Asterisk Jobs

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Hourly - Intermediate ($$) - Est. Time: Less than 1 week, Less than 10 hrs/week - Posted
HI, We are adding a new SIP provider, and I just can't seem to get the INBOUND connection working. Outbound is fine. The Trunk is OK. But no inbound. Need help. Ok to pay by the hour, or if fixed immediate will pay a fixed cost.
Skills: Asterisk FreePBX SIP
Hourly - Intermediate ($$) - Est. Time: 1 to 3 months, Less than 10 hrs/week - Posted
Looking for an experienced hacker help me out with a pretty complex project. This job is 100% legal so you don't need to worry. I will provide more details once we start a conversation going. But I can say that this is a very challenging job and whoever gets it needs to be self-motivated and up for the challenge.
Skills: Asterisk C++ Computer Networking Excel VBA
Fixed-Price - Expert ($$$) - Est. Budget: $400 - Posted
have on site GRANSTREAM box plus a number of brandnew GRANDSTREAM handsets; have message prompts recorded as .wav files; have a PTSN line in place to assist with out bound calls; have inbound numbers booked through sonetel; also a sipcity Australia account for outbound voip calls; require at least 5 separate autoattendant setups installed; caller ID on outbound calls; features such as 'screened calls', 'queued calls'; ';message bank' and messages sent by email etc; teleconference etc; overflow back up calls routed through to back up SIP in Phillipines; handsets hooked through OUTLOOK so that call history etc kept etc; and calls can be dialled from within OUTLOOK contacts - do through TEAMVIEWER connection.... user ASTERISK/ ELASTIX software - set up system completely with basic instruction notes......
Skills: Asterisk Elastix VOIP Administration VOIP Software
Fixed-Price - Expert ($$$) - Est. Budget: $150 - Posted
I like to make using about the desine off this site with about the same functionality this site will need to be interfaced with the A2billing where the client will be able to register or login create account add ip and topup online we need to be sure to don't have any security issue so please use Oop contact me if you can do it Cordialy
Skills: Asterisk A2Billing CSS HTML
Fixed-Price - Intermediate ($$) - Est. Budget: $800 - Posted
This project is projected at 60-90 days it will pay a total of 800 and a bonus if completed within 60 days You are building a VOIP We will provide 2 dedicated dicated servers(CloudSouth) when you get to that point we will also provide a terminating company (Voxbeam) IT will work as a desktop download item In the Android store In the apple store Window app BlackBerry app The target is all devices This is part 1 of a 2 part project. Part two will be building a call center program that will work from phones and tablets, and handled devices using this phone network
Skills: Asterisk Android App Development HTML5 iOS Development
Fixed-Price - Intermediate ($$) - Est. Budget: $30 - Posted
Install freepbx on vps, configure 3 lines including IVR for each of the 3 numbers, validate customer via, support pin. secure freepbx server. configure, xlite client. only bid if you are a registered freelancer/contractor able of issuing and signing invoices.
Skills: Asterisk FreePBX
Fixed-Price - Expert ($$$) - Est. Budget: $300 - Posted
Hello, I am looking for someone that can build me a feature in Vicidial where each individual user can create and upload their own voicemail and with the press of a button, leave their voicemail message at the sound of the beep by hitting the button, it plays the recorded message and simultaneously leaves the disposition as voicemail and moves onto the next call. This feature is not campaign specific, it's attached to the users in all campaigns. Also, the ability to upload different voicemails, name each one, and have the choice to select which one to play.
Skills: Asterisk vicidial
Hourly - Intermediate ($$) - Est. Time: Less than 1 month, Less than 10 hrs/week - Posted
We are looking for experienced developer/development team with prior experience in developing SIP audio/video clients. Features required: - TCP or UDP SIP transport - Codecs: G.722, G.711u, G729, Speex - Remote phonebook synchronization - Remote provisioning - Branding - VoiceMail and VoiceMail Message Waiting indication - DTMF modes - STUN support for NAT traversal - SRTP audio and TLS encryption
Skills: Asterisk Android App Development Android SDK API Development