Cisco SIP problems to IP paging speaker "cyberdata model"

Discussion in 'VOIP' started by jivor50@gmail.com, Nov 16, 2007.

  1. Guest

    I have CCM 5.1 with all phones using SCCP. I have four of these SIP
    based ip speakers to setup for paging. I have used the cyberdata
    website for their support setup and have also called CISCO TAC and the
    engineers their could not get this to work either. Im hoping that
    someone here can provide some help as both Cisco and cyberdata dont
    seem to know how get this thing to work.
    Basically here is the sceneario:

    SPEAKER:
    IP SIP based speaker with IP address: 192.168.0.195
    SIP server: 10.10.10.2 (this is ccm5.1)
    sip ports of 5060
    sip id: 60
    no authentication as was told by TAC engineer that ccm doesnt
    authenticate
    http://www.cyberdata.net/support/voip/sip_trunk.html

    CCM:
    created a sip trunk with the IP 192.168.0.195
    created a route pattern (DN 60) to use the SIP trunk
    all CSSs and partitions are correct and matching
    along with sip profiles and everything else

    So basically me and the TAC engineer worked on this for about an hour
    with no success. we did a trace and just noticed an unsuccessful. I
    also used wireshark to sniff packets to and from the ip speaker. I did
    not get one packet when i dialed DN 60 and i dialed the dn over 20
    times to see if anything was being sent to the speaker. so ithen
    removed the route pattern and created a a 3rd party sip phone profile
    with css, partitions and everything else matching but no success and
    could not sniff any packets using that approach either.

    So I then setup a SIP route pattern. BINGO i get a couple of packets
    sniffed between the 2 devices as soon as i hit save on SIP route
    pattern setup.

    My problem is there no way to translate a DN to use the SIP route
    pattern. I have not messed wth the SIP dial rules as I dont quite
    understand them yet and really dont think they are necessary as
    basically all that should need to be happen is a DN needs to be
    associated with a SIP route pattern. I am currently at a loss on how
    to get this to work. It should be no different than setting up any sip
    phone except that i would check unattended port. CANY ANYONE PLEASE
    HELP PLEASE! :)

    J
     
    , Nov 16, 2007
    #1
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