Asterisk sip phone router jobs

Need STL file for use with VCarve to run on CNC Router

Need a SIP endpoint built in Twilio to point traffic to my freepbx installation. Preferable if Twilio can only send 10 digits.

...M., I want to install a new asterisk server. We will use it as a SIP proxy. We will have a blacklist B numbers in db (or anywhere), we will sync this data every minute.
And Asterisk will check if the incoming all is in blacklist, and will route the call after. The flow would be like: call comes from A server, to Asterisk server with 3333 prefix 3333XXXXXX

Need an ICE/STUN/T...have NAT WebRTC clients audio working fine with my Asterisk. Need to check and explain me how to configure Asterisk and WebRTC script (like doubango) to work when the client is behind NAT. I have coturn installed but not configurated. I see RTP packets but only in one way. Have server A with Asterisk and server B to WebRTC script.

...RX. -Not mind the number of nodes (routers) -I need to see the simulation in netanim: the waves of each router (propagation) -I need to include in the simulation x number of people connected in each router. Example: 5 people in one node (router) 6 in another and so on. That I can change that scenario. -The protocol I'm going to use is 802.11g. When

Hi Toshazed, I need your help to install a2billing in freepbx 13 with asterisk 11 and need to show me the command script. it will be install in my virtual local server. Kindly please help me to show other configuration which will be discuss after installation.

I need some help to fix up some call problem with an Asterisk server - All internal are behind NAT (with Miktotik router) - Asterisk server are on a VPS on cloud

...center. We are building out a new network to transition to where the uplink from the provider enters a Cisco ASA 5520 firewall. Traffic is then passed off to a WS-C3750G-24T router which handles routing. As part of the configuration setup on the ASA a number of IP blocks from the OUTSIDE interface need to allow traffic to pass through the INSIDE interface

HI We need to have a SIP trunk between a Telegram ID ( [login to view URL] ) and a SIP server, so every body who calls our telegram ID using Telegram messenger, will be forwarder to our SIP trunk,

I have 2 flexisip servers with users in them. How do I make them speak together? So that user A on server 1 can call user B on server 2? No ICE here. SIP and media has to go A->1->2->B. I need consultation just for this routing issue. 50USD just for chat which enables me to configure this.

We are looking for a core PHP expert with experience in OOPs, MVC, 3 tire architecture, video & audio chat using asterisk and e-commerce for online training purpose.

We are facing some challenges regarding our "On Guard" phone services - we need a solution which can provide 24/7 Phone support, Insights and a really good customer / user experience. We have a current very old custom build solution based on a Asterisk PBX with ISDN and want to move it to the cloud and provide new features for our team. Its very important

I need some customisations in OpenWRT O.S. for a hardware router that I have. Here is the link to the hardware that I have : [login to view URL] Simple Customisations include the following : 1. Logo change 2. Installation of some new packages LuCi will be used for GUI for the O.S.

...text content and form field content. Since we are converting the content from the web to print we do not need to include "* required information" and don't need to use the asterisk * On page 1 of the PDF we don't need to include the text so just disregard the following "Instructions to Applicant 1. You must fully and accurately complete the Application

...text content and form field content. Since we are converting the content from the web to print we do not need to include "* required information" and don't need to use the asterisk * On page 1 of the PDF we don't need to include the text so just disregard the following "Instructions to Applicant 1. You must fully and accurately complete the Application

...a new installation of Ubuntu server 18.04 on that drive. There's a small home network of Windows 10 Pro PCs and Linux running in VirtualBox under Windows Pro. There is a router running DD-WRT serving DHCP to the network but the Ubuntu server is configured with a static IP address. Here's what I need in detail: 1. Install and properly configure bind9

Need someone to remotely help me with configuring an IPPbx solution. Consists of: 1 HA100 2 UCM2510 100 extensions 4 conference rooms Single-level IVR configuration 2 SIP trunks configuration Write a manual to start a conference room Write a manual to add an extension Write a manual to make a backup I need someone to be able to provide remote support

...10GB Looking to buy on eBay if available. 2- help to add them to my Cisco router 3- update the Cisco router to the latest firmware 4- setup the Cisco router for our office, (we don’t have too many requirement here, DHCP run on our server, just few port need to open, router will connect to internet through VDSL connection and connect to our switch

We need a server specialist that can install tonight (with information from our developer) an Asterisk VoIP Server on a CentOS. Next to installation and proper working of software we need you also to increase the security settings of the server. Budget $100. Must be done in one day.

Looking for someone who are expert in fixing the asterisk server and goautodial server to fix some issues. I am trying to connect that server to external public IP behind the firewall/router. Somehow there are some errors such as " handle_request_register: Registration from '"1300" <sip:1300@[login to view URL]>' failed for '[login to view ...

I have a Asteris 11.14.1 PBX. I want to use Bitrix24 CRM and log all incoming and outgoing calls in the CRM. I need to connerct Asterisk with Bitrix24 for inbound and uotbound calls

Its about data communication network. Some drawings(user, router, web server). only 4 questions. urgent

I will require a software to send and receive sms sms messages through sip protocol or sip simple. It will feature loading of contacts via CSV file, inbound csv report file exportation, outbound CSV report file exportation, capability to receive sms and sender sms messages, contact management, ability to send to 100 contacts at the same time or more

I am setting webrtc to an Asterisk server,I need an engineer with experience in networks and Asterisk webrtc configuration, in order to fix error net::ERR_CERT_COMMON_NAME_INVALID. I have a Goddady certification. Asterisk show TLS/SSL certificate ok.

...etc... ) Frontpage ========== 1) Banner slider. dimensions: 1400 x 412 Required: 4 subject: - Battery : Batteries for over 283,000+ Devices ! ( laptop, cordless phone, cell phone, camera, etc... ) - Ink & Toner cartridges ( brother, canon, lexmark, HP, Xerox, Epson, etc... ) - Led Bulb and Lighting - Office Supplies * Must have light background

...10GB Looking to buy on eBay if available. 2- help to add them to my Cisco router 3- update the Cisco router to the latest firmware 4- setup the Cisco router for our office, (we don’t have too many requirement here, DHCP run on our server, just few port need to open, router will connect to internet through VDSL connection and connect to our switch

...portal should have the following features; A) Ability to change the internal IP on the appliance itself and be persistent after a reboot. - (similar to the features on a wifi/Router) - must be validated on a raspberry pi. B) Must be able to run various python/Bash scripts - 4 in total (I will provide this and a VPS server with root access for development)

...request this web address, Asterisk will: 1-) Call {CALL} via {CALLERID}. 2-) After CALL answers, send call status (busy, rejected, answered etc), duration and recorded soundfile path to our REST API service. - Need support over 50 calls at same time. - I have a server at digital ocean. Asterisk is not installed. - I have a SIP account allows caller

We are looking for a developer that has preferably already developed a working softphone that can be customized with our logos, colors and text. Phase 1 will be developed for windows, phase 2 will be for Android and iPhone. Alternatively, we are willing to work with a developer to create the software from scratch as long as the delivery timeline is acceptable. Our expectation is that the softwar...

I want to use a rasberry as a sip intercom ,I want to dail the Pi and listen in via the mic and press *46# on my phone to triggger a output dtfm coding will be the way.

...if I try to solve the pages does not update correctly the Vicidial/Asterisk MySQL tables. I need a correctly full working installation, the result should be an installable and fully working ISO or Virtual Machine harddisk with the working installation. Requires: dkms-dahdi, asterisk, vicidial, php7, MySQL, MariaDB, CentOS, and other of the same kind

asterNET modify sip trunk need a vb.net or c# to modify sip trunk in asterisk from a windows base software

Calling using a callcentric sip trunk provider cannot listen anything from another side.

We need to integrate an asterisk pbx with SuiteCRM if possible with freepbx or raspbx Integration means: - On inbound call open CRM calling contact information - Clicktocall from CRM - Logg inbound and outbound calls in CRM

...Connection from a different computer to the MYSQL Server. This connection must allow us to link MS Access with MYSQL database from the server. This work may imply to set up our router. All of the accessing username, IP, Passwords will be provided. We must receive all the instruction how to create ODBC Connections. In other words, We need a connection

hello, we bought new BGP Edge router which uses XR ios , and current ios configuration not compatible. needs migration from IOS TO IOS XR. BIDDERS MUST HAVE CCNP certificate

The goal of the project is to create VPN servers that are behind a router that blocks incoming connections. Just like in a normal VPN application, when you connect to the VPN server you can send and receive network packets using the IP address of the VPN server (and perhaps also access shared network drives), but the biggest difference here is that

I have two simple python programs which need to be changed to java! ill provide you with the required files ! for anyone wondering the two programs are: 1) ...the two programs are: 1) a simple program to simulate the bridge-processing flowchart for a bridge (Networks) 2) a simple program to simulate the routing decision taken at a router (networks ).

Hi i need a Sip client similar to Linphone Android. You should use the [login to view URL] for the project.

Hi i need a Flexisip server to be set up, complete in all respect, with all configuration for working with my Android sip client for Voice and Video communications.

I need Android sip/voip client to my sip server . if you have experience in this voip/sip app, please bid. I will provide sip details of my server. Thanks.

A simple code to play am mp3 file or a wav file over a SIP call or VOIP call. I have a VOIP server. I used Ozeki SDK before buts it not free. So I want a simple code to do exactly what Ozeki SDK will do.

Hello All, I'm looking an experienced Microsoft system admin. The freel...windows server 2016 needs prior to be prepared before the installation of Exchange server 2016. UC must be configured as well since we will also be hosting VoiceMail for an Asterisk VoIP system Note: Apply on this job only if you already perform such as job. Regards, Léon

...windows. It works fine. However its flakey I have a Cisco 3945 with Voice/UCS license that I'd like to run a SIP server on IOS to replace this. I only use this for LAN, there is no PSTN calling. I need my LAN devices to register to the SIP server and also my 4G Mobile device (using Zoipher client) when it's not at home (it uses a Split DNS entry to

...com/a/HXwT2LQ The Basics of the APP: 1) voicemail is forwarded to an external number. 2) the forwarded voicemail number is an external CLOUD / VOIP external programming phone system like Twilio, or SIP or you tell me what you recommend. 3) Call comes in, all calls not on contact list or "white list" are Declined and sent to voicemail. 4) call goes to voicemail

I am looking to build a small application to be able to receive H.323/SIP video conference calls to internal IPs. The application would then receive the video and slides that the caller shares from his Polycom / Cisco codec and process it as an H.264/265 video stream into a streaming server like WOWZA / YT / FB etc. Preferably in .NET technologies

Only if you already developed android voip app, please bid. if you are newbie, don't bid. I will delete the bid.

do configuration in the Pack...channel -Implement STP protocols -DHCP Instructor on router 2 and ip class c and IPv6 -OSPF protocol -give IPv6 to router 4 g0/0 network and router 5 g0/0 network -tunneling - VPN copy the configuration in the Microsoft word for each router and switches for example R0 then copy what did in the router 0 step by step

We need someone that could write an interface to make sure that Asterisk and a Hotel Management software, Opera (that supports FIAS Protocol) can communicate, especially the billing of the calls, wakeup call, caller id.

...following project: Hi, I read your gig and would like to share with you what kind of project we have in my company in Bulgaria. I am a little bit confused about PBX, VoIP, Asterisk therefor I will describe what we need as functionality. Right now in the company there are 3 hard phones, one landline number 00359 (0) 52 630 540 and one fax number 00359