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"resource-pool enable" + "pots -> voip gateway" = FAIL

Hi!

I am observing a strange problem, which appears to be never solved by anyone (I did some internet research with no luck).

What I have is AS5350XM with roughly the following configuration (I'll post the relevant "sh run" snippet below, now it's just the textual description):

- 2 E1 interfaces which are connected to outside exchange and inside PBX, using ISDN (PRI)

- one of the numbers in the numbering plan, dialable from the outside, is dial-in modem (dial-peer data xxxx pots)

- Other dialpeers and translations are configured in such way that calls from the outside are cross-connected to an internal PBX and vice versa.

Now I'm adding another voip dialpeer which is basically this:

dial-peer voice 1000 voip

huntstop

preference 10

destination-pattern .T

voice-class codec 1

session target ipv4:xxx.xxx.xxx.xxx

Now, if I disengage the outside E1 interface, my calls will be routed to my backup setup at some gateway.

At some point I discovered that external dial-up users can max out the E1 line. So I'm trying to use resource-pools for limiting:

resource-pool enable

resource-pool call treatment resource busy

resource-pool call treatment profile busy

resource-pool call treatment discriminator busy

!

resource-pool group resource isdn-ports

range limit 60

!

resource-pool group resource spe-ports

range port 1/0 1/55

!

resource-pool profile customer dialup

limit base-size 25

limit overflow-size 0

resource isdn-ports digital

resource spe-ports digital

resource spe-ports speech

dnis group dialup

!

resource-pool profile customer customer-default

limit base-size all

limit overflow-size 0

resource spe-ports digital

resource isdn-ports digital

resource isdn-ports speech

resource spe-ports speech

dnis group default

(and dnis group dialup defines the dialup phone number).

When I enable resource-pool, the limiting starts working, but calls from PSTN to VOIP stop.

It looks like outgoing calls also got resource-limited somehow but I cannot find anything relevant in documentation (which explicitly says that outgoing calls are not subject to RPM).

Can anyone give any clues, why I'm failing? I'm sure that the answer is pretty obvious I just can't see it... thanks.

The other relevant bits of my config:

controller E1 3/0

framing NO-CRC4

pri-group timeslots 1-31

cablelength short

description VT

!

controller E1 3/1

framing NO-CRC4

pri-group timeslots 1-31

description PBX

interface Serial3/0:15

no ip address

encapsulation hdlc

isdn switch-type primary-net5

isdn incoming-voice modem

isdn send-alerting

isdn negotiate-bchan resend-setup

no isdn outgoing display-ie

!

interface Serial3/1:15

no ip address

encapsulation hdlc

isdn switch-type primary-net5

isdn protocol-emulate network

isdn incoming-voice modem

isdn send-alerting

isdn negotiate-bchan resend-setup

no cdp enable

voice-port 3/0:D

translation-profile incoming normalize-pstn

translation-profile outgoing drop-pstn

disc_pi_off

cptone RU

bearer-cap Speech

!

voice-port 3/1:D

translation-profile incoming normalize-os500

translation-profile outgoing drop-os500

disc_pi_off

cptone RU

bearer-cap Speech

1 REPLY
Silver

Re: "resource-pool enable" + "pots -> voip gateway" = FAIL

AS5400 platforms are normally installed as dial-in data, voice, fax, or modem access servers. In order to terminate speech type calls (voice, fax, or modem) the access server needs appropriate any service, any port (ASAP) Digital Signal Processor (DSP) resources to be installed.

If the modem, fax, or voice calls are not actually required to be terminated on the access server, but for some reason need to be switched back out to alternate ports, it is possible to configure the AS5400 to act in a purely TDM application where voice call switching is controlled via ISDN D-channel signaling. Data or speech calls can be switched based on the DNIS (called number) through to another interface. Effectively, the access server becomes a TDM voice/data switch. This feature is often called TDM switching, although other names such as hairpinning, tromboning, or dial-grooming are also applied to the technique. Generally, the terms are interchangeable and for this document, the term TDM switching is used. There are no dual tone multifrequency (DTMF) or multifrequency (MF) signaling tones passed with ISDN. Call control is done with High-Level Data Link Control (HDLC) encapsulated D-channel messages. Therefore, there is no need for the DSP resources for voice calls when in the TDM mode of operation.

The access server uses an incoming DNIS (called number) to match on an outgoing POTS dial-peer destination pattern and routes the call out an appropriate port. It is possible to use IOS translation rules to manipulate the called and calling numbers for call routing decisions as well.

Applications of TDM switching can include an access server acting as a small ISDN data/voice exchange (using ISDN network side protocol emulation), or call rerouting via alternate carriers (least cost).

This document describes how to configure an AS5400 to perform TDM switching for voice and data calls. Based on matches made on the DNIS for the incoming call (provided in the ISDN Q.931 setup message), the call is switched from one interface to an alternate interface. The technique also works on other platforms that use TDM backplanes such as the AS5350 and the AS5850.

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