Hi!
I've configured LDAP to read both users and extensions from LDAP server.
However, I'm experiencing problems with state tracking. Previously when
using static files, I was able to map extension number with channel
state using:
[sip_phones]
exten = 100,hint,SIP/user
exten =
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On 2010-10-14 11:29, Maciej Paszta wrote:
Hi!
I've configured LDAP to read both users and extensions from LDAP
server. However, I'm experiencing problems with state tracking.
Previously when using static files, I was able to map extension number
with channel state using:
[sip_phones]
Tanks
I want to create a channel bank with TDMoE. I have not to buy a product.
Best, Regards
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Hello community,
I have successfully set up asterisk free PBX server and I am also able to
connect to it by softphone.
Now as next step I want to extend this to PSTN ,
My Required scenario:
I need a number which will connect outside PSTN world to my PBX and by
applying extension particular
Hi Joshi,
To connect with PSTN line you need FXO / FXS card. FXO is used to connect CO
line and FXS is used to connect internal station line. With help of FXO you
can connect the outside world and with help of FXS you can connect normal
analog phones. Inspite of normal analog phones you can
Hi,I cannot get this to work..I have two application maps that call these two
macro's...transfer is done on sip phone and transfer2 is done on the incoming
dahdi linethats all workingbut the value stored in dtmf12 is never
passed into the second macro so I get in the NoOp..
So how
Hi all,
I am planning to do clustering for my company's asterisk servers. I dont
know much about it, just read some articles on the internet and learned how
to use DUNDi and some basic information about clustering.
What I need to know is:
1. can i register end user with multiple asterisk servers
Hi,
I've read in http://www.voip-info.org/wiki/view/Asterisk+codecs but I still
have questions about core show translation.
How are values replied by core show translation computed in the the first
place ?
I've got 2 machines using a 3GHz Intel CPU : 1 is a Xeon, 1 is a Pentium 4
(gathered with
or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
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From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
Sent: Thursday, October 14, 2010 10:43 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Explain core show translation
Hi,
I've
[ text with links can be found on
http://blogs.digium.com/2010/10/14/astricon-update/
]
AstriCon is less than two weeks away! If you haven’t booked your
flight to Washington DC, now’s your chance! The main hotel (the
Gaylord) is pretty booked, but that’s OK – there are still rooms a few
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
Sent: Thursday, October 14, 2010 11:12 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Default MOH not working on 1.6.1
I have a customer that has a Trendnet TEW-435BRM router which has the
bad habit of rewriting all external connections so the Asterisk server
only sees the IP address of the router itself. Up to today this has not
been a problem since all extensions are on the local network but now
they
We have a T1 of sorts, ATT ip flex reach basically voip over a t1
line i think. I will ask them and see what they say, I'm already able to
set our outgoing callerID to any number we own, just no other ones..
there some other way to handle this?
It depends on the technology and the carrier.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
Sent: Thursday, October 14, 2010 11:11 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Explain core show translation
On 10/13/10 14:52, Danny Nicholas wrote:
I think FOLLOWME is going to fix this for you
Can you elaborate please? is this a feature from our carrier? or
something that will be built into asterisk? sounds like a useful fix :)
--
Gerard Saraber
Network Admin.
Rarcoa, Inc
(630) 654-2580 x199
On 10-10-14 12:18 PM, Carlos Chavez wrote:
I have a customer that has a Trendnet TEW-435BRM router which has the
bad habit of rewriting all external connections so the Asterisk server
only sees the IP address of the router itself. Up to today this has not
been a problem since all
2010/10/14 Danny Nicholas da...@debsinc.com
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Olivier
*Sent:* Thursday, October 14, 2010 11:12 AM
*To:* Asterisk Users Mailing List -
On Thu, Oct 14, 2010 at 11:18:59AM -0500, Carlos Chavez wrote:
I opened up the ports on the router and my phone can register. The
problem is that I have no audio because Asterisk thinks that the phone
is on the internal network and does not use the NAT and externip
settings. How do you
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gerard
Sent: Thursday, October 14, 2010 11:24 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] call forwarding callerID
On 10/13/10 14:52,
On Thu, 2010-10-14 at 18:35 +0200, Daniel Tryba wrote:
On Thu, Oct 14, 2010 at 11:18:59AM -0500, Carlos Chavez wrote:
I opened up the ports on the router and my phone can register. The
problem is that I have no audio because Asterisk thinks that the phone
is on the internal network and
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To UNSUBSCRIBE or update options visit:
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You are missing the point completely. Maybe I did not explain myself
clearly. The problem is that when you connect to the server from
outside the network (Internet), Asterisk does not see the IP address of
the device, it thinks the device is connecting from the IP address of
the
- Stefan Schmidt s...@sil.at wrote:
This is not a problem with Asterisk. The router rewrites all
external
connections with its own IP so even a SSH connection will seem to
be
coming from the router (the 'w' command will say you are connected
from
the router and not from the IP
ah-ha,
thank you very much, that's what I found when googling, I'll ask my user
and see if Asterisk announcing the call is acceptable to him, if I can't
spoof the callerID.
Followme would alternatively work pretty well, press 1 to accept the
call etc. is a pretty nice feature, I'll see if that
Hi Karim,
Here you find a basic example for configuration:
http://www.voip-info.org/wiki/index.php?page=Asterisk+TDMoE
First step is to connect two asterisk using this basic configuration.
did you get that ?
There are other possibilites when using two asterisks like connecting them
using IAX
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gerard
Sent: Thursday, October 14, 2010 1:47 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] call forwarding callerID
ah-ha,
thank you very
Oh right...
MP-118
Thanks.
On 10/14/2010 03:38 PM, Bryant Zimmerman wrote:
For which device models?
From: "Mark Murawski"
markm-li...@intellasoft.net
Sent: Thursday, October 14,
Am 14.10.2010 21:06, schrieb Tim Nelson:
The TCP header is exactly what the NAT changes, no?
--Tim
to the outside yes but not inside.
for example thats how a typical nat table looks like. (its from a zyxel
adsl router with nat)
Nat session
2010/10/14 Olivier oza_4...@yahoo.fr
2010/10/14 Olivier oza_4...@yahoo.fr
Hello,
I've configured with the very same script 1 Intel Xeon and 1 Intel
Pentium4 machines.
On one MOH is working properly
On the other, I can read on console, lines such as those bellow but I
can't hear
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
Sent: Thursday, October 14, 2010 3:34 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Default MOH not working on 1.6.1
2010/10/14 Danny Nicholas da...@debsinc.com
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Olivier
*Sent:* Thursday, October 14, 2010 3:34 PM
*To:* Asterisk Users Mailing List - Non-Commercial
On Thu, Oct 14, 2010 at 3:25 PM, Mark Murawski
markm-li...@intellasoft.net wrote:
Does anyone have links to the most recent audiocodes firmware?
Why not contact Audiocodes?
--
Paul Belanger | dCAP
Polybeacon | Consultant
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
Because audiocodes does not provide support to end users and will tell
you to contact your vendor that sold you the box.
The problem is, the vendor that sold me the box is really hard to deal
with and has been brushing me off all week on getting firmware.
On 10/14/2010 05:14 PM, Paul
From: Paul Belanger paul.belan...@polybeacon.com
Sent: Thursday, October 14, 2010 5:17 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Audiocodes firmware
On Thu, Oct 14, 2010
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bryant
Zimmerman
Sent: Thursday, October 14, 2010 4:29 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Audiocodes firmware
_
On Thu, Oct 14, 2010 at 5:27 PM, Mark Murawski
markm-li...@intellasoft.net wrote:
Because audiocodes does not provide support to end users and will tell
you to contact your vendor that sold you the box.
That is ridiculous, how hard is it to provide a download link and
disclaimer about no
Here is the sip log
ns*CLI sip set debug peer hkbn2b
SIP Debugging Enabled for IP: 203.80.89.139:5060
[Oct 15 06:35:19] NOTICE[2462]: chan_sip.c:18334 handle_response_register:
Outbound Registration: Expiry for sip.voipuser.org is 120 sec (Scheduling
reregistration in 105 s)
== Using SIP RTP
On Thu, Oct 14, 2010 at 6:46 PM, asterisk asterisk aster...@ck-lee.com wrote:
Here is the sip log
487 Request Terminated, the far end is killing your session. Talk to your ITSP.
--
Paul Belanger | dCAP
Polybeacon | Consultant
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
We are being forced to move away from audiocodes ATA's because they refuse
to fix a few minor bugs unless we commit to a 1000 piece order. This is on
their 2 port ATA's. Their response to us is that ATA's are intended for
serious carriers that are using them in conjunction with their higher end
On Oct 14, 2010, at 5:40 PM, Paul Belanger paul.belan...@polybeacon.com wrote:
On Thu, Oct 14, 2010 at 5:27 PM, Mark Murawski
markm-li...@intellasoft.net wrote:
Because audiocodes does not provide support to end users and will tell
you to contact your vendor that sold you the box.
That is
Hi,
Embarrassed as I am to write this, I am hoping for some advice. One of
our very first PBX installs, now six years old, was taken advantage of
over the past few weeks. A victim of sipvicious, I assume, that managed
to guess one of the SIP passwords. 4000 calls to various middle eastern
As a practical matter, on anything that can generate endless billings, there
should be a dumb trap that compares current usage to history (last month)
and if usage exceeds 2/1 or 3/1 for instance then usage is choked or denied
enough to cause the user to complain or perhaps generate a message to
Jeff,
I suggest talking to your PSTN/VoIP provider. We had a large amount going
through TATA communications and have not accepted their word for payment
because they had a duty to not allow traffic if our credit went down to $1k
while the calls charged were actually more than that.
We have a queue that agents log into through the dial plan. Extension
Sip/101 logs in as Agent/101
We have 'ringinuse = no' in the queues.conf file.
The issue is that when Ext 101 is on a 'non queue' call (they placed a
call, someone called their DID, etc) they still receive queue calls.
Is
Crazy. What do you plan on using for an ATA now?
The problems I'm having are getting 500 "Server Internal Error" on
just about every other call placed out of this mp-118. The box has
been installed and in use for quite some time and recently started
having
What version of asterisk are you using and method are you using to login your
agents? I recently had this issue with a 1.4.33 install where the agents
logged in with agentcallbacklogin. In the end I had to move them away from
chan_agent altogether, using dynamic agents and AddQueueMember,
Warren,
I tried using AddQueueMember to add agents.
If they a user is on a call asterisk shows:
Members:
SIP/101 (dynamic) (Not in use) has taken no calls yet
No Callers
We are using 1.4.36.
What did you use to keep track of the extension state? Didn't see any
option for that at
On Thu, Oct 14, 2010 at 6:04 PM, Warren Selby wcse...@selbytech.com wrote:
What version of asterisk are you using and method are you using to login your
agents? I recently had this issue with a 1.4.33 install where the agents
logged in with agentcallbacklogin. In the end I had to move them
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