[asterisk-bugs] [Asterisk 0016959]: SIP-Provider without "SIP/2.0 180 Ringing" makes trouble with call file
Asterisk Bug Tracker
noreply at bugs.digium.com
Thu Mar 4 21:21:16 CST 2010
A NOTE has been added to this issue.
======================================================================
https://issues.asterisk.org/view.php?id=16959
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Reported By: fa_bian
Assigned To:
======================================================================
Project: Asterisk
Issue ID: 16959
Category: General
Reproducibility: always
Severity: minor
Priority: normal
Status: feedback
Asterisk Version: 1.6.2.2
JIRA:
Regression: No
Reviewboard Link:
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Request Review:
======================================================================
Date Submitted: 2010-03-04 06:15 CST
Last Modified: 2010-03-04 21:21 CST
======================================================================
Summary: SIP-Provider without "SIP/2.0 180 Ringing" makes
trouble with call file
Description:
I make a auto dial via call file.
If I use a SIP-Provider who sends a "SIP/2.0 180 Ringing" all is fine!
If I use a SIP-Provider who DOESN'T send a "SIP/2.0 180 Ringing" Asterisk
DOESN'T proceed in context.
======================================================================
----------------------------------------------------------------------
(0118998) fa_bian (reporter) - 2010-03-04 21:21
https://issues.asterisk.org/view.php?id=16959#c118998
----------------------------------------------------------------------
###########################
### call file (sip-trace) ###
### provider 1, WITHOUT ringing ###
###########################
INVITE sip:00491712345678 at sip.provider2.tld SIP/2.0
Via: SIP/2.0/UDP 123.123.123.123:5060;branch=z9hG4bK2de9667e;rport
Max-Forwards: 70
From: "49301234567" <sip:445566 at 123.123.123.123>;tag=as2f441153
To: <sip:00491712345678 at sip.provider2.tld>
Contact: <sip:445566 at 123.123.123.123>
Call-ID: 0b7cb2ad625457b03835a8c60214c934 at 123.123.123.123
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.2
Date: Fri, 05 Mar 2010 01:34:45 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 287
v=0
o=root 293346299 293346299 IN IP4 123.123.123.123
s=Asterisk PBX 1.6.2.2
c=IN IP4 123.123.123.123
t=0 0
m=audio 19358 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
<--- SIP read from UDP:12.12.12.12:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 123.123.123.123:5060;branch=z9hG4bK2de9667e;rport
From: "49301234567" <sip:445566 at 123.123.123.123>;tag=as2f441153
To: <sip:00491712345678 at sip.provider2.tld>
Contact: sip:00491712345678 at 12.12.12.12:5060
Call-ID: 0b7cb2ad625457b03835a8c60214c934 at 123.123.123.123
CSeq: 102 INVITE
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
WWW-Authenticate: Digest
realm="sip.provider2.tld",nonce="3323756578",algorithm=MD5
Content-Length: 0
<------------->
ACK sip:00491712345678 at sip.provider2.tld SIP/2.0
Via: SIP/2.0/UDP 123.123.123.123:5060;branch=z9hG4bK2de9667e;rport
Max-Forwards: 70
From: "49301234567" <sip:445566 at 123.123.123.123>;tag=as2f441153
To: <sip:00491712345678 at sip.provider2.tld>
Contact: <sip:445566 at 123.123.123.123>
Call-ID: 0b7cb2ad625457b03835a8c60214c934 at 123.123.123.123
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.2.2
Content-Length: 0
INVITE sip:00491712345678 at sip.provider2.tld SIP/2.0
Via: SIP/2.0/UDP 123.123.123.123:5060;branch=z9hG4bK16921415;rport
Max-Forwards: 70
From: "49301234567" <sip:445566 at 123.123.123.123>;tag=as2f441153
To: <sip:00491712345678 at sip.provider2.tld>
Contact: <sip:445566 at 123.123.123.123>
Call-ID: 0b7cb2ad625457b03835a8c60214c934 at 123.123.123.123
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.6.2.2
Authorization: Digest username="445566", realm="sip.provider2.tld",
algorithm=MD5, uri="sip:00491712345678 at sip.provider2.tld",
nonce="3323756578", response="595cb53585ff6e80d41456757e8228f9"
Date: Fri, 05 Mar 2010 01:34:45 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 287
v=0
o=root 293346299 293346300 IN IP4 123.123.123.123
s=Asterisk PBX 1.6.2.2
c=IN IP4 123.123.123.123
t=0 0
m=audio 19358 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
<--- SIP read from UDP:12.12.12.12:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 123.123.123.123:5060;branch=z9hG4bK16921415;rport
From: "49301234567" <sip:445566 at 123.123.123.123>;tag=as2f441153
To: <sip:00491712345678 at sip.provider2.tld>
Contact: sip:00491712345678 at 12.12.12.12:5060
Call-ID: 0b7cb2ad625457b03835a8c60214c934 at 123.123.123.123
CSeq: 103 INVITE
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
Content-Length: 0
<------------->
<--- SIP read from UDP:12.12.12.12:5060 --->
SIP/2.0 183 Session progress
Via: SIP/2.0/UDP 123.123.123.123:5060;branch=z9hG4bK16921415;rport
From: "49301234567" <sip:445566 at 123.123.123.123>;tag=as2f441153
To: <sip:00491712345678 at sip.provider2.tld>;tag=120113ac4b5da89a85a5c2
Contact: sip:00491712345678 at 12.12.12.12:5060
Call-ID: 0b7cb2ad625457b03835a8c60214c934 at 123.123.123.123
CSeq: 103 INVITE
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
Content-Type: application/sdp
Content-Length: 201
v=0
o=445566 1267752884 1267752884 IN IP4 13.13.13.13
s=SIP Call
c=IN IP4 13.13.13.13
t=0 0
m=audio 25570 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=ptime:20
<------------->
<--- SIP read from UDP:12.12.12.12:5060 --->
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 123.123.123.123:5060;branch=z9hG4bK16921415;rport
From: "49301234567" <sip:445566 at 123.123.123.123>;tag=as2f441153
To: <sip:00491712345678 at sip.provider2.tld>;tag=120113ac4b5da89a85a5c2
Contact: sip:00491712345678 at 12.12.12.12:5060
Call-ID: 0b7cb2ad625457b03835a8c60214c934 at 123.123.123.123
CSeq: 103 INVITE
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
Content-Type: application/sdp
Content-Length: 201
v=0
o=445566 1267752901 1267752901 IN IP4 13.13.13.13
s=SIP Call
c=IN IP4 13.13.13.13
t=0 0
m=audio 25570 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=ptime:20
<------------->
ACK sip:00491712345678 at 12.12.12.12:5060 SIP/2.0
Via: SIP/2.0/UDP 123.123.123.123:5060;branch=z9hG4bK21e476bc;rport
Max-Forwards: 70
From: "49301234567" <sip:445566 at 123.123.123.123>;tag=as2f441153
To: <sip:00491712345678 at sip.provider2.tld>;tag=120113ac4b5da89a85a5c2
Contact: <sip:445566 at 123.123.123.123>
Call-ID: 0b7cb2ad625457b03835a8c60214c934 at 123.123.123.123
CSeq: 103 ACK
User-Agent: Asterisk PBX 1.6.2.2
Content-Length: 0
---
OPTIONS sip:sip.provider2.tld SIP/2.0
Via: SIP/2.0/UDP 123.123.123.123:5060;branch=z9hG4bK28767e3a;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk at 123.123.123.123>;tag=as02d3179c
To: <sip:sip.provider2.tld>
Contact: <sip:asterisk at 123.123.123.123>
Call-ID: 0c8d900b1816ea6229d62a09565a8982 at 123.123.123.123
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.2
Date: Fri, 05 Mar 2010 01:35:13 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
<--- SIP read from UDP:12.12.12.12:5060 --->
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 123.123.123.123:5060;branch=z9hG4bK28767e3a;rport
From: "asterisk" <sip:asterisk at 123.123.123.123>;tag=as02d3179c
To: <sip:sip.provider2.tld>
Contact: sip:12.12.12.12:5060
Call-ID: 0c8d900b1816ea6229d62a09565a8982 at 123.123.123.123
CSeq: 102 OPTIONS
Supported: foo
User-Agent: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
Accept: application/sdp
<------------->
BYE sip:00491712345678 at 12.12.12.12:5060 SIP/2.0
Via: SIP/2.0/UDP 123.123.123.123:5060;branch=z9hG4bK6a6f395a;rport
Max-Forwards: 70
From: "49301234567" <sip:445566 at 123.123.123.123>;tag=as2f441153
To: <sip:00491712345678 at sip.provider2.tld>;tag=120113ac4b5da89a85a5c2
Call-ID: 0b7cb2ad625457b03835a8c60214c934 at 123.123.123.123
CSeq: 104 BYE
User-Agent: Asterisk PBX 1.6.2.2
Authorization: Digest username="445566", realm="sip.provider2.tld",
algorithm=MD5, uri="sip:00491712345678 at 12.12.12.12:5060",
nonce="3323756578", response="8d0d81af3175f24f0b9a3053b7d349f1"
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
---
<--- SIP read from UDP:12.12.12.12:5060 --->
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 123.123.123.123:5060;branch=z9hG4bK6a6f395a;rport
From: "49301234567" <sip:445566 at 123.123.123.123>;tag=as2f441153
To: <sip:00491712345678 at sip.provider2.tld>;tag=120113ac4b5da89a85a5c2
Contact: sip:00491712345678 at 12.12.12.12:5060
Call-ID: 0b7cb2ad625457b03835a8c60214c934 at 123.123.123.123
CSeq: 104 BYE
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
Content-Length: 0
<------------->
Issue History
Date Modified Username Field Change
======================================================================
2010-03-04 21:21 fa_bian Note Added: 0118998
======================================================================
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