VOIP SWITCH CONFIGURATION AND INTEGRATION TO WEBSITE We are a Voip Telecommunication Company, offering mainly international calling solutions to our customers. Our main services include, Pinless Calling Cards, Wholesale carrier services, International Mobile Top-Up, Voip Residential and PBX solution. We have just migrated our customers to the voipswitch
Hello, Adapt a Twilio Application for Voip Innovations, a competing provider. [url removed, login to view] You can use the following Voip Innovation's documents: [url removed, login to view] [url removed, login to view] Please contact me with any questions. Thank you.
i have local server with pri card, i have DID configured through the PRI card , now i need to move the local DIDs to cloud asterisk server and configure there sip accounts to receive the calls and setup ivr there.I am willing to pay 50$ for this project ,please do not bid if you think this amount is less.
...Sales F). Extension 1006 having name Customer Care G). Extension 1007 having name Technical Support ii). Need to configure a sip trunk from Callcentric for incoming calls. iii). Need to configure a sip trunk from Localphone for incoming calls. iv). I have a grandstram ATA that is need to be configured with my free PBX server for taking
We are KareXpert Technologies a HealthCloud...KareXpert Technologies a HealthCloud Platform where a doctor can make video call with their patients. We have our sip client(android), webrtc(webapp) and kamailio as sip server. We need an expertise for debugging and to give solution to make sip call work. Problem is webrtcToSip call freezing at web side.
We need an experienced developer that has worked before with OZEKI SDK specially the VOIP part of it to fix some problem a sample of OZEKI sdk has. 1) When the button refresh on all devices (Mic, Camera, Speakers) is pressed as long as you have not selected other item that the item 0 (default item) everything works well. but if you select other than
I need few answers for topic.I am basically networ...from Basic stuff, since i am new for this technology.I have interview ,so need to prepare for those answer. 1- Manager baghto bolla 2- Lte cdma 3-Handover reselections 4-Sip call flow 5-Earfcn downlink 6- Lte master information block 7- Types of registration in cdma 8-Tracking area update
We are looking for VoIP expert to join our noc team for monitoring and analysis of our current routes, you must be have previous experience. You will be dealing with VoIP routes, identifiying bad numbers on the routes using the CDR and Human behaviour data and call filter, adjusting the dynamic call filter and Human behaviour server.
...and sms 5- modeC=with capcha, customer loged in to see created sip account and code to copy and paste to their web page 6- modeD=if mobile telephone number is all ready in database say sorry and if email is all ready in database say sorry 7- mysql database name wsip 8- to create sip account will use asterisk database 9- delete account user we must
I have a VoIP and web design project that requires the worker I hire to be skilled in: VoIP – This includes experience with VoIP phone system features and functionality, VoIP phone service provisioning, IP phones, VoIP Billing systems and all VoIP terminology. Web Design – Skilled in web design such as Dreamweaver. Knows PHP and ecommerc...
I need a SIP Client programmed in python. This software should be/have: -Both, the SRTP protocol and the ZRTP protocols are needed. -It should have video calling and graphical interface (GTK ...). -I also need to have the ability to configure 2 extra buttons (relays) for different purposes. -I also want to be able to add a logo inside the graphic
Prince Themed Stencil with Prince and his symbol
I require to develop VOIP (voice over internet application) on preferably android platform, but windows OS is acceptable. Project is not meant to be done from scratch and I expect freelancer to be adept at utilizing already available open source projects to drastically cut development costs and time only modifying already existing solutions to my needs
We need someone who can write an application to be used via the voice control of Alexa (Amazon) The voice control needs to listen to s certain request and execute it then and this can then send in a sip instruction to a telephone switch to be answered by an operator.
I have my own DIDs bought them from a site has only one option to forward the incoming calls and SMS to SIP URI. I need you to setup Asterisk and configure it to receive the incoming SMS and to get these SMS's through an API via my website so i can check them via my site.