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Sip Jobs

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Fixed-Price - Intermediate ($$) - Est. Budget: $30 - Posted
We are looking for a highly experience FreePBX expert to configure our already well installed and secured VOIP FreePBX system. The task that need doing are as follows: 1. Configure Trunks 2. configure 8 incoming DIDs from 3 providers 3. Configure outbound and incoming routes 4. Setup 3 IVR, 5 extensions each, 3 Queues and call forwarding This is an urgent requirement that need to be completed ASAP. Only apply if you are free and can start straight away. You must also be familiar with the following DID and Voip careers: Voxbeam, Flowroute, Dinumbers.com. Happy bidding.
Skills: SIP Asterisk CentOS Elastix
Fixed-Price - Intermediate ($$) - Est. Budget: $200 - Posted
Developer will prepare server, install and configure A2Billing and relevant components. Absolutely required features: - Callback - web callback; Ani callback - Calling card - Call shop - Reseller agent accounts - Trucking - Origination and termination - Asterisk PBS programming - Custom IVR upload and configuration - In addition to standard features. There is a potential for service support on an ongoing basis upon successful completion of project. Developer must be flexible and punctual.
Skills: SIP Asterisk FreePBX HTML5
Fixed-Price - Intermediate ($$) - Est. Budget: $500 - Posted
We are seeking a Kamailio or OpenSIPS expert to help us install and configure a SIP proxy in front of multiple Asterisk servers. Currently, we have a cluster of 4 asterisk servers in a multi-tenant setup, but the end user needs to be told which one to connect to. ... The Asterisk servers connect to a MYSQL-based back-end (MariaDB Galera) The SIP proxy must: 1. Perform a simple MYSQL lookup based on the user ID to determine which Asterisk server this user belongs to 2. ... Pass through registration to the correct Asterisk server 3. Block mass SIP Registration Attacks 4. Provide NAT translation assistance 5.
Skills: SIP Asterisk MySQL Programming OpenSIPS
Fixed-Price - Intermediate ($$) - Est. Budget: $900 - Posted
For tunnel server it is expected the developer to use plain C language to write a flexible tunnel server . Its overall objective is to communicate with sip server to translate encrypted data passed from dialers. ... Android client api will be developed by java to integrate with android sip dialer. Iphone client api will be developed by Objective C to integrate with iphone sip dialer.
Skills: SIP Android App Development iPhone App Development VOIP Administration
Fixed-Price - Intermediate ($$) - Est. Budget: $200 - Posted
Compatible with any SIP based Softwitch Connectivity through WIFI/3G/4G/Edge/HSDPA Login by Your SIP server, SIP user & Password Outgoing and incoming VoIP calls Address book integration Calls History Loudspeaker Microphone on / off work in background mode Excellent Voice Quality Audio Codecs Support, G729, G711a, G711u, G722, AMR & GSM,PCMA/G.711 A-law,PCMU/G.u-law Video Codecs Support, H.263, H.264, VP8 AMR Speex16khz Sppex8khz Speex 32 Khz Opus 48khz Silk 24khz Silk 16khz iLBC AAC-ELD 22khz AAC-ELD 44khz Compatible with Android Phones & Tablets OS version 2.6 and above ETC....... ... Codec Section for Voice: Codec Section: Speex 32 kHz >> Automatic Selected Speex16khz >> automatic Selected Speex8khz Opus 48khz Silk 8khz Silk 12khz Silk 16khz Silk 24khz >> automatic Selected GSM 8 kHz AAC-ELD 22khz AAC-ELD 44khz ISAC 16 kHz ILBC 8 kHz AMR 8 kHz AMR-WB 16 kHz CODE2 8 kHz G729a >> automatic Selected G729 >> automatic Selected G711a G711u G722 G726-32 8 kHz G726-24 8 kHz G726-16 8 kHz PCMU 8 kHz >> automatic Selected PUCMA 8 kH8 >> automatic Selected RFC-2833 >> Automatic Selected QUAD-BAND >>GSM850, automatic Selected GSM900, >> automatic Selected GSM1800 ,>> automatic Selected GSM1900 >> Automatica Selected Support: Description : This app must be a customize SIP/XMPP based VOIP/IM app as my requirement and have ready for use without no Delay Registration when Consumers/Customers download the application.In all Mobile-voip-app Registration is depending on the Developer with code implementation during the production of the application You can review the following above in ( Description ) which will be needed from you to make the application running as a app in perfect condition. ... The application must work on open source Android/iPhone/web/windows mobile 8/mac/blackberry/Symbian SIP apps those work with Sip server like asterisk. For Example the app must also work on asterisk AMI (Asterisk management interface),AGI (Asterisk gateway interface). 2) Below Must be integrated as a Default when application download, See Below: Registration expire time >> 3600 Transport type >> UDP=5060 to 5080 Stunt Refresh Period == 30000 Send RTP DTMF Send SIP DTMF Send SIP INFO DTMF Noise suppression 2) Under >> Network, See Below: Registration expire time >> 3600 Transport type >> UDP=5060 to 5080 and UDP 10000 to 30000) RFC-2833 Stunt Refresh Period == 30000SIP INFO == automatically Selected 2+) We will Explain you about Complete network. 3) I already Design the presentation of the app how it should presented on a mobile-phone etc..
Skills: SIP Adobe Photoshop Android App Development Asterisk
Fixed-Price - Expert ($$$) - Est. Budget: $2,000 - Posted
Hi there create a VOIP site i want my own voip with all the features like other voip providers i even want my own chatting apps , pc apps, and website it should have voice, video, text and other features and can we create sdk, minutes or no like other voip sites for example like skype, oovoo, vsee so on it should be fully automated system and do you provide bug fix, version upgrade, support and maintenance are you going to use open source or create from start or you already have or your choice and if possible try to make it fast please let me know what you require and how much time and cost me please note this project is for limited time i am posting if i don't get good response i will remove or with in time i am looking i don't get thanks
Skills: SIP Android App Development Blackberry app development HTML5
Fixed-Price - Intermediate ($$) - Est. Budget: $1,000 - Posted
Hello We want to develop and automated telephone system we have an asterisk server which is not operational yet. We have an SQL server table where Vehicle and its pin code is saved. What we need is? When an entry comes to an SQL Server table lets say alarm (I can copy that to MYSQL or Asterisk Database) then aasterisk should automatically calls the number and ask "PLEASE ENTER PIN CODE" as user enters it would authenticates the PIN and then Ask the USER"Are you travelling in vehicle?" If he press one then No issues if press 2 then logs then entry in the database from where I will copy it to SQL SERVER BACK.
Skills: SIP Asterisk FreePBX VOIP Software
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