Sip Jobs

61 were found based on your criteria {{ paging.total|number:0 }} were found based on your criteria

show all
  • Hourly ({{ jobTypeController.getFacetCount("0")|number:0}})
  • Fixed Price ({{ jobTypeController.getFacetCount("1")|number:0}})
Fixed-Price - Expert ($$$) - Est. Budget: $200 - Posted
I need a software (similar to a Skype Dialer) that automatically calls phone numbers given from a list of numbers in CSV format. The software should record the call statuses like, Call Answered, Call Disconnected, Call Un-Answered, Call forwarded to Answering Machine , number not in service etc.
  • Number of freelancers needed: 3
Skills: SIP Interactive Voice Response Skype skype development
Fixed-Price - Intermediate ($$) - Est. Budget: $300 - Posted
Hello, We are looking for a programmer to complete different tasks such as: Tie our billing to: 1. Different monitoring systems 2. PBX 3. Automated backups Also do custom solutions for email to fax / fax to email using freeswitch or asterisk. We need a programmer with STRONG perl skills for the billing or PHP for the rest of the tasks. The budget is actually different for each task, please talk to us and negotiate.
  • Number of freelancers needed: 2
Skills: SIP Asterisk Freeswitch Linux System Administration
Hourly - Expert ($$$) - Est. Time: More than 6 months, 30+ hrs/week - Posted
We are looking for creative and motivated software developers to improve our XMPP-based front-end and back-end software stack in a long term partnership. Knowledge of various debugger tools like Firebug and IE Developers tools, Mobile UI development, Cross browser designing and developing experience. We work with cross-cultural and global teams, which require virtual communication skills, pro active behavior and the willing to think and act solution oriented. The developers must have experience working with responsive UI, cross browser front-end development. Linux, bash scripting, Prosody server and XMPP know how is vital. * good English skills (Both reading and writing) * experienced with GIT * fast learner, self-directed, work well with others * experienced in Ajax / JavaScript, willing to work with a new framework * experienced in Java / J2EE / JSP / servlets (Zimbra) * experienced JavaScript, HTML, DHTML, XHTML, CSS skills * Backend and Frontend experience is appreciated * XMPP, WebRTC, VoIP experience * Prosody server experience vital * good Linux skills required * bash scripting experience important * nice to have: Java, Perl, Python * Networking (IPv4, IPv6, HTTP, TLS, network security, encryption, TCP/UDP, server-client architectures, mesh architectures, peer2peer, VPNs) Also Zimbra knowledge is beneficial.
Skills: SIP AJAX AngularJS Bash shell scripting
Fixed Price Budget - Expert ($$$) - $1,000 to $1,500 - Posted
We are looking android sip client sip client should support, texting,audio and video calls. CALLS - yes - no - work in progress - not planned due to EOL SIP XMPP Audio calls Video Calls Desktop streaming Audio conference calls Audio level display Call recording Attended transfer Blind transfer Call encryption Mute Hold Noise suppression Echo cancellation INSTANT MESSAGING XMPP SIP Presence One-to-one chats Multi-user chats File transfer SECURITY Encrypted password storage Password protection with a master password Encrypted Instant Messaging with Off-the-Record Messaging (OTRv4) Chat authentication with the Socialist Millionaire Protocol over OTR Call encryption with SRTP and ZRTP for XMPP and SIP Call encryption with SRTP and SDES for XMPP and SIP CODECS Audio: Opus, SILK, Speex, G.722, PCMU/PCMA (G.711), iLBC, GSM, G.729 Annex C (requires compilation and licenses) Video: H.264, H.263-1998 / H.263+, (VP8 coming soon…) MISCELLANEOUS On-line provisioning Provisioning server discovery via DHCP and mDNS (Bonjour) IPv6 fully supported by SIP and XMPP Call history Missed call notifications Support for Google Contacts On-line contact list storage with XCAP Secure signalling with TLS DTMF (SIP INFO, RTP RFC 2833/4733, inband) Message Waiting Indication (RFC 3842) Certificate-based client authentication XMPP SPECIFIC DTMF (RTP RFC 2833/4733, inband) ... CALLS - yes - no - work in progress - not planned due to EOL SIP XMPP Audio calls Video Calls Desktop streaming Audio conference calls Audio level display Call recording Attended transfer Blind transfer Call encryption Mute Hold Noise suppression Echo cancellation INSTANT MESSAGING XMPP SIP Presence One-to-one chats Multi-user chats File transfer SECURITY Encrypted password storage Password protection with a master password Encrypted Instant Messaging with Off-the-Record Messaging (OTRv4) Chat authentication with the Socialist Millionaire Protocol over OTR Call encryption with SRTP and ZRTP for XMPP and SIP Call encryption with SRTP and SDES for XMPP and SIP CODECS Audio: Opus, SILK, Speex, G.722, PCMU/PCMA (G.711), iLBC, GSM, G.729 Annex C (requires compilation and licenses) Video: H.264, H.263-1998 / H.263+, (VP8 coming soon…) MISCELLANEOUS On-line provisioning Provisioning server discovery via DHCP and mDNS (Bonjour) IPv6 fully supported by SIP and XMPP Call history Missed call notifications Support for Google Contacts On-line contact list storage with XCAP Secure signalling with TLS DTMF (SIP INFO, RTP RFC 2833/4733, inband) Message Waiting Indication (RFC 3842) Certificate-based client authentication XMPP SPECIFIC DTMF (RTP RFC 2833/4733, inband) ... CALLS - yes - no - work in progress - not planned due to EOL SIP XMPP Audio calls Video Calls Desktop streaming Audio conference calls Audio level display Call recording Attended transfer Blind transfer Call encryption Mute Hold Noise suppression Echo cancellation INSTANT MESSAGING XMPP SIP Presence One-to-one chats Multi-user chats File transfer SECURITY Encrypted password storage Password protection with a master password Encrypted Instant Messaging with Off-the-Record Messaging (OTRv4) Chat authentication with the Socialist Millionaire Protocol over OTR Call encryption with SRTP and ZRTP for XMPP and SIP Call encryption with SRTP and SDES for XMPP and SIP CODECS Audio: Opus, SILK, Speex, G.722, PCMU/PCMA (G.711), iLBC, GSM, G.729 Annex C (requires compilation and licenses) Video: H.264, H.263-1998 / H.263+, (VP8 coming soon…) MISCELLANEOUS On-line provisioning Provisioning server discovery via DHCP and mDNS (Bonjour) IPv6 fully supported by SIP and XMPP Call history Missed call notifications Support for Google Contacts On-line contact list storage with XCAP Secure signalling with TLS DTMF (SIP INFO, RTP RFC 2833/4733, inband) Message Waiting Indication (RFC 3842) Certificate-based client authentication XMPP SPECIFIC DTMF (RTP RFC 2833/4733, inband)
Skills: SIP Android App Development
Fixed Price Budget - Intermediate ($$) - $500 to $1,000 - Posted
Freelancer must know: Objective-C Native programming Have used SignalR for RPC calls Skype Integration SIP Integration This is a FAST turnaround! 7-10 days.
Skills: SIP Skype
Hourly - Expert ($$$) - Est. Time: 1 to 3 months, Less than 10 hrs/week - Posted
There are utilities which will convert a VirtualBox image to hyper-v image of a VM. I'll let pick which one to use. Also we need a SIP trunk provider. I am thinking about Sigmavoip but I am open to suggestions. ... We are almost there as it is and I am pretty sophisticated technically. I just haven't done this before. 1. SIP-T48G Ultra-elegant Gigabit IP Phone http://yealink.com/product_info.aspx?
Skills: SIP FreePBX VOIP Administration VOIP Software
Fixed-Price - Expert ($$$) - Est. Budget: $70 - Posted
We have a freepbx system hosted in the cloud, latest version which comes with a VPN server. The VPN server uses OpenVPN. We would like to connect our customers to this pbx using a router with OpenWRT/DD-WRT firmware installed, their phones would then be plugged into this router and should be found by End Point Manager. We would like someone to write us a guide for us to setup the VPN on the freepbx, then setup the router, so that the phones we plug into the router appear in End Point Manager. We have already setup VPN on the freepbx and connected a desktop computer to this vpn, which runs a softphone, although we are struggling with connecting up the router and having phones appear in End Point Manager. We know this is possible but there is very limited documentation on the internet. Please only apply for this job if you have an excellent knowledge of freepbx, openVPN and the firmware (openwrt / DD-wrt). We would expect to be able to give this guide to our engineers and for them to be able to setup the VPN from start to finish.
Skills: SIP Asterisk FreePBX OpenVPN
Hourly - Intermediate ($$) - Est. Time: 1 to 3 months, Less than 10 hrs/week - Posted
We're seeking a FreeSWITCH / Kazoo / WebRTC expert that can assist us with multiple projects. This is a list of experience and requirements you need to have: - Kazoo configuration experience across multiple nodes is required. - Expertise using the Kazoo API's - Experience with event handling - Previous experience with WebRTC - DID provisioning - Setting up call flows - JavaScript experience is a very strong plus.
Skills: SIP Freeswitch OpenSIPS Telecommunications Engineering