[Asterisk-Users] Channel 0/1, span 1 got hangup request

[Anton Bakulev]

Dear Users,

I have an Digium Wildcard TE110P T1/E1 Card inserted in Linux box runnig
Asterisk 1.2.0
All incoming calls from E1 interface to SIP-phone goes exellent, but
calls from SIP to E1 gives the errors:

     -- Executing Dial("SIP/anton-6cf4", "Zap/g1/100") in new stack
-- Making new call for cr 32775
     -- Requested transfer capability: 0x00 - SPEECH
> Protocol Discriminator: Q.931 (8)  len=43
> Call Ref: len= 2 (reference 7/0x7) (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 05 41 6e 74 6f 6e]
> Display (len= 5) ╫)│@▒│@hm╫@┤0─@&│@>[ Anton ]
> [6c 0d 21 81 38 34 37 37 33 36 31 38 31 38 33]
> Calling Number (len=15) [ 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) '84773618183' ]
> [70 04 a1 31 30 30]
> Called Number (len= 6) [ Ext: 1  TON: National Number (2)  NPI: 
ISDN/Telephony Numbering Plan (E.164/E.163) (1) '100' ]
     -- Called g1/100
< Protocol Discriminator: Q.931 (8)  len=9
< Call Ref: len= 2 (reference 7/0x7) (Terminator)
< Message type: DISCONNECT (69)
< [08 02 80 90]
< Cause (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0) 0: 0
Location: User (0)
<                  Ext: 1  Cause: Unknown (16), class = Normal Event (1) ]
-- Processing IE 8 (cs0, Cause)
     -- Channel 0/1, span 1 got hangup request
Dec  5 15:30:12 WARNING[30946]: app_dial.c:706 wait_for_answer: Unable
to forward voice
NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Disconnect Indication,
peerstate Disconnect Request
> Protocol Discriminator: Q.931 (8)  len=9
> Call Ref: len= 2 (reference 7/0x7) (Originator)
> Message type: RELEASE (77)
> [08 02 81 90]
> Cause (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0) 0: 0 
Location: Private network serving the local user (1)
>                  Ext: 1  Cause: Unknown (16), class = Normal Event (1) ]
     -- Hungup 'Zap/1-1'
   == No one is available to answer at this time (1:0/0/0)
< Protocol Discriminator: Q.931 (8)  len=9
< Call Ref: len= 2 (reference 7/0x7) (Terminator)
< Message type: RELEASE COMPLETE (90)
< [08 02 80 d1]
< Cause (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0) 0: 0
Location: User (0)
<                  Ext: 1  Cause: Unknown (81), class = Invalid message
(5) ]
-- 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
     -- Timeout on SIP/anton-6cf4
   == CDR updated on SIP/anton-6cf4
     -- Executing Hangup("SIP/anton-6cf4", "") in new stack


/etc/zaptel.conf
span=1,1,5,ccs,hdb3
bchan=1-15,17-31
dchan=16
loadzone = nl
defaultzone=nl

/etc/asterisck/zapata.conf
[trunkgroups]
[channels]
language=en
signalling=pri_cpe
switchtype=euroisdn
echocancel=32
echocancelwhenbridged=yes
usecallerid=yes
callerid=asreceived
transfer=yes
overlapdial=yes
cancallforward=yes
group=1
context=zapata
channel => 1-15,17-31

Has anybody resolve this problem?

--
SY,
Anton V Bakulev.
MIPT-telecom.
bakulev at mipt.ru