Linphone is available for all major OS platforms.
Then there is PortGo as well
Regards,
Amit Patkar
On April 29, 2017 9:05:22 PM GMT+05:30, Thomas <thomasit...@gmail.com> wrote:
>Hello,
>Iam lookong for an Softphone for iPhor oder Android smartphone using
>togehter
>with an
Thanks Mathew. I understand that there is no coordination between
AsyncAGI & AMI.
Is there any dial plan function which can tell us if there is active AMI
session?
Thanks & Regards,
Amit Patkar
On 9/21/2016 6:27 PM, Matthew Jordan wrote:
On Tue, Sep 20, 2016 at 10:49 PM, Ami
It means, AMI application is no more running or crashed or lost network
connection with asterisk server.
In such cases call is neither answered nor disconnected by Asterisk. I want to
detect such state and jump to next dial plan to answer or reject the calls
Regards
Amit Patkar
On September 20
Regards,
Amit Patkar
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Hi
Is there any way to timeout AsyncAGI if there is no activity on channel
for defined period? I wish to send call to alternate route if there is
no activity on channel for defined period.
Thanks & Regards,
Amit Pa
Hi
Your extensions.conf should have +17775551212 extension and not 17775551212
Add + sign before your number. This should solve your issue.
[from-external]
exten => +17775551212,1,Log(WARNING, TWILIO)
same => n,Hangup()
*Thanks & Regards,*
A
instance.
How should I choose EC2 instance?
How many vCPU, RAM should be selected? I am assuming that server with ssd is
required as all 500+ calls needs to be recorded.
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referring to AWS deployment. Please help me to choose
AWS server instance.
*Thanks Regards,*
Amit Patkar
On 3/7/2015 12:19 AM, Jeff LaCoursiere wrote:
Why use Amazon? With that kind of load I would want dedicated
servers. Call Rackspace or Softlayer.
j
On 03/06/2015 11:59 AM, Amit
You can use following command to check
netstat -an
This will show host and ports in numeric format.*
Regards,*
Amit Patkar
On 2/27/2015 6:33 AM, Rusty Newton wrote:
On Mon, Feb 23, 2015 at 5:51 AM, Raj Roy Ghandhi roy.gan...@gmail.com
mailto:roy.gan...@gmail.com wrote:
Hi Friends
. It should show bidirectional traffic. If
not, you surely have an issue with media IP or ports.
*Thanks Regards,*
Amit Patkar
On 11/27/2014 10:01 AM, Marie Fischer wrote:
On 22.11.2014, at 13:40, Yves A. yves...@gmx.de wrote:
I have a really strange problem which is driving me crazy for days now
or these are IVR calls?
How many cards are installed in this server? (Total PRI terminated on
this server).
What is average call duration?
*Regards,*
Amit
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Hi Rusty
There are many posts on this list regarding Firefox interoperability.
Its not determined if problem is with Firefox or Asterisk.
Can someone share working configuration on this list? This will help
many users to configure Asterisk to support DTLS-SRTP.
*Thanks Regards,*
Amit Patkar
Hi Gary
You need to check if ICE / STUN is configured.
How are these extensions configured? If you are in private network, you
might have to disable DirectMedia / reInvite for calls going between 2
asterisk boxes.
I hope this helps to resolve your issue.
*Thanks Regards,*
Amit Patkar
On 5
Please check rtp.conf
Look for stunaddr setting. You can try with google STUN server
stunaddr = stun.l.google.com:19302
*Thanks Regards,*
Amit Patkar
On 5/21/2014 9:13 PM, Gary Shergill wrote:
Hi again,
Just noticed this is being sent to the wrong thread... first time using a
mailing list
it with Asterisk, I have successfully deployed
application on other SIP platforms and interoperability with SIP-I/SIP-T
was not an issue.
*Regards,*
Amit Patkar
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Hi Dhaval,
If you capture and share SIP traces for inbound and outbound calls
separately, experts on this list can guide to achieve objective.
You can enable SIP trace on asterisk by executing following command in
Asterisk console
*sip set debug on*
*Thanks Regards,*
Amit Patkar
On 3/12
again to
normal.
*Thanks Regards,*
Amit Patkar
On 1/28/2014 12:32 AM, Ron Wheeler wrote:
Can you get a reading of the total number of I/Os during your test?
Peak IOPS?
That might tell you very quickly about the storage pattern that
Asterisk uses.
Can you configure a RAM drive to see
I am using Monitor function. Let me try with MixMonitor and update.
After 80 calls, I see retransmission of SIP messages, unanswered calls..
*Thanks Regards,*
Amit Patkar
On 1/28/2014 3:25 AM, Matthew Jordan wrote:
On Mon, Jan 27, 2014 at 1:02 PM, Ron Wheeler
rwhee...@artifact-software.com
Will check and update.
*Thanks Regards,*
Amit Patkar
On 1/28/2014 5:45 AM, Tiago Geada wrote:
Hi,
MixMonitor takes a parameter of a system command to run when the
recording finishes. Like Chris said, you can write to ramdisk, and run
a script that will move the file into final position
test with simple IVR, I achieved 400+ calls with same server.
So write seem to be an issue.
Is there any way to tune / optimize / configure for better write performance?
I am not sure if I need to post this query on developers list? Please guide...
Regards
Amit Patkar
Message: 1
Date: Fri, 24
, as and when received, I will need disk IO system with
approx 25000 IOPS assuming 20 ms RTP packet.
Please assist me on this requirement.
*Thanks Regards,*
Amit Patkar
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test with simple IVR, I achieved 400+ calls with same server.
So write seem to be an issue.
Is there any way to tune / optimize / configure for better write performance?
I am not sure if I need to post this query on developers list? Please guide...
Regards
Amit Patkar
they're not a member of
any queue.
Thanks Regards,
Amit Patkar
after 40 sec only. This way comminication sync is completely lost.
Has some one come across such situation? Please help me to solve this issue.
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throwing Cal from Asterisk to application box i have to use SIP
request which having some string in R-URI. Please let me know is this
possible with configuration example.
Regards
Amit
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::. = date:AdBY 'digits/at' IMp:${SAY}
_date::. = date:AdBY:${SAY}
_time::. = date:IMp:${SAY}
Thanks Regards,
Amit Patkar
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New to Asterisk
] netsock2.c: Splitting '192.168.2.18:7490'
into...
[Jul 2 15:54:44] DEBUG[2737] netsock2.c: ...host '192.168.2.18' and
port '7490'.
[Jul 2 15:54:44] DEBUG[2737] chan_sip.c: Trying to put 'BYE sip:100'
onto UDP socket destined for 192.168.2.18:7490
Thanks Regards,
Amit Patkar
On 7/2/2013 5
,
Amit Patkar
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Thanks Matt.
app_playback module is loaded. I am able to play numbers.Issue is only
with date time and only with SAY DATETIME function. If I use SAY DATE,
date is getting played.
Do I need to check some other settings?
Thanks Regards,
Amit Patkar
On 7/1/2013 6:03 PM, Matthew Jordan wrote
Hi
I want to can we use asterisk as TTS server. Which can support mrcpv2 and
ssml.
Im looking for tts server with above requirement will asterisk 1.8 is
useful for me. Any configuration available.
Any opensource tts available.
Amit
On Mar 13, 2013 10:16 PM, Amit Salunkhe amitsalunkh...@gmail.com wrote:
Hi
I want to know asterisk 1.8 as text to speech server.
If we can use as TTS server then it support SSML.
Any sample configuration available for this requirement. Plz help me with
support asterisk as tts server
please guide
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used to get this.
I am using Asterisk 1.8.11 and Dahdi 2.4
Thanks Regards,
Amit Patkar
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Hi
Its surprising that no one is responded. Does this mean, that nobody
has ever used FastAGI and AsyncAGI?
Does that also mean, that FastAGI AsyncAGI should not be used?
I am using Asterisk 1.8.xx
Thanks Regards,
Amit Patkar
On 10/8/2012 11:26 AM, Amit Patkar | ATPL wrote:
Hi
commands will be passed by AMI session.
Regards,
Amit Patkar
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I have been doing a lot of reading forums and elsewhere but am somehow
unable to connect the dots.
Here is what I am trying to accomplish initially and then wish for it to
grow bigger from here on.
I have two POTS (Analog) line that would connect to the Asterisk Box.
I have, to begin with 5 IP
) modules too and if you are looking for a best sound qulity get
hardware echo canceller too.
I just didnt get why your are going to set 5 ext on each IP Phone!!!
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On Sat, Jun 16, 2012 at 4:34 PM, Amit Patel pistolfir...@gmail.comwrote:
I have been doing a lot of reading forums
deployment?
Thanks Regards,
Amit Patkar
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Hi,
Appreciate everyone for your valuable inputs. All these inputs provided by
you are really useful.
Thanks Regards,
Amit Patkar
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New to Asterisk
to be recorded.
What kind of capacity are you looking to achieve?
[Amit Patkar] Some where 2400 G.711 sessions with recording. So approx 1200
calls.
From my experience, Asterisk is not really much of a RAM hog. A couple
GB
is good for a couple hundred simultaneous calls.
With 4 'Intel(R) Xeon(TM) CPU
Hi Kevin,
Thank for your views. Where as no one is ready to share real numbers. I am
looking at benchmarks so that I can plan for resources.
Since asterisk project is active for so many years, I was expecting some
published numbers.
Thanks Regards,
Amit Patkar
On 03/12/2012 03:38 PM, Steve
needs to be
recorded.
What will be impact on no of session when G729a is used?
Thanks Regards,
Amit Patkar
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Hi,
Can someone share how can I configure asterisk to get P-Associated-Uri header
in 200 Ok to the REGISTER.
Thanks,
Amit
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You can use controlplayback
On Jan 25, 2012 9:00 PM, Eyal e...@mcr-m.com wrote:
Hi,
How can I play a sound file from the middle and end it after a certain
number of seconds?
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Whats the error msg
On Dec 22, 2011 5:45 AM, Klaverstyn, David C
david.klavers...@intergraph.com wrote:
Hi All,
** **
I have installed newt and newt_devel but dahdi_tool will not
compile/install. I’m trying this with
dahdi-linux-complete-2.5.0.2+2.5.0.2. does anyone have any
/listinfo/asterisk-users
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Hi
You can make the call from all. For that u need not to register but to
receive the call you need to register one and that can be done by any one
iphone app
On Nov 15, 2011 7:03 PM, Faraj Khasib fkha...@iconnecths.com wrote:
Hi guys,
I want to ask if its possible to make calls using one SIP
Remove all other codec
On Nov 15, 2011 8:17 PM, Jaap Winius jwin...@umrk.nl wrote:
Hi folks,
How can I take advantage of a high-bandwidth CODEC, like G.722, for
internal communications at my site, but use G.711 (alaw/ulaw) for all other
outgoing calls? I need G.711 to support Inband DTMF
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Hi you can use Absoulte timeout to set the time limit feature for the
channel
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Hello members,
I am looking for USA incoming DID which can be registered on softphone/IP
Phone/ Pap2 devices.
The DID will only be required to receive inbound calls and no outbound
calls.
Let me know your best per month prices/cost for the above.
Regards,
Amit Mehta
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Try sipdroid for android
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Dear All
Can anyone let me know where i can free sound file whcih i can use for
system monitoring alrams.
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Dear All
Can anyone let me know where i can free sound file whcih i can use for
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is changed to kamailio I guess. It had a fork, but now they have merged
together.
Thank You
Amit Nepal
Systems Administrator
Phoenix Internet
Phone: 602-385-0731
602-234-0917#112
http://www.phoenixinternet.net
On 3/4/2011 11:49 AM, Steve Edwards wrote:
I'm starting a new project similar
.
Thank You
Amit Nepal
Systems Administrator
Phoenix Internet
Phone: 602-385-0731
602-234-0917#112
http://www.phoenixinternet.net
On 1/20/2011 4:11 PM, Bryant Zimmerman wrote:
Amit
Make sure that the trunk you have between the two servers has the t.38
enabled on it. Do you have any NAT
I have a setup of asterisk 1.6 in one box and asteirsk 1.4 in another. I
can send recieve faxes from both boxes fine to and from pstn. But the
faxing between 1.6 and 1.4 extensions does fail. Any ideas please ?
--
Thank You
Amit Nepal
1.4 and ast 1.6.
ATA (T.38 capable) AST 1.6 AUDIO CODEAST
1.4ATA (t.38 Capable)
Thank You
Amit Nepal
On 1/20/2011 1:56 PM, David Backeberg wrote:
On Thu, Jan 20, 2011 at 3:14 PM, Amit Nepalami...@phoenixinternet.net wrote:
I have a setup of asterisk 1.6 in one box and asteirsk 1.4 in another
Yes Tom,
I am sending via the PSTN gateway which is audio code in my case.
Thank You
Amit Nepal
On 1/20/2011 3:07 PM, Tom Rymes wrote:
On 01/20/2011 4:26 PM, Amit Nepal wrote:
I have an Audio code gateway between two asterisk servers. The audio
code has PRI connected for PSTN. I can
. We want to disable this codecs, but form available GUI we not able to
see any option to disble it.
If anybody having any experince with this device plz share the same to
disable the scuh codecs.
Regards
Amit
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application need to isntall/integrate to work for speech to test. Please
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Hello,
I am looking for Voip providers for voip minutes to Mali(South Africa)
Kindly provide the ratesheet for the same.
Regards,
Amit Mehta
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Hi Gareth,
Thanks for the swift reply.
Kindly provide A-Z price list.
Regards,
Amit
On Mon, Jul 19, 2010 at 3:42 PM, Gareth Blades
list-aster...@skycomuk.comwrote:
amit mehta wrote:
Hello,
I am looking for Voip providers for voip minutes to Mali(South Africa)
Kindly provide
Dear ALl
Can we use Asterisk for only for transcoding?. if yes how many concurent
call we can transcode with help of Astetrisk?
For this we only need to config SIP.conf or any other file too.
Thanks
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Hi
Can someone help me in configuring say.conf file for Indian Languages?
I want to play numbers and dates in regional languages. I need if for
Bengali, Kannada, Telugu, Hindi, Marathi, Malayalam, Tamil, Gujrathi.
Thanks Regards,
Amit Patkar
sessions. No call
recording. No IVR. Pure gateway functionality. Can I achieve this capacity
with given server configuration?
If not, what kind of server is required to achieve this capacity.
Has anyone done this? Please share results.
Thanks Regards,
Amit Patkar
sipp is the device :), we are trying to test some implementation of tel uri. So
was wondering if someone has already supported tel uri.
Regards,
Amit
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olle E
Hi All,
Can some help me with how to run Asterisk with gdb.
Thanks,
Amit
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(${EPOCH},,%Y%m%d-%H%M%S)}-${CALLERID(num)})
*which results after completion is time-calledid number
how we can set file name which give us result with extension who attend the
call even that extension is member of specific Queue, not matter agent or
static member.
Regards
Amit
can use with FLASH key on analog
phone, it not wok with such confing, its only give dial tone to callee.
Any idea how we can do with Asterisk+Audio codes analog g/w + analog phone.
Thanks
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on
growing.
Thanks
Amit
On Tue, May 19, 2009 at 1:49 AM, ContactTel Business
li...@contacttel.comwrote:
This is a global message to all to announce our callcenter / dialer /
predictive dialer / vicidial program is now open.
Codecs: G711, GSM, G729, G723
Protocols: SIP
Duration Rate : 30/6 (6/6
,minivm.conf sip.conf
2. same way for MWI on respective IP Phone what we require to config in
Asterisk 1.6 for respective config file?
Help me with config examples or settings.
Thanks in Advance.
Regards
Amit
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is answered)
9) Opensips---200ok-àAsterisk (Since the call to 7010 by
asterisk is answered)
10) Asterisk does not ACK the 200ok in step 9 and instead keeps sending
the 200ok in step 8 until maximum retransmission is reached and then the
call is hung up.
w/regards,
Amit
Hello Members,
Sorry for hijacking the earlier thread and asking the question last time.
Is anyone aware about a solution to call incoming number and dictate
the files by using Dictate feature of Asterisk used for Medical
Transcription industry.
Thanks Regards,
Amit Mehta
tags to file
when it is stored like authid,name,time.
Thanks for all the suggestions you are going to give.
Regards,
Amit
On Thu, Feb 26, 2009 at 9:53 PM, Brent Davidson br...@texascountrytitle.com
wrote:
amit mehta wrote:
Hello Members,
Sorry for hijacking the earlier thread
Hello Users,
Is anyone aware about a solution to call incoming number and dictate the
files by using Dictate feature of Asterisk used for Medical Transcription
industry.
Looking forward for help.
Thanks,
Amit
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Thanks Regards
Amit
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to
call the asterisk internal agent and the other line will call the
number that was input by the customer and bridge the call.
Hope this might help you.
Regards,
Amit Mehta
Cell: +91 9898340962
On Tue, Jan 6, 2009 at 11:41 AM, Rilawich Ango maillist...@gmail.com wrote:
Hi all,
I want to build
provide me config exmple?
I am using Asterisk 1.4.9. Plz help me
Regards
Amit Salunkhe
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Scott,
Login as root user and start asterisk by typing asterisk and then give
command asterisk -r
Amit Mehta
Cell: +91 9898340962
On Wed, Dec 10, 2008 at 8:35 PM, Scott Berry [EMAIL PROTECTED] wrote:
Have a nice day,
Scott Berry
E-mail: [EMAIL PROTECTED]
I am studying out of the book
Hello Friends,
Sorry to hijack the thread,but i am Asterisk beginner and am facing
problem with eyebeam getting registered.
If i am selecting Domain with register and receive incoming calls then
i am not able to get register but if i remove the tick then i am able
to register with the server but
help me for this.I
am using Asterisk 1.4.9 SIP channels.
Regards
Amit
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as Extenspy. But result is same. So Plz Help me.
Thanks
Amit
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as follows---
[inbound context]
exten = 1,4,Set(SPYGROUP=techsupport)
exten = 1,5,Queue(tech_support|t|||120)
[call barg context]
exten = _555,1,Authenticate(123)
exten = _555,n,chanSpy(|g(techsupport)bv(4))--- same with
extenspy cmd
Thanks Regards
Amit
solution.
Thankx,
Amit.
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is not a problem for me.
Is there anything more I need to do apart from setting outboundproxy
to my local OpenSER? Because it doesn't seem to work.
Any pointers that you provide me will greatly help. Thanks in advance.
Regards,
Amit.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED
remote domain
user to be called via our local outbound. Right?
Regards,
Amit.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Grey Man
Sent: Friday, April 18, 2008 11:53 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk
configure Asterisk to use my OpenSER as an
Outboundproxy for all outgoing call legs?
Thanks in advance,
Regards,
Amit.
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Is the Asterisk server yours? I am trying to figure out if Asterisk is in
your control and if it could be a problem at Asterisk, rather than your
SJPhone or your script, because I don't see any glaring problems in the
script.
Regards,
Amit.
-Original Message-
From: [EMAIL PROTECTED
a separate NAT.
I made a call from SJPhone to XLite and vice versa - I am getting
Audio in both directions.
I used SJPhone 1.65 on Windows, and Asterisk 1.4.17.
Your problem lies somewhere else. Your script looks just fine.
Regards,
Amit.
-Original Message-
From: [EMAIL PROTECTED]
[mailto
doing something wrong? Or is there a bug in Asterisk, wherein, while
calling out to non-locally subscribed users, it blindly trusts the notion of
their IP address when it comes to RTP.
Any help is highly appreciated.
Regards,
Amit.
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