Displaying 20 results from an estimated 225 matches for "ishfaq".
2010 Mar 29
5
Continue a dialplan when the client hang up the call
Hi all,
When a user make a call to Asterisk, and when user hang up the call at any point of the conversation,? Asterisk will stop Diaplan intermediately.
At this situation,? Are there any way to make? Asterisk continue execute the Diaplan ?, so Asterisk can do something like that delete temporary file, .. etc.
Thanks in advance,
Giang
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An HTML
2011 May 19
3
Manager logged on/off messages
Hi
Is there a way I can stop Manager logged on/off messages from going to
the console/logs without losing all the other information I need?
Regards
Ish
--
Ishfaq Malik
Software Developer
PackNet Ltd
Office: 0161 660 3062
2011 Aug 11
5
Trouble with *8 Pickup
...er_data is NULL
[Aug 8 11:51:53] WARNING[19314] chan_sip.c: No SIP tech_pvt! Fixup of SIP/XXXX-00000404 failed.
[Aug 8 11:51:53] WARNING[19314] channel.c: Fixup failed on channel SIP/XXXX-00000404<MASQ>, strange things may happen.
Does anyone know what this warning means?
Thanks
Ish
--
Ishfaq Malik
Software Developer
PackNet Ltd
Office: 0161 660 3062
2015 Dec 30
2
Signaling ringing on other extension
Ishfaq Malik <ish at pack-net.co.uk> 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/asterisk-DeviceStates-SECT-1.html
Thanks a lot, but it does not seems to work......
2012 Jul 24
2
Finding the position of a character in a string
It there a native asterisk dialplan function which will tell me the
position of a specific character in a given string?
eg if I wanted to find what position the '@' was at in ${SIPURI}
Thanks in advance
Ish
--
Ishfaq Malik <ish at pack-net.co.uk>
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, 2A ENTERPRISE HOUSE, LLOYD STREET
NORTH, MANCHESTER
SCIENCE PARK, MANCHESTE...
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
Registered Address: PACKNET LIMITED, Duplex 2, Ducie House
37 Ducie Street
Manchester, M1 2JW
COMPANY REG NO. 04920552
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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
Manchester, M1 2JW
COMPANY REG NO. 04920552
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2009 Aug 20
8
mysql sip realtime
Hi
I have some question about mysql realtime.
1) Anyone know exactly if there is a specific order to declare sip table
column for realtime ? In which file can I find that order ?
2) In my extconfig.conf, [settings] are :
sipusers => mysql,general,siptable
sippeers => mysql,general,siptable
so means that I use realtime dynamic exactly ?
Is it normal if some parameters from sip.conf still
2011 Aug 02
3
MixMonitor and attended transfers
...A then does an attended transfer of incoming call to extension
B
I'm finding that the recording only lasts up to the point that the
transfer is made.
Is this the correct behaviour? Is there any way I could make this
inbound call into a single continuous recording?
Thanks in advance
Ish
--
Ishfaq Malik
Software Developer
PackNet Ltd
Office: 0161 660 3062
2011 Feb 22
3
Asterisk 1.4.39.2, 1.6.1.22, 1.6.2.16.2, and 1.8.2.4 Now Available
The Asterisk Development Team has announced security releases for Asterisk
branches 1.4, 1.6.1, 1.6.2, and 1.8. The available security releases are
released as versions 1.4.39.2, 1.6.1.22, 1.6.2.16.2, and 1.8.2.4.
These releases are available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/releases
The releases of Asterisk 1.4.39.2, 1.6.1.22, 1.6.2.16.2, and
2011 Feb 22
3
Asterisk 1.4.39.2, 1.6.1.22, 1.6.2.16.2, and 1.8.2.4 Now Available
The Asterisk Development Team has announced security releases for Asterisk
branches 1.4, 1.6.1, 1.6.2, and 1.8. The available security releases are
released as versions 1.4.39.2, 1.6.1.22, 1.6.2.16.2, and 1.8.2.4.
These releases are available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/releases
The releases of Asterisk 1.4.39.2, 1.6.1.22, 1.6.2.16.2, and
2015 Dec 30
2
Signaling ringing on other extension
Ishfaq Malik <ish at 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 program running on my PC to receive a message from
Asterisk if someone calls the other phone...
I'll think about that...
Or may...
2015 Dec 30
2
Signaling ringing on other extension
Ishfaq Malik <ish at 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?
Thanks
Luca Bertoncello
(lucabert at lucabert.de)
2009 Aug 25
2
Authenticating SIP peer on IP address only
Hi
I know this is far from best practice but is it possible to authenticate
a sip peer on the IP address it's coming from so that it doesn't need to
use a UN and Pass?
Ish
--
Ishfaq Malik
Software Developer
PackNet Ltd
Office: 0161 660 3062
2009 Nov 10
1
Call audio leaking between calls
...of phones on the same office network
being able to hear other concurrent call's audio whilst on calls of
their own? We're getting this for the first time and I'm at a bit of a
loss as to where to start to look.
We're using 1.4.17
Any pointers would be much appreciated!
Ish
--
Ishfaq Malik
Software Developer
PackNet Ltd
Office: 0161 660 3062
2010 Dec 17
2
Asterisk Freeze In 1.4 realtime
...this, all the channels
which are in use at that time are simply cut off or frozen.
The only way for us to get everything back to normal is via a hard
restart of asterisk.
If you have seen this behaviour or know of a bug which is associated to
this can you advise, or point us at a fix.
Thanks
--
Ishfaq Malik
Software Developer
PackNet Ltd
Office: 0161 660 3062
2011 Feb 11
3
Asterisk 1.8.3
Hi
Does anyone have any rough idea how far away 1.8.3 is?
We can't deploy 1.8 yet because of this issue
https://issues.asterisk.org/bug_view_advanced_page.php?bug_id=18403
--
Ishfaq Malik
Software Developer
PackNet Ltd
Office: 0161 660 3062
2011 Mar 16
2
chan_sip.c:3115 __sip_xmit of 0x108d33c0 (len 523) to xxx.xxx.xxx.xxx:0 returned -1: Invalid argument
Hi
Does anyone know what this error is about?
I've had 0 success in trying to find any reference to it on the internet
Thanks in advance
Ish
--
Ishfaq Malik
Software Developer
PackNet Ltd
Office: 0161 660 3062
2011 May 26
3
UK English sounds packs
Hi
Does anyone know if there are any free UK accented English sounds packs?
Thanks
Ish
--
Ishfaq Malik
Software Developer
PackNet Ltd
Office: 0161 660 3062
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, M1 2JW
COMPANY REG NO. 04920552
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