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Opensips Jobs

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Hourly - Est. Time: Less than 1 week - Posted
Configure a new OpenSIPs ITSP solution that will allow multiple SIP trunks to operate through our network of VoIP carriers. Requirements: - CDR reporting for traffic, tracked by SIP trunk and which carrier call goes-out on - Dynamic routing based on LCR for a number dialed - Incoming carrier numbers routed to correct SIP trunk with CDR - Install appropriate web GUI Nice-to-have: - Fraud detection - 911 (emergency) routing Will require directions to make updates to LCR table, add additional SIP trunks, etc. We have a high availability MariaDB cluster already in place. Ongoing support is expected. A successful candidate will quote on hours required for configuration and be given the opportunity for maintenance work as required.
Fixed-Price - Est. Budget: $ 100 Posted
Tehnology we use: -Asterisk -FreePBX -WebRTC (Tried with Sip5ML, JsSip, Sip.js) - 4 different trunks - One stationary number (our ISP) - 4 GSM numbers Problem we have: When calling trough stationary trunk on stationary trunk we get sound delay when answering late (Delay is not always the same it depends on the time when user pick up the call) Explanation: Ok so we start off with what is working. If we make a phone call through GSM trunks on any number (GSM or Stationary) everything works fine. The important thing here is that when connection is established and call starts to progress we get response 183 (With early media) and so on... The 183 response is sent by our GSM VoIP terminal. The problems appear when calling through our ISP trunk, since provider can't fake 183 response if it is not sent to them from dialed number. When we call through ISP trunk on GSM numbers things work fine, when connection is established GSM providers send 183 response with early media...