Asterisk avaya trunk sip jobs

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    18,723 asterisk avaya trunk sip jobs found, pricing in INR
    Telegram to SIP Calling 6 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,

    ₹16289 (Avg Bid)
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    2 bids

    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.

    ₹5741 (Avg Bid)
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    4 bids
    Core PHP expert 6 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.

    ₹1252 (Avg Bid)
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    9 bids

    ...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

    ₹90629 (Avg Bid)
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    10 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

    ₹4951 (Avg Bid)
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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

    ₹789 / hr (Avg Bid)
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    35 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

    ₹16289 (Avg Bid)
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    4 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.

    ₹7750 (Avg Bid)
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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]�...

    ₹2296 (Avg Bid)
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    9 bids

    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

    ₹14151 (Avg Bid)
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    6 bids

    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

    ₹22460 (Avg Bid)
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    10 bids

    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.

    ₹9544 (Avg Bid)
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    9 bids

    ...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

    ₹12557 (Avg Bid)
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    15 bids
    VoIP / SIP Softphone 4 days 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...

    ₹2153 - ₹17939
    ₹2153 - ₹17939
    0 bids

    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.

    ₹42336 (Avg Bid)
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    7 bids

    ...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

    ₹16707 (Avg Bid)
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    4 bids
    asterNET modify sip trunk 3 days left
    VERIFIED

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

    ₹15212 (Avg Bid)
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    2 bids

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

    ₹13993 (Avg Bid)
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    4 bids

    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

    ₹69976 (Avg Bid)
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    16 bids
    Develop a Sip client 1 day left
    VERIFIED

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

    ₹34802 (Avg Bid)
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    6 bids
    Flexisip server setup 1 day left
    VERIFIED

    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.

    ₹9400 (Avg Bid)
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    4 bids

    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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    15 bids

    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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    8 bids

    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

    ₹34658 (Avg Bid)
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    4 bids

    ...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

    ₹5956 (Avg Bid)
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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),

    ₹15643 (Avg Bid)
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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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    2 bids

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

    ₹44202 (Avg Bid)
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    6 bids

    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.

    ₹90987 (Avg Bid)
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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

    ₹7176 (Avg Bid)
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    1 bids

    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

    ₹31142 (Avg Bid)
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    8 bids

    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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    7 bids

    ...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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    2 bids

    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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    17 bids

    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

    ₹2153 (Avg Bid)
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    8 bids

    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.

    ₹1794 / hr (Avg Bid)
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    8 bids

    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.

    ₹65514 (Avg Bid)
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    5 bids

    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.

    ₹12844 (Avg Bid)
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    Trophy icon Make an Event Flyer Ended

    I am in need of a flyer design for a Sip & Paint event. I want the flyer to feel like an upscale event and not a club party. I have attached all the information need as well as an example from our last event. I have attached also a jpg and psd file of our sponsors, use which ever one is easier to work with I need the final version to be in a photoshop

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    121 entries

    I have asterisk server with whitelist number and make any number hit my asterisk to pass on this whitelist and if number in whiltelist it pass if not it hangup . now i need to add some options for non whitelist numbers like playback press 1 to continue and if he press add it to my whiltelist also need to send 503 code when call hangup after not passing

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    need to update outgoing calls, currently set to use SIP, but we are cancelling SIP for International calling and use the ISDN for all calls. the PABX is setup to use ISDN for Local and Mobile calls only and since we are terminating the SIP trunk for International calling we need to have it routed via ISDN

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    hello i have voipswich server and i want android sip dialer with the following requiremnet In-app registration for new users using their mobile no. (similar to Whatsapp) Customized with company name / logo SIP protocol support Making outgoing calls Codecs supported:g729, g711, Display Balance , Display Rate Call Log Call Status Indicator Android

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    we need Asterisk Call Center System developer to work with us project base .. with full experience of : - Basic Call System - Call Center System - Call Rating System - Call Details Reports - Branding The system for us .. - CRM Integration

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    I need the full Amazon AMI Install with Freepbx tweaking in order to make this 100% functional. [login to view URL]

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    I need a programmer that is familiar with webooks and the LaML language. I need DIDs programmed to ring a SIP URI and or forward to a PSTN line and SMS/Text redirected to another number. All programming needs to be done on [login to view URL]

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    Trophy icon WebRTC help needed Ended

    I use [login to view URL] v0.11.6 for my WebRTC project. Everything works well but there is no sound on reciever side. Only caller can hear. Asterisk 16 is installed on Centos 7. Need help to fix the issue.

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    3 entries

    We are looking for a Wireshark and VoIP Troubleshooting Expert to help us troubleshoot, find the source of, and resolve a choppy voice issue on a LAN. Someone who is a VoIP, SIP, MGCP, and QoS Expert. The switches in place are Dell L3, and Cisco and Meraki L2 Switches. All with QoS enabled, but maybe not properly.

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    We make an all natural cotton candy that is maple flavoured and we want to play up our Canadia...is maple flavoured and we want to play up our Canadian heritage. This is what I envision; A maple tree with a cotton candy top design done with a maple leaf carved in the trunk - go all Canadian with it.. I’ve attached our current logo for inspiration.

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    all sip calls via vpn calls will come from countries that are blocking sip and ports

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