On 11 May 2016 at 10:59, Ishfaq Malik <i...@pack-net.co.uk> wrote:
>
>
> On 11 May 2016 at 10:24, Israel Gottlieb <isr...@gmail.com> wrote:
>
>>
>> Hi all
>>
>> How is avg hold time and avg talktime calculated and over long a period
>> of time
age[1].
[1] http://en.wikipedia.org/wiki/Moving_average;
If you want to find an average over a fixed period of time, your best
bet is analysing the queue log. We had to do this ourselves when
implementing a Dashboard with figures for the day.
We found the figures outputted by the queue show command
to be
I've just spotted this line in apps/app_queue.c
unsigned int relativeperiodicannounce:1;
So I'm going to assume the default is yes. Please let me know if that
assumption is wrong.
On 12 April 2016 at 16:10, Ishfaq Malik <i...@pack-net.co.uk> wrote:
> Hi
>
> Using asterisk 1.8.
Hi
Using asterisk 1.8.23.1 on CentOS6
If I do not explicitly set a value for relative-periodic-announce, what
default value will all the queues inherit?
Regards
Ish
--
Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)161 660 2350
f: +44 (0)161 660 9825
e: i...@pack
efixes such as 16, 17, 18 to indicate dialling out as different
> companies;
> strip out the prefix using ${EXTEN:2} to recover the number by skipping two
> digits from the beginning, and Set(CALLERID(num)=) as appropriate.
>
>
>
>
You can also use the A option in the Dial application
d the contact successfully
>
>
>> [2016-03-21 20:42:09] VERBOSE[28019] res_pjsip/pjsip_configuration.c:
>> Contact 17367/sip:17367@46.39.229.18:60105 is now Unreachable. RTT:
>> 0.000 msec
>>
> At the next qualify, we couldn't reach the contact
>
> This looks l
?
Thanks in Advance
Ish
--
Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)161 660 2350
f: +44 (0)161 660 9825
e: i...@pack-net.co.uk
w: http://www.pack-net.co.uk
Registered Address: PACKNET LIMITED, Duplex 2, Ducie House
37 Ducie Street
Manchester, M1 2JW
COMPANY REG
Hi
We are using asterisk 1.8.23.1 on CentOS 6
Is there a way that transferring by SIP REFER can be blocked on a call by
call basis?
Regards
Ish
--
Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)161 660 2350
f: +44 (0)161 660 9825
e: i...@pack-net.co.uk
w: http
On 30 December 2015 at 10:41, Luca Bertoncello <lucab...@lucabert.de> wrote:
> Ishfaq Malik <i...@pack-net.co.uk> schrieb:
>
> > BLF is an interaction between the phones and the server. You need to
> > configure function buttons on the phones to display the presence
risk? Join us for a live introductory webinar every Thurs:
>http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
--
Ishfaq Malik
Department
On 30 December 2015 at 10:03, Luca Bertoncello <lucab...@lucabert.de> wrote:
> Ishfaq Malik <i...@pack-net.co.uk> schrieb:
>
> Hi Ishfaq
>
> > Look into Busy Lamp Field/Presence
> >
> > Here's a starting point:
> >
> >
> http://www.asterisk
On 30 December 2015 at 10:19, Luca Bertoncello <lucab...@lucabert.de> wrote:
> Ishfaq Malik <i...@pack-net.co.uk> schrieb:
>
> > The hints have to be in the same contexts in extensions.conf as defines
> in
> > the sip.conf subscribecontext which can be set pe
On 30 December 2015 at 15:04, Luca Bertoncello <lucab...@lucabert.de> wrote:
> Patrick Laimbock <patr...@laimbock.com> schrieb:
>
> > On 12/30/15 12:24, Luca Bertoncello wrote:
> > > Ishfaq Malik <i...@pack-net.co.uk> schrieb:
> > >
> > >&g
On 30 December 2015 at 15:09, Luca Bertoncello <lucab...@lucabert.de> wrote:
> Ishfaq Malik <i...@pack-net.co.uk> schrieb:
>
> > Looks like your phones do not support it. And it is a very common
> feature.
>
> I think so...
> Maybe I can write a little
On 30 December 2015 at 15:16, Luca Bertoncello <lucab...@lucabert.de> wrote:
> Ishfaq Malik <i...@pack-net.co.uk> schrieb:
>
> > Look up fop2
>
> Thank you very much, but I prefer a standalone application, if it's
> possibile...
> Any other suggestion?
&g
> --
> _
> -- Bandwidth and Colocation Provided by <http://www.api-digital.com>
> http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
> http://www.
s for a live introductory webinar every Thurs:
> http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
https://wiki.asterisk.org/wiki/display/AST/SIP+TLS+Transpo
added via the AMI are
forgotten.
Is there any issues in trying to share a single astdb over 2 machines that
we are unaware of?
Thanks in Advance
Ish
--
Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)161 660 2350
f: +44 (0)161 660 9825
e: i...@pack-net.co.uk
w: http
ll
allow=ulaw
; Again, more simply:
;allow=!all,ulaw
; and finally instantiate a few phones
;
; [2133](natted-phone,my-codecs)
;secret = peekaboo
; [2134](natted-phone,ulaw-phone)
;secret = not_very_secret
; [2136](public-phone,ulaw-phone)
;secret = n
t the queue to avoid sending calls to members whose devices are
; known to be 'in use' (via the channel driver supporting that device state)
; uncomment this option. (Note: only the SIP channel driver currently is
able
; to report 'in use'.)
;
; ringinuse = no
Regards
Ish
--
Ishfaq Malik
Departme
to the referrer peer.
Regards
Ish
--
Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)161 660 2350
f: +44 (0)161 660 9825
e: i...@pack-net.co.uk
w: http://www.pack-net.co.uk
Registered Address: PACKNET LIMITED, Duplex 2, Ducie House
37 Ducie Street
Manchester, M1 2JW
COMPANY
for this by adding your own filter string specific
for that user. It would have the advantage of blocking further calls as
well as alerting you by email.
Regards
Ish
--
Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)161 660 2350
f: +44 (0)161 660 9825
e: i...@pack
in the
default context (see below).
Regards
Ish
--
Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)161 660 2350
f: +44 (0)161 660 9825
e: i...@pack-net.co.uk
w: http://www.pack-net.co.uk
Registered Address: PACKNET LIMITED, Duplex 2, Ducie House
37 Ducie Street
Manchester
a core show channels and grep it for the peer name.
Ish
--
Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)161 660 2350
f: +44 (0)161 660 9825
e: i...@pack-net.co.uk
w: http://www.pack-net.co.uk
Registered Address: PACKNET LIMITED, Duplex 2, Ducie House
37 Ducie Street
Hi
Can asterisk handle asterisk variable variables?
For example:
If I were to set
FOO300=BAR111
and I had something in a dialplan like:
_3XX,1,NoOp(${FOO${EXTEN}})
And the user had entered 300, it would output BAR111
We are using asterisk 1.8
Thanks in advance
Ish
--
Ishfaq Malik
Hi
Is there any way to set the presence state of a peer to in-use in asterisk
1.8?
The idea is to integrate DND buttons on phones to BLF.
Regards
--
Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)161 660 2350
f: +44 (0)161 660 9825
e: i...@pack-net.co.uk
w: http
, Jun 9, 2015 at 9:19 AM, Ishfaq Malik i...@pack-net.co.uk wrote:
Hi
Is there any way to set the presence state of a peer to in-use in
asterisk 1.8?
The idea is to integrate DND buttons on phones to BLF.
Regards
--
Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0
for people to
see as it appears to be too big to mail?
Ish
--
Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44 (0)161 660 9825
e: i...@pack-net.co.uk
w: http://www.pack-net.co.uk
Registered Address: PACKNET LIMITED, Duplex 2, Ducie House
37 Ducie Street
://lists.digium.com/mailman/listinfo/asterisk-users
Reduce the timeout in the queue configuration (but not in the Queue
application in the dialplan), when the timeout (and the retry) value has
elapsed, all available members will be rung again.
Regards
Ish
--
Ishfaq Malik
Department: VOIP Support
Company
.
Thanks,
Daniel Gonzalez
--
_
Have you set
endbeforehexten=yes
in your cdr.conf ?
--
Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44 (0)161 660 9825
e: i...@pack-net.co.uk
w
Hello people
What are the cons, if any, of enabling a jitterbuffer?
We are currently using version 1.8
Thanks in advance
Ish
--
Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44 (0)161 660 9825
e: i...@pack-net.co.uk
w: http://www.pack-net.co.uk
'.
--
Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44 (0)161 660 9825
e: i...@pack-net.co.uk
w: http://www.pack-net.co.uk
Registered Address: PACKNET LIMITED, Duplex 2, Ducie House
37 Ducie Street
Manchester, M1 2JW
COMPANY REG NO. 04920552
to read a bit more and evaluate my pcap traces and possibly ask
the router vendors.
Thank you for your efforts.
jg
Some firewalls have a 'consistent NAT' option that needs to be enabled,
otherwise you get the symptoms described.
--
Ishfaq Malik
Department: VOIP Support
Company: Packnet
, this one can be
a (mild) problem on Draytek routers and can be resolved by telnetting into
the router and using the portmaptime command.
Also, turn of stateful packet inspection if it is an option.
--
Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44 (0)161
the files to
all the servers.
--
Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44 (0)161 660 9825
e: i...@pack-net.co.uk
w: http://www.pack-net.co.uk
Registered Address: PACKNET LIMITED, Duplex 2, Ducie House
37 Ducie Street
Manchester, M1 2JW
COMPANY REG
time
DATE(NOW()) and queuename='queue_name' and event='CONNECT';
I get the vastly different figure of 92.4.
So, is the queue show figure wrong due to a bug or am I making an incorrect
assumption as to what it means?
Thanks in advance
Ish
--
Ishfaq Malik
Department: VOIP Support
Company
://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Ishfaq Malik
Hi
I'm using asterisk 1.8
Does anyone know how to use the queue reload command. The built in help
doesn't really help.
queue reload {parameters|membe Reload queues, members, queue rules, or
parameters
Regards
Ish
--
Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0
reload reception
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Ishfaq Malik
*Sent:* Thursday, January 8, 2015 2:10 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] queue reload command
On 11 November 2014 15:27, Tech Support aster...@voipbusiness.us wrote:
Unless of course the database server is not running at all for some reason.
Regards;
JVC
Surely that should be monitored by some system designed for that purpose
such as Nagios?
--
Ishfaq Malik
Department: VOIP
Kind regards,
Jonas.
Are you using mysql_realtime or odbc with a mysql back end?
--
Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44 (0)161 660 9825
e: i...@pack-net.co.uk
w: http://www.pack-net.co.uk
Registered Address: PACKNET LIMITED, Duplex 2
On 24 October 2014 16:51, Ishfaq Malik i...@pack-net.co.uk wrote:
Hi
I'm using asterisk 1.8 but I'm sure this applies to other versions.
If someone puts a call divert on a handset such as a Snom phone I get this
type of SIP message on receipt of an inbound call:
Got SIP response 302 Moved
a local channel to make the call.
Is there any way I can access that IP address inside my dialplan? I've done
a ChanDump and there's no sign of it.
Regards
Ish
--
Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44 (0)161 660 9825
e: i...@pack-net.co.uk
w
On 23 September 2014 15:04, Rusty Newton rnew...@digium.com wrote:
On Mon, Sep 22, 2014 at 9:43 AM, Ishfaq Malik i...@pack-net.co.uk wrote:
Hi Guys
I'm using asterisk 1.8.23.1
When I add a SIP Header from inside the extensions.conf
(SIPAddHeader(Alert-Info:http://www.notused.com
Ish
--
Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44 (0)161 660 9825
e: i...@pack-net.co.uk
w: http://www.pack-net.co.uk
Registered Address: PACKNET LIMITED, Duplex 2, Ducie House
37 Ducie Street
Manchester, M1 2JW
COMPANY REG NO. 04920552
:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44 (0)161 660 9825
e: i...@pack-net.co.uk
w: http://www.pack-net.co.uk
Registered Address: PACKNET LIMITED, Duplex 2, Ducie House
37 Ducie
On 28 August 2014 07:56, Leandro Dardini ldard...@gmail.com wrote:
Can you post an example?
Leandro
2014-08-28 0:47 GMT+02:00 Ishfaq Malik i...@pack-net.co.uk:
Do the pause/unpause in a Macro or Gosub and reference that from the
features.conf
Also, make sure you put the filename
-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Ishfaq Malik
Department: VOIP
: [asterisk-users] Asterisk on CentOS7
On Wed, Aug 13, 2014 at 3:10 AM, Ishfaq Malik i...@pack-net.co.uk wrote:
Hi
Is anyone using asterisk on CentOS 7?
If so, is it working fine and as expected?
Random data point: the Asterisk project's build agents are still on CentOS
6.
Your mileage may
Hi
Is anyone using asterisk on CentOS 7?
If so, is it working fine and as expected?
Regards
Ish
--
Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44 (0)161 660 9825
e: i...@pack-net.co.uk
w: http://www.pack-net.co.uk
Registered Address: PACKNET
models of Snom and Yealink phones.
--
Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44 (0)161 660 9825
e: i...@pack-net.co.uk
w: http://www.pack-net.co.uk
Registered Address: PACKNET LIMITED, Duplex 2, Ducie House
37 Ducie Street
Manchester, M1 2JW
The next time the endpoint registers it will pick up the new configuration.
On 25 July 2014 12:38, Robin Kipp mli...@robin-kipp.net wrote:
Hi Ishfaq,
Am 24.07.2014 um 09:57 schrieb Ishfaq Malik i...@pack-net.co.uk:
It supplements it.
In fact, you can define some peers in the sip.conf
in the sip.conf directly, you'll have to
do a sip reload which will clear your realtime cache.
Regards
Ish
--
Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44 (0)161 660 9825
e: i...@pack-net.co.uk
w: http://www.pack-net.co.uk
Registered Address
at contrib/realtime/mysql/sippeers.sql from the source code
shows that the encryption column is completely absent and tls is not an
option for transport.
Does this mean I can't configure a peer to use TLS and SRTP if using ARA?
Are there any workarounds?
Thanks in advance
Ish
--
Ishfaq Malik
encryption enum ('yes','no') default 'no';
On 21 July 2014 11:31, Ishfaq Malik i...@pack-net.co.uk wrote:
Hi
I'm just about to upgrade to version 1.8.29.0 and have compiled with SRTP.
However, we exclusively use the asterisk realtime architecture using the
mysql connector.
Looking
mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44 (0)161 660 9825
e: i...@pack-net.co.uk
w: http://www.pack-net.co.uk
Registered
stream of the
AMI which would show you any end point going unreachable.
Regards
Ish
--
Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44 (0)161 660 9825
e: i...@pack-net.co.uk
w: http://www.pack-net.co.uk
Registered Address: PACKNET LIMITED, Duplex 2
is
the reasoning behind it?
Thanks in advance
Ish
--
Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44 (0)161 660 9825
e: i...@pack-net.co.uk
w: http://www.pack-net.co.uk
Registered Address: PACKNET LIMITED, Duplex 2, Ducie House
37 Ducie Street
Manchester
/var/spool/asterisk
Regards
Ish
--
Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44 (0)161 660 9825
e: i...@pack-net.co.uk
w: http://www.pack-net.co.uk
Registered Address: PACKNET LIMITED, Duplex 2, Ducie House
37 Ducie Street
Manchester, M1 2JW
Ish
--
Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44 (0)161 660 9825
e: i...@pack-net.co.uk
w: http://www.pack-net.co.uk
Registered Address: PACKNET LIMITED, Duplex 2, Ducie House
37 Ducie Street
Manchester, M1 2JW
COMPANY REG NO. 04920552
Hi
Is there any harm in using res_mysql for some things and res_odbc for
others?
We already use res_mysql for ARA but could do with having CEL logged to
MySQL.
Thanks in Advance
Ish
--
Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44 (0)161 660
-boun...@lists.digium.com] *Per conto di *Ishfaq Malik
*Inviato:* martedì 10 giugno 2014 12:05
*A:* Asterisk Users Mailing List - Non-Commercial Discussion
*Oggetto:* [asterisk-users] Mixing res_mysql and res_odbc
Hi
Is there any harm in using res_mysql for some things and res_odbc
--
Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44 (0)161 660 9825
e: i...@pack-net.co.uk
w: http://www.pack-net.co.uk
Registered Address: PACKNET LIMITED, Duplex 2, Ducie House
37 Ducie Street
Manchester, M1 2JW
COMPANY REG NO. 04920552
HI there
I was wondering if anyone has implemented voicemail to text and if so, what
package is being used to do so?
Thanks in Advance
Ish
--
Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44 (0)161 660 9825
e: i...@pack-net.co.uk
w: http://www.pack
On 15 May 2014 16:04, Ishfaq Malik i...@pack-net.co.uk wrote:
On 15 May 2014 16:03, Ishfaq Malik i...@pack-net.co.uk wrote:
Hi
I'm using asterisk 1.8.25.0 on CentOS 6.
I have compiled it with all the calendar modules:
*CLI module show like calendar
Module
.
Thanks in Advance
Ish
--
Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44 (0)161 660 9825
e: i...@pack-net.co.uk
w: http://www.pack-net.co.uk
Registered Address: PACKNET LIMITED, Duplex 2, Ducie House
37 Ducie Street
Manchester, M1 2JW
COMPANY REG
On 15 May 2014 16:03, Ishfaq Malik i...@pack-net.co.uk wrote:
Hi
I'm using asterisk 1.8.25.0 on CentOS 6.
I have compiled it with all the calendar modules:
*CLI module show like calendar
Module Description
Use Count
res_calendar.soAsterisk
.
Is this discrepancy an oversight? Is it possible to set the system default
to pai but an individual peer to rpid via a realtime table?
I have tried setting the system value to pai and a single peer value to yes
but it still sent pai rather than rpid.
Thanks in Advance
Ish
--
Ishfaq Malik
Department
and that is quite messy.
Would others agree that this behaviour is incorrect? Has anyone else seen
this or be able to replicate it? Am I just missing something obvious?
Thanks in Advance
Ish
--
Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44 (0)161 660 9825
e: i
billsec.
On 2 May 2014 11:23, Ishfaq Malik i...@pack-net.co.uk wrote:
Hi
I'm using asterisk 1.8.23.1 but I've seen this same issue in previous
versions of 1.8. I have created some work arounds but the behaviour is
incorrect.
This is the scenario:
Call comes in and goes to appropriate dialplan
?
--
Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44 (0)161 660 9825
e: i...@pack-net.co.uk
w: http://www.pack-net.co.uk
Registered Address: PACKNET LIMITED, Duplex 2, Ducie House
37 Ducie Street
Manchester, M1 2JW
COMPANY REG NO. 04920552
Hi
Using asterisk 1.8
NoOp and Verbose both put messages into the logs as VERBOSE, is there any
way to put a message into the logs as NOTICE from within a dial plan?
Thanks in advance
Ish
--
Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44 (0)161
On 1 May 2014 15:19, Matthew Jordan mjor...@digium.com wrote:
On Thu, May 1, 2014 at 7:17 AM, Ishfaq Malik i...@pack-net.co.uk wrote:
Hi
I'm using asterisk 1.8.
How are channel names constructed. I always thought they were
technology/peer-hex counter
but I've had a lot of instances
That works a treat, thank you.
On 1 May 2014 15:28, Steven Wheeler swhee...@usinternet.com wrote:
On Thu, May 1, 2014 at 8:37 AM, Ishfaq Malik i...@pack-net.co.uk
wrote:
Hi
Using asterisk 1.8
NoOp and Verbose both put messages into the logs as VERBOSE, is there any
way to put
. Thanks!
Tony
--
Tony Mountifield
Work: t...@softins.co.uk - http://www.softins.co.uk
Play: t...@mountifield.org - http://tony.mountifield.org
Just about every SIP ALG (Watchguard included) makes things worse or simply
not work. Have you tried to simply disable it?
--
Ishfaq Malik
On 14 April 2014 16:34, Matthew Jordan mjor...@digium.com wrote:
On Thu, Apr 10, 2014 at 9:14 AM, Ishfaq Malik i...@pack-net.co.uk wrote:
Does anyone on this list use pyst for AMI purposes?
If so, can you point me in the direction of some simple examples. There
seems to be none anywhere
to define the host and port
address in your peer config and then secure it with ACL.
Regards
Ish
--
Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44 (0)161 660 9825
e: i...@pack-net.co.uk
w: http://www.pack-net.co.uk
Registered Address: PACKNET LIMITED
Does anyone on this list use pyst for AMI purposes?
If so, can you point me in the direction of some simple examples. There
seems to be none anywhere online. Probably doesn't help that I'm not that
experienced at python but not insurmountably so.
Thanks in Advance
Ish
--
Ishfaq Malik
network always connect from the same
IP addresses? If so, you can just lock down your SIP port to those 7 IPs
explicitly in your IPTables configuration.
Another option would be to change which port you're running SIP on.
--
Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845
.
On Fri, Apr 4, 2014 at 8:03 AM, Ishfaq Malik i...@pack-net.co.uk wrote:
On 4 April 2014 15:22, motty cruz motty.c...@gmail.com wrote:
thank you all for your support. I am using Linux, I only have about 7
users outside our home network. I will learn fail2ban and will use it
accordingly.
again
://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44 (0)161 660 9825
e: i...@pack-net.co.uk
w
Hi people
Just having a quick check to see if anyone is using any AMI proxies and
which are the most popular. For our purposes it must be able to connect to
multiple asterisk instances.
Thanks for the help.
Ish
--
Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845
might be helpful
https://www.youtube.com/watch?v=GHFduPTNE1Qindex=9list=PLighc-2vlRgSwgJCxEh6NZwC8lE6XogaP
Not sure it's as detailed as you'd like though.
Regards
Ish
--
Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44 (0)161 660 9825
e: i...@pack
with a
dynamic host. This is no in between to the best of my knowledge.
Regards
Ish
--
Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44 (0)161 660 9825
e: i...@pack-net.co.uk
w: http://www.pack-net.co.uk
Registered Address: PACKNET LIMITED, Duplex 2, Ducie
the error.
On 16 March 2011 18:19, Tilghman Lesher tilgh...@meg.abyt.es wrote:
On Wednesday 16 March 2011 06:09:33 Ishfaq Malik wrote:
Does anyone know what this error is about?
I've had 0 success in trying to find any reference to it on the internet
Well, the most obvious problem is that you
:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44 (0)161 660 9825
e: i
for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845
://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44 (0)161 660 9825
e: i...@pack-net.co.uk
w: http
Hi
Is there any way to change the preferred audio playback format in asterisk
(I'm using 1.8.25.0)
i.e. first check for gsm, if doesn't exits then check for slin?
Regards
Ish
--
Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44 (0)161 660 9825
e: i
:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44 (0)161 660 9825
e: i...@pack
/voicemail/dialplan configuration.
https://wiki.asterisk.org/wiki/display/AST/Realtime+Database+Configuration
--
Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44 (0)161 660 9825
e: i...@pack-net.co.uk
w: http://www.pack-net.co.uk
Registered Address
Hi people
I'm just mailing to see what people are using for CTI solutions with
asterisk. Aslos, has anyone managed to integrate asterisk with Salesforce?
Thanks in Advance
Ish
--
Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44 (0)161 660 9825
e: i
=force_rport,comedia
Have you added directmedia=no?
Ish
--
Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44 (0)161 660 9825
e: i...@pack-net.co.uk
w: http://www.pack-net.co.uk
Registered Address: PACKNET LIMITED, Duplex 2, Ducie House
37 Ducie Street
to that part yet.
Thank you for your patience, I am looking forward to your feedback,
Alex
You could create your own feature in features.conf that executes a
Macro/Gosub defined in sip.conf...
Ish
--
Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44
show peer peer-name load.
Has anyone got any experience of connecting to Lync using ARA?
Thanks in advance
Ish
--
Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44 (0)161 660 9825
e: i...@pack-net.co.uk
w: http://www.pack-net.co.uk
Registered Address
-
From: asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] On Behalf Of Ishfaq Malik
Sent: Thursday, December 05, 2013 9:21 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Lync and Asterisk Realtime Architecture
not been able to correlate this
happening with any other events that are going on at the time.
Can anyone think of any reason why doing the asterisk -rx command might not
disconnect cleanly?
Thanks in Advance
Ish
--
Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845
On 7 November 2013 15:26, Gareth Blades mailinglist+aster...@dns99.co.ukwrote:
On 07/11/13 11:20, Ishfaq Malik wrote:
Hi
We are using asterisk 1.8.23.1
We have a script that runs on a minute cron which polls the asterisk
server for 3 bits of information by using
asterisk -rx 'command
else noticed this phenomenon?
Thanks in Advance
Ish
--
Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44 (0)161 660 9825
e: i...@pack-net.co.uk
w: http://www.pack-net.co.uk
Registered Address: PACKNET LIMITED, Duplex 2, Ducie House
37 Ducie Street
Hi
Thanks for the quick response. I'll read all the change logs from now on, I
promise!
Ish
On 4 November 2013 15:29, Joshua Colp jc...@digium.com wrote:
Ishfaq Malik wrote:
Hi
Ever since we upgraded our asterisk servers to 1.8.23.1, we no longer
get the 'no matching peer' error when we
1 - 100 of 484 matches
Mail list logo
