[asterisk-users] Fixing an old bug related to extension "s" - feedback wanted

[Olle E. Johansson]

Friends,

While working with the manager interface, I noticed that an originate action to 
a non-existing extension had a strange behaviour. Instead of generating an 
error, which would happen in most VoIP channels and Dahdi, Asterisk started 
looking for extension "s" as a "fallback". 

For as long as I've worked with Asterisk, the definition of extension "s" has 
been that it is used when *NO EXTENSION* has been given (and in the macro 
command). There are two good examples - immediate answer in Dahdi and calling a 
SIP domain without a username part - like "sip:digium.com". In my trainings I 
always repeat (with a loud voice) that extension "s" is *NOT* a wildcard.

Obviously this behaviour is a bug. It's been around for a long time and has 
been hidden by most apps and channel drivers that handle a bad extension in a 
correct way and report errors before the PBX is started in order to handle the 
channel.

The question is - how do we fix this? There might be applications out there 
that depend on this buggy behaviour.

What I've proposed are two separate fixes:

1) Change the manager Originate action


In Asterisk 1.8, there will be a warning if an extension given doesn't exist, 
but the behaviour will not change. A flag in Asterisk.conf [compat] section 
will be implemented so that you can change this behaviour and get an error 
response in manager if the extension does not exist.
In Asterisk 10 the error response will be the default behaviour. If an 
application using AMI needs a fallback, it needs to be controlled by the 
application. It needs to know that an extension does not exist and that the 
call can't be fulfilled.

2) Change the PBX core
===

The bug actually exists in the PBX core, in ast_pbx_start(). We will not change 
this in Asterisk 1.8. 

In Asterisk 10, the core pbx will report that the extension does not exist and 
no longer fall back to s in current context or s@default. This will, as we see 
it now, not affect most channel drivers and thus most dialplans. If you rely 
heavily on the originate function (AMI, CLI and dialplan)  and use the fallback 
behaviour, you will need to modify your dialplans.

Final question
===

My question now is what you think about these changes. Do you need a flag for 
Asterisk 10 to revert to the old behaviour? Is this bug something you actually 
rely on in your application?

Thanks for your response!

/O


Edvina SIP Masterclass covering SIP, Asterisk & Kamailio - Oxford, UK, Nov 
7-11. *  http://www.telespeak.co.uk



--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] single registration per user

[Olle E. Johansson]

18 sep 2011 kl. 22:23 skrev Catalin S.:

> Hello Eric,
>  
> Is about outgoing calls from multiple devices with the same username at aprox 
> same time. The overwritten is for incomming calls. I want to prevent using 
> the same account in multiple devices at same time. The solution with IP will 
> not apply because users may be behind nat or will change everytime multiple 
> access points. Do you have any other clues?

There is no real good way to prevent this. How can Asterisk now which 
registration that is the valid one? If a device reboots and gets a new IP from 
DHCP, we do not want to prevent that new registration to prevent the old one 
from another IP, but the very same device. There's no device ID used in the 
registration, only the SIP account. 

This also applies to OpenSER/kamailio/OpenSIPS. We can prevent multiple 
simultaneous registrations in those, but that will also mean that phones that 
reboot will be blocked until all registrations expire in the server.

/O


Edvina SIP Masterclass covering SIP, Asterisk & Kamailio - Oxford, UK, Nov 
7-11. *  http://www.telespeak.co.uk
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] single registration per user

[Ishfaq Malik]

On Tue, 2011-09-20 at 11:26 +0200, Olle E. Johansson wrote:
> 18 sep 2011 kl. 22:23 skrev Catalin S.:
> 
> > Hello Eric,
> >  
> > Is about outgoing calls from multiple devices with the same username at 
> > aprox same time. The overwritten is for incomming calls. I want to prevent 
> > using the same account in multiple devices at same time. The solution with 
> > IP will not apply because users may be behind nat or will change everytime 
> > multiple access points. Do you have any other clues?
> 
> There is no real good way to prevent this. How can Asterisk now which 
> registration that is the valid one? If a device reboots and gets a new IP 
> from DHCP, we do not want to prevent that new registration to prevent the old 
> one from another IP, but the very same device. There's no device ID used in 
> the registration, only the SIP account. 
> 
> This also applies to OpenSER/kamailio/OpenSIPS. We can prevent multiple 
> simultaneous registrations in those, but that will also mean that phones that 
> reboot will be blocked until all registrations expire in the server.
> 
> /O
> 
> 
> Edvina SIP Masterclass covering SIP, Asterisk & Kamailio - Oxford, UK, Nov 
> 7-11. *  http://www.telespeak.co.uk
> --
Surely the only way to prevent this is tight control of the usernames
and passwords, i.e. configuring all devices yourself without the user(s)
knowing what the un/pass for them are. Is there a good reason you can't
do this?

Regards

Ish
-- 
Ishfaq Malik
Software Developer
PackNet Ltd

Office:   0161 660 3062


--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] DTMF problem

[Olle E. Johansson]

19 sep 2011 kl. 01:51 skrev Zeeshan A Zakaria:

> This DTMF problem has always been there and there is no real solution for it, 
> other than using those expensive PBX systems like that from Avaya, Cisco, 
> etc. This problem happens when you are sending inband DTMF tones. Via 
> softphone you are sending out-of-band DTMF which is basically SIP messages.

Just to correct the latest part of your statement:

The default way to send DTMF in SIP calls is using DTMF as a codec called 
telephony-event in the RTP stream. This sends DTMF as events. Most hard and 
soft phones support this - usually called RFC2833 DTMF mode. Asterisk supports 
it by default. 

Sending DTMF in the audio usually gets messy when using an IP network. 
Especially if you use codecs that compress the audio. I do recommend you to use 
RFC2833. We have built very large IVR services and have no issues with DTMF 
being received in Asterisk so it's doable.

There are other issues with Asterisk DTMF, but that's another issue :-)

/O




--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Ghost DID in System

[Aaron Krohn]

Thanks for your reply, but it was an issue with upstream provider. The 
number got stuck in the middle of being ported.


On 09/20/2011 01:34 AM, C F wrote:

Without your dialplan there isnt much that can be done to help.
Can you please post your relevant dialplans?
Whats voip1 and voip2?
When you say outside the voip system call goes thru, to where?
Who has the number currently?
Any sip debug you care sharing?


On Mon, Sep 19, 2011 at 6:51 PM, Aaron Krohn  wrote:

This is going to sound ridiculous, but there appears to be a ghost DID in
our system. We are going to get the number ported to us, but it has not
happened yet. From a phone outside of our voip system, the call still goes
through. When calling the did from a phone within our system, there is just
dead air.

In the asterisk CLI, I can see our primary server, voip1 trying to do pass
the call to voip2 after it complains about not knowing what to do with the
call.

I have removed all references to this number from all dialplans and sip-did
lists and restarted many times. I simply don't understand why our voip1
server believes it should try to route the call instead of passing it to the
outside world. Does anyone have an explanation or know where I could look?
(dialplans, obviously =)


--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
  http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users


--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users






--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
  http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] NC DATA FINDOUT IN AUTO DIALER

[Nasir Iqbal]

Please check offline message

Regards
Nasir Iqbal

ICT Innovations
http://www.ictinnovations.com/



On Tue, Sep 20, 2011 at 2:47 AM, mahesh katta wrote:

> Thanks for reply,
> I had check it. in auto dialer whenever dial the number there is no voice
> to get agent. dialer will dial the number asterisk not getting voice like
> swo,NA. how we can get the voice in there.
> Best Regards,
>
> Mahesh Katta
> *BUZZ**WORKS* Business Services Private Limited
> BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI
> 201, Crystal Tower, 75 Gundavali Cross Lane, Andheri Kurla Road, Andheri
> (E) Mumbai 400069
> GSM +91.97029.70779 | Phone +91.22.4229.2634 | Fax +91.22.4229.2634
> Web http://www.buzzworks.com
>
>
>
> On Mon, Sep 19, 2011 at 6:23 PM, Nasir Iqbal wrote:
>
>> Please check our voice sms and fax broadcasting / smart autodialler /
>> smart predictive dialler based on asterisk  named ictbroadcast , it provide
>> real time report of busy, answered, congestion , failed, no answer call
>> statistics of running campaign
>>
>> HTTP://www.ictinnovations.com/ictbroadcast
>>
>> Regards
>>
>> On 19-Sep-2011 7:13 PM, "mahesh katta"  wrote:
>> > Hi List,
>> >
>> > I have one query, I am using Go autodial in this using auto dialing.
>> > autodial can do only whenever customer pick the call that call will go
>> to
>> > agents.
>> > but problem autodial dialing the database in that I am not getting NC
>> data
>> > means, not reachable,switch off ,outofservice data. how can I get this
>> data.
>> > is there any software get the telcovoice and give report ?
>> >
>> > Best Regards,
>> >
>> > Mahesh Katta
>> > *BUZZ**WORKS* Business Services Private Limited
>>
>> > BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI
>> > 201, Crystal Tower, 75 Gundavali Cross Lane, Andheri Kurla Road, Andheri
>> (E)
>> > Mumbai 400069
>> > GSM +91.97029.70779 | Phone +91.22.4229.2634 | Fax +91.22.4229.2634
>> > Web http://www.buzzworks.com
>>
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>   http://www.asterisk.org/hello
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>   http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Fixing an old bug related to extension "s" - feedback wanted

[Danny Nicholas]

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olle E.
Johansson
Sent: Tuesday, September 20, 2011 4:23 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Fixing an old bug related to extension "s" -
feedback wanted

Friends,

While working with the manager interface, I noticed that an originate action
to a non-existing extension had a strange behaviour. Instead of generating
an error, which would happen in most VoIP channels and Dahdi, Asterisk
started looking for extension "s" as a "fallback". 

For as long as I've worked with Asterisk, the definition of extension "s"
has been that it is used when *NO EXTENSION* has been given (and in the
macro command). There are two good examples - immediate answer in Dahdi and
calling a SIP domain without a username part - like "sip:digium.com". In my
trainings I always repeat (with a loud voice) that extension "s" is *NOT* a
wildcard.

Obviously this behaviour is a bug. It's been around for a long time and has
been hidden by most apps and channel drivers that handle a bad extension in
a correct way and report errors before the PBX is started in order to handle
the channel.

The question is - how do we fix this? There might be applications out there
that depend on this buggy behaviour.

What I've proposed are two separate fixes:

1) Change the manager Originate action


In Asterisk 1.8, there will be a warning if an extension given doesn't
exist, but the behaviour will not change. A flag in Asterisk.conf [compat]
section will be implemented so that you can change this behaviour and get an
error response in manager if the extension does not exist.
In Asterisk 10 the error response will be the default behaviour. If an
application using AMI needs a fallback, it needs to be controlled by the
application. It needs to know that an extension does not exist and that the
call can't be fulfilled.

2) Change the PBX core
===

The bug actually exists in the PBX core, in ast_pbx_start(). We will not
change this in Asterisk 1.8. 

In Asterisk 10, the core pbx will report that the extension does not exist
and no longer fall back to s in current context or s@default. This will, as
we see it now, not affect most channel drivers and thus most dialplans. If
you rely heavily on the originate function (AMI, CLI and dialplan)  and use
the fallback behaviour, you will need to modify your dialplans.

Final question
===

My question now is what you think about these changes. Do you need a flag
for Asterisk 10 to revert to the old behaviour? Is this bug something you
actually rely on in your application?

Thanks for your response!

/O

Just my .02 - fix Originate since the "Original Asterisk" book, page 125
paragraph 1 says "s" = "start".  If "s" is not really "start", I'm going to
scrap my 3+ years of dialplan writing and change all of my simple dialplans
to read exten=> start,1,blah instead of exten => s,1,blah.  To me exten=>
s,1,blah is more intuitive and less vulnerable than exten => _X.,1,blah.


--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Add PinCode on my dialplan

[Malvin Rito]

Hi List,
I currently have a asterisk server running used for dialing-out for IDD but I 
want to Put a pincode wherein only users with the right pin code will be 
allowed to dial IDD. Any sample dialplan you can suggest pls?

Thanks,
Malvin--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Add PinCode on my dialplan

[Danny Nicholas]

That's what the DISA function is for.

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Malvin Rito
Sent: Tuesday, September 20, 2011 8:34 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Add PinCode on my dialplan

 

Hi List,
I currently have a asterisk server running used for dialing-out for IDD but
I want to Put a pincode wherein only users with the right pin code will be
allowed to dial IDD. Any sample dialplan you can suggest pls?

Thanks,
Malvin

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Add PinCode on my dialplan

[bakko]

Or authenticate aplication.

If you want use a database with a user and pin table, so each user have a pin 
asigned, you can look a func_odbc function.

Regards
  - Original Message - 
  From: Danny Nicholas 
  To: 'Asterisk Users Mailing List - Non-Commercial Discussion' 
  Sent: Tuesday, September 20, 2011 8:38 AM
  Subject: Re: [asterisk-users] Add PinCode on my dialplan


  That's what the DISA function is for.

   

  From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Malvin Rito
  Sent: Tuesday, September 20, 2011 8:34 AM
  To: asterisk-users@lists.digium.com
  Subject: [asterisk-users] Add PinCode on my dialplan

   

  Hi List,
  I currently have a asterisk server running used for dialing-out for IDD but I 
want to Put a pincode wherein only users with the right pin code will be 
allowed to dial IDD. Any sample dialplan you can suggest pls?

  Thanks,
  Malvin



--


  --
  _
  -- Bandwidth and Colocation Provided by http://www.api-digital.com --
  New to Asterisk? Join us for a live introductory webinar every Thurs:
 http://www.asterisk.org/hello

  asterisk-users mailing list
  To UNSUBSCRIBE or update options visit:
 http://lists.digium.com/mailman/listinfo/asterisk-users--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Add PinCode on my dialplan

[Danny Nicholas]

+1 bakko

 

Using DISA might open a "hole" you don't want to have

asterisk -rx "core show application disa"

 

  -= Info about application 'DISA' =-

 

[Synopsis]

DISA (Direct Inward System Access)

 

[Description]

DISA([|]) or DISA()

The DISA, Direct Inward System Access, application allows someone from

outside the telephone switch (PBX) to obtain an "internal" system

dialtone and to place calls from it as if they were placing a call from

within the switch.

DISA plays a dialtone. The user enters their numeric passcode, followed by

the pound sign (#). If the passcode is correct, the user is then given

system dialtone on which a call may be placed. Obviously, this type

of access has SERIOUS security implications, and GREAT care must be

taken NOT to compromise your security.

 

There is a possibility of accessing DISA without password. Simply

exchange your password with "no-password".

 

Example: exten => s,1,DISA(no-password|local)

 

Be aware that using this compromises the security of your PBX.

 

The arguments to this application (in extensions.conf) allow either

specification of a single global passcode (that everyone uses), or

individual passcodes contained in a file. It also allows specification

of the context on which the user will be dialing. If no context is

specified, the DISA application defaults the context to "disa".

Presumably a normal system will have a special context set up

for DISA use with some or a lot of restrictions.

 

The file that contains the passcodes (if used) allows specification

of either just a passcode (defaulting to the "disa" context, or

passcode|context on each line of the file. The file may contain blank

lines, or comments starting with "#" or ";". In addition, the

above arguments may have |new-callerid-string appended to them, to

specify a new (different) callerid to be used for this call, for

example: numeric-passcode|context|"My Phone" <(234) 123-4567> or

full-pathname-of-passcode-file|"My Phone" <(234) 123-4567>.  Last

but not least, |mailbox[@context] may be appended, which will cause

a stutter-dialtone (indication "dialrecall") to be used, if the

specified mailbox contains any new messages, for example:

numeric-passcode|context||1234 (w/a changing callerid).  Note that

in the case of specifying the numeric-passcode, the context must be

specified if the callerid is specified also.

 

If login is successful, the application looks up the dialed number in

the specified (or default) context, and executes it if found.

If the user enters an invalid extension and extension "i" (invalid)

exists in the context, it will be used. Also, if you set the 5th argument

to 'NOANSWER', the DISA application will not answer initially.

 

Using Authenticate is a "safer" option

asterisk -rx "core show application authenticate"

 

  -= Info about application 'Authenticate' =-

 

[Synopsis]

Authenticate a user

 

[Description]

  Authenticate(password[|options[|maxdigits]]): This application asks the
caller

to enter a given password in order to continue dialplan execution. If the
password

begins with the '/' character, it is interpreted as a file which contains a
list of

valid passwords, listed 1 password per line in the file.

  When using a database key, the value associated with the key can be
anything.

Users have three attempts to authenticate before the channel is hung up. If
the

passsword is invalid, the 'j' option is specified, and priority n+101
exists,

dialplan execution will continnue at this location.

  Options:

 a - Set the channels' account code to the password that is entered

 d - Interpret the given path as database key, not a literal file

 j - Support jumping to n+101 if authentication fails

 m - Interpret the given path as a file which contains a list of account

 codes and password hashes delimited with ':', listed one per line
in

 the file. When one of the passwords is matched, the channel will
have

 its account code set to the corresponding account code in the file.

 r - Remove the database key upon successful entry (valid with 'd' only)

 maxdigits  - maximum acceptable number of digits. Stops reading after

 maxdigits have been entered (without requiring the user to

 press the '#' key).

 Defaults to 0 - no limit - wait for the user press the '#' key.

 

Since you (OP) asked for a "fish" today, here it is

[dial-w-pass]

Exten => s,1,noop(get pass and dial)

Exten => s,n,authenticate(1234,j,4)

Exten => s,n,dial(.)

Exten => s,n,hangup()

Exten => s,101,playback(tt-monkeys)

Exten => s,102,hangup

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bakko
Sent: Tuesday, September 20, 2011 8:44 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Add PinCode on my dialplan

 

Or authenticate aplication.

 

If you want use a databas

Re: [asterisk-users] Fixing an old bug related to extension "s" - feedback wanted

[Olle E. Johansson]

20 sep 2011 kl. 15:34 skrev Danny Nicholas:

> Just my .02 - fix Originate since the "Original Asterisk" book, page 125
> paragraph 1 says "s" = "start".  If "s" is not really "start", I'm going to
> scrap my 3+ years of dialplan writing and change all of my simple dialplans
> to read exten=> start,1,blah instead of exten => s,1,blah.  To me exten=>
> s,1,blah is more intuitive and less vulnerable than exten => _X.,1,blah.

I am sorry that the "Original Asterisk" book was wrong and do hope that they 
corrected that part in later editions.

I don't think any official docs have pointed out that "s" was anything else 
than a default extension for situations where there is no extension given.

Using "start" makes your dialplans much easier to read :-) and makes them more 
secure as no app will end up there by accident, which may happen in your 
current systems.

/O
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Fixing an old bug related to extension "s" - feedback wanted

[Olle E. Johansson]

20 sep 2011 kl. 15:34 skrev Danny Nicholas:

> 
> Just my .02 - fix Originate since the "Original Asterisk" book, page 125
> paragraph 1 says "s" = "start".  If "s" is not really "start", I'm going to

In the first edition, page 82, it actually says ""When a call enter a context 
without a specific destination extension, they are handled automatically by the 
s extension". Which is correct. It continues "(The s stands for start, as most 
calls start in the s extension)" which is very wrong.

In the edition you have, page 125, the "most calls" part is deleted and the 
text explains that "this is where a call will start if no extension information 
was passed with the call". 

So they got it right in the end :-)

/O
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] [Asterisk-Users]Using same extension number for outgoing and incoming both internal and PSTN

[Samuel Sappa]

Sorry if this question already asked.
I'm implementing Voip with asterisk and grandstream gxw4108, according
from the manual, for connecting with PSTN I must configure one SIP
account and assign that for dialing the PSTN so in my sip.conf I
configure SIP account(extension) :

[1401]
type=friend
username=1401
secret=1401
host=dynamic
context=my-office
insecure=port

in my extension.con
[my-office]
exten=>1401,1,Dial(SIP/1401,60)
exten=>_NXXN,1,Dial(SIP/${EXTEN}@1401)

but the problem is when I dial the number for the PSTN it's run/dial
on internal extension, from the asterisk guru website it's wrote to
separate the incoming and out going
in sip.conf
[1401]
type=friend
username=1401
secret=1401
host=dynamic
context=my-office-in
insecure=port

[1401]
type=friend
username=1401
secret=1401
host=dynamic
context=my-office-out
insecure=port

in extension.conf
[my-office-in]
exten=>1401,1,Dial(SIP/1001,60)
[my-office-out]
exten=>_NXXN,1,Dial(SIP/${EXTEN}@1401)

but still with this won't work too
My question it's
Is it my configuration true/correct or if there any other way for my problem
I'm using 1 Stage Dialing and the asterisk server and Grandstream
using different IP Address 192.168.101.xxx (for asterisk server) and
192.168.14.xxx (for grandstream gateway)
thank you for helping
-- 
Regards
Samuel Sappa,

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] How to add new Module in existed Asterisk

[tran quoc tuan]

Hi all,
I have an Asterisk Server in version 1.4.36 , it runs stable. Now I want to
add 2 new modules : jabber and chan_gtalk.
How to add these modules and not change anything of configuration existed
Asterisk ?


Best regards,
Ryan.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] How to add new Module in existed Asterisk

[Danny Nicholas]

Just do make menuselect 

Then 

Make && make install

As long as you don't do any of the other steps after make install, no
configuration files should be updated.

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of tran quoc tuan
Sent: Tuesday, September 20, 2011 10:37 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] How to add new Module in existed Asterisk

 

Hi all, 
I have an Asterisk Server in version 1.4.36 , it runs stable. Now I want to
add 2 new modules : jabber and chan_gtalk. 
How to add these modules and not change anything of configuration existed
Asterisk ? 


Best regards, 
Ryan. 

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] How to add new Module in existed Asterisk

[tran quoc tuan]

Thank Danny Nicholas for your reply ,
It means I may re-make menuselect in Asterisk version 1.4.36 to add 2 new
modules : jabber and chan_gtalk ? Can version Asterisk 1.4.36 support these
module : jabber and chan_gtalk ?

Thank in advance for all helps!
B.R
Ryan.



On Tue, Sep 20, 2011 at 10:39 PM, Danny Nicholas  wrote:

> Just do make menuselect 
>
> Then 
>
> Make && make install
>
> As long as you don’t do any of the other steps after make install, no
> configuration files should be updated.
>
> ** **
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *tran quoc tuan
> *Sent:* Tuesday, September 20, 2011 10:37 AM
> *To:* asterisk-users@lists.digium.com
> *Subject:* [asterisk-users] How to add new Module in existed Asterisk
>
> ** **
>
> Hi all,
> I have an Asterisk Server in version 1.4.36 , it runs stable. Now I want to
> add 2 new modules : jabber and chan_gtalk.
> How to add these modules and not change anything of configuration existed
> Asterisk ?
>
>
> Best regards,
> Ryan. 
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>   http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] How to add new Module in existed Asterisk

[Danny Nicholas]

To the best of my knowledge, these modules have been supported at least
since 1.4.22.

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of tran quoc tuan
Sent: Tuesday, September 20, 2011 10:46 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to add new Module in existed Asterisk

 

Thank Danny Nicholas for your reply ,  
It means I may re-make menuselect in Asterisk version 1.4.36 to add 2 new
modules : jabber and chan_gtalk ? Can version Asterisk 1.4.36 support these
module : jabber and chan_gtalk ? 

Thank in advance for all helps! 
B.R 
Ryan. 




On Tue, Sep 20, 2011 at 10:39 PM, Danny Nicholas  wrote:

Just do make menuselect 

Then 

Make && make install

As long as you don't do any of the other steps after make install, no
configuration files should be updated.

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of tran quoc tuan
Sent: Tuesday, September 20, 2011 10:37 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] How to add new Module in existed Asterisk

 

Hi all, 
I have an Asterisk Server in version 1.4.36 , it runs stable. Now I want to
add 2 new modules : jabber and chan_gtalk. 
How to add these modules and not change anything of configuration existed
Asterisk ? 


Best regards, 
Ryan. 


--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
  http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

 

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] How to add new Module in existed Asterisk

[tran quoc tuan]

Thank you very much. I will try to install these modules.

B.R.
Ryan.

On Tue, Sep 20, 2011 at 10:52 PM, Danny Nicholas  wrote:

> To the best of my knowledge, these modules have been supported at least
> since 1.4.22.
>
> ** **
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *tran quoc tuan
> *Sent:* Tuesday, September 20, 2011 10:46 AM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] How to add new Module in existed Asterisk*
> ***
>
> ** **
>
> Thank Danny Nicholas for your reply ,
> It means I may re-make menuselect in Asterisk version 1.4.36 to add 2 new
> modules : jabber and chan_gtalk ? Can version Asterisk 1.4.36 support these
> module : jabber and chan_gtalk ?
>
> Thank in advance for all helps!
> B.R
> Ryan.
>
>
> 
>
> On Tue, Sep 20, 2011 at 10:39 PM, Danny Nicholas 
> wrote:
>
> Just do make menuselect 
>
> Then 
>
> Make && make install
>
> As long as you don’t do any of the other steps after make install, no
> configuration files should be updated.
>
>  
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *tran quoc tuan
> *Sent:* Tuesday, September 20, 2011 10:37 AM
> *To:* asterisk-users@lists.digium.com
> *Subject:* [asterisk-users] How to add new Module in existed Asterisk
>
>  
>
> Hi all,
> I have an Asterisk Server in version 1.4.36 , it runs stable. Now I want to
> add 2 new modules : jabber and chan_gtalk.
> How to add these modules and not change anything of configuration existed
> Asterisk ?
>
>
> Best regards,
> Ryan. 
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>   http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
> ** **
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>   http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] mISDN Vs Dahdi

[Gopal krishnan]

What is the difference between using mISDN for BRI and using Dahdi without
mISDN?

Regards
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Fixing an old bug related to extension "s" - feedback wanted

[Ira]

At 07:09 AM 9/20/2011, you wrote:
Using "start" makes your dialplans much easier to read :-) and makes 
them more secure as no app will end up there by accident, which may 
happen in your current systems.


When I went and read version 3 it seemed to indicate that "start" has 
no actual meaning and I could just as well call it cow or fish.


Am I reading it correctly or does the word start actually have a 
special meaning?


Ira 



--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
  http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Fax from FXS to PRI

[Adam Moffett]

If I have a 4 port Digium FXS card and a single port PRI card on the 
same asterisk box, is it expected that I'd be able to plug a fax machine 
into the analog FXS port and have no problems sending or receiving 
faxes?  Our connection to the Telco is on the PRI obviously.


I don't recall the specific card models that we have, but I can check if 
it matters.


Does the version of asterisk or Zaptel matter?

My related question is this: In the scenario described above does the 
audio pass directly from one card to the other through the PCI bus or 
does it have to somehow be processed by software?


--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
  http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Fax from FXS to PRI

[Steve Totaro]

On Tue, Sep 20, 2011 at 4:43 PM, Adam Moffett wrote:

> If I have a 4 port Digium FXS card and a single port PRI card on the same
> asterisk box, is it expected that I'd be able to plug a fax machine into the
> analog FXS port and have no problems sending or receiving faxes?  Our
> connection to the Telco is on the PRI obviously.
>
> I don't recall the specific card models that we have, but I can check if it
> matters.
>
> Does the version of asterisk or Zaptel matter?
>
> My related question is this: In the scenario described above does the audio
> pass directly from one card to the other through the PCI bus or does it have
> to somehow be processed by software?
>
>
Nobody can say for sure.  It is not a supported configuration.  I can tell
you that I have had great success and wasted days messing around with this
configuration.

It is usually the other side's fax machine, a cheap all in-one, or it is TX
and RX gains, or IRQs, or..

Questions to ask are
1.  Is this for your system or are you installing for someone else?  You
could look very bad if the proper expectations are not set.  It may take a
great deal of trial and error to get to an acceptable level, if you can even
do that based on need.

2.  Needs, if fax is part of the lifeblood, then this route may not be the
best.  If it doesn't hurt to ask someone to resend or whatever, then go for
it.

Just remember the gotchas, IRQs, TX RX gain settings, echo can when bridged
=no.

Again, it has never been a supported configuration by Digium, and everyone
that has dealt with faxing in Asterisk especially on different systems will
tell you that you won't know until you try.  And even then, is it worth days
of your time trying to get it as close to a POTS line as possible?

Another issue I have run into are the Digium FXS daughter boards getting
fried somehow.  I punch on a 66 block now and put on surge protection after
frying six modules in as many years.  Something like this
http://www.digitaltele.com/ProductInfo.aspx?productid=HCO

I have never had to do that on the FXS side but now it is just standard for
all single pairs I do.  I will know in the next year or two if it helps.  No
idea what is frying the daughterboards.  Must be the fax machine.

Thanks,
Steve T

Thanks,
Steve T
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] mISDN Vs Dahdi

[Tamer Higazi]

Am 20.09.2011 19:47, schrieb Gopal krishnan:
> What is the difference between using mISDN for BRI and using Dahdi

mISDN was at 1st done for ISDN Services and channel driver as I know. It
supported like call routing (switch based, not your side on the pbx level).


> without mISDN?

you can use DAHDI without ISDN, for other related telephony interfaces.
Like analogue cards. and if there are other telephony hardware
interfaces that has nothing common todo you can use it too.

and genereal:
DAHDI is the native digium hardware telephony interface. Beside mISDN
you have the native support for an echo cancellor you could use.



for example: oslec or the hpec (high performance echo cancellor) which
you can't make use of it natively with mISDN.



As long you have no hardware interface boards as PCI modules, and you
connect only throug the network to your enddevices, there is no need to
startup dahdi at all.

end beside: dahdi is an extra service that starts up, mISDN is a channel
driver you must activate in the modules.conf.


Are all your question answered that far?!

> 
> Regards
> 
> 
Tamer

> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>http://www.asterisk.org/hello
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users


--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Log for voicemail to email?

[Kevin Oravits]

I am having a problem with one of my sites where they are not receiving the 
voicemail to email. I've done a lot of troubleshooting and can't find the 
issue. It would be helpful if there was a log I could look at so that I could 
see perhaps where the email is being rejected. Does anyone know of a log that 
runs on Asterisk that would have this history?

I'm running Asterisk 1.6 on CentOS 5.6. The server is not behind a firewall, 
the Firewall on the box is disabled, SELinux is disabled and I've added the IP 
to our filters. Oddly, we have the same setup at other sites but this is the 
only site it is not working at.

Any ideas would be great.

Thanks,

Kevin

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Log for voicemail to email?

[Leif Madsen]

On 20/09/11 06:53 PM, Kevin Oravits wrote:

I am having a problem with one of my sites where they are not receiving
the voicemail to email. I’ve done a lot of troubleshooting and can’t
find the issue. It would be helpful if there was a log I could look at
so that I could see perhaps where the email is being rejected. Does
anyone know of a log that runs on Asterisk that would have this history?


Well Asterisk isn't sending the email, the email service on the server 
is doing that. You'll need to enable the logging for email delivery on 
the email service itself.


(It's possible you haven't installed an MTA or have it disabled. Or 
perhaps the other ends are rejecting due a missing MX record, or some 
other email configuration issue.)


--
Leif Madsen
http://www.oreilly.com/catalog/asterisk

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
  http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Fixing an old bug related to extension "s" - feedback wanted

[Leif Madsen]

On 20/09/11 09:34 AM, Danny Nicholas wrote:

Just my .02 - fix Originate since the "Original Asterisk" book, page 125
paragraph 1 says "s" = "start".  If "s" is not really "start", I'm going to
scrap my 3+ years of dialplan writing and change all of my simple dialplans
to read exten=>  start,1,blah instead of exten =>  s,1,blah.  To me exten=>
s,1,blah is more intuitive and less vulnerable than exten =>  _X.,1,blah.


The 's' extension does stand for 'start' but I don't think we've ever 
implied it was a catch-all extension.


--
Leif Madsen
http://www.oreilly.com/catalog/asterisk

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
  http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Fixing an old bug related to extension "s" - feedback wanted

[Leif Madsen]

On 20/09/11 03:37 PM, Ira wrote:

At 07:09 AM 9/20/2011, you wrote:

Using "start" makes your dialplans much easier to read :-) and makes
them more secure as no app will end up there by accident, which may
happen in your current systems.


When I went and read version 3 it seemed to indicate that "start" has no
actual meaning and I could just as well call it cow or fish.

Am I reading it correctly or does the word start actually have a special
meaning?


No, that extension 'start' (literal) has no special meaning. You 
absolutely could call it cow, fish, pig, or farmer_john. How you get 
there is by implicitly calling it.


exten => s...  on the other hand has always had special meaning as Olle 
has pointed out, and typically has meant "start" (for analog lines). 
Outside of that you shouldn't really be using the 's' extension as your 
default extensions. The 's' extension has never been a "catch-all" 
extension. Olle has found a situation where the 's' extension is being 
used as a fallback, which is not right, and is suggesting we make 
Asterisk consistent in it's usage of 's'. I agree with his proposal.


But because this functionality (bug) has been around for quite some 
time, he is asking the community for feedback on who may have 
inadvertently used the functionality in their dialplans.


Apologies for anyone who may have read some documentation that appeared 
to imply that the 's' extension was a catch-all. In the first and second 
editions of Asterisk: The Future of Telephony we were mostly using 
analog lines, and thus the usage of the extension 's' was fairly 
prominent. There are many other single letter extensions that have extra 
meaning, such as 'i', 't', etc..., but we never intended to imply that 
's' was a catch-all extension.


--
Leif Madsen
http://www.oreilly.com/catalog/asterisk

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
  http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Log for voicemail to email?

[Lyle Giese]

On 09/20/11 17:53, Kevin Oravits wrote:

I am having a problem with one of my sites where they are not receiving
the voicemail to email. I’ve done a lot of troubleshooting and can’t
find the issue. It would be helpful if there was a log I could look at
so that I could see perhaps where the email is being rejected. Does
anyone know of a log that runs on Asterisk that would have this history?

I’m running Asterisk 1.6 on CentOS 5.6. The server is not behind a
firewall, the Firewall on the box is disabled, SELinux is disabled and
I’ve added the IP to our filters. Oddly, we have the same setup at other
sites but this is the only site it is not working at.

Any ideas would be great.

Thanks,

*Kevin *



--


/var/log/mail  on any of the SuSE or RedHat boxes I have looked at.

Lyle Giese
LCR Computer Services, Inc.

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
  http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] RESEND: Mixmonitor command parameter problem on Asterisk 1.8.4

[Ikka - Mitra Kreasindo]

Is anyone can help me with this ? I'm really desperate.

 

Thx in ad.

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ikka - Mitra
Kreasindo
Sent: Wednesday, September 14, 2011 5:02 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] Mixmonitor command parameter problem on Asterisk
1.8.4

 

Dear all.

 

I'm using MixMonitor command in my dialplan, and I used the "command"
parameter to execute some thing after recording the file.

 

I used the command parameter to convert the wav file that created earlier to
MP3 and than deleted the WAV file.

 

It worked fine with asterisk 1.4.21.2. and 1.6x

But than I have a new asterisk server with asterisk 1.8.4. The command
parameter doesn't work. It's trimed for about 297 character only. The rest
was gone. 

 

This is part of the log with Asterisk 1.4.21.2

 

  -- Executing [08129981925@speedy:7] MixMonitor("SIP/10001-b7d71bd0",
"/var/spool/asterisk/recording/speedy/2011/09/14/-08129981925-Ikka_Testing-1
0001-20110914-163803.wav|bW(2)|/usr/bin/lame
"/var/spool/asterisk/recording/speedy/2011/09/14/-08129981925-Ikka_Testing-1
0001-20110914-163803.wav"
"/var/spool/asterisk/recording/speedy/2011/09/14/-08129981925-Ikka_Testing-1
0001-20110914-163803.mp3" -b 16 -s 9.6 -m m --bitwidth 8 --lowpass 9.6
--resample 8 --lowpass-width 1 && rm -f
"/var/spool/asterisk/recording/speedy/2011/09/14/-08129981925-Ikka_Testing-1
0001-20110914-163803.wav"") in new stack

 

This is part of the log with Asterisk 1.8.4

 

  -- Executing [08129981925@speedy:7] MixMonitor("SIP/10001-001a",
"/var/spool/asterisk/recording/speedy/2011/09/14/ACCOUNT-08129981925-Admin_I
T-10001-20110914-165248.wav,bW(2),/usr/bin/lame
"/var/spool/asterisk/recording/speedy/2011/09/14/ACCOUNT-08129981925-Admin_I
T-10001-20110914-165248.wav" "/var/spool/asterisk") in new stack

 

 

As you can see, with 1.8.4 the command paramater is trimed. 

 

Is there some changes / bug with MixMonitor in Asterisk 1.8.4 ? Is there a
quick workaround for this problem ? 

 

Please help

 

Thx

 

 

Ikka Vertika

Jakarta -Indonesia

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] redundant traffic (Tarek Sawah)

[Claude Hayn]

Tarek,

Thank you for your response.  I am going with the load balancing idea.

Claude
 
-- next part --
An HTML attachment was scrubbed...
URL:

--

Message: 4
Date: Sat, 17 Sep 2011 21:54:07 +
From: Tarek Sawah 
Subject: Re: [asterisk-users] redundant traffic
To: Asterisk Users 
Message-ID: 
Content-Type: text/plain; charset="windows-1252"


I would suggest using a Vyatta based server to Run Asterisk on or behind..
and use the load balance feature to forward your incoming connections to the
Asterisk server this will create one default gateway for your asterisk
server so you won't have to have two separate networks identified.. nor two
NICs. or identify two ports on the server forwarding one of them to the
original binding port of Asterisk.

if it wasn't for the Default gateway .. it would have been easy to do some
port forwarding on the "internet router" side. but Asterisk needs to
communicate with the internet to send packets back. 
this is one of the scenarios i can think of. and can be done in 20 minutes.
well it can be expensive if you calculate the costs of an additional
computer on the network. :S

Tarek Sawah

Information Technology  Adviser

Integrated Digital Systems

CCNP, MCSE, RHCE, TELECOM

USA: +1 386 492 9993

From: chayn...@gmail.com
To: asterisk-users@lists.digium.com
Date: Sat, 17 Sep 2011 17:31:56 -0400
Subject: [asterisk-users] redundant traffic

Hello, I?ve got a customer that wants me to set up their single Asterisk
server so that they can receive redundant traffic streams from their
origination provider.  They want the traffic broadcast to 2 static IP
addresses on the Asterisk server for redundancy.  Their they want to be sure
to receive traffic if one of their subnets/gateways goes down. As I
understand it, having the two IP's set up to receive redundant information
as possible in Linux, but I wonder how (or if it's even possible) to address
this in Asterisk. As anybody ever done this?  Claude
--
_


--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] How to add new Module in existed Asterisk

[James zhu]

hi:
please check the chan_gtalk.conf, add your account info.

Best regards,
James.zhu
Doing asterisk/PRI/ss7/dahdi, linux, asterisk cards, gateway(fxs/fxo/pri<->SIP).
website: www.voipviews.com 


Date: Tue, 20 Sep 2011 22:37:00 +0700
From: tuant...@gmail.com
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] How to add new Module in existed Asterisk

Hi all, 
I have an Asterisk Server in version 1.4.36 , it runs stable. Now I want to add 
2 new modules : jabber and chan_gtalk. 
How to add these modules and not change anything of configuration existed 
Asterisk ? 



Best regards, 
Ryan. 


--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users  
  --
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Log for voicemail to email?

[Steve Edwards]

On Tue, 20 Sep 2011, Kevin Oravits wrote:

I am having a problem with one of my sites where they are not receiving 
the voicemail to email.


They are not receiving or you are not sending?

Unless you've changed 'mailcmd' in voicemail.conf, Asterisk will execute 
'/usr/sbin/sendmail -t' to send the email.


You can test this from a (single) shell command line as follows:

printf "To: korav...@rcolegal.com\nThis is a test.\n"\
| sendmail -t

If you don't get anything clueful at this point, start looking in the 
logs. (Personnaly, I tell syslogd to output everything to a single file so 
I always know where to look.)


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

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
  http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] RESEND: Mixmonitor command parameter problem on Asterisk 1.8.4

[Dale Noll]

I am not real familiar with the size of MixMonitor parameters, but just 
looking at the output, I would suggest you change the logic to call a 
script with a single argument.

something like this,

MixMonitor(${FILENAME},bW(2),/usr/local/bin/convert_to_mp3 ^{FILENAME})

--- /usr/local/bin/convert_to_mp3 --
#!/bin/bash

WAV=$1
MP3=$(echo $1 | sed 's/\.wav$/.mp3/')
/usr/bin/lame "${WAV}" "${MP3}" -b 16 -s 9.6 -m m --bitwidth 8 --lowpass 
9.6 --resample 8 --lowpass-width 1 && rm -f "${WAV}"


--- end of script ---
Set the permissions so it is executable by the asterisk owner.

Note:  This has not been tested and is intended as a starting point.


Dale


On 09/20/2011 07:53 PM, Ikka - Mitra Kreasindo wrote:


Is anyone can help me with this ? I'm really desperate...

Thx in ad.

*From:*asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Ikka - 
Mitra Kreasindo

*Sent:* Wednesday, September 14, 2011 5:02 PM
*To:* 'Asterisk Users Mailing List - Non-Commercial Discussion'
*Subject:* [asterisk-users] Mixmonitor command parameter problem on 
Asterisk 1.8.4


Dear all...

I'm using MixMonitor command in my dialplan, and I used the "command" 
parameter to execute some thing after recording the file.


I used the command parameter to convert the wav file that created 
earlier to MP3 and than deleted the WAV file.


It worked fine with asterisk 1.4.21.2. and 1.6x

But than I have a new asterisk server with asterisk 1.8.4. The command 
parameter doesn't work. It's trimed for about 297 character only. The 
rest was gone.


This is part of the log with Asterisk 1.4.21.2

  -- Executing [08129981925@speedy:7] MixMonitor("SIP/10001-b7d71bd0", 
"/var/spool/asterisk/recording/speedy/2011/09/14/-08129981925-Ikka_Testing-10001-20110914-163803.wav|bW(2)|/usr/bin/lame 
"/var/spool/asterisk/recording/speedy/2011/09/14/-08129981925-Ikka_Testing-10001-20110914-163803.wav" 
"/var/spool/asterisk/recording/speedy/2011/09/14/-08129981925-Ikka_Testing-10001-20110914-163803.mp3" 
-b 16 -s 9.6 -m m --bitwidth 8 --lowpass 9.6 --resample 8 
--lowpass-width 1 && rm -f 
"/var/spool/asterisk/recording/speedy/2011/09/14/-08129981925-Ikka_Testing-10001-20110914-163803.wav"") 
in new stack


This is part of the log with Asterisk 1.8.4

  -- Executing [08129981925@speedy:7] MixMonitor("SIP/10001-001a", 
"/var/spool/asterisk/recording/speedy/2011/09/14/ACCOUNT-08129981925-Admin_IT-10001-20110914-165248.wav,bW(2),/usr/bin/lame 
"/var/spool/asterisk/recording/speedy/2011/09/14/ACCOUNT-08129981925-Admin_IT-10001-20110914-165248.wav" 
"/var/spool/asterisk") in new stack


As you can see, with 1.8.4 the command paramater is trimed...

Is there some changes / bug with MixMonitor in Asterisk 1.8.4 ? Is 
there a quick workaround for this problem ?






--
"The truth speaks for itself. I'm just the messenger."
 Lyta Alexander - Babylon 5

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Fax from FXS to PRI

[Don Kelly]

On Tue, Sep 20, 2011 at 4:43 PM, Adam Moffett 
wrote:

If I have a 4 port Digium FXS card and a single port PRI card on the same
asterisk box, is it expected that I'd be able to plug a fax machine into the
analog FXS port and have no problems sending or receiving faxes?  Our
connection to the Telco is on the PRI obviously.




Nobody can say for sure.  It is not a supported configuration.  I can tell
you that I have had great success and wasted days messing around with this
configuration.  



 

Again, it has never been a supported configuration by Digium, and everyone
that has dealt with faxing in Asterisk especially on different systems will
tell you that you won't know until you try.  And even then, is it worth days
of your time trying to get it as close to a POTS line as possible?




Thanks,
Steve T

This is a scary answer-you're saying that what should be simple "TDM" FXS to
PRI does not work?

 

Are you suggesting this is an Asterisk problem or a Digium hardware problem?

 

Is this really everyone's experience?

--Don

Don Kelly

PCF Corp
People Come First
651 842-1000
888 Don Kell(y)



 

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Fax from FXS to PRI

[Jeff LaCoursiere]

On Tue, 2011-09-20 at 20:57 -0500, Don Kelly wrote:
> On Tue, Sep 20, 2011 at 4:43 PM, Adam Moffett
>  wrote:
> 
> If I have a 4 port Digium FXS card and a single port PRI card on the
> same asterisk box, is it expected that I'd be able to plug a fax
> machine into the analog FXS port and have no problems sending or
> receiving faxes?  Our connection to the Telco is on the PRI obviously.
> 
> 
> 
> 
> Nobody can say for sure.  It is not a supported configuration.  I can
> tell you that I have had great success and wasted days messing around
> with this configuration.  
> 
> 
> 
>  
> 
> Again, it has never been a supported configuration by Digium, and
> everyone that has dealt with faxing in Asterisk especially on
> different systems will tell you that you won't know until you try.
>  And even then, is it worth days of your time trying to get it as
> close to a POTS line as possible?
> 
> 
> 
> 
> Thanks,
> Steve T
> 
> This is a scary answer—you’re saying that what should be simple “TDM”
> FXS to PRI does not work?
> 
>  
> 
> Are you suggesting this is an Asterisk problem or a Digium hardware
> problem?
> 

Like most faxing issues, at it's root it is a timing problem IMO.
Sangoma makes a special timing cable to link their cards so you can do
exactly what you are asking to do.  I've never purchased it, but last I
looked into the issue, that is what they suggested.

j



--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Add PinCode on my dialplan

[Kyle Sexton]

Something like this should work:

exten => _011.,1,Answer
exten => _011.,n,Wait(1)
exten => _011.,n,Read(password,enter-password,5)
exten => _011.,n,GotoIf($[${password} = 12345]?5:9)

exten => _011.,n,NoOp(Matched _9011 - CheckRec-InternationalCall)
exten => _011.,n,Dial(SIP/+${EXTEN:3}@outbound)

exten => _011.,n,Hangup
exten => _011.,n,Playback(invalid)
exten => _011.,n,Hangup

Could be cleaned up (the GotoIf isn't very descriptive about where it's going), 
but it's a starting point.


On Sep 20, 2011, at 8:34 AM, Malvin Rito wrote:

> Hi List,
> I currently have a asterisk server running used for dialing-out for IDD but I 
> want to Put a pincode wherein only users with the right pin code will be 
> allowed to dial IDD. Any sample dialplan you can suggest pls?
> 
> Thanks,
> Malvin
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>   http://www.asterisk.org/hello
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk PRI hangup

[Kyle Sexton]

Just to help with troubleshooting you could try to reproduce the same problem 
with a different set of SIP endpoints.  Setup a soft phone as the destination 
and see if the problem occurs there.  That way you can eliminate the handset as 
a potential problem.


On Sep 15, 2011, at 10:31 AM, Claudio Prono wrote:

> Hello all,
> 
> Form 2-3 weeks i have some problems with incoming ISDN calls, it
> interrupts after 1-2 minutes of call. I have tried to debug this with
> pri set debug on span 1, i have noticied much of this messages:
> 
> -- Timeout occured, restarting PRI
> q921.c:468 t200_expire: q921_state now is Q921_LINK_CONNECTION_RELEASED
> q921.c:858 q921_reset: q921_state now is Q921_LINK_CONNECTION_RELEASED
> Sending TEI management message 1, TEI=127
> Received MDL message
> TEI assiged to 71
> q921.c:858 q921_reset: q921_state now is Q921_LINK_CONNECTION_RELEASED
> Sending Set Asynchronous Balanced Mode Extended
> q921.c:211 q921_send_sabme: q921_state now is Q921_AWAITING_ESTABLISH
> -- Got UA from network peer  Link up.
> q921.c:858 q921_reset: q921_state now is Q921_LINK_CONNECTION_RELEASED
> q921.c:805 q921_dchannel_up: q921_state now is
> Q921_LINK_CONNECTION_ESTABLISHED
> -- Timeout occured, restarting PRI
> q921.c:468 t200_expire: q921_state now is Q921_LINK_CONNECTION_RELEASED
> q921.c:858 q921_reset: q921_state now is Q921_LINK_CONNECTION_RELEASED
> Sending TEI management message 1, TEI=127
> Received MDL message
> TEI assiged to 72
> q921.c:858 q921_reset: q921_state now is Q921_LINK_CONNECTION_RELEASED
> Sending Set Asynchronous Balanced Mode Extended
> q921.c:211 q921_send_sabme: q921_state now is Q921_AWAITING_ESTABLISH
> -- Got UA from network peer  Link up.
> q921.c:858 q921_reset: q921_state now is Q921_LINK_CONNECTION_RELEASED
> q921.c:805 q921_dchannel_up: q921_state now is
> Q921_LINK_CONNECTION_ESTABLISHED
> 
> And there is a debug session of an hanged-up incoming call:
> 
> 
> 
>> Progress Indicator (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard
> (0)  0: 0  Location: Private network serving the local user (1)
>>  Ext: 1  Progress Description: Inband
> information or appropriate pattern now available. (8) ]
>  == Extension Changed 215[ext-local] new state Ringing for Notify User 202
>-- SIP/203-0017 is ringing
>-- SIP/206-0019 is ringing
>-- SIP/210-001a is ringing
>-- SIP/205-0018 is ringing
>-- SIP/201-0015 is ringing
>-- SIP/215-001b is ringing
>-- SIP/202-0016 is ringing
>-- SIP/201-0015 answered DAHDI/1-1
>  == Extension Changed 201[ext-local] new state InUse for Notify User 202
>  == Extension Changed 201[ext-local] new state InUse for Notify User 215
>  == Extension Changed 215[ext-local] new state Idle for Notify User 202
>  == Extension Changed 210[ext-local] new state Idle for Notify User 202
>  == Extension Changed 210[ext-local] new state Idle for Notify User 215
>  == Extension Changed 206[ext-local] new state Idle for Notify User 202
>  == Extension Changed 206[ext-local] new state Idle for Notify User 215
>  == Extension Changed 205[ext-local] new state Idle for Notify User 202
>  == Extension Changed 205[ext-local] new state Idle for Notify User 215
>  == Extension Changed 203[ext-local] new state Idle for Notify User 202
>  == Extension Changed 203[ext-local] new state Idle for Notify User 215
>-- Executing [s@macro-auto-blkvm:1] Set("SIP/201-0015",
> "__MACRO_RESULT=") in new stack
>  == Extension Changed 202[ext-local] new state Idle for Notify User 215
>-- Executing [s@macro-auto-blkvm:2] NoOp("SIP/201-0015",
> "Deleting: BLKVM/600/DAHDI/1-1 TRUE") in new stack
>-- Stopped music on hold on DAHDI/1-1
> q931.c:2951 q931_connect: call 93 on channel 1 enters state 8 (Connect
> Request)
>> Protocol Discriminator: Q.931 (8)  len=11
>> Call Ref: len= 1 (reference 93/0x5D) (Terminator)
>> Message type: CONNECT (7)
>> [18 01 89]
>> Channel ID (len= 3) [ Ext: 1  IntID: Implicit  Other  Spare: 0 
> Exclusive  Dchan: 0
>>   ChanSel: B1 channel
> ]
>> [1e 02 81 82]
>> Progress Indicator (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard
> (0)  0: 0  Location: Private network serving the local user (1)
>>  Ext: 1  Progress Description: Called
> equipment is non-ISDN. (2) ]
> < Protocol Discriminator: Q.931 (8)  len=4
> < Call Ref: len= 1 (reference 93/0x5D) (Originator)
> < Message type: CONNECT ACKNOWLEDGE (15)
> q931.c:3711 q931_receive: call 93 on channel 1 enters state 10 (Active)
> -- Got SABME from network peer.
> Sending Unnumbered Acknowledgement
> q921.c:858 q921_reset: q921_state now is Q921_LINK_CONNECTION_RELEASED
> q921.c:805 q921_dchannel_up: q921_state now is
> Q921_LINK_CONNECTION_ESTABLISHED
> < Protocol Discriminator: Q.931 (8)  len=4
> < Call Ref: len= 1 (reference 93/0x5D) (Originator)
> < Message type: STATUS ENQUIRY (117)
> YYY Here we get reset YYY
>> Protocol Discriminato

Re: [asterisk-users] Console Stereo - One call per ear

[Kyle Sexton]

I have no solution, but my head hurts thinking about listening to separate 
calls simultaneously in each ear.


On Sep 9, 2011, at 9:22 AM, fhirschberg wrote:

> Hi list!
> 
> I'm using the latest Asterisk 1.8.6.0 cross compiled for an i.MX27 board
> and it works really good.
> But I need a feature and don't know how to do this. 
> What I need is the ability to have 2 separate calls on each ear on the
> console channel. 
> Is there a way to get this working? It should be possible to have one call
> on both ears or, if another call is made, to hear this on one (selectable
> L/R) ear, while the other call stays on the other ear.
> Do I need a new console driver? I'm currently using chan_alsa and I already
> have Alsa devices for left, right and left + right output. 
> It would be great if anybody can help with informations or tips where to
> start with my problem.
> 
> Greetings
> Florian
> 
> 
> 
> 
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>   http://www.asterisk.org/hello
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users


--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] [Asterisk-Users]Using same extension number for outgoing and incoming both internal and PSTN

[Sam Govind]

Hey,

I don;t think asterisk-guru could've been wrong on this one - possibly
different scenario than your's. Anyway I see what you did there ! There is
no need for separate context for  incoming or outgoing if you don't want.
What you are doing is *exten=>_NXXN,1,Dial(SIP/${EXTEN}@1401**) *
*
*
When you defined the SIp user/peer [1401] you stated context for handling
dial request as "my-office" and when you tried dialling out you told
asterisk to dial the requested number located at 1401 which should've been
@ if calls need to be dialed to gateway and If your
gateway just accepts SIP based (w/o auth) calls.

*exten=>_NXXN,1,Dial(SIP/${EXTEN}@192.168.14.???**)  *
*
*
If your gateway shows attitude in serving direct request you may need to
create user in gateway and telling asterisk to register on Grandstream as a
user and dial-out using that user like.

*exten=>_NXXN,1,Dial(SIP/${EXTEN}@gstream-user**)  *
*
*
There could be more possible alternatives to successfully dial-out using one
context for handling incoming an out going/ preferred is you create separate
contexts.

Regards,
- Sammy

On Tue, Sep 20, 2011 at 8:13 PM, Samuel Sappa  wrote:

> Sorry if this question already asked.
> I'm implementing Voip with asterisk and grandstream gxw4108, according
> from the manual, for connecting with PSTN I must configure one SIP
> account and assign that for dialing the PSTN so in my sip.conf I
> configure SIP account(extension) :
>
> [1401]
> type=friend
> username=1401
> secret=1401
> host=dynamic
> context=my-office
> insecure=port
>
> in my extension.con
> [my-office]
> exten=>1401,1,Dial(SIP/1401,60)
> exten=>_NXXN,1,Dial(SIP/${EXTEN}@1401)
>
> but the problem is when I dial the number for the PSTN it's run/dial
> on internal extension, from the asterisk guru website it's wrote to
> separate the incoming and out going
> in sip.conf
> [1401]
> type=friend
> username=1401
> secret=1401
> host=dynamic
> context=my-office-in
> insecure=port
>
> [1401]
> type=friend
> username=1401
> secret=1401
> host=dynamic
> context=my-office-out
> insecure=port
>
> in extension.conf
> [my-office-in]
> exten=>1401,1,Dial(SIP/1001,60)
> [my-office-out]
> exten=>_NXXN,1,Dial(SIP/${EXTEN}@1401)
>
> but still with this won't work too
> My question it's
> Is it my configuration true/correct or if there any other way for my
> problem
> I'm using 1 Stage Dialing and the asterisk server and Grandstream
> using different IP Address 192.168.101.xxx (for asterisk server) and
> 192.168.14.xxx (for grandstream gateway)
> thank you for helping
> --
> Regards
> Samuel Sappa,
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>   http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Add PinCode on my dialplan

[Sam Govind]

DISA and DB based Auth could be an overkill.

Kyle showed the very simplistic dial plan if Dial-out pin is common for the
whole system.
See application *Authenticate(password[,options[,maxdigits[,prompt]]] *and
if Voicemail PIN are required to be used use application
*MAuthenticate([mailbox][@context][,options]
*

Regards,

- Sammy

On Wed, Sep 21, 2011 at 8:32 AM, Kyle Sexton  wrote:

> Something like this should work:
>
> exten => _011.,1,Answer
> exten => _011.,n,Wait(1)
> exten => _011.,n,Read(password,enter-password,5)
> exten => _011.,n,GotoIf($[${password} = 12345]?5:9)
>
> exten => _011.,n,NoOp(Matched _9011 - CheckRec-InternationalCall)
> exten => _011.,n,Dial(SIP/+${EXTEN:3}@outbound)
>
> exten => _011.,n,Hangup
> exten => _011.,n,Playback(invalid)
> exten => _011.,n,Hangup
>
> Could be cleaned up (the GotoIf isn't very descriptive about where it's
> going), but it's a starting point.
>
>
> On Sep 20, 2011, at 8:34 AM, Malvin Rito wrote:
>
> Hi List,
> I currently have a asterisk server running used for dialing-out for IDD but
> I want to Put a pincode wherein only users with the right pin code will be
> allowed to dial IDD. Any sample dialplan you can suggest pls?
>
> Thanks,
> Malvin
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>   http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>   http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] RESEND: Mixmonitor command parameter problem on Asterisk 1.8.4

[Sam Govind]

+1 Dale
Alternatively I'd troubles using the MixMonitor() command execution, so what
I did is used System(my commands here) just after the StopMixMonitor().
Using StopMixMonitor() is always recommended to guarantee save the recorded
file and using any commands via System() is easy.

On Wed, Sep 21, 2011 at 6:57 AM, Dale Noll  wrote:

> **
> I am not real familiar with the size of MixMonitor parameters, but just
> looking at the output, I would suggest you change the logic to call a script
> with a single argument.
> something like this,
>
> MixMonitor(${FILENAME},bW(2),/usr/local/bin/convert_to_mp3 ^{FILENAME})
>
> --- /usr/local/bin/convert_to_mp3 --
> #!/bin/bash
>
> WAV=$1
> MP3=$(echo $1 | sed 's/\.wav$/.mp3/')
> /usr/bin/lame "${WAV}" "${MP3}" -b 16 -s 9.6 -m m --bitwidth 8 --lowpass
> 9.6 --resample 8 --lowpass-width 1 && rm -f "${WAV}"
>
> --- end of script ---
> Set the permissions so it is executable by the asterisk owner.
>
> Note:  This has not been tested and is intended as a starting point.
>
>
> Dale
>
>
>
> On 09/20/2011 07:53 PM, Ikka - Mitra Kreasindo wrote:
>
>  Is anyone can help me with this ? I’m really desperate…
>
> ** **
>
> Thx in ad.
>
> ** **
>
> *From:* asterisk-users-boun...@lists.digium.com [
> mailto:asterisk-users-boun...@lists.digium.com]
> *On Behalf Of *Ikka - Mitra Kreasindo
> *Sent:* Wednesday, September 14, 2011 5:02 PM
> *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion'
> *Subject:* [asterisk-users] Mixmonitor command parameter problem on
> Asterisk 1.8.4
>
> ** **
>
> Dear all…
>
> ** **
>
> I’m using MixMonitor command in my dialplan, and I used the “command”
> parameter to execute some thing after recording the file.
>
> ** **
>
> I used the command parameter to convert the wav file that created earlier
> to MP3 and than deleted the WAV file.
>
> ** **
>
> It worked fine with asterisk 1.4.21.2. and 1.6x
>
> But than I have a new asterisk server with asterisk 1.8.4. The command
> parameter doesn’t work. It’s trimed for about 297 character only. The rest
> was gone. 
>
> ** **
>
> This is part of the log with Asterisk 1.4.21.2
>
> ** **
>
>   -- Executing [08129981925@speedy:7] MixMonitor("SIP/10001-b7d71bd0",
> "/var/spool/asterisk/recording/speedy/2011/09/14/-08129981925-Ikka_Testing-10001-20110914-163803.wav|bW(2)|/usr/bin/lame
> "/var/spool/asterisk/recording/speedy/2011/09/14/-08129981925-Ikka_Testing-10001-20110914-163803.wav"
> "/var/spool/asterisk/recording/speedy/2011/09/14/-08129981925-Ikka_Testing-10001-20110914-163803.mp3"
> -b 16 -s 9.6 -m m --bitwidth 8 --lowpass 9.6 --resample 8 --lowpass-width 1
> && rm -f
> "/var/spool/asterisk/recording/speedy/2011/09/14/-08129981925-Ikka_Testing-10001-20110914-163803.wav"")
> in new stack
>
> ** **
>
> This is part of the log with Asterisk 1.8.4
>
> ** **
>
>   -- Executing [08129981925@speedy:7] MixMonitor("SIP/10001-001a",
> "/var/spool/asterisk/recording/speedy/2011/09/14/ACCOUNT-08129981925-Admin_IT-10001-20110914-165248.wav,bW(2),/usr/bin/lame
> "/var/spool/asterisk/recording/speedy/2011/09/14/ACCOUNT-08129981925-Admin_IT-10001-20110914-165248.wav"
> "/var/spool/asterisk") in new stack
>
> ** **
>
> ** **
>
> As you can see, with 1.8.4 the command paramater is trimed… 
>
> ** **
>
> Is there some changes / bug with MixMonitor in Asterisk 1.8.4 ? Is there a
> quick workaround for this problem ? 
>
> ** **
>
>
> --
> "The truth speaks for itself. I'm just the messenger."
>  Lyta Alexander - Babylon 5
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>   http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Add PinCode on my dialplan

[Malvin Rito]

Thanks. ?If I want to use unique PIN for every user that dials out how 
can I implement it using Authenticate app?


Regards,
Malvin

On 9/21/2011 12:42 PM, Sam Govind wrote:

DISA and DB based Auth could be an overkill.

Kyle showed the very simplistic dial plan if Dial-out pin is common 
for the whole system.
See application *Authenticate(password[,options[,maxdigits[,prompt]]] 
*and if Voicemail PIN are required to be used use application 
*MAuthenticate([mailbox][@context][,options] *


Regards,

- Sammy

On Wed, Sep 21, 2011 at 8:32 AM, Kyle Sexton > wrote:


Something like this should work:

exten => _011.,1,Answer
exten => _011.,n,Wait(1)
exten => _011.,n,Read(password,enter-password,5)
exten => _011.,n,GotoIf($[${password} = 12345]?5:9)

exten => _011.,n,NoOp(Matched _9011 - CheckRec-InternationalCall)
exten => _011.,n,Dial(SIP/+${EXTEN:3}@outbound)

exten => _011.,n,Hangup
exten => _011.,n,Playback(invalid)
exten => _011.,n,Hangup

Could be cleaned up (the GotoIf isn't very descriptive about where
it's going), but it's a starting point.


On Sep 20, 2011, at 8:34 AM, Malvin Rito wrote:


Hi List,
I currently have a asterisk server running used for dialing-out
for IDD but I want to Put a pincode wherein only users with the
right pin code will be allowed to dial IDD. Any sample dialplan
you can suggest pls?

Thanks,
Malvin
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users



--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users




--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Add PinCode on my dialplan

[Sam Govind]

See "core show application authe"
If passwords are already the same as those of voicemail.conf go for
application VMAuthenticate() - DIA generates a dial-tone which I don't think
is suitable for dialling out from users(insiders)

  -= Info about application 'Authenticate' =-

[Synopsis]
Authenticate a user

[Description]
This application asks the caller to enter a given password in order to
continue
dialplan execution.
If the password begins with the '/' character,  it is interpreted as a file
which contains a list of valid passwords, listed 1 password per line in the
file.
When using a database key, the value associated with the key can be
anything.
Users have three attempts to authenticate before the channel is hung
up.

[Syntax]
Authenticate(password[,options[,maxdigits[,prompt]]])

[Arguments]
password
Password the user should know
options
a: Set the channels' account code to the password that is entered
d: Interpret the given path as database key, not a literal file
m: Interpret the given path as a file which contains a list of account
codes and password hashes delimited with ':', listed one per line in the
file. When one of the passwords is matched, the channel will have its
account code set to the corresponding account code in the file.
r: Remove the database key upon successful entry (valid with 'd'
only)
maxdigits
maximum acceptable number of digits. Stops reading after maxdigits
have been entered (without requiring the user to press the '#' key).
Defaults to 0 - no limit - wait for the user press the '#' key.
prompt
Override the agent-pass prompt file.

[See Also]
VMAuthenticate(), DISA()


On Wed, Sep 21, 2011 at 9:47 AM, Malvin Rito
wrote:

>  Thanks. ?If I want to use unique PIN for every user that dials out how can
> I implement it using Authenticate app?
>
> Regards,
> Malvin
>
>
> On 9/21/2011 12:42 PM, Sam Govind wrote:
>
> DISA and DB based Auth could be an overkill.
>
> Kyle showed the very simplistic dial plan if Dial-out pin is common for the
> whole system.
> See application *Authenticate(password[,options[,maxdigits[,prompt]]] *and
> if Voicemail PIN are required to be used use application 
> *MAuthenticate([mailbox][@context][,options]
> *
>
>  Regards,
>
>  - Sammy
>
> On Wed, Sep 21, 2011 at 8:32 AM, Kyle Sexton  wrote:
>
>> Something like this should work:
>>
>>  exten => _011.,1,Answer
>> exten => _011.,n,Wait(1)
>> exten => _011.,n,Read(password,enter-password,5)
>> exten => _011.,n,GotoIf($[${password} = 12345]?5:9)
>>
>>  exten => _011.,n,NoOp(Matched _9011 - CheckRec-InternationalCall)
>> exten => _011.,n,Dial(SIP/+${EXTEN:3}@outbound)
>>
>>  exten => _011.,n,Hangup
>> exten => _011.,n,Playback(invalid)
>> exten => _011.,n,Hangup
>>
>>  Could be cleaned up (the GotoIf isn't very descriptive about where it's
>> going), but it's a starting point.
>>
>>
>>  On Sep 20, 2011, at 8:34 AM, Malvin Rito wrote:
>>
>>  Hi List,
>> I currently have a asterisk server running used for dialing-out for IDD
>> but I want to Put a pincode wherein only users with the right pin code will
>> be allowed to dial IDD. Any sample dialplan you can suggest pls?
>>
>> Thanks,
>> Malvin
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>   http://www.asterisk.org/hello
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>>
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>   http://www.asterisk.org/hello
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>   http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to As

Re: [asterisk-users] Console Stereo - One call per ear

[Gohar Ahmed]

Couldn't help LOL on Kyle's remarks. But it could be two users listening to
two different streams/calls. Obviously both can't share single mic on their
call(if they ever need it).

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kyle Sexton
Sent: Wednesday, September 21, 2011 8:50 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Console Stereo - One call per ear

I have no solution, but my head hurts thinking about listening to separate
calls simultaneously in each ear.


On Sep 9, 2011, at 9:22 AM, fhirschberg wrote:

> Hi list!
> 
> I'm using the latest Asterisk 1.8.6.0 cross compiled for an i.MX27 board
> and it works really good.
> But I need a feature and don't know how to do this. 
> What I need is the ability to have 2 separate calls on each ear on the
> console channel. 
> Is there a way to get this working? It should be possible to have one call
> on both ears or, if another call is made, to hear this on one (selectable
> L/R) ear, while the other call stays on the other ear.
> Do I need a new console driver? I'm currently using chan_alsa and I
already
> have Alsa devices for left, right and left + right output. 
> It would be great if anybody can help with informations or tips where to
> start with my problem.
> 
> Greetings
> Florian
> 
> 
> 
> 
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>   http://www.asterisk.org/hello
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users


--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Asterisk-Radius integration

[Ronald Cepres]

Hi all,

I'm trying to setup a system such that when a call comes in to Asterisk, it
first checks the account balance of the caller via Radius and then determine
if the call should go through or not.

I have an average experience in Asterisk but I'm quite new to Radius so I'm
not sure if this setup is possible. Has anyone achieved this kind of setup?

Thanks!

Regards,
Ronald
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk-Radius integration

[amit anand]

Hi

for this you need to write some agi script that will handle the other
feature.

Also you can try for A2billing, its a complete solution for Billing with
asterisk

On Wed, Sep 21, 2011 at 12:08, Ronald Cepres  wrote:

> Hi all,
>
> I'm trying to setup a system such that when a call comes in to Asterisk, it
> first checks the account balance of the caller via Radius and then determine
> if the call should go through or not.
>
> I have an average experience in Asterisk but I'm quite new to Radius so I'm
> not sure if this setup is possible. Has anyone achieved this kind of setup?
>
> Thanks!
>
> Regards,
> Ronald
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>   http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 

Amit Anand


+91 9818559898
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users