Audiocodes Internal Document "How do I set up Call Termination on FXO?"
Debug Level - The default is 0 (off) so the first thing you should do is turn it up to the highest. Then you can browse to the "Message Log" page and see every event that happens on the gateway.
Relevant ini file parameters:
GWDEBUGLEVEL = 5 (<=4.4)
GWDEBUGLEVEL = 6 (>=4.6)
Max Digits - The default is 3 so when the a TEL->IP phone call is made, an FXS gateway sends the SIP INVITE after the 3rd digit is dialed.
Change this to the maximum you allow.
Relevant ini file parameters:
MAXDIGITS = 3
Registration - There are 2 trains of thought on setting up FXO Gateways:
Gateway registers with Proxy/Registrar and sends all PSTN->IP calls to that entity.
Option 1 - Gateway registers itself to the Registrar
Subscription Mode = "Per Gateway"
Option 2 - Gateway registers it's End Points with the Registrar
Subscription Mode = "Per Endpoint"
Gateway does not register but sends PSTN->IP calls based on the "Tel to IP Routing" table.
Call Setup PSTN->IP calls - An analog Gateway has to do one of 2 things when a Ring signal comes on its line.
Hotline to a certain SIP URI (this is the "Automatic Dialing" table).
Provide "2nd Dial-tone" and let the User input more digits to route on.
Call Setup IP->PSTN - Two methods available:
Two Stage Dialing - Seize the "PSTN line" and play that dial tone back to the IP user to enter digits to the PSTN switch.
One Stage Dialing - Seize the line, optionally wait for dial tone, and outpulse digits to the PSTN switch for the user.
DTMF - Two methods available:
(RFC2833) - Preferred - Also called Out of Band.
Both sides have to support this.
Make sure that your DTMF Payload type matches the far end. It should be automatic but some devices have to be explicitly assigned.
(INBAND) - Doesn't work well with compressed codecs.
Call Termination when the PSTN caller hangs up first (Or on a Voicemail/Conference Server)
Users Manual Section 8.4 says there are 6 methods supported by the MP1xx FXO.
In order of preference:
Detection of polarity reversal / current disconnect - This is the recommended method. The call is immediately disconnected after polarity reversal or current disconnect is detected on the Tel side (assuming the PBX / CO produces this signal).
Relevant ini file parameters:
EnableReversalPolarity
OR
EnableCurrentDisconnect Activating both at the same time could cause problems. Also, verify that the PSTN line or PBX is in fact providing these signals.
Detection of Reorder / Busy tones - The call is immediately disconnected after a Reorder / Busy tone is detected on the Tel side (assuming the PBX / CO produces this tone). This method requires the correct tone frequencies and cadence to be defined in the Call Progress Tones file. If these frequencies are not known, define them in the CPT file (the tone produced by the PBX / CO must be recorded and its frequencies analyzed). This method is slightly less reliable than the previous one.
Relevant ini file parameters:
TimeForReorderTone
The CD and new firmware come with a "Call Progress Tones Wizard" to record the sounds played by the PSTN line. It then builds a custom INI/DAT file you can load that contains the tones that your network is playing (Ringback, Reorder, Busy, etc). It might still need tweaking though.
Detection of silence - The call is disconnected after silence is detected on both call directions for a specific (configurable) amount of time. The call isn’t disconnected immediately; therefore, this method should only be used as a backup.
Relevant ini file parameters:
EnableSilenceDisconnect
FarEndDisconnectPeriod (with DSP templates number 2 or 3)
A Special DTMF code - A digit pattern that, when received from the Tel side, indicates the gateway to disconnect the call.
Relevant ini file parameters:
TelDisconnectCode
Interruption of RTP stream - This method works correctly only if silence suppression is not used.
Relevant ini file parameters:
BrokenConnectionEventTimeout
DisconnectOnBrokenConnection
Protocol-based termination of the call from the IP side - This method is controlled by the FAR IP end only.
Relevant ini file parameters:
None
The best advice we can give is to set your "Debug Level" to 5, then watch the "Message Log" page for what the gateway is doing. As a VAR, this will be invaluable if anything goes awry in 6 months.
Configure the gateway in 5 steps:
Upload Auxiliary Files
Registration
PSTN->IP Calls
IP->PSTN Calls
Security
Hints:
Auxiliary Files:
Call Progress Tones - See User Manual for more detail.
This helps the MP recognize what the PSTN line is doing (ie. Ringing, Busy, Reorder). Upload the file that matches your PSTN/PBX tones.
You can also build your own .dat with the Call Progress Tones Wizard.
Coefficients File - See User Manual for more detail.
Helps the MP match the electrical characteristics of the PSTN line.
Registration:
Watch the "Message Log" and the debug output of your proxy to see what registration messages it is sending to the proxy/registrar.
You can set your "expire time" down low so that you can see the MP re-registering every so many seconds in the "Message Log" instead of the default 3600.
PSTN-IP calls:
For FXO PSTN->IP calls, you have to set the "Automatic Dialing Table" to tell the MP1xx where to call.
Watch the "Message Log" and the debug output of your proxy to see what Messages the MP it is sending[receiving] to[from] the proxy/registrar.
Remember that until the MP sends an INVITE to the Proxy, nothing will work.
Hang up the PSTN side first to make sure the MP recognizes that the PSTN line was disconnected and drops the IP side.
If it doesn't, look into Current Disconnect, Polarity Reversal, and/or Call Progress Tone Detection in that order.
IP-PSTN calls:
For FXO IP->PSTN calls, you can set up either Two-Stage or One-Stage.
For One-Stage, the gateway will outpulse whatever it gets in the INVITE to the PSTN line.
Watch the "Message Log" and the debug output of your proxy to see what Messages the MP it is sending[receiving] to[from] the proxy/registrar.
Remember that until the MP receives an INVITE from the Proxy, nothing will work.
After you get one port working, activate a Hunt Group to utilize multiple ports at the same time.
Configuration Security:
Change the Web Admin password.
You can also make the Web Pages read-only which would only allow updates by the bootP server. (See the documentation)
Call Security:
Audiocodes Gateways can be set up to only accept SIP Messages from "Known Hosts", Ie. the proxy or what is in the Routing Tables.
Some folks set up a "passcode" that is inserted by the Proxy into the "Dialed Digits", then deleted by the MP before it outpulses those digits to the PSTN line.
Last, after you get it working, download your working INI file and put it (and the Auxiliary files) in a safe place so you don't have to go through this again if you have a catastrophe.
Contacting ABP for Support on Audiocodes Gateways:
4 things are always needed (as attachments, not pasted into an email):
1. The INI file you download from the gateway.
2. The output of the "Message Log" of your failure (with debug level = 5)
3. An ethereal trace of the failure in #2.
4. Details of what you are trying to do and what is happening.
If you don't already have one pending, open a Support Request with #4 only in the description. You will receive an automatic email reply from doing this.
Reply to that email with #1(board.ini), #2(message_log.txt), and #3(trace.pcap) attached.
Written : 2005/03/01
Modified: 2006/03/07
by: Shanon Swafford
ABP information supplied as a best effort only.
ABP does not guarantee suitability of products or provide any guarantees or warranties on our vendors products.