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I am working on a MAUI app that needs to implement outgoing calls. I have looked at multiple solutions, but all that I find is for .NET 4.8 or lower, using Java / C++ bindings. No solutions for .NET 9?...
fARcRY's user avatar
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0 answers
70 views

I'm working on a SIP application in C++ using PJSIP/PJSUA2 inside a Qt project. Basic outgoing calls and registration are working fine, and I can already manipulate or configure things like the Allow-...
2 votes
0 answers
104 views

I am working on an iOS project that uses the PJSIP library for video and audio calls. I have an Objective-C wrapper that accesses the PJSIP C functions directly. Now, I want to get statistics for each ...
Kalpesh's user avatar
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1 answer
120 views

I was hoping to setup a simple IMS call using Kamailio and establish the call between two IMS clients such as Boghe or Linphone. I've tried following this tutorial from Open5gs. The clients register ...
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0 answers
79 views

I’m working with the Linphone SDK on an Android 7.1.1 (API 25) watch. Outgoing SIP calls register and connect successfully. I can hear the other side, but when I speak, all they hear is distorted/...
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0 answers
109 views

I have been trying to generate an auth-url for quite some time now to make an outbound call from my n8n instance using HTTP Request node. But Zadarma wants a signature key for every call made. I ...
0 votes
0 answers
55 views

When I am connecting with the linphone free service (sip.linphone.org), the call is connected and its perfectly working, we can hear the other side and talk each other. But if I do with my own SIP ...
1 vote
1 answer
172 views

I'm trying to set up a SIP infrastructure using OpenSIPS as a load balancer for multiple backend SIP servers. The goal is to route incoming INVITE requests to a backend server and have OpenSIPS manage ...
0 votes
0 answers
129 views

I’m currently setting up a scalable VoIP architecture using FreeSWITCH for media handling and Kamailio as a SIP signaling proxy and load balancer. I'm trying to achieve the following: Distribute SIP ...
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0 answers
49 views

I am using Pjsua2 on C++ with voip.ms as my provider. Voip.ms allows sharing multiple phone numbers on a single SIP account by setting the outbound caller id: "Choose this option if you are using ...
Sophie's user avatar
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0 answers
64 views

I am currently facing an issue with the call conference feature while creating a meeting. Basically, I am trying to create a meeting after both extensions have been successfully connected. After the ...
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0 answers
70 views

I have Avaya Orchestration Designer 8.1.2 and we made an app to Transfer call to external Genesys Infrastructure. But we need to send a SIP REFER TO with User-to-User hex some information, like this: ...
0 votes
0 answers
146 views

I’m currently developing a SIP phone using the JsSIP library and would appreciate some guidance on implementing a conference call feature. So far, I’ve successfully implemented the basic call controls ...
1 vote
1 answer
205 views

I’m building a web-based SIP phone application using JsSIP (version 3.10.0) to handle VoIP calls over WebRTC. My setup works fine for inbound calls—audio streams both ways—but I’m facing an issue with ...
1 vote
1 answer
66 views

System Details: macOS Version: macOS 13 Chip: Apple M2 Conda Version: 23.7.4 Python Version: 3.11 RELION Version: 5.0 PyQt5 Version: 5.15.9 Questions: Why is PyQt5 failing to build metadata, and how ...
Melissa's user avatar
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