Commend Audiocodes MP-11 Series User manual

Audiocodes MP-11x FXO Example Setup Guide v1.6
NOTE: For rapid setup follow all settings suggestions below in RED for a typical configuration.
NOTE: This document is for reference only. The configuration examples should work equally with a
serverless or VirtuoSIS based environment. Modifications to the settings may need to be made to
match the needs of the application.
NOTE: All screenshots were taken with firmware 6.60A in the device.
NOTE: For more information on the parameters described below contact Audiocodes the manufacturer of
this device.
MISC: The unit takes approximately 2 minutes to boot up after power is applied. The Ready LED will
illuminate Green.
MISC: To permanently save configuration settings you must “Burn” them by pressing the button at the
top. All non-burned settings will be lost after are power cycle or reboot.
MISC: Some configuration changes require a reboot of the unit. This is performed by choosing ‘Reset’
from the Device Actions at the top. This will not actually “reset” any configuration.

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Index
1. Web Interface Page 3
2. Default Account Page 3
3. Static IP Address Page 3
4. DHCP Page 3
5. Endpoint Phone Number (Channel Assignment) Page 4
6. Hunt Group Settings (Number Matching) Page 5
7. IP to Trunk Group Routing (SIP call handling) Page 5
8. FXO Settings (Dial Mode) Page 6
9. Tel to IP Routing (Analog call handling) Page 7
10. Automatic Dialing Page 8
11. Coders (Codec Settings) Page 8
12. DTMF (Button Events) Page 9
13. IP->TEL Manipulations (Changing of digits dialed) Page 9
14. TEL->IP Manipulations (Changing of digits dialed) Page 10
15. VirtuoSIS Manipulations Page 11
16. SIP General Parameters Page 11
17. Volume Level Adjustments Page 12
18. On-Hook Detection (Call End) Page 12
19. Voicemail Detection Page 13
20. Caller ID Page 14
21. Registration Page 14
22. Firmware Updating Page 16
23. Troubleshooting Page 17

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1. Web Interface
Default IP: 10.1.10.10 (FXS/FXO) or 10.1.10.11 (FXO only) with subnet mask of 255.255.0.0.
If unknown see Section 23a for troubleshooting.
2. Default Account
Username: Admin
Password: Admin
(Case sensitive)
3. Static IP Address
a. Configuration >VoIP >Network >IP Interfaces Table
DHCP (disabled by default) will override any configured static IP Address!
4. DHCP
a. Configuration >System >Application Settings
b. Unless your network is using a DHCP server with Static Leases then it is recommended
to leave this option disabled.
c. Example FXO configuration:
Enable DHCP: Disable

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5. Endpoint Phone Number (Channel Assignment)
a. Configuration >VoIP >GW and IP to IP >Hunt Group >Endpoint Phone
Number
b. Channel references the physical ports on the back of the device. If your phone line is
connected to Port 1 FXO then enter Channel=1 below. If you have a 2FXS/2FXO Device
then reference Channel=3 as this is the first FXO port on the back.
c. Phone Number is only really meaningful for FXS connected devices. It is still
recommended to enter a value as this will show up in SIP Traces and will help to
reference activity. For this example, the numbers are made up.
d. Hunt Group ID matches the phone numbers that are dialed. It is paired with the Channel
number to choose which interface that they are attempted on. (defined later)
e. Tel Profile ID is which profile should be in use. A profile of 0 means that the global
settings will be used. Otherwise it will use the named profile (created later).
f. Example FXO configuration:
Channel = 1
Phone Number = 7001
Hunt Group ID = 1
Tel Profile ID = 1

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6. Hunt Group Settings (Number Matching)
a. Configuration >VoIP >GW and IP to IP >Hunt Group >Hunt Group
Settings
b. Hunt Group ID is the row number in Step 5. This links the Hunt Group to the Channel.
c. Channel Select Mode is how it hunts for a match. It is recommended to use ‘Cyclic
Ascending’ or ‘Cyclic Descending’ to have the gateway route the call to the first available
POTS line within the group configured in Step 5 (Endpoint Phone Number).
By Dest Phone Number –Matching based on called (destination) number
Cyclic Ascending –Next available channel (number) from low to high
Cyclic Descending –Next available channel (number) from high to low
By Source Phone Number –Matching based on calling (source) number
d. Registration Mode is only required if the Audiocodes box is to register to another SIP
Server. For example, “Per Account” can be set. Also for registration, you must also go to
Configuration > VoIP > SIP Definitions submenu > Account Table. Normally, registration
can be ignored.
e. Example FXO configuration:
Hunt Group ID = 1
Channel Select Mode = Cyclic Ascending
7. IP to Trunk Group Routing (SIP call handling)
a. Configuration >VoIP >GW and IP to IP >Routing >IP to Hunt Group
Routing
b. This table will route inbound traffic from the IP side to the appropriate ‘Hunt Group ID’
c. Dest. Phone Prefix matches the beginning (prefix) or ending (suffix) digits of the called
(destination) number. A special notation syntax is used. For example, [100-199] would
match any number starting in that range. An * would match any and $ would match none.
d. Source Phone Prefix matches the beginning (prefix) or ending (suffix) digits of the
calling (source) number. A special notation syntax is used. For example, (101,103,109)
would match only numbers ending with 101, 103, or 109. An * would match any and $

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would match none.
e. Source IP Address matches the IP Address of the source of the call. This is obtained
from the CONTACT header in the SIP INVITE. An * would match any number (0-255)
and x would match a single digit. For example, 10.10.x.* would match any Source IP from
10.10.0.000 to 10.10.9.255.
f. Hunt Group ID indicates the Hunt Group from Step 6 (Hunt Group Settings).
g. IP Profile ID indicates the IP Profile to use (created later).
h. Example FXO configuration:
Dest. Phone Prefix = *
Source Phone Prefix = *
Source IP Address = *
Hunt Group = 1
IP Profile ID = 1
8. FXO Settings (Dial Mode)
a. Configuration >VoIP >GW and IP to IP >Analog Gateway >FXO Settings
b. Dialing Mode setting determines how calls are dialed on the phone line.
One-stage dialing seizes one of the available lines and dials based on Step 6
Channel Select Mode. It dials the destination phone number received in the SIP
INVITE. Consider the “Waiting for Dial Tone”value to specify whether the
dialing will start immediately or only after detection of the dial tone.
Two-stages dialing seizes one of the available lines without performing any
dialing and connects the remote SIP device. All further signaling (Dialing and Call
Progress Tones) is/are performed directly from the remote SIP device. This gives
flexibility to the remote SIP device to choose what dialing happens next. For
example, a local PBX extension or 9 (for an outside line) is dialed.
c. Example FXO configuration:
Dialing Mode = One Stage

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Waiting for Dial Tone = No
9. Tel to IP Routing (Analog call handling)
a. Configuration > VoIP >GW and IP to IP >Routing >Tel to IP Routing
b. Here the incoming routing from the FXO ports to a specific IP address is defined. This is
only needed if calls should originate from the telephone side bound for SIP.
c. Example FXO configuration:
Src. Hunt Group ID = *
Dest. Phone Prefix = *
Source Phone Prefix = *
Dest. IP Address = [SIP UA, host, or device] (for example Linux IP of VirtuoSIS)
Port = 5060 (default)
Transport Type = UDP (default of any Commend device)
Dest. IP Group ID = -1 (any group)
IP Profile ID = 0 (use global)
Cost Group ID = None (only for FXS)

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10.Automatic Dialing
a. Configuration >VoIP >GW and IP to IP >Analog Gateway >Automatic
Dialing
b. Here you can define phone numbers that are automatically dialed as soon as the line is
hooked-off. By default, when a call is received from the analog side it will play dial tone
and wait for digits to be dialed. In some cases, it is desired to have the gateway
automatically send digits to a predetermined IP endpoint. For example, the analog side
hooks-off and 0 is immediately dialed (placing a Call Request in the Intercom Server).
c. Example FXO configuration:
FXO port #1 will automatically dial ‘12015292425’
FXO ports #2-4 will provide dial tone to the analog side
11.Coders (Codec Settings)
a. Configuration >VoIP >Coders and Profiles >Coders
b. Here you can define the codecs used and offered (in SDP). Depending on the codec
used it will determine the quality and bandwidth of the resulting call.
c. The G.722 codec affords the possibility of the highest possible speech quality (7kHz).
This will not be an available option until you change the DSP Template in effect under
Configuration->VoIP->Media->General Media Settings from 0 to 1. Be aware that this will
limit the maximum number of channels in some models.
d. Example FXO configuration:
Coder Name = G.711A-law
Coder Name = G.711U-law

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12.DTMF (Button Events)
a. Configuration >VoIP >GW and IP to IP >DTMF and Supplementary >DTMF
& Dialing
b. When you press a button on either side (SIP or analog) the event must be transported to
the other side. The method determines the type of delivery that these DTMF events are
transmitted by. Both sides must support and be using the same method. All Commend
SIP devices and servers support ‘RFC2833’. Additionally, some support ‘SIP Info’.
c. Example FXO configuration:
1st Tx DTMF Option = RFC2833
13.IP->TEL Manipulations (Changing of digits dialed)
a. Configuration > VoIP > GW and IP to IP > Manipulations > Dest Number IP-
>Tel
b. Manipulations change what digits are dialed or presented on TEL bound calls. This can
affect what needs to be dialed and what is actually dialed. For example, adding a 9 in
front of the number because it might be required to dial an outside line. This will override
whatever is already setup in Step 10 (Automatic Dialing).
c. For adding a pause in the dialing try using either ‘p’ or ‘,’.

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d. During my testing my PBX required dialing 9 first to redirect dialing to a POTS line
instead of dialing other direct PBX extensions. My PBX was quirky and required an extra-
long pause. Therefore, I had to add the double “9p9p” to get it working. This is not
normally required. Typically adding “9,” is enough.
e. Example FXO configuration:
Index 0
Number dialed by VirtuoSIS = 8
Actual telephone number dialed=9,Pause,9,Pause,18005551212
Index 1
Number dialed by VirtuoSIS = 911
Actual telephone number dialed = 911
Index 2
Number dialed by VirtuoSIS = 6
Actual telephone number dialed = 9,Pause,9,Pause,12015292425
14.TEL->IP Manipulations (Changing of digits dialed)
a. Manipulations change what digits are dialed or presented on IP bound calls. This can
affect what needs to be dialed and what is actually dialed. For example, removing the
digits dialed and instead choosing different numbers.
b. Configuration > VoIP > GW and IP to IP > Manipulations > Dest Number
TEL->IP
c. Example FXO configuration:
Index = 1, Any incoming number will dial 102 instead of what digits were
originally dialed on the telephone side.

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15.VirtuoSIS Manipulations
a. VirtuoSIS uses Asterisk to handle SIP related activities. Asterisk is a VoIP Server and
can also make manipulations to the dialstring. In the Commend CCT800 software go to
Project/Interfaces/SIS-SIP-T/Trunks/[Your Trunk] and under Dial Plan Configuration click
on the ‘Add Outgoing’ button. A new window (Dial plan configuration) will appear.
b. Example: Take the phone number dialed by VirtuoSIS on the trunk and prepend a 9 and
one second pause to it:
In the Dial plan Configuration window enter:
1,Dial(SIP/SIP-trunk01/9\,${EXTEN:2})
c. Example: Take the outgoing Caller-ID and change it to custom text:
In the Dial plan Configuration window enter:
1,Set(CALLERID(name)=From-Intercom Site ABC)
16.SIP General Parameters
a. Configuration > VoIP > SIP Definitions > General Parameters
b. Here you can setup other SIP specific parameters like Port Number, External NAT IP,
and Transport.
c. Example FXO configuration:
SIP Transport Type = UDP

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17.Volume Level Adjustments
a. Configuration > VoIP > Media > Voice Settings
b. One some telephone lines or telephony systems the volume of the audio on the call can
be too low. The typical symptom is loud ringing indication on the SIP Intercom Device
(set locally by the device itself), but when the call starts the volume is low. Therefore, an
adjustment on the Audiocodes box to the input/output gain can be one way to adjust for
this.
c. Voice Volume adjusts the IP-to-TEL volume level. Input Gain adjusts the TEL-to-IP
volume level.
18.On-Hook Detection (Call End)
a. The antiquated telephone system does not send data on the line. Therefore, there is no
standard way to indicate call progress on the far end (call ringing, call answered,
voicemail, etc.). Many telephony systems attempt to deal with this by requiring DTMF
digits to be dialed during the call. For example, voice prompts in a menu.

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b. To indicate that a phone call has ended the far end may indicate this by one of several
different methods. These are only indications and cannot actually do anything. A chosen
method must be supported by both ends to work!
DTMF Event (button pressed)
Reorder / Dial Tone
SIT (Special Information Tone) Tone (3 rising tones followed by an error)
Off-Hook “Howler” Tone (loud 0.1s ON / 0.1s OFF tone)
Polarity Reversal / Current Disconnect
c. Additionally, the following can also be used by the SIP FXO side (near end) to
additionally detect a dead call:
RTP stream interruption
Silence detection
d. The Commend SIP Devices or Server also require some kind of call progress indication.
SIP supports signaling during the call to indicate call progress. However, when
connecting SIP to the analog telephone side this issue must be considered with one of
the above methods.
e. The above issue can cause the Audiocodes device to not hang up automatically after
the call has ended on the analog telephone side. If the Audiocodes device still believes
the call to be activate then it will not in turn send a SIP BYE to cause the SIP side to hang
up either. Therefore, the SIP Device or Server still shows the call as active.
f. In the Search field enter the terms DisconnectOnBusyTone, DisconnectOnDialTone,
EnableCurrentDisconnect, and EnableReversalPolarity to find the relevant settings.
g. Try loading country specific or custom dial tone files. This is found under Maintenance >
Software Update > Load Auxiliary Files > Call Progress Tones.
h. As a backup Silence Detection may also be used. If both ends are silent for a definable
amount of time the call can also be disconnected automatically. In the Search field enter
the terms EnableSilenceDisconnect and FarEndDisconnectSilencePeriod.
i. These settings are described in more detail in the Audiocodes MP-11x Manual under
section 26.14.3.1 Calls Termination by PBX.
19.Voicemail Detection
a. As mentioned in Section 18. On-Hook Detection (Call End) there is no way to detect
Voicemail as there is no ability for the antiquated telephone system to send Call
Progress.
b. As a work around the following may be helpful. The greeting on the phone is re-recorded
to start with a DTMF event (button 2) sound. This triggers the Audiocodes device to
disconnect and consider it a failed attempt. The SIP Intercom device is given call
progress and will initiate another call attempt to the next number as programmed.
Commend SIP Intercom programmed with Phonebook Sequence:
oLine Row 1:
a. Call: [e-mail] (IP of Audiocodes)
b. Wait = 30s
oLine Row 2:
a. Call: [e-mail] (IP of Audiocodes)

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b. Wait = 30s
oIntercom will advance to next phone number if it does not get a SIP 200
OK within 30 seconds of first call attempt
Audiocodes device programmed with:
oAnswer Supervision=yes (Search: ENABLEVOICEDETECTION)
oVoiceMailInterface=DTMF (Search: VOICEMAILINTERFACE)
oDisconnect Call Digit Pattern=2 (Search: TELDISCONNECTCODE)
Voicemail on phone re-recorded to start with DTMF 2 playing following by
standard voice greeting.
20.Caller ID
a. Caller ID from the SIP side comes from the SIP From header. For example:
From: “John” <SIP:[email protected]>;tag=35dfsgasd45dg
On the analog telephone side Caller ID is CNAME (name portion) and CNUM (phone
number).
b. Bellcore FSK is the type of Caller ID standard used in North America. This is the default
method set by the device (“Standard Bellcore”).
c. On analog telephone POTS lines PSTN telephone carriers will ignore the CNAME and
CNUM in Caller ID; this is a fixed setting by the carrier. Therefore, you must connect to
the carrier by SIP or PRI ISDN to get custom Caller ID.
d. By default, Caller ID is disabled for the analog telephone side. Search:
ENABLECALLERID or use path above. Set “Enable Caller ID” to ‘Enabled’.
e. Configuration > VoIP > GW and IP to IP >DTMF and Supplementary >
Supplementary Services (Radio Button must be set to Full at the top)
21.Registration
a. Registration of a device to the Audiocodes is optional. This could be helpful to keep track
of what stations are reachable in serverless setups.

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b. Enable Stand Alone Survivability (SAS) to allow local registrations. Normally, all
registration requests are forwarded to an external registrar server. By not entering an
external Registrar Server (or if one cannot be found) this box will go into Emergency
Mode.
c. Configuration > VoIP > Applications Enabling > Applications Enabling
d. Configure additional SAS related settings. For example, if unregistered stations are
allowed to place calls. SIP Registration must be done (by default) to SIP port 5080.
e. Configuration > VoIP > SAS > Configuration
f. Check Registration. SAS will send 200 OK to ANY registration attempt regardless of
password.
g. Status & Diagnostics > VoIP Status > SAS/SBC Registered Users

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22.Firmware Updating
Tip: The latest firmware guarantees the most error free operation of the device and is
always recommended.
When updating older versions of firmware be sure to observe the following upgrade path:
5.6 to 5.8 to 6.0. Once at any version 6.0 or later you can jump right to the latest 6.6.
It is important that the firmware process not be interrupted! Otherwise, you may have to
perform a BootP recovery to gain access to the device again.
a. Maintenance > Software Update > Software Upgrade Wizard
b. Locate the CMP firmware file for the device and press ‘Load File’. Then press ‘Next’
when loaded.
c. Leave the ‘Use existing configuration’ box checked unless you want the settings to retrun
to factory default. Press ‘Next’.
d. Leave the loaded CPT (Tone File) and Press ‘Next’. By default the usa_tones_13.dat file.
e. Commonly no PRT (Pre-recorded Tone) file is loaded so just press ‘Next’.
f. Commonly no UserInfo file is loaded so just press ‘Next’.
g. Arriving at the end of the wizard you must press the ‘Reset’ button to commence. It takes
approximately 120 seconds to finish. Do not Interrupt!

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23.Troubleshooting
a. Unknown IP Address programmed
1.On PC open Wireshark and start listening on Ethernet Port
2.Connect Audiocodes’Ethernet directly to PC
3.Look for GARP announcement of current IP Address. (see below)
b. Install a SIP Soft Client on your PC to place test calls to the FXO Gateway
For example: Install MicroSIP found at http://www.microsip.org.
From the SIP Client place a call to the Audiocodes Gateway.
Dial using standard SIP URI.
For example: [email protected].x
(where x.x.x.x is the IP Address of the Audiocodes device)

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c. View the Status Screen from the Audiocodes Web GUI
Found under Status & Diagnostics button at the top left.
Tip: the screen will not display anything (other than “Log is Activated”) until traffic
is received. It may take a few seconds for the screen to update.
Example Messages to look for:
oWhen a phone line is plugged/plugged into the back of the device.
oSIP INVITE signaling the start of a call from SIP Intercom or Audiocodes.
oSIP BYE signaling hook-off/end of call from SIP Intercom or Audiocodes.
d. Front Panel physical LEDs
While ringing to a telephone the Channel Status LED flashes GREEN once per 5
seconds.

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e. Syslog output for remote viewing and capturing of events and logging information
Configuration > System > Syslog Settings
oEnable Syslog = Enable
oSyslog Server IP Address = Your_PC_IP_Address
oDebug Level = 5
Install syslogViewer-setup.exe (or comparable) program on your PC.
Ensure that your Windows Firewall is not blocking Port 514!
oWhen Pause Icon ( || ) is shown new entries are allowed. Press to stop
live capture.
f. Parallel RTP stream sent to PC for capture and analysis
Configuration > Logging > Logging Settings
oDebug Recording Destination IP = Your_PC_IP_Address
oDebug Recording Status = Start
Configuration > Logging > Logging Filters Table
oClick ‘Add’ button

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oIndex = 1
oFilter Type = FXO
oValue = 1 (for the first FXO port)
oCapture Type = Signaling & Media & PCM
Capture RTP traffic with Wireshark
oExample filter: udp.port == 925
oWithout special Audiocodes Wireshark Plugins installed the traffic will not
be recognized as RTP. Therefore, you must decode it as RTP.
Additionally, there might also be other traffic types mixed in (like syslog)
as well. RTP traffic (in my test) was always a length of 242 and 1033.