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
t35CountryCode: United States (181)
H.221 Manufacturer: Cisco (0xb5000012)
Data (10 bytes)
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
fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711ulaw
no telephony-service ccm-compatible
voice class codec 1
codec preference 1 g711ulaw
dial-peer voice 931 voip
description H323 Trunk to CM SLC
voice-class codec 1
session target ipv4:10.1.50.5
dtmf-relay h245-alphanumeric h245-signal rtp-nte
I have also attached the wireshark, the packet with the message in question is 206.
I appreciate any assistance.
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.
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.