Difference between revisions of "Call hangs after 15 minutes"
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. | 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. | |||
<br><br> | |||
= POSSIBLE SOLUTIONS = | |||
<br> | |||
=== Change Provider === | |||
Change Provider and check if issue is gone to confirm the origin of the problem. | |||
<br> | |||
=== Canreinvite setting === | |||
Set '''canreinvite''' to '''yes''' in the Device's Edit window. | |||
<br> | |||
=== 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 | |||
** check if Public IP addresses are placed in SIP headers | |||
<br> | |||
=== Adjust session timers === | |||
In Device or Provider settings page you can change these parameters: | |||
session-timers | |||
session-expires | |||
session-minse | |||
session-refresher | |||
Check what settings User or Provider is requiring and set accordingly. | |||
<br> | |||
=== Disable session timers (not recommended) === | |||
In the general section in /etc/asterisk/sip.conf: session-timers=refuse | |||
It can be done per Device in Device Settings page. | |||
<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> | |||
= See also = | |||
* [https://tools.ietf.org/html/rfc4028 Session Timers in the Session Initiation Protocol (SIP) RFC 4028] | |||
* https://andrewjprokop.wordpress.com/2015/02/10/understanding-sip-re-invite | |||
Latest revision as of 16:22, 31 October 2020
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
Change Provider
Change Provider and check if issue is gone to confirm the origin of the problem.
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
- check if Public IP addresses are placed in SIP headers
Adjust session timers
In Device or Provider settings page you can change these parameters:
session-timers
session-expires
session-minse
session-refresher
Check what settings User or Provider is requiring and set accordingly.
Disable session timers (not recommended)
In the general section in /etc/asterisk/sip.conf: session-timers=refuse
It can be done per Device in Device Settings page.
IMPORTANT: Drawback is that a call can show as going on forever if the BYE message is lost due to network problems.
