Difference between revisions of "Call hangs after 15 minutes"
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= POSSIBLE SOLUTIONS = | = POSSIBLE SOLUTIONS = | ||
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=== Canreinvite setting === | === Canreinvite setting === | ||
Set '''canreinvite''' to '''yes''' in the Device's Edit window. | Set '''canreinvite''' to '''yes''' in the Device's Edit window. | ||
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=== Solve NAT issues === | === 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. | ||
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** check the domain in the request URI | ** check the domain in the request URI | ||
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=== Increase session timers === | === Increase session timers === | ||
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And restart Asterisk. | And restart Asterisk. | ||
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=== Disable session timers (not recommended) === | === Disable session timers (not recommended) === | ||
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
