Sipgate with Trixbox

Has anyone here got Sipgate working with Trixbox?

I am banging my head against a wall here!

If I call my Sipgate number form my normal phone, I am getting the message
(From my trixbox) "The number you have dialled is not in service, please
check the number and try again" beep beep beep etc...

From a fresh install I have done the following...

I was previously on the last build of Asterisk@home without problem

---------

In sip.conf, added the following...
externip=MY IP ADDRESS
outside_addr=MY IP ADDRESS

---------

Added my extensions, 200 and 201 - these can call each other.

---------

Added a new trunk with the following details (Copied from my working
Asterisk@home setup)

Outbound Caller ID - MY SIPGATE NUMBER
Max Channels - 1
Dial Rules - 0044+XXXXXXXXXX (Plus a load more other rules)
Trunk Name - Sipgate
PEER Details - allow=ulaw&alaw
authuser=SIPGATE ID
canreinvite=no
context=ext-did
disallow=all
dtmfmode=info
fromdomain=sipgate.co.uk
fromuser=SIPGATE ID
host=sipgate.co.uk
insecure=very
qualify=yes
secret=SIPGATE PASSWORD
type=peer
username=SIPGATE ID

Register String - SIPGATE ID:PASSWORD@sipgate.co.uk/SIPGATE ID

--------------------

Inbound Routes

DID Number - SIPGATE ID

Destination - Office <200>

-------------------


Any ideas what else I need to do!

Here is what the terminal spews out when I call my sipgate PSTN number from
a normal phone

(I have change my sipgate ID to MY SIPGATE ID)


-- Executing NoOp("SIP/gw02.uk.sipgate.net-09b0c6f8", "Received incoming
SIP connection from unknown peer to MY SIPGATE ID") in new stack
-- Executing Set("SIP/gw02.uk.sipgate.net-09b0c6f8", "DID=MY SIPGATE
ID") in new stack
-- Executing Goto("SIP/gw02.uk.sipgate.net-09b0c6f8", "s|1") in new
stack
-- Goto (from-sip-external,s,1)
-- Executing GotoIf("SIP/gw02.uk.sipgate.net-09b0c6f8", "0?from-trunk|MY
SIPGATE ID|1") in new stack
-- Executing Set("SIP/gw02.uk.sipgate.net-09b0c6f8",
"TIMEOUT(absolute)=15") in new stack
-- Channel will hangup at 2006-08-04 14:31:06 UTC.
-- Executing Answer("SIP/gw02.uk.sipgate.net-09b0c6f8", "") in new stack
-- Executing Wait("SIP/gw02.uk.sipgate.net-09b0c6f8", "2") in new stack
-- Executing Playback("SIP/gw02.uk.sipgate.net-09b0c6f8",
"ss-noservice") in new stack
-- Playing 'ss-noservice' (language 'en')
-- Executing Congestion("SIP/gw02.uk.sipgate.net-09b0c6f8", "") in new
stack

So the call is getting to asterisk, but it is refusing to route it!

Thanks!!

Sparks...

"Sparks" <postmaster@127.0.0.1> wrote in message
news:44d35b63$0$638$bed64819@news.gradwell.net...
> Has anyone here got Sipgate working with Trixbox?

....I have now :-)

I had the go into the general settings page, and set the
"Allow Anonymous Inbound SIP Calls" to YES

Next problem...

I can dial in now, but can't seem to dial out!
It tells me the following...
(Dialed number changed to 888888)


-- Executing Macro("SIP/201-43c3", "dialout-trunk|2|888888||") in new
stack
-- Executing GotoIf("SIP/201-43c3", "1?3:2") in new stack
-- Goto (macro-dialout-trunk,s,3)
-- Executing Macro("SIP/201-43c3", "user-callerid") in new stack
-- Executing GotoIf("SIP/201-43c3", "0?report") in new stack
-- Executing GotoIf("SIP/201-43c3", "0?start") in new stack
-- Executing Set("SIP/201-43c3", "REALCALLERIDNUM=201") in new stack
-- Executing NoOp("SIP/201-43c3", "REALCALLERIDNUM is 201") in new stack
-- Executing Set("SIP/201-43c3", "AMPUSER=201") in new stack
-- Executing Set("SIP/201-43c3", "AMPUSERCIDNAME=Office") in new stack
-- Executing GotoIf("SIP/201-43c3", "0?report") in new stack
-- Executing Set("SIP/201-43c3", "CALLERID(all)=Office <201>") in new
stack
-- Executing NoOp("SIP/201-43c3", "Using CallerID "Office" <201>") in
new stack
-- Executing Macro("SIP/201-43c3", "record-enable|201|OUT") in new stack
-- Executing GotoIf("SIP/201-43c3", "0 > 0?2:4") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing AGI("SIP/201-43c3",
"recordingcheck|20060804-172904|1154708944.4") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20060804-172904|1154708944.4: Outbound recording not
enabled
-- AGI Script recordingcheck completed, returning 0
-- Executing NoOp("SIP/201-43c3", "No recording needed") in new stack
-- Executing Macro("SIP/201-43c3", "outbound-callerid|2") in new stack
-- Executing GotoIf("SIP/201-43c3", "1?start") in new stack
-- Goto (macro-outbound-callerid,s,3)
-- Executing NoOp("SIP/201-43c3", "REALCALLERIDNUM is 201") in new stack
-- Executing Set("SIP/201-43c3", "USEROUTCID=") in new stack
-- Executing Set("SIP/201-43c3", "EMERGENCYCID=") in new stack
-- Executing Set("SIP/201-43c3", "TRUNKOUTCID=020777777") in new stack
-- Executing GotoIf("SIP/201-43c3", "1?trunkcid") in new stack
-- Goto (macro-outbound-callerid,s,11)
-- Executing GotoIf("SIP/201-43c3", "0?usercid") in new stack
-- Executing Set("SIP/201-43c3", "CALLERID(all)=020777777") in new stack
-- Executing GotoIf("SIP/201-43c3", "1?report") in new stack
-- Goto (macro-outbound-callerid,s,15)
-- Executing NoOp("SIP/201-43c3", "CallerID set to "" <020777777>") in
new stack
-- Executing Set("SIP/201-43c3", "GROUP()=OUT_2") in new stack
-- Executing GotoIf("SIP/201-43c3", "0?108") in new stack
-- Executing Set("SIP/201-43c3", "DIAL_NUMBER=888888") in new stack
-- Executing Set("SIP/201-43c3", "DIAL_TRUNK=2") in new stack
-- Executing AGI("SIP/201-43c3", "fixlocalprefix") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
fixlocalprefix: Added prefix. New number: 00442088888888
-- AGI Script fixlocalprefix completed, returning 0
-- Executing Set("SIP/201-43c3", "OUTNUM=00442088888888") in new stack
-- Executing Set("SIP/201-43c3", "custom=SIP/Sipgate1") in new stack
-- Executing GotoIf("SIP/201-43c3", "0?16") in new stack
-- Executing Dial("SIP/201-43c3", "SIP/Sipgate1/00442088888888|120|r")
in new stack
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing Goto("SIP/201-43c3", "s-CHANUNAVAIL|1") in new stack
-- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
-- Executing NoOp("SIP/201-43c3", "Dial failed due to CHANUNAVAIL") in
new stack
-- Executing Macro("SIP/201-43c3", "outisbusy|") in new stack
-- Executing Playback("SIP/201-43c3", "all-circuits-busy-now") in new
stack
-- Playing 'all-circuits-busy-now' (language 'en')

Sparks wrote:
> Has anyone here got Sipgate working with Trixbox?
>
> I am banging my head against a wall here!
>
> If I call my Sipgate number form my normal phone, I am getting the message
> (From my trixbox) "The number you have dialled is not in service, please
> check the number and try again" beep beep beep etc...
>
> From a fresh install I have done the following...
>
> I was previously on the last build of Asterisk@home without problem
>
> ---------
>
> In sip.conf, added the following...
> externip=MY IP ADDRESS
> outside_addr=MY IP ADDRESS
>
> ---------
>
> Added my extensions, 200 and 201 - these can call each other.
>
> ---------
>
> Added a new trunk with the following details (Copied from my working
> Asterisk@home setup)
>
> Outbound Caller ID - MY SIPGATE NUMBER
> Max Channels - 1
> Dial Rules - 0044+XXXXXXXXXX (Plus a load more other rules)
> Trunk Name - Sipgate
> PEER Details - allow=ulaw&alaw
> authuser=SIPGATE ID
> canreinvite=no
> context=ext-did
> disallow=all
> dtmfmode=info
> fromdomain=sipgate.co.uk
> fromuser=SIPGATE ID
> host=sipgate.co.uk
> insecure=very
> qualify=yes
> secret=SIPGATE PASSWORD
> type=peer
> username=SIPGATE ID
>
> Register String - SIPGATE ID:PASSWORD@sipgate.co.uk/SIPGATE ID
>
> --------------------
>
> Inbound Routes
>
> DID Number - SIPGATE ID
>
> Destination - Office <200>
>
> -------------------
>
>
> Any ideas what else I need to do!

My trixbox setup looks a bit different from yours, also you have a
couple of things I don't. Note: I only use Sipgate for an incoming
number as I use other providers for outgoing calls.

- my setup looks like this (where USERNAME is your sipgate ID):
Max Channels - blank
trunk name - sipgate
Peer details:
authuser=USERNAME
canreinvite=no
context=from-trunk
dtmfmode=info
fromdomain=sipgate.co.uk
fromuser=USERNAME
host=sipgate.co.uk
insecure=very
secret=PASSWORD
type=friend
username=USERNAME

INCOMING
Usercontext: USERNAME
USER Details:
context=from-trunk
dtmfmode=info
insecure=very
secret=PASSWORD
type=peer

Reg String USERNAME:PASSWORD@sipgate.co.uk/USERNAME
---
Inbound route - as you have it
DID: USERNAME
Destination: 200 or whatever

"Sparks" <postmaster@127.0.0.1> wrote in message
news:44d35b63$0$638$bed64819@news.gradwell.net...
> Has anyone here got Sipgate working with Trixbox?
>


Yep. My sip_additional.conf looks like this:

[sipgate]
username=PASSWORD
type=friend
srvlookup=yes
secret=PASSWORD
qualify=no
qualify=yes
port=5060
nat=yes
insecure=very
host=sipgate.co.uk
fromuser=USERNAME
fromdomain=sipgate.co.uk
disable=all
context=from-trunk
bindaddr=0.0.0.0
allow=alaw
allow=ulaw
allow=g729
allow=gsm
allow=slinear

register=user:password@sipgate.co.uk/user

I just put everything (except the register string) in the first box (the
incoming box I think it was) - it doesn't matter. It still gets written to
the same sip_additional.conf.

remember to set "allow anonymous inbound SIP calls" in general, and set an
"ANY CID / ANY DID" inbound route.

"Carl Farrington" <carl@css--nomorespamplease--networks.commy.invalid> wrote
in message news:eb0efn$hn3$1$8300dec7@news.demon.co.uk...
>
> "Sparks" <postmaster@127.0.0.1> wrote in message
> news:44d35b63$0$638$bed64819@news.gradwell.net...
>> Has anyone here got Sipgate working with Trixbox?
>>
>
>
> Yep. My sip_additional.conf looks like this:
>
> [sipgate]
> username=PASSWORD

^^^ oops, that should of course be USERNAME. :blush: