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Home > Archive > Voice Over IP in UK > December 2007 > ASTERISK@HOME 2.7 PROBLEMS (2)
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ASTERISK@HOME 2.7 PROBLEMS (2)
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| phpguy 2007-12-10, 7:11 am |
| in responce to ASTERISK@HOME 2.7 PROBLEMS (part 1) to juno
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
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| After serious thinking phpguy wrote :
> in responce to ASTERISK@HOME 2.7 PROBLEMS (part 1) to juno
>
>
> hello tere, yes juno is rtight , i dont now what yum
You is simply the updater (like Windows update) for CentOS. If you yum
CentOS, it breaks Zaptel.
> 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
To avoid confusion - are you saying that both boxes have Aterisk@Home
on them? However, one is running a@h v1.4 and the other running a@h
v2.7.
>
> any help ?
As Desk Rabbit says, make the same call on both boxes whilst logged
into the Asterisk CLI.
Copy & paste the output of the CLI for both here.
> jono remembers me well i see hello mate
I do, I do.
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| Jono expressed precisely :
> After serious thinking phpguy wrote :
>
> You is simply the updater (like Windows update) for CentOS. If you yum
Duh! "Yum" is simply the updater
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| phpguy 2007-12-10, 7:11 pm |
| hi yes 2 boxes have both fresh install of asterisk@home but different
versions one has 1.4 the other 2.7
the one im using is 1.4 now and the 2.7 im test driving it before i
move to that so if anything happens my phones wont all be down
the log from 2.7 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>
///////////////////////////////////
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>
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