[asterisk-users] GotoIfTime days query

Hi I'm using 1.8. Is there a way you can specify staggered days in a single GotoIfTime command e.g. mon|wed|fri? Thanks in Advance Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth

Re: [asterisk-users] GotoIfTime days query

So pipes can be used as a secondary delimiter? On Fri, 2011-12-23 at 15:08 +0530, virendra bhati wrote: Hi , make variable and then put in funtion GotoIf() like set(day=mon|wed|fri) GotoIfTime(*,$day,1,jan?happynewyears,s,1); On Fri, Dec 23, 2011 at 3:03 PM, Ishfaq Malik i...@pack

Re: [asterisk-users] Realtime Registration

the updated DB entry. Hope this helps Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live

[asterisk-users] Mixing asterisk.conf, asterisk.ael and asterisk realtime

Hi I'm already mixing asterisk.conf and asterisk realtime architecture (Macros go in .conf) but is it possible to have a Macro in .ael, another in .conf and have them both be callable from the realtime database? Thanks Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062

Re: [asterisk-users] Best VoIP conferencing phone ?

://lists.digium.com/mailman/listinfo/asterisk-users -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live

Re: [asterisk-users] CDR mysql with asterisk 1.4

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 Software Developer PackNet Ltd Office

Re: [asterisk-users] CDR mysql with asterisk 1.4

options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New

[asterisk-users] DONT_OPTIMISE, BETTER_BACKTRACES and performance

Hi How much impact on performance do DONT_OPTIMISE and BETTER_BACKTRACES have on a busy (13000+ entries in cdr for yesterday) server? I'm trying to decide whether to have them on in case of crashes or not. -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062

[asterisk-users] Dependencies for BETTER_BACKTRACES on Centos 5.6

Hi I'm struggling to find the dependencies to allow me to tick BETTER_BACKTRACES while installing asterisk 1.8.7 on CentOS 5.6 Does anyone know what I need to install to do this? Regards Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062

[asterisk-users] Call files and spool directiory shared amongst several asterisk servers

servers share a single spool directory? We are using 1.8 Thanks Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New

Re: [asterisk-users] Frequent Asterisk Restarts

to the asterisk service? -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar

[asterisk-users] Unknown warning

in advance Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs

Re: [asterisk-users] Concurrent call monitoring

Thanks for the tips, I'll try this method now I know. On Tue, 2011-10-25 at 18:52 +0500, Sammy Govind wrote: We used Mix of Nagios, Zabbix, OpenNMS. Best one for this was Zabbix. On Tue, Oct 25, 2011 at 6:49 PM, Ishfaq Malik i...@pack-net.co.uk wrote: Which monitoring tool were you

Re: [asterisk-users] Unknown warning

Is it anything to worry about? there are about 8 a second happening. On Thu, 2011-10-27 at 06:10 -0400, Alex Balashov wrote: It means Asterisk is enqueueing a failed reinvite for retransmission. On 10/27/2011 06:04 AM, Ishfaq Malik wrote: Hi Can anyone shed some light on what

Re: [asterisk-users] Unknown warning

On Thu, 2011-10-27 at 06:16 -0400, Alex Balashov wrote: On 10/27/2011 06:15 AM, Ishfaq Malik wrote: Is it anything to worry about? there are about 8 a second happening. Maybe. Can't say without more data/context/packet capture/etc. Well it doesn't seem to be having too big an impact

Re: [asterisk-users] Unknown warning

On Thu, 2011-10-27 at 06:32 -0400, Alex Balashov wrote: On 10/27/2011 06:30 AM, Ishfaq Malik wrote: On Thu, 2011-10-27 at 06:16 -0400, Alex Balashov wrote: On 10/27/2011 06:15 AM, Ishfaq Malik wrote: Is it anything to worry about? there are about 8 a second happening. Maybe. Can't

Re: [asterisk-users] Unknown warning

, Ishfaq Malik wrote: Is it anything to worry about? there are about 8 a second happening. On Thu, 2011-10-27 at 06:10 -0400, Alex Balashov wrote: It means Asterisk is enqueueing a failed reinvite for retransmission. On 10/27/2011 06:04 AM, Ishfaq Malik wrote: Hi Can anyone shed some

Re: [asterisk-users] voicemail

) -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http

Re: [asterisk-users] dahdi_dummy required?

So am I correct in assuming dahdi_dummy isn't needed/useful anymore? Application MeetMe will not work without it. -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth and Colocation

Re: [asterisk-users] single registration per user

the only way to prevent this is tight control of the usernames and passwords, i.e. configuring all devices yourself without the user(s) knowing what the un/pass for them are. Is there a good reason you can't do this? Regards Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660

Re: [asterisk-users] SNMP problem

folder on your distro? I see people forgetting about that step. On Wed, Sep 14, 2011 at 2:18 AM, Ishfaq Malik i...@pack-net.co.uk wrote: Hi I'm using Asterisk 1.8.3.2 and am trying to configure snmp using this as my resource http

Re: [asterisk-users] realtime goto/gotoif/dial

| seperator instead of the , separator. So i was wondering, is this issue been solved? (I presume so, but can not find any confirmation about it) Hans -- Hi It's pipe in 1.4 and comma in 1.6 and 1.8 Regards Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062

[asterisk-users] SNMP problem

as to where I'm going wrong? Thanks in advancde Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live

Re: [asterisk-users] realtime goto/gotoif/dial

On Wed, 2011-09-14 at 12:01 +0200, Hans Witvliet wrote: On Wed, 2011-09-14 at 09:00 +0100, Ishfaq Malik wrote: On Wed, 2011-09-14 at 00:44 +0200, Hans Witvliet wrote: Hi all, I presume i made a silly mistake while filling a database But while googling on the results, i came

Re: [asterisk-users] SIP Realtime Templates (!)

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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Ishfaq

Re: [asterisk-users] Queue agent login notification

options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New

Re: [asterisk-users] Call drop in 10 seconds without disconnecting a-party call

/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth

Re: [asterisk-users] Queue agent login notification

: 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 Software Developer PackNet Ltd Office: 0161 660 3062

Re: [asterisk-users] Queue Breakout Input being Ignored

(num)} = anonymous or ${CALLERID(num)} = 0]?collect) same = n(collect),Goto(app-helpdesk-callback-collect,s,1) -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth and Colocation

Re: [asterisk-users] Trouble with *8 Pickup

On Mon, 2011-08-15 at 15:32 +0100, Paul Hayes wrote: is this bug already reported at the issue tracker/jira? Is someone working on it? Karsten https://issues.asterisk.org/jira/browse/ASTERISK-18225 That's a different issue to what we have been discussing... -- Ishfaq Malik

Re: [asterisk-users] Trouble with *8 Pickup

On Thu, 2011-08-11 at 16:38 +0100, Paul Hayes wrote: 2011/8/11 Ishfaq Malik i...@pack-net.co.uk mailto:i...@pack-net.co.uk On Thu, 2011-08-11 at 14:47 +0100, --[ UxBoD ]-- wrote: Ah, now this is interesting as one of our clients had the same problem the other day; in our

Re: [asterisk-users] Trouble with *8 Pickup

On Fri, 2011-08-12 at 09:46 +0100, Paul Hayes wrote: On 12/08/11 08:46, Ishfaq Malik wrote: Have you seen it in any other versions of 1.8 or is it something that has happened in the latest release? I've not specifically seen this issue with other versions of Asterisk but then I've never

Re: [asterisk-users] Asterisk reporting

to make pulling reports from it easier. -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live

[asterisk-users] Trouble with *8 Pickup

-0404MASQ, strange things may happen. Does anyone know what this warning means? Thanks Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth and Colocation Provided by http://www.api

Re: [asterisk-users] Trouble with *8 Pickup

with Snom phones. In the case of this server I was looking at, the only time this error occurred was when the pickup request happened in the same second as a dialplan step change so by the time the pick up of the channel was attempted, it no longer existed. -- Ishfaq Malik Software Developer PackNet Ltd

Re: [asterisk-users] MixMonitor and attended transfers [SOLVED]

On Tue, 2011-08-02 at 10:58 +0100, Ishfaq Malik wrote: Hi I'm using asterisk 1.8.3.2 (with a couple of patches) I have the following scenario... SIP call comes in and gets answered by extension A (MixMonitor is executed as part of this inbound dial plan of the number being called

[asterisk-users] CEL and MySQL

Is anyone using CEL with a MySQL backed at all? I've found a table schema but I'm guessing I need some sort of cel_mysql.conf and don't even have a sample for that. Can anyone give me any pointers as to what files I need to change to get this logging to my MySQL table? -- Ishfaq Malik Software

Re: [asterisk-users] CEL and MySQL

I ended up using the patch here https://issues.asterisk.org/jira/browse/ASTERISK-17638?focusedCommentId=180838#comment-180838 and recompiling. It seems to work fine so far On Mon, 2011-08-08 at 17:17 +0100, --[ UxBoD ]-- wrote: cel_odbc.conf and then use adapative odbc I think. -- Ishfaq

Re: [asterisk-users] pickupgroup

within a dialplan but in an office with lots of phones, it can be hard to discern which extension it is that is ringing... -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth and Colocation

[asterisk-users] Asterisk 1.8.5 eventfilter not working in manager.conf

Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs

[asterisk-users] trustrpid in sip.conf

Hi Are there any security issues I need to be aware of if I set trustrpid to yes in my sip.conf? Thanks Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth and Colocation Provided

[asterisk-users] 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

Re: [asterisk-users] MixMonitor and attended transfers

have this problem with 1.4, it just recorded the whole message as a matter of course. I'll have a look into using local channels for this but I think it has more to do with the way that 1.8 as treating attended transfers and how it joins the 2 channels involved. -- Ishfaq Malik Software

Re: [asterisk-users] MixMonitor and attended transfers

On Tue, 2011-08-02 at 10:58 +0100, Ishfaq Malik wrote: Hi I'm using asterisk 1.8.3.2 (with a couple of patches) I have the following scenario... SIP call comes in and gets answered by extension A (MixMonitor is executed as part of this inbound dial plan of the number being called

Re: [asterisk-users] Strange network issue

, are they doing any scheduled backups? Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live

[asterisk-users] [Fwd: Re: Strange network issue]

Forwarded Message From: Ishfaq Malik i...@pack-net.co.uk To: Mike Diehl mdi...@diehlnet.com Subject: Re: [asterisk-users] Strange network issue Date: Fri, 22 Jul 2011 09:55:53 +0100 On Fri, 2011-07-22 at 02:53 -0600, Mike Diehl wrote: On Friday 22 July 2011 2:42:12 am Ishfaq

Re: [asterisk-users] Is there a protocol to be used to communicate between different Asterisk Boxes to distribute load and tasks?

help? Regrads Bilal -- You could look into using DUNDi for locating which box an extension is registered too but you would also have to go realtime with your dialplans for that. Not sure how well it would apply to queues though. -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660

[asterisk-users] *8 causing large number of channels to go stale (possible bug)

channels I have seen so far have been picked up using *8 It would be interesting to see if anyone can replicate this. Thanks Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth

Re: [asterisk-users] *8 causing large number of channels to go stale (possible bug)

Please ignore, I should have looked at issues.asterisk.org first https://issues.asterisk.org/view.php?id=18654 Apologies Ish On Fri, 2011-07-15 at 09:03 +0100, Ishfaq Malik wrote: Hi We're using asterisk 1.8.3.2 with the patch from issue 18818 Were finding a high incidence of channels

Re: [asterisk-users] SoftHangup on asterisk 1.8.3.2 (renamed)

On Tue, 2011-07-12 at 09:13 +0100, Ishfaq Malik wrote: On Thu, 2011-07-07 at 14:23 -0400, Jeremy Kister wrote: On 7/7/2011 9:32 AM, Ishfaq Malik wrote: I'm having the same issue on 1.8.3.2 (with a couple of patches) exten = s,1,Set(CHAN=${SHELL(asterisk -rx core show channels | awk

[asterisk-users] How to Hang up a stale SIP channel?

is written to or read from but as the channel is stale, it will not be written to or read from so the command will not instigate the hangup. Does anyone know of any way we can hangup a stale channel via the console? Thanks in Advance Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660

Re: [asterisk-users] SoftHangup on asterisk 1.8.3.2 (renamed)

On Thu, 2011-07-07 at 14:23 -0400, Jeremy Kister wrote: On 7/7/2011 9:32 AM, Ishfaq Malik wrote: I'm having the same issue on 1.8.3.2 (with a couple of patches) exten = s,1,Set(CHAN=${SHELL(asterisk -rx core show channels | awk '/^SIP\/vgw1-/ { print $1 }' | head -1

Re: [asterisk-users] Queue Issue : Duration between 2 agents call

asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth

Re: [asterisk-users] SoftHangup on asterisk 1.8.2.3

, the console shows: -- Executing [s@nineoneone:10] SoftHangup(SIP/111-00a3, SIP/vgw1-00a2) in new stack but the SIP/vgw1-00a2 is still active. If I use 'channel request hangup SIP/vgw1-00a2', the call is dropped instantly. Am I using SoftHangup incorrectly? -- Ishfaq

Re: [asterisk-users] Asterisk 1.6.1 Realtime SIP Users

/asterisk-users -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every

Re: [asterisk-users] asterisk recording problem

. Best regards, Peter Gelencser Hi You are using the wrong application, you need to use MixMonitor Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth and Colocation Provided by http

Re: [asterisk-users] not find files in asterisk 1.8

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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Ishfaq Malik

Re: [asterisk-users] not find files in asterisk 1.8

Sorry, read your problem properly this time yum install asterisk18-configs On Mon, 2011-06-27 at 13:53 +0100, Ishfaq Malik wrote: If you're installing from source you need to do make samples On Mon, 2011-06-27 at 13:38 +0200, Paolo De Michele wrote: the file already installed

[asterisk-users] Unable to include switch 'Realtime/@' in context

Has anyone ever seen this error before and have any idea why it happened? Thanks Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth and Colocation Provided by http://www.api

[asterisk-users] AMI Suddenly not giving full response to 'Command'

@default NewMessages: 2 OldMessages: 0 This is using Asterisk 1.4.17~dfsg-2ubuntu1 Has anyone ever experienced anything like this before? -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth

Re: [asterisk-users] Queue Log in Mysql

), `data2` VARCHAR(100), `data3` VARCHAR(100), `data4` VARCHAR(100), `data5` VARCHAR(100), PRIMARY KEY (`id`) )ENGINE=InnoDB ; Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062

Re: [asterisk-users] RealTime Queue Logging in 1.8

/Asterisk+log+queue_log [SATISH] On Mon, Jun 6, 2011 at 1:49 PM, Ishfaq Malik i...@pack-net.co.uk wrote: On Thu, 2011-06-02 at 16:03 +0100, Ishfaq Malik wrote: Hi Does anyone know of an accurate resource I could refer to for this? The best I can

Re: [asterisk-users] RealTime Queue Logging in 1.8

. http://www.voip-info.org/wiki/view/Asterisk+log+queue_log [SATISH] On Mon, Jun 6, 2011 at 1:49 PM, Ishfaq Malik i...@pack-net.co.uk wrote: On Thu, 2011-06-02 at 16:03 +0100, Ishfaq Malik wrote: Hi Does anyone know of an accurate resource I could refer

Re: [asterisk-users] RealTime Queue Logging in 1.8

On Thu, 2011-06-02 at 16:03 +0100, Ishfaq Malik wrote: Hi Does anyone know of an accurate resource I could refer to for this? The best I can find is http://www.voip-info.org/wiki/view/Asterisk+queue_log+on+MySQL And that table wont create in my database... Thanks Ish Can someone

Re: [asterisk-users] benefits of asterisk 1.8

-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 Software Developer

[asterisk-users] RealTime Queue Logging in 1.8

Hi Does anyone know of an accurate resource I could refer to for this? The best I can find is http://www.voip-info.org/wiki/view/Asterisk+queue_log+on+MySQL And that table wont create in my database... Thanks Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062

Re: [asterisk-users] Audio dropping

on the broadband connections involved is smokeping http://oss.oetiker.ch/smokeping/ We find it an absolute godsend. -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth and Colocation Provided by http

[asterisk-users] 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 -- _ -- Bandwidth and Colocation Provided by http://www.api

Re: [asterisk-users] AJAM XML output not valid xml

On Mon, 2011-05-23 at 15:41 +0100, Ishfaq Malik wrote: Hi I'm using asterisk 1.8.3.2 and have been implementing AJAM. I've noticed the final '' is missing from every response I've had so far. Here is an example ajax-response response type='object' id='unknown'generic response='Success

Re: [asterisk-users] AstManProxy

On Tue, 2011-05-24 at 10:02 +0100, Steve Davies wrote: On 20 May 2011 16:16, Ishfaq Malik i...@pack-net.co.uk wrote: On Fri, 2011-05-20 at 10:58 -0400, Leif Madsen wrote: On 11-05-20 09:37 AM, Ishfaq Malik wrote: Do many people use this? Is it reliable and safe? It may still work

Re: [asterisk-users] click to call with php

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 Software Developer PackNet Ltd Office: 0161

[asterisk-users] *8 pickup and CLI presentation

? We are using Snom phones but I'm sure this is an asterisk rather than phone issue... Thanks is advance Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth and Colocation Provided

[asterisk-users] AstManProxy

Hi Do many people use this? Is it reliable and safe? Tanks Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New

Re: [asterisk-users] AstManProxy

On Fri, 2011-05-20 at 10:58 -0400, Leif Madsen wrote: On 11-05-20 09:37 AM, Ishfaq Malik wrote: Do many people use this? Is it reliable and safe? It may still work, but that code is quite old, and I'm not even sure it's necessary any more. Leif. The reasons I'm considering

Re: [asterisk-users] AstManProxy

problems. In your case, one request per 10-30 seconds should be just fine. Regards, Is this the same as AJAM? http://www.voip-info.org/wiki/view/Aynchronous+Javascript+Asterisk +Manager+(AJAM) -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062

Re: [asterisk-users] AstManProxy

an paleontological feel now. If you have any doubts, let me know in order to take some notes about common questions. I'll be writing some docs soon about using AJAM with PHP and Javascript (JQuery). Regards, Thanks Jose, usually all I need is pointing in the right direction :) -- Ishfaq Malik

[asterisk-users] 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

Re: [asterisk-users] Manager logged on/off messages

On Thu, 2011-05-19 at 12:15 -0400, Jose P. Espinal wrote: On 05/19/2011 12:05 PM, Ishfaq Malik wrote: 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? Hi Ishfaq, I think that you might use a proxy

Re: [asterisk-users] Asterisk 1.8 latest branch safe for production ?

when my colleges get in but it was only last week that I tested *8 pickup on this particular installation. -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth and Colocation Provided by http

Re: [asterisk-users] Asterisk 1.8 latest branch safe for production ?

the same ringing extensions using *8. Warning do it out of office hours. Alec Davis Hi I do NOT have those 2 options enabled. I'll give the 2 simultaneous calls scenario a test when there are bodies available to assist me. -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660

Re: [asterisk-users] Asterisk 1.8 latest branch safe for production ?

On Fri, 2011-05-06 at 20:21 +0100, --[ UxBoD ]-- wrote: Are you not seeing issues with *8 call pick up then ? Nope, I double checked it after seeing someone saying they had issues with it and it is fine on the installation I have. -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161

Re: [asterisk-users] asterisk syntax highlighting for gedit

of putting one together myself, but don't want to reinvent the wheel. So I'm just enquiring if anyone knows of one that already exists that i've missed. Thanks Naomi Rosenberg www.servicesforasterisk.co.uk The .ini file type does the job -- Ishfaq Malik Software Developer PackNet

Re: [asterisk-users] Realtime SIP peer status

On Wed, 2011-04-13 at 10:28 +0100, Andrew Thomas wrote: BTW - extensions.conf has mySQL functions built in - so no external script is actually needed. Could you point me in the right direction for that? -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062

Re: [asterisk-users] Realtime SIP peer status

On Wed, 2011-04-13 at 11:24 +0200, Jonas Kellens wrote: On 04/13/2011 11:20 AM, Ishfaq Malik wrote: On Wed, 2011-04-13 at 11:09 +0200, Jonas Kellens wrote: On 04/13/2011 10:57 AM, Ishfaq Malik wrote: On Wed, 2011-04-13 at 10:15 +0200, Jonas Kellens wrote: Hello

Re: [asterisk-users] Realtime SIP peer status

On Wed, 2011-04-13 at 10:32 +0100, Ishfaq Malik wrote: On Wed, 2011-04-13 at 10:28 +0100, Andrew Thomas wrote: BTW - extensions.conf has mySQL functions built in - so no external script is actually needed. Could you point me in the right direction for that? Ignore that, I

Re: [asterisk-users] Asterisk 1.8.3

not deleting propperly and hanging up the mail box so users can't check them. 1.8.2 is unusable if you use RealTime without the patch in this issue https://issues.asterisk.org/bug_view_advanced_page.php?bug_id=18403 -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062

Re: [asterisk-users] Asterisk 1.8.3

://issues.asterisk.org/view.php?id=18818 Mind you, this one will only affect you if you use RealTime architecture -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth and Colocation Provided by http

[asterisk-users] Finding out asterisk settings from console

get with sip show settings but there isn't one. Thanks Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join

Re: [asterisk-users] CDR fields not being written from h extension after Dial command completes.

endbeforehexten=yes in your cdr.conf? Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory

Re: [asterisk-users] using ${EXTEN} with waitexten

://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062

Re: [asterisk-users] Sox and bad quality when converting to 8 kHz

regards Thomas The best results I have had have been by using the following mpg123 -q -w ${TEMP} ${INPUT} sox ${TEMP} -c 1 -s -r 8000 ${OUTPUT} Regards Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062

Re: [asterisk-users] wrong time retrieved from system command

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 Software Developer PackNet Ltd Office: 0161 660 3062

Re: [asterisk-users] [1.6.2.5] Asterisk can't find MOH file

) exten = ,n,MusicOnHold(manolo_camp-morning_coffee) = Actually, how can Asterisk know that a file is MOH and hence, should be found in /var/lib/asterisk/moh/, rather than a regular prompt/sound file located in /var/lib/asterisk/sounds? Thank you. -- Ishfaq Malik

[asterisk-users] 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

Re: [asterisk-users] One Way Audio

or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com

Re: [asterisk-users] One Way Audio

Just fixed our problem with directmedia=no but this only applies if your extensions are behind a nat Ish On Thu, 2011-03-10 at 09:40 +, Ishfaq Malik wrote: I've been having a similar (well exactly the same) problem this last week and have been bashing my head trying to fix it. Just

Re: [asterisk-users] Converting MP3 files to wav for Asterisk

apart? I had been having the same issue and this above method has really improved the quality of my wav files (I had previously been using sox -V ${INPUT} -r 8000 -c 1 -s ${OUTPUT} resample -ql) Thanks for that Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062

[asterisk-users] Asterisk 1.8 SIP realtime and NAT

in asterisk 1.8.3? Is there a bug stopping this value being picked up? Can someone even point me to the correct source files so I can attempt to try and work out the correct 1.8 sip table definition from there as I can't find one anywhere at all? Thanks in advance Ish -- Ishfaq Malik Software Developer

Re: [asterisk-users] Asterisk 1.8 SIP realtime and NAT

Hi Scratch that The value name has changed from Nat to Force Rport Back to the drawing board On Wed, 2011-03-02 at 08:57 +, Ishfaq Malik wrote: Hi After recently upgrading to 1.8.3 I have noticed that the nat setting for my peer in my sip table is not making it into the realtime cache

Re: [asterisk-users] Asterisk 1.8.3-rc3 1.8.3 and one way audio

On Mon, 2011-02-28 at 13:40 +, Ishfaq Malik wrote: I've just installed 1.8.3-rc3 on a test server as we really needed that deadlock involving REFER fix on our server but now I'm having an odd issue with one way audio with a specific type of call. If I do extension to extension calls

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