[Asterisk-Dev] sunday's CVS-STABLE breaks SIP

Tue Mar 22 10:49:49 MST 2005


> Matthew Simpson wrote:
>> 1.0.7 CVS-STABLE breaks SIP with certain gateways.  Particularly Nex-tone 
>> and Sysmaster.
>>
>> Rolling back to 1.0.6 fixes the problem.
>
> Yes, there have been other reports of SIP problems with 1.0.7 as well. SIP 
> debug traces and Asterisk log traces would be most helpful!

Kevin, see the attached SIP trace.  This is a toll-free number terminating 
to the sysmaster.  The spurious BYE seems to be part of the problem. 
-------------- next part --------------

Sip read: 
INVITE sip:8668692267 at 67.137.224.13 SIP/2.0
Via: SIP/2.0/UDP 67.137.224.34:5060;branch=z9hG4bK73c4eeca
From: "9726179231" <sip:9726179231 at 67.137.224.34>;tag=as2c993cc2
To: <sip:8668692267 at 67.137.224.13>
Contact: <sip:9726179231 at 67.137.224.34>
Call-ID: 00e19f2171cb25c0757a3e5d53590794 at 67.137.224.34
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Tue, 22 Mar 2005 01:18:30 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 218

v=0
o=root 28959 28959 IN IP4 67.137.224.34
s=session
c=IN IP4 67.137.224.34
t=0 0
m=audio 14294 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

12 headers, 10 lines
Using latest request as basis request
Sending to 67.137.224.34 : 5060 (non-NAT)
Found no matching peer or user for '67.137.224.34:5060'
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 67.137.224.34:14294
Found description format PCMU
Found description format telephone-event
Capabilities: us - 0x106 (gsm|ulaw|g729), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723)
Looking for 8668692267 in noauth
list_route: hop: <sip:9726179231 at 67.137.224.34>
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 67.137.224.34:5060;branch=z9hG4bK73c4eeca
From: "9726179231" <sip:9726179231 at 67.137.224.34>;tag=as2c993cc2
To: <sip:8668692267 at 67.137.224.13>;tag=as68cbfb62
Call-ID: 00e19f2171cb25c0757a3e5d53590794 at 67.137.224.34
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:8668692267 at 67.137.224.13>
Content-Length: 0


 to 67.137.224.34:5060
We're at 67.137.224.13 port 14298
Answering/Requesting with root capability 0x4 (ulaw)
Answering with preferred capability 0x100 (g729)
Answering with preferred capability 0x2 (gsm)
Answering with non-codec capability 0x1 (telephone-event)
12 headers, 12 linesg
Reliably Transmitting:
INVITE sip:8668692267 at 67.137.224.102 SIP/2.0
Via: SIP/2.0/UDP 67.137.224.13:5060;branch=z9hG4bK48b82637
From: "9726179231" <sip:9726179231 at 67.137.224.13>;tag=as04a9d8ca
To: <sip:8668692267 at 67.137.224.102>
Contact: <sip:9726179231 at 67.137.224.13>
Call-ID: 6fcc94a072e1e3003087c22153de269d at 67.137.224.13
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Tue, 22 Mar 2005 01:23:47 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 267

v=0
o=root 31804 31804 IN IP4 67.137.224.13
s=session
c=IN IP4 67.137.224.13
t=0 0
m=audio 14298 RTP/AVP 0 18 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
 (no NAT) to 67.137.224.102:5060
    -- Called 8668692267 at 67.137.224.102

Sip read: 
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 67.137.224.13:5060;branch=z9hG4bK48b82637
From: "9726179231" <sip:9726179231 at 67.137.224.13>;tag=as04a9d8ca
To: <sip:8668692267 at 67.137.224.102>;tag=6dc7803e
Call-ID: 6fcc94a072e1e3003087c22153de269d at 67.137.224.13
CSeq: 102 INVITE
Contact: <sip:Mayberry at 67.137.224.102:5060>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, INFO, REFER, SUBSCRIBE
User-Agent: SysMaster VoIP Gateway v1.1.3
Content-Length: 0


10 headers, 0 lines

Sip read: 
SIP/2.0 200 OK
Via: SIP/2.0/UDP 67.137.224.13:5060;branch=z9hG4bK48b82637
From: "9726179231" <sip:9726179231 at 67.137.224.13>;tag=as04a9d8ca
To: <sip:8668692267 at 67.137.224.102>;tag=6dc7803e
Call-ID: 6fcc94a072e1e3003087c22153de269d at 67.137.224.13
CSeq: 102 INVITE
Contact: <sip:Mayberry at 67.137.224.102:5060>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, INFO, REFER, SUBSCRIBE
Allow-Events: message-summary
User-Agent: SysMaster VoIP Gateway v1.1.3
Content-Type: application/sdp
Content-Length: 218

v=0
o=admin 17807 17807 IN IP4 67.137.224.102
s=session
c=IN IP4 67.137.224.102
t=0 0
m=audio 5116 RTP/AVP 18 101
a=rtpmap:18 G729/8000/1
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

12 headers, 10 lines
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 67.137.224.102:5116
Found description format G729
Found description format telephone-event
Capabilities: us - 0x106 (gsm|ulaw|g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729)
Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723)
list_route: hop: <sip:Mayberry at 67.137.224.102:5060>
set_destination: Parsing <sip:Mayberry at 67.137.224.102:5060> for address/port to send to
set_destination: set destination to 67.137.224.102, port 5060
Transmitting:
ACK sip:8668692267 at 67.137.224.102 SIP/2.0
Via: SIP/2.0/UDP 67.137.224.13:5060;branch=z9hG4bK2808f821
From: "9726179231" <sip:9726179231 at 67.137.224.13>;tag=as04a9d8ca
To: <sip:8668692267 at 67.137.224.102>;tag=6dc7803e
Contact: <sip:9726179231 at 67.137.224.13>
Call-ID: 6fcc94a072e1e3003087c22153de269d at 67.137.224.13
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

 (no NAT) to 67.137.224.102:5060
    -- SIP/67.137.224.102-47b1 answered SIP/67.137.224.34-09077418
set_destination: Parsing <sip:Mayberry at 67.137.224.102:5060> for address/port to send to
set_destination: set destination to 67.137.224.102, port 5060
Reliably Transmitting:
BYE sip:Mayberry at 67.137.224.102:5060 SIP/2.0
Via: SIP/2.0/UDP 67.137.224.13:5060;branch=z9hG4bK10350095
From: "9726179231" <sip:9726179231 at 67.137.224.13>;tag=as04a9d8ca
To: <sip:8668692267 at 67.137.224.102>;tag=6dc7803e
Contact: <sip:9726179231 at 67.137.224.13>
Call-ID: 6fcc94a072e1e3003087c22153de269d at 67.137.224.13
CSeq: 103 BYE
User-Agent: Asterisk PBX
Content-Length: 0

 (no NAT) to 67.137.224.102:5060
  == Spawn extension (noauth, 8668692267, 2) exited non-zero on 'SIP/67.137.224.34-09077418'

Sip read: 
SIP/2.0 200 OK
Via: SIP/2.0/UDP 67.137.224.13:5060;branch=z9hG4bK10350095
From: "9726179231" <sip:9726179231 at 67.137.224.13>;tag=as04a9d8ca
To: <sip:8668692267 at 67.137.224.102>;tag=6dc7803e
Call-ID: 6fcc94a072e1e3003087c22153de269d at 67.137.224.13
CSeq: 103 BYE
Contact: <sip:Mayberry at 67.137.224.102:5060>
User-Agent: SysMaster VoIP Gateway v1.1.3
Content-Length: 0


9 headers, 0 lines
Destroying call '6fcc94a072e1e3003087c22153de269d at 67.137.224.13'
Reliably Transmitting (no NAT):
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 67.137.224.34:5060;branch=z9hG4bK73c4eeca
From: "9726179231" <sip:9726179231 at 67.137.224.34>;tag=as2c993cc2
To: <sip:8668692267 at 67.137.224.13>;tag=as68cbfb62
Call-ID: 00e19f2171cb25c0757a3e5d53590794 at 67.137.224.34
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:8668692267 at 67.137.224.13>
Content-Length: 0


 to 67.137.224.34:5060

Sip read: 
ACK sip:8668692267 at 67.137.224.13 SIP/2.0
Via: SIP/2.0/UDP 67.137.224.34:5060;branch=z9hG4bK73c4eeca
From: "9726179231" <sip:9726179231 at 67.137.224.34>;tag=as2c993cc2
To: <sip:8668692267 at 67.137.224.13>;tag=as68cbfb62
Contact: <sip:9726179231 at 67.137.224.34>
Call-ID: 00e19f2171cb25c0757a3e5d53590794 at 67.137.224.34
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0


9 headers, 0 lines
Destroying call '00e19f2171cb25c0757a3e5d53590794 at 67.137.224.34'


Sip read: 
CANCEL sip:8668692267 at 67.137.224.13 SIP/2.0
Via: SIP/2.0/UDP 67.137.224.34:5060;branch=z9hG4bK73c4eeca
From: "9726179231" <sip:9726179231 at 67.137.224.34>;tag=as2c993cc2
To: <sip:8668692267 at 67.137.224.13>
Contact: <sip:9726179231 at 67.137.224.34>
Call-ID: 00e19f2171cb25c0757a3e5d53590794 at 67.137.224.34
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Content-Length: 0


9 headers, 0 lines
Sending to 67.137.224.34 : 5060 (non-NAT)
Transmitting (no NAT):
SIP/2.0 481 Call Leg Does Not Exist
Via: SIP/2.0/UDP 67.137.224.34:5060;branch=z9hG4bK73c4eeca
From: "9726179231" <sip:9726179231 at 67.137.224.34>;tag=as2c993cc2
To: <sip:8668692267 at 67.137.224.13>;tag=as01558593
Call-ID: 00e19f2171cb25c0757a3e5d53590794 at 67.137.224.34
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: 
Content-Length: 0


 to 67.137.224.34:5060
Destroying call '00e19f2171cb25c0757a3e5d53590794 at 67.137.224.34'
ns3*CLI> sip no debug