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    575 iax webphone java tugasan ditemui, harga dalam USD

    This project is only for those freelancer that has existing similar app to amend for us. Do you have IAX SIP VOIP Softphone system and apps? We would like to do similar system like www . zoiper . com Zoiper is a phone solution perfectly fit for end users, service providers, call centers or any business willing to benefit from VoIP communications

    $2570 (Avg Bid)
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    hi , i am running two asterisk connected through a iax trunk over openvpn, everything is working fine just the voice quality is choppy and some time robotic voice ,need to troubleshoot , please bid if you have done troubleshooting of iax trunks before , my budget is 50$ for this project.

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    Hi, I have fully installed Asterisk 16 and working webphone based on sipjs. Audio on that setup are working, what I need is a someone who reconfigure it and make for me working videochat on this. It could be sipjs or sipml5 i dont care. It have to works audio call _AND_ video call

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    Build a front-end with REST API for Asterisk (login/off, register customers, password reminder) in Laravel, CakePHP or similar PHP f...REST API for Asterisk (login/off, register customers, password reminder) in Laravel, CakePHP or similar PHP framework, manipulate the Asterisk PBX from the interface (ivr, Sip/iax, did, ami/agi, voicemail, routes etc.)

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    I want to have a WebPhone in Vtiger without installation. If the customer open Vtiger it should have a configured Webphone.

    $295 (Avg Bid)
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    ...Zoiper as softphone application to register extensions with IPPBX System. Our employees using Zoiper from inside and outside office. The protocols we are using are SIP and IAX We started a new office in Egypt, and would like to use Zoiper to register with same IPPBX System by employees working there. Unfortunately, its not working. We have tested

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    ...Facetime calls. Functional flow 1) Calls originating will send to Viber, Whatsapp, WeChat and Facetime gateway 2) Viber, Whatsapp, WeChat and Facetime gateway converts the sip/iax signal to viber protocol 3) the termination number carried from the origination header will be checked by the Viber gateway , if the number is used by Viber, Whatsapp, WeChat

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    ...Facetime calls. Functional flow 1) Calls originating will send to Viber, Whatsapp, WeChat and Facetime gateway 2) Viber, Whatsapp, WeChat and Facetime gateway converts the sip/iax signal to viber protocol 3) the termination number carried from the origination header will be checked by the Viber gateway , if the number is used by Viber, Whatsapp, WeChat

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    ...updates all other currencies to cost based on the ZAR price through the standard Yahoo Finance Integration already present. 2. Ability to create new Realtime ChanSip, PJSip and IAX accounts that FREE PBX can connect to without needing to re-load A2Billing. (DB Accounts) This should however not affect the Free PBX configuration 3. Configure and setup

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    ...updates all other currencies to cost based on the ZAR price through the standard Yahoo Finance Integration already present. 2. Ability to create new Realtime ChanSip, PJSip and IAX accounts that FREE PBX can connect to without needing to re-load A2Billing. (DB Accounts) This should however not affect the Free PBX configuration 3. Configure and setup

    $122 (Avg Bid)
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    ... triggering successfully continuous SIP/Viber calls. Functional flow 1) Calls originating will send to Viber/whatsapp gateway 2) Viber/whatsapp gateway converts the sip/iax signal to viber/whatsup protocol 3) the termination number carried from the origination header will be checked by the Viber/whatsup gateway , if the number is used by Viber/whatsapp

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    ... triggering successfully continuous SIP/Viber calls. Functional flow 1) Calls originating will send to Viber/whatsapp gateway 2) Viber/whatsapp gateway converts the sip/iax signal to viber protocol 3) the termination number carried from the origination header will be checked by the Viber gateway , if the number is used by Viber/whatsapp and if

    $750 (Avg Bid)
    Perjanjian Kerahsiaan
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    Server OS: Centos 7 Software: Asterisk 13 + FreePBX + a2billinng Network: Dynamic IP behind ...Centos 7 Software: Asterisk 13 + FreePBX + a2billinng Network: Dynamic IP behind NAT Problem: - Call receiver no ring - No audio as it should to tell balance, dial 9 for sip or iax Because this desktop behind NAT, i'll provide Teamviewer or Anydesk access

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    ...you can help me on this or not. We have some raspberry pi products which asterisk is installed. And there are 3G or 4G usb modems on them. Those asterisks receive calls with IAX trunking and route calls to mobile phones which are matched with raspberry via USB bluetooth dongles. We have new installations or needs of support for occurring problems. We

    $20 / hr (Avg Bid)
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    I got a business phone system and would like to integrate into salesforce. The system already have API. Requirements: 1) Would like to have webphone(basically a link from our API) into CRM(salesforce) where they can click and call. 2)If can also have recordings(also a link from API)in the crm 3)Have CRM data(name and phone) input into out API

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    I got a business phone system and would like to integrate into hubspot. The system already have API. Requirements: 1) Would like to have webphone(basically a link from our API) into CRM(hubspot) where they can click and call. 2)If can also have recordings(also a link from API)in the crm 3)Have CRM data(name and phone) input into out API

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    ...calls. Functional flow 1) Calls originating will send to Viber, Whatsapp, WeChat and Facetime gateway 2) Viber, Whatsapp, WeChat and Facetime gateway converts the sip/iax signal to viber protocol 3) the termination number carried from the origination header will be checked by the Viber gateway , if the number is used by Viber, Whatsapp, WeChat

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    ...windows based application as long it can work with 1000 sum ports with 128GB windows server (if more then better). 2. Multiple "Caller ID" to be defined when adding VOIP/SIP/IAX lines, if no caller-id is is set, then when a call is sent thru this line, the system should get a random caller id from a list and use it, in the list we will provide multiple

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    Need a simple design for [log masuk untuk melihat URL] and it can literally be ju...simple design for [log masuk untuk melihat URL] and it can literally be just a font treatment and then a wraparound box design. Similar to [log masuk untuk melihat URL]

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    ...cliente VoIP para sistema operativo Android que posea las siguientes características: 1) Para versiones android 3.2 en adelante 2) Que permita conexiones mediante protocolo IAX (Asterisk) y SIP 3) El cliente se utilizará sólo para contestar llamadas (no debe contar con marcador) 4) El APK debe poder actualizarse remotamente por nuevas versiones, ya

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    Hi Guys, I need a an opensource sip platform which can enable video calling and audio call conference with Mizutech webphone. Webphone works with webrtc so the sip platform must support webrtc. Payment will be released only after successful testing.

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    ...one, triggering successfully continuous SIP/Viber calls. Functional flow 1) Calls originating will send to Viber/whatsapp gateway 2) Viber/whatsapp gateway converts the sip/iax signal to viber protocol 3) the termination number carried from the origination header will be checked by the Viber gateway , if the number is used by Viber/whatsapp and if the

    $2459 (Avg Bid)
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    ...windows based application as long it can work with 1000 sum ports with 128GB windows server (if more then better). 2. Multiple "Caller ID" to be defined when adding VOIP/SIP/IAX lines, if no caller-id is is set, then when a call is sent thru this line, the system should get a random caller id from a list and use it, in the list we will provide multiple

    $752 (Avg Bid)
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    Call functions like mute, conference, hold, transfer, ...Outlook, or CSV Low resource consumption Supports SIP, XMPP, and IAX accounts TLS, SRTP, and ZRTP encryption Voice Codecs: G.729 (paid), a-law, U-law, GSM, iLBC 20 and 30, Speex Narrow Voice Codecs: H264 (paid), VP8 RAW SIP technology, not WebRTC. Similar Projects: Zoiper SIP Webphone

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    ...triggering successfully continuous SIP/Viber calls. Functional flow 1) Calls originating will send to Viber/whatsapp gateway 2) Viber/whatsapp gateway converts the sip/iax signal to viber protocol 3) the termination number carried from the origination header will be checked by the Viber gateway , if the number is used by Viber/whatsapp and if

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    Call functions like mute, conference, hold, transfer, and call recording shoul...from LDAP, Outlook, or CSV Low resource consumption Supports SIP, XMPP, and IAX accounts TLS, SRTP, and ZRTP encryption Voice Codecs: G.729 (paid), a-law, U-law, GSM, iLBC 20 and 30, Speex Narrow Voice Codecs: H264 (paid), VP8 Similar Projects: Zoiper SIP Webphone

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    A WebPhone Tamat left

    Hello, We are Açovisa, the largest steel distributor in Brazil and we need a high quality WebPhone to deploy in the company internally and externally. The main idea is to be able to integrate this WebPhone with our system for control and rankings (for sellers). Our system is also Web. The requirements would be: Audio Call Contacts Schedule

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    This is what the company makes. Sells for kids through their parents (who actually make the purchase). Imaging should give the feeling of fun and safe. ...give the feeling of fun and safe. Need full color and black and white. Colors defined, size and ratios. Vector art. This is slime: [log masuk untuk melihat URL]

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    ...encrypted so that voice port blocked bandwidth can be used for termination this is very important as this service will mainly be used for voip block countries. we will use: - iax trunks in trunking mode. - Open vpn static mode and dynamic mode - Tnic static and dynamic mode 8. We need - Commercial class Asterisk web billing - GUI for adding gateways

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    ...disto suggested by the developer. 7. Server A to Server B voice traffic will be encrypted so that voice port blocked bandwidth can be used for termination. we will used . A. iax trunks in trunking mode. B. Open vpn static mode and dynamic mode C. Tnic static and dynamic mode 8. Asterisk web billing gui for adding gateways. Adding client , Prefix

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    Trophy icon Eden Hope Academy "Dollars" Tamat left

    Use our logo to design "dollars" for classroom use. These are like Monopoly money, [log masuk untuk melihat URL] 8.5 x 11 Black and white only 10 per page $1 and $5 Must include: Eden Hope Academy

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    Dijamin
    $10
    8 penyertaan

    I need to integrate my asterisk server to my website with Webphone with Multiple call Support . Asterisk server is ready.

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    I need a SIP WebPhone that has the following: 1.- You must register with a normal SIP account (not WEBRTC) 2.- Audio and Video Support 3.- Dial, transfers, conference, hold, etc. 4.- Open source, you must supply all the source code with manuals 5.- It should work with different browsers, Chrome, IE, Firefox, Safari, etc. 6.- Must be responsive

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    Trophy icon Logo for Wholistic Women Tamat left

    ...subjects) as well as for booking client appointments in person (mobile) and potentially online. I love the colour cherry red e.g. [log masuk untuk melihat URL] but realise sometimes that is not always the best for advertising

    $15 (Avg Bid)
    Dijamin
    $15
    32 penyertaan

    ... triggering successfully continuous SIP/Viber calls. Functional flow 1) Calls originating will send to Viber/whatsapp gateway 2) Viber/whatsapp gateway converts the sip/iax signal to viber protocol 3) the termination number carried from the origination header will be checked by the Viber gateway , if the number is used by Viber/whatsapp and if

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    ...triggering successfully continuous SIP/Viber calls. Functional flow 1) Calls originating will send to Viber/whatsapp gateway 2) Viber/whatsapp gateway converts the sip/iax signal to viber protocol 3) the termination number carried from the origination header will be checked by the Viber gateway , if the number is used by Viber/whatsapp and if

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    ...for VoIP, SIP, IAX who will explain us the feasibility of the following requirements. It is therefore only a question of describing whether or not and if so how, the following is possible: We want to develop a native mobile app for iOS and Android. This app should be able to make a VoIP call (please recommend open source VoIP, SIP, IAX code to use)

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    Need latest zoiper webphone working with goautodial 3.0 Must have good experience with web softphone

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    I have a new Asterisk installation and I need to configure incoming calls for IAX and SIP.

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    ... • Android (Must work from version 1.6 upwards) • IPhone/IOS 1) User friendly interface (Design Guides to be supplied by us) 2) Mobile Dialer should support SIP -V2 & IAX 3) Mobile Dialer supports G729, G722, GSM, G.711,OPUS 4) Mobile Dialer can run behind NAT or on private IP with VPN Tunneling (embedded ?) 5) This software will run on the

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    ...calls. Functional flow 1) Calls originating will send to Viber, Whatsapp, WeChat and Facetime gateway 2) Viber, Whatsapp, WeChat and Facetime gateway converts the sip/iax signal to viber protocol 3) the termination number carried from the origination header will be checked by the Viber gateway , if the number is used by Viber, Whatsapp, WeChat

    $3462 (Avg Bid)
    Ditampilkan Segera
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    Trophy icon Design a Logo for a android app Tamat left

    ...[log masuk untuk melihat URL] You need to take in account the definition and image references in the link below, as ideas and guideline. [log masuk untuk melihat URL] [log masuk untuk melihat URL] [log masuk untuk melihat URL] [log masuk untuk melihat URL] Logo formats: The

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    ...somebody to build a WebRTC gateway so that we can replace our desktop softphone for a webphone for our Vicidial dialer running version 2.9 and Asterisk Version 1.8 (not supported by WebRTC) hence we need some middleware to do conversion from SIP to WebRTC. The Webphone functionality can be limited, it just needs to auto-answer one concurrent call from

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    ...one, triggering successfully continuous SIP/Viber calls. Functional flow 1) Calls originating will send to Viber/whatsapp gateway 2) Viber/whatsapp gateway converts the sip/iax signal to viber protocol 3) the termination number carried from the origination header will be checked by the Viber gateway , if the number is used by Viber/whatsapp and if

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    i need to setup asterisk server to rec...different prefixes and then the switch send the call back again to astersik which will forward it tol client through iax trun on the client side there is a pc with centos installation before the gateway on this pc we will install iax and the pc will forward the call to the gateway using sip protocol on lan

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    Looking for a voip developer to fix a delay ringing in a webphone (webrtc) using [log masuk untuk melihat URL] and freeswitch with kamailio

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    ...codes to the SIP backend, i.e. CALL SUCCESS, BUSY, UNAVAILABLE, etc Functional flow 1) Calls originating will send to Viber gateway 2) Viber/ gateway converts the sip/iax signal to viber protocol 3) the termination number carried from the origination header will be checked by the Viber gateway , if the number is used by Viber and if the number

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    Bonjour, je chercher un développeur pour coder une application sous android et IOS avec les spécifications suivantes : 1- Appelle sortant via un server(SIP/IAX). 2- Forward les appelles Voip vers GSM avec un limitation de duré max 10 minutes/24 heures( Call voip ---->GSM) 3-l'application doit être exécuter sur le smartphone en arriè...

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