On 11 May 2016 at 10:59, Ishfaq Malik wrote:
>
>
> On 11 May 2016 at 10:24, Israel Gottlieb wrote:
>
>>
>> Hi all
>>
>> How is avg hold time and avg talktime calculated and over long a period
>> of time?
>>
>> Thanks,
>> Israel
>
/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 misleading.
Regar
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 wrote:
> Hi
>
> Using asterisk 1.8.23.1 on Cent
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
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Department: VOIP Support
Company: Packnet Limited
t: +44 (0)161 660 2350
f: +44 (0)161 660 9825
e: i...@pack
ialling 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 to play an audio file
to the ca
act 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 like a
unce plays?
Thanks in Advance
Ish
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Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)161 660 2350
f: +44 (0)161 660 9825
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37 Ducie Street
Manchester, M1 2JW
COMP
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
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On 30 December 2015 at 15:16, Luca Bertoncello wrote:
> Ishfaq Malik schrieb:
>
> > Look up fop2
>
> Thank you very much, but I prefer a standalone application, if it's
> possibile...
> Any other suggestion?
>
> Thanks
> Lu
On 30 December 2015 at 15:09, Luca Bertoncello wrote:
> Ishfaq Malik 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 program running on my PC to receive a message
> from
>
On 30 December 2015 at 15:04, Luca Bertoncello wrote:
> Patrick Laimbock schrieb:
>
> > On 12/30/15 12:24, Luca Bertoncello wrote:
> > > Ishfaq Malik schrieb:
> > >
> > >> Do you have a link to the user guide for your exact phone model?
> > >
On 30 December 2015 at 10:41, Luca Bertoncello wrote:
> Ishfaq Malik schrieb:
>
> > BLF is an interaction between the phones and the server. You need to
> > configure function buttons on the phones to display the presence state of
> > individual peers that have b
On 30 December 2015 at 10:19, Luca Bertoncello wrote:
> Ishfaq Malik schrieb:
>
> > The hints have to be in the same contexts in extensions.conf as defines
> in
> > the sip.conf subscribecontext which can be set per peer.
>
> Well, [anika_incoming] will be inclu
On 30 December 2015 at 10:03, Luca Bertoncello wrote:
> Ishfaq Malik schrieb:
>
> Hi Ishfaq
>
> > Look into Busy Lamp Field/Presence
> >
> > Here's a starting point:
> >
> >
> http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/
or 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
>
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n Provided by <http://www.api-digital.com>
> http://www.api-digital.com --
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https://wiki.asterisk.org/wiki/display/AST/SIP+TLS+Transport
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Ishfaq Malik
Depart
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
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Department: VOIP Support
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w: http
; 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 = not_very_secret_either
;
g 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
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Ishfaq Malik
Department:
ame.
You can use dialplan logic to check if it's a transfer. If it is, you can
send the call back to the referrer peer.
Regards
Ish
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Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
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w: http://www.pack-net.co.
ou could use fail2ban 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
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Department: VOIP Support
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f: +44 (0)161
nt to check which services you
offer everyone
; out there, by enabling them in the
default context (see below).
Regards
Ish
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Department: VOIP Support
Company: Packnet Limited
t: +44 (0)161 660 2350
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e: i...@pack-net.
gt;
>
>
>
You could do a core show channels and grep it for the peer name.
Ish
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Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)161 660 2350
f: +44 (0)161 660 9825
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Registered Address: PACKNET LIMITED,
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${EXTEN}})
And the user had entered 300, it would output BAR111
We are using asterisk 1.8
Thanks in advance
Ish
--
Ishfaq Malik
;
>
> Then you can simply hint on your device like:
>
> exten => _70X,hint,SIP/${EXTEN}&Custom:DND${EXTEN}
>
>
> On Tue, Jun 9, 2015 at 9:19 AM, Ishfaq Malik wrote:
>
>> Hi
>>
>> Is there any way to set the presence state of a peer to in-use in
>> as
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
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Department: VOIP Support
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w: http
ce. It does seem to happen at least once a day, however.
What is the best way of getting the core show locks output for people to
see as it appears to be too big to mail?
Ish
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iling list
> To UNSUBSCRIBE or update options visit:
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>
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 a
channel has been deleted.
>
> Thanks,
> Daniel Gonzalez
>
> --
> _
>
>
Have you set
endbeforehexten=yes
in your cdr.conf ?
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Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)
Hello people
What are the cons, if any, of enabling a jitterbuffer?
We are currently using version 1.8
Thanks in advance
Ish
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Company: Packnet Limited
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e: i...@pack-net.co.uk
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-pbx-in
> 172.16.1.7 5060 OK (1
> ms)
>
> 2 sip peers [Monitored: 2 online, 0 offline Unmonitored: 0 online, 0
> offline]
>
>
>
> *Is this normal behavior of SIP realtime ?*
>
>
>
> -
>
> Best re
with firewalls
(admittedly limited and once removed experience). Actually, 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.
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Ishfaq Malik
Depart
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.
lience in your shared
storage device.
Alternatively, you could use something like Puppet to deploy the files to
all the servers.
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Ishfaq Malik
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Company: Packnet Limited
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Regis
time >
DATE(NOW()) and queuename='' 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
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Department: VOIP Support
ovided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>http://www.asterisk.org/hello
>
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ame) or queue reload all
>
>
>
> for example
>
>
>
> queue 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
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
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Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
On 11 November 2014 15:27, Tech Support 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?
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Ishfaq Malik
Department: VOIP Support
C
store
> data in 1 column or in seperate columns ?
>
> Using Asterisk 1.8.12.2
>
>
>
> Kind regards,
>
> Jonas.
>
>
>
Are you using mysql_realtime or odbc with a mysql back end?
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Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 49
On 24 October 2014 16:51, Ishfaq Malik 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:
>
hich then triggers 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
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Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44 (0)161 660
On 23 September 2014 15:04, Rusty Newton wrote:
> On Mon, Sep 22, 2014 at 9:43 AM, Ishfaq Malik 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:<htt
le for it to
be expressed correctly? I'm using MySQL.
Thanks in Advance
Ish
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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
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sk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
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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
Reg
On 28 August 2014 07:56, Leandro Dardini wrote:
> Can you post an example?
>
> Leandro
>
>
> 2014-08-28 0:47 GMT+02:00 Ishfaq Malik :
>
> Do the pause/unpause in a Macro or Gosub and reference that from the
>> features.conf
>>
>> Also, make sure you pu
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>http://www.asterisk.org/hello
>
> asterisk-users m
esday, August 13, 2014 12:31 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Asterisk on CentOS7
>
> On Wed, Aug 13, 2014 at 3:10 AM, Ishfaq Malik wrote:
> > Hi
> >
> > Is anyone using asterisk on CentOS 7?
> >
>
Hi
Is anyone using asterisk on CentOS 7?
If so, is it working fine and as expected?
Regards
Ish
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Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44 (0)161 660 9825
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Registered Address: PACKNET
e Polycom phones I
> use certainly do. The Digium branded phones do as well.
>
Also certain models of Snom and Yealink phones.
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Department: VOIP Support
Company: Packnet Limited
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w: http://www.pack-net.co
the endpoint registers it will pick up the new configuration.
On 25 July 2014 12:38, Robin Kipp wrote:
> Hi Ishfaq,
>
> Am 24.07.2014 um 09:57 schrieb Ishfaq Malik :
>
>
>>
>>
> It supplements it.
>
> In fact, you can define some peers in the sip.conf and so
conf and some in the MySQL
table. However, if you do add any in the sip.conf directly, you'll have to
do a sip reload which will clear your realtime cache.
Regards
Ish
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Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
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e: i..
table sip add column encryption enum ('yes','no') default 'no';
On 21 July 2014 11:31, Ishfaq Malik 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 architectu
look 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
--
Is
;
>
> If you set qualify on your peers you could monitor the event stream of the
AMI which would show you any end point going unreachable.
Regards
Ish
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Department: VOIP Support
Company: Packnet Limited
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e: i...@pack-net.co.uk
w: http
>>>
>>>>>> 8109413462
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> --
>>>>>>
>>>>>> Regards
>>>>&
at is
the reasoning behind it?
Thanks in advance
Ish
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Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44 (0)161 660 9825
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w: http://www.pack-net.co.uk
Registered Address: PACKNET LIMITED, Duplex 2, Ducie House
37 Ducie Street
Manch
l move
all the directories that normally live under /var/spool/asterisk
Regards
Ish
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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, Du
ts.digium.com [mailto:
> asterisk-users-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
>
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
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Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44 (0)161 660
uot; "yes" \N \N \N \N \N \N \N \N \N \N \N
> \N \N \N \N \N \N "XX" \N "voipdiscount_out" \N \N \N \N \N \N \N
> \N \N \N \N \N \N \N \N
>
> I enabled also sip debug, but I don't see any attempt towards
> sip.voipaccount.com
> What am I
indication of how many calls each member has taken)
What happens when you choose rrmemory as the stratergy?
Have you read and fully understood the joinempty parameter?
Regards
Ish
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Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
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e:
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
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Department: VOIP Support
Company: Packnet Limited
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On 15 May 2014 16:04, Ishfaq Malik wrote:
>
>
> On 15 May 2014 16:03, Ishfaq Malik 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
>
On 15 May 2014 16:03, Ishfaq Malik 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.
to debug as I'm struggling
a touch right now.
Thanks in Advance
Ish
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Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
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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
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Ishfaq Malik
Depar
billsec.
On 2 May 2014 11:23, Ishfaq Malik 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 goe
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
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Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44 (0)161 6
That works a treat, thank you.
On 1 May 2014 15:28, Steven Wheeler wrote:
>On Thu, May 1, 2014 at 8:37 AM, Ishfaq Malik
> wrote:
>
> Hi
>
>
>
> Using asterisk 1.8
>
>
>
> NoOp and Verbose both put messages into the logs as VERBOSE, is there any
>
On 1 May 2014 15:19, Matthew Jordan wrote:
>
>
>
> On Thu, May 1, 2014 at 7:17 AM, Ishfaq Malik wrote:
>
>> Hi
>>
>> I'm using asterisk 1.8.
>>
>> How are channel names constructed. I always thought they were
>>
>> /-
>>
>
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
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Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44 (0)161
Hi
I'm using asterisk 1.8.
How are channel names constructed. I always thought they were
/-
but I've had a lot of instances where a channel name doesn't have the
correct peer as part of it.
Is it unwise to use channel names to extract the peers involved in a call?
--
interested to hear from you. 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 w
On 14 April 2014 16:34, Matthew Jordan wrote:
> On Thu, Apr 10, 2014 at 9:14 AM, Ishfaq Malik 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 any
ed to define the host and port
address in your peer config and then secure it with ACL.
Regards
Ish
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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
--
Is
>
>
> On Fri, Apr 4, 2014 at 8:03 AM, Ishfaq Malik wrote:
>
>>
>>
>>
>> On 4 April 2014 15:22, motty cruz wrote:
>>
>>> thank you all for your support. I am using Linux, I only have about 7
>>> users outside our home network. I will learn
of your home 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
ory 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
>
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Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44
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
cular session might be helpful
https://www.youtube.com/watch?v=GHFduPTNE1Q&index=9&list=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
ither fixed IP address or username and password 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
g the error.
On 16 March 2011 18:19, Tilghman Lesher 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 ob
idth and Colocation Provided by http://www.api-digital.com --
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y 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
>
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Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845
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
l.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
e architecture and use your favourite
database to hold peer/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
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
orking?
> We already have nat=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: P
he same machine as the Asterisk server
> itself is not possible, because both won't be able to bind to port
> 5060. My guess is that the solution is to originate a call from the
> CLI; but I haven't gotten to that part yet.
>
>
>
>
> Thank you for your patience, I a
al Message-
> 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
e sip show peer 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: PA
On 7 November 2013 15:26, Gareth Blades wrote:
> 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 usin
etely randomly, I've 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: Pack
to register
>
>
>
> -
> Regards,
> AJ Stanfield
>
> t: 0161-850-4001
> e: a...@dmcip.com
> w: http://www.dmcip.com
>
> - Original Message -
> From: "Ishfaq Malik"
> To: "Asterisk Users Mailing List - Non-Commercial Discussion" &
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 wrote:
> Ishfaq Malik wrote:
>
>> Hi
>>
>> Ever since we upgraded our asterisk servers to 1.8.23.1, we no longer
>> get the &#x
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