similar to: Working on web based IVR Designer for asterisk and Freeswitch

@Olivior I agree that seting up WebRTC is hard, however when done it is smooth to use. For replication you can build RPMs with working configurations. Regarding stability, it is not being used widly, so can't say it is mature. However we have no complain so far regarding audio or connectivity. sometime we provide support for "allow media / mic" type issues, but you know it is

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 then those soft / sip phone. the big plus is that they are easy to customize and developer can use the power of browser and web to build / offer features which are not possible with regular phones. Regarding your concern about BLF or call

Following new features are now supported by asterisk based telemarketing software Auto subscription / registration after call recipient press a key in voice broadcasting https://www.ictbroadcast.com/Subscription-Campaign-to-automatically-register-customers-at-website-with-Voice-broadcasting-Autodialer There will be restriction to call a number in off time accordingly to timezone of

Though asterisk support AMD which is based on silence detection but I did not found support of tone / beep detection in asterisk to record a voice message for answering machines after detecting tone Will appreciate any help in this regard Best Regards *Tahir Almas* Managing Partner ICT Innovations http://www.ictinnovations.com Leveraging open source in ICT Unified Communication Telemarketing

Sorry , I forget it for another monitoring tool monit that we have used in our production systems to restart asterisk in case of asterisk crash or halt. I have attached a monit configuration for your reference. it will work almost in all cases This configuration will check Asterisk for following 1. will check for Asterisk process. 2. will check Asterisk via AMI 3. will check

Why not to use Fail2ban https://www.voip-info.org/wiki/view/Fail2Ban+%28with+iptables%29+And+Asterisk *Tahir Almas* Managing Partner ICT Innovations http://www.ictinnovations.com http://www.ictbroadcast.com Leveraging open source in ICT On Tue, Feb 21, 2017 at 12:28 AM, Tzafrir Cohen <tzafrir.cohen at xorcom.com> wrote: > On Mon, Feb 20, 2017 at 11:36:24AM -0300, Victor

what is the difference between FreeSwitch and Asterisk , whitch one is more scalable and reliable? _________________________________________________________________ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, Please help me understand the following applications and what are its advantages if we compare between each of them. Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. Regards, Kaushal -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120103/ffad2be6/attachment.htm>

On 08/06/2013 07:40 PM, Jorge N??ez wrote: > Hi I am trying to use icecast to broadcast a realtime conference from > freeswitch. But I am having a delay like 20 seconds then I reduced it to > 12s. But I don't know if somebody can help me how to reduce it as lower > as possible. > > Thanks > > Jorge Jorge, first I'd like to know what you did to reduce the delay

This is slightly OT in that it isn't specifically *-related, but I was wondering what the members of the * user community felt about these two subjects. I've been perusing the OpenPBX.org mail list and the current hot topic is the fact that their project has come to a grinding halt. They are concerned that they don't have enough people working on their project. They feel that * has

Hello, One of our client company is providing hosted contact center solutions with Cisco IPCC. To keep the Call Recording cost at low, they are planning to use Asterisk / Free Switch. Can anyone integrate Cisco IPCC with Asterisk for call recording ? Regards, Kashif Naeem Business Development Manager Hadi Telecom www.haditelecom.com Cell: +92 (0)345 4226006 Office: +92 (0)42 5692766 Email:

Thanks for your answer, well I changed this parameters on icecast.xml and the the delay reduce from 20s to 12s <burst-on-connect>0</burst-on-connect> <burst-size>4096</burst-size> Well I was trying to reproduce mp3 and ogg but both have 12 s of delay. How can I reduce to maybe 1 or 2 seconds. -------------- next part -------------- An HTML attachment was

Hi I am trying to use icecast to broadcast a realtime conference from freeswitch. But I am having a delay like 20 seconds then I reduced it to 12s. But I don't know if somebody can help me how to reduce it as lower as possible. Thanks Jorge -------------- next part -------------- An HTML attachment was scrubbed... URL:

what-he-said On 08/07/2013 06:48 AM, Basil Mohamed Gohar wrote: > On 08/06/2013 07:40 PM, Jorge N??ez wrote: >> Hi I am trying to use icecast to broadcast a realtime conference from >> freeswitch. But I am having a delay like 20 seconds then I reduced it to >> 12s. But I don't know if somebody can help me how to reduce it as lower >> as possible. >> >>

We tried to use the CURL fn to POST json, but it's sent as form data and there seems no support for changing the Content-Type header. We switched to invoking curl in the shell. All the documentation I could find says there is just one parameter for the url and an optional second for POST body. Is there an undocumented way to set Content-Type? -------------- next part -------------- An HTML

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

Hi I am trying to deploy freeswitch with Digium TE121 card for my office setup, but it is continuously showing Signaling is up and channels are down except D channel. Our Architecture is like We have freeswitch installed with libpri1.4 and Dahdi. I am from India and here we are having E1 trunk. Dahdi Configuration is cat system.conf # Autogenerated by /usr/sbin/dahdi_genconf on Wed Aug 7

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

The Asterisk Development Team would like to announce the release of Asterisk 13.28.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.28.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release:

The Asterisk Development Team would like to announce the release of Asterisk 16.5.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 16.5.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release: