Re: [asterisk-users] Duplicate DTMF

[John A. Sullivan III]

On Thu, 2009-09-10 at 15:26 -0400, Kristian Kielhofner wrote:
> On Wed, Sep 9, 2009 at 10:22 PM, John A. Sullivan III
>  wrote:
> > Hello, all.  I've come across a nasty problem just as we are ready to
> > put our first system into production.  During our final testing, we were
> > plagued with several "invalid extension" or "password incorrect"
> > messages when we knew the information entered was correct.  Upon
> > investigation, we saw that DTMF signals were occasionally but not
> > consistently duplicated.  We might dial extension 1234, see 1234 on the
> > phone from which we dialed, but see 112334 on the Asterisk console.
> >
> > We have seen this from cell phones calling via the PSTN (we use a SIP
> > trunking carrier and do not handle the PSTN interface ourselves); we've
> > seen it from land line phones via the PSTN, and have even seen it
> > internally from our own Snom SIP phones.
> >
> > dtmfmode=auto but we have also tried setting it to rfc2833 and we have
> > tried relaxdtmf set to both yes and no.
> >
> > We are running Asterisk 1.6.1.6 on CentOS 5.3.  We really don't know
> > what more to do to fix it.  Googling shows that others have had this
> > problem but I haven't seen a clear resolution other than playing with
> > relaxdtmf.  How do we solve this problem? Thanks - John
> 
>   Fairly typical for most SIP carriers...  My blog entry may be able
> to illuminate this a bit:
> 
> http://blog.krisk.org/2009/02/update-youve-been-waiting-for.html
> 
>   In short RFC2833 DTMF issues are fairly common.  It's troublesome
> enough when trying to go directly to the Tier 1 carriers themselves.
> More than likely you're dealing with a reseller ("carrier") that most
> likely inherits issues from their carrier and adds their own.
> 
>   A couple of weeks ago someone e-mailed me asking for RFC2833
> assistance with Asterisk and a carrier using Sonus.  Turns out:
> 
> a) The "carrier" was a reseller of various other carriers that use Sonus.
> b)  The "carrier" proxied RTP (and therefor RFC2833 events) through an
> Asterisk 1.2 machine; further mangling the RFC2833 events.
> 
>   Other than some drastic changes at the "carrier" there wasn't much
> that could be done...
> 
>   Sorry I can't offer more specific advice to your situation bit
> without an RTP packet capture there isn't much I (we) can do.
> 
> P.S. - Ignore any suggestions for gain, etc.  These are for Zap
> channels and do not apply to sip.  Changing anything in zapata.conf
> will not affect your situation.  I'm not even sure of the existence or
> purpose of relaxdtmf in sip.conf in Asterisk 1.4 or later.
> 
This may indeed be the case.  I hesitated to ask our carrier (with whom
we are quite happy thus far) since I believe we have seen this issue on
internal calls (but only once as opposed to the consistent problem with
external calls).  I did anyway and they put us on a different "switch."
That seems to have solved the problem.  Thanks - John
-- 
John A. Sullivan III
Open Source Development Corporation
+1 207-985-7880
jsulli...@opensourcedevel.com

http://www.spiritualoutreach.com
Making Christianity intelligible to secular society


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] MySql and custom CDR

[Tilghman Lesher]

On Friday 11 September 2009 16:45:30 Steve Edwards wrote:
> On Fri, 11 Sep 2009, Steve Edwards wrote:
> >> I'm a 1.2 Luddite, but I found cdr_addon_mysql.c pretty easy to hack
> >> on. For example, I added a channel variable named PRODUCT. Here's the
> >> code I used:
> >>
> >>channel_pointer = ast_get_channel_by_name_locked(channel);
> >>product_pointer = pbx_builtin_getvar_helper(channel_pointer
> >>, "PRODUCT");
> >>
> >> and then add ",product", ",'%s'" and product_pointer to the sprintf
> >> that builds the insert statement.
>
> On Fri, 11 Sep 2009, Tilghman Lesher wrote:
> > 1) There's no guarantee that the channel will still be alive at the time
> > the CDR is posted.  In fact, if you're doing bulk posting of CDRs, it's
> > pretty much guaranteed the channel will be gone.
>
> I do check for ast_get_channel_by_name_locked() and
> pbx_builtin_getvar_helper() returning 0 and syslog it. It never happens in
> this environment. I just didn't include it in the snippet.
>
> > 2) CDR variables were created for exactly this reason -- they are
> > allocated to the CDR, not the channel, and thus, they are available even
> > after the channel is destroyed.  They are available to use in 1.2.
>
> Well shame on me. I assumed that the CDR function only operated on the
> "standard" set of CDR variables. Thanks for the clarification. That would
> clean up that section of code a bit. Unfortunately I doubt the client will
> pay for fixing something that isn't failing.
>
> So if a CDR exists after the channel is destroyed, when is the CDR
> destroyed?

The in-memory CDR is destroyed after it is posted to the CDR backends (which
presumably go to disk, somewhere, although the core isn't concerned with
that).

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

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] No 64 bit binary for Fax for Asterisk

[Kevin P. Fleming]

Matt Riddell wrote:

> Consequently, on load it presents an error:
> 
> [Sep 12 10:25:35] WARNING[31306]: loader.c:368 load_dynamic_module: 
> Error loading module 'res_fax.so': /usr/lib/asterisk/modules/res_fax.so: 
> wrong ELF class: ELFCLASS32
> 
> Is there any plan to provide a 64 bit binary for my architechture?
> 
> I've installed the ia32-libs package as well, but no success.

There is a plan, but it will probably be at least another 60 days before
such modules are available for testing; it's unfortunate that the FAX
protocol stack that is used in Fax For Asterisk is not natively 64-bit
clean, but we're working on improving it. As you've found, you can't
load it into a copy of Asterisk that was compiled as a 64-bit binary...
your only choices are to learn how to cross-compile to produce 32-bit
binaries on your 64-bit system, or just install the 32-bit flavor of
your distribution.

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

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

[asterisk-users] No 64 bit binary for Fax for Asterisk

[Matt Riddell]

Hi,

Last night I was working on trying to get Fax for Asterisk installed.

I followed all the instructions, but noted that the download selector 
only provider an X86 32 Bit binary.

The machine I'm using is:

2.6.26-1-amd64 #1 SMP Fri Mar 13 17:46:45 UTC 2009 x86_64 GNU/Linux

Consequently, on load it presents an error:

[Sep 12 10:25:35] WARNING[31306]: loader.c:368 load_dynamic_module: 
Error loading module 'res_fax.so': /usr/lib/asterisk/modules/res_fax.so: 
wrong ELF class: ELFCLASS32

Is there any plan to provide a 64 bit binary for my architechture?

I've installed the ia32-libs package as well, but no success.

-- 
Cheers,

Matt Riddell
Director
___

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

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] All the four lights blinking

[Christian Victor]

2009/9/11 ABBAS SHAKEEL 

> Thanks you very much Kevin.I will try it by connecting one end of
>  Ethernet cable to one slot and other to second slot . Configuring one
> as pri_net and the other as pri_cpe.
>
> I will provide you feed on monday either i succed or not
>
> Remember that you CANT NOT use an Ethernet cross-over cable. You need to
get a E1 cross-over cable. Google for the pinout.

Christian
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] MySql and custom CDR

[Steve Edwards]

On Fri, 11 Sep 2009, Steve Edwards wrote:

>> I'm a 1.2 Luddite, but I found cdr_addon_mysql.c pretty easy to hack 
>> on. For example, I added a channel variable named PRODUCT. Here's the 
>> code I used:
>>
>>  channel_pointer = ast_get_channel_by_name_locked(channel);
>>  product_pointer = pbx_builtin_getvar_helper(channel_pointer
>>  , "PRODUCT");
>>
>> and then add ",product", ",'%s'" and product_pointer to the sprintf 
>> that builds the insert statement.

On Fri, 11 Sep 2009, Tilghman Lesher wrote:

> 1) There's no guarantee that the channel will still be alive at the time 
> the CDR is posted.  In fact, if you're doing bulk posting of CDRs, it's 
> pretty much guaranteed the channel will be gone.

I do check for ast_get_channel_by_name_locked() and 
pbx_builtin_getvar_helper() returning 0 and syslog it. It never happens in 
this environment. I just didn't include it in the snippet.

> 2) CDR variables were created for exactly this reason -- they are 
> allocated to the CDR, not the channel, and thus, they are available even 
> after the channel is destroyed.  They are available to use in 1.2.

Well shame on me. I assumed that the CDR function only operated on the 
"standard" set of CDR variables. Thanks for the clarification. That would 
clean up that section of code a bit. Unfortunately I doubt the client will 
pay for fixing something that isn't failing.

So if a CDR exists after the channel is destroyed, when is the CDR 
destroyed?

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

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] Call getting stucked !!

[David @ULC]

I am GETTING tired of CALLS getting stucked !!!

Any help ?

On Thu, Sep 10, 2009 at 2:52 AM, David @ULC  wrote:

>
> 8186223080 : No entry in the log file !!!
>
>
>
> On Thu, Sep 10, 2009 at 2:06 AM, David @ULC  wrote:
>
>>
>> Local/718186223...@d 718186223...@default Up
>> Dial(SIP/18186223...@sip209||t
>>
>>
>> I see this in my Asterisk when I do
>>
>> show channels
>>
>>
>>
>>
>> On Thu, Sep 10, 2009 at 1:49 AM, David @ULC  wrote:
>>
>>>
>>> I don't know where is the problem. May be with VOIPSwitch OR may be with
>>> Asterisk..
>>>
>>> Call getting stuck : My agent hang up the call but in Active calls , I
>>> see call connected and getting charged
>>>
>>> I use VOIP and NOT PSTN
>>>
>>> Didnt check the Asterisk CLI. Can I get any history of what asterisk
>>> REALLY had ?
>>>
>>>
>>>
>>>
>>> On Wed, Sep 9, 2009 at 11:41 PM, David @ULC  wrote:
>>>
 I am using asterisk.

 I also have an access to VOIPSwitch ver 2 where I can see live calls.

 Many times I have seen that my calls are getting strucked and then it
 gets disconneected after 59 mins ( as settings are done accordingly in
 VOIPSwitch)

 What could be the reason ?

>>>
>>>
>>
>
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] MySql and custom CDR

[Tilghman Lesher]

On Friday 11 September 2009 02:04:05 pm Steve Edwards wrote:
> > On Friday 11 September 2009 11:05:36 am Patrick wrote:
> >> I've migrated from CSV CDR to MySql CDR and the customization of my
> >> CDR's is not working anymore.
> >>
> >> Do you know if the cdr_mysql is supporting custom cdr's ? If not, is
> >> there any alternative/workaround ?
>
> On Fri, 11 Sep 2009, Tilghman Lesher wrote:
> > This module doesn't support custom CDRs, but starting in 1.6.0, it
> > supports something even better:  the ability to automatically map CDR
> > variables directly into columns of the same name.  You can also alias
> > various CDR variables into columns of different names, with the
> > [aliases] configuration section.  See the sample configuration file for
> > more details.
>
> I'm a 1.2 Luddite, but I found cdr_addon_mysql.c pretty easy to hack on.
> For example, I added a channel variable named PRODUCT. Here's the code I
> used:
>
>   channel_pointer = ast_get_channel_by_name_locked(channel);
>   product_pointer = pbx_builtin_getvar_helper(channel_pointer
>   , "PRODUCT");
>
> and then add ",product", ",'%s'" and product_pointer to the sprintf that
> builds the insert statement.

1) There's no guarantee that the channel will still be alive at the time the
CDR is posted.  In fact, if you're doing bulk posting of CDRs, it's pretty
much guaranteed the channel will be gone.

2) CDR variables were created for exactly this reason -- they are allocated
to the CDR, not the channel, and thus, they are available even after the
channel is destroyed.  They are available to use in 1.2.

-- 
Tilghman

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] Asterisk & Faxing

[Danny Nicholas]

I don't do international faxing, but I'll share these tidbits:
Asterisk Faxing is a hit-or-miss proposition; either you're a genius or a
fool (sometimes both in the same day).  
Faxing using POTS is a pretty simple proposition; SIP and T.38 add
complexity to the equation.  Simplification of the process as much as
possible is as good as Tylenol.
The timing on the VM may or may not bite you dependent upon the frame you
select to use for outgoing (POTS, T.38, etc.).

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Martin W.
Capdevielle
Sent: Friday, September 11, 2009 2:17 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Asterisk & Faxing

Greetings!

I have Asterisk 1.4.25.1 running as a Xen virtual machine using 64-bit 
domU and Dom0 with faxing and was curious about other users' experiences 
with it.  So far receiving fax works just fine, but I'm curious if 
anyone else is doing international faxing with asterisk.  If yes, could 
you please post your experiences?  Were there any issues you may have 
encountered that I should be looking out for?

Best regards,
Martin W. Capdevielle

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

[asterisk-users] www.chan_mobile.org seems dead ?

[hbk]

Any tip on where to find install doc on chan_mobile ?


Thnak you!
HB

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

[asterisk-users] Asterisk & Faxing

[Martin W. Capdevielle]

Greetings!

I have Asterisk 1.4.25.1 running as a Xen virtual machine using 64-bit 
domU and Dom0 with faxing and was curious about other users' experiences 
with it.  So far receiving fax works just fine, but I'm curious if 
anyone else is doing international faxing with asterisk.  If yes, could 
you please post your experiences?  Were there any issues you may have 
encountered that I should be looking out for?

Best regards,
Martin W. Capdevielle

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] MySql and custom CDR

[Steve Edwards]

> On Friday 11 September 2009 11:05:36 am Patrick wrote:
>>
>> I've migrated from CSV CDR to MySql CDR and the customization of my 
>> CDR's is not working anymore.
>>
>> Do you know if the cdr_mysql is supporting custom cdr's ? If not, is 
>> there any alternative/workaround ?

On Fri, 11 Sep 2009, Tilghman Lesher wrote:

> This module doesn't support custom CDRs, but starting in 1.6.0, it 
> supports something even better:  the ability to automatically map CDR 
> variables directly into columns of the same name.  You can also alias 
> various CDR variables into columns of different names, with the 
> [aliases] configuration section.  See the sample configuration file for 
> more details.

I'm a 1.2 Luddite, but I found cdr_addon_mysql.c pretty easy to hack on. 
For example, I added a channel variable named PRODUCT. Here's the code I 
used:

channel_pointer = ast_get_channel_by_name_locked(channel);
product_pointer = pbx_builtin_getvar_helper(channel_pointer
, "PRODUCT");

and then add ",product", ",'%s'" and product_pointer to the sprintf that 
builds the insert statement.

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

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] Parser for Asterisk Queue Logs

[Maria Cristina Bayno]

Thank you Miguel.

Cristina

--- On Fri, 9/11/09, Miguel Molina  wrote:

From: Miguel Molina 
Subject: Re: [asterisk-users] Parser for Asterisk Queue Logs
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Date: Friday, September 11, 2009, 4:50 PM




  
Maria Cristina Bayno escribió:

  

  
Hello Team,



Can you help me on this? I have attached here the
queue logs of my asterisk. I've searching a parser for this. I do not
know what are the meaning of that logs. Thank you so much.



Your response is highly appreciated.



Regards,

Cristina


  

  

Hi,



This should help:
http://www.voip-info.org/wiki/view/Asterisk+log+queue_log



At the end you will find the meaning of every field.



Cheers,

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

 

-Inline Attachment Follows-

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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


  ___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] MySql and custom CDR

[Tilghman Lesher]

On Friday 11 September 2009 11:05:36 am Patrick wrote:
> I've migrated from CSV CDR to MySql CDR and the customization of my
> CDR's is not working anymore.
>
> Do you know if the cdr_mysql is supporting custom cdr's ? If not, is
> there any alternative/workaround ?

This module doesn't support custom CDRs, but starting in 1.6.0, it supports
something even better:  the ability to automatically map CDR variables
directly into columns of the same name.  You can also alias various CDR
variables into columns of different names, with the [aliases] configuration
section.  See the sample configuration file for more details.

-- 
Tilghman

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] Parser for Asterisk Queue Logs

[Maria Cristina Bayno]

thanks for the reply..I will see it...


--- On Fri, 9/11/09, Daniel - Asterisk  wrote:

From: Daniel - Asterisk 
Subject: Re: [asterisk-users] Parser for Asterisk Queue Logs
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Date: Friday, September 11, 2009, 4:50 PM

Hi Cristina,

You can find meanings in queuelog.txt (or queuelog.tex in * 1.6), it's attached.

Daniel

On Fri, Sep 11, 2009 at 11:14 AM, Maria Cristina Bayno  
wrote:


Hello Team,

Can you help me on this? I have attached here the
queue logs of my asterisk. I've searching a parser for this. I do not
know what are the meaning of that logs. Thank you so much.

Your response is highly appreciated.

Regards,
Cristina



  
___

-- Bandwidth and Colocation Provided by http://www.api-digital.com --



AstriCon 2009 - October 13 - 15 Phoenix, Arizona

Register Now: http://www.astricon.net



asterisk-users mailing list

To UNSUBSCRIBE or update options visit:

   http://lists.digium.com/mailman/listinfo/asterisk-users



-Inline Attachment Follows-

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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


  ___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

[asterisk-users] Aastra 51i and PAP2T behind NAT

[John Millican]

OK this is the RTFM question of the day but I need a sanity check.
I have 2 Astra 51i phone and a linksys PAP2 on a single DSL connection.

2 Aastra 51i-|
 |-NAT on dsl moden--(Internet)--Asterisk
PAP2t|

The DSL modem/router which has QOS set for the src and dest to the * box
the PAP2 has both lines registered @ ports 5060 and 5061 and work like a
charm.  one of the the aastra's registered at port 1025 worked all day
but the showed no service and lost registration over night sometime.
this happens with much regularity. I am looking for the docs on these
phones to see if they have a NAT keep alive option.  Does this sound
like a reasonable place to start for a solution?

Thanks in advance
-- 
JohnM


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] Hiding voiemailbox/entry from directory

[Darrick Hartman]

Use the 'hidefromdir' option in voicemail.conf for that particular 
entry.  This should be clearly documented in the example voicemail.conf 
file.

Michelle Dupuis wrote:
> I have internal mailboxes that I don't want visible to callers going 
> through the directory.  Is it possible (in * 1.4) to hide mailboxes fom 
> the directory, without creating a new context?

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

[asterisk-users] Hiding voiemailbox/entry from directory

[Michelle Dupuis]

I have internal mailboxes that I don't want visible to callers going through
the directory.  Is it possible (in * 1.4) to hide mailboxes fom the
directory, without creating a new context?
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] Parser for Asterisk Queue Logs

[Daniel - Asterisk]

Hi Cristina,

You can find meanings in queuelog.txt (or queuelog.tex in * 1.6), it's
attached.

Daniel

On Fri, Sep 11, 2009 at 11:14 AM, Maria Cristina Bayno  wrote:

> Hello Team,
>
> Can you help me on this? I have attached here the queue logs of my
> asterisk. I've searching a parser for this. I do not know what are the
> meaning of that logs. Thank you so much.
>
> Your response is highly appreciated.
>
> Regards,
> Cristina
>
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> AstriCon 2009 - October 13 - 15 Phoenix, Arizona
> Register Now: http://www.astricon.net
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
Queue Log Information
=

In order to properly manage ACD queues, it is important to be able to
keep track of details of call setups and teardowns in much greater detail
than traditional call detail records provide.  In order to support this,
extensive and detailed tracing of every queued call is stored in the
queue log, located (by default) in /var/log/asterisk/queue_log.

These are the events (and associated information) in the queue log:

ABANDON(position|origposition|waittime)
The caller abandoned their position in the queue.  The position is the
caller's position in the queue when they hungup, the origposition is
the original position the caller was when they first entered the
queue, and the waittime is how long the call had been waiting in the 
queue at the time of disconnect.

AGENTDUMP
The agent dumped the caller while listening to the queue announcement.

AGENTLOGIN(channel)
The agent logged in.  The channel is recorded.

AGENTCALLBACKLOGIN(ex...@context)
The callback agent logged in.  The login extension and context is recorded.

AGENTLOGOFF(channel|logintime)
The agent logged off.  The channel is recorded, along with the total time
the agent was logged in.

AGENTCALLBACKLOGOFF(ex...@context|logintime|reason)
The callback agent logged off.  The last login extension and context is
recorded, along with the total time the agent was logged in, and the
reason for the logoff if it was not a normal logoff 
(e.g., Autologoff, Chanunavail)

COMPLETEAGENT(holdtime|calltime|origposition)
The caller was connected to an agent, and the call was terminated normally
by the *agent*.  The caller's hold time and the length of the call are both
recorded.  The caller's original position in the queue is recorded in
origposition.

COMPLETECALLER(holdtime|calltime|origposition)
The caller was connected to an agent, and the call was terminated normally
by the *caller*.  The caller's hold time and the length of the call are both
recorded.  The caller's original position in the queue is recorded in
origposition.

CONFIGRELOAD
The configuration has been reloaded (e.g. with asterisk -rx reload)

CONNECT(holdtime|bridgedchanneluniqueid)
The caller was connected to an agent.  Hold time represents the amount
of time the caller was on hold. The bridged channel unique ID contains
the unique ID of the queue member channel that is taking the call. This
is useful when trying to link recording filenames to a particular
call in the queue.

ENTERQUEUE(url|callerid)
A call has entered the queue.  URL (if specified) and Caller*ID are placed
in the log.

EXITEMPTY(position|origposition|waittime)
The caller was exited from the queue forcefully because the queue had no
reachable members and it's configured to do that to callers when there
are no reachable members. The position is the caller's position in the
queue when they hungup, the origposition is the original position the 
caller was when they first entered the queue, and the waittime is how 
long the call had been waiting in the queue at the time of disconnect.

EXITWITHKEY(key|position)
The caller elected to use a menu key to exit the queue.  The key and
the caller's position in the queue are recorded.

EXITWITHTIMEOUT(position)
The caller was on hold too long and the timeout expired.

QUEUESTART
The queueing system has been started for the first time this session.

RINGNOANSWER(ringtime)
After trying for ringtime ms to connect to the available queue member,
the attempt ended without the member picking up the call. Bad queue
member!

SYSCOMPAT
A call was answered by an agent, but the call was dropped because the 
channels were not compatible.

TRANSFER(extension|context|holdtime|calltime)
Caller was transferred to a different extension.  Context and extension
are recorded. The caller's hold time and the length of the call are both
recorded. PLEASE remember that transfers performed by SIP UA's by way
of a reinvite may not always be caught by Asterisk and trigger off this
event. The only way to be 100% sure that you will get this event when
a transfer is performed by a queue member is to use the built-in transfer
functionality of Asterisk.

___
-- Ba

Re: [asterisk-users] Parser for Asterisk Queue Logs

[Miguel Molina]

Maria Cristina Bayno escribió:

Hello Team,

Can you help me on this? I have attached here the queue logs of my 
asterisk. I've searching a parser for this. I do not know what are the 
meaning of that logs. Thank you so much.


Your response is highly appreciated.

Regards,
Cristina


Hi,

This should help: http://www.voip-info.org/wiki/view/Asterisk+log+queue_log

At the end you will find the meaning of every field.

Cheers,

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

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

[asterisk-users] Parser for Asterisk Queue Logs

[Maria Cristina Bayno]

Hello Team,

Can you help me on this? I have attached here the
queue logs of my asterisk. I've searching a parser for this. I do not
know what are the meaning of that logs. Thank you so much.

Your response is highly appreciated.

Regards,
Cristina



  1252661855|1252661843.25443|7200|NONE|ENTERQUEUE||3026918112
1252661868|1252661843.25443|7200|Local/7...@from-internal-f6ec,1|CONNECT|13
1252661890|1252661843.25443|7200|Local/7...@from-internal-f6ec,1|COMPLETECALLER|13|22|1
1252661970|1252661957.25449|7200|NONE|ENTERQUEUE||6022790400
1252661974|1252661957.25449|7200|Local/7...@from-internal-ce8a,1|CONNECT|4
1252662010|1252661957.25449|7200|Local/7...@from-internal-ce8a,1|COMPLETECALLER|4|36|1
1252662220|1252662207.25455|7200|NONE|ENTERQUEUE||3027406891
1252662228|1252662207.25455|7200|Local/7...@from-internal-caaa,1|CONNECT|8
1252662232|1252662207.25455|7200|Local/7...@from-internal-caaa,1|COMPLETECALLER|8|4|1


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

[asterisk-users] MySql and custom CDR

[Patrick]

Hello,

I've migrated from CSV CDR to MySql CDR and the customization of my
CDR's is not working anymore.

Do you know if the cdr_mysql is supporting custom cdr's ? If not, is
there any alternative/workaround ?

Best regards,
Patrick

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] Voicemail by email with HTML

[Patrick]

Hello Danny,

I've also the same question :-)

I've tried to find more information on the "pup" mime enabled program
but I haven't find something on the internet (every search refers to
puppy linux :-( )

Can you give more info ? Where can I find it ?

Thanks in advance
Patrick

On Fri, Sep 11, 2009 at 16:42, Danny Nicholas  wrote:
> Change the client in voicemail.conf from sendmail to a “MIME-enabled” client
> like pup.
>
>
>
> 
>
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gabriel Ortiz
> Lour
> Sent: Friday, September 11, 2009 9:36 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] Voicemail by email with HTML
>
>
>
> Hi all,
>
>   I'm trying to send an email with the voicemail details and I want to send
> a HTML link on it to make a click2call to the voicemail main, but the email
> is send with 'text/plain' encoding and thus it will not show the link, but
> the HTML in plain text on the body of the email,
>
>   How can I change the enconding to 'text/html' so the link will get
> displayed correctly?
>
> Thanks,
> Gabriel
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> AstriCon 2009 - October 13 - 15 Phoenix, Arizona
> Register Now: http://www.astricon.net
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

[asterisk-users] Voicemail by email with HTML

[Gabriel Ortiz Lour]

Hi all,

  I'm trying to send an email with the voicemail details and I want to send
a HTML link on it to make a click2call to the voicemail main, but the email
is send with 'text/plain' encoding and thus it will not show the link, but
the HTML in plain text on the body of the email,

  How can I change the enconding to 'text/html' so the link will get
displayed correctly?

Thanks,
Gabriel
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] DIAL IAX2 vs. SIP

[Steve Edwards]

On Fri, 11 Sep 2009, Thomas Winter wrote:

> documentation shows me:
>
> Dial(Tech/User:passw...@host/Extension,Timeout,Optionen)
>
> This is working for IAX2.
>
> If Iam using
> DIAL(SIP/u...@secret@sip.domian.tls/123456)
> Asterisk shoes no host with name "sip.domian.tls/123456"

You need better documentation.

http://www.voip-info.org/tiki-index.php?page=Asterisk%20SIP%20Channels

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

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] Voicemail by email with HTML

[Danny Nicholas]

Change the client in voicemail.conf from sendmail to a "MIME-enabled" client
like pup.

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gabriel Ortiz
Lour
Sent: Friday, September 11, 2009 9:36 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Voicemail by email with HTML

 

Hi all,

  I'm trying to send an email with the voicemail details and I want to send
a HTML link on it to make a click2call to the voicemail main, but the email
is send with 'text/plain' encoding and thus it will not show the link, but
the HTML in plain text on the body of the email,

  How can I change the enconding to 'text/html' so the link will get
displayed correctly?

Thanks,
Gabriel

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] Caller ID from POTS lines

[C F]

and how are those POTS lines connected to Asterisk?
In any event doing something like:
Set(CALLERID(num)=${CALLERID(num):0:10}) should do the trick.

On Tue, Sep 8, 2009 at 12:27 PM, Jeremy Taylor  wrote:
>
> Hi,
>
> I'm using asterisk 1.4.22-4 in Trixbox with snom 360 phones. When
> calls come in on our POTS lines, the caller id shows up like
> "555-555-1...@192.168.1.10" where 555-555-1234 is the correct phone
> number and 192.168.1.10 is my pbx server IP. This format does not work
> for redialing on outbound calls.
>
> While there may be an outbound dialing change that could be made, it
> seems like the correct solution would be to change the format of the
> caller id string sent to the phones. I verified from the snom sip
> trace that the caller id is always sent with "@192.168.1.10" on it.
>
> What configuration change can be made in asterisk to correct this and
> only send the phone number as the caller id to the VOIP phone?
>
> Thanks, Jeremy
>
>
>
>
>
>
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> AstriCon 2009 - October 13 - 15 Phoenix, Arizona
> Register Now: http://www.astricon.net
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] All the four lights blinking

[ABBAS SHAKEEL]

Thanks you very much Kevin.I will try it by connecting one end of  Ethernet
cable to one slot and other to second slot . Configuring one
as pri_net and the other as pri_cpe.

I will provide you feed on monday either i succed or not

On Fri, Sep 11, 2009 at 4:23 PM, Kevin P. Fleming wrote:

> ABBAS SHAKEEL wrote:
>
> > But I cant generate calls using the loop back connector and get the
> > following error
> >
> >
> > *CLI> [Sep  9 14:42:55] NOTICE[9981]: channel.c:3749
> > __ast_request_and_dial: Unable to request channel DAHDI/1/123
>
> You cannot use a loopback connector for a PRI interface; PRI signaling
> has a 'network' end and a 'customer' end, and using a loopback connector
> will make the system try to be both at the same time, which is not
> possible.
>
> You can use CAS (FXS/FXO/etc.) signaling on a loopback connector, or you
> can make a crossover cable and connect two ports of your card together,
> configuring one of them as pri_net and the other as pri_cpe.
>
> --
> Kevin P. Fleming
> Digium, Inc. | Director of Software Technologies
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> skype: kpfleming | jabber: kpflem...@digium.com
> Check us out at www.digium.com & www.asterisk.org
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> AstriCon 2009 - October 13 - 15 Phoenix, Arizona
> Register Now: http://www.astricon.net
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 
Best Regards
Shakeel Abbas
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

[asterisk-users] Asterisk 1.6.1.6 Crash when accessing Directory

[Jason Martin]

Hello,

I may have found a serious issue with 1.6.1.6. I just compiled it  
yesterday on our server. When anyone tries to access the name  
directory through the Directory app, the asterisk process completely  
dies.

Our extensions are in a realtime MySQL table, and the directory has  
worked fine with previous versions of asterisk.


Jason Martin
Metrix Matrix, Inc.
785 Elmgrove Rd, Bldg 1
Rochester, NY 14624
Office: 888-865-0065 x202
Mobile: 585-705-1400




___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

[asterisk-users] 183 early media

[Khaled W Chehab]

HI all , 

I am using ,Dial(SIP/Gateway/${EXTEN},m)

how can i  modify asterisk, if it detects two early media to stop OR MUTE
the first RTP early media  AND let the user hear the second early media

any one developed something like that or know from where I can do this  from
chan_sip.c?

 

regards

 



*
No employee or agent is authorized to conclude any binding agreement on behalf 
of Xplorium with another party by e-mail without express written confirmation 
by an officer of Xplorium. Any views expressed by an individual in this 
electronic message do not necessarily reflect views of Xplorium or its 
subsidiaries and associates.

This electronic message and its attachments are solely addressed to the 
addressee(s), and contain confidential information protected from disclosure 
belonging to Xplorium.

If you are not the intended addressee of this electronic message and its 
attachments, kindly delete it immediately from your system and notify the 
sender by electronic mail. You must not copy this message or attachment or 
disclose its content to any other person.

Xplorium does not guarantee the integrity of this electronic message and any of 
its attachments, or that they are free from computer viruses or other defects.
*

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] All the four lights blinking

[Kevin P. Fleming]

ABBAS SHAKEEL wrote:

> But I cant generate calls using the loop back connector and get the
> following error
> 
> 
> *CLI> [Sep  9 14:42:55] NOTICE[9981]: channel.c:3749
> __ast_request_and_dial: Unable to request channel DAHDI/1/123

You cannot use a loopback connector for a PRI interface; PRI signaling
has a 'network' end and a 'customer' end, and using a loopback connector
will make the system try to be both at the same time, which is not possible.

You can use CAS (FXS/FXO/etc.) signaling on a loopback connector, or you
can make a crossover cable and connect two ports of your card together,
configuring one of them as pri_net and the other as pri_cpe.

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

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] ASR & ACD

[adamk]

> Is there any program Asterisk users use to calculate ASR and ACD ??
> 

i calculate them from CDR, using php in a fancy webpage.  But now as i think 
of it, probably an sql query would do just fine.

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] CLI file convert from alaw to g729 with TC400B transcoding card results to an empty file

[Marc Leurent]

Thank you Shaun for your answer!
Indeed, I have made some basic tests to convert a file to g729 using the 
software codec and it works!
Have a nice day!
-- 
-- --
Marc LEURENT
lf...@leurent.eu

Le mercredi, 9 septembre 2009 19.32:30, Shaun Ruffell a écrit :
> On 09/09/2009 09:33 AM, Marc Leurent wrote:
> > Good afternoon,
> > I'm trying to use the CLI command file convert on an Asterisk 1.4.26
> > server with a TC400B transcoding card. The transcoding card is working
> > well for calls but I have some trouble converting sound files from alaw
> > to g729. The command creates empty file as you can see below...
> >
> > CLI>  file convert /var/lib/asterisk/sounds/fr/service_notactivated.alaw
> > /var/lib/asterisk/sounds/fr/service_notactivated.g729
> >
> > (SCREEN):r...@nutella:[/var/../fr]# file service_notactivated*
> > service_notactivated.alaw: RIFF (little-endian) data, WAVE audio, ITU
> > G.711 A-law, mono 8000 Hz service_notactivated.g729: empty
> > service_notactivated.gsm:  data
> >
> > I was able to create the gsm file with the command, but the g729 one is
> > empty. Have you got any idea how I can solve this? Thanks
> > PS: I'm able to place call in g729 without problem and the TC400B works
> > well
>
> The problem is that 'convert' finishes when it receives back a NULL
> frame.  For the software based codecs, this isn't a problem because the
> transcoding happens synchronously with the caller.  However, codec_dahdi
> doesn't block the caller while the hardware is transcoding the audio,
> and therefore can return NULL frames if the hardware hasn't had enough
> time to transcode the frame.
>
> Probably what is needed is a standalone utility in dahdi-tools that can
> be used to transcode audio files with the hardware.
>
> Cheers,


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] Fwd: Patton smartnode 463x (BRI) 25ms tail echo cancellation

[Gaëtan Minet]

Thanks a lot for your help.

Since we decreased tx on the patton gateway (-3db iirc) , no more echo  
report... so far. But this is a bit too early to conclude everything  
is fine now.
Now I guess we would have had similar results with the pci card we  
replaced by the gateway (Digium B410p with misdn, but the card didn't  
seem to react to gain adjustments, even after a complete asterisk and  
misdn module unload/reload... maybe I should have rebooted the whole  
server...)

We tried BRI callback some weeks ago and indeed we never got echo. We  
clearly only have echo when we call/are called from an analog line  
(legacy line or "twin" as they call residential isdn "atas" here). So  
indeed so far it seems you're right, I guess there was no point trying  
to handle far-end echo on our side, but just avoid helping the other  
side to generate that echo with too much tx gain.

We however had another problem with the Digium card when the HW EC was  
on, even in a callback situation ("crispy echo" even when used in  
callback), but no echo at all and no sound quality problem when EC was  
off. Maybe the HW EC chips also had a problem with tx gain being too  
high... We are still waiting for an answer from Digium Support, as at  
least the last 5 cards we bought are showing the same symptoms.
That's why we switched to the patton as no other pci card but  
Sangoma's or Eicon's (currently unsupported in our systems) seemed to  
have HW EC.


Regarding gains, in this callback scenario,  I guess we should try to  
have rx/tx amplitude almost perfectly balanced (while keeping tx  
around -3db as 0db causes echo) ?
In this scenario, don't some cards/drivers use a passthrough  based on  
isdn call id to optimize the channel (I remember seeing some messages  
in the logs about passthrough mode being activated when we made such  
calls) ?


Thanks

Gaetan





On 09/09/2009, at 22:32, Jorge Mendoza wrote:

> Gaëtan,
>
> The echo arise at 4w/2w conversion point, normally at the far end. Try
> to call another phone number assigned to your BRI, i.e. call back to
> your Asterisk server. If you use a Polycom phone to initiate the call
> and another to receive the call, you have the perfect link: 4 wires  
> with
> not 4w/2w conversion. Under this test, theoretically you must not have
> echo. If so, it is necessary to look elsewhere, at your Asterisk box  
> maybe.
> Modifying the rx/tx gain is a good practice too.
>
> Best Regards
>
> Jorge Mendoza
>
> Gaëtan Minet wrote:
>> Thanks !
>>
>> We installed it in the interim and have a lot of calls with far-end
>> echo :(.
>> But it seems the solution could be to reduce the TX gain on our side
>> (these are using polycom phones, and  indeed  I can see a big
>> amplitude  imbalance between tx/rx on a recording). It's under  
>> test, I
>> hope it'll solve it.
>>
>> Kind regards,
>>
>> Gaetan
>>
>>
>> On 09/09/2009, at 16:22, Jorge Mendoza wrote:
>>
>>
>>> Gaëtan,
>>>
>>> They are using as gateway to the pstn. In fact, they are remote
>>> gateways
>>> for a centralized callcenter:
>>>
>>> [pstn] «--» [BRI] «--» [internet] «--» [callcenter]
>>>
>>> Regards
>>>
>>> Jorge Mendoza
>>>
>>> Gaëtan Minet wrote:
>>>
 Thanks

 Are you using these to connect isdn phones to the voip or to as a
 gateway to the pstn for a voip system ?

 Kind regards

 Gaetan


 On 08/09/2009, at 19:40, Jorge Mendoza wrote:



> We have some installations with smartnode 4554, (same tail echo
> cancellation) without problems so far.
>
> Jorge Mendoza
>
> Gaëtan Minet wrote:
>
>
>> Hi
>>
>> Is anybody using these ?
>>
>> Gaetan
>>
>>
>> Begin forwarded message:
>>
>>
>>
>>> *From: *Gaëtan Minet mailto:gminet-
>>> m...@mcit.be>>
>>> *Date: *Sat 22 Aug 2009 16:29:42 GMT+02:00
>>> *To: *Asterisk Users Mailing List - Non-Commercial Discussion
>>> >> >
>>> *Subject: **[asterisk-users] Patton smartnode 463x (BRI) 25ms  
>>> tail
>>> echo cancellation*
>>> *Reply-To: *Asterisk Users Mailing List - Non-Commercial
>>> Discussion
>>> >> >
>>>
>>> Hi all
>>>
>>> We use pci/pci-e BRI cards in our installations. Due to echo
>>> problems
>>> (that was before Oslec and others), we quickly switched to cards
>>> with
>>> hardware-based EC.
>>> So we use exclusively Digium B410p cards that provide 64ms tail
>>> EC.
>>>
>>> For several reasons we'd like to switch to external BRI gateways
>>> like
>>> the Patton smartnodes (the price is getting really close to a
>>> B410p).
>>>
>>> I'm however curious about their HW EC. I see in the datasheets
>>> that it
>>> only has 25ms tail per channel (pri are 128ms, but not BRI).
>>> Are some of you using these gateway and do your experience  
>>> (m

Re: [asterisk-users] Looking for a way to show caller id information on the desktop

[Lenz Emilitri]

As an ultra cheap way of doing it, you could simply output the caller-id to
a log file and display a "tail 20" of it on a web page.
Something like this:

exten => s,1,System( echo"${EPOCH}|${CALLERID(num)}"
>>/var/log/asterisk/incoming )

It should be trivial to display the last n lines of it on a web page that
reloads automagically.
l.



2009/9/10 Jonathan Moore 

> Hi there.
>
> My problem, I can't figure out how to ask this question.  So,
> hopefully someone out here can point me to the FM on this.
>
> I would like to have either a web page or an application that I can
> view that whenever a call arrives on the Asterisk server
> the application will display the callerid information.  I've found
> quite a few examples of the reverse of this.  To where a
> script is called to GET the callerid information, but that's not what
> I'm looking for.
>
> Is it possible, and if so, where should I start looking to find a
> solution to this?  I've failed at google so far, and I think I'm just
> not asking the right question.
>
> Thanks for any help or pointers.
>
> -jonathan
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> AstriCon 2009 - October 13 - 15 Phoenix, Arizona
> Register Now: http://www.astricon.net
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 
Loway - home of QueueMetrics - http://queuemetrics.com
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

[asterisk-users] asterisk addons don't compile using non standart prefix

[Fred]

My system is Debian
I have successfully installed Asterisk 1.6.1.6 to a non standart
folder /opt/asterisk

Then for asterisk-addons-1.6.1.1 I run:
./configure --prefix=/opt/asterisk/
make menuselect

And then command "make" fails with following errors:

Generating embedded module rules ...
make[1]: Entering directory `/usr/src/asterisk-addons-1.6.1.1'
make[1]: Leaving directory `/usr/src/asterisk-addons-1.6.1.1'
make[1]: Entering directory `/usr/src/asterisk-addons-1.6.1.1'
make[1]: Leaving directory `/usr/src/asterisk-addons-1.6.1.1'
make[1]: Entering directory `/usr/src/asterisk-addons-1.6.1.1'
make[1]: Leaving directory `/usr/src/asterisk-addons-1.6.1.1'
make[1]: Entering directory `/usr/src/asterisk-addons-1.6.1.1/channels'
make[1]: Nothing to be done for `all'.
make[1]: Leaving directory `/usr/src/asterisk-addons-1.6.1.1/channels'
make[1]: Entering directory `/usr/src/asterisk-addons-1.6.1.1/apps'
   [CC] app_addon_sql_mysql.c -> app_addon_sql_mysql.o
app_addon_sql_mysql.c:19:22: error: asterisk.h: No such file or directory
app_addon_sql_mysql.c:31:27: error: asterisk/file.h: No such file or directory
app_addon_sql_mysql.c:32:29: error: asterisk/logger.h: No such file or directory
app_addon_sql_mysql.c:33:30: error: asterisk/channel.h: No such file
or directory
app_addon_sql_mysql.c:34:26: error: asterisk/pbx.h: No such file or directory
app_addon_sql_mysql.c:35:29: error: asterisk/module.h: No such file or directory
app_addon_sql_mysql.c:36:34: error: asterisk/linkedlists.h: No such
file or directory
app_addon_sql_mysql.c:37:31: error: asterisk/chanvars.h: No such file
or directory
app_addon_sql_mysql.c:38:27: error: asterisk/lock.h: No such file or directory
app_addon_sql_mysql.c:39:30: error: asterisk/options.h: No such file
or directory
app_addon_sql_mysql.c:40:26: error: asterisk/app.h: No such file or directory
app_addon_sql_mysql.c:86: warning: data definition has no type or storage class
app_addon_sql_mysql.c:86: warning: parameter names (without types) in
function declaration
app_addon_sql_mysql.c:96: warning: ‘struct ast_channel’ declared
inside parameter list
app_addon_sql_mysql.c:96: warning: its scope is only this definition
or declaration, which is probably not what you want
(stripped of)

It would be great to recieve any help, if anyone else experienced this
problem or just know the cause and solution of this error.

Thanks in advance

Best regards,
Fedor

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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