On 03/18/2012 08:07 PM, Matt Hamilton wrote:
Kevin,thanks for your response.
Here is the more detailed Wireshark capture of the SIP traffic between
phone (10.0.1.57) and Asterisk (10.0.1.103). The numbers between
parentheses are Request Frames. I don't see an ACK for the 200 OK to the
INVITE (491) for the dialplan that gives us Retransmission errors
(without WAIT), but there is also no ACK for the same INVITE for the
dialplan that works (with WAIT).
If you still want to take a look at the full packet capture, I'll post it.
Matt
---------------------------------------------------------------------------------------------
Without WAIT(1) - we get Retransmisson errors
486 10.0.1.57 10.0.1.103 Request: INVITE sip:8*104_line104@10.0.1.103,
with SDP
487 10.0.1.103 10.0.1.57 Status: 401 Unauthorized
490 (486) 10.0.1.57 10.0.1.103 Request: ACK sip:8*104_line104@10.0.1.103
491 10.0.1.57 10.0.1.103 Request: INVITE sip:8*104_line104@10.0.1.103,
with SDP
492 10.0.1.103 10.0.1.57 Status: 100 Trying
493 10.0.1.103 10.0.1.57 Request: MESSAGE sip:104@10.0.1.57:5060
(text/plain)
*500 (for 491) 10.0.1.103 10.0.1.57 Status: 200 OK, with SDP*
501 10.0.1.103 10.0.1.57 Request: NOTIFY sip:104@10.0.1.57:5060
502 10.0.1.103 10.0.1.57 Request: CANCEL sip:104@10.0.1.57:5060
Why did Asterisk CANCEL the call here?
*503 (for 493) 10.0.1.57 10.0.1.103 Status: 200 OK*
524 (503) 10.0.1.57 10.0.1.103 Request: ACK
sip:8*104_line104@10.0.1.103:5060
This appears to be broken. The listing here claims this ACK is in
response to the '200 OK' in packet 503, which itself was a final
response to the MESSAGE request in packet 493. However, MESSAGE requests
do not use ACK for a three-way handshake like INVITE requests do. In
addition, this packet is going the wrong direction to be an ACK for
packet 503, since it's going the same direction as packet 503 did.
Whatever method you used to generate this report seems to be broken. I
can't tell you exactly how it is broken without seeing the headers in
the messages, though.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com & www.asterisk.org
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
