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uk.telecom.voip (UK VOIP) (uk.telecom.voip) Discussion of topics relevant to packet based voice technologies including Voice over IP (VoIP), Fax over IP (FoIP), Voice over Frame Relay (VoFR), Voice over Broadband (VoB) and Voice on the Net (VoN) as well as service providers, hardware and software for use with these technologies. Advertising is not allowed.

[email protected] 2.7 PROBLEMS



 
 
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  #1  
Old December 6th 07, 10:02 PM posted to uk.telecom.voip
phpguy
external usenet poster
 
Posts: 32
Default [email protected] 2.7 PROBLEMS

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
  #2  
Old December 6th 07, 10:08 PM posted to uk.telecom.voip
Jono
external usenet poster
 
Posts: 1,539
Default [email protected] 2.7 PROBLEMS

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


  #3  
Old December 7th 07, 05:48 AM posted to uk.telecom.voip
Nick
external usenet poster
 
Posts: 128
Default [email protected] 2.7 PROBLEMS


"phpguy" wrote in message
...
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.


  #4  
Old December 7th 07, 09:57 AM posted to uk.telecom.voip
Desk Rabbit
external usenet poster
 
Posts: 169
Default [email protected] 2.7 PROBLEMS

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 [email protected] or
Trixbox??

How about some error messages from the logs?

  #5  
Old December 7th 07, 06:57 PM posted to uk.telecom.voip
Jono
external usenet poster
 
Posts: 1,539
Default [email protected] 2.7 PROBLEMS

on 07/12/2007, Desk Rabbit supposed :
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


Read the subject line ;-)

did you mean [email protected] 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?



  #6  
Old December 8th 07, 06:03 PM posted to uk.telecom.voip
phpguy
external usenet poster
 
Posts: 32
Default [email protected] 2.7 PROBLEMS

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

  #7  
Old December 10th 07, 10:35 AM posted to uk.telecom.voip
Paul Hayes
external usenet poster
 
Posts: 50
Default [email protected] 2.7 PROBLEMS

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 [email protected] 2.7 could
possibly have the same configuration. [email protected] adds it's own extremely
customised configuration behind the web interface it also uses. Why not
just use [email protected] 2.7 on both machines?

cheers,
Paul.
  #8  
Old December 10th 07, 11:18 AM posted to uk.telecom.voip
Desk Rabbit
external usenet poster
 
Posts: 169
Default [email protected] 2.7 PROBLEMS

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 [email protected] 2.7
on another?

any help ?

Make the same call on both boxes, look at the error logs, compare and
contrast and post here.
  #9  
Old December 10th 07, 11:06 PM posted to uk.telecom.voip
phpguy
external usenet poster
 
Posts: 32
Default [email protected] 2.7 PROBLEMS

you got it here it is:


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

Welcome to [email protected]
-------------------------------------------------

For access to the [email protected] web GUI use this URL
http://192.168.100.70

For help on [email protected] commands you can use from this
command shell type help-aah.

[[email protected] ~]# asterisk -r
Asterisk 1.2.5, Copyright (C) 1999 - 2006 Digium, Inc. and others.
Created by Mark Spencer
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
  #10  
Old December 10th 07, 11:12 PM posted to uk.telecom.voip
phpguy
external usenet poster
 
Posts: 32
Default [email protected] 2.7 PROBLEMS

and log from 1.4 (that works) is:


login as: root
's password:
Last login: Thu Dec 6 01:46:20 2007 from 192.168.100.6
[[email protected] root]# asterisk -r
Asterisk 1.0.9, Copyright (C) 1999-2004 Digium.
Written by Mark Spencer
================================================== =======================
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....995ff9bc1dea81
 




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