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Asterisk & Avaya h.323

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Arklite

Technical User
Dec 3, 2004
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Can anybody help or point me in the right direction as i think I have a configuration mistake somewhere?
I have an Asterisk server with SIP Phones attached, this is connected to Avaya CM 3.1.2 using h.323 trunk. This CM 3.1.2 is connected in turn to another Avaya CM 3.1.2 via another h.323 trunk. On this CM there is Media gateway where the ISDN trunks reside.
Problem is this:
Out bound calls from the SIP phones to the public network via the ISDN on the second CM fail with the following denial messages:
From the Tandem CM:
denial event 1189: Normal, unspecified D1=0x83002a D2=0x31f
From the CM with the ISDN:
denial event 1218: Invalid IE contents D1=0x830036 D2=0x364

I have checked my asterisk.conf and sip.conf files and they appear correct, also I have made various changes to the oh323.conf file to try and resolve but this has proved unsuccessful. I am sure its just a config mismatch somewhere
but i have looking at this for so long now I cant se the wood for the trees?!

any ideas would be greatly appreciated.


 
Hear is a trace i have taken of the attempted ISDN call to an 06 number:note that is should be read from bottom to top.

1 16:40:32.535 71 ^^^^^^^^^^^^^^^^^^^^^^^^manual_disable^^^^^ GAP.00001

2 16:40:16.010 60 4001080f ==> REL_COM crv 1_6420 bad ie content !
D-channel port number: 4001080f
Protocol Discriminator: 08 call control
Call Reference Value: 02 e4 20 Destination 6420
Q.931 Message Type: 5a Release Complete
=> CAUSE transit network: 100. Invalid info element contents

3 16:40:15.913 62 4001080f <-- SETUP crv 0_6420 <<!
D-channel port number: 4001080f
Protocol Discriminator: 08 call control
Call Reference Value: 02 64 20 Origination 6420
Q.931 Message Type: 05 Setup
<- SENDING COMPLETE (a1)
<- BEARER CAPABILITY unrestr'd, (packet-mode), <<unrecog layer 1 protocol>>
coding standard: CCITT
| transfer mode: circuit-mode
| layer 1 indent: layer 1
<- CHANNEL IDENT a1 83 96 B-ch num 22, pref
<- CALLED PARTY NUMBER .................................. 0625081954
| type of number: Level 2 Regional Number
| numbering plan: private plan





 
It looks like you are sending calls out the PRI as a private number plan and your telco provider is expecting something else... Most likely public number plan.

On your route pattern on the Definity that the calls are being routed out of, you need to specify the call as pub-unk. Here's a hint, look at the bottom right.
 
Redphone
I saw this myself and have already tried the different options in both the route pattern and the ARS table. The clue that I find more important is the msg that states there is an unrecognized Layer 1 protocol. This seams more fundamental to me.
I am wondering if the Protocol of h323 at the asterisk needs to be changed before it reaches the ACM where the E1 ISDN is?
 
Try a trace on a good call and see what it shows. My guess is it will state the same layer 1 error. The call is being denied by mismatching IEs.

Can you make a 06 call from both CMs?

Did you try changing the trunk-group connected to the Asterisk to make the numbering Format unk-pvt or unknown?
 
Thanks Chkbrt
About a month ago i used this thread to set this up.
And for 90% of it it works fine. But, the Asterisk is
connected to 1 ACM which has a h.323 connection to another
ACM, this in turn has a LSP Gateway connected across a WAN
which houses the E1 circuit used for PSTN calls, its the PSTN
calls that fail from the SIP phone on the Asterisk.

And I have now found another problem.
DTMF is not recieved by the 1st Avaya from a SIP phone dialing
a Meetme Conf access code (meetme vector is on the 2nd ACM).

 
I am trying to establish a h.323 trunk between my Asterisk Trixbox 2.2 and an Avaya CM 3.1.2. I then realized I do not have the 00h323 addons installed. I have downloaded the addon package however when running the ./configure command I get the following error:
configure: error: No asterisk installation found
When I do a show channels or channeltypes i see no information associated to H.323 at all. Can the asterisk gurus please point the old telephone man in the right direction?
 
I have not seen this problem before. I would try
to re install the Asterisk, check version levels for latest.
Your download could of been corrupt or not installed clean.

best of luck
 
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