Im having an issue where an invite is getting cancelled to quickly before the call can be forwarded on, it gets cancelled 13s on each test call i make but i need it the keep trying for more than 13s.
Is there a way to adjust this so the call can keep trying for longer before we get the cancel.
Any Asterisk variable can be used in two places in FreePBX.
1 - General settings are added in the SIP settings module
2 - Peer settings are added in the trunk details
3 - Do not edit any files by hand.
Below is a link for the Asterisk SIP variables, should find what you want. Nothing else on the page is relevant to FreePBX other than the parameter list on the bottom.
Thanks for you response after checking the parameters on the link provided i can not locate one for the invite timeout. I do see rtp and register timeout parameters but not one for an invite.
also do we add this paramter in the trunk peer? or in the sip.conf file?
I know how to add it in the trunk peer but not the SIP settings module.
Thanks for your help very much appreciated.
I don’t know what variable either. This is more an Asterisk issue.
Don’t edit sip.conf. Did you look at the sip settings module? It has a place at the bottom of the page for custom variables.
can you kindly let me know where the sip settings module is? I have found ‘Asterisk SIP settings’ under the tools tab which seems to have some settings for registration timers etc… but nothing that will help with the 13 second timer that freepbx takes to CANCEL an INVITE which hasn;t been answered
That is the module, at the bottom of the page you can add any Asterisk channel variable.
I don’r know if any of the channel settings will help you, I was simply telling you the correct place to add general settings in FreePBX. I also sent you the link to the site with all the SIP settings.
If you can find one that assists your situation you can post of at asterisk.org and see if anyone can offer any assistance.