FreeSwitch

Fusionpbx - Remote Extensions No Audio!
Dear All, I have a fresh installed FusionPBX on a Cloud [Ubuntu 20.04] and I have configured it to use TLS And the remote registered extensions haven't audio Note: the remote registered extensions are behind a NAT and are located in a country is restricted the SIP traffic so I should use TLS with SRTP to bypass the SIP blockage. Note: Award will be to the one have an experience only to complete this project successful (no wasting time) Thanks…
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Freeswitch Failover On Certain Termination Causes
Hi I have a quite straightforward freeswitch configuration and I'm currently using freeswitch's failover feature with the directive: ... it works decently but I'd like to be able to failover (to the next trunk) only on a given list of 'dispositions' and/or 'hangup causes' to be more clean, i'd like to discriminate based on the following variables: ${originate_disposition} ${endpoint_disposition} ${hangup_cause} ${last_bridge_hangup_cause} ${bridge_hangup_cause} please contact me if you are a freeswitch billing expert and consider that I'm open…
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Freeswitch Expert Voip
I have an occasionally issue with audio between Freeswitch and our PBX with PBX (Sip Wise) . CAll enter Freeswitch and then continue to PBX. This issue is that we cannot hear a greetings from our PBX and it looks more random since in most of the cases it is working fine. We are looking for expert who can analyze and fix the issue within the Freeswitch.
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Sbc And Mediaproxy
We need to install and configure an SBC (opensis or kamalio) in front of our softswitch, the SBC should relay the registrations to our softswitch/pbx on a local interface, proxy the calls to them and proxy the media through a mediaproxy server of your choice, as you will install and configure it as well. Let me know if you have any questions
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Astpp Customization
I have an old astpp customizations and I need to update my customizations on the new server. I have the file with all the customizations and file edited, just need someone to re-apply on the new server.
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Cloud Based Conference Bridge
Need to create a super scaleable conference bridge using an open source PBX. The user management and auth will be handled separately. Knowledge of Asterisk or FreeSwitch is relevant.
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Freeswitch Mod_gsmopen Setup
I'm using freeswitch and want to configure mod_gsmopen so calls can go via sim
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Opensips Astpp Community Edition
I need to escalate my ASTPP server, putting an opensips in front to send calls to more freeswitch servers. I need to handle a minumum of 150 CPS with this setup. Maybe 8 freeswitch servers or more.
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Part-time Job At Home
Due to the company's business development, a large number of clerical jobs are now recruited from the society, and customer service processes orders. All orders are from Taobao and gaming platforms. The work is simple and easy to operate, and someone guides the whole process. Do not delay normal work and study! It only takes one or two hours a day. The salary is 40 dollars/hour, negotiable! 1: Bring your own mobile phone to work, regardless of gender, but you…
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Need Sms Platform For Text To Voice Call
hello, i need a sms platform , where my client send only DIGIT sms like , 123456 with user mobile number and this platform convert it to voice and dial that user number and play the convert file.
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Fusionpbx - Build Custom Module
We are looking for someone slightly familiar with FusionPBX to build us a module. We want to add to device_edit.php to connect to Polycom’s ZTP API. When a new MAC address is added, it will need to connect to the polycom ZTP (zero touch provisioning) API to add the MAC to their server along with some other provisioning information. More details to be provided once I find the right developer If we are happy with your work, we have other…
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