Creation trunk sip sur asterisk jobs

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    20,949 creation trunk sip sur asterisk jobs found, pricing in INR

    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.

    ₹13677 (Avg Bid)
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    8 bids

    Precisely cut out the background around the leaves, branches and trunk. Save as PNG file with transparent background.

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    13 bids

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

    ₹1800 (Avg Bid)
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    1 bids

    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

    ₹1800 (Avg Bid)
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    9 bids

    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.

    ₹12669 (Avg Bid)
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    A2billing installation 5 days left
    VERIFIED

    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

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    6 bids

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

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    ...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

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    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.

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    5 bids

    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.

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    1 bids

    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

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    6 bids
    Telegram to SIP Calling 3 days left
    VERIFIED

    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,

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    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.

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    5 bids
    Core PHP expert 3 days left

    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.

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    ...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 that you have indept knowledge arround Asterisk PBX and integration with SQL / External scripts from within Asterisk PBX We can accept that you just provide the SQL and we can

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    ...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

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    35 bids

    ...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

    ₹792 / hr (Avg Bid)
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    34 bids

    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

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    5 bids

    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.

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    20 bids

    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 URL]�...

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    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

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    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

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    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.

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    ...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

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    15 bids
    VoIP / SIP Softphone 1 day left
    VERIFIED

    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...

    ₹2160 - ₹17996
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    0 bids
    Sip Audio Intercom 15 hours left

    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.

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    ...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

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    asterNET modify sip trunk 12 hours left
    VERIFIED

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

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

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    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

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

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    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.

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    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.

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    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.

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    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

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    ...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

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    ...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 (the VOIP),

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    chan_sip.c:10880 process_sdp: No compatible codecs, not accepting this offer! ... compatible codecs, not accepting this offer! I am redirecting my calls (old provider) to my new freepbx, I have a trunk added but there is an error .

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    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

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

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    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.

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    ...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

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    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

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    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.

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    ...fields ... the only field I have doubts about is the one labeled as follows: The difference between your server time and the [login to view 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

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    We have an app available for android and iOS. We need to add Video calling (VOIP) feature native in the app. Most of configuration stuff should be handled on the back-end by the app. Configuration and account settings will be provisioned by accessing the server and getting the configuration for the user who has logged in. We will start with one of the platform Android or iOS and once that is fina...

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    I need help to set-up 1 ringgroup to 1 external number. This ringgroup should then use a predefined trunk to make the outgoing call. So when I call the ringgroup I should be connected to the external number using the predefined trunk. This should also work when using inbound routes to a ringgroup. I need your remote help. Please give me a fixed

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    we offer a cl...possibility to make phone calls directly in the browser and by sip (soft)-phones. For that we would like to implement sip servlets, for example [login to view URL] and forward calls to sip trunk providers. It would be great, if you could tell us your experience in sip / telecom solutions and java / tomcat / grails.

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    we have an IOS native app with full source code connecting to our online s...source code connecting to our online shop, we want to add calling feature within the app to call our call center. We have a fully functional PBX and the calls will be purely SIP calls and also some chat with call center. we need you to buil the calling part on top of the app.

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    Hi! I need help with Freepbx CRM and asterisk API's: I need customers to be able to make a call from the CRM on the browser with a click on a button using an API, not with Zulu software or plug in.

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