Sip Jobs

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Fixed Price Budget - Intermediate ($$) - $200 to $500 - Posted
A base ubuntu box will be provided I need to be able to do the following 1) Setup 2 sip trunks to my suppliers - this will be used as the agteways for the clients 2) Configure clients - the clients will be setup as trunks on my asterisk pbx's 3) Showen how to setup routeing for cost savings 4) Showen how to pull cdrs for billing purposes 5) The gateway will need to hangle the G729 codec, instructions on setup and liceinsing must also be provided 6) What changes have to be made for when i move the machine to site with new ip addresses etc 7) I would like to know how to see active calls through the system.
Skills: SIP Asterisk VOIP Administration VOIP Software
Fixed-Price - Expert ($$$) - 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: SIP Asterisk FreePBX VOIP Administration
Fixed Price Budget - Expert ($$$) - $50 to $100 - Posted
I have a VICIdial server fresh install and want to use it for auto-dialing bulk lists and removing disconnects from these lists automatically and without generating complaints. No script or message needed. Please only apply if you understand 100% what we are looking to do here and have suggestions for how to best accomplish it.
Skills: SIP Asterisk Elastix FreePBX
Fixed Price Budget - Expert ($$$) - $200 to $400 - Posted
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.
Skills: SIP Asterisk Call Center Management CRM
Fixed-Price - Intermediate ($$) - Est. Budget: $30 - Posted
I have Ubuntu server on which FreeSwitch VoIP conference calls I organise from time to time. I also have DID SIP Trunk that will allow simultaneous calls to a number that I have. ... I also have DID SIP Trunk that will allow simultaneous calls to a number that I have. I want the incoming traffic of DID SIP Trunks to be configured to the FreeSwitch server on my Ubuntu server so that incoming calls will get the ivr prompt and ask for conference number/PIN before it gets connected to the right conference call.
Skills: SIP Freeswitch VOIP Software
Hourly - Expert ($$$) - Est. Time: Less than 1 week, 10-30 hrs/week - Posted
We want an expert on FreeSwitch (Fussion PBX) who can help us on Configuration of our Remote AWS FreeSwitch (Fusion PBX) Server to our GoIP GSM Gateway. Job work: 1) Setting up outbound and inbound call for example Line one is allocated to FreeSwitch Extension 1000 then all outbound and Inbound calls from/on line 1 comes are done via FreeSwitch Extension 1000 and Direct dialing number from FreeSwitch Extension 1000 are dialed directly from Line 1. Simillary way Line 2 for 1001 Line 3 for 1002 till line 8. 2) Setting up FreeSwitch on Windows server 3) Caller ID Configuration.
Skills: SIP Freeswitch VOIP Software
Hourly - Entry Level ($) - Est. Time: Less than 1 week, Less than 10 hrs/week - Posted
Hello, We're in need of a Freeswitch expert to consult with in establishing extremely cheap calling or free calling from USA to the Philippines for family and friends and businesses. We currently have 1 installed but not configured Freeswitch Server with FusionPBX in USA. We are in need of a server provider in Southeast Asia or in the Philippines for the other half of the project, we can install Freeswitch and FusionPBX, we only require consultation and/or configuration assistance in connecting the 2 servers. PROJECT SUMMARIZED: Freeswitch Consultation Establish connection with Southeast Asia Freeswitch PBX Server with USA Freeswitch PBX Server to allow for cheap or free calling between the 2 countries. Please offer a soft negotiable offer and we will continue discussion about the project and costs privately before offering. Thank you.
Skills: SIP Freeswitch VOIP Administration VOIP Software