1751V account SIP su BRI

Voice su IP

Moderatore: Federico.Lagni

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weblaboratory
Cisco fan
Messaggi: 71
Iscritto il: gio 17 dic , 2009 5:27 pm

Buongiorno a tutti. Oggi mi è capitato di dover configurare un 1751V come gateway SIP-BRI dove di solito metto dei PATTON o simili.
Sono riuscito a configurare gli account SIP e le chiamate in ingresso funzionano. Purtroppo non vanno quelle in uscita.
Dietro il Cisco c'è un centralino ISDN obsoleto e non posso farci molto :-(

Sembra che non riesca a capire la rotta per indirizzare la chiamata verso il SIP SERVER.

Potete darmi una mano?
Vi posto la mia attuale configurazione

Codice: Seleziona tutto

isdn switch-type basic-net3
!
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 2 g711ulaw
 codec preference 3 g729r8
!
voice translation-rule 1
 rule 1 /^014300001$/ /\0/ type any unknown plan any unknown
 rule 2 /^0143000002$/ /\0/ type any unknown plan any unknown
 rule 15 /^.*/ /0143000003/ type any unknown plan any unknown
!
voice translation-profile CALLING
 translate calling 1
!
interface BRI1/0
 no ip address
 no logging event link-status
 isdn switch-type basic-net3
 isdn overlap-receiving T302 5000
 isdn not-end-to-end 64
 isdn protocol-emulate network
 isdn layer1-emulate network
 isdn incoming-voice voice
 isdn map address .* plan isdn type national
 isdn send-alerting
 isdn sending-complete
 isdn skipsend-idverify
 line-power
!
interface BRI1/1
 no ip address
 no logging event link-status
 isdn switch-type basic-net3
 isdn overlap-receiving T302 5000
 isdn not-end-to-end 64
 isdn protocol-emulate network
 isdn layer1-emulate network
 isdn incoming-voice voice
 isdn map address .* plan isdn type national
 isdn send-alerting
 isdn sending-complete
 isdn static-tei 0
 isdn skipsend-idverify
 line-power
!
voice-port 1/0
 echo-cancel coverage 32
 compand-type a-law
 playout-delay nominal 100
 playout-delay mode fixed
 cptone IT
 timeouts initial 30
 timeouts interdigit 3
 music-threshold -70
!
voice-port 1/1
 echo-cancel coverage 32
 compand-type a-law
 playout-delay nominal 100
 playout-delay mode fixed
 cptone IT
 timeouts initial 30
 timeouts interdigit 3
 music-threshold -70
!
dial-peer voice 66 pots
 incoming called-number .T
 direct-inward-dial
!
dial-peer voice 1 pots
 description P VOCE 0143000003
 destination-pattern 0143000003
 port 1/0
 prefix 143000003
 authentication username 0143000003 password XXXXXXXX
!
dial-peer voice 3 pots
 description P VOCE SEC FAX 0143000002
 destination-pattern 0143000002
 port 1/1
 prefix 143000002
 authentication username 0143000002 password XXXXXXX
!
dial-peer voice 2 pots
 description P PRIM VOCE 014300001
 destination-pattern 014300001
 port 1/1
 prefix 14300001
 authentication username 014300001 password XXXXXXX
!
dial-peer voice 65 voip
 tone ringback alert-no-PI
 translation-profile outgoing CALLING
 max-conn 4
 destination-pattern .T
 voice-class codec 1
 session protocol sipv2
 session target ipv4:_____IPSERVER____:5060
 incoming called-number .T
 dtmf-relay rtp-nte
 fax-relay ecm disable
 fax rate 14400 bytes 40
 ip qos dscp cs5 media
 ip qos dscp cs5 signaling
 no vad
!
sip-ua 
 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:_____IPSERVER____:5060 expires 180
 sip-server ipv4:_____IPSERVER____:5060
 g729-annexb override
!

Gli account sono registrati

Codice: Seleziona tutto

Line          peer           expires(sec)  registered 
============  =============  ============  ===========
014300002    2                141           yes         
014300001    1                37            yes         
014300003    3                37            yes    
Se la chiamata la eseguo mettendo il mio numero e cliccando CALL allora viene fuori questo log:

Codice: Seleziona tutto

ROUTER(config-dial-peer)#
Mar  6 15:20:33.306: ISDN BR1/1 Q931: RX <- SETUP pd = 8  callref = 0x01
        Bearer Capability i = 0x8090A3
                Standard = CCITT
                Transfer Capability = Speech
                Transfer Mode = Circuit
                Transfer Rate = 64 kbit/s
        Channel ID i = 0x83
                Preferred, Any B-channel
        Calling Party Number i = 0x0980, '201'
                Plan:Private, Type:Unknown
        Called Party Number i = 0x80, '_____MIONUMERODITELEFONO_________'
                Plan:Unknown, Type:Unknown
        High Layer Compat i = 0x9181
Mar  6 15:20:33.330: ISDN BR1/1 Q931: TX -> SETUP_ACK pd = 8  callref = 0x81
        Channel ID i = 0x8A
                Exclusive, B2
Mar  6 15:20:38.378: ISDN BR1/1 Q931: TX -> CALL_PROC pd = 8  callref = 0x81
Mar  6 15:20:38.518: ISDN BR1/1 Q931: TX -> DISCONNECT pd = 8  callref = 0x81
        Cause i = 0x82B9 - Bearer capability not presently authorized
Mar  6 15:20:38.542: ISDN BR1/1 Q931: RX <- RELEASE pd = 8  callref = 0x01
Mar  6 15:20:38.554: ISDN BR1/1 Q931: TX -> RELEASE_COMP pd = 8  callref = 0x81
ROUTER#
Se invece alzo la cornetta e digito ogni numero singolarmente:

Codice: Seleziona tutto

Mar  6 15:17:15.841: ISDN BR1/1 Q931: RX <- INFORMATION pd = 8  callref = 0x01
        Called Party Number i = 0x80, '3'
                Plan:Unknown, Type:Unknown
Mar  6 15:17:16.301: ISDN BR1/1 Q931: RX <- INFORMATION pd = 8  callref = 0x01
        Called Party Number i = 0x80, '8'
                Plan:Unknown, Type:Unknown
Mar  6 15:17:16.661: ISDN BR1/1 Q931: RX <- INFORMATION pd = 8  callref = 0x01
        Called Party Number i = 0x80, '7'
                Plan:Unknown, Type:Unknown
Mar  6 15:17:16.981: ISDN BR1/1 Q931: RX <- INFORMATION pd = 8  callref = 0x01
        Called Party Number i = 0x80, '5'
                Plan:Unknown, Type:Unknown
Mar  6 15:17:17.361: ISDN BR1/1 Q931: RX <- INFORMATION pd = 8  callref = 0x01
        Called Party Number i = 0x80, '5'
                Plan:Unknown, Type:Unknown
Mar  6 15:17:17.801: ISDN BR1/1 Q931: RX <- INFORMATION pd = 8  callref = 0x01
        Called Party Number i = 0x80, 'XXXX'
                Plan:Unknown, Type:Unknown
Mar  6 15:17:18.061: ISDN BR1/1 Q931: RX <- INFORMATION pd = 8  callref = 0x01
        Called Party Number i = 0x80, 'XXXX'
                Plan:Unknown, Type:Unknown
Mar  6 15:17:18.301: ISDN BR1/1 Q931: RX <- INFORMATION pd = 8  callref = 0x01
        Called Party Number i = 0x80, 'XXXX'
                Plan:Unknown, Type:Unknown
Mar  6 15:17:18.581: ISDN BR1/1 Q931: RX <- INFORMATION pd = 8  callref = 0x01
        Called Party Number i = 0x80, 'XXXXX'
                Plan:Unknown, Type:Unknown
Mar  6 15:17:23.630: ISDN BR1/1 Q931: TX -> CALL_PROC pd = 8  callref = 0x81
Mar  6 15:17:26.530: ISDN BR1/1 Q931: RX <- DISCONNECT pd = 8  callref = 0x01
        Cause i = 0x8090 - Normal call clearing
Mar  6 15:17:26.542: ISDN BR1/1 Q931: TX -> RELEASE pd = 8  callref = 0x81
Dove le XXXXX sono i singoli numeri digitati....

Forse sto sbagliando qualcosa sulle translation-rule ma non so bene come utilizzarla

Grazie per l'aiuto!!!! :-)
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crisalf76
Cisco fan
Messaggi: 52
Iscritto il: dom 25 ott , 2009 11:21 pm
Località: Napoli
Contatta:

Ciao

Prova a modificare la conf come segue :

!
voice service voip
fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback none
sip
bind control source-interface <WAN o Interfaccia Voce>
bind media source-interface <WAN o Interfaccia Voce>
!
!
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
max-conn 4
no vad
!

fammi sapere se poi funziona o se il problema permane con ulteriore debug.
quello ccsip
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