2011/11/2 giovanni.v i...@keybits.org
On 02/11/2011 18.45, Olivier wroteo:
1. As the above line comes from libpri, how can one be certain the telco
side didn't send the weird NUL byte ?
Assuming libpri debug doesn't mess nothing is quite sure the NUL come in
from the telco.
So, would
On Tue, 2011-11-01 at 12:08 -0500, Tim Nelson wrote:
Greetings-
I'm about to dive into the process of virtualizing some of my Asterisk
(primarily 1.4.x) infrastructure. In the past, when looking at virt
solutions, the primary issue preventing me from moving was the lack of proper
timing.
On 03/11/2011 9.08, Olivier wrote:
Then, how should such a valid but unuseful character, if I may rate it
as such, be handed, then ?
To me:
A. we should not expect any called number to include any character but
those in the 0 to 9 range.
B. we should notify sysadmin anytime an unuseful
Hi,
We're trying to time-limit some calls by specifying L(x:y:z) as an
option to the Dial command.
If we set the limit to a fairly short duration (eg 120 seconds) then
Asterisk seems to issue the hangup at about the right time.
However, for longish calls we're seeing quite a bit of overspill.
2011/11/3 giovanni.v i...@keybits.org
p.s.: sorry for my /Spaghetti-English-language/.
Don't worry about your english : I'm not pround of mine ;-)
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New
Please elaborate on your flavor of DAHDI and LIBPRI and what type of DAHDI
service you are using (PSTN, T1, etc). Speaking from a POTS line point of
view, there can easily be a 7-10 second delay in the processing of DAHDI
information (which would make your 1347 second call within tolerance).
Hi,
I'm still strugling with my CallerID presentation problem.
Let me remind it :
My setup is:
Alice cellphone --GSMISDN-- Asterisk -- ISDN GSM-- Bob
cellphone
Ive configured Asterisk so that whenever Bob forwards its incoming call to
its cellphone, the later phone should present
What version of Asterisk? Is the forwarding done using Followme, attended
transfer or blind transfer?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
Sent: Thursday, November 03, 2011 8:14 AM
To: Asterisk Users Mailing List -
On Thu, Nov 3, 2011 at 18:44, Danny Nicholas da...@debsinc.com wrote:
Please elaborate on your flavor of DAHDI and LIBPRI and what type of
DAHDI
service you are using (PSTN, T1, etc). Speaking from a POTS line point of
view, there can easily be a 7-10 second delay in the processing of DAHDI
If you dial to a Local/Context and use your time limits on that and then do
your dial to your DAHDI device inside that context does that have any
effect on the time limits working. We have used time limits with
Local/Context dials and had them work with out any known issues.
Thanks
Bryant
+1 Bryant - by using the Local/Context you are introducing some overhead to
the process, but eliminating the dependence on DAHDI timing (not that
there's anything wrong with that per se, but you can't control the Space
Shuttle with a Bearcat Scanner (or can you?) ).
From:
2011/11/3 Danny Nicholas da...@debsinc.com
What version of Asterisk?
1.6.1.18
Is the forwarding done using Followme, attended transfer or blind
transfer?
a plain Answer plus Dial
** **
*From:* asterisk-users-boun...@lists.digium.com [mailto:
Something like this?
[callbob]
Exten = start,1,answer
Exten = start,n,Dial(DAHDI/1/5551212,30)
If that is the case, Bob should always get the Caller ID of your asterisk
installation - I would suggest this instead
[callbob]
Exten = start,1,answer
Exten = start,n,Set(CALLERID(num)=${EXTEN})
In your example the CallerID number will always be start. Not what he is
looking for.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Thursday, November 03, 2011 9:38 AM
To: 'Asterisk
Trying to save a few keystrokes - better example
[callbob]
Exten = _XX.,1,answer
Exten = _XX.,n,Dial(DAHDI/1/5551212,30)
If that is the case, Bob should always get the Caller ID of your asterisk
installation - I would suggest this instead
[callbob]
Exten = _XX.,1,answer
Exten =
Anyone?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Henry
Dogger
Sent: dinsdag 1 november 2011 13:00
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] bug in queuemanager?
Sorry it took
Hi list,
Could anyone tell me what is the recommended hardware to a system for
following configuration:
SBC -- Asterisk (SS) -- Carrier GW
Asterisk should work as a Class 4 SoftSwitch, with following functionalists:
- Do the IP Authentication
- All communications on RTP/G729 (no transcoding
Shouldn't you be using a Proxy?
Nick.
On Thu, Nov 3, 2011 at 1:04 PM, Sunny no7f...@gmail.com wrote:
Hi list,
Could anyone tell me what is the recommended hardware to a system for
following configuration:
SBC -- Asterisk (SS) -- Carrier GW
Asterisk should work as a Class 4 SoftSwitch, with
I was thinking in Kamailio, but this sip proxy handles only the
SIP signalling traffic, no media processing.
On 3 November 2011 17:07, Nick Khamis sym...@gmail.com wrote:
Shouldn't you be using a Proxy?
Nick.
On Thu, Nov 3, 2011 at 1:04 PM, Sunny no7f...@gmail.com wrote:
Hi list,
Could
Sunny-
I was thinking in Kamailio, but this sip proxy handles only the
SIP signalling traffic, no media processing.
Kamailio + rtpproxy.
-Jeff
On 3 November 2011 17:07, Nick Khamis sym...@gmail.com wrote:
Shouldn't you be using a Proxy?
Nick.
On Thu, Nov 3, 2011 at 1:04 PM, Sunny
if i run let's say
1 pbx running on my main linux box
and a another on my windows box
if a person dial my main number and press lets say 1
are it possible to transfer the call over to my other pbx
hope anyone understand--
_
--
Yes. If you have two asterisk boxes running you can trunk them together and
place calls from one to to the other.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Nov 3, 2011, at 11:36 AM, mattias wrote:
if i run let's say
1 pbx running on my main linux box
and a
ok
so if i have a automatic phonesystem on the first i can e.g
press 1 forpersonal service
sounds cool
- Original Message -
From: Jim Dickenson
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Thursday, November 03, 2011 8:40 PM
Subject: Re: [asterisk-users] 2
Hello Everyone,
Unlike going through DIDx, DIDLogic etc.., we have an option of
getting DIDs directly
from local telco Bell Canada. Currently our SIP Trunk provider
assigned a DID to us,
and as you know, they just redirect requests it to our PBX.
However, when dealing directly with a telco, what
Hi asterisk users, can any recommend me a live queue monitor for asterisk
queues?
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New to Asterisk? Join us for a live introductory webinar every Thurs:
Hi all, can any tell me why queue status don't shows the statistics or
number of calls with exitwithkey status?
it only shows the number of waiting, completed and abandoned calls in a
queue.
Regards,
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On 11/03/2011 07:20 PM, Nick Khamis wrote:
Hello Everyone,
Unlike going through DIDx, DIDLogic etc.., we have an option of
getting DIDs directly
from local telco Bell Canada. Currently our SIP Trunk provider
assigned a DID to us,
and as you know, they just redirect requests it to our PBX.
Hello James,
Thank you so much for your response. We just purchased an AudioCodes
MP124 for this job. And setting
up OpenSIPS as the proxy. As I mentioned earlier, Bell Canada is the
Telco here in Toronto. As for other
Telcos around the world, for example Bell South in the states, is it
possible
On 11/03/2011 09:16 PM, Nick Khamis wrote:
Hello James,
Thank you so much for your response. We just purchased an AudioCodes
MP124 for this job. And setting
up OpenSIPS as the proxy. As I mentioned earlier, Bell Canada is the
Telco here in Toronto. As for other
Telcos around the world, for
Fair enough,
In regards to the the diagram discussed earlier:
Telco Lines - Gateway T1 - SIP Proxy - Media Servers - Customer
I understand that a T1 Gateway that has 480 channels, can handle up to
240 calls.
That is more than enough for the Gateway T1 - SIP Proxy part of
the diagram. I just
on 10/31/2011 11:59 PM Thanasis wrote the following:
I need your help in implementing the following scenario:
A certain extension will ring two sip phones simultaneously and when one
of them answers, the other keeps ringing until it answers too, and then
all three (the caller and the other
The mp124 is a analog gateway and doesn't support t1's I think
A T1 is a digital line which has 24 channels per port which means 24 calls
concurrently if you want more channels you need more ports
DID's are incoming numbers the telco sends down your trunk(port) you could have
thousands of
One FXS port can only handle one call. A PRI T1 gateway can handle 23 call
channels. A single T1 Data line with SIP can handle about 18 call channels
running G711, 37 channels running g729
Thanks
Bryant Zimmerman (ZK Tech Inc.)
616-855-1030 Ext. 2003
Thanks bryant ur right 23 channels I'm used to E1's where you a get 30 channels
a even number
-Original Message-
From: Bryant Zimmerman brya...@zktech.com
Sender: asterisk-users-boun...@lists.digium.com
Date: Thu, 3 Nov 2011 22:32:41
To: Asterisk Users Mailing List - Non-Commercial
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