Grosso problema con IVR e segreteria

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Grosso problema con IVR e segreteria

Postby bfeel » Tue Aug 09, 2011 9:14 am

Salve a tutti,
abbiamo intenzione di acquistare la versione PRO di 3CX perché pensiamo che attualmente sia il software più valido a disposizione. Però prima di acquistarlo lo dobbiamo testare per almeno un mesetto in tutte le sue funzionalità e non abbiamo avuto particolari problemi se non che gli IVR e la segreteria sono muti!

Attualmente il software si trova su un server rack in una serverfarm, l'hardware è molto potente e non ha nessun firewall attivo.

Queste sono le caratteristiche del software e del server:
- Firewall: NO
- Nat: NO
- IP pubblico statico: SI
- Provider SIP: Messagenet
- Scheda audio: NO
- OS: Win 2003 Standard
- Audio abilitato nei servizi: SI
- Codec audio: qualunque possibile ed immaginabile

Abbiamo notato che la cosa strana è che sei volte su dieci, i due IVR che abbiamo creato non si registrano (rimangono col pallino rosso) e infatti se chiami da occupato. Quando le poche volte (a forza di riavviare il servizio) gli IVR si registrano, la chiamata viene sempre instradata correttamente (infatti diventa giallo) ma è completamente muto!
Come mai, qui in locale su Windows 7 funziona tutto?
Inoltre abbiamo notato, che quando è muto, comunque funzionano tutte le opzioni che abbiamo specificato (premi 1 per... premi 2 per.. ecc..) e anche la segreteria funziona, perché puoi lasciare un messaggio dopo dieci secondi che lo riceviamo via mail, ma non si sente mai nulla.
Leggendo in rete, pare che sia obbligatoria la scheda audio fisica, è possibile? Anche installando una virtual soundcard ed abilitandola non funziona.

Qui di seguito postiamo lo status di una chiamata con IVR attivo:

Code: Select all
10:03:00.454  [CM504004]: Registration succeeded for: 10000@BF-MESSAGENET
10:03:00.282  [CM504003]: Sent registration request for 10000@BF-MESSAGENET
10:01:30.448  [CM503008]: Call(5): Call is terminated
10:01:13.385  [CM503007]: Call(5): Device joined: sip:995@127.0.0.1:40600;rinstance=32a5a97b933c49cd
10:01:13.369  [CM503007]: Call(5): Device joined: sip:538***9@sip.messagenet.it:5060
10:01:13.353  [CM505001]: Ext.995: Device info: Device Identified: [Man: 3CX Ltd.;Mod: 3CX IVR;Rev: General] Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [3CX IVR] PBX contact: [sip:995@127.0.0.1:5060]
10:01:13.353  [CM503002]: Call(5): Alerting sip:995@127.0.0.1:40600;rinstance=32a5a97b933c49cd
10:01:13.181  [CM503025]: Call(5): Calling Ext:Ext.995@[Dev:sip:995@127.0.0.1:40600;rinstance=32a5a97b933c49cd]
10:01:13.181  Cannot open stream source 'c:\documents and settings\all users\dati applicazioni\3cx\data\ivr\prompts\onhold.wav'
10:01:13.119  [CM503004]: Call(5): Route 1: Ext:Ext.995@[Dev:sip:995@127.0.0.1:40600;rinstance=32a5a97b933c49cd]
10:01:13.119  [CM503010]: Making route(s) to <sip:995@109.***.121.***:5060>
10:01:13.119  [CM505003]: Provider:[BF-MESSAGENET] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [victor] PBX contact: [sip:5382219@109.***.121.***:5060]
10:01:13.103  [CM503001]: Call(5): Incoming call from +3934735***@(Ln.10000@BF-MESSAGENET) to <sip:995@109.168.121.131:5060>



Ed ora la configurazione globale:

Code: Select all
   Nome   Descrizione   Valore
   ENL   Extension number length   3
   PEC   Prefix for external call   0
   MYPHONEVERSION   MyPhone Version   10.0.20401.1947
   MLSRVADDR   SMTP server address   mail.******.co.uk
   MLSRVPORT   SMTP server port   
   MLFROMADDR   Senders e-mail address for notifications   noreply@******.co.uk
   MLSRVUSER   Username for accessing SMTP server   info@******.co.uk
   MLSRVPASSWORD   Password for accessing SMTP server   zerocool130383
   PRMUPDTM   Time of the last changing within the prompts settings   05/08/2011 14.43.45
   ACPRMSET   The name of currently active prompts set   307392E1-F915-4f3a-9362-5049AADC242C
   STUNSRV   Default STUN server address/port   
   STUNSRVPORT   Default STUN server port   3478
   STUNSRV2   Default second STUN server address/port   
   STUNSRV2PORT   Default second STUN server port   3478
   STUNSRV3   Default third STUN server address/port   
   STUNSRV3PORT   Default third STUN server port   3478
   STUNTOUT   Timeout time for STUN response, ms   3000
   STUNTCHECK   Recheck external IP every, sec   1200
   MSE_USEIVR   Using MS Exchange IVR   0
   MSE_ADDRESS   Address of the MS Exchange IVR   :
   CALLBYNAME_MINLEN   Minimal length of number for call by name   3
   CALLBYNAME_MAXLEN   Maximal length of number for call by name   -1
   ALLOW_CALL_TAKEOVER   Allow takeover of established calls using Re-invites with replaces. Available options 0 off and 1 on   1
   PARK   Park dial code   *0
   UNPARK   Unpark dial code   *1
   PICKUP   Pickup dial code   *20*
   VMAIL   VMail dial code   *4
   SETAVAILABLE   SetAvailable dial code   *60
   SETAWAY   SetAway dial code   *61
   LOGGEDINQUEUE   LoggedInQueue dial code   *62
   LOGGEDOUTQUEUE   LoggedOutQueue dial code   *63
   DIALCODEOUTOFFICE   Deactivate Office Hours - Calendar settings and time used as set in the global Office hours in Settings/General.*641 IN Office *642 OUT OF Office irrespective of current time   
   INTERCOM   Intercom dial code   *9
   BILLINGCODE   BillingCode dial code   **
   SIPPINGPERIOD   Time intervall to poll endpoints with SIP INFO messages, in seconds. 0 - disabled   0
   EXPIRATIONGRACEPERIOD   Expiry Grace Period   0
   ALLOWUSEBUSYOPTFORGROUP   ="1" will enable use of phone 'Busy' status instead of 'PBX status' for members of ring groups   0
   CALLHISTTEMPLSINGLE   Path to CallTemplate.xml for single file output   C:\Documents and Settings\All Users\Dati applicazioni\3CX\Data\CDRTemplates\CDRTemplate-Single.xml
   CALLHISTTEMPLPERCALL   Path to CallTemplate.xml for Per Call file output   C:\Documents and Settings\All Users\Dati applicazioni\3CX\Data\CDRTemplates\CDRTemplate-PerCall.xml
   CALLHISTTEMPLSOCK   Path to CallTemplate.xml for Socket text output   C:\Documents and Settings\All Users\Dati applicazioni\3CX\Data\CDRTemplates\CDRTemplate-Socket.xml
   CALLHISTTEMPLSOCKLISTEN   Path to CallTemplate.xml for listening Socket text output   C:\Documents and Settings\All Users\Dati applicazioni\3CX\Data\CDRTemplates\CDRTemplate-SocketListen.xml
   CALLHISTTODB   Flag to write callhistory to Database or not   true
   CALLHISTITERIVL   callhistory Interval in seconds for processing iteration   60
   CALLHISTWEEKLY   Flag true=1 file per Week or false=per day   true
   CDRENABLE   Enabling of CDR records output 0 = disabled and 1 = enabled. Default=0   0
   SMDRHOSTPORT   SMDR Listening host and port, empty host = disabled(default), use 0.0.0.0:1752   
   FORCEREAUTH   forces PBX to re-request authentication if it has nonce that has already been used.  If value = 1, PBX will force the check. If set to 0 - PBX will not check reuse of nonce   1
   NONCEEXPIRATION   defines nonce expiration time (seconds). By default it is 10 sec. After this time has passed since nonce issue -- PBX will treat that nonce as expired and will force re-authentication (will issue 407 response with new nonce freshly generated.   10
   VMPINREQUIRED   Determines whether VM will ask for pin. By default it is 1. Set to 0 to access the VM menu directly(e.g. Hotel extensions).   1
   MSADDRFC2833FORINBANDDTMF   Allows Media server to distribute recognized in-band DTMF as RFC2833 to other participants of the call   1
   CALLRECORDSROOT   Define the root path of call recordings.   C:\Documents and Settings\All Users\Dati applicazioni\3CX\Data\Recordings
   PREPENDNAMETOCID   name of queue / inbound rule will be pre-pended rather than appended   1
   APPENDCIDQUEUE   enable   1
   FAXDIRECTSDP   modifies behavior of SDP negotiation when the FAX extension is the destination   1
   SECURESIPPATH   Secure SIP Certificate Path   C:\Documents and Settings\All Users\Dati applicazioni\3CX\Bin\Cert
   DISABLE_SYSCALL_AUTH   This option controls authentication of INVITE requests from IVR (makecall), Queue ("polling" call) and Paging group (intercom call to member of paging group). 0 (default)=requires authentication. 1=does not require authentication   0
   MSRTPTOS   This parameter instructs the Media Server to mark all the packets it generates with a value between 0 and 63 in the DSCP field in the IP Header. Default Value: 0 - Valid Values: 0-63 Please note that Windows XP and Server 2003 disables IP Header tagging by default. You will need to set a DWORD registry value called DWORD registry value called DisableUserTOSSetting in the following location "HKEY_LOCAL_MACHINE\SYSTEM\CurrentControlSet\Services\Tcpip\Parameters" and give it value "0". You will need to restart the machine for this setting to kick in.   0
   IVRSIPINTERFACE   Binding address of NIC for IVR component   127.0.0.1
   POSIPINTERFACE   Binding address of NIC for Parking orbit component   127.0.0.1
   CPSIPINTERFACE   Binding address of NIC for conferenceplace component   127.0.0.1
   BUSYPROMPT   Replaces busy tone with Busy prompt. This will answer the call. 0 = busy signal on busy. 1 = Busy prompt played out when busy   0
   BMCALLTOUT   Boomerang caller timeout in seconds   120
   BMACCEPTTOUT   Boomerang acceptor timeout in seconds   5
   VERSION   Version   10.0.20401.1947
   MUSICONHOLDBEHAVIOUR   Controls the behaviour of Music on Hold Default 0 OFF   0
   RANDOMMUSICPERCALL   Randomizes Music on Hold per call made Default value 0 OFF ie MOH is randomized Per Day. When 1 option is on and MOH will be randomized per call   1
   CONFPLACE_SIPPORT   specifies SIP port for Conference Room component.   40300
   CONFPLACE_RTPFIRST   RTP range of ports allocated for conference room   40310
   CONFPLACE_RTPLAST   RTP range of ports allocated for conference room   40438
   IVR_SIPPORT   specifies SIP port for IVR component.   40600
   ENABLECONFERENCEPIN   Enables the Conference Server Password Pin for authentication for new created conferences Default Off 0/On 1   0
   CONFERENCEEXTPIN   Default Password Pin number for authentication for new created conferences Default value 0000   0000
   IVR_RTPFIRST   RTP range of ports allocated for IVR. Default 128 sim calls. 2 ports per call   40610
   IVR_RTPLAST   RTP range of ports allocated for IVR. Default 128 sim calls. 2 ports per call   40866
   PARKORB_SIPPORT   specifies SIP port for ParkingOrbit omponent.   40000
   PARKORB_RTPFIRST   RTP range of ports allocated for Parking Orbit   40010
   PARKORB_RTPLAST   RTP range of ports allocated for Parking Orbit   40138
   CALLASSISTANTVERSION   Current Version of the Call assistant Server   20401
   PHONEDEVICESVERSION   Version number of Phone Devices Auto Detection   20401
   MYPHONESIPAUTH   Hide or show SIP Authentication Tab in myPhone. 0 hide 1 show   0
   TIME_NTP_SERVER   NTP server   time.ien.it
   TIME_TIMEZONE_AASTRA   Timezone in Aastra-compatible format   IT-Rome CET
   TIME_TIMEZONE_GRANDSTREAM   Timezone in Grandstream-compatible format   GMT-5:00 (US Eastern Time, New York)
   TIME_TIMEZONE_LINKSYS   Timezone in Linksys-compatible format   GMT-5:00 (US Eastern Time, New York)
   TIME_TIMEZONE_SNOM   Timezone in Snom-compatible format   -5 United States - Eastern Time
   TIME_UTC_OFFSET_MINUTES   Difference from UTC in minutes   60
   TIME_UTC_OFFSET_SECONDS   Difference from UTC in seconds- Used for Polycom phones   GMT-5:00 (US Eastern Time, New York)
   TIME_DST_START_MONTH   Month when DST settings start   3
   TIME_DST_START_DAY   Day when DST settings start   28
   TIME_DST_START_HOUR   Hour when DST settings start   2
   TIME_DST_END_MONTH   Month when DST settings end   10
   TIME_DST_END_DAY   Day when DST settings end   25
   TIME_DST_END_HOUR   Hour when DST settings end   3
   TIME_DST_SAVING_MINUTES   Number of minutes to add while DST is active   60
   PERS_PHBK_ONLY   System wide option. Currently available for Polycom only. Available values 0 1. 1 removes company phonebook leaving personal phonebook only.   0
   WEBSERVER   Webserver used in 3CX Phone System   ABYSS
   VMDIALOUTENABLED   Enable dialout calls from voicemail menu. By default it is 0. Set to 1 to enable dialout calls.   0
   TIME_TIMEZONE_YEALINK   Set timezone for Yealink - Available options are -13 to 12 Default -5 US Eastern Standard Time   GMT-5:00 Cuba(Havana)
   ALLOWSOURCEASOUTBOUND   Used only for remote extensions. If 1 (on) PBX saves the source IP:port of last successful OK to REGISTER message and constructs target requests to that IP:port, except those that have FQDN as target. If set to 0, ACK will be sent to IP:port specified in Contact header of 200/INV   1
   MSEXCH_SPECIALMENU   Value is a string. You can enter the value MNU for compatability with Exchange 2007 or voicemail special number example 999. Default value is Blank which defaults to 999   
   FAXOVEREMAILGATEWAY   The default 3cx fax extension. The fax over mail extension   888
   DUMMYGLOBALOFFHOURSEV   The dummy parameter used for office hours change event   
   BLACKLISTCHANGED   The dummy parameter used for black list change event   
   SYSTEMLOGGINGLEVEL   The current logging level   2
   ECHOTEST   Echo Test Extension used to test echo, delay and round trip timing   *777
   SLA_ENABLED   Enable or disable SLA feature - Available values 0 off or 1 on Default Value Off   0
   SLA_ONSUBSCRIBE   Defines how many SLA slots are available for each shared appearance when SLA is activated by device subscription. 0 means none. Positive N means N slots are allocated on the first subscription   0
   CHECK_CALL_DIVERSION_LOOP   Algorithm to check for infinite system routing loops. If set to 1 Internal call history will be used for loop detection. If set to 0 loop detection works for forward all type rules   1
   EXT_REFMT_LOCAL_PREFIX   This prefix (outbound rule) will be added to dialled outgoing numbers from extensions for LOCAL formatted calls   
   EXT_REFMT_NATL_PREFIX   This prefix (outbound rule) will be added to dialled outgoing numbers from extensions for NATIONAL formatted calls   
   EXT_REFMT_INTL_PREFIX   This prefix (outbound rule) will be added to dialled outgoing numbers from extensions for INTERNATIONAL formatted calls   
   SEC_RATE_TIME_ALLOW   Allowed free time - Abuse counting starts but no checks are made in this period - After this period expires, checks are enforced. Default value in ms   200
   SEC_RATE_BARRIER_1   Protection Layer 1 Triggered when Server receives X pkts/sec. Further requests will be rejected and will respond with 503 to many requests-resend after value SEC_RETRY_AFTER parameter   2000
   SEC_RETRY_AFTER   Value specified in 503 Too many requests triggered for Barrier 1 responses. Default value in seconds   5
   SEC_RATE_BARRIER_2   Protection Layer 2 triggers when layer 1 is abused. Default value in Packets per second. Abusing IP will be immediately blacklisted for value specified in SEC_BLACKLIST_TIME parameter   4000
   SEC_BLACKLIST_TIME   Amount in seconds to keep the abusing IP address in blacklist. Default value in seconds   1800
   SEC_MAX_FAILED_AUTH   Authentication protection. If attacker,device sends failed register/proxy auth required, entity is blacklisted. SEC_BLACKLIST_TIME parameter will be loaded. Default value 25   25
   SHOWEXTWEAKPASS   Enabled by default - Highlights in red extensions that have weak passwords and voicemail pin numbers   1
   REP_CMDTIMEOUT   This is the max time allocated to the reporter to query the database for generation of reports. On large systems should be increased if exceptions are detected. Default time in seconds   120
   QM_RTPFIRST   RTP range of ports allocated for Queue Manager   32000
   QM_RTPLAST   RTP range of ports allocated for Queue Manager   32399
   ALLOWFWDTOEXTFROMGRP   Allows members of ring groups and queue to forward calls to external numbers. Requires that the gateway must be able to accept a call ONLY when the callee has answered - Early media OFF. Default enabled   1
   PHONEBOOK_MIN_MATCH   Available values 0 disabled,-1 exact match only, N>0 minimum match length required for best-match strategy, numbers are compared from tail to head   0
   PHONEBOOK_LAST_FIRST   Defines caller name replacement format. Available values 0=FirstName LastName 1=LastName FirstName Default 1   1
   DIALCODEPROFILE   Change profile using dial code Default *3. Options *30 Available *31 Away *32 OutOfOffice *33 Custom1 *34 Custom2   *3
   TIME_TIMEZONE_GRANDSTREAMEXEC   Timezone in Grandstream-compatible format for new Grandstream Executive phone models   GMT-5:00 (Eastern Time)
   DISABLE_OUTBOUND_IN_OOH   Disables Outbound Calls in VP, Bridges, Gateways when Out of office hours is triggered. If 0 feature is disabled. If set to 1 Feature is enabled. Default 0   0
   ALLOWSOURCEASOUTBOUNDVP   Used only for VOIPProviders. Default 0 If 1 (on) PBX saves the source IP:port of last successful OK to REGISTER message and constructs target requests to that IP:port, except those that have FQDN as target. If provider needs this to 1 to work, then this provider is unreliable   0
   VMN_RETRY_MINUTES   Time interval in minutes to resend email notifications   15
   VMN_STOP_RETRY_HOURS   Time interval in hours to stop sending a failed email notification Default 3 days   72
   CALL_TRANSFER_TRACKING_TIME   Time interval in seconds in which Transferer will track status of blind transfer. If transfer will not be completed during this time, the transferer will be removed from the call and the result will depend only on the destination   60
   VMN_TEMPLATE   Path where Voice Mail Email Notification template is located. This will be used to send voicemail email notifications upon receipt of a voicemail. Template name is VMNTemplate.xml   C:\Documents and Settings\All Users\Dati applicazioni\3CX\Data\Http\Templates\VMNTemplate.xml
   SEC_MAX_CHALLENGES   When IP Challenges 3CX and does not answer to 407 with authentication, these requests are counted per IP and when value is reached, IP is blacklisted for Blacklist time specified   1000
   USERAGENTSTRING   User Agent that 3CX will send can be customized. This will hide the identity of the VoIP PBX so attackers cannot know what is behind port 5060 Needs full restart of services   
   RESPONSE_ERROR_CODES   These are all the error codes that will be reported with Event ID 12294 when received from VoIP Prov Bridge PSTN Trunk. Comma separated list of sip error codes   400,401,403,404,405,406,408,409,415,480,481,484,486,488,500,501,502,503,504,513,600,603,604,606
   SPUPDATE_EMAIL_NOTIFICATION   Sends an email notification to the administrator when there is a new service pack update   1
   ENABLEEARLYMEDIA   Switch off Early Media Support on 1 or off 0   1
   SP_VERSDESC   The installed service pack version description   SP.
   MAXNOANSWERTIMEOUT   This is the maximum timeout for ringing calls used for backup purposes - Default is 5 minutes   300
   SIP_FORKED_ID_BUSY   Available options 0 and 1. If set to 0, when one of the sip forked entities sends a busy signal, all invites to all participants are canceled at once. If set to 1, each participant must send a busy respectively. Other SiP Forked devices remain on target   0
   STUNDISABLED   Flag used to inform PBX about the Stun disabled status   1
   WEBSERVERUSER   The admin ID   admin
   WEBSERVERPASS   The admin pass   *************
   APPPATH   Application Installed Path   C:\Programmi\3CX PhoneSystem\
   IVRPROMPTPATH   Path for the ivr prompt file   C:\Documents and Settings\All Users\Dati applicazioni\3CX\Data\Ivr\Prompts
   MUSICONHOLDFILE   Music file for music on hold   C:\Documents and Settings\All Users\Dati applicazioni\3CX\Data\Ivr\Prompts\onhold.wav
   OPERATOR   Operator extension number   100
   SIPDOMAIN   SIP Domain Name   109.***.121.***
   LANGUAGE   Installation language code   it
   BUILD   Build version   1947
   PBXPUBLICIP   The public Ip/domain of the PBX   109.***.121.***
   DEFAULTLOCALIP   The default local IP of the PBX   109.***.121.***
   SIPPORT   Sip Port   5060
   FIRSTLOCALPORT   First port to use for internal calls   7000
   LASTLOCALPORT   Last port to use for internal calls   7499
   FIRSTEXTPORT   First port to use for external calls   9000
   LASTEXTPORT   Last port to use for external calls   9049
   DST   DST option   0
   MS_BLINDEXTPORTS   Flag used to inform PBX about the Stun disabled status   1
   MSALLOWNONSEQUENTIALPORTS   Allow non-sequental RTP port   0
   EXTCRNOTIFY   Notify the User when Extension is Added   1
   SWITCHSRTP   Switch on/off SRTP stream   0
   LIMIT_PICKUP_TO_GROUP   Only extensions from group can pickup the call.   0
   KEEPALIVEENABLED   Allow 3CX to send keep alives to a VoIP provider and is
                                                                        used in situations were static port mappings are not possible.   1
   COUNTRYCODE   Country index for e164 Number processing   39
   COUNTRYINDEXNAME   Country name for the country index   Italy
   REMOVECOUNTRYCODE   Remove if same Country   1
   REMOVEAREACODE   Remove if same Area Code   0
   ALLOWOUTBOUNDSIP   Allow calls to external SIP URIs   1
   SHOWPASSW   Specify if the extension password should be shown   1
   STUNSRVTMP   Default Temporary STUN server address/port   stun.3cx.com
   STUNSRV2TMP   Default Temporary second STUN server address/port   stun2.3cx.com
   STUNSRV3TMP   Default Temporary third STUN server address/port   stun3.3cx.com
   MSEXTERNALADDRESS   Public IP to specify in Contact and SDP for external entities   109.***.121.***
   CMEXTERNALADDRESS   Public IP to specify in Contact and SDP for external entities   109.***.121.***
   MSEXTERNALINTERFACE   Network card to use to communicate with external entities.   109.***.121.***
   CMEXTERNALINTERFACE   Network card to use to communicate with external entities.   109.***.121.***
         


Ovviamente per motivi di privacy, alcuni dati tipo IP pubblico, nomi host ed account SIP sono stati oscurati.

Qualcuno ci illumina?
bfeel
New User
 
Posts: 16
Joined: Sat Aug 06, 2011 4:05 pm

Re: Grosso problema con IVR e segreteria

Postby bfeel » Tue Aug 09, 2011 12:26 pm

Dimenticavamo, questo è il log del check-firewall:

Code: Select all
3CX Firewall Checker, v1.0. Copyright (C) 3CX Ltd. All rights reserved.

<13:24:57>: Phase 1, checking servers connection, please wait...
<13:24:57>: Stun Checker service is reachable. Phase 1 check passed.
<13:24:57>: Phase 2a,  Check Port Forwarding to UDP SIP port, please wait...
<13:24:57>: UDP SIP Port is set to 5060. Response received correctly with no translation. Phase 2a check passed.

<13:24:57>: Phase 2b. Check Port Forwarding to TCP SIP port, please wait...
<13:24:57>: TCP SIP Port is set to 5060. Response received correctly with no translation. Phase 2b check passed.

<13:24:57>: Phase 3. Check Port Forwarding to TCP Tunnel port, please wait...
<13:24:57>: TCP TUNNEL Port is set to 5090. Response received correctly with no translation. Phase 3 check passed.

<13:24:57>: Phase 4. Check Port Forwarding to RTP external port range, please wait...
<13:25:03>: UDP RTP Port 9000. Response received correctly with no translation. Phase 4-01 check passed.
<13:25:03>: UDP RTP Port 9001. Response received correctly with no translation. Phase 4-02 check passed.
<13:25:03>: UDP RTP Port 9002. Response received correctly with no translation. Phase 4-03 check passed.
<13:25:03>: UDP RTP Port 9003. Response received correctly with no translation. Phase 4-04 check passed.
<13:25:03>: UDP RTP Port 9004. Response received correctly with no translation. Phase 4-05 check passed.
<13:25:03>: UDP RTP Port 9005. Response received correctly with no translation. Phase 4-06 check passed.
<13:25:03>: UDP RTP Port 9006. Response received correctly with no translation. Phase 4-07 check passed.
<13:25:03>: UDP RTP Port 9007. Response received correctly with no translation. Phase 4-08 check passed.
<13:25:03>: UDP RTP Port 9008. Response received correctly with no translation. Phase 4-09 check passed.
<13:25:03>: UDP RTP Port 9009. Response received correctly with no translation. Phase 4-10 check passed.
<13:25:03>: UDP RTP Port 9010. Response received correctly with no translation. Phase 4-11 check passed.
<13:25:03>: UDP RTP Port 9011. Response received correctly with no translation. Phase 4-12 check passed.
<13:25:03>: UDP RTP Port 9012. Response received correctly with no translation. Phase 4-13 check passed.
<13:25:03>: UDP RTP Port 9013. Response received correctly with no translation. Phase 4-14 check passed.
<13:25:03>: UDP RTP Port 9014. Response received correctly with no translation. Phase 4-15 check passed.
<13:25:03>: UDP RTP Port 9015. Response received correctly with no translation. Phase 4-16 check passed.
<13:25:03>: UDP RTP Port 9016. Response received correctly with no translation. Phase 4-17 check passed.
<13:25:03>: UDP RTP Port 9017. Response received correctly with no translation. Phase 4-18 check passed.
<13:25:03>: UDP RTP Port 9018. Response received correctly with no translation. Phase 4-19 check passed.
<13:25:03>: UDP RTP Port 9019. Response received correctly with no translation. Phase 4-20 check passed.
<13:25:03>: UDP RTP Port 9020. Response received correctly with no translation. Phase 4-21 check passed.
<13:25:03>: UDP RTP Port 9021. Response received correctly with no translation. Phase 4-22 check passed.
<13:25:03>: UDP RTP Port 9022. Response received correctly with no translation. Phase 4-23 check passed.
<13:25:03>: UDP RTP Port 9023. Response received correctly with no translation. Phase 4-24 check passed.
<13:25:03>: UDP RTP Port 9024. Response received correctly with no translation. Phase 4-25 check passed.
<13:25:03>: UDP RTP Port 9025. Response received correctly with no translation. Phase 4-26 check passed.
<13:25:03>: UDP RTP Port 9026. Response received correctly with no translation. Phase 4-27 check passed.
<13:25:03>: UDP RTP Port 9027. Response received correctly with no translation. Phase 4-28 check passed.
<13:25:03>: UDP RTP Port 9028. Response received correctly with no translation. Phase 4-29 check passed.
<13:25:03>: UDP RTP Port 9029. Response received correctly with no translation. Phase 4-30 check passed.
<13:25:03>: UDP RTP Port 9030. Response received correctly with no translation. Phase 4-31 check passed.
<13:25:03>: UDP RTP Port 9031. Response received correctly with no translation. Phase 4-32 check passed.
<13:25:03>: UDP RTP Port 9032. Response received correctly with no translation. Phase 4-33 check passed.
<13:25:03>: UDP RTP Port 9033. Response received correctly with no translation. Phase 4-34 check passed.
<13:25:03>: UDP RTP Port 9034. Response received correctly with no translation. Phase 4-35 check passed.
<13:25:03>: UDP RTP Port 9035. Response received correctly with no translation. Phase 4-36 check passed.
<13:25:03>: UDP RTP Port 9036. Response received correctly with no translation. Phase 4-37 check passed.
<13:25:03>: UDP RTP Port 9037. Response received correctly with no translation. Phase 4-38 check passed.
<13:25:03>: UDP RTP Port 9038. Response received correctly with no translation. Phase 4-39 check passed.
<13:25:03>: UDP RTP Port 9039. Response received correctly with no translation. Phase 4-40 check passed.
<13:25:03>: UDP RTP Port 9040. Response received correctly with no translation. Phase 4-41 check passed.
<13:25:03>: UDP RTP Port 9041. Response received correctly with no translation. Phase 4-42 check passed.
<13:25:03>: UDP RTP Port 9042. Response received correctly with no translation. Phase 4-43 check passed.
<13:25:03>: UDP RTP Port 9043. Response received correctly with no translation. Phase 4-44 check passed.
<13:25:03>: UDP RTP Port 9044. Response received correctly with no translation. Phase 4-45 check passed.
<13:25:03>: UDP RTP Port 9045. Response received correctly with no translation. Phase 4-46 check passed.
<13:25:03>: UDP RTP Port 9046. Response received correctly with no translation. Phase 4-47 check passed.
<13:25:03>: UDP RTP Port 9047. Response received correctly with no translation. Phase 4-48 check passed.
<13:25:03>: UDP RTP Port 9048. Response received correctly with no translation. Phase 4-49 check passed.
<13:25:03>: UDP RTP Port 9049. Response received correctly with no translation. Phase 4-50 check passed.


Application exit code is 0
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Re: Grosso problema con IVR e segreteria

Postby eurylink.com » Tue Aug 09, 2011 3:33 pm

Il problema al 99% è di installazione locale ed è relativo alla versione di windows server utilizzata e dai serizi o software contemporaneamente attivati sul server. Se gli interni virtuali dell'IVR non si registrano il problema può essere sia sul tipo di servizi attivati in parallelo sul server in questione che sulla particolare versione di sistema operativo utilizzato (windows 2003 web server ad esempio). Occorrerebbe una verifica diretta sul server in questione per capire esattamente come procedere.
Se già avete verificato il funzionamento dei telefoni remoti e che la configurazione sia bidirezionale (il che dovrebbe escludere problemi di NAT sul lato server) il problema dovrebbe essere risolvibile. Va ovviamente sottolineato che una configurazione di questo tipo non è ufficialmente supportata da 3CX e quindi richiede un'attezione particolare sia sulla impostazioni degli interni remoti che del server 3cx stesso.
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Re: Grosso problema con IVR e segreteria

Postby bfeel » Tue Aug 09, 2011 3:51 pm

Salve e grazie per la risposta.
Software reinstallato e verificato possibili contrasti. Ora sembra che gli IVR si registrino tranquillamente ma il problema persiste.

Per le prove noi utilizziamo un telefono PSTN normalissimo, e l'interno 100 registrato riceve tranquillamente le chiamate.
Anche gli IVR ricevono la chiamata, ma sono muti (sia sul telefono PSTN che su altri interni).

Abbiamo notato inoltre che l'interno 100 trasmette il suono della voce al telefono PSTN, ma al contrario, sull'interno 100 non si sente niente.

Cosa ne pensate? Dobbiamo abbandonare l'idea di tenerlo su un 2k3 Server?
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Re: Grosso problema con IVR e segreteria

Postby eurylink.com » Tue Aug 09, 2011 4:15 pm

Per esperienza la gestione di un sistema remoto comporta molte problematiche che normalmente sono risolvibili a patto di operare opportunamente con le configurazioni, la scelta dell'hardware, la configurazione di router/firewall sulle sedi remote e molti altri aggiustamenti che in genere non sono necessari sulle configurazioni in locali.
Anche le prescrizioni sull'hardware da utilizzare variano molto rispetto alle installazioni completamente in locale.
Non ultima è la valutazione sul provider utilizzato, sulla banda disponibile (sia locale sia sul server remoto) sulle eventuali limitazioni sulle macchine server e sui sistemi di virtualizzazione utilizzati. Molto spesso è sufficiente un piccolo dettaglio a livello di configurazione per creare problemi e ciò vale maggiormente sulle configurazioni remote.
Il problema della monodirezionalità dell'audio in caso di installazioni remote è più critico e può essere dovuto ad una molteplicità di fattori sia locali (router/firewall locali) sia remote (errata configurazione di 3CX, errata configurazione della macchina server, sistema di virtualizzazione utilizzato.....)
La scelta di windows 2003 server in se non crea problemi se non sulla versione web server che richiede qualche accorgimento particolare in più. Per una eventuale verifica sulle funzionalità commerciali potete utilizzare la chiave demo di attivazione indicata in questo articolo: http://www.3cx.com/blog/releases/3cx-ph ... ce-pack-2/
Prestate inoltre attenzione che la versione Free di 3CX (e anche la demo) non contemplano l'utilizzo del codec G729 e quindi obbligano ad utilizzare un codec tipo G711 oppure GSM a più elevatro consumo di banda.
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Re: Grosso problema con IVR e segreteria

Postby bfeel » Tue Aug 09, 2011 5:39 pm

Noi siamo un ISP, abbiamo banda illimitata oltre 155Mbit, non è quello il problema. E neanche il firewall software/hardware perché la macchina è allacciata direttamente alla rete senza ostacoli.

Se non risolviamo, opteremo per un centralino onsite.
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Re: Grosso problema con IVR e segreteria

Postby ip19 » Tue Aug 09, 2011 5:44 pm

Avete provato banalmente con un altro operatore?
la prova migliore sarebbe con VoipVoice che è certificato, ma anche con eutelia che è gratuito può andare..
La macchina è fisica o virtuale? è un server dedicato al 3cx o gestisce anche altri servizi?
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Re: Grosso problema con IVR e segreteria

Postby bfeel » Tue Aug 09, 2011 6:22 pm

E' un server rack dedicato IBM Xeon QuadCore. Penso possa bastare.
Non è un problema di operatore, su Win7 in locale funziona tutto.
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Re: Grosso problema con IVR e segreteria

Postby ip19 » Tue Aug 09, 2011 6:43 pm

bfeel wrote:Non è un problema di operatore, su Win7 in locale funziona tutto.

senza provare non è detto :) probabilmente hai ragione, ma magari stai impazzendo e il problema è lì.. una prova io la farei
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Re: Grosso problema con IVR e segreteria

Postby bfeel » Tue Aug 09, 2011 6:45 pm

Stanotte provo e domattina vedremo.

Stiamo impazzendo!
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Re: Grosso problema con IVR e segreteria

Postby ip19 » Tue Aug 09, 2011 7:37 pm

un po' di domande:
Consegna audio al PBX sull'operatore voip è attivo?
il problema ce l'avete solo con gli ivr o anche se chiamate tra due interni registrati al centralino?
e se chiamate da un interno a un telefono tramite provider voip?
e se chiamate da un esterno al numero del provider voip assegnato direttamente ad un interno al posto che all'ivr?

8|

ciao
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Re: Grosso problema con IVR e segreteria

Postby bfeel » Wed Aug 10, 2011 11:32 am

Tra due interni registrati non sussiste alcun problema.
Se settiamo la regola di ingresso direttamente sull'interno, l'audio sull'interno VoIP non si sente, sul PSTN sì.
Da VoIP a PSTN funziona tutto.

Come si spiega?!
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Re: Grosso problema con IVR e segreteria

Postby bfeel » Wed Aug 10, 2011 1:54 pm

Rettifico e chiedo scusa.

Anche tra due interni il problema sussiste! La musica di attesa non si sente affatto!
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Re: Grosso problema con IVR e segreteria

Postby ip19 » Wed Aug 10, 2011 2:50 pm

bfeel wrote:Rettifico e chiedo scusa.

Anche tra due interni il problema sussiste! La musica di attesa non si sente affatto!


fatico un po' a capire.. il problema è SOLO sulla musica d'attesa (quindi i due interni si parlano, ma quando uno mette in attesa l'altro c'è il "muto") o su tutta la chiamata? si sente solo da un lato? se invertite il chiamante e il chiamato, si inverte anche il lato che sente?
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Re: Grosso problema con IVR e segreteria

Postby damiano » Wed Aug 10, 2011 3:40 pm

una possibile soluzione te l'ha data Matteo nei post precedenti,
se ignori gli aiuti e continui a porre problematiche non credo arriverai mai ad un dunque

:)

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