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VPN and remote networks no audio - General Help - FreePBX ...

    https://community.freepbx.org/t/vpn-and-remote-networks-no-audio/19582
    hi all , and good day , i’m having a very annoying issues with elastix PBX servers. here is the setup I have : total of two server and five offices PBX server at office ( A ) with static IP and local IP and its connected to another PBX server at site ( B ) ( local IP ) with IAX tunnel and VPN tunnel. Office ( A ) SIP phones connected to Office ( A ) PBX server either using static IP …

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. amell2020 (Estados Unidos) October 4, 2021, 2:28pm #1. Hello, i have a FreePBX (asterisk) system as my pbx. It is connected to my Mikrotik. PBX: 10.0.0.210. Mikrotik: 10.0.0.1/24. I have two Mikrotik i have setup server l2tp VPN and client VPN. Server Mikrotik VPN Adr local: 10.100.0.1.

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 ...

SOLVED: No audio on remote extension - General Help ...

    https://community.freepbx.org/t/solved-no-audio-on-remote-extension/38501
    Hi, after replacing (an old) Freebpx installation with 13, the remote extensions are able to register, intitiate calls, but there is no audio. During the call setup I can see a 401 error, and after a few seconds the line is dropped because no response from the external extension. We are running a NAT setup, no SIP ALG, same NAT setting as the old freepbx system. I’m sure it’s …

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.

No Sound on External SIP in Asterisk|FreePBX

    https://www.sysfix.co.uk/Blog/No-sound-on-external-sip-asterisk.html
    No sound with SIP | No Sound With Asterisk| No Sound With NAT. To correctly troubleshoot your issue, make sure that the following steps are taken. On your router, open up the following ports and forward them to your asterisk box. Your asterisk box may be in the form of (FreePBX, AsteriskNOW, or TrixBox)- They all use the same format

Remote Extension no audio! (Even after HOW-TO) - General ...

    https://community.freepbx.org/t/remote-extension-no-audio-even-after-how-to/17744
    1)In SIP settings: set nat to yes and in dynamic ip I’ve set my dyndns hostname.In local network I put 192.168.1.0/255.255.255.0 (my gateway is 192.168.1.1) 2)Opened ports UDP 5000-5082 e 10000-20000 on my router. 3)In the remote extension I put nat=yes

Remote SIP, no audio when using asterisk - Asterisk

    https://forums.whirlpool.net.au/archive/925040
    The general problem with the no audio on remote extensions with asterisk is where you put the sip nat settings. Most people put it in the sip_nat.conf file but as these settings would be added at the very end of your SIP configuration it wont work. Some machines it …

PJSIP no audio on calls - Asterisk SIP - Asterisk Community

    https://community.asterisk.org/t/pjsip-no-audio-on-calls/85165
    Everything worked perfectly on chansip. I moved to PJSIP and I can’t hear audio on any of my calls. NAT was nat=force_rport,comedia for extensions 200 and 300 and nat=no for 100. It worked well and I used the python script to convert sip.conf to pjsip.conf. If I enable direct media, I’m able to hear one-way. PJSIP config: removed RTP debug shows no log, nothing.

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