Softphones srtp jobs
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,...
...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 - ...
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.
Creation of a softfone SIP VOIP Iphone application, features needed: •Full SIP Compliance - enabling use of any SIP-compliant server •Single Account Support. •Pre-defined locked single VoIP Provider (locked to our server) •Multitasking capabilities ...Via RFC 2833, SIP INFO and Inband •VPN Support •Language: English •Network traversal o Application Managed, Server Managed or User Specified o Global IP Support o STUN & ICE • Media Efficiency and Quality o Noise Reduction o VAD (Voice Activation Detection) • QoS (Quality of Service) o ToS Marking o DSCP (Differentiated Services Code Point) • Security and Encryption o TLS and SRTP (Secure call signaling and audio encryption) • DNS SRV record lookups &bu...
Creation of a softfone SIP VOIP Iphone application, features needed: •Full SIP Compliance - enabling use of any SIP-compliant server •Single Account Support. •Pre-defined locked single VoIP Provider (locked to our server) •Multitasking capabilities ...Via RFC 2833, SIP INFO and Inband •VPN Support •Language: English •Network traversal o Application Managed, Server Managed or User Specified o Global IP Support o STUN & ICE • Media Efficiency and Quality o Noise Reduction o VAD (Voice Activation Detection) • QoS (Quality of Service) o ToS Marking o DSCP (Differentiated Services Code Point) • Security and Encryption o TLS and SRTP (Secure call signaling and audio encryption) • DNS SRV record lookups &bu...
Dear All Programmers, We are looking for Team of Programmers or Individual Freelancers to Develop our own branded Softphone based on following technologies Please, Bid or comment Serious developers/teams. 1. Framework 2. Supports G729 Codec 3. Supports VPN 4. Using 3G or Data Networks 5. Ex. - MobileVoip Application, SiPhone, iDoubs (Open Source) Thanks, Priorcom
HARDWARE: Yeastar IP PABX Model: MyPBX SOHO One PSTN Line into an FXO interface. Two Analog Telephones (LCD Display available for Caller ID) Extensions terminated into two FXS interfaces. There are no other Incoming Trunks or Extensions such as VOIP, IP Phones, Softphones, IAX, etc. In other words just the POTS only. A PC will be connected via LAN Port as & when need to retrieve CDR's etc. ie PC will not be permanently connected to PABX. The PABX will not be connected to the Internet. The MyPBX SOHO IP PABX has already been configured and the two extensions are working complete with an IVR facility to answer incoming calls with a welcome message and route the call to either of the two extensions by pressing one or two. CDR facility is also working. To initiate an o...
...a home phone. _______ "ONE SHEET" (Include graphic!) HEADLINE: AlphaPhone® Encryption by Data Clone Labs: Protected Business Voice Communications Near Headline, larger: • Designed from the ground-up to add layers of security to the confidentiality, integrity and availability to important telephone conversations • Supports a range of mobile phones, desk phones and computer-based “softphones” • Military-strong using 4096 bit RSA and 256 bit AES, and a layered security approach to countering man-in-the-middle eavesdropping • The only encrypted voice solution with standard system control protected by AlphaKey® mult-factor authentication using one-time passwords. The Challenge Business or nation-state competitors, ...
HARDWARE EQUIPMENT & CONFIGURATION: Yeastar IP PABX Model: MyPBX SOHO. One Incoming PSTN Line from Telecoms Exchange terminated into one FXO interface. Two Analog Telephones (LCD Display available for Caller ID) Extensions terminated into two FXS interfaces. There are no other Incoming Lines or Extensions such as VOIP, IP Phones, Softphones, etc. A PC will be connected via the LAN Port as and when need ie to retrieve CDR's or Call Recordings etc. The PABX will not be connected to the Internet ie the WAN port will not be used. The MyPBX SOHO IP PABX has already been configured and the two analog extensions are working satisfactorily complete with an IVR facility to answer incoming calls with a welcome message and route the call to either of the two extensions by pre...
HARDWARE EQUIPMENT & CONFIGURATION: Yeastar IP PABX Model: MyPBX SOHO. One Incoming PSTN Line from Telecoms Exchange terminated into one FXO interface. Two Analog Telephones (LCD Display available for Caller ID) Extensions terminated into two FXS interfaces. There are no other Incoming Lines or Extensions such as VOIP, IP Phones, Softphones, etc. A PC will be connected via the LAN Port as and when need ie to retrieve CDR's or Call Recordings etc. The PABX will not be connected to the Internet ie the WAN port will not be used. The MyPBX SOHO IP PABX has already been configured and the two analog extensions are working satisfactorily complete with an IVR facility to answer incoming calls with a welcome message and route the call to either of the two extensions by p...
...1.) Phone-Server (Asterisk, Voip) for crypted telephone phone calls - which will be for maximum 100 users (200 clients, which are 100 softphones and 100 iphones) from our company. Server must accept only apps and softphones which have a aceess code and password, we need to block these anytime, when anybody quits. Server must have a frontend, where anybody can make the changes (also a secretary!). 2.) IPhone App, where we can make crypted calls from Iphone over our Server, to other IPhone or Softphone. Iphone App must have a security code provides by us for login to our server. 3.) Softphone, that allow me to call over our Server to other Softphones or IPhones. Softphone must have a security code provides by us for login to our server. Iphone App sould be...
...with our other SIP applications , So we are looking for Skilled Developers who can Run with us For longer . Required features: 1. Based on SIP. 2. Should support Voice and SMS. 3. Codec G.723, G.729 with silence supresseion. 4. Balance inquiry, real time balance display. 5. DTMF support. 6. Traverse NAT/Firewall/Blocked network through Encryption (VPN or Tunneling or P2P...any) 7. SoftPhones to communicate with one encryption server for both media and signaling. 8. Encryption Server will decrypt the encrypted traffic and use SIP for signaling and RTP for Voice. 9. The whole system should work with standard softswitch. 10. Registration of softphone clients through the encryption server in the softswitch. More detail will be provided at start of contract Regards...
hi, I need help configuring astrisk freepbx. need to configure the following : 1. 2 extentions and 2 softphones 2. auto attendant 3. 1 sip account at the end of project i need to be able to : 1. make calls between extentions 2. make outbound calls 3. receive calls 4. to configure all of the above by myself.
So, i need someone to configure an elastix server to forward the calls that it receives via internet from softphones (like x-lite) throught a 4 way GSM gateway, based on the destination, of the call. I will need the conversations recorded. All the setting made to the elastix server and at the GSM gateway, i need to be documented.
Hello I have SIP server running under FQDN domain name (ie. ) Most of softphones and hardware ATA's can register with no problem when they use domain name. The problem is that some softphones and ATA's could only register using server IP address. Best example is very popular softphone X-Lite. When configured to use IP address it works perfectly, when domain name is entered it could not register to server. Server is running Yate 3.3 softswitch, but exactly the same problem was when I've used asterisk PBX. Server has also configured DNS SRV records but it does not help. Your tasks: - discover problem with configuration (remotely, I can not give access to this server, it's production enviroment) - provide instructions how to correct this p...
...You will be working directly with me, my customers and my programmer. I am looking for an individual to assist me in following up on email request for my various support areas, scheduling request, creating support tickets, pricing requests, and sending job submissions and create my appointment and design schedule. You will answer approximately 30 phone calls per week. You must be familiar with softphones and have a DSL connection. TIME: Working Days Range: Mon to Fri Working Hours Range: 9am - 6pm USA Eastern Time Zone IMPORTANT: You must always be available online and ready for our next project or task during your "Working Days & Hours Range". This is because customer will contact you any time via Telephone or email when we have a project or task for you ...
...you to maintain this, set things up, make changes etc. for me. In your response, please confirm that you have read through this whole list and confirm you are experienced in doing these things. Quotations not specifically mentioning that you acknowledge having read the whole list will be disqualified. Some of the features that we would like to set up are - About 10-15 extensions, some softphones, some desktop phones probably grandstream (I have not purchased these yet and would like your suggestions if you have any experience with multiple phone types types – they should be quick to provision), and a couple atas, probably pap2 or grandstream. A couple of the desktop phones will be cisco phones 7940 series, I know these are hard to get working normally but I did get...
We need an asterisk programm...programmer or a team to handle our on going tasks. We need someone who is RELIABLE and can turn around quotes and projects on the fly. We have experience outsourcing and know what we want and don't want. We are not interested in people that disappear and dont work EST time. We have projects and support requests that can be handled by you or your team ongoing. Experience needed: -zrtp and srtp install and config -sip/tls -asterisk troubleshooting and programming This first project is to setup Hylifax or similar using VoiP and t.38 (no analog). We need it to be multi-tenant as I think Hylifax already is. You can be a single person or team is better of course for availability. Show us what you can do and we look forward to working with ...
Hi All, We require a simple SIP client for Symbian OS. Following is a list of the basic and only things required: 1) SIP Server/Domain will be hardcoded. User only needs to enter Username/Password. This means the client will only register to one server. 2) SIP Client should support both transports UDP/TLS and both transports RTP/SRTP as very basic requirements. 3) Client should support all latest Symbian (Nokia) phones. Mostly and 3rd and 5th editions devices like Nokia 5800 Xpressmusic which will be used for testing your work. Please only quote if you have experience in Symbian.
