vicidial.org

Guys,

Could somebody help me on this issue?

Everything is working, I can do dial out through Dahdi for my E1 Trunks, but have not success for my FXO Trunk.

Here some debug to show up, so maybe someone has a clue on what could be happening.

I have done full recompile for libpri, dahdi, openr2, asterisk.

Code: Select all

[Jan 11 11:31:39] DEBUG[3259] pbx.c: Launching 'Dial'
[Jan 11 11:31:39] VERBOSE[3259] logger.c: [Jan 11 11:31:39]     -- Executing [1532074123@default:1] Dial("SIP/cc100-00000001", "DAHDI/8/1532074123||To") in new stack
[Jan 11 11:31:39] DEBUG[3259] chan_dahdi.c: Using channel 8
[Jan 11 11:31:39] DEBUG[3259] rtp.c: Channel 'DAHDI/8-1' has no RTP, not doing anything
[Jan 11 11:31:39] DEBUG[3259] channel.c: Not copying variable DIALEDTIME.
[Jan 11 11:31:39] DEBUG[3259] channel.c: Not copying variable ANSWEREDTIME.
[Jan 11 11:31:39] DEBUG[3259] channel.c: Not copying variable DIALEDPEERNAME.
[Jan 11 11:31:39] DEBUG[3259] channel.c: Not copying variable DIALEDPEERNUMBER.
[Jan 11 11:31:39] DEBUG[3259] channel.c: Not copying variable DIALSTATUS.
[Jan 11 11:31:39] DEBUG[3259] channel.c: Not copying variable SIPCALLID.
[Jan 11 11:31:39] DEBUG[3259] channel.c: Not copying variable SIPDOMAIN.
[Jan 11 11:31:39] DEBUG[3259] channel.c: Not copying variable SIPURI.
[Jan 11 11:31:39] DEBUG[3259] chan_dahdi.c: Dialing '1532074123'
[Jan 11 11:31:39] DEBUG[3259] chan_dahdi.c: Deferring dialing... (res -1)
[Jan 11 11:31:39] DEBUG[3259] devicestate.c: Notification of state change to be queued on device/channel DAHDI/8
[Jan 11 11:31:39] VERBOSE[3259] logger.c: [Jan 11 11:31:39]     -- Called 8/1532074123
[Jan 11 11:31:39] DEBUG[3259] chan_dahdi.c: Exception on 55, channel 8
[Jan 11 11:31:39] DEBUG[3259] chan_dahdi.c: Got event Hook Transition Complete(12) on channel 8 (index 0)
[Jan 11 11:31:39] DEBUG[3259] chan_dahdi.c: Sent deferred digit string: T153207412
[Jan 11 11:31:41] DEBUG[3259] chan_dahdi.c: Exception on 55, channel 8
[Jan 11 11:31:41] DEBUG[3259] chan_dahdi.c: Got event Dial Complete(9) on channel 8 (index 0)
[Jan 11 11:31:41] DEBUG[3259] chan_dahdi.c: Enabled echo cancellation on channel 8
[Jan 11 11:31:41] DEBUG[3259] chan_dahdi.c: Engaged echo training on channel 8
[Jan 11 11:31:43] DEBUG[3259] chan_dahdi.c: Exception on 55, channel 8
[Jan 11 11:31:43] DEBUG[3259] chan_dahdi.c: Got event Dial Complete(9) on channel 8 (index 0)
[Jan 11 11:31:43] DEBUG[3259] chan_dahdi.c: Echo cancellation already on
[Jan 11 11:31:43] DEBUG[3259] chan_sip.c: Setting framing from config on incoming call
[Jan 11 11:31:43] DEBUG[3259] chan_sip.c: ** Our capability: 0x6 (gsm|ulaw) Video flag: True
[Jan 11 11:31:43] DEBUG[3259] chan_sip.c: ** Our prefcodec: 0x0 (nothing)
[Jan 11 11:31:43] DEBUG[3259] chan_sip.c: -- Done with adding codecs to SDP
[Jan 11 11:31:43] DEBUG[3259] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=67)
[Jan 11 11:31:43] DEBUG[3259] chan_sip.c: Done building SDP. Settling with this capability: 0x6 (gsm|ulaw)
[Jan 11 11:31:43] DEBUG[3259] rtp.c: Got RTCP report of 52 bytes
[Jan 11 11:31:43] DEBUG[3259] rtp.c: Ooh, format changed from unknown to ulaw
[Jan 11 11:31:43] DEBUG[3259] rtp.c: Created smoother: format: 4 ms: 20 len: 160
[Jan 11 11:31:50] DEBUG[3259] rtp.c: Got RTCP report of 96 bytes
[Jan 11 11:31:57] DEBUG[3259] rtp.c: Got RTCP report of 96 bytes
[Jan 11 11:31:58] DEBUG[3259] rtp.c: Channel '<unspecified>' has no RTP, not doing anything
[Jan 11 11:31:58] DEBUG[3259] channel.c: Hanging up channel 'DAHDI/8-1'
[Jan 11 11:31:58] DEBUG[3259] chan_dahdi.c: dahdi_hangup(DAHDI/8-1)
[Jan 11 11:31:58] DEBUG[3259] chan_dahdi.c: Hangup: channel: 8 index = 0, normal = 55, callwait = -1, thirdcall = -1
[Jan 11 11:31:58] DEBUG[3259] chan_dahdi.c: disabled echo cancellation on channel 8
[Jan 11 11:31:58] DEBUG[3259] chan_dahdi.c: Set option TDD MODE, value: OFF(0) on DAHDI/8-1
[Jan 11 11:31:58] DEBUG[3259] chan_dahdi.c: Updated conferencing on 8, with 0 conference users
[Jan 11 11:31:58] VERBOSE[3259] logger.c: [Jan 11 11:31:58]     -- Hungup 'DAHDI/8-1'
[Jan 11 11:31:58] DEBUG[3259] devicestate.c: Notification of state change to be queued on device/channel DAHDI/8
[Jan 11 11:31:58] DEBUG[3259] app_dial.c: Exiting with DIALSTATUS=CANCEL.
[Jan 11 11:31:58] DEBUG[3259] pbx.c: Spawn extension (default,1532074123,1) exited non-zero on 'SIP/cc100-00000001'
[Jan 11 11:31:58] VERBOSE[3259] logger.c: [Jan 11 11:31:58]   == Spawn extension (default, 1532074123, 1) exited non-zero on 'SIP/cc100-00000001'
[Jan 11 11:31:58] DEBUG[3259] channel.c: Soft-Hanging up channel 'SIP/cc100-00000001'
[Jan 11 11:31:58] DEBUG[3259] pbx.c: Launching 'DeadAGI'
[Jan 11 11:31:58] VERBOSE[3259] logger.c: [Jan 11 11:31:58]     -- Executing [h@default:1] DeadAGI("SIP/cc100-00000001", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0-----CANCEL----------") in new stack
[Jan 11 11:31:58] DEBUG[3259] res_agi.c: Wow, connected!
[Jan 11 11:31:58] VERBOSE[3259] logger.c: [Jan 11 11:31:58]     -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0-----CANCEL---------- completed, returning 0
[Jan 11 11:31:58] DEBUG[3259] channel.c: Hanging up channel 'SIP/cc100-00000001'
[Jan 11 11:31:58] DEBUG[3259] chan_sip.c: Hangup call SIP/cc100-00000001, SIP callid d8d8444c-08ff-4cbd-80cc-b7c044d03027)
[Jan 11 11:31:58] DEBUG[3259] chan_sip.c: Hanging up channel in state Ring (not UP)
[Jan 11 11:31:58] DEBUG[3259] devicestate.c: Notification of state change to be queued on device/channel SIP/cc100


I wonder if the "Sent deferred digit string: T153207412" with the last phone digit missing could be a reason for call not dialing out. I'm not sure.

Thanks guys.

has this device ever worked? what's the model number on the card?

Hi william,

Yes, it has worked like a charm. It's a Yeastar TDM1600 Analog Interface Card. FXO modules connected to FCT (Fixed Cellular Terminal) GSM.

So I have a GSMTRUNK like this:

Dahdi system.conf

Code: Select all

# Span 1: WCTDM/16 "YSTDM16xx REV E Board 17" (MASTER)
fxsks=1
echocanceller=mg2,1
fxsks=2
echocanceller=mg2,2
fxsks=3
echocanceller=mg2,3
fxsks=4
echocanceller=mg2,4
fxsks=5
echocanceller=mg2,5
fxsks=6
echocanceller=mg2,6
fxsks=7
echocanceller=mg2,7
fxsks=8
echocanceller=mg2,8
# channel 9, WCTDM/16/8, no module.
# channel 10, WCTDM/16/9, no module.
# channel 11, WCTDM/16/10, no module.
# channel 12, WCTDM/16/11, no module.
# channel 13, WCTDM/16/12, no module.
# channel 14, WCTDM/16/13, no module.
# channel 15, WCTDM/16/14, no module.
# channel 16, WCTDM/16/15, no module.


chan_dahdi.conf

Code: Select all

; Configuration file

[trunkgroups]
[channels]
language=es
signalling=fxs_ks
answeronpolarityswitch=yes
hanguponpolarityswitch=yes
busydetect=yes
busycount=6
echocancel=yes
echocancelwhenbridged=no
echotraining=600
txgain=0
rxgain=0
group=0
callgroup=0
pickupgroup=0
immediate=no

#include dahdi-channels.conf


included dahdi-channels.conf

Code: Select all

; Span 1: WCTDM/16 "YSTDM16xx REV E Board 17" (MASTER)
;;; line="1 WCTDM/16/0 FXSLS  (In use) (SWEC: MG2)"
signalling=fxs_ks
callerid=asreceived
group=0
context=from-telular
channel => 1
context=default

;;; line="2 WCTDM/16/1 FXSLS  (In use) (SWEC: MG2)"
signalling=fxs_ks
callerid=asreceived
group=0
context=from-telular
channel => 2
context=default

;;; line="3 WCTDM/16/2 FXSLS  (In use) (SWEC: MG2)"
signalling=fxs_ks
callerid=asreceived
group=0
context=from-telular
channel => 3
context=default

;;; line="4 WCTDM/16/3 FXSLS  (In use) (SWEC: MG2)"
signalling=fxs_ks
callerid=asreceived
group=0
context=from-telular
channel => 4
context=default

;;; line="5 WCTDM/16/4 FXSLS  (In use) (SWEC: MG2)"
signalling=fxs_ks
callerid=asreceived
group=0
context=from-telular
channel => 5
context=default

;;; line="6 WCTDM/16/5 FXSLS  (In use) (SWEC: MG2)"
signalling=fxs_ks
callerid=asreceived
group=0
context=from-telular
channel => 6
context=default

;;; line="7 WCTDM/16/6 FXSLS  (In use) (SWEC: MG2)"
signalling=fxs_ks
callerid=asreceived
group=0
context=from-telular
channel => 7
context=default

;;; line="8 WCTDM/16/7 FXSLS  (In use) (SWEC: MG2)"
signalling=fxs_ks
callerid=asreceived
group=0
context=from-telular
channel => 8
context=default


Status from Dahdi for this span:

Code: Select all

asterisk:~ # cat /proc/dahdi/1
Span 1: WCTDM/16 "YSTDM16xx REV E Board 17"

      1 WCTDM/16/0 FXSKS (In use) (SWEC: MG2)
      2 WCTDM/16/1 FXSKS (In use) (SWEC: MG2)
      3 WCTDM/16/2 FXSKS (In use) (SWEC: MG2)
      4 WCTDM/16/3 FXSKS (In use) (SWEC: MG2)
      5 WCTDM/16/4 FXSKS (In use) (SWEC: MG2)
      6 WCTDM/16/5 FXSKS (In use) (SWEC: MG2)
      7 WCTDM/16/6 FXSKS (In use) (SWEC: MG2)
      8 WCTDM/16/7 FXSKS (In use) (SWEC: MG2)
      9 WCTDM/16/8 Reserved
     10 WCTDM/16/9 Reserved
     11 WCTDM/16/10 Reserved
     12 WCTDM/16/11 Reserved
     13 WCTDM/16/12 Reserved
     14 WCTDM/16/13 Reserved
     15 WCTDM/16/14 Reserved
     16 WCTDM/16/15 Reserved



status from CLi

Code: Select all

asterisk*CLI> dahdi show status
Description                              Alarms     IRQ        bpviol     CRC4     
YSTDM16xx REV E Board 17          OK         0          0          0         
T2XXP (PCI) Card 0 Span 1           OK         0          0          0         
T2XXP (PCI) Card 0 Span 2      NCONFIGUR 0          0          0


So everything from HW and drivers seems to be OK, but asterisk refuse to dial correctly through this span.

Any clue?....

Forgot to post this info,

CLi: dahdi show channel 1

Code: Select all

dahdi show channel 1
Channel: 1LI>
File Descriptor: 18
Span: 1k*CLI>
Extension:
Dialing: no
Context: from-telular
Caller ID:
Calling TON: 0
Caller ID name:
Destroy: 0LI>
InAlarm: 0
Signalling Type: FXS Kewlstart
Radio: 0
Owner: <None>
Real: <None>
Callwait: <None>
Threeway: <None>
Confno: -1
Propagated Conference: -1
Real in conference: 0
DSP: no
Relax DTMF: no
Dialing/CallwaitCAS: 0/0
Default law: ulaw
Fax Handled: no
Pulse phone: no
Echo Cancellation: 128 taps, currently OFF
Actual Confinfo: Num/0, Mode/0x0000
Actual Confmute: No
Hookstate (FXS only): Onhook


CLi: dahdi show channel 1

Code: Select all

asterisk*CLI> dahdi show channel 8
Channel: 8LI>
File Descriptor: 25
Span: 1k*CLI>
Extension:
Dialing: no
Context: from-telular
Caller ID:
Calling TON: 0
Caller ID name:
Destroy: 0LI>
InAlarm: 0
Signalling Type: FXS Kewlstart
Radio: 0
Owner: <None>
Real: <None>
Callwait: <None>
Threeway: <None>
Confno: -1LI>
Propagated Conference: -1
Real in conference: 0
DSP: no
Relax DTMF: no
Dialing/CallwaitCAS: 0/0
Default law: ulaw
Fax Handled: no
Pulse phone: no
Echo Cancellation: 128 taps, currently OFF
Actual Confinfo: Num/0, Mode/0x0000
Actual Confmute: No
Hookstate (FXS only): Offhook


do you have any CLI from when this worked properly for comparison?

can you install it in a pure plain asterisk and make it work, so we can then have something to compare? (i expect it has instructions available for that ... and vicidial is just somehow complicating the environment too much and something misfires ...)

Yes William,

I have a image from the system with this card working.

This image was made previous setting Vicidial parameters specifics for client.
Also was made because client needed a iaxmodem+hylafax config added to his Dialer.

Maybe something was broken in the process.

I will restore this basic service image, and make comparison or diff some files.

Any way I will post results.

Thanks

well, iaxmodema and hylafax being added would certainly add ... complexity to the system. so definitely get a deep debug of the "successful" run, then see if you can dupe it in the new system. then see if JUST changing the configuration files "back" fixes it if so, you have but to change the configuration file settings one at a time to find your culprit.

if it's a package that installed and is not compatible, that may take more time to identify.

William,

Here is a DEBUG of working fxsks call, from restored system

Code: Select all

[Jan 12 14:10:38] DEBUG[11263] pbx.c: Launching 'Dial'
[Jan 12 14:10:38] VERBOSE[11263] logger.c: [Jan 12 14:10:38]     -- Executing [1532074679@default:1] Dial("SIP/cc100-00000000", "DAHDI/g0/1/1532074679||To") in new stack
[Jan 12 14:10:38] DEBUG[11263] chan_dahdi.c: Using channel 1
[Jan 12 14:10:38] DEBUG[11263] dsp.c: dsp busy pattern set to 0,0
[Jan 12 14:10:38] DEBUG[11263] rtp.c: Channel 'DAHDI/1-1' has no RTP, not doing anything
[Jan 12 14:10:38] DEBUG[11263] channel.c: Not copying variable DIALEDTIME.
[Jan 12 14:10:38] DEBUG[11263] channel.c: Not copying variable ANSWEREDTIME.
[Jan 12 14:10:38] DEBUG[11263] channel.c: Not copying variable DIALEDPEERNAME.
[Jan 12 14:10:38] DEBUG[11263] channel.c: Not copying variable DIALEDPEERNUMBER.
[Jan 12 14:10:38] DEBUG[11263] channel.c: Not copying variable DIALSTATUS.
[Jan 12 14:10:38] DEBUG[11263] channel.c: Not copying variable SIPCALLID.
[Jan 12 14:10:38] DEBUG[11263] channel.c: Not copying variable SIPDOMAIN.
[Jan 12 14:10:38] DEBUG[11263] channel.c: Not copying variable SIPURI.
[Jan 12 14:10:38] DEBUG[11263] chan_dahdi.c: Dialing '1/1532074679'
[Jan 12 14:10:38] DEBUG[11263] chan_dahdi.c: Deferring dialing... (res -1)
[Jan 12 14:10:38] DEBUG[11263] devicestate.c: Notification of state change to be queued on device/channel DAHDI/1
[Jan 12 14:10:38] VERBOSE[11263] logger.c: [Jan 12 14:10:38]     -- Called g0/1/1532074679
[Jan 12 14:10:39] DEBUG[11263] chan_dahdi.c: Exception on 48, channel 1
[Jan 12 14:10:39] DEBUG[11263] chan_dahdi.c: Got event Hook Transition Complete(12) on channel 1 (index 0)
[Jan 12 14:10:39] DEBUG[11263] chan_dahdi.c: Sent deferred digit string: T1/153207467
[Jan 12 14:10:41] DEBUG[11263] chan_dahdi.c: Exception on 48, channel 1
[Jan 12 14:10:41] DEBUG[11263] chan_dahdi.c: Got event Dial Complete(9) on channel 1 (index 0)
[Jan 12 14:10:41] DEBUG[11263] chan_dahdi.c: Enabled echo cancellation on channel 1
[Jan 12 14:10:41] DEBUG[11263] chan_dahdi.c: Engaged echo training on channel 1
[Jan 12 14:10:42] DEBUG[11263] chan_dahdi.c: Exception on 48, channel 1
[Jan 12 14:10:42] DEBUG[11263] chan_dahdi.c: Got event Dial Complete(9) on channel 1 (index 0)
[Jan 12 14:10:42] DEBUG[11263] chan_dahdi.c: Echo cancellation already on
[Jan 12 14:10:42] DEBUG[11263] chan_sip.c: Setting framing from config on incoming call
[Jan 12 14:10:42] DEBUG[11263] chan_sip.c: ** Our capability: 0x6 (gsm|ulaw) Video flag: True
[Jan 12 14:10:42] DEBUG[11263] chan_sip.c: ** Our prefcodec: 0x0 (nothing)
[Jan 12 14:10:42] DEBUG[11263] chan_sip.c: -- Done with adding codecs to SDP
[Jan 12 14:10:42] DEBUG[11263] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=64)
[Jan 12 14:10:42] DEBUG[11263] chan_sip.c: Done building SDP. Settling with this capability: 0x6 (gsm|ulaw)
[Jan 12 14:10:42] DEBUG[11263] rtp.c: Got RTCP report of 52 bytes
[Jan 12 14:10:42] DEBUG[11263] rtp.c: Ooh, format changed from unknown to ulaw
[Jan 12 14:10:42] DEBUG[11263] rtp.c: Created smoother: format: 4 ms: 20 len: 160
[Jan 12 14:10:49] DEBUG[11263] rtp.c: Got RTCP report of 96 bytes
[Jan 12 14:10:54] DEBUG[11263] chan_dahdi.c: Exception on 48, channel 1
[Jan 12 14:10:54] DEBUG[11263] chan_dahdi.c: Got event Polarity Reversal(17) on channel 1 (index 0)
[Jan 12 14:10:54] DEBUG[11263] chan_dahdi.c: Answering on polarity switch!
[Jan 12 14:10:54] DEBUG[11263] devicestate.c: Notification of state change to be queued on device/channel DAHDI/1
[Jan 12 14:10:54] DEBUG[11263] chan_dahdi.c: Polarity Reversal event occured - DEBUG 1: channel 1, state 6, pol= 1, aonp= 1, honp= 1, pdelay= 600, tv= 0
[Jan 12 14:10:54] DEBUG[11263] chan_dahdi.c: Polarity Reversal detected but NOT hanging up (too close to answer event) on channel 1, state 6
[Jan 12 14:10:54] DEBUG[11263] chan_dahdi.c: Polarity Reversal event occured - DEBUG 2: channel 1, state 6, pol= 1, aonp= 1, honp= 1, pdelay= 600, tv= 0
[Jan 12 14:10:54] VERBOSE[11263] logger.c: [Jan 12 14:10:54]     -- DAHDI/1-1 answered SIP/cc100-00000000
[Jan 12 14:10:54] DEBUG[11263] devicestate.c: Notification of state change to be queued on device/channel SIP/cc100
[Jan 12 14:10:54] DEBUG[11263] chan_sip.c: SIP answering channel: SIP/cc100-00000000
[Jan 12 14:10:54] DEBUG[11263] chan_sip.c: Setting framing from config on incoming call
[Jan 12 14:10:54] DEBUG[11263] chan_sip.c: ** Our capability: 0x6 (gsm|ulaw) Video flag: True
[Jan 12 14:10:54] DEBUG[11263] chan_sip.c: ** Our prefcodec: 0x0 (nothing)
[Jan 12 14:10:54] DEBUG[11263] chan_sip.c: -- Done with adding codecs to SDP
[Jan 12 14:10:54] DEBUG[11263] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=64)
[Jan 12 14:10:54] DEBUG[11263] chan_sip.c: Done building SDP. Settling with this capability: 0x6 (gsm|ulaw)
[Jan 12 14:10:54] DEBUG[11263] chan_dahdi.c: Requested indication 20 on channel DAHDI/1-1
[Jan 12 14:10:54] DEBUG[11263] rtp.c: Got RTCP report of 120 bytes
[Jan 12 14:10:57] DEBUG[11263] channel.c: Didn't get a frame from channel: SIP/cc100-00000000
[Jan 12 14:10:57] DEBUG[11263] chan_dahdi.c: Requested indication 20 on channel DAHDI/1-1
[Jan 12 14:10:57] DEBUG[11263] channel.c: Bridge stops bridging channels SIP/cc100-00000000 and DAHDI/1-1
[Jan 12 14:10:57] DEBUG[11263] pbx.c: Launching 'DeadAGI'
[Jan 12 14:10:57] VERBOSE[11263] logger.c: [Jan 12 14:10:57]     -- Executing [h@default:1] DeadAGI("SIP/cc100-00000000", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0-----ANSWER-----19-----3") in new stack
[Jan 12 14:10:57] DEBUG[11263] res_agi.c: Wow, connected!
[Jan 12 14:10:57] VERBOSE[11263] logger.c: [Jan 12 14:10:57]     -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0-----ANSWER-----19-----3 completed, returning 0
[Jan 12 14:10:57] DEBUG[11263] channel.c: Hanging up channel 'DAHDI/1-1'
[Jan 12 14:10:57] DEBUG[11263] chan_dahdi.c: dahdi_hangup(DAHDI/1-1)
[Jan 12 14:10:57] DEBUG[11263] chan_dahdi.c: Hangup: channel: 1 index = 0, normal = 48, callwait = -1, thirdcall = -1
[Jan 12 14:10:57] DEBUG[11263] chan_dahdi.c: disabled echo cancellation on channel 1
[Jan 12 14:10:57] DEBUG[11263] chan_dahdi.c: Set option TDD MODE, value: OFF(0) on DAHDI/1-1
[Jan 12 14:10:57] DEBUG[11263] chan_dahdi.c: Updated conferencing on 1, with 0 conference users
[Jan 12 14:10:57] VERBOSE[11263] logger.c: [Jan 12 14:10:57]     -- Hungup 'DAHDI/1-1'
[Jan 12 14:10:57] DEBUG[11263] devicestate.c: Notification of state change to be queued on device/channel DAHDI/1
[Jan 12 14:10:57] DEBUG[11263] rtp.c: Channel '<unspecified>' has no RTP, not doing anything
[Jan 12 14:10:57] DEBUG[11263] app_dial.c: Exiting with DIALSTATUS=ANSWER.
[Jan 12 14:10:57] DEBUG[11263] pbx.c: Spawn extension (default,1532074679,1) exited non-zero on 'SIP/cc100-00000000'
[Jan 12 14:10:57] VERBOSE[11263] logger.c: [Jan 12 14:10:57]   == Spawn extension (default, 1532074679, 1) exited non-zero on 'SIP/cc100-00000000'
[Jan 12 14:10:57] DEBUG[11263] channel.c: Soft-Hanging up channel 'SIP/cc100-00000000'
[Jan 12 14:10:57] DEBUG[11263] channel.c: Hanging up channel 'SIP/cc100-00000000'
[Jan 12 14:10:57] DEBUG[11263] chan_sip.c: Hangup call SIP/cc100-00000000, SIP callid 8d459211-cbbb-46a0-8b0a-a9cb1b2850c5)
[Jan 12 14:10:57] DEBUG[11263] devicestate.c: Notification of state change to be queued on device/channel SIP/cc100



I found some diff on dsp.c & rtp.c. I'm not sure about it's meaning.

so ... your system, tell me what the first difference is? (i hate reading)

Well after debugging some calls, I found that asterisk is dialing out only the first 9 numbers.

So if I have a phone number that reads '1532074679', asterisk will call '153207467', even If I add more digits to phone number, it will dial only first 9 of them.

I think it is a redux bug, since this system (dialer) got nothing more than TDM1600 drivers loaded.

Aaaand ... your dial plan for the carrier you set up? (do not fool yourself into thinking this must be done in extensions.conf directly, as that is not true even if others do it ...).

Aaaand ... your dial plan for the carrier you set up? (do not fool yourself into thinking this must be done in extensions.conf directly, as that is not true even if others do it ...).

If not sure where it is, you can get a look at it via the asterisk console:

Code: Select all

show dialplan 1532074679@default

replacing the number above with your actual dialstring including dial prefix and phone_code/phone_number all together.

Yeaa..

it is what

Code: Select all

show dialplan 1532074679@default


is using from extensions.conf

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[gsm]
;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
;;;;;context included on default;;;;
;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
exten => _15XXXX.,1,Dial(${GSMTRUNK}/${EXTEN},,To)
exten => _015XXXX.,1,Dial(${GSMTRUNK}/${EXTEN},,To)

exten => i,1,Playback(busycell)
exten => i,n,wait(1)
exten => i,n,Hangup()
exten => t,1,Playback(busycell)
exten => t,n,wait(1)
exten => t,n,Hangup();


Those are supposed to be THREE LINES EACH. These extra lines are not discardable. Note lines 1 and three and add them to both of your existing versions using your extension instead of _91NXXNXXXXXX in each.

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exten => _91NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91NXXNXXXXXX,2,Dial(${GLOBALVARIABLEHERE}/${EXTEN:1},,tTor)
exten => _91NXXNXXXXXX,3,Hangup


Holy Jesus, what a fool of me.

I didn't realize that these lines must be there.

I will add them to extensions.conf and try again.

Thanks.

Well, after deep testing this isn't working.

Dial trough FXO span end with deferred calls, maybe not passing DTMF to PSTN or something.

For some reason it work's fine on ubuntu + asterisk 1.4.21.2-vici + zaptel 1.4.12.1.

But it doesn't on Redux 3.0.9 - 3.x or GoAutodial 2.0.

Main differences are Asterisk 1.4.27 (Goautodial) or 1.4.38 (redux) + Dahdi + Kernel

Are your configuration files otherwise identical?

Have you tried setting up without the FXO, testing to be sure the system works, and then manually adding the FXO with the manufacturer's directions after?