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...wered 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 want to detect whether application handling calls using AMI & AGI >has >> stopped responding. > >What do you mean by "stopped responding"?...

Hi I have a server with Quad Core Xeon 2.4GHz and 4GB RAM. I want to use it for PSTN-IP gateway. What is the maximum call handling capacity I can achieve with this server? I want at least 480 concurrent PSTN-IP calls. That mean I will have to install minimum 4 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

Hi I am using following say.conf file. Its a default file, which comes with Asterisk installation. When I call SAY DATETIME AGI function, it simply returns without playing date & time. Where as if I use mode=old setting, it works. Is this a bug or mode=new is not implemented for SAY DATETIME AGI function? [general] mode=new ; method for playing numbers and dates ;

...(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 available for all major...

Hi Is there 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:

...BUG[23332] manager.c: Running action 'Redirect' [Nov 12 16:53:02] DEBUG[3612] app_queue.c: Device 'SIP/1003' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Nov 12 16:53:02] VERBOSE[27410] pbx.c: == Spawn extension (avhan, 100, 1) exited non-zero on 'DAHDI/i1/110-8f' [Nov 12 16:53:02] DEBUG[27410] channel.c: Soft-Hanging up channel 'DAHDI/i1/110-8f' [Nov 12 16:53:02] DEBUG[27410] channel.c: Hanging up channel 'DAHDI/i1/110-8f' [Nov 12 16:53:02] DEBUG[27410] chan_dahdi.c: dahdi_hangup(DAHDI...

...; 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 > > > > > &gt...

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

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

...debug on' and see what they (Asterisk and the registrar) are talking about just before the call drops. > > -- > > marie > > > > > ------------------------------ > > Message: 6 > Date: Thu, 27 Nov 2014 10:49:23 +0530 > From: Amit Patkar <amit at avhan.com> > To: asterisk-users at lists.digium.com > Subject: Re: [asterisk-users] SIP call drops after 32 seconds, but > only when.... > Message-ID: <5476B45B.4020400 at avhan.com> > Content-Type: text/plain; charset="iso-8859-1"; Format="flowed" >...

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

Hi We have come across a situation where we are loosing synch of party 1 & party 2 voice in call recording. Here is the scenario Party 1 initiate a call to Party 2 using AMI commands When both calls are connected, we bridge these 2 calls. Then we start recording of this bridged call using AMI Monitor command. Monitor command is invoked on Party 1 only. Then we put party 2 on hold. We

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 will be used to determine active instance. How should I choose EC2 instance? How many vCPU, RAM should be selected? I am assuming that

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 traffic. I see many posts referring to AWS deployment. Please help me to choose AWS server instance. *Thanks & Regards,*

Hi What are the disk IOPS required for Asterisk call recording? I am trying to find out number of disks required in RAID array to record 500 calls. Is there any formula to calculate IOPS required by Asterisk call recording? This will help me to find IOPS for different scale. If I assume that Asterisk will write data on disk every second for each call, I will need disk array to support minimum