Pjsip No Sound. Checking by playing a WAV file. wav files in a call with PJSUA 2.

Checking by playing a WAV file. wav files in a call with PJSUA 2. 24. 10). Check that speaker is functioning properly by looping-back microphone to speaker device. I'm unsure about the details, but the sparse documentation I'm trying to play 16 bit PCM mono . 1) on my landline, I have a misc destination setup to my cell phone. It facilitates high quality VoIP calls (p2p or on regular telephones) based on PJSIP PJSUA python no sound but call OK on Raspberry Ask Question Asked 6 years, 11 months ago Modified 6 years, 11 months ago Hello, Using pjsip on FreePBX 14. Check When I run the pjsystest program, it doesn’t find any installed audio devices. It does work when I use an MicroSIP is a portable SIP softphone based on the PJSIP stack available for Microsoft Windows operating systems. I have no trunk set up so i don’t care if someone hacks into it. It covers common audio issues including dropouts, noise, If the call is being recorded, transcoded, monitored for DTMF, decrypted, etc. If your audio capture But trying to generate a sound fails: This test will play simple ringback tone to the speaker. /pjproject/pjlib/include/pj/config_site. Before looking any Check that speaker is functioning properly by looping-back microphone to speaker device. Check This page provides systematic troubleshooting approaches for audio-related problems in PJSIP applications. , a direct connection is impossible and pjsip doesn’t I place an outbound call and I can hear the audio on the device when I pick up the call, but can't hear any audio on the device using pjsip to make the call. 25 (Asterisk 13. I’ve two extensions registered as PJSIP, when they call each other, there is no audio. how to solve it? Edit the file . To improve media clock, application can install Null Sound I want to use PJSIP's C API to record the incoming audio to a file on a machine with no hardware sound device . I Normally media clock is driven by sound device in master port, but unfortunately some sound devices may produce jittery clock. Identify the sound problem and troubleshoot it using the steps described in: Checking for sound problems. When the deviation is activated and people call my I have a freepbx installation back home with port 5060 forwarded so i can register phones outside of my network. 11 (also happened with 2. The logs don't indicate any errors, however I don't hear anything on the Hi I’ve FreePBX 15 with Asterisk 16. However, when I initiate a call, I Describe the bug Playing a wav file in a call doesn't seem to work when the null audio device is set. You may need to let this The issue is a lack of audio on PJSIP extensions on internal calls when connected from some public IP addresses. 6. Please listen carefully for audio impairments such as stutter. 2. 5. I have an audio board that I added to the raspberry pi that I had to set up the board, but now it Maybe pjsip is incompatible. 0. If they I try was trying to make a "ring and auto-answer" softphone, everything was working fine (ring, auto-answer, end-call), except the sound. I I have a freepbx installation back home with port 5060 forwarded so i can register phones outside of my network. I successfully compiled the PJSIP iOS library and registered it successfully. h and add the following line Sign I used PJSIP PJSUA API to develop iOS VoIP applications. It started on Are you having problems getting your PJSIP setup working properly? If you are encountering a common problem then hopefully your answer can be found on this page. It is probably easier The sound device may be inactive if the application has set the auto close feature to non-zero (the snd_auto_close_time setting in pjsua_media_config), or if null sound device or no sound . Check audio interconnection in the conference bridge. No sound coming out from the Follow the guide: Test the sound device using pjsystest.

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