481 response to BYE request

Asterisk Asterisk SIP
1

Hello,
I’m new to this forum, so I’m not sure if it’s the place where I should ask this question.

I’m developing a simple voip application. I can call to other users, but can’t hang up. When client sends BYE request, server answers with 481 - call leg transaction does not exist. Here are client logs:

    INVITE sip:2@asteriskip:51110;transport=UDP SIP/2.0
    Via: SIP/2.0/UDP asteriskip:5060;branch=z9hG4bK06952c7a;rport
    Max-Forwards: 70
    From: "First" <sip:1@asteriskip>;tag=as746cc61d
    To: <sip:2@asteriskip:51110;transport=UDP>
    Contact: <sip:1@asteriskip:5060>
    Call-ID: 425bb181009f366c499b10f362d29ac6@asteriskip:5060
    CSeq: 102 INVITE
    User-Agent: FPBX-AsteriskNOW-12.0.76.4(11.16.0)
    Date: Tue, 07 Mar 2017 11:52:32 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
    Supported: replaces, timer
    P-Asserted-Identity: "First" <sip:1@asteriskip>
    Content-Type: application/sdp
    Content-Length: 310

v=0
o=root 2015785808 2015785808 IN IP4 asteriskip
s=Asterisk PBX 11.16.0
c=IN IP4 asteriskip
t=0 0
m=audio 13952 RTP/AVP 0 8 3 111 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP asteriskip:51110;branch=z9hG4bKnCqg
Contact: <sip:2@asteriskip:51110;transport=UDP>
To: <sip:2@asteriskip;transport=UDP>;tag=YU2R
From: <sip:2@asteriskip;transport=UDP>;tag=as746cc61d
Call-ID: 425bb181009f366c499b10f362d29ac6@asteriskip:5060
CSeq: 102 INVITE
Allow-Events: presence, kpml, talk
Content-Length: 0

SIP/2.0 200 OK
Via: SIP/2.0/UDP asteriskip:51110;branch=z9hG4bK06952c7a;rport
Contact: <sip:2@asteriskip:51110;transport=UDP>
To: <sip:2@asteriskip;transport=UDP>;tag=YU2R
From: "First" <sip:1@asteriskip;transport=>;tag=as746cc61d
Call-ID: 425bb181009f366c499b10f362d29ac6@asteriskip:5060
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Accept: application/sdp, application/sdp
Accept-Language: en
Content-Type: application/sdp
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
Allow-Events: presence, kpml, talk
Content-Length: 247

v=0
o=Z 13 13 IN IP4 myip
s=Test
c=IN IP4 myip
t=0 0
m=audio 50000 RTP/AVP 3 110 8 0 97 101
a=rtpmap:110 speex/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode = 30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
ACK sip:2@asteriskip:51110;transport=UDP SIP/2.0
Via: SIP/2.0/UDP asteriskip:5060;branch=z9hG4bK54ba94a0;rport
Max-Forwards: 70
From: "First" <sip:1@asteriskip>;tag=as746cc61d
To: <sip:2@asteriskip:51110;transport=UDP>;tag=YU2R
Contact: <sip:1@asteriskip:5060>
Call-ID: 425bb181009f366c499b10f362d29ac6@asteriskip:5060
CSeq: 102 ACK
User-Agent: FPBX-AsteriskNOW-12.0.76.4(11.16.0)
Content-Length: 0

BYE sip:1@asteriskip;transport=UDP SIP/2.0
Via: SIP/2.0/UDP asteriskip:51110;branch=z9hG4bKAZsQ
Max-Forwards: 70
From: <sip:2@asteriskip;transport=UDP>;tag=as746cc61d
To: <sip:1@asteriskip;transport=UDP>;tag=YU2R
Contact: <sip:2@asteriskip:51110;transport=UDP>
CSeq: 2 BYE
User-Agent: TestSoftphone
Call-ID: 425bb181009f366c499b10f362d29ac6@asteriskip:5060
Content-Length: 0

SIP/2.0 481 Call leg/transaction does not exist
Via: SIP/2.0/UDP asteriskip:51110;branch=z9hG4bKAZsQ;received=172.20.1.40;rport=51110
From: <sip:2@asteriskip;transport=UDP>;tag=as746cc61d
To: <sip:1@asteriskip;transport=UDP>;tag=YU2R
Call-ID: 425bb181009f366c499b10f362d29ac6@asteriskip:5060
CSeq: 2 BYE
Server: FPBX-AsteriskNOW-12.0.76.4(11.16.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

I read that 481 can occur if To tag or From tag or Call-ID is outside of dialog, but looks like tags and ids are ok. I don’t receive a tag for To, so I generate Ringing packet where To tag is set, that may be the problem.
EDIT. I added “pedantic=no” to sip_custom.conf and it works now. Though, I don’t know what’s wrong with this requests.

2

Please mark your log as unformatted text. The contents of the Contact header are being interpreted as markup, but may well be relevant to the problem.

3

Done. I also found a workaround, but have no idea why it didn’t work with default asterisk settings.

4

The BYE has been sent to the wrong port and its via contains the wrong IP address.

The Contact header from Asterisk says use port 51110, but the BYE has been addressed to 5060. Asterisk would only send 51110 if that was the port it used for all rquests, so I wounder if the log is from the far side of a broken router.

It might be giving 481 because the ACK was sent to the requested port but never arrived.

The IP address should be that of the request sender, not Asterisk.

5

51110 is my port, 5060 is server port. I think the problem was with the headers, because setting “pedantic” to “no” fixed it.
Thanks for your response.