Why not write the file to /tmp using MixMonitor, then use the command
option to trigger an AGI script that will move the data into your
database then delete the original file?
John
On 24 September 2010 04:23, Govind, Mahesh (NSN - IN/Bangalore)
mahesh.gov...@nsn.com wrote:
The reason is when
Hello Community,
I need to test or simulate many calls through dahdi/wanpipe, i have a
Sangoma A108D, and i need to test the stability of the
card/drivers/firmwares with a test environment, do you think is possible?
What should i do? using some loopback cable maybe?
Thanks in advance
DD
--
Before I reinvent the wheel, I'm looking for a script then when run will
- launch tcpdump (or equivalent) on the server and capture all SIP and
UDP traffic to an IP address
- then, rather than me manually analysing with wireshark, will analyze
the cap file and produce stats on jitter, lag, delta
Hi,
I've configured a new Asterisk 1.6.1 box to replace an old Bristuffed 1.2
Asterisk.
Since then, it happens that forwarded calls are not presented the way they
used to be.
It seems that now, some endpoints are displaying the original caller id
(that's what I'm trying to achive), while some
Thanks , I was not knowing about Mix Monitor . Whether MixMonitor is faster
than record ?
Both uses same mechanism to write to the file .
Regards
Mahesh
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of ext
Hi!
traffic to an IP address - then, rather than me manually analysing with
wireshark, will analyze the cap file and produce stats on jitter, lag,
delta etc.
This is what RTCP was made for.
Philipp
--
_
-- Bandwidth and
Hi All,
Is there anyone who ever implemented successfully Fax On Demand on Asterisk
1.4.29 ?
I've tried to look from Google about this issue and could not find any
satisfying about this.
Thanks in advance for all of you who willing to help
And Sorry if there's any mistake in my
Hi DD,
We usually use loopback cables and use the open source SIP test tool
SIPp to initiate SIP calls that are sent from one group of 4 ports to
another group of 4 ports.
Met vriendelijke groet,
Ingmar Steen
Teleknowledge
Van: asterisk-users-boun...@lists.digium.com
ummm but how do you do that?
SIPp is only for SIP calls...i need to check in some way the dahdi driver, i
need in someway stress de card, is that possible? may be it has no sence at
all :(
Thanks!
2010/9/24 Ingmar Steen i.st...@teleknowledge.nl
Hi DD,
We usually use loopback cables and
i don't see any mistakes in your question.. but i still don't get it.
what do you need exactly from Fax on demand? sending faxes? receiving faxes?
From: zoelha...@yahoo.co.id
To: asterisk-users@lists.digium.com
Date: Fri, 24 Sep 2010 17:27:57 +0700
Subject: [asterisk-users] Fax On Demand -
As the previous poster said use the sip software to make test calls.
Have the number it dials go out of the sangoma card and back into
another port via a crossover cable to an extension which answers and
plays back a file for a second or so before hanging up.
You can then make lots of calls
On 10-09-23 05:40 PM, cov...@ccs.covici.com wrote:
Hi. I am having a very strange problem --aren't they all -- with the
release candidate. I have softphone which talks to asterisk from behind
nat -- the asterisk is on a public ip -- and when I hit mute on the
softphone, all rtp traffic
On 10-09-23 07:01 PM, Mike wrote:
Hi,
I have a server with multiple IP address, Asterisk binding with all of
them. I'd like Asterisk to reply to a SIP peer from the same IP address
as the peer used to register to Asterisk (as opposed to using the main
IP address all the time regardless of
Hi,
We usually stress test with asterisk using dialplans like:
[sipp]
exten = service,1,1,Dial(DAHDI/r1/12345678)
[incoming-1]
exten = 12345678,1,Dial(DAHDI/r2/12345678)
[incoming-2]
exten = 12345678,1,Answer()
exten = 12345678,n,WaitMusicOnHold(30)
exten = 12345678,n,HangUp()
cov...@ccs.covici.com writes:
Hi. I am having a very strange problem --aren't they all -- with the
release candidate. I have softphone which talks to asterisk from behind
nat -- the asterisk is on a public ip -- and when I hit mute on the
softphone, all rtp traffic ceases! Now, a version
Hi folks,
could someone please try to confirm the following (mis)behaviour of my
asterisk?
Imagine the following scenario:
Caller A calls the central.
Central picks up, talks to Caller A which wants to be connected to
employee X.
Central puts Caller A on hold by Redirecting the Channel to a
Garet,
MANY thanks my friend...can you believe that my brain was stucked :(
So simple ;)
THANKS for your valuable help!
DD
2010/9/24 Gareth Blades list-aster...@skycomuk.com
As the previous poster said use the sip software to make test calls.
Have the number it dials go out of the sangoma
Leif Madsen leif.mad...@asteriskdocs.org wrote:
On 10-09-23 05:40 PM, cov...@ccs.covici.com wrote:
Hi. I am having a very strange problem --aren't they all -- with the
release candidate. I have softphone which talks to asterisk from behind
nat -- the asterisk is on a public ip -- and
Benny Amorsen benny+use...@amorsen.dk wrote:
cov...@ccs.covici.com writes:
Hi. I am having a very strange problem --aren't they all -- with the
release candidate. I have softphone which talks to asterisk from behind
nat -- the asterisk is on a public ip -- and when I hit mute on the
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Yves A.
Sent: Friday, September 24, 2010 6:57 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Redirecting a Channel more
No ideas ?
Just give me the way if possible
Sebastien
Hi,
Excuse me if I'm late to reply but my first response has been blocked by the
Hi Tarek,
what do you need exactly from Fax on demand? sending faxes? receiving
faxes?
In simple explanation is like this, Caller goes through IVR (After having
been validated), Then Caller Choose Fax On Demand option and hang up, and
then Asterisk Send the Caller a Fax that already been
Hi All,
How can we develop a differential charging setup using asterisk like for 1st
min we charge 1 cent, for 2nd min we charge 0.5 cent, for next 30 sec charge
@15cent, etc?
Any idea, suggestion.
--
Regards,
Abdul Basit | +92 32 1416 4196
--
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Abdul Basit
Sent: Friday, September 24, 2010 8:13 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] differential billing
Hi All,
How
It has nothing to do with asterisk. A separate billing system has to
be made, where the billing / rate policies are defined.
I can help you out further, so feel free to contact me.
Regards,
Nasir.
0333-2302834
On 24-09-2010 18:13, Abdul Basit wrote:
Hi All,
How can we develop a
On Fri, Sep 24, 2010 at 9:11 AM, IMS ims77@gmail.com wrote:
No ideas ?
Just give me the way if possible
Download the latest asterisk version (1.4.36) and retry, if it fails
create a new issue on https://issues.asterisk.org
--
Paul Belanger | dCAP
Polybeacon | Consultant
Jabber:
Benny Amorsen wrote:
cov...@ccs.covici.com writes:
Hi. I am having a very strange problem --aren't they all -- with the
release candidate. I have softphone which talks to asterisk from behind
nat -- the asterisk is on a public ip -- and when I hit mute on the
softphone, all rtp traffic
Thank you Danny.
I am thinking for AMI events. Do we need some code level change?
As i want asterisk to push events to some listener rather than i ask via
AMI.
For hight call volume read from AMI may be an over head on asterisk, i
think.
On Fri, Sep 24, 2010 at 6:19 PM, Danny Nicholas
First I've tryed with the version 1.4.36
But it didn't worked so I supposed it should be ok with the last version
1.6.2... but not
= I will create a new issue for this if you think it should be. Just hope
it will not be too long to have a correction.
Thanks a lot.
Sebastien
On Fri, Sep 24,
Hi Danny,
I decided against Parking Calls, because it seemed quite complicated and
useless
for me... as far as i remember, parkedcalls return automagically after a
timeout which was not desirable.
I would have to rewrite a lot of code, if i have to change... but there
must be a reason for
Lyle Giese l...@lcrcomputer.net wrote:
Benny Amorsen wrote:
cov...@ccs.covici.com writes:
Hi. I am having a very strange problem --aren't they all -- with the
release candidate. I have softphone which talks to asterisk from behind
nat -- the asterisk is on a public ip -- and
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Abdul Basit
Sent: Friday, September 24, 2010 8:41 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] differential billing
Thank you
Hi!
Why is it a problem? It sounds like Asterisk does silence suppression.
1) With no rtp traffic, the nat device will drop the connection in it's
nat table and thus disconnecting the softphone from Asterisk. (after
the router's timeout period of course)
2) The other issue is you are
Philipp von Klitzing klitz...@pool.informatik.rwth-aachen.de wrote:
Hi!
Why is it a problem? It sounds like Asterisk does silence suppression.
1) With no rtp traffic, the nat device will drop the connection in it's
nat table and thus disconnecting the softphone from Asterisk. (after
Somehow I can't get 1.6.2.13 to compile with DEBUG_CHANNEL_LOCKS.
Downloaded latest tgz and extracted
$ ./configure
$ make menuselect
(select the needed options from compiler flags)
$ grep DEBUG_CHANNEL_LOCKS menuselect.makeopts
MENUSELECT_CFLAGS=DONT_OPTIMIZE LOADABLE_MODULES
NEWBIE alert: i'm a linux person, not an asterisk person so i'm
certainly capable of handling any linux-flavoured solution you can
suggest. here's a note i got from a local company i know (some proper
names removed):
= start =
Now and again our ISP goes down and when it does give us
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Robert P. J.
Day
Sent: Friday, September 24, 2010 9:55 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] should trixbox system hang when ISP
On Fri, Sep 24, 2010 at 9:55 AM, Robert P. J. Day rpj...@crashcourse.cawrote:
so, is there an easy fix for this? if the ISP goes down, does that
necessarily mean that trixbox has to go down as well? or should i be
asking this question on a trixbox-specific list? thanks.
rday
Try
On 24 Sep 2010, at 16:09, Danny Nicholas wrote:
The BOBW solution I would suggest is that you run your
Trixbox/Asterisk using a local DCHP provider/server so you aren't as
vulnerable to how efficient your ISP is at staying up.
DNS. Not DHCP.
S
--
Is your ISP doing DNS resolutions for you? If yes, then I also think it has
something to do with the DNS queries which hangs asterisk. But it should not
bring the server down.
On CentOS, caching name server should be very easy to install by doing:
yum install caching-nameserver
I don't remember
On Fri, 24 Sep 2010, Warren Selby wrote:
Try installing a local caching nameserver on the same box that runs
asterisk, and have that handle DNS queries for you. I remember at
one point that trixbox would hang if you had any SIP trunks
configured and you lost internet connectivity, but a
On Thu, Sep 23, 2010 at 11:23 PM, Govind, Mahesh (NSN - IN/Bangalore)
mahesh.gov...@nsn.com wrote:
The reason is when doing a load balancing , We cannot confine the
recording to a particular asterisk machine ( If we have more than one
asterisk machine in the topology ).
Yes you can. You can
A quick answer? A2billing.
It has what you call it differential billing.. but they call it progressive
billing.. 3 steps .. for 3 different rates ..
Go for it.. easy to setup and quick to learn and use.
Regards
From: asterisk-users-boun...@lists.digium.com
A quick answer? A2billing.
It has what you call it differential billing.. but they call it progressive
billing.. 3 steps .. for 3 different rates ..
Go for it.. easy to setup and quick to learn and use.
Regards
From: asterisk-users-boun...@lists.digium.com
Greetings fellow listers,
I have an application where I have
approximately 300 files that I playback individually or in blocks to
simulate text-to-speech in a less mechanical voice than normal Allison
files provide. These files are presently in GSM format and
The best format would be in whatever format asterisk is sending the
final audio out in. Even if you store it in the highest quality asterisk
may have to transcode it on the fly so its best to store it in an
already transcoded format to reduce the cpu load.
For dahdi you would want to use the
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David
Backeberg
Sent: Friday, September 24, 2010 11:28 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Record() Cmd
If your sip provider supports gsm, then it is fine to send them your
existing format, but I am sure by the time voice reaches an end user, it is
transcoded at least once or twice again, so you can never guarantee what
quality the end user is getting. I would stay with ulaw, as it has more
chances
Still I have the connection loss when internet goes down, I have to restart
the Asterisk machine or need to remove the VoIP trunk accessing internet...
DNSmasq is the only option by losing the connection when internet goes
down...is there any other way...
Thanks
On Fri, Feb 12, 2010 at 4:20 AM,
On Fri, Sep 24, 2010 at 1:32 PM, Don Kelly d...@donkelly.biz wrote:
Don sez: I don't know how to make Outlook indent. I usually top-post, but I
don't like getting yelled at.
Why do you say Don't do that? Is there a real reason that it would be bad?
Performance is a real reason. Multiple
Its a long and old thread, haven't read it all, but just to let you know
this happens when there is no reply from the DNS. So change DNS or install
it locally on your asterisk server. At least caching name server should be
installed.
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-09-24 1:51
snip
To add to this laundry list
#1. It is much simpler to get a path from a database and load that file than
to try and process a MYSQL BLOB of any size.
#2. If you should eventually leave MYSQL, blobs don't always play nicely (no
pun intended) with other DB's like PostgreSQL.
#3. You can always
I hadn't considered writing to the db real-time; was actually planning on
recording locally and moving it to the db.
Thanks for the suggestions.
--Don
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David
On Fri, Sep 24, 2010 at 10:47 AM, Daniel Tryba dan...@tryba.nl wrote:
Am I missing something?
DEBUG_THREADS
--
Paul Belanger | dCAP
Polybeacon | Consultant
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
blog.polybeacon.com
--
Hi,
I've been getting regular CPU usage spikes(50%-80%), due to asterisk
(according to top). I never noticed this on 1.4, and I have top running in
the background pretty much all the time. In between those spikes Asterisk
stays under 10% CPU usage (I have a transcoder card, which helps).
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike
Sent: Friday, September 24, 2010 2:41 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] Asterisk 1.62.13 - CPU spikes every 10
sip / other registrations...
~
Andrew lathama Latham
lath...@gmail.com
* Learn more about OSS http://en.wikipedia.org/wiki/Open-source_software
* Learn more about Linux http://en.wikipedia.org/wiki/Linux
* Learn more about Tux http://en.wikipedia.org/wiki/Tux
On Fri, Sep 24, 2010 at 3:40 PM,
Thanks guys for caring enough to write.
Danny: I did check /var/log/messages/full . Nothing out of the ordinary.
Andrew: many hundreds of SIP peers are registering every 60 seconds (and have
done so since 1.4). No problem there and it doesn't coincide with the 10 minute
spikes anyways.
Core
snip
Check out this (old) link about 1.6.1
https://issues.asterisk.org/view.php?id=16158
you might want to recreate /dev/null and /dev/random and see if that helps.
--
_
-- Bandwidth and Colocation Provided by
I found that bug before I wrote, and I was hoping you were right, but
recreating those two missing files didn't help. I wasn't running 1.6.1
anyways, but I figured I'd try.
There must be a way (Linux or Asterisk-centric) to see if a particular
thread/module is doing this?
Mike
-Original
cov...@ccs.covici.com writes:
But it surpresses in both directions! I still want to hear the other
end. For a test is there a way to turn off that feature to see if that
is the cause?
Ah, so it isn't Asterisk doing silence suppression, it's Asterisk being
unable to handle that other devices
On Thu, Sep 23, 2010 at 10:06 AM, khalid touati khalidtou...@gmail.com wrote:
do you guys know how i can
turn debug on or just know why it's not getting enabled?
Thanks a lot for your help!
Abdullah
*CLI set debug 15
*CLI reload
--
Paul Belanger | dCAP
Polybeacon | Consultant
Jabber:
On 09/24/2010 03:52 PM, Mike wrote:
I found that bug before I wrote, and I was hoping you were right, but
recreating those two missing files didn't help. I wasn't running
1.6.1 anyways, but I figured I'd try.
There must be a way (Linux or Asterisk-centric) to see if a
particular
On Thu, Sep 23, 2010 at 10:06 AM, khalid touati khalidtou...@gmail.com
wrote:
do you guys know how i can turn debug on or just know why it's not
getting enabled?
On Fri, 24 Sep 2010, Paul Belanger wrote:
*CLI set debug 15
*CLI reload
If you change these lines in the '[logfiles]'
All I have to do to make it work is to use 1.8.0 revision 281875 --
after that something is broke. I was hoping someone could look and see
what changed just after that rev and see if it makes sense.
Benny Amorsen benny+use...@amorsen.dk wrote:
cov...@ccs.covici.com writes:
But it
Thanks Shaun, but I'm not sure I understand everything you wrote...I can
understand that blaming Asterisk might be a Linux error, but it still
doesn't explain what does make the CPU usage shoot up like this.
I am using 2.6.18-194.3.1.el5 (64 bits, CentOs), Asterisk 1.6.2.13 and DAHDI
Version:
Hi Gurus,
We have configured asterisk to trunk with avaya with ooh323 channel driver. The
sip phone registered on asterisk
can dial the extensions registered on avaya via this trunk , and vice versa
works too. Even we can make the avaya branch to dial asterisk’s extension and
then this
В Чтв, 23/09/2010 в 14:21 -0500, Danny Nicholas пишет:
FWIW, the current state of Speech-to-text will let you do a 70-95% accurate
translation of
incoming voicemails depending on clarity/dialect/training. Also depends on
language of
native speakers. For 100% reliability, this still
Yes I read the one more thread
http://lists.digium.com/pipermail/asterisk-users/2010-February/244256.html
also..
Thanks for your comments...:)
On Fri, Sep 24, 2010 at 11:27 PM, Zeeshan Zakaria zisha...@gmail.comwrote:
Its a long and old thread, haven't read it all, but just to let you know
At 01:14 PM 9/23/2010, you wrote:
The Asterisk Development Team has announced the second release candidate of
Asterisk 1.8.0. This release candidate is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/
I downloaded this, ran ./configure followed by make
On Fri, Sep 24, 2010 at 10:25:01PM -0700, Ira wrote:
At 01:14 PM 9/23/2010, you wrote:
The Asterisk Development Team has announced the second release candidate of
Asterisk 1.8.0. This release candidate is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/
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