Ok.. will be there...
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Today's Topics:
1. Secondary Dialtone and selecting a specific line from Zap/g
(bilal ghayyad)
2. Re: Infuriating problems: no dial tone, dropped calls, no
voice: 1.2.13 and 1.4.11 (Steve Totaro)
3. Re: extensions.conf vs. AEL (Barzilai Spinak)
4. Re: Configuration files inside SQLite3 (Karsten Wemheuer)
5. About Megaco (Floyd)
6. Re: extensions.conf vs. AEL (Tilghman Lesher)
7. Line State (Gustavo Gonzalez)
8. Re: About Megaco (Steve Totaro)
9. Re: About Megaco (Brian West)
----------------------------------------------------------------------
Message: 1
Date: Thu, 4 Oct 2007 05:01:39 -0700 (PDT)
From: bilal ghayyad <bilmar_gh at yahoo.com>
Subject: [asterisk-users] Secondary Dialtone and selecting a specific
line from Zap/g
To: asterisk-users at lists.digium.com
Message-ID: <779244.20020.qm at web53908.mail.re2.yahoo.com>
Content-Type: text/plain; charset=iso-8859-1
Dear Walt;
Maybe I did not understand any thing from below :) -
Are you talking about configuration to be done on the
Telephone device is self or on the AVAYA server it
self? If it is on the telephone device, so how you
will give a second dial tone and you do not know if
there is available channel :) -
I am looking to have a second dial tone by doing such
configuration at AVAYA server itself, and that to be
used by all users of different IP Phones models (not
link sys only).
Can you help?
Regards
Bilal
--------------------------
For another tone frequency for the outside dialtone,
try putting this
value "0 at 0,350 at -14,440 at -14;*(.4/0/1),10(*/0/2+3)" in
the Outside
Dialtone field. It will give you a slight pause
followed by a different
dialtone frequency. On a Linksys/Siprua 941, that
would be at the top
of the Regional page.
However, you won't hear any secondary dialtone unless
you put a comma
after EVERY initial '9' in the dialplan string for
each line in use.
On a 941, that would be at the bottom of the Ext 1 and
Ext 2 pages of
the web interface. I suggest the dialplan string of:
(*xx|[1-7]xx|9,[3469]11|98|99|9,[2-9]xxxxxx|9,11|9,[2-9]xxxxxx|9,1[2-9]xx[2-
9]xxxxxx|9,011xxxxxxxxxxx.)
- Walt Joyce
Eric "ManxPower" Wieling wrote:> I can't help you with that. I only wanted to point
out that
ignoreopat > is not what you need.
>
> On Polycom SIP phones you continue dialtone by
placing a , in the > phone's dialplan. SIP phones have their own
internal dialplan that
is > not part of Asterisk's dialplan. You would have to
check the docs
for > your phone. Not all SIP phones can continue
dialtone.>
> bilal ghayyad wrote:
>
>>I need to select a line from the Zap group channel
>>using the SIP Phone (not FXO and not FXS ports).
>>
>>ignorepat does not work?
>>
>>Also, what is the method to let the second dial tone
>>has another tone frequency?
>>
>>Regards
>>Bilal
____________________________________________________________________________
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------------------------------
Message: 2
Date: Thu, 04 Oct 2007 07:53:14 -0400
From: Steve Totaro <stotaro at mail.schoffstall.com>
Subject: Re: [asterisk-users] Infuriating problems: no dial tone,
dropped calls, no voice: 1.2.13 and 1.4.11
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <4704D42A.3010102 at mail.schoffstall.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Christian Weeks wrote:> Hi
> I've had an asterisk setup for the past 15 months, based on the debian
> asterisk packaging. Until late August of this year, I had no problems
> once initial setup was complete- the system worked essentially
> flawlessly.
>
> Since August I have been having exceedingly infuriating intermittent
> problems that are causing me occasional periods of nasty trouble:
> 1. No Dial Tone. Every Sunday night at just prior to midnight ( about
> the last second before monday), the Dial tone ceases on all zap
> handsets. Investigation shows that the zaptel layer is not transmitting
> sound to the handset- ztmonitor shows it being sent, but nothing is
> arriving at the handset. (You can also see sound being rx from the
> handset). This problem has occured spontaneously at other times but the
> midnight thing is just plain odd (nothing is happening on that box at
> the time).
> 2. I tried to upgrade from the asterisk packages in debian etch (1.2.13)
> to sid (1.4.11). This added new problems to the mix. Suddenly, no voice
> was being offered through asterisk at all to any zap channel. I tracked
> it down to my use of T/t in the Dial strings- this was somehow
> preventing native bridging from occuring. This can be verified because
> other call routings (e.g. IAX->zap; SIP->zap) have the same problem
(and
> cannot be corrected by removing a t/T because there's no native
bridge).
> Reverting to 1.2.13 seems to have fixed this problem. (Is this some kind
> of sound path regression? Debug logging has shown nothing).
> 3. Finally, with 1.4.11 especially, the system seems to have been quite
> unstable, with asterisk crashing (and ringing every phone in the house
> incessantly- which my wife was NOT happy about), especially when two
> simultaneous calls overlap in some way (this hasn't crashed 1.2.13 as
> badly, but asterisk seemed to need a reset afterwards).
>
> I suspected a dodgy channel bank ( I had a really old eBay special for
> $20 which had timing problems from day 1 ) so I upgraded a little ( the
> Zhone - more expensive and not super, but at least it has firmware and a
> console for mgmt ). This has had some effect, but nothing has changed
> about the fundamental problem (1 above).
>
> Other hardware: the T1 interface card is the R1T1 from rhino, there is a
> Wildcard TDM400P REV I, with a single FXO port for the incoming line
> from the POTS. The computer itself seems quite fine, no sign of
> interrupt errors or other problems with the hardware (I ran a memtest
> and a cpuburn neither of which showed any issues). zttest shows nothing
> unusual (99.87% iirc over about 10 minutes).
>
> I am happy to share anything that will help resolve the issue- my feeble
> C skills in attempting some printf in the ast_channel_bridge command to
> see what was being chucked about pretty much failed entirely because the
> timing went badly off... Trying to chuck ast_log calls in there didn't
> work very well either :(
>
> Thanks
> Christian
>
I think you should stick with 1.2 for now at least until all the bugs
are worked out. Stability seems to be a real issue although many would
have you believe otherwise.
The other thing that jumps out is the exact timing of your issue in
number 1. Random problems can be very difficult to track down but
yours is consistent, so it should be pretty easy to find the culprit.
Check your cron jobs. If you have something running at almost midnight
on Sunday, that is probably your issue. Logs may be of help.
Thanks,
Steve Totaro
------------------------------
Message: 3
Date: Thu, 04 Oct 2007 09:07:47 -0300
From: Barzilai Spinak <barcho at creacion.com.uy>
Subject: Re: [asterisk-users] extensions.conf vs. AEL
To: murf at parsetree.com, Asterisk Users Mailing List - Non-Commercial
Discussion <asterisk-users at lists.digium.com>
Message-ID: <4704D793.1010708 at creacion.com.uy>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
All this discussion is pointless. As pointless as the discussion of
assembly versus high-level languages decades ago.
Except most people rooting for "extension.conf" don't even have
the
technical and conceptual amplitude to understand what they are talking
about... they just want some telephony system to make a quick buck, or
save in their LD calls...
A lot of Asterisk is technically and architecturally twisted, and
"spaghettied", and with many redundant ways of doing the same thing
(in
different stages of obsolescence, incompleteness, and (un)documented).
At least AEL is a step in the right direction (even though it has to
adapt itself to the ugliness that exists below..)
BarZ
Steve Murphy wrote:> On Wed, 2007-10-03 at 09:33 -0600, Anthony Francis wrote:
>
>> Eric "ManxPower" Wieling wrote:
>>
>>>> Let us not forget that AEL cannot be stored in a database
therefore
>>>> rendering you unable to utilize realtime.
>>>>
>>>>
>>> AEL converted into standard extensions.conf syntax in the dialplan.
>>>
>>>
>>>
>> Doesn't this render having used AEL pointless?
>>
>>
>
> Absolutely not!
>
> Reasons to use AEL:
>
> 1. Several semantic checks are done on the AEL that are NOT done if you
> go straight to extensions.conf. We try to protect you... from yourself.
>
> 2. At least one security issue in USAGE is avoided by having AEL compile
> the corresponding code; as to how many more issues will automatically be
> handled via
> AEL in the future, is impossible to say. We'll see. If you keep coding
> via
> extensions.conf, be prepared to make corrections... if you do it in AEL,
> a restart of Asterisk will hopefully suffice, after AEL is updated.
>
> 3. Syntax errors are reported by AEL. It is pretty good at catching all
> omissions
> and commissions. Better than the extensions.conf parser is. For example,
> I don't
> know if we catch it now, but if you accidentally say "extem =>
3,..."
> instead of
> "exten => 3,..." in extensions.conf, that line will silently
be dropped.
> Sure, we
> could fix this, but to fix ALL possible problems will require an
> expensive rewrite of the config file parser, from the ground up.
>
> 4. You are insulated against any mods to extensions.conf; like the
> change to ',' instead of '|' in app arguments. No changes
to AEL code
> are necessary.
>
> 5. In extensions.conf, you have to feed your dialplan to asterisk to
> find any problems. AEL provides the standalone parser, aelparse, so you
> can correct any problems BEFORE feeding it to a living asterisk.
>
> 6. AEL is easier to read, IF you take advantage of the ability to use
> tabs, etc. wisely. Especially for nested code. Staying away from goto as
> much as possible,
> and using the flow of control and looping statements will make your code
> easier to read, compose, and maintain in the future. It means fewer bugs
> in your code,
> and overall this all means lower cost. And higher profits.
>
> 7. Repetitious entry of "extenname, priority," in your tabular
> extensions.conf can lead to subtle errors that could be hard to find,
> ESPECIALLY if you resort to using priority NUMBERS instead of
"n". And,
> if you ARE so foolish as to use just raw numbers, and you have to insert
> or delete a line or two, you have to renumber
> the remaining lines, and heaven help you if you make a simple error, and
> accidentally skip a number.
>
> 8. Work flow. Since aelparse allows you to dump the compiled dialplan in
> extensions.conf format, you can still use stuff like realtime. You can
> use this output against machines that don't even have pbx_ael loaded,
> then, and you should be able to use 1.4 compiled dialplans on 1.2
> machines, as long as you are careful about what apps you call, and how
> you call them.
>
> 9. Easier to write code. Good Code. using Goto's in extensions.conf
will
> allow you to do anything you need to do, but it also results in
> spaghetti style code.
> While the original author might be able to decrypt it, and maintain it,
> unless it's really well commented, the next guy to play with it, is
> going to have a hard time. Following the flow of control thru spaghetti
> can get your adrenalin flowing-- and side affects from strange cases and
> leakage in the spaghetti can make some devilishly hard to solve
> problems.
>
> Think of and treat extensions.conf like assembly code.
>
> Think of and treat AEL like a high(er) level language. For those who
> never did the computer science thing, I have just one piece of advise,
> and ignore this at your peril: your dialplan is a work of computer
> programming. It's software. If you don't treat it that way, and use
good
> software methodologies, you'll pay your price.
>
> murf
>
>
>
>
> ------------------------------------------------------------------------
>
> _______________________________________________
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>
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------------------------------
Message: 4
Date: Thu, 04 Oct 2007 14:29:01 +0200
From: Karsten Wemheuer <kwem at gmx.de>
Subject: Re: [asterisk-users] Configuration files inside SQLite3
To: asterisk-users at lists.digium.com
Message-ID: <1191500941.1267.4.camel at baghira.temp.temp>
Content-Type: text/plain
Hi Mark,
Am Mittwoch, den 03.10.2007, 11:15 -0500 schrieb Mark
Michelson:> GNUbie wrote:
> > Hello all,
> >
> > Is it possible to store, read and write configuration files in an
> > SQLite3 database instead of using the configuration files inside the
> > /etc/asterisk/ directory? If it is then can you point me to the right
> > documentation on how to do this or probably hints on how to do this?
> >
> > Thank you in advance.
> >
> > GNUbie
> >
>
> It is possible to store configuration files in any relational database
> which has ODBC compatibility. Thus, sqlite qualifies. If you are using
> trunk, you won't even need to use ODBC, because Asterisk has native
> support for sqlite.
Are You shure the native support of asterisk is for SQLite3 as the
original poster asks for? AFAIK * supports SQlite (Version 2, not 3),
which has a completely different API.
Karsten Wemheuer
------------------------------
Message: 5
Date: Thu, 4 Oct 2007 08:08:18 -0500 (CDT)
From: Floyd <evekeko1 at yahoo.com>
Subject: [asterisk-users] About Megaco
To: asterisk-users at lists.digium.com
Message-ID: <531897.52561.qm at web90513.mail.mud.yahoo.com>
Content-Type: text/plain; charset=iso-8859-1
Hi all,
I've been searching for a while and haven't found if
asterisk supports already or if it's going to support
h.248. ????
thanks
Eve
____________________________________________________________________________
________
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Encuentra consejos para cuidar el lugar donde vivimos.
http://telemundo.yahoo.com/promos/mejorambientalista.html
------------------------------
Message: 6
Date: Thu, 4 Oct 2007 08:13:28 -0500
From: Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
Subject: Re: [asterisk-users] extensions.conf vs. AEL
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <200710040813.28610.tilghman at mail.jeffandtilghman.com>
Content-Type: text/plain; charset="iso-8859-1"
On Thursday 04 October 2007 07:07:47 Barzilai Spinak
wrote:> All this discussion is pointless. As pointless as the discussion of
> assembly versus high-level languages decades ago.
As one of the main architects, I don't find this discussion pointless. My
personal opinion of AEL is that it's coming along nicely, but it's still
not
up to the point where I would consider using it for most dialplans. That
day
will come, and I'm working with Steve Murphy to ensure that it does. One
thing that you did not see in the language wars of yesteryear was of the
assembly language changing in subtle ways, to make development in the
higher level language easier or more consistent, as is the case with AEL and
extensions.conf.
> Except most people rooting for "extension.conf" don't even
have the
> technical and conceptual amplitude to understand what they are talking
> about... they just want some telephony system to make a quick buck, or
> save in their LD calls...
This seems like a rather harsh indictment, when it really comes down to the
fact that writing in extensions.conf works today, and while AEL does work to
a certain extent, many people would rather not have to rewrite their
dialplans
every time an architectural flaw is found in AEL that limits what they can
do;
ergo, they write their stuff in extensions.conf until the point where AEL
becomes more trusted.
> A lot of Asterisk is technically and architecturally twisted, and
> "spaghettied", and with many redundant ways of doing the same
thing (in
> different stages of obsolescence, incompleteness, and (un)documented).
As a maintainer and architect, I would very much like to hear specific
criticisms on how you think this could be improved. We try to deprecate
specific functionality that doesn't work correctly or which could be
expressed
in better ways, which allows users of the system to transition away from
those
expressions to better methods over a period of time, instead of immediately
at
an upgrade; we believe this facilitates adoptions and upgrade processes.
> At least AEL is a step in the right direction (even though it has to
> adapt itself to the ugliness that exists below..)
All high level languages have to adapt themselves to the ugliness below.
That
is part of what makes them high-level languages.
--
Tilghman
------------------------------
Message: 7
Date: Thu, 4 Oct 2007 10:43:53 -0300
From: Gustavo Gonzalez <GGonzalez at grupo-gestion.com.ar>
Subject: [asterisk-users] Line State
To: "'asterisk-users at lists.digium.com'"
<asterisk-users at lists.digium.com>
Message-ID:
<E14F809A418076499DD160A2431B58E819F382 at mx.grupo-gestion.com.ar>
Content-Type: text/plain; charset="iso-8859-1"
Hello all, I need a little help to check the state of the line from
asterisk on aa TDM400P because when the telco lines goes down, asterisk get
that line for outgoing calls. There is a way to check it out?
And when all lines are busy to do outgoing calls how can i do to callback
the people that call when a line is free?
Thanks
Alejandro Gonz?lez
Grupo Gesti?n
4384-0660
www.grupo-gestion.com.ar
GGonzalez at grupo-gestion.com.ar
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Message: 8
Date: Thu, 04 Oct 2007 09:39:50 -0400
From: Steve Totaro <stotaro at first-notification.com>
Subject: Re: [asterisk-users] About Megaco
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <4704ED26.6070405 at first-notification.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Floyd wrote:> Hi all,
> I've been searching for a while and haven't found if
> asterisk supports already or if it's going to support
> h.248. ????
>
> thanks
> Eve
>
>
Try searching using MGCP which is what Megaco evolved into.
http://www.voip-info.org/wiki-Asterisk+MGCP+channels
Thanks,
Steve Totaro
------------------------------
Message: 9
Date: Thu, 4 Oct 2007 08:59:11 -0500
From: Brian West <brian.west at mac.com>
Subject: Re: [asterisk-users] About Megaco
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <C93EAA6C-BC02-44D3-B413-7ACFCA6A55C2 at mac.com>
Content-Type: text/plain; charset="us-ascii"
On Oct 4, 2007, at 8:39 AM, Steve Totaro wrote:
> Try searching using MGCP which is what Megaco evolved into.
>
> http://www.voip-info.org/wiki-Asterisk+MGCP+channels
>
> Thanks,
> Steve Totaro
Too bad the MGCP channel isn't the full implementation.
/b
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