Trending PRI usage on an H.323 Gateway

Hey guys!!
Does anyone here know of any software or tools or any cisco "in house" method of tracking PRI usage on an H.323 gateway? We would like to see here what usage we're getting out of our PRIs.
Thanks,
BR

Hi Brent
Yes, you can use SNMP counters on an H.323 Gateway to track the number of concurrent calls.
A sample OID is 1.3.6.1.4.1.9.10.19.1.1.4.0 (you will need to check this)
I use MRTG/PRTG to track this and produce pretty graphs.
HTH. Barry

Similar Messages

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.

MGCP and H.323 gateways not working together

We have a 3640 in Mexico defined to our Call Manager here in Atlanta as a H.323 gateway. Also in the same Call Manager is a T1 PRI defined to our PBX as a MGCP gateway. When people in Mexico call phones connected to our PBX we get one way conversations. We have a couple of other H.323 to MGCP gateways situations that don't work. It seems the two are not compatible. Has anyone else ran into this. If so, what was your solution.

According to your description, I believe that you have a topology like the following diagram.
MexicoPSTN---?---3640---h.323---CCM---mgcp---gateway?---PRI---PBX
You didn't mentioned what type of circuit you had between the 3640 and the telco in Mexico or the type of gateway used for MGCP in Atlanta.
However, there should be no problems with this type of configuration causing one-way audio.
In this case CallManager talks H.323 to the 3640 in Mexico and talks MGCP to the gateway in Atlanta for the call control. When a call is placed from a PSTN phone in Mexico to a PBX phone in Atl, the call setup will first take place via H.323 between the 3640 and CCM and via MGCP b/w the Atl gw and CCM. After the call setup is complete, the audio stream should be connected directly between the 2 endpoints (ie 3640 gateway and the Atl gateway).
You should check for IP connectivity between the Mexico 3640 and the Atl router. Note, the call setup could complete successfully b/c both devices can reach the CCM, but can they reach each other when CCM steps out of the picture?
The following URL will also provide more good info on troubleshooting one-way audio.
http://www.cisco.com/en/US/tech/tk652/tk698/technologies_tech_note09186a008009484b.shtml
Hope this helps.
Shane

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

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

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

H.323 MCU 3515 as H.323 Gateway in CCM

Hi,
I have a little bit confusion and i hope u can help me clear it up!!
In which cases the MCU 3115(H.323 enabled) is configured in CCM as a H.323 Gateway??
Plz help me to clear this up, i need help urgently.
Thank you.

Hi Mikheil,
Could u please inform if the call to the originating endpoint is released, before you get the "recovery on timer expiry message"??
Also, the restrictions on Call Redirection Web page requires IVR to be setup for this feature to work. Is this setup?? Could u give more explanation regarding the problem.
thanks and regards,
Amit.

H.323 Gateway Registration

Hi,
I am trying to configure H.323 to a CUCM cluster. I was able to configure using the IP address. However when I tried using the hostname, it is unable to get registered. According to the Cisco doc help:
Device Name: Enter a unique name that Cisco Unified Communications Manager uses to identify the device. Use either the IP address or the host name as the device name.
Should we enter the hostname as configured on the IOS Gateway or hostname as configured on DNS server?

In the device you would be using the hostname that must be resolved by DNS.
hostname# resolve to IP address of the GW.
Br,
Nadeem
Please rate all useful post.

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

Remote h.323 gateway drop conference calls

Recently I went to H.323 mode on a 1760 remote gateway. This was done to better support caller ID and SRST. This gateway services 6 POTS lines locally and has a T1 point to point to get back to the CCM cluster.
When I set up a MeetMe conference and then try connect a call on one of the POTS lines to the MeetMe, it drops the call.
The same thing happens when trying to directly conference a local extension and a caller on a POTS line.
I can however directly conference a local extension and a caller that can be reached out the T1 point to point line, which would be in the area codes local to the CCM cluster at headquarters.
I can also call in to headquartes and be transfered in to an existing MeetMe conference.
So, it seems that the issue is with the local POTS lins or the gateway. Any help would be greatly appreciated.
TIA - Mike

I'm not using a SW Conference Bridge that I am aware of. Just the default conference and MeetMe feature of CCM 4.1.
I'll look at the G711 in the dial-peers. I know that have it defined globally as the first prefernece. Also, I will look at regions and sees what I can see.
Thanks - Mike

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

Supporting Nortel Remote Shelf with VOIP

Does anyone have any experience using H.323 gateways for supporting a Nortel remote shelf to headend PBX across a WAN.
The connection requires a clear channel t1 trunk between shelf and PBX. This is a bit different I think than trunking between PBXs.
Can this be set up with standard Voice PRI/T1 modules in H.323 gateways?
Any insights would be helpful

Yes it can.. However, the programming on the router for the clear channel trunk is extensive. What is the gateway?
I have three clear channels trunks between a Nortel Opt81 through a 7206 router at one end going to a 3745 to an Option 11 at the remote end.
Try this link:
http://www.cisco.com/en/US/customer/tech/tk652/tk653/technologies_tech_note09186a00800a96c1.shtml#tccctopic

Gateway Migration from H.323 to SIP

Hi All,
Currently all my Cisco Gateways registered with CUCM using H.323 protocol, for now i would like to migrate all these gateways registered protocol to SIP, simply i would like to Migrate all my H.323 gateways to SIP.
Can anyone have any best procedure to do this Migration, as well share the related documents if you have any.
Thanks in Advance for your Help.
Regards,
Madhu

Madhu,
There are a few things you need to consider..
1. You need to create sip trunks on cuc and point the trunks to the ip address of your gateway. If you have multiple IP address on the gateway, you need to carefully consider which ip address you will bind your sip signalling to.
2. You need to ensure that the cucm group in the device pool you assign to your sip trunk is the same as youconfigure for your dial-peers or you use the feature "run on all active cucm" on your sip trunk and RL.
3. Configure inbound dial-peer from cucm to the gateway and ensure it is enabled for sip (by using sip protocol sipv2 command)
4. Configure outbound dial-peer to cucm and ensure it is sip enabled.
5. configure rtp-nte on both your inbound and outbound dial-peers fir dtmf
6. ensure your sip trunk is set to use "no preference for DTMF
To understand more on sip signalling please refer to this document
https://supportforums.cisco.com/blog/154506

Usage Reports and Popularity Trends

So I've set up my site and the Search Crawl is running. I've also got the Reporting feature activated and configured Usage and Health Data Collection. The WSS_Admin and WSS_WPG have access to the Log files.
However, the Popularity Reports are either disabled on some lists and libraries or the Popular Items continually shows NIL data. Likewise, the Popularity Trends and Usage Reports under Site Settings also show NIL data. I know that data is being
collected though, as the Search Reports do contain some (limited) data.
Is there a guide somewhere that shows exactly how to set this up because I can't seem to get it to work no matter what steps I follow.

1st thing make sure you properly configured the Usage and health "http://technet.microsoft.com/en-us/library/ee663480.aspx"
Check the ULS logs for more details about error.
check some post having similar issue:
http://social.technet.microsoft.com/Forums/en-US/1b42b517-79cc-43b9-b6f0-2e4639461cb1/empty-usage-data-in-sharepoint-2013
http://social.technet.microsoft.com/Forums/sharepoint/en-US/b94d2114-48a2-4ac8-aa10-b32762275611/popularity-trends-report-is-empty?forum=sharepointsearch
http://www.sharepointsecurity.com/sharepoint/sharepoint-development/empty-usage-and-health-reports-in-sharepoint-2013/
Please remember to mark your question as answered &Vote helpful,if this solves/helps your problem. ****************************************************************************************** Thanks -WS MCITP(SharePoint 2010, 2013) Blog: http://wscheema.com/blog

Trending PRI usage on an H.323 Gateway

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