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Anton Bakulev

2005-Dec-10 05:34 UTC

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

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@mipt.ru

Steve Totaro

2005-Dec-10 06:39 UTC

head link

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

Just a couple guesses on things to try.

Zapata.conf
1.  Changing switchtype variables (doubtful but give it a try).  
2.  Add a variable to define "pridialplan" (I remember someone setting
this to "unknown" to solve a similar issue)  Try pridialplan=unknown
and/or prilocaldialplan=local or some other valid option.

Zaptel.conf
1.  span=1,1,5,ccs,hdb3

I think that your dial statement or the pridialplan is your issue.  If
you look at the debug info 

Here is something suspicious:  "-- Called g1/100" unless 100 is the
number you are trying to dial outbound.

If the above fails, then try below.

Try tweaking your settings here like span=1,0,0,ccs,hdb3

What is the provider expecting?

Thanks,
Steve

> 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@mipt.ru

Steve Totaro

2005-Dec-11 12:23 UTC

head link

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

What are you doing in between making changes and testing the changes?

Thanks,
Steve
> 
> >Just a couple guesses on things to try.
> >
> >Zapata.conf
> >1.  Changing switchtype variables (doubtful but give it a try).
> >2.  Add a variable to define "pridialplan" (I remember
someone
setting> >this to "unknown" to solve a similar issue)  Try
pridialplan=unknown
> >and/or prilocaldialplan=local or some other valid option.
> 
> A do this config, but no effects....
> 
> >Zaptel.conf
> >1.  span=1,1,5,ccs,hdb3
> >
> >I think that your dial statement or the pridialplan is your issue.
If> >you look at the debug info
> >Here is something suspicious:  "-- Called g1/100" unless 100
is the
> >number you are trying to dial outbound.
> >If the above fails, then try below.
> >Try tweaking your settings here like span=1,0,0,ccs,hdb3
> >What is the provider expecting?
> 
> No effect on settings:
> span=1,0,0,ccs,hdb3
> span=1,1,5,ccs,hdb3
> span=1,2,4,ccs,hdb3
> 
> >Thanks,
> >Steve
> 
> 
> > 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@mipt.ru
> 
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asterisk users - Dec 2005 - Channel 0/1, span 1 got hangup request

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

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

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