NO JOY [SOLVED] Piaf 3 / Incrediblepbx11 - TDM410 - Incoming (ok), but no outgoing?

[Shadowfire]

Hope everyone is doing well. I got both the Piaf3 and Incrediblepbx11 install. My six phone and extensions seem to be working internally. I am able to call in from mobile or external lines. My issue I am having is the outgoing calls - the system tells me "Your call can not be completed as dialed. Please check the number and dial again.".

here is Asterisk CLI when I try to dial out -

[May 26 10:46:34] Connected to Asterisk 11.16.0 currently running on localhost (pid = 1997)
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Executing [8632554436@from-internal:1] ResetCDR("SIP/700-00000002", "") in new stack
-- Executing [8632554436@from-internal:2] NoCDR("SIP/700-00000002", "") in new stack
-- Executing [8632554436@from-internal:3] Progress("SIP/700-00000002", "") in new stack
-- Executing [8632554436@from-internal:4] Wait("SIP/700-00000002", "1") in new stack
> 0xb7305ea8 -- Probation passed - setting RTP source address to 192.168.1.92:11784
-- Executing [8632554436@from-internal:5] Progress("SIP/700-00000002", "") in new stack
-- Executing [8632554436@from-internal:6] Playback("SIP/700-00000002", "silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer") in new stack
-- <SIP/700-00000002> Playing 'silence/1.gsm' (language 'en')
-- <SIP/700-00000002> Playing 'cannot-complete-as-dialed.gsm' (language 'en')
-- <SIP/700-00000002> Playing 'check-number-dial-again.gsm' (language 'en')
== Spawn extension (from-internal, 8632554436, 6) exited non-zero on 'SIP/700-00000002'
-- Executing [h@from-internal:1] Hangup("SIP/700-00000002", "") in new stack
== Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/700-00000002'


System Info and configs below:
----------------------------------------

CentOS release 6.7 (Final)
PIAF 3.0
IncrediblePBX11
Digium, Inc. Wildcard TDM410 4-port analog card
(6) Yealink - T21 E2 Phones

Using DAHDI Config DID's

Analog Hardware
Type Ports Action
FXO Ports 1,2,3 Edit
FXS Ports 4 Edit

Analog FXO Ports

Port 1 Settings:
Signaling: Kewl Start
Group: 0
Context: from-pstn
Port 2 Settings:
Signaling: Kewl Start
Group: 0
Context: from-pstn

Port 3 Settings:
Signaling: Kewl Start
Group: 0
Context: from-pstn

Using DAHDI Channels

Channel 1 - Description: Main Line 1 / DID: 8636448241
Channel 2 - Description: Main Line 2 / DID: 8636499217

-----------------------------
chan_dahdi.conf
;--------------------------------------------------------------------------------;
; Do NOT edit this file as it is auto-generated by FreePBX. All modifications to ;
; this file must be done via the web gui. There are alternative files to make ;
; custom modifications, details at: http://freepbx.org/configuration_files ;
;--------------------------------------------------------------------------------;
;

[general]

; generated by module
#include chan_dahdi_general.conf

; for user additions not provided by module
#include chan_dahdi_general_custom.conf

[channels]
language=en
busydetect=yes
busycount=10
usecallerid=yes
callwaiting=yes
usecallingpres=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=no
echotraining=no
immediate=no
faxdetect=no
rxgain=0.0
txgain=0.0
module_name=wctdm24xxp
opermode_checkbox=0
opermode=USA
alawoverride_checkbox=0
alawoverride=0
fxs_honor_mode_checkbox=0
fxs_honor_mode=0
boostringer_checkbox=0
boostringer=0
lowpower_checkbox=0
lowpower=0
fastringer_checkbox=0
fastringer=0
ringdetect_checkbox=0
ringdetect=0
mwi_checkbox=0
mwi=none
neon_voltage=
neon_offlimit=
echocan_nlp_type=0
echocan_nlp_threshold=
echocan_nlp_max_supp=

; for user additions not provided by module
#include chan_dahdi_channels_custom.conf

; include dahdi groups defined by DAHDI module of FreePBX
#include chan_dahdi_groups.conf

; include dahdi extensions defined in FreePBX
#include chan_dahdi_additional.conf

----------------

chan_dahdi_groups.conf
;--------------------------------------------------------------------------------;
; Do NOT edit this file as it is auto-generated by FreePBX. All modifications to ;
; this file must be done via the web gui. There are alternative files to make ;
; custom modifications, details at: http://freepbx.org/configuration_files ;
;--------------------------------------------------------------------------------;
;


signalling=fxs_ks
context=from-pstn
group=0
channel=>1

signalling=fxs_ks
context=from-pstn
group=0
channel=>2

signalling=fxs_ks
context=from-pstn
group=0
channel=>3

signalling=fxo_ks
context=from-analog
group=0
channel=>4


----------------

dahdi-channels.conf

----------------------------------------------
; Autogenerated by /usr/sbin/dahdi_genconf on Mon May 16 14:06:16 2016
; If you edit this file and execute /usr/sbin/dahdi_genconf again,
; your manual changes will be LOST.
; Dahdi Channels Configurations (chan_dahdi.conf)
;
; This is not intended to be a complete chan_dahdi.conf. Rather, it is intended
; to be #include-d by /etc/chan_dahdi.conf that will include the global settings
;

; Span 1: WCTDM/0 "Wildcard TDM410P" (MASTER)
;;; line="1 WCTDM/0/0 FXSKS"
signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
channel => 1
callerid=
group=
context=default

;;; line="2 WCTDM/0/1 FXSKS"
signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
channel => 2
callerid=
group=
context=default

;;; line="3 WCTDM/0/2 FXSKS"
signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
channel => 3
callerid=
group=
context=default

;;; line="4 WCTDM/0/3 FXOKS"
signalling=fxo_ks
callerid="Channel 4" <4004>
mailbox=4004
group=5
context=from-internal
channel => 4
callerid=
mailbox=
group=
context=default

---------------


Any help would be appreciated.

Thanks,

-SF-

 

[Shadowfire]

Okay.... Looks like it has something to do with Outbound Route, I have got it to ring, and seems to be dailing out, but it just keeps ringing and no pickup.

Ex: I can dial my cell number and the Yealink T21 E2 rings like it is dialing out, but on my cell phone I get no call.

When I originally setup outbound (no outbound working) the pattern was as listed below:

Dial Patterns that will use this Route?
(prepend) + | prefix | [1800NXXXXXX / CallerID ]

(prepend) + | prefix | [1844NXXXXXX / CallerID ]

(prepend) + | prefix | [1855NXXXXXX / CallerID ]

(prepend) + | prefix | [1866NXXXXXX / CallerID ]

(prepend) + | prefix | [1877NXXXXXX / CallerID ]

(prepend) + | prefix | [1888NXXXXXX / CallerID ]

(prepend) + | prefix | [NXXNXXXXXX / CallerID]

(prepend) + | prefix | [NXXXXXX / CallerID]

(prepend) + | prefix | [411 / CallerID]

(prepend) + | prefix | [311 / CallerID ]

(prepend) + | prefix | [911 / CallerID]

--------------------------------------

Then I dropped all of those and just did a Dial Patter like below and it started to dial, but never pickedup:

(prepend) + | prefix | [NXXNXXXXXX / CallerID]

(prepend) + | prefix | [NXXXXXX / CallerID]

________________________

In the Asterisk CLI, I get the following -

http://pastebin.com/U81xkrsS

____________

Okay, I fixed the Dial Pattern to the correct prfixes and all. It acts like it is outbound, but still no ring on my cell that I call from the phone system side with a Yealink T21E2 phone. ( get calls coming in to Yealink, dials out but doesn't connect to cell or other phone externally.)


Any thoughts?

Thanks,
-SF-

 

[Shadowfire]

Okay... This is to funny now.

Saddly I feel like the old calgone commercial... Calgone take me away!!! LOL!! it's the silly things in life eh?

I finally fixed the outbound by dial pattern and rebuilding the DAHDI Trunk and now I can dial out on the Yealink T21PE2,

BUT wait for it! Wait for it!!! I can not call in now.

It tells me " The number you have dialed is not in service, please check the number and try again"

Here is the Asterisk CLI:


== Spawn extension (macro-dialout-trunk, s, 22) exited non-zero on 'SIP/700-0000000b' in macro 'dialout-trunk'
== Spawn extension (from-internal, 8632554436, 8) exited non-zero on 'SIP/700-0000000b'
-- Starting simple switch on 'DAHDI/1-1'
-- Executing [s@from-pstn:1] NoOp("DAHDI/1-1", "No DID or CID Match") in new stack
-- Executing [s@from-pstn:2] Answer("DAHDI/1-1", "") in new stack
-- Executing [s@from-pstn:3] Wait("DAHDI/1-1", "2") in new stack
-- Executing [s@from-pstn:4] Playback("DAHDI/1-1", "ss-noservice") in new stack
-- <DAHDI/1-1> Playing 'ss-noservice.gsm' (language 'en')
-- Executing [s@from-pstn:5] SayAlpha("DAHDI/1-1", "") in new stack
-- Executing [s@from-pstn:6] Hangup("DAHDI/1-1", "") in new stack
== Spawn extension (from-pstn, s, 6) exited non-zero on 'DAHDI/1-1'
-- Executing [h@from-pstn:1] Macro("DAHDI/1-1", "hangupcall,") in new stack
-- Executing [s@macro-hangupcall:1] GotoIf("DAHDI/1-1", "1?theend") in new stack
-- Goto (macro-hangupcall,s,3)
-- Executing [s@macro-hangupcall:3] ExecIf("DAHDI/1-1", "0?Set(CDR(recordingfile)=)") in new stack
-- Executing [s@macro-hangupcall:4] Hangup("DAHDI/1-1", "") in new stack
== Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'DAHDI/1-1' in macro 'hangupcall'
== Spawn extension (from-pstn, h, 1) exited non-zero on 'DAHDI/1-1'
-- Hanging up on 'DAHDI/1-1'
-- Hungup 'DAHDI/1-1'


I think it has to do with -

-- Executing [s@from-pstn:1] NoOp("DAHDI/1-1", "No DID or CID Match") in new stack

Probably the DID setup I have.. I am going to check that now.


If you know what is the problem hit me up with a reply I would be super appreciative!

 

[Shadowfire]

By the way I had orginally used from-analog for DAHDI and was working with that a first, but I wasn't getting any where with outbound calls so I switch it to from-pstn context.

I am once again going to switch it to from-analog and see if I get some love.

Anywho have a great day everyone. Let you know what happens.

Thanks,

-SF-

 

[Shadowfire]

Well.. I love it! GOD is so good!!! I got it! With help for my provider! and no I am not talking about any earthly provider.

I switched it back to from-analog and it was working like it should!

Thanks anyway.

-SF-