Re: [asterisk-users] database queries from extensions.conf

[Al Baker]

Klaus Darilion wrote:
> Hi Jared!
>
> Thanks for the info - looks very flexible - you only have to edit 4 
> configuration files for a simple query :-)
>
> just a few questions:
> The ODBC library is unixodbc?
>
> How does it compare to the other solutions in terms of performance? e.g. 
> (I have to make several queries for each call (caller preferences, LNP, 
> LCR...)
>
> regards
> klaus
>
> Jared Smith schrieb:
>   
>> On Thu, 2008-11-13 at 15:16 +0100, Klaus Darilion wrote:
>> 
>>> What is the preferred way to make database lookups from within the dialplan?
>>>   
>> The preferred method is to use func_odbc, which takes SQL queries and
>> builds custom dialplan functions from them.  I've used it quite a bit,
>> and am very happy with it.
>>
>> I also presented on func_odbc at AstriCon, and you can download my
>> presentation:
>>
>> http://www.astricon.net/2008/glendale/web/presentations/DatabaseDriven_JSmith.pdf
>>
>>
>>
>> 
>   

Quote "
The preferred method is to use func_odbc, which takes SQL queries and 
builds custom dialplan functions from them. I've used it quite a bit,

and am very happy with it."

How can you be VERY HappY with  something that allows ONLY single statemts of 
SQL, ipso-facto you CANNOT
do
Begin Transaction
SQL Statement
SQL Statement
End Transaction

This isSOOO much more limited that the MYSQl add-on


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Re: [asterisk-users] database queries from extensions.conf

[Al Baker]

Klaus Darilion wrote:
> Wolfgang Pichler schrieb:
>   
>> Hi,
>>
>> you yould also use DBQuery (does only support mysql) - take a look at 
>> http://www.voip-info.org/wiki/view/Asterisk+cmd+DBQuery (it does also 
>> contain a cdr backend to write customzied cdr entries to the database)
>> 
>
> hi wolfgang!
>
> Have you programmed this yourself?
>
> Do you know how it compares to MYSQL function and func_odbc?
>
> regards
> klaus
>
>
>   
>> regards,
>> Wolfgang
>>
>> Klaus Darilion schrieb:
>> 
>>> Hi!
>>>
>>> What is the preferred way to make database lookups from within the dialplan?
>>>
>>> I only know the MYSQL function from asterisk-addons. Are the other 
>>> methods too? (e.g. for postgresql, unixodbc)
>>>
>>> thanks
>>> klaus
>>>
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>   

func_odb only allows a SINGLE  database statement
Ergo you cannot do Transactions or Multi-statement
SQL
It is a MAJOR  Backstep in DB Access.
The MYSQL add-on is  the BEST way to access DB from Dial Plan
Digium should support and ADD to this rather than non putting a SINGLE 
mention of it in the
last book and making no mention of it at Astricon.

With this Add-on, and if DIGIUM would fix the brain dead implement ion 
of REAL-TIME
for Exstensions.conf, things would/could be Soo Sweet. [ Can I get a 
Amen for
having LABELS for steps in exstensions.conf when it is in Real Time ?  
Why the Heck do I have to use Different Format for Applications in 
exstensions.conf ]


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Re: [asterisk-users] SendImage()

[Rob Hillis]

Philipp Kempgen wrote:
> SendImage() in 1.4:
>
> ---cut---
>   SendImage(filename): Sends an image on a channel.
> If the channel supports image transport but the image send
> fails, the channel will be hung up. Otherwise, the dialplan
> continues execution.
> The option string may contain the following character:
> 'j' -- jump to priority n+101 if the channel doesn't support image 
> transport
> This application sets the following channel variable upon completion:
> SENDIMAGESTATUSThe status is the result of the attempt as a text 
> string, one of
> OK | NOSUPPORT
> ---cut---
>
> in 1.6:
>
> ---cut---
>   SendImage(filename): Sends an image on a channel.
> Result of transmission will be stored in SENDIMAGESTATUS
> channel variable:
> SUCCESS  Transmission succeeded
> FAILURE  Transmission failed
> UNSUPPORTED  Image transmission not supported by channel
> ---cut---
>
> Is there any reason to break backwards compatibility?
> Why is "SUCCESS" better than "OK" and "UNSUPPORTED" better than
> "NOSUPPORT"?
> IMHO there was no need to change anything except for adding
> the "FAILURE" return status.
>
> Might be a -dev question though

This is typical of the criticism that has been levelled at Digium time 
and time and time again - making changes that don't really add any 
functionality, but break compatibility.

I had a hell of a time migrating a couple of systems from 1.2 to 1.4 - 
so much that I have no plans at all in the near future of migrating them 
from 1.4 to 1.6.

Even the comments made at the time suggesting a parsing tool be provided 
to point out where changes to dialplan code would be required got a 
"nice idea" response, but nothing has been forthcoming.

This habit of breaking functionality for limited or no reason, plus 
making the results from functions far /less/ useful (note my previous 
complaints about the REALTIME() function) and more difficult to use is 
the biggest problem with Asterisk bar none.

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[asterisk-users] How does IMAP notify Asterisk that I've read a message?

[Barry L. Kline]

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I have an Asterisk box sitting between the PSTN and a legacy PBX.  I
have successfully configured Asterisk to use IMAP for voicemail and have
written the necessary script to turn the MWI indicator (via a .call file
to the PBX) on and off.  I have two issues still outstanding:

1) When the user listens to his voice mail via the phone, it will be
announced that the caller is unknown, in spite of the fact that the
email headers show the appropriate Callerid(num) information.  I can
live with that, but I'll eventually need to get it fixed.

2) If I "listen" to the voicemail using my email client, the MWI on the
phone is not turned off, which isn't surprising given that my script
needs to be called to generate the .call file.  What I don't know is
how, exactly, Asterisk is notified that I've listened to my voicemail
via email.  Does Asterisk poll the server?  If so, where is the
frequency of the poll set?  Can Asterisk be configured to call the
script again when the messages are read and the MWI should be turned off?

The docs don't say anything about this and I've not found anything in my
googling that has given me any leads?

I'm currently using Asterisk 1.4.22.

Thanks for any information that you can provide.

Barry


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Re: [asterisk-users] MOH Realtime Problem

[Sebastian]

Someone?? Any idea??

 

 

De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Sebastian
Enviado el: Friday, November 21, 2008 9:09 PM
Para: asterisk-users@lists.digium.com
Asunto: [asterisk-users] MOH Realtime Problem

 

Hi,

 

I'm having 2 problems:

 

1)  MOH in realtime is not working, I have configured it but never go to
look at the database, no warning or error found and I can do a query using
realtime and the family from the cli. 

2)  I have SIP phones via realtime, if I register one of them and a call
to a queue comes the call is never delivered to the phone, I have to make a
call from the phone so the phone start getting calls from de queue.
(RESOLVED)

 

 

Any ideas??

 

 

Ast 1.6.0.1

 

 

 

 

 

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[asterisk-users] SendImage()

[Philipp Kempgen]

SendImage() in 1.4:

---cut---
  SendImage(filename): Sends an image on a channel.
If the channel supports image transport but the image send
fails, the channel will be hung up. Otherwise, the dialplan
continues execution.
The option string may contain the following character:
'j' -- jump to priority n+101 if the channel doesn't support image transport
This application sets the following channel variable upon completion:
SENDIMAGESTATUSThe status is the result of the attempt as a text 
string, one of
OK | NOSUPPORT
---cut---

in 1.6:

---cut---
  SendImage(filename): Sends an image on a channel.
Result of transmission will be stored in SENDIMAGESTATUS
channel variable:
SUCCESS  Transmission succeeded
FAILURE  Transmission failed
UNSUPPORTED  Image transmission not supported by channel
---cut---

Is there any reason to break backwards compatibility?
Why is "SUCCESS" better than "OK" and "UNSUPPORTED" better than
"NOSUPPORT"?
IMHO there was no need to change anything except for adding
the "FAILURE" return status.

Might be a -dev question though.


   Philipp Kempgen

-- 
http://www.das-asterisk-buch.de  -  http://www.the-asterisk-book.com
Amooma GmbH - Bachstr. 126 - 56566 Neuwied  ->  http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
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Re: [asterisk-users] Large Asterisk installations (~10, 000 extensions), preferably at universities

[William Muriithi]

Bourvine,


>
> So, why won't we save the big bucks we pay them, hire two professionals
> (who cost less) and support an open source code by ourselves? This way
> we depend on ourselves only.
>
>
>
> Thanks, __Yehavi:

I remember hearing University of Pennsylvania have been using Asterisk
for sometime. I am not certain where I came across that information,
but google confirmed it as a fact. And you may need to ask for more
details from Digium as they worked together, or call the school. I am
relatively certain they would share their experience.  The deployment
was of 15,000 extensions, just about what you have in mind. Below is
some articles.

http://www.networkworld.com/news/2007/071707-open-source-voip.html

http://www.digium.com/en/company/casestudies/viewcasestudies/University-of-Pennsylvania

William
>
>
>
>
> 2008/11/21 Grygoriy Dobrovolskyy <[EMAIL PROTECTED]>
>
>
>
>2008/11/21 Yehavi Bourvine <[EMAIL PROTECTED]>
>
>Hello,
>
>
>
>  Our university has to upgrade soon its old Nortel PBX's which
> holds around 10,000 extensions tied to 5 PBXes. Up to now we thought
> about commercial solutions but now there is a window openning for open
> source solution. However, I need examples to convince that this solution
> is feasible, and preferably at other universities.
>
>
>
>Are there any pointers for such installations?
>
>
>
>   Thanks! __Yehavi:
>
>
>
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>
>Hello very interesting project you have, however asterisk is not
> a registry server, i suggest that you use opensips/opense/kamalio for
> your registrar, from where you dispatch to you asterisk servers, inside
> a good environment with a controlled network and nice tagged voip flow
> you could acheve a good results.
>
>
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>
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> Message: 9
> Date: Fri, 21 Nov 2008 09:46:13 -0500
> From: Alex Balashov <[EMAIL PROTECTED]>
> Subject: Re: [asterisk-users] Large Asterisk installations (~10, 000
>extensions), preferably at universities
> To: Asterisk Users Mailing List - Non-Commercial Discussion
>
> Message-ID: <[EMAIL PROTECTED]>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
> Jason Aarons (US) wrote:
>
>> Just switching from Nortel to something else may not eliminate
>> hardware/software failures, or prevent those without experience from
>> pushing the enter key at the wrong time.
>
> One also has to keep in mind - Asterisk, like any large open-source
> project, gets a lot more QA, patches and bug fixes than any commercial
> product sold in the intra-industrial channel (i.e. excluding consumer
> mass-market stuff) ever will!  It has a massive installed base, many
> users reporting bugs through an open and easy to understand process, and
> a large community either directly or derivatively involved in
> contributing fixes and testing code.
>
> How much installed base from which to harness that kind of large-scale
> technical feedback does Nortel have?  Avaya?  Cisco?
>
> Asterisk has by far the best QA mechanism.  In terms of potential bugs
> that impact "mission-critical" availability, I would feel better using
> it than any of these black-box, proprietary vendor solutions any day.
>
> --
> Alex Balashov
> Evariste Systems
> Web: http://www.evaristesys.com/
> Tel: (+1) (678) 954-0670
> Direct : (+1) (678) 954-0671
> Mobile : (+1) (706) 338-8599
>
>
>
> --
>
> Message: 10
> Date: Fri, 21 Nov 2008 15:46:59 +0100
> From: Philipp Kempgen <

Re: [asterisk-users] Asterisk Instant message passing with eyebeam

[Alex Balashov]

Asterisk does not support the MESSAGE method.

Max Alex wrote:

> Hi All,
> I am searching about asterisk IM message passing with eyebeam.
> but i am not able to send instant message to another registered users.
> i am working in asterisk 1.4 branch.
> i have tested within call and without call but there is no message recieved.
> and every time i got error user not found in eye beam.
> and in asterisk i got Method is not implemented.
> 
> Can anybody helps me in this?
> If any patches are there then please let me know.
> Thanks in advance!!
> Thanks,
> Max Alex
> Voip Developer
> 
> 
> 
> 
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-- 
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599

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Re: [asterisk-users] IMAP voicemail with Exchange (was: A way to run extenrnotify when IMAP events take place...)

[Sigma Networks]

Noah Miller wrote:

Hi Jeff -

  

I have IMAP voicemail working with Exchange 2003 using a single username and
password for multiple mailboxes.



Sorry to hijack this thread (at least I changed the Subject), but this
really caught my eye.  I was under the impression that Exchange's IMAP
doesn't have the master user feature and therefore can't do single
username authentication for multiple mailboxes.  Care to share how you
accomplished this?


Thanks,
Noah

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from my recent research Exchange2003 does have a master user that can be 
given write access to all mailboxes.   Exchange2007, though removes the 
MasterUser capability.



# Asterisk/Exchange Voicemail 

# Asterisk 1.6.0 + Exchange 2007 SP1 Unified Messaging 





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Re: [asterisk-users] presence with polycom DND

[Paul Hales]

You might have to look at writing a forward macro on the server that
would be dialed by the DND button - that also changed the device status
to busy(via the devstate app?).

My guess is that it would be less than 10 lines of dialplan code, but
maybe 1.6 only

PaulH


cfh wrote:
> hi,
>
> I have configured asterisk 1.4.21 to control the presence BLF (hint + 
> watch buddy parameter)  of Polycom phones (650,550,330) and it works good.
>
> But when I set the phones on Do Not Disturb (DND) on the server there 
> arent sip notifications and the presence doesnt change.
>
> On the Polycom configuration I have try to use the server based DND 
> option but i dont know how to use this with asterik.
>
> What can i do ? Are there some workaround to use the DND button and the 
> BLF on asterisk?
>
> thanks
>
> cfh
>
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[asterisk-users] CDR Desgin

[Grey Man]

I've taken the liberty of starting a new thread to discuss the design
of the Asterisk CDR mechanism. The discussion has been kindly
initiated by murf putting together a proposal:
http://svn.digium.com/svn/asterisk/team/murf/RFCs.

After reading the proposal I still don't think it's the right way to
go. To my mind adding more channel variables increases the complexity
in a situation that is already overly so. I think it's a mistake to
try and think about all the different call scenarios and come up with
little tricks for the more complicated ones. There will always be
something missed; app_shotgun initiates calls to 100 random numbers
and as soon as three or more calls are answered it will start randonly
transferring them amongst each other at 2 second intervals.

I think it's important to clarify at the outset what a CDR should be.
The most fundamental requirement for CDRs is that they accurately
record the following pieces of information for EVERY call entering or
leaving the system (note every means every and not; "channel" calls
but not "peer" calls).

1. Destination (aka as A Number)
2. AccountCode (aka as B Number)
3. Call Start Time (answer time),
4. Duration.

Of course adding extra information can be very useful and I'm not
proposing any fields be removed from the current implementation
(although for pity's sake one change that should be made it to use a
GUID/UUID for the CDR's uniqueid and save endless confusion).

People that really do need verbose or enhanced CDRs to do things like
tracking a call's flow as it travels in and out of queues, parking
lots etc. would be better off using AMI or the new CEL and not CDRs.
At the very least if problems arise with their call flow tracking they
will still be able to rely on the accuracy of the CDRs to piece it
altogether to work out what's going wrong.

My proposal of creating a 1-to-1 relationship between CDRs and
Asterisk channels already exsits but somewhere along the line it's
going awry. As an experiment, and to actually do something instead of
continually moaning about it, I started commenting out the blocks of
code in res_featrures.c and sip_channel.c that muck around with the
channel CDRs when a transfer occurs. The results of that were that the
CDRs for blind and attended transfers actually got better! They're
still not quite right but are pretty close with only one CDR on each
having a wrong detstination.

Regards,

Greyman.

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Re: [asterisk-users] MozIAX - Mozilla IAX2 soft-phone 3sec delay

[Jean-Denis Girard]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Hi,

Joseph a écrit :
| Did anybody tried MozIAX extension? It is Mozilla IAX2 soft-phone.
| http://moziax.mozdev.org/
|
| I tried it yesterday on eee pc, connected to asterisk on local LAN and
the performance is terrible!
| The delay is about 2sec or 3sec. and very bad echo.
| I think it is the implementation of their IAX2 in their add on, as I
have tried external mic. and the same delay problem.
|
| As a comparison I've tried DIAX over dial-up connection and the voice
quality was acceptable with very little delay.
|

MozIAX and DIAX share the same IAX2 implementation (libiaxclient, see
http://iaxclient.wiki.sourceforge.net/projects), so I doubt this is the
problem. I'm not aware of such delay problems; maybe there is a sound
daemon running on the EEE PC. Maybe you could ask on the MozIAX mailing
list.


Thanks,
- --
Jean-Denis Girard

SysNux  Systèmes  Linux  en Polynésie française
http://www.sysnux.pf/   Tél: +689 50 10 40 / GSM: +689 79 75 27
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Re: [asterisk-users] Need Recording Solution in Asterisk

[David Cook]

> One of our client Bank has 900 employees working in different locations.
> They need to record all internal and external calls. Can any body suggest
Call Recording Solution for this 
> requirement. We need to know the Hardware / Bandwidth and  all
requirements and costing.

Few questions first 
1. Why are they being recorded (business need)?
2. Does the value of the recording remain constant over time or diminish?
3. What criteria will you be required to retrieve the recording with?
4. Do you expect users to retrieve their own recordings or make requests of
a records management operations staff?
5. Does everything need to be on-line or near-line/off-line and do you
require a data management and migration solution?
6. Do you need to do word spotting and trend analysis on the content of
these recordings (target marketing and customer service analysis typically)?

Recording the call is quite easy. Storing it for retrieval which is
acceptable to the business under their potentially diverse requirements is
the tough part to nail down.

There are commercial products like Witness out there which do a good job of
this at a premium price. If the business drivers have low impact, you could
simply record in asterisk and archive the files with some creative scripting
and database work.

You said this is a bank so I'm presuming they will have a formal risk
analysis methods in place which would guide you through qualifying the
requirements. Check out what the IT/CIO folks have to help you out in this
manner.

-dbc. 


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Re: [asterisk-users] hint priority with 50 channels

[Loic Didelot]

Thanks for the help. 

Loic.

On Sat, 2008-11-22 at 11:41 -0600, Tilghman Lesher wrote:
> On Saturday 22 November 2008 10:31:34 Loic Didelot wrote:
> > Thanks for that idea. That what I had in mind. Now I just need to figure
> > out where to change and test for side effects.
> 
> In main/pbx.c, search for "ast_add_extension2" (in my branch, it's at line
> 1917, but it may vary slightly).  The top line of the function has:
> char hint[AST_MAX_EXTENSION].  Change AST_MAX_EXTENSION to something
> larger (say, 512) and recompile.  BTW, I'm going to make this change in trunk,
> so starting in 1.6.1, it will no longer be as much of a problem.
> 


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Re: [asterisk-users] Setting up to reveive faxes.

[Gordon Henderson]

On Sat, 22 Nov 2008, Tilghman Lesher wrote:

> On Saturday 22 November 2008 09:10:39 Gordon Henderson wrote:
>> On Sat, 22 Nov 2008, Noah Miller wrote:
>>> Hi Ken -
>>>
 Hey, all.  When I last was heavily into Asterisk (1.0.x), setting up to
 receive faxes was, well, a PITA, what with having to patch the Asterisk
 install with various driver patches and this, that, and the other.

 Is that still true?  Is there a fax HOWTO out there that reflects
 Asterisk 1.4.x?
>>>
>>> Not sure if you mean IP faxing or TDM faxing, but I don't think you'll
>>> need to do any patching.  In general check out:
>>> http://www.voip-info.org/wiki-Asterisk+fax
>>>
>>> For IP faxes, check out the wiki here:
>>> http://www.voip-info.org/wiki/view/Asterisk+T.38
>>>
>>> AFAIK: 1.4.x can't terminate IP faxes - only pass through, while 1.6.x
>>> can pass through and terminate with the help of spandsp.
>>
>> Does this mean that 1.4 lost an ability that 1.2 currently has? Right now,
>> with 1.2, I use spandsp (or rather the RxFAX application which uses
>> spandsp) to terminate faxes and save them as TIF files, then outside
>> asterisk email them to the appropriate destination... I'm currently
>> looking at moving to 1.4 and this is something that's essential for me...
>> I guess it's time to look deeper into this.
>
> You missed the part about IP faxes.  Yes, you can receive faxes over a Zaptel
> or DAHDI channel on your system.  That feature has not gone away.  It is the
> feature of being able to terminate T.38 faxes that was never in 1.2, isn't in
> 1.4, but is in 1.6.

OK. We were talking about T.38, and I must have missed that.

>  RxFax has never worked with IP faxes.  It requires the
> use of a Zaptel/DAHDI channel.

Er, wrong, and didn't we discuss this some weeks back?

Maybe what you need to say is: "RxFax will never work reliably with IP 
faxes."

Myself and a colleague have sent several faxes via VoIP in the UK between 
servers at the end of ADSL lines at opposite ends of the country, and 
they've worked just fine. It's not something I'd sell or recommend because 
it's not reliable, but it can work.

I also know people who've got FAX machines connected to ATAs VoIP 
connected to PBXs with ISDN lines and those FAX machines work OK too, so 
although it's better IP over a LAN than over the public Internet, it's 
still FAX over IP, as is the case where people are running IAXMODEM on a 
separate PC to the Asterisk box connected to HylaFax...

Gordon

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Re: [asterisk-users] Setting up to reveive faxes.

[Tilghman Lesher]

On Saturday 22 November 2008 09:10:39 Gordon Henderson wrote:
> On Sat, 22 Nov 2008, Noah Miller wrote:
> > Hi Ken -
> >
> >> Hey, all.  When I last was heavily into Asterisk (1.0.x), setting up to
> >> receive faxes was, well, a PITA, what with having to patch the Asterisk
> >> install with various driver patches and this, that, and the other.
> >>
> >> Is that still true?  Is there a fax HOWTO out there that reflects
> >> Asterisk 1.4.x?
> >
> > Not sure if you mean IP faxing or TDM faxing, but I don't think you'll
> > need to do any patching.  In general check out:
> > http://www.voip-info.org/wiki-Asterisk+fax
> >
> > For IP faxes, check out the wiki here:
> > http://www.voip-info.org/wiki/view/Asterisk+T.38
> >
> > AFAIK: 1.4.x can't terminate IP faxes - only pass through, while 1.6.x
> > can pass through and terminate with the help of spandsp.
>
> Does this mean that 1.4 lost an ability that 1.2 currently has? Right now,
> with 1.2, I use spandsp (or rather the RxFAX application which uses
> spandsp) to terminate faxes and save them as TIF files, then outside
> asterisk email them to the appropriate destination... I'm currently
> looking at moving to 1.4 and this is something that's essential for me...
> I guess it's time to look deeper into this.

You missed the part about IP faxes.  Yes, you can receive faxes over a Zaptel
or DAHDI channel on your system.  That feature has not gone away.  It is the
feature of being able to terminate T.38 faxes that was never in 1.2, isn't in
1.4, but is in 1.6.  RxFax has never worked with IP faxes.  It requires the
use of a Zaptel/DAHDI channel.

-- 
Tilghman

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Re: [asterisk-users] hint priority with 50 channels

[Tilghman Lesher]

On Saturday 22 November 2008 10:31:34 Loic Didelot wrote:
> Thanks for that idea. That what I had in mind. Now I just need to figure
> out where to change and test for side effects.

In main/pbx.c, search for "ast_add_extension2" (in my branch, it's at line
1917, but it may vary slightly).  The top line of the function has:
char hint[AST_MAX_EXTENSION].  Change AST_MAX_EXTENSION to something
larger (say, 512) and recompile.  BTW, I'm going to make this change in trunk,
so starting in 1.6.1, it will no longer be as much of a problem.

-- 
Tilghman

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Re: [asterisk-users] MozIAX - Mozilla IAX2 soft-phone 3sec delay

[Joseph]

On 11/22/08 13:29, Tim Panton wrote:
>
>On 21 Nov 2008, at 21:12, Joseph wrote:
>
>> Did anybody tried MozIAX extension? It is Mozilla IAX2 soft-phone.
>> http://moziax.mozdev.org/
>>
>> I tried it yesterday on eee pc, connected to asterisk on local LAN  
>> and the performance is terrible!
>> The delay is about 2sec or 3sec. and very bad echo.
>> I think it is the implementation of their IAX2 in their add on, as I  
>> have tried external mic. and the same delay problem.
>
>>
>>
>> As a comparison I've tried DIAX over dial-up connection and the  
>> voice quality was acceptable with very little delay.
>
>
>
>Sounds to me as if you'll need to tweak the audio settings on the eee .
>DIAX probably does that for you, it might be worth looking in /proc/snd
>while DIAX is running to see how it configures the audio device, then
>getting moziax to do the same.
>
>I'm surprised you got reasonable response over a dialup connection -
>which codec are you using ?
>
>Tim.

Over dial up, I used GSM;
I'll try to install Ubuntu and try it again. 
Any idea what to look for in snd file?  
If there is snd file they must have put it somewhere else as it is not in 
/proc/snd.

-- 
#Joseph
GPG KeyID: ED0E1FB7

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Re: [asterisk-users] ISDN Cause codes

[Robert Boardman]

I have found that the messages are not played as the hangup cause clears 
down the channel and passed hangup to the other end

should I have progress() before the dial command?

Robb

Martin Smith wrote:
> Hi Robert,
>
> I'd recommend the following options for Dial() so that you corroborate
> operator messages w/ cause codes:
>
>  1. remove R and r - we've found this can supress operator recordings on
> early audio
>  2. likewise, remove m to disable MOH
>
> Also, check the values of DIALSTATUS to compare to HANGUPCAUSE.
>
> Good luck,
>
> Martin Smith, Systems Developer
> [EMAIL PROTECTED]
> Bureau of Economic and Business Research
> University of Florida
> (352) 392-0171 Ext. 221 
>
>  
>
>   
>> -Original Message-
>> From: [EMAIL PROTECTED] 
>> [mailto:[EMAIL PROTECTED] On Behalf Of 
>> Robert Boardman
>> Sent: Friday, November 21, 2008 3:07 PM
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> Subject: Re: [asterisk-users] ISDN Cause codes
>>
>> Thanks for the reply
>>
>> Could you be a little more specific?
>>
>> Thanks
>> Robb
>>
>> Martin Smith wrote:
>> 
>>> Hi Robert,
>>>
>>> I'd suggest tweaking the Dial() arguments so that you (1) 
>>>   
>> allow early
>> 
>>> audio, (2) don't force it play ringing to the calling party, and (3)
>>> modify any other options to be as relaxed as possible. if 
>>>   
>> you make those
>> 
>>> changes, you'll start hearing the operator message 
>>>   
>> recordings and those
>> 
>>> are sometimes easier to reference against the cause codes.
>>>
>>> Cheers,
>>>
>>>
>>> Martin Smith, Systems Developer
>>> [EMAIL PROTECTED]
>>> Bureau of Economic and Business Research
>>> University of Florida
>>> (352) 392-0171 Ext. 221 
>>>
>>>  
>>>
>>>   
>>>   
 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Robert Boardman
 Sent: Thursday, November 20, 2008 5:56 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [asterisk-users] ISDN Cause codes

 Hi All

 Just been looking at stats for one of my sites, and I'm 
 conserned about 
 the number of error cause codes being returned from the telco

 for example

 12000 calls processed

 131 are cause code 31* normal. unspecified.*

 139 are cause code 28 * invalid number format (address 
 
>> incomplete).*
>> 
 112 are cause code 1 *Unallocated (unassigned) number.

 *this adds up to about 3% of calls not completing.

 there are various other codes including 17 busy 34 channel 
 unavaliable 
 and 44 requested channel unavaliable, which add up to another 1%.*
 *
 the telco says there is no problem with the line, I'm trying to 
 understand what the problem could be

 now  alot of calls complete OK so I don't think is my configs

 Any advice would be appriciated

 Versions
 asterisk 1.4.21.1
 zaptel 1.4.12.1


 Robb

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Re: [asterisk-users] hint priority with 50 channels

[Loic Didelot]

Thanks for that idea. That what I had in mind. Now I just need to figure
out where to change and test for side effects.

Loic

On Sat, 2008-11-22 at 08:19 -0700, Anthony Francis wrote:
> Loic Didelot wrote:
> > Why do we put 80 character limits, computers have GB or memory? 
> >
> >
> > Loic
> >
> >   
> Asterisk was written in c and they do have to declare how much memory 
> should be reserved for a variable in c, so the programmer arbitrarily 
> chose a number. They may have put some logic or investigation behind it, 
> but thats pretty much it. If you don't like that chose, edit the 
> definition in the source code and then recompile and voila! you have 
> your longer string handling.
> 
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Re: [asterisk-users] Setting up to reveive faxes.

[Anthony Messina]

On Saturday 22 November 2008 09:10:39 am Gordon Henderson wrote:
> On Sat, 22 Nov 2008, Noah Miller wrote:
> > Hi Ken -
> >
> >> Hey, all.  When I last was heavily into Asterisk (1.0.x), setting up to
> >> receive faxes was, well, a PITA, what with having to patch the Asterisk
> >> install with various driver patches and this, that, and the other.
> >>
> >> Is that still true?  Is there a fax HOWTO out there that reflects
> >> Asterisk 1.4.x?
> >
> > Not sure if you mean IP faxing or TDM faxing, but I don't think you'll
> > need to do any patching.  In general check out:
> > http://www.voip-info.org/wiki-Asterisk+fax
> >
> > For IP faxes, check out the wiki here:
> > http://www.voip-info.org/wiki/view/Asterisk+T.38
> >
> > AFAIK: 1.4.x can't terminate IP faxes - only pass through, while 1.6.x
> > can pass through and terminate with the help of spandsp.
>
> Does this mean that 1.4 lost an ability that 1.2 currently has? Right now,
> with 1.2, I use spandsp (or rather the RxFAX application which uses
> spandsp) to terminate faxes and save them as TIF files, then outside
> asterisk email them to the appropriate destination... I'm currently
> looking at moving to 1.4 and this is something that's essential for me...
> I guess it's time to look deeper into this.

with 1.6, i'm using this: http://messinet.com/?page_name=AsteriskFAXGateway to 
do email-fax gatewaying in both directions.

i suppose it could also work with 1.4's tx_fax and rx_fax which i think are in 
the agx-ast-addons package available at 
http://sourceforge.net/projects/agx-ast-addons/

-- 
Anthony - http://messinet.com - http://messinet.com/~amessina/gallery
8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E


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Re: [asterisk-users] hint priority with 50 channels

[Anthony Francis]

Loic Didelot wrote:
> Why do we put 80 character limits, computers have GB or memory? 
>
>
> Loic
>
>   
Asterisk was written in c and they do have to declare how much memory 
should be reserved for a variable in c, so the programmer arbitrarily 
chose a number. They may have put some logic or investigation behind it, 
but thats pretty much it. If you don't like that chose, edit the 
definition in the source code and then recompile and voila! you have 
your longer string handling.

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Re: [asterisk-users] Setting up to reveive faxes.

[Gordon Henderson]

On Sat, 22 Nov 2008, Noah Miller wrote:

> Hi Ken -
>
>> Hey, all.  When I last was heavily into Asterisk (1.0.x), setting up to
>> receive faxes was, well, a PITA, what with having to patch the Asterisk
>> install with various driver patches and this, that, and the other.
>>
>> Is that still true?  Is there a fax HOWTO out there that reflects Asterisk
>> 1.4.x?
>
> Not sure if you mean IP faxing or TDM faxing, but I don't think you'll
> need to do any patching.  In general check out:
> http://www.voip-info.org/wiki-Asterisk+fax
>
> For IP faxes, check out the wiki here:
> http://www.voip-info.org/wiki/view/Asterisk+T.38
>
> AFAIK: 1.4.x can't terminate IP faxes - only pass through, while 1.6.x
> can pass through and terminate with the help of spandsp.

Does this mean that 1.4 lost an ability that 1.2 currently has? Right now, 
with 1.2, I use spandsp (or rather the RxFAX application which uses 
spandsp) to terminate faxes and save them as TIF files, then outside 
asterisk email them to the appropriate destination... I'm currently 
looking at moving to 1.4 and this is something that's essential for me... 
I guess it's time to look deeper into this.

Gordon

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Re: [asterisk-users] Setting up to reveive faxes.

[Noah Miller]

Hi Ken -

> Hey, all.  When I last was heavily into Asterisk (1.0.x), setting up to
> receive faxes was, well, a PITA, what with having to patch the Asterisk
> install with various driver patches and this, that, and the other.
>
> Is that still true?  Is there a fax HOWTO out there that reflects Asterisk
> 1.4.x?

Not sure if you mean IP faxing or TDM faxing, but I don't think you'll
need to do any patching.  In general check out:
http://www.voip-info.org/wiki-Asterisk+fax

For IP faxes, check out the wiki here:
http://www.voip-info.org/wiki/view/Asterisk+T.38

AFAIK: 1.4.x can't terminate IP faxes - only pass through, while 1.6.x
can pass through and terminate with the help of spandsp.


- Noah

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[asterisk-users] [SOLVED] Re: Upgrade 1.4.19 to 1.6 => segementation fault

[Ronald Wiplinger (Lists)]

Ronald Wiplinger (Lists) wrote:
> During compiling I have not seen an error, however, when I start
> asterisk again it ends with:
>
>
> app_morsecode.so => (Morse code)
>   == Registered custom function 'SYSINFO'
>  func_sysinfo.so => (System information related functions)
> Segmentation fault (core dumped)
>
>
> How can I figure out what is wrong?
>   
I removed all modules, which were left from the 1.4 installation and now
it works!



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[asterisk-users] IMAP voicemail with Exchange (was: A way to run extenrnotify when IMAP events take place...)

[Noah Miller]

Hi Jeff -

> I have IMAP voicemail working with Exchange 2003 using a single username and
> password for multiple mailboxes.

Sorry to hijack this thread (at least I changed the Subject), but this
really caught my eye.  I was under the impression that Exchange's IMAP
doesn't have the master user feature and therefore can't do single
username authentication for multiple mailboxes.  Care to share how you
accomplished this?


Thanks,
Noah

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Re: [asterisk-users] MozIAX - Mozilla IAX2 soft-phone 3sec delay

[Tim Panton]

On 21 Nov 2008, at 21:12, Joseph wrote:

> Did anybody tried MozIAX extension? It is Mozilla IAX2 soft-phone.
> http://moziax.mozdev.org/
>
> I tried it yesterday on eee pc, connected to asterisk on local LAN  
> and the performance is terrible!
> The delay is about 2sec or 3sec. and very bad echo.
> I think it is the implementation of their IAX2 in their add on, as I  
> have tried external mic. and the same delay problem.

>
>
> As a comparison I've tried DIAX over dial-up connection and the  
> voice quality was acceptable with very little delay.



Sounds to me as if you'll need to tweak the audio settings on the eee .
DIAX probably does that for you, it might be worth looking in /proc/snd
while DIAX is running to see how it configures the audio device, then
getting moziax to do the same.

I'm surprised you got reasonable response over a dialup connection -
which codec are you using ?

Tim.


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Re: [asterisk-users] Large Asterisk installarions (~10, 000 extensions), preferably at universities

[Tim Panton]

On 22 Nov 2008, at 00:06, Michael Collins wrote:

>> Date: Fri, 21 Nov 2008 16:20:28 -0600
>> From: Terry Wilson <[EMAIL PROTECTED]>
>> Subject: Re: [asterisk-users] Large Asterisk installarions (~10,
> 000
>>  extensions), preferably at universities
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>>  
>> Message-ID: <[EMAIL PROTECTED]>
>> Content-Type: text/plain; charset=US-ASCII; format=flowed; delsp=yes
>>
>>> Yehavi Bourvine wrote:
>>>
 OK, but I still did not get a reply to my original question: Why
 using
 SIP registrar in front of Asterisk and not simply use bare
> Astersik?
 can't it handle the load? (remember - in my case it doesn't handle
 the
 RTP, only signalling). Can't it handle so much registrations? (I am
 using realtime DB, it is has any relevance).
>>>
>>> My experience has shown that using a dedicated registrar for large
>>> installs is more effective;  it doesn't tie up resources on the
>>> Asterisk
>>> box with all those registration refreshes, for one.  A product built
>>> to
>>> be a high-throughput standalone registrar will handle the
> concurrency
>>> requirements and perform better.
>>
>> I've looked at doing various things to chan_sip to improve signaling
>> performance (hash tables for call lookups, etc.)  I gave up when I
>> realized that the overhead of handling the RTP was so far above the
>> overhead of processing SIP signaling that it didn't really matter
>> much.  The only reason I have ever had to use a SIP registrar  
>> (OpenSER
>> in my case) was if I needed to load balance calls across multiple
>> asterisk servers.  If most of the phones are not separated by a NAT
>> from Asterisk (as would be the case in something like a University
>> network), the registration timeout could be set to a relatively high
>> value w/o causing any problems which would cut down on some of the  
>> SIP
>> traffic from registrations.
>>
>> In fact, I just ran some tests using SIPp and w/o any audio, using
>> realtime w/ 10k accounts I can register 100/second while doing 10
>> calls/second.  If you are looking just at registrations every 15
>> minutes or so, that is 90k devices that could register to asterisk.
>> This was using 1.6.0.1 on my little HP amd64 development box--not
>> anything near the kind of machine that you would probably install  
>> in a
>> large installation.  Asterisk just gets faster and faster.  Some of
>> the "it isn't good at x" stuff comes from experiences with older
>> releases.
>
> In a HA and/or high volume scenario I worry about stuff like this that
> has been in tree since 1.0 or earlier and is in 1.6, channel.c lines
> 3825~3828:
>
>/* XXX This is a seriously wacked out operation.  We're
> essentially putting the guts of
>   the clone channel into the original channel.  Start by
> killing off the original
>   channel's backend.   I'm not sure we're going to keep this
> function, because
>   while the features are nice, the cost is very high in terms
> of pure nastiness. XXX */
>
> That's not something I want in my high-end, high-capacity,
> high-availability production system!

Actually that's exactly the kind of comment I _do_ want to see in an  
opensource
platform. It is honest, open and an encouragement to others to think  
of a better fix.

Discourage poor coding, critique the
design etc  - but please don't discourage this kind of commenting, it  
is the kind
of thing that helps one find a bug _infinitely_ faster that you could  
without the
clue the original author left for you.

Tim.



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Re: [asterisk-users] hint priority with 50 channels

[Loic Didelot]

Here is the scenario. My customer has 3 groups each with 10 users. Every
user has the users of his group programmed on the snom phone for pickup
etc... This works perfectly. 

Now they want one global button for each group. so if any phone in
another group rings then they can pick it up.

Why do we put 80 character limits, computers have GB or memory? 


Loic


On Fri, 2008-11-21 at 13:19 -0700, Anthony Francis wrote:
> Just curious but why would you want to have a lot of devices all have 
> the exact same state information?
> 
> Philipp Kempgen wrote:
> > Loic Didelot schrieb:
> >
> >   
> >> I noticed that my hint priority stops working when I add to many
> >> extensions/channels. It looks like everything exceeding 80 characters is
> >> discarded. 
> >>
> >> By stop working I mean the status is and stays "Unavailable".
> >>
> >>
> >> This works
> >> exten => *1,hint,SIP/loicvoip1_1&IAX2/loicvoip1_1&SIP/loicvoip1_1_a1
> >>
> >> This does not work:
> >> exten =>
> >> *1,hint,SIP/bla1&SIP/bla2&SIP/bla3&SIP/bla4&SIP/bla9&SIP/bla5&SIP/bla6&SIP/bla7&SIP/loicvoip1_1&IAX2/loicvoip1_1&SIP/loicvoip1_1_a1
> >>
> >>
> >> I tested on several asterisk 1.4 versions like 1.4.21*.
> >>
> >>
> >> Is this a bug or something like working as designed?
> >> 
> >
> > It's by design. 80 characters is likely to be the limit.
> >
> >   
> >> Is there another
> >> possibility to monitor a bigger number of channels?
> >> 
> >
> > In Asterisk 1.6 you could build something with "Custom" hints
> > and DEVICE_STATE().
> >
> >Philipp Kempgen
> >
> >   
> 


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