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Carlos Jerónimo

2007-Mar-29 03:56 UTC

[asterisk-users] error in FreePBX

Ive installed asterisk and freepbx. Through the interface ive
configured 2 extensions, 6000 and 6001.
My problem is that when i try to call from extension 6000 to 6001, i
hear this msg "Im-sorry&an-error-has-occured" and the call is
terminated.
As expected if i call to another number i get an error.
i thought the problem might been related with the NAT but if checked
and changed some NAT configuration parameters, it didnt worked aswell.
As this ever happened to anyone before? Any hints are very appreciated.

Thank you very much




-- 
Carlos Jer?nimo

Remco Barendse

2007-Mar-29 04:26 UTC

head link

[asterisk-users] error in FreePBX

On Thu, 29 Mar 2007, Carlos Jer?nimo wrote:
> Ive installed asterisk and freepbx. Through the interface ive
> configured 2 extensions, 6000 and 6001.
> My problem is that when i try to call from extension 6000 to 6001, i
> hear this msg "Im-sorry&an-error-has-occured" and the call is
> terminated.
> As expected if i call to another number i get an error.
> i thought the problem might been related with the NAT but if checked
> and changed some NAT configuration parameters, it didnt worked aswell.
> As this ever happened to anyone before? Any hints are very appreciated.
>
> Thank you very much
I have the same problem, it seems to occur when an extension is busy here.

All my extensions are on local lan with phones having ip addresses in a 
private range without NAT or anything so that is not the problem.

Sounds like an error in the dial pan FreePBX generated.

Brad Sumrall

2007-Mar-29 04:34 UTC

head link

[asterisk-users] Need help making a voice record server $$$

Hey there folks,

Looking to my favorite mailing list for assistance and have a few bucks to
pay you for your time.

Me: Played with asterisk for a while in the early days and getting stuck on
silly stuff on a time sensitive project for a friend.

Project:
PSTN incoming call to asterisk and then back to PSTN again, asterisk will
hold and record the RTP stream.

Upon disconnect, asterisk will name the record file by CID and Date.


That's it!

E mail me with how much you want for your time and this will surely grow
into other project that are later going to be implemented on this server.

Sincerely,
Brad
BradS@ftnco.com

Alex Robar

2007-Mar-29 05:19 UTC

head link

[asterisk-users] error in FreePBX

What output does the CLI generate when you try to make a call? It will tell
you what the system is doing, so it will usually give you a good indicator
of what is causing the call to fail.

Alex

On 3/29/07, Carlos Jer?nimo <carjer@gmail.com>
wrote:>
> Ive installed asterisk and freepbx. Through the interface ive
> configured 2 extensions, 6000 and 6001.
> My problem is that when i try to call from extension 6000 to 6001, i
> hear this msg "Im-sorry&an-error-has-occured" and the call is
> terminated.
> As expected if i call to another number i get an error.
> i thought the problem might been related with the NAT but if checked
> and changed some NAT configuration parameters, it didnt worked aswell.
> As this ever happened to anyone before? Any hints are very appreciated.
>
> Thank you very much
>
>
>
>
> --
> Carlos Jer?nimo
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>


-- 
Alex Robar
alex.robar@gmail.com
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Steve Murphy

2007-Mar-29 06:59 UTC

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[asterisk-users] error in FreePBX

On Thu, 2007-03-29 at 13:26 +0200, Remco Barendse wrote:> On Thu, 29 Mar 2007, Carlos Jer?nimo wrote:
> 
> > Ive installed asterisk and freepbx. Through the interface ive
> > configured 2 extensions, 6000 and 6001.
> > My problem is that when i try to call from extension 6000 to 6001, i
> > hear this msg "Im-sorry&an-error-has-occured" and the
call is
> > terminated.
> > As expected if i call to another number i get an error.
> > i thought the problem might been related with the NAT but if checked
> > and changed some NAT configuration parameters, it didnt worked aswell.
> > As this ever happened to anyone before? Any hints are very
appreciated.
> >
> > Thank you very much
> 
> I have the same problem, it seems to occur when an extension is busy here.
> 
> All my extensions are on local lan with phones having ip addresses in a 
> private range without NAT or anything so that is not the problem.
> 
> Sounds like an error in the dial pan FreePBX generated.
My suggestion: try a FreePBX mailing list first; the problem *is* more
likely to be in their stuff.

murf

-- 
Steve Murphy
Software Developer
Digium
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Carlos Jerónimo

2007-Mar-29 10:36 UTC

head link

[asterisk-users] error in FreePBX

Hi Steve, your sugestion is correct, but i registed 2 times in FreePbx
foruns this week, and my login is inactive yet. In the mail i receive
this msg:

********
Welcome to FreePBX Forums Forums

Please keep this email for your records. Your account information is as follows:


Your account is currently inactive, the administrator of the board
will need to activate it before you can log in. You will receive
another email when this has occured.
....

**********

because this i post this here.
Regards


2007/3/29, Steve Murphy <murf@digium.com>:> On Thu, 2007-03-29 at 13:26 +0200, Remco Barendse wrote:
> > On Thu, 29 Mar 2007, Carlos Jer?nimo wrote:
> >
> > > Ive installed asterisk and freepbx. Through the interface ive
> > > configured 2 extensions, 6000 and 6001.
> > > My problem is that when i try to call from extension 6000 to
6001, i
> > > hear this msg "Im-sorry&an-error-has-occured" and
the call is
> > > terminated.
> > > As expected if i call to another number i get an error.
> > > i thought the problem might been related with the NAT but if
checked
> > > and changed some NAT configuration parameters, it didnt worked
aswell.
> > > As this ever happened to anyone before? Any hints are very
appreciated.
> > >
> > > Thank you very much
> >
> > I have the same problem, it seems to occur when an extension is busy
here.
> >
> > All my extensions are on local lan with phones having ip addresses in
a
> > private range without NAT or anything so that is not the problem.
> >
> > Sounds like an error in the dial pan FreePBX generated.
>
> My suggestion: try a FreePBX mailing list first; the problem *is* more
> likely to be in their stuff.
>
> murf
>
> --
> Steve Murphy
> Software Developer
> Digium
>
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>

-- 
Carlos Jer?nimo

Philippe Lindheimer

2007-Mar-29 12:37 UTC

head link

[asterisk-users] error in FreePBX

something is broken in your configuration. dialparties is returning with no
extension to dial (which could be DND, could be the phone is occupied and no CW
active, or several other factors). And your call to voicemail is failing which
implies something is broken in that setup since it wouldn't go there unless
you configured it with voicemail.

Get onto the IRC (online help in freepbx or #freepbx with another IRC client)
and find one of the user who might be able to step you through the help. (or -
delete all your extensions, re-install and recreate them). If the only problem
were with the phone configuration (which is also a possibility) you would still
hit voicemail properly.

p

asterisk-users-request@lists.digium.com wrote:From: "Carlos Jer?nimo"
<carjer@gmail.com>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
 <asterisk-users@lists.digium.com>
Date: Thu, 29 Mar 2007 18:20:46 +0100
Subject: Re: [asterisk-users] error in FreePBX

 I haved read this but i not understand what the problem. mybe you
understand why this error, for all call, in a extensions configured in
asterisk/FreePbx.

ieeta-proj-04*CLI> sip show peers
Name/username              Host            Dyn Nat ACL Port     Status
6009/6009                  192.168.9.54     D   N      5062     OK (1 ms)
6000/6000                  192.168.1.211    D   N      5061     OK (3 ms)
2 sip peers [2 online , 0 offline]
    -- Executing Macro("SIP/6009-08197f70",
"exten-vm|6000|6000") in new stack
    -- Executing Macro("SIP/6009-08197f70", "user-callerid")
in new stack
    -- Executing NoOp("SIP/6009-08197f70", "user-callerid: device
6009") in new stack
    -- Executing GotoIf("SIP/6009-08197f70", "0?report") in
new stack
    -- Executing GotoIf("SIP/6009-08197f70", "0?start") in
new stack
    -- Executing Set("SIP/6009-08197f70",
"REALCALLERIDNUM=6009") in new stack
    -- Executing NoOp("SIP/6009-08197f70", "REALCALLERIDNUM is
6009")
in new stack
    -- Executing Set("SIP/6009-08197f70", "AMPUSER=6009") in
new stack
    -- Executing Set("SIP/6009-08197f70",
"AMPUSERCIDNAME=6009") in new stack
    -- Executing GotoIf("SIP/6009-08197f70", "0?report") in
new stack
    -- Executing Set("SIP/6009-08197f70", "CALLERID(all)=6009
<6009>")
in new stack
    -- Executing Set("SIP/6009-08197f70",
"REALCALLERIDNUM=6009") in new stack
    -- Executing NoOp("SIP/6009-08197f70", "TTL:  ARG1:
6000") in new stack
    -- Executing GotoIf("SIP/6009-08197f70", "0?continue")
in new stack
    -- Executing Set("SIP/6009-08197f70", "_TTL=64") in new
stack
    -- Executing GotoIf("SIP/6009-08197f70", "1?continue")
in new stack
    -- Goto (macro-user-callerid,s,21)
    -- Executing NoOp("SIP/6009-08197f70", "Using CallerID
"6009"
<6009>") in new stack
    -- Executing Set("SIP/6009-08197f70",
"FROMCONTEXT=exten-vm") in new stack
    -- Executing Set("SIP/6009-08197f70", "VMBOX=6000") in
new stack
    -- Executing Set("SIP/6009-08197f70", "EXTTOCALL=6000")
in new stack
    -- Executing Set("SIP/6009-08197f70", "CFUEXT=") in new
stack
    -- Executing Set("SIP/6009-08197f70", "CFBEXT=") in new
stack
    -- Executing Set("SIP/6009-08197f70", "RT=15") in new
stack
    -- Executing Macro("SIP/6009-08197f70",
"record-enable|6000|IN")
in new stack
    -- Executing GotoIf("SIP/6009-08197f70", "0?2:4") in new
stack
    -- Goto (macro-record-enable,s,4)
    -- Executing DeadAGI("SIP/6009-08197f70",
"recordingcheck|20070329-181220|1175188340.3") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
    -- AGI Script recordingcheck completed, returning 0
    -- Executing NoOp("SIP/6009-08197f70", "No recording
needed") in new stack
    -- Executing Macro("SIP/6009-08197f70",
"dial|15|tr|6000") in new stack
    -- Executing DeadAGI("SIP/6009-08197f70",
"dialparties.agi") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
    -- AGI Script dialparties.agi completed, returning 0
    -- Executing NoOp("SIP/6009-08197f70", "Returned from
dialparties
with no extensions to call") in new stack
    -- Executing NoOp("SIP/6009-08197f70", "DIALSTATUS is ")
in new stack
    -- Executing GosubIf("SIP/6009-08197f70", "0?docfu|1")
in new stack
    -- Executing GosubIf("SIP/6009-08197f70", "0?docfb|1")
in new stack
    -- Executing NoOp("SIP/6009-08197f70", "Voicemail is
6000") in new stack
    -- Executing GotoIf("SIP/6009-08197f70", "0?s-|1") in
new stack
    -- Executing NoOp("SIP/6009-08197f70", "Sending to Voicemail
box
6000") in new stack
    -- Executing Macro("SIP/6009-08197f70", "vm|6000|") in
new stack
    -- Executing Macro("SIP/6009-08197f70",
"user-callerid|SKIPTTL")
in new stack
    -- Executing NoOp("SIP/6009-08197f70", "user-callerid: 6009
6009")
in new stack
    -- Executing GotoIf("SIP/6009-08197f70", "0?report") in
new stack
    -- Executing GotoIf("SIP/6009-08197f70", "1?start") in
new stack
    -- Goto (macro-user-callerid,s,5)
    -- Executing NoOp("SIP/6009-08197f70", "REALCALLERIDNUM is
6009")
in new stack
    -- Executing Set("SIP/6009-08197f70", "AMPUSER=6009") in
new stack
    -- Executing Set("SIP/6009-08197f70",
"AMPUSERCIDNAME=6009") in new stack
    -- Executing GotoIf("SIP/6009-08197f70", "0?report") in
new stack
    -- Executing Set("SIP/6009-08197f70", "CALLERID(all)=6009
<6009>")
in new stack
    -- Executing Set("SIP/6009-08197f70",
"REALCALLERIDNUM=6009") in new stack
    -- Executing NoOp("SIP/6009-08197f70", "TTL: 64 ARG1:
SKIPTTL") in
new stack    -- Executing GotoIf("SIP/6009-08197f70",
"1?continue") in
new stack
    -- Goto (macro-user-callerid,s,21)
    -- Executing NoOp("SIP/6009-08197f70", "Using CallerID
"6009"
<6009>") in new stack
    -- Executing Set("SIP/6009-08197f70", "VMGAIN=") in new
stack
    -- Executing GotoIf("SIP/6009-08197f70", "1?s-|1") in
new stack
    -- Goto (macro-vm,s-,1)
    -- Executing Macro("SIP/6009-08197f70",
"get-vmcontext|6000") in new stack
    -- Executing Set("SIP/6009-08197f70", "VMCONTEXT=") in
new stack
    -- Executing GotoIf("SIP/6009-08197f70", "1?200:300") in
new stack
    -- Goto (macro-get-vmcontext,s,200)
    -- Executing Set("SIP/6009-08197f70",
"VMCONTEXT=default") in new stack
    -- Executing VoiceMail("SIP/6009-08197f70",
"6000@default|u") in new stack
    -- Executing Goto("SIP/6009-08197f70", "exit-FAILED|1")
in new stack
    -- Goto (macro-vm,exit-FAILED,1)
    -- Executing Playback("SIP/6009-08197f70",
"im-sorry&an-error-has-occured") in new stack
    -- Playing 'im-sorry' (language 'en')
    -- Playing 'an-error-has-occured' (language 'en')
    -- Executing Hangup("SIP/6009-08197f70", "") in new
stack
  == Spawn extension (macro-vm, exit-FAILED, 2) exited non-zero on
'SIP/6009-08197f70' in macro 'vm'
  == Spawn extension (macro-vm, exit-FAILED, 2) exited non-zero on
'SIP/6009-08197f70' in macro 'exten-vm'
  == Spawn extension (macro-vm, exit-FAILED, 2) exited non-zero on
'SIP/6009-08197f70'
ieeta-proj-04*CLI>
********************************************

thanks

Carlos Jer?nimo




 
---------------------------------
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Francesco Peeters (Asterisk)

2007-Mar-29 14:51 UTC

head link

[asterisk-users] error in FreePBX

On Thu, March 29, 2007 19:36, Carlos Jer?nimo wrote:> Hi Steve, your sugestion is correct, but i registed 2 times in FreePbx
> foruns this week, and my login is inactive yet. In the mail i receive
> this msg:
>
> ********
> Welcome to FreePBX Forums Forums
>
> Please keep this email for your records. Your account information is as
> follows:
>
>
> Your account is currently inactive, the administrator of the board
> will need to activate it before you can log in. You will receive
> another email when this has occured.
> ....
>Same here... Been waiting a week since my last attempt, but still nothing!...

-- 
F Peeters
  PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch
  2 Sweex HFC-PCI modes=0 sync_slave=0 timer_card=0
  AMD Duron 1GHz - 1GB - * 1.2.1 - vISDN
  2 Sweex HFC-PCI cards

Jaswinder Singh

2007-Mar-30 06:32 UTC

head link

[asterisk-users] error in FreePBX

disable voicemail for  that extension .. apply settings .. re-enable
voicemail .. re-apply settings . this helped me once before.
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