missing destination extension in chan_capi, incoming calls fail

Hello everyone,

A user of AskoziaPBX is having trouble receiving incoming
calls from
another PBX connected via ISDN, outgoing calls are working.
One MSN is
assigned to the s0 bus coming from that PBX and connected to
his
system's port which is configured to use
point-to-multipoint, terminal
equipment signaling.

When he attaches a telephone to this line everything works
correctly.

The appropriate Asterisk logs which show no destination
being set or
recognized are:

Mar 5 18:08:13 --
capi_handle_connect_indication:6065:ENTRY=:PLCI=0x0b00:PBX_C
HAN=**Unknown**:
-- Incoming call from '302' to '', CIP=0x0010,
sending_complete=no
Mar 5 18:08:13 ==
cd_start_pbx:5950:ENTRY=0:PLCI=0x0b00:PBX_CHAN=CAPI/0/-6: ==
Started
PBX
Mar 5 18:08:13 == Starting CAPI/0/-6 at
isdn-unit-0-incoming,,1 failed
so falling back to exten 's'
Mar 5 18:08:13 == Starting CAPI/0/-6 at
isdn-unit-0-incoming,s,1 still
failed so falling back to context 'default'
Mar 5 18:08:13 pbx.c:2470 in __ast_pbx_run: Channel
'CAPI/0/-6' sent
into invalid extension 's' in context 'default', but no
invalid
handler
Mar 5 18:08:13 ==
cd_free:2150:ENTRY=0:PLCI=0x0b00:PBX_CHAN=CAPI/0/-6: ==
Mar 5 18:08:13 > CAPI: Command=INFO_IND, 0x8492: no call
descriptor
for PLCI=0x0b00, MSGNUM=0x0000:
Mar 5 18:08:13 > CAPI: Command=DISCONNECT_IND, 0x848c: no
call
descriptor for PLCI=0x0b00, MSGNUM=0x0000:


I could work around this problem by generating 's'
extensions in the
isdn-unit-0-incoming context but this still does not explain
to me why
there is no destination information present. Perhaps this is
simply
the case if only one MSN is assigned to an s0 bus?

The system is FreeBSD 6.2-p10, Asterisk 1.4.17 and isdn4bsd
revision 606.

Attachments (via link, sending bounced once due to size):
 - extensions.conf
 - capi.conf
 - isdn trace of failed call

http://www.askozia.com/tmp/capi_problem_debug_info.zip



Any help on this is greatly appreciated,
-Michael

_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.c
om--

Asterisk-BSD mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-bsd

Hi Michael,

With regard to chan_capi there is an "immediate"
option which will make an "s" 
out of empty destination numbers. That might be your
problem, to make the 
dialplan match against an empty number:

        if (cd->cep) {

            start_immediate = (cd->options.immediate
&& (cd->dst_telno[0] == 
0));

            if (cd->flags.sending_complete_received ||
start_immediate) {

                /* number is complete, try to start the PBX
*/

                cd_start_pbx(pp_cd, start_immediate ?
"s" : 
&(cd->dst_telno[0]));
                cd = *pp_cd;
            }

capi.conf NOTE:

USB devices currently start at controller unit 63.

--HPS

On Saturday 15 March 2008, Michael Iedema wrote:
> Hello everyone,
>
> A user of AskoziaPBX is having trouble receiving
incoming calls from
> another PBX connected via ISDN, outgoing calls are
working. One MSN is
> assigned to the s0 bus coming from that PBX and
connected to his
> system's port which is configured to use
point-to-multipoint, terminal
> equipment signaling.
>
> When he attaches a telephone to this line everything
works correctly.
>
> The appropriate Asterisk logs which show no destination
being set or
> recognized are:
>
> Mar 5 18:08:13 --
>
capi_handle_connect_indication:6065:ENTRY=:PLCI=0x0b00:PBX_C
HAN=**Unknown**
>: -- Incoming call from '302' to '', CIP=0x0010,
sending_complete=no Mar 5
> 18:08:13 ==
>
cd_start_pbx:5950:ENTRY=0:PLCI=0x0b00:PBX_CHAN=CAPI/0/-6: ==
Started
> PBX
> Mar 5 18:08:13 == Starting CAPI/0/-6 at
isdn-unit-0-incoming,,1 failed
> so falling back to exten 's'
> Mar 5 18:08:13 == Starting CAPI/0/-6 at
isdn-unit-0-incoming,s,1 still
> failed so falling back to context 'default'
> Mar 5 18:08:13 pbx.c:2470 in __ast_pbx_run: Channel
'CAPI/0/-6' sent
> into invalid extension 's' in context 'default', but no
invalid
> handler
> Mar 5 18:08:13 ==
cd_free:2150:ENTRY=0:PLCI=0x0b00:PBX_CHAN=CAPI/0/-6: ==
> Mar 5 18:08:13 > CAPI: Command=INFO_IND, 0x8492: no
call descriptor
> for PLCI=0x0b00, MSGNUM=0x0000:
> Mar 5 18:08:13 > CAPI: Command=DISCONNECT_IND,
0x848c: no call
> descriptor for PLCI=0x0b00, MSGNUM=0x0000:
>
>
> I could work around this problem by generating 's'
extensions in the
> isdn-unit-0-incoming context but this still does not
explain to me why
> there is no destination information present. Perhaps
this is simply
> the case if only one MSN is assigned to an s0 bus?
>
> The system is FreeBSD 6.2-p10, Asterisk 1.4.17 and
isdn4bsd revision 606.
>
> Attachments (via link, sending bounced once due to
size):
>  - extensions.conf
>  - capi.conf
>  - isdn trace of failed call
>
> http://www.askozia.com/tmp/capi_problem_debug_info.zip

>
>
> Any help on this is greatly appreciated,
> -Michael
>
> _______________________________________________
> --Bandwidth and Colocation Provided by http://www.api-digital.c
om--
>
> Asterisk-BSD mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-bsd



_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.c
om--

Asterisk-BSD mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-bsd

On 3/16/08, Hans Petter Selasky <hselaskyc2i.net> wrote:
> Hi Michael,
>
>  With regard to chan_capi there is an
"immediate" option which will make an
"s"
>  out of empty destination numbers. That might be your
problem, to make the
>  dialplan match against an empty number:
>
>         if (cd->cep) {
>
>             start_immediate = (cd->options.immediate
&& (cd->dst_telno[0] ==
>  0));
>
>             if (cd->flags.sending_complete_received
|| start_immediate) {
>
>                 /* number is complete, try to start the
PBX */
>
>                 cd_start_pbx(pp_cd, start_immediate ?
"s" :
>  &(cd->dst_telno[0]));
>                 cd = *pp_cd;
>             }
>

Thanks for the tip!

I've added immediate=yes to the generated configs and am
now
generating an 's' extension for the provider's incoming
extension.
I'll post back again once some users can test these
changes.

Regards,
-Michael

_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.c
om--

Asterisk-BSD mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-bsd

Hi,

Could you send me your updated "extensions.conf"
?

Does the following extension exist: 'ISDN Out-incoming' ?

I think it is something in the dialplan that is wrong.

--HPS

_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.c
om--

Asterisk-BSD mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-bsd

On Mon, Mar 17, 2008 at 9:56 AM, Michael Iedema
<michaelaskozia.com> wrote:
> On 3/16/08, Hans Petter Selasky <hselaskyc2i.net> wrote:
>  > Hi Michael,
>  >
>  >  With regard to chan_capi there is an
"immediate" option which will make an
"s"
>  >  out of empty destination numbers. That might be
your problem, to make the
>  >  dialplan match against an empty number:
>  >
>  >         if (cd->cep) {
>  >
>  >             start_immediate =
(cd->options.immediate && (cd->dst_telno[0]
==
>  >  0));
>  >
>  >             if
(cd->flags.sending_complete_received || start_immediate)
{
>  >
>  >                 /* number is complete, try to
start the PBX */
>  >
>  >                 cd_start_pbx(pp_cd,
start_immediate ? "s" :
>  >  &(cd->dst_telno[0]));
>  >                 cd = *pp_cd;
>  >             }
>  >
>
>  Thanks for the tip!
>
>  I've added immediate=yes to the generated configs and
am now
>  generating an 's' extension for the provider's
incoming extension.
>  I'll post back again once some users can test these
changes.
>

Unfortunately, still no luck here. Incoming calls using
PTP-TE mode
are still failing for multiple people. We've tried:

 - setting immediate=yes / no
 - setting isdnmode=did / msn

Every time, there is no destination extension set. Since
these are
incoming calls which are failing, I can rule out mistakes in
the
Dial() settings. I have a stack of capi.conf,
extensions.conf and
d-channel traces from a few different people but cannot
locate what's
going wrong. The latest of these submissions is here[1].

More documentation about how capi.conf and isdnconfig should
be
configured using PTP-TE mode specifically would be great if
there are
some floating around. Any and all help is appreciated.

Regards,
-Michael

[1] http://lists.askozia.com/pbx-users/showmsg.php?id=10/08

_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.c
om--

Asterisk-BSD mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-bsd

On Wed, Mar 26, 2008 at 7:12 PM, Hans Petter Selasky
<hselaskyc2i.net> wrote:
> Hi,
>
>  Could you send me your updated
"extensions.conf" ?
>
>  Does the following extension exist: 'ISDN
Out-incoming' ?
>
>  I think it is something in the dialplan that is
wrong.
>
>  --HPS
>

That log was copied from the web interface which replaces
the internal
names with the user given names. I've disabled it in future
releases.
The context is below:

; incoming provider : ISDN Out
[ISDN-PROVIDER-159382343347c6f0e545af0-incoming]


When there is only one destination for incoming calls from
that line,
things work with immediate=yes. That sets an 's' extension
and
AskoziaPBX generates an 's' extension to be used. For
multiple
destinations, no number is sent and AskoziaPBX does not
generate 's'
extensions as the call must be routed.

I will have access next week to a PTP line to do my own
testing but
for now here are his extensions.conf and interface status
info.

Regards,
-Michael


extensions.conf:
---------------------------
[macro-main];
exten => s,1,Dial($,$,$)
exten => s,n,NoOp(DIALSTATUS: $)
exten => s,n,Goto(s-$,1)

;exten => s-ANSWER
;exten => s-CANCEL
;exten => s-CHANUNAVAIL
;exten => s-CONGESTION
;exten => s-DONTCALL
;exten => s-TORTURE
;exten => s-INVALIDARGS
exten => s-BUSY,1,Goto(busy,1)
exten => s-NOANSWER,1,Goto(noanswer,1)
exten => _s-.,1,Goto(noanswer,1)

; busy
exten => busy,1,GotoIf($["$" !=
""]?vm-busy,1)
exten => busy,n,Busy(10)
exten => vm-busy,1,Wait(1)
exten => vm-busy,n,Set(MISSEDCAUSE=busy)
exten =>
vm-busy,n,GosubIf($["$" !=
""]?mailnotification,1)
exten =>
vm-busy,n,ExecIf($["$" !=
""]|Busy|10)
exten => vm-busy,n,VoiceMail($,b,$)
exten => vm-busy,n,Playback(vm-goodbye)
exten => vm-busy,n,GotoIf($["$" =
"SUCCESS"]?vm-message-exit,1:vm-nomessage-noexit,1
)

; no answer
exten => noanswer,1,GotoIf($["$"
!= ""]?vm-noanswer,1)
exten => noanswer,n,Hangup()
exten => vm-noanswer,1,Wait(1)
exten => vm-noanswer,n,Set(MISSEDCAUSE=no-answer)
exten => vm-noanswer,n,VoiceMail($,u,$)
exten => vm-noanswer,n,Playback(vm-goodbye)
exten => vm-noanswer,n,GotoIf($["$"
=
"SUCCESS"]?vm-message-exit,1:vm-nomessage-noexit,1
)

; voicemail exit
exten => vm-message-exit,1,Set(MISSEDCAUSE=didntmiss)
exten => vm-message-exit,n,Hangup()
exten =>
vm-nomessage-exit,1,GosubIf($["$&quo
t; !=
""]?mailnotification,1)
exten => vm-nomessage-exit,n,Hangup()

; hangup
exten => h,1,GotoIf($["$" !=
""]?vm-h,1)
exten => h,n,Hangup()
exten => vm-h,1,ExecIf($["$" =
"ANSWER"]|Hangup)
exten => vm-h,n,ExecIf($["$" =
""]|Set|MISSEDCAUSE=hungup)
exten =>
vm-h,n,GosubIf($[("$" !=
"") &
("$" !=
"didntmiss")]?mailnotification,1)
exten => vm-h,n,Hangup()

; mail missed call notification
exten =>
mailnotification,1,Set(TMPMAILFILE=/var/spool/asterisk/tmp/e
mail-${MD5($)})
exten => mailnotification,n,System(touch $)
exten => mailnotification,n,System(echo "date:
`date`" > $)
exten => mailnotification,n,System(echo "to:
$" >>
$)
exten => mailnotification,n,System(echo "subject:
Missed Call
($)" >> $)
exten => mailnotification,n,System(echo "from:
AskoziaPBX
Notifications <mkeuterweb.de>" >>
$)
exten => mailnotification,n,System(echo >>
$)
exten => mailnotification,n,System(echo
"${STRFTIME($,,%Y-%m-%d - %H:%M)} from
${CALLERID(all)}" >>
$)
exten => mailnotification,n,System(/usr/local/bin/msmtp
-C
/etc/msmtp.conf -t < $)
exten => mailnotification,n,System(rm $)
exten => mailnotification,n,Return()

;i - Invalid
;s - Start
;h - Hangup
;t - Timeout
;T - AbsoluteTimeout
;a - Asterisk extension
;o - Operator
; Echo
[APPLICATION-MAPPING-83606076846f835d8828a4]
exten => _[*#0-9]!,1,Answer()
exten => _[*#0-9]!,2,Echo()
exten => _[*#0-9]!,3,Hangup()

; Voicemail Main
[APPLICATION-MAPPING-202623056347c6fc9bd8a97]
exten => _[*#0-9]!,1,Answer()
exten => _[*#0-9]!,2,Wait(1)
exten => _[*#0-9]!,3,VoiceMailMain()

; WakeMe
[APPLICATION-MAPPING-202074715246f8385a7a820]
exten => _[*#0-9]!,1,Answer()
exten => _[*#0-9]!,2,Set(CHANNEL(language)=en)
exten => _[*#0-9]!,3,WakeMe()
exten => _[*#0-9]!,4,Hangup()

; Milliwatt
[APPLICATION-MAPPING-48856682646f8381701993]
exten => _[*#0-9]!,1,Answer()
exten => _[*#0-9]!,2,Milliwatt()
exten => _[*#0-9]!,3,Hangup()


[public-direct-dial]

[internal]
; internal phone : AEG Tosca Fritzbox - <26>
exten => 26,1,NoOp(internal calling internal phone: AEG
Tosca Fritzbox - <26>)
exten => 26,n,Macro(main|SIP/26|26|26|tTo|20)
; internal phone : Euracom P4 - <25>
exten => 25,1,NoOp(internal calling internal phone:
Euracom P4 - <25>)
exten => 25,n,Macro(main|SIP/25|25|25|tTo|20)
; internal phone : Snom 360 - <22>
exten => 22,1,NoOp(internal calling internal phone: Snom
360 - <22>)
exten => 22,n,Macro(main|SIP/22|22|22|tTo|)
; application : Echo
exten => 00003246,1,NoOp(internal calling application:
Echo)
exten =>
00003246,n,Dial(Local/$APPLICATION-MAPPING-83606076846f835d8828a4/n)
; application : Voicemail Main
exten => 7000,1,NoOp(internal calling application:
Voicemail Main)
exten =>
7000,n,Dial(Local/$APPLICATION-MAPPING-202623056347c6fc9bd8a97/n)
; application : WakeMe
exten => 00009253,1,NoOp(internal calling application:
WakeMe)
exten =>
00009253,n,Dial(Local/$APPLICATION-MAPPING-202074715246f8385a7a820/n)
; application : Milliwatt
exten => 000064554,1,NoOp(internal calling application:
Milliwatt)
exten =>
000064554,n,Dial(Local/$APPLICATION-MAPPING-48856682646f8381701993/n)
; conference room : Default Conference - <2663>
exten => 2663,1,NoOp(internal calling conference: Default
Conference - <2663>)
exten => 2663,n,Answer()
exten => 2663,n,MeetMe(2663,M)
exten => 2663,n,Hangup()
; parking lot
include => parkedcalls
exten => 701,1,ParkedCall(701)
exten => 701,hint,park:701parkedcalls
exten => 702,1,ParkedCall(702)
exten => 702,hint,park:702parkedcalls
exten => 703,1,ParkedCall(703)
exten => 703,hint,park:703parkedcalls
exten => 704,1,ParkedCall(704)
exten => 704,hint,park:704parkedcalls
exten => 705,1,ParkedCall(705)
exten => 705,hint,park:705parkedcalls
exten => 706,1,ParkedCall(706)
exten => 706,hint,park:706parkedcalls
exten => 707,1,ParkedCall(707)
exten => 707,hint,park:707parkedcalls
exten => 708,1,ParkedCall(708)
exten => 708,hint,park:708parkedcalls
exten => 709,1,ParkedCall(709)
exten => 709,hint,park:709parkedcalls
exten => 710,1,ParkedCall(710)
exten => 710,hint,park:710parkedcalls
exten => 711,1,ParkedCall(711)
exten => 711,hint,park:711parkedcalls
exten => 712,1,ParkedCall(712)
exten => 712,hint,park:712parkedcalls
exten => 713,1,ParkedCall(713)
exten => 713,hint,park:713parkedcalls
exten => 714,1,ParkedCall(714)
exten => 714,hint,park:714parkedcalls
exten => 715,1,ParkedCall(715)
exten => 715,hint,park:715parkedcalls
exten => 716,1,ParkedCall(716)
exten => 716,hint,park:716parkedcalls
exten => 717,1,ParkedCall(717)
exten => 717,hint,park:717parkedcalls
exten => 718,1,ParkedCall(718)
exten => 718,hint,park:718parkedcalls
exten => 719,1,ParkedCall(719)
exten => 719,hint,park:719parkedcalls
exten => 720,1,ParkedCall(720)
exten => 720,hint,park:720parkedcalls
; sip device hints
exten => 22,hint,SIP/22
exten => 25,hint,SIP/25
exten => 26,hint,SIP/26

; incoming provider : ISDN Out
[ISDN-PROVIDER-159382343347c6f0e545af0-incoming]
include => public-direct-dial
exten => 22,1,NoOp(incoming call from provider: ISDN Out
to $)
exten => 22,n,Ringing()
exten => 22,n,Wait(1)
exten => 22,n,Macro(main|SIP/22|22|22|to|)

exten => 7000,1,NoOp(incoming call from provider: ISDN
Out to $)
exten => 7000,n,Ringing()
exten => 7000,n,Wait(1)
exten =>
7000,n,Dial(Local/$APPLICATION-MAPPING-202623056347c6fc9bd8a97/n)

; outgoing provider : ISDN Out
[ISDN-PROVIDER-159382343347c6f0e545af0]
exten => _9.,1,NoOp(outgoing call to provider: ISDN Out)
exten => _9.,n,Dial(CAPI/0/${EXTEN:1}/bl,,To)
exten => _9.,n,Hangup()


; AEG Tosca Fritzbox - <26>
[SIP-PHONE-125793563347e7fe6d229b2]
include => internal
include => ISDN-PROVIDER-159382343347c6f0e545af0

; Euracom P4 - <25>
[SIP-PHONE-11808374047e7f46f6d80d]
include => internal
include => ISDN-PROVIDER-159382343347c6f0e545af0

; Snom 360 - <22>
[SIP-PHONE-191033468047c6f09a41bbc]
include => internal
include => ISDN-PROVIDER-159382343347c6f0e545af0

; isdn te interface "HFC PCI Card" incoming
[isdn-unit-0-incoming]
exten => _[0-9a-zA-Z*#]!,1,Set(CHANNEL(language)=de)
exten =>
_[0-9a-zA-Z*#]!,n,Goto(ISDN-PROVIDER-159382343347c6f0e545af0
-incoming,$,1)
exten => s,1,Set(CHANNEL(language)=de)
exten =>
s,n,Goto(ISDN-PROVIDER-159382343347c6f0e545af0-incoming,${EX
TEN},1)

[outgoing-uri]
exten => _[0-9a-zA-Z*#]!,1,NoOp(Direct dialing via SIP
uri :
$$)
exten => _[0-9a-zA-Z*#]!,n,Dial(SIP/$$,,To)




-----------------------------------------------------------
Status: Interfaces

ISDN Unit 0 (HFC PCI Card)
Attached	yes
PH State	F3: Deactivated
Dialtone	enabled
Description	HFC-2BDS0 128K PCI ISDN adapter
Type	passive ISDN (Basic Rate, 2xB)
Driver Type	DRVR_DSS1_P2P_TE (point-to-point, terminal
equipment)
Channels	0x3
Serial	0xabcd
Power Save	off

_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.c
om--

Asterisk-BSD mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-bsd