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New Member

No VoIP forwarding on incomming calls on PRI on AS5350

Hi we have a problem with VoIP calls not being routed. The call is supposed be a GW to GW call and we can't even see the call being registred as a VoIP call. We get a Requested circuit/channel not available error on the ISDN debug. No h323 packet exits the gateway. Incomming calls from Internet works, though.

The IOS version we are using is 12.2(2)XB5, config:

interface FastEthernet0/0

ip address xx.xx.xx.xx 255.255.255.224

no ip redirects

no ip proxy-arp

duplex full

speed 100

no cdp enable

h323-gateway voip interface

h323-gateway voip id GK-1-AraT ipaddr xx.xx.xx.xx

h323-gateway voip h323-id gw-1-madrid

h323-gateway voip bind srcaddr xx.xx.xx.xx

interface Serial3/4:15

no ip address

no snmp trap link-status

isdn switch-type primary-net5

isdn timer t321 40000

isdn protocol-emulate network

isdn incoming-voice modem

isdn T310 60000

isdn send-alerting

isdn sending-complete

no isdn outgoing display-ie

fair-queue 64 256 0

no cdp enable

dial-peer voice 1006 voip

destination-pattern 00%

progress_ind connect enable 2

translate-outgoing called 8100

voice-class codec 3

session target ipv4:xx.xx.xx.xx

We can ping the session target. We are at a loss here and have looked in and searched the CCO and nothing works.

Any help will be much appreciated.

Thanks,

Niels

1 REPLY
New Member

Re: No VoIP forwarding on incomming calls on PRI on AS5350

Your problem is in the destination-pattern of the dial-peer.

Percent sign (%), which indicates that the previous digit/pattern occurred zero or multiple times, similar to the wild card usage in the regular expression.

You should have destination-pattern 00T. Also, after you receive the call on the isdn side you should verify with show call history voice brief. You should see two call legs : one listed with pid:X (where X is the dial-peer for pots) and one with pid:1006.

Anyway, your pots dial-peer should look like this

dial-peer voice 1 pots

application your-iwr-appl (or session)

port 3/4:D

so you get the call answered when calling from the pstn side :)

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