Sip Jobs

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Hourly - Intermediate ($$) - Est. Time: Less than 1 month, Less than 10 hrs/week - Posted
Any experience with FreeSwitch is a huge asset, but not necessarily required. At the very least, a healthy understanding of SIP and telephony is required. Please apply for additional details on the bug.
Skills: SIP C
Hourly - Expert ($$$) - Est. Time: Less than 1 week, 10-30 hrs/week - Posted
I need a US or Canada based Avaya IP Office expert. We have some issues we are running into with some customers on configuring their Avaya system to forward calls to an external extension and I am hoping to find someone who can help us resolve these issues and some other work we may have.
Skills: SIP Avaya
Hourly - Intermediate ($$) - Est. Time: Less than 1 week, Less than 10 hrs/week - Posted
We have already configured a Asterisk 13.6.0 with FreePBX 12.0.76.2 in a Ubuntu 14.04.3 LTS (cloud server in Linode) but we need to finish setting some things like the following: -The biggest current problem we want to solve is that external calls are working properly, now sometimes when we make a call to the external number the other person listens to us but we can not do it and sometimes it happens the other way around, so we researched and apparently the best solution is to configure the NAT of our asterisk or maybe some other solucion would be to configure a server STUN or ICE -Fax Configuration (spandsp) -Queue configuration as completely as possible. -Setting times and schedules (work time and time off work) a Time Conditions. -We also need to configure the application in a good way GS Wave (IP telephony application), currently this does not work very well, very stuck listening. -IVR configuration. -Call forwarding. -Service "Do Not Disturb": To display as no connected if someone calls us, but we can make calls. -Call groups: to ring all phones in a group when someone calls, or to ring sequentially in order, etc. -Pickup calls from any telephone or just from some specific telephone. -Check how to call forwarding, call waiting with single key combination, for example to do with the cordless phone These are the task in order of importance that we need to solve.
Skills: SIP Asterisk FreePBX Ubuntu
Hourly - Expert ($$$) - 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
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