asterisk users - Jul 2004 - Connecting Asterisk and Avaya Definity By E1 . Incoming work, but not outgoing

Yes, I can make a call on that extension from other definity phone, if you
mean it.

-----Original Message-----
From: Ken Godee [mailto:ken@perfect-image.com] 
Sent: 30 ÉÀÌÑ 2004 Ç. 19:14
To: asterisk-users@lists.digium.com
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?

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:ken@perfect-image.com] 
> Sent: 30 ???? 2004 ?. 19:14
> To: asterisk-users@lists.digium.com
> 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.

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       4    ext
        1       4    ext
        2       4    ext
        3       4    ext

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:ken@perfect-image.com] 
Sent: 30 ÉÀÌÑ 2004 Ç. 20:10
To: asterisk-users@lists.digium.com
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:ken@perfect-image.com] 
> Sent: 30 êáíó 2004 ú. 19:14
> To: asterisk-users@lists.digium.com
> 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.

_______________________________________________
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Thanks for help.
All works now.

Problem was in codecs on different sides

Definity: display ds1 1b14 CRC? n 
Interface Companding: mulaw                             

And when making call via asterisk
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

So I can't make call. But incoming call (Definity -> Asterisk) works,
because asterisk understand ulaw.
                                                                         
So, I have once more question.
How can I change codec on Digium card on Asterisk side?
I configure asterisk and definity with this page
(http://www.voip-info.org/wiki-Asterisk+Avaya), and here no one word about
what codec asterisk use.