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Zeeshan A Zakaria
IT Consultant
www.zeeshanz.com
855-ZEESHAN
"asterisk asterisk" wrote:
>From time to time, I have a DTMF problem. The asterisk cannot recognize
>my
>handset key pressed if I press 9 to start with. However, if I press
>with 6,
>it is ok.
>
>
t;
> do this
>
> [bigbluebutton]
> exten => _.,1,Set(VOLUME(TX)=10)
> exten => _.,1,Set(VOLUME(RX)=10)
> exten => _.,n,Goto(start-dialplan,s,1)
> exten => _.,n,Hangup
>
>
>
> On Thu, Aug 4, 2011 at 4:33 PM, Zeeshan Ali Shah
> wrote:
>
&g
any hint since it seems asterisk treat it as unknown directive
On Thu, Aug 4, 2011 at 12:22 PM, Zeeshan Ali Shah
wrote:
> but got these as well
>
> [Aug 4 12:21:08] WARNING[3082]: pbx_config.c:1588 pbx_load_config: ==!!==
> Unknown directive: Set(VOLUME(TX) at line 9 -- IGNORING!!!
-- IGNORING!!!
[Aug 4 12:21:08] WARNING[3082]: pbx_config.c:1588 pbx_load_config: ==!!==
Unknown directive: SetGlobalVar(SetVOLUME(RX) at line 11 -- IGNORING!!!
On Thu, Aug 4, 2011 at 12:20 PM, Zeeshan Ali Shah
wrote:
> Yes i tried the followings one by one with differnet values..
> Set(VOLUME(
PM, Zeeshan Ali Shah wrote:
>
>>
>> Tried below, but it still no improvement
>>
>>
>> Zeeshan
>> SetGlobalVar(VOLUME(TX)=10)
>> SetGlobalVar(VOLUME(RX)=10)
>>
>
> Have you tried just doing
>
> Set(VOLUME(TX)=10)
>
> and then 5 etc to mak
Tried below, but it still no improvement
Zeeshan
SetGlobalVar(VOLUME(TX)=10)
SetGlobalVar(VOLUME(RX)=10)
On Thu, Aug 4, 2011 at 3:47 AM, Matt Riddell wrote:
> On 4/08/11 2:12 AM, Zeeshan Ali Shah wrote:
>
>> Hi, I am running asterisk with konference . tried to increase the
&
any way to increase volume in dialplan ? i attached the extensions.conf in
privious post
Zeeshan
On Wed, Aug 3, 2011 at 4:33 PM, Zeeshan Ali Shah wrote:
>
> I tried few times this, but no improvement .
>
> meeting*CLI> konference volume 72193 up
>
> any way to do it fo
I tried few times this, but no improvement .
meeting*CLI> konference volume 72193 up
any way to do it form extensions.conf in dialplan ? since konfernece name is
dynamic
Zeeshan
On Wed, Aug 3, 2011 at 4:30 PM, virendra bhati wrote:
> Hi,
>
> In CLI please press Konference t
; Do the echo test
exten => 600,n,Playback(demo-echodone) ; Let them know it's over
exten => 600,n,Goto(s,6); Start over
-
Zeeshan
On Wed, Aug 3, 2011 at 4:16 PM, Danny Nicholas wrote:
> You need to provide more information
Hi, I am running asterisk with konference . tried to increase the
conference voice but not success
i tried to add in diaplain
SetGlobalVar(Set(VOLUME(TX)=10))
SetGlobalVar(Set(VOLUME(RX)=10))
but it does not effect..
any hint ?
Zeeshan
Thanks a lot guys for your answers. I'll go ahead with NI-2. I didn't know
its the same thing as National.
Thanks again,
Zeeshan A Zakaria
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On 2011-01-21 10:36 AM, "Bruce B" wrote:
Yes, it does. Bell provides the
opinion if you
have used these protocols on an Asterisk box and if there were any things to
consider. If anybody has experience with Primus, it'll be more helpful.
Thanks
Zeeshan A Zakaria
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ved by
the apache user to its destination.
Zeeshan A Zakaria
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On 2010-12-21 3:29 PM, "Danny Nicholas" wrote:
PERL has a move() command; I wouldn’t expect less out of PHP.
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*From:* asterisk-users-boun...@list
Thanks for this info. It seems like good hardware and software solution
provider. I'll explore it a bit more and see if it fits my client's need.
Zeeshan A Zakaria
--
www.ilovetovoip.com
www.pbxforall.com
On 2010-12-20 9:56 AM, "marvin horst" wrote:
I'm not certain wh
tion, your
guidance will be highly appreciated.
Thanks,
Zeeshan A Zakaria
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uipment at big telcos is better
at fixing up bad tones and send you the correct tones.
Zeeshan A Zakaria
--
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www.pbxforall.com
On 2010-11-05 11:24 AM, "Danny Nicholas" wrote:
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
How about 'show channels'.
As for filtering, you'll have to do it separately using a format like:
asterisk -rx 'show channels' | grep ''
You can filter the output further using awk. But each filtering will take a
second or two based on what yo
Its good to know the MATH function because it can do much more and also deal
with floating point numbers. However in your case a simple addition would be
suffice as other posters posted, or try Danny's GotoIf if it fits your
scenario.
Set(vgLabel=vg${MATH(${vg}+1,i)})
Zeeshan A Za
auses hickups
in the new system, and effects quality of service to the customers. As they
say, if its not broken, don't fix it.
Zeeshan A Zakaria
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On 2010-11-03 11:30 AM, "Tilghman Lesher" wrote:
On Wednesday 03 November 2010 09:32:10 Danny Nicho
You'll have to change it in a zaptel source file, and recompile zaptel. I'll
confirm name of the file later today. I have done it a few times.
Zeeshan A Zakaria
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On 2010-11-02 5:37 AM, "Giampaolo TUCCI" wrote:
Hi
I'm
them? Apparently
wilderness of the Internet is protected by law and law makers everywhere
want to keep it this way.
Zeeshan A Zakaria
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www.ilovetovoip.com
www.pbxforall.com (beta)
On 2010-11-01 1:56 PM, "jon pounder" wrote:
On 11/01/2010 01:44 PM, Nyamul Hassan wrote:
I think the only rea
Too late, now switching to attack level: lethal :)
No, I am not one of these losers, and don't ever plan to be.
Zeeshan A Zakaria
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www.ilovetovoip.com
www.pbxforall.com (beta)
On 2010-11-01 1:49 PM, "Jeff LaCoursiere" wrote:
On Mon, 1 Nov 2010, Zeeshan Zakaria wrot
Hi Cary,
Can you email me off the list to point it out?
Zeeshan A Zakaria
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www.pbxforall.com (beta)
On 2010-11-01 1:37 PM, "Cary Fitch" wrote:
I was going to point out a failing of the attackers, but figured they read
the list and don’t need any more tip
And obviously these attackers read our emails on lists like this and adjust
their sick strategies accordingly.
Zeeshan A Zakaria
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On 2010-11-01 12:02 PM, "Jamie A. Stapleton" <
jstaple...@computer-business.com> wrote:
Only 100?
activity, and refuse to do anything against it, what can rest of us do?
Nothing, but suffer.
Unless main Internet routers will identify these attackers and block their
IPs, there is no real way to control this criminal activity.
Zeeshan A Zakaria
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www.ilovetovoip.com
www.pbxforall.com (beta
Unsuccessful attempts are recorded, however SIP-s is not easily doable on
asteridk 1.4. I tried once without any success. Maybe somebody who has
successfully implemented it can write a little how-to on it.
Zeeshan A Zakaria
--
www.ilovetovoip.com
www.pbxforall.com (beta)
On 2010-11-01 4:48 AM
My count has reached 100 for the day. The server serves doesn't serve
international calls anyways, I wonder how would it benefit any hacker in any
way.
--
Zeeshan
Sat, Oct 30, 2010 at 9:33 PM, Joel Maslak wrote:
> No. It seems that opening up some sort of automatic blocking could ca
?
Zeeshan A Zakaria
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Yes, it works fine in 1.4.22 and 1.4.27 and 1.4.35.
Zeeshan A Zakaria
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www.pbxforall.com (beta)
On 2010-10-29 5:15 AM, "Asterisk User" wrote:
Hello everybody,
does anybody know if BLF is correctly working in Asterisk 1.4.*? I'm
particularly interested in
iable back to 0.
Using a macro for dialing would be even a better idea, but that would make
it more complicated for you at this time. Keep it simple for only two
trunks.
Sincerely,
Zeeshan A Zakaria
www.ilovetovoip.com
www.pbxforall.com (beta)
On Thu, Oct 28, 2010 at 1:12 PM, Tim King wrote:
>
p-info.org website and also in my blogs at
ilovetovoip.com.
Regards,
Zeeshan A Zakaria
--
www.ilovetovoip.com
www.pbxforall.com (beta)
On 2010-10-28 3:48 AM, "Per Jessen" wrote:
Over the last two weeks, we have had at least two "incidents" where our
asterisk server got
Do you have canreinvite=yes anywhere? If yes, try setting it to no. Also
pasting your sip.conf here would be helpful.
Zeeshan A Zakaria
--
www.ilovetovoip.com
www.pbxforall.com (beta)
On 2010-10-27 6:16 PM, "Mike Diehl" wrote:
There are NO ACL's in place, either at the ne
Chapters 4, 5 and 6 is a good start.
Zeeshan A Zakaria
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www.pbxforall.com (beta)
On 2010-10-25 2:01 PM, "Jigar Joshi" wrote:
Ok Thanks Guys.
Can you guyz suggest me upto which chapters orwhat are the chapters I should
cover for my requirement.
Because Its too
Do you recommend using wav files instead? Will there be any downside of
using wav?
Zeeshan A Zakaria
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www.ilovetovoip.com
www.pbxforall.com (beta)
--
> _
> -- Bandwidth and Col
The more detail you post, the more chances are there to get help. For
example here you should have posted your sip.conf, devices used, and
probably also the context doing the communication.
Zeeshan A Zakaria
--
www.ilovetovoip.com
www.pbxforall.com (beta)
On 2010-10-21 11:45 AM, "Zakir Mah
Thanks Kevin to verify this. This would really solve a very big problem for
me as E1-T1 conversions has been a big part of my work lately, with no
satisfactory and reliable solution yet. I'll propose this card to my client
and would love to try it.
Zeeshan A Zakaria
--
www.ilovetovoi
every asterisk user
must to know. Otherwise seeking help here won't help because you won't be
able to even understand the answers here.
Zeeshan A Zakaria
--
www.ilovetovoip.com
www.pbxforall.com (beta)
On 2010-10-24 7:59 AM, "Rayan Smith" wrote:
Hi Jigar
> I am facing
transcodings at cell phone providers' ends. Bad though because many
customers use cell phones exclusively. Maybe if I convert them to gsm format
before playing, they'll play better, but will add delay and additional
processing because they are converted and played in real time.
Zeeshan
I am using app_swift.
As a side note, demo on their website also generates sounds which at places
sounds like robotic.
Zeeshan A Zakaria
--
www.ilovetovoip.com
www.pbxforall.com (beta)
On 2010-10-23 6:03 PM, "Darren Sessions" wrote:
Are you using app_swift or wav files?
On Oc
x27;t seem good enough system to
be used in a commercial system. Is there any better quality text-to-voice
engine?
Zeeshan A Zakaria
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www.ilovetovoip.com
www.pbxforall.com (beta)
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_
-- Bandwidth and Colocation Provided by
ntion aculab or adtran, dealt with them in the past, won't deal again.)
I talked to Digium and the sales guy said their TE420 card can *supposedly*
do it as it can have ports configured as a mix of E1s and T1s. Has anybody
used this card for the purpose of T1 to E1 conversion?
Regards
Zeeshan
Hello list,
I need to do E1 to T1 conversion for a project, and was wondering if there
exists a card with both E1 and T1 on it. Or is it possible to use two
separate cards in an asterisk box, one for E1 and one for T1?
(Please don't mention aculab or adtran)
Zeeshan A Za
Rob, you are the man. Thanks for pointing me in the right direction.
Zeeshan A Zakaria
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www.pbxforall.com (beta)
On 2010-10-22 12:28 PM, "Rob Coward" wrote:
Any reason you cant change the asterisk server to bond the 2 nics together ?
We use bonded nics a lot
Thanks for this info.
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-10-22 7:45 AM, "Andrew Latham" wrote:
Have a look...
http://www.digium.com/en/company/casestudies/
Contact John Todd jt...@digium.com with your case studies...
~
Andrew "lathama" Latham
lath...@gm
I didn't know about Digium's cool case studies. Will my realtime virtual PBX
with partially javascript based GUI and Voice Reminder service fit into cool
case study?
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-10-22 7:17 AM, "Andrew Latham" wrote:
The sound files fo
I think you are the first person ever to ask this question. Of course you
can use them, they are royalty free for a purpose.
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-10-22 5:53 AM, "Aurimas Skirgaila" wrote:
Hi,
I wonder if I may freely use the default soundfiles that
be sent to this server with the updated bad IP address.
But the problem is how to make sure that only legitimate users are
contributing to this list. Contributors to this list somehow need to verify
to an admin that they are not hackers, and this the hard part.
Zeeshan A Zakaria
Maybe you should post this portion for your dialplan. I have done the same
thing several times and never had this timeout issue.
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-10-21 4:08 AM, "GBR Icasiano, Ryan A." <
raicasi...@globalbridgeresources.com> wrote:
Hi,
Here is
this setup.
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-10-21 6:56 AM, "Andrew Latham" wrote:
No transcoding? OK, this will work...
http://www.supermicro.com/products/system/1U/5015/SYS-5015A-PHF.cfm?typ=E
~
Andrew "lathama" Latham
lath...@gmail.com
* Learn more about OS
there is no quick and easy way for this purpose. Similarly
to be able to not proceed with the call if account balance is zero, you need
to check account balance before calling the meetme application.
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-10-21 1:32 AM, "DHAVAL INDRODIYA" wro
Yes, one server will do it all. It will not be in a data center but at
customer premisis, so doesn't have to be 1U.
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-10-20 10:07 PM, "Bruce B" wrote:
I am assuming you will rack this in a data-center.For a *server* I won
Any suggestions?
On Wed, Oct 20, 2010 at 9:52 AM, Zeeshan Zakaria wrote:
> Hello list,
>
> What servers would you suggest for:100 concurrent SIP calls, 4xT1 card, and
> a not much busy website, i.e. getting 500-1000 hits a day.
>
> Thanks,
>
> Zeeshan A Zakaria
>
Hello list,
What servers would you suggest for:100 concurrent SIP calls, 4xT1 card, and
a not much busy website, i.e. getting 500-1000 hits a day.
Thanks,
Zeeshan A Zakaria
--
www.ilovetovoip.com
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_
-- Bandwidth and
I didn't design the network, it was already here at clien't site. It is
designed for redundancy. I am trying to come up with a solution to make
asterisk work in it. I am looking into opensips how it can help me.
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-10-18 5:00 PM, &quo
Will OpenSIPs do the job?
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-10-18 4:43 PM, "Paul Belanger" wrote:
On Mon, Oct 18, 2010 at 3:40 PM, Zeeshan Zakaria wrote:
> Is there a way/softwa...
DNS SRV or a SIP proxy.
--
Paul Belanger | dCAP
Polybeacon | Consultant
Jabbe
middle man between asterisk and
the ethernet ports, and by default sends registrations to asterisk only from
eth0, and if this port fails, sends registration coming in from eth1?
Zeeshan A Zakaria
--
www.ilovetovoip.com
two more redundancy
solutions, and will try some new techniques, and probably try three server
setup too. At that time I plan to post a tutorial on redundancy solution on
my blog, because seems like a lot of people want to know how to do it, yet
guidance is very limited.
Zeeshan A Zakaria
--
www
thing but in Proxmox with
DRBD. Somebody told me it could be setup so that even the active calls are
not dropped. I haven't set it up yet, but will try it when get time.
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-10-18 10:59 AM, "Danny Nicholas" wrote:
--
est to use
tcpdump to find out the IP of this service. AMI uses TCP port 5038.
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-10-17 3:37 AM, "Dan Journo" wrote:
Nope,
Its a totally normal self-built Asterisk.
Dan
Zeeshan Zakaria wrote:
Do you use FreePBX by any chance
Do you use FreePBX by any chance?
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-10-16 6:38 PM, "Dan Journo" wrote:
> Serious answer:
> Looks like a process running asterisk -r. Do you have any sort of
> AGI, cron j...
Thanks for lightning my day!
Is there any way to de
deals with international calls. I get a few IPs blocked everyday by
fail2ban, though by default no new connections are allowed international
calls on my system.
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-10-15 10:40 AM, "Steve Edwards" wrote:
On Thu, 14 Oct 2010, bruce bruce wro
I never had this problem, and this is certainly not asterisk's fault.
Probably your conversion is not good. Can you email me a file and I'll do
conversion on my end, and if sounds good, let you know how I did it. Then a
script can be written to convert them all.
Zeeshan
bill them, works perfectly fine. Though it is confusing.
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-10-13 1:21 PM, "Danny Dias" wrote:
Hello Asterisk Community,
Is there a way to check in asterisk cdrs and extension forwarded?
I mean, i'm calling to a ISDN number, wich go
DTMF setting you should have on your system. Usually all
of them support RFC2833, so if in your sip.conf where you have defined the
trunk, dtmfmode is set to rfc2833, your provider should receive it and pass
on to the next carrier or trunk.
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-10-13 10
It depends upon whether you are receiving DTMF or sending, and whether you
are using a VoIP protocol or using DAHDI/Zaptel.
Could you explain a bit what type of setup you have?
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-10-13 9:15 AM, "Dan Journo" wrote:
Hi,
Which DT
Check sip_buddies table for the correct context entry.
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-10-13 5:31 AM, "Oguzhan Kayhan" wrote:
Hello,
I have a default installation of asterisk 1.6.1.9-2
When i create a user in users.conf via asterisk-gui,
calls, voicemail etc works
You need to create a dialplan context to achieve it and then access it using
originate.
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-10-11 5:54 AM, "施铁泉" wrote:
I use the action Originate，i want the called first ringing，the called
answer,callee ringing.it can achieve?
Be
could
guide me to any references, links, books, or other learning sources.
Sincerely,
--
Zeeshan A Zakaria
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elves
------
Zeeshan
On Wed, Oct 6, 2010 at 12:12 PM, Rizwan Hisham wrote:
> Back in the days i heard that they have changed the architecture in 1.6 and
> its a lot better than 1.4 (6 times better call handling and robust
> architecture, someone told me). If they have decided t
be my
next jump once it'll be out of beta, but at this time it should not be used
in a production environment. Many of us still use 1.4 in production and if
you are just starting, this'll be your best choice.
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-10-06 11:54 AM, "Danny Nich
gives you option for hardware level virtulization, called KVM.
I haven't tried it. With only OpenVZ you shall not be able to use
zaptel/dahdi hardware though, and I don't know if KVM allows for it.
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-10-05 10:57 AM, "Steve Howes"
Seems like anonymous SIP calls which end up in from-sip-external context
with a dead end. This is usually how hackers start their hack attempts.
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-10-02 3:05 PM, "bruce bruce" wrote:
Hi Everyone,
Like always, here are IPs from China t
Its a long and old thread, haven't read it all, but just to let you know
this happens when there is no reply from the DNS. So change DNS or install
it locally on your asterisk server. At least caching name server should be
installed.
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-09-24
to retain a better quailty even after a few transcodings, plus
almost every sip provider will be able to receive it as it is and pass it on
as received.
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-09-24 1:02 PM, "Gareth Blades" wrote:
The best format would be in whatever format a
ember if it also sets up the required config files.
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-09-24 11:15 AM, "Warren Selby" wrote:
On Fri, Sep 24, 2010 at 9:55 AM, Robert P. J. Day
wrote:
>
>
> so, is there...
Try installing a local caching nameserver on the sam
>From what you explained it seems to me that your mobile provider has blocked
your sip communication altogether. Have you tried changing IP address of
your asterisk server? If changing IP works, then probably your provider has
blocked you sip communication by IP only.
Zeeshan A Zaka
issue with
spa3000 device, not asterisk.
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-09-20 11:02 AM, "Arie Skliarouk" wrote:
Hi,
On Mon, Sep 20, 2010 at 16:39, Zeeshan Zakaria wrote:
>
> Have you tried removi...
Of course, with the same result.
--
Ari
Have you tried removing option 'g' from your Dial command?
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-09-20 7:45 AM, "Arie Skliarouk" wrote:
Hi,
I use asterisk with sip3000 device with "sip-aho" connected to PSTN and
"sip-ahi" connected to a
It means that fail2ban is not configured correctly on your machine. For me
it works fine, and in fact lately these registration/hack attempts have gone
up significantly, thanks to cloud computing I guess.
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-09-17 5:28 PM, "dave george"
easy.
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-09-16 11:38 AM, "Benny Amorsen"
>
wrote:
Chris Owen writes:
> So I guess my question is what is the real purpose of the q...
The purpose is simply to see if the phone is available. For your
particular use it is likely best
r every 60
seconds, or if they don't on a realtime network, depending upon the
bandwidth, they should be made to do so. Or in some cases you can send a
reboot signal to a sip device too. The bottom line is, try not to do a
'reload' as it would affect everybody else to
You can do 'extensions reload' or 'ael reload' if you don't want to lose
real-time sip registrations. I only reload what is needed to be reloaded
instead of reloading everything.
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-09-15 4:28 PM, "Leif Madsen"
ists in
the sound folder of asterisk), and you'll see the call activity on the CLI.
For the rest, you need to consult chapters 5 and 6 of 'Asterisk-The Future
of Telephony' book.
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-09-15 8:59 AM, "Kevin P. Fleming" wrote:
which come to mind:
1. Is your MySQL up to date?
2. Software versions on your test system are the same as on the production
system?
3. Can you post a MySQL query from your dialplan which works fine.
Regards,
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-09-15 9:20 AM, "Jonas Kellens&qu
I'll keep this all in mind. I don't plan to become a Cisco expert over
night. Flirts I'll try to make them use Asterisk. I don't know the details
yet. But some of these big organizations don't even want to consider
anything other than the proprietary sy
I also thought that they should get it from an official Cisco reseller if
they wanted support. Maybe at this stage they themselves don't know what
they want.
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-09-14 4:14 PM, "Peder" wrote:
My best advice would be “don’t do i
heir networking equipment, IOS, their 7960 series
phones and making them work with asterisk, and also using Cisco press's
wonderful book 'Taking charge of Your VoIP Project'.
Sincerely,
Zeeshan A Zakaria
--
www.ilovetovoip.com
--
__
True, that is even better.
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-09-14 12:49 PM, "Steve Howes" wrote:
On 14 Sep 2010, at 17:32, Dan Journo wrote:
> I'm trying to view a list of the active calls to see i...
Don't?. 'core restart when convenient&#
mands. I
prefer to be brief, and used to this shorter syntax.
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-09-14 12:39 PM, "Dan Journo" wrote:
Hi,
I'm trying to view a list of the active calls to see if I can restart
Asterisk.
When I do 'sip show channels',
Or
1234 => {
Verbose (ID is ${UNIQUEID});
};
:)
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-09-14 12:27 PM, "Danny Nicholas" wrote:
*>From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Zeeshan Zakaria
Can you post what are you doing to see UNIQUEID? And also what version of
Asterisk you are using?
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-09-14 12:11 PM, "Dan Journo" wrote:
> ${UNIQUEID} is going to be realtivly unique certnely in the short term
I dont understand some
This might help to answer poster's question. It tells how the allow
anonymous sip connections work in FreePBX, and shows the code.
http://www.geekzone.co.nz/sbiddle/7183
<http://www.geekzone.co.nz/sbiddle/7183>--
Zeeshan
On Sun, Sep 12, 2010 at 12:11 AM, Paul Belanger &
I use ${UNIQUEID} with ${CDR(accountcode)} and it works great. But this is
when you have an accountcode for each user. As the last poster suggest, you
can append it with date and time and it'll be truly unique and also help you
keep track of the recording.
Zeeshan A Za
good voice
recognition engine should do it.
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-09-14 7:09 AM, "DHAVAL INDRODIYA" wrote:
is it possible with lumenvox i will purchase liceance
regards
Dhaval
On Tue, Sep 14, 2010 at 4:20 PM, Zeeshan Zakaria wrote:
>
&
In theory it should work but in real life it doesn't. Converting reliably
half an hour of speech into text is simply a dream.
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-09-14 4:52 AM, "Nickolay V. Shmyrev" wrote:
В Втр, 14/09/2010 в 14:00 +0530, DHAVAL INDRODIYA пишет
It is simply not possible, though it might be in the distant future.
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-09-14 1:50 AM, "DHAVAL INDRODIYA" wrote:
Thanks Paul,
i think still i have some problem to understand , i mean to say that i have
30 minutes audio file in
WAV fo
believe in making guesses. Troubleshooting
requires some good detail of the problem. And yes, answering non-asterisk
related issues is not the goal of this mailing list.
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-09-11 9:24 PM, "Jeff LaCoursiere" wrote:
> --
> www.ilovetovo
Actually it is a very easy to understand and fix issue, but looking at the
code taking care of anonymous sip calls is the key. Those who post third
party GUI related issues should at least post the underlying asterisk config
or code here, so the asterisk part of the problem can be fixed.
Zeeshan
whatever you like.
--
Zeeshan
On Sat, Sep 11, 2010 at 2:43 PM, Jeff LaCoursiere wrote:
>
> On Sat, 2010-09-11 at 03:53 -0400, Zeeshan Zakaria wrote:
> > This is not elastix or FreePBX forum and asking non-asterisk related
> > questions here is misusing this mailing list. Allo
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