Calling *21*NUMBER# for telephone redirection

Asterisk Asterisk Support
1

Hi,

I’m setting up an asterisk in our company but have a problem with it:
From our ISDN provider, we get a free service to redirect incomming calls when we are not available in the building. To enable this we used to dial 021TELNUMBER#, the first 0 being a number to ask an ouside line (on our old PBX).

Whne I try to do this from within asterisk I allways get CHAN unavailable.

More info:
– Executing [s@macro-dialout-trunk:26] Set(“SIP/41-08252748”, “the_num=21TELNUMBER%23”) in new stack
– Executing [s@macro-dialout-trunk:27] Dial(“SIP/41-08252748”, “mISDN/g:isdn/21TELNUMBER%23|300|”) in new stack
– Called g:isdn/21TELNUMBER%23
P[ 1] empty_chan_in_stack: cannot empty channel 255
== Everyone is busy/congested at this time (1:0/0/1)
– Executing [s@macro-dialout-trunk:28] Goto(“SIP/41-08252748”, “s-CHANUNAVAIL|1”) in new stack

I also tried calling to an internal number and links that to the outgoing number (so the %23 thing doesn’t occur).

– Executing [s@macro-dial:7] Dial(“SIP/41-08214f10”, “mISDN/g:isdn/210498309617#||tr”) in new stack
– Called g:isdn/21TELNUMBER#
P[ 1] empty_chan_in_stack: cannot empty channel 255
== Everyone is busy/congested at this time (1:0/0/1)
– Executing [s@macro-dial:8] Set(“SIP/41-08214f10”, “DIALSTATUS=CHANUNAVAIL”) in new stack
– Executing [s@macro-exten-vm:10] Set(“SIP/41-08214f10”, “SV_DIALSTATUS=CHANUNAVAIL”) in new stack
– Executing [s@macro-exten-vm:11] GosubIf(“SIP/41-08214f10”, “0?docfu|1”) in new stack
– Executing [s@macro-exten-vm:12] GosubIf(“SIP/41-08214f10”, “0?docfb|1”) in new stack
– Executing [s@macro-exten-vm:13] Set(“SIP/41-08214f10”, “DIALSTATUS=CHANUNAVAIL”) in new stack
– Executing [s@macro-exten-vm:14] NoOp(“SIP/41-08214f10”, “Voicemail is novm”) in new stack
– Executing [s@macro-exten-vm:15] GotoIf(“SIP/41-08214f10”, “1?s-CHANUNAVAIL|1”) in new stack

2

Are you using asterisknow? or Trixbox? Either way I would assume that you have built a calling rule telling the system that when you dial 21N to dial that through your ISDN line?

3

We are using Asterisk 1.6.1.6 with mISDN. So o asterisknow or trixbox.

We have also tried with mISDNv2 en chan_lcr but then I got following error:

[Sep 15 08:47:04] NOTICE[4348]: chan_lcr.c:1720 lcr_request: [call=NULL ast=NULL] Received request from Asterisk. (data=outgoing/21TELNR#)
[Sep 15 08:47:04] NOTICE[4348]: chan_lcr.c:315 alloc_call: [call=0 ast=NULL] Call instance allocated.
[Sep 15 08:47:04] NOTICE[4348]: chan_lcr.c:1822 lcr_call: [call=NULL ast=lcr/4] Received call from Asterisk.
[Sep 15 08:47:04] NOTICE[4348]: chan_lcr.c:351 send_message: [call=NULL ast=NULL] Sending MESSAGE_NEWREF to socket.
[Sep 15 08:47:04] NOTICE[4239]: chan_lcr.c:1269 receive_message: [call=NULL ast=NULL] Received new ref by LCR, as requested from chan_lcr. (ref=4)
[Sep 15 08:47:04] NOTICE[4239]: chan_lcr.c:583 send_setup_to_lcr: [call=4 ast=lcr/4] Sending setup to LCR. (interface=outgoing dialstring=21TELNR#, cid=40)
[Sep 15 08:47:04] NOTICE[4239]: chan_lcr.c:351 send_message: [call=NULL ast=NULL] Sending MESSAGE_SETUP to socket.
[Sep 15 08:47:05] NOTICE[4239]: chan_lcr.c:1000 lcr_in_release: [call=4 ast=lcr/4] Incomming release from LCR, releasing ref. (cause=28)
[Sep 15 08:47:05] NOTICE[4239]: chan_lcr.c:1577 queue_send: [call=0 ast=lcr/4] Sending queued HANGUP to Asterisk.
[Sep 15 08:47:05] NOTICE[4348]: chan_lcr.c:2038 lcr_hangup: [call=0 ast=lcr/4] Received hangup from Asterisk thread.
[Sep 15 08:47:05] NOTICE[4348]: chan_lcr.c:2061 lcr_hangup: [call=0 ast=lcr/4] Freeing call instance, because we have no ref AND we are requesting no ref.
[Sep 15 08:47:05] NOTICE[4348]: chan_lcr.c:290 free_call: [call=0 ast=NULL] Call instance freed.
[Sep 15 08:47:05] WARNING[4348]: pbx.c:3839 __ast_pbx_run: Channel ‘SIP/40-09cb3178’ sent into invalid extension ‘600’ in context ‘from-sip’, but no invalid handler

4

Does nobody has any idea of how to solve this issue

5

Can you please paste your extensions.conf in here, so we can see what you have set for the context?

6

We are now trying with Dahdi and an analog line:

[extensions.conf]:

[general]
static=yes
writeprotect=no
clearglobalvars=no

[globals]

[default]
include => from-sip

[from-sip]
include => misdnintern
include => misdnextern
include => out
include => omschakeling
include => queues
include => parkedcalls
include => conference

exten => 1234,1,Answer()
exten => 1234,2,Playback(pbx-invalid)
exten => 1234,3,Hangup()

exten => _XX,1,Answer()
exten => _XX,n,Dial(SIP/${EXTEN},,TtrkK)
exten => _XX,n,Hangup()


[out]
exten => 1,1,Answer()
exten => 1,n,Dial(DAHDI/1/0495042298,,A(hello-world)M(dtmf))
exten => 1,n,Goto(s-${DIALSTATUS},1)

exten => s-ANSWER,1,Background(hello-world)
exten => s-ANSWER,2,Hangup()

exten => s-NOANSWER,1,Background(pbx-invalidpark)
exten => s-NOANSWER,2,Hangup()

exten => s-UNKNOWN,1,Background(pbx-invalidpark)
exten => s-UNKNOWN,2,Hangup()

exten => s-.,1,BackGround(pbx-invalid)
exten => s-.,2,Hangup()

[misdnintern]
exten => _0X.,1,Answer()
exten => _0X.,n,Dial(DAHDI/g0/${EXTEN:1})
exten => _0X.,n,Hangup()

[misdnextern]

exten => 700,1,Dial(DAHDI/7/*21*0495042298#)

;exten => 11214911,1,Answer()
;exten => 11214911,2,Dial(SIP/50)
;exten => 11214911,3,Hangup()

exten => 11220419,1,Answer()
exten => 11220419,n,Set(CALLERID(num)=0${CALLERID(num)})
exten => 11220419,n,Queue(all,tr,,,45)
exten => 11220419,n,Hangup

[macro-dtmf]
exten => s,1,Background(vm-press)
exten => s,n,Background(digits/1)
exten => s,n,Read(NUMBER,,1)
exten => s,n,SayNumber($NUMBER)
exten => s,n,GotoIf($[ ${NUMBER} = 1 ]?100:200)
exten => s,100,Playback(auth-thankyou)
exten => s,200,Playback(vm-sorry)
exten => s,n,Hangup()

[omschakeling]

exten => _*21*X.,1,Dial(DAHDI/8/${exten})
exten => 600,1,Dial(DAHDI/8/*21*0498506822#)
exten => 604,1,Dial(dahdi/8/0495042298,,r)

It contains also a bit of test data