[OpenSIPS-Users] ACK never reach UAS

[Max Mühlbronner]

If i remember correctly "building telephony systems with opensips" (flavios
great book) suggests to disable record routing in opensips for testing with
sipp.


Best Regards


Max M.

-----Ursprüngliche Nachricht-----
Von: users-bounces at lists.opensips.org
[mailto:users-bounces at lists.opensips.org] Im Auftrag von M.Abdulaziz
Gesendet: Freitag, 30. Dezember 2011 20:33
An: users at lists.opensips.org
Betreff: Re: [OpenSIPS-Users] ACK never reach UAS



Thank you very much Vald & Juse for your valuable help but it didn't work
either

I attached a sipp trace of the messages for one call
http://opensips-open-sip-server.1449251.n2.nabble.com/file/n7138850/SIPTrace
SIPTrace 

(Call-ID: 12-2298 at 192.168.1.69)

UAC Port: 5063
opensips: 5060
UAS Port: 5061

*for UAS I used the embedded scenario*
*Here is my scenario file of UAC*

<scenario name="Basic Sipstone UAC">
  
  
  <send retrans="500" >
    
  </send>

  <recv response="100" rrs="true" optional="true">
  </recv>

  <recv response="180"  rrs="true" optional="true">
  </recv>

  
  
  
  <recv response="200"  rrs="true" rtd="true">
  </recv>

  
  
  <send >
    
  </send>

  
  
  <pause milliseconds = "4000"/>

  
  <send retrans="500">
    
  </send>

  <recv response="200"  rrs="true" crlf="true">
  </recv>

  
  <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>

  
  <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>

</scenario>


Thank you in advance for your help,

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