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Asterisk Jobs

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Fixed-Price - Est. Budget: $ 200 Posted
Hi I need help with eSpeak TTS, i am having issue to play it via Asterisk, if you have experience with this , then please reply to this job posting Thanks Aron
Skills: Asterisk
Hourly - Est. Time: Less than 1 month, 10-30 hrs/week - Posted
We are seeking a consultant to provide ongoing customization and support of an existing Asterisk phone system on a part-time, as-needed bases. Some of the current issues with which we need assistance: Troubleshooting issues with VOIP trunks causing performance issues for Asterisk (currently disabled.) Troubleshooting dial plans to determine why some calls are delayed for long periods (10 seconds or more in some cases) before ringing starts. Troubleshooting local long distance calls ringing with No Answer rather than returning recording from telco stating that a 1 must be dialed. Customizing Voicemail to no longer say "Comedian Mail" but to say company name instead. Functionality for deleting voicemail from URL in email (email link is fine, but needs to be secure/nonguessable ID in link) and Asterisk changes message IDs with each deletion. Clicktodial application for identifying phone numbers in a web page and click to dial (Like Noojee for Asterisk but this does not work...
Fixed-Price - Est. Budget: $ 200 Posted
We are using Asterisk 1.6.2.9 on a Ubuntu2.1 server with Yealink T28 handsets. We are experiencing intermittent drop outs on incoming calls (outgoing calls never drop out). We probably get on average 20 incoming calls a day and approximately 2 calls would drop up within 2 to 3 minutes of the call being answered. We are using a Digium Asterisk TDM411B card for incoming calls from an existing POTs number and also have a virtual number from Voxbone. Both numbers seem to be experiencing this incoming drop out problem. What we need you to do? 1) We need someone to turn on all debugging on asterisk to find out what is causing the drop outs on these incoming calls and fix it. 2) Once debugging is turned on we will tell you the phone numbers of the incoming calls that drop out each day so that you can trouble shoot and fix the problem. We know that it is not the handset or router that is causing the drop out as we have changed the handset and router and still have the same problems. Upgrading...
Skills: Asterisk
Hourly - Est. Time: Less than 1 month, 30+ hrs/week - Posted
i need some kind of bandwidth compression system ( upto 60-80% than usual SIP calls )from Server A to Server B. Server A = Asterisk server Server B = Asterisk Client server Explanation of scenario: 1. server A ( asterisk server, with static IP) receiving VoIP calls , with sip protocol, using G711,G729 and/or G723.1 codec and sending calls to Server B 2. Server B ( Asterisk server with PRIVATE NETWORK IP), receiving calls from server A and sending to gateways (quintum gateway for example) or E1 cards. 3. Number of Server B can be unlimited. 4. Number of Gateways/E1 cards per server B can be unlimited 7. Server A to Server B voice traffic will be encrypted so that voice port blocked bandwidth can be used for termination. we will used . A. iax2 trunks in trunking mode. ( May be need to customize IAX2) B. Open vpn static mode and dynamic mode 8. Asterisk web billing gui for adding gateways. Adding client , Prefix , dialing plan viewing active calls, billing cdr ,etc. we...