Symbian codectrabajos
...Speakerphone, mute and hold functions Call history with a list of received, missed and dialed calls Call recording Ringtones and contact avatars Dial plan support Multiple call support Swap between two active calls Merge and split calls (three-way conferencing support) Call transfer (attended and unattended) Audio codecs include G.711 a/u, G.722 (HD), iLBC, GSM, OPUS and SILK Automatic codec selection to ensure optimal call quality Support for DTMF: the ability to enter numbers to use an auto attendant Via RFC 2833, SIP INFO and in-band Advanced Features Network Traversal Application Managed, Server Managed or Custom Configuration Global IP Support STUN & ICE Media Efficiency and Quality Noise Reduction Echo Cancellation Voice Activity Detection Qualit...
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Hi, find below the requirement and if you have any doubt plz reply me. We are looking for a fully featured mobile phone dialer with built-in mini-SIP tunneling capability. We need full source code of the whole project. Project Description: We are looking for the following features and functionalities. · Android , iphone , windows, symbian and BB OS platforms must be supported. That means it must run well in major mobile phones in the market. · The dialer must support SIP platform completely and has to be compatible with major SIP softswitches. · It must have mini-SIP tunnel to bypass VOIP blockage. It must work from behind the NAT or from private IP. · It must have dial, redial, receive, hangup, call hold, mute, transfer, conf...
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...main goal to minimize the BW in client side with good quality of voice . We 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 5. For server B installation need easy to use ISO image that could be booted from USB flash drive, and ...
Setup a Linux media streaming Server using Http 5 so it can be viewed on all platforms and most browsers (Computers, Mac, Linux, Androids OS devices, iphone, ipad, Blackberry, symbian, Java Phones, WindowsMobile , Tablet pc etc..). Provide a means to load the Media Music Videos or Movies into the Server Provide a means to categorize the Videos or Movies. Many Directories and Sub-directories can be Created by Admin Provide 2 Streaming Options. 1. Provide A Play List Version So videos to be played can be added to the play list by the adminitrator and Streamed to audience Viewers has no control over what is played. They will be directed to a web page where they just click and viewlike a TV Station. 2. Provide A VOD version so users can make their own selections but will need a userna...
...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 5. For server B installation need easy to use ISO image that could be booted from USB flash drive, and those USB flash drive will be delivered to our Server B type client (ther termination provider) 6. For server B need portion of asterisk web gui for adding gateways..prefix or so, viewing active calls, billing cdr,etc. Specific Need: 1. Server A to convert all calls in g723.1 codec and server B to receive all calls in...
I need someone to tell me how to start learning Symbian programming and send me some tutorial and where to find more tutorial to build my Symbian application.
...main goal to minimize the BW in client side with good quality of voice . We 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 5. For server B installation need easy to use ISO image that could be booted from USB flash drive, and ...
...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 5. For server B installation need easy to use ISO image that could be booted from USB flash drive, and those USB flash drive will be delivered to our Server B type client (ther termination provider) 6. For server B need portion of asterisk web gui for adding gateways..prefix or so, viewing active calls, billing cdr,etc. Specific Need: 1. Server A to convert all calls in g723.1 codec and server B to receive all calls in g723.1 ...
...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 5. For server B installation need easy to use ISO image that could be booted from USB flash drive, and those USB flash drive will be delivered to our Server B type client (ther termination provider) 6. For server B need portion of asterisk web gui for adding gateways..prefix or so, viewing active calls, billing cdr,etc. Specific Need: 1. Server A to convert all calls in g723.1 codec and server B to receive all calls in...
...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 5. For server B installation need easy to use ISO image that could be booted from USB flash drive, and those USB flash drive will be delivered to our Server B type client (ther termination provider) 6. For server B need portion of asterisk web gui for adding gateways..prefix or so, viewing active calls, billing cdr,etc. Specific Need: 1. Server A to convert all calls in g723.1 codec and server B to receive all calls in g723.1 ...
Hello, I would need a simple mobile application made for three platforms, i.e. iOS/Android/Symbian. The functionality of the mobile application are as follows: 1. Accept mobile user registration/login 2. Collect mobile number manually or automatically 3. Collect Geo location 4. Collect order details for laundry I need a simple application and less of hassle, we can compromise on looks but not on functionality. Person who wins this project will also be hired after 6 months for up gradation, so quote accordingly. At the moment we need a trial functional application at low budget. Do mail me for any specific queries.
Need to install AMR-NB codec for any stable version of Asterisk 1.6.x or 1.8.x Were you able to do this?
I need someone to send me Symbian tutorial and tell me how and where to start build my Symbian application. and where to find more tutorial and help in future , am android developer and have a very good knowledge of mobile application development but need to start developing Symbian.
I need someone to send me Symbian tutorial and tell me how and where to start build my Symbian application. and where to find more tutorial and help in future , am android developer and have a very good knowledge of mobile application development but need to start developing Symbian.
I need someone to tell me how to start learning Symbian programming and send me some tutorial and where to find more tutorial to build my Symbian application.
...interfere with the running application performance. You can use the FFMPEG libraries or you can use your own approach (it's required that you point which approach you will follow in order to get considered), the video encoding it's not important while it have any (so no raw video, which would yield a very big file size), my preference for the encoding is AVI as the container and H.264 for the codec. The file should be saved locally and the folder can be hardcoded into the application, but always in the external storage. Please, advice which should be minimum Android version for this application, also write 'Android Video' at the start of your message you I can know you have read the whole specifications, messages without this keyword will not be even ...
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 5. For server B installation need easy to use ISO image that could be booted from USB flash drive, and those USB flash drive will be delivered to our Server B type client (ther termination provider) 6. For server B need portion of asterisk web gui for adding gateways..prefix or so, viewing active calls, billing cdr,etc.
Need to install AMR-NB codec for any stable version of Asterisk 1.6.x or 1.8.x Were you able to do this?
please refer the cicso offer. I need this but cisco's offer is not ready now. the price is negiotiable. Thanks! or your would like to contact ma at .
please refer the cicso offer. I need this but cisco's offer is not ready now. the price is negiotiable. Thanks! or your would like to contact ma at .
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 5. For server B installation need easy to use ISO image that could be booted from USB flash drive, and those USB flash drive will be delivered to our Server B type client (ther termination provider) 6. For server B need portion of asterisk web gui for adding gateways..prefix or so, viewing active calls, billing cdr,etc.
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...minimize the BW in client side with good quality of voice . We 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 PC And DDWRTRouter firmware 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 5. For server B installation need easy to use ISO image that could be booted from USB flash...
...minimize the BW in client side with good quality of voice . We 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 PC And DDWRTRouter firmware 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 5. For server B installation need easy to use ISO image that could be booted from USB flash...
...powers phones costing Rs 3,000 to Rs 45,000 and that helps it attract a lot of consumers," added Manasi. Apart from the wide adoption by phone companies, Android has also been helped by the demise of Symbian operating system, which was earlier used by Nokia. After Nokia partnered with Microsoft for Windows Phone it decided to stop making Symbian-based phones. Manasi said the first quarter of this year was the last time her research firm tracked shipment of Symbian phones. In the absence of any other viable low-cost smartphone OS, Android has seen its popularity surge. In 2010, Symbian had over 40% share in India's smartphone market. Manasi said that Windows Phone was doing well but it required more time. "Windows Phone is steadily grow...
...BW in client side with good quality of voice . We 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 PC And DDWRTRouter firmware 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 5. For server B installation need easy to use ISO image that could be booted from USB flash d...
...goal to minimize the BW in client side with good quality of voice . We 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 5. For server B installation need easy to use ISO image that could be booted from USB flash dri...
...Voicemail and more. Video support. A calling card platform with IP IVR including pin or pin-less scenarios. Callback solution supports all types of triggering methods: SMS, missed-call, web Callback. Web Callshop interface. Web self-care portal for end users and support for Online payments.. SMS features for both wholesale and retail services, using HTTP, SIP and SMPP protocols. Mobile SIP for Symbian, Windows Mobile, iPhone, Android and BlackBerry users. Windows communicator with Instant Messaging, SMS, Voicemail and many other modules. Multi-tenant, multi-user IP PBX platform for offering hosted services. Instant messanger functionality, XMAPP protocol support, presence, chat, file transfer. Unified messaging, Voicemail/SMS forwarding to SMS or email, Voicemail trans...
i want to transcode high definition video using x264 codec to produce smaller size better quality video file. i have purchased a program that does what i want but the vendor seems to have issues,so i would like someone to make me a similair program for personal private use
I am writing a html5 page for chrome which can simply playback vp9 and h264 videos for a video quality analysis study which involves participants to watch the video. The rationale of using HTML5 to playback the videos is that only Chrome support the latest VP9 codec from Google. This HTML5 page only loads videos locally and it's a OFFLINE page. What I have already done: I have stored the video playlist in an array. Each video last 10 seconds. A 10 seconds countdown video will be played after each test video is finished as participants suppose to rate the viewed video in 10 seconds. What I need you to do: I have a popup box (I have used ) which will be prompted asking for user feedback after each test video is viewed during the 10 seconds evaluation time while countdow...
I need a Freeswitch autodialer which can send Each calls disconnection @ 4secs OR whatever duration I set it will follow the same SIP CAlls Codec 729,723 It should send each call with a random cli IT shud pick the calling numbers from DB randomly not in sequence IT supports multiple campaigns at real time I have CENTOS OS ready.
...cards [] Setup Skype connect to connect PBX via pin authenticate.. So the customer able to dial given Skype contact and authenticate pin then make calls. (3 Rates card will be applied) > Generate 500 pins for Skype connect [] Create 500 sip accounts for soft phone connection. It should be pin less authenticate. Customer able to dial define country local phone number and billed. [] G729 codec setup [] Security implementing to server. [] Basic training (maximum 1 hr) how to operate ,add more cards and basic trouble shooting. [] In addition a2biling must be install to another asterisk server. (only install) I'm looking for long term relationship. Near future I'll add more servers to my business. So there will be required more technical support. I need t...
Setup a Linux media streaming Server using Http 5 so it can be viewed on all platforms and most browsers (Computers, Mac, Linux, Androids OS devices, iphone, ipad, Blackberry, symbian, Java Phones, WindowsMobile , Tablet pc etc..). Provide a means to load the Media Music Videos or Movies into the Server Provide a means to categorize the Videos or Movies. Many Directories and Sub-directories can be Created by Admin Provide 2 Streaming Options. 1. Provide A Play List Version So videos to be played can be added to the play list by the adminitrator and Streamed to audience Viewers has no control over what is played they just click and view 2. Provide A VOD version so users can make their own selections but will need a username and pass to do so. A Linux Test Server will b...
...features must be available in the dialer: Making & Receiving VoIP Calls Over 3G,4G, WiFi, GPRS. -Video calling -Voice calling -Messaging -Filesharing The dialer should also provide Vpn/tunneling support for working in SIP/Voip blocked countries. Mobile Dialer should support SIP/H323 for signaling and should work with any standard SIP server. Mobile Dialer supports G729, G723.1 and G.711 codec for sending audio stream. Mobile Dialer can run behind NAT or on private IP with VPN Tunneling embedded. Integrate with phonebook contacts Dial, Hang up, Redial Call Hold Call Mute Call Forwarding Call Waiting Speaker Phone Keypad Display Call History: All, Recent, Dialed, Missed Account Setup Work with all standard SIP servers Supports all standard SIP supported swi...
...%{HTTP_USER_AGENT} "sch-|sec-|send|seri|sgh-|shar|sie-|siem|smal|smar|sony|sph-|symb|t-mo" [NC,OR] RewriteCond %{HTTP_USER_AGENT} "teli|tim-|tosh|tsm-|upg1|upsi|vk-v|voda|w3cs|wap-|wapa|wapi" [NC,OR] RewriteCond %{HTTP_USER_AGENT} "wapp|wapr|webc|winw|winw|xda|xda-" [NC,OR] RewriteCond %{HTTP_USER_AGENT} "||windowssce|iemobile|mini|mmp" [NC,OR] RewriteCond %{HTTP_USER_AGENT} "symbian|midp|wap|phone|pocket|mobile|pda|psp" [NC] RewriteCond %{HTTP_USER_AGENT} !macintosh [NC] # Check if we're not already on the mobile site RewriteCond %{HTTP_HOST} !^m. # Can not read and write cookie in same request, must duplicate condition RewriteCond %{QUERY_STRING} !(^|&)m=0(&|$) # Check to make sure we haven&...
...Our main goal to minimize the BW in client side with good quality of voice . We 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 5. For server B installation need easy to use ISO image that could be booted from USB flash drive, an...
This is a repost! I need an application that can send/receive calls through SIP (Can use any sip stack like sipdroid) and forward it to the GSM network...background o Must be very lightweight to run on small memory devices o Configure SIP Accounts o Create a mini dialplan to choose which chip to use based on techprefix Sip Requirements o Register on Sip Proxy/Gateway o Receive call authenticated by IP, user/pass or no authentication. o Make calls with or without authentication o Forward DTMF digits using RFC2833 or inband o Use codec G711 and GSM o Be able to use codec G729 o Create a mini dialplan to choose which chip to use based on techprefix, per SIP Account o Hability to register more than 1 SIP account (but receive just 1 simultaneous call) o Be able to store a CSV wi...
This is a repost! I need an application that can send/receive calls through SIP (Can use any sip stack like sipdroid) and forward it to the GSM network...background o Must be very lightweight to run on small memory devices o Configure SIP Accounts o Create a mini dialplan to choose which chip to use based on techprefix Sip Requirements o Register on Sip Proxy/Gateway o Receive call authenticated by IP, user/pass or no authentication. o Make calls with or without authentication o Forward DTMF digits using RFC2833 or inband o Use codec G711 and GSM o Be able to use codec G729 o Create a mini dialplan to choose which chip to use based on techprefix, per SIP Account o Hability to register more than 1 SIP account (but receive just 1 simultaneous call) o Be able to store a CSV wi...
This is a repost! I need an application that can send/receive calls through SIP (Can use any sip stack like sipdroid) and forward it to the GSM network...background o Must be very lightweight to run on small memory devices o Configure SIP Accounts o Create a mini dialplan to choose which chip to use based on techprefix Sip Requirements o Register on Sip Proxy/Gateway o Receive call authenticated by IP, user/pass or no authentication. o Make calls with or without authentication o Forward DTMF digits using RFC2833 or inband o Use codec G711 and GSM o Be able to use codec G729 o Create a mini dialplan to choose which chip to use based on techprefix, per SIP Account o Hability to register more than 1 SIP account (but receive just 1 simultaneous call) o Be able to store a CSV wi...
This is a repost! I need an application that can send/receive calls through SIP (Can use any sip stack like sipdroid) and forward it to the GSM network...background o Must be very lightweight to run on small memory devices o Configure SIP Accounts o Create a mini dialplan to choose which chip to use based on techprefix Sip Requirements o Register on Sip Proxy/Gateway o Receive call authenticated by IP, user/pass or no authentication. o Make calls with or without authentication o Forward DTMF digits using RFC2833 or inband o Use codec G711 and GSM o Be able to use codec G729 o Create a mini dialplan to choose which chip to use based on techprefix, per SIP Account o Hability to register more than 1 SIP account (but receive just 1 simultaneous call) o Be able to store a CSV wi...
...background o Must be very lightweight to run on small memory devices o Configure SIP Accounts o Create a mini dialplan to choose which chip to use based on techprefix Sip Requirements o Register on Sip Proxy/Gateway o Receive call authenticated by IP, user/pass or no authentication. o Make calls with or without authentication o Forward DTMF digits using RFC2833 or inband o Use codec G711 and GSM o Be able to use codec G729 o Create a mini dialplan to choose which chip to use based on techprefix, per SIP Account o Hability to register more than 1 SIP account (but receive just 1 simultaneous call) o Be able to store a CSV with all calls made Skills required: Android, Software Architecture, VoIP See more: sip gsm gateway android, android, sip gateway android, gsm gateway an...
This is a repost! I need an application that can send/receive calls through SIP (Can use any sip stack like sipdroid) and forward it to the GSM network...background o Must be very lightweight to run on small memory devices o Configure SIP Accounts o Create a mini dialplan to choose which chip to use based on techprefix Sip Requirements o Register on Sip Proxy/Gateway o Receive call authenticated by IP, user/pass or no authentication. o Make calls with or without authentication o Forward DTMF digits using RFC2833 or inband o Use codec G711 and GSM o Be able to use codec G729 o Create a mini dialplan to choose which chip to use based on techprefix, per SIP Account o Hability to register more than 1 SIP account (but receive just 1 simultaneous call) o Be able to store a CSV wi...
This is a repost! I need an application that can send/receive calls through SIP (Can use any sip stack like sipdroid) and forward it to the GSM network...background o Must be very lightweight to run on small memory devices o Configure SIP Accounts o Create a mini dialplan to choose which chip to use based on techprefix Sip Requirements o Register on Sip Proxy/Gateway o Receive call authenticated by IP, user/pass or no authentication. o Make calls with or without authentication o Forward DTMF digits using RFC2833 or inband o Use codec G711 and GSM o Be able to use codec G729 o Create a mini dialplan to choose which chip to use based on techprefix, per SIP Account o Hability to register more than 1 SIP account (but receive just 1 simultaneous call) o Be able to store a CSV wi...
...Our main goal to minimize the BW in client side with good quality of voice . We 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 5. For server B installation need easy to use ISO image that could be booted from USB flash drive, an...
...Our main goal to minimize the BW in client side with good quality of voice . We 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 5. For server B installation need easy to use ISO image that could be booted from USB flash drive, an...
...Our main goal to minimize the BW in client side with good quality of voice . We 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 5. For server B installation need easy to use ISO image that could be booted from USB flash drive, an...
...Our main goal to minimize the BW in client side with good quality of voice . We 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 5. For server B installation need easy to use ISO image that could be booted from USB flash drive, an...
...asterisk servers to compress and bypass voice packets, we will provide some demonstration of the existing services to the candidates after interviewing and once we believed that you can do it. Server A = Asterisk Server Server B = Asterisk Client 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 5. For server B installation need easy to use ISO image that could be booted from USB flash dri...
