I’m sure that my userid and password are correct.
nat=yes option in sip.conf is enabled.
Incoming calls are allright.
This is the output of the asterisk cli (I censored the phone numbers).
Using SIP RTP CoS mark 5
– Executing [-----------@Internal:1] Dial(“SIP/24-0000004d”, “SIP/---------@sip.messagenet.it,TtKk”) in new stack
== Using SIP RTP CoS mark 5
– Called -----------@sip.messagenet.it
[Feb 15 11:49:14] NOTICE[2090]: chan_sip.c:16913 handle_response_invite: Failed to authenticate on INVITE to ‘“telefonoIP” sip:----------@192.168.0.200;tag=as2b0448b5’
– SIP/sip.messagenet.it-0000004e is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
– Executing [-------------@Internal:2] Hangup(“SIP/24-0000004d”, “”) in new stack
== Spawn extension (Internal, ---------, 2) exited non-zero on ‘SIP/24-0000004d’
Sometimes i had this other kind of problem, so i loose the registration and incoming call won’t work…
I replaced my account name with "------"
This message is repeated a lot of times.
[Feb 15 16:54:56] NOTICE[1982]: chan_sip.c:10669 sip_reg_timeout: – Registration for '-------@sip.messagenet.it’ timed out, trying again (Attempt #13284)
[Feb 15 16:54:57] WARNING[1982]: chan_sip.c:10793 transmit_register: Probably a DNS error for registration to ---------@sip.messagenet.it, trying REGISTER again (after 20 seconds)
Related to the previous problem, could it be a nat issue?
I disabled the firewall and I opened 5060 and 5061 tcp port.
I can’t understand…
When you make a call, if the service provider requires authentication, they will need your user ID somewhere. The default location of that is the same as the default location for outgoing calling line ID. Even if you have a correct user ID for REGISTER, you also need to make sure that the user ID for outgoing calls is in the right place. This might mean using RPID (remote party ID) for CLI, or it might mean that you cannot set the CLI to anything except the name by which you are registered.
Your service provider ought to be able to tell you how to configure Asterisk for use with their service. It is also possible, if you tell them that you are using Asterisk, they will make adjustments at their end.