AudioCodes MediaPack 114 and 118 SIP Gateway with Asterisk

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THIS APPLIES TO FXO ONLY IF YOU HAVE FXS CHANNEL ISSUES IT IS NOT ADDRESSED IN THIS DOCUMENTATION

If you are having problems setting up these products then you are not alone. Here is a List of issues that I've run into and How I solved these issues.

1. Getting the darn thing to work at all.
2. Getting CallerID to work correctly
3. Echo and Static issues
4. Volume issues
5. Call Pickup Issues
6. Support Issues

First of all make sure you have the documentation open and handy as you read this you can reference the commands I will give you in there and get more in depth information.

LINK IS BROKEN (leaving it in just in case someone finds it again)
http://www.audiocodes.com/asp/DisplayFoldersFiles2.asp?FolderID=1135
Click on the full Users Manual

The First Issue I had was solved by making the Gateway work as a whole. I tried doing all the things I did previously with other gateways like the Grandstream and the Clippcomm (btw never get a Clippcomm if you want a working product) to no avail. I did everything I though was right so I ended up finding a Config file as a starting point to move forward. Which you may need so I will give you it here.

Example Config File:
;**************
;** Ini File **
;**************

;;Board[:] MP-114 FXO
;;Serial Number[:] 875328
;;Slot Number[:] 1
;;Software Version[:] 4.80A.025.004
;;Board IP Address[:] 192.168.6.10
;;Board Subnet Mask[:] 255.255.255.0
;;Board Default Gateway[:] 192.168.6.1
;;Ram size[:] 32M   Flash size[:] 8M 
;;Num DSPs[:] 1  Num DSP channels[:] 4
;;Profile[:] NONE 
;------------------------------


[SYSTEM Params]

SyslogServerIP = 10.1.1.89

[BSP Params]

PCMLawSelect = 3
LocalOAMIPAddress = 192.168.6.10
RoutingTableHopsCountColumn = 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0

[ATM Params]


[Analog Params]

FarEndDisconnectSilenceMethod = 0
CallProgressTonesFilename = 'usa_tones_12.dat'

[ControlProtocols Params]


[MGCP Params]


[MEGACO Params]

EP_Num_0 = 0
EP_Num_1 = 1
EP_Num_2 = 0
EP_Num_3 = 0
EP_Num_4 = 0

[SS7 Params]


[Voice Engine Params]

IdlePCMPattern = 85
ECDCremoval = 1
ECNLPMode = 0
ECNlpSensitivity = 1
EchoCancellerAggressiveNLP = 1
InputGainLocation = 1
InputGain = 0
VoiceVolume = 4

[WEB Params]

LogoWidth = '339'

[SIP Params]

ENABLECALLERID = 1
MAXDIGITS = 14
TIMEBETWEENDIGITS = 5
REGISTRATIONTIME = 3600
ISPROXYUSED = 1
ISREGISTERNEEDED = 1
AUTHENTICATIONMODE = 1
ISTWOSTAGEDIAL = 0
CDRREPORTLEVEL = 1
GWDEBUGLEVEL = 5
PROXYNAME = '192.168.6.5'
SIPGATEWAYNAME = 'audiocodes.com'
USERNAME = 'mp_fxo'
CNONCE = '0a123bcf'
PASSWORD = '***'
PROGRESSINDICATOR2IP = 1
ISFAXUSED = 1
CODERNAME = g711Ulaw64k,20,0,$$,0
CODERNAME = g729,20,0,$$,0
PREFIX = 10,192.168.6.5,*,0,255
PREFIX = 20,192.168.6.5,*,0,255
PREFIX = *,192.165.6.5,*,0,255
PSTNPREFIX = *,1,*,*,0
TARGETOFCHANNEL0 = 600,1
TARGETOFCHANNEL1 = 600,1
TARGETOFCHANNEL2 = 600,1
TARGETOFCHANNEL3 = 600,1
TRUNKGROUP_1 = 1-4,201,0
PROXYIP = 192.168.6.5
AUTHENTICATION_0 = mp_fxo,250
AUTHENTICATION_1 = mp_fxo,250
AUTHENTICATION_2 = mp_fxo,250
AUTHENTICATION_3 = mp_fxo,250
TRUNKGROUPSETTINGS = 1,1
TRUNKGROUPSETTINGS = 2,1
TRUNKGROUPSETTINGS = 3,1
TXDTMFOPTION = 4
;;TelProfile[:] ProfileName, Preference, CodersGroupID, IsFaxUsed, DJBufMinDelay, JBufOptFactor, IPDiffServ, SigIPDiffServ, DtmfVolume, InputGain, VoiceVolume, EnableReversePolarity, EnableCurrentDisconnect, EnableDigitDelivery, ECE, MWIAnalog, MWIDisplay, FlashHookPeriod, EnableEarlyMedia, ProgressIndicator2IP
TELPROFILE_1 = Default Tel Profile,1,0,1,70,7,46,46,-11,0,1,1,0,0,1,0,0,400,0,1
ENABLECALLERID_0 = 1
ENABLECALLERID_1 = 1
ENABLECALLERID_2 = 1
ENABLECALLERID_3 = 1

[VXML Params]


[IPsec Params]


[Audio Staging Params]


[PSTN-SDH Params]





Okay. So what I am going to focus on are two areas. Voice Engine Params and Sip Params

Those are the two pertinent areas. There are several How-To's out there that tell you to turn off autodial. Not sure why because thats what ended up getting everything to work for me in the end.

Under Sip params I want you to notice a few settings. The Username and password and the ProxyIP.

Image


Those are the settings that will make this device register with your asterisk server. If you put those in correctly then you should setup this context in your
SIP.CONF file

[mp_fxo]
type=friend
username=mp_fxo
host=dynamic
secret=***
context=inbound
allow=all
qualify=no
canreinvite=no
insecure=very
nat=yes


As well you must make sure that using this that you are registering "Per Gateway" instead of "Per Endpoint" The setting is located under Protocol Management on the left and then under Protocol Definition / Proxy & Registration

Image


This should get you up and running on the asterisk side of things.

2.) CALLERID
You may notice however that your Caller ID does not work. You can change that setting in the INI file under the SIP Params

ENABLECALLERID = 1

There seems to be other places to set Caller ID under the menu. But I cannot hash out how those settings for each Channel do anything different than setting this one setting in the INI file

just as a side note there is an undocumented page you can go to to change these settings without reloading the INI file each time

http://192.168.6.10/AdminPage will give you something that looks like this

Image


Doesnt Look like much but if you click on ini Parameters You get this

Image


If you've uploaded your config file a few hundred times i'm sure you see the value of this page.


3.) STATIC and ECHO
Moving along to Static/Echo issues. This was my biggest issue the start off with. Probably because the users of the Voip Phones heard it not the person on the other end. Regardless this issue was strange to me and I admit I was ready to throw the thing out the window and get a Grandstream that drops a call every now and again. Much easier to deal with. On the other hand I love a challenge.

So that lead me to call AudioCodes. First of all I wouldnt suggest this because they want you to pay an arm and 3 legs to get support but I was able to sneak in the mix somehow. And got the support I needed from them. lol

Here are the settings that effect static/echo..


ECDCremoval = 1
ECNLPMode = 0
ECNlpSensitivity = 1
EchoCancellerAggressiveNLP = 1
InputGainLocation = 1
InputGain = 0 ;; or 1 if you want to play with it.
DJBufMinDelay = 10
ECHybridLoss = -6db
Voicevolume = 4

These settings should be put under Voice Engine Params. The very first ECDCremoval = 1. Seemed to be the one key setting for the echo cancellation

It first seemed to me the the Static Issue was the Same as the Echo Issue because it would either be one or the other. The static seemed to be the echo being cancelled badly!. If you put some of these settings in and do not see them in your ini later don't stress it. Its because if it was the default setting it won't keep in the ini.. but you can see them on the AdminPage mentioned earlier.

The InputGainLocation seems to be where you want the gain to happen before or after echo cancellation.

4.) Volume Issues.

You may notice above InputGain and Voicevolume.
Voicevolume= what the person on the far end hears
Inputgain = what the person using the gateway hears

Inputgain has the most effect on echo and such but if you start getting higher up in Voicevolume you can start hearing some issues as well. My suggestion would be to start Voicevolume=7 and Inputgain=1 and move Voicevolume down if you get some weird feedback. Untill it goes away and Inputgain ... only go 1 or 0. I have one of my sites at 0 and one at 1. It really depends I think on the lines and dc voltage at the location in question.

5.) Call Pickup Issues
This was a problem for me only because I had my MaxDigits set to 7 and thats way too low but its worth mentioning that if your MaxDigits is set too low then when you place a call. It may never pickup for you. It could just keep ringing .. even though the party on the other end has already answered. You should probably set MaxDigits=14 or something like that.

6.) Support
This device does seem to me to be the best out there. It takes a lot to setup and it the Toughest out of 4 brands of Gateways that I have ever setup.

AudioCodes will not support you unless you pay them directly. They sell it to vendors and expect these vendors to support you. I have found there are very few who will support well. That is and still get it for a decent price. So If you are looking for support look no further than forums.

Thanks
you can email me at schapman1974@gmail.com if you have any questions.


38181 Views
THIS APPLIES TO FXO ONLY IF YOU HAVE FXS CHANNEL ISSUES IT IS NOT ADDRESSED IN THIS DOCUMENTATION

If you are having problems setting up these products then you are not alone. Here is a List of issues that I've run into and How I solved these issues.

1. Getting the darn thing to work at all.
2. Getting CallerID to work correctly
3. Echo and Static issues
4. Volume issues
5. Call Pickup Issues
6. Support Issues

First of all make sure you have the documentation open and handy as you read this you can reference the commands I will give you in there and get more in depth information.

LINK IS BROKEN (leaving it in just in case someone finds it again)
http://www.audiocodes.com/asp/DisplayFoldersFiles2.asp?FolderID=1135
Click on the full Users Manual

The First Issue I had was solved by making the Gateway work as a whole. I tried doing all the things I did previously with other gateways like the Grandstream and the Clippcomm (btw never get a Clippcomm if you want a working product) to no avail. I did everything I though was right so I ended up finding a Config file as a starting point to move forward. Which you may need so I will give you it here.

Example Config File:
;**************
;** Ini File **
;**************

;;Board[:] MP-114 FXO
;;Serial Number[:] 875328
;;Slot Number[:] 1
;;Software Version[:] 4.80A.025.004
;;Board IP Address[:] 192.168.6.10
;;Board Subnet Mask[:] 255.255.255.0
;;Board Default Gateway[:] 192.168.6.1
;;Ram size[:] 32M   Flash size[:] 8M 
;;Num DSPs[:] 1  Num DSP channels[:] 4
;;Profile[:] NONE 
;------------------------------


[SYSTEM Params]

SyslogServerIP = 10.1.1.89

[BSP Params]

PCMLawSelect = 3
LocalOAMIPAddress = 192.168.6.10
RoutingTableHopsCountColumn = 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0

[ATM Params]


[Analog Params]

FarEndDisconnectSilenceMethod = 0
CallProgressTonesFilename = 'usa_tones_12.dat'

[ControlProtocols Params]


[MGCP Params]


[MEGACO Params]

EP_Num_0 = 0
EP_Num_1 = 1
EP_Num_2 = 0
EP_Num_3 = 0
EP_Num_4 = 0

[SS7 Params]


[Voice Engine Params]

IdlePCMPattern = 85
ECDCremoval = 1
ECNLPMode = 0
ECNlpSensitivity = 1
EchoCancellerAggressiveNLP = 1
InputGainLocation = 1
InputGain = 0
VoiceVolume = 4

[WEB Params]

LogoWidth = '339'

[SIP Params]

ENABLECALLERID = 1
MAXDIGITS = 14
TIMEBETWEENDIGITS = 5
REGISTRATIONTIME = 3600
ISPROXYUSED = 1
ISREGISTERNEEDED = 1
AUTHENTICATIONMODE = 1
ISTWOSTAGEDIAL = 0
CDRREPORTLEVEL = 1
GWDEBUGLEVEL = 5
PROXYNAME = '192.168.6.5'
SIPGATEWAYNAME = 'audiocodes.com'
USERNAME = 'mp_fxo'
CNONCE = '0a123bcf'
PASSWORD = '***'
PROGRESSINDICATOR2IP = 1
ISFAXUSED = 1
CODERNAME = g711Ulaw64k,20,0,$$,0
CODERNAME = g729,20,0,$$,0
PREFIX = 10,192.168.6.5,*,0,255
PREFIX = 20,192.168.6.5,*,0,255
PREFIX = *,192.165.6.5,*,0,255
PSTNPREFIX = *,1,*,*,0
TARGETOFCHANNEL0 = 600,1
TARGETOFCHANNEL1 = 600,1
TARGETOFCHANNEL2 = 600,1
TARGETOFCHANNEL3 = 600,1
TRUNKGROUP_1 = 1-4,201,0
PROXYIP = 192.168.6.5
AUTHENTICATION_0 = mp_fxo,250
AUTHENTICATION_1 = mp_fxo,250
AUTHENTICATION_2 = mp_fxo,250
AUTHENTICATION_3 = mp_fxo,250
TRUNKGROUPSETTINGS = 1,1
TRUNKGROUPSETTINGS = 2,1
TRUNKGROUPSETTINGS = 3,1
TXDTMFOPTION = 4
;;TelProfile[:] ProfileName, Preference, CodersGroupID, IsFaxUsed, DJBufMinDelay, JBufOptFactor, IPDiffServ, SigIPDiffServ, DtmfVolume, InputGain, VoiceVolume, EnableReversePolarity, EnableCurrentDisconnect, EnableDigitDelivery, ECE, MWIAnalog, MWIDisplay, FlashHookPeriod, EnableEarlyMedia, ProgressIndicator2IP
TELPROFILE_1 = Default Tel Profile,1,0,1,70,7,46,46,-11,0,1,1,0,0,1,0,0,400,0,1
ENABLECALLERID_0 = 1
ENABLECALLERID_1 = 1
ENABLECALLERID_2 = 1
ENABLECALLERID_3 = 1

[VXML Params]


[IPsec Params]


[Audio Staging Params]


[PSTN-SDH Params]





Okay. So what I am going to focus on are two areas. Voice Engine Params and Sip Params

Those are the two pertinent areas. There are several How-To's out there that tell you to turn off autodial. Not sure why because thats what ended up getting everything to work for me in the end.

Under Sip params I want you to notice a few settings. The Username and password and the ProxyIP.

Image


Those are the settings that will make this device register with your asterisk server. If you put those in correctly then you should setup this context in your
SIP.CONF file

[mp_fxo]
type=friend
username=mp_fxo
host=dynamic
secret=***
context=inbound
allow=all
qualify=no
canreinvite=no
insecure=very
nat=yes


As well you must make sure that using this that you are registering "Per Gateway" instead of "Per Endpoint" The setting is located under Protocol Management on the left and then under Protocol Definition / Proxy & Registration

Image


This should get you up and running on the asterisk side of things.

2.) CALLERID
You may notice however that your Caller ID does not work. You can change that setting in the INI file under the SIP Params

ENABLECALLERID = 1

There seems to be other places to set Caller ID under the menu. But I cannot hash out how those settings for each Channel do anything different than setting this one setting in the INI file

just as a side note there is an undocumented page you can go to to change these settings without reloading the INI file each time

http://192.168.6.10/AdminPage will give you something that looks like this

Image


Doesnt Look like much but if you click on ini Parameters You get this

Image


If you've uploaded your config file a few hundred times i'm sure you see the value of this page.


3.) STATIC and ECHO
Moving along to Static/Echo issues. This was my biggest issue the start off with. Probably because the users of the Voip Phones heard it not the person on the other end. Regardless this issue was strange to me and I admit I was ready to throw the thing out the window and get a Grandstream that drops a call every now and again. Much easier to deal with. On the other hand I love a challenge.

So that lead me to call AudioCodes. First of all I wouldnt suggest this because they want you to pay an arm and 3 legs to get support but I was able to sneak in the mix somehow. And got the support I needed from them. lol

Here are the settings that effect static/echo..


ECDCremoval = 1
ECNLPMode = 0
ECNlpSensitivity = 1
EchoCancellerAggressiveNLP = 1
InputGainLocation = 1
InputGain = 0 ;; or 1 if you want to play with it.
DJBufMinDelay = 10
ECHybridLoss = -6db
Voicevolume = 4

These settings should be put under Voice Engine Params. The very first ECDCremoval = 1. Seemed to be the one key setting for the echo cancellation

It first seemed to me the the Static Issue was the Same as the Echo Issue because it would either be one or the other. The static seemed to be the echo being cancelled badly!. If you put some of these settings in and do not see them in your ini later don't stress it. Its because if it was the default setting it won't keep in the ini.. but you can see them on the AdminPage mentioned earlier.

The InputGainLocation seems to be where you want the gain to happen before or after echo cancellation.

4.) Volume Issues.

You may notice above InputGain and Voicevolume.
Voicevolume= what the person on the far end hears
Inputgain = what the person using the gateway hears

Inputgain has the most effect on echo and such but if you start getting higher up in Voicevolume you can start hearing some issues as well. My suggestion would be to start Voicevolume=7 and Inputgain=1 and move Voicevolume down if you get some weird feedback. Untill it goes away and Inputgain ... only go 1 or 0. I have one of my sites at 0 and one at 1. It really depends I think on the lines and dc voltage at the location in question.

5.) Call Pickup Issues
This was a problem for me only because I had my MaxDigits set to 7 and thats way too low but its worth mentioning that if your MaxDigits is set too low then when you place a call. It may never pickup for you. It could just keep ringing .. even though the party on the other end has already answered. You should probably set MaxDigits=14 or something like that.

6.) Support
This device does seem to me to be the best out there. It takes a lot to setup and it the Toughest out of 4 brands of Gateways that I have ever setup.

AudioCodes will not support you unless you pay them directly. They sell it to vendors and expect these vendors to support you. I have found there are very few who will support well. That is and still get it for a decent price. So If you are looking for support look no further than forums.

Thanks
you can email me at schapman1974@gmail.com if you have any questions.


38181 Views
Created by: schapman, Last modification: Thu 19 of Mar, 2015 (02:36 UTC)
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