Elastix + Patton 4554 -> IVR passa la chiamata e... NO HANGUP!

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	Autore	Discussione: Elastix + Patton 4554 -> IVR passa la chiamata e... NO HANGUP! (Letto 1471 volte)
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Elastix + Patton 4554 -> IVR passa la chiamata e... NO HANGUP!

lovelord83

Beginner

Messaggi: 24

Giugno 23, 2009, 11:25:46 am

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?

Che?!?

Andrea


« Ultima modifica: Luglio 03, 2009, 01:35:37 pm da lovelord83 »	Registrato

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Elastix + Patton 4554 -> IVR passa la chiamata e... NO HANGUP!

« inserita:: Giugno 23, 2009, 11:25:46 am »

Centralino IP PBX - 3CX. Scarica la versione Gratuita!


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#1 Re: Elastix + Patton 4552 -> IVR passa la chiamata e... NO HANGUP!

mdmlink

PrimoLivello

Messaggi: 10

Luglio 03, 2009, 11:15:22 am

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


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#2 Re: Elastix + Patton 4554 -> IVR passa la chiamata e... NO HANGUP!

lovelord83

Beginner

Messaggi: 24

Luglio 03, 2009, 01:12:05 pm

Citazione di: mdmlink - Luglio 03, 2009, 11:15:22 am

Ciao mdmlink,

cosa intendi per settaggio dell'indication.conf?

Questo?

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


« Ultima modifica: Luglio 03, 2009, 01:33:34 pm da lovelord83 »	Registrato

#3 Re: Elastix + Patton 4554 -> IVR passa la chiamata e... NO HANGUP!

lovelord83

Beginner

Messaggi: 24

Luglio 03, 2009, 01:32:56 pm

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


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#4 Re: Elastix + Patton 4554 -> IVR passa la chiamata e... NO HANGUP!

mdmlink

PrimoLivello

Messaggi: 10

Luglio 03, 2009, 01:50:10 pm

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


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#5 Re: Elastix + Patton 4554 -> IVR passa la chiamata e... NO HANGUP!

lovelord83

Beginner

Messaggi: 24

Luglio 03, 2009, 01:54:09 pm

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.

Che?!?


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#6 Re: Elastix + Patton 4554 -> IVR passa la chiamata e... NO HANGUP!

starlab

Moderator
Guru

Messaggi: 1624

Luglio 03, 2009, 02:33:50 pm

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

early-disconnect


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#7 Re: Elastix + Patton 4554 -> IVR passa la chiamata e... NO HANGUP!

lovelord83

Beginner

Messaggi: 24

Luglio 03, 2009, 02:48:08 pm

Citazione di: starlab - Luglio 03, 2009, 02:33:50 pm

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.


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#8 Re: Elastix + Patton 4554 -> IVR passa la chiamata e... NO HANGUP!

mdmlink

PrimoLivello

Messaggi: 10

Luglio 03, 2009, 02:51:39 pm

Grazie Starlab

ora finalmente mi chiude la linea


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#9 Re: Elastix + Patton 4554 -> IVR passa la chiamata e... NO HANGUP!

lovelord83

Beginner

Messaggi: 24

Luglio 03, 2009, 02:52:53 pm

Citazione di: mdmlink - Luglio 03, 2009, 02:51:39 pm

Grazie Starlab

ora finalmente mi chiude la linea

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


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#10 Re: Elastix + Patton 4554 -> IVR passa la chiamata e... NO HANGUP!

lovelord83

Beginner

Messaggi: 24

Luglio 03, 2009, 03:14:23 pm

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:

#----------------------------------------------------------------#
# #
# 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:

#----------------------------------------------------------------#
# #
# 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? Occhi al cielo


« Ultima modifica: Luglio 03, 2009, 03:16:18 pm da lovelord83 »	Registrato

#11 Re: Elastix + Patton 4554 -> IVR passa la chiamata e... NO HANGUP!

mdmlink

PrimoLivello

Messaggi: 10

Luglio 03, 2009, 03:22:29 pm

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

Codice:

#----------------------------------------------------------------#
# #
# 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


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#12 Re: Elastix + Patton 4554 -> IVR passa la chiamata e... NO HANGUP!

starlab

Moderator
Guru

Messaggi: 1624

Luglio 03, 2009, 03:24:20 pm

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


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#13 Re: Elastix + Patton 4554 -> IVR passa la chiamata e... NO HANGUP!

lovelord83

Beginner

Messaggi: 24

Luglio 03, 2009, 03:37:41 pm

Citazione di: starlab - Luglio 03, 2009, 03:24:20 pm

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