CallManager Express SIP trunk problem

Hi all here.

I have a problem with my SIP trunk CallManager Express (version 10.0). On my site already configured trunk SIP between CME and CUE. I have configured DN for SIP phone and SIP phones recorded on CME successfully, but... As soon as SIP phone registered on CME, DIAL-PEER for CME and CUE connection changes. Especially 'session target ipv4' automatically changes to my IP phone SIP and all calls go to my SIP phone and does not reach the auto attendant.

How can I solve this problem?

Advertisement

Hello

Following is the error of CUCM;

WARNING: 399 UnicompCM "cannot find a Device Manager for the request received on port 55549 leave 192.168.10.2.

This probably means that CME use 192.168.10.2 for source packages SIP CUCM however SIP trunk in CUCM is not directed to this IP address, it is useful to point out to another IP interface of the GUY and therefore dismiss this appeal.

Can you please check what is the IP of CMF address you configured on the SIP trunk in CUCM?

-Vivek

Tags: Cisco Support

Similar Questions

  • DTMF in SIP Trunk problem

    Hello

    I have a problem in case of detection of the DTMF

    We have a SIP of the ITSP Trunk and everything is ok except DTMF.

    The sip trunk is between ITSP and router 3945

    ITSP <->3945 <->CUCM 10.5

    I have try all method such as rtp - nte, h245 alphanumeric and h245-signal, info-sip, sip-kpml,... in line-peer to ITSPs

    ITSP say that he send with standard RFC2833 dtmf (it equals to rtp-net, I know), but when I am 'debug ccsip message', the results show the dtmf with rfc2833 sends us

    16 August 13:52:28.991: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received:
    INVITE sip: [email protected] / * /: 5060. user = phone SIP/2.0
    Via: SIP/2.0/UDP 10.105.40.34:5060; branch = z9hG4bKejhh8jixkobhnb7ykvi87vuj8
    Call ID: [email protected]/ * /.
    From: sip: [email protected] / * />; tag = c3cx3vcw-CC-40
    To: sip: [email protected] / * /; user = phone >
    CSeq: 1 INVITE
    Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REGISTRY, PRACK, INFO SUBSCRIBE, NOTIFY, updates, MESSAGE, see
    Max-Forwards: 69
    Supported: 100rel, timer
    User-Agent: Huawei SoftX3000 V300R010
    Session time-out: 300
    Min - SE: 90
    Contact: sip: [email protected] / * /: 5060; user = phone >
    Content-Length: 374
    Content-Type: application/sdp
    v = 0
    o = HuaweiSoftX3000 11042757 11042757 IN IP4 10.105.40.34
    s = call Sip
    c = IN IP4 10.105.40.34
    t = 0 0
    m = audio 27762 RTP / AVP 8 0 18 4 2 98 99 102
    a = rtpmap:8 PCMA/8000
    a = rtpmap:0 PCMU/8000
    a G729/8000 rtpmap:18 =
    a = rtpmap:4 G723/8000
    a = rtpmap:2 G726-32/8000
    a = rtpmap:98 G726-40/8000
    a = rtpmap:99 G726-32/8000
    a = rtpmap:102 G726-24/8000
    a = ptime:20
    a = fmtp:18 annex b = No.
    It is a message to guest (with sdp) of ITSP
    As you can see the line with red color must have a code with number of 101 but rather a code with number of 18
    In my "ccsip media debug' output show that the method of negotiation between me and itsp 'incoming voice. '
    It's my router config:
    voip phone service
    No IP trust to authenticate
    allow connections h323 to SIP
    allow connections sip h323
    allow sip to sip connections
    SIP
    interface FastEthernet0/0/1 source control binding
    bind media source interface FastEthernet0/0/1
    min - to 300 session expires-300
    !

    Dial-peer voice 2 voip---> router CUCM and vice versa
    translation-profile outgoing toos
    destination-model 42584...
    session protocol sipv2
    session target ipv4:10.20.30.70
    Codec g711ulaw
    DTMF-relay rtp - nte
    !
    VoIP voice 10 Dial - peer---> router for ITSP and vice versa
    destination-model. T
    session protocol sipv2
    session target ipv4:10.105.40.34
    incoming called-number. T
    DTMF-relay rtp - nte
    Codec g711ulaw
    I have configured cucm with a sip section to my favorite router with active PSG and RFC2833
    BUT HE DIDN'T THERE WAS NO DETECTION OF DTMF IN INCOMING CALLS AND OUTGOING
    I even test dial-position 10 without any method of dtmf-relay because I want to configure it with the incoming voice method, but it does not work
    I change the codec but does not solve the problem
    There is an interesting point and that is, if use Elastix 3945 of the router and configure dtmf among elastix and itsp as "inbound" method and method dtmf among elastix and cucm as 'rfc2833' everything is OK (ITSP<--Inbound--> Elastix <--rfc2833-->CUCM)
    Please give me a solution to solve the problem between Cisco 3945 and ITSP
    Concerning

    It would be more accurate to say that the problem has been bypassed by using a transcoding resource. The correct resolution here would be to open a ticket to trouble with the ITSP. I agree that the PROMPT message you show doesn't include advertising RFC2833 in the SDP offer. It is a problem that only the carrier can rectify.

  • Problem with a SIP Trunk between VCS and CUCM

    Hello world

    I created a SIP Trunk between control VCS (X7.1) and a CUCM 8.6.

    I followed this Cisco_VCS_Cisco_Unified_Communications_Manager_Deployment_Guide_CUCM_6-1_7_8_and_X7 - 0.pdf deployment guide.

    The trunk is in place, but I have the following problem: I can ring since an EX90 recorded on the VCS for a registered on the CUCM 9951 but when I take the call, I have a busy signals (audio and video). When I on the EX90 9951 ring, I have the following message on VC: can not reach the contact.

    I have a 503 service unavailable status on the call history and "call rejected" on the event log.

    My transformation is OK and the call is sent to my neighbor CUCM zone

    You have an idea to solve this problem?

    Thanks for your help

    Best regards

    Bruno Z.

    Hello

    Yes, it's true! It seems at first glance a problem with codec negotiation when I see the reason 47 code.

    in any case set the region to G.711 trunk phone in case if you have different device pool and the trunk region and ip-phones.

    make G.711 end to end and test again.

    Thank you

    Alok

  • secure sip trunk cucm driver

    Hello

    I have set up non-secure sip trunk using tcp of cucm to the driver

    It's up on top of the cucm but in the driver call control is inaccessible

    so the Conference made from an endpoint works but not the scheduled conference or composed auo

    also, I set up a secure safe using tls

    This time sip trunk is down in the side cucm and so it's on the driver side

    I tried to download the temporary driver's certificate in the callmanager-trust and the tomcast-trust

    Download the certificate of call manager to the conductor

    but still does not work

    I don't know if I need to generate a certificate authority certificates or just, I missed something?

    CUCM: 10.5.2

    conductor: XC4.0

    Thank you

    Hello

    I recently deployed the CUCM and conductor (same version) as mentioned above.

    My case was a little different as my certificate CUCM management was made by the internal certification authority, similarly to the conductor, I did the management of certificates, made sure the root certification authority is present in the two server to trust each other.

    So I configured the location with port 5061 with ip address CM on the conductor.

    Similarly on CM SIP trunk pointing to ad-hoc conductor and meeting woth port 5060 and security profile, device security mode an encrypted and the subject name X.509 to match the FQDN or Cluster COMPLETE domain name to match the domain the driver's FULL name, which allows the TLS communication code.

    If I remember I was too faced the same problem when you face, however, after appropriate management of certificates and security profile, it has been resolved properly.

    This guide should be useful

    http://www.Cisco.com/c/dam/en/us/TD/docs/Telepresence/infrastructure/con...

    Please let me know if you need more help.

    Kind regards

    RACLOT

  • SIP trunk behind a router using NAT

    Hello

    Is it possible to use a SIP trunk to a provider SIP ITSP having the CUBE / router gateway behind a firewall using a NAT?

    Does anyone do this?

    I ask because I'm having problems to make my SIP trunk to work and my router for cube is behind my generic service provider router, which makes the NAT. I just want to rule this out as a problem.

    Has anyone else done this? Or is it really impossible?

    Thank you very much

    Tom

    Hello

    As NAT works fine SIP would work properly as the Protocol.

    Here is the RFC for "NAT Traversal practices for Client - Server SIP"

    https://Tools.ietf.org/html/rfc6314

    HTH

    JB

  • Source IP address of the originating call to sip trunk

    I'm unable to set address ip source from trunk sip call. I read somewhere that the source ip address of the call must be the ip address that call device registered to. I also checked "run in all active nodes such as suggested by cisco. We have 3 Sub cluster and 1 pub however still call orgianate to a particular Sub.

    I also undergoes several changes i - e DP and list route but think not worked for me. Can someone help to sortout this problem?

    Let's say if your ip phone registered in sub - and "run on all nodes" enabled, then the source ip address is always sup - A s ip address regardless of the device pool config on the sip trunk.

    Could please explain you the device pool config in phones IP & SIP trunk?

    you have enabled the "run on all nodes" for a list of course also?

    Suresh

    Please note all useful messages.

  • SPAN and SIP Trunk recording in parallel

    I'm looking to get away using a dictaphone SPAN and use SIP automatic trunk of the record (by using a recorder DMS Verint pool UDP) calls.

    My question is, if I apply SIP trunk recording simultaneously with recording SPAN, this mind? I need to make a CEP of the solution, but cannot stop the current recording.

    Thanks for any help you can give me!

    With the traditional port SPAN record in the Callmanager ignores the phone is registered, so when you activate the current record there will be no impact.

    However you can finish by double flow at the end of voice recording, so not sure what verint would do that.

  • SIP trunk CUBE with Callcentric - incoming unanswered call

    I'm doing some tests with a Sip trunk with a provider called Callcentric.
    It is a CUBE scenario. I use a SIP to the CUCM trunk.

    I have a Cisco Callmanager and a virtual router 7200 (GNS3) as a gateway (C7200-ADVENTERPRISEK9-M), Version 12.4 (24).

    A CPIC connected to Callmanager, I call out to PSTN and it works perfectly.

    When I do an incoming call to the PSTN to the softphone destination, she sounds, but when I press the answer button, the call is not connected and the analog phone in RTC can listen back ringtone.

    Do you have any idea what it could be?
     

    Some relevant configurations:
     

    voip phone service
    allow sip to sip connections
    Fax protocol cisco
    SIP
    interface FastEthernet0/0 source control binding
    bind media source interface FastEthernet0/0
    Registrar Server

    voice class codec 1
    g711ulaw codec preference 1

    translation of the voice-rule 1
    rule 1 / ^ 8 / /0056/
    !
    voice translation-rule 2
    rule 1 5.0 / /17772114zzz/
    !
    voice translation-rule 3
    rule 1 /17772114zzz/ /500/

     

    voice translation-profile IN
    definition of 3 called
    !
    FLIGHT voice translation-profile
    definition of call 2
    translate 1 called

    Dial-peer voice 1 voip
    CALLCENTRIC description
    entrants IN translation-profile
    translation-profile outgoing OUT
    destination-model 8.T
    codec voice-class 1
    session protocol sipv2
    session target sip-Server
    incoming called-number 17772114zzz
    SIP DTMF-relay-notify rtp - nte
    !
    Dial-peer voice 2 voip
    CUCM description
    destination-model 500
    media stream-autour
    codec voice-class 1
    session protocol sipv2
    session target ipv4:192.168.10.116
    incoming called number 8.T
    SIP DTMF-relay-notify rtp - nte

    SIP - ua
    credentials of username, password 7 17772114zzz 094F471B100928 callcentric.com Kingdom
    authentication username 17772114zzz password 7 121A0C051B0803 callcentric.com Kingdom
    no remote-party-id
    Registrar dns:callcentric.com expires 3600
    DNS:callcentric.com SIP server
    Home-Office


    Thank you guys.
     
     

    Hello

    Please configure the sip requires low profile to remove the header and apply this profile in the dial-peer 1 outgoing.

    SIP-class voice profiles 1

    response header 200 sip requires DELETE

    If this does not work under Dial-peers, try to apply globally.

    voip phone service

    SIP

    SIP profiles 1

    Suresh

    Please note all useful posts

  • Caller ID of VCS Jabber Video on SIP Trunk on the phone

    Hello, Netpros.

    I am trying to find a way to show a caller E.164 ID when a call to a video client Jabber a UCM or a PSTN phone.

    VCSC/E 7.1; 13.2 TMS, configured TMSPE.

    We have a VCSC > SIP Trunk > UCM (8,6) > H.323 GW w / PRI voice configured for the composition of and the PSTN destination.  We do this, so the Jabber client video can be used as a softphone, and so we can have transparent FindMe to voice and SNR to the video in either sense.  In TMS, we have a model of implementation configured with an address of the video, ILA and unit address.  He pulls ad office phone number for video of Jabber users.

    From what I read here:

    http://www.cisco.com/en/US/docs/telepresence/infrastructure/vcs/config_guide/Cisco_VCS_FindMe_Express_Deployment_Guide_X7-1.pdf, we should be able to see the caller ID when you call send out an ISDN GW.

    «This section describes how to use FindMe with calls are routed through an ISDN gateway (for example, when you call a mobile phone, or any other destination accessible ISDN).» If VCS has Caller ID (Applications > FindMe > Configuration) defined to use the ID FindMe, the identification of the appellant presented will be the user E.164 phone number. »

    UCM SIP config is vanilla, by

    ttp: / /www.tandberg.com/collateral/documentation/Deployment_Guides/Cisco_VCS_Ci...

    However, when a video call Jabber is directed to a University Complutense of MADRID registered phone, we get "unknown caller".  When to call us directly to the PSTN, we see that the BTN from the PRI.

    What is strange, is that if we call another client of Jabber of VCS (with all the same problems), directly into a phone UCM via Internet on URI (for example [email protected] / * /) through the VCSE, we see the caller ID name.

    Thank you, all!

    Joshua

    Hi Joshua,.

    At present the VCS will only send caller E.164 id to a GW of H.323 ISDN, which is registered for VCS.

    There a feature request in the order book to define if E.164 Caller ID or name/Video address on a per-zone basis. This will also be sove the question when the call is to go on a SIP connection, as in this case, a remote ISDN GW.

    Thank you

    Guy

  • Configure (fxs) analog phones with Sip trunk

    Hi all

    IS - this configuration FXS possiple to knit on SIP TRUNK? I HAVE 16 FXS ports and voip gateway 2921 cisco.

    Configuring fxs to work with her?

    any help thanks.

    Is there a PBX at stake here, i.e. CUCM or CME?

    In both cases, you have set good dial-peers to point to the FXS ports to match the destination of the SIP trunk, which is pretty simple.

  • CME: assign extension sip trunk

    Hello

    Instead of using a prefix to use a separate sip trunk, I would like an IP phone to use a separate sip for its 2nd line trunk.

    Then I set up a 2nd line on an IP phone to use extention 96:

    ePhone 1
    Mac address *.
    name *.
    button 2:96

    ePhone-dn 96 double line
    Number of 432

    and then I would this extension (with the number ending in 432) always use a SIP server.

    I think I need a dial-peer to achieve this, similar to:

    Dial-peer voice voip 432
    destination-model?
    codec voice-class 1
    voice-class sip dtmf-relay rtp - nte force
    session protocol sipv2
    session target sip-Server
    DTMF-relay rtp - nte
    No vad

    How can I join the dial-peer name extension? (for analog, I would put "monitor trunk 1 * voice port number *"on the name extension ").

    Any help appreciated,

    Jonathan

    Hello.

    You can try with answer address under your dial-position 432 432.

    HTH

    Concerning

    Carlo

  • CUCM v8.5 with 3 SIP Trunks to the Lync Server - Route algorithm of Distribution for the Group

    I CUCM connected to three different Lync server via 3 different SIP trunks.

    RG is composed of the following elements:

    LYNC SIP TRUNK 1 (1.1.1.1)

    LYNC SIP TRUNK 2 (2.2.2.2)

    LYNC SIP TRUNK 3 (3.3.3.3)

    The route group was built with "Top Down" the algorithm of distribution. The first SIP trunk knows congestion and some calls are never routed to secondary and tertiary SIP trunks.

    Based on all the forum posts I've seen - it seems that I have to configure the algorithm of group distribution of ranges as 'circular '.

    If I change the algorithm group of "Circular" lines - can I expect the following results:

    1. first call will go through LYNC SIP TRUNK 1

    2. second call will cross LYNC SIP TRUNK 2

    3. third call will cross LYNC SIP TRUNK 3

    When I change the algorithm of distribution of the route to the 'circular' group and click 'SAVE', I am invited on "RESET".  This service assigns to the existing calls through SIP Trunks?

    TIA,

    Amir

    Hello

    the circular algorithm will work the way you mentioned, but I suggest to try and make sure that is not perform integration

    to reset the SIP trunk actually active calls should not be made because the gose media directly between endpoint unles syou use something like trust rely wher evous enforce calls to go to the SIP

    Therefore, in most cases is should be fine just try configuration at the time of the calls will face service disruption

    hope this helps

  • Sip Trunk design question

    Hello

    I have a requirement to pass an h323 to SIP environment environment. I'm looking for good practices, especially around security. I have 2 servers CUCM (8.5) in cities separated for redundancy. I have also 2 voice gateways which, at the present time, h323 to the PSTN, are each located in different cities.

    My requirements are:

    1. create a sip trunk instead of the supplier of the use of PRI.

    2 If the Wan link fails on a gateway provider, router replacing in the other location should be able to receive installation messages and if a user connects via extension mobility, should be able to answer the call.

    Is there a simplified design docos on for this? I hesitate to create a SIP trunk directly to the supplier for safety, thus thinking to end the call on the routers of voice with the CUBE. I am sure that it is managed from the factory and would appreciate comments.

    See you soon!

    Pieter

    Simple answer use ALWAYS the CUBE.  With IOS 15.1 T and more you have security against fraud free of charge that you can use to restrict which can address IP contacted the CUBE, that's all you need.

    HTH,

    Chris

  • CallManager Express

    Hello

    I use CallManager Express Version 15.5 (M). How can I limit some ip phones emit an external calls to the PSTN. I use 2 lines I want to affect every line to a pone ip only 2 ip phones can make external calls. I tried to configure that but all ip phones emit an external calls

    Kind regards

    Hi Moussa, you must configure the Corinthians to refer the link for more information below: http://www.cisco.com/c/en/us/support/docs/voice/call-routing-dial-plans/... HTH Fox

  • How a dial-peer SIP Trunk using a Registrar?

    Hi all

    I'll put up a GUY using two records.  With the generous help of people on this forum, I got records in doubles to work.  But now I need to know how to configure the dial-peer to use registration information.  For example, I have this set up:

    SIP - ua

    Password 08114342101A0A1A43 7 authentication username 5555555555

    ...

    IPv4:11.11.11.11:6034 at the office 1 expires 3600

    Registrar 2 ipv4:22.22.22.22:6035 expires 3600

    How to configure a dial-peer to send traffic to one of the recordings?  I tried this, and it does not work:

    Dial-peer voice voip 105

    Description * outgoing SIP Trunk call *.

    translation-profile outgoing PSTN_Outgoing

    destination-model 91%...

    session protocol sipv2

    Registrar of target session? WHAT SHOULD I USE HERE?

    codec voice-class 2

    DTMF-relay rtp - nte

    No vad

    Thanks in advance.

    Hi Tod,

    Please go through this post, hope that it answers your question:

    http://tekcert.com/blog/2011/02/03/CME-configuration-example-SIP-trunks-ViaTalk-and-voipms

    Specify the article accordingly.

    Kind regards

    Kevin

Maybe you are looking for


HashFlare