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No Sound on External SIP in Asterisk|FreePBX
https://www.sysfix.co.uk/Blog/No-sound-on-external-sip-asterisk.html#:~:text=No%20sound%20with%20SIP%20%7C%20No%20Sound%20With,or%20TrixBox%29-%20They%20all%20use%20the%20same%20format
No audio on Asterisk SIP call - Stack Overflow
https://stackoverflow.com/questions/5554837/no-audio-on-asterisk-sip-call
3) Multiple answers in a single call: a call can be answered only a single time, in some asterisk versions you won't receive audio if a call is answered twice or more times so make sure you don't. Anyways, why is Asterisk placing 2 calls?
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
SIP with NAT or Firewalls - Asterisk Guru
http://asteriskguru.com/tutorials/sip_nat_oneway_or_no_audio_asterisk.html
1) Call coming from behind nat, Asterisk sends audio to a private ip.
No audio when bridging two trunk calls - Asterisk SIP ...
https://community.asterisk.org/t/no-audio-when-bridging-two-trunk-calls/85714
But when I am trying to do the same between two mobile numbers (through my sip provider), I get all the signals (ringing, hangup) on both phones, but no audioon both phone. Sample PJSIP config for my sip provider I am using as: [sipprovider] type=endpoint transport=transport-udp aors=sipprovider auth=sipprovider outbound_auth=sipprovider
Can't hear other person · Issue #2 · i-p-tel/sipdroid · GitHub
https://github.com/i-p-tel/sipdroid/issues/2
Asus Transformer Prime, latest Sipdroid from Play Store, connecting to Asterisk 1.8.7 on same IP subnet (so no NAT or firewall concerns). If I call from Asus to another extension on the same subnet connected to same asterisk server then remote end can not hear my voice, but they _can_ hear dialpad buttons if I press then.
No audio on PJSIP channels - Asterisk WebRTC - Asterisk ...
https://community.asterisk.org/t/no-audio-on-pjsip-channels/90281
I don’t understand how Astertisk can match the local IP address of my client, since In my softphone code I use my public IP address to log to asterisk I’m using WebRTC for my sofphones Here is my config ;====================== registration ======= [transport-udp-nat] type=transport protocol=udp bind=0.0... No audio on PJSIP channels Asterisk
No audio on incoming calls from messagenet - General Help ...
https://community.freepbx.org/t/no-audio-on-incoming-calls-from-messagenet/11446
Hi all, i have an issue with incoming calls from messagenet with asterisk+freepbx. My situation is: A server with archlinux, with asterisk 1.8.4 and freepbx 2.9 installed on it. This server is behind a nat. There are configured some extensions (100, 101, 102…) There are some trunks, voipcheap, megavoip, messagenet I use voipcheap and megavoip for outgoing calls, …
Sipdroid Android VoIP app SIP Client Setup Configuration
https://www.voipvoip.com/sipdroid/
1.) Install the Sipdroid app for your Android phone, open the application and go to SIP Account Settings. 2.) Enter the following information; Authorization User Name:555xxxxxxx (Instead of 555xxxxxxx account number example, use the account number assigned to you - e.g. 5551231234- when signed up for VoIPVoIP service.
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