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

Sip Jobs

16 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
Hourly - Expert ($$$) - Est. Time: Less than 1 week, Less than 10 hrs/week - Posted
On sites where we deploy Mikrotik routers we have discovered NAT issues for sip clients registering to freepbx hosted servers in remote data centres. ... It seems Mikrotik Nat is broken for sip. With routeros "sip helper" enabled or disabled, we have the problem of sip end points periodically showing "Un-Registered" If we replace Mikrotik with any other router like a snapgear or cheap dlink the problems go away.
Skills: SIP Asterisk Mikrotik RouterOS VOIP Administration
Hourly - Entry Level ($) - Est. Time: 1 to 3 months, Less than 10 hrs/week - Posted
We're looking to have SMS/MMS t.38 faxing and Conference Bridge Apps made. SMS/MMS would need to send SMS replies back through email and possibly even start from an email if possible. Would be nice also to have a conference bridge as well with some basic feature sets. t.38 faxing could work with either Plivo or Voip Innovations.
Skills: SIP API Development iPhone App Development PHP
Hourly - Entry Level ($) - Est. Time: 1 to 3 months, 10-30 hrs/week - Posted
What i'm interesting in is developing a custom audio gateway plugin that would allow audio to flow in and out to SIP enabled topologies (like asterisk/Etherstack and many others) . ... The AUDIO IN/OUT and Push-To-Talk and Incoming Voice Call is exposed in the API layer, which may allow use to code to an open SIP stack. I have contacted Etherstack and they love the IDEA, although there server application runs on LINUX, which is no problem running under a VM, they can licence us to support SIP (Direct RTP streams, G711 codecs). ... I have contacted Etherstack and they love the IDEA, although there server application runs on LINUX, which is no problem running under a VM, they can licence us to support SIP (Direct RTP streams, G711 codecs). Note the current development will be in windows, but possibly moved into windows embedded later.
Skills: SIP Asterisk Microsoft Visual Studio OpenSIPS
Hourly - Intermediate ($$) - Est. Time: Less than 1 week, Less than 10 hrs/week - Posted
We need to foward on a one to one basis, any incoming call from PORT 1 on de GXW4108 as a SIP TRUNK to ring at the customer's UCM-6102; when the customers request dialout on their UCM-6102, the outbound call should be routed outbound via the SIP TRUNK to PORT 1 on the GXW4108 which is connected to the FreePBX Asterisk system ... We need to foward on a one to one basis, any incoming call from PORT 1 on de GXW4108 as a SIP TRUNK to ring at the customer's UCM-6102; when the customers request dialout on their UCM-6102, the outbound call should be routed outbound via the SIP TRUNK to PORT 1 on the GXW4108 which is connected to the FreePBX Asterisk system
Skills: SIP Asterisk CentOS FreePBX
Hourly - Intermediate ($$) - Est. Time: Less than 1 month, Less than 10 hrs/week - Posted
Customize vicidial according to our requirement for inbound callcenter. Customization includes 1. Look and feel of the agent and admin/supervisor interface. 2. Simplified method to login to the system 3. Simplified screen to create/manage agents and extensions.
  • Number of freelancers needed: 3
Skills: SIP Asterisk FreePBX JavaScript
Hourly - Entry Level ($) - Est. Time: More than 6 months, Less than 10 hrs/week - Posted
We are looking for a Linux Server Administrator that specialises in both VOIP and VPN services. The person must have knowledge and experience with Asterisk VOIP and OpenVPN services. The job will be on a per needed bases. The administrator must have excellent English communication skills and be able to respond to technical support tickets from time to time.
Skills: SIP Asterisk CentOS Debian OS
Hourly - Intermediate ($$) - 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
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