VoIP Provider Configuration

Data Collection for HiPath 3000 and Op Scape Office MX

Version: 1.0.2 Date: 2010-09-29 SEN VA SME SD 33 Siemens Enterprise Communications GmbH & Co. KG

Communication for the open minded

Siemens Enterprise Communications www.siemens.com/open

Table of Content
1 2 2.1 2.2 2.3 3 4 5 5.1 5.1.1 5.1.2 5.1.3 5.1.4 5.1.5 5.1.6 5.2 5.2.1 5.2.2 5.3 5.3.1 5.3.2 5.3.3 5.3.4 5.3.5 5.3.6 5.3.7 5.3.8 5.3.9 5.3.10 5.3.11 5.3.12 6 6.1 6.2 Scope.......................................................................................................................................... 3 List of Configuration parameters ........................................................................................... 3 Basic ITSP Configuration........................................................................................................ 4 User Account configuration..................................................................................................... 4 Special configuration SIP Profile Data................................................................................ 5 ITSP questionaire..................................................................................................................... 8 Restrictions and known limitations ........................................................................................ 8 Details for ITSP configuration parameters.......................................................................... 10 Basic ITSP configuration....................................................................................................... 10 Provider Identification / Domain .......................................................................................... 10 Provider Registrar ................................................................................................................. 10 Provider Proxy ...................................................................................................................... 11 Provider Outbound Proxy ..................................................................................................... 12 Provider STUN ..................................................................................................................... 12 DDI or MSN type of ITSP .................................................................................................... 13 Account configuration............................................................................................................ 13 Registration information ....................................................................................................... 13 Call numbers ......................................................................................................................... 13 Special configuration - SIP Profile Data .............................................................................. 14 Format of From:, PAI: and PPI: for Basic Call..................................................................... 14 Diverted Calls: Format of From:........................................................................................... 15 Diverted Calls: Format of Diversion:.................................................................................... 15 Anonymous Calls: Format of From: ..................................................................................... 16 Anonymous Calls: Format of Privacy:.................................................................................. 16 Call number formatting......................................................................................................... 16 Settings for Registration: ...................................................................................................... 17 Parameter for Authentication:............................................................................................... 17 Routing parameter:................................................................................................................ 17 Media handling: .................................................................................................................... 18 Supported Methods in sending direction............................................................................... 18 Supported Methods in receiving direction ............................................................................ 18 Modifying the ITSPs profile settings................................................................................... 19 HG1500 (HiPath 3000)........................................................................................................... 19 OpenScape Office MX ........................................................................................................... 23

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Scope

This document will provide a guideline how to configure OpenScape Office MX V2 and HiPath 3000 V8 for a new Internet Telephony Service Provider (ITSP). For this purpose it guides through collecting relevant data from the ITSP and gives explanation on the SIP functionality of the system. The content of the configuration lists is necessary for successful interworking with the ITSP and to implement the new provider interface in the OpenScape Office MX and HiPath 3000 administration. The document is structured in the parts: Comprehensive table with all parameters which are needed for the ITSP connection ITSP questionnaire List of known restrictions and limitations Detailed description of the ITSP configuration parameters Guideline for entering profile data in HG1500 or OpenScape Office
Anastasios Gkinis Siemens Enterprise Communications S.A. SME QA - SIP Certification 15, Metaxa Str. GR 145 64 Nea Kifissia / Athens, Greece Tel.: 0030 210 8196419 Fax: 0030 210 8075412 mailto : anastasios.ginis@siemens-enterprise.com

For all questions regarding certification and test please contact:

List of Configuration parameters

1. Basic configuration: information how to connect to the SIP infrastructure of the ITSP. This information may change due to new SIP servers at the ITSP and can also be changed by the end user. 2. User Account configuration: information about the user account. This information is provided by the ITSP and has to be entered by the end user. 3. Special configuration: information on specific needs inside the SIP protocol. This parameters are configured to meet the specific requirements of a certain provider and should not be changed by the end user.

For a connection to an ITSP various configuration parameters are needed according to the profile of the used ITSP. The parameters can be divided into three major parts:

Please enter the data of your ITSP into the following lists. For detailed explanations for all parameters, see chapter 5.

This document contains examples for SIP messages and how they are influenced by the configured profile values. In these examples the following placeholders for call numbers or accounts are used:
Calling party number Calling party account Called party number Called party account 023026672695 sip-acc1 004970070 sip-acc2

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The list contains entries written in red which needs to be filled or confirmed and entries in black which must not be changed, see explanation for details.

2.1

Basic ITSP Configuration

Configured value New provider sip.provider.com true false

Provider Name Domain Name Provider Registrar Use Registrar IP Address / Host name Port Reregistration Interval at Provider (sec) Provider Proxy Use Proxy IP Address / Host name Port Provider Outbound Proxy Use Outbound Proxy IP Address / Host name Port Provider STUN IP Address / Host name Port Type of ITSP DDI or MSN type of ITSP

true 0.0.0.0 5060 120 true 0.0.0.0 5060

true false

false 0.0.0.0 0 0.0.0.0 3478

MSN-Subscriber DDI-Subscriber

MSN-Subscriber

2.2

User Account configuration

Term used in ITSP documentation, web portal .
e.g. SIP-ID e.g. SIP-ID or Account-name or empty e.g. SIP-Password e.g. Display name

Please document the wording which is used by the ITSP.

Term used in WBM Internet telephony station Authorization name Password Internet telephony station number

Please document the required format here.

Call number format E.164 International number E.164 International number E.164 national number Other format used by ITSP (e.g. +4923026672695) (e.g. 004923026672695) (e.g. 023026672695) (e.g. 23026672695)

This covers format only, the individual account data is customer specific.

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2.3

Special configuration SIP Profile Data

Possible values 0 = omit 1 = CallNo 2 = Account Configured value

Configuration for From: DisplayPart<sip:UserPart@HostPart>

From-DisplayPart
(P_SipProvProfileCLIPFromNameAddr)

From: <sip: From: 023026672695 <sip:... From: sip-acc1 <sip:... From-UserPart

(P_SipProvProfileCLIPFromAddrSpec)

From: ... <sip:023026672695@... From: ... <sip:sip-acc1@... From-HostPart

(P_SipProvProfileFromURIDomain)

1 = CallNo 2 = Account

From: ... <sip:...@192.168.138.1>;... From: ... <sip:...@domainName>;...

0 = localIPAddr 1 = DomainName

Configuration for P-Asserted-Identity: DisplayPart<sip:UserPart@domain>

PAI-DisplayPart
(P_SipProvProfileCLIPPAssertedIdNamedAddr)

P-Asserted-Identity: <sip:... P-Asserted-Identity: 023026672695 <sip:... P-Asserted-Identity: sip-acc1 <sip:... PAI-UserPart

(P_SipProvProfileCLIPPAssertedIdAddrSpec)

0 = omit 1 = CallNo 2 = Account

No PAI header generated P-Asserted-Identity: ... <sip:023026672695@... P-Asserted-Identity: ... <sip:sip-acc1@...

0 = omit 1 = CallNo 2 = Account

Configuration for P-Preferred-Identity: DisplayPart<sip:UserPart@domain>

PPI-DisplayPart
(P_SipProvProfileCLIPPPreferredNameAddr)

P-Preferred-Identity: <sip:... P-Preferred-Identity: 023026672695 <sip:... P-Preferred-Identity: sip-acc1 <sip:... PPI-UserPart

(P_SipProvProfileCLIPPPreferredAddrSpec)

0 = omit 1 = CallNo 2 = Account

No PPI header generated P-Preferred-Identity: ... <sip:023026672695@... P-Preferred-Identity: ... <sip:sip-acc1@...

0 = omit 1 = CallNo 2 = Account

Configuration for Diversion: DisplayPart<sip:UserPart@domain>

Diversion-DisplayPart
(P_SipProvProfileDiversionNameAddr)

Possible values 0 = omit 1 = CallNo 2 = Account

Configured value

Diversion: <sip: Diversion: 023026672695 <sip:... Diversion: sip-acc1 <sip:... Diversion-UserPart

(P_SipProvProfileDiversionAddrSpec)

No Diversion header generated Diversion: ... <sip:023026672695@... Diversion: ... <sip:sip-acc1@...

0 = omit 1 = CallNo 2 = Account

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diverted calls: contents of From:

OriginalCallingNumber in From
(P_SipProvProfileRedirNrInFrom)

From: sip:A-number@... The outgoing invite contains INVITE C From: A To: C Diversion: B From: sip:B-number@... The outgoing invite contains INVITE C From: B To: C Diversion: B From-AnonymousDisplayPart
(P_SipProvProfileCLIRFromNameAddr)

true

false

false

Anonymous calls (number restricted) contents of From:

From: From: From: From:

<sip:... 023026672695 <sip:... sip-acc1 <sip:... Anonymous <sip:...

0 = omit 1 = CallNo 2 = Account 3 = Anonymous

From-AnonymousUserPart
(P_SipProvProfileCLIRFromAddrSpec)

From: ... <sip:023026672695@... From: ... <sip:sip-acc1@... From: ... <sip:anonymous@anonymous.invalid From: ... <sip:anonymous@... Send Privacy header
(P_SipProvProfileCLIRPrivacy)

1 = CallNo 2 = Account 3 = sipURIAnonymous 4 = sipURIAnonymousUser

Anonymous calls (number restricted) contents of Privacy:

Header not generated Privacy: id

0 = omit 1 = Id

Call number formatting

Incoming call Called party number is taken from:
(P_SipProvProfileCalledPartyNumber)

Possible values 0 = requestLine 1 = ToNameAddr 2 = ToAddrSpec

Configured value

INVITE sip: 004970070@... SIP/2.0 To: 004970070 <sip:...@....> To: <sip:004970070@....> Incoming call Calling party number is taken from:
(P_SipProvProfileCallingPartyNumber)

From: 023026672695 <sip:...@...> From: ... <sip:023026672695@...> P_Asserted_Identity: 023026672695 <sip:...@...> P_Asserted_Identity: <sip:023026672695@...> Incoming call Calling party Type of Number (TON):
(P_SipProvProfileCallingPartyNumberTON)

0 = automatic 1 = FromNameAddr 2 = FromAddrSpec 3 = PAssertedIdNameAddr 4 = PAssertedIdAddrSpec

number is classified automatically number is classified as international number

0 = automatic 1 = international

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Settings for Registration

Contact header setting
(P_SipProvProfileRegContactAddr)

Contact: <sip:@192.168.138.1:5060>; Contact: <sip:@tel.t-online.de>; Contact header setting

(P_SipProvProfileContactUriWithProtocol)

0 = localIPAddr 1 = domainName

Contact: <sip:...@...>;transport=udp; +u.www.siemens.com/icn/en/oscar/sip/sec-level false = w/o Transport & OSC parameter

true false false

Parameter for Authentication

Authentication header handling (P_SipProvProfileUseRouteURIAuthentication) use the route URI handling true use the request URI (always) false

true

Routing parameter
Include rport in Via: header
(P_SipProvProfileUseViaRPort)

Via: SIP/2.0/UDP 192.168.138.1:5060;rport; Via: SIP/2.0/UDP 192.168.138.1:5060

true false

true

Media handling
Treatment of received 100rel in supported header field
(P_SipProvProfileIgnore100Rel)

ignore a received Supported: 100rel header Direct payload connection

(P_SipProvProfileDirectPayload)

true false

true

Payload is transported directly between endpoints Paylaod is terminated by RTP-Proxy

true false

false

Supported methods / sending direction

Support of UPDATE method
(P_SipProvProfileUPDATESupported)

system will never send UPDATE

true false

true

Supported methods / Receiving direction

Support of REFER
(P_SipProvProfileREFERAllowed)

REFER is accepted REFER is rejected with 405 Method not allowed

true false

false

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ITSP questionaire

This questionnaire is intended to identify the basic requirements for interconnecting the Enterprise communication system with a SIP Service Provider. This table is for information purposes only, if the information is not available please skip the question. The table is filled with the default values which are supported by the system, please change if necessary.
Information provided by ITSP General Info ITSP company name Country Homepage / URL Used SIP products Please list the relevant products used in the SIP core network (proxy, registrar, mediagateways) Engineering contact person(s) Sales contact person interface specification Transport UDP TCP TLS IPSec Tunnel Codecs G.711 a-law G.711 -law G.729A G.729A/B Other codecs Fax G.711 based fax T.38 DTMF transmission RFC2833 / Inband / INFO Security Authentication (e.g. Digest Authentication, other) Special Features Does ITSP perform Session refresh / Session supervision by sending: Sessiontimer - UPDATE / Sessiontimer - reINVITE / OPTIONS / INFO Proxy or B2BUA, Mediahandling Forking Transfer with REFER The Company Germany www.xyz.net Sonus SW 7.2.4

Provide url here yes no no no yes yes yes yes (Sample (Sample (Sample (Sample rate rate rate rate = = = = 20ms) 20ms) 20ms) 20ms)

no yes Yes / - / Digest Authentication - / - / - / B2BUA no no

Restrictions and known limitations

Due to different reasons, HiPath and OpenScape Office do not support some features, which may be offered by an ITSP. This section contains a list of feature limitations at the ITSP access.

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Basic Call using LateSDP For 3rd party call control some ITSPs use the so called lateSDP-call where the system receives an initial INVITE without SDP. This feature is released in HiPath 3000 V8 starting with MR4 and will be supported in OSO V3. Media Types: HiPath and OpenScape Office offers support for the media types audio (Voice and voice band data) and image (fax) on the ITSP interface. All other media types (e.g. Video) are NOT supported on the ITSP interface. This limitation does not apply to the internal SIP interfaces. DTMF: The system supports DTMF according to the RFC2833 standard. Sending and receiving of DTMF-relay in the body of an INFO or NOTIFY message is NOT supported Forking: HiPath and OpenScape Office does not support forked calls, if the called party that connects the call does not send a provisional response (e.g. 180 Ringing) before connect (200 OK). (Receiving a 180 from target A followed by a 200 OK from a different target is not supported) Early media in combination with forking The HiPath and OpenScape Office systems cannot handle early media in combination with forking. Most ITSP filter out early media in responses and Supported: 100rel in sent INVITE messages, if a call was forked. Subscribe/Notify for MessageWaitingIndication As HiPath and OpenScape Office systems have internal voicemail systems they do not support subscriptions for message waiting to public voicemail systems provided by the ITSP. Transfer and redirection REFER and 302 (Redirect/Diversion) handling are deactivated on the ITSP interface due to security considerations. Handling of these messages would mean to create new calls to a number that was provided by an external, possibly untrusted, party, which may result in high costs (toll fraud). The HiPath/ OpenScape Office system does not send REFER or 302 to the ITSP leg, too. STUN The STUN mode and used STUN servers are the same for all active ITSP. It is not possible to enable STUN for one ITSP only. If no STUN server is configured for an ITSP, the system uses a server of a deactivated ITSP. To switch off STUN it is necessary to set the STUN mode to Off. Feature activation using Keypad Some ITSP allow feature activation by sending of the characters * and/or # followed by a featurecode (stimulus feature activation). Using * and # on the ITSP interface has to be explicitly allowed by the system configuration (system flag), which is available for HiPath 3000 and planned for OpenScape Office V3. In a SIP-URI "#" is an invalid character which has to be escaped. Therefore the ASCII representation in Hex is sent on the line: # is 0x23 -> %23 in SIP-URI

Note: If features are invoked by stimulus procedures, no indication might be given to the user (e.g. no special dial tone, no display information). Thus using stimulus features is not recommended and requires a careful handling by the user.

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Usage of + in callnumbers The system supports receiving of a call number starting with +. Nevertheless the system does NOT support sending of numbers starting with +.

Details for ITSP configuration parameters

The following chapter explains all parameters for basic configuration, user account configuration and special configuration in detail. In the SIP examples the following placeholders for call numbers or accounts are used:
Calling party number Calling party account Called party number Called party account 023026672695 sip-acc1 004970070 sip-acc2

5.1
5.1.1

Basic ITSP configuration

Provider Identification / Domain This is a unique name which is used in the WBM of the system to identify the ITSP. The name has no effect on the SIP protocol.

Provider Name

Domain Name The domain name is provided by the ITSP. It can contain a name or an IP-address. Usage examples: The configured Domain Name is used in the Host part of e.g. From:, To: and PAI-header fields
From: sip:023026672695@DomainName To: sip: 023026672695@DomainName P-Asserted-Identity: sip: 023026672695@DomainName

5.1.2 Provider Registrar Registration is used by ITSP for two different purposes. 1. Addressing: With the registration the ITSPs Registrar is informed about the IPAddress of the system. This is useful if the system is located behind an internet access using a dynamic IP address. 2. Monitoring: Registrations have to be repeated periodically. The ITSPs Registrar will monitor the registrations and thus knows about availability of the system. If ITSP provide an infrastructure with static IP addresses and availability is monitored by other means a Registration might not be necessary. Thus using a Registrar is configurable with the following parameters: Use Registrar Default : true

With this flag registration for an ITSP is activated. If the ITSP works without registration this flag is set to false and the following data are ignored. IP Address / Host name Default : 0.0.0.0

The address of the ITSPs registrar server. This can be an IP-address or the host name.

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For a better flexibility ITSPs usually provide a FQDN here (e.g. sip.provider.com). If your ITSP provides you with a fixed IP address, please clarify if this is an intermediate solution or if they intend to stick to this server address. If your provider is using geographically separated servers, each to be used in a certain location (e.g. sip-south.provider.com, sip-north.provider.com) please list all possible servers in the table. It has to be decided if one of these servers is preconfigured as a default server or if the field is left empty so that the end user has to enter the correct server address. Port Port of the ITSP registrar. If DNSSRV shall be used to query the IP address the port MUST be set to 0 In case of DNSSRV the registrar host name is usually the same as the ITSP domain name. Reregistration Interval at Provider (sec) Default(sec) : 120 Default : 5060

Registrations have to be repeated in regular intervals. The system proposes the configured interval which is sent in the expires header field. The ITSP can accept this value or answers with a different one which will be used instead. Note: The configuration of this timer will affect the amount of messages sent periodically and the time in which a loss of connection is detected. A short registration interval has the advantage that a loss of connection to the ITSP is detected after a short time, but this needs more messages to sent on the interface. If a long interval is configured it takes a long time to detect the connection loss. Note that if STUN is used it monitors the connection to the STUN server (and thus the internet connection in general). If it detects an IP address change (e.g. due to DSL reconnect) or loss of connection, it unregisters the old IP address and registers the new one at the ITSP. Usage examples: The configured data will be used to perform the registration of the user accounts (see also user account data 5.2)
REGISTER sip:RegistrarIp/HostName:RegistrarPort;transport=udp SIP/2.0 Via: SIP/2.0/UDP SystemIp:SystemPort;rport;branch=z9hG4bK122b40148b4a2d146 ... Contact: <sip:UserAccount@SystemIp:SystemPort>;expires=Reregistrationinterval

The values of SystemIp and SystemPort depend on the deployment of the system. They are taken either from the IP configuration or will be determined using the STUN protocol. For details see Background information about STUN and Network configuration 5.1.3 Provider Proxy true With this flag usage of a proxy server is activated. This flag cannot be configured and MUST be set to true. IP Address / Host name Default : 0.0.0.0

Use Proxy

The address of the ITSPs proxy server. This can be an IP-address or the host name. Please see the comments for IP Address listed for Provider Registrar too. Port Default : 5060

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Port of the ITSP registrar. If DNSSRV shall be used to query the IP address of the proxy server the port MUST be set to 0 In case of DNSSRV the registrar host name is usually the same as the ITSP domain name. Usage examples: The configured data will be used to address the ITSPs SIP server
INVITE sip:004970070@ProviderProxyIp/HostName:ProviderProxyPort SIP/2.0

5.1.4

Provider Outbound Proxy Default : false In some rare cases an outbound proxy may be used. Usually no outbound proxy is used for ITSP connections and the flag SHOULD be set to false. If your deployment requires an outbound proxy, please give some background information about the deployment of the ITSP. With this flag usage of an outbound proxy is activated.

Use Outbound Proxy

IP Address / Host name

Default : 0.0.0.0

The address of the ITSPs outbound proxy server. This can be an IP-address or the host name. Port Default : 0 Port of the ITSP outbound proxy server. If DNSSRV shall be used to query the IP address of the proxy server the port MUST be set to 0 In case of DNSSRV the registrar host name is usually the same as the ITSP domain name. Usage examples: If an outbound proxy is used, all SIP messages are sent via this proxy. Note: The SIP Request-URI and the header fields contain the data configured for SIP proxy. In addition a Route: header field containing the outbound proxy is sent
INVITE sip:004970070@ProviderProxyIp/HostName:ProviderProxyPort SIP/2.0 Route: sip:ProviderOutboundProxyIp/HostName:ProviderOutboundProxyPort

5.1.5 Provider STUN STUN may be needed, if the system is connected via an external router. It depends on the deployment of the ITSP if STUN has to be used. Some ITSPs provide so called far end NAT traversal where STUN is not needed to traverse the router. Thus it has to be checked with the ITSP if STUN is required and/or if a STUN server is provided.

If STUN is required for a certain provider, we MUST have an entry for the STUN server. A certification is NOT possible without having the information about the STUN server to be used. This STUN server might be provided by the provider itself or by a cooperation partner.

Note: If you do not provide a STUN server address and operate with STUN activated, the system will search for an alternative STUN server. Usually call establishment will be possible without any problems, but due to possible delays in the network, the service may be limited.

If the ITSP provides far end NAT traversal, please ensure that STUN is switched off in the STUN configuration dialogue.

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IP Address / Host name

Default : 0.0.0.0

The address of the ITSPs STUN server. This can be an IP-address or the host name. Port Port of the ITSPs STUN server. 5.1.6 DDI or MSN type of ITSP This value is used to provide a convenient user interface for the administration of an ITSP account in the configuration wizards. This parameter has no effect to the SIP signaling and is not visible in the provider configuration. For a MSN type each number has to be entered separately and for each ITSP number an internal number has to be assigned. For a DDI type a range of number will be configured and internal number can be assigned automatically. Default : 3478

5.2

Account configuration

5.2.1 Registration information Regardless if an ITSP uses registration or not, one or more user accounts has to be configured. The following data has to be entered and will be used as described in the SIP protocol
Field in WBM/Wizard Internet telephony station Authorization name Password Used in UserAccount Sip URI in From: , To: and Contact: UserAuthorizationName in Authorization: Leave empty, if no extra name is needed Used to calculate the response=CalculatedHash in Authorization

If no Authorization name is given, the value of Internet telephony station is used as username. These values have to be provided by the ITSP. Example:
REGISTER sip:RegistrarIp/HostName:RegistrarPort;transport=udp SIP/2.0 Via: SIP/2.0/UDP SystemIp:SystemPort;rport;branch=z9hG4bK... From: <sip: UserAccount@ RegistrarIp/HostName:RegistrarPort To: <sip: UserAccount@ RegistrarIp/HostName:RegistrarPort> ... Contact: <sip:UserAccount@SystemIp:SystemPort>;expires=Reregistrationinterval Authorization: Digest username="UserAuthorizationName", realm="ReceivedIn401", nonce="ReceivedIn401",uri="RURI",response="CalculatedHash",algorithm=MD5

5.2.2 Call numbers For each account one or more call numbers can be assigned depending on the ITSP. These numbers have to be configured in the WBM and will be used as they are configured. For correct display information it is important to send the right call number format to the ITSP. Example:
From: <sip:+4923026672695@DomainName> From: <sip:004923026672695@DomainName>

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From: <sip:023026672695@DomainName>

5.3

Special configuration - SIP Profile Data

023026672695 sip-acc1 004970070

In the following section the examples use the following values:

Calling party number Calling party account Called party number

5.3.1 Format of From:, PAI: and PPI: for Basic Call The system provides various profile parameters to control format of SIP header fields according to the needs of the SIP ITSP. The contents of the following header fields which describe the source of a call can be controlled:
From: DisplayPart <sip:UserPart@HostPart> P-Asserted-Identity: DisplayPart <sip:UserPart@domain> P-Preferred-Identity: DisplayPart <sip:UserPart@domain>

Each header field has an own parameter to set the Display part and the User part independently. From-DisplayPart From-UserPart PAI-DisplayPart PAI-UserPart PPI-DisplayPart PPI-UserPart Possible settings are Call number the number assigned by the ITSP is sent in the format as configured in the User Account data under MSN Account the user name assigned by the ITSP is sent in the format as configured in the User Account data under Internet telephony station Omit field is omitted The From: header field is mandatory and thus configured by default to user call number in display, account in user part and domain name in the host part. For some rare configurations its possible to set the host part with the systems IP address. PAI and PPI are disabled by default. PAI is used by several provides to provide better information about the caller. Usage of PPI is uncommon, but as it is requested by some providers the system can make use of this header field too. If the ITSP does NOT support the PAI or PPI both displayPart AND userPart MUST be set to omit for the header field concerned.

Even if your provider might work with the configured default, please check carefully which options are supported. Only a correct usage of the header fields will enable the usage of the transferred addressing information for features (e.g. caller list)

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5.3.2 Diverted Calls: Format of From: For regular outgoing calls the ITSP usually checks if the caller is allowed to place outgoing calls. This is done by means of account information provided in the outgoing INVITE as configured in 5.3.1. If an outgoing call is established due to call forwarding, the system can provide information about the original calling party as well. Therefore a profile parameter (OriginalCallingNumber in From) is provided to control the contents of the From: header field for diverted calls.

In default (OriginalCallingNumber in From=false) the From: header field contains information which represents the B-Ext. (Redirecting party) and is known by the ITSP (callnumber/account). Setting OriginalCallingNumber in From=true causes the system to sent the original calling party (A-Ext) in the From: header field.

Note: If OriginalCallingNumber in From is set to true the Format settings FromDisplayPart and From-UserPart must be set to calling party as well.

5.3.3 Diverted Calls: Format of Diversion: By default the outgoing INVITE for diverted calls contains a diversion: header field. The diversion header field represents always the B-Ext. (Redirecting party). In default the call number is used in the UserPart and NO displayPart is sent.
Diversion: DisplayPart <sip:UserPart@domain>;reason=unconditional;counter=1

Example:
Diversion: <sip:023026672695@domain>;reason=unconditional;counter=1

DisplayPart and UserPart may be configured Diversion-DisplayPart Diversion-UserPart

If the ITSP does NOT support the diversion header field both displayPart AND userPart MUST be set to omit. If B has invoked call number suppression (presentation restricted) the diversion-header field contains:
Diversion: Anonymous <sip:UserPart@domain>;reason=unconditional;counter=1

Example:
Diversion: Anonymous <sip: 23026672695@domain>;reason=unconditional;...

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Note: The current implementation does not support configuration in WBM (will be implemented soon) If your provider does not support the diversion header filed or needs other settings as supported forward this request to the certification team.

5.3.4 Anonymous Calls: Format of From: If a user has activated call number suppression and the provider doesnt use the PAI header field this can be signaled by using anonymous in the display as well as in the user part of the From: header field according to the configuration of: From-AnonymousDisplayPart From-AnonymousUserPart

5.3.5 Anonymous Calls: Format of Privacy: If the ITSP supports the P-Asserted-Identity or P-Preferred-Identity header field, call number suppression can be signaled by the presence of the privacy header containing the tag-value id.
Privacy: id

5.3.6 Call number formatting Various parameters are provided to control the treatment of header fields on incoming INVITE messages. Incoming call Called party number is taken from: With this parameter it is determined where the destination address of a call is derived from. In default this is taken from the user part of the To: header field.
To: 004970070 <sip:004970070@sip.provider.de >

If the provider requires using the Display part of the To: header filed or the user-part of the request URI this can be used too.
INVITE sip:004970070@80.144.242.235:61901 SIP/2.0 To: 004970070 <sip:004970070@sip.provider.de>

For ITSPs this parameter SHALL be set to ToAddrSpec. Incoming call Calling party number is taken from: With this parameter it is determined where the source address of a call is derived from. In default the system searches first in P-Asserted-Identity, if present. If no PAsserted-Identity is present the From: header field is taken.
INVITE sip:004970070@80.144.242.235:61901 SIP/2.0 From: <sip:023026672695@sip.provider.de > P-Asserted-Identity: <sip:023026672695@sip.provider.de>

If the provider requires a special treatment here it can be determined which part of a the P-Asserted-Identity or From: header field should be used to derive the callers identity.
INVITE sip:004970070@80.144.242.235:61901 SIP/2.0 From: 023026672695 <sip:023026672695@sip.provider.de> P-Asserted-Identity: 023026672695 <sip:023026672695@sip.provider.de>

For ITSPs this parameter SHALL be set to automatic.

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Incoming call Calling party Type of Number (TON): With this parameter the treatment off the calling party number format is controlled. If the provider sends the calling party number in a dialable format including prefixes the parameter MUST not be changed. If the provider uses an international number format without prefixes (e.g. 498970070 instead of 00498970070 or +498970070) this number cannot be processed automatically by the system, thus the classification of the number has to be configured. The current implementation supports the setting of international number. For ITSPs this parameter SHALL be set to automatic. 5.3.7 Settings for Registration: P_SipProvProfileRegContactAddr IpAddress or domain name may be set in the hostPart of the sipUri in the Contact header in REGISTER.
Contact: <sip:@192.168.138.1:5060>; Contact: <sip:@sip.provider.de>;

For ITSPs this parameter MUST be set to IpAddress. P_SipProvProfileContactUriWithProtocol Used to add the transport protocol and an OwnSecurity parameter in the Contact header in REGISTER.
Contact: <sip:..@..>;transport=udp;+u.www.siemens.com/icn/en/oscar/

For ITSPs this parameter MUST be set to false. 5.3.8 Parameter for Authentication: P_SipProvProfileUseRouteURIAuthentication True sets the behavior for using the Route URI instead of the request URI in the (Proxy-) Authorization header for INVITE or INFO (not for REGISTER). The request URI will be used when no Route URI is present For ITSPs this parameter MUST be set to true. 5.3.9 Routing parameter: RFC3581 defines a rport parameter for the Via header: .. When used with UDP, responses to requests are returned to the source address the request came from, and to the port written into the topmost Via header field value of the request. This behavior is not desirable in many cases, most notably, when the client is behind a Network Address Translator (NAT). "rport" allows a client to request that the server send the response back to the source IP address and port from which the request originated. P_SipProvProfileUseViaRPort If set to true, the rport parameter is added to Via-Header.
Via: SIP/2.0/UDP 192.168.138.1:5060;rport;

If set to false, no rport parameter is added to Via-Header.

Via: SIP/2.0/UDP 192.168.138.1:5060;

Note: The current implementation does NOT support the configuration of rport. rport is sent regardless of the setting of this parameter. It will be supported starting with OSO V3. If the ITSP does NOT support the rport parameter and the system MUST omit it, forward this

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request to the certification team. 5.3.10 Media handling: P_SipProvProfileIgnore100Rel Ignore a 'Supported: 100rel' header field if received in INVITE. Because 100rel may interfere with interworking to non-SIP subscribers, this parameter SHOULD be set to true. P_SipProvProfileDirectPayload Usually in SIP networks payload is transported end-to-end. Using third party call control techniques in conjunction with ITSP often causes problems in payload transport. To avoid such problems HP3k/OSO uses the integrated RTP-Proxy to terminate the media exchanged with the ITSP. For ITSPs this parameter MUST be set to false. 5.3.11 Supported Methods in sending direction With the parameters in this section the used methods on the SIP interface can be controlled P_SipProvProfileUPDATESupported If set to true UPDATE is used depending on the contents of the Allow header field. If UPDATE is not set in the Allow header field received, the SIP stack never sends an UPDATE. If set to false the UPDATE method will never be used (a re-INVITE is used instead of this) For ITSP this parameter SHALL be set to true (default) Note: Currently UPDATE is sent to an ITSP for the Session-Timer feature only, thus there is no need to set this flag to false. 5.3.12 Supported Methods in receiving direction P_SipProvProfileREFERAllowed With this flag the treatments of a received REFER is controlled. Excerpt of SIPconnect 1.1 Technical Recommendation (Draft v14) Call transfer can be accomplished by the use of REFER requests (a proxy model), or by the use of one or more INVITE requests (a third-party call control model). Both are supported in SIPconnect. Service providers using the proxy model with REFER are cautioned to examine carefully possible interactions with charging considerations. Unless configured otherwise, the SIP-PBX should reject all REFER requests from the SP-SSE. This mode of operation is intended to protect unsophisticated enterprise networks from unexpected charges due to REFERed calls. As HP3k and OSO has implemented the third-party call control model they do NOT support receiving a REFER. A REFER is always rejected by 405_Method_not_allowed For ITSPs this parameter MUST be set to false.

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Modifying the ITSPs profile settings

It depends on the provider what kind of format is required in the SIP messages. Every new ITSP profile in our systems (HiPath 3000 and OpenScape Office MX) comes with some default values which are the most common used among the providers already certified. But some times it is necessary to change these values to comply with the providers requirements. If you need to change the format in outgoing SIP messages and/or the treatment of the incoming you need to change the Special Configuration - SIP Provider Data. For the moment it is not possible to edit these expert settings after a creation of a profile, so you have to create a new profile. Only during a profile creation you are allowed to modify these settings. Below you can find instructions to modify these settings in HiPath 3000 and OpenScape Office MX.

Note: Before starting with the creation of a new profile please verify that your provider profile is deactivated. Please be very careful in providing the entries. If a profile with wrong entries is created, the provider may not go into service or calls will fail. Caution: For access to this wizard developer access rights are required. This user role is reserved for experienced technicians and development personnel. With this user role a lot of (usually hidden) configuration possibilities are offered. Using these possibilities in a wrong way may result in a malfunction of the system. It can cause a RESTART and may lead to a situation where you need a factory reset to recover.

!
6.1

HG1500 (HiPath 3000)

Go within Web-based Management (WBM) to Maintenance > Appl. Diagnostics > Developer Settings > SIP Provider Profiles > click Add Here you can select an already predefined/created profile as base template. In this case all the settings will be filled in automatically and the only thing you have to do is to give it a different Provider Name e.g. Provider1. Alternatively you can choose the default template and create a new profile from the scratch. In both cases you have to enter a unique 5 digits random number in the serial number (this field is used only for internal sorting).

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Add the profile settings to the relevant fields (or verify them if you have chosen a base template and fill in where necessary). Click OK & Next

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Here you can edit the Extended SIP Provider Data. Make the changes and Click OK & Next

Here you can see the profile Provider 1 you have just created. Click OK & Next again

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Press Transfer to SIP and then press Finish.

Now this new profile is copied to the ITSPs list in the Voice Gateway menu.

Go to your new profile and create again the accounts, the DIDs and the MSNs.

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If an old profile e.g. with name: Provider was created (and used as a base template above) and you want to have only the new profile then delete the old profile (after deleting the MSNs and accounts first) and then rename the Provider1 to Provider. You can do all these directly at the ITSP menu under Voice Gateway. After you have done all these, you can enable this new profile.

6.2

OpenScape Office MX

The procedure for the OSO MX is exactly the same as in HG1500. The only difference is that the SIP Provider Wizard is not located at the same location in the WBM. It is at the following location: Expert Mode > Maintenance > Appl. Diagnostics > Central Box > Developer Settings > SIP Provider Profiles

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About Siemens Enterprise Communications Group (SEN Group) The SEN Group is a premier provider of enterprise communications solutions. More than 14,000 employees in 80 countries carry on the tradition of voice and data excellence started more than 160 years ago with Werner von Siemens and the invention of the pointer telegraph. Today the company leads the market with its "Open Communications" approach that enables teams working within any IT infrastructure to improve productivity through a unified collaboration experience. SEN Group is a joint venture between the private equity firm, The Gores Group, and Siemens AG and incorporates Siemens Enterprise Communications, Enterasys Networks, SER Solutions, Cycos and iSEC. For more information about Siemens Enterprise Communications, please visit www.siemens.com/open Siemens Enterprise Communications GmbH & Co. KG Siemens Enterprise Communications GmbH & Co. KG is a Trademark Licensee of Siemens AG Status 09/2010 The information provided in this brochure contains merely general descriptions or characteristics of performance which in case of actual use do not always apply as described or which may change as a result of further development of the products. An obligation to provide the respective characteristics shall only exist if expressly agreed in the terms of contract. Availability and technical specifications are subject to change without notice. OpenScape, OpenStage and HiPath are registered trademarks of Siemens Enterprise Communications GmbH & Co. KG. All other company, brand, product and service names are trademarks or registered trademarks of their respective holders. Printed in Germany.

Communication for the open minded

Siemens Enterprise Communications www.siemens.com/open