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Sip Trunk between Avaya Aura and Asterisk Voicemail Server

Sip Trunk between Avaya Aura and Asterisk Voicemail Server

(OP)
Hi,
We are using Avaya Aura CM 6.3 and voicemail is Asterisk.
When I dial the mail VM, I get a dialling-error tone.I don't know if the Asterisk VM has ever worked. I'm trying to figure out what is wrong.

The Coverage-Path, aar analysis, hunt-group, signalling group ,trunk, and node-names all look OK to me , but these are some things that don't look right:

1. In System Manager, Location 1 is defined for our main site but there is no Location defined for the Astersik.
2. In System Manager, there is no Entity Link for the SIP trunk between Avaya CM and Asterisk
3. In System Manager there is no Domain defined for the Asterisk VM.
4. In Asterisk, in extensions.conf, the main voicemail phone number does not appear to be configured.

So my questions at this point are:
a) Does the SIP Trunk require it's own location in System Manager?
b) Does the SIP Trunk require an Entity Link in System Manager?
c) In System Manager, does a Domain need to be defined for the Asterisk?
d) Does the main VM phone number get defined in extensions.conf and if so, how?

Thanks for any help with this.

RE: Sip Trunk between Avaya Aura and Asterisk Voicemail Server

a.No
b.Yes, if you're trunking it to Session Manager
c.No, you can use "domain.com" for everything, nothing wrong with that
d.beats me

So, you sound like you're caught between two ways of doing things. With no SM, you'd have a CM SIP sig group to Asterisk - that's it. With Session Manager, you'd have a CM SIP sig group to SM and an entity link from CM to SM and from SM to Asterisk. SM is the hub.

You might still have different trunk group numbers in CM for different things SM does for you - trunk 99 to SM but for voicemail, trunk 10 to SM but to get to PSTN SIP trunks, etc..

RE: Sip Trunk between Avaya Aura and Asterisk Voicemail Server

(OP)
Thanks Kyle555.
We do have an SM but the SIP trunk is not going that way. The SIP trunk is going directly to Asterisk.
The tech who configured it is no longer with us. I am trying to determine why the voice mail is not working.I'm not sure if this has ever worked.
When I call one of the stations and it is N/A, it goes to coverage but voicemail does not answer.It goes to turkey-tone.
When I dial the Asterisk voicemail phone number, that too goes to turkey-tone.
Not sure if the Asterisk depends on a main voicemail phone number to be configured or if only the 'User' mailboxes have to be configured to answer and take a message.
Any ideas?

RE: Sip Trunk between Avaya Aura and Asterisk Voicemail Server

Lookup a devconnect note for a 3rd party SIP voicemail. Short version is the station has a coverage path to a hunt group for voicemail that goes to AAR and dials the voicemail pilot.
When 1111 calls 2222 and covers to voicemail hunt 3333, hunt 3333 sends the call to AAR for voicemail pilot number 3333. The "TO" header is 3333, the "FROM" header is 1111 and either the history-info or diversion header will say "2222" and that's how voicemail should know to answer with 2222's greeting.

RE: Sip Trunk between Avaya Aura and Asterisk Voicemail Server

Do you need MWI on your Avaya phones?
If not, I have an easy way of running an Asterisk box as a voice mail server to your Avaya system.
I am doing this myself.

RE: Sip Trunk between Avaya Aura and Asterisk Voicemail Server

(OP)
No, we do not need MWI. I'm interested in your "easy way".

RE: Sip Trunk between Avaya Aura and Asterisk Voicemail Server

(OP)
When I trace station 22222 calling to station 33333...
33333 No answer.... forwards to voicemail (44444)
This is the part of the trace that forwards to voicemail.
I get a : SIP<SIP/2.0 404 Not Found
and: denial event 1166: Unassigned number

I do not see 44444 programmed in the Asterisk.


LIST TRACE Station 22222

no answer station 12345 cid 0x11e1
16:27:15 coverage-path 98 point 1 cid 0x11e1
16:27:15 call-forwarding *0044444
16:27:15 term trunk-group 98 cid 0x11e1
16:27:15 call-forwarding *0044444
16:27:15 route-pattern 98 preference 1 location 1/ALL cid 0x11e1
16:27:15 seize trunk-group 98 member 4 cid 0x11e1
16:27:15 Calling Number & Name NO-CPNumber NO-CPName
16:27:15 SIP>INVITE sip:44444@X.X.X.X SIP/2.0
16:27:15 Call-ID: xxxxxxxxxxxxx
16:27:15 Setup digits 44444
time data
16:27:15 Calling Number & Name 22222 Smith, J
16:27:15 SIP<SIP/2.0 404 Not Found
16:27:15 Call-ID: xxxxxxxxxxxxxx
16:27:15 SIP>ACK sip:44444@X.X.X.X SIP/2.0
16:27:15 Call-ID: xxxxxxxxxxxxxxxxx
16:27:15 denial event 1166: Unassigned number D1=0x8c93 D2=0x201
16:27:15 idle trunk-group 98 member 4 cid 0x11e1

RE: Sip Trunk between Avaya Aura and Asterisk Voicemail Server

Well, my numbers were made up as an example.

In your case, you sent your call to voicemail to pilot number 44444 and voicemail answered that it knows nothing about 44444 - 404 not found.

So, what's your Asterisk pilot number?

RE: Sip Trunk between Avaya Aura and Asterisk Voicemail Server

(OP)
Good Question. How would I find that? I don't see 44444 programmed in the Asterisk as a User. I am new to Asterisk. I am learning as I go along.

RE: Sip Trunk between Avaya Aura and Asterisk Voicemail Server

Try that: Asterisk as VM for Avaya CM
I am the guy who provided the solution in that thread, so please ask if that's not clear.

RE: Sip Trunk between Avaya Aura and Asterisk Voicemail Server

(OP)
Thanks telecomadmin12. I looked at the link.
"The way VM works, Avaya dials 6600 and then Asterisk has rules that deal with inbound calls for 6600 in extensions_override_freepbx.conf:

[Avaya-vm]
exten => 6600,1,NoOp(${CALLERID(num)})
exten => 6600,2,NoOp(${CALLERID(rdnis)})
exten => 6600,3,GotoIf($["${CALLERID(rdnis)}" != ""]?4:400)
exten => 6600,4,Playback(silence/1)
exten => 6600,5,Voicemail(${CALLERID(rdnis)}@Avaya-vm)
exten => 6600,6,Hangup
exten => 6600,400,VoicemailMain(${CALLERID(num)}@Avaya-vm)"


As per your example, I couldn't find extensions_override_freepbx.conf. This is what I found out:
" This is for "FreePBX", not "Asterisk". Asterisk natively doesn't use / come-with "extensions_additional.conf"."


Our trunk between Avaya and Astersik is SIP.
As you can see in my List Trace Station 2222, the call to 33333 is not answered. It goes to coverage path 98 and points to trunk group 98, call forwarding to voicemail 44444, then to route=pattern 98 where a trunk member is seized. It is in the next steps where the failure occurs:
16:27:15 SIP>INVITE sip:44444@X.X.X.X SIP/2.0
16:27:15 Call-ID: xxxxxxxxxxxxx
16:27:15 Setup digits 44444
time data
16:27:15 Calling Number & Name 22222 Smith, J
16:27:15 SIP<SIP/2.0 404 Not Found
16:27:15 Call-ID: xxxxxxxxxxxxxx
16:27:15 SIP>ACK sip:44444@X.X.X.X SIP/2.0
16:27:15 Call-ID: xxxxxxxxxxxxxxxxx
16:27:15 denial event 1166: Unassigned number D1=0x8c93 D2=0x201

What causes the SIP<SIP/2.0 404 Not Found
and what causes the denial event 1166: Unassigned number D1=0x8c93 D2=0x201 ?

It seems like something is missing in the Asterisk, not the Avaya.

As I have said, I do not see a mailbox 44444 in Asterisk. The Avaya is looking for 44444.
Where in the Asterisk should it be found?

RE: Sip Trunk between Avaya Aura and Asterisk Voicemail Server

Sorry, my solution was for FreePBX, not Vanilla Asterisk.
So you need an entry in voicemail.conf for each Avaya extension.
E.g.: exten => 2000,1,VoiceMail(2000,u)

Then you also need the proper dial plan in extension.conf to access the voicemail application. Make sure your trunk context can access the context in extensions.conf where you have your voicemail dial plan under (or it's the same context).
You also need to be able to reach the context in voicemail.conf from extensions.conf.
Here is a tutorial: Voicemailapp

I am using the voice mail direct dial prefix in FreePBX that I insert in the route pattern on Avaya when sending the digits out the trunk to Asterisk via uniform dial plan, AAR, route pattern, Asterisk trunk.
In other words, from Avaya, I am sending the dialed digits out a trunk, just like I was calling extensions on a remote system, but prefix the dialed digits with a FreePBX feature code, that allows accessing voice mail box directly.
In FreePBX, I have the Avaya extensions set up as custom extensions. They can also be virtual if Asterisk is only acting as voice mail server and has no exiting phones running off of it. Then the voice mail direct dial prefix wouldn't be required.
However, with a vanilla Asterisk install, this does not apply.

RE: Sip Trunk between Avaya Aura and Asterisk Voicemail Server

(OP)
Thanks telecomadmin12,

I viewed the tutorial. It was helpful but I still am trying to figure out why an unanswered call in not being answered by voicemail.

Here is an example of what is configured in our Asterisk Voicemail:

voicemail.conf
[Default}
45678 => 45678,Rm2011,


extensions.conf
exten => 45678,1,NoOp() ;Rm2011
same => n,Voicemail(45678)

There are no sections for [Internal] or for [Voicemail]

Somewhere I read that NoOp means No Operation = do nothing.
Is this why the mailbox does not answer?

RE: Sip Trunk between Avaya Aura and Asterisk Voicemail Server

NoOp means do nothing, but then dial plan execution continues with the next line of code. It's a troubleshooting tool and not the reason why your mailbox doesn't answer.

Why don't you register a soft client on your Asterisk server and try to dial the mailbox from internal.
If that works then you know it's the path from Avaya where the problem is.

Leave the last comma out in 45678 => 45678,Rm2011

Why don't you show some Asterisk logs with a failed call. That'll tell us why.

RE: Sip Trunk between Avaya Aura and Asterisk Voicemail Server

(OP)
Really appreciate your help telecomadmin12.

Not sure if there is a softphone configured. The previous Administrator who set up the Asterisk voicemail likely had one. Would it be defined in sip.conf ?
Not using FreePBX... What softphone would you suggest I download?


Another thing, where can I find the log files?

RE: Sip Trunk between Avaya Aura and Asterisk Voicemail Server

Sip endpoints are defined in sip.conf.
You can try Zoiper.

Look at the Asterisk CLI and watch the dial plan debug that's presented when you make a call. Copy and paste that here.
Access the Asterisk CLI via asterisk -vvvvr.
This info gets also logged into /var/log/asterisk/full, but that might be different on your Asterisk install.

You can also do a sip debug if necessary. On the Asterisk CLI enter sip set debug on (if using chan_sip and not pjsip.)
Then sip set debug off.

RE: Sip Trunk between Avaya Aura and Asterisk Voicemail Server

(OP)
Realized that any soft phone would work. Using Avaya OneX Communicator...

From my station 2222 on my OneX softphone, I dialled station 33333.
33333 on no answer should forward to asterisk voicemail.

This is from the Asterisk cli:
CLI> sip show history 22222
No such SIP Call ID starting with '22222'
== Using SIP RTP CoS mark 5
[Sep 11 12:06:29] NOTICE[17704][C-000001b6]: chan_sip.c:25791 handle_request_invite: Call from 'avaya' (X.X.X.X:13421) to extension '44444' rejected because extension not found in context 'Office_vm'.




RE: Sip Trunk between Avaya Aura and Asterisk Voicemail Server

(OP)
I removed the comma, as you said to, in voicemail.conf
voicemail.conf
[Default}
47956 => 45678,Rm4104

and here's the debug output:
[Sep 12 09:59:38] NOTICE[17704][C-000001ba]: chan_sip.c:25791 handle_request_invite: Call from 'avaya' (X.X.X.X:13421) to extension '44444' rejected because extension not found in context 'Office_vm'.

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