Softphones srtpεργασίες

Hello, I need customize or create a SIP Softphone Custom Branding. Follow the examples softphones you can use to customize: I need only a function customize (static IP for example) Logo customize Template (colors, skins, etc) - Optionally We will analyze offers with only Windows App and with Mobile app (android and IOS.

... Home: 1. About the product 2. FAQ 3. Partner program: (just an invitation to distributors to participate in the program telling them is a good product and very profitable) Technology: 1. How it works? 2. Use cases 3. Why privacy? An introduction about the product: Communication between two or more Smartphones using our App is done confidentially and securely through AES and SRTP. AES, or Advanced Encryption Standard, is a symmetric-key cypher that will replace the commonly used Data Encryption Standard (DES). It is recognized by the U.S government and military institutions as the standard for protecting classified information: “The design and strength of all key lengths of the AES algorithm (i.e., 128, 192 and 256) are sufficient to protect classified i...

... Home: 1. About the product 2. FAQ 3. Partner program: (just an invitation to distributors to participate in the program telling them is a good product and very profitable) Technology: 1. How it works? 2. Use cases 3. Why privacy? An introduction about the product: Communication between two or more Smartphones using our App is done confidentially and securely through AES and SRTP. AES, or Advanced Encryption Standard, is a symmetric-key cypher that will replace the commonly used Data Encryption Standard (DES). It is recognized by the U.S government and military institutions as the standard for protecting classified information: “The design and strength of all key lengths of the AES algorithm (i.e., 128, 192 and 256) are sufficient to protect classified i...

Hi, we have a small ip-bx server working here, its needs to be setup with some trunks, and set up some roles for users and softphones, basically weekly -3-4 hours needed. Pls only apply IP/VOIP Professionals

Details on chat, but configure perfectly for use WAN connection and softphones, trunk lines, etc.

I am unable to receive call using my bria/ xlite account using 1800 toll free. I need help on this

I need a softphone setup today on asterisk and also learn how to set them up. I additionally need my pbx setup for greetings and queues. Most of all I need this happening like right now so any help would be appreciated. Please place your bid then we can go from there,

Hello, I need customize a opensource SIP Softphone. Follow the examples softphones you can use to customize: I need only a function customize (static IP for example) Logo customize Template (colors, skins, etc) - Optionally Mobile Softphone - Optionally

I have three open source Android applications that do the following: 1. Jabber instant messaging based on XMPP protocol (with secure OTR encryption support) 2. SIP VoIP calling based on SIP (with SRTP/ZRTP for call media encryption support) 3. Email (with OpenPGP based email encryption support) I would like to change for each of the apps above: 1) the look and feel (aesthetic and visual elements including UI/UX) and make it conform to my company's branding and design philosophy. 2) improve performance and debug any issues/incompatibilities in the open source versions 3) optimize for use on different OEM hardware (e.g. Nexus, Samsung, HTC), different physical screen sizes and resolutions, and on both Tablets and phones In next phase I would also like to take th...

I currently have a PBXww account set up and softphones (Bria, Zoiper etc) working on the system. I have purchased a CISCO unified wireless hand set 7921G which I would like to add to the configuration. I have a TP-Link wireless router in my home. I need the phone configuring for SIP (software update or what ever is required there) and connecting to my PBXww network.

...website is: Home: 1. About the product 2. FAQ 3. Partner program: (just an invitation to distributors to participate in the program telling them is a good product and very profitable) Technology: 1. How it works? 2. Use cases 3. Why privacy? An introduction about the product: Communication between two or more Smartphones is done confidentially and securely through AES and SRTP. AES, or Advanced Encryption Standard, is a symmetric-key cypher that will replace the commonly used Data Encryption Standard (DES). It is recognized by the U.S government and military institutions as the standard for protecting classified information: “The design and strength of all key lengths of the AES algorithm (i.e., 128, 192 and 256) are sufficient to protect classified i...

We need to have a very simple FreePBX system set up for us. Here is as "exoti...from 8am - 5pm ET M - F 6. When we are not open any calls that come in go directly to voicemail (but, since we have three different company names under which we operate there are three different after hours messages depending on which DID/800# is dialed) 7. We have 6 or 7 Polycom IP670 phones that need to be programmed to work with the FreePBX system 8. We want to also have Zoper or Bria4 softphones for a few of the extensions 9. 3 of our DID numbers are presently with (but we aren't "married" to them and the numbers aren't public so if you have another preferred SIP, fine) We are hoping to have this finished by the end of next week at the latest. Thank you for cons...

Requirement is for a multisite PBX (Sydney, Singapore, Seattle, Ireland) with domestic trunks to offices in Australia, Philippines/Indonesia, Canada/USA and UK/Europe. Approximately 100x hardphones (Cisco SPA 5xx) plus 50x softphones. This is not a simple/small setup but it is not massive either. Some complexities with latencies and trunk codecs are expected. You will provide:- - advice on which Asterisk frontend, Elastix, FreeSwitch etc - setup of the EC2 instances and the configuration of the Asterisk front end - assist with sourcing appropriate SIP trunk providers in each domestic location - setup of calling plans, least cost path routing, inter PBX call routing etc - setup of call queues, ring groups and other calling features as required You will have:- -...

Research on Voip Plans & Pricing in India & Design a Pricing Plan for our New VOIP Services We are looking for some one to conduct research and design pricing plan for the ...supports prepaid, postpaid or pre-authorized payment accounts. Customers can monitor their call history and account balance information online. Reduce telecommunication costs Unlimited long distance International calling Mobile VoIP People are increasingly using VoIP on their mobile devices. To make VoIP calls from such devices people use software applications called ‘mobile dialers’ or ‘softphones.’ These apps run on devices such as smartphones and tablets and use high speed wireless networks to send and receive data. As these technologies become increasingly more acce...

1. Brand linphone all versions (or any other opensource softphones), allow client registration/login and balance display. 2. Create simple website where users can register, and download the branded voip client. 3. Integrate the website to an Elastix MT server with A2billing

Hi Freelancers, looking for softphones software expert (zoiper or eyebeam) with experience in PBX systems so he can Provide me a Toll Free number with SIP profile including a plan of unlimited minutes and unlimited Extensions for a monthly fee and its configuration and network setup. I also have elastix open source Call Center Software(execution file) if you need it for PBX/extension integration center is made up of 7-8 computers in total running on xp or windows7. Lowest bidders will be considered only. Happy Bidding

Hello Rioplp. We need your help to add the SIP Trunket to our FreePBX and then to make active the "Call on Click" to our vTiger CRM server. We have an account of VOIP to call just Italian telephone numbers. Where may you be able to work for us? Both the servers are active. On the FreePBX we have created users and they may call each other via XLite softphones. Now we need to do outgoing calls. Thank You! Dritan. drbakaj@

Hello Rioplp. We need your help to add the SIP Trunket to our FreePBX and then to make active the "Call on Click" to our vTiger CRM server. We have an account of VOIP to call just Italian telephone numbers. Where may you be able to work for us? Both the servers are active. On the FreePBX we have created users and they may call each other via XLite softphones. Now we need to do outgoing calls. Thank You! Dritan. drbakaj@

Develop a Cross Platform VOIP client in C code to use SRTP with modification which allows both the recipient IP address and the AES encryption Key to be passed in via the host application. The developer must compile for iOS, android, windowsphone, windows and OSx and also create an EDK library for each of these platforms for use with madewithmarmalade development software. Initial scope is for voice but pricing is invited for video calling as well as streaming

I have a freeswitch softswitch that receives calls from softphones and sends the calls to a couple if Cisco gateways. The problem is that th INVTE sent to the Ciscos contains non-standard headers. I need the Freeswitch to use only standard SIP headers so the Ciscos can yerminate the call properly.

softphone dialers with source code example unification voice, video, chat, twin dilaers

Estamos buscando un profesional que configure una central telefonica, la cual debe correr desde una maquina virtual e...profesional que configure una central telefonica, la cual debe correr desde una maquina virtual en un iMac con Yosemite. La configuración de la central se tiene que hacer remotamente sobre la maquina virtual. El profesional debe poder brindar soporte del proyecto en un futuro. El hardware con el que se cuenta es: GXP1610 Small-Medium Business IP Phone (4) GXW400x Series IP Analog Gateway (1) Softphones en iOS y android. En la actualidad contamos con una solución que instalamos nosotros mismos con Asterisk to go, lo cual es muy sencillo y solo viendo los tutoriales se logra configurar, estamos buscando son funciones especificas relacionadas...

List Linphone App: Customize GUI -Rename Linphone to IronCall -Change Logo -Change Task logo -Change app background -Change Keypad to normal black one -Change About -Remove in contact option *all* only Sip contact available Customized setup assistant with 1 option: -i Already have a Ironcall account Standard settings: -ZRTP/SRTP Enabled (ALWAYS) -TLS/SSL Enabled (ALWAYS)

Hi, we have a new freeswitch server that is working fine with some SIP clients and one trunk working succesfully. We can register an android phone and softphones, but for any reason, we can't register none of the both Linksys PAP2 devices that we have. This is not the famous G729 issue, also tested using just PCMU and PCMA (ulaw /alaw codecs) but nothing. Also, those PAP2 devices work fine with other external SIP accounts of other providers. The log in freeswitch say this: 2015-03-23 18:28:03.842108 [WARNING] sofia_reg.c:1742 SIP auth challenge (REGISTER) on sofia profile 'default' for [1000@] from ip And nothing else, the PAP2 show this in the main info page: "Can't connect to login server" Also, this PAP2 device can connect to oth...

Hi, we have a new freeswitch server that is working fine with some SIP clients and one trunk working succesfully. We can register an android phone and softphones, but for any reason, we can't register none of the both Linksys PAP2 devices that we have. This is not the famous G729 issue, also tested using just PCMU and PCMA (ulaw /alaw codecs) but nothing. Also, those PAP2 devices work fine with other external SIP accounts of other providers. The log in freeswitch say this: 2015-03-23 18:28:03.842108 [WARNING] sofia_reg.c:1742 SIP auth challenge (REGISTER) on sofia profile 'default' for [1000@] from ip And nothing else, the PAP2 show this in the main info page: "Can't connect to login server" Also, this PAP2 device can connect to oth...

Softphone for G729, I need suggestions on softphones working in UAE, OMAN, etc

List Linphone App: Customize GUI -Rename Linphone to IronCall -Change Logo -Change Task logo -Change app background -Change Keypad to normal black one -Change About -Remove in contact option *all* only Sip contact available Customized setup assistant with 2 options -i Already have a Ironcall account Standard settings: -ZRTP/SRTP Enabled (ALWAYS) -TLS/SSL Enabled (ALWAYS)

...-Change Keypad to normal black one -Change About -Remove in contact option *all* only Sip contact available Customized setup assistant with 2 options -Make new IronCall account (No email needed,) -i Already have a Ironcall account Standard settings: -ZRTP/SRTP Enabled (ALWAYS) -TLS/SSL Enabled (ALWAYS) ------------------------------------------------------------------------------ Freeswitch Server: Account setup with IronCall App -Registration from the Ironcall app without email Server configuration: -ZRTP/SRTP Enabled (ALWAYS) -TLS/SSL Enabled (ALWAYS) -Only Internal call (extension to extension) -Sending media files enabled -Video calls via ZRTP enabled -Install SSL Certificate (We supply one) -Forward secrecy (SSL) ----------...

...as following, but not limited to: -dial/receive calls -call recording -call transfer -call forwarding -call conference -send DTMF tones -detect DTMF tones -mic and speaker volume and mute control -multiple lines support -at least G711 U-Law, G711 A-Law, GSM 6.10 codec support -PLC -noise cancellation, silence detection, autogain and other standard softphone features found in most softphones on the market -and basically everything else a SIP siftphone needs to have You can take a look at or or similar products as a reference on what we need, except we need less features and we need it to work on Linux. The exact list of features will be discussed and agreed upon before development phase starts. We're opened to

...as following, but not limited to: -dial/receive calls -call recording -call transfer -call forwarding -call conference -send DTMF tones -detect DTMF tones -mic and speaker volume and mute control -multiple lines support -at least G711 U-Law, G711 A-Law, GSM 6.10 codec support -PLC -noise cancellation, silence detection, autogain and other standard softphone features found in most softphones on the market -and basically everything else a SIP siftphone needs to have You can take a look at or or similar products as a reference on what we need, except we need less features and we need it to work on Linux. The exact list of features will be discussed and agreed upon before development phase starts. We're opened to

i already have a free US number from Ipkall and I configured with a sip using softphones like X-lite, yate etc .. i am able to recieve calls but unable to hear the sound and couldnt check outgoing.. I want someone to debug the issue as voip is blocked in my country,... Hope someone efficient in this can help me with free incoming US numbered configured in sip

Hola necesito programador con experiencia en Protocolo SIP, el cual este familiarizado con Linphone SDK, PjSip o otras SDK el cual permitan el desarrollo de Dialer, o softphones para Android y Iphone.

Im recently starting my first Asterisk project. After install it, everything is already configured and working properly thru softphones (X-Lite), but Im completely stuck how to configure SIP phones as Polycom IP650 with my Asterisk. Looking for somebody to help me configure and learn how to configure those phones now on. I have a small IT company in Miami, starting installing Asterisk servers to some clients. Recien estoy empezando a instalar mi primer servidor Asterisk. Todo esta funcionando bien con los telefonos X-Lite, pero no conozco nada de LINUX para poder configurar los archivos, el TFTP Server y demas cosas que segun veo se necesitan para conectar telefonos duros SIP. Quisiera ayuda para configurar un ejemplo y aprender a como configurarlos en lo adelante. Tengo una pequ...

добрый вечер мне надо немного консультация по по воду SIP server, так как я новичок нечего не понимаю в этом Проблема в следующем, поставщик номеров пишет: We don't provide SIP trunks to customers for softphones. If however the customer has his own server he will need to allow the following IPs through the firewall. ( - ) ( - ) ======= тоест выходить чтобы принимать звонки на их номера мне надо свой SIP server? Вы можете советовать как можно решить эту проблему?

My setup: Vtiger 5.4, Asterisk, 4 x Cisco 502g, 1 x Cisco 508, softphones. CTI not working, need help on this issue.

Need integrated Softphone for vicidial/goauto dial. Right now we need a softphone to be configured to make it work with vicidial, I need something integrated is possible. You can you open source softphones if needed. If you think you can do it please mention the time frame & cost.

Dear all, I would like to have a SIP Dialer for Windows desktop with the following features: HD Audio and video call Audio conference Call transfer Call recording Languages : English Call History Presence Address Book Echo Cancellation Quality of Service Secure communications (TLS, SRTP, zRTP) All Major codecs: G729, G723, G711,SILK, G722, ...etc Ability to send SMS via SMPP Set Caller ID Call control (Call ID, Speaker, hold, conference, call forwarding, VoiceMail, ... etc) ** The dialer will be used by my clients, each client would have his own server (whether on local network or using a collocated server) So I need to have the option to pre-define the server's IP address for each client, I do not want it to be like 3 fields (extension, password, and IP) so...

I had installed and setup a fresh vicidial call center. But it gives some issues and does not work as expected. My softphones and sip trunk doesn't get registered and I can't dial calls. You must fix this issue and any other issues and make everything run smoothly. Server is in my office. Everything is already installed, need to find issues and solve them. I had installed it on my VPS successfully, but when I try to install it on my server at office it gives there issues. I have a SIP trunk directly linked to the server via ethernet. I'm also a developer but this is the first time I worked on asterisk, vicidial etc. I need a more experienced developer to take over and configure it correctly. It must be done within the day, please don't apply if you think...

I had installed and setup a fresh vicidial call center. But it gives some issues and does not work as expected. My softphones and sip trunk doesn't get registered and I can't dial calls. You must fix this issue and any other issues and make everything run smoothly. Server is in my office. Everything is already installed, need to find issues and solve them. I had installed it on my VPS successfully, but when I try to install it on my server at office it gives there issues. I have a SIP trunk directly linked to the server via ethernet. I'm also a developer but this is the first time I worked on asterisk, vicidial etc. I need a more experienced developer to take over and configure it correctly. It must be done within the day, please don't apply if you think...

I had installed and setup a fresh vicidial call center. But it gives some issues and does not work as expected. My softphones and sip trunk doesn't get registered and I can't dial calls. You must fix this issue and any other issues and make everything run smoothly. Server is in my office. Everything is already installed, need to find issues and solve them. I had installed it on my VPS successfully, but when I try to install it on my server at office it gives there issues. I have a SIP trunk directly linked to the server via ethernet. I'm also a developer but this is the first time I worked on asterisk, vicidial etc. I need a more experienced developer to take over and configure it correctly. It must be done within the day, please don't apply if you think...

Setup remote extensions on Elastix pbx system with billion router and zoipim softphones

We have our own telephony infrastructure and we now require a SIP softphone App to connect to our platform for our team to use. I need a sip softphone application for Android and IOS, with support for TLS,SRTP,ZRTP and XMMP(encrypted). Features required. * Full support for TLS,SRTP,ZRTP and XMMP(encrypted). * Full Source Code to make mods after project completion. * Support G729, G723, Codecs * Application should work all Networks 3G/Wi-Fi/2G or GPRS. * Phone full contact book sync * Real time Balance Display * Loudspeaker * Account to Account Balance transfer * Call Duration * Call Log * Redial/Call from Call logs * Simple to use User Friendly Interface. I need only expert and very experienced in SIP Phones to integrate with our existent FreeSwitch+A2Bill...

We have our own telephony infrastructure and we now require a SIP softphone App to connect to our platform for our team to use. I need a sip softphone application for Android and IOS, with support for TLS,SRTP,ZRTP and XMMP(encrypted). Features required. * Full support for TLS,SRTP,ZRTP and XMMP(encrypted). * Full Source Code to make mods after project completion. * Support G729, G723, Codecs * Application should work all Networks 3G/Wi-Fi/2G or GPRS. * Phone full contact book sync * Real time Balance Display * Loudspeaker * Account to Account Balance transfer * Call Duration * Call Log * Redial/Call from Call logs * Simple to use User Friendly Interface. I need only expert and very experienced in SIP Phones to integrate with our existent FreeSwitch+A2Bill...

We have our own telephony infrastructure and we now require a SIP softphone App to connect to our platform for our team to use. I need a sip softphone application for Android and IOS, with support for TLS,SRTP,ZRTP and XMMP(encrypted). Features required. * Full support for TLS,SRTP,ZRTP and XMMP(encrypted). * Full Source Code to make mods after project completion. * Support G729, G723, Codecs * Application should work all Networks 3G/Wi-Fi/2G or GPRS. * Phone full contact book sync * Real time Balance Display * Loudspeaker * Account to Account Balance transfer * Call Duration * Call Log * Redial/Call from Call logs * Simple to use User Friendly Interface. I need only expert and very experienced in SIP Phones to integrate with our existent FreeSwitch+A2Bill...

...veces. En otros momentos la llamada se escucha perfectamente. La troncal esta configurada, las llamadas salen pero , A VECES, no se escuchan bien. Hay que colgar y volver a llamar y al hacerlo , se escucha bien y a veces no. Es decir, el problema es que se establecen las llamadas pero , a veces, no se escucha bien. Las llamadas se realizan desde republica dominicana conectandonos a traves de softphones zoiper a la centralita Voip Elastix en Barcelona (España). El problema tendria que estar en la centralita en España. Llamamos sin problema por voip desde la oficina en republica dominican (via skype o conectandonos a una empresa de telefonia ip directamente). Por tanto creo que esta relacionado con la configuracion del portech o algun problema con este equipò ...

Hi, I have FusionPBX installed on Ubuntu 14.04 Server. It works fine, and I want to add TLS & sRTP. This is beyond my scope of understanding. Can you help? Tasks: -Install commercial SSL certificate -Generate files for TLS for server & client -Install files for sRTP -Test using softphone -Provide advice for adding multiple profile TLS (3) Kind regards, Yakov Nissen

We require Android and IOS VOIP Dialer based on RCS features RCS is installed on server Dialer should thereby support all features as can be seen at Do NOT OVERBID , better stay away

I own a Synology nas on which I installed the asterisk package. It works correctly, but I have to add a sip voip service that requires SRTP and tls but are not currently installed. I am looking for someone who can install it via teamview. No compilers installed yet.

Hi, I have FreePBX12/Asterisk11 setup. I want to enable TLS & SRTP. I have a self-signed certificate installed (/etc/pki/tls), but that needs to be changed to a commercial certificate (I will purchase). The PBX is already setup with a fqdn (), port 443 is forced. I need: 1) Do you have experience setting up TLS/SRTP? 2)Can you complete this project for me (see below)? Needs: A)Install commercial SSL certificate (I will purchase & email to you B)Ensure asterisk/freepbx is configured for TLS & SRTP (TLS files placed in - need verify) C)Test using "Blink" softphone D)Demonstrate setting up TLS on softphone (Blink is OK) via Teamviewer (15 minutes or less)

I have a very simple setup of an Linksys SPA9000 and SPA400 with a couple of Cicso SPA5xx phones and a couple of softphones. I'm using a PSTN line for all incoming and outgoing calls but I want to take advantage of the features a IP telephony system provides. I've set it up to receive and make incoming, outgoing and internal calls but I need help with the caller ID and the voicemail system.