Allowguest Issue

Asterisk Asterisk Support
1

Hi Support,

When I set allowguest=no in sip.conf like

[general]
allowguest=no

the incoming calls won’t work ( no extensions ring ) most of the time. ( It was noticed that sometimes it works, sometimes it won’t ).

The error that I get is,

[Dec 12 12:14:02] NOTICE[13054]: chan_sip.c:22518 handle_request_invite: Sending fake auth rejection for device <sip:XXXXXXXX@123.123.123.123:5060>;tag=as01b5d246

When debug is on,

Using INVITE request as basis request - 2523181-3595848916-125555@ms
No matching peer for 'yyyyyyyyyyy' from '123.123.123.123:5060'
[Dec 12 15:54:21] NOTICE[22405]: chan_sip.c:22518 handle_request_invite: Sending fake auth rejection for device <sip:xxxxxxxxx@123.123.123.123:5060>;tag=as2498b19c

<--- Reliably Transmitting (NAT) to 123.123.123.123:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 123.123.123.123:5060;branch=z9hG4bK-ea504d19c5e893a2e709bac179b7dd9c;received=123.123.123.123;rport=5060

If I set allowguest=yes, then the issue is fixed, and all incoming calls work all times.

I see that allowguest when enabled is a security risk. How can I make the calls work, when allowguest is set to “no”? Please help.

2

Register your peers properly to avoid issues or enable allowguest=yes.

3

I think you meant “or” not “and”, or you meant allowguest=no.

4

Right, it was “or”

5

The peers are registered properly. I confirm it using “sip show registry” and “sip show peers”.

What else can go wrong?

6

“sip show registry” shows to which servers ASTERISK registers to (does not show the registration status of the SIP endpoints).

Can you copy/paste the output of “sip show peers”?

7

This is my result of “sip show peers”

asterisk*CLI> sip show peers Name/username Host Dyn Forcerport ACL Port Status 7001/7001 203.203.203.203 D N A 16941 Unmonitored 7002/7002 (Unspecified) D N 0 Unmonitored 7003/7003 (Unspecified) D N 0 Unmonitored callcentric/1777236XXXX 204.204.204.204 N 5080 Unmonitored 4 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 2 offline] asterisk*CLI>

8

Neither 7002 nor 7003 are registered.

9

7002 and 7003 are softphones which are not connected.

( But it has worked in the past with only 7001 connected )

10

Can you do a “sip set debug on”, make an incomming call and copy/paste the output here?