Re: [asterisk-users] Seg Faults with 1.6.2.19

[Lee Archer]

Hi Kevin, the ticket below was closed as it doesn't happen with 1.8.  It
can't be related to my ODBC connections if others are having it.  Should
a new ticket be opened?

Regards

Lee

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P.
Fleming
Sent: 18 July 2011 15:10
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Seg Faults with 1.6.2.19

On 07/18/2011 08:07 AM, Steve Davies wrote:
> On 18 July 2011 14:05, Lee Archer  wrote:
>> Seems to be an already reported problem but since no more fixes for 
>> 1.6 it's back to 1.6.2.18.2 for me.
>>
>> https://issues.asterisk.org/jira/browse/ASTERISK-18103
>>
>> Regards
>>
>> Lee
>>
>
> If it is a regression introduced in 1.6.2.19, then it should still be
fixed.
>
> At least I believe that's the rules.

That should be the case, yes.

--
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Re: [asterisk-users] Seg Faults with 1.6.2.19

[Lee Archer]

Seems to be an already reported problem but since no more fixes for 1.6
it's back to 1.6.2.18.2 for me.

https://issues.asterisk.org/jira/browse/ASTERISK-18103

Regards

Lee

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lee Archer
Sent: 18 July 2011 14:01
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Seg Faults with 1.6.2.19

Hi Eric, are you using ODBC?

Regards

Lee

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric
Wieling
Sent: 18 July 2011 13:54
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Seg Faults with 1.6.2.19



Sent from my Computer

> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lee 
> Archer
> Sent: Monday, July 18, 2011 7:04 AM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] Seg Faults with 1.6.2.19
>
> Hi, is anyone else having problems with the reload command crashing 
> Asterisk 1.6.2.19?  I've run a few tests and
> 1.6.2.18.2 doesn't have this problem but 1.6.2.19 after a few reloads 
> is just dumping and restarting.

We experienced the same thing.  After a few reloads, Asterisk crashes.

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Re: [asterisk-users] Seg Faults with 1.6.2.19

[Lee Archer]

Hi Eric, are you using ODBC?

Regards

Lee

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric
Wieling
Sent: 18 July 2011 13:54
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Seg Faults with 1.6.2.19



Sent from my Computer

> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lee 
> Archer
> Sent: Monday, July 18, 2011 7:04 AM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] Seg Faults with 1.6.2.19
>
> Hi, is anyone else having problems with the reload command crashing 
> Asterisk 1.6.2.19?  I've run a few tests and
> 1.6.2.18.2 doesn't have this problem but 1.6.2.19 after a few reloads 
> is just dumping and restarting.

We experienced the same thing.  After a few reloads, Asterisk crashes.

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Re: [asterisk-users] Seg Faults with 1.6.2.19

[Lee Archer]

Hi Steve, I think it's related to my ODBC connection.  I started with a random 
hang where it looked ODBC related which led me to try a few things.  Reloading 
the config a few times is causing core dumps which 1.6.2.18.2 just doesn't 
have, however my main reason for using 1.6.2.19 is a fix to ODBC so I don't 
really want to downgrade.  I will try and get some traces from one of my test 
boxes.

Thanks

Lee

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Davies
Sent: 18 July 2011 12:52
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Seg Faults with 1.6.2.19

On 18 July 2011 12:03, Lee Archer  wrote:
> Hi, is anyone else having problems with the reload command crashing 
> Asterisk 1.6.2.19?  I've run a few tests and 1.6.2.18.2 doesn't have 
> this problem but
> 1.6.2.19 after a few reloads is just dumping and restarting.
>
> Thanks
>
> Lee
>

I've not had a problem here with 1.6.2.19.

What are you reloading that causes the issue, and can you post the usual gdb 
backtrace somewhere? Perhaps on the bug tracker.

Regards,
Steve

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[asterisk-users] Seg Faults with 1.6.2.19

[Lee Archer]

Hi, is anyone else having problems with the reload command crashing
Asterisk 1.6.2.19?  I've run a few tests and 1.6.2.18.2 doesn't have
this problem but 1.6.2.19 after a few reloads is just dumping and
restarting.

Thanks

Lee
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Re: [asterisk-users] Recording SIP history

[Lee Archer]

Hi, can anyone help with this?

 

Thanks

 

Lee

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lee Archer
Sent: 05 July 2011 16:27
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Recording SIP history

 

Hi all, can someone explain what siphistory is supposed to do as it
appears to record nothing to my log files.  When I sip show history
 it's fine but it's not logging anything.  My logger.conf has
debug => debug and the debug file grows.  Is my understanding correct in
that at the end of the call the entire sip show history  should
be dumped to the debug file?  I am using 1.6.2.19.

;--- SIP DEBUGGING
---

sipdebug=yes ; Turn on SIP debugging by default, from

; the moment the channel loads this
configuration

recordhistory=yes  ; Record SIP history by default

; (see sip history / sip no history)

dumphistory=yes; Dump SIP history at end of SIP dialogue

; SIP history is output to the DEBUG
logging channel

Thanks

Lee

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[asterisk-users] Recording SIP history

[Lee Archer]

Hi all, can someone explain what siphistory is supposed to do as it
appears to record nothing to my log files.  When I sip show history
 it's fine but it's not logging anything.  My logger.conf has
debug => debug and the debug file grows.  Is my understanding correct in
that at the end of the call the entire sip show history  should
be dumped to the debug file?  I am using 1.6.2.19.

;--- SIP DEBUGGING
---
sipdebug=yes ; Turn on SIP debugging by default, from
; the moment the channel loads this
configuration
recordhistory=yes  ; Record SIP history by default
; (see sip history / sip no history)
dumphistory=yes; Dump SIP history at end of SIP dialogue
; SIP history is output to the DEBUG
logging channel

Thanks

Lee
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[asterisk-users] Vestec for Asterisk

[Lee Archer]

Hi, I installed the Vestec module to one of my development Asterisk
servers a few months ago but now I need to move the license to another
host.  Does anyone know how to do this?  I've had a look on my Account
page on the Digium website but it only shows the Language Pack, and I
can't do anything with this either.

Can anyone point me in the right direction please?

Thanks

Lee
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[asterisk-users] QUEUE_PRIO

[Lee Archer]

Hi, does QUEUE_PRIO work the Queues and Asterisk 1.6.2?  I've found some
documentation on Google but it looks like it's old Asterisk and not
current.

Thanks

Lee
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Re: [asterisk-users] No MOH with parked call

[Lee Archer]

Hi, try unloading res_timing_dahdi.so then trying again.

Lee

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve
Davies
Sent: 07 December 2010 12:54
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] No MOH with parked call

Hi,

Has anybody else noticed that MOH does not play on parked calls in
1.6.2.14? Or is it just my setup? MOH seems to work in every other
respect (Call Held or in-Queue), but when a call is parked, the logs
show MOH being started, but the parked party hears nothing.

The verbose logs show the following. Any thoughts on whet to check next?

Thanks,
Steve


### Call comes in here and is answered
-- SIP/snom360-0d6f answered DAHDI/2-1
-- Executing [...@macro-set-moh-call:1] GotoIf("SIP/snom360-0d6f",
"0?done") in new stack
-- Executing [...@macro-set-moh-call:2] Set("SIP/snom360-0d6f",
"CHANNEL(musicclass)=m-default") in new stack
-- Executing [...@macro-set-moh-call:3] NoOp("SIP/snom360-0d6f",
"") in new stack

### Here the call is being blind transferred to the Park number
-- Started music on hold, class 'default', on DAHDI/2-1
-- Stopped music on hold on DAHDI/2-1
  == Spawn extension (local, 210, 1) exited non-zero on 'DAHDI/2-1'
-- Executing [...@local:1] ForkCDR("DAHDI/2-1", "") in new stack
-- Executing [...@local:2] Set("DAHDI/2-1", "CDR(userfield)=") in
new stack

### Not sure why I send "Ringing" here, but I tried NoOP() and
Answer() too just in case
-- Executing [...@local:3] Ringing("DAHDI/2-1", "") in new stack
-- Executing [...@local:4] Set("DAHDI/2-1",
"CHANNEL(musicclass)=default") in new stack
-- Executing [...@local:5] Set("DAHDI/2-1",
"CHANNEL(parkinglot)=default") in new stack
-- Executing [...@local:6] Goto("DAHDI/2-1",
"parkedcalls_default,park,1") in new stack
-- Goto (parkedcalls_default,park,1)
-- Executing [p...@parkedcalls_default:1] Park("DAHDI/2-1", "") in
new stack
  == Parked DAHDI/2-1 on 211 (lot default). Will timeout back to
extension [parkedcalls_default] s, 1 in 90 seconds
-- Added extension '211' priority 1 to parkedcalls_default
(0xbe2e528)

# The "211" announcement is heard perfectly
--  Playing 'digits/2.alaw' (language 'en')
  == Extension Changed 211[extensions] new state InUse for Notify User
steve
--  Playing 'digits/1.alaw' (language 'en')
--  Playing 'digits/1.alaw' (language 'en')

# The system claims to start MOH "default" which works elsewhere, but
the caller gets silence
-- Started music on hold, class 'default', on DAHDI/2-1
  == Spawn extension (parkedcalls_default, s, 1) exited non-zero on
'Parked/DAHDI/2-1'

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[asterisk-users] CDR updating

[Lee Archer]

Hi, I am using Asterisk 1.6.2.13 and have an issue but I'm not sure if
it's a bug or not.  I am using the cdr_adaptive_odbc logging module and
writing my CDR records to a MS-SQL server.  I need to log which end
hangs the call up and have placed the relevant
CDR(myfield)=caller/callee commands where they need to be.  

When I watch the call on the console I can see the CDR field being set
properly but when I check the CDR record it is incorrect.  It appears
that when one end hangs up the CDR is being written immediately instead
of waiting until the h exten.  I have had a look in cdr.conf and set
endbeforehexten=no, but this doesn't seem to make any difference.  

Does anyone have any ideas or is it a problem with the cdr_adaptive_odbc
module?

Thanks

Lee


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Re: [asterisk-users] Use of AGISIGHUP

[Lee Archer]

Thanks for the replies.  I am already ignoring the signal but it doesn't
seem to be making much difference on this system with this script.  It's
an old legacy script I should hopefully be dropping and writing properly
within the dial plan.

I will keep trying!

Thanks

Lee

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve
Edwards
Sent: 26 August 2010 21:02
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Use of AGISIGHUP

>> On Thu, 26 Aug 2010, Lee Archer wrote:
>
>>> I am setting AGISIGHUP=no before running a Perl script via AGI but 
>>> it doesn?t seem to be doing anything as the script is still exiting 
>>> on a hangup and not completing properly.

> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve 
> Edwards
>
>> I'm just a 1.2 Luddite, so I've never seen AGISIGHUP and I think it's

>> a bad idea to protect lazy programmers :)

On Thu, 26 Aug 2010, Danny Nicholas wrote:

> Here's a one-liner that should "fix" the problem
>
> local $SIG{HUP} = 'IGNORE';
>
> Does that make me lazy?

Not at all. If that is the correct "response" for your program, it's
perfect:

1) The program will execute correctly on your system, my system, any
system regardless of the configuration.

2) It tells the next guy explicitly what you intended to happen upon
receiving the signal.

--
Thanks in advance,

-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867
PST
Newline  Fax:
+1-760-731-3000

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[asterisk-users] Use of AGISIGHUP

[Lee Archer]

Hi, I am setting AGISIGHUP=no before running a Perl script via AGI but
it doesn't seem to be doing anything as the script is still exiting on a
hangup and not completing properly.  I am using 1.4.35 and have tried
various combinations.  Can anyone shed any light on this?

Regards

Lee
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Re: [asterisk-users] Adding a context from the console

[Lee Archer]

Should I log this as a bug since it doesn't work?

Regards

Lee

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lee Archer
Sent: 20 May 2010 16:28
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Adding a context from the console

Hi, this didn't seem to work.  Is there something I am missing?

dialplan add extension 1234,1,NoOp,hello into default
Extension '1234,1,NoOp,hello' added into 'default' context
-- Added extension '1234' priority 1 to default (0x8e8f520)

dialplan add extension 1234,1,NoOp,hello into test
Failed to add '1234,1,NoOp,hello' extension into 'test' context

Regards

Lee

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lee Archer
Sent: 19 May 2010 16:54
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Adding a context from the console

Many thanks.

Regards

Lee

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman
Lesher
Sent: 19 May 2010 16:44
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Adding a context from the console

On Wednesday 19 May 2010 02:28:02 Lee Archer wrote:
> Hi, is it possible to add a context from the console using the
dialplan
> command?

Yes, just add an extension to it.  The context will be created as
needed.

-- 
Tilghman Lesher
Digium, Inc. | Senior Software Developer
twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
Check us out at: www.digium.com & www.asterisk.org

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Re: [asterisk-users] Help with IP Routing

[Lee Archer]

Try a Cisco ASA.  It will rewrite the headers if configured properly.

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Motiejus Jakštys
Sent: 26 May 2010 14:17
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Help with IP Routing

Assume previous IP is LAN. Forwarding public IP ports to LAN is
straighforward. However, with SIP headers you will (don't know H323)
have to modify outgoing SIP headers: replace LAN ip with WAN ip.
For callers you have to substitute RTP destination IP
For callees you have to substitute RTP source IP.

I`m afraid you will have to check more details here:
http://www.ietf.org/rfc/rfc3261.txt
Maybe client sends server it's own IP address?

However, dumb header substitution + port range forwarding should work
in all cases for SIP.

On Wed, May 26, 2010 at 3:55 PM, Nivin Kumar  wrote:
>
> Hello,
>
> I'm in a bit of a fix. We have a particular Windows based softswitch which is 
> has its SIP and H323 ports hardcoded to listen on a particular IP address. 
> The problem is that the ISP is having major issues and we can no longer 
> depend on them for service. The softswitch will not listen on any other IP 
> address and this can not be fixed. I was thinking of creating a NAT network 
> wherein we will forward all traffic from another public ip address to this 
> server, however I'm not sure how this will work. Do I need to modify the sip 
> headers? Any thoughts or suggestions?
>
> Thanks,
> Nivin
>
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Re: [asterisk-users] Adding a context from the console

[Lee Archer]

Hi, this didn't seem to work.  Is there something I am missing?

dialplan add extension 1234,1,NoOp,hello into default
Extension '1234,1,NoOp,hello' added into 'default' context
-- Added extension '1234' priority 1 to default (0x8e8f520)

dialplan add extension 1234,1,NoOp,hello into test
Failed to add '1234,1,NoOp,hello' extension into 'test' context

Regards

Lee

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lee Archer
Sent: 19 May 2010 16:54
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Adding a context from the console

Many thanks.

Regards

Lee

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman
Lesher
Sent: 19 May 2010 16:44
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Adding a context from the console

On Wednesday 19 May 2010 02:28:02 Lee Archer wrote:
> Hi, is it possible to add a context from the console using the
dialplan
> command?

Yes, just add an extension to it.  The context will be created as
needed.

-- 
Tilghman Lesher
Digium, Inc. | Senior Software Developer
twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
Check us out at: www.digium.com & www.asterisk.org

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Re: [asterisk-users] Adding a context from the console

[Lee Archer]

Many thanks.

Regards

Lee

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman
Lesher
Sent: 19 May 2010 16:44
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Adding a context from the console

On Wednesday 19 May 2010 02:28:02 Lee Archer wrote:
> Hi, is it possible to add a context from the console using the
dialplan
> command?

Yes, just add an extension to it.  The context will be created as
needed.

-- 
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Digium, Inc. | Senior Software Developer
twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
Check us out at: www.digium.com & www.asterisk.org

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Re: [asterisk-users] Adding a context from the console

[Lee Archer]

Hi, anyone know?

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lee Archer
Sent: 17 May 2010 11:29
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Adding a context from the console

 

Hi, is it possible to add a context from the console using the dialplan
command?

Thanks

Lee

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[asterisk-users] Adding a context from the console

[Lee Archer]

Hi, is it possible to add a context from the console using the dialplan
command?

Thanks

Lee
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[asterisk-users] Have a macro update a channel variable

[Lee Archer]

Hi, I wonder if anyone can help me with a macro issue I have.  I need to
set a variable which tells me whether a call has been authenticated
properly.  However this authentication is taking place inside of a macro
and I don't want to use a global variable if it will apply to other
channels.  I've tried using _ and __ with no real success.  Is there a
way of having a macro update a channel variable so when the call ends I
can check the variable and handle according?  I can NoOp the variable in
the macro prior to changing it and it shows what it should.  I then
change the variable to AUTH for a successful authentication and a NoOp
shows the correct value again.  But when the call ends the variable
going back to the original value.

Thanks

Lee
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Re: [asterisk-users] Continue dialplan is source channel hangs up

[Lee Archer]

I upgraded to 1.6 and tried F and it didn't do the same as the g option.  I 
will have to use the h extension to finish the logging.

 

Thanks

 

Lee

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sebastian
Sent: 10 May 2010 14:45
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Continue dialplan is source channel hangs up

 

  F([[context^]exten^]priority): When the caller hangs up, transfer

the called party to the specified destination and continue execution at

that location.

 

Also just F will continue to the next priority on the dialplan.

 

 

De: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] En nombre de Lee Archer
Enviado el: lunes, 10 de mayo de 2010 9:36
Para: asterisk-users@lists.digium.com
Asunto: [asterisk-users] Continue dialplan is source channel hangs up

 

Hi, does anyone know if there is an equivalent dial option for the source 
channel to the g option?  I've had a good look and can't find one.  

g- Proceed with dialplan execution at the current extension if the 
destination channel hangs up.

Thanks

Lee

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Re: [asterisk-users] Records sets and ODBC

[Lee Archer]

Thanks, I figured it out.  I was using 1.4 but have had to move to 1.6.1

Regards

Lee

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman
Lesher
Sent: 10 May 2010 17:27
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Records sets and ODBC

On Monday 10 May 2010 07:19:34 Lee Archer wrote:
> Hi, I have a system using ODBC and connecting to a MS-SQL database.
> Does anyone know if it is possible to return a record set consisting
of
> several rows from SQL back into Asterisk?  I have tried using ARRAY
but
> only the contents of the last row are being stored.

Only in 1.6.x (mode=multirow).

-- 
Tilghman

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Re: [asterisk-users] Continue dialplan is source channel hangs up

[Lee Archer]

Thanks.  Is there no 1.4 equivalent or is this a feature of 1.6 only?

 

Lee

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sebastian
Sent: 10 May 2010 14:45
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Continue dialplan is source channel hangs up

 

  F([[context^]exten^]priority): When the caller hangs up, transfer

the called party to the specified destination and continue execution at

that location.

 

Also just F will continue to the next priority on the dialplan.

 

 

De: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] En nombre de Lee Archer
Enviado el: lunes, 10 de mayo de 2010 9:36
Para: asterisk-users@lists.digium.com
Asunto: [asterisk-users] Continue dialplan is source channel hangs up

 

Hi, does anyone know if there is an equivalent dial option for the source 
channel to the g option?  I've had a good look and can't find one.  

g- Proceed with dialplan execution at the current extension if the 
destination channel hangs up.

Thanks

Lee

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[asterisk-users] Continue dialplan is source channel hangs up

[Lee Archer]

Hi, does anyone know if there is an equivalent dial option for the
source channel to the g option?  I've had a good look and can't find
one.  

g- Proceed with dialplan execution at the current extension if the
destination channel hangs up.

Thanks

Lee
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[asterisk-users] Records sets and ODBC

[Lee Archer]

Hi, I have a system using ODBC and connecting to a MS-SQL database.
Does anyone know if it is possible to return a record set consisting of
several rows from SQL back into Asterisk?  I have tried using ARRAY but
only the contents of the last row are being stored.

Thanks

Lee  
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Re: [asterisk-users] CDR: Add Dialed Number Identifierfield (DNID)field into MySQL

[Lee Archer]

Do we need an update to cdr_addon_mysql for this to work?

Lee

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of RSCL Mumbai
Sent: 16 March 2010 07:35
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] CDR: Add Dialed Number Identifierfield 
(DNID)field into MySQL

>> I have read 2 solutions
>
>> (a) Changing the Dial plan and capturing DNID and inserting it into
>> one of the existing column in CDR table.
>
>> (b) Copy new CDR related .c & .h files which have added the
>> functionality of recording DNID into MySQL.
>> For this, CDR table structure needs to be changed and a new field has
>> be created in CDR table.
>
>> But I am still not very sure on how to go about doing this.
>> Since I only have a production server, I do not have the options of
>> experimenting.
>> Can someone help with a step-by-step?
>
>> Thx
>> Sanjay
>
>
>
>
>>> On Mon, Mar 15, 2010 at 3:08 PM, Lee Archer  
>>> wrote:
>>> Isn't the use of DNID separate to the userfield?  I'd like to have this
>>> working also.
>>>
>>> Lee
>>>
>>> -Original Message-
>>> From: asterisk-users-boun...@lists.digium.com
>>> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alex
>>> Balashov
>>> Sent: 15 March 2010 08:34
>>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>>> Subject: Re: [asterisk-users] CDR: Add Dialed Number Identifierfield
>>> (DNID) field into MySQL
>>>
>>> Use the userfield.
>>>
>>> On 03/15/2010 04:25 AM, RSCL Mumbai wrote:
>>>
>>>> Hi,
>>>>
>>>> I would like to see the DNID in my MySQL CDR logs.
>>>>
>>>> I have read one big thread in the Asterisk Developer List, but I could
>>>> not figure out how to implement it ?
>>>> Is there a simple step-by-step.
>
>
> If this is Asterisk 1.6.*, then you can use the adaptive ODBC, which is 
> configured using /etc/asterisk/cdr_adaptive_odbc.conf.  If you compiled 
> Asterisk with samples, you will find a sample file that has pretty much 
> everything that you need.  From there, simply set the fieldname that you wish 
> to write to the CDR, like this:
>
> ; Using Adaptive ODBC CDR's, sets the caller ID DNID to the CDR's custom 
> field named "DNID"
> Set(CDR(DNID)=${CALLERID(DNID)})
>
> Personally, I like to set the DNID to a variable, just in case, when the 
> inbound call first hits Asterisk from the trunk.  This probably isn't 
> necessary, but I am always afraid that the CALLERID(DNID) value will change 
> with a transfer or a channel redirect, which we use.  From there I write the 
> variable to the CDR.
>
> For more information on the adaptive concept, please see 
> http://www.asterisk.org/node/48492.  There is also more detail from Tilghman 
> Lesher here: 
> http://www.mail-archive.com/asterisk-users@lists.digium.com/msg210573.html
>
> It's very elegant in it's design and it works like a champ- we use it in 
> production.
>
> If you are using Asterisk 1.4.*, you can use the the CDR's userfield. This is 
> an optional, user defined field that can store just about whatever data you 
> wish depending on the data type defined in the database.  You will have to 
> google around to find out more information on how to enable it, although I 
> believe that it's an option in the /etc/asterisk/cdr.conf configuration file 
> that you are using.
>
> Again, if you are using Asterisk 1.6.* I would strongly recommend that you 
> take advantage of the Adaptive CDR system.


I am using Asterisk 1.4.*

My cdr_mysql.conf has only the following:

[global]
hostname = localhost
dbname=asteriskcdrdb
password = amp109
user = asteriskuser
userfield=1
;port=3306
;sock=/tmp/mysql.sock
---

I could not much info on the net on this subject.

Thx
Sanjay

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Re: [asterisk-users] CDR: Add Dialed Number Identifierfield (DNID) field into MySQL

[Lee Archer]

Isn't the use of DNID separate to the userfield?  I'd like to have this
working also.

Lee

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alex
Balashov
Sent: 15 March 2010 08:34
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] CDR: Add Dialed Number Identifierfield
(DNID) field into MySQL

Use the userfield.

On 03/15/2010 04:25 AM, RSCL Mumbai wrote:

> Hi,
>
> I would like to see the DNID in my MySQL CDR logs.
>
> I have read one big thread in the Asterisk Developer List, but I could
> not figure out how to implement it ?
> Is there a simple step-by-step.
>
> Thx in advance.
>
> Vai
>


-- 
Alex Balashov - Principal
Evariste Systems LLC

Tel: +1 678-954-0670
Direct : +1 678-954-0671
Web: http://www.evaristesys.com/

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[asterisk-users] Increasing the dahdi chunk size with Sangoma cards

[Lee Archer]

Hi, does anyone run non HWEC Sangoma PRI cards with an increased dahdi
chunk size?  I tested it at 2ms and it seemed fine with no noticeable
loss in audio quality, and it reduced the interrupt processing to 50%.

Regards

Lee
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[asterisk-users] Error and call drops

[Lee Archer]

Hi, does anyone have an info into what could cause

[Nov 28 14:24:48] ERROR[11964] utils.c: write() returned error: Broken
pipe
[Nov 28 14:25:08] ERROR[12540] utils.c: write() returned error: Broken
pipe
[Nov 28 14:25:08] ERROR[12540] utils.c: write() returned error: Broken
pipe
[Nov 28 14:26:23] ERROR[13098] utils.c: write() returned error: Broken
pipe

Is it a write process or a problem with one of the scripts I am running?
I am seeing this over and over again and experience call drops on a
percentage of calls.

Thanks

Lee
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Re: [asterisk-users] ASTERISK and SNMP

[Lee Archer]

I use CentOS, and it works fairly well.   But I had to piece together info from 
several places.  I've tried it several different wants and this way worked, as 
long as asterisk is run as root.

 

Copy asterisk-mib.txt and digium-mib.txt from /doc to 
/usr/share/snmp/mibs/

 

mkdir /var/agentx

touch /var/agentx/master

 

My /etc/asterisk/res_snmp.conf

 

;

; Configuration file for res_snmp

;

 

[general]

; We run as a subagent per default -- to run as a full agent

; we must run as root (to be able to bind to port 161)

;subagent = yes

; SNMP must be explicitly enabled to be active

enabled = yes

 

My snmp.conf

 

rwcommunity private 127.0.0.1

rocommunity public

disk /

master agentx

agentXperms 0660 0550 root root

 

restart snmp and the /var/agentx/master should look like srw-rw 1 root root 
0 Nov 25 11:31 /var/agentx/master

 

restart asterisk manually and you see a net-snmp connect.

 

export MIBS=+ASTERISK-MIB

 

You should be able to to do a snmpwalk -v 2c -c public localhost asterisk

 

Regards

 

Lee

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of mickael ropars
Sent: 27 November 2009 11:58
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] ASTERISK and SNMP

 

Michal

please wait I found some issues in my con file

2009/11/27 mickael ropars 

I am running on Trixbox so my OS is Cent0S 5.4 and the Asterisk version is 
1.4.22-4

on asterisk side Snmp module is running:

> module load res_snmp.so
  == Parsing '/etc/asterisk/res_snmp.conf': Found
 Loading [Sub]Agent Module
 Loaded res_snmp.so => (SNMP [Sub]Agent for Asterisk)

see below my snmpd.conf file (I remove commented line for an easy reading)

regards

Mickael




###
# Access Control
###


# First, map the community name (COMMUNITY) into a security name
# (local and mynetwork, depending on where the request is coming
# from):

#   sec.name  source  community
com2sec local localhost   COMMUNITY
com2sec mynetwork NETWORK/24  COMMUNITY


rwcommunity local
rocommunity local


# Second, map the security names into group names:

#   sec.model  sec.name
group MyRWGroup v1 local
group MyRWGroup v2clocal
group MyRWGroup usmlocal
group MyROGroup v1 mynetwork
group MyROGroup v2cmynetwork
group MyROGroup usmmynetwork


# Third, create a view for us to let the groups have rights to:

#   incl/excl subtree  mask
view allincluded  .1   80


# Finally, grant the 2 groups access to the 1 view with different
# write permissions:

#context sec.model sec.level match  read   write  notif
access MyROGroup ""  any   noauthexact  allnone   none
access MyRWGroup ""  any   noauthexact  allallnone


###
# System contact information
#

syslocation Right here, right now.
syscontact Me 



###
# Process checks.
#
#  Make sure mountd is running
proc mountd

#  Make sure there are no more than 4 ntalkds running, but 0 is ok too.
proc ntalkd 4

#  Make sure at least one sendmail, but less than or equal to 10 are running.
proc sendmail 10 1


###
# Executables/scripts
#

# a simple hello world
exec echotest /bin/echo hello world

###
# disk checks
#

disk / 1


###
# load average checks
#

# Check for loads:
load 12 14 14


###
# Extensible sections.
#



###
# Pass through control.
#

###
# Subagent control


#

master agentx
agentXperms 0660 0550 nobody asterisk
SNMPD_FLAGS="${SNMPD_FLAGS} -x /var/agentx/master"
mibs +ASTERISK-MIB

###
# Further Information







2009/11/27 michal kalinowski 

What operating system do You have ? What asterisk version You compile ?
After install net-snmp do You recompile asterisk with res_snmp module ?

I'm used instruction from here
http://voxilla.com/2009/02/03/configuring-asterisk-snmp-support-1131
and everything work correctly.

BR,
Michał
W dniu 27 listopada 2009 11:18 użytkownik mickael ropars
 napisał:

> Hi Michal,
>
> thanks a lot for you quick answer I appreciate.
>

Re: [asterisk-users] odbc to ms-sql server

[Lee Archer]

If you are want CDR's to go to MS-SQL try cdr_tds.

Regards

Lee

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Neeraj
Chand
Sent: 06 November 2009 07:04
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] odbc to ms-sql server


 Gotcha! Missed libtool! :)

-Original Message-
From: Neeraj Chand 
Sent: Friday, 6 November 2009 6:43 PM
To: 'asterisk-users@lists.digium.com'
Subject: RE: odbc to ms-sql server

Hi all, 

I'm trying to set up an odbc connection to a ms-sql server from an
asterisk 1.6.1 install

My problem is that I cannot get asterisk to build func_odbc &
res_odbc.so

I installed yum -y install unixODBC unixODBC-devel libtool-ltdl
libtool-ltdl-devel

And then went on to reconfigure / recompile asterisk

after a ./configure --with-odbc=/usr/lib/

I get
###
checking for mandatory modules:  UNIXODBC... ok
configure: creating ./config.status


And then when I go to make menuselect;

[XXX]Res_odbc 

[XXX] func_odbc

[XXX] cdr_odbc

Can anyone help out with what I am missing? 

[I've gotten to a stage where tsql and isql connections to my sql db
work, however, getting odbc right is making me pull my hair out a bit]

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Re: [asterisk-users] Extra CDR fields

[Lee Archer]

Thanks, I had already tried these and no luck.  I will have to try a few
more things.

 

Regards

 

Lee

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny
Nicholas
Sent: 03 November 2009 16:51
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Extra CDR fields

 

Try this link

http://www.voip-info.org/wiki/view/Asterisk+func+cdr

 

YMMV - good luck

 



From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lee Archer
Sent: Tuesday, November 03, 2009 10:42 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Extra CDR fields

 

Do you have any info on multiple userfields as that's exactly what I
would be looking for?

 

Regards

 

Lee

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny
Nicholas
Sent: 03 November 2009 16:21
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Extra CDR fields

 

Depends on your Asterisk release and CDR type.  According to what I've
read, you can have multiple user fields.

 



From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lee Archer
Sent: Tuesday, November 03, 2009 10:09 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Extra CDR fields

 

Hi, is userfield the only extra CDR field that can be added or can
others?

Regards

Lee

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Re: [asterisk-users] Extra CDR fields

[Lee Archer]

I'm having to use 1.4 and the cdr_tds module.  I already use userfield for some 
data but need a second field that I can write some info to.  Can't move to 1.6 
just yet.

Regards

Lee

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Carlos Chavez
Sent: 03 November 2009 16:42
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Extra CDR fields

On Tue, 2009-11-03 at 16:09 +, Lee Archer wrote:
> Hi, is userfield the only extra CDR field that can be added or can 
> others?
> 
> Regards
> 
> Lee

With Asterisk 1.6.X there is adaptive CDR where you can add as many 
fields as you want to the CDR, even using a database through ODBC or Mysql.

--
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez Prats
Director de Tecnología
+52-55-91169161 ext 2001
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Re: [asterisk-users] Extra CDR fields

[Lee Archer]

Do you have any info on multiple userfields as that's exactly what I
would be looking for?

 

Regards

 

Lee

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny
Nicholas
Sent: 03 November 2009 16:21
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Extra CDR fields

 

Depends on your Asterisk release and CDR type.  According to what I've
read, you can have multiple user fields.

 



From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lee Archer
Sent: Tuesday, November 03, 2009 10:09 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Extra CDR fields

 

Hi, is userfield the only extra CDR field that can be added or can
others?

Regards

Lee

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[asterisk-users] Extra CDR fields

[Lee Archer]

Hi, is userfield the only extra CDR field that can be added or can
others?

Regards

Lee
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Re: [asterisk-users] SIP Headers

[Lee Archer]

SPA921 isn't an Aastra phone though is it?  I would expect the Linksys
manual to list some of the ones you can use.

 

Regards

 

Lee

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle
Dupuis
Sent: 19 October 2009 01:07
To: 'Asterisk Users List'
Subject: Re: [asterisk-users] SIP Headers

 

There is an admin manual you can download from Aastra..have you checked
there?  (Not the user manual)

 



From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo
Sent: Sunday, October 18, 2009 6:18 PM
To: Asterisk Users List
Subject: [asterisk-users] SIP Headers

Hi,

 

I was wondering where I can find a list of SIP Headers which can be used
for the SPA 921 to customise the ringtone and other features for each
call.

 

Here is an example taken from
http://www.voip-info.org/wiki/view/Asterisk+cmd+Page

 

exten => s,3,SIPAddHeader(answer-after=0)

 

Many thanks

Dan Journo

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Re: [asterisk-users] Asterisk Monitoring

[Lee Archer]

Zenoss has something that hits the manager port.  I run Asterisk 1.4 boxes and 
are using SNMP to monitor.  Asterisk 1.6 has a couple of extra SNMP OID’s that 
show the number of calls processed.  It’s a shame 1.4 doesn’t have this OID as 
it could be really useful.

 

Regards

 

Lee

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of rea...@gmail.com
Sent: 18 October 2009 17:39
To: Asterisk Users Mailing List - Non-Commercial Discussion; abc005
Subject: Re: [asterisk-users] Asterisk Monitoring

 

 

-- Sent from my Palm Unknown

 



Jai Rangi wrote:

Nagios has a plugin "check_sip" that can be used for this.


-Jai 




On Sat, Oct 17, 2009 at 5:30 PM, Dan Journo  
wrote:

Hello,

 

I was wondering if anyone has any insights on the best way to automatically 
monitor an asterisk box to check it is constantly available and processing 
calls.

 

Many thanks

Dan

 



See original image

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receive immediate remote IT support. Click the chat link below for support.
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Dan Journo
IT Business Consultant
Kesher Communications Ltd

Tel: 07957 233 599
Web: http://www.KesherCommunications.com  
Live Chat/Instant Support: Click Here 

  

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[asterisk-users] Aastra IP phone configuration generator

[Lee Archer]

For anyone who is interested I've recently created an Aastra IP Phone
config generator.  I don't know if one existed but thought I'd create it
anyways.  It can be found at http://www.lraweb.pwp.blueyonder.co.uk/.
If you have any problems or stuff you want adding then please contact at
the address listed on the web page.

Regards

Lee
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[asterisk-users] G729 codec problems

[Lee Archer]

I have a system that has had 5 G729 licenses for over a year and I've
come to install the v31 G729 codec from the Digium ftp server but it
won't see the license.  Does anyone know how to get around this problem?
It is registered and I do have newer systems running this v31 version of
the codec but in the license file the product line is different.  On the
older system it says Product: Digium-G729 but on the newer systems it
says Product: G.729 Codec.  I've tried Digium support but had no reply
and thought I'd try the list.

Regards

Lee
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RE: [asterisk-users] Problem with SuSe 10.0 and zaptel 1.2.17

[Lee Archer]

It was fixed in 1.2.17.1. 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir
Cohen
Sent: 26 April 2007 21:05
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Problem with SuSe 10.0 and zaptel 1.2.17

On Wed, Apr 25, 2007 at 08:57:37AM +0100, Lee Archer wrote:
> I installed zaptel 1.2.17 and shortly afterwards got a problem of 
> calls not clearing properly.  I ran dmesg which showed
> 
>   Unable to handle kernel NULL pointer dereference at virtual
address 009c
>   printing eip:
>   f8a79fa8
>   *pde = 
>   Oops:  [#1]
>   Modules linked in: zttranscode button battery ac ipv6 edd
wcte11xp zaptel crc_ccitt i2c_i801 i2c_core tg3 generic shpchp
pci_hotplug parport_pc lp parport dm_mod ext3 jbd sg fan thermal
processor 3w_ piix sd_mod scsi_mod ide_disk ide_core
>   CPU:0
>   EIP:0060:[]Tainted: G U VLI
>   EFLAGS: 00010082   (2.6.13-15.15-default)
>   EIP is at zt_chanandpseudo_ioctl+0xd28/0xf70 [zaptel]
>   eax:    ebx: f74403ac   ecx:    edx: 
>   esi: b723f2b0   edi: f749ca78   ebp: 0046   esp: f50b3e28
>   ds: 007b   es: 007b   ss: 0068
>   Process asterisk (pid: 5430, threadinfo=f50b2000 task=f7bbf060)
>   Stack: 462f0587  41a0d314  01ff 0001
0246 0001
>    f50b3f38  005b 0001
dfcf089c f50b3ebc
>  f50b3efc f61ae400 f6a0d3b4 005b 005b f6a0d314
 0001
>   Call Trace:
>[] generic_file_aio_write+0x58/0xc0
>[] ext3_file_write+0x1b/0x93 [ext3]
>[] do_sync_write+0xb6/0x110
>[] zt_ioctl+0x93/0x100 [zaptel]
>[] zt_ioctl+0x0/0x100 [zaptel]
>[] do_ioctl+0x4e/0x60
>[] vfs_ioctl+0x4f/0x1c0
>[] sys_ioctl+0x37/0x70
>[] sysenter_past_esp+0x54/0x79
>   Code: ff 89 f8 89 f1 e8 75 88 77 c7 31 ff c7 85 94 06 00 00 00
00 00 
> 00 e9 77 f4 ff ff 8b 4c 24 24 e9 e4 f8 ff ff 8b 04 95 20 0d aa f8 <8b>

> 80 9c 00 00 00 e8 5d a9 6c c7 8b 44 24 20 8b 04 85 20 0d aa
> 
> I've since installed zaptel 1.2.16 again and it's fine.  Is anyone
else getting this problem?

Not me, but others do. Try 1.2.17.1 .


-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
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[asterisk-users] Problem with SuSe 10.0 and zaptel 1.2.17

[Lee Archer]

I installed zaptel 1.2.17 and shortly afterwards got a problem of calls not 
clearing properly.  I ran dmesg which showed

Unable to handle kernel NULL pointer dereference at virtual address 
009c
printing eip:
f8a79fa8
*pde = 
Oops:  [#1]
Modules linked in: zttranscode button battery ac ipv6 edd wcte11xp 
zaptel crc_ccitt i2c_i801 i2c_core tg3 generic shpchp pci_hotplug parport_pc lp 
parport dm_mod ext3 jbd sg fan thermal processor 3w_ piix sd_mod scsi_mod 
ide_disk ide_core
CPU:0
EIP:0060:[]Tainted: G U VLI
EFLAGS: 00010082   (2.6.13-15.15-default)
EIP is at zt_chanandpseudo_ioctl+0xd28/0xf70 [zaptel]
eax:    ebx: f74403ac   ecx:    edx: 
esi: b723f2b0   edi: f749ca78   ebp: 0046   esp: f50b3e28
ds: 007b   es: 007b   ss: 0068
Process asterisk (pid: 5430, threadinfo=f50b2000 task=f7bbf060)
Stack: 462f0587  41a0d314  01ff 0001 0246 
0001
     f50b3f38  005b 0001 dfcf089c 
f50b3ebc
   f50b3efc f61ae400 f6a0d3b4 005b 005b f6a0d314  
0001
Call Trace:
 [] generic_file_aio_write+0x58/0xc0
 [] ext3_file_write+0x1b/0x93 [ext3]
 [] do_sync_write+0xb6/0x110
 [] zt_ioctl+0x93/0x100 [zaptel]
 [] zt_ioctl+0x0/0x100 [zaptel]
 [] do_ioctl+0x4e/0x60
 [] vfs_ioctl+0x4f/0x1c0
 [] sys_ioctl+0x37/0x70
 [] sysenter_past_esp+0x54/0x79
Code: ff 89 f8 89 f1 e8 75 88 77 c7 31 ff c7 85 94 06 00 00 00 00 00 00 
e9 77 f4 ff ff 8b 4c 24 24 e9 e4 f8 ff ff 8b 04 95 20 0d aa f8 <8b> 80 9c 00 00 
00 e8 5d a9 6c c7 8b 44 24 20 8b 04 85 20 0d aa 

I've since installed zaptel 1.2.16 again and it's fine.  Is anyone else getting 
this problem?

Regards

Lee
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RE: [asterisk-users] Asterisk -> Streaming Audio Bridge

[Lee Archer]

I used mpg123 to stream air traffic control as a MOH class but I also
found it didn't always work with the shoutcast servers. 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric
Germann
Sent: 27 February 2007 02:17
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] Asterisk -> Streaming Audio Bridge

Greetings,

Does anyone know of a tool that can act as a VoIP client and stream to a
streaming server such as shoutcast/icecast, etc.

I've got a client interested in doing basketball play by plays during
tourney season.  They have * in place now and the bandwidth to burn for
streaming out.  In the old world, I did an analog phone patch -> mixer
-> encoder -> streaming server.  What I'm thinking of is more along the
lines of a client that registers as a SIP/IAX client, answers the phone
and patches it to a streaming server.

Thoughts/suggestions?

Thanks

Eric

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RE: [asterisk-users] freepbx with ASTERISK 1.4

[Lee Archer]

I said what to do before.

http://freepbx.org/trac/ticket/1610 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Guillermo
Salas M.
Sent: 16 February 2007 14:51
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] freepbx with ASTERISK 1.4

On Fri, 2007-02-16 at 09:33 -0500, McGhee, Stefano wrote:
> > it's possible to configure freepbx 2.2 with asterisk 1.4?
> 
> Look here for the archives:
> 
> http://lists.digium.com/pipermail/asterisk-users/
> 
> Search for the subject "FreePBX 2.2.0 and Asterisk 1.4.0".
> 
> You'll find EXACTLY what you're looking for. :-)
> 


Look at:

http://article.gmane.org/gmane.comp.telephony.pbx.asterisk.amportal.user
/5377


Regards,


> Stefano
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--
Guillermo Salas M.
Telconet S.A.
Calle 15 y Avenida 24 Esq
Edificio Barre #2 Primer Piso
Telefono : +593 5 262 8071
Celular  : +593 9 985 5138
e-mail   : [EMAIL PROTECTED]
www  : http://www.manta.telconet.net
   http://www.telcocarrier.net

Linux User: 255902

Beat me, whip me, make me use Windows!

Please avoid sending me Word or PowerPoint attachments.
See http://www.fsf.org/philosophy/no-word-attachments.html

Please avoid the Top Posting, see
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RE: [asterisk-users] freepbx with ASTERISK 1.4

[Lee Archer]

Yes check the freepbx website, and in particular trac bug #1610.

Lee 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of younss
azzayani
Sent: 16 February 2007 11:23
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] freepbx with ASTERISK 1.4

Hi everybody,
it's possible to configure freepbx 2.2 with asterisk 1.4?

Have a nice day

Younss AZ
KASTERISK.COM
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RE: [asterisk-users] TE110P and HDLC problems

[Lee Archer]

I had this problem and in the end it appeared to be slot timing on the mobo.  I 
had to put the TE110P in the 1st slot - which happened to be a PCI-X slot.  
That was using a Supermicro motherboard too. 

Regards

Lee

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew 
Fredrickson
Sent: 25 January 2007 20:58
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] TE110P and HDLC problems

There was a recent driver fix that *might* help you.  It's not in an official 
1.x.x release yet, but if you check out 1.2 from svn, you should get the latest 
version of the driver with the fix.

Matthew Fredrickson

On Jan 25, 2007, at 9:15 AM, Marc Patino Gómez wrote:

> Hi!,
>
> this issue makes me crazy. I read a lot of docs, also * mailling list 
> and I try a lot of things  without success.
>
> Any help will be appreciated. Here is the info:
>
> Hardware:
> 
> Supermicro Server with motherboard X7DB8, chipset Intel 5000P, Xeon 
> 5050 Digium TE110P
>
> Software
> -
> Asterisk version 1.2.12.1
> Zaptel version 1.2.8
>
> /etc/zaptel.conf
>
> loadzone=es
> defaultzone=es
> span=1,1,0,ccs,hdb3,crc4
> bchan=1-15
> dchan=16
> bchan=17-31
>
> The dammed errors:
>
> Jan 25 14:11:52 NOTICE[4934]: chan_zap.c:8316 pri_dchannel: PRI got
> event: HDLC Bad FCS (8) on Primary D-channel of span 1 Jan 25 14:11:52 
> NOTICE[4934]: chan_zap.c:8316 pri_dchannel: PRI got
> event: HDLC Abort (6) on Primary D-channel of span 1 Jan 25 14:11:52 
> NOTICE[4934]: chan_zap.c:8316 pri_dchannel: PRI got
> event: HDLC Abort (6) on Primary D-channel of span 1 ...
>
> I tried the following without success:
>
> - Disable Hyper Threading.
> - Disable NIC's, RAID controller, usb, serial... to avoid shared IRQ, 
> so TE110P has his own IRQ as shows lspci -vb.
> - Also I tried with APIC and without APIC.
> ..
>
>
> These HDLC errors appear when I physically loop the E1 interface in 
> the Card and also appear, and more frequently, when I connect the E1 
> circuit (from the Telco) to the interface of the Card.
>
>
> Thanks a lot
>
> --  
> --- 
> -
>
> Marc Patino Gómez
> Dpto. Sistemas
>
> Claranet España. Servicios Internet
> C/General Almirante 2-28, Torres Cerdá
> 08014 Barcelona
> Tel. Información General: 902 884 633 Tel. Soporte Técnico: 902 884 622
> Fax: +34 93 445 19 20
> www.claranet.es
>
> Claranet Group: United Kingdom - Spain - France - Germany - Portugal -  
> Netherlands - USA
>
> --- 
> -
>
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> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
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RE: [asterisk-users] Music on Hold on IP Phones with FreePBX 2.2.0

[Lee Archer]

Have you tried the #freepbx IRC channel or the freepbx mailing list?

Regards

Lee 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt
Arnilo S. Baluyos (Mailing Lists)
Sent: 23 January 2007 01:57
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Music on Hold on IP Phones with FreePBX 2.2.0

Hello everyone,

We just installed a new Trixbox 2.0 server and updated FreePBX to 2.2.0
from 2.2.0rc3.

We are having some problems with regards to Music on Hold on IP phones.
When we press the "Hold" button, the caller doesn't hear the MOH sound.
This functionality used to work with the older [EMAIL PROTECTED]
installation on the same hardware and configuration.

However, we don't have any problems with softphones only on IP phones.

Is there anyone also having the same problem?

Best regards,
Matt

--
Stand before it and there is no beginning.
Follow it and there is no end.
Stay with the ancient Tao,
Move with the present.
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RE: [asterisk-users] 7 points of comparison Polycom 430/501 andAastra480i. Which one to choose ?

[Lee Archer]

Aren't Aastra due to release new phones and some form of
operator/reception addon?  The Aastra user/admin guides are a lot more
up2date that they used to be.  I'd like Aastra to add a GSM codec to
their phone and have a more regular firmware release schedule.  I agree
with the list below though that Polycom does have a better line up
currently, and especially point 7 - when rebooting the phone please
don't drop the network ports.

Regards

Lee

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michelle
Dupuis
Sent: 23 January 2007 02:44
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [asterisk-users] 7 points of comparison Polycom 430/501
andAastra480i. Which one to choose ?

Here are another $0.02

We too have put in a lot of polycoms and aastras.  I agree with a lot of
what you noted below...but there are two big strikes against aastra:

1.  Firmware bugs.  Even some basic functions of the 480i are
unusable/unstable due to firmware bugs.  The word from support is always
"wait for the next firmware"
2.  Poor documentation.  Their documentation is out of date and lacking
a LOT of critical functions.  (eg: Try to setup a hold button on the
wireless handset using a config file)

We're steering more customers towards polycom now.

MD

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of C F
Sent: Monday, January 22, 2007 9:36 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] 7 points of comparison Polycom 430/501
andAastra 480i. Which one to choose ?

With over 300 Polycoms, and around 80 Aastra 480i under my belt here is
my $0.02.
1. Sound quality, Polycom wins but the Aastra has excellent sound
quality as well.
2. Complete product line, Polycom wins.
3. Cordless Aastra, although it's not the best cordless.
4. Backlit, Aastra
5. PoE, Aastra
6. Speakerphone, they both have good speaker phones. Although in general
the answer to 1 goes here as well.
7. They both have 2 network ports, but I havnt' done any tests on the
speed, I did however notice that when restarting the phone, the Polycom
will not shut the network ports down, while the Aastra will.

On another note, in general the Polycoms give me less problems. The
Aastras are not yet that stable. See my next post to the list.

On 1/22/07, Bruce Reeves <[EMAIL PROTECTED]> wrote:
> Your list seems to lean heavily to the Aastra, while I choose the 
> Polycom
> 501/601 over the Aastra, I did like the unit I tested and the 
> cordless. In the end the fact that most of the people using the phones

> would use the speaker phone, Polycom and their history of conference 
> phones made the choice. We rolled 75 phones at one site and another 30

> now at remote locations. As far a a receptionist phone, we choose to 
> use a software operator panel instead of a phone that took up most of 
> the desk, there were initial concerns but the results have been 
> excellent. If you have not already done so grab a few people from 
> different parts of the office and have them give their 2 cents, it 
> will help to have their perspectives on the quality and feel of the
phones.
>
>
> On 1/22/07, Vikas <[EMAIL PROTECTED]> wrote:
> > I need to provide a 80 people office with VOIP.
> >
> > I want to commit to one vendor Polycom or Aastra. Price of the 
> > phones is not a factor in the decision. The quality of the phones is

> > the factor.
> >
> > Some of the features that I am evaluating on are: (arranged in order

> > of priority) 1. Sound quality 2. complete product line with 
> > conference phone and receptionist phone (not on Aastra) 3. cordless 
> > (not on 501/430) 4. backlit LCD (not on 501/430) 5. Inbuilt POE (not

> > on 501) 6. speaker phone 7. 2 network ports.
> >
> > Which one will you choose ?
> >
> > Vikas
> > ___
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> > asterisk-users mailing list
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> >
> http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
>
>
> --
> Bruce
> Nortex Networks
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>
>
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RE: [asterisk-users] chan_zap.c: Failed to read gains: Invalidargument

[Lee Archer]

So anyone else any ideas? 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir
Cohen
Sent: 05 January 2007 09:30
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] chan_zap.c: Failed to read gains:
Invalidargument

On Fri, Jan 05, 2007 at 10:53:17AM +0200, Tzafrir Cohen wrote:
> On Fri, Jan 05, 2007 at 07:47:15AM -0000, Lee Archer wrote:
> > I am running Asterisk 1.2.14 and Zaptel 1.2.12 and using a Digium 
> > TE110P card in E1 mode.  I've recently noticed in my logs the 
> > following
> > 
> > Jan  5 01:27:11 VERBOSE[22490] logger.c:  [chan_zap.so]Jan  5 
> > 01:27:11 VERBOSE[22490] logger.c:  [chan_zap.so] => (Zapata
Telephony w/PRI)
> > Jan  5 01:27:11 VERBOSE[22490] logger.c:   == Parsing
> > '/etc/asterisk/zapata.conf': Jan  5 01:27:11 VERBOSE[22490]
logger.c:
> > == Parsing '/etc/asterisk/zapata.conf': Found Jan  5 01:27:11 
> > DEBUG[22490] chan_zap.c: Failed to read gains: Invalid argument Jan

> > 5 01:27:11 DEBUG[22490] chan_zap.c: Failed to read gains: Invalid 
> > argument
> 
> This is a debug message and not even a warning message. I'm not sure 
> that this is something to worry about.

Sorry, my stupid misreading of the code. If this message was given,
ZT_SETGAINS will not be called.

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
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RE: [asterisk-users] chan_zap.c: Failed to read gains: Invalidargument

[Lee Archer]

Yes I get the same message after reload chan_zap.so

Regards

Lee 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir
Cohen
Sent: 05 January 2007 08:53
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] chan_zap.c: Failed to read gains:
Invalidargument

On Fri, Jan 05, 2007 at 07:47:15AM -, Lee Archer wrote:
> I am running Asterisk 1.2.14 and Zaptel 1.2.12 and using a Digium 
> TE110P card in E1 mode.  I've recently noticed in my logs the 
> following
> 
> Jan  5 01:27:11 VERBOSE[22490] logger.c:  [chan_zap.so]Jan  5 01:27:11

> VERBOSE[22490] logger.c:  [chan_zap.so] => (Zapata Telephony w/PRI)
> Jan  5 01:27:11 VERBOSE[22490] logger.c:   == Parsing
> '/etc/asterisk/zapata.conf': Jan  5 01:27:11 VERBOSE[22490] logger.c:
> == Parsing '/etc/asterisk/zapata.conf': Found Jan  5 01:27:11 
> DEBUG[22490] chan_zap.c: Failed to read gains: Invalid argument Jan  5

> 01:27:11 DEBUG[22490] chan_zap.c: Failed to read gains: Invalid 
> argument

This is a debug message and not even a warning message. I'm not sure
that this is something to worry about.

The code there tries to first read the gains and set the gains based on
them. The return value from the ioctl that sets the gains does not seem
to be checked in several code pathes, though. So it may actually fail
silently.

Do you get the same debug messages on 'reload chan_zap.so' ?

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
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RE: [asterisk-users] chan_zap.c: Failed to read gains: Invalidargument

[Lee Archer]

Sorry I should have stated that I've tried +x, -x, x.y and x and I still
get the same.

Regards

Lee 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric
"ManxPower" Wieling
Sent: 05 January 2007 08:10
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] chan_zap.c: Failed to read gains:
Invalidargument

Lee Archer wrote:
> I am running Asterisk 1.2.14 and Zaptel 1.2.12 and using a Digium 
> TE110P card in E1 mode.  I've recently noticed in my logs the 
> following
> 
> Jan  5 01:27:11 VERBOSE[22490] logger.c:  [chan_zap.so]Jan  5 01:27:11

> VERBOSE[22490] logger.c:  [chan_zap.so] => (Zapata Telephony w/PRI)
> 
> Jan  5 01:27:11 VERBOSE[22490] logger.c:   == Parsing 
> '/etc/asterisk/zapata.conf': Jan  5 01:27:11 VERBOSE[22490] logger.c:

> == Parsing '/etc/asterisk/zapata.conf': Found
> 
> Jan  5 01:27:11 DEBUG[22490] chan_zap.c: Failed to read gains: Invalid

> argument Jan  5 01:27:11 DEBUG[22490] chan_zap.c: Failed to read 
> gains: Invalid

> 
> Which seems to suggest that I've done something wrong with the rx and 
> txgain option in /etc/asterisk/zapata.conf.  But these haven't been 
> changed in 18 months and still say
> 
> ; You may also set the default receive and transmit gains (in dB) ; 
> rxgain=4.0 txgain=0.0
> 
> Have I done something wrong or has something changed?

Don't use fractional gains.  i.e. use rxgain=4 and txgain=0

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[asterisk-users] chan_zap.c: Failed to read gains: Invalid argument

[Lee Archer]

I am running Asterisk 1.2.14 and Zaptel 1.2.12 and using a Digium TE110P
card in E1 mode.  I've recently noticed in my logs the following

Jan  5 01:27:11 VERBOSE[22490] logger.c:  [chan_zap.so]Jan  5 01:27:11
VERBOSE[22490] logger.c:  [chan_zap.so] => (Zapata Telephony w/PRI)
Jan  5 01:27:11 VERBOSE[22490] logger.c:   == Parsing
'/etc/asterisk/zapata.conf': Jan  5 01:27:11 VERBOSE[22490] logger.c:
== Parsing '/etc/asterisk/zapata.conf': Found
Jan  5 01:27:11 DEBUG[22490] chan_zap.c: Failed to read gains: Invalid
argument
Jan  5 01:27:11 DEBUG[22490] chan_zap.c: Failed to read gains: Invalid
argument
Jan  5 01:27:11 DEBUG[22490] chan_zap.c: Updated conferencing on 1, with
0 conference users
Jan  5 01:27:11 VERBOSE[22490] logger.c: -- Registered channel 1,
PRI Signalling signalling
Jan  5 01:27:11 DEBUG[22490] chan_zap.c: Failed to read gains: Invalid
argument
Jan  5 01:27:11 DEBUG[22490] chan_zap.c: Failed to read gains: Invalid
argument
Jan  5 01:27:11 DEBUG[22490] chan_zap.c: Updated conferencing on 2, with
0 conference users
Jan  5 01:27:11 VERBOSE[22490] logger.c: -- Registered channel 2,
PRI Signalling signalling
Jan  5 01:27:11 DEBUG[22490] chan_zap.c: Failed to read gains: Invalid
argument
Jan  5 01:27:11 DEBUG[22490] chan_zap.c: Failed to read gains: Invalid
argument
Jan  5 01:27:11 DEBUG[22490] chan_zap.c: Updated conferencing on 3, with
0 conference users
Jan  5 01:27:11 VERBOSE[22490] logger.c: -- Registered channel 3,
PRI Signalling signalling
Jan  5 01:27:11 DEBUG[22490] chan_zap.c: Failed to read gains: Invalid
argument
Jan  5 01:27:11 DEBUG[22490] chan_zap.c: Failed to read gains: Invalid
argument
Jan  5 01:27:11 DEBUG[22490] chan_zap.c: Updated conferencing on 4, with
0 conference users
Jan  5 01:27:11 VERBOSE[22490] logger.c: -- Registered channel 4,
PRI Signalling signalling
Jan  5 01:27:11 DEBUG[22490] chan_zap.c: Failed to read gains: Invalid
argument
Jan  5 01:27:11 DEBUG[22490] chan_zap.c: Failed to read gains: Invalid
argument
Jan  5 01:27:11 DEBUG[22490] chan_zap.c: Updated conferencing on 5, with
0 conference users
Jan  5 01:27:11 VERBOSE[22490] logger.c: -- Registered channel 5,
PRI Signalling signalling
Jan  5 01:27:11 DEBUG[22490] chan_zap.c: Failed to read gains: Invalid
argument
Jan  5 01:27:11 DEBUG[22490] chan_zap.c: Failed to read gains: Invalid
argument
Jan  5 01:27:11 DEBUG[22490] chan_zap.c: Updated conferencing on 6, with
0 conference users
Jan  5 01:27:11 VERBOSE[22490] logger.c: -- Registered channel 6,
PRI Signalling signalling
Jan  5 01:27:11 DEBUG[22490] chan_zap.c: Failed to read gains: Invalid
argument
Jan  5 01:27:11 DEBUG[22490] chan_zap.c: Failed to read gains: Invalid
argument
Jan  5 01:27:11 DEBUG[22490] chan_zap.c: Updated conferencing on 7, with
0 conference users
Jan  5 01:27:11 VERBOSE[22490] logger.c: -- Registered channel 7,
PRI Signalling signalling
Jan  5 01:27:11 DEBUG[22490] chan_zap.c: Failed to read gains: Invalid
argument
Jan  5 01:27:11 DEBUG[22490] chan_zap.c: Failed to read gains: Invalid
argument
Jan  5 01:27:11 DEBUG[22490] chan_zap.c: Updated conferencing on 8, with
0 conference users
Jan  5 01:27:11 VERBOSE[22490] logger.c: -- Registered channel 8,
PRI Signalling signalling
Jan  5 01:27:11 VERBOSE[22490] logger.c: -- Automatically generated
pseudo channel
Jan  5 01:27:11 VERBOSE[22490] logger.c:   == Starting D-Channel on span
1

Which seems to suggest that I've done something wrong with the rx and
txgain option in /etc/asterisk/zapata.conf.  But these haven't been
changed in 18 months and still say

; You may also set the default receive and transmit gains (in dB)
;
rxgain=4.0
txgain=0.0

Have I done something wrong or has something changed?

Thanks

Lee
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[asterisk-users] IAX trunk problem

[Lee Archer]

> I wonder if anyone can help me with this.  I have 4 sites running
> Asterisk and these are linked via IAX trunks and ADSL lines.  Calls
> coming into any of these sites are received locally and forwarded to a
> central operator.  E.g.  Call comes in on site A and is forwarded to
> the operator on site B.  99 out of 100 times the operator will send
> the call back to someone at the site from where it came but site B's
> Asterisk server seems to be staying in the loop.  E.g. A > B > A.
> I've had a look and can't see anything obvious as I had assumed that
> Asterisk would pass the call off.  I've tried notransfer on the trunks
> but site B's Asterisk server doesn't seem to be joining the endpoints
> and staying in the loop and therefore the call is going over the
> trunks twice.
> 
> Thanks
> 
> Lee
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[asterisk-users] IAX trunk problem

[Lee Archer]

I wonder if anyone can help me with this.  I have 4 sites running
asterisk and calls coming into any of these sites are received locally
and forwarded to a central operator.  E.g.  Call comes in on site A and
is forwarded to the operator on site B.  99/100 the operator will send
the call back to the site from where it came but site B's Asterisk
server seems to be staying in the loop.  E.g. A > B > A.  I've had a
look and can't see anything obvious as I had assumed that asterisk would
pass the call off.

Thanks

Lee
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RE: [asterisk-users] Queue forks asterisk and then leaves theextraprocesses lying around

[Lee Archer]

Hi, have a look at http://www.freepbx.org/trac/ticket/1174 it's
currently in the bug list. 

Regards

Lee

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nigel
Roberts
Sent: 08 November 2006 09:14
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Queue forks asterisk and then leaves
theextraprocesses lying around

Hi Lee,

On Wed, 08 Nov 2006 at 09:00:27 -0000, Lee Archer wrote:

> Are you using freePBX by any chance? 

Yes, version 2.1.1.

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RE: [asterisk-users] Queue forks asterisk and then leaves the extraprocesses lying around

[Lee Archer]

Are you using freePBX by any chance? 

Regards

Lee

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nigel
Roberts
Sent: 08 November 2006 08:55
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Queue forks asterisk and then leaves the
extraprocesses lying around

Hi,

I have a problem with Queue where by a call comes in to the queue and if
all the phones are busy and the queue reaches the timeout, it will fork
a process and leave it sitting there before going off to the next step
in the dial plan and continuing normally. This doesn't cause any
problems except for I assume that it will eventually use up all the
memory on the machine and it messes with my process monitoring.

It doesn't seem to matter what I have as the next step after the Queue
command and it happens only sometimes. It seems like it might even be a
timing issue given that it's less likely to happen if any one of the
phones ring.

The new asterisk processes that get started up look like they think
they're new asterisk instances or though they don't actually do anything
or interfere with the first asterisk instance.

Has anyone had any problems like this? Am I doing something wrong?

The appropriate part of my dial plan looks like this:

exten => 101,1,Answer
exten => 101,n,GotoIf($["${CONTEXT}"="from-internal"]?USERCID:SETCID)
exten => 101,n(USERCID),Macro(user-callerid,)
exten => 101,n(SETCID),Set(CALLERID(name)=${CALLERIDNAME})
exten =>
101,n,Set(MONITOR_FILENAME=/var/spool/asterisk/monitor/q${EXTEN}-${TIMES
TAMP}-${UNIQUEID})
exten => 101,n,Queue(101|tr|||30)
exten => 101,n,Goto(ext-local,83,1)
exten => 101*,1,Macro(agent-add,101,)
exten => 101**,1,Macro(agent-del,101,101)

and from queues.conf

[101]
wrapuptime=0
timeout=15
strategy=ringall
retry=5
queue-youarenext=
queue-thereare=
queue-thankyou=queue-thankyou
queue-callswaiting=
music=default
monitor-join=yes
monitor-format=
member=Local/[EMAIL PROTECTED],0
member=Local/[EMAIL PROTECTED],0
maxlen=2
leavewhenempty=no
joinempty=Yes
context=
announce-holdtime=no
announce-frequency=0

and some logs to show what I mean by the new asterisk process thinking
that it is actually a new asterisk.

-- snip --

Nov  8 21:44:38 DEBUG[25896] channel.c: Hanging up channel
'Local/[EMAIL PROTECTED],2'
Nov  8 21:44:38 DEBUG[25627] devicestate.c: Changing state for
Local/[EMAIL PROTECTED] - state 0 (Unknown) Nov  8 21:44:38 DEBUG[25904]
app_queue.c: Device 'Local/[EMAIL PROTECTED]' changed to state '0'
(Unknown) Nov  8 21:44:38 DEBUG[25897] cdr_addon_mysql.c: cdr_mysql:
inserting a CDR record.
Nov  8 21:44:38 DEBUG[25897] cdr_addon_mysql.c: cdr_mysql: SQL command
as follows: INSERT INTO cdr
(calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,dura
tion,billsec,disposition,amaflags,accountcode,uniqueid) VALUES
('2006-11-08 21:44:38','49761450','49761450','83','from-internal',
'Local/[EMAIL PROTECTED],2','','AGI','recordingcheck|20061108-214438
|1162975478.49',0,0,'NO ANSWER',3,'','1162975478.49') Nov  8 21:44:38
DEBUG[25897] channel.c: Hanging up channel
'Local/[EMAIL PROTECTED],2'
Nov  8 21:44:38 DEBUG[25627] devicestate.c: Changing state for
Local/[EMAIL PROTECTED] - state 0 (Unknown) Nov  8 21:44:38 DEBUG[25905]
app_queue.c: Device 'Local/[EMAIL PROTECTED]' changed to state '0'
(Unknown)
Nov  8 21:44:38 VERBOSE[25902] logger.c:   == Parsing
'/etc/asterisk/extconfig.conf': Nov  8 21:44:38 DEBUG[25902]
config.c:Parsing /etc/asterisk/extconfig.conf
Nov  8 21:44:38 VERBOSE[25902] logger.c:   == Parsing
'/etc/asterisk/extconfig.conf': Found
Nov  8 21:44:38 VERBOSE[25902] logger.c:   == Parsing
'/etc/asterisk/manager.conf': Nov  8 21:44:38 DEBUG[25902] config.c:
Parsing /etc/asterisk/manager.conf

... lots of asterisk start up logs ...

-- snip --

Regards,
Nigel

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[asterisk-users] Manager interface

[Lee Archer]

Title: Manager interface






This has probably been discussed before but I need to do a screen pop and I'm looking for ways to do it.  I am assuming I need to use the manager interface, which is ok cos I'm using that for calling out but I'm not quite what to pick up on.

Regards


Lee


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RE: [asterisk-users] can no more compile zaptel !!!

[Lee Archer]

http://bugs.digium.com/view.php?id=7536

Lee 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: 17 July 2006 15:25
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] can no more compile zaptel !!!

Hi all,
I was refreshing a running asterisk with last versions.

I am no more able to compile zaptlel package; make hung on vpm450 I saw
it was introduced last 7/7/2006
(http://ftp.digium.com/pub/telephony/zaptel/releases/ChangeLog-1.2.7)

I don't know which is the purpose of this driver, but obviously
something is missing im my box.

first lines of error output

/usr/src/zaptel-1.2/vpm450m.c:34:20: error: octdef.h: No such file or
directory
/usr/src/zaptel-1.2/vpm450m.c:36:36: error: apilib/octapi_largmath.h: No
such file or directory
/usr/src/zaptel-1.2/vpm450m.c:38:40: error:
oct6100api/oct6100_defines.h:
No such file or directory
/usr/src/zaptel-1.2/vpm450m.c:39:39: error: oct6100api/oct6100_errors.h:
No such file or directory
/usr/src/zaptel-1.2/vpm450m.c:40:38: error: oct6100api/oct6100_apiud.h:
No such file or directory
/usr/src/zaptel-1.2/vpm450m.c:42:33: error: apilib/octapi_llman.h: No
such file or directory
/usr/src/zaptel-1.2/vpm450m.c:44:41: error:
oct6100api/oct6100_tlv_inst.h:
No such file or directory
/usr/src/zaptel-1.2/vpm450m.c:45:47: error:
oct6100api/oct6100_chip_open_inst.h: No such file or directory
/usr/src/zaptel-1.2/vpm450m.c:46:48: error:
oct6100api/oct6100_chip_stats_inst.h: No such file or directory
/usr/src/zaptel-1.2/vpm450m.c:47:48: error:
oct6100api/oct6100_interrupts_inst.h: No such file or directory
/usr/src/zaptel-1.2/vpm450m.c:48:45: error:
oct6100api/oct6100_channel_inst.h: No such file or directory
/usr/src/zaptel-1.2/vpm450m.c:49:50: error:
oct6100api/oct6100_remote_debug_inst.h: No such file or directory
/usr/src/zaptel-1.2/vpm450m.c:50:43: error:
oct6100api/oct6100_debug_inst.h: No such file or directory
/usr/src/zaptel-1.2/vpm450m.c:51:41: error:
oct6100api/oct6100_api_inst.h:
No such file or directory
/usr/src/zaptel-1.2/vpm450m.c:52:48: error:
oct6100api/oct6100_adpcm_chan_inst.h: No such file or directory
/usr/src/zaptel-1.2/vpm450m.c:54:47: error:
oct6100api/oct6100_interrupts_pub.h: No such file or directory
/usr/src/zaptel-1.2/vpm450m.c:55:46: error:
oct6100api/oct6100_chip_open_pub.h: No such file or directory
/usr/src/zaptel-1.2/vpm450m.c:56:44: error:
oct6100api/oct6100_channel_pub.h: No such file or directory
/usr/src/zaptel-1.2/vpm450m.c:57:47: error:
oct6100api/oct6100_adpcm_chan_pub.h: No such file or directory
/usr/src/zaptel-1.2/vpm450m.c:59:36: error: oct6100_chip_open_priv.h: No
such file or directory
/usr/src/zaptel-1.2/vpm450m.c:60:40: error:
oct6100_miscellaneous_priv.h:
No such file or directory
/usr/src/zaptel-1.2/vpm450m.c:61:33: error: oct6100_memory_priv.h: No
such file or directory
/usr/src/zaptel-1.2/vpm450m.c:62:31: error: oct6100_tsst_priv.h: No such
file or directory
/usr/src/zaptel-1.2/vpm450m.c:63:34: error: oct6100_channel_priv.h: No
such file or directory
/usr/src/zaptel-1.2/vpm450m.c:64:37: error: oct6100_adpcm_chan_priv.h:
No such file or directory

Actually I have no one of these files.
Is it a svn problem ?

svn checkout http://svn.digium.com/svn/zaptel/branches/1.2 zaptel-1.2

thanks in advance,
Andrea

Chi ricevesse questa mail per errore e' gentilmente pregato di
cancellarla.

Visitate il sito http://www.frameweb.it

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RE: [Asterisk-Users] Compiling mpg123 under 1.2.9.1 / Ubuntu 6.06

[Lee Archer]

Try make on its own and read what it says.  You probably want make linux

Regards

Lee 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Marc
Rohlfing
Sent: 13 June 2006 12:09
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Compiling mpg123 under 1.2.9.1 / Ubuntu 6.06

  Hi,

I made the mistake of upgrading both my Linux box (to Ubuntu 6.06) and
Asterisk (to 1.2.9.1) at the same time. Now, when trying to compile
mpg123 - using the tried and true "make mpg123" -, the build fails with
an error

make[3]: Entering directory `/usr/src/asterisk-1.2.9.1/mpg123-0.59r'
make[3]: *** No rule to make target `\
', needed by `mpg123'.  Stop.

Maybe there's someone out there more versed in Linux who has an idea
what might have gone wrong. Thanks!

  Marc

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[Asterisk-Users] Duplicate asterisk processes

[Lee Archer]

Title: Duplicate asterisk processes






I'm still getting duplicate process but the results of gdb are different.  Can anyone shed any light onto what is causing this?

(gdb) info threads

  1 Thread 1091845040 (LWP 31287)  0xe410 in __kernel_vsyscall ()

(gdb) thread apply all bt


Thread 1 (Thread 1091845040 (LWP 31287)):

#0  0xe410 in __kernel_vsyscall ()

#1  0x4004f13e in __lll_mutex_lock_wait () from /lib/tls/libpthread.so.0

#2  0x4004be41 in _L_mutex_lock_191 () from /lib/tls/libpthread.so.0

#3  0x40224600 in _IO_wide_data_2 () from /lib/tls/libc.so.6

#4  0x4112d1a0 in ?? ()

#5  0x4112d1a0 in ?? ()

#6  0x4112d258 in ?? ()

#7  0x4112d234 in ?? ()

#8  0x08061555 in ast_get_channel_tech (name=0x4112d1a0 "SIP") at lock.h:601

#9  0x080c9013 in ast_device_state (device=0x7a37 ) at devicestate.c:118

Previous frame inner to this frame (corrupt stack?)

#0  0xe410 in __kernel_vsyscall ()


Regards


Lee


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RE: [Asterisk-Users] Multiple processes

[Lee Archer]

I don't have any ODBC CDR stuff.  I unloaded the ODBC Asterisk modules
and the problem occurred again about an hour later.

Lee 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rodney G.
McDuff
Sent: 01 June 2006 01:32
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Multiple processes

Temporarily turn off your ODBC CDR stuff and see if the problem is still
there.

Lee Archer wrote:
>
> Can someone shed any light on the following.  I have 2 identical 
> systems, 1 of which seems to spawn multiple processes which have to be

> killed manually.  It recently kicked up 2 so I ran gdb on them and 
> this is the thread output.  I current use FreePBX with these systems.
>
> 1st extra process
>
> (gdb) info thread
>   1 Thread 1095261104 (LWP 14213)  0xe410 in __kernel_vsyscall ()
> (gdb) thread apply all bt
>
> Thread 1 (Thread 1095261104 (LWP 14213)):
> #0  0xe410 in __kernel_vsyscall ()
> #1  0x4004f13e in __lll_mutex_lock_wait () from 
> /lib/tls/libpthread.so.0
> #2  0x4004be41 in _L_mutex_lock_191 () from /lib/tls/libpthread.so.0
> #3  0x0001 in ?? ()
> #4  0x40e16778 in ?? () from /usr/lib/asterisk/modules/cdr_odbc.so
> #5  0x40e16818 in __dso_handle () from 
> /usr/lib/asterisk/modules/cdr_odbc.so
> #6  0x0002 in ?? ()
> #7  0x in ?? ()
> #8  0x080a20b7 in ast_cdr_unregister (name=0x40e1455c "ODBC") at
> lock.h:592
> #9  0x40e13299 in odbc_unload_module () at cdr_odbc.c:240 #10 
> 0x40e13978 in reload () at cdr_odbc.c:465
> #11 0x0805be32 in ast_module_reload (name=0x0) at loader.c:257
> #12 0x080b4623 in hup_handler (num=-4) at asterisk.c:754
> #13 
> #14 0xe410 in __kernel_vsyscall ()
> #15 0x401afd99 in sched_setscheduler () from /lib/tls/libc.so.6
> #16 0x080b4743 in ast_set_priority (pri=0) at asterisk.c:803
> #17 0x40445ee8 in agi_exec_full (chan=0x82782b0, data= out>, enhanced=0, dead=0) at res_agi.c:300
> #18 0x0808d521 in pbx_extension_helper (c=0x82782b0, con= optimized out>, context=0x8278400 "macro-record-enable",
>
> exten=0x82784f4 "s", priority=4, label=0x0, callerid=0x8159f38 
> "0163861", action=1) at pbx.c:553
> #19 0x40c44851 in macro_exec (chan=0x82782b0, data=0x4147c768) at 
> app_macro.c:210 #20 0x0808d521 in pbx_extension_helper (c=0x82782b0, 
> con=, context=0x8278400 "macro-record-enable",
>
> exten=0x82784f4 "s", priority=7, label=0x0, callerid=0x8159f38 
> "0163861", action=1) at pbx.c:553
> #21 0x40c44851 in macro_exec (chan=0x82782b0, data=0x41482fd8) at 
> app_macro.c:210
> #22 0x0808d521 in pbx_extension_helper (c=0x82782b0, con= optimized out>, context=0x8278400 "macro-record-enable",
>
> exten=0x82784f4 "s", priority=1, label=0x0, callerid=0x8159f38 
> "0163861", action=1) at pbx.c:553
> #23 0x0808ead4 in __ast_pbx_run (c=0x82782b0) at pbx.c:2227
> #24 0x0808f6cc in pbx_thread (data=0x0) at pbx.c:2514
> #25 0x4004a297 in start_thread () from /lib/tls/libpthread.so.0
> #26 0x401c737e in clone () from /lib/tls/libc.so.6
> #27 0x41485bb0 in ?? ()
> #0  0xe410 in __kernel_vsyscall ()
>
> 2nd extra process
>
> (gdb) info thread
>   1 Thread 1096059824 (LWP 14214)  0xe410 in ?? ()
> (gdb) thread apply all bt
>
> Thread 1 (Thread 1096059824 (LWP 14214)):
> #0  0xe410 in ?? ()
> #1  0x41533594 in ?? ()
> #2  0x0002 in ?? ()
> #3  0x in ?? ()
> #4  0x4004f13e in __lll_mutex_lock_wait () from 
> /lib/tls/libpthread.so.0
> #5  0x4004be41 in _L_mutex_lock_191 () from /lib/tls/libpthread.so.0
> #6  0x0001 in ?? ()
> #7  0x40e16778 in ?? () from /usr/lib/asterisk/modules/cdr_odbc.so
> #8  0x40e16818 in __dso_handle () from 
> /usr/lib/asterisk/modules/cdr_odbc.so
> #9  0x0002 in ?? ()
> #10 0x in ?? ()
> #11 0x080a20b7 in ast_cdr_unregister (name=0x40e1455c "ODBC") at
> lock.h:592
> #12 0x40e13299 in odbc_unload_module () at cdr_odbc.c:240
> #13 0x40e13978 in reload () at cdr_odbc.c:465
> #14 0x0805be32 in ast_module_reload (name=0x0) at loader.c:257
> #15 0x080b4623 in hup_handler (num=-4) at asterisk.c:754
> #16 
> #17 0xe410 in ?? ()
> #0  0xe410 in ?? ()
>
> Regards
>
> Lee
>
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RE: [Asterisk-Users] Multiple processes

[Lee Archer]

It's only been a problem since I updated to Asterisk 1.2 a 
few months ago.  It was a fresh install of OS, Asterisk, FreePBX and other 
scripts.  I've recently just updating FreePBX but the problem hasn't 
gone.
 
Regards
 
Lee


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Anthony 
CennamiSent: 31 May 2006 16:34To: Asterisk Users Mailing 
List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] 
Multiple processes
Were these upgrades or fresh installs?  Earlier versions of 
asterisk ran with multiple threads.  If you upgraded asterisk versions, but 
did not upgrade the associated startup scripts, multiple processes would still 
be spawned even if not appropriate. Anthony
On 5/31/06, Lee 
Archer <[EMAIL PROTECTED]> 
wrote:

  
  
  Can someone shed any light on the following.  
  I have 2 identical systems, 1 of which seems to spawn multiple processes which 
  have to be killed manually.  It recently kicked up 2 so I ran gdb on them 
  and this is the thread output.  I current use FreePBX with these systems. 
  
  1st extra process 
  (gdb) info thread   1 Thread 1095261104 (LWP 14213)  0xe410 in 
  __kernel_vsyscall () (gdb) thread apply all 
  bt 
  Thread 1 (Thread 1095261104 (LWP 14213)): 
  #0  0xe410 in __kernel_vsyscall () 
  #1  0x4004f13e in __lll_mutex_lock_wait () 
  from /lib/tls/libpthread.so.0 #2  
  0x4004be41 in _L_mutex_lock_191 () from /lib/tls/libpthread.so.0 
  #3  0x0001 in ?? () #4  0x40e16778 in ?? () from 
  /usr/lib/asterisk/modules/cdr_odbc.so #5  0x40e16818 in __dso_handle () from 
  /usr/lib/asterisk/modules/cdr_odbc.so #6  0x0002 in ?? () #7  0x in ?? () #8  0x080a20b7 in ast_cdr_unregister (name=0x40e1455c "ODBC") at 
  lock.h:592 #9  0x40e13299 in 
  odbc_unload_module () at cdr_odbc.c:240 #10 
  0x40e13978 in reload () at cdr_odbc.c:465 #11 0x0805be32 in ast_module_reload (name=0x0) at loader.c:257 
  #12 0x080b4623 in hup_handler (num=-4) at 
  asterisk.c:754 #13  #14 0xe410 in 
  __kernel_vsyscall () #15 0x401afd99 in 
  sched_setscheduler () from /lib/tls/libc.so.6 #16 0x080b4743 in ast_set_priority (pri=0) at asterisk.c:803 
  #17 0x40445ee8 in agi_exec_full (chan=0x82782b0, 
  data="" optimized out>, enhanced=0, dead=0) at res_agi.c:300 
  #18 0x0808d521 in pbx_extension_helper 
  (c=0x82782b0, con=, context=0x8278400 
  "macro-record-enable",
      exten=0x82784f4 "s", priority=4, 
  label=0x0, callerid=0x8159f38 "0163861", action="" at pbx.c:553 
  #19 0x40c44851 in macro_exec (chan=0x82782b0, 
  data="" at app_macro.c:210 #20 
  0x0808d521 in pbx_extension_helper (c=0x82782b0, con=, context=0x8278400 "macro-record-enable",
      exten=0x82784f4 "s", priority=7, 
  label=0x0, callerid=0x8159f38 "0163861", action="" at pbx.c:553 
  #21 0x40c44851 in macro_exec (chan=0x82782b0, 
  data="" at app_macro.c:210 #22 
  0x0808d521 in pbx_extension_helper (c=0x82782b0, con=, context=0x8278400 "macro-record-enable",
      exten=0x82784f4 "s", priority=1, 
  label=0x0, callerid=0x8159f38 "0163861", action="" at pbx.c:553 
  #23 0x0808ead4 in __ast_pbx_run (c=0x82782b0) at 
  pbx.c:2227 #24 0x0808f6cc in pbx_thread 
  (data="" at pbx.c:2514 #25 0x4004a297 in 
  start_thread () from /lib/tls/libpthread.so.0 #26 0x401c737e in clone () from /lib/tls/libc.so.6 #27 0x41485bb0 in ?? () #0  0xe410 in __kernel_vsyscall () 
  2nd extra process 
  (gdb) info thread   1 Thread 1096059824 (LWP 14214)  0xe410 in ?? () 
  (gdb) thread apply all bt 
  Thread 1 (Thread 1096059824 (LWP 14214)): 
  #0  0xe410 in ?? () #1  0x41533594 in ?? () #2  0x0002 in ?? () #3  0x in ?? () #4  0x4004f13e in __lll_mutex_lock_wait () from 
  /lib/tls/libpthread.so.0 #5  
  0x4004be41 in _L_mutex_lock_191 () from /lib/tls/libpthread.so.0 
  #6  0x0001 in ?? () #7  0x40e16778 in ?? () from 
  /usr/lib/asterisk/modules/cdr_odbc.so #8  0x40e16818 in __dso_handle () from 
  /usr/lib/asterisk/modules/cdr_odbc.so #9  0x0002 in ?? () #10 
  0x in ?? () #11 0x080a20b7 in 
  ast_cdr_unregister (name=0x40e1455c "ODBC") at lock.h:592 #12 0x40e13299 in odbc_unload_module () at 
  cdr_odbc.c:240 #13 0x40e13978 in reload () 
  at cdr_odbc.c:465 #14 0x0805be32 in 
  ast_module_reload (name=0x0) at loader.c:257 #15 0x080b4623 in hup_handler (num=-4) at asterisk.c:754 
  #16  #17 0xe410 in ?? () #0  0xe410 in ?? () 
  Regards 
  Lee 
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RE: [Asterisk-Users] Multiple processes

[Lee Archer]

I get NPTL 2.3.5.  It's only on 1 box and after a while there are so
many that it stops calls.  On the other box and the other test boxes I
have its only 1 asterisk process.

Regards

Lee 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Moises
Silva
Sent: 31 May 2006 14:56
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Multiple processes

check the output of "getconf GNU_LIBPTHREAD_VERSION"

If you see as output "linuxthreads-version"

then is highly probably that those "extra" processes that you say, are
nothing more than some of the threads Asterisk needs for other services.

If you see as output "nptl-version"

then I think you should see only one Asterisk process.

Regards

On 5/31/06, Lee Archer <[EMAIL PROTECTED]> wrote:
>
>
>
> Can someone shed any light on the following.  I have 2 identical 
> systems, 1 of which seems to spawn multiple processes which have to be
killed manually.
>  It recently kicked up 2 so I ran gdb on them and this is the thread
output.
>  I current use FreePBX with these systems.
>
> 1st extra process
>
> (gdb) info thread
>   1 Thread 1095261104 (LWP 14213)  0xe410 in __kernel_vsyscall ()
> (gdb) thread apply all bt
>
> Thread 1 (Thread 1095261104 (LWP 14213)):
> #0  0xe410 in __kernel_vsyscall ()
> #1  0x4004f13e in __lll_mutex_lock_wait () from 
> /lib/tls/libpthread.so.0
> #2  0x4004be41 in _L_mutex_lock_191 () from /lib/tls/libpthread.so.0
> #3  0x0001 in ?? ()
> #4  0x40e16778 in ?? () from
> /usr/lib/asterisk/modules/cdr_odbc.so
> #5  0x40e16818 in __dso_handle () from 
> /usr/lib/asterisk/modules/cdr_odbc.so
> #6  0x0002 in ?? ()
> #7  0x in ?? ()
> #8  0x080a20b7 in ast_cdr_unregister (name=0x40e1455c "ODBC") at 
> lock.h:592
> #9  0x40e13299 in odbc_unload_module () at cdr_odbc.c:240 #10 
> 0x40e13978 in reload () at cdr_odbc.c:465
> #11 0x0805be32 in ast_module_reload (name=0x0) at loader.c:257
> #12 0x080b4623 in hup_handler (num=-4) at asterisk.c:754
> #13 
> #14 0xe410 in __kernel_vsyscall ()
> #15 0x401afd99 in sched_setscheduler () from /lib/tls/libc.so.6
> #16 0x080b4743 in ast_set_priority (pri=0) at asterisk.c:803
> #17 0x40445ee8 in agi_exec_full (chan=0x82782b0, data= out>, enhanced=0, dead=0) at res_agi.c:300
> #18 0x0808d521 in pbx_extension_helper (c=0x82782b0, con= optimized
> out>, context=0x8278400 "macro-record-enable",
>
> exten=0x82784f4 "s", priority=4, label=0x0, callerid=0x8159f38 
> "0163861", action=1) at pbx.c:553
> #19 0x40c44851 in macro_exec (chan=0x82782b0, data=0x4147c768) at 
> app_macro.c:210 #20 0x0808d521 in pbx_extension_helper (c=0x82782b0, 
> con= out>, context=0x8278400 "macro-record-enable",
>
> exten=0x82784f4 "s", priority=7, label=0x0, callerid=0x8159f38 
> "0163861", action=1) at pbx.c:553
> #21 0x40c44851 in macro_exec (chan=0x82782b0, data=0x41482fd8) at 
> app_macro.c:210
> #22 0x0808d521 in pbx_extension_helper (c=0x82782b0, con= optimized
> out>, context=0x8278400 "macro-record-enable",
>
> exten=0x82784f4 "s", priority=1, label=0x0, callerid=0x8159f38 
> "0163861", action=1) at pbx.c:553
> #23 0x0808ead4 in __ast_pbx_run (c=0x82782b0) at pbx.c:2227
> #24 0x0808f6cc in pbx_thread (data=0x0) at pbx.c:2514
> #25 0x4004a297 in start_thread () from /lib/tls/libpthread.so.0
> #26 0x401c737e in clone () from /lib/tls/libc.so.6
> #27 0x41485bb0 in ?? ()
> #0  0xe410 in __kernel_vsyscall ()
>
> 2nd extra process
>
> (gdb) info thread
>   1 Thread 1096059824 (LWP 14214)  0xe410 in ?? ()
> (gdb) thread apply all bt
>
> Thread 1 (Thread 1096059824 (LWP 14214)):
> #0  0xe410 in ?? ()
> #1  0x41533594 in ?? ()
> #2  0x0002 in ?? ()
> #3  0x in ?? ()
> #4  0x4004f13e in __lll_mutex_lock_wait () from 
> /lib/tls/libpthread.so.0
> #5  0x4004be41 in _L_mutex_lock_191 () from /lib/tls/libpthread.so.0
> #6  0x0001 in ?? ()
> #7  0x40e16778 in ?? () from
> /usr/lib/asterisk/modules/cdr_odbc.so
> #8  0x40e16818 in __dso_handle () from 
> /usr/lib/asterisk/modules/cdr_odbc.so
> #9  0x0002 in ?? ()
> #10 0x in ?? ()
> #11 0x080a20b7 in ast_cdr_unregister (name=0x40e1455c "ODBC") at 
> lock.h:592
> #12 0x40e13299 in odbc_unload_module () at cdr_odbc.c:240
> #13 0x40e13978 in reload () at cdr_odbc.c:465
> #14 0x0805be32 in ast_module_reload (name=0x0) at loader.c:257
> #15 0x080b4623 in hup_handler (num=-4) at asterisk.c:754
> #16 
> #17 0xe410 in ?? ()
> #0  0xe410 in ?? ()
>
> Regards
>
> Lee 

[Asterisk-Users] Multiple processes

[Lee Archer]

Title: Multiple processes






Can someone shed any light on the following.  I have 2 identical systems, 1 of which seems to spawn multiple processes which have to be killed manually.  It recently kicked up 2 so I ran gdb on them and this is the thread output.  I current use FreePBX with these systems.

1st extra process


(gdb) info thread

  1 Thread 1095261104 (LWP 14213)  0xe410 in __kernel_vsyscall ()

(gdb) thread apply all bt


Thread 1 (Thread 1095261104 (LWP 14213)):

#0  0xe410 in __kernel_vsyscall ()

#1  0x4004f13e in __lll_mutex_lock_wait () from /lib/tls/libpthread.so.0

#2  0x4004be41 in _L_mutex_lock_191 () from /lib/tls/libpthread.so.0

#3  0x0001 in ?? ()

#4  0x40e16778 in ?? () from /usr/lib/asterisk/modules/cdr_odbc.so

#5  0x40e16818 in __dso_handle () from /usr/lib/asterisk/modules/cdr_odbc.so

#6  0x0002 in ?? ()

#7  0x in ?? ()

#8  0x080a20b7 in ast_cdr_unregister (name=0x40e1455c "ODBC") at lock.h:592

#9  0x40e13299 in odbc_unload_module () at cdr_odbc.c:240

#10 0x40e13978 in reload () at cdr_odbc.c:465

#11 0x0805be32 in ast_module_reload (name=0x0) at loader.c:257

#12 0x080b4623 in hup_handler (num=-4) at asterisk.c:754

#13 

#14 0xe410 in __kernel_vsyscall ()

#15 0x401afd99 in sched_setscheduler () from /lib/tls/libc.so.6

#16 0x080b4743 in ast_set_priority (pri=0) at asterisk.c:803

#17 0x40445ee8 in agi_exec_full (chan=0x82782b0, data="" optimized out>, enhanced=0, dead=0) at res_agi.c:300

#18 0x0808d521 in pbx_extension_helper (c=0x82782b0, con=, context=0x8278400 "macro-record-enable",

    exten=0x82784f4 "s", priority=4, label=0x0, callerid=0x8159f38 "0163861", action="" at pbx.c:553

#19 0x40c44851 in macro_exec (chan=0x82782b0, data="" at app_macro.c:210

#20 0x0808d521 in pbx_extension_helper (c=0x82782b0, con=, context=0x8278400 "macro-record-enable",

    exten=0x82784f4 "s", priority=7, label=0x0, callerid=0x8159f38 "0163861", action="" at pbx.c:553

#21 0x40c44851 in macro_exec (chan=0x82782b0, data="" at app_macro.c:210

#22 0x0808d521 in pbx_extension_helper (c=0x82782b0, con=, context=0x8278400 "macro-record-enable",

    exten=0x82784f4 "s", priority=1, label=0x0, callerid=0x8159f38 "0163861", action="" at pbx.c:553

#23 0x0808ead4 in __ast_pbx_run (c=0x82782b0) at pbx.c:2227

#24 0x0808f6cc in pbx_thread (data="" at pbx.c:2514

#25 0x4004a297 in start_thread () from /lib/tls/libpthread.so.0

#26 0x401c737e in clone () from /lib/tls/libc.so.6

#27 0x41485bb0 in ?? ()

#0  0xe410 in __kernel_vsyscall ()


2nd extra process


(gdb) info thread

  1 Thread 1096059824 (LWP 14214)  0xe410 in ?? ()

(gdb) thread apply all bt


Thread 1 (Thread 1096059824 (LWP 14214)):

#0  0xe410 in ?? ()

#1  0x41533594 in ?? ()

#2  0x0002 in ?? ()

#3  0x in ?? ()

#4  0x4004f13e in __lll_mutex_lock_wait () from /lib/tls/libpthread.so.0

#5  0x4004be41 in _L_mutex_lock_191 () from /lib/tls/libpthread.so.0

#6  0x0001 in ?? ()

#7  0x40e16778 in ?? () from /usr/lib/asterisk/modules/cdr_odbc.so

#8  0x40e16818 in __dso_handle () from /usr/lib/asterisk/modules/cdr_odbc.so

#9  0x0002 in ?? ()

#10 0x in ?? ()

#11 0x080a20b7 in ast_cdr_unregister (name=0x40e1455c "ODBC") at lock.h:592

#12 0x40e13299 in odbc_unload_module () at cdr_odbc.c:240

#13 0x40e13978 in reload () at cdr_odbc.c:465

#14 0x0805be32 in ast_module_reload (name=0x0) at loader.c:257

#15 0x080b4623 in hup_handler (num=-4) at asterisk.c:754

#16 

#17 0xe410 in ?? ()

#0  0xe410 in ?? ()


Regards


Lee


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RE: [Asterisk-Users] mpg123 or asterisk

[Lee Archer]

I don't know as I never experienced any system crashes with either.  But
I did notice many times that mpg123 was still running after asterisk had
been shutdown.

Lee 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of MBIT
Technologies
Sent: 30 May 2006 08:29
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] mpg123 or asterisk

Can MAD crash a server like mpg123 can?




-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Lee Archer
Sent: Tuesday, 30 May 2006 5:06 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] mpg123 or asterisk

Can't you use mpg123 as compiled under x86_32?  I do on a few servers I
have.  I found madplay better process wise than mpg123.

Regards

Lee 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Erick
Perez
Sent: 29 May 2006 21:37
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] mpg123 or asterisk

Well, being unable to compile mpg123 under x86_64 i installed lame and
transformed the mp3-->wav-->raw.
and using "files" as the format player.

Are there any good scripts to stress test MoH?
I want to test this machine for 1000 "calls" on hold.

http://www.asteriskguru.com/tutorials/astertest.html
AsterTest is good but i dont have access to another * installation.


On 5/27/06, Steve Totaro <[EMAIL PROTECTED]> wrote:
> Please let us know your results.  I cannot really test this in 
> production system since it is a $16,000/hr call center.  I was using 
> madplay but it was crashing and creating zombie processes, I figured 
> native was not the way to go since all of the different audio streams.
> Mpg123 works perfectly for me under a load of sixty channels, I can 
> confirm that for sure.
>
> Thanks,
> Steve
>
> Erick Perez wrote:
> > Interesting.
> > So, i will have to test then...
> >
> >
> > On 5/27/06, Steve Totaro <[EMAIL PROTECTED]> wrote:
> >> In my very limited testing of native, each channel was receiving a 
> >> different stream (each caller heard something different).  Under a 
> >> high volume of calls, which is going to hurt performance more?
> >> Transcoding MP3s but sending a single stream or separate streams 
> >> per call under native?
> >>
> >> When I say high, I mean 1,000+ calls.
> >>
> >> Thanks,
> >> Steve
> >>
> >>
> >> Erick Perez wrote:
> >> > Thanks to all. Native format will be.
> >> >
> >> > On 5/27/06, Matt Riddell (IT) <[EMAIL PROTECTED]> wrote:
> >> >> Vahan Yerkanian wrote:
> >> >> > Erick Perez wrote:
> >> >> >> should I use mpg123 with asterisk 1.2.7 or should i use the 
> >> >> >> native player asterisk has?
> >> >> >> the target machine will receive heavy load.
> >> >> >
> >> >> > mpg123 was used back when asterisk didn't have native format
> >> >> support. If
> >> >> > you are expecting heavy load, the native format is the way to
> >> go. You
> >> >> > might decide not to use mp3 format at all, recompressing your 
> >> >> > MoH
> >> >> files
> >> >> > using sox to the formats you gonna use, such as .al, .ul, 
> >> >> > .gsm, or
> >> >> leave
> >> >> > it at .sln to cut the decoding leg only.
> >> >>
> >> >> Heh, damn this GPRS connection.  In order to pass the time while

> >> >> downloading messages I reply before they are all in, and yet by
> >> the time
> >> >> I have received all the messages I note that your question has
> >> already
> >> >> been answered!
> >> >>
> >> >> :)
> >> >>
> >> >> --
> >> >> Cheers,
> >> >>
> >> >> Matt Riddell
> >> >> ___
> >> >>
> >> >> http://www.sineapps.com/news.php (Daily Asterisk News - html) 
> >> >> http://freevoip.gedameurope.com (Free Asterisk Voip Community) 
> >> >> http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) 
> >> >> ___
> >> >> --Bandwidth and Colocation provided by Easynews.com --
> >> >>
> >> >> Asterisk-Users mailing list
> >> >> To UNSUB

RE: [Asterisk-Users] mpg123 or asterisk

[Lee Archer]

Can't you use mpg123 as compiled under x86_32?  I do on a few servers I
have.  I found madplay better process wise than mpg123.

Regards

Lee 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Erick
Perez
Sent: 29 May 2006 21:37
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] mpg123 or asterisk

Well, being unable to compile mpg123 under x86_64 i installed lame and
transformed the mp3-->wav-->raw.
and using "files" as the format player.

Are there any good scripts to stress test MoH?
I want to test this machine for 1000 "calls" on hold.

http://www.asteriskguru.com/tutorials/astertest.html
AsterTest is good but i dont have access to another * installation.


On 5/27/06, Steve Totaro <[EMAIL PROTECTED]> wrote:
> Please let us know your results.  I cannot really test this in 
> production system since it is a $16,000/hr call center.  I was using 
> madplay but it was crashing and creating zombie processes, I figured 
> native was not the way to go since all of the different audio streams.
> Mpg123 works perfectly for me under a load of sixty channels, I can 
> confirm that for sure.
>
> Thanks,
> Steve
>
> Erick Perez wrote:
> > Interesting.
> > So, i will have to test then...
> >
> >
> > On 5/27/06, Steve Totaro <[EMAIL PROTECTED]> wrote:
> >> In my very limited testing of native, each channel was receiving a 
> >> different stream (each caller heard something different).  Under a 
> >> high volume of calls, which is going to hurt performance more?  
> >> Transcoding MP3s but sending a single stream or separate streams 
> >> per call under native?
> >>
> >> When I say high, I mean 1,000+ calls.
> >>
> >> Thanks,
> >> Steve
> >>
> >>
> >> Erick Perez wrote:
> >> > Thanks to all. Native format will be.
> >> >
> >> > On 5/27/06, Matt Riddell (IT) <[EMAIL PROTECTED]> wrote:
> >> >> Vahan Yerkanian wrote:
> >> >> > Erick Perez wrote:
> >> >> >> should I use mpg123 with asterisk 1.2.7 or should i use the 
> >> >> >> native player asterisk has?
> >> >> >> the target machine will receive heavy load.
> >> >> >
> >> >> > mpg123 was used back when asterisk didn't have native format
> >> >> support. If
> >> >> > you are expecting heavy load, the native format is the way to
> >> go. You
> >> >> > might decide not to use mp3 format at all, recompressing your 
> >> >> > MoH
> >> >> files
> >> >> > using sox to the formats you gonna use, such as .al, .ul, 
> >> >> > .gsm, or
> >> >> leave
> >> >> > it at .sln to cut the decoding leg only.
> >> >>
> >> >> Heh, damn this GPRS connection.  In order to pass the time while

> >> >> downloading messages I reply before they are all in, and yet by
> >> the time
> >> >> I have received all the messages I note that your question has
> >> already
> >> >> been answered!
> >> >>
> >> >> :)
> >> >>
> >> >> --
> >> >> Cheers,
> >> >>
> >> >> Matt Riddell
> >> >> ___
> >> >>
> >> >> http://www.sineapps.com/news.php (Daily Asterisk News - html) 
> >> >> http://freevoip.gedameurope.com (Free Asterisk Voip Community) 
> >> >> http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) 
> >> >> ___
> >> >> --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
> >> >>
> >> >
> >> >
> >>
> >> ___
> >> --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
> >>
> >
> >
>
> ___
> --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
>


-- 

---
Erick Perez
Linux User 376588
http://counter.li.org/  (Get counted!!!) Panama, Republic of Panama
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RE: [Asterisk-Users] GXP2k and BLF problem

[Lee Archer]

Think you need to contact Grandstream support then.  
I've got the same version of * and GXP fw and I get no problems.  Sorry I 
can't help you any further.
 
Lee


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of 
asteriskSent: 24 May 2006 13:30To: 'Asterisk Users 
Mailing List - Non-Commercial Discussion'Subject: RE: 
[Asterisk-Users] GXP2k and BLF problem


Phone is manually 
configured through web interface.
 
Thanks to all those who 
are helping with this.
 
Fadge
 




From: 
[EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Lee ArcherSent: 24 May 2006 13:18To: Asterisk Users Mailing List - 
Non-Commercial DiscussionSubject: RE: [Asterisk-Users] GXP2k and BLF 
problem
 
Are you using 
preconfiged scripts?  If so what happens if you manually config the phone 
then restart asterisk and then the phone?
 
Lee
 



From: 
[EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of asteriskSent: 24 May 2006 12:56To: 'Asterisk Users Mailing List - 
Non-Commercial Discussion'Subject: RE: [Asterisk-Users] GXP2k and BLF 
problem
No,
    
Rebooting the phones does not fix the problem. As previously stated the only way 
to get them working again is to default the config of the phone and 
rebuild.
 
Fadge
 




From: 
[EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Lee ArcherSent: 24 May 2006 12:41To: Asterisk Users Mailing List - 
Non-Commercial DiscussionSubject: RE: [Asterisk-Users] GXP2k and BLF 
problem
 
I run 1.1.0.13 on my 
GXP's and after stopping and starting the server I either wait for the phones to 
re-reg or I reboot the phones.  After restarting asterisk does rebooting 
the phones does fix the problem?
 
Lee
 



From: 
[EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of asteriskSent: 24 May 2006 12:25To: 'Asterisk Users Mailing List - 
Non-Commercial Discussion'Subject: RE: [Asterisk-Users] GXP2k and BLF 
problem
Phones are configured 
for “asterisk BLF” . Like I said, the config works when the phone is defaulted 
and rebuilt but stops working if the server is rebooted. Show hints 
Gives:
 
-= Registered Asterisk 
Dial Plan Hints =-
   
310 
: 
SIP/310   
State:Unavailable Watchers  0
   
305 
: 
SIP/305   
State:Idle    
    Watchers  0
   
304 
: 
SIP/304   
State:Idle    
    Watchers  0
   
303 
: 
SIP/303   
State:Idle   
  Watchers  0
   
302 
: 
SIP/302   
State:Idle   
  Watchers  0
   
301 
: 
SIP/301   
State:Unavailable Watchers  0

-  
6 hints 
registered
 
Thanks
 
fadge
 




From: 
[EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Lee ArcherSent: 24 May 2006 12:01To: Asterisk Users Mailing List - 
Non-Commercial DiscussionSubject: RE: [Asterisk-Users] GXP2k and BLF 
problem
 
Are the GXP's 
configured properly for BLF and what does show hints 
print?
 
Lee
 



From: 
[EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of asteriskSent: 24 May 2006 11:57To: 'Asterisk Users Mailing List - 
Non-Commercial Discussion'Subject: RE: [Asterisk-Users] GXP2k and BLF 
problem
It does not happen with 
a reload of asterisk but when the server is rebooted. I have set the register 
timeout to 1 min but this has no effect. The BLF still does not work. The * is V 
1.2.7.1. I can see the phone register first time when phone reboots and would 
expect to see it re-register every 1 min on the console if I understand things 
correctly, but it does not.
 
Fadge
 




From: 
[EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Lee ArcherSent: 24 May 2006 11:30To: Asterisk Users Mailing List - 
Non-Commercial DiscussionSubject: RE: [Asterisk-Users] GXP2k and BLF 
problem
 
Stopping and restarting 
Asterisk will lose the hints, then you will have to wait until the phone 
registers again.  With 1.2.7.1 a reload shouldn't lose anything.  
Change the register time on the phones to something less that 60 minutes if it's 
a big problem.  Instead of factory defaulting the phones you might find a 
simple restart will re-register the phone and BLF will work on the 
phone.
 
Regards
 
lee
 



From: 
[EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of picciuXSent: 24 May 2006 10:49To: Asterisk Users Mailing List - 
Non-Commercial DiscussionSubject: Re: [Asterisk-Users] GXP2k and BLF 
problem
asterisk version? until 1.2.7.1, sip subscriptions get lost on reload of 
asterisk. In this case, have to wait until phone re-registers and re-subscribe 
for BLFs...

2006/5/24, asterisk <[EMAIL PROTECTED]>: 

Hi,Is anyone else having problems with the 
Grandstream GXP2000 BLF? When Irestart * the status lights stop working but 
the buttons still work as onetouch dials and call the correct number. The 
only way I cen get the lights working again is to reset to factory default 
and re-build the config.I am running I upgraded t

RE: [Asterisk-Users] GXP2k and BLF problem

[Lee Archer]

Are you using preconfiged scripts?  If so what happens 
if you manually config the phone then restart asterisk and then the 
phone?
 
Lee


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of 
asteriskSent: 24 May 2006 12:56To: 'Asterisk Users 
Mailing List - Non-Commercial Discussion'Subject: RE: 
[Asterisk-Users] GXP2k and BLF problem


No,
    
Rebooting the phones does not fix the problem. As previously stated the only way 
to get them working again is to default the config of the phone and 
rebuild.
 
Fadge
 




From: 
[EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Lee ArcherSent: 24 May 2006 12:41To: Asterisk Users Mailing List - 
Non-Commercial DiscussionSubject: RE: [Asterisk-Users] GXP2k and BLF 
problem
 
I run 1.1.0.13 on my 
GXP's and after stopping and starting the server I either wait for the phones to 
re-reg or I reboot the phones.  After restarting asterisk does rebooting 
the phones does fix the problem?
 
Lee
 



From: 
[EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of asteriskSent: 24 May 2006 12:25To: 'Asterisk Users Mailing List - 
Non-Commercial Discussion'Subject: RE: [Asterisk-Users] GXP2k and BLF 
problem
Phones are configured 
for “asterisk BLF” . Like I said, the config works when the phone is defaulted 
and rebuilt but stops working if the server is rebooted. Show hints 
Gives:
 
-= Registered Asterisk 
Dial Plan Hints =-
   
310 
: 
SIP/310   
State:Unavailable Watchers  0
   
305 
: 
SIP/305   
State:Idle    
    Watchers  0
   
304 
: 
SIP/304   
State:Idle    
    Watchers  0
   
303 
: 
SIP/303   
State:Idle   
  Watchers  0
   
302 
: 
SIP/302   
State:Idle   
  Watchers  0
   
301 
: 
SIP/301   
State:Unavailable Watchers  0

-  
6 hints 
registered
 
Thanks
 
fadge
 




From: 
[EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Lee ArcherSent: 24 May 2006 12:01To: Asterisk Users Mailing List - 
Non-Commercial DiscussionSubject: RE: [Asterisk-Users] GXP2k and BLF 
problem
 
Are the GXP's 
configured properly for BLF and what does show hints 
print?
 
Lee
 



From: 
[EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of asteriskSent: 24 May 2006 11:57To: 'Asterisk Users Mailing List - 
Non-Commercial Discussion'Subject: RE: [Asterisk-Users] GXP2k and BLF 
problem
It does not happen with 
a reload of asterisk but when the server is rebooted. I have set the register 
timeout to 1 min but this has no effect. The BLF still does not work. The * is V 
1.2.7.1. I can see the phone register first time when phone reboots and would 
expect to see it re-register every 1 min on the console if I understand things 
correctly, but it does not.
 
Fadge
 




From: 
[EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Lee ArcherSent: 24 May 2006 11:30To: Asterisk Users Mailing List - 
Non-Commercial DiscussionSubject: RE: [Asterisk-Users] GXP2k and BLF 
problem
 
Stopping and restarting 
Asterisk will lose the hints, then you will have to wait until the phone 
registers again.  With 1.2.7.1 a reload shouldn't lose anything.  
Change the register time on the phones to something less that 60 minutes if it's 
a big problem.  Instead of factory defaulting the phones you might find a 
simple restart will re-register the phone and BLF will work on the 
phone.
 
Regards
 
lee
 



From: 
[EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of picciuXSent: 24 May 2006 10:49To: Asterisk Users Mailing List - 
Non-Commercial DiscussionSubject: Re: [Asterisk-Users] GXP2k and BLF 
problem
asterisk version? until 1.2.7.1, sip subscriptions get lost on reload of 
asterisk. In this case, have to wait until phone re-registers and re-subscribe 
for BLFs...

2006/5/24, asterisk <[EMAIL PROTECTED]>: 

Hi,Is anyone else having problems with the 
Grandstream GXP2000 BLF? When Irestart * the status lights stop working but 
the buttons still work as onetouch dials and call the correct number. The 
only way I cen get the lights working again is to reset to factory default 
and re-build the config.I am running I upgraded the firmware to 1.1.0.13 but still have the 
sameproblem.Any ideas 
?Fadge___--Bandwidth 
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RE: [Asterisk-Users] GXP2k and BLF problem

[Lee Archer]

I run 1.1.0.13 on my GXP's and after stopping and starting 
the server I either wait for the phones to re-reg or I reboot the phones.  
After restarting asterisk does rebooting the phones does fix the 
problem?
 
Lee


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of 
asteriskSent: 24 May 2006 12:25To: 'Asterisk Users 
Mailing List - Non-Commercial Discussion'Subject: RE: 
[Asterisk-Users] GXP2k and BLF problem


Phones are configured 
for “asterisk BLF” . Like I said, the config works when the phone is defaulted 
and rebuilt but stops working if the server is rebooted. Show hints 
Gives:
 
-= Registered Asterisk 
Dial Plan Hints =-
   
310 
: 
SIP/310   
State:Unavailable Watchers  0
   
305 
: 
SIP/305   
State:Idle    
    Watchers  0
   
304 
: 
SIP/304   
State:Idle    
    Watchers  0
   
303 
: 
SIP/303   
State:Idle   
  Watchers  0
   
302 
: 
SIP/302   
State:Idle   
  Watchers  0
   
301 
: 
SIP/301   
State:Unavailable Watchers  0

-  
6 hints 
registered
 
Thanks
 
fadge
 




From: 
[EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Lee ArcherSent: 24 May 2006 12:01To: Asterisk Users Mailing List - 
Non-Commercial DiscussionSubject: RE: [Asterisk-Users] GXP2k and BLF 
problem
 
Are the GXP's 
configured properly for BLF and what does show hints 
print?
 
Lee
 



From: 
[EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of asteriskSent: 24 May 2006 11:57To: 'Asterisk Users Mailing List - 
Non-Commercial Discussion'Subject: RE: [Asterisk-Users] GXP2k and BLF 
problem
It does not happen with 
a reload of asterisk but when the server is rebooted. I have set the register 
timeout to 1 min but this has no effect. The BLF still does not work. The * is V 
1.2.7.1. I can see the phone register first time when phone reboots and would 
expect to see it re-register every 1 min on the console if I understand things 
correctly, but it does not.
 
Fadge
 




From: 
[EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Lee ArcherSent: 24 May 2006 11:30To: Asterisk Users Mailing List - 
Non-Commercial DiscussionSubject: RE: [Asterisk-Users] GXP2k and BLF 
problem
 
Stopping and restarting 
Asterisk will lose the hints, then you will have to wait until the phone 
registers again.  With 1.2.7.1 a reload shouldn't lose anything.  
Change the register time on the phones to something less that 60 minutes if it's 
a big problem.  Instead of factory defaulting the phones you might find a 
simple restart will re-register the phone and BLF will work on the 
phone.
 
Regards
 
lee
 



From: 
[EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of picciuXSent: 24 May 2006 10:49To: Asterisk Users Mailing List - 
Non-Commercial DiscussionSubject: Re: [Asterisk-Users] GXP2k and BLF 
problem
asterisk version? until 1.2.7.1, sip subscriptions get lost on reload of 
asterisk. In this case, have to wait until phone re-registers and re-subscribe 
for BLFs...

2006/5/24, asterisk <[EMAIL PROTECTED]>: 

Hi,Is anyone else having problems with the 
Grandstream GXP2000 BLF? When Irestart * the status lights stop working but 
the buttons still work as onetouch dials and call the correct number. The 
only way I cen get the lights working again is to reset to factory default 
and re-build the config.I am running I upgraded the firmware to 1.1.0.13 but still have the 
sameproblem.Any ideas 
?Fadge___--Bandwidth 
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RE: [Asterisk-Users] GXP2k and BLF problem

[Lee Archer]

Are the GXP's configured properly for BLF and 
what does show hints print?
 
Lee


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of 
asteriskSent: 24 May 2006 11:57To: 'Asterisk Users 
Mailing List - Non-Commercial Discussion'Subject: RE: 
[Asterisk-Users] GXP2k and BLF problem


It does not happen with 
a reload of asterisk but when the server is rebooted. I have set the register 
timeout to 1 min but this has no effect. The BLF still does not work. The * is V 
1.2.7.1. I can see the phone register first time when phone reboots and would 
expect to see it re-register every 1 min on the console if I understand things 
correctly, but it does not.
 
Fadge
 




From: 
[EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Lee ArcherSent: 24 May 2006 11:30To: Asterisk Users Mailing List - 
Non-Commercial DiscussionSubject: RE: [Asterisk-Users] GXP2k and BLF 
problem
 
Stopping and restarting 
Asterisk will lose the hints, then you will have to wait until the phone 
registers again.  With 1.2.7.1 a reload shouldn't lose anything.  
Change the register time on the phones to something less that 60 minutes if it's 
a big problem.  Instead of factory defaulting the phones you might find a 
simple restart will re-register the phone and BLF will work on the 
phone.
 
Regards
 
lee
 



From: 
[EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of picciuXSent: 24 May 2006 10:49To: Asterisk Users Mailing List - 
Non-Commercial DiscussionSubject: Re: [Asterisk-Users] GXP2k and BLF 
problem
asterisk version? until 1.2.7.1, sip subscriptions get lost on reload of 
asterisk. In this case, have to wait until phone re-registers and re-subscribe 
for BLFs...

2006/5/24, asterisk <[EMAIL PROTECTED]>: 

Hi,Is anyone else having problems with the 
Grandstream GXP2000 BLF? When Irestart * the status lights stop working but 
the buttons still work as onetouch dials and call the correct number. The 
only way I cen get the lights working again is to reset to factory default 
and re-build the config.I am running I upgraded the firmware to 1.1.0.13 but still have the 
sameproblem.Any ideas 
?Fadge___--Bandwidth 
and Colocation provided by Easynews.com 
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   http://lists.digium.com/mailman/listinfo/asterisk-users
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RE: [Asterisk-Users] GXP2k and BLF problem

[Lee Archer]

Stopping and restarting Asterisk will lose the hints, then 
you will have to wait until the phone registers again.  With 1.2.7.1 a 
reload shouldn't lose anything.  Change the register time on the phones to 
something less that 60 minutes if it's a big problem.  Instead of factory 
defaulting the phones you might find a simple restart will re-register the phone 
and BLF will work on the phone.
 
Regards
 
lee


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of 
picciuXSent: 24 May 2006 10:49To: Asterisk Users 
Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] 
GXP2k and BLF problem
asterisk version? until 1.2.7.1, sip 
subscriptions get lost on reload of asterisk. In this case, have to wait until 
phone re-registers and re-subscribe for BLFs...
2006/5/24, asterisk <[EMAIL PROTECTED]>:
Hi,Is 
  anyone else having problems with the Grandstream GXP2000 BLF? When 
  Irestart * the status lights stop working but the buttons still work as 
  onetouch dials and call the correct number. The only way I cen get the 
  lights working again is to reset to factory default and re-build the 
  config.I am running I upgraded the firmware to 1.1.0.13 but still have the 
  sameproblem.Any ideas 
  ?Fadge___--Bandwidth 
  and Colocation provided by Easynews.com 
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RE: [Asterisk-Users] CallerID

[Lee Archer]

I don't have a PBX sitting between Asterisk and the telco.  Asterisk is
the PBX.  I'm using a TE110P card.

Regards

Lee 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of C F
Sent: 23 May 2006 17:56
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] CallerID

It appears that the PBX sitting between Asterisk and your provider is
not passing on the calling pres flags.

On 5/23/06, Lee Archer <[EMAIL PROTECTED]> wrote:
> I have a problem with BT in the UK.  Using setcallerpres I can change 
> the number shown on the recipents phones to Private or unknown but no 
> matter what I set my asterisk cid and callerpres to it still displays 
> the base number of my PRI ddi range.
>
> Regards
>
> Lee
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of C F
> Sent: 23 May 2006 15:05
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] CallerID
>
> You should set the presentation flags to private.
> http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+CallingPres
>
> On 5/23/06, Greg Oliver <[EMAIL PROTECTED]> wrote:
> > I am trying to set CIDNum to nothing, but my outgoing PRI controlled

> > by another PBX seems to fill in something when asterisk does not..  
> > If
>
> > I set a number either in the sip channel for the phone, or from 
> > extensions.con, it is realized..  If I try to leave them blank, or 
> > even Not Defined, the main number of the pri gets sent out..
> >
> > I am trying to debug a glitvh in or software and I need to be able 
> > to make a test call with unknown (blank callerid)..  I can 
> > successfully set it to private, but that is not the same..
> >
> > Any ideas?
> >
> > TIA
> >
> > -Greg
> >
> > ___
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> >
> > Asterisk-Users mailing list
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> >
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>
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RE: [Asterisk-Users] CallerID

[Lee Archer]

I have a problem with BT in the UK.  Using setcallerpres I can change
the number shown on the recipents phones to Private or unknown but no
matter what I set my asterisk cid and callerpres to it still displays
the base number of my PRI ddi range. 

Regards

Lee

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of C F
Sent: 23 May 2006 15:05
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] CallerID

You should set the presentation flags to private.
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+CallingPres

On 5/23/06, Greg Oliver <[EMAIL PROTECTED]> wrote:
> I am trying to set CIDNum to nothing, but my outgoing PRI controlled 
> by another PBX seems to fill in something when asterisk does not..  If

> I set a number either in the sip channel for the phone, or from 
> extensions.con, it is realized..  If I try to leave them blank, or 
> even Not Defined, the main number of the pri gets sent out..
>
> I am trying to debug a glitvh in or software and I need to be able to 
> make a test call with unknown (blank callerid)..  I can successfully 
> set it to private, but that is not the same..
>
> Any ideas?
>
> TIA
>
> -Greg
>
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> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
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[Asterisk-Users] Future pickup feature

[Lee Archer]

Title: Future pickup feature






Can anyone say whether call pickup with the ability to transfer the callers details is going to be part of any Asterisk release?  I'd like to pick up calls but also know roughly who it is I'm talking.

Regards


Lee


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RE: [Asterisk-Users] Asterisk with SuSe 10

[Lee Archer]

I downloaded the source and built it from that.  SuSE10 comes with a
version of asterisk 1.0.X on the DVD.

Regards

Lee 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Yu Safin
Sent: 01 May 2006 16:31
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk with SuSe 10

On 1/24/06, Lee Archer <[EMAIL PROTECTED]> wrote:
> Thanks, I've got it running on my test box but didn't know if there 
> was any global objection to using it.  I've had a few funnies with it 
> but that might be down to Supermicro and P4's with the EM64T thing.
>
> Regards
>
> Lee
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Ben 
> Klang
> Sent: 24 January 2006 15:49
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Asterisk with SuSe 10
>
> On Tuesday 24 January 2006 09:26, Lee Archer wrote:
> > Has anyone had any experience with the Asterisk on a SuSe 10
platform?
> > I'm currently using FC3 but because we use SuSe within other parts 
> > of the business I'm being pushed to changed the OS.
> Just about all of my production Asterisk servers are on SuSE 9.3.  My 
> development and demo boxes are SuSE 10.  Both run great.  I do however

> usually tweak the RPM that came with it to add in a few patches.  If 
> you are comfortable with running Asterisk 1.0.9 then the RPM works 
> very well.  SuSE always seems to really think things through when they

> package applications.
>
> For running something newer than Asterisk 1.0.9 SuSE 10 is also works 
> fine.
> For your own sanity you'll want to not install/uninstall the SuSE 
> Asterisk RPMs.  One possible gotcha: be careful of possibly 
> conflicting kernel modules in /lib/modules/`uname -r`/extra as the 
> Zaptel drivers are not part of any Asterisk package but rather the 
> kernel.  The zaptel compile from source installs modules to 
> /lib/modules/`uname -r`/misc so you'll want to delete the files in 
> extra.  You'll also have to remember that each time you update the
kernel RPM.
>
> Hope that helps.  The bottom line from me is Thumbs Up.
>
> /BAK/
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>
did you have to install asterisk from source or from rpm?
I have installed asterisk under RH and I am switching over to SuSE OSS
10.0.  I could not find the "rpm" for asterisk.  My searches show that
the rpm is available for the commercial version of SuSE.
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RE: [Asterisk-Users] multiple asterisk process ?

[Lee Archer]

All I can figure is that something I haven't yet figured is causing
these processes to be created, and after a while there is so many that
outgoing calls over zap can't be made.  It only applies to 1 system out
of 7, running Suse 10 and a 2.6 kernel.

Lee 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dave
Cotton
Sent: 18 April 2006 10:02
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] multiple asterisk process ?

On Tue, 2006-04-18 at 09:33 +0100, Lee Archer wrote:
> Yes it is a problem cos after a while of just leaving it the system is

> unable to make calls out via the PSTN, which is why I have spent time 
> with the telco, more like wasted time, and played with zaptel's 
> make options.  After trying a few things I came to the temporary 
> conclusion that it was the zaptel watchdog trying and failing to 
> restart a hung channel.  I recompiled zaptel without the watchdog and 
> a few days later it did the same so I'm back to sq 1.

Ok, I'll ask it another way.

Is it _the_ problem because I've an uptime of 209 days on a system with
no problems and multiple asterisk processes.


--
Dave Cotton <[EMAIL PROTECTED]>

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RE: [Asterisk-Users] multiple asterisk process ?

[Lee Archer]

Yes it is a problem cos after a while of just leaving it the system is
unable to make calls out via the PSTN, which is why I have spent time
with the telco, more like wasted time, and played with zaptel's make
options.  After trying a few things I came to the temporary conclusion
that it was the zaptel watchdog trying and failing to restart a hung
channel.  I recompiled zaptel without the watchdog and a few days later
it did the same so I'm back to sq 1.

Regards

Lee

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dave
Cotton
Sent: 18 April 2006 09:27
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] multiple asterisk process ?

On Tue, 2006-04-18 at 09:13 +0100, Lee Archer wrote:
> Any thoughts as to why only 1 of my boxes has this problem?

Is it really a problem?

>  I'm on a
> 2.6 kernel so any more ideas?

Can someone answer what was the original purpose of the "export
LD_ASSUME_KERNEL=2.4.1" in the asterisk script?

Perhaps Gregory Boehnlein, the author, will be able to tell us.

--
Dave Cotton <[EMAIL PROTECTED]>

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RE: [Asterisk-Users] multiple asterisk process ?

[Lee Archer]

Any thoughts as to why only 1 of my boxes has this problem?  I'm on a
2.6 kernel so any more ideas?

Regards

Lee

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dave
Cotton
Sent: 18 April 2006 09:00
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] multiple asterisk process ?

On Tue, 2006-04-18 at 08:29 +0100, Tim Panton wrote:
> 
> I'd guess you have a startup script for asterisk that is setting the 
> LD_ASSUME_KERNEL environment variable.
> 
> To check, find the 'main' asterisk process id (almost always the 
> lowest numbered asterisk process) then look (as root) in the /proc 
> entry, eg:
> 
> cat /proc/13098/environ | strings | grep LD_ASSUME_KERNEL

Now we're getting somewhere.

In some old contribs/init.d  asterisk scripts there is the following:-

# Leave this set unless you know what you are doing.
#export LD_ASSUME_KERNEL=2.4.1

While others have nothing or this

# Uncomment this ONLY if you know what you are doing.
# export LD_ASSUME_KERNEL=2.4.1

--
Dave Cotton <[EMAIL PROTECTED]>

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RE: [Asterisk-Users] multiple asterisk process ?

[Lee Archer]

I had this and no one could really answer it.  I only get it 1 of my
systems.  I've tried a few things, from removing zaptel watchdog - since
I contacted the telco and they said I had a hung channel, to rebuilding
* with different options.  Are you configuring * manually or using a
GUI?

Lee

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of stevanus
Sent: 17 April 2006 10:10
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] multiple asterisk process ?

Hi,

Why does my asterisk keep forking instances at random times everyday?

When I do ps aux, I got this:

asterisk 13068  2.2  5.1 25924 12276 ?   Sl   06:00  13:18 asterisk 
-vvvg -c
asterisk 23558  0.0  5.1 26040 12248 ?   S09:57   0:00 asterisk 
-vvvg -c
asterisk 29832  0.0  5.1 25924 12208 ?   S11:48   0:00 asterisk 
-vvvg -c
asterisk 31872  0.0  5.1 25924 12208 ?   S12:32   0:00 asterisk 
-vvvg -c
asterisk  2520  0.0  5.1 25928 12228 ?   S13:21   0:00 asterisk 
-vvvg -c
asterisk  4638  0.0  5.1 25924 12232 ?   S13:50   0:00 asterisk 
-vvvg -c
asterisk  5126  0.0  5.1 25932 12240 ?   S13:57   0:00 asterisk 
-vvvg -c
asterisk  6487  0.0  5.1 26016 12336 ?   S14:23   0:00 asterisk 
-vvvg -c

Is this normal?
Does anyone experience this?

Thanks..

Regards,

Stevanus
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RE: [Asterisk-Users] Dial out on Zap: Can't fix up channel from 31 to 30 because 30 is already in use??

[Lee Archer]

When you find out what's causing it can you let me know as I have 1
system that gets this error and the telco tells me everything is fine
with their equipment.

Regards

Lee 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Pimjai
Wesnarat
Sent: 11 April 2006 15:33
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Dial out on Zap: Can't fix up channel from 31
to 30 because 30 is already in use??

Hi,

I still cant dial out on Zap and I really have no clue why.
I'm using Asterisk Asterisk 1.2.6 and Zaptel 1.2.5 with Digium card 4
ports, 31 channels each and able to receive incoming calls and fax
perfectly.

I've done this in my dial plan.

exten => 111,1,Answer()
exten => 111,n,Ringing()
exten => 111,n,Wait(2)
exten => 111,n,AbsoluteTimeout(30)
exten => 111,n,Dial(Zap/G1/002212601574) exten =>
111,n,NoOp(${DIALSTATUS}) exten => 111,n,Busy() exten => 111,n,Hangup()

My zapata.conf is like this


[channels]
context=from-pstn
group=0
switchtype=euroisdn
overlapdial=yes
faxdetect=no
echocancel=yes
echocancelwhenbridged=yes


; PRI port 1 (E1)
; context=1
group=1
signalling=pri_cpe
channel=>1-15,17-31


And I've got this on my CLI:

   -- Accepting overlap call from '2212601571' to '111' on channel 0/31,
span 1
-- Starting simple switch on 'Zap/31-1'
-- Executing Answer("Zap/31-1", "") in new stack
-- Executing Ringing("Zap/31-1", "") in new stack
-- Executing Wait("Zap/31-1", "2") in new stack
-- Executing AbsoluteTimeout("Zap/31-1", "30") in new stack
-- Set Absolute Timeout to 30
-- Executing Dial("Zap/31-1", "Zap/G1/002212601574") in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called G1/002212601574
-- Moving call from channel 31 to channel 30 Apr 11 16:27:06
WARNING[10322]: chan_zap.c:7745 pri_fixup_principle: 
Can't fix up channel from 31 to 30 because 30 is already in use Apr 11
16:27:06 WARNING[10322]: chan_zap.c:9046 pri_dchannel: Unable to move
channel 30!
-- Channel 0/30, span 1 got hangup request Apr 11 16:27:06
WARNING[10966]: app_dial.c:706 wait_for_answer: Unable to forward voice
-- Hungup 'Zap/30-1'
  == Everyone is busy/congested at this time (1:0/0/1)
-- Executing NoOp("Zap/31-1", "CHANUNAVAIL") in new stack
-- Executing Busy("Zap/31-1", "") in new stack
-- Channel 0/31, span 1 got hangup request
  == Spawn extension (from-pstn, 111, 7) exited non-zero on 'Zap/31-1'
-- Executing NoOp("Zap/31-1", "") in new stack
-- Executing Goto("Zap/31-1", "999") in new stack
-- Goto (from-pstn,h,999)
-- Hungup 'Zap/31-1'



Could somebody give me a clue?


Pim

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RE: [Asterisk-Users] Double Call Progress tones

[Lee Archer]

I found progressinband=no in sip.conf fixed my problem when I had this.

Regards

Lee 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of kevin ling
Sent: 10 April 2006 12:24
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Double Call Progress tones

Hi,

I have the same problem on TE110P and Taiwan telco PRI line. I think to
fine tune the rigntone frequencies not resolve this problem. 

For example.
When I make a call to mobile. I can hear one ringtone like geneate by
asterisk or device. And another ringtone like from telco. You known,
some mobile will pickup the call and play music before user really
answer the call. So I can hear music and mix with ringtone. 

Regards,
kevin 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Simone
Cittadini
Sent: Wednesday, March 22, 2006 11:53 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Double Call Progress tones

Ron Wellsted ha scritto:

> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
>
> This is slowly driving me nuts!
>
> I have several Cisco 7960s with SIP 8.2/7.5 fw connecting to Asterisk
> 1.2.5 driving a TE110P on a BT EuroISDN PRI line.  On all outgoing 
> calls I get a double ring tone (UK style + US style).  I also have a 
> DECT phone on a Sipura SPA-3000 configured with UK tones.  This gives 
> me a double ring of UK + UK, so this suggests the call progress tones 
> are being generated by the SIP device.
>
> As a result I have edited sip.conf to set "progressinband=never" but 
> this has made no difference (even after a total restart).
>
> Previously I was running 1.0.7 without this problem, I upgraded to fix

> a problem with Monitor (the call stopped monitoring when transfered,
> 1.2.5 has fixed this).
>
> Does any one have any suggestions?
>
Configure the ringing frequencies on the sip devices so that is
something not udible by human hears (we did that as a quick fix before
discovering progressinband some time ago, worked for linksys pap)
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RE: [Asterisk-Users] Possible PRI fault?

[Lee Archer]

Inbound and outbound calls seem to work fine, but I noticed this due to
trying to figure out why these extra asterisk process as being started.
When the system was running * 1.0.10 I did notice that after a while
channels 1 and 2 would stop receiving and making calls.  The PRI is with
BT and on the other systems the PRI is also with BT and they don't have
this problem. 

Regards

Lee

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew
Kohlsmith
Sent: 04 April 2006 15:58
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Possible PRI fault?

On Tuesday 04 April 2006 10:39, Lee Archer wrote:
> I've been looking through the logs of a system trying to figure out 
> why it sometimes starts extra asterisk processes.  In the logs I keep 
> seeing

Define "starts extra asterisk processes."

> Apr  4 15:22:18 WARNING[5054] chan_zap.c: Can't fix up channel from 1 
> to
> 2 because 2 is already in use
> Apr  4 15:22:18 WARNING[5054] chan_zap.c: Unable to move channel 2!
> Apr  4 15:22:55 WARNING[5054] chan_zap.c: Can't fix up channel from 1 
> to
> 4 because 4 is already in use

This sounds like the telco is trying to specify which B channel to use.
My understanding is that Asterisk does not currently support this.
Asterisk chooses the B channel for outgoing calls.

Did it ever work, or is this a new problem?

-A.
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[Asterisk-Users] Possible PRI fault?

[Lee Archer]

Title: Possible PRI fault?






I've been looking through the logs of a system trying to figure out why it sometimes starts extra asterisk processes.  In the logs I keep seeing

Apr  4 15:22:18 WARNING[5054] chan_zap.c: Can't fix up channel from 1 to 2 because 2 is already in use

Apr  4 15:22:18 WARNING[5054] chan_zap.c: Unable to move channel 2!

Apr  4 15:22:55 WARNING[5054] chan_zap.c: Can't fix up channel from 1 to 4 because 4 is already in use

Apr  4 15:22:55 WARNING[5054] chan_zap.c: Unable to move channel 4!

Apr  4 15:26:49 WARNING[5054] chan_zap.c: Call specified, but not found?

Apr  4 15:26:49 WARNING[5054] chan_zap.c: Unable to move channel 1!

Apr  4 15:26:53 WARNING[5054] chan_zap.c: Call specified, but not found?

Apr  4 15:26:53 WARNING[5054] chan_zap.c: Unable to move channel 2!


Does this indicate a PRI problem?  I am running with a TE110P card and I have identical systems running that don't have this problem.

Regards


Lee


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RE: [Asterisk-Users] Loading module chan_zap.so failed! PLZ help me!

[Lee Archer]

context=from-pstn
signalling=fxs_ks
faxdetect=incoming
rxgain=6.0
txgain=2.0
usecallerid=yes
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
group=1
channel=1

Is my /etc/asterisk/zapata.conf

Lee 

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of ali asma
Sent: 04 April 2006 11:05
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Loading module chan_zap.so failed! PLZ help me!

ztcfg is ok, but asterisk still can't load chan_zap.so module


--- Lee Archer <[EMAIL PROTECTED]> a écrit :

> Try signalling=fxs_ks in /etc/asterisk/zapata.conf
> 
> Lee
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of ali asma
> Sent: 04 April 2006 10:50
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] Loading module chan_zap.so failed! PLZ 
> help me!
> 
> Sorry, now I have this:
> linux:~ # ztcfg -vv
> 
> Zaptel Configuration
> ==
> 
> 
> Channel map:
> 
> Channel 01: FXS Kewlstart (Default) (Slaves: 01)
> 
> 1 channels configured.
> 
> But the same error when running asterisk
> 
> 
> --- Lee Archer <[EMAIL PROTECTED]> a écrit :
> 
> > Just modprobe wcfxo then do ztcfg -vv after to see
> what it found.  Did
> > you make the changes to /etc/zaptel.conf?
> > 
> > Lee
> > 
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED]
> On Behalf Of ali asma
> > Sent: 04 April 2006 10:38
> > To: Asterisk Users Mailing List - Non-Commercial
> Discussion
> > Subject: RE: [Asterisk-Users] Loading module
> chan_zap.so failed! PLZ
> > help me!
> > 
> > I modified the configuration but I still have the
> same error.
> > Please tell me in whach directory should I
> execute:
> > modprobe zaptel
> > modprobe wcfxo
> > becose it seems that my card not had been detected
> > 
> > Thanks,
> > 
> > --- Lee Archer <[EMAIL PROTECTED]> a
> écrit :
> > 
> > > I run suse 10 and have an X100p.  But I use
> > fxsks=1 in the
> > > /etc/zaptel.conf not /etc/asterisk/zaptel.conf.
> > > 
> > > Lee
> > > 
> > > -Original Message-
> > > From: [EMAIL PROTECTED]
> > > [mailto:[EMAIL PROTECTED]
> > On Behalf Of ali asma
> > > Sent: 04 April 2006 10:13
> > > To: Asterisk Users Mailing List - Non-Commercial
> > Discussion
> > > Subject: RE: [Asterisk-Users] Loading module
> > chan_zap.so failed! PLZ
> > > help me!
> > > 
> > > Hi,
> > > Sorry my card is X101P. 
> > > My config is :
> > > 
> > > /etc/asterisk/zaptel.conf :
> > > loadzone=us
> > > defaultzone=us
> > > fxoks=1
> > > 
> > > and
> > > /etc/asterisk/zapata.conf :
> > > [trunkgroups]
> > > [channels]
> > > context=mainmenu
> > > signalling=fxo_ks
> > > faxdetect=incoming
> > > usecallerid=yes
> > > echocancel=yes
> > > echocancelwhenbridged=no
> > > echotraining=800
> > > language=en
> > > channel=>1
> > > 
> > > 
> > > please help me
> > > 
> > > 
> > > --- ali asma <[EMAIL PROTECTED]> a écrit :
> > > 
> > > > Hi,
> > > > I' ve just connected a carte X100M to my
> > asterisk
> > > server running
> > > > zaptel-1.2.5, libpri-1.2.2 and
> > > > asterisk-1.2.6 on SUSE 10.0.
> > > > When I make modprobe wcfxo and modprobe zaptel
> I
> > > haven't any error, I
> > > > have also chan_zap.so module existing in
> > > /usr/lib/asterisk/modules.
> > > > But, when i run ztcfg, it shows me this:
> > > > 
> > > > Zaptel Configuration
> > > > ==
> > > > Channel map:
> > > > 0 channels configured.
> > > > 
> > > > and when I run asterisk it shows me this:
> > > > 
> > > > Asterisk Dynamic Loader Starting:
> > > >   == Parsing '/etc/asterisk/modules.conf':
> Found
> > > [chan_zap.so]Apr  4
> > > > 09:45:58 WARNING[9975]:
> > > > loader.c:325 __load_resource:
> > > > /usr/lib/asterisk/modules/chan_zap.so:
> undefined
> > > > symbol: ast_pickup_call
> > > > Apr  4 09:45:58 WARNING[9975]: loader.c:499
> > > > load_modules: Loading module chan_zap.so
>

RE: [Asterisk-Users] Loading module chan_zap.so failed! PLZ help me!

[Lee Archer]

Try signalling=fxs_ks in /etc/asterisk/zapata.conf

Lee

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of ali asma
Sent: 04 April 2006 10:50
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Loading module chan_zap.so failed! PLZ help me!

Sorry, now I have this:
linux:~ # ztcfg -vv

Zaptel Configuration
==


Channel map:

Channel 01: FXS Kewlstart (Default) (Slaves: 01)

1 channels configured.

But the same error when running asterisk


--- Lee Archer <[EMAIL PROTECTED]> a écrit :

> Just modprobe wcfxo then do ztcfg -vv after to see what it found.  Did 
> you make the changes to /etc/zaptel.conf?
> 
> Lee
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of ali asma
> Sent: 04 April 2006 10:38
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] Loading module chan_zap.so failed! PLZ 
> help me!
> 
> I modified the configuration but I still have the same error.
> Please tell me in whach directory should I execute:
> modprobe zaptel
> modprobe wcfxo
> becose it seems that my card not had been detected
> 
> Thanks,
> 
> --- Lee Archer <[EMAIL PROTECTED]> a écrit :
> 
> > I run suse 10 and have an X100p.  But I use
> fxsks=1 in the
> > /etc/zaptel.conf not /etc/asterisk/zaptel.conf.
> > 
> > Lee
> > 
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED]
> On Behalf Of ali asma
> > Sent: 04 April 2006 10:13
> > To: Asterisk Users Mailing List - Non-Commercial
> Discussion
> > Subject: RE: [Asterisk-Users] Loading module
> chan_zap.so failed! PLZ
> > help me!
> > 
> > Hi,
> > Sorry my card is X101P. 
> > My config is :
> > 
> > /etc/asterisk/zaptel.conf :
> > loadzone=us
> > defaultzone=us
> > fxoks=1
> > 
> > and
> > /etc/asterisk/zapata.conf :
> > [trunkgroups]
> > [channels]
> > context=mainmenu
> > signalling=fxo_ks
> > faxdetect=incoming
> > usecallerid=yes
> > echocancel=yes
> > echocancelwhenbridged=no
> > echotraining=800
> > language=en
> > channel=>1
> > 
> > 
> > please help me
> > 
> > 
> > --- ali asma <[EMAIL PROTECTED]> a écrit :
> > 
> > > Hi,
> > > I' ve just connected a carte X100M to my
> asterisk
> > server running
> > > zaptel-1.2.5, libpri-1.2.2 and
> > > asterisk-1.2.6 on SUSE 10.0.
> > > When I make modprobe wcfxo and modprobe zaptel I
> > haven't any error, I
> > > have also chan_zap.so module existing in
> > /usr/lib/asterisk/modules.
> > > But, when i run ztcfg, it shows me this:
> > > 
> > > Zaptel Configuration
> > > ==
> > > Channel map:
> > > 0 channels configured.
> > > 
> > > and when I run asterisk it shows me this:
> > > 
> > > Asterisk Dynamic Loader Starting:
> > >   == Parsing '/etc/asterisk/modules.conf': Found
> > [chan_zap.so]Apr  4
> > > 09:45:58 WARNING[9975]:
> > > loader.c:325 __load_resource:
> > > /usr/lib/asterisk/modules/chan_zap.so: undefined
> > > symbol: ast_pickup_call
> > > Apr  4 09:45:58 WARNING[9975]: loader.c:499
> > > load_modules: Loading module chan_zap.so failed!
> > >  
> > > 
> > > Where do i look, how can i debug?
> > >  
> > >  Thanks in advance,
> > > 
> > > 
> > >   
> > > 
> > >   
> > >   
> > >
> >
>
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> > >   
> > >
> >
>
http://lists.digium.com/mailman/listinfo/asterisk-users
> > > 
> > 
> > 
> > 
> > 
> > 
> > 
> > 
> >
>
___
> > Nouvea

RE: [Asterisk-Users] Loading module chan_zap.so failed! PLZ help me!

[Lee Archer]

What does it say when you do a ztcfg -vv?  On my system I have

Loadzone = uk
Defaultzone = uk
fxsks=1

On a ztcfg -vv I get

Channel map:

Channel 01: FXS Kewlstart (Default) (Slaves: 01)

1 channels configured.

Lee

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of ali asma
Sent: 04 April 2006 10:45
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Loading module chan_zap.so failed! PLZ help me!

yes I make it but I still have the same error


--- Lee Archer <[EMAIL PROTECTED]> a écrit :

> Just modprobe wcfxo then do ztcfg -vv after to see what it found.  Did 
> you make the changes to /etc/zaptel.conf?
> 
> Lee
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of ali asma
> Sent: 04 April 2006 10:38
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] Loading module chan_zap.so failed! PLZ 
> help me!
> 
> I modified the configuration but I still have the same error.
> Please tell me in whach directory should I execute:
> modprobe zaptel
> modprobe wcfxo
> becose it seems that my card not had been detected
> 
> Thanks,
> 
> --- Lee Archer <[EMAIL PROTECTED]> a écrit :
> 
> > I run suse 10 and have an X100p.  But I use
> fxsks=1 in the
> > /etc/zaptel.conf not /etc/asterisk/zaptel.conf.
> > 
> > Lee
> > 
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED]
> On Behalf Of ali asma
> > Sent: 04 April 2006 10:13
> > To: Asterisk Users Mailing List - Non-Commercial
> Discussion
> > Subject: RE: [Asterisk-Users] Loading module
> chan_zap.so failed! PLZ
> > help me!
> > 
> > Hi,
> > Sorry my card is X101P. 
> > My config is :
> > 
> > /etc/asterisk/zaptel.conf :
> > loadzone=us
> > defaultzone=us
> > fxoks=1
> > 
> > and
> > /etc/asterisk/zapata.conf :
> > [trunkgroups]
> > [channels]
> > context=mainmenu
> > signalling=fxo_ks
> > faxdetect=incoming
> > usecallerid=yes
> > echocancel=yes
> > echocancelwhenbridged=no
> > echotraining=800
> > language=en
> > channel=>1
> > 
> > 
> > please help me
> > 
> > 
> > --- ali asma <[EMAIL PROTECTED]> a écrit :
> > 
> > > Hi,
> > > I' ve just connected a carte X100M to my
> asterisk
> > server running
> > > zaptel-1.2.5, libpri-1.2.2 and
> > > asterisk-1.2.6 on SUSE 10.0.
> > > When I make modprobe wcfxo and modprobe zaptel I
> > haven't any error, I
> > > have also chan_zap.so module existing in
> > /usr/lib/asterisk/modules.
> > > But, when i run ztcfg, it shows me this:
> > > 
> > > Zaptel Configuration
> > > ==
> > > Channel map:
> > > 0 channels configured.
> > > 
> > > and when I run asterisk it shows me this:
> > > 
> > > Asterisk Dynamic Loader Starting:
> > >   == Parsing '/etc/asterisk/modules.conf': Found
> > [chan_zap.so]Apr  4
> > > 09:45:58 WARNING[9975]:
> > > loader.c:325 __load_resource:
> > > /usr/lib/asterisk/modules/chan_zap.so: undefined
> > > symbol: ast_pickup_call
> > > Apr  4 09:45:58 WARNING[9975]: loader.c:499
> > > load_modules: Loading module chan_zap.so failed!
> > >  
> > > 
> > > Where do i look, how can i debug?
> > >  
> > >  Thanks in advance,
> > > 
> > > 
> > >   
> > > 
> > >   
> > >   
> > >
> >
>
___
> > > 
> > > Nouveau : téléphonez moins cher avec Yahoo!
> > > Messenger ! Découvez les tarifs exceptionnels
> pour
> > appeler la France
> > > et l'international.
> > > Téléchargez sur http://fr.messenger.yahoo.com 
> > > ___
> > > --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
> > > 
> > 
> > 
> > 
> > 
> > 
> > 
> > 
> >
>
___
> > Nouvea

RE: [Asterisk-Users] Loading module chan_zap.so failed! PLZ help me!

[Lee Archer]

Just modprobe wcfxo then do ztcfg -vv after to see what it found.  Did you make 
the changes to /etc/zaptel.conf?

Lee 

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of ali asma
Sent: 04 April 2006 10:38
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Loading module chan_zap.so failed! PLZ help me!

I modified the configuration but I still have the same error.
Please tell me in whach directory should I execute:
modprobe zaptel
modprobe wcfxo
becose it seems that my card not had been detected

Thanks,

--- Lee Archer <[EMAIL PROTECTED]> a écrit :

> I run suse 10 and have an X100p.  But I use fxsks=1 in the 
> /etc/zaptel.conf not /etc/asterisk/zaptel.conf.
> 
> Lee
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of ali asma
> Sent: 04 April 2006 10:13
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] Loading module chan_zap.so failed! PLZ 
> help me!
> 
> Hi,
> Sorry my card is X101P. 
> My config is :
> 
> /etc/asterisk/zaptel.conf :
> loadzone=us
> defaultzone=us
> fxoks=1
> 
> and
> /etc/asterisk/zapata.conf :
> [trunkgroups]
> [channels]
> context=mainmenu
> signalling=fxo_ks
> faxdetect=incoming
> usecallerid=yes
> echocancel=yes
> echocancelwhenbridged=no
> echotraining=800
> language=en
> channel=>1
> 
> 
> please help me
> 
> 
> --- ali asma <[EMAIL PROTECTED]> a écrit :
> 
> > Hi,
> > I' ve just connected a carte X100M to my asterisk
> server running
> > zaptel-1.2.5, libpri-1.2.2 and
> > asterisk-1.2.6 on SUSE 10.0.
> > When I make modprobe wcfxo and modprobe zaptel I
> haven't any error, I
> > have also chan_zap.so module existing in
> /usr/lib/asterisk/modules.
> > But, when i run ztcfg, it shows me this:
> > 
> > Zaptel Configuration
> > ==
> > Channel map:
> > 0 channels configured.
> > 
> > and when I run asterisk it shows me this:
> > 
> > Asterisk Dynamic Loader Starting:
> >   == Parsing '/etc/asterisk/modules.conf': Found
> [chan_zap.so]Apr  4
> > 09:45:58 WARNING[9975]:
> > loader.c:325 __load_resource:
> > /usr/lib/asterisk/modules/chan_zap.so: undefined
> > symbol: ast_pickup_call
> > Apr  4 09:45:58 WARNING[9975]: loader.c:499
> > load_modules: Loading module chan_zap.so failed!
> >  
> > 
> > Where do i look, how can i debug?
> >  
> >  Thanks in advance,
> > 
> > 
> > 
> > 
> > 
> > 
> >
>
___
> > 
> > Nouveau : téléphonez moins cher avec Yahoo!
> > Messenger ! Découvez les tarifs exceptionnels pour
> appeler la France
> > et l'international.
> > Téléchargez sur http://fr.messenger.yahoo.com 
> > ___
> > --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
> > 
> 
> 
> 
>   
> 
>   
>   
>
___
> Nouveau : téléphonez moins cher avec Yahoo!
> Messenger ! Découvez les tarifs exceptionnels pour appeler la France 
> et l'international.
> Téléchargez sur http://fr.messenger.yahoo.com 
> ___
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> --
> 
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>   
>
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> 
> 
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>   
>
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> 







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RE: [Asterisk-Users] Loading module chan_zap.so failed! PLZ help me!

[Lee Archer]

I run suse 10 and have an X100p.  But I use fxsks=1 in the /etc/zaptel.conf not 
/etc/asterisk/zaptel.conf.

Lee

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of ali asma
Sent: 04 April 2006 10:13
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Loading module chan_zap.so failed! PLZ help me!

Hi,
Sorry my card is X101P. 
My config is :

/etc/asterisk/zaptel.conf :
loadzone=us
defaultzone=us
fxoks=1

and
/etc/asterisk/zapata.conf :
[trunkgroups]
[channels]
context=mainmenu
signalling=fxo_ks
faxdetect=incoming
usecallerid=yes
echocancel=yes
echocancelwhenbridged=no
echotraining=800
language=en
channel=>1


please help me


--- ali asma <[EMAIL PROTECTED]> a écrit :

> Hi,
> I' ve just connected a carte X100M to my asterisk server running 
> zaptel-1.2.5, libpri-1.2.2 and
> asterisk-1.2.6 on SUSE 10.0.
> When I make modprobe wcfxo and modprobe zaptel I haven't any error, I 
> have also chan_zap.so module existing in /usr/lib/asterisk/modules.
> But, when i run ztcfg, it shows me this:
> 
> Zaptel Configuration
> ==
> Channel map:
> 0 channels configured.
> 
> and when I run asterisk it shows me this:
> 
> Asterisk Dynamic Loader Starting:
>   == Parsing '/etc/asterisk/modules.conf': Found  [chan_zap.so]Apr  4 
> 09:45:58 WARNING[9975]:
> loader.c:325 __load_resource:
> /usr/lib/asterisk/modules/chan_zap.so: undefined
> symbol: ast_pickup_call
> Apr  4 09:45:58 WARNING[9975]: loader.c:499
> load_modules: Loading module chan_zap.so failed!
>  
> 
> Where do i look, how can i debug?
>  
>  Thanks in advance,
> 
> 
>   
> 
>   
>   
>
___
> 
> Nouveau : téléphonez moins cher avec Yahoo!
> Messenger ! Découvez les tarifs exceptionnels pour appeler la France 
> et l'international.
> Téléchargez sur http://fr.messenger.yahoo.com 
> ___
> --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
> 







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exceptionnels pour appeler la France et l'international.
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RE: [Asterisk-Users] 1.2.6 doesn't use mpg123?

[Lee Archer]

What's the spec of the box?

Lee 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt Roth
Sent: 03 April 2006 18:48
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] 1.2.6 doesn't use mpg123?

Matt wrote:

>Ok.. see it... so now my question is  which should I use? 
>Obviously a hold system using ulaw for hold files is going to use less 
>CPU, but is it more stable to have Asterisk playing the sound files?
>Especially since it has to start a seperate stream for every on hold 
>person?  Seems like in a busy call center.. it would be more efficient 
>to have 1 stream going to every caller, rather then multiple streams.
>  
>
Matt,

Before switching our system from the rawplayer method to native MOH, I
consulted Kevin Fleming.  He said the impact on the system would be "not
much, more memory usage though." 


Right now we have seventy calls waiting in queues (all with native MOH)
and 120 calls connected to agents.  The box is jumping between 50%-60%
idle.  "ps auxm" shows 241 threads for Asterisk, but none of them take
more than 0.8% CPU.

Personally, I wouldn't mind seeing an option that allowed native MOH to
pull from a single thread for each class.  It would probably yield lower
CPU utilization, but I'm not sure how difficult it would be to
implement.  If enough people are interested in this feature, it should
be submitted through Mantis as a feature request.

Matthew Roth
InterMedia Marketing Solutions
Software Engineer and Systems Developer
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RE: Re[2]: [Asterisk-Users] 1.2.6 doesn't use mpg123?

[Lee Archer]

Madplay did work fine with * 1.0.

Lee 

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andre Ruiz
Sent: 03 April 2006 15:28
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: Re[2]: [Asterisk-Users] 1.2.6 doesn't use mpg123?

Mpg123 was the only way back in 1.0 versions.

In asterisk 1.2, they´ve implemented a new "format" module for mp3 codec, 
making it possible to do "native" mp3 streaming, but the old way is still ok.

When I first saw the native one, there were some limitations, like the
mp3 could not have any ID3 tag and had to be on certain rate, and be mono. I 
quickly discarded it because I would not like to convert my
mp3 collection to a new format, and would rather keep mpg123, which was fine.

Those limitations may be gone, please check. In that case, it may be better to 
use the native format due to CPU use concerns, but until testing I don´t think 
it´ll be much of a big difference.


andre

On 4/3/06, Matt <[EMAIL PROTECTED]> wrote:
> Ok.. see it... so now my question is  which should I use?
> Obviously a hold system using ulaw for hold files is going to use less 
> CPU, but is it more stable to have Asterisk playing the sound files?
> Especially since it has to start a seperate stream for every on hold 
> person?  Seems like in a busy call center.. it would be more efficient 
> to have 1 stream going to every caller, rather then multiple streams.
>
> On 4/1/06, Lee Archer <[EMAIL PROTECTED]> wrote:
> > Check the musiconhold.conf.sample in the asterisk/configs directory.
> > That will tell you what you need to know.
> >
> > Lee
> >
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] On Behalf Of Matt
> > Sent: 01 April 2006 16:41
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: Re: Re[2]: [Asterisk-Users] 1.2.6 doesn't use mpg123?
> >
> > How did you switch from native to mpg123 on 1.2.x?  That's what I 
> > can't figure out.
> >
> >
> > On 4/1/06, Lee Archer <[EMAIL PROTECTED]> wrote:
> > > Has anyone else had a problem with asterisk creating multiple threads?
> > > I'm still testing but I've move from native to mpg123 for the 
> > > machine with the problem and the problem hasn't come back.
> > >
> > > Lee
> > >
> > > -Original Message-
> > > From: [EMAIL PROTECTED]
> > > [mailto:[EMAIL PROTECTED] On Behalf Of Matt
> > > Sent: 01 April 2006 15:07
> > > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > > Subject: Re: Re[2]: [Asterisk-Users] 1.2.6 doesn't use mpg123?
> > >
> > > Ok this is great... but I just noticed this morning while doing 
> > > some tests that asterisk seems to start a new stream for every caller
> > > With mpg123 it would just start one and all calls would hear the same
> > > stream.Unless something was seriously lagging, my test calls this
> > > morning all were in different spots in the hold music.   Isn't this
> > > less efficient?
> > >
> > >
> > > On 4/1/06, Melcon Moraes <[EMAIL PROTECTED]> wrote:
> > > > You don't have to use it in newer versions. Get your mp3, ant 
> > > > convert to slin format with sox.
> > > >
> > > > Ex: sox -V file.mp3 [-c1] file.slin
> > > >
> > > > -V: just to show you what's going on
> > > > -c1: convert to 1 channel, if your mp3 is stereo
> > > >
> > > > Then edit your musiconhold.conf like this:
> > > >
> > > > [native]
> > > > mode=files
> > > > directory=/var/lib/asterisk/moh-native
> > > >
> > > > and you'll have a nice native streaming. You can convert your 
> > > > stuff to
> > >
> > > > another formats, like "sox file.mp3 [-c1] file.gsm" or "sox 
> > > > file.mp3
> >
> > > > [-c1] file.ul" and let asterisk decide which one best fits given
> > > channel.
> > > >
> > > > []'s
> > > > MM
> > > >
> > > >  -Original Message-
> > > > From:   "Lee Archer" <[EMAIL PROTECTED]>
> > > > To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> > > 
> > > > Cc:
> > > > Sent:  Sat, 1 Apr 2006 10:34:42 +0100
> > > > Delivered:  Sat,  01 Apr 2006 06:28:16 Subject:[Asterisk-Users]
> > > > 1.2.6 doesn

RE: Re[2]: [Asterisk-Users] 1.2.6 doesn't use mpg123?

[Lee Archer]

Have you tried madplay?  I used to use it instead of mpg123 for MOH.
Can't remember whether it does start a new process for every instance or
not.

Lee 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt
Sent: 03 April 2006 15:03
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: Re[2]: [Asterisk-Users] 1.2.6 doesn't use mpg123?

Ok.. see it... so now my question is  which should I use? 
Obviously a hold system using ulaw for hold files is going to use less
CPU, but is it more stable to have Asterisk playing the sound files? 
Especially since it has to start a seperate stream for every on hold
person?  Seems like in a busy call center.. it would be more efficient
to have 1 stream going to every caller, rather then multiple streams.

On 4/1/06, Lee Archer <[EMAIL PROTECTED]> wrote:
> Check the musiconhold.conf.sample in the asterisk/configs directory.
> That will tell you what you need to know.
>
> Lee
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Matt
> Sent: 01 April 2006 16:41
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: Re[2]: [Asterisk-Users] 1.2.6 doesn't use mpg123?
>
> How did you switch from native to mpg123 on 1.2.x?  That's what I 
> can't figure out.
>
>
> On 4/1/06, Lee Archer <[EMAIL PROTECTED]> wrote:
> > Has anyone else had a problem with asterisk creating multiple
threads?
> > I'm still testing but I've move from native to mpg123 for the 
> > machine with the problem and the problem hasn't come back.
> >
> > Lee
> >
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] On Behalf Of Matt
> > Sent: 01 April 2006 15:07
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: Re: Re[2]: [Asterisk-Users] 1.2.6 doesn't use mpg123?
> >
> > Ok this is great... but I just noticed this morning while doing some

> > tests that asterisk seems to start a new stream for every caller
> > With mpg123 it would just start one and all calls would hear the
same
> > stream.Unless something was seriously lagging, my test calls
this
> > morning all were in different spots in the hold music.   Isn't this
> > less efficient?
> >
> >
> > On 4/1/06, Melcon Moraes <[EMAIL PROTECTED]> wrote:
> > > You don't have to use it in newer versions. Get your mp3, ant 
> > > convert to slin format with sox.
> > >
> > > Ex: sox -V file.mp3 [-c1] file.slin
> > >
> > > -V: just to show you what's going on
> > > -c1: convert to 1 channel, if your mp3 is stereo
> > >
> > > Then edit your musiconhold.conf like this:
> > >
> > > [native]
> > > mode=files
> > > directory=/var/lib/asterisk/moh-native
> > >
> > > and you'll have a nice native streaming. You can convert your 
> > > stuff to
> >
> > > another formats, like "sox file.mp3 [-c1] file.gsm" or "sox 
> > > file.mp3
>
> > > [-c1] file.ul" and let asterisk decide which one best fits given
> > channel.
> > >
> > > []'s
> > > MM
> > >
> > >  -Original Message-
> > > From:   "Lee Archer" <[EMAIL PROTECTED]>
> > > To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> > 
> > > Cc:
> > > Sent:  Sat, 1 Apr 2006 10:34:42 +0100
> > > Delivered:  Sat,  01 Apr 2006 06:28:16 Subject:[Asterisk-Users]
> > > 1.2.6 doesn't use mpg123?
> > >
> > > I use mpg123 for streaming but I can't get it to compile under 
> > > SuSe10 and x86_64 CPU.  Does anyone have any recommendations for 
> > > other programs that allow streaming and will compile on this arch?
> > >
> > > Regards
> > >
> > > Lee
> > >
> > > -Original Message-
> > > From: [EMAIL PROTECTED]
> > > [mailto:[EMAIL PROTECTED] On Behalf Of Matt
> > > Sent: 31 March 2006 22:36
> > > To: [EMAIL PROTECTED]; Asterisk Users Mailing List - 
> > > Non-Commercial Discussion
> > > Subject: Re: [Asterisk-Users] 1.2.6 doesn't use mpg123?
> > >
> > > > >
> > > > And isn't mpg123 ( or some replacement ) required when using a 
> > > > stream for MOH I couldn't get streaming to work without it in
1.2?
> > >
> > > Yes.. mpg123 is required for streaming... I had it working in
> 1.0.9...

RE: Re[2]: [Asterisk-Users] 1.2.6 doesn't use mpg123?

[Lee Archer]

Check the musiconhold.conf.sample in the asterisk/configs directory.
That will tell you what you need to know.

Lee 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt
Sent: 01 April 2006 16:41
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: Re[2]: [Asterisk-Users] 1.2.6 doesn't use mpg123?

How did you switch from native to mpg123 on 1.2.x?  That's what I can't
figure out.


On 4/1/06, Lee Archer <[EMAIL PROTECTED]> wrote:
> Has anyone else had a problem with asterisk creating multiple threads?
> I'm still testing but I've move from native to mpg123 for the machine 
> with the problem and the problem hasn't come back.
>
> Lee
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Matt
> Sent: 01 April 2006 15:07
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: Re[2]: [Asterisk-Users] 1.2.6 doesn't use mpg123?
>
> Ok this is great... but I just noticed this morning while doing some 
> tests that asterisk seems to start a new stream for every caller
> With mpg123 it would just start one and all calls would hear the same
> stream.Unless something was seriously lagging, my test calls this
> morning all were in different spots in the hold music.   Isn't this
> less efficient?
>
>
> On 4/1/06, Melcon Moraes <[EMAIL PROTECTED]> wrote:
> > You don't have to use it in newer versions. Get your mp3, ant 
> > convert to slin format with sox.
> >
> > Ex: sox -V file.mp3 [-c1] file.slin
> >
> > -V: just to show you what's going on
> > -c1: convert to 1 channel, if your mp3 is stereo
> >
> > Then edit your musiconhold.conf like this:
> >
> > [native]
> > mode=files
> > directory=/var/lib/asterisk/moh-native
> >
> > and you'll have a nice native streaming. You can convert your stuff 
> > to
>
> > another formats, like "sox file.mp3 [-c1] file.gsm" or "sox file.mp3

> > [-c1] file.ul" and let asterisk decide which one best fits given
> channel.
> >
> > []'s
> > MM
> >
> >  -Original Message-
> > From:   "Lee Archer" <[EMAIL PROTECTED]>
> > To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> 
> > Cc:
> > Sent:  Sat, 1 Apr 2006 10:34:42 +0100
> > Delivered:  Sat,  01 Apr 2006 06:28:16 Subject:[Asterisk-Users] 
> > 1.2.6 doesn't use mpg123?
> >
> > I use mpg123 for streaming but I can't get it to compile under 
> > SuSe10 and x86_64 CPU.  Does anyone have any recommendations for 
> > other programs that allow streaming and will compile on this arch?
> >
> > Regards
> >
> > Lee
> >
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] On Behalf Of Matt
> > Sent: 31 March 2006 22:36
> > To: [EMAIL PROTECTED]; Asterisk Users Mailing List - 
> > Non-Commercial Discussion
> > Subject: Re: [Asterisk-Users] 1.2.6 doesn't use mpg123?
> >
> > > >
> > > And isn't mpg123 ( or some replacement ) required when using a 
> > > stream for MOH I couldn't get streaming to work without it in 1.2?
> >
> > Yes.. mpg123 is required for streaming... I had it working in
1.0.9...
> > though have not tried in 1.2.
> > ___
> > --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
> >
> >
> > ###
> >
> > This message has been scanned by F-Secure Anti-Virus for Microsoft
> Exchange.
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> > --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
> >
> > E-mail classificado pelo Identificador de Spam Inteligente Terra.
> > Para alterar a categoria classificada, visite 
> > http://mail.terra.com.br/protected_email/imail/imail.cgi?+_u=levelz&;
> > _l
> > =1,1143884189.96596.438.aldavila.hst.terra.com.br,5146,Des15,Des15
> >
> >
> >  --Original Message Ends--
> >
> > --
> > Melcon Moraes <[EMAIL PROTECTED]>
> >
> > ___

RE: Re[2]: [Asterisk-Users] 1.2.6 doesn't use mpg123?

[Lee Archer]

Has anyone else had a problem with asterisk creating multiple threads?
I'm still testing but I've move from native to mpg123 for the machine
with the problem and the problem hasn't come back.

Lee 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt
Sent: 01 April 2006 15:07
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: Re[2]: [Asterisk-Users] 1.2.6 doesn't use mpg123?

Ok this is great... but I just noticed this morning while doing some
tests that asterisk seems to start a new stream for every caller 
With mpg123 it would just start one and all calls would hear the same
stream.Unless something was seriously lagging, my test calls this
morning all were in different spots in the hold music.   Isn't this
less efficient?


On 4/1/06, Melcon Moraes <[EMAIL PROTECTED]> wrote:
> You don't have to use it in newer versions. Get your mp3, ant convert 
> to slin format with sox.
>
> Ex: sox -V file.mp3 [-c1] file.slin
>
> -V: just to show you what's going on
> -c1: convert to 1 channel, if your mp3 is stereo
>
> Then edit your musiconhold.conf like this:
>
> [native]
> mode=files
> directory=/var/lib/asterisk/moh-native
>
> and you'll have a nice native streaming. You can convert your stuff to

> another formats, like "sox file.mp3 [-c1] file.gsm" or "sox file.mp3 
> [-c1] file.ul" and let asterisk decide which one best fits given
channel.
>
> []'s
> MM
>
>  -Original Message-
> From:   "Lee Archer" <[EMAIL PROTECTED]>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"

> Cc:
> Sent:  Sat, 1 Apr 2006 10:34:42 +0100
> Delivered:  Sat,  01 Apr 2006 06:28:16 Subject:[Asterisk-Users] 1.2.6 
> doesn't use mpg123?
>
> I use mpg123 for streaming but I can't get it to compile under SuSe10 
> and x86_64 CPU.  Does anyone have any recommendations for other 
> programs that allow streaming and will compile on this arch?
>
> Regards
>
> Lee
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Matt
> Sent: 31 March 2006 22:36
> To: [EMAIL PROTECTED]; Asterisk Users Mailing List - 
> Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] 1.2.6 doesn't use mpg123?
>
> > >
> > And isn't mpg123 ( or some replacement ) required when using a 
> > stream for MOH I couldn't get streaming to work without it in 1.2?
>
> Yes.. mpg123 is required for streaming... I had it working in 1.0.9...
> though have not tried in 1.2.
> ___
> --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
>
>
> ###
>
> This message has been scanned by F-Secure Anti-Virus for Microsoft
Exchange.
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> To UNSUBSCRIBE or update options visit:
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> E-mail classificado pelo Identificador de Spam Inteligente Terra.
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> =1,1143884189.96596.438.aldavila.hst.terra.com.br,5146,Des15,Des15
>
>
>  --Original Message Ends--
>
> --
> Melcon Moraes <[EMAIL PROTECTED]>
>
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> To UNSUBSCRIBE or update options visit:
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RE: Re[2]: [Asterisk-Users] 1.2.6 doesn't use mpg123?

[Lee Archer]

I want to stream shoutcast etc. but mpg123 won't compile.  I use native
moh with files but it won't work with streams.

Lee 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Melcon
Moraes
Sent: 01 April 2006 14:33
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re[2]: [Asterisk-Users] 1.2.6 doesn't use mpg123?

You don't have to use it in newer versions. Get your mp3, ant convert to
slin format with sox. 

Ex: sox -V file.mp3 [-c1] file.slin

-V: just to show you what's going on
-c1: convert to 1 channel, if your mp3 is stereo

Then edit your musiconhold.conf like this:

[native]
mode=files
directory=/var/lib/asterisk/moh-native

and you'll have a nice native streaming. You can convert your stuff to
another formats, like "sox file.mp3 [-c1] file.gsm" or "sox file.mp3
[-c1] file.ul" and let asterisk decide which one best fits given
channel.

[]'s
MM

 -Original Message-
From:   "Lee Archer" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"

Cc: 
Sent:  Sat, 1 Apr 2006 10:34:42 +0100
Delivered:  Sat,  01 Apr 2006 06:28:16
Subject:[Asterisk-Users] 1.2.6 doesn't use mpg123?

I use mpg123 for streaming but I can't get it to compile under SuSe10
and x86_64 CPU.  Does anyone have any recommendations for other programs
that allow streaming and will compile on this arch? 

Regards

Lee

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt
Sent: 31 March 2006 22:36
To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: Re: [Asterisk-Users] 1.2.6 doesn't use mpg123?

> >
> And isn't mpg123 ( or some replacement ) required when using a stream 
> for MOH I couldn't get streaming to work without it in 1.2?

Yes.. mpg123 is required for streaming... I had it working in 1.0.9...
though have not tried in 1.2.
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 --Original Message Ends--

--
Melcon Moraes <[EMAIL PROTECTED]>

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RE: [Asterisk-Users] 1.2.6 doesn't use mpg123?

[Lee Archer]

I use mpg123 for streaming but I can't get it to compile under SuSe10
and x86_64 CPU.  Does anyone have any recommendations for other programs
that allow streaming and will compile on this arch? 

Regards

Lee

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt
Sent: 31 March 2006 22:36
To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: Re: [Asterisk-Users] 1.2.6 doesn't use mpg123?

> >
> And isn't mpg123 ( or some replacement ) required when using a stream 
> for MOH I couldn't get streaming to work without it in 1.2?

Yes.. mpg123 is required for streaming... I had it working in 1.0.9...
though have not tried in 1.2.
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[Asterisk-Users] Multiple processes

[Lee Archer]

Title: Multiple processes






Does anyone have any ideas why my recently updated * 1.2.5 system should spawn multiple * process at seemingly random intervals?

Regards


L:ee


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RE: [Asterisk-Users] Double-ring tone

[Lee Archer]

Why not just set it for the affected extensions in sip.conf?  I did it
globally and my GXP's didn't mind.

Lee

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Julian
Lyndon-Smith
Sent: 16 March 2006 09:41
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Double-ring tone

That's in the [general] section of sip.conf, yes ?

How would that affect the 7.4 phones ? Wouldn't want to annoy them ;)

Julian.

Lee Archer wrote:
> Could be the same problem I had with my Aastra - progressinband=no 
> worked for me.
> 
> Lee
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Julian 
> Lyndon-Smith
> Sent: 15 March 2006 18:10
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Double-ring tone
> 
> Not sure it's that weird :O
> 
> Douglas Garstang wrote:
>> The phone must have transported you to Australia... :)
>>
>> -Original Message-
>> From: Julian Lyndon-Smith [mailto:[EMAIL PROTECTED]
>> Sent: Wednesday, March 15, 2006 10:05 AM
>> To: asterisk Users Mailing List - Non-Commercial Discussion
>> Subject: [Asterisk-Users] Double-ring tone
>>
>>
>> I upgraded my Cisco7960 to SIP 8-2 from 7-4. Everything seems ok, 
>> works fine. Except that when I make an outbound call, I get a 
>> double-ring sound. I also found that if the target number is engaged,

>> I get a ring sound and at the same time get a busy sound.
>>
>> If I revert back to 7-4, there is no problem.
>>
>> Anyone else had this, or any clues on how to fix it ? All of our 
>> other
> 
>> phones are still on 7-4.
>>
>> TIA.
>>
>> Julian
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>>
> 
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RE: [Asterisk-Users] Double-ring tone

[Lee Archer]

Could be the same problem I had with my Aastra - progressinband=no
worked for me. 

Lee

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Julian
Lyndon-Smith
Sent: 15 March 2006 18:10
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Double-ring tone

Not sure it's that weird :O

Douglas Garstang wrote:
> The phone must have transported you to Australia... :)
> 
> -Original Message-
> From: Julian Lyndon-Smith [mailto:[EMAIL PROTECTED]
> Sent: Wednesday, March 15, 2006 10:05 AM
> To: asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] Double-ring tone
> 
> 
> I upgraded my Cisco7960 to SIP 8-2 from 7-4. Everything seems ok, 
> works fine. Except that when I make an outbound call, I get a 
> double-ring sound. I also found that if the target number is engaged, 
> I get a ring sound and at the same time get a busy sound.
> 
> If I revert back to 7-4, there is no problem.
> 
> Anyone else had this, or any clues on how to fix it ? All of our other

> phones are still on 7-4.
> 
> TIA.
> 
> Julian
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> 

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RE: [Asterisk-Users] How to change Budgetone dialtone?

[Lee Archer]

Sorry... Just ignore me.

Lee 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dmitry
Ivanov
Sent: 07 March 2006 14:16
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] How to change Budgetone dialtone?

On Tuesday 07 March 2006 15:49, Lee Archer wrote:
> Download the IP Phone Custom Ringtones Generation Tool Unzip and read 
> the readme

Ringtone != dialtone.


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RE: [Asterisk-Users] How to change Budgetone dialtone?

[Lee Archer]

Hi try http://www.grandstream.com/y-downloads.htm

Download the IP Phone Custom Ringtones Generation Tool
Unzip and read the readme

Regards

Lee 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dmitry
Ivanov
Sent: 07 March 2006 13:40
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] How to change Budgetone dialtone?

Good day!

Is is possible to change dialtone (and other tones as well) in BT-102?


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[Asterisk-Users] HDLC error

[Lee Archer]

Title: HDLC error






Can anyone help and point me in a useful direction.  I'm using * 1.2.4 with Zaptel 1.2.4.  I have a TE110P card and it’s a Supermicro P8SCT mobo.  If I run the PRI card in the PCI-X slot it shares an interupt with eth0 but I don't get problems.  I've been trying to move it onto it's own IRQ, by moving the card to a regular PCI slot but I now get the errors

Mar  2 08:43:09 NOTICE[6272]: chan_zap.c:8203 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1

Mar  2 08:43:20 NOTICE[6272]: chan_zap.c:8203 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1

I've done a few searches and tried a few things but still get it.  


My zaptel.conf looks like

span=1,1,0,ccs,hdb3

bchan=1-8

dchan=16


The system works but there is popping and the above messages.  I'd rather run the cards on different IRQ's but I'm not sure if it's the mobo I'm using or the something in the config I need to change.

Regards


Lee


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