Looks like version 11.3 did not fix my issue.
http://pastebin.com/gd291Bqz
On Thu, Apr 4, 2013 at 1:23 PM, Duane Larson duane.lar...@gmail.com wrote:
Thanks Jim. Searched through the change log for deadlock but nothing
really stuck out. I'll upgrade to 11.3 and see if that makes
Thanks Jim. Searched through the change log for deadlock but nothing
really stuck out. I'll upgrade to 11.3 and see if that makes a difference.
On Thu, Apr 4, 2013 at 10:59 AM, Jim Lucas li...@cmsws.com wrote:
On 04/03/2013 08:15 PM, Duane Larson wrote:
So it just happened again on both
It just happened again on the 11.0.1 box and I was able to grab a debug. I
am hoping someone can tell me if this is a bug or something wrong with my
config.
gdb asterisk-bin/sbin/asterisk 29048
Go here for the debug output
http://pastebin.com/DGXx0BSk
On Tue, Apr 2, 2013 at 7:42 PM, Duane
with
version 11.2.1
http://pastebin.com/mbjSSAWM
This has to be a bug right? I am thinking of opening an issue on the
Asterisk JIRA system
On Wed, Apr 3, 2013 at 4:45 PM, Duane Larson duane.lar...@gmail.com wrote:
It just happened again on the 11.0.1 box and I was able to grab a debug.
I am
I am currently running two different versions of Asterisk
11.0.1
11.2.1
I have noticed the bug occur on both servers.
The issue is that when I try to dial a phone number sometimes the call will
never go out. I will check the Asterisk server with NGREP and see that the
SIP messages are making
?
On Tue, Jun 14, 2011 at 5:18 PM, Duane Larson duane.lar...@gmail.comwrote:
One more piece to add. I had mentioned before that I could get a call from
a PSTN user to work the first time. So here is all the output of a Good
call from a PSTN user after I have performed a RELOAD on asterisks CLI
or not.
[SATISH]
On Mon, Jun 13, 2011 at 9:12 PM, Duane Larson duane.lar...@gmail.comwrote:
Yesterday I rebooted the server and it seems to be working again. Not
sure what the reboot might have changed. Hopefully it doesn't happen again
but I can't be sure. To answer your question I have
all calls from the PSTN
afterward get put in the queue automatically and the agent never gets
called.
On Tue, Jun 14, 2011 at 4:37 PM, Duane Larson duane.lar...@gmail.comwrote:
Ok. Something isn't right. With a user that is local to my SIP user
database calls the queue phone number everything
every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
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--
*--*--*--*--*--*
Duane
Queue not sending call to Agent
I am having an issue and i am not sure if it is a bug or a config issue. I
was originally running Asterisk 1.8.1.1 when I noticed this issue. I
upgraded to 1.8.4.2 to see if that would fix it but it didn't.
The issue is that I have a call queue and the
mailing list
To UNSUBSCRIBE or update options visit:
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Duane
*--*--*--*--*--*
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Patch worked like a charm. Thanks Chad. Thought I had done something wrong
when installing. Really appreciate it.
On Mon, Jan 10, 2011 at 2:27 PM, Duane Larson duane.lar...@gmail.comwrote:
Thanks Chad. I will try the patch.
On Mon, Jan 10, 2011 at 2:27 PM, Chad Wallace cwall
I look in 2011 spool directory I don't see any message at all. It
is just not being copied. What could be the issue?
--
--
*--*--*--*--*--*
Duane
*--*--*--*--*--*
--
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I have asterisk 1.8.0 installed and I am not able to forward a voicemail
from one users mailbox to another user.
I have the user log into their mailbox
press 8 to forward a message
enter the extension of the user I wish to forward too
I don't prepend a audio message
and press # to send the
I have configured my mysql database by following this link
http://www.voip-info.org/wiki/view/Asterisk+RealTime+Queue
The only difference is that I am using ODBC instead of MySQL with Realtime.
Within extensions.conf I have the following for my queue
exten = 9**2**1611,1,Answer
exten =
Snom
Sent from Droid
On Dec 17, 2010 12:36 PM, Matt mhop...@gmail.com wrote:
I'm looking for a wireless desktop VoIP phone. Does any exist?
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New to
Awesome. Didn't notice that, but that is my fault for not reading the
changelog or the updated sample configs. I will try this out.
Thanks all for the comments.
On Wed, Dec 1, 2010 at 10:09 AM, Tilghman Lesher tles...@digium.com wrote:
On Tuesday 30 November 2010 18:34:17 Duane Larson wrote
I have MySQL Cluster set up for OpenSIPS which allows for the best Redundant
High-Availability. I was wondering if it's possible for Asterisk to also
use multiple database servers for Realtime? Currently with Realtime I am
only able to point to a single IP address for a database. If that
automated failovers in
MySQL in the 1-2 second range.
Singer
On Tue, Nov 30, 2010 at 19:51, David Backeberg dbackeb...@gmail.comwrote:
On Tue, Nov 30, 2010 at 7:34 PM, Duane Larson duane.lar...@gmail.com
wrote:
I have MySQL Cluster set up for OpenSIPS which allows for the best
Thats sounds interesting too. I will look into that also.
On Wed, Dec 1, 2010 at 1:30 AM, Stefan Schmidt s...@sil.at wrote:
Am 01.12.10 05:10, schrieb Duane Larson:
For me OpenSIPS will do most of the work. Asterisk will only handle Hunt
Groups/Queues, IVRs, and Voicemail when OpenSIPS
Your router would have to do per-destination when it came to load balancing
between the two dsl circuits. That way a single call could only use one dsl
path.
On Nov 2, 2010 7:36 PM, Dan Journo d...@keshercommunications.com wrote:
Hi,
I've got a client with two ADSL connections for
I am trying to set up Hunt Groups and I am having some issues. Here is what
I am trying to do. All my users actually register with OpenSIPS. Asterisk
is using Realtime and I have set up a MySQL View Table so that Asterisk
see's all the SIP users info that OpenSIPS has. This is what I have
We want to allow users to record calls,
have the recording go to the users voicemail box, and then email the user
with the recording.
We are running Asterisk 1.2
Best regards,
Duane Pudenz
Network Infrastructure Manager
Shasta Industries
o. 602.532.3706
c
should we be looking for ?
Best regards,
Duane Pudenz
Network Infrastructure Manager
Shasta Industries
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I do not think so by reading the documentation
however I have changed the settings and still get the same error when starting
Asterisk
Best regards,
Duane Pudenz
Network Infrastructure Manager
Shasta Industries
- Message from [EMAIL PROTECTED] on Mon, 20 Feb 2006 07:08:05
+0100
Issue resolved, Thanks Digium!
The module slot closest to the bracket
that contains the connector is the last slot. I assumed that the
first slot would be at the bracket.
Silly user error, never assume.
Best regards,
Duane Pudenz
Network Infrastructure Manager
Shasta Industries
o
=outstation
channel= 1-4
signalling=fxs_ks
echocancel=yes
echocancelwhenbridged=yes
usecallerid=yes
group=2
context=incomingpstn
channel= 5-16
Best regards,
Duane Pudenz
Network Infrastructure Manager
Shasta Industries___
--Bandwidth
We are testing our Asterisk server prior to deployment. The server has
a 4 port T1 Digium adapter (WCT4XXP) with 3 Long Distance (LD) T1s and
one PRI for local calls.
We are using sipp from two different stations routing a test number out
the LD lines and another test number out the PRI line.
and a single number on a business card
and all the other information can be referenced from our DNS zone.
--
Best regards,
Duane
http://www.cacert.org - Free Security Certificates
http://www.nodedb.com - Think globally, network locally
http://www.sydneywireless.com - Telecommunications Freedom
http
[FATAL: Module zaptel not found.]
Best regards,
Duane
Pudenz
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did you follow the directions in doc/cdr.txt in the asterisk distro?
Duane Cox
- Original Message -
From: PA [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Thursday, May 26, 2005 3:16 PM
Subject: [Asterisk-Users] tds_CDR and MS SQL Server troubleshooting
I am trying
Right, your ATA (cable modem?) is using the NCS 1.0 profile of MGCP.
Asterisk does not currently support the NCS profile from cable labs.
It would be very nice to see someday, as well as backwards support for MGCP 0.1
Duane Cox
- Original Message -
From: Huddleston, Robert [EMAIL
I know others have this hardware config, but they use SIP between the two.
- Original Message -
From: list [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Monday, May 16, 2005 12:02 PM
Subject: [Asterisk-Users] Lucent TNT ASTERISK
Anybody using asterisk to talk to a
question is, why is your phone sending RSIP during a call?
Maybe your phone isn't 100% setup yet, but enough
to get calls up.
See if you can capture an mgcp debug from the * CLI.
Duane
- Original Message -
From: Ben Dugdale [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent
to change that; aaln/1 and phone/1 are typically used
for endpoint definitions, but I see that most manufactures use aaln/x
I would try the 2 above recommendations and then see if that helps clear up
the problem.
Duane Cox
- Original Message -
From: Ben Dugdale [EMAIL PROTECTED]
To: Duane Cox
]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, May 12, 2005 6:51 PM
Subject: Re: [Asterisk-Users] Asterisk with ShoreTel 210 (MGCP)
Duane Cox wrote:
Yes * can work with MGCP phones directly. You have a configuration issue.
Glad to hear
that your end device is setup looking for host/domain name
convention and not IP.
If so, change * to match or change your device to IP and not dns.
GoodLuck,
Duane Cox
- Original Message -
From: Ben Dugdale [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Thursday, May 12
Do you get 2-way audio that sometimes drops off to 1-way audio then picks
back up as 2-way? (Thats what I see)
Not sure if my problem is a lost packet issue as I am sending IAX off net.
Duane Cox
- Original Message -
From: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent
What specific Dell Servers are you having trouble with, and what specific
TDM card?
I'm running dell 2650 and the TE410P and haven't had any problems... yet...
Duane Cox
- Original Message -
From: Matt Schulte [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Thursday
post your mgcp.conf (mask out any ips if you need to)
duane cox
- Original Message -
From: [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, March 16, 2005 3:00 PM
Subject: [Asterisk-Users] MGCP Channel Lockup
at the end of the day spam filters aren't just good of getting
rid of one type of menace...
--
Best regards,
Duane
http://www.cacert.org - Free Security Certificates
http://www.nodedb.com - Think globally, network locally
http://www.sydneywireless.com - Telecommunications Freedom
http
On Fri, March 18, 2005 16:01, Jay Milk said:
Hi folks, I think my little agi script is ready for the big one-oh-oh.
Feel free to check the e164.org zone for TXT records...
--
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Duane
http://www.cacert.org - Free Security Certificates
http://www.nodedb.com - Think globally
of stateless UDP connections...
--
Best regards,
Duane
http://www.cacert.org - Free Security Certificates
http://www.nodedb.com - Think globally, network locally
http://www.sydneywireless.com - Telecommunications Freedom
http://happysnapper.com.au - Sell your photos over the net!
http://e164.org - Using
MGCP devices with
asterisk and the AUEP works just fine.
You should be able to audit a wildcard or and endpoint without any trouble.
Duane Cox
- Original Message -
From: Fabio Margarido [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Wednesday, March 09, 2005 11:12 AM
Subject
tired of not having proper enum routing in asterisk I hacked
up a php script ages ago to handle it...
http://www.e164.org/enum.phps
--
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Duane
http://www.cacert.org - Free Security Certificates
http://www.nodedb.com - Think globally, network locally
http://www.sydneywireless.com
, catch being that they're not really allocated by
anyone/body except our DNS zone... Any number ranges in our zone are
also accessible from FWD etc etc etc...
--
Best regards,
Duane
http://www.cacert.org - Free Security Certificates
http://www.nodedb.com - Think globally, network locally
http
that for those of you wanting to send/receive calls to/from
FWD subscribers you can now do so, easily and without needing to stay
registered to their servers... Just prefix your caller ID with **164 if
you want to make it easy for people to hit redial...
--
Best regards,
Duane
http://www.cacert.org - Free
, but the stable version
waits a long time before attempting to send, the strange thing is the dial
plans for disa are identical and it was the same sipura connecting to
both...
--
Best regards,
Duane
http://www.cacert.org - Free Security Certificates
http://www.nodedb.com - Think globally, network
://www.asterisk.net.au/voip%20in%203%20beers.ppt
Anyone is free to use the slides etc as long as both John Todd and I get
credit where credit is due etc...
--
Best regards,
Duane
http://www.cacert.org - Free Security Certificates
http://www.nodedb.com - Think globally, network locally
http
to
another one if one isn't reachable instead of failing like that...
--
Best regards,
Duane
http://www.cacert.org - Free Security Certificates
http://www.nodedb.com - Think globally, network locally
http://www.sydneywireless.com - Telecommunications Freedom
http://happysnapper.com.au - Sell your
...
--
Best regards,
Duane
http://www.cacert.org - Free Security Certificates
http://www.nodedb.com - Think globally, network locally
http://www.sydneywireless.com - Telecommunications Freedom
http://happysnapper.com.au - Sell your photos over the net!
http://e164.org - Using Enum.164
for the information on
that website...
--
Best regards,
Duane
http://www.cacert.org - Free Security Certificates
http://www.nodedb.com - Think globally, network locally
http://www.sydneywireless.com - Telecommunications Freedom
http://happysnapper.com.au - Sell your photos over the net!
http://e164.org
hear a full word,
You could always just use e164 for toll free numbers, we have sip urls
for about 11 countries and international toll free in our zone, and I've
never had an issue with call quality to the US toll free numbers...
--
Best regards,
Duane
http://www.cacert.org - Free Security
if it could be possible to make it
so that after hitting 9.. The tone would change to something else letting
the user know that they are dialing on an outside line.
Yes, you can do this, stick a extension in your dial plan for 9, then
point that to app_disa...
--
Best regards,
Duane
http
On Sun, February 20, 2005 21:56, Peter Svensson said:
Or have the 9 dial an outside line and get the external dialtone.
Which will only work if you're actually sending the call to an outside
line...
--
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Duane
http://www.cacert.org - Free Security Certificates
http
regards,
Duane
http://www.cacert.org - Free Security Certificates
http://www.nodedb.com - Think globally, network locally
http://www.sydneywireless.com - Telecommunications Freedom
http://happysnapper.com.au - Sell your photos over the net!
http://e164.org - Using Enum.164 to interconnect asterisk
somenumber@host where some number is an existing number in your
incoming dial plan...
http://www.e164.org has more examples of this because that's how it works...
--
Best regards,
Duane
http://www.cacert.org - Free Security Certificates
http://www.nodedb.com - Think globally, network locally
Colin Anderson wrote:
http://advancedippipeline.com/60400413
I don't know why vonage isn't installing/using SRV records so they can
stuff about with ports, rather then having a static port 5060...
--
Best regards,
Duane
http://www.cacert.org - Free Security Certificates
http://www.nodedb.com
of and ended just using
xten lite under wine...
--
Best regards,
Duane
http://www.cacert.org - Free Security Certificates
http://www.nodedb.com - Think globally, network locally
http://www.sydneywireless.com - Telecommunications Freedom
http://happysnapper.com.au - Sell your photos over the net
Bruno Hertz wrote:
On Mon, 2005-02-14 at 22:29 +1100, Duane wrote:
I gave up trying to use linux soft clients they all seem to have some
fatal flaws or issues I could never fully get rid of
While I'd second that, Gnomemeeting is still pretty good and by far the
best softphone I've used on Linux
,
Duane
http://www.cacert.org - Free Security Certificates
http://www.nodedb.com - Think globally, network locally
http://www.sydneywireless.com - Telecommunications Freedom
http://happysnapper.com.au - Sell your photos over the net!
http://e164.org - Using Enum.164 to interconnect asterisk
channels...
--
Best regards,
Duane
http://www.cacert.org - Free Security Certificates
http://www.nodedb.com - Think globally, network locally
http://www.sydneywireless.com - Telecommunications Freedom
http://happysnapper.com.au - Sell your photos over the net!
http://e164.org - Using Enum.164
, not to
mentioned biased towards their own services, for things such as
internet...
--
Best regards,
Duane
http://www.cacert.org - Free Security Certificates
http://www.nodedb.com - Think globally, network locally
http://www.sydneywireless.com - Telecommunications Freedom
http://happysnapper.com.au
Joseph wrote:
Can anybody confirm if IAX on FWD is down again?
I can not register IAX with FWD.
I got fed up with the yo-yo, which then led me to dump fwd and install
asterisk and start playing with inter-asterisk routing via e164.org...
--
Best regards,
Duane
http://www.cacert.org - Free
Joseph wrote:
I wander what is causing the problem, I was thinking that it was
something on my part but I did not change any settings and IAX2 registry
At the time the only thing I could put it down to was congestion...
--
Best regards,
Duane
http://www.cacert.org - Free Security Certificates
iax network on demand or fail over to pstn etc... examples can be found
on both www.e164.org and www.asterisk.net.au...
--
Best regards,
Duane
http://www.cacert.org - Free Security Certificates
http://www.nodedb.com - Think globally, network locally
http://www.sydneywireless.com
one, you can take iaxclient and compile it as a DLL.
I did that and now I'm using it with Delphi. My phone is almost done :)
I'll post it here when it's ready (really soon)
What about a kylx(sp?) version for linux?
--
Best regards,
Duane
http://www.cacert.org - Free Security Certificates
http
César Davi Ávila do Nascimento wrote:
Hi All,
I have a question for you:
- SIP doesn't work behind NAT very well
Do you agree with this sentence?
Depends on the NAT/firewall in question, you can also alleviate some of
these issues using STUN and sip proxing...
--
Best regards,
Duane
http
have 2 zones, e164.org for people putting their names or other
info and cc.e164.org stores a whole bunch of exchange information, we
are always looking for more sources of data to make the information more
granular as well...
--
Best regards,
Duane
http://www.cacert.org - Free Security
answering in some circumstances.
Has anyone been able to make firefly work under wine at all? If so how?
A decent linux client is the only thing skype has over SIP/IAX...
--
Best regards,
Duane
http://www.cacert.org - Free Security Certificates
http://www.nodedb.com - Think globally, network locally
Adam Hart wrote:
Few people have claimed success, I'm not sure how though.
Any chance of a native linux version then? :)
--
Best regards,
Duane
http://www.cacert.org - Free Security Certificates
http://www.nodedb.com - Think globally, network locally
http://www.sydneywireless.com
a a-z on how to setup both
linux (debian) and asterisk from cvs etc... http://www.asterisk.net.au
--
Best regards,
Duane
http://www.cacert.org - Free Security Certificates
http://www.nodedb.com - Think globally, network locally
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http
to quickly get
something in place now that the base code exists.
--
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Duane
http://www.cacert.org - Free Security Certificates
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in asterisk on the main box then
don't allow transfers on the remote boxes and don't use transfer buttons...
--
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Duane
http://www.cacert.org - Free Security Certificates
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to the remote SIP
clients _always_ go thru the co-located box (with its extra bandwidth).
Erm that's the assumption I was making, there was a centralised box
somewhere...
--
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Duane
http://www.cacert.org - Free Security Certificates
http://www.nodedb.com - Think globally, network
yourself with e164.org.
--
Best regards,
Duane
http://www.cacert.org - Free Security Certificates
http://www.nodedb.com - Think globally, network locally
http://www.sydneywireless.com - Telecommunications Freedom
http://happysnapper.com.au - Sell your photos over the net!
http://e164.org - Using
even test it, I just wrote it up using c+p from your original
version in the email window off the top of my head... :)
--
Best regards,
Duane
http://www.cacert.org - Free Security Certificates
http://www.nodedb.com - Think globally, network locally
http://www.sydneywireless.com
Emanuele Venditti wrote:
Hi Howard,
could you share your indications.conf settings as well?
I appreaciate that.
manny
Correct indications for Australia was merged into the CVS long ago...
--
Best regards,
Duane
http://www.cacert.org - Free Security Certificates
http://www.nodedb.com - Think
,
Duane
http://www.cacert.org - Free Security Certificates
http://www.nodedb.com - Think globally, network locally
http://www.sydneywireless.com - Telecommunications Freedom
http://happysnapper.com.au - Sell your photos over the net!
http://e164.org - Using Enum.164 to interconnect asterisk servers
, we have an API to handle injecting, updating
and removing these and are happy to customise it if people already use a
similar system and would prefer not to need modifying their end.
--
Best regards,
Duane
http://www.cacert.org - Free Security Certificates
http://www.nodedb.com - Think globally
,Playback(invalid)
exten = _44808.,4,Hangup
exten = _44808.,103,Busy
--
Best regards,
Duane
http://www.cacert.org - Free Security Certificates
http://www.nodedb.com - Think globally, network locally
http://www.sydneywireless.com - Telecommunications Freedom
http://happysnapper.com.au - Sell your
about after you say my
computer did it).
Personally, Bellster is zero benefit to me, as my local calls cost more
than what I get from our VoIP carrier. I suppose to save a few bucks
here and there it could potentially be worth it for some people.
-Michael
--
Best regards,
Duane
http
Duane wrote:
I was discussing bellster with a friend of mine, and he made another
point about this service...
I can't imagine how unsettling it would be for my girlfriend to pick up the
phone and hear somebody else on the line. The first time that happened, that'd
be the end of me sharing
?
As for the original question, the 2 ports on the 2000 and the 3000 are
both seperate SIP identities and you have to configure them as 2
seperate lines...
--
Best regards,
Duane
http://www.cacert.org - Free Security Certificates
http://www.nodedb.com - Think globally, network locally
http
Nathan Alberti wrote:
Duane,
My apologies if I have misunderstood but is this an error ?
Dialing a 1300XX, number would make it 611300XX, then jumping to
StripMSD(3) would make it 300XX ?
I must update that, I completely re-wrote it and the more up to date
version is at:
http
Robert Rozman wrote:
I'd like to setup little private enum server. Any more info on how to do
that ?
You just need bind or any other name server that supports NAPTR records
and to setup a zone with NAPTR records...
--
Best regards,
Duane
http://www.cacert.org - Free Security Certificates
http
of a SIP or IAX provider
exists?
Just do a if goto call...
exten = s,2,GotoIf($[$[${ENUM:0:3} = SIP] | $[${ENUM:0:3} = IAX]] ? 3 : 52)
That traps both, but obviously can be altered to trap one or the other
and handle them seperately...
--
Best regards,
Duane
http://www.cacert.org - Free Security
What area of IL are you in?
- Original Message -
From: Jay Milk [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
[EMAIL PROTECTED]
Sent: Tuesday, December 07, 2004 11:01 AM
Subject: [Asterisk-Users] IAX DIDs, Illinois
I have been looking at moving from
with any other calling location, but she
can duplicate it every time.
Anybody know what's up with that?
Thanks
Duane Cox
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There are mgcp examples in configs/mgcp.conf.sample
in the source dir.
What do you mean by "...I found some problems with
the MGCP messages..."
You have to be more specific than that if you would
like help on the topic.
Duane Cox
- Original Message -
From:
We use several Dell 2650 servers. Order them
with the dual DC power supply option.
Buy a row of -48 batteries and a -48 power source,
your servers will stay up for hours.
- Original Message -
From:
TinKoon
To: 'Asterisk Users Mailing List -
Non-Commercial Discussion'
I wish I was wrong, but I think right now MGCP in Asterisk is CallAgent
only.
http://www.voip-info.org/tiki-index.php?page=Asterisk%20MGCP%20channels
- Original Message -
From: Tim Jackson [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, November 23, 2004 2:48 PM
Subject:
Michael Vogel wrote:
I just downloaded it. Now I only need to know, how to include it in
asterisk. The documention is ... hmm ... ;-)
http://www.e164.org/enum.phps
Little script I whipped up a while back that doesn't need anything but
the php binary to work...
--
Best regards,
Duane
http
(after verification) and non-real numbers...
--
Best regards,
Duane
http://www.cacert.org - Free Security Certificates
http://www.nodedb.com - Think globally, network locally
http://www.sydneywireless.com - Telecommunications Freedom
http://happysnapper.com.au - Sell your photos over the net!
http
so people can keep their numbers and hostnames etc up to date
themselves... /2cents...
--
Best regards,
Duane
http://www.cacert.org - Free Security Certificates
http://www.nodedb.com - Think globally, network locally
http://www.sydneywireless.com - Telecommunications Freedom
http
to a file?
exten = 200,2,System(/bin/echo -e 'Incoming Call From: ${CALLERID}\n
Received: ${DATETIME}'|/usr/bin/smbclient -M target_netbios_name)
--
Best regards,
Duane
http://www.cacert.org - Free Security Certificates
http://www.nodedb.com - Think globally, network locally
http
Thomas Hutton wrote:
Hi Duane,
You asked Why dump to a file? - I don't know if this is possible or
not, but can you send a ctrld to the smbclient -M command? I believe
the way you wrote the command it will just hang, no?
the equiv of ctrl+d is hit when it runs out of things to echo
...
--
Best regards,
Duane
http://www.cacert.org - Free Security Certificates
http://www.nodedb.com - Think globally, network locally
http://www.sydneywireless.com - Telecommunications Freedom
http://happysnapper.com.au - Sell your photos over the net!
http://e164.org - Using Enum.164 to interconnect
not really obtain it
from the PBX. Does anyone know how I can tackle this issue?
app_disa
--
Best regards,
Duane
http://www.cacert.org - Free Security Certificates
http://www.nodedb.com - Think globally, network locally
http://www.sydneywireless.com - Telecommunications Freedom
http://happysnapper.com.au
if the call will go through or not, I usually get round this by
having fail over, if it fails to connect to the first route, drop to the
second, then the 3rd etc...
--
Best regards,
Duane
http://www.cacert.org - Free Security Certificates
http://www.nodedb.com - Think globally, network locally
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