Difference between revisions of "Call hangs after 15 minutes"

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= POSSIBLE SOLUTIONS =
= POSSIBLE SOLUTIONS =


<br><br>
<br>
=== Change Provider ===
 
Change Provider and check if issue is gone to confirm the origin of the problem.
 
<br>
=== Canreinvite setting ===
=== Canreinvite setting ===


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


<br><br>
<br>
=== 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.  
* Do pcap capture and analyze the logs.
* Do pcap capture and analyze the logs.
** check the domain in the request URI
** check the domain in the request URI
** check if Public IP addresses are placed in SIP headers
<br>


<br><br>
=== Adjust session timers ===
=== Increase session timers ===
 
In Device or Provider settings page you can change these parameters:
 
session-timers
 
session-expires
 
session-minse
 
session-refresher


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


session-timers=originate
Check what settings User or Provider is requiring and set accordingly.
session-expires=10800
session-minse=300
session-refresher=uas


And restart Asterisk.
<br>


<br><br>
=== Disable session timers (not recommended) ===
=== Disable session timers (not recommended) ===


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



See also