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    23,231 j2me sip blackberry iş bulundu, ücretlendirmeleri EUR

    Hi thepasserby82 Merhaba Ankara'da bir şirketiz.,Bir VoIP-SIP programı geliştirmek istiyoruz. Farklı olarak speech to text ile entegre olacak. Bunun örnek kodları github ta mevcut. Bunu imkan veren en iyi voip open source çözümü olarak asteriks bulduk. Türkiye de biri ile çalışmak bizim için daha avantalı olacaktır. İletişim bilgilerini pay...

    €893 (Avg Bid)
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    Belirli bir prefixle ayrıştırılmış belirli bir IP den H323 ve/veya SIP protokollerinden gelen toplu çağrıları elemek için whitelist ve blacklist yazılımının oluşturulması. Whitelist ve blacklist te bulk ve/veya tek tek data girişi için kullanıcı ara yüzü oluşturulması. Filitreden geçen trafiği tekrar ilgili prefix ile belirli bir IP ye yönlendirilm...

    €496 (Avg Bid)
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    Belirli bir prefixle ayrıştırılmış belirli bir IP den H323 ve/veya SIP protokollerinden gelen toplu çağrıları elemek için whitelist ve blacklist yazılımının oluşturulması. Whitelist ve blacklist te bulk ve/veya tek tek data girişi için kullanıcı ara yüzü oluşturulması. Filitreden geçen trafiği tekrar ilgili prefix ile belirli bir IP ye yönlendirilm...

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    iphone-ipad-samsung-blackberry cep telefonlarının telefon kapakları giydirme tasarımı için, vereceğimiz ürünlerin 3 Boyutlu modellemesi yapılarak bu modellerin sadece ön ve yan kısımlarına UV mapping yapılarak, OBJ formatında teslim edilmesi gerekmektedir. Ortalama modellemesi yapılması istenen ürün sayısı 10-20 adet arasıdır. En uygun ve en hızlı teklifi

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    i want the installation of the goautodial v4 and the sip set up in my vps

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    ...professional in sip and asterisk. You can try the software before of apply for this project. Just clone linphone android 4.1 from this link: [login to view URL] Then provide an asterisk switch and try to run sip messaging. Note: I’m using MessageSend() to send SIP MESSAGE requests between my registered SIP endpoints

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    We need and expert with Opensips in order to configure and connect it to Asterisk PBX to configure Opensips for handling SIP messaging to SMS provider and vice versa. a complete Opensips solution to send/receive sms from peer-to-peer SMS provider. Encoding and splitting the messages by Opensips with following guideline: All software must auto start

    €20 - €166
    €20 - €166
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    ...using Linux Deploy app . Now we have Asterisk on an Android mobile phone . As i saw many calling app , like skype or other calling app's , can give choices to user to call over SIP or internally through app or use GSM (local GSM network on mobile phone) if GSM option is selected ,call will be processed over GSM network . My main Goal is not to Build

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    I am working on configuring Adtran TA924e voice gateway to connect to a SIP trunk provider over the internet and have a few analog FXS stations call out and all show the same caller id (phone number). I have configured everything with the trunk, it registers, can call in and out, but cannot figure out how to override the caller id for all FSX ports

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    ...minimum of 20 calls at the same time, but preferably many more than that. - The calls can be terminated (sent to) a US phone number +1585-XXX-XXXX or to an IP address using SIP. We would be receiving up to 1,000 calls per day on this number. Each call would average less than 1 minute in length. We need at least 5 telephone service providers'

    €27 - €223
    €27 - €223
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    ... 2. To have a failover SIP server set for each trunk. We currently use a combination of service providers for Trunks on our systems, each with a different primary and secondary SIP server. Currently our trunk settings only accommodate for a single SIP server to be entered, but we would like to also include the secondary SIP server as a failover in

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    We are seeking someone who is Very Experienced in assisting us getting our VoIP Telco off the ground. We have wholesale access to SIP Trunks/Channels and DID's We need someone who is experienced in Softswitches, Cloud PBX, Website to PBX integration so customers can sign up online to our plans, self manage their own PBX. The system creates an PBX

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    Build a PBX with Twilio 2 gün left
    ONAYLI

    I am currently using Freepbx on the cloud as my VOIP service. I am wanting to build applications on T...Freepbx on the cloud as my VOIP service. I am wanting to build applications on Twilio Programmable Voice platform that will essentially behave as a PBX in which I can connect my SIP endpoints directly to. Please provide a demo of a similar project!

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    I am currently using Freepbx on the cloud as my VOIP service. I am wanting to build applications on ...Freepbx on the cloud as my VOIP service. I am wanting to build applications on Twilio Programmable Voice platform that will essentially behave as a PBX in which I can connect my SIP endpoints directly to. Please provide a demo of a similar project!

    €223 - €670
    Yerel
    €223 - €670
    0 teklifler

    Configure Twilio Elastic SIP Trunking with FreePBX. Required to Share Screen Virtual and Troubleshoot together: I have followed all the steps in this link and my Aterisk is still not working: [login to view URL] I think it could be a PHP5 error or maybe something I'm not

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    ...serves as a prelude to the trip. The main idea is to convey the Teleport in the shape of the takeaway coffee cup, which brings you to different part of the world with every new sip. I’d like to implement a very simple and understandable graphics, but, at the same, enrich it with interesting and attractive details....

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    ...is used for all outbound calls Type – this is typically set to ‘Friend’ DTMF mode – this is typically set to “rfc2833’ Default User – authentication username to contact your SIP provider NAT – this is typically set to ‘force_report, comedia’ Qualify – this is typically set to ‘Yes’ Allowed Codecs – this i...

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    ...using Linux Deploy app . Now we have Asterisk on an Android mobile phone . As i saw many calling app , like skype or other calling app's , can give choices to user to call over SIP or internally through app or use GSM (local GSM network on mobile phone) if GSM option is selected ,call will be processed over GSM network . My main Goal is not to Build

    €2396 (Avg Bid)
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    6 teklifler

    i have a dialer installed but i dont know how to set up the sip trunk, so help me to set up the sip trunk and make a test call

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    I am looking for an asterisk specialist for realizing this special project. A (SIP-)telephone should be placed in a small museum. Visitors can call up and listen to various audio announcements by entering a telephone number (two-digit? three-digit? tbd). My idea is to use a raspberry Pi 3 B+ and RasPBX on it. The installation is not connected to the

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    Need a voip server where I can call on SIP clients App

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    ...customized version of CSipSimple with xmpp and sip tls/srtp features and full remote provisioning system. the app should bind with server once with user/pass and form then receive all configuration by api and check frequently based on a time interval. VoIP application (android) - Based on opensource CSipSimple - SIP-TLS / SRTP encryption support - Remote

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    I’m looking for an MDM/UEM for BlackBerry consultant in Australia preferably with security clearance or in process of being cleared.

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    ...company based in the North East of the UK. developing bespoke Telecommunication SoftSWITCH and Billing modules, to integrate with the third parties existing software for VoIP/SIP phone system with many rich features and is suitable for small to the midsize contact centre. SoftSWITCH - We are interested in welcoming a person with a profound interest in

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    magnusbilling Bitti left

    configuration spa112, pap2t, check codec, test calls on sip devices, set trunk, rates, etc

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    I need sip, swp, lumbsum calculator source code code anyone have contact me asap

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    Connect ONT Pot line from Huawei Router HG8245 to 3CX IP PBX via SIP IP Based, VoIP line is from a huawei Router and need to connect it to a local PBX and then the second scenario will connect to a 3CX PBX from cloud. Actual POTS Line have a private IP , Need to also configure inbound and outbound rules. Only experienced engineer. Need to be done

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    ...project can be in two parts. A mobile application running on the Android platform and able to register on a SIP registration server (Already existing). The Mobile application will register to a server and accept call from that server with SIP protocol. After that call will terminate to a GSM network. (this part just like a traditional GMS getaway)

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    i need an expert in voip to help me with a telemarketing campaign knowlage in sip trunk dialer pre record messages

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    i need to set up it to make calls to usa for telemarketing i will need a sip trunk provider

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    Need a FreePBX expert to make an SIP server and connect it to Dinstar GSM Gateway

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    ...Linphone API calls and answers messages. A2 Billing has an existing SOAP interface which provides already some functionality such as creation of user accounts with DID and SIP account. So one can adapt some code for the new Linphone API. [login to view URL] Verification should work

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    Instalação e funcionamento cliente x servidor SIPCAPTURE HOMER

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    I'm starting a new company. We need a company logo for a website and an application icon. The company is called ...those events. We love the logos of Airbnb, LimeBike, SoundCloud, Spoitfy, etc for their simplicity. I don't want to be too prescriptive but since this is very close to "Blackberry" phones, we'd like to stay away from that reference.

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    Create an android app Softphone to automatically call a list of contacts and also create batch conference calls up to 16 contacts at once

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    Voip dialer to work using sip account

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    VoIP Articles Bitti left

    ...titles/ideas but I'm open to suggestions. Articles should be geared toward people searching for a VoIP solution provider. Budget is $20 per article. What is VoIP and How Does It Work SIP Trunking Cool Things Your VoIP System Can Do (features) VoIP Sound Quality Issues and How To Fix Them Use VoIP to Reduce Cell Phone Cost What Happens to Your Business Phone

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    ...Communications Platform, for Web and Mobile Developers to rapidly build and scale telco applications using an enhanced set of features for Voice, Video, and Messaging Apps for WebRTC, SIP and PSTN networks. 2 Restcomm USSD gateway [login to view URL] SSC gateway [login to view URL] GMLC This is to be installed on our VPS. [login to view URL]

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    im looking for someone to set up kamailio sp server for voip project

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    i have voip softswitch. now I am looking for mobile dialer to use sip account and password

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    Desenvolvimento de um cliente desktop para captura de dados SIP(Áudio de entrada e zaida, numero discado e etc) usando endereço, usuário e senha do agente no sistema como asterisk, opensip e etc... Sem preferencia de linguagem para desenvolvimento o único requisito é seu funcionamento nos sistemas operacionais windows e linux.

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    Take inputs of header content of a L3 Packet of atleast 15 packets. Store it in a file Read it from this file and sort the packet in SIP / SPORT order and store it in another file. Read from the new file and display as a packet content By this you would be exposed to file access Memory storage while sorting Link list while sorting Visualization

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    We have a Dynamics application listed in the BlackBerry Marketplace. We need to test that whitelisting the on premise app server that this dynamics mobile app should connect to allows access to this connection, and removing from the whitelist restricts access. To do this we need to setup a demo environment using the cloud UEM and a AWS hosted environment

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    ...SSL, Predictive dialing, IVR, Incoming calls, and potentially GSM hardware upgrade (need to evaluate solution) with Asterisk expert. - Server health - IP phone connection - SIP line - Bandwidth and latency of system to network - Connectivity Other task related to asterisk server that will be needed in future: We need to have a setup, where we are

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    I have 4 analog pstn lines, connected to grandstream gateway then to LAN, FreePBX call center installed in the same LAN, brand new MikroTIK router to...pstn lines, connected to grandstream gateway then to LAN, FreePBX call center installed in the same LAN, brand new MikroTIK router to be configured as PPPoE client and allow SIP traffic through network

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    As extension to our existing SIP Android App, we need to add support for P2P-SIP, so end user can decide to use SIP or P2P-SIP.

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    I use [login to view URL] for SIP trunking, but they also offer an API for SMS and MMS, where we can use our business phone number to send and receive such messages. I would like to develop a mobile app with user and password privileges for my clients. Here is documentation on the API. [login to view URL]

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    Trophy icon Logo drawn Bitti left

    Nonprofit group needing a simple logo. Needing something similar to what’s attached, but needing the hair to be either a big Afro or long locks instead as well. The group is called Sip & Sage. PLEASE make hair into afro or locks....also please make skin brown.

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    Garantili
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    27 girdi
    VoIP Leads Bitti left

    We are looking for business to business B2B telesales agents to generate qualified leads of companies interes...features as part of our package: Hosted PBX Call Recording Call Monitoring Real Time centre reporting (call costs, current spend, agent dials etc...) Voice-mail to email SIP Trunk Support (Remote and customer service) Number porting.

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    We have Multiple Cisco AS5400 Universal Gateway which need to be configured purely as SIP Media Gateway to process the RTP. This Cisco AS5400 won't interface with any PSTN E1/T1. It would be controlled and communicated to Media Gateway Controller via MGCP protocol. The following need to configured in-order to communicate with Media Gateway Controller

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