Hi i have already working sip trunk with my provider
and its works ok for inccomming and outgoing
but they told me that i still send them invite signal to dst 400 their side .
and they want me to stop that
this occurs for the in coming calls only .
here is my sip trunk info :
type=friend
canredirect=no
canreinvite=no
insecure=very
host=10.0.0.01
dtmfmode=rfc2833
disallow=all
allow=ulaw&alaw&gsm
If you mean port 400, you need to fix your NAT router, or tell them not to specify it in the Contact header on the INVITE response. With your configuration, Asterisk will only send to port 400 if that Contact header specifies it, either because that is what they sent, or that is what your router corrupted it to.
If you mean the request URI, that will depend on your dialplan and what was dialled, but you would want that to be variable
Note that type=friend is bad practice, use type=peer, and that canreinvite and insecure=very are deprecated or unsupported in currently supported versions of Asterisk, although neither affects the port used. Use directmedia for canreinvite. Use insecure=invite, or insecure=port,invite, depending on the minimum that is really needed. It is clearer to use remotesecret, instead of secret, rather than to use insecure=invite.
Maybe the ISP is talking about the SIP response code 400:
400 Bad Request
The request could not be understood due to malformed syntax. The Reason-Phrase SHOULD identify the syntax problem in more detail, for example, “Missing Call-ID header field”
Ask them to capture a packet and send it to you, so you can anylize it. Maybe it helps.
You should get a SIP trace, if they are talking about destination port this is the answer:
[quote=“david55”]If you mean port 400, you need to fix your NAT router, or tell them not to specify it in the Contact header on the INVITE response. With your configuration, Asterisk will only send to port 400 if that Contact header specifies it, either because that is what they sent, or that is what your router corrupted it to.
[/quote]