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Obsolete Releases => SME VoIP (Asterisk, SAIL etc) => Topic started by: rrkelly on March 10, 2008, 11:24:04 AM

Title: sail route config questions
Post by: rrkelly on March 10, 2008, 11:24:04 AM
How does sail differentiate between an in bound and out bound route?
in the trunk conf screen what is an open inbound route ? what is a closed inbound route?
also on the trunk screen the two text boxes on the bottom
the left one is for what? the right one is for what? i don't want to guess.

an alias can be marked as a ring group but where is it hooked to an inbound route?



thanks
rob

Title: Re: sail route config questions
Post by: SARK devs on March 10, 2008, 05:42:23 PM
Hi there

Quote
How does sail differentiate between an in bound and out bound route?

Outbound is handled through the route panel.  It is sensitive to dial patterns and it routes calls over the correct trunk based upon those patterns.  There are a couple of exceptions which can subvert the route abstraction, these are aliased outbound dials (which seize the trunk directly) and trunk-preselect where a trunk is deliberately seized by prepending the dialstring with a two digit code (defined in the trunk).

Inbound is handled through the trunk structure (usually through a DiD).  There are essentially three inbound pathways; the open inbound route (which is used if the system is in an OPEN state), the closed inbound route (which is used if the system is in a CLOSED  state) and the natural route which is used during extension-to-extension dialing and also by privileged siblings. The natural route cannot be altered or influenced except by natural routings such as call forwards, DNDs and so forth. 

Quote
also on the trunk screen the two text boxes on the bottom the left one is for what? the right one is for what?

I assume you mean the peer and user entries you see when when you clcik to edit a trunk.  These are the sip.conf (or iax.conf) peer and user entries which SAIL has generated for you.  You may freely modify these entries to suit particular circumstances but in most cases they should just work.


Quote
an alias can be marked as a ring group but where is it hooked to an inbound route?

In the inbound route drop-down menu.  By convention, you should give aliases numeric names (keys - 4 digit).  The reason for this is that you can then test them by the simple expedient of dialing them. :-)

Full info on trunk fields and what they do can be found here...

http://www.selintra.com/docs/cgi-bin/view/Main/DocChapter096

and for routes...

http://www.selintra.com/docs/cgi-bin/view/Main/DocChapter112

Kind Regards

S


Kind Regards

S

 

 





Title: Re: sail route config questions
Post by: rrkelly on March 11, 2008, 12:05:07 PM
open state is simply if your are within  hours of normal operation and closed is after hours? provided
you set a time period for the route to be open within the automation menu

so how is it you are supposed to tie a given trunk  to a ring group (alias)
i have an inbound sip trunk that is used by my sip provider for a local DID i have all of my extensions
in an  alias marked as a ring group and it keeps trying to use the callid as an extention which of course does not exist. I did not see where to set the DID currently the open route is set to my ring group alias.

if i read your reply correctly in coming routes are set thru the trunk config ?

Title: Re: sail route config questions
Post by: SARK devs on March 11, 2008, 02:44:29 PM
You can use automation or you can use *30*, *31* to throw the system to open or closed.  See the feature key settings in the /docs appendices.


How have you named the trunk?  Did you use the DiD you are trying to route, or some other name? Here's a trunks panel from one of our servers...

(http://selintra.com/trunks.jpg)

Notice that the names of the trunks (extreme left) are the actual DiD numbers which those trunks are going to receive and route.  In the case where the DiD number is NOT the same as the trunk name then you can create a special trunk called a PTT_DiD_Group (we have two ISDN DiDs defined in the example).   In such cases, it is the inbound route of the PTT_DiD_Group which will decide where your call goes.


S