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...plan. Is there any recommendation? Should I split it to multiple instances and balance traffic across multiple small server instances? I can use Kamailio to balance traffic. I see many posts 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 Patkar wrote: >> Hi >> >> I plan to host Asterisk instances on A...

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, 2016 8:07:23 PM GMT+05:30, Matthew Jordan <mjordan at digium.com> wrote: >On Sat, Sep 17, 2016 at 6:26 AM, Amit Patkar <amit at avhan.com> wrote: >> Hi >> >> Is there any way to detect inactivity on channel when AsyncAGI is >used? >> I wan...

...ad you are talking about. I'm assuming with such a giant load you are making a decent profit. Buy some hefty hardware and do the architecture properly. You can rent half a rack at lots of high end datacenters for less than $1000/month. > > j > >> On 03/07/2015 12:43 AM, Amit Patkar wrote: >> Hi Jeff >> >> Are you aware of any challenges of hosting it on AWS? It will help me to work out alternate plan. Is there any recommendation? Should I split it to multiple instances and balance traffic across multiple small server instances? I can use Kamailio to balance...

...here any way to detect inactivity on channel when AsyncAGI is used? I want to detect whether application handling calls using AMI & AGI has stopped responding. Alternatively, how can dialplan check if there is any AMI user connected and decide dial plan execution? Thanks & Regards, Amit Patkar -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160917/39bebca0/attachment.html>

...expected are 500+. 2 instances will be configured for 100% redundancy. Heart beat will be used to determine active 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. Regards, Amit Patkar -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150306/9354768b/attachment.html>

...nt messaging, group chat (channels), voice and video conferencing, and screen sharing. It integrates with a variety of applications and CRMs such as Salesforce, Zoho, Zendesk, Redmine, etc. Try it out! -- Alex Epshteyn web: www.thirdlane.com ----- Original Message ----- > From: "Amit Patkar" <amit at avhan.com> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at lists.digium.com> > Sent: Saturday, April 29, 2017 9:16:05 AM > Subject: Re: [asterisk-users] softphone instead of desktop phones > > > Linphone is...

...it for the kind of load you are talking about. I'm assuming with such a giant load you are making a decent profit. Buy some hefty hardware and do the architecture properly. You can rent half a rack at lots of high end datacenters for less than $1000/month. j On 03/07/2015 12:43 AM, Amit Patkar wrote: > Hi Jeff > > Are you aware of any challenges of hosting it on AWS? It will help me > to work out alternate plan. Is there any recommendation? Should I > split it to multiple instances and balance traffic across multiple > small server instances? I can use Kamailio to ba...

...ing with such a >> giant load you are making a decent profit. Buy some hefty hardware >> and do the architecture properly. You can rent half a rack at lots >> of high end datacenters for less than $1000/month. >> >> j >> >> On 03/07/2015 12:43 AM, Amit Patkar wrote: >>> Hi Jeff >>> >>> Are you aware of any challenges of hosting it on AWS? It will help >>> me to work out alternate plan. Is there any recommendation? Should I >>> split it to multiple instances and balance traffic across multiple >>>...

...nd screen sharing. It > integrates with a variety of applications and CRMs such as Salesforce, > Zoho, > Zendesk, Redmine, etc. > > Try it out! > > > -- > > Alex Epshteyn > web: www.thirdlane.com > > > ----- Original Message ----- > > From: "Amit Patkar" <amit at avhan.com> > > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > <asterisk-users at lists.digium.com> > > Sent: Saturday, April 29, 2017 9:16:05 AM > > Subject: Re: [asterisk-users] softphone instead of desktop phones >...

...x 4E1 cards and run 480 G.711 RTP 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

...I can deal with array supporting 250 IOPS. But if I assume that Asterisk will write data on disk for every RTP packet received, 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140124/cbbbcf4b/attachment-0001.html>

Hello, I am running Asterisk 13.6.0 in an AWS instance, and I set it up with Twilio SIP trunk using pjsip_wizard.conf (nice feature!). I see that the calls actually "reach" the PBX, but for some reason, they are not caught by any of my extensions context. Here's what I observe when I test this from a non-PBX connected E164 number (a landline), say 555-666-1212. My Twilio number is

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 Patkar wrote: > Hi > > I plan to host Asterisk instances on AWS/EC2 servers. > Requirement is to run asterisk instance with transcoding (g.729 + > g.711) and full recording. Number of concurrent calls expected are > 500+. 2 instances will be configured for 100% redundancy. Heart beat...

...1}0, digits/h-${SAY:1} _e[n]um:[1-9]00 => num:${SAY:0:1}, digits/h-hundred _e[n]um:[1-9]XX => num:${SAY:0:1}, digits/h-hundred, enum:${SAY:1} [en_GB](date-base,digit-base,en-base) _[n]um:XXX => num:${SAY:0:1}, digits/hundred, vm-and, num:${SAY:1} Thanks & Regards, Amit Patkar -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130701/81753231/attachment.htm>

Hello, Iam lookong for an Softphone for iPhor oder Android smartphone using togehter with an headset. I tried Zoiper and CSipSimple but quality was bad compared to an desktop SIP phone. Is there an better softphone? Or are there softphone solutions for PC desktop MAC or Android with an headset? I want to save cost for desktop phones. thanks Thomas

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

...c. Where asparty 2 voice continue even after 25 sec even though party 2 was kept on hold. And party 2 side recording ends 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. -- Thanks & Regards, Amit Patkar -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20131029/ab10e4d5/attachment.html>

Hello Group Members, I have one question regarding SIP-I/SIP-T support in any of Asterisk versions. We have client which send SIP-I/SIP-T request can asterisk handle it and serve as a normal SIP call. As per mine analysis SIP-I/SIP-T are variant of SIP protocol with adding of ISUP/SS7 packets to original SIP request. If we want to support it then how do we implement it and support it with

Hi, I've run into a slight issue when using WebRTC and two Asterisk boxes. I am using SIPml as the test WebRTC client. My two asterisk boxes, one of them is configured for WebRTC with websockets, etc (asteriskrtc.local) and the other is just a standard asterisk server (asteriskgary.local). Dealing with just the WebRTC asterisk server, asteriskrtc.local, I am able to log in to the SIPml

hi, I have a really strange problem which is driving me crazy for days now. If I register my asterisk (tried all versions from 1.6 up to 13.x) with one sip registrar, everything works... calls go out and call come in... no 32 seconds limit. but as soon as I configure another sip registration on another server, outgoing calls drop after 32 seconds. as far as I know, there is no firewall in