9.................
IP Transfer Point
denism@ot.ru - OPEN TECHNOLOGIES
Aug 13, 2003, 12:54am PST
Can I use Cisco 7206 and 7507 as IP-MPLS routers with IP Transfer Point feature simultaneously (port adapter PA-MCX-8TE1-M).
And which release of IOS should I order.
Thanks for any help.
mmellet
Aug 19, 2003, 7:59am PST
I think this is possible:
Check this out....
http://www.cisco.com/warp/public ... rodlit/sts70_wp.htm
10...............
connect-time == disconnect-time
asp13 - administrator,
Aug 18, 2003, 1:19am PST
Hello!
I have an AS5350 as a voip gateway running c5350-is-mz.122-13.T5.bin. When i mak
e a call somewhere i can see strange thing in radius log:
-------------radius log, quoted---------
Acct-Session-Time = 0
h323-call-type = "h323-call-type=VoIP"
h323-connect-time = "h323-connect-time=10:44:50.091 MSD Mon Aug 18 2003"
h323-disconnect-time = "h323-disconnect-time=10:44:50.091 MSD Mon Aug 18 2003"
------------End---------------
I.e. connect-time exactly equals disconnect-time for Voip-leg. Though all statistic for Telephony-leg looks OK.
Could you tell me why such thing happens and what to do?
Thank you!
----------------sh run, quoted--------
aaa authentication login h323 group Voip
aaa authorization exec h323 group Voip
aaa accounting connection h323 start-stop group Voip
gw-accounting aaa
-----------------End---------------
thusain - CISCO SYSTEMS, CCIE
Aug 18, 2003, 11:48am PST
THat probably reflects a call that was never answered. Most billing servers actually look at the duration which is connect
time - disconnect time, and if it is 0 it is not billed for the call. The telephony leg will show connected earlier,
especially if you have an ivr setup. The ip leg will show connected when the call is answered.
Hope this helps.
Taimoor
asp13 - administrator,
Aug 18, 2003, 9:47pm PST
Hope this helps.
Absolutely! My limited mind translated 'one telephony number with answer machine' to 'somewhere' :-)
I've call a real man and everything is OK
Thanks a lot!
11................
AS5350 E1 PRI call setup question
lyew
Aug 15, 2003, 5:14am PST
Hi, I am about to connect a AS5350 to a soft switch via E1 PRI. I am being posted question by the soft switch vendor as below
that have no idea to answer. Any help is appreciated.
Incoming call (via the PRI) from As5300 is actually 1 stage call or 2 stage call.
1 stage call:
The destination number will be in the Called Address signal field with respect to ITU Q.931.
No connect call is needed on our gateway unless called party answer the call during out dial.
2 stage call:
The destination number will be send to our gateway in the respective voice channel when call is answered.
Then, our gateway will make the out going call to called party with respect to the received digit.
If it's a 2 stage call, please let me know what is the time frame when Cisco box start sending destination number once the
call is answered on our gateway. And also the inter digit time frame in between digit to send.
thusain - CISCO SYSTEMS, CCIE
Aug 15, 2003, 12:47pm PST
The short and simple answer to your questions is that it will be a 1 stage call, where we will provide the called number in
the q931 setup message and the call should be placed using that.
However you can get both stages to work. If we make a setup and ure switch is set for a 2 stage call, then it is likely the
switch will send back dialtone, and a connect which will cut through the audio in both directions. Then the originating side
can enter digits and make the call to any other extension. SO both scenarios can work, jsut depends on what you are trying to
do.
Taimoor
12.................
VOIP Problem
whamri
Aug 13, 2003, 6:33am PST
Hi. I have some problems with VoIP
If you know to connect any telephone to router we need to VICs. As you know each module in the router can hold two VICs, so
these two VICs will have the same IP.
The problem:
For example
VIC 1: 222 333 888
VIC 2: 222 333 999
When any body dial the first number the call will received in the first line or the second line randomly
jtufail - CISCO SYSTEMS, CCIE
Aug 14, 2003, 1:24pm PST
You can assign preferences to the dial peers that are associated to the voice ports assuming that the called number is the
same (a match occurs on the dial peer) for these voice ports.
In case you have different numbers, then the call would only be routed to the port which the router finds a match.
13................
Abnormal # of calls in progress
wetzelj
Aug 12, 2003, 1:37am PST
Hi!
I'm new to CallManager but I'm monitoring the network late-night, and I'm noticing the number of calls in progress is weird.
Right now, at 3:30am, there are 74 supposed calls in progress. There is no way this could be true since during the day there
are only about 115 calls in progress at any time.
It's like these are left over from the day's activity. Any hints on what is going on here? What are these 74 phantom calls?
How can I clear them?
I'm running call manager 3.2
wetzelj
Aug 14, 2003, 1:25am PST
Timeouts? Good idea. I checked and they seem ok.
So what I decided to do was stop and restart the CCM process at 4am.
That put an abrupt end to all the phantom calls. System has been running properly since.
I wonder what it was... maybe, late at night, the CallManager was merely dreaming....
14...............
How do you set up voice over ip infrastructure
monteroam@stewart.army.mil - INFORMATION SYSTEMS TECH, US ARMY
Aug 13, 2003, 12:44pm PST
I currently have 2 Cisco 3745 routers. I know I need a call manager and a gateway. Is there a module I can use for the 3745
to make it one or do I need to use a VG200 or VG248. What I want to do is allow an ip phone at a remote site to call a
regular phone in the states. The Wan links are no issue. How do I have the IP phones talk to the existing infrastructue?
Thanks
thusain - CISCO SYSTEMS, CCIE
Aug 13, 2003, 1:25pm PST
You can use either analo gor digital modules in the 3745 to interface with the PSTN or PBX infrastructure. You can get an NM
-HDV Module if you need a PRI or a Voice T1, or NM-HDA/NM-2V modules for analog lines.
You can also look at the ITS solution depending on the size of your phone deployment, which would allow the router to be the
call manager itself. If your phone deployments is rather large, then you would need a call manager server.
Taimoor
