Audiocodes

From Pbxnsip Wiki

Audiocodes Gateways are very robust and easy to use once you get over the initial learning. The older MP10x gateways came in 4 and 8 ports on the fxo and 2,4, 8, and 24 on the fxs. fxs gateways provide dial tone (foreign exchange station) and fxo expect a dial tone and plug into the PSTN (foreign exchange office). By default the audiocodes gateways will respond to bootp requests but not DHCP servers. There is a bootp program that comes in the box. If there are no responses to bootp requests then the fxs defaults to 10.1.10.10 and the fxo to 10.1.10.11. A good idea is not to plug in the Ethernet cable once you power it up and once the ready light is green plug in Ethernet. Once you figure out the IP address then http into the unit and the default U/P is Admin/Admin and it is case sensitive. Once you are log in do the following steps.

1) skip the quick setup and go right to Protocol Management. Then click General parameters. Set the channel select mode to cyclic ascending. This means that when a call goes from the IP side to the gateway it will grab the first available port and go up from there. The rest of the defaults should be fine in this screen.

2) Proxy and Registration tab set the enable proxy to use proxy and put the IP address of the pbxnsip server in there. The name is not important unless you make them dns resolvable so put the IP address in there. Gateway name should be the IP address of the Audiocodes gateway. If you have redundant servers you can set them but usually there is just one pbxnsip server. On fxo gateways you don't usually enable the registrations unless you wanted to two stage dial and dial an extension which would route to a port on the Audiocodes gateway. This is useful if you had a mesh of gateways and say you had one in india and if you dialed extension 111 it grabbed dial tone for the Indian PSTN and you could then dial what you would like. This is also useful if one wanted to front end a TDM pbx and let remote IP phones connect to it. So fxs gateways enable the registration but on fxo don't unless you are doing something non standard. Authentication mode is the other parameter to concern yourself with. On FXS you want to register each port and usually with unique username/passwords. On FXO usually you want one U/P for the entire box so set it to per gateway and set the u/p on this screen and make sure it matches the trunk on the pbxnsip side.

3)Coders depends on what you like. pbxnsip supports G.711, GSM, G.726.

4) DTMF and Dialing - usually increase the maximum digits from 3 and enable DTMF RFC 2833 Negotiation.

5) Advanced General Parameters Figuring out how the remote disconnect happens on an analog switch is always a challenge. It is usually either current disconnect or reverse polarity. You can't set both at the same time so you have to either ask the telco (ha ha) or try one and then the other. Call in from a remote phone hang up and make sure the line frees up. Set the debug level to 6 here so when you are installing the system you can see verbose messages in the logfile.

6) Supplementary services make sure you enable caller ID and bellcore is fine.

7) Endpoint phone numbers are the next important tab. If you want to use a port on the gateway by entering the number here will "enable" it. Usually you will put the DID number in here the telco gave you. However if no caller ID is present this is the number that will show up.

8) Endpoint settings

   Authentication is used mostly on fxs ports for the u/p that is in the challenge response. On FXO you typically don't set this.  
   Automatic dialing is very important on fxo ports since when the call comes in you want it   automatically to dial the pbxnsip system and usually go to an Auto Attendant extension that is configured there.  So enable it and then put the extension number of the AA which in our default config is 70.  If you want to ring an extension directly then put that extension number in there.  
   Leave the caller ID blank since this is what will go in there if nothing is presented but you want to set the next field.
   Detect caller ID from Telco.  Enable this and you need to set how long to wait for detecting caller ID which usually comes between the 1st and 2nd ring on analog lines.  

9) FXO settings

  Dialing mode change it to one stage since two stage will give you a dial tone when you call it and not forward the call along.  This is for IP to Tel direction.  Set the rings for detecting caller ID to 1 and if you don't see caller ID then change it to 2.  The other defaults are fine. 

10) This should be enought to get a call into the system. At this point go into the pbxnsip server into the domain trunk settings and create a SIP Gateway trunk, name it Audiocodes and put the IP address of the Audiocodes gateway in the domain settings. If you set a U/P put it in the trunk as well. In the extension field of the trunk put the extension that you want to take the call. Usually the AA.

Installation is Pretty simple. My existing config file is as below: (User Parameters) to be edited as per requirements

Simply log in to the gatway, and uplod the config file after making proper changes from the browser.

;**************
;** Ini File **
;**************

;Board: MP-108 FXS
;Serial Number: 378322
;Slot Number: 0
;Software Version: 4.20.354.571
;Board IP Address: 10.10.10.11
;Board Subnet Mask: 255.255.255.0
;Board Default Gateway: 10.10.10.250
;------------------------------

[SYSTEM Params]

DNSPRISERVERIP = x.x.x.x (DNS Server)
SNMPMANAGERIP = x.x.x.x (SNMP Manager, Optional)
SNMPMANAGERTABLEIP_0 = x.x.x.x (SNMP TRAP, Optional)

[Voice Engine Params]

CALLERIDTYPE = 1
RFC2833PAYLOADTYPE = 96

[SIP Params]

MAXDIGITS = 7 (Maximum number of digits, after this the gateway will not accept any further digits)
NUMBEROFWAITINGINDICATIONS = 2 (Call waiting indications)
ALWAYSUSEROUTETABLE = 0
ENABLECALLWAITING = 0 (Enable call waiting, 0=off, 1=on)
TIMEBETWEENWAITINGINDICATIONS = 10 (delay between call waiting indications)
TIMEBETWEENDIGITS = 4 (Interdigit timeout)
TIMEFORDIALTONE = 16
REMOVEPREFIX = 0
OUTOFBANDDTMFFORMAT = 2
PLAYRBTONE2IP = 1
ISFALLBACKUSED = 0 (Use internal call routing tableas fallback)
ISHOOKFLASHUSED = 0 (Allow hook flash for hold/transfer etc)
REGISTRATIONTIME = 600 (Max registration in seconds)
SIPT1RTX = 500
SIPT2RTX = 4000
ISPROXYUSED = 1 (Use Proxy for calls)
ALWAYSSENDTOPROXY = 0 (Always send calls to proxy, 0=off, 1=on, use 1 for NAT Traversal)
ISPROXYHOTSWAP = 0
PROXYHOTSWAPRTX = 3
SIPMAXRTX = 7
ISREGISTERNEEDED = 1 (Registration required)
AUTHENTICATIONMODE = 2
ADDTRUNKGROUPASPREFIX = 0
ADDPORTASPREFIX = 0
SIPDESTINATIONPORT = 5060
PLAYRBTONE2TEL = 1
ISFAXUSED = 0
ROUTEMODEIP2TEL = 0
ROUTEMODETEL2IP = 0
ENABLECALLERID = 1
ENABLECURRENTDISCONNECT = 0
ENABLEDIDWINK = 0
ENABLEREVERSALPOLARITY = 1
SECURECALLSFROMIP = 0
ENABLEBUSYOUT = 0
ENABLEHOLD = 0
ENABLETRANSFER = 1
ENABLEFORWARD = 0
CDRREPORTLEVEL = 1
CHANNELSELECTMODE = 0
REGRETTIME = 0
GWDEBUGLEVEL = 0
PROXYKEEPALIVETIME = 60
PROXYREDUNDANCYMODE = 0
ENABLEPROXYKEEPALIVE = 1
ENABLERPIHEADER = 1
ISPRACKREQUIRED = 1
PRACKMODE = 1
ENABLEEARLYMEDIA = 1
ISUSERPHONE = 1
ISINFOUSED4DTMF = 0
SIPSESSIONEXPIRES = 60
PROXYNAME = '10.10.10.1' (Replace with PBXnSIP IP)
REGISTRARIP = 10.10.10.1 (Replace with PBXnSIP IP)
SIPGATEWAYNAME = '10.10.10.1' (Replace with PBXnSIP IP)
USERNAME = 
CNONCE = 'Default_Cnonce'
PASSWORD = 'Default_Passwd'
ALTROUTINGTEL2IPENABLE = 0
ALTROUTINGTEL2IPMODE = 0
CODERNAME = g711Ulaw64k,20
CODERNAME = g711Alaw64k,20
CODERNAME = g729,20
CODERNAME = g7231,30
CALLERDISPLAYINFO0 = Port-1 (CallerID for port 1 to 8)
CALLERDISPLAYINFO1 = port2
CALLERDISPLAYINFO2 = port3
CALLERDISPLAYINFO3 = port4
CALLERDISPLAYINFO4 = port5
CALLERDISPLAYINFO5 = port6
CALLERDISPLAYINFO6 = port7
CALLERDISPLAYINFO7 = port8
TRUNKGROUP = 1-8,111 (Replace 111 with your phone number, eg 111 to 118 is defined here)
PROXYIP = 10.10.10.1
AUTHENTICATION_0 = 111,111 (Username,password for port 1 to 8)
AUTHENTICATION_1 = 112,112
AUTHENTICATION_2 = 113,113
AUTHENTICATION_3 = 114,114
AUTHENTICATION_4 = 115,115
AUTHENTICATION_5 = 116,116
AUTHENTICATION_6 = 117,117
AUTHENTICATION_7 = 118,118

[IPsec Params]

--Vitrag 03:22, 12 February 2006 (PST)