Re: [asterisk-users] Deleting voicemail by program

[Mike Diehl]

John, that is some serious script-fu!  I does exactly what I was going to do 
in perl.  However, my initial testing indicates that asterisk will renumber 
voicemail boxes to eliminate holes.  But I'm still testing.

Thanks again,

Mike.

On Tuesday, October 10, 2023 11:47:35 AM EDT John Harragin wrote:
> Here is something I wrote years ago. I expect you can adjust it for your
> needs
> 
> 
> 
> # cat remove_blank_vmail
> #!/bin/bash
> # remove_blank_vmail takes arguments as voicemail boxes and removes
> messages with audio files shorter then MINSIZE (in bytes)
> #--
> # Description:
> # Author: John Harragin Monroe-Woodbury CSD
> # Created at: Thu Nov  6 12:27:35 EST 2008
> #
> # Copyright: None. Modify and use however you like...
> #
> #--
> # Configure section:
> 
> BASEDIR=/var/spool/asterisk/voicemail/default/  # default
> context
> MINSIZE=12000   # 1.5
> seconds
> 
> #--subroutines:
> 
> ProcessDir () {
>  lastfile=""
>  delcnt=0
>  for file in $(ls -A ${msgdir}/msg*.txt 2>/dev/null); do   # the
> redirect supresses msg when dir empty
>if [ $(stat --format=%s ${file/.txt/.wav}) -lt ${MINSIZE} ]; then
>  rm ${file/.txt/.*}
>  let delcnt++
>fi
>lastfile=${file}
>  done
>  if [ $delcnt -gt 0 ]; then echo "$delcnt short messages deleted from
> ${msgdir}"; fi
>  partfilename=${lastfile/*\/msg/}  # get number
> from file name
>  highest=${partfilename/.txt/}
>  while [[ $highest = 0* ]]; do highest=${highest#0}; done  # bash does
> not like leading zeros
>  if [ ${#highest} -eq 0 ]; then highest=0; fi  # ...or
> blanks for math
>  realcount=0
>  for ((x=0;x<=${highest};x+=1)); do
>chkname=msg$(printf "%04d" $x)  # build name
> - pad with zeros...
>if [ -e ${msgdir}/${chkname}.txt ]; then
>  if [ $realcount -ne $x ];then
>newname=msg$(printf "%04d" $realcount)
>for idivfile in $(ls -A ${msgdir}/${chkname}.*); do
>  mv ${idivfile} ${msgdir}/${newname}.${idivfile/*\/*./}
>done
>  fi
>  let realcount++
>fi
>  done
> }
> 
> #--main:
> 
> for ext in "$@"; do
>  if [ -d ${BASEDIR}${ext} ];then
>for msgdir in $(ls -d ${BASEDIR}${ext}/*); do
>  ProcessDir ${msgdir}
>done
>  else
>echo "${BASEDIR}${ext} is not a valid directory"
>  fi
>  echo "Processed extension $ext"
> done
> 
> On Mon, Oct 9, 2023 at 3:06 PM Mike Diehl  wrote:
> > Hi all,
> > 
> > I need to be able to delete a voicemail message using a program.
> > 
> > Is is sufficient to simply delete the .wav and .txt files in the spool
> > directory?
> > Or do I need to also renumber the remaining files?
> > 
> > For example, let say a given mailbox has 20 messages in it and I want to
> > delete message number 5.  Can I just delete the 2 files and expect that
> > asterisk will renumber them?  Or do I need to?
> > 
> > Also, is the answer the same when I migrate to storing voicemails in a
> > database?
> > 
> > Thanks in advance.
> > 
> > Mike
> > 
> > 
> > 
> > --
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> > https://community.asterisk.org/
> > 
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> >   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
> > 
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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] Deleting voicemail by program

[Mike Diehl]

Unfortunately, I'm using a version of asterisk that is old enough to not 
benefit from this... 

Mike.

On Monday, October 9, 2023 3:15:45 PM EDT Michael Bradeen wrote:
> Hi Mike,
> 
> New AMI actions were recently added to app_voicemail to let you remotely
> manipulate a mailbox:
> https://github.com/asterisk/asterisk/issues/181
> 
> Hope this helps.
> 
> BR,
> -Mike
> 
> On Mon, Oct 9, 2023 at 1:06 PM Mike Diehl  wrote:
> > Hi all,
> > 
> > I need to be able to delete a voicemail message using a program.
> > 
> > Is is sufficient to simply delete the .wav and .txt files in the spool
> > directory?
> > Or do I need to also renumber the remaining files?
> > 
> > For example, let say a given mailbox has 20 messages in it and I want to
> > delete message number 5.  Can I just delete the 2 files and expect that
> > asterisk will renumber them?  Or do I need to?
> > 
> > Also, is the answer the same when I migrate to storing voicemails in a
> > database?
> > 
> > Thanks in advance.
> > 
> > Mike
> > 
> > 
> > 
> > --
> > _
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> > 
> > Check out the new Asterisk community forum at:
> > https://community.asterisk.org/
> > 
> > New to Asterisk? Start here:
> >   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
> > 
> > asterisk-users mailing list
> > 
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users





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[asterisk-users] Deleting voicemail by program

[Mike Diehl]

Hi all,

I need to be able to delete a voicemail message using a program.  

Is is sufficient to simply delete the .wav and .txt files in the spool 
directory?  
Or do I need to also renumber the remaining files?

For example, let say a given mailbox has 20 messages in it and I want to 
delete message number 5.  Can I just delete the 2 files and expect that 
asterisk will renumber them?  Or do I need to?

Also, is the answer the same when I migrate to storing voicemails in a 
database?

Thanks in advance.

Mike



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Re: [asterisk-users] MWI with PJSIP - unsollicitated works fine, solicitated doesn't

[Mike]

Hi Marek,

Thank you - I figured out my issue, which was that the MWI subscribes to a 
PJSIP AOR, which in turns monitors a mailbox, not directly an actual 
mailbox.


Mike
-Original Message-
From: asterisk-users  On Behalf Of 
Marek Greško
Sent: November 19, 2021 03:57
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Subject: Re: [asterisk-users] MWI with PJSIP - unsollicitated works fine, 
solicitated doesn't

Hello Michael,

I was also struggling with solicited MWI after moving to pjsip. My problem 
was I was defining mailbox=111@extensioncontext. But the correct context in 
the mailbox command is to be defined by context in voicemail.conf. My 
voicemails were all defined in the context default (see voicemail.conf) and 
the mailbox command should look like this:
mailbox=111@default.

Hope this helps. I do not know whether this is also your problem.

Marek


2021-11-14 16:38 GMT+01:00, Mike :
> Hi,
>
>
>
> Just  recently moved over from chan_sip to PJSIP and am slowly
> cleaning up whatever needs to be.
>
>
>
> I can't seem to make sollicitated MWI work, but unsollicitated works fine.
>
>
>
>
> I got my phones subscribing to mailbox@context (i.e. 100@whatever)
>
>
>
> I have my related AOR entry (realtime, in a DB) set to
> mailboxes=100@whatever . I can see it is set properly by using the
> command "pjsip show aor "
>
>
>
> But when I turn pjsip logger on, I see messages from the phones
> subscribing and SIP/2.0 401 Unauthorized messages back.
>
>
>
> If I put the same column in my realtime DB (mailboxes) for ENPOINT to
> the same value (100@whatever) then it works fine, MWI works on the phone.
>
>
>
> For a few reasons I'd like to get MWI working in sollicitated mode
> instead.  Is there a trick to it?
>
>
>
> I upgraded to Asterisk 18.8.0 just to see if a later patch fixed
> anything, so I am current.
>
>
>
>
>
>
>
>
>
>
>
>
>
> Michael
>
>

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Re: [asterisk-users] MWI with PJSIP - unsollicitated works fine, solicitated doesn't

[Mike]

So I think I am halfway there.

 

It seems configuring 100@whatever in the aor turns the MWI subscription
from a 401 unauthorized into a 404 not found.

 

So I'm guessing the MWI subscribe goes through, since the aor now allows
it, but then fails when asterisk actually looks for the mailbox once
passes the "security" of mailboxes=100@whatever.

 

The thing is, the mailbox is only in a table  but asterisk definitely sees
it (and saves msg with no issues). "Voicemail show users for whatever"
lists it as being there.

 

But the mailbox is neither in voicemail.conf nor users.conf (by design).
Is this needed?

 

Is there a better place to ask this sort of question?

 

 

 

 

From: asterisk-users  On Behalf
Of Mike
Sent: November 14, 2021 10:38
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'

Subject: [asterisk-users] MWI with PJSIP - unsollicitated works fine,
solicitated doesn't

 

Hi,

 

Just  recently moved over from chan_sip to PJSIP and am slowly cleaning up
whatever needs to be.

 

I can't seem to make sollicitated MWI work, but unsollicitated works fine.


 

I got my phones subscribing to mailbox@context (i.e. 100@whatever)

 

I have my related AOR entry (realtime, in a DB) set to
mailboxes=100@whatever . I can see it is set properly by using the command
"pjsip show aor "

 

But when I turn pjsip logger on, I see messages from the phones
subscribing and SIP/2.0 401 Unauthorized messages back.

 

If I put the same column in my realtime DB (mailboxes) for ENPOINT to the
same value (100@whatever) then it works fine, MWI works on the phone.

 

For a few reasons I'd like to get MWI working in sollicitated mode
instead.  Is there a trick to it?

 

I upgraded to Asterisk 18.8.0 just to see if a later patch fixed anything,
so I am current.

 

 

 

 

 

 

Michael

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[asterisk-users] MWI with PJSIP - unsollicitated works fine, solicitated doesn't

[Mike]

Hi,

 

Just  recently moved over from chan_sip to PJSIP and am slowly cleaning up
whatever needs to be.

 

I can't seem to make sollicitated MWI work, but unsollicitated works fine.


 

I got my phones subscribing to mailbox@context (i.e. 100@whatever)

 

I have my related AOR entry (realtime, in a DB) set to
mailboxes=100@whatever . I can see it is set properly by using the command
"pjsip show aor "

 

But when I turn pjsip logger on, I see messages from the phones
subscribing and SIP/2.0 401 Unauthorized messages back.

 

If I put the same column in my realtime DB (mailboxes) for ENPOINT to the
same value (100@whatever) then it works fine, MWI works on the phone.

 

For a few reasons I'd like to get MWI working in sollicitated mode
instead.  Is there a trick to it?

 

I upgraded to Asterisk 18.8.0 just to see if a later patch fixed anything,
so I am current.

 

 

 

 

 

 

Michael

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[asterisk-users] Server loses sip registrations after converting to vm to mysql storage.

[Mike Diehl]

Hi all,

I've got an old server (Asterisk 13.28.0) that I'm trying to configure to store 
voicemail in a 
mysql database.  

I have sip realtime working via odbc and it's been working well for years.

However, when I recompile Asterisk in order to store voicemail in the database, 
I have 
problems.  (That is the ONLY thing I change.)

The server seems to run for a while and voicemail seems to work.  Then, the 
server loses 
ALL of it's sip registrations.  I have a script that I can run to reload the 
registrations, but 
the server eventually loses them again.

Any ideas as to where I should start looking?

Thanks in advance,



-- 
Mike Diehl


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Re: [asterisk-users] TON values

[Mike]

Never mind I just saw it - thank you.

Mike
-Original Message-
From: asterisk-users  On Behalf Of 
Doug Lytle
Sent: March 12, 2021 15:30
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Subject: Re: [asterisk-users] TON values

Mike,

The below link turned up for me in a Google Search

https://www.voip-info.org/asterisk-config-chandahdiconf/

Doug

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Re: [asterisk-users] TON values

[Mike]

Thank you - but this doesn't explain how I can change it on a call-by-call 
basis (it actually lets me think this is impossible)


-Original Message-
From: asterisk-users  On Behalf Of 
Doug Lytle
Sent: March 12, 2021 15:30
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Subject: Re: [asterisk-users] TON values

Mike,

The below link turned up for me in a Google Search

https://www.voip-info.org/asterisk-config-chandahdiconf/

Doug

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[asterisk-users] TON values

[Mike]

HI,

 

I am ask by my PRI provider to set up certain calls using the following
values:

TON for Called number National (2).
TON for Calling Number International (1).

My google-talents fail me.  The only "ton" reference I can find is an old
deprecated one, and it's not clear what it corresponds two among these two
and how to change the other.

Is there a Set(called number ton) equivalent I can use here?

My use case is calling a local (:i.e. North American for me) number and
sending an European CallerID (I'm bridging a European call to a NA number
and want to forward original callerid). 

 

Mike

 

 

 

 

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Re: [asterisk-users] CallerID presentation - presentation prohibited but still passing number

[Mike]

THANK YOU! Case closed, that was indeed the problem.

Michael





From: asterisk-users  On Behalf Of 
Joshua C. Colp
Sent: March 11, 2021 15:52
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Subject: Re: [asterisk-users] CallerID presentation - presentation 
prohibited but still passing number



On Thu, Mar 11, 2021 at 4:50 PM Mike mailto:mich...@virtutel.ca> > wrote:

Thank you for taking the time.  I believe you misunderstood my question.
Callerid presence is passed perfectly already, as shown through Verbose
commands on both sides of the SIP call. The CALLERID name and numbers
aren't passed properly ONLY when presence is "hidden".

As if Asterisk decided that since this is a hidden number, to replace the
number with "Anynomous " as opposed to letting the receiving
Asterisk process it as desired with whatever logic I choose.

I just tested without any u() or f() or s() functions - same result. No
improvement or degradation with my issue. (Not sure why I had these
options)



You probably want to set the "trust_id_outbound" option[1] to "yes".



[1] 
https://github.com/asterisk/asterisk/blob/master/configs/samples/sip.conf.sample#L377



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Asterisk Technical Lead

Sangoma Technologies

Check us out at www.sangoma.com <http://www.sangoma.com/>  and 
www.asterisk.org <http://www.asterisk.org/>

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Re: [asterisk-users] CallerID presentation - presentation prohibited but still passing number

[Mike]

Thank you for taking the time.  I believe you misunderstood my question.
Callerid presence is passed perfectly already, as shown through Verbose
commands on both sides of the SIP call. The CALLERID name and numbers
aren't passed properly ONLY when presence is "hidden".

As if Asterisk decided that since this is a hidden number, to replace the
number with "Anynomous " as opposed to letting the receiving
Asterisk process it as desired with whatever logic I choose.

I just tested without any u() or f() or s() functions - same result. No
improvement or degradation with my issue. (Not sure why I had these
options)




-Original Message-
From: phr...@phreaknet.org 
Sent: March 11, 2021 15:33
To: Mike ; asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] CallerID presentation - presentation
prohibited but still passing number

I've been able to pass presentation status between tandems without needing
to do anything explicitly. This seems to be part of the Caller ID that is
transmitted without explicit intervention. Have you tested without using
the u option? I've never used the u option and not had issues with
presentation transmitting as supposed to. This is with manual
Set(CALLERID(pres)=something) and then seeing if it gets honored on the
other end. The remote Asterisk gets the full number, of course, but the
called telephone does not display it. Perhaps the u option is intended for
lines, not trunks, and so the number never gets sent?

The only time I've found I (may) need to explicitly account for
presentation is if I am regenerating the call, and then this needs to be
accounted for in the call file (ignore that if that makes no sense).

The only inconsistency I've encountered has to do with presentation
mismatches between the name and the number. If, for instance, I want the
number to display but not the same, setting the presentations as the
documentation would suggest does not work. The behavior is inconsistent
between different SIP clients and it didn't work for me in any logical
way. I didn't bother to a file a bug report about it, as I worked around
this by simply doing Set(CALLERID(name)=) to empty the name and write the
original name back into the variable after the call. Your mileage may
vary.

NA

On 3/11/2021 2:22 PM, Mike wrote:
>
> Hi,
>
> Using Asterisk 13.36.0
>
> I have a bit of a technical issue with hidden caller IDs.  My setup,
> at the moment, is composed of two Asterisk boxes. In some instance,
> calls arrive on Asterisk A, and are then sent to Asterisk B for
> further processing. The link between them is SIP (both on the same
> switch/LAN). Asterisk A has a Digium PRI card (recent one) and a PRI
link.
>
> When I receive a hidden number (i.e. “presentation prohibited”) call
> on Asterisk A through PRI, I get the following Caller ID information
> (using 444-555- as example):
>
> “ <444555>”
>
> And
>
> CallerID presence is received as “prohibited_not_screened”.
>
> Which is fine – I know the incoming number BUT I am told not to show
> it to the end user. All good.
>
> The problem is when calls are not processed on Asterisk A, but sent to
> Asterisk B for further processing. The dial command I used on Asterisk
> A to send calls to AsterisB is the following:
>
> exten => s,n,Dial(SIP/AsteriskB/123,,f(""
> <444555>)u(prohib_not_screened))
>
> Again, so far so good. But, on Asterisk B in the appropriate context,
> on extension 123, my first command is a Verbose to show Callerid(all)
> and the received called id is shown as “Anonymous ” with
> CALLERID presence still “prohib_not_screened”. I would like Asterisk B
> to receive the actual callerid (“ <444555>”) along with the
> appropriate CallerID presence value (which is correct already).
>
> Basically I want to “pass forward” both CALLERID and CALLERIDPRES
> exactly as received on AteriskA to AsteriskB so that AsteriskB gets
> the exact same info AsteriskA had in the first place.
>
> How do I accomplish this?
>
> Michael
>

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[asterisk-users] CallerID presentation - presentation prohibited but still passing number

[Mike]

Hi,

 

Using Asterisk 13.36.0

 

I have a bit of a technical issue with hidden caller IDs.  My setup, at
the moment, is composed of two Asterisk boxes. In some instance, calls
arrive on Asterisk A, and are then sent to Asterisk B for further
processing. The link between them is SIP (both on the same switch/LAN).
Asterisk A has a Digium PRI card (recent one) and a PRI link.

 

When I receive a hidden number (i.e. "presentation prohibited") call on
Asterisk A through PRI, I get the following Caller ID information (using
444-555- as example):

" <444555>" 

And 

CallerID presence is received as "prohibited_not_screened".

 

Which is fine - I know the incoming number BUT I am told not to show it to
the end user. All good.

 

The problem is when calls are not processed on Asterisk A, but sent to
Asterisk B for further processing.  The dial command I used on Asterisk A
to send calls to AsterisB is the following:

exten => s,n,Dial(SIP/AsteriskB/123,,f(""
<444555>)u(prohib_not_screened))

 

Again, so far so good. But, on Asterisk B in the appropriate context, on
extension 123, my first command is a Verbose to show Callerid(all) and the
received called id is shown as "Anonymous " with CALLERID
presence still "prohib_not_screened". I would like Asterisk B to receive
the actual callerid (" <444555>") along with the appropriate CallerID
presence value (which is correct already). 

 

Basically I want to "pass forward" both CALLERID and CALLERIDPRES exactly
as received on AteriskA to AsteriskB so that AsteriskB gets the exact same
info AsteriskA had in the first place.

 

How do I accomplish this?

 

 

Michael

 

 

 

 

 

 

 

 

 

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Re: [asterisk-users] Forcing mwi update

[Mike Diehl]

On Thursday, May 16, 2019 05:12:17 PM Joshua C. Colp wrote:
> On Thu, May 16, 2019, at 5:00 PM, Mike Diehl wrote:
> > Hi all,
> > 
> > 
> > I've got a program that connects via AMI and acts upon the voicemail
> > message waiting event.
> > 
> > 
> > I'd like to be able to force one of those events at will instead of
> > having to wait for the voicemail app to cause the event to get emitted.
> > 
> > 
> > Is this possible? AMI or asterisk command?
> 
> Do you mean something like the MailboxCount AMI action[1]?
> 
> [1]
> https://wiki.asterisk.org/wiki/display/AST/Asterisk+16+ManagerAction_Mailbo
> xCount

Um, ya.  That one!  

Thank you so much.  Now I'm feeling pretty silly.


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[asterisk-users] Forcing mwi update

[Mike Diehl]

Hi all,

I've got a program that connects via AMI and acts upon the voicemail message 
waiting event.

I'd like to be able to force one of those events at will instead of having to 
wait for the 
voicemail app to cause the event to get emitted.

Is this possible?  AMI or asterisk command?

Thanks

-- 
Mike Diehl



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Re: [asterisk-users] Odd one-way audio problem (Mike Diehl)

[Mike Diehl]

Hi, and thank you for your suggestion!

As it turns out, my server didn't even HAVE an rtp.conf file...  (No, I don't 
know 
how that happened...)

So I created one with:

rtpstart=1
rtpend=2

and reloaded chan_sip.

I hope that is sufficient. Or do I need to restart asterisk completely?

Anyway, my user tested later that day and they are still having problems

Any other ideas?

Mike.


On Friday, March 22, 2019 08:32:39 AM Stefan Viljoen wrote:
> Hi Mike
> 
> In rtp.conf, what are the port ranges you specify?
> 
> I had almost exactly the same problem not too long ago. People will phone,
> and sometimes it will work, sometimes not - one way audio would happen,
> then start working, then stop working.
> 
> The problem turned out to be that the port specification for RTP traffic in
> /etc/asterisk/rtp.conf was too wide.
> 
> It was set to
> 
> rtpstart=1
> rtpend=65535
> 
> (apparently by a previous maintainer / technician who worked on the system.)
> 
> The high port number was too high, and only after I investigated in detail
> with our trunk provider, were they able to determine that somtimes the
> Asterisk on my side was negotiating too high port numbers for RTP with
> their system.
> 
> I changed rtp.conf to read
> 
> rtpstart=1
> rtpend=2
> 
> and all the random one-way audio problems have been gone for more than two
> months. This client now has had thousads of successful calls so far after
> this change was made.
> 
> I also had the issue where MOST calls in their office was fine (with
> rtp.conf at 1 to 65535) though some would still fail, I'm guessing that
> was due to NATing not being done in the office (e. g. a wider "range" of
> RTP ports worked) vs. when they connected to their provider's SIP trunk on
> the internet to negotiate calls where it was ignoring the higher ports
> ("too high" ports) or their local firewall wasn't allowing some high ports
> to be opened that were "too high".
> 
> Restricting the RTP port range between 1 and 2 in this case solved
> their problem definitively and forever.
> 
> E. g. something similar given that you start that "most of the time" things
> worked fine - which is exactly the symptom I had with this client.
> 
> Just a thought...
> 
> Regards
> 
> Stefan
> 
> ---
> 
> Hi all,
> 
> I have a user who is reporting one-way audio, but only when a call is made
> to or from particular PSTN (cell) numbers.
> 
> Their phones are behind a NAT router and my server is on the open Internet.
> 
> Calls within their office sound fine.  Calls to/from most numbers sound
> fine.
> 
> When they took their phones home, those same phone numbers still had
> problems.
> 
> So, I don't think it's their network.  I've taken pcaps of both legs of
> example calls.  On the provider-side, I see 2-way audio.  On the
> client-side, I only hear one side.
> 
> Most of the time, though, their phones work correctly.
> 
> Any ideas where to look to fix this?
> 
> Thanks in advance.

-- 
Mike Diehl
Diehlnet Communications, LLC. 
Sales: (800) 254-6105   
Support: (505) 903-5700 
Fax: (505) 903-5701  

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Re: [asterisk-users] Odd one-way audio problem

[Mike Diehl]

My comments below:

On Wednesday, March 20, 2019 12:19:08 AM Antony Stone wrote:
> On Tuesday 19 March 2019 at 21:36:53, Mike Diehl wrote:
> > Hi all,
> > 
> > I have a user who is reporting one-way audio, but only when a call is made
> > to or from particular PSTN (cell) numbers.
> 
> I'm assuming you're using a PSTN trunking provider to connect to those
> numbers (ie: you don't have your own on-site gateway device).
> 
> Do you use only a single trunking provider, through which some calls show
> this problem, but most don't, or do you use several trunking providers, and
> the call numbers showing this problem always go via the same one?

I only have the one inbound provider, and all numbers go through that provider.


> > Their phones are behind a NAT router and my server is on the open
> > Internet.
> > 
> > Calls within their office sound fine.  Calls to/from most numbers sound
> > fine.
> 
> I think that basically rules out the common NAT reasons for one-way audio.
> 
> > When they took their phones home,
> 
> ...almost certainly also behind NAT...
> 
> > those same phone numbers still had problems.
> 
> But presumably other numbers didn't?  (Important to check!)
> 
> > So, I don't think it's their network.
> 
> From what you've said, I think you're right.
> 
> > I've taken pcaps of both legs of example calls.  On the provider-side, I
> > see 2-way audio.  On the client-side, I only hear one side.
> 
> Please explain that in a bit more detail.

I use voipmonitor to create sniffer traces of calls on my server.  I get 2 pcap 
files for 
each call: 

1.  Traffic from the phone to my server.
2.  Traffic from my server to the trunk provider.

In the cases I'm concerned about, one of these legs (only) will exhibit the 
one-way 
audio.


> You have an Asterisk server on the Internet, presumably with one IP address
> (or maybe two, but one IP4 and one IP6).
> 
> Where are you capturing "the provider side" and "the client side"?
> 
> Can you show us the tshark / tcpdump / whatever commands you are actually
> using to perform these captures, and make clear which machine/s you're
> running those commands on?
> 
> > Most of the time, though, their phones work correctly.
> > 
> > Any ideas where to look to fix this?
> 
> Only two things spring to mind so far:
> 
> 1. Transcoding?


There is no transcoding.  I ONLY allow ulaw for voice.


> 2. IPv4 on one side and IPv6 on the other (although I'm hard pushed to see
> how this could create one-way audio rather than no audio)?

My server doesn't have IPv6 configured at all.


> I think the key thing I would look for in the pcaps is for any re-invites -
> is one side telling the other "oh, you can get my audio from here" and
> that's not an accessible address?


I will take a more thorough look at the pcaps.


> However, why this would be specific to particular _numbers_ rather than
> particular SIP connections puzzles me too.
> 
> 
> Antony.


Thank you for your time, and for validating/confirming my conclusions.

Any other ideas would be most welcome!

-- 
Mike Diehl


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[asterisk-users] Odd one-way audio problem

[Mike Diehl]

Hi all,

I have a user who is reporting one-way audio, but only when a call is made to 
or from 
particular PSTN (cell) numbers.

Their phones are behind a NAT router and my server is on the open Internet.  

Calls within their office sound fine.  Calls to/from most numbers sound fine.

When they took their phones home, those same phone numbers still had problems.

So, I don't think it's their network.  I've taken pcaps of both legs of example 
calls.  On 
the provider-side, I see 2-way audio.  On the client-side, I only hear one side.

Most of the time, though, their phones work correctly.

Any ideas where to look to fix this?

Thanks in advance.


-- 
Mike Diehl


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[asterisk-users] Question about packet counts in voipmonitor

[Mike Diehl]

Hi all,

I'm not sure this is the place to ask, but here goes...

I'm using voipmonitor to gather call statistics such as packet counts, average 
jitter, etc.  
Eventually, I want to use those stats to detect and alert on poor call quality.

However, I'm finding that the packet counts for each leg of a given call can 
vary quite a 
bit.  

For example, I have a call that was connected for 84 seconds.  At 50 
frames/sec, I 
expect to see about 4200 frames.  However, on one side I see 4187 (which is 
good) and 
on the other side, I only see 2577 frames sent.

Am I doing something wrong?  Or is this approach simply doomed?

Any thoughts would be welcome.

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Re: [asterisk-users] Trying to add MoH to conference bridge

[Mike Diehl]

Well, it SEEMS to be working now.  I don't know what I did, and frankly,
don't have time to back track to find out.

Thanks for your time.

Mike.

On Thu, May 24, 2018 at 4:33 AM, Doug Lytle  wrote:

> On 05/23/2018 05:23 PM, Mike Diehl wrote:
>
>
> However, my user isn't hearing anything.  MoH does work otherwise.
>
>
> The only difference between your setup and mine is that I'm having them
> wait for the marked user.  In that case, MOH does play.
>
> What does your console output look like?
>
> Doug
>
>
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[asterisk-users] Trying to add MoH to conference bridge

[Mike Diehl]

Hi all,

I've got an AGI script that launches the conference bridge with a line like:

"$main::agi->exec(ConfBridge,$conf,default_bridge,default_user,$menu_profile)"

The $conf variable contains the room number.

I'm trying to configure it so that when only one person is in the
conference, they hear moh.

My /etc/asterisk/confbridge.conf looks like:

===
[general]

[default_bridge]
type=bridge

[default_user]
type=user
quiet=no
announce_join_leave=yes
music_on_hold_class=default
music_on_hold_when_empty=yes

[default_menu]
type=menu
0=playback_and_continue(/none)
1=increase_listening_volume
2=toggle_mute
3=increase_talking_volume
4=reset_listening_volume
5=admin_toggle_mute_participants
6=reset_talking_volume
7=decrease_listening_volume
8=admin_toggle_conference_lock
9=decrease_talking_volume
*=admin_kick_last
\#=participant_count
===

However, my user isn't hearing anything.  MoH does work otherwise.

What am I missing?

Thanks in advance,

Mike.
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[asterisk-users] Streaming MoH from iHeart radio?

[Mike Diehl]

Hi all,

I have a user who would like to stream their favorite radio station from
iHeart radio for their music on hold.

It this TECHNICALLY possible?  If so, any pointers would be appreciated.

Is this LEGAL in the US?

Thanks in advance,

Mike.
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[asterisk-users] Reject call from Asterisk dialplan

[Mike]

Hi,

 

I'm looking for a way to reject a call remotely using the Asterisk
dialplan.

 

For example, phone A is ringing - I'm at the other end of the room next to
phone B, and I want to reject the call to Phone A by dialing an extension.

 

I'm basically trying to reproduce the Polycom "reject" action but through
the Asterisk dialplan.

 

Reasons:

1.  It would allow me to log through Asterisk who's rejecting calls
2.  It would allow rejecting calls from another phone  (see above
scenario)

 

I thought there could be a "SendSIPCode 486 to SIP peer phoneA"
application, but a quick scan of the documentation does not bring obvious
answers.

 

Mike

 

 

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[asterisk-users] Problems with app_cdr writing CDRs nowhere

[Mike]

Hi,

 

I've been having an issue with Asterisk 13 (multiple versions, 13.18.2 is
one of them) logging CDRs in mysql. I am using app_cdr, and it generally
works fine, but when I run occasional maintenance that require Mysql to be
rebooted (generally during downtime) on the main mysql server (the one
logging the CDRs) sometimes (I can't find the pattern yet) it just stops
logging, but Asterisk keeps on happily thinking it is logging it's CDRs.
CDR show status is showing everything working fine:

 

CLI> cdr show status

Call Detail Record (CDR) settings

--

  Logging:Enabled

  Mode:   Simple

  Log unanswered calls:   No

  Log congestion: Yes

 

* Registered Backends

  ---

mysql

Adaptive ODBC

cdr_manager (suspended)

cdr-custom

 

CLI> cdr mysql status

Connected to mysql_server@cdr_db port 3306 using table cdr for 6 minutes,
4 seconds.

  Wrote 9238 records since last restart and 93 records since last
reconnect.

 

 

The only thing I can do when that happens is module reload app_cdr, which
wakes it up and it starts writing.  The CLI command "cdr submit" does
nothing.

 

So here are my questions:

1.  Where are my missing CDR records? Have they been written anywhere?
I am not using any other CDR backends, as cdr-custom created issues when
large amount of data needed to be written to local disk (queue problems).
Are they gone for good?
2.  How can I avoid this or mitigate this?

 

Any help is appreciated.

 

Mike

 

 

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[asterisk-users] Duplicate CDR's in Mysql

[Mike Diehl]

Hi all,
 
I have a problem I've not seen before.  My Asterisk server stores CDR's via 
mysql, and I'm getting duplicate records.
 
For example:
 
mysql> select uniqueid,count(*) from cdr  group by uniqueid having count(*)>1;
+--+--+
| uniqueid | count(*) |
+--+--+
| server12-1515090905.2182 |5 |
| server12-1515091190.2215 |3 |
+--+--+
2 rows in set (0.68 sec)
 
 
If I query for each uniqueid, I see that the records are identical.  I have a 
Perl script that goes through and removes the duplicates.  Otherwise, EVERY CDR 
would be duplicated.
 
Now, my Asterisk server was configured with multiple CDR backends, but I 
unloaded those modules.  Here is what I have configured during run-time:
 
*CLI> cdr show status
 
Call Detail Record (CDR) settings
--
  Logging:Enabled
  Mode:   Simple
  Log unanswered calls:   No
  Log congestion: No
 
* Registered Backends
  ---
cdr-custom
Adaptive ODBC
 
Any ideas would be appreciated.
 
 
 
-- 
Mike Diehl
Diehlnet Communications, LLC. 
Sales: (800) 254-6105   
Support: (505) 903-5700 
Fax: (505) 903-5701  
 


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[asterisk-users] Duplicate CDR's in mysql

[Mike Diehl]

Hi all,

I have a problem I've not seen before.  My Asterisk server stores CDR's via 
mysql, and I'm getting duplicate records.

For example:

mysql> select uniqueid,count(*) from cdr  group by uniqueid having count(*)>1;
+--+--+
| uniqueid | count(*) |
+--+--+
| server12-1515090905.2182 |5 |
| server12-1515091190.2215 |3 |
+--+--+
2 rows in set (0.68 sec)


If I query for each uniqueid, I see that the records are identical.  I have a 
Perl script that goes through and removes the duplicates.  Otherwise, EVERY CDR 
would be duplicated.

Now, my Asterisk server was configured with multiple CDR backends, but I 
unloaded those modules.  Here is what I have configured during run-time:

*CLI> cdr show status

Call Detail Record (CDR) settings
--
  Logging:Enabled
  Mode:   Simple
  Log unanswered calls:   No
  Log congestion: No

* Registered Backends
  ---
cdr-custom
Adaptive ODBC

Any ideas would be appreciated.



-- 
Mike Diehl
Diehlnet Communications, LLC. 
Sales: (800) 254-6105   
Support: (505) 903-5700 
Fax: (505) 903-5701  

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[asterisk-users] Queues - different moh for queue waiting and subsequent onhold

[Mike]

Hello,

Quick one here: is it possible to setup my dialplan on such a way that the
MoH while waiting to be answered by an agent on a Queue be different than
the one that I will hear when that agent puts me manually on hold for a
few minutes AFTER I finally got a hold of someone?

I seem to recall that setting up a musicclass in the dialplan was what was
used for onhold MoH, while the "music" field of the Queue was the "queue
waiting" MoH.  But that as back on 1.8, I am on Asterisk 13 right now, and
the "music" field of the Queue seems to overwrite the dialplan MoH
definition.

Mike





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Re: [asterisk-users] speech-recog.agi

[Mike Diehl]

If you'll release it for python, I'll take a stab at porting it to perl.

Mike


On October 19, 2017 4:53:52 PM EDT, Jonathan H <lardconce...@gmail.com> wrote:
>That's because it uses a deprecated API and endpoint.
>
>However, funny you should ask this, because I've just finished
>updating my Google TTS routine to take advantage of the new
>streamlined API.
>
>If you can wait a couple of days, I've stick it up on the repo -
>BUT... it's going to require python3.5+, the way I do it...
>
>Would that work for you?
>
>On 19 October 2017 at 18:41, Carlos Chavez <cur...@telecomab.mx> wrote:
>> I want to try using google for speech recognition in Asterisk and
>I
>> found a ready made AGI:
>>
>> http://zaf.github.io/asterisk-speech-recog/
>>
>> I have followed all the steps listed in the web site but I keep
>getting
>> this error:
>>
>> AGI Tx >> 200 result=99981 (timeout)
>endpos=22720
>> AGI Rx << VERBOSE "Unable to get recognition
>data." 3
>>
>> I made sure all the dependencies are met and that my API key for
>Google
>> Cloud Speech is correct (cut and paste).  Any pointers to get this to
>work
>> or any other quick waysto start using Google for speech recognition
>in
>> Asterisk?  Thanks.
>>
>>
>> --
>> Telecomunicaciones Abiertas de México S.A. de C.V.
>> Carlos Chávez
>> +52 (55)8116-9161
>>
>>
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Re: [asterisk-users] ERROR during high volume MoH dialplan

[Mike]

I had that problem before - I believe "task processor queue reached 500
scheduled tasks" crashing means your CDR records (queue) are being written
as the call ends, and if you had many thousands of entries being written
to disk it crashes asterisk (each ring to one phone is an entry, so it
goes up fast - for example 10 busy phones, with a between-ring delay of 1
second means every second there are 10 entries being put in memory)

 

I was using a MySQL CDR, but I had left the "CSV" type of CDR on. I
removed/disabled the CSV CDR module, kept on the SQL CDR only and things
have been working fine ever since.

 

Mike

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joseph Smith
Sent: September 1, 2017 16:41
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] ERROR during high volume MoH dialplan

 

Thanks for the suggestion Tony,

 

I installed each codec for MoH, core sounds, and extra sound packages.
Unfortunately the tests produce the same results. 

 

[Sep  1 20:36:45] ERROR[10081][C-7fe5]: frame.c:343 ast_frdup: FRACK!,
Failed assertion Excessive refcount 10 reached on ao2 object 0x20380b0
(

 

continuously for a while followed by a 

 

[Sep  1 20:36:46] WARNING[7761][C-770d]: taskprocessor.c:888
taskprocessor_push: The 'subp:PJSIP/sipp-0020' task processor queue
reached 500 scheduled tasks.

 

Then this time Asterisk actually crashed. :(

 

  _  

From: asterisk-users-boun...@lists.digium.com
<asterisk-users-boun...@lists.digium.com> on behalf of Tony Mountifield
<t...@softins.co.uk>
Sent: Friday, September 1, 2017 11:01 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] ERROR during high volume MoH dialplan 

 

In article
<cy4pr2201mb14643c2177c953fa27ac9e2ba8...@cy4pr2201mb1464.namprd22.prod.ou
tlook.com>,
Joseph Smith <warlock1...@hotmail.com> wrote:
> 
> Thanks for the feedback.
> 
> I do agree with having multiple smaller servers.  When I was first
approached with this task I mentioned as much. 
> However, the current desire is to work with already existing hardware.
That is out of my hands at the moment unless it
> just can't be done.  I will explore Freeswitch a bit soon to compare it
as well.
> 
> 
> I am struggling to find what the bottle neck is in this scenario.  Does
anyone have any advice on what that could be or
> on steps to discover it?   Do you think that tasks are pooling up
because of transcoding?  If so would it help to change
> the codec that is being used?  I am not sure about the MoH but the audio
files I am using are gsm.

You will find it less taxing on the server if you have MoH files and
sounds files
available in all the possible native formats. Then Asterisk can use the
appropriate
one for the channel without transcoding.

On my systems I have MoH and sounds installed in wav, ulaw, alaw, gsm and
g729.

They will also sound better than transcoding from the gsm versions.

Cheers
Tony
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Re: [asterisk-users] Detecting DoS attacks via SIP

[Mike Diehl]

I appreciate the discussion on the question I asked.

I currently listen for failed registration attempts via AMI and
automatically block the offending IP address at the firewall.  I was hoping
to find another AMI event that would be the magic bullet I need, but it
doesn't sound like that's going to happen.

I understand that fail2ban is probably not what I want and probably
wouldn't detect the attacks I'm seeing.

It turns out that not all of the attacks are from the "friendly scanner,"
but enough of them are that it's a good start.

So, I really like the idea of the IP geo location firewall rules coupled
with the "friendly scanner" filter, as provided by a few of you guys.  It
was mentioned that this is a broad hammer, but I'm kinda looking for a
broad hammer! ;^)

Looks like I need to do some research, but I think I have what I need.

Thanks again,

Mike Diehl.

On Sat, Aug 19, 2017 at 4:36 PM, Telium Technical Support <supp...@telium.ca
> wrote:

> I think you missed the point of the Digium post.  Fail2ban can ONLY ban
> IP’s if Asterisk records a failure to register.  Asterisk does not detect
> malformed SIP packets, buffer overflow attacks, suspicious dialing
> patterns, connection attempts outside geofenced areas, use of stolen
> credentials (rapid  ramp of calls using one set of credentials), etc.
>
>
>
> Asterisk only gives you a rudimentary “failed” message for a failure to
> register / wrong credentials.  And of course fail2ban only responds to
> Asterisk log messages, so it does little more than ban the annoying script
> kiddies.
>
>
>
> Have a good look at that Voip-Info page and read what actual SIP security
> systems do.  Then compare that to fail2ban and it’s night & day
> difference.  People still think fail2ban is a security system, and Digium
> is very clear that it is NOT.
>
>
>
>
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com] *On Behalf Of *Kseniya Blashchuk
> *Sent:* Thursday, August 17, 2017 12:41 AM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion <
> asterisk-users@lists.digium.com>
> *Subject:* Re: [asterisk-users] Detecting DoS attacks via SIP
>
>
>
> Well, correct me if I'm wrong, but I would say this conversation you have
> posted is a bit outdated, now fail2ban can be used with asterisk security
> log https://wiki.asterisk.org/wiki/display/AST/Asterisk+
> Security+Event+Logger.
>
>
>
> On Thu, Aug 17, 2017, 4:53 AM Telium Technical Support <supp...@telium.ca>
> wrote:
>
> Keep in mind that the attacks you are seeing in the log are ONLY the ones
> that Asterisk is detecting and rejecting.  All other attacks aren't even
> showing up!
>
> There's a good discussion of how to secure your PBX here:
> https://www.voip-info.org/wiki/view/asterisk+security
>
> In general, don't let the malevolent traffic get as far as the PBX (block
> at
> the firewall).  Also, Digium regularly warns users that fail2ban is NOT a
> security system: http://forums.asterisk.org/viewtopic.php?p=159984
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of mdiehl
> Sent: Tuesday, August 15, 2017 3:38 PM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] Detecting DoS attacks via SIP
>
> Hi all,
>
> Lately, I've seen an increase in the number of attacks against my system
> from the so-called "Friendly Scanner."  When one of these script kiddies
> targets my server, all I see for symptoms is a few of my trunks become
> lagged due to server load and a stream of messages on the console that
> resemble this:
>
> [Aug  2 20:27:50]   == Using SIP VIDEO CoS mark 6
> [Aug  2 20:27:50]   == Using SIP RTP TOS bits 24
> [Aug  2 20:27:50]   == Using SIP RTP CoS mark 5
> [Aug  2 20:32:47]   == Using SIP VIDEO TOS bits 24
> [Aug  2 20:32:47]   == Using SIP VIDEO CoS mark 6
> [Aug  2 20:32:47]   == Using SIP RTP TOS bits 24
> [Aug  2 20:32:47]   == Using SIP RTP CoS mark 5
> [Aug  2 20:34:26]   == Using SIP VIDEO TOS bits 24
> [Aug  2 20:34:26]   == Using SIP VIDEO CoS mark 6
>
>
> I have to turn on sip debugging to find out who's hitting me.  However, I
> can't just leave it on because it would kill my logging system.
>
> So, how are other people handling this?  Is there an AMI event I want watch
> for?  I watch for PeerStatus, but since there's no actual peer in the
> attack, I don't seem to get an event from AMI.
>
> Any ideas?
>
> Mike Diehl.
>
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Re: [asterisk-users] MoH via AGI broken after upgrade.

[Mike Diehl]

Man, I was hoping it was something like that.  I did read the release notes; I 
must have missed that part.

This should solve the problem, so thanks again.

Mike

On July 20, 2017 1:09:08 PM EDT, Richard Mudgett <rmudg...@digium.com> wrote:
>On Thu, Jul 20, 2017 at 11:50 AM, mdiehl <mdiehlena...@gmail.com>
>wrote:
>
>> I recently upgraded Asterisk from 1.8.x to 13.x and am now finding
>that
>> music on hold isn't working like it used to.
>>
>> It seems that even though the correct MoH class is being set, the
>system
>> still plays the "default" music.
>>
>> All of my call handling is done with an AGI script.  When a call is
>made,
>> the AGI script sets the MoH class before dialing.
>>
>> The log indicates that the correct class is being set:
>> [Jul 18 15:14:57] -- AGI Script Executing Application:
>> (SetMusicOnHold) Options: (jazz)
>>
>> However, when the call is placed on hold, the "default" MoH class is
>used:
>> [Jul 18 15:15:50] -- Started music on hold, class 'default', on
>> channel 'SIP/trunk-bfa9'
>>
>>
>> My AGI script is writen in Perl.  Here is the line that does the MuH
>class
>> setting:
>>
>> $agi->exec("SetMusicOnHold", $o->{musiconhold});
>>
>> I have verified that $o->{musiconhold} contains the name of a valid
>MoH
>> class.
>>
>> Is there a different/new way to set the MoH class in version 13?
>>
>> Any advise would be welcome.
>>
>
>The SetMusicOnHold application was deprecated in v1.6 and removed in
>v13.
>Use
>Set(CHANNEL(musicclass)=class) instead to set the music class on the
>channel.
>
>The change was documented in the UPGRADE.txt files.
>
>Richard

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[asterisk-users] Asterisk crashes when storing voicemail via odbc

[Mike Diehl]

Hi all,

I'm working on migrating all of my servers to store voicemail in a mysql 
database via 
odbc.

I've got a development server that I can reconfigure and test at will.  When 
it's 
configured to store vm on the file system, it seems to be rock solid.

However, when I ONLY change it to store vm in the database, it becomes very 
unstable.  

Here's what it's doing.  When I attempt leave a voicemail, I am prompted to 
leave a 
message.  Once I have left a message, the console locks up and I have to 
killall -9 to get 
it to restart and become responsive again.

I'm running Asterisk 13.14.0 built by root @ server on a x86_64 running Linux 
on 
2017-06-20 14:27:06 UTC

For odbc, I've got unixODBC 2.3.2-r2.

Are these the versions I should be using?  If so, any recommendations as to how 
to 
troubleshoot this would be most welcome.

TIA,

-- 
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Re: [asterisk-users] CallerId presence issue

[Mike]

Thank you - At first glance it seems to have done the trick.

Mike


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Daniel Tryba
Sent: June 14, 2017 10:41
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] CallerId presence issue

On Wed, Jun 14, 2017 at 10:18:19AM -0400, Mike wrote:
> I have a PRI coming in PBX_A and PBX_A is connected to PBX_B via SIP.
> PBX_A gets PRI calls on a 4 port Digium card, and each call naturally 
> has its own callerid values and presence.  I pass on those calls to 
> PBX_B via SI, and I'm trying to pass on this CALLERID info to PBX_B as
well.
> 
> My relevant dialplan snippet on PBX_A is:
> exten =>
> 1,1,Dial(SIP/pbx_b/55,,f(${CALLERID(all)})u(${CALLERID(pres)})
> ))
...
> I'm clearly missing something to pass on the callerid presence state 
> via the SIP link, but I can't figure out what.

Never heard of this method, are you sure this works for SIP, sound more
like for ISDN (look at packet captures).

But the/a standardized method is to use the P-Asserted-Identity and
Privacy headers (rfc3325). This should work if you set in the peer configs
in sip.conf on both sides:
sendrpid=pai
trustrpid=yes

Or you can do header manipulation/getting/setting manualy if desired.

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Re: [asterisk-users] CallerId presence issue

[Mike]

Actually, a correction: the callerid isn't passed on properly either: on
SIP_B I get "Anonymous " instead of "  <514-555-1234>"  that my
dial app is sending.

 

The exact dial command that is used, once variables are evaluated, is
this:

Dial(SIP/pbx3/555,,f("" <5145551234>)u(prohib_not_screened))"

 

While the log value found on the other end of the sip link are evaluated,
I get this:

Callerid name: Anonymous callerid number: number: anonymous  Presence
information : allowed_not_screened - allowed_not_screened -
allowed_not_screened

 

Somewhere in this Dial(SIP/) command callerid info is changed.  An
asterisk verbose check does not show me anything that would change
callerid info.

 

Mike

 

 

 

 

 

 

 

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike
Sent: June 14, 2017 10:18
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] CallerId presence issue

 

Hi,

 

I've run into a minor snag trying to pass on CALLERID presence from one
Asterisk to another via SIP (both running 13.16.0)

 

I have a PRI coming in PBX_A and PBX_A is connected to PBX_B via SIP.
PBX_A gets PRI calls on a 4 port Digium card, and each call naturally has
its own callerid values and presence.  I pass on those calls to PBX_B via
SI, and I'm trying to pass on this CALLERID info to PBX_B as well. 

 

My relevant dialplan snippet on PBX_A is:

 

exten =>
1,1,Dial(SIP/pbx_b/55,,f(${CALLERID(all)})u(${CALLERID(pres)})))

 

*the u() value being dynamically taken from the channel itself.

 

On pbx_b, I  have a simply verbose line like this:

exten => 55,1,Verbose(1,Presence information :
${CALLERID(num-pres)} - ${CALLERID(name-pres)} - ${CALLERPRES()})

 

 

Here is my experience with this: whenever "prohib_not_screened" (tested
via a cell phone with hidden caller id info) is sent in the u() value of
the Dial application, pbx_b always gets "allowed_not_screened" as presence
state.Short version: the callerid presence seems lost on the SIP link.
The callerid info isn't, name and number are fine.

 

I'm clearly missing something to pass on the callerid presence state via
the SIP link, but I can't figure out what.

 

Any help or hint would be appreciated.

 

Michael

 

 

 

 

 

 

 

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[asterisk-users] CallerId presence issue

[Mike]

Hi,

 

I've run into a minor snag trying to pass on CALLERID presence from one
Asterisk to another via SIP (both running 13.16.0)

 

I have a PRI coming in PBX_A and PBX_A is connected to PBX_B via SIP.
PBX_A gets PRI calls on a 4 port Digium card, and each call naturally has
its own callerid values and presence.  I pass on those calls to PBX_B via
SI, and I'm trying to pass on this CALLERID info to PBX_B as well. 

 

My relevant dialplan snippet on PBX_A is:

 

exten =>
1,1,Dial(SIP/pbx_b/55,,f(${CALLERID(all)})u(${CALLERID(pres)})))

 

*the u() value being dynamically taken from the channel itself.

 

On pbx_b, I  have a simply verbose line like this:

exten => 55,1,Verbose(1,Presence information :
${CALLERID(num-pres)} - ${CALLERID(name-pres)} - ${CALLERPRES()})

 

 

Here is my experience with this: whenever "prohib_not_screened" (tested
via a cell phone with hidden caller id info) is sent in the u() value of
the Dial application, pbx_b always gets "allowed_not_screened" as presence
state.Short version: the callerid presence seems lost on the SIP link.
The callerid info isn't, name and number are fine.

 

I'm clearly missing something to pass on the callerid presence state via
the SIP link, but I can't figure out what.

 

Any help or hint would be appreciated.

 

Michael

 

 

 

 

 

 

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Re: [asterisk-users] Upgraded server crashes on voicemail storage

[Mike Diehl]

Well, I guess my assumption has been proven wrong.  It is NOT the odbc drive.
I recompiled Asterisk w/o odbc voicemail storage and I'm still getting 
crashes when someone leave voicemail.

I tried to run strace on the server, but didn't get much:

=
voip11 ~ # ps -auxw | grep asterisk
root  9339  0.0  0.0   9604  2548 ?Ss   10:35   0:00 /bin/sh 
/home/phones/commands/safe_asterisk
root  9346 13.1 10.9 104155439880 443704 ? Sl   10:35   0:13 
/usr/sbin/asterisk -v
root  9480  0.0  0.0  12824  2372 pts/11   S+   10:36   0:00 grep 
--colour=auto asterisk
root 11129  0.0  0.2 104153027592 10600 pts/6 S+ Jun07   0:08 rasterisk 
Rv

voip11 ~ # strace -p 9346
strace: Process 9346 attached
restart_syscall(<... resuming interrupted poll ...>

=

So, if I could find out what syscall was being interrupted That MIGHT 
tell me what was wrong, but this is all I get from strace.

Any ideas would be welcome.

Mike.

On Wednesday, June 07, 2017 04:34:10 PM Mike Diehl wrote:
> Thank you for your time.  I've put my replies to your questions in-line, 
> below.
> 
> 
> On Wednesday, June 07, 2017 10:19:41 AM Antony Stone wrote:
> > On Tuesday 06 June 2017 17:54:59 Mike Diehl wrote:
> > 
> > > Hi all,
> > > 
> > > I'm upgrading to Asterisk 13.14.0 x86_64.  During my beta testing, I've
> > > discovered that my server crashes as soon as I leave a voicemail message. 
> > > I'm using odbc voicemail storage as well as mysql dynamic configuration.
> > > 
> > > I'm using unixODBC 2.3.2-r2 with myodbc 5.2.7-r1
> > > 
> > > I suspect that the odbc drivers are the problem.  Is ther an alternative
> > > drive that I should be using?
> > > 
> > > Failing that, any other ideas?
> > 
> > Give us more details of what you mean by "crashes".
> 
> My remote console gets disconnected from the Asterisk server, waits a few 
> seconds, 
> reconnects and shows me the start-up log.  It's just like if you told 
> asterisk to 
> restart now.
> 
> 
> > What happens, what do you get in the Asterisk logs, what do you get in 
> > syslog, 
> > what state is the machine in afterwards, is there a kernel panic, what 
> > information leads you to suspect the ODBC drivers...?
> 
> What I see in the log is:
> 
> 
> [Jun  7 14:23:58] VERBOSE[11347][C-0001] app_dial.c: Everyone is 
> busy/congested at this time (1:0/0/1)
> [Jun  7 14:23:58] VERBOSE[11347][C-0001] res_agi.c: magic_switch.pl: 
> --- jmd (CHANUNAVAIL)
> [Jun  7 14:23:58] VERBOSE[11347][C-0001] res_agi.c: AGI Script Executing 
> Application: (voicemail) Options: (1505903@default,su)
> [Jun  7 14:23:58] VERBOSE[11347][C-0001] file.c: 
>  Playing 
> '/var/spool/asterisk/voicemail/default/15059035700/unavail.slin' (language 
> 'en')
> [Jun  7 14:24:08] VERBOSE[11347][C-0001] file.c: 
>  Playing 'beep.ulaw' (language 'en')
> [Jun  7 14:24:09] VERBOSE[11347][C-0001] app_voicemail.c: Recording the 
> message
> [Jun  7 14:24:09] VERBOSE[11347][C-0001] app.c: x=0, open writing:  
> /var/spool/asterisk/voicemail/default/15059035700/tmp/x8hgQD format: wav, 
> 0x7d380013d750
> [Jun  7 14:24:12] VERBOSE[11347][C-0001] app.c: User ended message by 
> pressing #
> [Jun  7 14:24:12] VERBOSE[11347][C-0001] file.c: 
>  Playing 'auth-thankyou.ulaw' (language 'en')
> [Jun  7 14:24:13] VERBOSE[11347][C-0001] config.c: Parsing 
> '/var/spool/asterisk/voicemail/default/15059035700/INBOX/msg0004.txt': Found
> 
> +++ CRASH! +++
> 
> [Jun  7 14:24:15] Asterisk 13.14.0 built by root @ voip11 on a x86_64 running 
> Linux on 2017-06-06 21:26:05 UTC
> [Jun  7 14:24:15] VERBOSE[11362] config.c: Parsing 
> '/etc/asterisk/logger.conf': Found
> 
> 
> I am thinking it's the odbc driver because I believe the server was stable 
> before 
> I rebuilt it with odbc voicemail storage support; it had been using the file 
> system
> for storage.  I'm in the process of migrating all of my servers to database 
> storage.
> 
> 
> 
> > Also, what have you upgraded from, what machine specs are you running on, 
> > what's the dialplan section dealing with leaving voicemail...?
> 
> The ONLY thing I changed from the previous configuration was to convert to 
> odbc voicemail 
> storage.
> 
> > The more info you give us, the more likely it is we can suggest something 
> > useful.
> 
> Ya, I understand; I was just tired... and frustrated.  Thanks 

Re: [asterisk-users] Upgraded server crashes on voicemail storage

[Mike Diehl]

Thank you for your time.  I've put my replies to your questions in-line, below.


On Wednesday, June 07, 2017 10:19:41 AM Antony Stone wrote:
> On Tuesday 06 June 2017 17:54:59 Mike Diehl wrote:
> 
> > Hi all,
> > 
> > I'm upgrading to Asterisk 13.14.0 x86_64.  During my beta testing, I've
> > discovered that my server crashes as soon as I leave a voicemail message. 
> > I'm using odbc voicemail storage as well as mysql dynamic configuration.
> > 
> > I'm using unixODBC 2.3.2-r2 with myodbc 5.2.7-r1
> > 
> > I suspect that the odbc drivers are the problem.  Is ther an alternative
> > drive that I should be using?
> > 
> > Failing that, any other ideas?
> 
> Give us more details of what you mean by "crashes".

My remote console gets disconnected from the Asterisk server, waits a few 
seconds, 
reconnects and shows me the start-up log.  It's just like if you told asterisk 
to 
restart now.


> What happens, what do you get in the Asterisk logs, what do you get in 
> syslog, 
> what state is the machine in afterwards, is there a kernel panic, what 
> information leads you to suspect the ODBC drivers...?

What I see in the log is:


[Jun  7 14:23:58] VERBOSE[11347][C-0001] app_dial.c: Everyone is 
busy/congested at this time (1:0/0/1)
[Jun  7 14:23:58] VERBOSE[11347][C-0001] res_agi.c: magic_switch.pl: 
--- jmd (CHANUNAVAIL)
[Jun  7 14:23:58] VERBOSE[11347][C-0001] res_agi.c: AGI Script Executing 
Application: (voicemail) Options: (1505903@default,su)
[Jun  7 14:23:58] VERBOSE[11347][C-0001] file.c: 
 Playing 
'/var/spool/asterisk/voicemail/default/15059035700/unavail.slin' (language 'en')
[Jun  7 14:24:08] VERBOSE[11347][C-0001] file.c: 
 Playing 'beep.ulaw' (language 'en')
[Jun  7 14:24:09] VERBOSE[11347][C-0001] app_voicemail.c: Recording the 
message
[Jun  7 14:24:09] VERBOSE[11347][C-0001] app.c: x=0, open writing:  
/var/spool/asterisk/voicemail/default/15059035700/tmp/x8hgQD format: wav, 
0x7d380013d750
[Jun  7 14:24:12] VERBOSE[11347][C-0001] app.c: User ended message by 
pressing #
[Jun  7 14:24:12] VERBOSE[11347][C-0001] file.c: 
 Playing 'auth-thankyou.ulaw' (language 'en')
[Jun  7 14:24:13] VERBOSE[11347][C-0001] config.c: Parsing 
'/var/spool/asterisk/voicemail/default/15059035700/INBOX/msg0004.txt': Found

+++ CRASH! +++

[Jun  7 14:24:15] Asterisk 13.14.0 built by root @ voip11 on a x86_64 running 
Linux on 2017-06-06 21:26:05 UTC
[Jun  7 14:24:15] VERBOSE[11362] config.c: Parsing '/etc/asterisk/logger.conf': 
Found


I am thinking it's the odbc driver because I believe the server was stable 
before 
I rebuilt it with odbc voicemail storage support; it had been using the file 
system
for storage.  I'm in the process of migrating all of my servers to database 
storage.



> Also, what have you upgraded from, what machine specs are you running on, 
> what's the dialplan section dealing with leaving voicemail...?

The ONLY thing I changed from the previous configuration was to convert to odbc 
voicemail 
storage.

> The more info you give us, the more likely it is we can suggest something 
> useful.

Ya, I understand; I was just tired... and frustrated.  Thanks again for your 
time.


-- 
Mike Diehl
Diehlnet Communications, LLC. 
Sales: (800) 254-6105   
Support: (505) 903-5700 
Fax: (505) 903-5701  


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[asterisk-users] Upgraded server crashes on voicemail storage

[Mike Diehl]

Hi all,

I'm upgrading to Asterisk 13.14.0 x86_64.  During my beta testing, I've 
discovered that my server crashes as soon as I leave a voicemail message.  I'm 
using odbc voicemail storage as well as mysql dynamic configuration.

I'm using unixODBC 2.3.2-r2 with myodbc 5.2.7-r1

I suspect that the odbc drivers are the problem.  Is ther an alternative drive 
that I should be using?  

Failing that, any other ideas?

Thanks in advance.

-- 
Mike Diehl



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Re: [asterisk-users] Asterisk 13 queue and DND phones

[Mike]

This makes sense, thank you, although this is applicable to Polycom phones 
only (I was hoping for a more universal solution, as current phones are not 
an indicator of phones we may get in the future)



Mike





From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Héctor Royo
Sent: May 17, 2017 08:55
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 13 queue and DND phones



Hi. I will try to give you an idea:

You can remap de 'Do not disturb' key to do some actions.


(Best examples I've found so far: 
http://community.polycom.com/t5/VoIP/FAQ-Using-Enhanced-Feature-Keys-EFK-macros-to-change-key/td-p/5705)



I once tried to remap de DND on a Polycom IP 550 to something like:



key.SPIP550.9.function.prim="^PauseExten$Tinvite$FDoNotDisturb$"



That would place a call to extension "PauseExten" and also sets the phone on 
DND.



The extension 'PauseExten' has the dialplan logic to Pause/Unpause the 
phone. And this executes everytime you press de DND key.



This "works" but it is not the best solution.



Hope it helps. Sorry for my bad english.









2017-05-17 13:22 GMT+01:00 Mike <mich...@virtutel.ca>:

Hi,



I’ve noticed that when I set a phone on DND (phone-side DND, meaning it 
rejects calls with a busy status, SIP 486 response code I believe) the queue 
keeps on trying the phone over and over again.



This creates issues in terms of CDR entries – in a scenario where there is 
only one phone on DND, and a delay between attempts of 1 second, the queue 
will attempt to ring the single phone every second, creating one CDR entry 
per second.  It also creates one event per second in the queue_log file.



Is there any strategy to avoid this? I’m trying to avoid autopause=yes 
(because them the employee needs to take action when turned DND off -  I 
want the simple act of turning DND off to mean the phone starts ringing 
again).



-  Can I have an autoUNpause after x seconds?

-  Can the queue detect that the phone is returning response code 
486, and not ring it for either x second or until the next call?

-  Can the CDR engine/queue_log engine be told to not log more than 
one RINGNOANSWER status? Having one helps, as it tells us the agent was on 
DND at the time, but having hundreds does not help anyone.



Specifically this is with Polycom phones, but I don’t think it makes a 
difference.



Thank you for taking the time to help me,





Mike






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Tfno: 922 845401  Móvil: 628 568773

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[asterisk-users] Asterisk 13 queue and DND phones

[Mike]

Hi,

 

I've noticed that when I set a phone on DND (phone-side DND, meaning it
rejects calls with a busy status, SIP 486 response code I believe) the
queue keeps on trying the phone over and over again.

 

This creates issues in terms of CDR entries - in a scenario where there is
only one phone on DND, and a delay between attempts of 1 second, the queue
will attempt to ring the single phone every second, creating one CDR entry
per second.  It also creates one event per second in the queue_log file.

 

Is there any strategy to avoid this? I'm trying to avoid autopause=yes
(because them the employee needs to take action when turned DND off -  I
want the simple act of turning DND off to mean the phone starts ringing
again).

 

-  Can I have an autoUNpause after x seconds?

-  Can the queue detect that the phone is returning response code
486, and not ring it for either x second or until the next call?

-  Can the CDR engine/queue_log engine be told to not log more
than one RINGNOANSWER status? Having one helps, as it tells us the agent
was on DND at the time, but having hundreds does not help anyone.

 

Specifically this is with Polycom phones, but I don't think it makes a
difference.

 

Thank you for taking the time to help me,

 

 

Mike

 

 

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Re: [asterisk-users] 100% CPU after upgrade. (Solved)

[Mike Diehl]

I had meant to post a follow up to this, but just... didn't.  Sorry.

Anyway, I had made a silly change to my safe_asterisk script that caused it to 
start asterisk in the background, but also with a console.  This caused 
asterisk to try to write to a non-existent console tty.

Dumb mistake on my part.

Hope this helps someone else.

Mike.


On Thursday, April 06, 2017 10:28:03 AM you wrote:
> On Thu, Apr 6, 2017 at 10:20 AM, Mike Diehl <mdiehlena...@gmail.com> wrote:
> > I found it!
> >
> > I had customized the safe_asterisk script and managed to slip in a -c on 
the asterisk command line.
> >
> > So, when I ran strace on the running process, I saw a bunch of messages 
indicating an invalid IOCTL on file handle 1, which is always STDOUT.  A 
background process shouldn't be writing to STDOUT, so I knew I had dorked 
something up.
> >
> > I appreciate your time.
> >
> 
> Thanks so much for letting me know.  Would you mind posting this
> resolution publicly so that anybody following it can learn from what
> happened?
> 
> Best wishes,
> Matthew Fredrickson
> 
> > Mike.
> >
> >
> >
> > On Tuesday, April 04, 2017 09:18:26 AM you wrote:
> >> On Mon, Apr 3, 2017 at 4:45 PM, Mike Diehl <mdiehlena...@gmail.com> 
wrote:
> >> > Those are all rational questions, so here we go:
> >> >
> >> > We upgraded from 11.x, though the system was a backup server, so it was 
never
> >> > actually used.
> >> >
> >> > The system is a 2.4Gh quad-core Xenon with 4G of RAM, so it should have 
plenty
> >> > of power for what I'm asking it to do.  The system is configured via RT 
using
> >> > a local Mysql database.
> >> >
> >> > We only use the native SIP channel driver at this time.
> >> >
> >> > I honestly don't see any reason for this server to eat 100% of it's 
cpu, and
> >> > am hesitant to roll it out to production until I understand why it is.
> >>
> >> I don't either.  Is there any Asterisk logging that indicates
> >> something that might be going on?  If you can't see anything, try
> >> increasing the core debug level and core verbose level (core set
> >> verbose 10, core set debug 10) at the Asterisk CLI and see if you get
> >> anything more out of logging to see what's going on.
> >>
> >>
> >
> > --
> > Mike Diehl
> > Diehlnet Communications, LLC.
> > Sales: (800) 254-6105
> > Support: (505) 903-5700
> > Fax: (505) 903-5701
> >
> 
> 
> 
> 

-- 
Mike Diehl



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Re: [asterisk-users] asterisk-users Digest, Vol 152, Issue 31

[Mike Codjoe]

Dear Saint Michael,

I will be grateful if you could introduce me to the Company that
offers the translation service.

I am really interested in google voice.

Sincerely,

Michael Codjoe

On 29 March 2017 at 17:00,   wrote:
> Send asterisk-users mailing list submissions to
> asterisk-users@lists.digium.com
>
> To subscribe or unsubscribe via the World Wide Web, visit
> http://lists.digium.com/mailman/listinfo/asterisk-users
> or, via email, send a message with subject or body 'help' to
> asterisk-users-requ...@lists.digium.com
>
> You can reach the person managing the list at
> asterisk-users-ow...@lists.digium.com
>
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of asterisk-users digest..."
>
>
> Today's Topics:
>
>1. Bounty on Google Voice (Saint Michael)
>
>
> --
>
> Message: 1
> Date: Wed, 29 Mar 2017 12:45:16 -0400
> From: Saint Michael 
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> 
> Subject: [asterisk-users] Bounty on Google Voice
> Message-ID:
> 

Re: [asterisk-users] 100% CPU after upgrade.

[Mike Diehl]

Those are all rational questions, so here we go:

We upgraded from 11.x, though the system was a backup server, so it was never 
actually used.

The system is a 2.4Gh quad-core Xenon with 4G of RAM, so it should have plenty 
of power for what I'm asking it to do.  The system is configured via RT using 
a local Mysql database.

We only use the native SIP channel driver at this time.

I honestly don't see any reason for this server to eat 100% of it's cpu, and 
am hesitant to roll it out to production until I understand why it is.

Once again, any suggestions will be welcome.

Thanks,

Mike Diehl.

On Friday, March 31, 2017 01:51:07 PM Matt Fredrickson wrote:
> One thing you didn't mention was what version you previously upgraded
> from...  Also, more information about the system in general would
> help.  (Endpoints, is it realtime or flat file configured, if
> realtime, what type of database, what channel drivers (SIP or PJSIP,
> and others).
> 
> Matthew Fredrickson
> 
> On Fri, Mar 31, 2017 at 12:08 PM, Mike Diehl <mdiehlena...@gmail.com> wrote:
> > Hi all,
> >
> > I've upgraded to Asterisk 13.14.0 and now I'm seeing that Asterisk is 
using 100% CPU.
> >
> > I have one AMI agent connected that is acting rationally.  I've got a hand 
full of SIP (RT) registrations.  There is no other call activity.
> >
> > I've tried to unload various modules; nothing resolved the issue.
> >
> > Any suggestions?
> >
> > --
> > Mike Diehl
> >
> >
> >
> >
> > --
> > _
> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> >
> > Check out the new Asterisk community forum at: 
https://community.asterisk.org/
> >
> > New to Asterisk? Start here:
> >   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> 
> 
> 

-- 
Mike Diehl



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Re: [asterisk-users] Issue with Asterisk 13, multiple CDR per queue and arbitrary upper limit

[Mike]

I'm going to answer my own question, since someone may one day have a 
similar issue.

I had some default Asterisk 13 settings in place, so my CDR's were written 
simultaneously  to csv, sqlite3 and Mysql.

Once I removed those I did  not use, the CDR's were dumped into the Mysql DB 
in the blink of an eye, with no impact whatsoever to Asterisk. I do not know 
(nor care at this point) whether the CSV was the issue of sqlite3 but one 
(or both) of them must have been slowing things down and created the issue.

Regards,

Mike






-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michaël 
Gaudette
Sent: April 2, 2017 1:45 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Issue with Asterisk 13, multiple CDR per queue 
and arbitrary upper limit

Joshua,

After playing with this issue I'm starting to think this has little to do
with the 5000 limit - at least not directly.

The amount of CDR entries to be written to the DB is just too high for
either Asterisk or the Database to keep up, and it possibly creates issues
around DB access (on which my Asterisk dialplan relies). Is there any way to
"slow down" the writing of all the CDR entries? Or, on the contrary, to have
the CDR entries be flushed at every 100 "entries to be written" instead of
5000, so that the hit is relatively small?

Regards,

----
Mike

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joshua Colp
Sent: April 1, 2017 6:56 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Issue with Asterisk 13, multiple CDR per queue
and arbitrary upper limit

On Fri, Mar 31, 2017, at 10:55 PM, Michaël Gaudette wrote:
>
>
> Hi,
>
>
>
> I`ve recently upgraded a server from 1.8 to Asterisk 13.  While
> everything
> is under control, I have one issue with the way CDRs are kept for queues.
> And I don`t mean “I don`t like it”. I mean it crashes the server.
>
>
>
> I realize there are multiple CDRs per queue call – one per ring/per
> phone,
> basically.  The issue is that whenever the number of CDRs “to  be
> recorded” for a call exceeds 5000, Asterisk becomes unresponsive for a
> few
> minute. I get this message in the console:
>
> “taskprocessor_push: The 'subm:cdr_engine-0003' task processor queue
> reached 5000 scheduled tasks again.”
>
>
>
> This scenario is trivial to reproduce: a queue, with simultaneous ring,
> 20
> phones, all unreachable, 1 second between attempts.  After 250 (5000
> divided by 20) seconds of waiting asterisk partially breaks down.
>
>
>
> This seems to be because while multiple CDR`s are written per queue call,
> it`s only done at the  end of the call, so CDRs accumulate in
> memory/cacher/whatever and break some limit.
>
>
>
> So, my question is:  is there any way to force the CDR`s to be written as
> the queue app is working it`s magic, instead of at the very end of the
> call? Or anyway to work around this limit? Or any fix for this?

There is not. If you are running the latest version I'd suggest filing
an issue[1] as we definitely should not crash under the scenario.

[1] https://issues.asterisk.org/jira

-- 
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org

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[asterisk-users] 100% CPU after upgrade.

[Mike Diehl]

Hi all,

I've upgraded to Asterisk 13.14.0 and now I'm seeing that Asterisk is using 
100% CPU.  

I have one AMI agent connected that is acting rationally.  I've got a hand full 
of SIP (RT) registrations.  There is no other call activity.

I've tried to unload various modules; nothing resolved the issue.

Any suggestions?

-- 
Mike Diehl




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[asterisk-users] Asterisk/FFA version upgrade recommendation

[Mike Diehl]

Hi all,

I'm needing to upgrade Asterisk from 10.x to whatever the recommended version 
is that will allow me to continue to use Fax For Asterisk.

I don't have many upgrade windows, I'd like to get the most bang for my buck, 
but I can't afford to be a beta tester on this server.

The FFA site says that it's supported by Asterisk version 12 and lower, but 
version 12 doesn't seem to be supported.  Perhaps my information is 
outdated?

Anyway, I can't go with the spandsp route because my system listens for AMA 
events that spandsp doesn't seem to produce and I can't emulate easily.

Any recommendations would be very welcome.

-- 
Mike Diehl




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[asterisk-users] Asterisk/FFA version upgrade recommendation

[Mike Diehl]

Hi all,

I'm needing to upgrade Asterisk from 10.x to whatever the recommended version 
is that will allow me to continue to use Fax For Asterisk.

I don't have many upgrade windows, I'd like to get the most bang for my buck, 
but I can't afford to be a beta tester on this server.

The FFA site says that it's supported by Asterisk version 12 and lower, but 
version 12 doesn't seem to be supported.  Perhaps my information is 
outdated?

Anyway, I can't go with the spandsp route because my system listens for AMA 
events that spandsp doesn't seem to produce and I can't emulate easily.

Any recommendations would be very welcome.

-- 
Mike Diehl



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Re: [asterisk-users] iptables for SIP talk to other port

[Mike]

I'm by no means an iptables guru...

Not sure if it's necessary to enable forwarding via:
echo "1" > /proc/sys/net/ipv4/ip_forward

Also have you tried without the "POSTROUTING" rule?

I seem to recall that "iptables" is smart enough to correctly route 
packets back out without that rule.



On Sat, 15 Oct 2016, Jerry Geis wrote:


I have a host 192.168.1.3 that wants to run SIP on 5068 (long story).My host is 
192.168.10.201.
My host needs to stay on 5060 because of all the other devices I have connected.

I tried putting port=5068 in my SIP extension definition but that did not work.

So I thought about using iptables to accomplish this:

iptables -t nat -A PREROUTING  -p tcp --dport 5068                -j REDIRECT 
--to-port 5060
iptables -t nat -A POSTROUTING -p tcp --dport 5060 -d 192.168.1.3 -j REDIRECT 
--to-port 5068


Do I not have the right format of the command?
Anything incoming destined for 5068 redirect to 5060...
Anything going out to 192.168.1.3 and port 5060 redirect to 5068.

Seems like that should have worked?

Thoughts?  sip show peers still says unreachable.

Thanks,

Jerry

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[asterisk-users] SPA112 flapping

[Mike Diehl]

Hi all,

I've got a device that seems to become unreachable for about 2 minutes, every 
hour.  From what I can tell, it isn't due to network or server issues.  Any 
ideas?

TIA.


-- 
Mike Diehl
Diehlnet Communications, LLC.
Voice: (505) 903-5700
Fax: (505) 903-5701


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Re: [asterisk-users] confbridge setup

[Mike Diehl]

Hello,

Thanks for getting back to me.  I didn't know that the conferences wouldn't 
show up on the list until they were "active;"  I thought that was meant to show 
the defined conferences.

However, when I try to dial into the conference room that I (think) have 
defined, I see:

-- [Apr 18 14:33:27] -- AGI Script Executing Application: (ConfBridge) 
Options: (5340)
[Apr 18 14:33:27] ERROR[3048][C-b16d]: app_confbridge.c:1201 
join_conference_bridge: Conference '5340' mixing bridge could not be created.

What do I need to do in order for a mixing bridge to be created?

Thanks again,

Mike.

On Saturday, April 16, 2016 04:18:44 PM Bobby Hakimi wrote:
> You can't see them until someone joins the bridge, might be able to put in
> db using the asterisk live setup
> 
> On Apr 16, 2016 1:36 PM, "Mike Diehl" <mdiehlena...@gmail.com> wrote:
> > Hi all,
> > 
> > I'm trying to configure a few conference bridges.  I've started with the
> > very
> > basic:
> > 
> > [general]
> > 
> > [default_bridge]
> > type=bridge
> > 
> > [default_user]
> > type=user
> > 
> > [default_bridge]
> > type=bridge
> > 
> > [5340]
> > type=bridge
> > 
> > 
> > However:
> > 
> > confbridge list
> > Conference Bridge Name   Users  Marked Locked?
> >  == == 
> > *CLI>
> > 
> > 
> > It doesn't seem to be creating any bridges and I'm sure I've left
> > something
> > basic out.  What am I missing?
> > 
> > Finally, once I've got this working, can all of this be put into a
> > database?
> > If so, what table structure do I use?
> > 
> > Thanks in advance,
> > 
> > 
> > --
> > Mike Diehl
> > Diehlnet Communications, LLC.
> > Voice: (505) 903-5700
> > Fax: (505) 903-5701
> > 
> > 
> > --
> > _
> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> > 
> > New to Asterisk? Join us for a live introductory webinar every Thurs:
> >http://www.asterisk.org/hello
> > 
> > asterisk-users mailing list
> > 
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users

-- 
Mike Diehl
Diehlnet Communications, LLC.
Voice: (505) 903-5700
Fax: (505) 903-5701


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[asterisk-users] confbridge setup

[Mike Diehl]

Hi all,

I'm trying to configure a few conference bridges.  I've started with the very 
basic:

[general]

[default_bridge]
type=bridge

[default_user]
type=user

[default_bridge]
type=bridge

[5340]
type=bridge


However:

confbridge list
Conference Bridge Name   Users  Marked Locked?
 == == 
*CLI> 


It doesn't seem to be creating any bridges and I'm sure I've left something 
basic out.  What am I missing?

Finally, once I've got this working, can all of this be put into a database?  
If so, what table structure do I use?

Thanks in advance,


-- 
Mike Diehl
Diehlnet Communications, LLC.
Voice: (505) 903-5700
Fax: (505) 903-5701


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Re: [asterisk-users] ODBC crashing asterisk

[Mike Diehl]

Normally, SQL errors don't result in a segfault.  I understand that this is a 
problem with a particular version of the ODBC driver.  I just can't find a 
reference to it at the moment.

On Thursday, March 24, 2016 09:54:35 AM Антон Сацкий wrote:
> You have an error in your SQL syntax; check the manual that corresponds
> 
> On Mar 23, 2016 11:38 PM, "Mike Diehl" <mdiehlena...@gmail.com> wrote:
> > Hi all,
> > 
> > I've got a new server up, but it's not staying up
> > 
> > After a day or so, it segfaults with:
> > 
> > [Mar 22 23:17:49] WARNING[12177]: res_odbc.c:1406 _ast_odbc_request_obj2:
> > SetConnectAttr (Txn isolation) returned an error: HY000: [MySQL][ODBC
> > 5.2(a)
> > Driver]You have an error in your SQL syntax; check the manual that
> > corresponds
> > to your MySQL server version for the right syntax to use near '7' at line
> > 1
> > 
> > 
> > I'm using ODBC for sip and voice mail configuration.
> > 
> > I'm running Asterisk 11.20.0-rc3.
> > 
> > I've been told that there is a particular version of odbc that is stable.
> > In
> > the mean time, I'm trying to run unixODBC 2.3.2.
> > 
> > What version SHOULD I use?
> > 
> > TIA,
> > 
> > 
> > --
> > Mike Diehl
> > Diehlnet Communications, LLC.
> > Voice: (505) 903-5700
> > Fax: (505) 903-5701
> > 
> > 
> > --
> > _
> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> > 
> > New to Asterisk? Join us for a live introductory webinar every Thurs:
> >http://www.asterisk.org/hello
> > 
> > asterisk-users mailing list
> > 
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users

-- 
Mike Diehl
Diehlnet Communications, LLC.
Voice: (505) 903-5700
Fax: (505) 903-5701


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[asterisk-users] Can't create confbridge

[Mike Diehl]

Hi all,

I'm trying to get mod_confbridge working from an AGI script.  When I dial the 
appropriate extension, I get:

[Mar 24 17:10:08] ERROR[14310][C-0019]: app_confbridge.c:1201 
join_conference_bridge: Conference '1505xxx' mixing bridge could not be 
created.

The AGI script looks, essentially, like:

$main::agi->exec("ConfBridge","1505xxx");


I've got a dummy /etc/asterisk/confbridge.conf file:

[general]

[default_bridge]
type=bridge

[default_user]
type=user

[default_bridge]
type=bridge

[1505xxx]
type=bridge


Any suggestions would be welcome.

-- 
Mike Diehl
Diehlnet Communications, LLC.
Voice: (505) 903-5700
Fax: (505) 903-5701


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[asterisk-users] ODBC crashing asterisk

[Mike Diehl]

Hi all,

I've got a new server up, but it's not staying up

After a day or so, it segfaults with:

[Mar 22 23:17:49] WARNING[12177]: res_odbc.c:1406 _ast_odbc_request_obj2: 
SetConnectAttr (Txn isolation) returned an error: HY000: [MySQL][ODBC 5.2(a) 
Driver]You have an error in your SQL syntax; check the manual that corresponds 
to your MySQL server version for the right syntax to use near '7' at line 1


I'm using ODBC for sip and voice mail configuration.

I'm running Asterisk 11.20.0-rc3.

I've been told that there is a particular version of odbc that is stable.  In 
the mean time, I'm trying to run unixODBC 2.3.2.

What version SHOULD I use?

TIA,


-- 
Mike Diehl
Diehlnet Communications, LLC.
Voice: (505) 903-5700
Fax: (505) 903-5701


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[asterisk-users] spa112 can't get line 2 to register

[Mike Diehl]

Hi all,

I've got a couple of SPA112's that are having problems registering line 2.  
Line 1 registers just fine.  All of them are behind a NAT.

Here is a sample provisioning file that the devices are using.

(Any help would be most appreciated.)




Yes

xxx



syslog.example.com

3
Yes
Yes
2
3600
3600
300
Yes
Yes


Yes
 http://example.com/index.pl?mac=$MA 





 $PN $MAC -- Requesting resync $SCHEME://$SERVIP:
$PORT$PATH 
 $PN $MAC -- Successful resync $SCHEME://$SERVIP:
$PORT$PATH 
$PN $MAC -- Resync failed: 
$ERR


Yes
3600

 http://example.com/fw/spa112-132.bin 
 $PN $MAC -- Requesting upgrade $SCHEME://$SERVIP:
$PORT$PATH 
 $PN $MAC -- Successful upgrade $SCHEME://$SERVIP:
$PORT$PATH -- $ERR 
$PN $MAC -- Upgrade failed: 
$ERR






















70
5
2
$VERSION
$VERSION


application>dtmf-relay
application>hook-flash
No
No
No
Yes
Yes
5060
5080
.5
4
5
32
16
32
32
32
240
30
1
30
30
1200
5
0
0






16384
16482
0.030
0
0
No
Yes
100
101

2
112
113
G711u
NSE
telephone-event
PCMU
PCMA
G726-32
G729a
G722
encaprtp
No
No
No
No
No
No
No
No



15
Yes
No
30

no
yes
$NOTIFY
$PROXY
0x68
3
0xb8
6
high
Yes
UDP
No

No

Yes
No
none
0
No
4
0
0
No
No
No
No
Yes
No
No

Yes

3
No


example.com
example.com
No
Yes
Yes
No
 7200 
No
no
no
3600
Normal


2147483647

None
username-1
username-1
secret
Yes
username-1

yes
Yes
Yes
Yes
No
No
No
No
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
No
No
Yes
Yes
Yes
Yes
Yes
Yes
No

G711u
Unspecified
Unspecified
No
Yes
No
Yes
medium
Yes
Yes
Yes
Yes
G711u
Yes
Yes
Yes
ReINVITE
Yes
No
Auto
Yes
Strict
70
Yes
None
1
Yes
caller or callee
No
Yes
 (*xx|[3469]11|0|00|[2-9]xx|1xxx[2-9]xxS0|.) 

5000
Forward
Forward
Forward
Yes
No
30

no
yes
$NOTIFY
$PROXY
0x68
3
0xb8
6
high
Yes
UDP
No

No

Yes
No
none
0
No
4
0
0
No
No
No
No
Yes
No
No

Yes

3
No


example.com
example.com
No
Yes
Yes
No
 7200 
No
NO
NO
3600
Normal


2147483647

None
username-2
username-2
secret
Yes
username-2

yes
Yes
Yes
Yes
No
No
No
No
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
No
G711u
Unspecified
Unspecified
No
Yes
No
Yes
medium
Yes
Yes
Yes
Yes
G711u
Yes
Yes
Yes
ReINVITE
Yes
No
Auto
Yes
Strict
70
Yes
None
1
Yes
caller or callee
No
Yes
 (*xx|[3469]11|0|00|[2-9]xx|1xxx[2-9]xxS0|.) 

5000
Forward
Forward
Forward
No 


20




























Yes
No
No
No
Yes
Yes
Yes
No
No
automatic
source
media








1
1
8
7
0
0
New VM Available
0
No
No 


20




























Yes
No
No
No
Yes
Yes
Yes
No
No
automatic
source
media








1
1
8
7
0
0
New VM Available
0
No
350@-19,440@-19;10(*/0/1+2)
420@-19,520@-19;10(*/0/1+2)
420@-16;10(*/0/1)
520@-19,620@-19;10(*/0/1+2)
480@-19,620@-19;10(.5/.5/1+2)
480@-19,620@-19;10(.25/.25/1+2)
480@-10,620@0;10(.125/.125/1+2)
440@-19,480@-19;*(2/4/1+2)
440@-19,480@-19;*(1/1/1+2)
600@-16;1(.25/.25/1)

985@-16,1428@-16,1777@-16;20(.380/0/1,.380/0/2,.380/0/3,0/4/0)


914@-16,1371@-16,1777@-16;20(.274/0/1,.274/0/2,.380/0/3,0/4/0)


914@-16,1371@-16,1777@-16;20(.380/0/1,.380/0/2,.380/0/3,0/4/0)


985@-16,1371@-16,1777@-16;20(.380/0/1,.274/0/2,.380/0/3,0/4/0)

350@-19,440@-19;2(.1/.1/1+2);10(*/0/1+2)
350@-19,440@-19;2(.2/.2/1+2);10(*/0/1+2)
600@-19;*(.1/.1/1,.1/.1/1,.1/9.5/1)
350@-19;20(.1/.1/1,.1/9.7/1)
397@-19,507@-19;15(0/2/0,.2/.1/1,.1/2.1/2)
350@-16;*(.1/.1/1)
60(2/4)
60(.8/.4,.8/4)
60(.4/.2,.4/.2,.8/4)
60(.3/.2,1/.2,.3/4)
1(.5/.5)
60(.2/.4,.2/.4,.2/4)
60(.4/.2,.4/.2,.4/4)
60(0.25/9.75)
30(.3/9.7)
30(.1/.1, .1/9.7)
30(.1/.1, .1/.1, .1/9.7)
30(.1/.1,.3/.1,.1/9.3)
1(.5/.5)
30(.1/.1,.3/.2,.3/9.1)
30(.3/.1,.3/.1,.1/9.1)
2.3(.3/2)
Bellcore-r1
Bellcore-r2
Bellcore-r3
Bellcore-r4
Bellcore-r5
Bellcore-r6
Bellcore-r7
Bellcore-r8
Trapezoid
20
85
440@-10
No
.1
.9
0
5
1800
30
.5
0
10
3
2
.5
*69
*07
*98
*66
*86
*05
*72
*73
*90
*91
*92
*93
*63
*83
*60
*80
*64
*84
*56
*57
*71
*70
*67
*68
*81
*82
*77
*87
*78
*79
*65
*85
*25
*45
*26
*46
*74
*96
*16
*17
*18
*19


*99
#99
*03




*017110
*027110
*017111
*027111
*0172632
*0272632
*01729
*02729
*01722
*02722


GMT-07:00
600
-3
-3
-16
.1
Yes
Yes
Bellcore(N.Amer,China)
bell 202
Default
No
No


dh
0
0
0.0.0.0:0.0.0.0:0.0.0.0
0
0



0:5:30
0
0


1
auto


0



username




0
1


0
0.0.0.0/0.0.0.0
public
private
0
v3rwuser
MD5
11
DES
11


-08 1 1
1
auto

3600
0
1



0
1




0
0.0.0.0

1

1
0

1


1
1
1


1
0
1
0
1
1
0.0.0.0 0
80


0






86400
1
0


0
200
syslog.example.com
514




25
100
60



0
3
0
0
0


0
3
0
0
0


0
admin

cisco









-- 
Mike Di


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[asterisk-users] WaitForSilence NEVER detects silence,,Post

[Mike A. Leonetti]

I have a call server that runs on a few custom AGI scripts initiating 
calls and then managing the calls. I'm getting stuck on the detecting 
silence functions. I wanted to use the silence detecting as a quick 
method of substituting Answering Machine Detection.


However, whenever WaitForSilence is supposed to be detecting silence, it 
always just ends the interval whether or not there is actual silence. If 
I'm still talking, it will consider that as silence also.


My AEL dialplan associated with the calling is:

100 = {
Answer();
WaitForSilence(5000,2,60);
AGI(agi://127.0.0.1/playmessage,${CALLID});
AGI(agi://127.0.0.1/saytext,Goodbye.);
Hangup();
}



And the CLI just outputs:

== Using SIP RTP CoS mark 5
 Channel SIP/twilio-006e was answered
-- Executing [100@makeCall:1] Answer(SIP/twilio-006e, ) in 
new stack
-- Executing [100@makeCall:2] 
WaitForSilence(SIP/twilio-006e, 5000,2,60) in new stack

-- Waiting 2 time(s) for 5000 ms silence with 60 timeout
-- Exiting with 5000ms silence = 5000ms required
-- Exiting with 5000ms silence = 5000ms required
-- Executing [100@makeCall:3] AGI(SIP/twilio-006e, 
agi://127.0.0.1/playmessage,45) in new stack
-- Playing '/var/nam/data/outgoing/60' (escape_digits=#) 
(sample_offset 0)
 0x7f2179cf7990 -- Probation passed - setting RTP source address 
to 54.172.61.251:18920
-- Playing '/var/nam/data/tts/9eccb3f2ed77972157becdfbbac7232c' 
(escape_digits=1#) (sample_offset 0)
-- SIP/twilio-006eAGI Script agi://127.0.0.1/playmessage 
completed, returning 4
== Spawn extension (makeCall, 100, 3) exited non-zero on 
'SIP/twilio-006e'



In my test above, it waits for 5 seconds of silence twice, but even if 
I'm talking for the 5 seconds it will still just figure that I'm being 
silent when I'm not.


I also tried using AMD to see if that would do a good job of detecting 
an answering machine, but it thinks that everything is a MACHINE.


I know Asterisk is knowledgeable abut detecting silence because I have 
another AGI script that uses the RECORD FILE command that will 
successfully record somebody's voice and stop recording when there is 5 
seconds of silence (which is what I set).


Is there a setting somewhere that I'm missing somewhere for a silence 
threshold for WaitForSilence or am I misunderstanding its use?


The Asterisk version is

Asterisk 11.7.0~dfsg-1ubuntu1


And it's Asterisk installed from an Ubuntu package.

Thanks so much!

--
Mike A. Leonetti
As warm as green tea


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[asterisk-users] WaitForSilence NEVER detects silence

[Mike A. Leonetti]

I have a call server that runs on a few custom AGI scripts initiating 
calls and then managing the calls. I'm getting stuck on the detecting 
silence functions. I wanted to use the silence detecting as a quick 
method of substituting Answering Machine Detection.


However, whenever WaitForSilence is supposed to be detecting silence, it 
always just ends the interval whether or not there is actual silence. If 
I'm still talking, it will consider that as silence also.


My AEL dialplan associated with the calling is:

100 = {
Answer();
WaitForSilence(5000,2,60);
AGI(agi://127.0.0.1/playmessage,${CALLID});
AGI(agi://127.0.0.1/saytext,Goodbye.);
Hangup();
}



And the CLI just outputs:

== Using SIP RTP CoS mark 5
 Channel SIP/twilio-006e was answered
-- Executing [100@makeCall:1] Answer(SIP/twilio-006e, ) in 
new stack
-- Executing [100@makeCall:2] 
WaitForSilence(SIP/twilio-006e, 5000,2,60) in new stack

-- Waiting 2 time(s) for 5000 ms silence with 60 timeout
-- Exiting with 5000ms silence = 5000ms required
-- Exiting with 5000ms silence = 5000ms required
-- Executing [100@makeCall:3] AGI(SIP/twilio-006e, 
agi://127.0.0.1/playmessage,45) in new stack
-- Playing '/var/nam/data/outgoing/60' (escape_digits=#) 
(sample_offset 0)
 0x7f2179cf7990 -- Probation passed - setting RTP source address 
to 54.172.61.251:18920
-- Playing '/var/nam/data/tts/9eccb3f2ed77972157becdfbbac7232c' 
(escape_digits=1#) (sample_offset 0)
-- SIP/twilio-006eAGI Script agi://127.0.0.1/playmessage 
completed, returning 4
== Spawn extension (makeCall, 100, 3) exited non-zero on 
'SIP/twilio-006e'



In my test above, it waits for 5 seconds of silence twice, but even if 
I'm talking for the 5 seconds it will still just figure that I'm being 
silent when I'm not.


I also tried using AMD to see if that would do a good job of detecting 
an answering machine, but it thinks that everything is a MACHINE.


I know Asterisk is knowledgeable abut detecting silence because I have 
another AGI script that uses the RECORD FILE command that will 
successfully record somebody's voice and stop recording when there is 5 
seconds of silence (which is what I set).


Is there a setting somewhere that I'm missing somewhere for a silence 
threshold for WaitForSilence or am I misunderstanding its use?


The Asterisk version is

Asterisk 11.7.0~dfsg-1ubuntu1


And it's Asterisk installed from an Ubuntu package.

Thanks so much!

--
Mike A. Leonetti
As warm as green tea


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Re: [asterisk-users] Asterisk 12 - Security Fix Only Notice

[Mike Diehl]

On Monday, December 08, 2014 02:21:16 PM Matthew Jordan wrote:
 Hey everyone!
 
 This is a friendly reminder that Asterisk 12 will be entering security fix
 only mode soon. As a Standard release of Asterisk, Asterisk 12 received one
 year of maintenance fixes, and will receive one year of security fixes.
 Asterisk 12 was first released on 2013-12-20 - the one year anniversary of
 which is just around the corner! After 2014-12-20, additional releases of
 Asterisk 12 will no longer be made. The final bug fix release of Asterisk
 12 will therefore be 12.8.0. Users of Asterisk 12 are encouraged to move to
 the next major version, Asterisk 13, as soon as possible. Asterisk 13 is a
 Long Term Support (LTS) and has maintenance support through 2018-10-24,
 with its full End of Life occurring on 2019-10-24.
 
 For more information on Asterisk versions and their supported lifetimes,
 please see the following wiki page:
 
 https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions
 
 Thank you for your continued support of Asterisk!

Is there any time frame for when FFA will be available for 13?

-- 
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Diehlnet Communications, LLC.
Voice: (505) 903-5700
Fax: (505) 903-5701

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[asterisk-users] RT voicemail greetings not played

[Mike]

Hi all.

I've got an odd situation with my RT asterisk server.  I've got a number of 
users who are reporting that their voicemail greeting isn't being played 
anymore.  This used to work before a recent asterisk restart.  The dialplan is 
in AGI, so it wasn't changed.  I'm storing voicemail in a mysql database and 
that is working properly.  It's just the greeting message that isn't working 
properly.  And, there are not file not found type errors on the console with 
verbose=25.

Any ideas as to where I should look?

-- 
Mike



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[asterisk-users] RT voicemail greetings not played

[Mike]

Hi all.

I've got an odd situation with my RT asterisk server.  I've got a number of 
users who are reporting that their voicemail greeting isn't being played 
anymore.  This used to work before a recent asterisk restart.  The dialplan is 
in AGI, so it wasn't changed.  I'm storing voicemail in a mysql database and 
that is working properly.  It's just the greeting message that isn't working 
properly.  And, there are not file not found type errors on the console with 
verbose=25.

Any ideas as to where I should look?

-- 
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Re: [asterisk-users] Playback/background audio from MySQL BLOB

[Mike]

On Tue, 23 Sep 2014, Steve Edwards wrote:


 On 09/23/2014 02:17 PM, Steve Edwards wrote:



  For some applications, storing recorded audio (prompts and caller
  recordings) as a BLOB in MySQL has advantages.


On Tue, 23 Sep 2014, Don Kelly wrote:


 I'm curious about what the advantages are of storing audio in a blob.
 Wouldn't it be more efficient to store it in a file and just put the
 filename in the database?


Multiple web servers, multiple Asterisk servers, multiple DB servers, 
synchronizing filesystems vs db, etc.


It appears to eliminate some problems, but Asterisk limiting audio playback 
to files seems like a tough obstacle.


Maybe make the audio files available to all servers via a single NFS 
directory?  Probably not a good solution if the servers aren't co-located.


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[asterisk-users] Unregistered ports on SPAxxxx

[Mike Diehl]

Hi all,

I've got a few devices, SPA112's and SPA8000's, that are giving me problems.

Each device has a separate SIP credential for each port, but sometimes, only a 
few of the ports register.  

So, the device will be running fine for a while, then suddenly one or more of 
the ports will become Unreachable.  These ports will stay unreachable until 
the device is power cycled.

I'm presuming that there was a momentary interruption in connectivity that 
caused the registrations to fail/timeout.  But the ports should have become 
Reachable by the time the registration period elapses.  But they don't.

Any ideas?

Mike.

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Re: [asterisk-users] Unregistered ports on SPAxxxx

[Mike Diehl]

On Tuesday, August 05, 2014 11:01:01 AM Kevin Larsen wrote:

 I've got a few devices, SPA112's and SPA8000's, that are giving me problems.
 
 Each device has a separate SIP credential for each port, but 
 sometimes, only a 
 few of the ports register.  
 
 So, the device will be running fine for a while, then suddenly one or more 
of 
 the ports will become Unreachable.  These ports will stay unreachable until 
 the device is power cycled.
 
 I'm presuming that there was a momentary interruption in connectivity that 
 caused the registrations to fail/timeout.  But the ports should have become 
 Reachable by the time the registration period elapses.  But they don't.
 
 Any ideas?

Interesting you should mention this. I have an SPA-112 that is giving me fits 
right now. Multiple times per week it goes down and has to be power cycled. 
When it is down, it is not registered with Asterisk, I cannot reach its 
configuration web page, but I can ping it. Mine is running 1.2.1 (004) on the 
firmware, but I see that 1.3.3 (015) is out. That was going to be my next 
change to see if it helps. 
All of my SPA112's are running 1.3.2(014).  My SPA8000's are running 5.1.10.


-- 
Mike Diehl
Diehlnet Communications, LLC.
Voice: (505) 903-5700
Fax: (505) 903-5701
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Re: [asterisk-users] Unregistered ports on SPAxxxx

[Mike Diehl]

On Tuesday, August 05, 2014 05:19:55 PM Steven Howes wrote:



On 5 Aug 2014, at 17:10, Mike Diehl mdiehlena...@gmail.com wrote:
All of my SPA112's are running 1.3.2(014).  My SPA8000's are running 5.1.10.



If you do firmware upgrade your 8000s, don’t go past 6.1.3 or it’ll go badly… 
Freezing and requiring power-cycle, clocks stopping (and showing minus 
figures!) and major struggles downgrading again. Had about a dozen of them 
doing the same, eventual downgrade to 6.1.3 and it’s all happy.


Steve

Thanks for the warning!  Ouch!

-- 
Mike Diehl
Diehlnet Communications, LLC.
Voice: (505) 903-5700
Fax: (505) 903-5701
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Re: [asterisk-users] Terrible dahdi_test results

[Mike Leddy]

Thanks again Russ,

I did as you described:

[ 1777.393179] dahdi: Version: 2.7.0.1
[ 1777.393995] dahdi: Telephony Interface Registered on major 196
[ 1789.943167] dahdi: Warning: Span DAHDI_DUMMY/1 didn't specify a
spantype. Please fix driver!
[ 1789.943865] dahdi_dummy: Trying to load High Resolution Timer
[ 1789.943869] dahdi_dummy: Initialized High Resolution Timer
[ 1789.943872] dahdi_dummy: Starting High Resolution Timer
[ 1789.943876] dahdi_dummy: High Resolution Timer started, good to go
[ 2816.078038] Setting up DMA (write/read = 2000/2200)
[ 2816.078054] Controller version: 24
[ 2816.07] FALC version: 
[ 2816.080284] TE110P: Setting up global serial parameters for E1 FALC
V1.2
[ 2816.080775] TE110P: Successfully initialized serial bus for card
[ 2816.084698] Found a Wildcard: Digium Wildcard TE110P T1/E1

There is a significant improvement:

# dahdi_test 
Opened pseudo dahdi interface, measuring accuracy...
79.256% 99.401% 86.288% 95.736% 77.951% 94.636% 87.010% 97.638% 
90.234% 97.651% 97.648% 91.744% 90.541% 92.671% 91.090% 91.118% 
95.832% 96.503% 83.753% 97.642% 98.419% 81.845% 92.441% 99.142% 
96.645% 88.332% 77.918% 97.651% 87.029% 69.357% 98.153% 89.980% 
93.939% 86.579% 78.785% 98.497% 77.916% 82.414% 96.838% 97.061% 
99.392% 94.414% 97.094% 99.095% 90.329% 97.678% 90.087% 93.631% 
99.130% 97.379% 94.361% 97.653% 97.670% 77.946% 78.312% 95.465% 
84.429% 97.640% 78.185% 97.910% 77.938% 78.314% 89.535% 83.030% 
85.246% 84.128% 74.614% 87.055% 90.382% 79.477% 97.096% 90.822% ^C
--- Results after 72 passes ---
Best: 99.401% -- Worst: 69.357% -- Average: 90.260025%
Cummulative Accuracy (not per pass): 92.402

I suspect it may not be good enough yet for production but a
step in the right direction :-)

Timer Stats Version: v0.2
Sample period: 10.013 s

 10014,  9027 modprobe init_module (dahdi_dummy_hr_int)


I will test it on a live E1 soon.

Best regards,

Mike


On Wed, 2014-05-14 at 16:53 -0500, Russ Meyerriecks wrote:
 On Wed, May 14, 2014 at 3:41 PM, Mike Leddy m...@loop.com.br wrote:
 Hi Eric,
 
 I plugged an E1 into the card and it doesn't make any
 difference.
 
 
 Check to see if the card is interrupting 1000 times per second with
 something like:
 cat /proc/interrupts | grep wc  sleep 1  cat /proc/interrupts |
 grep wc
 
 
 You could also try manually compiling dahdi_dummy by commenting it
 back in, in the file:
 drivers/dahdi/Kbuild
 Then modprobe dahdi_dummy
 This module forces the use of the high res timers
 
 
 -- 
 Russ Meyerriecks
 Digium, Inc. | Linux Kernel Developer
 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
 direct: +1 256-428-6025
 Check us out at: www.digium.com  www.asterisk.org



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Re: [asterisk-users] Terrible dahdi_test results

[Mike Leddy]

Hi Russ,

I rebooted the machine loading dahdi_dummy in /etc/modules before
the /etc/init.d/dahdi.

Now dahdi_test shows a nearly perfect score:

# dahdi_test 
Opened pseudo dahdi interface, measuring accuracy...
99.998% 99.990% 99.998% 99.996% 99.998% 99.998% 99.997% 99.997% 
99.998% 99.997% 99.998% 99.997% 99.998% 99.998% 99.997% 99.998% 
99.997% 99.997% 99.997% 99.998% 99.997% 99.998% 99.998% 99.997% ^C
--- Results after 24 passes ---
Best: 99.998% -- Worst: 99.990% -- Average: 99.997188%
Cummulative Accuracy (not per pass): 99.997

When I connect a live E1 to the card it does work but I get a fair
number of:

[May 15 15:10:39] NOTICE[4017] chan_dahdi.c: PRI got event: HDLC Bad FCS (8) on 
D-channel of span 1
[May 15 15:10:42] NOTICE[4017] chan_dahdi.c: PRI got event: HDLC Bad FCS (8) on 
D-channel of span 1
[May 15 15:10:43] NOTICE[4017] chan_dahdi.c: PRI got event: HDLC Bad FCS (8) on 
D-channel of span 1
[May 15 15:10:43] NOTICE[4017] chan_dahdi.c: PRI got event: HDLC Bad FCS (8) on 
D-channel of span 1
[May 15 15:10:43] NOTICE[4017] chan_dahdi.c: PRI got event: HDLC Bad FCS (8) on 
D-channel of span 1
[May 15 15:10:45] NOTICE[4017] chan_dahdi.c: PRI got event: HDLC Bad FCS (8) on 
D-channel of span 1
[May 15 15:11:01] NOTICE[4017] chan_dahdi.c: PRI got event: HDLC Bad FCS (8) on 
D-channel of span 1
[May 15 15:11:01] NOTICE[4017] chan_dahdi.c: PRI got event: HDLC Bad FCS (8) on 
D-channel of span 1
[May 15 15:11:06] NOTICE[4017] chan_dahdi.c: PRI got event: HDLC Bad FCS (8) on 
D-channel of span 1
[May 15 15:11:12] NOTICE[4017] chan_dahdi.c: PRI got event: HDLC Bad FCS (8) on 
D-channel of span 1
[May 15 15:11:12] NOTICE[4017] chan_dahdi.c: PRI got event: HDLC Bad FCS (8) on 
D-channel of span 1
[May 15 15:11:12] NOTICE[4017] chan_dahdi.c: PRI got event: HDLC Bad FCS (8) on 
D-channel of span 1
[May 15 15:11:13] NOTICE[4017] chan_dahdi.c: PRI got event: HDLC Bad FCS (8) on 
D-channel of span 1

Resulting in:

[May 15 15:10:34] NOTICE[4017] chan_dahdi.c: PRI got event: Alarm (4) on 
D-channel of span 1
[May 15 15:10:39] NOTICE[4017] chan_dahdi.c: PRI got event: Alarm (4) on 
D-channel of span 1
[May 15 15:10:43] NOTICE[4017] chan_dahdi.c: PRI got event: Alarm (4) on 
D-channel of span 1
[May 15 15:10:44] NOTICE[4017] chan_dahdi.c: PRI got event: Alarm (4) on 
D-channel of span 1
[May 15 15:10:45] NOTICE[4017] chan_dahdi.c: PRI got event: Alarm (4) on 
D-channel of span 1
[May 15 15:11:13] NOTICE[4017] chan_dahdi.c: PRI got event: Alarm (4) on 
D-channel of span 1
[May 15 15:11:17] NOTICE[4017] chan_dahdi.c: PRI got event: Alarm (4) on 
D-channel of span 1
[May 15 15:11:22] NOTICE[4017] chan_dahdi.c: PRI got event: Alarm (4) on 
D-channel of span 1

Not usable in production but getting a lot closer.

Is there anything else that can be done to improve this ?

Best regards,

Mike


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Re: [asterisk-users] Terrible dahdi_test results

[Mike Leddy]

Hi Russ,

Sorry I didn't send that... Here's a few runs:

# cat /proc/interrupts | grep wc  sleep 1  cat /proc/interrupts | grep wc
  28:1682906  0  0  0  0  0 
 0  0  0  0  0  0  IR-IO-APIC-fasteoi   
wcte11xp
  28:1683913  0  0  0  0  0 
 0  0  0  0  0  0  IR-IO-APIC-fasteoi   
wcte11xp

= 1007

# cat /proc/interrupts | grep wc  sleep 1  cat /proc/interrupts | grep wc
  28:1690587  0  0  0  0  0 
 0  0  0  0  0  0  IR-IO-APIC-fasteoi   
wcte11xp
  28:1691594  0  0  0  0  0 
 0  0  0  0  0  0  IR-IO-APIC-fasteoi   
wcte11xp

= 1007

# cat /proc/interrupts | grep wc  sleep 1  cat /proc/interrupts | grep wc
  28:1695091  0  0  0  0  0 
 0  0  0  0  0  0  IR-IO-APIC-fasteoi   
wcte11xp
  28:1696100  0  0  0  0  0 
 0  0  0  0  0  0  IR-IO-APIC-fasteoi   
wcte11xp

= 1009

# cat /proc/interrupts | grep wc  sleep 1  cat /proc/interrupts | grep wc
  28:1700363  0  0  0  0  0 
 0  0  0  0  0  0  IR-IO-APIC-fasteoi   
wcte11xp
  28:1701370  0  0  0  0  0 
 0  0  0  0  0  0  IR-IO-APIC-fasteoi   
wcte11xp

= 1007


Best regards,

Mike

On Thu, 2014-05-15 at 10:13 -0500, Russ Meyerriecks wrote:

 On Thu, May 15, 2014 at 9:10 AM, Mike Leddy m...@loop.com.br wrote:
 --- Results after 72 passes ---
 Best: 99.401% -- Worst: 69.357% -- Average: 90.260025%
 Cummulative Accuracy (not per pass): 92.402
 
 I suspect it may not be good enough yet for production but a
 step in the right direction :-)
 
 
 What was the result of the /proc/interrupt test? How many interrupts did the 
 te110 fire in 1 second?
 What happens if you rmmod the wct1xxp driver and ran dahdi_test against just 
 dahdi  dahdi_dummy modules?
  
 -- 
 Russ Meyerriecks
 Digium, Inc. | Linux Kernel Developer
 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
 direct: +1 256-428-6025
 Check us out at: www.digium.com  www.asterisk.org
 


On Thu, 2014-05-15 at 10:13 -0500, Russ Meyerriecks wrote:
 
 On Thu, May 15, 2014 at 9:10 AM, Mike Leddy m...@loop.com.br wrote:
 --- Results after 72 passes ---
 Best: 99.401% -- Worst: 69.357% -- Average: 90.260025%
 Cummulative Accuracy (not per pass): 92.402
 
 I suspect it may not be good enough yet for production but a
 step in the right direction :-)
 
 
 What was the result of the /proc/interrupt test? How many interrupts did the 
 te110 fire in 1 second?
 What happens if you rmmod the wct1xxp driver and ran dahdi_test against just 
 dahdi  dahdi_dummy modules?
  
 -- 
 Russ Meyerriecks
 Digium, Inc. | Linux Kernel Developer
 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
 direct: +1 256-428-6025
 Check us out at: www.digium.com  www.asterisk.org



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Re: [asterisk-users] Terrible dahdi_test results

[Mike Leddy]

Thanks for the suggestion Gareth,

The span line I'm using is:

span=1,1,0,ccs,hdb3,crc4

Its the same as used with the TE12xP that is normally used on this
E1 line in production which is extremely stable.

I did as you suggested using:

span=1,0,0,ccs,hdb3,crc4

After a dahdi_cfg pretty much the same results:

[May 15 17:35:24] NOTICE[4017] chan_dahdi.c: PRI got event: Alarm (4) on 
D-channel of span 1
[May 15 17:35:24] NOTICE[4017] chan_dahdi.c: PRI got event: Alarm (4) on 
D-channel of span 1
[May 15 17:35:32] NOTICE[4017] chan_dahdi.c: PRI got event: Alarm (4) on 
D-channel of span 1
[May 15 17:35:36] NOTICE[4017] chan_dahdi.c: PRI got event: Alarm (4) on 
D-channel of span 1
[May 15 17:36:01] NOTICE[4017] chan_dahdi.c: PRI got event: Alarm (4) on 
D-channel of span 1
[May 15 17:36:05] NOTICE[4017] chan_dahdi.c: PRI got event: Alarm (4) on 
D-channel of span 1
[May 15 17:36:12] NOTICE[4017] chan_dahdi.c: PRI got event: Alarm (4) on 
D-channel of span 1
[May 15 17:36:13] NOTICE[4017] chan_dahdi.c: PRI got event: Alarm (4) on 
D-channel of span 1
[May 15 17:36:21] NOTICE[4017] chan_dahdi.c: PRI got event: Alarm (4) on 
D-channel of span 1
[May 15 17:36:21] NOTICE[4017] chan_dahdi.c: PRI got event: Alarm (4) on 
D-channel of span 1
[May 15 17:36:25] NOTICE[4017] chan_dahdi.c: PRI got event: Alarm (4) on 
D-channel of span 1

Best regards,

Mike

On Thu, 2014-05-15 at 17:53 +0100, Gareth Blades wrote:
 On 15/05/14 16:28, Mike Leddy wrote:
  Hi Russ,
 
  I rebooted the machine loading dahdi_dummy in /etc/modules before
  the /etc/init.d/dahdi.
 
  Now dahdi_test shows a nearly perfect score:
 
  # dahdi_test
  Opened pseudo dahdi interface, measuring accuracy...
  99.998% 99.990% 99.998% 99.996% 99.998% 99.998% 99.997% 99.997%
  99.998% 99.997% 99.998% 99.997% 99.998% 99.998% 99.997% 99.998%
  99.997% 99.997% 99.997% 99.998% 99.997% 99.998% 99.998% 99.997% ^C
  --- Results after 24 passes ---
  Best: 99.998% -- Worst: 99.990% -- Average: 99.997188%
  Cummulative Accuracy (not per pass): 99.997
 
  When I connect a live E1 to the card it does work but I get a fair
  number of:
 
  [May 15 15:10:39] NOTICE[4017] chan_dahdi.c: PRI got event: HDLC Bad FCS 
  (8) on D-channel of span 1
  [May 15 15:10:42] NOTICE[4017] chan_dahdi.c: PRI got event: HDLC Bad FCS 
  (8) on D-channel of span 1
  [May 15 15:10:43] NOTICE[4017] chan_dahdi.c: PRI got event: HDLC Bad FCS 
  (8) on D-channel of span 1
  [May 15 15:10:43] NOTICE[4017] chan_dahdi.c: PRI got event: HDLC Bad FCS 
  (8) on D-channel of span 1
  [May 15 15:10:43] NOTICE[4017] chan_dahdi.c: PRI got event: HDLC Bad FCS 
  (8) on D-channel of span 1
  [May 15 15:10:45] NOTICE[4017] chan_dahdi.c: PRI got event: HDLC Bad FCS 
  (8) on D-channel of span 1
  [May 15 15:11:01] NOTICE[4017] chan_dahdi.c: PRI got event: HDLC Bad FCS 
  (8) on D-channel of span 1
  [May 15 15:11:01] NOTICE[4017] chan_dahdi.c: PRI got event: HDLC Bad FCS 
  (8) on D-channel of span 1
  [May 15 15:11:06] NOTICE[4017] chan_dahdi.c: PRI got event: HDLC Bad FCS 
  (8) on D-channel of span 1
  [May 15 15:11:12] NOTICE[4017] chan_dahdi.c: PRI got event: HDLC Bad FCS 
  (8) on D-channel of span 1
  [May 15 15:11:12] NOTICE[4017] chan_dahdi.c: PRI got event: HDLC Bad FCS 
  (8) on D-channel of span 1
  [May 15 15:11:12] NOTICE[4017] chan_dahdi.c: PRI got event: HDLC Bad FCS 
  (8) on D-channel of span 1
  [May 15 15:11:13] NOTICE[4017] chan_dahdi.c: PRI got event: HDLC Bad FCS 
  (8) on D-channel of span 1
 
  Resulting in:
 
  [May 15 15:10:34] NOTICE[4017] chan_dahdi.c: PRI got event: Alarm (4) on 
  D-channel of span 1
  [May 15 15:10:39] NOTICE[4017] chan_dahdi.c: PRI got event: Alarm (4) on 
  D-channel of span 1
  [May 15 15:10:43] NOTICE[4017] chan_dahdi.c: PRI got event: Alarm (4) on 
  D-channel of span 1
  [May 15 15:10:44] NOTICE[4017] chan_dahdi.c: PRI got event: Alarm (4) on 
  D-channel of span 1
  [May 15 15:10:45] NOTICE[4017] chan_dahdi.c: PRI got event: Alarm (4) on 
  D-channel of span 1
  [May 15 15:11:13] NOTICE[4017] chan_dahdi.c: PRI got event: Alarm (4) on 
  D-channel of span 1
  [May 15 15:11:17] NOTICE[4017] chan_dahdi.c: PRI got event: Alarm (4) on 
  D-channel of span 1
  [May 15 15:11:22] NOTICE[4017] chan_dahdi.c: PRI got event: Alarm (4) on 
  D-channel of span 1
 
  Not usable in production but getting a lot closer.
 
  Is there anything else that can be done to improve this ?
 
  Best regards,
 
  Mike
 
 
 
 Check your span= line in you configuration. If your telco is providing 
 clocking and you are set to generate it yourself then they go out of 
 sync which generally causes errors like these. If you are set to be the 
 clock master try changing it to see if it improves. It should either 
 improve or not work at all.
 



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Re: [asterisk-users] Terrible dahdi_test results

[Mike Leddy]

I remembered I have an older box with a Wildcard TE12xP that 
uses the wcte12xp module with a newer 3.9.11 kernel that works 
perfectly.

I setup the problematic machine with the same kernel in the hope 
that this might be relevant. Unfortunately the same situation 
persists.

I used the /proc/timer_stats to see how the timers were used:

Timer Stats Version: v0.2
Sample period: 10.002 s

  311, 15081 kworker/u:0  mod_timer (te12xp_timer)


With the TE110P I couldn't find any entry It seems that the
timing mechanism is different, it doesn't use mod_timer.

I'm running out of ideas. Please help.

Thanks,

Mike

On Tue, 2014-05-13 at 17:56 -0300, Mike Leddy wrote:
 Thanks again Russ,
 
 Just a quick reply for now.
 
 No virtualization, but yes I am running a tickless kernel:
 
 #
 # Processor type and features
 #
 CONFIG_NO_HZ=y
 
 Standard for debian kernels. I booted with nohz=off and the behaviour
 changed. Unfortunately for the worse:
 
 # dahdi_test 
 Opened pseudo dahdi interface, measuring accuracy...
 66.653% 66.683% 66.683% 66.807% 67.705% 66.666% 66.651% 66.679% 
 67.516% 66.882% 66.649% 66.657% 66.678% 66.668% 66.672% 66.664% 
 66.675% 66.675% 66.659% 66.692% 66.631% 66.187% 66.650% 66.710% 
 66.648% 66.633% 66.714% 66.638% 66.688% 66.794% 66.645% 66.696%
 --- Results after 32 passes ---
 Best: 67.705% -- Worst: 66.187% -- Average: 66.726523%
 
 Comparing the boot messages without nohz=off:
 
 [0.00] hpet clockevent registered
 [0.00] Fast TSC calibration failed
 [0.00] TSC: Unable to calibrate against PIT
 [0.00] TSC: using HPET reference calibration
 [0.00] Detected 2593.456 MHz processor.
 
 and with nohz=off:
 
 [0.00] hpet clockevent registered
 [0.00] Fast TSC calibration using PIT
 [0.00] Detected 2593.225 MHz processor.
 
 I am encouraged that we seem to be homing in on the problem. I need to
 read up a bit more on the subject and look at possible power 
 saving issues on this machine.
 
 Best regards,
 
 Mike
 
 
 On Tue, 2014-05-13 at 15:26 -0500, Russ Meyerriecks wrote:
  On Tue, May 13, 2014 at 7:28 AM, Mike Leddy m...@loop.com.br wrote:
  But on examination the /etc/init.d/dahdi start was only
  loading
  the dahdi module.
  
  
  With this in mind I might start looking around the system for things
  which might cause jitter in the servicing of system timer interrupts:
  
  
  Are you running under a virtualized environment?
  Are you running a tickless kernel? (maybe try adding nohz=off to your
  kernel boot parameters)
  Is there some sort of processor power saving or frequency scaling
  going on that interrupts the system timer?
  
  
  -- 
  Russ Meyerriecks
  Digium, Inc. | Linux Kernel Developer
  445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
  direct: +1 256-428-6025
  Check us out at: www.digium.com  www.asterisk.org
 
 
 



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Re: [asterisk-users] Terrible dahdi_test results

[Mike Leddy]

Hi Eric,

I plugged an E1 into the card and it doesn't make any difference.

# dahdi_test 
Opened pseudo dahdi interface, measuring accuracy...
88.829% 87.806% 88.988% 88.854% 88.944% 88.952% 88.967% 88.841% 
88.889% 88.946% 88.933% 88.841% 88.885% 89.050% 88.904% 87.933% 
88.912% 88.949% 88.913% 88.886% 88.891% 88.798% 88.746% 89.009% 
88.934% 88.870% 88.875% 89.003% 88.925% 88.863% 89.018% 88.093% 
88.447% 88.691% 89.034% 88.703% 88.815% 89.011% 88.919% 88.825% 
etc.

I will try the card in an older machine tomorrow.

Ironic is that i bought this card because it has a PCI express
interface so I can use it in recent servers but it uses an
older chipset and driver than I was using.

Thanks for the help,

Mike


On Wed, 2014-05-14 at 15:54 -0400, Eric Wieling wrote:
 Try the card in another machine with a different brand of motherboard.  If it 
 works you know it is a hardware issue. 
 
 Do you have an actual T-1 plugged into your card?  If not, try that and see 
 if there is any difference.
 
 -Original Message-
 From: asterisk-users-boun...@lists.digium.com 
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Leddy
 Sent: Wednesday, May 14, 2014 3:43 PM
 To: asterisk-users@lists.digium.com
 Subject: Re: [asterisk-users] Terrible dahdi_test results
 
 
 I remembered I have an older box with a Wildcard TE12xP that uses the 
 wcte12xp module with a newer 3.9.11 kernel that works perfectly.
 
 I setup the problematic machine with the same kernel in the hope that this 
 might be relevant. Unfortunately the same situation persists.
 
 I used the /proc/timer_stats to see how the timers were used:
 
 Timer Stats Version: v0.2
 Sample period: 10.002 s
 
   311, 15081 kworker/u:0  mod_timer (te12xp_timer)
 
 
 With the TE110P I couldn't find any entry It seems that the timing 
 mechanism is different, it doesn't use mod_timer.
 
 I'm running out of ideas. Please help.
 
 Thanks,
 
 Mike
 
 On Tue, 2014-05-13 at 17:56 -0300, Mike Leddy wrote:
  Thanks again Russ,
  
  Just a quick reply for now.
  
  No virtualization, but yes I am running a tickless kernel:
  
  #
  # Processor type and features
  #
  CONFIG_NO_HZ=y
  
  Standard for debian kernels. I booted with nohz=off and the behaviour 
  changed. Unfortunately for the worse:
  
  # dahdi_test
  Opened pseudo dahdi interface, measuring accuracy...
  66.653% 66.683% 66.683% 66.807% 67.705% 66.666% 66.651% 66.679% 
  67.516% 66.882% 66.649% 66.657% 66.678% 66.668% 66.672% 66.664% 
  66.675% 66.675% 66.659% 66.692% 66.631% 66.187% 66.650% 66.710% 
  66.648% 66.633% 66.714% 66.638% 66.688% 66.794% 66.645% 66.696%
  --- Results after 32 passes ---
  Best: 67.705% -- Worst: 66.187% -- Average: 66.726523%
  
  Comparing the boot messages without nohz=off:
  
  [0.00] hpet clockevent registered
  [0.00] Fast TSC calibration failed
  [0.00] TSC: Unable to calibrate against PIT
  [0.00] TSC: using HPET reference calibration
  [0.00] Detected 2593.456 MHz processor.
  
  and with nohz=off:
  
  [0.00] hpet clockevent registered
  [0.00] Fast TSC calibration using PIT
  [0.00] Detected 2593.225 MHz processor.
  
  I am encouraged that we seem to be homing in on the problem. I need to 
  read up a bit more on the subject and look at possible power 
  saving issues on this machine.
  
  Best regards,
  
  Mike
  
  
  On Tue, 2014-05-13 at 15:26 -0500, Russ Meyerriecks wrote:
   On Tue, May 13, 2014 at 7:28 AM, Mike Leddy m...@loop.com.br wrote:
   But on examination the /etc/init.d/dahdi start was only
   loading
   the dahdi module.
   
   
   With this in mind I might start looking around the system for things 
   which might cause jitter in the servicing of system timer interrupts:
   
   
   Are you running under a virtualized environment?
   Are you running a tickless kernel? (maybe try adding nohz=off to 
   your kernel boot parameters) Is there some sort of processor power 
   saving or frequency scaling going on that interrupts the system 
   timer?
   
   
   --
   Russ Meyerriecks
   Digium, Inc. | Linux Kernel Developer
   445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
   direct: +1 256-428-6025
   Check us out at: www.digium.com  www.asterisk.org
  
  
  
 
 
 
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asterisk

Re: [asterisk-users] Terrible dahdi_test results

[Mike Leddy]

Thanks Russ,

I blacklisted wcopenpci, and i noticed an improvement:

# dahdi_test 
Opened pseudo dahdi interface, measuring accuracy...
91.973% 96.242% 94.423% 77.932% 97.654% 87.114% 68.750% 87.417% 
75.768% 95.693% 93.086% 77.936% 87.283% 92.996% 77.942% 77.926% 
77.943% 89.545% 93.949% 87.032% 91.939% 71.877% 90.648% 88.163% 
77.173% 97.651% 77.932% 77.942% 77.929% 76.548% 80.922% 90.292% 
81.977% 89.354% 79.203% 76.665% 77.935% 91.000% 68.182% 95.018%
--- Results after 40 passes ---
Best: 97.654% -- Worst: 68.182% -- Average: 84.673880%
Cummulative Accuracy (not per pass): 86.629

But on examination the /etc/init.d/dahdi start was only loading 
the dahdi module. 

# lsmod | egrep 'wc|dahdi'
dahdi 192295  0 
crc_ccitt  12347  1 dahdi

After a modprobe wcte11xp the situation is:

# lsmod | egrep 'wc|dahdi'
dahdi_echocan_oslec12578  30 
echo   12652  1 dahdi_echocan_oslec
wcte11xp   21535  0 
dahdi 192295  2 wcte11xp,dahdi_echocan_oslec
crc_ccitt  12347  1 dahdi

The test returns to as it was before:

# dahdi_test 
Opened pseudo dahdi interface, measuring accuracy...
89.100% 89.212% 88.871% 89.078% 89.286% 89.559% 89.358% 89.423% 
89.184% 89.083% 88.931% 89.070% 89.110% 88.818% 88.573% 89.091% 
89.006% 88.978% 89.157% 89.069% 89.112% 88.890% 89.374% 88.900% 
89.042% 89.043% 88.946% 88.786% 88.865% 89.259% 88.951% 88.763% 
88.944% 89.123% 88.956% 88.976% 89.044% 89.040% 88.970% 89.148%
--- Results after 40 passes ---
Best: 89.559% -- Worst: 88.573% -- Average: 89.052215%
Cummulative Accuracy (not per pass): 89.052

Still experimenting. 

Best regards,

Mike


On Mon, 2014-05-12 at 17:23 -0500, Russ Meyerriecks wrote:
 On Mon, May 12, 2014 at 4:57 PM, Mike Leddy m...@loop.com.br wrote:
 [   39.223031] wcopenpci: Module loaded
 [   39.751770] wcte1xxp: Setting yellow alarm
 
 # dahdi_hardware
 pci::06:00.0 wcte11xp+e159:0001 Digium Wildcard
 TE110P T1/E1 Board
 
 
 Mike,
   This stuff predates my time on this project, but it looks like
 wcopenpci and wcte1xxp might be conflicting on your system because
 they register against the same hardware. I might try blacklisting the
 wcopenpci driver in /etc/modprobe.d/dahdi.blacklist.conf and rebooting
 or reloading dahdi.
 
 
 -- 
 Russ Meyerriecks
 Digium, Inc. | Linux Kernel Developer
 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
 direct: +1 256-428-6025
 Check us out at: www.digium.com  www.asterisk.org



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Re: [asterisk-users] Terrible dahdi_test results

[Mike Leddy]

Thanks again Russ,

Just a quick reply for now.

No virtualization, but yes I am running a tickless kernel:

#
# Processor type and features
#
CONFIG_NO_HZ=y

Standard for debian kernels. I booted with nohz=off and the behaviour
changed. Unfortunately for the worse:

# dahdi_test 
Opened pseudo dahdi interface, measuring accuracy...
66.653% 66.683% 66.683% 66.807% 67.705% 66.666% 66.651% 66.679% 
67.516% 66.882% 66.649% 66.657% 66.678% 66.668% 66.672% 66.664% 
66.675% 66.675% 66.659% 66.692% 66.631% 66.187% 66.650% 66.710% 
66.648% 66.633% 66.714% 66.638% 66.688% 66.794% 66.645% 66.696%
--- Results after 32 passes ---
Best: 67.705% -- Worst: 66.187% -- Average: 66.726523%

Comparing the boot messages without nohz=off:

[0.00] hpet clockevent registered
[0.00] Fast TSC calibration failed
[0.00] TSC: Unable to calibrate against PIT
[0.00] TSC: using HPET reference calibration
[0.00] Detected 2593.456 MHz processor.

and with nohz=off:

[0.00] hpet clockevent registered
[0.00] Fast TSC calibration using PIT
[0.00] Detected 2593.225 MHz processor.

I am encouraged that we seem to be homing in on the problem. I need to
read up a bit more on the subject and look at possible power 
saving issues on this machine.

Best regards,

Mike


On Tue, 2014-05-13 at 15:26 -0500, Russ Meyerriecks wrote:
 On Tue, May 13, 2014 at 7:28 AM, Mike Leddy m...@loop.com.br wrote:
 But on examination the /etc/init.d/dahdi start was only
 loading
 the dahdi module.
 
 
 With this in mind I might start looking around the system for things
 which might cause jitter in the servicing of system timer interrupts:
 
 
 Are you running under a virtualized environment?
 Are you running a tickless kernel? (maybe try adding nohz=off to your
 kernel boot parameters)
 Is there some sort of processor power saving or frequency scaling
 going on that interrupts the system timer?
 
 
 -- 
 Russ Meyerriecks
 Digium, Inc. | Linux Kernel Developer
 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
 direct: +1 256-428-6025
 Check us out at: www.digium.com  www.asterisk.org



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[asterisk-users] Terrible dahdi_test results

[Mike Leddy]

Hello, I am trying to get a Wildcard TE110P to work in a relatively
modern HP Proliant DL385p Gen8 server. Being a potent 12 core Opteron
server I expected no problems.

Much to my dismay the dahdi_test results are constantly terrible:

# dahdi_test 
Opened pseudo dahdi interface, measuring accuracy...
89.101% 89.195% 89.142% 88.957% 88.953% 89.115% 89.089% 89.134% 
89.066% 89.021% 88.933% 89.044% 89.200% 89.017% 89.425% 89.014% 
89.140% 89.814% 89.379% 89.185% 88.943% 89.000% 89.090% 89.067% 
88.975% 88.875% 89.095% 89.130% 89.049% 89.046% 89.040% 88.945% 
89.211% 89.021% 89.091% 88.972% 88.973% 89.147% 89.003% 88.970%
--- Results after 40 passes ---
Best: 89.814% -- Worst: 88.875% -- Average: 89.089184%
Cummulative Accuracy (not per pass): 89.089

Trying to use the card results in constant 'HDLC Bad FCS' and
consequent 'HDLC Abort'.

As far as I can tell everything should be fine. The card has its 
own IO-APIC interrupt (28). I tried setting its smp_affinity to 
one cpu, changed the pci card latency... to no avail, always
the same terrible dahdi_test results.

Some info:

# uname -a
Linux mundau 3.2.0-4-amd64 #1 SMP Debian 3.2.51-1 x86_64 GNU/Linux

# dmesg | egrep -i 'dahdi|te110|wcop|wcte1'
[   39.054098] dahdi: Version: 2.7.0.1
[   39.054775] dahdi: Telephony Interface Registered on major 196
[   39.128206] TE110P: Setting up global serial parameters for E1 FALC V1.2
[   39.128376] TE110P: Successfully initialized serial bus for card
[   39.131355] Found a Wildcard: Digium Wildcard TE110P T1/E1
[   39.223031] wcopenpci: Module loaded 
[   39.714308] dahdi_echocan_oslec: Registered echo canceler 'OSLEC'
[   39.716015] TE110P: Span configured for CCS/HDB3/CRC4
[   39.751770] wcte1xxp: Setting yellow alarm

# dahdi_hardware 
pci::06:00.0 wcte11xp+e159:0001 Digium Wildcard TE110P T1/E1 Board

# dahdi_scan 
[1]
active=yes
alarms=RED
description=Digium Wildcard TE110P T1/E1 Card 0
name=WCT1/0
manufacturer=Digium
devicetype=Digium Wildcard TE110P T1/E1
location=PCI Bus 06 Slot 01
basechan=1
totchans=31
irq=0
type=digital-E1
syncsrc=0
lbo=0 db (CSU)/0-133 feet (DSX-1)
coding_opts=HDB3
framing_opts=CCS,CRC4
coding=HDB3
framing=CCS/CRC4

Here is the /proc/interrupt info with CPU1-CPU10 removed for space:

CPU0 ... CPU11 
   0: 71   0   IO-APIC-edge  timer
   1:  3   0   IO-APIC-edge  i8042
   3:  8   0   IO-APIC-edge  serial
   7:  1   0   IO-APIC-edge  
   8:  1   0   IO-APIC-edge  rtc0
   9:  0   0   IO-APIC-fasteoi   acpi
  12:  5   0   IO-APIC-edge  i8042
  14:  0   0   IO-APIC-edge  pata_atiixp
  15:  0   0   IO-APIC-edge  pata_atiixp
  16: 52   0   IO-APIC-fasteoi   ahci
  22:   1189   0   IO-APIC-fasteoi   ehci_hcd:usb2, ohci_hcd:usb4, 
ohci_hcd:usb5
  23:157   0   IO-APIC-fasteoi   ehci_hcd:usb1, ohci_hcd:usb6, 
ohci_hcd:usb7
  28:6217028   0   IO-APIC-fasteoi   wcte11xp
  44:  0   0   IO-APIC-fasteoi   uhci_hcd:usb3, hpilo
  72:  0   0   PCI-MSI-edge  AMD-Vi
  73:  29799   0   PCI-MSI-edge  hpsa0
  77:  2   0   PCI-MSI-edge  eth1-0
  78:  17413   0   PCI-MSI-edge  eth1-1
  79:   3502   0   PCI-MSI-edge  eth1-2
  80:  10967   0   PCI-MSI-edge  eth1-3
  81:   3765   0   PCI-MSI-edge  eth1-4
  82:  1   0   PCI-MSI-edge  eth2-0
  83:  1   0   PCI-MSI-edge  eth2-1
  84:  1   0   PCI-MSI-edge  eth2-2
  85:  1   0   PCI-MSI-edge  eth2-3
  86:  1   0   PCI-MSI-edge  eth2-4
  87:  1   0   PCI-MSI-edge  eth3-0
  88:  1   0   PCI-MSI-edge  eth3-1
  89:  1   0   PCI-MSI-edge  eth3-2
  90:  1   0   PCI-MSI-edge  eth3-3
  91:  1   0   PCI-MSI-edge  eth3-4
 NMI:  0   0   Non-maskable interrupts
 LOC:  363816631   Local timer interrupts
 SPU:  0   0   Spurious interrupts
 PMI:  0   0   Performance monitoring interrupts
 IWI:  0   0   IRQ work interrupts
 RES:  361248213   Rescheduling interrupts
 CAL:599 671   Function call interrupts
 TLB:157 226   TLB shootdowns
 TRM:  0   0   Thermal event interrupts
 THR:  0   0   Threshold APIC interrupts
 MCE:  0   0   Machine check exceptions
 MCP: 24  24   Machine check polls
 ERR:  1
 MIS:  0

Should I just give up on using the card in this server ? 
Is there anything else I can try ? 
What other information may be relevant ?

Many thanks in advance.

Mike


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[asterisk-users] Ghost calls on PBX

[Mike Diehl]

Hi all,

I have a user with an old Mitel PBX connected to a couple of SPA112's.  The
user is reporting that their phones ring several times a day and when they
answer the call, all they hear is dial tone or busy signal.

Their PBX guy says that the SPA112's aren't providing line supervision and
the PBX requires it.

Does anyone know how to fix this?  I'd also like to fix it from a
provisioning file, if possible.

Thank you!

Mike.
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[asterisk-users] SPA112 provisioning file questions

[Mike Diehl]

Hi all,

I've got a provisioning file that I use to configure Cisco SPA112's.

I'm wanting to get this file to do 3 things for me.  I want to turn T.38
on, Call forwarding off, and Call waiting, off for both lines.  but it's
not working.

This is what I'm using to enable T.38 for line 1.

FAX_Enable_T38_1_Yes/FAX_Enable_T38_1_
FAX_T38_Redundancy_1_1/FAX_T38_Redundancy_1_
FAX_T38_ECM_Enable_1_Yes/FAX_T38_ECM_Enable_1_
FAX_Tone_Detect_Mode_1_caller or callee/FAX_Tone_Detect_Mode_1_

This is what I'm using to turn cfwd off on line 1.

Cfwd_All_Serv_1_No/Cfwd_All_Serv_1_
Cfwd_Busy_Serv_1_No/Cfwd_Busy_Serv_1_
Cfwd_No_Ans_Serv_1_No/Cfwd_No_Ans_Serv_1_
Cfwd_Sel_Serv_1_Yes/Cfwd_Sel_Serv_1_
Cfwd_Last_Serv_1_Yes/Cfwd_Last_Serv_1_


This is what I'm using to turn call waiting off on line 1.

Call_Waiting_Serv_1_No/Call_Waiting_Serv_1_

However, these setting don't seem be be getting set on the device, even
after a reboot.

Any ideas what I'm doing wrong?

TIA,

Mike.
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Re: [asterisk-users] SPA112 provisioning file questions

[Mike Diehl]

Well, I went to an online xml validation site and found an error.  After
correcting the error, my problem is gone!

Thank you.

Mike.


On Thu, Mar 27, 2014 at 2:56 PM, Noah Engelberth
nengelbe...@team-meta.netwrote:

  To me, the settings you've sent look correct.  However, one thing I've
 found with SPA configuration files is that they're very picky - if they
 don't parse as valid XML anywhere in the file, it will pretty much silently
 discard the entire file.  The first troubleshooting step I use for SPA
 provisioning is to run all the configuration files a phone should be
 pulling through an XML validator, or pull them up in a browser and see if
 the browser handles it as XML (Chrome or IE seem to work equally well for
 this in my experience, but Firefox can get a bit cranky since the file
 isn't really an XML file with all the normal headers  tags).



 Also, have you verified with logging on the provisioning server that the
 configuration file is actually being pulled?





 Thank you,



 Noah Engelberth

 MetaLINK Technologies



 *From:* asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] *On Behalf Of *Mike Diehl
 *Sent:* Thursday, March 27, 2014 2:47 PM
 *To:* Asterisk Users Mailing List - Non-Commercial Discussion
 *Subject:* [asterisk-users] SPA112 provisioning file questions



 Hi all,

 I've got a provisioning file that I use to configure Cisco SPA112's.

 I'm wanting to get this file to do 3 things for me.  I want to turn T.38
 on, Call forwarding off, and Call waiting, off for both lines.  but it's
 not working.

 This is what I'm using to enable T.38 for line 1.

 FAX_Enable_T38_1_Yes/FAX_Enable_T38_1_
 FAX_T38_Redundancy_1_1/FAX_T38_Redundancy_1_
 FAX_T38_ECM_Enable_1_Yes/FAX_T38_ECM_Enable_1_
 FAX_Tone_Detect_Mode_1_caller or callee/FAX_Tone_Detect_Mode_1_

 This is what I'm using to turn cfwd off on line 1.

 Cfwd_All_Serv_1_No/Cfwd_All_Serv_1_
 Cfwd_Busy_Serv_1_No/Cfwd_Busy_Serv_1_
 Cfwd_No_Ans_Serv_1_No/Cfwd_No_Ans_Serv_1_
 Cfwd_Sel_Serv_1_Yes/Cfwd_Sel_Serv_1_
 Cfwd_Last_Serv_1_Yes/Cfwd_Last_Serv_1_

  This is what I'm using to turn call waiting off on line 1.

 Call_Waiting_Serv_1_No/Call_Waiting_Serv_1_

 However, these setting don't seem be be getting set on the device, even
 after a reboot.

 Any ideas what I'm doing wrong?

 TIA,

 Mike.

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[asterisk-users] Strange call transfer problem.

[Mike Diehl]

Hi again,

I've got a user who's using a bunch of Grandstream GXP2xxx's.  For the most
part, they work, except for when they try to do a phone-based call transfer.

Here's what it looks like is happening:

Phone A is on a call with phone B.   (B could be another phone, or a PSTN
endpoint.)

The user on phone A hits the phones transfer button.
The user on phone B hears moh, as they should.
When the user on phone A dials a number, the Asterisk server sees the dial
come in as though it came from phone B.

This really dorks up my call routing.

I really need the transfer dial to come from phone A.  What can I do?  I
really dread putting each phone into their own context and parameterizing
their ID...

Any ideas?

Mike.
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[asterisk-users] Strange dropped calls

[Mike Diehl]

Hi all,

I have a user who is reporting dropped calls at his site.  We don't have
any other users complaining of this.

So far, this is what we know:

1.  The manager bought all new Polycom phones. (POE)

2.  They replaced the network switch with a POE version.

3.  It's not just one or two of the phones that have problems.

4.  It doesn't matter if they use the headset or the cordless set.

5.  The ISP reports a very clean circuit.  (Ethernet from the CLEC.)

6.  We don't see their phones become unavailable very often.

7.  They are the only site that seems to be having trouble.

So, where else can/should I look?

Mike.
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[asterisk-users] IAXModem or T38Modem?

[Mike Diehl]

Hi all,

I'm installing Hylafax on my Asterisk system.  From what I've read, I can
either use IAXModem or T38Modem to provide the virtual fax device.  So at
the risk of starting a religious war, which one should I use?

I don't mind running IAX if I have to.  I want as much flexibility and
stability as I can get.

So, what are your recommendations?

Mike.
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Re: [asterisk-users] Oddity with FFA

[Mike Diehl]

Steve,

I appreciate you elaborating on my problem.  I don't suppose this is as
easy as putting a wait(3) in my dial plan before hangup.?  (Didn't
think so.)

Aside from checking (and hoping) for a newer version of FFA that fixes this
issue, I guess there's not much I can do, then.

Thanks again.

Mike.


On Tue, Mar 11, 2014 at 12:27 AM, Steve Underwood ste...@coppice.orgwrote:

 Hi Mike,

 If the sending machine keeps trying it might be the call has been hung up
 by asterisk before its own acknowledgement message has finished being sent.
 There have been bugs like this in the past, and people can be pretty casual
 about making changes which hang up aggressively. A FAX system should really
 wait for the final DCN message before disconnecting, to ensure both sides
 have seen what they need. Spandsp does that, but I am not sure about FFA.

 Regards,
 Steve

 On 03/11/2014 03:03 AM, Mike Diehl wrote:

 Steve,

 I BELIEVE the fax is complete because the fax image is a form that
 appears to only be a single page.

 But, since FFA isn't providing acknowledgement, the sending fax machine
 is resending the document multiple times!

 Mike.


 On Mon, Mar 10, 2014 at 12:49 PM, Steve Underwood ste...@coppice.orgmailto:
 ste...@coppice.org wrote:

 On 03/11/2014 12:36 AM, Mike Diehl wrote:

 Hi all,

 For the most part, we are finding that Fax for Asterisk works
 pretty
 well.  However, we have seen some wierdness that we'd like to
 try to
 fix.

 Once in a while, we will get a partial result report for a
 given fax
 but when we look at the actual .tiff image, it looks to be
 complete.
 This is causing our users to not get a positive
 acknowledgement when
 they send the fax.

 Is there anything we can do to mitigate this?

 Mike.

 How do you know the FAX is complete? If a page was received, the
 sending machine said more pages were to follow, and then it
 dropped the call, is that a complete FAX?

 Steve


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[asterisk-users] Oddity with FFA

[Mike Diehl]

Hi all,

For the most part, we are finding that Fax for Asterisk works pretty
well.  However, we have seen some wierdness that we'd like to try to
fix.

Once in a while, we will get a partial result report for a given fax
but when we look at the actual .tiff image, it looks to be complete.
This is causing our users to not get a positive acknowledgement when
they send the fax.

Is there anything we can do to mitigate this?

Mike.

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Re: [asterisk-users] Oddity with FFA

[Mike Diehl]

Steve,

I BELIEVE the fax is complete because the fax image is a form that appears
to only be a single page.

But, since FFA isn't providing acknowledgement, the sending fax machine is
resending the document multiple times!

Mike.


On Mon, Mar 10, 2014 at 12:49 PM, Steve Underwood ste...@coppice.orgwrote:

 On 03/11/2014 12:36 AM, Mike Diehl wrote:

 Hi all,

 For the most part, we are finding that Fax for Asterisk works pretty
 well.  However, we have seen some wierdness that we'd like to try to
 fix.

 Once in a while, we will get a partial result report for a given fax
 but when we look at the actual .tiff image, it looks to be complete.
 This is causing our users to not get a positive acknowledgement when
 they send the fax.

 Is there anything we can do to mitigate this?

 Mike.

  How do you know the FAX is complete? If a page was received, the sending
 machine said more pages were to follow, and then it dropped the call, is
 that a complete FAX?

 Steve


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[asterisk-users] Call transfer problem.

[Mike Diehl]

Hi all,

I have a user who is having trouble transferring calls, using a
Grandstream GXP2xxx.

Here's the use case that I've seen:

I call the user from phone A and he answers on phone B.

Then, he hits the transfer button on his phone and dials an extension
that is reachable by him, but not by me, based on administrative
policy.

However, the Asterisk logs indicate that the new call is being
initiated by phone A, not phone B!  Thus the call transfer fails.

I have other users, with other phones, that are able to transfer just
fine.  What could be different with this particular user?

Any ideas?

Mike.

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Re: [asterisk-users] Call transfer problem.

[Mike Diehl]

I'm sorry, I should have mentioned that he's doing a phone-based
transfer, not an asterisk-based transfer.

Mike.

On Mon, Feb 24, 2014 at 1:30 PM, Don Kelly d...@donkelly.biz wrote:
 Does he complete the call as a supervised transfer--waits for the called
 party to answer before completing the transfer?

   --Don


 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Diehl
 Sent: Monday, February 24, 2014 12:24 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [asterisk-users] Call transfer problem.

 Hi all,

 I have a user who is having trouble transferring calls, using a Grandstream
 GXP2xxx.

 Here's the use case that I've seen:

 I call the user from phone A and he answers on phone B.

 Then, he hits the transfer button on his phone and dials an extension that
 is reachable by him, but not by me, based on administrative policy.

 However, the Asterisk logs indicate that the new call is being initiated by
 phone A, not phone B!  Thus the call transfer fails.

 I have other users, with other phones, that are able to transfer just fine.
 What could be different with this particular user?

 Any ideas?

 Mike.

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Re: [asterisk-users] h extension isn't processed after call file finishes.

[Mike Diehl]

Matthew,

I don't think I've been as clear as I'd like.

I've got some fax-connected TA's that make outbound calls.  The dial
plan routes those calls to an AGI script that captures the fax image,
the destination phone number, and creates a call file to deliver the
image to the destination.

The first outbound call executes the h extension when it is hung up.
The second call, created by the call file, doesn't execute the h
extension,  even though I use the dialplan to actually route the
outbound call.

So, I'm ending up with statistics on the reception of the fax, but not
the final delivery.

Does that make more sense?

Mike.

On Wed, Feb 19, 2014 at 6:10 PM, Matthew Jordan mjor...@digium.com wrote:
 On Tue, Feb 18, 2014 at 2:13 PM, Steve Edwards
 asterisk@sedwards.com wrote:
 On Mon, 17 Feb 2014, Mike Diehl wrote:

 Is there something I need to do in order to get the h extension to get
 called?


 Would the 'g' dial() option help?

 Proceed with dialplan execution at the current extension if the destination
 channel hangs up.

 It won't take you to h, but it may allow you to do what you need to do --
 even if the next dialplan priority just says 'goto h.'


 I'm actually a bit confused about what channel(s) are executing the
 'h' extension. From the description in OP's e-mail, it sounds as if at
 least one channel is dropping into the 'h' extension, and some
 channels are not. Which channels are they? If it is the outbound
 channel, then since that channel doesn't execute dialplan, it will
 never get put into the 'h' extension, unless you use the Dial
 application's 'e' option. If you want hangup logic and you're using
 Asterisk 11+, you could also use a hangup handler on the outbound
 channel.

 But otherwise, I would expect that the 'h' extension would always be
 fired for a channel executing dialplan, so long as it is in the same
 context.

 --
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 Digium, Inc. | Engineering Manager
 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
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[asterisk-users] h extension isn't processed after call file finishes.

[Mike Diehl]

Hi all,

I'm trying to build a fax relay mechanism where faxes come in and get
relayed out to their final destination.  I'm using the h extension to store
various results from both legs.  This data is being saved correctly for the
first (receiving) leg. The second leg isn't calling the h extension when
it's finished.  The second leg is being initiated by a .call file like:

Channel: local/1505xxx@context
Application: sendfax
Data: /tmp/voice11-voice11-1392668806.182025.tiff,zfds
WaitTime: 90
MaxRetries: 2
Account: vFax
CallerID: Fax 505xxx

The h extension calls an agi scrip that logs a bunch of information about
the fax attempt.  Works just fine when I receive a fax.  But there is no
sign of it in the logs for the sending leg of the fax.

Is there something I need to do in order to get the h extension to get
called?

Mike.
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[asterisk-users] Strange incoming call issue.

[Mike Diehl]

Hi all,

I've got a customer who's reporting ghost calls. Essentially, the phone
rings, they pick up, and there's no body there.

It is NOT one-way audio, and it doesn't happen all the time.

We use voipmonitor to watch calls, and this is what we saw for the call in
question:

| calldate| caller | called | duration | whohanged |
+-++++-+
| 2014-02-12 09:28:06 | 575xxx | CCD539F38...-1 |   60 | NULL  |
| 2014-02-12 09:29:06 | 575xxx | CCD539F38...-2 |1 | NULL  |

So, it looks like my customer received a call, which lasted a minute, and
then they  hung up.  Then their phone rang again, but there was no one
there.
Based on what I'm seeing in my log, the first call was never hung up, even
though both parties claim to have hung up the call.  My logs only indicate
that the 'h' extension was called once, at 9:29:07

My question is, how can a call not get hung up when both parties hang up
the call?  I know that sounds odd, but that's what I'm seeing.

Any ideas?

Mike.
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Re: [asterisk-users] Lots of calls, less memory

[Mike]

On 14-02-10 10:37 AM, Justin Sherrill wrote:
 We're running Asterisk 1.8 on a 32-bit Debian machine, and it has been fine 
 for some time now.  But!  We've got such a incoming call volume over the few 
 weeks that we'll have Asterisk occasionally restart itself.  My hunch is that 
 it is in part memory pressure.


What log entries are leading you to think that you're running out of RAM?

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Re: [asterisk-users] file.c:1160 ast_writefile: Unable to open file /var/spool/asterisk/monitor/11Feb2014/_11-Feb-2014-17-44-01.wav: No such file or directory

[Mike]

On 14-02-11 03:00 AM, akhilesh chand wrote:

 file.c:1160 ast_writefile: Unable to open file
 /var/spool/asterisk/monitor/11Feb2014/_11-Feb-2014-17-44-01.wav: No
 such file or directory
 app_mixmonitor.c:286 mixmonitor_thread: Cannot open
 /var/spool/asterisk/monitor/11Feb2014/_11-Feb-2014-17-44-01.wav


Does the path to that file exist, and can asterisk write to it?


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Re: [asterisk-users] SPA112 Won't stay up

[Mike Diehl]

Based on what we're hearing, we've decided to replace the SPA112.  Thank
you for your input.

Mike.


On Thu, Feb 6, 2014 at 4:39 PM, Andres and...@telesip.net wrote:

  On 2/6/14, 11:18 AM, Mike Diehl wrote:

  Hi all,

  I have an SPA112 that in sitting behind a Ubee cable modem.  The internet
 link is solid, but the device becomes unreachable within a day or so of
 being rebooted.  Then the customer goes to reboot the device, they report
 that all 4 lights are lit.  The ISP reports that the device does respond to
 ping, so it's not completely dead.  I've had the same symptoms with
 SPA303's sitting behind Ubee modems.

 So, is there some configuration setting on the SPA that I can set to make
 this device more stable?

  I have seen similar behavior before on the SPA122.  I could ping it,
 open the web page, etc... but it would not register until I rebooted it.
 Upon closer examination I could see that the SPA122 was only working
 partially.  The voice modules appeared to be dead thus it would not
 register.  You could see this by looking at the stats page and the lines
 would not show any stats at all or even if they were ON or OFF hook.  A
 reboot would fix it for a few days.  The solution was to get a new SPA122.
 My take on this is that it was a hardware issue, not a software one that
 could be fixed with configuration settings.

 What I hate about these units is that they take more that 1 minute to boot
 and register.   The SPA2102 only took about 15 seconds.   That really sucks
 when you have a customer on the line and are troubleshooting an issue that
 requires a reboot.

 Mike.




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[asterisk-users] SPA112 Won't stay up

[Mike Diehl]

Hi all,

I have an SPA112 that in sitting behind a Ubee cable modem.  The internet
link is solid, but the device becomes unreachable within a day or so of
being rebooted.  Then the customer goes to reboot the device, they report
that all 4 lights are lit.  The ISP reports that the device does respond to
ping, so it's not completely dead.  I've had the same symptoms with
SPA303's sitting behind Ubee modems.

So, is there some configuration setting on the SPA that I can set to make
this device more stable?

Mike.
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Re: [asterisk-users] SPA112 Won't stay up

[Mike Diehl]

I've got the registration period set to 15 minutes.  However, I've got
similar devices all over the place that don't seem to have this
unreliability issue.  The way I solved it with the SPA303 that I had in the
office was to replace the Ubee modem with a different make/model.  That's
not an option in this particular case, though.

Mike.


On Thu, Feb 6, 2014 at 11:27 AM, Leandro Dardini ldard...@gmail.com wrote:

 How long is the registration timeout? If the device is behind a
 router/firewall, then you need to set a registration timeout lower than the
 state table life in the router/firewall. I usually set my devices to just
 2 minutes and it works almost all the time. Most Cisco devices have a very
 long timeout of 3600 seconds.

 Leandro


 2014-02-06 17:18 GMT+01:00 Mike Diehl mdiehlena...@gmail.com:

 Hi all,

 I have an SPA112 that in sitting behind a Ubee cable modem.  The internet
 link is solid, but the device becomes unreachable within a day or so of
 being rebooted.  Then the customer goes to reboot the device, they report
 that all 4 lights are lit.  The ISP reports that the device does respond to
 ping, so it's not completely dead.  I've had the same symptoms with
 SPA303's sitting behind Ubee modems.

 So, is there some configuration setting on the SPA that I can set to make
 this device more stable?

 Mike.

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Re: [asterisk-users] SPA112 Won't stay up

[Mike Diehl]

Unfortunately, we plug straight into the Ubee and the ISP will not support
any other modem.

GRRr..

Mike.


On Thu, Feb 6, 2014 at 12:34 PM, David Wessell da...@ringfree.biz wrote:

 Is there another router in the mix? Put the cable modem in bridge mode and
 attAch a real router.

 http://401stblow.wordpress.com/2012/10/21/fixing-time-warner-cable-ubee-modem-connectivity-issues/


 On Thursday, February 6, 2014, Mike Diehl mdiehlena...@gmail.com wrote:

 I've got the registration period set to 15 minutes.  However, I've got
 similar devices all over the place that don't seem to have this
 unreliability issue.  The way I solved it with the SPA303 that I had in the
 office was to replace the Ubee modem with a different make/model.  That's
 not an option in this particular case, though.

 Mike.


 On Thu, Feb 6, 2014 at 11:27 AM, Leandro Dardini ldard...@gmail.comwrote:

 How long is the registration timeout? If the device is behind a
 router/firewall, then you need to set a registration timeout lower than the
 state table life in the router/firewall. I usually set my devices to just
 2 minutes and it works almost all the time. Most Cisco devices have a very
 long timeout of 3600 seconds.

 Leandro


 2014-02-06 17:18 GMT+01:00 Mike Diehl mdiehlena...@gmail.com:

 Hi all,

 I have an SPA112 that in sitting behind a Ubee cable modem.  The
 internet link is solid, but the device becomes unreachable within a day or
 so of being rebooted.  Then the customer goes to reboot the device, they
 report that all 4 lights are lit.  The ISP reports that the device does
 respond to ping, so it's not completely dead.  I've had the same symptoms
 with SPA303's sitting behind Ubee modems.

 So, is there some configuration setting on the SPA that I can set to
 make this device more stable?

 Mike.

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 828-575-0030


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Re: [asterisk-users] IOPS required by Asterisk for Call Recording

[Mike]

On 14-01-29 08:34 AM, Amit wrote:
 Thanks Ron.
 I will try to get these readings. About RAM disk, I will study on how
 to create RAM disk and conduct this test again.
 There is no bottleneck on network.

To create a ramdisk under Linux, assuming you have enough ram -


# mkdir /ramdisk
# mount -t tmpfs tmpfs /ramdisk


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Re: [asterisk-users] IOPS required by Asterisk for Call Recording

[Mike]

On 14-01-25 01:26 AM, Amit wrote:
 250GB SATA disk (No RAID)
If you care enough to record the calls, you should care enough to get
some fast and redundant storage. SSDs would be best,  15K SAS drives
second choice. Even a good RAID10 of SATA drives would help a lot.

A RAID card with battery backed cache would be helpful as well.

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Re: [asterisk-users] IOPS required by Asterisk for Call Recording

[Mike]

On 14-01-24 11:16 AM, Amit wrote:

 If I assume that Asterisk will write data on disk every second for
 each call, I will need disk array to support minimum of 500 IOPS.
 Where as if Asterisk push data every 2 seconds, I can deal with array
 supporting 250 IOPS.
 But if I assume that Asterisk will write data on disk for every RTP
 packet received, as and when received, I will need disk IO system with
 approx 25000 IOPS assuming 20 ms RTP packet.
You're assuming that asterisk will perform an fsync() after each write.
If asterisk writes without an fsync after each write, then the OS will
schedule writes intelligently based on RAM/disk IO available rather than
scheduling each one as a separate write.

Looking at the code for ast_writestream() there doesn't appear to be an
fsync() type call after each write, but someone more familiar with the
internals of Asterisk would be better able to verify that.

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[asterisk-users] SIP Mass exodus

[Mike Diehl]

Hi all,

I've been seeing some strangeness lately on my 10.2.1 server.  It's
gotten to the point that a few times each day, I see masses of SIP
clients becoming unreachable.  They're not all on the same network,
and we don't see any calls drop.  In a few seconds, they all come
back.

I don't think it's a connectivity issue because we don't drop calls,
and the endpoints aren't on the same networks.  We don't see excessive
CPU load when it happens.

It does SEEM to happen most right after someone accesses their voicemail.

We are using RT SIP registration as well as database voicemail storage
(mysql).  The database is on the same machine as the asterisk server.

Have we grown beyond the ability to host both the db and * on the same
hardware?  Or is this a known issue with a (hopefully) known fix?

TIA,

Mike Diehl.

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Re: [asterisk-users] Tired of dropouts and garbled phone calls - where to go next?

[Mike]

On Mon, 28 Oct 2013, Eddie Mikell wrote:


All,
The users in our organization are well, quite frankly, sick of phone service 
that is being provided.  The choppy phone
calls, and drop outs are detrimental to our sales force.

I've tried about everything I can think of.  

  Moved the asterisk server from VM machine to dedicated machine

  More than enough bandwidth

  Setting 802.1p = 7

  Set Dedicated voice traffic 35% of bandwidth.

Not sure what option would be the best

  Put analog lines in the conference room to avoid the dropouts - leave the 
sip lines in place for day to day use

  Hire a consultant

  Ditch the system and buy a pre-packaged system - RingCentral or some such.

There are no local asterisk professionals who can help, and we are a little 
leery of opening up our system to outside
consultants.

Anyone else face the above, and finally abandoned Asterisk for a commercial 
system?  

We have 167 users.
I use Grandstream GXP 2100 on the desktop and Polycom ip6000 for the conference 
rooms.

Suggestions welcome.

Best

Eddie
--


As stated in previous replies if you haven't already I would certainly try 
to isolate the problem, e.g., are extension to extension calls good, is 
the problem only on outside calls etc.


We are starting our 4th year of VoIP service and have had two seemingly 
similar episodes to yours during that time.  We are on a non-symmetric 
cable connection, 20/4 (I believe).  After a few days of crappy audio I 
started looking for some way to characterize/correlate bad audio with 
something I could measure.  I found iperf (http://iperf.sourceforge.net/) 
to be a free and easy starting point, which actually turned out to be all 
I needed.


I simply ran a server instance on our cloud server roughly 1K miles 
away and a client instance locally.  I used the command line swithces 
that forced udp mode.  This allowed me to see jitter and packet loss in 
both directions.  We had terrible packet loss in the outbound direction. 
This didn't show up in normal browsing, emailing etc., kinds of things as 
I suspect TCP retries masked the problem.  With a little persistence with 
the cable company the second tech found a bad tap (I believe) outside at 
the cable drop.  Replacing that solved our issue for almost two years.


The next time this happened iperf showed a similar packet loss problem. 
This time it turned out to be noise in the system according to the cable 
tech.  He said it could be from any number of sources but a different team 
would be out to hunt it down the next day.  In the mean time he changed 
out our old Moto SB5101 modem for a more modern DOCSIS 3.0 modem.  The 
multiple channel bonding that it offered was much better at punching 
through the noise.  That change alone ended crappy audio as well as packet 
loss as shown by iperf.-- 
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Re: [asterisk-users] Access PBX from internet - best practice

[Mike]

On 13-10-17 08:13 AM, richard.seg...@marisec.ca wrote:
 The endpoints do not have a fixed IP, and a VPN tunnel wouldn't work under 
 this scenario.  Basically this setup is for people who are traveling, and may 
 be using a smart phone at an airport (or something similar).  The idea is 
 that our system can be used to reduce toll costs, and provide access to 
 internal resources. 


A VPN would be perfect for this situation - you certainly don't need
fixed IPs on the endpoints. I quite happily pass calls over my VPN from
my smartphone.

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[asterisk-users] Grnvoip

[Mike Diehl]

Does anyone know if Grnvoip is still in business, or what's going on with
them?  I had an account with them, but they no longer terminate calls.

Mike.
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