ATA (SCCP)

[mightyking]

Hi Experts,

We have some issues with our ATAs since we migrated our GWs from MGCP to H323.

-Line 2 on ATA has issues both inbound and outbound

-Instance of not getting dial tone on one or both ports

Here're some of the settings:

LBR Codec          3

AudioMode          0x00350035

ConnectMode     0x90000400

Thanks,

MK

[mightyking]

Anybody with any ideas?

[mightyking]

Still looking for some help.

[Avner Izhar]

Have you tried skinny? It may make things easier then mgcp or h.323 .

In any case, no dial tone has nothing to do with codec since the dial tone is generated locally by the ATA.

If the problem happens after dialing, then I would look at the call termination point (pstn gateway perhaps) and try to find out what is going on.

Kind of hard to be specific without more details here, hope that helps.

Sent from Cisco Technical Support iPhone App

[mightyking]

Thanks for the reply. As I mentioned the ATA is skinny (SCCP) and the PSTN GW is H323. Please let me know if you need more details.

[Avner Izhar]

ok, in that case what directory number is assigned to one of the ATA that has issues?

also, please post the output of 'show dial-peer voice summary' from the H.323 gateway.

thx, Avner

[mightyking]

Hi,

Here's the dial peer configuration

------------------------------------------

dial-peer voice 1 voip

  destination-pattern 234567....

progress_ind setup enable 3

modem passthrough nse codec g711ulaw

voice-class h323 1

session target ipv4:X.X.X.X

dtmf-relay h245-signal h245-alphanumeric

fax-relay ecm disable

fax rate 14400

fax protocol t38 nse ls-redundancy 0 hs-redundancy 0 fallback pass-through g711ulaw

ip qos dscp cs3 signaling

no vad

!

dial-peer voice 2 voip

preference 1

destination-pattern 234567....

progress_ind setup enable 3

modem passthrough nse codec g711ulaw

voice-class h323 1

session target ipv4:X.X.X.X

dtmf-relay h245-signal h245-alphanumeric

fax-relay ecm disable

fax rate 14400

fax protocol t38 nse ls-redundancy 0 hs-redundancy 0 fallback pass-through g711ulaw

ip qos dscp cs3 signaling

no vad

!

dial-peer voice 130 voip

modem passthrough nse codec g711ulaw

voice-class h323 1

session target ipv4:X.X.X.X

incoming called-number .

dtmf-relay h245-signal h245-alphanumeric

fax-relay ecm disable

fax rate 14400

fax protocol t38 nse ls-redundancy 0 hs-redundancy 0 fallback pass-through g711ulaw

ip qos dscp cs3 signaling

no vad

!

dial-peer voice 110 voip

modem passthrough nse codec g711ulaw

voice-class h323 1

session target ipv4:X.X.X.X

incoming called-number .

dtmf-relay h245-signal h245-alphanumeric

fax-relay ecm disable

fax rate 14400

fax protocol t38 nse ls-redundancy 0 hs-redundancy 0 fallback pass-through g711ulaw

ip qos dscp cs3 signaling

no vad

[Avner Izhar]

Ok thx, looks like u are using the default codec (g.729) under the dial peers. The ATA can only support 729 on port 1, if it's not a problem u should add a 'voice class codec' config, enable 711 and 729 in it and assign it to the dial peers.

This should help, if not send the output of 'debug VoIP dialpeer' for a failed call .

Thanks

[mightyking]

Thanks Avner,

I will apply the config and let you know.

MK