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.
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OpenSIPS/Sippy Carrier Routing - Stage 1 either an OpenSIPS or Sippy box will act as an outbound traffic router for multiple carriers. Calls will be sent from already running Asterisk boxes to the OpenSIPS/Sippy/SippyGO box (we will prepare base OS, OpenSIPs/SIPPY core installs) Media needs to be proxied with RTPProxy/etc through the OpenSIPS/Sippy/SippyGO
Asterisk - CTI (Click To Call) API This will work on a Asterisk 13, FreePBX 13, and Asternic Pro 2.1.x platform. We need to provide a REST API for several actions e.g. a) Click To Call - e.g. /[url removed, login to view] authid = preset hash value to authenticate the user against. Return in JSON - Asterisk
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...syncing for both server A and B. If server A goes down server B must take care of all the things. Like MX records DNS entry. 3. We have a newly installed Asterisk server. Need to configure the Asterisk server. Here are the following things you need to do. i). Configure all extensions. There are total 7 extensions need to create. Those are given below
We have all ready own [url removed, login to view] first all comments are will be clea...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 delete from [url removed, login to view] if success than we delete useraccount
I need an experience free lancer that can fix connectivity issues and Vicidial asterisk based system. If you don;t know Vicidial please don't even send your proposal. Long term project with some support needs on demand.
- Routes calls to Route1 or Route2 based on 2 simple database que...2 - If Query 2 is negative we route the call to Route1, if positive we route to Route2 - Telephone numbers are 12 digits long, no special characters - Should work with Asterisk 13.14 and MySQL 5.5 - Should handle 5-10 calls per second without issues and need for special hardware
I need someone who lives in the United States to create a voicemail for a business and answer 3 phone calls over the next 2 days. The phone calls shouldn’t be more than 2 minutes long. Bids from a call center will not be considered. Bids from people outside the United States will not be accepted.
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.
I have the solution and all the call flow and everything is written i atacht part...everything is written i atacht part of the file the rest of the file is similar Just need to write in the correct syntax without errors in DIALPLAN so if you are good at asterisk dilplan it is very simple for you and if it is not simple for you . dont bid for it
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