For questions and discussion regarding the use and development of Asterisk, the free PBX in software

Jeremy Kister | 1 Jan 2011 05:34

Re: Base memory usage

On 12/31/2010 9:11 AM, Larry Wimble wrote:
> Removing modules one by one seemed to have virtually no effect until I
> got to chan_iax2.so.  Removing this module dropped memory consumption
> from 209mb to 16mb (looking at the RES column in the output of `top').

Apparently, it's a known issue: 
https://issues.asterisk.org/view.php?id=18194

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http://jeremy.kister.net./

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Gilles | 1 Jan 2011 18:30

Re: Base memory usage

On Fri, 31 Dec 2010 08:11:18 -0600, "Danny Nicholas"
<danny <at> debsinc.com> wrote:
>Incidently, is there a sure-fire way (eg. checking error messages in
>Asterisk's log file) to know which modules a given Asterisk setup
>needs, so we can safely not load unneeded modules?
>
>Check /var/log/asterisk/full from your last 1.6 startup.  The list of
>modules from there should be what you "need" in 1.8.

Thanks for the tip. The appliance doesn't log messages in ./full, but
I'll check how to enable it.

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Gilles | 1 Jan 2011 18:32

Re: Log and forward calls to cellphone?

On Wed, 29 Dec 2010 16:55:46 +0100, Administrator TOOTAI
<admin <at> tootai.net> wrote:
>I wouldn't be one of your friend: when I'm calling you I call a landline 
>but finally will be charged for a mobile call (imagine I have free calls 
>to landlines from my ISP). I give you an information: in France you 
>don't have the right to do this unless you have it precise *before* 
>redirection.

I checked with the VOSP: Apparently, it doesn't support getting an SIP
message to forward calls on the fly, and I pay for the forwarded leg
of the call (the caller will pay his part).

Thanks guys.

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bilal ghayyad | 1 Jan 2011 18:43

Saving the monitor file on new file always using Monitor(wav, Record1, m)

Dear List;

For each call (in specific case), I need to do a record and save in a spearated file, so I am thinking the best
thing is to save based on the time.

Monitor(wav,Record1,m)

So, how can I make the file name to be based on the current time (which is changed always, or based on the some
unique paramter (related to the call it self).

Any advise?

Regards
Bilal

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bilal ghayyad | 1 Jan 2011 18:50

Cisco IP Phones and AVAYA IP Phones: How to configure in Asterisk

Hi All;

How to configure the buttons in the Cisco IP Phones to be used for different functionalities like "Call
Forward, Call Pickup, ... etc"?

For example, if I need to assign one of the buttons existed at Cisco IP Phone to be used for CallFrw, how to do
this in Asterisk?

Regards
Bilal

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Robert Fantini | 1 Jan 2011 19:04

Re: Base memory usage

did you check :
/var/log/asterisk/full

On Sat, Jan 1, 2011 at 12:30 PM, Gilles <codecomplete <at> free.fr> wrote:

On Fri, 31 Dec 2010 08:11:18 -0600, "Danny Nicholas"
<danny <at> debsinc.com> wrote:
>Incidently, is there a sure-fire way (eg. checking error messages in
>Asterisk's log file) to know which modules a given Asterisk setup
>needs, so we can safely not load unneeded modules?
>
>Check /var/log/asterisk/full from your last 1.6 startup. The list of
>modules from there should be what you "need" in 1.8.

Thanks for the tip. The appliance doesn't log messages in ./full, but
I'll check how to enable it.

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Bryant Zimmerman | 1 Jan 2011 19:15

Re: DIALSTATUS on CANCEL

Vandar

I know understand what you are saying here. Once I turned on CEL I was able to see when and where each hangup was firing for each channel and the order of operations here. I am now moving very aggressively to get to CEL as I now see why CDR's are so broken. I have my CEL to CDR translator in testing and this is looking very promising.

Thanks for your help.
Bryant

From: BryantZ <at> zktech.com
Sent: Friday, December 24, 2010 9:28 AM
To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users <at> lists.digium.com>
Subject: Re: [asterisk-users] DIALSTATUS on CANCEL

If a call is hung up before an answer our "h" extension is not running in our dial macro

Bryant

On Dec 24, 2010, at 3:47 AM, Vardan Harutyunyan <hvardan71 <at> gmail.com> wrote:

> Hello Bryant
> Extension "h" is worked in any case of hangup.
> It not important to that the call was answered or no.
> It also be more flexible, if you use instead of ${DIALSTATUS}use ${HANGUPCAUSE}, while in most of cause ${DIALSTATUS} is returned the same return code.
> http://www.voip-info.org/wiki/view/Asterisk+variable+hangupcause
>
>
> --
> Vardan Harutyunyan,
> Senior System Administrator
>
> Enterprise Incubator Foundation
> 123 Hovsep Emin Street,
> Yerevan 0051, Republic of Armenia
> Tel: + 374 10 219735
> Fax: + 374 10 219777
> E-mail: info <at> eif.am
> www.eif-it.com
>
> Bryant Zimmerman wrote:
>> Vardan
>>
>> I have not use AEL so it is a bit hard to follow with the formatting the
>> way it is but it looks like correct.
>> Please note the "h" extension only appears to run if a call is connected
>> so I do not know when the "CANCEL" would ever be set.
>> There may be someone else who can speak to this. It also appears thet
>> ${DIALSTATUS} may not be set if the call is not allowed to time out or
>> dialed. To me it would make sense to set the inital state of the
>> ${DIALSTATUS} to CANCEL and if nothing changes it that would stand, but
>> I may be missing the point on this can anyone else speak to it?
>>
>> Bryant
>>
>> ------------------------------------------------------------------------
>> *From*: "Vardan Harutyunyan" <hvardan71 <at> gmail.com>
>> *Sent*: Thursday, December 23, 2010 2:11 AM
>> *To*: asterisk-users <at> lists.digium.com
>> *Subject*: Re: [asterisk-users] DIALSTATUS on CANCEL
>>
>> I have make test in AEL.
>>
>> context fu {
>>
>> _000./userN => {
>> Dial(SIP/${EXTEN:3} <at> Prov);
>> Noop(${DIALSTATUS});
>> };
>> h => {
>> Noop(${DIALSTATUS});
>> };
>> };
>>
>> And look CLI
>> -- Executing [00018185402020 <at> fu:1] NoOp("SIP/userN-b6317738", "")
>> in new stack
>> -- Executing [00018185402020 <at> fo:2] Dial("SIP/user3-b6317738",
>> "SIP/18185402020 <at> Prov") in new stack
>> -- Called 18185402020 <at> Prov
>> -- SIP/Prov-082a83b8 is making progress passing it to
>> SIP/userN-b6317738
>> == Spawn extension (fu, 00018185402020, 2) exited non-zero on
>> 'SIP/user3-b6317738'
>> -- Executing [h <at> fu:1] NoOp("SIP/userN-b6317738", "CANCEL") in new stack
>>
>> I think, I am right
>>
>> --
>> Vardan Harutyunyan,
>> Senior System Administrator
>>
>> Enterprise Incubator Foundation
>> 123 Hovsep Emin Street,
>> Yerevan 0051, Republic of Armenia
>> Tel: + 374 10 219735
>> Fax: + 374 10 219777
>> E-mail: info <at> eif.am
>> www.eif-it.com
>>
>> Bryant Zimmerman wrote:
>>> The Dial Status is not set when accessing it from the h extension.
>>>
>>> Bryant
>>>
>>> ------------------------------------------------------------------------
>>> *From*: "Vardan Harutyunyan" <hvardan71 <at> gmail.com>
>>> *Sent*: Wednesday, December 22, 2010 10:39 AM
>>> *To*: asterisk-users <at> lists.digium.com
>>> *Subject*: Re: [asterisk-users] DIALSTATUS on CANCEL
>>>
>>> Try to use h extension
>>>
>>> --
>>> Vardan Harutyunyan,
>>> Senior System Administrator
>>>
>>> Enterprise Incubator Foundation
>>> 123 Hovsep Emin Street,
>>> Yerevan 0051, Republic of Armenia
>>> Tel: + 374 10 219735
>>> Fax: + 374 10 219777
>>> E-mail: info <at> eif.am
>>> www.eif-it.com
>>>
>>> Michael wrote:
>>> > Hi Nikhil,
>>> >
>>> > Both debug and verbose are set to 20. That's all I got, but as you can
>>> > see, for the other types of reasons, the DIALSTATUS got a value (and we
>>> > see the events). I'm pretty sure it's a bug.
>>> >
>>> > Michael
>>> >
>>> > On Wed, Dec 22, 2010 at 9:01 AM, Nikhil <d.nikhil <at> cem-solutions..net
>>> > <mailto:d.nikhil <at> cem-solutions.net>> wrote:
>>> >
>>> > Hi
>>> > Enable debug level to more than 1 ,you may get something.
>>> >
>>> > Thanks
>>> > Nikhil
>>> >
>>> > On 12/22/2010 11:26 AM, Michael wrote:
>>> >
>>> > Spawn extension (incoming-private, 11111111, 3) exited non-zero
>>> > on 'SIP/Proxy-00000031'
>>> >
>>> >
>>> >
>>> >
>>> > --
>>> > _____________________________________________________________________
>>> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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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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>>
>>
>> --
>> _____________________________________________________________________
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>>
>>
>> --
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>
>
> --
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Nic Colledge | 1 Jan 2011 19:36

Re: Saving the monitor file on new file always using Monitor(wav, Record1, m)

Try using ${UNIQUEID} to get the unique id of the current call. That or something like CDR(uniqueid).
Forget which off the top of my head.
Nic.
-----Original Message-----
From: asterisk-users-bounces <at> lists.digium.com
[mailto:asterisk-users-bounces <at> lists.digium.com] On Behalf Of bilal ghayyad
Sent: 01 January 2011 17:43
To: asterisk-users <at> lists.digium.com
Subject: [asterisk-users] Saving the monitor file on new file always using Monitor(wav, Record1, m)

Dear List;

For each call (in specific case), I need to do a record and save in a spearated file, so I am thinking the best
thing is to save based on the time.

Monitor(wav,Record1,m)

So, how can I make the file name to be based on the current time (which is changed always, or based on the some
unique paramter (related to the call it self).

Any advise?

Regards
Bilal

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Steve Edwards | 1 Jan 2011 19:50

Re: Saving the monitor file on new file always using Monitor(wav, Record1, m)

On Sat, 1 Jan 2011, bilal ghayyad wrote:

> For each call (in specific case), I need to do a record and save in a 
> spearated file, so I am thinking the best thing is to save based on the 
> time.

Read up on the STRFTIME function.

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-- 
Thanks in advance,
-------------------------------------------------------------------------
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Newline                                              Fax: +1-760-731-3000

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Bryant Zimmerman | 1 Jan 2011 19:46

Re: Saving the monitor file on new file always using Monitor(wav, Record1, m)

Use a combination of ${EPOCH} with a format string and the unique call / channel id.

Example:

exten => s,1,Set(MY_TIMEVAR=:${STRFTIME(${EPOCH},,%d%mNaVH:NaVS)})
exten => s,n,Monitor(wav,${MY_TIMEVAR}~${CHANNEL},m)

From: "bilal ghayyad" <bilmar_gh <at> yahoo.com>
Sent: Saturday, January 01, 2011 1:16 PM
To: asterisk-users <at> lists.digium.com
Subject: [asterisk-users] Saving the monitor file on new file always using Monitor(wav, Record1, m)

Dear List;

For each call (in specific case), I need to do a record and save in a spearated file, so I am thinking the best thing is to save based on the time.

Monitor(wav,Record1,m)

So, how can I make the file name to be based on the current time (which is changed always, or based on the some unique paramter (related to the call it self).

Any advise?

Regards
Bilal

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