UK based but guess that does not matter.
I have managed to get my hardphone to work with sipgate, incomming and
outgoing calls OK.
But when I try to switch on the second account or line within the phone to
talk to astlinux then Sipgate becomes incomming calls only. It could be
because I am using port 5060 for sip on both accounts not sure?
The end game is to have Astlinux register with sipgate then route to internal
phones, but, I get confused with all the examples out there, some refer to
trunks for incomming calls only!
I have a simple residential sipgate account, so I would assume that I make the
entry into sip.conf, I think I could use type=peer to match on the
registration to sipgate, is that correct or should it be friend as calls will
go both ways, this is one area I get confused.
I want to do the usual 9 to dial an outside line and direct these over
sipgate, but first steps are to get an incomming call to go to an internal
phone.
Anyway following an example from sipgate as below:
In sip.conf I have the following insertion
with bogus id and password
;--------------------------------------------
register=>1234567:[email protected]/1234567
[sipgate]
type=peer
secret=abcdefg
insecure=invite
username=1234567
defaultuser=1234567
fromuser=1234567
context=sipgate_in
fromdomain=sipgate.co.uk
qualify=yes
disallow=all
allow=alaw
dtmfmode=rfc2833
;---------------------------------------------------
in extensions.conf I have:
;----------------------------------------------------------------------
[sipgate_in]
exten => 1234567,1,Playback(hello-world)
;exten => 1234567,1,Dial(SIP/000b822d88d8)
same => n,Hangup()
;----------------------------------------------------------------------
The hope was that I would get "Hello World" but all that happens is like a
direct hangup.
The sipgate webpage says that there is a successful registration and just
shows calls to my account as not answered. When I check the status page of
astlinux there is no messages to show it is even hitting asterisk.
Could I have router problems?
I have not set any port forwards at this point in my router, or placed
astlinux in any DMZ.
I have also changed my hardphones so that each one operates on a different
local SIP port and RTP port, such as 5160, 5260, 5360 etc and the RTP's 5104,
5204, 5304 etc. Calls between hardphones still works OK.
Can anyone assist or point me in a direction?
Cheers
Adrian
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