Asterisk PBX Jobs
Asterisk PBX, like any other PBX, is a complicated subject that is best handled by experts. If you are a pro in this field, then you should bid on the many jobs at Freelancer.com.
Asterisk PBX (private branch exchange) is implementation software. Created by Mark Spencer in 1999, the software simply allows connected telephones to make calls to each other and also to connect to other services. The name is based on the symbol asterisk, (*). For Asterisk PBX to function as it should, the configurations must be on point, which is why this should be done by an expert.
Asterisk PBX is a topic that needs skill and if you are an expert in this, then you should earn money through what you know best. There are thousands of jobs posted on Freelancer.com related to Asterisk PBX and if you at a pro in this particular field, then Freelancer.com will offer you a chance to work on projects you understand. The site attracts some of the best-paying clients and offers an easy-to-use platform, where freelancers can browse and bid on jobs they are interested in. You can simply start your career in Asterisk PBX at Freelancer.com today.
Hire Asterisk PBX Developersasterisk 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 ...
I am looking for an off-the-shelf voice messaging platform like [login to view URL] has developed.
Can you create a feature so that my users call face time each other on my website? #VOIP #video
I have an Asterisk server for private use already running, and would like to acidify a Trunk using an FXO VoIP Gateway, for this is necessary to create a sip trunk in Asterisk and I do not know how to do. In my attempts, I can even connect SIP between them, but I can not complete calls.
I need a server administrator with experience on VoIP technology. FreeBPX, Asterisk, SIP Clients Cisco SIP phone provisioning and some other SIP phones. Developer most commute to office in Rupnaghar India. If you don't leave in India please don't apply.
I am using for our office a pbx with regular sip phones and some softphones. The softphones work using Bria Mobile with Push notifications enabled. However, the phone doesn’t work properly on background meaning that push notifications don’t work as expected for receiving calls. Here is the page where Bria mobile explains the settings that needs to be completed in order for push notific...
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
We'd like to allow a parked caller to press a DTMF which sends the caller to the user that originally parked them. Please provide a full project plan and how much time you'll need to accomplish this.
Hi, I have Php house made CRM, want to integrated 3CX PBX VoIP with CRM, Adding extra field in the customer profile (email) and generate report. More information are available in the attachment.
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.
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.
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 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.
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
I have the a2billing installed and Elastix MT, but cant make it works.
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.
