PRIMARIO e Account SIP su 2801

[weblaboratory]

Buongiorno a tutti
ho montato una scheda scheda VWIN2-1MFT-T1/E1 con relativo DSP e configurato tutto quanto...

Codice: Seleziona tutto

card type e1 0 2
network-clock-participate wic 2
network-clock-select 1 E1 0/2/0
isdn switch-type primary-net5
!
voice-card 0
!
voice rtp send-recv
!
voice service voip
 fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback none
 sip
!
voice class codec 1
 codec preference 1 g711alaw
 codec preference 2 g729r8
!
controller E1 0/2/0
 pri-group timeslots 1-16
!
interface Serial0/2/0:15
 no ip address
 ip mroute-cache
 no logging event link-status
 isdn switch-type primary-net5
 isdn overlap-receiving T302 5000
 isdn protocol-emulate network
 isdn incoming-voice voice
 isdn send-alerting
 isdn sending-complete
 no isdn outgoing display-ie
 no fair-queue
 no cdp enable
!
voice-port 0/2/0:15
 echo-cancel coverage 32
 playout-delay nominal 100
 playout-delay mode fixed
 cptone IT
 timeouts initial 30
 timeouts interdigit 3
 music-threshold -70
!
dial-peer voice 1 pots
 description PROVIAMO
 destination-pattern 010XXXXX
 port 0/2/0:15
 authentication username 010XXXXX password XXXXXXX
!
dial-peer voice 10 voip
 destination-pattern .+T
 session protocol sipv2
 session target ipv4:XXXXXXXXXX:5060
 session transport udp
 dtmf-relay sip-notify
 codec g711ulaw
 no vad
!
sip-ua
 nat symmetric role passive
 nat symmetric check-media-src
 retry invite 4
 retry response 3
 retry bye 2
 retry cancel 2
 registrar ipv4:XXXXXXXX:5060 expires 180
 sip-server ipv4:XXXXXXXXXXXXXX:5060
!

solitamento uso una configurazione simile per account su BRI o PSTN

il router naviga correttamente e il relativo account funziona.
ma sembra che non lanci proprio la registrazione SIP.
HO provato anche a mettere l'autenticazione direttamente su SIP-UA ma non cambia nulla

il register status non appare nulla

Codice: Seleziona tutto

PRIMARIO#show sip-ua register status
Line          peer           expires(sec)  registered
============  =============  ============  ===========

Cosa sto sbagliando?
Grazie a tutti

Codice: Seleziona tutto

PRIMARIO#show isdn status
Global ISDN Switchtype = primary-net5
ISDN Serial0/2/0:15 interface
        ******* Network side configuration *******
        dsl 0, interface ISDN Switchtype = primary-net5
    Layer 1 Status:
        DEACTIVATED
    Layer 2 Status:
        TEI = 0, Ces = 1, SAPI = 0, State = TEI_ASSIGNED
    Layer 3 Status:
        0 Active Layer 3 Call(s)
    Active dsl 0 CCBs = 0
    The Free Channel Mask:  0x00000000
    Number of L2 Discards = 0, L2 Session ID = 0

[weblaboratory]

Nessuno sa come aiutarmi per configurare ad-hoc un PRIMARIO su un 2801 con account SIP?

Aiutatemi please

[crisalf76]

Ciao

Io userei questa configurazione :

card type e1 0 2
network-clock-participate wic 2
network-clock-select 1 E1 0/2/0
!
isdn switch-type primary-net5
!
controller E1 0/2/0
clock source internal
pri-group timeslots 1-31
!
!
interface Serial0/2/0:15
shutdown
no ip address
ip mroute-cache
no logging event link-status
isdn switch-type primary-net5
isdn overlap-receiving T302 5000
isdn protocol-emulate network
isdn incoming-voice voice
isdn map address .* plan isdn type national
isdn send-alerting
isdn sending-complete
no fair-queue
no cdp enable
no shutdown
!
!
voice-port 0/2/0:15
echo-cancel coverage 32
playout-delay nominal 100
playout-delay mode fixed
cptone IT
timeouts initial 30
music-threshold -70
!
!
dial-peer voice 1 pots
destination-pattern 010XXXXX
port 0/2/0:15
authentication username 010XXXXX password XXXXXXX
prefix 10XXXXX
!
!
isdn switch-type basic-net3
!
voice-card 0
!
!
voice rtp send-recv
!
voice service voip
fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback none
sip
!
voice class codec 1
codec preference 1 g711alaw
codec preference 2 g729r8
!
dial-peer voice 65 voip
translation-profile outgoing CALLING
tone ringback alert-no-PI
destination-pattern .T
voice-class codec 1
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
incoming called-number .T
fax-relay ecm disable
fax rate 14400 bytes 40
ip qos dscp cs5 media
ip qos dscp cs5 signaling
no vad
!
!
voice translation-profile CALLING
translate calling 1
!
!
!
sip-ua
no remote-party-id
set pstn-cause 18 sip-status 408
set pstn-cause 27 sip-status 502
set pstn-cause 31 sip-status 480
set pstn-cause 58 sip-status 503
set pstn-cause 65 sip-status 488
set pstn-cause 87 sip-status 403
set pstn-cause 88 sip-status 503
set pstn-cause 95 sip-status 500
set pstn-cause 102 sip-status 504
set pstn-cause 111 sip-status 500
registrar ipv4:XXXXXXXX:5060 expires 4400
sip-server ipv4:XXXXXXXXXXXXXX:5060
g729-annexb override
!
!
voice translation-rule 1
rule 1 /^010XXXXX[0-9][0-9]$/ /\0/ type any unknown plan any unknown
rule 15 /^.*/ /010XXXXX/ type any unknown plan any unknown
!

ti ricordo che per ogni canale voce con codec g711 su router devi mettere almeno 2M di pvdm quindi un primario a 15 canali 32 M di pvdm 30 canali 64M.

fammi sapere comeva