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Summary:ASTERISK-19864: Asterisk replying to Session progress with an Ack then an Invite
Reporter:Brian (iptel)Labels:
Date Opened:2012-05-12 02:07:25Date Closed:2012-06-22 08:52:04
Priority:MajorRegression?
Status:Closed/CompleteComponents:Channels/chan_sip/General
Versions:1.8.12.0 Frequency of
Occurrence		Constant
Related
Issues:
Environment:Debian SqueezeAttachments:
Description:Asterisk sends invite as normal to SIP provider.

When provider sends back 183 progress - asterisk responds with an Ack then reinvites again.

Provider quite rightly sends back 491 Request Pending.

Sanitized relevant parts of trace:

#
U 2012/05/11 23:43:17.262756 10.10.10.50:5060 -> 192.10.10.81:1024
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.10.10.81:47324;received=192.10.10.81;branch=z9hG4bK05cd0aed;rport=1024
Record-Route: <sip:10.10.10.50:5060;nat=yes;ftag=as05265827;lr;cudg=07c.d5dd1884>
From: "01234567" <sip:087654321@sip.blahblah.co>;tag=as05265827
To: <sip:9876543@sip.blahblah.co>;tag=as233404c8
Call-ID: 23d467e42a7a0c9530b859a920428ecb@sip.blahblah.co
CSeq: 103 INVITE
Server:  MEDIA GW
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:20039876543@10.10.10.110:5060>
Content-Type: application/sdp
Content-Length: 258

v=0
o=root 1216464932 1216464932 IN IP4 10.10.10.110
s= MEDIA
c=IN IP4 10.10.10.11
t=0 0
m=audio 58376 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=ptime:20
a=sendrecv

#
U 2012/05/11 23:43:17.370426 192.10.10.81:1024 -> 10.10.10.50:5060
ACK sip:9876543@sip.blahblah.co SIP/2.0
Via: SIP/2.0/UDP 192.10.10.81:47324;branch=z9hG4bK05cd0aed;rport
Max-Forwards: 70
From: "01234567" <sip:087654321@sip.blahblah.co>;tag=as05265827
To: <sip:9876543@sip.blahblah.co>;tag=86dcc0529716e713f1be0da1dbbcaa0e.df63
Contact: <sip:087654321@192.10.10.81:47324>
Call-ID: 23d467e42a7a0c9530b859a920428ecb@sip.blahblah.co
CSeq: 103 ACK
User-Agent: ASTERISK 1.8.12.0
Content-Length: 0


#
U 2012/05/11 23:43:17.372987 192.10.10.81:1024 -> 10.10.10.50:5060
INVITE sip:9876543@sip.blahblah.co SIP/2.0
Via: SIP/2.0/UDP 192.10.10.81:47324;branch=z9hG4bK754a6dac;rport
Max-Forwards: 70
From: "01234567" <sip:087654321@sip.blahblah.co>;tag=as05265827
To: <sip:9876543@sip.blahblah.co>
Contact: <sip:087654321@192.10.10.81:47324>
Call-ID: 23d467e42a7a0c9530b859a920428ecb@sip.blahblah.co
CSeq: 104 INVITE
User-Agent: ASTERISK 1.8.12.0
Proxy-Authorization: Digest username="087654321", realm="sip.blahblah.co", algorithm=MD5, uri="sip:9876543@sip.blahblah.co", nonce="4fad96230001120a304dba74debde21ea297de778462597c", response="4bcd32d61c2280157c86daaf035d1374"
Date: Fri, 11 May 2012 22:43:17 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 281


Perhaps I am doing something silly but I have the exact same setup working fine with 1.8.9.3

Thanks for your help.

Regards
Brian.
Comments:By: Rusty Newton (rnewton) 2012-05-21 16:42:55.347-0500

Thank you for taking the time to report this bug and helping to make Asterisk better. Unfortunately, we cannot work on this bug because your description did not include enough information. You may find it helpful to read the Asterisk Issue Guidelines http://www.asterisk.org/developers/bug-guidelines. We would be grateful if you would then provide a more complete description of the problem. At a minimum, we need:

1. the specific steps or actions you took that caused you to encounter the problem,
2. the behavior you expected, and
3. the behavior you actually encountered (in as much detail as possible).

This likely includes output from the console with debug level logging, a SIP trace (if this is SIP related), and configuration information such as dialplan (e.g. extensions.conf) and channel configuration (e.g. sip.conf). Thanks!


I realize you sanitized part of the submitted debug. If you are able, please provide a full pcap from the beginning of the first INVITE through to the end of the call.
By: Walter Doekes (wdoekes) 2012-05-22 07:07:36.077-0500

Looks to me like you've been sanitizing too much:

from media gw
{noformat}
From: "01234567" <sip:087654321@sip.blahblah.co>;tag=as05265827
To: <sip:9876543@sip.blahblah.co>;tag=as233404c8
Call-ID: 23d467e42a7a0c9530b859a920428ecb@sip.blahblah.co
{noformat}

from asterisk
{noformat}
From: "01234567" <sip:087654321@sip.blahblah.co>;tag=as05265827
To: <sip:9876543@sip.blahblah.co>;tag=86dcc0529716e713f1be0da1dbbcaa0e.df63
Call-ID: 23d467e42a7a0c9530b859a920428ecb@sip.blahblah.co
{noformat}

Observe the diffeent To-tag.

I'd say that there is a parallel branched response that asterisk is correctly replying to.
By: Matt Jordan (mjordan) 2012-06-22 08:52:04.413-0500

Per Walter's comment, closing as "not a bug".