Hi all, Asterisk is great but I'm having issues with setting up
realtime for our call center, which is needed for login integration
with the rest of our applications (telephonists' web interface, etc.).
I have reviewed a large number of previous posts to the mailing list
and the voip-info wiki to
Hello,
I am trying to figure out which soft echo canceller I could use.
There is OSLEC, HPEC from Digium and Octware from Octasic. I have
problems to find details about their CPU needs. Can anyone share his
experience. What CPU and Memory is required for 2,4,8 and 16 channels?
Any help is
Matt Riddell wrote:
Erik Anderson wrote:
Good evening all - for the first time, I'm implementing my first-ever
queue in asterisk. Overall, it's a pretty simple setup, 4 static
members, very low call volume, etc. The one thing that has stumped me
so far, though, is the following...
This is
Hi!
I'm trying to build an HA system using heartbeat for failover.
Everything works fine with SIP, but I cannot connect my IAX phone to
the asterisk server using the managed IP address. Here is the
configuration of the server (asterisk and the IP address are up, 'ip
addr' and 'netstat'
2008/7/21 Loic Didelot [EMAIL PROTECTED]:
Hello,
I am trying to figure out which soft echo canceller I could use.
There is OSLEC, HPEC from Digium and Octware from Octasic.
I thought HPEC was licenced by Digium from Octasic (ie those 2 software are
the same).
Maybe someone should correct me
Olivier wrote:
I thought HPEC was licenced by Digium from Octasic (ie those 2 software
are the same).
Maybe someone should correct me ...
That is not correct; HPEC is a G.168 line echo canceller from Adaptive
Digital Technologies. The same algorithm (but not the same source code)
is used on
On Mon, 21 Jul 2008, Loic Didelot wrote:
Hello,
I am trying to figure out which soft echo canceller I could use.
There is OSLEC, HPEC from Digium and Octware from Octasic. I have
problems to find details about their CPU needs. Can anyone share his
experience. What CPU and Memory is required
On Mon, Jul 21, 2008 at 9:11 AM, Walter Stanish
[EMAIL PROTECTED] wrote:
[Jul 21 15:28:21] DEBUG[2028] chan_sip.c: Received REGISTER (2) -
Command in SIP REGISTER
[Jul 21 15:28:21] DEBUG[2028] chan_sip.c: SIP message could not be
handled, bad request:
Gordon Henderson wrote:
So at worst, it's saying it can handle 29 incarnations, and at best, 37 -
that's assuming no other CPU load such as transcoding.
So it's well capable of handing your requirements of 16 channels - more-so
if you're using a server class box, and not the embedded type
Kevin P. Fleming wrote:
Olivier wrote:
I thought HPEC was licenced by Digium from Octasic (ie those 2 software
are the same).
Maybe someone should correct me ...
That is not correct; HPEC is a G.168 line echo canceller from Adaptive
Digital Technologies. The same algorithm (but
Thank you for you answers.
So what tail would be suggested for
SIP - LOCAL LAN - Asterisk - ISDN/BRI ?
Is HPEC more or less resource hungry compared to OSLEC?
Best regards,
Loic Didelot.
On Mon, 2008-07-21 at 06:54 -0500, Kevin P. Fleming wrote:
Gordon Henderson wrote:
So at worst, it's
Hi all,
I have upgraded my Asterisk box from 1.2.x to 1.4.x version: it seems
that sometimes some phones become paused and cannot receive calls
anymore. I tried to set autopause = no in every section of my
queues.conf but nothing changes
Anybody knows why a phone becomes paused? Is it an
Loic Didelot wrote:
Thank you for you answers.
So what tail would be suggested for
64ms
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety.
Kevin P. Fleming wrote:
Gordon Henderson wrote:
So at worst, it's saying it can handle 29 incarnations, and at best, 37 -
that's assuming no other CPU load such as transcoding.
So it's well capable of handing your requirements of 16 channels - more-so
if you're using a server class
Hello!
I'm using Asterisk 1.4.18 (I've tried 1.4.19,1.4.21 too) and zaptel
version 1.4.11. Card is Digium Wildcard TDM800P, with driver
wctdm24xxp. From Asterisk side this card has FXS ports, and FXO from
outside. I've connect to them GSM-FXO gateway Benq C5 APC-868
(http://www.kontec.ru/c5.php).
Will Tatam wrote:
The docs state that the AGI is run when the caller is connected but this
does not appear to be true with 1.4.21.1
What I see is
1) caller enters queue
2) agent is found for call
3) agent1's call begins to ring
4) AGI is executed
5) agent does not answer the call
Giorgio Incantalupo wrote:
Hi all,
I have upgraded my Asterisk box from 1.2.x to 1.4.x version: it seems
that sometimes some phones become paused and cannot receive calls
anymore. I tried to set autopause = no in every section of my
queues.conf but nothing changes
Anybody knows why
Steve Underwood wrote:
G.168 is not an algorithm. Its a test spec. These cancellers all use
related, but different, algorithms.
Yeah, that's what I get for emailing before breakfast :-) There is a
missing 'compliant' in that sentence...
--
Kevin P. Fleming
Director of Software Technologies
Hi all,
Panic! Panic!
When I get a call over mISDN to my IAX extension and try to transfer it
to another IAX/SIP, I get this message:
Dropping incompatible voice frame on ... of format ulaw since our
native format has
changed to alaw
Immediately followed by one almost the same:
Dropping
Florian Hackenberger wrote:
Hi!
I'm trying to build an HA system using heartbeat for failover.
Everything works fine with SIP, but I cannot connect my IAX phone to
the asterisk server using the managed IP address.
I've had a similar issue with HA, although in my case SIP wouldn't
register
Hi,
I encountered something i can't understand. I've setup 2 extensions.
[100]
type=friend
host=dynamic
nat=yes
secret=100
[101]
type=friend
host=dynamic
nat=yes
secret=101
and on extensions.conf
exten = _1XX,1,Dial(SIP/${EXTEN}|30|t)
exten = _1XX,n,Hangup
This dial plan is ok, audio
Can anyone recommend decent quality as close to pay-as-you-go SIP
wholesale termination providers in both Singapore and Sydney,
Australia? I will be in both places and want a local carrier while I'm
there. It needs to be easy in and easy out and if it's not $0 base or
close I'll need to be
Hi Mark,
it is show queues I use to see if phones are paused or not. The phones
I'm using for tests are all SIP phones.
Yes, what you are supposing could be right...Asterisk could see the
phones as stuck.
I'm still investigating, making test on my 1.4 box and I have noticed
some other strange
Nhadie wrote:
Hi,
I encountered something i can't understand. I've setup 2 extensions.
[100]
type=friend
host=dynamic
nat=yes
secret=100
[101]
type=friend
host=dynamic
nat=yes
secret=101
and on extensions.conf
exten = _1XX,1,Dial(SIP/${EXTEN}|30|t)
exten = _1XX,n,Hangup
Giorgio Incantalupo wrote:
Hi Mark,
it is show queues I use to see if phones are paused or not. The phones
I'm using for tests are all SIP phones.
Yes, what you are supposing could be right...Asterisk could see the
phones as stuck.
I'm still investigating, making test on my 1.4 box and I
Hi list,
Have installed trixbox and I am working with a fxo gateway to get fxo
calls to trixbox. I am using sip to send the calls from the gateway to
trixbox. I have an extension 3000 on trixbox
on [from-sip-external] on extensions.conf ,I have put the dial plan below.
exten =
Hola,
Estoy de vacaciones hasta el 1 de Agosto.
Para dar soporte sobre la centralita de telefonia: [EMAIL PROTECTED]
Perdonen las molestias.
Ruth Llaneza Lapausa - Tecnico de VoIP.
[EMAIL PROTECTED]
Tlf: 902 199 384
Mildmac SA – www.mildmac.es – [EMAIL PROTECTED]
C/ Hnos. García Noblejas 41,
Hi
I have set up an asterisk system which allows the use of Overlap Dialing from
SIP handsets. In order to do this I had to list the various patterns of numbers
which can be dialed in the UK. We also dial with a prefix of '9' for and outside
line so much of my dialplan looks like this :-
[084x]
Hello all I am looking for a recording tool for large environment, searching
on the web I found that oreka is a great tool for this issue, anyone knows
other tool or web gui to access to asterisk recordings? Anyone have
installed successfully oreka recording tool? Thanks for any data.
Cheers!
[EMAIL PROTECTED] schrieb:
Estoy de vacaciones hasta el 1 de Agosto.
Auto-responders should not reply to messages with any of the
headers in following:
Precedence: list
or
Precedence: bulk
or
List-*
Grüße,
Philipp Kempgen
--
http://www.das-asterisk-buch.de -
Gustavo,
You may want to try out Druid (http://www.voiceroute.org) Open Source
Edition which has free recording abilities for conference, queues,
individual extensions controllable by the admin individual user.
Druid Open Source Edition is free and open source.
Ming
On Tue, Jul 22, 2008 at 12:19
I need to increase the ringing timeout on the AA50 appliance. How do I
accomplish this?
I need the phones to ring a bit more before the caller gets to the
voicemail.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon
Ask George Bush what he uses!
On Mon, Jul 21, 2008 at 11:19 PM, Gustavo A Gonzalez [EMAIL PROTECTED]
wrote:
Hello all I am looking for a recording tool for large environment,
searching on the web I found that oreka is a great tool for this issue,
anyone knows other tool or web gui to access
I am upgrading a box from 1.4 to 1.6 and my console/dsp stopped working.
I am getting a SIP/401 Unauthorized error and then a SIP/404 error.
I changed nothing in the configs.
Is there a particular parameter needed for 1.6 that 1.4 did not care about?
If I drop back to 1.4 it starts working
This is almost standard with voip calls. The echo-cancellation has to
train up to the call parameters. Some hardware is better with it than
others and you can try tweaking the value for the echo canceler up and
down. What type hardware are you using - both phone and server?
Hi,
I have Astra
Hi to All, I have a PBX (MAINPBX) from a Telecomm Provider, which have the
feature to transfer calls (Incoming call - Answer - FLASH - Dial Number
to transfer - Answer - FLASH+4) and the call is transferred, but I have
the need to implement an internal ACD using Asterisk as the PBX, the trunks
Gustavo A Gonzalez wrote:
Hello all I am looking for a recording tool for large environment,
searching on the web I found that oreka is a great tool for this issue,
anyone knows other tool or web gui to access to asterisk recordings?
Anyone have installed successfully oreka recording tool?
[Jul 21 15:28:21] DEBUG[2028] chan_sip.c: Received REGISTER (2) -
Command in SIP REGISTER
[Jul 21 15:28:21] DEBUG[2028] chan_sip.c: SIP message could not be
handled, bad request: ZjFhZjZlNmZmZjM3OWFlYzE0MGYzZDYwYzJmODAwNDg.
It looks like Asterisk is unhappy with the SIP REGISTER request
Hi,
If 't' is set on Dial command, but then i set canreinvite=yes on the account
[100]
type=friend
host=dynamic
nat=yes
secret=100
canreinvite=yes --- if i set this
would asterisk still stay in the path?
regards,
nhadie
Mark Michelson wrote:
Nhadie wrote:
Hi,
I encountered something i
Nhadie wrote:
Hi,
If 't' is set on Dial command, but then i set canreinvite=yes on the account
[100]
type=friend
host=dynamic
nat=yes
secret=100
canreinvite=yes --- if i set this
would asterisk still stay in the path?
regards,
nhadie
Yes, the Dial option will override the
Thanks Olle. How do I use it? What's the parameters???
Doug.
- Original Message
From: Johansson Olle E [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Sunday, July 20, 2008 1:36:24 AM
Subject: Re: [asterisk-users]
On Mon, Jul 21, 2008 at 01:36:10PM -0400, Alex Balashov wrote:
OrecX comes with a GUI.
Now, I won't refrain from allegations of braindeath related to its
design; it is some gargantuan JSP/servlet-driven monstrosity that could
have been reproduced in probably 50 lines of PHP or Perl. I've
Jerry Geis wrote:
I am upgrading a box from 1.4 to 1.6 and my console/dsp stopped working.
I am getting a SIP/401 Unauthorized error and then a SIP/404 error.
I changed nothing in the configs.
How are you getting SIP-related errors from Console/DSP? Posting a
console log would be most
Jay R. Ashworth wrote:
On Mon, Jul 21, 2008 at 01:36:10PM -0400, Alex Balashov wrote:
OrecX comes with a GUI.
Now, I won't refrain from allegations of braindeath related to its
design; it is some gargantuan JSP/servlet-driven monstrosity that could
have been reproduced in probably 50
ow are you getting SIP-related errors from Console/DSP? Posting a
console log would be most helpful, as many people on the mailing list
are not telepathic :-)
--
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - The Genuine Asterisk Experience (TM)
Kevin,
below is the log
Jerry Geis wrote:
�Looking for mediaport_audio_visual in smvoice-mediaport (domain
192.168.1.25)
Do you have an extension called 'mediaport_audio_visual' in a context
called 'smvoice-mediaport'? If so, can you post that context so we can
see how it looks?
--
Kevin P. Fleming
Director of
Do you have an extension called 'mediaport_audio_visual' in a context
called 'smvoice-mediaport'? If so, can you post that context so we can
see how it looks?
Kevin,
I mentioned that 1.4 works - 1.6 did not, going back to 1.4 works again.
Here are the pieces:
my sip.conf has context
Hi Joseph -
I have Astra 480i's and Snom M3's. I am using a SIP provider so I do
not have any peripheral cards.
I am on voip-wiki now reading about the echo canceller tuning, thanks!
For your particular case, you're probably not going to find much
useful info on the wiki about echo
I have a user behind a firewall who's had no issues in the past connecting
though his firewall. He's registered just fine. But when he places a call,
a large number of them have no audio on either side of the connection. No
one can hear him, he can't hear anyone as well. After a lot of poking
Hi everybody! I'm have to install some Asterisks in heavy load
scenario with a load balance schema. The question is not very
technical nor how to do it. I jut want to know if any of you have ever
done an installation like this. Let me be more precise: 10 Asterisk
servers, 2 OpenSer servers. I
I have used the OpenSer dispatcher module to load the calls (hash by
caller id) to a group of asterisk boxes (In my case, 2 servers).
The Asterisk boxes both use ARA and MySQL Master/Master replication.
In a case like yours, I think you can use MySQL cluster, and you can
still use Dispatcher to
We also have the similar setup, 2 ser server with heartbeat doing the load
balance and 4 asterisk servers handling the media. Of course the data is on
MySQL Cluster.
Jai Rangi
www.bingotelecom.com
On Mon, Jul 21, 2008 at 5:13 PM, Edgar Guadamuz [EMAIL PROTECTED] wrote:
I have used the
On Fri, 2008-07-18 at 13:02 -0400, Bill Michaelson wrote:
After much checking and puzzling, I cannot get my Polycom 601 to
toggle call recording with my Asterisk 1.4.21.1.
I can see this in the feature*.conf file set:
automon=*1
and I can see a 'Ww' in the logged/traced call to dial().
Hi,
Try to delete whole column 'md5secret' from DB peers table.
Leave only 'secret'. And try then.
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Walter Stanish
Sent: Monday, July
On Mon, 21 Jul 2008, Nicholas Blasgen wrote:
I have a user behind a firewall who's had no issues in the past connecting
though his firewall. He's registered just fine. But when he places a call,
a large number of them have no audio on either side of the connection. No
one can hear him, he
Hi list,
Have installed trixbox and I am working with a fxo gateway to get fxo
calls to trixbox. I am using sip to send the calls from the gateway to
trixbox. I have an extension 3000 on trixbox
on [from-sip-external] on extensions.conf ,I have put the dial plan below.
exten =
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