You've landed at the right place. oDesk is now Upwork. Learn about the new platform.

Freeswitch Jobs

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

show all
  • Hourly ({{ jobTypeController.getFacetCount("hourly") | number:0}})
  • Fixed Price ({{ jobTypeController.getFacetCount("fixed") | number:0}})
show all
only
only
only
show all
only
only
only
only
only
show all
only
only
only
Looking for the Team App?
Download the New Upwork Team App
Fixed-Price - Intermediate ($$) - Est. Budget: $3,000 - Posted
We need an experienced developer in server in the following ; opensips +asterisk +a2billing.... loadbalancing,failover , media proxy, natting.... Summary ............... The idea is to build a redundant system that can scale with a click of a button You can also use Kazoo It will be a portal to purchase phone numbers //Toll free Number //Premium numbers and option to forward to SIP, Softswitch, Regular number,fax2email Highly secure to Brute force and other attacks. We would also need custom PBX with user friendly interface Both Admin panel and end user panel has to be neatly designed ( custom made based on our requirements ) and attractive and at same time upgrades to Opensips/Asterisk/a2billing/kazoo should be possible from time to time without affecting front hand or backhand working Details of project will be shared before awarding project.
Skills: Freeswitch A2Billing Asterisk
Fixed-Price - Intermediate ($$) - Est. Budget: $150 - Posted
Installation and configuration Asterisk v13 FreePBX v13 and A2Billing v2.2 on a remote vps or Installation and configuration freeswitch fusionpbx with billing installation on the vps if you're available.
Skills: Freeswitch A2Billing Asterisk CentOS
Fixed-Price - Intermediate ($$) - Est. Budget: $1,800 - Posted
we are looking for VOIP Engineer/Administrator who can work with Kazoo, Kamailio, FreeSwitch, CGRates. complete configuration and modification to make full class 5 switch and also experience in network and app development for cloud base VOIP system o Voice and Video Telephony o Instant Messaging (IM) and Presence o Voice/IM conferencing o Interactive Voicebox and Voice-to-mail o Fax-to-mail, Webfax and Fax Clients o Serial and Parallel Call Forking o Periodic, time-based Call Forwarding (CFU, CFB, CFNA, CFT) o Inbound and outbound Call Blocking and anonymous Call Rejection o CLIP/CLIR, DDI, DID and Extension Dialing o Signaling over UDP, TCP, TLS, WS, WSS o DTLS-SRTP transcoding for WebRTC bridging
Skills: Freeswitch Asterisk SIP VOIP Administration
Fixed-Price - Expert ($$$) - Est. Budget: $650 - Posted
OVERVIEW: My lead engineer has run into a problem with our lead product. He was tasked to install TLSv1.2 on a freeswitch server and then to produce a wireshark trace showing that the server was using TLSv1.2 for SIP traffic. ... He was tasked to install TLSv1.2 on a freeswitch server and then to produce a wireshark trace showing that the server was using TLSv1.2 for SIP traffic. We configured freeswitch like this: ./configure --prefix="$TS" --with-soundsdir="/storage/sounds" CFLAGS="-I /usr/local/ssl/include" LDFLAGS="-L/usr/local/ssl/lib" TASK: I need a qualified SIP developer to see why our freeswitch SIP is using TLS 1.0 for SIP connections, even though v1.2 is being specified. ... /configure --prefix="$TS" --with-soundsdir="/storage/sounds" CFLAGS="-I /usr/local/ssl/include" LDFLAGS="-L/usr/local/ssl/lib" TASK: I need a qualified SIP developer to see why our freeswitch SIP is using TLS 1.0 for SIP connections, even though v1.2 is being specified.
Skills: Freeswitch SSL
Fixed-Price - Intermediate ($$) - Est. Budget: $100 - Posted
Hi I think freeswitch has mod_portaudio and mod_alsa that can do this task. I need someone who is experienced with this to help me. ... I am able to capture audio with arecord from sound card and aplay a wav audio to sound card. I need to send the audio in/out of RTP via Freeswitch to the sound card. Getting audio passing is the 1st step. ... Then, you will need to: 1 - configure ring tone within Freeswitch 2 - send UDP signal to open up audio to the sound card.
Skills: Freeswitch
Fixed-Price - Intermediate ($$) - Est. Budget: $2,000 - Posted
Modified solution should provide SIP interface (SIP Trunks) with T.38 (we require the usage of Freeswitch - www.freeswitch.org). Modifications should be transparent for other system functionalities.
Skills: Freeswitch Asterisk SIP
Looking for the Team App?
Download the New Upwork Team App
Fixed Price Budget - ${{ job.amount.amount | number:0 }} to ${{ job.maxAmount.amount | number:0 }} Fixed-Price - Est. Budget: ${{ job.amount.amount | number:0 }} Open to Suggestion Hourly - Est. Time: {{ [job.duration, job.engagement].join(', ') }} - Posted
Skills: {{ skill.prettyName }}
Looking for the Team App?
Download the New Upwork Team App