Voip Software Jobs

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Fixed Price Budget - Intermediate ($$) - $200 to $500 - Posted
Good day. I need someone to setup and configure Kamailio And Siremis sever Ipaddress and username will be supplied. A base ubuntu box will be provided I need to be able to do the following 1) Setup 2 sip trunks to my suppliers - this will be used as the agteways for the clients 2) Configure clients - the clients will be setup as trunks on my asterisk pbx's 3) Showen how to setup routeing for cost savings 4) Showen how to pull cdrs for billing purposes 5) The gateway will need to hangle the G729 codec, instructions on setup and liceinsing must also be provided 6) What changes have to be made for when i move the machine to site with new ip addresses etc 7) I would like to know how to see active calls through the system. Why im doing this I currently have 50+ asterisk PBXs And i have 2-3 trunks setup on each oneand im doing the routing at the pabx level. this is becoming way to hard to manage. I want to setup one trunk to my gateway and handle the routing o the gateway it this way. What will need to be tested before payment 1) Setup a trunk on a test pbx system. 2) From a call from my asterisk system i must route the call through the different providers 3) I will need to see the cdr;'s for testing 4) While doing the calls i would like to see live view of active calls If you have any questions please ask
Skills: VOIP Software Asterisk SIP VOIP Administration
Fixed-Price - Intermediate ($$) - Est. Budget: $100 - Posted
We are looking for an experienced landing page copywriter/optimizer to take a look at 1 of our existing landing page (There is 2 variations) and make advise and changes to improve the conversions. The industry for the landing pages is in the "Online Marketing Services" The landing page is http://ask-alvin.com/sem-ppc-agency-singapore/ You can propose how you want to help improve the conversion rates. You can do pure advisory with screenshot on what are the changes required. You can do advisory and execution. Etc... You propose and quote accordingly. The landing page are deployed using unbounce . Have prior experience is advantageous as you would be familiar with execution constraints. If you interested in getting some 5* feedback on your profile and potential long term work. *Shortlisted vendor will be provided the landing page urls to preview prior to project confirmation. About Us * Freelancer.com: Total 350+ Projects, 100 5* Employer Reviews, Over USD$50,000 Spend http://screencast.com/t/5fUQx8BSIEMR * Upwork/Elance : Total 220+ Projects , Over USD$40,000 Spend http://screencast.com/t/5rJfxKAo0UB Submission Requirements 1. Indicate past experience in landing page optimization for marketing firms Vendor Selection 1. Price Competitiveness based on hourly rates 2. Past Track Record and upwork History 3. All interested vendor must indicate their past experience 4. Excellent command of English
Skills: VOIP Software Customer service Email Handling Telephone Handling
Fixed-Price - Expert ($$$) - Est. Budget: $50 - Posted
When making an inbound call there is a long sound delay (5+ sec) on the webRTC client side. We need you to fix this delay issue. We will give you all the access needed to our environment after we select you for the job. You will get paid if issue is resolved. ### Problem described more in detail ### Enviroment: Asterisk 13.7.2 FreePBX 13.0.98 Our server A is connected to trunk B and to WebRTC agent C Problem in short. When making an inbound call, audio in WebRTC agent has a delay (X seconds silence before we get Audio). Step by step explanation of the problem: Note: The number before the sentence represents approximately seconds from the moment the PBX gets signal from the trunk in other words when the end user starts making an inbound call. 0 sec: End user is dialing our phone number 0 sec: Trunk B sends invite SDP to our server A. 0 sec: Server A respond with SDP 100 Trying to trunk B 0 sec: Server A sends SDP 183 Progress (immediately after 100 Trying) to trunk B 0 sec: Trunk B start sending RTP sound packets to our server A. 1 sec - 14 sec: The recipient of the call doesn't answer immediately (for instance he doesn't have time to do it) and he waits 15 seconds before answering... 15 sec: Server A sends SDP 200 (ok) to trunk B (in this moment connection is established from phone to WebRTC agent) 15 sec: Trunk B responds immediately with ACK 15 sec - 21 sec: During this period we have approximately 6 seconds of silence (this varies from 6-11 seconds) before sound comes from A to C. On A we have sound from start but is somehow not passed to WebRTC agent C. 21 sec: First sound from A to C and vice versa. 21 sec: From here on A and C both can hear each other without any problems. Upon your request we will give you access to the server or any data you might need. If you're an expert in pbx / webrtc I am sure you will nail this immediately.
Skills: VOIP Software Asterisk FreePBX SIP
Fixed Price Budget - Expert ($$$) - $50 to $100 - Posted
I have a VICIdial server fresh install and want to use it for auto-dialing bulk lists and removing disconnects from these lists automatically and without generating complaints. No script or message needed. Please only apply if you understand 100% what we are looking to do here and have suggestions for how to best accomplish it.
Skills: VOIP Software Asterisk Elastix FreePBX
Fixed Price Budget - Expert ($$$) - $200 to $400 - Posted
Need someone very familiar with the setup, customization, administration, and usage of VICIdial software to assist with the following list of items. Please be sure to read through the entire post and understand clearly the entire scope of this listing. - Need the system to be completely configured and ready for production usage as soon as humanly possible - ** ANY AND ALL IMPROVEMENTS THAT CAN BE MADE TO THE SYSTEM FOR SCALE AND PERFORMANCE ARE APPRECIATED - THE IDEAL CANDIDATE WILL HAVE SEVERAL TIPS FOR US TO IMPROVE THE OPERATION AND WILL BE ABLE TO WALK US THROUGH THE MAIN CONCEPTS SO WE HAVE A BETTER UNDERSTANDING OF THE SYSTEM ** Currently, we have a mostly fresh install of VICIdial Express with test campaign and lists loaded up and most of the basic configuration completed. We need the system setup completed for inbound and outbound campaigns and everything working as soon as possible. We would like to have an expert look over the configuration and finish up the remaining steps ASAP so that we can begin production operations by the end of business day Pacific Standard time tomorrow. Right away for this initial job, we need the following main items completed as soon as possible: - VICIdial configured to work with our carrier with both SIP and DID - 5 Agent profiles/phones created completely and ready for use in campaigns - Successful inbound and outbound call tests with integrated ZoIPer web phone on Vici agent UI - Template full production-ready campaign setup that we can easily copy and change the contact list - Any improved agent interface that you have that would be nice, will be addressing that further after system is dialing and production-ready - Setup and test incoming call-routing for inbound campaigns - Setup press-1 campaign for outbound dialing on the test lists - Test agent flow for calls inbound and outbound and make sure everything is working properly We will be scaling up to approximately 30-50 agents in the short term future, so any optimizations that can be made for that level of volume are needed as well during configuration. After the calls are all working and routing is set, we will be integrating vTiger with our install and will need help to improve the following integrations between them: - Disposition text comments from comments box need to transfer to the corresponding lead in Vtiger. - Call recordings from VICIdial are linked and accessible on Vtiger. - Direct calling from Vtiger (integrate and make sure it works 100% properly with our hosted version of ZoIPer web) Looking for a smart candidate with good availability and English communication It will help if you are a perfectionist and a patient person ready to assist on Skype. Please get in touch we are looking for somebody skilled in this area who would be available on short notice. We are looking to establish long lasting relationships. After the initial configuration is done and everything is working properly in production, we will be looking for an experienced and highly skilled VICI developer who also has good design taste and can customize the Admin and agent interface for VICIdial extensively to be responsive and modernly pleasing to the eye - utilizing Bootstrap, CSS, PHP and other modern web libraries. You are welcome to bid with or without this additional design work, and we can negotiate the project bid based on what work you believe you can accomplish best. Ongoing administration and support will be needed and you'll be the first person we turn to if you can demonstrate your knowledge of the system and its more complex configuration options.
Skills: VOIP Software Asterisk Call Center Management CRM
Fixed-Price - Intermediate ($$) - Est. Budget: $20 - Posted
I need a person to look up call centers across South America (with focus on Mexico, Colombia and Brazil), find the decision maker that selects the voice provider for their termination. These call centers will either call back to the USA or back into South American Countries. I am looking for a list of 50 names to start. If results are good, I will ask for more. These leads do NOT have to be English speakers as they will be targeted in Spanish. I will need the name of call center, name of decision maker, title and email and LinkedIn profile. This person is usually responsible of selecting a VoIP provider for their SIP Trunking needs. The right candidate for this job has some knowledge of VoIP and Call Centers. List can be submitted in Excel format. Necesito una persona para buscar los centros de llamadas en América del Sur (con un enfoque en México, Colombia y Brasil), encontrar la persona que toma las decisiones para selecionar el proveedor de voz para su terminación/llamadas. Estos centros de llamadas generalmente llaman a los EE.UU. y tambien a otros países de América del Sur. Busco a una lista de 50 nombres para empezar. Si los resultados son buenos, voy a pedir más. El contacto no tiene que hablar ingles ya que seran parte de una cmpana de marketing en español. Voy a necesitar el nombre de centro de llamadas, el nombre de quien toma las decisiones, el título y el correo electrónico y el perfil de LinkedIn. Esta persona es generalmente responsable de la selección de un proveedor de VoIP para sus necesidades de SIP Trunking. El candidato adecuado para este trabajo tiene un conocimiento de los centros de llamadas, VoIP y como operan dichos centros de forma general. La lista puede ser presentada en formato Excel.
Skills: VOIP Software Lead generation Outbound Sales Spanish
Fixed-Price - Expert ($$$) - Est. Budget: $300 - Posted
We have an old PBX system that is running on PIAF. We want to migrate our current setup over to something new on a new computer with better performance. We would like advise on what open source system we should migrate to. Please explain why you think it is a good fit. Once we decide we would also need you to set up the new system and migrate everything over. We would also need a backup system install that uploads backups to the cloud. Office setup: We have Sales Department, Accounting department, purchasing department and warehouse. Sales has 5 phones, Purchasing is 2 phones Accounting is 2 phones. Warehouse is 1 phone and has an intercom extension. We have an IVR set up with our Business hours. We have a total of 10 extensions. Please as any questions needed. Thanks Joe
Skills: VOIP Software Asterisk Elastix FreePBX