Are you of that ?
I'm not 100% certain, but I think Thomson phones wouldn't query centralized
directory for outbound calls.
I think centralized directory is only queried when using Directory key.
well, our customers use it by opening phonebook and selecting
number to call ... so don't know
On Saturday 04 October 2008 11:47:01 Guillermo V. Salas wrote:
How can I prevent a remote DoS as described on the following site? :
http://www.voip0day.com/news/remote-denial-of-service-exploit-effects-the-a
sterisk-pbx/
This has already been addressed in the following advisory:
2008/10/5 Mr Shunz [EMAIL PROTECTED]
Are you of that ?
I'm not 100% certain, but I think Thomson phones wouldn't query
centralized
directory for outbound calls.
I think centralized directory is only queried when using Directory key.
well, our customers use it by opening phonebook and
On October 3, 2008 04:15:26 pm Tariq .. wrote:
it is FRING i'm sorry for the mistype...
www.fring.com
I just downloaded it for the iphone... it's pretty cheap looking, crashes
occasionally and appears to force all audio through their server, but I have
to say that yes, it does have potential.
On October 3, 2008 08:56:34 pm Philipp Kempgen wrote:
I could live with 1 or maybe 2 of these issues but 5 is a bit
much. You didn't even notice these problems, so, ok, sorry for
being rude. But for people who are used to email in ages it feels
like a punch in the face. It's a real culture
2008/10/4 Tzafrir Cohen [EMAIL PROTECTED]
On Sat, Oct 04, 2008 at 02:02:48PM +0200, Olivier wrote:
Hi,
You can see here and there, several new SIP RFCs relying on SIP Events
Framework.
For example, RFC3680 with which a registration server would notify
endpoints
with relevant events.
Hello.
Have you ever tried updating your GCC version?
Thanks.
I am using cross compile so i can't update GCC other wise it will effect on
my other packages anyway... tell me one thing i have host system kernel
version is 2.6.18 and i am compiling ARM embedded rootbuild with other
On Thu, Oct 02, 2008 at 11:33:01AM -0400, Satish Patel wrote:
I wanted to show you what option i used now i have download zaptel-1.4.12.1
clfs:/mnt/clfs/sources/zaptel-1.4.12.1$ ./configure --host=${CLFS_TARGET}
--prefix=/usr
configure: WARNING: If you wanted to set the --build type,
On Sun, Oct 05, 2008 at 11:28:47AM -0400, satish patel wrote:
Hello.
Have you ever tried updating your GCC version?
Thanks.
I am using cross compile so i can't update GCC other wise it will effect on
my other packages anyway... tell me one thing i have host system kernel
Andrew Kohlsmith (lists) schrieb:
Having been a user of email and a staunch advocate of text-only messages,
minimal signature lines, proper trimming and bottom posting for well over 15
years, I have to say that I've never felt punched in the face nor
experienced any kind of culture clash.
2008/10/5 Andrew Kohlsmith (lists) [EMAIL PROTECTED]
On October 3, 2008 04:15:26 pm Tariq .. wrote:
it is FRING i'm sorry for the mistype...
www.fring.com
I just downloaded it for the iphone... it's pretty cheap looking, crashes
occasionally and appears to force all audio through their
On October 5, 2008 12:22:37 pm Philipp Kempgen wrote:
Thunderbird could probably render his text/html part just fine but
I don't want it to. (Nothing is wrong with preferring text/plain in
the MUA.)
Thus it renders his text/plain part which lacks line breaks.
I posted some links to the list
Andrew Kohlsmith (lists) schrieb:
On October 5, 2008 12:22:37 pm Philipp Kempgen wrote:
---cut---
http://lists.digium.com/pipermail/asterisk-users/2008-October/219538.html
http://lists.digium.com/pipermail/asterisk-users/2008-October/219541.html
---cut---
That quoted text is not very
On Thu, Oct 02, 2008 at 11:33:01AM -0400, Satish Patel wrote:
I wanted to show you what option i used now i have download
zaptel-1.4.12.1
clfs:/mnt/clfs/sources/zaptel-1.4.12.1$ ./configure
--host=${CLFS_TARGET} --prefix=/usr
configure: WARNING: If you wanted to set the --build
Hello,
I've this situation: 300+ simultaneous calls and dialplan like this:
exten = _X.,1,Answer()
exten = _X.,2,DEADAGI(check_status.php)
exten = _X.,3,Dial(SIP/other/${NUMBER})
exten = _X.,4,Hangup
exten = h,1,DEADAGI(cdr.php)
When project is running , I had a lot of defunct php scripts
Giedrius Augys schrieb:
I've this situation: 300+ simultaneous calls and dialplan like this:
exten = _X.,1,Answer()
exten = _X.,2,DEADAGI(check_status.php)
exten = _X.,3,Dial(SIP/other/${NUMBER})
exten = _X.,4,Hangup
exten = h,1,DEADAGI(cdr.php)
When project is running , I had a
On Sun, 5 Oct 2008, Giedrius Augys wrote:
I've this situation: 300+ simultaneous calls and dialplan like this:
exten = _X.,1,Answer()
exten = _X.,2,DEADAGI(check_status.php)
exten = _X.,3,Dial(SIP/other/${NUMBER})
exten = _X.,4,Hangup
exten = h,1,DEADAGI(cdr.php)
When project is
Very well put.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Philipp
Kempgen
Sent: October 5, 2008 1:07 PM
To: Asterisk Users
Subject: Re: [asterisk-users] OT: text/plain
Andrew Kohlsmith (lists) schrieb:
On October 5, 2008 12:22:37 pm Philipp Kempgen
Philipp Kempgen wrote:
Andrew Kohlsmith (lists) schrieb:
On October 5, 2008 12:22:37 pm Philipp Kempgen wrote:
---cut---
http://lists.digium.com/pipermail/asterisk-users/2008-October/219538.html
http://lists.digium.com/pipermail/asterisk-users/2008-October/219541.html
Some users at a new Asterisk installation with Polycom IP330 phones are
complaining about echo with the amplified headsets they used to use with
their Nortel phones. I listened myself, and I here my own voice
annoyingly loudly, and no headset/phone combination of volume control
manipulation
Olivier wrote:
I suspect my understanding of it is incorrect as I would say that if
an extension is on call with someone else, curcalls shall return 1
(which it doesn't here as it returns 0).
From what I can see, this is a counter for active calls.
From one sip phone to another I called
Kevin P. Fleming wrote:
Olivier wrote:
2. R Hook-flash key is now available to transfer calls.
In s450IP web management server, its defaults settings are :
Application-type: dtmf-relay
Application-signal: 16
Is there anything to configure in features.conf, extensionsconf or
elsewhere
Regards,
Satish Patel
Quoting Tzafrir Cohen [EMAIL PROTECTED]:
On Sun, Oct 05, 2008 at 11:28:47AM -0400, satish patel wrote:
Hello.
Have you ever tried updating your GCC version?
Thanks.
I am using cross compile so i can't update GCC other wise it will effect on
my other
I'm looking at the app_voicemail.c from both 1.4 and 1.6.1 and seeing
that neither allows an individual mailbox to override the imapfolder
value. It seems entirely intuitive to me that one might want to do
that, not to mention how trivial it looks to add that to
app_voicemail.c.
Maybe my
Yes, you can set moh in sip.conf or zapata.conf. The options are
mohinterpret= mohsuggest=. I think last time I used them (1.2.x)
they were just moh= but it seems mohsuggest=class will do what
you want it to.
On Sat, Oct 4, 2008 at 2:57 PM, carl Lougher [EMAIL PROTECTED] wrote:
This seems
How much further than 300m? It might be very well possible to just
lower the speed to 10M and just use that If you already have some
quality Cat5 cable between both points it's worth a shot. I support
some sites with this arrangement and I've had to find 10M hubs for
replacement hardware (the
Yes. Disable VAD in your Cisco as Asterisk does not (fully) support it.
On Wed, Oct 1, 2008 at 9:21 PM, Gabriel Ortiz Lour
[EMAIL PROTECTED] wrote:
Hi all,
I'm experiencing problems with VAD activated on a cisco router doing the
bridge between an PBX and de asterisk server. The calls are
Has anyone successfully used the IMAP voicemail storage with Microsoft
Exchange 2003? Can someone provide a working example configuration?
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25
Isn't IMAP IMAP? Does MS not actually follow the protocol? Why would
it be different?
On Sun, Oct 5, 2008 at 8:38 PM, Andrew Joakimsen [EMAIL PROTECTED] wrote:
Has anyone successfully used the IMAP voicemail storage with Microsoft
Exchange 2003? Can someone provide a working example
Yes, IMAP is IMAP... at least it is supposed to.
But not all IMAP servers use the same configuration. Not all IMAP
servers will use the same Master User IMAP setup, what works in
Dovecot might not work in UW or Exchange due to a prefix or some other
fairly trivial setting. Remember there are two
I'm attempting to convert from ZAP to DAHDI with 1.6.0.
I was using 1.6.0-beta9.
I followed the directions I could find.
I moved /etc/zapata to /etc/dahdi/system.conf
I moved /etc/asterisk/zapata.conf to /etc/asterisk/chan_dahdi.conf
I don't undestand how to deal with extensions.conf?
I
Here's a couple of distances I'm looking to cover (distances are +- 10%):
1 at 400M
1 at 600M
1 at 1800M
1 at 2400M
some of these links may already have pots circuits complete with occasional
ringing voltage in the same conduit (but likely not the same cable). how
far can I push the distance of
I don't know how to explain this.
After receiving 1 inbound call on the DAHDI channel attached
to the PSTN, outbound calls to the PSTN start working with
getting the unable to create channel if type DAHDI message.
If I restart *, the problem returns until I get 1 inbound call.
Jim
d calls from
Even that isn't always true.
Sometimes dial out on DAHDI works, sometimes it doesn't.
I'm not sure what makes it start working, but once it does,
it appears to stay working.
Jim
Jim Duda wrote:
I don't know how to explain this.
After receiving 1 inbound call on the DAHDI channel attached
I was going to write a blog once about the non-existent T.38 support
in asterisk hence my purchase of the above domain. It expires in 10
days. T.38 support in asterisk still does not exist but I don't have
any time. If someone wants this domain I will offer it for free and
can send push it to your
only condition would be that you do not use it for a
commercial use, i.e. you don't try to sell a t.38 module for asterisk.
If you want to retain any control of what it is used for, you better
re-register it. Once it expires and some one else gets it, you have no say
in the matter.
John
I think the problem is that every [Dead]AGI call is still a distinct
invocation of the script, even if the interpreter stays loaded as an ELF
module or whatnot.
A good solution to this problem would be to use a FastAGI service,
wherein a daemon runs persistently with a reusable DB handle.
On Fri, 3 Oct 2008 12:00:16 -0800, Babcock, Michael Alex wrote:
is it frig or fring?
On Oct 3, 2008, at 11:49 AM, Tariq .. wrote:
try using Frig.. it's a great client with an SIP client.. i tried it
on IPhone and on my N82 Nokia phone.. it works great on GPRS and Wi-
Fi... and i DO
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