Sip Jobs

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Fixed-Price - Expert ($$$) - Est. Budget: $50 - Posted
We are looking for someone to setup the Twilio trunk (already created on the Twilio side) into our 3CX server as a call queue to the 6 extensions that will be remote Xlite SIP phones.
Skills: SIP VOIP Administration
Fixed-Price - Expert ($$$) - Est. Budget: $100 - Posted
I'm looking for expert who have built and customized the linphone for android version open source. Now I don't need additional customization, only built open source. If you provide me, I'll check and pay. Please don't bid if you haven't experiment for it.
Skills: SIP Android App Development GitHub Java
Fixed-Price - Intermediate ($$) - Est. Budget: $100 - Posted
I would like somebody to help me with the initial setup of a Freeswitch server. I want to use this as a conference call server (using mod_conference). https://freeswitch.org/confluence/display/FREESWITCH/mod_conference I want: * To be able to dial in using POTS to a conference call on any of the 3 DID numbers I provide which can be configured at DIDLogic. - The dialplan should then pass control to a script in the language of your choice (mod_perl, mod_lua, mod_v8 ... any will do) - All this script needs to do is play a prompt (can use say) asking for the room number (9 digits) and then patch the user into that conference room. There doesn't need to be a concept of leader / leader PIN or database connection or anything complex at this stage, I am just trying to get the basic setup right. * To be able to 'originate' a call from the freeswitch CLI (fs_cli) using the 'originate' command which should dial out to a specified POTS number and patch them into a specified mod_conference room. - I just need an example of the syntax for this, you can test it works with the credit I will give you in the DIDlogic account. I will provide: * root SSH access to a Linux server (Redhat) with stock Freeswitch 1.6 installed. * a DIDlogic account with 3 DID numbers and a small amount of credit to test outgoing calls. I will require: * A brief description of the configuration changes you needed to make in freeswitch and any issues you faced. * I should be able to glean your precise changes by doing a `git diff` on the /etc/freeswitch directory
Skills: SIP Asterisk Freeswitch Linux System Administration
Fixed-Price - Intermediate ($$) - Est. Budget: $500 - Posted
I am looking for a person who can help in creating a sip server to allow sip softphone registration with and eventually create a basic intercom system.
Skills: SIP OpenSIPS
Fixed-Price - Expert ($$$) - Est. Budget: $100 - Posted
if you listen the notifications events in Android OS of skype app, you will see that it uses a long numeric ID to identify the owner of the skype running app and also for every caller. I need you to find my long numeric ID for skype username "minutizer.com". Then you need to provide me the scripts or the methodology to identify this long numeric number for each skype username (or vice-versa). Extra notes: Try to listen the notification events in Android OS for Skype app. Then receive/make a Skype call and you will see that the users are identified as a long number and not as their Skype username . we need a way to map those numbers to the actual Skype username.
Skills: SIP Android Android App Development Android SDK
Hourly - Intermediate ($$) - Est. Time: More than 6 months, 10-30 hrs/week - Posted
Looking to set up SIP appliances to send and receive calls triggered by automation systems or via custom button presses. ... For example, someone presses a doorbell, and it triggers a SIP device to call customer's cell phone. Also general help needed to set up grandstream UCM devices to handle small office and residential systems.
Skills: SIP REST VOIP Software