Wengo sipcông việc

Mình cần hỗ trợ config Elastix, inbound, outbound và sip trunk

Hi, I need support setting up a basic on premise (RasPBX) Asteriks installation. I have the following devices: - Raspberry PI 4 with RasPBX installed - 2N technolgies Doorbell with Sip Support (works already and is registered to sipgate) - Several Devices with Sip Softphone installed - Home Assistant on several devices (I want to connect via webRTC (Browser) to the Asteriks) So I need: - ...

looking for customizing Linphone for android, Need "AUTO LOGIN WITHOUT USERNAME AND PASSWORD" (username and password from a2billing table), PUSH NOTIFICATION SHOULD WORK TO RECEIVE INCOMING CALLS AND FCM MESSAGES, SMS OPTION SHOULD WORK WITH ASTERISK SIP [đăng nhập để xem URL] SOME OPTIONS OF LINPHONE to simplify Linphone interface. we have a sample source code and customized Linphone di...

We are currently facing VoIP issue on Mikrotik RB750G3. The incoming & outgoing calls can be done but no voice from both sides come out. The PBX system is NEC SL2100 & the SIP Phones are Flying Voice IP622CP.

need a cloud pbx that can be connected to my grandstream ip phone. I need to be able to get calls in one area and then calls can be rerouted via extensions to other phones. This will include softphone solution that can act as a call centre. Multiple lines must be housed. This will not include SIP. It should be able to run all on the cloud and be housed on our server.

I need a cloud pbx that can be connected to my grandstream ip phone. I need to be able to get calls in one area and then calls can be rerouted via extensions to other phones. This will include softphone solution that can act as a call centre. Multiple lines must be housed. This will not include SIP. It should be able to run all on the cloud and be housed on our server.

looking for customizing Linphone for iPhone and android, Need "AUTO LOGIN WITHOUT USERNAME AND PASSWORD" (username and password from a2billing table), PUSH NOTIFICATION SHOULD WORK TO RECEIVE INCOMING CALLS AND FCM MESSAGES, SMS OPTION SHOULD WORK WITH ASTERISK SIP [đăng nhập để xem URL] SOME OPTIONS OF LINPHONE to simplefy linphone interface. we have a sample source code and customized ...

Hi Erewin, I noticed your profile and would like to see if you are interested in helping with my project. I require FreeSwitch / Asterisk expertise to help debug a SIP trunk issue. There is further work with push notification later. We can discuss any details over chat.

I have an IVR with 3 avaya 9608 sip phones. I would like: - To have a call queue so when people call during business hours it puts them on hold until we can answer the call - To be able to transfer calls to another phone - make the voice recognition work. Right now it doesnt work and I dont know why. For example say "two" rather than press 2.

i need checker sip protocol with user, pass and ip list

i need someone to teach me how to use the magnusbilling 1. the DIDs all menu train. 2. set Plans, Tariffs, the Prefixes, if it's short number like 1250, can not call out. 3. set sound record, and now some line didn't have the sound record issues 4. the Routes all submenus 5. Voice Broadcasting 6. time in the system not right. 7. if we want to made a call, create sip account, the ta...

We are migrating our VoiP services from hardware to cloud-based servers. Currently running Class 4 and Class 5 hardware softswitches. All authentication is done via our existing JeraSoft billing server. Have set up a Freeswitch with FusionPBX. The aim of this softswitch is to migrate over the existing Class 5 services. Will keep JeraSoft billing for authentication. Have completed basic configura...

I currently have a script --- that runs on a SIP server -- to DIP my provider for the CNAM for the call. I am changing providers and only need the script modified to use the new provider. The new provider's directions are below -- [đăng nhập để xem URL] Attached is the existing script that needs modified.

The business model is a PlayCafe & Spa, it will be designed with boutique play spaces for children. The play areas will be hands on, most spaces will be facilitated by a Playpal Instructor. Some areas will allow independent play and equipment exploration. The business will have an area that feels upscale, and modern for parents; especially for moms to enjoy a chair massage, tan, exercise, glam...

React native VoIP( SIP ) client app needed

93˚ Coffee was founded with the aim of taking coffee to unprecedented standards in India. We are independent coffee roasters based in Delhi NCR. We give value to innovation and creating products that bring about change. Our vision stems from our love for coffee and inspiring others to bring a change to the world. Traveling the world, meeting new people, and hearing their stories made us realize th...

Hi Aleksandar, I have read about the project "Create SIP Server with WebRTC"..and seems exactly what I need. please could you tell me more? thanks

I am planning an startup company and looking for Freelancer who can help me build an operational API for my Video conferencing solution. Our requirements are as under: Sr.# Module Name 1 One to One Video Conferencing (using Webrtc/Mobilertc/??) 2 One to Many Video Conferencing (Group Chat and Webinar) 3 Record, Video meeting (before, during and after the call) Or Snapshot to the local drive...

Please recreate the attached logo, I took a screenshot of it, I need it to be designed and sent in good quality please. Once completed, Please use the logo to create labels for each of the following variants. Note that you will need to remove the word "Juice" and replace it with "Dessert" or "Fruit" etc. Dimensions of Label: Height=53mm Length=130mm -Dr. Dessert ...

Olá, Preciso de um profissional para desenvolver um Softphone SIP que eu consiga ler QR Code do meu PABX (Asterisk) e configure automaticamente o ramal no softphone sem a necessidade de configuração manual. Este softphone tem que funcionar para IOS e Android.

The PJSIP is an open source SIP stack , we want to compile it for Android NDK The PJSIP is developed in C and support all SIP features as per the RFC standards. SIP, SDP, RTP, STUN, TURN, and ICE. It combines signalling protocol (SIP) with multimedia framework (WebRTC) and NAT traversal functionality. Please bid only if you have good hands on experience in compiling SDK and good knowledge on VO...

Hi. I need opensips expert for hiding or modifying the RTP IP before reaching the termination site. Means I need to modify the RTP IP from the sip trunk running on Opensips as a different IP, before it reaches the termination side. So the termination site could not see the actual IP address.

i want developer for my own mutualfund software who make this kind of below report 1. SIP Performance Report Based on XIRR Calculations: which effectively calculates internal rate of returns where there are multiple transactions happening at different times (SIP) 2. Capital Gains Report: Category split of investment in long term and short term based on holding period of investment 3. Total Returns...

I want developer for my own mutualfund software who makes below report 1. SIP Performance Report Based on XIRR Calculations: which effectively calculates internal rate of returns where there are multiple transactions happening at different times (SIP) 2. Capital Gains Report: Category split of investment in long term and short term based on holding period of investment 3. Total Returns Till date E...

i want frelancer for my mutualfund software who can import data from cams & karvy and make report for client as valuation report, capital gain report & sip report ,Aum report any many more report any one who can work before this type of report pls contact us

Design me a logo for cafe brand Shake & Sip

i have a problem with my 3cx phone system, I have added my IP and get way and my sip server ip but the numbers still not work

Develop a custom WHMCS module to integrate with PortaOne VOIP Billing. The following features needs to be integrated as required: Admin Area: Create, Suspend / Un-suspend, & terminate PortaOne account View Call Logs with Detailed Dialed Information (Report of call history made to view or download) DID/SIP Number Pool (Fetched from PortaOne DID Allocation) Passes client number as CallerID &am...

I have just set up a Raspbx and Freepbx.. I have the extensions set up and working, but need assistance with the inbound and outbound settings/ I am using mynetfone as the provider of the 2 SIP lines I need to connect to the system.

I am looking for a developer to create a code that will allow Acrobits softphone (Groundwire) to send and receive MMS Messaging from a SIP provider and deliver them to FreePBX / Asterisk. For example Joe sends a picture message from his personal cell phone to a SIP, DID which will come through as a picture message received by the Groundwire softphone application on the phone Below is a link to...

CSipSimple or Zoiper Sip application developer required

We need a box label for our product the "Sip-Line." it will need to include our logo, a barcode, and a design. The label will go on the front face of a 4 x 4 x 5 box. As we sell multiple colors of this product we will need the label to reflect those colors (navy, pink, grey, black, bright orange, camoflauge, teal)

Compile zoiper sip application and run it for me.

I have a Grandstream ATA with 2 sip accounts working, but the line sometimes drops or is unavailable, other times when a user is on hold, or is transferred they get disconnected.

Hi, I’m sisca from Grahacomm. Right now, we have a project to connect SIP Trunk with several IP Segments from TELKOM (Telecommunication provider) to Yeastar PBX S300. Kindly inform me if you are interested with this project. Many thanks.

Need 2-3 buttons/functions added & change UI. Website works with app. It allows you to pick & call 5 people to call in the Philippines with unlimited minutes & monthly billing. Also lets you load/top up cell phones worldwide. I'd love calling cards as well. Software for the AUTOREPLY & CARD & PIN INSTANTLY UPON PMT WITH ACCESS NUMBERS & PIN IN [đăng nhập để xem URL] 2 ...

We need a PBX on a Virtual Machine based on Asterisk (freepbx or whatever you think is best). - Create 5 extensions. - Configure sip trunk (with linksys spa300). outgoing pstn / incoming connections . - Configure a GOIP Gateway (1SIM - GOIP) for outgoing / incoming calls. - Configure outgoing calls with international prefixes. - Record calls. - Enable video connection with h264. Thanks,

We are a ISP providing Internet, Network, SIP and Hosted PBX services to business customers. We are looking for somebody to join our Help Desk team to provide Level 1 configuration, diagnostics and support service to our customers. CCNA certification required. Experience with supporting SIP PBX/Telephony and Firewalls desirable. Looking for somebody who is flexible and can work Australian Easter...

build a sip application for me

I need Zoiper sip application expert

Zoiper SIP application developer needed -Ukraine ,Russian and Chinese developers needed

We need someone to have a look at our VIOP system as calls are not going through. It seems as though they are being blocked. Are SIP trunks are registered. We are running a 3CX system on AWS.

[đăng nhập để xem URL] Hello, Above, you will find the link for the solution i'm looking for my vtiger 7.2. The extension allows to automate calls for a marketing prospection for example. An agent of a call center could make more calls efficient thanks to this extension. We work with 3CX for the plateform SIP. Thanks you for your reply, Geoffrey

I need Zoiper SIP application expert . I will do quick hiring

Zoiper SIP application-ONLY experienced SIP application developers apply

Name is Second Sip Coffee We want something smart, simple , trendy and cool.

Hello, I am looking for someone who can do these works for me which are given below: a) Install vicidial on vps b) add voip/sip c) create agents d) create press 1 demo campaign (provide me tutorial how to create that type of campaign) e) upload samples leads

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...

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

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 [đăng nhập để xem URL] file Configure [đăng nhập...