Re: [asterisk-users] How to enable DTLS

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,*

Re: [asterisk-users] One Way Audio with WebRTC (with external asterisk)

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

Re: [asterisk-users] One Way Audio with WebRTC (with external asterisk)

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

Re: [asterisk-users] Moving from Redfone's Fonebridge to Allo 2nd Gen PRI card

Hi Allo also offer gateways - similar to Fonebridge. Have you tried/evaluated that? It will help us to know those results. What is CPU & RAM utilization on this server? What kind of work load you run? Does it involve transcoding (codecs used)? Are these calls passed on to SIP client or these a

Re: [asterisk-users] SIP call drops after 32 seconds, but only when....

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. wrote: I have a really strange problem which is driving me crazy for days now. If I regi

Re: [asterisk-users] Asterisk does not listed to port 5060

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 <mailto:roy.gan...@gmail.com>> wrote: Hi Friends, I enco

[asterisk-users] AWS/EC2 server selection

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-- _ -- Bandwidth and Colocation

Re: [asterisk-users] AWS/EC2 server selection

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,

Re: [asterisk-users] Issues with Twilio number incoming call and context matching

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

[asterisk-users] AysncAGI - timeout

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

[asterisk-users] AsyncAGI - How to jump in dial plan when no action initiated on channel or AMI user is disconnected

Regards, Amit Patkar -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016 http://www.asterisk.org/community/astricon-user-confere

Re: [asterisk-users] AsyncAGI - How to jump in dial plan when no action initiated on channel or AMI user is disconnected

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

Re: [asterisk-users] AsyncAGI - How to jump in dial plan when no action initiated on channel or AMI user is disconnected

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, Amit Patka

Re: [asterisk-users] softphone instead of desktop phones

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 wrote: >Hello, >Iam lookong for an Softphone for iPhor oder Android smartphone using >togehter >with an headset. >I tried Zoiper and

[asterisk-users] Capacity of single instance of Asterisk

needs to be recorded. What will be impact on no of session when G729a is used? Thanks & Regards, Amit Patkar -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a

[asterisk-users] PSTN termination in Virtualized Asterisk Environment

deployment? Thanks & Regards, Amit Patkar -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk

[asterisk-users] Queries regarding FastAGI

commands will be passed by AMI session. Regards, Amit Patkar -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http

Re: [asterisk-users] Queries regarding FastAGI

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 wr

[asterisk-users] Actual DAHDI channel number

Earlier we used to get this. I am using Asterisk 1.8.11 and Dahdi 2.4 Thanks & Regards, Amit Patkar -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory

[asterisk-users] Asterisk 1.8.20 AGI function SAY DATETIME does not play anything when mode in say.conf is changed to "new"

_e[n]um:[2-9][1-9] => num:${SAY:0:1}0, digits/h-${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:${

Re: [asterisk-users] Asterisk 1.8.20 AGI function SAY DATETIME does not play anything when mode in say.conf is changed to "new"

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, Matthe

Re: [asterisk-users] Asterisk 1.8.20 AGI function SAY DATETIME does not play anything when mode in say.conf is changed to "new"

01 - state 1 (Not in use) [Jul 2 15:54:44] DEBUG[2698] devicestate.c: device 'SIP/1001' state '1' [Jul 2 15:54:44] DEBUG[2737] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x98ac7f0' [Jul 2 15:54:44] DEBUG[2737] netsock2.c: Splitting '192.168.2.

Re: [asterisk-users] Asterisk 1.8.20 AGI function SAY DATETIME does not play anything when mode in say.conf is changed to "new"

Y:1} _[n]um:XX0 => num:${SAY:0:2}, digits/billion _[n]um:XXX => num:${SAY:0:2}, digits/billion, num:${SAY:2} _[n]um:XXX0 => num:${SAY:0:3}, digits/billion _[n]um: => num:${SAY:0

[asterisk-users] Loosing synch between party 1 & party 2 voice in monitor recording

de 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 -- _ -- Bandwidth and Coloca

[asterisk-users] Asterisk 1.8.20 crashing

er found, checking channel drivers for DAHDI - pseudo [Nov 12 16:53:03] DEBUG[3573] devicestate.c: Changing state for DAHDI/pseudo - state 0 (Unknown) [Nov 12 16:53:03] DEBUG[3573] devicestate.c: device 'DAHDI/pseudo' state '0' [Nov 12 16:53:03] DEBUG[3612] app_queue.c: Device 'meetme:65001' changed to state '1' (Not in use) but we don't

Re: [asterisk-users] Capacity of single instance of Asterisk

needs 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 

Re: [asterisk-users] Capacity of single instance of Asterisk

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, S

Re: [asterisk-users] Capacity of single instance of Asterisk

Hi, Appreciate everyone for your valuable inputs. All these inputs provided by you are really useful. Thanks & Regards, Amit Patkar -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Aste

[asterisk-users] Maximum call handling capacity on single server

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 Pa

[asterisk-users] say.conf implementation of Indian Languages to play numbers and dates

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 Pa