Hi,
I have a problem about sound but I don’t know how can I fix it. When I watched to CLI log I can’t see anything.
Could you give me a suggestion .
Problem is;
I have two extensions that are 1001(Yealink T28) and 1002(Xpeech Dect). When I tried to call from 1002 to 1001 there is no problem but when I tried to call from 1001 to 1002. 1002 ishearing me but I can’t hear it.
One way audio normally results from NAT or firewall problems.
Hello, if your server is on live IP on cloud, try below settings in sip.conf in general section:
It seems NAT issue.
[general]
callcounter=yes ; enable device states for SIP devices
canreinvite=no
trustrpid=yes
sendrpid=no
qualifyfreq=60
transport=Auto
avpf=no
force_avp=no
icesupport=no
encryption=no
bindaddr=0.0.0.0
bindport=5060
externip=52.25.x.x
localnet=172.31.45.162/255.255.240.0
nat=force_rport,comedia
nat=yes
qualify=yes
@ankitjayswal87 some of your configurations are deprecated on current releases of Asterisk for example.
nat=yes
canreinvite=no
externip
check the sample configuration file for sip http://svn.digium.com/svn/asterisk/trunk/configs/samples/sip.conf.sample
also you have the icesupport set to no, and in a nat environment should be set to yes.
ICE is a standardized mechanism for establishing communication suitable for live media streams between software agents running behind NAT firewalls. Establishing connections through NATs is referred to as traversing the NAT
check this document for more reference. https://wiki.asterisk.org/wiki/display/AST/Interactive+Connectivity+Establishment+(ICE)+in+Asterisk