How can i found out what the name is for a SIP channel if i know the username, context, extension?
i want to be able to force a call to drop using the manager but need to obtain the Channel name in order to force the drop.
Thanks
Dan
___
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In the manager, in order to force a call to end by using the Hangup command, i need to know the channel name.
how do i find out the channel name if i know the extension/username?
Thanks
Dan
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--Bandwidth and Colocation sponsored by Easynews.com
difficult.
Dan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wayne
Sent: Sunday, October 02, 2005 2:22 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Asterisk on windows
Hiyall,
been following this for a while
on a Unix like OS. About the M$ part,
well, it's silly decisions like that that contribute to 3Com's fading
away.
Regards,
Patrick
Dan
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media streams. So the software does not need
to deal with high priority real-time traffic
Regards,
Steve
Dan
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Dan - as a thought - I am messing with a H323 'capable' IP Phone and I
am (maybe foolishly) trying to use ooh323 with no gateway, gatekeeper,
or anything else and I am not getting it to work too well. It seems
'sometimes' it does work.
I'm using it to connect to Cisco Call Manager. I set
Does anyone know where i can find out how powerful a machine has to be to handle a certain amount of call volume?
Eg, 2Ghz is enough processing powerto maintain100 calls at a time.
4Ghz is engouh to process 250 calls etc etc.
Thanks
Dan
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little bugs that related to our use of Cisco Call
Manager, and expected little or no interest in getting them resolved.
I had a test version made available to me in just over a day and
complete resolution a few hours later.
Dan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
Is there a guy somewhere on how much bandwidth each codec uses, along with the advantages and disadvantages of each one?
Dan Journo
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I dont believe its limiting but if you want to block users in real time when their credit runs out, you need to use the realtime config. Thats what i assume anyway.
Dan
On 9/21/05, Steven [EMAIL PROTECTED] wrote:
I found configuration via MySQL too limiting.I went back to text files.I do not know
What is this sip-nat-helper thing, is there a website were we can get
some info on it, partly thinking as the question before was relating to
open source software, I would assume that I could download this thing.
Dan
On Wed, 14 Sep 2005 [EMAIL PROTECTED] wrote:
If you have a linux box
On 9/20/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Ok.
I was sucessful in installing ODBC storage
I'm using MySQL in the backend as it is. but all my drivers are now ODBC.
I am running asterisk-cvs head as of last night 9/19/05
My question is this... the old voicemail.cgi script
Is there a guide anywhere which runs through how to set up asterisk with mysql?
I've looked and almost all the document misses out relevant information.
Thanks
Dan Journo
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Ive already set up the cdr mysql.
Now im trying to add realtime now but stuck on how to do it. those links didnt really help much. and the cli doesnt provide much info on what is going on.
any help would be appreciated.
Thanks
Dan
On 9/20/05, Nathan Pralle [EMAIL PROTECTED] wrote:
Dan Journo
in Perl or PHP are not
perfect, but If you have to setup a socket and have the overhead, what
are the advantages?
Dan
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Hi Jason,
Is there a specific version of DIAX that I should use? I grabbed the
latest
release...Looking at the DIAX site, 910g has the URL feature fixed.
Is it
broken again in 915a?
URL feature works in 0.9.15a.
Take care that it is implemented JUST for the Dial command.
Best regards,
Dan
it makes more sense to track
the participant by caller-id. I have a patch for 1.0.X on my
site, but have not polished one for CVS-Head or the 1.2.0beta
release.
Thanks and enjoy,
Dan
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the feedback I have received since the last announcement, and
apologize that I allowed work to get in the way of development.
Thanks and enjoy,
Dan
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Asterisk
tested the new channel with
Cisco Call Manager?
Thanks,
Dan
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. If you would like it to work for just
contacts it will need to be modified.
Regards;
Dan
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useful.
http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current/docs-html/x1695.html
Dan
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channels used up)
exten = s-BUSY,1,NoOp(Trunk is reporting BUSY)
exten = s-BUSY,2,Busy()
exten = s-BUSY,3,Wait(60)
exten = s-BUSY,4,NoOp()
exten = _s-.,1,NoOp(Dial failed due to ${DIALSTATUS})
Any ideas?
Thanks,
Dan
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?
TYIA
Dan
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://www.newegg.com/OldVersion/app/ViewProductDesc.asp?description=25-180-004DEPA=0
I'll sell it to you for $6 plus $3 shipping.
Dan
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On 8/9/05, Dan Littlejohn [EMAIL PROTECTED] wrote:
On 8/9/05, Douglas Logan [EMAIL PROTECTED] wrote:
Now that the X100P is no longer being offered by Digium, what is the
best solution? I seem to have run into a few posts where people talk
about problems they've had with their X100P clone
Interface).
Download it here:
http://www.littlejohnconsulting.com/?q=ari
Place it in /var/www/html/recordings. AMP is including it in their
distribution and I will make updates there and on my website.
Regards;
Dan Littlejohn
(512) 791-0137
www.littlejohnconsulting.com
option ntp-servers x.x.x.x;
Works great. Polycom 501.
- Dan
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When you say local asterisk box and a remote asterisk box, are you
referring to having NAT in between the two boxes in any way?
I am just curious if you are and how it may work.
Dan
On Mon, 25 Jul 2005, Duracom ISP Lists wrote:
We are setting up a test bed and I am curious to know what type
Just a guess on 31, but have you made sure you have a different IP Number
on each of the two phones. With some of my own observations in watching
them boot up, I have remembered that even if it is assigned a static IP,
it still has a phase that says Obtaining IP
Just a thought.
Dan
On Mon
DISA
- Dan
Min Hwan Chang wrote:
Hm, I'm wondering if its possible for someone to call in the POTS
line, dial an extension, then be able to dial a number of their
choosing out the IAX line?
So let's say I'm here in california and I dial into the office. Dial
which gets me a message
-exten-vm, s, 7) exited non-zero on
'SIP/440-35ab' in macro 'exten-vm'
== Spawn extension (from-internal, 440, 1) exited non-zero on
'SIP/440-35ab'
-- Executing Macro(SIP/440-35ab, hangupcall) in new stack
Thanks,
Dan
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Dan
On Thu, 21 Jul 2005, Ivan Fetch wrote:
Hello,
On Wed, 20 Jul 2005, Wiley Siler wrote:
For the fella who wanted MOH music
Royalty free stuff can be found here.. The Acoustic Guitar is a nice
collection...
http://www.freeplaymusic.com/
Cheers,
W
I spoke with Scott
running.
Dan
On Sat, 16 Jul 2005, Ted Serreyn wrote:
This is not a problem. I do this and a bit more. The IAX protocol helps
quite a bit to go thru the NAT.
--
Ted Serreyn Phone:262-432-0260 Fax:262-432-0232
Serreyn Network Services, LLChttp://www.serreyn.com
to be enlightened.
- Dan
Eric Rees wrote:
Could you pass along the information you used to get the Polycom lights
to work.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tom Hayden
Sent: Wednesday, July 20, 2005 11:57 AM
To: Asterisk Users Mailing List
out.
If you still have problems, what do you see on the Asterisk CLI at a
decently high verbosity, when you set Watch Buddy to enabled?
- Dan
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I asked them a while ago (month or so?) about International rates.
They responded that they got burned on International call fraud, and
only allow International termination under special circumstances (or
something to that effect).
- Dan
Jay Milk wrote:
That's odd -- they used to be here
Jerry Geis wrote:
Ken,
Point to a different proxy. I had the same issue with chicago...
Same here with DCA. Now using MIA.
- Dan
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simultaneously. 80*4 = 320. You'd be using 320kbps down and
320kbps up, which is within your 1500kbps down / 384 kbps up.
Someone please correct me if I'm wrong.
- Dan
Tim Pushor wrote:
Of course - ISDN is bi-directional. I guess saying that ULAW takes
130K+ bandwidth depending on the framing
to use mixed case username... I'm picky).
I got my config files from what was posted on the wiki.
At what point are you having problems?
- Dan
chris gamble wrote:
I just received my first polycom 501, tried my best to follow all of the
documentation and configs on the wiki ( though some
.
Yeah... Those are the ones I used (Thanks, by the way!).
- Dan
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website or other places that might be
of use in this idea? Or does anyone have suggestions or knowledge of this
working for them?
--
Dan Adams - [EMAIL PROTECTED]
http://www.infochi.net
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website or other places that might be
of use in this idea? Or does anyone have suggestions or knowledge of this
working for them?
--
Dan Adams - [EMAIL PROTECTED]
http://www.infochi.net
pgpJn1WRUdWig.pgp
Description: PGP signature
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the speaker phone, but don't need the XHTML microbrowser and/or the
extra lines, go with the 501.
- Dan
Chris Gamble wrote:
From their website, the key difference between the polycom 500 and 600 phones
is the number of lines they support. What does this mean in terms of
asterisk? Do I have to have
be recommending them.
My USD0.02.
- Dan
List Receiver wrote:
According to voipsupply.com
http://www.voipsupply.com/product_info.php?cPath=95_112products_id=817
--Please Note: Polycom phones are not supported under Asterisk Open Source
PBX. Polycom certified platform partners include Path
Brian Roy wrote:
On 7/9/05, Dan Perik [EMAIL PROTECTED] wrote:
PJ,
You should check out the Polycom 500/501/600. I'm quite sure it has all
that (although I don't use all of what you listed).
IIRC, the 500's browser is crippled. I think you have to go up to the
600
PJ,
You should check out the Polycom 500/501/600. I'm quite sure it has all
that (although I don't use all of what you listed).
- Dan
Pavel Jezek wrote:
Still looking for cheaper (under $250,-) alternative to cisco 7940
with features needed for corporate use, mainly:
- shared phone book
VoIP phone.
- Dan
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Hi, I am sorta a newbie to the asterisk community at least in the realm of
hardware types. I was wondering, what type of card is used to allow asterisk,
on a slackware installation to talk to a standard phone line so that asterisk
can call out?
Dan
Jeffrey Starin wrote:
911 Help!
I accidentially deleted all directories under /var/spool/asterisk
I did use the backup facility not too long ago but cannot find the
process for restore.
However, I don't believe a full restore is needed -- I just need to know
the names of the
please send me a mail directly.
Thank you in advance for your support in DIAX development.
Best regards,
Dan
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explaining it.
- Dan
Andrew Latham wrote:
I think the $10 is setup, as you will notice all the others mention
the monthly next to the rate.
I was confused also. (Hint Teliax)
On 6/27/05, John Goerzen [EMAIL PROTECTED] wrote:
I'm looking for someone that sells minutes in bulk like LiveVoip used
I'm not sure which is funnier... that someone would offer something like
that for sale on ebay, or that someone would pay $10.56 + $4.50 shipping
to buy it.
rofl
- Dan
Steven Kalcevich wrote:
I for one will not be using anymore live voip...I found my own provider.
http://cgi.ebay.com/ws
No one has any idea? Even a NO it cant
be done would be appreciated.
Thanks in advance.
Dan
From:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
On Behalf Of Dan Morin
Sent: Monday, June 20, 2005 7:24
PM
To: Asterisk
Users Mailing List - Non-Commercial Discussion
Subject
between the two running uLaw. It seems to work fairly well.
Dan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andy Rosen
Sent: Sunday, June 26, 2005 2:22 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users
conference. I noticed 10 second delays after 5 minutes
of conference.
Let me know if you find anything else out about this.
Dan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tony
Mountifield
Sent: Sunday, June 26, 2005 5:35 PM
To: asterisk-users
Not always. Some use a www capture page. When you log in through that
page, it opens up that mac/ip for a specified length of time. We're
doing that here using nocat (http://nocat.net) Without logging in, no
traffic goes through from that mac/ip.
- Dan
Denis Galvão - iSolve wrote:
Hi Steve
Title: Normal
If someone has a minute, I would appreciate their help
configuring my dialplan. I am using 2 Sipura-2000s to connect to the CO
ports on my legacy PBX. Im tyring to setup the dialplan so that
when someone enters an extension (1XX), it will determine which of the 4 sip
and date as
number).
The help file is updated to.
As usual, please send me your feedback.
For any questions or features requests do not hesitate to send me a
mail.
Best regards,
Dan
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Young asked which is why this is so
difficult.
Can anyone confirm that dialing 8 + the Trunk Group number will select a
CO line in that trunk? Thanks in advance.
Dan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Peter
Svensson
Sent: Monday, June 20
supervision, so this tone is the only
thing that will detect a disconnect. It is not a standard fast busy or offhook
tone.
Please see this post for more info: http://www.voip-info.org/tiki-index.php?page=Asterisk-Panasonic1232vm
Any help would be greatly appreciated. Thanks in
advance.
Dan
In the queues.conf file, under your queue
you can add the following:
member=sip/ExtensionNumber
where ExtensionNumber is the extension.
Then they should always be part of the queue.
Hope this helps.
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Asterisk only runs on 5060/udp.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Moises
Silva
Sent: Monday, June 20, 2005 7:38 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] service scan
i want to make script in
all
I cannot get ChanIsAvail to work with sip or iax on
v1.0.3. It does work fine on a zap channel. I am trying with Sipuras and
PAP2s.
It appears I am not the only one having this
problem. Has anyone gotten it to work?
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If anyone has any experience with a Panasonic KX-TD1232
phone system, I would really like to talk to you for a few minutes.
I have asterisk connected to a Panasonic system via FXS
- CO ports. Im trying to get the Panasonic configured so that
if someone dials a number (9) while Intercom
password).
Thanks;
Dan
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Bad form to post to your own mailing, but I found the flash panel docs
(http://www.asternic.org/) Oh well (I was blind or something)
If someone could point me to the incoming/outfoing call recording
feature for AMP it would be greatly appreciated.
Dan
On 6/16/05, Dan Littlejohn [EMAIL
In /etc/asterisk/modules.conf
noload = chan_alsa.so
noload = chan_oss.so
- Dan
Conrad Beckert wrote:
Hi
... probably one of those RTFM kind of questions (while I'd be happy to know
where a good reference FM is :-) )
Has anyone an idea on how to disable the console sound driver. My problem
be defective.
Would not recommend it because of the low sound volume problem.
Talking on the phone is actually the point of the device so who cares
how configurable it is if you cannot hear anything. I purchased a
Digium TDM400P and have had very good luck with it.
Dan
On 6/15/05, Rich Adamson
Sipura support is nonexistant. Just our experience.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Dan
Littlejohn
Sent: Wednesday, June 15, 2005 9:55 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users
Hi
Everyone,
Is it possible to
have a SIP Phone work remotely if it's behind a Router performing NAT without
connecting the Router to a VPN? The Asterisk Box will be in the
DMZ.
Thanks
Dan
CYTEXONE
Dan Levine
[EMAIL PROTECTED]
CYTEXONE | Your Technology Specialists
877
I would be willing to Pay $500 for a good Asterisk / Exchange
Integration
Dan Levine
[EMAIL PROTECTED]
CYTEXONE | Your Technology Specialists R
877.CYTEXONE x 810
212.477.0990 x 810
212.208.6889 FAX
502 Laguardia Place, Suite 2B
New York, NY 10012
http://www.cytexone.com
Hi
Everyone,
Is it possible to
have a SIP Phone work remotely if it's behind a Router performing NAT without
connecting the Router to a VPN? The Asterisk Box will be in the
DMZ.
Thanks
Dan
CYTEXONE
Dan Levine
[EMAIL PROTECTED]
CYTEXONE | Your Technology Specialists
877
i am a newbie, but have you tried
genzaptelconf -s -d
Dan
On 6/8/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
All,
I have an [EMAIL PROTECTED] installation with a TDM40B card. I can make
internal
IP calls with no problems, but when I try to dial out I get a message that
All
I got these errors and my hardware is working so I do not think they
are an issue
Hint: insmod errors
Removing zaptel module: zaptel: Device or resource busy
What about the ztdummy module?
Dan
On 6/8/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Dean,
Here are the results
Just got the same thing here.
- Dan
Chris Coulthurst wrote:
Anyone else unable to get to www.voip-info.org? Site is returning
'connection refused' here.
Chris Coulthurst
[EMAIL PROTECTED]
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%26tid%3D106+dvorak+site:slashdot.orghl=en
I think you can safely discard this one too.
Dan
On 6/6/05, Colin Anderson [EMAIL PROTECTED] wrote:
http://www.pcmag.com/article2/0,1759,1812887,00.asp
Specifically, his assertion that ISP's would sniff traffic and block, say,
the SIP port. You
I'd be interesting in the same thing. Are they posted anywhere on the
web, or anything (I've looked, but not found).
Thanks,
- Dan
Luis Diaz wrote:
hi all, can anyone emailme the .conf of asterisk at home, i cant
download the full size tar or iso because of a network problem that
fu*** every
Folks!
I discovered some serious problem with several
Sipuras 3000 but I don't know if the problem is with them or Asterisk.
Basically, if I call a Sipura PSTN line, when there is a call already in
progress, generally I get a 503 Sevice Unavailable, but if I try hard enough, I
am able to
. For example if the dst
field was 5551212 the lastapp would be SIP/905/5551211.
Any help would be greatly appreciated.
Dan
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/zaptel start
Then do your modprobe.
Let us know what happens.
- Dan
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I haven't been having any problems. Since it's for home use (and my
daughters aren't teenagers yet :-) ), I don't have a lot of traffic, but
I've had good success.
- Dan
Sean Kennedy wrote:
Hi all,
Can any broadvoice customers give me their opinions on the service
recently? It's actually
to offload all the
non-client stuff, but until then, it all goes on this one machine. Yes,
this is a home setup, but with ties to work functions.
- Dan
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experience.
Dan
On 5/26/05, Brian Roy [EMAIL PROTECTED] wrote:
On 5/26/05, Andres Paglayan [EMAIL PROTECTED] wrote:
I am about to start building my first ever * production server and would
be nice to have some input from the list.
My personal vote would be for the Sipura's.
Pro's
While not widely used in the Cisco product line, SCCP can be used
for more than handsets. Newer VoIP gateways support SCCP trunking.
SIP in the 4.X series of CCM is a nice addition, but it is rather
limited at the moment. G7.11 only, requires a MTP for DTMF, hold
and transfer, etc.
Dan
. Not sure if current
browsers like it or not. I've never tried it, but came across this
document, and thought it may be something useful.
- Dan
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not been able to
find anything to troubleshoot the problem. I have seen adjustments to
gain etc, but not for the spa3000. In the US (Texas).
Any help would be appreciated. Thanks;
Dan
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.
If anyone has any ideas, I would really appreciate it. Thanks in
advance.
Dan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wilson
Pickett
Sent: Wednesday, May 25, 2005 1:06 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re
.
I have one runing for more than one month continously for Home
Automation purpose and not a single crash in between.
If it is realy a bug, then it would be solved ASAP.
Thank you and best regards,
Dan
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. In the budgetone, I can not. Im running version
1.0.6.2 firmware in the budgetone.
Please let me know if you have any
suggestions. Thanks in advance.
Dan
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it wrong.
Dan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Flash Love
Sent: Sunday, May 22, 2005 2:45 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Using patch -p0
meetme-diff-cbmysql_1.txtproduces 'malformed patch' message
I have
.
Dan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark
Johnson
Sent: Friday, May 20, 2005 8:09 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Stange question...
Ok, guys... Please be gentle with me. I have what is going
I cannot answer the loader problem, but would suggest that if you
are looking for web based control that you look at the Web-MeetMe
package. It works against the standard app_meetme.
Dan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John
Melody
Sent
it in one box if possible?
thanks
Mike
http://lartc.org/wondershaper/
That's a good place for starting with traffic shaping, in addition to
Luki's link.
- Dan
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buffers
cached
Mem: 904752 881060 23692 0
94384 413448
-/+ buffers/cache: 373228 531524
Swap: 524280 3600 520680
But most of it is being used for buffers/cache.
- Dan
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http
to route outgoing email for your * machine
through your mail hub. Asterisk doesn't have it's own smtp client, it
just uses the local machines MTA.
- Dan
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the
plug. It also sounded like the carrier is initiating a lawsuit against
BroadVoice.
So to sum up, it seems like a basic contract dispute.
- Dan
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size
2. Limit conference duration
3. Restrict monitoring access to the conference owner
The design goal is to keep the authentication optional, with sane
defaults if no authentication is required.
Thanks,
Dan
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Sounds like a kernel lock up. After you've rebooted, check out
/var/log/messages to see what happened.
- Dan
[EMAIL PROTECTED] wrote:
I don't know if this is related, but the last two mornings I've come
in, the newer AAH 1.0 computer has been locked-up. The Caps Lock and
Scroll Lock lights
) standard (0) 0: 0
Location: Private network serving the local user (1)
Ext: 1 Cause: Unallocated (unassigned) number (1),
class = Normal Event (0) ]
-- Hungup 'Zap/1-1'
--
Dan Goscomb [EMAIL PROTECTED]
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I am using euroisdn
this is a BT ISDN30e (a euroisdn circuit)
On Thu, 2005-05-05 at 01:53 -0700, Kris Boutilier wrote:
-Original Message-
From: Dan Goscomb [mailto:[EMAIL PROTECTED]
Sent: Thursday, May 05, 2005 1:48 AM
To: Asterisk-Users@lists.digium.com
Subject: [Asterisk
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