Salve,un bel po' di tempo fa via avevo chiesto alcuni pareri sulla soluzione Voice di Cisco e, grazie ad una piacevole offerta fattaci da Cisco stessa (non ricordo il programma commerciale), ora stiamo migrando la nostra telefonia su un Cisco UC520 e una trentina di telefoni 7911. Il problema e' il seguente: quando il CCME riceve una chiamata da un "trunk" SIP, correttamente lo redirige al Cisco Unity Express che fa da Auto Attendant (in seguito vorrei realizzare una soluzione mista CUE/B-ACD con lo scenario 2 variazione 3 del documento 67907 di Cisco) ma il messaggio che viene riprodotto bruscamente si interrompe. Questo comportamento e' casuale, delle volte va fino alla fine, altre volte non dura che pochi secondi.
Uno stralcio di configurazione: Current configuration : 22736 bytes ! ! Last configuration change at 11:53:14 ITA Fri Feb 29 2008 by cisco ! NVRAM config last updated at 11:51:12 ITA Fri Feb 29 2008 by cisco ! version 12.4 parser config cache interface parser config interface no service pad service timestamps debug datetime msec service timestamps log datetime msec no service password-encryption service internal ! hostname telefonista ! boot-start-marker boot system flash uc500-advipservicesk9-mz.124-11.XW6 boot-end-marker ! [...] ! ! stcapp ccm-group 1 stcapp ! stcapp feature access-code ! multilink bundle-name authenticated isdn switch-type basic-net3 ! ! voice call send-alert voice rtp send-recv ! voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip supplementary-service h450.12 no supplementary-service sip refer sip ! ! voice class codec 1 codec preference 1 g729r8 codec preference 2 g711ulaw ! ! voice register global max-dn 160 max-pool 40 ! ! voice translation-rule 1110 rule 14 /^9\(.*\)/ /\1/ rule 15 /^0\(.*\)/ /\1/ ! voice translation-rule 1111 rule 15 /.*/ /08554105/ ! voice translation-rule 1121 rule 15 /.*/ /0854315192/ ! voice translation-rule 1131 rule 15 /.*/ /0872596500/ ! voice translation-rule 2001 rule 10 /.*/ /88/ ! ! voice translation-profile AA_Profile translate called 2001 ! voice translation-profile ISDN1 translate calling 1111 translate called 1110 ! voice translation-profile ISDN2 translate calling 1121 translate called 1110 ! voice translation-profile VOIP translate calling 1131 translate called 1110 ! ! voice-card 0 dspfarm dsp services dspfarm ! ! ! username cisco privilege 15 secret 5 $1$YY/.$NSlk4XFf6x9b2ncJBwMqO. archive log config logging enable logging size 600 hidekeys ! process-max-time 50 ! ip tftp source-interface Loopback0 ! ! ! interface Loopback0 description CCM ip address 10.1.10.2 255.255.255.252 ip nat inside ip virtual-reassembly ! interface FastEthernet0/0 description Wan no ip address ip nat outside ip virtual-reassembly shutdown duplex auto speed auto ! interface Integrated-Service-Engine0/0 description cue is initialized with default IMAP group ip unnumbered Loopback0 ip nat inside ip virtual-reassembly service-module ip address 10.1.10.1 255.255.255.252 service-module ip default-gateway 10.1.10.2 ! [...] ! interface BRI0/1/0 no ip address isdn switch-type basic-net3 isdn point-to-point-setup isdn incoming-voice voice isdn sending-complete ! interface BRI0/1/1 no ip address isdn switch-type basic-net3 isdn point-to-point-setup isdn incoming-voice voice isdn sending-complete ! ! interface Vlan1 description LAN ip address xxx.yyy.xxx.26 255.255.255.192 ip nat outside ip virtual-reassembly ! interface Vlan100 description ePhones ip address 10.254.0.1 255.255.255.0 ip nat inside ip virtual-reassembly ! ip route 0.0.0.0 0.0.0.0 xxx.yyy.xxx.1 ip route 10.1.10.1 255.255.255.255 Integrated-Service-Engine0/0 ! ip http server ip http authentication local ip http secure-server ip http path flash: ip nat inside source list 1 interface FastEthernet0/0 overload ip nat inside source static tcp 10.1.10.1 143 interface Vlan1 143 ip nat inside source static tcp 10.1.10.1 80 interface Vlan1 8080 ! access-list 1 remark SDM_ACL Category=2 access-list 1 permit 10.1.1.0 0.0.0.255 access-list 1 permit 192.168.10.0 0.0.0.255 access-list 1 permit 10.1.10.0 0.0.0.3 snmp-server community public RO ! ! ! ! tftp-server apps11.8-2-2TR2.sbn tftp-server apps31.8-2-2TR2.sbn tftp-server apps41.8-2-2TR2.sbn tftp-server apps70.8-2-2TR2.sbn [...] ! control-plane ! ! ! voice-port 0/0/0 timeouts ringing infinity ! voice-port 0/0/1 timeouts ringing infinity ! voice-port 0/0/2 timeouts ringing infinity ! voice-port 0/0/3 timeouts ringing infinity ! voice-port 0/1/0 translation-profile incoming AA_Profile compand-type a-law connection plar 88 bearer-cap Speech ! voice-port 0/1/1 translation-profile incoming AA_Profile compand-type a-law connection plar 88 bearer-cap Speech ! [...] ! sccp local Loopback0 sccp ccm 10.254.0.1 identifier 1 sccp ! sccp ccm group 1 bind interface Loopback0 associate ccm 1 priority 1 associate profile 2 register DSP-TRANS ! dspfarm profile 2 transcode codec g711ulaw codec g711alaw codec g729r8 maximum sessions 6 associate application SCCP ! ! dial-peer voice 1 pots service stcapp port 0/0/0 ! dial-peer voice 2 pots service stcapp port 0/0/1 ! dial-peer voice 3 pots service stcapp port 0/0/2 ! dial-peer voice 4 pots service stcapp port 0/0/3 ! [...] ! dial-peer voice 2000 voip description ** cue voicemail pilot number ** destination-pattern 77 b2bua session protocol sipv2 session target ipv4:10.1.10.1 dtmf-relay sip-notify rtp-nte codec g711ulaw no vad ! dial-peer voice 2001 voip description ** cue auto attendant number ** destination-pattern 88 b2bua session protocol sipv2 session target ipv4:10.1.10.1 dtmf-relay sip-notify rtp-nte codec g711ulaw no vad ! dial-peer voice 50 pots description ** incoming dial peer ** incoming called-number .% direct-inward-dial port 0/1/0 ! dial-peer voice 51 pots description ** incoming dial peer ** incoming called-number .% direct-inward-dial port 0/1/1 ! dial-peer voice 52 pots description ** BRI pots dial-peer ** translation-profile outgoing ISDN1 preference 5 destination-pattern 9T port 0/1/0 no sip-register ! dial-peer voice 55 pots description ** BRI pots dial-peer ** translation-profile outgoing ISDN2 preference 5 destination-pattern 9T port 0/1/1 no sip-register ! dial-peer voice 100 voip description VOIP translation-profile outgoing VOIP preference 3 destination-pattern 0T voice-class codec 1 session protocol sipv2 session target sip-server dtmf-relay rtp-nte ! dial-peer voice 101 voip translation-profile incoming AA_Profile b2bua session protocol sipv2 session target ipv4:10.1.10.1 incoming called-number 0872596500 dtmf-relay rtp-nte codec g711ulaw no vad ! ! no dial-peer outbound status-check pots sip-ua authentication username TTTTTYYYYY password 7 XXXXYYYYY no remote-party-id registrar dns:voip.xxxxx.it expires 3600 sip-server dns:voip.xxxxx.it host-registrar ! ! ! telephony-service sdspfarm units 1 sdspfarm transcode sessions 3 sdspfarm tag 1 DSP-TRANS load 7960-7940 P00308000500 load 7914 S00105000200 load 7902 CP7902080002SCCP060817A load 7921 CP7921G-1.0.4 load 7931 SCCP31.8-2-2SR2S load 7941GE SCCP41.8-2-2SR2S load 7941 SCCP41.8-2-2SR2S load 7961GE SCCP41.8-2-2SR2S load 7961 SCCP41.8-2-2SR2S load 7975 SCCP75.8-3-2S load 7965 SCCP45.8-3-2S load 7945 SCCP45.8-3-2S load 7942 SCCP42.8-3-2S load 7962 SCCP42.8-3-2S load 7971 SCCP70.8-2-2SR2S load 7970 SCCP70.8-2-2SR2S load 7936 cmterm_7936.3-3-13-0 load 7906 SCCP11.8-2-2SR2S load 7911 SCCP11.8-2-2SR2S max-ephones 40 max-dn 160 ip source-address 10.254.0.1 port 2000 max-redirect 20 auto assign 5 to 8 type anl calling-number initiator service dnis overlay service dnis dir-lookup timeouts interdigit 5 system message Benvenuto in Micso! url services http://10.1.10.1/voiceview/common/login.dourl authentication http://10.1.10.1/voiceview/authentication/authenticate.do
cnf-file location flash: user-locale IT network-locale IT time-zone 23 time-format 24 date-format dd-mm-yy voicemail 77 max-conferences 8 gain -6 call-forward pattern .T call-forward system redirecting-expanded moh music-on-hold.au multicast moh 239.10.16.16 port 2000 web admin system name cisco secret 5 zzzzzzzzzzzzzzzz dn-webedit time-webedit transfer-system full-consult dss transfer-pattern 0.T transfer-pattern .T secondary-dialtone 0 create cnf-files version-stamp 7960 Feb 28 2008 19:10:36 ! ! ephone-dn 100 dual-line number 0872596500 ! ! ephone-dn 159 number A801.. no-reg primary mwi off ! ! ephone-dn 160 number A800.. no-reg primary mwi on ! ! [...] ! ! ! ntp logging ntp clock-period 17180288 ntp master ntp update-calendar ntp server 193.204.114.233 ! webvpn cef end La configurazione di CUE e' piu' breve e credo ci sia poco da vedere: Generating configuration: clock timezone Europe/Rome hostname telefonista-cue ip domain-name micso.it ip name-server 195.32.69.20 195.32.69.14 ntp server 10.1.10.2 prefersoftware download server url "ftp://127.0.0.1/ftp"; credentials hidden "6u/dKTN/hsEuSAEfw40XlF2eFHnZfyUTSd8ZZNgd+Y9J3xlk2B35j0nfGWTYHfmPSd8ZZNgd+Y9J3xlk2B35j0nfGWTYHfmPSd8ZZNgd+Y9J3xlk2B35j0nfGWTYHfmP"
groupname Administrators create groupname Broadcasters create groupname IMAPgrp create username cisco create groupname Administrators member cisco groupname Administrators privilege ManagePrompts groupname Administrators privilege broadcast groupname Administrators privilege local-broadcast groupname Administrators privilege ManagePublicList groupname Administrators privilege ViewPrivateList groupname Administrators privilege vm-imap groupname Administrators privilege superuser groupname Broadcasters privilege broadcast groupname IMAPgrp privilege vm-imap restriction msg-notification min-digits 1 restriction msg-notification max-digits 30 restriction msg-notification dial-string preference 1 pattern * allowed smtp server address relay.micso.it authentication nonebackup server url "ftp://127.0.0.1/ftp"; credentials hidden "EWlTygcMhYmjazXhE/VNXHCkplVV4KjescbDaLa4fl4WLSPFvv1rWUnfGWTYHfmPSd8ZZNgd+Y9J3xlk2B35j0nfGWTYHfmPSd8ZZNgd+Y9J3xlk2B35j0nfGWTYHfmP"
calendar biz-schedule systemschedule closed day 1 from 00:00 to 24:00 open day 2 from 09:00 to 13:30 open day 2 from 15:00 to 18:30 open day 3 from 09:00 to 13:30 open day 3 from 15:00 to 18:30 open day 4 from 09:00 to 13:30 open day 4 from 15:00 to 18:30 open day 5 from 09:00 to 13:30 open day 5 from 15:00 to 18:30 open day 6 from 09:00 to 13:30 open day 6 from 15:00 to 18:30 closed day 7 from 00:00 to 24:00 end schedule ccn application autoattendant description "autoattendant" enabled maxsessions 6 script "aa.aef" parameter "busClosedPrompt" "AABusinessClosed.wav" parameter "holidayPrompt" "AAHolidayPrompt.wav" parameter "welcomePrompt" "AAWelcome.wav" parameter "disconnectAfterMenu" "false" parameter "allowExternalTransfers" "false" parameter "MaxRetry" "3" parameter "busOpenPrompt" "AABusinessOpen.wav" parameter "businessSchedule" "systemschedule" parameter "operExtn" "73" end application ccn application ciscomwiapplication description "ciscomwiapplication" enabled maxsessions 6 script "setmwi.aef" parameter "CallControlGroupID" "0" parameter "strMWI_OFF_DN" "A801" parameter "strMWI_ON_DN" "A800" end application ccn application msgnotification description "msgnotification" enabled maxsessions 6 script "msgnotify.aef"parameter "logoutUri" "http://localhost/voicemail/vxmlscripts/mbxLogout.jsp";
parameter "DelayBeforeSendDTMF" "1" end application ccn application promptmgmt description "promptmgmt" enabled maxsessions 1 script "promptmgmt.aef" end application ccn application voicemail description "voicemail" enabled maxsessions 6 script "voicebrowser.aef"parameter "logoutUri" "http://localhost/voicemail/vxmlscripts/mbxLogout.jsp";
parameter "uri" "http://localhost/voicemail/vxmlscripts/login.vxml"; end application ccn engine end engine ccn subsystem jtapi ccm-manager address 0.0.0.0 end subsystem ccn subsystem sip gateway address "10.1.10.2" dtmf-relay sip-notify rtp-nte end subsystem ccn trigger sip phonenumber 77 application "voicemail" enabled maxsessions 6 end trigger ccn trigger sip phonenumber 88 application "autoattendant" enabled maxsessions 6 end trigger service imap enable end imap service phone-authentication end phone-authentication service voiceview enable end voiceview voicemail callerid voicemail default mailboxsize 775 voicemail broadcast recording time 300 endIl tutto funziona perfettamente dai telefoni interni e dall'ISDN (anche se si mangia tipo mezzo secondo alla risposta, insomma la vocina registrata parte mezzo secondo dopo). Domanda 1: ma la DSP farm serve? Era disabilitato di default. Vedo che le conference call si possono fare lo stesso... Mi indicate il documento che spiegava quante risorse DSP servivano per ogni tipo di interfaccia? Non lo trovo piu'! La bestiolina ha un PVDM2-64. Domanda 2: L'ISDN e' configurata "giusta" per l'Italia? Sembra fungere ma non vorrei sorprese e soprattutto c'e' un modo per ritardare la risposta? Non fa nemmeno uno squillo che gia' risponde. Domandone 3: perche' la vocina si blocca??? Potete provare a fare lo 0872596500 lo tengo ancora per qualche giorno su questo numero. Vi si ferma? Ho gia' provato a cambiare i codec nella voice class e a bloccarli sul dial-peer incoming. Ma alle volte va alle volte no. Se digitate in modo da volere l'interno 29 (io) poi la comunicazione e' assolutamente perfetta.
Grazie, buon weekend e saluti a tutti, Normando -- | Normando Marcolongo | Micso s.r.l. | | via Tiburtina, 318 | I-65128 Pescara, Italy| |tel/fax (+39)08554105 | mob. (+39)3386296362 |
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