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Asterisk 1.6.1.0 NAT problems, Remote Client No Sound ...
https://community.asterisk.org/t/asterisk-1-6-1-0-nat-problems-remote-client-no-sound-audio/27050
But, there’s no sound can be heard from the X-Lite client. There’s no problem when connecting with VPN. Our setting is, X-Lite <—> NAT (broadand router) <—> INTERNET <—> NAT (broadband router)(fix ip) <—> Asterisk 1.6.1.0 (192.168.0.1) SIP.CONF file has configured for using nat already [general] udpbindaddr=192.168.0.1:5060 tcpenable=no
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 ta… Hi all, I’m new in voip and asterisk and i really need your help. ... s=CounterPath X-Lite 3.0 c=IN IP4 192.168.1.181 t=0 0 m=audio 44448 RTP/AVP 107 119 100 106 0 105 98 8 101 a=alt:1 2 ...
No sound: X-Lite -> Asterisk -> VoIP Provider ->Cellphone ...
http://readlist.com/lists/lists.digium.com/asterisk-users/8/40157.html
X-Lite (build 34025) at home behind NAT -> Internet -> Asterisk at work behind NAT -> Internet -> VoIP provider -> GSM gateway -> cellphone. I just tried calling my own cellphone, but there is no sound either way. Here's what I did on the X-Lite at home in the Topology section: IP address : Discover global address STUN server : Discover server
No sound - POTS to X-Lite - Asterisk
https://forums.whirlpool.net.au/archive/1255907
Pots line – Asterisk (IP04 appliance) – X-Lite softphone. The Asterisk box and PC running X-Lite are on the same subnet & switch. Asterisk also has another analogue phone connected to an FXS port, and has itself registered as a SIP client to iiNet. Outbound calls from X-lite via Pots: Works Outbound calls from X-lite via iiNet: Works Inbound calls via iiNet, picking …
SIP with NAT or Firewalls - Asterisk Guru
http://asteriskguru.com/tutorials/sip_nat_oneway_or_no_audio_asterisk.html
DrWho's Solution to Asterisk behind NAT/Firewall (techiegz at gmail dot com) 06 July 2006 05:53:50 On your router NAT/firewall, forward SIP ports 5060 - 5082 and RTP ports 8000 - 20000 to your * server IP address. Then edit the "rtpstart" value in rtp.conf - from rtpstart=10000 to rtpstart=8000 since 8000 is the default RTP port on x-lite phones.
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 audio on outgoing trunk call - General Help - FreePBX ...
https://community.freepbx.org/t/no-audio-on-outgoing-trunk-call/56697
Right now I’m 10.0.0.2 connecting from home administering the Asterisk box and doing tests from my X-Lite/Zoiper clients as well. I have tried disconnecting the Asterisk box from the internet, set it’s default gateway to the provider’s “Cisco VPN Gateway” LAN-side IP, externip to WAN-side IP. And still the same problem.
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