An Overview of RADIUS on the IMG

Size: px
Start display at page:

Download "An Overview of RADIUS on the IMG"

Transcription

1 An Overview of RADIUS on the IMG The IMG uses Remote Authentication Dial In User Service (RADIUS) protocol for streaming the Call Detail Records (CDR). The implementation is compliant with RFC 2865 and RFC The RADIUS messages are sent to external RADIUS servers. The IMG RADIUS interface generates an ACCESS, a START & a STOP Request for the inbound leg and a START & STOP Request for the outbound leg of the call, as well as data associated with the INVITE, the 200 OK, the BYE and the CANCEL methods for those legs utilizing a SIP protocol. Specifications The IMG implementation of RADIUS is based on the following RADIUS RFCs: RFC Remote Authentication Dial-In User Service (RADIUS) RFC RADIUS Accounting Formats The IMG 1010 supports the Cantata RADIUS formats, which Includes some attributes defined by RFC 2865 and RFC 2866, as well as Cantata Vendor Specific Attributes (VSA). Scenarios The IMG 1010 supports RADIUS Authentication and Accounting. IMG 1010 customer has the option of using one of the following scenarios: Authentication and Accounting In this case an Authentication Server and an Accounting Server are both assigned to the RADIUS client on the IMG. Accounting only In this case only an Accounting Server is assigned to the RADIUS client on the IMG. Authentication only In this case only an Authentication Server is assigned to the RADIUS client on the IMG See RADIUS Scenarios for more details. As per RFC 2865 and RFC 2866, the IMG 1010 uses port 1812 for Authentication and port 1813 for Accounting by default. But these ports are also configurable. The Authentication and Accounting servers could be the same entity, in which case both servers will have the same IP address. Or they could be different entities with different IP addresses. The RADIUS attributes and VSA s included in the messages will vary based on the protocol for used for a specific side of the call, depending on whether it is a TDM protocol (SS7 or ISDN) or IP protocol (SIP or H.323). The User name and Password values configured for the Authentication Server used will be included in the user name and password attributes in the Access Request message sent from the IMG. 1

2 RADIUS RADIUS Server Redundancy The IMG 1010 supports an Active Standby redundancy scheme. Redundancy logic is independent for Authentication and Accounting Servers. When configuring RADIUS servers you may create them with an initial priority preference. The IMG will begin using the preferred Server(s) and switchover to an alternate server after detecting a communication failure to the currently active server. Once the switchover occurs all future Radius messages will flow to the newly active server until a failure occurs on this server. If an error is detected in trying to send a Radius message to this newly active server, the IMG will attempt to switch again back to the previously active server. This behaviour is repeated, until a working server is detected. If the IMG fails to connect to a RADIUS Server an alarm will be sent. You can monitor alarms using EventView. Typically when a RADIUS message needs to be sent to a server, it is assembled and passed to the OS for transport to the currently active server. These servers are configured to send the message, wait 2 seconds and then retry sending the message an additional 3 times. Therefor a RADIUS message will be sent a total of 4 times, with 2 second intervals, before attempting a switchover to the next server, if one is configured. The switchover behaviour is coupled to the message type. Therefore an Accounting Server switchover is independent of an Authentication Server switchover. Under typical call load it will take a while for the switchover to complete since the IMG may have many RADIUS messages queued up to the failed server. Each of these messages must fail and be retried on the newly active server following notification of the send failure. NOTE: A negative response does not constitute a server failure. Supported Packet Types Access-Request Sent to a RADIUS server - conveys information used to determine whether a user is allowed access to a specific NAS, and any special services requested for that user. Access-Accept Sent by the RADIUS server - provides specific configuration information necessary to begin delivery of service to the user. Access-Reject Sent by the RADIUS Server if any value of the received Attributes is not acceptable Accounting-start Accounting-stop RADIUS Server Debug Mode 2

3 An Overview of RADIUS on the IMG You can configure your RADIUS Client in Debug Mode so that calls will be completed whether the RADIUS server is active or not. The IMG will not require authentication for the RADIUS server to complete a call and no billing information will be logged. You enable RADIUS Debug Mode using the RADIUS Client screen. RADIUS for Pre-Paid Applications The IMG now accepts and acts upon data received in RADIUS Authentication Response messages that the Radius Server may send pertaining to prepaid application. This will allow the IMG1010 to be used in a prepaid application environment. You enable this feature in the RADIUS Client pane. NOTE: Radius Prepaid Support Mode will be disabled if Radius Debug Mode is enabled. The two modes cannot be enabled at the same time. The IMG sends the following VSAs to the RADIUS Server (NOTE: These apply to ER2 release and above only): Cantata-call-type = Cantata-voip-dst-sig-ip-in = Cantata-trunk-grp-in = and receives the following attributes in return: Call Duration Number of seconds for which the call is authorized. Call Tracing: Will log the termination [Normal Call Clearing (16)] as a result of the max duration of the call being exceeded. VSA: h323-credit-time (VSA # 102) New Dialed Number Phone number to which the call is redirected; for example, to a toll-free number or a customer service. Call Tracing: Will log the use of the new Dialed Number, which overrides the Dialed number received in the incoming call. VSA: h323-redirect-number (VSA # 106) See RADIUS CDR Example: Pre-paid Support RADIUS Server Failure Alarm The IMG provides automatic alarming notification to IMG users when a Radius Server has changed states and can no longer be accessed. The alarm, reported in 3

4 RADIUS EventView, will include the RADIUS Server Type (Access, Accounting), the Server ID, the mode of the Radius Server (normal, debug), the state of the Radius Server and the IP address. Related Topics Basic RADIUS Call Flow Generic RADIUS Attributes Cantata RADIUS VSAs RADIUS Call Flow: SS7 to H.323 RADIUS CDR Example: SIP-to-ISDN Configuring Billing and Authentication 4

5 RADIUS Scenarios The IMG 1010 supports RADIUS Authentication and Accounting. IMG 1010 customer has the option of using one of the following scenarios: Authentication and Accounting In this case an Authentication Server and an Accounting Server are both assigned to the RADIUS client on the IMG. Accounting only In this case only an Accounting Server is assigned to the RADIUS client on the IMG. Authentication only In this case only an Authentication Server is assigned to the RADIUS client on the IMG

6

7 Generic RADIUS Attributes RADIUS Attributes carry the specific authentication, authorization, information and configuration details for the request and reply. Some Attributes may be included more than once. IETF Attribute # Attribute Name Description Values Example 1 User-Name Account number or calling party number String User- Password 16 octets user password String cantata 4 NAS-IP- Address IP Address of the requesting IMG Numeric (4 octets) NAS-Port The Physical Port Number of the NAS (Network Access Server) that is authenticating the user. Numeric (4 octets) Service- Type The Type of Service the user has requested, or the type of service to be provided Numeric Values Login-User 14 Login-IP- Host This Attribute indicates the system with which to connect the user, when the Login- Service Attribute is included. It MAY be used in Access- Numeric (4 octets) Values

8 RADIUS Accept packets. It MAY be used in an Access- Request packet as a hint to the server that the NAS would prefer to use that host, but the server is not required to honor the hint. 29 Termination- Action 0 Default 1 RADIUS- Request RADIUS-Request 30 Called- Station-Id This Attribute allows the NAS to send in the Access- Request packet the phone number that the user called, using Dialed Number Identification (DNIS) or similar technology. String The String field is one or more octets, containing the phone number that the user's call came in on Note that this may be different from the phone number the call comes in on. It is only used in Access- Request packets. 31 Calling- Station-Id This Attribute allows the NAS to send String The String

9 Generic RADIUS Attributes in the Access- Request packet the phone number that the call came from, using Automatic Number Identification (ANI) or similar technology. It is only used in Access- Request packets. field is one or more octets, containing the phone number that the user placed the call from. 32 NAS- Identifier This Attribute contains a string identifying the NAS originating the Access- Request. It is only used in Access- Request packets. String The String field is one or more octets, and should be unique to the NAS within the scope of the RADIUS server. For example, a fully qualified domain name would be suitable as a NAS- Identifier. 40 Acct-Status- Type Indicates whether this Accounting- Request marks the beginning of the user service (Start) or the Numeric (4 octets) Values Start 9

10 RADIUS end (Stop). 41 Acct-Delay- Time This attribute indicates how many seconds the client has been trying to send this record for, and can be subtracted from the time of arrival on the server to find the approximate time of the event generating this Accounting- Request. (Network transit time is ignored.) Numeric (4 octets) 0 42 Acct-Input- Octets Indicates how many octets have been received from the port over the course of this service being provided, and can only be present in Accounting- Request records where the Acct- Status-Type is set to Stop. Numeric (4 octets) 1 43 Acct- Output- Octets indicates how many octets have been sent to the port in the course of delivering this service, and can only be present in Numeric (4 octets) 1 10

11 Generic RADIUS Attributes Accounting- Request records where the Acct- Status-Type is set to Stop. 44 Acct- Session-ID This attribute is a unique Accounting ID to make it easy to match start and stop records in a log file. String The String field SHOULD be a string of UTF-8 encoded [7] characters c0405b e48 46 Acct- Session- Time This attribute indicates how many seconds the user has received service for, and can only be present in Accounting- Request records where the Acct- Status-Type is set to Stop. Numeric (4 octets) Acct-Input- Packets This attribute indicates how many packets have been received from the port over the course of this service being provided to a Framed User, and can only be present in Accounting- Request records where the Acct- Status-Type is set to Stop. Numeric (4 octets) 1 48 Acct- This attribute Numeric 0 11

12 RADIUS Output- Packets indicates how many packets have been sent to the port in the course of delivering this service to a Framed User, and can only be present in Accounting- Request records where the Acct- Status-Type is set to Stop. (4 octets) 49 Acct- Terminate- Cause This attribute indicates how the session was terminated, and can only be present in Accounting- Request records where the Acct- Status-Type is set to Stop. Values NAS-Request 60 Chap- Challenge This Attribute contains the CHAP Challenge sent by the NAS to a PPP Challenge- Handshake Authentication Protocol (CHAP) user. It is only used in Access- Request packets. String The String field contains the CHAP Challenge. 61 NAS-Port- Type This Attribute indicates the type of the physical port of the NAS Values Ethernet 12

13 Generic RADIUS Attributes which is authenticating the user. 13

14

15 Cantata VSAs Cantata RADIUS Vendor Code: 2754 IETF Attribute # for all VSAs: 26 NOTE: As of ER2, the word Cantata appears at the beginning of all Cantata VSA names. Attribute Name VSA # Description Value Format Example Cantataani-posttranslate Cantataani-pretranslate 42 Calling number to be sent out of the IMG. 40 Incoming Automatic Number Identification String String Cantatacalldirection 43 The direction of the call. String Calling Party Called Party 141 Value of the Call-ID header. String Syntax is as per RFC 3261 "SIP:Session Initiation protocol" Cantatacall-id Cat- 0@ Cantata- Call-Origin 26 Gateway's behavior in relation to the connection that is active for this leg. answer = Legs 1 and 3 originate = Legs 2 and 4 callback = Legs 1 and 3 originate Cantata- Call-Type 27 Protocol type or family used on this leg of the call. Telephony VOIP VOFR h323 Call Type = SS7 For example, answer on a leg 1; originate on a leg 2; callback on leg 1. Cantataconnecttime 28 Connect time in Network Time Protocol (NTP) format: hour, minutes, seconds, hh:mm:ss:mmm ZON DDD MMM ## YYYY 12:30: EST Fri Mar

16 RADIUS microseconds, time_zone, day, month, day_of_month, and year. Cantatacredit-time * (NOTE: In ER1, this VSA was named h.323- credittime) 102 Number of seconds for which the call is authorized Integer in decimal notation Valid Range: sec = unlimited seconds 3200 Cantatadisconnecttime 29 Disconnect time in NTP format: hour, minutes, seconds, microseconds, time_zone, day, month, day_of_month, year. hh:mm:ss:mmm ZON DDD MMM ## YYYY 12:30: EST Fri Mar Cantatadnis-posttranslate Cantatadnis-pretranslate 41 Called number to be sent out of the IMG. 39 Incoming Dialed Number Identification Service String String Cantatah323-confid 24 Unique call identifier generated by the gateway. Used to identify the separate billable events (calls) within a single calling session 16-byte number in hexadecimal notation with one space between each 4-byte integer A69E11D6 808D87CA 50D5A43C Cantatah323-gwid 33 Domain name server (DNS) name or local name of the Character string boston.cantata.com 16

17 Cantata RADIUS VSAs voice gateway that is sending the VSA Cantata- h323- incomingconf-id 35 Unique number for identifying a calling session on a gateway, where a session is closed when the calling party hangs up 16-byte number in hexadecimal notation with one space between each 4-byte integer A69E11D6 808D87CA 50D5A43C Cantata- h.323- redirectnumber * 106 Phone number to which the call is redirected; for example, to a toll-free number or a customer service E.164 format (decimal digits with no spacing characters) Max # of digits is Cantataincomingreq-uri 146 For inbound Radius mess. both Start & Stop. Access to the value after the RFS on the inbound side string. Syntax is as per RFC 3261"SIP:Session Initiation protocol" Cantatalostpackets 47 Number of lost voice packets during the call. Unsigned integer 0 Cantatamedia-dstrtp-ip 37 Remote media gateway IP address. Numerals in dotted decimal notation: nnn.nnn.nnn.nnn Cantatanext-hopdn 149 The Domain Name (DN) or Fully Qualified Domain Name (FQDN) where the request is forwarded. string FQDN[:port][/protocol] Where FQDN is a host, domain name or dotted IP address. next-hop-dn=company.com Cantatanext-hopip 148 Next-hop IP address where the request is Numerals in dotted decimal notation: nnn.nnn.nnn.nnn

18 RADIUS forwarded. Cantataoutgoingreq-uri 147 For outbound Radius mess.both Start & Stop Access to the value after outseize of the Invite string. Syntax is asper RFC 3261"SIP:SessionInitiation protocol" outgoing-req-uri = "OUTGOING SIP URI(in) = 145 Previous hop IP address, as seen by the proxy. What would normally be placed in the received parameter when the proxy detected that the sender does not agree with the top-most via string of the form prev-hopip ipaddress[:port][/protocol] Where "port" is an optional parameter giving the transport layer port number and the default is5060. Where "protocol" is an optional parameter giving the transport layer protocol and the default is UDP. Valid values are TCP and UDP, but since the proxy does not support TCP, this parameter Is never included :5061/UDP Cantataprev-hopvia 144 Sent-by portion of topmost via when the request arrived at the proxy. String Syntax is as per RFC 3262 "SIP:Session Initiation protocol" :5060 Cantata- Q931- disconnectcause 30 Q.931 disconnect cause code retrieved from CCAPI. The source of the code is the disconnect location such as a PSTN, terminating gateway, or SIP. 2-character, ASCIIencoded hexadecimal number representing a Q.931 code. Range: 01 to A0 (which is 1 to 160 decimal) 3 18

19 Cantata RADIUS VSAs Cantatareleasesource 38 If a call was released by the calling party, called party, or an internal or external source. Values 1 Cantata- Setup- Time 25 Setup time in NTP format: hour, minutes, seconds, microseconds, time_zone, day, month, day_of_month, year. hh:mm:ss.mmm ZON DDD MMM ## YYYY 12:30: EST Fri Mar Cantatasipattemptinfo 151 In case IMG tries the contacts returned with 3xx, IMG will log to the CDR each attempt. Character String sip-attempt-info ="ATTEMPT 1 : @ :506 status 302" Cantatasip-respcode 150 Sip Cause Code before translation to the Q931 value Character String sip-resp-code = "486 Busy Here" Cantatatimestamp 105 Time of day at the dialed number or at the remote gateway hour, minutes, seconds. Decimal number hh:mm:ss 8:00:30 Cantatatranscause-code 53 Translated Cause Code when translate cause table is used. Character String trans-cause-code = "16" Cantatatrunk-grpin 44 The trunk group name from where the incoming call has come. Character string SS7_IMG Cantata- trunk-grp- 45 The trunk group name to which the Character string SS7_IMG 19

20 RADIUS out outgoing call is routed. Cantatavoip-dstrtp-ip-in 36 Inbound Remote VoIP gateway IP address. Numerals in dotted decimal notation: nnn.nnn.nnn.nnn Cantatavoip-dstrtp-ip-out 50 Outbound Remote VoIP gateway IP address. Numerals in dotted decimal notation: nnn.nnn.nnn.nnn Cantatavoip-dstsig-ip-in 23 Inbound IP address of the remote gateway. Numerals in dotted decimal notation: nnn.nnn.nnn.nnn Cantatavoip-dstsig-ip-out 49 Outbound IP address of the remote gateway Numerals in dotted decimal notation: nnn.nnn.nnn.nnn Cantatavoip-srcrtp-ip-in 46 Inbound Source VoIP RTP IP Address Numerals in dotted decimal notation: nnn.nnn.nnn.nnn Cantatavoip-srcrtp-ip-out 51 Outbound Source VoIP RTP IP Address Numerals in dotted decimal notation: nnn.nnn.nnn.nnn Cantatavoip-srcsig-ip-in 48 Outbound IP address of the source gateway Numerals in dotted decimal notation: nnn.nnn.nnn.nnn Cantatavoip-srcsig-ip-out 52 Inbound IP address of the source gateway Numerals in dotted decimal notation: nnn.nnn.nnn.nnn * ER1 or later 20

21 RADIUS Call Flow: SS7 to SIP 21

22

23 RADIUS Call Flow: SS7 to H Release from SS7 The following call flow shows a call that is released by the calling party (SS7). 23

24

25 RADIUS Call Flow: SS7 to H Release from SS7 25

26

27 RADIUS CDR Example: SS7 to H.323 Thu Aug 31 13:51: NAS-IP-Address = NAS-Port = 1812 Calling-Station-Id = " " Called-Station-Id = " " Acct-Session-Id = "00201c0403c1008c fb1d155d24" Acct-Status-Type = Start NAS-Port-Type = Ethernet Service-Type = Login-User call-origin = "originate" call-type = "SS7" Acct-Delay-Time = 0 Login-IP-Host = Tunnel-Client-Endpoint:0 = " " setup-time = "THU AUG 31 17:51:25: " dnis-pre-translate = " " ani-pre-translate = " " call-direction = "INCOMING LEG" trunk-grp-in = "ANSI_CICs" Client-IP-Address = Acct-Unique-Session-Id = "3641e eb86" Timestamp = Thu Aug 31 13:51: NAS-IP-Address = NAS-Port = 1812 Calling-Station-Id = " " Called-Station-Id = " " Acct-Session-Id = "00201c0403c1008c fb1d155d24" Acct-Status-Type = Start NAS-Port-Type = Ethernet Service-Type = Login-User h323-conf-id = "0023c1fd c0001ff" call-origin = "answer" 27

28 RADIUS call-type = "H323" Acct-Delay-Time = 0 Login-IP-Host = Tunnel-Client-Endpoint:0 = " " setup-time = "THU AUG 31 17:51:26: " connect-time = "THU AUG 31 17:51:30: " dnis-post-translate = " " ani-post-translate = " " call-direction = "OUTGOING LEG" trunk-grp-out = "H323_sw137" voip-dst-sig-ip-out = " " voip-dst-rtp-ip-out = " " voip-src-rtp-ip-out = " " voip-src-sig-ip-out = " " Client-IP-Address = Acct-Unique-Session-Id = "3641e eb86" Timestamp = Thu Aug 31 13:51: NAS-IP-Address = NAS-Port = 1812 Calling-Station-Id = " " Called-Station-Id = " " Acct-Session-Id = "00201c0403c1008c fb1d155d24" Acct-Session-Time = 20 Acct-Status-Type = Stop Acct-Terminate-Cause = NAS-Request NAS-Port-Type = Ethernet Service-Type = Login-User h323-conf-id = "0023c1fd c0001ff" call-origin = "originate" call-type = "SS7" Q931-disconnect-cause = "10" Acct-Delay-Time = 0 Login-IP-Host = Tunnel-Client-Endpoint:0 = " " setup-time = "THU AUG 31 17:51:25: " 28

29 RADIUS CDR Example: H.323-to-SS7 connect-time = "THU AUG 31 17:51:30: " disconnect-time = "THU AUG 31 17:51:50: " release-source = "CALLING PARTY RELEASE" dnis-pre-translate = " " ani-pre-translate = " " call-direction = "INCOMING LEG" trunk-grp-in = "ANSI_CICs" Client-IP-Address = Acct-Unique-Session-Id = "3641e eb86" Timestamp = Thu Aug 31 13:51: NAS-IP-Address = NAS-Port = 1812 Calling-Station-Id = " " Called-Station-Id = " " Acct-Session-Id = "00201c0403c1008c fb1d155d24" Acct-Session-Time = 20 Acct-Status-Type = Stop Acct-Terminate-Cause = Service-Unavailable NAS-Port-Type = Ethernet Service-Type = Login-User h323-conf-id = "0023c1fd c0001ff" call-origin = "answer" call-type = "H323" Q931-disconnect-cause = "10" Acct-Delay-Time = 0 Login-IP-Host = Tunnel-Client-Endpoint:0 = " " setup-time = "THU AUG 31 17:51:26: " connect-time = "THU AUG 31 17:51:30: " disconnect-time = "THU AUG 31 17:51:50: " Acct-Input-Octets = 0 Acct-Output-Octets = Acct-Input-Packets = 0 Acct-Output-Packets = 669 release-source = "CALLING PARTY RELEASE" 29

30 RADIUS dnis-post-translate = " " ani-post-translate = " " call-direction = "OUTGOING LEG" trunk-grp-out = "H323_sw137" lost-packets = "0" voip-dst-sig-ip-out = " " voip-dst-rtp-ip-out = " " voip-src-rtp-ip-out = " " voip-src-sig-ip-out = " " Client-IP-Address = Acct-Unique-Session-Id = "3641e eb86" Timestamp =

31 RADIUS CDR Example: SIP-to-ISDN Tue Mar 6 16:45: NAS-IP-Address = NAS-Port = 1812 Calling-Station-Id = " " Called-Station-Id = " " Acct-Session-Id = "00201c0403c ddec1e0002" Acct-Status-Type = Start NAS-Port-Type = Ethernet Service-Type = Login-User Cantata-call-origin = "originate" Cantata-call-type = "SIP" Acct-Delay-Time = 0 Login-IP-Host = Tunnel-Client-Endpoint:0 = " " Cantata-setup-time = "TUE MAR 06 21:45:01: " Cantata-voip-dst-sig-ip-in = " " Cantata-voip-dst-rtp-ip-in = " " Cantata-dnis-pre-translate = " " Cantata-ani-pre-translate = " " Cantata-call-direction = "INCOMING LEG" Cantata-trunk-grp-in = "EB_SIP" Cantata-voip-src-rtp-ip-in = " " Cantata-voip-src-sig-ip-in = " " Cantata-call-id = "NDMzNGY0Yjc2Y2FhMDZmNzVlZTQ4ODJiZWZlNWRjNWM." Cantata-prev-hop-ip = " :5061/UDP" Cantata-prev-hop-via = "sip: :5061" Cantata-incoming-req-uri = "sip: @ " Client-IP-Address = Acct-Unique-Session-Id = "4366fe3b438fd016" Timestamp = Tue Mar 6 16:45: NAS-IP-Address = NAS-Port = 1812 Calling-Station-Id = " " 31

32 RADIUS Called-Station-Id = " " Acct-Session-Id = "00201c0403c ddec1e0002" Acct-Status-Type = Start NAS-Port-Type = Ethernet Service-Type = Login-User Cantata-call-origin = "answer" Cantata-call-type = "ISDN" Acct-Delay-Time = 0 Login-IP-Host = Tunnel-Client-Endpoint:0 = " " Cantata-setup-time = "TUE MAR 06 21:45:02: " Cantata-connect-time = "TUE MAR 06 21:45:05: " Cantata-dnis-post-translate = " " Cantata-ani-post-translate = " " Cantata-call-direction = "OUTGOING LEG" Cantata-trunk-grp-out = "ISDN_1" Client-IP-Address = Acct-Unique-Session-Id = "4366fe3b438fd016" Timestamp = Tue Mar 6 16:45: NAS-IP-Address = NAS-Port = 1812 Calling-Station-Id = " " Called-Station-Id = " " Acct-Session-Id = "00201c0403c ddec1e0002" Acct-Session-Time = 20 Acct-Status-Type = Stop Acct-Terminate-Cause = NAS-Request NAS-Port-Type = Ethernet Service-Type = Login-User Cantata-call-origin = "answer" Cantata-call-type = "ISDN" Cantata-Q931-disconnect-cause = "10" Acct-Delay-Time = 0 Login-IP-Host = Tunnel-Client-Endpoint:0 = " " Cantata-setup-time = "TUE MAR 06 21:45:02: " 32

33 RADIUS CDR Example: SIP-to-ISDN Cantata-connect-time = "TUE MAR 06 21:45:05: " Cantata-disconnect-time = "TUE MAR 06 21:45:25: " Cantata-release-source = "CALLED PARTY RELEASE" Cantata-dnis-post-translate = " " Cantata-ani-post-translate = " " Cantata-call-direction = "OUTGOING LEG" Cantata-trunk-grp-out = "ISDN_1" Client-IP-Address = Acct-Unique-Session-Id = "4366fe3b438fd016" Timestamp = Tue Mar 6 16:45: NAS-IP-Address = NAS-Port = 1812 Calling-Station-Id = " " Called-Station-Id = " " Acct-Session-Id = "00201c0403c ddec1e0002" Acct-Session-Time = 20 Acct-Status-Type = Stop Acct-Terminate-Cause = NAS-Request NAS-Port-Type = Ethernet Service-Type = Login-User Cantata-call-origin = "originate" Cantata-call-type = "SIP" Cantata-Q931-disconnect-cause = "10" Acct-Delay-Time = 0 Login-IP-Host = Tunnel-Client-Endpoint:0 = " " Cantata-setup-time = "TUE MAR 06 21:45:01: " Cantata-connect-time = "TUE MAR 06 21:45:05: " Cantata-disconnect-time = "TUE MAR 06 21:45:25: " Cantata-voip-dst-sig-ip-in = " " Acct-Input-Octets = Acct-Output-Octets = Acct-Input-Packets = 1149 Acct-Output-Packets = 1156 Cantata-voip-dst-rtp-ip-in = " " Cantata-release-source = "CALLED PARTY RELEASE" 33

34 RADIUS Cantata-dnis-pre-translate = " " Cantata-ani-pre-translate = " " Cantata-call-direction = "INCOMING LEG" Cantata-trunk-grp-in = "EB_SIP" Cantata-voip-src-rtp-ip-in = " " Cantata-lost-packets = "0" Cantata-voip-src-sig-ip-in = " " Cantata-call-id = "NDMzNGY0Yjc2Y2FhMDZmNzVlZTQ4ODJiZWZlNWRjNWM." Cantata-prev-hop-ip = " :5061/UDP" Cantata-prev-hop-via = "sip: :5061" Cantata-incoming-req-uri = "sip: @ " Client-IP-Address = Acct-Unique-Session-Id = "4366fe3b438fd016" Timestamp =

35 RADIUS CDR Example: SIP to SS7 Fri Oct 27 11:47: NAS-IP-Address = NAS-Port = 1812 Calling-Station-Id = " " Called-Station-Id = " " Acct-Session-Id = "00201c0403d9009e de1b459d4" Acct-Status-Type = Start NAS-Port-Type = Ethernet Service-Type = Login-User Cantata-Call-Origin = "originate" Cantata-Call-Type = "SIP" Acct-Delay-Time = 0 Login-IP-Host = Tunnel-Client-Endpoint:0 = " " Cantata-Setup-Time = "FRI OCT 27 15:47:39: " Cantata-voip-dst-sig-ip-in = " " Cantata-voip-dst-rtp-ip-in = " " Cantata-dnis-pre-translate = " " Cantata-ani-pre-translate = " " Cantata-call-direction = "INCOMING LEG" Cantata-trunk-grp-in = "SIP_CSP" Cantata-voip-src-rtp-ip-in = " " Cantata-voip-src-sig-ip-in = " " Cantata-call-id = "CSP b @ " Cantata-prev-hop-ip = " :1053/UDP" Cantata-prev-hop-via = "sip: :5060" Cantata-incoming-req-uri = "sip: @ " Client-IP-Address = Acct-Unique-Session-Id = "d34eddf4b59683df" Timestamp = Fri Oct 27 11:47: NAS-IP-Address = NAS-Port =

36 RADIUS Calling-Station-Id = " " Called-Station-Id = " " Acct-Session-Id = "00201c0403d9009e de1b459d4" Acct-Status-Type = Start NAS-Port-Type = Ethernet Service-Type = Login-User Cantata-Call-Origin = "answer" Cantata-Call-Type = "SS7" Acct-Delay-Time = 0 Login-IP-Host = Tunnel-Client-Endpoint:0 = " " Cantata-Setup-Time = "FRI OCT 27 15:47:40: " Cantata-connect-time = "FRI OCT 27 15:47:43: " Cantata-dnis-post-translate = " " Cantata-ani-post-translate = " " Cantata-call-direction = "OUTGOING LEG" Cantata-trunk-grp-out = "SS7_E1_CSP" Client-IP-Address = Acct-Unique-Session-Id = "d34eddf4b59683df" Timestamp = Fri Oct 27 11:48: NAS-IP-Address = NAS-Port = 1812 Calling-Station-Id = " " Called-Station-Id = " " Acct-Session-Id = "00201c0403d9009e de1b459d4" Acct-Session-Time = 22 Acct-Status-Type = Stop Acct-Terminate-Cause = NAS-Request NAS-Port-Type = Ethernet Service-Type = Login-User Cantata-Call-Origin = "originate" Cantata-Call-Type = "SIP" Cantata-Q931-disconnect-cause = "10" Acct-Delay-Time = Login-IP-Host =

37 RADIUS CDR Example: SS7-to-SIP Tunnel-Client-Endpoint:0 = " " Cantata-Setup-Time = "FRI OCT 27 15:47:39: " Cantata-connect-time = "FRI OCT 27 15:47:43: " Cantata-disconnect-time = "FRI OCT 27 15:48:05: " Cantata-voip-dst-sig-ip-in = " " Acct-Input-Octets = Acct-Output-Octets = Acct-Input-Packets = 1236 Acct-Output-Packets = 1237 Cantata-voip-dst-rtp-ip-in = " " Cantata-release-source = "CALLING PARTY RELEASE" Cantata-dnis-pre-translate = " " Cantata-ani-pre-translate = " " Cantata-call-direction = "INCOMING LEG" Cantata-trunk-grp-in = "SIP_CSP" Cantata-voip-src-rtp-ip-in = " " Cantata-lost-packets = "0" Cantata-voip-src-sig-ip-in = " " Cantata-call-id = "CSP b @ " Cantata-prev-hop-ip = " :1053/UDP" Cantata-prev-hop-via = "sip: :5060" Cantata-incoming-req-uri = "sip: @ " Client-IP-Address = Acct-Unique-Session-Id = "d34eddf4b59683df" Timestamp = Fri Oct 27 11:48: NAS-IP-Address = NAS-Port = 1812 Calling-Station-Id = " " Called-Station-Id = " " Acct-Session-Id = "00201c0403d9009e de1b459d4" Acct-Session-Time = 22 Acct-Status-Type = Stop Acct-Terminate-Cause = NAS-Request NAS-Port-Type = Ethernet Service-Type = Login-User 37

38 RADIUS Cantata-Call-Origin = "answer" Cantata-Call-Type = "SS7" Cantata-Q931-disconnect-cause = "10" Acct-Delay-Time = 0 Login-IP-Host = Tunnel-Client-Endpoint:0 = " " Cantata-Setup-Time = "FRI OCT 27 15:47:40: " Cantata-connect-time = "FRI OCT 27 15:47:43: " Cantata-disconnect-time = "FRI OCT 27 15:48:05: " Cantata-release-source = "CALLING PARTY RELEASE" Cantata-dnis-post-translate = " " Cantata-ani-post-translate = " " Cantata-call-direction = "OUTGOING LEG" Cantata-trunk-grp-out = "SS7_E1_CSP" Client-IP-Address = Acct-Unique-Session-Id = "d34eddf4b59683df" Timestamp =

39 RADIUS CDR Example - SIP to SIP with Proxy and DNS Wed Aug 30 14:41: NAS-IP-Address = NAS-Port = 1812 Calling-Station-Id = " " Called-Station-Id = "4005" Acct-Session-Id = "00201c0403c d61c08fe" Acct-Status-Type = Start NAS-Port-Type = Ethernet Service-Type = Login-User call-origin = "originate" call-type = "SIP" Acct-Delay-Time = 0 Login-IP-Host = Tunnel-Client-Endpoint:0 = " " setup-time = "WED AUG 30 18:41:26: " voip-dst-sig-ip-in = " " voip-dst-rtp-ip-in = " " dnis-pre-translate = "4005" ani-pre-translate = " " call-direction = "INCOMING LEG" trunk-grp-in = "4001_SIP_Snom200" voip-src-rtp-ip-in = " " voip-src-sig-ip-in = " " call-id = "3c2e91d9e407-7za42unmvgcb@ " prev-hop-ip = " :5060/UDP" prev-hop-via = "sip: :5060" incoming-req-uri = "sip:4005@ " Client-IP-Address = Acct-Unique-Session-Id = "9318c8cea8c1fa1f" Timestamp = Wed Aug 30 14:41: NAS-IP-Address = NAS-Port =

40 RADIUS Calling-Station-Id = " " Called-Station-Id = "4005" Acct-Session-Id = "00201c0403c d61c08fe" Acct-Status-Type = Start NAS-Port-Type = Ethernet Service-Type = Login-User call-origin = "answer" call-type = "SIP" Acct-Delay-Time = 0 Login-IP-Host = Tunnel-Client-Endpoint:0 = " " setup-time = "WED AUG 30 18:41:26: " connect-time = "WED AUG 30 18:41:30: " dnis-post-translate = "4005" ani-post-translate = " " call-direction = "OUTGOING LEG" trunk-grp-out = "SIP_IBM_DNS" next-hop-ip = " " outgoing-req-uri = "sip:4005@mg-ibm" next-hop-dn = "linux-sip" call-id = "7cf5-40b Miami-1@ " voip-dst-sig-ip-out = " " voip-dst-rtp-ip-out = " " voip-src-rtp-ip-out = " " voip-src-sig-ip-out = " " Client-IP-Address = Acct-Unique-Session-Id = "9318c8cea8c1fa1f" Timestamp = Wed Aug 30 14:41: NAS-IP-Address = NAS-Port = 1812 Calling-Station-Id = " " Called-Station-Id = "4005" Acct-Session-Id = "00201c0403c d61c08fe" Acct-Session-Time = 6 Acct-Status-Type = Stop 40

41 RADIUS CDR Example - SIP to SIP with Proxy and DNS Acct-Terminate-Cause = NAS-Request NAS-Port-Type = Ethernet Service-Type = Login-User call-origin = "answer" call-type = "SIP" Q931-disconnect-cause = "10" Acct-Delay-Time = 0 Login-IP-Host = Tunnel-Client-Endpoint:0 = " " setup-time = "WED AUG 30 18:41:26: " connect-time = "WED AUG 30 18:41:30: " disconnect-time = "WED AUG 30 18:41:36: " Acct-Input-Octets = Acct-Output-Octets = Acct-Input-Packets = 300 Acct-Output-Packets = 304 release-source = "CALLED PARTY RELEASE" dnis-post-translate = "4005" ani-post-translate = " " call-direction = "OUTGOING LEG" trunk-grp-out = "SIP_IBM_DNS" lost-packets = "0" next-hop-ip = " " outgoing-req-uri = "sip:4005@mg-ibm" next-hop-dn = "linux-sip" call-id = "7cf5-40b Miami-1@ " voip-dst-sig-ip-out = " " voip-dst-rtp-ip-out = " " voip-src-rtp-ip-out = " " voip-src-sig-ip-out = " " Client-IP-Address = Acct-Unique-Session-Id = "9318c8cea8c1fa1f" Timestamp = Wed Aug 30 14:41: NAS-IP-Address = NAS-Port =

42 RADIUS Calling-Station-Id = " " Called-Station-Id = "4005" Acct-Session-Id = "00201c0403c d61c08fe" Acct-Session-Time = 6 Acct-Status-Type = Stop Acct-Terminate-Cause = NAS-Request NAS-Port-Type = Ethernet Service-Type = Login-User call-origin = "originate" call-type = "SIP" Q931-disconnect-cause = "10" Acct-Delay-Time = 0 Login-IP-Host = Tunnel-Client-Endpoint:0 = " " setup-time = "WED AUG 30 18:41:26: " connect-time = "WED AUG 30 18:41:30: " disconnect-time = "WED AUG 30 18:41:36: " voip-dst-sig-ip-in = " " Acct-Input-Octets = Acct-Output-Octets = Acct-Input-Packets = 471 Acct-Output-Packets = 473 voip-dst-rtp-ip-in = " " release-source = "CALLED PARTY RELEASE" dnis-pre-translate = "4005" ani-pre-translate = " " call-direction = "INCOMING LEG" trunk-grp-in = "4001_SIP_Snom200" voip-src-rtp-ip-in = " " lost-packets = "0" voip-src-sig-ip-in = " " call-id = "3c2e91d9e407-7za42unmvgcb@ " prev-hop-ip = " :5060/UDP" prev-hop-via = "sip: :5060" incoming-req-uri = "sip:4005@ " Client-IP-Address =

43 RADIUS CDR Example - SIP to SIP with Proxy and DNS Acct-Unique-Session-Id = "9318c8cea8c1fa1f" Timestamp =

44

45 RADIUS CDR Example: CAS to SS7 Tue Oct 24 16:54: NAS-IP-Address = NAS-Port = 1812 Calling-Station-Id = " " Called-Station-Id = " " Acct-Session-Id = "00201c e00f a6b26" Acct-Status-Type = Start NAS-Port-Type = Ethernet Service-Type = Login-User Cantata-call-origin = "originate" Cantata-call-type = "CAS T1" Acct-Delay-Time = 0 Login-IP-Host = Tunnel-Client-Endpoint:0 = " " Cantata-setup-time = "TUE OCT 24 20:54:44: " Cantata-dnis-pre-translate = " " Cantata-ani-pre-translate = " " Cantata-call-direction = "INCOMING LEG" Cantata-trunk-grp-in = "IncomingCAS" Client-IP-Address = Acct-Unique-Session-Id = "6331a70cd9eff252" Timestamp = Tue Oct 24 16:54: NAS-IP-Address = NAS-Port = 1812 Calling-Station-Id = " " Called-Station-Id = " " Acct-Session-Id = "00201c e00f a6b26" Acct-Status-Type = Start NAS-Port-Type = Ethernet Service-Type = Login-User Cantata-call-origin = "answer" Cantata-call-type = "SS7" Acct-Delay-Time = 0 Login-IP-Host =

46 RADIUS Tunnel-Client-Endpoint:0 = " " Cantata-setup-time = "TUE OCT 24 20:54:44: " Cantata-connect-time = "TUE OCT 24 20:54:44: " Cantata-dnis-post-translate = " " Cantata-ani-post-translate = " " Cantata-call-direction = "OUTGOING LEG" Cantata-trunk-grp-out = "ss7_out" Client-IP-Address = Acct-Unique-Session-Id = "6331a70cd9eff252" Timestamp = Tue Oct 24 16:54: NAS-IP-Address = NAS-Port = 1812 Calling-Station-Id = " " Called-Station-Id = " " Acct-Session-Id = "00201c e00f a6b26" Acct-Session-Time = 7 Acct-Status-Type = Stop Acct-Terminate-Cause = NAS-Request NAS-Port-Type = Ethernet Service-Type = Login-User Cantata-call-origin = "answer" Cantata-call-type = "SS7" Cantata-Q931-disconnect-cause = "11" Acct-Delay-Time = 0 Login-IP-Host = Tunnel-Client-Endpoint:0 = " " Cantata-setup-time = "TUE OCT 24 20:54:44: " Cantata-connect-time = "TUE OCT 24 20:54:44: " Cantata-disconnect-time = "TUE OCT 24 20:54:51: " Cantata-release-source = "CALLED PARTY RELEASE" Cantata-dnis-post-translate = " " Cantata-ani-post-translate = " " Cantata-call-direction = "OUTGOING LEG" Cantata-trunk-grp-out = "ss7_out" Client-IP-Address = Acct-Unique-Session-Id = "6331a70cd9eff252" 46

47 RADIUS CDR Example: CAS to SS7 Timestamp = Tue Oct 24 16:54: NAS-IP-Address = NAS-Port = 1812 Calling-Station-Id = " " Called-Station-Id = " " Acct-Session-Id = "00201c e00f a6b26" Acct-Session-Time = 9 Acct-Status-Type = Stop Acct-Terminate-Cause = NAS-Request NAS-Port-Type = Ethernet Service-Type = Login-User Cantata-call-origin = "originate" Cantata-call-type = "CAS T1" Cantata-Q931-disconnect-cause = "11" Acct-Delay-Time = 0 Login-IP-Host = Tunnel-Client-Endpoint:0 = " " Cantata-setup-time = "TUE OCT 24 20:54:44: " Cantata-connect-time = "TUE OCT 24 20:54:44: " Cantata-disconnect-time = "TUE OCT 24 20:54:51: " Cantata-release-source = "CALLED PARTY RELEASE" Cantata-dnis-pre-translate = " " Cantata-ani-pre-translate = " " Cantata-call-direction = "INCOMING LEG" Cantata-trunk-grp-in = "IncomingCAS" Client-IP-Address = Acct-Unique-Session-Id = "6331a70cd9eff252" Timestamp =

48

49 RADIUS CDR Example: SS7 to CAS Wed Oct 25 13:38: NAS-IP-Address = NAS-Port = 1812 Calling-Station-Id = " " Called-Station-Id = " " Acct-Session-Id = "00201c f004d f8305baaf4" Acct-Status-Type = Start NAS-Port-Type = Ethernet Service-Type = Login-User Cantata-call-origin = "originate" Cantata-call-type = "SS7" Acct-Delay-Time = 0 Login-IP-Host = Tunnel-Client-Endpoint:0 = " " Cantata-setup-time = "WED OCT 25 17:38:56: " Cantata-dnis-pre-translate = " " Cantata-ani-pre-translate = " " Cantata-call-direction = "INCOMING LEG" Cantata-trunk-grp-in = "ss7_in" Client-IP-Address = Acct-Unique-Session-Id = "77debfa2fbe236b0" Timestamp = Wed Oct 25 13:39: NAS-IP-Address = NAS-Port = 1812 Calling-Station-Id = " " Called-Station-Id = " " Acct-Session-Id = "00201c f004d f8305baaf4" Acct-Status-Type = Start NAS-Port-Type = Ethernet Service-Type = Login-User Cantata-call-origin = "answer" Cantata-call-type = "CAS T1" Acct-Delay-Time = 0 Login-IP-Host =

50 RADIUS Tunnel-Client-Endpoint:0 = " " Cantata-setup-time = "WED OCT 25 17:39:00: " Cantata-connect-time = "WED OCT 25 17:39:07: " Cantata-dnis-post-translate = " " Cantata-ani-post-translate = " " Cantata-call-direction = "OUTGOING LEG" Cantata-trunk-grp-out = "IncomingCAS" Client-IP-Address = Acct-Unique-Session-Id = "77debfa2fbe236b0" Timestamp = Wed Oct 25 13:39: NAS-IP-Address = NAS-Port = 1812 Calling-Station-Id = " " Called-Station-Id = " " Acct-Session-Id = "00201c f004d f8305baaf4" Acct-Session-Time = 10 Acct-Status-Type = Stop Acct-Terminate-Cause = NAS-Request NAS-Port-Type = Ethernet Service-Type = Login-User Cantata-call-origin = "originate" Cantata-call-type = "SS7" Cantata-Q931-disconnect-cause = "10" Acct-Delay-Time = 0 Login-IP-Host = Tunnel-Client-Endpoint:0 = " " Cantata-setup-time = "WED OCT 25 17:38:56: " Cantata-connect-time = "WED OCT 25 17:39:07: " Cantata-disconnect-time = "WED OCT 25 17:39:17: " Cantata-release-source = "CALLING PARTY RELEASE" Cantata-dnis-pre-translate = " " Cantata-ani-pre-translate = " " Cantata-call-direction = "INCOMING LEG" Cantata-trunk-grp-in = "ss7_in" Client-IP-Address = Acct-Unique-Session-Id = "77debfa2fbe236b0" 50

51 RADIUS CDR Example: SS7 to CAS Timestamp = Wed Oct 25 13:39: NAS-IP-Address = NAS-Port = 1812 Calling-Station-Id = " " Called-Station-Id = " " Acct-Session-Id = "00201c f004d f8305baaf4" Acct-Session-Time = 10 Acct-Status-Type = Stop Acct-Terminate-Cause = NAS-Request NAS-Port-Type = Ethernet Service-Type = Login-User Cantata-call-origin = "answer" Cantata-call-type = "CAS T1" Cantata-Q931-disconnect-cause = "10" Acct-Delay-Time = 0 Login-IP-Host = Tunnel-Client-Endpoint:0 = " " Cantata-setup-time = "WED OCT 25 17:39:00: " Cantata-connect-time = "WED OCT 25 17:39:07: " Cantata-disconnect-time = "WED OCT 25 17:39:17: " Cantata-release-source = "CALLING PARTY RELEASE" Cantata-dnis-post-translate = " " Cantata-ani-post-translate = " " Cantata-call-direction = "OUTGOING LEG" Cantata-trunk-grp-out = "IncomingCAS" Client-IP-Address = Acct-Unique-Session-Id = "77debfa2fbe236b0" Timestamp =

52

53 RADIUS CDR Example: Pre-Paid Support The following CDR shows a SIP to SIP call with Pre-paid Support enabled. Important lines related to Pre-paid Support are in red. 18:07: CALL(SIP) (01:0001:00) RCVD INVITE from :5060 UDP 18:07: CALL(SIP) (01:0001:00) with Via sent-by: :07: CALL(SIP) (01:0001:00) SENT 100 Trying [] to : :07: CALL(SIP) (01:0001:00) SENT Route Ctrl to L4 18:07: CALL(L4) (01:0001:00) RCVD Route Control from SIP 18:07: CALL(L4) (01:0001:00) Accessing Route Table 2 18:07: CALL(L4) (01:0001:00) Accessing Resource Table 1 18:07: CALL(L4) (01:0001:00) SENT Route Control Ack to SIP 18:07: CALL(SIP) (01:0001:00) RCVD Route Ctrl Ack from L4 18:07: CALL(SIP) (01:0001:00) Session Group Profile ID is 0 18:07: CALL(SIP) (01:0001:00) SENT Route Ctrl to L4 18:07: CALL(L4) (01:0001:00) RCVD Route Control from SIP 18:07: CALL(L4) (01:0001:00) Accessing IP Bearer Profiles 18:07: CALL(L4) (01:0001:00) Profile Id 1 (RG 769) 18:07: CALL(L4) (01:0001:00) SENT Route Control Ack to SIP 18:07: CALL(SIP) (01:0001:00) RCVD Route Ctrl Ack from L4 18:07: CALL(SIP) (01:0001:00) m line codec list: :07: CALL(SIP) (01:0001:00) RTP Type: 0, name: PCMU, clk: :07: CALL(SIP) (01:0001:00) RTP Type: 101, name: telephoneevent, clk: :07: CALL(SIP) (01:0001:00) Leg 0 associated with hndl(19999), LTS(1792) 18:07: CALL(SIP) (01:0001:00) SENT Setup to L4 18:07: CALL(L4) (01:0001:00) RCVD Setup Ind from SIP 18:07: CALL(L4) (01:0001:00) SENT RFS to GCL 18:07: CALL(GCL) (00:0000:00) SENT RADIUS AUTH REQUEST 18:07: CALL(GCL) (01:0001:00) RCVD RFS DN=[4004] ANI=[ ] from L4 18:07: CALL(GCL) (01:0001:00) ANI APRI=[0],SI=[0],Category=[10] 18:07: CALL(GCL) (01:0001:00) Incoming Channel Group = 10 [4001_SIP_Snom200] 18:07: CALL(GCL) (01:0001:00) RCVD RADIUS AUTH ACCEPT ACK 53

54 RADIUS 18:07: CALL(GCL) (01:0001:00) Using Radius MaxDurCall=1 seconds 18:07: CALL(GCL) (01:0001:00) SENT RADIUS ACCT START 18:07: CALL(L4) (01:0001:00) RCVD RFS response from GCL 18:07: CALL(GCL) (01:0001:00) RCVD RADIUS ACCT START 18:07: CALL(GCL) (01:0001:01) SENT Route Control to L4 18:07: CALL(L4) (01:0001:00) RCVD Route Control from GCL 18:07: CALL(L4) (01:0001:00) Accessing Route Table 5 18:07: CALL(GCL) (01:0001:01) RCVD Mid Stream Router Response 18:07: CALL(GCL) (01:0001:01) Outgoing Channel Group = 13 [SIP_IBM_5060] 18:07: CALL(GCL) (01:0001:01) SENT Route Control to L4 18:07: CALL(L4) (01:0001:01) RCVD Route Control from GCL 18:07: CALL(L4) (01:0001:01) Accessing Resource Table 1 18:07: CALL(L4) (01:0001:01) Resource Group ID is 1 18:07: CALL(L4) (01:0001:01) SENT Outseize Ctrl to SIP 18:07: CALL(GCL) (01:0001:01) RCVD CPE of ADDRESS INFO from L4 18:07: CALL(GCL) (01:0001:01) Leg 1 associated with LTS(1024) 18:07: CALL(SIP) (01:0001:01) RCVD Outseize Ctrl from L4 18:07: CALL(SIP) (01:0001:01) SENT Route Ctrl to L4 18:07: CALL(L4) (01:0001:01) RCVD Route Control from SIP 18:07: CALL(L4) (01:0001:01) Accessing GatewayId to IP Tables 18:07: CALL(L4) (01:0001:01) Gateway ID is :07: CALL(L4) (01:0001:01) SENT Route Control Ack to SIP 18:07: CALL(SIP) (01:0001:01) RCVD Route Ctrl Ack from L4 18:07: CALL(SIP) (01:0001:01) Session Group Profile ID is 0 18:07: CALL(SIP) (01:0001:01) SENT Route Ctrl to L4 18:07: CALL(L4) (01:0001:01) RCVD Route Control from SIP 18:07: CALL(L4) (01:0001:01) Accessing IP Bearer Profiles 18:07: CALL(L4) (01:0001:01) Profile Id 1 (RG 769) 18:07: CALL(L4) (01:0001:01) SENT Route Control Ack to SIP 18:07: CALL(SIP) (01:0001:01) RCVD Route Ctrl Ack from L4 18:07: CALL(SIP) (01:0001:01) SENT INVITE to :5060 UDP 18:07: CALL(SIP) (01:0001:01) with R-URI: :5060 UDP 18:07: CALL(SIP) (01:0001:01) RCVD 180 Ringing from :5060 UDP 18:07: CALL(SIP) (01:0001:01) SENT Outseize Ack to L4 18:07: CALL(SIP) (01:0001:01) SENT Alerting to L4 54

55 RADIUS CDR Example: Pre-paid Support 18:07: CALL(L4) (01:0001:01) RCVD Outseize ACK from SIP 18:07: CALL(L4) (01:0001:01) RVCD Alerting from SIP 18:07: CALL(GCL) (01:0001:00) SENT Connect Tone to L4 18:07: CALL(L4) (01:0001:00) RCVD Connect Tone from GCL 18:07: CALL(L4) (01:0001:00) SENT Request DSP Service to SYSRM 18:07: CALL(MED) (01:0001:00) Transmitting tone 0x2 on (0x1,0x1,0x0) 18:07: CALL(MED) (01:0001:00) RCVD OUTPULSE CP 18:07: CALL(GCL) (01:0001:00) SENT Connect w/pad Response to L4 18:07: CALL(L4) (01:0001:00) SENT Progress to SIP 18:07: CALL(SIP) (01:0001:01) RCVD Host Connect from L4 18:07: CALL(SIP) (01:0001:00) RCVD Progress from L4 18:07: CALL(SIP) (01:0001:00) Set Fax Type to Bypass 18:07: CALL(SIP) (01:0001:00) SENT L3-L3 Outseize Ctrl to VPPL 18:07: CALL(IP) (01:0001:00) RCVD Outseize from L3, [vts 768] [m1.0.1] 18:07: CALL(IP) (01:0001:00) RTP: (Src) : :07: CALL(IP) (01:0001:00) RTP: (Dst) : :07: CALL(IP) (01:0001:00) VoIP Codec is G711Mulaw; Payload Size is 20ms 18:07: CALL(IP) (01:0001:00) RFC2833 DTMF Relay in use, Dynamic Payload Type is :07: CALL(IP) (01:0001:00) SENT Outseize ACK to L3P, topvid: x66 18:07: CALL(SIP) (01:0001:00) RCVD Outseize Ack from VPPL 18:07: CALL(SIP) (01:0001:00) SENT 183 Session Progress to :5060 UDP 18:07: CALL(SIP) (01:0001:00) RCVD Cut Thru from VPPL 18:07: CALL(SIP) (01:0001:00) RCVD Connect from VPPL 18:07: CALL(SIP) (01:0001:01) RCVD 200 OK from :5060 UDP 18:07: CALL(SIP) (01:0001:01) SENT ACK to :5060 UDP 18:07: CALL(SIP) (01:0001:01) with R-URI: :5060 UDP 18:07: CALL(SIP) (01:0001:01) m line codec list: 0 18:07: CALL(SIP) (01:0001:01) RTP Type: 0, name: PCMU, clk: :07: CALL(SIP) (01:0001:01) Set Fax Type to Bypass 18:07: CALL(SIP) (01:0001:01) SENT L3-L3 Outseize Ctrl to VPPL 18:07: CALL(IP) (01:0001:01) RCVD Outseize from L3, [vts 0] [m0.0.1] 55

Voice Over IP Information

Voice Over IP Information Voice Over IP Information Basic CISCO information The links below contain information specific to Cisco about VoIP: Cisco RADIUS Vendor-Specific Attributes for VoIP Call Authorization http://www.cisco.com/warp/public/cc/so/neso/vvda/pctl/distrib/radus_ov.htm

More information

RADIUS Authentication and Accounting

RADIUS Authentication and Accounting 5 RADIUS Authentication and Accounting Contents Overview...................................................... 5-2 Terminology................................................... 5-3 Switch Operating Rules

More information

WiNG 4.X / WiNG 5.X RADIUS Attributes

WiNG 4.X / WiNG 5.X RADIUS Attributes Configuration Guide for RFMS 3.0 Initial Configuration XXX-XXXXXX-XX WiNG 4.X / WiNG 5.X RADIUS Attributes Part No. TME-08-2011-01 Rev. C MOTOROLA and the Stylized M Logo are registered in the US Patent

More information

Configuring RADIUS Servers

Configuring RADIUS Servers CHAPTER 13 This chapter describes how to enable and configure the Remote Authentication Dial-In User Service (RADIUS), that provides detailed accounting information and flexible administrative control

More information

RADIUS Vendor-Specific Attributes (VSA)

RADIUS Vendor-Specific Attributes (VSA) The Internet Engineering Task Force (IETF) draft standard specifies a method for communicating vendor-specific information between the network access server and the RADIUS server by using the vendor-specific

More information

VOIP-211RS/210RS/220RS/440S. SIP VoIP Router. User s Guide

VOIP-211RS/210RS/220RS/440S. SIP VoIP Router. User s Guide VOIP-211RS/210RS/220RS/440S SIP VoIP Router User s Guide Trademarks Contents are subject to revise without prior notice. All trademarks belong to their respective owners. FCC Warning This equipment has

More information

Avaya IP Office 8.1 Configuration Guide

Avaya IP Office 8.1 Configuration Guide Avaya IP Office 8.1 Configuration Guide Performed By tekvizion PVS, Inc. Contact: 214-242-5900 www.tekvizion.com Revision: 1.1 Date: 10/14/2013 Copyright 2013 by tekvizion PVS, Inc. All Rights Reserved.

More information

Provisioning and configuring the SIP Spider

Provisioning and configuring the SIP Spider Provisioning and configuring the SIP Spider Administrator Guide Table of Contents 1. Introduction... 3 2. Manual Provisioning... 4 3. Automatic Provisioning... 5 3.1 Concept... 5 3.2 Preparing the configuration

More information

SIP Trunking Manual 05.15. Technical Support Web Site: http://ws1.necii.com (registration is required)

SIP Trunking Manual 05.15. Technical Support Web Site: http://ws1.necii.com (registration is required) SIP Trunking Manual 05.15 Technical Support Web Site: http://ws1.necii.com (registration is required) This manual has been developed by NEC Unified Solutions, Inc. It is intended for the use of its customers

More information

NAT TCP SIP ALG Support

NAT TCP SIP ALG Support The feature allows embedded messages of the Session Initiation Protocol (SIP) passing through a device that is configured with Network Address Translation (NAT) to be translated and encoded back to the

More information

SIP : Session Initiation Protocol

SIP : Session Initiation Protocol : Session Initiation Protocol EFORT http://www.efort.com (Session Initiation Protocol) as defined in IETF RFC 3261 is a multimedia signaling protocol used for multimedia session establishment, modification

More information

nexvortex Setup Guide

nexvortex Setup Guide nexvortex Setup Guide CUDATEL COMMUNICATION SERVER September 2012 510 S P R I N G S T R E E T H E R N D O N V A 2 0 1 7 0 + 1 8 5 5. 6 3 9. 8 8 8 8 Introduction This document is intended only for nexvortex

More information

Session Initiation Protocol (SIP) The Emerging System in IP Telephony

Session Initiation Protocol (SIP) The Emerging System in IP Telephony Session Initiation Protocol (SIP) The Emerging System in IP Telephony Introduction Session Initiation Protocol (SIP) is an application layer control protocol that can establish, modify and terminate multimedia

More information

Application Notes for Configuring SIP Trunking between McLeodUSA SIP Trunking Solution and an Avaya IP Office Telephony Solution 1.

Application Notes for Configuring SIP Trunking between McLeodUSA SIP Trunking Solution and an Avaya IP Office Telephony Solution 1. Avaya Solution & Interoperability Test Lab Application Notes for Configuring SIP Trunking between McLeodUSA SIP Trunking Solution and an Avaya IP Office Telephony Solution 1.0 Abstract These Application

More information

SIP Trunking Quick Reference Document

SIP Trunking Quick Reference Document SIP Trunking Quick Reference Document Publication Information SAMSUNG TELECOMMUNICATIONS AMERICA reserves the right without prior notice to revise information in this publication for any reason. SAMSUNG

More information

SIP Trunking. Service Guide. www.megapath.com. Learn More: Call us at 877.634.2728.

SIP Trunking. Service Guide. www.megapath.com. Learn More: Call us at 877.634.2728. Service Guide Learn More: Call us at 877.634.2728. www.megapath.com What is MegaPath SIP Trunking? SIP Trunking enables your business to reduce costs and simplify IT management by combining voice and Internet

More information

nexvortex Setup Guide

nexvortex Setup Guide nexvortex Setup Guide CISCO UC500 March 2012 Introduction This document is intended only for nexvortex customers and resellers as an aid to setting up the Cisco PBX software to connect to the nexvortex

More information

Link2VoIP SIP Trunk Setup

Link2VoIP SIP Trunk Setup DSX Link2VoIP SIP Trunk Setup April 23, 2011 Issue 1.00 NEC Corporation of America 4 Forest Parkway, Shelton, CT 06484 Copyright 2011 NEC Corporation of America 6535 N. State Highway 161 Irving, TX 75039-2402

More information

Application Notes for Configuring Cablevision Optimum Voice SIP Trunking with Avaya IP Office - Issue 1.1

Application Notes for Configuring Cablevision Optimum Voice SIP Trunking with Avaya IP Office - Issue 1.1 Avaya Solution & Interoperability Test Lab Application Notes for Configuring Cablevision Optimum Voice SIP Trunking with Avaya IP Office - Issue 1.1 Abstract These Application Notes describe the procedures

More information

Application Notes for Avaya IP Office 7.0 Integration with Skype Connect R2.0 Issue 1.0

Application Notes for Avaya IP Office 7.0 Integration with Skype Connect R2.0 Issue 1.0 Avaya Solution & Interoperability Test Lab Application Notes for Avaya IP Office 7.0 Integration with Skype Connect R2.0 Issue 1.0 Abstract These Application Notes describe the steps to configure an Avaya

More information

Technical Configuration Notes

Technical Configuration Notes MITEL SIP CoE Technical Configuration Notes Configure MCD for use with OpenIP SIP Trunking service SIP CoE 11-4940-00186 NOTICE The information contained in this document is believed to be accurate in

More information

White paper. SIP An introduction

White paper. SIP An introduction White paper An introduction Table of contents 1 Introducing 3 2 How does it work? 3 3 Inside a normal call 4 4 DTMF sending commands in sip calls 6 5 Complex environments and higher security 6 6 Summary

More information

Configuring SIP Trunking and Networking for the NetVanta 7000 Series

Configuring SIP Trunking and Networking for the NetVanta 7000 Series 61200796L1-29.4E July 2011 Configuration Guide Configuring for the NetVanta 7000 Series This configuration guide describes the configuration and implementation of Session Initiation Protocol (SIP) trunking

More information

Application Notes for Configuring Intelepeer SIP Trunking with Avaya IP Office 7.0 - Issue 1.0

Application Notes for Configuring Intelepeer SIP Trunking with Avaya IP Office 7.0 - Issue 1.0 Avaya Solution & Interoperability Test Lab Application Notes for Configuring Intelepeer SIP Trunking with Avaya IP Office 7.0 - Issue 1.0 Abstract These Application Notes describe the procedures for configuring

More information

SIP Trunking Service Configuration Guide for Time Warner Cable Business Class

SIP Trunking Service Configuration Guide for Time Warner Cable Business Class SIP Trunking Service Configuration Guide for Time Warner Cable Business Class NDA-31669 Issue 1.0 NEC Corporation of America reserves the right to change the specifications, functions, or features at

More information

SBC 1000/2000 Configuration Guide with Lync 2013 for Windstream/ LPAETEC SIP Trunk Deployments

SBC 1000/2000 Configuration Guide with Lync 2013 for Windstream/ LPAETEC SIP Trunk Deployments SBC 1000/2000 Configuration Guide with Lync 2013 for Windstream/ LPAETEC SIP Trunk Deployments Application Notes Rev. 1.0 Last Updated: April 10, 2015 Revision Date Revised By Comments 0.1 12/03/2015 Roman

More information

Connecting with Vonage

Connecting with Vonage Connecting with Vonage Vonage (http://www.vonage.com/) offers telephone service using the VoIP (Voice over Internet Protocol) standard SIP (Session Initiation Protocol). The service allow users making

More information

AGLARBRI PROJECT AFRICAN GREAT LAKES RURAL BROADBAND RESEARCH INFRASTRUCTURE. RADIUS installation and configuration

AGLARBRI PROJECT AFRICAN GREAT LAKES RURAL BROADBAND RESEARCH INFRASTRUCTURE. RADIUS installation and configuration AGLARBRI PROJECT AFRICAN GREAT LAKES RURAL BROADBAND RESEARCH INFRASTRUCTURE RADIUS installation and configuration Project Manager: Miguel Sosa (mesc@kth.se) Member Email Position and number of credits

More information

Application Notes Rev. 1.0 Last Updated: January 9, 2015

Application Notes Rev. 1.0 Last Updated: January 9, 2015 SBC 1000/2000 Series Configuration Guide with Cisco Unified Call Manager v9.1 for Level 3 Voice Complete SM SIP Trunk Deployments Application Notes Rev. 1.0 Last Updated: January 9, 2015 Contents 1 Document

More information

Mediatrix 4404 Step by Step Configuration Guide June 22, 2011

Mediatrix 4404 Step by Step Configuration Guide June 22, 2011 Mediatrix 4404 Step by Step Configuration Guide June 22, 2011 Proprietary 2011 Media5 Corporation Table of Contents First Steps... 3 Identifying your MAC Address... 3 Identifying your Dynamic IP Address...

More information

Oracle Communications Session Border Controller. Accounting Guide Release S-CZ7.1.2

Oracle Communications Session Border Controller. Accounting Guide Release S-CZ7.1.2 Oracle Communications Session Border Controller Accounting Guide Release S-CZ7.1.2 May 2015 Notices Copyright 2014, 2013, Oracle and/or its affiliates. All rights reserved. This software and related documentation

More information

Application Notes for Configuring Broadvox SIP Trunking with Avaya IP Office - Issue 1.0

Application Notes for Configuring Broadvox SIP Trunking with Avaya IP Office - Issue 1.0 Avaya Solution & Interoperability Test Lab Application Notes for Configuring Broadvox SIP Trunking with Avaya IP Office - Issue 1.0 Abstract These Application Notes describe the procedures for configuring

More information

IP Office Technical Tip

IP Office Technical Tip IP Office Technical Tip Tip no: 188 Release Date: September 27, 2007 Region: GLOBAL Verifying IP Office SIP Trunk Operation IP Office back-to-back SIP Line testing IP Office Release 4.0 supports SIP trunking.

More information

SIP Trunking Service Configuration Guide for Skype

SIP Trunking Service Configuration Guide for Skype SIP Trunking Service Configuration Guide for Skype NDA-31154 Issue 1.0 NEC Corporation of America reserves the right to change the specifications, functions, or features at any time without notice. NEC

More information

bintec Workshop WAN Partner Configuration Copyright November 8, 2005 Funkwerk Enterprise Communications GmbH Version 0.9

bintec Workshop WAN Partner Configuration Copyright November 8, 2005 Funkwerk Enterprise Communications GmbH Version 0.9 bintec Workshop WAN Partner Configuration Copyright November 8, 2005 Funkwerk Enterprise Communications GmbH Version 0.9 Purpose Liability Trademarks Copyright Guidelines and standards How to reach Funkwerk

More information

NetComm V90 VoIP Phone Quick Start Guide Draft Release 0.1

NetComm V90 VoIP Phone Quick Start Guide Draft Release 0.1 NetComm V90 VoIP Phone Quick Start Guide Draft Release 0.1 Copyright NetComm Ltd Overview NetComm V90 SIP VoIP Phone User Guide Table of Contents Overview... 3 V90 VoIP Phone Specification...4 Shipping

More information

Application Notes Rev. 1.0 Last Updated: February 3, 2015

Application Notes Rev. 1.0 Last Updated: February 3, 2015 SBC 1000/2000 Series Configuration Guide with Cisco Unified Call Manager v8.6 for Level 3 Voice Complete SM Deployments Application Notes Rev. 1.0 Last Updated: February 3, 2015 Contents 1 Document Overview...

More information

SIP Trunking Service Configuration Guide for MegaPath

SIP Trunking Service Configuration Guide for MegaPath Notice Note that when converting this document from its original format to a.pdf file, some minor font and format changes may occur. When viewing and printing this document, we cannot guarantee that your

More information

AV@ANZA Formación en Tecnologías Avanzadas

AV@ANZA Formación en Tecnologías Avanzadas SISTEMAS DE SEÑALIZACION SIP I & II (@-SIP1&2) Contenido 1. Why SIP? Gain an understanding of why SIP is a valuable protocol despite competing technologies like ISDN, SS7, H.323, MEGACO, SGCP, MGCP, and

More information

ESI SIP Trunking Installation Guide

ESI SIP Trunking Installation Guide ESI SIP Trunking Installation Guide 0450-1227 Rev. B Copyright 2009 ESI (Estech Systems, Inc.). Information contained herein is subject to change without notice. ESI products are protected by various U.S.

More information

MITEL SIP CoE Technical. Configuration Note. Configure MCD for use with Thinktel SIP Trunking Service. SIP CoE 12-4940-00197

MITEL SIP CoE Technical. Configuration Note. Configure MCD for use with Thinktel SIP Trunking Service. SIP CoE 12-4940-00197 MITEL SIP CoE Technical Configuration Note Configure MCD for use with SIP Trunking Service SIP CoE NOTICE The information contained in this document is believed to be accurate in all respects but is not

More information

Abstract. Avaya Solution & Interoperability Test Lab

Abstract. Avaya Solution & Interoperability Test Lab Avaya Solution & Interoperability Test Lab Application Notes for Configuring SIP Trunking Using Verizon Business IP Contact Center VoIP Inbound and Avaya IP Office Release 8 with Avaya Session Border Controller

More information

Creating your own service profile for SJphone

Creating your own service profile for SJphone SJ Labs, Inc. 2005 All rights reserved SJphone is a registered trademark. No part of this document may be copied, altered, or transferred to, any other media without written, explicit consent from SJ Labs

More information

EZLoop IP-PBX Enterprise SIP Server

EZLoop IP-PBX Enterprise SIP Server EZLoop IP-PBX Enterprise SIP Server Copyright 2007 Teletronics International, Inc. 2 Choke Cherry Road, Rockville, MD 20850 sales@teletronics.com www.teletronics.com CH1. Overview...4 1.1 Specifications...4

More information

Trapeze Networks Integration Guide

Trapeze Networks Integration Guide Trapeze Networks Integration Guide Revision Date 0.9 27 May 2009 Copyright 2007 amigopod Pty Ltd amigopod Head Office amigopod Pty Ltd Suite 101 349 Pacific Hwy North Sydney, NSW 2060 Australia ABN 74

More information

EE4607 Session Initiation Protocol

EE4607 Session Initiation Protocol EE4607 Session Initiation Protocol Michael Barry michael.barry@ul.ie william.kent@ul.ie Outline of Lecture IP Telephony the need for SIP Session Initiation Protocol Addressing SIP Methods/Responses Functional

More information

Configuring the Synapse SB67070 SIP Gateway from AT&T for Clearfly SIP Trunking. January 2013

Configuring the Synapse SB67070 SIP Gateway from AT&T for Clearfly SIP Trunking. January 2013 Configuring the Synapse SB67070 SIP Gateway from AT&T for Clearfly SIP Trunking January 2013 1 Introduction This guide was created to assist Synapse partners with configuring the Synapse SB67070 SIP Gateway

More information

SIP Trunking Service Configuration Guide for Broadvox Fusion

SIP Trunking Service Configuration Guide for Broadvox Fusion Notice Note that when converting this document from its original format to a.pdf file, some minor font and format changes may occur. When viewing and printing this document, we cannot guarantee that your

More information

Configuring SIP Support for SRTP

Configuring SIP Support for SRTP Configuring SIP Support for SRTP This chapter contains information about the SIP Support for SRTP feature. The Secure Real-Time Transfer protocol (SRTP) is an extension of the Real-Time Protocol (RTP)

More information

Voice over IP. Demonstration 1: VoIP Protocols. Network Environment

Voice over IP. Demonstration 1: VoIP Protocols. Network Environment Voice over IP Demonstration 1: VoIP Protocols Network Environment We use two Windows workstations from the production network, both with OpenPhone application (figure 1). The OpenH.323 project has developed

More information

MITEL SIP CoE. Technical. Configuration Notes. Configure the Mitel 3300 MCD 4.1 for use with Paetec Broadworks Softswitch. SIP CoE 08-4940-00035

MITEL SIP CoE. Technical. Configuration Notes. Configure the Mitel 3300 MCD 4.1 for use with Paetec Broadworks Softswitch. SIP CoE 08-4940-00035 MITEL SIP CoE Technical Configuration Notes Configure the Mitel 3300 MCD 4.1 for use with Broadworks Softswitch SIP CoE 08-4940-00035 NOTICE The information contained in this document is believed to be

More information

Feature and Technical

Feature and Technical BlackBerry Mobile Voice System for SIP Gateways and the Avaya Aura Session Manager Version: 5.3 Feature and Technical Overview Published: 2013-06-19 SWD-20130619135120555 Contents 1 Overview...4 2 Features...5

More information

Fonality. Optimum Business Trunking and the Fonality Trixbox Pro IP PBX Standard Edition V4.1.2- p13 Configuration Guide

Fonality. Optimum Business Trunking and the Fonality Trixbox Pro IP PBX Standard Edition V4.1.2- p13 Configuration Guide Fonality Optimum Business Trunking and the Fonality Trixbox Pro IP PBX Standard Edition V4.1.2- p13 Configuration Guide Fonality Table of Contents 1. Overview 2. SIP Trunk Adaptor Set-up Instructions 3.

More information

Configuration guide for Switchvox and Cbeyond.

Configuration guide for Switchvox and Cbeyond. Configuration guide for Switchvox and Cbeyond. This document will guide a Switchvox administrator through configuring the system to utilize Cbeyond s BeyondVoice with SIPconnect service. After you have

More information

Silent Monitoring and Recording Using Unified Communications Manager

Silent Monitoring and Recording Using Unified Communications Manager Silent Monitoring and Recording Using Unified Communications Manager Call centers are expected to guarantee the quality of customer service their agents provide to callers. To that end, the ability to

More information

Voxitas SIP Trunk Setup

Voxitas SIP Trunk Setup DSX Voxitas SIP Trunk Setup April 23, 2011 Issue 1.05 NEC Corporation of America 4 Forest Parkway, Shelton, CT 06484 Copyright 2011 NEC Corporation of America 6535 N. State Highway 161 Irving, TX 75039-2402

More information

nexvortex Setup Template

nexvortex Setup Template nexvortex Setup Template ZULTYS, INC. April 2013 5 1 0 S P R I N G S T R E E T H E R N D O N V A 2 0 1 7 0 + 1 8 5 5. 6 3 9. 8 8 8 8 Introduction This document is intended only for nexvortex customers

More information

Configuration Notes 290

Configuration Notes 290 Configuring Mediatrix 41xx FXS Gateway with the Asterisk IP PBX System June 22, 2011 Proprietary 2011 Media5 Corporation Table of Contents Introduction... 3 About Mediatrix 41xx Series FXS Gateways...

More information

IP PBX. SD Card Slot. FXO Ports. PBX WAN port. FXO Ports LED, RED means online

IP PBX. SD Card Slot. FXO Ports. PBX WAN port. FXO Ports LED, RED means online 1 IP PBX SD Card Slot FXO Ports PBX LAN port PBX WAN port FXO Ports LED, RED means online 2 Connect the IP PBX to Your LAN Internet PSTN Router Ethernet Switch FXO Ports 3 Access the PBX s WEB GUI The

More information

UX5000 with CommPartners SIP Trunks

UX5000 with CommPartners SIP Trunks UX5000 with CommPartners SIP Trunks SECTION 1 NEC S UX5000 AND CommPartners SETUP GUIDE This guide provides example entries for the required fields. The actual data will be e- mailed to you in the following

More information

IP Office Technical Tip

IP Office Technical Tip IP Office Technical Tip Tip no: 200 Release Date: January 23, 2008 Region: GLOBAL IP Office Session Initiation Protocol (SIP) Configuration Primer There are many Internet Telephony Service Providers (ITSP)

More information

SIP Trunking Service Configuration Guide for PAETEC (Broadsoft Platform)

SIP Trunking Service Configuration Guide for PAETEC (Broadsoft Platform) Notice Note that when converting this document from its original format to a.pdf file, some minor font and format changes may occur. When viewing and printing this document, we cannot guarantee that your

More information

Sample Configuration for SIP Trunking between Avaya IP Office R8.0 and Cisco Unified Communications Manager 8.6.2 Issue 1.0

Sample Configuration for SIP Trunking between Avaya IP Office R8.0 and Cisco Unified Communications Manager 8.6.2 Issue 1.0 Avaya Solution & Interoperability Test Lab Sample Configuration for SIP Trunking between Avaya IP Office R8.0 and Cisco Unified Communications Manager 8.6.2 Issue 1.0 Abstract These Application Notes describe

More information

SIP Trunk Configuration Guide. using

SIP Trunk Configuration Guide. using SIP Trunk Configuration Guide using www.cbeyond.net 1-877-441-9783 The information contained in this document is specific to setting up SIP connections between Vertical SBX IP 320 and Cbeyond. If you require

More information

Application Notes for Configuring Microsoft Office Communications Server 2007 R2 and Avaya IP Office PSTN Call Routing - Issue 1.0

Application Notes for Configuring Microsoft Office Communications Server 2007 R2 and Avaya IP Office PSTN Call Routing - Issue 1.0 Avaya Solution & Interoperability Test Lab Application Notes for Configuring Microsoft Office Communications Server 2007 R2 and Avaya IP Office PSTN Call Routing - Issue 1.0 Abstract These Application

More information

Application Notes for BT Wholesale/HIPCOM SIP Trunk Service and Avaya IP Office 8.0 Issue 1.0

Application Notes for BT Wholesale/HIPCOM SIP Trunk Service and Avaya IP Office 8.0 Issue 1.0 Avaya Solution & Interoperability Test Lab Application Notes for BT Wholesale/HIPCOM SIP Trunk Service and Avaya IP Office 8.0 Issue 1.0 Abstract These Application Notes describe the procedures for configuring

More information

Technical Manual 3CX Phone System for Windows

Technical Manual 3CX Phone System for Windows Technical Manual 3CX Phone System for Windows This technical manual is intended for those who wish to troubleshoot issues encountered with implementing 3CX Phone System. It is not intended to replace the

More information

GW400 VoIP Gateway. User s Guide

GW400 VoIP Gateway. User s Guide GW400 VoIP Gateway User s Guide P/N: 956YD30001 Copyright 2006. All Rights Reserved. Document Version: 1.0 All trademarks and trade names are the properties of their respective owners. i Table of Contents

More information

SIP Trunking using Optimum Business SIP Trunk Adaptor and the Cisco Call Manager Express Version 8.5

SIP Trunking using Optimum Business SIP Trunk Adaptor and the Cisco Call Manager Express Version 8.5 CISCO SIP Trunking using Optimum Business SIP Trunk Adaptor and the Cisco Call Manager Express Version 8.5 Goal The purpose of this configuration guide is to describe the steps needed to configure the

More information

Mediatrix 3000 with Asterisk June 22, 2011

Mediatrix 3000 with Asterisk June 22, 2011 Mediatrix 3000 with Asterisk June 22, 2011 Proprietary 2011 Media5 Corporation Table of Contents Introduction... 3 Network Topology... 3 Equipment Detail... 3 Configuration of the Fax Extension... 4 Configuration

More information

Media Gateway Controller RTP

Media Gateway Controller RTP 1 Softswitch Architecture Interdomain protocols Application Server Media Gateway Controller SIP, Parlay, Jain Application specific Application Server Media Gateway Controller Signaling Gateway Sigtran

More information

An Introduction to VoIP Protocols

An Introduction to VoIP Protocols An Introduction to VoIP Protocols www.netqos.com Voice over IP (VoIP) offers the vision of a converged network carrying multiple types of traffic (voice, video, and data, to name a few). To carry out this

More information

6.40A AudioCodes Mediant 800 MSBG

6.40A AudioCodes Mediant 800 MSBG AudioCodes Mediant 800 MSBG Page 1 of 66 6.40A AudioCodes Mediant 800 MSBG 1. Important Notes Check the SIP 3 rd Party Validation Website for current validation status. The SIP 3 rd party Validation Website

More information

Prestige 310. Cable/xDSL Modem Sharing Router. User's Guide Supplement

Prestige 310. Cable/xDSL Modem Sharing Router. User's Guide Supplement Prestige 310 Cable/xDSL Modem Sharing Router User's Guide Supplement Domain Name Support Enhanced WAN Setup Remote Node Support PPPoE Support Enhanced Unix Syslog Setup Firmware and Configuration Files

More information

Mediatrix Gateway 440x Series Quick Configuration Guide

Mediatrix Gateway 440x Series Quick Configuration Guide Mediatrix Gateway 440x Series Quick Configuration Guide All BRI Mediatrix gateways are pre-configured on ETH1 port with DHCP and ETH2 port with static IP 192.168.0.10. All PRI Mediatrix gateways are pre-configured

More information

MITEL SIP CoE. Technical. Configuration Note. Configure MCD for use with Intelepeer Service provider SIP Trunking. SIP CoE 14-4940-00313

MITEL SIP CoE. Technical. Configuration Note. Configure MCD for use with Intelepeer Service provider SIP Trunking. SIP CoE 14-4940-00313 MITEL SIP CoE Technical Configuration Note Configure MCD for use with Intelepeer Service provider SIP Trunking SIP CoE 14-4940-00313 NOTICE The information contained in this document is believed to be

More information

Session Border Controller

Session Border Controller CHAPTER 13 This chapter describes the level of support that Cisco ANA provides for (SBC), as follows: Technology Description, page 13-1 Information Model Objects (IMOs), page 13-2 Vendor-Specific Inventory

More information

Unit 23. RTP, VoIP. Shyam Parekh

Unit 23. RTP, VoIP. Shyam Parekh Unit 23 RTP, VoIP Shyam Parekh Contents: Real-time Transport Protocol (RTP) Purpose Protocol Stack RTP Header Real-time Transport Control Protocol (RTCP) Voice over IP (VoIP) Motivation H.323 SIP VoIP

More information

Let's take a look at another example, which is based on the following diagram:

Let's take a look at another example, which is based on the following diagram: Chapter 3 - Voice Dial Peers In order to understand the concept of dial peers, it is important to understand call legs. A voice call over a packet network is segmented into discrete call legs. A call leg

More information

This specification this document to get an official version of this User Network Interface Specification

This specification this document to get an official version of this User Network Interface Specification This specification describes the situation of the Proximus network and services. It will be subject to modifications for corrections or when the network or the services will be modified. Please take into

More information

MyIC setup and configuration (with sample configuration for Alcatel Lucent test environment)

MyIC setup and configuration (with sample configuration for Alcatel Lucent test environment) MyIC setup and configuration (with sample configuration for Alcatel Lucent test environment) N.B. Goto MyIC Preferences in the System Toolbar. Description: this may be any appropriate description of the

More information

KX-UT113/KX-UT123 KX-UT133/KX-UT136 KX-UT248

KX-UT113/KX-UT123 KX-UT133/KX-UT136 KX-UT248 Model No. SIP Phone KX-UT113/KX-UT123 KX-UT133/KX-UT136 KX-UT248 Thank you for purchasing this Panasonic product. Please read this manual carefully before using this product and save this manual

More information

SV9100 SIP Trunking Service Configuration Guide for Time Warner Cable Business Class

SV9100 SIP Trunking Service Configuration Guide for Time Warner Cable Business Class SV9100 SIP Trunking Service Configuration Guide for Time Warner Cable Business Class NDA-31660 Issue 1.0 NEC Corporation of America reserves the right to change the specifications, functions, or features

More information

VoIP Application Note:

VoIP Application Note: VoIP Application Note: Configure NEC UX5000 w/ BroadVox SIP Trunking Service P/N 0913226 Date: 8/12/09 Table of Contents: GOAL... 3 PREREQUISITES... 3 SIP TRUNKING INFORMATION PROVIDED BY BROADVOX:...

More information

Release Notes for NeoGate TE100 16.18.0.X

Release Notes for NeoGate TE100 16.18.0.X Release Notes for NeoGate TE100 16.18.0.X ===Firmware Version: V16.18.0.2==== Applicable Model: NeoGate TE100 Release Date: October 25th, 2014 http://www.yeastar.com 1/6 1. New Features 1. Added support

More information

Link Gate SIP. (Firmware version 1.20)

Link Gate SIP. (Firmware version 1.20) Link Gate SIP (Firmware version 1.20) User guide v1.0 1 Content 2 1. Technical parameters - Dimensions 133 x 233 x 60 mm - Weight 850 g - Operating position various - Operating condition temperature: +5

More information

EarthLink Business SIP Trunking. NEC SV8100 IP PBX Customer Configuration Guide

EarthLink Business SIP Trunking. NEC SV8100 IP PBX Customer Configuration Guide EarthLink Business SIP Trunking NEC SV8100 IP PBX Customer Configuration Guide Publication History First Release: Version 1.0 August 30, 2011 CHANGE HISTORY Version Date Change Details Changed By 1.0 8/30/2011

More information

Avaya IP Office SIP Trunk Configuration Guide

Avaya IP Office SIP Trunk Configuration Guide Valcom Session Initiation Protocol (SIP) VIP devices are compatible with SIP-enabled versions of Avaya IP Office (5.0 and higher). The Valcom device can be added to the IP Office system as a SIP Trunk.

More information

ARCHITECTURES TO SUPPORT PSTN SIP VOIP INTERCONNECTION

ARCHITECTURES TO SUPPORT PSTN SIP VOIP INTERCONNECTION ARCHITECTURES TO SUPPORT PSTN SIP VOIP INTERCONNECTION 10 April 2009 Gömbös Attila, Horváth Géza About SIP-to-PSTN connectivity 2 Providing a voice over IP solution that will scale to PSTN call volumes,

More information

VoIP Server Reference

VoIP Server Reference IceWarp Server VoIP Server Reference Version 10 Printed on 12 August, 2009 i Contents VoIP Service 1 Introduction... 1 V10 New Features... 3 SIP REFER... 3 SIP Call Transfer Agent Settings... 3 NAT traversal

More information

Chapter 29 User Authentication

Chapter 29 User Authentication Chapter 29 User Authentication Introduction...29-3 Privilege Levels...29-3 User Level... 29-3 Manager Level... 29-4 Security Officer Level... 29-5 Remote Security Officer Level... 29-6 Operating Modes...29-6

More information

Application Notes for Configuring Avaya IP Office 9.0 with HIPCOM SIP Trunk Issue 1.0

Application Notes for Configuring Avaya IP Office 9.0 with HIPCOM SIP Trunk Issue 1.0 Avaya Solution & Interoperability Test Lab Application Notes for Configuring Avaya IP Office 9.0 with HIPCOM SIP Trunk Issue 1.0 Abstract These Application Notes describe the procedures for configuring

More information

How To Connect A Phone To An Ip Trunk On A Cell Phone On A Sim Sim Simlia (Vizon) Or Ip Office (Izon) On A Ppl (Telnet) On An Ip Office Softphone On A Vnet (V

How To Connect A Phone To An Ip Trunk On A Cell Phone On A Sim Sim Simlia (Vizon) Or Ip Office (Izon) On A Ppl (Telnet) On An Ip Office Softphone On A Vnet (V Avaya Solution & Interoperability Test Lab Application Notes for SIP Trunking Using Verizon Business IP Trunk SIP Trunk Service and Avaya IP Office Release 7.0 Issue 1.1 Abstract These Application Notes

More information

SIP Trunking Application Notes V1.3

SIP Trunking Application Notes V1.3 SIP Trunking Application Notes V1.3 Publication Information SAMSUNG TELECOMMUNICATIONS AMERICA reserves the right without prior notice to revise information in this publication for any reason. SAMSUNG

More information

Session Initiation Protocol Gateway Call Flows and Compliance Information

Session Initiation Protocol Gateway Call Flows and Compliance Information Session Initiation Protocol Gateway Call Flows and Compliance Information Corporate Headquarters Cisco Systems, Inc. 170 West Tasman Drive San Jose, CA 95134-1706 USA http://www.cisco.com Tel: 408 526-4000

More information

VoIP Interkonnektion Test Specification

VoIP Interkonnektion Test Specification VoIP Interkonnektion Specification Ausgabedatum 310.2015 Ersetzt Version - Gültig ab 012015 Vertrag Vertrag betreffend Verbindung von VoIP Fernmeldeanlagen und -diensten Gültig ab 012015 1/21 Table of

More information

Connecting with Free IP Call

Connecting with Free IP Call Connecting with Free IP Call Free IP Call (http://www.freeipcall.com/) offers telephone service using the VoIP standard SIP. The service allow users making/receiving VoIP calls to/from VoIP telephone numbers

More information

Updated Since : 2007-02-09

Updated Since : 2007-02-09 Microsoft Exchange Server 2007 Unified Messaging PBX Configuration Note: Avaya S8300 with AudioCodes Mediant 2000 using T1 CAS (In-band DTMF Tones) By : AudioCodes Updated Since : 2007-02-09 READ THIS

More information

MITEL SIP CoE. Technical. Configuration Notes. Configure MCD 4.1 for use with SKYPE SIP Trunking. SIP CoE 10-4940-00120

MITEL SIP CoE. Technical. Configuration Notes. Configure MCD 4.1 for use with SKYPE SIP Trunking. SIP CoE 10-4940-00120 MITEL SIP CoE Technical Configuration Notes Configure MCD 4.1 for use with SKYPE SIP Trunking SIP CoE 10-4940-00120 NOTICE The information contained in this document is believed to be accurate in all respects

More information

DSX. ATC SIP Trunk Setup. April 22, 2011 Issue 1.01. NEC Corporation of America 4 Forest Parkway, Shelton, CT 06484

DSX. ATC SIP Trunk Setup. April 22, 2011 Issue 1.01. NEC Corporation of America 4 Forest Parkway, Shelton, CT 06484 DSX ATC SIP Trunk Setup April 22, 2011 Issue 1.01 NEC Corporation of America 4 Forest Parkway, Shelton, CT 06484 Copyright 2011 NEC Corporation of America 6535 N. State Highway 161 Irving, TX 75039-2402

More information