[asterisk-users] SendText causes Retransmission errors

Matt Hamilton mistral9999 at hotmail.com
Sun Mar 18 20:07:47 CDT 2012


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 at 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 at 10.0.1.103             
 491            10.0.1.57    10.0.1.103   Request: INVITE sip:8*104_line104 at 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 at 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 at 10.0.1.57:5060      
 502            10.0.1.103   10.0.1.57    Request: CANCEL sip:104 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 10.0.1.103     
 430            10.0.1.57    10.0.1.103   Request: INVITE sip:8*104_line104 at 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 at 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 at 10.0.1.57:5060      
 647            10.0.1.103   10.0.1.57    Request: CANCEL sip:104 at 10.0.1.57:5060      
 667 (443)      10.0.1.57    10.0.1.103   Request: ACK sip:8*104_line104 at 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 at 10.0.1.57:5060 

--------------------------------------------------------------------------------------------------------






> Date: Fri, 16 Mar 2012 10:22:49 -0500
> From: kpfleming at digium.com
> To: asterisk-users at 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: kfleming at digium.com | SIP: kpfleming at digium.com | Skype: kpfleming
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> Check us out at www.digium.com & www.asterisk.org
> 
> --
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