Mandatory Skills: 1) Hands on expertise in SIP, RTP and RTCP 2) Good knowledge on TUN,STUN, NAT 3) Free-switch working knowledge on audio conference using SIP and Web RTC 4) Good knowledge on RTP Proxy and routed audio conferences concept where media would flow via free switch RTP Proxy 5) Working experience of High Availability and Cluster 6) SDP
Looking for Experienced agents to generate Final expense leads. Data and dialer will be provided. need to dial on Regular basis. Its an ongoing project If out come is good will be happy to Rehire again. $2 to $3 per lead will be paid. Sample Recording has been attached.
I'm looking for programming help with [login to view URL] and Kamailio. Must have experience with SIP and preferably experience with [login to view URL] library. Will need 1 - 4 hours per week for 3 - 5 weeks.
We would like you debug and fix our sip trunking /Ext to Ext calling setup to work with our providers so that we may use Elastix MT 3.0 to the fullest with multiple organizations and users per organization, Elastix MT.3.0 has been installed already just need to fix some bugs. We are currently able to have our handsets receive registration, and our
Hello, I want an article of two hundred words on the above subject line within the next one hour. Please bid if you are having knowledge about the VoIP industry. My budget is 10USD. Any bids higher than 10 USD will not be considered.
My Name is Fayssal Daoud, my company TAMQEEN works as a business consultant with small and medium size enterprises automate their workflow to render them more efficient and profitable. I am currently working on an AGILE CRM project for an Auto Repair Workshop in the UAE. The required job is as follows; 1- Design the telephony integration optimum system 2- Generate a block diagram and BOQ of requir...
MAKE UP TO ...getting any other documents that are needed Sending final documents and closing deal. Qualifications and Skills Previous telemarketing or phone skills Basic computer knowledge Dialer experience a plus Must be coachable Must be available M-F 9 A.M. EST to 5 P.M. EST you aren't necessarily working 9-6 but you must be available these hours
...0/24 (administrative network). There is a firewall that prevents connections from 4.0/24 to 5.0/24 We have VOIP SIP Server running at [login to view URL] Currently the server runs well but we will like to make possible users on 4.0/24 to access the VOIP SIP Server If you have long experience working with this type of routers and you know how to make the required
KAMAILIO SIP + MEDIA PROXY WITH SIP CAPTURE INTEGRATION INSTALL INSTRUCTIONS + CONFIG FILES + TECHNICAL SUPPORT We need a KAMAILIO CONFIGURATION accepting calls from one NIC with public IP address and redirect passing the call to 3 different SIP providers connect to 3 different NIC’s with “local” IP’s (load balance - round robin). Media must be redir...
...brokers. Requirements: Basic knowledge of life insurance/financial services NO EXCEPTIONS MUST be able to speak perfect English with little or no accent Have access to a dialer/VOIP. CRM/ or other means of making calls Outbound calls must display a Florida area code (561 or 954) Have experience with Excel files, Google Sheets and Google Drive Must BE
Client needs Large Call Centers experienced in Medicare Supplement Warm ...Agencies Must be Call Centers with 10 agents to start,Not less,and not Home based Agents. Payouts will be on Milestones at Fixed Payouts,Not Pay Per Hour. Client will provide Dialer and VoIP Minutes and Training Client will select Call-centers based on Mock-call with 10 agents
...- Aplicativo Mobile, destinado ao cliente final, desenvolvido por profissional freelancer. - Aplicativo Mobile, destinado ao profissional técnico que irá atender o cliente final, desenvolvido por profissional freelancer. O código-fonte desenvolvido será de propriedade da Percentus. O desenvolvedor que irá criar os aplicativos mobile dev...
We need a standard single zone seven server cluster installation of Kazoo. Full setup of Kazoo with working modules
Follow this link: [login to view URL] We will like to add the functionality in our mobile dialer that will enable our customer’s account to be auto recharged a variable amount once their current balance is less than 2 USD. Our app is built on the Linphone Open source and we use the MOR Switch & PayPal.
Seeking professional advice in assessing and configuring existing Free...shown. - Ability to create new Accounts, Contacts or Leads at anytime from the Call Popup results. We use softphone Linphone / Zoiper for communicating, will add dedicated VOIP phone in future plan. Offer for USD 80-100 for completing the project. Work length less than 3-5 days.
...life insurance/financial services MUST be able to speak perfect English- no heavy accents please- it won't work Have access to a dialer/VOIP. CRM/ Skype or other means of making calls You are responsible for your dialer/ subscription fees Have experience with Excel files, Google Sheets and Google Drive Must BE ABLE TO MAKE CALLS FROM 8:30 -10:30 and/
We provide VOIP telephony services to both residential and business customers. I added an idea of what represents the company: the attached infinity sign and clouds for cloud networking. I'd like to have some blue and something very catchy. Here are some ideas what you can do [login to view URL] The winner should provide PSD source
Rewrite existing App for Android and iOS adding a small extensi...GWT The features of the App include • Login and account maintenance optionally from Facebook Connect, Google , OAuth 2.0, Yahoo, AOL, etc. including OpenID connect. • Phone Dialer like Rebtel’s user interface. • Communication with Enterprise servers using JSON (mostly existing)
I am looking for a Asterisk developer to quickly update existing Asterisk code for dialer, inbound/ outbound/ press 1, and other advanced features. Need to configure advanced features as follows and show how to use. Initial work / payment, and possible ongoing work. Voice Broadcasting Interactive Voice Broadcasting ( Press-1 Campaign) Multi- Campaign
...Day per Agent. Agents MUST have experience in VICI Dialer “Call with Customer” Transfer Methods and “Leave 3 Way Call” after 2 Mins Duration. Do note that as per the TCPA The Marketing US Company(US Client) has to Provide Leads based on their SAN Number from the Federal Trade Commission(FTC) VICI Dialer and Login Credentials are provided by us(Client)
I have 3 x Cisco SG250-26P 60 pcs and 60 Voip Phones the pc's will be plugged into the phones for internet connection and access to the local network. The phones will be tagged with a different Vlan. What i need is: The cisco SG250-26P setup so i can have a the phones patched into the switch, phones to operate independently on a separate vlan (ip)
...with B to C experience calling on American consumers /companies. You will be provided with scripts, and files to call from. Pay will be $5/hr US +bonus We do NOT pay any dialer/subscription fees. We are looking to hire ONE FREELANCER AT A TIME, but will eventually need more. Our services: life insurance, mortgage protection, Medicare Supplements
.../companies. You will be provided with scripts, files to call from and ongoing mentoring. Spanish and/or French a huge plus. Pay will be $5/hr US +bonus We do NOT pay any dialer/subscription fees. We are looking to hire ONE FREELANCER AT A TIME, but will eventually need more Our services: life insurance, mortgage protection, Medicare Supplements
...are freeswitch and fusionpbx FusionPBX CDR billing requirements Problem statement Currently i have a single FusionPBX configured in a multitenant environment with different SIP trunks from multiple providers configured per tenant. Given that there are multiple providers, the upstream CDR reports are not suitable for processing the clients billing for
...3945 router and it does not work fully. I have: Cisco 3945 with Voice license 1 x 4G Samsung phone with Zoipher that registers to my SIP server over the WAN 2 x LAN connected SIP clients 1 x FreeSwitch Server on Windows You're free to change the setup, but this was working using FreeSwitch fine on my lan until i changed things over. Youll need: NAT
...Minimum Seats : 5 or more Training - 7 days and Provided Data, Dialer and VOIP Provided by client Biweekly Payments b. USA CREDIT FINANCE PAYOUT - $5 per hour per agent Minimum Seats : 5 or more Training - 1 Day and Provided Data, Dialer and VOIP Provided by client Biweekly Payments Interested call centers/BPO
We are a consultant for businesses providing them with primarily business phone service; but also, business internet connectivity as well as mobile phone service, SD-WAN, and other telecom and IoT services We need content, articles, blog posts that are relative to our business "Carrier Solutions".. Our website is www.carrier-solutions.com. I Also would
...making calls so you'll need to represent my company professionally and develop a great repore with phone prospects. Experience with Mojo dialer is a plus *Previous cold calling, appointment setting and MOJO dialer experience is a huge PLUS! To be considered for the position please send me the following 3 items: (1) your resume, (2) a copy of your
...of life insurance/financial services NO EXCEPTIONS MUST be able to speak perfect English- no heavy accents, please. Have access to a dialer/VOIP. CRM/ Skype or other means of making calls- Be responsible for any dialer/subscription fees Have experience with Excel files, Google Sheets and Google Drive Must BE ABLE TO MAKE CALLS FROM 8:30 -10:30 and/
i need someone experienced about issabel (formerly elastix) to set up an maintain our vps based sip switches. serving to multiple clients using multiple sip operators,trunk or sip user based.
I need an Avaya IP Office expert who is familiar with SIP Trunk configuration with fax server products. Currently we have the licensing installed for 4 SIP Trunk Connections, but the Avaya IP Office is responding with a SIP "Temporary Error".
...experience calling on American consumers /companies. You will be provided with scripts, files to call from and ongoing mentoring. Pay will be $5/hr US +bonus We do NOT pay any dialer/subscription fees. We are looking to hire ONE FREELANCER AT A TIME, but will eventually need more. The Premise: There are 10,000 people a day turning 65 which represents
...should have a universal form of support for mjpeg H264 formats.h264+. H265+ support IPv4 / IPv6, TCP, UDP, RTP, RTSP, RTCP, HTTP, HTTPS, DNS, DDNS, DHCP, FTP, NTP, SMTP, UPnP, SIP, SNMP, PPPoE, VLAN, 802.1 x, QoS, ONVIF Profile S & G, and to integrate via the SDK P2P, video Analytics modules [login to view URL]
Hi, we need VoIP Softphone With Shadowsocks VPN. We need Very Simple VoIP Softphone. you can use Opensource VoIP Softphone SDK & use Opensource Shadowsocks VPN SDK. If you have Great Idea then contact me. If you have not idea then you don't need contact
...APPOINTMENT/AGENT/DAY VALIDITY : QUALITY PASS PAYOUT FREQ : Bi-Monthly PAYOUT METHOD : PAYONEER, WESTERN UNION, MONEYGRAM, PAYPAL REQUIREMENT : CALL RECORDING OTHERS : DATA, DIALER & VOIP – NOT PROVIDED THIS PROJECT IS APPLICABLE ONLY FOR CALL CENTERS. Your agents will be tested if they will qualify on the program....
I have third party Linux Server, SIP/VOIP, when an IP phones registered with server and are assigned with extension number like 501,502 etc, now these phones make communication with each other by dialing their number. What I want to do is, remove the IP phone and use Raspberry module, and it should work as phone, and connecting push button switch on
Hello, I'm doing a national telemarketing campaign and need to load up a dialer with telephone numbers. I'm looking for someone that has a list of all the businesses in the United States and will sell the data to me for a one-time fee. These can't be residential numbers that could possilbly be on the Do Not Call list. Ideally, the businesses would
...of work but getting frustrated with what looks like sip/nat issue and that I couldn't find a way to be able to call between local extensions. It's mainly for personal use, I wanted a simple PBX that could handle a small group of sip devices that could call each other and that could use an external SIP gateway. The aim was to have better sound quality