Avaya IX Messaging: H323 phones (9620) cannot access their voicemail (CM + SM 8.1)

[Dardan]

Hello everyone,

I wish you a happy new year.

When an H323 user tries to access its voicemail (Avaya IX Messaging - Officelinx 10.8, this message appears:

CM trace

11:13:37 TRACE STARTED 01/09/2020 CM Release String cold-01.0.890.0-25578
11:13:40 active station 3495 cid 0xd34
11:13:40 Calling party uses private-numbering
11:13:40 G711A ssff ps:20
rgn:1 [192.168.201.29]:2980
rgn:1 [192.168.232.100]:2054
11:13:40 call-forwarding
11:13:40 term hunt-group 99 cid 0xd34
11:13:40 call-forwarding #993535
11:13:40 term trunk-group 89 cid 0xd34
11:13:40 call-forwarding #993535
11:13:40 route-pattern 99 preference 1 location 1/ALL cid 0xd34
11:13:40 seize trunk-group 89 member 4 cid 0xd34
11:13:40 Calling Number & Name 3495 David Westcom
11:13:40 SIP>INVITE sip:3535@test.local SIP/2.0
11:13:40 Call-ID: df9dca2a339141eaaaf605056b4ba32
11:13:40 Setup digits 3535
11:13:40 Calling Number & Name 3495 David Westcom
11:13:40 SIP<SIP/2.0 100 Trying
11:13:40 Call-ID: df9dca2a339141eaaaf605056b4ba32
11:13:40 Proceed trunk-group 89 member 4 cid 0xd34
11:13:40 SIP<SIP/2.0 407 Proxy Authentication Required <<<<<<<<<<<<<<<<<<<<<
11:13:40 Call-ID: df9dca2a339141eaaaf605056b4ba32
11:13:40 SIP>ACK sip:3535@test.local SIP/2.0
11:13:40 Call-ID: df9dca2a339141eaaaf605056b4ba32
11:13:40 idle trunk-group 89 member 4 cid 0xd34
11:13:40 denial event 2291: Rem off-net covpt is busy D1=0x8c9a D2=0xd34 <<<<<<<<<<
11:13:40 no answer station 3495 cid 0xd34
11:13:40 coverage-path 0 point 0 cid 0xd34
VOIP data from: [192.168.232.100]:2054
11:13:51 Jitter:0 0 0 0 0 0 0 0 0 0: Buff:11 WC:0 Avg:0

It only happens with H323 stations. SIP clients are working correctly.

Any ideas?

Thank you for your support!

 

[kyle555]

The sip part of the call is just the trunk side from CM to SM to IXM right?

If so, why is there a 407 proxy authentication required? You only see that from SM if the SIP invite isn't from/to a IP/port pair that matches a known entity link. If the invite is from a known IP/port pair of an entity link, then that is used to determine routing. If it isn't, it is assumed to be a phone which must provide its username and password. You see 407 all the time on SIP phones calling and it's perfectly normal.

In your case, I think it might point to entity links being configured incorrectly.