[OpenSIPS-Users] Un recognized sip header from asterisk to OpenSIPS

[sevpal]

Hi, do a normal sip trace using the sip trace module in opensips and post the output here. From what that can be understood from this log though, is that there are headers “x-asterisk-hangupcause:” “x-asterisk-hangupcausecode:”  that are being rejected due to validity or case sensitivity, possibly the response from the user agent. 

Where are these headers being added?

From: Travis Manson-Drake 
Sent: Thursday, March 31, 2016 1:47 PM
To: OpenSIPS users mailling list 
Subject: [OpenSIPS-Users] Un recognized sip header from asterisk to OpenSIPS

Hello everyone.

 

Hope your all doing well!

 

I seem to be having an issue in which when a call is sent through OpenSIPS to my Asterisk PBX asterisk with eventually send a BYE with a hang up Cause of 111/unrecognized sip header. I looked at the headers of all my packets but can’t find anything out of the norm. has anyone experienced this before and ideas on what it might be or what I might check?

 



 

I found a few article on asterisk forums mention NAT issues, but I’ve implemented a NAT helper into my routing logic so that shouldn’t be the case.

 

Thank you all for your time

 

      Travis Manson-Drake
      Voice Systems Analyst
     

 

      Simply Bits, LLC
      T: 520.545.0311  F: 520.545.7252
      E: travism at simplybits.com 
      5225 N. Sabino Canyon Road
      Tucson, AZ 85750
      Support Hotline: 520.545.0333
     

 

 



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