the calleridname of Nursery and the calleridnum of the *1XX number thatwas put in minus the *. Now I know how to do it individually but I now there must be an easier
way to simply
Hello,
we are building an asterisk cluster. Here is what we are trying.
Four Asterisk Servers
AT1, AT2, AT3, AT4.
Two service providers (SIP accounts). One for call origination (CO) and
one for call termination. (CT)
1. Say some one dials a number 18xx xxx , CO forwards the call to
our
exten = _*1XX,1,Set(CALLERID(all) = Nursery ${EXTEN:1})
exten = _*1XX,2,Dial(SIP/400)
Tom Vile wrote:
I have a need for a dialplan that call for the ability for people to
dial *1XX and it send a call
to extension 400 with the calleridname of Nursery and the calleridnum
of the *1XX number
In the astdb, there will be a record once a sip user register.
However, I found that the record will still stay in the astdb even
when the user not register for a long long time. Can I refresh the
astdb by some command such that it will get the update status of the
system?
Thanks.
jk wrote:
Hello,
we are building an asterisk cluster. Here is what we are trying.
Four Asterisk Servers
AT1, AT2, AT3, AT4.
Two service providers (SIP accounts). One for call origination (CO) and
one for call termination. (CT)
1. Say some one dials a number 18xx xxx , CO forwards the call
I just upgraded to asterisk 1.2.12 from 1.0.1 and I can not get cdr -
mysql to load
I've install asterisk-addon and have cdr_addon_mysql.so in module
directory
In modules.conf I have:
preload = cdr_addon_mysql.so
Should it be load or preload?
In cdr_mysql.conf (nothing has changed)
[global]
On Sun, Oct 08, 2006 at 10:04:51PM -0600, Joseph wrote:
What am I missing?
Maybe your /etc/mysql/my.cnf ?
# Instead of skip-networking you can listen only on
# localhost which is more compatible and is not less secure.
# bind-address = 127.0.0.1
#skip-networking
Martin Joseph wrote:
I am seeing occasional stuck SIP channels that seem to occur when the
fricking Nokia E60 drifts out of WIFI range in the midst of a call.
This is particularly annoying when the stuck channels include my PSTN
gateway (wellgate 3701a), which leaves incoming and outgoing
Hi,
I can't get my INX line working for incoming (outgoing is working fine).
When I dial this number from my home phone, asterisk sends the call
straight to extension 101, for some reason it doens't read what my
extensions.conf is saying.
I had the same dial plan and same extensions.conf
On Mon, 2006-10-09 at 01:13 -0300, Hermann Wecke wrote:
On Sun, Oct 08, 2006 at 10:04:51PM -0600, Joseph wrote:
What am I missing?
Maybe your /etc/mysql/my.cnf ?
# Instead of skip-networking you can listen only on
# localhost which is more compatible and is not less secure.
On Sun, Oct 08, 2006 at 10:39:26PM -0600, Joseph wrote:
I have bind-address = 127.0.0.1 in my.cnf
the cdr was working find with asterisk 1.0.1 just after upgrade
something is not connecting.
I don't know if asterisk will use the localhost or the network IP to
connect. Just try to comment your
On Mon, 2006-10-09 at 01:45 -0300, Hermann Wecke wrote:
On Sun, Oct 08, 2006 at 10:39:26PM -0600, Joseph wrote:
I have bind-address = 127.0.0.1 in my.cnf
the cdr was working find with asterisk 1.0.1 just after upgrade
something is not connecting.
I don't know if asterisk will use the
Hey Folks,
Been wrestling with the 601 and the expansion module. Finally
figured out how to populate both with speed dial entries. Also
hints are showing in Asterisk with the show hints command.
But how do I get the LEDs to light when one of these other
extensions is either off-hook, or
how does one force mandatory password change on login? and a period of time to pass before mandating a password change?im using trixbox so if you have that info. that would be good, if you have it for asterisk, im sure i could figure it out as wellthx.
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