Re: [asterisk-users] DTMF digits falsely detected

[Vladimir Mikhelson]

On 9/14/2012 11:04 PM, Matthew Jordan wrote:
> - Original Message -
>> From: "Vladimir Mikhelson" 
>> To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
>> 
>> Sent: Friday, September 14, 2012 10:39:30 PM
>> Subject: Re: [asterisk-users] DTMF digits falsely detected
>>
>>
>> On 9/14/2012 10:11 PM, Matthew Jordan wrote:
>>> - Original Message -
 From: "Vladimir Mikhelson" 
 To: "Asterisk Users Mailing List - Non-Commercial Discussion"
 
 Sent: Friday, September 14, 2012 9:24:41 PM
 Subject: Re: [asterisk-users] DTMF digits falsely detected


 On 9/14/2012 6:04 PM, Alec Davis wrote:
>> -Original Message-
>> From: asterisk-users-boun...@lists.digium.com
>> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
>> Vieri
>> Sent: Saturday, 15 September 2012 8:45 a.m.
>> To: asterisk-users@lists.digium.com
>> Subject: [asterisk-users] DTMF digits falsely detected
>>
 Can it be related to
 https://issues.asterisk.org/jira/browse/ASTERISK-19610 ??

 -Vladimir
>>> Most likely not.  If the SIP peer is using rfc2833 DTMF, its most
>>> likely
>>> related to r370252.
>>>
>>> Please file an issue on the issue tracker,
>>> https://issues.asterisk.org/jira.
>>> Please include a pcap of the RTP stream and a DEBUG log with RTP
>>> debug
>>> enabled, using 'rtp set debug on'.
>>>
>>> Thanks,
>>>
>>> --
>>> Matthew Jordan
>>>
>> Matt,
>>
>> I have created the issue.  See
>> https://issues.asterisk.org/jira/browse/ASTERISK-20424?focusedCommentId=197108#comment-197108
>>
>> Sorry, I will be unable to produce pcap and rtp debug as I have fixed
>> the issue by uninstalling the Soft Phone I used for multiple years
>> with
>> no issues.
>>
>> -Vladimir
> Well, it'd be appreciated if someone who is experiencing this would be
> willing to reproduce it and attach a pcap and DEBUG log to the issue.
> The bug fixed by that commit dealt with out of order DTMF; I suspect
> that the problem is your soft phone is sending re-transmits of the end
> event of the DTMF digit with an increasing timestamp.  The previous
> behavior in Asterisk would most likely have been more tolerant of
> this non-compliant scenario, but didn't handle the out of order packets
> as well.
>
> Unfortunately, without evidence confirming that, there isn't much I can
> do.
>
> --
> Matthew Jordan
>
Hopefully the initial poster still has the configuration to produce the
files for you.

Are you saying the DTMF logs I attached do not provide enough evidence
to support the theory of the DTMF length being the cause of this issue?

-Vladimir


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Re: [asterisk-users] Asterisk on VM with NO DAHDI hardware

[Shaun Ruffell]

On Fri, Sep 14, 2012 at 09:42:23PM -0400, Mark Robinson wrote:
> I did some research on this subject and still do not understand.
> Why we use modules if asterisk can obtain timing directly from
> kernel?

Probably longer than it needs to be but...

Asterisk needs timing information via a kernel file descriptor
that can be passed into to the poll() [1] or select() system calls.
This allows Asterisk to sleep while waiting for either a network
packet to come in or one of several time intervals to expire in the
same system call which is very efficient.

DAHDI historically provided this service via /dev/dahdi/timer.
DAHDI was a good choice to provide a timing service since Asterisk
typically wanted VOIP traffic synchronized with any telephony
hardware installed. If there was not telephony hardware DAHDI could
use internal kernel interfaces to find the best source of timing for
Asterisk (generic kernel timers, Real-Time Clock, High Resolution
timers, etc..).

Using DAHDI was (and is) a natural choice but updating kernels can
be more difficult since DAHDI must be recompiled every time the
kernel is updated. This unnecessarily increases the administration
burden when there is not any telephony hardware to synchronize with.

To reduce the dependency on DAHDI, res_timing_pthread was created in
Asterisk version 1.6.1 [2]. This implementation could provide timing
via file descriptors without requiring DAHDI to be installed. It
uses a pipe()[3] and a thread. The thread will sleep for a period of
time and then write to one end of the pipe when the timer fires.
res_timing_pthread is more system intensive--creates a thread that
must be scheduled, the scheduling of that thread determines the
timeliness of the timer firing, and the extra system calls required
to accomplish the same task--and I believe it is advisable to avoid
res_timing_pthread unless you have no other choice.

It was not until kernel version 2.6.25 that Linux provided a
standard interface for configuring timers that signaled via file
descriptors, the timerfd interface [4]. Now Asterisk had a standard
kernel interface which provided essentially the same service that
DAHDI's /dev/dahdi/timer did without needing to install DAHDI or
creating a separate thread which wrote to a pipe and
res_timing_timerfd was first release in Asterisk 1.6.2 [5]. The
problem is that timerfd_create() is not available on all platforms
Asterisk must support. Also, if you have telephony hardware
installed it is still generally best to synchronize to the clock on
the telephony hardware to minimize audio problems caused by
mismatches in clock rates. But if available and you don't have any
other dependency on DAHDI (app_meetme, app_page, etc...), timerfd
should be your first choice for timing source.

So given these different methods of obtaining timing, Asterisk
needed a way to abstract them so that other parts of the system had
a standard way to get a file descriptor which could be configured to
fire at certain intervals. That is why the timing interface [6] and
various timing modules were created. It allows the Asterisk
administrator to use the timing source that works best for them.

That's why, to the best of my knowledge, Asterisk uses modules even
though it can now obtain timing directly from the kernel.

Cheers,
Shaun

[1] http://linux.die.net/man/2/poll
[2] http://svnview.digium.com/svn/asterisk?view=revision&revision=122928
[3] http://linux.die.net/man/2/pipe
[4] http://linux.die.net/man/2/timerfd_create
[5] http://svnview.digium.com/svn/asterisk?view=revision&revision=157820
[6] http://svnview.digium.com/svn/asterisk?view=revision&revision=122062 

-- 
Shaun Ruffell
Digium, Inc. | Linux Kernel Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: www.digium.com & www.asterisk.org

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Re: [asterisk-users] DTMF digits falsely detected

[Matthew Jordan]

- Original Message -
> From: "Vladimir Mikhelson" 
> To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
> 
> Sent: Friday, September 14, 2012 10:39:30 PM
> Subject: Re: [asterisk-users] DTMF digits falsely detected
> 
> 
> On 9/14/2012 10:11 PM, Matthew Jordan wrote:
> >
> > - Original Message -
> >> From: "Vladimir Mikhelson" 
> >> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> >> 
> >> Sent: Friday, September 14, 2012 9:24:41 PM
> >> Subject: Re: [asterisk-users] DTMF digits falsely detected
> >>
> >>
> >> On 9/14/2012 6:04 PM, Alec Davis wrote:
>  -Original Message-
>  From: asterisk-users-boun...@lists.digium.com
>  [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
>  Vieri
>  Sent: Saturday, 15 September 2012 8:45 a.m.
>  To: asterisk-users@lists.digium.com
>  Subject: [asterisk-users] DTMF digits falsely detected
> 
> >> Can it be related to
> >> https://issues.asterisk.org/jira/browse/ASTERISK-19610 ??
> >>
> >> -Vladimir
> > Most likely not.  If the SIP peer is using rfc2833 DTMF, its most
> > likely
> > related to r370252.
> >
> > Please file an issue on the issue tracker,
> > https://issues.asterisk.org/jira.
> > Please include a pcap of the RTP stream and a DEBUG log with RTP
> > debug
> > enabled, using 'rtp set debug on'.
> >
> > Thanks,
> >
> > --
> > Matthew Jordan
> >
> 
> Matt,
> 
> I have created the issue.  See
> https://issues.asterisk.org/jira/browse/ASTERISK-20424?focusedCommentId=197108#comment-197108
> 
> Sorry, I will be unable to produce pcap and rtp debug as I have fixed
> the issue by uninstalling the Soft Phone I used for multiple years
> with
> no issues.
> 
> -Vladimir

Well, it'd be appreciated if someone who is experiencing this would be
willing to reproduce it and attach a pcap and DEBUG log to the issue.
The bug fixed by that commit dealt with out of order DTMF; I suspect
that the problem is your soft phone is sending re-transmits of the end
event of the DTMF digit with an increasing timestamp.  The previous
behavior in Asterisk would most likely have been more tolerant of
this non-compliant scenario, but didn't handle the out of order packets
as well.

Unfortunately, without evidence confirming that, there isn't much I can
do.

--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org

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Re: [asterisk-users] DTMF digits falsely detected

[Vladimir Mikhelson]

On 9/14/2012 10:11 PM, Matthew Jordan wrote:
>
> - Original Message -
>> From: "Vladimir Mikhelson" 
>> To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
>> 
>> Sent: Friday, September 14, 2012 9:24:41 PM
>> Subject: Re: [asterisk-users] DTMF digits falsely detected
>>
>>
>> On 9/14/2012 6:04 PM, Alec Davis wrote:
 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
 Vieri
 Sent: Saturday, 15 September 2012 8:45 a.m.
 To: asterisk-users@lists.digium.com
 Subject: [asterisk-users] DTMF digits falsely detected

>> Can it be related to
>> https://issues.asterisk.org/jira/browse/ASTERISK-19610 ??
>>
>> -Vladimir
> Most likely not.  If the SIP peer is using rfc2833 DTMF, its most likely
> related to r370252.
>
> Please file an issue on the issue tracker, https://issues.asterisk.org/jira.
> Please include a pcap of the RTP stream and a DEBUG log with RTP debug
> enabled, using 'rtp set debug on'.
>
> Thanks,
>
> --
> Matthew Jordan
>

Matt,

I have created the issue.  See
https://issues.asterisk.org/jira/browse/ASTERISK-20424?focusedCommentId=197108#comment-197108

Sorry, I will be unable to produce pcap and rtp debug as I have fixed
the issue by uninstalling the Soft Phone I used for multiple years with
no issues.

-Vladimir


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Re: [asterisk-users] DTMF digits falsely detected

[Matthew Jordan]

- Original Message -
> From: "Vladimir Mikhelson" 
> To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
> 
> Sent: Friday, September 14, 2012 9:24:41 PM
> Subject: Re: [asterisk-users] DTMF digits falsely detected
> 
> 
> On 9/14/2012 6:04 PM, Alec Davis wrote:
> >> -Original Message-
> >> From: asterisk-users-boun...@lists.digium.com
> >> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
> >> Vieri
> >> Sent: Saturday, 15 September 2012 8:45 a.m.
> >> To: asterisk-users@lists.digium.com
> >> Subject: [asterisk-users] DTMF digits falsely detected
> >>
> 
> Can it be related to
> https://issues.asterisk.org/jira/browse/ASTERISK-19610 ??
> 
> -Vladimir

Most likely not.  If the SIP peer is using rfc2833 DTMF, its most likely
related to r370252.

Please file an issue on the issue tracker, https://issues.asterisk.org/jira.
Please include a pcap of the RTP stream and a DEBUG log with RTP debug
enabled, using 'rtp set debug on'.

Thanks,

--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org

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Re: [asterisk-users] DTMF digits falsely detected

[Vladimir Mikhelson]

On 9/14/2012 6:04 PM, Alec Davis wrote:
>> -Original Message-
>> From: asterisk-users-boun...@lists.digium.com 
>> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vieri
>> Sent: Saturday, 15 September 2012 8:45 a.m.
>> To: asterisk-users@lists.digium.com
>> Subject: [asterisk-users] DTMF digits falsely detected
>>
>> Hi,
>>
>> I have a context that basically does:
>>
>> Wait(1)
>> Background(message)
>> WaitExten(10)
>>
>> _6XX,1,DoSomething
>>
>> The problem is that when I reach this context and press some 
>> digits (eg. 6566604) then I can see in the log that Asterisk 
>> reads 6655666.
>> So it's actually reading the digits twice.
>> How can I avoid this?
>> Incoming channel type is ISDN (mISDN).
>>
> Are you saying every digit twice, or some digits twice.
> Where is the call originating from, GSM cell phone or landline?
>
> Which version of asterisk are you using?
>
> Alec Davis
>
>
Alec,

Can it be related to
https://issues.asterisk.org/jira/browse/ASTERISK-19610 ??

-Vladimir
 

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Re: [asterisk-users] DTMF digits falsely detected

[Vladimir Mikhelson]

On 9/14/2012 6:04 PM, Alec Davis wrote:
>> -Original Message-
>> From: asterisk-users-boun...@lists.digium.com 
>> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vieri
>> Sent: Saturday, 15 September 2012 8:45 a.m.
>> To: asterisk-users@lists.digium.com
>> Subject: [asterisk-users] DTMF digits falsely detected
>>
>> Hi,
>>
>> I have a context that basically does:
>>
>> Wait(1)
>> Background(message)
>> WaitExten(10)
>>
>> _6XX,1,DoSomething
>>
>> The problem is that when I reach this context and press some 
>> digits (eg. 6566604) then I can see in the log that Asterisk 
>> reads 6655666.
>> So it's actually reading the digits twice.
>> How can I avoid this?
>> Incoming channel type is ISDN (mISDN).
>>
> Are you saying every digit twice, or some digits twice.
> Where is the call originating from, GSM cell phone or landline?
>
> Which version of asterisk are you using?
>
> Alec Davis
>
>
>
Hi,

Started seeing similar abnormality today after the 1.8.16.0 upgrade.

In my case digits were repeated 3 times.

Several observations:

  * The problem only manifested itself on SIP channel.  OOH323 and DAHDI
did not exhibit this problem.
  * I was able to dial the extension with no issues, the problem started
in the Voice Mail application.

For example, in the case below I entered "430#"

[2012-09-14 11:50:06] VERBOSE[32019] app_read.c: -- User entered
'444333000'

Following is the excerpt from the DTMF log:

[2012-09-14 11:50:05] DTMF[32019] channel.c: DTMF begin '4' received on
SIP/462-
[2012-09-14 11:50:05] DTMF[32019] channel.c: DTMF begin ignored '4' on
SIP/462-
[2012-09-14 11:50:05] DTMF[32019] channel.c: DTMF end '4' received on
SIP/462-, duration 300 ms
[2012-09-14 11:50:05] DTMF[32019] channel.c: DTMF end passthrough '4' on
SIP/462-
[2012-09-14 11:50:05] DTMF[32019] channel.c: DTMF end '4' received on
SIP/462-, duration 300 ms
[2012-09-14 11:50:05] DTMF[32019] channel.c: DTMF end passthrough '4' on
SIP/462-
[2012-09-14 11:50:05] DTMF[32019] channel.c: DTMF end '4' received on
SIP/462-, duration 300 ms
[2012-09-14 11:50:05] DTMF[32019] channel.c: DTMF end passthrough '4' on
SIP/462-
[2012-09-14 11:50:05] DTMF[32019] channel.c: DTMF begin '3' received on
SIP/462-
[2012-09-14 11:50:05] DTMF[32019] channel.c: DTMF begin ignored '3' on
SIP/462-
[2012-09-14 11:50:06] DTMF[32019] channel.c: DTMF end '3' received on
SIP/462-, duration 300 ms
[2012-09-14 11:50:06] DTMF[32019] channel.c: DTMF end passthrough '3' on
SIP/462-
[2012-09-14 11:50:06] DTMF[32019] channel.c: DTMF end '3' received on
SIP/462-, duration 300 ms
[2012-09-14 11:50:06] DTMF[32019] channel.c: DTMF end passthrough '3' on
SIP/462-
[2012-09-14 11:50:06] DTMF[32019] channel.c: DTMF end '3' received on
SIP/462-, duration 300 ms
[2012-09-14 11:50:06] DTMF[32019] channel.c: DTMF end passthrough '3' on
SIP/462-
[2012-09-14 11:50:06] DTMF[32019] channel.c: DTMF begin '0' received on
SIP/462-
[2012-09-14 11:50:06] DTMF[32019] channel.c: DTMF begin ignored '0' on
SIP/462-
[2012-09-14 11:50:06] DTMF[32019] channel.c: DTMF end '0' received on
SIP/462-, duration 300 ms
[2012-09-14 11:50:06] DTMF[32019] channel.c: DTMF end passthrough '0' on
SIP/462-
[2012-09-14 11:50:06] DTMF[32019] channel.c: DTMF end '0' received on
SIP/462-, duration 300 ms
[2012-09-14 11:50:06] DTMF[32019] channel.c: DTMF end passthrough '0' on
SIP/462-
[2012-09-14 11:50:06] DTMF[32019] channel.c: DTMF end '0' received on
SIP/462-, duration 300 ms
[2012-09-14 11:50:06] DTMF[32019] channel.c: DTMF end passthrough '0' on
SIP/462-
[2012-09-14 11:50:06] DTMF[32019] channel.c: DTMF begin '#' received on
SIP/462-
[2012-09-14 11:50:06] DTMF[32019] channel.c: DTMF begin ignored '#' on
SIP/462-
[2012-09-14 11:50:06] DTMF[32019] channel.c: DTMF end '#' received on
SIP/462-, duration 300 ms
[2012-09-14 11:50:06] DTMF[32019] channel.c: DTMF end passthrough '#' on
SIP/462-
[2012-09-14 11:50:06] DTMF[32019] channel.c: DTMF end '#' received on
SIP/462-, duration 300 ms
[2012-09-14 11:50:06] DTMF[32019] channel.c: DTMF end '#' received on
SIP/462-, duration 300 ms
[2012-09-14 11:50:06] DTMF[32019] channel.c: DTMF begin emulation of '#'
with duration 300 queued on SIP/462-
[2012-09-14 11:50:06] DTMF[32019] channel.c: DTMF end '#' received on
SIP/462-, duration 300 ms
[2012-09-14 11:50:07] DTMF[32019] channel.c: DTMF end emulation of '#'
queued on SIP/462-
[2012-09-14 11:50:07] DTMF[32019] channel.c: DTMF end '#' received on
SIP/462-, duration 300 ms
[2012-09-14 11:50:07] DTMF[32019] channel.c: DTMF begin emulation of '#'
with duration 300 queued on SIP/462-

As a comparison here is an excerpt from the DTMF log of a similar call
from the same extension before the upgrade (Asterisk 1.8.15.1):

[2

Re: [asterisk-users] Asterisk on VM with NO DAHDI hardware

[Mark Robinson]

I did some research on this subject and still do not understand. Why we use
modules if asterisk can obtain timing directly from kernel?
On Sep 13, 2012 11:25 PM, "Mark Robinson"  wrote:

> Thanks Shaun. Very usefully head-up.
>
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Re: [asterisk-users] DTMF digits falsely detected

[Alec Davis]

> -Original Message-
> From: asterisk-users-boun...@lists.digium.com 
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vieri
> Sent: Saturday, 15 September 2012 8:45 a.m.
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] DTMF digits falsely detected
> 
> Hi,
> 
> I have a context that basically does:
> 
> Wait(1)
> Background(message)
> WaitExten(10)
> 
> _6XX,1,DoSomething
> 
> The problem is that when I reach this context and press some 
> digits (eg. 6566604) then I can see in the log that Asterisk 
> reads 6655666.
> So it's actually reading the digits twice.
> How can I avoid this?
> Incoming channel type is ISDN (mISDN).
> 

Are you saying every digit twice, or some digits twice.
Where is the call originating from, GSM cell phone or landline?

Which version of asterisk are you using?

Alec Davis


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[asterisk-users] DTMF digits falsely detected

[Vieri]

Hi,

I have a context that basically does:

Wait(1)
Background(message)
WaitExten(10)

_6XX,1,DoSomething

The problem is that when I reach this context and press some digits (eg. 
6566604) then I can see in the log that Asterisk reads 6655666.
So it's actually reading the digits twice.
How can I avoid this?
Incoming channel type is ISDN (mISDN).

Thanks,

Vieri


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[asterisk-users] opus codec

[Vieri]

Hi,

Will Asterisk support the OPUS codec?

http://opus-codec.org/

Thanks,

Vieri


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Re: [asterisk-users] Digium AEX410, MTNL Mumbai Caller-ID problems

[Raj Mathur (राज माथुर)]

On Friday 14 Sep 2012, Richard Mudgett wrote:
> > Continuing with the saga of Digium vs MTNL Mumbai, looking for
> > suggestions on handling incoming Caller-ID issues.  The card
> > manages to
> > grab a couple of (random) digits of the incoming CID, but they're
> > more
> > or less useless.  Is there any way to fix this?
> > 
> > Asterisk 1.8.13, Dahdi 2.5.0.1 on Debian Testing (Wheezy), MTNL
> > Mumbai.
> > Digium, Inc. Wildcard AEX410 4-port analog card (PCI-Express)
> > 
> > chan_dahdi.conf contains:
> > usecallerid = yes
> > cidsignalling=dtmf
> > cidstart=polarity_in
> > 
> > Signalling is fxsks.
> > 
> > Log (calling number is 9811066XXX):

> You appear to be suffering form
> https://issues.asterisk.org/jira/browse/ASTERISK-19610
> 
> It is fixed in the just released v1.8.17.0-rc1.

Outstanding!  Just tried with 1.8.17.0-rc1 and the old Dahdi (2.5.0.1) 
and it delivered at least two CIDs just fine.  Many thanks.

Now to sort out the Asterisk packaged for Debian issue.  Worst case, 
will package it myself :(

Regards,

-- Raj
-- 
Raj Mathur  || r...@kandalaya.org   || GPG:
http://otheronepercent.blogspot.com || http://kandalaya.org || CC68
It is the mind that moves   || http://schizoid.in   || D17F

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Re: [asterisk-users] MySQL Query : Calls Answered for < 5 sec

[Phil Frost]

On 09/14/2012 12:45 PM, RSCL Mumbai wrote:

I need a list of calls Answered and Disconnected in less than 5 sec.


http://dev.mysql.com/doc/refman/5.6/en/select.html
http://www.google.com/search?q=sql+tutorial
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Re: [asterisk-users] MySQL Query : Calls Answered for < 5 sec

[RSCL Mumbai]

I need a list of calls Answered and Disconnected in less than 5 sec.

Thx



On Fri, Sep 14, 2012 at 10:07 PM, Warren Selby wrote:

> On Fri, Sep 14, 2012 at 11:33 AM, RSCL Mumbai wrote:
>
>> @Raj
>>
>> I tried your query and variation by using replacing duration with billsec.
>> In both cases, I get results including disposition "NO ANSWER"
>>
>
>
> If you don't want the "NO ANSWER" disposition, add an AND NOT DISPOSITION
> = 'NO ANSWER' to your query.  This is all pretty basic SQL Query writing,
> not specific to asterisk...
>
> --
> Thanks,
> --Warren Selby, dCAP
> http://www.SelbyTech.com 
>
>
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Re: [asterisk-users] MySQL Query : Calls Answered for < 5 sec

[Warren Selby]

On Fri, Sep 14, 2012 at 11:33 AM, RSCL Mumbai  wrote:

> @Raj
>
> I tried your query and variation by using replacing duration with billsec.
> In both cases, I get results including disposition "NO ANSWER"
>


If you don't want the "NO ANSWER" disposition, add an AND NOT DISPOSITION =
'NO ANSWER' to your query.  This is all pretty basic SQL Query writing, not
specific to asterisk...

-- 
Thanks,
--Warren Selby, dCAP
http://www.SelbyTech.com 
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Re: [asterisk-users] MySQL Query : Calls Answered for < 5 sec

[Danny Nicholas]

Change to this:

select * from cdr
where calldate >= '2012-09-01' and calldate < '2012-09-08'
and duration < 5 and disposition<>’NO ANSWER’;



 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of RSCL Mumbai
Sent: Friday, September 14, 2012 11:34 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] MySQL Query : Calls Answered for < 5 sec

 

@Raj

I tried your query and variation by using replacing duration with billsec.
In both cases, I get results including disposition "NO ANSWER"





On Fri, Sep 14, 2012 at 9:58 PM, Raj Mathur (राज माथुर)  
wrote:

On Friday 14 Sep 2012, RSCL Mumbai wrote:
> I am trying to construct MySQL query(s) to get a list of calls which
> lasted for less than 5 seconds between a given date range.
> Any help is appreciated.

On the CDR database, to get all calls that lasted < 5 seconds between
2012-09-01 and 2012-09-07 (inclusive), the MySQL query would be:

select * from cdr
where calldate >= '2012-09-01' and calldate < '2012-09-08'
and duration < 5;

Regards,

-- Raj
--
Raj Mathur  || r...@kandalaya.org   || GPG:
http://otheronepercent.blogspot.com || http://kandalaya.org || CC68
It is the mind that moves   || http://schizoid.in   || D17F


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Re: [asterisk-users] MySQL Query : Calls Answered for < 5 sec

[RSCL Mumbai]

@Raj

I tried your query and variation by using replacing duration with billsec.
In both cases, I get results including disposition "NO ANSWER"




On Fri, Sep 14, 2012 at 9:58 PM, Raj Mathur (राज माथुर) <
r...@linux-delhi.org> wrote:

> On Friday 14 Sep 2012, RSCL Mumbai wrote:
> > I am trying to construct MySQL query(s) to get a list of calls which
> > lasted for less than 5 seconds between a given date range.
> > Any help is appreciated.
>
> On the CDR database, to get all calls that lasted < 5 seconds between
> 2012-09-01 and 2012-09-07 (inclusive), the MySQL query would be:
>
> select * from cdr
> where calldate >= '2012-09-01' and calldate < '2012-09-08'
> and duration < 5;
>
> Regards,
>
> -- Raj
> --
> Raj Mathur  || r...@kandalaya.org   || GPG:
> http://otheronepercent.blogspot.com || http://kandalaya.org || CC68
> It is the mind that moves   || http://schizoid.in   || D17F
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>http://www.asterisk.org/hello
>
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> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
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Re: [asterisk-users] MySQL Query : Calls Answered for < 5 sec

[RSCL Mumbai]

The following query gives me calls with disposition "NO ANSWER"



On Fri, Sep 14, 2012 at 9:50 PM, Danny Nicholas  wrote:

> Select * from cdr where duration < 5 and (calldate=> date1 and calldate <=
> date2)
>
> ** **
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *RSCL Mumbai
> *Sent:* Friday, September 14, 2012 11:16 AM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* [asterisk-users] MySQL Query : Calls Answered for < 5 sec
>
> ** **
>
> Hello,
>
> I am trying to construct MySQL query(s) to get a list of calls which
> lasted for less than 5 seconds between a given date range.
> Any help is appreciated.
>
>
> Thank you in advance.
>
> Regards,
> Sans
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>
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>
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Re: [asterisk-users] MySQL Query : Calls Answered for < 5 sec

[Raj Mathur (राज माथुर)]

On Friday 14 Sep 2012, RSCL Mumbai wrote:
> I am trying to construct MySQL query(s) to get a list of calls which
> lasted for less than 5 seconds between a given date range.
> Any help is appreciated.

On the CDR database, to get all calls that lasted < 5 seconds between 
2012-09-01 and 2012-09-07 (inclusive), the MySQL query would be:

select * from cdr
where calldate >= '2012-09-01' and calldate < '2012-09-08'
and duration < 5;

Regards,

-- Raj
-- 
Raj Mathur  || r...@kandalaya.org   || GPG:
http://otheronepercent.blogspot.com || http://kandalaya.org || CC68
It is the mind that moves   || http://schizoid.in   || D17F

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Re: [asterisk-users] MySQL Query : Calls Answered for < 5 sec

[Mitul Limbani]

question worth asking in mysql user list then here !!

Regards,
Mitul Limbani,
Chief Architech & Founder,
Enterux Solutions Pvt. Ltd.
110 Reena Complex, Opp. Nathani Steel,
Vidyavihar (W), Mumbai - 400 086. India
http://www.enterux.com/
http://www.entvoice.com/
email: mi...@enterux.in
DID: +91-22-71967121
Cell: +91-9820332422




On Fri, Sep 14, 2012 at 9:46 PM, RSCL Mumbai  wrote:

> Hello,
>
> I am trying to construct MySQL query(s) to get a list of calls which
> lasted for less than 5 seconds between a given date range.
> Any help is appreciated.
>
>
> Thank you in advance.
>
> Regards,
> Sans
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>http://www.asterisk.org/hello
>
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> To UNSUBSCRIBE or update options visit:
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>
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Re: [asterisk-users] MySQL Query : Calls Answered for < 5 sec

[Danny Nicholas]

Select * from cdr where duration < 5 and (calldate=> date1 and calldate <= 
date2)

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of RSCL Mumbai
Sent: Friday, September 14, 2012 11:16 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] MySQL Query : Calls Answered for < 5 sec

 

Hello,

I am trying to construct MySQL query(s) to get a list of calls which lasted for 
less than 5 seconds between a given date range.
Any help is appreciated.


Thank you in advance.

Regards,
Sans

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Re: [asterisk-users] kernel: dahdi: Master changed to TE2/0/2 --- Is a normal message

[Shaun Ruffell]

[friendly request that you inline or bottom post and trim your replies.
See http://brooksreview.net/2011/01/interleaved-email/ which makes
most of the points I would make on the subject]

On Fri, Sep 14, 2012 at 09:00:39AM -0300, equis software wrote:
> My test were...
> 
> SIEMENS -> LTG1 <--- cable1 - SS7 ---> IVR1 (have errors)
>|-> LTG2 <--- cable2 - SS7 ---> IVR2 (OK)
> 
> minutes later...
> 
> SIEMENS -> LTG1 <--- cable1 - SS7 ---> IVR2 (OK)
>|-> LTG2 <--- cable2 - SS7 ---> IVR1 (have same errors)

And your problem follows the server not the card or the cable?  Have
you tried other slots? Are you screwing the cards down?

-- 
Shaun Ruffell
Digium, Inc. | Linux Kernel Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: www.digium.com & www.asterisk.org

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[asterisk-users] MySQL Query : Calls Answered for < 5 sec

[RSCL Mumbai]

Hello,

I am trying to construct MySQL query(s) to get a list of calls which lasted
for less than 5 seconds between a given date range.
Any help is appreciated.


Thank you in advance.

Regards,
Sans
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Re: [asterisk-users] kernel: dahdi: Master changed to TE2/0/2 --- Is a normal message

[Shaun Ruffell]

On Fri, Sep 14, 2012 at 11:27:14AM -0300, equis software wrote:
> Shaun, could it be related to
> http://lists.digium.com/pipermail/asterisk-users/2012-March/271304.html ??
> I'm using DAHDI Version: 2.4.1.2

No, those committs were to address an issue that only existed
between 2.6.0 and 2.6.1. It's not present in your version of the
firmware. However, I would recommend you update to 2.6.1 when you
get the chance.

-- 
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Digium, Inc. | Linux Kernel Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: www.digium.com & www.asterisk.org

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Re: [asterisk-users] Digium AEX410, MTNL Mumbai Caller-ID problems

[Raj Mathur (राज माथुर)]

On Friday 14 Sep 2012, Richard Mudgett wrote:
> > Continuing with the saga of Digium vs MTNL Mumbai, looking for
> > suggestions on handling incoming Caller-ID issues.  The card
> > manages to
> > grab a couple of (random) digits of the incoming CID, but they're
> > more
> > or less useless.  Is there any way to fix this?
> > 
> > Asterisk 1.8.13, Dahdi 2.5.0.1 on Debian Testing (Wheezy), MTNL
> > Mumbai.
> > Digium, Inc. Wildcard AEX410 4-port analog card (PCI-Express)
> > 
> > chan_dahdi.conf contains:
> > usecallerid = yes
> > cidsignalling=dtmf
> > cidstart=polarity_in
> > 
> > Signalling is fxsks.
> > 
> > Log (calling number is 9811066XXX):

> You appear to be suffering form
> https://issues.asterisk.org/jira/browse/ASTERISK-19610
> 
> It is fixed in the just released v1.8.17.0-rc1.

Thanks, will test that on the weekend then.

Regards,

-- Raj
-- 
Raj Mathur  || r...@kandalaya.org   || GPG:
http://otheronepercent.blogspot.com || http://kandalaya.org || CC68
It is the mind that moves   || http://schizoid.in   || D17F

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Re: [asterisk-users] Digium AEX410, MTNL Mumbai Caller-ID problems

[A J Stiles]

On Friday 14 September 2012, Mitul Limbani wrote:
> Operator sends callerId after 1st small ring (actually this is not audible
> since its very small duration ring) post which all the data flows.
> 
> However, sometimes due to line distrubance this first small ring is missed.

Are you sure caller ID in India is not done as it is in the UK, with a 
reversal of the line polarity before the first ring and the ident sent after 
the reversal?

-- 
AJS

Answers come *after* questions.

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Re: [asterisk-users] Digium AEX410, MTNL Mumbai Caller-ID problems

[Richard Mudgett]

> Continuing with the saga of Digium vs MTNL Mumbai, looking for
> suggestions on handling incoming Caller-ID issues.  The card manages
> to
> grab a couple of (random) digits of the incoming CID, but they're
> more
> or less useless.  Is there any way to fix this?
> 
> Asterisk 1.8.13, Dahdi 2.5.0.1 on Debian Testing (Wheezy), MTNL
> Mumbai.
> Digium, Inc. Wildcard AEX410 4-port analog card (PCI-Express)
> 
> chan_dahdi.conf contains:
> usecallerid = yes
> cidsignalling=dtmf
> cidstart=polarity_in
> 
> Signalling is fxsks.
> 
> Log (calling number is 9811066XXX):
> [Sep 14 08:21:11] DEBUG[9337]: chan_dahdi.c:11895 do_monitor: Monitor
> doohicky
> got event Ring Begin on channel 1
> [Sep 14 08:21:11] DEBUG[9337]: sig_analog.c:3621
> analog_handle_init_event:
> channel (1) - signaling (5) - event (ANALOG_EVENT_RINGBEGIN)
> [Sep 14 08:21:11] DEBUG[9337]: chan_dahdi.c:11895 do_monitor: Monitor
> doohicky
> got event Ring/Answered on channel 1
> [Sep 14 08:21:11] DEBUG[9337]: sig_analog.c:3621
> analog_handle_init_event:
> channel (1) - signaling (5) - event (ANALOG_EVENT_RINGOFFHOOK)
> [Sep 14 08:21:11] DEBUG[9337]: dsp.c:471 ast_tone_detect_init: Setup
> tone 1100
> Hz, 500 ms, block_size=160, hits_required=21
> [Sep 14 08:21:11] DEBUG[9337]: dsp.c:471 ast_tone_detect_init: Setup
> tone 2100
> Hz, 2600 ms, block_size=160, hits_required=116
> [Sep 14 08:21:11] DEBUG[9337]: dsp.c:1576 ast_dsp_set_busy_pattern:
> dsp busy
> pattern set to 0,0
> [Sep 14 08:21:11] DEBUG[9315]: devicestate.c:340 _ast_device_state:
> No provider
> found, checking channel drivers for DAHDI - 1
> [Sep 14 08:21:11] DEBUG[9315]: devicestate.c:458 do_state_change:
> Changing state
> for DAHDI/1 - state 2 (In use)
> [Sep 14 08:21:11] DEBUG[9315]: devicestate.c:438 devstate_event:
> device 'DAHDI/1'
> state '2'
> [Sep 14 08:21:11] DEBUG[11186]: sig_analog.c:1769 __analog_ss_thread:
> __analog_ss_thread 1
> -- Starting simple switch on 'DAHDI/1-1'
> [Sep 14 08:21:11] DEBUG[11186]: sig_analog.c:2392 __analog_ss_thread:
> Receiving
> DTMF cid on channel DAHDI/1-1
> [Sep 14 08:21:11] DEBUG[9350]: app_queue.c:1487 handle_statechange:
> Device
> 'DAHDI/1' changed to state '2' (In use) but we don't care because
> they're not a
> member of any queue.
> [Sep 14 08:21:12] DEBUG[11186]: sig_analog.c:1602 analog_handle_dtmf:
> Begin DTMF
> digit: 0x31 '1' on DAHDI/1-1
> [Sep 14 08:21:12] DEBUG[11186]: chan_dahdi.c:2026 my_handle_dtmf:
> Begin DTMF
> digit: 0x31 '1' on DAHDI/1-1
> [Sep 14 08:21:12] DEBUG[11186]: dsp.c:1424 ast_dsp_process: DTMF
> Detected - Reset
> busydetector
> [Sep 14 08:21:12] DEBUG[11186]: sig_analog.c:1602 analog_handle_dtmf:
> End DTMF
> digit: 0x31 '1' on DAHDI/1-1
> [Sep 14 08:21:12] DEBUG[11186]: chan_dahdi.c:2026 my_handle_dtmf: End
> DTMF digit:
> 0x31 '1' on DAHDI/1-1
> [Sep 14 08:21:12] DEBUG[11186]: sig_analog.c:2426 __analog_ss_thread:
> CID got
> digit '1'
> [Sep 14 08:21:12] DEBUG[11186]: sig_analog.c:1602 analog_handle_dtmf:
> Begin DTMF
> digit: 0x36 '6' on DAHDI/1-1
> [Sep 14 08:21:12] DEBUG[11186]: chan_dahdi.c:2026 my_handle_dtmf:
> Begin DTMF
> digit: 0x36 '6' on DAHDI/1-1
> [Sep 14 08:21:12] DEBUG[11186]: dsp.c:1424 ast_dsp_process: DTMF
> Detected - Reset
> busydetector
> [Sep 14 08:21:12] DEBUG[11186]: sig_analog.c:1602 analog_handle_dtmf:
> End DTMF
> digit: 0x36 '6' on DAHDI/1-1
> [Sep 14 08:21:12] DEBUG[11186]: chan_dahdi.c:2026 my_handle_dtmf: End
> DTMF digit:
> 0x36 '6' on DAHDI/1-1
> [Sep 14 08:21:12] DEBUG[11186]: sig_analog.c:2426 __analog_ss_thread:
> CID got
> digit '6'
> [Sep 14 08:21:13] DEBUG[11186]: sig_analog.c:3509 analog_exception:
> analog_exception 1
> [Sep 14 08:21:13] DEBUG[11186]: sig_analog.c:3603 analog_exception:
> Exception on
> 16, channel 1
> [Sep 14 08:21:13] DEBUG[11186]: sig_analog.c:2660
> __analog_handle_event:
> __analog_handle_event 1
> [Sep 14 08:21:13] DEBUG[11186]: sig_analog.c:2687
> __analog_handle_event: Got
> event ANALOG_EVENT_RINGBEGIN(12) on channel 1 (index 0)
> [Sep 14 08:21:14] DEBUG[11186]: sig_analog.c:3509 analog_exception:
> analog_exception 1
> [Sep 14 08:21:14] DEBUG[11186]: sig_analog.c:3603 analog_exception:
> Exception on
> 16, channel 1
> [Sep 14 08:21:14] DEBUG[11186]: sig_analog.c:2660
> __analog_handle_event:
> __analog_handle_event 1
> [Sep 14 08:21:14] DEBUG[11186]: sig_analog.c:2687
> __analog_handle_event: Got
> event ANALOG_EVENT_RINGOFFHOOK(2) on channel 1 (index 0)
> [Sep 14 08:21:14] DEBUG[11186]: sig_analog.c:3043
> __analog_handle_event: Ring
> detected
> [Sep 14 08:21:14] DEBUG[11186]: sig_analog.c:2441 __analog_ss_thread:
> CID got
> string '16'
> [Sep 14 08:21:14] WARNING[11186]: callerid.c:243 callerid_get_dtmf:
> Couldn't
> detect start-character. CID parsing might be unreliable
> [Sep 14 08:21:14] DEBUG[11186]: sig_analog.c:2443 __analog_ss_thread:
> CID is
> '16', flags 0
> [Sep 14 08:21:14] DEBUG[9315]: devicestate.c:340 _ast_device_state:
> No provider
> found, checking channel drivers for DAHDI - 1
> [Sep 14 08:21:14] DEBUG

Re: [asterisk-users] Digium AEX410, MTNL Mumbai Caller-ID problems

[Mitul Limbani]

Operator sends callerId after 1st small ring (actually this is not audible
since its very small duration ring) post which all the data flows.

However, sometimes due to line distrubance this first small ring is missed.

Regards,
Mitul Limbani,
Chief Architech & Founder,
Enterux Solutions Pvt. Ltd.
110 Reena Complex, Opp. Nathani Steel,
Vidyavihar (W), Mumbai - 400 086. India
http://www.enterux.com/
http://www.entvoice.com/
email: mi...@enterux.in
DID: +91-22-71967121
Cell: +91-9820332422




On Fri, Sep 14, 2012 at 7:49 PM, Warren Selby  wrote:

> On Fri, Sep 14, 2012 at 9:02 AM, Raj Mathur (राज माथुर) <
> r...@linux-delhi.org> wrote:
>
>> So if there's a good chance that the latest Asterisk and Dahdi packages
>> will give better results in testing or might actually solve the problem,
>> I'll be glad to compile from source.  If not, then perhaps it's not
>> worth polluting a production box with locally-compiled packages.
>>
>>
> Try adding a Wait(2) between your NoOp and your Verbose lines.  I don't
> know about your telco, but sometimes the CID is not sent with the first
> ring, and you have to add a Wait(2) to grab it.
>
> You may even want to call your upper level support at your telco and ask
> them how and when they send your callerid information...
>
>
> --
> Thanks,
> --Warren Selby, dCAP
> http://www.SelbyTech.com 
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Re: [asterisk-users] Need to record user voice while play background music

[Steve Edwards]

Un-top-posting...


On Fri, 14 Sep 2012, RAJNI VANZA wrote:

I was wondering if anyone has any experience for recording user voice 
while play background music?



On Fri, Sep 14, 2012 at 11:13 AM, Steve Edwards wrote:

What methods have you tried?


On Fri, 14 Sep 2012, RAJNI VANZA wrote:

I have tried with Monitor(), MixMonitor() and conference (meetme) in 
dilaplan. By using Monitor() in dialplan recording is done through in, 
out two file mix in one recording file created but its not so good 
result.


I'd vote for mixmonitor(). I use it to record calls in a 'third party 
verification' system. Interestingly (at least in 1.2), you can record the 
entire call in 1 file (for debugging) while recording parts in other files 
for delivery to the client. The 'parts' are assembled using sox in an AGI 
at the completion of the call.


--
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] kernel: dahdi: Master changed to TE2/0/2 --- Is a normal message

[equis software]

Shaun, could it be related to
http://lists.digium.com/pipermail/asterisk-users/2012-March/271304.html ??
I'm using DAHDI Version: 2.4.1.2


On Fri, Sep 14, 2012 at 9:00 AM, equis software wrote:

> My test were...
>
> SIEMENS -> LTG1 <--- cable1 - SS7 ---> IVR1 (have errors)
>|-> LTG2 <--- cable2 - SS7 ---> IVR2 (OK)
>
> minutes later...
>
> SIEMENS -> LTG1 <--- cable1 - SS7 ---> IVR2 (OK)
>|-> LTG2 <--- cable2 - SS7 ---> IVR1 (have same errors)
>
> Errors:
> Sep 12 11:49:42 call3 kernel: [1018444.069418] dahdi: Master changed to
> TE2/0/1
> Sep 12 11:49:48 call3 kernel: [1018449.724411] dahdi: Master changed to
> TE2/0/2
> Sep 12 11:49:48 call3 kernel: [1018449.823093] dahdi: Master changed to
> TE2/0/1
> Sep 12 11:49:52 call3 kernel: [1018454.175277] dahdi: Master changed to
> TE2/0/2
> Sep 12 11:49:52 call3 kernel: [1018454.198138] dahdi: Master changed to
> TE2/0/1
> Sep 12 11:49:53 call3 kernel: [1018455.493002] dahdi: Master changed to
> TE2/0/2
> Sep 12 11:49:53 call3 kernel: [1018455.593648] dahdi: Master changed to
> TE2/0/1
>
>
>
>
> On Fri, Sep 14, 2012 at 5:31 AM, Patrick Lists <
> asterisk-l...@puzzled.xs4all.nl> wrote:
>
>> On 09/14/2012 10:25 AM, A J Stiles wrote:
>> [snip]
>>
>>  It could be nothing more than a dry solder joint on one of the RJ45s.
>>>  For the
>>> sake of five minutes' work with a soldering iron, that's got to be worth
>>> eliminating.
>>>
>>
>> Wouldn't that void your warranty? I would take it up with Digium support
>> and let them sort it out.
>>
>> Regards,
>> Patrick
>>
>>
>>
>>
>> --
>> __**__**_
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>>   http://www.asterisk.org/hello
>>
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>> To UNSUBSCRIBE or update options visit:
>>   
>> http://lists.digium.com/**mailman/listinfo/asterisk-**users
>>
>
>
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Re: [asterisk-users] Fax Detect on Demand

[Eric Wieling]

No.  While the Wait(4) is waiting Asterisk should detect the fax tone of the 
sending fax machine and the call should be sent to the fax extension and never 
ring the phone.

Fax detect must be configured on the INCOMING channel.  In my case that is 
DAHDI.

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
Sent: Friday, September 14, 2012 10:19 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Fax Detect on Demand



2012/9/13 Eric Wieling 


Yes.  Trivial

Off the top of my head:

exten => 12125551212,1,Set(EMAIL=john@example.com)
exten => 12125551212,n,Answer
exten => 12125551212,n,Wait(4)
exten => 12125551212,n,Dial(SIP/1212)

exten => fax,1,ReceiveFax(/tmp/myfax.tiff)
exten => fax,n,AGI(yourfax2emailagi.php,${FAXEXTEN},${EMAIL})

The Answer and Wait are required for faxdetection to work.




It doesn't work for me : using your example, SIP phone 1212 user would pick its 
handset up when it starts to ring, he would hear CNG tone but nothing happens 
and dialplan's fax extension is not entered.
In my setup, I used a fax machine connected to an Asterisk FXS port.
This machine dialed the SIP phone.

I also used SendFAX application with the very same SIP phone and it did work 
the same way: callee could hear fax tone and branching to fax extension didn't 
happen.
Nothing in CLI showed fax tone recognition.


In your setup, which type of incoming channel was calling the extension 
12121212 ?

You did set SIP1212 faxdetect to yes, didn't you ?
Did you enter any value sup t38p_udptl parameter ?


As a side note, if I tweak FXS port setting to detect fax, it works.

 



-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
Sent: Thursday, September 13, 2012 6:20 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Fax Detect on Demand



2012/8/16 Eric Wieling 


Using Asterisk 1.8.mumble.  We would like to use fax detect on 
demand.

Both chan_dahdi and chan_sip support setting fax detetect on a 
static basis,


For curiosity's sake, could you make it work first using static 
settings ?




but no way I've been able to find to enable/disable it on 
demand in the dialplan.

In 1.4 we used the NVFaxDetect 3rd party app, but that no 
longer appears to be maintained.

Does anyone have any suggestions?

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Re: [asterisk-users] Digium AEX410, MTNL Mumbai Caller-ID problems

[Warren Selby]

On Fri, Sep 14, 2012 at 9:02 AM, Raj Mathur (राज माथुर) <
r...@linux-delhi.org> wrote:

> So if there's a good chance that the latest Asterisk and Dahdi packages
> will give better results in testing or might actually solve the problem,
> I'll be glad to compile from source.  If not, then perhaps it's not
> worth polluting a production box with locally-compiled packages.
>
>
Try adding a Wait(2) between your NoOp and your Verbose lines.  I don't
know about your telco, but sometimes the CID is not sent with the first
ring, and you have to add a Wait(2) to grab it.

You may even want to call your upper level support at your telco and ask
them how and when they send your callerid information...


-- 
Thanks,
--Warren Selby, dCAP
http://www.SelbyTech.com 
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Re: [asterisk-users] Fax Detect on Demand

[Olivier]

2012/9/13 Eric Wieling 

> Yes.  Trivial
>
> Off the top of my head:
>
> exten => 12125551212,1,Set(EMAIL=john@example.com)
> exten => 12125551212,n,Answer
> exten => 12125551212,n,Wait(4)
> exten => 12125551212,n,Dial(SIP/1212)
>
> exten => fax,1,ReceiveFax(/tmp/myfax.tiff)
> exten => fax,n,AGI(yourfax2emailagi.php,${FAXEXTEN},${EMAIL})
>
> The Answer and Wait are required for faxdetection to work.
>


It doesn't work for me : using your example, SIP phone 1212 user would pick
its handset up when it starts to ring, he would hear CNG tone but nothing
happens and dialplan's fax extension is not entered.
In my setup, I used a fax machine connected to an Asterisk FXS port.
This machine dialed the SIP phone.

I also used SendFAX application with the very same SIP phone and it did
work the same way: callee could hear fax tone and branching to fax
extension didn't happen.
Nothing in CLI showed fax tone recognition.


In your setup, which type of incoming channel was calling the extension
12121212 ?

You did set SIP1212 faxdetect to yes, didn't you ?
Did you enter any value sup t38p_udptl parameter ?


As a side note, if I tweak FXS port setting to detect fax, it works.



>
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
> Sent: Thursday, September 13, 2012 6:20 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Fax Detect on Demand
>
>
>
> 2012/8/16 Eric Wieling 
>
>
> Using Asterisk 1.8.mumble.  We would like to use fax detect on
> demand.
>
> Both chan_dahdi and chan_sip support setting fax detetect on a
> static basis,
>
>
> For curiosity's sake, could you make it work first using static settings ?
>
>
>
>
> but no way I've been able to find to enable/disable it on demand
> in the dialplan.
>
> In 1.4 we used the NVFaxDetect 3rd party app, but that no longer
> appears to be maintained.
>
> Does anyone have any suggestions?
>
> --
>
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com--
> New to Asterisk? Join us for a live introductory webinar every
> Thurs:
>http://www.asterisk.org/hello
>
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>
>
>
>
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Re: [asterisk-users] Digium AEX410, MTNL Mumbai Caller-ID problems

[Raj Mathur (राज माथुर)]

On Friday 14 Sep 2012, Patrick Lists wrote:
> On 09/14/2012 05:26 AM, Raj Mathur (राज माथुर) wrote:
> [snip]
> 
> > Asterisk 1.8.13, Dahdi 2.5.0.1 on Debian Testing (Wheezy), MTNL
> > Mumbai. Digium, Inc. Wildcard AEX410 4-port analog card
> > (PCI-Express)
> 
> Your DAHDI and Asterisk versions are old so for starters I would
> update everything to the latest releases. See asterisk.org.

I could compile the latest from source for the sake of testing, but will 
eventually have to move back to a packaged Asterisk for production.  
Since the Digium Debian Asterisk packages don't have a maintainer any 
more, that means production will eventually return back to 1.8.13.

So if there's a good chance that the latest Asterisk and Dahdi packages 
will give better results in testing or might actually solve the problem, 
I'll be glad to compile from source.  If not, then perhaps it's not 
worth polluting a production box with locally-compiled packages.

Regards,

-- Raj
-- 
Raj Mathur  || r...@kandalaya.org   || GPG:
http://otheronepercent.blogspot.com || http://kandalaya.org || CC68
It is the mind that moves   || http://schizoid.in   || D17F

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Re: [asterisk-users] kernel: dahdi: Master changed to TE2/0/2 --- Is a normal message

[equis software]

My test were...

SIEMENS -> LTG1 <--- cable1 - SS7 ---> IVR1 (have errors)
   |-> LTG2 <--- cable2 - SS7 ---> IVR2 (OK)

minutes later...

SIEMENS -> LTG1 <--- cable1 - SS7 ---> IVR2 (OK)
   |-> LTG2 <--- cable2 - SS7 ---> IVR1 (have same errors)

Errors:
Sep 12 11:49:42 call3 kernel: [1018444.069418] dahdi: Master changed to
TE2/0/1
Sep 12 11:49:48 call3 kernel: [1018449.724411] dahdi: Master changed to
TE2/0/2
Sep 12 11:49:48 call3 kernel: [1018449.823093] dahdi: Master changed to
TE2/0/1
Sep 12 11:49:52 call3 kernel: [1018454.175277] dahdi: Master changed to
TE2/0/2
Sep 12 11:49:52 call3 kernel: [1018454.198138] dahdi: Master changed to
TE2/0/1
Sep 12 11:49:53 call3 kernel: [1018455.493002] dahdi: Master changed to
TE2/0/2
Sep 12 11:49:53 call3 kernel: [1018455.593648] dahdi: Master changed to
TE2/0/1



On Fri, Sep 14, 2012 at 5:31 AM, Patrick Lists <
asterisk-l...@puzzled.xs4all.nl> wrote:

> On 09/14/2012 10:25 AM, A J Stiles wrote:
> [snip]
>
>  It could be nothing more than a dry solder joint on one of the RJ45s.
>>  For the
>> sake of five minutes' work with a soldering iron, that's got to be worth
>> eliminating.
>>
>
> Wouldn't that void your warranty? I would take it up with Digium support
> and let them sort it out.
>
> Regards,
> Patrick
>
>
>
>
> --
> __**__**_
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>   http://www.asterisk.org/hello
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>   
> http://lists.digium.com/**mailman/listinfo/asterisk-**users
>
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Re: [asterisk-users] kernel: dahdi: Master changed to TE2/0/2 --- Is a normal message

[Patrick Lists]

On 09/14/2012 10:25 AM, A J Stiles wrote:
[snip]

It could be nothing more than a dry solder joint on one of the RJ45s.  For the
sake of five minutes' work with a soldering iron, that's got to be worth
eliminating.


Wouldn't that void your warranty? I would take it up with Digium support 
and let them sort it out.


Regards,
Patrick



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Re: [asterisk-users] kernel: dahdi: Master changed to TE2/0/2 --- Is a normal message

[A J Stiles]

On Thursday 13 September 2012, equis software wrote:
> After exchanging the cable with other equipment was running smoothly in my
> computer the problem persisted while the other team the cable that could be
> bad worked.
> With this test done, now suspect the problem I have it on the Digium card.

It could be nothing more than a dry solder joint on one of the RJ45s.  For the 
sake of five minutes' work with a soldering iron, that's got to be worth 
eliminating.

-- 
AJS

Answers come *after* questions.

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Re: [asterisk-users] Need to record user voice while play background music

[RAJNI VANZA]

Hi Steve,

Thanks for reply.

I have tried with Monitor(), MixMonitor() and conference (meetme) in
dilaplan.
By using Monitor() in dialplan recording is done through in, out two file
mix in one recording file created but its not so good result.

-- 
Best Regards,

Rajnikant Vanza
Call : +91-96999 64656
Software Engineer
---
Working On Linux,C/C++,VoIP Technology
Mumbai

On Fri, Sep 14, 2012 at 11:13 AM, Steve Edwards
wrote:

> On Fri, 14 Sep 2012, RAJNI VANZA wrote:
>
>  I was wondering if anyone has any experience for recording user voice
>> while play background music?
>>
>
> What methods have you tried?
>
> --
> Thanks in advance,
> --**--**
> -
> Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
> Newline  Fax: +1-760-731-3000
>
> --
> __**__**_
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>   http://www.asterisk.org/hello
>
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>   
> http://lists.digium.com/**mailman/listinfo/asterisk-**users
>
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Re: [asterisk-users] Digium AEX410, MTNL Mumbai Caller-ID problems

[Patrick Lists]

On 09/14/2012 05:26 AM, Raj Mathur (राज माथुर) wrote:
[snip]

Asterisk 1.8.13, Dahdi 2.5.0.1 on Debian Testing (Wheezy), MTNL Mumbai.
Digium, Inc. Wildcard AEX410 4-port analog card (PCI-Express)


Your DAHDI and Asterisk versions are old so for starters I would update 
everything to the latest releases. See asterisk.org.


Regards,
Patrick


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