Patton SmartNode: Deutsche Telekom SIP-Trunk

Veröffentlicht am März 29, 2018 von admin

Dieses Config Snippet zeigt, wie ein Patton SmartNode mit Trinity-Firmware korrekt an einem DeutschlandLAN SIP-Trunk der Deutschen Telekom betrieben wird:

profile voip TELEKOM
  codec 1 g711alaw64k rx-length 20 tx-length 20
  dtmf-relay rtp

context cs SWITCH
 
   interface sip IF_SIP_TELEKOM
     bind context sip-gateway GW_SIP_WAN
     route call dest-table RT_FROM_WAN
     remote reg.sip-trunk.telekom.de
     early-disconnect
     privacy
     address-translation outgoing-call to-header user-part call host-part fix sip-trunk.telekom.de
     address-translation outgoing-call request-uri user-part call host-part fix sip-trunk.telekom.de target-param none
     address-translation outgoing-call identity-header user-part call e164 single-primary host-part fix sip-trunk.telekom.de
     address-translation incoming-call calling-e164 from-header
     address-translation incoming-call called-e164 p-called-party-id
     use profile voip TELEKOM

authentication-service AUTH_TELEKOM
  username <Zugangsnummer> password <Kennwort>

location-service LOC_TELEKOM
  domain 1 sip-trunk.telekom.de
  match-any-domain

  identity <Registrierungsrufnummer +49...>
    alias expression .+
    user phone

    authentication outbound
      authenticate 1 authentication-service AUTH_TELEKOM username <Zugangsnummer>

    registration outbound
      registrar sip-trunk.telekom.de
      transport-protocol force tcp
      proxy 1 reg.sip-trunk.telekom.de
      lifetime 600
      register auto
      flows
      nat-traversal crlf 55

    call outbound
      proxy 1 reg.sip-trunk.telekom.de
      transport-protocol force tcp
      request-uri dynamic-registrar
      flows

    call inbound
      use profile voip TELEKOM

profile voip TELEKOM

codec 1 g711alaw64k rx-length 20 tx-length 20

dtmf-relay rtp

context cs SWITCH

interface sip IF_SIP_TELEKOM

bind context sip-gateway GW_SIP_WAN

route call dest-table RT_FROM_WAN

remote reg.sip-trunk.telekom.de

early-disconnect

privacy

address-translation outgoing-call to-header user-part call host-part fix sip-trunk.telekom.de

address-translation outgoing-call request-uri user-part call host-part fix sip-trunk.telekom.de target-param none

address-translation outgoing-call identity-header user-part call e164 single-primary host-part fix sip-trunk.telekom.de

address-translation incoming-call calling-e164 from-header

address-translation incoming-call called-e164 p-called-party-id

use profile voip TELEKOM

authentication-service AUTH_TELEKOM

username <Zugangsnummer> password <Kennwort>

location-service LOC_TELEKOM

domain 1 sip-trunk.telekom.de

match-any-domain

identity <Registrierungsrufnummer +49...>

alias expression .+

user phone

authentication outbound

authenticate 1 authentication-service AUTH_TELEKOM username <Zugangsnummer>

registration outbound

registrar sip-trunk.telekom.de

transport-protocol force tcp

proxy 1 reg.sip-trunk.telekom.de

lifetime 600

flows

nat-traversal crlf 55

call outbound

proxy 1 reg.sip-trunk.telekom.de

transport-protocol force tcp

request-uri dynamic-registrar

flows

call inbound

use profile voip TELEKOM

Patton SmartNode: umgeleiteten ISDN-Anruf an bestimmtes Ziel routen

Veröffentlicht am März 8, 2018 von admin

Wenn ein ankommender Anruf aufgrund einer Rufumleitung (CF) des Anschlusses 030123456789 am SmartNode eintrifft, soll dieser an die Rufnummer 089234567890 an die SIP-PBX geroutet werden.

  routing-table called-e164 RT_FROM_ISDN
    route default dest-interface IF_SIP_PBX MT_CF

  mapping-table calling-redir-e164 to called-e164 MT_CF
    map 030123456789 to 089234567890

routing-table called-e164 RT_FROM_ISDN

route default dest-interface IF_SIP_PBX MT_CF

mapping-table calling-redir-e164 to called-e164 MT_CF

map 030123456789 to 089234567890

Patton SmartNode: Network provided CGPN aus ISDN an SIP-PBX übermitteln

Veröffentlicht am März 8, 2018 von admin

Im ISDN wird neben der vom Anrufer gesetzten Rufnummer (user-provided, not screened) auch die tatsächliche Anschlussrufnummer (network-provided) übermittelt. Diese beiden Rufnummern weichen z.B. voneinander ab, wenn der Anrufer mit clip no screening eine komplett von der Anschlussrufnummer abweichende Rufnummer spooft (in diesem Beispiel 08001234567).

SETUP (DSS1 User)
  [04038090A3]
  Bearer capability : speech - CCITT
    circuit mode - 64kBit/s - G.711 A-law
  [1803A1838B]
  Channel id : 11 - preferred
    other interface - is not d-channel - CCITT - b-channel units
  [6C0C218032313138313136373033]
  Calling party number : 8001234567
    national number - E.164 numbering plan
    presentation allowed - user provided not screened
  [6C082183323131383130]
  Calling party number : 301234567890
    national number - E.164 numbering plan
    presentation allowed - network provided
  [7007C1323132313231]
  Called party number : 30567890
    subscriber number - E.164 numbering plan
  [7D029181]
  High layer compatibility : telephony
    CCITT

SETUP (DSS1 User)

[04038090A3]

Bearer capability : speech - CCITT

circuit mode - 64kBit/s - G.711 A-law

[1803A1838B]

Channel id : 11 - preferred

other interface - is not d-channel - CCITT - b-channel units

[6C0C218032313138313136373033]

Calling party number : 8001234567

national number - E.164 numbering plan

presentation allowed - user provided not screened

[6C082183323131383130]

Calling party number : 301234567890

national number - E.164 numbering plan

presentation allowed - network provided

[7007C1323132313231]

Called party number : 30567890

subscriber number - E.164 numbering plan

[7D029181]

High layer compatibility : telephony

CCITT

Mit der nachfolgenden Konfiguration kann ein Patton SmartNode Gateway beide Rufnummern an die SIP-PBX weiterleiten.

  interface sip IF_SIP
    bind context sip-gateway GW_SIP
    route call dest-table RT_TO_ISDN
    remote 10.10.10.10
    early-disconnect
    privacy
    address-translation outgoing-call identity-header user-part call e164 double-primary-first host-part fix isdn;user=phone

interface sip IF_SIP

bind context sip-gateway GW_SIP

route call dest-table RT_TO_ISDN

remote 10.10.10.10

early-disconnect

privacy

address-translation outgoing-call identity-header user-part call e164 double-primary-first host-part fix isdn;user=phone

Im SIP-INVITE werden die Rufnummern dann so zugeordnet:

P-Asserted-Identity = ISDN network provided
P-Preferred-Identity = ISDN user provided, not screened

So sieht das Ganze dann im SIP-INVITE aus:

INVITE sip:030567890@10.10.10.10:5060 SIP/2.0
Via: SIP/2.0/UDP 10.10.10.20:5060;branch=z9hG4bK78ff5ef91a725aa3b
Max-Forwards: 70
From: <sip:08001234567@10.10.10.20:5060>;tag=aed58e3281
To: <sip:030567890@10.10.10.10:5060>
Call-ID: 10de514fc9ca5984
CSeq: 19829 INVITE
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, INFO, UPDATE, REFER, REGISTER
Contact: <sip:08001234567@10.10.10.20:5060;transport=udp>
P-Asserted-Identity: <sip:0301234567890@isdn;user=phone>
P-Preferred-Identity: <sip:08001234567@isdn;user=phone>
Supported: replaces
User-Agent: Patton SN4970 4E30VR 00A0BA09D858 R6.10 2018-01-09 H323 RBS SIP M5T SIP Stack/4.2.14.18
Content-Type: application/sdp
Content-Length: 185

INVITE sip:030567890@10.10.10.10:5060 SIP/2.0

Via: SIP/2.0/UDP 10.10.10.20:5060;branch=z9hG4bK78ff5ef91a725aa3b

Max-Forwards: 70

From: <sip:08001234567@10.10.10.20:5060>;tag=aed58e3281

To: <sip:030567890@10.10.10.10:5060>

Call-ID: 10de514fc9ca5984

CSeq: 19829 INVITE

Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, INFO, UPDATE, REFER, REGISTER

Contact: <sip:08001234567@10.10.10.20:5060;transport=udp>

P-Asserted-Identity: <sip:0301234567890@isdn;user=phone>

P-Preferred-Identity: <sip:08001234567@isdn;user=phone>

Supported: replaces

User-Agent: Patton SN4970 4E30VR 00A0BA09D858 R6.10 2018-01-09 H323 RBS SIP M5T SIP Stack/4.2.14.18

Content-Type: application/sdp

Content-Length: 185

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VoIP & Real Time Communications

Schlagwort-Archive: Patton SmartNode

Patton SmartNode: Deutsche Telekom SIP-Trunk

Patton SmartNode: umgeleiteten ISDN-Anruf an bestimmtes Ziel routen

Patton SmartNode: Network provided CGPN aus ISDN an SIP-PBX übermitteln