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Obi110 RJ11 pinouts for line and phone.

Started by superpat, January 11, 2014, 07:45:07 AM

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superpat

Hi,

I have just bought a used OBi110 via ebay

I reside in the  UK and  am trying to install the OBi110 to work with Freepbx.

Does anyone have a DEFINITE pinout for the RJ11 sockets, line and phone, on the
OBI110, please?

Once I  know the pin outs I can purchase the correct cables!

Looking at RJ11 to RJ11 and BT to RJ11 cables, the possible permutations are daunting! It will be so much easier if I knew exactly what I require.

My analogue phone needs a RJ11 6P2C, wired with 3=a and 4=b (according to it's manual)  So I need the correct RJ11 to RJ11 cable, Obi110 phone port to phone.

Does the OBi110 provide the correct ring voltages?

many thanks

Patrick


ianobi

#1
Patrick - welcome to the forum.

The general principle for the RJ11 is that the two wire line ("tip & ring" or "a leg and b leg") always uses the middle two pins. This is normally pins 3 and 4. If there's a second line it will use pins 2 and 5. The OBi110 has only one phone line and one PSTN line so they both use pins 3 and 4 at the OBi end. It looks like your phone cable can be a straight RJ11 to RJ11 cable.

The British Telecom style 431A socket uses pins 2 and 5 for the two wire line. If your PSTN line is coming in from a British Telecom socket, then you need a connector or cable that connects pins 2 and 5 of the UK style 431A wall socket to pins 3 and 4 of the US style RJ11 plug that connects into the OBi. Adaptors and cables are easy to find on Amazon or Maplins.

Ringing voltages are no problem. However, by default the OBi110 is set up for North American defaults. As you bought it on Ebay, then I guess it was set up however the previous owner wanted it! It may be wise to do a factory reset first to get back to default settings. I suggest these changes first:

I suggest setting the line port impedance to 370+(620||310nF).

For the phone port impedance I would probably select 320+(1050||230nF) which looks closest to BT's nominal values.

Both Line and Phone ports should be set to FSK(V.23) method for CallerID.

Line Port RingDelay can be set to 0.

Have a look around the forum for UK dial plans / digit maps. Getting OBi set up to match the UK numbering formats will speed up dialling a lot.

Post back if you have other UK questions. There are a few of us Brits here on this forum   :)

superpat

#2
Hi Ian

Many thanks for advice. I have obtained both straight and cross over cables. With Obi110 connected but powered down,  I can make and receive calls and get ringing on my connected DECT phone.

I have seet the OBi110 as per your parameters.

With Obi110 and Freepbx in use, phone will not ring out on either set of cables.

I set th Asterisk CLI to sip debug and recorded the attached ouput, which does not mean a lot to me. although I can see ring invite ack and then error 503 all lines busy!

Please could you have a look and see if it throws any light on the problem.

URL xxx.xxx.xxx.x24 is the OBi110 and x20 is the asterisk freepbx. my home BT phone  number is 11111111111

<------------>

<--- Transmitting (no NAT) to xxx.xxx.xxx.x24:5061 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP xxx.xxx.xxx.x24:5061;branch=z9hG4bK-5d667a77;received=xxx.xxx.xxx.x24;rport=5061
From: <sip:07502146341@xxx.xxx.xxx.x20>;tag=SP239487a77396c20db
To: <sip:xxx.xxx.xxx.x20>;tag=as7284ef0c
Call-ID: 0a29788b@xxx.xxx.xxx.x24
CSeq: 8002 INVITE
Server: FPBX-2.11.0(11.6.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:s@xxx.xxx.xxx.x20:5060>
Content-Length: 0


<------------>
 == Using SIP RTP TOS bits 184
 == Using SIP RTP CoS mark 5
Audio is at 12272
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding codec 100002 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to xxx.xxx.xxx.x24:5060:
INVITE sip:11111111111@xxx.xxx.xxx.x24:5060 SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.x20:5060;branch=z9hG4bK777d9a5b
Max-Forwards: 70
From: "07502146341" <sip:07502146341@xxx.xxx.xxx.x20>;tag=as0c019365
To: <sip:11111111111@xxx.xxx.xxx.x24:5060>
Contact: <sip:07502146341@xxx.xxx.xxx.x20:5060>
Call-ID: 7057a3c2247483fd0eb020882af41929@xxx.xxx.xxx.x20:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.11.0(11.6.0)
Date: Thu, 16 Jan 2014 12:54:11 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 307

v=0
o=root 474641648 474641648 IN IP4 xxx.xxx.xxx.x20
s=Asterisk PBX 11.6.0
c=IN IP4 xxx.xxx.xxx.x20
t=0 0
m=audio 12272 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---

<--- Transmitting (no NAT) to xxx.xxx.xxx.x24:5061 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP xxx.xxx.xxx.x24:5061;branch=z9hG4bK-5d667a77;received=xxx.xxx.xxx.x24;rport=5061
From: <sip:07502146341@xxx.xxx.xxx.x20>;tag=SP239487a77396c20db
To: <sip:xxx.xxx.xxx.x20>;tag=as7284ef0c
Call-ID: 0a29788b@xxx.xxx.xxx.x24
CSeq: 8002 INVITE
Server: FPBX-2.11.0(11.6.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:s@xxx.xxx.xxx.x20:5060>
Content-Length: 0


<------------>

<--- SIP read from UDP:xxx.xxx.xxx.x24:5060 --->
SIP/2.0 100 Trying
Call-ID: 7057a3c2247483fd0eb020882af41929@xxx.xxx.xxx.x20:5060
CSeq: 102 INVITE
Content-Length: 0
From: "07502146341" <sip:07502146341@xxx.xxx.xxx.x20>;tag=as0c019365
To: <sip:11111111111@xxx.xxx.xxx.x24:5060>;tag=SP137630d8a2e4639ad
Via: SIP/2.0/UDP xxx.xxx.xxx.x20:5060;branch=z9hG4bK777d9a5b;received=xxx.xxx.xxx.x20;rport=5060
Server: OBIHAI/OBi110-1.3.0.2824

<------------->
--- (8 headers 0 lines) ---

<--- SIP read from UDP:xxx.xxx.xxx.x24:5060 --->
SIP/2.0 503 Service Unavailable
Call-ID: 7057a3c2247483fd0eb020882af41929@xxx.xxx.xxx.x20:5060
CSeq: 102 INVITE
Content-Length: 0
From: "07502146341" <sip:07502146341@xxx.xxx.xxx.x20>;tag=as0c019365
To: <sip:11111111111@xxx.xxx.xxx.x24:5060>;tag=SP137630d8a2e4639ad
Via: SIP/2.0/UDP xxx.xxx.xxx.x20:5060;branch=z9hG4bK777d9a5b;received=xxx.xxx.xxx.x20;rport=5060
Server: OBIHAI/OBi110-1.3.0.2824

<------------->
--- (8 headers 0 lines) ---
Transmitting (no NAT) to xxx.xxx.xxx.x24:5060:
ACK sip:11111111111@xxx.xxx.xxx.x24:5060 SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.x20:5060;branch=z9hG4bK777d9a5b
Max-Forwards: 70
From: "07502146341" <sip:07502146341@xxx.xxx.xxx.x20>;tag=as0c019365
To: <sip:11111111111@xxx.xxx.xxx.x24:5060>;tag=SP137630d8a2e4639ad
Contact: <sip:07502146341@xxx.xxx.xxx.x20:5060>
Call-ID: 7057a3c2247483fd0eb020882af41929@xxx.xxx.xxx.x20:5060
CSeq: 102 ACK
User-Agent: FPBX-2.11.0(11.6.0)
Content-Length: 0


---
 == Everyone is busy/congested at this time (1:0/1/0)

<--- Reliably Transmitting (no NAT) to xxx.xxx.xxx.x24:5061 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP xxx.xxx.xxx.x24:5061;branch=z9hG4bK-5d667a77;received=xxx.xxx.xxx.x24;rport=5061
From: <sip:07502146341@xxx.xxx.xxx.x20>;tag=SP239487a77396c20db
To: <sip:xxx.xxx.xxx.x20>;tag=as7284ef0c
Call-ID: 0a29788b@xxx.xxx.xxx.x24
CSeq: 8002 INVITE
Server: FPBX-2.11.0(11.6.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-Asterisk-HangupCause: Circuit/channel congestion
X-Asterisk-HangupCauseCode: 34
Content-Length: 0


<------------>
Really destroying SIP dialog '7057a3c2247483fd0eb020882af41929@xxx.xxx.xxx.x20:5060' Method: INVITE
 == Spawn extension (macro-exten-vm, s-CONGESTION, 3) exited non-zero on 'SIP/OBITRUNK1-00000008' in macro 'exten-vm'
 == Spawn extension (from-did-direct, 11111111111, 2) exited non-zero on 'SIP/OBITRUNK1-00000008'

<--- SIP read from UDP:xxx.xxx.xxx.x24:5061 --->
ACK sip:xxx.xxx.xxx.x20:5060 SIP/2.0
Call-ID: 0a29788b@xxx.xxx.xxx.x24
Content-Length: 0
CSeq: 8002 ACK
From: <sip:07502146341@xxx.xxx.xxx.x20>;tag=SP239487a77396c20db
Max-Forwards: 70
To: <sip:xxx.xxx.xxx.x20>;tag=as7284ef0c
Via: SIP/2.0/UDP xxx.xxx.xxx.x24:5061;branch=z9hG4bK-5d667a77;rport
Authorization: DIGEST algorithm=MD5,nonce="272f9374",realm="asterisk",response="f5b8d5528a717b7b7990a0ce14ebf29d",uri="sip:xxx.xxx.xxx.x20:5060",username="OBITRUNK1"
User-Agent: OBIHAI/OBi110-1.3.0.2824

<------------->
--- (10 headers 0 lines) ---
 == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'SIP/OBITRUNK1-00000008' in macro 'hangupcall'
 == Spawn extension (from-did-direct, h, 1) exited non-zero on 'SIP/OBITRUNK1-00000008'
Really destroying SIP dialog '0a29788b@xxx.xxx.xxx.x24' Method: ACK

<--- SIP read from UDP:xxx.xxx.xxx.x24:5060 --->
keep-alive
<------------->

<--- SIP read from UDP:xxx.xxx.xxx.x24:5061 --->
keep-alive
<------------->

<--- SIP read from UDP:xxx.xxx.xxx.x24:5060 --->
keep-alive
<------------->

<--- SIP read from UDP:xxx.xxx.xxx.x24:5061 --->
keep-alive
<------------->
raspbx*CLI>



I have edited the ouput  down to get under the posting limit, but I think I have included pertinent entries.    If there are any other tests I can make please advise me.


Many thanks

regards

Patrick

ianobi

QuoteWith Obi110 connected but powered down,  I can make and receive calls and get ringing on my connected DECT phone.

There's some good news here. Because you have bought an older version of OBi110 from Ebay, it still has a mains fail relay, the newer versions no longer have one. With the OBi110 powered down and the DECT phone plugged into the OBi Phone Port, if it rings and you can make calls ok, then the pin-outs are almost certainly correct.

I would take a step by step approach and start by putting the OBi110 to its default settings by doing a factory reset. Get the OBi110 and PSTN line working with your DECT phone first. Default is for the PSTN line to be the Primary Line, so you should be able to pick up the phone and dial normally. Incoming calls to the OBi110 should route to the Phone Port in its default condition. You could put the settings I suggested back in at this point. You will need to fine tune digit maps etc to get the best from the PSTN Line. Best to do all that first, then add in Freepbx.

If all that works, then it's time to add in the Freepbx. I'm no expert on Freepbx, so others may wish to post here with advice. I'm guessing that you will wish for incoming PSTN calls to be routed to the Freepbx. This can be done by changing the Line Port InboundCallRoute. Then I guess you wish to make the DECT phone an extension off the Freepbx. There's a few posts about that on this forum - as I say I'm not the expert on that.