VG 2900 more Voice VLANS which Interface H323 Binding ?

[armin.wagenfuehr]

Hello Community, we have several constelation regarding networkconfig on several BR site VG2921. The VG2921 also perform the routing on Branch site. But at all BR sites the WAN0 is on VG2921 GE0/0 --- > HQ Router --- > Switch ---> CUCM.

At some sites the VG2921 have 4ESG Switchportmodules and the VLAN Interface XXX is the GW IP for the Phones and GE0/0 is WAN.

At some sites GE0/1 and GE0/2 on VG2921 are connected to analog ( Interface <-- > Interface) VG2XX. The GE0/1 and GE0/2 are the Gateways for the analog VG2XX.

Here the constelations:

BR1:

WAN1 is GE0/0 (IP Route and GW to HQ Router)

Voice LAN130 GW IP is on GE0/1 --- > Connect to a VG202 ( IP Route and GW to the IP address on VGW2921 GE0/1)

Voice LAN131 GW IP is on GE0/2 --- > Connect to a VG204 ( IP Route and GW to the IP address on VGW2921 GE0/2)

BR2:

WAN1 is GE0/0 (IP Route and GW to HQ Router)

Voice LAN140 GW IP is on GE0/1 --- > Connect to a VG224 ( IP Route and GW to IP on VGW2921 GE0/1)

Voice VLAN141

4ESG Switch Modul in VG2921 4x 6941 Connected

Interface VLAN 140

ip address xxx.xxx.xxx.xxx. Subnet xxx.xxx.xxx.xxx

The main Question is, which Interface IP AddressI should make to the

h323-gateway voip bind srcaddr ?

call-manager-fallback ip source-address 172.16.114.22 port 2000 ?

Thanks in ahead

Armin

[Chris Deren · Accepted solution]

Does not really matter as long as the IP is routable from all involved devices, i.e. phones/CUCM.

For SRST you should use the same interface as the H323 bind interface.

Why you have analog gateways plugged into router and not switch is beyond my comprehension as I've never seen that done and don't see how that would make for a good idea.

HTH,

Chris

[armin.wagenfuehr]

Hi Chris, many thanks for your answer. I will use the GE0/1 and on Routers w/ Switchmoduls the Interface VLAN XXX for H323 Binding. it is all the time the first Network behind the WAN GE0/0.

Why we do that? Good question, it is not my Idea! We are work for a Consulter it is not our direct Customer. The Consulter would not spend more Switchmodules for the rest of the VGW. I don´t know why. I prefer also to plug the analog VG2XX into a switchport. Further I have also now Idea, how we could tag the Interface - Interface connection into a VLAN !!!!???

Cheers Armin

[Chris Deren]

Armin,

Thanks for nice rating,  I am not sure if you are asking another question or if you are all set.

"Further I have also now Idea, how we could tag the Interface - Interface connection into a VLAN !!!!???""

Chris

[armin.wagenfuehr]

Hi Chris, thanks to you for confirm my objection.

Maybe you know that, how we tag the Interface Interface connection into the Voice VLAN? The next Switchport Modul is at the HQ site, but we need also a VLAN for QOS on BR site.

At this customer we also provide a CME SRST functionality. It means the Customer would also have full functionality in SRST mode regarding for example Hunting (Huntpilots -- Huntgroups') and Pick up Groups, as the same they have in CUCM.

We should work with Callmanager - Fallback mode, CME mode or we should start telephonie service and create also all Phones, Huntgrpoups and Pick up Groups on the VGW?

How should work that?

Cheers Armin

[Chris Deren · Accepted solution]

Well, are you using CME as SRST or Enhanced SRST?

With CME as SRST it is manual process which is documented in CME Admin guide, on the other hand E-SRST synchronized everything but requires UMG module:

http://www.cisco.com/en/US/products/sw/voicesw/ps2169/

http://www.cisco.com/en/US/prod/collateral/voicesw/ps6788/vcallcon/ps2169/data_sheet_c78-678873.html

Chris

[armin.wagenfuehr]

Hi Chris, we must use CME SRST, because we have not this kind of Moduls. Thanks to point me to the right documentation.

So we should start telephonie-service not call-manager-fallback, right?

What means the command below regarding mode cme?

voice register global

mode cme

source-address 10.27.32.64 port 5060

max-dn 25

max-pool 10

[Chris Deren]

Correct, telephony-service. Check out this guide for detailed configuration:

http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/admin/configuration/guide/cmesrst.html

HTH,

Chris

[armin.wagenfuehr]

Hi, thanks a lot! Cheers Armin

[armin.wagenfuehr]

Hi Chris, really thanks to point me to the right docu. CME SRST mode WORKS fine!!!!

But we run in a stopping issue and I can´t find a solution. Maybe you could help me again.

We have the following Setup.

PSTN Called Party 938240 --> VIC2BRI  --> VGW 2921 Dial Peer Voice Voip --- > CUCM Strip 938 and prefix 90 and 8961SIP Phone "DN 90240" rings.

All DN´s are 5 digits long. We have also SCCP Phones with 5 digits DN. Internal the User dial only 3 digits for example 240. Works all fine, when register at CUCM. We work with TransPattern in CUCM.

Problem:

But when the SIP and SCCP Phone goes in CME SRST mode and register at the VGW2921, the phones also register with the 5 digits DN. But in CME SRST mode for external incoming calls, I need still to strip 938 and prefix 90 to reach the Ephones with the 5 digits DN and for internal calls I would only dial the last 3 digits.

I have try many ways (dial-pattern TransRules additional dial-peer´s etc...), but all was not sucessfull.

The goal should be, also in CME SRST mode, prefix 90 so that the User should dial only the last 3 digits for internal and for external incoming calls to strip 938 and prefix 90 to reach the 5 digits DN. In normal mode the calls should route to the CUCM.

Please, could you point me to the right solution within an CME Config example for SIP and SCCP Phones?

Cheers Armin

[Chris Deren]

Use dialplan-pattern under SRST to translate 5 digits from PSTN to 3 digit internal.

Use Translation rules to allow 3 digit dialing, apply the profile to SRST in incoming direction.

HTH

Chris

[armin.wagenfuehr]

Dialplan-pattern und telephony-service works fine.

telephony-service

dialplan-pattern 1 938... extension-length 5 extension-pattern 90...

Incoming call from PSTN Called Party 938240 rings at Phone 90240.

But for internal Calls Dial 240 and translate to 90240 works not in telephony-service. You can´t assign a translation profile to telephony-service, only to CCM Fallback SRST, but we work wit CME SRST.

[Chris Deren]

You could build the 3 digit number as secondary number on the ephone.

HTH, please rate all useful posts!

Chris

[armin.wagenfuehr]

Good Idea!!! Thanks for help me.

Meantime I found a solution like that,

voice translation-rule 4

rule 1 /\(^[1-5]..\)/ /90\1/

voice translation-profile intern_srst

translate called 4

ephone-dn  1  dual-line

number 90240

label 90240

description 9842938240

name 9842938240

translation-profile incoming intern_srst

voice register dn  1

translation-profile incoming intern_srst

number 90297

name Laitstelle

label 90297

HTH, please rate all useful posts!

[Chris Deren]

Armin, +5 for posting a good solution.

Chris