Setup sip phone asteriskpekerjaan

Saya membutuhkan yg paham dan pengalaman utk melakukan setting/configurasi asterisk saat ini sdh terpasang akan tetapi masih mengkonsumsi cpu server boros sekali rencanaakan dipakai utk call center dg jumlah agent 100 orang.

ccnp voice training and configuration for voice gateway mgcl sip and h323

...to write the installation guide commands to establish it completely on CENTOS 6.7 VPS starting with Asterisk, a2billing & freePBX , I will be using it for PC2Phone AND calling card, so the installation manual you write should indicate the proper setup. I know that there are alot of websites that shows the installation guide like http://www.powerpbx.

Hello we need a person that can setup the ASTERISK / FREEPBX server for us + make manual how to use it. Including all security + usability settings. Budget $100. Must be done in one day.

I need an app for Android and IOS that basically takes and makes sip calls. It can be an open source phone labeled with the name that I need.

...Centrales telefonica Asterisk y GrandStream (con acceso al personal de soporte directo del fabricante). -Configuraciones de troncales SIP dinamico y estatico. -Facilidades de troncales SIP con las prestadoras de República Dominicana. -Configuraciones de IVR. -Cola de llamadas. -Extensiones SIP. -Configuraciones de PBX redundante Asterisk -Configuracione...

Fix bug and implement call in asterisk application using ari. The application is built on node ari package

Asterisk has stopped running on my version of Freepbx. I do not know how to access the command line or the commands to restart. I just need someone to show me, preferably by doing a screen share and walking me through the steps on how to 1. access the command line on my server 2. enter the commands to stop, and restart asterisk

Tengo un gateway goip gsm de 8 sim, 4 voipeadores tplink y un servidor en asterisk pero necesito crear la troncal sip y enlazarlos con los voipeadores para realizar llamadas salientes por el gateway gsm

I want to setup click2Dial function on my PHP website with Asterisk, the open source communications framework. There should be call button on press call should be the dial. and store the recording of that call. Have to setup Asterisk on AWS.

I need to install a2billing in freepbx 13 with asterisk 11 and have to verified the command script by using teamviewer in my virtual local server. Kindly also need to show some configuration, like in freepbx IVR has two option and one of them is calling card option. so if customer chose the calling card option then how to divert them to a2billing and

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

Install & Setup Free PBX on Digital Ocean Droplet using Centos Setup with carrier, all basic features etc. I am new to this so will need walk thru after all is setup.

...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.

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.

...conectar nuestra Central telefónica Issabel (Asterisk) con el CRM Bitrix 24. Lo que queremos lograr es poder realizar llamadas telefónicas desde el mismo Bitrix a través de las centrales Issabel ya configuradas. El bitrix cuenta con instructivo para conectarlo con FreePBX. Si bien este se maneja también con Asterisk, es necesario hacer la configuraci&oa...

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

HI We need to have a SIP trunk between a Telegram ID ( [log masuk untuk melihat 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

...you package up the files and send them all back to me. I will attach a .zip file which has an example InDesign file that you can use which already has the header and footer setup for you. In this file has the swatches of color that can be used for this project which is basically a dark blue and light blue with some grey. This zip also contains the logo

...you package up the files and send them all back to me. I will attach a .zip file which has an example InDesign file that you can use which already has the header and footer setup for you. In this file has the swatches of color that can be used for this project which is basically a dark blue and light blue with some grey. This zip also contains the logo

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

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@[log masuk untuk melihat URL]>' failed for '[log masuk u...

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

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.

...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.

Hi i need a Sip client similar to Linphone Android. You should use the [log masuk untuk melihat 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.

...freelancer has to setup a Microsoft Exchanger 2016 hosting for multiple domains (Multi-tenancy) on a single new server, microsoft server 2016. Keep in mind that the 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

...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.

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

Create a Survey FreePBX/Asterisk module to measure customer satisfaction levels for our call center agents services. Current software versions: Elastix 4 Here is what is needed. After a call, our Call Center Agents will transfer a customer call to an extension where the survey should be executed. Survey will consist of 1 or more fixed questions. We

I have a personal asterisk server for my self and kids and I see lots of messages like this. --STATUS: HANGUP on h from from ,,XXXXXX") in new stack Looks like some one trying to make calls but I dont have outgoing plans just a few extensions. I have fail2ban installed but it doesnt seem to be catching them.

...fields ... the only field I have doubts about is the one labeled as follows: The difference between your server time and the [log masuk untuk melihat URL] server time (in seconds)*: The asterisk (*) at the end of the field name corresponds to the following: * Please ensure your system clock is set to proper time and time zone. AuthorizeNet gateway will validate the