Freepbx Jobs

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Fixed Price Budget - Expert ($$$) - $10 to $50 - Posted
Hi all, I am facing with one issue in my system that call DROP, low ACD when calls are sending from main server (openvpn server) to router(asterisk, openvpn client), it getting drop after 4-5 minutes. Low ACD also. as i am very busy and cant fix it now. I need one expert have a look and fix. It will not take more than one hour. I think. So, please help with best price. thanks,
Skills: FreePBX Asterisk Linux System Administration VOIP Administration
Hourly - Expert ($$$) - Est. Time: Less than 1 month, 30+ hrs/week - Posted
The qualified candidate must have extensive experience in Freeswitch and Asterisk and understanding of FreePBX, MYSQL and PHP and some RubyOnRails. Must be able to communicate in English and be available for frequent voice,and chat communications as well Elance PMB and Email.
Skills: FreePBX Asterisk Freeswitch
Fixed-Price - Intermediate ($$) - Est. Budget: $200 - Posted
Hello, This is a Example Description Description Kdeals-app is a free VoIP audio and video Dialer to make VoIP call from the Android Phones,Iphone-Phones,Blackberry,window/Nokia & Tablets. It works over Wi-Fi/2G/3G/4G/EDGE/HSDPA kdeals Dialer Must support high quality audio & video codecs like, G729,G729a, G.726 ,G711a, G711u, g722, VP8, H.263-1998, H.264 etc. 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.. when downloaded by consumers, So in this case you would only need to Create your apk and implement the functionality of the application, in this email attachment you will see a picture which should be in the application which will come under Codec. ( Tab), when consumers click on codec's which should give them a option to Choose the codec which they wish to use during they Call. and the other image/picture in the attachment Must come under ( settings ) which give the consumer option to Configure the application for use of Voip-calls Chat-flow 4)Regarding My VOIP/ SIP app, as I discussed/mention the implementation of my requirements for chat-plow. Please check features list and let me know, I if you understand: - Personal Chat/ Group Chat - Send/ Receive Pictures/ Audio/ Video files through app - Conference audio/ video calls. Video call will be upto 4 persons at a time - Send / Receive emails similar to BlackBerry Messenger app. User can configure unlimited email accounts in the app and send/ receive email through - Offline message, user can send SMS from itself but received SMS will be saved into the native massage app of the device. Will use messages APIs to send unlimited messages. -Free Calling within the app. - Sending and receiving picture within app. - SIP support: app user can make call on mobile and landline numbers by getting services from SIP providers. Please get back to me with your latest feedback and do let me know if you understand the features list of the app. Look forward to hear . 5) No Mile-stones are no Escrow will be accepted because 99.9% of developers are not Honest they are only/telling lies,when the job is completed and applications is tested the money will be Paid to the Developer Through ELANCE. Thank you Regards Rakesh Singh
Skills: FreePBX Adobe Photoshop Android App Development Asterisk
Hourly - Entry Level ($) - Est. Time: Less than 1 week, 30+ hrs/week - Posted
• Prepare Mikrotik Router. • Prepare SIP Server Elastix (IP Phone PBX) • Hardware & Network Troubles Shooting- winNT, winXP,Vista, windows7,windows10 • ISP Router “Mikrotik” Configure & Bandwidth Manage. • Network - LAN, WAN,VLAN, Dial-Up, Hotspot Vpn-(PPTP, PPPOE, EoIP-Tunnel). • IP Subnet, Routing, Data Link (Intranet), Global IP Camera. • Server - FTP, File, Proxy, windows 2013 server.
Skills: FreePBX Computer Networking Microsoft Active Directory Network Administration
Hourly - Intermediate ($$) - Est. Time: More than 6 months, Less than 10 hrs/week - Posted
Cisco to Asterisk , I need a CISCO voice person to work with me daily, jobs can be daily of weekly from change a name on a phone to build a new PBX. I need daily availability on skyp or whatsapp. RATES are on a flat rate from $10 to $200 per ticket request
Skills: FreePBX Asterisk cisco routers Cisco CallManager