VoIP Jobs

i have a server at home sitting around doing nothing. i would like to know if i can run a reliable iptv service from this server

I require a ccna trainer who can teach online , you must have industry experience and be able to design labs to the highest quality and have good speaking [login to view URL] should be able to work either 10 to 14:00 or/and 18:00 to 22:00 UK time monday, tuesday, wednesday, Thursday and weekends , this days are all flexible you can choose which days you want to commit to

i have voip device and i have small router it have 32mb ram os Openwrt i want to run 32calls so you need to use SIP2SIP register USA SIP server >>Bangladesh Openwrt Router>>>VOIP Device we cant use asterisk cz its need too much space i think best libre [login to view URL] or yate or besip [login to view URL] up to you as you like i just need pass a call on g729 codec under 15KBPS n...

Step 1: Homepage with logins Step2: Login page with details step3: VOIP or Live Stream features

We look for Developers skilled to programm IVR's - Design Modules - Skilled in JS VoiceIP - PHP - Nuance - and more. We want to built a cognitive IVR Speech recognition. Dynamic call flow pre uploaded audio. Multiple call flows random play on incomming calls.

Application that allows people to instantly phone a specialist, on a pay-per-minute basis, to receive recommendations

I have 2 FreePBX Servers [A & B] Server A: On Cloud And SIP Trunk has been created SSL is configured by "Let's Encrypt" Server B: On Local And SIP Trunk has been configured SSL is configured by "Self Signed" Result: Server A - SIP Trunk is appeared as not registered Server B - SIP Trunk is appeared as registered! I need to get all Registered, and the call is go thr...

We have a company based in kuwait and we want to set a " Autodial" management system for our sales team of 5 people where all the numbers will be uploaded or via CRM which we already have called HIGHRISE CRM ......We need our sales team should freeze there time and mandatory should call 3 hours non stop with all the customers which is assigned by the system to each extensions call will b...

configuracion de internos y de central asterisk

Hi, we need a professional Asterisk / freepbx sysadmin who can fix the nating in dmz using opnsense. currently we have done 90% of work Lan 192.168.X.0/24 WAN A.B.C.D (static public) DMZ 192.168.Y.0/24 PBX is here (also VIP to L.M.N.P public IP) using 1:1 Nating now the odd thing is that we are using linphone as softphone and we are having problem while being both client outside over 3G etc. call...

we wish to have someone connect and configure our [login to view URL] to our SIP and Trunk

I have a FreePBX/Asterisk System working at Amazon. I can access it directly or via a VPN. Normal telephony works as expected. 1) Trying to get WebRTC phone (via the UCP) working 2) Trying to integrate external WebRTC Can you help debug?

I need someone to install vicidial on one of our servers.

International Call center campaign : Australian Solar appointment setting campaign. Per seen appointment $35 AUD. Sale close by client commission will be your. Centers just have to book appointments. Data will be provided by client. Training will be provided by client Directly in your call center. Dialer and voip provided. Long term campaign. Weekly Payout. Interested Call centers and cal...

Please review requirements below: REQUIREMENTS : -we use API with plivo -We need dial pad after agent places call, currently agents cannot input extension. -We need an autodialer from the list we input (currently agents need to dial manually) and plivo doesn't have auto dial capability (CHECK on this) if not we will have to go open source with asterisk pbx, -Company with PBX need to be abl...

We have an asterisk PBX integrated with Zoho CRM but it's delivering the channel ID instead of the dialed number to the the CRM extension. We need some one to fix the coding of the asterisk PBX to deliver the correct needed information

Necesito servidor de canales para RESTREAMING IPTV

We are looking for a person that can write some scripts to integrate our Free PBX server to integrate with a hotel property Management system.

We Need somene that have some skills over ISSABEL Platform VOIP , Asterisk and Call Center Platform . We need to implemented some news features into reports database .

Voip Calling App with unlimited Bandwidth, It will Be used for calling East African Countries, it must be functional.

I want to create a ARI program that will play a message from URL in cloud and record the sentence spoken in channel and repeat what the person spoke after record is done Key thing here is I do not want any button pressing...Using ARI functions code should detetc when a person has spoken and when there is silence of more than 2 secs , consider the recording done Ideally i would like to implement t...

Hi Dear Sir How Are You. Dear Sir I want to develop a soft switch from Asterisk PBX. Please Contact me and Give me your contact number Thanks [Removed by Freelancer.com Admin - please see Section 13 of our Terms and Conditions]

Hi Dear Sir How Are You. Dear Sir I want to develop a soft switch from Asterisk PBX. Please Contact me and Give me your contact number Thanks [Removed by Freelancer.com Admin - please see Section 13 of our Terms and Conditions]

Asterisk based Soft Switch [Removed by Freelancer.com Admin - please see Section 13 of our Terms and Conditions]

Looking for a Tech Blogger to cover all the topics around VoIP, SIP trunking, DIDs, different countries legislations and Etc. Candidate needs to be familiar with the industry and love tech-savvy content Please when apply, send your samples

We need to bridge standard SIP calls to/from our iOs/Android app written in Adobe AIR Actionscript. In other words: handle the RTP media part of SIP to/from spk/mic. If you don't know by now what is required, PLEASE DO NOT RESPOND! Fixed payment $200.

I am using vicidial for doing automated robocalls at night when the B2B customer is not open and I will get their voicemail. I have 110,000 phone numbers to call bi-weekly. The problem I need to overcome is detection as to whether I am calling: 1. a basic residential type answering machine or 2. a PBX style answering machine which has 4 different ways of asking you to leave a message by pressing...

I have installed Dinstar GSM gateway, but I have voice quality problem. Who can solve it, please contact with me

We already having third party integration for voip out going calling for any phone number in world. We want to create our own solution and wanted to integrate Asterisk server with incoming and outgoing call center for our business in US. 1. Need a person who knows exactly how to do that using asterisk and what other external integration options we will needed to integrate. 2. Hardware requiremen...

Vorrei integrare un sistema VoIP nel mio sito web così cliente possono vedere la loro fattura e possiamo vedere i loro menù.

Voip Calling App with unlimited Bandwidth, It will Be used for calling East African Countries, it must be functional.

具备sip协议，支持30路以上fxs，可以实现点对点通信，需要提供相关技术支持

I need to create and confing server. i need to forward calls to it and to see the header and make tests can be done on linuxcpanelwinvps

Basicamente no puedo terminnar de encontrar la configuracion ideal para que el GXW4108 permita enviar y recibir llamadas administrado por una FreePBX , se logro configurar pero inmediatamente se cuelga. Yo creo que son los parametros GXW4108 lo que trae conflicto.