Tek-Tips is the largest IT community on the Internet today!

Members share and learn making Tek-Tips Forums the best source of peer-reviewed technical information on the Internet!

  • Congratulations strongm on being selected by the Tek-Tips community for having the most helpful posts in the forums last week. Way to Go!

Asterisk sip trunk to LG iPECS

Status
Not open for further replies.

eddiebullman

Programmer
Jun 24, 2008
140
GB
Hi

i am having problems setting up a sip trunk between a LG iPECS UCP 100 to a AsteriskNOW server

i can call from the Asterisk server to the iPECS with now problems but when i dial from the iPECS to the asterisk in comes up on the display normal clearing and drops the call

i know this probably a setting in the asterisk server but if some one can help it would much appreciated

i use the web GUI so if you give me pointer for this as i don't have a clue when it comes to command line

thanks
 
eddiebullman,
it sounds like maybe there isn't an extension (or pattern match) setup to catch calls in the context that you have your sip trunk set in. Do you know how many digits your LG system is sending across the trunk?

if 4 it could be as simple as something like
Code:
exten => _XXXX,1,Goto(usercontext,${EXTEN},1)

where "usercontext" is whatever context your endpoints are set in.

Alternatively, and I'm not sure if this is a good practice or not, you could probably just point your SIP trunk to whatever context your extensions are in and so long as what the other system is passing matches the extension number it should connect the call.

I'm used to CLI so I'm not sure how to tell you to do this in the GUI but I'll give it a shot if this doesn't help.


Edit: If you need to do some digit manipulation (like say your extensions on Asterisk side are 4 digits and your LG system is sending 7 or 10 or whatever I can help with that. Not tough on CLI. I'll see about downloading Asterisk NOW in a VM and try and figure out how the GUI interacts with the flat files.


 
Hi

this is what i have set on the trunk

sipoutgoing2_n9btn8.jpg


sipincoming2_ywqyja.jpg



if you could help with this it would be great

thanks
 
Ok, can you post one of the endpoints you're trying to call to as well. not being able to take a look at the system myself we'll have to figure a way to get some CLI data of a call that failed to try and figure out what's amiss. I'm installing AsteriskNOW in vmware as I type this so hopefully that will allow me to be able to point you in the right direction to get the info to troubleshoot.
-Drew

 
Depending on your current log settings we might be able to just pull historical data if they are verbose enough to tell what happened. Will respond again in a bit when I get the system up and familiarize myself with the GUI. Could walk you through the command line version easily but I'm not yet familiar with freepbx interface. Hopefully there's a way to watch the Asterisk command line and see the call flow real-time. I imagine it's just a case of either wrong context or one system sending one thing and the other expecting something else.

 
Alright, I've found how to find the info we need. Go ahead and make a call that will fail so it'll be fresh in the logs. Then in the freepbx webgui click on "Reports" at the top and scroll down to "Asterisk Log Files"

You should see the "File" box at top says "full"

Highlight and select all 500 lines (hopefully that's enough if the call was made just before you pulled the logs) and copy and paste that into some sort of pastebin. I personally use for most things or if I'm feeling more paranoid. Hastebin is supposed to disappear after 30 days though so probably fine too.



 
Hi ddickenson

thanks for the reply i will have a go at what you said when i am back at work which is 26/04/17 will chat then and once again thanks

Eddie
 
Status
Not open for further replies.

Part and Inventory Search

Sponsor

Back
Top