Static trunk not allowing incoming calls

Hi im setting up a new pbx server using peerless as the providers, so far i have gotten outbound calls to work but i am unable to get in bound calls to work, when i call in and look at the logs in freepbx nothing pops up, but i was able to do a packet capture and i got this

 [ ] #2 03/08/2022 16:40:09.736 len:583/625 in:X0*(i) out:X1 UDP 10.XXX.XXX.75:5060->208.XXX.XXX.160:5060 [flags:]
Forwarded
 *SIP/2.0 401 Unauthorized..Via: SIP/2.0/UDP 208.XXX.XXX.160:5060;rport=5060;received=208.XXX.XXX.160;*
 *branch=z9hG4bK-XXXXXXX-XXXX-XXXXXX-7ff6121b69a0..Call-ID: 5260e523-XXXXXXXXXXXX-XXXXXXX7ff60878910*
 *0-e34cdfcf-13c4-7225..From: <sip:[email protected]>;tag=s5322XXXXXXXXXXX77013..To: <s*
 *ip:[email protected]>;tag=z9hG4bK-XXXXXXX-XXXXXX-74d7d25-7ff6121b69a0..CSeq: 1 INVITE..*
 *WWW-Authenticate: Digest realm="asterisk",nonce="164XXX9209/f1e1a15ab82f8XXXXXXXXXXb4b",op*
 *aque="2357cXXXXXXXX23b",algorithm=md5,qop="auth"..Server: FPBX-16.0.19(16.24.0)..Content-Length:*
 *  0....*

my outbound sip settings are

type=peer
host=208.XXX.XXX.160
port=5060
insecure=port,invite
dtmfmode=rfc2833
disallow=all
allow=ulaw

my inbound sip settings per there documentation is blank

It is not clear where your log was taken from, it is garbled, and the redaction rules have not been explained. Generally people want logs taken from /var/log/asterisk/full, with ‘sip set debug on’ enabled, and uploaded to pastebin.freepbx.org/ SIP protocol is formed or carriage return-line feed printable text lines.

You quote your “outbound” settings, but this is an inbound call, and insecure only affects inbound calls. Also, insecure=invite does nothing unless there is a secret, and insecure=port is rarely needed for UDP. The fact that Asterisk is asking for authentication for a source which has either or both of no secret and insecure=invite, suggest it is not matching the peer entry, for some reason.

chan_sip is deprecated, unsupported, and scheduled for removal next year.

(In simple cases, FreePBX doesn’t need separate inbound settings, so outbound can be a misnomer.)

SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 208.XXX.XXX.160:5060;rport=5060;received=208.XXX.XXX.160;branch=z9hG4bK-XXXXXXX-XXXX-XXXXXX-7ff6121b69a0
Call-ID: 5260e523-XXXXXXXXXXXX-XXXXXXX7ff608789100-e34cdfcf-13c4-7225
From: <sip:[email protected]>;tag=s5322XXXXXXXXXXX77013
To: <sip:[email protected]>;tag=z9hG4bK-XXXXXXX-XXXXXX-74d7d25-7ff6121b69a0
CSeq: 1 INVITE
WWW-Authenticate: Digest realm="asterisk",nonce="164XXX9209/f1e1a15ab82f8XXXXXXXXXXb4b",opaque="2357cXXXXXXXX23b",algorithm=md5,qop="auth"
Server: FPBX-16.0.19(16.24.0)
Content-Length:  0


The log first log was taken off of our firewall, as for the redaction its just to hide our ip, phone numbers and personal information to prevent spam. I will try to get a full log with sip debugging on, also i will try seeing what happens when i remove those outbound settings that you suggested, but that was what was in the documentation for setting up the trunk by peerless

What I was getting at regarding redaction is that you need to say what the various redacted strings represent, and confirm that different addresses don’t get transformed to the same redacted form. (It can also be important to distinguish between public and private addresses, but I think all yours are public.)

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