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 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 525 (501) 10.0.1.57 10.0.1.103 Status: 200 OK 526 10.0.1.57 10.0.1.103 Status: 487 Request Terminated 527 (for 502) 10.0.1.57 10.0.1.103 Status: 200 OK 528 (502) 10.0.1.103 10.0.1.57 Request: ACK sip:104@10.0.1.57:5060 585 (524) 10.0.1.103 10.0.1.57 Status: 200 OK, with SDP (resend of 500) 588 (524) 10.0.1.57 10.0.1.103 Request: ACK sip:8*104_line104@10.0.1.103:5060 803 (588) 10.0.1.103 10.0.1.57 Status: 200 OK, with SDP (resend of 500) 806 (588) 10.0.1.57 10.0.1.103 Request: ACK sip:8*104_line104@10.0.1.103:5060 1223 (806) 10.0.1.103 10.0.1.57 Status: 200 OK, with SDP (resend of 500) 1229 (806) 10.0.1.57 10.0.1.103 Request: ACK sip:8*104_line104@10.0.1.103:5060 2042 (1229) 10.0.1.103 10.0.1.57 Status: 200 OK, with SDP (resend of 500) 2044 10.0.1.57 10.0.1.103 Request: ACK sip:8*104_line104@10.0.1.103:5060 2886 10.0.1.103 10.0.1.57 Status: 200 OK, with SDP 2888 10.0.1.57 10.0.1.103 Request: ACK sip:8*104_line104@10.0.1.103:5060 3752 10.0.1.103 10.0.1.57 Status: 200 OK, with SDP 3755 10.0.1.57 10.0.1.103 Request: ACK sip:8*104_line104@10.0.1.103:5060 --------------------------------------------------------------------------------------------------------- with WAIT(1). There is no more messages beyond 672 until the call is over. Everything is normal. There is no ACK for the OK for INVITE in 430 here either. 425 10.0.1.57 10.0.1.103 Request: INVITE sip:8*104_line104@10.0.1.103, with SDP 426 10.0.1.103 10.0.1.57 Status: 401 Unauthorized 429 (425) 10.0.1.57 10.0.1.103 Request: ACK sip:8*104_line104@10.0.1.103 430 10.0.1.57 10.0.1.103 Request: INVITE sip:8*104_line104@10.0.1.103, with SDP 431 10.0.1.103 10.0.1.57 Status: 100 Trying 432 10.0.1.103 10.0.1.57 Request: MESSAGE sip:104@10.0.1.57:5060 (text/plain) 443 (for 432) 10.0.1.57 10.0.1.103 Status: 200 OK 645 (for 430) 10.0.1.103 10.0.1.57 Status: 200 OK, with SDP 646 10.0.1.103 10.0.1.57 Request: NOTIFY sip:104@10.0.1.57:5060 647 10.0.1.103 10.0.1.57 Request: CANCEL sip:104@10.0.1.57:5060 667 (443) 10.0.1.57 10.0.1.103 Request: ACK sip:8*104_line104@10.0.1.103:5060 668 (646) 10.0.1.57 10.0.1.103 Status: 200 OK 670 10.0.1.57 10.0.1.103 Status: 487 Request Terminated 671 (647) 10.0.1.57 10.0.1.103 Status: 200 OK 672 (for 647) 10.0.1.103 10.0.1.57 Request: ACK sip:104@10.0.1.57:5060 -------------------------------------------------------------------------------------------------------- > Date: Fri, 16 Mar 2012 10:22:49 -0500 > From: kpflem...@digium.com > To: asterisk-users@lists.digium.com > Subject: Re: [asterisk-users] SendText causes Retransmission errors > > On 03/16/2012 09:43 AM, Matt Hamilton wrote: > > Hi, > > > > I'm using SendText to send a text message when the user picks up a line > > in a SLA setup (even though I'm not sure the problem is related to SLA). > > I'm on Asterisk 10.2.1 (same in 1.8.9) > > > > > > [from-office] > > .. > > same => n,SendText(hi) > > same => n,SLAStation(line1234) > > .. > > > > Here is a simplified version of the SIP messages: > > > > 1 phone => Asterisk INVITE > > 2 Asterisk => phone Trying > > 3 Asterisk => phone MESSAGE > > 4 Asterisk => phone OK (for the INVITE at 1) > > 5 phone => Asterisk OK (for the MESSAGE at 3) > > > > 6 Asterisk => phone OK (for the INVITE at 1)*** RESEND of 4 > > 7 Asterisk => phone OK (for the INVITE at 1)*** RESEND of 4 > > .. > > Did the phone send an ACK for message 4? If not, that explains why > Asterisk is retransmitting the '200 OK'. Posting a packet capture of > this problem occurring would probably provide the details necessary to > figure out what is going on. > > -- > 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
-- _____________________________________________________________________ -- 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