PS> Here is an email reply with my setup might help anyone looking to integrate STS and asterisk. I like STS but need to have a few remote phones at home offices etc.. Asterisk is a cheap way to do this using one of our existing linux servers and a $14 X100p card from ebay connected to a SLT port on the STS:
I thought about using CO but then you would not be able to dial say "100" from VoIP and send the call to STS extension 100. Right now I have something like this on my asterisk:
Extensions.conf------
[general]
static=yes
writeprotect=no
[globals]
TRUNK=Zap/g1 ; Trunk interface
[sip]
ignorepat => 9
exten => _1XX,1,Dial(Zap/1/w${EXTEN})
exten => 200,1,Dial(SIP/voip0,20)
exten => 201,1,Dial(SIP/voip1,20)
exten => 202,1,Dial(SIP/voip2,20)
exten => 203,1,Dial(SIP/voip3,20)
exten => 210,1,Dial(SIP/officetrail0,20)
exten => 211,1,Dial(SIP/officetrail1,20)
exten => 212,1,Dial(SIP/officetrail2,20)
.....
exten => 8,1,Dial(Zap/1/)
exten => 8,2,Congestion
exten => 9,1,Dial(Zap/1/w802)
exten => 9,2,Congestion
----------
So currently this works and if a VoIP user wants to dial a user on STS they just dial the extension, say 100 or 102 ..etc from the voip phone. Then it sends the call out the trunk (X100p) which is connected to the SLT port on the STS.
So hitting 9 on voip will get them an outside PSTN line to dial an outside line. Hitting 8 will get the internal bbbbbbbbbbbbeeeeeee noise when u pick up a Vodavi STS phone which then will let you dial say voicemail etc... If a VoIP user wants to dial another Voip they just dial the 211 or 201 etc...
So my only remaining complaints with this setup are:
A) No CLID into/out of VoiP due to SLT port not supporting
B) STS phone users cannot dial VoIP users from the STS phones.
To fix B, I am thinking about setting up another line as a CO Line going to my asterisk then whenever a starplus user wants to dial a VoIP user they just have to pick up that CO Line. My other option, is to just create a incoming call menu system on the VoIP / SLT port on the asterisk. Then STS user will just have to dial the SLT port extension then will be prompted to dial the VoIP extension they want (200,201, etc) ....'
I started playing with the later and have something like this thus far for a test:
exten => 40,1,Answer
exten => 40,2,Background(this-call-may-be)
exten => 40,3,Background(recorded)
exten => 40,4,Background(for-quality-purposes)
exten => 40,5,Goto(submenu,s,1)
exten => 1,1,Goto(default,voip1,1)
This is just a test so I dial 40 from VoIP then if I hit 1 it will send me to voip1 user. Still needs some work as it's not working right.
Let me know if you can add anything to any of this and hope it helps.
-Phil
