SIP to SIP works, FXO to SIP works, SIP to FXO :-(

64_Bit_Hacker

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#1
Hi Guys/Girls/Quadrupeds ,

I reposted this to the help section from FreePBX.

I have got an interesting predicament. I have scoured the forums and the web, but to no avail. Any help or insight would greatly be appreciated.

Quick History:

Elastix core version 1.6.2.10
Server has SIP trunk. Calls are coming in and out over it just fine.

Users:
I have multiple users on SIP and PSTN.

PSTN is handled by two YSTDM8xx REV E Boards with FXS modules. Now PSTN users can phone other PSTN users on the network, they can also phone SIP users and make calls to external numbers. SIP users can do the same except they cannot phone PSTN users.

I am getting a Channel unavailable on the PSTN user to which the SIP call is being passed.

Interesting thing:

I see this on my SIP:

dialparties.agi: Methodology of ring is 'none'
-- dialparties.agi: Added extension 10 to extension map
-- dialparties.agi: Extension 10 cf is disabled
-- dialparties.agi: Extension 10 do not disturb is disabled
dialparties.agi: EXTENSION_STATE: 0 (NOT_INUSE)
dialparties.agi: Extension 10 has ExtensionState: 0

but on PSTN I get an extenionstate of 4?

dialparties.agi: Methodology of ring is 'none'
-- dialparties.agi: Added extension 17 to extension map
-- dialparties.agi: Extension 17 cf is disabled
-- dialparties.agi: Extension 17 do not disturb is disabled
dialparties.agi: EXTENSION_STATE: 4 (UNAVAILABLE)
dialparties.agi: Extension 17 has ExtensionState: 4

Pasting some info below:

dahdi show channels

Chan Extension Context Language MOH Interpret Blocked State
pseudo default default In Service
1 from-internal default In Service
2 from-internal default In Service
3 from-internal default In Service
4 from-internal default In Service
5 from-internal default In Service
6 from-internal default In Service
7 from-internal default In Service
8 from-internal default In Service
9 from-internal default In Service
10 from-internal default In Service
11 from-internal default In Service
12 from-internal default In Service

Example Channel:

The odd thing here is that signalling shows as FXO, but the web gui and chan_dadi indicates FXS. Also on a side note, where can I set the default 'law' to g729 from ulaw?

dahdi show channel 1

Channel: 1
File Descriptor: 18
Span: 1
Extension:
Dialing: no
Context: from-internal
Caller ID: 13
Calling TON: 0
Caller ID name: device
Mailbox: 13@device
Destroy: 0
InAlarm: 0
Signalling Type: FXO Kewlstart
Radio: 0
Owner: <None>
Real: <None>
Callwait: <None>
Threeway: <None>
Confno: -1
Propagated Conference: -1
Real in conference: 0
DSP: no
Busy Detection: no
TDD: no
Relax DTMF: no
Dialing/CallwaitCAS: 0/0
Default law: ulaw
Fax Handled: no
Pulse phone: no
DND: no
Echo Cancellation:
128 taps
(unless TDM bridged) currently OFF
Wait for dialtone: 0ms
Actual Confinfo: Num/0, Mode/0x0000
Actual Confmute: No
Hookstate (FXS only): Onhook

cat chan_dahdi.conf

[channels]
context=from-pstn
signalling=fxs_ks
rxwink=300 ; Atlas seems to use long (250ms) winks
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=no
faxdetect=incoming
echotraining=800
rxgain=0.0
txgain=0.0
callgroup=1
pickupgroup=1

;Uncomment these lines if you have problems with the disconection of your analog lines
;busydetect=yes
;busycount=3


immediate=no

#include dahdi-channels.conf
#include chan_dahdi_additional.conf

As you can see below, I think it's because my state shows as unavailable.

core show hint 1
10@ext-local : SIP/10&Custom:DND10 State:InUse Watchers 0
13@ext-local : DAHDI/1&Custom:DND13 State:Unavailable Watchers 0
14@ext-local : SIP/14&Custom:DND14 State:Idle Watchers 0
15@ext-local : DAHDI/2&Custom:DND15 State:InUse Watchers 0
16@ext-local : DAHDI/3&Custom:DND16 State:Unavailable Watchers 0
17@ext-local : DAHDI/4&Custom:DND17 State:Unavailable Watchers 0
18@ext-local : DAHDI/5&Custom:DND18 State:Unavailable Watchers 0
19@ext-local : DAHDI/6&Custom:DND19 State:Unavailable Watchers 0

dahdi_scan

[1]
active=yes
alarms=OK
description=YSTDM8xx REV E Board 9
name=WCTDM/8
manufacturer=YEASTAR
devicetype=YSTDM8xx REV E
location=PCI Bus 02 Slot 01
basechan=1
totchans=8
irq=169
type=analog
port=1,FXS
port=2,FXS
port=3,FXS
port=4,FXS
port=5,FXS
port=6,FXS
port=7,FXS
port=8,FXS
[2]
active=yes
alarms=OK
description=YSTDM8xx REV E Board 9
name=WCTDM/8
manufacturer=YEASTAR
devicetype=YSTDM8xx REV E
location=PCI Bus 02 Slot 02
basechan=9
totchans=8
irq=233
type=analog
port=9,FXS
port=10,FXS
port=11,FXS
port=12,FXS
port=13,none
port=14,none
port=15,none
port=16,none

Thx
Carl
 

64_Bit_Hacker

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#2

trymes

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#3
If I understand correctly, you have multiple Dahdi extensions and multiple SIP extensions. The Sip Extensions cannot dial the Dahdi extensions. Can the Dahdi extensions dial the other dahdi extensions?

How have you set up the extensions in FreePBX?

Also, I do not believe that you can use G.729 over the Dahdi Channels, they are G.711 by definition. You would not gain anything from using G.729, either, except less transcoding, I suppose. Unless you are talking about a massive amount of traffic, though, I would imagine that your server can handle it just fine.
 

64_Bit_Hacker

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#4
Hi Trymes,
Thank you for the reply. You are correct on my setup, the Dahdi extensions can phone other Dahdi's and can also phone out over the SIP trunk. I understand it uses ulaw, I don't care if it stays ulaw, as long as SIP can phone the extension. All extensions were setup via the GUI.

My SIP codec preference in Tools > Asterisk SIP is g729, alaw, ulaw. This is the order at my SIP provider G.729a, G.711 A-law, G.711 u-law for my Trunk. The only way I could get calls in and out fine was if I set the order like that.

Not even my PBX can phone the Dahdi extensions, if I route an inbound route to it then I receive the standard extension unavailable. One of the SIP phones(switchboard) codecs is set as PCMU, PCMA, G729 and G722, which cannot phone any Dahdi. Could it have something to do with dahdi_transcode? I do not have it listed below, and I don't know how to install it.

lsmod | grep dahdi
dahdi_echocan_oslec 36096 12
echo 38912 1 dahdi_echocan_oslec
dahdi 239408 28 dahdi_echocan_oslec,ystdm8xx
crc_ccitt 35265 1 dahdi

I'm running out of ideas here. Is there anymore info I can post which would help?
 

trymes

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#5
Asterisk should transcode between the SIP and Dahdi extensions, but it can't hurt to enable ulaw & alaw for the SIP extensions and verify that isn't the problem (I really doubt it is, though).

Can the Dahdi extensions phone the SIP extensions?

As for more info, can you post the contents of your dahdi_channels file?

Lastly, you ought to be able to contact your card vendor and get support from them, unless you bought one of the el cheapo cards.

Tom
 

64_Bit_Hacker

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#6
Dahdi can phone SIP yes, basically Dahdi can phone all, except for SIP which can't phone Dahdi and the Server (on inbound routes) which can't go to Dahdi.

Dahdi_channels and chan_dahdi files below. I have contacted the supplier, their unaware of the issue but will tests it. And also tried contacted the manufacturer, have not heard anything from them yet. My card is a Yeastar TDM800 : http://www.yeastar.com/Products/TDM800.asp.

cat /etc/asterisk/dahdi-channels.conf
; Autogenerated by /usr/sbin/dahdi_genconf on Thu Oct 14 09:49:08 2010
; 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/8 "YSTDM8xx REV E Board 9" (MASTER)
;;; line="1 WCTDM/8/0 FXOKS"
signalling=fxo_ks
callerid="Channel 1" <4001>
mailbox=4001
group=5
context=from-internal
channel => 1
callerid=
mailbox=
group=
context=default

;;; line="2 WCTDM/8/1 FXOKS"
signalling=fxo_ks
callerid="Channel 2" <4002>
mailbox=4002
group=5
context=from-internal
channel => 2
callerid=
mailbox=
group=
context=default

;;; line="3 WCTDM/8/2 FXOKS"
signalling=fxo_ks
callerid="Channel 3" <4003>
mailbox=4003
group=5
context=from-internal
channel => 3
callerid=
mailbox=
group=
context=default

;;; line="4 WCTDM/8/3 FXOKS"
signalling=fxo_ks
callerid="Channel 4" <4004>
mailbox=4004
group=5
context=from-internal
channel => 4
callerid=
mailbox=
group=
context=default

;;; line="5 WCTDM/8/4 FXOKS"
signalling=fxo_ks
callerid="Channel 5" <4005>
mailbox=4005
group=5
context=from-internal
channel => 5
callerid=
mailbox=
group=
context=default

;;; line="6 WCTDM/8/5 FXOKS"
signalling=fxo_ks
callerid="Channel 6" <4006>
mailbox=4006
group=5
context=from-internal
channel => 6
callerid=
mailbox=
group=
context=default

;;; line="7 WCTDM/8/6 FXOKS"
signalling=fxo_ks
callerid="Channel 7" <4007>
mailbox=4007
group=5
context=from-internal
channel => 7
callerid=
mailbox=
group=
context=default

;;; line="8 WCTDM/8/7 FXOKS"
signalling=fxo_ks
callerid="Channel 8" <4008>
mailbox=4008
group=5
context=from-internal
channel => 8
callerid=
mailbox=
group=
context=default


; Span 2: WCTDM/8 "YSTDM8xx REV E Board 9"
;;; line="9 WCTDM/8/0 FXOKS"
signalling=fxo_ks
callerid="Channel 9" <4009>
mailbox=4009
group=5
context=from-internal
channel => 9
callerid=
mailbox=
group=
context=default

;;; line="10 WCTDM/8/1 FXOKS"
signalling=fxo_ks
callerid="Channel 10" <4010>
mailbox=4010
group=5
context=from-internal
channel => 10
callerid=
mailbox=
group=
context=default

;;; line="11 WCTDM/8/2 FXOKS"
signalling=fxo_ks
callerid="Channel 11" <4011>
mailbox=4011
group=5
context=from-internal
channel => 11
callerid=
mailbox=
group=
context=default

;;; line="12 WCTDM/8/3 FXOKS"
signalling=fxo_ks
callerid="Channel 12" <4012>
mailbox=4012
group=5
context=from-internal
channel => 12
callerid=
mailbox=
group=
context=default

cat /etc/asterisk/chan_dahdi.conf
; Auto-generated by /usr/sbin/hardware_detector
[trunkgroups]

[channels]
context=from-internal
signalling=fxs_ks
rxwink=300 ; Atlas seems to use long (250ms) winks
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=no
faxdetect=incoming
echotraining=800
rxgain=0.0
txgain=0.0
callgroup=1
pickupgroup=1

;Uncomment these lines if you have problems with the disconection of your analog lines
;busydetect=yes
;busycount=3


immediate=no

#include dahdi-channels.conf
#include chan_dahdi_additional.conf


Thank you again for the help and insights.
 

fraggle4

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#7
It looks like you are using Oslec echo cancellation.

Per http://www.rowetel.com/blog/oslec.html
Make sure echotraining is disabled when using Oslec – this is not supported and if enabled will cause the channel to be silent (i.e. no audio will pass through)
;echotraining=800

Can't hurt to try...
 

64_Bit_Hacker

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#8
I just tried this. I think the main issue could be the State. Where when the SIP users tries to phone Dhadi, the server then tries to contact Dahdi to route the call and according to it, it's unavailable. Which makes sense why they can phone out over Dahdi, because the signal is coming into the TDM card from the phone.

Could one set the state, or change something to detect state?

13@ext-local : DAHDI/1&Custom:DND13 State:Unavailable Watchers 0
14@ext-local : SIP/14&Custom:DND14 State:Idle Watchers 0
15@ext-local : DAHDI/2&Custom:DND15 State:Unavailable Watchers 0
16@ext-local : DAHDI/3&Custom:DND16 State:Unavailable Watchers 0
17@ext-local : DAHDI/4&Custom:DND17 State:Unavailable Watchers 0
18@ext-local : DAHDI/5&Custom:DND18 State:Unavailable Watchers 0
19@ext-local : DAHDI/6&Custom:DND19 State:Unavailable Watchers 0
20@ext-local : DAHDI/7&Custom:DND20 State:Unavailable Watchers 0
21@ext-local : DAHDI/8&Custom:DND21 State:Unavailable Watchers 0
23@ext-local : DAHDI/9&Custom:DND23 State:Unavailable Watchers 0
 

fraggle4

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#9
I'm afraid your barking up the wrong tree with your last, what you are seeing is normal.

-= Registered Asterisk Dial Plan Hints =-
1101@ext-local : DAHDI/1&Custom:DND11 State:Unavailable Watchers 0
1152@ext-local : DAHDI/38&Custom:DND1 State:Unavailable Watchers 0
1155@ext-local : SIP/1155&Custom:DND1 State:Idle Watchers 0
 

64_Bit_Hacker

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#10
Ah well. I thought so much as my FXO's at the office show Unavail all the time aswell. I see I never answered echotraining fully, uncommented restarted asterisk. Tested. Still the same.
 

trymes

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#11
I notice that your channels are set in group 5. How have you set up your extensions in FreePBX?
 

64_Bit_Hacker

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#12
I just went to extensions, Chose generic ZAP. Entered User extension example 13, Display Name like Secretary and Sip Alias 13 and then at channel, simply the channel number into which it's plugged in 13's case it is 1.

I recently to no change installed the DAHDi Config 2.8.0.1 module, this gives me an option of generic dahdi but makes no difference, set extensions to test, same issue.
 

fraggle4

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#13
If it helps, here's some call progress from a SIP to dahdi extension. You should see something very similar.....

dialparties.agi: Extension 1210 has ExtensionState: 4
-- dialparties.agi: Checking CW and CFB status for extension 1210
-- dialparties.agi: dbset CALLTRACE/1210 to 1155
-- dialparties.agi: Filtered ARG3: 1210
== Manager 'admin' logged off from 127.0.0.1
-- AGI Script dialparties.agi completed, returning 0
-- Executing [s@macro-dial:7] Dial("SIP/1155-000003e0", "DAHDI/30r2||trwW") in new stack
-- Called 30r2
-- DAHDI/30-1 is ringing

Ignore the r2 bit, that's just distictive ringing.
 

trymes

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#14
Other than eliminating the SIP Alias, I am out of ideas here. You could start fresh with your DAHDi config, I suppose. However, especially because you are on FreePBX 2.8, I would suggest that you ask your question at freepbx.org as well. You might get more help there.

Tom
 

64_Bit_Hacker

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#15
Hi Tom + Fraggle,

Full call test below, maybe i'm missing something.

== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using UDPTL TOS bits 184
== Using UDPTL CoS mark 5
-- Executing [13@from-internal:1] Macro("SIP/10-00000071", "exten-vm,novm,13") in new stack
-- Executing [s@macro-exten-vm:1] Macro("SIP/10-00000071", "user-callerid,") in new stack
-- Executing [s@macro-user-callerid:1] Set("SIP/10-00000071", "AMPUSER=10") in new stack
-- Executing [s@macro-user-callerid:2] GotoIf("SIP/10-00000071", "0?report") in new stack
-- Executing [s@macro-user-callerid:3] ExecIf("SIP/10-00000071", "1?Set(REALCALLERIDNUM=10)") in new stack
-- Executing [s@macro-user-callerid:4] Set("SIP/10-00000071", "AMPUSER=10") in new stack
-- Executing [s@macro-user-callerid:5] Set("SIP/10-00000071", "AMPUSERCIDNAME=Switchboard") in new stack
-- Executing [s@macro-user-callerid:6] GotoIf("SIP/10-00000071", "0?report") in new stack
-- Executing [s@macro-user-callerid:7] Set("SIP/10-00000071", "AMPUSERCID=10") in new stack
-- Executing [s@macro-user-callerid:8] Set("SIP/10-00000071", "CALLERID(all)="Switchboard" <10>") in new stack
-- Executing [s@macro-user-callerid:9] ExecIf("SIP/10-00000071", "0?Set(CHANNEL(language)=)") in new stack
-- Executing [s@macro-user-callerid:10] GotoIf("SIP/10-00000071", "0?continue") in new stack
-- Executing [s@macro-user-callerid:11] Set("SIP/10-00000071", "__TTL=64") in new stack
-- Executing [s@macro-user-callerid:12] GotoIf("SIP/10-00000071", "1?continue") in new stack
-- Goto (macro-user-callerid,s,19)
-- Executing [s@macro-user-callerid:19] NoOp("SIP/10-00000071", "Using CallerID "Switchboard" <10>") in new stack
-- Executing [s@macro-exten-vm:2] Set("SIP/10-00000071", "RingGroupMethod=none") in new stack
-- Executing [s@macro-exten-vm:3] Set("SIP/10-00000071", "VMBOX=novm") in new stack
-- Executing [s@macro-exten-vm:4] Set("SIP/10-00000071", "__EXTTOCALL=13") in new stack
-- Executing [s@macro-exten-vm:5] Set("SIP/10-00000071", "CFUEXT=") in new stack
-- Executing [s@macro-exten-vm:6] Set("SIP/10-00000071", "CFBEXT=") in new stack
-- Executing [s@macro-exten-vm:7] Set("SIP/10-00000071", "RT=""") in new stack
-- Executing [s@macro-exten-vm:8] Macro("SIP/10-00000071", "record-enable,13,IN") in new stack
-- Executing [s@macro-record-enable:1] GotoIf("SIP/10-00000071", "1?check") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing [s@macro-record-enable:4] ExecIf("SIP/10-00000071", "0?MacroExit()") in new stack
-- Executing [s@macro-record-enable:5] GotoIf("SIP/10-00000071", "0?Group:OUT") in new stack
-- Goto (macro-record-enable,s,15)
-- Executing [s@macro-record-enable:15] GotoIf("SIP/10-00000071", "1?IN") in new stack
-- Goto (macro-record-enable,s,20)
-- Executing [s@macro-record-enable:20] ExecIf("SIP/10-00000071", "1?MacroExit()") in new stack
-- Executing [s@macro-exten-vm:9] Macro("SIP/10-00000071", "dial-one,,tr,13") in new stack
-- Executing [s@macro-dial-one:1] Set("SIP/10-00000071", "DEXTEN=13") in new stack
-- Executing [s@macro-dial-one:2] Set("SIP/10-00000071", "DIALSTATUS_CW=") in new stack
-- Executing [s@macro-dial-one:3] GosubIf("SIP/10-00000071", "0?screen,1") in new stack
-- Executing [s@macro-dial-one:4] GosubIf("SIP/10-00000071", "0?cf,1") in new stack
-- Executing [s@macro-dial-one:5] GotoIf("SIP/10-00000071", "1?skip1") in new stack
-- Goto (macro-dial-one,s,8)
-- Executing [s@macro-dial-one:8] GotoIf("SIP/10-00000071", "0?nodial") in new stack
-- Executing [s@macro-dial-one:9] GotoIf("SIP/10-00000071", "0?continue") in new stack
-- Executing [s@macro-dial-one:10] Set("SIP/10-00000071", "EXTHASCW=") in new stack
-- Executing [s@macro-dial-one:11] GotoIf("SIP/10-00000071", "1?next1:cwinusebusy") in new stack
-- Goto (macro-dial-one,s,12)
-- Executing [s@macro-dial-one:12] GotoIf("SIP/10-00000071", "0?docfu:skip3") in new stack
-- Goto (macro-dial-one,s,16)
-- Executing [s@macro-dial-one:16] GotoIf("SIP/10-00000071", "1?next2:continue") in new stack
-- Goto (macro-dial-one,s,17)
-- Executing [s@macro-dial-one:17] GotoIf("SIP/10-00000071", "1?continue") in new stack
-- Goto (macro-dial-one,s,25)
-- Executing [s@macro-dial-one:25] GotoIf("SIP/10-00000071", "0?nodial") in new stack
-- Executing [s@macro-dial-one:26] GosubIf("SIP/10-00000071", "1?dstring,1:dlocal,1") in new stack
-- Executing [dstring@macro-dial-one:1] Set("SIP/10-00000071", "DSTRING=") in new stack
-- Executing [dstring@macro-dial-one:2] Set("SIP/10-00000071", "DEVICES=13") in new stack
-- Executing [dstring@macro-dial-one:3] ExecIf("SIP/10-00000071", "0?Return()") in new stack
-- Executing [dstring@macro-dial-one:4] ExecIf("SIP/10-00000071", "0?Set(DEVICES=3)") in new stack
-- Executing [dstring@macro-dial-one:5] Set("SIP/10-00000071", "LOOPCNT=1") in new stack
-- Executing [dstring@macro-dial-one:6] Set("SIP/10-00000071", "ITER=1") in new stack
-- Executing [dstring@macro-dial-one:7] Set("SIP/10-00000071", "THISDIAL=ZAP/1") in new stack
-- Executing [dstring@macro-dial-one:8] GosubIf("SIP/10-00000071", "1?zap2dahdi,1") in new stack
-- Executing [zap2dahdi@macro-dial-one:1] ExecIf("SIP/10-00000071", "0?Return()") in new stack
-- Executing [zap2dahdi@macro-dial-one:2] Set("SIP/10-00000071", "NEWDIAL=") in new stack
-- Executing [zap2dahdi@macro-dial-one:3] Set("SIP/10-00000071", "LOOPCNT2=1") in new stack
-- Executing [zap2dahdi@macro-dial-one:4] Set("SIP/10-00000071", "ITER2=1") in new stack
-- Executing [zap2dahdi@macro-dial-one:5] Set("SIP/10-00000071", "THISPART2=ZAP/1") in new stack
-- Executing [zap2dahdi@macro-dial-one:6] ExecIf("SIP/10-00000071", "1?Set(THISPART2=DAHDI/1)") in new stack
-- Executing [zap2dahdi@macro-dial-one:7] Set("SIP/10-00000071", "NEWDIAL=DAHDI/1&") in new stack
-- Executing [zap2dahdi@macro-dial-one:8] Set("SIP/10-00000071", "ITER2=2") in new stack
-- Executing [zap2dahdi@macro-dial-one:9] GotoIf("SIP/10-00000071", "0?begin2") in new stack
-- Executing [zap2dahdi@macro-dial-one:10] Set("SIP/10-00000071", "THISDIAL=DAHDI/1") in new stack
-- Executing [zap2dahdi@macro-dial-one:11] Return("SIP/10-00000071", "") in new stack
-- Executing [dstring@macro-dial-one:9] Set("SIP/10-00000071", "DSTRING=DAHDI/1&") in new stack
-- Executing [dstring@macro-dial-one:10] Set("SIP/10-00000071", "ITER=2") in new stack
-- Executing [dstring@macro-dial-one:11] GotoIf("SIP/10-00000071", "0?begin") in new stack
-- Executing [dstring@macro-dial-one:12] Set("SIP/10-00000071", "DSTRING=DAHDI/1") in new stack
-- Executing [dstring@macro-dial-one:13] Return("SIP/10-00000071", "") in new stack
-- Executing [s@macro-dial-one:27] GotoIf("SIP/10-00000071", "0?nodial") in new stack
-- Executing [s@macro-dial-one:28] GotoIf("SIP/10-00000071", "1?skiptrace") in new stack
-- Goto (macro-dial-one,s,30)
-- Executing [s@macro-dial-one:30] Set("SIP/10-00000071", "D_OPTIONS=tr") in new stack
-- Executing [s@macro-dial-one:31] ExecIf("SIP/10-00000071", "0?SIPAddHeader(Alert-Info: )") in new stack
-- Executing [s@macro-dial-one:32] ExecIf("SIP/10-00000071", "0?SIPAddHeader()") in new stack
-- Executing [s@macro-dial-one:33] ExecIf("SIP/10-00000071", "0?SetMusicOnHold()") in new stack
-- Executing [s@macro-dial-one:34] GosubIf("SIP/10-00000071", "0?qwait,1") in new stack
-- Executing [s@macro-dial-one:35] Set("SIP/10-00000071", "__CWIGNORE=") in new stack
-- Executing [s@macro-dial-one:36] Set("SIP/10-00000071", "__KEEPCID=TRUE") in new stack
-- Executing [s@macro-dial-one:37] Dial("SIP/10-00000071", "DAHDI/1,,tr") in new stack
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [s@macro-dial-one:38] ExecIf("SIP/10-00000071", "0?Set(DIALSTATUS=)") in new stack
-- Executing [s@macro-dial-one:39] GosubIf("SIP/10-00000071", "0?s-CHANUNAVAIL,1") in new stack
-- Executing [s@macro-dial-one:40] MacroExit("SIP/10-00000071", "") in new stack
-- Executing [s@macro-exten-vm:10] GotoIf("SIP/10-00000071", "0?exit,return") in new stack
-- Executing [s@macro-exten-vm:11] Set("SIP/10-00000071", "SV_DIALSTATUS=CHANUNAVAIL") in new stack
-- Executing [s@macro-exten-vm:12] GosubIf("SIP/10-00000071", "0?docfu,1") in new stack
-- Executing [s@macro-exten-vm:13] GosubIf("SIP/10-00000071", "0?docfb,1") in new stack
-- Executing [s@macro-exten-vm:14] Set("SIP/10-00000071", "DIALSTATUS=CHANUNAVAIL") in new stack
-- Executing [s@macro-exten-vm:15] NoOp("SIP/10-00000071", "Voicemail is 'novm'") in new stack
-- Executing [s@macro-exten-vm:16] GotoIf("SIP/10-00000071", "1?s-CHANUNAVAIL,1") in new stack
-- Goto (macro-exten-vm,s-CHANUNAVAIL,1)
-- Executing [s-CHANUNAVAIL@macro-exten-vm:1] NoOp("SIP/10-00000071", "IVR_RETVM: IVR_CONTEXT: ") in new stack
-- Executing [s-CHANUNAVAIL@macro-exten-vm:2] GotoIf("SIP/10-00000071", "0?exit,1") in new stack
-- Executing [s-CHANUNAVAIL@macro-exten-vm:3] PlayTones("SIP/10-00000071", "congestion") in new stack
== Spawn extension (macro-exten-vm, s-CHANUNAVAIL, 3) exited non-zero on 'SIP/10-00000071' in macro 'exten-vm'
== Spawn extension (from-internal, 13, 1) exited non-zero on 'SIP/10-00000071'
-- Executing [h@from-internal:1] Macro("SIP/10-00000071", "hangupcall") in new stack
-- Executing [s@macro-hangupcall:1] GotoIf("SIP/10-00000071", "1?noautomon") in new stack
-- Goto (macro-hangupcall,s,3)
-- Executing [s@macro-hangupcall:3] NoOp("SIP/10-00000071", "TOUCH_MONITOR_OUTPUT=") in new stack
-- Executing [s@macro-hangupcall:4] GotoIf("SIP/10-00000071", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,7)
-- Executing [s@macro-hangupcall:7] GotoIf("SIP/10-00000071", "1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,10)
-- Executing [s@macro-hangupcall:10] GotoIf("SIP/10-00000071", "1?theend") in new stack
-- Goto (macro-hangupcall,s,12)
-- Executing [s@macro-hangupcall:12] Hangup("SIP/10-00000071", "") in new stack
== Spawn extension (macro-hangupcall, s, 12) exited non-zero on 'SIP/10-00000071' in macro 'hangupcall'
== Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/10-00000071'

Also Fraggle, if you do a cat /etc/asterisk/dahdi-channels.conf, what do you see on your channels. "Does it show (In use)"? ;;; line="12 WCTDM/8/3 FXOKS (In use)"

Thx
Carl
 

fraggle4

Joined
Apr 22, 2009
Messages
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#16
The dialplan in freepbx 2.8 must be a lot different from 2.7, but I think this is the key item:

-- Executing [s@macro-dial-one:37] Dial("SIP/10-00000071", "DAHDI/1,,tr") in new stack
== Everyone is busy/congested at this time (1:0/0/1)

I am beginning to think that dahdi is not loaded correctly, even though it appears to be OK.
My setup is so different that it wouldn't do any good to post my dahdi-channels.conf, anyway I don't use this file.
However, this is part of my /etc/dahdi/system.conf that should look similar, these are four fxs ports.

# Span 3: WCTDM/4 "Wildcard TDM400P REV I Board 5"
fxoks=49
echocanceller=oslec,49
fxoks=50
echocanceller=oslec,50
fxoks=51
echocanceller=oslec,51
fxoks=52
echocanceller=oslec,52

I would be tempted to remove the second fxs card with the unused channels, and reconfigure, just to see if it works with just the one card. I know that's a pain, but I can't think of anything else.
Incidentally there is some good info on dahdi setup here:
http://www.cadvision.com/blanchas/Aster ... hdiHW.html
 

trymes

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Aug 19, 2009
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#17
This may be a stupid question, but have you configured these cards via the Hardware detector and told it to replace the dahdi_channels.conf file?
 

vegremy

Joined
Feb 2, 2011
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#18
I have the same problem, but after i seek on this post, and used asterisk CLI, i found is just a matter of change in the extension config page of elastix, where it says "dial" and have "ZAP/1" it must be "ZAP/channel number here" in my case: "ZAP/25" and works!!
 

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