2,832 questions
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54
views
How to disable or modify Supported headers (100rel/timer) in PJSUA2 before sending INVITE
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-...
- 139
0
votes
1
answer
96
views
Setting up an IMS call between two legacy IMS clients
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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answers
76
views
How to authorize Zadarma API?
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 ...
1
vote
1
answer
130
views
How to correctly route a final ACK to a backend server with OpenSIPS load_balancer and rtpproxy?
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 ...
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answers
42
views
Pjsua2 - setting outbound caller ID
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 ...
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0
answers
61
views
Avaya OD Property Refer-To in Blind Transfer Node
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:
...
1
vote
1
answer
185
views
JsSIP "peerconnection" Event Not Firing for Outbound Calls - Why and How to Fix?
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 ...
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63
views
How can I setup SIP_UA in Flutter
I am trying to build a softphone application and I am having a hard time implementing sip_ua and make calls. There's not a lot of resource online about this specific library. This is the state of the ...
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answers
43
views
One-way audio only on iOS softphone, need more complete solution
I have developed an application for both iOS and Android using Flutter/DART and using the sip_ua package. I have everything working great on both platforms, with one small exception. On iOS, the audio ...
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answers
43
views
sip in background flutter
I have a problem when I try to pass my sip service in the background.
I'm using the flutter_background_service: ^5.0.10 plugin, I've followed the documentation scrupulously for the implementation of ...
- 37
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votes
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135
views
How to resolve Twilio Error 32011: "Error Communicating with Your SIP Communications Infrastructure"?
Issue with Incoming Calls on Twilio SIP Trunk - Error 32011
Hello everyone,
I'm experiencing an issue with incoming calls on my Twilio SIP trunk setup. Here are the details of the scenario:
The ...
1
vote
0
answers
43
views
No Way to Reconnect Call When Internet is Lost in sip_ua
I have noticed that there is currently no clear way in the sip_ua package to handle scenarios where the internet connection is lost during an active call. Specifically, there seems to be no built-in ...
- 11
0
votes
1
answer
264
views
How to work with navigation and flutter background service
Goal
I want to creating a background service to listen for registration. And if user successfully logged in then it will redirect to home page widget
Code snippet
Here is my current code snippet work ...
- 11
0
votes
1
answer
152
views
Invalid description, no ice-ufrag attribute at level 0 Error
I was trying to implement a sipjs session.
initializeSip() {
const uri = UserAgent.makeURI('sip:[email protected]'); // Replace with your SIP URI
if (!uri) {
throw new Error('Failed to create URI');
...
- 17
0
votes
1
answer
97
views
CallerID not working using SIPSorcery .net 6 with Voip.ms
When using SIPSorcery with .NET 6 to make SIP calls via VoIP.ms, the custom Caller ID is not being passed correctly, despite being set in the code.
We have the setting configured in VoIP.ms as:
"...
- 1
0
votes
2
answers
222
views
Unable to register sip in Twilio sip domain from my flutter mobile application using the sip_us package
I am using the sip_ua package in my flutter application to make a sip connection with twilio sip domain. I have used Zoiper to test my sip domain and it works fine, and I can see the registered sip ...
- 1
0
votes
0
answers
97
views
I'm trying to use pjsip for video calling in react native. But it doesn't work
In my React native project, I want to access freeswitch data using pjsip and make video calls between two phones. Actually, one is a phone and the other is a doorbell panel with camera. Does anyone ...
0
votes
1
answer
418
views
Use custom generated call_id in the Sipp scenarion send Invite instead
I'm using Sipp to run some test cases. In my use case, the Sipp scenario tests a SIP Invite sent to a remote server and validate for 100, 180 and 200 OK finally, a basic uac.xml
The remote end point I'...
- 86
0
votes
1
answer
306
views
Twilio <Client> and <SIP> with <Enqueue> and <Queue>: BLF Not Glowing on Yealink T31 When Call is Parked
I'm currently working on a project using Twilio and Yealink T31 hardphones. I'm leveraging Twilio's <Client> and <SIP> verbs for handling calls, and using <Enqueue> and <Queue> ...
- 3
2
votes
1
answer
101
views
Accessing Twilio Voicemail Recordings in Polycom VVX250 Phones
We're using Twilio for calls and Polycom VVX250 phones. We have an SIP domain and credentials configured on Twilio and the Polycom phones. Our system checks a database to route calls based on linked ...
- 1,346
-1
votes
1
answer
152
views
Remove "Anonymous" text from From header - SIP invite
I need to remove "Anonymous" text from below From header in INVITE request. Done following configuration but no luck.
Used Asterisk 18.23.1 built by root @ pilrh-noc-dynamicconivr-app01 on a ...
- 6,572
0
votes
0
answers
181
views
TCP sequence number is incremented though the previous segment was not acked
Let me brief my scenario on SIP REGISTER, There is an endpoint behind NAT and gets connected to my application server over TCP via NAT device.
TCP Handshake done, endpoint(A) sent SIP REGISTER(let's ...
- 315
0
votes
1
answer
55
views
SIP INVITE to the client having 2 lines of same accounts but different PBXs
INVITE URI is not matching with line1 and line 2 accounts completely (username + proxy). How to handle INVITE now? What is the possible response from the phone?
Behavior expected is to route the call ...
- 1
0
votes
1
answer
451
views
Integration of flutter application into linphone codebase
Actually, I want to integrate my Flutter app with the existing linphone (VoIP) project (it is open source) and while I am trying to do this I am getting multiple errors. If any of you know the ...
0
votes
1
answer
327
views
Send SIP traces over TCP from Kamailio to Heplify
I am using siptrace module to send traces to Heplify from Kamailio. By default traces are send using UDP, but we would like to send them over TCP.
Reading siptrace module documentation does not easily ...
- 1,175