IP 406V2 with 2 Subnets

[malchick]

I've got port 8 enabled as LAN 2 on an IP 406 V2. Connected to an MPLS, I'm trying to place calls on a voip trunk to a second branch. I am able to ping from the private lan (LAN 1) to the branch, and from the branch to the private lan.

With H323 tracing enabled, when I try to place a call to the branch, I get a H323 RX back from the branch:

H323Rx: 71 src=192.168.x.x:1720
H323 Pcol=08(Q931) Reflen=2 ref=001E(Local)
Message Type = ReleaseComplete

The destination address on the RX from the branch is also to the Public interface, (LAN2 at the main site). I'm thinking it should be LAN 1, where my call originated?

Also should the H323 gatekeeper be enabled on LAN2?

Thanks for any suggestions.

Interestingly, calls are successful from the branch to the main office. Using shortcodes for the moment to route the calls.

 

[amriddle01]

It will be an issue with your IP routes, or does ping traverse the link via the IP office rather than some other equipment?

ACSS (SME)
APSS (SME)

http://www.ipoffice.tel

 

[malchick]

LAN 2 on the IP office is connected directly to the MPLS provider router. I've disabled/enabled fw on LAN2 but has not change outcome. At the moment have TCP UDP and ICMP enabled in and out.

 

[amriddle01]

Let's call them site A and B, are you saying site A can call and ping site B but site B can't call site A but can ping it? And both site A and B are using the IP offices to route all traffic accross the MPLS link ?

ACSS (SME)
APSS (SME)

http://www.ipoffice.tel

 

[malchick]

That is correct> But lets reverse it: Site A has two interfaces LAN1 and LAN2, with the IPO gatekeeper on LAN 1.
I can ping Site B from Site A but can't call. From Site B I can call or ping Site A. IPOs at both sites are connected directly to MPLS router.

 

[amriddle01]

List the IP routes from Site A and B and the addresses of their LAN ports (even edited slightly will do)

ACSS (SME)
APSS (SME)

http://www.ipoffice.tel

 

[amriddle01]

Also, the configuration of the IP trunks from both sites and the SC's used to dial site to site may help

ACSS (SME)
APSS (SME)

http://www.ipoffice.tel

 

[malchick]

IP routes:

SiteA

Destination Subnet NextHop IFName Metric RouteType
A.B.2.0 /24 LAN1 Attached
0.0.0.0 /0 A.B.2.1 LAN1 1 Static
E.F.40.0 /24 LAN2 Attached
E.F.20.0 /24 E.F.40.1 LAN2 1 Static


SiteB

Destination Subnet NextHop IFName Metric RouteType
E.F.20.0 /24 LAN1 Attached
0.0.0.0 /0 E.F.20.1 LAN1 1 Static

From Config @ SiteA:
IP Line :
Line Number 10
Incoming Group ID=10
Outgoing Group ID=10
Channels=5
Outgoing Channels=5
VoiceChannels=5
Data Channels=5
No SCs for Line

Voip Settings:
Gateway= E.F.20.10
G729
H450 support
Call initiation timeout=4

Options:
Local Tones Enabled, all others disabled.

Call Routing ShortCode
86xx
DIAL
86N
Line ID 10

 

[malchick]

Sorry forgot to add:

Site A LAN ports:

LAN1
A.B.2.3
LAN2
E.F.40.2


Site B LAN port:
E.F.20.2

 

[amriddle01]

In site A's VoIP connection settings you have the gateway as 20.10 but it needs to be 20.2 (site B's LAN port)

ACSS (SME)
APSS (SME)

http://www.ipoffice.tel

 

[malchick]

Thanks for your time amr. Sorry I made a typo in that last post. Site B's LAN address is E.F.20.10.

 

[malchick]

A followup question re: LAN1 and LAN2 on IPO 406V2
Should the H323 gatekeeper be disabled on LAN2 if that interface is being only being used to route to a gatekeeper in another office. Call processing is already being initiated by H323 gatekeeper on LAN1. I'm having a problem getting calls from a main office to a branch office. If I look sysmon output from the branch office, the call setup seems to proceed up to a point, where the h245 handshaking is about to begin, and then there is an error: "didn't find extension by name ..." referring to the extension that the call was initiated from at the main site. thanks for any suggestions.

 

[amriddle01]

No, Gatekeeer is there to serve registering and running IP Phones not a link between two sites, send a copy of the .cfg file(config) for each site and I will look at them for you, send them to my home address from the site in my signature. Feel free to edit Inoming call routes and system name etc but leave shortcodes and relavant things in place (there isn't much I could do with an IP address even if I wanted to)

ACSS (SME)
APSS (SME)

http://www.ipoffice.tel

 

[Bas1234]

Explain more here;

"The destination address on the RX from the branch is also to the Public interface, (LAN2 at the main site)."

___________________________________________
It works! Now if only I could remember what I did...

Dain Bramaged
___________________________________________

 

[malchick]

Thanks amriddle, Will do.

BAS1234 the sysmon trace files from the remote site indicate that reply to the h323 call setup request, is addressed to the gatekeeper on LAN2, instead of the gatekeeper on LAN1, where the call originated. Why I was wondering if I should disable it.

 

[Bas1234]

Post the trace here.

___________________________________________
It works! Now if only I could remember what I did...

Dain Bramaged
___________________________________________

 

[amriddle01]

The config looks all ok to me, you need to post a trace/traces as Bas says, post a monitor trace showing a working call from B to A and then a failed call from A to B

ACSS (SME)
APSS (SME)

http://www.ipoffice.tel

 

[Bas1234]

You only need 1 H323 trunk programmed in site A and B, also check incomming/outgoing group ID on both sites. Make them both 11 for site A and 12 for site B.

___________________________________________
It works! Now if only I could remember what I did...

Dain Bramaged
___________________________________________

 

[amriddle01]

That side of things is set up correctly Bas, as are the shortcodes and IP routes


ACSS (SME)
APSS (SME)

http://www.ipoffice.tel

 

[TeleEngineer]

Do you have direct media path on or off, it should be off over routed destinations