Hi Guys,
Not typed here for many a year - and its a question I find myself sincerely embarrassed to ask!
But the man who didn't ask a question didn't learn anything and I might help someone else in doing so.
Been building and working with numerous builds of asterisk as a small part of my role since about V1.4
but have always spent 98% of my config time in either the Shell/CLI - not really FreePBX since running AAH
back in the day.
I now find myself supporting an up-to-date incredible build in a "hobbyist" role, but because certain colleagues are only familiar with the FreePBX GUI , I have been asked to ....curb....my desire to just go to straight to CLI followed by looking at configs in /etc/asterisk/*.* to solve an issue / tweak a feature.
Whilst me doing so means things get fixed quickly, there is a group fear that I might:
* Break FreePBX - preventing the team from administering it when I'm not around;
* Do something that doesn't appear in FreePBX - so other team members dont realise its there;
* Make config changes that get overwritten by FreePBX blocking a service when someone else
makes a change;
So I have no choice but to get as familiar with FreePBX as I can . Which gives rise to this question:
Trunks menu
SIP settings (outgoing / incoming)
Am I correct in thinking that whatever I type relating to the sip settings in either the user or peer fields of the trunk menu, is just concatenated into one block of text in the SIP config file ?
And therefore I <could> (not <should>) put any parameter in either field without impacting trunk operation?
If I am correct, is there a definitive guide to which trunk definition elements, by FreePBX convention, should
normally go into which GUI menu field please ? I acknowledge I could just prove this by building a dummy trunk but there is presently only the production platform (Don't go there - this is neither a Telco nor IT - in fact its not a "business" as such)
The long term goal is to get the rest of the team as comfortable in CLI / shell as I can, but in this early stage
there has to be give and take.
Regards
BB
Not typed here for many a year - and its a question I find myself sincerely embarrassed to ask!
But the man who didn't ask a question didn't learn anything and I might help someone else in doing so.
Been building and working with numerous builds of asterisk as a small part of my role since about V1.4
but have always spent 98% of my config time in either the Shell/CLI - not really FreePBX since running AAH
back in the day.
I now find myself supporting an up-to-date incredible build in a "hobbyist" role, but because certain colleagues are only familiar with the FreePBX GUI , I have been asked to ....curb....my desire to just go to straight to CLI followed by looking at configs in /etc/asterisk/*.* to solve an issue / tweak a feature.
Whilst me doing so means things get fixed quickly, there is a group fear that I might:
* Break FreePBX - preventing the team from administering it when I'm not around;
* Do something that doesn't appear in FreePBX - so other team members dont realise its there;
* Make config changes that get overwritten by FreePBX blocking a service when someone else
makes a change;
So I have no choice but to get as familiar with FreePBX as I can . Which gives rise to this question:
Trunks menu
SIP settings (outgoing / incoming)
Am I correct in thinking that whatever I type relating to the sip settings in either the user or peer fields of the trunk menu, is just concatenated into one block of text in the SIP config file ?
And therefore I <could> (not <should>) put any parameter in either field without impacting trunk operation?
If I am correct, is there a definitive guide to which trunk definition elements, by FreePBX convention, should
normally go into which GUI menu field please ? I acknowledge I could just prove this by building a dummy trunk but there is presently only the production platform (Don't go there - this is neither a Telco nor IT - in fact its not a "business" as such)
The long term goal is to get the rest of the team as comfortable in CLI / shell as I can, but in this early stage
there has to be give and take.
Regards
BB