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

Sip Jobs

46 were found based on your criteria

show all
show all
only
only
only
show all
only
only
only
only
only
show all
only
only
only
Fixed-Price - Est. Budget: $ 150 Posted
We are looking for someone who can do the below mentioned tasks for us. 1- Build a website using our given template, and integrate it to Mysql database of our voip server. 1- Website must have signup and member/reseller portals, you can use the pre-designed portals provided by VoiP Vendor and modify them according to website design as well. 2- Integrate user and reseller to add payment using paypal,skrill,ecoPayz,and credit card. and in case of bank wire user can add transaction ID about his payment for verification. 3- Vendor API for DID Numbers must be integrated, User can Buy DID numbers from Inventory as well as from vendor push/pull queries. 4- Email must be configured to sent on different events (Signup/Email Confirmation, Balance Low, Balance Added, DID Purchase, DID Renew, DID Expire, DID Billing Coming Soon, Maintenance email to all users etc.) 5- Ability to order VPS from the portal, and can choose from predefined vps hardware, VPS price must increase/decrease...
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: $ 400 Posted
Evaluating the performance of Voip in congested LTE network, by using parameters like Packet loss and End to end delay. the LTE network should be congested with services like FTP, HTTP and email services while the voip service is also being run. Also the performance of voip in non congested network will be evaulated, so as to be able to compare and contrast the reliabilty of voip when there is congestion and where there is not. Also codecs that can be used to improve voip whilst the lte network is being congested. the simulator tool i would like to use is OPNET. thank you
Fixed-Price - Est. Budget: $ 150 Posted
We are looking for someone who can do the below mentioned tasks for us. 1- Build a website using our given template, and integrate it to Mysql database of our voip server. 1- Website must have signup and member/reseller portals, you can use the pre-designed portals provided by VoiP Vendor and modify them according to website design as well. 2- Integrate user and reseller to add payment using paypal,skrill,ecoPayz,and credit card. and in case of bank wire user can add transaction ID about his payment for verification. 3- Vendor API for DID Numbers must be integrated, User can Buy DID numbers from Inventory as well as from vendor push/pull queries. 4- Email must be configured to sent on different events (Signup/Email Confirmation, Balance Low, Balance Added, DID Purchase, DID Renew, DID Expire, DID Billing Coming Soon, Maintenance email to all users etc.) 5- Ability to order VPS from the portal, and can choose from predefined vps hardware, VPS price must increase/decrease...
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...
Hourly - Est. Time: Less than 1 week, Less than 10 hrs/week - Posted
I have an Asterisks PBX and a Vicidial predictive dialer and I need help configuring these servers and some special functions that I need the PBX to Show Local Caller IDs when calling Local area codes. I also need help with recording which of my call back number(s) the inbound callers use to contact us. We have some code to use our Asterisks PBX to recognize the inbound callers ID and pop up their Caller ID data and all information we have in our data base to know which agent spoke to the caller and what notes we have in the db for this potential client... I would like to speak to some knowledgeable programmers to and assign some tasks to know who can help us with these servers and programs long term.