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VoIP settings (no voice/audio) - Network Protection ...
https://community.sophos.com/utm-firewall/f/network-protection-firewall-nat-qos-ips/40449/voip-settings-no-voice-audio
Firewall = VoIP (network) > VoIP Protocols (H323) + (RTP 16384:16482) + (SIP 5060:5080) Global SIP Settings > SIP server networks (VoIP Provider x.x.x.x and x.x.x.x) Global SIP Settings > SIP client networks (VoIP Network) With this …
VoIP IP Phone No Audio issues over VPN - Network ...
https://community.sophos.com/utm-firewall/f/network-protection-firewall-nat-qos-ips/40994/voip-ip-phone-no-audio-issues-over-vpn
We can ring the phone, but there is no audio when we answer. Same thing if we ring the main office - they can take the call, but no audio. We can access the web interface of the polycom phone from the main office no problem. ... Try googling site:astaro.org voip "no audio" and I bet you'll find some other suggestions. Cheers - Bob . Sophos UTM ...
SIP Proxy no audio on outgoing - Management, Networking ...
https://community.sophos.com/utm-firewall/f/management-networking-logging-and-reporting/32017/sip-proxy-no-audio-on-outgoing
I have a Zyxel P2002 for my VoIP and using Astaro 6.303. My problem is that i have no audio on outgoing calls, incomming calls works as it should, just not the out going. Anyone have a clue here? I have tryed with any any any in the packet filter to make sure that wasent the problem Found som log from the outgoing call.
Voip No Outgoing Audio - Network Protection: Firewall, NAT ...
https://community.sophos.com/utm-firewall/f/network-protection-firewall-nat-qos-ips/40711/voip-no-outgoing-audio
Voip No Outgoing Audio. 1 external internet 91.208.*.*. We have added a new internal subnet of 172.1.1.0/24 to allow for more internal IP addresses and to section off the Voip phones. The phones on the internal 192.168.1.0/24 range works perfectly for voice both in and out without problems or running the VOIP security.
SIP voice call issue "no audio" - Network Protection ...
https://community.sophos.com/utm-firewall/f/network-protection-firewall-nat-qos-ips/84841/sip-voice-call-issue-no-audio
SIP voice call issue "no audio" Fabio Ermotti over 5 years ago. Hello, we have 3CX PBX server on internal network and enabled VoIP (Network Protection-->VoIP-->SIP) on our UTM v9 (9.409-9) to allow SIP comunication with our VoIP provider: all seem works fine but sometimes a phone call is established without audio or a call is broken unexpectly ...
NO AUDIO on calls SOLVED - voip problem caused by GV ...
https://support.google.com/voice/thread/7375647/no-audio-on-calls-solved-voip-problem-caused-by-gv-updates?hl=en
Google Hangouts is no longer needed to make VoIP calls. I don't know why you are not hearing audio on your calls. You've got some sort of problem …
VoIP issue: no audio - Network Protection: Firewall, NAT ...
https://community.sophos.com/utm-firewall/f/network-protection-firewall-nat-qos-ips/42114/voip-issue-no-audio
When I dial out I get no audio. The only way can get VoIP to work properly is by creating a firewall rule: FreePBX ->any->any. (this works regardless if VoIP is enabled or not under Network Protection). If I change the protocol to SIP or the destination to the SIP provider I get no audio.
How to solve VoIP one way audio- step by step.
https://www.voipmechanic.com/voip-one-way-audio.htm
Step One- Check your equipment. The first thing that you need to eliminate is a faulty phone, handset earpiece or a headset on a softphone. To do the do the following: Softphone - Check that you have audio in both the earpiece and microphone. Verify that your microphone is connected properly.
SIP - No audio or one way audio :: Zoiper
https://www.zoiper.com/en/support/answer/for/android/11/SIP_-_No_audio_or_one_way_audio
SIP - No audio or one way audio. When using SIP protocol one way or missing auido issues mostly appear due to configuration problems. The issues usually appear if the configuration is not adjusted accordingly to the wireless or 3g/4g network. In most cases this can be resolved by altering the account configuration.
SIP with NAT or Firewalls - Asterisk Guru
http://asteriskguru.com/tutorials/sip_nat_oneway_or_no_audio_asterisk.html
Without STUN support, you will also need NAT=yes or NAT=route, and you will have no incoming audio on the natted phone until the asterisk server received audio from that natted phone. Please note that without STUN support, the registrar and proxy server have to be on the same IP.
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