Difference between revisions of "Call hangs after 15 minutes"

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Some providers perform a SIP re-INVITE every 15 minutes for any active calls. It's a way of checking if the call is indeed still active. If an appropriate 200 OK is not received back, they disconnect the call.
Some providers perform a SIP re-INVITE every 15 minutes for any active calls. It's a way of checking if the call is indeed still active. If an appropriate 200 OK is not received back, they disconnect the call.


'''POSSIBLE SOLUTIONS:'''
<br><br>
= POSSIBLE SOLUTIONS =


Refuse SIP re-invites: In the general section in /etc/asterisk/sip.conf: session-timers=refuse
<br><br>
=== Canreinvite setting ===


Set '''canreinvite''' to '''yes''' in the Device's Edit window.


Could be NAT issues:
<br><br>
=== Solve NAT issues ===
* Make sure network is configured properly and all packets reach the Softswitch.  
* Make sure network is configured properly and all packets reach the Softswitch.  
* Do pcap capture and analyze the logs.
* Do pcap capture and analyze the logs.
** check the domain in the request URI
<br><br>
=== Increase session timers ===
Add in /etc/asterisk/sip.conf [general] section:
session-timers=originate
session-expires=10800
session-minse=300
session-refresher=uas
And restart Asterisk.
<br><br>
=== Disable session timers (not recommended) ===
In the general section in /etc/asterisk/sip.conf: session-timers=refuse
<br><br>
= See also =
* [https://tools.ietf.org/html/rfc4028 Session Timers in the Session Initiation Protocol (SIP) RFC 4028]

Revision as of 16:22, 22 February 2016

Switch may not be responding to the SIP re-INVITE.

Some providers perform a SIP re-INVITE every 15 minutes for any active calls. It's a way of checking if the call is indeed still active. If an appropriate 200 OK is not received back, they disconnect the call.



POSSIBLE SOLUTIONS



Canreinvite setting

Set canreinvite to yes in the Device's Edit window.



Solve NAT issues

  • Make sure network is configured properly and all packets reach the Softswitch.
  • Do pcap capture and analyze the logs.
    • check the domain in the request URI



Increase session timers

Add in /etc/asterisk/sip.conf [general] section:

session-timers=originate session-expires=10800 session-minse=300 session-refresher=uas

And restart Asterisk.



Disable session timers (not recommended)

In the general section in /etc/asterisk/sip.conf: session-timers=refuse




See also