BT HH 2.0B and VoIP

[jimjones]

I've got a HH2.0B that has been hacked and I want to set up the hub phone with a new sip provider. Need to know the steps to enter my providers details and the usual stuff

SIP_URI = nnnnnnn(this is your 7 digit SIP ID)
[username] = nnnnnnn@sipgate.co.uk(replace nnnnnnn with your SIP ID)
[password] = ********(enter your password NOT the website password)
[password] = ********
[display name] = (whatever you want as your name)
voiceport = (I use COMMON, you can select using up/down arrows)
[abbr] = (nothing is needed here, not sure what it is)

Like I did in the HH2.0A

P.S. I’m not that knowledgeable with this stuff so step by step would be better. Thanks in advance.

[Technodevotee]

These are the instructions for configuring a SIP connection on a 2.0B which were posted by Psidoc.

Quote:

Sip Based Voip service On BTHUB 2B
The SIP service on BTHUB 2B can be activated and used without activating BBTalk service from BT. The HUB must be unlocked with CLI access to it. We need following information regarding the VOIP service provider

1. Username: and Auth Username (if different from username)
2. Password:
3. Registration Server: and port
4. Outgoing Proxy Address: and port

For simplicity, I provide here example of my service provider "sip2sip.info"

1. username & auth name = "test_user"
2. Password = "test_pass"
3. Registration Serverort = sip2sip.info:5060
4. Outgoing proxyort = proxy.sipthor.net:5060

NOTE: If your service provider does not provide outbound proxy, you can safely use registration servers address in place of outbound proxy.

Now the actual configuration starts. Please, note that we are going to configure a SIP TRUNK to the service provider. In an Unlocked BTHUB 2B there is a default TRUNK already configured and indexed at voip/trunk/0. This is used by the LANDLINE port.

We will be adding a new trunk index at voip/trunk/1 and of type "sip"
Here are the steps:

SSH into the BTHUB and get full CLI mode ready.

BT Home Hub 2.0B>

Enter the following CLI sequence..

>conf set voip/trunk/1/name "any name you like"

>conf set voip/trunk/1/type "sip"

>conf set voip/trunk/1/sip/dtmf_mode "rfc2833_negotiated"

>conf set voip/trunk/1/sip/username "test_user" <--- change this as per your requirement

>conf set voip/trunk/1/sip/auth_name "test_user <--- change this as per your requirement

>conf set_obscure voip/trunk/1/sip/auth_password "password" <--- change accordingly & NOTE the set_obscure

>conf set voip/trunk/1/sip/proxy/address "sip2sip.info" <--- change accordingly

>conf set voip/trunk/1/sip/proxy/port 5060 <--- change accordingly 

>conf set voip/trunk/1/sip/proxy/register_with_proxy 1 

>conf set voip/trunk/1/sip/outbound_proxy/address "proxy.sipthor.net" <--- change accordingly 

>conf set voip/trunk/1/sip/outbound_proxy/port 5060 <--- change accordingly 

>conf set voip/trunk/1/sip/outbound_proxy/enabled 1

>conf set voip/trunk/1/enabled 1

>conf reconf 1

That's it. The HUB, by this time may already have registered to your SIP provider (Hopefully).

Now, access the Web-GUI. You will get BT Broadband Talk status from there (connected Or Not).Goto "setting" -> "telephones" and change the setting for any incoming call on this newly created Voip line".
If you are using a HUBPHONE, you will get first dialtone from SIP provider. The default dialplan allows you to access landline by prefixing a '5' to any number.

All configuration were done as per the OPENRG Manual provided by PsiDoc. Emails, queries are welcome.

end quote:

I found that by adding the following:

conf set voip/trunk/1/ring_extension/0/id 0
conf set voip/trunk/1/ring_extension/1/id 1

Inbound SIP calls ring all phones and there is no need to perform the last steps of the procedure.

Additional SIP connections can be made by incrementing the trunk ID. It is also possible to add dialplan rules to enable selection of the connection by dialling a prefix. If anyone would like help on this just ask.

note to admins:
This procedure could do with being made a sticky.

technodevotee

[jimjones]

Thank you so much

[NobbyJones]

Hi all,

Thought Id almost got this working but cant receive incoming calls... The HH2b is registering with Sipgate and I can make calls but the HH is giving Sipgate its local IP address on the LAN, not my WAN IP. Any ideas?

I read on another post that the HH IP must be in a different range to my router, can someone confirm this? My router has a local IP of 192.168.1.1 and assigns DHCP address in the range of 192.168.1.2 - 192.168.1.254. What should I have the HH's IP as and what should the client DHCP range be set to?

Thanks.

[jimjones]

@ technodevotee
I would be interested in adding a second connection on my HH2.0B, can you tell me the steps for "adding dial plan rules by dialling a prefix"
I would like the original connection, that I made using your steps, to not have a prefix and a second connection to have a prefix of 1 or 6.
Is this possible or do both have to have a prefix?
I would be interested to know if this can be done on the HH2.0A also as I already have one connection on it and would like a second.
Thanks for your help.

[Technodevotee]

If you look at your config under VOIP/dialplan you should see a bunch of blocks of code entries something like this:

      (1
        (pattern(5))
        (trunk_index(0))
        (fallback_trunk_index(-1))
        (remove_digits(1))
        (num_digits_to_remove(1))
        (min_digits(2))
        (max_digits(31))
        (call_type(0))
        (narrowband(0))
      )

I have highlighted this one because it is the one that is used to select the PSTN line.

For mine, I added new code blocks (each one representing a rule in the dial plan) for each of my trunks. In my case, the highest code block number was 15 and the SIP trunk indexes were 1 & 2 so I incremened the number of the code block, pattern (prefix number) and trunk index for each SIP trunk like this:

 
conf set voip/dial_plan/16/pattern 6
conf set voip/dial_plan/16/trunk_index 1
conf set voip/dial_plan/16/fallback_trunk_index -1
conf set voip/dial_plan/16/remove_digits 1
conf set voip/dial_plan/16/num_digits_to_remove 1
conf set voip/dial_plan/16/min_digits 3
conf set voip/dial_plan/16/max_digits 31
conf set voip/dial_plan/16/call_type 0
conf set voip/dial_plan/16/narrowband 0
conf set voip/dial_plan/17/pattern 7
conf set voip/dial_plan/17/trunk_index 2
conf set voip/dial_plan/17/fallback_trunk_index -1
conf set voip/dial_plan/17/remove_digits 1
conf set voip/dial_plan/17/num_digits_to_remove 1
conf set voip/dial_plan/17/min_digits 3
conf set voip/dial_plan/17/max_digits 31
conf set voip/dial_plan/17/call_type 0
conf set voip/dial_plan/17/narrowband 0

This allows me to select my outbound connection by dialing a prefix, i.e. 5 for PSTN, 6 for the first SIP trunk and 7 for for the second SIP trunk.

Obviously, I don't know what numbers are used in other Hubs but this works fine for me.

I hope I made this clear enough.

technodevotee

[jimjones]

@ technodevotee
Thanks, I will try this on my 2.0B. I am also trying to add one on my 2.0A but am stuck on what to add to the dialplan part
http://www.psidoc.com/showthread.php...=3286#post3286
you may not be as fimilar with the 2.0A hub

[dzidek23]

Hi,

unfortunately I'll have to refresh this topic... With a lot of help from Technodevotee I did manage to force HH2b (ADSL) to connect over ETH port 4... everything seams to be ok but,

I have two SIP providers and two trunks (one handset thou), both outgoing calls are ok but on one, incoming calls are rejected with "this call can not be answered" (error 404)

Web-GUI shows two handsets registered (but there should be only one)

I've populated Technodevotee code:

conf set voip/trunk/1/ring_extension/0/id 0
conf set voip/trunk/1/ring_extension/1/id 1

conf set voip/trunk/2/ring_extension/0/id 0
conf set voip/trunk/2/ring_extension/1/id 1

and it still does not work..

any suggestions are really really welcome

dzidek23