Skill Pre-Requisites: Knowledge of WebRTC Gateway and not WebRTC web-client. Have developed SIP Servers applications (SBC or PBX) Good knowledge in WebRTC, Web Sockets, HTTP Knowledge of Asterisk server and Jitsi Media. Experience in product development and system engineering for WebRTC. Knowledge on JAVA will be good to have (Not mandatory)
We have our own sip / xmpp server and we have compiled the latest build of csipsimple and Linphone. We want a secured voice over zrtp. This is a peer to peer encryption but i dont get it work, not under csipsimple and not under Linphone. I need support to get this work. Linphone and csipsimple can be downloaded on playstore.
...org/display/FOP/Installing+FreePBX+14+on+CentOS+7#InstallingFreePBX14onCentOS7-FirewalldBasicConfiguration I want Observium to continue operating at http://<hostname/ and asterisk admin panel to run at http://<hostname>/admin I expect this to be a very straight forward task. To be completed via teamviewer. Please express the amount of time (minutes)
Hello, All! We have ASTERISK with realtime. We need develop RestAPI service that will extract peer information from ASTERISK 1.6 and send it to client. This service must have limits for some parameters like queries per second, answer with care. so it should affect on system performance. Write your way to do this in your bid, please. Or use chat
Call functions like mute, conference, hold, transfer, and call rec...integration from LDAP, Outlook, or CSV Low resource consumption Supports SIP, XMPP, and IAX accounts TLS, SRTP, and ZRTP encryption Voice Codecs: G.729 (paid), a-law, U-law, GSM, iLBC 20 and 30, Speex Narrow Voice Codecs: H264 (paid), VP8 Similar Projects: Zoiper SIP Webphone
company name: Aleo Design a simple logo using cursive letter and a stylish handwriting font. font can be picked up from goo...cursive letter and a stylish handwriting font. font can be picked up from google fonts : [url removed, login to view] logo can have a small stylish star ( asterisk ) on the up right side close to the "o"
i simply want to convert my android phone to behave as sip voip gsm gateway like GOIP, DINSTAR. android phone will register with my softswitch via 3g internet and when my softswitch will send call to android it will forward to gsm network. so it will work like DIALER(DIALS NUMBER)--> MY SOFTSWITCH --> ANDROID PHONE --> TO NUMBER DIALED BY DIALER
...website comprising most of the items in this diagram attached: 'schematic minus BTS [url removed, login to view]' . The diagram should comprise a "MultiDSLA System; DUT UE; IMS; VPP+ (ReferenceSIP Client); DSLA (Analogue test interface); a cloud". similar to '[url removed, login to view]'. diagram should be in a style similar to the 3 examples given be...
I need you to develop some software for me. I would like this software to be developed for Windows using .NET. Sample C# Project to :1. Connect to SIP trunk2. Make a call : play a file when pickup , hangup after playing , run event when hangup(by system or user) or no answer3. Receive call : play a file , run event when hang-up or no answer (by system
Hi, I'm trying to setup a Debian Jessie SIP server with Kamailio. Everything works but the TLS handshake doesn't complete. It stops at the client handshake, so the server doesn't send it's certificate. I would like someone who has experience setting up Kamailio with TLS and unix server administration. The deliverable would be to let me know
...build key features and testing. Works on Debian® 2.8 distribution, some of the key FOSS components that support the unified communication and management functionality are Asterisk®, FreePBX®, Chan_Dongle, and RaspAP Web GUI. Build a IP-PBX product that allows users to make VoIP calls to each other and give access to a long list of features.
It's simple I will rent a Linux Server (Centos u Ubuntu) and I need to install a dialler in it. The dialler is an Asterisk based one and all it's suppose to do is send call automatically to a telecom server by using a SIP account with 30 concurrent calls capabilities. The account has already been set in the Telecom Server
...I am new on ASTPP. I use the link([url removed, login to view]) to manual install the ASTPP. I configure the trunks, rates, sip, gateways, etc. All are configured. When I try to make a call, show me error on fs_cli and the call hangup: 2018-01-13 13:00:25.568487 [ERR] switch_odbc.c:368 STATE: IM002