Marco's Informatik Blog

Il perchè del topic è presto detto…rispondono a caso sapendo di non poter essere in qualche modo cazziati!
Da un cliente che ha tre borchie ISDN, ho chiesto un numero aggiuntivo su una di esse e dato che queste tre borchie erano in ricerca automatica, la borchia con il numero aggiuntivo è stata tolta dalla r.a., questo perchè le borchie multinumero non funzionano in modalità PBX (di solito borchie punto punto, ma anche punto multipunto, capita).
Comunque sia ovviamente c’era da riprogrammare anche quella del capofila per modificare la r.a. solo su due borchie…e qui comincia l’odissea.
Ieri la borchia capofila non andava affatto, l’altra si, mentre da oggi pomeriggio nemmeno questa, ne ingresso ne uscita.
Ovviamente la Telecom chi ha incolpato? ME! Il tecnico del centralino, con la solita storiella “c’è da ripogrammare il sistema…..” UEEE! Apparte che quella configurazione sul patton gira su N apparati, ma se prima che LORO mettessero le mani andava il tutto con 3 borchie, non vedo perchè non deve andare con due!!!!
Poi siccome loro pensano SEMPRE di aver a che fare con dei piccioni dall’altra parte del telefono (cose del tipo: non ho portante ADSL e loro “controllato i DNS?”), ma a questo punto mi pare chiaro il problema dal debug isdn del patton:

00:19:34  ISDN  > # 665 p: 1 R: sapi: 0 cr=1 ea=0 tei:  0 ea=1 SABME                p =1
00:19:35  ISDN  > # 666 p: 1 R: sapi: 0 cr=1 ea=0 tei:  0 ea=1 SABME                p =1
00:19:36  ISDN  > # 667 p: 1 R: sapi: 0 cr=1 ea=0 tei:  0 ea=1 SABME                p =1
00:19:37  ISDN  > # 668 p: 1 R: sapi: 0 cr=1 ea=0 tei:  0 ea=1 SABME                p =1
00:19:38  ISDN  > # 669 p: 1 R: sapi: 0 cr=1 ea=0 tei:  0 ea=1 SABME                p =1
00:19:39  ISDN  > # 670 p: 1 R: sapi: 0 cr=1 ea=0 tei:  0 ea=1 SABME                p =1
00:19:40  ISDN  > # 671 p: 1 R: sapi: 0 cr=1 ea=0 tei:  0 ea=1 SABME                p =1
00:19:41  ISDN  > # 672 p: 1 R: sapi: 0 cr=1 ea=0 tei:  0 ea=1 SABME                p =1
00:19:42  ISDN  > # 673 p: 1 R: sapi: 0 cr=1 ea=0 tei:  0 ea=1 SABME                p =1
00:19:43  ISDN  > # 674 p: 1 R: sapi: 0 cr=1 ea=0 tei:  0 ea=1 SABME                p =1
00:19:44  ISDN  > # 675 p: 1 R: sapi: 0 cr=1 ea=0 tei:  0 ea=1 SABME                p =1
00:19:45  ISDN  > # 676 p: 1 R: sapi: 0 cr=1 ea=0 tei:  0 ea=1 SABME                p =1

Ovviamente questa musica (tentativo di riconnessione) si ripete all’infinito su entrambe le porte atatccate alle borchie incriminate.
Quindi che dire, vediamo domani se rispondono di nuovo al mio sollecito di gusto con qualche altra storiella!
Come chiusura di post vi ricordo come attivare il debug su uan determinata porta isdn con il patton

login: {username qui}
password: {password qui}
enable
configure
debug ccisdn signaling
debug isdn event {slot} {porta} all

per esempio per il debug sulla bri 0/0

debug isdn event 0 0 all
mentre per disabilitare il debug, no debug all

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Sto attrezzandomi per un altra iniziativa, ovvera la realizzazione di video corsi.
Per adesso ho realizzato un video per la creazione di un trunk custom su Trixbox, utile ad esempio per configurare un Patton qualisiasi.
Questo può affiancarsi alla guida che propongo con una piccola contribuzione, in alternativa alla guida cartacea + il file di configurazione del Patton.
Come esempio in questo caso uso Trixbox 2.6.0.7, se eventualmente siete interessati vi comunico “la contribuzione” su consulenze at ciacci.biz
Ciao!

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Brevissimo post per dirvi che da qualche giorno è pronto (forse ha bisogno di qualche aggiustatina, ma mi pare che sia ok per le cose base!) il configuratore web per i Patton 4552,4634 e 4638.
Per maggiori info registratevi su www.ilmiovoip.com, il mio portale (nuovo e ancora acerbo!) dedicato completamente al VoIP.

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Piccolo aggiornamento sul post precedente, ho dovuto fare una piccola correzione alla configurazione postata in precedenza, in quanto non era necessario dtmf-dialing, ed il sintomo che dava era curioso, ovvero il numero di telefono digitato da alcuni apparecchi collegati al PBX classico, erano “storpiati”.
Ad esempio se io digitavo 0577123456 questo mi diventava 0505777711223345455656 ad esempio!!!
Togliendolo tutto ok, inoltre ho creato un nuovo utente per usarlo come “friend” lato pbx ed inoltre ho aggiunto delle rotte per chiamare i cellulari, 8XX e 7XX

#—————————————————————-#
# #
# SN4638/5BIS/UI #
# R3.20 2006-11-17 H323 SIP BRI #
# 2007-07-27T06:39:21 #
# Generated configuration file #
# #
#—————————————————————-#

cli version 3.20
dns-client server 88.149.128.12
webserver port 80 language en
snmp community public ro

system

ic voice 0

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

profile voip ASTERISK
codec 1 g729 rx-length 20 tx-length 20
codec 2 g711ulaw64k rx-length 20 tx-length 20
codec 3 g711alaw64k rx-length 20 tx-length 20

profile pstn default

profile sip default

profile sip asterisk

profile aaa default
method 1 local
method 2 none

context ip router

interface IF_IP_WAN
ipaddress 192.168.1.101 255.255.255.0
no napt-inside

interface IF_IP_LAN
ipaddress 192.168.1.1 255.255.255.0
no napt-inside

context cs switch
digit-collection timeout 4
national-prefix 0
international-prefix 00

routing-table called-e164 RT_2_ISDN
route .%T dest-service HUNTING MT_ITC

routing-table called-e164 RT_ISDN_2_SIP
route 99[1-9].T3 dest-interface IF_S0_PSTN
route 1[1-9].T3 dest-service SER_HUNT_OUT
route 0[1-9].T3 dest-service SER_HUNT_OUT
route 00[1-9].T3 dest-service SER_HUNT_OUT
route 8[0-9].T3 dest-service SER_HUNT_OUT
route 7[0-9].T3 dest-service SER_HUNT_OUT
route 3[0-9].T3 dest-service SER_HUNT_OUT
route default dest-service SER_HUNT_OUT

mapping-table itc to itc MT_ITC
map default to speech

interface isdn IF_S0_PSTN
route call dest-interface IF_S0_PHONE
dtmf-dialing

interface isdn IF_S0_PHONE1
route call dest-table RT_ISDN_2_SIP
use profile tone-set IT

interface isdn IF_S0_PHONE2
route call dest-table RT_ISDN_2_SIP
use profile tone-set IT

interface isdn IF_S0_PHONE3
route call dest-table RT_ISDN_2_SIP
use profile tone-set IT

interface isdn IF_S0_PHONE4
route call dest-table RT_ISDN_2_SIP
use profile tone-set IT

interface sip IF_SIP_ASTERISK
bind gateway GW-ASTERISK
service default
route call dest-table RT_2_ISDN
early-disconnect
remote-party-id called-party
remote-party-id calling-party
address-translation outgoing-call from-header user-part fix 210 host-part call
use profile voip ASTERISK

service hunt-group SER_HUNT_OUT
timeout 6
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_SIP_ASTERISK
route call 2 dest-interface IF_S0_PSTN

service hunt-group HUNTING
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
drop-cause user-busy
route call 1 dest-interface IF_S0_PHONE1
route call 2 dest-interface IF_S0_PHONE2
route call 3 dest-interface IF_S0_PHONE3
route call 4 dest-interface IF_S0_PHONE4

context cs switch
no shutdown

gateway sip GW-ASTERISK
bind interface IF_IP_WAN router

service default
domain 192.168.1.00
realm 192.168.1.100
authentication 210 password 210
default-server 192.168.1.100 loose-router
registrar 192.168.1.100 5060
user 210
session-timer 1800

gateway sip GW-ASTERISK
no shutdown

port ethernet 0 0
medium auto
encapsulation ip
bind interface IF_IP_WAN router
no shutdown

port ethernet 0 1
medium auto
encapsulation ip
bind interface IF_IP_LAN router
no shutdown

port bri 0 0
clock auto
encapsulation q921
power-feed

q921
protocol pp
uni-side auto
encapsulation q931

q931
protocol dss1
uni-side net
encapsulation cc-isdn
bind interface IF_S0_PHONE1 switch

port bri 0 0
no shutdown

port bri 0 1
clock auto
encapsulation q921
power-feed

q921
protocol pp
uni-side auto
encapsulation q931

q931
protocol dss1
uni-side net
encapsulation cc-isdn
bind interface IF_S0_PHONE2 switch

port bri 0 1
no shutdown

port bri 0 2
clock auto
encapsulation q921
power-feed

q921
protocol pp
uni-side auto
encapsulation q931

q931
protocol dss1
uni-side net
encapsulation cc-isdn
bind interface IF_S0_PHONE3 switch

port bri 0 2
no shutdown

port bri 0 3
clock auto
encapsulation q921
power-feed

q921
protocol pp
uni-side auto
encapsulation q931

q931
protocol dss1
uni-side net
encapsulation cc-isdn
bind interface IF_S0_PHONE4 switch

port bri 0 3
no shutdown

port bri 0 4
clock auto
encapsulation q921

q921
protocol pmp
uni-side auto
encapsulation q931

q931
protocol dss1
uni-side user
encapsulation cc-isdn
bind interface IF_S0_PSTN switch

port bri 0 4
no shutdown
Lo username 210 in sip.conf

[210]
username=210
type=friend
secret=210
record_out=Adhoc
record_in=Adhoc
qualify=no
port=5060
nat=never
mailbox=210@device
host=dynamic
dtmfmode=inband
context=from-internal
canreinvite=no
disallow=all
allow=alaw

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Tempo addietro ho fatto un post in cui parlavo dell’uso dei Patton SmartNode 4638 con Trixbox, in particolare una configurazione che prevedeva un 4638 collegato a borchie Telecom, un TRixbox in mezzo con funzioni di IVR ed un altro 4638 a valle collegato al vecchio PBX, con le porte in modalità NT.
Bene, la seconda parte della configurazione era errata, o meglio in parte, perchè è vero che le telefonate potevano transitare, ma solo impegnando la linea e poi alzando la cornetta, facendo subito il numero.
Mi spiego meglio…come voi saprete una periferica isdn, un TA con funzioni voce invia direttamente la stringa del numero impegnando il canale D, andando poi ad occupare il canale B al momento della risposta, mentre un telefono ISDN o un PBX si aspettano un tono libero, continuo o ritmato.
Per questo i toni italiani sono importanti nella configurazione, ma io non avevo tenuto conto di creare le tabelle di routing e i service di hunt group, quindi documentandomi sul sito Patton sono riuscito a fare una configurazione adeguata per questo 4638 a monte del centralino classico, ed eccola qua

#—————————————————————-#
# #
# SN4638/5BIS/UI #
# R3.20 2006-11-17 H323 SIP BRI #
# 2007-07-27T06:39:21 #
# Generated configuration file #
# #
#—————————————————————-#

cli version 3.20
dns-client server 88.149.128.12
webserver port 80 language en
snmp community public ro

system

ic voice 0

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

profile voip ASTERISK
codec 1 g729 rx-length 20 tx-length 20
codec 2 g711ulaw64k rx-length 20 tx-length 20
codec 3 g711alaw64k rx-length 20 tx-length 20

profile pstn default

profile sip default

profile sip asterisk

profile aaa default
method 1 local
method 2 none

context ip router

interface IF_IP_WAN
ipaddress dhcp
no napt-inside

interface IF_IP_LAN
ipaddress 192.168.1.1 255.255.255.0
no napt-inside

context cs switch
digit-collection timeout 4
national-prefix 0
international-prefix 00

routing-table called-e164 RT_2_ISDN
route .%T dest-service HUNTING MT_ITC

routing-table called-e164 RT_ISDN_2_SIP
route 99[1-9].T3 dest-interface IF_S0_PSTN
route 0[1-9].T3 dest-service SER_HUNT_OUT
route 00[1-9].T3 dest-service SER_HUNT_OUT
route default dest-service SER_HUNT_OUT

mapping-table itc to itc MT_ITC
map default to speech

interface isdn IF_S0_PSTN
route call dest-interface IF_S0_PHONE
dtmf-dialing

interface isdn IF_S0_PHONE1
route call dest-table RT_ISDN_2_SIP
dtmf-dialing
use profile tone-set IT

interface isdn IF_S0_PHONE2
route call dest-table RT_ISDN_2_SIP
dtmf-dialing
use profile tone-set IT

interface isdn IF_S0_PHONE3
route call dest-table RT_ISDN_2_SIP
dtmf-dialing
use profile tone-set IT

interface isdn IF_S0_PHONE4
route call dest-table RT_ISDN_2_SIP
dtmf-dialing
use profile tone-set IT

interface sip IF_SIP_ASTERISK
bind gateway GW-ASTERISK
service default
route call dest-table RT_2_ISDN
early-disconnect
remote-party-id called-party
remote-party-id calling-party
address-translation outgoing-call from-header user-part fix 210 host-part call
use profile voip ASTERISK

service hunt-group SER_HUNT_OUT
timeout 6
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_SIP_ASTERISK
route call 2 dest-interface IF_S0_PSTN

service hunt-group HUNTING
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
drop-cause user-busy
route call 1 dest-interface IF_S0_PHONE1
route call 2 dest-interface IF_S0_PHONE2
route call 3 dest-interface IF_S0_PHONE3
route call 4 dest-interface IF_S0_PHONE4

context cs switch
no shutdown

gateway sip GW-ASTERISK
bind interface IF_IP_WAN router

service default
domain 192.168.0.25
realm 192.168.0.25
authentication 210 password t6knzpUB0cY= encrypted
default-server 192.168.0.25 loose-router
registrar 192.168.0.25 5060
user 210
session-timer 1800

gateway sip GW-ASTERISK
no shutdown

port ethernet 0 0
medium auto
encapsulation ip
bind interface IF_IP_WAN router
no shutdown

port ethernet 0 1
medium auto
encapsulation ip
bind interface IF_IP_LAN router
no shutdown

port bri 0 0
clock auto
encapsulation q921
power-feed

q921
protocol pp
uni-side auto
encapsulation q931

q931
protocol dss1
uni-side net
encapsulation cc-isdn
bind interface IF_S0_PHONE1 switch

port bri 0 0
no shutdown

port bri 0 1
clock auto
encapsulation q921
power-feed

q921
protocol pp
uni-side auto
encapsulation q931

q931
protocol dss1
uni-side net
encapsulation cc-isdn
bind interface IF_S0_PHONE2 switch

port bri 0 1
no shutdown

port bri 0 2
clock auto
encapsulation q921
power-feed

q921
protocol pp
uni-side auto
encapsulation q931

q931
protocol dss1
uni-side net
encapsulation cc-isdn
bind interface IF_S0_PHONE3 switch

port bri 0 2
no shutdown

port bri 0 3
clock auto
encapsulation q921
power-feed

q921
protocol pp
uni-side auto
encapsulation q931

q931
protocol dss1
uni-side net
encapsulation cc-isdn
bind interface IF_S0_PHONE4 switch

port bri 0 3
no shutdown

port bri 0 4
clock auto
encapsulation q921

q921
protocol pmp
uni-side auto
encapsulation q931

q931
protocol dss1
uni-side user
encapsulation cc-isdn
bind interface IF_S0_PSTN switch

port bri 0 4
no shutdown

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Data la penuria su Intenet di esempi di configurazione funzionanti, propongo qui la configurazione di uno Smartnode 4552 presente nella sede remota di un mio cliente, quindi capirete il perchè delle regole di routing nel contesto router.
Di seguito la conf

#—————————————————————-#
# #
# SN4552/2BIS/EUI #
# R3.20 2006-11-17 SIP #
# 2007-07-04T18:53:09 #
# Generated configuration file #
# #
#—————————————————————-#

cli version 3.20
dns-relay
webserver port 80 language en
sntp-client
sntp-client server primary 129.132.2.21 port 123 version 4
system hostname Montarioso

system

ic voice 0

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_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 g711ulaw64k rx-length 20 tx-length 20
codec 2 g711alaw64k rx-length 20 tx-length 20
dejitter-max-delay 200

profile pstn default

profile sip default

profile aaa default
method 1 local
method 2 none

context ip router

interface WAN
ipaddress 192.168.3.96 255.255.255.0
use profile napt NAPT_WAN
tcp adjust-mss rx mtu
tcp adjust-mss tx mtu

interface LAN
ipaddress 192.168.1.1 255.255.255.0
tcp adjust-mss rx mtu
tcp adjust-mss tx mtu

context ip router
route 192.168.2.100 255.255.255.255 192.168.3.1 1
route 0.0.0.0 0.0.0.0 192.168.3.1 1

context cs switch
national-prefix 0
international-prefix 00

interface isdn IF-TELCO
route call dest-interface IF_ASTERISK
use profile tone-set IT

interface sip IF_ASTERISK
bind gateway GW_ASTERISK
service default
route call dest-interface IF-TELCO
early-disconnect
remote-party-id called-party
remote-party-id calling-party

context cs switch
no shutdown

gateway sip GW_ASTERISK
bind interface WAN router

service default
domain 192.168.2.100
realm 192.168.2.100
authentication 1004 password 36ocYTYpKxk= encrypted
default-server 192.168.2.100 loose-router
registration-lifetime 500
registrar 192.168.2.100 5060
user 1004
session-timer 1800

gateway sip GW_ASTERISK
no shutdown

port ethernet 0 0
encapsulation ip
bind interface WAN router
no shutdown

port ethernet 0 1
bind interface LAN router
no shutdown

port bri 0 0
clock auto
encapsulation q921

q921
permanent-layer2
protocol pp
uni-side user
encapsulation q931

q931
protocol dss1
uni-side user
encapsulation cc-isdn
bind interface IF-TELCO switch

port bri 0 0
no shutdown

port bri 0 1
clock auto
encapsulation q921

q921
protocol pmp
uni-side auto
encapsulation q931

q931
protocol dss1
uni-side net
encapsulation cc-isdn

port bri 0 1
shutdown

Mentre lato Asterisk solo questo

[1004]
type=friend
username=1004
fromuser=1004
secret=1234
context=from-pstn
qualify=yes
insecure=very
host=192.168.3.96
disallow=all
allow=ulaw
allow=alaw

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