trixbox - Comments

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KhunIT — Sun, 08 Nov 2009 02:31:48 -0800

In reply to Sangoma A200 2nd Card & New Server / DAHDI Driver setup:

linux cant see the card.

nttranbao — Sun, 08 Nov 2009 01:46:23 -0800

In reply to digium wildcard card 400p not able to configure:

linux cant see the card. This might be because of non-proper plugged or mal-functioned TDM card.

Try lspci to see if linux detects the card presence

Sangoma A200 2nd Card & New Server / DAHDI Driver setup

KhunIT — Sun, 08 Nov 2009 00:57:18 -0800

In reply to Sangoma A200 2nd Card & New Server / DAHDI Driver setup:

I have been using A200 product quite successfully for 6 months now.
We have bought another card and put the two A200 cards in a new server with the latest version of TrixBox and are attempting to use the DAHDI drivers.

The Problem:
• I’m able to dial in from an out outside line but it does not seem to hang up after I’m done with the call.
• Also all OUTBOUND calls reports that all circuits are busy now.

I’m a bit confused because at the command line there seems to be three different options for configuring the WANPIPE and DAHDI
Setup-sangoma
Wancfg
Wancfg_dahdi (This is the last one I tried and I seem to be having some limited success with).

Since all three of these exists I’m wondering if there may a few different versions installed on this box.

Rick

________________________________________

Copies of Configuration & Log files:

[voip.safecoms.com ~]# wanrouter status

Devices currently active:
wanpipe1 wanpipe2

Wanpipe Config:

Device name | Protocol Map | Adapter | IRQ | Slot/IO | If's | CLK | Baud rate |
wanpipe1 | N/A | A200/A400/B600/B700| 50 | 2 | 1 | N/A | 0 |
wanpipe2 | N/A | A200/A400/B600/B700| 74 | 1 | 1 | N/A | 0 |

Wanrouter Status:

[voip.safecoms.com ~]# wanrouter summary

Configuration File Summary in : /etc/wanpipe/

Device Protocol Type Cpu/Io Slot/Irq Bus State
------------------------------------------------------------------------
wanpipe1 WAN_AFT_ANALOG PCI A 2 5 Connected
wanpipe2 WAN_AFT_ANALOG PCI A 1 5 Connected
[voip.safecoms.com ~]# wanrouter list

Devices currently active:
wanpipe1 wanpipe2

[voip.safecoms.com ~]# wanrouter hwprobe

-------------------------------
| Wanpipe Hardware Probe Info |
-------------------------------
1 . AFT-A200-SH : SLOT=2 : BUS=5 : IRQ=14 : CPU=A : PORT=PRI : HWEC=32 : V=11
2 . AFT-A200-SH : SLOT=1 : BUS=5 : IRQ=3 : CPU=A : PORT=PRI : HWEC=32 : V=11

Card Cnt: A200=2

[voip.safecoms.com ~]# wanrouter version

WANPIPE Release: 3.5.4.8

[voip.safecoms.com ~]#

voip*CLI> dahdi show channels
Chan Extension Context Language MOH Interpret Blocked State
pseudo from-zaptel default In Service
1 from-zaptel default In Service
2 from-zaptel default In Service
3 from-zaptel default In Service
4 from-zaptel default In Service
25 from-zaptel default In Service
26 from-zaptel default In Service
27 from-zaptel default In Service
28 from-zaptel default In Service

voip*CLI> dahdi show status
Description Alarms IRQ bpviol CRC4 Fra Codi Options LBO
wrtdm Board 1 OK 0 0 0 CAS Unk YEL 0 db (CSU)/0-133 feet (DSX-1)
wrtdm Board 2 OK 0 0 0 CAS Unk YEL 0 db (CSU)/0-133 feet (DSX-1)

voip*CLI> core show channels
Channel Location State Application(Data)
0 active channels
0 active calls
2 calls processed

-- Unregistered SIP '711'
-- Registered SIP '711' at 10.254.1.209 port 4310
> Saved useragent "X-Lite Beta release 4.0 Beta 2 stamp 55091" for peer 711
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using SIP VRTP TOS bits 136
== Using SIP VRTP CoS mark 6

-- Executing [90855573000@from-internal:1] Macro("SIP/711-b76aa008", "user-callerid,SKIPTTL,") in new stack
-- Executing [s@macro-user-callerid:1] Set("SIP/711-b76aa008", "AMPUSER=711") in new stack
-- Executing [s@macro-user-callerid:2] GotoIf("SIP/711-b76aa008", "0?report") in new stack
-- Executing [s@macro-user-callerid:3] ExecIf("SIP/711-b76aa008", "1?Set(REALCALLERIDNUM=711)") in new stack
-- Executing [s@macro-user-callerid:4] Set("SIP/711-b76aa008", "AMPUSER=711") in new stack
-- Executing [s@macro-user-callerid:5] Set("SIP/711-b76aa008", "AMPUSERCIDNAME=Mobile Rick") in new stack
-- Executing [s@macro-user-callerid:6] GotoIf("SIP/711-b76aa008", "0?report") in new stack
-- Executing [s@macro-user-callerid:7] Set("SIP/711-b76aa008", "AMPUSERCID=711") in new stack
-- Executing [s@macro-user-callerid:8] Set("SIP/711-b76aa008", "CALLERID(all)="Mobile Rick" <711>") in new stack
-- Executing [s@macro-user-callerid:9] Set("SIP/711-b76aa008", "REALCALLERIDNUM=711") in new stack
-- Executing [s@macro-user-callerid:10] ExecIf("SIP/711-b76aa008", "0?Set(CHANNEL(language)=)") in new stack
-- Executing [s@macro-user-callerid:11] GotoIf("SIP/711-b76aa008", "1?continue") in new stack
-- Goto (macro-user-callerid,s,20)
-- Executing [s@macro-user-callerid:20] NoOp("SIP/711-b76aa008", "Using CallerID "Mobile Rick" <711>") in new stack
-- Executing [90855573000@from-internal:2] Set("SIP/711-b76aa008", "_NODEST=") in new stack
-- Executing [90855573000@from-internal:3] Macro("SIP/711-b76aa008", "record-enable,711,OUT,") in new stack
-- Executing [s@macro-record-enable:1] GotoIf("SIP/711-b76aa008", "1?check") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing [s@macro-record-enable:4] AGI("SIP/711-b76aa008", "recordingcheck,20091108-143644,1257665803.2") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck,20091108-143644,1257665803.2: Outbound recording not enabled
-- AGI Script recordingcheck completed, returning 0
-- Executing [s@macro-record-enable:5] MacroExit("SIP/711-b76aa008", "") in new stack
-- Executing [90855573000@from-internal:4] Macro("SIP/711-b76aa008", "dialout-trunk,3,0855573000,,") in new stack
-- Executing [s@macro-dialout-trunk:1] Set("SIP/711-b76aa008", "DIAL_TRUNK=3") in new stack
-- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/711-b76aa008", "0?sub-pincheck,s,1") in new stack
-- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/711-b76aa008", "0?disabletrunk,1") in new stack
-- Executing [s@macro-dialout-trunk:4] Set("SIP/711-b76aa008", "DIAL_NUMBER=0855573000") in new stack
-- Executing [s@macro-dialout-trunk:5] Set("SIP/711-b76aa008", "DIAL_TRUNK_OPTIONS=tr") in new stack
-- Executing [s@macro-dialout-trunk:6] Set("SIP/711-b76aa008", "OUTBOUND_GROUP=OUT_3") in new stack
-- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/711-b76aa008", "1?nomax") in new stack
-- Goto (macro-dialout-trunk,s,9)
-- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/711-b76aa008", "0?skipoutcid") in new stack
-- Executing [s@macro-dialout-trunk:10] Set("SIP/711-b76aa008", "DIAL_TRUNK_OPTIONS=") in new stack
-- Executing [s@macro-dialout-trunk:11] Macro("SIP/711-b76aa008", "outbound-callerid,3") in new stack
-- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/711-b76aa008", "0?Set(CALLERPRES()=)") in new stack
-- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/711-b76aa008", "0?Set(REALCALLERIDNUM=711)") in new stack
-- Executing [s@macro-outbound-callerid:3] GotoIf("SIP/711-b76aa008", "1?normcid") in new stack
-- Goto (macro-outbound-callerid,s,6)
-- Executing [s@macro-outbound-callerid:6] Set("SIP/711-b76aa008", "USEROUTCID=") in new stack
-- Executing [s@macro-outbound-callerid:7] Set("SIP/711-b76aa008", "EMERGENCYCID=") in new stack
-- Executing [s@macro-outbound-callerid:8] Set("SIP/711-b76aa008", "TRUNKOUTCID=2") in new stack
-- Executing [s@macro-outbound-callerid:9] GotoIf("SIP/711-b76aa008", "1?trunkcid") in new stack
-- Goto (macro-outbound-callerid,s,12)
-- Executing [s@macro-outbound-callerid:12] ExecIf("SIP/711-b76aa008", "0?Set(CALLERID(all)=2)") in new stack
-- Executing [s@macro-outbound-callerid:13] ExecIf("SIP/711-b76aa008", "0?Set(CALLERID(all)=)") in new stack
-- Executing [s@macro-outbound-callerid:14] ExecIf("SIP/711-b76aa008", "0?Set(CALLERPRES()=prohib_passed_screen)") in new stack
-- Executing [s@macro-dialout-trunk:12] ExecIf("SIP/711-b76aa008", "0?AGI(fixlocalprefix)") in new stack
-- Executing [s@macro-dialout-trunk:13] Set("SIP/711-b76aa008", "OUTNUM=0855573000") in new stack
-- Executing [s@macro-dialout-trunk:14] Set("SIP/711-b76aa008", "custom=DAHDI/2") in new stack
-- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/711-b76aa008", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^))") in new stack
-- Executing [s@macro-dialout-trunk:16] Macro("SIP/711-b76aa008", "dialout-trunk-predial-hook,") in new stack
-- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/711-b76aa008", "") in new stack
-- Executing [s@macro-dialout-trunk:17] GotoIf("SIP/711-b76aa008", "0?bypass,1") in new stack
-- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/711-b76aa008", "0?customtrunk") in new stack
-- Executing [s@macro-dialout-trunk:19] Dial("SIP/711-b76aa008", "DAHDI/2/0855573000,300,") in new stack
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [s@macro-dialout-trunk:20] Goto("SIP/711-b76aa008", "s-CHANUNAVAIL,1") in new stack
-- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
-- Executing [s-CHANUNAVAIL@macro-dialout-trunk:1] GotoIf("SIP/711-b76aa008", "1?noreport") in new stack
-- Goto (macro-dialout-trunk,s-CHANUNAVAIL,3)
-- Executing [s-CHANUNAVAIL@macro-dialout-trunk:3] NoOp("SIP/711-b76aa008", "TRUNK Dial failed due to CHANUNAVAIL (hangupcause: 0) - failing through to other trunks") in new stack
-- Executing [90855573000@from-internal:5] Macro("SIP/711-b76aa008", "dialout-trunk,2,0855573000,,") in new stack
-- Executing [s@macro-dialout-trunk:1] Set("SIP/711-b76aa008", "DIAL_TRUNK=2") in new stack
-- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/711-b76aa008", "0?sub-pincheck,s,1") in new stack
-- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/711-b76aa008", "0?disabletrunk,1") in new stack
-- Executing [s@macro-dialout-trunk:4] Set("SIP/711-b76aa008", "DIAL_NUMBER=0855573000") in new stack
-- Executing [s@macro-dialout-trunk:5] Set("SIP/711-b76aa008", "DIAL_TRUNK_OPTIONS=tr") in new stack
-- Executing [s@macro-dialout-trunk:6] Set("SIP/711-b76aa008", "OUTBOUND_GROUP=OUT_2") in new stack
-- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/711-b76aa008", "1?nomax") in new stack
-- Goto (macro-dialout-trunk,s,9)
-- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/711-b76aa008", "0?skipoutcid") in new stack
-- Executing [s@macro-dialout-trunk:10] Set("SIP/711-b76aa008", "DIAL_TRUNK_OPTIONS=") in new stack
-- Executing [s@macro-dialout-trunk:11] Macro("SIP/711-b76aa008", "outbound-callerid,2") in new stack
-- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/711-b76aa008", "0?Set(CALLERPRES()=)") in new stack
-- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/711-b76aa008", "0?Set(REALCALLERIDNUM=711)") in new stack
-- Executing [s@macro-outbound-callerid:3] GotoIf("SIP/711-b76aa008", "1?normcid") in new stack
-- Goto (macro-outbound-callerid,s,6)
-- Executing [s@macro-outbound-callerid:6] Set("SIP/711-b76aa008", "USEROUTCID=") in new stack
-- Executing [s@macro-outbound-callerid:7] Set("SIP/711-b76aa008", "EMERGENCYCID=") in new stack
-- Executing [s@macro-outbound-callerid:8] Set("SIP/711-b76aa008", "TRUNKOUTCID=1") in new stack
-- Executing [s@macro-outbound-callerid:9] GotoIf("SIP/711-b76aa008", "1?trunkcid") in new stack
-- Goto (macro-outbound-callerid,s,12)
-- Executing [s@macro-outbound-callerid:12] ExecIf("SIP/711-b76aa008", "0?Set(CALLERID(all)=1)") in new stack
-- Executing [s@macro-outbound-callerid:13] ExecIf("SIP/711-b76aa008", "0?Set(CALLERID(all)=)") in new stack
-- Executing [s@macro-outbound-callerid:14] ExecIf("SIP/711-b76aa008", "0?Set(CALLERPRES()=prohib_passed_screen)") in new stack
-- Executing [s@macro-dialout-trunk:12] ExecIf("SIP/711-b76aa008", "0?AGI(fixlocalprefix)") in new stack
-- Executing [s@macro-dialout-trunk:13] Set("SIP/711-b76aa008", "OUTNUM=0855573000") in new stack
-- Executing [s@macro-dialout-trunk:14] Set("SIP/711-b76aa008", "custom=DAHDI/1") in new stack
-- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/711-b76aa008", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^))") in new stack
-- Executing [s@macro-dialout-trunk:16] Macro("SIP/711-b76aa008", "dialout-trunk-predial-hook,") in new stack
-- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/711-b76aa008", "") in new stack
-- Executing [s@macro-dialout-trunk:17] GotoIf("SIP/711-b76aa008", "0?bypass,1") in new stack
-- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/711-b76aa008", "0?customtrunk") in new stack
-- Executing [s@macro-dialout-trunk:19] Dial("SIP/711-b76aa008", "DAHDI/1/0855573000,300,") in new stack
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [s@macro-dialout-trunk:20] Goto("SIP/711-b76aa008", "s-CHANUNAVAIL,1") in new stack
-- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
-- Executing [s-CHANUNAVAIL@macro-dialout-trunk:1] GotoIf("SIP/711-b76aa008", "1?noreport") in new stack
-- Goto (macro-dialout-trunk,s-CHANUNAVAIL,3)
-- Executing [s-CHANUNAVAIL@macro-dialout-trunk:3] NoOp("SIP/711-b76aa008", "TRUNK Dial failed due to CHANUNAVAIL (hangupcause: 0) - failing through to other trunks") in new stack
-- Executing [90855573000@from-internal:6] Macro("SIP/711-b76aa008", "dialout-trunk,4,0855573000,,") in new stack
-- Executing [s@macro-dialout-trunk:1] Set("SIP/711-b76aa008", "DIAL_TRUNK=4") in new stack
-- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/711-b76aa008", "0?sub-pincheck,s,1") in new stack
-- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/711-b76aa008", "0?disabletrunk,1") in new stack
-- Executing [s@macro-dialout-trunk:4] Set("SIP/711-b76aa008", "DIAL_NUMBER=0855573000") in new stack
-- Executing [s@macro-dialout-trunk:5] Set("SIP/711-b76aa008", "DIAL_TRUNK_OPTIONS=tr") in new stack
-- Executing [s@macro-dialout-trunk:6] Set("SIP/711-b76aa008", "OUTBOUND_GROUP=OUT_4") in new stack
-- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/711-b76aa008", "1?nomax") in new stack
-- Goto (macro-dialout-trunk,s,9)
-- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/711-b76aa008", "0?skipoutcid") in new stack
-- Executing [s@macro-dialout-trunk:10] Set("SIP/711-b76aa008", "DIAL_TRUNK_OPTIONS=") in new stack
-- Executing [s@macro-dialout-trunk:11] Macro("SIP/711-b76aa008", "outbound-callerid,4") in new stack
-- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/711-b76aa008", "0?Set(CALLERPRES()=)") in new stack
-- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/711-b76aa008", "0?Set(REALCALLERIDNUM=711)") in new stack
-- Executing [s@macro-outbound-callerid:3] GotoIf("SIP/711-b76aa008", "1?normcid") in new stack
-- Goto (macro-outbound-callerid,s,6)
-- Executing [s@macro-outbound-callerid:6] Set("SIP/711-b76aa008", "USEROUTCID=") in new stack
-- Executing [s@macro-outbound-callerid:7] Set("SIP/711-b76aa008", "EMERGENCYCID=") in new stack
-- Executing [s@macro-outbound-callerid:8] Set("SIP/711-b76aa008", "TRUNKOUTCID=3") in new stack
-- Executing [s@macro-outbound-callerid:9] GotoIf("SIP/711-b76aa008", "1?trunkcid") in new stack
-- Goto (macro-outbound-callerid,s,12)
-- Executing [s@macro-outbound-callerid:12] ExecIf("SIP/711-b76aa008", "0?Set(CALLERID(all)=3)") in new stack
-- Executing [s@macro-outbound-callerid:13] ExecIf("SIP/711-b76aa008", "0?Set(CALLERID(all)=)") in new stack
-- Executing [s@macro-outbound-callerid:14] ExecIf("SIP/711-b76aa008", "0?Set(CALLERPRES()=prohib_passed_screen)") in new stack
-- Executing [s@macro-dialout-trunk:12] ExecIf("SIP/711-b76aa008", "0?AGI(fixlocalprefix)") in new stack
-- Executing [s@macro-dialout-trunk:13] Set("SIP/711-b76aa008", "OUTNUM=0855573000") in new stack
-- Executing [s@macro-dialout-trunk:14] Set("SIP/711-b76aa008", "custom=DAHDI/3") in new stack
-- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/711-b76aa008", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^))") in new stack
-- Executing [s@macro-dialout-trunk:16] Macro("SIP/711-b76aa008", "dialout-trunk-predial-hook,") in new stack
-- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/711-b76aa008", "") in new stack
-- Executing [s@macro-dialout-trunk:17] GotoIf("SIP/711-b76aa008", "0?bypass,1") in new stack
-- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/711-b76aa008", "0?customtrunk") in new stack
-- Executing [s@macro-dialout-trunk:19] Dial("SIP/711-b76aa008", "DAHDI/3/0855573000,300,") in new stack
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [s@macro-dialout-trunk:20] Goto("SIP/711-b76aa008", "s-CHANUNAVAIL,1") in new stack
-- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
-- Executing [s-CHANUNAVAIL@macro-dialout-trunk:1] GotoIf("SIP/711-b76aa008", "1?noreport") in new stack
-- Goto (macro-dialout-trunk,s-CHANUNAVAIL,3)
-- Executing [s-CHANUNAVAIL@macro-dialout-trunk:3] NoOp("SIP/711-b76aa008", "TRUNK Dial failed due to CHANUNAVAIL (hangupcause: 0) - failing through to other trunks") in new stack
-- Executing [90855573000@from-internal:7] Macro("SIP/711-b76aa008", "outisbusy,") in new stack
-- Executing [s@macro-outisbusy:1] Playback("SIP/711-b76aa008", "all-circuits-busy-now,noanswer") in new stack
-- Playing 'all-circuits-busy-now.ulaw' (language 'en')
-- Executing [s@macro-outisbusy:2] Playback("SIP/711-b76aa008", "pls-try-call-later,noanswer") in new stack
-- Playing 'pls-try-call-later.ulaw' (language 'en')
-- Executing [s@macro-outisbusy:3] Macro("SIP/711-b76aa008", "hangupcall") in new stack
-- Executing [s@macro-hangupcall:1] ResetCDR("SIP/711-b76aa008", "vw") in new stack
-- Executing [s@macro-hangupcall:2] NoCDR("SIP/711-b76aa008", "") in new stack
-- Executing [s@macro-hangupcall:3] GotoIf("SIP/711-b76aa008", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,6)
-- Executing [s@macro-hangupcall:6] GotoIf("SIP/711-b76aa008", "1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s@macro-hangupcall:9] GotoIf("SIP/711-b76aa008", "1?theend") in new stack
-- Goto (macro-hangupcall,s,11)
-- Executing [s@macro-hangupcall:11] Hangup("SIP/711-b76aa008", "") in new stack
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/711-b76aa008' in macro 'hangupcall'
== Spawn extension (macro-outisbusy, s, 3) exited non-zero on 'SIP/711-b76aa008' in macro 'outisbusy'
== Spawn extension (from-internal, 90855573000, 7) exited non-zero on 'SIP/711-b76aa008'
-- Executing [h@from-internal:1] Macro("SIP/711-b76aa008", "hangupcall") in new stack
-- Executing [s@macro-hangupcall:1] ResetCDR("SIP/711-b76aa008", "vw") in new stack
-- Executing [s@macro-hangupcall:2] NoCDR("SIP/711-b76aa008", "") in new stack
-- Executing [s@macro-hangupcall:3] GotoIf("SIP/711-b76aa008", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,6)
-- Executing [s@macro-hangupcall:6] GotoIf("SIP/711-b76aa008", "1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s@macro-hangupcall:9] GotoIf("SIP/711-b76aa008", "1?theend") in new stack
-- Goto (macro-hangupcall,s,11)
-- Executing [s@macro-hangupcall:11] Hangup("SIP/711-b76aa008", "") in new stack
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/711-b76aa008' in macro 'hangupcall'
== Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/711-b76aa008'
voip*CLI>

/etc/wanpipe/

#================================================
# WANPIPE1 Configuration File
#================================================
#
# Date: Mon Jul 31 17:10:23 EDT 2006
#
# Note: This file was generated automatically
# by /usr/local/sbin/setup-sangoma program.
#
# If you want to edit this file, it is
# recommended that you use wancfg program
# to do so.
#================================================
# Sangoma Technologies Inc.
#================================================

[devices]
wanpipe1 = WAN_AFT_ANALOG, Comment

[interfaces]
w1g1 = wanpipe1, , TDM_VOICE, Comment

[wanpipe1]
CARD_TYPE = AFT
S514CPU = A
CommPort = PRI
AUTO_PCISLOT = NO
PCISLOT = 2
PCIBUS = 5
FE_MEDIA = FXO/FXS
TDMV_LAW = MULAW
TDMV_OPERMODE = FCC
RM_BATTTHRESH = 3
RM_BATTDEBOUNCE = 16
FE_NETWORK_SYNC = NO
MTU = 1500
UDPPORT = 9000
TTL = 255
IGNORE_FRONT_END = NO
TDMV_SPAN = 1
TDMV_HW_DTMF = NO
TDMV_HW_FAX_DETECT = NO

[w1g1]
ACTIVE_CH = ALL
MTU = 8
TDMV_HWEC = YES

[devices]
wanpipe2 = WAN_AFT_ANALOG, Comment

[interfaces]
w2g1 = wanpipe2, , TDM_VOICE, Comment

[wanpipe2]
CARD_TYPE = AFT
S514CPU = A
CommPort = PRI
AUTO_PCISLOT = NO
PCISLOT = 1
PCIBUS = 5
FE_MEDIA = FXO/FXS
TDMV_LAW = MULAW
TDMV_OPERMODE = FCC
RM_BATTTHRESH = 3
RM_BATTDEBOUNCE = 16
FE_NETWORK_SYNC = NO
MTU = 1500
UDPPORT = 9000
TTL = 255
IGNORE_FRONT_END = NO
TDMV_SPAN = 2
TDMV_HW_DTMF = NO
TDMV_HW_FAX_DETECT = NO

[w2g1]
ACTIVE_CH = ALL
MTU = 8
TDMV_HWEC = YES

#-------------------------
#WANROUTER.RC
#-------------------------
#!/bin/sh
# router.rc WAN router meta-configuration file.
#
# This file defines variables used by the router shell scripts
# and should be located in /etc/wanpipe directory. These are:
#
# ROUTER_BOOT = Boot flag (YES/NO).
# WAN_CONF_DIR = Where to put wanpipe config files.
# WAN_INTR_DIR = Where to put wanpipe interface files.
# WAN_LOG = Where to put start-up log file.
# WAN_LOCK = File used as a lock.
# WAN_LOCK_DIR =
# WAN_IP_FORWARD = Enable IP Forwarding on startup.
# WAN_DEVICES="wanpipe1 wanpipe2 "
# loaded on 'wanrouter start'
# (ex: "wanpipe1 wanpipe2 wanpipe3...")
#
# Note: Name of wanpipe devices correspond
# to the configuration files in
# WANPIPE_CONF_DIR directory:
# (ex. /etc/wanpipe/wanpipe1.conf )
#
# Note: This file is 'executed' by the shell script, so
# the usual shell syntax must be observed.
ROUTER_BOOT=YES
WAN_CONF_DIR=/etc/wanpipe
WAN_INTR_DIR=/etc/wanpipe/interfaces
WAN_BIN_DIR=/usr/sbin
WAN_LOG=/var/log/wanrouter
WAN_LOCK=/var/lock/subsys/wanrouter
WAN_LOCK_DIR=/var/lock/subsys
WAN_IP_FORWARD=NO
NEW_IF_TYPE=NO
WAN_LIB_DIR=/etc/wanpipe/lib
WAN_ADSL_LIST=/etc/wanpipe/wan_adsl.list
WAN_ANNEXG_LOAD=NO
WAN_SCTP_LOAD=NO
WAN_LIP_LOAD=NO
WAN_DYN_WANCONFIG=NO
WAN_SCRIPTS_DIR=/etc/wanpipe/scripts
WAN_FIRMWARE_DIR=/etc/wanpipe/firmware
WAN_DEVICES_REV_STOP_ORDER=YES
WAN_DEVICES="wanpipe1 wanpipe2 "
#-------------------------
#END WANROUTER.RC
#-------------------------

ZAPATA-AUTO.CONF
; Autogenerated by /usr/local/sbin/sangoma/setup-sangoma -- do not hand edit
; Zaptel Channels Configurations (zapata.conf)
;
; This is not intended to be a complete zapata.conf. Rather, it is intended
; to be #include-d by /etc/zapata.conf that will include the global settings
;
callerid=asreceived

;Sangoma AFT-A200 [slot:2 bus:5 span:1]
context=from-zaptel
group=0
echocancel=yes
signalling = fxs_ks
channel => 1

context=from-zaptel
group=0
echocancel=yes
signalling = fxs_ks
channel => 2

context=from-zaptel
group=0
echocancel=yes
signalling = fxs_ks
channel => 3

context=from-zaptel
group=0
echocancel=yes
signalling = fxs_ks
channel => 4

;Sangoma AFT-A200 [slot:1 bus:5 span:1]
context=from-zaptel
group=0
echocancel=yes
signalling = fxs_ks
channel => 25

context=from-zaptel
group=0
echocancel=yes
signalling = fxs_ks
channel => 26

context=from-zaptel
group=0
echocancel=yes
signalling = fxs_ks
channel => 27

context=from-zaptel
group=0
echocancel=yes
signalling = fxs_ks
channel => 28

ZAPTEL.CONF
#autogenerated by /usr/sbin/wancfg_dahdi do not hand edit
#autogenrated on 2009-10-29
#Dahdi Channels Configurations
#For detailed Dahdi options, view /etc/dahdi/system.conf.bak
loadzone=us
defaultzone=us

#Sangoma AFT-A200 [slot:2 bus:5 span:1]
fxsks=1
echocanceller=mg2,1
fxsks=2
echocanceller=mg2,2
fxsks=3
echocanceller=mg2,3
fxsks=4
echocanceller=mg2,4

#Sangoma AFT-A200 [slot:1 bus:5 span:1]
fxsks=25
echocanceller=mg2,25
fxsks=26
echocanceller=mg2,26
fxsks=27
echocanceller=mg2,27
fxsks=28
echocanceller=mg2,28

/etc/dahdi/

---------------------------
#
# /etc/dahdi/genconf_parameters
#
# This file contains parameters that affect the
# dahdi_genconf configuration generator.
#
# Syntax:
# * A comment from '#' to end of line
# * Blank lines ignored
# * Whitespace at end of line trimmed
# * Single valued items:
# key value
# * List valued items:
# key
# value1
# value2
# ...
#

# When generating extensions for chan_dahdi.conf or users.conf etc: the
# extension number will be channel_number+base_exten . The default is:
#base_exten 4000
#
# Make FXS (analog phones) extensions answer immediately (sets
# 'immediate = yes' for them in chan_dahdi.conf). Don't enable this before
# you're read documentation about this option.
#fxs_immediate yes
#
# For FXS (analog phones) - use KS or LS? ks is the only method for
# Asterisk to provide disconnect supervision and thus it would normally
# be preferred and is the default.
#fxs_default_start ls
#
# For FXO (analog lines) - use KS or LS? KS is the default and is
# normally the better choice as it allows detecting hang-ups on many
# lines.
#fxo_default_start ls

# Set tone zone values. This is used for playing tones (busy, dial-tone
# and such). The default is 'us'. This sets the value for both loadzone
# and defaultzone in system.conf .
#lc_country il

# The dialplan context into which to send trunks in chan_dahdi.conf or
# users.conf. The default value is:
#context_lines from-pstn
#
# The dialplan context into which to send extensions in chan_dahdi.conf or
# users.conf. The default value is:
#context_phones from-internal
#
# Two extra contexts for the input ports and output ports of an
# Astribank. Default values are:
#context_input astbank-input
#context_output astbank-output

# A group to put all analog phones in. By default 0, so you can dial to
# the 'first phone available' using Dahdi/g5 .
#group_phones 5
#
# A group in which to put all the channels belonging to some trunk.
# Thus you can dial through "some trunk" using Dahdi/G0/NUMBER
#group_lines 0

# Channels of digital trunk of span N are also added to group 10+N (that
# is: 14 for channels of span 4).

# Do we want to use PtP ('bri') or PtMP ('bri_ptmp') for BRI? PtMP
# allows connecting several CPE devices on the same network device
# (several BRI phones on the same line, kind of like several analog
# phones on the same analog line). However it is generally brings
# unnecessary complexity for a pbx-pbx connection. It is still the
# default as this is normally what you get for a BRI PSTN connection.
#bri_sig_style bri
#
# If this option is set (that is: not remmed-out), BRI NT ports will
# also be set as overlap. This is useful if you want to connect ISDN
# phones.
#brint_overlap

# The echo canceler to use. If you have a hardware echo canceler, just
# leave it be, as this one won't be used anyway.
#
# The default is mg2, but it may change in the future. E.g: a packager
# that bundles a better echo canceler may set it as the default, or
# dahdi_genconf will scan for the "best" echo canceler.
#
#echo_can hpec
#echo_can oslec
#echo_can none # to avoid echo canceler altogether

# bri_hardhdlc:
# 'yes' - forces BRI cards to use 'hardhdlc' signalling.
# 'no' - forces BRI cards to use 'dchan' (an alias for 'fcshdlc').
# It is usefull only for dahdi with the bristuff patch.
#
# If it is left out or set to 'auto':
# * Information supplied by the driver is used to decide:
# - Currently implemented for Astribanks.
# - Taken from /sys/bus/xpds/drivers/bri/dchan_hardhdlc.
# * Without this info, falls back to 'hardhdlc'.
#bri_hardhdlc auto

# For MFC/R2 Support: 'R2' will make E1 spans CAS and with the
# 'r2_idle_bits' bit in system.conf . It will also make dahdi_genconf default
# to generating the channels of this card in unicall.conf rather than in
# chan_dahdi.conf . The meaning of this may be extended somehow to support
# R2 through openr2/chan_dahdi later on.
#pri_connection_type R2
#r2_idle_bits 1101

# pri_termtype contains a list of settings:
# Currently the only setting is for TE or NT (the default is TE). This
# sets two different but normally related configuration items:
#
# A TE span will have *_cpe signalling in Asterisk and will also get
# timing from the remote party.
#
# A NT span will have *_new signalling in Asterisk and will provide
# timing to the remote party.
#
# pri_termtype is a list if span specs and configuration (TE/NT) for
# them. The first spec that matches is used. The matching is of perl
# regular expressions, but with '*' and '?' have their meaning from
# basic regular expressions.
#pri_termtype
# SPAN/2 NT
# SPAN/4 NT
#
#pri_termtype
# SPAN/* NT
#

---------------------------
#INIT.CONF
# Shell settings for Dahdi initialization scripts.
# This replaces the old/per-platform files (/etc/sysconfig/zaptel,
# /etc/defaults/zaptel)
#

# The maximal timeout (seconds) to wait for udevd to finish generating
# device nodes after the modules have loaded and before running dahdi_cfg.
#DAHDI_DEV_TIMEOUT=40

# Override settings for xpp_fxloader
#XPP_FIRMWARE_DIR=/usr/share/dahdi
#XPP_HOTPLUG_DISABLED=yes

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

#MODULES
# Contains the list of modules to be loaded / unloaded by /etc/init.d/dahdi.
#
# NOTE: Please add/edit /etc/modprobe.d/dahdi or /etc/modprobe.conf if you
# would like to add any module parameters.
#
# Format of this file: list of modules, each in its own line.
# Anything after a '#' is ignore, likewise trailing and leading
# whitespaces and empty lines.

# Digium TE205P/TE207P/TE210P/TE212P: PCI dual-port T1/E1/J1
# Digium TE405P/TE407P/TE410P/TE412P: PCI quad-port T1/E1/J1
# Digium TE220: PCI-Express dual-port T1/E1/J1
# Digium TE420: PCI-Express quad-port T1/E1/J1
wct4xxp

# Digium TE120P: PCI single-port T1/E1/J1
# Digium TE121: PCI-Express single-port T1/E1/J1
# Digium TE122: PCI single-port T1/E1/J1
wcte12xp

# Digium T100P: PCI single-port T1
# Digium E100P: PCI single-port E1
wct1xxp

# Digium TE110P: PCI single-port T1/E1/J1
wcte11xp

# Digium TDM2400P/AEX2400: up to 24 analog ports
# Digium TDM800P/AEX800: up to 8 analog ports
# Digium TDM410P/AEX410: up to 4 analog ports
wctdm24xxp

# OpenVOx
opvxa1200

# X100P - Single port FXO interface
# X101P - Single port FXO interface
wcfxo

# Digium TDM400P: up to 4 analog ports
wctdm

# Digium B410P: 4 NT/TE BRI ports
wcb4xxp

# Digium TC400B: G729 / G723 Transcoding Engine
wctc4xxp

# Xorcom Astribank Devices
xpp_usb

--------------------------------
SYSTEM.CONF

#autogenerated by /usr/sbin/wancfg_dahdi do not hand edit
#autogenrated on 2009-11-08
#Dahdi Channels Configurations
#For detailed Dahdi options, view /etc/dahdi/system.conf.bak
loadzone=us
defaultzone=us

#Sangoma AFT-A200 [slot:2 bus:5 span:1]
fxsks=1
echocanceller=mg2,1
fxsks=2
echocanceller=mg2,2
fxsks=3
echocanceller=mg2,3
fxsks=4
echocanceller=mg2,4
#Sangoma AFT-A200 [slot:1 bus:5 span:2]
fxsks=25
echocanceller=mg2,25
fxsks=26
echocanceller=mg2,26
fxsks=27
echocanceller=mg2,27
fxsks=28
echocanceller=mg2,28

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

/etc/asterisk/chan_dahdi.conf

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

;autogenerated by /usr/sbin/wancfg_dahdi do not hand edit
;autogenrated on 2009-11-08
;Dahdi Channels Configurations
;For detailed Dahdi options, view /etc/asterisk/chan_dahdi.conf.bak

[trunkgroups]

[channels]
context=default
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
relaxdtmf=yes
rxgain=0.0
txgain=0.0
group=1
callgroup=1
pickupgroup=1
immediate=no

;Sangoma AFT-A200 [slot:2 bus:5 span:1]
context=from-zaptel
group=0
echocancel=yes
signalling = fxs_ks
channel => 1

context=from-zaptel
group=0
echocancel=yes
signalling = fxs_ks
channel => 2

context=from-zaptel
group=0
echocancel=yes
signalling = fxs_ks
channel => 3

context=from-zaptel
group=0
echocancel=yes
signalling = fxs_ks
channel => 4

;Sangoma AFT-A200 [slot:1 bus:5 span:2]
context=from-zaptel
group=0
echocancel=yes
signalling = fxs_ks
channel => 25

context=from-zaptel
group=0
echocancel=yes
signalling = fxs_ks
channel => 26

context=from-zaptel
group=0
echocancel=yes
signalling = fxs_ks
channel => 27

context=from-zaptel
group=0
echocancel=yes
signalling = fxs_ks
channel => 28

Thanks!

joshuamarius — Sat, 07 Nov 2009 22:44:52 -0800

In reply to Solution to: trixbox install hangs on Loading ata_piix driver:

Had the same problem on a Dell Vostro 200.

Thanks again for taking the time to post this solution.

----------------
Joshua Marius
www.letheonline.net

Trix 2.8.0.x - I will pass til it gets better

jdwebcc — Sat, 07 Nov 2009 21:14:42 -0800

In reply to TB 2.8:

I have been using Trix now for about 4 years.. back to the Asterisk@Home -- and it works very well. I have never had a problem I couldn't deal with.. never a problem that wasn't just typical.

I downloaded the 2.8 and this wasn't so typical - A vanilla install -- it seemed to work as I am used to - but just as others in this thread posted issues I too had the same problems and more. One so bad that one day - while doing a demo it just all the sudden wouldn't let my phones dial internally. BLF mis-reported phones in use.

I decided at that point it just isn't ready.

Reinstalled 2.6.3 and let it at that for now.

I am sure the powers that be will get it past these issues and it will become the stable product we are used to.

Thats my two cents.

No CDR report for any call on New TB 2.8.0.2

thomashang — Sat, 07 Nov 2009 21:12:37 -0800

In reply to TB 2.8:

Install clean trixbox 2.8.0.2 (10/21/09),
setup only 2 extension and try to call each other,
go to check the CDR report,( even on Report of FREEPBX)
there is not any record.

Is it a bug ?

Try to install 2 times, not help.
For TB 2.8.0.1 was working ok. but not on TB 2.8.0.2
Need solve the problem.......

Thomas

More than 4 GIG

jdwebcc — Sat, 07 Nov 2009 21:04:07 -0800

In reply to How do i get Trixbox 2.6.2.3 to recognize more than 4GB of ram:

That article clearly states some Processors and appropriate OS. This is not definitive at all. Plus for regular users it still stands.. you need to goto 64bit CentOS to easily benefit.

There is also no telling the impact or how well the whole Trixbox LAMP installation would run... ?

I would agree ALL things possible at some point - but here for most users especially new users to nix - it is far easier to just goto 64bit CentOS.. wouldn't you agree ?

When dealing with questions here on the forums I always try to offer simple easy to follow directions or solutions.

You should not have the web

SkykingOH — Sat, 07 Nov 2009 20:56:14 -0800

In reply to Can't reach trixbox http properly with new static IP address?:

You should not have the web server exposed to the Internet on any port. If you have to expose SIP you must employ strict security.

It's not a matter of if the system will be hacked it's a matter of when.

There are several great articled on securing trixbox. The best is by engineer Tim.

Asterisk does. You will

SkykingOH — Sat, 07 Nov 2009 20:51:17 -0800

In reply to Does trixbox support session timer?:

Asterisk does. You will need a short custom extension to accomplish this.

Here are the directions

jdwebcc — Sat, 07 Nov 2009 20:42:07 -0800

In reply to Sharing Voicemail with 2 extensions:

https://support.rhinoequipment.com/index.php?_m=knowledgebase&_a=...

Using FXO / USB ?

jdwebcc — Sat, 07 Nov 2009 20:37:02 -0800

In reply to Majic Jack Setup:

How do you plan on setting it up ? More specifically how do you plan to use it ?

There is a chance

antidelldude — Sat, 07 Nov 2009 20:20:51 -0800

In reply to trixbox on an EEE BOX b202:

yum install kernel-headers kernel-devel gcc ncurses ncurses-devel

Here is the kernel source (Maybe, I think the trixbox team forks the kernel that is installed with trixbox): ftp://ftp.redhat.com/pub/redhat/linux/enterprise/5Server/en/os/SR...

Here is the patch you will need http://cooldavid.org/download/jme.netdev-2.6.20080916.patch
You will have to edit the patch above to get it to patch correctly. You will have to compile a module for the current kernel. It is a headache. I was bored and spent a few hours trying, but I have never successfully rebuilt a centos kernel from their source rpm. The kernel doesn't have to be fully rebuilt, the build directory just needs to have everything to build the module against your kernel. If someone could tell me how to get a working build directory for the current kernel, i'll try again.

Off to the

mkleine — Sat, 07 Nov 2009 19:18:41 -0800

In reply to setup-sangoma on 2.8?:

Off to the wiki...

http://wiki.sangoma.com/wanpipe-asterisk-trixbox-rpms#fstinall

shows the steps required.

It works as advertised. However, I've got some oddities (issues where the send and receive packets seem to 'crash' with simultaneous sends on the D-channel) to my TSP. The guilty party seems to be librpri-1.4.10-1 and libpri-1.4.10-2 is supposed to fix it, but the RPM isn't yet available from the trixbox repository.

no luck

ahmershuja — Sat, 07 Nov 2009 17:10:54 -0800

In reply to callback + disa not working with ce 2.8 (newbie):

i tried what u mentioned, but still not working,
any other solution you know,

ahmer

why you need softphone...

ahmershuja — Sat, 07 Nov 2009 16:59:02 -0800

In reply to xlite softphone for nokia e90:

You can use callback with disa, I am using CE2.6.x wonderful, but some problem with CE2.8.x callback doesn't work, coz. of dahdi, I am still trying to find how to setup with CE2.8 dahdi :(((

goodluck...
ahmer

Thanks for the reply. The

uncanny — Sat, 07 Nov 2009 15:48:53 -0800

In reply to Auto Config for Llinksys spa942?:

Thanks for the reply. The command "tail -v /var/log/atftp" brings up an error "tail: cannot open `/var/log/atftp' for reading: No such file or directory"
Should I be running that from somewhere specific? I am logged into the box at that point as root.
Is there a way to check if the Option66 is coming through to the phone? I have been through the phone menu a few times now and cannot seem to find any entry that is either proxy or tftp - shouldn't this be populated somewhere on the phone?
Checking the web admin panel for the phone doesn't seem to show this data either.

hum!

rrichiez — Sat, 07 Nov 2009 14:36:20 -0800

In reply to Echo problems using Nettalk please help:

Well I know for a fact that i have olsec echo cancellation which comes in Trixbox 2.6 but don't know how to tweak it. Talking to the support at nettalk is like talking to magic jacks wich i dont think i will get anywhere.

thanks

rrichiez

Does your card have echo

b14ck — Sat, 07 Nov 2009 14:18:16 -0800

In reply to Echo problems using Nettalk please help:

Does your card have echo cancellation? Are you using it? Have you called your nettalk for troubleshooting help?

Pfsense, IPcop, Smoothwall

datu503 — Sat, 07 Nov 2009 13:55:29 -0800

In reply to small business firewall for Trixbox? :

That's my choice, in that order. I hear many of our buyers like Smoothwall, very friendly, nice GUI.
But pfsense still tops!

I only installed Trixbox CE

raymonvdm — Sat, 07 Nov 2009 12:58:26 -0800

In reply to /etc/init.d/asterisk restart kill`s Trixbox:

I only installed Trixbox CE on my box how did i get pbxtra on it then ?

Me = Confused

Thank you so much for all

ningSF — Sat, 07 Nov 2009 09:31:31 -0800

In reply to How to get real IP address of each extension number in trixbox?:

Thank you so much for all reply :)

jdwebcc,
I can see all ip addresses of each extension from Tools Tab > Asterisk Info > Peers that you tell me :) thanks again. But I have a question, how to get ip address from web page by programming? because I want to get ip address by my program for bring it to use in some function within my program. Could you please tell me about that?

thanks :)

Write a simple custom

nttranbao — Sat, 07 Nov 2009 07:23:45 -0800

In reply to Callback from voicemail:

Write a simple custom context, which automatically adds desired prefix (i.e. 9), and dial desired trunks. Then assign that custom context into vm options.

No raid on mine.

hkgonra — Sat, 07 Nov 2009 06:56:44 -0800

In reply to Dell has discontinued the SC440, anyone used the T100 yet?:

No raid on mine.

Did you do a clean install

andrew — Sat, 07 Nov 2009 06:54:15 -0800

In reply to TB 2.8:

Did you do a clean install of trixbox 2.8.0.2.?

This is starting to turn

andrew — Sat, 07 Nov 2009 06:51:47 -0800

In reply to What's to become of the trixbox CE project?:

This is starting to turn into an unproductive thread. I'm going to address a few of the issues brought up here then I'm not going to post to this thread anymore.

1. 64-bit version - It doesn't make sense to build a 64 bit version. Testing has shown that you get 0 performance gain from it in telephony applications. I realize that other distros are doing it and I'm not sure why. You need to maintain two build machines and two completely separate QA processes. Maybe this is why distros like Elastix haven't gone to Asterisk 1.6 yet. They have been busy building a 64 bit version instead. I asked the trixbox community to prioritize what they wanted and Asterisk 1.6 was much higher than a 64 bit version. We have limited resources and we can only handle so much at once. I also realize that 64 bit is the future. Now with small Atom boxes supporting 64 bit I hope it's not long before CentOS drops 32 bit support completely. I would say we are about 2 years away from that now. At that point trixbox will drop 32-bit support and go to 64-bit for everything.

2. FreePBX fork - This has been hashed over many, many, many times in other threads. The result is we need control over FreePBX upgrades and we need to be able to make changes to FreePBX code to support things like the EndPoint manager. We are not the only distro that does this Elastix also has their own fork of FreePBX. It just makes sense. We have our own fork of Asterisk and our own fork of the Linux kernel. We need all these to provide the features and stability that trixbox users want.

3. Backup - A comprehensive backup system was the most requested feature by the trixbox community that last time I asked. We have been spending about 80% of our resources for the past few months creating one. A Beta will be out shortly.

Please try to keep this forum productive. "trixbox sucks" psots don't really help anybody. trixbox is a free product used and support by the trixbox community. If you don't like something in trixbox please post constructive criticism as to what you would like to see changed and why. If you find a problem with trixbox please post a bug report so we can resolve it.

We have had about 50,000 download of trixbnox 2.8 and for the most part things seem to be going smoothly. After we finish the backup system I'll start a new thread where we can discuss what features we want in the next version of trixbox.

Thanks Guys!

techman — Sat, 07 Nov 2009 06:36:12 -0800

In reply to Backup Module with 2.6.2:

Life savers! Works and rescued me!

http://www.devenlink.com
Advertise your business in Style- Free link exchange
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Had EXACTLY the same problem - Disable/enable trunk FIXED

theyollom — Sat, 07 Nov 2009 06:06:42 -0800

In reply to sipgate.co.uk stopped working:

Just thought I would let you know that I had EXACTLY the same problem today. Tried everything to get it working and saw this thread. As a last resort I tried disabling the trunk - save - reload configuration - enable to trunk - reload configuration and IT WORKED - problem solved - FIXED. I have no idea why just thought I would let others know !

prior link

dbaddorf — Sat, 07 Nov 2009 05:59:37 -0800

In reply to Hudlite Server & Client Installation on trixbo 2.6.x /2.6X. (Centos 5.x):

Had you checked everything out in the link: http://www.trixbox.org/forums/vendor-moderated-forums/hudlite-tri...? I don't know if there are other links or not on why the hud server wouldn't start, but that was one that i found.
I'm just guessing, but I wonder if maybe you have the wrong version of hudlite installed? I think that I would try removing the rpm's, and install hudlite with either the command line:

install-hudlite

Or go to Packages and Install “tbm-hudadmin”.
This is just a guess, but I hope maybe you will find something with this...

try these setting they are

safa786 — Sat, 07 Nov 2009 05:07:17 -0800

In reply to Inbound Sipgate - tearing my hair out!!!!:

try these setting they are working in my system
allow=alaw&ulaw&g729&gsm&slinear
bindaddr=0.0.0.0
context=from-trunk
disable=all
fromdomain=sipgate.co.uk
fromuser=SIPID
host=sipgate.co.uk
insecure=very
nat=yes
port=5060
qualify=yes
secret=SipagtePassword
srvlookup=yes
type=friend
username=SIPID

i had the same problem but after adding these into sip_nat.con sipgate works ok for incoming

nat=yes
externip=your external Ip
localnet=192.168.0.0 or 192.168.1.0 / 255.255.255.0

hope it will help

I use m0n0wall

johnf — Sat, 07 Nov 2009 04:36:29 -0800

In reply to small business firewall for Trixbox? :

pfsense is a fork of m0n0wall. Both are highly dependable, but m0n0 has less bells and whistles. Typically my m0n0 uptime is the time since the last power failure, sometimes months. The learning curve is a lot less steeper than for trixbox, it's almost a fit-and- forget type deal.