Can not Call site to site

[vanagon2tdi]

We have a Cisco 3745 as our PRI gateway and are having troubles with not being able to call site to site. The sites can call out to PSTN and PSTN can call site with no problem.

The system is setup as below with Ethernet between Cisco GW and Sites.

PSTN------Cisco GW-------SiteA Phone Number 5551234

!

!

SiteB Phone number 4441234

Sorry Diagram not working, SiteB should be connected directly to Cisco GW.

On the GW there are the correct dialpeers, otherwise SiteA would not be able to make calls to PSTN and other way around.

Any ideas why I cannot call from SiteA to SiteB?

[Brandon Buffin]

Do you have voip dial peers on each gateway pointing to the other gateway for calls between sites? Can you post the config for the two gateways?

Brandon

[vanagon2tdi]

Here is the configs from Site A, Site B and the Cisco GW.

Site A

dial-peer voice 200 voip

huntstop

max-conn 2

destination-pattern .T

session target ipv4:10.254.253.2 (Cisco GW)

dtmf-relay cisco-rtp

fax rate 9600

ip qos dscp cs5 media

no vad

!

dial-peer voice 100 pots

huntstop

destination-pattern 5551234

port 2/0

!

Site B

dial-peer voice 200 voip

huntstop

max-conn 2

destination-pattern .T

session target ipv4:10.254.253.2 (Cisco GW)

dtmf-relay cisco-rtp

fax rate 9600

ip qos dscp cs5 media

no vad

!

dial-peer voice 100 pots

huntstop

destination-pattern 4441234

port 2/0

!

Cisco GW

dial-peer voice 555 voip

huntstop

destination-pattern 5551234

session target ipv4:10.7.7.222 (Site A)

dtmf-relay cisco-rtp

fax rate 7200

ip qos dscp cs5 media

no vad

!

dial-peer voice 444 voip

huntstop

destination-pattern 4441234

session target ipv4:10.5.5.100 (Site B)

dtmf-relay cisco-rtp

fax rate 7200

ip qos dscp cs5 media

no vad

!

[vanagon2tdi]

What seems to be happening is that the call is matching on the outgoing dial-peer on the Cisco GW and trying to go to the PSTN instead of Site A (or B). Even though, when I am on the GW and do a

Show dialplan number 5551234 it gives me the dialpeer 555 as it's first match?

Here is the debug

May 11 19:27:37.189: //-1/E8307843-2BF5-11D6-808C-CED82A058B01/DPM/dpAssociateIncomingPeerCore:

Match Rule=DP_MATCH_INCOMING_DNIS; Called Number=5551234

May 11 19:27:37.189: //-1/E8307843-2BF5-11D6-808C-CED82A058B01/DPM/dpMatchPeertype:

Is Incoming=TRUE, Number Expansion=FALSE

May 11 19:27:37.189: //-1/E8307843-2BF5-11D6-808C-CED82A058B01/DPM/dpMatchCore:

Dial String=5551234, Expanded String=5551234, Calling Number=

Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH

May 11 19:27:37.189: //-1/E8307843-2BF5-11D6-808C-CED82A058B01/DPM/dpMatchCore:

Result=-1

May 11 19:27:37.189: //-1/E8307843-2BF5-11D6-808C-CED82A058B01/DPM/dpAssociateIncomingPeerCore:

Match Rule=DP_MATCH_ANSWER; Calling Number=4441234

May 11 19:27:37.189: //-1/E8307843-2BF5-11D6-808C-CED82A058B01/DPM/dpMatchPeertype:

Is Incoming=TRUE, Number Expansion=FALSE

May 11 19:27:37.189: //-1/E8307843-2BF5-11D6-808C-CED82A058B01/DPM/dpMatchCore:

Dial String=, Expanded String=, Calling Number=4441234T

Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH

May 11 19:27:37.193: //-1/E8307843-2BF5-11D6-808C-CED82A058B01/DPM/dpMatchCore:

Result=-1

May 11 19:27:37.193: //-1/E8307843-2BF5-11D6-808C-CED82A058B01/DPM/dpAssociateIncomingPeerCore:

Match Rule=DP_MATCH_ORIGINATE; Calling Number=4441234

May 11 19:27:37.193: //-1/E8307843-2BF5-11D6-808C-CED82A058B01/DPM/dpMatchPeertype:

Is Incoming=TRUE, Number Expansion=FALSE

May 11 19:27:37.193: //-1/E8307843-2BF5-11D6-808C-CED82A058B01/DPM/dpMatchCore:

Dial String=, Expanded String=, Calling Number=4441234T

Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH

May 11 19:27:37.193: //-1/E8307843-2BF5-11D6-808C-CED82A058B01/DPM/MatchNextPeer:

Result=Success(0); Incoming Dial-peer=999 Is Matched

May 11 19:27:37.193: //-1/E8307843-2BF5-11D6-808C-CED82A058B01/DPM/MatchNextPeer:

Result=Success(0); Incoming Dial-peer=991 Is Matched

May 11 19:27:37.193: //-1/E8307843-2BF5-11D6-808C-CED82A058B01/DPM/MatchNextPeer:

Result=Success(0); Incoming Dial-peer=992 Is Matched

May 11 19:27:37.193: //-1/E8307843-2BF5-11D6-808C-CED82A058B01/DPM/MatchNextPeer:

Result=Success(0); Incoming Dial-peer=990 Is Matched

May 11 19:27:37.193: //-1/E8307843-2BF5-11D6-808C-CED82A058B01/DPM/MatchNextPeer:

Result=Success(0); Incoming Dial-peer=555 Is Matched

May 11 19:27:37.193: //-1/E8307843-2BF5-11D6-808C-CED82A058B01/DPM/dpAssociateIncomingPeerCore:

Match Rule=DP_MATCH_INCOMING_DNIS; Called Number=5551234

I have the config posted now.

[Brandon Buffin]

If Site A calls Site B (and vice versa), where should the call go?

Brandon

[vanagon2tdi]

It should go to the site as the GW has a dialpeer pointing that phone number to the ip of that site.

In some instences if i put a dialpeer on Site A telling it where Site B's phone numbers are (ip adress) then it will work. The thing is we have over 1000 different sites each with an average of 4 phone numbers, so it would be impossible to put all these dialpeers on all these routers.

[Brandon Buffin]

The functionality that you want is provided by a gatekeeper. A gatekeeper provides address translation as well as call admission control. Take a look at the following links.

VoIP with Gatekeeper

http://www.cisco.com/en/US/tech/tk1077/technologies_configuration_example09186a0080117768.shtml

Understanding Cisco IOS Gatekeeper Call Routing

http://www.cisco.com/en/US/tech/tk1077/technologies_tech_note09186a00800a8928.shtml

Brandon

[comsatbrazil]

We also found the same behavior connecting ATAs to a Cisco gateway. When connection comes from IP, and even if there is a dial-peer voip for the called number, the call is not sent to the destination ATA. If this dial-peer is a pots one, the call is sent to the voice port.

It seems that the router doesn´t support IP to IP calls through the gateway. Is there any way to bypass this restriction without the Gatekeeper?

I tested it also with SIP, pointing the ATAs to the router, not using a SIP server, and found the same situation. NO ATA to ATA calls through router. If we have a PBX in the gateway, this call can be routed through the PBX, but you occupy 2 PBX lines to make it work.

[comsatbrazil]

I found in Cisco Web site a document called

"Understanding Dial Peers and Call Legs on Cisco IOS Platforms" that clearly tells that this configuration is not supported. Calls originating and terminating on the same router are called "hair-pinning",and are supported only for POTS to POTS, but not for VOIP to VOIP, which is our case. Following is the part of document where it is described:

"Note: Hair-Pinning is the name given to calls that originate and terminate on the same router/gateway. On POTS-to-POTS Hair-Pinning calls, the router/gateway matches an inbound POTS dial-peer and an outbound POTS dial-peer to terminate the call. This is supported on POTS interfaces. However, VoIP-to-VoIP Hair-Pinning is not supported on Cisco IOS voice-enabled platforms except in CallManager Express with certain IOS releases"

[vanagon2tdi]

Wow, Thanks everyone for your feed back. Good to hear that I have not just made an error.

So can anyone think of a work around for this? My Gateway is connected straight to the PSTN via several PRI T1 interfaces, so a router in between them would be costly.

[Brandon Buffin]

The cost would probably be less than you might imagine. All you need is an IOS router with the proper IOS/feature set capable of gatekeeper functionality. A single gatekeeper can control multiple gateways. The upfront cost starts to make financial sense when compared to the costs associated with configuring and maintaining a dial plan for 1000 sites. Another benefit the gatekeeper will provide is call admission control. Without some form of CAC, your voice quality will quickly degrade as WAN links become oversubscribed.

Brandon

[vanagon2tdi]

So I looked at those links that othters send and they say that my IOS must have an "x" in the name to have the Gatekeeper functionality. I am having trouble finding out which router to use (hoping for 2801) and what IOS version.

Think i just found out that I need at least a 2811.