Difference between revisions of "Call hangs after 15 minutes"
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In the general section in /etc/asterisk/sip.conf: session-timers=refuse | In the general section in /etc/asterisk/sip.conf: session-timers=refuse | ||
<span style="color: red"> | |||
IMPORTANT: Drawback is that a call can show as going on forever if the BYE message is lost due to network problems.</span> | |||
<br><br> | <br><br> | ||
= See also = | = See also = | ||
* [https://tools.ietf.org/html/rfc4028 Session Timers in the Session Initiation Protocol (SIP) RFC 4028] | * [https://tools.ietf.org/html/rfc4028 Session Timers in the Session Initiation Protocol (SIP) RFC 4028] | ||
Revision as of 16:25, 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
IMPORTANT: Drawback is that a call can show as going on forever if the BYE message is lost due to network problems.
