Tribox and 2talk - Incoming Call Problems (newbie)

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sfrasernz

227 posts

Master Geek

#33668 9-May-2009 09:54

I am new to the voip game but I've spent a lot time reading and have got things almost working. I have a Trixbox PBX configured and working internally. I can call ext to ext and have an inbound route configured routing all calls to a ring group. I assume this is working because I can dial 7777 and the extensions all ring.

I have recently signed up for a free 2Talk account and have been allocated an 028 number. I'm trying to get inbound calling working but everytime I call the 028 number from my vodafone mobile it goes straight to the 2talk voicemail. There isn't any credit on the account but i think they still allow calls to be received.

I have set up a SIP Trunk with Registration within Trixbox and it looks like it is connecting. Here are is the status being reported:

SIP REGISTRY
--------------
2talk.co.nz:5060 0282XXXXXXX 105 Registered

SIP PEERS
------------
2talkFree/028XXXXXXX       202.180.76.166              5060     Unmonitored
210/210                    10.1.1.4         D   N   A 40440    OK (171 ms)
201                        (Unspecified)    D   N   A 0        UNKNOWN
200/200                    10.1.1.3         D   N   A 21822    OK (19 ms)
028XXXXXXXX                202.180.76.166              5060     Unmonitored
5 sip peers [Monitored: 2 online, 1 offline Unmonitored: 2 online, 0 offline]

Is there anything I'm missing?? Is there further config I need to do with 2talk? Do I need to tell them I'm using Trixbox?

I am working behind a NAT firewall but it is an outbound connection so I don't think that is the problem.

For completeness here are my Trunk settings:

PEER DETAILS:
fromuser=028XXXXXXX
host=2talk.co.nz
insecure=invite,port
qualify=no
secret=password
type=peer
username=028XXXXXXX

USER Conext: 028XXXXXXX

USER DETAILS:
host=2talk.co.nz
secret=password
type=peer
context=from-trunk

Any help would be greatly appreciated! Thanks.

sfrasernz

227 posts

Master Geek

#213690 9-May-2009 16:47

UPDATE

So I've managed to get outbound calling to work. Very pleased.

The final (hopefully) problem is inbound calling. The call successfully connects to my PBX but an automated message is played saying "the number you have dialed is not in service. Please check the number and try again". The inbound routing should send it to ring group 600 and ring all extensions.

I think this has something to do with Contexts but am unsure where to look next. I can see from the call log the call is connected and sent to Destination S. Not sure what that means either.

sbiddle

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#213694 9-May-2009 17:08

What are you using for your inbound route?

kiwiscoota

237 posts

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#213737 9-May-2009 20:16

sfrasernz: UPDATE
The final (hopefully) problem is inbound calling.......

If you need someone to call you multiple times for testing purposes, I'm more than happy to assist. I've messaged privately with my contact details.

sfrasernz

227 posts

Master Geek

#213738 9-May-2009 20:26

Thanks for the replies.

I have two inbound routes....

The first is pretty much all the defaults with a destination set. I thought this would route any incoming call to the programmed destination.

DID Number: blank
Caller ID Number: blank
Source: None
Destination: Extensions 200

The other is:

DID Number: _028XXXXXXXX I've also tried 028XXXXX
Caller ID Number: blank
Source: None
Destination: Extensions 200

Do I have these setup right?

sbiddle

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#213744 9-May-2009 20:46

Are you sure calls are hitting your Asterisk box?

sfrasernz

227 posts

Master Geek

#213807 10-May-2009 09:27

I'm pretty certain they are. If I look at the 2talk site I can see calls of 8-15 seconds being answered which would roughly equate to the duration of the "the number you have dialed is not in service" message.

Within the Trixbox call log I can see calls coming in and are being answered. An example is:

2009-05-09 17:30:56 SIP/028255... 0282XXXXXXX "Joe Bloggs" s ANSWERED 00:08

The Dst says "s" which is apparantly some default call route related to "contexts" but I'm still trying to understand how this works.0

The context setup in my SIP Trunk is context=from-trunk. The context for my extensions are from-internal. I've tried setting them the same by still no go.

sbiddle

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#213808 10-May-2009 09:36

sfrasernz: I'm pretty certain they are. If I look at the 2talk site I can see calls of 8-15 seconds being answered which would roughly equate to the duration of the "the number you have dialed is not in service" message.

Within the Trixbox call log I can see calls coming in and are being answered. An example is:

2009-05-09 17:30:56 SIP/028255... 0282XXXXXXX "Joe Bloggs" s ANSWERED 00:08

The Dst says "s" which is apparantly some default call route related to "contexts" but I'm still trying to understand how this works.0

The context setup in my SIP Trunk is context=from-trunk. The context for my extensions are from-internal. I've tried setting them the same by still no go.

The contexts need to be different. Think of contexts as a security level - giving access to certain things.

from-trunk is the default for inbound calls from external trunks. from-internal is for local extensions.

My 2talk trunk settings are very different and I only have my settings in peer settings because there is really no need for both.

One obvious ommission is any form of codec settings, try putting

disallow=all
allow=ulaw

in your peer settings.

Is the voice you are hearing Alison (who is the "voice" of Asterisk) or a Callplus message?

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sfrasernz

227 posts

Master Geek

#213810 10-May-2009 09:56

I'm pretty sure it's Alison. It is a female voice. The times that I haven't been able to get through because the SIP trunk was down I heard a male voice which must be 2talk.co.nz.

I deleted all the USER settings under the SIP trunk and modified my peer settings to:

username=0282XXXXXXX
fromuser=0282XXXXXXX
secret=password
host=2talk.co.nz
insecure=very
qualify=no
nat=yes
type=peer
disallow=all
allow=ulaw

How do these compare to yours?

I have another laptop with a default install so I might go through the configuration process again and see if I have any better luck.

sfrasernz

227 posts

Master Geek

#213832 10-May-2009 11:54

My other Trixbox is doing exactly the same thing.

I fired up the CLI and watched an incoming call. Here is the log

Executing [<my_number>@from-sip-external:1] NoOp("SIP/<my_number>-09e3fb88", "Received incoming SIP connection from unknown peer to <my_number>") in new stack
Executing [<my_number>@from-sip-external:2] Set("SIP/<my_number>-09e3fb88", "DID=<my_number>") in new stack
Executing [<my_number>@from-sip-external:3] Goto("SIP/<my_number>-09e3fb88", "s|1") in new stack
Goto (from-sip-external,s,1)
Executing [s@from-sip-external:1] GotoIf("SIP/<my_number>-09e3fb88", "0?from-trunk|<my_number>|1") in new stack
Executing [s@from-sip-external:2] Set("SIP/<my_number>-09e3fb88", "TIMEOUT(absolute)=15") in new stack
Channel will hangup at 2009-05-09 23:45:01 UTC.
Executing [s@from-sip-external:3] Answer("SIP/<my_number>-09e3fb88", "") in new stack
Executing [s@from-sip-external:4] Wait("SIP/<my_number>-09e3fb88", "2") in new stack
Executing [s@from-sip-external:5] Playback("SIP/<my_number>-09e3fb88", "ss-noservice") in new stack
<SIP/<my_number>-09e3fb88> Playing 'ss-noservice' (language 'en')

sbiddle

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#213833 10-May-2009 12:05

That's clearly showing the calls are hitting trixbox but not matching an inbound route so are being rejected which plays the ss-noservice audio that you are hearing.

Create an inbound route with your 028 number as the DID.

sfrasernz

227 posts

Master Geek

#213853 10-May-2009 13:41

I tried setting an inbound route with the DID set to my number and a ring group as a destination but still have the same problem.

Under my SIP trunk settings there is a section titled INCOMING SETTINGS specifying User Context and User Settings. Do I need to complete this section?

sfrasernz

227 posts

Master Geek

#213860 10-May-2009 14:21

UPDATE...I have enabled "anonymous SIP connections" on the general settings and calls are now coming in as they should. I guess I need to figure out why the connection is anonymous...must be something with 2talk.co.nz.

I appreciate your help sbiddle.

A little off topic but how do you find your VFX connection? I'll be dropping Telecom when we move house in a few months and am looking for a VoIP and Naked DSL provider.

sbiddle

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#213869 10-May-2009 15:00

Under no circumstances leave the allow anonymous connection unticked unless you really know what it means (and no disrespect to you but you probably don't!).

This leaves your box wide open to any inbound SIP traffic including anonymous URI calling and poses a large number of potential security risks.

Looking at your post above you mention

Executing [@from-sip-external:1] NoOp("SIP/-09e3fb88", "Received incoming SIP connection from unknown peer to ") in new stack

Is your DDI in the inbound route set to this same number in the exact same format? Did you remove the initial generic route when you created a new route with the DDI set?

VFX works fine for me and I have no issues with the service. 2talk have plans including airtime which makes it an appealing alternative for many people.

sfrasernz

227 posts

Master Geek

#213889 10-May-2009 18:12

Yeah the inbound route with the DID number is the only route that I have configured. The number format is exactly what I'm seeing in the logs. I'm really not sure what to try next.

sfrasernz

227 posts

Master Geek

#214000 11-May-2009 08:11

RESOLVED

Turns out the problem was with my PEER settings. After adding the fromdomain= everything began to work as it should.

Thanks for your assistance.