Is there a product to bring in more than one POTS line short of a full
T1? It just seems silly that the technology hasn't advanced any
further than to have a single line per card.
We are working on an FXO module for the TDM400P and hope to have it ready
in a couple of months for initial
[snip]
And to DISA:
* Properly handle extensions with multiple matches and dots
Please let me know on or off list about any feedback you have regarding
these changes.
Mark
OK, this patch seems to work after some tests. DISA no longer
jumps to wildcarded strings that might match in the called
Is there a product to bring in more than one POTS line short of a full
T1? It just seems silly that the technology hasn't advanced any
further than to have a single line per card.
We are working on an FXO module for the TDM400P and hope to have it ready
in a couple of months for initial
HI,
can somebody tell me how and where must I put the SIP register line? I
think is in [general] section of the sip.conf and that I have to put:
register = user:[EMAIL PROTECTED]:port/localextension
but, user and password of the SIP gateway? Because I'm trying this and
doesn't work...
thanks
Hi all,
I am sorry to ask you this newbie question again but I did not get any answer:
If I have installed a asteriskpbx at the public internet what would be my own
sip address ? I guess it depends on the callerid . ??
sip.conf
[phone2]
type=friend
host=dynamic
dtmfmode=rfc2833
Hi Olaf,
It all depends on the dailplan you have setup in extensions.conf and the
general context in your sip.conf.
In your sip.conf in the general section is the context in which all non
specified sip connections will be dropped, for example:
context=sip-inbound
then make the sip-inbound
On Sun, 15 Jun 2003, John Laur wrote:
I do not think it is necessarily a hardware issue, as the line-in-use
lights do not light until the wcfxo kernel module is loaded. It would be
very nice for asterisk to be able to share these lines via the PBX..
That is very interesting. I have
I would probably want to buy 2 or 3 licences to test with and then
later as I needed more add then on as needed one or two at a time.. Is
this possible??
It doesn't support multiple licenses (thanks again, Voiceage) and it's a
pain in the asterisk to upgrade a license from one size to another,
Hey all,
I have a E1 setup with a E400P digium card. Everything works just great
except for the callerid. When i make an outgoing call via the E1 to a
hardphone somewhere, all i get is private number. In my zapata.conf
however , i have defined the following:
context=localE1
group = 1
Hello group,
Just got this reply from Sasha @ Smart Link, a soft modem chipset
company with a very nice attitude towards Linux. BTW, You can check the
status of most modem chipsets and Linux support here:
http://www.idir.net/~gromitkc/dips/roster.html
The good news is that the current 2.7.x
As far as SMP and single T400P - we'll try and report the results
but the idea was to go with as high density as possible ...
Right, I'm just trying to narrow down the problem. I'm theorizing that
the problem is some sort of spinlock deadlock. Does it only occur if
there is activity or even
Hello everyone.
I am having problems getting speex support.
It seems * is not loading speex. When i did a make in the codecs sub dir,
the following error pops up when making speex:
codec_speex.c:34:19: speex.h: No such file or directory
is this file missing in the cvs as i just removed the
U need to have speex libraries and header file allready installed on your box.
otherwhiles * will not find it and not use it :)
On Tuesday 17 Jun 2003 1:27 pm, Tjardick van der Kraan wrote:
Hello everyone.
I am having problems getting speex support.
It seems * is not loading speex. When i
does anyone know what compression uses Speex ? and which gateways used it so
they can work with Asterisk.
Thanks,
- Original Message -
From: Emanuele Pucciarelli [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, June 17, 2003 8:49 AM
Subject: Re: [Asterisk-Users] Speex
On Tue,
On Tue, 17 Jun 2003, Mark Spencer wrote:
I'm in Paris right now and can't test this change, but I've been
researching the DAA and there are a few international settings I can
change, so I've changed the driver in CVS so that you can specify
That's encouraging news.
the operational mode.
Thank you! Downloaded, removed my commented out area in the asterisk code,
re-built everything and no problems. Now, if my overnight delivery really is
overnight I'll have my quad FXS and single FXO card to start playing with
today and of course my Altigen phones to attempt to use with H.323
If
Try to explicitly add this line
,1,SetCallerid,(somename 12345)
,2,Dial,Zap/g1/${phonenumber}
regards
Martin
On Tue, 17 Jun 2003, Tom De Wispelaere wrote:
Hey all,
I have a E1 setup with a E400P digium card. Everything works just great
except for the callerid. When i make an outgoing call
Marcus Adolfsson wrote:
Message
Just a quick note to people looking for SIP firmware images for Cisco
phones:
To access these files from Cisco's website, you need to have a Service
Contract (SmartNet) on at least on of your phones. I though a contract
was several hundred dollars, but it is
Did you cvs update zaptel and recompiled ?
Martin
On Tue, 17 Jun 2003, K. C. Li wrote:
On Tue, 17 Jun 2003, Mark Spencer wrote:
I'm in Paris right now and can't test this change, but I've been
researching the DAA and there are a few international settings I can
change, so I've changed
I am using a grandstream phone with g729 and alaw odecs and in both modes I
cannot seem to pass dtmf's, neither inband nor out of band, neither wthrough
a lcoal server nor through a natted connection. Am I missing something ?
___
Asterisk-Users
On Tue, 17 Jun 2003, Martin Pycko wrote:
Did you cvs update zaptel and recompiled ?
Yes. I followed the instructions on the Digium download page, namely:
export CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot
cvs login [password anoncvs]
cvs checkout zaptel libpri asterisk
cvs logout
cd
On 17/06/2003 at 10:23 Rushowr wrote:
Is it possible to set up Asterisk without any of the cards? I'm
interested in setting it up for the company I work for, but I would like
to set it up and see how difficult it will be before I start having the
company spend a chunk on equipment.
Yes, you can
Rushowr wrote:
Is it possible to set up Asterisk without any of the cards?
Absolutely. Use IP Phones and a VoIP termination provider... Hell, you
may not even need your own Asterisk box if your just looking to see how
simple it is to make an IP phone work.
Jeremy McNamara
Klaus Darilion wrote:
Is it possible to set up a VoIP Gateway using a soft modem instead of an
X100P card?
No, you need the 1000hz interrupt provided by the X100P to properly sync
the audio streams in the conference.
Jeremy McNamara
___
Did you cvs update zaptel and recompiled ?
Yes. I followed the instructions on the Digium download page, namely:
I looked at the log file and there was no commit on this recently. It
seems that if this change has been made, it's just not in CVS yet :)
Looking forward to trying it though...
You mean you don't have access to a machine on the otherside of your
firewall where you could use CVS for yourself? What kind of linux person
doesn't have multiple machines around the net to use, or test from when
you believe something is wrong with your setup?
On Tue, 2003-06-17 at 11:17, Low,
On Tuesday 17 June 2003 08:49, Francisco Perez-Landaeta wrote:
does anyone know what compression uses Speex ? and which gateways
used it so they can work with Asterisk.
Speex is a codec, not a protocol. It is analogous to the functionality
of ulaw, gsm, and ilbc.
-Tilghman
I generate nightly tarballs of all the asterisk related sources. You can
get them from http://asterisk.gnuinter.net/files/cvsnightly
There are also nightly cvs changelog files at
http://asterisk.gnuinter.net/files/changelogs
James
On Tue, 17 Jun 2003, Low, Adam wrote:
Hi All,
Our FW
Oops I just now commited it, sorry for the delay!
Mark
On Tue, 17 Jun 2003, Mark Spencer wrote:
On Sun, 15 Jun 2003, John Laur wrote:
I do not think it is necessarily a hardware issue, as the line-in-use
lights do not light until the wcfxo kernel module is loaded. It would be
very
Hey Martin, thnx for your reply,
I've tried it out as you said by setting it explicitly in my
extensions.conf as follows :
exten = number,1,Wait,1
exten = number,2,Answer
exten = number,3,SetCallerID(somename 0)
exten = number,4,Dial,Zap/g2/${phonenumber}
in the console it says
--
I do have that in modules.conf
Anything else?
Il mar, 2003-06-17 alle 18:59, WipeOut . ha scritto:
Have I left something out or done something wrong??
Yes, in modules.conf:
[global]
chan_capi.so=yes
You need it in order to export its symbols to the applications!
Bye,
--
E.
Hello to all,
I want to put incoming calls in a queue and that user hear a beauty song
:-)
But, although I think that parameters in the file queues.conf are
corrected; is not possible to listen any melody ... and nevertheless,
the message of Started music on hold, class 'random', on
Hi,
I have been working through getting chan_capi installed..
I think the AVM capi driver is loading fine...
I think I compiled chan_capi correctly...
I have updated modules.conf as required...
When I run asterisk -vvvc I get the following at the end after which asterisk exits..
Il mar, 2003-06-17 alle 18:59, WipeOut . ha scritto:
Have I left something out or done something wrong??
Yes, in modules.conf:
[global]
chan_capi.so=yes
You need it in order to export its symbols to the applications!
Bye,
--
E.
___
Asterisk-Users
[...]
Hi,
No, you need the 1000hz interrupt provided by the X100P to properly sync
the audio streams in the conference.
Could you explain a bit more please ? :)
Best regards,
Piotr Adamiak
--
We demand 40 million helicopters, and a dollar! -- WAIT!
Just do the pri debug span 1 and see for yourself that asterisk sends
that. You might however send it without one digit or something ... or
maybe your telco doesn't support it. Just give then a call.
Martin
On Tue, 17 Jun 2003, Tom De Wispelaere wrote:
Hey Martin, thnx for your reply,
I've
Do you have '-z' option with the definition of random in musiconhold.conf
? actually I just did see the options of mpg123 and it has to be an
uppercase Z:
-Z
Martin
On Tue, 17 Jun 2003, Rafael Gonzalez Lomeña wrote:
Hello to all,
I want to put incoming calls in a queue and that user hear
It would be really nice if the DISA allowed one to make follow-on calls
by pressing some key sequence (say, press * three times in a row within
one second).
This especially helps when you are at a hotel that charges for each
call. We put our DISA on a toll-free number, and as long as it
supports
James, thanks I appreciate it.
-Original Message-
From: James Golovich
To: '[EMAIL PROTECTED]'
Sent: 17/06/03 18:41
Subject: Re: [Asterisk-Users] Firewall Silly - anyone can help with a CVS tar ball ?
* DISCLAIMER *
This message and any attachment are confidential and
We have it done at Digium so it can be done.
Just record your name I guess with voicemail but I'm not entirely sure
about that you can record that in voicemail.
Martin
On Tue, 17 Jun 2003, Derek Beaumont wrote:
I'm wondering if I can do the following:
Caller activates the Directory
Has this been solved? When I park a call, the caller hears a second of
music on hold
and then the whole system crashes.
I can restart with a simple (asterisk -cvvv), I don't have to reboot or
anything
John
___
Asterisk-Users mailing list
[EMAIL
When in voicemail they need to go into the record name section and
record their name. Then it will play their name.
-Original Message-
From: Derek Beaumont [mailto:[EMAIL PROTECTED]
Sent: Tuesday, June 17, 2003 3:33 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Directory
Where is this time that is sent to the phones with the callerid info coming
from ?
If I do date at the command line I get the correct time as set by ntp
yet the time the phones get set to is 50 minutes slow.
___
Asterisk-Users mailing list
[EMAIL
ok forget it
I was telling a story of what inspired me to write this module
that embeds a perl interpreter into the asterisk process and
to see if anyone was interested in it (IT'S AN EXPERIMENT)
What I get are several replies telling me to RTFM and I would no longer need to solve the
Can you provide some more information about the problem? How are you
parking the call (with #transfer, or with a hookflash on zaptel)?
There was a problem with app_agent, where a segfault would occur when
transfering but we fixed this late last week.
If you cvs update and the problem still
Describe that a little bit.
The call came on what interface and then you dialed what interface
and how did you park it ? You pressed a flash button or '#' key ?
Martin
On Tue, 17 Jun 2003, John Congdon wrote:
Has this been solved? When I park a call, the caller hears a second of
music on
Maybe the hardware date?
-d
At 05:07 PM 6/17/2003 -0400, you wrote:
Where is this time that is sent to the phones with the callerid info
coming from ?
If I do date at the command line I get the correct time as set by ntp
yet the time the phones get set to is 50 minutes slow.
Yo Iain,
On Tue, Jun 17, 2003 at 21:48:34 +0100, Iain McWilliams wrote:
Hi,
I'm trying to get asterisk running on kernel 2.4.20 however trawling through
the archives I've found a few references to patches to remove i4l's dtmf
detection, but have been unable to find the patch itself (I
Title: RE: [Asterisk-Users] Dual T400P, SMP, performance issues
I believe this is related to the load, there are always calls in our test.
Attached is a part of /var/log/messages file with SysRq memory info - in
case you can see something in it. The box was rebooted 06-16 17:08 and
the
Hello,
I've commited the new busydetect routine to CVS.
You need to cvs update asterisk of course and then choose it
in asterisk/Makefile and recompile asterisk.
All you X100P users that had the problems
with false hangups or the card not being able to detect the busy tone
please check that.
In
Anthony,
Don't take the comments about your work too seriously.
We like to do things the easiest way, and its their way of helping you g.
I liked your work. Could you show us a database example of your work.
Peter
At 14:19 17/06/2003 -0700, you wrote:
ok forget it I was telling a
Quite frequently, outgoing calls from the X100P cards here will not dial
properly. Instead of hearing the ringing after the Zap interface picks
up, I'll hear silence for a while then the 'If you'd like to make a call
please hang up and try again' recording as if zaptel picked up the line,
Do your phones have NTP clients built into them, like ATA-186 or
Cisco 79xx systems? If so, you'll need to set the phone to pull the
correct date from a valid NTP server.
JT
Maybe the hardware date?
-d
At 05:07 PM 6/17/2003 -0400, you wrote:
Where is this time that is sent to the phones
Yo Martin,
On Tue, Jun 17, 2003 at 17:03:15 -0500, Martin Pycko wrote:
Hello,
I've commited the new busydetect routine to CVS.
You need to cvs update asterisk of course and then choose it
in asterisk/Makefile and recompile asterisk.
[...]
It fails to compile here (Redhat 9, gcc version
Did you try to use 'w' as a digit before dialing the number like this:
exten = _X.,1,Dial,Zap/1/w${NUMBER}
You could also try to put 'w' inbetween the digits.
regards
Martin
On Tue, 17 Jun 2003, John Laur wrote:
Quite frequently, outgoing calls from the X100P cards here will not dial
You can do a pri debug span 1 and confirm that we're sending using the
same parameters as when they send to us. Might be one more step to do
before calling the telco.
Mark
On Tue, 17 Jun 2003, Tom De Wispelaere wrote:
Hey Martin, thnx for your reply,
I've tried it out as you said by setting
This doesn't work for me. Voicemail says the extension number but
does not play the user's name. (Asterisk CVS-04/30/03-22:57:49)
On Tue, 17 Jun 2003, Benjamin Miller wrote:
When in voicemail they need to go into the record name section and
record their name. Then it will play their name.
Hi,
I'm experimenting with the dev kit lite and now past the USB
unpleasantness it's working great with standard phones and
lines.
The priority right now is getting soft phones (under Windows
XP) working well.
So far, I've only been able to get the XTEN Lite phone working
and I really don't
tried the ilbc and speex yet ??
On Wed, 18 Jun 2003 00:15:38 -0400, Reed Wade wrote:
I've got the XTEN Lite soft phone mostly working with * but it's
dropping out like a very bad cell phone call.
The GSM codec is worst (unusable), G711u and G711a are best but
not good enough to use.
I
I was about to say 'yes' and they were worse but now I can't
remember for certain. I'll try those again to make sure.
Are they likely to be better?
-reed
At 02:27 PM 6/18/2003 +1000, Gary wrote:
tried the ilbc and speex yet ??
On Wed, 18 Jun 2003 00:15:38 -0400, Reed Wade wrote:
I've
60 matches
Mail list logo
