Existing jsSIP dialer needs guru troubleshooter to help resolve bug. Upon a successful connected call from web app through fpbx to sip trunk using webrtc jsSIP. We have a lag upon connection where no sound between caller and called. The lag/no sound lasts 5 seconds usually then both parties on call can hear each other. We require all work be
We have outbound dialer, with FPBX to SIP trunk working. Just small delay on some #'s, not all calls, when connects until each person can hear each other. Work MUST be done via AnyDesk or TeamViewer. Here is where the delay occurs in CLI: (the lag is between these two lines) Sequence: 1. Line Posts in CLI: 0xb7503018 -- Probation passed -
...install FreeSwitch on my server and install ASTTP on that. then doing this configs: 1- Activate G729 Codec on that 2- Change transfer protocol to UDP 3- activate just SIP protocol on server. (disable another protocols) 4- Enable Video Call feature on freeswich if you not robot please add "Swh" in your bid. Thanks for your attention
I am using Express Talk VoIP Softphone in my Windows PC (WIN7 OS)But the Sip is not connecting properly all time Need Guidence to connect quicky all time to sip account
...have a custom andriod voip client done using linphone. the job was not 100% completed. I need to: 1- fix the logo and 2- turn off access to settings , 3- turn off the sip address when dialing a number, and 4- make the app recognize only the last 5 digits of the phone number. 5- Publish to google play. I only have the app no source code
...CISCO VG224 en SIP contra un Server Asterisk. Actualmente lo tenemos configurado pero con algunos problemas como que solo algunos puertos se registran y otro no. /////////////////////////////////////////////////////////////////////////////// We are looking for someone who has the ability to configure a CISCO VG224 gateway using SIP with an Asterisk
Montar un PBX en Amazon. Ramales internos líneas SIP Salas de conferencias telefónicas Menú inicial Contestadoras para cada ramal I need to set up a FreePBX in an Amazon EC2 instance Set SIP lines. Internal extensions for my team (13) I need conference call rooms An initial menu Voice mail boxes por each extension
The main part of a project is a creation of software which will help to manage GoIP equipment (Hybertone manufacturer - gateways and sim banks). All the features which will be included should be available in the interface - Some basic features are: 1:Simulation of Human behavior -Generation of the flow of incoming calls -The list of 'preferred'
...develop some software for me. I would like this software to be developed . I need help standing up the backend and code for a video conferencing functionality. Would need to be able to host up to 35 users on a single meeting and many meetings concurrently. I am open to the use of function Technology like web RTC or turn servers/sip. Please provide
I need someone to configure a SNOM 710 phone so that the openvpn will communicate with my openvpn server. My openvpn server is working a...SNOM 710 phone so that the openvpn will communicate with my openvpn server. My openvpn server is working and I have other devices connecting remotely. If you can config the sip settings that is an added bonus.
i need a prank calling app for my google play account i need the project to be done in 4 hours
We just had a fresh install of Voipswitch with modules, need someone to setup and configure software. - Payment processor setup Strip and PayPal (Customer should be able to buy credit online or by prepaid calling cards) - Setup Onlineshop module (credit card payments, plans, subscriptions, auto recharge etc.) - Set up gateways - Create tariffs
...I need a FreeSWITCH (FS) configuration for the following functionality: 1. Interconnect internal endpoints 2. Link with an existing Asterisk PBX 3. Link with an external SIP trunk for incoming and outgoing calls 4. Configure basic CDR, voicemail, IVR and conference 5. Configure presence/BNL 6. Configure chat on Linphone endpoints (optional)
Need a developer that has a PPT (Walkie Talkie ) application that we can embbed into other appllications. The PTT platform must work with a Sip Server like OpenSip or Asterisk. Users can connect to server via any Mobile Internet, Wifi Internet or via Bluetooth hotspots that will be setup for users connection. This project is not a Design from scratch
...post. I have a dedicated Linux box with Ubuntu [url removed, login to view] installed on it. Have also installed Asterisk but have not configured it. I would be looking for someone to setup the SIP information and install a soft phone and get it to dial out. Furthermore, to protect my PBX server I would want whatever possible basic safeguards to be in place (ie. changing
...products involved are needed. Inbound calls will be delivered using VOIP so a computer with a SIP client such as X-Lite or a mobile phone with a SIP client such as SoftPhone will be required. If you do not have previous knowledge of working with SIP phones we can assist you in every way. A reasonably fast internet connection is also required to
I have a custom andriod voip client done using linphone. the job was not 100% completed. I need to fix the logo and turn off settings , turn off the sip address when dialing a number, and make the app recognize only the last 5 digits of the phone number. I only have the app no source code.
...need a intercom software which is run on Orange pi zero model embedded [url removed, login to view] Ui [url removed, login to view] runs on gnu/linux as a service and has some hardware button handlers for PTT,change room,mute. Software must be P2P serverless or use SIP or mumble voip [url removed, login to view] least 9 user connection supported. We need just SIP ...