H323 Trunk to Avaya with Direct IP Media

[hastings_c]

I have been tasked with configuring a cisco IP to IP voice gateway to bring a SIP trunk in from carrier and pass it into an Avaya CM as an H323 trunk. That was relatively easy to get working, but now I have been told we need to set up direct IP media to bypass the Avaya media gateway and have the RTP stream go directly from the Cisco to the Phone.

The only change that I made was to the Avaya side. As I understand the Cisco will send the audio stream wherever the H323 OLC message dictates.

What is occuring is the call comes in from carrier, rings the phone, and it is answered. There is no audio, and the call disconnects after 15 seconds. A wireshark of the call shows the call completes, the fast-start OLC channel opens, then closes when the media should move directly between the gateway and the phone. The Cisco sends an information message with a non-standard parameter. and the Avaya responds with a status, then never opens a new channel.

Avaya support claims that the issue is the information message with a non-standard parameter. They can't process this message then never contiue with the call.

The information message shows up in wireshark as 

     h221NonStandard

          t35CountryCode: United States (181)

          t35Extension: 0

          manufactureCode: 18

     H.221 Manufacturer: Cisco (0xb5000012)

     Data (10 bytes)

          Data :c0010018800003000121

How do I configure the Cisco IP-IP gateway to not send this non-standard message?

I have made all kinds of changes to the config that I don't understand. Here are the current settings on the Cisco, I have no idea what half this config does or what is needed.

voice rtp send-recv

!

voice service voip

allow-connections h323 to h323

allow-connections h323 to sip

allow-connections sip to h323

allow-connections sip to sip

supplementary-service h450.12

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

h323

  emptycapability

  no telephony-service ccm-compatible

sip

  early-offer forced

  midcall-signaling passthru

voice class codec 1

codec preference 1 g711ulaw

dial-peer voice 931 voip

description H323 Trunk to CM SLC

huntstop

destination-pattern 8013663883

voice-class codec 1

session target ipv4:10.1.50.5

dtmf-relay h245-alphanumeric h245-signal rtp-nte

no vad

I have also attached the wireshark, the packet with the message in question is 206.

I appreciate any assistance.

[Jagpreet Singh]

Hi,

     Please run the following debugs on the IP to IP gateway and make s test call and share the debugs with us:

> debugs ccsip messages

> debug h225 asn 1

> debug h245 asn 1

If possible try the cal out when the call flow is minimal. Also, mention the calling and the called number when you post the debugs.

Regards,

Jagpreet

[hastings_c]

I have grabbed the debugs, but there are some other active calls.

The calling number is 801-261-0505

the called number is 801-366-3883

If I need to get calls during a lower volume time I can do so this evening.

[Jagpreet Singh]

I was able to download the debug and can see the SIP INVITE come in, however cant see the H225 setup message go out. Could you collect the above debugs again and also add the putput for " debug voice ccapi inout".

And these debugs are bulky and put load on your router, thats the reason I wanted you to collect them during low call volume. But, if there was no ill effect when you collected the above, you can collect those again.

Regards,

Jagpreet

[hastings_c]

Here are the debugs.

Thank you for you help.