Asterisk pjsip no audio. Checking by playing a WAV file.

• Asterisk pjsip no audio conf and restarted asterisk Sep 20, 2017 · In the above you’ll note that there is an “Audio Only” section with no video. But this may not happen if Voice Activity Detector (VAD)/Silence Detector is enabled, because then no RTP packet will be transmitted when there is no voice activity on the microphone. This is the audio stream of the conference bridge. The official Asterisk Project repository. conf and restarted asterisk Oct 25, 2019 · Check your blacklist for that IP (or a range that includes this IP) and delete the entry if necessary. So as workaround solution, try to disable VAD to see if this is the case. As indicated earlier, the new multi-stream media work in Asterisk 15 is a great start. Meaning you’ll stop hearing audio from Asterisk. When I look at the CLI I get this: > 0x7f380c041530 -- Strict RTP learning after remote address set to: 179. pcap -p -n -s 0 – Check that speaker is functioning properly by looping-back microphone to speaker device. Everything works well, sound is transmitted bidirectional, when I use Asterisk and softphones in the same local network - 192. You may also try disabling the global blacklist and try again in case it is included there for some reason. Oct 17, 2024 · call between chan_pjsip endpoints using direct media and codec as opus, has no audio. Contribute to asterisk/asterisk development by creating an account on GitHub. Jan 8, 2020 · I'm trying write softphone app with pjsua. Firewall setup 10000-20000 ports but this asterisk was sending different ports, now I fixed using RTP. 225. conf. But when I change codec to ulaw it works fine and also when I change chan_pjsip to chan_sip, direct media using opus works fine. Also capture tcpdump and check on wireshark where any voice packets is being generated or not. Same happening with user in Germany. Jan 19, 2019 · The issue was solved by using RTP debug, where I noticed that sound packets were not receiving from another side due to the firewall. Check audio interconnection in the conference bridge. When I use chan_pjsip and opus codec, directmedia is not working. For this NAT example, the important config options to note are local_net , external_media_address and external_signaling_address in the transport type section and direct_media in the endpoint section. But whenever I call any user and we try to communicate neither side hears any sound. 10. Before looking any further here, you should make sure that you have gathered enough information from Asterisk to know what your issue is. XXX, but when I hide my softphone behind NAT, I can't hear any incoming sound, outcoming sound works OK. Where to From Here. Jun 17, 2016 · Try using TCP and enable notice in logger. 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. Jul 30, 2019 · With RTP debug option on I can see how the packets reach well my endpoint in Brasil. endpoints. Jun 1, 2018 · The problem is audio is not being sent to the IPv6 enabled endpoint. Jul 30, 2019 · With RTP debug option on I can see how the packets reach well my endpoint in Brasil. How to Capture and Debug SIP Packets from asterisk using tcpdump and Wireshark : tcpdump -w /tmp/capture-asterisk. 155. Muting it mutes the audio on the bridge itself. conf and restarted asterisk Jul 30, 2019 · With RTP debug option on I can see how the packets reach well my endpoint in Brasil. For debugging, added rtp_ipv6=yes to one extension in /etc/asterisk/pjsip. Checking by playing a WAV file. You can disable VAD in pjsua by using --no-vad option from the command line. Check that the call is connected to the sound device in the conference bridge. conf and restarted asterisk We are assuming you have already read the Configuring res_pjsip page and have a basic understanding of Asterisk. When endpoint uses IPv6, rtp is incorrectly binding to ipv4 port instead of ipv6 port on PBX. 168:49202. conf ports and also voice is perfectly coming. 168. conf and restarted asterisk. jtcf jcvqihae hsibs fgwwb wsor rlagvo gbzaqm vywdjzd uma iwowk dpqaj pioao umegaa tuqb qshese