[asterisk-bugs] [Asterisk 0011800]: asterisk 1.6-beta1 destroys cisco 7960 (sip firmware 7.4) outbound calls after 20sec due to no response to 200 OK
noreply at bugs.digium.com
noreply at bugs.digium.com
Mon Jan 21 11:57:25 CST 2008
A NOTE has been added to this issue.
======================================================================
http://bugs.digium.com/view.php?id=11800
======================================================================
Reported By: hmodes
Assigned To:
======================================================================
Project: Asterisk
Issue ID: 11800
Category: Channels/chan_sip/Interoperability
Reproducibility: always
Severity: major
Priority: normal
Status: feedback
Asterisk Version: 1.6.0-beta1
SVN Branch (only for SVN checkouts, not tarball releases): 1.6
SVN Revision (number only!):
Disclaimer on File?: N/A
Request Review:
======================================================================
Date Submitted: 01-19-2008 23:04 CST
Last Modified: 01-21-2008 11:57 CST
======================================================================
Summary: asterisk 1.6-beta1 destroys cisco 7960 (sip firmware
7.4) outbound calls after 20sec due to no response to 200 OK
Description:
Version tag on this report is incorrect as there is no 1.6-beta1 tag yet.
1.6-beta1 appears to have an issue with sip timers that causes cisco 7960
sip firmware 7.4 clients to be disconnected after 20 seconds for not
responding to 200 OK (marked as 'critical packet') once call setup is
complete. See debug. Incoming calls to the client do not experience the
disconnect. This is reproducible on two phones both on the lan with
asterisk and at a remote site.
======================================================================
----------------------------------------------------------------------
hmodes - 01-21-08 11:57
----------------------------------------------------------------------
Huh, looks like sip debug peer is only displaying the outgoing messages.
Here's the same call w/ debug ip and everything reverted back to the
vanilla tarball.
<--- SIP read from UDP://172.27.28.127:50718 --->
INVITE sip:12152223456 at 172.27.28.2;user=phone SIP/2.0
Via: SIP/2.0/UDP 172.27.28.127:5068;branch=z9hG4bK270f21cf
From: "hmodes"
<sip:hmodes at 172.27.28.2>;tag=0002fdaeee6f057f0ac8fb0d-275aa1c1
To: <sip:12152223456 at 172.27.28.2;user=phone>
Call-ID: 0002fdae-ee6f000f-262c94c0-1bfbee42 at 172.27.28.127
Max-Forwards: 70
Date: Mon, 21 Jan 2008 17:53:49 GMT
CSeq: 101 INVITE
User-Agent: Cisco-CP7960G/8.0
Contact: <sip:hmodes at 172.27.28.127:5068;transport=udp>
Expires: 180
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
Supported: replaces,join,norefersub
Content-Length: 277
Content-Type: application/sdp
Content-Disposition: session;handling=optional
v=0
o=Cisco-SIPUA 5885 0 IN IP4 172.27.28.127
s=SIP Call
t=0 0
m=audio 16610 RTP/AVP 0 8 18 101
c=IN IP4 172.27.28.127
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (17 headers 13 lines) ---
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Sending to 172.27.28.127 : 5068 (no NAT)
Using INVITE request as basis request -
0002fdae-ee6f000f-262c94c0-1bfbee42 at 172.27.28.127
Found user 'hmodes' for 'hmodes'
<--- Reliably Transmitting (no NAT) to 172.27.28.127:5068 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
172.27.28.127:5068;branch=z9hG4bK270f21cf;received=172.27.28.127
From: "hmodes"
<sip:hmodes at 172.27.28.2>;tag=0002fdaeee6f057f0ac8fb0d-275aa1c1
To: <sip:12152223456 at 172.27.28.2;user=phone>;tag=as01fdc2db
Call-ID: 0002fdae-ee6f000f-262c94c0-1bfbee42 at 172.27.28.127
CSeq: 101 INVITE
User-Agent: matrix.gs asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="matrix.gs",
nonce="6a066103"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog
'0002fdae-ee6f000f-262c94c0-1bfbee42 at 172.27.28.127' in 32000 ms (Method:
INVITE)
<--- SIP read from UDP://172.27.28.127:50782 --->
ACK sip:12152223456 at 172.27.28.2;user=phone SIP/2.0
Via: SIP/2.0/UDP 172.27.28.127:5068;branch=z9hG4bK270f21cf
From: "hmodes"
<sip:hmodes at 172.27.28.2>;tag=0002fdaeee6f057f0ac8fb0d-275aa1c1
To: <sip:12152223456 at 172.27.28.2;user=phone>;tag=as01fdc2db
Call-ID: 0002fdae-ee6f000f-262c94c0-1bfbee42 at 172.27.28.127
Date: Mon, 21 Jan 2008 17:53:49 GMT
CSeq: 101 ACK
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
<--- SIP read from UDP://172.27.28.127:50718 --->
INVITE sip:12152223456 at 172.27.28.2;user=phone SIP/2.0
Via: SIP/2.0/UDP 172.27.28.127:5068;branch=z9hG4bK7afb1e94
From: "hmodes"
<sip:hmodes at 172.27.28.2>;tag=0002fdaeee6f057f0ac8fb0d-275aa1c1
To: <sip:12152223456 at 172.27.28.2;user=phone>
Call-ID: 0002fdae-ee6f000f-262c94c0-1bfbee42 at 172.27.28.127
Max-Forwards: 70
Date: Mon, 21 Jan 2008 17:53:49 GMT
CSeq: 102 INVITE
User-Agent: Cisco-CP7960G/8.0
Contact: <sip:hmodes at 172.27.28.127:5068;transport=udp>
Authorization: Digest
username="hmodes",realm="matrix.gs",uri="sip:12152223456 at 172.27.28.2;user=phone",response="b379158727f2be3a5c60a82730c03b39",nonce="6a066103",algorithm=MD5
Expires: 180
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
Supported: replaces,join,norefersub
Content-Length: 277
Content-Type: application/sdp
Content-Disposition: session;handling=optional
v=0
o=Cisco-SIPUA 5885 0 IN IP4 172.27.28.127
s=SIP Call
t=0 0
m=audio 16610 RTP/AVP 0 8 18 101
c=IN IP4 172.27.28.127
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (18 headers 13 lines) ---
Sending to 172.27.28.127 : 5068 (no NAT)
Using INVITE request as basis request -
0002fdae-ee6f000f-262c94c0-1bfbee42 at 172.27.28.127
Found user 'hmodes' for 'hmodes'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 172.27.28.127:16610
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - 0x4 (ulaw), peer - audio=0x10c
(ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4
(ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 172.27.28.127:16610
Looking for 12152223456 in userauth (domain 172.27.28.2)
list_route: hop: <sip:hmodes at 172.27.28.127:5068;transport=udp>
<--- Transmitting (no NAT) to 172.27.28.127:5068 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
172.27.28.127:5068;branch=z9hG4bK7afb1e94;received=172.27.28.127
From: "hmodes"
<sip:hmodes at 172.27.28.2>;tag=0002fdaeee6f057f0ac8fb0d-275aa1c1
To: <sip:12152223456 at 172.27.28.2;user=phone>
Call-ID: 0002fdae-ee6f000f-262c94c0-1bfbee42 at 172.27.28.127
CSeq: 102 INVITE
User-Agent: matrix.gs asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:12152223456 at 172.27.28.2:51731>
Content-Length: 0
<------------>
-- Executing [12152223456 at userauth:1]
SetMusicOnHold("SIP/hmodes-083e0188", "default") in new stack
-- Executing [12152223456 at userauth:2] NoOp("SIP/hmodes-083e0188", "")
in new stack
-- Executing [12152223456 at userauth:3] Dial("SIP/hmodes-083e0188",
"SIP/12152223456 at atlas,60,r") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called 12152223456 at atlas
<--- Transmitting (no NAT) to 172.27.28.127:5068 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP
172.27.28.127:5068;branch=z9hG4bK7afb1e94;received=172.27.28.127
From: "hmodes"
<sip:hmodes at 172.27.28.2>;tag=0002fdaeee6f057f0ac8fb0d-275aa1c1
To: <sip:12152223456 at 172.27.28.2;user=phone>;tag=as2aca97e5
Call-ID: 0002fdae-ee6f000f-262c94c0-1bfbee42 at 172.27.28.127
CSeq: 102 INVITE
User-Agent: matrix.gs asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:12152223456 at 172.27.28.2:51731>
Content-Length: 0
<------------>
-- SIP/atlas-084186e0 is making progress passing it to
SIP/hmodes-083e0188
--- set_address_from_contact host '69.59.227.94'
-- SIP/atlas-084186e0 answered SIP/hmodes-083e0188
Audio is at 172.27.28.2 port 49324
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Reliably Transmitting (no NAT) to 172.27.28.127:5068 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
172.27.28.127:5068;branch=z9hG4bK7afb1e94;received=172.27.28.127
From: "hmodes"
<sip:hmodes at 172.27.28.2>;tag=0002fdaeee6f057f0ac8fb0d-275aa1c1
To: <sip:12152223456 at 172.27.28.2;user=phone>;tag=as2aca97e5
Call-ID: 0002fdae-ee6f000f-262c94c0-1bfbee42 at 172.27.28.127
CSeq: 102 INVITE
User-Agent: matrix.gs asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:12152223456 at 172.27.28.2:51731>
Content-Type: application/sdp
Content-Length: 248
v=0
o=root 1900596219 1900596219 IN IP4 172.27.28.2
s=session
c=IN IP4 172.27.28.2
t=0 0
m=audio 49324 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------>
-- Packet2Packet bridging SIP/hmodes-083e0188 and SIP/atlas-084186e0
Retransmitting http://bugs.digium.com/view.php?id=1 (no NAT) to
172.27.28.127:5068:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
172.27.28.127:5068;branch=z9hG4bK7afb1e94;received=172.27.28.127
From: "hmodes"
<sip:hmodes at 172.27.28.2>;tag=0002fdaeee6f057f0ac8fb0d-275aa1c1
To: <sip:12152223456 at 172.27.28.2;user=phone>;tag=as2aca97e5
Call-ID: 0002fdae-ee6f000f-262c94c0-1bfbee42 at 172.27.28.127
CSeq: 102 INVITE
User-Agent: matrix.gs asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:12152223456 at 172.27.28.2:51731>
Content-Type: application/sdp
Content-Length: 248
v=0
o=root 1900596219 1900596219 IN IP4 172.27.28.2
s=session
c=IN IP4 172.27.28.2
t=0 0
m=audio 49324 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
Retransmitting http://bugs.digium.com/view.php?id=2 (no NAT) to
172.27.28.127:5068:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
172.27.28.127:5068;branch=z9hG4bK7afb1e94;received=172.27.28.127
From: "hmodes"
<sip:hmodes at 172.27.28.2>;tag=0002fdaeee6f057f0ac8fb0d-275aa1c1
To: <sip:12152223456 at 172.27.28.2;user=phone>;tag=as2aca97e5
Call-ID: 0002fdae-ee6f000f-262c94c0-1bfbee42 at 172.27.28.127
CSeq: 102 INVITE
User-Agent: matrix.gs asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:12152223456 at 172.27.28.2:51731>
Content-Type: application/sdp
Content-Length: 248
v=0
o=root 1900596219 1900596219 IN IP4 172.27.28.2
s=session
c=IN IP4 172.27.28.2
t=0 0
m=audio 49324 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
Retransmitting http://bugs.digium.com/view.php?id=3 (no NAT) to
172.27.28.127:5068:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
172.27.28.127:5068;branch=z9hG4bK7afb1e94;received=172.27.28.127
From: "hmodes"
<sip:hmodes at 172.27.28.2>;tag=0002fdaeee6f057f0ac8fb0d-275aa1c1
To: <sip:12152223456 at 172.27.28.2;user=phone>;tag=as2aca97e5
Call-ID: 0002fdae-ee6f000f-262c94c0-1bfbee42 at 172.27.28.127
CSeq: 102 INVITE
User-Agent: matrix.gs asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:12152223456 at 172.27.28.2:51731>
Content-Type: application/sdp
Content-Length: 248
v=0
o=root 1900596219 1900596219 IN IP4 172.27.28.2
s=session
c=IN IP4 172.27.28.2
t=0 0
m=audio 49324 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
Retransmitting http://bugs.digium.com/view.php?id=4 (no NAT) to
172.27.28.127:5068:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
172.27.28.127:5068;branch=z9hG4bK7afb1e94;received=172.27.28.127
From: "hmodes"
<sip:hmodes at 172.27.28.2>;tag=0002fdaeee6f057f0ac8fb0d-275aa1c1
To: <sip:12152223456 at 172.27.28.2;user=phone>;tag=as2aca97e5
Call-ID: 0002fdae-ee6f000f-262c94c0-1bfbee42 at 172.27.28.127
CSeq: 102 INVITE
User-Agent: matrix.gs asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:12152223456 at 172.27.28.2:51731>
Content-Type: application/sdp
Content-Length: 248
v=0
o=root 1900596219 1900596219 IN IP4 172.27.28.2
s=session
c=IN IP4 172.27.28.2
t=0 0
m=audio 49324 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
Retransmitting http://bugs.digium.com/view.php?id=5 (no NAT) to
172.27.28.127:5068:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
172.27.28.127:5068;branch=z9hG4bK7afb1e94;received=172.27.28.127
From: "hmodes"
<sip:hmodes at 172.27.28.2>;tag=0002fdaeee6f057f0ac8fb0d-275aa1c1
To: <sip:12152223456 at 172.27.28.2;user=phone>;tag=as2aca97e5
Call-ID: 0002fdae-ee6f000f-262c94c0-1bfbee42 at 172.27.28.127
CSeq: 102 INVITE
User-Agent: matrix.gs asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:12152223456 at 172.27.28.2:51731>
Content-Type: application/sdp
Content-Length: 248
v=0
o=root 1900596219 1900596219 IN IP4 172.27.28.2
s=session
c=IN IP4 172.27.28.2
t=0 0
m=audio 49324 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
Retransmitting http://bugs.digium.com/view.php?id=6 (no NAT) to
172.27.28.127:5068:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
172.27.28.127:5068;branch=z9hG4bK7afb1e94;received=172.27.28.127
From: "hmodes"
<sip:hmodes at 172.27.28.2>;tag=0002fdaeee6f057f0ac8fb0d-275aa1c1
To: <sip:12152223456 at 172.27.28.2;user=phone>;tag=as2aca97e5
Call-ID: 0002fdae-ee6f000f-262c94c0-1bfbee42 at 172.27.28.127
CSeq: 102 INVITE
User-Agent: matrix.gs asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:12152223456 at 172.27.28.2:51731>
Content-Type: application/sdp
Content-Length: 248
v=0
o=root 1900596219 1900596219 IN IP4 172.27.28.2
s=session
c=IN IP4 172.27.28.2
t=0 0
m=audio 49324 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
[Jan 21 12:54:12] WARNING[19363]: chan_sip.c:2688 retrans_pkt: Maximum
retries exceeded on transmission
0002fdae-ee6f000f-262c94c0-1bfbee42 at 172.27.28.127 for seqno 102 (Critical
Response)
[Jan 21 12:54:12] WARNING[19363]: chan_sip.c:2715 retrans_pkt: Hanging up
call 0002fdae-ee6f000f-262c94c0-1bfbee42 at 172.27.28.127 - no reply to our
critical packet.
== Spawn extension (userauth, 12152223456, 3) exited non-zero on
'SIP/hmodes-083e0188'
Really destroying SIP dialog
'0002fdae-ee6f000f-262c94c0-1bfbee42 at 172.27.28.127' Method: INVITE
Issue History
Date Modified Username Field Change
======================================================================
01-21-08 11:57 hmodes Note Added: 0080954
======================================================================
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