Sip Jobs

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Hourly - Expert ($$$) - Est. Time: Less than 1 week, Less than 10 hrs/week - Posted
Hi we are looking for "multi-homed" asterisk configuration for a SIP trunk to send/receive traffic using the private IP through the IPsec tunnel.
Skills: SIP Asterisk VOIP Software
Hourly - Intermediate ($$) - Est. Time: More than 6 months, 10-30 hrs/week - Posted
My company does outbound calling from overseas. If you apply for this job, please put your favorite M&M color in your reply. I need some one who has max experience with these dialer systems. I need you to help my team in configuring it and give ongoing support in my Vicidial/Goautodial. You will be responsible to load and change lists, to reboot the server, coordinate your work with our team and submitting daily and weekly reports. Please let us know of your experience with these dialers.
Skills: SIP Asterisk Lead generation vicidial
Hourly - Entry Level ($) - Est. Time: Less than 1 week, 10-30 hrs/week - Posted
We are using Asterisk as Fax Server (FoIP) with T.38. We need some help to customize outgoing faxes via Microsoft Outlook. Currently, the outgoing is working but we need to make some specific changes. Thanks
Skills: SIP Asterisk Fax
Fixed-Price - Intermediate ($$) - Est. Budget: $100 - Posted
One of the middle sized companies needs to make the following configurations on its Elastix (Asterisk) server: 1) Configure Specific set of extentions to use the specific trunk 2) Create a limited user with an ability to monitor calls for specific extentions (default settings do not allow that) This is a one time job. Service needed ASAP. Please send your bids according to all tasks specified above.
Skills: SIP Asterisk Elastix
Fixed-Price - Expert ($$$) - Est. Budget: $100 - Posted
hi i like to create a SIP softphone for windows and web softphone for any web browser please contact me if you think to be able todo it
Skills: SIP Java JavaScript
Fixed-Price - Expert ($$$) - Est. Budget: $800 - Posted
Functional Flow 1) Calls originating will send to gateway 2) gateway converts the sip/iax signal to OTT protocol 3) the termination number carried from the origination header will be checked by the OTT gateway , if the number is used by an OTT and if the number is online, the call will terminated on OTT. 4) if the number is not used in OTT it sends 503 error and rerouted to our other premium routes as planned . 5) PDD Would be 12 sec
Skills: SIP VOIP Software
Hourly - Intermediate ($$) - Est. Time: Less than 1 month, Less than 10 hrs/week - Posted
We are looking for experienced developer/development team with prior experience in developing SIP audio/video clients. Features required: - TCP or UDP SIP transport - Codecs: G.722, G.711u, G729, Speex - Remote phonebook synchronization - Remote provisioning - Branding - VoiceMail and VoiceMail Message Waiting indication - DTMF modes - STUN support for NAT traversal - SRTP audio and TLS encryption
Skills: SIP Android App Development Android SDK API Development