Cisco SPA 112 outbound call issue.

Firstly, i applogize if i havent posted this in the correct section but this was my best educated guess.
Recently upgraded to a Cisco SPA 112 which works fine.. but only for a few days,
then If i try to call a number i get the dial tone but the number fails to
connect & all i hear is silence.  If i hang up the phone to retry it fails to
disconnect correctly and i get no dial tone for several minutes. After leaving the phone
hungup for a few mins the dail tone returns but the same problem is still present.
A hard/soft reset of the 112 fixes the issue temporarily, but reoccures again withinn a few days.
  Running Sipgate on lines 1 and 2 on Virgin media UK ISP.
Model:
SPA112, 2 FXS
Hardware Version:
1.0.0
Boot Version:
1.0.1 (Oct  6 2011 - 20:04:00)
Firmware Version:
1.3.1 (003) Dec 17 2012
Recovery Firmware:
1.0.2 (001)
Im no expert and now at a loss and could do with some expert help tbh.

Dan, i will see how it goes over the next week or so with the updated F/W & shall report back, and thanks for the link as i wasnt sure how to save a syslog.
Should i set the verb as high as 6, or is that an over kill.
Gabriel, I have an FTP running on my sat receiver which i will attempt to use to catch the log, but as for the the "sipgate registration", when i loose conection my ata reports its still registered & the sipgate URL, isnt in real time as far as i can tell.
My ata would have normally shown its fault since my first post 6 days ago, but unfortunately a recent power cut has reset my testing period back to 1 day.

Similar Messages

  • Cisco SPA 112 adapter Issue

    Hi All,
    We are using Cisco SPA 112 adapter and it is having 2 lines,I configured both of the lines.
    There is no issue in line 1 but line 2 is automatically went to disable mode.After that i have to manually enabled that option.
    Please help me to fix this issue

    Dan, i will see how it goes over the next week or so with the updated F/W & shall report back, and thanks for the link as i wasnt sure how to save a syslog.
    Should i set the verb as high as 6, or is that an over kill.
    Gabriel, I have an FTP running on my sat receiver which i will attempt to use to catch the log, but as for the the "sipgate registration", when i loose conection my ata reports its still registered & the sipgate URL, isnt in real time as far as i can tell.
    My ata would have normally shown its fault since my first post 6 days ago, but unfortunately a recent power cut has reset my testing period back to 1 day.

  • Weird Outbound calling issue

    Lync server 2010/ Lync 2010 client / Audiocodes Gateway and Avaya PBX.
    Lync infra is all up to date.   Not sure where the issue is related but here we go:
    User exist in AD, but is not configure for Lync.  User had a DID.
    From a  lync client 2010 :
    If I dial the user extension (3307)  it translate the number to DID :  +1xxx6623307  All good
    If I dial the DID : xxx6623307  all is good
    If I dial the DID with the 1 :   1xxx6623307 all is good
    but If I select the WORK phone number in Lync 2010 :  You can see in the screen shoot, it will try +1xxx6623307 x3307
    And the CALL will fail.
    In AD, user WORK number attribute is configured with only the 4 digit extension : 3307
    Any hints?
    regards

    It seems a synchronization issue.
    What is your normalization rule in Company_Phone_Number_Normalization_Rules.txt for extersion 3307?
    Check which phone number is the right number should be normalized first.
    Lisa Zheng
    TechNet Community Support

  • Use Cisco CUCM for outbound "call me at" feature on Lync meetings

    I'm trying to find a step by step to enable users (non enterprise voice users) to use the dial me at feature in Lync conference meetings. I only want the user to have the ability to tell Lync to dial a number to place that number into the conference call,
    the feature is easy to enable but i can't get the routing right between CUCM and Lync. I've looked all around the net but I can't seem to find anything that matches what i'm trying to do, other docs cover enterprise voice and that's out of my scope. Any assistance
    here would be nice. Thanks

    Hi,
    In Lync Server 2010, it is not supported with “call me at” function for non-Enterprise Voice users.
    However, Lync Server 2013 now allows participants that are not Enterprise Voice enabled to initiate dial-out calls from a meeting conference, called “Dial-out Conferencing for non-Enterprise Voice users”.
    This can be configured by setting the Conferencing policy to allow this feature (Set-CSConferencingPolicy –AllowNonEnterpriseVoiceUsersToDialOut:$true). After enabling this, then assign a voice policy to the users who need the function.
    Best Regards,
    Eason Huang
    Please remember to mark the replies as answers if they help, and unmark the answers if they provide no help. If you have feedback for TechNet Support, contact [email protected]
    Eason Huang
    TechNet Community Support

  • SPA 112/122 FW 1.1.0 (011) provisioning problem

    I have problem with provisionig Cisco SPA 112/122 after upgrade FW to 1.1.0 (011).
    Is there any solution or just we should not use this FW?
    We use provisioning by entering url to
    Profile Rule:   http://prov.802.cz/prov.xml
    set short times and Submit, but nothing happens :-(
    After downgrade FW to 1.0.1 (022) provisionign work with no problem.
    Thanks for your advice
    Tomas

    We have identified some provisioning issues and those are fixed in the upcoming release which should be out on cisco.com in the next 2 weeks.  Thank you.

  • Cisco Jabber for Windows in Extend and Connect mode and making outbound calls

    Hi guys,
    I've set up Cisco Jabber for Windows to use Extend and Connect to control a remote PBX endpoint. I've configured the required CTI-RD device, remote destinations, associated the users to the line and added the devices to end-user controlled device. The extend and connect part is working flawlessly without any issues. I'm able to receive inbound calls on the remote PBX endpoint and control the call (hold, resume, transfer etc.) using the Jabber call window that pops up.
    However, I'm unable to make any outbound calls via the Jabber client when in extend and Connect mode. Reading the Extend and Connect guide, I need to configure Dial Via Office (DVO) Reverse. So when the user initiates a Dial-Via-Office reverse call, CUCM calls and connect to the Extend and Connect device (CTI-RD). CUCM then calls and connects to the number the user dialled and finally connects the two call legs.
    After attempting to configure DVO-R for Jabber for Windows in Extend and Connect mode following the CUCM feature services guide, i'm unable to get any outbound calls working. From RTMT, i am receiving the following Termination Cause Code: (27) Destination out of order. What i also notice is that there is no calling number for that trace either. I would've thought that the calling party would've been the Enterprise Feature Access (EFA) number.
    Has anyone got this working or can provide some guidance?
    Thanks.

    Hi guys,
    I've set up Cisco Jabber for Windows to use Extend and Connect to control a remote PBX endpoint. I've configured the required CTI-RD device, remote destinations, associated the users to the line and added the devices to end-user controlled device. The extend and connect part is working flawlessly without any issues. I'm able to receive inbound calls on the remote PBX endpoint and control the call (hold, resume, transfer etc.) using the Jabber call window that pops up.
    However, I'm unable to make any outbound calls via the Jabber client when in extend and Connect mode. Reading the Extend and Connect guide, I need to configure Dial Via Office (DVO) Reverse. So when the user initiates a Dial-Via-Office reverse call, CUCM calls and connect to the Extend and Connect device (CTI-RD). CUCM then calls and connects to the number the user dialled and finally connects the two call legs.
    After attempting to configure DVO-R for Jabber for Windows in Extend and Connect mode following the CUCM feature services guide, i'm unable to get any outbound calls working. From RTMT, i am receiving the following Termination Cause Code: (27) Destination out of order. What i also notice is that there is no calling number for that trace either. I would've thought that the calling party would've been the Enterprise Feature Access (EFA) number.
    Has anyone got this working or can provide some guidance?
    Thanks.

  • Site to Site calling issue - Cisco 2911 Dial Peer Configuration

    My customer dials from remote site to main site to their main site number, the call by-passes their auto attendant and goes directly to any random available party. 
    At first fingers were pointing to the their PBX, however we noticed one of their sites that wasn't managed by our company did not have the issue.   We cut that site over to our service and the issue started right up.  I believe it is possibly due to the way the dial peers are configured and how the calls route into the PBX.  Unfortunately I do not understand much about them and curious to know if anyone has any history on a issue similiar to this or any input whatsoever?
    Cisco equipment/Dialpeer config below ........
    co IOS Software, C2900 Software (C2900-UNIVERSALK9-M), Version 15.2(4)M4, RELEASE SOFTWARE (fc2) - Cisco CISCO2911/K9
    dial-peer voice 100 voip
     description --- VoIP Dial-Peer ---
     translation-profile outgoing 7digit
     huntstop
     preference 1
     service session
     destination-pattern .T
     progress_ind setup enable 3
     session protocol sipv2
     session target sip-server
     incoming called-number .T
     voice-class codec 99  
     dtmf-relay rtp-nte
     fax-relay ecm disable
     fax rate 14400
     fax nsf 000000
     ip qos dscp af41 signaling
     no vad
    dial-peer voice 150 voip
     permission none
     description 900 block
     huntstop
     destination-pattern 1900T
     session protocol sipv2
     session target sip-server
     voice-class codec 99  
     dtmf-relay rtp-nte
     ip qos dscp af41 signaling
     no vad
    dial-peer voice 151 voip
     permission none
     description 900 block
     huntstop
     destination-pattern 900T
     session protocol sipv2
     session target sip-server
     voice-class codec 99  
     dtmf-relay rtp-nte
     ip qos dscp af41 signaling
     no vad
    dial-peer voice 101 pots
     description --- INCOMING Calls from PBX ---
     incoming called-number .T
     direct-inward-dial
    dial-peer voice 1001 pots
     description --- Calls to the PBX ---
     preference 3
     destination-pattern .T
     port 0/0/1:23
     forward-digits 4
    Here is some ISDN debug information
    BAD CALL
    Protocol Profile = Networking Extensions
    0xA11C0201420201008014484152545F20484F54454C535F434C4159544F4E
    Component = Invoke component
    Invoke Id = 66
    Operation = CallingName
    Name Presentation Allowed Extended
    Name = XXXXXXXXXXX
    Display i = ''XXXXXXXXXXX''
    Calling Party Number i = 0x2180, ''XXXXXXXXXX''
    Plan:ISDN, Type:National
    Called Party Number i = 0x80, ''6551''
    Plan:Unknown, Type:Unknown
    Aug 19 16:10:47.242 GMT: ISDN Se0/0/1:23 Q931: RX <- ALERTING pd = 8 callref = 0xAB15
    Channel ID i = 0xA98381
    Exclusive, Channel 1
    Aug 19 16:11:02.634 GMT: ISDN Se0/0/1:23 Q931: RX <- CONNECT pd = 8 callref = 0xAB15
    Channel ID i = 0xA98381
    Exclusive, Channel 1
    Aug 19 16:11:02.634 GMT: ISDN Se0/0/1:23 Q931: TX -> CONNECT_ACK pd = 8 callref = 0x2B15
    GOOD CALL
    Protocol Profile = Networking Extensions
    0xA116020144020100800E475245454E204D4F554E5441494E
    Component = Invoke component
    Invoke Id = 68
    Operation = CallingName
    Name Presentation Allowed Extended
    Name = XXXXXXXXXXXXXXXXXX
    Display i = ''XXXXXXXXXXX''
    Calling Party Number i = 0x2180, ''XXXXXXXXXX''
    Plan:ISDN, Type:National
    Called Party Number i = 0x80, 'XXXX''
    Plan:Unknown, Type:Unknown
    Aug 19 16:15:07.999 GMT: ISDN Se0/0/1:23 Q931: RX <- ALERTING pd = 8 callref = 0xAB17
    Channel ID i = 0xA98381
    Exclusive, Channel 1

    I done the configration via CCA  and the running conf i can see two voip dial peer. this is the site where all trunk line roured. Customer from other site2 needs to call outside by taking line from site1.
    dial-peer voice 2100 voip
    corlist incoming call-internal
    description **CCA*INTERSITE inbound call to SITE 1
    translation-profile incoming multisiteInbound
    incoming called-number 82...
    voice-class h323 1
    dtmf-relay h245-alphanumeric
    fax protocol cisco
    no vad
    dial-peer voice 2101 voip
    corlist incoming call-internal
    description **CCA*INTERSITE outbound calls to SITE2
    translation-profile outgoing multisiteOutbound
    destination-pattern 81...
    session target ipv4:192.168.50.1
    voice-class h323 1
    dtmf-relay h245-alphanumeric
    fax protocol cisco
    no vad
    no dial-peer outbound status-check pots

  • Cisco Mobile 8.1 w/iPhone 4 - Outbound Call Fails

    Hello Everyone,
    Using CUCM 7.1.5 and Cisco Mobile iPhone app, I set up my boss's iPhone 4 and it registers and can receive calls. I am getting a "Call Failed" when trying to make an outbound call using internal 5-digit dialing. Has anyone set this up? Is it necessary to configure these Application Dialing Rules (and install the COP file for them) that are referenced in the Cisco Mobile Admin Guide? I interpretted those rules as optional, if you want to set up things like bypassing dialing 9 for PSTN access, etc. Shouldn't the internal dialing work since the phone is registered?
    Grateful for any help,
    Kelly

    No, you do not need Application Dial Rules for Iphone to IPPhone calls. ADR are for
    like you said to add a 9, remove the + and manipluate digits.
    I suggest you open a case with the TAC to help you see why you cannot call. Maybe issue
    with ASA or something.
    George

  • SPA 112 PROBLEM WITH SIMULTANEOUS CALLS

    HELLO,I HAVE A PROBLEM IN THE ATA SPA 112 ABOUT SIMLUTANEOUS CALLS. The ata has 2 lines,when a call falls in  line 1, the call on line 2 falls at the same time...and when I'm on line 1 and I recevive a call on line 2 the call in line 1 becomes mute and then drops.I didn' have these problems with the old pap2t .
    Can  you please help me to solve this problem ? Do I have to change something in the configuration ? Thanks,Mario

    Hi,
    thanks for responding.
    I had found out about the timing problem in the meantime, but did not find a way to mark this thread as solved.
    There are two timing values: one is the PSTN answer delay, and the other one the PSTN ring timeout.
    It seems that ring timeout should be longer than ring time + ring pause. As long as this conndition is not met. the dial plan is not even considered

  • Outbound Call in Cisco Finesse

    Anybody please help what all configuration to be done in Finesse to do the outbound calls.
    I am using Cisco UCCX 10.0, in which i have both the option for Cisco CAD and Finesse. When i run a outbound campaign its coming as an incoming call in Finesse.
    When i answer the call, its showing the error "Unable to communicate with Enterprise Server. Outbound option not available".
    Please help how to enable the outbound controls in Finesse

    This is ineed the case, from RN (somewhat cryptic :-) ):
    Cisco Finesse
    Cisco Finesse is the next generation browser-based agent and supervisor desktop for Unified CCX. Finesse is an alternative to Cisco Agent Desktop, Cisco Supervisor Desktop, and Cisco Desktop Administrator. Finesse is available with Enhanced and Premium license packages and provides typical inbound voice contact center functionality. It supports Unified Communications Manager-based silent monitoring and workflow-based recording with MediaSense and Work Force Optimization (WFO).
    Chris

  • Does Cisco UC 500 Support Call Center Outbound Calls?

    does cisco uc 500 support call center outbound calls ?

    I have developed a "Campaign Dialer" solution for UC500 and CME. More details are available on the website referenced in my profile.

  • Incoming calls issue in Third Party SIP Phone

    Hi,
    Yesterday I configured my third party sip phone which is yealink in this case on cucm and successfully registered it with cucm, despite of registration i have some calling issue in this phone. I am able to make outbound calls from this phone to any other phone however issue is related to inbound calls.I tried calling its DN from anywhere but call disconnect after sometime. Also didnt get any proper sip session trace in RTMT. Kindly suggest some step to sortout this issue.
    Thanks

    Dear Manish,
    Call normally dicsonnected after 30-40 sec with termination code 102 in session trace. PFB SDI  trace with 5030 is Thirdparty sip phone and 5033 is c7945. Looking forward for your suggestion.
    CallingPartyNumber=5033
    |DialingPartition=
    |DialingPattern=5030
    |FullyQualifiedCalledPartyNumber=5030
    |DialingPatternRegularExpression=(5030)
    |DialingWhere=
    |PatternType=Enterprise
    |PotentialMatches=NoPotentialMatchesExist
    |DialingSdlProcessId=(0,0,0)
    |PretransformDigitString=5030
    |PretransformTagsList=SUBSCRIBER
    |PretransformPositionalMatchList=5030
    |CollectedDigits=5030
    |UnconsumedDigits=
    |TagsList=SUBSCRIBER
    |PositionalMatchList=5030
    |VoiceMailbox=
    |VoiceMailCallingSearchSpace=PT-LHR-LOCAL:PT-Local:Unityvmpt:PT-F6-Local:PT-ISL-LOCAL:PT-KHI-LOCAL:PT_Operator_LHR:PT_Operator_KHI:PT_Operator_ISL
    |VoiceMailPilotNumber=7103
    |RouteBlockFlag=RouteThisPattern
    |RouteBlockCause=0
    |AlertingName=Syed Ahmer
    |UnicodeDisplayName=Syed Ahmer
    |DisplayNameLocale=1
    |OverlapSendingFlagEnabled=0
    12:17:38.028 |//SIP/SIPUdp/wait_SdlSPISignal: Outgoing SIP UDP message to 172.16.200.21:[5062]:
    [23928282,NET]
    INVITE sip:[email protected]:5062 SIP/2.0
    Via: SIP/2.0/UDP 10.100.200.11:5060;branch=z9hG4bK1ca0cc6e317649
    From: "Syed Ahmer" ;tag=8787406~039e2a80-8561-4586-8954-d01ed2aa12c8-246211918
    To:
    Date: Thu, 30 Jan 2014 07:17:38 GMT
    Call-ID: [email protected]
    Supported: timer,resource-priority,replaces
    Min-SE:  1800
    User-Agent: Cisco-CUCM8.5
    Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
    CSeq: 101 INVITE
    Expires: 180
    Allow-Events: presence
    Send-Info: conference, x-cisco-conference
    Alert-Info:
    Contact:
    Remote-Party-ID: "Syed Ahmer" ;party=calling;screen=yes;privacy=off
    Max-Forwards: 70
    Content-Length: 0
    |14,100,50,1.14103336^10.163.14.4^SEP00230432C828
    12:17:38.028 |EnvProcessUdpPort - EnvProcessUdpHandler::fireSignal() varId = 0|14,100,50,1.14103336^10.163.14.4^SEP00230432C828
    12:17:38.028 |EnvProcessUdpHandler::fireSignal - SEND: index = 0, handler = 0xaf299320|*^*^*
    12:17:38.028 |EnvProcessUdpPort::fireSignal - SEND, destination = 172.16.200.21:5062|*^*^*
    12:17:38.028 |EnvProcessUdpPort - EnvProcessUdpHandler::send(buff, 850, 172.16.200.21:5062)|*^*^*

  • CME - Sending outbound calls to FXO port

    Hi Guys,
    Need your help for the below scenario.
     Our customer has a CME where 4 FXO ports are already connected and working. Customer has added 2 more FXO port and few IP phones.
    The requirement is when ever an outbound call is made from the newly configured IP phones, the call should go through the newly added FXO lines.
    For eg ext 3001 , the outbound call should go through port 0/1/0
    Already the prefix 9 is used for dialing the number and I guess only one prefix number can be used in CME.
    I tried translation rule , cor list but none worked , the call is default going through the old fxo port and not to the new fxo port.
    Can you guys help me with the configuration.
    Regards
    Sathya

    Previous post on similar issue might be helpful - 
    https://supportforums.cisco.com/discussion/11431746/h323-choose-outbound-fxo-port-based-calling-number
    Thnx

  • Make a Cisco SPA 303 ring by sending a packet through your network?

    Hey Guys,
    I was wondering, and I need to know for my business, is there any way at all for me to make my Cisco SPA 303 VOIP Phone to ring by sending a packet through my local network?
    I would like to just be able to click a button or send a command throught the command prompt and make it ring, but I don't know if there is any way for this to happen.
    Thank!

    Do you know perl?
    I had same issue and I wrote a simple perl script that works as wake up service.
    PERL is an interpreted language and so can be executed on Linux and  Windows operating systems. Linux can interpret perl natively while for  Windows you can download many free interpreters like Activeperl or  Strawberry perl. To run the script you must use a third party server.
    In my configuration the script runs on a linux server in background as a service and checks every minute the  directory called "alarm", reads files and uses the file name as called  number and checks the content to verify if is the time to call. At the moment the script uses SIP and handles 4 call responses: 404 user  not found, 486 busy, 487 not answer and 200 answer ok.  In every cases sends an email and deletes files. Only for the answer  case plays a nice music.
    Files have this particular format: file name is equal to calling  party number and file content is the alarm time in 24 hours format  with : as separator between hours and minutes.
    e.g.
    ext. 101 must be called at 8 am ---> write the file 101.txt with the content 08:00
    ext. 101 must be called at 8:30 am ---> write the file 101.txt with the content 08:30
    There is a limitation: if you activate the Authentication for SIP messages and there are more  then two simultaneous calls, the script sends some INVITEs without  authentication or with wrong checksum and so not all phones ring. This  problem is under investigation.
    Are you intresting?
    Regards.

  • Error messages in 2651XM GW, cause outbound call failure, reboot fix it

    Cisco 2651XM as Gateway, it keep posting these error message and after a period of time, it cause outbound call failure.
    Reboot fix it but there're still error messages...
    How to fix it? It's IOS bug or hardware issue? How to identify?
    Cisco IOS Software, C2600 Software (C2600-IPVOICE-M), Version 12.3(8)T10, RELEASE SOFTWARE (fc2)
    Technical Support: http://www.cisco.com/techsupport
    Copyright (c) 1986-2005 by Cisco Systems, Inc.
    Compiled Wed 03-Aug-05 20:45 by hqluong
    ROM: System Bootstrap, Version 12.2(7r) [cmong 7r], RELEASE SOFTWARE (fc1)
    cpchn1-g1 uptime is 6 hours, 56 minutes
    System returned to ROM by reload at 03:52:44 NZST Tue Apr 17 2007
    System restarted at 03:56:27 NZST Tue Apr 17 2007
    System image file is "flash:c2600-ipvoice-mz.123-8.T10.bin"
    Cisco 2651XM (MPC860P) processor (revision 0x100) with 118784K/12288K bytes of memory.
    Processor board ID JAE072000AJ (1555074759)
    M860 processor: part number 5, mask 2
    2 FastEthernet interfaces
    62 Serial interfaces
    2 Channelized E1/PRI ports
    32K bytes of NVRAM.
    32768K bytes of processor board System flash (Read/Write)
    See attach detail error messages

    Cisco 2651XM as Gateway, it keep posting these error message and after a period of time, it cause outbound call failure.
    Reboot fix it but there're still error messages...
    How to fix it? It's IOS bug or hardware issue? How to identify?
    Cisco IOS Software, C2600 Software (C2600-IPVOICE-M), Version 12.3(8)T10, RELEASE SOFTWARE (fc2)
    Technical Support: http://www.cisco.com/techsupport
    Copyright (c) 1986-2005 by Cisco Systems, Inc.
    Compiled Wed 03-Aug-05 20:45 by hqluong
    ROM: System Bootstrap, Version 12.2(7r) [cmong 7r], RELEASE SOFTWARE (fc1)
    cpchn1-g1 uptime is 6 hours, 56 minutes
    System returned to ROM by reload at 03:52:44 NZST Tue Apr 17 2007
    System restarted at 03:56:27 NZST Tue Apr 17 2007
    System image file is "flash:c2600-ipvoice-mz.123-8.T10.bin"
    Cisco 2651XM (MPC860P) processor (revision 0x100) with 118784K/12288K bytes of memory.
    Processor board ID JAE072000AJ (1555074759)
    M860 processor: part number 5, mask 2
    2 FastEthernet interfaces
    62 Serial interfaces
    2 Channelized E1/PRI ports
    32K bytes of NVRAM.
    32768K bytes of processor board System flash (Read/Write)
    See attach detail error messages

Maybe you are looking for