2,831 questions
0
votes
0
answers
63
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
2
votes
0
answers
99
views
How to get call statistics for a pjsip call in iOS
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 ...
- 4,039
0
votes
1
answer
103
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 ...
0
votes
0
answers
73
views
Microphone only records distortion on Linphone SIP calls (Android 7.1.1, Watch device)
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/...
- 725
0
votes
0
answers
82
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 ...
0
votes
0
answers
49
views
Call initiated and connected but can't hear anything
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
140
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 ...
0
votes
0
answers
103
views
How to Configure Kamailio as a Load Balancer for FreeSWITCH Clusters?
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 ...
0
votes
0
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 ...
0
votes
0
answers
61
views
Issue with N-Way Conference Feature and addMeetNowForGeneral API
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 ...
- 1
0
votes
0
answers
67
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:
...
0
votes
0
answers
127
views
How to Implement Conference Call Feature Using JsSIP with FusionPBX/Freeswitch
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
190
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 ...
1
vote
1
answer
64
views
Conda Environment Update Fails Due to PyQt5 Metadata Generation Error on macOS (ARM)
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 ...
- 59
0
votes
0
answers
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 ...
0
votes
2
answers
235
views
PJSUA2 Python Library RTP packet capturing
I have started a simple project that is capable of making a SIP establishment for voice call and then forwarding RTP packet through another endpoint. I used PJSUA2 python library and right now I can ...
- 101
0
votes
0
answers
46
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 ...
0
votes
0
answers
60
views
varying RTP stream result from custom SIP implementation
I am in the process of creating my own SIP implementation in Node.js. As well as a b2bua as a learning project.
Finding people wise in the ways of SIP has proved to be difficult elsewhere but here I ...
- 304
0
votes
0
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
0
votes
0
answers
91
views
Twilio SIP Interface - use for Outbound calls
I’ve used Twilio Flow with the “Make Outbound Call” widget to call users and send requests to my API based on their input, which was straightforward (my dev skills are ok, but with infra I am just ...
- 87
0
votes
0
answers
138
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 ...
0
votes
0
answers
135
views
How to Ensure H264 Codec is Negotiated in WebRTC SDP on React Native?
I’m working on a React Native app that uses react-native-webrtc and SIP.js for video calls. The SIP server sends an INVITE SDP that includes support for multiple codecs, including H264, VP8, and VP9. ...
- 372
1
vote
0
answers
47
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
268
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
272
views
Flutter receive web rtc calls when app is terminated (detached state)
Websocket connection is closed abruptly in asterisk using Web RTC connection when my flutter app terminated (User swipe to close app)
Flutter version
Flutter 3.24.0 • channel stable • https://github....
- 11
0
votes
1
answer
153
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
98
views
SIP: How to compile a TRYING response to an INVITE
I'm developing an application in Python able to manage SIP transactions.
I don't need to answer or send/receive media, I'd just like to take track of an incoming call and send an INIVTE with a CANCEL ...
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
1
vote
0
answers
430
views
PyVoIP client doesn't answer to SIP OPTIONS sent by the SIP Server
I'm trying to create a SIP client that logs all the received calls.
I'm using Python with the pyVoIP library, this is the code:
from pyVoIP.VoIP import VoIPPhone, CallState, PhoneStatus, ...
0
votes
2
answers
241
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
1
answer
370
views
PJSUA2 auto-declining recieved calls
I'm unable to receive calls on my PJSUA2 python script. I've simplified my script as much as possible to replicate the issue below. This simplified script is largely sourced from the PJSUA2 docs ...
- 47
0
votes
0
answers
98
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
3
answers
552
views
Using JSSIP to test SIP login credentials
In my web application, the user provides his SIP credentials and I want to test that they are valid using JSSIP to establish a connection to his VOIP provider and then unregister. This is the code I ...
- 52.3k
0
votes
1
answer
426
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
536
views
Integrating SIP.js or similar library with react-native-webrtc
Please, HELP.
I am trying like for few weeks or months already to make outbound call with sip.js or jssip and react-native-webrtc.
I have stun and turn server from telnyx.
I have default sip.js with ...
0
votes
1
answer
313
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
0
votes
1
answer
68
views
twilio sip registraion - BE node, FE react native (Twilio Register SIP Endpoint)
I have configured twilio SIP domain, TwiML app and credentials list for users.
But I am unable to register users in Registered SIP endpoints. Is it possible to register from react-native app?
I am ...
- 940
2
votes
1
answer
107
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
0
votes
1
answer
89
views
SIP Call error: pjmedia_endpt_create_base_sdp: Assertion `!"Invalid address family"' failed
I am a college student who is new to VOIP and SIP. I have a SIP server setup and am able to register myself, but when I try to make a call, there is this error:
10:43:18.461 pjsua_acc.c .....
-1
votes
1
answer
153
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
1
vote
0
answers
60
views
How to use sip signling to Implement audio and video calls to webrtc
In Android and iOS client, I use PJSip as signaling, with webrtc to achieve audio and video communication, do not know how to achieve signaling conversion, someone help me？
I am sending events to ...
- 11
0
votes
0
answers
187
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
127
views
Kamailio SBC Record Route header first line is empty, only the second line contains the IP
Record Route header line 1 is empty for 180 Ringing and 200 OK messages, the second line of the Record Route contains the IP. For the INVITE is not the case, both lines contain the IPs.
This causes ...
0
votes
1
answer
58
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
310
views
Registering SIP when receiving a firebase message in terminated flutter app
I'm trying to make a call notification on my flutter app. This is almost working except the following thing. When the app is terminated, I receive the firebase message and I can display my ...
0
votes
1
answer
464
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 ...
-1
votes
2
answers
234
views
How to send dtmf and play audio stream like ivr using websocket send to freeswitch?
I'm building a call application using webrtc to freeswitch using java as sip gateway, I want to implement ivr function but don't know how to send dtmf to freeswitch via websocket on java. I cannot ...
0
votes
1
answer
328
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
-2
votes
1
answer
79
views
Amı Originate Calls Diffrent Context From Sended Context
I'm using ami-io library with Node.js. I'm sending originate call with a context for redirect answered calls to ivr. There is 2 ivrs for diffrent purpose. Also I'm setting the lead_id to channel for ...
0
votes
1
answer
122
views
SIP messages are not visible after Phone rooting(samsung A33)
I am trying to get the sip messages in the rooted samsung a33.I took the tcpdump and still no sip packets can seen. Anyone can help here.
Took the tcpdump from phone side using PCapdroid and also tcp ...
- 21