My mistake, CURLOPT(header) is only to retrieve headers, not to sent. sorry.
Nasir Iqbal
ICTBroadcast - an Auto Dialer software for ITSP
<https://www.ictbroadcast.com/how-become-internet-telephony-service-provider-itsp-using-ictbroadcast-sp-edition>
SMS, Fax and Voice broadcasting &am
nnect for each query.
For example with following odbc settings you can achieve 500+ concurrent
channels (approx 2500 queries / minute) without any performance issue.
pooling => yes
limit => 16
pre-connect => yes
Regards
Nasir Iqbal
ICTBroadcast - an Auto Dialer software f
Hi David,
Have you tried CURLOPT function.
i.e
Set(CURLOPT(header)=Content-Type: application/json)
Regards
Nasir Iqbal
ICTBroadcast - an Auto Dialer software for ITSP
<https://www.ictbroadcast.com/how-become-internet-telephony-service-provider-itsp-using-ictbroadcast-sp-edition>
SMS, F
s Carlos
> reported earlier ?
>
> Le sam. 29 sept. 2018 à 13:32, Nasir Iqbal a
> écrit :
>
>> Hi Olivior,
>>
>> We have recently worked on a WebRTC based agent panel. As based on my
>> experience I think that WebRTC based phones are far better and cheaper the
features
which are not possible with regular phones.
Regarding your concern about BLF or call history, for me as a being
developer it is just a matter of customization.
Regards
Nasir Iqbal
ICTBroadcast - an Auto Dialer software for ITSP
<https://www.ictbroadcast.com/how-become-inter
Can you confirm if said device is being provisioned correctly by
provisioning server. and if there is no Firewall between said device and
auto provisioning HTTP server ? And what is PSTN line status ? you can
check both from SPA3102 web interface.
Regards
Nasir Iqbal
ICTBroadcast - an Auto
server or can freely signup for Community Koji Build
system being hosted by CentOs and Fedora
After you have access to koji build system it is very simple and quick to
make your own custom RPM for any target distribution and architecture of
your choice!
Regards
Nasir Iqbal
ICTBroadcast - an Auto
rate a syntax error
in asterisk, My question is if there anyway to fix this syntax error
without compromising extra parameters in application data.
Nasir Iqbal
ICTBroadcast - an Auto Dialer soft
ware for ITSP
<https://www.ictbroadcast.com/how-become-internet-telephony-service-provider-itsp-us
are unable to get any coredump ! (even running with -g) Any help
will be appreciated.
Thanks in advance
Nasir Iqbal
ICTBroadcast - an Auto Dialer software for ITSP
<https://www.ictbroadcast.com/how-become-internet-telephony-service-provider-itsp-using-ictbroadcast-sp-edition>
SMS, Fax and
check this http://www.voip-info.org/wiki/view/Asterisk+cmd+ControlPlayback
Nasir Iqbal
ICTBroadcast
SMS, Fax and Voice broadcasting solution
http://www.ictbroadcast.com/
On Wed, Jan 25, 2012 at 8:29 PM, Eyal wrote:
> Hi,
>
> How can I play a sound file from the middle and end i
Check our asterisk based software ICTBroadcast at
http://www.ictbroadcast.com, it might fulfil your requirements, It support
multiple campaign types including playing voice message to recipients
and forwarding call to live agent
Regards
Nasir Iqbal
ICTBroadcast
SMS, Fax and Voice
We are working to develop a web based IVR Designer that will work with
Asterisk as well as Freeswitch using Raphaejs library, Click following link
for detail
http://sourcecodemania.com/ivr-designer-using-raphaeljs-for-asterisk/
Looking for your valuable suggestions
Regards
Nasir Iqbal
for which user/number sip reinvite is for? ooh! you are running a direct
application without any dialplan or user, may be that is the cause! I think
you should first write fax dialplan, reload asterisk and test again with
originate but this time with extension not application.
Nasir Iqbal
ICT
Check firewall and NAT settings!
Monitoring sip and media flow from asterisk cli can help, use "sip set debug
on", "rtp set debug on" and "udptl set debug on"
Nasir Iqbal
ICT Innovations
http://www.ictinnovations.com/
On Fri, Oct 7, 2011 at 1:37 AM, James Sharp
What about waiting in "queues"?
Nasir Iqbal
ICT Innovations
http://www.ictinnovations.com/
On Wed, Oct 5, 2011 at 1:35 PM, Yaroslav Panych wrote:
> Hello, everyone
>
> Here part of my dialplan context
> [globals]
> CMD_NOOP=0
> CMD_DOSTUFF1=1
> CMD_DOSTUF
You can do this by an AMI based transfer (Redirect) to Local channel, and
then in local channel's dialplan you need to add your desired custom sip
header followed by a dial command.
Nasir Iqbal
ICT Innovations
http://www.ictinnovations.com/
On Wed, Oct 5, 2011 at 11:36 AM, Olivier
have you tried with MYSQL command?
http://www.voip-info.org/wiki/view/Asterisk+cmd+MYSQL
Nasir Iqbal
ICT Innovations
http://www.ictinnovations.com/
On Tue, Oct 4, 2011 at 11:25 PM, Bryant Zimmerman wrote:
> Hey all
>
> I wanted to get some input on what you all think is the be
On some analogs systems caller id is sent after first ring, so
removing "callerid=asreceived"
may help
Nasir Iqbal
ICT Innovations
http://www.ictinnovations.com/
On Tue, Oct 4, 2011 at 4:38 AM, neo haux wrote:
> Hi
>
> I am testing a degium TDP400P (2fxo+2fxs) o
Yes, Zoip support T.38 faxing but It is only client application and you need
FOIP gateway (asterisk) to transmit a fax to your FXO port
Nasir Iqbal
ICT Innovations
http://www.ictinnovations.com/
On Thu, Sep 22, 2011 at 3:20 AM, Olivier wrote:
>
>
> 2011/9/21 Ian Pilcher
>
>
You can use ictfax HTTP://www.ictfax.org web interface to send faxes, Ictfax
is pure foip software based on t.38 as compared to hylafax
No need for iaxmodem and client application
On 22-Sep-2011 4:00 AM, "Larry Moore" wrote:
> On 22/09/2011 4:12 AM, Ian Pilcher wrote:
>> I am looking for a simpl
Please check offline message
Regards
Nasir Iqbal
ICT Innovations
http://www.ictinnovations.com/
On Tue, Sep 20, 2011 at 2:47 AM, mahesh katta wrote:
> Thanks for reply,
> I had check it. in auto dialer whenever dial the number there is no voice
> to get agent. dialer will dial t
Please check our voice sms and fax broadcasting / smart autodialler / smart
predictive dialler based on asterisk named ictbroadcast , it provide real
time report of busy, answered, congestion , failed, no answer call
statistics of running campaign
HTTP://www.ictinnovations.com/ictbroadcast
Regar
Hi,
As zfone download server is offline, is there anyone who can provide me copy
/ link of libzrtp SDK source?
Thanks in Advance
Nasir Iqbal
ICT Innovations
http://www.ictinnovations.com/
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Is there an way (asteisk command / AMI / Agi ) to process incoming SIP
messages like ( 100 trying , 183 session progress , 200 Ack) ,
I am intersted to findout delay between 183 and 200 message
Regards
Nasir Iqbal
Try open souce solution "ICTFAX" for T.38 faxing developed by us available
at http://www.sourceforge.net/projects/ictfax
Nasir Iqbal
ICT Innovations
http://www.ictinnovations.com/
On Sat, Sep 4, 2010 at 3:03 AM, Joel Maslak wrote:
> g711 across a network without perfect
In simple words , Paddy should go with my trick, that is what i got from
this reply
Regards
On Sat, Aug 21, 2010 at 5:14 AM, Sherwood McGowan <
sherwood.mcgo...@gmail.com> wrote:
> Nasir Iqbal wrote:
> > With all honor and respect you deserve, Do I need your permission to
ice gained from user
> > experience
> > is always welcome.
> >
> > Paddy
> >
> >
> >
> >
> > -Original Message-
> > From: asterisk-users-boun...@lists.digium.com
> > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
stom CLI while using the same trunk?
Regards
--
Nasir Iqbal
ICT Innovations
http://www.ictinnovations.com/
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New to Asterisk? Join us for a live introductory
little syntax mistake, try this
exten => s,1,Dial(SIP/Ext400&Local/${ext...@home-context)
[home-context]
exten => s,1,Set(CALLERID(num)=44112233445566)
exten => s,n,Dial(SIP/TheWorld/441234567890)
--
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tal.com --
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> http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
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>
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Nasir Iqbal
ICT Inn
Hi
to convert wav file use following
sox 'orgFile' -w -r 8000 -c 1 -s 'fixedFile'
while replace orgFile and fixedFile with actual filenames
If still now luck try with mp3
Regards
--
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in 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
>
--
Nasir Iqbal
ICT Innovations
http://www.icti
dwidth and Colocation Provided by http://www.api-digital.com --
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> http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http
o UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
--
Nasir Iqbal
ICT Innovations
http://www.ictinnovations.com/
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I and stuff..
>
>
>
>
> On Sun, Aug 8, 2010 at 5:00 AM, Nasir Iqbal wrote:
>
>> Hi
>>
>> following asterisk cli commands can help
>>
>> show channels, show uptime and show sysinfo
>>
>> here is an example
>>
>> asterisk -x "c
ive 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
>
--
Nasir Iqbal
ICT Innovations
http://www.ictinnovations.com/
--
o 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
ive
> method for this ?
>
>
> --
> _
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org/hello
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>
--
Nasir Iqbal
ICT Innovations
http://www.ictinnovations.com/
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idth and Colocation 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/mail
http://www.api-digital.com --
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> http://www.asterisk.org/hello
>
> asterisk-users mailing list
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> http://lists.digium.com/mailman/listinfo/as
ery easy to install. just upload it into Elastix using module
installation interface!!. for further information you can check user manual.
No. you have to download it separately from sourceforge.
>
> Thanks
>
>
> On Tue, Aug 3, 2010 at 2:01 AM, Nasir Iqbal wrote:
>
>>
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> 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
ello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
--
Nasir Iqbal
ICT Innovations
http://www.ictinnovations.com/
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___
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> http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http:
http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
--
Nasir Iqbal
ICT Innovations
http://www.ictinnovations.com/
--
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> http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
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> http://www.asterisk.org/hello
>
>
ed off, which could also be
> a problem. (Row 21 to 26 in your log)
>
> best regards
>
> steve
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join
ww.api-digital.com --
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> http://www.asterisk.org/hello
>
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al.com --
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>
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as a quick option you can use
Dial(IAX2/server1/${EXTEN}&IAX2/server2/${EXTEN}&IAX2/server3/${EXTEN})
call will connect to whichever is available & answer, others simply ignored!
On Sun, Jul 11, 2010 at 8:09 PM, wrote:
> > But how can i determine on which physical server user B is registered?
Hi List,
Recently I tried sending sms using app_sms (hardware TDM400P) in Singapore
with land line telco provider singtel
it worked fine and can send sms in Latin characters 7-bits/8-bits
but I am unable to send Unicode (UCS-2 or 16-bits) sms in Arabic or Chinese.
the problem is that my mobil
ands in macro. as soon as your
macro finished your call will be connected to "Leg B"
you can read more at
http://www.voip-info.org/wiki-Asterisk+cmd+Dial#Dialmacros
and
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Macro
Nasir Iqbal
ICT Innovations
http://www.ictinnovations.com
Hi,
Have you tried Callweaver http://www.callweaver.org
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art mysql;
4. reload / restart asterisk.
Regards
Nasir Iqbal
http://www.ictinnovations.com
On Wed, 2007-09-26 at 23:25 -0500, RENZZO SOTOMAYOR wrote:
> Peder, I have all the permissions in mysql user. I can query my
> database from the local box.
> Mik Cheez, yes, it is. mysql.sock is
cript has been working before...
yes you are right.
but this is only true if your are using FastAGI. its not available in
regular AGI
please visit http://www.voip-info.org/wiki-Asterisk+FastAGI for more
info.
Regards
Nasir Iqbal
http://www.ictinnovations.com
>
>
> regards,
> michael
Hi,
uncomment "immediate=no"
Regards
Nasir Iqbal
ICT Innovations
http://ictinnovations.com
On Sat, 2007-09-15 at 13:18 -0500, Guillermo Salas M. wrote:
> Hello,
>
> I've one astribank with 8 FXO unit and 8 pstn lines connected to the
> astribank. When I receiv
to Script)
$callerID = $argv[1];
Regards
Nasir Iqbal
ICT Innovations
http://ictinnovations.com
On Sat, 2007-09-15 at 18:21 +0200, Michael Kamleitner wrote:
> hi folks,
>
> I've built a simple PHP-script utilizing the AGI-interface. in
> extensions.conf I trigger the script
Hi List,
I wonder that how I can check that FAX is delivered successfully or not,
in my dialplan while using TxFAX.
Which variable contain fax status? as PHASEESTATUS and PHASEESTRING in
Callweaver.
Regards
Nasir Iqbal
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Regards
Nasir Iqbal
ICT Innovations
On Fri, 2007-08-10 at 12:19 -0400, Mike wrote:
> Well, if you really must know (this is OT for everybody else I guess) I have
> a custom Web GUI used for my customers, and when some settings are modified,
> a conf file is created.
(Hand Shaking). But how I can solve it?
I am currently using Asterisk 1.2
Thanks
On Tue, 2007-08-07 at 18:24 +0500, Nasir Iqbal wrote:
> Hi List,
>
> Me setup for faxing is
>
> Asterisk (TxFAX) => ATA => FAX Machine
>
> And SIP setting is
>
> Codec uLaw
> dtm
Hi List,
Me setup for faxing is
Asterisk (TxFAX) => ATA => FAX Machine
And SIP setting is
Codec uLaw
dtmfmode inband
but I am facing a problem
when I send a FAX of one page from Asterisk to ATA+FAX Machine the FAX
Machine Print two pages (Enlarging the page) but shows it received one
page.
or Supervisor
exten => 2,1,Dial(SIP/xo-out/$manager_num) ;for Manager
exten => 3,1,Voicemail(your_voice_mail_box)
Regards
Nasir Iqbal
On Tue, 2007-07-31 at 12:21 -0400, Nitesh Divecha wrote:
> Thanks Nasir,
>
> By putting "'Exten'=> your_extensions_here" i
ns_here,
'Priority'=>1,
'Callerid'=>$cid));
or you must put an "s" extensions in your desired context in this case
it is "default".
Regards
Nasir Iqbal
On Tue, 2007-07-31 at 10:08 -0400, Nitesh Divecha wrote:
> Hello
also have a look on
http://www.voip-info.org/wiki/view/Asterisk+standard+extensions
On Thu, 2007-07-26 at 20:57 -0600, Al lists wrote:
> features.conf
>
> On 7/26/07, GNUbie <[EMAIL PROTECTED]> wrote:
> Hello all,
>
> Where can I find the complete list of default Asteri
Hi Saqib,
Architecture is depend on what service you want to deliver.
Voip is more cheaper then pstn for interoffice connectivity.
But consider regulatory issue before using it.
visit http://www.voip-info.org/wiki-Asterisk for complete detail.
Regards
Nasir iqbal
On Wed, 2007-07-25 at 22
Hi dave,
you can use AMD application
for more info please visit
www.voip-info.org/wiki/index.php?page=Asterisk+cmd+AMD
Regards
Nasir Iqbal
On Tue, 2007-07-24 at 17:22 -0400, dave cantera wrote:
> hi,
> can anyone point me to answering machine beep detection methods or writeups
you.
Regards
Nasir Iqbal
On Tue, 2007-07-24 at 16:03 -0500, Eric "ManxPower" Wieling wrote:
> Noah Miller wrote:
>
> > 2) Set callprogress=yes in zapata.conf (if you haven't already done that).
>
> If you set callprogress=yes you will have the opposite proble
Hi,
please see your ./configure output especially few last lines.
and note missing thins.
Regards
Nasir Iqbal
On Tue, 2007-07-24 at 10:45 -0400, hugolivude wrote:
> Thanks Tahir. I already got the asterisk-addons though - that's what
> I'm having trouble with! BTW - aste
//This Line
Changed
I think that you know how to get arg from shell script.
cheers
Nasir Iqbal
ICT Innovations
On Thu, 2007-07-19 at 08:43 -0400, [EMAIL PROTECTED] wrote:
> I'm in the process of writing a simple autodialer to dial a list of numbers
> and play a message. One of the opti
nging()
exten => 605,n,set(SIP_CODEC=ulaw)
exten => 605,n,RxFAX(/tmp/nasir.tiff|ecm)
exten => 605,n,hangup()
but Thanks for your answer
Thanks
Nasir Iqbal
> [userX]
> ...
> context=internal
> disallow=all
> allow=gsm
> allow=ulaw
> ...
>
> [fax]
> ...
> dis
gt; "dialplan add extension" command.
> >
> > but how we can dynamically create a context in dialplan. is that
> > possible?
> >
Nasir Iqbal
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tch. You can see the ca
Try with underscore before extension like.
exten => _5000/19256002182,1,Answer
Nasir Iqbal
ICT Innovations
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response should be in G711 Codec
Actually I want to setup an extension for FAX receiving (rx_fax) and
other for IVR. when your call FAX extension the codec will be G711 and
when user call IVR the codec must be GSM
Please help me
Thanks
Nasir Iqbal
Hi everybody,
>From asterisk CLI we can add extensions in dial-plan dynamically using
"dialplan add extension" command.
but how we can dynamically create a context in dialplan. is that
possible?
Nasir Iqbal
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mber(s with ring group) as agent.
Regards
Nasir Iqbal
ICT Innovations
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r requirement and also check noise and quality
your PSTN lines.
Regards
Nasir Iqbal
ICT Innovations
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r requirement and also check noise and quality
your PSTN lines.
Regards
Nasir Iqbal
ICT Innovations
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Hi everybody,
>From asterisk CLI we can add extensions in dial-plan dynamically using
"dialplan add extension" command.
but how we can dynamically create a context in dialplan. is that
possible?
Nasir Iqbal
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Hi,
I want to use festival with asterisk to play a text with sable tags.
have some body any idea about it
Nasir Iqbal
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Hi,
I want to use festival with asterisk to play a text with sable tags.
have some body any idea about it
Nasir Iqbal
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Hi,
I want to use festival with asterisk to play a text with sable tags.
have some body any idea about it
Nasir Iqbal
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Hi,
I want to use festival with asterisk to play a text with sable tags.
have some body any idea about it
Nasir Iqbal
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