Inbound Calls Terminate with "Bye" vocal message

Hi all,

This is my first post here, but I’ve been using Asterisk Now almost since the begging.
I’m no Linux expert since AsteriskNOW is my only interaction with Linux, but through careful hardware selection I’ve managed to reduce problems to what I can solve by knowledge and/or searching.
This time though I’ve come to a dead end on how to solve this one problem, and I’m starting to bang my head on the wall. :stuck_out_tongue:

I’m using AsteriskNOW 1.7.1 32-bit with Asterisk 1.8 and freepbx 2.8.1, and a Digium TDM410P with 4 FXO ports without EC hardware.
I’m using 3 of the 4 FXO ports. Each port is supposed to both receive and make calls. Also a VoIP channel is configured only to make calls.

The problem resides on the fact that I’m able to make calls through all 3 analog channels but can’t receive through any of them. All I can hear is the message “bye” and then it puts the call on hold. As of now for testing purposes all 3 channels are routed to a single extension (201).

I’ve trimmed the log to what I assume is the relevant section, but if more is needed please tell me. Here it is:

[Feb 3 02:15:31] VERBOSE[3428] pbx.c: -- Executing [[email protected]:1] Macro("SIP/201-0000000e", "user-callerid,SKIPTTL,") in new stack [Feb 3 02:15:31] VERBOSE[3428] pbx.c: -- Executing [[email protected]:1] Set("SIP/201-0000000e", "AMPUSER=201") in new stack [Feb 3 02:15:31] VERBOSE[3428] pbx.c: -- Executing [[email protected]:2] GotoIf("SIP/201-0000000e", "0?report") in new stack [Feb 3 02:15:31] VERBOSE[3428] pbx.c: -- Executing [[email protected]:3] ExecIf("SIP/201-0000000e", "1?Set(REALCALLERIDNUM=201)") in new stack [Feb 3 02:15:31] VERBOSE[3428] pbx.c: -- Executing [[email protected]:4] Set("SIP/201-0000000e", "AMPUSER=201") in new stack [Feb 3 02:15:31] VERBOSE[3428] pbx.c: -- Executing [[email protected]:5] Set("SIP/201-0000000e", "AMPUSERCIDNAME=Ivone") in new stack [Feb 3 02:15:31] VERBOSE[3428] pbx.c: -- Executing [[email protected]:6] GotoIf("SIP/201-0000000e", "0?report") in new stack [Feb 3 02:15:31] VERBOSE[3428] pbx.c: -- Executing [[email protected]:7] Set("SIP/201-0000000e", "AMPUSERCID=201") in new stack [Feb 3 02:15:31] VERBOSE[3428] pbx.c: -- Executing [[email protected]:8] Set("SIP/201-0000000e", "CALLERID(all)="Ivone" <201>") in new stack [Feb 3 02:15:31] VERBOSE[3428] pbx.c: -- Executing [[email protected]:9] ExecIf("SIP/201-0000000e", "0?Set(CHANNEL(language)=)") in new stack [Feb 3 02:15:31] VERBOSE[3428] pbx.c: -- Executing [[email protected]:10] GotoIf("SIP/201-0000000e", "1?continue") in new stack [Feb 3 02:15:31] VERBOSE[3428] pbx.c: -- Goto (macro-user-callerid,s,19) [Feb 3 02:15:31] VERBOSE[3428] pbx.c: -- Executing [[email protected]:19] Set("SIP/201-0000000e", "CALLERID(number)=201") in new stack [Feb 3 02:15:31] VERBOSE[3428] pbx.c: -- Executing [[email protected]:20] Set("SIP/201-0000000e", "CALLERID(name)=Ivone") in new stack [Feb 3 02:15:31] VERBOSE[3428] pbx.c: -- Executing [[email protected]:21] NoOp("SIP/201-0000000e", "Using CallerID "Ivone" <201>") in new stack [Feb 3 02:15:31] VERBOSE[3428] pbx.c: -- Executing [[email protected]:2] NoOp("SIP/201-0000000e", "Calling Out Route: Outbound9519") in new stack [Feb 3 02:15:31] VERBOSE[3428] pbx.c: -- Executing [[email protected]:3] Set("SIP/201-0000000e", "MOHCLASS=default") in new stack [Feb 3 02:15:31] VERBOSE[3428] pbx.c: -- Executing [[email protected]:4] ExecIf("SIP/201-0000000e", "1?Set(TRUNKCIDOVERRIDE=214689519)") in new stack [Feb 3 02:15:31] VERBOSE[3428] pbx.c: -- Executing [[email protected]:5] Set("SIP/201-0000000e", "_NODEST=") in new stack [Feb 3 02:15:31] VERBOSE[3428] pbx.c: -- Executing [[email protected]:6] Macro("SIP/201-0000000e", "record-enable,201,OUT,") in new stack [Feb 3 02:15:31] VERBOSE[3428] pbx.c: -- Executing [[email protected]:1] GotoIf("SIP/201-0000000e", "1?check") in new stack [Feb 3 02:15:31] VERBOSE[3428] pbx.c: -- Goto (macro-record-enable,s,4) [Feb 3 02:15:31] VERBOSE[3428] pbx.c: -- Executing [[email protected]:4] ExecIf("SIP/201-0000000e", "0?MacroExit()") in new stack [Feb 3 02:15:31] VERBOSE[3428] pbx.c: -- Executing [[email protected]:5] GotoIf("SIP/201-0000000e", "0?Group:OUT") in new stack [Feb 3 02:15:31] VERBOSE[3428] pbx.c: -- Goto (macro-record-enable,s,15) [Feb 3 02:15:31] VERBOSE[3428] pbx.c: -- Executing [[email protected]:15] GotoIf("SIP/201-0000000e", "0?IN") in new stack [Feb 3 02:15:31] VERBOSE[3428] pbx.c: -- Executing [[email protected]:16] ExecIf("SIP/201-0000000e", "1?MacroExit()") in new stack [Feb 3 02:15:31] VERBOSE[3428] pbx.c: -- Executing [821468951[email protected]:7] Macro("SIP/201-0000000e", "dialout-trunk,3,214689517,") in new stack [Feb 3 02:15:31] VERBOSE[3428] pbx.c: -- Executing [[email protected]:1] Set("SIP/201-0000000e", "DIAL_TRUNK=3") in new stack [Feb 3 02:15:31] VERBOSE[3428] pbx.c: -- Executing [[email protected]:2] GosubIf("SIP/201-0000000e", "0?sub-pincheck,s,1") in new stack [Feb 3 02:15:31] VERBOSE[3428] pbx.c: -- Executing [[email protected]:3] GotoIf("SIP/201-0000000e", "0?disabletrunk,1") in new stack [Feb 3 02:15:31] VERBOSE[3428] pbx.c: -- Executing [[email protected]:4] Set("SIP/201-0000000e", "DIAL_NUMBER=214689517") in new stack [Feb 3 02:15:31] VERBOSE[3428] pbx.c: -- Executing [[email protected]:5] Set("SIP/201-0000000e", "DIAL_TRUNK_OPTIONS=tr") in new stack [Feb 3 02:15:31] VERBOSE[3428] pbx.c: -- Executing [[email protected]:6] Set("SIP/201-0000000e", "OUTBOUND_GROUP=OUT_3") in new stack [Feb 3 02:15:31] VERBOSE[3428] pbx.c: -- Executing [[email protected]:7] GotoIf("SIP/201-0000000e", "1?nomax") in new stack [Feb 3 02:15:31] VERBOSE[3428] pbx.c: -- Goto (macro-dialout-trunk,s,9) [Feb 3 02:15:31] VERBOSE[3428] pbx.c: -- Executing [[email protected]:9] GotoIf("SIP/201-0000000e", "0?skipoutcid") in new stack [Feb 3 02:15:31] VERBOSE[3428] pbx.c: -- Executing [[email protected]:10] Set("SIP/201-0000000e", "DIAL_TRUNK_OPTIONS=") in new stack [Feb 3 02:15:31] VERBOSE[3428] pbx.c: -- Executing [[email protected]:11] Macro("SIP/201-0000000e", "outbound-callerid,3") in new stack [Feb 3 02:15:31] VERBOSE[3428] pbx.c: -- Executing [[email protected]:1] ExecIf("SIP/201-0000000e", "0?Set(CALLERPRES()=)") in new stack [Feb 3 02:15:31] VERBOSE[3428] pbx.c: -- Executing [[email protected]:2] ExecIf("SIP/201-0000000e", "0?Set(REALCALLERIDNUM=201)") in new stack [Feb 3 02:15:31] VERBOSE[3428] pbx.c: -- Executing [[email protected]:3] GotoIf("SIP/201-0000000e", "1?normcid") in new stack [Feb 3 02:15:31] VERBOSE[3428] pbx.c: -- Goto (macro-outbound-callerid,s,6) [Feb 3 02:15:31] VERBOSE[3428] pbx.c: -- Executing [[email protected]:6] Set("SIP/201-0000000e", "USEROUTCID=") in new stack [Feb 3 02:15:31] VERBOSE[3428] pbx.c: -- Executing [[email protected]:7] Set("SIP/201-0000000e", "EMERGENCYCID=") in new stack [Feb 3 02:15:31] VERBOSE[3428] pbx.c: -- Executing [[email protected]:8] Set("SIP/201-0000000e", "TRUNKOUTCID=214689519") in new stack [Feb 3 02:15:31] VERBOSE[3428] pbx.c: -- Executing [[email protected]:9] GotoIf("SIP/201-0000000e", "1?trunkcid") in new stack [Feb 3 02:15:31] VERBOSE[3428] pbx.c: -- Goto (macro-outbound-callerid,s,12) [Feb 3 02:15:31] VERBOSE[3428] pbx.c: -- Executing [[email protected]:12] ExecIf("SIP/201-0000000e", "1?Set(CALLERID(all)=214689519)") in new stack [Feb 3 02:15:31] VERBOSE[3428] pbx.c: -- Executing [[email protected]:13] ExecIf("SIP/201-0000000e", "0?Set(CALLERID(all)=)") in new stack [Feb 3 02:15:31] VERBOSE[3428] pbx.c: -- Executing [[email protected]:14] ExecIf("SIP/201-0000000e", "1?Set(CALLERID(all)=214689519)") in new stack [Feb 3 02:15:31] VERBOSE[3428] pbx.c: -- Executing [[email protected]:15] ExecIf("SIP/201-0000000e", "0?Set(CALLERPRES()=prohib_passed_screen)") in new stack [Feb 3 02:15:31] VERBOSE[3428] pbx.c: -- Executing [[email protected]:12] GosubIf("SIP/201-0000000e", "0?sub-flp-3,s,1") in new stack [Feb 3 02:15:31] VERBOSE[3428] pbx.c: -- Executing [[email protected]:13] Set("SIP/201-0000000e", "OUTNUM=214689517") in new stack [Feb 3 02:15:31] VERBOSE[3428] pbx.c: -- Executing [[email protected]:14] Set("SIP/201-0000000e", "custom=DAHDI/2") in new stack [Feb 3 02:15:31] VERBOSE[3428] pbx.c: -- Executing [[email protected]:15] ExecIf("SIP/201-0000000e", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default))") in new stack [Feb 3 02:15:31] VERBOSE[3428] pbx.c: -- Executing [[email protected]:16] Macro("SIP/201-0000000e", "dialout-trunk-predial-hook,") in new stack [Feb 3 02:15:31] VERBOSE[3428] pbx.c: -- Executing [[email protected]:1] MacroExit("SIP/201-0000000e", "") in new stack [Feb 3 02:15:31] VERBOSE[3428] pbx.c: -- Executing [[email protected]:17] GotoIf("SIP/201-0000000e", "0?bypass,1") in new stack [Feb 3 02:15:31] VERBOSE[3428] pbx.c: -- Executing [[email protected]:18] GotoIf("SIP/201-0000000e", "0?customtrunk") in new stack [Feb 3 02:15:31] VERBOSE[3428] pbx.c: -- Executing [[email protected]:19] Dial("SIP/201-0000000e", "DAHDI/2/214689517,300,") in new stack [Feb 3 02:15:31] VERBOSE[3428] app_dial.c: -- Called 2/214689517 [Feb 3 02:15:34] WARNING[3428] chan_dahdi.c: Unable to enable echo cancellation on channel 2 (No such device) [Feb 3 02:15:34] VERBOSE[3428] app_dial.c: -- DAHDI/2-1 answered SIP/201-0000000e [Feb 3 02:15:34] VERBOSE[3429] sig_analog.c: -- Starting simple switch on 'DAHDI/1-1' [Feb 3 02:15:35] DEBUG[3429] chan_dahdi.c: CallerID number: 214687564, name: (null), flags=4 [Feb 3 02:15:35] VERBOSE[3429] pbx.c: == Starting DAHDI/1-1 at ,s,1 failed so falling back to exten 's' [Feb 3 02:15:35] VERBOSE[3429] pbx.c: == Starting DAHDI/1-1 at ,s,1 still failed so falling back to context 'default' [Feb 3 02:15:35] VERBOSE[3429] pbx.c: -- Executing [[email protected]:1] Playback("DAHDI/1-1", "vm-goodbye") in new stack [Feb 3 02:15:35] WARNING[3429] chan_dahdi.c: Unable to enable echo cancellation on channel 1 (No such device) [Feb 3 02:15:35] VERBOSE[3429] file.c: -- Playing 'vm-goodbye.ulaw' (language 'en') [Feb 3 02:15:36] VERBOSE[3430] manager.c: == Manager 'admin' logged on from 127.0.0.1 [Feb 3 02:15:36] VERBOSE[3430] manager.c: == Manager 'admin' logged off from 127.0.0.1 [Feb 3 02:15:36] VERBOSE[3429] pbx.c: -- Executing [[email protected]:2] Macro("DAHDI/1-1", "hangupcall") in new stack [Feb 3 02:15:36] VERBOSE[3429] pbx.c: -- Executing [[email protected]:1] GotoIf("DAHDI/1-1", "1?skiprg") in new stack [Feb 3 02:15:36] VERBOSE[3429] pbx.c: -- Goto (macro-hangupcall,s,4) [Feb 3 02:15:36] VERBOSE[3429] pbx.c: -- Executing [[email protected]:4] GotoIf("DAHDI/1-1", "1?skipblkvm") in new stack [Feb 3 02:15:36] VERBOSE[3429] pbx.c: -- Goto (macro-hangupcall,s,7) [Feb 3 02:15:36] VERBOSE[3429] pbx.c: -- Executing [[email protected]:7] GotoIf("DAHDI/1-1", "1?theend") in new stack [Feb 3 02:15:36] VERBOSE[3429] pbx.c: -- Goto (macro-hangupcall,s,9) [Feb 3 02:15:36] VERBOSE[3429] pbx.c: -- Executing [[email protected]:9] Hangup("DAHDI/1-1", "") in new stack [Feb 3 02:15:36] VERBOSE[3429] app_macro.c: == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'DAHDI/1-1' in macro 'hangupcall' [Feb 3 02:15:36] VERBOSE[3429] pbx.c: == Spawn extension (default, s, 2) exited non-zero on 'DAHDI/1-1' [Feb 3 02:15:36] VERBOSE[3429] pbx.c: -- Executing [[email protected]:1] Macro("DAHDI/1-1", "hangupcall,") in new stack [Feb 3 02:15:36] VERBOSE[3429] pbx.c: -- Executing [[email protected]:1] GotoIf("DAHDI/1-1", "1?skiprg") in new stack [Feb 3 02:15:36] VERBOSE[3429] pbx.c: -- Goto (macro-hangupcall,s,4) [Feb 3 02:15:36] VERBOSE[3429] pbx.c: -- Executing [[email protected]:4] GotoIf("DAHDI/1-1", "1?skipblkvm") in new stack [Feb 3 02:15:36] VERBOSE[3429] pbx.c: -- Goto (macro-hangupcall,s,7) [Feb 3 02:15:36] VERBOSE[3429] pbx.c: -- Executing [[email protected]:7] GotoIf("DAHDI/1-1", "1?theend") in new stack [Feb 3 02:15:36] VERBOSE[3429] pbx.c: -- Goto (macro-hangupcall,s,9) [Feb 3 02:15:36] VERBOSE[3429] pbx.c: -- Executing [[email protected]:9] Hangup("DAHDI/1-1", "") in new stack [Feb 3 02:15:36] VERBOSE[3429] app_macro.c: == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'DAHDI/1-1' in macro 'hangupcall' [Feb 3 02:15:36] VERBOSE[3429] pbx.c: == Spawn extension (default, h, 1) exited non-zero on 'DAHDI/1-1' [Feb 3 02:15:36] VERBOSE[3429] sig_analog.c: -- Hanging up on 'DAHDI/1-1' [Feb 3 02:15:36] VERBOSE[3429] chan_dahdi.c: -- Hungup 'DAHDI/1-1' [Feb 3 02:15:43] VERBOSE[3435] manager.c: == Manager 'admin' logged on from 127.0.0.1 [Feb 3 02:15:44] VERBOSE[3435] manager.c: == Manager 'admin' logged off from 127.0.0.1 [Feb 3 02:15:55] VERBOSE[3428] pbx.c: -- Executing [[email protected]:1] Macro("SIP/201-0000000e", "hangupcall,") in new stack [Feb 3 02:15:55] VERBOSE[3428] pbx.c: -- Executing [[email protected]:1] GotoIf("SIP/201-0000000e", "1?skiprg") in new stack [Feb 3 02:15:55] VERBOSE[3428] pbx.c: -- Goto (macro-hangupcall,s,4) [Feb 3 02:15:55] VERBOSE[3428] pbx.c: -- Executing [[email protected]:4] GotoIf("SIP/201-0000000e", "1?skipblkvm") in new stack [Feb 3 02:15:55] VERBOSE[3428] pbx.c: -- Goto (macro-hangupcall,s,7) [Feb 3 02:15:55] VERBOSE[3428] pbx.c: -- Executing [[email protected]:7] GotoIf("SIP/201-0000000e", "1?theend") in new stack [Feb 3 02:15:55] VERBOSE[3428] pbx.c: -- Goto (macro-hangupcall,s,9) [Feb 3 02:15:55] VERBOSE[3428] pbx.c: -- Executing [[email protected]:9] Hangup("SIP/201-0000000e", "") in new stack [Feb 3 02:15:55] VERBOSE[3428] app_macro.c: == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/201-0000000e' in macro 'hangupcall' [Feb 3 02:15:55] VERBOSE[3428] sig_analog.c: -- Hanging up on 'DAHDI/2-1' [Feb 3 02:15:55] VERBOSE[3428] chan_dahdi.c: -- Hungup 'DAHDI/2-1' [Feb 3 02:15:55] VERBOSE[3428] app_macro.c: == Spawn extension (macro-dialout-trunk, s, 19) exited non-zero on 'SIP/201-0000000e' in macro 'dialout-trunk' [Feb 3 02:15:55] VERBOSE[3428] pbx.c: == Spawn extension (from-internal, 8214689517, 7) exited non-zero on 'SIP/201-0000000e'

I’d be very grateful for any possible assistance.

Best Regards,
jb

I had the same problem as you, I could get it to work temporarily by adding context=from-zaptel to chan_dahdi_groups.conf but it would revert.

This is how I solved.

  1. run dahdi_genconf . this will generate (or overwrite) a file, /etc/asterisk/dahdi-channels.conf

  2. Here, you change the context from-pstn to from-zaptel. Save.

  3. Now, in the file /etc/asterisk/chan_dahdi.conf , make sure you have this line, if not, add it.
    #include dahdi-channels.conf

Restart asterisk, and you should be good to go.

Note: if you ever run dahdi_genconf again, you have to repeat the process as changes will be overwritten.

Good luck!

Ok, new problem…

I did change the settings on the file chan_dahdi_groups.conf and it did solve my problem, but it changed back the settings from “context=from-zaptel” to “context=” over night.

If I change it back to “context=from-zaptel” and reboot the system, it comes back to working order, but only for so long.

I suppose this is happening because it’s a system generated file, as stated at the file header, and as such it will change the file back whenever it is updated.

My question now is where, in freepbx webui, can I change these settings so they become permanent?

Thank you,
jb

Hi,

Here are the files:

chan_dahdi.conf

; Copied from DAHDI Module of FreePBX

[general]

#include chan_dahdi_general.conf

[channels]

; 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

As it seems, this file only points to chan_dahdi_general.conf, chan_dahdi_groups.conf and chan_dahdi_additional.conf, so here they are as well:

chan_dahdi_general.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 ;
;--------------------------------------------------------------------------------;
;

chan_dahdi_general.conf appears to be empty.

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=
group=1
channel=>1

signalling=fxs_ks
context=
group=2
channel=>2

signalling=fxs_ks
context=
group=3
channel=>3

signalling=fxs_ks
context=
group=4
channel=>4

chan_dahdi_additional.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 ;
;--------------------------------------------------------------------------------;
;

chan_dahdi_additional.conf appears to be empty.

dahdi-channels.conf

; Autogenerated by /usr/sbin/dahdi_genconf on Sun Jun 30 21:12:55 2011
; 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 Board 1” (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 FXSKS"
signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
channel => 4
callerid=
group=
context=default

What I’ve pasted here are the original file. In dahdi-channels.conf the first “context” entry of each “line” was already “from-pstn” and the second was “default”.
So, on dahdi-channels.conf I tried every permutation of both “context” of every “line” group (1 through 4) with both “context=from-pstn” and “context=from-zaptel”, but with no results.

In chan_dahdi_groups.conf I also tried to put “from-pstn” and the message change from “Bye” to “The number you have dialled is not in service”.
Finnaly I tried “from-zaptel” in this last file and wonder of wonders it worked.

Note that I did reboots afters every change, and, after the problem was solved, I defaulted everything I changed manually, except, of course, the changes that worked.

Whats bugging me is that it worked by changing parameters on a file that specifically says not to be altered manually, as this might mislead many people as it did me.

So the problem is SOLVED! :slight_smile:

I post this here for future reference as so anyone having the same problem as me can get some help from this.

Thank you so very much to sanjayws for the sugestions.

Best Regards to all,
jb

Hi can you post your chan_dahdi.conf and/or dahdi_channels.conf files? i think your contexts in there may be incorrect. Just test sake, if you see anything context=default in chan_dahdi.conf try to change that to context=from-pstn or from-zaptel…

Hi all,
I have run into the same problem, but the solution here doesn’t help me unfortunately. Outgoing calls work, incoming call ends with “bye” message and a line in a log “ERROR: FreePBX Does not use the [default] context, confguration error”. I can see just a “default” context of channels in a CLI. Yes, I ran dahdi_genconf and changed context in dahdi_channels.conf from from-pstn to from-zaptel.

I have just a simple configuration - one ZAP trunk, one inbound route with destination for custom announcement message.

Could you bring any idea please? I have slightly different configuration than first post, updated system, etc. Probably I am missing something obvious for an Asterisk expert.

Some info for the start:
CentOS release 5.6 (Final), asterisk16-1.6.2.19-1_centos5, FreePBX 2.9.0.7, DAHDI Version: 2.4.1.2, rhino-linux-0.99.4rc1

incoming call log

– Starting simple switch on ‘DAHDI/2-1’
– Executing [[email protected]:1] Playback(“DAHDI/2-1”, “vm-goodbye”) in new stack
– <DAHDI/2-1> Playing ‘vm-goodbye.gsm’ (language ‘en’)
– Executing [[email protected]:2] NoOp(“DAHDI/2-1”, “ERROR: FreePBX Does not use the [default] context, confguration error”) in new stack
– Executing [[email protected]:3] Macro(“DAHDI/2-1”, “hangupcall”) in new stack
– Executing [[email protected]:1] GotoIf(“DAHDI/2-1”, “1?theend”) in new stack
– Goto (macro-hangupcall,s,3)
– Executing [[email protected]:3] Hangup(“DAHDI/2-1”, “”) in new stack
== Spawn extension (macro-hangupcall, s, 3) exited non-zero on ‘DAHDI/2-1’ in macro ‘hangupcall’
== Spawn extension (default, s, 3) exited non-zero on ‘DAHDI/2-1’
– Executing [[email protected]:1] Macro(“DAHDI/2-1”, “hangupcall,”) in new stack
– Executing [[email protected]:1] GotoIf(“DAHDI/2-1”, “1?theend”) in new stack
– Goto (macro-hangupcall,s,3)
– Executing [[email protected]:3] Hangup(“DAHDI/2-1”, “”) in new stack
== Spawn extension (macro-hangupcall, s, 3) exited non-zero on ‘DAHDI/2-1’ in macro ‘hangupcall’
== Spawn extension (default, h, 1) exited non-zero on ‘DAHDI/2-1’
– Hungup ‘DAHDI/2-1’

CLI> dahdi show channels

Chan Extension Context Language MOH Interpret Blocked State
pseudo default default In Service
1 default default In Service
2 default default In Service
3 default default In Service
4 default default In Service
5 default default In Service
6 default default In Service
7 default default In Service
8 default default In Service

dahdi_channels.conf

; Autogenerated by /usr/sbin/dahdi_genconf on Fri Aug 5 07:30:41 2011
; 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: Rhino RCB24FXX/1 “Rhino RCB24FXX/1” (MASTER)
;;; line="1 FXO/1/0"
signalling=fxs_ks
callerid=asreceived
group=0
context=from-zaptel
channel => 1
callerid=
group=
context=default

;;; line="2 FXO/1/1"
signalling=fxs_ks
callerid=asreceived
group=0
context=from-zaptel
channel => 2
callerid=
group=
context=default
…etc for * channels

chan_dahdi.conf

;
; DAHDI telephony
;
; Configuration file

[channels]
language=en
context=from-dahdi
;context=default
;context=from-pstn

signalling=fxs_ks
usecallerid=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
group=1
callgroup=1
pickupgroup=1
callerid=asreceived
cidsignalling=bell,dmft
cidstart=ring,
channel => 1-8

#include dahdi-channels.conf

dmesg for rcbfx driver

rcbfx 1: Rhino PCI BAR0 b8a10000 IOMem mapped at f8e60000
rcbfx 1: Waiting for response from card …
rcbfx 1: Firmware Version 2.0
rcbfx 1: firmware rcbfx.fw not available from userspace
rcbfx 1: Hardware version 10
rcbfx 1: G168 DSP App file size = 50928 c6f0
rcbfx 1: G168 DSP Ping DSP Version 106
rcbfx 1: G168 DSP Active and Servicing 8 Channels - ff
rcbfx 1: Starting DMA
rcbfx 1: Spotted a Rhino: Rhino RCB24FXX (4 modules)
dahdi_echocan_mg2: Registered echo canceler 'MG2’
dahdi: Registered tone zone 0 (United States / North America)
rcbfx 1: Released a Rhino
dahdi: Telephony Interface Unloaded
dahdi: Telephony Interface Registered on major 196
dahdi: Version: 2.4.1.2
alloc chan 0 ec 0
alloc chan 1 ec 1
alloc chan 2 ec 2
alloc chan 3 ec 3
alloc chan 4 ec 4
alloc chan 5 ec 5
alloc chan 6 ec 6

alloc chan e ec e
alloc chan f ec f
alloc chan 10 ec 10
alloc chan 11 ec 11
alloc chan 12 ec 12
alloc chan 13 ec 13
alloc chan 14 ec 14
alloc chan 15 ec 15
alloc chan 16 ec 16
alloc chan 17 ec 17

the whole restart sequence of Asterisk and dahdi

[[email protected] ~]# amportal stop

Please wait…

STOPPING ASTERISK
All calls will be dropped once the timer hits 0. To cancel, press CTL-C
Asterisk ended with exit status 0
Asterisk shutdown normally.
Asterisk Stoppedrisk to Stop 120

STOPPING FOP SERVER
FOP Server Stopped
[[email protected] ~]# /etc/init.d/dahdi restart
Unloading DAHDI hardware modules: done
Loading DAHDI hardware modules:
rcbfx: DAHDI Tools Version - 2.4.1

DAHDI Version: 2.4.1.2
Echo Canceller(s):
Configuration

Channel map:

Channel 01: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 01)
Channel 02: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 02)
Channel 03: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 03)
Channel 04: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 04)
Channel 05: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 05)
Channel 06: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 06)
Channel 07: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 07)
Channel 08: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 08)

8 channels to configure.

Changing signalling on channel 1 from Unused to FXS Kewlstart
Setting echocan for channel 1 to mg2
Changing signalling on channel 2 from Unused to FXS Kewlstart
Setting echocan for channel 2 to mg2
Changing signalling on channel 3 from Unused to FXS Kewlstart
Setting echocan for channel 3 to mg2
Changing signalling on channel 4 from Unused to FXS Kewlstart
Setting echocan for channel 4 to mg2
Changing signalling on channel 5 from Unused to FXS Kewlstart
Setting echocan for channel 5 to mg2
Changing signalling on channel 6 from Unused to FXS Kewlstart
Setting echocan for channel 6 to mg2
Changing signalling on channel 7 from Unused to FXS Kewlstart
Setting echocan for channel 7 to mg2
Changing signalling on channel 8 from Unused to FXS Kewlstart
Setting echocan for channel 8 to mg2
[ OK ]

Running dahdi_cfg: [ OK ]
[[email protected] ~]# amportal start

Please wait…

SETTING FILE PERMISSIONS
Permissions OK

STARTING ASTERISK
Asterisk Started

STARTING FOP SERVER
FOP Server Started

I’d be very grateful for any possible assistance.
Best Regards,
Milan

You have a second context declaration that is still in ‘default’

The last context variable in the channel will take precedence, change them all not just one.

Thanks for a reply, but still the same.

I have tried to change it and than comment it out. Now it is like this

;;; line="1 FXO/1/0" signalling=fxs_ks callerid=asreceived group=0 context=from-zaptel channel => 1 ;callerid= ;group= ;context=from-default

Commented out for each channel.

Any other ideas?

First context declaration in chan_dahdi.conf is =from-dahdi and channel => 1-8, than in dahdi_channels.conf is context=from-zaptel for each channel. Everywhere is written that a line

#include dahdi-channels.conf

is important. Why, when it is - probably - already declared in chan_dahdi.conf for all channels.

Well, just a question.

Hi, should I provide some more info, logs, configs, etc?

Don’t need any logs, did you restart DAHDI?

Post output of DAHDI show channels

Yes I did.

here is the output

whistler*CLI> dahdi show channels
Chan Extension Context Language MOH Interpret Blocked State
pseudo default default In Service
1 default default In Service
2 default default In Service
3 default default In Service
4 default default In Service
5 default default In Service
6 default default In Service
7 default default In Service
8 default default In Service

One more note. After complete system reinstall I got it working - just changed context=from-pstn to from-zaptel. Even with overwriting context=default it was working. I had a different error in a log that time, but I think still default context from “dahdi show channels”

Than I installed cdr extension, restart asterisk, got a core dump error, reinstall asterisk and did update of the kernel + recompile rhino driver and - here is where I am. Stacked with a “FreePBX Does not use the [default] context” message.

Would be sweet to get from “dahdi show channels” something like context “from-dahdi” or “from-zaptel” but I have no idea where I am heaving mistake.

I am not sure where you are screwing up but I need to see all of your DAHDI files from /etc/dahdi and /etc/asterisk not just parts of them, post the whole file.

I need /etc/dahdi/system.conf

and any file that show up in the output of this command:

[[email protected] asterisk]# ls | grep -i dahdi
chan_dahdi_additional.conf
chan_dahdi.conf
dahdi-channels.conf
dahdi-channels.conf.bak
[[email protected] asterisk]#

No more ideas? That’s sad. Probably I have to reinstall it completely again and hope its gonna to work. It’s so fragile.

What I have different in comparison with working system is latest FreePBX (not a version from repository) but it probably doesn’t matter as far as it is only web configuration interface.

Maybe could anybody post how should look like conf files for simple environment + short description how to configure (trunk, inbound route) for Rhino card and one analog line in FreePBX? Maybe I am missing something obvious.

Thank you guys.

Why can’t you post the config files like I asked you to?

You don’t need to reinstall.

If you did what I asked you would be up in running in 10 minutes.

I need all the config files for DAHDI as specified in my previous message.

Thank you SkykingOH for a help.
I’ve got it working finally with Asterisk 1.8 from distro repository.

Incoming calls are working now even with context = from-pstn what generated dahdi_genconf command and I can see them in CLI. Such a relieve.

Just for record and for others what I did:
#yum remove asterisk16*
#yum install asterisk18 asterisk18-addons asterisk18-addons-mysql asterisk18-configs asterisk18-dahdi asterisk18-voicemail

#dahdi_genconf

chan_dahdi.conf changed:

;
; DAHDI telephony
;
; Configuration file

[channels]
language=en

;context=from-dahdi
;context=default
context=from-pstn
;context=from-zaptel

signalling=fxs_ks
usecallerid=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes

;group=1
;group=0

callgroup=1
pickupgroup=1

;callerid=asreceived
;cidsignalling=bell,dmft
;cidstart=ring,
;channel => 1-8

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

Still few issues with FreePBX, but they are for another thread.