On Mon, Apr 02, 2007 at 03:38:18PM +1000, Joel Hill wrote:
Give this a try it fixes a problem we have had with a couple of Via
boxes.
modprobe wcte11xp
modprobe wcte11xp
ztfcg -vv
zttool
We found that probing the card twice before running ztcfg helped alot.
Are you sure that this
After working with the Grandstream GXP 2000 series phones, I have
decided that I am quite unhappy with their problems, both voice quality,
volume, features and others. For their price now, there are plenty of
phones to choose from as well.
So subjectively what would be the best Hardphone for
Lspci does show:
03:02.0 Ethernet controller: Digium, Inc.: Unknown device 0120 (rev 11)
dmesg | tail
ip_tables: (C) 2000-2002 Netfilter core team
ip_tables: (C) 2000-2002 Netfilter core team
tg3: eth0: Link is up at 100 Mbps, full duplex.
tg3: eth0: Flow control is on for TX and on for RX.
lp:
On 4/2/07, Corporate IT Solutions - Michael Dunne
[EMAIL PROTECTED] wrote:
So subjectively what would be the best Hardphone for a small/medium
business with multiple line support, BLF, etc.
Does _anyone_ read the archives anymore? This is like a weekly question or
something.
On Mon, Apr 02, 2007 at 04:21:45PM +1000, Klaverstyn, David C wrote:
Lspci does show:
03:02.0 Ethernet controller: Digium, Inc.: Unknown device 0120 (rev 11)
dmesg | tail
ip_tables: (C) 2000-2002 Netfilter core team
ip_tables: (C) 2000-2002 Netfilter core team
tg3: eth0: Link is up at 100
On Mon, 2 Apr 2007, Corporate IT Solutions - Michael Dunne wrote:
After working with the Grandstream GXP 2000 series phones, I have
decided that I am quite unhappy with their problems, both voice quality,
volume, features and others. For their price now, there are plenty of
phones to choose
I have got a couple of Snom 360 and am very pleased with them.
On Mon, 2 Apr 2007 09:09:43 +0100 (BST), Gordon Henderson [EMAIL PROTECTED]
wrote:
On Mon, 2 Apr 2007, Corporate IT Solutions - Michael Dunne wrote:
After working with the Grandstream GXP 2000 series phones, I have
decided that
Hi Guys,
I started getting this error only from one of our ITSP's once we upgraded from
1.2.16 to 1.2.17.
Can anyone shed light ?
--- (12 headers 0 lines) ---
Transmitting (NAT) to 209.212.93.53:5060:
SIP/2.0 603 Declined (no dialog)
Via: SIP/2.0/UDP
You could use an rebranded (OEM) idefisk - does sip and IAX and uses
XML for the config files, not the registry - making it possible to use
it on a usb stick.
More info : http://www.asteriskguru.com/idefisk/oem/
(But its not open source, nor free).
Joachim
Mike Lynchfield wrote:
sip
single server dual xeon 3.2 raid 1 2 gig mem be overkill. i've looked at
the server dimensioning but would like comments / opinions from anyone who
has deployed a system for this many sip users.
TIA
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Hi,
What is the best way to implement Automatic Redial on No Answer ?
Looking at
http://www.ietf.org/internet-drafts/draft-ietf-sipping-service-examples-12.txtI
can see how Automatic Redial on Busy could (should) be done.
How would you do it on No Answer ?
Is there any event you should
Good morning
I am new with Astersik and I want to know how can I configure my LDK to
communicate with an Asterisk server via SIP. I don't know how to procede and
which configuration files should I be interested in.
If anyone could help me, I would be extremely grateful
thanks a lot in advance.
HI,
Here is my setup:
USERS - PSTN - Service Provider - Asteriskbox1 - IAX2 trunk - Internet
- IAX2 trunk - Asteriskbox2 -Sip Clients
between asteriskbox1 and asterisk box2, I've VPN configured. from
Asteriskbox2 to internet my line speed is 1MB.
Is there any why that I can calculate how
Have a look at:
http://www.voip-info.org/wiki-Bandwidth+consumption
and
http://www.asteriskguru.com/tools/bandwidth_calculator.php
- Original Message -
From: Arun Kumar
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Monday, April 02, 2007 12:04 PM
Subject:
Hi,
I have a requirement for sending and receiving faxes and was wondering the best
way to achieve it with Asterisk as I only have one phone line.
I currently have a TDM11B in my server (1 x FXO, 1 x FXS), so I was thinking
that I would need to get a additional FXS module, connect that to a
Hi All,
Anybody have experience on LDAPget in Asterisk? Is it authentication
mechanism or not? After getting the details from LDAP, how can use those
details in Asterisk? Can anyone explain mail purpose of LDAPget in Asterisk?
Thanks in advance
--
Sravana
Hi all,
anyone knows if libpri has implemented Path Replacement Operation
(ANF-PR) for Q.SIG Protocol(ETS 300 258)?
I would like give this operation in an integretion with an Alcatle 4400,
but I don't know if this operation is implement.
Any idea?
Best Regards,
Tron
Anybody done LDAP authentication in Asterisk? can you explain how?
Thanks in advance
--
Sravana
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Klaverstyn, David C wrote:
Lspci does show:
03:02.0 Ethernet controller: Digium, Inc.: Unknown device 0120 (rev 11)
You have a TE120P, not a TE110P, so you are loading the wrong driver module.
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I've deployed about 40 GXP's and also haven't had the issues some are
reporting. The one issue is if I have to restart the server for any
reason I have to reboot all the phones for the BLF to light up properly.
This is easily accomplished by a script on the server, allows me to
reboot all the
Hi,
I have a problem with a Queue.
When a caller is first in queue and announcement starts the agent gets
open line or the 2nd in the line, not the first.
What can I do with that?
--
Suich
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Je suis absent du 2/04/2007 au 11/04/2007.
Je répondrai à votre message dès mon retour. Pour toute urgence, contacter
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To
Thanks for the response Ola. Turned out, I needed to add a Wait(1). That
extra second gave the provider of the PRI time to send the callerid info
as well, and then it could be sent correctly to the phones. :)
Rob
Ola Lidholm wrote:
On 29 mar 2007, at 23.01, Rob Schall wrote:
We have the
I have my system set up to check the cid of the calling number and if
the room number the user inputs matches the calling extension (the last
4 digits in my case) then the number is considered admin. This does have
the same downside that Dovid pointed out, the admin must be in the room
for
Hi,
I have installed the above two, and have two questions:
* Is there a reason (or better=a fix), why the chan-capi-1.0.0 does
not compile together with Asterisk 1.4.2?
* Anyone else experiencing problems with chan-capi-HEAD not seeing
the controller? If I run asterisk with verbose setting to
I am in the process of buying a TDM800 card from Yeastar (
http://www.yeastar.com/products.asp?TypeName=TDM800%20PCI%20CardcTypeName=1 )
Any one has tested this cards? How reliable are them? I am specially interested
in the FXO/FXS module.
--
Gustavo Felisberto
(HumpBack)
Web:
Hi, i registered freeworlddialup, and follow steps by the ebook
trixbox made easy.
i can make outbound calls, in my network and out (i can call a gizmo
number), and echo test, but i can not receive incomming calls, out of
network extensions, for example in the page freeworlddialup, i go to
the
Thanks Jason,
Could you post some sample code? What do you do if the CLI is not present ?
(i.e. international callee..)
Thanks,
Adrian Marsh
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jason Fuermann
Sent: 02 April 2007 14:32
To: Asterisk Users
On 4/1/07, Mark Hennessy [EMAIL PROTECTED] wrote:
Hi, I'm using Asterisk with two Cisco 7960 phones using SIP.
I'm seeing the following weird behavior:
SIP Phome 1 is extension 4002
SIP Phone 2 is extension 4003
So, did you name your SIP user/peers 4002 and 4003? It doesn't matter, but
the
Jason,
Ok, the 30VIP template seems to be working ok as far as button
assignment goes. I can define speeddial numbers to the speeddial
buttons. However, it appears that there is no code to support the
STIMULUS_SPEEDDIAL case. Is this correct?
Debug output from chan_skinny shows the following
How do I do that? Doesn't Ds create dynamic room numbers?
Thanks,
Adrian Marsh
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dovid B
Sent: 01 April 2007 11:02
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
On Sun, 2007-04-01 at 10:49 +0200, Philipp Kempgen wrote:
Priority jumping is deprecated anyways. Better use something
like Goto(s-${DIALSTATUS},1). See extensions.conf for examples.
Regards,
Philipp
I totally agree! While you can get what you want with +101
jumping, I highly suggest
- Chris Nighswonger [EMAIL PROTECTED] wrote:
Jason,
Ok, the 30VIP template seems to be working ok as far as button
assignment goes. I can define speeddial numbers to the speeddial
buttons. However, it appears that there is no code to support the
STIMULUS_SPEEDDIAL case. Is this correct?
Trying to create an extension that will toggle an enum value in our
database...
exten = s,1,MYSQL(Connect connid localhost myuser tmppass asterisk)
exten = s,n,MYSQL(Query resultid ${connid} UPDATE\ night_service\ SET\
status=(SELECT\ CASE\ status\ WHEN\ \'y\'\ THEN\ \'n\'\ ELSE\ \'y\'\ END));
hello friends,
is there any way to simplify that extensions.conf file?
[miprimerejemplo]
exten = 2,1,Dial(SIP/2,30,Ttm)
exten = 2,2,Hangup
exten = 2,102,Voicemail(2)
exten = 2,103,Hangup
exten = 20100,1,Dial(SIP/20100,30,Ttm)
exten = 20100,2,Hangup
exten =
I don't know if it's quite what you're looking for but the Linksys NSLU2
(slug) might do the job. It runs Linux natively and you have a choice of
2 or 3 distros and many pre-packaged apps. Not sure about connecting it
directly to a PC via USB but it has a 10/100 ethernet port. I paid
CDN$90
Use a macro
[macro-stdexten]
exten = s,1,Dial(${ARG1},30,Ttm)
exten = s,2,Hangup
exten = s-NOANSWER,1,Voicemail(${ARG2})
exten = s-NOANSWER,n,Hangup
then call it like
exten = 2,1,Macro(stdexten,SIP/2,2)
Josu Lazkano Lete wrote:
hello friends,
is there any way to simplify that
[miprimerejemplo]
exten = _X.,1,Dial(SIP/${exten},30,Ttm)
...
exten = s,1,Dial(SIP/${exten},30,Ttm)
...
Josu Lazkano Lete wrote:
hello friends,
is there any way to simplify that extensions.conf file?
[miprimerejemplo]
exten = 2,1,Dial(SIP/2,30,Ttm)
exten = 2,2,Hangup
exten =
Hi Josu,
[miprimerejemplo]
exten = 2,1,Dial(SIP/${EXTEN},3-,Ttm)
exten = 2,2,Hangup
exten = 2,102,Voicemail(${EXTEN})
exten = 2,103,Hangup
... is all you need in that context. Asterisk will match any called number
that starts with a 2 and is 5 digits long. ${EXTEN} carries the
On 4/2/07, Josu Lazkano Lete [EMAIL PROTECTED] wrote:
hello friends,
is there any way to simplify that extensions.conf file?
[miprimerejemplo]
exten = 2,1,Dial(SIP/2,30,Ttm)
exten = 2,2,Hangup
exten = 2,102,Voicemail(2)
exten = 2,103,Hangup
exten =
On Mon, 2 Apr 2007, Josu Lazkano Lete wrote:
hello friends,
is there any way to simplify that extensions.conf file?
Write a macro to do your 4-lines of 'dial'. This is in the book - get the
book read it. (Asterisk, The future of Telephony - it's free in PDF form
- pages 110 onwards)
On Mon, 2 Apr 2007, Peer Oliver Schmidt wrote:
Hi,
I have installed the above two, and have two questions:
* Is there a reason (or better=a fix), why the chan-capi-1.0.0 does
not compile together with Asterisk 1.4.2?
Actually chan-capi-1.0.0 does compile with Asterisk 1.4.2. The problem
Je suis absent du 2/04/2007 au 11/04/2007.
Je répondrai à votre message dès mon retour. Pour toute urgence, contacter
Emmanuelle Parache Moga ou Cédric Buzay.
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Steve, I was hoping for something native to Asterisk, ie something not
requiring a new process.
Steve Totaro wrote:
Madplay
Doug Garstang wrote:
Oh poo. No one seems to know. :(
Doug Garstang wrote:
All,
Is there a dial plan command that can stream uncompressed audio from
another source?
I think that the Polycom phones are really good, but their lack of support for
Asterisk is legendary. Of those I prefer the IP600/601 for the better display.
The Aastra 480i and 480i CT are my other favorite. This is currently what lives
on my desk. It supports multiple lines but only 2
Hello Armin,
Actually chan-capi-1.0.0 does compile with Asterisk 1.4.2. [..]but you [..]
just
change the line
if grep -q ASTERISK_VERSION_NUM 0104 $INCLUDEDIR/version.h; then
to
if grep -q ASTERISK_VERSION_NUM .*104 $INCLUDEDIR/version.h; then
in create_config.sh
thanks, will try.
BJ Weschke wrote:
On 3/29/07, Jordan Novak [EMAIL PROTECTED] wrote:
What is the most stable version supporting queue priority. I have had
many
crashes, I am using 1.2.11 and have set the weight in queues.conf. is
there
a better way or a patch. I can't seem to find much. Any suggestions?
Jordan Novak wrote:
I am running 1.2.11, I have had the queue priority lock the system up
twice in the last week. I will forgo how I know this is what caused
it, just trust me. I used the weight function in queues.conf to add
priority to the queue. Is there another way to do it that will make
Philipp Kempgen ha scritto:
Edoardo Serra wrote:
The purpose of thisi implementation is to deal with some carriers that
give us the call as ANSWERED when the called party is still ringing.
Our billing software is billing the user (and the carrier is billing us)
even with unsuccessful calls.
On 3/30/07, Rizwan Hisham [EMAIL PROTECTED] wrote:
If you are using sip then you should look for the call-limit option in
sip.conf file.
Using IAX. Is that a problem?
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After working with the Grandstream GXP 2000 series phones, I have
decided that I am quite unhappy with their problems, both voice
quality, volume, features and others. For their price now, there are
plenty of phones to choose from as well.
A couple of years ago when we first started
Andrew Joakimsen wrote:
The logic of the macro is totally opposite of what it should be. I do
recall sending a corrected version of the script to someone a while
back, it might be on the mailing lists archive.
However, there is an option for the Dial() command to do exactly what you wish
p:
Hi,
I have FRITZ!Card PCI card. I have installed misdn-1.1.0 on stable debian
3.1 with 2.6.8. kernel. Then I reboot system, and it doesn't boot, it stops
near Apache2 starting I started my system with recovery kernel,
and tun off misd, then my system works fine. I think it's problem with
I found a subtle difference between the two traces you sent (the call that
works and the call that gets dropped). This may or may not be what's causing
the problem.
The call that gets dropped had a retransmission of INVITE from UAC to
UAS (and therefore retransmission of 200 OK from UAS to UAC).
Anyone ever tried pointing app_playback at a named pipe ?
On 2 Apr 2007, at 17:11, Doug Garstang wrote:
Steve, I was hoping for something native to Asterisk, ie something
not requiring a new process.
Steve Totaro wrote:
Madplay
Doug Garstang wrote:
Oh poo. No one seems to know. :(
On 4/2/07, Jason Parker [EMAIL PROTECTED] wrote:
Yes, you are (mostly) correct. Speeddials can be added to the phone, but they
can't actually be used.. There is code there that is #ifdef'd out, because it
(mostly) does not work.
If this thread should be moved to the -dev list, just let me
Somewhere I found a link to gsutil utility that someone wrote. A quick
google looks like
http://freshmeat.net/projects/gsutil/?branch_id=59227release_id=219046
is probably the best place to get it.
This utility actually has quite a few nifty options, here's a copy of
the help:
Version 3.0 of
For build quality and footprint, the SPA-942s are hard to fault, but the web
interface isn't the most pleasant and remote provisioning is hit-and-miss (mainly
dependent on how cooperative your supplier is at getting you the provisioning
documentation/software).
As for Linksys, their
First-sale doctrine, unless your vendor did something illicit to
obtain Polycom phones there is nothing they can do about it. It's good
to know Polycom has anti-competitive business practices. I also
dislike that they refuse to give out anything but old firmware
versions too.
On 3/30/07, Stephen
Getting no service display on aastra 480i. Sip debug shows an unathorized
blub when the aastra tries to register.
Some reading indicates that 1.4 firmware wants aastra.cfg and mac.cfg in
/tftpboot/. There are none.
Anyone have basic config files? Or can point me to a good link? All links I
I have an FXS port and an FXO port on my Sangoma, can I detect inbound Fax
calls on my FXO port and route them automatically to the FXS port (connected
to a fax machine) while allowing normal voices to ring the main extension
like normal?
I looked through archive but didn't see this exact
Je suis absent du 2/04/2007 au 11/04/2007.
Je répondrai à votre message dès mon retour. Pour toute urgence, contacter
Emmanuelle Parache Moga ou Cédric Buzay.
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On Mon, 2 Apr 2007, Chris Bagnall wrote:
This is easily accomplished by a script on the server, allows me to
reboot all the phones with one command and takes about 2 minutes. I'm
very happy with these phones.
How would you feel about sharing that script with the list? I'm sure
it'd be of
You don't need the cfg files (or a tftp) to boot the phones or register.
There are some sample configs lying around, but Aastra's are very poorly
documented (and their firmware still has big bugs - so don't modify from
default too much). We've setup a number of 480i's and got very frustrated
with
Salvatore Giudice wrote:
You should be aware that flash memory is generally not the best medium to
store data when you have a high number of read/writes. Flash memory will
fail much more quickly under these conditions.
Does this mean that devices such as the samsung Flash SSD (part #
Does the GotoIfTime application work in the local time of the server, or
UTC?
--
Alan Chandler
http://www.chandlerfamily.org.uk
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On 4/1/07, Matthew Rubenstein [EMAIL PROTECTED] wrote:
I need a USB microprocessor *device* on which the Linux and Asterisk
will run (even if very slowly), not just a storage drive from which to
run it on the PC. MonteVista is a good distro, though there are other
minimal embedded
David Ruggles wrote:
I have an FXS port and an FXO port on my Sangoma, can I detect inbound Fax
calls on my FXO port and route them automatically to the FXS port (connected
to a fax machine) while allowing normal voices to ring the main extension
like normal?
Fax detection is spotty at
Does any one know if there's an mechanism (internal to asterisk or
otherwise) to replicate dynamic SIP device registrations across a pool
of asterisk servers?
I'm in the process of creating a asterisk cluster using a SIP hardware
load balancer and so far this is one of the challenges I'm
Joe Acquisto [EMAIL PROTECTED] Wrote on: 4/2/2007 3:03 PM:
Getting no service display on aastra 480i. Sip debug shows an
unathorized blub when the aastra tries to register.
Some reading indicates that 1.4 firmware wants aastra.cfg and mac.cfg
in /tftpboot/. There are none.
Anyone have
On Mon, 2 Apr 2007, David Ruggles wrote:
I have an FXS port and an FXO port on my Sangoma, can I detect inbound Fax
calls on my FXO port and route them automatically to the FXS port (connected
to a fax machine) while allowing normal voices to ring the main extension
like normal?
Yes.
On Mon, 2 Apr 2007, Alan Chandler wrote:
Does the GotoIfTime application work in the local time of the server, or
UTC?
Looking at the code, I'd say it was localtime of the server.
From pbx.c, in ast_check_timing() :
time(t);
localtime_r(t,tm);
Gordon
On Mon, 2007-04-02 at 15:03 -0400, Joe Acquisto wrote:
Getting no service display on aastra 480i. Sip debug shows an
unathorized blub when the aastra tries to register.
Some reading indicates that 1.4 firmware wants aastra.cfg and mac.cfg in
/tftpboot/. There are none.
Anyone have
Michelle Dupuis [EMAIL PROTECTED] Wrote on: 4/2/2007 3:23 PM:
You don't need the cfg files (or a tftp) to boot the phones or register.
There are some sample configs lying around, but Aastra's are very poorly
documented (and their firmware still has big bugs - so don't modify from
default too
On Mon, 2007-04-02 at 15:07 -0400, David Ruggles wrote:
I have an FXS port and an FXO port on my Sangoma, can I detect inbound Fax
calls on my FXO port and route them automatically to the FXS port (connected
to a fax machine) while allowing normal voices to ring the main extension
like normal?
I'm building a dialplan for use with a bunch of GXP2000 desk sets. During
testing, we had some user issues surrounding the lack of an on-phone
dialplan. Users would hit 9 and sit there waiting for a redial tone, and
the GXP would time out, sending just '9' to *, which couldn't do much other
Hello,
I'm using Realtime to select extensions out of a database so that we can
provision inbound tollfree on the fly. Once I 'catch' the inbound, I
want to get out of realtime and use the regular extensions again. I
thought I could just use the goto statement and go to another
On 4/2/07, John C. Wolosuk Jr. [EMAIL PROTECTED] wrote:
Does any one know if there's an mechanism (internal to asterisk or
otherwise) to replicate dynamic SIP device registrations across a pool
of asterisk servers?
I'm in the process of creating a asterisk cluster using a SIP hardware
load
On Mon, 2 Apr 2007, Peer Oliver Schmidt wrote:
Hello Armin,
Actually chan-capi-1.0.0 does compile with Asterisk 1.4.2. [..]but you [..]
just
change the line
if grep -q ASTERISK_VERSION_NUM 0104 $INCLUDEDIR/version.h; then
to
if grep -q ASTERISK_VERSION_NUM .*104
Thanks for answering my question. I apologize for not looking harder, I'll
look harder next time before asking the list.
Thanks,
David Ruggles
CCNA MCSE (NT) CNA A+
Network EngineerSafe Data, Inc.
(910) 285-7200 [EMAIL PROTECTED]
___
Hello
I'd like to know too
On 4/2/07, Gustavo Felisberto [EMAIL PROTECTED] wrote:
I am in the process of buying a TDM800 card from Yeastar (
http://www.yeastar.com/products.asp?TypeName=TDM800%20PCI%20CardcTypeName=1)
Any one has tested this cards? How reliable are them? I am specially
Hello Armin,
[EMAIL PROTECTED]:~# grep capi /var/log/asterisk/messages
[Mar 31 16:17:03] ERROR[3850] chan_capi.c: Unable to listen on contr1
(error=0x100f)
Is this helpful, or do you need more information?
Yes, at this state it might be possible that less CPU power causes
problems. The
Hi all !
I'm trying to make a automonitor generated filename to make its way
into CRD(usrefiled), so I can keep track of recorded conversations in
CDR logs. Looking how to do that, I have found cool (but almost
undocumented) option of res_monitor: if you set monitor format in form
of
What is the driver for the TE120P
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kevin P.
Fleming
Sent: Monday, 2 April 2007 10:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Problems with TE110P
wcte12xp
- David C Klaverstyn [EMAIL PROTECTED] wrote:
What is the driver for the TE120P
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kevin
P.
Fleming
Sent: Monday, 2 April 2007 10:59 PM
To: Asterisk Users Mailing List - Non-Commercial
That information is listed in the README file that comes with the Zaptel
source code.
Klaverstyn, David C wrote:
What is the driver for the TE120P
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kevin P.
Fleming
Sent: Monday, 2 April 2007 10:59 PM
To:
From what I can tell the Zaptel 1.2.16 does not have a driver for the
TE120P. Is this correct and if so how do I get it?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric
ManxPower Wieling
Sent: Tuesday, 3 April 2007 9:11 AM
To: Asterisk Users Mailing
Je suis absent du 2/04/2007 au 11/04/2007.
Je répondrai à votre message dès mon retour. Pour toute urgence, contacter
Emmanuelle Parache Moga ou Cédric Buzay.
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To
On Mon, 2007-04-02 at 15:03 -0400, Joe Acquisto wrote:
Getting no service display on aastra 480i. Sip debug shows an
unathorized blub when the aastra tries to register.
Some reading indicates that 1.4 firmware wants aastra.cfg and mac.cfg in
/tftpboot/. There are none.
Anyone have
OK I now have
MODULES:=zaptel wcte12xp
In my makefile and the driver loads but the card still fails.
lsmod | grep -i wct
wcte12xp 39360 0
zaptel184996 1 wcte12xp
ztcfg -vv
Notice: Configuration file is /etc/zaptel.conf
line 0: Unable to open master device
Permissions?
Sds,
Gustavo
From: Klaverstyn, David C [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
Subject: RE: [asterisk-users]
OK,
Found the problem.
It looks like the configuration file is not correct.
I added the following line to /etc/sysconfig/zaptel
MODULES=$MODULES wcte12xp # TE120P - Single Span T1 Card
Once I did this all is now working.
Editing the zaptel.sysconfig file in the zaptel source code will
I had a similar issue. Aastra tech support told me I needed to configure
the NTP server for UTC which solved my problem.
-Original Message-
From: Joe Acquisto [mailto:[EMAIL PROTECTED]
Sent: Monday, April 02, 2007 12:44 PM
To: asterisk-users@lists.digium.com
Subject: Re:
Seems you don't know what your doing then. Everything you say does not work
works for everyone else! EVERYTHING!
-Original Message-
From: Joe Acquisto [mailto:[EMAIL PROTECTED]
Sent: Monday, April 02, 2007 1:15 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject:
On Mon, 2007-04-02 at 16:30 -0700,
[EMAIL PROTECTED] wrote:
Date: Mon, 02 Apr 2007 20:26:09 +0100
From: Thomas Kenyon [EMAIL PROTECTED]
Subject: [asterisk-users] Re: On Topic: Cheapest Asterisk USB Key?
To: Asterisk Users Mailing List - Non-Commercial Discussion
I have a brand new TE120P card that I have installed and asterisk is not
starting as I am getting the error: ERROR[5054] chan_zap.c: Unknown
signalling method 'pri_cpe'
It seems it does not matter what I change the vaule for signalling= to,
it always returns it as invalid.
I have tried the
Does any one know if there's an mechanism (internal to asterisk or
otherwise) to replicate dynamic SIP device registrations across a pool
of asterisk servers?
I'm in the process of creating a asterisk cluster using a SIP hardware
load balancer and so far this is one of the challenges I'm
Andrew Joakimsen wrote:
First-sale doctrine, unless your vendor did something illicit to
obtain Polycom phones there is nothing they can do about it.
What they can do is refuse to keep supplying the vendor, and that's a
threat the vendors tend to take seriously, especially if the product is
any
I upgraded to 1.4.1 and my DTMF has stopped working, I tried
rfc2833compensate=yes and relaxdtmf=yes etc but none working.
Everything seems to work fine with 1.2.10
Is there any way I dump the dtmf data packets received by asterisk on
console?
Any idea or pointers to debug the issue will be
I can reproduce these symptoms with both a Sipura 2002 and a Polycom 501.
On Friday 30 March 2007 02:27, Rizwan Hisham wrote:
What kind of phone are you using? some devices which connect you pstn phone
to voip network have their own special dtmf keys defined inside their
configuration to
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