Re: [asterisk-users] ACR Anonymous Call Rejection

[Gordon Henderson]

On Wed, 16 Sep 2009, Danny Nicholas wrote:

> What do you want your message to say?  I'd just use busy-pls-hold and the
> caller would eventually get the idea that you weren't going to talk to them.
> You could also consider these
> Off-duty
> Not-auth-pstn
> Not-taking-your-call
> Number-not-answering

tt-allbusy

Gordon

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Re: [asterisk-users] G729

[Gordon Henderson]

On Wed, 16 Sep 2009, Tilghman Lesher wrote:

> On Wednesday 16 September 2009 06:46:13 Khaled W Chehab wrote:
>> What g729 module should  I download ?
>
> You should download only the licensed g.729 module from Digium, after paying a
> $10 license per concurrent user.  All other modules have various problems (GPL
> violation or lack of paying the associated patent license).  You cannot use
> G.729 without paying the license fee until after all associated patents expire
> (sometime in 2014).

... or be in in a country that doesn't honour software patents or 
copyrights - which the OP might well be...

And there is a choice if the OP wants to pay the license fees - the OP 
does not have to dwonload the Digium one, there is another, competing one 
here:

   http://www.howlertech.com/products/howlets

It's cheaper too.

Gordon



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Re: [asterisk-users] limit concurrent calls on trunk supporting multiple DID

[Patrick]

Thank you Alex, I'll handle this programatically if there is no other way.

Best regards,
Patrick




On Thu, Sep 17, 2009 at 07:51, Alex Balashov  wrote:
> You can set some kind of counter in the dial plan, call an AGI script,
> use func_odbc to make database calls, or otherwise achieve this
> programatically.
>
> --
> Sent from mobile device
>
> On Sep 17, 2009, at 1:16 AM, Patrick 
> wrote:
>
>> Hello guys,
>>
>> I've one SIP trunk that support multiple DID. Only the trunk is
>> documented in sip.conf (called DID is taken from the sip-header in
>> real time).
>> I would like to limit the number of simultaneous calls on each DID. Is
>> there a way to achieve this ?
>> My understanding is that the SIP configuration parameter
>> "limitonpeers" will limit at the trunk level, right ?
>>
>> Thanks in advance
>> Patrick
>>
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[asterisk-users] ZAP and line disconnection detection

[M Shokuie]

Dear Folks,

Im looking for a way to detect if an analog line is connected to card or not
(Im using Sangoma A200). Im using the dialtone detection when dialing but
need a way to detect the disconnection of the line when it actually happens.

Anyone have any hints or tricks for this?

Regards.
--
Mohammad Sh.
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Re: [asterisk-users] limit concurrent calls on trunk supporting multiple DID

[Alex Balashov]

You can set some kind of counter in the dial plan, call an AGI script,  
use func_odbc to make database calls, or otherwise achieve this  
programatically.

--
Sent from mobile device

On Sep 17, 2009, at 1:16 AM, Patrick   
wrote:

> Hello guys,
>
> I've one SIP trunk that support multiple DID. Only the trunk is
> documented in sip.conf (called DID is taken from the sip-header in
> real time).
> I would like to limit the number of simultaneous calls on each DID. Is
> there a way to achieve this ?
> My understanding is that the SIP configuration parameter
> "limitonpeers" will limit at the trunk level, right ?
>
> Thanks in advance
> Patrick
>
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[asterisk-users] limit concurrent calls on trunk supporting multiple DID

[Patrick]

Hello guys,

I've one SIP trunk that support multiple DID. Only the trunk is
documented in sip.conf (called DID is taken from the sip-header in
real time).
I would like to limit the number of simultaneous calls on each DID. Is
there a way to achieve this ?
My understanding is that the SIP configuration parameter
"limitonpeers" will limit at the trunk level, right ?

Thanks in advance
Patrick

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Re: [asterisk-users] custom voicemail e-mail

[Patrick]

Hello Ron,

I was thinking also to replace the email sent by the voicemail by a php script.
My questions is simple, how do you manage to get the voicemail
variables from the php script ?
Or, maybe, you get from stdin the content of the "email" that should
be send via sendmail ?

Thanks in advance

Best regards,
Patrick

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Re: [asterisk-users] Connected Line ID for Asterisk 1.4

[Matt Riddell]

On 17/09/09 1:57 PM, Jeff LaCoursiere wrote:
>
> On Wed, 16 Sep 2009, Doug Lytle wrote:
>
>> Matt Riddell wrote:
>>> Basically, the phones are displaying 79 on the screen (the number the
>>> dial for pickup) - as you'd expect, but they'd like to see the CID of
>>> the person who called in.
>>>
>>>
>> There are patches against 1.4 that allow you to change the display to
>> anything that the phone will support and it'll be as a standard in the
>> 1.6.3 (??) series of Asterisk as a standard.
>>
>> The only problem is that the patches don't apply beyond 1.4.20.x
>>
>> You can read bug 8824
>>
>> https://issues.asterisk.org/view.php?id=8824
>>
>> I'll be staying with Asterisk 1.4.20.1 until I've had time to test out
>> the latest 1.6 series.
>>
>
> The last patch for RPID is marked for 1.4.23.1 (2/10/09) :
> https://issues.asterisk.org/file_download.php?file_id=21601&type=bug
>
> I've been running it on 1.4.23.1 since Feb.
>
> I just did an upgrade to 1.4.26.2 and their are rpid options showing in
> the sip peer info in the CLI (though I haven't tried to use it yet), so I
> expect that the reason there are no patches past 1.4.23.2 is that it has
> already been included in the release branch.

Wow, patch and a half!

:)

It doesn't apply cleanly, but with a bit of munging I've managed to get 
it in there.

Compiles without any warnings or problems.

Thanks for the help and I'll let you know how it goes!

-- 
Cheers,

Matt Riddell
Director
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Re: [asterisk-users] dCAP Exam

[Neeraj Chand]

Hi All,

Thanks for all the wonderful contributions, from cell phones right up to
proxies, etc...

Many thanks also to Tony Turner for the great advice.

As for Jared, what can I say...simply legend... :) 

I believe this is what I was after.

:)

For all those attending AstriconSee you there! 




> The most helpful thing would be a past scenario, something that has
come
> up in previous dCAP exams.
> 
> Can anyone send in a short descriptor of the final prac [real scenario
> that has happened before?]



Without going into too much detail on the exact details of the dCAP
exam, the general idea is this:  A small company has hired you to build
a typical small-business PBX using Asterisk, and you have 90 minutes to
get it up and running.  Given the time constraint, we really stick to
the basics, so there shouldn't be anything unexpected during the test.


-- 
Jared Smith
Training Manager
Digium, Inc.




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Re: [asterisk-users] Connected Line ID for Asterisk 1.4

[Jeff LaCoursiere]

On Wed, 16 Sep 2009, Doug Lytle wrote:

> Matt Riddell wrote:
>> Basically, the phones are displaying 79 on the screen (the number the
>> dial for pickup) - as you'd expect, but they'd like to see the CID of
>> the person who called in.
>>
>>
> There are patches against 1.4 that allow you to change the display to
> anything that the phone will support and it'll be as a standard in the
> 1.6.3 (??) series of Asterisk as a standard.
>
> The only problem is that the patches don't apply beyond 1.4.20.x
>
> You can read bug 8824
>
> https://issues.asterisk.org/view.php?id=8824
>
> I'll be staying with Asterisk 1.4.20.1 until I've had time to test out
> the latest 1.6 series.
>

The last patch for RPID is marked for 1.4.23.1 (2/10/09) :
https://issues.asterisk.org/file_download.php?file_id=21601&type=bug

I've been running it on 1.4.23.1 since Feb.

I just did an upgrade to 1.4.26.2 and their are rpid options showing in 
the sip peer info in the CLI (though I haven't tried to use it yet), so I 
expect that the reason there are no patches past 1.4.23.2 is that it has 
already been included in the release branch.

j

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[asterisk-users] web-meetme cbEnd.php not running - error

[Glen Ganderton]

Hey,

Ive installed web meetme and everything is working fine except no records
are being written to the cdr and participants tables, this is because the
cbEnd.php script is not running. Below is the output of the cbEnd.php when I
run in manually. I am running asterisk 1.4.20.1 and web meetme 3.1.0 and the
latest release's of PEAR,PHP and MySQL.


./cbEnd.php
PHP Strict Standards:  Assigning the return value of new by reference is
deprecated in /usr/share/php5/PEAR/PEAR.php on line 569
PHP Strict Standards:  Assigning the return value of new by reference is
deprecated in /usr/share/php5/PEAR/PEAR.php on line 572
PHP Strict Standards:  Non-static method DB::connect() should not be called
statically in /usr/local/apache2/htdocs/web-meetme/lib/database.php on line
8
PHP Strict Standards:  Non-static method DB::parseDSN() should not be called
statically in /usr/share/php5/PEAR/DB.php on line 520
PHP Strict Standards:  Non-static method PEAR::raiseError() should not be
called statically in /usr/share/php5/PEAR/DB.php on line 543
PHP Strict Standards:  Non-static method DB::errorMessage() should not be
called statically, assuming $this from incompatible context in
/usr/share/php5/PEAR/DB.php on line 965
PHP Strict Standards:  Non-static method DB::isError() should not be called
statically, assuming $this from incompatible context in
/usr/share/php5/PEAR/DB.php on line 688
PHP Strict Standards:  is_a(): Deprecated. Please use the instanceof
operator in /usr/share/php5/PEAR/DB.php on line 594
PHP Strict Standards:  Non-static method PEAR::getStaticProperty() should
not be called statically, assuming $this from incompatible context in
/usr/share/php5/PEAR/PEAR.php on line 867
PHP Strict Standards:  Non-static method DB::isError() should not be called
statically in /usr/local/apache2/htdocs/web-meetme/lib/database.php on line
9
PHP Strict Standards:  is_a(): Deprecated. Please use the instanceof
operator in /usr/share/php5/PEAR/DB.php on line 594
DB Error: not found
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Re: [asterisk-users] Connected Line ID for Asterisk 1.4

[Doug Lytle]

Matt Riddell wrote:
> Basically, the phones are displaying 79 on the screen (the number the 
> dial for pickup) - as you'd expect, but they'd like to see the CID of 
> the person who called in.
>
>   
There are patches against 1.4 that allow you to change the display to 
anything that the phone will support and it'll be as a standard in the 
1.6.3 (??) series of Asterisk as a standard.

The only problem is that the patches don't apply beyond 1.4.20.x

You can read bug 8824 

https://issues.asterisk.org/view.php?id=8824

I'll be staying with Asterisk 1.4.20.1 until I've had time to test out 
the latest 1.6 series.

Doug



-- 
Ben Franklin quote:

"Those who would give up Essential Liberty to purchase a little Temporary 
Safety, deserve neither Liberty nor Safety."


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[asterisk-users] H323 RTP Transmission error of packet

[Ruddy Gbaguidi]

Using H323 to reach another h323 switch, I have no audio and the following
error:

[Sep 16 15:45:55] DEBUG[10528]: rtp.c:2760 ast_rtp_raw_write: RTP
Transmission error of packet 21282 to XXX.XXX.XXX.XXX:6064: Invalid argument

[Sep 16 15:45:55] DEBUG[10528]: rtp.c:2760 ast_rtp_raw_write: RTP
Transmission error of packet 21283 to XXX.XXX.XXX.XXX:6064: Invalid argument

[Sep 16 15:45:55] DEBUG[10528]: rtp.c:2760 ast_rtp_raw_write: RTP
Transmission error of packet 21284 to XXX.XXX.XXX.XXX:6064: Invalid argument

[Sep 16 15:45:55] DEBUG[10528]: rtp.c:2760 ast_rtp_raw_write: RTP
Transmission error of packet 21285 to XXX.XXX.XXX.XXX:6064: Invalid argument

[Sep 16 15:45:55] DEBUG[10528]: rtp.c:2760 ast_rtp_raw_write: RTP
Transmission error of packet 21286 to XXX.XXX.XXX.XXX:6064: Invalid argument

 

Can you please tell me what I`m missing

I`m doing a quick dial like

Dial(h323/1514...@xxx.xxx.xxx.xxx)

 

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Re: [asterisk-users] Music on Hold

[Matt Riddell]

On 17/09/09 10:40 AM, Dan Saul wrote:
> This might be another piece of the puzzle:
>
> It would appear any application using playback functionality exits
> immediately. For example anything involving voicemail or playback. Phone
> calls work with no problem but not if asterisk must play something back.

Do you get an error in the console?

What do you have in /etc/asterisk/modules.conf

-- 
Cheers,

Matt Riddell
Director
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Re: [asterisk-users] res-crypto dependencies

[Michelle Dupuis]

Thanks all.  Turns out a package proplem was causing a conflict...I hard to
mess with packages to get all in and happy. 

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman
Lesher
Sent: Wednesday, September 16, 2009 4:43 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] res-crypto dependencies

On Wednesday 16 September 2009 15:10:57 Michelle Dupuis wrote:
> On Wednesday, September 16, 2009 4:03 PM, Tilghman Lesher wrote:
> > On Wednesday 16 September 2009 14:34:05 Michelle Dupuis wrote:
> > > I'm trying to enable res_crypto on a 1.4 installation, but 
> > > menuconfig says ssl is needed.  I've installed openssl, 
> > > openssl-devel, openssl-perl but it's still not happy.
> > >
> > > Anyone know what else is needed?
> >
> > Try libopenssl-devel
>
> No such package (under fedora 9)...
>
> Should I be lookin in other repo's or do you know what inside that 
> package it wants?  (In case fedora packages it somewhere else)

You may need to re-run ./configure after installing packages, if you haven't
already.

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

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Re: [asterisk-users] Music on Hold

[Dan Saul]

This might be another piece of the puzzle:

It would appear any application using playback functionality exits
immediately. For example anything involving voicemail or playback. Phone
calls work with no problem but not if asterisk must play something back.

The modules are loaded however...

Tsunami*CLI> module show like voicemail
Module Description  Use
Count
app_voicemail.so   Comedian Mail (Voicemail System)
0

I'm begining to think that the problem lies with my vendor's package.

On Wed, Sep 16, 2009 at 5:30 PM, Dan Saul  wrote:

> The files used to be "Frederic Chopin – Polonaised Op. 40-2.raw" I have
> since replaced the raw files with the original mp3s They are now as follows:
>
> [r...@tsunami musiconhold]# ls -l .
> total 13320
> -rw-r--r-- 1 asterisk asterisk 5415926 2009-09-16 17:18 hm1.mp3
> -rw-r--r-- 1 asterisk asterisk 8217974 2009-09-16 17:18 hm2.mp3
>
> I also have the same issue with the default files in /var/lib/asterisk/moh
> .
>
>
> On Wed, Sep 16, 2009 at 5:02 PM, Danny Nicholas  wrote:
>
>>  What are your actual file names (/etc/asterisk/musiconhold/Frederic
>> Chopin – Polonaised Op. 40-2.wav?)
>>
>>
>>  --
>>
>> *From:* asterisk-users-boun...@lists.digium.com [mailto:
>> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Dan Saul
>> *Sent:* Wednesday, September 16, 2009 4:50 PM
>> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
>> *Subject:* Re: [asterisk-users] Music on Hold
>>
>>
>>
>> That was a good shot in the dark, but sadly renaming it to something
>> simple (and removing all non ascii in the process) does not correct this.
>>
>> On Wed, Sep 16, 2009 at 4:37 PM, Danny Nicholas 
>> wrote:
>>
>> Just a “shot in the dark” but could MOH be choking on the “long file
>> names”?  (does it work on fred_chopin_pol_1)?
>>
>>
>>  --
>>
>> *From:* asterisk-users-boun...@lists.digium.com [mailto:
>> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Dan Saul
>> *Sent:* Wednesday, September 16, 2009 4:18 PM
>> *To:* asterisk-users@lists.digium.com
>> *Subject:* [asterisk-users] Music on Hold
>>
>>
>>
>> Hi,
>>
>> I have trouble getting MOH to work after an upgrade from asterisk 1.4 to
>> 1.6.1.4. The call goes on hold, MOH is started, and then stops right away.
>>
>> Here are the files both of type .raw:
>>
>> Tsunami*CLI> moh show files
>> Class: default
>> File: /etc/asterisk/musiconhold/Frédéric Chopin - Polonaises Op. 40-2
>> File: /etc/asterisk/musiconhold/Frédéric Chopin - Polonaises Op. 40-1
>>
>> These files were generated by SoX:
>> Channels   : 1
>> Sample Rate: 8000
>> Precision  : 16-bit
>> Sample Encoding: 16-bit Signed Integer PCM
>> Endian Type: little
>> Reverse Nibbles: no
>> Reverse Bits   : no
>> Comment: 'Processed by SoX'
>>
>> This prints in the asterisk console when you attempt to put someone in
>> hold:
>>
>> -- Started music on hold, class 'default', on
>> SIP/link2voip-sw1-02477668
>> -- Stopped music on hold on SIP/link2voip-sw1-02477668
>>
>> No errors are printed, however the other side just hears silence.
>>
>> Here is the full debug output (asterisk -rv):
>>
>>  == Using SIP RTP CoS mark 5
>> -- Executing [...@phones:1]
>> Goto("SIP/ATA-xx-L1-024b6d88", "1xx,1") in new stack
>> -- Goto (phones,1xx,1)
>> -- Executing [1xxx...@phones:1]
>> MSet("SIP/ATA-xx-L1-024b6d88", "oldcidnum=0") in new stack
>> -- Executing [1xxx...@phones:2]
>> MSet("SIP/ATA-xx-L1-024b6d88", "CALLERID(name)=""") in new stack
>> -- Executing [1xxx...@phones:3]
>> MSet("SIP/ATA-xx-L1-024b6d88", "CALLERID(num)=xx") in new
>> stack
>> -- Executing [1xxx...@phones:4]
>> Monitor("SIP/ATA-xx-L1-024b6d88", "wav,/tmp/out 0 2009-09-17 03h 04m
>> 51s CST xx,m") in new stack
>> -- Executing [1xxx...@phones:5]
>> Gosub("SIP/ATA-xx-L1-024b6d88", "ExternalDial,s,1(1xx)") in
>> new stack
>> -- Executing [...@externaldial:1]
>> MSet("SIP/ATA-xx-L1-024b6d88", "LOCAL(num)=1xx") in new
>> stack
>> -- Executing [...@externaldial:2]
>> MSet("SIP/ATA-xx-L1-024b6d88", "~~EXTEN~~=s") in new stack
>> -- Executing [...@externaldial:3]
>> Dial("SIP/ATA-xx-L1-024b6d88", "SIP/1xxx...@link2voip-sw1,120")
>> in new stack
>>   == Using SIP RTP CoS mark 5
>> -- Called 1xxx...@link2voip-sw1
>> -- SIP/link2voip-sw1-02477668 is making progress passing it to
>> SIP/ATA-xx-L1-024b6d88
>> -- SIP/link2voip-sw1-02477668 answered SIP/ATA-xx-L1-024b6d88
>> -- Started music on hold, class 'default', on
>> SIP/link2voip-sw1-02477668
>> -- Stopped music on hold on SIP/link2voip-sw1-02477668
>>> doing dnsmgr_lookup for 'sip.ca2.link2voip.com'
>>> doing dnsmgr_lookup for 'sip.ca1.link2voip.com'
>>   == Spawn ex

Re: [asterisk-users] Music on Hold

[Dan Saul]

The files used to be "Frederic Chopin – Polonaised Op. 40-2.raw" I have
since replaced the raw files with the original mp3s They are now as follows:

[r...@tsunami musiconhold]# ls -l .
total 13320
-rw-r--r-- 1 asterisk asterisk 5415926 2009-09-16 17:18 hm1.mp3
-rw-r--r-- 1 asterisk asterisk 8217974 2009-09-16 17:18 hm2.mp3

I also have the same issue with the default files in /var/lib/asterisk/moh .

On Wed, Sep 16, 2009 at 5:02 PM, Danny Nicholas  wrote:

>  What are your actual file names (/etc/asterisk/musiconhold/Frederic
> Chopin – Polonaised Op. 40-2.wav?)
>
>
>  --
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Dan Saul
> *Sent:* Wednesday, September 16, 2009 4:50 PM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] Music on Hold
>
>
>
> That was a good shot in the dark, but sadly renaming it to something simple
> (and removing all non ascii in the process) does not correct this.
>
> On Wed, Sep 16, 2009 at 4:37 PM, Danny Nicholas  wrote:
>
> Just a “shot in the dark” but could MOH be choking on the “long file
> names”?  (does it work on fred_chopin_pol_1)?
>
>
>  --
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Dan Saul
> *Sent:* Wednesday, September 16, 2009 4:18 PM
> *To:* asterisk-users@lists.digium.com
> *Subject:* [asterisk-users] Music on Hold
>
>
>
> Hi,
>
> I have trouble getting MOH to work after an upgrade from asterisk 1.4 to
> 1.6.1.4. The call goes on hold, MOH is started, and then stops right away.
>
> Here are the files both of type .raw:
>
> Tsunami*CLI> moh show files
> Class: default
> File: /etc/asterisk/musiconhold/Frédéric Chopin - Polonaises Op. 40-2
> File: /etc/asterisk/musiconhold/Frédéric Chopin - Polonaises Op. 40-1
>
> These files were generated by SoX:
> Channels   : 1
> Sample Rate: 8000
> Precision  : 16-bit
> Sample Encoding: 16-bit Signed Integer PCM
> Endian Type: little
> Reverse Nibbles: no
> Reverse Bits   : no
> Comment: 'Processed by SoX'
>
> This prints in the asterisk console when you attempt to put someone in
> hold:
>
> -- Started music on hold, class 'default', on
> SIP/link2voip-sw1-02477668
> -- Stopped music on hold on SIP/link2voip-sw1-02477668
>
> No errors are printed, however the other side just hears silence.
>
> Here is the full debug output (asterisk -rv):
>
>  == Using SIP RTP CoS mark 5
> -- Executing [...@phones:1] Goto("SIP/ATA-xx-L1-024b6d88",
> "1xx,1") in new stack
> -- Goto (phones,1xx,1)
> -- Executing [1xxx...@phones:1]
> MSet("SIP/ATA-xx-L1-024b6d88", "oldcidnum=0") in new stack
> -- Executing [1xxx...@phones:2]
> MSet("SIP/ATA-xx-L1-024b6d88", "CALLERID(name)=""") in new stack
> -- Executing [1xxx...@phones:3]
> MSet("SIP/ATA-xx-L1-024b6d88", "CALLERID(num)=xx") in new
> stack
> -- Executing [1xxx...@phones:4]
> Monitor("SIP/ATA-xx-L1-024b6d88", "wav,/tmp/out 0 2009-09-17 03h 04m
> 51s CST xx,m") in new stack
> -- Executing [1xxx...@phones:5]
> Gosub("SIP/ATA-xx-L1-024b6d88", "ExternalDial,s,1(1xx)") in
> new stack
> -- Executing [...@externaldial:1] MSet("SIP/ATA-xx-L1-024b6d88",
> "LOCAL(num)=1xx") in new stack
> -- Executing [...@externaldial:2] MSet("SIP/ATA-xx-L1-024b6d88",
> "~~EXTEN~~=s") in new stack
> -- Executing [...@externaldial:3] Dial("SIP/ATA-xx-L1-024b6d88",
> "SIP/1xxx...@link2voip-sw1,120") in new stack
>   == Using SIP RTP CoS mark 5
> -- Called 1xxx...@link2voip-sw1
> -- SIP/link2voip-sw1-02477668 is making progress passing it to
> SIP/ATA-xx-L1-024b6d88
> -- SIP/link2voip-sw1-02477668 answered SIP/ATA-xx-L1-024b6d88
> -- Started music on hold, class 'default', on
> SIP/link2voip-sw1-02477668
> -- Stopped music on hold on SIP/link2voip-sw1-02477668
>> doing dnsmgr_lookup for 'sip.ca2.link2voip.com'
>> doing dnsmgr_lookup for 'sip.ca1.link2voip.com'
>   == Spawn extension (ExternalDial, s, 3) exited non-zero on
> 'SIP/ATA-xx-L1-024b6d88'
>
> Any thoughts or ideas? If there were an error I could work on solving that,
> but there is none... Thanks.
>
>
> ___
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[asterisk-users] call-limit on dahdi channel

[Alex Samad]

Hi

how do i set the call-limit on a dahi line - its connected to the pstn
network - shared fax line.  How do i tell asterisk not to send more than
1 call there !


Alex

-- 
"Drug therapies are replacing a lot of medicines as we used to know it."

- George W. Bush
10/18/2000
St. Louis, MO
during the third presidential debate


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Re: [asterisk-users] How to configure a coverage pathfor anextension???

[Alex Samad]

On Wed, Sep 16, 2009 at 12:24:22PM -0700, Steve Edwards wrote:
> On Wed, 16 Sep 2009, Danny Nicholas wrote:
> 
> > I'd try this:
> > - exten => 4000,1,Dial(SIP/4000,20,ikKtT)
> > - exten => s-NOANSWER,1,Dial(SIP/4001,20,ikKtT)
> > - exten => s-NOANSWER,2,Voicemail(4000)
> > - exten => s-BUSY,1,Dial(SIP/4001,20,iKkTt)
> > - exten => s-BUSY,2,Voicemail(4000)
> > - exten => h,1,hangup
> 
> Don't you need a "goto(s-${DIALSTATUS},1)" in there somewhere?

I am curios as well, what tell it to do the  jump


-- 
If at first you do succeed, try to hide your astonishment.


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Re: [asterisk-users] Music on Hold

[Danny Nicholas]

What are your actual file names (/etc/asterisk/musiconhold/Frederic Chopin –
Polonaised Op. 40-2.wav?)

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Saul
Sent: Wednesday, September 16, 2009 4:50 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Music on Hold

 

That was a good shot in the dark, but sadly renaming it to something simple
(and removing all non ascii in the process) does not correct this.

On Wed, Sep 16, 2009 at 4:37 PM, Danny Nicholas  wrote:

Just a “shot in the dark” but could MOH be choking on the “long file names”?
(does it work on fred_chopin_pol_1)?

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Saul
Sent: Wednesday, September 16, 2009 4:18 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Music on Hold

 

Hi,

I have trouble getting MOH to work after an upgrade from asterisk 1.4 to
1.6.1.4. The call goes on hold, MOH is started, and then stops right away.

Here are the files both of type .raw:

Tsunami*CLI> moh show files
Class: default
File: /etc/asterisk/musiconhold/Frédéric Chopin - Polonaises Op. 40-2
File: /etc/asterisk/musiconhold/Frédéric Chopin - Polonaises Op. 40-1

These files were generated by SoX:
Channels   : 1
Sample Rate: 8000
Precision  : 16-bit
Sample Encoding: 16-bit Signed Integer PCM
Endian Type: little
Reverse Nibbles: no
Reverse Bits   : no
Comment: 'Processed by SoX'

This prints in the asterisk console when you attempt to put someone in hold:

-- Started music on hold, class 'default', on SIP/link2voip-sw1-02477668
-- Stopped music on hold on SIP/link2voip-sw1-02477668

No errors are printed, however the other side just hears silence.

Here is the full debug output (asterisk -rv):

 == Using SIP RTP CoS mark 5
-- Executing [...@phones:1] Goto("SIP/ATA-xx-L1-024b6d88",
"1xx,1") in new stack
-- Goto (phones,1xx,1)
-- Executing [1xxx...@phones:1]
MSet("SIP/ATA-xx-L1-024b6d88", "oldcidnum=0") in new stack
-- Executing [1xxx...@phones:2]
MSet("SIP/ATA-xx-L1-024b6d88", "CALLERID(name)=""") in new stack
-- Executing [1xxx...@phones:3]
MSet("SIP/ATA-xx-L1-024b6d88", "CALLERID(num)=xx") in new
stack
-- Executing [1xxx...@phones:4]
Monitor("SIP/ATA-xx-L1-024b6d88", "wav,/tmp/out 0 2009-09-17 03h 04m
51s CST xx,m") in new stack
-- Executing [1xxx...@phones:5]
Gosub("SIP/ATA-xx-L1-024b6d88", "ExternalDial,s,1(1xx)") in
new stack
-- Executing [...@externaldial:1] MSet("SIP/ATA-xx-L1-024b6d88",
"LOCAL(num)=1xx") in new stack
-- Executing [...@externaldial:2] MSet("SIP/ATA-xx-L1-024b6d88",
"~~EXTEN~~=s") in new stack
-- Executing [...@externaldial:3] Dial("SIP/ATA-xx-L1-024b6d88",
"SIP/1xxx...@link2voip-sw1,120") in new stack
  == Using SIP RTP CoS mark 5
-- Called 1xxx...@link2voip-sw1
-- SIP/link2voip-sw1-02477668 is making progress passing it to
SIP/ATA-xx-L1-024b6d88
-- SIP/link2voip-sw1-02477668 answered SIP/ATA-xx-L1-024b6d88
-- Started music on hold, class 'default', on SIP/link2voip-sw1-02477668
-- Stopped music on hold on SIP/link2voip-sw1-02477668
   > doing dnsmgr_lookup for 'sip.ca2.link2voip.com'
   > doing dnsmgr_lookup for 'sip.ca1.link2voip.com'
  == Spawn extension (ExternalDial, s, 3) exited non-zero on
'SIP/ATA-xx-L1-024b6d88'

Any thoughts or ideas? If there were an error I could work on solving that,
but there is none... Thanks.


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[asterisk-users] Connected Line ID for Asterisk 1.4

[Matt Riddell]

Hi,

We're using the Pickup app on a customer's site with Linksys phones.

Is there any way to display the callerid of the phone call you've picked 
up in 1.4?

I assume (rightly or wrongly) that this is connected line ID.

Basically, the phones are displaying 79 on the screen (the number the 
dial for pickup) - as you'd expect, but they'd like to see the CID of 
the person who called in.

I've managed to get the CID passed along during subsequent transfer of 
the call, but would love to be able to provide them with the icing on 
the cake :)

-- 
Cheers,

Matt Riddell
Director
___

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http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer)
http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems)

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Re: [asterisk-users] Music on Hold

[Miguel Molina]

Dan Saul escribió:
> Hi,
>
> I have trouble getting MOH to work after an upgrade from asterisk 1.4 
> to 1.6.1.4. The call goes on hold, MOH is started, and then stops 
> right away.
>
> Here are the files both of type .raw:
>
> Tsunami*CLI> moh show files
> Class: default
> File: /etc/asterisk/musiconhold/Frédéric Chopin - Polonaises Op. 40-2
> File: /etc/asterisk/musiconhold/Frédéric Chopin - Polonaises Op. 40-1
I would use a more friendly filename. That special accents and spaces 
maybe are confusing asterisk when it tries to read the files. Try 
renaming to chopin_op40-1 and chopin_op40-2 for example.

Cheers,

-- 
Ing. Miguel Molina
Grupo de Tecnología
Millenium Phone Center


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Re: [asterisk-users] Music on Hold

[Dan Saul]

That was a good shot in the dark, but sadly renaming it to something simple
(and removing all non ascii in the process) does not correct this.

On Wed, Sep 16, 2009 at 4:37 PM, Danny Nicholas  wrote:

>  Just a “shot in the dark” but could MOH be choking on the “long file
> names”?  (does it work on fred_chopin_pol_1)?
>
>
>  --
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Dan Saul
> *Sent:* Wednesday, September 16, 2009 4:18 PM
> *To:* asterisk-users@lists.digium.com
> *Subject:* [asterisk-users] Music on Hold
>
>
>
> Hi,
>
> I have trouble getting MOH to work after an upgrade from asterisk 1.4 to
> 1.6.1.4. The call goes on hold, MOH is started, and then stops right away.
>
> Here are the files both of type .raw:
>
> Tsunami*CLI> moh show files
> Class: default
> File: /etc/asterisk/musiconhold/Frédéric Chopin - Polonaises Op. 40-2
> File: /etc/asterisk/musiconhold/Frédéric Chopin - Polonaises Op. 40-1
>
> These files were generated by SoX:
> Channels   : 1
> Sample Rate: 8000
> Precision  : 16-bit
> Sample Encoding: 16-bit Signed Integer PCM
> Endian Type: little
> Reverse Nibbles: no
> Reverse Bits   : no
> Comment: 'Processed by SoX'
>
> This prints in the asterisk console when you attempt to put someone in
> hold:
>
> -- Started music on hold, class 'default', on
> SIP/link2voip-sw1-02477668
> -- Stopped music on hold on SIP/link2voip-sw1-02477668
>
> No errors are printed, however the other side just hears silence.
>
> Here is the full debug output (asterisk -rv):
>
>  == Using SIP RTP CoS mark 5
> -- Executing [...@phones:1] Goto("SIP/ATA-xx-L1-024b6d88",
> "1xx,1") in new stack
> -- Goto (phones,1xx,1)
> -- Executing [1xxx...@phones:1]
> MSet("SIP/ATA-xx-L1-024b6d88", "oldcidnum=0") in new stack
> -- Executing [1xxx...@phones:2]
> MSet("SIP/ATA-xx-L1-024b6d88", "CALLERID(name)=""") in new stack
> -- Executing [1xxx...@phones:3]
> MSet("SIP/ATA-xx-L1-024b6d88", "CALLERID(num)=xx") in new
> stack
> -- Executing [1xxx...@phones:4]
> Monitor("SIP/ATA-xx-L1-024b6d88", "wav,/tmp/out 0 2009-09-17 03h 04m
> 51s CST xx,m") in new stack
> -- Executing [1xxx...@phones:5]
> Gosub("SIP/ATA-xx-L1-024b6d88", "ExternalDial,s,1(1xx)") in
> new stack
> -- Executing [...@externaldial:1] MSet("SIP/ATA-xx-L1-024b6d88",
> "LOCAL(num)=1xx") in new stack
> -- Executing [...@externaldial:2] MSet("SIP/ATA-xx-L1-024b6d88",
> "~~EXTEN~~=s") in new stack
> -- Executing [...@externaldial:3] Dial("SIP/ATA-xx-L1-024b6d88",
> "SIP/1xxx...@link2voip-sw1,120") in new stack
>   == Using SIP RTP CoS mark 5
> -- Called 1xxx...@link2voip-sw1
> -- SIP/link2voip-sw1-02477668 is making progress passing it to
> SIP/ATA-xx-L1-024b6d88
> -- SIP/link2voip-sw1-02477668 answered SIP/ATA-xx-L1-024b6d88
> -- Started music on hold, class 'default', on
> SIP/link2voip-sw1-02477668
> -- Stopped music on hold on SIP/link2voip-sw1-02477668
>> doing dnsmgr_lookup for 'sip.ca2.link2voip.com'
>> doing dnsmgr_lookup for 'sip.ca1.link2voip.com'
>   == Spawn extension (ExternalDial, s, 3) exited non-zero on
> 'SIP/ATA-xx-L1-024b6d88'
>
> Any thoughts or ideas? If there were an error I could work on solving that,
> but there is none... Thanks.
>
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Re: [asterisk-users] Music on Hold

[Danny Nicholas]

Just a “shot in the dark” but could MOH be choking on the “long file names”?
(does it work on fred_chopin_pol_1)?

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Saul
Sent: Wednesday, September 16, 2009 4:18 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Music on Hold

 

Hi,

I have trouble getting MOH to work after an upgrade from asterisk 1.4 to
1.6.1.4. The call goes on hold, MOH is started, and then stops right away.

Here are the files both of type .raw:

Tsunami*CLI> moh show files
Class: default
File: /etc/asterisk/musiconhold/Frédéric Chopin - Polonaises Op. 40-2
File: /etc/asterisk/musiconhold/Frédéric Chopin - Polonaises Op. 40-1

These files were generated by SoX:
Channels   : 1
Sample Rate: 8000
Precision  : 16-bit
Sample Encoding: 16-bit Signed Integer PCM
Endian Type: little
Reverse Nibbles: no
Reverse Bits   : no
Comment: 'Processed by SoX'

This prints in the asterisk console when you attempt to put someone in hold:

-- Started music on hold, class 'default', on SIP/link2voip-sw1-02477668
-- Stopped music on hold on SIP/link2voip-sw1-02477668

No errors are printed, however the other side just hears silence.

Here is the full debug output (asterisk -rv):

 == Using SIP RTP CoS mark 5
-- Executing [...@phones:1] Goto("SIP/ATA-xx-L1-024b6d88",
"1xx,1") in new stack
-- Goto (phones,1xx,1)
-- Executing [1xxx...@phones:1]
MSet("SIP/ATA-xx-L1-024b6d88", "oldcidnum=0") in new stack
-- Executing [1xxx...@phones:2]
MSet("SIP/ATA-xx-L1-024b6d88", "CALLERID(name)=""") in new stack
-- Executing [1xxx...@phones:3]
MSet("SIP/ATA-xx-L1-024b6d88", "CALLERID(num)=xx") in new
stack
-- Executing [1xxx...@phones:4]
Monitor("SIP/ATA-xx-L1-024b6d88", "wav,/tmp/out 0 2009-09-17 03h 04m
51s CST xx,m") in new stack
-- Executing [1xxx...@phones:5]
Gosub("SIP/ATA-xx-L1-024b6d88", "ExternalDial,s,1(1xx)") in
new stack
-- Executing [...@externaldial:1] MSet("SIP/ATA-xx-L1-024b6d88",
"LOCAL(num)=1xx") in new stack
-- Executing [...@externaldial:2] MSet("SIP/ATA-xx-L1-024b6d88",
"~~EXTEN~~=s") in new stack
-- Executing [...@externaldial:3] Dial("SIP/ATA-xx-L1-024b6d88",
"SIP/1xxx...@link2voip-sw1,120") in new stack
  == Using SIP RTP CoS mark 5
-- Called 1xxx...@link2voip-sw1
-- SIP/link2voip-sw1-02477668 is making progress passing it to
SIP/ATA-xx-L1-024b6d88
-- SIP/link2voip-sw1-02477668 answered SIP/ATA-xx-L1-024b6d88
-- Started music on hold, class 'default', on SIP/link2voip-sw1-02477668
-- Stopped music on hold on SIP/link2voip-sw1-02477668
   > doing dnsmgr_lookup for 'sip.ca2.link2voip.com'
   > doing dnsmgr_lookup for 'sip.ca1.link2voip.com'
  == Spawn extension (ExternalDial, s, 3) exited non-zero on
'SIP/ATA-xx-L1-024b6d88'

Any thoughts or ideas? If there were an error I could work on solving that,
but there is none... Thanks.

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[asterisk-users] Music on Hold

[Dan Saul]

Hi,

I have trouble getting MOH to work after an upgrade from asterisk 1.4 to
1.6.1.4. The call goes on hold, MOH is started, and then stops right away.

Here are the files both of type .raw:

Tsunami*CLI> moh show files
Class: default
File: /etc/asterisk/musiconhold/Frédéric Chopin - Polonaises Op. 40-2
File: /etc/asterisk/musiconhold/Frédéric Chopin - Polonaises Op. 40-1

These files were generated by SoX:
Channels   : 1
Sample Rate: 8000
Precision  : 16-bit
Sample Encoding: 16-bit Signed Integer PCM
Endian Type: little
Reverse Nibbles: no
Reverse Bits   : no
Comment: 'Processed by SoX'

This prints in the asterisk console when you attempt to put someone in hold:

-- Started music on hold, class 'default', on SIP/link2voip-sw1-02477668
-- Stopped music on hold on SIP/link2voip-sw1-02477668

No errors are printed, however the other side just hears silence.

Here is the full debug output (asterisk -rv):

 == Using SIP RTP CoS mark 5
-- Executing [...@phones:1] Goto("SIP/ATA-xx-L1-024b6d88",
"1xx,1") in new stack
-- Goto (phones,1xx,1)
-- Executing [1xxx...@phones:1]
MSet("SIP/ATA-xx-L1-024b6d88", "oldcidnum=0") in new stack
-- Executing [1xxx...@phones:2]
MSet("SIP/ATA-xx-L1-024b6d88", "CALLERID(name)=""") in new stack
-- Executing [1xxx...@phones:3]
MSet("SIP/ATA-xx-L1-024b6d88", "CALLERID(num)=xx") in new
stack
-- Executing [1xxx...@phones:4]
Monitor("SIP/ATA-xx-L1-024b6d88", "wav,/tmp/out 0 2009-09-17 03h 04m
51s CST xx,m") in new stack
-- Executing [1xxx...@phones:5]
Gosub("SIP/ATA-xx-L1-024b6d88", "ExternalDial,s,1(1xx)") in
new stack
-- Executing [...@externaldial:1] MSet("SIP/ATA-xx-L1-024b6d88",
"LOCAL(num)=1xx") in new stack
-- Executing [...@externaldial:2] MSet("SIP/ATA-xx-L1-024b6d88",
"~~EXTEN~~=s") in new stack
-- Executing [...@externaldial:3] Dial("SIP/ATA-xx-L1-024b6d88",
"SIP/1xxx...@link2voip-sw1,120") in new stack
  == Using SIP RTP CoS mark 5
-- Called 1xxx...@link2voip-sw1
-- SIP/link2voip-sw1-02477668 is making progress passing it to
SIP/ATA-xx-L1-024b6d88
-- SIP/link2voip-sw1-02477668 answered SIP/ATA-xx-L1-024b6d88
-- Started music on hold, class 'default', on SIP/link2voip-sw1-02477668
-- Stopped music on hold on SIP/link2voip-sw1-02477668
   > doing dnsmgr_lookup for 'sip.ca2.link2voip.com'
   > doing dnsmgr_lookup for 'sip.ca1.link2voip.com'
  == Spawn extension (ExternalDial, s, 3) exited non-zero on
'SIP/ATA-xx-L1-024b6d88'

Any thoughts or ideas? If there were an error I could work on solving that,
but there is none... Thanks.
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Re: [asterisk-users] res-crypto dependencies

[Tilghman Lesher]

On Wednesday 16 September 2009 15:10:57 Michelle Dupuis wrote:
> On Wednesday, September 16, 2009 4:03 PM, Tilghman Lesher wrote:
> > On Wednesday 16 September 2009 14:34:05 Michelle Dupuis wrote:
> > > I'm trying to enable res_crypto on a 1.4 installation, but menuconfig
> > > says ssl is needed.  I've installed openssl, openssl-devel,
> > > openssl-perl but it's still not happy.
> > >
> > > Anyone know what else is needed?
> >
> > Try libopenssl-devel
>
> No such package (under fedora 9)...
>
> Should I be lookin in other repo's or do you know what inside that package
> it wants?  (In case fedora packages it somewhere else)

You may need to re-run ./configure after installing packages, if you haven't
already.

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

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Re: [asterisk-users] [SPAM] RE: dCAP Exam

[Pascal Bruno]

On Wed, Sep 16, 2009 at 2:37 PM, Zoa  wrote:

>
>  What if i send my twin brother to take the exam instead of me... ?
>
> z
>
>
If you think you cannot pass the test yourself, your twin wont be able to
pass it neither, he can be even worst than you

lol
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Re: [asterisk-users] res-crypto dependencies

[Danny Nicholas]

My .02 - you're probably going to have to modify
build_tools/menuselect-deps.  Tilghman would know this answer better than
me.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle
Dupuis
Sent: Wednesday, September 16, 2009 3:11 PM
To: 'Asterisk Users List'
Subject: Re: [asterisk-users] res-crypto dependencies

No such package (under fedora 9)...

Should I be lookin in other repo's or do you know what inside that package
it wants?  (In case fedora packages it somewhere else)

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman
Lesher
Sent: Wednesday, September 16, 2009 4:03 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] res-crypto dependencies

On Wednesday 16 September 2009 14:34:05 Michelle Dupuis wrote:
> I'm trying to enable res_crypto on a 1.4 installation, but menuconfig 
> says ssl is needed.  I've installed openssl, openssl-devel, 
> openssl-perl but it's still not happy.
>
> Anyone know what else is needed?

Try libopenssl-devel

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

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Re: [asterisk-users] res-crypto dependencies

[Michelle Dupuis]

No such package (under fedora 9)...

Should I be lookin in other repo's or do you know what inside that package
it wants?  (In case fedora packages it somewhere else)

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman
Lesher
Sent: Wednesday, September 16, 2009 4:03 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] res-crypto dependencies

On Wednesday 16 September 2009 14:34:05 Michelle Dupuis wrote:
> I'm trying to enable res_crypto on a 1.4 installation, but menuconfig 
> says ssl is needed.  I've installed openssl, openssl-devel, 
> openssl-perl but it's still not happy.
>
> Anyone know what else is needed?

Try libopenssl-devel

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

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Re: [asterisk-users] Meetme feature

[Steve Edwards]

On Wed, 16 Sep 2009, Anahi Ludue?a wrote:


Hi People, I want to do the following steps:



- Create a meetme between 2 persons.
- First, 1 person (user1) is entered into the meetme.
- Second, user2 is entered into the meetme. User2 is the marked user and 
also he is able to exit the conference by pressing #.
- If user2 exited by pressing #, I want the user1 would be able to save 
a voicemail to the user2.

How can I know if the user2 exited the conference by pressing # ?


Use "X" and MEETME_EXIT_CONTEXT.

--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
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Re: [asterisk-users] res-crypto dependencies

[Tilghman Lesher]

On Wednesday 16 September 2009 14:34:05 Michelle Dupuis wrote:
> I'm trying to enable res_crypto on a 1.4 installation, but menuconfig says
> ssl is needed.  I've installed openssl, openssl-devel, openssl-perl
> but it's still not happy.
>
> Anyone know what else is needed?

Try libopenssl-devel

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

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[asterisk-users] res-crypto dependencies

[Michelle Dupuis]

I'm trying to enable res_crypto on a 1.4 installation, but menuconfig says
ssl is needed.  I've installed openssl, openssl-devel, openssl-perl
but it's still not happy.
 
Anyone know what else is needed?
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[asterisk-users] Meetme feature

[Anahi Ludueña]

Hi People, I want to do the following steps:
- Create a meetme between 2 persons.
- First, 1 person (user1) is entered into the meetme.
- Second, user2 is entered into the meetme. User2 is the marked user and also 
he is able to exit the conference by pressing #.
- If user2 exited by pressing #, I want the user1 would be able to save a 
voicemail to the user2.
How can I know if the user2 exited the conference by pressing # ?
Thanks a lot, bye!



_
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Re: [asterisk-users] How to configure a coverage pathfor anextension???

[Steve Edwards]

On Wed, 16 Sep 2009, Danny Nicholas wrote:

> I'd try this:
> - exten => 4000,1,Dial(SIP/4000,20,ikKtT)
> - exten => s-NOANSWER,1,Dial(SIP/4001,20,ikKtT)
> - exten => s-NOANSWER,2,Voicemail(4000)
> - exten => s-BUSY,1,Dial(SIP/4001,20,iKkTt)
> - exten => s-BUSY,2,Voicemail(4000)
> - exten => h,1,hangup

Don't you need a "goto(s-${DIALSTATUS},1)" in there somewhere?

BTW, everybody seems to do "s-${DIALSTATUS}." Why not just 
"${DIALSTATUS}?"

"s-" doesn't seem to add any value to me.

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

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Re: [asterisk-users] [SPAM] RE: dCAP Exam

[Zoaaaaa]

 What if i send my twin brother to take the exam instead of me... ?

z

C. Savinovich wrote:
>
> What about if I use the browser from my cellular phone?
>
>  
>
> CS
>
>  
>
> *From:* asterisk-users-boun...@lists.digium.com 
> [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Pascal 
> Bruno
> *Sent:* Wednesday, September 16, 2009 10:21 PM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] dCAP Exam
>
>  
>
> I believe the administrator can see what is on your screen with screen 
> with those screen sharing stuff, this makes it harder a lil bit, and 
> www.boratproxy.com  becomes useless in that 
> case.
>
>  
>
> On Wed, Sep 16, 2009 at 1:28 PM, Steve Totaro 
>  > wrote:
>
>  
>
> On Wed, Sep 16, 2009 at 11:26 AM, Tilghman Lesher  > wrote:
>
> On Tuesday 15 September 2009 20:14:32 Neeraj Chand wrote:
> > Hmm...so by open book, that means access to the internet? Possible to
> > get own notes ?
>
> Yes, you have access to the Internet, but your access is proxied, and the
> administrator of the test can see everything that you access.  So it's 
> best
> for you stick with only general guides and not look for crib notes. 
>  If your
> test proctor believes you cheated, you fail.
>
>
> --
> Tilghman Lesher
> Digium, Inc. | Senior Software Developer
> twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
> Check us out at: www.digium.com  & 
> www.asterisk.org 
>
>
> Just tunnel your HTTP traffic over an SSH link and go to some dCAP 
> brain dump sites.  
>
> Or go to www.boratproxy.com  and confuse 
> their proxy.  ah too fun.
>
>
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>
>
>
> -- 
> Pascal B.
> http://www.kameleonlabs.com/
> Mike Ditka 
>   - "If 
> God had wanted man to play soccer, he wouldn't have given us arms."
>
> 
>
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Re: [asterisk-users] custom voicemail e-mail

[Ron]

thank you tilghman...that did the trick.. thanks again!

Tilghman Lesher wrote:
> On Wednesday 16 September 2009 12:38:59 Ron wrote:
>> yup my scripts starts with that line, is there anyway to check on the
>> logs if asterisk voicemail app is executing that command? thanks
> 
> Okay, next sanity check is that your script is chmod 755 (executable).
> 

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Re: [asterisk-users] [SPAM] RE: dCAP Exam

[John Millican]

C. Savinovich wrote:
> What about if I use the browser from my cellular phone?
> 
>  
> 
> CS
> 
>  
> 
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Pascal Bruno
> Sent: Wednesday, September 16, 2009 10:21 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] dCAP Exam
> 
>  
> 
> I believe the administrator can see what is on your screen with screen with
> those screen sharing stuff, this makes it harder a lil bit, and
> www.boratproxy.com becomes useless in that case.
> 
>  
> 
> On Wed, Sep 16, 2009 at 1:28 PM, Steve Totaro
>  wrote:
> 
>  
> 
> On Wed, Sep 16, 2009 at 11:26 AM, Tilghman Lesher 
> wrote:
> 
> On Tuesday 15 September 2009 20:14:32 Neeraj Chand wrote:
>> Hmm...so by open book, that means access to the internet? Possible to
>> get own notes ?
> 
> Yes, you have access to the Internet, but your access is proxied, and the
> administrator of the test can see everything that you access.  So it's best
> for you stick with only general guides and not look for crib notes.  If your
> test proctor believes you cheated, you fail.
> 
> 
> --
> Tilghman Lesher
> Digium, Inc. | Senior Software Developer
> twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
> Check us out at: www.digium.com & www.asterisk.org
> 
> 
> Just tunnel your HTTP traffic over an SSH link and go to some dCAP brain
> dump sites.   
> 
> Or go to www.boratproxy.com and confuse their proxy.  ah too fun.
> 
I can't resist:
After having taken many MS and Cisco tests in the past, it would seem
rather apparent to me that for the dcap, as with any other test, if you
know what you are doing you are all set and you don't have to try and
find ways to cheat that don't  look like cheating.

Disclaimer: I have not taken the dcap test yet!

JohnM


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Re: [asterisk-users] dCAP Exam

[Jared Smith]

On Wed, 2009-09-16 at 13:14 +1200, Neeraj Chand wrote:
> Hmm...so by open book, that means access to the internet? Possible to
> get own notes ? 

You get access to voip-info.org and searching Google to use as a
reference.  We don't allow copying/pasting of config files, or copying
files via the internet or USB sticks.

> The most helpful thing would be a past scenario, something that has come
> up in previous dCAP exams.
> 
> Can anyone send in a short descriptor of the final prac [real scenario
> that has happened before?]

Without going into too much detail on the exact details of the dCAP
exam, the general idea is this:  A small company has hired you to build
a typical small-business PBX using Asterisk, and you have 90 minutes to
get it up and running.  Given the time constraint, we really stick to
the basics, so there shouldn't be anything unexpected during the test.


-- 
Jared Smith
Training Manager
Digium, Inc.


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Re: [asterisk-users] dCAP Exam

[Jared Smith]

On Wed, 2009-09-16 at 13:28 -0400, Steve Totaro wrote:
> Just tunnel your HTTP traffic over an SSH link and go to some dCAP
> brain dump sites.   

Yes, there are all kinds of technical ways of trying to cover your
tracks... I've certainly seen a number of them.

That being said, it's pretty easy for me to tell whether someone
understands Asterisk or are just copying/pasting configurations from a
website.  Again, the emphasis on the dCAP exam is real-world knowledge
of how to build a simple small-business PBX with Asterisk.  If you've
used Asterisk in a professional capacity, it should be very
straightforward to pass the practical portion of the exam.  If you're an
Asterisk novice, you probably won't pass (even if you do copy/paste
configs from a website).

If you have further questions about the dCAP exam, I'd be happy to do
what I can to answer them.

-- 
Jared Smith
Training Manager
Digium, Inc.


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Re: [asterisk-users] custom voicemail e-mail

[Tilghman Lesher]

On Wednesday 16 September 2009 12:38:59 Ron wrote:
> yup my scripts starts with that line, is there anyway to check on the
> logs if asterisk voicemail app is executing that command? thanks

Okay, next sanity check is that your script is chmod 755 (executable).

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

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Re: [asterisk-users] [SPAM] RE: dCAP Exam

[Jared Smith]

On Thu, 2009-09-17 at 14:00 +0430, C. Savinovich wrote:
> What about if I use the browser from my cellular phone?

Sorry, cell phone use is not permitted during the testing.  We've had
students try to snap pictures of the exam with their cell phone cameras,
so we had to institute a policy against cell phone use.

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Training Manager
Digium, Inc.


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[asterisk-users] [SPAM] RE: dCAP Exam

[C. Savinovich]

What about if I use the browser from my cellular phone?

 

CS

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Pascal Bruno
Sent: Wednesday, September 16, 2009 10:21 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] dCAP Exam

 

I believe the administrator can see what is on your screen with screen with
those screen sharing stuff, this makes it harder a lil bit, and
www.boratproxy.com becomes useless in that case.

 

On Wed, Sep 16, 2009 at 1:28 PM, Steve Totaro
 wrote:

 

On Wed, Sep 16, 2009 at 11:26 AM, Tilghman Lesher 
wrote:

On Tuesday 15 September 2009 20:14:32 Neeraj Chand wrote:
> Hmm...so by open book, that means access to the internet? Possible to
> get own notes ?

Yes, you have access to the Internet, but your access is proxied, and the
administrator of the test can see everything that you access.  So it's best
for you stick with only general guides and not look for crib notes.  If your
test proctor believes you cheated, you fail.


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


Just tunnel your HTTP traffic over an SSH link and go to some dCAP brain
dump sites.   

Or go to www.boratproxy.com and confuse their proxy.  ah too fun.


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-- 
Pascal B.
http://www.kameleonlabs.com/
Mike Ditka    -
"If God had wanted man to play soccer, he wouldn't have given us arms." 

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Re: [asterisk-users] dCAP Exam

[Pascal Bruno]

I believe the administrator can see what is on your screen with screen with
those screen sharing stuff, this makes it harder a lil bit, and
www.boratproxy.com becomes useless in that case.

On Wed, Sep 16, 2009 at 1:28 PM, Steve Totaro <
stot...@totarotechnologies.com> wrote:

>
>
> On Wed, Sep 16, 2009 at 11:26 AM, Tilghman Lesher wrote:
>
>> On Tuesday 15 September 2009 20:14:32 Neeraj Chand wrote:
>> > Hmm...so by open book, that means access to the internet? Possible to
>> > get own notes ?
>>
>> Yes, you have access to the Internet, but your access is proxied, and the
>> administrator of the test can see everything that you access.  So it's
>> best
>> for you stick with only general guides and not look for crib notes.  If
>> your
>> test proctor believes you cheated, you fail.
>>
>> --
>> Tilghman Lesher
>> Digium, Inc. | Senior Software Developer
>> twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
>> Check us out at: www.digium.com & www.asterisk.org
>>
>
> Just tunnel your HTTP traffic over an SSH link and go to some dCAP brain
> dump sites.
>
> Or go to www.boratproxy.com and confuse their proxy.  ah too fun.
>
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http://www.kameleonlabs.com/
Mike Ditka   -
"If God had wanted man to play soccer, he wouldn't have given us arms."
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Re: [asterisk-users] custom voicemail e-mail

[Danny Nicholas]

Change /usr/bin/sendmail to /usr/bin/sh in voicemail.conf.  That will
disable the function in voicemail.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ron
Sent: Wednesday, September 16, 2009 12:45 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] custom voicemail e-mail

Hi Danny,

if the voicemail function is called then the AGI, wont the vooicemail 
function already send an e-mail before going to the AGI? Thanks!

Regards
Ron

Danny Nicholas wrote:
> The ODBC isn't having an effect, otherwise you couldn't run it
stand-alone.
> Voicemail.conf states that changing the /usr/bin/sendmail -t is done at
your
> own risk.   You could just do a system or AGI command to run your PHP
script
> whenever the voicemail application is called, like this:
> - exten => s,1,dial...
> - exten => s,2,voicemail...
> - exten => s,3,AGI(php.agi) or
> - exten => s,3,system("/usr/bin/php /usr/bin/voicemail.php)
> 
> 
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ron
> Sent: Wednesday, September 16, 2009 11:36 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] custom voicemail e-mail
> 
> Hi All,
> 
> I'm trying to use a php script to send voicemail e-mail so i can send 
> custom e-mail message based on what mailbox.
> 
> on my voicemail.conf i have
> 
> mailcmd=/var/www/voicemail.php
> 
> but when i tried to call an extension and goe to voicemail i'm not 
> receiving the e-mail.
> 
> but when i execute "php /var/www/voicemail.php" on the shell, i can 
> receive the e-mail.
> 
> how would i know the asterisk is actually executing that command? also 
> i'm using unixodbc for my voicemail
> 
> odbcstorage=mydb
> odbctable=mytable
> 
> not sure if that has an effect or not. TIA
> 
> Regards
> Ron
> 
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Re: [asterisk-users] custom voicemail e-mail

[Ron]

Hi Danny,

if the voicemail function is called then the AGI, wont the vooicemail 
function already send an e-mail before going to the AGI? Thanks!

Regards
Ron

Danny Nicholas wrote:
> The ODBC isn't having an effect, otherwise you couldn't run it stand-alone.
> Voicemail.conf states that changing the /usr/bin/sendmail -t is done at your
> own risk.   You could just do a system or AGI command to run your PHP script
> whenever the voicemail application is called, like this:
> - exten => s,1,dial...
> - exten => s,2,voicemail...
> - exten => s,3,AGI(php.agi) or
> - exten => s,3,system("/usr/bin/php /usr/bin/voicemail.php)
> 
> 
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ron
> Sent: Wednesday, September 16, 2009 11:36 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] custom voicemail e-mail
> 
> Hi All,
> 
> I'm trying to use a php script to send voicemail e-mail so i can send 
> custom e-mail message based on what mailbox.
> 
> on my voicemail.conf i have
> 
> mailcmd=/var/www/voicemail.php
> 
> but when i tried to call an extension and goe to voicemail i'm not 
> receiving the e-mail.
> 
> but when i execute "php /var/www/voicemail.php" on the shell, i can 
> receive the e-mail.
> 
> how would i know the asterisk is actually executing that command? also 
> i'm using unixodbc for my voicemail
> 
> odbcstorage=mydb
> odbctable=mytable
> 
> not sure if that has an effect or not. TIA
> 
> Regards
> Ron
> 
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Re: [asterisk-users] custom voicemail e-mail

[Ron]

Hi Tilghman,

yup my scripts starts with that line, is there anyway to check on the 
logs if asterisk voicemail app is executing that command? thanks

Regards
Ron

Tilghman Lesher wrote:
> On Wednesday 16 September 2009 11:35:31 Ron wrote:
>> Hi All,
>>
>> I'm trying to use a php script to send voicemail e-mail so i can send
>> custom e-mail message based on what mailbox.
>>
>> on my voicemail.conf i have
>>
>> mailcmd=/var/www/voicemail.php
>>
>> but when i tried to call an extension and goe to voicemail i'm not
>> receiving the e-mail.
>>
>> but when i execute "php /var/www/voicemail.php" on the shell, i can
>> receive the e-mail.
> 
> Make sure the first line of your script is:
> #!/usr/bin/php
> 

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Re: [asterisk-users] dCAP Exam

[Steve Totaro]

On Wed, Sep 16, 2009 at 11:26 AM, Tilghman Lesher wrote:

> On Tuesday 15 September 2009 20:14:32 Neeraj Chand wrote:
> > Hmm...so by open book, that means access to the internet? Possible to
> > get own notes ?
>
> Yes, you have access to the Internet, but your access is proxied, and the
> administrator of the test can see everything that you access.  So it's best
> for you stick with only general guides and not look for crib notes.  If
> your
> test proctor believes you cheated, you fail.
>
> --
> Tilghman Lesher
> Digium, Inc. | Senior Software Developer
> twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
> Check us out at: www.digium.com & www.asterisk.org
>

Just tunnel your HTTP traffic over an SSH link and go to some dCAP brain
dump sites.

Or go to www.boratproxy.com and confuse their proxy.  ah too fun.
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Re: [asterisk-users] ACR Anonymous Call Rejection

[Danny Nicholas]

What do you want your message to say?  I'd just use busy-pls-hold and the
caller would eventually get the idea that you weren't going to talk to them.
You could also consider these
Off-duty
Not-auth-pstn
Not-taking-your-call
Number-not-answering


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Barton Fisher
Sent: Wednesday, September 16, 2009 10:31 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] ACR Anonymous Call Rejection

Does any have or can point me to /ACR/ Anonymous Call Rejection message 
I can download?  The one I found was not not too clear.

Thanks, Bart


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Re: [asterisk-users] custom voicemail e-mail

[Tilghman Lesher]

On Wednesday 16 September 2009 11:35:31 Ron wrote:
> Hi All,
>
> I'm trying to use a php script to send voicemail e-mail so i can send
> custom e-mail message based on what mailbox.
>
> on my voicemail.conf i have
>
> mailcmd=/var/www/voicemail.php
>
> but when i tried to call an extension and goe to voicemail i'm not
> receiving the e-mail.
>
> but when i execute "php /var/www/voicemail.php" on the shell, i can
> receive the e-mail.

Make sure the first line of your script is:
#!/usr/bin/php

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

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Re: [asterisk-users] asterisk-users Digest, Vol 62, Issue 44

[adolfo]

No me encuentro en la oficina,
Volvere el proximo 28 de Septiembre.
Utilice los siguientes contactos:

v...@mildmac.es
rafael.mara...@mildmac.es
edua...@mildmac.es

Muchas Gracias



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Re: [asterisk-users] custom voicemail e-mail

[Danny Nicholas]

The ODBC isn't having an effect, otherwise you couldn't run it stand-alone.
Voicemail.conf states that changing the /usr/bin/sendmail -t is done at your
own risk.   You could just do a system or AGI command to run your PHP script
whenever the voicemail application is called, like this:
- exten => s,1,dial...
- exten => s,2,voicemail...
- exten => s,3,AGI(php.agi) or
- exten => s,3,system("/usr/bin/php /usr/bin/voicemail.php)


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ron
Sent: Wednesday, September 16, 2009 11:36 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] custom voicemail e-mail

Hi All,

I'm trying to use a php script to send voicemail e-mail so i can send 
custom e-mail message based on what mailbox.

on my voicemail.conf i have

mailcmd=/var/www/voicemail.php

but when i tried to call an extension and goe to voicemail i'm not 
receiving the e-mail.

but when i execute "php /var/www/voicemail.php" on the shell, i can 
receive the e-mail.

how would i know the asterisk is actually executing that command? also 
i'm using unixodbc for my voicemail

odbcstorage=mydb
odbctable=mytable

not sure if that has an effect or not. TIA

Regards
Ron

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[asterisk-users] custom voicemail e-mail

[Ron]

Hi All,

I'm trying to use a php script to send voicemail e-mail so i can send 
custom e-mail message based on what mailbox.

on my voicemail.conf i have

mailcmd=/var/www/voicemail.php

but when i tried to call an extension and goe to voicemail i'm not 
receiving the e-mail.

but when i execute "php /var/www/voicemail.php" on the shell, i can 
receive the e-mail.

how would i know the asterisk is actually executing that command? also 
i'm using unixodbc for my voicemail

odbcstorage=mydb
odbctable=mytable

not sure if that has an effect or not. TIA

Regards
Ron

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[asterisk-users] ACR Anonymous Call Rejection

[Barton Fisher]

Does any have or can point me to /ACR/ Anonymous Call Rejection message 
I can download?  The one I found was not not too clear.


Thanks, Bart
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Re: [asterisk-users] MeetMe in Macro

[Miguel Molina]

You can pass variables in the Originate Action, see 
http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+Originate.


Taken from there:

*Variable*: Channels variables to set (max 32). Variables will be set 
for both channels (local and connected).



 Example(Placing a call from a SIP channel to an extension, this
 will cause the outside call not to be placed until the SIP channel
 has picked up):


Action: Originate
Channel: SIP/101test
Context: default
Exten: 8135551212
Priority: 1
Callerid: 3125551212
Timeout: 3
*Variable: var1=23|var2=24|var3=25*
ActionID: ABC45678901234567890

That variables to send in the Originate action will become channel 
variables that can be retrieved on the context where the call goes. You 
can check that the variables exist on the context when the call is 
answered, for example:


[default]
exten => 8135551212,1,Noop(var1 is ${var1})
exten => 8135551212,n,Noop(var2 is ${var2})
exten => 8135551212,n,Noop(var3 is ${var3})
...

and so on... with no need to call Macro() or Gosub().

Cheers,

--
Ing. Miguel Molina
Grupo de Tecnología
Millenium Phone Center


Anahi Ludueña escribió:

Thanks,
I asked you to execute the GoSub from the Originate action, because I 
need to pass some parameters.
First, I created a macro since I could pass the parameters from 
originate. But the macro's problem is it doesn't jump to the 
particular extension (for example: h extension). So, when you told me 
that GoSub could "replace" the Macro, I thought it could be called 
from the Originate...
Do you know if there is another way to pass some parameters to a 
context from the Originate?

Thank you!


*

*

*Anahi Ludueña*

 




Date: Wed, 16 Sep 2009 10:27:26 -0500
From: mmol...@millenium.com.co
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] MeetMe in Macro

Hi,

The GoSub() application is intended for use in the dialplan, not to 
call it from a Originate Action. What is your specific need? You can 
Originate to a extension instead of an application an then if you need 
to execute a subroutine, you can use GoSub() and Return() then you 
need to on the called context.


You can check 
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Gosub but 
the example using the same context is not very clear.


A better example would be this:

[incoming]
exten => s,1,Answer()
exten => s,n,Noop(one)
exten => s,n,Noop(two)
exten => s,n,GoSub(mysub,s,1)
exten => s,n,Noop(I returned!)
exten => s,n,Hangup

[mysub]
exten => s,1,Noop(So I'm at a subroutine)
exten => s,n,Noop(I need to do special steps)
exten => s,n,Playback(tt-monkeys)
exten => s,n,Return()

Cheers,
--
Ing. Miguel Molina
Grupo de Tecnología
Millenium Phone Center
  


Anahi Ludueña escribió:

Thanks Miguel, It was my mistake.
So, my question is:
if I want to call the GoSub application from the Originate
Action (using AMI), what I need to put in the context parameter?
The GoSub will jump to a special context.
Thanks,




Date: Wed, 16 Sep 2009 09:34:31 -0500
From: mmol...@millenium.com.co 
To: asterisk-...@lists.digium.com
;
asterisk-users@lists.digium.com

Subject: Re: [asterisk-dev] MeetMe in Macro

Hi,

I didn't notice on my first answer, but we are on the -dev list
and this is not related to asterisk code developing. I will answer
you on the -users list, so we can continue the discussion there.

Cheers,

-- 
Ing. Miguel Molina

Grupo de Tecnología
Millenium Phone Center
  

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Re: [asterisk-users] MeetMe in Macro

[Anahi Ludueña]

Thanks,
I asked you to execute the GoSub from the Originate action, because I need to 
pass some parameters.
First, I created a macro since I could pass the parameters from originate. But 
the macro's problem is it doesn't jump to the particular extension (for 
example: h extension). So, when you told me that GoSub could "replace" the 
Macro, I thought it could be called from the Originate...
Do you know if there is another way to pass some parameters to a context from 
the Originate?
Thank you!





Anahi Ludueña
 
Date: Wed, 16 Sep 2009 10:27:26 -0500
From: mmol...@millenium.com.co
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] MeetMe in Macro






  


Hi,



The GoSub() application is intended for use in the dialplan, not to
call it from a Originate Action. What is your specific need? You can
Originate to a extension instead of an application an then if you need
to execute a subroutine, you can use GoSub() and Return() then you need
to on the called context.



You can check
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Gosub but the
example using the same context is not very clear.



A better example would be this:



[incoming]

exten => s,1,Answer()

exten => s,n,Noop(one)

exten => s,n,Noop(two)

exten => s,n,GoSub(mysub,s,1)

exten => s,n,Noop(I returned!)

exten => s,n,Hangup



[mysub]

exten => s,1,Noop(So I'm at a subroutine)

exten => s,n,Noop(I need to do special steps)

exten => s,n,Playback(tt-monkeys)

exten => s,n,Return()



Cheers,

-- 
Ing. Miguel Molina
Grupo de Tecnología
Millenium Phone Center



Anahi Ludueña escribió:

  Thanks Miguel, It was my mistake.

So, my question is:

if I want to call the GoSub application from
the Originate Action (using AMI), what I need to put in the context
parameter? The GoSub will jump to a special context.

Thanks,

  

  

  

  Date: Wed, 16 Sep 2009 09:34:31 -0500

From: mmol...@millenium.com.co

To: asterisk-...@lists.digium.com; asterisk-users@lists.digium.com

Subject: Re: [asterisk-dev] MeetMe in Macro

  

  
Hi,

  

I didn't notice on my first answer, but we are on the -dev list and
this is not related to asterisk code developing. I will answer you on
the -users list, so we can continue the discussion there.

  

Cheers,

  -- 
Ing. Miguel Molina
Grupo de Tecnología
Millenium Phone Center
  


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Re: [asterisk-users] G729

[Tilghman Lesher]

On Wednesday 16 September 2009 06:46:13 Khaled W Chehab wrote:
> What g729 module should  I download ?

You should download only the licensed g.729 module from Digium, after paying a
$10 license per concurrent user.  All other modules have various problems (GPL
violation or lack of paying the associated patent license).  You cannot use
G.729 without paying the license fee until after all associated patents expire
(sometime in 2014).

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

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Re: [asterisk-users] dCAP Exam

[Tilghman Lesher]

On Tuesday 15 September 2009 20:14:32 Neeraj Chand wrote:
> Hmm...so by open book, that means access to the internet? Possible to
> get own notes ?

Yes, you have access to the Internet, but your access is proxied, and the
administrator of the test can see everything that you access.  So it's best
for you stick with only general guides and not look for crib notes.  If your
test proctor believes you cheated, you fail.

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

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Re: [asterisk-users] How to configure a coverage path for anextension???

[Erik de Wild]

it should look something like

exten => 4000,1,Dial(SIP/4000,30,t)
exten => 4000,2,Goto(4001,1)

exten => 4001,1,Dial(SIP/4001,30,t)

If 4000,1 is answered it will never reach 4000,2

if 4000 is busy or not available for another reason it wil goto 4001,1

hope this is useful

Erik de Wild
Tripple-o

Verstuurd vanaf mijn iPhone

Op 16 sep 2009 om 16:24 heeft Ioan Indreias  het  
volgende geschreven:\



Hi Juan,

1. Please use the semicolon (;) character to comment your dialplan.  
Your choice (#) is intended for something else.


2. Probably you have to add the "j" option of Dial application (show  
application Dial), like:


exten => 4000,1,Dial(SIP/4000,20,iKkTtj)
exten => 4000,102,Dial(SIP/4001,20,iKkTtj)

3. For more hints you could check voip-info page.

HTH
Ioan Indreias
www.modulo.ro


On Wed, Sep 16, 2009 at 4:52 PM, Juan Cardoza   
wrote:
I comment all the lines in my extensions.conf file to work only with  
the

lines you provide me Danny:

Extensions.conf

[local-sip]

#exten => _4XXX,1,Dial(SIP/${EXTEN},10,tTr)
#exten => _5XXX,1,Dial(Dahdi/1/${EXTEN})
#exten => 164,1,Dial(Dahdi/1/${EXTEN})
#exten => 0550,1,Dial(Dahdi/1/${EXTEN})
#exten => _4XXX,3,Hangup()

[incoming]

exten => 4000,1,Dial(SIP/4000,20,iKkTt) <- I test this line only  
and it

works
exten => 4000,s-BUSY,Dial(SIP/4001,20,iKkTt) < When I add this  
line the

call arrives to the 4000
#exten => _4xxx,1,Dial(SIP/${EXTEN},10,tTr)

I dont answer the call and the Asterisk server drop the call.

[Sep 16 08:50:40] WARNING[1823]: chan_dahdi.c:4035 dahdi_enable_ec:  
Unable

to enable echo cancellation on channel 23 (No such device)
   -- Executing [4...@incoming:1] Dial("DAHDI/23-1", "SIP/ 
4000,20,iKkTt")

in new stack
   -- Called 4000
   -- SIP/4000-08a41440 is ringing
   -- SIP/4000-08a41440 answered DAHDI/23-1
   -- Accepting call from '' to '4000' on channel 0/22, span 1
   -- Executing [4...@incoming:1] Dial("DAHDI/22-1", "SIP/ 
4000,20,iKkTt")

in new stack
[Sep 16 08:50:50] WARNING[1823]: chan_dahdi.c:4035 dahdi_enable_ec:  
Unable

to enable echo cancellation on channel 22 (No such device)
   -- Called 4000
   -- SIP/4000-08a359c8 is ringing
[Sep 16 08:50:52] NOTICE[3394]: chan_sip.c:21804  
handle_request_subscribe:

Received SIP subscribe for peer without mailbox: 4000
   -- Nobody picked up in 2 ms
   -- Auto fallthrough, channel 'DAHDI/22-1' status is 'NOANSWER'
   -- Hungup 'DAHDI/22-1'
tp2asterisk01*CLI>

What could I need to fix this???
Thanks a lot for your help.
Jhon



-Mensaje original-
De: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] En nombre de Danny  
Nicholas

Enviado el: Miércoles, 16 de Septiembre de 2009 08:09 a.m.
Para: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Asunto: Re: [asterisk-users] How to configure a coverage path for
anextension???

In regular configuration (extensions.conf) this is one way to do it:
- exten => 4000,1,Dial(SIP/4000,20,iKkTt)
- exten => 4000,s-BUSY,Dial(SIP/4001,20,iKkTt)

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Juan  
Cardoza

Sent: Wednesday, September 16, 2009 8:04 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] How to configure a coverage path for
anextension???

I have been checking but nothing that clear my idea...

I have the extension 4000 and the idea is when this extension  
receive a call
and the extension 4000 is busy, the call from PSTN could be send to  
a second
extension, example: 4001, this need to happen only if the first  
extension is

busy.

If not, the call need to be take by the first station.
Please any one how can help me on this???

Best regards
Jhon


Teleperformance values: Integrity - Respect - Professionalism -  
Innovation -

Commitment

The information contained in this communication is privileged and
confidential.  The content is intended only for the use of the  
individual or

entity named above. If the reader of this message is not the intended
recipient, you are hereby notified that any dissemination,  
distribution or
copying of this communication is strictly prohibited.  If you have  
received
this communication in error, please notify me immediately by  
telephone or

e-mail, and delete this message from your systems.
Please consider the environmental impact of needlessly printing this  
e-mail.


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ast

Re: [asterisk-users] MeetMe in Macro

[Miguel Molina]

Hi,

The GoSub() application is intended for use in the dialplan, not to call 
it from a Originate Action. What is your specific need? You can 
Originate to a extension instead of an application an then if you need 
to execute a subroutine, you can use GoSub() and Return() then you need 
to on the called context.


You can check 
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Gosub but the 
example using the same context is not very clear.


A better example would be this:

[incoming]
exten => s,1,Answer()
exten => s,n,Noop(one)
exten => s,n,Noop(two)
exten => s,n,GoSub(mysub,s,1)
exten => s,n,Noop(I returned!)
exten => s,n,Hangup

[mysub]
exten => s,1,Noop(So I'm at a subroutine)
exten => s,n,Noop(I need to do special steps)
exten => s,n,Playback(tt-monkeys)
exten => s,n,Return()

Cheers,

--
Ing. Miguel Molina
Grupo de Tecnología
Millenium Phone Center


Anahi Ludueña escribió:

Thanks Miguel, It was my mistake.
So, my question is:
if I want to call the GoSub application from the Originate Action 
(using AMI), what I need to put in the context parameter? The GoSub 
will jump to a special context.

Thanks,




Date: Wed, 16 Sep 2009 09:34:31 -0500
From: mmol...@millenium.com.co
To: asterisk-...@lists.digium.com; asterisk-users@lists.digium.com
Subject: Re: [asterisk-dev] MeetMe in Macro

Hi,

I didn't notice on my first answer, but we are on the -dev list and 
this is not related to asterisk code developing. I will answer you on 
the -users list, so we can continue the discussion there.


Cheers,
--
Ing. Miguel Molina
Grupo de Tecnología
Millenium Phone Center
  


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Re: [asterisk-users] How to configure a coverage path for anextension???

[Juan Cardoza]

It works, thanks a lot, I also change the character for comments.

 

I am familiar with that page, I had been looking for the information in that
page also in google but noting.

 

Thanks to all for your help on this, let me continue doing some tests to
complete the task to do.

Best regards

John

 

De: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] En nombre de Ioan Indreias
Enviado el: Miércoles, 16 de Septiembre de 2009 09:24 a.m.
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Asunto: Re: [asterisk-users] How to configure a coverage path for
anextension???

 

Hi Juan,

 

1. Please use the semicolon (;) character to comment your dialplan. Your
choice (#) is intended for something else.

 

2. Probably you have to add the "j" option of Dial application (show
application Dial), like:

 

exten => 4000,1,Dial(SIP/4000,20,iKkTtj)
exten => 4000,102,Dial(SIP/4001,20,iKkTtj)

 

3. For more hints you could check voip-info
  page.

 

HTH

Ioan Indreias

www.modulo.ro

 

On Wed, Sep 16, 2009 at 4:52 PM, Juan Cardoza  wrote:

I comment all the lines in my extensions.conf file to work only with the
lines you provide me Danny:

Extensions.conf

[local-sip]

#exten => _4XXX,1,Dial(SIP/${EXTEN},10,tTr)
#exten => _5XXX,1,Dial(Dahdi/1/${EXTEN})
#exten => 164,1,Dial(Dahdi/1/${EXTEN})
#exten => 0550,1,Dial(Dahdi/1/${EXTEN})
#exten => _4XXX,3,Hangup()

[incoming]

exten => 4000,1,Dial(SIP/4000,20,iKkTt) <- I test this line only and it
works
exten => 4000,s-BUSY,Dial(SIP/4001,20,iKkTt) < When I add this line the
call arrives to the 4000
#exten => _4xxx,1,Dial(SIP/${EXTEN},10,tTr)

I dont answer the call and the Asterisk server drop the call.

[Sep 16 08:50:40] WARNING[1823]: chan_dahdi.c:4035 dahdi_enable_ec: Unable
to enable echo cancellation on channel 23 (No such device)
   -- Executing [4...@incoming:1] Dial("DAHDI/23-1", "SIP/4000,20,iKkTt")
in new stack
   -- Called 4000
   -- SIP/4000-08a41440 is ringing
   -- SIP/4000-08a41440 answered DAHDI/23-1
   -- Accepting call from '' to '4000' on channel 0/22, span 1
   -- Executing [4...@incoming:1] Dial("DAHDI/22-1", "SIP/4000,20,iKkTt")
in new stack
[Sep 16 08:50:50] WARNING[1823]: chan_dahdi.c:4035 dahdi_enable_ec: Unable
to enable echo cancellation on channel 22 (No such device)
   -- Called 4000
   -- SIP/4000-08a359c8 is ringing
[Sep 16 08:50:52] NOTICE[3394]: chan_sip.c:21804 handle_request_subscribe:
Received SIP subscribe for peer without mailbox: 4000
   -- Nobody picked up in 2 ms
   -- Auto fallthrough, channel 'DAHDI/22-1' status is 'NOANSWER'
   -- Hungup 'DAHDI/22-1'
tp2asterisk01*CLI>

What could I need to fix this???
Thanks a lot for your help.
Jhon



-Mensaje original-
De: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] En nombre de Danny Nicholas
Enviado el: Miércoles, 16 de Septiembre de 2009 08:09 a.m.
Para: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Asunto: Re: [asterisk-users] How to configure a coverage path for

anextension???

In regular configuration (extensions.conf) this is one way to do it:
- exten => 4000,1,Dial(SIP/4000,20,iKkTt)
- exten => 4000,s-BUSY,Dial(SIP/4001,20,iKkTt)

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Juan Cardoza
Sent: Wednesday, September 16, 2009 8:04 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] How to configure a coverage path for
anextension???

I have been checking but nothing that clear my idea...

I have the extension 4000 and the idea is when this extension receive a call
and the extension 4000 is busy, the call from PSTN could be send to a second
extension, example: 4001, this need to happen only if the first extension is
busy.

If not, the call need to be take by the first station.
Please any one how can help me on this???

Best regards
Jhon


Teleperformance values: Integrity - Respect - Professionalism - Innovation -
Commitment

The information contained in this communication is privileged and
confidential.  The content is intended only for the use of the individual or
entity named above. If the reader of this message is not the intended
recipient, you are hereby notified that any dissemination, distribution or
copying of this communication is strictly prohibited.  If you have received
this communication in error, please notify me immediately by telephone or
e-mail, and delete this message from your systems.
Please consider the environmental impact of needlessly printing this e-mail.

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Re: [asterisk-users] How to list ongoing calls from dialplan

[Miguel Molina]

Olivier escribió:



2009/9/16 Danny Nicholas mailto:da...@debsinc.com>>

Core show channels offers this on releases of asterisk that
use/display bridging (1.4.26 does bridging but does not show
bridged in status).

1. Here is an example

> core show channels
Channel  Location State   Application(Data)
0 active channels
0 active calls
21 calls processed

I don't see anything saying that channelA et is bridged to channelB

If there's no live channels you obviously can't see anything :-)

Try "core show channels verbose", the last field "Brigded to" will tell 
you if it's bridged or not, and to what channel.


Cheers,

--
Ing. Miguel Molina
Grupo de Tecnología
Millenium Phone Center

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[asterisk-users] MeetMe in Macro

[Anahi Ludueña]

Thanks Miguel, It was my mistake.
So, my question is:
if I want to call the GoSub application from
the Originate Action (using AMI), what I need to put in the context
parameter? The GoSub will jump to a special context.
Thanks,



Date: Wed, 16 Sep 2009 09:34:31 -0500
From: mmol...@millenium.com.co
To: asterisk-...@lists.digium.com; asterisk-users@lists.digium.com
Subject: Re: [asterisk-dev] MeetMe in Macro






  
  


Hi,



I didn't notice on my first answer, but we are on the -dev list and
this is not related to asterisk code developing. I will answer you on
the -users list, so we can continue the discussion there.



Cheers,

-- 
Ing. Miguel Molina
Grupo de Tecnología
Millenium Phone Center




_
Hay tantos ordenadores como personas. ¡Descubre ahora cuál eres tú!
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Re: [asterisk-users] [asterisk-dev] MeetMe in Macro

[Miguel Molina]

Hi,

I didn't notice on my first answer, but we are on the -dev list and this 
is not related to asterisk code developing. I will answer you on the 
-users list, so we can continue the discussion there.


Cheers,

--
Ing. Miguel Molina
Grupo de Tecnología
Millenium Phone Center


Anahi Ludueña escribió:

Hi, thanks Miguel.
I have another question: if I want to call the GoSub application from 
the Originate Action (using AMI), what I need to put in the context 
parameter? The GoSub will jump to a special context.

Thanks again





Date: Tue, 15 Sep 2009 17:17:38 -0500
From: mmol...@millenium.com.co
To: asterisk-...@lists.digium.com
Subject: Re: [asterisk-dev] MeetMe in Macro

Anahi Ludueña escribió:

Hi people, I have the following context in my dialplan:

[macro-meetme-part]
exten => s,1,Set(PART_OPT=${ARG1})
exten => s,n,Set(CONFNRO=${ARG2})
exten => s,n,GotoIf($[${ISNULL(${ARG3})}]?start:rec)
exten => s,n(rec),Set(MEETME_RECORDINGFILE=${ARG3}-${CONFNRO})
exten => s,n(start),MeetMe(${CONFNRO},${PART_OPT})
exten => s,n,Noop(Next Line)

The problem I have is that after the user hangs up the phone, the
dial plan doesn't go to the next line, but it shows the message:
/" Spawn extension (macro-meetme-part, s, 5) exited non-zero on
'SIP/2001-083238e8' in macro 'meetme-part'"/

Why does it happen? Is it right? Or I'm wrong?

Thanks a lot...

No, when a hangup occurs, going to the next line is not the default 
behavior on the dialplan. If there's a hangup that's where the channel 
ends just like the message reads and the CDR is proof of that. The 
only exception is when you define the special 'h' extension, which is 
executed after the channel is hungup, but I'm not sure if the 'h' 
extension works within a Macro. On that case I would play it safe and 
define it in a normal context. It is known that some special 
extensions, like the 'i' one, doesn't work within a Macro. That's one 
of the many reasons why the use of Gosub() as a replacement for 
Macro() is encouraged.


Cheers,

--
Ing. Miguel Molina
Grupo de Tecnología
Millenium Phone Center
PBX: (+57 1)6500800 ext. 1201
Fax: (+57 1)6500816
Móvil: (+57)3138873587 
  


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Re: [asterisk-users] How to list ongoing calls from dialplan

[Danny Nicholas]

Core show channels only works when a call is active;  here's an example:

Channel  Location State   Application(Data)

DAHDI/2-1(None)   Up  AppDial((Outgoing Line))

SIP/104-08461bd0 1-d...@macro-trunkdi Up
Dial(DAHDI/R1/w2975000|20|kKtT

2 active channels

1 active call

 

>From this example you can see that SIP/104 is bridged to DAHDI/2.  An AGI
might be your best bet to get the information you want.  I'd look at
voip-info.org for information.

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
Sent: Wednesday, September 16, 2009 9:20 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to list ongoing calls from dialplan

 

 

2009/9/16 Danny Nicholas 

Core show channels offers this on releases of asterisk that use/display
bridging (1.4.26 does bridging but does not show bridged in status).

1. Here is an example

> core show channels
Channel  Location State   Application(Data)
0 active channels
0 active calls
21 calls processed

I don't see anything saying that channelA et is bridged to channelB (I can
guess from the order used to display calls but can I rely on this ?)
Is there an option showing more explicitly those bridges ?

2. How can you access to the same data from dialplan ?
Funtion CHANNELS() is fine but I can't see a way to infere two channels are
bridged from it.

Regards

 


  _  


From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
Sent: Tuesday, September 15, 2009 5:45 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] How to list ongoing calls from dialplan

 

Hello,

Function CHANNELS() list current channels but how can you tell when 2
channels are bridged ?

Regards


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Re: [asterisk-users] How to configure a coverage path for anextension???

[Ioan Indreias]

Hi Juan,
1. Please use the semicolon (;) character to comment your dialplan. Your
choice (#) is intended for something else.

2. Probably you have to add the "j" option of Dial application (show
application Dial), like:

exten => 4000,1,Dial(SIP/4000,20,iKkTt*j*)
exten => 4000,102,Dial(SIP/4001,20,iKkTt*j*)

3. For more hints you could check
voip-info
 page.

HTH
Ioan Indreias
www.modulo.ro


On Wed, Sep 16, 2009 at 4:52 PM, Juan Cardoza  wrote:

> I comment all the lines in my extensions.conf file to work only with the
> lines you provide me Danny:
>
> Extensions.conf
>
> [local-sip]
>
> #exten => _4XXX,1,Dial(SIP/${EXTEN},10,tTr)
> #exten => _5XXX,1,Dial(Dahdi/1/${EXTEN})
> #exten => 164,1,Dial(Dahdi/1/${EXTEN})
> #exten => 0550,1,Dial(Dahdi/1/${EXTEN})
> #exten => _4XXX,3,Hangup()
>
> [incoming]
>
> exten => 4000,1,Dial(SIP/4000,20,iKkTt) <- I test this line only and it
> works
> exten => 4000,s-BUSY,Dial(SIP/4001,20,iKkTt) < When I add this line the
> call arrives to the 4000
> #exten => _4xxx,1,Dial(SIP/${EXTEN},10,tTr)
>
> I dont answer the call and the Asterisk server drop the call.
>
> [Sep 16 08:50:40] WARNING[1823]: chan_dahdi.c:4035 dahdi_enable_ec: Unable
> to enable echo cancellation on channel 23 (No such device)
>-- Executing [4...@incoming:1] Dial("DAHDI/23-1", "SIP/4000,20,iKkTt")
> in new stack
>-- Called 4000
>-- SIP/4000-08a41440 is ringing
>-- SIP/4000-08a41440 answered DAHDI/23-1
>-- Accepting call from '' to '4000' on channel 0/22, span 1
>-- Executing [4...@incoming:1] Dial("DAHDI/22-1", "SIP/4000,20,iKkTt")
> in new stack
> [Sep 16 08:50:50] WARNING[1823]: chan_dahdi.c:4035 dahdi_enable_ec: Unable
> to enable echo cancellation on channel 22 (No such device)
>-- Called 4000
>-- SIP/4000-08a359c8 is ringing
> [Sep 16 08:50:52] NOTICE[3394]: chan_sip.c:21804 handle_request_subscribe:
> Received SIP subscribe for peer without mailbox: 4000
>-- Nobody picked up in 2 ms
>-- Auto fallthrough, channel 'DAHDI/22-1' status is 'NOANSWER'
>-- Hungup 'DAHDI/22-1'
> tp2asterisk01*CLI>
>
> What could I need to fix this???
> Thanks a lot for your help.
> Jhon
>
>
>
> -Mensaje original-
> De: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] En nombre de Danny
> Nicholas
> Enviado el: Miércoles, 16 de Septiembre de 2009 08:09 a.m.
> Para: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Asunto: Re: [asterisk-users] How to configure a coverage path for
> anextension???
>
> In regular configuration (extensions.conf) this is one way to do it:
> - exten => 4000,1,Dial(SIP/4000,20,iKkTt)
> - exten => 4000,s-BUSY,Dial(SIP/4001,20,iKkTt)
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Juan Cardoza
> Sent: Wednesday, September 16, 2009 8:04 AM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: [asterisk-users] How to configure a coverage path for
> anextension???
>
> I have been checking but nothing that clear my idea...
>
> I have the extension 4000 and the idea is when this extension receive a
> call
> and the extension 4000 is busy, the call from PSTN could be send to a
> second
> extension, example: 4001, this need to happen only if the first extension
> is
> busy.
>
> If not, the call need to be take by the first station.
> Please any one how can help me on this???
>
> Best regards
> Jhon
>
>
> Teleperformance values: Integrity - Respect - Professionalism - Innovation
> -
> Commitment
>
> The information contained in this communication is privileged and
> confidential.  The content is intended only for the use of the individual
> or
> entity named above. If the reader of this message is not the intended
> recipient, you are hereby notified that any dissemination, distribution or
> copying of this communication is strictly prohibited.  If you have received
> this communication in error, please notify me immediately by telephone or
> e-mail, and delete this message from your systems.
> Please consider the environmental impact of needlessly printing this
> e-mail.
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Re: [asterisk-users] How to list ongoing calls from dialplan

[Olivier]

2009/9/16 Danny Nicholas 

>  Core show channels offers this on releases of asterisk that use/display
> bridging (1.4.26 does bridging but does not show bridged in status).
>
1. Here is an example

> core show channels
Channel  Location State   Application(Data)
0 active channels
0 active calls
21 calls processed

I don't see anything saying that channelA et is bridged to channelB (I can
guess from the order used to display calls but can I rely on this ?)
Is there an option showing more explicitly those bridges ?

2. How can you access to the same data from dialplan ?
Funtion CHANNELS() is fine but I can't see a way to infere two channels are
bridged from it.

Regards

>
>  --
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Olivier
> *Sent:* Tuesday, September 15, 2009 5:45 PM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* [asterisk-users] How to list ongoing calls from dialplan
>
>
>
> Hello,
>
> Function CHANNELS() list current channels but how can you tell when 2
> channels are bridged ?
>
> Regards
>
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>
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>   http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [asterisk-users] How to configure a coverage pathfor anextension???

[Danny Nicholas]

I'd try this:
- exten => 4000,1,Dial(SIP/4000,20,ikKtT)
- exten => s-NOANSWER,1,Dial(SIP/4001,20,ikKtT)
- exten => s-NOANSWER,2,Voicemail(4000)
- exten => s-BUSY,1,Dial(SIP/4001,20,iKkTt)
- exten => s-BUSY,2,Voicemail(4000)
- exten => h,1,hangup

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Juan Cardoza
Sent: Wednesday, September 16, 2009 8:53 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] How to configure a coverage pathfor
anextension???

I comment all the lines in my extensions.conf file to work only with the
lines you provide me Danny:

Extensions.conf

[local-sip]

#exten => _4XXX,1,Dial(SIP/${EXTEN},10,tTr)
#exten => _5XXX,1,Dial(Dahdi/1/${EXTEN})
#exten => 164,1,Dial(Dahdi/1/${EXTEN})
#exten => 0550,1,Dial(Dahdi/1/${EXTEN})
#exten => _4XXX,3,Hangup()

[incoming]

exten => 4000,1,Dial(SIP/4000,20,iKkTt) <- I test this line only and it
works
exten => 4000,s-BUSY,Dial(SIP/4001,20,iKkTt) < When I add this line the
call arrives to the 4000
#exten => _4xxx,1,Dial(SIP/${EXTEN},10,tTr)

I dont answer the call and the Asterisk server drop the call.

[Sep 16 08:50:40] WARNING[1823]: chan_dahdi.c:4035 dahdi_enable_ec: Unable
to enable echo cancellation on channel 23 (No such device)
-- Executing [4...@incoming:1] Dial("DAHDI/23-1", "SIP/4000,20,iKkTt")
in new stack
-- Called 4000 
-- SIP/4000-08a41440 is ringing
-- SIP/4000-08a41440 answered DAHDI/23-1
-- Accepting call from '' to '4000' on channel 0/22, span 1
-- Executing [4...@incoming:1] Dial("DAHDI/22-1", "SIP/4000,20,iKkTt")
in new stack
[Sep 16 08:50:50] WARNING[1823]: chan_dahdi.c:4035 dahdi_enable_ec: Unable
to enable echo cancellation on channel 22 (No such device)
-- Called 4000 
-- SIP/4000-08a359c8 is ringing
[Sep 16 08:50:52] NOTICE[3394]: chan_sip.c:21804 handle_request_subscribe:
Received SIP subscribe for peer without mailbox: 4000
-- Nobody picked up in 2 ms
-- Auto fallthrough, channel 'DAHDI/22-1' status is 'NOANSWER'
-- Hungup 'DAHDI/22-1'
tp2asterisk01*CLI>

What could I need to fix this???
Thanks a lot for your help.
Jhon



-Mensaje original-
De: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] En nombre de Danny Nicholas
Enviado el: Miércoles, 16 de Septiembre de 2009 08:09 a.m.
Para: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Asunto: Re: [asterisk-users] How to configure a coverage path for
anextension???

In regular configuration (extensions.conf) this is one way to do it:
- exten => 4000,1,Dial(SIP/4000,20,iKkTt)
- exten => 4000,s-BUSY,Dial(SIP/4001,20,iKkTt)

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Juan Cardoza
Sent: Wednesday, September 16, 2009 8:04 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] How to configure a coverage path for
anextension???

I have been checking but nothing that clear my idea...

I have the extension 4000 and the idea is when this extension receive a call
and the extension 4000 is busy, the call from PSTN could be send to a second
extension, example: 4001, this need to happen only if the first extension is
busy.

If not, the call need to be take by the first station.
Please any one how can help me on this???

Best regards
Jhon


Teleperformance values: Integrity - Respect - Professionalism - Innovation -
Commitment

The information contained in this communication is privileged and
confidential.  The content is intended only for the use of the individual or
entity named above. If the reader of this message is not the intended
recipient, you are hereby notified that any dissemination, distribution or
copying of this communication is strictly prohibited.  If you have received
this communication in error, please notify me immediately by telephone or
e-mail, and delete this message from your systems.
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Re: [asterisk-users] Reproducible crash - known bug?

[Jared Smith]

On Tue, 2009-09-15 at 22:41 -0500, Ian Pilcher wrote:
> Running asterisk-1.6.1-0.23.rc1.fc11.i586 on Fedora 11. I can
> reproducibly crash Asterisk by associating a single voicemail mailbox
> with two SIP extensions. For example:

Please open a report on our issue tracker at http://issues.asterisk.org/


-- 
Jared Smith
Training Manager
Digium, Inc.


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Re: [asterisk-users] How to configure a coverage path for anextension???

[Juan Cardoza]

I comment all the lines in my extensions.conf file to work only with the
lines you provide me Danny:

Extensions.conf

[local-sip]

#exten => _4XXX,1,Dial(SIP/${EXTEN},10,tTr)
#exten => _5XXX,1,Dial(Dahdi/1/${EXTEN})
#exten => 164,1,Dial(Dahdi/1/${EXTEN})
#exten => 0550,1,Dial(Dahdi/1/${EXTEN})
#exten => _4XXX,3,Hangup()

[incoming]

exten => 4000,1,Dial(SIP/4000,20,iKkTt) <- I test this line only and it
works
exten => 4000,s-BUSY,Dial(SIP/4001,20,iKkTt) < When I add this line the
call arrives to the 4000
#exten => _4xxx,1,Dial(SIP/${EXTEN},10,tTr)

I dont answer the call and the Asterisk server drop the call.

[Sep 16 08:50:40] WARNING[1823]: chan_dahdi.c:4035 dahdi_enable_ec: Unable
to enable echo cancellation on channel 23 (No such device)
-- Executing [4...@incoming:1] Dial("DAHDI/23-1", "SIP/4000,20,iKkTt")
in new stack
-- Called 4000 
-- SIP/4000-08a41440 is ringing
-- SIP/4000-08a41440 answered DAHDI/23-1
-- Accepting call from '' to '4000' on channel 0/22, span 1
-- Executing [4...@incoming:1] Dial("DAHDI/22-1", "SIP/4000,20,iKkTt")
in new stack
[Sep 16 08:50:50] WARNING[1823]: chan_dahdi.c:4035 dahdi_enable_ec: Unable
to enable echo cancellation on channel 22 (No such device)
-- Called 4000 
-- SIP/4000-08a359c8 is ringing
[Sep 16 08:50:52] NOTICE[3394]: chan_sip.c:21804 handle_request_subscribe:
Received SIP subscribe for peer without mailbox: 4000
-- Nobody picked up in 2 ms
-- Auto fallthrough, channel 'DAHDI/22-1' status is 'NOANSWER'
-- Hungup 'DAHDI/22-1'
tp2asterisk01*CLI>

What could I need to fix this???
Thanks a lot for your help.
Jhon



-Mensaje original-
De: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] En nombre de Danny Nicholas
Enviado el: Miércoles, 16 de Septiembre de 2009 08:09 a.m.
Para: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Asunto: Re: [asterisk-users] How to configure a coverage path for
anextension???

In regular configuration (extensions.conf) this is one way to do it:
- exten => 4000,1,Dial(SIP/4000,20,iKkTt)
- exten => 4000,s-BUSY,Dial(SIP/4001,20,iKkTt)

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Juan Cardoza
Sent: Wednesday, September 16, 2009 8:04 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] How to configure a coverage path for
anextension???

I have been checking but nothing that clear my idea...

I have the extension 4000 and the idea is when this extension receive a call
and the extension 4000 is busy, the call from PSTN could be send to a second
extension, example: 4001, this need to happen only if the first extension is
busy.

If not, the call need to be take by the first station.
Please any one how can help me on this???

Best regards
Jhon


Teleperformance values: Integrity - Respect - Professionalism - Innovation -
Commitment

The information contained in this communication is privileged and
confidential.  The content is intended only for the use of the individual or
entity named above. If the reader of this message is not the intended
recipient, you are hereby notified that any dissemination, distribution or
copying of this communication is strictly prohibited.  If you have received
this communication in error, please notify me immediately by telephone or
e-mail, and delete this message from your systems.
Please consider the environmental impact of needlessly printing this e-mail.

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Re: [asterisk-users] IVR seleCtion

[Matt Florell]

Hello,

ViciDial has IVR logging(pre-Queue) of IVRs set up through our web
interface(we call them Call Menus), but ViciDial does not use Asterisk
queues at all and it's logging is done entirely in a MySQL database. As a
side note, the logging done by ViciDial (non-IVR of course) is also fully
compatible with QueueMetrics.

MATT---


On 9/16/09, Maria Cristina Bayno  wrote:
>
> Hello Team,
>
> IVR selection of QUEUEMETRICS
>
> As we know queuemetrics had an IVR selection functionality where it can get
> the IVR keypress of a caller.
>
> We saw this link
> http://forum.queuemetrics.com/index.php?action=printpage;topic=503.0
>
> and upon checking, its only determined the Queue, I want to get is the per
> IVR of a caller.
>
> Can you help me guys regarding this? I want to implement this with the
> trixbox asterisk.
>
> Any idea? Thank you
>
> Cristina Bayno
> Technical Support
> Bitstop Network Services, inc.
>
>
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Re: [asterisk-users] How to configure a coverage path for anextension???

[Danny Nicholas]

In regular configuration (extensions.conf) this is one way to do it:
- exten => 4000,1,Dial(SIP/4000,20,iKkTt)
- exten => 4000,s-BUSY,Dial(SIP/4001,20,iKkTt)

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Juan Cardoza
Sent: Wednesday, September 16, 2009 8:04 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] How to configure a coverage path for
anextension???

I have been checking but nothing that clear my idea...

I have the extension 4000 and the idea is when this extension receive a call
and the extension 4000 is busy, the call from PSTN could be send to a second
extension, example: 4001, this need to happen only if the first extension is
busy.

If not, the call need to be take by the first station.
Please any one how can help me on this???

Best regards
Jhon


Teleperformance values: Integrity - Respect - Professionalism - Innovation -
Commitment

The information contained in this communication is privileged and
confidential.  The content is intended only for the use of the individual or
entity named above. If the reader of this message is not the intended
recipient, you are hereby notified that any dissemination, distribution or
copying of this communication is strictly prohibited.  If you have received
this communication in error, please notify me immediately by telephone or
e-mail, and delete this message from your systems.
Please consider the environmental impact of needlessly printing this e-mail.

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[asterisk-users] How to configure a coverage path for an extension???

[Juan Cardoza]

I have been checking but nothing that clear my idea...

I have the extension 4000 and the idea is when this extension receive a call
and the extension 4000 is busy, the call from PSTN could be send to a second
extension, example: 4001, this need to happen only if the first extension is
busy.

If not, the call need to be take by the first station.
Please any one how can help me on this???

Best regards
Jhon


Teleperformance values: Integrity - Respect - Professionalism - Innovation - 
Commitment

The information contained in this communication is privileged and confidential. 
 The content is intended only for the use of the individual or entity named 
above. If the reader of this message is not the intended recipient, you are 
hereby notified that any dissemination, distribution or copying of this 
communication is strictly prohibited.  If you have received this communication 
in error, please notify me immediately by telephone or e-mail, and delete this 
message from your systems.
Please consider the environmental impact of needlessly printing this e-mail.

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Re: [asterisk-users] How to list ongoing calls from dialplan

[Danny Nicholas]

Core show channels offers this on releases of asterisk that use/display
bridging (1.4.26 does bridging but does not show bridged in status).

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
Sent: Tuesday, September 15, 2009 5:45 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] How to list ongoing calls from dialplan

 

Hello,

Function CHANNELS() list current channels but how can you tell when 2
channels are bridged ?

Regards

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[asterisk-users] IVR seleCtion

[Maria Cristina Bayno]

Hello Team,

IVR selection of QUEUEMETRICS

As we know queuemetrics had an IVR selection functionality where it can get the 
IVR keypress of a caller.

We saw this link 
http://forum.queuemetrics.com/index.php?action=printpage;topic=503.0

and upon checking, its only determined the Queue, I want to get is the per IVR 
of a caller.

Can you help me guys regarding this? I want to implement this with the trixbox 
asterisk.

Any idea? Thank you

Cristina Bayno
Technical Support
Bitstop Network Services, inc.  



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[asterisk-users] G729

[Khaled W Chehab]

I have problemin g729 codec compatibility,I get the g729 module from
http://asterisk.hosting.lv/   and I have Asterisk 1.4.22-3 RPM

What g729 module should  I download ?

I already downloaded 

codec_g723-ast14-icc-glibc-pentium4.so

 

[trixbox1.localdomain asterisk]# cat /proc/cpuinfo 

processor   : 0

vendor_id   : GenuineIntel

cpu family  : 15

model   : 4

model name  : Intel(R) Xeon(TM) CPU 3.40GHz

stepping: 1

cpu MHz : 3399.733

cache size  : 1024 KB

fdiv_bug: no

hlt_bug : no

f00f_bug: no

coma_bug: no

fpu : yes

fpu_exception   : yes

cpuid level : 5

wp  : yes

flags   : fpu vme de pse tsc msr pae mce cx8 apic mtrr pge mca cmov
pat pse36 clflush dts acpi mmx fxsr sse sse2 ss constant_tsc up pni

bogomips: 6813.20

 

please advice

regards

 



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