[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 21:16:39 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 21:16 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.
======================================================================
----------------------------------------------------------------------
(0118997) fa_bian (reporter) - 2010-03-04 21:16
https://issues.asterisk.org/view.php?id=16959#c118997
----------------------------------------------------------------------
######################
### call file (sip-trace) ###
### provider 1, WITH ringing ###
######################
INVITE sip:00491712345678 at sip.provider1.tld SIP/2.0
Via: SIP/2.0/UDP 123.123.123.123:5060;branch=z9hG4bK2cdca6b8;rport
Max-Forwards: 70
From: "49301234567" <sip:112233 at 123.123.123.123>;tag=as6ec65564
To: <sip:00491712345678 at sip.provider1.tld>
Contact: <sip:112233 at 123.123.123.123>
Call-ID: 3df2e69d5319f26668e00eae14319fa5 at 123.123.123.123
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.2
Date: Fri, 05 Mar 2010 01:37:12 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 289
v=0
o=root 2138980316 2138980316 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 14422 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:34.34.34.34:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 123.123.123.123:5060;branch=z9hG4bK2cdca6b8;rport=5060
From: "49301234567" <sip:112233 at 123.123.123.123>;tag=as6ec65564
To:
<sip:00491712345678 at sip.provider1.tld>;tag=4fa8f7eb71cc68cca91a14abea886308.16dc
Call-ID: 3df2e69d5319f26668e00eae14319fa5 at 123.123.123.123
CSeq: 102 INVITE
Proxy-Authenticate: Digest realm="sip.provider1.tld",
nonce="4b906174eb50796385f55a94796d29fa73bd824d"
Content-Length: 0
<------------->
ACK sip:00491712345678 at sip.provider1.tld SIP/2.0
Via: SIP/2.0/UDP 123.123.123.123:5060;branch=z9hG4bK2cdca6b8;rport
Max-Forwards: 70
From: "49301234567" <sip:112233 at 123.123.123.123>;tag=as6ec65564
To:
<sip:00491712345678 at sip.provider1.tld>;tag=4fa8f7eb71cc68cca91a14abea886308.16dc
Contact: <sip:112233 at 123.123.123.123>
Call-ID: 3df2e69d5319f26668e00eae14319fa5 at 123.123.123.123
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.2.2
Content-Length: 0
INVITE sip:00491712345678 at sip.provider1.tld SIP/2.0
Via: SIP/2.0/UDP 123.123.123.123:5060;branch=z9hG4bK0d6698d9;rport
Max-Forwards: 70
From: "49301234567" <sip:112233 at 123.123.123.123>;tag=as6ec65564
To: <sip:00491712345678 at sip.provider1.tld>
Contact: <sip:112233 at 123.123.123.123>
Call-ID: 3df2e69d5319f26668e00eae14319fa5 at 123.123.123.123
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.6.2.2
Proxy-Authorization: Digest username="112233", realm="sip.provider1.tld",
algorithm=MD5, uri="sip:00491712345678 at sip.provider1.tld",
nonce="4b906174eb50796385f55a94796d29fa73bd824d",
response="ac3e7d021dfbf8a384686cd702010d7f"
Date: Fri, 05 Mar 2010 01:37:12 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 289
v=0
o=root 2138980316 2138980317 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 14422 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:34.34.34.34:5060 --->
SIP/2.0 100 Giving a try
Via: SIP/2.0/UDP 123.123.123.123:5060;branch=z9hG4bK0d6698d9;rport=5060
From: "49301234567" <sip:112233 at 123.123.123.123>;tag=as6ec65564
To: <sip:00491712345678 at sip.provider1.tld>
Call-ID: 3df2e69d5319f26668e00eae14319fa5 at 123.123.123.123
CSeq: 103 INVITE
Content-Length: 0
<------------->
<--- SIP read from UDP:34.34.34.34:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 123.123.123.123:5060;branch=z9hG4bK0d6698d9;rport=5060
Record-Route: <sip:35.35.35.35;lr=on>
Record-Route: <sip:34.34.34.34;lr=on;ftag=as6ec65564>
From: "49301234567" <sip:112233 at 123.123.123.123>;tag=as6ec65564
To: <sip:00491712345678 at sip.provider1.tld>;tag=as003d1501
Call-ID: 3df2e69d5319f26668e00eae14319fa5 at 123.123.123.123
CSeq: 103 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:2400491712345678 at 36.36.36.36>
Content-Type: application/sdp
Content-Length: 262
v=0
o=root 16906 16906 IN IP4 36.36.36.36
s=session
c=IN IP4 36.36.36.36
t=0 0
m=audio 18484 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:34.34.34.34:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 123.123.123.123:5060;branch=z9hG4bK0d6698d9;rport=5060
Record-Route: <sip:35.35.35.35;lr=on>
Record-Route: <sip:34.34.34.34;lr=on;ftag=as6ec65564>
From: "49301234567" <sip:112233 at 123.123.123.123>;tag=as6ec65564
To: <sip:00491712345678 at sip.provider1.tld>;tag=as003d1501
Contact: <sip:2400491712345678 at 36.36.36.36>
Call-ID: 3df2e69d5319f26668e00eae14319fa5 at 123.123.123.123
CSeq: 103 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 262
v=0
o=root 16906 16907 IN IP4 36.36.36.36
s=session
c=IN IP4 36.36.36.36
t=0 0
m=audio 18484 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
<------------->
ACK sip:2400491712345678 at 36.36.36.36 SIP/2.0
Via: SIP/2.0/UDP 123.123.123.123:5060;branch=z9hG4bK5ad9fbda;rport
Route: <sip:34.34.34.34;lr=on;ftag=as6ec65564>,<sip:35.35.35.35;lr=on>
Max-Forwards: 70
From: "49301234567" <sip:112233 at 123.123.123.123>;tag=as6ec65564
To: <sip:00491712345678 at sip.provider1.tld>;tag=as003d1501
Contact: <sip:112233 at 123.123.123.123>
Call-ID: 3df2e69d5319f26668e00eae14319fa5 at 123.123.123.123
CSeq: 103 ACK
User-Agent: Asterisk PBX 1.6.2.2
Content-Length: 0
---
<--- SIP read from UDP:34.34.34.34:5060 --->
BYE sip:112233 at 123.123.123.123 SIP/2.0
Via: SIP/2.0/UDP 34.34.34.34:5060;branch=z9hG4bKf30f.1c23c1b2.0
Via: SIP/2.0/UDP 35.35.35.35;branch=z9hG4bKf30f.1c23c1b2.0
Via: SIP/2.0/UDP 36.36.36.36:5060;branch=z9hG4bK559edaa9;rport=5060
Max-Forwards: 68
From: <sip:00491712345678 at sip.provider1.tld>;tag=as003d1501
To: "49301234567" <sip:112233 at 123.123.123.123>;tag=as6ec65564
Call-ID: 3df2e69d5319f26668e00eae14319fa5 at 123.123.123.123
CSeq: 102 BYE
X-hint: rr-enforced
Content-Length: 0
<------------->
<--- Transmitting (no NAT) to 34.34.34.34:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
34.34.34.34:5060;branch=z9hG4bKf30f.1c23c1b2.0;received=34.34.34.34
Via: SIP/2.0/UDP 35.35.35.35;branch=z9hG4bKf30f.1c23c1b2.0
Via: SIP/2.0/UDP 36.36.36.36:5060;branch=z9hG4bK559edaa9;rport=5060
From: <sip:00491712345678 at sip.provider1.tld>;tag=as003d1501
To: "49301234567" <sip:112233 at 123.123.123.123>;tag=as6ec65564
Call-ID: 3df2e69d5319f26668e00eae14319fa5 at 123.123.123.123
CSeq: 102 BYE
Server: Asterisk PBX 1.6.2.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
<------------>
Issue History
Date Modified Username Field Change
======================================================================
2010-03-04 21:16 fa_bian Note Added: 0118997
======================================================================
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