Symbian codecprojekty

We need to have the software for transcoding the live satellite streams. It has to have next specs: Incoming streams: multicast MPTS signal; able to select a/v tracks bi PID; MPEG-2 video codec; MP3, AC3, AAC audio codecs; CloseCaptions format - CEA-608; EPG info; UDP/RTP transport protocols. Output streams: H.254, H.265 video codecs; AAC, AC3 audio codecs; CEA-608, AMF CloseCaptions; EPG Info; RTMP, HLS transport protocols. Also it must support NVIDIA Kepler GPU Cards.

I Need a Gerber file to be sent to my Manufacturer for a specific system. at this point, all I have is a PDF version of schematics & Bill of material. however, some of the component...however, some of the component are obsolete. what i need you to do is to re-engineer the schematics using newer component (available for at least in the next 10 years), do some modification to suit the system requirement, Update the Bill of Material, design the new PCB, and produce the Gerber file. (Manufacturer is Using Altium). details: Major component is: 1 LQFP-144 IC, 1 Flash Memory 1 CODEC IC Some switching regulator for power supply, Interfacing : 1 Microphone / Line In 1 Audio Out / Speaker 1 Ethernet Controller (Currently designed with Mikro ETH 100 Board from Mikroelektronika...

as similar to your previous completed project, mainly i need to compare these three codec to findout the best codec for different environment, implementing voip over cognitive radio network. also please do let me know the time required, here this is what i required for my project and my deadline is very near, hope to hear you soon. thank you.

Native apps that works in the plattform listed below without with high speed platform order of priority arrangement 1: Java Phones(Via Java2ME .jar extention client users 45%) 2: Blackberry (client users 15%) 3: Android (client users 15%) 4: Windows Phone (client users 5%) 5: symbian (client users 10%) 6: Iphone (client users 10%) Features Users section 1: Report link on every page, forget password, login and sign up with auto generated unique ID Pin for for different users 2: update status,comments,shares,likes, Users profiles and edit users profile, users pictures uploads 3: message system,notification system, friends list, friends request by unique search using unique Id Pin,phone contact,meet people and by usernames 4: members and friend...

...limitation that licenced user (Primary User) should not be interfered by using the channel (spectrum) and in general condition this could be the main cause of degradation in quality of service on VoIP. Hence, it is important to develop a perfect model that will minimise the interference by sensing the idle state of PU and transmit SU through error free channel. There are other factors like; selected codec, payloads, protocols etc. to increase the efficiency of VoIP. In this project, several published model will be analysed and accordingly tested using the simulator Network Simulator 2 (NS2) implementing VoIP on CRN then validated the best proposed model by mathematical model. Aims: Develop model for implementing VoIP on CRN, which maximise the quality of service. Objective...

Hi Guys, We are looking for an Excel database containing mobile application development companies specifically based out of US/Canada. The companies should work on the technologies like iOS, Android, Blackberry, Windows mobile, J2ME, Symbian & HTML5. The target market would be anything from Hospitality or Mobile Gaming, Health Care or Nonprofit, Product or Service company (everything that relates to mobile development). Fields Required: Company Name, Website, Phone No & Address. We are not looking for hiring people nor we're looking for scraping out data from a website. ONLY BID IF YOU ALREADY HAVE A DATABASE as the requirement is URGENT. And, most importantly, please send a sample containing at least 50 leads. A BID WITHOUT A SAMPLE WILL NOT BE ENTERTAINED! ...

Project Description: This is a repost! I need an application that can send/receive calls through SIP (Can use any sip stack like sipdroid) and fo...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...

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...phone. Stealth SMS do exist i.e. check HushSMS Full Version and Ping SMS / Silent SMS in Android market. Then Target phone should return at least Location area Code (LAC) and its Cell ID (CID) data. Optionally it would be a great future to receive GPS coordinates for devices with integrated GPS. NO application to be installed on the target phone. This should work on all mobile phone platforms (Symbian,Blacbkerry, Apple, Android etc ) and all networks. NO milestone payments will be released until testing is complete and full payment will be made on completion. I will not add skype id’s to chat for payments in advance and so on. If you believe of being capable of completing this project please mail me the technical details on how you plan to complete this project. Plea...

...phone. Stealth SMS do exist i.e. check HushSMS Full Version and Ping SMS / Silent SMS in Android market. Then Target phone should return at least Location area Code (LAC) and its Cell ID (CID) data. Optionally it would be a great future to receive GPS coordinates for devices with integrated GPS. NO application to be installed on the target phone. This should work on all mobile phone platforms (Symbian,Blacbkerry, Apple, Android etc ) and all networks. NO milestone payments will be released until testing is complete and full payment will be made on completion. I will not add skype id’s to chat for payments in advance and so on. If you believe of being capable of completing this project please mail me the technical details on how you plan to complete this project. Plea...

I am looking for someone to write a SIP dialer that will be on the iPhone platform that will register to my SIP server and be able to make calls to users on my network or out. This dialer will need to have the G729a/b codec integrated (I will pay for the licence for the codec). Regards,

I am looking for someone to write a SIP dialer that will be on the iPhone platform that will register to my SIP server and be able to make calls to users on my network or out. This dialer will need to have the G729a/b codec integrated (I will pay for the licence for the codec). Regards,

I am looking for someone to write a SIP dialer that will be on the iPhone platform that will register to my SIP server and be able to make calls to users on my network or out. This dialer will need to have the G729a/b codec integrated (I will pay for the licence for the codec). Regards,

...features must be available on the dialler: Making & Receiving VoIP Calls Over 3G,4G, WiFi, GPRS. • Video calling • Voice calling • Messaging • Filesharing The dialer should also provide Vpn/tunnelling support for working in SIP/VoIP blocked countries. Mobile Dialer should support SIP for signalling 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 Tunnelling embedded. 1. Integrate with phonebook contacts 2. Dial, Hang up, Redial 3. Call Hold 4. Call Mute 5. Call Forwarding 6. Call Waiting 7. Speaker Phone 8. Keypad Display 9. Call History: All, Recent, Dialled, Missed 10. Account Setup 11. Work with all st...

Hi, We are a company based in Mumbai. Name of the company- Rud Creations We are looking for serious Newcomers people who will work in our Company. You will need to work Online. You Should have one of one of this Skills to Qualify- -HTML/CSS -PHP/MySQL -Flash -Graphics Design -Develop Native Apps and Games for Android, iOS, Windows and Symbian -Web Design -SEO -Custom Software Development -Branding and Brand Promotion -Advertising -Marketing and Managing Please Bid with your Skills, We don't need any Qualifications or Education, If you have the Skill then you are most Welcome. More details will be given by PM later.

...that provider.. just check the Example voip-app (Zoiper ) in config section > (Outbound Prox) this must appear automatic> (MY SIP) B).Another Example a Voip-mobile must have a (Features) Built in > ( DTMF) which Contain ( Disable/RFC-2833/sip info/inban, C).And Video-codec should be set on Automatic' Mobile voip Video Phone/apps is a next generation stylish SIP video phone with large adjustable color LCD, VGA resolution many phones contain different camera, and real-time bandwidth-efficient H.264/H.263 video codec that renders stunning picture quality over mobile voip-apps and public Internet. D) Similar to zyper, the voice configuration ======================================================================================== FLOW:- 1) After ...

...Contain ( Disable/RFC-2833/sip info/inban, C) I see you already send me Proper Example of the Above in you doc-files a few Days ago'So please don't miss understand me and think I am saying/explaining as if you don't know the Right spec for My requested App. D).And Video-codec should be set on Automatic' Mobile voip Video Phone/apps is a next generation stylish SIP video phone with large adjustable color LCD, VGA resolution many phones contain different camera, and real-time bandwidth-efficient H.264/H.263 video codec that renders stunning picture quality over mobile voip-apps and public Internet. E) Similar to zyper, the voice configuration ======================================================================================== FLOW:- 1...

...Contain ( Disable/RFC-2833/sip info/inban, C) I see you already send me Proper Example of the Above in you doc-files a few Days ago'So please don't miss understand me and think I am saying/explaining as if you don't know the Right spec for My requested App. D).And Video-codec should be set on Automatic' Mobile voip Video Phone/apps is a next generation stylish SIP video phone with large adjustable color LCD, VGA resolution many phones contain different camera, and real-time bandwidth-efficient H.264/H.263 video codec that renders stunning picture quality over mobile voip-apps and public Internet. E) Similar to zyper, the voice configuration ======================================================================================== FLOW:- 1...

First you have to set the parameter 'qualify=yes' on all the sip extensions at sip.conf. This parameter will become your extension reachable to your server. Check your dialplan rule at extensions.conf. Must Have something like this: exten => _2X,1,Dial(SIP/${EXTEN},300,Ttr) exten => _2X,n,Hangup() At the end, but very important, check the codec you are using. You are using the 3G network, your connection probably will have an high latency. Set on your softphone codecs that will give you a better compression like GSM or G729. Just a last note: If you have firewall, check if the sip ports are opened in two ways, and not only out-side to in-side.

Hi, I need unlimited freelancers,Team from Hyderabad,Andhra Pradesh,India to work on my ongoing projects.I need you people to give me the updates daily through phone,skype,emails etc...I'll provide you my client on conference all you need to deal with technical stuffs as I'll deal with managerial should be keen i...provide you my client on conference all you need to deal with technical stuffs as I'll deal with managerial should be keen in understanding my client provide me you contact mobile number,skype id,email id in order to get in conversations with this will be a long term my projects relates with ERP,Oracle,VoIP,Iphone,Ipad,Blackberry,Palm,Android,Symbian,etc...My projects will be very complicated. Thanks for bidding!!!

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I want an application which can be used for screen sharing by my clients (i have a website where people interact, one is a learner and other one a coach) the application must be - able to run on windows, mac and linux screen sharing with audio(both sides) start of shar...red5 as our server for our previous application(we had to scrap it as there was huge delay in video and no audio ) the delay must not be more than 3-4 seconds in video we prefer h.264 encoding for video and aac for audio, for h.264 you can use GPL library the quality of video must be absolutely clear like that of skype sharing or teamviewer, no compromise with that and please no flash sv2 codec as we have tried it earlier it didn't worked well we are not sure about budget so please bid what you th...

...features must be available on the dialler: Making & Receiving VoIP Calls Over 3G,4G, WiFi, GPRS. • Video calling • Voice calling • Messaging • Filesharing The dialer should also provide Vpn/tunnelling support for working in SIP/VoIP blocked countries. Mobile Dialer should support SIP for signalling 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 Tunnelling embedded. 1. Integrate with phonebook contacts 2. Dial, Hang up, Redial 3. Call Hold 4. Call Mute 5. Call Forwarding 6. Call Waiting 7. Speaker Phone 8. Keypad Display 9. Call History: All, Recent, Dialled, Missed 10. Account Setup 11. Work with all st...

have a symbian whatsapp back-up file () under S60 and need to decrypt to a readable file due to change of sim card and phone number. chat history readable under text or excel or other software is fine

Hi, I need unlimited freelancers,Team from Hyderabad,Andhra Pradesh,India to work on my ongoing projects.I need you people to give me the updates daily through phone,skype,emails etc...I'll provide you my client on conference all you need to deal with technical stuffs as I'll deal with managerial should be keen i...provide you my client on conference all you need to deal with technical stuffs as I'll deal with managerial should be keen in understanding my client provide me you contact mobile number,skype id,email id in order to get in conversations with this will be a long term my projects relates with ERP,Oracle,VoIP,Iphone,Ipad,Blackberry,Palm,Android,Symbian,etc...My projects will be very complicated. Thanks for bidding!!!

I need an application that runs on Android phones (4.0 or above) and can send/receive calls through SIP (Can use any sip stack like sipdroid) and forward it to the GSM network. The application should then...the APP - Enable to route calls from SIP to GSM - Must run on background - Must be very lightweight to run on small memory devices - Configure SIP Accounts. Sip Requirements - Register on Sip Proxy/Gateway - Receive call authenticated by IP, user/pass or no authentication. - Make calls with or without authentication - Forward DTMF digits using RFC2833 or inband - Use codec G711 and GSM - Be able to use codec G729 - Receive through GSM one 1 simultaneous call and rout it to a predefined SIP client - Be able to store a CSV with all calls made Skills required...

...under your own brand name. Clients can download this mobile SIP app with full configuration of your service (i.e. switch IP, port, brand name, balance display etc). Your client does not need to configure anything except inserting PIN of their account. Available on Multiple Platforms: This SIP dialer runs on Symbian, iPhone, Windows Mobile 5 & 6, Blackberry (GSM and CDMA) and Android platforms Technical Specifications: Protocol: Session Initiation Protocol (SIP). Supported Platforms: Symbian, Windows Mobile, Blackberry, iPhone and Android. Supported Codecs: g729, AMR, GSM and g711. Compatibility with various Softswitches: this mobile SIP app is compatible with all standard SIP softswitches. Note : In Market Many Mobile dialers already running so if someone have e...

...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 ...

...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...

...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 ...

...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 ...

...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 ...

...calling -Messaging (ability to enable/disable) -Filesharing (ability to enable/disable) Making calls over local numbers (calling card mode) (pinless mode) if internet is not present and user wants to use VoIP calls. . Mobile Dialer should support SIP/H323 for signaling and should work with any standard SIP server and based on RFC3261. Mobile Dialer most supports G729, G723.1 and G.711 codec for sending audio stream. Mobile Dialer most be able to run behind NAT or on private IP with VPN Tunneling embedded. Account Signup Account Setup Integrate with phonebook contacts and favorites Call History: All, Recent, Dialed, Missed (with redialing) Registration Status: (Registred /Unregistred /Calling Card Mode / CID Auth Mode Call Status - Time of call Speaker Phone ...

...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 (Teles 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. 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 (the termination provider) 6. For server B need it to be installed at any kind of routers such as Mikrotik router, or any other router u recommend that would be easy to use, and easy to be installed just plug and play method. Specific Need: A to convert all calls in G729 codec and server B to re...

...my cometchat web editions look according to this design. 2. You have to integrate this chat with the mobile version of my site as like facebook mobile edition has a chat system integrated with, you have to also made a mobile edition of this cometchat with new design (cometchat already has a mobile version), I need a mobile version of this chat system for IOS, Android, Blackberry, Windows, Symbian Mobile with a exclusive design (in that case you have to make the design) 3. You know that cometchat has a desktop version it is build with php it works on MAC and Windows you have to develop it also take a look on here- I need my messenger for windows, mac, linux will be like that. Please bid if you can do it only you will be paid well and if you can do it I will

Freelance Linux Engineer Needed ASAP (Amazon AMI) with experience in FFMPEG with Codec configuration for PHP.

Searching for a talented app developer who is extremely knowledgeable with how Skype, tango, google hangouts, and oovoo work. This app will require the video call feature, so the developer must have extensive knowledge in this area...there will not be any profiles. The users will only be able to make video calls over Wi-Fi or using cellular data like oovoo, up to 4 people at a time (think oovoo). It is a very simple app. The developer must be familiar with using SIP protocol, must know the best codecs for audio and video encoding, and know FreePBX, VOIPSwitch, a2billing system for my app. It is helpful to understand H264 codec for the video encoding and G722, G711, G712 codecs for audio encoding. Please ask any clarification questions before bidding. Thank you and ...

i need installer of following video codecs. Xvideo Codec divx codec mkv codec you can use opensource codecs .more will be discussed with selected bidder -must include 3 installers -must include intallers sourcecode/script etc

...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...

38 unique content articles, mostly review type. Keyword desnity should ...gimp - review (use the idea of free alternative to photoshop) download manager - review facebook messenger - review oovoo - review winamp - review gta san andreas - - review part could be about gta san andreas and other half of it would be around the fact that this is an update for GTa4 counter strike - review nero - (DVD CD burning software) review xvid codec - review, why you need it firefox - review spider solitaire - (standard windows game) viber for windows - review - ad blocker - review steam - (steam engine, for playing CS and etc) review spotify - review, how to use it cheat engine - what is, review weather channel desktop - review

...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 ...