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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> > 
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> >   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
> > 
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> > 
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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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> > 
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> >   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
> > 
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> > 
> > To UNSUBSCRIBE or update options visit:
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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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[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] 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.


-- 
Mike Diehl



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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.

-- 
Mike Diehl


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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] 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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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  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  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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>>
>> 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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>
>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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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  wrote:
>On Thu, Jul 20, 2017 at 11:50 AM, mdiehl 
>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,

-- 
Mike Diehl


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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] 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] 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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[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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[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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_
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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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_
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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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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?

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

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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.


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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!

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Mike Diehl
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Fax: (505) 903-5701
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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.

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

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[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] 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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[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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[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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[asterisk-users] VM notification to multiple email recipients

[Mike Diehl]

Hi all,

I've got a user who wants to receive voicemail notifications at two
different email addresses.  I could probably setup an alias in
/etc/aliases, but then I'd have to manage that across multiple servers,
which I don't want to do.

Is there a way I can tell Asterisk to send to multiple addresses?

Mike
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Re: [asterisk-users] VM notification to multiple email recipients

[Mike Diehl]

OK, and to make things even more difficult, I store my voicemail and
voicemail configuration in MySql.

Looks like, for now, I will be creating aliases in /etc/aliases and
sync'ing that across my servers

Thank you for your suggestions.

Mike.


On Wed, Sep 11, 2013 at 12:14 PM, Carlos Rojas crt.ro...@gmail.com wrote:

 Hi

 You can do this,
 http://mike.eire.ca/2012/02/03/asterisk-1-8-vm-multiple-emails/

 If you are using asterisk 1.8


 On Wed, Sep 11, 2013 at 1:55 PM, Mike Diehl mdiehlena...@gmail.comwrote:

 Hi all,

 I've got a user who wants to receive voicemail notifications at two
 different email addresses.  I could probably setup an alias in
 /etc/aliases, but then I'd have to manage that across multiple servers,
 which I don't want to do.

 Is there a way I can tell Asterisk to send to multiple addresses?

 Mike

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[asterisk-users] Cisco SPA303 won't ring for more than 60 seconds

[Mike Diehl]

Hi all,

I've got a user with a couple of Cisco SPA303's.  When I dial their phones
with a dial string like:

dial(sip/phone-a,300,rwkxttT)

The phone rings, as expected.

However after exactly 60 seconds, I get:

[Aug 21 02:09:56] -- Got SIP response 480 Temporarily not available
back from a.b.c.d:5062
[Aug 21 02:09:56] -- SIP/phone-a-6a9d is circuit-busy
[Aug 21 02:09:56]   == Everyone is busy/congested at this time (7:0/6/1)

Clearly I'm asking the phone to ring for 5 minutes, but it's giving up
after 1. Does anyone know how to fix this?

TIA,

Mike.
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Re: [asterisk-users] Cisco SPA303 won't ring for more than 60 seconds

[Mike Diehl]

Does anyone know what knob I need to turn to adjust how long the phone will
ring before giving up?

Mike.


On Wed, Aug 21, 2013 at 8:18 AM, Eric Wieling ewiel...@nyigc.com wrote:

 Asterisk is not timing out.   The phone is rejecting the call after 60
 seconds.   This is a phone configuration issue.

 -Original Message-
 From: asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Diehl
 Sent: Wednesday, August 21, 2013 4:15 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [asterisk-users] Cisco SPA303 won't ring for more than 60 seconds

 Hi all,


 I've got a user with a couple of Cisco SPA303's.  When I dial their phones
 with a dial string like:


 dial(sip/phone-a,300,rwkxttT)

 The phone rings, as expected.

 However after exactly 60 seconds, I get:

 [Aug 21 02:09:56] -- Got SIP response 480 Temporarily not available
 back from a.b.c.d:5062
 [Aug 21 02:09:56] -- SIP/phone-a-6a9d is circuit-busy
 [Aug 21 02:09:56]   == Everyone is busy/congested at this time (7:0/6/1)


 Clearly I'm asking the phone to ring for 5 minutes, but it's giving up
 after 1. Does anyone know how to fix this?

 TIA,

 Mike.


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[asterisk-users] 811

[Mike Diehl]

Hi all,

I have a customer that tried to use the Texas One-Call number (a
toll-free call) to have the utility company come out and mark buried
pipes and cables.  That call resulted in a recording telling her to
dial 811, instead.

So, as a service provider, how do I terminate a call to 811?  In NM, I
send it to NM's One call local number.  I wasn't able to find such a
number for TX.

Is there a list somewhere?

How do other people handle this situation?

Mike

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Re: [asterisk-users] 811

[Mike Diehl]

Fantastic!  Thank you!

Mike.

On Thu, Aug 15, 2013 at 3:21 PM, Shane Young asteri...@shaneyoung.com wrote:
 Quoting Mike Diehl mdiehlena...@gmail.com:



 Is there a list somewhere?


 There is a list by state here:
 http://www.call811.com/state-specific.aspx



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[asterisk-users] Backing up DB kills RT peers

[Mike Diehl]

Hi all,

I had a scary thing happen to me when I did my database backup last
night.  Within seconds of starting the backup script, ALL of my real
time sip peers disappeared!

All of my voicemailboxes, voicemail messages, and sip peers are in a
Mysql database.

I was able to do a database backup 3 days ago without any incedent.
Now suddenly, the whole thing comes crashing down.

I also notice that while the backup is running, my other queries
block, which is probably why my peers disappear.

I'll be posting to the mysql list as well, but has any Asterisk user
seen this before?

TIA,

Mike Diehl.

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Re: [asterisk-users] Strange issues with newly rebooted machine

[Mike Diehl]

Yes, it should have, but I don't think it's turned on  by default...
Anyway, this lower MTU setting is just temporary, so I'll just live
with it for now.

Thanks for your assistance.

Mike.

On Wed, Aug 7, 2013 at 2:28 AM, Steven Howes steve-li...@geekinter.net wrote:
 On 6 Aug 2013, at 19:28, Mike Diehl wrote:
 We got it fixed!  Our co-lo is in the process of doing a network
 reconfiguration/relocation and had changed their MTU to 1400 during
 the transition.  Once we did the same, everything started to work.

 PMTU should take care of that. Are you blocking ICMP somewhere?

 S

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[asterisk-users] Strange issues with newly rebooted machine

[Mike Diehl]

Hi all,

After being up and running for almost 2 years, we finally had to
reboot one of our servers.  Now, however, it's having problems.

We're using real-time configuration for SIP peers and voicemail, via ODBC.

But when I run sip show peers I don't get anything.  Even when I
load a known peer by name, nothing happens:

sip show peer voice12 load

This command just returns, with no output.

It did occur to me that I might be having a problem with ODBC.
However, when I show the status, I get good results:

 odbc show db

ODBC DSN Settings
-

  Name:   db
  DSN:db
Last connection attempt: 1969-12-31 17:00:00
  Pooled: No
  Connected: In use

I'm using 10.2.1.  Also, I've noticed that tab command completion
doesn't work on the Asterisk console.

Any ideas what is wrong here?

Mike Diehl.

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Re: [asterisk-users] Strange issues with newly rebooted machine

[Mike Diehl]

I appreciate your quick response.  I issued the commands specified and
got NO output!

===
CLI core set verbose 10
Verbosity was 25 and is now 10
CLI core set debug 10
Core debug was 25 and is now 10
CLI module unload chan_sip.so
CLI module load chan_sip.so
CLI
===

The reason we had to reboot the machine is that we changed it's
physical location, but didn't change it's IP address.  As part of the
restart, I also took the opportunity to rebuild a RAID-1 array.  Other
than that, there have been no configuration changes since the last
time this worked.

Any other ideas?

Mike


On Tue, Aug 6, 2013 at 4:36 AM, Jeremy Kister
asterisk...@jeremykister.com wrote:
 On 8/6/13 5:30 AM, Mike Diehl wrote:

 sip show peer voice12 load

 This command just returns, with no output.


 throwing out a random idea since it's early in the morning and you might be
 in a big jam...

 assuming the sip isnt working correctly at all (and its not just a console
 issue),

 after asterisk is started, perhaps try core set verbose 10, core set debug
 10, module unload chan_sip.so, and module load chan_sip.so .  if there are
 any errors loading the module it may be easy to spot them.

 --

 Jeremy Kister
 http://jeremy.kister.net./


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Re: [asterisk-users] Strange issues with newly rebooted machine

[Mike Diehl]

No, my phones aren't getting a response from the server.  I can't even
get any output from the server if I do:

sip show peer name load

This command usually loads the peer from the db and shows me it's
configuration.  In this case, I get nothing.

I do have rtcachefriends=yes in my sip.conf.  In fact, this server has
a virtually identical configuration to one that is already running.
(I sync the configurations using unison.)

I don't THINK this is a configuration issue.  Any ideas, though?

Mike.

On Tue, Aug 6, 2013 at 10:08 AM, Warren Selby wcse...@selbytech.com wrote:
 On Tue, Aug 6, 2013 at 10:47 AM, Mike Diehl mdiehlena...@gmail.com wrote:

 I appreciate your quick response.  I issued the commands specified and
 got NO output!

 ===
 CLI core set verbose 10
 Verbosity was 25 and is now 10
 CLI core set debug 10
 Core debug was 25 and is now 10
 CLI module unload chan_sip.so
 CLI module load chan_sip.so
 CLI
 ===

 The reason we had to reboot the machine is that we changed it's
 physical location, but didn't change it's IP address.  As part of the
 restart, I also took the opportunity to rebuild a RAID-1 array.  Other
 than that, there have been no configuration changes since the last
 time this worked.

 Any other ideas?


 Are the phone still working?  I've noticed that realtime registered peers
 don't always show when I do sip show peers or even sip show peer name.
 I usually only see the peer if I make a call to the peer or the peer makes a
 call first.

 Do you have rtcachefriends=yes in your sip.conf?

 --
 Thanks,
 Warren Selby, dCAP
 http://www.SelbyTech.com


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Re: [asterisk-users] Strange issues with newly rebooted machine

[Mike Diehl]

We got it fixed!  Our co-lo is in the process of doing a network
reconfiguration/relocation and had changed their MTU to 1400 during
the transition.  Once we did the same, everything started to work.

Thank you all for your time and quick responses.

Mike.

On Tue, Aug 6, 2013 at 10:44 AM, Tim Nelson tnel...@rockbochs.com wrote:
 - Original Message -
 No, my phones aren't getting a response from the server.  I can't
 even
 get any output from the server if I do:

 sip show peer name load

 This command usually loads the peer from the db and shows me it's
 configuration.  In this case, I get nothing.

 I do have rtcachefriends=yes in my sip.conf.  In fact, this server
 has
 a virtually identical configuration to one that is already running.
 (I sync the configurations using unison.)

 I don't THINK this is a configuration issue.  Any ideas, though?


 It sounds like Asterisk is hung in general. Next step, stop asterisk 
 altogether, edit your /etc/asterisk/logger.conf to output all to a logfile:

 full = notice,warning,error,debug,verbose,dtmf

 Then, do a 'tail -F /var/log/asterisk/full', and startup Asterisk.

 I'm guessing you'll be able to see some errors flow by, but more importantly, 
 maybe the log will stop, showing you exactly what is hanging.

 Good luck!

 --Tim

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[asterisk-users] Can' correlate AMI MonitorStart with CDR

[Mike Diehl]

Hi all.

I'm trying to write an AMI handler to take care of cases where a user
uses the feature code to start recording a call.

When a user starts a recording, my AMI listener receives this event:

$VAR1 = {
  'Server' = 'example.com',
  'Event' = 'MonitorStart',
  'Uniqueid' = 'server-1374906101.132305',
  'Privilege' = 'call,all',
  'Channel' = 'SIP/trunk-000200d0'
};

However, when I then go and look up that call in the cdr table using
Uniqueid = 'server-1374906101.132305', my query doesn't find anything.

If I then ask the cdr database for the Uniqueid of the call in
question, I get something slightly different, such as:

server-1374906100.132304

In general, it seems that the two strings only differ in the last character.

So, how am I supposed to correlate this?

TIA,

Mike Diehl.

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[asterisk-users] Turning off CFWD on an SPA112?

[Mike Diehl]

Hi all,

I'm not sure how this happened, but one of my customers managed to
turn call forwarding on on his spa112.  I thought I had that turned
off in the provisioning file.

I have this in the provisioning file:

Cfwd_All_Serv_1_No/Cfwd_All_Serv_1_
Cfwd_Busy_Serv_1_No/Cfwd_Busy_Serv_1_

And I have a similar entry for line 2.

When I dial the device, I use this line to dial it:

exten = 888,n,dial(sip/000E084B8xxx-1,60,Trit)

Here is what I get on the server console:

[Jul 19 15:49:42] -- Called sip/000E084B8xxx-1
[Jul 19 15:49:42] -- Got SIP response 302 Moved Temporarily back
from 99.114.250.142:5060
[Jul 19 15:49:42] -- Now forwarding SIP/3CCE73D31786-1-00017bdb to
'Local/2146647512@magic' (thanks to SIP/000E084B8xxx-1-00017bdc)
[Jul 19 15:49:42] -- Forwarding SIP/3CCE73D31xxx-1-00017bdb to
'Local/214xx@magic' prevented.
[Jul 19 15:49:42]   == Everyone is busy/congested at this time (1:1/0/0)
[Jul 19 15:49:42] -- Auto fallthrough, channel
'SIP/3CCE73D31xxx-1-00017bdb' status is 'BUSY'

So, my question is, how do I turn this feature off on this device?  My
customer really isn't in a position to be able to plug a handset into
the device and dial *73.  Can this be disabled from the provisioning
file?  Is there anything else I can do to prevent this?

TIA,

Mike Diehl.

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Re: [asterisk-users] Upgrade from 10.2.4 to 11.4.x, the Reader's Digest version?

[Mike Diehl]

Thank you!  That was very helpful.

Mike.

On Wed, Jul 10, 2013 at 7:38 PM, Matthew Jordan mjor...@digium.com wrote:

 On Wed, Jul 10, 2013 at 5:35 PM, Mike Diehl mdiehlena...@gmail.com wrote:

 Hi all,

 I'm contemplating an upgrade from 10.2.4 to 11.4.x.  However, the
 1.8.x to 10.4.x upgrade was painful; some of the modules had been
 renamed, if I recall correctly.

 So, is there a list of MAJOR changes and GOTCHA's between 10.x and
 11.x?  I'm hoping for something a little less granular than the
 release notes from 10.2.x to 11.4.x.  I don't mind reading, but that
 is almost as long as War and Peace!

 Does such a document exist, or do I need to start reading..


 Upgrade notes:

 https://wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+11

 While the upgrade notes cover changes to configuration and module status, it
 is also a good idea to read through what is new:

 https://wiki.asterisk.org/wiki/display/AST/New+in+11

 I wouldn't say it is War and Peace, but yes, there is some content in
 there.

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

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[asterisk-users] Upgrade from 10.2.4 to 11.4.x, the Reader's Digest version?

[Mike Diehl]

Hi all,

I'm contemplating an upgrade from 10.2.4 to 11.4.x.  However, the
1.8.x to 10.4.x upgrade was painful; some of the modules had been
renamed, if I recall correctly.

So, is there a list of MAJOR changes and GOTCHA's between 10.x and
11.x?  I'm hoping for something a little less granular than the
release notes from 10.2.x to 11.4.x.  I don't mind reading, but that
is almost as long as War and Peace!

Does such a document exist, or do I need to start reading..

TIA,

Mike Diehl.

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[asterisk-users] Converting from FXO to SIP?

[Mike Diehl]

I have a customer who has an analog PBX that is able to be put in
away mode such that when an inbound call comes in, it rings their
cordless phone. This lets them leave the desk without risking missing
a call.

However, we'd like to find a black box that would act like the
cordless phone as far as the PBX was concerned, but instead of ringing
a handset, we want it to dial an extension on our SIP network. This
will allow us to handle the call much more flexibly than their PBX
can.

I was thinking that a TA with an FXO port might do the trick. But, I'm
not sure how to get the device to redirect an incoming call on the FXO
port to a sip destination. Is this something that gets done in the
device's dialplan?

Does anyone have any insight into how to do this?

TIA,

Mike Diehl

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Re: [asterisk-users] Converting from FXO to SIP?

[Mike Diehl]

Thank you!

Mike.

On Tue, Jul 2, 2013 at 1:37 PM, Administrator TOOTAI ad...@tootai.net wrote:
 Le 02/07/2013 21:06, Mike Diehl a écrit :

 [...]

 I was thinking that a TA with an FXO port might do the trick. But, I'm
 not sure how to get the device to redirect an incoming call on the FXO
 port to a sip destination. Is this something that gets done in the
 device's dialplan?

 Does anyone have any insight into how to do this?


 Hi Mike,

 you can do this without problem with any ATAs GW like Linksys SPA3102,
 SIPURA 3000, Tiger G102 aso. You just have to redirect FXO port to a SIP
 extension in the device and it's done. You can even redirect outgoing calls
 to this extension, for urgency calls like police, fire service, etc.

 Beronet (Berofix device) and others manufacturers have also gateways with
 FXO modules that you can use to fill your needs.

 --
 Daniel

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Re: [asterisk-users] AudioCodes MP-112

[Mike Diehl]

Funny you should ask!  I have an MP-202 in front of me right now that I'm
working on.  When I get it working, I'll let you know.  In the mean time,
what symptoms are you getting?

Mike Diehl.

On Mon, Jul 1, 2013 at 4:07 PM, David Wessell da...@ringfree.biz wrote:

   Does anyone have experience setting up an AudoCodes MP-X with an
 asterisk (FreePBX based) system? I would be willing to pay a reasonable
 amount for assistance with the MP-X device. I have remote access setup, so
 no one should have to leave their comfy chair..

  Thanks
 David


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[asterisk-users] Questions about sRTP

[Mike Diehl]

Hi all,

I'm getting ready to setup SIP/TLS and SRTP.  But I have a few questions.
The first one is that I was reading an article at:

https://supportforums.cisco.com/docs/DOC-15381

That indicated that Asterisk doesn't support TLS as an OPTIONAL transport.
It's either all or nothing.  Specifically, this is what it said:

==
*Note: There is no optional SRTP mode in Asterisk, i.e. if encryption is
active on peer, it will not accept non-ciphered audio and viceversa. On the
IP phones, however, it is possible to have unsecure calls if the other peer
does not support SRTP, i.e. incoming calls may work, but not outgoing
calls. This is an Asterisk limitation (Snom supports also the
“optional”mode on SRTP sending two m=audio attributes, but Asterisk does
not know how to handle those descriptors).*
==

This is from a quite dated article (2011), so I'm hoping that I newer
versions of Asterisk will fall back on plaintext if TLS isn't available for
some reason.

Secondly, is there any way to detect if a call is secure from inside the
dialplan or AGI script?

I think that's all for now.

Thanks in advance,

Mike Diehl.
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Re: [asterisk-users] Questions about sRTP

[Mike Diehl]

On Thu, Jun 20, 2013 at 2:05 PM, Joshua Colp jc...@digium.com wrote:

 Mike Diehl wrote:

 Hi all,

 I'm getting ready to setup SIP/TLS and SRTP.  But I have a few
 questions.  The first one is that I was reading an article at:

 https://supportforums.cisco.com/docs/DOC-15381

 That indicated that Asterisk doesn't support TLS as an OPTIONAL
 transport.  It's either all or nothing.  Specifically, this is what it
 said:


 Your statement is incorrect. Asterisk supports TLS as an optional
 signaling transport (although if you do SDES SRTP without it then someone
 can snoop on your keys and ultimately decrypt your media).

 What it does not support is optional *SRTP*. If a device requests SRTP and
 it's not possible, the call will fail.


So then, is it safe to say that Asterisk will ALLOW a secure phone call,
but the client hast to REQUEST it?

I understand that requesting SRTP without SIP/TLS is evil; I just
misunderstood what I was reading.

I'm also thinking that the AGI script I use to route calls can check if
either leg of a call comes from or goes to port 5061 and play a sound file
to indicate that the cal is 'secure.'  Does this seem reasonable?

Thanks,

Mike.
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[asterisk-users] My new Polycom 450's can't xfer to 4-digit extension

[Mike Diehl]

=
dialplan.6.routing.server.1.port=/
 emergency dialplan.1.routing.emergency.1.value=
dialplan.1.routing.emergency.1.server.1=
dialplan.2.routing.emergency.1.value=
dialplan.2.routing.emergency.1.server.1=
dialplan.3.routing.emergency.1.value=
dialplan.3.routing.emergency.1.server.1=
dialplan.4.routing.emergency.1.value=
dialplan.4.routing.emergency.1.server.1=
dialplan.5.routing.emergency.1.value=
dialplan.5.routing.emergency.1.server.1=
dialplan.6.routing.emergency.1.value=
dialplan.6.routing.emergency.1.server.1=/
  /routing
   /dialplan

The MAC-specific provisioning file does not have a dialplan  section.

All of my Polycom users share these files and many of them can
transfer to 4-digit extensions.  Is there something I need to do for
the 450 to make this work?

Thank you in advance.

Mike Diehl.

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[asterisk-users] Getting DIALSTATUS via agi

[Mike Diehl]

Hi all,

Hopefully, I just need a second set of eyes on this one, but I just can't
figure out what I'm doing wrong.  I'm using an agi script to dial a number,
check the dial result, and act accordingly.

The problem is that I'm not getting anything back from DIALSTATUS, or
HANGUPCAUSE.

Here is the relevant perl code:

===

$main::agi-verbose(
, 1);
$main::agi-exec(dial,
sip/$t-{trunk}/$t-{prefix}$o-{number},240,KrTWX);
$result = $main::agi-get_variable(DIALSTATUS);
$cause = $main::agi-get_variable(HANGUPCAUSE);

$o-{DIALSTATUS} = $result;
$o-{HANGUPCAUSE} = $cause;
$main::agi-verbose(= The
result of the last call was ($o-{DIALSTATUS}) ($o-{HANGUPCAUSE}), 1);
$main::agi-verbose(
, 1);

===

Here is the relevant output with agi debug turned on:

===

[Apr  1 15:01:55] SIP/CCEF485BC606-2-7787AGI Tx  200 result=0
[Apr  1 15:01:55] SIP/CCEF485BC606-2-7787AGI Rx  VERBOSE

 1
[Apr  1 15:01:55]  magic_switch.pl:


[Apr  1 15:01:55] SIP/CCEF485BC606-2-7787AGI Tx  200 result=1
[Apr  1 15:01:55] SIP/CCEF485BC606-2-7787AGI Rx  EXEC dial
sip/didlogic/15058457900,240,KrTWX
[Apr  1 15:01:55] -- AGI Script Executing Application: (dial) Options:
(sip/didlogic/15058457900,240,KrTWX)
[Apr  1 15:01:55]   == Using SIP RTP TOS bits 24
[Apr  1 15:01:55]   == Using SIP RTP CoS mark 5
[Apr  1 15:01:55] -- Called sip/didlogic/15058457900
[Apr  1 15:01:55] -- Got SIP response 603 Declined back from
178.63.143.236:5060
[Apr  1 15:01:55] -- SIP/didlogic-7788 is busy
[Apr  1 15:01:55]   == Everyone is busy/congested at this time (1:1/0/0)
[Apr  1 15:01:55] SIP/CCEF485BC606-2-7787AGI Tx  200 result=0
[Apr  1 15:01:55] SIP/CCEF485BC606-2-7787AGI Rx  GET VARIABLE
DIALSTATUS
[Apr  1 15:01:55] SIP/CCEF485BC606-2-7787AGI Tx  200 result=1 (BUSY)
[Apr  1 15:01:55] SIP/CCEF485BC606-2-7787AGI Rx  GET VARIABLE
HANGUPCAUSE
[Apr  1 15:01:55] SIP/CCEF485BC606-2-7787AGI Tx  200 result=1 (21)
[Apr  1 15:01:55] SIP/CCEF485BC606-2-7787AGI Rx  VERBOSE
= The result of the last call was () () 1
[Apr  1 15:01:55]  magic_switch.pl: = The
result of the last call was () ()
[Apr  1 15:01:55] SIP/CCEF485BC606-2-7787AGI Tx  200 result=1
[Apr  1 15:01:55] SIP/CCEF485BC606-2-7787AGI Rx  VERBOSE

 1
[Apr  1 15:01:55]  magic_switch.pl:



===



So, the result of my previous AGI command was 200.  Then I can clearly see
that the result of this dial is BUSY/21, which I understand.  But, my perl
script isn't getting this value.  I do the assignment to/from $result and
$cause because I use those values later in the script, but have to pass
them back to a global routine, as well.

Can anyone see what I'm doing wrong?

Mike Diehl.
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[asterisk-users] Long voicemails not being stored in database

[Mike Diehl]

Hi all,

I've been using mysql, via odbc,  to store voicemails for some time now. 
However, we've recently noticed that when someone leave a message of about
45 seconds or loinger, the message doesn't get stored in the database.  We
do, however, receive the email notification with attachment intact.

My server is pretty busy, so I'm not really able to get a console output,
untill much later tonight, perhaps.

I'm thinking it's a tuning parameter on the mysql database that sets the
size of a blob, but I don't know.

Any help would be appreciated.

Mike Diehl.
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Mike Diehl.

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Re: [asterisk-users] Long voicemails not being stored in database

[Mike Diehl]

I went and checked; my database has the recording field defined as a
longblob.

Any other ideas would be most appreciated.

Mike Diehl.


Danny Nicholas da...@debsinc.com wrote:
 IIRC blobs are normally set to a limit of 65 Kb.  You may need to redefine
 as medium blob (16Mb) or long blob (4 Gb).  28 seconds at 44 Khz takes up
 around 2.5 Mb.
 
  
 
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Diehl
 Sent: Tuesday, November 06, 2012 3:16 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [asterisk-users] Long voicemails not being stored in database
 
  
 
 Hi all,
 
 I've been using mysql, via odbc,  to store voicemails for some time now.
 However, we've recently noticed that when someone leave a message of about
 45 seconds or loinger, the message doesn't get stored in the database.  We
 do, however, receive the email notification with attachment intact.
 
 My server is pretty busy, so I'm not really able to get a console output,
 untill much later tonight, perhaps.
 
 I'm thinking it's a tuning parameter on the mysql database that sets the
 size of a blob, but I don't know.
 
 Any help would be appreciated.
 
 Mike Diehl.
 
   _  
 
 
 Take care and have fun,
 Mike Diehl. 
 
 


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Mike Diehl.



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[asterisk-users] Failover router recommendation

[Mike Diehl]

I hope no one considers this off topic...

I have a phone customer who wants 2 Internet connections so that if one
goes down, he can use the other for phone service.

So, I'd like to get a recommendation for a relatively inexpensive router
that can perform this function.

Also, when the failover occurs, the phone's IP address will obviously
change.  So, how can/should I configure this to minimize my customer's
down-time?

TIA,

Mike Diehl.
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[asterisk-users] AMI disconnects

[Mike Diehl]

Hi all.

I've got a perl script that connects to Asterisk's management interface using 
Asterisk::AMI.  So far, its proven to be very useful. 

I'm hoping to use this to detect and respond to asterisk restarts and sip 
reloads.

However, my script gets disconnected quite frequently, causing false alarms in 
my monitoring.

Here's what the code looks like:

=
while ($more) {
$change = 0;

while (!$ami-connected(5)) {
Event::Shutdown(Event=Shutdown);
send_to_subscribers( {Event = Shutdown, customer_id = 
0}, Connection to Asterisk lost on $server);

$change++;
sleep(1);

ami_connect();

next;
}

$watchdog = 10;
while ($ami-get_event(1)  $watchdog--) {
if ($watchdog  2) { print Watchdog timer too short.\n; }
$change++;
}

Do other stuff from here.

if ($change == 0) {
sleep(2);
}

}

exit 1;
=

When this scrip disconnects, the asterisk console simply says that the manager 
logged off from 127.0.0.1 and got logged back on.  No reall 
error indication given.

Any suggestions on how to make this script keep it's connection?

TIA.


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Re: [asterisk-users] AMI disconnects

[Mike Diehl]

On Thursday 03 May 2012 1:47:09 pm Paul Belanger wrote:
 On 12-05-03 01:45 PM, Mike Diehl wrote:
  Hi all.
  
  I've got a perl script that connects to Asterisk's management interface
  using Asterisk::AMI.  So far, its proven to be very useful.
  
  I'm hoping to use this to detect and respond to asterisk restarts and sip
  reloads.
  
  However, my script gets disconnected quite frequently, causing false
  alarms in my monitoring.

  When this scrip disconnects, the asterisk console simply says that the
  manager logged off from 127.0.0.1 and got logged back on.  No reall
  error indication given.
  
  Any suggestions on how to make this script keep it's connection?
 
 This belongs on the asterisk-users list.  

I posted this to the asterisk-users list...  Did I miss something?

 However, something like StarPY
 (Python) has this functionality by using the Twisted.  It simply
 monitors events on the AMI, and if the connect break, can be setup to
 re-connect.

My code doesn't have any problems getting re-connected in a timely fashion.  
The issue is that it's getting disconnected for no apparent reason!

-- 

Take care and have fun,
Mike Diehl.

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[asterisk-users] No UDPTL ports remaining

[Mike Diehl]

Hi all,

Lately, I've been seeing more and more instances where I get a flood of warning 
messages like this:

[Apr 26 14:09:50] WARNING[21054] udptl.c: No UDPTL ports remaining

The next thing I know, my server is dropping calls and starting to misbehave.

I use fax via T.38, so I can't just turn udptl off.  I could expand the port 
range, but I suspect that will just mask the situation.

What can I do to prevent this from happening?

TIA,

-- 

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Re: [asterisk-users] Transfer to fax

[Mike Diehl]

So I'm still trying to get this to work... (I'm top posting, but the details 
are below, if you want/need background info)

I'd like Asterisk to detect incoming faxes and redirect them elsewhere.  The 
details aren't important, as long as I get the detection working. 

I've added this to my sip.conf file.  Probably overkill, but I'll tune it once 
it works:

[general]
faxdetect=both

My sip registrations are all in a Mysql RT database, so I added this column to 
my table:

faxdetect char(3) default 'no'

I've set faxdetect to 'yes' for the devices that I expect to be receiving fax 
calls.

I did a sip reload from the console after adding and updating this column.

I've added a fax extension to the appropriate context in extensions.conf:
exten = fax,1,noop(I hear a fax!)

Since most of my dialplan is in an AGI script, I've added this to the code 
that handles my test number:

$main::agi-answer();
$main::agi-exec(ringing);
$main::agi-exec(wait,5);


So, now that all of this is in place, I call the extension from my fax 
machine... and I don't get any indication on the console that Asterisk heard a 
fax.  My extension simply rings and I answer it.

What am  missing?

TIA,
Mike Diehl.


On Friday 24 February 2012 4:22:07 pm Kevin P. Fleming wrote:
 On 02/24/2012 05:20 PM, Mike Diehl wrote:
  On Friday 24 February 2012 4:06:19 pm Kevin P. Fleming wrote:
  On 02/24/2012 05:00 PM, Mike Diehl wrote:
  On Friday 24 February 2012 3:17:04 pm Mike Diehl wrote:
  On Friday 24 February 2012 2:39:48 pm Kevin P. Fleming wrote:
  On 02/24/2012 03:32 PM, Mike Diehl wrote:
  Hi all,
  
  I've got a user that has one phone number an wants to be able to us
  it for both voice and fax.
  
  When a fax call comes in, he wants to do some incantation on the
  keypad and have the call go to the fax machine.
  
  As I see it, he has 3 options.
  
  1.  (blind?) Transfer it to the fax extension.
  
  2.  Use features.conf to create a key sequence, say *2, to
  dial/transfer to a fax extension.
  
  3.  Use fax detect (SIP) to do it automatically.  However I'm also
  using FFA, so I suspect these are mutually exclusive.
  
  They are not. Enabling faxdetect should do exactly what you want; it
  will redirect the call to the 'fax' extension in the current context,
  and you can then Dial() your FAX machine (or send the call to
  ReceiveFAX).
  
  Thank you.  Then, that's what I'll do.
  
  On second though, I think my suggestion that FFA and fax detect were
  mutually exclusive stemmed from the idea that a call that was being
  originated/answered/handled by FFA would have it's call disconnected
  and redirected by fax detect.
  
  If this is the case, it changes my dial plan logic, and I'm not sure I
  fully understand what changes I'll need to make.
  
  For all I know, it might even simplify things by isolating all fax
  handling in one block.
  
  Well, first you should not have faxdetect enabled on outbound channels.
  That takes care of the 'originating' part.
  
  If you have an inbound channel that you *know* you are sending to
  ReceiveFAX, then you can just disable faxdetect on that channel before
  doing so (this is why we made 'faxdetect' configurable from the
  dialplan). Alternatively, you can just let calls that you *know* are
  going to go to ReceiveFAX (dedicated FAX DIDs, for example) just 'idle'
  in the dialplan listening to silence until faxdetect kicks in and sends
  them to ReceiveFAX.
  
  Note that the usage of FFA is not relevant here; whether you are using
  Fax for Asterisk, the free version of it, or res_fax_spandsp, the
  behavior and scenarios would be the same.
  
  Very nice.
  
  Sounds like I need to add a faxdetect column to my SIP real-time
  configuration. Once I've done a sip reload or pruned/loaded my user
  agents, I should be good to go!
  
  Didn't know faxdetect was configurable in the dialplan...  Pointer to how
  to do it?
 
 The CHANNEL() dialplan function with the 'faxdetect' option. Not sure
 which releases have it; it might only be Asterisk 10.

-- 

Take care and have fun,
Mike Diehl.

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Re: [asterisk-users] Transfer to fax

[Mike Diehl]

On Tuesday 13 March 2012 3:21:58 pm Danny Nicholas wrote:
 #1 you might need a progress() statement after answer

I'll try that.  Thank you.

 #2 what does sip show peer xxx look like on this peer?

I'm testing against my office phone, a Polycom 501:

  * Name   : 0004F211F1D0-2
  Realtime peer: Yes, cached
  Secret   : Set
  MD5Secret: Not set
  Remote Secret: Not set
  Context  : customers
  Subscr.Cont. : Not set
  Language : 
  Accountcode  : 1
  AMA flags: Unknown
  Transfer mode: open
  CallingPres  : Presentation Allowed, Not Screened
  Callgroup: 
  Pickupgroup  : 
  Mailbox  : 7001@context
  VM Extension : asterisk
  LastMsgsSent : 32767/65535
  Call limit   : 0
  Dynamic  : Yes
  Callerid : Mike Diehl 5051234567
  MaxCallBR: 384 kbps
  Expire   : 172
  Insecure : no
  Nat  : Always
  ACL  : Yes
  T.38 support : Yes
  T.38 EC mode : FEC
  T.38 MaxDtgrm: -1
  DirectMedia  : Yes
  PromiscRedir : No
  User=Phone   : No
  Video Support: Yes
  Text Support : No
  Ign SDP ver  : Yes
  Trust RPID   : No
  Send RPID: No
  Subscriptions: Yes
  Overlap dial : Yes
  DTMFmode : rfc2833
  Timer T1 : 500
  Timer B  : 32000
  ToHost   : 
  Addr-IP : 173.10.242.192 Port 1811
  Defaddr-IP  : 0.0.0.0 Port 5060
  Prim.Transp. : UDP
  Allowed.Trsp : UDP
  Def. Username: 0004F211F1D0-2
  SIP Options  : (none)
  Codecs   : 0x4 (ulaw)
  Codec Order  : (ulaw:20)
  Auto-Framing :  No 
  100 on REG   : Yes
  Status   : OK (88 ms)
  Useragent: PolycomSoundPointIP-SPIP_501-UA/3.1.4.0070
  Reg. Contact : sip:0004F211F1D0-2@10.0.1.81
  Qualify Freq : 6 ms
  Variables:
 line_id = 0004F211F1D0-2
  Sess-Timers  : Accept
  Sess-Refresh : uas
  Sess-Expires : 1800 secs
  Min-Sess : 90 secs
  Parkinglot   : 



 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Diehl
 Sent: Tuesday, March 13, 2012 4:18 PM
 To: asterisk-users@lists.digium.com
 Subject: Re: [asterisk-users] Transfer to fax
 
 So I'm still trying to get this to work... (I'm top posting, but the
 details are below, if you want/need background info)
 
 I'd like Asterisk to detect incoming faxes and redirect them elsewhere. 
 The details aren't important, as long as I get the detection working.
 
 I've added this to my sip.conf file.  Probably overkill, but I'll tune it
 once it works:
 
 [general]
 faxdetect=both
 
 My sip registrations are all in a Mysql RT database, so I added this column
 to my table:
 
 faxdetect char(3) default 'no'
 
 I've set faxdetect to 'yes' for the devices that I expect to be receiving
 fax calls.
 
 I did a sip reload from the console after adding and updating this column.
 
 I've added a fax extension to the appropriate context in extensions.conf:
 exten = fax,1,noop(I hear a fax!)
 
 Since most of my dialplan is in an AGI script, I've added this to the code
 that handles my test number:
 
 $main::agi-answer();
 $main::agi-exec(ringing);
 $main::agi-exec(wait,5);
 
 
 So, now that all of this is in place, I call the extension from my fax
 machine... and I don't get any indication on the console that Asterisk
 heard a fax.  My extension simply rings and I answer it.
 
 What am  missing?
 
 TIA,
 Mike Diehl.
 
 On Friday 24 February 2012 4:22:07 pm Kevin P. Fleming wrote:
  On 02/24/2012 05:20 PM, Mike Diehl wrote:
   On Friday 24 February 2012 4:06:19 pm Kevin P. Fleming wrote:
   On 02/24/2012 05:00 PM, Mike Diehl wrote:
   On Friday 24 February 2012 3:17:04 pm Mike Diehl wrote:
   On Friday 24 February 2012 2:39:48 pm Kevin P. Fleming wrote:
   On 02/24/2012 03:32 PM, Mike Diehl wrote:
   Hi all,
   
   I've got a user that has one phone number an wants to be able
   to us it for both voice and fax.
   
   When a fax call comes in, he wants to do some incantation on
   the keypad and have the call go to the fax machine.
   
   As I see it, he has 3 options.
   
   1.  (blind?) Transfer it to the fax extension.
   
   2.  Use features.conf to create a key sequence, say *2, to
   dial/transfer to a fax extension.
   
   3.  Use fax detect (SIP) to do it automatically.  However I'm
   also using FFA, so I suspect these are mutually exclusive.
   
   They are not. Enabling faxdetect should do exactly what you
   want; it will redirect the call to the 'fax' extension in the
   current context, and you can then Dial() your FAX machine (or
   send the call to ReceiveFAX).
   
   Thank you.  Then, that's what I'll do.
   
   On second though, I think my suggestion that FFA and fax detect
   were mutually exclusive stemmed from the idea that a call that was
   being originated/answered/handled by FFA would have it's call
   disconnected and redirected by fax detect.
   
   If this is the case, it changes my dial plan logic, and I'm not
   sure I fully understand what changes I'll need to make.
   
   For all I know, it might

Re: [asterisk-users] Transfer to fax

[Mike Diehl]

On Tuesday 13 March 2012 3:45:18 pm Kevin P. Fleming wrote:
 On 03/13/2012 04:18 PM, Mike Diehl wrote:

  I've set faxdetect to 'yes' for the devices that I expect to be receiving
  fax calls.
 
 'faxdetect' is not a chan_sip configuration option (unlike chan_dahdi).
 It's a feature that can be enabled on a channel via the CHANNEL()
 dialplan function. In the dialplan itself, you'd use something like this:
 exten = 1234,5,Set(CHANNEL(faxdetect)=yes)

This function was implemented somewhere in the 10.x code base, I believe.  I'm 
running 1.6.x.  So, it sounds like I need to plan an upgrade in order to get 
this to work.

 To do this in a configuration file, so that it will be applied to
 channels as soon as they are created, use 'setvar':
 
 [peer1]
 setvar=CHANNEL(faxdetect)=yes
 
 I'm not sure how this would be done using Realtime configuration, but it
 should be possible. I'd encourage you to test it out using a
 non-Realtime peer first, just to make sure that it works the way you
 expect.

I've used setvar in my RT config and it works very well.

WRT the upgrade, I've gone in and made some code changes to the voicemail 
module which I'll have to port over to version 10.x.  Sounds like I should 
sign up to be a developer so I can pass those patches on...

-- 

Take care and have fun,
Mike Diehl.

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Re: [asterisk-users] Transfer to fax

[Mike Diehl]

On Tuesday 13 March 2012 4:04:31 pm Kevin P. Fleming wrote:
 On 03/13/2012 04:56 PM, Mike Diehl wrote:
  On Tuesday 13 March 2012 3:45:18 pm Kevin P. Fleming wrote:
  On 03/13/2012 04:18 PM, Mike Diehl wrote:
  I've set faxdetect to 'yes' for the devices that I expect to be
  receiving fax calls.
  
  'faxdetect' is not a chan_sip configuration option (unlike chan_dahdi).
  It's a feature that can be enabled on a channel via the CHANNEL()
  dialplan function. In the dialplan itself, you'd use something like
  this: exten =  1234,5,Set(CHANNEL(faxdetect)=yes)
  
  This function was implemented somewhere in the 10.x code base, I believe.
   I'm running 1.6.x.  So, it sounds like I need to plan an upgrade in
  order to get this to work.
 
 Right, so prior to that version, the *only* channel driver that had
 'faxdetect' functionality was chan_dahdi.

So, I have a few long nights ahead of me!

Thanks for your time.

-- 

Take care and have fun,
Mike Diehl.

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[asterisk-users] Ongoing attack from 188.138.100.16

[Mike Diehl]

I've been logging sip registrations from this IP address for 2 days now.  I've 
emailed the domain's admin, but nothing seems to come of it.

I've routed him into oblivion, but still, I think 50 requests a second for 2 
days is a bit much.

Any ideas?

-- 

Take care and have fun,
Mike Diehl.

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Re: [asterisk-users] Ongoing attack from 188.138.100.16

[Mike Diehl]

route add -host 188.138.100.16 dev lo

Good bye.  But it shouldn't come to this.

On Tuesday 06 March 2012 5:48:26 pm Matt Desbiens wrote:
 iptables -A INPUT --src 188.138.100.16 -j DROP
 
 On Mar 6, 2012 7:29 PM, Mike Diehl mdi...@diehlnet.com wrote:
  I've been logging sip registrations from this IP address for 2 days now.
  
   I've
  
  emailed the domain's admin, but nothing seems to come of it.
  
  I've routed him into oblivion, but still, I think 50 requests a second
  for 2
  days is a bit much.
  
  Any ideas?
  
  --
  
  Take care and have fun,
  Mike Diehl.
  
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Take care and have fun,
Mike Diehl.

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Re: [asterisk-users] Ongoing attack from 188.138.100.16

[Mike Diehl]

On Tuesday 06 March 2012 5:47:39 pm Patrick Lists wrote:
 On 07-03-12 01:28, Mike Diehl wrote:
  I've been logging sip registrations from this IP address for 2 days now. 
  I've emailed the domain's admin, but nothing seems to come of it.
  
  I've routed him into oblivion, but still, I think 50 requests a second
  for 2 days is a bit much.
  
  Any ideas?
 
 Did you talk to the upstream provider? A quick whois on that IP address
 suggests mentions PlusServer. Send them an email at ab...@plusserver.de

I did send them an email at that address.  No response.

 or use the chat link on the left side on this page:
 http://www.plusserver.de/produkte/ or call them at +49-2233-6124300.

I tried the chat as well with no effect.  My German is a bit rusty, or I'd call 
them


-- 

Take care and have fun,
Mike Diehl.

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[asterisk-users] Transfer to fax

[Mike Diehl]

Hi all,

I've got a user that has one phone number an wants to be able to us it for 
both voice and fax.

When a fax call comes in, he wants to do some incantation on the keypad and 
have the call go to the fax machine.

As I see it, he has 3 options.

1.  (blind?) Transfer it to the fax extension.  

2.  Use features.conf to create a key sequence, say *2, to dial/transfer to a 
fax extension.

3.  Use fax detect (SIP) to do it automatically.  However I'm also using FFA, 
so I suspect these are mutually exclusive.

Does anyone have any insight as to which would work better?

TIA,

-- 

Take care and have fun,
Mike Diehl.

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Re: [asterisk-users] Transfer to fax

[Mike Diehl]

On Friday 24 February 2012 2:39:48 pm Kevin P. Fleming wrote:
 On 02/24/2012 03:32 PM, Mike Diehl wrote:
  Hi all,
  
  I've got a user that has one phone number an wants to be able to us it
  for both voice and fax.
  
  When a fax call comes in, he wants to do some incantation on the keypad
  and have the call go to the fax machine.
  
  As I see it, he has 3 options.
  
  1.  (blind?) Transfer it to the fax extension.
  
  2.  Use features.conf to create a key sequence, say *2, to dial/transfer
  to a fax extension.
  
  3.  Use fax detect (SIP) to do it automatically.  However I'm also using
  FFA, so I suspect these are mutually exclusive.
 
 They are not. Enabling faxdetect should do exactly what you want; it
 will redirect the call to the 'fax' extension in the current context,
 and you can then Dial() your FAX machine (or send the call to ReceiveFAX).

Thank you.  Then, that's what I'll do.

-- 

Take care and have fun,
Mike Diehl.

--
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Re: [asterisk-users] Transfer to fax

[Mike Diehl]

On Friday 24 February 2012 3:17:04 pm Mike Diehl wrote:
 On Friday 24 February 2012 2:39:48 pm Kevin P. Fleming wrote:
  On 02/24/2012 03:32 PM, Mike Diehl wrote:
   Hi all,
   
   I've got a user that has one phone number an wants to be able to us it
   for both voice and fax.
   
   When a fax call comes in, he wants to do some incantation on the keypad
   and have the call go to the fax machine.
   
   As I see it, he has 3 options.
   
   1.  (blind?) Transfer it to the fax extension.
   
   2.  Use features.conf to create a key sequence, say *2, to
   dial/transfer to a fax extension.
   
   3.  Use fax detect (SIP) to do it automatically.  However I'm also
   using FFA, so I suspect these are mutually exclusive.
  
  They are not. Enabling faxdetect should do exactly what you want; it
  will redirect the call to the 'fax' extension in the current context,
  and you can then Dial() your FAX machine (or send the call to
  ReceiveFAX).
 
 Thank you.  Then, that's what I'll do.

On second though, I think my suggestion that FFA and fax detect were mutually 
exclusive stemmed from the idea that a call that was being 
originated/answered/handled by FFA would have it's call disconnected and 
redirected by fax detect.

If this is the case, it changes my dial plan logic, and I'm not sure I fully 
understand what changes I'll need to make.

For all I know, it might even simplify things by isolating all fax handling in 
one block.

Comments?

-- 

Take care and have fun,
Mike Diehl.

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Re: [asterisk-users] Transfer to fax

[Mike Diehl]

On Friday 24 February 2012 4:06:19 pm Kevin P. Fleming wrote:
 On 02/24/2012 05:00 PM, Mike Diehl wrote:
  On Friday 24 February 2012 3:17:04 pm Mike Diehl wrote:
  On Friday 24 February 2012 2:39:48 pm Kevin P. Fleming wrote:
  On 02/24/2012 03:32 PM, Mike Diehl wrote:
  Hi all,
  
  I've got a user that has one phone number an wants to be able to us it
  for both voice and fax.
  
  When a fax call comes in, he wants to do some incantation on the
  keypad and have the call go to the fax machine.
  
  As I see it, he has 3 options.
  
  1.  (blind?) Transfer it to the fax extension.
  
  2.  Use features.conf to create a key sequence, say *2, to
  dial/transfer to a fax extension.
  
  3.  Use fax detect (SIP) to do it automatically.  However I'm also
  using FFA, so I suspect these are mutually exclusive.
  
  They are not. Enabling faxdetect should do exactly what you want; it
  will redirect the call to the 'fax' extension in the current context,
  and you can then Dial() your FAX machine (or send the call to
  ReceiveFAX).
  
  Thank you.  Then, that's what I'll do.
  
  On second though, I think my suggestion that FFA and fax detect were
  mutually exclusive stemmed from the idea that a call that was being
  originated/answered/handled by FFA would have it's call disconnected and
  redirected by fax detect.
  
  If this is the case, it changes my dial plan logic, and I'm not sure I
  fully understand what changes I'll need to make.
  
  For all I know, it might even simplify things by isolating all fax
  handling in one block.
 
 Well, first you should not have faxdetect enabled on outbound channels.
 That takes care of the 'originating' part.
 
 If you have an inbound channel that you *know* you are sending to
 ReceiveFAX, then you can just disable faxdetect on that channel before
 doing so (this is why we made 'faxdetect' configurable from the
 dialplan). Alternatively, you can just let calls that you *know* are
 going to go to ReceiveFAX (dedicated FAX DIDs, for example) just 'idle'
 in the dialplan listening to silence until faxdetect kicks in and sends
 them to ReceiveFAX.
 
 Note that the usage of FFA is not relevant here; whether you are using
 Fax for Asterisk, the free version of it, or res_fax_spandsp, the
 behavior and scenarios would be the same.

Very nice.  

Sounds like I need to add a faxdetect column to my SIP real-time configuration. 
 
Once I've done a sip reload or pruned/loaded my user agents, I should be good 
to go!

Didn't know faxdetect was configurable in the dialplan...  Pointer to how to do 
it?

BTW, FFA rocks!



-- 

Take care and have fun,
Mike Diehl.

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Re: [asterisk-users] Transfer to fax

[Mike Diehl]

On Friday 24 February 2012 4:22:07 pm Kevin P. Fleming wrote:
 On 02/24/2012 05:20 PM, Mike Diehl wrote:
  On Friday 24 February 2012 4:06:19 pm Kevin P. Fleming wrote:
  On 02/24/2012 05:00 PM, Mike Diehl wrote:
  On Friday 24 February 2012 3:17:04 pm Mike Diehl wrote:
  On Friday 24 February 2012 2:39:48 pm Kevin P. Fleming wrote:
  On 02/24/2012 03:32 PM, Mike Diehl wrote:
  Hi all,
  
  I've got a user that has one phone number an wants to be able to us
  it for both voice and fax.
  
  When a fax call comes in, he wants to do some incantation on the
  keypad and have the call go to the fax machine.
  
  As I see it, he has 3 options.
  
  1.  (blind?) Transfer it to the fax extension.
  
  2.  Use features.conf to create a key sequence, say *2, to
  dial/transfer to a fax extension.
  
  3.  Use fax detect (SIP) to do it automatically.  However I'm also
  using FFA, so I suspect these are mutually exclusive.
  
  They are not. Enabling faxdetect should do exactly what you want; it
  will redirect the call to the 'fax' extension in the current context,
  and you can then Dial() your FAX machine (or send the call to
  ReceiveFAX).
  
  Thank you.  Then, that's what I'll do.
  
  On second though, I think my suggestion that FFA and fax detect were
  mutually exclusive stemmed from the idea that a call that was being
  originated/answered/handled by FFA would have it's call disconnected
  and redirected by fax detect.
  
  If this is the case, it changes my dial plan logic, and I'm not sure I
  fully understand what changes I'll need to make.
  
  For all I know, it might even simplify things by isolating all fax
  handling in one block.
  
  Well, first you should not have faxdetect enabled on outbound channels.
  That takes care of the 'originating' part.
  
  If you have an inbound channel that you *know* you are sending to
  ReceiveFAX, then you can just disable faxdetect on that channel before
  doing so (this is why we made 'faxdetect' configurable from the
  dialplan). Alternatively, you can just let calls that you *know* are
  going to go to ReceiveFAX (dedicated FAX DIDs, for example) just 'idle'
  in the dialplan listening to silence until faxdetect kicks in and sends
  them to ReceiveFAX.
  
  Note that the usage of FFA is not relevant here; whether you are using
  Fax for Asterisk, the free version of it, or res_fax_spandsp, the
  behavior and scenarios would be the same.
  
  Very nice.
  
  Sounds like I need to add a faxdetect column to my SIP real-time
  configuration. Once I've done a sip reload or pruned/loaded my user
  agents, I should be good to go!
  
  Didn't know faxdetect was configurable in the dialplan...  Pointer to how
  to do it?
 
 The CHANNEL() dialplan function with the 'faxdetect' option. Not sure
 which releases have it; it might only be Asterisk 10.

Version 10 is on my radar, but I'm using 1.6.3 now. I'll do some research.  If 
it's not in 1.6, I'll just try to make this work in the sip config, for now.

Thanks for your time.

-- 

Take care and have fun,
Mike Diehl.

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[asterisk-users] Turning off splash ring on PAP2T

[Mike Diehl]

Hi all,

I'd like to know how I can turn off the splash ring voicemail waiting 
indication on a PAP2T from the provisioning XML file.  I can do it from the web 
interface, but I need to do it on a lot of machines

TIA,

-- 

Take care and have fun,
Mike Diehl.

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