Sip Jobs

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Hourly - Expert ($$$) - Est. Time: More than 6 months, 30+ hrs/week - Posted
We are looking to hire someone who is proficient with PHP-SIP and knows how to make web calls. This way when a user enters a number into the "POP UP" it will connect the call between the user and visitor. ... But we want to charge per minute and we aren't sure if twilio is the correct route. So we are looking for an open SIP for the click to call is any available. Please experts only.
Skills: SIP JavaScript Laravel Framework OpenSIPS
Hourly - Intermediate ($$) - Est. Time: Less than 1 week, Less than 10 hrs/week - Posted
And to use IP PBX, i need to integrate my voip thing with SIP account no. Currently I am using www.call2india.com voip So who can help me to setup VOIP SIP account? ... Currently I am using www.call2india.com voip So who can help me to setup VOIP SIP account? Thanks Gurmeet skype- live:gurmeet.chahal
Skills: SIP VOIP Administration VOIP Software
Hourly - Intermediate ($$) - Est. Time: Less than 1 week, Less than 10 hrs/week - Posted
We are running an customer care operation on SIP telephony currently. This means we need to have locally installed SIPclients or phone. ... We want this to be more flexible in the future and are therefor looking into integrating Twillios WebRCT client into our backend to allow agents to place calls and recieve calls from their browser (so we don't need to do any installation any more) We are looking for someone who can assist us/help us on how to add this (we are coding in PHP), and has experience into potentially also integrating existing SIP accounts into this.
Skills: SIP Twilio API
Fixed Price Budget - Entry Level ($) - $25 to $50 - Posted
I'm having difficulty setting up a sip trunk for incoming and outgoing calls from a service provider called "Mynetfone - mynetfone.com.au" I have successfully setup the Sip trunk in Elastix (Asterisk) I'm studying the FusionPBX for it's multi-tenant feature. ... I'm having difficulty setting up a sip trunk for incoming and outgoing calls from a service provider called "Mynetfone - mynetfone.com.au" I have successfully setup the Sip trunk in Elastix (Asterisk) I'm studying the FusionPBX for it's multi-tenant feature. ... Requirement : Need an expert help to setup the trunk in FusionPBX (Freeswitch) for Gateway, Inbound and Outbound Setup. I will provide you the Sip trunk settings.
Skills: SIP Freeswitch
Fixed Price Budget - Intermediate ($$) - $10 to $350 - Posted
We need to setup and e-mail2fax solution for our cloud based pbx. Fax should be work on T38 protocol aka FOIP. The idea is pretty simple and there are a few old opensource scripts out there. server will accept an email with format faxnumber@ourfaxserver.com and then sent the attached file (mainly pdf or jpeg, ability to do also office files is a welcome) We can provide a test setup VPS server.
Skills: SIP A2Billing Asterisk FreePBX
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