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Correcting One-way Audio with a VoIP Call - Asterisk
https://www.asteriskpbxsystems.com/troubleshoot-oneway-audio.html
One-way audio can occur in either direction, however in-bound audio failure (lack of audio from the outside caller reaching the inside network (LAN) phone) is probably the most common. In many of these cases routers and firewalls could be the …
One-way audio, codecs, and native_rtp? - Asterisk Support ...
https://community.asterisk.org/t/one-way-audio-codecs-and-native-rtp/69551
(Office phone can’t hear the other end.) However, everything works fine when my cell calls into the trunk and Asterisk server. The first thing you associate with one-way audio is NAT problems and I thought I solved those by forwarding UDP 5060, my RTP port range and by setting localnet and externaddr in sip.conf.
Asterisk vmware esxi deployment RTP problems with one-way ...
https://community.asterisk.org/t/asterisk-vmware-esxi-deployment-rtp-problems-with-one-way-audio/81481
When I had problems with audio or no audio or no audio in one direction that was affecting ALL calls and not some. Asterisk 16.6.0 is installed on Debian 10, has 16 CPU, 12GB of RAM and 500GB disk space on fast Netapp storage and esxi, where Asterisk VM is located, are connected over 4 10Gbps iface fiber optics.
One way audio problem in trunk - Asterisk SIP - Asterisk ...
https://community.asterisk.org/t/one-way-audio-problem-in-trunk/72567
Outgoing call : Everything is ok Incoming call : Call signaling is ok but audio RTP is one way. Peer A,B and C can send audio packet to Peer X but it isn’t able to receive audio packet from Peer X. Network : [Asterisk trunk] ----- [VoIP Provider's Router] ----- [VoIP Provider's SBC] ----- [Internet]-----[Peer X] IP: 192.168.0.5 Local IP : 192.168.0.1 I...
RTCPinterval Affecting One-way audio - Asterisk SIP ...
https://community.asterisk.org/t/rtcpinterval-affecting-one-way-audio/90290
RTCPinterval Affecting One-way audio. cable October 19, 2021, 7:40am #1. I’m getting one way audio on outbound (they can always hear me, but I cannot hear them). If I set rtcpinterval=4922 or less the problem goes away. If it is something larger or commented out (5000) there is no return audio. Newest Git-master, also tried 18.7 both with pjsip.
sip - Asterisk one way audio - Stack Overflow
https://stackoverflow.com/questions/55643399/asterisk-one-way-audio
This results always in a one way audio connenction. I use the odbc database, and can't really find the problem. Can anybody help me in the right direction. There seems to be no errors at all. [general] context=public allowguest=no allowoverlap=no udpbindaddr=0.0.0.0:15060 tcpenable=no tcpbindaddr=0.0.0.0 transport=udp srvlookup=yes language=ja ...
PJSIP One Way Audio - Asterisk SIP - Asterisk Community
https://community.asterisk.org/t/pjsip-one-way-audio/84926
If you are behind NAT, then are you able to forward ports on your firewall for RTP audio ? It might be 10000-20000 UDP port range that you need to forward to your PBX - but check port settings in /etc/asterisk/rtp.conf file. Also the Asterisk CLI command “rtp set debug on” might help you see more info.
One way audio remote vpn based extension - Asterisk ...
https://community.asterisk.org/t/one-way-audio-remote-vpn-based-extension/44314
Dear all, I’m facing a problem on a Asterisk based box (Pika Warp 2) with one of our clients. We are trying to setup a remote extension, using a openvpn connection on a hardphone (YealinkT28 equivalent). The VPN is up, I can ping the phone from the asterisk box, I can manage the phone from the openVPN server, the phone is registering, we can call the …
How to troubleshoot one-way / no audio issues - Cisco ...
https://community.cisco.com/t5/collaboration-voice-and-video/how-to-troubleshoot-one-way-no-audio-issues/ta-p/3164442
How to check really quick if the phones are sending / receiving RTP (audio). * Open the web page for 2 test phones, then click the 'stream 1' link located at the left handed side of the page, and check if the IP address and port match the information on both sides, keep pressing the 'stream 1' link and you will notice that the Tx and Rx stats ...
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