Troubleshooting one-way audio with H.323 trunk between Asterisk and Avaya IP Office

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I’m attempting to configure an H.323 trunk (using chan_h323) between an Asterisk box and an Avaya IP office. It mostly works. Calls from Polycom SIP devices registered to Asterisk can place calls over the trunk to IP Office extensions and everything works great. However, calling from an IP Office handset to any of the Polycoms results in a one-way call where the Polycom can not hear the Avaya.

I think what’s happening is somehow, the Polycom is receiving two RTP streams, and one of them is silence. I think this because if I place the call on hold, I will hear, occasionally, short bursts of what could be the hold music on the Polycom. Also, when I look at packet captures taken from the Polycom’s port, I see two streams when it’s working (the Polycom calls Avaya) and three when it’s not (Avaya calls Polycom).

I do notice that this “extra” stream has a unique SSRC. If I do the packet captures from the Asterisk box, it looks like the extra stream is generated by Asterisk. Graphically, the RTP streams and their SSRCs I see on a working call (Polycom calls Avaya) looks like this:

Polycom –0x123-> Asterisk –0x123-> IP Office
< -0x456-- <-0x456 The non-working call (Avaya calls Polycom) looks like: <-0x789--
Polycom –0x123-> Asterisk –0x123-> IP Office
< -0x456-- <-0x456 However, I'm new to Asterisk, and I'm not very familiar with any of these VoIP protocols, so I find myself stuck. Can anyone suggest some troubleshooting steps or material I might read which would help me to determine if my hypothesis is correct, and if so, how I can determine what's responsible for this extra stream? Console output from a non-working call with sip and h323 debug and trace on follows. Here, 207 is the Avaya handset originating the call, and 216 is the Polycom receiving it. 172.20.20.233 is the IP Office, 172.20.20.205 is Asterisk, and 172.20.32.70 is the Polycom. asterisk01*CLI>
== New H.323 Connection created.