Symbian codectrabajos

1. SIP dialer working in Android/IOS/Windows/Symbian environment for blocked countries. 2. Integration with my soft switch or you can also offer if you have something similar. 3. We want to resell this dialer to another users with changing the brand logo and name. So we can make a revenue share for this.

Required to write a Simple application of Generation of SMS based on User Selection. Show the received information from the specific user and display on screen. Basically this is like a small app used for convenience rather than typing an SMS and send button.

i want to port an app named SIMPLERSERVER original made for symbian to android.(full details and download sis version here ). The app was later made for pc.(file attached). The app is a tunneler used to apply some tweaks to free website provided by my ISP so i can surf the internet for free.

i need a app for VOIP calls (dialer) i have a example of, but i need to redesign to my name company and certain features on my IPswitch please give me a phone number including country code and city etc. but we can be in contact over this site and email to provide to you the pictures and we need to talk to explained to you. thanks Ubaldo

I'd like an SDP parser () written with scala's parser combinators. I will provide a lot of different examples from a server log, that the parser must be able to parse. Required features: -Parse test messages -Create SDP from scratch -Insert and audio codec at a certain position -Remove audio codec -Filter audio codecs with a provided list

i need a app for VOIP calls (dialer) i have a example of, but i need to redesign to my name company and certain features on my IPswitch please give me a phone number including country code and city etc. but we can be in contact over this site and email to provide to you the pictures and we need to talk to explained to you. thanks Ubaldo

Dear All, I want to transcode all of my calls to g723 , whatever codec is coming , system should be capable of it to transcode this to g723 , anyone who have done this before will be appreciated if you have not done this before please do no waste your and mine time thanks

.Its An Instant messaging app localised for Nigeria There are many apps with similar functionality to that which i have described But i want this one to have .."Our Own Humorous Lingua " The difference with this instant messaging app will be the content in funny icons, it will be in the local vernacular way of speaking Engl...little circles or any shape or just short abbreviation in miniature forms.. It should be proprietary,A cross-platform instant messaging subscription service for smartphones with Internet access. In addition to text messaging, users can send each other images, video, and audio media messages. The software should be available for Google Android, BlackBerry OS, Apple iOS, selected Nokia Series 40, Symbian, selected Nokia Asha platform and Micr...

...logo, and Company Name on the application. 2. Options on account signup page: I already have account I already have sip account Sign up for VibeVoip option for contacts. 4. Call logs will be only numbers - not the whole sip URI. 5. Under Settings -> the options to enable some or all of the options by default. 6. Help and about us Links to our web site. 7. support for codec G729 (VibeVoip Customers only ) 8. Provide source code and detailed documentation to customize basic info like splash screen, icon, links etc with proper reference to image and xml files. NO upfront payments, 50% on upload to appstore and playstore and 50% after testing and bug fixing. Must have prior experience Will give work only after initial interview, Apply only if you agree. Communic...

...unlimited. 5. For server B installation may an ISO of USB image mounting of easy with some free tools which can be done using Puppy Linux OS. 6. For For configuration of both A and B there will some GUI (i.e. Web Interface) for adding gateways prefixes, 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 and forward them to gateways g729 codec. 2. Some kind of bandwidth compression is required for better performance on low bandwidth connection. i.e. Traffic Squeezer. 3. server A to Server B voice traffic will be encrypted so that voice port blocked bandwidth can be used for termination OR you can develop this using wrt or openwrt for tp link or any router 1 ) We ne...

...be unlimited. 5. For server B installation may an ISO of USB image mounting of easy with some free tools which can be done using Puppy Linux OS. 6. For For configuration of both A and B there will some GUI (i.e. Web Interface) for adding gateways prefixes, 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 and forward them to gateways g729 codec. 2. Some kind of bandwidth compression is required for better performance on low bandwidth connection. i.e. Traffic Squeezer. 3. server A to Server B voice traffic will be encrypted so that voice port blocked bandwidth can be used for termination OR you can develop this using wrt or openwrt for tp link or any router 1 ) We need Compl...

I need a softphone for all platforms. To register sip with codec; speex, gsm, g729 and other that you may suggest. It must be put in a open source to have g729 as free to use. the android can be downloaded on

Hi, I have an Asterisk server but calls are crackling. I upgraded the server to 2GB ram and still calls are crackling. top in Linux shell shows 0 load; the server is performing very well, but asterisk calls are breaking up. Need to find the root cause and fix this. Possible causes: - Network (mis-)configuration - Codec used (am using ulaw) - Termination provider - other issue(s)

Hi, I have an Asterisk server but calls are crackling. I upgraded the server to 2GB ram and still calls are crackling. top in Linux shell shows 0 load; the server is performing very well, but asterisk calls are breaking up. Need to find the root cause and fix this. Possible causes: - Network (mis-)configuration - Codec used (am using ulaw) - Termination provider - other issue(s)

...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 ...

...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 ...

The project is to provide German TV channel link. Good quality garanty - HLS - 6 segments per playlist (m3u8) - Minimum segment time 5 seconds, must contain at least one R-frame - advised average bit rate for Video 900 Kbps VBR (= variable bit rate) including audio (advised bit ...project is to provide German TV channel link. Good quality garanty - HLS - 6 segments per playlist (m3u8) - Minimum segment time 5 seconds, must contain at least one R-frame - advised average bit rate for Video 900 Kbps VBR (= variable bit rate) including audio (advised bit rate 96 Kbps) for SD quality. - R-frames settings is 3 or 4 - B-frames per GOP (group of pictures) is 2 - Codec .H264 for video - Codec AAC for audio, lc (low complexity) - 360p Minimun 25 channels We pay every...

The project is to developp a solution ready to work, install on Linux server who will convert live realtime streaming from RTMP into HLS. The input area is a link RTMP, and the output is a link HLS. We must be able to convert several link in same time, the limit will be the power of ...RTMP, and the output is a link HLS. We must be able to convert several link in same time, the limit will be the power of the server. Set up of HLS : - 6 segments per playlist (m3u8) - Minimum segment time 5 seconds, must contain at least one R-frame - advised average bit rate for Video 900 Kbps VBR - R-frames settings is 3 or 4 - B-frames per GOP (group of pictures) is 2 - Codec .H264 for video - Codec AAC for audio The project include is to develop and to install. Option : Ac...

...be unlimited. 5. For server B installation may an ISO of USB image mounting of easy with some free tools which can be done using Puppy Linux OS. 6. For For configuration of both A and B there will some GUI (i.e. Web Interface) for adding gateways prefixes, 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 and forward them to gateways g729 codec. 2. Some kind of bandwidth compression is required for better performance on low bandwidth connection. i.e. Traffic Squeezer. 3. server A to Server B voice traffic will be encrypted so that voice port blocked bandwidth can be used for termination OR you can develop this using wrt or openwrt for tp link or any router 1 ) We need Compl...

The project is very simple. I need a simple Number Scanner application An application which scans numbers(digits) FYI : This Project should be compatible with all the mobile devices (IPhone, Android, Symbian etc) Budget : $150 (Max)

...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 ...

I am looking for a bit of software to work on windows desktop .. I need a icecast2 encoder to connect to the icecast 2 server's streaming mp3 or aac codec, We need to be able to have a direct line input to the encoder from the record in on the sound card to the software .. Also will wound need a MP3 player on the encoder that will play the mp3 over the line in from the live dj.. Will need some kind of UV meter for the audio coming in. The config file will need to an external file as well to it can be changed.. Hope this is very clear if not send me a message ..

...unlimited. 5. For server B installation may an ISO of USB image mounting of easy with some free tools which can be done using Puppy Linux OS. 6. For For configuration of both A and B there will some GUI (i.e. Web Interface) for adding gateways prefixes, 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 and forward them to gateways g729 codec. 2. Some kind of bandwidth compression is required for better performance on low bandwidth connection. i.e. Traffic Squeezer. 3. server A to Server B voice traffic will be encrypted so that voice port blocked bandwidth can be used for termination OR you can develop this using wrt or openwrt for tp link or any router 1 ) We ne...

Project Description: Project...unlimited. 5. For server B installation may an ISO of USB image mounting of easy with some free tools which can be done using Puppy Linux OS. 6. For For configuration of both A and B there will some GUI (i.e. Web Interface) for adding gateways prefixes, 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 and forward them to gateways g729 codec. 2. Some kind of bandwidth compression is required for better performance on low bandwidth connection. i.e. Traffic Squeezer. 3. server A to Server B voice traffic will be encrypted so that voice port blocked bandwidth can be used for termination OR you can develop this using wrt or openwrt for tp l...

...for SIP dialer software for PC, Mobile Phones & Web more detail are as under: 1. User friendly interface 2. Support SIP for signaling 3. Supports G729, G723.1 and G.711 codec for sending audio stream 4. Can run behind NAT or on private IP with proxy tunneling embedded ( *** Most Important *** If you really don't know how to integrate a proxy server inside app please do not bid for this project ) 5. This software will run on the phone having Nokia Symbian, Android, iPhone, iPad, blackberry, and windows Mobile. 6. Must be running from the application layer. The most important part is to add G729 and G723.1 codec in it. 7. Show user balance details using HTTP API or Get Method 8. Auto Sync of Balance. 9. Real Time Sip status messages. 10. Call ...

...integration, change the them, colors, etc... 2. Show customer balance in the top of the dialler screen by using get data from our webservice (a2billing). 3. Automatically register with a2billing use IMEI as SIP User right after installation. 4. Automatically credit X amount to the new created account. 5. Option to refeel the account through Paypal and Google Wallet. 6. Use G.729 codec and fall back to codec that use less bandwidth. 7. Option for customer to find the rate based on the selected currency. 8. Option for customer to contact support via email and subject their IMEI. 9. Option to Keep or delete the call history. 10. Option to create favourite numbers. 11. Option to educate customer with dialling method accepted on the a2billing. 12. Random pop-up messa...

Recently We have installed Freeswitch with ASTPP billing system. At this time we need a sip softphone that should support above listed smartphones. Also we need configure VPN ( PPTP or L2TP . etc) inside the app in order to avoid blocking from ISP. If any one who worked with the subjected matter with excellent skills please contact me. SKILL Required ---------------------- Ability to develop sip softphone for listed smartphone. Ability to configure PPTP/L2TP or other VPN Protocol in CentOS 6.5 server (It should support more then 500 user connection to VPN ) Thank you

I will provide a compiled video codec as libs with a test interface as C++ source code. The objective is to replace present jpeg2000 modules (j2kenc.c, jpeg2000.c, jpeg2000dec.c, jpeg2000dwt.c) in libavcodec with my libs for encoding/decoding j2k files. The project is about implementing a new interface in ffmpeg module style for my compiled, running codec lib. The FFMEG project comes in source code as a VS2010 solution, compiling with Intel Compiler. You can download a one month trial from Intel for free and install it under VS2010. The source code compiles on Windows and Linux as ffmpeg does. For testing you should have a GPU Cuda graphic card installed.

...logo integration, change the them, colors, etc... 2. Show customer balance in the top of the dialer screen by using get data from our webservice (a2billing). 3. Automatically register with a2billing use IMEI as SIP User right after installation. 4. Automatically credit X amount to the new created account. 5. Option to refeel the account through Paypal and Google Wallet. 6. Use G.729 codec and fall back to codec that use less bandwidth. 7. Option for customer to find the rate based on the selected currency. 8. Option for customer to contact support via email and subject their IMEI. 9. Option to Keep or delete the call history. 10. Option to create favorite numbers. 11. Option to educate customer with dialing method accepted on the a2billing. 12. Random pop-up message to ask...

...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...

We currently have PJSIP compiled in xcode but we need help integrating to your application. Integration would include 1. Register and Unregister SIP user (Digest authentication) 2. Outgoing and Incoming calls(Push notification) 3. Call hold 4. Call Transfer 5. Calls should go though codec G729. You do not need to make GUI the only thing we need is to integrate this features to our current app. Specified OS: IOS 6 and up Development of graphic design and UI elements: I will provide all assets.

...managing bandwidth queue and QoS with proprietary congestion management technology. Works as a third party device/software with any Soft Switch. “Point-To-Multi Point” technology, requires native devices at both end. It works as like tunnel to secure packets and bypass all kind of firewall. Protocols & Codecs SBO Multipath V.3 uses lossless compression method and do not use codec for providing better compression, rather it compresses transport (RTP). So it works with all commonly used codecs i.e. G.729, G.723, GSM, AMR etc. It works with Session Initiation Protocol (SIP). Why Use SBO Multipath? For Reducing Operating Expenditures: Transmission cost is one of the big expenditure for any type of telecommunication business. For VoIP...

Project Description: 1. server...unlimited. 5. For server B installation may an ISO of USB image mounting of easy with some free tools which can be done using Puppy Linux OS. 6. For For configuration of both A and B there will some GUI (i.e. Web Interface) for adding gateways prefixes, 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 and forward them to gateways g729 codec. 2. Some kind of bandwidth compression is required for better performance on low bandwidth connection. i.e. TrafficSqeezer. 3. server A to Server B voice traffic will be encrypted so that voice port blocked bandwidth can be used for termination OR you can develop this using wrt or openwrt for tp l...

Project Description: 1. server...unlimited. 5. For server B installation may an ISO of USB image mounting of easy with some free tools which can be done using Puppy Linux OS. 6. For For configuration of both A and B there will some GUI (i.e. Web Interface) for adding gateways prefixes, 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 and forward them to gateways g729 codec. 2. Some kind of bandwidth compression is required for better performance on low bandwidth connection. i.e. TrafficSqeezer. 3. server A to Server B voice traffic will be encrypted so that voice port blocked bandwidth can be used for termination OR you can develop this using wrt or openwrt for tp l...

Hi, I understand that you were working on a project related to extracting Whatsapp messages from a Backup on a Symbian phone, specifically "msgstore.bak.1", where you able to complete that job? Are you able to do it? Thanks and Regards

I need to have xsplit or ffsplit like screen capture video broadcasting . - Screen region setting and broadcasting with audio - audio / video input select - RTMP protocol - AAC / MP4 codec - interface like xsplit or ffsplit ( will be provided ) - compatible from windows xp . - UI will be sent Applied need to prove if he has done similar project. Thanks

...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 ...

Project Description: Description Description 1) User friendly interface 2) Dialer supports SIP for signaling. 3) Dialer supports G729, G.711, G723.1, iLBC codec for sending audio & Video data. 4) Dialer should be able to run behind NAT or on private IP. 5) client can see the remaining balance in real time. 6) Dialer should be fully customizable. 7) Dialer with any kind of Soft switch which support SIP (SER/OpenSER/Asterisk) 8) Mobile SIP dialer for OS Mobile: Symbian (ALL), iPhone & iPod, Windows, Android OS, (communication via either GPRS, EDGE, 3G or Wi-Fi connection), Voice & Video 9) Full source code must be provided with proper comments on the code much more

We are looking for an experienced video engineer to quickly convert, compress with a great codec which reduces the size of our files and makes it perfect for playback on mobile device. Target: Size reduction but maintaining quality for mobile viewing. We want them as small in file size as possible COMPRESSION SETTINGS WILL BE PROVIDED Number of files: 22 Original File Type: MOV Required File Type: MP4 Required Devices: iPhone & iPad Budget: $35

Recently We have installed Freeswitch with ASTPP billing system. At this time we need a sip softphone that should support above listed smartphones. Also we need configure VPN ( PPTP or L2TP . etc) inside the app in order to avoid blocking from ISP. If any one who worked with the subjected matter with excellent skills please contact me. SKILL Required ---------------------- Ability to develop sip softphone for listed smartphone. Ability to configure PPTP/L2TP or other VPN Protocol in CentOS 6.5 server (It should support more then 500 user connection to VPN ) Thank you

Project Description: Description Description 1) User friendly interface 2) Dialer supports SIP for signaling. 3) Dialer supports G729, G.711, G723.1, iLBC codec for sending audio & Video data. 4) Dialer should be able to run behind NAT or on private IP. 5) client can see the remaining balance in real time. 6) Dialer should be fully customizable. 7) Dialer with any kind of Soft switch which support SIP (SER/OpenSER/Asterisk) 8) Mobile SIP dialer for OS Mobile: Symbian (ALL), iPhone & iPod, Windows, Android OS, (communication via either GPRS, EDGE, 3G or Wi-Fi connection), Voice & Video 9) Full source code must be provided with proper comments on the code much more

We want to run goip gsm gateway with asterisk server, asterisk will be installed on server and client both end & goip will be connected with client pc as trunk gateway mode, Need Asterisk with compression and encryption both for low bandwidth IAX2 codec will be used. if you know please contact us. we will give you good money

...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 ...

I need the attached audio files converted into this format - IMA ADPCM codec, 8.000 kHz, 4-bit, mono. - the finished product needs to sound good. If you don't know this format don't bid. Need this in the next hour! Budget is $30.

1. server A ( asterisk server, with static IP) receiving VoIP calls , with SIP and IAX protocol, u...unlimited. 5. For server B installation may an ISO of USB image mounting of easy with some free tools which can be done using Puppy Linux OS. 6. For For configuration of both A and B there will some GUI (i.e. Web Interface) for adding gateways prefixes, 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 and forward them to gateways g729 codec. 2. Some kind of bandwidth compression is required for better performance on low bandwidth connection. i.e. TrafficSqeezer. 3. server A to Server B voice traffic will be encrypted so that voice port blocked bandwidth can be use...

...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 ...

Hello, Recently we have configured Asterisk and Billing system one of our dedicated server. Now i need sip softphone to make calls from mobile. sip softphone that should work with multiple mobile platforms. such as IOS,ANDROID,NOKIA SYMBIAN, with G729 codec support. any person who technically skilled with this subjected work please contact me. Thank you