Asterisk sip opensip jobs
We're looking to work with someone to develop a VoIP / SIP client, initially for iOS (Apple Mobile) but with scope to increase this to Android and desktop should that initial work be a success. It will need to support the following: - Taking login details from a user with the realm/server box locked to a specific pattern (regex) - Logging the user in to their extension and storing details on phone, implementing Touch/FaceID for unlocking after X period - Backgrounding so that when the app is not front and centre, it retains a connection to the server for any incoming calls - Native integration so that calls are shown (optionally) within the call history on the phone - DND Status so the user can set their extension to reject calls but where they can remain logged in - AMI indic...
We are Intraden Systems, presently we offer web design solutions and I am looking for a resource who is export in configuring SIP and Twilio for VOIP demands. In addition, we would also need someone who can help config whatsapp chatbot and IVR
Hi Iurii M., I noticed your profile and would like to offer you my project. We can discuss any details over chat. I am interested in asterisk based multi tenant PBX.
We are a communications company in south africa. We are looking at migrating our hosted services from asterisk to fusionpbx. We require assistance in configuration and require training so we can configure future deployments.
Hi NetworkLab, I noticed your profile and would like to offer you my project. We are a telecoms company in South Africa looking to migrate our hosted platform from asterisk to fusionpbx. I am just stuck in NAT-HELL and am in need of an expert in not only solving my NAT issues but also training me on future deployments
We need to convert a volume of Cisco IP Phone from SCCP to SIP, mainly the model are 7911 and 7940 We search someone able to quickly provide a working configuration and step by step to be able able to convert this phone to a sip firmware Scope of works: Using a TFTPD64 server, Prepare a TFTP Server to deploy the firmware files to the phone Configure file Configure Configure file for one test cisco phone, we have 800 phone, each phone need to have this own file, but for you works, we only need to provide one demo SEP... file, each phone need to registered in a session border controller, in the sip file, all the configuration need to be provided (extension, calling name, sip registrar ip) quick step by step guide
Need to configure a local Free PBX using a sip trunk and create Backup to install in second FreePBX for emegency
Hi I need to build a softphone to connect to common pbx systems like issabel, freepbx and asterisk . The softphone app will work on all platforms desktop and mobile . Windows MAC LINUX IOS Android Source code delivering is a must . Webrtc technology is required.
We need urgent PTT Solution like Zello and EVO We need both IOS & Android Solution. You may use SIP /VOIP Open source Server. Please NO WebRTC. If you already developed a project similar just offer this bid.. and show your sample PTT project.
Kindly request Please read deeply before bid. Integrate or create application for the voice calling and SMS Messaging for Zoiper. for Windows. The requirement is to replace SIP voice calling and SMS gateway (already included in Zoiper) to make the personal mobile to be the gateway for voice calling and SMS only (no need for other features) by connecting the device by USB cable or any other way (I think it will be easier by connecting the mobile to PC by Your Phone App by Microsoft ( Your Phone App URL: :overviewtab This project is very simple and easy when you use Zoiper SDK 2.0. Please visit this link: From this link you can read all the details about SDK. You can download:
I want to make java sip call for desktop app. Here is github url for my project. Will provide sip user creds and server info. If you can finish with my requirements, contact me.
I want to make java sip call for desktop app. Here is github url for my project. Will provide sip user creds and server info. If you can finish with my requirements, contact me.
This task is straight forward. I need a iOS linphone app that recieve VoIP push and work in background. Only bid if you have done this before. No time waster. I will need to see evidence you have done this or we start with a proof of concept. Open to other iOS mobile sip client that can work with freeswitch and recieve push. I will provide server detail - i use fusionpbx
Hello, I want a xamarin application to receive and make VOIP calls, but I don't want to use paid libraries.
I want hire a C developer to convert some of our interpreted LUA scripts to compiled C or C++ modules. The current modules have close to 1000 lines. The copyright will belong to our company, but the name of the author will appear in the source code. Demonstrate previous knowledge of FreeSwitch and C/C++ development is mandatory. We will eventually build a similar module for Asterisk.
...mutual fund investment app which would be similar to scripbox/orowealth kind of app in terms of capability. The key functionalities needed are- 1. Investor onboarding, Profile/Settings/Notifications/Refer,share with friends 2. Investor KYC 3. Bank account registration and validation 4. Auto debit authrization for SIP transactions 5. Investor's risk tolerance check through set of questionnaire 6. Enable all kind of transactions in mutual fund schemes- buy/redeem/switch/stp/swp/SIP 7. Displaying schemes with its latest NAV, updating daily NAV of schemes 8. Represent tabular & Graphical performance of schemes across different time period like YTD, 1yr, 3yr, 5yr etc, 9. Investor's portfolio report, portfolio overall performance (gain/loss). Need intuitive desi...
I need android app developed with react native. The functionality will similar to scripbox, Orowealth offering - 1. Investor's risk taking tolerance through set of questionaire 2. Enable all kind of transactions in mutual fund schemes- buy/redeem/switch/stp/swp/SIP 3. Displaying daily NAV 4. Tabular & Graphical performance of schemes across different time period like YTD, 1yr, 3yr, 5yr etc, 5. Investor's portfolio report, portfolio overall performance (gain/loss). 6. KYC document submission Interface with the AMC's will be through API which we will provide. My timeline for completion is 2 to 3 weeks.
I am trying to migrate an old Asterisk/PHP/MySQL billing server data to a new Open Source platform with similar functions, and Asterisk based. I found 1 candidate: MAGNUSBILLING. When I tested Magnusbilling, I found some missing things I need: 1) This platform does not offer virtual FAX. 2) I see there is a STRIPE option under “payment Method” menu, however, is not functional since it uses an old version of its API. 3) Some other customizations within the PHP code like the LOGIN part (Yii framewok), Calling plans, etc. Experience with this platforms is a PLUS. I will provide a document with the detailed points I need to be added to the platform, for a proper quote of this job. Thanks
I look for professionals who can integrate my VOIP solution with my Auction system. The main objective is: Have an area in my current system where those responsible for care: 1) Be warned that there is a connection 2) The system recognizes through the telephone number that the person is already registered in the system and displays the data on screen 3) If the Person is not registered, he must inform on screen 4) If the registration exists and the person responsible for the service is approved, it will be possible to access another screen and assign the bid to the person in the desired lot. The Auctions system uses PHP and Mysql and it is complete the intention is that this tool is developed in a way that allows my team to make necessary updates.
We need to build our own SIP/ VoIP/ tfn server to setup the toll free numbers to sale the client.
Hi, I am looking for someone who can Install / Configure Asterisk based Free PBX 1. I want to use GSM gateway 2. Click to call API 3. Condition based IVR
expert in vo ip with experience in pbx control asterisk centos php angular css page development
I am looking to hire someone whom can prepare instructions for configuration the EdgeMarc 4806. We have many units to deploy and are looking for someone who can prepare a configuration guide. The guide must cover steps to configure the SIP-UA for the FXS ports and connect them to Hosted PBX service, as well as how to configure the T1 interface as a Network-side PRI interface for a fractional (8-channel) PRI with a phone system. The PRI should be connected to a remote SIP Trunk and include how to configure DID mapping/translations.
Hello Guys, I have small issue the asterisk is rejecting call... due to Tel URI .... INVITE sip:+9714602XXXX@xxxxxxxxx SIP/2.0 Via: SIP/2.0/UDP ;branch=z9hG4bK34zzdzv9zdt3bthz6bzh6d6zc;Role=3;Hpt=8e42_36;TRC=ffffffff-ffffffff Call-ID: asbcqd3893uim9dfu8m1bxokwmf3duu8tf3m@ From: <tel:0559625523;noa=national;srvattri=national>;tag=kx1t8kw1 To: <sip:046022600@xxxxxxxx;user=phone> can you help about this
The Project requires a developer with knowledge in the VoIP industry and utilizing Vue and node.js the web App will deal with carrier APIs ( Like Bandwidth ) grab the services, like SIP TRUNK, Number ( DID ) selling, SMS Selling, Voice and e911 reselling) the user will subscribe to our portal, select the product ( e.g Buy Phone Number ( DID ) will go through filtering by selecting Country, City, State, area code, and add to cart the selected number from the API response. - Allowing the user to create a SIP connection where he will be able to route the bought phone number, Voice, SMS, and so on. more details will be shared with the winner applicant .
I am looking to develop API for Grand stream / Open Vox IP PBX system using the Asterisk Manager Interface (AMI) or using the API document which is given by the Grand Stream (find enclosed) I need the following API Click to Call API Auto call forwarding to extension based on the mobile number Getting call logs data - CDR / CLI
migrated freepbx server and now gateway sip wont register. 4 pots connect to gateway then to freepbx based server. thanks.
Hi. I need someone to manipulate the receivers response to the caller according to the time of answering the call. My goal is. If a call is received within "5" second then the receiver will be considered offline but the caller will still listen the ringing tone. So the caller will think the phone is ringing when it's actually unreachable. Budget $50 Thank you.
Hi Friends, I need professional mobile had experienced working with OTT APP (have features call look like messenger/Viber/Whatapp) , can make call and wakeup call anytime when receive push notification of server. We use SIP Server ( Freeswitch) to develop system. Each time have call from user A -> user B, system will push notification to user B User B install mobile app ( Android/IOS) and wakeup and receive call. We can make this flow work with some device but not all device work smooth, some device can not wakeup ( android, ios too) i think the app like that ( voip ) need some special skill and tech to implement. My requirement: please see the attact file
NEED HELP TO SETUP VOIP SERVER - RUNS ASTERISK AND ALSO ONGOING SUPPORT
I do have VOS3000 2.1.2.4 Security Authentication Panel installed on my server, I need help to configure it so that it will automatically block the SIP IP Address that has 3 failed login attempt. Also the Admin should be able to unblock the IP Address once the it has been verified that IP Address in question is not a hacker!
Hi. I need someone to manipulate the receivers response to the caller according to the time of answering the call. My goal is. If a call is received within "5" second then the receiver will be considered offline but the caller will still listen the ringing tone. So the caller will think the phone is ringing when it's actually unreachable. Thank you.
need a logo for our new brand "Wow Tea Wow" which is going to be different tea products brand. I have attached a sample logo for your reference. brand name : " Wow Tea Wow" tag line : A healthy sip!
We are seeking someone that Speaks English and Spanish. Sends Emails and Reaches out to talk to Bars and Restaurant Owners to Schedule appointments to demo a product.
I am looking for softphone for Windows just like linphone. Let me know if you have any questions.
...Implementation • Managing and Ensuring Security • Ongoing Monitoring of Systems and Processes • Developing Documentation Required Skills and knowledge: • Source Code Management and Configuration Management Tools: GIT and CVS • Continuous Integration • Infrastructure Management and Monitoring: Nagios, Solar Winds, and NetFlow • Programming: PHP, Perl, Bash, Microsoft SQL, and MySQL • VoIP: SIP protocol, Asterisk, and Kamailio • Operating Systems: Windows fundamentals, Linux fundamentals, CentOS, and RPM management • Virtualization and Containerization • Good Communications Skills and Conversational English ABOUT DLS: DLS Internet is a privately owned Voice Over IP and Information Technology Services provider wit...
I am looking for a mobile SIP app for ios and android tailor made for my company
Hi Sunil C., I noticed your profile and would like to offer you my project. I have React Native project, and i stuck in call functionallity, Hopefully you can help me use SIP, the plan is to use pjsip as voip, and i already have a sip server We can discuss any details over chat.
Hello, I'm looking for an expert in AstLinux VoIP. I need to build a testing system and have a expert advice & tips on SIP Providers and other topics. This project needs documentation, this means you will have to document all the configuration steps. I want to use AstLinux because it's stable and doesn't have a lot of moving parts. I'm interested in these two SIP providers but open to better suggestions: We can host this system on Cloud or on a PBX on prem. It's your choice. ----------------------------- Features I need for this project: - Auto-attendant based on day and night time - Redirect extensions to a different phone number, call forwarding - Request a phone call back - Background music - Conference call
I am after a module for Asterisk or any other open source PBX that will allow me to remotely request and automated telephone call to deliver an automated pre-recorded message. For example http://192.168.1.1/call=01623661624&message=1 This will insert into a database, the number with a status of pending. It will then call the number above and play the pre-recorded message 1 (as i plan to have multiple messages). The receiver can then press 1 to confirm (which will update the status on the database record too "Confirmed") The receiver can also press 2 (which will divert the call to another number and mark the status on the database record to "Pending") The module/development will need to be able to tell if the customer has answer or if its a voicema...
Hi Karuna S., I noticed your profile and would like to offer you my project. I am developing an online CRM system with webRTC SIP embedded to make phone calls. The problem I'm facing is to keep the active SIP session "alive" between pages as the call gets cut when refreshing the browser or navigating to other pages. From what I've read is I will need to port the current webRTC to a PWA.
please contact me if you know how to make change on the sip Header... the provider needs to receive the calling number @ Domain instead of the host name.
I have small issue the asterisk is rejecting call... due to Tel URI .... INVITE sip:+9714602XXXX@xxxxxxxxx SIP/2.0 Via: SIP/2.0/UDP ;branch=z9hG4bK34zzdzv9zdt3bthz6bzh6d6zc;Role=3;Hpt=8e42_36;TRC=ffffffff-ffffffff Call-ID: asbcqd3893uim9dfu8m1bxokwmf3duu8tf3m@ From: <tel:0559625523;noa=national;srvattri=national>;tag=kx1t8kw1 To: <sip:046022600@xxxxxxxx;user=phone> can you help about this
I am looking for an audio , SIP , WEBRTC, EXPERT to build a product like 1000% of features and operations. with some additions. with my own dashboard design and home page layout. You must know sip snd webrtc. This is NOT for first timer. I will NOT pay for you to learn. Apply ONLY if you have done similar and can provide sample link and code. If you don't like to listen and follow instructions DO NOT APPLY. if you are inexperienced and cannot work under pressure DO NOT APPLY.. This is only for serious professionals who are willing to work long term and not try to make money now and disappear. I am in this business 40 years and have heard all. I will provide sample layout of homepage and dashboard to NOT CONTACT ME IF YOU DONT HAVE SAMPLE OF SIMILAR
I need a php to query a mysql database and originate calls for numbers (result from mysql query) using AMI . i use a SIP channel to outbound calls with only 1 simoultaneus call, the MAIN GOAL is to get calls wait until channel available or retry if CONGESTION.
