No sound to call Asterisk SIP

Question

I almost managed to launch a two-way call (click to call): 1st to my office and 2nd to my camera using the answer of Michal Niklas (thanks Michal) Asterisk click to call .

The main question is that 2 participants in the call cannot hear each other, I used because of the internal context for both of them. The system status web interface shows me 4 active channels and 2 external calls when a call is connected to both sides.

I configured Channel: local / MY OFFICE PHONE @ from internal Extension: MY CELL PHONE

when I install Channel Sip / 1 and test it with a soft phone, it works great!

Thanks for the help...

Answer 1

seems like a NAT problem. here are some suggestions for sip.conf

put nat = yes in the user definition as well as in the [common] tag, put externip and localnet

eg.

[general] externip=<your global IP> localnet=192.168.2.0/255.255.0.0 

Answer 2

Sounds like a NAT problem; Have you redirected RTP ports correctly? Have you configured the STUN server in the softphone (if any)?

Answer 3

configure in /etc/asterisk/rtp.conf

 [general] rtpstart=10000 rtpend=20000 

go to cli star

 rasterisk -vvvvvvvvvvvvvvvv core reload 

Answer 4

In my case, I had to change chan sip settings ( freepbx.tld/admin/config.php?display=sipsettings ) from nat to route

Answer 5

Try adding / changing your IP address in sip_nat.conf

 vi /etc/asterisk/sip_nat.conf 

Add / Modify

 nat=yes externip=XXX.XXX.XXX.XXX 

Save, quit and restart