Third Party SIP & MeetMe numbers

Hello,
Was just wondering if anyone has an elegant way of getting a Third Party SIP phone to initiate a MeetMe?

Well that's not entirely accurate; a Cisco SIP phone can initiate a MeetMe conference.
If you look in the SIP Profile on UCM, I believe there is a setting for this. I've never done it but you'd need to configure the SIP phone to send the xcisco-??? message in the invite probably. Hopefully that'll move you in the right direction at least.

Similar Messages

  • CUCM: Third Party SIP Phone "Caller ID" is not displaying for outgoing calls

    Hi Team,
    we are running CUCM 9.1(2a),
    we have integrated Third Party SIP Phone(Avaya 1230 SIP Phone) with CUCM,
    Issue: Third Party SIP Phone "Caller ID" is not displaying for outgoing calls, we are able to see only the dailed Number,
    When "A" calls to "B", "A" can see only the dailed number of "B" but not the "Caller ID"
    Regards
    Ananthakumar

    Are A and B both Avaya phones?
    So it looks like you're not seeing the alerting name/connected name getting updated then?  Do you have alerting names configured on the directory numbers?  Might need to take a look at the SIP messaging to see if the alerting name/connected name is being sent to the Avaya phones and maybe they just aren't displaying it.  Might just be something that needs to be tweaked in the 46xxsettings.txt file.

  • 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)|*^*^*

  • MOH in third party sip phones

    Hello , 
    I would like to know if the CUCM support MOH in third party sip phones such as x lite or other ? 
    Now I can only hear silent . 
    Thanx 

    Hi ben Zecharia,
     I found your post looking for MoH in 3rd Party SIP Phone and also found another post that said that CUCM 8.x do not support MoH in 3rd Party SIP Phone (check this link).
    Hope this helps (you and others).

  • Video call from Cisco 8945 SCCP to Polycom VVX 1500 Third party Sip Phones between intercluster trunks is not working

    Dear All,
                   We have two cucm Clusters in Different Locations between that clusters i created
    Inter-Cluster Trunk (Non-Gatekeeper Controlled) Now all are working fine Between Clusters
    audio calls & Video calls between sccp 8945 phones  , but iam facing a Problem with third party
    Video Phones (Polycom VVX 1500 ) Third Party SIP Phones located in second cluster, From 1 st cluster cisco 8945 Video
    phone to  2nd cluster Polycom Video phone all calls are works for voice call only, but no video ,
    Please Suggest me Solution.
    Thank you,
    Sriman

    Try setting up a SIP trunk between the two clusters and set a route patten just to the VVX 1500 and check how that goes. 
    From memory inter-cluster trunks are a H.323 like protocol which might have video inter-op issues with the Polycom device.

  • Activate a call forward with a Third-party SIP Device or with a analog device

    Hi,
    In a CUCMv9, how i can activate a call forward (all, busy, no anwser...) with Third-party SIP Device or with a analog device connected to a fxs?
    I want to activate a call forward like a Alcatel or Aastra PBX with a code.
    For exemple, i pick up the phone, with the code *95 followed by the destination number and hangs up the phone. And use the #95 for désactivate this call forward.
    It's possible?
    Thanks.

    No codes for 3rd party SIP phones, no way to do it. Or for that matter, not even for Cisco Phones, other than CFA.
    Anything besides CFA needs to be done via CCMadmin or CCMuser for any kind of phone.
    For FXS that's only doable if you're running SCCP
    http://www.cisco.com/en/US/partner/docs/ios/voice/fxs/configuration/guide/fxssccpsplmft.html
    HTH
    java
    if this helps, please rate
    www.cisco.com/go/pdihelpdesk

  • RTP Media Stream with "Third Party SIP Device" always through CUCM

    Hello,
    i have i quite strage problem on one of my customers locations:
    we have a cucm 7.1.5(SU4) with the cucm in the datacenter. And we have a small location(branch office) which has a small wan connection to the datacenter (1MBit/s).
    In this location we have several Kirk (Polycom) Dect phones which register as "Third Party SIP Device - Basic" on the CUCM.
    (The problem is the same if i use the X-Lite SIP Client instead)
    When this SIP Phones or the X-Lite Client dials a internal Number of the same location the RTP Media Streams goes directly from the SIP Client to the phone. But if they dial an external number the RTP Stream goes from the SIP Client via the wan connection to the CUCM and back via the wan connection to the 2901 H.323 Gateway (on the same location).
    and of course if i now start a big download or upload i'm no longe able to complete the phone call because we have no QOS on the wan connection, because we don't want to make calls over this connection.
    When i look at the Sniffer files with Wireshark i see that the CUCM sends his own ip adresse in the SDP Header for the RTP Stream to the SIP Client. And this of course is wrong because the RTP Stream should always reside on the branch office.
    i tested this in my lab (CUCM 8.5) and it is the same. i used the "Standard SIP Profile" and a Basic Third Party SIP Device"
    The SCCP Phones on the same location which are configured with the same Region, Location, Device Pools, Media Resources and which use the same Gateway for external calls do not have this problem.
    In the Gateway configuration "MTP Required" is not activated and i tested it in my lab with some Cisco SIP Phones (9971) and they are also not affacted with this problem.
    any ideas?

    do you have a SIP trunk to the external devices with MTP required checked ?

  • CUOM with Third-party SIP Device

    Hi everyone,
    I have installed CUOM in environment with CUCM, Cisco IPPhones (SCCP), and IPPhones (Third-party SIP Device), I can monitor CUCM and SCCP IPPhones but the Third-party SIP Device can't be monitored with CUOM, Please someone have a solution for this issue
    Thanks in advance,

    do you have a SIP trunk to the external devices with MTP required checked ?

  • Paging Third Party SIP Phones connected to CUCM

    Current SetUp: CUCM - Cisco 3925, Two MCS 7816 (Call Control Server) and One MCS 7825 (Voice mail server)
    We have third party SIP phones configured in auto answer mode. These phones are used to make live announcements.
    To Do:
    There are approximately 80 phones in the system and the requirement is to select any combination of these phones to make Public announcement (or Paging).
    Is there an application that enables us to select any combination of phones on the fly to do paging? How can we select a mp3 file to play on a phone in an auto answer mode?
    Any help will be appreciated.
    Thanks
    Sid

    There's nothing built into call manager to do this.  You could investigate using the Cisco Unified Application Environment (CUAE) and write a script to do this, or there are some 3rd party applications that might work for you such as Berbee's Informacast.

  • Add third party SIP Phone to CCM 5

    'm not able to register this SIP Phone to the CCM5.0. I have device license that cater all IP Phone models.(LIC-CM-DL-100=)
    I got error message " Login Forbidden" "timeout" in the IP Phone.
    In the CCM, I got this message in Phone COnfig Window
    Registration: Rejected.
    Can you explain on how to register this 3rd party IP phone to CCM?
    Is it CCM able to support SIP Phone?

    Hi,
    This is most likely because of the following...
    Because third-party SIP phones do not send a MAC address, they must identify themselves by using digest authentication.
    The REGISTER message includes the following header:
    Authorization: Digest username="swhite",realm="ccmsipline",nonce="GBauADss2qoWr6k9y3hGGVDAqnLfoLk5",uri="sip:172.18.197.224",algorithm=MD5,response="126c0643a4923359ab59d4f53494552e"
    The username, swhite, must match an end user that is configured in the End User Configuration window of Cisco Unified CallManager Administration. The administrator configures the SIP third-party phone with the user; for example, swhite, in the Digest User field of Phone Configuration window.
    See the following document.
    http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/5_1_3/ccmcfg/b09sip3p.html
    Hope this helps, if so please rate.
    Regards,
    Dave

  • Third Party SIP Phone Alerting name

    Hello,
    We are having cisco ip phones & third party sip phones in our company. both are registered to same cucm 9.X.
    Now when cisco phones calls sip Phones. we are able to see the alerting name on cisco phone.
    Say, Cisco Phone "Phone-A" calls Third Party Sip phone "Phone-B". So, on cisco ip Phone display we are able to see "Phone-B".
    But same when we try from SIP phones to Cisco phones, we are unable to see the Alert name on the Sip Phone. only the Number we can see on the Sip phone.
    Any help will be highly appriciated.
    Thanks,

    Hi Amod.
    I'm terribly sorry cause I got what you are asking only now :(
    This behaviour depends on client capability.
    CUCM always send RPID ( remote-party ID) to the third party SIP client/phone on ring.
    If the client is capable to update the calling number into what it receives as RPID, than you'll be able to see calling name.
    I've read some release notes of XLITE and other  SIP Desk phone and it seems to be not mentioned.
    Sorry again
    Regards
    Carlo

  • MOH for third party sip phones

    Hello , 
    I using CUCM version 9.1.1.2000-5  . 
    Does this version support MOH for third party sip phones ? 
    Thank you 

    Hi,
    I couldn't found better piece of information which list all cases how MoH is implemented for various terminals and hence thought of testing the same. My observations;
    SCCP phone -> Hold -> SIP Phone -> MoH plays to SIP phone
    SIP Phone -> Hold -> SCCP phone - > Doesn't play MoH
    SIP Phone -> Hold -> SIP phone -> Doesn't play MoH
    Please note that I have checked with both Xlite and 3CX, results are same.
    I have verified in wireshark also,  call manager is not sending RTP packets to held party when call is hold by third party SIP phone.
    Checked in CUCM 9.1
    Thanks
    Vivek

  • Are CMBE 3000 support third party SIP phones?

    Hi everybody!
    Are CMBE 3000 support third party SIP phones?

    Hi Yang,
    The typical IP phone today costs the same or less than an equivalent digital desk phone set. When you factor in the lower overall total cost of ownership (TCO) that results from an IP Communications solution running on a converged IP network for voice, video, and data, an IP based solution can save a company a substantial amount of money in the medium to long term.
    Thanks

  • CM5 support the third party SIP phone?

    Does CCM5 support the third party SIP phone? what manufacturer? all manufacturer?
    Does IP phone support the third party SIP server? what manufacturer? all manufacturer?

    thanks for ananddiwakar.
    I agree your the first answer: CCM5.0 support the third party IP phone, but be some limitations with the features.
    But for cisco sip phone supporting standard SIP server, I'm confused that because the cisco SIP phone need download Firmware from SIP server, and for the third party SIP server, is the mechanism of downloading the Firmware from the third party SIP server right?

  • Third party SIP Endpoints in BE6K

    Hi
    I have a doubt about BE6K,  Is this solutions support Third part SIP Endpoints, like grandstream or polycom? And if is true, What license support thjis pohone?
    Best regards

    Third Party ATA is another story... But a third Party IP phone will work fine.
    I have a Polycom conference register with our BE6K.
    Just need to find out witch phone model config to push down the GrandStream or the Polycom.
    Good Luck

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