Softphone iax jobs
I need someone to develope a SIP client softphone for blackberry with Requirments below. Requirements: Codec: g729, gsm and g711-ulaw Three way calling Call hold Second incoming and outgoing calls Recent calls (dialed call, received call, missed calls, at least 20 calls on each log and be able to make calls from the logs) Phone book, calling from phone-book DTMF Features that will make the software user friendly Software should be connected to the server at all time (3G/EDGE/WIFI/GPRS) I need exact coding you build the software with.
...HFC Card Cologne ISDN card- Requested functions are: - Let callers call Asterisk PBX and start the script to find the first free appointement date and time (from google calendar) and to be able to reserve it. (If they are new, we ask them to enter their phone number, their name, we compare the phone number and the callerID. If they are similar, we accept. If not the same we forward to a SIP/IAX extension) - Each reserved appointment is stored in google calendar account with all the details : CID + link to the recorded name + expected duration - each appointment is a 15 minute appointment but an option should let the caller select duration from 15 minutes to 1 hours if they know. If they do not know, we define to 15 min. all saved appointement are given an id and password se...
We are VOIP company in wholesale business, we currently starting our business in reta...Create customer/reseller portal + backend 2) Create payment integration with several payment operators 3) Customer sign up system 4) fraude protection system (sign,payments,etc...) 5) API/SOAP integration with our voip system 6) create voucher system (prepaid payments) 7) SMS integration for customer verification (callerid, account) 8) Tarif plan administration 9) Build of Custom Softphone application *(if possible) we have about 2 months to complete the pilot project... To launch full project next year ## Deliverables We preparing document from all features, but most details will be enclosed once project is accepted and if u are up to the task, and have knownless t...
...using FreePBX and Tribox. 1. We need the first system to run freepbx to handle our voip service configured this way: a) Each voip subscriber is located on an extension without specific did assigned to them, but rather we will have regional did numbers coming into the system where inbound callers can key in the extension they want to reach. b) Each subscriber can dial-out through their softphone or voip phone, but will have a 2 hour per day limit they can use the service for outbound calls outside the system. c) all toll calls will be blocked (i.e. 976, 411, 800, 888, etc) d) system needs to auto provision popular voip phones. 2. The second system running tribox will be used to issue did fax only service to subscribers with only IN BOUND FAX service , no outbou...
we are looking experts in the area of VOIP. We have spent 2 years developing a PC to PC softphone and now looking add video conferencing facility and would also like to fine tune our media handler for better audio quality. i am looking for people having extensive knowledge in SIP, RTP and Handling Audio/Video Codecs + extensive knowledge about UDP ports. Thanks
I would like someone to develope a sip client softphone for iphone using , with Intel IPP codecs( g729, gsm, g711-ulaw). Example sites:
need to set up this system using elastix, have used virtual elastix before but have not learned how use all features. i need a system set up with multi tenant abilities a...and give to my clients. i want to learn how to set up a adisa when calling from external to dial into a certain extesion and show that extensions caller id and also to dial into system externally and dial the cid i need to show on the termanating location. also i need to understand how to get my polycom330 sip phones to connect with my current system and how to config a multi extension softphone like x lite which i have currently. would like to understand the difference with pstn and sip trunks and using pri, also the cheapeast and best way to host my system. almost forgot need to setup skype trunk on...
Hello, I need a softphone using IAX protocol. Please go to this link : to get a opensource softphone. I need it customized and also unique look. balance display. Thanks.
Hi I am looking for a Voip expert in India preferrably Delhi who has expertise in Voip regulations require a CAT A ITSP liscence to sell service all over India, I DO NOT have an ITSP liscence and will require a tie up with an existing ITSP category A for colocation and bandwidth. I will also require the complete set up such as , billing software,private label softphone,server space ,hosted platform and flexibility to purchase minutes from the most competitive source I can who has the know how please contact and bid. Thanks
i need software of softphone like X-Lite. this must be written in **visual studio 2008 C#**. there are so many open sources in internet. most impotant thing is that must be correct and it must work efficiently (use SIP) . there is no need to use skin directory. ## Deliverables finally u mast sent solution file in C# 2008
I have asterisk (pbxinaflash) + A2Billing up and running. I need the following: 1) Fine the current system 2) Fine tune Calling card and softphone/ATA Calling 3) Setup for ATA package with free calls to USA/Canada (no IVR) and bill all other locations with IVR 4) Basic website with for customer access with rates, FAQ, contact form, Banner and basic shopping cart. 5) Add rate list to sites that can be easily updated 6) Fine tune PayPal payment 7) Add web click to call 8) Add call back and SMS call back 9) Integrate with DIDX and IPKALL for DID selling and did assignment to customers at signup 10) Account low balance notification by email/SMS
Integrate VtigerCRM and Vicidial by making softphone that would be loaded as part of Vtiger. The softphone would connect to the local user's soundcard for audio input and output and would display a popup inside vtiger when a call comes into asterisk, showing the caller's account information if it exists and if not, would prompt the user to create a new account/contact. The softphone would also manage outgoing calls for the user. every phone number would be a link that upon clicking, the softphone would dial.
I am looking for someone who can assist in the customization of Freeswitch on a Linux system, with a few softphone endpoints, a Linksys ATA, setup simple IVR menus, 3 mailboxes, and dialplans. Currently freeswitch (trunk version) is installed along with WikiPBX on a CentOS 5.2 box, with a couple internal endpoints registered. There are intermittent problems with audio, and inbound/outbound dialing ( trunking) needs proper configuration. Preference will be given to those with Python skills, and experience configuring various freeswitch modules (eg: dingaling, pocketsphinx, cepstral, etc.) for follow-on projects. Ideal person would be available after project completion for consultation or further modifications, at hourly rate. Please bid only if you have direct Freeswitch
...and asterisk by making a java based softphone that would be loaded as part of Vtiger. The softphone would connect to the local user's soundcard for audio input and output and would display a popup inside vtiger when a call comes into asterisk, showing the caller's account information if it exists and if not, would prompt the user to create a new account/contact. The softphone would also manage outgoing calls for the user. every phone number would be a link that upon clicking, the softphone would dial. the Vtiger 5.1.0 beta already has a lot of this functionality. We are mainly looking for someone to write a vtiger module that would provide a java-based softphone. I have seen a similar integration with crm and cisco's pbx. the s...
...and asterisk by making a java based softphone that would be loaded as part of Vtiger. The softphone would connect to the local user's soundcard for audio input and output and would display a popup inside vtiger when a call comes into asterisk, showing the caller's account information if it exists and if not, would prompt the user to create a new account/contact. The softphone would also manage outgoing calls for the user. every phone number would be a link that upon clicking, the softphone would dial. the Vtiger 5.1.0 beta already has a lot of this functionality. We are mainly looking for someone to write a vtiger module that would provide a java-based softphone. I have seen a similar integration with crm and cisco's pbx. the ...
...asterisk by making a java based softphone that would be loaded as part of Vtiger. The softphone would connect to the local user's soundcard for audio input and output and would display a popup inside vtiger when a call comes into asterisk, showing the caller's account information if it exists and if not, would prompt the user to create a new account/contact. The softphone would also manage outgoing calls for the user. every phone number would be a link that upon clicking, the softphone would dial. the Vtiger 5.1.0 beta already has a lot of this functionality. We are mainly looking for someone to write a vtiger module that would provide a java-based softphone. I have seen a similar integration with crm and cisco's pbx. the ...
We are interested in a SIP softphone for Windows 2000, XP and Vista. The softphone should be able to work with Asterisk. We need the following : - Simple design. - Easy wizard when installing to select the sound device and network settings. - Very important : built in VPN client to connect to a VPN network if selected. - Compatible with the SIP protocol (g723, g729, ulaw, gsm ...) In the future we will need porting it to mobile phone platforms (another project).
We are a voip service provider and want to build our own softphone which can interact with our asterisk based sip servers. The softphone has to be very similar with MizuPhone from with gsm and g.711 codec, call forward and call transfer. g.729 codec would be a plus. Please take a look at We would like as much features as possibile.
We require your assistance to set up from scratch an Asterisk System. ## Deliverables 1) An Asterisk System up an running 2) This system should be able to receive phone calls from land lines and forward them to a softphone 3) User using a softphone should be able to place a call using land line connected to Asterisk. 4) Greeting message 5) IVR simple options example , press 1 for Sales press 2 to leave a message. Press 3 for a FAX Press 4 forward call to mobile phone. 6) This is a SOHO office, so very few IVR opcions are needed ## Platform Asterisk
We need a step by step, how to document, to setup a basic campaign using Vicidialnow. We already have vicidialnow installed. What we want are the step by step instructions for: How to setup trunks How to connect to the system and make calls via softphone how to setup a basic campaign, with a working example we can follow. how to setup a basic incoming call The instructions should be simple, so that we can easily follow them and have the above working with no additional information needed. In the end, we should be able to take our base install of vicidial now, and your how to document and have a complete running and working campaign. Thank you
make a software like this but with microsft crm intrgration and phone call tranfering aslo all other softphone features <> ## Deliverables it also need to work on ms outlook 03 and 07 ? and the sip login to the phone system is not the same account as the outlook account.? the phone system is free switchs
...small radio station for callers to call in, be placed in the conference call. When they are in the conference they are muted and hear the show playing in the background. When they are unmuted their voice is heard in the conference and is also heard on the air. The best way I can think to accomplish this is to automatically join (call) a user to the conference. This user will be a X-lite softphone configured to Auto Answer and will be piping in the shows audio from the encoder (sound card mic in) and outputting the voices/callers who are unmuted to the sound mixer (sound card aux out). What we need in short: 1) A way to automatically dial an "Extension" and have that extension when answered automatically tied into the bridge "unmuted". 2) Brid...
HI, I NEED TO BUILD A MOBILE VOIP CLIENT THAT WORKS ON SYMBIAN OX 9.X , IT SHOULD SUPPORT SIP & IAX PROTOCOLS & SUPPORT CALLS OVER WIFI/3G/GPRS/HSPDA NETWORKS , IT SHOULD OFFER SIP PRESENCE FEATURE & IM CHAT OVER MSN, YAHOO & GTALK SERVICES. IT SHOULD SUPPORT SMS & SENDING MAIL FEATURES. IT COULD BE WRITTEN WITH ANY SUITABLE PROGRAMING LANGUAGE.
We need a step by step, how to document, to setup a basic campaign using Vicidialnow. We already have vicidialnow installed. What we want are the step by step instructions for: How to setup trunks How to connect to the system and make calls via softphone how to setup a basic campaign, with a working example we can follow. how to setup a basic incoming call The instructions should be simple, so that we can easily follow them and have the above working with no additional information needed. In the end, we should be able to take our base install of vicidial now, and your how to document and have a complete running and working campaign. Thank you
I have pbxinaflash with a2billing hosted. My location is Germany Who can configure it to me? The following things are needed: The currency must be converted into euros. I do PhoneSex. My customers should buy credits in order to be able to call me. I have a IAX account. The IAX account to be integrated into the PBX inside. If the customer is calling me the telephone number of the PBX are displayed. If the customer has paid, the PBX call forwarding to my landline number. Time following packages should the customer can buy: 1 hour 42 Euros 30 min 22 euros 15 min 12 euro 5 min 4.50 euros Quote 40 minutes 20 euros 50% I pay now for the job, 50% after completion. I would max 120 $ pay for the configuration.
The Rhino SIP-IAX2 Proxy is a software product that interfaces with Rhino's PCI and PCI Express cards and provides IP connectivity to the “last mile.?? The software will be av...PCI and PCI Express cards and provides IP connectivity to the “last mile.?? The software will be available for Linux, Windows, and Mac OS X and will allow a connection from the local host to the application as a driver and operating-system agnostic interface to the Rhino hardware. Communications will be over industry-standard protocols, specifically: • SIP (RFC 3261) • IAX2 (RFC draft-guy-iax-04) ## Deliverables The products that this software will support are all Rhino Equipment Corp. products, three drivers: r1t1, rxt1 and rcbfx, which are all currently Linux based imp...
I am looking for someone to help me install and configure a Vicidial predictive dialer, with all the features and configure SIP or IAX trunks so that Vicidial is working properly. i need the pbx system as well We will be using the dialer primarily for voice broadcasting in the US and will need my agents in the Philippines to use xlite to receive and place calls. We will also need the ability to handle inbound and predictive outbound calls. We need the install to be done remotely. For the server, I want this dialer to be installed on local server accunt. I can give you instructions on how to do this when needed. We plan on using the same consultant for maintenance and other project installs if the test goes well. 80% of the payment will be given once the job is compl...
...software should be able to check if the connection is open or block by some ISPs in case of blocking it should use different way to connect for example if it can connect directly do so other wise try to use L2TP VPN, in case of L2TP blocking using PPTP VPN, in case of blocking using Open VPN to go through. 13-? ? For protocol again it should be able to check if it can use SIP other wise using IAX....
...software should be able to check if the connection is open or block by some ISPs in case of blocking it should use different way to connect for example if it can connect directly do so other wise try to use L2TP VPN, in case of L2TP blocking using PPTP VPN, in case of blocking using Open VPN to go through. 13-? ? For protocol again it should be able to check if it can use SIP other wise using IAX....
Configuration of trunks and extensions. Configure DID to point to an extension Configure extension to use a particular trunk Setting Up Dial Plan for Europe Setting up Prefix for different trunks VOIP provider callwithus. All hardware IAX Most of the extensions behind NAT
Need a Windows Application that will act as a call monitor for asterisk. I need someone to program a piece of software that will tie into our asterisk boxes and: -Display all users availability -If they are on a call or free -Users can set custom messages if away from their phone -Transfer...boxes and: -Display all users availability -If they are on a call or free -Users can set custom messages if away from their phone -Transfer calls -Put calls on hold -Pop when a new user calls -etc. Should be attractive and very functional. This will install on a users computer that already has a hard phone. It can act as a softphone but the two devices will have the same extension and even though the call is picked up on the hardphone the softphone still should have ul...
We need a Flex/Flash Coder to take our ideas to a software. It's a phone client (softphone) that is developed by us that need to get a good looking representation on the desktop. For example with reflections, iPhone-look etc. It takes someone experienced in Flash with design capabilities and interested in artistic work. Like to see examples of your work. ## Deliverables * * *This broadcast message was sent to all bidders on Saturday Jan 31, 2009 8:59:16 AM: Hi, Thank you for your interest in our softphone project. Ref. Flex/flash-design for softphone. 1. We need a softphone design for a web site. Preferably in Flex. Enclosed with this document you will find 1. A flash component with icons for navigation. 2. Powerpoint sketches of what is requi...
We need an Asterisk based application for training our staff. Very briefly, this web application should be developed using Asterisk with MySQL and CGI Perl. You Should have prior experience in building application based on Asterisk There should be a provision to create calls using pre-recorded voice to a? telephone/softphone number (of the staff), and to record the reply to each question in the call. The manager should be able to see a report on each of the staff and play back the conversation to mark his? evaluation .
Actually, we are looking for: 1) User friendly interface 2) Mobile Dialer supports SIP for signaling. 3) Mobile Dialer supports G729, G723.1 codec for sending audio data. 4) Must support DTMF dialing during a call 5) Mobile Dialer can run behind NAT or on private IP. 6) Mobile Dialer runs on application layer so it supports GPRS, WiFi, 3G Network and Bluetooth for internet connectivity. 7) Ability lock the service provider's configuration (provision). 8) Check / notification of new voice-mail. 9) This application has to be compatible with Flash-UI(Samsung,LG),Symbian (S60) 9.1 and/or Windows Mobile 5 and 6.
Simple IP video phone system prototype required: windows CE/Mobile 6 camera PDA with IP video softphone and gsm connection dials in to SIP server extension (server having sip/gsm gateway) and starts the video call...the other end is web based softphone and shows the audio/ video on a webpage. You could use all the open source components in the world to do this prototype/demo. I need the prototype delivered on my server in 2 days. If its successful, you would be in for many projects to follow. So some of the inputs I have: 1 - windows CE/Mobile 6 and Microsoft IP video phone stack. 2 - server (Microsoft RTC or Asterisk etc.,) 3 - Web video softphone - ActiveX or flash 4 - Communication will have to be through gsm between pda and server and ...
We need a SIP VoIP Softphone with HTTP (Port 80) Function. We need the voice to be able to travel on port 80 TCP. We also need the softphone to register to our SIP Proxy Server from Port 80. Basically, we would need something like an http tunnel to register and send voice calls out to port 5060 on our sip proxy server. We need this because our ISP has blocked all UDP and TCP Ports except port 80. We are also behind firewall and web proxy of ISP and we dont have a public/static ip. Skype and Pfingo are working since they have this HTTP (Port 80) function. Please check them - and
I am looking for someone to help me install and configure a Vicidial predictive dialer, with all the features and configure SIP or IAX trunks so that Vicidial is working properly. We will be using the dialer primarily for voice broadcasting in the US and will need my agents in the Philippines to use xlite to receive and place calls. We will also need the ability to handle inbound and predictive outbound calls. We need the install to be done remotely. For the server, I want this dialer to be installed on my Amazon cloud server account. I can give you instructions on how to do this when needed. We plan on using the same consultant for maintenance and other project installs if the test goes well. 75% of the payment will be given once the job is completed and working. The ...
...axvoice as VoIP provider, this has already been configured in Asterisk 1.6. Using PBX in a flash 1.3 with Asterisk 1.6 with SIP over TCP support. Using Microsoft Speech Server 2007 for IVR. I have a simple outbound workflow application to call US phone numbers. I've configured an outbound route on the PBX to send all calls to the VoIP provider. I've verified that it works fine with a softphone calling external phone numbers. I've also setup a TCP SIP trunk from the PBX and MSS2007. I'm in need of some configuration work form someone who has experience with MSS2007 to implement a configuration on the PBX/MSS2007 and/or possibly change how the application works that will allow my workflow application to make outbound calls. I'm performing...
Hi - We need a SIP softphone built using Flash, so that the phone can be embedded in a web page and be used to make calls directly from that page. We'll also need the corresponding server side work done to bridge between the phone and SIP, and Flash's streaming audio format and G.711. Configurationwise, the Flash part will be passed a parameter when it is downloaded, which identifies the phone, and this parameter will be passed back to the server when a call is made. The server will use this parameter to look up the SIP server and account details. The deliverable is the phone and the server side, installed on a VPS which we'll give you access to. Google for Red5Phone for something which does most of what's needed.
The Project, We are looking for a design of a skin for a soft phone, that can be overlay on our C# application. You can find some examples of what other people have done for soft phones here All Images on the softphone must be a separate image, using photoshop and placed on layers. If you can suggest a better way of doing this that is fine. Please find below a link to a soft phone as and example. You will not be required to do any programming, only to take your design and put it on a C# application and ensure that each button generates and event. ## Deliverables 1) Complete and fully-functional working program(s) in executable form as well as complete source code of
We are looking to build a Flash based SIP softphone to use with our customers portal. The softphone must have : - No installation of modules - Work on Linux, Windows, MacOS - Softphone will be integrated in a PHP page - Softphone domain, port, login and pass will be configurated from javascript cookies - Softphone have just to RECEIVE calls so no keypad - Softphone have to "Auto Answer Mode" and "Manual Mode" through CheckBox - Only 1 line is necessary (2 lines will be appreciated) - Must support Alaw, Ulaw, G729 codecs. - Softphone must be skinned through CSS The softphone will be connected with asterisk server.
We are looking to build a Flash based SIP softphone to use with our customers portal. The softphone must have : - No installation of modules - Work on Linux, Windows, MacOS - Softphone will be integrated in a PHP page - Softphone domain, port, login and pass will be configurated from javascript cookies - Softphone have just to RECEIVE calls so no keypad - Softphone have to "Auto Answer Mode" and "Manual Mode" through CheckBox - Only 1 line is necessary (2 lines will be appreciated) - Must support Alaw, Ulaw, G729 codecs. - Softphone must be skinned through CSS The softphone will be connected with asterisk server.
