Voice Over IP in UK - ASTERISK@HOME 2.7 PROBLEMS

Author

ASTERISK@HOME 2.7 PROBLEMS

phpguy

2007-12-06, 7:11 pm

HELLO ALL I GOT 2 ASTERISK BOXES WITH X100P CARDS ONE IS ASTERISK 1.4
OTHER IS 2.7

CONFIGURATION IS THE SAME ONE WORKS GREAT AND CAN DIAL OUT WHEN I SET
UP THE ZAP INTERFACE (THE 1.4) BUT THE 2.7 FAILS IT HAS A VOICE THAT
SAYS ALL CIRCUITS ARE BUSY NOW AND TRY AGAIN LATER

I DONT SEE WHY THE 1.4 WORKS IMMEDIATELY AND THE 2.7 GIVES ME THAT
WQITH SAME CARDS AND CONFIGYRATION

I USE THEM FOR DIALING OUT ON REGULAR LANDLINES

ANYONE KNOW WHY? CHEERS

Jono

2007-12-06, 7:11 pm

phpguy laid this down on his screen :
> HELLO ALL I GOT 2 ASTERISK BOXES WITH X100P CARDS ONE IS ASTERISK 1.4
> OTHER IS 2.7
>
> CONFIGURATION IS THE SAME ONE WORKS GREAT AND CAN DIAL OUT WHEN I SET
> UP THE ZAP INTERFACE (THE 1.4) BUT THE 2.7 FAILS IT HAS A VOICE THAT
> SAYS ALL CIRCUITS ARE BUSY NOW AND TRY AGAIN LATER
>
> I DONT SEE WHY THE 1.4 WORKS IMMEDIATELY AND THE 2.7 GIVES ME THAT
> WQITH SAME CARDS AND CONFIGYRATION
>
> I USE THEM FOR DIALING OUT ON REGULAR LANDLINES
>
> ANYONE KNOW WHY? CHEERS

Have you run yum? Installed any updates?

Check out "Rebuild Zaptel" on this page
<http://nerdvittles.com/index.php?p=123>

Nick

2007-12-07, 1:11 am

"phpguy" <ksmaka2@hotmail.com> wrote in message
news:2a12c728-778b-4837-a1c5-35ea0981a6ac@b40g2000prf.googlegroups.com...
> HELLO ALL I GOT 2 ASTERISK BOXES WITH X100P CARDS ONE IS ASTERISK 1.4
> OTHER IS 2.7
>
> CONFIGURATION IS THE SAME ONE WORKS GREAT AND CAN DIAL OUT WHEN I SET
> UP THE ZAP INTERFACE (THE 1.4) BUT THE 2.7 FAILS IT HAS A VOICE THAT
> SAYS ALL CIRCUITS ARE BUSY NOW AND TRY AGAIN LATER
>
> I DONT SEE WHY THE 1.4 WORKS IMMEDIATELY AND THE 2.7 GIVES ME THAT
> WQITH SAME CARDS AND CONFIGYRATION
>
> I USE THEM FOR DIALING OUT ON REGULAR LANDLINES
>
> ANYONE KNOW WHY? CHEERS

Please don't shout, and I'll reply when I get my breath back.

Desk Rabbit

2007-12-07, 7:11 am

phpguy wrote:
> HELLO ALL I GOT 2 ASTERISK BOXES WITH X100P CARDS ONE IS ASTERISK 1.4
> OTHER IS 2.7
>
> CONFIGURATION IS THE SAME ONE WORKS GREAT AND CAN DIAL OUT WHEN I SET
> UP THE ZAP INTERFACE (THE 1.4) BUT THE 2.7 FAILS IT HAS A VOICE THAT
> SAYS ALL CIRCUITS ARE BUSY NOW AND TRY AGAIN LATER
>
> I DONT SEE WHY THE 1.4 WORKS IMMEDIATELY AND THE 2.7 GIVES ME THAT
> WQITH SAME CARDS AND CONFIGYRATION
>
> I USE THEM FOR DIALING OUT ON REGULAR LANDLINES
>
> ANYONE KNOW WHY? CHEERS

Very confusing message and why shout?

There is no such thing as Asterisk 2.7 did you mean Asterisk@Home or
Trixbox??

How about some error messages from the logs?

Jono

2007-12-07, 1:11 pm

on 07/12/2007, Desk Rabbit supposed :
> phpguy wrote:
>
> Very confusing message and why shout?
>
> There is no such thing as Asterisk 2.7

Read the subject line ;-)

> did you mean Asterisk@Home or
> Trixbox??

The former. From what I remember, he chose v2.7 as it seems to be the
last version that hasn't broken the call recording of inbound calls to
a ring group.

I keep meaning to try Ward Mundy's PBX-In-A-Flash
<http://nerdvittles.com>

> How about some error messages from the logs?

phpguy

2007-12-08, 7:11 pm

hello tere, yes juno is rtight , i dont now what yum is and what
updates to install ad why revuilt zaptel as both are fresh
installation and not upgrade

1.4 is fresh install on one pc and 2.7 fresh install on another

any help ? juno remembers me well i see hello mate

Paul Hayes

2007-12-10, 7:11 am

phpguy wrote:
> HELLO ALL I GOT 2 ASTERISK BOXES WITH X100P CARDS ONE IS ASTERISK 1.4
> OTHER IS 2.7
>
> CONFIGURATION IS THE SAME ONE WORKS GREAT AND CAN DIAL OUT WHEN I SET
> UP THE ZAP INTERFACE (THE 1.4) BUT THE 2.7 FAILS IT HAS A VOICE THAT
> SAYS ALL CIRCUITS ARE BUSY NOW AND TRY AGAIN LATER
>
> I DONT SEE WHY THE 1.4 WORKS IMMEDIATELY AND THE 2.7 GIVES ME THAT
> WQITH SAME CARDS AND CONFIGYRATION
>
> I USE THEM FOR DIALING OUT ON REGULAR LANDLINES
>
> ANYONE KNOW WHY? CHEERS

I don't really see how vanilla Asterisk 1.4 and Asterisk@Home 2.7 could
possibly have the same configuration. A@H adds it's own extremely
customised configuration behind the web interface it also uses. Why not
just use Asterisk@Home 2.7 on both machines?

cheers,
Paul.

Desk Rabbit

2007-12-10, 7:11 am

phpguy wrote:
> hello tere, yes juno is rtight , i dont now what yum is and what
> updates to install ad why revuilt zaptel as both are fresh
> installation and not upgrade
Yum is an updater program. Type man yum on the command line.

> 1.4 is fresh install on one pc and 2.7 fresh install on another
So you have plain vanilla Asterisk 1.4 on one box and Asterisk@home 2.7
on another?

> any help ?
Make the same call on both boxes, look at the error logs, compare and
contrast and post here.

phpguy

2007-12-10, 7:11 pm

you got it here it is:

login as: root
root@192.168.100.70's password:
Last login: Wed Dec 5 18:52:07 2007 from 192.168.100.6

Welcome to Asterisk@Home
-------------------------------------------------

For access to the Asterisk@Home web GUI use this URL
http://192.168.100.70

For help on Asterisk@Home commands you can use from this
command shell type help-aah.

[root@asterisk1 ~]# asterisk -r
Asterisk 1.2.5, Copyright (C) 1999 - 2006 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for
details.
This is free software, with components licensed under the GNU General
Public
License version 2 and other licenses; you are welcome to redistribute
it under
certain conditions. Type 'show license' for details.
========================================
=================================
Connected to Asterisk 1.2.5 currently running on asterisk1 (pid =
2659)
NIX connection
Verbosity is at least 3
asterisk1*CLI> set verbose 5
Verbosity was 3 and is now 5
-- Saved useragent "Sipura/SPA921-4.1.10(b)" for peer 200
-- Executing Macro("SIP/200-7adb", "dialout-trunk|1|2107236397|")
in new stack
-- Executing GotoIf("SIP/200-7adb", "1?3:2)") in new stack
-- Goto (macro-dialout-trunk,s,3)
-- Executing Macro("SIP/200-7adb", "user-callerid") in new stack
-- Executing DBget("SIP/200-7adb", "AMPUSER=DEVICE/200/user") in
new stack
-- DBget: varname=AMPUSER, family=DEVICE, key=200/user
-- DBget: set variable AMPUSER to 200
-- Executing DBget("SIP/200-7adb", "AMPUSERCIDNAME=AMPUSER/200/
cidname") in new stack
-- DBget: varname=AMPUSERCIDNAME, family=AMPUSER, key=200/cidname
-- DBget: set variable AMPUSERCIDNAME to KOSTAS
-- Executing GotoIf("SIP/200-7adb", "0?5") in new stack
-- Executing SetCallerID("SIP/200-7adb", ""KOSTAS" <200>") in new
stack
-- Executing NoOp("SIP/200-7adb", "Using CallerID "KOSTAS" <200>")
in new stack
-- Executing Macro("SIP/200-7adb", "record-enable|200|OUT") in new
stack
-- Executing GotoIf("SIP/200-7adb", "0 > 0?2:4") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing AGI("SIP/200-7adb", "recordingcheck|20071210-180545|
1197327945.0") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20071210-180545|1197327945.0: Outbound recording not
enabled
-- AGI Script recordingcheck completed, returning 0
-- Executing NoOp("SIP/200-7adb", "No recording needed") in new
stack
-- Executing Macro("SIP/200-7adb", "outbound-callerid|1") in new
stack
-- Executing DBget("SIP/200-7adb", "USEROUTCID=AMPUSER/200/
outboundcid") in new stack
-- DBget: varname=USEROUTCID, family=AMPUSER, key=200/outboundcid
-- DBget: set variable USEROUTCID to
-- Executing GotoIf("SIP/200-7adb", "0?4") in new stack
-- Executing SetCallerID("SIP/200-7adb", "2107292105") in new
stack
-- Executing GotoIf("SIP/200-7adb", "1?6") in new stack
-- Goto (macro-outbound-callerid,s,6)
-- Executing NoOp("SIP/200-7adb", "CallerID set to 2107292105") in
new stack
-- Executing SetGroup("SIP/200-7adb", "OUT_1") in new stack
-- Executing CheckGroup("SIP/200-7adb", "1") in new stack
-- Executing SetVar("SIP/200-7adb", "DIAL_NUMBER=2107236397") in
new stack
-- Executing SetVar("SIP/200-7adb", "DIAL_TRUNK=1") in new stack
-- Executing AGI("SIP/200-7adb", "fixlocalprefix") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
fixlocalprefix: Could not parse /etc/asterisk/localprefixes.conf
-- AGI Script fixlocalprefix completed, returning 0
-- Executing SetVar("SIP/200-7adb", "OUTNUM=2107236397") in new
stack
-- Executing Cut("SIP/200-7adb", "custom=OUT_1|:|1") in new stack
-- Executing GotoIf("SIP/200-7adb", "0?16") in new stack
-- Executing Dial("SIP/200-7adb", "ZAP/1/2107236397") in new stack
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing Goto("SIP/200-7adb", "s-CHANUNAVAIL|1") in new stack
-- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
-- Executing NoOp("SIP/200-7adb", "Dial failed due to
CHANUNAVAIL") in new stack
-- Executing Macro("SIP/200-7adb", "outisbusy") in new stack
-- Executing Playback("SIP/200-7adb", "all-circuits-busy-now") in
new stack
-- Playing 'all-circuits-busy-now' (language 'en')
-- Executing Playback("SIP/200-7adb", "pls-try-call-later") in new
stack
-- Playing 'pls-try-call-later' (language 'en')
-- Executing Macro("SIP/200-7adb", "hangupcall") in new stack
-- Executing ResetCDR("SIP/200-7adb", "w") in new stack
-- Executing NoCDR("SIP/200-7adb", "") in new stack
-- Executing Wait("SIP/200-7adb", "5") in new stack
== Spawn extension (macro-hangupcall, s, 3) exited non-zero on 'SIP/
200-7adb' in macro 'hangupcall'
== Spawn extension (macro-hangupcall, s, 3) exited non-zero on 'SIP/
200-7adb' in macro 'outisbusy'
== Spawn extension (macro-hangupcall, s, 3) exited non-zero on 'SIP/
200-7adb'
asterisk1*CLI>

phpguy

2007-12-10, 7:11 pm

and log from 1.4 (that works) is:

login as: root
root@192.168.100.69's password:
Last login: Thu Dec 6 01:46:20 2007 from 192.168.100.6
[root@asterisk1 root]# asterisk -r
Asterisk 1.0.9, Copyright (C) 1999-2004 Digium.
Written by Mark Spencer <markster@digium.com>
========================================
=================================
Connected to Asterisk 1.0.9 currently running on asterisk1 (pid =
1328)
Verbosity is at least 3
asterisk1*CLI> set verbose 5
Verbosity was 3 and is now 5
-- Registered SIP '200' at 192.168.100.15 port 5060 expires 3600
-- Executing Macro("SIP/200-93f9", "dialout-trunk|1|2107236397|")
in new stack
-- Executing GotoIf("SIP/200-93f9", "1?3:2)") in new stack
-- Goto (macro-dialout-trunk,s,3)
-- Executing Macro("SIP/200-93f9", "record-enable|200|OUT") in new
stack
-- Executing GotoIf("SIP/200-93f9", "0 > 0?2:4") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing GotoIf("SIP/200-93f9", "1?5:8") in new stack
-- Goto (macro-record-enable,s,5)
-- Executing DBget("SIP/200-93f9", "RecEnable=RECORD-OUT/200") in
new stack
-- DBget: varname=RecEnable, family=RECORD-OUT, key=200
-- DBget: Value not found in database.
-- Executing SetVar("SIP/200-93f9",
"CALLFILENAME=OUT200-20071211-012759-1197354479.262") in new stack
-- Executing Goto("SIP/200-93f9", "s|14") in new stack
-- Goto (macro-record-enable,s,14)
-- Executing GotoIf("SIP/200-93f9", "0?15:99") in new stack
-- Goto (macro-record-enable,s,99)
-- Executing NoOp("SIP/200-93f9", "NO RECORDING NEEDED") in new
stack
-- Executing GotoIf("SIP/200-93f9", "1?7") in new stack
-- Goto (macro-dialout-trunk,s,7)
-- Executing GotoIf("SIP/200-93f9", "0?9") in new stack
-- Executing SetCallerID("SIP/200-93f9", "2107292105") in new
stack
-- Executing SetGroup("SIP/200-93f9", "OUT_1") in new stack
-- Executing CheckGroup("SIP/200-93f9", "1") in new stack
-- Executing SetVar("SIP/200-93f9", "DIAL_NUMBER=2107236397") in
new stack
-- Executing SetVar("SIP/200-93f9", "DIAL_TRUNK=1") in new stack
-- Executing AGI("SIP/200-93f9", "fixlocalprefix") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
fixlocalprefix: Could not parse /etc/asterisk/localprefixes.conf
-- AGI Script fixlocalprefix completed, returning 0
-- Executing SetVar("SIP/200-93f9", "OUTNUM=2107236397") in new
stack
-- Executing Cut("SIP/200-93f9", "custom=OUT_1|:|1") in new stack
-- Executing GotoIf("SIP/200-93f9", "0?19") in new stack
-- Executing Dial("SIP/200-93f9", "ZAP/1/2107236397") in new stack
-- Called 1/2107236397
-- Zap/1-1 answered SIP/200-93f9
-- Hungup 'Zap/1-1'
== Spawn extension (macro-dialout-trunk, s, 17) exited non-zero on
'SIP/200-93f9' in macro 'dialout-trunk'
== Spawn extension (from-internal, 992107236397, 1) exited non-zero
on 'SIP/200-93f9'
-- Executing Macro("SIP/200-93f9", "hangupcall") in new stack
-- Executing ResetCDR("SIP/200-93f9", "w") in new stack
-- Executing NoCDR("SIP/200-93f9", "") in new stack
-- Executing Wait("SIP/200-93f9", "5") in new stack
== Spawn extension (macro-hangupcall, s, 3) exited non-zero on 'SIP/
200-93f9' in macro 'hangupcall'
== Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/
200-93f9'
asterisk1*CLI>

see also the thread
http://groups.google.co.uk/group/uk...isk%40home+2.7+(2)#88995ff9bc1dea81

Desk Rabbit

2007-12-11, 7:11 am

phpguy wrote:
> you got it here it is:
<Snipped>
On both boxes:-
In the asterisk console type show channels
At the OS command line type lsmod

Post both outputs.

I think your zaptel is screwed on the faulty box.

phpguy

2007-12-12, 1:11 pm

hi . zaptel you mean my fxo card ? i put the card of the 2.7 box in
the 1.,4 box and works great tho

ill try this in 2 days time as im out on business trip and post again
cheers

Desk Rabbit

2007-12-12, 1:11 pm

phpguy wrote:
> hi . zaptel you mean my fxo card ? i put the card of the 2.7 box in
> the 1.,4 box and works great tho
>
> ill try this in 2 days time as im out on business trip and post again
> cheers
By zaptel I mean the software that makes your card actually work. No
offence but you seriously need to do some research and reading if you
are playing with Asterisk systems. Start here
http://www.voip-info.org/wiki/view/Zaptel

phpguy

2007-12-12, 7:11 pm

hi i dont know if that means nothing but i reinstalled twice

cheers

Desk Rabbit

2007-12-13, 7:11 am

phpguy wrote:
> hi i dont know if that means nothing but i reinstalled twice
>
> cheers
Thats like saying you drove in a nail and it bent so you pulled it out
and drove it in again and was still bent.

Why didn't you do what I asked you to do earlier instead? It would have
been more useful. In case you lost it here it is again:-

On both boxes:-
In the asterisk console type show channels
At the OS command line type lsmod

Post both outputs.

phpguy

2007-12-14, 1:11 pm

hi thaks for your help ill try it tomorrow as im away for business
frofm those boxes

ill let you know tomorrow thanks appreciate it

phpguy

2007-12-15, 1:11 pm

hi thanks for your patience

the working 1.4 box

login as: root
root@192.168.100.69's password:
Last login: Tue Dec 11 01:26:01 2007 from 192.168.100.6
[root@asterisk1 root]# asterisk -r
Asterisk 1.0.9, Copyright (C) 1999-2004 Digium.
Written by Mark Spencer <markster@digium.com>
========================================
=================================
Connected to Asterisk 1.0.9 currently running on asterisk1 (pid =
1328)
Verbosity is at least 5
asterisk1*CLI> show channels
Channel (Context Extension Pri ) State Appl.
Data
0 active channel(s)
asterisk1*CLI> lsmod
No such command 'lsmod' (type 'help' for help)
asterisk1*CLI> exit
[root@asterisk1 root]# lsmod
Module Size Used by Not tainted
audit 89880 2 (autoclean)
autofs4 15832 0 (autoclean) (unused)
r8169 12168 1
crc32 3712 0 [r8169]
wcusb 19552 0 (unused)
wcfxo 9216 1
zaptel 179808 6 [wcusb wcfxo]
floppy 56624 0 (autoclean)
sg 36236 0 (autoclean) (unused)
scsi_mod 106664 1 (autoclean) [sg]
microcode 5688 0 (autoclean)
ide-cd 33920 0 (autoclean)
cdrom 32416 0 (autoclean) [ide-cd]
keybdev 2944 0 (unused)
mousedev 5524 0 (unused)
hid 22244 0 (unused)
input 5888 0 [keybdev mousedev hid]
usb-uhci 25740 0 (unused)
usbcore 77376 1 [wcusb hid usb-uhci]
ext3 85736 2
jbd 50668 2 [ext3]
[root@asterisk1 root]#

//////////////////////////////////////////

the other box

login as: root
root@192.168.100.70's password:
Last login: Mon Dec 10 18:03:40 2007 from 192.168.100.6

Welcome to Asterisk@Home
-------------------------------------------------

For access to the Asterisk@Home web GUI use this URL
http://192.168.100.70

For help on Asterisk@Home commands you can use from this
command shell type help-aah.

[root@asterisk1 ~]# asterisk -r
Asterisk 1.2.5, Copyright (C) 1999 - 2006 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for
details.
This is free software, with components licensed under the GNU General
Public
License version 2 and other licenses; you are welcome to redistribute
it under
certain conditions. Type 'show license' for details.
========================================
=================================
Connected to Asterisk 1.2.5 currently running on asterisk1 (pid =
2635)
Verbosity is at least 3
asterisk1*CLI> show channels
Channel Location State Application(Data)
0 active channels
0 active calls
asterisk1*CLI> exit
[root@asterisk1 ~]# lsmod
Module Size Used by
md5 4033 1
ipv6 234881 12
autofs4 23237 0
i2c_dev 11329 0
i2c_core 22081 1 i2c_dev
sunrpc 159269 1
ztdummy 3924 0
wctdm 34880 0
wcfxo 13088 0
wcte11xp 27936 0
wct1xxp 19488 0
wct4xxp 65600 0
tor2 91936 0
zaptel 206852 11
ztdummy,wctdm,wcfxo,wcte11xp,wct1xxp,wct
4xxp,tor2
crc_ccitt 2113 1 zaptel
microcode 6881 0
dm_mirror 27824 0
dm_mod 56661 1 dm_mirror
button 6481 0
battery 8901 0
ac 4805 0
uhci_hcd 31065 0
ehci_hcd 30917 0
snd_intel8x0 33897 0
snd_ac97_codec 63889 1 snd_intel8x0
snd_pcm_oss 49017 0
snd_mixer_oss 17985 1 snd_pcm_oss
snd_pcm 96841 2 snd_intel8x0,snd_pcm_oss
snd_timer 29893 1 snd_pcm
snd_page_alloc 9673 2 snd_intel8x0,snd_pcm
snd_mpu401_uart 8769 1 snd_intel8x0
snd_rawmidi 26597 1 snd_mpu401_uart
snd_seq_device 8137 1 snd_rawmidi
snd 55461 9
snd_intel8x0,snd_ac97_codec,snd_pcm_oss,
snd_mixer_oss,snd_pcm,snd_timer,snd_mpu4
01_uart,snd_rawmidi,snd_seq_device
soundcore 9889 1 snd
e100 41793 0
mii 4673 1 e100
ext3 116809 2
jbd 71385 1 ext3
[root@asterisk1 ~]#

///////////////////////////

help me god !