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[faza_404 at yahoo.com]

Hi :
asterisk didn't send voice message to my mail(fatemefatah2000 at yahoo.com).My main configured files are:
extensions.conf:
[from-pstn]
exten => 9711315,1,Dial(SIP/3000,30)
exten => 9711315,2,VoiceMail(3000 at ff_tutorial)
exten => 9711315,3,PlayBack(vm-goodbye)
exten => 9711315,4,HangUp()
sip.conf:
[3000]
type=friend
username=3000
secret=1234567
host=dynamic
context=from-pstn
mailbox=3000 at ff_tutorial
voicemail.conf:
[ff_tutorial]
3000 => 1234567,3000,fatemefatah2000 at yahoo.com

And these are in console:

Jun 29 12:08:11 WARNING[24207]: chan_zap.c:8093 zt_pri_error: Call Reference Length not supported: 0
??? -- Accepting call from '3322000' to '9711315' on channel 0/1, span 1
Jun 29 12:08:15 DEBUG[24207]: chan_zap.c:1554 zt_enable_ec: Enabled echo cancellation on channel 1
??? -- Executing Dial("Zap/1-1", "SIP/3000|30") in new stack
Jun 29 12:08:15 DEBUG[24257]: chan_sip.c:1889 create_addr_from_peer: Setting NAT on RTP to 0
Jun 29 12:08:15 DEBUG[24257]: chan_sip.c:2085 sip_call: Outgoing Call for 3000
??? -- Called 3000
Jun 29 12:08:16 DEBUG[24203]: chan_sip.c:1468 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '3f4c7d54725d24f1635f9b920f5c6508 at 192.168.200.65' Request 102: Found
??? -- SIP/3000-08941d28 is ringing
Jun 29 12:08:16 DEBUG[24257]: chan_zap.c:4981 zt_indicate: Requested indication 3 on channel Zap/1-1
Jun 29 12:08:21 WARNING[24207]: chan_zap.c:8093 zt_pri_error: Call Reference Length not supported: 0
Jun 29 12:08:31 WARNING[24207]: chan_zap.c:8093 zt_pri_error: Call Reference Length not supported: 0
Jun 29 12:08:41 WARNING[24207]: chan_zap.c:8093 zt_pri_error: Call Reference Length not supported: 0
??? -- Nobody picked up in 30000 ms
Jun 29 12:08:47 DEBUG[24257]: chan_zap.c:4981 zt_indicate: Requested indication -1 on channel Zap/1-1
Jun 29 12:08:47 DEBUG[24257]: chan_sip.c:2450 sip_hangup: update_call_counter(3000) - decrement call limit counter
Jun 29 12:08:47 DEBUG[24257]: chan_sip.c:1392 __sip_ack: Acked pending invite 102
Jun 29 12:08:47 DEBUG[24257]: chan_sip.c:1415 __sip_ack: Stopping retransmission on '3f4c7d54725d24f1635f9b920f5c6508 at 192.168.200.65' of Request 102: Match Found
Jun 29 12:08:47 DEBUG[24257]: app_dial.c:1661 dial_exec_full: Exiting with DIALSTATUS=NOANSWER.
??? -- Executing VoiceMail("Zap/1-1", "3000 at ff_tutorial") in new stack
Jun 29 12:08:47 DEBUG[24257]: channel.c:1777 ast_settimeout: Scheduling timer at 160 sample intervals
??? -- Playing 'vm-intro' (language 'en')
Jun 29 12:08:47 DEBUG[24203]: chan_sip.c:1415 __sip_ack: Stopping retransmission on '3f4c7d54725d24f1635f9b920f5c6508 at 192.168.200.65' of Request 102: Match Found
Jun 29 12:08:47 DEBUG[24203]: chan_sip.c:1415 __sip_ack: Stopping retransmission on '3f4c7d54725d24f1635f9b920f5c6508 at 192.168.200.65' of Request 102: Match Not Found
Jun 29 12:08:51 WARNING[24207]: chan_zap.c:8093 zt_pri_error: Call Reference Length not supported: 0
Jun 29 12:08:52 DEBUG[24257]: channel.c:1777 ast_settimeout: Scheduling timer at 0 sample intervals
Jun 29 12:08:52 DEBUG[24257]: channel.c:1777 ast_settimeout: Scheduling timer at 0 sample intervals
Jun 29 12:08:52 DEBUG[24257]: app.c:1235 ast_lock_path: Locked path '/var/spool/asterisk/voicemail/ff_tutorial/3000/INBOX'
Jun 29 12:08:52 DEBUG[24257]: app.c:1256 ast_unlock_path: Unlocked path '/var/spool/asterisk/voicemail/ff_tutorial/3000/INBOX'
Jun 29 12:08:52 DEBUG[24257]: channel.c:1777 ast_settimeout: Scheduling timer at 160 sample intervals
??? -- Playing 'beep' (language 'en')
Jun 29 12:08:53 DEBUG[24257]: channel.c:1777 ast_settimeout: Scheduling timer at 0 sample intervals
Jun 29 12:08:53 DEBUG[24257]: channel.c:1777 ast_settimeout: Scheduling timer at 0 sample intervals
??? -- Recording the message
Jun 29 12:08:53 DEBUG[24257]: app.c:568 ast_play_and_record_full: play_and_record: <None>, /var/spool/asterisk/voicemail/ff_tutorial/3000/tmp/GE15v9, 'wav49|gsm|wav'
Jun 29 12:08:53 DEBUG[24257]: app.c:585 ast_play_and_record_full: Recording Formats: sfmts=wav49
??? -- x=0, open writing:? /var/spool/asterisk/voicemail/ff_tutorial/3000/tmp/GE15v9 format: wav49, 0x88b0f48
??? -- x=1, open writing:? /var/spool/asterisk/voicemail/ff_tutorial/3000/tmp/GE15v9 format: gsm, 0x88b12a0
??? -- x=2, open writing:? /var/spool/asterisk/voicemail/ff_tutorial/3000/tmp/GE15v9 format: wav, 0x88b15e0
Jun 29 12:08:53 DEBUG[24257]: chan_zap.c:4981 zt_indicate: Requested indication 18 on channel Zap/1-1
Jun 29 12:09:01 WARNING[24207]: chan_zap.c:8093 zt_pri_error: Call Reference Length not supported: 0
Jun 29 12:09:11 DEBUG[24257]: chan_zap.c:3669 zt_handle_dtmfup: DTMF digit: # on Zap/1-1
??? -- User ended message by pressing #
Jun 29 12:09:11 DEBUG[24257]: channel.c:1777 ast_settimeout: Scheduling timer at 160 sample intervals
??? -- Playing 'auth-thankyou' (language 'en')
Jun 29 12:09:11 WARNING[24207]: chan_zap.c:8093 zt_pri_error: Call Reference Length not supported: 0
Jun 29 12:09:12 DEBUG[24257]: channel.c:1777 ast_settimeout: Scheduling timer at 0 sample intervals
Jun 29 12:09:12 DEBUG[24257]: channel.c:1777 ast_settimeout: Scheduling timer at 0 sample intervals
Jun 29 12:09:12 DEBUG[24257]: app.c:1235 ast_lock_path: Locked path '/var/spool/asterisk/voicemail/ff_tutorial/3000/INBOX'
Jun 29 12:09:12 DEBUG[24257]: app.c:1256 ast_unlock_path: Unlocked path '/var/spool/asterisk/voicemail/ff_tutorial/3000/INBOX'
Jun 29 12:09:12 DEBUG[24257]: app_voicemail.c:1695 sendmail: Attaching file '/var/spool/asterisk/voicemail/ff_tutorial/3000/INBOX/msg0011', format 'WAV', uservm is '2048', global is 2048
Jun 29 12:09:12 DEBUG[24257]: app_voicemail.c:1832 sendmail: Sent mail to fatemefatah2000 at yahoo.com with command '/usr/sbin/sendmail -t'
??? -- Executing Playback("Zap/1-1", "vm-goodbye") in new stack
Jun 29 12:09:12 DEBUG[24257]: channel.c:1777 ast_settimeout: Scheduling timer at 160 sample intervals
??? -- Playing 'vm-goodbye' (language 'en')
Jun 29 12:09:18 DEBUG[24257]: channel.c:1777 ast_settimeout: Scheduling timer at 0 sample intervals
Jun 29 12:09:18 DEBUG[24257]: channel.c:1777 ast_settimeout: Scheduling timer at 0 sample intervals
??? -- Executing Hangup("Zap/1-1", "") in new stack
? == Spawn extension (from-pstn, 9711315, 4) exited non-zero on 'Zap/1-1'
Jun 29 12:09:18 DEBUG[24257]: chan_zap.c:3015 zt_setoption: Set option AUDIO MODE, value: ON(1) on Zap/1-1
Jun 29 12:09:18 DEBUG[24257]: chan_zap.c:2496 zt_hangup: Hangup: channel: 1 index = 0, normal = 19, callwait = -1, thirdcall = -1
Jun 29 12:09:18 DEBUG[24257]: chan_zap.c:2645 zt_hangup: Not yet hungup...? Calling hangup once with icause, and clearing call
Jun 29 12:09:18 DEBUG[24257]: chan_zap.c:1586 zt_disable_ec: disabled echo cancellation on channel 1
Jun 29 12:09:18 DEBUG[24257]: chan_zap.c:2936 zt_setoption: Set option TDD MODE, value: OFF(0) on Zap/1-1
Jun 29 12:09:18 DEBUG[24257]: chan_zap.c:1523 update_conf: Updated conferencing on 1, with 0 conference users
Jun 29 12:09:18 DEBUG[24257]: chan_zap.c:3011 zt_setoption: Set option AUDIO MODE, value: OFF(0) on Zap/1-1
Jun 29 12:09:18 DEBUG[24257]: chan_zap.c:1586 zt_disable_ec: disabled echo cancellation on channel 1
??? -- Hungup 'Zap/1-1'
Jun 29 12:09:18 DEBUG[24257]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is '3322000'
Jun 29 12:09:18 DEBUG[24257]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is '3322000'
Jun 29 12:09:18 DEBUG[24257]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is '9711315'
Jun 29 12:09:18 DEBUG[24257]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is 'from-pstn'
Jun 29 12:09:18 DEBUG[24257]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is 'Zap/1-1'
Jun 29 12:09:18 DEBUG[24257]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is 'SIP/3000-08941d28'
Jun 29 12:09:18 DEBUG[24257]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is 'Hangup'
Jun 29 12:09:18 DEBUG[24257]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is '(null)'
Jun 29 12:09:18 DEBUG[24257]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is '2008-06-29 12:08:15'
Jun 29 12:09:18 DEBUG[24257]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is '2008-06-29 12:08:47'
Jun 29 12:09:18 DEBUG[24257]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is '2008-06-29 12:09:18'
Jun 29 12:09:18 DEBUG[24257]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is '63'
Jun 29 12:09:18 DEBUG[24257]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is '31'
Jun 29 12:09:18 DEBUG[24257]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is 'ANSWERED'
Jun 29 12:09:18 DEBUG[24257]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is 'DOCUMENTATION'
Jun 29 12:09:18 DEBUG[24257]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is '(null)'
Jun 29 12:09:18 DEBUG[24257]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is '1214728695.0'
Jun 29 12:09:18 DEBUG[24257]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is '(null)'
Jun 29 12:09:21 WARNING[24207]: chan_zap.c:8093 zt_pri_error: Call Reference Length not supported: 0
Jun 29 12:09:22 DEBUG[24203]: chan_sip.c:1415 __sip_ack: Stopping retransmission on '02b2d2de37422458059ff58f0b0cf1d6 at 192.168.200.65' of Request 102: Match Found
Jun 29 12:09:31 WARNING[24207]: chan_zap.c:8093 zt_pri_error: Call Reference Length not supported: 0
I'd appreciate any help.




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[ben4asterisk at yahoo.com]

Use SendDTMF.

--- On Thu, 7/3/08, Neha Punia <Neha_Punia at infosys.com> wrote:

[Neha_Punia at infosys.com]

But if I m using this SendDTMF it does not send anything



I m using it like this in extension.conf

exten => 205,1,Answer



exten => 205,n,Wait(20)



exten => 205,n,Playback(dtmf-1)



exten => 205,n,Wait(20)



exten => 205,n,SendDTMF(9)



exten => 205,n,Wait(5)



exten => 205,n,Read(digito)



exten => 205,n,SayDigits(${digito})



exten => 205,n,Hangup



on the console it only shows tht the call completed and no message about the DTMF and in the log files it shows like :



Jul 3 17:21:01 DEBUG[896] chan_sip.c: Stopping retransmission on '0b2e4fb4092a2c897558760351afa503 at 10.152.119.125' of Request 102: Match Found

Jul 3 17:21:27 DEBUG[896] chan_sip.c: Setting NAT on RTP to 0

Jul 3 17:21:27 DEBUG[896] chan_sip.c: Outgoing Call for 205

Jul 3 17:21:27 DEBUG[896] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '1ee8aac6271e35d87646b01325e09297 at 10.152.119.125' Request 102: Found

Jul 3 17:21:27 DEBUG[896] chan_sip.c: Acked pending invite 102

Jul 3 17:21:27 DEBUG[896] chan_sip.c: Stopping retransmission on '1ee8aac6271e35d87646b01325e09297 at 10.152.119.125' of Request 102: Match Found

Jul 3 17:21:27 DEBUG[896] chan_sip.c: build_route: Contact hop: <sip:205 at 10.152.119.74>

Jul 3 17:21:47 DEBUG[896] chan_sip.c: * Detected inband DTMF '1'

Jul 3 17:22:18 DEBUG[896] chan_sip.c: Stopping retransmission on '0314444d2adfe2a3776d58197149704f at 10.152.119.125' of Request 102: Match Found

Jul 3 17:22:23 DEBUG[896] pbx.c: Function result is '(null)'

Jul 3 17:22:23 DEBUG[896] pbx.c: Function result is '(null)'

Jul 3 17:22:23 DEBUG[896] pbx.c: Function result is '205'

Jul 3 17:22:23 DEBUG[896] pbx.c: Function result is 'default'

Jul 3 17:22:23 DEBUG[896] pbx.c: Function result is 'SIP/3001-008d8ce0'

Jul 3 17:22:23 DEBUG[896] pbx.c: Function result is '(null)'

Jul 3 17:22:23 DEBUG[896] pbx.c: Function result is 'Hangup'

Jul 3 17:22:23 DEBUG[896] pbx.c: Function result is '(null)'

Jul 3 17:22:23 DEBUG[896] pbx.c: Function result is '2008-07-03 17:21:27'

Jul 3 17:22:23 DEBUG[896] pbx.c: Function result is '2008-07-03 17:21:27'

Jul 3 17:22:23 DEBUG[896] pbx.c: Function result is '2008-07-03 17:22:23'

Jul 3 17:22:23 DEBUG[896] pbx.c: Function result is '56'

Jul 3 17:22:23 DEBUG[896] pbx.c: Function result is '56'

Jul 3 17:22:23 DEBUG[896] pbx.c: Function result is 'ANSWERED'

Jul 3 17:22:23 DEBUG[896] pbx.c: Function result is 'DOCUMENTATION'

Jul 3 17:22:23 DEBUG[896] pbx.c: Function result is '(null)'

Jul 3 17:22:23 DEBUG[896] pbx.c: Function result is '1215085887.0'

Jul 3 17:22:23 DEBUG[896] pbx.c: Function result is '(null)'

Jul 3 17:22:23 DEBUG[896] chan_sip.c: update_call_counter(205) - decrement call limit counter

Jul 3 17:22:23 NOTICE[896] pbx_spool.c: Call completed to SIP/3001

Jul 3 17:22:23 DEBUG[896] chan_sip.c: Stopping retransmission on '1ee8aac6271e35d87646b01325e09297 at 10.152.119.125' of Request 103: Match Found

Jul 3 17:22:24 DEBUG[896] chan_sip.c: Auto destroying call '35b080fa74bccebd47f949512fd4f324 at 10.152.119.74'

Jul 3 17:23:57 DEBUG[896] chan_sip.c: Stopping retransmission on '00641a2d0698e0610317b7a412d5c88b at 10.152.119.125' of Request 102: Match Found

Jul 3 17:24:09 DEBUG[896] chan_sip.c: Auto destroying call '35b080fa74bccebd47f949512fd4f324 at 10.152.119.74'

Jul 3 17:25:47 DEBUG[896] chan_sip.c: Stopping retransmission on '3fd49efb5e3009fd5bf480584b91febe at 10.152.119.125' of Request 102: Match Found

Jul 3 17:25:54 DEBUG[896] chan_sip.c: Auto destroying call '35b080fa74bccebd47f949512fd4f324 at 10.152.119.74'



It says "detected inband dtmf 1 but says nothing about 9.

Am I doing anything wrong in the extension.conf.





-----Original Message-----
From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Benjamin Jacob
Sent: Thursday, July 03, 2008 5:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] (no subject)





Use SendDTMF.







--- On Thu, 7/3/08, Neha Punia <Neha_Punia at infosys.com> wrote:

[bikrish at w2sindia.com]

Hello everybody
I have configures asterisk server
and i
am using TE220P digium card.? Here is the content of
the
/etc/zaptel.conf file
###########################
span=1,1,0,ccs,hdb3
bchan=1-15,17-31
dchan=16

span=2,2,0,ccs,hdb3
bchan=32-46,48-62
dchan=47


loadzone??????? = in
defaultzone???? = in

############################

the content of
/etc/asterisk/zapata.conf is as follow

############################
[channels]
context=incoming
switchtype=national
;pridialplan=national
usecallerid=yes
hidecallerid=no
callwaiting=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
echocancel=yes
rxgain=0.0
txgain=0.0
immediate=no
callprogress=no
callerid=asreceived
group=1
channel=>1-15,17-31
#############################

output of zttool is as follow

????????????????????????????????????????????????????????????????????

???????????????????????????????
│????
Alarms?????????
Span??????????????????????????????????????????????
│
???????????????????????????????
│????
RED????????????
T2XXP (PCI) Card 0 Span
1?????????????????????

???????????????????????????????
│????
OK?????????????
T2XXP (PCI) Card 0 Span
2??????????????????????

???????????????????????????????
│?????????????????????????????????????????????????????????????????
???????????????????????????????


Output of? cat /prox/zaptel/1 is as follow


??? Span 1: TE2/0/1 "T2XXP (PCI) Card 0 Span
1"
HDB3/CCS RED

?????????? 1
TE2/0/1/1
Clear (In use) RED
?????????? 2
TE2/0/1/2
Clear (In use) RED
?????????? 3
TE2/0/1/3
Clear (In use) RED
?????????? 4
TE2/0/1/4
Clear (In use) RED
?????????? 5
TE2/0/1/5
Clear (In use) RED
?????????? 6
TE2/0/1/6
Clear (In use) RED
?????????? 7
TE2/0/1/7
Clear (In use) RED
?????????? 8
TE2/0/1/8
Clear (In use) RED
?????????? 9
TE2/0/1/9
Clear (In use) RED
????????? 10 TE2/0/1/10
Clear (In use) RED
????????? 11 TE2/0/1/11
Clear (In use) RED
????????? 12 TE2/0/1/12
Clear (In use) RED
????????? 13 TE2/0/1/13
Clear (In use) RED
????????? 14 TE2/0/1/14
Clear (In use) RED
????????? 15 TE2/0/1/15
Clear (In use) RED
????????? 16 TE2/0/1/16
HDLCFCS (In use) RED
????????? 17 TE2/0/1/17
Clear (In use) RED
????????? 18 TE2/0/1/18
Clear (In use) RED
????????? 19 TE2/0/1/19
Clear (In use) RED
????????? 20 TE2/0/1/20
Clear (In use) RED
????????? 21 TE2/0/1/21
Clear (In use) RED
????????? 22 TE2/0/1/22
Clear (In use) RED
????????? 23 TE2/0/1/23
Clear (In use) RED
????????? 24 TE2/0/1/24
Clear (In use) RED
????????? 25 TE2/0/1/25
Clear (In use) RED
????????? 26 TE2/0/1/26
Clear (In use) RED
????????? 27 TE2/0/1/27
Clear (In use) RED
????????? 28 TE2/0/1/28
Clear (In use) RED
????????? 29 TE2/0/1/29
Clear (In use) RED
????????? 30 TE2/0/1/30
Clear (In use) RED
????????? 31 TE2/0/1/31
Clear (In use) RED
??????
I
am
new to asterisk and googled around , configured the asterisk
server. Now
when i make a call from outside , it give me busy
tone..? and when i
call from softphone .. it shows me as show
below


?? ??? -- Executing
[9999600833 at incoming:1]
Dial("SIP/bikrish-09b21980",
"Zap/g1/9999600833") in
new stack
[Jul? 3
19:14:34] WARNING[6018]: app_dial.c:1183
dial_exec_full: Unable to
create channel of type 'Zap' (cause 34 -
Circuit/channel
congestion)
? == Everyone is busy/congested at
this time
(1:0/1/0)
? == Auto fallthrough, channel
'SIP/bikrish-09b21980' status is 'CONGESTION'

I am not able
to
figure out the problem. Any kind of help would be appericiated.

Thanking you

bikrish

[shmaltz at gmail.com]

The number one skill for setting up asterisk is learn how to
communicate since it's a communication application

As for your problem looks like you are trying to use the wrong span
for dial out.
On Thu, Jul 3, 2008 at 8:50 AM, Bikrish Amatya <bikrish at w2sindia.com> wrote:

[abalashov at evaristes...]

C F wrote:

[asterisk.org at sedwar...]

On Thu, 3 Jul 2008, Alex Balashov wrote:

[abalashov at evaristes...]

Steve Edwards wrote:

[asterisk.org at sedwar...]

On Thu, 3 Jul 2008, Alex Balashov wrote:

[abalashov at evaristes...]

Steve Edwards wrote:

[asterisk.org at sedwar...]

On Fri, 4 Jul 2008, Peter Lindquist wrote:

[brianc at palaver.net]

Alex Balashov wrote: