Softclient sip jobs
I need a new website. I need you to design and build it. It contains SIP, MONTHLY saving, fixed deposit, Loan, profile tracking, complete bank website
I need you to develop some software for me. I would like this software to be developed for Windows using Python.
We search for a very experienced developer, who did already integrations with: - SIP/PBX into vtiger open source Our consutancy requirements are: to discuss deeply about the options of how to - configure a existing vtiger installation (in a remote datacenter like EC2) to work with SIP/PBX - the onsite-phone integration, so that users can have a call from vtiger - and far more interesting, to have a phone call tracking in the contact on vtiger, based on the given calls to the client you job would be: - to answer our questions in a call - to collect our requirements and to write them down in a document If you do a good job here, you will be our preferred candidate for the implementation into our vtiger. Our budget? we do not disclose our budget. We have placed here a wide r...
...full viewing history. Bulk import and export. Blacklist to be kept by client for each account. Start/Send times to be specified by client, Monday to Sunday 00.00 - 24.00; enable to be changed as required. I'm sure if you've read this far you know whats required in order to build this. We want to make use of VoIP/SIP Trunks to send SMS as we were doing previously with another business partner. Each SIP trunk could send approc 12,000 SMS per hour which we want to replicate with this. The SIP trunks must be 'pooled' rather than allocated to each account, this way they are maximised across all clients. Clients to have a 'sending rating' also so that a priority can be applied to SMS in queue making use of the trunks available. Priority 1 - t...
Move Gatekeeper site to Sip Cisco call manager and CME
...hold a small footprint while providing meetings, conference rooms, and webinar (named presenter mode) options. Both services provide apps for android and IOS, and use webrtc as the base platform for delivery. We require a php front end that connects to Jitsi and provides all the features of zoom. Features needed Connection by phone (SIP trunk) This will be a SIP number that terminates on asterisk (freepbx) IVR which will request conference room number and pin based on the information (numeric only) provided by the script. This will require using asterisks API or .call file. Interface Main interface for users should be controlled by Username and password User should be able to create conferences, schedule by time, and so on. Create a free account
...hold a small footprint while providing meetings, conference rooms, and webinar (named presenter mode) options. Both services provide apps for android and IOS, and use webrtc as the base platform for delivery. We require a php front end that connects to Jitsi and provides all the features of zoom. Features needed Connection by phone (SIP trunk) This will be a SIP number that terminates on asterisk (freepbx) IVR which will request conference room number and pin based on the information (numeric only) provided by the script. This will require using asterisks API or .call file. Interface Main interface for users should be controlled by Username and password User should be able to create conferences, schedule by time, and so on. Create a free account
...one-directional from the server to the subscriber placing the call, but it could be two-ways with numerical input only from PhoneA to My Server by server prompt. Once the transaction is done in the server, I want to achieve one of these: 1. My Server returns control to phone dialer (PhoneA) which places the call using the regular phone network (PSTN) or 2. Call is made / connected from my server via SIP solution to PhoneB without using the regular PSTN The target is for the app to be used by millions of subscribers in placing calls. So the solutions have to put that into consideration. The app is gong to be designed and built by me. If you can do it, we can discuss on that as well. I explained the whole operation because I would like to get some insight (or input) on h...
We are looking for a SIP Load Balancer based on extension range. Ex. Extension 2000 -2499 goes to server A - Extension 2500 - 2999 goes to server B and so on. Server A and B are Asterisk Server. The SIP Load balancer must run in a Linux Centos Environment. Documentation on how to change Load Balancing Parameters is required.
Need someone that already work on GOAUTODIAL for some fixes to have working configuration. I have installed it on my server and followed installation guide and inserted credential of my sip account but dialer not working.
We want a freelancer to make a presentation for our concept - Stock SIP. The content for the presentation including brand logo's will be provided by us.
Our company Globetel Consultancy Service is based in turkey. We are looking for a candidate who can help to develop a SIP to viber / whatsapp gateway. The gateway should be able to convert the incoming voice calls over SIP and forward that through viber / whatsapp in order to complete the call to the receiver number. In addition to that it should be able to return the correct call error codes such as "Callee Busy", "Callee Unavailable", "Call connected Successfully" to the SIP backend. The candidate can choose any platform.
hi, i want to set my asterisk server to receive calls from another asterisk server which will then pass it to sip
project is for adding proxy to my sip server for outgoing calls . im not sure how its done or how to do it but i want the ip of the outgoing call to have a proxy ip
I need to use an android cellphone as an asterisk channel/Gateway. This will make p...APK format Full source code Simple manual for compiling and generating the application from source Features : -Route call from SIP to GSM -Convert audio from/to SIP and GSM networks -Able to run on Android 4.0 and up. - Must run on background - Must be very lightweight to run on small memory devices Sip Requirements : - Register on Sip Proxy/Gateway - Receive call authenticated by IP, user/pass or no authentication. - Make calls with or without authentication - Forward DTMF digits using RFC2833 or inband - Use codec G711 and GSM - Be able to use codec G729 - Receive through GSM one 1 simultaneous call and rout it to a predefined SIP client - Be a...
Hello I Want A Professional Windows Desktop application Developer Who able to create sip softphone dailer i want to create a special one with upgrades and fetures and i can help with sample but no open source in the attachments there is image for program sample
Hello , My name is Nathanael , I am the CTO of Tacticall, that's a call center, i want someone to configure twilio sip trunk on my server vicidial to dial multiple region like USA,Haiti,Guyana,ect... Vicidial work perfectly with other trunks , now we would like to switch to twilio sip trunk instead! can you done it in one day?
Hello , My name is Nathanael , I am the CTO of Tacticall, that's a call center, i want someone to configure twilio sip trunk on my server vicidial to dial multiple region like USA,Haiti,Guyana,ect... Vicidial work perfectly with other trunks , now we would like to switch to twilio sip trunk instead! can you done it in one day?
Need to enhance calls, by using open source SIP Servers
Due within 7 days. All components must be completed to excellent/expert level. No grammar mistakes. DUE 3/20 Something Good 2 Eat LLC (Founder Chef Tiffany Swinson) in partnership with Dream North Entertainment (Founder Nychol Lyna) -we work with Sip & Sonder (Inglewood, CA) and they are a vegan restaurant where we consult, cater and facilitate our delivery service through EZ Cater Use somethinggood2eat@ Password GrantWriterLance09 (Twitter @sumthinggud2eat Instagram/Facebook @somethinggood2eat) (Instagram @dreamnorth_ent) 2020: $10,300 Current Revenues Monthly Expenses=$3000 Projected Revenue=$200,000 2019: 47 paid clients/7
Dear Mr. Evigeny I’m looking for an expert to building and install the Signal server sources by Open Whisper Systems on Freebsd 12 system and integrate that server with Android, iOS platforms both white label and customized using signal API app. I plan to be able sell phone numbers, call credits and subscriptions plan for call over SIP protocol managed by ASTPP billing system. I have more details. Let me know if this can be done. Best regards, Lucio Burnett
Required a SIP application for raspberry pi, based on node-red. Node-red dashboard is required to allow user to enter sip server information, name, IP, port, username and password. Sip calls will use onboard 2.5mm Jack and a usb microphone.
Hello, We have a new SIP provider uses RFC 3261, Proxy-to-User Authentication We need assistance setting up outbound calls. See documentation attached. Thank you
we need to use api with our device from asterisk or voipswitch or any way pleas check attach file
I need someone to install vicidial on my vps server and apply SIP configuration for inbound calling
Scope: Install Open source SIP Client Install Open source softphone Integrate into application to properly record calls and texts
Bid if you are familiar with softphone and web app
Are you an asterisk specialist? I pay you as a consultant per hour done to configure and form one of our IT member. Around 5 hours a week will be done. We will chat on skype during you performing your work
Hi I am looking for a developer who can develop a sip mobile dialer for android and iPhone
I am looking for Sip mobile dialer For Android and iOs Tunneling the Sip Mobile Dialer With Tunneling Server where it has to anti block in blocked countries it has to work perfectly .
Looking for experienced Linux Administrator with great TR069 knowledge. The goal is to have all our routers, sip phones and ATAs configured remotely using the GenieACS platform and the project is completed once the devices are configured, monitored and are provided the provisioning templates. For all devices, we need to be able to plan single & mass firmware upgrades/downgrades, CPE replacement, CPE reboot and have a CPE Factory reset option. Tasks: 1.- Define hardware specifications for a VM, linux distro and version and we'll provide corresponding ssh credentials. 2.- Setup GenieACS 3.- Generate a SSL and install it (let's encrypt) 4.- Create initial config of GenieACS 5.- Create provisioning templates for Mikrotik(2 models), SmartRG devices (3 models), Yealink ph...
I want to learn how to set dial plan and sip account in a server vos3000 or VoipSwitch !
Hi, I want to learn how to make server softwitch for call termination ( exactly i want to know how to set dial plan and sip account in server). ( vos3000 .......) Feel free to contact me. Regards.
...hardware unit running a SIP Client that can connect to RTSP or http camera streams. The idea is to call into the sip client, the client would then access the video /audio stream from the camera and send it to PBX connected to a Video phone. I have this working with video using a client called Baresip, v4l2loopback and FFMPEG. Right now I trying to figure out how to send the sound to a sound loopback at the same time send the received video/ audio from the video phones to a stream. The inbound now comes to the local sound and video port but II would like to output it as a RTSP or HTTP instead. So basically its a SIP to RTSP and RTSP to SIP converter. Once we get it working correctly, then create a web portal for configuration. Transcoding options would me...
I am looking for SfB experience. We have : Instant Messaging/Chat Desktop/PPT Sharing File sharing Audioconferencing Dial-in Audioconferencing(Masergy/Toll free) Voice Mail With Exchange SBC - SIP trunk/PSTN Exchange 2013 integrated for Voice mail, Resource booking Open External Federation Contact me, if you have experience back end troubleshooting experience.
Maintain customer details Login for clients to see there investment details Login for lead providers to see there leads details...us Login for Admin to see complete details Maintaining Customer's Investment details Sending Renewal notice to them Sending seasonal greetings to all Maintaining Mutual Portfolio with folio update option Allowing Mutual Funds Transactions to client Simple website to display all of my products with update features so few products can be updated from our end Few useful tools [calculators] [i.e. SIP/ SWP returns. Loan calc etc..] Domain we will purchase separately to keep it in our control Note: Few more things to be add which can be discussed before finalizing We deal with all types of investment products: INSURANCE, INVESTMENT, REAL ESTATE, LOANS...
Maintain customer details Login for clients to see there investment details Login for lead providers to see there leads details...us Login for Admin to see complete details Maintaining Customer's Investment details Sending Renewal notice to them Sending seasonal greetings to all Maintaining Mutual Portfolio with folio update option Allowing Mutual Funds Transactions to client Simple website to display all of my products with update features so few products can be updated from our end Few useful tools [calculators] [i.e. SIP/ SWP returns. Loan calc etc..] Domain we will purchase separately to keep it in our control Note: Few more things to be add which can be discussed before finalizing We deal with all types of investment products: INSURANCE, INVESTMENT, REAL ESTATE, LOANS...
We are looking for a virtual phone solution to be accessed with iPhone that can have an auto attendant (option 1: Sales and 2: Support). I need to build our own solution, especially we don't receive many calls. I expect that the solution would be in the cloud. The system will have a US number. The developer may utilize technologies and system components like soft PBX, SIP, Asterisk, DID, etc. I need a stable system that would have a cheap running cost. Please give me a plan of what you will do and a fixed rate. Example of optional technologies: Asterisk, FreeSwitch, Kazoo VoIP Cluster, Kamailio, callweaver (faxing), ASTPP, Auto Dialer, opensips, kamailio, Call Center, Elastix, Vicidial, VoIP, Ringless VoiceMail, 3CX, FusionPBX, Issabel, OSDial, GoAutodial, PABX
I am running a FreePBX instance of asterisk and I was hacked, lots of outbound calls, I think I blacklisted the IP that hijacked an extension and made the calls. Now I see a lot of attempt in my CDR Report with "Congestion" under app and s[from-sip-external] under destination. System number keeps changing and the CallerIDs are all 4 digits. Need someone reliable that I can use for this PBX and other Asterisk enhancements and improvements.
...- Existing SIP account successfully registered on MicroSIP application on Windows machine; thus the SIP account works 100% fine using MicroSIP. > By this I mean: I received SIP information from my phone company. Using this SIP information I can place outgoing and incoming calls using MicroSip. - The SIP account is NOT blocked anyhow (like for example certain domains or whatever) - SIP account contains an “@ sign” in the username - SIP account needs to registered using UDP only I'm looking for a fixed price. Installation of the SIP account as trunk in my freepbx should be done via remote desktop (teamviewer) Note: I’m NOT asking to install a trunk, just a working SIP account in the “trunk se...
We use Asterisk PBX along with Free PBX using SIP Protocol. We would like a developer to integrate Whats App calls to our PBX System. Either directly by configuring the Gateway on the PBX itself or as a trunk which is piped from an Android Phone (in this case the phone will be ON and connected on WiFi) to receive and make calls and pipe it as a SIP trunk to our PBX. Requirements: Inbound and Outbound Calling. Caller ID must be passed on incoming calls. Ability to have more than one Whats App number to work simultaneously
We're looking to build a SIP based softphone with react-native using and react-native-webrtc. Inbound calls should be able to utilize PushKit and CallKit on iOS due to new restrictions in iOS 13, the backend for push notifications is already available just need to build out the frontend. Needs to be a full functioning phone and phone screen should be able to hold a call, transfer a call, conference, mute, activate speaker phone, etc. We are not married to react-native-webrtc but felt it was the most logical way to support media cross platform. If there is a better way we are open. React-native React-native-webrtc react-native-websockets sip over websockets webrtc Skills Required Javascript Mobile App Development iPhone
Are you an asterisk specialist? I pay you as a consultant per hour done to configure and form one of our IT member. Around 5 hours a week will be done. We will chat on skype during you performing your work
...because we live in a very rural area. We've come up with the idea to be able to send them a link to Google maps directly to the phone from the phone call itself. This way we are hoping that the other was received the link click on it, Confirm they know where they are going, and we won't have to deal with repeat calls later on as they get closer and not being able to find a spot. We use the Twilio SIP service to drive our PBX - and those same number are available on the messaging side of we will be looking for is to make sure that when call or a calls employee be, and employee be respond with a message, the origin of the message (in terms of number remains intact) A few other notes: ------------------------------------------------ 1. The link is not specific to the c...
We have freepbx running in production but would like to have the server optimized. Here are few things for a start that need to be done: 1) Securing freepbx server - fail2ban, iptables etc ipv4 and ipv6 2) ensuing that secure channel is used - iax, tls etc 3) ensure that the SIP setup is optimal 4) ensure that codecs and sip settings are right 5) configuring DID for extensions 6) Ensuring that backup trunk is setup as failsafe for primary trunk 7) Review and provide other feedback based on the configuration
Need help with Freeswitch troubleshooting realtime configuration and various issues. Current configuration includes Debian OS, Freeswitch using LUA for subscriber configuration, SIP profile configuration and dial plan all stored in database for realtime functionality. Also python used for socket connectivity into Freeswitch for event drive functions. Must have knowledge with Freeswitch, LUA, mySql, Python all working together. Successful candidate will be called upon to assist with design, troubleshooting and best practices at various non preset times. Depending on current issues and request the hours committed will vary and this will be on an ongoing basis.
Need help with Freeswitch troubleshooting realtime configuration and various issues. Current configuration includes Debian OS, Freeswitch using LUA for subscriber configuration, SIP profile configuration and dial plan all stored in database for realtime functionality. Also python used for socket connectivity into Freeswitch for event drive functions. Must have knowledge with Freeswitch, LUA, mySql, Python all working together. Successful candidate will be called upon to assist with design, troubleshooting and best practices at various non preset times. Depending on current issues and request the hours committed will vary and this will be on an ongoing basis.
