I am looki...connections (Upstream Providers) and 1 x PABX / Opensips / downstream. Initial network configuration is completed. Configuration is required for the above + basic call routing and SIP headers. with the requirement for a basic configuration document (outlining works completed) Initial configuration could lead to additional future works.
i have set up a new freepbx server it is online but i cannot get the SIP registration to go through and actually connect to the provider this is not a big project i am missing a setting somewhere and i need someone to connect to my server via team viewer and fix it for me
Need the Integration of Yeaster S100 with Zoho CRM for 32 Sip Accounts and 10 Lines
I have a 3CX server hosted on Azure with a public IP address. However, the client IP phones are behind NAT and common NAT traversal techniques such as STUN and TURN are cannot be used. I am configuring an outbound SIP proxy server using Kamailio. There is no database or authentication required. The Kamailio server should perform the following functions:
Looking to have a application that registers to a sip server based on sip 2.0 and or direct ip calling have the ability to make calls. open source can be used. up to 10 favorite contacts can be created in tile format. does not need to incorportate with native android phone contacts(for now) each contact(favorite should have a place to put in rtsp string
...will bring voice to text-based chatbots. The Gateway accepts SIP VoIP and then communicates with a chatbot over a JSON/REST interface. The Gateway also communicates with Google over gRPC for Speech to Text and Amazon Poly over JSON/REST for Text to Speech. Interfaces SIP Interface - SIP signalling interface. RTP Interface - G.711 and G.722 RTP media
I would like to use the uvc camera on my linphone app to make video connections. If the ...would like to use the uvc camera on my linphone app to make video connections. If the linphone app gives you the source that is normally built, register it according to the sip information I provided and make the video connection possible through the uvc camera.
...using Technology such as NODE.js/ React native app and react native web When building the app like whatsapp we need to make sure that all in network calls stay on the app server , we a place on the admin to add all the millions of numbers we have in our network now and the ability to keep adding to the list very easily. Any calls that are going
We need a solution in asterisk that help us block incoming calls with the same numbers (robot calls). We are star...if any new call is coming in, then look into the Database to see if the incoming number is in the database, and if it is, then just block the call and sent a "486 Busy Here" Sip response, otherwise asterisk can place the call as usual.
... Its all about Fun Art, not Fine Art! And remember, we are a BYO studio, so bring along your favourite bottle or two of liquid creativity to enjoy while you paint and sip with us on the day. You supply the wine and we’ll provide all the art materials, glassware and good vibes. Leave the rest to us! This is not your typical painting class
...not the best with Asterisk and I am unsure how the control panel would communicate with Asterisk. This is what we need the control to do: - Add a SIP Trunk Channel (For PSTN connectivity) - Remove a SIP Trunk Channel - Add Telephone Numbers to the database which can be used by extensions for outbound and inbound calls - Remove telephone numbers - Create
...can be achieved with a call conference or call bridging. when dialing user (A) calls user (B) they are put into a call conference or bridge. if the dialer(user a) hangs up(sip 487) before user b answers, a disconnects but the conference / bridge channel stays open until b hangs up or for a max of 30 seconds cli needs to be passed from a to be etc