[asterisk-users] Asterisk DIAL() premature timeout on a PRI trunk to legacy PBX

Fri Sep 14 01:55:59 CDT 2007

Thank you,
I did what you mentioned below.
It seems that I'm getting a hangupcause of 0 which I
believe is "not defined".
Is Alcatel the first party that is trying to
disconnect or is it Asterisk? (Not sure how to
interpret the debug info I'm posting below)

Whether it's Alcatel or Asterisk, what could be the
actual cause? (or where should I start looking?)

Thanks

INF-VOIP*CLI> pri debug span 1
Enabled debugging on span 1
    -- Executing NoOp("SIP/4053-083189e8", "[ALCATEL
TEST] Start") in new stack
    -- Executing Dial("SIP/4053-083189e8",
"Zap/g1/5900") in new stack
1 -- Making new call for cr 32781
    -- Requested transfer capability: 0x00 - SPEECH
1 > Protocol Discriminator: Q.931 (8)  len=32
1 > Call Ref: len= 2 (reference 13/0xD) (Originator)
1 > Message type: SETUP (5)
1 > [04 03 80 90 a3]
1 > Bearer Capability (len= 5) [ Ext: 1  Q.931 Std: 0 
Info transfer capability: Speech (0)
1 >                              Ext: 1  Trans
mode/rate: 64kbps, circuit-mode (16)
1 >                              Ext: 1  User
information layer 1: A-Law (35)
1 > [18 04 e9 81 83 81]
1 > Channel ID (len= 6) [ Ext: 1  IntID: Explicit, PRI
Spare: 0, Exclusive Dchan: 0
1 >                        ChanSel: Reserved
1 >                       Ext: 1  DS1 Identifier: 1
1 >                       Ext: 1  Coding: 0   Number
Specified   Channel Type: 3
1 >                       Ext: 1  Channel: 1 ]
1 > [6c 06 21 80 34 30 35 33]
1 > Calling Number (len= 8) [ Ext: 0  TON: National
Number (2)  NPI: ISDN/Telephony Numbering Plan
(E.164/E.163) (1)
1 >                           Presentation:
Presentation permitted, user number not screened (0)
'4053' ]
1 > [70 05 a1 35 39 30 30]
1 > Called Number (len= 7) [ Ext: 1  TON: National
Number (2)  NPI: ISDN/Telephony Numbering Plan
(E.164/E.163) (1) '5900' ]
1 > [a1]
1 > Sending Complete (len= 1)
    -- Called g1/5900
1 < Protocol Discriminator: Q.931 (8)  len=10
1 < Call Ref: len= 2 (reference 13/0xD) (Terminator)
1 < Message type: CALL PROCEEDING (2)
1 < [18 03 a9 83 81]
1 < Channel ID (len= 5) [ Ext: 1  IntID: Implicit, PRI
Spare: 0, Exclusive Dchan: 0
1 <                        ChanSel: Reserved
1 <                       Ext: 1  Coding: 0   Number
Specified   Channel Type: 3
1 <                       Ext: 1  Channel: 1 ]
1 -- Processing IE 24 (cs0, Channel Identification)
    -- Zap/1-1 is proceeding passing it to
SIP/4053-083189e8
1 < Protocol Discriminator: Q.931 (8)  len=5
1 < Call Ref: len= 2 (reference 13/0xD) (Terminator)
1 < Message type: ALERTING (1)
    -- Zap/1-1 is ringing
    -- Zap/1-1 is busy
1 NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Call
Delivered, peerstate Call Received
1 > Protocol Discriminator: Q.931 (8)  len=9
1 > Call Ref: len= 2 (reference 13/0xD) (Originator)
1 > Message type: DISCONNECT (69)
1 > [08 02 81 90]
1 > Cause (len= 4) [ Ext: 1  Coding: CCITT (ITU)
standard (0) 0: 0   Location: Private network serving
the local user (1)
1 >                  Ext: 1  Cause: Normal Clearing
(16), class = Normal Event (1) ]
    -- Hungup 'Zap/1-1'
  == Everyone is busy/congested at this time (1:1/0/0)
    -- Executing NoOp("SIP/4053-083189e8", "[ALCATEL
TEST] hangupcause: 0") in new stack
    -- Executing Hangup("SIP/4053-083189e8", "") in
new stack
  == Spawn extension (custom-TEST_ALCATEL, s, 4)
exited non-zero on 'SIP/4053-083189e8'
1 < Protocol Discriminator: Q.931 (8)  len=9
1 < Call Ref: len= 2 (reference 13/0xD) (Terminator)
1 < Message type: RELEASE (77)
1 < [08 02 81 90]
1 < Cause (len= 4) [ Ext: 1  Coding: CCITT (ITU)
standard (0) 0: 0   Location: Private network serving
the local user (1)
1 <                  Ext: 1  Cause: Normal Clearing
(16), class = Normal Event (1) ]
1 -- Processing IE 8 (cs0, Cause)
1 NEW_HANGUP DEBUG: Calling q931_hangup, ourstate
Null, peerstate Release Request
1 > Protocol Discriminator: Q.931 (8)  len=9
1 > Call Ref: len= 2 (reference 13/0xD) (Originator)
1 > Message type: RELEASE COMPLETE (90)
1 > [08 02 80 90]
1 > Cause (len= 4) [ Ext: 1  Coding: CCITT (ITU)
standard (0) 0: 0   Location: User (0)
1 >                  Ext: 1  Cause: Normal Clearing
(16), class = Normal Event (1) ]
1 NEW_HANGUP DEBUG: Calling q931_hangup, ourstate
Null, peerstate Null
1 NEW_HANGUP DEBUG: Destroying the call, ourstate
Null, peerstate Null
    -- Channel 1/1, span 1 received AOC-E charging 0
units
INF-VOIP*CLI> quit

--- "Eric \"ManxPower\" Wieling" <eric at fnords.org>
wrote:

> Looks like the Alcatel is sending back a busy. 
> Check the value of 
> HANGUPCAUSE with a Noop as the priority after the
> Dial.  You may also 
> want to do a pri debug span X to see the actual
> Q.931 ISDN messages that 
> are exchanged.
> 
> Vieri wrote:
> > An Asterisk extension calls an Alcatel extension
> via a
> > PRI link which rings 4 times for about 10-15
> seconds
> > and then drops.
> > So if the Alcatel user doesn't answer within 10-15
> > seconds the call is aborted.
> > (A timeout is *not* specified in the Asterisk Dial
> > command.)
> > It seems however that either Asterisk or Alcatel
> drop
> > the call prematurely (it's more likely to be on
> the
> > Asterisk side).
> > 
> > What could I try?
> > 
> > The Asterisk log displays (* ext is 4053; Alcatel
> ext
> > is 5900):
> > 
> > -- Executing
> > Dial("SIP/4053-08311988","Zap/g1/5900||tTW") in
> new
> > stack
> > -- Requested transfer capability: 0x00 - SPEECH
> > -- Called g1/5900
> > -- Zap/2-1 is proceeding passing it to
> > SIP/4053-08311988
> > -- Zap/2-1 is ringing
> > -- Zap/2-1 is busy
> > -- Hungup 'Zap/2-1'
> > == Everyone is busy/congested at this time
> (1:1/0/0)
> > -- Executing Hangup("SIP/4053-08311988", "") in
> new
> > stack
> > == Spawn extension (from-internal, 5900, 4) exited
> > non-zero on 'SIP/4053-08311988'
> > -- Executing Macro("SIP/4053-08311988",
> "hangupcall")
> > in new stack
> > ...etc...
> > 
> > The Alcatel board is configured as:
> > 
> > Interface Type + PRA2
> > CRC4 + YES
> > Retransmission Timer : 100
> > TEI Identity Check Timer : 100
> > Polling Timer : 1000
> > No. Of Retransmissions : 3
> > Max Frame Size (Bytes) : 260
> > Passive board + NO
> > SS7 signaling + NO
> > 
> > (I also tried to increase the above "Timer" values
> but
> > that did not change anything)
> > 
> > In Asterisk's /etc/zaptel.conf I have:
> > 
> > # TE120P (PRI):
> > span=1,1,0,ccs,hdb3,crc4
> > 
> > bchan=1-15
> > dchan=16
> > bchan=17-31
> > 
> > What could be the problem here?
> > 
> > Thanks

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