Displaying 20 results from an estimated 5000 matches similar to: "Putting a notice in the logs from the dialplan"
2014 May 01
1
Channel names
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 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?
--
Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
2014 Jul 21
1
TLS, STRP and ARA
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 at tutorials we have to set encryption=yes and transport=tls for
any peer we want encrypted traffic for.
Having a look at contrib/realtime/mysql/sippeers.sql from the source code
shows that the encryption column is
2014 May 20
2
Voicemail message to text
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: ish at pack-net.co.uk
w: http://www.pack-net.co.uk
Registered Address: PACKNET LIMITED, Duplex 2, Ducie House
37 Ducie Street
Manchester,
2013 Dec 05
1
Lync and Asterisk Realtime Architecture
Hi guys
We're using asterisk 1.8.23.1 on CentOS 5 and are trying to create a trunk
to MS Lync server.
If I create the peer in sip.conf the trunk connects with no problem.
However, we prefer to use ARA.
Whenever we define the peer in our peers table, the trunk does not work,
even if we use sip show peer <peer-name> load.
Has anyone got any experience of connecting to Lync using ARA?
2014 Jan 10
1
CTI
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: ish at pack-net.co.uk
w: http://www.pack-net.co.uk
Registered Address: PACKNET LIMITED, Duplex 2,
2014 Jun 10
1
Mixing res_mysql and res_odbc
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 9825
e: ish at pack-net.co.uk
w: http://www.pack-net.co.uk
Registered Address: PACKNET LIMITED, Duplex
2014 Oct 24
1
Call forwarding from Phones and getting the referrer IP
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 Temporarily" back from xxx.xxx.xxx.xxx:xxxxx
Which then triggers a local channel to make the call.
Is there any way I can access that IP address inside
2015 Jan 08
2
queue reload command
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)845 004 4994
f: +44 (0)161 660 9825
e: ish at pack-net.co.uk
w: http://www.pack-net.co.uk
2014 May 02
1
CDR billsec issue with calls forwarded through the Local channel
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
In the dialplan the call is forwarded to another number using a Local
channel (and using /n ) e.g.
Dial(Local/<my-number>@outbound-context/n,60)
The number is
2014 Jul 07
1
CDR dcontext not updated on FAILED and BUSY calls
Hi
We're using asterisk 1.8.23.1. Our inbound calls are routed into the
default context with explicit number matching. If found they are passed on
to a distinct context for the number being called using the Goto
application.
If the call is successful or even if it has no answer, the cdr dcontext
field has the correct second context.
However, if the call fails or is busy, and even though we
2014 May 15
1
Asterisk 1.8 and calendar intergration
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.so Asterisk Calendar integration 4
res_calendar_ews.so Asterisk MS Exchange Web Service Calenda 0
res_calendar_caldav.so
2015 Jun 09
2
Manipulate extension state in 1.8.x
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: ish at pack-net.co.uk
w: http://www.pack-net.co.uk
Registered Address: PACKNET LIMITED, Duplex 2, Ducie House
37 Ducie Street
2015 Jan 08
0
queue reload command
Hi
queue reload(queue name) or queue reload all
for example
queue reload reception
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at 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
Hi
I'm using
2013 Nov 04
1
No matching peers message has gone (1.8.23.1)
Hi
Ever since we upgraded our asterisk servers to 1.8.23.1, we no longer get
the 'no matching peer' error when we get a dictionary SIP attack.
Now the logs always show a 'wrong password' when there actually isn't a
matching peer.
We even have alwaysauthreject = yes in our sip.conf.
Has anyone else noticed this phenomenon?
Thanks in Advance
Ish
--
Ishfaq Malik
2014 Aug 13
1
Asterisk on CentOS7
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: ish at 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
2013 Oct 15
1
PJSIP and ARA
Hi
This is a bit of an exploratory question for groundwork before I start
playing with asterisk 12.
I've spotted the very useful looking file
contrib/realtime/mysql/mysql_config.sql
in the source. Are the table names starting ps_ all to do with PJSIP?
Direct MySQL connection has been deprecated for quite a while, will I need
to use ODBC for PJSIP or will it be supported by the
old
2014 Jan 23
1
Change the preferred audio playback format
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: ish at pack-net.co.uk
w: http://www.pack-net.co.uk
Registered Address: PACKNET LIMITED, Duplex
2014 Mar 24
1
AMI Proxy
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)845 004 4994
f: +44 (0)161 660 9825
e: ish at pack-net.co.uk
w: http://www.pack-net.co.uk
Registered
2014 Apr 10
1
AMI and pyst
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
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44 (0)161 660 9825
e:
2015 Feb 10
2
IAX port
On 10 February 2015 at 09:02, jg <webaccounts173 at jgoettgens.de> wrote:
>
>
>>
>> I get an occasional similar problem, we have Mikrotik firewalls and from
>> tcpdump monitoring on the asterisk boxes I can see that the firewall
>> (unbidden) has changed the IAX port. Usually a firewall reset and sometimes
>> PBX reset combination fixes it.
>>