I have just installed Asterisk on my ubuntu server and trying to get it running
I have defined two accounts in sip.conf and now want to do a basic successful call using X-lite sip phone. The registration is fine, the ringing is OK and the call in general looks normal
but still no voice can be heard and lead the call to be disconnected after!.. checking the logs
chan_sip.c:3641 retrans_pkt: Retransmission timeout reached on transmission Mjk3MGU1NjgxZWQwM2E3MjhjZmFiNzhjOGVjZjg5ZTc for seqno 2 (Critical Response)
I have a very simple sip configuration
What could be the cause for this error?
One more question, I can see in the configuration an option to add a registrar using the data of SIP service provider, but what if I want my server to be the service provider… should I add this part in my configuration and why should we need another SIP provider?