Fw: [Asterisk-Users] Possible Sip logic bug?

Rich Adamson radamson at routers.com
Thu Feb 5 08:41:14 MST 2004 

Clif and all...

At the bottom of this post is the "sip show debug" for the problem.
The underlying problem (again): when C7960 hangs up on working conversation,
the C7960 sends a BYE to *, but * never sends a BYE to the sip gateway.

Any suggestions would be greatly appreciated.

Rich

> Try it again after executing: "sip debug" and give us the actual SIP 
> messages.  The devil
> is usually in the details. 
> 
> Rich Adamson wrote:
> 
> >Anyone have comments on this? Really could use some suggestions or ideas
> >why this is happening.  Thanks.
> >Rich
> >
> >------------------------
> >  
> >
> >>Anyone recognize this as a sip logic bug?
> >>
> >>Example Case:
> >> C7960 -> * -> sip gateway -> pstn
> >> (sip gateway config'ed with canreinvite=no, but shouldn't have an
> >>  impact on this.)
> >>
> >>Outgoing call initiated from C7960. Call is completed and conversation
> >>is very much normal. All equipment on the same wire; no nat.
> >>
> >>The C7960 user hangs up the phone. Pkt flows (as observed by sniffer)
> >>are:
> >>
> >>C7960 sends sip BYE packet to *
> >>  * returns 200 OK
> >>* sends INVITE to sip gateway    <<================ where is BYE?
> >>  sip gateway responds with 100 Trying
> >>  sip gateway responds with 200 OK
> >>  sip gateway responds with 200 OK
> >>  sip gateway responds with 200 OK
> >>
> >>The end result, the sip gateway does not drop the pstn line until the
> >>"called" number hangs up.
> >>
> >>It would appear that asterisk has an issue dropping the call. When the
> >>C7960 issues the BYE, I would expect * to send a BYE to the sip g/w.
> >>Is this a * logic problem (or my logic problem)?
> >>
> >>(I'm actually running CVS-12/04/03-14:24:40 and has been very stable
> >>in this production environment. Is it time to update this one even
> >>though it is 99% sip hardphone based?)
> >>
> >>Rich

----------------------------------------------------
Note: Call is already established from C7960 (193.92) via * (193.101) to
the sip gateway (193.109) which called cell phone 444-1234. The "sip show 
channels" was executed, followed by "sip debug", then hung up the C7960 
watching the results below. Note the C7960 sends the BYE and * confirms, 
but * never sends a BYE to the gateway. Sniffer on the wire confirms the 
exact same thing.

phoenix*CLI>
phoenix*CLI> sip show channels
Peer             User/ANR    Call ID      Seq (Tx/Rx)  Lag      Jitter  Format
222.111.193.109  4441234     66841295427  00103/00000  00000ms  0000ms  ULAW
222.111.193.92   3000        00036bc3-8b  00102/00103  00000ms  0000ms  ULAW
2 active SIP channel(s)

  == Spawn extension (from-sip, 64441234, 2) exited non-zero on 'SIP/3000-375c'
    -- Executing SetCIDNum("SIP/3000-ead2", "1111") in new stack
    -- Executing Dial("SIP/3000-ead2", "SIP/4441234 at 222.111.193.109") in new sta
ck
    -- Called 4441234 at 222.111.193.109
    -- SIP/222.111.193.109-fb6e answered SIP/3000-ead2
    -- Attempting native bridge of SIP/3000-ead2 and SIP/222.111.193.109-fb6e

SIP Debugging Enabled
Sip read: I>
BYE sip:64441234 at 222.111.193.101:5060 SIP/2.0
Via: SIP/2.0/UDP 222.111.193.92:5060
From: "NPI-Rich" <sip:3000 at 222.111.193.101>;tag=00036bc38b88045b25941469-0a0c5ae
b
To: <sip:64441234 at 222.111.193.101>;tag=as751f96fc
Call-ID: 00036bc3-8b886103-14d092b4-65542d42 at 222.111.193.92
Date: Thu, 05 Feb 2004 15:13:50 GMT
CSeq: 103 BYE
User-Agent: CSCO/6
Content-Length: 0
Proxy-Authorization: Digest username="3000",realm="asterisk",uri="sip:222.111.19
3.101",response="bb01af8f1eac65d392b68147867e79e6",nonce="7660e36e",algorithm=md
5
10 headers, 0 lines
Sending to 222.111.193.92 : 5060 (non-NAT)
Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 222.111.193.92:5060
From: "NPI-Rich" <sip:3000 at 222.111.193.101>;tag=00036bc38b88045b25941469-0a0c5ae
b
To: <sip:64441234 at 222.111.193.101>;tag=as751f96fc
Call-ID: 00036bc3-8b886103-14d092b4-65542d42 at 222.111.193.92
CSeq: 103 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:64441234 at 222.111.193.101>
Content-Length: 0
 to 222.111.193.92:5060
set_destination: Parsing <sip:464-3072 at 222.111.193.109:5060> for address/port to
 send to
set_destination: set destination to 222.111.193.109, port 5060
We're at 222.111.193.101 port 14308
Answering with preferred capability 4
Answering with capability 8
Answering with non-codec capability 1
11 headers, 10 lines

Reliably Transmitting:
INVITE sip:464-3072 at 222.111.193.109:5060 SIP/2.0
Via: SIP/2.0/UDP 222.111.193.101:5060;branch=z9hG4bK40ac3b17
From: "NPI-Rich" <sip:1111 at 222.111.193.101>;tag=as3310fadb
To: <sip:4441234 at 222.111.193.109>;tag=8c44b610-98bad313
Contact: <sip:1111 at 222.111.193.101>
Call-ID: 1132075c1322edc13a2df10e32eb2491 at 222.111.193.101
CSeq: 104 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 219
v=0
o=root 14743 14745 IN IP4 222.111.193.101
s=session
c=IN IP4 222.111.193.101
t=0 0
m=audio 14308 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
 (no NAT) to 222.111.193.109:5060
  == Spawn extension (from-sip, 64441234, 2) exited non-zero on 'SIP/3000-ead2'
Sip read: I>
SIP/2.0 100 Trying
Call-ID: 1132075c1322edc13a2df10e32eb2491 at 222.111.193.101
CSeq: 104 INVITE
From: NPI-Rich <sip:1111 at 222.111.193.101>;tag=as3310fadb
To: <sip:4441234 at 222.111.193.109>;tag=8c44b610-98bad313
Via: SIP/2.0/UDP 222.111.193.101:5060;branch=z9hG4bK40ac3b17
Content-Length: 0
7 headers, 0 lines

Sip read: I>
SIP/2.0 200 OK
Call-ID: 1132075c1322edc13a2df10e32eb2491 at 222.111.193.101
CSeq: 104 INVITE
From: NPI-Rich <sip:1111 at 222.111.193.101>;tag=as3310fadb
To: <sip:4441234 at 222.111.193.109>;tag=8c44b610-98bad313
Via: SIP/2.0/UDP 222.111.193.101:5060;branch=z9hG4bK40ac3b17
Content-Length: 212
Content-Type: application/sdp
Contact: <sip:464-3072 at 222.111.193.109:5060>
Allow: INVITE, ACK, BYE, CANCEL, REFER
v=0
o=MxSIP 0 0 IN IP4 222.111.193.109
s=SIP Call
c=IN IP4 222.111.193.109
t=0 0
m=audio 5004 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
10 headers, 10 lines

Sip read: I>
SIP/2.0 200 OK
Call-ID: 1132075c1322edc13a2df10e32eb2491 at 222.111.193.101
CSeq: 104 INVITE
From: NPI-Rich <sip:1111 at 222.111.193.101>;tag=as3310fadb
To: <sip:4441234 at 222.111.193.109>;tag=8c44b610-98bad313
Via: SIP/2.0/UDP 222.111.193.101:5060;branch=z9hG4bK40ac3b17
Content-Length: 212
Content-Type: application/sdp
Contact: <sip:464-3072 at 222.111.193.109:5060>
Allow: INVITE, ACK, BYE, CANCEL, REFER
v=0
o=MxSIP 0 0 IN IP4 222.111.193.109
s=SIP Call
c=IN IP4 222.111.193.109
t=0 0
m=audio 5004 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
10 headers, 10 lines

Sip read: I>
SIP/2.0 200 OK
Call-ID: 1132075c1322edc13a2df10e32eb2491 at 222.111.193.101
CSeq: 104 INVITE
From: NPI-Rich <sip:1111 at 222.111.193.101>;tag=as3310fadb
To: <sip:4441234 at 222.111.193.109>;tag=8c44b610-98bad313
Via: SIP/2.0/UDP 222.111.193.101:5060;branch=z9hG4bK40ac3b17
Content-Length: 212
Content-Type: application/sdp
Contact: <sip:464-3072 at 222.111.193.109:5060>
Allow: INVITE, ACK, BYE, CANCEL, REFER
v=0
o=MxSIP 0 0 IN IP4 222.111.193.109
s=SIP Call
c=IN IP4 222.111.193.109
t=0 0
m=audio 5004 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
10 headers, 10 lines

Sip read: I>
SIP/2.0 200 OK
Call-ID: 1132075c1322edc13a2df10e32eb2491 at 222.111.193.101
CSeq: 104 INVITE
From: NPI-Rich <sip:1111 at 222.111.193.101>;tag=as3310fadb
To: <sip:4441234 at 222.111.193.109>;tag=8c44b610-98bad313
Via: SIP/2.0/UDP 222.111.193.101:5060;branch=z9hG4bK40ac3b17
Content-Length: 212
Content-Type: application/sdp
Contact: <sip:464-3072 at 222.111.193.109:5060>
Allow: INVITE, ACK, BYE, CANCEL, REFER
v=0
o=MxSIP 0 0 IN IP4 222.111.193.109
s=SIP Call
c=IN IP4 222.111.193.109
t=0 0
m=audio 5004 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
10 headers, 10 lines

Sip read: I> 
SIP/2.0 200 OK
Call-ID: 1132075c1322edc13a2df10e32eb2491 at 222.111.193.101
CSeq: 104 INVITE
From: NPI-Rich <sip:1111 at 222.111.193.101>;tag=as3310fadb
To: <sip:4441234 at 222.111.193.109>;tag=8c44b610-98bad313
Via: SIP/2.0/UDP 222.111.193.101:5060;branch=z9hG4bK40ac3b17
Content-Length: 212
Content-Type: application/sdp
Contact: <sip:464-3072 at 222.111.193.109:5060>
Allow: INVITE, ACK, BYE, CANCEL, REFER
v=0
o=MxSIP 0 0 IN IP4 222.111.193.109
s=SIP Call
c=IN IP4 222.111.193.109
t=0 0
m=audio 5004 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
10 headers, 10 lines

SIP Debugging Disabled
phoenix*CLI>

More information about the asterisk-users mailing list