H323 gateway with PABX Avaya

Hi
Which are the parameter that I need to configure H323 signaling system between my gateway CISCO2811 bundle CME and Avaya PABX
Thanks

Try:
http://www.cisco.com/en/US/tech/tk1077/technologies_tech_note09186a00803d22cd.shtml

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H323 Gateway Intermittent MOH Silence

We are using a 3925 router - c3900-universalk9-mz.SPA.150-1.M4.bin - as an H323 gateway with 2 PRI's. We have a PUB and a SUB running -
System version: 8.5.1.11900-21 When we place people on hold we get MOH very intermittently. Out of 10 calls 8 might get MOH and 2 get silence. I have checked codecs (all calls G711), I have restarted IPMediaStreaming. The weird thing is that some calls work and some do not. We cannot isolate any factors to be the same or possible causes. I uploaded the voice portion of the router to see if it helps. Thanks for any help.
Thanks,
Donnie

HEllo,
I had the same issue. Check the BUG ID. My IOS version is 15.0(1)M4 on a Cisco 2911. The affected IOS version described on the Bug ID.
I've made the Option2 Workaround and the issue got solved.
Thanks for the info.
Rui

Problem with H323 gateway

We have a CUCM ver 8.6.2.22900-9.
For our PSTN access we have the following scheme:
CUCM <========> Cisco2911 <======> ISP-------> PSTN
H323 tk SIP tk
Our Cisco2911 is running ver 150-1.M4.
We configure Cisco 2911 as H323 gateway on CUCM, and we established a SIP tk with our ISP.
When we tested incoming calls we received them at router but they are unable to acceess CUCM,
When we tested outgoing calls we got reorder tone and they did not reach our voice gateway.
We run "debug cch323 all" and tested with one incoming and one outgoing call, and we got the out on attached file "debug h323 forum"
CUCM Ip : 10.1.2.2
Router's H323 int: 10.2.2.1
Do you have any idea about the root cause?
Can be an access list problem preventing proper H323 traffic??
Thanks in advanced
Enrique

Do you have voip dial-peers pointing back to your cucm from your gateway? If not then the call wouldn't route onward to the cucm, but the gateway would also ignore communications from that souce because of toll fraud prevention.
http://www.cisco.com/c/en/us/support/docs/voice/call-routing-dial-plans/112083-tollfraud-ios.html

Help on set up branch office with 2921 H323 gateway

I setup a new branch office with 2921 H323 gateway and cucm in HQ. When I call a number in remote office, I get dead silence and busy tone. However, user can hear ring at the remote location and able to answer the phone. I was able to talk to him. Any place I need to check?
Question #2, should cucm in HQ handle all calls between HQ and remote office? I tried to call from my VoIP phone to remote office VoIP phone and monitored remote office GW running "debug voip ccapi inout". I saw messages like gateway is handling calls. Is this normal?
I'm fairly new to VoIP environment, still trying to learn. Thanks. Let me know if you need anything to troubleshoot this.

I think that's where I'm confused. I'd like to have CUCM to handle every calls for remote sites. when I searched for the number I'm dialing for remote office, there is a route pattern that covers this number and it points to gateway. This route pattern covers all of their local numbers including our remote office numbers. For example, I have 9.1201456XXXX point to H323 GW. That I got it. However, I want the numbers belong to our office like 1111 don't go to GW for call processing. Do I make sense? I want only their local call to remote office go through GW not our internal call between our offices. I'm sorry if I don't make much sense. Thanks for your help.

Adding MGCP FXS Ports to H323 Gateways

Currently all of our Gateways are H.323 gateways. Due to a business requirement we are now going to be enforcing our users to use forced authorization codes to place LD calls. In order to facilitate this on our analog phones it seems the only option is to use MGCP gateways.
From what I understand we can run multiple signalling protocols on voice gateways. We have a variety of gateway models but by and large most of these gateways are VG224 models. I think what I would like to do is keep the current h.323 dial-peer and voice-port settings for the PLAR emergency phones that we have on these gateways and only change the analog phones to MGCP.
Most of the route patterns to these h323 gateways look like this... 102[0-5] and then the dial peers on the individual gateways route to the appropriate voice port like this...
dial-peer voice 1020 pots
huntstop
destination-pattern 1002
port 2/21
The Voice port config looks like this...
voice-port 2/21
timeouts interdigit 7
description tie pr 1520
station-id name PTRM 1020
station-id number 1020
caller-id enable
My plan is to create the MGCP Gateways in CUCM as wells as the DN's... in this example x1020. I will then enable MGCP on the gateways. After that my assumption is that I can individually remove the Voice-port and dial-peer configurations and then add the MGCP dial peers with the port and "service MGCPAPP" commands.
My other option is to redo the entire gateway at the same time and schedule after-hours down-times to make the change. I want to avoid this if possible as we have 40+ gateways that need to be changed.
Basically I just need some guidance or confirmation if my plan will work or if there is a better way to do this? Are there any caveats or known issues I should look out for when running multiple signalling protocols on the same gateway?
Thanks,
Trav Moore

Thanks Aaron,
I was wondering about the MGCP ccm-config command but was worried it would re-write the entire h.323 gateway to MGCP. Good to know that it won't and that this is a potential option.
I actually do prefer the idea of only having one signalling protocol (I would like to go all SIP if not for the FAC codes needed). Unfortunately any maintenance that I do that impacts end-users requires a lot of after-hours scheduling and maintenance alerts. These gateways have a combination of fax-machines, PLAR's (emergency phones and overhead paging), and analog phones. Maybe eventually I can migrate all of these ports to MGCP. For now the analog phones are the only ones that must be converted and if I can quickly convert them without anyone noticing aside from the minimal reset in CUCM then this would be ideal.
Thanks!

Why we dont' see H323 gateway and SIP trunk "registered" in CCM?

in CCM admin, we see the status of H323 gateways and SIP trunks have no "registered to xxxx" status.

Unlike devices with other protocols, there is no registration mechanism for these.

Gatekeeper showing priority 0 on h323 gateway

Hello,
I have an H323 gateway configured with primary and alternate gatekeepers. Priorities were not configured, so both gatekeepers SHOULD show as 127. However, the first gatekeeper configured shows a priority of 127, the other gatekeeper shows a priority of 0.
I am able to connect to both gatekeepers from the gateway.
I did some research and see that there are some bugs out there on earlier versions which cause the priority to show incorrectly. but i'm wondering if the bug would only be cosmetic (show the priority as incorrect) or if the priority is actually incorrect.
This is happening on multiple gateway / gatekeeper configurations all with the same gatekeeper priority default configuration and showing the same 0 / 127 priorities in the show gateway
I am wondering if the registration failure alerts is due to the priority showing as 0, and if the fix might be to manually configure the gatekeepers' priority.
Has anyone else seen this issue?
H.323 ITU-T Version: 4.0 H323 Stack Version: 0.1
H.323 service is up
Gateway GATEWAY is registered to Gatekeeper GATEKEEPER-PRI
Alias list (CLI configured)
H323-ID GATEWAY
Alias list (last RCF)
H323-ID GATEWAY
H323 resource thresholding is Disabled
Permanent Alternate Gatekeeper List
priority 0 id GATEKEEPER-ALT ipaddr 10.76.155.172 1719 register needed
priority 127 id GATEKEEPER-PRI ipaddr 10.76.155.171 1719 register needed
Primary gatekeeper ID GATEKEEPER-PRI ipaddr 10.76.155.171 1719
Showing alerts in logs:
074975: Mar 14 10:45:51.195 EDT: %CCH323-2-GTWY_REGSTR_FAILED_ALT_GK: Gateway GATEWAY failed attempt to register with Alternate Gatekeeper GATEKEEPER-ALT
074976: Mar 14 10:45:51.251 EDT: %CCH323-6-REGSTR: Gateway GATEWAY registered with Gatekeeper GATEKEEPER-PRI
074994: Mar 14 10:46:06.715 EDT: %CCH323-2-GTWY_REGSTR_FAILED_ALT_GK: Gateway GATEWAY failed attempt to register with Alternate Gatekeeper GATEKEEPER-ALT
074995: Mar 14 10:46:06.771 EDT: %CCH323-6-REGSTR: Gateway GATEWAY registered with Gatekeeper GATEKEEPER-PRI
Configuration:
interface Loopback0
ip address 10.174.102.171 255.255.255.255
ip pim sparse-dense-mode
h323-gateway voip interface
h323-gateway voip id GATEKEEPER-PRI ipaddr 10.176.155.171 1719
h323-gateway voip id GATEKEEPER-ALT ipaddr 10.176.155.172 1719
h323-gateway voip h323-id GATEWAY
h323-gateway voip tech-prefix 700#
h323-gateway voip bind srcaddr 10.174.102.171
Thank you!

BTW - my router is a 3845, IOS version: 12.4(15)T9

Can not add H323 gateway again

Hi all,
I am doing lab with CUCM 7.1. So, I added one h323 gateway (10.15.242.59) and then I deleted it. Now, I add new h323 gateway (same Ip address 10.15.242.59), one message displays:
"Add failed. One of the required fields on the page has the same value as an entry that already exists in the database. Please check the corresponding Find List page to verify your entry does not exist."
But I also check in route group and gateway list, there are no one which has this IP address.
Thanks,
Thuc

Hi.
You can also check if device still exist in the DB by issuing:
run sql select * from device where name='10.15.242.59'
and see if you obtain some result.
HTH
Regards
Carlo
Please rate all helpful posts
"The more you help the more you learn"

Callmanager 3.3(5), H323 gateways not rolling over

Outbound calls are not overflowing from one PRI on one h323 gateway to another. Once all 23 channels are used, I get a busy tone.
I have two h323 gateways. Both are cisco 3725 routers and each has a PRI on it. Each H323 is in theri own route group. And the Route-List contains only these two Route-groups. So it's a pure h323 route-list.
I have tried reversing the order to verify that outbound calls work on both PRI and they do.
I opened a case with cisco and mentioned changing the service parameter of "Stop routing on user busy flag" from true to false. We did that and it's still not working.
Wondering if anyone is having the same experience or have any ideas.

This looks like a pretty old post but this has bitten me once before and there is a fix. In CUCM Service Parameters under Cisco Callmanager Active/ Clusterwide Parameters/Route Plan - Change
"Stop Routing on Unallocated Number Flag" and "Stop Routing on User Busy Flag" to False. This will allow H323 gateways and SIP Trunks to advance to the next Gateway or Trunk in the Route Group/Route List. If it is set to True the backup Gateways and or SIP Trunks will never be used.
Thanks.

Valcom h323 gateway.

Has anyone connected a Valcom intercom system to a h323 gateway. I have a Valcom v-2000a connected to the speakers and then it is connected to the gateway. I have tried FXO port and FXS, when i dial the page number, i can see the FXO port pickup but then it goes to fast busy, when connected to the FXS port, i hear silence, when i hang up and try again all i get is a busy signal. Can someone assist me in getting this to work properly?
thanks,

My solution was FXO, created a dial-peer, the catch with mine is that the Valcom installer left the power brick connected to the 66 block, once I removed their power brick everything started working with mine.

H323-gateway voip bind srcaddr

Friends,
I am working on voip setup. I need to know about the command h323-gateway voip bind srcaddr. In which scenario this command is used? Actually i have an ospf core network. Each router is accessible from other router through multiple links. Regarding this command i think, if i create a loopback interface on all routers and this interface would be the source of voice packet and destination would be the loopback of the target router then my voice call will never drop if any link down the network. Because loopback would be accessible from everywhere.
Need your responses
Best Regards

This command is used on an H.323 gateway when you want to specify the source address for H.323 packets. Yes, you can certainly create a loopback address and source H.323 packets from this interface. This would help with the resiliency of your network as you describe.
Hope this helps. If so, please rate the post.
Brandon

H323 Gateway Redundancy In CCM

HI
I would like to configure 2 h323 gateway on the cisco call manager,Each gateway's have got 2 E1's.
I would like to know is it possible to configure redundancy on the ccm ? if we can how the ccm will come to know the first gateway is down or fully utilized..?
Please advise me...
Thanks.
Nazeer

yes, it's possible, this is from a similar post gogasca replied:
The default behavior in IOS, when there are no valid POTS dial-peers for a call to go out of, is to return âunallocated numberâ as a cause code to CCM. This can happen even if a t1 goes down. When the T1 goes down, the POTS peer is marked as down, and when CCM sends a call to the gateway, it'll return a UAN cause code to the ccm, causing CCM (by default) to stop hunting for other available gateways. There are ways to change the behavior using service parameters in CCM, but this behavior didn't make sense to me, so we looked for other ways to do it.
It turns out that you can issue the global command "no dial-peer outbound status-check pots" on the IOS GW, if you're opposed to changing the CCM behavior. What this command will do is cause the dial-peer to stay up. IOS will try and route the call, and when the T1 is down, it returns "No circuit available" to CCM. When CCM receives this cause code, it knows there's been a non-user error, and continues hunting, achieving the desired behavior.
In your case since it is H323 it will be retruning only H225 release comp message with proper code to continue re-routing.
The IOS code will translate any code we get from pots side to IP side
also, if possible, please try searching within netpro first as it's very probable someone already asked the same before
HTH
javalenc
if this helps, please rate

Plugging fax server into h323 gateway

Hi All
we currently have a h323 gateway used for our outgoing calls via ISDN
We have a requirement for a fax server which has an isdn30 card in it to connect using our main isdn lines
Is it possible to use the other e1 port on this h323 gateway so the fax server can dial out ?
cheers

Hi Carl.
With a correct call routing configuration on VG, you can use a second E1 port to let your fax to dial out through another available E1 port.
Regards
Carlo

Reg. Integrating Gateway with SAML.

Hi All,
Is it possible to integrate NetWeaver Gateway with 2 different SAML.
For example let us assume that we have 2 different AD and 2 different SAML on top of AD ( Please assume that these SAML/AD will not converge).
Is it possible for to configure one gateway to multiple SAML?
Can any one help me on this.
Thanks
-Arun

Hi,
I have a similiar solution running
callmanager 3.3(4)SR2
Siemens DX PABX
2651XM 12.3(6a) AIM Mobule
MGCP
Router is configured as master
PBX is connected via E1 using Q931 NOT QSIG
I did not install this solution but i understand that they had issues running QSIG and then back tracked to Q931.I don't know the exact issues as it was before my time.
It all works fine , but obviously we lose supplementary services.
Mark

Caller ID not working on H323 Gateway

Hi,
I have a CCM Version 4.1.3 set-up to cater for 200 phones, I have a VG gateway in one of the sites that uses an FXO card for local calls, this VG is set-up as an H323 gateway, however calls coming throguh the PSTN line onto the FXO card and then onto an IP phone are not displaying Caller ID.
Just wondering if anyone has seen this before, I know that on an MGCP gateway the FXO cards don't support Caller-ID but we have it set-up as H323.
Thanks
Paul

Funny, this is a question straight from one of the CCVP tests, although I won't say which one.
Try "caller-id enable" in global mode and "station-id name" on the voice-port.
http://www.cisco.com/en/US/tech/tk652/tk653/technologies_configuration_example09186a00800a9a49.shtml

H323 gateway with PABX Avaya

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