What’s this?

A little back story and terse summary of the settings I am using to enable VoIP (Voice over IP) telephony from my existing Obihai OBi110 bridge unit, so my analogue phones still work without an analogue line any more..

See also: my earlier post on using A&A for a static IP separated from my ISP

TL;DR? config below

The Obihai OBi110

PC-Magazine wrote a review back in 2012 when these were a shiny new toy explaining what they are for: bridging telephone technologies together.

My Current Use

I bought the OBi110 to use as a super configurable call screening device, with a built-in auto-assistant to inform callers that they are being screened and to press a specific DTMF key, or be diverted to a voicemail system on my local server (this bit never happened!). The call screening has been in place for several years, working well, and allowing me to block many nusiance calls.

The Plan

It’s simple - as the OBi110 supports VoIP via the SIP standard, and A&A can cost-effectively port my existing BT landline number to their SIP VoIP service, I just have to figure out the configuration required to make it work. Behind a Network Address Translation (NAT) router. When nobody has ever done this before with an OBi110 (at least not written it up!). Without b0rking my very well known home phone number that I have had for almost 35 years.

The detailed plan to protect my number!

  • Buy a temporary test number (they’re super cheap!).
  • Use that temporary number to get the OBi110 working with A&A SIP.
  • ..or.. if that fails: buy a known-good SIP to Analogue Telephone Adapter (ATA). Get that working.
  • Once something is working, port my real number in and switch over to it.
  • Release the temporary number.

The side effects

Asking A&A to port my existing number in will have dramtic side effects on my landline services: as this is the only number, BT will assume I no longer want their telephony and issue a PSTN cease on my line, which will also go to my old broadband provider (Plusnet), who will immediately cease their service too. A non-reversible and final cessation of my landline and old broadband.

The Config

By section, top-to-bottom in the side menu. NB: only shows non-default settings

System Management, Network Settings (these are specific to my internal network - adjust for yours!):

Parameter Value
AddressingType Static
LocalTimeZone GMT+0

System Management, Auto Provisioning:

Parameter Value
ITSP Provisioning/Method Disabled

System Management, Device Admin (I collect syslog on my server, you may not want to):

Parameter Value
Syslog/Level 7

Service Providers, ITSP Profile A, General:

Parameter Value
Name Andrews & Arnold
STUNEnable Yes
STUNServer stun.aa.net.uk

Service Providers, ITSP Profile A, SIP:

Parameter Value
ProxyServer voiceless.aa.net.uk
RegistrarServer voiceless.aa.net.uk
UserAgentDomain voiceless.aa.net.uk

NB: Setting a proxy server is against A&A recommendation but OBi110 considers a service disabled without it, it provides a default if RegistrarServer is not set.

Voice Services, SP1 Service (as above, I collect detailed syslogs, you may not want to):

Parameter Value
X_RingProfile B
X_SipDebugOption Log All Except REGISTER Messages
AuthUserName <allocated number>

NB: Ring profile B is for UK ringing tones, which are less weird for us.

Voice Services, OBITALK Service:

Parameter Value
Enable No

Voice Services, Auto Attendant:

Parameter Value
Enable No

Physical Interfaces, PHONE Port:

Parameter Value
OutboundCallRoute {([1-9]x?*(Mpli)):pp},{(<#:>|999):pli},{**0:aa},{***:aa2},{(<**1:>(Msp1)):sp1},{(<**2:>(Msp2)):sp2},{(<**8:>(Mli)):li},{(<**9:>(Mpp)):pp},{(Mpli):pli}
PrimaryLine SP1
RingFrequency 25
CallerIDMethod FSK(V.23)
CallerIDTrigger After Polarity Reversal

NB: Outbound call route replaces 911 with 999 and routes to pli instead of li.
NBB: CallerID settings are for my UK, BT provided analogue phones. YMMV.

Physical Interfaces, LINE Port:

Parameter Value
Enable No

The Result

With the above settings, I can dial a standard number (no prefixes) and it calls out on the SP1 SIP service, connects and can send / receive audio. Incoming calls to the authenticated number are routed to the analogue phones, which correctly answer and can send / receive audio. Success!

NB: I can still force use of the PSTN line by using the **8 prefix in the outbound call route.