Am Dienstag, den 23.10.2007, 22:21 -0700 schrieb satish patel:
there is no special requiremnt to use g.729 but day to day my sip
client incressing thats why some time i got breaking voice or voice
quality not much better i think in LAN there is lots of brodcat on
lan
If your LAN is congested
I have 5 GXP 2000's with firmware 1.1.4.25 running with Asterisk 1.4.13.
Is anyone else getting the following error in the asterisk console:
[Oct 22 11:39:01] WARNING[7100]: rtp.c:1142 ast_rtp_read: RTP Read too short
every couple of seconds when a handset is in a call?
I didn't notice this
From time to time, various ways of connecting asterisk SIP channels to
skype has been discussed here. This Friday, one of the subjects of our
weekly Voip Users Conference will be about trying to connect our
asterisk pbx with Skype.
I have no nexus with Skype, Paypal or Ebay. In fact, (Note for
On 10/24/07, randulo [EMAIL PROTECTED] wrote:
From time to time, various ways of connecting asterisk SIP channels to
skype has been discussed here. This Friday, one of the subjects of our
weekly Voip Users Conference will be about trying to connect our
asterisk pbx with Skype.
I have no
Hello,
apologizes if the email looks too off-topic...
Last minute arrangements allowed to host one day of OpenSER Admin
Training session within VoN Fall Boston, Nov 1, 2007, course that will
cover openser and asterisk integration for basic media services. I
believe the event could bring more
SB == BerkHolz, Steven [EMAIL PROTECTED] writes:
SB [..]
SB This way I can test different versions of the features of Server2
SB (clone with different IP) without affecting production. I assume
SB that I just use an IAX or SIP trunk between the two asterisk
SB servers.
SB Does this make sense?
P == Patrick [EMAIL PROTECTED] writes:
P There is a Xen page called something like cool configurations. It
P has information how you can configure access to a PCI card. Iirc it
P is even possible to assign one PCI slot/card to one virtual client
P and another PCI slot to another virtual client.
Joseph Begumisa wrote:
Has anyone had any compatibility issues with a TE110P card installed
on a Dell Poweredge 1950? I noted the following error on the LCD
display of the Dell Poweredge 1950:
E1711 PCI PErr Slot 1 E171F PCIE Fatal Error B0 D4 F0.
The Dell hardware owners manual
Hi
I have a situation where I want to be able to count how many times a
caller goes round a loop of Please hold..., please continue to hold.
I have found an example on voip-info but I can't get it to work. Not
sure if I've got some syntax wrong somewhere? All that happens at the
moment, is I
Hi
I believe that
exten = s,7,GotoIf($[${trips}=4]?,8)
the , should be :
On 10/24/07, Phil Knighton [EMAIL PROTECTED] wrote:
Hi
I have a situation where I want to be able to count how many times a
caller goes round a loop of Please hold..., please continue to hold. I
have found an
Thomas Kenyon wrote:
I have 5 GXP 2000's with firmware 1.1.4.25 running with Asterisk 1.4.13.
Is anyone else getting the following error in the asterisk console:
[Oct 22 11:39:01] WARNING[7100]: rtp.c:1142 ast_rtp_read: RTP Read too short
every couple of seconds when a handset is in a
Dear Alex;
Thanks for your great help and nice replies.
I would like to confirm that I understood your request
very well, so please advise me for the following:
1) If no need for registering asterisk with the
softswitch, then no need to use register = but we
will configure the section with
Hi,
I wanted to know if anyone has experience in integration asterisk with
IBM Sametime server (by implementing TCSPI).
Any pointers for this would be very helpful.
Have been reading/googling around a bit and I get to understand that the
communication between the Sametime server and Asterisk
Phil Knighton wrote:
exten = i,1,Set(trips=$[${trips} + 1])
exten = i,2,Goto(s,7)
i=invalid, t=timeout
exten = t,1,Set(trips=$[${trips} + 1])
You'll also want to initialize ${trips} with a Set(trips=0) at the beginning of
your routine.
Doug
--
Ben Franklin quote:
Those who would give
Dear guys,
many people have been using Snom with Subscription/notify lights I
tried almost every tip in the google.
But there's one thing related to the snom phones and asterisk I
didn't see in any forum
The Asterisk console show very often a message like:
fail to extend from xx to xxx
This
On 10/24/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Hi,
I wanted to know if anyone has experience in integration asterisk with IBM
Sametime server (by implementing TCSPI).
Any pointers for this would be very helpful.
Have been reading/googling around a bit and I get to understand that
Hi,
I am trying to setup a conference between Sametime users using
conferencing infrastructure of asterisk.
Sametime server has a component called TCSPI, which we can implement to
talk to any PBX, including asterisk (as per documentation). I was trying
to implement the TCSPI for Asterisk.
On 10/24/07, Steve Totaro [EMAIL PROTECTED] wrote:
Joseph Begumisa wrote:
Has anyone had any compatibility issues with a TE110P card installed
on a Dell Poweredge 1950? I noted the following error on the LCD
display of the Dell Poweredge 1950:
E1711 PCI PErr Slot 1 E171F PCIE
Hi all,
I have an internal echo problem on my LAN only. I replaced the LAN
switch with a new linksys 2024 with QOS and seemed to help but not fix
the problem. Any ideas? Here in my setup - Dell PE6400 Dual 700,
Asterisk 1.2.24/FreePBX, 2-NIC cards, one with a public ip and one with
an
randulo said on 24.10.2007 10:17:
From time to time, various ways of connecting asterisk SIP channels to
skype has been discussed here. This Friday, one of the subjects of our
weekly Voip Users Conference will be about trying to connect our
asterisk pbx with Skype.
Has anyone tried
Shaun wrote:
I've been trying to get the polycom 550 phones to show a idle display bitmap
but have not been successful. Anybody have any experience with this? The
manual gives instructions
Hi all,
After reading great things about the OSLEC Echo Canceller
(http://www.rowetel.com/ucasterisk/oslec) and seeing the reactions of
people who have tried it on a recent Trixbox thread
(http://www.trixbox.org/forums/trixbox-forums/open-discussion/need-people-echo-problems),
it sounds like
We are having an issue where DTMF is not being sent out right away and the
tone duration is inconsistent. For a test we send a '5', then a second
later we send a '9', and then five seconds later we send a '5'. If you look
at the logs below you can see the first '5' is played right away, then the
Hello,
I would like to have whisper channel spy (not private whisper) in
Asterisk 1.2. I see here:
http://www.the-asterisk-book.com/unstable/applikationen-chanspy.html
That is only available for Asterisk 1.4.
I wonder if there is any way to emulate it in Asterisk 1.2. For example,
to join two
Hello Everyone,
Can someone point me to reliable links on how to tweak Asterisk AMD
I am calling a number and have to two files to play depending if it is a
real person or an
answering machine.
Most everytime Asterisk calls it thinks it is an Answering Machine and it
starts playing
the AMD
Replies/Comments inline...
Alan Lord wrote:
Hi all,
After reading great things about the OSLEC Echo Canceller
(http://www.rowetel.com/ucasterisk/oslec) and seeing the reactions of
people who have tried it on a recent Trixbox thread
Carles Pina i Estany wrote:
Hello,
I would like to have whisper channel spy (not private whisper) in
Asterisk 1.2. I see here:
http://www.the-asterisk-book.com/unstable/applikationen-chanspy.html
That is only available for Asterisk 1.4.
I wonder if there is any way to emulate it in
Sorry, it's clear my question was far too vague.
To clarify, is there a recipe to make * record voicemail in a format
that allows playback on iPhone's media/music player playback for
voicemails that are received say, in an email message.
It seems the * voicemail defaults don't work. This
Jason,
I think there is a bit of terminology confusion here,
you can easily convert from format to another using
sox, so if your * server is going to record and email
you a voicemail file, it can surely sox the file to whatever
format the iphone needs it in and then send the email.
It
It is doable. The iPhone uses a subset of the Apple OS. Sometime ago I
reviewed the file structure of the iPhone. It is just a matter of placing
the voicemail files from * into the voicemail folder of the iPhone.
Somebody with more time than me though :)
CS
-Original Message-
Thomas Kenyon wrote:
Thomas Kenyon wrote:
I have 5 GXP 2000's with firmware 1.1.4.25 running with Asterisk 1.4.13.
Is anyone else getting the following error in the asterisk console:
[Oct 22 11:39:01] WARNING[7100]: rtp.c:1142 ast_rtp_read: RTP Read too short
every couple of seconds when
On 10/24/07, Costa Dinoteli [EMAIL PROTECTED] wrote:
Most everytime Asterisk calls it thinks it is an Answering Machine and it
starts playing
the AMD message, instead of the delivering the 1st real message
Why is it thinking that it's a machine? If you're on the console at verbose
3 or
Video on HD Voice. Worth a watch but nothing you wouldn't already know
about.
http://www.eweek.com/article2/0,1895,2193922,00.asp
My question however is this - when are ITSP's going to start offering
digital voice services with HD codecs?
It's crazy that calls to my clients via skype are
Hi all,
I need a fully developed web based remote provisioning system. I cant find
anything reliable on the internet. Have already checked ataconfig.com and
voxilla-ays.com. have tried to contact them but got no response. So if
anybody knows a good provisioning system then plz tell me about it.
What is your setup, hardware wise?
If you have the digium cards- FXO or FXS, you must make sure you tune them. I
had issues with DTMF's, when I went live with my Asterisk system. Once I tune
them, everything worked great.
Date: Wed, 24 Oct 2007 09:05:35 -0500
From: [EMAIL PROTECTED]
To:
Hi
I have an ISDN connection with 100 DIDs assigned to it...
What I'm trying to achieve is set the proper outgoing callerID while
showing the local caller's extension in the CDR.
There is a behaviour that I just can't explain.
the callerid field in sip.conf is set as :
callerid=Jean-Yves/E 300
We've experienced the same problem twice now in the past month. The
asterisk pid stops responding. We aren't able to connect to the CLI and
all calls are dropped. The lots are pretty bare as well.
This is the message log:
Oct 24 09:12:35 WARNING[20711] channel.c: Avoided initial deadlock for
Hello, I am using Asterisk SVN, a cellular phone, and chan_mobile to
run a small home PBX with two analog telephones connected to a Linksys
ATA using SIP. It works great (except for some Bluetooth adapter bugs
that I am still trying to beat...seems the misaligned audio detection
still needs work),
Hi All,
Ingnorant question, how do you apply the backport func_odbc to 1.2 branch?
Thanks.
JR
--
JR Richardson
Engineering for the Masses
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To
I guess what I'm asking is if there is a recipe anyone has used to
allow a voicemail message (once recorded by asterisk) to playback on
iPhone when said recorded voicemail is received as an email
attachment. I understand you can convert using sox, so I guess that's
the ingredient and some
Dave Fullerton wrote:
Replies/Comments inline...
Ditto :-)
Alan Lord wrote:
Hi all,
After reading great things about the OSLEC Echo Canceller
(http://www.rowetel.com/ucasterisk/oslec) and seeing the reactions of
people who have tried it on a recent Trixbox thread
On 10/24/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Hi,
I am trying to setup a conference between Sametime users using
conferencing infrastructure of asterisk.
Sametime server has a component called TCSPI, which we can implement to
talk to any PBX, including asterisk (as per
Hi Alan.
I've installed OSLEC with zaptel-1.4.5.1 applying the patches made for the 1.4
version and I have had the same problem.
Looking at the compiler options I've found that the symbol ECHO_CAN_FROMENV is
defined by default and this prevents the echo selection from zconfig.h.
I've solved
This is semi-related, but I have a Tmobile MDA and I couldn't play the
files either. The issue was not a codec issue, it was an email encoding
issue. If I sent the message to an email account and it was then
downloaded to my desktop via outlook and then forwarded on to my phone,
I can listen
Jason Lixfeld wrote:
I guess what I'm asking is if there is a recipe anyone has used to
allow a voicemail message (once recorded by asterisk) to playback on
iPhone when said recorded voicemail is received as an email
attachment. I understand you can convert using sox, so I guess that's
Can the Asterisk Manager Proxy, AstManProxy, prefix the host name that output
applies to, to the start of each line? If you are proxying multiple systems,
how can it uniquely identify the output from each system?
Thanks,
Doug.
__
Do You
It plays wav, but as far as I understand, * encodes the wav using
something like ulaw which iPhone doesn't support. If I can switch the
codec to pcm, that may work - is that possible?
On 24-Oct-07, at 12:25 PM, Darrick Hartman (lists) wrote:
Jason Lixfeld wrote:
I guess what I'm asking is
On Wednesday 24 October 2007 10:50:47 JR Richardson wrote:
Ingnorant question, how do you apply the backport func_odbc to 1.2 branch?
ASTSRC=/path/to/downloaded/asterisk/source make install
--
Tilghman
___
--Bandwidth and Colocation Provided by
Seems the answer was simple enough - set format=wav and it works
fine. Mine was set at wav49.
On 24-Oct-07, at 1:02 PM, Jason Lixfeld wrote:
It plays wav, but as far as I understand, * encodes the wav using
something like ulaw which iPhone doesn't support. If I can switch the
codec to
marcotasto wrote:
Hi Alan.
I've installed OSLEC with zaptel-1.4.5.1 applying the patches made for the
1.4 version and I have had the same problem.
Looking at the compiler options I've found that the symbol ECHO_CAN_FROMENV
is defined by default and this prevents the echo selection from
marcotasto wrote:
Hi Alan.
I've installed OSLEC with zaptel-1.4.5.1 applying the patches made for the
1.4 version and I have had the same problem.
Looking at the compiler options I've found that the symbol ECHO_CAN_FROMENV
is defined by default and this prevents the echo selection from
On Oct 24, 2007, at 12:25 PM, [EMAIL PROTECTED]
wrote:
This is semi-related, but I have a Tmobile MDA and I couldn't play the
files either. The issue was not a codec issue, it was an email
encoding
issue. If I sent the message to an email account and it was then
downloaded to my desktop
Douglas Garstang wrote:
Can the Asterisk Manager Proxy, AstManProxy, prefix the host name that output
applies to, to the start of each line? If you are proxying multiple systems,
how can it uniquely identify the output from each system?
Thanks,
Doug.
each Event block should have a
Your best bet may be to write your own. That's what we ended up doing and
it isn't that hard.
On 10/24/07, Rizwan Hisham [EMAIL PROTECTED] wrote:
Hi all,
I need a fully developed web based remote provisioning system. I cant find
anything reliable on the internet. Have already checked
We have Digium PRI cards, TE110 and TE420 (with hardware echo cancellation).
On 10/24/07, John Meksavan [EMAIL PROTECTED] wrote:
What is your setup, hardware wise?
If you have the digium cards- FXO or FXS, you must make sure you tune
them. I had issues with DTMF's, when I went live with
Thanks, just realised that...
- Original Message
From: Richard Lyman [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, October 24, 2007 10:45:25 AM
Subject: Re: [asterisk-users] AstManProxy Host Prefix?
Is it well known that setting the ActionID when connecting to AMI has
absolutely no effect?
Is this fixed in Asterisk 1.4?
If you add an ActionID to your Originate command for example, it looks like the
only events that come back with an ActionID associated are the initial
response,
Any ideas ?
Jonn
Original Message
Subject:[asterisk-users] Internal LAN echo problem
Date: Wed, 24 Oct 2007 08:34:32 -0500
From: Jonn R Taylor [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
Any echo you hear on pure IP calls is caused by the endpoint phone. You
cannot do ANYTHING about it on Asterisk.
Jonn Taylor wrote:
Any ideas ?
Jonn
Original Message
Subject: [asterisk-users] Internal LAN echo problem
Date: Wed, 24 Oct 2007 08:34:32
I would have thought an LGPL version wouldn't be out of the question.
I hope not! LGPL is perfect for library-ish FOSS. Releasing libraries
under standard GPL, while making Richard Stallman's heart go
pitter-patter, limits what they can do since they can only go into other
GPL projects.
Eric ManxPower Wieling wrote:
Any echo you hear on pure IP calls is caused by the endpoint phone. You
cannot do ANYTHING about it on Asterisk.
Jonn Taylor wrote:
Any ideas ?
Jonn
Original Message
Subject: [asterisk-users] Internal LAN echo problem
Date:
On Wed, Oct 24, 2007 at 03:03:01PM +0100, Alan Lord wrote:
Hi all,
After reading great things about the OSLEC Echo Canceller
(http://www.rowetel.com/ucasterisk/oslec) and seeing the reactions of
people who have tried it on a recent Trixbox thread
Buy a Polycom 301 off ebay and see if it echos on your LAN.
Thanks,
Steve Totaro
Jonn Taylor wrote:
Any ideas ?
Jonn
Original Message
Subject: [asterisk-users] Internal LAN echo problem
Date: Wed, 24 Oct 2007 08:34:32 -0500
From: Jonn R Taylor
On Wed, Oct 24, 2007 at 01:20:31PM -0400, Dave Fullerton wrote:
This looks like it is isolated to 1.4.5.x.
Right.
It looks like digium added
Just to set the record straight, it was me who added it, and thus caused
hte changed behaviour you noticed here. The behaviour was restored in later
Douglas Garstang wrote:
Is it well known that setting the ActionID when connecting to AMI has
absolutely no effect?
Is this fixed in Asterisk 1.4?
If you add an ActionID to your Originate command for example, it looks like
the only events that come back with an ActionID associated are
Tzafrir Cohen wrote:
On Wed, Oct 24, 2007 at 01:20:31PM -0400, Dave Fullerton wrote:
This looks like it is isolated to 1.4.5.x.
Right.
It looks like digium added
Just to set the record straight, it was me who added it, and thus caused
hte changed behaviour you noticed here. The
The bug tracker seems to be down.
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety.
___
--Bandwidth and Colocation Provided by
What would be nice if it you could specify the host per user in
astmanproy.users
Anyone interested in making the change? $$$
Doug
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Richard
Lyman
Sent: Wednesday, October 24, 2007 1:45 PM
To: Asterisk Users
I've made a change to my manager.conf file in asterisk 1.2.18
Is there a way to reload that config file from the CLI without
restarting asterisk?
Bob
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
On 15:38, Wed 24 Oct 07, Doug Lytle wrote:
The bug tracker seems to be down.
And so is the public svn and downloads.digium.com and
ftp.digium.com and the websvn.
They are working on it.
--
Michiel van Baak
[EMAIL PROTECTED]
http://michiel.vanbaak.eu
GnuPG key:
Jonn Taylor wrote:
Eric ManxPower Wieling wrote:
Any echo you hear on pure IP calls is caused by the endpoint phone. You
cannot do ANYTHING about it on Asterisk.
Jonn Taylor wrote:
Any ideas ?
Jonn
Original Message
Subject:[asterisk-users] Internal LAN
I can do it for $10,000
On 10/24/07, asterisk [EMAIL PROTECTED] wrote:
What would be nice if it you could specify the host per user in
astmanproy.users
Anyone interested in making the change? $$$
Doug
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf
On 22:06, Wed 24 Oct 07, Michiel van Baak wrote:
On 15:38, Wed 24 Oct 07, Doug Lytle wrote:
The bug tracker seems to be down.
And so is the public svn and downloads.digium.com and
ftp.digium.com and the websvn.
They are working on it.
And it's working again for me
--
Michiel van Baak
Bob Pierce wrote:
I've made a change to my manager.conf file in asterisk 1.2.18
Is there a way to reload that config file from the CLI without
restarting asterisk?
Bob
every time there is a new connection to the asterisk manager interface,
the manager.conf file is reread.
(meaning, it
I have a box with a TE210P. Things work for a while then stop when
making call files.
I get NOANSWER as the return code (right away).
I am running asterisk 1.2.12.1, libpri 1.2.3 and zap 1.2.9.1
When I try to update to newer zaptel the machine locks when loading the
zaptel drivers.
I tried to
Alan,
I'm glad to see that you are able to run zaptel and OSLEC following my tweak!
Some days ago I've sent to David Rowe a little patch that preserves the echo
cancel status between calls.
I'm using it since several weeks with my TDM400P home based PBX and I think
that's a really effective
On 10/24/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:
Jonn Taylor wrote:
Eric ManxPower Wieling wrote:
Any echo you hear on pure IP calls is caused by the endpoint phone. You
cannot do ANYTHING about it on Asterisk.
Jonn Taylor wrote:
Any ideas ?
Jonn
Calling Digium. Post your /var/log/messages and /var/log/asterisk/full
(just anything that looks relevant).
Try a Sangoma card.
Thanks,
Steve
Jerry Geis wrote:
I have a box with a TE210P. Things work for a while then stop when
making call files.
I get NOANSWER as the return code (right
On Wed, 2007-10-24 at 13:31 -0700, Richard Lyman wrote:
every time there is a new connection to the asterisk manager
interface, the manager.conf file is reread.
(meaning, it reloads itself)
Great. Thanks for your help!
___
--Bandwidth and Colocation
Let me screw this thread up by top posting now.
Could echo be caused by late packets if jitterbuffer is on or something
or would that just cause lag?
Thanks,
Steve
kevin bergner wrote:
On 10/24/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:
Jonn Taylor wrote:
Eric ManxPower
On 10/24/07, Steve Totaro [EMAIL PROTECTED] wrote:
Let me screw this thread up by top posting now.
Could echo be caused by late packets if jitterbuffer is on or something
or would that just cause lag?
Thanks,
Steve
So, does this qualify as an in-line reply, or a top post? Maybe it's a
On 10/24/07, David Gomillion [EMAIL PROTECTED] wrote:
On 10/24/07, Steve Totaro [EMAIL PROTECTED] wrote:
Let me screw this thread up by top posting now.
Could echo be caused by late packets if jitterbuffer is on or something
or would that just cause lag?
Thanks,
Steve
So,
See response in-random-lined.
David Gomillion wrote:
On 10/24/07, *David Gomillion* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
On 10/24/07, *Steve Totaro* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Let me screw this thread up by top posting now.
Convert the voicemail to a mp3 file.
As of firmware version 1.1.1, the iPhone mail application will recognize, but
not play wav attachments. But the mail application does, recognize and play
mp3 file attachments.
--
Eric Chamberlain, CISSP
Chief Technical Officer
Voxilla - http://voxilla.com/
Hi.
I'd like to get access to the CDR's generated by Asterisk (1.4) in real-time
from a remote connection coming in on TCP. Basically what I have is a
Windows application that is used to process incoming, outgoing and missed
call records putting them into a database for some analysing etc.
Hello,
I am not sure if I totally understand the question but if your looking
to stream the connection you could create a simple bash script like this
#!/bin/bash
while true; do
tail -f /var/log/asterisk/cdr-custom/Master.csv | nc -p 1024 -l
done
There probably is a better solution
I’m no expert in this field bit I would have though logging the calls to MySQL
and then queering the MySQL database would be the best not to mention the
easiest way to get the details you are looking for.
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Hass
Sent:
I tested this again, and wav files do play as attachments with firmware 1.1.1.
--
Eric Chamberlain, CISSP
Chief Technical Officer
Voxilla - http://voxilla.com/
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Eric Chamberlain
Sent:
Jason Parker wrote:
See response in-random-lined.
David Gomillion wrote:
On 10/24/07, *David Gomillion* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
On 10/24/07, *Steve Totaro* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Let me screw this thread up by top
On Wed, Oct 24, 2007 at 10:29:41PM -, [EMAIL PROTECTED] wrote:
Hi.
I'd like to get access to the CDR's generated by Asterisk (1.4) in real-time
from a remote connection coming in on TCP. Basically what I have is a
Windows application that is used to process incoming, outgoing and missed
#!/bin/bash
while true; do
tail -f /var/log/asterisk/cdr-custom/Master.csv | nc -p 1024 -l
done
Thank you John, this bash script is exactly what I was looking for. Very
simple, yet works.
As for doing this with insert into database and then polling for it... well I
don‘t like
On Wed, Oct 24, 2007 at 06:32:45PM -0500, Jonn R Taylor wrote:
Jason Parker wrote:
Will the madness never end?
Aparantly, not. The message I have quoted told me three times how to
unsubscribe from the mailing list (not counting the fourth one added to
the post by the mailman after posting).
Did you count the number of $'s? ;-)
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sean
Bright
Sent: Wednesday, October 24, 2007 4:11 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] AstManProxy
Am Mittwoch, den 24.10.2007, 11:19 -0500 schrieb Peder @
NetworkOblivion:
This is semi-related, but I have a Tmobile MDA and I couldn't play the
files either. The issue was not a codec issue, it was an email encoding
issue. If I sent the message to an email account and it was then
Anyone know what SIP phones support RTP multicast intercom or MOH.
I am working on a project that a client needs to page 150 phones at
the same time. I have clients that have 40 phones working with a
custom script that I wrote that checks to see if there on the phone
and if not puts them in a
Our testing has yielded pretty good results. We had 10 simultaneous
calls with ulaw and quality was very good. We are pure VOIP also.
How many VMs were you running at the time, and what load were they under?
We've setups running between 3 and 5 VMs on a single box (multi-core, lots of
RAM,
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Costa Dinoteli wrote:
Hello Everyone,
Can someone point me to reliable links on how to tweak Asterisk AMD
I am calling a number and have to two files to play depending if it is a
real person or an
answering machine.
Most everytime Asterisk
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Rob Schall wrote:
We've experienced the same problem twice now in the past month. The
asterisk pid stops responding. We aren't able to connect to the CLI and
all calls are dropped. The lots are pretty bare as well.
Asterisk 1.2.13-r1
Has anyone had any compatibility issues with a TE110P card installed
on a Dell Poweredge 1950? I noted the following error on the LCD
display of the Dell Poweredge 1950:
E1711 PCI PErr Slot 1 E171F PCIE Fatal Error B0 D4 F0.
Yes, I have had this problem with a dell PE1650, 1850, SC1400,
I use TE212P, it shoudl work without errors.
I use it with Asterisk 1.2.18 + zaptel-1.2.17.1
On RHEL 4.4
On Dell PowerEdge 850
It may be that the card is bad, try contacting Asterisk support.
I had one bad card when I first got it, the 2nd one worked .
--
Deepak
Jerry Geis [EMAIL
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