Hi
*MSISDN* is a number uniquely identifying a subscription in a
GSMhttp://en.wikipedia.org/wiki/GSMor a
UMTS http://en.wikipedia.org/wiki/UMTS mobile network.
On Sat, Apr 16, 2011 at 1:01 AM, Pezhman Lali l...@lopl.net wrote:
yes, ami is your unique answer.
what is msisdns ?
On Wed, Apr
I've been using Gmail and thought you might like to try it out. Here's an
invitation to create an account.
You're Invited to Gmail!
virendra bhati has invited you to open a Gmail account.
Gmail is Google's free email service, built on the idea that email can be
intuitive, efficient, and fun.
Hi,*
How can I obtain msisdns of current calls ?*
Get first 4 digits with cut function of asterisk then cross check with
database by odbc connection then you will get the MSISDN details.
MSISDN list will be provided my govt of all country. So make your local
database to cross check.
On Sat, Apr
google for adaptive cdr. in asterisk.
On Sun, Apr 17, 2011 at 3:58 AM, John Jolly jgjo...@gmail.com wrote:
I have a particular DID that when called will prompt the user to enter the
caller id that they want to be displayed followed by it prompting for the
phone number to dial. How would I go
Hello list,
I have in sip.conf :
/maxexpiry=60 ; Maximum allowed time of incoming
registrations
; and subscriptions (seconds)
minexpiry=60 ; Minimum length of
registrations/subscriptions (default 60)
defaultexpiry=120
Hi All;
I am using Asterisk for Call Center (so agents login, logout, ready, not ready,
... etc). To be able to have a good call center reporting, on what I have to
depend? On the CDR of Asterisk or there is another way?
Is there a good open source tool to be used for Asterisk call center
Hi,
this is an old outstanding problem, unfortunately I don't remember how
to walkaround it. I use asterisk 1.8.3 and I have a public IP in my
network interface. As soon as the Internet connection goes down, phones
stop working. I want to be able to use pstn, isdn and the gsm gateway
even if the
Disable DNS lookups. Chan_sip crashes asterisk if you have that enabled and
internet is offline.
Regards,
Alex
-Messaggio originale-
Da: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Per conto di Niccolò Belli
Inviato: lunedì 18 aprile 2011
On Monday 18 Apr 2011, Niccolò Belli wrote:
As soon as the Internet connection goes down, phones
stop working. I want to be able to use pstn, isdn and the gsm gateway
even if the Internet connection goes down, how can I achieve it?
You most probably are using a nameserver somewhere else on the
Could 'dnsmgr' help?
On 18.04.2011 14:30, A J Stiles wrote:
On Monday 18 Apr 2011, Niccolò Belli wrote:
As soon as the Internet connection goes down, phones
stop working. I want to be able to use pstn, isdn and the gsm gateway
even if the Internet connection goes down, how can I achieve it?
On 18 Apr 2011, at 11:06, bilal ghayyad wrote:
I am using Asterisk for Call Center (so agents login, logout, ready, not
ready, ... etc). To be able to have a good call center reporting, on what I
have to depend? On the CDR of Asterisk or there is another way?
Is there a good open source
On 18/04/11 12:16, Niccolò Belli wrote:
Hi,
this is an old outstanding problem, unfortunately I don't remember how
to walkaround it. I use asterisk 1.8.3 and I have a public IP in my
network interface. As soon as the Internet connection goes down, phones
stop working. I want to be able to use
On Mon, Apr 18, 2011 at 5:49 AM, Steven Howes steve-li...@geekinter.netwrote:
On 18 Apr 2011, at 11:06, bilal ghayyad wrote:
I am using Asterisk for Call Center (so agents login, logout, ready, not
ready, ... etc). To be able to have a good call center reporting, on what I
have to depend? On
If you want to know where I got the starting idea for the methods that I've
developed, check out
http://www.voip-info.org/wiki/view/Asterisk+queue_log+on+MySQL and look at
the section about using triggers
--
_
-- Bandwidth and
On Mon, Apr 18, 2011 at 2:16 AM, virendra bhati virbh...@gmail.com wrote:
I've been using Gmail and thought you might like to try it out. Here's an
invitation to create an account.
Just a guess, but I'm pretty sure that invitation will not work for everyone
on the list ;-)
--
On Monday 18 Apr 2011, bilal ghayyad wrote:
Hi All;
I am using Asterisk for Call Center (so agents login, logout, ready, not
ready, ... etc). To be able to have a good call center reporting, on what I
have to depend? On the CDR of Asterisk or there is another way?
Is there a good open
Anyone know a good IAX2 softphone for Windows that has g729 and it is free?
att
Eduardo--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every
On Mon, Apr 18, 2011 at 8:18 AM, A J Stiles
asterisk_l...@earthshod.co.ukwrote:
If all the details you need to compile your reports can be found in
existing
databases (Asterisk's CDR database stores the details of calls; you may
need
to get user login/out events from a separate database),
Hello list,
I've got a whole lot of these in my debug log :
[Apr 18 15:12:44] DEBUG[26973] audiohook.c: Read factory 0x1cea33a0 and
write factory 0x1cea3dd8 both fail to provide 160 samples
[Apr 18 15:12:44] DEBUG[26973] audiohook.c: Failed to get 160 samples
from read factory 0x1cea33a0
[Apr
http://lmgtfy.com/?q=audiohook.c%3A+Failed+to+get+160+samples+from+read+factory
From: Jonas Kellens
Sent: Mon 4/18/2011 9:46 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Asterisk unresponsive
Hello list,
I've got a whole lot of these in my debug
On 04/18/2011 03:58 PM, Terry Brummell wrote:
http://lmgtfy.com/?q=audiohook.c%3A+Failed+to+get+160+samples+from+read+factory
This should tell me that there are others who experience this same
problem in some kind of form and that there is no real answer to it ?
Hence why I seek for an
On Mon, Apr 18, 2011 at 4:54 AM, Jonas Kellens jonas.kell...@telenet.bewrote:
Hello list,
I have in sip.conf :
snip
So are my settings wrong ?
What does sip show settings look like from the CLI?
--
Thanks,
--Warren Selby, dCAP
http://www.selbytech.com
--
On 04/18/2011 05:33 PM, Warren Selby wrote:
On Mon, Apr 18, 2011 at 4:54 AM, Jonas Kellens
jonas.kell...@telenet.be mailto:jonas.kell...@telenet.be wrote:
Hello list,
I have in sip.conf :
snip
So are my settings wrong ?
What does sip show settings look like from the CLI?
Is there a way to place a hangup time on a dynamic Meetme conference. I am
using Page() with a Meetme conf and I have had a few instances where
someone from a wifi voip phone looses ip while doing a page and the page
never hangs up. I have to kill it. I need to somehow limit the page so
after
Quick question out of curiosity: Did you googled your problem, and read through
all the results, and made an exhaustive research on line of the error message
before you opted to post your question here?
CS
On April 18, 2011 at 10:16 AM Jonas Kellens jonas.kell...@telenet.be wrote:
On
Hey Guys!
I have a stupid question about canreinvite. We are using asterisk 1.8.3.2 as a
PBX we don't have NAT or firewall thing in between asterisk and phone. so i
should use conreinvite=no right ? what is the default value of conreinvite in
asterisk 1.8.3.2 ? i meant yes or no ?
-S
On 11-04-18 09:46 AM, Jonas Kellens wrote:
Asterisk freezed and only a reboot of the whole server fixed this. Any
command on the Asterisk CLI was not executed because Asterisk was too
busy processing all of these messages that you see in the debug log.
What is the origin of these messages ?
I ran tcpdump on version 1.6 and 1.8 and compare sip header and i found in 1.8
asterisk if you call non-exiting peer/exten its waiting for ACK packet for 100
Tying message and in 1.6 its not that why i am getting following messages
__sip_xmit: sip_xmit blah..blah
See following header of sip
Hi,
I have A101DE Sangoma Card(
http://www.sangoma.com/products/hardware_products/digital_voice_and_data_networking/a101.html
)
lspci shows as 03:04.0 Network controller: Sangoma Technologies Corp.
A200/Remora FXO/FXS Analog AFT card
[root@asterisk ~]# lspci -vvv -s 03:04.0
03:04.0 Network
When I do core show channels concise over the AMI interface
how do I specify that I want to see the actual channel number like
DAHDI/4/xxx
where 4 is the actual channel.
RIght now I am seeing DAHDI/i1/x where i1 is the span.
Thanks,
Jerry
--
On 11-04-18 02:47 PM, Jerry Geis wrote:
When I do core show channels concise over the AMI interface
how do I specify that I want to see the actual channel number like
DAHDI/4/xxx
where 4 is the actual channel.
RIght now I am seeing DAHDI/i1/x where i1 is the span.
I could have
On Mon, Apr 18, 2011 at 3:06 PM, Leif Madsen
leif.mad...@asteriskdocs.org wrote:
On 11-04-18 02:47 PM, Jerry Geis wrote:
When I do core show channels concise over the AMI interface
how do I specify that I want to see the actual channel number like
DAHDI/4/xxx
where 4 is the actual
On 4/14/11 5:03 PM, m...@tdiehl.org wrote:
Yes, I noticed that also. For some reason the latest Dahdi rpms are
sitting in
the top level dir at http://packages.asterisk.org/centos/5/current/
but they are
not signed. They need to be signed and moved into the approiate arch
directory
and the
On 11-04-18 03:06 PM, Leif Madsen wrote:
On 11-04-18 02:47 PM, Jerry Geis wrote:
When I do core show channels concise over the AMI interface
how do I specify that I want to see the actual channel number like
DAHDI/4/xxx
where 4 is the actual channel.
RIght now I am seeing DAHDI/i1/x
On Mon, Apr 18, 2011 at 11:42 PM, Kaushal Shriyan
kaushalshri...@gmail.comwrote:
Hi,
I have A101DE Sangoma Card(
http://www.sangoma.com/products/hardware_products/digital_voice_and_data_networking/a101.html
)
lspci shows as 03:04.0 Network controller: Sangoma Technologies Corp.
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