Yes, with isql it's working fine, I can see the database and all fields.
On Sun, Jun 20, 2010 at 9:11 PM, Tilghman Lesher tles...@digium.com wrote:
On Sunday 20 June 2010 13:15:11 Andraž wrote:
If I use MySQL with the same fields it's working fine. I think that is
something wrong with
Hi listusers,
I am using call file to dial out the sip on a different machine. The
problem is whenever i dial the call lands up on the softphone but i have to
pick it up 2 times, for both line 1 and line 2. If i reject it in the 1st
time only then both are rejected.
channel: SIP/2001
Hello dear list.
I am having issues on parkedcalls.
I am using a Cisco SPA525G as a test phone, and I have the transfer button
there when I am in a call,
But when I want to transfer the current call I am in, I push the transfer
button, and onscreen I se Enter Number, and if I enter ex sip
On Sun, Jun 20, 2010 at 07:55:07PM -0400, Michelle Dupuis wrote:
And after cleaning everything 323/pwlib/ptlib related, and reinstalling ptlib
and h323plus, I can't even get asterisk to compile chan_h323 anymore.
Perhaps something old was left over.
My .configure run shows:
checking
On 18/06/10 20:22, Eddie Mikell wrote:
All:
I am using the standard voicemail in asterisk. Everything works well,
except, if a users wants to record their own personal greeting, it
doesn't playback.
I can see the soundfile being created. I suspect it is a setting in the
voicemail.conf, or an
thanks for reply.
how can i give the root permission to
apache ?
sudo.
i also tried sudo .
However, without
careful configuration you will probably be giving root
access
to any process that runs as your apache user.
I've
never done it, but I'm guessing you could create a
On Monday 21 June 2010 01:16:30 Andraž wrote:
Yes, with isql it's working fine, I can see the database and all fields.
On Sun, Jun 20, 2010 at 9:11 PM, Tilghman Lesher tles...@digium.com wrote:
On Sunday 20 June 2010 13:15:11 Andraž wrote:
If I use MySQL with the same fields it's working
Now it's workin fine. It was problem with drivers, because it doesn't
support all kind of fields. I just changed from varblog to picture data type
and now it's working fine.
Tnx for help.
On Mon, Jun 21, 2010 at 2:08 PM, Tilghman Lesher tles...@digium.com wrote:
On Monday 21 June 2010 01:16:30
I saw the following lines in the log this morning. From my router logs
I see that the connection went down as my ISP was doing maintenance
for a few minutes last night. I can understand the external
registrations timing out, but why do the phones become unreachable.
They are on the internal lan
Hello
I'm learning how to work with Asterisk on an embedded system (MMU-less
Blackfin processor, 64MB RAM and 256MB NAND), and was wondering what
people use as scripting language to handle calls through the dialplan
and AGI, considering the hardware limitations?
Ideally, I'd rather use a rich
HI list-users,
Greetings!!
I have been using call file, i playback my file using *
application:playback*
and once the playback is over the call is disconnected. Is there any way it
can wait and also record the dtmf inputs once the playback is over.
Thanks in advace
Nikhil Kumar
summer
Use a context instead of the playback command. Like this
[playit]
exten = s,1,NoOp(Answer)
exten = s,n,SetMusicOnHold(default)
exten = s,n,Waitexten(5,m)
exten = s,n,Verbose(play ${ARG1})
exten = s,n,Playback(${ARG1})
So you replace playback(file) with playit(file).
_
From:
On Mon, 21 Jun 2010, Gilles wrote:
Hello
I'm learning how to work with Asterisk on an embedded system (MMU-less
Blackfin processor, 64MB RAM and 256MB NAND), and was wondering what
people use as scripting language to handle calls through the dialplan
and AGI, considering the hardware
On Mon, 21 Jun 2010, Gilles wrote:
I'm learning how to work with Asterisk on an embedded system (MMU-less
Blackfin processor, 64MB RAM and 256MB NAND), and was wondering what
people use as scripting language to handle calls through the dialplan
and AGI, considering the hardware
On Mon, 21 Jun 2010 14:06:22 +0100 (BST), Gordon Henderson
gordon+aster...@drogon.net wrote:
You could always type
asterisk blackfin
into google and see what it suggests.
Here, I'll save you the effort:
Thanks but I already know this (uCasterisk is deprecated). And can't
stand Perl ;-)
--
Even though I'm a PERL Weenie, I'll second this suggestion because you have
to have gcc present for PERL or Micro PERL.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: Monday, June 21, 2010
If you can install python or PHP in that machine (in means of
storage), you are free to run it there. 64 RAM is really enough to run
python, so you have to just try if it suits in the application. If it
takes too slow to initialize - try to find some embedded versions.
openwrt, for instance, has
Yuk!
I did manage to get it compiling again, but same error. I found an environment
variable which makes the loader tell you what it's doing, and when I load
chan_h323.so I see that it is running the init code when it segfaults. So I'm
ok up to that point...
What a mess the entire H.323 is!
On Mon, 21 Jun 2010 17:25:09 +0300, Motiejus Jaktys
desired@gmail.com wrote:
If you can install python or PHP in that machine (in means of
storage), you are free to run it there. 64 RAM is really enough to run
python, so you have to just try if it suits in the application. If it
takes too
Hey Gilles,
for whatever reason your messages appear twice twice on this list.
Philipp
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Hi Everyone,
I want to know if a specific codec type is used at least one. For example, I
want to know if out of the 100 calls on the system if there is a 1 channel
that is running G.729 codec right now. If using dial-plan and I dial in, I
can use this to obtain information about CURRENT channel.
On Mon, Jun 21, 2010 at 2:27 AM, Aksel Celasun ak...@abacus-it.no wrote:
I am using a Cisco SPA525G as a test phone, and I have the transfer
button there when I am in a call,
But when I want to transfer the current call I am in, I push the transfer
button, and onscreen I se “Enter Number”,
Hello all,
Anybody could point me any clue about an Open Source or licensed
switchboard for my users?
ARI or FOP is not enought for my users.
Thanks in advance.
VoipCrazy
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All:
Still trying to get Grandstream to play personal greetings recorded by
user - no luck. Someone mentioned the u option. What is that?
Something in voicemail.conf?
Eddie
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RTFM - B = busy, U = unavailable, S = Silent. User can record custom busy
and unavailable messages; if you use S you can just playback any kind of
message you want.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf
Eddie Mikell wrote:
user - no luck. Someone mentioned the u option. What is that?
core show application voicemail
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety.
--
Hello-
I have a system with one D410P and one B200P (both OpenVox). All is
well with the D410P, inbound and outbound, and I can initiate calls on
the B200P BRI span, but there may be something wrong with my inbound
BRI setup: there is no indication of an inbound call when I dial in to
it
Uhmmm.. remember for each channel you run perl or php interpreter so with that
amount of memory maybe this can be a problem. For that kind of project I'd use
C or java as fastagi protocol
From: desired@gmail.com
Date: Mon, 21 Jun 2010 17:25:09 +0300
To: asterisk-users@lists.digium.com
Hi,
I am using Trixbox trixbox CE 2.6.2.3 (Stable) using Asterisk 1.4.22-4
I am looking for the following functionality:
``
I receive a call from Mr. A.
I put Mr. A on hold.
I dial Mr. B
I connect Mr. A's
Hello everyone.
I am wondering whether there is a certain technique I should use to identify
all log lines in the asterisk/full logfile that are related to a single call.
If a user reports that something strange happened with a certain call, I'd like
to be able to easily go back and look at
http://downloads.polycom.com/voice/voip/sip_sw_releases_matrix.html
Howdy, all. What's the difference between split and combined
firmware, which can be seen at the above link? I've googled to no avail,
I'm afraid.
Thanks!
-Ken
--
This message has been scanned for viruses and
dangerous
At 12:27 AM 6/21/2010, you wrote:
Almost 10 seconds, before the transfer to sip
200 is made, can I reduce that timer?
And I cant see any button on the Cisco phone
which will function like a direct transfer now, do I have to wait
?
On my Aastra phones, I press Transfer 101
Transfer. So
On Mon, Jun 21, 2010 at 12:10 PM, Ken D'Ambrosio k...@jots.org wrote:
http://downloads.polycom.com/voice/voip/sip_sw_releases_matrix.htmlHowdy,
all. What's the difference between split and combined
firmware, which can be seen at the above link? I've googled to no avail,
I'm afraid.
The
From polycom web site:
PLEASE NOTE:
Combined download should be used where phones may be running pre-4.0 BootROM.
Split download file is recommended, but requires that all phones are running
BootROM 4.0 or newer.
-Original Message-
From: k...@jots.org
Sent: Mon, 21 Jun 2010
Read:
http://downloads.polycom.com/voice/voip/relnotes/spip_ssip_3_1_2_relnote
s.pdf
starting page 11/88:
1.4 Distribution Files
1.4.1 Release using individual (split) files
1.4.2 Release using Combined Image
HTH,
JDB
-Original Message-
From: asterisk-users-boun...@lists.digium.com
On Mon, Jun 21, 2010 at 10:19 AM, Warren Selby wcse...@selbytech.com wrote:
On Mon, Jun 21, 2010 at 12:10 PM, Ken D'Ambrosio k...@jots.org wrote:
Howdy, all. What's the difference between split and combined
firmware, which can be seen at the above link? I've googled to no avail,
I'm afraid.
,
recordingcheck|20100621-085328|1277132008.6419) in new stack
[Jun 21 08:53:28] VERBOSE[21559] logger.c: -- Launched AGI Script
/var/lib/asterisk/agi-bin/recordingcheck
[Jun 21 08:53:29] VERBOSE[21559] logger.c:
recordingcheck|20100621-085328|1277132008.6419: Outbound recording not enabled
:28] VERBOSE[21559] logger.c: -- Goto
(macro-record-enable,s,4)
[Jun 21 08:53:28] DEBUG[21559] app_macro.c: Executed application: GotoIf
[Jun 21 08:53:28] VERBOSE[21559] logger.c: -- Executing
[...@macro-record-enable:4] AGI(SIP/611-b7b9ae38,
recordingcheck|20100621-085328
This is a really rookie question: when should i use TE110P ISDN PRI Card?
--
Necati DEMİR
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On Mon, Jun 21, 2010 at 3:04 PM, Necati Demir nde...@demir.web.tr wrote:
This is a really rookie question: when should i use TE110P ISDN PRI Card?
--
Necati DEMİR
When you have a single PRI / BRI line you wish to terminate into an
asterisk system.
--
On Mon, Jun 21, 2010 at 03:12:40PM -0400, David Backeberg wrote:
On Mon, Jun 21, 2010 at 3:04 PM, Necati Demir nde...@demir.web.tr wrote:
This is a really rookie question: when should i use TE110P ISDN PRI Card?
When you have a single PRI / BRI line you wish to terminate into an
asterisk
El 21/06/10 14:04, Necati Demir escribió:
This is a really rookie question: when should i use TE110P ISDN PRI Card?
--
Necati DEMİR
---
When you need to...
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-- Bandwidth and
On Mon, 21 Jun 2010, Necati Demir wrote:
This is a really rookie question: when should i use TE110P ISDN PRI Card?
From an economic standpoint? When you have more than x POTS lines where x
depends on where you are in the world. Generally speaking, somewhere
around 8 to 12.
There are many
On Mon, Jun 21, 2010 at 7:32 AM, Ryan Wagoner rswago...@gmail.com wrote:
I saw the following lines in the log this morning. From my router logs
I see that the connection went down as my ISP was doing maintenance
for a few minutes last night. I can understand the external
registrations timing
And I can't see any button on the Cisco phone which will function like a
direct transfer now, do I have to wait...?
Thank you for your reply.
In my Dialplan menu on the SPA525g, I have a field where the input are, and I
must say, I don't know if this is the right one, but the field contains
Hello, and thank you for your response.
When I push transfer, the buttons with the function transfer disappears, and
then I enter the sip number,
Wait 10 seconds and then it transfers with the MOH in the background, when the
connection/channel is made,
Then transfer button is revealed again
I need to access number received after a I dial a SIP or H323 call?
suppose I get one of these:
*404 Not found
**486 Busy here
**408 Request Timeout
**480 Temporarily unavailable
**480 Temporarily unavailable
**403 Forbidden (+) **
410 Gone
**301 Moved Permanently
**410 Gone **
404 Not Found (=)
I am sure you'll have to write your own dialplan for it in asterisk.
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-06-21 12:31 PM, RSCL Mumbai rscl.mum...@gmail.com wrote:
Hi,
I am using Trixbox trixbox CE 2.6.2.3 (Stable) using Asterisk 1.4.22-4
I am looking for the following
Every call is assigned a unique SIP channel id. I usually look for this id
and then grep the log file by this id. It looks something like
SIP/201-a08rfr7... if I remember correctly.
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-06-21 1:13 PM, Douglas Mortensen d...@impalanetworks.com wrote:
Hi
I am using the AMD application in a power dialing.
All works well when I use an internal extension but when I try to use an
external number, the AMD every times returns non human status. Also the
AMDCAUSE returns Total-Time-5500. I am using the default parameters in
AMD.CONF.
Anybody has
On Mon, Jun 21, 2010 at 12:25 PM, RSCL Mumbai rscl.mum...@gmail.com wrote:
What is the simplest way to achieve this ??
Use the transfer button on your phone?
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Polybeacon | Consultant
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
blog.polybeacon.com
On Monday 21 June 2010 16:09:22 CDR wrote:
I need to access number received after a I dial a SIP or H323 call?
suppose I get one of these:
*404 Not found
**486 Busy here
**408 Request Timeout
**480 Temporarily unavailable
**480 Temporarily unavailable
**403 Forbidden (+) **
410 Gone
On Thu, Jun 3, 2010 at 6:16 AM, Gilles codecompl...@free.fr wrote:
Hello
I just read this article and would like some feedback from
experienced Asterisk users:
===
Failed open source VoIP deployment leads to hosted VoIP strategy By
Jessica Scarpati
snip
On Mon, 21 Jun 2010 16:47:08 -0700, CunningPike
cunningp...@gmail.com wrote:
Not in our experience as a 500-phone, 20-site install for a municipal
government. We are just migrating from our first generation install to
replacement hardware (to new blades from servers that are now 5 years
old) and
Sometimes you have to play some audio before calling AMD or other audio
functions for whatever reason... Like play 100ms of silence in a .wav file
immediately after answer. This causes RTP to be sent out to the carrier.
John
From: asterisk-users-boun...@lists.digium.com
I see that objective systems has updated their ooh323 stack, but it is not
compatible with the latest chan_ooh323 wrapper available on their site.
Has anyone update the chan_ooh323 wrapper for Asterisk 1.6.2.x ?
Michelle
--
_
Hi Guys,
An 8 channel Astribank is connected to Trixbox 2.8 and I ran
freepbx-module-zapauto but I get the following when running these
commands and can't make calls out:
[Trixbox]# dahdi_genconf xpporder
/usr/sbin/dahdi_genconf: warning - OLD DRIVER: missing
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