[Asterisk-Users] PRI: This number has been disconnected

Fri Dec 30 08:52:08 MST 2005

I restarted as you say.

PRI Debug bellow

[Kasterisk1*CLI> 
    -- Executing
[1;36;40mMacro[0;37;40m("[1;35;40mSIP/225-99e9[0;37;40m",
"[1;35;40mdialout-trunk|1|2514990|[0;37;40m") in new stack

[Kasterisk1*CLI> 
    -- Executing
[1;36;40mGotoIf[0;37;40m("[1;35;40mSIP/225-99e9[0;37;40m",
"[1;35;40m1?3:2)[0;37;40m") in new stack

[Kasterisk1*CLI> 
    -- Goto (macro-dialout-trunk,s,3)

[Kasterisk1*CLI> 
    -- Executing
[1;36;40mMacro[0;37;40m("[1;35;40mSIP/225-99e9[0;37;40m",
"[1;35;40muser-callerid[0;37;40m") in new stack

[Kasterisk1*CLI> 
    -- Executing
[1;36;40mDBget[0;37;40m("[1;35;40mSIP/225-99e9[0;37;40m",
"[1;35;40mAMPUSER=DEVICE/225/user[0;37;40m") in new stack

[Kasterisk1*CLI> 
    -- DBget: varname=AMPUSER, family=DEVICE, key=225/user

[Kasterisk1*CLI> 
    -- DBget: set variable AMPUSER to 225

[Kasterisk1*CLI> 
    -- Executing
[1;36;40mDBget[0;37;40m("[1;35;40mSIP/225-99e9[0;37;40m",
"[1;35;40mAMPUSERCIDNAME=AMPUSER/225/cidname[0;37;40m") in new stack

[Kasterisk1*CLI> 
    -- DBget: varname=AMPUSERCIDNAME, family=AMPUSER, key=225/cidname

[Kasterisk1*CLI> 
    -- DBget: set variable AMPUSERCIDNAME to sipura Linksys

[Kasterisk1*CLI> 
    -- Executing
[1;36;40mGotoIf[0;37;40m("[1;35;40mSIP/225-99e9[0;37;40m",
"[1;35;40m0?5[0;37;40m") in new stack

[Kasterisk1*CLI> 
    -- Executing
[1;36;40mSetCallerID[0;37;40m("[1;35;40mSIP/225-99e9[0;37;40m",
"[1;35;40m"sipura Linksys" <225>[0;37;40m") in new stack

[Kasterisk1*CLI> 
    -- Executing
[1;36;40mNoOp[0;37;40m("[1;35;40mSIP/225-99e9[0;37;40m",
"[1;35;40mUsing CallerID "sipura Linksys" <225>[0;37;40m") in new stack

[Kasterisk1*CLI> 
    -- Executing
[1;36;40mMacro[0;37;40m("[1;35;40mSIP/225-99e9[0;37;40m",
"[1;35;40mrecord-enable|225|OUT[0;37;40m") in new stack

[Kasterisk1*CLI> 
    -- Executing
[1;36;40mGotoIf[0;37;40m("[1;35;40mSIP/225-99e9[0;37;40m", "[1;35;40m0
> 0?2:4[0;37;40m") in new stack

[Kasterisk1*CLI> 
    -- Goto (macro-record-enable,s,4)

[Kasterisk1*CLI> 
    -- Executing [1;36;40mAGI[0;37;40m("[1;35;40mSIP/225-99e9[0;37;40m",
"[1;35;40mrecordingcheck|20051229-145347|1135878827.11[0;37;40m") in new
stack

[Kasterisk1*CLI> 
    -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck

[Kasterisk1*CLI> 
  recordingcheck|20051229-145347|1135878827.11: Outbound recording not
enabled

[Kasterisk1*CLI> 
    -- AGI Script recordingcheck completed, returning 0

[Kasterisk1*CLI> 
    -- Executing
[1;36;40mNoOp[0;37;40m("[1;35;40mSIP/225-99e9[0;37;40m", "[1;35;40mNo
recording needed[0;37;40m") in new stack
     -- Executing
[1;36;40mMacro[0;37;40m("[1;35;40mSIP/225-99e9[0;37;40m",
"[1;35;40moutbound-callerid|1[0;37;40m") in new stack
     -- Executing
[1;36;40mDBget[0;37;40m("[1;35;40mSIP/225-99e9[0;37;40m",
"[1;35;40mUSEROUTCID=AMPUSER/225/outboundcid[0;37;40m") in new stack
     -- DBget: varname=USEROUTCID, family=AMPUSER, key=225/outboundcid
     -- DBget: set variable USEROUTCID to 
     -- Executing
[1;36;40mGotoIf[0;37;40m("[1;35;40mSIP/225-99e9[0;37;40m",
"[1;35;40m1?4[0;37;40m") in new stack
     -- Goto (macro-outbound-callerid,s,4)
     -- Executing
[1;36;40mGotoIf[0;37;40m("[1;35;40mSIP/225-99e9[0;37;40m",
"[1;35;40m1?6[0;37;40m") in new stack
     -- Goto (macro-outbound-callerid,s,6)
     -- Executing
[1;36;40mNoOp[0;37;40m("[1;35;40mSIP/225-99e9[0;37;40m",
"[1;35;40mCallerID set to "sipura Linksys" <225>[0;37;40m") in new stack
     -- Executing
[1;36;40mSetGroup[0;37;40m("[1;35;40mSIP/225-99e9[0;37;40m",
"[1;35;40mOUT_1[0;37;40m") in new stack
     -- Executing
[1;36;40mCheckGroup[0;37;40m("[1;35;40mSIP/225-99e9[0;37;40m",
"[1;35;40m[0;37;40m") in new stack
     -- Executing
[1;36;40mSetVar[0;37;40m("[1;35;40mSIP/225-99e9[0;37;40m",
"[1;35;40mDIAL_NUMBER=2514990[0;37;40m") in new stack
     -- Executing
[1;36;40mSetVar[0;37;40m("[1;35;40mSIP/225-99e9[0;37;40m",
"[1;35;40mDIAL_TRUNK=1[0;37;40m") in new stack
     -- Executing
[1;36;40mAGI[0;37;40m("[1;35;40mSIP/225-99e9[0;37;40m",
"[1;35;40mfixlocalprefix[0;37;40m") in new stack

[Kasterisk1*CLI> 
    -- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix

[Kasterisk1*CLI> 
  fixlocalprefix: Could not open /etc/asterisk/localprefixes.conf

[Kasterisk1*CLI> 
    -- AGI Script fixlocalprefix completed, returning 0
     -- Executing
[1;36;40mSetVar[0;37;40m("[1;35;40mSIP/225-99e9[0;37;40m",
"[1;35;40mOUTNUM=2514990[0;37;40m") in new stack
     -- Executing
[1;36;40mCut[0;37;40m("[1;35;40mSIP/225-99e9[0;37;40m",
"[1;35;40mcustom=OUT_1|:|1[0;37;40m") in new stack
     -- Executing
[1;36;40mGotoIf[0;37;40m("[1;35;40mSIP/225-99e9[0;37;40m",
"[1;35;40m0?16[0;37;40m") in new stack
     -- Executing
[1;36;40mDial[0;37;40m("[1;35;40mSIP/225-99e9[0;37;40m",
"[1;35;40mZAP/g0/2514990[0;37;40m") in new stack
 -- Making new call for cr 32772
     -- Requested transfer capability: 0x00 - SPEECH
 > Protocol Discriminator: Q.931 (8)  len=49
 > Call Ref: len= 2 (reference 4/0x4) (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 81]
 > 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: 1 ]

> [28 0e 73 69 70 75 72 61 20 4c 69 6e 6b 73 79 73]
 > Display (len=14) ˆ>Ù[ sipura Linksys ]

> [6c 05 21 80 32 32 35]
 > Calling Number (len= 7) [ Ext: 0  TON: National Number (2)  NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1)
 >                           Presentation: Presentation permitted, user
number not screened (0) '225' ]

> [70 08 a1 32 35 31 34 39 39 30]
 > Called Number (len=10) [ Ext: 1  TON: National Number (2)  NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1) '2514990' ]

> [a1]
 > Sending Complete (len= 1)
     -- Called g0/2514990

[Kasterisk1*CLI> 
< Protocol Discriminator: Q.931 (8)  len=13

[Kasterisk1*CLI> 
< Call Ref: len= 2 (reference 4/0x4) (Terminator)
 < Message type: STATUS (125)

< [08 03 80 e3 28]
 < Cause (len= 5) [ Ext: 1  Coding: CCITT (ITU) standard (0) 0: 0
Location: User (0)
 <                  Ext: 1  Cause: Unknown (99), class = Protocol Error (6)
]
 <              Cause data 1: 28 (40, Display IE)

< [14 01 01]
 < Call State (len= 3) [ Ext: 0  Coding: CCITT (ITU) standard (0) Call
state: Call Initiated (1)
 -- Processing IE 8 (cs0, Cause)
 -- Processing IE 20 (cs0, Call State)

[Kasterisk1*CLI> 
< Protocol Discriminator: Q.931 (8)  len=10
 < Call Ref: len= 2 (reference 4/0x4) (Terminator)
 < Message type: CALL PROCEEDING (2)

< [18 03 a9 83 81]
 < 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: 1 ]
 -- Processing IE 24 (cs0, Channel Identification)

[Kasterisk1*CLI> 
    -- Zap/1-1 is proceeding passing it to SIP/225-99e9

[Kasterisk1*CLI> 
< Protocol Discriminator: Q.931 (8)  len=13
 < Call Ref: len= 2 (reference 4/0x4) (Terminator)
 < Message type: DISCONNECT (69)

< [08 02 80 81]
 < Cause (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0) 0: 0
Location: User (0)
 <                  Ext: 1  Cause: Unallocated (unassigned) number (1),
class = Normal Event (0) ]

< [1e 02 80 88]

[Kasterisk1*CLI> 
< Progress Indicator (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0) 0:
0   Location: User (0)
 <                               Ext: 1  Progress Description: Inband
information or appropriate pattern now available. (8) ]
 -- Processing IE 8 (cs0, Cause)
 -- Processing IE 30 (cs0, Progress Indicator)
     -- Channel 0/1, span 1 got hangup request

[Kasterisk1*CLI> 
NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Disconnect Indication,
peerstate Disconnect Request
 > Protocol Discriminator: Q.931 (8)  len=16
 > Call Ref: len= 2 (reference 4/0x4) (Originator)
 > Message type: RELEASE (77)

> [08 02 81 81]
 > Cause (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0) 0: 0
Location: Private network serving the local user (1)
 >                  Ext: 1  Cause: Unallocated (unassigned) number (1),
class = Normal Event (0) ]

> [7e 05 04 d8 e3 bc 08]

> User-User Information (len= 7) [ 04 58 63 3c 08 ]
     -- Hungup 'Zap/1-1'
   == No one is available to answer at this time (1:0/0/0)
     -- Executing
[1;36;40mGoto[0;37;40m("[1;35;40mSIP/225-99e9[0;37;40m",
"[1;35;40ms-NOANSWER|1[0;37;40m") in new stack
     -- Goto (macro-dialout-trunk,s-NOANSWER,1)
     -- Executing
[1;36;40mNoOp[0;37;40m("[1;35;40mSIP/225-99e9[0;37;40m", "[1;35;40mDial
failed due to NOANSWER[0;37;40m") in new stack
     -- Executing
[1;36;40mMacro[0;37;40m("[1;35;40mSIP/225-99e9[0;37;40m",
"[1;35;40moutisbusy[0;37;40m") in new stack
     -- Executing
[1;36;40mPlayback[0;37;40m("[1;35;40mSIP/225-99e9[0;37;40m",
"[1;35;40mall-circuits-busy-now[0;37;40m") in new stack

[Kasterisk1*CLI> 
    -- Playing 'all-circuits-busy-now' (language 'es')

[Kasterisk1*CLI> 
< Protocol Discriminator: Q.931 (8)  len=10
 < Call Ref: len= 2 (reference 4/0x4) (Terminator)
 < Message type: RELEASE COMPLETE (90)

< [08 03 80 ab 7e]
 < Cause (len= 5) [ Ext: 1  Coding: CCITT (ITU) standard (0) 0: 0
Location: User (0)
 <                  Ext: 1  Cause: Unknown (43), class = Network Congestion
(2) ]
 <              Cause data 1: 7e (126)
 -- Processing IE 8 (cs0, Cause)
 NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null
 NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null

[Kasterisk1*CLI> 
    -- Executing
[1;36;40mPlayback[0;37;40m("[1;35;40mSIP/225-99e9[0;37;40m",
"[1;35;40mpls-try-call-later[0;37;40m") in new stack

[Kasterisk1*CLI> 
    -- Playing 'pls-try-call-later' (language 'es')

[Kasterisk1*CLI> 
    -- Executing
[1;36;40mMacro[0;37;40m("[1;35;40mSIP/225-99e9[0;37;40m",
"[1;35;40mhangupcall[0;37;40m") in new stack
     -- Executing
[1;36;40mResetCDR[0;37;40m("[1;35;40mSIP/225-99e9[0;37;40m",
"[1;35;40mw[0;37;40m") in new stack

[Kasterisk1*CLI> 
    -- Executing
[1;36;40mNoCDR[0;37;40m("[1;35;40mSIP/225-99e9[0;37;40m",
"[1;35;40m[0;37;40m") in new stack
     -- Executing
[1;36;40mWait[0;37;40m("[1;35;40mSIP/225-99e9[0;37;40m",
"[1;35;40m5[0;37;40m") in new stack

[Kasterisk1*CLI> 
  == Spawn extension (macro-hangupcall, s, 3) exited non-zero on
'SIP/225-99e9' in macro 'hangupcall'
   == Spawn extension (macro-outisbusy, s, 3) exited non-zero on
'SIP/225-99e9' in macro 'outisbusy'
   == Spawn extension (from-internal, 2514990, 2) exited non-zero on
'SIP/225-99e9'
     -- Executing
[1;36;40mMacro[0;37;40m("[1;35;40mSIP/225-99e9[0;37;40m",
"[1;35;40mhangupcall[0;37;40m") in new stack
     -- Executing
[1;36;40mResetCDR[0;37;40m("[1;35;40mSIP/225-99e9[0;37;40m",
"[1;35;40mw[0;37;40m") in new stack
     -- Executing
[1;36;40mNoCDR[0;37;40m("[1;35;40mSIP/225-99e9[0;37;40m",
"[1;35;40m[0;37;40m") in new stack
     -- Executing
[1;36;40mWait[0;37;40m("[1;35;40mSIP/225-99e9[0;37;40m",
"[1;35;40m5[0;37;40m") in new stack
   == Spawn extension (macro-hangupcall, s, 3) exited non-zero on
'SIP/225-99e9' in macro 'hangupcall'
   == Spawn extension (from-internal, h, 1) exited non-zero on
'SIP/225-99e9'

-----Mensaje original-----
De: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] En nombre de Adam Goryachev
Enviado el: Jueves, 29 de Diciembre de 2005 23:40
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Asunto: RE: [Asterisk-Users] PRI: This number has been disconnected

On Thu, 2005-12-29 at 16:22 -0300, Javier Ergas wrote:
> I have tried both inband and outofband too unsuccessfully. I think the
> priindication parameter says how Asterisk reports Busy and Congestion
> to the PSTN, not the other way around.
> 
> In the Asterisk config sirrix.conf
> (http://www.voip-info.org/wiki/index.php?page=Asterisk+config
> +sirrix.conf) there is a providetones parameter, witch I think handles
> the way that interface receives the signalization from the PSTN, but I
> think it won’t work for zaptel/Zapata.
> 
>  
> 
> Today I tried Asterisk 1.2 in another Telco and I experienced the same
> behavior. I’m starting to think this is a bug in the Asterisk E1
> signalization. 

Just remember to restart asterisk, not just reload...

I had this problem, and changing this value seemed to fix it for me.
Your mileage may vary...

Maybe providing a pri debug output during an example call would get you
a comment from someone else ....

Regards,
Adam

> 

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