INBOUND /DID failed authentication

[ashis103]

hello team

would you please help me out DID authentication issue NB ERR:Failed to authenticate use. If i dialed from outside in my did asterisk cli showing below message.

[Jan 15 09:20:49] NOTICE[2705]: chan_sip.c:15733 handle_request_invite: Failed to authenticate user <sip:+442033720303@79.99.193.70:5060>;tag=1501201314281033047238
[Jan 15 09:20:49] NOTICE[2705]: chan_sip.c:15733 handle_request_invite: Failed to authenticate user <sip:+442033720303@79.99.193.70:5060>;tag=1501201314281033047316

[ZoVoS]

host=79.99.193.70 in your carriers "Account Entry"???

[ashis103]

yes.IP 79.99.193.70 is our carrier

[steve745]

Your not registered correctly. I have seen this when registered at 2 locations when using 1 server as a temp to hold the other server traffic during upgrade. Try contacting your sip provider and ask if they do IP authenication. Else Check your carrier section in VICI or Goautodial and make sure you have register string...

register => XXXXXXX:xxxxxx@79.99.193.70/XXXXXXX
register => XXXXXXX:xxxxxx@79.99.193.70:5060/XXXXXXX

XXXXXXX= Sip user credentials
xxxxxxx=Sip secret

If you still have problems plz post full sys info build version type of install

[ZoVoS]

you don't need to be registered to accept inbound.

Below is the minimum required to handle an inbound call

insecure
http://www.voip-info.org/wiki/view/Aste ... p+insecure

type (atleast peer)
http://www.voip-info.org/wiki/view/Asterisk+sip+type

context=trunkinbound
So Asterisk knows where to send the call

host=***.***.***.***
http://www.voip-info.org/wiki/view/Asterisk+sip+host

For example a codex without registration;

[BlueIN]
insecure=invite
type=peer
context=trunkinbound
host=***.***.***.***
canreinvite=no
qualify=yes
disallow=all
allow=alaw



Can you please provide me with what your insecure and type options are set to?
If host=79.99.193.70 then I am unsure as to why his calls are not authenticating.

[williamconley]

It is likely the + on inbound is breaking something. (+442033720303)

You can modify [trunkinbound] in extensions.conf which has a section to remove the + (we have a couple posts dealing with this as well, just stripping the + so it will match a DID).

But that may never get a chance to fire since the system says it will not authenticate. Any time authentication becomes an issue, you should modify your carrier SIP context to include the IP address and NOT include a user/pass or to have "insecure=invite" as mentioned above.

Registration is at the discretion of the carrier. Some allow you to "point" the DID to your server's IP address, some do not. If they require registration that does not mean you need to put a user/pass in the SIP context for that carrier. The register => entry is completely separate from the [context] for that sip connection and can even have different authentication information. I have a carrier who requires this as we register to one server, get calls from another server (several, actually) and send calls through yet another server (or two, depending on the coast of OUR server).

Learning how to make it work for each individual carrier can be painstaking, but it often helps if you just call the carrier and ... ask.