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No audio on Asterisk SIP call - Stack Overflow
https://stackoverflow.com/questions/5554837/no-audio-on-asterisk-sip-call
No audio on Asterisk SIP call. Ask Question Asked 10 years, 10 months ago. Active 1 year, 10 months ago. Viewed 34k times 9 I almost managed to init a 2 sided call (click to call): 1st to my office and the 2nd to my cell using Michal Niklas answer (thanks Michal) on Asterisk click to call. The major ISSUE is that the 2 call participants can not ...
No Sound on External SIP in Asterisk|FreePBX
https://www.sysfix.co.uk/Blog/No-sound-on-external-sip-asterisk.html
SIP One Way Audio Troubleshooting Once these ports have been forwarded to the IP of your Asterisk server, give your router a reboot. Re-Login to your router just to reconfirm that the posts are in fact pointing to the correct IP of the Asterisk Server. The next step is to ensure that you configure your NAT settings on the Asterisk server correctly.
Calls with no audio in both sides - Asterisk SIP ...
https://community.asterisk.org/t/calls-with-no-audio-in-both-sides/73244
Calls with no audio in both sides Asterisk Asterisk SIP humber2 January 19, 2018, 11:02am #1 I’m trying to implement a new service on my Asterisk server network. We have a 3G mobile phone with a softphone app - Linphone, whose number is 9012 and is registered on the Asterisk server with IP # 10.192.124.101 (Sagres).
SIP with NAT or Firewalls - Asterisk Guru
http://asteriskguru.com/tutorials/sip_nat_oneway_or_no_audio_asterisk.html
Asterisk will send the audio to the port and ip where its receiving the audio from. Instead of relying on the addresses in the SIP and SDP messages. This will only work if the phone behind nat send and receive audio on the same port and if they …
No audio and no rtp traffic - Asterisk SIP - Asterisk ...
https://community.asterisk.org/t/no-audio-and-no-rtp-traffic/69420
I have installed magnusbilling on asterisk 11 and sometime no one of both end hear something, sometime I issued one way audio and sometimes I’m able to talk normally. In this third case i can see the output in “rtp set debug on”, otherwise no. ... – SIP/GO_VoIP_1-000000b9 answered SIP/VOIPTEST-000000b8 Audio is at 33636 Adding codec ...
asterisk: IP address order may cause no audio · Issue …
https://github.com/irontec/ivozprovider/issues/511
The first SIP message, the invite from trunk provider to my system shows no difference at all. The second SIP message, the INVITE from kamailio trunks to asterisk has a difference on the last SDP header. The IPs: xxx.xxx.xxx.194 kamailio trunks xxx.xxx.xxx.195 kamailio users / rtp proxies audio sock xxx.xxx.xxx.206 trunk provider SIP server
No audio on sip calls over VPN - Endpoints - FreePBX ...
https://community.freepbx.org/t/no-audio-on-sip-calls-over-vpn/78414
No audio on sip calls over VPN. i have a FreePBX (asterisk) system as my pbx. It is connected to my Mikrotik. I have two Mikrotik i have setup server l2tp VPN and client VPN. Inside my internal lan, 10.0.0.0/24, everything is working fine as voip telephony concerned. When i connected through VPN, i can register my sip phone and i can call every ...
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
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. Run Zoiper for Android and go to Config.
SIP no audio with FreePBX - Netgate Forum
https://forum.netgate.com/topic/163469/sip-no-audio-with-freepbx
I have a SIP client on internet which is configured to connect on x.x.x.218. The client registers on FreePBX and I can make and receive calls, but audio is a problem. If I place the call on the SIP client, there is no audio at all. If I place the call elsewhere and call the SIP client, the SIP client side has audio, but not the other side.
No Audio PJSIP - Endpoints - FreePBX Community Forums
https://community.freepbx.org/t/no-audio-pjsip/65068
As a quick test, set all the Recording Options for one of the extensions to Force and retest. If you still don’t get audio, report whether dialing *43 (echo test) from each of the extensions works. If setting Force does give you audio, add this to /etc/asterisk/pjsip_custom_post.conf disable_direct_media_on_nat=yes and if that doesn’t …
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