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analog to VoIP conversion

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telnettech

Vendor
May 5, 2005
207
US
need some help, i have a asterisk ip pbx with 140 ip pohnes and need to add 132 + analog phones to the system. I have "rhino" converters and i am not able to get the tip and ring down correctly.....i need to know what kind of card i need in my pbx and also what signalling i should use.....i have tried fxols and im able to go off hook and talk but i dont hear the phone ring.....i tried fxoks and the phone rings but i am not able to answer the call....i am not able to get dial tone with either one.....help would be greatly appreciated


Brian
 
Brian,

Not really familiar with rhinos (but they seem straightforward enough) - have you tried reversing the polarity on those lines?

Reversing the polarity ALWAYs worked in Star Trek, it's gotta work here!
 
the analogs are punched down on a leviton cat 5 data panel 568B....we then ran a rj 11 to an ortronic cat 3 patch panel....they are connected to the Rhino box by amphenol cable......r u saying to swap my tip and ring around?


Brian
 
If you are connecting to a "station" the correct interface is Foreign Exchange Station (FXS).

Here is how it was original designed

CO---<FXO>------(carrier)-----<FXS>---Station set

CO and FXS:

-Generate loop current/voltage
-Look for far end device to change to low impedence (e.g. allow loop current to flow) to indicate "off hook"
-Generate ringing voltage to indicate a call is present for far end
-Respond to addressing (dialing) from far end

Station set and FXO:

-Signal on hook and off hook by high impedence and low impedence states respectively (low impedence permits loop current, provide by the CO or FXS, to flow.
-Respond to ringing voltag eto indicate an incoming call
-Transmits adress digits to the CO or FXS

Both FXS and FXO can be LS (loop start, the usual type) or GS (ground start).


Some FXS cards require an external "ring generator" to provide the 20 Hz 90 VAC ring current. Some have it built in.

Best of luck.
 
yeah - i meant swap tip and ring - I think that might be responsible for your 'no ring tone' symptom.

ISDNman, can you comment? My analog signaling sucks.
 
Hey Telnetech,

Please fill n some details:

-You are connecting the Asterix to analog phones, or is there some other PBX involved? If this is simply setting up an analog port to connect to analog phones then the post should be FXS (since it is connecting to a station) and it should be loop start.

It is possible that Rhino has this confused, since most people lack the context on how this all fits together (see my previous post). Similar to how most people don't undertstand how whether an RS-232 port is considered DTE or DCE determines if "Transmit data" is and output or and input (it is an output on a DTE and an input on a DCE).

If you give me the model number and a link to the Rhino device you are using I can look to see if they have this backwards.

-When you dial this analog extension, do you hear "ring back tone" aka "audible ring" at the dialing phone or not?

-When you dial this analog extension does it ring?

Reversing Tip and Ring is worth a shot, but in this case it is a long shot. In this case I'd certainly expected balanced ringing so I don't think it should matter.

The usual case where polarity matters is that some sets (older 2500 sets for example ) have DTMF pads that are polarity sensitive.
 
ISDNMAN

1) there is no other PBX involved in the system. This is a Asterisk IP PBX running on its own server. we are connecting the Rhino to an additional T-1 port we have on the T-1 card in the server using a normal RJ48C T-1 crossover cable that was supplied by Rhino.

2)The Rhino channelbank has no model number. it is a 24 POTS FXS channelbank. the Rhino has a auto T-1 configuration that senses the signalling and sets itself up based on this(E&M,ESF,AMI,Loopstart,etc...). Also we have programmed the analog side to be loopstart. you can see on the LED Display that the line is getting ring voltage and when you go off hook you can see also that the Rhino detects it. the Rhino connects to an ortronics 24 port, cat 3 patch panel by 25 pair Telco connectors(amphenol cable)on each end.

3)when you call the ext, you DO get a ring on the phone and can hear the ring on the other end, but when you go off hook, you can not talk, and the other end continues to ring the cadence until it goes to voicemail. When you pick up the analog phone, you get no dial tone. I swapped my tip and ring as asked earlier in the post and was able to talk but still got no dial tone to make the call or was able to hear the phone ring. i verified that the cat 3 patch panel was pinned correctly (pins 4 and 5).
 
RTFMDUDE,

Loop signaling primer.

On any modern analog set made after mid 1980's or so, polarity is not an issue. As ISDNman stated, early touch-tone sets required power from the loop for the dialpad and polarity had to be observed.

For the record, a loop start subscriber line is a single pair, usually white/blue - blue/white or red and green in the case of quad wire. Red or White with a blue tracer is the ring side which is generally -48 volts DC on-hook from the CO battery. Green or Blue with a white tracer is at ground potential. Ground start trunks are similar.

Loop provides dialtone when an impedance load (about 200 ohms, IIRC)is placed on the line by going off-hook. Ground starts require briefly grounding the ring side, then dial tone is heard - Ground starts are for PBXes so that they will not sieze a trunk that is in use for outgoing calls.

Trivia: Tip and ring are named for the plugs used on old-time manual switchboards that had three contacts available, the tip, ring, and sleeve, just like a stereo headphone plug.

jsaxe

"When the going gets weird, the weird turn pro."
Hunter S. Thompson (R.I.P. Doc)
 
Sounds like the Rhino box is misconfigured on the station side.

Or it might be defective. Have you looked to see if they have a troubleshooting FAQ
 
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