Block External calls h.323 gateway CM4.1(3)

Is there a way to block external calls from getting through the gateway. The gateway is H.323, Callmanager 4.1(3)

You can use Class of Restriction on the gateway.
http://www.cisco.com/en/US/tech/tk652/tk90/technologies_configuration_example09186a008019d649.shtml

Similar Messages

Cannot make outside call (H.323 gateway and CUCM 6)

I cannot make outside calls. I am using H.323 gateway configuration and CUCM 6. I have attached configuration file and debug file. I configured H323 gateway and route pattern in CUCM. Please let me know if this is a configuration issue or telecom issue.

Hi 9tysixuae,
I can see that call is hitting the analog ports but from there on it generates error 34 which means circuit not available. You can try the following :
1. Plug Analog phone and verify if the circuit is fine ?
2. Try putting signal ground start on the voice port and see if that makes any difference.
Regards
Aditya Gupta

CME:how to block external call to external call

cme have the four fxo and AA,when the external calls come in,and dial 9+ pstn num,it can call from external call to another external call,how can blocking?

Hi,
try to use this command
#call application voice aa max-extension-length 5
This option declares the maximum length of the extension that the user can dial when dial-by-extension-option is chosen. The default value is 5. The value can be 0 with no restriction up to x digits.
or try
3.
Configure Class of Restriction (COR) to block call transfers from B-ACD to PSTN numbers. The sample configuration below prevents the B-ACD from transferring calls out to local and long distance PSTN numbers. The B-ACD can still transfer calls to internal extensions.
Below is an example of such a configuration:
dial-peer cor custom
name longdistance
name local
dial-peer cor list call-longdistance
member longdistance
dial-peer cor list call-local
member local
dial-peer cor list block-pstn
dial-peer voice 1 voip
corlist incoming block-pstn
application aa
destination-pattern 1000
session target ipv4:192.168.1.1
incoming called-number 1000
dtmf-relay h245-alphanumeric
codec g711ulaw
no vad
dial-peer voice 2 pots
corlist outgoing call-longdistance
destination-pattern 91..........
port 0/2/0
dial-peer voice 3 pots
corlist outgoing call-local
destination-pattern 9[2-9]......
port 0/2/0
Thanks
Najeeb

Call Forwarding / Displayed Number on Forwarding target with H.323 Gateway

Hi Community,
i´m wondering if there is sort of a simple way to get this working properly.
Scenario:
Germany, variable dial plan, no fixed NANP, and we have ClipNoScreening ;)
We use 0 for getting PSTN-dialing.
We have internal DNs, for example a 6789, 4 Digits. We use external phone number mask on our lines, 123456XXX
Our main number is 0123/456-xxx
When i call outside everything is displayed fine on the called target, +49 123/456789.
When i forward a call on my cellphone, with CFA target of 00111/222333444 (my cellphone example), and an internal colleague from within our office, is calling my office phone, everything is ALSO displayed fine.
Now here comes the BUT:
When someone calls from PSTN on my office-phone, i get displayed on my cellphone the +49 (0) 0xxxxxxxx, which means the caller number PLUS the added 0 from the gateway. Which is completely consequent and correct, since we add them on the gateway, when a call comes in, to be able to just answer directly on the office phone.
The rule on the H323 gateway:
voice translation-profile OUTGOING-VOIP
translate calling 1
translate called 2
voice translation-rule 1
rule 1 /^$.*$/ /0\1/ type subscriber unknown plan any unknown
rule 2 /^$.*$/ /00\1/ type national unknown plan any unknown
rule 3 /^$.*$/ /000\1/ type international unknown plan any unknown
voice translation-rule 2
rule 6 /4560$/ /6600/
rule 9 /^456$...$$/ /6\1/
voice translation-profile OUTGOING-POTS
translate calling 3
translate called 4
voice translation-rule 3
rule 1 /^00049/ /0/ type unknown national
rule 2 /^0/ // type unknown subscriber
rule 3 /^00/ /0/ type unknown national
rule 4 /^000/ /00/ type unknown international
voice translation-rule 4
rule 2 /^00049$.*$$/ /\1/ type unknown national
rule 3 /^000$.*$$/ /\1/ type unknown international
rule 4 /^00$.*$$/ /\1/ type unknown national
rule 5 /^0$.*$$/ /\1/ type unknown subscriber
dial-peer voice 10456 voip
translation-profile outgoing OUTGOING-VOIP
destination-pattern 456.T
progress_ind setup enable 3
modem passthrough nse codec g711ulaw
session target ipv4:<IP-OF-CUCM>
incoming called-number .
voice-class codec 1
voice-class h323 1
dtmf-relay h245-alphanumeric
fax-relay ecm disable
fax rate disable
fax protocol pass-through g711ulaw
no vad
no supplementary-service h225-notify cid-update
dial-peer voice 345000 pots
tone ringback alert-no-PI
translation-profile outgoing OUTGOING-POTS
destination-pattern 0.T
progress_ind alert enable 8
progress_ind progress enable 8
progress_ind connect enable 8
port 0/0/0:15
forward-digits all
In case of forwarding the external call to an external device, like for example a cellphone, this is crap.
Its obvious regarding the debugs all is working as designed ;), because my phone just forwards the full calling number including the added 0, since i put in to forward the originating calling DN.
My question now:
Can i simply correct this behavior somehow, also for international calls which would the 00 get added by the gateway?
Many thanks in advance for some input,
Andreas

Hi Community,
i´m wondering if there is sort of a simple way to get this working properly.
Scenario:
Germany, variable dial plan, no fixed NANP, and we have ClipNoScreening ;)
We use 0 for getting PSTN-dialing.
We have internal DNs, for example a 6789, 4 Digits. We use external phone number mask on our lines, 123456XXX
Our main number is 0123/456-xxx
When i call outside everything is displayed fine on the called target, +49 123/456789.
When i forward a call on my cellphone, with CFA target of 00111/222333444 (my cellphone example), and an internal colleague from within our office, is calling my office phone, everything is ALSO displayed fine.
Now here comes the BUT:
When someone calls from PSTN on my office-phone, i get displayed on my cellphone the +49 (0) 0xxxxxxxx, which means the caller number PLUS the added 0 from the gateway. Which is completely consequent and correct, since we add them on the gateway, when a call comes in, to be able to just answer directly on the office phone.
The rule on the H323 gateway:
voice translation-profile OUTGOING-VOIP
translate calling 1
translate called 2
voice translation-rule 1
rule 1 /^$.*$/ /0\1/ type subscriber unknown plan any unknown
rule 2 /^$.*$/ /00\1/ type national unknown plan any unknown
rule 3 /^$.*$/ /000\1/ type international unknown plan any unknown
voice translation-rule 2
rule 6 /4560$/ /6600/
rule 9 /^456$...$$/ /6\1/
voice translation-profile OUTGOING-POTS
translate calling 3
translate called 4
voice translation-rule 3
rule 1 /^00049/ /0/ type unknown national
rule 2 /^0/ // type unknown subscriber
rule 3 /^00/ /0/ type unknown national
rule 4 /^000/ /00/ type unknown international
voice translation-rule 4
rule 2 /^00049$.*$$/ /\1/ type unknown national
rule 3 /^000$.*$$/ /\1/ type unknown international
rule 4 /^00$.*$$/ /\1/ type unknown national
rule 5 /^0$.*$$/ /\1/ type unknown subscriber
dial-peer voice 10456 voip
translation-profile outgoing OUTGOING-VOIP
destination-pattern 456.T
progress_ind setup enable 3
modem passthrough nse codec g711ulaw
session target ipv4:<IP-OF-CUCM>
incoming called-number .
voice-class codec 1
voice-class h323 1
dtmf-relay h245-alphanumeric
fax-relay ecm disable
fax rate disable
fax protocol pass-through g711ulaw
no vad
no supplementary-service h225-notify cid-update
dial-peer voice 345000 pots
tone ringback alert-no-PI
translation-profile outgoing OUTGOING-POTS
destination-pattern 0.T
progress_ind alert enable 8
progress_ind progress enable 8
progress_ind connect enable 8
port 0/0/0:15
forward-digits all
In case of forwarding the external call to an external device, like for example a cellphone, this is crap.
Its obvious regarding the debugs all is working as designed ;), because my phone just forwards the full calling number including the added 0, since i put in to forward the originating calling DN.
My question now:
Can i simply correct this behavior somehow, also for international calls which would the 00 get added by the gateway?
Many thanks in advance for some input,
Andreas

Can Call Manager support network side Q.931 from H.323 gateways

With CCM 3.1, is it possible to support network side E1 Q.931 on a H.323 gateway. This would allow PABXs to connect to CCM, which then allow CCM to act as a tandem exchange.

The network side support on the H323 gateway
is transparent to the CCM. They are two seperate call legs.
1. pots for E1 PRI and
2. voip call leg to CCM.
The support for network side PRI is on 2600/3600/3700/7700/AS5300,AS5800
and its been available since 12.1(3)T and please use
a more recent image 12.2T train ip plus at the least.

Help with setting up Caller-id on FXO in 2911 H.323 gateway.

I have a remote site that has a couple of POTS lines terminating to a FXO on the 2911. This remote site is an H.323 gateway in a CUCM 8.6 cluster. Incoming local calls for that location ring all phones at the location.
What do I need to do to enable Caller-id on these POTS lines terminating to the FXO? I am pretty sure the carrier is sending the caller ID information.
Thanks

Hi
1- Please find the below table as the following link http://www.cisco.com/en/US/products/hw/routers/ps274/products_tech_note09186a00800b53c7.shtml
Caller ID
Requires VIC-2FXO-M1, VIC-2FXO-M2, VIC-4FXO-M1, VIC2-2FXO, VIC2-4FXO, or MRP3-8FXOM1
2- under fxo port
voice-port 0/3/0
caller-id enable
3-If the above configure and still have no caller id , please add the below commannds to the voice-port
caller-id alerting line-reversal
cptone ?              "based on your"
caller-id alerting ring 2    "the default is 1" maximum number of rings to be detected before a call is answered over an FXO voice port.
4-Do debus to make sure all ok
"debug vpm signal "
[0/3/0] get_fxo_caller_id:Caller ID received. Message type=128 length=31 checksum=74
Thank you
please rate all useful information

CUCM 8.6(2) migration problem with external calls

Hello all.
Yesterday we have migrated our telephone infrastructure from CM4.x to CUCM8.6(2), after some weeks of tests.
Yesterday night all seems to work properly, all phone updated and registered, external calls going out and in.
But from this morning, with all users at work, it appears a strange problem, that until now I couldn't solve: randomly all external calls go down.
I can't address this problem, since gateways (all cisco 2811 routers) are the same and with same configuration as yesterday.
All thing that I can think is that router that seems to cause the problem is configured not with mgcp by cucm, but with h323 route inside the router.
Any suggestions will be greatly appreciated.
Daniele

GW says normal call clearing.
But, maybe I've addressed the problem.
I've found a bug fixed into latest cucm release (8.6(2a)SU1) that say "h.323 calls improperly disconnected".
So I'm trying to upgrade from 8.6(2a) to 8.6(2a)SU1, but process fails :-(
I've tried from a dvd and also loading iso image from sftp, but after few minutes appears an error
08/04/2012 09:43:55 upgrade_install.sh|Started auditd...|
08/04/2012 09:43:56 upgrade_install.sh|Started setroubleshoot...|
08/04/2012 09:43:56 upgrade_install.sh|Changed selinux mode to enforcing|
08/04/2012 09:43:56 upgrade_install.sh|Cleaning up rpm_archive...|
08/04/2012 09:43:56 upgrade_install.sh|Removing /common/rpm-archive/8.6.2.21900-5|
08/04/2012 09:43:56 upgrade_install.sh|File:/usr/local/bin/base_scripts/upgrade_install.sh:599, Function: main(), Upgrade Failed -- (1)|
08/04/2012 09:43:56 upgrade_install.sh|set_upgrade_result: set to 1|
08/04/2012 09:43:56 upgrade_install.sh|is_upgrade_lock_available: Upgrade lock is not available.|
08/04/2012 09:43:56 upgrade_install.sh|is_upgrade_in_progress: Already locked by this process (pid: 1286).|
08/04/2012 09:43:56 upgrade_install.sh|release_upgrade_lock: Releasing lock (pid: 1286)|
I've rebooted server yet and problem remains.
Thanks for any other suggestions.
Daniele

Block Inbound Calls

Most of us are probably aware of the limited ability, via H323 gateways and translation rules, to block inbound calls from the PSTN.
Two questions:
1.) Is there another method I may not be aware of?
2.) I was reading about External Call Control Profiles this morning and it seems like a logical fit. Anyone experimented with this?
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/8_0_1/ccmfeat/fsextcallctrl.html

Here is great Doc from CCO that outline Voice Translation Profiles and has examples of
various situations. Letting the GW handle it to me seems most straight forward if its H323 GW
Why make CUCM have to even deal with it?
Has example for blocking ext
http://www.cisco.com/en/US/customer/tech/tk652/tk90/technologies_configuration_example09186a00803f818a.shtml
HTH, if it does not answer what your after pls give me some more details on what you want to accomplishand I will see if I can help
George

Differentiate ring tone for internal/external call

how to differentiate ring tone for internal call and external call?
rgds,
walad.

Floyd,
Those parameters were changed in 4.1 CCM. I don't believe they are going to affect Ring Type but could effect CFNA or CFB settings.
?Use the Call Classification field in the Cisco CallManager Administration Gateway Configuration window to configure H.323 and MGCP gateways with the option for OffNet, OnNet, or Use System Default.
?The default value for H.323 and MCGP gateways specifies OffNet.
?The default value for intercluster trunks (ICT), or trunks other than SIP trunks, specifies OnNet.
?In Cisco CallManager release 4.1, Call Classification replaces the H323 Network Location, MGCP Network Location, and MGCP Network Location OffNet for E1 and T1 service parameters.
?To configure trunks and gateways, the administrator can use the Call Classification clusterwide service parameter and choose the Use System Default option for the individual trunks and gateways.
?By default, the service parameter specifies OffNet.
?You can also configure trunks and gateways individually.
?The system considers FXS and phones to be OnNet; you cannot configure them.
The only way that I know of to achieve what you want is dual-line. Ring settings can be set per line or a default for the phone. You could use a unique inbound Gateway CSS with a 2nd line on the phones just for inbound calling. Then you can specify the ring on the phone to be different for the second line.
There is nothing in 4.1 and below (again to my knowledge) that will allow for distinctive ring per call. You should double-check the 4.2 and 5.x release notes to make sure there isn't a new feature though.
Please rate any helpful posts
Thanks
Fred

Block International Calls

How do I prevent my users from calling international phone numbers?
CUCM 10.0 + Cisco H.323 gateway
Take into consideration that according to the NL numbering plan:
0 - external
00 - long distance
000 - International
So matching for 000 is preceded by a match for either 0 and 00.
I gather I'll have to do something with a route filter, but so far I haven't been able to create one that actually works.

I wasn't able to use a 000 route pattern, the callmanager seemed to like the match for 0 better.
However, I was able to figure out how to use route filters.
- created a 2nd 0.@ with a 2nd route partition (same calling search space).
- created two route filters: 1: no international, 2: international
- set the two filters on the respective route patterns
- enabled Forced Authorization Code on the international route pattern
This way only users that know the code can call international.

Jabber for IPhone over AnyConnect VPN calls to SIP Gateway - One Way Audio

Hello,
I am wondering if anyone has seen this before. I have the latest version of Anyconnect for the IPhone and Jabber app, running on the newest version of CUCM 8.6 and Jabber 8.6.
The Jabber phone registers fine and I can make internal calls without an issue. I also can make external calls to H323 and HGCP PRI gateways without an issue. However, when I make a call from the Jabber client that goes out a SIP gateway and SIP trunk I get one way audio. I do not see any packets even trying to leave inside interface of the ASA headed toward the SIP trunk IP.
I've also verified that useing the same VPN profile I can make 2 way audio calls out the SIP trunk from a softphone running on windows.
Anyone have any ideas what could be the problem?
Thanks.

Hi,
try to disable early media on 180 under sip profile settings for jabber client.
HTH
Anas
please rate if itnis helpful
Sent from Cisco Technical Support Android App

H.323 Gateway is not Registering in CCM 4.1.2

H.323 Gateway. I am using cisco2620
========================================
nafay#sh run
Building configuration...
Current configuration : 966 bytes
version 12.2
service timestamps debug uptime
service timestamps log uptime
no service password-encryption
hostname nafay
ip subnet-zero
mgcp
mgcp call-agent 192.168.0.111
mgcp dtmf-relay codec all mode out-of-band
call rsvp-sync
voice class h323 1
h225 timeout tcp establish 3
ccm-manager mgcp
interface FastEthernet0/0
ip address 192.168.0.55 255.255.255.0
duplex auto
speed auto
interface Serial0/0
no ip address
shutdown
ip classless
ip http server
voice-port 1/0/0
voice-port 1/0/1
voice-port 1/1/0
voice-port 1/1/1
dial-peer cor custom
dial-peer voice 100 voip
preference 1
destination-pattern 1...
voice-class h323 1
session target ipv4:192.168.0.111
dtmf-relay h245-alphanumeric
ip precedence 5
dial-peer voice 101 voip
preference 2
destination-pattern 1...
session target ipv4:192.168.0.111
dtmf-relay h245-alphanumeric
ip precedence 5
line con 0
line aux 0
line vty 0 4
end
nafay#
=========================================

Is this config is OK. Still not registering
=====================================================
service timestamps debug uptime
service timestamps log uptime
hostname nafay
ip subnet-zero
call rsvp-sync
voice class h323 1
h225 timeout tcp establish 3
interface FastEthernet0/0
ip address 192.168.0.55 255.255.255.0
duplex auto
speed auto
interface Serial0/0
no ip address
shutdown
ip classless
ip http server
voice-port 1/0/0
voice-port 1/0/1
voice-port 1/1/0
voice-port 1/1/1
dial-peer cor custom
dial-peer voice 100 voip
preference 1
destination-pattern 1...
voice-class h323 1
session target ipv4:192.168.0.111
dtmf-relay h245-alphanumeric
ip precedence 5
dial-peer voice 101 voip
preference 2
destination-pattern 1...
session target ipv4:192.168.0.111
dtmf-relay h245-alphanumeric
ip precedence 5
line con 0
line aux 0
line vty 0 4

H.323 gateway behind NAT

i configued h.323 gateway (gateway is connected PSTN through FXO) behind internet NAT router and try to call that gateway from a softphone through internet. the dialed PSTN no is ringging but no voice for both ways. Pls refer the attached configuration. Is this a problem with NAT translation?
Thanks in advance!

Yes, you need a version of IOS that has NAT ALG. What IOS are you running?
NAT with ALG can translate the embedded addresses in H225/H245.
Cisco IOS NAT Application Layer Gateways
http://www.cisco.com/en/US/tech/tk648/tk361/technologies_white_paper09186a00801af2b9.shtml
http://www.cisco.com/en/US/products/ps6441/products_configuration_guide_chapter09186a00807819ce.html
Please rate helpful posts.
Dave

AIM/CUE not Picking-up in-bound external calls

I have a 2811 router with an AIM/CUE (12-VM) card. All of the telephony-service stuff works fine:
-All calls in house work fine
-All outbound calls work fine, to PSTN
-All inbound calls work fine, from PSTN
-Each phone can access it's VM internally
-the CME gui and the Unity-Express gui's are accessible on the LAN.
But, when I call from outside into the PRI with a DID number, I get four rings, then a pause, and then it never goes to voicemail. Sometimes; depending if I call with my celr or a land line, I get the message "Sorry your call cannot be completed as dialed, 000-000." Any clue why the vm does work for internal or external calls.
Here's a sample of the config.
interface FastEthernet0/0
no ip address
no ip mroute-cache
duplex full
speed 100
interface FastEthernet0/0.100
description SUB-IF for VOICE VLAN
encapsulation dot1Q 100
ip address 10.1.100.1 255.255.255.0
interface FastEthernet0/0.101
description GW FOR VOICEMAIL MODULE AIM/CUE
encapsulation dot1Q 101
ip address 10.1.101.1 255.255.255.0
interface Service-Engine0/1
discription AIM/CUE VOICEMAIL MODULE
ip unnumbered FastEthernet0/0.101
service-module ip address 10.1.101.2 255.255.255.0
service-module ip default-gateway 10.1.101.1
ip route 10.1.101.2 255.255.255.255 Service-Engin0/1
dial-peer voice 40 voip
description ** cue voicemail pilot **
destination-pattern 6000
session protocol sipv2
session target ipv4:10.1.101.2
dtmf-relay sip-notify
codec g711ulaw
no vad
dial-peer voice 41 voip
description ** cue auto attendant **
destination-pattern 6001
session protocol sipv2
session target ipv4:10.1.101.2
dtmf-relay sip-notify
codec g711ulaw
no vad
dial-peer voice 1 pots
description ** T1 PRI Emergency 911 **
destination-pattern 9911
port 0/0/0:23
forward-digits 3
dial-peer voice 2 pots
description ** T1 PRI OutBnd Calls **
destination-pattern 9T
port 0/0/0:23
dial-peer voice 3 pots
description ** T1 PRI InBnd Calls**
incoming called-number .
direct-inward-dial
port 0/0/0:23

Problem has to do with how many Digits are getting sent to voice mail on the router when it tries to make the call. You should be seeing 10 digits in the debug.
Next question when one ephone calls another ephone locally does voice mail work?
Try the following commands. This will create a translation-rule that will strip incoming 10 digits to the 4 digits you need. I assume you do not need all 10 digits.
voice translation-rule 1
rule 1 /^650$....$/ /\1/
voice translation-profile digitstrip
translate called 1
voice-port 0/0/0:23
translation-profile incoming digitstrip
*************Voice port were you Telco Line comes in Port 0/0/0:23 based on your config*************
telephony-service
no dialplan-pattern 1 6503926... extension-length 4
Ronnie

Non-blocking J2EE calls

We wish to connect fat clients (in our case Powerbuilder) to a J2EE environment (Weblogic). We also need to make some of the calls from the fat client non-blocking (asynchronous). With many clients, JMS was rejected as a mechanism.
Our first solution, which works as a prototype, was to package a Java bean as an OLE object and deploy it to the client. This bean could make the calls non-blocking. However there are registry issues (the packaging implies a pre-defined directory structure), we need a JRE on every client and 15Mb of weblogic jars (aaargh), so it only really works as a prototype.
The next (current) solution is a C dll that can be called from Powerbuilder and uses a socket to talk to a Java J2EE Gateway on the server. This Gateway makes the system non-blocking and calls the EJBs. This is clearly easier in terms of configuring the client.
Ideally we would like to drop the Gateway by finding a non-blocking way to get into J2EE from the C dll but the only methods we have thought of:
- simulating an HTTP servlet call
- SOAP
are synchronous (or appear to be, my knowledge of SOAP is limited).
Has anyone come across other ways of getting non-blocking communications with J2EE?

CORBA, this is what it was designed to do, but it is not not a light weight solution.

Block External calls h.323 gateway CM4.1(3)

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