Issue with ooh323 connecting Avaya <-> Asterisk H323 t


#1

I am using H323 trunking between an Avaya CM and an Asterisk v1.2 instance. I am dialing as follows:

Eyebeam -> Asterisk -> Avaya -> Asterisk -> Avaya 4602SW IP hardware phone

When the phone rings I answer with no audio and get the following:

[quote=“Asterisk CLI”]-- Executing Dial(“H323/Jose Luis Casta-1e35”, “SIP/50008|20|tT”)
– SIP Seeding peer from astdb: ‘50008’ at 50008@89.1.211.201:1098 for 180
– Called 50008
– SIP/50008-6755 is ringing
– SIP Seeding peer from astdb: ‘50008’ at 50008@89.1.211.201:1098 for 180
– H323/89.1.250.101-5d1b is ringing
– H323/89.1.250.101-5d1b answered SIP/50001-9394
– Attempting native bridge of SIP/50001-9394 and H323/89.1.250.101-5d1b
– SIP/50008-6755 answered H323/Jose Luis Casta-1e35
[color=red]Oct 10 16:21:50 WARNING[6465]: src/chan_h323.c:915 h323_indicate: Don’t know how to indicate condition -1 on ooh323c_4[/color]
– Attempting native bridge of H323/Jose Luis Casta-1e35 and SIP/50008-6755[/quote]

If I dial as follows:

Avaya Digital Set -> Avaya -> Asterisk -> Avaya 4602SW IP

I get audio but the same error as above. Ideas?

On a sidenote, all endpoints and switches have alaw enabled as I am in Spain.


#2

I am having the same issue as you are, but, in my case, I am always without audio and the call is droped right after the called extension picks up the phone to asnwer.

It seems to be related to something calling ooh323_indicate with condition = -1, but I could’nt find what is wrong until now.


#3

I am on the latest v1.2beta2/CVS HEADs and doing more tests. I will post as I progress this issue.