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Sip Jobs

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Fixed-Price - Entry Level ($) - Est. Budget: $1,000 - Posted
LET YOUR I.T. KNOWLEDGE GRAB YOUR SHARE OF THE GROWING INTERNATIONAL CELLULAR TELECOM REVENUE - This is a FT or PT opportunity for candidates located in AFRICA (mainland and Islands) and the other countries listed at the end of this message. - Earn a minimum of $ 1,000 USD for the first month. - Expect at least usd$ 5,000 monthly thereafter. Very likely to increase to $10,000+ - Your income depends solely on the quality, consistency and results of your work, which will be measured daily by our Network Operations Center staff. - This by no means involves sales; we have already a long time agreement with international carriers that provide the revenue. - Knowledge of the local cellular market and rates is a plus. - Top Quality internet bandwith is a must (you either have it, or you get it) - To qualify please note that you must be located in any of the following places: AFRICA (mainland and Islands), except for Southafrica; Algeria, Angola, Anguilla, Armenia, Ascension, Barbados, Belarus, Belize, Bolivia, Cambodia, Ecuador, El Salvador, Eritrea, Fiji, Georgia, Guatemala, Hhaiti, Honduras, Iraq, Jamaica, Jordan, Kyrgyzstan, Lebanon, Nepal, Nicaragua, Oman, Philippines, Qatar, Samoa, Sao Tome, Tajikistan, Ukraine, Yemen If you consider you meet this requirements, let us know promptly, and we will send you more details. Thank You
  • Number of freelancers needed: 5
Skills: SIP Cisco IOS IT Management Telecommunications Engineering
Fixed-Price - Intermediate ($$) - Est. Budget: $15 - Posted
Sofswitch Softswitch and GSM gateway are fully integrated. But the calls are not transfered to the PSTN with the configuration done. The VoIP calls that comes from the Softswitch will be terminated to the GSM gateway that forwards the call PSTN (To mobile phone) Now the VoIP expert is to correct the configuration and resolve problems so that calls get to the destination.
Skills: SIP Asterisk VOIP Administration
Fixed-Price - Expert ($$$) - Est. Budget: $1,000 - Posted
Kamalio as registration server extensions as load balancer and to use Elastix servers for voice traffic and voice gateway ... .It can be three servers Kamalio and 5 servers Elastix Appliance, Also We would like to have A2billing for billing purpose because we are prepaid, also we would like to Required Device Provisioning development in this, Cisco SPA Devices and Grandstream, Mediatrix TO 600 simultaneous calls: CPU: Quad Core 2.2 (2 processors) RAM: 16 GB Hard Drive: 1TB Disk array Network Interface: 4 Gigabit ports If you have better Recommendation as well please let me know,. We will do the all the basic installations and You have to do Configurations and make sure reliable system ,.
Skills: SIP Asterisk Linux System Administration MySQL Administration
Hourly - Entry Level ($) - 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
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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
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