Hi
I installed Asterisk and i successfully made a call from my phone (connected to my wifi) and my labtop and everything was excellent. but when I connected with my cellular data with my phone while labtop was still connected to wifi, I was able to establish a call, but no voice was in either side.
i have very simple configuration as below :
sip.conf :
[general]
transport=udp
allowguest=no
allowoverlap=no
bindport=5161
bindaddr=0.0.0.0
friends_internal
type=friend
host=dynamic
context=from-internal
qualify=yes
secret=12345
ali
reza
extensions.conf
[from-internal]
exten=>6001,1,Dial(SIP/ali,20)
exten=>6002,1,Dial(SIP/reza,20)
Please provide the SDP exchange for the failed call. This is most likely due to firewalling by the service provider, possibly to prevent loss of revenue.
Also note that VoIP and Wifi do not mix well in terms of voice quality, because latencies are too high and variable.
Thanks for your answer.
what do you mean by SDP exchange ? to debug information from asterisk CLI ?!
I also share my cellular data with my laptop, and voice had no problem and everything was OK. It seems it works only p2p.
The INVITE transaction, and in particular the body of the messages, in the VERBOSE Iand DEBUG) channels of the log files. Details will depend on whether you are using chan_sip or chan_pjsip. Please don’t screen scrape as the result is often truncated and lacks good time stamps.
https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information
Although it is very helpful if you remove all but the INVITE transaction.
Note. This will contain IP addresses, which may be sensitive, but we need to know, at least, where they differ from each other, and which are public and which private addresses.
1 Like
Thank you for your answers
I was making a rookie mistake… I hadn’t declared “externip” in my sip.conf
after adding this, everything gets OK