On Thu, Jan 01, 2004 at 09:48:36PM -0700, Steve Murphy wrote:
Hello--
[snip]
Trouble is, asterisk only sees the brain-dead interface. How do I
exorcise it from the kernel, or at least make the SB the first-priority
one? rmmod didn't seem to do anything. Playing with the Redhat sound
On Thursday 01 January 2004 12:57, Darren Nickerson wrote:
That worked a treat - thanks! Comedian Mail is now able to download
to the handset and there's a lot more functionality now.
There's a patch on the bugtracker that should allow you to specify these
codes per user, as requested.
Hi,
I've sent this to asterisk-dev recently, but seen no comments. Has anyone
else experienced this behaviour ? I've made a complete clean checkout of CVS
code, and it still happens
-Original Message-
I've just made a new update from cvs on my devel box to play
with, and I
On Thu, Jan 01, 2004 at 12:51:09PM -0700, Ken Godee wrote:
Darren Nickerson wrote:
That worked a treat - thanks! Comedian Mail is now able to download to the
handset and there's a lot more functionality now.
-d
I'd be interested in knowing if once you try to use Comedian mail
softkeys if
On 01/01/04 10:19, Olle E. Johansson wrote:
What I am looking for is a solution like this:
* Call comes in
* XXX on Line YYY answers
* A URL to a web page is transmitten on some channel, preferably the
VoIP channel
* The web page opens in a web window´
You're best off writing a separate
On Fri, 2004-01-02 at 08:31, Florian Overkamp wrote:
Hi,
I've sent this to asterisk-dev recently, but seen no comments. Has anyone
else experienced this behaviour ? I've made a complete clean checkout of CVS
code, and it still happens
-Original Message-
I've just made a new
Hello,
I need a way to record every call made to asterisk on a file.
The app_record application works but it is blocking, so I can't connect
a phone-operator and an user while recording.
I thought to use the MeetMe application and using a fake user to record
the call but in this way I can't know
Sounds like something nasty being printed. If you run asterisk in the
background (without -vvvgc) and don't attach to it do you hear it still?
Mark
On Fri, 2 Jan 2004, Patrick wrote:
On Fri, 2004-01-02 at 08:31, Florian Overkamp wrote:
Hi,
I've sent this to asterisk-dev recently, but
You must use Monitor Application
Happy New Year,
srsergio
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Edoardo
Borghesi [fabbricadigitale]
Enviado el: viernes, 02 de enero de 2004 12:33
Para: [EMAIL PROTECTED]
Asunto: [Asterisk-Users] Call recording
I have some strange question bout the asterisk (gpl license ...) but i'm not an
experienced linux user ...
What happens if for example a big company buys digium , do we have a garantuee that
asterisk stays opensource ???
Kind regards
Michael Devenijn
winmail.dat
xlite saying login timed out. contact network admin.
how to get rid of this. * is not behind NAT.
DIAX works fine
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On Fri, Jan 02, 2004 at 12:23:52PM +0100, Michael Devenijn wrote:
I have some strange question bout the asterisk (gpl license ...) but i'm not an
experienced linux user ...
What happens if for example a big company buys digium , do we have a garantuee that
asterisk stays opensource ???
xlite saying login timed out. contact network admin.
how to get rid of this. * is not behind NAT.
also, the grandstream SIP phone also seems to fail to register. IAX phones
are all ok.
DIAX works fine
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[EMAIL
What about you drop your beer, stand up from your couch (if your fat
belly allows you to), turn off the damn TV and try to learn some basic
C programming. Then maybe you can help us in solving those frequent
segmentation faults (if any).
-Original Message-
From: [EMAIL PROTECTED]
Hi there,
is this a problem with the Wiki software or the DB? The delay is still
tolerable, but not exactly nice to work with.
http://www.voip-info.org/tiki-index.php?page=Asterisk+billing
Page generated in: 2.35 seconds
Philipp
___
Asterisk-Users
On Fri, 2004-01-02 at 07:59, Ranga wrote:
unsubscribe
This is a function you do your self. You should see a URL in the footer
of this message that will show you how to do it. Had you been reading
copies of messages that where not in HTML, you would have seen this
message before.
--
Steven
Hi all,
I am trying to write a program that sends SIP requests to asterisk. My aim
is to make asterisk record as many voicemails it can at a time. The design
of the program is like this:
There are two processes: One main process and a child process (No flames
pls. I have very little idea about
On 02/01/04 14:24, Girish Gopinath wrote:
I gave the sip debug command, and one of the lines showed:Ignoring this
request
Can you log the SIP debug messages to a file and put it up on the web
somewhere? Or do an ethereal capture or similar. It's very hard to say
what the problem might be
On Fri, 2004-01-02 at 08:51, Ranga wrote:
Sorry...I missed it. I wanted to change my email id. So unsubscribed and
subscribed again.
Well then I guess it was a good thing I kept some composure instead of
flaming away as is the usual for those kinds of messages.
Welcome back.
- Original
This is bizzare the following was removed from this post
--
And the answers are standards, professionalism, comunications and
documentation
Note the section on Professionalism
On Fri, 2004-01-02 at 09:27, Asterisk List wrote:
Hello:
It has happened while I was making 1000 outgoing calls, at a sustained rate
of 2 calls per second.
Asterisk makes a SIP call to a CISCO router and this router is connected to
the PSTN line.
While putting files in the outgoig
Hi-
In trying to track down a possible memory leak in asterisk, I've discovered
that the show memory allocations command crashes asterisk (causes it to
stop handling calls, although it doesn't seg fault). The related show
memory summary works however.
Before I post this to the bugs list, can
Sergio Serrano Revuelto wrote:
You must use Monitor Application
Happy New Year,
srsergio
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Edoardo
Borghesi [fabbricadigitale]
Enviado el: viernes, 02 de enero de 2004 12:33
Para: [EMAIL PROTECTED]
Asunto:
Philipp von Klitzing wrote:
Hi there,
is this a problem with the Wiki software or the DB? The delay is still
tolerable, but not exactly nice to work with.
http://www.voip-info.org/tiki-index.php?page=Asterisk+billing
Page generated in: 2.35 seconds
The same page, 1.75 seconds for me.
The
Since i'm sure there are others out there in the same position as me, being disappointed that the original T400P and E400P cards are no longer available from Digium, I thought I would pass on a place I found to get them. I needed the older card, since I didn't want to have to get a motherboard
Michael,
I just got mine. Do you recall how you managed to priortize RTP? Or do
you rely on the 'priortized switching port' feature? I tried that, but
perhaps my TOS value does not match the one this router expects. Even
sending a single, large email can kill the voice stream. Leave alone
I have a similar problem, with GS phones, X-Lite or Kphone. I tried all
the codecs with the same result. Choppy sound in the direction SIP-Phone
- pstn, but crystal clear sound the other way around. The only
difference in my case is that I have two asterisks servers connected
together via IAX2,
Well, Eric and James have answered already. Personally, I use redhat
(will upgrade to fedora soon), but using an unmodified kernel.org kernel
compiled from source. Best regards,
On Thu, 2004-01-01 at 15:25, JR Richardson wrote:
Hey Nicolas,
That did it. I ran that export command you
On Fri, 2004-01-02 at 11:28, John Ternovas wrote:
Since i'm sure there are others out there in the same position as me,
being disappointed that the original T400P and E400P cards are no
longer available from Digium, I thought I would pass on a place I
found to get them. I needed the older
Title: Message
I
understand that there also is a new board from Digium, the TE405P, which is like
the 3.3vTE410P, but uses a 5-volt supply.
Just
so you have another choice!
regards,
Scott
Scott M.
Stingel Emerging Voice Technology Inc.Palo Alto, California and London, England
Email:
On Fri, 2004-01-02 at 11:35, Nicolas Gudino wrote:
I have a similar problem, with GS phones, X-Lite or Kphone. I tried all
the codecs with the same result. Choppy sound in the direction SIP-Phone
- pstn, but crystal clear sound the other way around. The only
difference in my case is that I
So I made the mistake of buying a Carrier Access channel bank without
noticing the page on the wiki about the fact that they don't support
disconnect supervision (bastards!). However, apart from that, I do have
it working fine for incoming calls.
Is there some trick to get asterisk to detect
Does anyone have recommendations for (or against) mini-ITX platforms to
be used with Wildcard X100P and TDM400P cards?
I am considering the use of systems using VIA EPIA CL and Epia M as
small, quiet platforms on which to host Asterisk.
___
On Fri, 2004-01-02 at 12:25, Sean Adams wrote:
So I made the mistake of buying a Carrier Access channel bank without
noticing the page on the wiki about the fact that they don't support
disconnect supervision (bastards!). However, apart from that, I do have
it working fine for incoming
Hi Steven,
On Fri, 2004-01-02 at 14:55, Steven Critchfield wrote:
What is the ping times between your 2 asterisk servers? In the archive I
have documented before that IAX jitter buffer sometimes has problems on
short ping time links. At the time we where on a private T1 with 4ms
ping times.
busydetect should help you. Set busycount=10 busydetect=yes in zapata.conf
and measure the length of the tone .. should be equal the pause too.
Then in dsp.c change the vaules BUSY_MIN and BUSY_MAX for example like
this: your result - 100, your result + 100 [ms]
regards
Martin
On Fri, 2 Jan
On Fri, 2004-01-02 at 12:34, Gary Gapinski wrote:
Does anyone have recommendations for (or against) mini-ITX platforms to
be used with Wildcard X100P and TDM400P cards?
I am considering the use of systems using VIA EPIA CL and Epia M as
small, quiet platforms on which to host Asterisk.
It
From: Scott Stingel [EMAIL PROTECTED]
I understand that there also is a new board from Digium,
the TE405P, which is like the 3.3v TE410P, but uses a 5-volt supply.
Any ideas if this is actually shipping yet though? If not when?
Linus
___
On Fri, 2004-01-02 at 12:46, Nicolas Gudino wrote:
Hi Steven,
On Fri, 2004-01-02 at 14:55, Steven Critchfield wrote:
What is the ping times between your 2 asterisk servers? In the archive I
have documented before that IAX jitter buffer sometimes has problems on
short ping time links. At
Dear Forum,
I'm using the AgentCallbackLogin function to log my
agents onto multiple call queues.
exten =
3001,1, AgenCallbackLogin(1001,@sip). This works very well.
I can not work out how to log them back out? On of the forum
members was kind enough to point me into the
Hi!
I'm using the AgentCallbackLogin function to log my agents onto multiple
call queues.
exten = 3001,1, AgenCallbackLogin(1001,@sip). This works very well.
I can not work out how to log them back out? On of the forum members was
kind enough to point me into the directions of 'dial a
Certainly a bit of Googling can lead to partial free documentation (I'd
recomment the Newbridge section of
http://www.at2.com/pages/DocumentLibrary.cfm) and I suspect the
information
overlaps between models particularly with respect to the console commands,
but alas the PDFs that are
Here's a little tidbit about the non-functional flash key on the
Budgetone 100's. I have 20 of these phones. On some, the flash key
works, and on some it does not. Since the problem is utterly independent
of the firmware revision, I suspected that it was hardware based. So,
in the interest
I don't know how I managed to mess up sending this last time, but
somehow it got attached to the AgentCallbackLogin thread. Since the
indended audience may not see it there, please indulge me by tolerating
this second copy:
Here's a little tidbit about the non-functional flash key on the
On Friday 02 January 2004 13:46, Nicolas Gudino wrote:
Hi Steven,
On Fri, 2004-01-02 at 14:55, Steven Critchfield wrote:
What is the ping times between your 2 asterisk servers? In the archive I
have documented before that IAX jitter buffer sometimes has problems on
short ping time links.
Okay, I'm an idiot. The tones are picked up just fine by asterisk with
no changes.
It helps if you understand the syntax of zapata.conf. I thought
busydetect=yes just had to be under the context line. I didn't realize
how the channels= is actually the delimiter that includes the stuff
above
John,
Try these files.
They work for me.
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of John Coll
Sent: 02 January 2004 23:42
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Newbie - getting two local phones to
communicate would be a good start
This is hard work :) I have read the Asterisk Handbook, BudgeTone User
Manual, Andy Powell's useful notes, Zac Sprackett's Asterisk Resource Pages
and more.
I am not a linux newbie but am new to Asterisk. I have failed to find any
docs that explain how to get a very very simple, minimal,
Hi all,
I have posted a copy of the 3624 manual on the web. It's 11MB and over
650 pages, so not exactly light reading! You can grab it at
http://www.caeveo.com/files/newbridge3624.pdf. Please be kind and save
it to your local machine instead of reading it from the web! Thanks!
Sean
On Fri, 2 Jan 2004, Steven Critchfield wrote:
On Fri, 2004-01-02 at 12:34, Gary Gapinski wrote:
Does anyone have recommendations for (or against) mini-ITX platforms to
be used with Wildcard X100P and TDM400P cards?
I am considering the use of systems using VIA EPIA CL and Epia M as
My sip.conf
[general]
port = 5060 ; Port to bind to
bindaddr = 0.0.0.0
disallow=all; Disallow all codecs
allow=ulaw ; Allow codecs in order of preference
dtmfmode=rfc2833
[xlite1]
type=user
host=dynamic
secret=xlite1
context=outgoing
Josh,
This is what I've understood it to be so far...
The phone(s) are available in two flavours:
79xx with Call Manager Single User License
79xx without Call Manager Single User License
These optional licenses (which can also be purchase separately, and are
approx £10/$15) are to upgrade
Hi John,
If your effort is to make calls between two GS phones via *, here is what
you need.
You need all three devices in the same LAN, so set both phones and * to
10.0.1.98/24.
After that from your asterisk Linux box ping both phones. If that is
successful you know your layer 1,2 and 3 are ok.
Just to second this post: I had the same symptoms and resolved them by
tweaking my firewall.
Hope this helps,
David Gomillion
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of SW
Sent: Friday, January 02, 2004 11:07 PM
To: John Coll; [EMAIL PROTECTED]
- Original Message -
From: Chandra [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, January 03, 2004 1:34 AM
Subject: Re: [Asterisk-Users] Call recording/SIP not loggin IN
My sip.conf
[general]
port = 5060 ; Port to bind to
bindaddr = 0.0.0.0
disallow=all
hi,
i can't seem to register my grandstream SIP to * server...
i have my grandstream IP as 192.168.0.11 want to register to * at
202.51.xx.xxx.
sip show peers says that my grand stream has unspecified IP but when i try
to dial a number it gets this error...
WARNING[5126]: File chan_sip.c, Line
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