Ciao ragazzi,

mi trovo in questa condizione:

Elastix 1.5.2-2.3 (ultima stable)
Patton SmartNode 4554 2 bri

Premettendo che le chiamate in ingresso e uscita funzionano regolarmente e che il pbx è configurato a modino, che di conseguenza la trunk pbx <-> patton è registrata ed è up....

vorrei sapere perchè il pbx non si rende conto di quando i chiamanti riagganciano dopo essere passati dall'IVR (poichè se reindirizzo direttamente su una extension e riaggancio funziona regolare).


Il percorso INBOUND è il seguente:

ring up -> DID check -> Day and Night Check -> Time Condition Check 1 -> Time Condition Check 2 -> IVR ingresso con direct extensions call abilitato e timeout 6 sec

percorso (t)  -> Ring group centralino
percorso (i) -> Ripeti IVR

Le follow me sono per ogni singola extensions

ring ext -> No answers? -> Follow me -> Queue Ext. -> retry ext. every 10 sec until 5 min. -> Announce (tutti utenti occupati) -> Hang Up.


Quello che succede di anomalo è che se una chiamata in ingresso passa attraverso l'IVR e magari riaggancia prima che l'operatore dell'estensione risponda la chiamata rimane attiva come se l'utente chiamante fosse sempre al telefono, l'operatore risponde e sente il classico tono di occupato continuo.

Cosa mi consigliate di fare?



Andrea

Ciao, ho anch'io lo stesso problema con patton 4634 e asterisk.
Senza IVR funziona tutto, con IVR non sente il Congestion.
Ho già settato tutte le segnalazioni del patton uguali al profilo IT dell'indication.conf di asterisk.

se qualcuno ha soluzioni si faccia avanti, grazie

mdmlink ha scritto:Ciao, ho anch'io lo stesso problema con patton 4634 e asterisk.
Senza IVR funziona tutto, con IVR non sente il Congestion.
Ho già settato tutte le segnalazioni del patton uguali al profilo IT dell'indication.conf di asterisk.

se qualcuno ha soluzioni si faccia avanti, grazie

Ciao mdmlink,

cosa intendi per settaggio dell'indication.conf?

Questo?

http://www.voip-info.org/wiki/view/Asterisk+config+indications.conf

Vi riporto di seguito la mia configurazione attuale del Patton se vi può dare una mano a capire:


#----------------------------------------------------------------#
#                                                                #
# SN4554/2BIS/EUI                                                #
# R4.2 2007-09-19 SIP                                            #
# 2009-07-01T09:05:28                                            #
# SN/00A0BA045848                                                #
# Generated configuration file                                  #
#                                                                #
#----------------------------------------------------------------#

cli version 3.20
gui type basic
clock local offset +01:00
dns-client server 172.16.200.11
dns-relay
webserver port 80 language en
sntp-client
sntp-client server primary 129.132.2.21 port 123 version 4

system

  ic voice 0

system
  clock-source 1 bri 0 0
  clock-source 2 bri 0 1

profile acl ACL_WAN_PERMIT_SEL_MGMT

profile service-policy SP_WAN_OUT
  no rate-limit

profile service-policy SP_WAN_IN
  no rate-limit

profile napt NAPT_WAN

profile ppp default

profile call-progress-tone IT_Dialtone
  play 1 200 425 -12
  pause 2 200
  play 3 600 425 -12
  pause 4 1000

profile call-progress-tone IT_Alertingtone
  play 1 1000 425 -12
  pause 2 4000

profile call-progress-tone IT_Busytone
  play 1 500 425 -12
  pause 2 500

profile tone-set default

profile tone-set IT
  map call-progress-tone dial-tone IT_Dialtone
  map call-progress-tone ringback-tone IT_Alertingtone
  map call-progress-tone busy-tone IT_Busytone
  map call-progress-tone release-tone IT_Busytone
  map call-progress-tone congestion-tone IT_Busytone

profile tone-set default

profile voip default
  codec 1 g711alaw64k rx-length 20 tx-length 20
  codec 2 g711ulaw64k rx-length 20 tx-length 20

profile voip VOIP
  codec 1 g729 rx-length 20 tx-length 20
  codec 2 g711alaw64k rx-length 20 tx-length 20
  codec 3 g711ulaw64k rx-length 20 tx-length 20
  dejitter-mode static
  dejitter-max-delay 120

profile pstn default

profile sip default

profile aaa default
  method 1 local
  method 2 none

context ip router

  interface IF_IP_WAN
    ipaddress 172.16.200.15 255.255.255.0
    use profile napt NAPT_WAN
    tcp adjust-mss rx mtu
    tcp adjust-mss tx mtu

  interface IF_IP_LAN
    ipaddress unnumbered

context ip router
  route 0.0.0.0 0.0.0.0 172.16.200.254 0

subscriber ppp SUB_PPPOE
  dial in
  no multilink

context cs switch
  national-prefix 0
  international-prefix 00

  interface isdn IF_S0_00
    route call dest-interface IF_ASTERISK_0

  interface isdn IF_S0_01
    route call dest-interface IF_ASTERISK_1

  interface sip IF_SIP_SERVICE
    service default

  interface sip IF_ASTERISK_0
    bind gateway GW_ASTERISK_0
    service patton_0
    route call dest-interface IF_S0_00

  interface sip IF_ASTERISK_1
    bind gateway GW_ASTERISK_1
    service patton_1
    route call dest-interface IF_S0_01

  service hunt-group SER_HG_PSTN_FALLBACK
    drop-cause normal-unspecified
    drop-cause no-circuit-channel-available
    drop-cause network-out-of-order
    drop-cause temporary-failure
    drop-cause switching-equipment-congestion
    drop-cause access-info-discarded
    drop-cause circuit-channel-not-available
    drop-cause resources-unavailable

context cs switch
  no shutdown

gateway sip GW_SIP

  service default

gateway sip GW_SIP
  no shutdown

gateway sip GW_ASTERISK_0
  call-signaling-port 5062
  bind interface IF_IP_WAN router

  service patton_0
    defaultserver manual 172.16.200.14 5060 loose-router
    registration manual 172.16.200.14 5060 use-default-server
    user PATTON-0 authenticate password xxxxxxxxxxxxxx encrypted default register display-name PATTON-0 phone-context from-trunk

gateway sip GW_ASTERISK_0
  no shutdown

gateway sip GW_ASTERISK_1
  call-signaling-port 5064
  bind interface IF_IP_WAN router

  service patton_1
    defaultserver manual 172.16.200.14 5060 loose-router
    registration manual 172.16.200.14 5060 use-default-server
    user PATTON-1 authenticate password xxxxxxxxxxxxxx encrypted default register display-name PATTON-1 phone-context from-trunk

gateway sip GW_ASTERISK_1
  no shutdown

port ethernet 0 0
  bind interface IF_IP_WAN router

  pppoe

    session SES_PPPOE
      shutdown

port ethernet 0 0
  no shutdown

port bri 0 0
  clock auto
  encapsulation q921

  q921
    uni-side auto
    encapsulation q931

    q931
      protocol dss1
      uni-side user
      bchan-number-order ascending
      encapsulation cc-isdn
      bind interface IF_S0_00 switch

port bri 0 0
  no shutdown

port bri 0 1
  clock auto
  encapsulation q921

  q921
    uni-side auto
    encapsulation q931

    q931
      protocol dss1
      uni-side user
      bchan-number-order ascending
      encapsulation cc-isdn
      bind interface IF_S0_01 switch

port bri 0 1
  no shutdown

per indication intendo esattamente indication.conf e io ho settato come default IT.
Nel patton ho creato come te i toni con le stesse indicazioni di asterisk:

profile call-progress-tone IT_DIALTONE
  play 1 200 425 -12
  pause 2 200
  play 3 600 425 -12
  pause 4 1000

profile call-progress-tone IT_ALERTINGTONE
  play 1 1000 425 -12
  pause 2 4000

profile call-progress-tone IT_BUSYTONE
  play 1 500 425 -12
  pause 2 500

profile call-progress-tone IT_RELEASETONE
  play 1 250 425 -12
  pause 2 250

profile call-progress-tone IT_CONGESTION
  play 1 200 425 -12
  pause 2 200

profile tone-set default
profile tone-set IT
  map call-progress-tone dial-tone IT_DIALTONE
  map call-progress-tone ringback-tone IT_ALERTINGTONE
  map call-progress-tone busy-tone IT_BUSYTONE
  map call-progress-tone release-tone IT_BUSYTONE
  map call-progress-tone congestion-tone IT_CONGESTION


il problema però è un altro perchè se sono fuori dall?IVR funziona tutto, con l?IVR non sente l'hangup

si infatti sono più che convinto che il problema non sia il patton come molti dicono, ma sia proprio un problema di riconoscimento del congestion in elastix, e più precisamente nel modulo dell'IVR.

Questo è ovvio perchè se si devìa la chiamata ad un'extension non ci sono problemi di sorta.

E' veramente fastidiosa come cosa perchè mi ritrovo diverse volte a prendere la chiamata e sentire il tono di occupato di qualcuno che ha riagganciato, oppure mi trovo messaggi in segreteria con toni di occupato continui.

Tra l'altro se in chiamata si riaggancia dall'esterno il sistema riconosce l'hangup correttamente e riaggancia. Non capisco perchè passando dall'IVR ciò non accade.

Nelle interfaccie SIP del patton nel context cs switch aggiungi il seguente parametro:

early-disconnect

starlab ha scritto:Nelle interfaccie SIP del patton nel context cs switch aggiungi il seguente parametro:

early-disconnect

ho appena provato,

se lo inserisco la context diventa

context cs switch
early-disconnect
no shutdown


ma così facendo il pbx non mi aggancia più le chiamate in ingresso, sento solo un tono di libero.

Grazie Starlab

ora finalmente mi chiude la linea 

mdmlink ha scritto:Grazie Starlab

ora finalmente mi chiude la linea 

Allora devo essere un pirla io... aspetta che riprovo... magari sono veramente un pirla...

C'è qualcosa che non va....

se inserisco early-disconnect nella mia config. e la importo nel patton smette di agganciarmi le chiamate in ingresso. Se lo rimuovo e faccio il reload funziona ma il problema ovviamente sussiste.

Posto la config. modificata:


COSI' IL PBX RISPONDE, MA IL PROBLEMA RIMANE

Codice: Seleziona tutto


#----------------------------------------------------------------#
#                                                                #
# SN4554/2BIS/EUI                                                #
# R4.2 2007-09-19 SIP                                            #
# 2009-07-01T09:05:28                                            #
# SN/00A0BA045848                                                #
# Generated configuration file                                   #
#                                                                #
#----------------------------------------------------------------#

cli version 3.20
gui type basic
clock local offset +01:00
dns-client server 172.16.200.11
dns-relay
webserver port 80 language en
sntp-client
sntp-client server primary 129.132.2.21 port 123 version 4

system

  ic voice 0

system
  clock-source 1 bri 0 0
  clock-source 2 bri 0 1

profile acl ACL_WAN_PERMIT_SEL_MGMT

profile service-policy SP_WAN_OUT
  no rate-limit

profile service-policy SP_WAN_IN
  no rate-limit

profile napt NAPT_WAN

profile ppp default

profile call-progress-tone IT_Dialtone
  play 1 200 425 -12
  pause 2 200
  play 3 600 425 -12
  pause 4 1000

profile call-progress-tone IT_Alertingtone
  play 1 1000 425 -12
  pause 2 4000

profile call-progress-tone IT_Busytone
  play 1 500 425 -12
  pause 2 500

profile tone-set default

profile tone-set IT
  map call-progress-tone dial-tone IT_Dialtone
  map call-progress-tone ringback-tone IT_Alertingtone
  map call-progress-tone busy-tone IT_Busytone
  map call-progress-tone release-tone IT_Busytone
  map call-progress-tone congestion-tone IT_Busytone

profile tone-set default

profile voip default
  codec 1 g711alaw64k rx-length 20 tx-length 20
  codec 2 g711ulaw64k rx-length 20 tx-length 20

profile voip VOIP
  codec 1 g729 rx-length 20 tx-length 20
  codec 2 g711alaw64k rx-length 20 tx-length 20
  codec 3 g711ulaw64k rx-length 20 tx-length 20
  dejitter-mode static
  dejitter-max-delay 120

profile pstn default

profile sip default

profile aaa default
  method 1 local
  method 2 none

context ip router

  interface IF_IP_WAN
    ipaddress 172.16.200.15 255.255.255.0
    use profile napt NAPT_WAN
    tcp adjust-mss rx mtu
    tcp adjust-mss tx mtu

  interface IF_IP_LAN
    ipaddress unnumbered

context ip router
  route 0.0.0.0 0.0.0.0 172.16.200.254 0

subscriber ppp SUB_PPPOE
  dial in
  no multilink

context cs switch
  national-prefix 0
  international-prefix 00

  interface isdn IF_S0_00
    route call dest-interface IF_ASTERISK_0

  interface isdn IF_S0_01
    route call dest-interface IF_ASTERISK_1

  interface sip IF_SIP_SERVICE
    service default

  interface sip IF_ASTERISK_0
    bind gateway GW_ASTERISK_0
    service patton_0
    route call dest-interface IF_S0_00

  interface sip IF_ASTERISK_1
    bind gateway GW_ASTERISK_1
    service patton_1
    route call dest-interface IF_S0_01

  service hunt-group SER_HG_PSTN_FALLBACK
    drop-cause normal-unspecified
    drop-cause no-circuit-channel-available
    drop-cause network-out-of-order
    drop-cause temporary-failure
    drop-cause switching-equipment-congestion
    drop-cause access-info-discarded
    drop-cause circuit-channel-not-available
    drop-cause resources-unavailable

context cs switch
  no shutdown

gateway sip GW_SIP

  service default

gateway sip GW_SIP
  no shutdown

gateway sip GW_ASTERISK_0
  call-signaling-port 5062
  bind interface IF_IP_WAN router

  service patton_0
    defaultserver manual 172.16.200.14 5060 loose-router
    registration manual 172.16.200.14 5060 use-default-server
    user PATTON-0 authenticate password xxxxxxxxxxxxxx encrypted default register display-name PATTON-0 phone-context from-trunk

gateway sip GW_ASTERISK_0
  no shutdown

gateway sip GW_ASTERISK_1
  call-signaling-port 5064
  bind interface IF_IP_WAN router

  service patton_1
    defaultserver manual 172.16.200.14 5060 loose-router
    registration manual 172.16.200.14 5060 use-default-server
    user PATTON-1 authenticate password xxxxxxxxxxxxxx encrypted default register display-name PATTON-1 phone-context from-trunk

gateway sip GW_ASTERISK_1
  no shutdown

port ethernet 0 0
  bind interface IF_IP_WAN router

  pppoe

    session SES_PPPOE
      shutdown

port ethernet 0 0
  no shutdown

port bri 0 0
  clock auto
  encapsulation q921

  q921
    uni-side auto
    encapsulation q931

    q931
      protocol dss1
      uni-side user
      bchan-number-order ascending
      encapsulation cc-isdn
      bind interface IF_S0_00 switch

port bri 0 0
  no shutdown

port bri 0 1
  clock auto
  encapsulation q921

  q921
    uni-side auto
    encapsulation q931

    q931
      protocol dss1
      uni-side user
      bchan-number-order ascending
      encapsulation cc-isdn
      bind interface IF_S0_01 switch

port bri 0 1
  no shutdown


COSI' IL PBX NON AGGANCIA LA CHIAMATA IN INGRESSO
contiene la modifica di starlab

Codice: Seleziona tutto



#----------------------------------------------------------------#
#                                                                #
# SN4554/2BIS/EUI                                                #
# R4.2 2007-09-19 SIP                                            #
# 2009-07-01T09:05:28                                            #
# SN/00A0BA045848                                                #
# Generated configuration file                                   #
#                                                                #
#----------------------------------------------------------------#

cli version 3.20
gui type basic
clock local offset +01:00
dns-client server 172.16.200.11
dns-relay
webserver port 80 language en
sntp-client
sntp-client server primary 129.132.2.21 port 123 version 4

system

  ic voice 0

system
  clock-source 1 bri 0 0
  clock-source 2 bri 0 1

profile acl ACL_WAN_PERMIT_SEL_MGMT

profile service-policy SP_WAN_OUT
  no rate-limit

profile service-policy SP_WAN_IN
  no rate-limit

profile napt NAPT_WAN

profile ppp default

profile call-progress-tone IT_Dialtone
  play 1 200 425 -12
  pause 2 200
  play 3 600 425 -12
  pause 4 1000

profile call-progress-tone IT_Alertingtone
  play 1 1000 425 -12
  pause 2 4000

profile call-progress-tone IT_Busytone
  play 1 500 425 -12
  pause 2 500

profile tone-set default

profile tone-set IT
  map call-progress-tone dial-tone IT_Dialtone
  map call-progress-tone ringback-tone IT_Alertingtone
  map call-progress-tone busy-tone IT_Busytone
  map call-progress-tone release-tone IT_Busytone
  map call-progress-tone congestion-tone IT_Busytone

profile tone-set default

profile voip default
  codec 1 g711alaw64k rx-length 20 tx-length 20
  codec 2 g711ulaw64k rx-length 20 tx-length 20

profile voip VOIP
  codec 1 g729 rx-length 20 tx-length 20
  codec 2 g711alaw64k rx-length 20 tx-length 20
  codec 3 g711ulaw64k rx-length 20 tx-length 20
  dejitter-mode static
  dejitter-max-delay 120

profile pstn default

profile sip default

profile aaa default
  method 1 local
  method 2 none

context ip router

  interface IF_IP_WAN
    ipaddress 172.16.200.15 255.255.255.0
    use profile napt NAPT_WAN
    tcp adjust-mss rx mtu
    tcp adjust-mss tx mtu

  interface IF_IP_LAN
    ipaddress unnumbered

context ip router
  route 0.0.0.0 0.0.0.0 172.16.200.254 0

subscriber ppp SUB_PPPOE
  dial in
  no multilink

context cs switch
  national-prefix 0
  international-prefix 00

  interface isdn IF_S0_00
    route call dest-interface IF_ASTERISK_0

  interface isdn IF_S0_01
    route call dest-interface IF_ASTERISK_1

  interface sip IF_SIP_SERVICE
    service default

  interface sip IF_ASTERISK_0
    bind gateway GW_ASTERISK_0
    service patton_0
    route call dest-interface IF_S0_00

  interface sip IF_ASTERISK_1
    bind gateway GW_ASTERISK_1
    service patton_1
    route call dest-interface IF_S0_01

  service hunt-group SER_HG_PSTN_FALLBACK
    drop-cause normal-unspecified
    drop-cause no-circuit-channel-available
    drop-cause network-out-of-order
    drop-cause temporary-failure
    drop-cause switching-equipment-congestion
    drop-cause access-info-discarded
    drop-cause circuit-channel-not-available
    drop-cause resources-unavailable

context cs switch
  early-disconnect          <--- ECCO LA MODIFICA
  no shutdown

gateway sip GW_SIP

  service default

gateway sip GW_SIP
  no shutdown

gateway sip GW_ASTERISK_0
  call-signaling-port 5062
  bind interface IF_IP_WAN router

  service patton_0
    defaultserver manual 172.16.200.14 5060 loose-router
    registration manual 172.16.200.14 5060 use-default-server
    user PATTON-0 authenticate password xxxxxxxxxxxxxx encrypted default register display-name PATTON-0 phone-context from-trunk

gateway sip GW_ASTERISK_0
  no shutdown

gateway sip GW_ASTERISK_1
  call-signaling-port 5064
  bind interface IF_IP_WAN router

  service patton_1
    defaultserver manual 172.16.200.14 5060 loose-router
    registration manual 172.16.200.14 5060 use-default-server
    user PATTON-1 authenticate password xxxxxxxxxxxxxx encrypted default register display-name PATTON-1 phone-context from-trunk

gateway sip GW_ASTERISK_1
  no shutdown

port ethernet 0 0
  bind interface IF_IP_WAN router

  pppoe

    session SES_PPPOE
      shutdown

port ethernet 0 0
  no shutdown

port bri 0 0
  clock auto
  encapsulation q921

  q921
    uni-side auto
    encapsulation q931

    q931
      protocol dss1
      uni-side user
      bchan-number-order ascending
      encapsulation cc-isdn
      bind interface IF_S0_00 switch

port bri 0 0
  no shutdown

port bri 0 1
  clock auto
  encapsulation q921

  q921
    uni-side auto
    encapsulation q931

    q931
      protocol dss1
      uni-side user
      bchan-number-order ascending
      encapsulation cc-isdn
      bind interface IF_S0_01 switch

port bri 0 1
  no shutdown




Mi aiutate a capire dove sbaglio? 

ciao,
ti posto la configurazione del mio patton così vedi dove inserire la modifica

Codice: Seleziona tutto

#----------------------------------------------------------------#
#                                                                #
# SN4634/3BIS                                                    #
# R5.2 2009-01-14 H323 SIP BRI                                   #
# 1970-01-02T21:51:43                                            #
# SN/00A0BA049B9E                                                #
# Generated configuration file                                   #
#                                                                #
#----------------------------------------------------------------#

cli version 3.20
administrator administrator password 6j/jXWVC9P/GLlvRBeEerQ== encrypted
dns-client server 192.168.1.1
webserver port 80 language en

system

  ic voice 0

system
  clock-source 1 bri 0 0
  clock-source 2 bri 0 1
  clock-source 3 bri 0 2

profile napt NAPT_WAN

profile ppp default

profile call-progress-tone IT_DIALTONE
  play 1 200 425 -12
  pause 2 200
  play 3 600 425 -12
  pause 4 1000

profile call-progress-tone IT_ALERTINGTONE
  play 1 1000 425 -12
  pause 2 4000

profile call-progress-tone IT_BUSYTONE
  play 1 500 425 -12
  pause 2 500

profile call-progress-tone IT_RELEASETONE
  play 1 250 425 -12
  pause 2 250

profile call-progress-tone IT_CONGESTION
  play 1 200 425 -12
  pause 2 200

profile tone-set default
profile tone-set IT
  map call-progress-tone dial-tone IT_DIALTONE
  map call-progress-tone ringback-tone IT_ALERTINGTONE
  map call-progress-tone busy-tone IT_BUSYTONE
  map call-progress-tone release-tone IT_BUSYTONE
  map call-progress-tone congestion-tone IT_CONGESTION

profile voip default
  codec 1 g711alaw64k rx-length 20 tx-length 20
  codec 2 g711ulaw64k rx-length 20 tx-length 20

profile pstn default

profile sip default

profile aaa default
  method 1 local
  method 2 none

context ip router

  interface eth0
    ipaddress 192.168.2.2 255.255.0.0
    use profile napt NAPT_WAN

context ip router
  route 0.0.0.0 0.0.0.0 192.168.1.254 0

context cs switch

  routing-table called-e164 rt_tel_to_ip
    route .T2 dest-interface if_sip_1

  routing-table called-e164 rt_ip_to_tel
    route .T2 dest-service hg_isdn

  interface isdn if_isdn_1
    route call dest-table rt_tel_to_ip
    use profile tone-set IT

  interface isdn if_isdn_2
    route call dest-table rt_tel_to_ip
    use profile tone-set IT

  interface sip if_sip_1
    bind context sip-gateway GW_SIP
    route call dest-table rt_ip_to_tel
    remote 192.168.2.1 5060
    local 192.168.2.1 5060
    early-disconnect
    use profile tone-set IT

  service hunt-group hg_isdn
    cyclic
    drop-cause normal-unspecified
    drop-cause no-circuit-channel-available
    drop-cause network-out-of-order
    drop-cause temporary-failure
    drop-cause switching-equipment-congestion
    drop-cause access-info-discarded
    drop-cause circuit-channel-not-available
    drop-cause resources-unavailable
    route call 1 dest-interface if_isdn_1
    route call 2 dest-interface if_isdn_2

context cs switch
  no shutdown

authentication-service AS_50
  username 50 password HVOrjMjGXnka81sl5oKPiQ== encrypted

location-service AsteriskPBX
  domain 1 192.168.2.1 5060

  identity 50
    display-name 50

    authentication outbound
      authenticate 1 authentication-service AS_50 username 50

    registration outbound
      lifetime 3600
      register auto
      retry-timeout on-system-error 10
      retry-timeout on-client-error 10
      retry-timeout on-server-error 10

context sip-gateway GW_SIP

  interface IF_SIP
    bind interface eth0 context router port 5060

context sip-gateway GW_SIP
  bind location-service AsteriskPBX
  no shutdown

port ethernet 0 0
  medium 10 half
  encapsulation ip
  bind interface eth0 router
  no shutdown

port ethernet 0 1
  medium 10 half
  shutdown

port bri 0 0
  clock auto
  encapsulation q921

  q921
    uni-side auto
    encapsulation q931

    q931
      protocol dss1
      uni-side net
      bchan-number-order ascending

port bri 0 0
  shutdown

port bri 0 1
  clock auto
  encapsulation q921

  q921
    permanent-layer2
    uni-side user
    encapsulation q931

    q931
      protocol dss1
      uni-side user
      bchan-number-order ascending
      encapsulation cc-isdn
      bind interface if_isdn_1 switch

port bri 0 1
  no shutdown

port bri 0 2
  clock auto
  encapsulation q921

  q921
    permanent-layer2
    uni-side user
    encapsulation q931

    q931
      protocol dss1
      uni-side user
      bchan-number-order ascending
      encapsulation cc-isdn
      bind interface if_isdn_2 switch

port bri 0 2
  no shutdown






La inserisci nel posto errato, rileggi con attenzione quello che ho scritto.

starlab ha scritto:La inserisci nel posto errato, rileggi con attenzione quello che ho scritto.

Starlab, che dire, a parte che effettivamente l'avevo inserito nel punto errato della context. Tu ti riferivi alla context SIP ed io l'ho inserito nel punto errato.

Grazie per avermi aiutato ancora una volta!

PS: un grazie a mdmlink per il supporto.

Andrea