Re: [asterisk-users] Quick Help, Anyone? EMERGENCY RESOLVED

[Mike Trest - Personal]

Wow!  Fast response.  Thanks.  I am back up now.  ..mike..


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[asterisk-users] Quick Help, Anyone? EMERGENCY

[Mike Trest - Personal]

Hi,

I have a disk crash on an 2006 vintage Asterisk box that has a g729 
license from Digium.
I have been able to re-install from media on the same chassis. 
Reactivation is in progress.
Good so far. . .

HOWEVER, I cannot locate my g729 files for the 'digits' portion of 
the "sounds".
I only need the ten digits  0.g729  1.g729 . . . 9.g729

Can some kind person zip them up and email attachment to me via 
[EMAIL PROTECTED]   or
give me a pointer of where to get them in next few minutes!  REALLY URGENT.

Thanks ..mike..


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Re: [asterisk-users] Next Move - Hosting

[Mike Trest - Personal]

Hi,
Your question is appropriate because you are asking about the best 
design approach.  Although it does have economic issues related.

First, let us dispatch the economics because they will impact your 
technical approach.  The real issue here is not cost of additional E1 
but exposure to fairly large liabilities for Trunk/International 
calls on the same E1 as your company.  If your shared customer is 
late on payment, your company will need to pay the E1 provider for 
all calls in any circumstances.  This could lead to cut off of your 
companies service if you do not pay.

Now the technical.   Avoid the MITEL unless it has features that you 
cannot manage on the Asterisk - - - which are very few if any ! ! !

IF YOU ARE GOING TO REMAIN E1 CARRIER BASED: I suggest you separate 
your clients onto a different E1 where the phone numbers assigned can 
be easily tracked separately by your Asterisk billing.   If your 
clients simultaneous calls justify the cost.  Put each client on a separate E1.

IF YOU ARE GOING TO ELECT VOIP CARRIER BASED:  I suggest you select a 
SIP provider who can issue local numbers to you and that you manage 
the clients on a pair of asterisks with "shared" service.   You can, 
with experience, deal with the over-lap extensions quite nicely since 
you are an IT guy.That pair of asterisks can both be served by a 
decent internet connection.   Depending on the class of machine(s) 
you use, you can support 200 ~ 400 simultaneous calls.

YES, pay greatest attention to the billing software you select 
because that will be you biggest "black hole" which can pull all your 
energy into it depths to resolve billing matters with your customers.

NO, I cannot tout one billing system over another.  Perhaps someone 
else will do that via private EMAIL
because that part of this discussions is NOT for this list.


Best wishes.  ..mike..

At 06:09 AM 4/4/2008, you wrote:
>I posted this to asterisk biz but didn't get a reply.. I didn't want to
>offend anyone being that this is kind of branching into hosting, and
>maybe outside of the remit of this list.
>
>Hi
>
>Been lurking on the user list for a while but I have some what of an
>immediate requirement and I'm wondering if you can suggest the best
>solution (if mines a rubbish idea)
>
>I have been testing Asterisk as a bolt on to our Mitel 3300.. its been
>doing some softphones for users abroad, etc and I'm happy with the fact
>I want to progress to a full system.
>
>However during this testing phase 2 customers of mine (I'm a IT Service
>Provider) have ask for some managed, collocated small business servers,
>which include the requirement for me to host their phones.
>
>No Problem I thought, I'm well on the way to this anyhow.
>
>So I'm thinking (although not tried it) that if I got my Asterisk box
>running for my company (E1 card for outside link) I would AIX the hosted
>PBX for the customers to my PBX to allow them to make outgoing calls. I
>would get my teleco to provide phone numbers for them and also get my
>PBX to redirect that number to the hosted PBX.
>
>Is this correct so far? Or should I keep their system separate on
>another E1? Or should I forget my PBX and push their incoming / outing
>calls out to a SIP / AIX provider on the net and wash my hands of it?
>
>Also I know you can run multi context on one host BUT can they also run
>the same extension numbers? Or would I have to let one company have
>401-410 and the next company have 411 to 420, etc, etc (I'm guessing
>that's the case)
>
>And lastly.. Call accounting.. Certainly found a lot of good info about
>certain call accounting applications but as anyone got any good feedback
>about one they personally use.. Id like to keep it GNU / Open Source /
>Free while I build myself up.. Although I don't want to compete with the
>big boys, Id like to think I could get 10-20 or so customers co-located.
>
>Cheers
>
>Tim
>
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Re: [asterisk-users] does the meetme module still require an external timing source?

[Mike Trest - Personal]

At 01:13 PM 4/1/2008, you wrote:
>Is app_conference stable now?
>
>I've never made it through a thousand calls without a crash.  (With a
>busy call center this doesn't take all that long.)
>
>-HJC


I have deployed a MEETME conference bridge based on a FARM of 
asterisks  with 6,000 conference ports active using basic meetme() 
with a very complex IVR front end that we wrote in perl for the 
customer specific needs.  Still in use after 3+ years.   Hundreds of 
thousands of total participants and millions of minutes later, still 
running.  Very happy with results.


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Re: [asterisk-users] Langugae issue

[Mike Trest - Personal]

Tzafrir, Anselm, and others.  Thanks for your comments on my 
suggestion to Ayman.

As one who is "familiar, but not-native speaker" with Arabic, Hebrew, 
and several other classical Semitic family languages,  it would 
require much more time to try to fit those into the linear structure 
of SAY.CONF  than to deal with it in a directly parsed manner.  I can 
say the same for some Asian languages too.  The results would 
recognized but would not be culturally acceptable.

OFF TOPIC COMMENTS:
I am constantly amazed at cross-language translations that try to 
follow the western language standards in computerized 
applications.  Historically, the use of numbers came relatively late 
to western languages.   While I am proud to be an American (as well 
as a computer-geek), I have crossed the multi-lingual & 
multi-cultural barriers many years back!
END OFF TOPIC COMMENTS.

..mike..

At 11:23 AM 3/30/2008, you wrote:
>On Sun, Mar 30, 2008 at 05:16:00PM +0200, Anselm Martin Hoffmeister wrote:
> > Am Sonntag, den 30.03.2008, 09:54 -0400 schrieb Mike Trest - Personal:
> > > Ayman,
> > >
> > > One solution is to write an AGI scrip to parse the number and read
> > > back in Arabic semantic order.  for the last two digits and for
> > > certain special numbers like 11 , 100 , 1000, ... .I must bring
> > > out my old Arabic language books to do this myself, but if you will
> > > share the language files with the asterisk group, then  I will make an
> > > example AGI for you that we can share with the list.
> > >
> > > If you are agreeable, let us  continue EMAIL messages privately until
> > > we have something working that we can share with the list.
> > >
> > > ..mike..
> >
> > Dear Mike,
> >
> > for me it seems that this is what say.conf is good for:
> > 
> http://svn.digium.com/view/asterisk/branches/1.6.0/configs/say.conf.sample?revision=105596&view=markup
> > (which seems to be considered the "new" format).
> >
> > Perhaps it would be better to implement Arabic there than by means of an
> > AGI script. Be sure to check with the developers wether this will be
> > relevant for Asterisk 1.4 or if you need to go with 1.6 SVN to benefit.
>
>say.conf works nicely for some languages. I was not able to make
>something useful enough with its syntax for Hebrew, and from the little
>I know of Arabic syntax, it will share the same problem.
>
>One basic problem is that there's no gender-form parameter anywhere in
>the interface.
>
>--
>Tzafrir Cohen
>icq#16849755  jabber:[EMAIL PROTECTED]
>+972-50-7952406   mailto:[EMAIL PROTECTED]
>http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
>
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Re: [asterisk-users] Langugae issue

[Mike Trest - Personal]

Ayman,

One solution is to write an AGI scrip to parse the number and read 
back in Arabic semantic order.  for the last two digits and for 
certain special numbers like 11 , 100 , 1000, ... .I must bring 
out my old Arabic language books to do this myself, but if you will 
share the language files with the asterisk group, then  I will make 
an example AGI for you that we can share with the list.


If you are agreeable, let us  continue EMAIL messages privately until 
we have something working that we can share with the list.


..mike..


At 09:20 AM 3/30/2008, aymen warfalli wrote:

Hi list

I add new directory for Arabic voices support and I 'd translated 
all the English voices files into Arabic , with language = ar ,and 
it is working fine ,except some problems in saying the number order 
,because the Arabic structure is quite different  for  numbers 
,where in  English language we can say twenty two while the order 
should be two and twenty  ,so please if you can guide me how to 
change the setting to do that .


regads

Ayman




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Re: [asterisk-users] Had it with Dell Garbage

[Mike Trest - Personal]

Steve,
I have fielded several hundred Asterisk and related VoIP boxes.
I buy SuperMicro 1-U units mostly.  I have also used their larger
units with RAID and a full load of ULTRA SCSI (for MySql application).

I like these because, after bad experience with DELL/COMPAQ/HP/IBM
compatibility issue, the supermicro systems always load and work with
all of the Fedora kernels will just with their RAID controllers.

..mike..


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Re: [asterisk-users] Pattern matching....

[Mike Trest - Personal]

[746][704][048]

[At 01:21 PM 2/21/2008, you wrote:
>On Thu, 2008-02-21 at 12:44 -0500, Michael Munger wrote:
> > Will this work to match any number from the 770,404, or 678 area
> > codes?
> >
> >
> >
> > _[404|770|678]NXX
> >
> >
> >
> > If this won’t work, is there a pattern that will do this?
> >
> >
>
>No, it won't work, there's no '|' for alternative matches, and no parens
>available for grouping, either. And your usage of char sets is off.
>Try something like this:
>
>_404NXX
>
>_770NXX
>
>_678NXX
>
>
>as three separate extensions.
>
>If you REALLY want to keep that as one extension, then you could:
>
>_NXXNXX  =>  {
> Set(areacode=${EXTEN:0:3})
> if ('${areacode}' = '404') {
> 
> } else if ('${areacode}' = '770') {
> 
> } else if ('${areacode}' = '678') {
> 
> }
>
>
>OR, you could do it this way, also:
>
>_NXXNXX  =>  {
> Set(areacode=${EXTEN:0:3})
> switch(${areacode})
> {
> case 404:
> 
> break;
> case 770:
> 
> break;
> case 678:
> 
> break;
> }
>
>This is, of course AEL code, and this stuff would be inside a context
>construct...
>
>murf
>
>
>
>
>
>
> >
> >
> > Yours,
> >
> > Michael Munger, dCAP
> >
> > 404-438-2128
> >
> > [EMAIL PROTECTED]
> >
> >
> >
> > Attachment encrypted? click here.
> >
> >
> >
> >
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>--
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>Software Developer
>Digium
>
>
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Re: [asterisk-users] Asterisk Scalability

[Mike Trest - Personal]

This is standard stuff.
I have switch over 200 simultaneous with g711 on a 1-U, Xeon-DualCore @ 3.0
using RH versions of Linux.  Even higher with "pass thru" (no-transcoding)
on g729.
..mike..


At 07:54 AM 2/8/2008, Femi wrote:
>This will be  closed service provider network with own VoIP phones and
>gateways so we can assume that there is no transcoding
>
>Femi
>
>
>-Original Message-
>From: [EMAIL PROTECTED]
>[mailto:[EMAIL PROTECTED] On Behalf Of Philipp
>Kempgen
>Sent: 08 February 2008 12:15
>To: Asterisk Users
>Subject: Re: [asterisk-users] Asterisk Scalability
>
>Femi wrote:
>
> > Does anyone have data on the switching capacity of Asterisk based on the
> > hardware?
> > I need to know what type of hardware would be required to switch 100
> > simultaneous calls as opposed to 1000 or 1 calls, no TDM just SIP to
>SIP
> > VoIP calls
>
>That largely depends on whether you need to do transcoding
>and between which codecs, etc.
>
>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
>
>Geschäftsführer: Stefan Wintermeyer
>Handelsregister: Neuwied B 14998
>
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Re: [asterisk-users] Bugs??

[Mike Trest - Personal]

When similar problem occurred, I traced the issue to remote GSM 
gateway with poor protocol stack.

The asterisk was doing exactly what it was supposed to do.

The IMMEDIATE work around we used was to put maximum call timer into 
extensions.conf


exten => s, 6,Set(TIMEOUT(absolute)=3660)

This  gives one hour+one minute.   With average call duration below 30 minutes
this worked quite well for our GSM traffic purposes.
You set to any value appropriate to your traffic.
..mike..
Currently I am running 120 VoIP SIP channels on my asterisk server 
but each day 2, 3 calls got hanged in asterisk,  and on asterisk CLI 
"show channels" showing us as call UP but in real there is no call.


When asterisk restarted the hanged calls removed from CLI with very 
high duration which damaged our billing system and customers 
accounts goes in high negative.


First I tried to get call info from asterisk mysql CDR using billsec 
field but the same result then I create PERL AGI to get duration 
from "ANSWEREDTIME" and same result.


Please have a look to the following URL which I put the result of 
"show channel " you can see the DIALSTATUS=CONGESTION 
but Elapsed Time: 20h54m16s which really strange and out of my mind 
to control such as call.


http://www.emafone.net/bugs.html

Please advice us if it is Bug and solved in some ver or its need 
some configuration to avoid this issue.


This is in both ver of asterisk 1.2 and 1.4





Regard,


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Re: [asterisk-users] Multi-SPAN (4xE1) Zap Group (Outbound)

[Mike Trest - Personal]

Thanks to all who replied privately as well!  ..mike..

At 03:41 PM 1/7/2008, you wrote:
>Mike Trest - Personal wrote:
> > Hi,
> > Can someone point me to a zapata.conf example that will create a
> > single DIAL OUT group including all 4 spans on a TE4XXP?



>Try:
>
>group=0,1
>channel  => 1-15,17-31
>group=0,2
>channel  => 32-46,48-62
>group=0,3
>channel  => 63-77,79-93
>group=0,4
>channel  => 94-103,110-124
>
>This allows you to use group 0 to dial out over all 4 spans, but 
>each span still has it's own
>group that you can use to troubleshoot.  You can break this down 
>even further if you need.


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[asterisk-users] Multi-SPAN (4xE1) Zap Group (Outbound)

[Mike Trest - Personal]

Hi,
Can someone point me to a zapata.conf example that will create a 
single DIAL OUT
group including all 4 spans on a TE4XXP?

One friend says to change the group number all to "1" on all 4 spans.
Another suggestions says it is possible to have these unique groups (1-4)
and to combine all 4 into a single group "5".

I like the second suggestion best.

Can you guide me to the correct changes for my current zapata.conf?
The 4 spans are stand alone E1/PRI trunks (Not NFAS).

The CURRENT channel and group statements are:
;Span  1group=1 channel  => 1-15,17-31
;Span  2group=2 channel  => 32-46,48-62
;Span  3group=3 channel  => 63-77,79-93
;Span  4group=4 channel  => 94-103,110-124




Thanks,  ..mike..


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