[asterisk-bugs] [Asterisk 0018399]: Call torn down upon connection when early media 183 used
Asterisk Bug Tracker
noreply at bugs.digium.com
Mon Nov 29 15:40:10 CST 2010
A NOTE has been added to this issue.
======================================================================
https://issues.asterisk.org/view.php?id=18399
======================================================================
Reported By: eeman
Assigned To:
======================================================================
Project: Asterisk
Issue ID: 18399
Category: Channels/chan_sip/General
Reproducibility: always
Severity: major
Priority: normal
Status: new
Asterisk Version: 1.8.1-rc1
JIRA:
Regression: No
Reviewboard Link:
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Request Review:
======================================================================
Date Submitted: 2010-11-29 15:27 CST
Last Modified: 2010-11-29 15:40 CST
======================================================================
Summary: Call torn down upon connection when early media 183
used
Description:
Asterisk 1.8.1-rc1 & Asterisk 1.6.2.14
Centos 5.5
have scenario as such
Asterisk-1.8.1-rc1 -SIP-> Asterisk 1.6.2.14 -SIP-> Broadvox (Sonus
Softswitch)
When calling a TF number that uses early media for their IVR (example
1-800-626-2001); once the call gets connected and the 200 OK message is
received, my 1.8.1-rc1 box issues a BYE message with a HangupCauseCode of
0. I can reproduce this with several numbers that are using early media for
their IVR's. Just as soon as my call gets connected to a call-center's ACD
Queue I hear 1-2 seconds of the recording before the call is torn down. I
have tested this using a linksys SPA-2102 ATA, A Polycom IP501, as well as
a Digium FXS module and get identical results.
======================================================================
----------------------------------------------------------------------
(0129201) eeman (reporter) - 2010-11-29 15:40
https://issues.asterisk.org/view.php?id=18399#c129201
----------------------------------------------------------------------
example when using sip set debug -
-- Executing [dial-SIP at macro-tl-dialout-base:7]
Dial("SIP/202-0000003c", "SIP/+18006262001 at GW01EEMAN,60,") in new stack
== Using UDPTL TOS bits 184
== Using UDPTL CoS mark 5
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Audio is at 5060
Adding codec 0x4 (ulaw) to SDP
Reliably Transmitting (no NAT) to 69.64.11.6:5060:
INVITE sip:+18006262001 at 69.64.11.6 SIP/2.0
Via: SIP/2.0/UDP 216.135.89.226:5060;branch=z9hG4bK11cb4eae
Max-Forwards: 70
From: "Erik Smith" <sip:5023152516 at 216.135.89.226>;tag=as3813e5d8
To: <sip:+18006262001 at 69.64.11.6>
Contact: <sip:5023152516 at 216.135.89.226:5060>
Call-ID: 0794caf82d792f245c6d346d37d2a924 at 216.135.89.226:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.1-rc1
Date: Mon, 29 Nov 2010 21:33:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Remote-Party-ID: "Erik Smith"
<sip:5023152516 at 216.135.89.226>;party=calling;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 184
v=0
o=root 276500518 276500518 IN IP4 216.135.89.226
s=Asterisk PBX 1.8.1-rc1
c=IN IP4 216.135.89.226
t=0 0
m=audio 12180 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=ptime:20
a=sendrecv
---
-- Called +18006262001 at GW01EEMAN
<--- SIP read from UDP:69.64.11.6:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
216.135.89.226:5060;branch=z9hG4bK11cb4eae;received=216.135.89.226
From: "Erik Smith" <sip:5023152516 at 216.135.89.226>;tag=as3813e5d8
To: <sip:+18006262001 at 69.64.11.6>;tag=as4e48411b
Call-ID: 0794caf82d792f245c6d346d37d2a924 at 216.135.89.226:5060
CSeq: 102 INVITE
Server: Asterisk PBX 1.6.2.14-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk",
nonce="669350ed"
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
Transmitting (no NAT) to 69.64.11.6:5060:
ACK sip:+18006262001 at 69.64.11.6 SIP/2.0
Via: SIP/2.0/UDP 216.135.89.226:5060;branch=z9hG4bK11cb4eae
Max-Forwards: 70
From: "Erik Smith" <sip:5023152516 at 216.135.89.226>;tag=as3813e5d8
To: <sip:+18006262001 at 69.64.11.6>;tag=as4e48411b
Contact: <sip:5023152516 at 216.135.89.226:5060>
Call-ID: 0794caf82d792f245c6d346d37d2a924 at 216.135.89.226:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.1-rc1
Content-Length: 0
---
Audio is at 5060
Adding codec 0x4 (ulaw) to SDP
Reliably Transmitting (no NAT) to 69.64.11.6:5060:
INVITE sip:+18006262001 at 69.64.11.6 SIP/2.0
Via: SIP/2.0/UDP 216.135.89.226:5060;branch=z9hG4bK02aac998
Max-Forwards: 70
From: "Erik Smith" <sip:5023152516 at 216.135.89.226>;tag=as3813e5d8
To: <sip:+18006262001 at 69.64.11.6>
Contact: <sip:5023152516 at 216.135.89.226:5060>
Call-ID: 0794caf82d792f245c6d346d37d2a924 at 216.135.89.226:5060
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.8.1-rc1
Authorization: Digest username="GW01EEMAN", realm="asterisk",
algorithm=MD5, uri="sip:+18006262001 at 69.64.11.6", nonce="669350ed",
response="8c8669c1bc9665d5ebeaed71ad6618a1"
Date: Mon, 29 Nov 2010 21:33:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Remote-Party-ID: "Erik Smith"
<sip:5023152516 at 216.135.89.226>;party=calling;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 184
v=0
o=root 276500518 276500519 IN IP4 216.135.89.226
s=Asterisk PBX 1.8.1-rc1
c=IN IP4 216.135.89.226
t=0 0
m=audio 12180 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=ptime:20
a=sendrecv
---
<--- SIP read from UDP:69.64.11.6:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
216.135.89.226:5060;branch=z9hG4bK02aac998;received=216.135.89.226
From: "Erik Smith" <sip:5023152516 at 216.135.89.226>;tag=as3813e5d8
To: <sip:+18006262001 at 69.64.11.6>
Call-ID: 0794caf82d792f245c6d346d37d2a924 at 216.135.89.226:5060
CSeq: 103 INVITE
Server: Asterisk PBX 1.6.2.14-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:+18006262001 at 69.64.11.6>
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
<--- SIP read from UDP:69.64.11.6:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP
216.135.89.226:5060;branch=z9hG4bK02aac998;received=216.135.89.226
From: "Erik Smith" <sip:5023152516 at 216.135.89.226>;tag=as3813e5d8
To: <sip:+18006262001 at 69.64.11.6>;tag=as35c589d8
Call-ID: 0794caf82d792f245c6d346d37d2a924 at 216.135.89.226:5060
CSeq: 103 INVITE
Server: Asterisk PBX 1.6.2.14-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:+18006262001 at 69.64.11.6>
Content-Type: application/sdp
Content-Length: 179
v=0
o=root 629951581 629951581 IN IP4 69.64.11.6
s=Asterisk PBX 1.6.2.14-rc1
c=IN IP4 69.64.11.6
t=0 0
m=audio 13978 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=ptime:20
a=sendrecv
<------------->
--- (12 headers 9 lines) ---
Found RTP audio format 0
Found audio description format PCMU for ID 0
Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0
(nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing),
combined - 0x0 (nothing)
Peer audio RTP is at port 69.64.11.6:13978
-- SIP/GW01EEMAN-0000003d is making progress passing it to
SIP/202-0000003c
Reliably Transmitting (no NAT) to 69.64.11.6:5060:
INFO sip:+18006262001 at 69.64.11.6 SIP/2.0
Via: SIP/2.0/UDP 216.135.89.226:5060;branch=z9hG4bK1f19538a
Max-Forwards: 70
From: "Erik Smith" <sip:5023152516 at 216.135.89.226>;tag=as3813e5d8
To: <sip:+18006262001 at 69.64.11.6>;tag=as35c589d8
Contact: <sip:5023152516 at 216.135.89.226:5060>
Call-ID: 0794caf82d792f245c6d346d37d2a924 at 216.135.89.226:5060
CSeq: 104 INFO
User-Agent: Asterisk PBX 1.8.1-rc1
Content-Type: application/dtmf-relay
Content-Length: 23
Signal=1
Duration=51
---
<--- SIP read from UDP:69.64.11.6:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
216.135.89.226:5060;branch=z9hG4bK1f19538a;received=216.135.89.226
From: "Erik Smith" <sip:5023152516 at 216.135.89.226>;tag=as3813e5d8
To: <sip:+18006262001 at 69.64.11.6>;tag=as35c589d8
Call-ID: 0794caf82d792f245c6d346d37d2a924 at 216.135.89.226:5060
CSeq: 104 INFO
Server: Asterisk PBX 1.6.2.14-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
Reliably Transmitting (no NAT) to 69.64.11.6:5060:
INFO sip:+18006262001 at 69.64.11.6 SIP/2.0
Via: SIP/2.0/UDP 216.135.89.226:5060;branch=z9hG4bK479acf86
Max-Forwards: 70
From: "Erik Smith" <sip:5023152516 at 216.135.89.226>;tag=as3813e5d8
To: <sip:+18006262001 at 69.64.11.6>;tag=as35c589d8
Contact: <sip:5023152516 at 216.135.89.226:5060>
Call-ID: 0794caf82d792f245c6d346d37d2a924 at 216.135.89.226:5060
CSeq: 105 INFO
User-Agent: Asterisk PBX 1.8.1-rc1
Content-Type: application/dtmf-relay
Content-Length: 23
Signal=3
Duration=40
---
<--- SIP read from UDP:69.64.11.6:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
216.135.89.226:5060;branch=z9hG4bK479acf86;received=216.135.89.226
From: "Erik Smith" <sip:5023152516 at 216.135.89.226>;tag=as3813e5d8
To: <sip:+18006262001 at 69.64.11.6>;tag=as35c589d8
Call-ID: 0794caf82d792f245c6d346d37d2a924 at 216.135.89.226:5060
CSeq: 105 INFO
Server: Asterisk PBX 1.6.2.14-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '4caf777a5e65552c61b28e3f2c8e2b4a at 69.64.11.6'
Method: OPTIONS
<--- SIP read from UDP:69.64.11.6:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
216.135.89.226:5060;branch=z9hG4bK02aac998;received=216.135.89.226
From: "Erik Smith" <sip:5023152516 at 216.135.89.226>;tag=as3813e5d8
To: <sip:+18006262001 at 69.64.11.6>;tag=as35c589d8
Call-ID: 0794caf82d792f245c6d346d37d2a924 at 216.135.89.226:5060
CSeq: 103 INVITE
Server: Asterisk PBX 1.6.2.14-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 179
v=0
o=root 629951581 629951582 IN IP4 69.64.11.6
s=Asterisk PBX 1.6.2.14-rc1
c=IN IP4 69.64.11.6
t=0 0
m=audio 13978 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=ptime:20
a=sendrecv
<------------->
--- (11 headers 9 lines) ---
Found RTP audio format 0
Found audio description format PCMU for ID 0
Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0
(nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing),
combined - 0x0 (nothing)
Peer audio RTP is at port 69.64.11.6:13978
list_route: no route
[Nov 29 16:33:38] WARNING[25528]: chan_sip.c:12909
__set_address_from_contact: Invalid contact uri (missing sip: or sips:),
attempting to use anyway
[Nov 29 16:33:38] ERROR[25528]: netsock2.c:245 ast_sockaddr_resolve:
getaddrinfo("Ã ", "(null)", ...): Name or service not known
[Nov 29 16:33:38] WARNING[25528]: chan_sip.c:12920
__set_address_from_contact: Invalid host name in Contact: (can't resolve in
DNS) : 'Ã '
Transmitting (no NAT) to 69.64.11.6:5060:
ACK sip:+18006262001 at 69.64.11.6 SIP/2.0
Via: SIP/2.0/UDP 216.135.89.226:5060;branch=z9hG4bK6b67a4d8
Max-Forwards: 70
From: "Erik Smith" <sip:5023152516 at 216.135.89.226>;tag=as3813e5d8
To: <sip:+18006262001 at 69.64.11.6>;tag=as35c589d8
Contact: <sip:5023152516 at 216.135.89.226:5060>
Call-ID: 0794caf82d792f245c6d346d37d2a924 at 216.135.89.226:5060
CSeq: 103 ACK
User-Agent: Asterisk PBX 1.8.1-rc1
Content-Length: 0
---
Reliably Transmitting (no NAT) to 69.64.11.6:5060:
BYE sip:+18006262001 at 69.64.11.6 SIP/2.0
Via: SIP/2.0/UDP 216.135.89.226:5060;branch=z9hG4bK1460ab81
Max-Forwards: 70
From: "Erik Smith" <sip:5023152516 at 216.135.89.226>;tag=as3813e5d8
To: <sip:+18006262001 at 69.64.11.6>;tag=as35c589d8
Call-ID: 0794caf82d792f245c6d346d37d2a924 at 216.135.89.226:5060
CSeq: 106 BYE
User-Agent: Asterisk PBX 1.8.1-rc1
Authorization: Digest username="GW01EEMAN", realm="asterisk",
algorithm=MD5, uri="sip:+18006262001 at 69.64.11.6", nonce="669350ed",
response="42e7818855fbe4b45995bd5bc2aba30a"
X-Asterisk-HangupCause: Unknown
X-Asterisk-HangupCauseCode: 0
Content-Length: 0
---
Scheduling destruction of SIP dialog
'0794caf82d792f245c6d346d37d2a924 at 216.135.89.226:5060' in 6400 ms (Method:
INVITE)
-- SIP/GW01EEMAN-0000003d answered SIP/202-0000003c
Scheduling destruction of SIP dialog
'0794caf82d792f245c6d346d37d2a924 at 216.135.89.226:5060' in 6400 ms (Method:
INVITE)
Reliably Transmitting (no NAT) to 69.64.11.6:5060:
BYE sip:+18006262001 at 69.64.11.6 SIP/2.0
Via: SIP/2.0/UDP 216.135.89.226:5060;branch=z9hG4bK063298a2
Max-Forwards: 70
From: "Erik Smith" <sip:5023152516 at 216.135.89.226>;tag=as3813e5d8
To: <sip:+18006262001 at 69.64.11.6>;tag=as35c589d8
Call-ID: 0794caf82d792f245c6d346d37d2a924 at 216.135.89.226:5060
CSeq: 107 BYE
User-Agent: Asterisk PBX 1.8.1-rc1
Authorization: Digest username="GW01EEMAN", realm="asterisk",
algorithm=MD5, uri="sip:+18006262001 at 69.64.11.6", nonce="669350ed",
response="42e7818855fbe4b45995bd5bc2aba30a"
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
---
== Spawn extension (macro-tl-dialout-base, dial-SIP, 7) exited non-zero
on 'SIP/202-0000003c' in macro 'tl-dialout-base'
== Spawn extension (macro-tl-dialout-1-trunk, s, 3) exited non-zero on
'SIP/202-0000003c' in macro 'tl-dialout-1-trunk'
== Spawn extension (from-inside-redir, 18006262001, 1) exited non-zero
on 'SIP/202-0000003c'
-- Executing [h at from-inside-redir:1] Hangup("SIP/202-0000003c", "") in
new stack
== Spawn extension (from-inside-redir, h, 1) exited non-zero on
'SIP/202-0000003c'
<--- SIP read from UDP:69.64.11.6:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
216.135.89.226:5060;branch=z9hG4bK1460ab81;received=216.135.89.226
From: "Erik Smith" <sip:5023152516 at 216.135.89.226>;tag=as3813e5d8
To: <sip:+18006262001 at 69.64.11.6>;tag=as35c589d8
Call-ID: 0794caf82d792f245c6d346d37d2a924 at 216.135.89.226:5060
CSeq: 106 BYE
Server: Asterisk PBX 1.6.2.14-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
<--- SIP read from UDP:69.64.11.6:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
216.135.89.226:5060;branch=z9hG4bK063298a2;received=216.135.89.226
From: "Erik Smith" <sip:5023152516 at 216.135.89.226>;tag=as3813e5d8
To: <sip:+18006262001 at 69.64.11.6>;tag=as35c589d8
Call-ID: 0794caf82d792f245c6d346d37d2a924 at 216.135.89.226:5060
CSeq: 107 BYE
Server: Asterisk PBX 1.6.2.14-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog
'0794caf82d792f245c6d346d37d2a924 at 216.135.89.226:5060' Method: INVITE
eeman*CLI> sip set debug off
SIP Debugging Disabled
Issue History
Date Modified Username Field Change
======================================================================
2010-11-29 15:40 eeman Note Added: 0129201
======================================================================
More information about the asterisk-bugs
mailing list