sip show peer ovh
* Name : ovh
Secret : Set
MD5Secret: Not set
Context : entrant-ovh
Subscr.Cont. : Not set
Language : fr
AMA flags: Unknown
Transfer mode: open
CallingPres : Presentation Allowed, Not Screened
Callgroup:
Pickupgroup :
7 apr 2009 kl. 18.26 skrev Florian Hackenberger:
On Tuesday 07 April 2009, Olle E. Johansson wrote:
I don't see any problems there. YOu still have devices with states,
as you would have with authentication. Of course, it still depends on
your configuration. But authentication should not
Hi all,
someone has used the voice recognition software named Sphinx??? Can he tell
me how to use and its performance???
Thanks
Marco
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To UNSUBSCRIBE
Hi, I know this is a little OT but there are many Asterisk users of the
excellent Siemens DECT/VOIP phones like the S685IP and 475IP and this is
probably newsworthy for them.
One of the biggest bug bears has been no mute function on the handset.
When I woke up this morning, the handset told me
Hello,
Using odbc voicemail and mysql, i have a problem.
After 12 seconds recording, asterisk stop recording and hangup.
I have changed the settings in voicemail.conf to allow 180 seconds but,...
Any hint?
Olivier
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On 8 Apr 2009, at 08:57, hh174 wrote:
Using odbc voicemail and mysql, i have a problem.
After 12 seconds recording, asterisk stop recording and hangup.
I have changed the settings in voicemail.conf to allow 180 seconds
but,...
Any hint?
Does it do it if you don't use odbc voicemail and
On Wed, Apr 8, 2009 at 9:05 AM, Alan Lord (News) alansli...@gmail.com wrote:
Hi, I know this is a little OT but there are many Asterisk users of the
excellent Siemens DECT/VOIP phones like the S685IP and 475IP and this is
probably newsworthy for them.
Oh, yes! This is the greatest news since
2009/4/8 Alan Lord (News) alansli...@gmail.com
* DHCP Option 114 implemented.*
* DHCP Option 120 implemented.*
http://lists.digium.com/mailman/listinfo/asterisk-users
What does it imply ?
Provisionning from DHCP server ?
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2009/4/6 Ed W li...@wildgooses.com:
Hi, got a Sangoma A200 with a bunch of extension cards and having real
problems getting it to deal with a normal single BT line
The A200 is a great card, and we use it quite a lot in the UK. Mostly
we use the A200D for the echo cancellation.
Symptoms are
Hi All,
Thanks for your reply.
I got this refer message in asterisk.
but there is not any active channel of blind transfer.
--
REFER sip:1...@192.168.1.25 sip%3a1...@192.168.1.25 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.30:5060;branch=z9hG4bK5880efa5cca586b0
From:
Hi
I have used the transfer operation this way.
When i got a call on grandstream phone, i will receive it
and press transfer button and enter transfer number and press send button.
My call is disconnected but no call transfer from asterisk.
Please advice me!!
Thanks,
Max Alex
Voip Developer
On
Hello,
I have a problem with Asterisk trunk billing. I have bought some number of
trunks from a VoIP provider with his own rates. I am planning to sell some
of these trunks to my clients with my own rates. The problem is: how to
process this trunk, Can I process it as a normal SIP/IAX client (if
On Tuesday 07 April 2009 11:28:52 Tilghman Lesher wrote:
The recent vulnerability had nothing to do with this, but with the ability
of an attacker to scan a SIP server for legitimate usernames and passwords.
This, by the way, merely took advantage of the SIP protocol, as written.
Normally, SIP
On Wednesday 08 April 2009 00:08:23 Olivier wrote:
I've updated http://www.voip-info.org/wiki/view/Asterisk+func+shell with an
example to test file existence.
Why not just use the STAT() function, in that case?
--
Tilghman
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On 8 Apr 2009, at 14:13, abdelkader wrote:
I have a problem with Asterisk trunk billing. I have bought some
number of trunks from a VoIP provider with his own rates. I am
planning to sell some of these trunks to my clients with my own
rates. The problem is: how to process this trunk,
Steve Davies wrote:
2009/4/6 Ed W li...@wildgooses.com:
We have found that using Residential settings as a starting point,and then
asking for Disconnect clear time to be set to 800ms is all that is needed.
That one setting allows the hangup to be detected reliably. We do also use
David Backeberg wrote:
Hello there:
I think I have a silly kernel configuration problem. I'm running:
* vanilla 2.6.27.10 kernel built from source
* dahdi-2.1.0.4 built from source
So far so good,
dahdi module loads just fine:
dahdi: Telephony Interface Registered on major 196
dahdi:
Hi,
You can start by looking here :
http://www.voip-info.org/wiki/view/Asterisk+billing
Jimmy
-Original Message-
From: abdelkader2...@gmail.com
Sent: Wed, 8 Apr 2009 15:13:21 +0200
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Asterisk Trunk billing
Hello,
I
On Tuesday 07 April 2009 23:38:08 Olivier wrote:
2009/4/7 Mark Michelson mmichel...@digium.com
Philipp Kempgen wrote:
BTW (developer's question) is there a reason why SendText() resp.
sendtext_exec() refuses to send zero-length data?
I can't point to any specific reason. I assume
All,
Im having a problem with ReceiveFax where its generating a ton of these
messages the entire time the receivefax app is running receiving my fax.
[Apr 7 22:16:06] ERROR[26918]: channel.c:2520 __ast_read: ast_read() called
with no recorded file descriptor.
Im running on Centos 5.2 with
Greg Kennedy wrote:
All,
Im having a problem with ReceiveFax where its generating a ton of these
messages the entire time the receivefax app is running receiving my fax.
[Apr 7 22:16:06] ERROR[26918]: channel.c:2520 __ast_read: ast_read()
called with no recorded file descriptor.
Im
Haven't you read my email?
1. Wrong list
2. Missing log entries (set debug 4, set verbose 4)
klaus
Max Alex schrieb:
Hi All,
Thanks for your reply.
I got this refer message in asterisk.
but there is not any active channel of blind transfer.
--
REFER
sangoma support are amazing, they've solved nearly all the problems i've
experienced with PRI, except for one which turned out to be a bug in SWIX
(some rubbish windows based voip pbx, full of bugs and generally crap!).
there also quite happy to log in to your systems and have a look themselves
Also, FC10 is out. You should probably grab that first.
Unless you are a strong Linux Guru, I would never recommend a Fedora release
for a production system. I have FC9 here and FC10. It took me months to
eliminate the bugs from FC9, and I still haven't gotten FC10 to install on the
machine
Does anyone have any first-hand experience with the Zoiper Business version
softphone? If so what has been your experience with it?
Thanks,
Greg
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Centos is a much more appropriate distro for production work. Nothing goes
into it until it is known to be rock solid, and update occur much more
slowly.
Yes, and that also means newer glib etc can be needed sometimes which are
not YET avail on centos, however if you are not a yum freak and
Gregory Malsack wrote:
Does anyone have any first-hand experience with the Zoiper Business
version softphone? If so what has been your experience with it?
Thanks,
Greg
I've been using it on my notebook. I've been happy with it but I'm not
a heavy user. The biggest reason I
On Tue, 7 Apr 2009, George Pajari wrote:
I have an Asterisk 1.4.18 with a mix of cordless phones connected using
Linksys SPA2102 ATAs and
Cisco 7940G phones. Unit obtains SIP trunking from an ITSP (server has
no PCI boards).
*8 Call Pickup works fine from any of the phones connected using
On Wed, 8 Apr 2009, Gregory Malsack wrote:
Does anyone have any first-hand experience with the Zoiper Business
version softphone? If so what has been your experience with it?
I've used it - the business edition supports more than 2 accounts. I think
that's the main difference. (Both now
On 4/8/2009 1:19 PM, Gregory Malsack wrote:
Does anyone have any first-hand experience with the Zoiper Business
version softphone? If so what has been your experience with it?
Thanks,
Greg
I am not a very heavy user of it either, but I'm a semi-regular user,
and I like it a lot. It's the
Gordon Henderson wrote:
On Wed, 8 Apr 2009, Gregory Malsack wrote:
Does anyone have any first-hand experience with the Zoiper Business
version softphone? If so what has been your experience with it?
I've used it - the business edition supports more than 2 accounts. I think
Gregory Malsack wrote:
Does anyone have any first-hand experience with the Zoiper Business
version softphone? If so what has been your experience with it?
I have some experience with it, as i'm on the team producing it :)
Can't give you unbiased comments though, so i guess its better if
Wilton Helm schrieb:
Also, FC10 is out. You should probably grab that first.
And by the way: Debian 5 Lenny is out. http://www.debian.org/
Philipp Kempgen
--
AMOOCON 2009, May 4-5, Rostock / Germany - http://www.amoocon.de
Asterisk: http://the-asterisk-book.com -
2009/4/8 Olivier oza-4...@myamail.com
2009/4/8 Alan Lord (News) alansli...@gmail.com
* DHCP Option 114 implemented.*
* DHCP Option 120 implemented.*
http://lists.digium.com/mailman/listinfo/asterisk-users
What does it imply ?
Provisionning from DHCP server ?
114 is for passing
On Wed, 8 Apr 2009, Philipp Kempgen wrote:
Wilton Helm schrieb:
Also, FC10 is out. You should probably grab that first.
And by the way: Debian 5 Lenny is out. http://www.debian.org/
I've been a Debian user since more or less the begining (of Debian that
is - there was sls before that!)
I
Hi all,
I have the below peace of my AGI script...the problem here is that I cannot
fetch the extension value to inside the script and assign it to another
variable...I highlighted it in red
#!/usr/bin/perl
#use DBD::mysql;
use DBI;
use DBD::mysql;
use Asterisk::AGI;
Here's what fail2ban service caught
The IP 89.111.184.221 has just been banned by Fail2Ban after
80 attempts against ASTERISK.
On Wed, Apr 8, 2009 at 7:01 PM, Tilghman Lesher
tilgh...@mail.jeffandtilghman.com wrote:
On Tuesday 07 April 2009 11:28:52 Tilghman Lesher wrote:
The recent
This is at least correct on my setup
$dest = $input{dnid}
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of michel freiha
Sent: Wednesday, April 08, 2009 4:28 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
It works d\Danny...Thanks a lot for your help
Regards
On Thu, Apr 9, 2009 at 12:37 AM, Danny Nicholas da...@debsinc.com wrote:
This is at least correct on my setup
$dest = $input{dnid}
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
Nice, share the knowledge and send the fail2ban rule ;) ill post mine's
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jaswinder
Singh
Sent: April-08-09 5:34 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
Hi,
I have installed Elastix 1.5.2 (Barranquilla, Colombia (TELCO: METROTEL))
with a TE220P (2xE1) and TDM2400P (16FXS), openr2 is included in 1.5.2
version. The outcoming calls are ok, but with incoming call i have an error:
ERROR*[*14972*]* chan_dahdi.c: Chan 2 - Protocol error. Reason = Multi
I'm afraid I already know the answer because I've done a lot of searching,
but does anyone know of a softphone that supports a central phone book and
paging (like the sip autoanswer option of some hardphones)
Thanks,
David Ruggles
CCNA MCSE (NT) CNA A+
Network EngineerSafe Data, Inc.
Xlite etc, counterpath.com have AA features, not sure about central phone
book.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Ruggles
Sent: April-08-09 10:50 PM
To: 'Asterisk Users Mailing List -
Hi Geraint,
My apologies for the very very late reply. But, I wasn't able to make the
incoming calls park in one extension and pick the call from there. The
agents are quite comfortable with the setup we discussed, as calls they make
will be made ring in their EyeBeam and then gets connected to
Sounds like a protocol variant issue. Is the telco supposed to send you ANI?
You have 2 options, the first option is to try with the ITU variant,
if that does not work, set the option mfcr2_skip_category=yes and see
if that helps.
Moy
On Wed, Apr 8, 2009 at 6:06 PM, Giovanny Magallanes
On 4/04/2009 2:22 a.m., John covici wrote:
The minute asterisk tries to execute an agi, it gets utils.c write
error broken pipe and so hangs up the call.
Anyone know what is going on?
I am using kernel 2.6.27 with dahdi trunk if that makes a difference.
thanks in advance for any ideas.
If I revert back to my old version of asterisk, it works just fine.
on Thursday 04/09/2009 Matt Riddell(li...@venturevoip.com) wrote
On 4/04/2009 2:22 a.m., John covici wrote:
The minute asterisk tries to execute an agi, it gets utils.c write
error broken pipe and so hangs up the call.
On 7/04/2009 9:41 p.m., Olivier wrote:
Hi,
I'm using a SIP phone (Thomson ST2030) which is able to display text
received though Asterisk's SendText() application.
I'm using this to display from Asterisk Forwarded to 0123456789 whenever a
user forwards his calls to another number or
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