Cisco SIP problems to IP paging speaker

Cisco SIP problems to IP paging speaker "cyberdata model"

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Subject Author Date
Cisco SIP problems to IP paging speaker "cyberdata model" jivor50 11-16-2007
Posted by on November 16, 2007, 1:02 am
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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

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other useful resources:
The Federal Communications Commission (FCC)
Telecommunications Industry Association
Electronic and Software Security Products and Services
International Telecommunication Union

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