May be that information can help...
On definity
display dialplan analysis Page 1 of
3
DIAL PLAN ANALYSIS TABLE
Percent Full:
6
Dialed Total Call Dialed Total Call Dialed Total Call
String Length Type String Length Type String Length Type
0 4ext
1 4ext
2 4ext
3 4ext
2073 is extension - i.e. normal digital phone connected to definity
on asterisk
pri debug span 1
-- Executing Dial(SIP/1015-db22, Zap/g1/2073) in new stack
-- Making new call for cr 32785
Protocol Discriminator: Q.931 (8) len=50
Call Ref: len= 2 (reference 17/0x11) (Originator)
Message type: SETUP (5)
[04 03 80 90 a3]
Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer
capability: Speech (0)
Ext: 1 Trans mode/rate: 64kbps, circuit-mode
(16)
Ext: 1 User information layer 1: A-Law (35)
[18 03 a9 83 82]
Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive
Dchan: 0
ChanSel: Reserved
Ext: 1 Coding: 0 Number Specified Channel Type:
3
Ext: 1 Channel: 2 ]
[1e 02 80 83]
Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0:
0 Location: User (0)
Ext: 1 Progress Description: Calling
equipment is non-ISDN. (3) ]
[28 0e b1 52 69 63 6b 20 41 74 72 65 69 64 65 73]
Display (len=14) Charset: 31 [ Rick Atreides ]
[6c 06 21 81 31 30 31 35]
Calling Number (len= 8) [ Ext: 0 TON: National Number (2) NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1)
Presentation: Presentation permitted, user
number passed network screening (1) '1015' ]
[70 05 a1 32 30 37 33]
Called Number (len= 7) [ Ext: 1 TON: National Number (2) NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1) '2073' ]
-- Called g1/2073
Protocol Discriminator: Q.931 (8) len=9
Call Ref: len= 2 (reference 32785/0x8011) (Terminator)
Message type: RELEASE COMPLETE (90)
[08 02 81 d8]
Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location:
Private network serving the local user (1)
Ext: 1 Cause: Incompatible destination (88), class =
Invalid message (5) ]
-- Processing IE 8 (cs0, Cause)
-- Channel 2, span 1 got hangup
NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null
NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null
-- Hungup 'Zap/2-1'
== No one is available to answer at this time
-Original Message-
From: Ken Godee [mailto:[EMAIL PROTECTED]
Sent: 30 ÉÀÌÑ 2004 Ç. 20:10
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Connecting Asterisk and Avaya Definity By E1 .
Incoming work, but not outgoing
Roman Bessyadovskii wrote:
Yes, I can make a call on that extension from other definity phone, if you
mean it.
-Original Message-
From: Ken Godee [mailto:[EMAIL PROTECTED]
Sent: 30 êáíó 2004 ú. 19:14
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Connecting Asterisk and Avaya Definity By
E1.
Incoming work, but not outgoing
Roman Bessyadovskii wrote:
Hi All.
I connect asterisk and definity by manual at
www.voip-info.org/tiki-index.php?page=Asterisk%20Avaya.
(I just only have E1, not T1 card).
I see, that card work (in definity trunk status, and at asterisk
Incoming call, from definity is work ok, but when I try outgoing call, I
recive
-- Executing Dial(SIP/1015-870f, Zap/g1/2073) in new stack
-- Called g1/2073
-- Channel 1, span 1 got hangup
-- Hungup 'Zap/1-1'
== No one is available to answer at this time
How fix it?
Do you have the Dial Plan set up properly
on the Definity side?
No, that's not what I mean.
You must understand how the Dial plan is used in the Definity.
Are all your extensions on the Definity 4 digits starting with a 2?
If you do not have first digit 2, length 4, type extension
set up in the dial plan, Definity will not know what to do with the four
digits your sending into the switch.
do a display dialplan and make sure it is set up correctly.
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