Salut !
J'ai un petit soucis et peut-être pouvez-vous m'aider.
Je dois migrer une voice GW H323 en Roumanie et hier j'effectuais des tests opérateur pour vérifier que la ligne a bien été installée et qu'elle fonctionne bien. J'ai pu vérifier que les appels sortant fonctionnent correctement autant en national qu'en international mais par contre impossible de faire marcher les appels entrant.
 
L'opérateur, lorsqu'on fait un test d'appel, envoie bien 7 digits que je transforme en N° routable en interne (4405001) et j'ai ensuite configuré des dial-peer voip (3 pour être précis) qui matchent avec la destination-pattern 440.... (pour ensuite router vers le CUCM).
 
Lorsque je fais un test de la translation pattern, le numéro est bien transformé:
 
RCB-VoiceGW-01#Test voice translation-rule 4010 2080464
Matched with rule 1
Original number: 2080464        Translated number: 4405001
Original number type: none      Translated number type: none
Original number plan: none      Translated number plan: none
 
Lorsque je teste les matchs avec les dia-peer configurés avec la commande show dialplan number 4405001, ça a l'air d'être bon aussi :
 
RCB-VoiceGW-01#show dialplan number 4405001
Macro Exp.: 4405001
 
VoiceOverIpPeer101
 peer type = voice, system default peer = FALSE, information type = voice,
 description = `--- Appels H323 vers CUCM1 - PUBLISHER ---',
 tag = 101, destination-pattern = `440....',
 voice reg type = 0, corresponding tag = 0,
 allow watch = FALSE
 answer-address = `', preference=0,
 CLID Restriction = None
 CLID Network Number = `'
 CLID Second Number sent  
 CLID Override RDNIS = disabled,
 rtp-ssrc mux = system
 source carrier-id = `', target carrier-id = `',
 source trunk-group-label = `', target trunk-group-label = `',
 numbering Type = `unknown'
 group = 101, Admin state is up, Operation state is up,
 incoming called-number = `', connections/maximum = 0/unlimited,
 DTMF Relay = enabled,
 modem transport = system,
 URI classes:
     Incoming (Called) =  
     Incoming (Calling) =  
     Destination =  
 huntstop = disabled,
 in bound application associated: 'DEFAULT'
 out bound application associated: ''
 dnis-map =  
 permission :both
 incoming COR list:maximum capability
 outgoing COR list:minimum requirement
 Translation profile (Incoming):
 Translation profile (Outgoing):
 incoming call blocking:
 translation-profile = `'
 disconnect-cause = `no-service'
 advertise 0x40 capacity_update_timer 25 addrFamily 4 oldAddrFamily 4
 mailbox selection policy: none
 type = voip, session-target = `ipv4:10.24.28.11',
 technology prefix:  
 settle-call = disabled
 ip media DSCP = ef, ip media rsvp-pass DSCP = ef
 ip media rsvp-fail DSCP = ef, ip signaling DSCP = af31,
 ip video rsvp-none DSCP = af41,ip video rsvp-pass DSCP = af41
 ip video rsvp-fail DSCP = af41,
 ip defending Priority = 0, ip preemption priority = 0
 ip policy locator voice:  
 ip policy locator video:  
 UDP checksum = disabled,
 session-protocol = cisco, session-transport = system,
 req-qos = best-effort, acc-qos = best-effort,
 req-qos video = best-effort, acc-qos video = best-effort,
 req-qos audio def bandwidth = 64, req-qos audio max bandwidth = 0,
 req-qos video def bandwidth = 384, req-qos video max bandwidth = 0,  
 dtmf-relay = h245-signal,  
 dtmf-relay = h245-alphanumeric,  
 RTP dynamic payload type values: NTE = 101
 Cisco: NSE=100, fax=96, fax-ack=97, dtmf=121, fax-relay=122
        CAS=123, TTY=119, ClearChan=125, PCM switch over u-law=0,
        A-law=8, GSMAMR-NB=117 iLBC=116, AAC-ld=114, lmr_tone=0, nte_tone=0
        h263+=118, h264=119
        G726r16 using static payload
        G726r24 using static payload
 RTP comfort noise payload type = 19
 fax rate = voice,   payload size =  20 bytes
 fax protocol = t38
 fax-relay ecm enable
 Fax Relay ans enabled
 Fax Relay SG3-to-G3 Enabled (by system configuration)
 fax NSF = 0xAD0051 (default)
 voice-class codec = 1
 codec = g729r8,   payload size =  20 bytes,
 video codec = None
 voice class codec = 1
 voice class sip rsvp-fail-policy voice post-alert mandatory keep-alive interval 30
 voice class sip rsvp-fail-policy voice post-alert optional keep-alive interval 30
 voice class sip rsvp-fail-policy video post-alert mandatory keep-alive interval 30
 voice class sip rsvp-fail-policy video post-alert optional keep-alive interval 30
 text relay = disabled
 Media Setting = forking (disabled) flow-through (global)
 Expect factor = 10, Icpif = 20,
 Playout Mode is set to adaptive,
 Initial 60 ms, Max 1000 ms
 Playout-delay Minimum mode is set to default, value 40 ms  
 Fax nominal 300 ms
 Max Redirects = 1, signaling-type = cas,
 VAD = disabled, Poor QOV Trap = disabled,  
 Source Interface = NONE
 voice class sip url = system,
 voice class sip tel-config = system,
 voice class sip rel1xx = system,
 tvoice class sip outbound-proxy = system,
 voice class sip asserted-id = system,
 voice class sip privacy = system,
 voice class sip e911 = system,
 voice class sip history-info = system,
 voice class sip pass-thru headers = system,
 voice class sip pass-thru content unsupp = system,
 voice class sip pass-thru content sdp = system,
 voice class sip anat = system,
 voice class sip g729 annexb-all = system,
 voice class sip early-offer forced = system,
 voice class sip negotiate cisco = system,
 voice class sip block 180 = system,
 voice class sip block 183 = system,
 voice class sip preloaded-route = system,
 voice class sip random-contact = system,
 voice class sip random-request-uri validate = system,
 voice class sip call-route p-called-party-id = system,
 voice class sip privacy-policy send-always = system,
 voice class sip privacy-policy passthru = system,
 voice class sip bandwidth audio  = system,
 voice class sip bandwidth video  = system,
 voice class sip authenticate redirecting-number = system,
 redirect ip2ip = disabled
 local peer = false
 probe disabled,
 Secure RTP: system (use the global setting)
 voice class perm tag = `'
 Time elapsed since last clearing of voice call statistics never
 Connect Time = 0, Charged Units = 0,
 Successful Calls = 0, Failed Calls = 0, Incomplete Calls = 0
 Accepted Calls = 0, Refused Calls = 3,
 Last Disconnect Cause is "1   ",
 Last Disconnect Text is "unassigned number (1)",
 Last Setup Time = 824248.
 Last Disconnect Time = 0.
Matched: 4405001   Digits: 3
Target: ipv4:10.24.28.11
 
VoiceOverIpPeer102
 peer type = voice, system default peer = FALSE, information type = voice,
 description = `--- Appels H323 vers CUCM2 - SUBSCRIBER 1 ---',
 tag = 102, destination-pattern = `440....',
 voice reg type = 0, corresponding tag = 0,
 allow watch = FALSE
 answer-address = `', preference=1,
 CLID Restriction = None
 CLID Network Number = `'
 CLID Second Number sent  
 CLID Override RDNIS = disabled,
 rtp-ssrc mux = system
 source carrier-id = `', target carrier-id = `',
 source trunk-group-label = `', target trunk-group-label = `',
 numbering Type = `unknown'
 group = 102, Admin state is up, Operation state is up,
 incoming called-number = `', connections/maximum = 0/unlimited,
 DTMF Relay = enabled,
 modem transport = system,
 URI classes:
     Incoming (Called) =  
     Incoming (Calling) =  
     Destination =  
 huntstop = disabled,
 in bound application associated: 'DEFAULT'
 out bound application associated: ''
 dnis-map =  
 permission :both
 incoming COR list:maximum capability
 outgoing COR list:minimum requirement
 Translation profile (Incoming):
 Translation profile (Outgoing):
 incoming call blocking:
 translation-profile = `'
 disconnect-cause = `no-service'
 advertise 0x40 capacity_update_timer 25 addrFamily 4 oldAddrFamily 4
 mailbox selection policy: none
 type = voip, session-target = `ipv4:10.24.28.12',
 technology prefix:  
 settle-call = disabled
 ip media DSCP = ef, ip media rsvp-pass DSCP = ef
 ip media rsvp-fail DSCP = ef, ip signaling DSCP = af31,
 ip video rsvp-none DSCP = af41,ip video rsvp-pass DSCP = af41
 ip video rsvp-fail DSCP = af41,
 ip defending Priority = 0, ip preemption priority = 0
 ip policy locator voice:  
 ip policy locator video:  
 UDP checksum = disabled,
 session-protocol = cisco, session-transport = system,
 req-qos = best-effort, acc-qos = best-effort,
 req-qos video = best-effort, acc-qos video = best-effort,
 req-qos audio def bandwidth = 64, req-qos audio max bandwidth = 0,
 req-qos video def bandwidth = 384, req-qos video max bandwidth = 0,  
 dtmf-relay = h245-signal,  
 dtmf-relay = h245-alphanumeric,  
 RTP dynamic payload type values: NTE = 101
 Cisco: NSE=100, fax=96, fax-ack=97, dtmf=121, fax-relay=122
        CAS=123, TTY=119, ClearChan=125, PCM switch over u-law=0,
        A-law=8, GSMAMR-NB=117 iLBC=116, AAC-ld=114, lmr_tone=0, nte_tone=0
        h263+=118, h264=119
        G726r16 using static payload
        G726r24 using static payload
 RTP comfort noise payload type = 19
 fax rate = voice,   payload size =  20 bytes
 fax protocol = t38
 fax-relay ecm enable
 Fax Relay ans enabled
 Fax Relay SG3-to-G3 Enabled (by system configuration)
 fax NSF = 0xAD0051 (default)
 voice-class codec = 1
 codec = g729r8,   payload size =  20 bytes,
 video codec = None
 voice class codec = 1
 voice class sip rsvp-fail-policy voice post-alert mandatory keep-alive interval 30
 voice class sip rsvp-fail-policy voice post-alert optional keep-alive interval 30
 voice class sip rsvp-fail-policy video post-alert mandatory keep-alive interval 30
 voice class sip rsvp-fail-policy video post-alert optional keep-alive interval 30
 text relay = disabled
 Media Setting = forking (disabled) flow-through (global)
 Expect factor = 10, Icpif = 20,
 Playout Mode is set to adaptive,
 Initial 60 ms, Max 1000 ms
 Playout-delay Minimum mode is set to default, value 40 ms  
 Fax nominal 300 ms
 Max Redirects = 1, signaling-type = cas,
 VAD = disabled, Poor QOV Trap = disabled,  
 Source Interface = NONE
 voice class sip url = system,
 voice class sip tel-config = system,
 voice class sip rel1xx = system,
 tvoice class sip outbound-proxy = system,
 voice class sip asserted-id = system,
 voice class sip privacy = system,
 voice class sip e911 = system,
 voice class sip history-info = system,
 voice class sip pass-thru headers = system,
 voice class sip pass-thru content unsupp = system,
 voice class sip pass-thru content sdp = system,
 voice class sip anat = system,
 voice class sip g729 annexb-all = system,
 voice class sip early-offer forced = system,
 voice class sip negotiate cisco = system,
 voice class sip block 180 = system,
 voice class sip block 183 = system,
 voice class sip preloaded-route = system,
 voice class sip random-contact = system,
 voice class sip random-request-uri validate = system,
 voice class sip call-route p-called-party-id = system,
 voice class sip privacy-policy send-always = system,
 voice class sip privacy-policy passthru = system,
 voice class sip bandwidth audio  = system,
 voice class sip bandwidth video  = system,
 voice class sip authenticate redirecting-number = system,
 redirect ip2ip = disabled
 local peer = false
 probe disabled,
 Secure RTP: system (use the global setting)
 voice class perm tag = `'
 Time elapsed since last clearing of voice call statistics never
 Connect Time = 0, Charged Units = 0,
 Successful Calls = 0, Failed Calls = 0, Incomplete Calls = 0
 Accepted Calls = 0, Refused Calls = 0,
 Last Disconnect Cause is "",
 Last Disconnect Text is "",
 Last Setup Time = 0.
 Last Disconnect Time = 0.
Matched: 4405001   Digits: 3
Target: ipv4:10.24.28.12
 
VoiceOverIpPeer103
 peer type = voice, system default peer = FALSE, information type = voice,
 description = `--- Appels H323 vers CUCM3 - SUBSCRIBER 2 ---',
 tag = 103, destination-pattern = `440....',
 voice reg type = 0, corresponding tag = 0,
 allow watch = FALSE
 answer-address = `', preference=2,
 CLID Restriction = None
 CLID Network Number = `'
 CLID Second Number sent  
 CLID Override RDNIS = disabled,
 rtp-ssrc mux = system
 source carrier-id = `', target carrier-id = `',
 source trunk-group-label = `', target trunk-group-label = `',
 numbering Type = `unknown'
 group = 103, Admin state is up, Operation state is up,
 incoming called-number = `', connections/maximum = 0/unlimited,
 DTMF Relay = enabled,
 modem transport = system,
 URI classes:
     Incoming (Called) =  
     Incoming (Calling) =  
     Destination =  
 huntstop = disabled,
 in bound application associated: 'DEFAULT'
 out bound application associated: ''
 dnis-map =  
 permission :both
 incoming COR list:maximum capability
 outgoing COR list:minimum requirement
 Translation profile (Incoming):
 Translation profile (Outgoing):
 incoming call blocking:
 translation-profile = `'
 disconnect-cause = `no-service'
 advertise 0x40 capacity_update_timer 25 addrFamily 4 oldAddrFamily 4
 mailbox selection policy: none
 type = voip, session-target = `ipv4:10.24.28.14',
 technology prefix:  
 settle-call = disabled
 ip media DSCP = ef, ip media rsvp-pass DSCP = ef
 ip media rsvp-fail DSCP = ef, ip signaling DSCP = af31,
 ip video rsvp-none DSCP = af41,ip video rsvp-pass DSCP = af41
 ip video rsvp-fail DSCP = af41,
 ip defending Priority = 0, ip preemption priority = 0
 ip policy locator voice:  
 ip policy locator video:  
 UDP checksum = disabled,
 session-protocol = cisco, session-transport = system,
 req-qos = best-effort, acc-qos = best-effort,
 req-qos video = best-effort, acc-qos video = best-effort,
 req-qos audio def bandwidth = 64, req-qos audio max bandwidth = 0,
 req-qos video def bandwidth = 384, req-qos video max bandwidth = 0,  
 dtmf-relay = h245-signal,  
 dtmf-relay = h245-alphanumeric,  
 RTP dynamic payload type values: NTE = 101
 Cisco: NSE=100, fax=96, fax-ack=97, dtmf=121, fax-relay=122
        CAS=123, TTY=119, ClearChan=125, PCM switch over u-law=0,
        A-law=8, GSMAMR-NB=117 iLBC=116, AAC-ld=114, lmr_tone=0, nte_tone=0
        h263+=118, h264=119
        G726r16 using static payload
        G726r24 using static payload
 RTP comfort noise payload type = 19
 fax rate = voice,   payload size =  20 bytes
 fax protocol = t38
 fax-relay ecm enable
 Fax Relay ans enabled
 Fax Relay SG3-to-G3 Enabled (by system configuration)
 fax NSF = 0xAD0051 (default)
 voice-class codec = 1
 codec = g729r8,   payload size =  20 bytes,
 video codec = None
 voice class codec = 1
 voice class sip rsvp-fail-policy voice post-alert mandatory keep-alive interval 30
 voice class sip rsvp-fail-policy voice post-alert optional keep-alive interval 30
 voice class sip rsvp-fail-policy video post-alert mandatory keep-alive interval 30
 voice class sip rsvp-fail-policy video post-alert optional keep-alive interval 30
 text relay = disabled
 Media Setting = forking (disabled) flow-through (global)
 Expect factor = 10, Icpif = 20,
 Playout Mode is set to adaptive,
 Initial 60 ms, Max 1000 ms
 Playout-delay Minimum mode is set to default, value 40 ms  
 Fax nominal 300 ms
 Max Redirects = 1, signaling-type = cas,
 VAD = disabled, Poor QOV Trap = disabled,  
 Source Interface = NONE
 voice class sip url = system,
 voice class sip tel-config = system,
 voice class sip rel1xx = system,
 tvoice class sip outbound-proxy = system,
 voice class sip asserted-id = system,
 voice class sip privacy = system,
 voice class sip e911 = system,
 voice class sip history-info = system,
 voice class sip pass-thru headers = system,
 voice class sip pass-thru content unsupp = system,
 voice class sip pass-thru content sdp = system,
 voice class sip anat = system,
 voice class sip g729 annexb-all = system,
 voice class sip early-offer forced = system,
 voice class sip negotiate cisco = system,
 voice class sip block 180 = system,
 voice class sip block 183 = system,
 voice class sip preloaded-route = system,
 voice class sip random-contact = system,
 voice class sip random-request-uri validate = system,
 voice class sip call-route p-called-party-id = system,
 voice class sip privacy-policy send-always = system,
 voice class sip privacy-policy passthru = system,
 voice class sip bandwidth audio  = system,
 voice class sip bandwidth video  = system,
 voice class sip authenticate redirecting-number = system,
 redirect ip2ip = disabled
 local peer = false
 probe disabled,
 Secure RTP: system (use the global setting)
 voice class perm tag = `'
 Time elapsed since last clearing of voice call statistics never
 Connect Time = 0, Charged Units = 0,
 Successful Calls = 0, Failed Calls = 0, Incomplete Calls = 0
 Accepted Calls = 0, Refused Calls = 0,
 Last Disconnect Cause is "",
 Last Disconnect Text is "",
 Last Setup Time = 0.
 Last Disconnect Time = 0.
Matched: 4405001   Digits: 3
Target: ipv4:10.24.28.14
 
 
 
Lorsque je fais un debug voip ccapi, j'ai des traces lorsque je passe l'appel entrant qui me montrent qu'apparemment il n'y a pas de match avec un dial peer ; il marque:
 
*Feb 23 17:46:45.138: //-1/B6FAEFD28021/CCAPI/cc_api_display_ie_subfields:
   cc_api_call_setup_ind_common:
   cisco-username=
   ----- ccCallInfo IE subfields -----
   cisco-ani=00033466806631
   cisco-anitype=1
   cisco-aniplan=1
   cisco-anipi=0
   cisco-anisi=0
   dest=4405001
   cisco-desttype=4
   cisco-destplan=1
   cisco-rdie=FFFFFFFF
   cisco-rdn=
   cisco-lastrdn=
   cisco-rdntype=-1
   cisco-rdnplan=-1
   cisco-rdnpi=-1
   cisco-rdnsi=-1
   cisco-redirectreason=-1   fwd_final_type =0
   final_redirectNumber =
   hunt_group_timeout =0
 
*Feb 23 17:46:45.138: //-1/B6FAEFD28021/CCAPI/cc_api_call_setup_ind_common:
   Interface=0x29DBDFE8, Call Info(
   Calling Number=00033466806631,(Calling Name=)(TON=International, NPI=ISDN, Screening=Not Screened, Presentation=Allowed),
   Called Number=4405001(TON=Subscriber, NPI=ISDN),
   Calling Translated=FALSE, Subscriber Type Str=RegularLine, FinalDestinationFlag=FALSE,
   Incoming Dial-peer=0, Progress Indication=ORIGINATING SIDE IS NON ISDN(3), Calling IE Present=TRUE,
   Source Trkgrp Route Label=, Target Trkgrp Route Label=, CLID Transparent=FALSE), Call Id=-1
*Feb 23 17:46:45.138: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
   
*Feb 23 17:46:45.138: :cc_get_feature_vsa malloc success
*Feb 23 17:46:45.138: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
   
*Feb 23 17:46:45.138:  cc_get_feature_vsa count is 1
*Feb 23 17:46:45.138: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
   
*Feb 23 17:46:45.138: :FEATURE_VSA attributes are: feature_name:0,feature_time:806751472,feature_id:38
*Feb 23 17:46:45.138: //38/B6FAEFD28021/CCAPI/cc_api_call_setup_ind_common:
   Set Up Event Sent;
   Call Info(Calling Number=00033466806631(TON=International, NPI=ISDN, Screening=Not Screened, Presentation=Allowed),
   Called Number=4405001(TON=Subscriber, NPI=ISDN))
*Feb 23 17:46:45.138: //38/B6FAEFD28021/CCAPI/cc_process_call_setup_ind:
   Event=0x296BE0F8
*Feb 23 17:46:45.138: //-1/xxxxxxxxxxxx/CCAPI/cc_setupind_match_search:
   Try with the demoted called number 4405001
...etc
 
 
Ce qui me gène c'est que je ne comprends pas du tout pourquoi ça ne match pas avec le dial-peer ; avez-vous une idée ?
tarmac