-- Executing [555@internal:1] Dial("SIP/username2-0000246e", "SIP/555") in new stack
-- Called SIP/555
-- SIP/555-0000246f is ringing
<--- Transmitting (no NAT) to 172.16.0.112:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 172.16.0.112;branch=z9hG4bKea11c1cf4C5903A;received=172.16.0.112
From: "username2" <sip:username2@172.16.0.189>;tag=5A4E4148-FBE49175
To: <sip:555@172.16.0.189;user=phone>;tag=as331bf488
Call-ID: fed0c14-3f5d3166-c1dfa16b@172.16.0.112
CSeq: 2 INVITE
Server: Asterisk PBX 13.6.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:555@172.16.0.189:5060>
Content-Length: 0
<------------>
<--- SIP read from UDP:172.16.0.112:5060 --->
REGISTER sip:172.16.0.189:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.0.112;branch=z9hG4bK52750173B936914E
From: "username2" <sip:username2@172.16.0.189>;tag=9AE6ED48-E971AD75
To: <sip:username2@172.16.0.189>
CSeq: 253 REGISTER
Call-ID: e0dbf814-b6437d66-1f52dd6b@172.16.0.112
Contact: <sip:username2@172.16.0.112>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
User-Agent: PolycomSoundPointIP-SPIP_301-UA/3.1.8.0070
Accept-Language: en
Authorization: Digest username="username2", realm="asterisk", nonce="76071db4", uri="sip:172.16.0.189:5060", response="04236469f2e07a3a004df77477cfa547", algorithm=MD5
Max-Forwards: 70
Expires: 60
Content-Length: 0
<------------->
--- (13 headers 0 lines) ---
Sending to 172.16.0.112:5060 (no NAT)
[Jul 11 12:32:08] NOTICE[100895]: chan_sip.c:16713 check_auth: Correct auth, but based on stale nonce received from '"username2" <sip:username2@172.16.0.189>;tag=9AE6ED48-E971AD75'
<--- Transmitting (no NAT) to 172.16.0.112:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.16.0.112;branch=z9hG4bK52750173B936914E;received=172.16.0.112
From: "username2" <sip:username2@172.16.0.189>;tag=9AE6ED48-E971AD75
To: <sip:username2@172.16.0.189>;tag=as66ba7f6d
Call-ID: e0dbf814-b6437d66-1f52dd6b@172.16.0.112
CSeq: 253 REGISTER
Server: Asterisk PBX 13.6.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2371669c", stale=true
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog 'e0dbf814-b6437d66-1f52dd6b@172.16.0.112' in 32000 ms (Method: REGISTER)
<--- SIP read from UDP:172.16.0.112:5060 --->
REGISTER sip:172.16.0.189:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.0.112;branch=z9hG4bK538bf1bd2BBDA070
From: "username2" <sip:username2@172.16.0.189>;tag=9AE6ED48-E971AD75
To: <sip:username2@172.16.0.189>
CSeq: 254 REGISTER
Call-ID: e0dbf814-b6437d66-1f52dd6b@172.16.0.112
Contact: <sip:username2@172.16.0.112>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
User-Agent: PolycomSoundPointIP-SPIP_301-UA/3.1.8.0070
Accept-Language: en
Authorization: Digest username="username2", realm="asterisk", nonce="2371669c", uri="sip:172.16.0.189:5060", response="8f9a41b584d5df43e47b3cb484c071ff", algorithm=MD5
Max-Forwards: 70
Expires: 60
Content-Length: 0
<------------->
--- (13 headers 0 lines) ---
Sending to 172.16.0.112:5060 (no NAT)
<--- Transmitting (no NAT) to 172.16.0.112:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.0.112;branch=z9hG4bK538bf1bd2BBDA070;received=172.16.0.112
From: "username2" <sip:username2@172.16.0.189>;tag=9AE6ED48-E971AD75
To: <sip:username2@172.16.0.189>;tag=as66ba7f6d
Call-ID: e0dbf814-b6437d66-1f52dd6b@172.16.0.112
CSeq: 254 REGISTER
Server: Asterisk PBX 13.6.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 60
Contact: <sip:username2@172.16.0.112>;expires=60
Date: Tue, 11 Jul 2017 09:32:08 GMT
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog 'e0dbf814-b6437d66-1f52dd6b@172.16.0.112' in 32000 ms (Method: REGISTER)