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/ast
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 "mak
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 wrote:
> Its a long and old thread, haven't read it all, but just to let you know
> this happens when th
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 sti
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 extensi
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: 2.3
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 wrote:
> cov...@ccs.covici.com writes:
>
> > But it surpresses in both directions
> On Thu, Sep 23, 2010 at 10:06 AM, khalid touati
> 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]' section of logger.c
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 th
On Thu, Sep 23, 2010 at 10:06 AM, khalid touati 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: paul.belan...@polybeaco
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 devic
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
> -Origin
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 http://www.api-digita
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 s
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,
_
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 minutes
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).
I
On Fri, Sep 24, 2010 at 10:47 AM, Daniel Tryba wrote:
> Am I missing something?
>
DEBUG_THREADS
--
Paul Belanger | dCAP
Polybeacon | Consultant
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
blog.polybeacon.com
--
_
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
B
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 use S
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 PM
On Fri, Sep 24, 2010 at 1:32 PM, Don Kelly 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 simultaneous writ
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,
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 t
-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 an
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 nat
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
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
[mailto:asterisk-users-boun...@li
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
[mailto:asterisk-users-boun...@li
On Thu, Sep 23, 2010 at 11:23 PM, Govind, Mahesh (NSN - IN/Bangalore)
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 record the file whe
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 c
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 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
--
_
On Fri, Sep 24, 2010 at 9:55 AM, Robert P. J. Day wrote:
>
> 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 installing a local
-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
dropscon
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
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 DEBUG_CHANNEL_LOCKS
Philipp von Klitzing 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
> > the router's timeout peri
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
_
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 D
Lyle Giese 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 when I h
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 th
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, 201
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 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 when I hit mute on the
>> softphone, all r
On Fri, Sep 24, 2010 at 9:11 AM, IMS 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: paul.belan...@polybeacon.co
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 differe
_
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
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
--
_
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
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
moderator
-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 th
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
Leif Madsen 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 when I hit mute on the
>
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
> 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
> anot
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
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 versio
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,Hang
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
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 ce
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 whi
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 - Aste
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
> Hi DD,
>
>
>
> We usually use loopback cables and use the open source SI
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
[mailto:asteris
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 que
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
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,
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 are
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 et
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
--
___
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)
wrote:
> The reason is when doing a load balancing
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