Re: [asterisk-users] where is 1.4.12?

[Diego Iastrubni]

Russel,

Please excuse me for saying it yet once more... (look for the thread "Stable 
Stable Asterisk", from Sunday). Build bots are nice to check and spot for 
compile errors (which is good). But I think that what people are looking here 
(well, specially me) is a set of automated tests for all of us to make before 
we bring our test PBX to production, something you do before release.

I have a been working on such a list, but it's more or less concentrated on 
channel banks (like duh... look at my email...). I would be more then happy 
to give you the list of tests I have made if you desire.

On Thursday 30 August 2007 06:33, Russell Bryant wrote:
> Brian West wrote:
> > I commend these efforts but if it compiles it doesn't mean it won't
> > crash in certain conditions much less run at all.  Proper unit testing
> > is hard to do trust me I have been reading up on the subject and in this
> > type of environment its hard to do proper unit tests without bring up
> > the environment and performing all tests.  That in itself is not easy.
>
> Thanks.  I understand that simply compiling only goes so far in regards to
> testing.  What I was trying to get across is that we were working on the
> setup to have an easily configurable automated build and test environment. 
> After working on setting up the infrastructure and verifying it using build
> tests, we can start adding automated runtime tests.  Then, over time, we
> can build smarter and smarter ways to automatically and routinely test
> functionality.

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

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

Re: [asterisk-users] where is 1.4.12?

[Tzafrir Cohen]

On Wed, Aug 29, 2007 at 09:48:18PM -0400, Matt wrote:

> I've seen one too many "security upgrades" take a system
> down because they induced new bugs.   

In this case, do a bit of extra work, and patch your version. Finding
the exact SVN commit that fixed the security issue is normally quite
easy, becase the commit message gets marked with a reference to the
respective ASA (Asterisk Security Anouncement).

You'll probably have to take some more time to actually read that ASA,
understand if it impacts your system, generate a patch and test it.

You would be deviating from the released version. Sure.

But then again, you always have the alternative of upgrading to the
latest version, which has been tested to fix the issue.

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir

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

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

Re: [asterisk-users] where is 1.4.12?

[Russell Bryant]

Brian West wrote:
> I commend these efforts but if it compiles it doesn't mean it won't
> crash in certain conditions much less run at all.  Proper unit testing
> is hard to do trust me I have been reading up on the subject and in this
> type of environment its hard to do proper unit tests without bring up
> the environment and performing all tests.  That in itself is not easy.

Thanks.  I understand that simply compiling only goes so far in regards to
testing.  What I was trying to get across is that we were working on the setup
to have an easily configurable automated build and test environment.  After
working on setting up the infrastructure and verifying it using build tests, we
can start adding automated runtime tests.  Then, over time, we can build smarter
and smarter ways to automatically and routinely test functionality.

-- 
Russell Bryant
Software Engineer
Digium, Inc.

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

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

Re: [asterisk-users] AsteriskNOW Web GUI

[Philipp Kempgen]

Steve Totaro wrote:

> Awesome, when you say end user do you mean the people sitting at the 
> phones or the person doing MACs, or both?

people at the phones: yes
the person doing MACs: what is that? (your obviously not talking
about the one sitting on my desk :)


Philipp

-- 
amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Let's use IT to solve problems and not to create new ones.
  Asterisk? -> http://www.das-asterisk-buch.de
  My pick of the month: rfc 2822 3.6.5

Geschäftsführer: Stefan Wintermeyer
Handelsregister: Neuwied B 14998

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

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

Re: [asterisk-users] where is 1.4.12?

[Brian West]

On Aug 29, 2007, at 9:35 PM, Russell Bryant wrote:

Another Digium software developer, Joshua Colp, has recently been  
working on an
automated build farm with virtual machines for all of the different  
operating
systems we support.  It already has 64 and 32 bit versions of Linux  
(glibc and
uclibc) and FreeBSD, building both asterisk 1.4 and trunk  
(development for the
next major version).  It is still growing, with planned support for  
Solaris 10

x86/x86-64/sparc, and Mac OSX PPC/Intel.



Russell,
	I commend these efforts but if it compiles it doesn't mean it won't  
crash in certain conditions much less run at all.  Proper unit  
testing is hard to do trust me I have been reading up on the subject  
and in this type of environment its hard to do proper unit tests  
without bring up the environment and performing all tests.  That in  
itself is not easy.


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

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

Re: [asterisk-users] where is 1.4.12?

[Russell Bryant]

Matt wrote:
> I guess my request is just that Digium maybe spend a little more time
> in QA before rolling a release out the door.   It's just annoying when
> you do what should be a dot upgrade, and find out  a feature that had
> worked just one dot below has now stopped working, or worse yet
> asterisk segfaults.And when it's on a production system you can't
> just "keep trying and get traces".

I appreciate your feedback.  The past few months has been an interesting
learning experience for the project in terms of release management.  The needs
of the project are changing, for sure.  I can assure you that the desire for
more testing is being worked on.

Another Digium software developer, Joshua Colp, has recently been working on an
automated build farm with virtual machines for all of the different operating
systems we support.  It already has 64 and 32 bit versions of Linux (glibc and
uclibc) and FreeBSD, building both asterisk 1.4 and trunk (development for the
next major version).  It is still growing, with planned support for Solaris 10
x86/x86-64/sparc, and Mac OSX PPC/Intel.

It uses the program, buildbot.  It has a web interface and IRC bot that we are
already using to control builds and receive updates on build status.  Another
feature it supports is having it run builds with a custom set of changes.  Once
things are running smoothly, I would like to start looking into ways we can have
it run some automated runtime tests to verify basic functionality.

-- 
Russell Bryant
Software Engineer
Digium, Inc.

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

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

Re: [asterisk-users] Unknown connection error: (2006) MySQL server has gone away

[Matt Riddell]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Shaun Hofer wrote:
> Hi,
> I get the following after a call has finished:
> 
>ERROR[6862]:mysql_log: cdr_mysql: Unknown connection error: (2006) 
> MySQL server has gone away
> 
> Does this error message only appear when asterisk makes a new connection 
> to mysql, because the old connection was stale (and dropped) ?

Correct.

> If so, is there a way to get asterisk to stop reporting this as an error 
> seeing it seems to write the CDR to database just fine ?

In my cdr_addon_mysql.c around line 160 there are the series of statements:

case CR_SERVER_LOST:
ast_log(LOG_ERROR, "cdr_mysql: Server has gone away. Attempting to
reconnect.\n");
break;

You can just comment out that line by putting // at the beginning of it.

I guess it should really be a debug message at that stage and then an
error further down:

ast_log(LOG_ERROR, "cdr_mysql: Retried to connect fives times, giving
up.\n");

Because it will retry 5 times and the "gone away" message is printed on
each of those times but doesn't necessarily indicate any problem (i.e.
stale connection).

- --
Kind Regards,

Matt Riddell
Director
___

http://www.venturevoip.com (Great new VoIP end to end solution)
http://www.venturevoip.com/news.php (Daily Asterisk News - html)
http://feeds.venturevoip.com/AsteriskNews (Daily Asterisk News - rss)
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.7 (MingW32)
Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org

iD8DBQFG1inoDQNt8rg0Kp4RArqhAJ9WfGJ9NvQwtxJn11HGi/FWMQvydwCdEtQ5
5kpz7xZ5FE8UAsii3l24mGQ=
=X1h1
-END PGP SIGNATURE-

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

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

Re: [asterisk-users] Channel Bank Recommendations

[Mark Bell]

Good to know thanks!

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Russ Price
> Sent: Wednesday, August 29, 2007 8:11 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Channel Bank Recommendations
> 
> Mark Bell wrote:
> > Need to add some fxs and fxo ports behind a fonebridge2 box any
> > recommendations a channel bank
> >
> 
> I have an Adtran Total Access 750 on my system, and it has worked very
> well.  However, if you live near a radio transmitter, you will need RF
> filters for your FXO ports.  I live about three miles from a
50,000-watt
> AM transmitter, and without filters, my FXOs pick up the radio station
> nicely. :)
> 
> Commonly-available DSL filters will work.
> 
>   Russ
> 
> ___
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users

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

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

Re: [asterisk-users] where is 1.4.12?

[Matt]

I guess that's my point.  I realize asterisk is open source and FREE,
however, I wouldn't expect a commercial application to crash as often
as I've seen asterisk go down.   Don't get me wrong (and we're kind of
going way off topic here), I really like asterisk, have done some bug
tracing but I don't posses any kind of programming know-how with
C... so fixing bugs is out of my court.

Again.. asterisk is an amazing product.  However, I guess what I'm
saying is, I've seen one too many "security upgrades" take a system
down because they induced new bugs.   Or a feature upgrade that causes
things to be broken (we're talking simple dot upgrades like 1.2.6 to
1.2.7 or something like that).

I guess my request is just that Digium maybe spend a little more time
in QA before rolling a release out the door.   It's just annoying when
you do what should be a dot upgrade, and find out  a feature that had
worked just one dot below has now stopped working, or worse yet
asterisk segfaults.And when it's on a production system you can't
just "keep trying and get traces".

On 8/29/07, shadowym <[EMAIL PROTECTED]> wrote:
> I have found the response to bug reports extremely impressive!  If something
> happens and I spend a bit of time to get good information to post to
> bugs.digium.com or put it in a bug thread that matches the problem I am
> having the response often can be very quick and sometimes resolutions can
> come with days or even hours.  Not just from Digium but 3rd party
> individuals as well.  These are usually not trivial bugs either but often
> very deep hard to reproduce bugs.
>
> I KNOW for a fact if I did have these problems with just about any other
> commercial product (they all have problems, you just don't know about them
> until they happen to you) out there I would be SOL or have to put in a lot
> more effort/time to get things moving forward towards a solution.
>
> This is a VERY powerful advantage of Asterisk that should NOT be overlooked
> IMHO.
>
> -Original Message-
> From: Russell Bryant [mailto:[EMAIL PROTECTED]
> Sent: Wednesday, August 29, 2007 1:45 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] where is 1.4.12?
>
> Matt wrote:
> > Just to chime in.. we still have a few systems running 1.2.6 because
> > of Digium's inability to fix bugs. Every version of Asterisk we've
> > ever tried has some sort of major bug that causes it to crash (it
> > being Asterisk) after being up for some period of time, or something
> > doesn't work right... then you'll have version X and version Y will
> > come out as a security fix only, yet stuff is broken in Y that wasn't
> > broken in X.
>
> "Digium's inability to fix bugs".  What a troll ...
>
> I'm sure you have never reported any of the issues you have experienced,
> either.
>  We surely can't fix them if they aren't reported.
>
> --
> Russell Bryant
> Software Engineer
> Digium, Inc.
>
>
>
>
> ___
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>

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

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

Re: [asterisk-users] Unknown connection error: (2006) MySQL server has gone away

[Atis]

On 8/30/07, Shaun Hofer <[EMAIL PROTECTED]> wrote:
> Hi,
> I get the following after a call has finished:
>
>ERROR[6862]:mysql_log: cdr_mysql: Unknown connection error: (2006)
> MySQL server has gone away

I also got those, approximately 20-30 per day (out of call volume
2000/day). I don't know what cause them, but asterisk manages to
reconnect - so everything is ok - i'm just ignoring them.

Regards,
Atis


-- 
Atis Lezdins,
IT Responsible of BEST Riga,
[EMAIL PROTECTED]
ICQ: 142239285
Skype: atis.lezdins
Cell Phone: +371 28806004 [Tele2, Latvia]
Work phone: +1 800 7502835 [Toll free, USA]
?BEST? -> www.BEST.eu.org

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

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

Re: [asterisk-users] where is 1.4.12?

[Steve Totaro]

Brian Roy wrote:
>
>
> On 8/29/07, *Steve Totaro* <[EMAIL PROTECTED] 
> > wrote:
>
>
> Kind of harsh for am employee of Digium on a public Asterisk mailing
> list, don't you think?
>
>  
> Enough The Digium/Aseterisk bashing seems to be at an all time 
> high recently. You seem to be involved in a lot of it. Russell has 
> given most of his time and life to this project over the years and to 
> see someone say "inability to fix bugs" I'm sure frustrates him. Not 
> that I view his remark uncalled-for, because I agree that was very 
> "trollish".
>  
> Let's get back to real problems. Like Russell says, if you got a bug, 
> submit it or fix it. If you got hardware problems call Digium. If you 
> don't like Asterisk/Digium go on to something else. The bashing is 
> ridiculous. 
>  
> My .02
>  

I just call em like I see em.  Shoot from the hip.  Trollish or not, 
check corporate email etiquette and never send something in anger. 

Personal emails are a different matter and should be addressed to the 
individual, not the list.

Thanks,
Steve

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

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

Re: [asterisk-users] AsteriskNOW Web GUI

[Steve Totaro]

Philipp Kempgen wrote:
> Steve Totaro wrote:
>
>   
>> In my haste (and wishful thinking), I got the impression that the setup 
>> included an end user GUI.  I see now that this is just an uptime thing 
>> and not a GUI of any sort.
>> 
>
> Are you talking about our project? (It's called "Gemeinschaft" btw
> as in http://en.wikipedia.org/wiki/Gemeinschaft )
>
> It *does* have a web based end user GUI.
>
> But "end user" does not include the admin. So the MySQL replication
> or cluster needs to be set up by hand - although we provide them
> with a detailed step by step manual.
>
> Regards,
>   Philipp Kempgen
>
>   
Awesome, when you say end user do you mean the people sitting at the 
phones or the person doing MACs, or both?

Please keep us posted on if/when the project gets released into the 
wild.  I am very interested as I am sure many others are.

Thanks,
Steve

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

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

Re: [asterisk-users] where is 1.4.12?

[shadowym]

I have found the response to bug reports extremely impressive!  If something
happens and I spend a bit of time to get good information to post to
bugs.digium.com or put it in a bug thread that matches the problem I am
having the response often can be very quick and sometimes resolutions can
come with days or even hours.  Not just from Digium but 3rd party
individuals as well.  These are usually not trivial bugs either but often
very deep hard to reproduce bugs.

I KNOW for a fact if I did have these problems with just about any other
commercial product (they all have problems, you just don't know about them
until they happen to you) out there I would be SOL or have to put in a lot
more effort/time to get things moving forward towards a solution. 

This is a VERY powerful advantage of Asterisk that should NOT be overlooked
IMHO.

-Original Message-
From: Russell Bryant [mailto:[EMAIL PROTECTED] 
Sent: Wednesday, August 29, 2007 1:45 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] where is 1.4.12?

Matt wrote:
> Just to chime in.. we still have a few systems running 1.2.6 because
> of Digium's inability to fix bugs. Every version of Asterisk we've
> ever tried has some sort of major bug that causes it to crash (it
> being Asterisk) after being up for some period of time, or something
> doesn't work right... then you'll have version X and version Y will
> come out as a security fix only, yet stuff is broken in Y that wasn't
> broken in X.

"Digium's inability to fix bugs".  What a troll ...

I'm sure you have never reported any of the issues you have experienced,
either.
 We surely can't fix them if they aren't reported.

-- 
Russell Bryant
Software Engineer
Digium, Inc.




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

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

Re: [asterisk-users] Channel Bank Recommendations

[Russ Price]

Mark Bell wrote:
> Need to add some fxs and fxo ports behind a fonebridge2 box any 
> recommendations a channel bank
>

I have an Adtran Total Access 750 on my system, and it has worked very 
well.  However, if you live near a radio transmitter, you will need RF 
filters for your FXO ports.  I live about three miles from a 50,000-watt 
AM transmitter, and without filters, my FXOs pick up the radio station 
nicely. :)

Commonly-available DSL filters will work.

Russ

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

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

[asterisk-users] Unknown connection error: (2006) MySQL server has gone away

[Shaun Hofer]

Hi,
I get the following after a call has finished:

   ERROR[6862]:mysql_log: cdr_mysql: Unknown connection error: (2006) 
MySQL server has gone away

Does this error message only appear when asterisk makes a new connection 
to mysql, because the old connection was stale (and dropped) ?
If so, is there a way to get asterisk to stop reporting this as an error 
seeing it seems to write the CDR to database just fine ?

Thanks
Shaun


(Second time sending this email, first one seemed have gone missing)

-- 
Email: [EMAIL PROTECTED]

***
If you receive this email by mistake, please notify us and do not make 
any use of the email.  We do not waive any privilege, confidentiality 
or copyright associated with it.


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

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

[asterisk-users] app_conference and asterisk 1.2.24

[Anton Krall]

Is app_conference designed only for 1.4? I tried compiling against 1.2.24
and but get a no such file while looking for autoconf.h which is a file only
used in 1.4... anybody running app_conference on 1.2?



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

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

Re: [asterisk-users] Members in 'Unknown' status in output of 'queue show'

[BJ Weschke]

 On 8/29/07, James FitzGibbon <[EMAIL PROTECTED]> wrote:
> Does anyone know what can cause queue members to go into a status of
> "Unknown"?
>
> pbxtel-01*CLI> queue show
>
> cshas 2 calls (max unlimited) in 'rrmemory' strategy (24s holdtime),
> W:0, C:447, A:20, SL: 91.7% within 60s
>Members:
>   SIP/1405 (dynamic) (Unknown) has taken no calls yet
>   SIP/1420 (dynamic) (paused) (Not in use) has taken no calls yet
>   SIP/1442 (dynamic) (paused) (Unknown) has taken 2 calls (last was 101
> secs ago)
>   SIP/1440 (dynamic) (In use) has taken 2 calls (last was 3071 secs ago)
>   SIP/1428 (dynamic) (paused) (Not in use) has taken 2 calls (last was
> 10818 secs ago)
>   SIP/1404 (dynamic) (paused) (Not in use) has taken 2 calls (last was
> 2228 secs ago)
>   SIP/1429 (dynamic) (paused) (Unknown) has taken 2 calls (last was 953
> secs ago)
>   SIP/1432 (dynamic) (Unavailable) has taken 5 calls (last was 1229 secs
> ago)
>   SIP/1430 (dynamic) (In use) has taken 2 calls (last was 22744 secs
> ago)
>   SIP/1435 (dynamic) (In use) has taken 3 calls (last was 13511 secs
> ago)
>   SIP/1434 (dynamic) (Unknown) has taken 6 calls (last was 9504 secs
> ago)
>   SIP/1424 (dynamic) (In use) has taken 4 calls (last was 16373 secs
> ago)
>   SIP/1408 (dynamic) (paused) (Not in use) has taken 2 calls (last was
> 8685 secs ago)
>   SIP/1203 (dynamic) (In use) has taken 3 calls (last was 16425 secs
> ago)
>   SIP/1410 (dynamic) (Unknown) has taken 2 calls (last was 8629 secs
> ago)
>Callers:
>   1. Zap/50-1 (wait: 11:15, prio: 0)
>   2. Zap/36-1 (wait: 0:41, prio: 0)
>
> That's just one queue, but I had nearly all my agents just go into Unknown
> status.  This is on * 1.4.10.1.  I had this happen once in the past, but
> couldn't reproduce it in the lab.
>
> When this happens, 'ringinuse=no' stops working, because app_queue considers
> "Unknown" to be a valid state to dispatch a caller to.  So my agents start
> getting flooded with calls while already on the phone, then the call-limit
> I've configured in sip.conf kicks in and my console fills up with this:
>
> pbxtel-01*CLI>
> [Aug 29 16:44:04] ERROR[22621]: chan_sip.c:3169 update_call_counter: Call to
> peer '1405' rejected due to usage limit of 2
>
> pbxtel-01*CLI>
> [Aug 29 16:44:04] ERROR[22621]: chan_sip.c:3169 update_call_counter: Call to
> peer '1410' rejected due to usage limit of 2
>
> pbxtel-01*CLI>
> [Aug 29 16:44:04] ERROR[22762]: chan_sip.c:3169 update_call_counter: Call to
> peer '1405' rejected due to usage limit of 2
>
> pbxtel-01*CLI>
> [Aug 29 16:44:04] ERROR[22762]: chan_sip.c:3169 update_call_counter: Call to
> peer '1410' rejected due to usage limit of 2
>
> pbxtel-01*CLI>
> [Aug 29 16:44:04] ERROR[22686]: chan_sip.c:3169 update_call_counter: Call to
> peer '1405' rejected due to usage limit of 2
>
> pbxtel-01*CLI>
> [Aug 29 16:44:04] ERROR[22686]: chan_sip.c:3169 update_call_counter: Call to
> peer '1410' rejected due to usage limit of 2
>
> I had to restart Asterisk to clear the states - sip reloads, app_queue
> reloads didn't do anything.
>
> Any thoughts as to where to start debugging this?  I killed * instead of
> stopping it so that I got a core file.  There is nothing in the log to
> indicate what went wrong prior to the first instance of "...rejected due to
> usage limit".
>
> Anything else I should gather before submitting a bug?
>

 I think we will want to see what state chan_sip is sending into
app_queue for it to be called "Uknown". What is the last state these
channels are in before they go to "Unknown" in app_queue?

-- 
Bird's The Word Technologies, Inc.
http://www.btwtech.com/

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

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

[asterisk-users] Hangup detection and trombining

[Ben Dinnerville]

Hi All,

I hate to post yet another "bloody hangup detection issue" on the list, but
I have been pulling my hair out no end of late with a hangup detection issue
on 1 system (have a few others out there with TDM400's and no issue but this
one is causing a real headache)

The scenario is - system with TDM04B, a call comes in on a analogue line,
rings internally and then diverts to a mobile on a second analogue line, so
we in effect have a trombone happening where a call comes in on 1 analogue
and back out on another analogue.

Hangup detection seems to be working most of the time, but on a regular
basis does not (about once every 2 days or so). We cannot get hangup
supervision / polarity reversal or any other smart way of detecting a
hangup, so are using busydetect. What seems to be happening is that on
trombone'd calls when both parties hangup, there is a busy tone being played
on each leg of the call back down each line. Some times we seem to get lucky
and the tones are played in sync and a hangup occurs, but other times the
tones are out of sync with each other and are overlapping, causing a
non-normal tone on the line(s) or a continuous tone rather than a 'beep beep
beep' which means the card / system cannot detect a hangup via busy detect.

Can anyone out there confirm if my assumptions are correct re the dual'ing
or the tones and the effect it will have on hangup detection?

And if correct, can anyone recommend a work around to get hangup detection
working in such a scenario?

Cheers,

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

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

Re: [asterisk-users] Queue stats

[zoachien]

Google for OrderlyQ or Queuemetrics. (in random order)

Zoa

Scott Wolfe wrote:
> What do you want? Maybe I can write it into ASTassistant.
>
> Scott
> http://www.astassistant.com
>
>
> - Original Message - 
> From: "Matt Riddell" <[EMAIL PROTECTED]>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
> 
> Sent: Wednesday, August 29, 2007 2:35 PM
> Subject: Re: [asterisk-users] Queue stats
>
>
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
>
> Ed Nuñez wrote:
>   
>> Can anyone recommend a good commercial solution for queue statistics?
>> 
>
> http://queuemetrics.loway.it/
>
> - --
> Kind Regards,
>
> Matt Riddell
> Director
> ___
>
> http://www.venturevoip.com (Great new VoIP end to end solution)
> http://www.venturevoip.com/news.php (Daily Asterisk News - html)
> http://feeds.venturevoip.com/AsteriskNews (Daily Asterisk News - rss)
> -BEGIN PGP SIGNATURE-
> Version: GnuPG v1.4.7 (MingW32)
> Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org
>
> iD8DBQFG1ea0DQNt8rg0Kp4RAqlwAJ94aTwBccuYEfO8rRUWQd9bJVakMgCgt4cA
> q6DH4NZFflCgxgqe1KX5iko=
> =sCz/
> -END PGP SIGNATURE-
>
> ___
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users 
>
>
> ___
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>   


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

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

Re: [asterisk-users] Queue stats

[Scott Wolfe]

What do you want? Maybe I can write it into ASTassistant.

Scott
http://www.astassistant.com


- Original Message - 
From: "Matt Riddell" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Sent: Wednesday, August 29, 2007 2:35 PM
Subject: Re: [asterisk-users] Queue stats


-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Ed Nuñez wrote:
> Can anyone recommend a good commercial solution for queue statistics?

http://queuemetrics.loway.it/

- --
Kind Regards,

Matt Riddell
Director
___

http://www.venturevoip.com (Great new VoIP end to end solution)
http://www.venturevoip.com/news.php (Daily Asterisk News - html)
http://feeds.venturevoip.com/AsteriskNews (Daily Asterisk News - rss)
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.7 (MingW32)
Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org

iD8DBQFG1ea0DQNt8rg0Kp4RAqlwAJ94aTwBccuYEfO8rRUWQd9bJVakMgCgt4cA
q6DH4NZFflCgxgqe1KX5iko=
=sCz/
-END PGP SIGNATURE-

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

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


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

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

Re: [asterisk-users] Authenticating SIP user in LDAP database instead of SIP.conf file

[Gavin Henry]

No probs.

On 29/08/2007, Abhishek M S <[EMAIL PROTECTED]> wrote:
> Dear Mr Gavin,
> Thank you once again. Will have to talk it over with my prof before
> upgrading to Asterisk 1.4. The productive system is currently running on
> 1.2.6.
> Thanks
> Abhishek
>
>
>  On 8/28/07, Gavin Henry <[EMAIL PROTECTED]> wrote:
> > On 27/08/07, Abhishek M S <[EMAIL PROTECTED]> wrote:
> > > Dear Mr Gavin,
> > >
> > > Sorry for having miss pelt  your name twice... Thank you once again for
> your
> > > prompt reply. Is this the correct version of the driver for Asterisk
> 1.2.x :
> > >  res_config_ldap-v0.7.tar.gz  from the link
> > > http://bugs.digium.com/view.php?id=5768
> >
> > If you use an old version of res_config_ldap with Asterisk version
> > 1.2, I'm pretty sure you'll be asked to upgrade to Asterisk 1.4 if you
> > seek any help via the lists or bug tracker.
> >
> > If you can use the latest release of Asterisk, you should.
> >
> > >
> > > Thank you for your time and patience,
> > >
> > > Abhishek
> > >
> > >
> > >
> > >
> > >  On 8/27/07, Gavin Henry <[EMAIL PROTECTED]> wrote:
> > > > On 27/08/07, Abhishek M S <[EMAIL PROTECTED]> wrote:
> > > > > Dear Mr Galvin,
> > > >
> > > > Gavin! ;-)
> > > >
> > > > >
> > > > > As of today I am using the res_config_ldap of Astirectory in my test
> > > > > Asterisk system to connect to a test LDAP database of my University.
> > > Things
> > > > > seem to be working fine so far. Now I'm faced with the task of
> > > installing
> > > > > this in the productive system. Before doing so, I'd sure like to
> > > consider
> > > > > trying the RealTime database driver that you people have developed.
> Why
> > > so?
> > > > > because I trust your judgment.
> > > >
> > > > Thanks, but you should still test it yourself.
> > > >
> > > > >
> > > > > > >I see it is res_config_ldap. You'd be much better using the
> latest
> > > > > > >version in the bug tracker.
> > > > >
> > > > > This would mean removing Astirectory module, installing the new
> driver
> > > and
> > > > > loading the new schema into LDAP. In my view, the latter part
> shouldn't
> > > be a
> > > > > concern because the old attributes and object classes (Astirectory)
> > > should
> > > > > in no way interfere with the new ones. Besides the old object
> classes
> > > could
> > > > > be deleted from LDAP. Also the former part shouldn't be of much
> concern
> > > > > either.
> > > >
> > > > Nope, you are correct.
> > > >
> > > > >
> > > > > My only concern as of now is in the installation of the RealTime
> > > database
> > > > > driver because the 'readme' file does not say anything about the
> > > > > installation. It only says about the configuration after
> installation.
> > > > > From the link:
> > > > >
> > >
> http://svn.digium.com/svn/asterisk/team/group/res_config_ldap/
> > > > > Would it be sufficiant if I were to copy the makefile and
> > > res_config_ldap.c
> > > > > to the res/ directory of my running Asterisk and do make; make
> install?
> > > or
> > > > > do I have to do LIBS=-lldap export LIBS ./configure before that? My
> > > asterisk
> > > > > version is 1.2.6.
> > > >
> > > > This Digium version is for 1.4.x, not 1.2
> > > >
> > > > >
> > > > > Thanks in advance,
> > > > > Abhishek
> > > > >
> > > > >
> > > > >
> > > > >
> > > > >
> > > > >
> > > > > On 8/27/07, Gavin Henry < [EMAIL PROTECTED] > wrote:
> > > > > > I see it is res_config_ldap. You'd be much better using the latest
> > > > > > version in the bug tracker.
> > > > > >
> > > > > > On 27/08/07, Gavin Henry < [EMAIL PROTECTED]> wrote:
> > > > > > > On 26/08/07, Abhishek M S < [EMAIL PROTECTED]> wrote:
> > > > > > > > Dear Mr Galvin,
> > > > > > >
> > > > > > > Gavin ;-)
> > > > > > >
> > > > > > > >
> > > > > > > > Thank you for the links. Had gone through the bug tracker
> before
> > > > > though. I
> > > > > > > > was specifically referring to the schema for the driver
> > > 'Astirectory'
> > > > > and
> > > > > > > > not the one related to the real time LDAP driver for Open
> LDAP.
> > > > > > >
> > > > > > > It's for any LDAP Compliant Directory Server.
> > > > > > >
> > > > > > >  In the
> > > > > > > > 'Astirectory'  documentation there's a file defining the
> schema
> > > for
> > > > > LDAP
> > > > > > > > which is incomplete. By incomplete I mean the Syntax and few
> other
> > > > > fields
> > > > > > > > are not defined let alone the schema being a static file. I do
> > > > > understand
> > > > > > > > that for Open LDAP a static file schema should be defined.
> > > > > > >
> > > > > > > Not really. in the RealTime driver you can specify which LDAP
> > > > > > > attributes map to which Asterisk Config settings.
> > > > > > >
> > > > > > > > The only reason why I preferred Astirectory over the LDAP real
> > > time
> > > > > driver
> > > > > > > > was the fact that there is no mapping required for SIP users
> and
> > > > > peers.
> > > > > > >
> > > > > > > OK, maybe I need to go and read more about Astirectory.
> > > > > > >
> > > > > > > >
> >

Re: [asterisk-users] AsteriskNOW Web GUI

[Philipp Kempgen]

Steve Totaro wrote:

> In my haste (and wishful thinking), I got the impression that the setup 
> included an end user GUI.  I see now that this is just an uptime thing 
> and not a GUI of any sort.

Are you talking about our project? (It's called "Gemeinschaft" btw
as in http://en.wikipedia.org/wiki/Gemeinschaft )

It *does* have a web based end user GUI.

But "end user" does not include the admin. So the MySQL replication
or cluster needs to be set up by hand - although we provide them
with a detailed step by step manual.

Regards,
  Philipp Kempgen

-- 
amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Let's use IT to solve problems and not to create new ones.
  Asterisk? -> http://www.das-asterisk-buch.de
  My pick of the month: rfc 2822 3.6.5

Geschäftsführer: Stefan Wintermeyer
Handelsregister: Neuwied B 14998

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

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

Re: [asterisk-users] Dell SC1430 + Digium TE110P = Digital Noise in PRI

[Tony Mountifield]

In article <[EMAIL PROTECTED]>,
Russell Bryant <[EMAIL PROTECTED]> wrote:
> Tony Mountifield wrote:
> > Will there be non-TigerJet TE2xx and TE4xx cards that are regular PCI
> > and not PCI Express?
> 
> The dual span and quad span cards never used a TigerJet PCI interface.  The 
> only
> cards that ever did use it are the TDM400P, T100P, E100P, and TE110P.

Ah, ok. Cool. Thanks!

Been using the TE110P a lot recently and forgot that the quad cards call
themselves Xilinx devices.

Tony
-- 
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org

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

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

Re: [asterisk-users] where is 1.4.12?

[Brian Roy]

On 8/29/07, Steve Totaro <[EMAIL PROTECTED]> wrote:
>
>
> Kind of harsh for am employee of Digium on a public Asterisk mailing
> list, don't you think?


Enough The Digium/Aseterisk bashing seems to be at an all time high
recently. You seem to be involved in a lot of it. Russell has given most of
his time and life to this project over the years and to see someone say
"inability to fix bugs" I'm sure frustrates him. Not that I view his remark
uncalled-for, because I agree that was very "trollish".

Let's get back to real problems. Like Russell says, if you got a bug, submit
it or fix it. If you got hardware problems call Digium. If you
don't like Asterisk/Digium go on to something else. The bashing is
ridiculous.

My .02
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

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

Re: [asterisk-users] where is 1.4.12?

[Russell Bryant]

Steve Totaro wrote:
> I don't see Matt as a troll, he is mostly helpful to people on these 
> lists (if memory servers me correctly).
> 
> Kind of harsh for am employee of Digium on a public Asterisk mailing 
> list, don't you think? 

I tend to make my passes through the mailing lists very quickly and don't have
much of a filter on the messages I send out, as I try to spend most of my time
on the bug tracker.  :)  I apologize if that one was considered over the line.

-- 
Russell Bryant
Software Engineer
Digium, Inc.

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

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

Re: [asterisk-users] Queue stats

[Matt Riddell]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Ed Nuñez wrote:
> Can anyone recommend a good commercial solution for queue statistics?  

http://queuemetrics.loway.it/

- --
Kind Regards,

Matt Riddell
Director
___

http://www.venturevoip.com (Great new VoIP end to end solution)
http://www.venturevoip.com/news.php (Daily Asterisk News - html)
http://feeds.venturevoip.com/AsteriskNews (Daily Asterisk News - rss)
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.7 (MingW32)
Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org

iD8DBQFG1ea0DQNt8rg0Kp4RAqlwAJ94aTwBccuYEfO8rRUWQd9bJVakMgCgt4cA
q6DH4NZFflCgxgqe1KX5iko=
=sCz/
-END PGP SIGNATURE-

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

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

Re: [asterisk-users] AsteriskNOW Web GUI

[bkruse]

Ahh,

I see, knew it :]

-bk
Steve Totaro wrote:
> In my haste (and wishful thinking), I got the impression that the setup 
> included an end user GUI.  I see now that this is just an uptime thing 
> and not a GUI of any sort.
>
> Thanks,
> Steve
>
> The system provides all the phones with configuration (mass
>   
>> deployment), i.e. SIP settings, programmable keys, ringtones, ...
>> All users are free to log in/out on their handsets or login at a
>> different handset and have their private phonebook etc. available
>> there (largely depends on the model).
>> 
>
>
>
> bkruse wrote:
>   
>> I have seen many setups like this, and have done many myself.
>>
>> With mysql-ndb you can do distributed databases and realtime
>> for numerous reasons. The point is to eliminate single point
>> of failure. Dual NIC cards, regular backups, clustering, load
>> balancing, there are tons of things you CAN do, if your willing.
>>
>> You can setup all this up yourself in a matter of hours. I am not
>> sure if your are implying that another GUI can set it up, because
>> I doubt it. Who knows, if I get my hands on like a serverside language
>> like php, you could probably even do the mysql setup and what not
>> over binding to SSH (tcl/expect) and SSH.
>>
>> Anyways, just my 2 cents,
>>
>> -bk
>>
>>
>> Steve Totaro wrote:
>>   
>> 
>>> bkruse wrote:
>>>   
>>> 
>>>   
> I am looking at Thirdlane's solution now.  Very impressive and modest 
> cost.
>   
> 
>   
> 
>   
 The asterisk GUI is free :]


   
 
   
 
>>> I am not making any GUI purchasing or GPL decisions until I see what 
>>> comes of this blurb from another thread.  It may change everything in a 
>>> major way.
>>>
>>> Philipp Kempgen wrote:
>>> ram wrote:
>>>
>>>  
>>>   
>>> 
>>>   
 any success stories of the setup

 kindly post your config and information
 
 
   
 
>>> That would be really difficult to understand because this is all
>>> integrated in a bigger project.
>>>
>>> Basically we use MySQL to replicate from a central database to
>>> many Asterisk "nodes". (Or you could use MySQL Cluster.) All
>>> Asterisk servers read the sip friends, queues, etc. from their
>>> local MySQL database via Realtime. The users are distributed
>>> across the nodes. A bunch of custom scripts generate parts of
>>> the dialplan or are called via AGI in order to tell it where
>>> to route calls etc.
>>>
>>> We constantly check if all the Asterisk nodes respond to SIP
>>> packets, and should one of them ever fail to do so repeatedly
>>> a standby server takes it's place. This is done in less than
>>> 10 seconds without any manual interaction.
>>>
>>> The configuration is mainly done via command line tools to be
>>> easily scriptable. Additionally every user has access to a web
>>> interface where they can change their callforwarding rules, look
>>> at phonebooks, dialed numbers, missed calls, access voicemail,
>>> program their phone's keys, monitor queue status and so on ...
>>>
>>> The system provides all the phones with configuration (mass
>>> deployment), i.e. SIP settings, programmable keys, ringtones, ...
>>> All users are free to log in/out on their handsets or login at a
>>> different handset and have their private phonebook etc. available
>>> there (largely depends on the model).
>>>
>>> We are probably going to release our software in about 3 months
>>> or so (can't promise that, don't nail me down to it) under the
>>> GNU GPL.
>>>
>>> Regards,
>>>   Philipp Kempgen
>>>
>>>
>>>
>>> ___
>>> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>>>
>>> asterisk-users mailing list
>>> To UNSUBSCRIBE or update options visit:
>>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>>   
>>> 
>>>   
>> ___
>> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>>
>>   
>> 
>
>
> ___
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>   


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

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

Re: [asterisk-users] where is 1.4.12?

[Steve Totaro]

Russell Bryant wrote:
> Matt wrote:
>   
>> Just to chime in.. we still have a few systems running 1.2.6 because
>> of Digium's inability to fix bugs. Every version of Asterisk we've
>> ever tried has some sort of major bug that causes it to crash (it
>> being Asterisk) after being up for some period of time, or something
>> doesn't work right... then you'll have version X and version Y will
>> come out as a security fix only, yet stuff is broken in Y that wasn't
>> broken in X.
>> 
>
> "Digium's inability to fix bugs".  What a troll ...
>
> I'm sure you have never reported any of the issues you have experienced, 
> either.
>  We surely can't fix them if they aren't reported.
>
>   
I don't see Matt as a troll, he is mostly helpful to people on these 
lists (if memory servers me correctly).

Kind of harsh for am employee of Digium on a public Asterisk mailing 
list, don't you think? 

Not that it is my business, but I try to keep all my business related 
email very cordial and avoid name calling.

Thanks,
Steve

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

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

Re: [asterisk-users] Dell SC1430 + Digium TE110P = Digital Noise in PRI

[Russell Bryant]

Tony Mountifield wrote:
> Will there be non-TigerJet TE2xx and TE4xx cards that are regular PCI
> and not PCI Express?

The dual span and quad span cards never used a TigerJet PCI interface.  The only
cards that ever did use it are the TDM400P, T100P, E100P, and TE110P.

-- 
Russell Bryant
Software Engineer
Digium, Inc.

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

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

Re: [asterisk-users] where is 1.4.12?

[Ira]

CentOS 4.5 Final
Kernel 2.6.9-55.0.2.EL  on an I686

At 01:16 PM 8/29/2007, you wrote:
>What Kernel are you using, pre 2.6.4?
>
>On 8/29/07, Ira <[EMAIL PROTECTED]> wrote:
> > At 12:45 PM 8/29/2007, you wrote:
> > >Restarting itself?  I assume you are using safe_asterisk?  It is probably
> > >crashing, in which case we'll need a backtrace posted to
> > >bugs.digium.com to get
> > >it fixed.
> >
> > I've tried 1.4 a few times, the latest being 1.4.11. All versions I
> > tried prior to 1.4.11 would cause a kernel panic within 4 calls,
> > 1.4.11 seemed OK, but 48 hours later I noticed the system was down
> > and there had been another kernel panic so I'm back to the most
> > current 1.2 version.  I'd love to help find this problem, but I have
> > no idea what I can do to help.
> >
> > Ira


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

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

Re: [asterisk-users] Dell SC1430 + Digium TE110P = Digital Noise in PRI

[Tony Mountifield]

In article <[EMAIL PROTECTED]>,
Russell Bryant <[EMAIL PROTECTED]> wrote:
> If the TE110P will not work out for you, Digium will trade it for a TE120P.  
> The
> 120 is the replacement for the 110 which uses a far superior PCI interface
> developed at Digium instead of the TigerJet, which has been the cause of
> compatability issues in the past.  Very soon, the TigerJet part will no longer
> be in use in any of the Digium cards.

Will there be non-TigerJet TE2xx and TE4xx cards that are regular PCI
and not PCI Express?

Cheers
Tony
-- 
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org

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

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

[asterisk-users] Members in 'Unknown' status in output of 'queue show'

[James FitzGibbon]

Does anyone know what can cause queue members to go into a status of
"Unknown"?

pbxtel-01*CLI> queue show

cshas 2 calls (max unlimited) in 'rrmemory' strategy (24s holdtime),
W:0, C:447, A:20, SL:91.7% within 60s
   Members:
  SIP/1405 (dynamic) (Unknown) has taken no calls yet
  SIP/1420 (dynamic) (paused) (Not in use) has taken no calls yet
  SIP/1442 (dynamic) (paused) (Unknown) has taken 2 calls (last was 101
secs ago)
  SIP/1440 (dynamic) (In use) has taken 2 calls (last was 3071 secs ago)
  SIP/1428 (dynamic) (paused) (Not in use) has taken 2 calls (last was
10818 secs ago)
  SIP/1404 (dynamic) (paused) (Not in use) has taken 2 calls (last was
2228 secs ago)
  SIP/1429 (dynamic) (paused) (Unknown) has taken 2 calls (last was 953
secs ago)
  SIP/1432 (dynamic) (Unavailable) has taken 5 calls (last was 1229 secs
ago)
  SIP/1430 (dynamic) (In use) has taken 2 calls (last was 22744 secs
ago)
  SIP/1435 (dynamic) (In use) has taken 3 calls (last was 13511 secs
ago)
  SIP/1434 (dynamic) (Unknown) has taken 6 calls (last was 9504 secs
ago)
  SIP/1424 (dynamic) (In use) has taken 4 calls (last was 16373 secs
ago)
  SIP/1408 (dynamic) (paused) (Not in use) has taken 2 calls (last was
8685 secs ago)
  SIP/1203 (dynamic) (In use) has taken 3 calls (last was 16425 secs
ago)
  SIP/1410 (dynamic) (Unknown) has taken 2 calls (last was 8629 secs
ago)
   Callers:
  1. Zap/50-1 (wait: 11:15, prio: 0)
  2. Zap/36-1 (wait: 0:41, prio: 0)

That's just one queue, but I had nearly all my agents just go into Unknown
status.  This is on * 1.4.10.1.  I had this happen once in the past, but
couldn't reproduce it in the lab.

When this happens, 'ringinuse=no' stops working, because app_queue considers
"Unknown" to be a valid state to dispatch a caller to.  So my agents start
getting flooded with calls while already on the phone, then the call-limit
I've configured in sip.conf kicks in and my console fills up with this:

pbxtel-01*CLI>
[Aug 29 16:44:04] ERROR[22621]: chan_sip.c:3169 update_call_counter: Call to
peer '1405' rejected due to usage limit of 2

pbxtel-01*CLI>
[Aug 29 16:44:04] ERROR[22621]: chan_sip.c:3169 update_call_counter: Call to
peer '1410' rejected due to usage limit of 2

pbxtel-01*CLI>
[Aug 29 16:44:04] ERROR[22762]: chan_sip.c:3169 update_call_counter: Call to
peer '1405' rejected due to usage limit of 2

pbxtel-01*CLI>
[Aug 29 16:44:04] ERROR[22762]: chan_sip.c:3169 update_call_counter: Call to
peer '1410' rejected due to usage limit of 2

pbxtel-01*CLI>
[Aug 29 16:44:04] ERROR[22686]: chan_sip.c:3169 update_call_counter: Call to
peer '1405' rejected due to usage limit of 2

pbxtel-01*CLI>
[Aug 29 16:44:04] ERROR[22686]: chan_sip.c:3169 update_call_counter: Call to
peer '1410' rejected due to usage limit of 2

I had to restart Asterisk to clear the states - sip reloads, app_queue
reloads didn't do anything.

Any thoughts as to where to start debugging this?  I killed * instead of
stopping it so that I got a core file.  There is nothing in the log to
indicate what went wrong prior to the first instance of "...rejected due to
usage limit".

Anything else I should gather before submitting a bug?

Thanks

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

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

Re: [asterisk-users] where is 1.4.12?

[Matt]

>
> "Digium's inability to fix bugs".  What a troll ...
>
> I'm sure you have never reported any of the issues you have experienced, 
> either.
>  We surely can't fix them if they aren't reported.

On the contrair, we have reported them.However, my concern is more
when a security release has ADDITIONAL bugs that were not in place in
the previous version.

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

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

Re: [asterisk-users] AsteriskNOW Web GUI

[Steve Totaro]

In my haste (and wishful thinking), I got the impression that the setup 
included an end user GUI.  I see now that this is just an uptime thing 
and not a GUI of any sort.

Thanks,
Steve

The system provides all the phones with configuration (mass
> deployment), i.e. SIP settings, programmable keys, ringtones, ...
> All users are free to log in/out on their handsets or login at a
> different handset and have their private phonebook etc. available
> there (largely depends on the model).



bkruse wrote:
> I have seen many setups like this, and have done many myself.
>
> With mysql-ndb you can do distributed databases and realtime
> for numerous reasons. The point is to eliminate single point
> of failure. Dual NIC cards, regular backups, clustering, load
> balancing, there are tons of things you CAN do, if your willing.
>
> You can setup all this up yourself in a matter of hours. I am not
> sure if your are implying that another GUI can set it up, because
> I doubt it. Who knows, if I get my hands on like a serverside language
> like php, you could probably even do the mysql setup and what not
> over binding to SSH (tcl/expect) and SSH.
>
> Anyways, just my 2 cents,
>
> -bk
>
>
> Steve Totaro wrote:
>   
>> bkruse wrote:
>>   
>> 
 I am looking at Thirdlane's solution now.  Very impressive and modest cost.
   
 
   
 
>>> The asterisk GUI is free :]
>>>
>>>
>>>   
>>> 
>>>   
>> I am not making any GUI purchasing or GPL decisions until I see what 
>> comes of this blurb from another thread.  It may change everything in a 
>> major way.
>>
>> Philipp Kempgen wrote:
>> ram wrote:
>>
>>  
>>   
>> 
>>> any success stories of the setup
>>>
>>> kindly post your config and information
>>> 
>>> 
>>>   
>> That would be really difficult to understand because this is all
>> integrated in a bigger project.
>>
>> Basically we use MySQL to replicate from a central database to
>> many Asterisk "nodes". (Or you could use MySQL Cluster.) All
>> Asterisk servers read the sip friends, queues, etc. from their
>> local MySQL database via Realtime. The users are distributed
>> across the nodes. A bunch of custom scripts generate parts of
>> the dialplan or are called via AGI in order to tell it where
>> to route calls etc.
>>
>> We constantly check if all the Asterisk nodes respond to SIP
>> packets, and should one of them ever fail to do so repeatedly
>> a standby server takes it's place. This is done in less than
>> 10 seconds without any manual interaction.
>>
>> The configuration is mainly done via command line tools to be
>> easily scriptable. Additionally every user has access to a web
>> interface where they can change their callforwarding rules, look
>> at phonebooks, dialed numbers, missed calls, access voicemail,
>> program their phone's keys, monitor queue status and so on ...
>>
>> The system provides all the phones with configuration (mass
>> deployment), i.e. SIP settings, programmable keys, ringtones, ...
>> All users are free to log in/out on their handsets or login at a
>> different handset and have their private phonebook etc. available
>> there (largely depends on the model).
>>
>> We are probably going to release our software in about 3 months
>> or so (can't promise that, don't nail me down to it) under the
>> GNU GPL.
>>
>> Regards,
>>   Philipp Kempgen
>>
>>
>>
>> ___
>> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>   
>> 
>
>
> ___
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>   


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

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

[asterisk-users] Cisco 7970G App Development?

[Matthew Rubenstein]

Do you know where I can find docs for developing apps that run locally
on a Cisco 7970G IP phone (with SIP firmware installed)? Apps that use
the phone's display, keys, and other local functions, as well as call
init/control, and other network features, including looking up directory
info in, say, an LDAP server? All development using Asterisk instead of
CallManager services, of course.
-- 

(C) Matthew Rubenstein


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

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

Re: [asterisk-users] where is 1.4.12?

[Russell Bryant]

Matt wrote:
> Just to chime in.. we still have a few systems running 1.2.6 because
> of Digium's inability to fix bugs. Every version of Asterisk we've
> ever tried has some sort of major bug that causes it to crash (it
> being Asterisk) after being up for some period of time, or something
> doesn't work right... then you'll have version X and version Y will
> come out as a security fix only, yet stuff is broken in Y that wasn't
> broken in X.

"Digium's inability to fix bugs".  What a troll ...

I'm sure you have never reported any of the issues you have experienced, either.
 We surely can't fix them if they aren't reported.

-- 
Russell Bryant
Software Engineer
Digium, Inc.

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

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

Re: [asterisk-users] ATrpms/Fritz FCPCI CAPI/Fedora 7

[Razza]

On 29/08/2007, Patrick <[EMAIL PROTECTED]> wrote:
>
> On Tue, 2007-08-28 at 20:51 +0100, Razza wrote:
> > HI all,
> > Has anyone succesfully installed an AVM Fritz! card on Fedora 7 using
> > the drivers at ATrpms recently? http://atrpms.net/dist/f7/fcpci/
> > I tried with a clean F7 build on my EPIA 5000 yesterday, after
> > modifying /etc/capi.conf (removing the coment # in front of fcpci
> > line) I received the following error when executing 'capiinit' -
> >
> > FATAL: Error inserting fcpci
> > (/lib/modules/2.6.22.4-65.fc7/updates/drivers/isdn/fritz/fcpci.ko):
> > Unknown symbol in module, or unknown parameter (see dmesg)
> > ERROR: failed to load driver fcpci
> >
> > After some searcing I found this article -
> >
> https://bugs.launchpad.net/ubuntu/+source/linux-restricted-modules-2.6.22/+bug/121978
> >
> > I am a little stumped however what to do next and indeed if this is
> > the cause of the problem, can anyone offer some guidance ?
> > Thanks in advance.
>
> The solution is at the end of this page:
> http://student.physik.uni-mainz.de/~reiffert/fcpci.php
> Basically you need to replace pci_module_init with pci_register_driver
>
> fritz/src/main.c
>
> #if defined (__fcpci__)
>/*  if (0 == (err = pci_module_init (&fcpci_driver))) { */
>if (0 == (err = pci_register_driver (&fcpci_driver))) {
>LOG("PCI driver registered.\n");
>
> Regards,
> Patrick
>

Within an RPM?
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

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

Re: [asterisk-users] AsteriskNOW Web GUI

[bkruse]

I have seen many setups like this, and have done many myself.

With mysql-ndb you can do distributed databases and realtime
for numerous reasons. The point is to eliminate single point
of failure. Dual NIC cards, regular backups, clustering, load
balancing, there are tons of things you CAN do, if your willing.

You can setup all this up yourself in a matter of hours. I am not
sure if your are implying that another GUI can set it up, because
I doubt it. Who knows, if I get my hands on like a serverside language
like php, you could probably even do the mysql setup and what not
over binding to SSH (tcl/expect) and SSH.

Anyways, just my 2 cents,

-bk


Steve Totaro wrote:
> bkruse wrote:
>   
>>> I am looking at Thirdlane's solution now.  Very impressive and modest cost.
>>>   
>>> 
>>>   
>> The asterisk GUI is free :]
>>
>>
>>   
>> 
>
> I am not making any GUI purchasing or GPL decisions until I see what 
> comes of this blurb from another thread.  It may change everything in a 
> major way.
>
> Philipp Kempgen wrote:
> ram wrote:
>
>  
>   
>> any success stories of the setup
>>
>> kindly post your config and information
>> 
>> 
>
> That would be really difficult to understand because this is all
> integrated in a bigger project.
>
> Basically we use MySQL to replicate from a central database to
> many Asterisk "nodes". (Or you could use MySQL Cluster.) All
> Asterisk servers read the sip friends, queues, etc. from their
> local MySQL database via Realtime. The users are distributed
> across the nodes. A bunch of custom scripts generate parts of
> the dialplan or are called via AGI in order to tell it where
> to route calls etc.
>
> We constantly check if all the Asterisk nodes respond to SIP
> packets, and should one of them ever fail to do so repeatedly
> a standby server takes it's place. This is done in less than
> 10 seconds without any manual interaction.
>
> The configuration is mainly done via command line tools to be
> easily scriptable. Additionally every user has access to a web
> interface where they can change their callforwarding rules, look
> at phonebooks, dialed numbers, missed calls, access voicemail,
> program their phone's keys, monitor queue status and so on ...
>
> The system provides all the phones with configuration (mass
> deployment), i.e. SIP settings, programmable keys, ringtones, ...
> All users are free to log in/out on their handsets or login at a
> different handset and have their private phonebook etc. available
> there (largely depends on the model).
>
> We are probably going to release our software in about 3 months
> or so (can't promise that, don't nail me down to it) under the
> GNU GPL.
>
> Regards,
>   Philipp Kempgen
>
>
>
> ___
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>   


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

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

Re: [asterisk-users] where is 1.4.12?

[Matt]

Just to chime in.. we still have a few systems running 1.2.6 because
of Digium's inability to fix bugs. Every version of Asterisk we've
ever tried has some sort of major bug that causes it to crash (it
being Asterisk) after being up for some period of time, or something
doesn't work right... then you'll have version X and version Y will
come out as a security fix only, yet stuff is broken in Y that wasn't
broken in X.

On 8/29/07, Ira <[EMAIL PROTECTED]> wrote:
> At 12:45 PM 8/29/2007, you wrote:
> >Restarting itself?  I assume you are using safe_asterisk?  It is probably
> >crashing, in which case we'll need a backtrace posted to
> >bugs.digium.com to get
> >it fixed.
>
> I've tried 1.4 a few times, the latest being 1.4.11. All versions I
> tried prior to 1.4.11 would cause a kernel panic within 4 calls,
> 1.4.11 seemed OK, but 48 hours later I noticed the system was down
> and there had been another kernel panic so I'm back to the most
> current 1.2 version.  I'd love to help find this problem, but I have
> no idea what I can do to help.
>
> Ira
>
>
> ___
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>

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

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

Re: [asterisk-users] where is 1.4.12?

[Andrew Latham]

What Kernel are you using, pre 2.6.4?

On 8/29/07, Ira <[EMAIL PROTECTED]> wrote:
> At 12:45 PM 8/29/2007, you wrote:
> >Restarting itself?  I assume you are using safe_asterisk?  It is probably
> >crashing, in which case we'll need a backtrace posted to
> >bugs.digium.com to get
> >it fixed.
>
> I've tried 1.4 a few times, the latest being 1.4.11. All versions I
> tried prior to 1.4.11 would cause a kernel panic within 4 calls,
> 1.4.11 seemed OK, but 48 hours later I noticed the system was down
> and there had been another kernel panic so I'm back to the most
> current 1.2 version.  I'd love to help find this problem, but I have
> no idea what I can do to help.
>
> Ira
>
>
> ___
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>


-- 
/*
 Andrew Latham
 LATHAMA (lay-th-ham-eh)
 [EMAIL PROTECTED]
 [EMAIL PROTECTED]
*/

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

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

Re: [asterisk-users] where is 1.4.12?

[Ira]

At 12:45 PM 8/29/2007, you wrote:
>Restarting itself?  I assume you are using safe_asterisk?  It is probably
>crashing, in which case we'll need a backtrace posted to 
>bugs.digium.com to get
>it fixed.

I've tried 1.4 a few times, the latest being 1.4.11. All versions I 
tried prior to 1.4.11 would cause a kernel panic within 4 calls, 
1.4.11 seemed OK, but 48 hours later I noticed the system was down 
and there had been another kernel panic so I'm back to the most 
current 1.2 version.  I'd love to help find this problem, but I have 
no idea what I can do to help.

Ira 


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

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

Re: [asterisk-users] Queue Agents on Remote Asterisk server?

[Aubrey Wells]

because this customer is still small enough that people wear many  
hats, and have other responsibilities than taking support calls, thus  
they need voicemail. I think I've worked out a way to get to work  
with James' suggestion.



--
Aubrey Wells
Senior Engineer
Shelton | Johns Technology Group
404.478.2790
www.sheltonjohns.com



Why do your agents have voicemail?

Thanks,
Steve

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

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


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

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

Re: [asterisk-users] Dell SC1430 + Digium TE110P = Digital Noise in PRI

[Russell Bryant]

Marc Patino Gómez wrote:
> I have a terrible noise issue with Dell SC1430 + Digium TE110P. The 
> digium card is not sharing interrupts with any other device, as I saw in 
> Dell's BIOS and also with "lspci -vb" command.
> 
> After changing coax wire, UTP, balum, digium card ... I have found that 
> the problem is in Dell box, so now I'm running the same Asterisk config 
> in other server with the same Digium card and there is no noise in PRI.
> 
> Any advice to solve the problem with Dell box?

Did you ever contact Digium technical support to give them a chance to fix your
problem?  It is really disappointing to see people go with another vendor
without even giving us a chance to resolve your issue.

If the TE110P will not work out for you, Digium will trade it for a TE120P.  The
120 is the replacement for the 110 which uses a far superior PCI interface
developed at Digium instead of the TigerJet, which has been the cause of
compatability issues in the past.  Very soon, the TigerJet part will no longer
be in use in any of the Digium cards.

-- 
Russell Bryant
Software Engineer
Digium, Inc.

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

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

Re: [asterisk-users] WARNING[11439]

[Russell Bryant]

Eugeniy Khvastunov wrote:
> Yesterday has established Asterisk 1.2.21.1 on Gentoo.
> Prompt the reason of the following message:
> Aug 29 14:06:24 WARNING[11439]: channel.c:780 channel_find_locked:
> Avoided initial deadlock for '0x815d548', 9 retries!

This is not a definite indication of a problem, but it could be.  Build Asterisk
with DEBUG_THREADS enabled.  When you see this happen, grab the output of the
"core show locks" CLI command.  That will let me see if there is something
deadlocked.

-- 
Russell Bryant
Software Engineer
Digium, Inc.

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

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

Re: [asterisk-users] where is 1.4.12?

[Russell Bryant]

Bruce Ferrell wrote:
> Several days ago an announcement came out for a SIP bug in versions
> below 1.4.12.  So far I don't see 1.4.12 available for download and I'm
> seeing something that may be the bug...  My asterisk is restarting
> itself about every 30 minutes.

Huh?  If the announcement said that, it was a typo.  The fix for the last
security issue is in 1.4.11, which was released with the security announcement.

Restarting itself?  I assume you are using safe_asterisk?  It is probably
crashing, in which case we'll need a backtrace posted to bugs.digium.com to get
it fixed.

Also, give the latest code in the 1.4 branch a try first.

$ svn co http://svn.digium.com/svn/asterisk/branches/1.4 asterisk-1.4

-- 
Russell Bryant
Software Engineer
Digium, Inc.

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

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

Re: [asterisk-users] Authenticating SIP user in LDAP database instead of SIP.conf file

[Abhishek M S]

Dear Mr Gavin,
Thank you once again. Will have to talk it over with my prof before
upgrading to Asterisk 1.4. The productive system is currently running on
1.2.6.
Thanks
Abhishek

On 8/28/07, Gavin Henry <[EMAIL PROTECTED]> wrote:
>
> On 27/08/07, Abhishek M S <[EMAIL PROTECTED]> wrote:
> > Dear Mr Gavin,
> >
> > Sorry for having miss pelt  your name twice... Thank you once again for
> your
> > prompt reply. Is this the correct version of the driver for Asterisk
> 1.2.x :
> >  res_config_ldap-v0.7.tar.gz  from the link
> > http://bugs.digium.com/view.php?id=5768
>
> If you use an old version of res_config_ldap with Asterisk version
> 1.2, I'm pretty sure you'll be asked to upgrade to Asterisk 1.4 if you
> seek any help via the lists or bug tracker.
>
> If you can use the latest release of Asterisk, you should.
>
> >
> > Thank you for your time and patience,
> >
> > Abhishek
> >
> >
> >
> >
> >  On 8/27/07, Gavin Henry <[EMAIL PROTECTED]> wrote:
> > > On 27/08/07, Abhishek M S <[EMAIL PROTECTED]> wrote:
> > > > Dear Mr Galvin,
> > >
> > > Gavin! ;-)
> > >
> > > >
> > > > As of today I am using the res_config_ldap of Astirectory in my test
> > > > Asterisk system to connect to a test LDAP database of my University.
> > Things
> > > > seem to be working fine so far. Now I'm faced with the task of
> > installing
> > > > this in the productive system. Before doing so, I'd sure like to
> > consider
> > > > trying the RealTime database driver that you people have developed.
> Why
> > so?
> > > > because I trust your judgment.
> > >
> > > Thanks, but you should still test it yourself.
> > >
> > > >
> > > > > >I see it is res_config_ldap. You'd be much better using the
> latest
> > > > > >version in the bug tracker.
> > > >
> > > > This would mean removing Astirectory module, installing the new
> driver
> > and
> > > > loading the new schema into LDAP. In my view, the latter part
> shouldn't
> > be a
> > > > concern because the old attributes and object classes (Astirectory)
> > should
> > > > in no way interfere with the new ones. Besides the old object
> classes
> > could
> > > > be deleted from LDAP. Also the former part shouldn't be of much
> concern
> > > > either.
> > >
> > > Nope, you are correct.
> > >
> > > >
> > > > My only concern as of now is in the installation of the RealTime
> > database
> > > > driver because the 'readme' file does not say anything about the
> > > > installation. It only says about the configuration after
> installation.
> > > > From the link:
> > > >
> > http://svn.digium.com/svn/asterisk/team/group/res_config_ldap/
> > > > Would it be sufficiant if I were to copy the makefile and
> > res_config_ldap.c
> > > > to the res/ directory of my running Asterisk and do make; make
> install?
> > or
> > > > do I have to do LIBS=-lldap export LIBS ./configure before that? My
> > asterisk
> > > > version is 1.2.6.
> > >
> > > This Digium version is for 1.4.x, not 1.2
> > >
> > > >
> > > > Thanks in advance,
> > > > Abhishek
> > > >
> > > >
> > > >
> > > >
> > > >
> > > >
> > > > On 8/27/07, Gavin Henry < [EMAIL PROTECTED] > wrote:
> > > > > I see it is res_config_ldap. You'd be much better using the latest
> > > > > version in the bug tracker.
> > > > >
> > > > > On 27/08/07, Gavin Henry < [EMAIL PROTECTED]> wrote:
> > > > > > On 26/08/07, Abhishek M S < [EMAIL PROTECTED]> wrote:
> > > > > > > Dear Mr Galvin,
> > > > > >
> > > > > > Gavin ;-)
> > > > > >
> > > > > > >
> > > > > > > Thank you for the links. Had gone through the bug tracker
> before
> > > > though. I
> > > > > > > was specifically referring to the schema for the driver
> > 'Astirectory'
> > > > and
> > > > > > > not the one related to the real time LDAP driver for Open
> LDAP.
> > > > > >
> > > > > > It's for any LDAP Compliant Directory Server.
> > > > > >
> > > > > >  In the
> > > > > > > 'Astirectory'  documentation there's a file defining the
> schema
> > for
> > > > LDAP
> > > > > > > which is incomplete. By incomplete I mean the Syntax and few
> other
> > > > fields
> > > > > > > are not defined let alone the schema being a static file. I do
> > > > understand
> > > > > > > that for Open LDAP a static file schema should be defined.
> > > > > >
> > > > > > Not really. in the RealTime driver you can specify which LDAP
> > > > > > attributes map to which Asterisk Config settings.
> > > > > >
> > > > > > > The only reason why I preferred Astirectory over the LDAP real
> > time
> > > > driver
> > > > > > > was the fact that there is no mapping required for SIP users
> and
> > > > peers.
> > > > > >
> > > > > > OK, maybe I need to go and read more about Astirectory.
> > > > > >
> > > > > > >
> > > > > > > Regards
> > > > > > > Abhishek
> > > > > > >
> > > > > > >
> > > > > > > On 8/24/07, Gavin Henry <[EMAIL PROTECTED]> wrote:
> > > > > > > >
> > > > > > > > Please see the official tracker in the Digium buglist:
> > > > > > > >
> > > > > > > > http://bugs.digium.com/view.php?id=5768
> > > > > > > >
> > > > > > > > Here are the s

[asterisk-users] where is 1.4.12?

[Bruce Ferrell]

Several days ago an announcement came out for a SIP bug in versions
below 1.4.12.  So far I don't see 1.4.12 available for download and I'm
seeing something that may be the bug...  My asterisk is restarting
itself about every 30 minutes.

HELP!!!

:)

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

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

Re: [asterisk-users] AsteriskNOW and config files

[Jared Smith]

On Wed, 2007-08-29 at 19:48 +0200, Olivier wrote:
> Is it possible to set things such as parts of config files are edited
> though AsteriskNOW GUI while other parts remain "hand editable" ?
> AsteriskNOW website include screenshots but not much information (such
> as user manual) beside that. 

One of the nice things about the AsteriskNOW GUI is that it leaves all
your configuration files "hand editable", and won't overwrite your
changes after you edit a configuration file by hand.  As far as I know,
it's fairly unique in that regard.

-- 
Jared Smith
Community Relations Manager
Digium, Inc.


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

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

[asterisk-users] Trying to use "Set Group" correctly

[Mike]

Sorry if this is a repost, but I never saw this appear on the mailing list
(I originally sent it 5 days ago)

Hi,
 
A while ago I inquired about controlling the number of channels used by each
of my customers, and I was told to use the Set Group command  to count the
number of used channels .  I did, and at first glance it seemed to work.
 
Now, the thing is my original plan was to only increment the count everytime
someone made  an external call.  In other words, a call from inside to
outside counts for 1, a call from outside to inside counts for 1 and a call
from inside to inside counts for 0.  Allowing me to sell me service on a
"per-external-line basis".
 
Now, what I noticed is that a call coming in counts for 1.  But if the
person, who is first sent to an IVR, eventually dials an extension and rings
a SIP phone, the count is incremented again (to 2 at this point).  Which is
not that I want.  
 
How can I make sure that only the "external leg" is counted?
 
 
Mike
 
 
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

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

Re: [asterisk-users] Queue Agents on Remote Asterisk server?

[Steve Totaro]

Aubrey Wells wrote:
> Hi,
> I have a main Asterisk server, and a server at a branch location 
> connected via a IAX2 trunk. I want to have a queue at the main 
> location that has people from both locations as members. I got this 
> working, but the trouble comes when the round-robin logic selects a 
> member at the branch office to call. If that user is unavailable, 
> their voicemail answers the call, and the main server detects this as 
> an answered call and assumes the agent answered. This is obviously not 
> what I want, as I would like for the call to roll to one of the other 
> agents. Has anyone come across this before? Solutions?
>
> Thanks!
> *
> *
> *--*
> *Aubrey Wells*
>
>

Why do your agents have voicemail? 

Thanks,
Steve

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

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

Re: [asterisk-users] AsteriskNOW and config files

[Tzafrir Cohen]

On Wed, Aug 29, 2007 at 07:48:42PM +0200, Olivier wrote:
> Hello,
> 
> Is it possible to set things such as parts of config files are edited though
> AsteriskNOW GUI while other parts remain "hand editable" ?
> AsteriskNOW website include screenshots but not much information (such as
> user manual) beside that.
> 
> This thing was the one that kept us from using freepbx (let me say I don't
> mean it's not possible with freepbx : I mean we couldn't find any practical
> way during the sort period of time we could dedicate to trials).

There are actually numerous ways you can add extra configuration
directives to freePBX using _custom.conf files.

Can you give an example of one such thing you were attempting to do?

> 
> Maybe cutting and copying AsteriskNOW code would be an answer but, beside
> other considerations, it would very difficult to maintain.

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir

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

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

Re: [asterisk-users] Fax Problems with SpanDSP

[Christian Peter]

Am Mittwoch, den 29.08.2007, 20:13 +0800 schrieb Steve Underwood:
> Christian Peter wrote:
> > Hi list,
> >
> > I'm running current SpanDSP
> > http://www.soft-switch.org/downloads/spandsp/spandsp-0.0.4pre6.tgz
> > with Asterisk 1.2.22 somewhat successfully.
> >
> > Most Fax machines do work but I have problems with people having 
> > Tobit FaxWare and Shamrock CapiFax.
> >
> > http://www.tobit.com/login/mrd.asp?CategoryID=120
> > http://www.shamrock.de/
> >
> > I've got black bars over the pages. In Tobit some content is Ok, other
> > is covered by the black bars. Anyone else has simliar problems?
> >
> > I talked to Tobit and they said there should be an option somewhere in
> > SpanDSP to disable Fax header crossbars. But I found none.
> >
> > Can anybody help me with this issue. Please no "switch to Hylafax"
> > mails, because I'm very happy with SpanDSP, it integrates nicely and
> > works most time.
> >
> > Thank you,
> > Regards
> >
> > Christian Peter
> >   
> I assume if you are using spandsp-0.0.4pre6 you have adapted 
> app_rxfax.c. and app_txfax.c to work with it.

Hi Steve,

thank you for your help. I'm using

http://www.soft-switch.org/downloads/snapshots/spandsp/test-apps-asterisk-1.2/

without modifications. 

> 
> I haven't heard from anyone using tobit or shamrock software (who on 
> earth wants to call their company tobit? weird). I have no idea what fax 
> header crossbars might be. Do they have some kind of bicycle integration 
> in their product? :-\

:) That was just a simple translation I looked up in a dictionary. Call
it blocks, or banners. I either don't know what they meant :)

> 
> Is your problem when sending from Asterisk or receiving? Can you enable 
> debug and e-mail me a log and (assuming its a receive problem) the 
> resulting TIFF file.

The problems occur on receiving. I'll do that and send you the email
soon.

Regards Christian

> 
> Steve
> 
> 
> ___
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users


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

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

Re: [asterisk-users] Queue Agents on Remote Asterisk server?

[James FitzGibbon]

On 8/29/07, Aubrey Wells <[EMAIL PROTECTED]> wrote:

> I have a main Asterisk server, and a server at a branch location connected
> via a IAX2 trunk. I want to have a queue at the main location that has
> people from both locations as members. I got this working, but the trouble
> comes when the round-robin logic selects a member at the branch office to
> call. If that user is unavailable, their voicemail answers the call, and the
> main server detects this as an answered call and assumes the agent answered.
> This is obviously not what I want, as I would like for the call to roll to
> one of the other agents. Has anyone come across this before? Solutions?
>

Don't contact the remote agents using a context that includes a call to
VoiceMail().  Contact a remote context that dials the agent using Dial()
with the appropriate timeout and hangs up if the agent is unavailable.  Then
app_queue () will do the right thing.

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

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

[asterisk-users] AsteriskNOW and config files

[Olivier]

Hello,

Is it possible to set things such as parts of config files are edited though
AsteriskNOW GUI while other parts remain "hand editable" ?
AsteriskNOW website include screenshots but not much information (such as
user manual) beside that.

This thing was the one that kept us from using freepbx (let me say I don't
mean it's not possible with freepbx : I mean we couldn't find any practical
way during the sort period of time we could dedicate to trials).

Maybe cutting and copying AsteriskNOW code would be an answer but, beside
other considerations, it would very difficult to maintain.

What do you think of that ?

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

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

Re: [asterisk-users] AsteriskNOW Web GUI

[Steve Totaro]

Tired eyes/brain mistake, sorry about that.

I am looking for something for 1.2.x for now.  Maybe 1.4.x in a few 
months I will revisit AsteriskNow.

Thanks,
Steve

bkruse wrote:
> Steve,
>
> That is http.conf, not html.conf.
>
> You can type make checkconfig to check your asterisk
> configuration now.
>
> -bk
>
> Steve Totaro wrote:
>   
>> The README is here: svn co 
>> http://svn.digium.com/svn/asterisk-gui/branches/asterisknow
>>
>> /Configuration
>> =
>> You may install sample configuration files by doing "make samples".  
>> Also you
>> will need to edit your Asterisk configuration files to enable the GUI 
>> properly,
>> specifically:
>>
>> 1) In http.conf:
>>
>> [general]
>> enabled = yes
>> enablestatic = yes/
>>
>> I am looking at Thirdlane's solution now.  Very impressive and modest cost.
>>
>> Thanks,
>> Steve
>>
>> bkruse wrote:
>>   
>> 
>>> As Tzafrir stated, it will NOT work with 1.2.x.
>>>
>>> Where is this html.conf, which README? I will update it.
>>>
>>> I will write a brief page on setting up the *GUI for all who want to 
>>> know..
>>>
>>> There are SOME GUI's that work with 1.2, however, I almost guarantee 
>>> none of them are client side, such as this one.
>>>
>>> -bk
>>>
>>>
>>> Steve Totaro wrote:
>>>   
>>> 
>>>   
 Will this work on 1.2.x?  I just installed it and did make samples. 

 The README references a file called html.conf which does not exist and 
 also abruptly ends with the word "to" on a blank line. 

 Besides that, what would the URL be for AsteriskNow?  Is that 
 customizable in the elusive html.conf file?

 Any GUIs that are easily installed on existing systems and work with 1.2.x?

 Thanks,
 Steve

 bkruse wrote:
   
 
   
 
> svn co http://svn.digium.com/svn/asterisk-gui/branches/asterisknow 
> thegui; cd thegui; sh configure; make && sudo make install ; clear ; 
> echo 'completed'
>
> -bk
> Yann JOUANIN wrote:
>   
> 
>   
> 
>   
>> You can do it from svn server , I think there is a page in the wiki
>>
>>  
>>
>> Best,
>>
>>  
>>
>> yann
>>
>> 


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

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

Re: [asterisk-users] AsteriskNOW Web GUI

[Steve Totaro]

bkruse wrote:
>> I am looking at Thirdlane's solution now.  Very impressive and modest cost.
>>   
>> 
> The asterisk GUI is free :]
>
>
>   

I am not making any GUI purchasing or GPL decisions until I see what 
comes of this blurb from another thread.  It may change everything in a 
major way.

Philipp Kempgen wrote:
ram wrote:

 
> any success stories of the setup
>
> kindly post your config and information
> 

That would be really difficult to understand because this is all
integrated in a bigger project.

Basically we use MySQL to replicate from a central database to
many Asterisk "nodes". (Or you could use MySQL Cluster.) All
Asterisk servers read the sip friends, queues, etc. from their
local MySQL database via Realtime. The users are distributed
across the nodes. A bunch of custom scripts generate parts of
the dialplan or are called via AGI in order to tell it where
to route calls etc.

We constantly check if all the Asterisk nodes respond to SIP
packets, and should one of them ever fail to do so repeatedly
a standby server takes it's place. This is done in less than
10 seconds without any manual interaction.

The configuration is mainly done via command line tools to be
easily scriptable. Additionally every user has access to a web
interface where they can change their callforwarding rules, look
at phonebooks, dialed numbers, missed calls, access voicemail,
program their phone's keys, monitor queue status and so on ...

The system provides all the phones with configuration (mass
deployment), i.e. SIP settings, programmable keys, ringtones, ...
All users are free to log in/out on their handsets or login at a
different handset and have their private phonebook etc. available
there (largely depends on the model).

We are probably going to release our software in about 3 months
or so (can't promise that, don't nail me down to it) under the
GNU GPL.

Regards,
  Philipp Kempgen



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

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

Re: [asterisk-users] G729 Confusion

[SIP]

Jay R. Ashworth wrote:
> 2) Asterisk will attempt to complete a call (rather than correctly
>>> returning reorder) when it can't allocate a codec for both directions
>>> of the call.
>>>
>>>   
>> Yes, Asterisk will complete the call and you will have no audio if you 
>> have no free licenses.
>> 
>
> Ok; am I the only person that thinks that's a bug?  :-)
>
> Cheers,
> -- jra
>   

No... but it's why, after purchasing 10 licenses for g729, we took it 
out of production.  It's just not worth the hassle. If we could have 
gotten a refund, we'd have done it in a heartbeat.

N.

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

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

[asterisk-users] Queue Agents on Remote Asterisk server?

[Aubrey Wells]

Hi,
I have a main Asterisk server, and a server at a branch location  
connected via a IAX2 trunk. I want to have a queue at the main  
location that has people from both locations as members. I got this  
working, but the trouble comes when the round-robin logic selects a  
member at the branch office to call. If that user is unavailable,  
their voicemail answers the call, and the main server detects this as  
an answered call and assumes the agent answered. This is obviously  
not what I want, as I would like for the call to roll to one of the  
other agents. Has anyone come across this before? Solutions?


Thanks!

--
Aubrey Wells



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

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

Re: [asterisk-users] Monitor System using AGI Scripts

[Jon Pounder]

Quoting Nitesh Divecha <[EMAIL PROTECTED]>:

> Thanks Jared,
>
> Basically, it would be a totally different system running Asterisk with
> AGI scripts and monitoring other systems (Web Servers, FTP, SMTP). Not
> specifically monitoring ports (80, 21, 25) but whole system. If system
> timeouts then AGI scripts are triggered and notify system admin.
>
> I saw one PHP-AGI example "ping.php", might able to modify abit and work
> around... but to ping a systems 24/7 its chaos...

I built something like this but just injected a call with a play file  
into the call queue.

Most times its just more damn annoying than anything else and is no  
longer in use. when you already know there is a problem you don't want  
to have to go answer the phone just to make it stop trying to get a  
call through.

SMS is more effective since the messages just pile up and you don't  
need to answer each one, especially if you are already busy dealing  
with it.

PS: another mistake I made was a second "everything is ok" call once  
the problem goes away, much better to be able to check status on  
demand than get that.


there are plenty of things like opennms that can generate the "down"  
events already, just use one of those and script whatever action you  
want to happen.




>
> Cheers,
> Nitesh
>
>
>
>
> Jared Smith wrote:
>> On Wed, 2007-08-29 at 10:46 -0400, Nitesh Divecha wrote:
>>
>>> Anyone using AGI scripts to monitor their systems?
>>>
>>> Something like if the system goes down, AGI script will be triggered and
>>> system admin will be notified saying "System XYZ has gone down"...
>>>
>>
>> If the system goes down, how would an AGI script get triggered?  I know
>> lots of people using the Asterisk Manager Interface to monitor their
>> Asterisk systems, or res_snmp on Asterisk 1.4.  You'd probably be better
>> suited to look at those first.
>>
>>
>>
>
>
> ___
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>



Jon Pounder

_/_/_/  _/_/  _/   _/_/_/  _/_/  _/_/_/_/
 _/_/_/  _/  _/ _/_/_/  _/  _/_/
_/_/  _/_/  _/ _/_/  _/_/  _/
_/_/_/  _/_/  _/_/_/_/ _/_/_/  _/_/  _/_/_/_/


Inline Internet Systems Inc.
Thorold, Ontario, Canada

Tools to Power Your e-Business Solutions
www.inline.net
www.ihtml.com
www.ihtmlmerchant.com
www.opayc.com


This message was sent using IMP, the Internet Messaging Program.



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

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

[asterisk-users] Asterisk with IM (instant messaging)

[Alejandro Cabrera Obed]

Hi people, I have Asterisk 1.2.13/DebianEtch as my VoIP server, using SIP.

I need to use IM (instant messaging) among X-Lite clients, but when I
send a message to any other client I get the error "Error: method not
allowed". I read Asterisk does not support instant messaging,
so.What's the best way to have instant messaging with Asterisk ???

Thanks a lot.

Alejandro

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

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

Re: [asterisk-users] understanding queues

[Drew Gibson]

Elliot Finley wrote:


Hello,

 

I feel like I understand how the dial plan works pretty well with one 
exception.  It seems like queues are using the stdexen macro to ring 
the agents/extensions.  Is this normal?  Is there anyway to configure 
this differently?



Hi Elliot,

The queue will use the dial method in the current (or default) context. 
If you want a different dial method for the queue, put the queue in a 
new context along with the code for the new dial method.


For example, assuming call centre phones are on extensions 100 through 
199 ...


[office]

... regular office dialplan here ...

;  Call any extension
exten => _XXX,1,Macro(stdexten,${EXT_${EXTEN}})


[call-centre]

...IVR, etc here 

;  Only call call centre extensions
exten => _1XX,1,macro(ccexten,${EXT_${EXTEN}})



[macro-stdexten]

... handle dialing this way 

[macro-ccexten]

... handle dialing that way ...



regards,

Drew

--
Drew Gibson

Systems Administrator
OANDA Corporation
416-593-6767 x322
www.oanda.com

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

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

Re: [asterisk-users] understanding queues

[Atis]

On 8/29/07, Elliot Finley <[EMAIL PROTECTED]> wrote:
> I feel like I understand how the dial plan works pretty well with one
> exception.  It seems like queues are using the stdexen macro to ring the
> agents/extensions.  Is this normal?  Is there anyway to configure this
> differently?

I'm not completely sure - which one is actually used, but there is
context for queue, and context for agent, when it makes login into
Agent/ interface. So you just have to play with two of them (or read
docs) - but you can  change that for sure.

Regards,
Atis



-- 
Atis Lezdins,
IT Responsible of BEST Riga,
[EMAIL PROTECTED]
ICQ: 142239285
Skype: atis.lezdins
Cell Phone: +371 28806004 [Tele2, Latvia]
Work phone: +1 800 7502835 [Toll free, USA]
?BEST? -> www.BEST.eu.org

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

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

Re: [asterisk-users] Monitor System using AGI Scripts

[Nitesh Divecha]

Thanks man,

That's what I was looking right now... to use Nagios with asterisk plug-ins.

Cheers,
Nitesh
 

James FitzGibbon wrote:
> On 8/29/07, *Nitesh Divecha* <[EMAIL PROTECTED] 
> > wrote:
>
>
> Basically, it would be a totally different system running Asterisk
> with
> AGI scripts and monitoring other systems (Web Servers, FTP, SMTP). Not
> specifically monitoring ports (80, 21, 25) but whole system. If system
> timeouts then AGI scripts are triggered and notify system admin.
>
>
> You'd be better to monitor using something like Nagios or one of the 
> other open-source monitoring systems, then have the notification 
> script (which should be customizable in your monitoring system) write 
> a .call file to make Asterisk dial out and tell the sysadmin.
>
> To use the Asterisk dialplan to schedule and cycle checks of services 
> . erm. no.
>
> -- 
> j.
> 
>
> ___
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users


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

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

[asterisk-users] understanding queues

[Elliot Finley]

Hello,

 

I feel like I understand how the dial plan works pretty well with one
exception.  It seems like queues are using the stdexen macro to ring the
agents/extensions.  Is this normal?  Is there anyway to configure this
differently?

 

I realize this is a newbie question, but I have searched google/archives
and haven't been able to find the answer.

 

Thanks,

Elliot

 

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

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

Re: [asterisk-users] E911 mf camma Trunks

[Andrew Ott]

I have tried that it doesn't seem to make a difference.  The way I
understand restrictcid=yes is that when you do that it will just send the
ANI caller id info, which is all 911 cares about anyway, with restrictcid=no
it then sends name and number caller id in addition to ANI, so either way
should work, but I'm not getting the ANI transmitted to them in the
KP-0-ANI-ST string, they say the switch is sending KP-911-ST and then
KP-0-911-ST instead of the ANI in there.

Not sure where to go from here.

==
Andrew Ott   Email: [EMAIL PROTECTED] or [EMAIL PROTECTED]
Network Admin/Webmaster  Web:   www.prairieweb.com or www.actcom.net
==
Death is Gods way of Dropping Carrier.
God is real, unless declared integer.
God, Root, What's the Difference?  -  UserFriendly.org
==

 

> >-Original Message-
> >From: [EMAIL PROTECTED] 
> >[mailto:[EMAIL PROTECTED] On Behalf 
> >Of Trevor Peirce
> >Sent: Tuesday, August 28, 2007 2:07 PM
> >To: Asterisk Users Mailing List - Non-Commercial Discussion
> >Subject: Re: [asterisk-users] E911 mf camma Trunks
> >
> >Andrew Ott wrote:
> >> ZAPATA.conf
> >> ===
> >> ;911 group
> >> group = 2
> >> restrictcid=yes
> >> signalling = e911
> >> channel => 25-26
> >> ===
> >>
> >> ...
> >>
> >> I've tried it with either one of those ${EXTEN} which just 
> >does 911, 
> >> and the ${CALLERID(ani)} both have the same result no number 
> >> transmitted over the
> >> 911 trunks, keep in mind we still get to the 911 center 
> >with no problem.
> >>   
> >
> >Have you tried restrictcid=no ? I believe it should still 
> >send the ANI (and only block the Caller ID information), but 
> >I'd suggest you try turning it off, especially since I doubt 
> >the call takers will complain if they get more information 
> >than they need, rather than no information at all...
> >
> >Trevor
> >
> >--
> >Does your Canadian VoIP service need CRTC-compliant 9-1-1 
> >services?  Please visit http://www.digitalcon.ca/voip9-1-1/ 
> >to find out more!
> >
> >
> >___
> >--Bandwidth and Colocation Provided by http://www.api-digital.com--
> >
> >asterisk-users mailing list
> >To UNSUBSCRIBE or update options visit:
> >   http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >



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

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

Re: [asterisk-users] Monitor System using AGI Scripts

[James FitzGibbon]

On 8/29/07, Nitesh Divecha <[EMAIL PROTECTED]> wrote:

Basically, it would be a totally different system running Asterisk with
> AGI scripts and monitoring other systems (Web Servers, FTP, SMTP). Not
> specifically monitoring ports (80, 21, 25) but whole system. If system
> timeouts then AGI scripts are triggered and notify system admin.


You'd be better to monitor using something like Nagios or one of the other
open-source monitoring systems, then have the notification script (which
should be customizable in your monitoring system) write a .call file to make
Asterisk dial out and tell the sysadmin.

To use the Asterisk dialplan to schedule and cycle checks of services .
erm. no.

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

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

Re: [asterisk-users] Zaptel hardware for timing (was: Re: app-conference)

[aLaN SaNcHeZ]

...

2007/8/28, Philipp Kempgen <[EMAIL PROTECTED]>:
>
> ram wrote:
>
> > app_meetme can use ztummy but on highload expect to use hardware source
>
> A thing that was on my mind for quite some time now:
> Would it be beneficial to have a Zaptel compatible card
> in a system just as a timing source, even if it's not
> connected to a PRI?
>
>
> Regards,
>   Philipp Kempgen
>
> --
> amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
> Let's use IT to solve problems and not to create new ones.
>   Asterisk? -> http://www.das-asterisk-buch.de
>   My pick of the month: rfc 2822 3.6.5
>
> Geschäftsführer: Stefan Wintermeyer
> Handelsregister: Neuwied B 14998
>
> ___
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

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

Re: [asterisk-users] Voicemail Password Issue

[John Meksavan]

The commas do work also. Thanks again, Moj.



From: "Mojo with Horan & Company, LLC" <[EMAIL PROTECTED]>
Reply-To: Asterisk Users Mailing List - Non-Commercial 
Discussion
To: Asterisk Users Mailing List - Non-Commercial 
Discussion

Subject: Re: [asterisk-users] Voicemail Password Issue
Date: Tue, 28 Aug 2007 14:53:26 -0800

John, glad it worked for you.  Since you didn't feel you needed a name
or email address, you might as well try just commas as delimiters:

201 => 1234,,

Moj

John Meksavan wrote:
> Mojo,
>
>  Thanks for helping me with this issue.  You must have a NAME and EMAIL
> address after putting in the voicemail pin.
>
>  I just migrate to Asterisk 1.4.x from 1.2.13, so I am still trying to
> get use to all the new stuff in the newer version.  In Asterisk 1.2.13,
> it is not necessary to have a name and email address.  Thanks again for
> your help in resolving this issue.
>
> Best Regards,
> John
>
>
>> From: "Mojo with Horan & Company, LLC" <[EMAIL PROTECTED]>
>> Reply-To: Asterisk Users Mailing List - Non-Commercial
>> Discussion
>> To: Asterisk Users Mailing List - Non-Commercial
>> Discussion
>> Subject: Re: [asterisk-users] Voicemail Password Issue
>> Date: Tue, 28 Aug 2007 14:10:19 -0800
>>
>> While I can't say this won't work the way you have it, I CAN say it's
>> not the way mine is set up and it's not a way I've SEEN it ever set up.
>>
>> Could it just be complaining that you've got nothing on the right side
>> of the => for mailbox 200?
>>
>> Or could it be complaining that you don't have anything past the pin
>> number on the other lines?
>> Try:
>> 201 => 1234,Name
>> or
>> 201 => 1234,Name,email
>>
>> I'm thinking it's my first suggestion, though.  To test that, try 
adding

>> another without a pin number:
>>
>> 199 =>
>>
>> and see if you then get two of the "variable has bad format" error
>> messages
>>
>> Moj
>>
>>
>> John Meksavan wrote:
>> > Here is my voicemail.conf file:
>> > [default]
>> > 200 =>
>> > 201 => 1234
>> > 225 => 1234
>>
>> ___
>> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>
> _
> More photos, more messages, more storage—get 2GB with Windows Live
> Hotmail.
> 
http://imagine-windowslive.com/hotmail/?locale=en-us&ocid=TXT_TAGHM_migration_HM_mini_2G_0507

>
>
>
>
> 
>
> ___
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users

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

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


_
Now you can see trouble…before he arrives 
http://newlivehotmail.com/?ocid=TXT_TAGHM_migration_HM_viral_protection_0507



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

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

Re: [asterisk-users] Monitor System using AGI Scripts

[Nitesh Divecha]

Thanks Jared,

Basically, it would be a totally different system running Asterisk with 
AGI scripts and monitoring other systems (Web Servers, FTP, SMTP). Not 
specifically monitoring ports (80, 21, 25) but whole system. If system 
timeouts then AGI scripts are triggered and notify system admin.

I saw one PHP-AGI example "ping.php", might able to modify abit and work 
around... but to ping a systems 24/7 its chaos...

Cheers,
Nitesh



 
Jared Smith wrote:
> On Wed, 2007-08-29 at 10:46 -0400, Nitesh Divecha wrote:
>   
>> Anyone using AGI scripts to monitor their systems?
>>
>> Something like if the system goes down, AGI script will be triggered and 
>> system admin will be notified saying "System XYZ has gone down"...
>> 
>
> If the system goes down, how would an AGI script get triggered?  I know
> lots of people using the Asterisk Manager Interface to monitor their
> Asterisk systems, or res_snmp on Asterisk 1.4.  You'd probably be better
> suited to look at those first.
>
>
>   


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

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

Re: [asterisk-users] sip authorization problem

[bilal ghayyad]

Dear Ryan;

I am also facing a problem with my SIP endpoint, but I
need also to know what commands and tools you used to
do the below debug as I need to do such thing for my
cases, can u help?

Regards
Bilal


Hi,

I am trying to setup a simple home voip service w/ *
I have compiled and installed the svn source
as a first step I am trying to configure SIP for
inside my network.
I have a handful of softphones and a few hardphones
that I want to all
 be
able to call each other
I have configured users.conf with a single
softphone(kphone) and have
 tried
calling itself (ext 6000) and the demo
from the asterisk install (ext 500), both give me a
401 Unauthorized
 error
below I have included some debugging output and all
the important
 config
files

***part of extensions.conf that was added by
asterisk-gui
 (svn)***
[asterisk_guitools]
exten = executecommand,1,System(${command})
exten = executecommand,n,Hangup()
exten = record_vmenu,1,Answer
exten = record_vmenu,n,Playback(vm-intro)
exten = record_vmenu,n,Record(${var1})
exten = record_vmenu,n,Playback(vm-saved)
exten = record_vmenu,n,Playback(vm-goodbye)
exten = record_vmenu,n,Hangup
exten = play_file,1,Answer
exten = play_file,n,Playback(${var1})
exten = play_file,n,Hangup
hasbeensetup = Y

[DID_trunk_1]
include = default

[numberplan-custom-1]
plancomment = DialPlan1
include = default
include = parkedcalls

[timebasedrules]
***part of extensions.conf that was added by
asterisk-gui
 (svn)***


***part of users.conf that was added by
asterisk-gui (svn)***
[trunk_1]
allow = all
context = DID_trunk_1
dialformat = ${EXTEN:1}
hasexten = no
hasiax = yes
hassip = no
host = iax2.fwdnet.net
port = 4569
registeriax = yes
registersip = no
secret = rycort4e
trunkname = Custom - fwd
trunkstyle = customvoip
username = 788694

[6000]
callwaiting = yes
cid_number = 6000
fullname = proton
hasagent = yes
hasdirectory = no
hasiax = no
hasmanager = no
hassip = yes
hasvoicemail = yes
host = dynamic
mailbox = 6000
secret = proton
threewaycalling = yes
vmsecret = 1234
registeriax = no
registersip = yes
canreinvite = yes
nat = no
dtmfmode = inband
disallow = all
allow = all
context = numberplan-custom-1
***part of users.conf that was added by
asterisk-gui (svn)***

the rest are straight from the samples that got
installed at build time

***debugging output*
*CLI> sip show peers
Name/username  HostDyn Nat ACL
Port Status
6000/6000  192.168.0.101D 
5060
 Unmonitored
1 sip peers [Monitored: 0 online, 0 offline
Unmonitored: 1 online, 0
offline]

debugging output from calling 500
<--- SIP read from 192.168.0.101:5060 --->
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 192.168.0.101;branch=z9hG4bK78B64BD
CSeq: 2212 INVITE
To: 
Content-Type: application/sdp
From: "6000" ;tag=327F7192
Call-ID: [EMAIL PROTECTED]
Subject: sip:[EMAIL PROTECTED]
Content-Length: 230
User-Agent: kphone/4.2
Contact: "6000" 

v=0
o=username 0 0 IN IP4 192.168.0.101
s=The Funky Flow
c=IN IP4 192.168.0.101
t=0 0
m=audio 33322 RTP/AVP 0 97 8 3
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30

<->
--- (11 headers 11 lines) ---
  == Using TOS bits 0
  == Using CoS mark 5
Sending to 192.168.0.101 : 5060 (no NAT)
Using INVITE request as basis request -
[EMAIL PROTECTED]
No user '6000' in SIP users list
Found peer '6000' for '6000' from 192.168.0.101:5060

<--- Reliably Transmitting (no NAT) to
192.168.0.101:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP

192.168.0.101;branch=z9hG4bK78B64BD;received=192.168.0.101
From: "6000" ;tag=327F7192
To: ;tag=as6b3f431e
Call-ID: [EMAIL PROTECTED]
CSeq: 2212 INVITE
User-Agent: Asterisk PBX SVN-trunk-r81159
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,
SUBSCRIBE, NOTIFY
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5,
realm="asterisk",
 nonce="5f450cef"
Content-Length: 0


<>
Scheduling destruction of SIP dialog
'[EMAIL PROTECTED]' in
 32000 ms
(Method: INVITE)

<--- SIP read from 192.168.0.101:5060 --->
ACK sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 192.168.0.101;branch=z9hG4bK78B64BD
CSeq: 2212 ACK
To: ;tag=as6b3f431e
From: "6000" ;tag=327F7192
Call-ID: [EMAIL PROTECTED]
Content-Length: 0
User-Agent: kphone/4.2
Contact: "6000" 


<->
--- (9 headers 0 lines) ---
Really destroying SIP dialog
'[EMAIL PROTECTED]' Method: ACK

***debugging output*


thanks in advanced
Ryan
-- next part --
An HTML attachment was scrubbed...
URL:

http://lists.digium.com/pipermail/asterisk-users/attachments/20070828/7d27056c/attachment-0001.htm




   

Take the Internet to Go: Yahoo!Go puts the Internet in your pocket: mail, news, 
photos & more. 
http://mobile.yahoo.com/go?refer=1GNXIC

___
-

Re: [asterisk-users] Distributed System

[Steve Totaro]

Philipp Kempgen wrote:
> ram wrote:
>
>   
>> any success stories of the setup
>>
>> kindly post your config and information
>> 
>
> That would be really difficult to understand because this is all
> integrated in a bigger project.
>
> Basically we use MySQL to replicate from a central database to
> many Asterisk "nodes". (Or you could use MySQL Cluster.) All
> Asterisk servers read the sip friends, queues, etc. from their
> local MySQL database via Realtime. The users are distributed
> across the nodes. A bunch of custom scripts generate parts of
> the dialplan or are called via AGI in order to tell it where
> to route calls etc.
>
> We constantly check if all the Asterisk nodes respond to SIP
> packets, and should one of them ever fail to do so repeatedly
> a standby server takes it's place. This is done in less than
> 10 seconds without any manual interaction.
>
> The configuration is mainly done via command line tools to be
> easily scriptable. Additionally every user has access to a web
> interface where they can change their callforwarding rules, look
> at phonebooks, dialed numbers, missed calls, access voicemail,
> program their phone's keys, monitor queue status and so on ...
>
> The system provides all the phones with configuration (mass
> deployment), i.e. SIP settings, programmable keys, ringtones, ...
> All users are free to log in/out on their handsets or login at a
> different handset and have their private phonebook etc. available
> there (largely depends on the model).
>
> We are probably going to release our software in about 3 months
> or so (can't promise that, don't nail me down to it) under the
> GNU GPL.
>
> Regards,
>   Philipp Kempgen
>
>   
Sweet.  I sure hope that happens.  Sounds like everything that one could 
possibly want in an enterprise PBX.

Thanks,
Steve

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

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

Re: [asterisk-users] Channel Bank Recommendations

[Brian Roy]

On 8/29/07, Mark Bell <[EMAIL PROTECTED]> wrote:
>
>  Need to add some fxs and fxo ports behind a fonebridge2 box any
> recommendations a channel bank
>


We're using a Rhino here and haven't had one problem with it. It's connected
to an analog fax server and lights up for hours at a time. Probably been up
300+ days since we bought it.
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

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

[asterisk-users] Cisco FXS Issue...

[William Stillwell (Ki4swy)]

Im sure this has been thrown around this list 1,000 times, and Im sure its been 
around the net too.. But I have done everything, and cannot seem to get inward 
calls to be processed on my asterisk box..

First, Let me tell you what works:

1) Softphones (ZoIPer using IAX2 Protocol) Can make calls behind a Natted 
Firewall to the FXS Port, and it rings, and calls work full duplex.

2) Soyo IP Phone (Loaded with IAX2 Firmware) Can make calls on the same subnet 
as the asterisk server, and can talk full duplex to the FXS Port.

3) I can call other FXS Ports on the same Switch.

 Hardware 

Cisco 3725 w/ Two FXO and Two FXS

Config:

voice-port 2/1/0
 mwi
 description FXS Port 0
 station-id name fxs_2_1_0
 station-id number 1000
!
voice-port 2/1/1
 mwi
 description FXS Port 1
 station-id name fxs_2_1_1
 station-id number 1001
!
!
!
!
!
dial-peer voice 1000 pots
 description Binds to FXS Port 2/1/0
 destination-pattern 1000
 port 2/1/0
 authentication username 1000 password 
!
dial-peer voice 1001 pots
 description Binds to FXS Port 2/1/1
 destination-pattern 1001
 port 2/1/1
 authentication username 1001 password 
!
dial-peer voice 200 voip
 destination-pattern .T
 progress_ind progress enable 8
 session protocol sipv2
 session target ipv4::5060
 session transport udp
 codec g711ulaw

sip-ua 
 retry invite 3
 retry response 3
 retry bye 3
 retry cancel 3
 timers trying 1000
 registrar ipv4: expires 60
 sip-server ipv4::5060
 notify telephone-event max-duration 500



 Asterisk Info -

sip.conf

[1000]
  username=1000
  type=friend
  secret=
  qualify=yes
  nat=no
  insecure=very
  Host=dynamic
  dTMFMode=rfc2833
  auth=md5,plaintext
  allow=ulaw


[1001]
  username=1001
  type=friend
  secret=
  qualify=yes
  nat=no
  insecure=very
  Host=dynamic
  auth=md5,plaintext
  allow=ulaw


--- 

The cisco box does register.

Dialing anything on the fxs port results in "Fast Busy" with no warnings/errors 
showing up console. 





Sent via the WebMail system at kotbh.net


 
   

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

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

Re: [asterisk-users] Monitor System using AGI Scripts

[Jared Smith]

On Wed, 2007-08-29 at 10:46 -0400, Nitesh Divecha wrote:
> Anyone using AGI scripts to monitor their systems?
> 
> Something like if the system goes down, AGI script will be triggered and 
> system admin will be notified saying "System XYZ has gone down"...

If the system goes down, how would an AGI script get triggered?  I know
lots of people using the Asterisk Manager Interface to monitor their
Asterisk systems, or res_snmp on Asterisk 1.4.  You'd probably be better
suited to look at those first.


-- 
Jared Smith
Community Relations Manager
Digium, Inc.


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

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

Re: [asterisk-users] G729 Confusion

[Jay R. Ashworth]

On Tue, Aug 28, 2007 at 01:46:18PM -0500, Andres wrote:
> >Does that mean   
> >
> >1) Don't buy 729 licenses in odd numbers ?
> >
> When you buy 1 license, you get 1 encoder and 1 decoder.  So odd or even 
> numbers are fine.

Yeah; I saw that in Matt's followup.

> >2) Asterisk will attempt to complete a call (rather than correctly
> >returning reorder) when it can't allocate a codec for both directions
> >of the call.
> >
> Yes, Asterisk will complete the call and you will have no audio if you 
> have no free licenses.

Ok; am I the only person that thinks that's a bug?  :-)

Cheers,
-- jra
-- 
Jay R. Ashworth   Baylink  [EMAIL PROTECTED]
Designer The Things I Think   RFC 2100
Ashworth & Associates http://baylink.pitas.com '87 e24
St Petersburg FL USA  http://photo.imageinc.us +1 727 647 1274

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

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

[asterisk-users] Queue stats

[Ed Nuñez]

Can anyone recommend a good commercial solution for queue statistics?  




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

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

Re: [asterisk-users] Ringing sound doesn't work

[Simon Perreault]

On Wednesday 29 August 2007 10:46:18 Eric "ManxPower" Wieling wrote:
> You do not have a /etc/asterisk/indications.conf  This file is used to
> provide ringing sounds AFTER a channel has been answered.

Thanks a million times!

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

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

[asterisk-users] Monitor System using AGI Scripts

[Nitesh Divecha]

Hello All,

Anyone using AGI scripts to monitor their systems?

Something like if the system goes down, AGI script will be triggered and 
system admin will be notified saying "System XYZ has gone down"...

Any suggestions...

Cheers,
Nitesh


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

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

Re: [asterisk-users] Ringing sound doesn't work

[Eric "ManxPower" Wieling]

Simon Perreault wrote:
> Hi,
> 
> I have these extensions:
> 
> exten => 101,1,Dial(SIP/101,15)
> exten => 102,1,Dial(SIP/102,15)
> exten => 0,1,Dial(SIP/101&SIP/102,15,r)
> 
> They work fine and I get the ringing sound if I dial them directly. However, 
> I 
> also have this extension:
> 
> exten => s,1,Answer()
> exten => s,2,Background(viagenie)
> exten => s,3,WaitExten()
> 
> The ringing sound doesn't work for any extension if I use this one. I just 
> get 
> silence until someone answers. How come?
> 
> I use Asterisk 1.4.10. I have attached my extensions.conf file to this email.

You do not have a /etc/asterisk/indications.conf  This file is used to 
provide ringing sounds AFTER a channel has been answered.

BTW, don't use "r" option to Dial.  It doesn't work.

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

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

[asterisk-users] (Asterisk_1.4.0 + rxfax + spandsp_0.0.4) - symbol lookup error

[Pirlouwi]

Hello Usersks,

foreword:
i saw a thread on Asterisk-Users list about this problem, but it seems
that there were no solution:
http://www.mail-archive.com/asterisk-users@lists.digium.com/msg167757.html
Therefore, I relaunch this thread.
---
Here it is:
I have the same behavior.
1) I never got installed spandsp before. The version that I have
installed is 0.0.4 (strangly labelled 0.0.3 inside the readme file),
and strangely too: the symbolic link inside /usr/lib refere to
libspandsp.so.0.0.2. It could be the beginning of the problem.

2) Asterisk is last version: 1.4.0, but not the svn version.

3) I got this error when using a very simple dialplan:
exten => 300, 1, answer()
exten => 300, 2, rxfax(/tmp/test.tif||debug)

The error is:
*CLI> -- Starting simple switch on 'Zap/1-1'
-- Executing [EMAIL PROTECTED]:1] Answer("Zap/1-1", "") in new stack
-- Executing [EMAIL PROTECTED]:2] RxFAX("Zap/1-1",
"/tmp/test.tif||debug") in new stack
/usr/sbin/asterisk: symbol lookup error:
/usr/lib/asterisk/modules/app_rxfax.so: undefined symbol:
span_set_message_handler
[EMAIL PROTECTED] apps]#

And after that Asterisk died with error code 127.

What did I after this error:
Actually, I am not able any more to compile app_rxfax.so. The
Asterisk-1.4 make menuselect tells me that rxfax and txfax have
dependencies problems (it is marked as XXX).
I don't know how to correct this, and I suppose that this correction
could correct the first problem mentioned in the beginning of this
mail.


It is exactly same error as former thread, but I got no solution.
Help should be very appreciated!

Thanks a lot in advance.
--Pirlouwi.

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

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

Re: [asterisk-users] Distributed System

[Philipp Kempgen]

ram wrote:

> any success stories of the setup
> 
> kindly post your config and information

That would be really difficult to understand because this is all
integrated in a bigger project.

Basically we use MySQL to replicate from a central database to
many Asterisk "nodes". (Or you could use MySQL Cluster.) All
Asterisk servers read the sip friends, queues, etc. from their
local MySQL database via Realtime. The users are distributed
across the nodes. A bunch of custom scripts generate parts of
the dialplan or are called via AGI in order to tell it where
to route calls etc.

We constantly check if all the Asterisk nodes respond to SIP
packets, and should one of them ever fail to do so repeatedly
a standby server takes it's place. This is done in less than
10 seconds without any manual interaction.

The configuration is mainly done via command line tools to be
easily scriptable. Additionally every user has access to a web
interface where they can change their callforwarding rules, look
at phonebooks, dialed numbers, missed calls, access voicemail,
program their phone's keys, monitor queue status and so on ...

The system provides all the phones with configuration (mass
deployment), i.e. SIP settings, programmable keys, ringtones, ...
All users are free to log in/out on their handsets or login at a
different handset and have their private phonebook etc. available
there (largely depends on the model).

We are probably going to release our software in about 3 months
or so (can't promise that, don't nail me down to it) under the
GNU GPL.

Regards,
  Philipp Kempgen

-- 
amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Let's use IT to solve problems and not to create new ones.
  Asterisk? -> http://www.das-asterisk-buch.de
  My pick of the month: rfc 2822 3.6.5

Geschäftsführer: Stefan Wintermeyer
Handelsregister: Neuwied B 14998

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

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

[asterisk-users] can anybody tell me about unicall.conf for incoming and outgoing

[sanchal . singh]

 Hi,
   I am using debian 4.0 with version 2.6.18-4-686
I have downloaded the required files form site
  asterisk-1.2.24.tar.gz
  libmfcr2-0.0.3-1.4.tar.bz2
  libsupertone-0.0.2.tar.gz
  libunicall-0.0.3-1.4.tar.bz2
  spandsp-20060903.tar.gz

  I downloaded and installed the files in the follwing sequence
  spandsp
  libsupertone
  libunicall
  libmfcr2-0.0.3

 then copying the chan_unicall.c and channels_Makefile.patch to
 channels subdirectory of asterisk-1.2.24 After building and installing
asterisk
 When I run 'asterisk -vvvgc' command it doesnot find dtmf_put during
linking time. So, I
 commented the dtmf_put function in chan_unicall.c. Than every thing worked
fine...
  Can any body tell me about how to configure unicall.conf for
incoming and outgoing call
 As asterisk is running on one end and E1 card running application (i.e
dialogic) is running at other end.
 So, through properly configuring unicall.conf I want to communicate through
E1 card running a[pplication for both incoming and outgoing calls.

Thanks and regard
sanchal




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

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

Re: [asterisk-users] Channel Bank Recommendations

[Jared Smith]

On Wed, 2007-08-29 at 09:38 -0400, Mark Bell wrote:
> Need to add some fxs and fxo ports behind a fonebridge2 box any
> recommendations a channel bank

Personally, I've had great success with Carrier Access (ADIT 600) and
Adtran (TA-750/TA-850) channel banks (even the ones I've bought at
bargain prices on eBay).  Just avoid the Carrier Access AccessBank 1
models with FXO ports... the FXO ports have no disconnect supervision.
(They're the ones that are shaped like a pizza box.)

-Jared



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

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

[asterisk-users] OT - Callto:// tags options

[Olivier]

Hello,

>From a previous thread, I learned Callto:// tags can be used inside web
pages to mean "dial a new call to the mentioned phone number".
Is there any pointer explaining available options ?

I'm after something meaning "transfer ongoing call to the mentioned phone
extension" instead of "dial a new call".

Google replied me this :
http://msdn2.microsoft.com/en-us/library/ms709071.aspx

Anything else relevant ?

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

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

[asterisk-users] Channel Bank Recommendations

[Mark Bell]

Need to add some fxs and fxo ports behind a fonebridge2 box any
recommendations a channel bank

 

Thanks 

Mark

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

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

[asterisk-users] Voicemail and fax detect

[Olivier]

Hi,

Has anyone experienced bundling fax detection and voicemail ?

My goal is to listen to incoming calls beginning before forwarding them to
spandsp or voicemail, accordingly.
This feature is not business critical as casual collective fax numbers
remain available for important faxes.
This is more for private or sensitive fax someone can receive from time to
time.

What did you experienced (user satisfaction, ...) ?

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

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

Re: [asterisk-users] Best text-to-speech

[Ron Joffe]

On Wednesday 29 August 2007 08:12, equis software wrote:
> Hi!
> I need to use text to speech, what is the best application?
>
> Thanks!

Cepstral


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

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

[asterisk-users] Ringing sound doesn't work

[Simon Perreault]

Hi,

I have these extensions:

exten => 101,1,Dial(SIP/101,15)
exten => 102,1,Dial(SIP/102,15)
exten => 0,1,Dial(SIP/101&SIP/102,15,r)

They work fine and I get the ringing sound if I dial them directly. However, I 
also have this extension:

exten => s,1,Answer()
exten => s,2,Background(viagenie)
exten => s,3,WaitExten()

The ringing sound doesn't work for any extension if I use this one. I just get 
silence until someone answers. How come?

I use Asterisk 1.4.10. I have attached my extensions.conf file to this email.

Thanks,
Simon
[globals]
SIPTRUNK=418555
IAXTRUNK=514555

[default]
exten => s,1,Answer()
exten => s,2,Background(viagenie)
exten => s,3,WaitExten()

exten => i,1,Background(invalid)
exten => i,n,Goto(s,1)

exten => t,1,Background(please-try-again)
exten => t,n,Goto(s,1)

[phones]
exten => 101,1,Dial(SIP/101,15)
exten => 101,n,Goto(201,1)

; Simon
exten => 102,1,Dial(SIP/102,15)
exten => 102,n,Voicemail(102)

exten => 201,n,Dial(SIP/[EMAIL PROTECTED],15)
exten => 201,n,Voicemail(101)

[ivr]
exten => 0,1,Dial(SIP/101&SIP/102,15,r)
exten => 0,n,Goto(201,1)

exten => 8,1,Directory(default)
exten => #,1,Directory(default)
exten => 500,1,VoiceMailMain()

[voip_incoming]
exten => ${SIPTRUNK},1,Goto(s,1)

[voip_outgoing]
exten => _NXX,1,Set(CALLERID(all)="Viagenie (418-555-)")
exten => _NXX,2,Dial(SIP/[EMAIL PROTECTED])

exten => _NXXNXX,1,Set(CALLERID(all)="Viagenie (418-555-)")
exten => _NXXNXX,2,Dial(SIP/[EMAIL PROTECTED])

exten => _1NXXNXX,1,Set(CALLERID(all)="Viagenie (418-555-)")
exten => _1NXXNXX,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED])

exten => _9NXX,1,Set(CALLERID(all)="Viagenie (418-555-)")
exten => _9NXX,2,Dial(SIP/418${EXTEN:[EMAIL PROTECTED])

exten => _9NXXNXX,1,Set(CALLERID(all)="Viagenie (418-555-)")
exten => _9NXXNXX,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED])

exten => _91NXXNXX,1,Set(CALLERID(all)="Viagenie (418-555-)")
exten => _91NXXNXX,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED])

exten => _9.,1,Set(CALLERID(all)="Viagenie (418-555-)")
exten => _9.,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED])

exten => N11,1,Set(CALLERID(all)="Viagenie (418-555-)")
exten => N11,2,Dial(SIP/[EMAIL PROTECTED])

[external]
include => default
include => phones
include => ivr
include => voip_incoming

[internal]
include => external
include => voip_outgoing
exten => 10,1,Goto(s,1)
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

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

Re: [asterisk-users] Best text-to-speech

[Atis]

On 8/29/07, equis software <[EMAIL PROTECTED]> wrote:
> Hi!
> I need to use text to speech, what is the best application?

The only one i know is Festival(). It is far from best, but it works,
and it's easy to add in asterisk. And it is free.

For more info see: http://www.voip-info.org/wiki-Asterisk+Festival+installation

Regards,
Atis


-- 
Atis Lezdins,
IT Responsible of BEST Riga,
[EMAIL PROTECTED]
ICQ: 142239285
Skype: atis.lezdins
Cell Phone: +371 28806004 [Tele2, Latvia]
Work phone: +1 800 7502835 [Toll free, USA]
?BEST? -> www.BEST.eu.org

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

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

Re: [asterisk-users] Fax Problems with SpanDSP

[mitcheloc]

Steve,

Are there any plans to get a newer version of spandsp working with Asterisk?

On 8/29/07, Steve Underwood <[EMAIL PROTECTED]> wrote:
> Christian Peter wrote:
> > Hi list,
> >
> > I'm running current SpanDSP
> > http://www.soft-switch.org/downloads/spandsp/spandsp-0.0.4pre6.tgz
> > with Asterisk 1.2.22 somewhat successfully.
> >
> > Most Fax machines do work but I have problems with people having
> > Tobit FaxWare and Shamrock CapiFax.
> >
> > http://www.tobit.com/login/mrd.asp?CategoryID=120
> > http://www.shamrock.de/
> >
> > I've got black bars over the pages. In Tobit some content is Ok, other
> > is covered by the black bars. Anyone else has simliar problems?
> >
> > I talked to Tobit and they said there should be an option somewhere in
> > SpanDSP to disable Fax header crossbars. But I found none.
> >
> > Can anybody help me with this issue. Please no "switch to Hylafax"
> > mails, because I'm very happy with SpanDSP, it integrates nicely and
> > works most time.
> >
> > Thank you,
> > Regards
> >
> > Christian Peter
> >
> I assume if you are using spandsp-0.0.4pre6 you have adapted
> app_rxfax.c. and app_txfax.c to work with it.
>
> I haven't heard from anyone using tobit or shamrock software (who on
> earth wants to call their company tobit? weird). I have no idea what fax
> header crossbars might be. Do they have some kind of bicycle integration
> in their product? :-\
>
> Is your problem when sending from Asterisk or receiving? Can you enable
> debug and e-mail me a log and (assuming its a receive problem) the
> resulting TIFF file.
>
> Steve
>
>
> ___
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>


-- 

Mitchel Constantin
Snap - A desktop user interface for Asterisk
www.snapanumber.com

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

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

Re: [asterisk-users] Fax Problems with SpanDSP

[Steve Underwood]

Christian Peter wrote:
> Hi list,
>
> I'm running current SpanDSP
> http://www.soft-switch.org/downloads/spandsp/spandsp-0.0.4pre6.tgz
> with Asterisk 1.2.22 somewhat successfully.
>
> Most Fax machines do work but I have problems with people having 
> Tobit FaxWare and Shamrock CapiFax.
>
> http://www.tobit.com/login/mrd.asp?CategoryID=120
> http://www.shamrock.de/
>
> I've got black bars over the pages. In Tobit some content is Ok, other
> is covered by the black bars. Anyone else has simliar problems?
>
> I talked to Tobit and they said there should be an option somewhere in
> SpanDSP to disable Fax header crossbars. But I found none.
>
> Can anybody help me with this issue. Please no "switch to Hylafax"
> mails, because I'm very happy with SpanDSP, it integrates nicely and
> works most time.
>
> Thank you,
> Regards
>
> Christian Peter
>   
I assume if you are using spandsp-0.0.4pre6 you have adapted 
app_rxfax.c. and app_txfax.c to work with it.

I haven't heard from anyone using tobit or shamrock software (who on 
earth wants to call their company tobit? weird). I have no idea what fax 
header crossbars might be. Do they have some kind of bicycle integration 
in their product? :-\

Is your problem when sending from Asterisk or receiving? Can you enable 
debug and e-mail me a log and (assuming its a receive problem) the 
resulting TIFF file.

Steve


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

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

[asterisk-users] Best text-to-speech

[equis software]

Hi!
I need to use text to speech, what is the best application?

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

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

Re: [asterisk-users] Fax Problems with SpanDSP

[Steve Underwood]

Carlos Chavez wrote:
> On Wed, 2007-08-29 at 00:03 +0300, Tzafrir Cohen wrote:
>   
>> On Tue, Aug 28, 2007 at 10:11:03PM +0200, Christian Peter wrote:
>> 
>>> Hi list,
>>>
>>> I'm running current SpanDSP
>>> http://www.soft-switch.org/downloads/spandsp/spandsp-0.0.4pre6.tgz
>>> with Asterisk 1.2.22 somewhat successfully.
>>>   
>> Shouldn't you have used spandsp 0.0.3 with asterisk 1.2 ?
>>
>> 
>   Actually, as Steve Underwood has gently reminded the list several
> times, he recommends SpanDsp 0.0.2 for Asterisk 1.2
>   
Well, its not so much that I recommend it. Its just that I have never 
done anything to adapt the app_rxfax.c and app_txfax.c for Asterisk to 
work with newer versions of spandsp. Compared to the current spandsp, 
the softFAX in 0.0.2 actually sucks.

Steve


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

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

Re: [asterisk-users] sip authorization problem

[Jared Smith]

On Tue, 2007-08-28 at 22:41 -0400, Ryan Murray wrote:
> I have configured users.conf with a single softphone(kphone) and have
> tried calling itself (ext 6000) and the demo
> from the asterisk install (ext 500), both give me a 401 Unauthorized
> error

You need to configure kphone to authenticate itself to Asterisk, using
6000 as the SIP username, and the password that you set in users.conf
(the line that says secret=).


-- 
Jared Smith
Community Relations Manager
Digium, Inc.


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

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

Re: [asterisk-users] Fax Problems with SpanDSP

[Patrick]

On Tue, 2007-08-28 at 16:23 -0500, Carlos Chavez wrote:
> On Wed, 2007-08-29 at 00:03 +0300, Tzafrir Cohen wrote:
> > On Tue, Aug 28, 2007 at 10:11:03PM +0200, Christian Peter wrote:
> > > Hi list,
> > > 
> > > I'm running current SpanDSP
> > > http://www.soft-switch.org/downloads/spandsp/spandsp-0.0.4pre6.tgz
> > > with Asterisk 1.2.22 somewhat successfully.
> > 
> > Shouldn't you have used spandsp 0.0.3 with asterisk 1.2 ?
> > 
>   Actually, as Steve Underwood has gently reminded the list several
> times, he recommends SpanDsp 0.0.2 for Asterisk 1.2

Yes he did but using asterisk-1.2.24 and spandsp-0.0.2pre26 with the
app_txfax and app_rxfax apps available from soft-switch.org[1] results
in the compilation failure below.

gcc  -pipe  -Wall -Wstrict-prototypes -Wmissing-prototypes
-Wmissing-declarations  -Iinclude -I../include -D_REENTRANT
-D_GNU_SOURCE  -O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions
-fstack-protector --param=ssp-buffer-size=4 -m32 -march=i686
-mtune=generic -fasynchronous-unwind-tables -DZAPTEL_OPTIMIZATIONS
-fomit-frame-pointer  -fPIC   -c -o app_rxfax.o app_rxfax.c
app_rxfax.c: In function 'phase_e_handler':
app_rxfax.c:105: error: 't30_stats_t' has no member named 'x_resolution'
app_rxfax.c:105: error: 't30_stats_t' has no member named 'y_resolution'
app_rxfax.c:116: error: 't30_stats_t' has no member named 'y_resolution'
app_rxfax.c:122: error: 't30_stats_t' has no member named 'y_resolution'
app_rxfax.c: In function 'phase_d_handler':
app_rxfax.c:147: error: 't30_stats_t' has no member named 'width'
app_rxfax.c:147: error: 't30_stats_t' has no member named 'length'
app_rxfax.c:148: error: 't30_stats_t' has no member named 'x_resolution'
app_rxfax.c:148: error: 't30_stats_t' has no member named 'y_resolution'
app_rxfax.c: In function 'rxfax_exec':
app_rxfax.c:171: error: 'fax_state_t' undeclared (first use in this
function)
app_rxfax.c:171: error: (Each undeclared identifier is reported only
once
app_rxfax.c:171: error: for each function it appears in.)
app_rxfax.c:171: error: expected ';' before 'fax'
app_rxfax.c:281: error: 'fax' undeclared (first use in this function)
app_rxfax.c:281: error: too few arguments to function 'fax_init'
app_rxfax.c:294: warning: implicit declaration of function
't30_set_ecm_capability'
app_rxfax.c:295: warning: implicit declaration of function
't30_set_supported_compressions'
app_rxfax.c:295: error: 'T30_SUPPORT_T4_1D_COMPRESSION' undeclared
(first use in this function)
app_rxfax.c:295: error: 'T30_SUPPORT_T4_2D_COMPRESSION' undeclared
(first use in this function)
app_rxfax.c:295: error: 'T30_SUPPORT_T6_COMPRESSION' undeclared (first
use in this function)
app_rxfax.c:346: warning: implicit declaration of function
't30_terminate'
make[1]: *** [app_rxfax.o] Error 1

Anyone (Steve?) know how to fix this?

Regards,
Patrick

[1] http://soft-switch.org/downloads/snapshots/spandsp/test-apps-asterisk-1.2/


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

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

Re: [asterisk-users] ATrpms/Fritz FCPCI CAPI/Fedora 7

[Patrick]

On Tue, 2007-08-28 at 20:51 +0100, Razza wrote:
> HI all,
> Has anyone succesfully installed an AVM Fritz! card on Fedora 7 using
> the drivers at ATrpms recently? http://atrpms.net/dist/f7/fcpci/
> I tried with a clean F7 build on my EPIA 5000 yesterday, after
> modifying /etc/capi.conf (removing the coment # in front of fcpci
> line) I received the following error when executing 'capiinit' -
>  
> FATAL: Error inserting fcpci
> (/lib/modules/2.6.22.4-65.fc7/updates/drivers/isdn/fritz/fcpci.ko):
> Unknown symbol in module, or unknown parameter (see dmesg)
> ERROR: failed to load driver fcpci 
>  
> After some searcing I found this article - 
> https://bugs.launchpad.net/ubuntu/+source/linux-restricted-modules-2.6.22/+bug/121978
>  
>  
> I am a little stumped however what to do next and indeed if this is
> the cause of the problem, can anyone offer some guidance ?
> Thanks in advance.

The solution is at the end of this page:
http://student.physik.uni-mainz.de/~reiffert/fcpci.php
Basically you need to replace pci_module_init with pci_register_driver

fritz/src/main.c

#if defined (__fcpci__)
/*  if (0 == (err = pci_module_init (&fcpci_driver))) { */
if (0 == (err = pci_register_driver (&fcpci_driver))) {
LOG("PCI driver registered.\n");

Regards,
Patrick



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

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

Re: [asterisk-users] Multiple servers using realtime

[Patrick]

On Tue, 2007-08-28 at 19:59 -0600, Edgar Guadamuz wrote:
> I have a confusion about using SER for balancing the load across the
> Asterisk boxes. The doubt is: once a user registers in a Asterisk box,
> all the calls from or to him are going to be done by the same Asterisk
> server or can a user make a call by one Asterisk server and then make
> another call by other Asterisk server?

I think the user registers with the SER box. With loadbalancing an
outgoing call can go through different Asterisk boxes:

call #1 --> SER box #1 --> Asterisk box #1 --> destination
call #2 --> SER box #1 --> Asterisk box #2 --> destination

Regards,
Patrick

> On 8/28/07, Dovid B <[EMAIL PROTECTED]> wrote:
> > We have a similar set up. I would recommend also using SER and load
> > balancing so you can load balance your calls out between your asterisk
> > box's.
> 
> ___
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users


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

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

[asterisk-users] Email to Voice

[voip crazy]

Hello all,

Anyone knows any solution  (Comercial or Free) to  listen  my email via a
phone call?

Thanks in advance.

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

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

[asterisk-users] WARNING[11439]

[Eugeniy Khvastunov]

Hi All!
Yesterday has established Asterisk 1.2.21.1 on Gentoo.
Prompt the reason of the following message:
Aug 29 14:06:24 WARNING[11439]: channel.c:780 channel_find_locked: 
Avoided initial deadlock for '0x815d548', 9 retries!


--
wbr. Eugeniy Khvastunov,
[FMGH-UANIC]

[EMAIL PROTECTED]

begin:vcard
fn:Eugeniy Khvastunov
n:Khvastunov;Eugeniy
email;internet:[EMAIL PROTECTED]
tel;work:+380675745646
version:2.1
end:vcard

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

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

Re: [asterisk-users] Distributed System

[ram]

On 8/29/07, Philipp Kempgen <[EMAIL PROTECTED]> wrote:
>
> Seysan wrote:
>
> > Is there anywhere that we can look into for Realtime + MySQL that you
> > mentioned?
>
> Maybe
> http://www.voip-info.org/wiki/view/Asterisk+RealTime
> http://www.asteriskguru.com/tutorials/realtime_pgsql.html



Hi

any success stories of the setup

kindly post your config and information

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

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