Hi,
Im a new user to freepbx and the purpose is to allow DISA on our new shoretel system.
from reading around i have tried several setups and it looks like the way for me is a sip trunk between the 2 systems to direct the DDI from the shoretel system to an extension on the freepbx and then a sip trunk to a user account on shoretel system to establish an external dial tone. The problem im having is that the freepbx will not connect to the shoretel system’s user account to give me an outside line.
This may be the wrong way to do it and if so could somebody please point me in the right direction.
The config of the sip trunk to the user account is as follows:
Trunk name: 2901
outbound callerid: 2901
trunkname: out2
PEER Details
disallow=all
allow=ulaw
canreinvite=no
dtmfmode=rfc2833
fromdomain=172.16.102.10
fromuser=2901
host=172.16.102.10
insecure=very
secret=12345678
type=peer
username=disa01
user=2901
authname=disa01
qualify=yes
dtmf=auto
Incoming settings
context=from-trunk
host=172.16.102.10
secret=12345678
type=user
username=disa01
disallow=all
allow=ulaw
Regisration String:
disa01:[email protected]/disa01
any help would be much appreciated.
I think the first thing to do is use valid configuration variables.
Also it looks like you simply invented settings as nothing makes sense, you have insecure=very set then you have every authentication mechanism activated and some that don’t even exist!
You want to use the minimum data necessary to authenticate.
With fixed IP addresses you do not need to register to the shoretel.
What is the Shoretel trunk config look like?
Yo have the following variables you used don’t even exist: user, DTMF, and authname.
You don’t need separate incoming peer when they are the same host.
Here is a link on Asterisk SIP variables and usage:
http://www.voip-info.org/wiki/view/Asterisk+config+sip.conf
I think it is funny that you paid all that money for the Shoretel and now have to use FreePBX to get the features you need. FreePBX is free and blows the Shoretel stuff away in terms of stability, scalability, reliability and of course features. You also don’t have to buy the overprices OEM phones they use and “ShoreGear” switches. Can you get your money back?
Hi,
Thanks for the reply. The variables i’ve used are more than likely muddled up as ive used several different approaches and i’m sure i copied some from someone else who said theyd getten the same thing working on theirs.
I’ll have a read through the variables but ive been able to get both systems communicating with each other although once disa tries to go back out through the shoretel system, then it treats the trunk as external coming in which then stops disa working. This was my basis of wanting to try using a trunk on the freepbx system as a sip extension on the shoretel system. This way the shoretel system will not need any abnormal settings (custom).
Unfortunately the choice to go with shoretel was a corporate decision and was based on the other bonuses it gives. Our old nec system could do disa fine but in our tender we specified that mobile phones should be recorded and they complied although we didnt realise that this meant with only a handful of devices and through software thats not as easy as holding 9 with a speed dial attached.
A long story short is that I could get this working if i could just set a freepbx trunk as a sip exension on shoretel.
Thanks
Until you clean out the crap in the trunk there is little chance of it working.
You do not need two seperate peers for inbound and outbound. Just 1 with type=friend.
You also have no context in the first peer that is a duplicate (subset) of the second peer. For the trunk to look like an extension in needs to be in the from-internal dial plan not from trunk. Problem with from-internal is it can’t be the target of an inbound route and hit the DISA.
Ok i’ve minimised the config to:
type=peer
port=5060
host=172.16.102.13
context=from-internal
and
type=user
host=172.16.102.13
context=from-internal
port=5060
Still the same issue but at least i’ve got the basics to work from. So as I call the offsite extension on the shoretel system I get the disa prompts and then when I try to call out, the shoretel system is seeing the call as an inbound call as its from a sip trunk. This is why I can only assume that a sip extension would be needed on the freepbx side so that shoretel will allow an outbound isdn trunk. I really dont mind getting someone to know what they are doing to do this (paying) but i was hoping that it would be a simple few lines to connect (freepbx trunk) - (shoretel sip extension). Also i would have liked to know if this is even possible.
Thanks