VoIP Jobs

asterisk person to work on an existing product to fix and update the product knowledge needed: PHP Asterisk database

We have iPECS-100 behind Chekpoint firewall. All customers use soft phones clients. Have periodically experience communication problems. Need a person for a one-time or permanent job

well , we need basically bandwith optimization. in client end they have termination gateways, each gateway can carry 30 simultaneous calls. and all gateway under DHCP network . we take a server which run in static IP . all client should send calls to server IP from his SIP server and we need to create separate account for each gateway . and calls should send to specific termination . call should ...

Can you create a feature so that my users call face time each other on my website? #VOIP #video

System requirements Registration screen Logon screen online Users screen Server php and database mysql Through the application you can make a voice call between registered users on the system The user is activated after registration through the database The IP number, MAC address, device type and user name are stored in the database The call is secure and encrypted between the parties Simple appli...

i need help in my freepbx i m not getting call waitng options

Only for gurus familiar with Bitrix who can setp Bitrix application to work with our PBX Need to use Telephony REST api

a2billing setup, calling card, PBX, Auto dialing, SMS

I would like someone to config a Cisco IAD 2435 for MGCP. I dont want to use Cisco call manager. I can give you remote access to this device for testing.

I'm stating voip system and calling card with soft-phone service. I want some one who can install freeswitch with fusionpbx billing solution.

i have a Linux server its running on Apache tomcat i have some .jsp file on webapp forder its an api for vos3000 like i want to insert update delete but i dont know how to do you have to fixed it check the attach file here is all info i dont release any fund without test

Looking for someone to pass asterisk logs to, to determine why some calls are dropped or why calls are not routing properly.

Configurar um server opensips/kamailio para controle de ddrs,chamadas para gateways externos , chamadas entre ramais , cdr com interface grafica.

i need to install yate on openwrt and pass calls server to my gateway we pass call useing sip to sip if you can make it please bid

Hello. We are Looking for Call Center Solution based on Freeswitch (Multitenant) (Not Vicidial or GoAutodial) along with Source Code. If anyone have ready, contact us and provide us demo. We will purchase instantly. For more information you can chat with us. Thank you

we need a phone system based on twilio platform.

I need an android application for video streaming. (Bigo live, [login to view URL] kind of app). Please review specifications document before bidding.

Ringless voice mail broadcast service. Automatically delivers your pre-recorded voice message straight into consumers’ voicemail, without ringing the telephone. *** Replicate the voicecasting capabilities found here - [login to view URL] called voicecasting*** Voicelogic has a service that allows you to make a call that will go straight to voicemail without ringing the customer. I want to ...

Need someone to configure Asterisk secure websocket for javascript library integration (such as jssip or SIPml). Asterisk is installed. Access will be provided.

1. REGISTER + SUBSCRIBE to SIP server, with authentication 2. Accept INVITES 3. play inbound SIP packets and convert microphone to outbound sip That's it. If you've done it before, it's 1 hour work. Fixed payment $100. We need Actionscript in AIR.

Signalling work with CISCO CUCM Understanding VOIP - SIP (including Blind and Attended Transfer implementation) - VOIP Call analysis – Wire Shark or similar - Visual Basic Script language - Proprietary Cisco SIP protocol extensions - Cisco CUCM - Call flows of the Attended Call Transfer - Cisco Finesse handling of the Call Transfer Budget as outlined Deadline: 4 weeks

Requirement two-way audio/video SIP based communications form the mobile phones to Asterisks on ARM based targets and web browser targets

Hi, i need assistance of an Asterisk expert on daily basis. I am looking for an experienced expert coder only. 5 star review will be provided for some one Good job! Happy Bidding!

I am looking for a SNOM phone guru that can create a dial plan for me with the following: 4 Digit Internal extension calling (extensions range from 2000-8000 & 9800) 7 Digit Local Calling 10 digit local calling (we are in Canada) 11 digit long distance calling Toll Free 800/855/844/866/877/888 International Calling with 011 Emergency calling with 911 Directory Assistance with 411 We use &quo...

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...

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

Preciso que configure um ATCOM Modelo IP2G4A como Trunk de um PBX Asterisk Issabel que já está instalado. Todos estão na mesma LAN já com a rede configurada.

Hola con todos Alguien ha podido integar infusionsoft con centrales telefonicas ip como Elastix or Supermicro-Asterisk. La integración debe permitir : - Conocer duración de llamada - Grabar llamada - Origen de la llamada - Numero de llamadas por Gestor y por campaña de llamadas - Consolidación general de las llamadas Para controlar al gestor de llamadas necesitamos...

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