SIP Trunk Solution Ecotel Germany (sipTrunk 2.0) Configuration Guideline

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SIP Trunk Solution Ecotel Germany (sipTrunk 2.0) Configuration Guideline
Technical Bulletin                                                                      OmniPCX Enterprise
 TC2570 ed.01                                                                                                  Release 12.2

                       SIP Trunk Solution Ecotel Germany
                     (sipTrunk 2.0) Configuration Guideline

 This document details how to set up an IPBX OXE R12.2 for enabling a public SIP trunk with SIP Provider Ecotel Germany

 Revision History
 Edition 1: February 18, 2019       creation of the document

Legal notice:
The Alcatel-Lucent name and logo are trademarks of Nokia used under license by ALE. To view other trademarks used by
affiliated companies of ALE Holding, visit: www.al-enterprise.com/en/legal/trademarks-copyright. All other trademarks are the
property of their respective owners. The information presented is subject to change without notice. Neither ALE Holding nor any
of its affiliates assumes any responsibility for inaccuracies contained herein. © 2019 ALE International. All rights reserved.
www.al-enterprise.com
SIP Trunk Solution Ecotel Germany (sipTrunk 2.0) Configuration Guideline
Table of contents
1 General .................................................................................................................................................. 4
   1.1 References ....................................................................................................................................... 4
   1.2 Scope & usage of the configuration guide ........................................................................................... 4
   1.3 Scope of Alcatel-Lucent Enterprise’s support ....................................................................................... 4
   1.4 RFCs supported by OmniPCX Enterprise ............................................................................................. 4
        SIP RFCs: ..................................................................................................................................................... 4
        RTP, T38 & DTMF (used for SIP) ..................................................................................................................... 5
        New RFCs in OXE R11.2.1 and R12.x ............................................................................................................... 5
   1.5 Software/ Hardware components on customer's infrastructure ............................................................. 6
   1.6 Supported topology:.......................................................................................................................... 6
   1.7 Feature List & Set Compatibility ......................................................................................................... 6
     1.7.1 Supported Features & Sets .......................................................................................................... 6

2 Ecotel SIP Trunk Solution Configuration .................................................................................................. 15

   2.1 Signaling protocol and number of physical channels .......................................................................... 15
   2.2 Omnipcx Enterprise configuration..................................................................................................... 15
     2.2.1 Entities configuration ............................................................................................................... 15
     2.2.2 Discriminator Selector .............................................................................................................. 16
     2.2.3 Prefix Plan ............................................................................................................................... 17
     2.2.4 External Call Back Translator .................................................................................................... 17
     2.2.5 Trunk Configuration .................................................................................................................. 17
        2.2.5.1 Trunk Group: ....................................................................................................................................17
        2.2.5.2 Trunk groups local parameters ...........................................................................................................19
        2.2.5.3 Trunk group NDP selector ..................................................................................................................22
     2.2.6 ARS Configuration ..................................................................................................................... 22
        2.2.6.1 NPD .................................................................................................................................................22
        2.2.6.2 DDI-Translation .................................................................................................................................23
        2.2.6.3 ARS Route List ..................................................................................................................................23
        2.2.6.4 ARS Route ........................................................................................................................................24
        2.2.6.5 Time Based Route List .......................................................................................................................25
        2.2.6.6 Numbering Command Table ...............................................................................................................25
     2.2.7 SIP Gateway and SIP Proxy Configuration ................................................................................... 26
        2.2.7.1 SIP Gateway .....................................................................................................................................26
        2.2.7.2 SIP Proxy ..........................................................................................................................................27
        2.2.7.3 SIP Registrar .....................................................................................................................................27
     2.2.8 SIP External Gateway Configuration .......................................................................................... 28
        2.2.8.1 SIP External Gateway ........................................................................................................................28
     2.2.9 SIP trunk configuration abstract ................................................................................................ 33

3 Troubleshooting .................................................................................................................................... 36
   3.1 information to be sent to TS contact before tests start up ................................................................. 36

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SIP Trunk Solution Ecotel Germany Configuration Guideline                                                                                            TC2570 ed.01
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3.2 Basic pieces of information to be sent for test failure analysis ............................................................ 37
  3.3 Service Request (SR) writing for effective assignment and analysis .................................................... 38

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SIP Trunk Solution Ecotel Germany Configuration Guideline                                                             TC2570 ed.01
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1 General
This document details the process for configuring from scratch a public SIP trunk of the SIP provider Ecotel
Germany (sipTrunk 2.0) on a system OXE R12.2

1.1 References
Alcatel-Lucent documentation available on the Business Partner Web Site:
[1] Alcatel-Lucent OmniPCX Enterprise Communication Server R12.2 – Technical Documentation
[2] Technical Bulletin TC2005 – Certified SIP providers for OpenTouch and/or OmniPCX Enterprise
[3] Troubleshooting Guide TG0069 – OmniPCX Enterprise – Session Initiation Protocol (SIP)

1.2 Scope & usage of the configuration guide
This guide is intended for engineers who are familiar with mgr, OmniVista 8770, OpenTouch and with the
very basic set up of the IPBX. Therefore, well-known configurations like that for the IP-LAN or for "Traffic
Sharing and Barring" are just reminded without any details.

1.3 Scope of Alcatel-Lucent Enterprise’s support
The support delivered for this SIP Trunk solution is strictly delimited by the approval context and the system
configuration detailed in this document. The protocol and the functional aspects of the SIP trunk are in the
scope, but not the audio quality of calls for the part incumbent on the SIP provider or on the client's
infrastructure.

1.4 RFCs supported by OmniPCX Enterprise
SIP RFCs:
   RFC 2543 (obsolete by RFC 3261,3262, 3263,3264, 3265): SIP: Session Initiation Protocol
   RFC 2782: A DNS RR for specifying the location of services (DNS SRV)
   RFC 2822: Internet Message Format
   RFC 3261: SIP: Session Initiation Protocol
   RFC 3262: Reliability of Provisional Responses in SIP (PRACK)
   RFC 3263: SIP: Locating SIP Servers
   RFC 3264: An Offer / Answer model with SDP
   RFC 3265: SIP-Specific Event Notification
   RFC 3311: The SIP UPDATE Method (session timer only)
   RFC 3323: Privacy Mechanism for the Session Initiation Protocol (SIP)
   RFC 3324: Short term requirements for network asserted identity
   RFC 3325: Private Extensions to the Session Initiation Protocol (SIP) for Asserted Identity within Trusted
    Networks
   RFC 3265: SIP-specific Event Notification
   RFC 3515: The Session Initiation Protocol (SIP) Refer method
   RFC 3891/3892: The Session Initiation Protocol (SIP) 'Replaces' Header/ Referred-By Mechanism
   RFC 3398: Integrated Services Digital Network (ISDN) User Part (ISUP) to SIP Mapping
   RFC 3966: The telephone URI for telephone numbers: since R11 only TEL URI is supported

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SIP Trunk Solution Ecotel Germany Configuration Guideline                                        TC2570 ed.01
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   RFC 4497: Inter-working between SIP and QSIG
   RFC 5373: Requesting Answering Modes for the Session Initiation Protocol
   RFC 4244: An Extension to the Session Initiation Protocol (SIP)for Request History Information
   RFC 3326: The Reason Header Field for the Session Initiation Protocol (SIP)
   RFC 3428: Session Initiation Protocol (SIP) Extension for Instant Messaging (partial)
   RFC 3608: Service Route header
   RFC 3327: Path Header
   RFC 1321: Authentication for Outgoing calls
   RFC 2246: The TLS Protocol Version 1.0
   RFC 3268: Advanced Encryption Standard (AES) Cipher suites for Transport Layer Security (TLS)
   RFC 3280/5280: Internet X.509 Public Key Infrastructure Certificate and Certificate Revocation List (CRL)
    Profile
   RFC 3842: A message Summary and Message Waiting Indication Event Package
   RFC 4028: The session timers in the Session Initiation Protocol
   RFC 3960: Early Media (partial): Gateway model not supported
   RFC 4568: Session Description Protocol (SDP) Security Descriptions for Media Streams
   RFC 5806: Diversion Indication in SIP
   RFC 3725: Invite without SDP (3pcc in SIP)
   RFC 3966: The tel URI
   RFC 5009: The P-Early-Media header

RTP, T38 & DTMF (used for SIP)
   RFC   2617: HTTP Authentication: Basic and Digest Access Authentication
   RFC   2833/4733: DTMF Transparency. RFC 2833 replaced by RFC 4733
   RFC   1889/1890: RTP: A transport protocol for Real-Time applications
   RFC   2198: RTP Payload for Redundant Audio data
   RFC   3550: RTP: A Transport Protocol for Real-Time application (audio only)
   RFC   3551: RTP Profile for Audio and Video Conferences with Minimal Control (audio only)
   RFC   3711: The Secure Real Time. Supported on A-LU IP Phone and Softphone
   RFC   3362: T38 ITU-T Procedures for real time Group3 Fax Relay / communications over IP
   RFC 3711: The Secure Real-time Transport Protocol (SRTP) (media integrity)

New RFCs in OXE R11.2.1 and R12.x
   RFC 4904: Representing Trunk Groups in tel/sip Uniform Resource Identifiers (URIs )
   RFC 6140: “Registration for Multiple Phone Numbers in the Session Initiation Protocol (SIP)”
   RFC 7433 A Mechanism for transporting User to User Call Control Information in SIP
   draft-ietf-cuss-sip-uui-isdn-08 Interworking ISDN Call Control User Information with SIP

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SIP Trunk Solution Ecotel Germany Configuration Guideline                                          TC2570 ed.01
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1.5 Software/ Hardware components on customer's infrastructure
          INFRA COMPONENT                        MODEL                      VERSION (min compatible)

                  OXE                      OmniPCX Enterprise                 R12.2-m3.402.22.c

1.6 Supported topology:
Test-Topology:

1.7 Feature List & Set Compatibility
1.7.1 Supported Features & Sets
The following tables list the main inter-operation features and the range of sets that are supported with this
SIP Trunk solution. For the different items, refer to the indication given in the support column which is
marked as "OK" (for full support), or "WR" (support With Restriction), or "NOK"/”NA” (for Not OK or Not
Applicable), or "NT" (for Not Tested).

Features not supported:

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PASSE
TEST
                                                                                                  D/               Comments
CASE                  FEATURE                         TEST / CHECK DESCRIPTION
                                                                                                 FAILE       Trace available YES/NO
 #
                                                                                                   D
         1. - START/Configuration
         1.1 - Registration in the
                                                           UNI mode tests only
         operator’s proxy

       Configure OXE in order to have it         Registration is OK. "Expires" timer is                  Expires value in REGISTER is
       registering a range of phone numbers      correctly used for refresh.                                       set to 300
       on the public network / (with or          - SIP UTI looks like:                                  Expires value in OK is set to 60
       without authentication) / set a           sip:+e164_number@My_sip_provider_do                              by provider
# 1100                                                                                           passed
       correct number range. Check               main                                                   Re-Registration is every 48 sec
       REGISTER messages exchanged               - contact header looks like:                               (80 percent of 60 sec)
       (check particularly "expires" values in   sip:+e164_installation_number@IP_addre
       request and answer)                       ss_of_IPPBX                                                    Traces available

       Configure OXE in order to have it
       registering a range of phone numbers
       on the public network / (with or                                                                     Registration not possible
                                                 Registration is NOT OK. Check OXE retries
       without authentication) / set a bad                                                                   2 retiries every 30 sec.
# 1101                                           to register after "retry-after" timer if sent   passed
       operator number range in OXE config.
                                                 by operator
       Check REGISTER messages                                                                                  Traces available
       exchanged (check particularly "retry-
       after" header in the answer)

         1.2 - Authentication

       Configure OXE and operator in order
       to have SIP trunk in authenticated
# 1200                                           Config is possible                              passed               done
       mode towards the operator (DIGEST
       authenticated on operator side)

                                                                                                         Outgoing call is authenicated
                                                 Outgoing and incoming calls are possible.                with challenge according to
       DIGEST authenticated on operator
# 1201                                           Check call flows (RFC 2617 and RFC 1321         passed RFC 2617 and RFC 1321 (MD5)
       side / correct secret
                                                 are used for outgoing calls)
                                                                                                                Traces available

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Outgoing call not possible
                                                 Outgoing call is impossible. Check call
       DIGEST authenticated on operator                                                              Incoming call still possible
# 1202                                           flows (RFC 2617 and RFC 1321 are used     passed
       side / incorrect secret
                                                 for outgoing calls)
                                                                                                          Traces available

         2. - VOICE CALLS : site 1 and
         PSTN or GSM
         2.1 - 1PCC OUTGOING CALLS
         2.1.0 - Basics to GSM sets

                                                                                                      Ringback Tone is remote,
                                                                                                    audio is ok, display on calling
                                                 Ring / (remote or local, please tell)                      and called ok
         OXE device calls to GSM set: outcall
# 2100                                           ringback, normal audio after answer.      passed   Destination tested: T-Mobile
         phase and com. establishment
                                                 Check the display of the calling number                       and O2

                                                                                                          Traces available

                                                                                                    Tested with T-Mobile and O2
         OXE device calls GSM set: end of
# 2101                                           Com. and trunk properly released          passed
         com. by local party
                                                                                                          Traces available

         OXE device calls GSM set: end of                                                           Tested with T-Mobile and O2
# 2102                                           Com. and trunk properly released          passed
         com. by remote party                                                                             Traces available

                                                                                                    Tested with T-Mobile and O2
                                                                                                    SIP-Flow: CANCELED sent by
                                                 Com. and trunk properly released
                                                                                                       OXE/OK answered by
         OXE device calls to GSM set: outcall    - Check SIP call flow. CANCEL sent by
# 2103                                                                                     passed      Network/487 sent by
         phase and call clearing before answer   OXE /OK answered by Network./ then 487
                                                                                                     Network/ACK sent by OXE
                                                 sent by OXE / Ack from network
                                                                                                          Traces available

                                                                                                    Tested with T-Mobile and O2
                                                                                                        Provider answers with
         OXE device calls to GSM set: outcall to Correct display and trunk properly                 announcement "Number not
# 2104                                                                                     passed
         a wrong number                          released                                            listed" in Progress Message

                                                                                                          Traces available

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 SIP Trunk Solution Ecotel Germany Configuration Guideline                                                           TC2570 ed.01
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Tested with T-Mobile and O2
                                                                                                           Provider sends "487 Busy
                                                                                                                      here"
                                                                                                         Display shows "besetzt" and
         OXE device calls to GSM set: outcall to Correct display and busy tone. busy tone
# 2105                                                                                          passed         busy tone is heard
         a busy set                              heard. (check SIP/SDP traces)
                                                                                                         After timeout calling set goes
                                                                                                                  in idle state

                                                                                                                Traces available

         2.1.1 - 1PCC Basics to PSTN sets

                                                                                                           Ringback Tone is remote,
                                                                                                         audio is ok, display on calling
         OXE device calls to PSTN set: outcall   Ring / remote ringback, normal audio
# 2110                                                                                          passed    and called ok. Destination
         phase and com. establishment            after answer
                                                                                                           tested: T-Mobile and O2
                                                                                                                Traces available

         2.1.3 - Anonymous outgoing calls

                                                                                                            Test towards different
                                                                                                            destinations Display on
                                                                                                          destination is dependent on
         OXE device to public number: outcall    Identity of caller not displayed on callee                destination's provider or
# 2130                                                                                          passed
         with CLIR                               side                                                    implementation (unbekannt,
                                                                                                                secret identity)

                                                                                                                Traces available
         2.2 - INCOMING CALLS
         2.2.1 - Incoming basic calls with
         CLI (no overflow to attendant
         configured)

                                                                                                           Tested with T-Mobile, O2,
                                                 Display CLI info on callee sides, normal                PSTNDisplay on callee side ok,
                                                 conversation / check SIP trace and from                  conversation ok, SDP offer is
       Incoming public call to OXE device:
                                                 parameter- Check display is consistent                     G711 A Law and Mu Law,
# 2210 com. Establishment. End call by any                                                      passed
                                                 with FROM or PAI parameter- check SDP                    G729, G726, Payload 96 and
       party.
                                                 Offer/Response exchange- Check Ring                       101Response is G11 A Law
                                                 back tone on public side                                  and Payload 101Ring back
                                                                                                         tone availableTraces available

                                                                                                                 Display ok
                                                 Display CLI info on caller and callee sides,            CANCEL sent by network, OK
                                                 normal conversation / check SIP trace.                    answered by OXE, 487
         Incoming public call to OXE device:
# 2211                                           - Check SIP call flow. CANCEL sent by          passed   Request Terminated sent by
         call clearing before answer
                                                 network /OK answered by OXE./ then 487                    OXE, ACK from network
                                                 sent by network/ Ack from OXE
                                                                                                                Traces available
         2.2.2 - Anonymous incoming calls
         from public

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Callee Display shows Trunk
                                                                                                        Group Name - Ecotel as
                                                   Check that identity is not displayed on
                                                                                                       configured in Trunk Group
                                                    Callee side in ringing phase, and in
                                                                                                             FROM-Header is:
                                                         connected phase neither.
                                                                                                     anonymous@anonymous.invali
                                                 - Check SIP traces : URI in FROM header
                                                                                                                    d
           Incoming call from public network                  should look like :
# 2220                                                                                        passed       PAI is not available
           with secret identity to OXE device.                  "Anonymous"
                                                                                                       It is necessary to have an
                                                 ;ta
                                                                                                         External Gateway with
                                                              g=1928301774
                                                                                                             Remote Domain:
                                                   - Check if P-Asserted ID is avalable in
                                                                                                           anonymous.invalid
                                                                 messages
                                                                                                              Traces available
         2.3 - 1PCC routing services
         2.3.2 - DDI call / subscriber's call
         forwarding (to internal or
         voicemail) :

                                                                                                          Tested with T-Mobile and
                                                 CLI and audio-msg in VM of the right                   O2Message is correct. Calling
         Incoming Public call: immediate         callee/ no SIP 302 sent to public network.               line correctly seen i.e. call
# 2320                                                                                        passed
         forward to Voice Mail                   / possible to navigate in the menus with                back is possible. Navigation
                                                 DTMF                                                        via DTMF is ok. Traces
                                                                                                                   available

         Incoming Public call: immediate         Audio OK / no SIP 302 message sent to
# 2321                                                                                        passed
         forward to other internal user          public network

         2.3.4 - DDI call / subscriber's call
         forwarding (to external)
                                                                                                        OK with NPD for external fwd
                                                                                                                     = -1
                                                                                                        # of forwarding station shown
                                                 Display # or name of initial callee on caller                    on callee
                                                 side, normal audio after answer / SDP
                                                 transparency (for supported codecs by                  OK with NPD for external fwd
         Incoming Public call to OXE device:
# 2340                                           OXE) for offer/answer /no SIP 302             passed               = 110
         immediate forward to public number
                                                 message sent to network. /tromboning                    # of caller shown on callee
                                                 done on OXE side / check header History
                                                 Info is present with OXE user number                   No 302 sent, SDP transparent
                                                                                                                  to OXE

                                                                                                              Traces available

                                                 Display # or name of initial callee on caller          OK with NPD for external fwd
                                                 side, normal audio after answer / SDP                   = 110 # of caller shown on
         Incoming Public call to OXE device:     transparency (for supported codecs by                     callee. No 302 sent, SDP
# 2341                                                                                         passed
         fwd on no answer to public number.      OXE) for offer/answer /no SIP 302                      transparent to OXE. Voice OK
                                                 message sent to network. /tromboning                    after answer of destination
                                                 done on OXE side                                               Traces available

         2.3.5 - DDI call / subscriber not
         available

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480 Temporarily not available
                                                                                                             is sent to network
                                                                                                          Behavior of network is
                                                                                                      different depending on calling
                                                In SIP traces, 480 temporarily unavailable
         Incoming Public call : OXE device Do                                                                     provider:
# 2350                                          message is sent to network. It is handled    passed
                    Not Disturb                                                                               ISDN: busy tone
                                                     by network in a correct manner
                                                                                                            O2: voice message
                                                                                                            T-Mobile: busy tone

                                                                                                             Traces available

                                                                                                      Dependent on configuration:
                                                                                                       484 sent by OXE in case not
                                                                                                      existing number but within a
                                                                                                              range of DDI
         Incoming Public call : OXE non         Call is freed by OXE (404) or overflow to
# 2351                                                                                       passed   404 is sent by OXE in case of
         attributed number                      operator (according to OXE config)
                                                                                                      not existing number in DDI-
                                                                                                                Translator

                                                                                                             Traces available

                                                Normal busy tone is heard on caller                   486 busy here is sent by OXE.
         Incoming Public call to busy OXE
# 2352                                          side.SIP 486 user busy sent to network. It   passed   Caller has busy tone. ACK sent
         device
                                                is handled by network in a correct manner              by provider. Traces available

         2.4 - Advanced FEATURES for
         COMMUNICATION
         2.4.1 - 1PCC Enquiry call (call
         hold / call retrieve)
                                                                                                         Caller receives MOH
                                                                                                     Re-Invite with SDP sent by
                                                                                                                  OXE
                                                                                                          and OK by network
                                                                                                      Hold: OXE sends Re-Invite
                                                Music on hold is heard on caller side.
       Incoming Public call to OXE device :                                                         with SDP address of GD, media
                                                Check SIP traces. RE-INVITEs - OK sent
# 2410 put on-hold and retrieve                                                              passed      attribute : sendonly
                                                between OXE and network. off-hold is OK
       communication                                                                                Retrieve from Hold: OXE sends
                                                also
                                                                                                    Re-Invite with SDP address of
                                                                                                        user, media attribute:
                                                                                                               sendrecv

                                                                                                             Traces available

                                                                                                      Call remains in progress, voice
                                                                                                                    ok
                                                                                                       Provider sends INFO after 10
       Ongoing call with OXE user : push the No RTP flow sent to external user. The
                                                                                                                   min,
# 2411 mute button on OXE device and wait 5 call remains in progress until hanging up        passed
                                                                                                        OXE sends Re-Invite after
       minutes before hanging up the call    manually
                                                                                                             additional 5 min

                                                                                                             Traces available

         2.4.3 - Early attended transfer
         (on ringing)

 OmniPCX Enterprise - Release 12.2
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Sendonly for hold needs to be
                                                                                                            set to FALSE in external
                                                 Display (no evolution of display on public
         Public call on going with OXE device:                                                           gateway otherwise in case of
# 2430                                           side), Ring back tone on public side. After    passed
         transfer to other OXE internal device                                                           outgoing call one way voice in
                                                 answer normal audio, clean release
                                                                                                                   some cases
                                                                                                                Traces available

                                                 Tromboning is done on OXE side, no
         Public call on going with OXE device:   display change on initial caller. Display on
# 2431                                                                                          passed          Traces available
         transfer to ext number                  second public callee is OXE user, even
                                                 after answer

         2.4.4 - Supervised call transfer
         (after answer).

         Public call ongoing with an OXE         Display (no evolution of display on public                 Display of callee shows
         device: transfer to other OXE device    side). After transfer normal audio, clean               external number after transfer
# 2440                                                                                          passed
         (wait for second callee answer before   release. Final callee has display of
         transfering)                            external                                                       Traces available

       Public call ongoing with an OXE                                                                   Display on second public callee
                                                 Tromboning is done on OXE side, no
       device: transfer to PUBLIC set (wait                                                                       is OXE-user
# 2441                                           display change on initial caller. Display on   passed
       for second callee answer before
                                                 second public callee is OXE user
       transfering)                                                                                             Traces available

         2.4.5 - Conference (3-party)

                                                                                                         Ending of conference connects
                                                 Displays are globally OK (the name of all                    to first external user,
         2 Public sets and 1 OXE set which has   participants is not displayed on public side              Transfer out of conference
# 2451                                                                                          passed
         generated conference                    => normal), normal audio. Check the end                           also possible
                                                 of conf is handled correctly
                                                                                                                Traces available

         2.4.6 - DTMF sent to public
         network

                                                                                                          Tested via "Loop" by calling
                                                 Check that the DTMF code are well                       own 4645 Mailbox externally.
         Outgoing call to an external IVR from
# 2460                                           accepted by the remote server / RFC2833        passed   Code is well accepted, seen in
         OXE device: DTMF to external server.
                                                 is used                                                 trace of incoming call.Traces
                                                                                                                    available

 OmniPCX Enterprise - Release 12.2
 SIP Trunk Solution Ecotel Germany Configuration Guideline                                                                TC2570 ed.01
 Copyright © ALE International 2019                                                                                         page 12/39
Outgoing call to an external IVR from     Check that the DTMF code are well
# 2461 OXE device: IN-BAND DTMF to               accepted by the remote server / IN-BAND     n/a
       external server.                          DTMF is used

         3 - 3PCC specificities
         3.1.2 - Piloting a public device
         from OTC PC

         OXE user: OTC PC launches a 3PCC      display, normal audio/ check display on
# 3121                                                                                       n/a
         call to public number from public set OTC PC application on caller side.

         3.1.3 - Call Admission Control
         related tests.

                                                                                                      OXE sends 503 - Service
         Incoming call from public network       Check call is rejected properly / 503 is                   unavailable
# 3130                                                                                      passed
         when CAC is saturated.                  corrrecly sent and accepted by network
                                                                                                          Traces available

         5 - FAX TRANSMISSION
         5.1.1 - Fax transmission with fax
         machine attached on OXE

                                                                                                       Tested Mode: G711 only
         Multiple pages (3) Fax transmit to
# 5110                                           Check fax transmission is done             passed   5 pages have been correctly
         PSTN from OXE fax machine
                                                                                                         sent in good quality

                                                                                                       Tested Mode: G711 only
         Multiple pages (3) Fax receive from
# 5111                                           Chek fax transmission is done              passed   5 pages have been correctly
         public net to OXE FAX machine
                                                                                                         sent in good quality

         5.2.1 - Fax transmission from Fax
         server
                                                Check that fax is correctly transmitted.
         A multiple pages fax is sent over PSTN
# 5210                                          Note the audio codec used during fax         n/a
         from fax server
                                                transmission
         5.2.2 - Fax reception by Fax
         server

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 Copyright © ALE International 2019                                                                                   page 13/39
Check that fax is correctly received (note
         A multiple page fax is received from
# 5220                                          which audio codec is used during call           n/a
         PSTN by fax server
                                                setup)
         7 - Inter-working with
         applications
         7.1.1 - OTMC

       Step 1 : Call the voice mail number
       from a public phone set through SIP      Step 1 : Check that a menu is heard
# 7110 trunk.                                   Step 2 : Check that it is possible to           n/a
       Step 2 :Press some touch keys to         browse the menu.
       navigate the menu.

       Step 1 : From a GUI client select a
                                                Step 1 : Check that public phone is called
       voice message that has previously
                                                back
# 7111 been left by a public phone. Ask for                                                     n/a
                                                Step 2 : Check that call is OK
       call back sender
                                                Step 3 : Check displays are consistent.
       Step 2 : Answer the call

       Step 1 : From a GUI client select a
       voice message that has previously
                                                Step 1 : Check that public phone is called
       been left by a public phone. Ask for
                                                backStep2 : Ring back tone is heard on
# 7112 call back sender from a public set                                                       n/a
                                                caller public set.Step 3 : Check that call is
       also. (other device or mobile)Step 2 :
                                                OKStep 4 : Check displays are consistent.
       answer your own public setStep 3 :
       Answer the call on callee side.

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2 Ecotel SIP Trunk Solution Configuration
2.1 Signaling protocol and number of physical channels
The SIP trunk uses a specific signaling protocol and some physical resources of the IPBX (i.e. DSP channels).
Obviously, it is required a board which provides the system with DSP channels (i.e: GD3/INTIP3 board). NGP
boards (INTIP3/GD3) are necessary for fax in G711. It is possible to check the number of DSP channels
available in the system by using the command “compvisu lio”.

2.2 Omnipcx Enterprise configuration
Screenshots taken from WBM of OXE

2.2.1 Entities configuration

In Test Entity 10 was used

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2.2.2 Discriminator Selector

Discriminator Selector for Entity 10:

Discriminator 10 chosen in Entity:

Entries point towards ARS Route-List Number 10
Example:

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2.2.3 Prefix Plan

2.2.4 External Call Back Translator

2.2.5 Trunk Configuration
To enable phone calls over the SIP trunk, it’s mandatory to have an ISDN trunk group declared with SIP
specification

2.2.5.1 Trunk Group:

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2.2.5.2 Trunk groups local parameters

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2.2.5.3 Trunk group NDP selector

2.2.6 ARS Configuration

To enable voice calls via the ARS system, it’s necessary to have ARS Route list created.

2.2.6.1 NPD

NPD 110:

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2.2.6.2 DDI-Translation

2.2.6.3 ARS Route List

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2.2.6.4 ARS Route

Route-List 10 used:

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2.2.6.5 Time Based Route List

2.2.6.6 Numbering Command Table

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2.2.7 SIP Gateway and SIP Proxy Configuration

2.2.7.1 SIP Gateway

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2.2.7.2 SIP Proxy

2.2.7.3 SIP Registrar

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2.2.8 SIP External Gateway Configuration
3 external gateways are needed:
GW 10: for registration and outgoing calls
GW 11: for incoming calls – Call from Ecotel arrives with “4c.gw” in domain part of FROM Header which is
used for selecting the external gateway and so the correct trunk group etc.
GW 12: for incoming secret id calls – Call arrives with anonymous.invalid in domain part of FROM Header
which is used for selecting the external gateway and so the correct trunk group etc (except if we change the
system parameter Via Header_Inbound_Calls Routing to True, so GW 11 will be used).

2.2.8.1 SIP External Gateway

External Gateway 10 :
Used for Registration and outgoing calls (see Numbering Command Table)

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External Gateway 11:
Used for incoming calls - Call from Ecotel arrives with “4c.gw” in domain part of FROM Header.

Rest is identical with External Gateway 10

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Copyright © ALE International 2019                                                                 page 31/39
External Gateway 12:
Used for incoming secret id calls – Call arrives with anonymous.invalid in domain part of FROM Header

Rest is identical with External Gatewy 10.

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2.2.9 SIP trunk configuration abstract
The following tables gather the overall system configuration. They only show the values to be modified, that
means that the values that are not appearing here will be the default system values.
NOTE: NPD here is meant for incoming calls. Outgoing calls will use NPD managed in ARS routes.

System SIP parameters
Path: System / Other System Param. / SIP Parameters   OXE default value              Carrier's value (if modified)
Packetization times per codec                         True
Via Header_ Inbound Calls Routing                     False
TLS signaling possible                                False
Local resources                                       True
Loose Route with RegID                                True
Reject unidentified proxy calls                       False
SRTP offer answer mode                                False
Hotel doorcam application                             False
Transfer : Refer using single step                    True
RE-INVITE delay for hold                              3
SIP Bearer Capability                                 Speech
Number of SIP trunks (UCaaS)                          0                               10
Enhanced codec negotiation                            Not Available                   Local Type
G722 for SIP trunking                                 True
Sipmotor restart delay                                5
Private SIP transit mode                              Mixed mode
SIP registered pseudo reservation                     False
Blind Transfer with direct RTP                        True
From Header For Anonymous Calls                       Anonymous
Maximum Trunk Group Overflow                          3
SIP video transit mode                                Not Available
Raise SIP Motor Incidents                             True
Enhanced Canonical Form                               False
SIP UUI Normal Transit                                False
Force NCT on Internal Route                           False
SIP diversion info for incoming                       False
Accept keys for unsecured GW                          False
FROM display name                                     Display name else caller number

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Copyright © ALE International 2019                                                                   page 33/39
System compression parameters
Path: System / Other System Param. / Compression      OXE default value   Carrier's value (if modified)
Voice Activity Detect (Comp Bds)                      False
Post Filter on Compressor Boards                      True
Compression Type                                      G 723               G 729
Volume on boards with compressors                     8
VRE on boards with compressors                        False
Multi. Algorithms for Compression                     False
Voice Activity Detection on G711                      False
G722 data rate                                        64 K
G722 Conference With OMS                              True

Additional System parameters
Path: System / Other System Param. / System
Parameters                                            OXE default value   Carrier's value (if modified)
Law                                                   A-Law
Accept Mu and A laws in SIP                           False
DTMF in RFC2833 Only                                  False
Path: System/Other System Param./External Signaling
Parameters                                            OXE default value   Carrier's value (if modified)
NPD for external forward                              -1                  110
Calling Name Presentation :                           False

IP parameters
Path: IP / IP Parameters                              OXE default value   Carrier's value (if modified)
Fast Start                                            True
Round trip delay request                              True
Jitter buff size(modem/fax transp)                    40
G711 VOIP Framing                                     20 ms
G729 VOIP Framing                                     20 ms
G723 VOIP Framing                                     30 ms
Jitter algorithm (voice)                              1
Jitter buffer size (voice)                            30
DTMF mode                                             0
CAC with OTMS/OTBE                                    False
Path: IP / Fax Parameters                             OXE default value   Carrier's value (if modified)
T38 only                                              True
Local T38 port number                                 RTP port number     RTP port number +3
NAT Support for FAX T38                               False

SIP trunk groups parameters
Path: Trunk Groups                                    OXE default value   Carrier's value (if modified)

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Trunk Group Type :                                   T2                           T2
Trunk Group Name :                                   -                            Ecotel
Number Compatible With :                             -1
Remote Network :                                     255                          15
Node number :                                        -                            1
Q931 Signal variant :                                ISDN All Countries           ISDN All Countries (mandatory)
Number Of Digits To Send :                           0
Channel selection type + Quantified                  Quantified
Auto.DTMF dialing on outgoing call :                 NO
T2 Specification :                                   SIP
Homogenous network for direct RTP :                  NO
Public Network COS :                                 31
DID transcoding :                                    False                        True
Associated Ext SIP gateway :                         -1
Path: Trunk Group / Trunk Group                      OXE default value            Carrier's value (if modified)
Entity Number :                                      -                            1
Quality profile for voice over IP :                  Profile #1
DTMF end-to-end signal. :                            NO
Trunk group used in DISA :                           NO
No. of digits unused (ISDN) :                        0                            0
B Channel Choice :                                   YES
IE External Forward                                  None                         None
Path: Trunk Group / Trunk group NPD selector         OXE default value            Carrier's value (if modified)
Public NPD ID :                                      2                            110
Path: Trunk Group / Trunk Group / Virtual accesses
for SIP                                              OXE default value            Carrier's value (if modified)
Number of SIP Accesses :                             2                            2

Type of Numbering Plan used on SIP trunk group (Check Trunk group NPD selector)
Path: Translator / External Numbering Plan /
Numbering Plan Description (NPD)                      OXE default value           Carrier's value (if modified)
Description identifier :                              2                           110
Calling Numbering plan ident.                         Unknown                     NPI/TON: ISDN Unknown
Called numbering plan ident.                          Unknown                     NPI/TON: ISDN Unknown

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Copyright © ALE International 2019                                                                        page 35/39
Timers used by SIP trunk group
Path: External Services / Trunk COS             OXE default value     Carrier's value (if modified)
Trunk COS :                                     31
T2 T0 ABC-F ISDN Trunks
Timer T303                                      100
Timer T304                                      300
Timer T310                                      200
Timer T313                                      40
Timer T305                                      40
Timer T308                                      40
Timer T309                                      900
Timer T302                                      150
Timer T386                                      200
ABC-F Trunks
Timer T306                                      60
Timer T314                                      20
Timer T381                                      1200
Timer T383                                      1200
Timer T384                                      300
Timer T386                                      200
Timer T388                                      350
Timer T389                                      150
Timer T390                                      1800
Timer T392                                      10
Timer T397                                      100

3 Troubleshooting

3.1 information to be sent to TS contact before tests start up
Before going through the test plan, we strongly recommend to place and inbound and outbound and log the
following commands/logs and send the results to SIP Tech Support in order to make sure the initial
configuration is correct.

              a. infocollect
              b. motortrace for both inbound and outbound calls
              motortrace 3

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traced > /tmpd/traceSIP&

3.2 Basic pieces of information to be sent for test failure analysis
        Call scenario description (calling, called number ….)
        infocollect
        Traces:
              -    For all cases:

                           Network traces of SIP dialogs (on OXE side in case of NAT translation)

                           OXE SIP motor traces:
                            killall traced

                            motortrace 3
                            traced -1 /tmpd/sipMotortrace_26062016 -s 4000000 -f 99 -d &

              -    Call Handling level 2 traces:
                           Use cases: Signaling problems: Unsuccessful calls, inconsistent SIP messages content,
                            bad codec negotiation, RTP stream setup failure…

                   These traces will help for call setup problems deeper analysis of VoIP problems
                   understanding.
                            tuner km
                            tuner ctr
                            trc i
                            tuner clear-traces
                            tuner all=off

                            actdbg all=off
                            tuner +at +tr +s +cpu +cpl hybrid=on

                            actdbg sip=on fct=on isdn=on abcf=on rtp=on
                            actdbg cnx=on voip=on
                            mtracer -ag -1 /tmpd/sipCHtrace_26062016 -f 99 -s 4000000 -d&

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Copyright © ALE International 2019                                                                     page 37/39
These are basic traces. Other traces may be requested after first problem description and
                    analysis.

3.3 Service Request (SR) writing for effective assignment and analysis
SR creation must follow the rules hereafter:

        A SR must be opened for any problem.
        One SR must describe one problem only.
        SR must contain detailed test case description:
            I.      Test reference in test plan if any, test summary otherwise.
           II.      Identification of the elements involved in the test (MCDU, IP address…).

          III.      Elements role in the test (callee, caller, OXE user, external user called/calling through SIP
                    trunk, automated attendant, SIP gateway, SBC…).
          IV.       Elements important settings in VoIP context (used encoding law, compression algorithm).

           V.       Step by step test description.
          VI.       Detailed test result: Call establishment failure, immediate call cut, half way audio from caller
                    to callee, no audio, bad audio quality, bad display on called set…

        For efficient routing to Technical Support structure dedicated to SIP interoperability tests follow-up:
                -   SR summary must contain the following tag: SBD-

                -   SR status summary/short description must contain: “SR followed by: Mr X”.
                -   Immediately after SR creation, a mail should be sent to your TS contact with SR identifier and
                    short summary.

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Copyright © ALE International 2019                                                                         page 38/39
Submitting a Service Request
Please connect to our eService Request application.

Before submitting a Service Request, please be sure:
    The application has been certified via the AAPP if a third party application is involved.
    You have read the release notes that list new features, system requirements, restrictions, and more,
       and are available in the Technical Documentation Library.
    You have read through the related troubleshooting guides and technical bulletins available in the
       Technical Documentation Library.
    You have read through the self-service information on commonly asked support questions and known
       issues and workarounds available in the Technical Knowledge Center.

                                      - END OF DOCUMENT -

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Copyright © ALE International 2019                                                             page 39/39
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