Re: [asterisk-users] play sound file

[Nasir Iqbal]

check this http://www.voip-info.org/wiki/view/Asterisk+cmd+ControlPlayback

Nasir Iqbal

ICTBroadcast
SMS, Fax and Voice broadcasting solution
http://www.ictbroadcast.com/



On Wed, Jan 25, 2012 at 8:29 PM, Eyal  wrote:

> Hi,
>
> How can I play a sound file from the middle and end it after a certain
> number of seconds?
>
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Re: [asterisk-users] play sound file

[Eyal]

Thanks

 

But this is not what I am looking for, in this way I can start the sound
file from some point in the file but the callers must hear the file
until the end.

I need something that allows me to start from some place in the file and
end it in some other place in the file (say song from time 01:32 until
01:57),

Or

Like the controlplayback doing fast-forward but without having to click
any key by caller.

 

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nasir
Iqbal
Sent: Thursday, January 26, 2012 10:53 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] play sound file

 

check this
http://www.voip-info.org/wiki/view/Asterisk+cmd+ControlPlayback


Nasir Iqbal

ICTBroadcast

SMS, Fax and Voice broadcasting solution

http://www.ictbroadcast.com/





On Wed, Jan 25, 2012 at 8:29 PM, Eyal  wrote:

Hi,

How can I play a sound file from the middle and end it after a certain
number of seconds?


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Re: [asterisk-users] play sound file

[Sammy Govind]

You can use a combination of ChanSpy() and a local extension playing the
required file to caller/callee.

On Thu, Jan 26, 2012 at 2:11 PM, Eyal  wrote:

> Thanks
>
> ** **
>
> But this is not what I am looking for, in this way I can start the sound
> file from some point in the file but the callers must hear the file until
> the end.
>
> I need something that allows me to start from some place in the file and
> end it in some other place in the file (say song from time 01:32 until
> 01:57),
>
> Or
>
> Like the *controlplayback* doing fast-forward but without having to click
> any key by caller.
>
> ** **
>
> ** **
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Nasir Iqbal
> *Sent:* Thursday, January 26, 2012 10:53 AM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] play sound file
>
> ** **
>
> check this http://www.voip-info.org/wiki/view/Asterisk+cmd+ControlPlayback
> 
>
>
> Nasir Iqbal
>
> ICTBroadcast
>
> SMS, Fax and Voice broadcasting solution
>
> http://www.ictbroadcast.com/
>
>
>
> 
>
> On Wed, Jan 25, 2012 at 8:29 PM, Eyal  wrote:
>
> Hi,
>
> How can I play a sound file from the middle and end it after a certain
> number of seconds?
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>
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> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
> ** **
>
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Re: [asterisk-users] play sound file

[Johan Wilfer]

2012-01-26 10:11, Eyal skrev:
>
> Thanks
>
>  
>
> But this is not what I am looking for, in this way I can start the
> sound file from some point in the file but the callers must hear the
> file until the end.
>
> I need something that allows me to start from some place in the file
> and end it in some other place in the file (say song from time 01:32
> until 01:57),
>
> Or
>
> Like the *controlplayback* doing fast-forward but without having to
> click any key by caller.
>

You can do that by combining ControlPlayback and use TIMEOUT function.

If you don't want the user to be able to use any keys you can use all
keys as stop-keys in ControlPlayback and have some logic restart the
playback at position ${CPLAYBACKOFFSET}.

For more details do:

core show application ControlPlayback
core show function TIMEOUT

/Johan



>  
>
>  
>
> *From:*asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Nasir
> Iqbal
> *Sent:* Thursday, January 26, 2012 10:53 AM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] play sound file
>
>  
>
> check this http://www.voip-info.org/wiki/view/Asterisk+cmd+ControlPlayback
>
>
> Nasir Iqbal
>
> ICTBroadcast
>
> SMS, Fax and Voice broadcasting solution
>
> http://www.ictbroadcast.com/
>
>
>
> On Wed, Jan 25, 2012 at 8:29 PM, Eyal  > wrote:
>
> Hi,
>
> How can I play a sound file from the middle and end it after a certain
> number of seconds?
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
>  
>
>
>
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-- 
Johan Wilfer email: jo...@jttech.se
JT Tech | Developer  webb: http://jttech.se

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Re: [asterisk-users] dialplan problem : not including context

[Jonas Kellens]

On 01/13/2012 06:58 PM, Administrator TOOTAI wrote:

Le 13/01/2012 14:32, Jonas Kellens a écrit :

On 01/13/2012 02:23 PM, Doug Lytle wrote:


Jonas Kellens wrote:

I have the following in dialplan :


[TrunkAccounts]


dialplan show TrunkAccounts

Make sure the sort order is what you're expecting.

Doug


Hello,

The order is correct for as far as I'm sure.

[TrunkAccounts]

exten => 32380837,1,GoTo(01,32380837,1)
exten => 32380838,1,GoTo(01,32380838,1)
exten => 32380839,1,GoTo(01,32380839,1)

[CheckOnNet]

include => TrunkAccounts

exten => _321[0-3],1,GoTo(context1,${EXTEN},1)

exten => 3214,1,GoTo(context2, ${EXTEN} ,1)

exten => _.,1,NoOp()
exten => _.,n,Return()


Are you sure about your _. exten? Typo in the mail? It means 9 
and more digits but your extensions are 8 digits ...


Include are always treated *after* context command. If _. is 
right, something is wrong with Asterisk as it should treat 
TrunkAccounts. If _XXX. (8 digits or more) is what you have in 
yourdialplan, than the behavior of Asterisk is OK


Try

[TrunkAccounts]

exten => 32380837,1,GoTo(01,32380837,1)
exten => 32380838,1,GoTo(01,32380838,1)
exten => 32380839,1,GoTo(01,32380839,1)

[TrunkNotTreated]

exten => _.,1,NoOp()
exten => _.,n,Return()

[CheckOnNet]

include => TrunkAccounts
include => TrunkNotTreated

exten => _321[0-3],1,GoTo(context1,${EXTEN},1)
exten => 3214,1,GoTo(context2, ${EXTEN} ,1)

[...]



Hello,

I confirm that this is working for me !

Thanks !

Jonas.



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Re: [asterisk-users] ChanSpy : how to know channel name ?

[Jonas Kellens]

On 01/25/2012 11:10 AM, Ishfaq Malik wrote:

I use ChanSpy successfully all the time. You do not have to specify the
full channel, just the prefix which is the peer name. As you can see it
also states 'This includes the audio  coming in and out of the channel
being spied on.'


I confirm that ChanSpy does not need the complete channel name. The peer 
name is enough for ChanSpy to find the channel.


I need further testing though to find out if both legs of the 
conversation can be listened to.


Thanks.

Jonas.

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Re: [asterisk-users] Pickup calls coming from queues

[Niccolò Belli]

Il 25/01/2012 22:52, Michael Keuter ha scritto:

Outcry! :-)


I'm outcrying too :)

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[asterisk-users] User hit f to disconnect call.

[Vieri]

Hi,

I was receiving fax calls just fine until recently. I'm now having random 
disconnections.

Faxes are received over an ISDN BRI line and Asterisk 1.4 detects it and sends 
it to a iaxmodem (exten 10025 below). All's apparently as expected except for 
the fact that the following message comes up in the Asterisk log:

User hit f to disconnect call.

The iaxmodem log also shows a premature hangup (see below).

I did a test fax call but I certainly didn't press any key to abort the call. 
What does that message mean?

Asterisk log (0X is destination, Y is sending fax machine):

[Jan 26 13:46:13] VERBOSE[619] logger.c: -- Executing 
[fax@from-pstn-deviate-custom:12] Dial("mISDN/6-u22326", 
"IAX2/10025/0971847022|20|d") in new stack
[Jan 26 13:46:13] DEBUG[619] chan_iax2.c: prepending 8 to prefs
[Jan 26 13:46:13] VERBOSE[15361] logger.c: -- Call accepted by 127.0.0.1 
(format alaw)
[Jan 26 13:46:13] VERBOSE[15361] logger.c: -- Format for call is alaw
[Jan 26 13:46:13] VERBOSE[619] logger.c: -- Called 10025/0X
[Jan 26 13:46:13] VERBOSE[619] logger.c: -- IAX2/10025-3460 is ringing
[Jan 26 13:46:13] VERBOSE[619] logger.c: -- User hit f to disconnect call.
[Jan 26 13:46:13] VERBOSE[619] logger.c: -- Hungup 'IAX2/10025-3460'
[Jan 26 13:46:13] VERBOSE[619] logger.c:   == Spawn extension 
(from-pstn-deviate-custom, f, 0) exited non-zero on 'mISDN/6-u22326'
[Jan 26 13:46:13] VERBOSE[619] logger.c: -- Executing 
[h@from-pstn-deviate-custom:1] Macro("mISDN/6-u22326", "hangupcall") in new 
stack

iaxmodem log:

[2012-01-26 13:46:13] Incoming call connected 0X, Y, (null).
[2012-01-26 13:46:13] Answering
[2012-01-26 13:46:13] Remote hangup.
[2012-01-26 13:46:14] Hanging Up
[2012-01-26 13:46:19] Hanging Up
[2012-01-26 13:46:22] Taking receiver off-hook.
[2012-01-26 13:46:22] Hanging Up

Thanks,

Vieri


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[asterisk-users] Softphones with SIP transfer

[Agustina Berretta]

Hello

how are you?
Can you give me advice on which are the best free or not (free prefered)
that use SIP Transfer.

Thanks a lot!!!
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[asterisk-users] Manager Originate and Callerid ?

[Russell Brown]

I'm using Manager API Originate to initiate calls from SIP channels (via
phpagi FWIW) and it all works well except

...the CallerID for the SIP channel specified in users.conf isn't set for
the call :-(

If I explicitly set the Callerid in the Manager Originate API call then
it works but the API is actually being run from another server which
doesn't 'know' the correct Callerid number and name for any given SIP
phone so can't set them.

I'm calling the Manager API with the following:-

Action: Originate
Channel: SIP/101
Context: from-sip
Exten: 01234567890
Priority: 1
Timeout: 2
ActionID: foo

This results in the Callerid(name) and Callerid(num) being blank for the
call.

The 'from-sip' context is exactly the same as my SIP phones are using
and when manually dialing the Callerid info is correctly picked up from
users.conf.

Any ideas why this is and how I can get the Manager API Originate call
to use the correct Callerid info?

-- 
 Regards,
 Russell
 
| Russell Brown  | MAIL: russ...@lls.com PHONE: 01780 471800 |
| Lady Lodge Systems | WWW Work: http://www.lls.com  |
| Peterborough, England  | WWW Play: http://www.ruffle.me.uk |
 

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Re: [asterisk-users] Sip Registration Hijacking

[Paul Hayes]

On 20/01/12 01:36, eherr wrote:


It is also register on an AudioCodes MP-118.



Thanks,

-E

Is the Audiocodes gateway accessible online?  Have you set a strong 
password for it's web interface (and cli if it has one)?  It is possible 
someone is breaking into that and getting the SIP password out of it.


cheers,
Paul.

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[asterisk-users] SLA for DAHDI FXO - Emulating Key System Functionality

[Tim Nelson]

Greetings-

I currently have a customer that *requires* key-system functionality in an 
Asterisk PBX. On a SIP phone, the BLF keys need to show the current state of 
the analog lines attached to the system (DAHDI FXO). By pressing one of these 
keys (for line 1 for example), the dialed number needs to be dialed out the 
correct port. Also, when that line is busy, the phone BLF key for that line 
needs to reflect the status.

I've been reading about SLATrunk but it doesn't seem quite what I'm looking 
for. Also, I'm looking at using Hints to supply such information, but I'm not 
sure exactly how it should look.

Has anyone done this before, and if so, how did you implement it? My target is 
to use Asterisk 1.8 but another version would suffice.

Looking forward to your comments!

--Tim

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Re: [asterisk-users] SLA for DAHDI FXO - Emulating Key System Functionality

[Kevin P. Fleming]

On 01/26/2012 09:46 AM, Tim Nelson wrote:

Greetings-

I currently have a customer that *requires* key-system functionality in an 
Asterisk PBX. On a SIP phone, the BLF keys need to show the current state of 
the analog lines attached to the system (DAHDI FXO). By pressing one of these 
keys (for line 1 for example), the dialed number needs to be dialed out the 
correct port. Also, when that line is busy, the phone BLF key for that line 
needs to reflect the status.

I've been reading about SLATrunk but it doesn't seem quite what I'm looking 
for. Also, I'm looking at using Hints to supply such information, but I'm not 
sure exactly how it should look.

Has anyone done this before, and if so, how did you implement it? My target is 
to use Asterisk 1.8 but another version would suffice.


The SLA functionality is exactly what you are looking for.

--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com & www.asterisk.org

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Re: [asterisk-users] User hit f to disconnect call.

[Kevin P. Fleming]

On 01/26/2012 07:22 AM, Vieri wrote:

Hi,

I was receiving fax calls just fine until recently. I'm now having random 
disconnections.

Faxes are received over an ISDN BRI line and Asterisk 1.4 detects it and sends 
it to a iaxmodem (exten 10025 below). All's apparently as expected except for 
the fact that the following message comes up in the Asterisk log:

User hit f to disconnect call.

The iaxmodem log also shows a premature hangup (see below).

I did a test fax call but I certainly didn't press any key to abort the call. 
What does that message mean?

Asterisk log (0X is destination, Y is sending fax machine):

[Jan 26 13:46:13] VERBOSE[619] logger.c: -- Executing [fax@from-pstn-deviate-custom:12] 
Dial("mISDN/6-u22326", "IAX2/10025/0971847022|20|d") in new stack
[Jan 26 13:46:13] DEBUG[619] chan_iax2.c: prepending 8 to prefs
[Jan 26 13:46:13] VERBOSE[15361] logger.c: -- Call accepted by 127.0.0.1 
(format alaw)
[Jan 26 13:46:13] VERBOSE[15361] logger.c: -- Format for call is alaw
[Jan 26 13:46:13] VERBOSE[619] logger.c: -- Called 10025/0X
[Jan 26 13:46:13] VERBOSE[619] logger.c: -- IAX2/10025-3460 is ringing
[Jan 26 13:46:13] VERBOSE[619] logger.c: -- User hit f to disconnect call.
[Jan 26 13:46:13] VERBOSE[619] logger.c: -- Hungup 'IAX2/10025-3460'
[Jan 26 13:46:13] VERBOSE[619] logger.c:   == Spawn extension 
(from-pstn-deviate-custom, f, 0) exited non-zero on 'mISDN/6-u22326'
[Jan 26 13:46:13] VERBOSE[619] logger.c: -- Executing [h@from-pstn-deviate-custom:1] 
Macro("mISDN/6-u22326", "hangupcall") in new stack


'f' is the fake DTMF control frame used inside Asterisk to indicate that 
a CNG tone was detected. Do you have 'faxdetect' enabled on the mISDN 
channel driver for that BRI?


Even if you do, though, I don't know why receiving an 'f' would 
disconnect the call, unless you've provided the 'd' option to app_dial. 
Even if you did, app_dial should be smart enough to not treat 'f' as a 
DTMF key, but it's not (at least not in Asterisk 1.4, this may have 
changed in later versions).


--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com & www.asterisk.org

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Re: [asterisk-users] Manager Originate and Callerid ?

[Kevin P. Fleming]

On 01/26/2012 09:23 AM, Russell Brown wrote:


I'm using Manager API Originate to initiate calls from SIP channels (via
phpagi FWIW) and it all works well except

...the CallerID for the SIP channel specified in users.conf isn't set for
the call :-(

If I explicitly set the Callerid in the Manager Originate API call then
it works but the API is actually being run from another server which
doesn't 'know' the correct Callerid number and name for any given SIP
phone so can't set them.

I'm calling the Manager API with the following:-

Action: Originate
Channel: SIP/101
Context: from-sip
Exten: 01234567890
Priority: 1
Timeout: 2
ActionID: foo

This results in the Callerid(name) and Callerid(num) being blank for the
call.

The 'from-sip' context is exactly the same as my SIP phones are using
and when manually dialing the Callerid info is correctly picked up from
users.conf.

Any ideas why this is and how I can get the Manager API Originate call
to use the correct Callerid info?


It can't. The Caller ID is provided by chan_sip when an incoming call 
*arrives* over SIP, and it can match the caller to the entry in sip.conf 
(or users.conf in your case).


An outgoing call originated by AMI will not be matched up to any 
sip.conf entries for Caller ID purposes.


--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com & www.asterisk.org

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Re: [asterisk-users] SLA for DAHDI FXO - Emulating Key System Functionality

[Tim Nelson]

- Original Message -
> On 01/26/2012 09:46 AM, Tim Nelson wrote:
> > Greetings-
> >
> > I currently have a customer that *requires* key-system functionality
> > in an Asterisk PBX. On a SIP phone, the BLF keys need to show the
> > current state of the analog lines attached to the system (DAHDI
> > FXO). By pressing one of these keys (for line 1 for example), the
> > dialed number needs to be dialed out the correct port. Also, when
> > that line is busy, the phone BLF key for that line needs to reflect
> > the status.
> >
> > I've been reading about SLATrunk but it doesn't seem quite what I'm
> > looking for. Also, I'm looking at using Hints to supply such
> > information, but I'm not sure exactly how it should look.
> >
> > Has anyone done this before, and if so, how did you implement it? My
> > target is to use Asterisk 1.8 but another version would suffice.
> 
> The SLA functionality is exactly what you are looking for.
> 

Fantastic, I'll revisit the usage and do some testing. However, can anyone 
point me to some working examples?

--Tim

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Re: [asterisk-users] SLA for DAHDI FXO - Emulating Key System Functionality

[Jeff LaCoursiere]

On Thu, 2012-01-26 at 10:48 -0600, Tim Nelson wrote:
> - Original Message -
> > On 01/26/2012 09:46 AM, Tim Nelson wrote:
> > > Greetings-
> > >
> > > I currently have a customer that *requires* key-system functionality
> > > in an Asterisk PBX. On a SIP phone, the BLF keys need to show the
> > > current state of the analog lines attached to the system (DAHDI
> > > FXO). By pressing one of these keys (for line 1 for example), the
> > > dialed number needs to be dialed out the correct port. Also, when
> > > that line is busy, the phone BLF key for that line needs to reflect
> > > the status.
> > >
> > > I've been reading about SLATrunk but it doesn't seem quite what I'm
> > > looking for. Also, I'm looking at using Hints to supply such
> > > information, but I'm not sure exactly how it should look.
> > >
> > > Has anyone done this before, and if so, how did you implement it? My
> > > target is to use Asterisk 1.8 but another version would suffice.
> > 
> > The SLA functionality is exactly what you are looking for.
> > 
> 
> Fantastic, I'll revisit the usage and do some testing. However, can anyone 
> point me to some working examples?
> 
> --Tim
> 

I'm also very interested in working examples, especially if someone has
set it up for SIP termination "trunks" rather than Dahdi.

Thanks,

j


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Re: [asterisk-users] Sip Registration Hijacking

[eherr]

It is accessible from HTTP.

However, the access list only allows access from my home and the password is 
strong.

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paul Hayes
Sent: Thursday, January 26, 2012 10:30 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Sip Registration Hijacking

On 20/01/12 01:36, eherr wrote:
>
> It is also register on an AudioCodes MP-118.

> Thanks,
>
> -E
>
Is the Audiocodes gateway accessible online?  Have you set a strong 
password for it's web interface (and cli if it has one)?  It is possible 
someone is breaking into that and getting the SIP password out of it.

cheers,
Paul.

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Re: [asterisk-users] SLA for DAHDI FXO - Emulating Key System Functionality

[Patrick Lists]

On 26-01-12 18:08, Jeff LaCoursiere wrote:
[snip]


I'm also very interested in working examples, especially if someone has
set it up for SIP termination "trunks" rather than Dahdi.


Maybe I am missing something here but why would you want to emulate a 
keysystem with analog (thus single call) lines on a SIP trunk which 
support multiple simultaneous calls on that single trunk?


Regards,
Patrick


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Re: [asterisk-users] SLA for DAHDI FXO - Emulating Key System Functionality

[Danny Nicholas]

Just a WAG, but I'm guessing they may have a limited number of lines and
don't want one phone hogging 2-3 at a time.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Patrick Lists
Sent: Thursday, January 26, 2012 11:21 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] SLA for DAHDI FXO - Emulating Key System
Functionality

On 26-01-12 18:08, Jeff LaCoursiere wrote:
[snip]
>
> I'm also very interested in working examples, especially if someone 
> has set it up for SIP termination "trunks" rather than Dahdi.

Maybe I am missing something here but why would you want to emulate a
keysystem with analog (thus single call) lines on a SIP trunk which support
multiple simultaneous calls on that single trunk?

Regards,
Patrick


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[asterisk-users] Too many open files

[Mike Diehl]

Hi all,

While trying to track down a T.38 issue, I came across a series of log
entries like this:

[Jan 26 10:23:31] WARNING[32508]: udptl.c:948 ast_udptl_new_with_bindaddr:
Unable to allocate socket: Too many open files
[Jan 26 10:23:31] ERROR[32508]: acl.c:488 ast_ouraddrfor: Cannot create
socket


I'm thinking this log entry is related to my fax failure, so I've go to ask:

What causes it and how do I fix/mitigate it?

Thanks in advance,

Mike.

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Re: [asterisk-users] Sip Registration Hijacking

[Steve Edwards]

On Thu, 26 Jan 2012, eherr wrote:


It is accessible from HTTP.

However, the access list only allows access from my home and the 
password is strong.


Can you configure it to 'syslog' accesses where you can monitor it.

Maybe your access lists are invalid, misunderstood or not being honored.

--
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-
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Newline  Fax: +1-760-731-3000

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Re: [asterisk-users] SLA for DAHDI FXO - Emulating Key System Functionality

[Jeff LaCoursiere]

On Thu, 2012-01-26 at 18:21 +0100, Patrick Lists wrote:
> On 26-01-12 18:08, Jeff LaCoursiere wrote:
> [snip]
> >
> > I'm also very interested in working examples, especially if someone has
> > set it up for SIP termination "trunks" rather than Dahdi.
> 
> Maybe I am missing something here but why would you want to emulate a 
> keysystem with analog (thus single call) lines on a SIP trunk which 
> support multiple simultaneous calls on that single trunk?
> 

Because we sell a product that limits the number of simultaneous calls
on that trunk, and many of our customers have asked for key system-like
functionality (since that is what they are used to), and to provide an
interface that "looks" like individual POTS lines would be more
comfortable for them.

Thanks,

j


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Re: [asterisk-users] SLA for DAHDI FXO - Emulating Key System Functionality

[bakko]

Hi

For this scenario you can use the group and group_count functions and create
a hint dialplan like this:

exten => trunkname,hint,custom:trunkname

When you reach the maximum number of available channels set the hint in use:

Set(DEVICE_STATE(Custom:confcorso)=INUSE)

To remove:

Set(DEVICE_STATE(Custom:confcorso)=NOT_INUSE)

After this, configure BLF on the phone top subscribe trunknumber "extension"

Regards

- Bakko


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[asterisk-users] unsubscribe

[Dietmar Zlabinger]

unsubscribe
Am 26.01.2012 18:43 schrieb "Steve Edwards" :

> On Thu, 26 Jan 2012, eherr wrote:
>
>  It is accessible from HTTP.
>>
>> However, the access list only allows access from my home and the password
>> is strong.
>>
>
> Can you configure it to 'syslog' accesses where you can monitor it.
>
> Maybe your access lists are invalid, misunderstood or not being honored.
>
> --
> Thanks in advance,
> --**--**
> -
> Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
> Newline  Fax: +1-760-731-3000
>
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Re: [asterisk-users] unsubscribe

[Doug Lytle]

Dietmar Zlabinger wrote:

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Re: [asterisk-users] Too many open files

[Chad Wallace]

On Thu, 26 Jan 2012 10:35:14 -0700
Mike Diehl  wrote:

> While trying to track down a T.38 issue, I came across a series of log
> entries like this:
> 
> [Jan 26 10:23:31] WARNING[32508]: udptl.c:948
> ast_udptl_new_with_bindaddr: Unable to allocate socket: Too many open
> files [Jan 26 10:23:31] ERROR[32508]: acl.c:488 ast_ouraddrfor:
> Cannot create socket
> 
> 
> What causes it and how do I fix/mitigate it?

In the script that runs asterisk, execute this command before running
asterisk:

ulimit -n 8192

Then restart asterisk.

Or, if you have the /etc/default/asterisk file on your system (like I
do), uncomment the MAX_FILES line (or increase it), and the default init
script should take care of it when you next restart asterisk.


-- 

C. Chad Wallace, B.Sc.
The Lodging Company
http://www.lodgingcompany.com/
OpenPGP Public Key ID: 0x262208A0


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Re: [asterisk-users] Executing Script after MixMonitor is called

[David Backeberg]

On Wed, Jan 25, 2012 at 10:29 AM, Faraj Khasib  wrote:
> Hello Guys,
> I am trying to convert files that are .wac to mp3 after mixmonitor command is 
> called but it doesnt execute the command, I tried the command in terminal it 
> worked, any help please ... below is my dial plan
> exten=6500,n,Set(MIXMONITOR_EXEC=&& nice -n 19 /usr/local/bin/lame -b 8 -t -F 
> -m m --bitwidth 8 --quiet "/var/spool/asterisk/monitor/${CALLFILENAME}.wav" 
> "/var/spool/asterisk/monitor/${CALLFILENAME}.mp3" && rm -f 
> "/var/spool/asterisk/monitor/${CALLFILENAME}.wav")
> exten=6500,n,MixMonitor(${CALLFILENAME}.wav,b)

One obvious thing to try is to make sure that when you're making your
shell test that you are running the shell as the same user that
asterisk is running as.

Set your console to at least verbose = 3, and run the dialplan. What
do you get for the error?

I will say just looking at that code, you have a mess. It would be
simpler to debug if you made a short shell script that consisted of
something like

shebang /path/to/bash

PATH=$1
lame --arguments $1.wav $1.mp3
if [ -f {$1}.mp3 ] ; then
 rm {$1}.wav

and then call that script with the argument that is the path to the
file. Will make your dialplan easier to debug, and still get you the
equivalent functionality.

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Re: [asterisk-users] Executing Script after MixMonitor is called

[David Backeberg]

On Thu, Jan 26, 2012 at 7:18 PM, David Backeberg  wrote:
> shebang /path/to/bash
>
> PATH=$1
> lame --arguments $1.wav $1.mp3
> if [ -f {$1}.mp3 ] ; then
>  rm {$1}.wav

And my silly code sample hasn't been debugged, and I can spot one
glaring bug, and another less important bug. (gotta close the if with
a fi, I declare PATH and never use it, and I'm not certain that the
declare will work without a 'declare' statement)

Your dialplan sample was missing an essential step, which was

System()

that actually kicks off your command. You could also use it to kick
off the bash script, like
System(path_to_script path_argument_for_script)

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Re: [asterisk-users] Executing Script after MixMonitor is called

[Steve Edwards]

On Thu, 26 Jan 2012, David Backeberg wrote:


On Thu, Jan 26, 2012 at 7:18 PM, David Backeberg  wrote:

shebang /path/to/bash

PATH=$1
lame --arguments $1.wav $1.mp3
if [ -f {$1}.mp3 ] ; then
 rm {$1}.wav


And my silly code sample hasn't been debugged, and I can spot one
glaring bug, and another less important bug. (gotta close the if with
a fi, I declare PATH and never use it, and I'm not certain that the
declare will work without a 'declare' statement)

Your dialplan sample was missing an essential step, which was

System()

that actually kicks off your command. You could also use it to kick
off the bash script, like
System(path_to_script path_argument_for_script)


+1 for anything that cleans up a bunch of ugly dialplan code :)

The OP was using MIXMONITOR_EXEC (although I wonder about the '&&' syntax) 
so he doesn't need to explicitly execute (via system()) his commands.


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[asterisk-users] upgraded 1.8.8.0 > 10.1.0-rc2: now db warnings

[sean darcy]

I've just upgraded from 1.8.8.0 to 10.1.0-rc2. Now I'm getting a flood of:

WARNING[5100]: db.c:295 ast_db_put: Couldn't execute statment: SQL logic 
error or missing database


AFAIK, I'm not doing any database puts (or gets). There were no such 
warnings in 1.8.8.0.


What do I need to do to silence these warnings?

sean


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Re: [asterisk-users] Executing Script after MixMonitor is called

[David Backeberg]

On Thu, Jan 26, 2012 at 7:36 PM, Steve Edwards
 wrote:
> The OP was using MIXMONITOR_EXEC (although I wonder about the '&&' syntax)
> so he doesn't need to explicitly execute (via system()) his commands.

Wow. Never knew that was possible. I still don't like the syntax, but
good to know.

For optimal debugging, I would suggest OP not try to build the
pyramids with a single line of commands joined by ampersands, but
rather do a separate System() call for either doing my script idea or
doing the individual commands in multiple separate System() calls.

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Re: [asterisk-users] upgraded 1.8.8.0 > 10.1.0-rc2: now db warnings

[Jim DeVito]

Are you by chance using templates (!) In your sip.con? Ive had access denied 
errors befor when running as non root.

- Original message -
> I've just upgraded from 1.8.8.0 to 10.1.0-rc2. Now I'm getting a flood
> of:
> 
> WARNING[5100]: db.c:295 ast_db_put: Couldn't execute statment: SQL logic 
> error or missing database
> 
> AFAIK, I'm not doing any database puts (or gets). There were no such 
> warnings in 1.8.8.0.
> 
> What do I need to do to silence these warnings?
> 
> sean
> 
> 
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[asterisk-users] Weird IPs in Fail2ban list

[asterisk jobs]

Hello everyone,

I have noticed getting wired IPs blocked by Fail2ban. Has anyone else seen
this or can explain this?

Chain fail2ban-ASTERISK (1 references)
num  target prot opt source   destination
1DROP   all  --  0.23.20.189  0.0.0.0/0

I also get things like, 0.0.5.2, etcFail2ban seems to be working when I
am testing. Are these numbers taken from the SIP packet or the TCP/IP
protocol source because they surely are not valid addresses.

Thanks
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Re: [asterisk-users] Executing Script after MixMonitor is called

[Jeremy Kister]

On 1/25/2012 10:29 AM, Faraj Khasib wrote:

I am trying to convert files that are .wac to mp3 after mixmonitor

> command is called but it doesnt execute the command, I tried the

command in terminal it worked, any help please ... below is my dial

> plan

what version of asterisk are you using ?

if it's an older version of 1.8 (< 1.8.4) and you're also recording the 
call, you may be encountering a known bug.

https://issues.asterisk.org/jira/browse/ASTERISK-17346


--
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http://jeremy.kister.net./

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Re: [asterisk-users] Executing Script after MixMonitor is called

[Satish Barot]

This is how I use a wav to mp3 script on Mixmonitor in my dialplan
(Asterisk 1.8.7.0).
...
same => n,MixMonitor(${FILENAME},W(4),/var/spool/asterisk/wav2mp3
^{FILENAME})
...
and my script is...

#!/bin/bash

WAV="/var/spool/asterisk/monitor/$1"
MP3=$(echo $1 | sed 's/\.wav$/.mp3/')
MP3DEST="/var/spool/asterisk/mp3/$MP3"
/usr/bin/lame "${WAV}" "${MP3DEST}" --silent -b 16 -s 9.6 -m m --bitwidth 8
--lowpass 9.6 --resample 8 --lowpass-width 1

--SATISH BAROT
Ahmedabad,India.

On Wed, Jan 25, 2012 at 8:59 PM, Faraj Khasib wrote:

> Hello Guys,
> I am trying to convert files that are .wac to mp3 after mixmonitor command
> is called but it doesnt execute the command, I tried the command in
> terminal it worked, any help please ... below is my dial plan
> exten=6500,n,Set(MIXMONITOR_EXEC=&& nice -n 19 /usr/local/bin/lame -b 8 -t
> -F -m m --bitwidth 8 --quiet
> "/var/spool/asterisk/monitor/${CALLFILENAME}.wav"
> "/var/spool/asterisk/monitor/${CALLFILENAME}.mp3" && rm -f
> "/var/spool/asterisk/monitor/${CALLFILENAME}.wav")
> exten=6500,n,MixMonitor(${CALLFILENAME}.wav,b)
>
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Re: [asterisk-users] Weird IPs in Fail2ban list

[Mikhail Lischuk]

 

asterisk jobs писал 27.01.2012 06:49: 

> Hello everyone, I
have noticed getting wired IPs blocked by Fail2ban. Has anyone else seen
this or can explain this? 
> 
> Chain fail2ban-ASTERISK (1 references)

> num target prot opt source destination 
> 1 DROP all -- 0.23.20.189
0.0.0.0/0 [1] 
> I also get things like, 0.0.5.2, etcFail2ban seems
to be working when I am testing. Are these numbers taken from the SIP
packet or the TCP/IP protocol source because they surely are not valid
addresses. 
> Thanks

Did you find those IPs in Asterisk log? 

If so -
it isn't Fail2Ban problem, for it just parses logs and extracts
substring 

-- 
With Best Regards
Mikhail Lischuk

 

Links:
--
[1]
http://0.0.0.0/0
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[asterisk-users] Strange how Asterisk know the updated information of log

[virendra bhati]

Hi All,

I want to make a new file of CLI log everyday. So I just make a shell
script in asterisk log directory. My file is working fine and making new
file with the name of *full_2012-01-27*. But strange I noticed that
asterisk is updating my newly crested files even i don't reload asterisk.

So how asterisk know that file name is changed ? why not asterisk make new
file with the name of *full* ?

Can someone please tell me this behaviour of Asterisk (1.6.2.20).

-- 

Thanks and regards

 Virendra Bhati
+91-8885268942
Software Engineer
E-mail-: virbh...@gmail.com
Skype id:- virbhati2
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