asterisk users - May 2005 - Asterisk@home 1.0 + Sipgate UK/SIP Provider

Hello, I've been looking at the DialPlans by some poeple using
Asterisk with SipGate, but the new Asterisk@home 1.0 allows you to
create Outbound routes etc, does using the web admin give the same
effects?

When I add a SIP Trunk with my Sipgate settings and use a pattern of
"8|." to place all calls with a 8 prefix tot he sipgate account the
softphones dial the number, the Asterisk console returns

-- Executing Dial("SIP/201-fcb3", "SIP/sipgate/###########")
in new stack
-- Called sipgate/##########

But the call is never made, and no errors reported.

I am behind a router (ipcop) but I would have thought I dont need to
set any ports as its outgoing, and there is no outgoing blocks on the
router.

Edit SIP Trunk
----------------------
Outbound Caller ID: <my sip number>
Dial Rules: 8|.
Trunk Name: sipgate
PEER Details
host=217.10.79.219
secret=******
type=peer
username=####### <Sipgate username>

Edit Route
---------------
Dial Patterns: 8|.
Trunk Sequence: SIP/sipgate

Thanks for your time.

Hello, I've been looking at the DialPlans by some poeple using
Asterisk with SipGate, but the new Asterisk@home 1.0 allows you to
create Outbound routes etc, does using the web admin give the same
effects?

When I add a SIP Trunk with my Sipgate settings and use a pattern of
"8|." to place all calls with a 8 prefix tot he sipgate account the
softphones dial the number, the Asterisk console returns

-- Executing Dial("SIP/201-fcb3", "SIP/sipgate/###########")
in new stack
-- Called sipgate/##########

But the call is never made, and no errors reported.

I am behind a router (ipcop) but I would have thought I dont need to
set any ports as its outgoing, and there is no outgoing blocks on the
router.

Edit SIP Trunk
----------------------
Outbound Caller ID: <my sip number>
Dial Rules: 8|.
Trunk Name: sipgate
PEER Details
host=3D217.10.79.219
secret=3D******
type=3Dpeer
username=3D####### <Sipgate username>

Edit Route
---------------
Dial Patterns: 8|.
Trunk Sequence: SIP/sipgate

Thanks for your time.

Hello, I've been looking at the DialPlans by some poeple using
Asterisk with SipGate, but the new Asterisk@home 1.0 allows you to
create Outbound routes etc, does using the web admin give the same
effects?

When I add a SIP Trunk with my Sipgate settings and use a pattern of
"8|." to place all calls with a 8 prefix tot he sipgate account the
softphones dial the number, the Asterisk console returns

-- Executing Dial("SIP/201-fcb3", "SIP/sipgate/###########")
in new stack
-- Called sipgate/##########

But the call is never made, and no errors reported.

I am behind a router (ipcop) but I would have thought I dont need to
set any ports as its outgoing, and there is no outgoing blocks on the
router.

Edit SIP Trunk
----------------------
Outbound Caller ID: <my sip number>
Dial Rules: 8|.
Trunk Name: sipgate
PEER Details
host=217.10.79.219
secret=******
type=peer
username=####### <Sipgate username>

Edit Route
---------------
Dial Patterns: 8|.
Trunk Sequence: SIP/sipgate

Thanks for your time.

I am using Sipgate with *@Home and this is how I have set mine up to
have it working perfectly. Using the AMP Interface my trunk is setup as
follows......

Under Trunk:
Outbound caller ID is your full sip number including area code.

Peer Detail:
allow=ulaw
authuser=539xxxx (your sip number)
canreinvite=no
disallow=all
dtmfmode=info
fromdomain=sipgate.co.uk
fromuser=539xxxx (your sip number)
host=sipgate.co.uk
insecure=very
nat=yes
secret=XXXXXXX (your sip password)
type=peer
username=539xxxx (your sip number)

User Details:
allow=ulaw
authuser=539xxxx (your sip number)
context=ext-did
disallow=all
dtmfmode=info
faxdetect=incoming
fromdomain=sipgate.co.uk
fromuser=539xxxx (your sip number)
host=sipgate.co.uk
insecure=very
secret=XXXXXXX (your sip password)
username=539xxxx (your sip number)

User Context:
Mine is ext-did

Register String:
539xxxx:XXXXXXX@sipgate.co.uk/539xxxx


Hope this helps...
Mark





-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Paul
Goodyear
Sent: 16 May 2005 11:15
To: Asterisk-Users@lists.digium.com
Subject: [Asterisk-Users] Asterisk@home 1.0 + Sipgate UK/SIP Provider

Hello, I've been looking at the DialPlans by some poeple using
Asterisk with SipGate, but the new Asterisk@home 1.0 allows you to
create Outbound routes etc, does using the web admin give the same
effects?

When I add a SIP Trunk with my Sipgate settings and use a pattern of
"8|." to place all calls with a 8 prefix tot he sipgate account the
softphones dial the number, the Asterisk console returns

-- Executing Dial("SIP/201-fcb3", "SIP/sipgate/###########")
in new
stack
-- Called sipgate/##########

But the call is never made, and no errors reported.

I am behind a router (ipcop) but I would have thought I dont need to
set any ports as its outgoing, and there is no outgoing blocks on the
router.

Edit SIP Trunk
----------------------
Outbound Caller ID: <my sip number>
Dial Rules: 8|.
Trunk Name: sipgate
PEER Details
host=217.10.79.219
secret=******
type=peer
username=####### <Sipgate username>

Edit Route
---------------
Dial Patterns: 8|.
Trunk Sequence: SIP/sipgate

Thanks for your time.
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> -- Executing Dial("SIP/201-fcb3",
"SIP/sipgate/###########") in new stack
> -- Called sipgate/##########
> 
Paul  I apreciate why you've #### the dialled digits out there, but
would you be good enough to include the first few, as if your asterisk
box is sending extra / unwanted / too few digits to sipgate its never
going to work :)

Other than that it seems someone else has posted config for your
reference to check.

David