Asterisk failed to authenticate- inbound call

jcolp · April 5, 2021, 5:13pm

I only see an outbound SIP REGISTER, I don’t see the responses to it.

basimkhan · April 5, 2021, 5:17pm

<--- Transmitting SIP request (582 bytes) to UDP:158.85.70.148:5060 --->
REGISTER sip:toronto1.voip.ms:5060 SIP/2.0
Via: SIP/2.0/UDP 172.31.29.184:5060;rport;branch=z9hG4bKPje3f7bf12-2436-4749-bcb6-e0bb476d555a
From: <sip:132688_test@toronto1.voip.ms>;tag=beeb27e3-2086-46fd-a05d-d294c66f6320
To: <sip:132688_test@toronto1.voip.ms>
Call-ID: 879fa14c-cd88-4b4a-acce-a6fbb737c30b
CSeq: 58602 REGISTER
Contact: <sip:s@172.31.29.184:5060;line=azabuve>
Expires: 3600
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Max-Forwards: 70
User-Agent: Asterisk PBX 16.16.2
Content-Length:  0


<--- Received SIP response (600 bytes) from UDP:158.85.70.148:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.31.29.184:5060;branch=z9hG4bKPje3f7bf12-2436-4749-bcb6-e0bb476d555a;received=35.183.128.250;rport=5060
From: <sip:132688_test@toronto1.voip.ms>;tag=beeb27e3-2086-46fd-a05d-d294c66f6320
To: <sip:132688_test@toronto1.voip.ms>;tag=as2cd5e305
Call-ID: 879fa14c-cd88-4b4a-acce-a6fbb737c30b
CSeq: 58602 REGISTER
Server: voip.ms
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="toronto1.voip.ms", nonce="3d018a6b"
Content-Length: 0


<--- Transmitting SIP request (765 bytes) to UDP:158.85.70.148:5060 --->
REGISTER sip:toronto1.voip.ms:5060 SIP/2.0
Via: SIP/2.0/UDP 172.31.29.184:5060;rport;branch=z9hG4bKPj0b24809f-2c85-4c6e-8325-049437eef852
From: <sip:132688_test@toronto1.voip.ms>;tag=beeb27e3-2086-46fd-a05d-d294c66f6320
To: <sip:132688_test@toronto1.voip.ms>
Call-ID: 879fa14c-cd88-4b4a-acce-a6fbb737c30b
CSeq: 58603 REGISTER
Contact: <sip:s@172.31.29.184:5060;line=azabuve>
Expires: 3600
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Max-Forwards: 70
User-Agent: Asterisk PBX 16.16.2
Authorization: Digest username="132688_test", realm="toronto1.voip.ms", nonce="3d018a6b", uri="sip:toronto1.voip.ms:5060", response="da9589734bbb1437db495a29544090f8", algorithm=MD5
Content-Length:  0


<--- Received SIP response (621 bytes) from UDP:158.85.70.148:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.31.29.184:5060;branch=z9hG4bKPj0b24809f-2c85-4c6e-8325-049437eef852;received=35.183.128.250;rport=5060
From: <sip:132688_test@toronto1.voip.ms>;tag=beeb27e3-2086-46fd-a05d-d294c66f6320
To: <sip:132688_test@toronto1.voip.ms>;tag=as2cd5e305
Call-ID: 879fa14c-cd88-4b4a-acce-a6fbb737c30b
CSeq: 58603 REGISTER
Server: voip.ms
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 3600
Contact: <sip:s@172.31.29.184:5060;line=azabuve>;expires=3600
Date: Mon, 05 Apr 2021 17:15:52 GMT
Content-Length: 0


<--- Transmitting SIP request (588 bytes) to UDP:158.85.70.148:5060 --->
REGISTER sip:toronto1.voip.ms:5060 SIP/2.0
Via: SIP/2.0/UDP 172.31.29.184:5060;rport;branch=z9hG4bKPjc7c3ea3b-279e-45d4-896d-cc564e52cfde
From: <sip:132688_atu-107@toronto1.voip.ms>;tag=56f0c5eb-9e22-40fb-b29c-a09a9cebb189
To: <sip:132688_atu-107@toronto1.voip.ms>
Call-ID: 51fdd8f6-b69a-4f68-9d23-def6ff9484f9
CSeq: 42310 REGISTER
Contact: <sip:s@172.31.29.184:5060;line=rgrwctv>
Expires: 3600
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Max-Forwards: 70
User-Agent: Asterisk PBX 16.16.2
Content-Length:  0


<--- Received SIP response (606 bytes) from UDP:158.85.70.148:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.31.29.184:5060;branch=z9hG4bKPjc7c3ea3b-279e-45d4-896d-cc564e52cfde;received=35.183.128.250;rport=5060
From: <sip:132688_atu-107@toronto1.voip.ms>;tag=56f0c5eb-9e22-40fb-b29c-a09a9cebb189
To: <sip:132688_atu-107@toronto1.voip.ms>;tag=as7dc54241
Call-ID: 51fdd8f6-b69a-4f68-9d23-def6ff9484f9
CSeq: 42310 REGISTER
Server: voip.ms
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="toronto1.voip.ms", nonce="1b876f37"
Content-Length: 0


<--- Transmitting SIP request (774 bytes) to UDP:158.85.70.148:5060 --->
REGISTER sip:toronto1.voip.ms:5060 SIP/2.0
Via: SIP/2.0/UDP 172.31.29.184:5060;rport;branch=z9hG4bKPjee1f1d74-1de7-4032-8326-3be52576b4c0
From: <sip:132688_atu-107@toronto1.voip.ms>;tag=56f0c5eb-9e22-40fb-b29c-a09a9cebb189
To: <sip:132688_atu-107@toronto1.voip.ms>
Call-ID: 51fdd8f6-b69a-4f68-9d23-def6ff9484f9
CSeq: 42311 REGISTER
Contact: <sip:s@172.31.29.184:5060;line=rgrwctv>
Expires: 3600
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Max-Forwards: 70
User-Agent: Asterisk PBX 16.16.2
Authorization: Digest username="132688_atu-107", realm="toronto1.voip.ms", nonce="1b876f37", uri="sip:toronto1.voip.ms:5060", response="e4ca494dfed207a2f9e75f6068aec099", algorithm=MD5
Content-Length:  0


<--- Received SIP response (627 bytes) from UDP:158.85.70.148:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.31.29.184:5060;branch=z9hG4bKPjee1f1d74-1de7-4032-8326-3be52576b4c0;received=35.183.128.250;rport=5060
From: <sip:132688_atu-107@toronto1.voip.ms>;tag=56f0c5eb-9e22-40fb-b29c-a09a9cebb189
To: <sip:132688_atu-107@toronto1.voip.ms>;tag=as7dc54241
Call-ID: 51fdd8f6-b69a-4f68-9d23-def6ff9484f9
CSeq: 42311 REGISTER
Server: voip.ms
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 3600
Contact: <sip:s@172.31.29.184:5060;line=rgrwctv>;expires=3600
Date: Mon, 05 Apr 2021 17:15:53 GMT
Content-Length: 0

This is all of the registration log I believe. I think the logs for the call were in my last reply, please let me know if I am still missing something.

jcolp · April 5, 2021, 5:21pm

According to the outbound registration the line parameter is present in the registration and in the Contact. If the SIP INVITE does not contain this (which it does not appear to) then it is being removed/discarded/not used and line support can’t be used.

All calls would have to use an identify. You can either match all traffic to a single endpoint based on source address, or you could try using the match_header option[1] on the identify and matching based on the “X-Dest-User” SIP header such as:

[voipms-atu-107]
type=identify
match_header=X-Dest-User: 132688_atu-107
endpoint=something

[1] asterisk/pjsip.conf.sample at master · asterisk/asterisk · GitHub

basimkhan · April 5, 2021, 7:02pm

Hi, Thanks a lot, that worked. I was able to get it to work with the match_header. Would X-dest-user always be the sub account I use with my voip provider?

Thank you again, you really helped a lot. I very much appreciate it.

david551 · April 5, 2021, 7:28pm

X- means it is non-standard. Most providers won’t send it and for those that do, it means whatever they want it to mean.

system · May 5, 2021, 7:29pm

This topic was automatically closed 30 days after the last reply. New replies are no longer allowed.

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Asterisk failed to authenticate- inbound call

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