Originally Posted by
warez_kid
Couple of weeks ago i setup a new trixbox system running the latest open source release. I had a fair bit of trouble getting the incomming pennytel trunk to work. It would show up as connected(on trixbox and through pennytel support) but go to a pennytel message bank when i tried to ring the number.
Thanks for posting that.
If I use exactly the same settings (trixbox 2.6.2.1) I can't make an outgoing call via PennyTel ("all circuits are busy now"). trixbox is registered with PT (PT|diagnostics) and in PBX|PBX Status. If I change the PBX|Settings|Trunks|SIP/PennyTel PEER Details section so that the line:
host=dynamic
changes to:
host=sip.pennytel.com
then I can make outgoing calls.
Register string is:
6173121XXXX:ZZZZZ@sip.pennytel.com/6173121XXXX
PT diagnostics say:
Status : Online
Agent : Asterisk PBX
Contact : sip:6173121XXXX@220.233.XXX.YY:5076
However, if I dial my PT number from a PSTN service, I get "the number you have dialled is not in service".
The asterisk CLI console says:
Code:
-- Executing [6173121XXXX@from-sip-external:1] NoOp("SIP/6173121XXXX-08e540d0", "Received incoming SIP connection from unknown peer to 6173121XXXX") in new stack
-- Executing [6173121XXXX@from-sip-external:2] Set("SIP/6173121XXXX-08e540d0", "DID=6173121XXXX") in new stack
-- Executing [6173121XXXX@from-sip-external:3] Goto("SIP/6173121XXXX-08e540d0", "s|1") in new stack
-- Goto (from-sip-external,s,1)
-- Executing [s@from-sip-external:1] GotoIf("SIP/6173121XXXX-08e540d0", "0?from-trunk|6173121XXXX|1") in new stack
-- Executing [s@from-sip-external:2] Set("SIP/6173121XXXX-08e540d0", "TIMEOUT(absolute)=15") in new stack
-- Channel will hangup at 2009-01-12 07:28:32 UTC.
-- Executing [s@from-sip-external:3] Answer("SIP/6173121XXXX-08e540d0", "") in new stack
-- Executing [s@from-sip-external:4] Wait("SIP/6173121XXXX-08e540d0", "2") in new stack
-- Executing [s@from-sip-external:5] Playback("SIP/6173121XXXX-08e540d0", "ss-noservice") in new stack
-- <SIP/6173121XXXX-08e540d0> Playing 'ss-noservice' (language 'en')
== Spawn extension (from-sip-external, s, 5) exited non-zero on 'SIP/6173121XXXX-08e540d0'
-- Executing [h@from-sip-external:1] NoOp("SIP/6173121XXXX-08e540d0", "Hangup") in new stack
-- Executing [h@from-sip-external:2] Set("SIP/6173121XXXX-08e540d0", "DID=s") in new stack
-- Executing [h@from-sip-external:3] Goto("SIP/6173121XXXX-08e540d0", "s|1") in new stack
-- Goto (from-sip-external,s,1)
-- Executing [s@from-sip-external:1] GotoIf("SIP/6173121XXXX-08e540d0", "0?from-trunk|s|1") in new stack
-- Executing [s@from-sip-external:2] Set("SIP/6173121XXXX-08e540d0", "TIMEOUT(absolute)=15") in new stack
-- Channel will hangup at 2009-01-12 07:28:38 UTC.
-- Executing [s@from-sip-external:3] Answer("SIP/6173121XXXX-08e540d0", "") in new stack
== Spawn extension (from-sip-external, s, 3) exited non-zero on 'SIP/6173121XXXX-08e540d0'
So we know the call is reaching the trixbox.
For Trunk Incoming USER Context I have 6173121XXXX.
If I change PBX|Settings|Trunks|SIP/PennyTel USER Details section from
context=from-sip
to:
context=from-trunk
then it doesn't seem to make any difference. Sometimes I get busy tone and nothing appears in the CLI.
I have one inbound route I've called "From PT" with the destingation set to extension 200 (the registered softphone). The extension details say it is "Used as Destination by 1 Object:" and when hovered over it says "Inbound Route: From PT (/)".
The extension device options are:
Code:
This device uses sip technology.
secret xxxx
dtmfmode rfc2833
canreinvite no
context from-internal
host dynamic
type friend
nat yes
port 5060
qualify yes
callgroup
pickupgroup
disallow
allow
dial SIP/200
accountcode
mailbox 200@default
Language
I'm using a Billion 7404VGP router, with its VoIP ports registered to VoXaLot (my current method of using VoIP). VoXaLot is not registered to PT.
For my testing of trixbox, I have SJPhone. It's registered to the trixbox, can make external calls, receive calls from another extension, dial the echo test and speaking clock, etc.
No ports in the Billion have been forwarded for VoIP purposes.
If I open DMZ to the trixbox, incoming still does not work.
What have I missed?
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