We are in a need of a customization of the opensource VoIP / SIP client LinPhone. The development has to be done in two phases: Phase 1: Customization of mobile + tablet apps (iOS + Android) // Objective C = iOS + Java = Android --> ready in two weeks. Phase 2: Customization of PC apps (Windows, Mac OSX and Linux) // Qt/QML = Windows, Mac OSX and Linux
1. REGISTER + SUBSCRIBE to SIP server, with authentication 2. Accept INVITES 3. play inbound SIP packets and convert microphone to outbound sip That's it. If you've done it before, it's 1 hour work. Fixed payment $100. We need Actionscript in AIR.
PHP SIP client: 1. REGISTER + SUBSCRIBE to SIP server, WITH authentication 2. Accept INVITEs to send only, NO recv (so no traffic inbound) 3. Return WAV file to caller; depending on called number either file 1 or file 2 4. BYE to disconnect, that's it. - FIXED $40 for working code (do NOT ASK FOR MORE) - It's not even 30 minutes work if you've done
Requirement two-way audio/video SIP based communications form the mobile phones to Asterisks on ARM based targets and web browser targets
I am looking for a SNOM phone guru that can create a dial plan for me with the following...strip the '"9". A long distance call would be 91XXXXXXXXXX. We do not want to use time outs for anything but the international calling. We require the complete string for SIP account 1, 2, & 3, Delivery in a text file. PBX: 3CX Professional Phone: SNOM D765
i have voip device and i have small router it have 32mb ram os Openwrt i want to run 32calls so you need to use SIP2SIP register USA SIP server >>Bangladesh Openwrt Router>>>VOIP Device we cant use asterisk cz its need too much space i think best libre [login to view URL] or yate or besip [login to view URL]
...A: On Cloud And SIP Trunk has been created SSL is configured by "Let's Encrypt" Server B: On Local And SIP Trunk has been configured SSL is configured by "Self Signed" Result: Server A - SIP Trunk is appeared as not registered Server B - SIP Trunk is appeared as registered! I need to get all Registered, and the call is go through SIP...
...out forms and documents on our Web Site. The documents are made with LibreOffice and changed to PDF for use. The server used has 3 drives. One for the Nginx/Apache Web Server and the SMTP email Server, one for the MySQL Data Base Server and one for the module with LibreOffice. Invoice information comes from another Asterisk Server. To start, Teamviewer
**Must speak both english and spanish** We are developing an inhouse an Asterisk based PBX solution to suit our needs. This is an ongoing project and we will need developers who are experts in PHP, NodeJS, CSS, MySQL, Optimization, Security. Bonus points for mobile development. We also need project managers. Previous experience managing development
I have a FreePBX/Asterisk System working at Amazon. I can access it directly or via a VPN. Normal telephony works as expected. 1) Trying to get WebRTC phone (via the UCP) working 2) Trying to integrate external WebRTC Can you help debug?
We have an asterisk PBX integrated with Zoho CRM but it's delivering the channel ID instead of the dialed number to the the CRM extension. We need some one to fix the coding of the asterisk PBX to deliver the correct needed information
...Tabelle (ca. 20 Wörter) Total geht es um ca. 100 Texte in diesem Format zu verschiedenen Hunderassen. Preis also für 100 Texte angeben. Bitte starte dein Angebot mit einem Asterisk (*), um zu zeigen, dass du das Inserat gelesen hast....