My calls are not going to voice mail, the phone just rings any suggestions?

Voice mail is installed, but incoming calls are not goint to voice mail.. What can I do?

Check with your Phone Carrier that Voice Mail is Setup and Activated...

Similar Messages

  • Calls not going to Voice Mail

    Hi,
    We've had skype for ages.  We use it for our business and have skype numbers and multiple skype users.
    For some reason, the past three days, any incoming calls on our user sales.munz1 is not going to voice mail.  It just rings, and rings and rings.
    Since the other users don't use voice mail, this is the only user with the issue.
    What's up Skype?  We haven't changed anything.
    At this moment, we've unticked go to Voice Mail, otherwise we can not hear the phone ring and ring and ring and pick up customer calls.
    Any solutions?

    It really can only be on the carrier side as it is their servers that have number of rings etc set before dumping to voice mail.  Voice mail again is on their servers, your phone doesn't record voice mail, they do.  Their servers then tell your phone you have voice mail and send a *copy* to visual voice mail.
    I would call them again.  If never dumps to voice mail, it is their servers not stopping the call and dumping it to voice mail.

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    Please do the needful.

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  • In Lync, if a call is not answered, and does not go to voice mail, is anything recorded in the Lync database ?

    We are using Lync 2010. This question relates to the LCSCDR database.
    In Lync, if a call is not answered, and does not go to voice mail, is anything recorded in the Lync database ? Are there any options to make Lync store a record of unanswered calls ?
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  • Calls are not getting thru in Cisco voice GW for a particular Number

    Cisco gateway is connecte to a PBX with an Qsig interface, for a particualr destination number the calls are not gettin estabilished.
    the output of the Q931 debug :
    Aug 16 16:17:46.145: ISDN Se0/0/0:23 Q931: RX <- SETUP pd = 8  callref = 0x7E05
            Bearer Capability i = 0x8090A2
                    Standard = CCITT
                    Transfer Capability = Speech
                    Transfer Mode = Circuit
                    Transfer Rate = 64 kbit/s
            Channel ID i = 0xA98396
                    Exclusive, Channel 22
            Facility i = 0x9FAA068001008201008B0100A16E0202070102011530650201010A010
    1800101A111A00FA50D0A010212083530303035393938A211A00FA50D0A010212083530303035393
    938A312801054454C45434F4D20574F524B524F4F4DA412801054454C45434F4D20574F524B524F4
    F4DA50C06062B0C02FF373730020500
            Facility i = 0x9FAA068001008201008B0100A11D0202010002010080144E455453202
    F204C4F4E472044495354414E4345
            Calling Party Number i = 0x2183, '8168911010'
                    Plan:ISDN, Type:National
            Called Party Number i = 0x89, '18553808521'
                    Plan:Private, Type:Unknown
            Sending Complete
    Aug 16 16:17:46.149: ISDN Se0/0/0:23 Q931: TX -> CALL_PROC pd = 8  callref = 0xF
    E05
            Channel ID i = 0xA98396
                    Exclusive, Channel 22
    Aug 16 16:17:55.709: ISDN Se0/0/0:23 Q931: TX -> DISCONNECT pd = 8  callref = 0x
    FE05
            Cause i = 0x80BF - Service/option not available, unspecified
    Aug 16 16:17:55.741: ISDN Se0/0/0:23 Q931: RX <- RELEASE pd = 8  callref = 0x7E0
    5
    Aug 16 16:17:55.741: ISDN Se0/0/0:23 Q931: TX -> RELEASE_COMP pd = 8  callref =
    0xFE05
    The Qsig and dial-peer configration :
    interface Serial0/0/0:23
    no ip address
    encapsulation hdlc
    isdn switch-type primary-qsig
    isdn overlap-receiving
    isdn incoming-voice voice
    isdn send-alerting
    no cdp enable
    dial-peer voice 1 voip
    description To CBTS GK
    destination-pattern +1T
    signaling forward rawmsg
    session protocol sipv2
    session target ipv4:10.9.5.10
    session transport tcp
    voice-class codec 1
    dtmf-relay rtp-nte
    no vad
    interface Serial0/0/0:23
    no ip address
    encapsulation hdlc
    isdn switch-type primary-qsig
    isdn overlap-receiving
    isdn incoming-voice voice
    isdn send-alerting
    no cdp enable
    dial-peer voice 1 voip
    description To CBTS GK
    destination-pattern +1T
    signaling forward rawmsg
    session protocol sipv2
    session target ipv4:10.9.5.10
    session transport tcp
    voice-class codec 1
    dtmf-relay rtp-nte
    no vad

    Hi Raj,
    My name is Edson Pineiro, I understand that your problem description is in regards to failed incoming calls from a qsig trunk.
    According to the received q931 setup message I can see the called party number is 18553808521 and as so the gateway should route the dnis based on the best match in destination-pattern. My first suggestion would be to ensure your outgoing dial-peers has a matching destination-pattern that matches the dialed number, for example:
    dial-peer voice 1 voip
    destination-pattern 1T
    The T is a wild card for any digit any length
    Or you can be very specific.
    dial-peer voice 1 voip
    destinaton-pattern 18553808521
    The next suggestion would be to ensure that your incoming pots dial-peers contains 'direct-inward dial'. This is so that you don't receive secondary dial tone when dialing in, which I don't think is happening here.
    Another suggestion would be to remove 'isdn overlap-receiving' from interface serial 0/0/0:23. Reason being is that the DNIS received is enbloc and not overlapping. You can clearly see that the complete e164 number is received within the setup and no further digits are needed.
    But overall the disconnect cause code is 0x80BF the 80 portion is related to the source of the disconnect which is the router and BF "Service/option not available, unspecified" which is described as:
    The network or remote equipment cannot provide the service option that the user requests, due to an unspecified reason. A subscription problem can cause this issue.
    Any ways seems like the router does not support the protocol or type of message included in the Setup. After decoding one of the facility message:
            Facility i = 0x9FAA068001008201008B0100A16E0202070102011530650201010A010
    1800101A111A00FA50D0A010212083530303035393938A211A00FA50D0A010212083530303035393
    938A312801054454C45434F4D20574F524B524F4F4DA412801054454C45434F4D20574F524B524F4
    F4DA50C06062B0C02FF373730020500
    decode -->
    Facility IE first byte (protocol profile): 0x9f(Network Extentions), depends on Network Protocol Profile
    **Note:
    **0x91/0x9f both be used by older qsig spec, including:
    **ISO 11582:1995, ETSI 300 239:1993/1995
    **newer qsig spec use 0x9f only, including:
    **ISO 11582:1995/Cor.1:1999, ECMA 165(4th), ETSI 300 239:2003
    **see CSCeb58118 for CCM compatibility issue
    NetworkFacilityExtension ::= {
    sourceEntity: 0
    destinationEntity: 0
    NetworkProtocolProfile not present
    APDU is a ROSE
    0
    DivertingLegInformation2Invoke ::= {
    invokeID: 1793
    operationValue: 21
    argument: DivertingLegInformation2Arg ::= {
    diversionCounter: 1
    diversionReason: 1
    originalDiversionReason: 1
    divertingNr: PrivatePartyNumber ::= {
    privateTypeOfNumber: 2
    privateNumberDigits: 50005998
    originalCalledNr: PrivatePartyNumber ::= {
    privateTypeOfNumber: 2
    privateNumberDigits: 50005998
    redirectingName: 54 45 4C 45 43 4F 4D 20 57 4F 52 4B 52 4F 4F 4D
    originalCalledName: 54 45 4C 45 43 4F 4D 20 57 4F 52 4B 52 4F 4F 4D
    Looks like this is a redirected call (call forward or transfer), the redireted number is "50005998" and the other end of the PRI maybe attempting to do either a 2 B channel transfer or B channel optimization, which is not supported certain gateways or needs the use of a tcl scripts. Any ways is it possible to confirm if such features are enabled on the other end of the qsig trunk? and what the number 50005998 is assigned too. This may warrant a TAC case.
    However please ensure your carry through the first three configuration changes before looking at the possible bad facility message.
    Here are some good documents on ISDN, IOS dial-peers and call legs:
    Understanding debug isdn q931 Disconnect Cause Codes
    http://www.cisco.com/en/US/tech/tk801/tk379/technologies_tech_note09186a008012e95f.shtml
    Configuring Telephony Call-Redirect Features
    Two B-Channel Transfer
    http://www.cisco.com/en/US/docs/ios/voice/ivr/pre12.3_14_t/configuration/guide/ivrapp.pdf
    Understanding Dial Peers and Call Legs on Cisco IOS Platforms
    http://www.cisco.com/en/US/partner/tech/tk652/tk90/technologies_tech_note09186a008010ae1c.shtml
    Understanding Direct-Inward-Dial (DID) on IOS Voice Digital (T1/E1) Interfaces
    http://www.cisco.com/en/US/partner/tech/tk652/tk653/technologies_tech_note09186a00801142f8.shtml
    Understanding Inbound and Outbound Dial Peers Matching on IOS Platforms
    http://www.cisco.com/en/US/partner/tech/tk652/tk90/technologies_tech_note09186a008010fed1.shtml#prereq
    Voice Translation Rules
    http://www.cisco.com/en/US/partner/tech/tk652/tk90/technologies_tech_note09186a0080325e8e.shtml
    Let me know how you go.
    Thanks again for asking the tuff questions.
    Cheers
    Edson

  • Mails are not going from apex application

    Hi,
    Mails are not going from my apex application. I had done the instance settings by logged in as admin. SMTP port is also connecting. Please help me to solve the issue.

    Hi,
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  • I can not receive messages and i can't send messages also if someone rings me it, it rings at their end and then goes to voice mail but nothing at my end

    i can not receive messages and i can't send messages also if someone rings me it, it rings at their end and then goes to voice mail but my phone doesn't ring

    Talk to your carrier.  Sounds like something is not provisioned right on your account.

  • HT4515 how do I set the number of rings before the caller goes to voice mail ?

    How do i set the number of rings before the coller goes to voice mail on my iphone?
    Thanks you.

    Determined by your carrier, so contact them.

  • Agents line rings only twice before going to voice mail

    I have our queues set up so that if the agent does not answer the phone or if the caller is stuck in the queue the call will transfer to VM. This all works fine except that the agent is telling me that the phone only rings 2 times before the call gets sent to voice mail. I want to increase the number of rings to 4 or so.
    I thought this was a factor of time. Say if the person is on hold for 16 seconds (I think that is equivalent to 4 rings) then kick out to the voice mail. Unfortunately I am looking at the code in the UCCX Editor and cannot seem to find the timing or anything saying "Number of rings" to set a value. I also went through the UCCX GUI but did not find anything. I was going through my books form the class I took on UCCX 7 and cannot find anything on this, though I could have sworn we covered this. But, that was 2 or 3 years ago and I am also on UCCX v8.5 now.
    Anyway, does anyone have any suggestions on where I can look or what I can change to get this working. On a side note, when a call comes to her personal line, it does ring 4 times before going to voice mail. The agent line is the only one that only rings 2 times.
    Thanks for any help and pointers on what to look for.
    David Graham

    I cannot determine what method you are using to technically extend the call to voicemail after the ringing, and therefore couldn't give you any tips to check.  Could you just clarify the call flow for both points:
    1. How do you extend the call to the Agent?
    2. How do you extend the call to Voicemail?
    Typically, when working with Agents and Ringing time, you adjust two settings:
    1. In UCCX Script - Select Resource Step (Or Connect Step if applicable) Timeout value
    2. In CUCM - Directory Number (Or Service Parameter if applicable) Ring No Answer timer
    Depending on what you're doing in the scripting it could be something like a Call Consult Transfer even.  But that's not clear at this time.  Let us know, and we'll get you fixed up.
    Anthony Holloway
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