Softphones srtptrabajos

...looking for call centers that can produce 30-50 sales a month, payment is paid weekly by bank wire, paypal or western union we will pay as you go as you meet our agreed bench marks. We have call centers making a lot of money with this ongoing project and you could make a lot also. Scripts, Rebuttals, Product Info provided. We will also provide you with all live incoming leads and voip softphones. It is important that all agents must speak Fluent English. Thank You Maria - Director About Us: We are an Australian company who growing day by day. Most of our dealings are online and we have new products launching every quarter. This new venture has a goal is to reach 500 listings by the end of the year. The basic listings are $250, and you will earn $30-$50...

The project is creating a softphone that supports: direct IP-to-IP calls, phonebook, SIP,SRTP. works on windows and android.

...(most produce 150 + sales per per month) payment is paid weekly by bank wire, paypal or western union we will pay as you go as you meet our agreed bench marks. We have call centers making alot of money with this ongoing project and you should make alot also. Scripts, Rebuttals, Product Info provided. rnWE will also provide you with all live incoming leads and voip softphones. It is important that all agents must speak Fluent English. rnrnView our exciting product video to understand what the system does @ rnVisit our website @ rnrnThank You. rnrnGary Parker

The project is creating a softphone that supports: direct IP-to-IP calls, phonebook, SIP,SRTP. works on windows and android.

Install ejabberd on Amzon server for XMPP service on Bria softphones I hav etried openfire with no luck and have been told ejabbered is the best product

I would like to have srtp installed on my synology to work with Asterisk. Asterisk is running. It is a standard Synology Nas, no compilers installed yet

I would like to have srtp installed on my synology to work with Asterisk.

Einrichtung einer kleinen TK-Anlage auf Basis von Asterisk und eines Patton Smartnode 4630. Betriebssystem ist Debian, wird mit OpenVZ virtualisiert. 2 ISDN Anschlüsse für 4 gleichzeitige Gespräche und 29er Nummernblock sind vorhanden. Telefoniert wird mit Softphones oder DECT-Telefonen, hierfür muss eine Fritzbox mit dem Asterisk verbunden werden. Es sollen 2 digitale Anrufbeantworter und die Möglichkeit Faxe per Mail zu empfangen/senden eingerichtet werden.

Hi, we need a sip dialer for Symbian S60 series and other possibly . Major features required are as follows 1. Stun support 2. SIP with TLS ( SRTP is not needed just SIPS) 3. G729 codec Core need is for Symbian , but in addition to Symbian Andriod and IPhone version are also welcome.

Object Description: We need someone to set up voice broadcasting software on our server. We need someone to: Develop, configure software using Asterisk, Vicidial or other to send voice broadcasting. You will set up through remote access to our Host Server You will show us how to Manage and Initiate campaigns Voice...maintenance then run the raided or mirrored backup till remedy for the primary drive/drives is complete. Any demo would be very helpful if available. My uses for the system are heavily utilized for robo calling thousands of phone numbers an hour delivering our automated messages to people that answer the phone then press 1 to be transferred to a live agent. Calls get routed to our voip softphones as the telemarketers communicate with the potential cl...

Object Description: We need someone to set up voice broadcasting software on our server. We need someone to: Develop, configure software using Asterisk, Vicidial or other to send voice broadcasting. You will set up through remote access to our Host Server You will show us how to Manage and Initiate campaigns Voice...maintenance then run the raided or mirrored backup till remedy for the primary drive/drives is complete. Any demo would be very helpful if available. My uses for the system are heavily utilized for robo calling thousands of phone numbers an hour delivering our automated messages to people that answer the phone then press 1 to be transferred to a live agent. Calls get routed to our voip softphones as the telemarketers communicate with the potential cl...

Object Description: We need someone to set up voice broadcasting software on our server. We need someone to: Develop, configure software using Asterisk, Vicidial or other to send voice broadcasting. You will set up through remote access to our Host Server You will show us how to Manage and Initiate campaigns Voice...maintenance then run the raided or mirrored backup till remedy for the primary drive/drives is complete. Any demo would be very helpful if available. My uses for the system are heavily utilized for robo calling thousands of phone numbers an hour delivering our automated messages to people that answer the phone then press 1 to be transferred to a live agent. Calls get routed to our voip softphones as the telemarketers communicate with the potential cl...

...running Asterisk 1.8.10.1 and FreePBX v2.11.0.0rc1.8. The system is setup and ready for configurations. I have Cisco 7960, 7971 SIP based IP phones and a Cisco SPA303 that I would like have configured with phone extensions. I have the most of the configuration files setup for the Cisco 79xx phones but need someone to review them. If we can get away with using the Cisco SPA303 and get some VOIP softphones like 3CX to register and work that would be great. Basic features needed: Extensions, Paging/Intercom, Ring Groups, and Voicemail Parking, Music on Hold. IVR would be nice. Just need someone familiar with the GUI and where to set things up to help. Remote access can be granted to the server and I can update any DHCP TFTP configs if needed. I already have the TFTP server running ...

I need some one to help me to create a PBX server using Asterisk Now, I already have a server, the card () the network is ready, the softphones, etc

I am looking for someone with experience in both Asterisk and Microsoft Lync Server. We need to be able to make/receive external phone calls from Microsoft Lync through our Elastix PBX (Asterisk). Currently we use Softphones (XLite and Bria) to make phone calls through Asterisk and we run Microsoft Lync for internal IM/VC/Presence. We want to be able to remove the need for the Softphones and make and receive phone calls directly through Lync. We have two Elastix machines, one in each of our offices (Sydney - Australia AND Manila - Philippines). We are running one Lync server in Sydney - Australia. This project is to enable calling from Sydney based Microsoft Lync users through the Sydney based Elastix machine (unless more advanced options are presented). For security reas...

REMEMBER: THIS PROJECT IS NOT FOR ANY FRESHERS/JUNIOR DEVELOPER..SO PLEASE DON'T BID IF YOU DON'T HAVE ANY PREVIOUS EXPERIENCE IN VOIP/SOFTPHONE.. WHAT I need is already developed code with following features. Basic softphone features , 1-registration(UDP,TLS,TCP),(RTP,SRTP) (BASIC) 2-dial a number ,receive a call (BASIC) 3-the normal feature that a softphone has(speaker ,voice mail button, conference,dial from phonebook)…. This will discuss it later (optional) Basic Main Feature Our GSM normal dial plan is 03XXXXXX What I want to see is the scenario below : (BASIC) PBX send a call(invite 03045212@UA Registered IP address) to the registered UA which is your softphone in this case ----à Registered UA(Your Softphone)---àt...

OBJETIVO Evaluar la calidad de los codecs de audio y video integrados en softphones SIP para PCs. DESCRIPCIÓN Los softphones son aplicaciones de software desarrolladas para PC que permiten establecer una comunicación de audio y video. Algunos vienen con otras capacidades adicionales pero éstas no importan al objetivo propuesto. El funcionamiento básico de un softphone consiste en establecer una llamada con otro terminal utilizando señalización SIP a través de la red IP, y luego de negociar las capacidades de medios con SDP, tanto de audio como de video, comenzar a enviar y recibir flujo RTP de ambos tipos. QUÉ DEBE HACER EL INSTRUMENTO Para evaluar la calidad de los codecs de audio y video, se deber...

Actually I am looking for a VOIP application..This project is not for time serious bidders i need.. Those who have made already basic soft phone ,I will give priority to select them..don't bid if you don't have any previous experience of voip project.I will report as a spam if you place only copy paste bid.. Basic softphone features which i need is--- 1-registration(UDP,TLS,TCP),(RTP,SRTP) (BASIC) 2-dial a number ,receive a call (BASIC) 3-the normal feature that a softphone has(speaker ,voice mail button, conference,dial from phonebook)…. This will discuss it later (optional) Basic Main Feature Our GSM normal dial plan in Lebanon is 03XXXXXX (03045212 as its my GSM local number) What I want to see is the scenario below : (BASIC) PBX send a call(invite...

...Actually I am looking for a VOIP application..This project is not for time serious bidders i need.. Those who have made already basic soft phone ,I will give priority to select them..don't bid if you don't have any previous experience of voip project.I will report as a spam if you place only copy paste bid.. Basic softphone features which i need is--- 1-registration(UDP,TLS,TCP),(RTP,SRTP) (BASIC) 2-dial a number ,receive a call (BASIC) 3-the normal feature that a softphone has(speaker ,voice mail button, conference,dial from phonebook)…. This will discuss it later (optional) Basic Main Feature Our GSM normal dial plan in Lebanon is 03XXXXXX (03045212 as its my GSM local number) What I want to see is the scenario below : (BASIC) PB...

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We need a complete voip system. - will work on all major platforms. - windows softphone - cross-platformbrowser webphone - Android Softphone (branded) - IPhone Softphone - Symbian dialer presales Any third party voip endpoint - Voip tunnel clients for third party softphones and devices - Voip clients size select features - Sip protocol stack and related protocols - H323 protocol stack (gateway and keeper) - Class4 features (transit, routing, billing, LCR, BRS, load balancing, failovering) - Class5 features (PBX features, ACD, call forward, transfer, DID, etc) - Codec support: all (G.729, G.723, G.711, G.722, G726, GSM, speex, iLBC, wideband, ultra-wideband, H264, DTMF, etc) - Video, Presence, IM support (SIP SIMPLE) IVR Basic calling cards (requires IVR basic) - P2P, ...

...phone numbers registered all of these will come back to the same main menu and extensions. The menu should change depending on who is logged in at the start users will use a soft phone with a cisco usb adapter and headset but we would be looking to get physical phones in the near future. I need email alerts sent from some extensions should a voicemail get left and a not ready button for the softphones that will not change the menu but place the caller on hold until the agent is ready again. I need to be able to set hold music to each extension individual playlists for each. The agents should not be able to call outside the network only to other extensions (as set list like manager, It support but no other agents) they should be able to leave their manager a voicemail but they ...

I have been able to compile linphone from source under centos 6.3, however I have not been able to get SRTP working. This project is to provide instructions for getting linphone working on linux including any steps like building keys. The notes should allow linphone to call at lest one other SRTP client, such as bria or another srtp client.

We looking for developers to help us out on our voip retail project. We currently have basic site setup using django interconnected with our billing system. Basic functionality is setup , but we like to do some additional functionality based on user and administration interfa...our billing system. Basic functionality is setup , but we like to do some additional functionality based on user and administration interface. few examples: - customer notifications to site - customer chat - custom email to registered users - sms sending via site - sms phone number verification - refer a friend - promotion functionality - banner system - download counter for our softphones - work on payment api - security on login and logout , session control Let us know. Thank...

I am looking for someone with experience in both Asterisk and Microsoft Lync Server. We need to be able to make/receive external phone calls from Microsoft Lync through our Elastix PBX (Asterisk). Currently we use Softphones (XLite and Bria) to make phone calls through Asterisk and we run Microsoft Lync for internal IM/VC/Presence. We want to be able to remove the need for the Softphones and make and receive phone calls directly through Lync. We have two Elastix machines, one in each of our offices (Sydney - Australia AND Manila - Philippines). We are running one Lync server in Sydney - Australia. This project is to enable calling from Sydney based Microsoft Lync users through the Sydney based Elastix machine (unless more advanced options are presented). For secur...

I am looking for help with branded softphones for Windows, Android, Iphone and Blackberry. I am also looking for installation of A2billing and adding calling card service to my company. Please let me know if you can assist me. Thank You

...preferred signaling component for setup/teardown but we are open to suggesteions. Prefer an open source application(s) tailored to meet our needs. If the solution is to be custom built all rights and code will be released to buyer. The limitations: No changes can be made in the firewall(s). Secure connection CANNOT require an installed VPN interface. Soft VPN is acceptable (SSL, SSH, RTMP, SRTP, etc.) as long as nothing has to be installed on the client like a layer 3 IPSEC VPN or TUN interface. If there is a client dependency it must be very small. Video has to stream with no more than 2 seconds delay, must be usable and synchronous. Server is Ubuntu 10.04 LTS and can support any packages required (NETCAT, SOCAT etc.). How it will be used: Secure video conferen...

...preferred signaling component for setup/teardown but we are open to suggesteions. Prefer an open source application(s) tailored to meet our needs. If the solution is to be custom built all rights and code will be released to buyer. ## The limitations: No changes can be made in the firewall(s). Secure connection CANNOT require an installed VPN interface. Soft VPN is acceptable (SSL, SSH, RTMP, SRTP, etc.) as long as nothing has to be installed on the client like a layer 3 IPSEC VPN or TUN interface. If there is a client dependency it must be very small. Video has to stream with no more than 2 seconds delay, must be usable and synchronous. Server is Ubuntu 10.04 LTS and can support any packages required (NETCAT, SOCAT etc.). ## How it will be used: Secure video co...

We have a server configured with latest elastix that we would like to add a few snom phone lines to and also final configuration of our fxo card(installed and showing). All the parts are there we just require a bit of guidance to make it all work (softphones work fine for in/out calls). what we would like ... 1) update endpoint config to include 370 phone settings for snom 2) fxo card configuration (already physically installed) The sooner the better.

We need to develop the mobile application on Multiple platform (iphone / ipad / Android) for wifi / 3G calls similar to other SIP softphones (like freephoo, 3CX etc application) We need to use our SIP service. Some of the application features are as mention below: - Sign Up - Forgot Password - Create user (sign up form username, password, re-type password, email, country, our SIP Server bades on Asterisk / FreePBX / A2Billing 1.7

...appropriately. The situation is our web panel is simply a website with a database at this point. We need a asterisk developer that can create an inbuilt softphone into our web application so that we can use the credentials of our PBXTRA phone extensions to login to the panel and receive, transfer, hold and conference calls coming in. This application should be scalable to eventually work with softphones, deskphones, etc . On our web based panel we have buttons for answer, dial, transfer, hold, conference, park, mute, etc. We need these buttons to work in such away that we can control the call from our desktop alone with headsets. There is an admin interface where we enter new customers in and this is where we associate the company with their assigned DID etc. We control the...

I am looking for someone who can provide complete training on Freepbx, voip, setting dialplans , exte, configuration , setting softphones along with the A2Billing. Person who can train over skype voice chat and also can provide documentation after every class. Only the experience person must contact. Two to three classes a day , 1-2 hours every class. This will be ongoing until the entire course is completed

I am looking for someone who can provide complete training on Asterisk installation, setting dialplans , exte, configuration , setting softphones along with the A2Billing. Person who can train over skype voice chat and also can provide documentation after every class. Only the experience person must contact. Two to three classes a day , 2-3 hours every class. This will be ongoing until the entire course is completed

Dear, We want to develop a VOIP / SIP Dialer for Symbian Mobile Phones which can work in blocked areas. It will use SRTP / TLS / TCP protocol with SIPS. The dialer must be able to work in all type of environments blocked / open on all type of data networks GPRS 3G, Wifi etc. The Dialer will support and will offer to select or auto select codec G711, G729, G723.1. The dialer will be able to work on all type Soft Switches / proxy servers. The dialer will support maximum number of Nokia handsets Symbian^3, 3rd Edition, 5th Edition etc. Developer will also develop server side but it is optional. The Developer will also provide complete source code of the project. With thanks,

Im reloading a freepbx system that sits between the telco and a nortel pbx so all calls can be recorded. There are two PRIs. I need the dialplan to forward nortel extensions the the nortel system and sip extensions to softphones. I have a script that will convert the recordings to mp3s, move and sort them into folders by date preferably to an external usb drive. The converted recordings will have to be accessible and downloadable from the recordings interface of freepbx. The system is setup and running in this configuration now but barely. You should be able to copy the custom dialplan work we've already done if needed. I can have the system loaded and the sangoma A104 configured. Easy job for the right person.

need help registering extension ... Install VPN Asterisk istalled and register to call centric 2 register Ip phone 7940 3 Register softphones

need help registering extension ... Install VPN 3cx istalled and register to call centric 2 register Ip phone 7940 3 Register softphones

...doesn’t work 100% as expected. Tried to use configuration from the official OpenSIPS book but with no success. There are issues with NAT’ed BYE requests and with ACK’s, so some modification is needed. Current config file is: Design goals: 1. OpenSIPS version 1.72 notls as registrar (MySQL) 2. NAT support for all popular NAT types, as much compatible with various softphones/devices as possible. Support for both NAT’ed and not NAT’ed clients. Also support for devices behind NAT but with public IP addresses in SIP requests (changed on client side by using STUN, ICE etc) 3. All inbound/outbound calls after authentication should be passed to Sippy B2BUA and it will handle routing logic and establish connections with B-leg of the call 4. Sippy i...

I ve test Native dialer on Android and also use it , there is no G729 codecs, or low rate bit codecs except gsm , and as usual gsm is not so well know on voip not used so much that why I need to ask some profesional people can add a simple way on android softphones a native sip dialer which is smilar to the original one on ZTE BLADE I ve TEST it g729 on default and when is connected on wifi g711u ,

Set up a Asterisk server with SRTP in a Virtual PC, 100% crypted calls with SRTP, set up everything that it works in my office in switzerland including dyndns and so on, that we can do crypted phone calls. Set up Free PBX, that i can create everything (extentions) myself. Make a detailled description how to start it and how to make extentions and use it.

Dear, i need somebody which will set me up a Asterisk Server with SRTP. I need only VOIP Communication over a Softphone between internal stuff in the company with SRTP - nothing else. Must be 100% secure phone calls! Let me know your offers.

We need a softphone that can hang up immediately if the line is busy, answering machine, or wrong number, and redirect these negative call results to a webpage as a querystring. Modifications of present softphones are wellcome.

Initial setup of asterisk based pbx for microbusiness, and assistance setting up associated softphones to use the pbx. We are a micro-business, and currently we have a main phone number with a legacy phone service provider that call forwards to a Skype number, that rings on our desks. We have additional skype accounts, with skype in/out. We are looking to improve on our phone service, and save money by switching to an Asterisk based pbx configured to work with our existing skype account, and google voice. Once we are familiar with that setup, we plan to move our main number to a VOIP provider (add a trunk for it to Asterisk), and then finally to start to sell a similar solution to our customers. We are a small IT company, but just don't have the time to do this in-house,...

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...dial plan for up to 10 different extensions and attach TWO of those to X-lite softphones on your windows machines. 3. The machine will be also an authorative DNS server for your domain: (where xx is your group number) 4. Configure Postfix MTA is such a way that all the voicemail for the extensions are forwarded to the appropriate mailboxes. 5. Set up the appropriate IAX trunk in order to connect your IP-PBX with the Asterisk servers outside your domain. 6. Test the system with numerous incoming and outgoing VoIP calls within the LAN. 7. Configure voice mail for the softphones used in step 2. 8. Complete list of the equipment used. 9. Use Wireshark to capture VoIP traffic between the softphones and properly analyse and record the findings regarding the protoc...

...those to X-lite softphones on your windows machines. The machine will be also an authorative DNS server for your domain: Configure Postfix MTA is such a way that all the voicemail for the extensions are forwarded to the appropriate mailboxes. Set up the appropriate IAX trunk in order to connect your IP-PBX with the Asterisk servers outside your domain. Test the system with numerous incoming and outgoing VoIP calls within the LAN. 7. Configure voice mail for the softphones used in step 2. Student Name: Student Surname: Student no: Group No: 1 2 3 4 5 Complete list of the equipment used. 9. Use Wireshark to capture VoIP traffic between the softphones...

Dear Freelancers, we have an Asterisk server to be configured for media encryption. We have installed FreePBX on Virtual PC and completed configuration for SRTP. It works, but it has too bad voice quality even when both ends are on a good bandwidth. We are not sure, but think the directmedia mode (direct RTP) might be a solution. We tried directmedia mode and it doesn't work in encryption mode when we think SRTP + directmedia. Any solution can be accepted if it improves the voice quality and video quality in encryption mode. You must have much experience in Asterisk and FreePBX configuration and administration in Linux. Contact me, if you have a solution or know the problem.

General requirements: - Protocol: SIP/SIPS RFC 3261 compliance - Operation system: iOS (iPhone), Android - Network interface: UMTS (3G) , GPRS, EDGE, Wi-Fi - Audio and video codecs: G711, GSM, HD G722, speex - Media-delivery: RTP/SRTP - Branding (startscreen, background, logo) - Dialer must work with our voip Platform (SER-like solution) - Default language is English. There must be a possibility to add another languages (eg. using xml file) - When exiting, the application have to unregister cleanly (expires=0) - Video calls - Predefined domain - only an username and a password should be provided by the user - User friendly interface Basic phone functions: - Making and receiving SIP calls - Integrated contacts from the native address book from the mobile phone - ...

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I'm looking for experienced voip specialist that can configure asterisk server to be able to: Connect to Skype Connect SIP Trunk. Configure 1-2 queues with callback members - over Skype Connect. Configure members with softphones over nat. Configure CDR so agent number and call duration will be visible in mysql database. I have ec2 server on ubuntu with asterisk 10. Currently I'm stuck with configuration and need help.