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Sip Jobs

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Hourly - Intermediate ($$) - Est. Time: Less than 1 week, 10-30 hrs/week - Posted
Using the tool SIPP, I need to have a list of phone numbers in CSV and/or TXT format. It will need to execute standard UAC scenario against each number, adhering to a concurrent call limit and produce results for the list, including times. The intent is to: 1) validate that each number properly connects and provide results on a per number basis as rows in a output. 2) that the destination host can handle X simultaneous calls I believe this is all already possible with variable injection, however I would like someone to just provide a complete example of how to achieve this. Additional SIPP work and SIPSAK work will be needed over the next few weeks. Please have skills for both of these programs.
Skills: SIP Linux System Administration
Fixed-Price - Intermediate ($$) - Est. Budget: $40 - Posted
I need full configuration for solution VOIP kamailio and asterisk. Kamailio sip proxy and nat transversal, dispatcher asterisk. Kamailio need lcr database, cdr, accept registar or P2P
Skills: SIP Asterisk OpenSIPS
Fixed-Price - Intermediate ($$) - Est. Budget: $500 - Posted
Rebrand Jitsi with new name/logo and package it for windows/linux/android and mac. 2. Remove facebook/XMPP/SIP and Google Talk login screen and just provide a username/password screen that pulls the configuration from the server to configure a SIP account. 3. ... Remove facebook/XMPP/SIP and Google Talk login screen and just provide a username/password screen that pulls the configuration from the server to configure a SIP account. 3. Add a mechanism to download a directory from the server and add those as contacts and subscribe for SIP presence NOTIFYs for those contacts. ... Add a mechanism to download a directory from the server and add those as contacts and subscribe for SIP presence NOTIFYs for those contacts.
Skills: SIP Android Java VOIP Software
Fixed-Price - Expert ($$$) - Est. Budget: $1,500 - Posted
Hello I m looking for SBO same as Syncwitch.com SipZip , but more advance with option of tunneling for pak , uae , Iran, Afghanistan routes Sipzip is also one of the advance tunneling plus also gives inter-calling on sims Please apply only if u have worked on Bandwidth optimizations for Voip Regards,
Skills: SIP Asterisk Linux System Administration PHP
Hourly - Intermediate ($$) - Est. Time: Less than 1 week, Less than 10 hrs/week - Posted
Hello I m looking for SBO same as Syncwitch.com www.sipZip.com , but more advance with option of tunneling for pak , uae , Iran, Afghanistan routes Sipzip is also one of the advance tunneling plus also gives inter-calling on sims Required Skill , Php , Astrisks , Linux and web development Please Apply if u have worked on Voip Bandwidth optimization Regards,
Skills: SIP Asterisk Java Linux System Administration
Fixed-Price - Expert ($$$) - Est. Budget: $350 - Posted
We have a requirement from a client who currently has a Panasonic KX-TES824 PABX and has a requirement to setup an IVR that will assist with handling calls and responding to common issues within the company. We are looking at deploying an Asterix-based solution and would like to get a specialist to work out the full scope and implementation for us remotely. Additional information: 22nd Jan 2016 The client has 1600 students registered for a course online and will need to be able to handle all the calls coming in by way of an IVR to tackle frequently asked questions, and then to allow the caller to go through to a department extension if they insist on talking to someone. In short they require: 1. Automated Response with info to Press 1 for x 2 for y etc. 2. Advert on Hold (i.e. when a caller is waiting for the call to be picked, there should be an informative audio playing in the background. 3. Currently client has 8 extensions that may go up to 15. 4. The Panasonic TES-824 PABX is linked to three networks (one landline and two mobile) and handles calls from all three. This will eventually be changed to a single Voice Shortcode activated on all networks. 5. FAQs: There are up to 20 FAQ Audios that will be available and can be chosen based on the query or option that the customer chooses. An example FAQ will be information on dates of opening for a particular school term or when the examination dates are set. Hope this helps.
Skills: SIP Asterisk FreePBX Telephone Handling
Fixed-Price - Intermediate ($$) - Est. Budget: $1,800 - Posted
we are looking for VOIP Engineer/Administrator who can work with Kazoo, Kamailio, FreeSwitch, CGRates. complete configuration and modification to make full class 5 switch and also experience in network and app development for cloud base VOIP system o Voice and Video Telephony o Instant Messaging (IM) and Presence o Voice/IM conferencing o Interactive Voicebox and Voice-to-mail o Fax-to-mail, Webfax and Fax Clients o Serial and Parallel Call Forking o Periodic, time-based Call Forwarding (CFU, CFB, CFNA, CFT) o Inbound and outbound Call Blocking and anonymous Call Rejection o CLIP/CLIR, DDI, DID and Extension Dialing o Signaling over UDP, TCP, TLS, WS, WSS o DTLS-SRTP transcoding for WebRTC bridging
Skills: SIP Asterisk Freeswitch VOIP Administration
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