NEED MORE INFO AudioCodes MP-114 FXS/FXO Configuration problems

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So as mentioned in the other thread, I went ahead and got the AudioCodes MP-114 FXS/FXO gateway to use with 3cx/whatever I pick next.
It's a 2 FXS + 2 FXO gateway.

I'm having a little difficulty with getting some of the configuration right (after all this is the first time I've setup a full gateway like this).

Basic Setup:


PBX:

Extension 1000 setup on pbx
Generic Sip Gateway setup and set to 2 physical ports, 4 simultaneous calls, IP authentication, both DIDs added.

Gateway:
FXS 1 (Port 1) = extension 1000
FXO 1 (Port 3) = POTS
FXO 2 (Port 4) = Ooma

I'm using per gateway authentication and using the credentials for extension 1000.
Each port is a different hunt group id (port 2 is nothing)

Tel to IP Routing:
Any source trunk id with any dest phone prefix and any source phone prefix to IP of PBX.

IP to Trunk Group Routing:
Dest phone prefix 1000# with any source phone prefix and any source IP address to hunt group 1 (port 1, extension 1000)
Any dest phone prefix, any source phone prefix, any source ip address to hunt group 2 (port 3, fxo POTS)


Incoming calls from POTS/Ooma go to PBX correctly.
Calls from SIP client to x1000 works.
Calls from x1000 to SIP client just rings fast busy (doesn't seem to ever make it to the PBX).

Calls out to the FXO need to be tested still (I did a test or two with POTS hooked up and it worked, but don't remember right now which phone I used - I'll add those results later when I test it - I think my test calls out that should have gone through Ooma failed..but again, I need to test more to be sure).

Right now my first concern is getting the FXS to IP working properly.
I'm guessing it has something to do with the Tel to IP routing (that's what log the test calls show up in), but I'm not sure what to try next.

Any help from those with more experience would be greatly appreciated.

This device has a massive number of options and there don't see to be many examples out there for it. Most are for the pure FXO versions or run through a basic setup and never get into how to actually configure the routing properly for FXS to SIP clients on the PBX. :)

Thanks as usual!
 
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Yeah. Here's the config:
The PBX is currently 192.168.130.104 (to be changed once I swap it for the old one) and the AudioCodes is 192.168.130.6.
acset1.png

acset2.png
 
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Just as a parallel effort have you contacted Audiocodes or the dealer? They will usually supply initial support to get a device installed.
 
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I haven't yet, it hasn't been super high on my priority list.

I did finally run through and test everything today to see where things are/aren't working:

FXS - cordless phone hooked up to the FXS port, X1000 on the server
IP - 3cx client on my phone X1100 on the server

POTS Incoming calls are routed to a call group that calls several extensions, including the FXS and my extension.
I had all outbound rules configured to go through the gateway for POTS testing.

POTS Incoming answered by FXS: Works fine
POTS Incoming by IP: Works fine

POTS Out from FXS: Busy
POTS Out from IP: Works

FXS calling IP: Busy
IP calling FXS: Works but terrible audio (lots of dropped audio and bad quality in both directions)

Looking at the call log in 3CX I don't even see the POTS Out from FXS or the FXS Calling IP calls, so those appear to never hit the server from the gateway..hmmmm (and unfortunately those are the most important ones other than POTS Incoming Answered by FXS)
 

phonebuff

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@Mark D. Montgomery II

I have not touched analog for almost three years, but when I do the AudioCodes MP platform was always my first choice.

The support from them was always great, even got a custom build to solve a disconnect issue from a PBX once. Anyway to me this sounds like the MP thinks the Asterisk box is not there and is trying to do a hair pin with the calls internally, probably with a dial pattern that does not work for your PSTN service. I would start in the MP not in the Asterisk service.

========
 

ou812

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It's also been years since I used pots but I still have a couple in service today, here are notes for the install I did.


AudioCodes MP-114 fxo Setup

1. GENERAL PARAMETERS
NETWORK
a. Quick Setup
b. IP Address = Enter the IP address of your Audio Codes device
c. Subnet Mask Address = Enter Your Subnet Mask
d. Default Gateway IP Address = Enter your Router’s IP Address

PROXY
e. Gateway Name = make this field blank
f. Working with Proxy = No
g. Proxy IP Address = 0.0.0.0
h. Proxy Name = make this field blank
i. Enable Registration = Disable

CODERS
j. Coders Table = Select G.711U-law (or whatever other codecs you use)

1. OUTBOUND CALLS:

PIAF
a. Prelimnary PIAF setup: Create Audio Codes Trunk
i. Trunks > add trunks > sip trunk
1. trunk name = Gateway1
2. peers details:
a. host = ip of your MP11x
b. type = peers
c. qualify = yes
3. submit changes and apply configuration changes

ii. outbound routes
1. “properly” configure an outbound route to use the Trunk created above

b. AudioCodes: Management > Routing Tables > IP to Trunk Group Routing
i. Dest. Phone Prefix = *
ii. Source Phone Prefix = *
iii. Source IP Address = IP address of your PIAF
iv. Hunt Group ID = 1 (I finally discovered that this setting is critical)
v. Profile ID = does not seam to matter, set to “0” or “1” (i used 0)
c. Endpoint Phone Number Table
i. Channel(s) = “1-8” (for MP118 or “1-4” for MP114)
ii. Phone Number = your POTS telephone number
iii. Hunt Group ID = 1 (must set)
iv. Profile ID = 1 (setting does not ‘seem’ to matter) (i used 0)
d. Hunt Group Management
i. Hunt Group ID = 1 (corresponds to “1” in step above)
ii. Channel Select Mode = cyclic ascending (i used descending)
iii. Registration Mode = don’t register (this setting does not ‘seem’ to matter)
e. Advanced Applications > FXO Settings
i. Dialing Mode = one stage (two stage will give you dial tone again, then allow to dial your PSTN phone number, good for ‘dial 9 for an outside’ call I guess)
ii. YOU ARE NOW READY TO MAKE A CALL!


1. INBOUND CALLS:
a. Preliminary Steps: in PIAF, in ‘general settings’, allow anonymous inbound SIP calls (NO SEE NOTE)
b. Protocol Management > Routing Tables > Tel to IP Routing
i. Dest. Phone Prefix = *
ii. Source Phone Prefix = *
iii. Dest. IP Address = IP Address of Your PIAF
iv. Hunt Group ID = 0 (does not ‘seam’ to matter for inbound calls)
v. Profile ID = 0
vi. Hit “Submit”
c. Endpoint Phone Number Table
i. Channel(s) = “1-8” (for MP118 or “1-4” for MP114)
ii. Phone Number = your POTS phone number or any arbitary extension or phone number
iii. Hunt Group ID = 1 (must set)
iv. Profile ID = 1 (setting does not ‘seem’ to matter) (i used 0)
d. End Point Settings > Automatic Dialing
i. Destination Phone Number = PIAF extension you want to ring when someone calls your phone number, most liking your main extension, a ring group extension, or a queue extension
ii. Auto Dial Status = enabled
iii. Click “Submit”
iv. YOU ARE NOW READY TO RECEIVE CALLS!

NOTE
Anonymous sip calls should already be terminated by freepbx
Use Inbound routes by D.I.D. Number
Set D.I.D. Numbers in Automatic Dialing
Set C.I.D. Numbers in EndPoint Phone Number.


Gary.
 
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Thanks!
That helped out a bit but I'm still having odd issues.
I think I'll be dropping a message to AudioCodes.

I did rebuild the gateway like yours which works it seems, but calls are still weird.
Calls from SIP phones to the extension on the Gateway works fine.
Calls from the extension on the Gateway to the SIP numbers rings the default inbound rule for the gateway instead of the extension dialed (I tried with no registration and with Per FXS registration and it behaved the same both ways).
When I made a call to an outside number from the extension on the gateway it also range the default inbound rule on the pbx. :p

Part of this is obviously configuration issues due to me having the FXO/FXS model not the pure FXO model since I need that Analog extension on the gateway...lol.
 

trumee

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Yeah. Here's the config:
The PBX is currently 192.168.130.104 (to be changed once I swap it for the old one) and the AudioCodes is 192.168.130.6.
acset1.png

acset2.png
Your config no longer appears on the forum. Please can you post it again.

I have a MP-114 FXO_FXS device and am trying to set it up. I want to send all the calls from FXO/POTS to PBX, unfortunately that part is not working for me. The device picks up the incoming FXO call after two rings and offers me a dialtone. I have enabled 'One Stage' dialing in Analog Gateway>FXO Settings, so dont why it is doing 'two stage'. In 'one stage' the call should be directly sent to the PBX.

Here is the log when a calls comes in through FXO/POTS port. I dont see any activity in the PBX.
Code:
Log is Activated

7d:12h:54m:28s (   lgr_psbrdex)(110445    )  recv <-- ANALOG_IF_RING_START Ch:2 type(0)

7d:12h:54m:28s (      lgr_flow)(110446    )  #2:RING_START_EV

7d:12h:54m:28s (      lgr_flow)(110447    )  |       #2:RING_START_EV State:PRECALL_IDLE Substate:sub_RING_END_SECOND_STATE

7d:12h:54m:29s (   lgr_psbrdex)(110448    )  recv <-- EV_ANALOG_IF_RING_END Ch:2 type(0)

7d:12h:54m:29s (      lgr_flow)(110449    )  #2:RING_END_EV

7d:12h:54m:29s (      lgr_flow)(110450    )  |       #2:RING_END_EV State:PRECALL_IDLE Substate:sub_RING_END_SECOND_STATE

7d:12h:54m:29s (   lgr_psbrdif)(110451    )  ActivateDigitMap for channel : 2, MaxDialStringLength = 20, MaxEndDialTimer = 4000,
MaxLongInterDigitTimer = 8000, MaxStartTimer = 16000, DigitMap = [0-9*#ABCD][0-9ABCD].T, DPIndex = -1, DPPriority = 0

7d:12h:54m:29s (      lgr_flow)(110452    )  #-100: StartDigitMapDetection with params:
<Pattern=[0-9*#ABCD][0-9ABCD].T>
<MaxStartTimer=16000>
<SendEachDigit=1>
<UseEndDialKey=0>
<MaxLongInterDigitTimer=8000>
<MaxEndDialTimer=4000>
<MaxDialStringLength=20>
<MaxShortInterDigitTimer=2000>
<MinInterDigitLen=-2>
<MinDigitLen=-2>
<EndDialWithHashMark=0>

7d:12h:54m:29s (lgr_digitmap_mngr)(110453    )  #2:Activate DigitMapMngr pattern:[0-9*#ABCD][0-9ABCD].T, Max Length is: 20, DPIndex: -1, DPPriority: 0

7d:12h:54m:29s (   lgr_psbrdif)(110454    )  UpdateChannelParams, Channel 2

7d:12h:54m:29s (   lgr_psbrdif)(110455    )  #2:ConfigFaxModemChannelParams NSEMode=0, CNGDetMode=0, FAXTranType=0, VxxTranType=0, VoiceVol= 0, DTMFVol=-11, InGain=0, RTPRedDepth=0, ECE=1,ECEType=0 SCE=0, ECNlpMode=0, DJBufMinDelay=10, DJBufOptFac=10, Result=1)

7d:12h:54m:29s (   lgr_psbrdif)(110456    )  Turn ringer ON for channel 2

7d:12h:54m:29s (      lgr_flow)(110457    )  |       #2:FXO Seize Line

7d:12h:54m:29s (   lgr_psbrdex)(110458    )  PCIIFChangeChannelParams failed  ECCNG ECNlpMode

7d:12h:54m:29s (   lgr_psbrdif)(110459    )  Changed ECNlpMOde to: 1
 
Last edited:

tbrummell

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I haven't set up an AC FXO on to Incredible/FreePBX in a very long time, but, you will need to have the Automatic dialing on that port(s) set to your ring group/extension/auto attendant. You are getting dialtone back because the gateway doesn't know what to do with you.

When you call your number and hear dialtone, if you were to dial a dialable number on your pbx, it will most likely ring.
 

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