[asterisk-users] Hello and Music on Hold question

from the console but that just reloads the conf file. Thanks Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list

Re: [asterisk-users] cant use h,1 at cancel!

rendering the cdr a bit useless? -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http

Re: [asterisk-users] snom mass deploy help

really know perl but I'm sure I'll work out what's going on. Thanks for the head start. Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk

Re: [asterisk-users] Asterisk + mySQL

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Re: [asterisk-users] error in playback of voiceprompt????

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Re: [asterisk-users] Asterisk + mySQL

you can. Ish jonas kellens wrote: Thanks for your reply. I saw that info also on voip-info.org. I was wondering if I could define other columns, like those used for billing (as defined in my sip.conf). Jonas. On Tue, 2009-06-23 at 09:22 +0100, Ishfaq Malik wrote: Hi The calldate column

Re: [asterisk-users] [extensions.conf] Any idea why not working as itshould?

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Re: [asterisk-users] Message Waiting Indication Astersk and kamailio

/listinfo/asterisk-users -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http

Re: [asterisk-users] GUI for Asterisk

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Re: [asterisk-users] Intercepting a Call while ringing a device

-digital.com -- 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 and Colocation

Re: [asterisk-users] Using a mobile phone via USB as an extension

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[asterisk-users] Lagged Extension

poke finds it's way there. Or is that lag too much to create a SIP channel? Thanks in advance Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com

Re: [asterisk-users] Lagged Extension

Oh, I forgot to mention, the customer can make outbound calls from this extension. Just calls cannot be routed back even though the IP Address and port are in the realtime cache. Ishfaq Malik wrote: Hi There I have an extension which is in a different country and is constantly lagged

Re: [asterisk-users] Lagged Extension

somewhere else. Ishfaq Malik wrote: I have an extension which is in a different country and is constantly lagged (about 800ms). When anyone tries to call this extension we get a No route to destination message. ___ -- Bandwidth

Re: [asterisk-users] Realtime difference sipusers sippeers

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 -- asterisk-users mailing list

Re: [asterisk-users] Delete voicemail after couple of days

or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Not 100% on this but I think you want to just delete the files and not the directories too so the command would be find /var/spool/asterisk/voicemail/ -mtime +2 -type f -exec rm {}\; Ish -- Ishfaq Malik

Re: [asterisk-users] CallerPres SIP headers Analog Phone

to my outbound SIP provider the RPID header is not correct privacy=off;screen=no instead of full and yes how can I correct this? Have you set sendrpid = yes in your sip.conf? -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062

Re: [asterisk-users] Asterisk and several clients behind NAT

-digital.com -- 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 and Colocation Provided

Re: [asterisk-users] Test Function if SIP Device is Still Alive

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Re: [asterisk-users] Music on hold based on user

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Re: [asterisk-users] sip trunk that fails over time

-- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing

Re: [asterisk-users] sip realtime with caching

://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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register

[asterisk-users] 20 seconds cut off problem

but that person got a new handset (they were previously using a very old and rubbish Grandstream) and the problem immediately stopped. Has anyone experienced anything like this before? -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062

Re: [asterisk-users] 20 seconds cut off problem

. Thanks again Ish Peter Johansson wrote: Hello. I think i've seen this problem, it was generated by a missing ACK on 200 OK. If that is the case try modifying session timer parameters in sip.conf so a missing ACK will not lead to call termination. Peter Ishfaq Malik wrote: Hi I'm

Re: [asterisk-users] No audio on remote SIP calls

Phoenix, Arizona Register Now: http://www.astricon.net 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] SNOM Phones Displays NR Frequently

registrar occasionally fail, you might want to specify registrar via IP rather than by name. Regards, Chris We always upgrade to v7.3.7 and find it very stable -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062

Re: [asterisk-users] Cisco 7960 Multiline phone

://www.astricon.net 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 and Colocation

Re: [asterisk-users] call drops after a few seconds

- October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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] Areski CDR + Mysql + asterisk 1.6

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 -- AstriCon 2009 - October 13 - 15 Phoenix

Re: [asterisk-users] mysql sip realtime

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Re: [asterisk-users] mysql sip realtime

/mailman/listinfo/asterisk-users -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http

Re: [asterisk-users] mysql sip realtime

and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Ishfaq Malik Software

Re: [asterisk-users] mysql sip realtime

: http://www.astricon.net 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] mysql sip realtime

://www.kolmisoft.com VoIP Billing and Routing Solutions -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ishfaq Malik Sent: 2009 m. rugpjūčio 20 d. 15:56 To: Asterisk Users Mailing List - Non-Commercial

Re: [asterisk-users] mysql sip realtime

Have you set the qualify column in the sip table? harry R wrote: 2009/8/21 Ishfaq Malik i...@pack-net.co.uk mailto:i...@pack-net.co.uk I have to disagree with you there, we use 1.4.17 and sip prune realtime works fine After a few test, I notice these events when I use

[asterisk-users] Set up SER as a SIP extension on asterisk server

have any pointers for me for setting up a sip extension in a SER that I can pass on as I've never looks at one? Cheers Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 ___ -- Bandwidth and Colocation Provided by http

[asterisk-users] 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

Re: [asterisk-users] Set up SER as a SIP extension on asterisk server

and no authentification. Do the necesssary stuff in your extension.conf to identify and bill your client. Olivier Ishfaq Malik a écrit : Hi People We have a client who want to route their outbound calls through our asterisk server. We need them to authenticate as a sip extension so we know

Re: [asterisk-users] DeadAgi

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 and Colocation Provided by http://www.api-digital.com

[asterisk-users] ding Dialled number down a sip channel to a PBX

CALLERID(dnid) and CALLERID(rdnis) to the dialled number, though not at the same time but the customers PBX box does not pick up the dialled number setting. Has anyone got any experience in this? Thanks Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062

Re: [asterisk-users] Sending Dialled number down a sip channel to a PBX

Bumping this in the hope that it is seen by people who missed it before. Ishfaq Malik wrote: We have a customer who connects PBX boxes (Avaya etc.) to our asterisk server (1.4.17) as a SIP extension. This customer needs the dialled number sent to the PBX as well as number that the call

Re: [asterisk-users] Is anyone doing real time updates to where asterisk registers?

-- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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] Calls hang up after 20 seconds

(incoming, 03411885583, 4) exited non-zero on 'SIP/03411885583-081e0d78' (peer 101 was not connected at this time, but Asterisk also hags up with all the peers connected) Any idea? Thanks in advance. -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062

Re: [asterisk-users] sporadic one-way audio

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 and Colocation Provided by http://www.api-digital.com

[asterisk-users] Check if a variable is set

Hi Is there any way to check if a variable is set in asterisk? I've had a look around and can't find a purpose built function for it. I'm going to be using it to see if an argument has been passed with a macro or not (e.g. see if ${ARG3} is set or not) Thanks in advance Ish -- Ishfaq Malik

Re: [asterisk-users] Check if a variable is set

On Fri, Oct 16, 2009 at 3:04 PM, Ishfaq Malik i...@pack-net.co.uk mailto:i...@pack-net.co.uk wrote: Hi Is there any way to check if a variable is set in asterisk? I've had a look around and can't find a purpose built function for it. I'm going

Re: [asterisk-users] Check if a variable is set

That fails to execute in both conditions ABBAS SHAKEEL wrote: Please try this xten = s,2,ExecIf( 0EXISTS(${ARG3})=1 0${ARG3}=1| On Fri, Oct 16, 2009 at 3:45 PM, Ishfaq Malik i...@pack-net.co.uk mailto:i...@pack-net.co.uk wrote: I'm basically trying to make an argument optional

Re: [asterisk-users] Check if a variable is set

itself? I think we might need to store the return value of isnull then test with execif. On 16/10/2009, Ishfaq Malik i...@pack-net.co.uk wrote: That fails to execute in both conditions ABBAS SHAKEEL wrote: Please try this xten = s,2,ExecIf( 0EXISTS(${ARG3})=1 0${ARG3}=1

Re: [asterisk-users] Check if a variable is set

Malik i...@pack-net.co.uk wrote: That fails to execute in both conditions ABBAS SHAKEEL wrote: Please try this xten = s,2,ExecIf( 0EXISTS(${ARG3})=1 0${ARG3}=1| On Fri, Oct 16, 2009 at 3:45 PM, Ishfaq Malik i...@pack-net.co.uk mailto:i...@pack-net.co.uk wrote: I'm basically

Re: [asterisk-users] Check if a variable is set

-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ishfaq Malik Sent: Friday, October 16, 2009 9:08 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Check if a variable is set Hi Here it is [macro-extcall

[asterisk-users] sip fullcontact and port values

we look in the DB the location always has the port we set but the port value is often something else? What do these 2 values actually do and where is the port value being generated? Thanks in advance Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062

Re: [asterisk-users] R: CDR(billsec)

/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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] R: CDR(billsec)

that hangup.php is operating (at least somewhat) independently of the dialplan. The OP seemed to want in-line knowledge of his billable seconds. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ishfaq Malik Sent

Re: [asterisk-users] R: R: R: CDR(billsec)

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Re: [asterisk-users] Voicemail file

://www.api-digital.com -- 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

[asterisk-users] Call audio leaking between calls

appreciated! Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http

Re: [asterisk-users] Call audio leaking between calls

wrote: Aslamoalikum Ishfaq Can you check this with asterisk 1.6.X ? On Tue, Nov 10, 2009 at 2:45 PM, Ishfaq Malik i...@pack-net.co.uk mailto:i...@pack-net.co.uk wrote: Hi Has anyone ever had experience of phones on the same office network being able to hear other concurrent

Re: [asterisk-users] Call audio leaking between calls

Doug Lytle wrote: Ishfaq Malik wrote: Has anyone ever had experience of phones on the same office network being able to hear other concurrent call's audio whilst on calls of It's called cross talk and yes, we've experienced it. But, it will only happen

Re: [asterisk-users] Call audio leaking between calls

or at least increasing the physical distance between them - in my experience this is the most common cause for cross talk. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ishfaq Malik Sent: Tuesday, November

Re: [asterisk-users] Call audio leaking between calls

Malik Sent: Tuesday, November 10, 2009 7:47 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call audio leaking between calls Doug Lytle wrote: Ishfaq Malik wrote: Has anyone ever had experience of phones on the same office network

[asterisk-users] Odd Local Channel and 0 billsec issue

proportion, but not all the calls show this issue. Has anyone had any experience of similar issues? Thanks in advance Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 ___ -- Bandwidth and Colocation Provided by http://www.api

Re: [asterisk-users] Odd Local Channel and 0 billsec issue

that go through the local channel instead of the defined channel would be useful to help diagnose what's going on here. Thanks, --Warren Selby On Mon, Nov 16, 2009 at 4:01 AM, Ishfaq Malik i...@pack-net.co.uk mailto:i...@pack-net.co.uk wrote: Hi I've been noticing an odd issue

[asterisk-users] Is Answer really needed

it from all my dial plans? Because it doesn't seem to to me. Thanks Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list

Re: [asterisk-users] Problem with Timeout

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Re: [asterisk-users] Recording from billsec

by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Use option b b - Only save audio to the file while the channel is bridged. *does not include conferences* -- Ishfaq Malik Software

Re: [asterisk-users] sip realtime question

rtcachefriends=yes Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http

Re: [asterisk-users] Calls Dropping

-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 and Colocation Provided by http://www.api

Re: [asterisk-users] Realtime mysql extensions mutiple queries for each priority?

this patch gets applied to Asterisk, the better. Thanks. JR Looks like I'll be giving it a go in week or so then! I'll let you know how I get on too. -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 ___ -- Bandwidth

Re: [asterisk-users] Per user voicemail greeting

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 and Colocation Provided by http://www.api-digital.com -- asterisk

Re: [asterisk-users] SOLVED IN PART Per user voicemail greeting

listu...@spamomania.co.uk wrote: On Wed, 2009-12-30 at 15:18 +, Ishfaq Malik wrote: Get the customer to log into their voicemail mailbox and follow the instructions to record an unavailable message (Options 0 then 1 if there are no messages I think) Then in the conf you need

Re: [asterisk-users] Extension Status

talk but it doesn't effect . could you help me please? Thanks -- Ahmed Magdy Mahmoud Hi You need to set a value for qualify in your sip.conf Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062

[asterisk-users] Ringing issue

be helpful and I will be trying to replicate this on out test system. Thanks in advance Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth and Colocation Provided by http://www.api

Re: [asterisk-users] Ringing issue

Ishfaq Malik wrote: Hi We run a hosted VoIP service for multiple customers off the same server and I'm having an odd issue with just one customer in particular. We're using realtime in a MySQL DB and this is their dialplan *** 1. row

Re: [asterisk-users] Ringing issue

Ishfaq Malik wrote: Ishfaq Malik wrote: Hi We run a hosted VoIP service for multiple customers off the same server and I'm having an odd issue with just one customer in particular. We're using realtime in a MySQL DB and this is their dialplan *** 1. row

Re: [asterisk-users] Snom vs Polycom

they looked a bit cheap (although costing much more!) and from an admin point of view were not nearly as good as the snom. But that's just my subjective opinion! Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062

Re: [asterisk-users] MySQL RealTime Error

: Invalid database specified: asterisk (check res_mysql.conf) the content of res_mysql.conf is: http://www.pastebin.org/81966 i've try command mysql -uroot -proot ,i can connect to mysql successfully. Could you tell me what's wrong with me ? -- Ishfaq Malik Software Developer PackNet Ltd

Re: [asterisk-users] Linux-based hard phones?

Ken D'Ambrosio wrote: Just wondering if there are any Linux-based hard phones out there -- if so, it'd be neat to see if I couldn't take advantage of the underlying OS. Thanks, -Ken Snom phones use Linux Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062

Re: [asterisk-users] asterisk and mysql connection

+RealTime Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit

[asterisk-users] Muted calls occasionally dropping after 30 seconds

that there was nothing else going on at the time. The call path when recreating this on our test platform was My Mobile - number/SIP provider - out asterisk server - SIP extension Has anyone else ever experienced anything like this? It's really got me rather frustrated! Thanks in advance Ish -- Ishfaq

Re: [asterisk-users] IP Phone recommendation

willing to admit that I may not always be right, but I am never wrong. -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk

Re: [asterisk-users] OpenVPN/SNOM 820: a review.

What firmware version were you using? -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE

Re: [asterisk-users] Is answer() necessary ?

plan applications that implicitly invoke it. Alex Hope all this helps Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062

Re: [asterisk-users] rtcachefriends qualify sip reload

need to do a sip prune realtime either directly from asterisk cli or using AMI. You really do not need to do a SIP reload when changing the config of one sip extension. Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062

Re: [asterisk-users] rtcachefriends qualify sip reload

jonas kellens wrote: On Tue, 2010-03-02 at 11:32 +, Ishfaq Malik wrote: In my experience, yes, that is normal behaviour. Generally any SIP phone will try to reconnect with the server within 2 mins anyway. In the Zoiper softphone, it is set to 3600 seconds... I don't want my

Re: [asterisk-users] rtcachefriends qualify sip reload

jonas kellens wrote: On Tue, 2010-03-02 at 11:32 +, Ishfaq Malik wrote: If you are changing RealTime config in your DB you need to do a sip prune realtime either directly from asterisk cli or using AMI. You really do not need to do a SIP reload when changing the config of one sip

Re: [asterisk-users] realtime call peers status

Thanks a lot in advance From asterisk cli core show channels core show channel channel name If you need to put it into a pretty front end you can use the AMI Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062

Re: [asterisk-users] realtime call peers status

:) ). I'm reading something about dymanic realtime: could be ok for my needs? Or is better spent my time on this docs : http://www.voip-info.org/wiki/view/Asterisk+manager+API ? 2010/3/2 Ishfaq Malik i...@pack-net.co.uk: lore wrote: Hi all, I need to check in realtime the calls

Re: [asterisk-users] rtcachefriends qualify sip reload

] rtcachefriends qualify sip reload On Tue, 2010-03-02 at 15:59 +0100, jonas kellens wrote: On Tue, 2010-03-02 at 11:32 +, Ishfaq Malik wrote: If you are changing RealTime config in your DB you need to do a sip prune realtime either directly from asterisk cli or using AMI. You

Re: [asterisk-users] Snom Provisioning

There is a setting for provisioning server for snom phones, if you set that as a server and script of your own you can set the settings remptely and also change the settings remotely, here's a good place to start http://wiki.snom.com/Functions/Phone/Mass_deployment Ish -- Ishfaq Malik Software

[asterisk-users] ExtenSpy Problem

-081acfe0 Playing 'beep' (language 'en') -- SIP/PACK501-081acfe0 Playing 'beep' (language 'en') Does anyone have any thought/experience of this? Also, if a call is already being recorded by MixMonitor, can it also be spied on? Thanks in advance Ish -- Ishfaq Malik Software Developer PackNet Ltd

Re: [asterisk-users] ExtenSpy Problem

Ishfaq Malik wrote: Hi I'm trying to get ExtenSpy to work but it wont, I'm dialling a number from my mobile which comes into our server and answering the number on a particular SIP extension which all works fine. I'm then dialling an exten from my own SIP extension which executes

Re: [asterisk-users] Android Phones ;-)

or to google, just a happy user ;) Conrad I did the same last week and agree totally, a nice little softphone, well integrated with the rest of the phone and took about 1 min to configure without looking at any instructions. -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161

Re: [asterisk-users] Android Phones ;-)

Jeff LaCoursiere wrote: On Mon, 15 Mar 2010, Ishfaq Malik wrote: Conrad Wood wrote: FWIW, just received an android-based phone and after installing sipdroid found that it works very well with asterisk ;). It automatically dials numbers through asterisk if available and otherwise

Re: [asterisk-users] ExtenSpy Problem

David Backeberg wrote: On Fri, Mar 12, 2010 at 11:31 AM, Ishfaq Malik i...@pack-net.co.uk wrote: Hi I'm trying to get ExtenSpy to work but it wont, I'm dialling a number from my mobile which comes into our server and answering the number on a particular SIP extension which all works fine

Re: [asterisk-users] ExtenSpy Problem

David Backeberg wrote: On Mon, Mar 15, 2010 at 1:19 PM, Ishfaq Malik i...@pack-net.co.uk wrote: David Backeberg wrote: On Fri, Mar 12, 2010 at 11:31 AM, Ishfaq Malik i...@pack-net.co.uk wrote: You didn't mention version. Could be relevant. Apologies for not adding

Re: [asterisk-users] ExtenSpy Problem

David Gibbons wrote: snip Bumping a thread without adding anything useful is annoying. If you do it again, I won't be helping. /snip Although I have gotten quite a chuckle from your posts, it's really going to hurt when you fall from that high horse. I thought that was a little harsh

Re: [asterisk-users] ExtenSpy Problem [SOLVED]

Ishfaq Malik wrote: David Backeberg wrote: On Mon, Mar 15, 2010 at 1:19 PM, Ishfaq Malik i...@pack-net.co.uk wrote: David Backeberg wrote: On Fri, Mar 12, 2010 at 11:31 AM, Ishfaq Malik i...@pack-net.co.uk wrote: You didn't mention version. Could be relevant

Re: [asterisk-users] voicemail problem

and hang up or press the pound key. Is there a way to delete the second part from the voicemail, that only my personal recorded message is played back and a signal tone comes to signal the caller to start talking?! Tamer Higazi -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161

Re: [asterisk-users] Continue a dialplan when the client hang up the call

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 -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062

[asterisk-users] Full transfer details on inbound calls

the searching in there. My question is is this a good way to go about what I'm trying to achieve or is there a simpler/less process intensive method that I'm missing. Thanks in advance Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062

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