Softphones srtp trabalhos

Build VOIP Application with SRTP encryption

I hired a technician to set up my Asterisk system and he is currently stealing from me . By using my voip account, IP, DID numbers to make international calls. I need to change all this info so this stops, not sure how he runs it but since he set my softphones up he can call through my phones.

...servers are provided at both sites for these reasons. Requirements The SoHo’s collection’ ops has two departments – Collection and Administration. There are no offsite operations. Your design must provide for: • Six employees in the Collection department (three per branch). • Two managers into Administration (one per branch). • Eight wired workstations equipped with MS Windows 8.1 x32 and eight softphones. • Installation and setup of up to two servers (one per branch), providing AD DS, DNS, and related services might be needed. • All networking devices must have IP v.4 addresses. • Use the private address 192.168.X.X/24 network for internal addressing throughout the company’s WAN and LAN networks. • Two public addre...

I have 2 teams at the end of a project and we are running into an issue with securing the web sockets for outbound calls through our CRM. Our CRM is Microsoft SQL and our dialer is Asterisk based. I need a consultant to help us find our issue and find a solution. PLEASE ONLY EXPERIENCED RESPONSES ONLY!!!!! Below I have listed the 1. Must be familiar with Asterisk Server 11 2....us find our issue and find a solution. PLEASE ONLY EXPERIENCED RESPONSES ONLY!!!!! Below I have listed the 1. Must be familiar with Asterisk Server 11 2. Must be familiar with WebSockets and Secure WebSockets 3. Must know how to use Secure WebSockets on SIPML5 and be able to receive inbound calls 4. Must be familiar with MeetMe conference Asterisk App 5. Must know how to detect RTP and SRTP...

hey guys i need helping to solve problem with my pbx i have freepbs 12 asterisk 13 and i want to h...guys i need helping to solve problem with my pbx i have freepbs 12 asterisk 13 and i want to have webrtc working . i compiled all stuff...all working . but the problem is once i call from phone to webrtc and answer it ... it hangup and i have error as : 2016-12-28 15:11:11] WARNING[28418][C-00000006]: res_rtp_asterisk.c:2141 dtls_srtp_setup: Could not set policies when setting up DTLS-SRTP on '0x7fd4bc006ea0' [2016-12-28 15:11:11] WARNING[28418][C-00000006]: res_rtp_asterisk.c:4120 ast_rtcp_read: RTCP Read error: Unspecified. Hanging up. if you are 100 % you can solve it ... bid else , don't bid plz , no try and error im willing to pay 100 $...

We have a PBX running on Asterisk and we have around 60 extensions configured. We use this in order to do remote voice alerts in softphones running on servers and some Axis IP speakers using SIP. We have many Axis speakers working and for this we had forwarded the necessary ports for them to work and we use the public IP address. All of the extensions are outside of the network our server is running. For the softphone we use Jitsi and we have preamplified speakers connected in the speaker plug of our sound card. So when from our office we dial an extension, we have Jitsi auto answer the call and then we speak and call talk and the sound is heard. When we use Jitsi, we use a VPN connexion (OpenVPN) so we use the VPN (local) ip address. But when we work with the Axis IP C3003-E ...

...MEDIA based in Register Username Protocols + scenarios : Ex: This for User 100 Forward to X-PBX UA (TLS + SRTP )--->LB-Register&MEDIA(TLS) Forward UDP +RTP ----> X-PBX and vice versa UA (TLS + SRTP )--->LB-Register&MEDIA(TLS) Forward TLS +SRTP ----> X-PBX and vice versa UA (TCP + SRTP )--->LB-Register&MEDIA(TLS) Forward UDP +RTP ----> X-PBX and vice versa UA (UDP + RTP )--->LB-Register&MEDIA(TLS) Forward UDP +RTP ----> X-PBX and vice versa Ex: This for User 200 Forward to Y-PBX UA (TLS + SRTP )--->LB-Register&MEDIA(TLS) Forward UDP +RTP ----> Y-PBX and vice versa UA (TLS + SRTP )--->LB-Register&MEDIA(TLS) Forward TLS +SRTP ----> Y-PBX and vice versa UA (TCP + ...

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 are looking for experienced developer/development team with prior experience in developing SIP audio/video clients. Features required: - TCP or UDP SIP transport - Codecs: G.722, G.711u, G729, Speex - Remote phonebook synchronization - Remote provisioning - Branding - VoiceMail and VoiceMail Message Waiting indication - DTMF modes - STUN support for NAT traversal - SRTP audio and TLS encryption -openvpn encryption

We've already got a running 3CX server, as well as a Twilio phone number. Everything is ready to go, the only thing left needing to be done is properly configuring Twilio credentials into 3CX to allow softphones to begin working with both incoming and outgoing calls.

I am looking for some one who has previosu experience in softphones. propsed job is to rectify errors in IOS and make to upload on apple store. connect apps with api and configured front end features with api.

We would like a get a minor configuration on asterisk. We have already created extensions and we are looking for two more features. When we call a conference number from our softphone other softphones should need to automatically get into the conference room. Another one, we can able to call locally with LAN but we are unable to call via Internet. We are in bit urgency. Can you please let us know how long will it take to complete. We are using linux platform. Thanks.

Hi Santhosh, We would like a get a minor configuration on asterisk. We have already created extensions and we are looking for two more features. When we call a conference number from our softphone other softphones should need to automatically get into the conference room. Another one, we can able to call locally with LAN but we are unable to call via Internet. We are in bit urgency. Can you please let us know how long will it take to complete. We are using linux platform. You can just share our screen and do the configurations or as per your compatibility. Thanks and Regards, Srinivas V.

We use freepbx 13 and we have users with softphones bria (counterpath). Currently we configure the bria phones manually. We want this to be automatically, i was reading brias instructions and it says we can configure a provisioning server. So this project is to create that server.

The current project would provide for the user audio news breaks. eneric framework: tow java programs, one as client on android and the other as server. THE INITIAL SRTP WILL BE CREATING THE FRAMEWORK

...account etc, with standard dialer features standard features, where customer can recharge their account using credit card and Paypal, refill history, rate search etc. more of less same features as mentioned in your website. * *Flexible Feature Set*- Add and remove features to either simplify the softphone or to add extra functionality. * *Branded*– Softphones branded to our name and logo. * *Contacts Integration*– integrated with the address book for simple dialling. * *Auto-provisioning*– Installed and provisioned via a QR code. * *Balance Display*– Display of available balance from A2Billing * *Flexible Design*– Flexible look and feel of the interface. * *Cod...

Need a mobile SoftPhone with source code. 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. 11. Wants to use in VoipSwitch like a pc2phone More detail will be provided at start of contract.

I would like to integrate Asterisk and Kamailio to provide: - Calling between srtp (websocket) and RTP (native softphone) - Load balancing and failed over between multiple asterisk servers via dispatcher - Enable IM and presence via Kamailio Requirement: - Users are registered to Kamailio. - INVITE requests are routed through the Asterisk server. - Asterisk RealTime user integration with Kamailio's subscriber table. - PUBLISH, SUBSCRIBE and MESSAGE requests are handled by Kamailio. What I have now: - 1 x Linux CentOS 6.8 64bit which running kamailio 4.4 - 2 x Linux CentOS 6.8 64bit which running asterisk-13.x, with realtime enabled, and a simple dialplan for testing handling call. - Web base SIP client

Необходима настройка sip транка Максифон от Белтелекома для Freepbx13 (FreePBX Distro, Asterisk 13). На данный момент без ограничений работают исходящие вызовы, входящие ограничены 5 минутами, по истечении которых пропадает звук в одну сторону. В логах появляется следующая ошибка: WARNING[4439][C-00000003] res_srtp.c: SRTP unprotect failed with: authentication failure 10 WARNING[4439][C-00000003] res_srtp.c: SRTP unprotect failed with: authentication failure 110

...main focus on the following markets: oil & gas, finance/banking, technology. Currently we need a FreePBX expert to setup FreePBX for us. We have the FreePBX system installed already on a VPS server and we want it to cover all our extensions around the world and we need to make all the settings, extensions, sips and trunks. - Set Fax, SMS, Video calls, conferences and instruction to set Softphones on Computer and Mobile. - Setup Trunk and Phones in the System for Outgoing Calls and a Dial plan for Calling to use and call in different countries. - Optimised the outcalls plan and use the best providers (sip providers, Google voice, …) - Set and manage DID numbers in multiple countries. - Advise on optimising the system or switch to better system. The right ...

I need an Android app. I already have a design for it, I just need it to be built. Implement tls signalling and use of srtp in a voip app with library pjsua

I need a project for school which implements a VoIP app on Android. It has to use TLS and SRTP and Asterisk as call center.

Hi nttranbao, I noticed your profile and would like to offer you my project. We can discuss any details over chat. I need help to connect my elastix pbx server to braodvoice sip provider and establish softphones which within usa and india. pleas let me know how much you charge.

*We have,* - setup a asteriskNow on AWS server. - hooked up a twillio account to asterisk - hooked up 2x cisco handsets (SPA504G) - hooked up 2x softphones *We can * - call between handset and softphones using extensions - make an external call via handsets *We can not* - receive an external call. (ie twillio inbound trunk) not receiving. We re-built the asterisk server twice and end up with the same outcome. I think it is likely a setting or configuration item we have missed. It is just a demo environment, so we can likely provide you with remote access.

Hey there! My company is a talent acquisition/recruiting firm with a very lean team (12 total). We have currently turned our entire business to cloud-based solutions, including our phone service. We use ShoreTel Cloud's VoIP softphones as our telephones, however, we don't have a network set up and the service is extremely poor/non-functional. As you can imagine, this hurts our business, and we'd like to get it taken care of as soon as possible. We would need the following: A router set up through our current Time Warner Cable modem The router should be set up with very specific firewalls (please see attached document) We *may* need an installation of ethernet throughout the office for optimized internet We can negotiate pricing, and we need this done as so...

Hi, We need iphone and android expert to customise linphone and rebrand it completely with logo by using Linphone opensource system including the following features below Audio & video calls Multiple calls ma...logo by using Linphone opensource system including the following features below Audio & video calls Multiple calls management Call transfer, pause and resume Audio conferencing (merge calls into a conference) Instant Messaging Pictures and files sharing Address Book Call History Display of advanced call statistics Echo Cancellation Quality of Service Secure communications: zRTP, TLS, SRTP Bluetooth headset support Languages: English, Arabic Account creation assistant Dedicated tablet user interface Push notification User registration via email...

Develop an android VoIP client. The app should permit to call somebody else who uses the app. Like Viber. I want to use webRTC, protocols like SIP and SRTP for security and a VoIP Platform.

...What will be sold: VoIP PBX with IVR (voice prompts) installation, fax2email and print2fax services, with remote installation in customer's office in Luxembourg and Alicante Spain; PBX hosting service at 20 EUR / month; VoIP phones will be purchased by customer (I have excel spreadsheet with product names, put to annex or within contract). Those named phones will be configured by us; VoIP softphones for IOS and android will be purchased by customer and we will send PDF guide how to configure them, together with authentication information. Maintenance: We will take care about security and PBX system upgrades for free during contract period; Support: We will do phone and email support with reaction within 4 business hours. Maximum three hours per month at specified...

...suppression Echo cancellation INSTANT MESSAGING XMPP SIP Presence One-to-one chats Multi-user chats File transfer SECURITY Encrypted password storage Password protection with a master password Encrypted Instant Messaging with Off-the-Record Messaging (OTRv4) Chat authentication with the Socialist Millionaire Protocol over OTR Call encryption with SRTP and ZRTP for XMPP and SIP Call encryption with SRTP and SDES for XMPP and SIP CODECS Audio: Opus, SILK, Speex, G.722, PCMU/PCMA (G.711), iLBC, GSM, G.729 Annex C (requires compilation and licenses) Video: H.264, H.263-1998 / H.263+, (VP8 coming soon…) MISCELLANEOUS On-line provisioning Provisioning server discovery via DHCP and mDNS (Bonjour) IPv6 fully supported by SIP ...

Hello i have asterisk server now i want to install SRTP you have to install SRTP on my server and teach me how to INSTAL SRTP why i m useing SRTP local isp blocking my ips thats why before i use openvpn if you have any Batter idea You can tell me I want to Install SRTP and connect IAX to IAX | SIP to SIP | sync to SYNC if you can please bid

I need full comprehensive and detailed explanation about packet traversal through NAT using STUN/TURN server, configuration of asterisk with STUN/ TURN server, machine requirements for asterisk (number of concurrent sessions, audio conference bridge), softphones that work with asterisk and STUN/TURN servers and so on. First you explain and if it's comprehensive detailed and full I pay.

We have set up Asterisk Server and the clients use linphone apps on Android mobile phones. We are able to do enctryped voice using TLS and SRTP. We also can do unencrypted chat. We would like you to have encrypted chat. You will need to show that it works, provide us the source codes and step by step guide to get it to work.

Hello freelancers, We are a Sydney and Melbourne based Photography Studio and are looking for a Person to help us with our Asterisk PBX server and everything connected to it. For PBX we run a Virtual Box with Linux Ubuntu and Asterisk as Telephony System. Host OS for this is Windows 7. In our call centre we have currently 12-15 Lines running, all using JITSI Softphones and Jabra USB Headsets to make outgoing calls. Outgoing numbers are set to mobile numbers, so clients call back each booker on their individual mobile phones. This all is based in our Sydney office. We are looking for somebody to help us with this system, to fix some problems we have, maintain it, improve stability and extend eventually. Our main Issues are that the System is not running stable, we have tro...

We are using Asterisk server and use TLS and sRTP to secure the communications. We were using self signed certificates. Now we have purchased a signed certificate from a recognised CA. We need to generate correct and and files.

We need to configure asterisk server such that our soft phones can use TLS and sRTP to communicate. The soft phones can be on computers or hand phones, and can have public IP or are behind routers.

...voIP provider will be placed with default configuration to avoid configuration requirements from users d) First SIP REGISTER message MUST contents some credentials as encoded IMEI and user phone number to provide back from our registration servers the first credentials e) Protocol must be SIPv2. TLS must be supported if available on our connection servers to encrypt registering communications. SRTP is not full required f) APP must be runs on Android, Symbian and IOS g) Codec g729 must be supported h) Preferable programming in C ( avoid Java as much as possible ) as will help you to implement other voice codecs. i) SIP Provider logo will be provide to show at startup the application j) App must be configurable to use WiFi and/or GPRS/3G/4G networks k) Dial pattern may ...

GOAL: want a secure inter connection encrypted on mobile phone TODO: 1. Install Asterisk PBX with TLS , SRTP, ZRTP support on ubuntu or centos server 2. create config and test on 2 phones using open source dialer (encryption support) Thanks

We are looking to develop a softphone like Zoiper, () or Zangi () some of the features are listed below. Please apply if you have understanding of the project. Access to the native contact list; - F...and registration status; - Blocked Numbers List. • Codecs: - Included: G.711 (uLaw & ALaw), iLBC; - Purchasable: G.722 (Wideband), iSAC (HD), G.729 Annex-B; - Support for UDP and TCP. • Other: - Call Waiting / 2nd Call / Call Toggle / Call Transfer / Conference; - Switch between Multiple SIP accounts (one registration at a time); - Secured SIP Transport (TLS); - Secured Encrypted Media (SRTP) with SDES; - Languages: English, French, German, Spanish, Portuguese, Arabic, Japanese, Chinese (Simplified), Russian, Turkish

We would like you debug and fix our sip trunking setup to work with our providers so that we may use Elastix MT 3.0 to the fullest with multiple organizations and users per organization...our providers so that we may use Elastix MT 3.0 to the fullest with multiple organizations and users per organization. We are currently able to have our handsets receive registration, and our trunks are showing registration from our sip provider, however we are not able to make or receive phone calls through the end points (Polycom) as well as we are not able to make or receive calls through softphones (Zoiper) We will provide shell access to the server. We will require a screen recording of all changes that are made by our freelancer. We would like to be able to access it through YouTube p...

We are looking for a Voip Pro that can setup a Freeswitch cluster using FusionPBX. Requirements: 1. Quick delivery - before Christmas. 2. SIP TLS + SRTP. 3. Centrally managed DB for both freeswitches 4. Presence enabled. 5. Documentation & knowledge transfer Attached file. (Please see SETUP A)

I need to setup an Asterisk voip server, with freepbx for webadmin, TLS/SRTP encryption for extensions, video calls, codec g729, I need recordings for some extensions, CDR to check call logs and recordings, a SIP trunk for international calls using + format instead 00. Also a Openfire server for XMPP chat with an easy app or plugin to audit chat of the users.

write ios/android/windows/linux/osx voip library will be used by our app for VOIP/VIDEO communication on supported platforms API exposed as easy socket interface (websockets for js) Traffic needs to be obfuscated strongly so it can work in countries like UAE (tunnel) app needs to go our our central servers first (tunneling) and then from there SRTP over SIP infrastructure needs to be ultra secure, need to think about how to exchange the keys best & how to encrypt lib needs to work with e.g. 10 tunnel end points & automatically chose the best tunnel SPECIALISTS ONLY PLEASE you need to be able to show working projects done in this domain

I need to setup an Asterisk voip server, with freepbx for webadmin, TLS/SRTP encryption for extensions, video calls, codec g729, I need recordings for some extensions, CDR to check call logs and recordings, a SIP trunk for international calls using + format instead 00. Also a Openfire server for XMPP chat with an easy app or plugin to audit chat of the users.

I have some blocking issues with voip softphones, need to change the sip port so it will be able to bypass ISP blocks, Centos6 with asterisk Box if you have experience in this please bid otherwise not. contact for information

Install, Deploy, and configure a Freeswitch/PbxFusion Voip server on a OpenVZ host. 1 ip with fallover, 3 virtual extensions with softphones, custom CID.

Hi i am looking for someone to setup a freepbx and A2billing system. The setup needs to include Secure Calling (SRTP) and TLS. Please note the secure calling is important. if you cannot set that up please do not bid on the work.

We need iphone and android expert to customise linphone and rebrand it completely with logo including the following features below Audio & video calls Multiple calls...linphone and rebrand it completely with logo including the following features below Audio & video calls Multiple calls management Call transfer, pause and resume Audio conferencing (merge calls into a conference) Instant Messaging Pictures and files sharing Address Book Call History Display of advanced call statistics Echo Cancellation Quality of Service Secure communications: zRTP, TLS, SRTP Bluetooth headset support Languages: English, French, Japanese, Arabic Account creation assistant Dedicated tablet user interface Push notification User registration via email and mobile number ...

... 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...

...version of AsteriskNOW to Deploy I will use the ISO image to deploy to server. Using my PC with TeamViewer, you have access to Asterisk server. Configure AsteriskNOW with FreePBX to: Two (2) PSTN Lines (FXO) Two (2) Extensions (FSX) We would like that in case of Internet or Power fault, the PSTN lines route to FSX extensions Six (6) Extensions using Softphones on Windows and Android One (1) Extension using Ealink SIP-T21P Two (2) SIP Trunk services (i apreciate if you can recommend one) Configure outgoing routes to use the better phone costs on SIP Trunks. During the Work Hour, incoming phone calls on PSTN must play a Welcome Message and ring in some extensions at same time. In case of none of these e...

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.