[asterisk-bugs] [JIRA] (ASTERISK-28243) Corrupted SIP after handling a 302 redirect

[lvl (JIRA)]

lvl created ASTERISK-28243:
------------------------------

             Summary: Corrupted SIP after handling a 302 redirect
                 Key: ASTERISK-28243
                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-28243
             Project: Asterisk
          Issue Type: Bug
      Security Level: None
          Components: Channels/chan_pjsip
    Affects Versions: 16.1.1
            Reporter: lvl


After handling a 302 redirect, Asterisk starts sending corrupted SIP headers for the remainder of that call.

This did not occur in 16.0.1 and is reproducible for me 95% of the time with a basic scenario of A calling B, B redirecting to C, and C answering.

A couple of examples of the SIP transmitted by Asterisk, taken directly from the console after "pjsip set logger on":

{quote}
SIP/2.0 200 OK
Via: SIP/2.0/UDP ip:8065;rport=8065;received=ip;branch=z9hG4bK1693.a19da1224aba5ecdfadfc7877b4ed0ed.0
Via: SIP/2.0/UDP ip;branch=z9hG4bK1693.19407339547949d0aa707e6d2062277c.0
Via: SIP/2.0/UDP ip:48708;rport=48708;received=ip;branch=z9hG4bKPjae31d1a5-ce0e-4075-9024-4ea10e2bf316
Record-Route: <sip:ip:8065;lr;ftag=d172a719-a571-41b4-ac0e-9426e6cdacfd;did=de9.8851>
Record-Route: <sip:ip;lr>
Call-ID: 85fc438f-e060-4235-a12b-8abf12d194ac
From: <sip:phone_17940_0 at domain.com>;tag=d172a719-a571-41b4-ac0e-9426e6cdacfd
To: <sip:202 at domain.com>;tag=b04b4eb8-e953-46a4-a00a-df91dadc8b14
CSeq: 4124 INVITE
Server: PBX
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REFER, MESSAGE

{quote}

{quote}
SIP/2.0 200 OK
Via: SIP/2.0/UDP ip:8065;rport=8065;received=ip;branch=z9hG4bK4087.5f49d7eba2c76f74284867007fbcd30a.0
Via: SIP/2.0/UDP ip;branch=z9hG4bK4087.606cdb260641f1475c946106a64bc3e6.0
Via: SIP/2.0/UDP ip:41084;rport=41084;received=ip;branch=z9hG4bKPj07877751-1fc3-4870-96f5-3f07228810da
Record-Route: <sip:ip:8065;lr;ftag=64708ee0-bcd2-4baa-854a-2bc12b2f210e;did=88b.ab92>
Record-Route: <sip:ip;lr>
Call-ID: 007d3cb6-8304-4ae6-a3c7-f65a8e42a3d9
From: <sip:phone_17940_0 at domain.com>;tag=64708ee0-bcd2-4baa-854a-2bc12b2f210e
To: <sip:202 at domain.com>;tag=23944ff9-c67b-417f-8b60-f38295921932
CSeq: 1568 INVITE
Server: PBX
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REFER, MESSAGE
phone_179: <sip:~~s~~@domain.com>;reason=unknown
{quote}

{quote}
SIP/2.0 200 OK
Via: SIP/2.0/UDP ip:8065;rport=8065;received=ip;branch=z9hG4bK18d.87385f4d3e56bdb07d67390e49ebbd65.0
Via: SIP/2.0/UDP ip;branch=z9hG4bK18d.f9ad907dee3ca8576b87347a65443bf8.0
Via: SIP/2.0/UDP ip:45827;rport=45827;received=ip;branch=z9hG4bKPjbadc1081-ef39-4ef2-9fc3-332b35b57d4e
Record-Route: <sip:ip:8065;lr;ftag=ed214f4a-56ae-4285-a911-dfc79925f0d1;did=0e.0be2>
Record-Route: <sip:ip;lr>
Call-ID: 6f608fd0-1355-427a-9a81-e07997730698
From: <sip:phone_17940_0 at domain.com>;tag=ed214f4a-56ae-4285-a911-dfc79925f0d1
To: <sip:202 at domain.com>;tag=053682e9-3c8a-4be2-962e-4cb9d51779b9
CSeq: 15645 INVITE
Server: PBX
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REFER, MESSAGE
P
{quote}



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