On 21:22, Sat 04 Mar 06, C F wrote:
vi here
vim :) Combined with the syntax file for asterisk.
--
Michiel van Baak
[EMAIL PROTECTED]
http://michiel.vanbaak.info
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D
Why is it drug addicts and computer afficionados are both
On 20:19, Sat 04 Mar 06, Ryan Laginski wrote:
Downgrade to 1.0.10. I was unable to get the 12sp+ to work reliably in
1.2.0-1.2.4 and had the same problem.
You could try the chan-sccp.org driver for skinny/sccp
The 12SP+ is listed as supported device.
--
Michiel van Baak
[EMAIL PROTECTED]
On 20:52, Sat 04 Mar 06, [EMAIL PROTECTED] wrote:
We're still waiting for a SIP-enabled 7970...
The newer model phones (7941g/ge, 7961g) are sccp-only. Seems a step
backwards to me.
why?
I had my phones running on SIP, got chan-sccp and started
experimenting with it.
All my phones are
On Fri, Mar 03, 2006 at 03:06:02PM -0500, S McGowan wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hey all,
First of all, hello again! Been a while since I've posted to the
list, but I've been here lurking and watching ;-)
Anyway, I wanted to pose a general question to the list
On Sun, 5 Mar 2006, Michiel van Baak wrote:
On 20:52, Sat 04 Mar 06, [EMAIL PROTECTED] wrote:
We're still waiting for a SIP-enabled 7970...
The newer model phones (7941g/ge, 7961g) are sccp-only. Seems a step
backwards to me.
why?
If cisco really is moving towards SIP as claimed earlier,
Hi
I am using asterisk 1.4 on RHEL4
I am sending this mail to the mailing list , to
enquire wheter any one had faced simillar problem
which I am facing now
I am facing a problem which is not able to solve
or understand , the problem is that I cannot log into
the mailbox from a VoIP
Hello everybody,
I'm experiencing a strange problem with my Asterisk. I hope you can help:
Asterisk is running at my company behind NAT. Ports 5060 and 1-2
are being forwarded to it. I have put the router's external IP-address
into externip in sip.conf. At home I'm using an AVM
I suppose you are using 1.2.4 asterisk version
Maybe is not sending dtmf tones as rfc2833 and inband mode is not being
detected by your asterisk box.
Im a wrong? Could you try to configure dtmf tones on your softphone?
John Joseph escribió:
Hi
I am using asterisk 1.4 on RHEL4
I am
Just trying to think - are you using the standard E1 setup from ATP?
I have found that the settings on their website work pretty well.
Also - have you tried to put an answer in your dialplan? That might keep the
dialplan open..
later,
PaulH
- Original Message -
From: Paul C
On 02:08, Sun 05 Mar 06, [EMAIL PROTECTED] wrote:
On Sun, 5 Mar 2006, Michiel van Baak wrote:
On 20:52, Sat 04 Mar 06, [EMAIL PROTECTED] wrote:
We're still waiting for a SIP-enabled 7970...
The newer model phones (7941g/ge, 7961g) are sccp-only. Seems a step
backwards to me.
why?
If cisco
Thanks Alberto
I am able to login now , I had used the option
dtmfmode=auto
thanks
Joseph John
http://www.voip-info.org/wiki-Asterisk+sip+dtmfmode
--- Alberto Sagredo [EMAIL
Hi James,
I am definitely interested in the card and also in the results of your
testing.
Regards,
David
LEXNET PTY LTD
[e] [EMAIL PROTECTED]
[m] 0411 172 667
Mail: PO Box R1180
Royal Exchange, Sydney NSW 1225
-Original Message-
From: [EMAIL
On Sun, 5 Mar 2006, Michiel van Baak wrote:
On 02:08, Sun 05 Mar 06, [EMAIL PROTECTED] wrote:
sccp and asterisk has some err.. real annoying bugs at the moment, where
ciscos running SIP don't have these problems.
Yeah, but still I can live with that because all the other
things make up for
Come on.! Don't tell me no one has ever had a problem on this model
with asterisk? Live it up guys... and make a few comments
Cheers
Stephen
Stephen Arulraj wrote:
Anyone knows how to hook this up with Asterisk?
___
--Bandwidth and
did you uncommnet # from before ztdummy ?
--- Sina Bahram [EMAIL PROTECTED] wrote:
Hi all,
I hope everyone is doing well. I just joined the
list, and I've really
enjoyed all I have read about asterisk so far.
Unfortunately, I'm having a
bit of trouble implementing this thing :).
By
make linux26
make install
worked for me
thanks
--- Dovid Bender [EMAIL PROTECTED] wrote:
did you uncommnet # from before ztdummy ?
--- Sina Bahram [EMAIL PROTECTED] wrote:
Hi all,
I hope everyone is doing well. I just joined the
list, and I've really
hi Zach,
i would use GOTOIF to forward the DID from within the [incoming] context to the other context. i would try :
exten = gotoif($[did]=DID1,goto did1|s|1,)
exten = gotoif($[did]=DID2,goto did2|s|1,)
On 3/4/06, Zach A [EMAIL PROTECTED] wrote:
Both DIDs are SIP and from the same provider.
VI as well but sometimes I use the editor built into WinSCP.
Thanks,
Steve Totaro
-Original Message-
From: C F [mailto:[EMAIL PROTECTED]
Sent: Saturday, March 04, 2006 9:22 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Preferred
I would look at the cost of the channle banks vs.
selling the analog phones and getting very basic voip
hardphones.
--- Conrad Wood [EMAIL PROTECTED] wrote:
Does anyone have any recommendations on how to
connect 160 analogue
phones to an asterisk PBX?
Background information:
A client
On Sun, 05 Mar 2006 19:56:13 +0800
Stephen Arulraj [EMAIL PROTECTED] wrote:
Come on.! Don't tell me no one has ever had a problem on this model
with asterisk? Live it up guys... and make a few comments
maybe you would get more answers if you wouldn't steal a thread, but would
create
Kevin Smith wrote:
Hey everyone,
We have a special mail box for certain customers when we are out of
the office. Basically they enter a pin number and if it is valid they
leave a message and it notifies the on call techs. My question is
regarding externnotify for the voice mail.conf. If I
Hello,
Anyone knows a way to show real-time content from a DB into the LCD display
of an IP phone, like any 79xx?
If someone knows which phone is capable of doing and how, like using XML
files, please advise.
Regards,
Max Glucksmann
e-mail: [EMAIL PROTECTED]
Web: http://www.comtel-networks.com
Hi folks,I've just updated my * and I noticed that from the update the uniqueid field into mysql, is not written and ASTPP do not charge the calls.I got an eye to cdr_mysql.c and I found that at line 212, into one insert query, uniqueid is missing.
But I can be wrong.In any case, somebody got same
I'm experiencing a strange problem with my Asterisk. I hope you can help:
Asterisk is running at my company behind NAT. Ports 5060 and 1-2
are being forwarded to it. I have put the router's external IP-address
into externip in sip.conf. At home I'm using an AVM FritzBox Fon WLAN
News!I've just replaced the cdr_addon_mysql.so with the old one, and it start to work properly!So I can suppose a bug into that module.I'll check the old cdr_addon_mysql.c and see difference of code, if any.
Thanks.2006/3/5, FaberK [EMAIL PROTECTED]:
Hi folks,I've just updated my * and I noticed
We're still waiting for a SIP-enabled 7970...
The newer model phones (7941g/ge, 7961g) are sccp-only. Seems a step
backwards to me.
why?
If cisco really is moving towards SIP as claimed earlier, then releasing
new phones which are sccp-only is a step backwards from that goal.
If
Conrad,
i would go with following solution:
1.6 sets of Audio Codes of 24 FXS ports conected by SIP accounts to the system. the type is MP 124. then you open the conector on the initial MDF and then the users have the same phone on their table
2. one dual Xeon system (or even stronger - 2 Dual
Might take a close look at group = 1 in your zapata.conf file. That
should be group=1.
Someone mentioned adding w into your outbound calls, like:
exten = _9XX,1,Dial(${OUTBOUNDTRUNK}/ww${EXTEN:1})
Did you try that in each of your Dial strings?
In our area code(703), and I am not
You need to compile asterisk-addons with CFLAGS+=-DMYSQL_LOGUNIQUEIDcheck:
http://www.voip-info.org/tiki-index.php?page=Asterisk%20cdr%20mysql
On 3/5/06, FaberK [EMAIL PROTECTED] wrote:
News!I've just replaced the cdr_addon_mysql.so with the old one, and it start to work properly!So I can suppose
I have to wait until Monday to test but I will make that change.thanks Rich Adamson [EMAIL PROTECTED] wrote: Might take a close look at group = 1 in your zapata.conf file. Thatshould be group=1.Someone mentioned adding "w" into your outbound calls, like: exten =
When I receive a call from fwd, I'd like to insert a prefix
prior to the caller ID - 1) to be able to look it up in a database
ofidentified numbers and 2) for the receiver to be able to dial it
back.
So what I need is to identify the DID and based on that,
insert the prefix.
Any pointers
SellVoIP appears to follow a US dialplan. A US numberis
dialled as 1NXXNXX whereas an international (to the US) numberis
dialled as 011X.
Frankly, I didn't ask whether international numbers like
Barbados where the code remains as 1 butare international (to the US) need
the 011 or can be
Hi,
Companies like DIDx and Sixtel,
when they state DIDs at $XX.XX per month and $XX.XX
per minute/monthly,
do these companies provide inbound and outbound routing
of calls, or are these rates strictly for inbound
Call routing of DIDs?
Thanks.
On Sat, Feb 25, 2006 at 11:24:36PM +0100, Thomas Artner wrote:
Am Saturday 25 February 2006 22:59 schrieb Anton Krall:
I cant get faxes right now with tdm, something is wrong but, what do I need
to have in order to convert from tiff to pdf?
I have the mailfax script that invokes tif2ps
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Martin Joseph
|Sent: Friday, March 03, 2006 1:46 PM
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: Re: [Asterisk-Users] dtmf tones problem with unicall and E1
|
|
|On Mar 3,
Did that too, same errors
Take care,
Sina
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John Joseph
Sent: Sunday, March 05, 2006 7:33 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Problem compiling
Hi all I'm a newbie in asterisk.I ant to know how i ca configure asterisk to work with the nathelper module of openser to fix the nat problem!Thanks in advance! bets regards Serge
Nouveau : téléphonez moins cher avec Yahoo! Messenger ! Découvez les tarifs exceptionnels pour
Inbound should be free as far as I am concerned unless you have a toll
free number.
Thanks,
Steve Totaro
_
From: VIC IP Communications [mailto:[EMAIL PROTECTED]
Sent: Sunday, March 05, 2006 11:28 AM
To: Asterisk-Users@lists.digium.com
Subject: [Asterisk-Users] re: Sixtel Services
At 07:57 AM 03/05/2006, you wrote:
How can I strip the 00 and insert 011 in one entry in the
dialplan. I'm stripping the 00 and passing the rest of the numbers
for numbers dialled as 001X. (as in: 00|1XX.) but in case of
numbers out of the US, how would I insert the 011 ?
exten = _011X. ,
i have AAH
connected to pstn via digium TDM01B
had been testing
it on telewest line (UK cable company) with very little issues.now moved to
a BT line and had several that i anticipated from infomation on this
list.the one that has caught me out is low volume from the caller via pstn.
Being a sixTel customer I can tell you how sixTel bills. They charge $X.XX
per month for a DID, they also charge per minute inbound (a certain rate)
and they charge outbound at another rate.
-Original Message-
Date: Sun, 5 Mar 2006 11:28:16 -0500
From: VIC IP Communications [EMAIL
what about this?
[incoming]
exten = DID1,1,Goto(incoming1,${EXTEN},1)
exten = DID2,1,Goto(incoming2,${EXTEN},1)
Julian.
On 3/5/06, Tele Cost Price Reducer [EMAIL PROTECTED] wrote:
hi Zach,
i would use GOTOIF to forward the DID from within the [incoming] context to
the other context. i
Andrea,
Thinking back to your question, Andrea, I'm really wondering whether or not
software solutions like FOP could or would ever scale to serve, for example,
a 200 seats, full time receptionist.
Obviously, software is flexible and with a suitable keyboard and a smart
software-hardware
While testing a
problem with "spontaeously" and "occasionally" rebooting asterisk, I came upon
this problem:
Program received signal SIGSEGV, Segmentation fault.
[Switching to Thread -1210770512 (LWP 11346)]
0x002e3fe1 in pri_release_timeout (data="" at
q931.c:2589
2589 q931.c: No such
[EMAIL PROTECTED] wrote:
On Sun, 5 Mar 2006, Michiel van Baak wrote:
On 21:22, Sat 04 Mar 06, C F wrote:
vi here
vim :) Combined with the syntax file for asterisk.
http://www.bemroses.net/images/curves.jpg
-Dan
Rotfl!!!
Looks like whoever drew the emacs curve couldn't program
We still have a seat open in the London
Introduction to Asterisk class.
TKS
Paul
Paul Mahler
[EMAIL PROTECTED]
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of McQuiggan, Mark xt46480
Sent: Sunday, March 05, 2006 12:20
PM
To:
Hi,
I'm going to send a Sipura SPA-3000 to one of my friends in Egypt.
Does anybody has experienced any difficulties configuring the SPA-3000 to meet
the Egyptian PSTN network norms.
Appreciate your help.
__
Do You Yahoo!?
Tired of spam? Yahoo!
We have a bunch of PAP2s, and using the # to send immediately does not
work as described in the manual. The PAP still waits for the
Interdigit_Short_Timer to expire before sending the dial string. In
addition, the dialplan does not cause the string to be sent
immediately as it should.
Here's
Thank you. works like a charm. I'm using [EMAIL PROTECTED] so I had to massage
it into AMP's structure.
Your example is actually the reverse of what I needed to do, but that's not
the issue.
AMP uses a macro to dial (syntax almost exactly the same).
I feel this should be documented somewhere
Find perl code attached:
while ($count = $BACK)
{
print STDERR $count\n;
@item = pop(@text);
print STDERR @item\n;
$count++;
}
regards,
PaulH
On Sat, 2006-03-04 at 07:54 -0800, Michael Collins wrote:
I actually got it all working - but it's great to see where we did the
same
I have a issue with two Zap clone cards where they used to work.I am using [EMAIL PROTECTED] 2.5 which includes Asterisk 1.24 and Zaptel drivers 1.2.4. The system is a new Intel Celerion machine. I used to have the same cards running in a Intel PIII system. in this system, they worked. In this
Comments inline:
a vim user myself. I don't use most of what you descvribe below,
however:
1. Syntax Highlighting, and ease of updating that highlighting
Update asterisk.vim
Good idea, primary issue being I'd have to learn vim, but it's looking like a
LOT of people agree with the concept of
When I bought two Polycom 501 SIP phones, I naively thought they were
Power-over-Ethernet (IEEE 802.3af) because they were powered over
ethernet. Silly me.
Polycom must have some odd voltage or funny way of injecting the
power, because the POE switch I bought for them (Netgear [EMAIL
For Polycom IP500/501's and IP300/301's you need a special polycom POE
cable.
When you buy Polycom phones you can usually specify POE or powerpack.
PaulH
On Sun, 2006-03-05 at 16:23 -0800, William M Conlon wrote:
When I bought two Polycom 501 SIP phones, I naively thought they were
I guess the way you want to do it should work, (over a long run you
might run into trouble, but only trial and error will confirm this).
However keep in mind that the polycom cables come keyed on one end of
the RJ45, so that you don't by mistake put the powered end into the
switch. What that
I saw that Polycom offered a cable (not stocked anywhere), at $40 a
pop for 802.3af connections. That's what made me think the phone
itself is NOT 802.3af compliant.
Presumably, for $40, there's more than a fuse in that special cable.
On Mar 5, 2006, at 4:31 PM, Paul Hales wrote:
For
As I understand 802.3af, the phones go through a negotiation with the
unit supplying the power. I don't think it's a matter of -48VDC on a
particular pair. I remember a schematic from years ago--it had each of
the receive pair and the transmit pair going into a transformer winding,
and that
My recollection of the marketing fluff was that we would just use our
legacy network (cables) and the devices at both ends would figure out
whether they were sourcing, sinking, or neither. In the case of the
501, it's the special Polycom cable, either with or without provision
for an AC
The IP501 does not have a power jack. You'll need one of the Polycom
cables.
William M Conlon wrote:
My recollection of the marketing fluff was that we would just use our
legacy network (cables) and the devices at both ends would figure out
whether they were sourcing, sinking, or neither.
Funnily enough upgrading to 1.2.x solved my problems! Well that and optus
changing some stuff as well. zaptel-trunk drivers also helped a lot with
my echo problems.
- Original Message -
From: James Sturges [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial
lookup info on RX and TX gain on voip-info.org
On 3/5/06, billy [EMAIL PROTECTED] wrote:
i have AAH connected to pstn via digium TDM01B
had been testing it on telewest line (UK cable company) with very little
issues.
now moved to a BT line and had several that i anticipated from infomation
In 1.0.9, if I Dial() with the M option, the specified macro executes just
fine, however there is several seconds of silence - no audio transmitted to
either caller or callee. After 5 or 6 seconds (in my installation) call
audio is transmitted normally. Is this known behavior? Can't seem to find a
Hi
I run all my Linux boxes without support for kernel modules. I'm in
the process of setting up an Asterisk PBX and I want to avoid enabling
modules on this box too. Is it possible to compile zaptel drivers
statically into the Linux kernel?
TIA
Paolo
Not true. Some do and some don't. Some have a place to plug a separate DC
adapter, and some have the inline power, where the adapter plugs into the
ethernet cable. Not sure which ones are newer, and which are older.
-Original Message-
From: Michael Welter [mailto:[EMAIL
The IP300/301 has the power jack, the IP500/501 the inline cable.
PaulH
On Sun, 2006-03-05 at 20:56 -0700, Douglas Garstang wrote:
Not true. Some do and some don't. Some have a place to plug a separate DC
adapter, and some have the inline power, where the adapter plugs into the
ethernet
Is there a variable to read to see how many calls are currently open?
(related to channel status?)
PaulH
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Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
SoX needs that libid3tag, libmad and madplay are installed before it can
read mp3 files and convert them into some other format.
Zach A
-Original Message-
From: Chris Stenton [mailto:[EMAIL PROTECTED]
Sent: Thursday, March 02, 2006 3:30 AM
To: Asterisk Users Mailing List -
Hi,
The 3 MP3 files which are installed with asterisk, what is their bit
rate, are they mono and do they have ID3 tags?
Zach A
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To UNSUBSCRIBE or update options
Zach A wrote:
Hi,
The 3 MP3 files which are installed with asterisk, what is their bit
rate, are they mono and do they have ID3 tags?
Zach A
{192}([EMAIL PROTECTED]:Desktop)# file /var/lib/asterisk/mohmp3/*.mp3
/var/lib/asterisk/mohmp3/fpm-calm-river.mp3: MPEG ADTS, layer III, v1,
128
Hi guys. Without having a FWD account, can Asterisk redirect calls to FWD?
For instance, an extension behind Asterisk dials 99751234, and Asterisk
says that starts with 99. let's strip off the 99 and call 751234 at FWD,
IE: sip:[EMAIL PROTECTED]:5060.
Is that possible, or would services such
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