[Asterisk-Users] Re: Meridian 808 Function

[Miguel Ruiz Velasco Sobrino]

>Message: 19
>Date: Mon, 30 May 2005 11:12:13 -0500
>From: "Carlos Chavez" <[EMAIL PROTECTED]>
>Subject: Re: [Asterisk-Users] Meridian 808 Function
>To: Asterisk Users Mailing List - Non-Commercial Discussion
>
>Message-ID: <[EMAIL PROTECTED]>
>Content-Type: text/plain;   charset=iso-8859-1
>
>On Mon, 30 May 2005 08:53:12 -0700 (PDT), Miguel Ruiz Velasco Sobrino wrote
>> Hi,
>> Some time ago, there was a discussion about the inability of nortel 
>> meridian pbx to dial analog tones thru an meridian ATA, and the work 
>> arround was to enable 808 function that makes the dtmf tones long 
>> for the current call.
>>
>> The nortel meridian is connected via a nortel ATA to a TDM400 to a 
>> FXO port.
>> 
>> Anyone can say me who to actually use that function (you dial 
>> something or is pbx programation)?
>> 
>> Thanks for the hint
>> 
>  The way to use is (which I do not really recommend) is that the nortel
> user will dial the extension that is connected to the Asterisk server and then
> press the FUNCTION button on his/her phone followed by 808.  After 1 or 2
> seconds the phone will say "Long Tones".  You can then dial the extension you
> want on the * server.
> 
>  The best way to integrate with a Meridian is to use FXS ports connected
> to trunk line ports and then configure the PBX to dial something like 7 to get
> a dial tone from the Asterisk server.  That way your users do not have to
> remember a complicated procedure.
>
> --
> Carlos Chavez
> Director de Tecnología
> Telecomunicaciones Abiertas de México S.A. de C.V.
> Tel: +52-55-91169161 Ext 2001

Thanks for the tip, it worked as seen on TV!
I also think the users are [somewhat] uncoperative and memoryless, but i can't 
connect
FXS to trunk because that pbx was already full. Also having some problems with 
disconnect
supervision. I know FXO is a hassle, but had no other means of solution.

Thanks

Miguel Ruiz Velasco

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[Asterisk-Users] Meridian 808 Function

[Miguel Ruiz Velasco Sobrino]

Hi,
Some time ago, there was a discussion about the inability of nortel meridian 
pbx to dial
analog tones thru an meridian ATA, and the work arround was to enable 808 
function that
makes the dtmf tones long for the current call.

The nortel meridian is connected via a nortel ATA to a TDM400 to a FXO port.

Anyone can say me who to actually use that function (you dial something or is 
pbx
programation)?

Thanks for the hint

Miguel Ruiz Velasco

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[Asterisk-Users] Re: Enhanced Queue App Revisited

[Miguel Ruiz Velasco Sobrino]

--- [EMAIL PROTECTED] wrote:
>Matt Roth wrote:
> 
>> Preferably, I would like an out-of-the-box solution, but custom-coding 
>> is an option as long as the necessary data is available from Asterisk. 
>> If anyone could point me in the right direction, it would be greatly 
>> appreciated.
>

You may want to see the aheeva offering  on call-center
management systems using asterisk as the call routing core.
They co-sponsored the asterisk von booth, so i guess they have a sort of 
agreement with digium.

Miguel Ruiz Velasco

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[Asterisk-Users] Re: wctdm and two tdm cards

[Miguel Ruiz Velasco Sobrino]

>> I have a server I'm working on here with two tdm cards in it.
>> 4 FXS and 4FX0. Both cards work fine on their own. The problem
>> lies with using both in the system at once. I have verified the
>> IRQ's are fine. I have tried switching the slots the cards reside in, no 
>> luck though. I am using ACPI but not APM. I am using gentoo latest, with 
>> vanilla 2.6(.10) kernel and udev. CVS as of CVS-HEAD-03/02/05-03:42:41.
>> 
>> The problem is as follows:
>> 
>> If I power up the system from system off, the cards both get detected
>> 
>> If I reboot the system with reset button, ctrl alt del, or 'reboot'
>> the TDM04P does not get detected.
>> 
>> If I then reboot, then hit the power button, and let it turn off, then
>> turn it back on again and boot, it detects both cards fine.
>> 
>> I have tried searchign the list archives, but I have not had much luck. 
>>   One person on IRC mentioned he's seen this before, but didn't have any
>> solutions.
>> 
>> Does anyone here know what might be the problem? or have a fix/work 
>> around? I know I shouldnt be rebooting servers, but I have to make sure 
>> it works upon reboot as it is going to be installed in a power-outtage 
>> happy part of the world :)
>
>I'm not having any problems like that with RHv9 (2.4 kernel), so I'd have
>to guess the issue is 'timing' related in whatever script that loads your
>tdm-zaptel drivers.
>
>As I recall (as a non-v2.6 user), there was an issue with timing and 
>someone added a sleep/wait statement in the startup script to bypass the
>problem. Might consider finding your startup and add some additional time
>to that sleep/wait.
>
>Another approach to isolating the problem is to load the drivers by hand
>paying close attention to error messages, delays, etc. If your not sure
>how to do that, read your startup script and simply do those steps manually.
>
>Someone mentioned unplugging power and/or removing the card. That approach
>is totally BS. The same startup process is run regardless of whether one
>is rebooting or starting from power-on. There is nothing on the tdm card
>that stores values (no flash, no battery backup mem, etc). If the startup
>script operates "one time" from any startup mode, it is setting the tdm
>registers, etc, correctly.
>
>I'm away from the office this week, but I recall there was a readme shipped
>with the zaptel source that discusses kernel 2.6 timing issues. Might look
>for that in your src directory.

The approach of non-rebooting but power off and then power on (cold reboot), is 
NOT BS.
For me had been the only way to "reboot" the server with the TDM cards. If you 
make a hot
reboot (al least with my cards) the modprobe will have fatal errors and won't 
load the
cards. I think it's MoBo related, maybe some kind of IRQ assignment not 
released.

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[Asterisk-Users] Re: List tips for new subscribers

[Miguel Ruiz Velasco Sobrino]

--- [EMAIL PROTECTED] wrote:
>[EMAIL PROTECTED] (Andrew Kohlsmith) writes:
>> You did type it yourself, but you replied to a message in a thread
>> and erased everything, thus screwing up the threading.  I think
>> that's what he was referring to.
>>
>Wouldn't it be nice if the mailinglist software were hacked to enforce
>some rules?
>
>  * reject all HTML email
>  * reject any mail with more quoted text than original text
>  * reject any mail that starts a totally new subject but 
>   threads to a different unrelated one.
>   eg. has references, but new subject with no "(was: oldsubject)" 
>  * reject any mail that has "re: " or the same subject line as
>   other msgs, but no references.  (This one needs to be done
>   very carefully.)
The last two ideas are really dangerous, given the very high volume of this 
list, a lot
of people is subscribed to the digest version of the list (myself included), 
and that
break the association between header and reference. The issue of being 
subscribed to the
digest was discussed in november.
So that would mean lot's of people would be marginated to post. Also, each MUA 
handles 
things sightly different and making post acceptance rules so strict would have 
a terrible
effect.


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[Asterisk-Users] Re: IAX2 Trunk Problems with NAT

[Miguel Ruiz Velasco Sobrino]

Simple! first: return the iax.h to it's original state because it's a config 
issue, not a
code one.

In the internal server, you need to put in the relevant section a host 
directive with the
name or IP of the external server AND a register under the [general] section of 
the
iax.conf

In the external server, in the definition you need to put host=dynamic.
Then hoy need to  dial with the definition section name

***internal:
*iax.conf
[general]
register => user:[EMAIL PROTECTED]
[iax-external]
host=external-hostname

*extensions.conf
 ... Dial (IAX2/iax-external/phonenumber)

***external:
*iax.conf
[iax-internal]
host=dynamic

*extensions.conf
... Dial (IAX2/iax-internal/phonenumber)

This also addresses the problem of someone having two iax2/udp streams, one of 
each call,
even with trunking enabled.


> From: Mustafa N. Deeb   
> 
> To: 'Asterisk Users Mailing List - 
> Non-Commercial Discussion' 
> 
> Sent: Monday, February 07, 2005 9:45 AM
> 
> Subject: [Asterisk-Users] 
> 
>  
> 
> Hi, 
> 
> I have successfully configured an IAX trunk between 2 asterisks, calls can
> go through both ways without any problems, NAT in the middle of course
> (iptables)
> 
> Now , leave them for a while ,   and make a call from the external server ,
> it doesnt go through, 
> 
> Dial from the internal one, everything works fine again..
>  
> 
> Now , it is clearly  a problem in the NAT engine, although IAX shouldnt
> have this problem
> 
> is there something I can do to prevent asterisk from tearing down the trunk
> while it is not used
> 
> or , modify the iptables so that  to allow the incoming UDP connections
> 
> Cheers


=
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[Asterisk-Users] Re: Re: load balancing 20 asterisk servers

[Miguel Ruiz Velasco Sobrino]

--- [EMAIL PROTECTED] wrote:

> > DNS based load ballancing has it's place, as dose using an
> > application level switch.  
> > 
> > Say an earthquake takes out your California data center.
> > Shortly the DNS servers will notice and pull that center's
> > record.  However do to caches and all this is not fast
> > and users will notice.
> > 
> > What the switch does is route at the protocol level between
> > local machines.  You can take a machine off line and no one
> > will notice.  Works great until the big quake a backhoe
> > takes out a fiber cable ro there is a fire flood or who
> > knows what.
> 
> You have fiber-seeking-backhoes in your area? Wow!

Once in the data center where the company i worked had their equipment, a car 
crashed in
the outside wall, and was just the place where 1 of the two whole floor power 
panel were,
so half of the equipment crashed due to lack of energy, and also broke a fibre
patchpanel, so the available bandwidth was minuscule.
And in other ocasion, one repair gang in the city literally cutted a fibre 
cable with a
street digger, and the repair lasted about 1 month.
Massive catastrophic failure due to external causes IS a possible scenario.
And yes, it IS a world class data-center.

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[Asterisk-Users] Re: load balancing 20 asterisk servers

[Miguel Ruiz Velasco Sobrino]

As being said, the cost of HW based solutions is in many cases too expensive to 
be
practical, leave alone to have a spare one to give you true high-availibility.
If you complain about DNS caching and timeouts not being respected, you can do 
a fairly
easy thing.

As said in the follow-up, you can drop in a data center many tiny boxes.
When one box stops working or crashes you can bring down it's interface and 
give a spare
box the same IP of the [now] defunct machine. If the box becomes irresponsive 
(a real OS
crash) and you have IMPI 2.0 capable MoBo's (all intel server boards have that, 
also some
other brands), you can remotely shutdown or reset the machine to avoid IP 
clashes. Intel
has a command line utility (also the graphical console) to manage that, i've 
using for a
while this and is absolutely wonderful.

Obiously, you will need a separate LAN with privates IP's to make much of the
administration and the DB access, and use the public LAN only for 
internet-related
things, so each box has it's own fixed private IP and only the public IP 
changes.

Indeed, with IMPI 2.0 is possible to remotely power-up a machine (if you are
enviromentally concerned... or if the datacenter metters the electricity you 
use), so you
don't need to be running all your spare servers waiting for a failure, maybe 
only one and
have the others shut-down until needed.


--- [EMAIL PROTECTED] wrote:
Message: 5
Date: Wed, 2 Feb 2005 21:57:42 -0600 (CST)
From: Joe Greco <[EMAIL PROTECTED]>
Subject: Re: [Asterisk-Users] Re: load balancing 20 asterisk servers
To: asterisk-users@lists.digium.com
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; charset=us-ascii

> I'm trying to stay away from a software based load balancer cause what
> happens if that server fails?
> Its far less likely for a piece of dedicated hardware to fail than an actual
> computer.

You really ought to open up one of those pieces of dedicated hardware
sometime and see what's inside.

Yep, it's software based.

Heck, many of the so-called pieces of dedicated hardware are in fact nothing
more than a fancy rack mount PC.  Open up something like a CacheFlow server
and you find an Intel server motherboard, some propietary software, and that
is about it.  Heck, go on eBay and pick yourself up some of those nice F5
BigIP ... rack mount PC's.

Some of the newer stuff is software based with some ASIC assistance for
SSL/compression.  I know that F5 has made an effort to not look like a PC
anymore, for example, and has integrated some switchlike capabilities in
their product.

Still, when it comes right down to it, the traffic direction logic in these
things is software based.

Incidentally: one of the /down/sides to these devices, aside from being
hellishly expensive, is that when it blows at 5:01PM on a Thursday evening
when Friday is Christmas, even if you have the best service contract, it
can be a trying experience to get service.  PC's have the distinct
advantage that you can actually plan to have spare parts available, and
on top of it, you can actually build high quality redundant equipment
fairly inexpensively.

AIC RMC2N-XP Chassis$150
EMACS R2G-6350P Power   $300
SuperMicro P4SC8$300
Intel P4-3.0 Prescott   $175
Memory  as desired
CF Adapter  $ 20
1GB CompactFlash Boot   $ 60

$1005

Toss in a monster passive heatsink and you have a system that isn't
particularly susceptible to the loss of any single moving part.

Of course, you have to be able to sysadmin your way out of a cardboard
box, so it's not like it's cost-free, but here's the thing:

If my hypothetical load balancer fails at 5:01PM on Xmas eve, I can:

1) Grab the cold spare I built because it's cheaper to do two of these
   than a single expensive HW based solution

2) Configure the hot spare I built into production (again because it's
   cheaper).

3) Grab a desktop PC and stick a few Intel GigE NIC's in it and go to
   town.

4) At least have a reasonable chance of figuring out some other way to
   fix things temporarily.

So.

What's really interesting is that even some "networking hardware" is
actually just computing gear on steroids.  I recently saw a SMC 8624T
24-port gigE switch, and it appears to be a bunch of Broadcom GigE chips
with a CPU that runs some (can't recall which) embedded OS.  VxWorks?

... JG


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[Asterisk-Users] Re: 911 and Cops knocking on my door

[Miguel Ruiz Velasco Sobrino]

It may be a problem of the PSTN not catching the initial 5

change your dial string from 
DIAL(zap/1/${EXTEN})
to
DIAL(zap/1/ww${EXTEN})

note a "w" before the number, each w makes a 1/2 sec pause and you can put many 
of them,
so if your PSTN lags a little to give you dialtone, that probably make it. Also 
test
(like some one else said) with the  5 "912" XXX number, to save you a cop visit.
In a panasonic PBX, i had to put 4 of them!!.

--- [EMAIL PROTECTED] wrote:

looks like an ignorepat problem on the first *number* (single) dialed 
(i.e., trying to ignore the number 9 on an outbound call.)

try to make a call to 591-2079.


On Wed, 2 Feb 2005, Andrew Niemantsverdriet wrote:

> Hi,
> I am quite new to asterisk so I am not sure what is needed to figure
> out this problem. If more information is needed and not provided I
> will gladly provide it.
>
> I have a very basic asterisk setup. 1 x100p card and a grandstream
> handytone 286.  I can make calls fine to most phone numbers from the
> handytone device the trouble seems to come when I dial this number
> 591-1079. It puts me through to the local 911 dispatch. Causing the
> police to show up at my doorstep and check to make sure everything is
> alright.
>
> I can see why I think; 5 "911" 079. But I don't understand why it is
> being handled this way. Can somebody offer me some guidance on how to
> get this to stop?
>
> TIA
> _
> /-\ ndrew

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[Asterisk-Users] Re: load balancing 20 asterisk servers

[Miguel Ruiz Velasco Sobrino]

--- [EMAIL PROTECTED] wrote:


> The DNS approach does not handle single or multiple system failures,
> only very elementary load balancing over a lengthy period of time.

Are you shure of that? I'm aware that the load criteria is trickier, but very 
possible.

If you use DDNS (dynamic DNS) using Bind 9. You have to run a health monitor 
(like a tcp
or ping monitor) in one server (like the dns one) if one server dies, a script 
removes
it's A record automagically from the pool, and even with a script that monitors 
the load
you can dynamically add and remove the entry of each individual server in the 
DNS server.
And you would not need to care if the load balancer sends the SIP stream to one 
server
and the RTP stream to an other or in the case of outgoing connections or 
whatever
extrange situation.

Use "nsupdate" utility for doing DDNS, it's really simple and incredibly 
powerfull. Also
because all the requests are digitally signed, you will likely don't have 
security
problems.



> You may want to consider a simpler aproach, why don't you balance the load 
> via DNS?
> If you put in a zone file various A records for the same machine, but with 
> different
> IP's, BIND will catch the trick and send a different IP (from the pool yo 
> defined) each
> time a DNS request arrives. That's a simple way of doing that, it will 
> definively work
> for termination, but you may have to think more who to cope with origiation 
> (outgoing
> calls), since different clients will be connected to different servers.
> 
> 
> --- [EMAIL PROTECTED] wrote:

> 
> We use it on our web and mail server to load ballance across multiple 
> hosts. The way we have it configured
> it will maintain a session for 15 minutes between a client and a 
> specific server. So long as you have
> qualify=yes in your configuration files, each client will continue to 
> talk to the one server until they are turned off/
> deactivated for at least 15 minutes (or whatever time period you 
> configure into it). I've not tested LVS with
> Asterisk, but it may be the right direction for you to take.
> 
> Cheers,
> -Shaun
> 
> Matthew Boehm wrote:
> 
> >I've read several other emails and pages on the wiki but none give any
> >deffinate answers. if you have 20 asterisk servers each with 4 pri's, all
> >running RealTime Extensions and RealTime SIPBuddies from the same MySQL
> >server, what prevents you from putting all 20 servers behind a single load
> >balancer? That way all of your UA's can use the same IP to register to; vs
> >maintaining which customer is assigned to which machine.
> >
> >perhaps its just that i am not that familiar with load balancers. i was
> >under the impression that a load balancer could/would send each recieved
> >packet to a different server.
> >this doesn't matter in the case of register requests since all asterisk
> >boxes share same SIP registry database.
> >
> >but what about invite requests and the rtp stream? you would have a majorly
> >broken conversation if each packet in the rtp stream went to a different
> >asterisk box.
> >
> >or are load balancers SIP aware? or is there some sort of session control
> >that the balancer is aware of and will send all packets in a "sip session"
> >to the same asterisk box?
> >
> >and then what about meet me conferences? if 10 UA's all dial a conference
> >DID number and all 10 get balanced to 10 different servers then they are all
> >sitting in seperate rooms right?
> >
> >hints, opinions, facts...all welcome and appreciated.
> >
> >-Matthew





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[Asterisk-Users] Re: load balancing 20 asterisk servers

[Miguel Ruiz Velasco Sobrino]

You may want to consider a simpler aproach, why don't you balance the load via 
DNS?
If you put in a zone file various A records for the same machine, but with 
different
IP's, BIND will catch the trick and send a different IP (from the pool yo 
defined) each
time a DNS request arrives. That's a simple way of doing that, it will 
definively work
for termination, but you may have to think more who to cope with origiation 
(outgoing
calls), since different clients will be connected to different servers.


--- [EMAIL PROTECTED] wrote:

Hi,

You may want to look into LVS (Linux Virtual Server). It allows load 
ballancing in a highly configurable way.
http://www.linuxvirtualserver.org/

We use it on our web and mail server to load ballance across multiple 
hosts. The way we have it configured
it will maintain a session for 15 minutes between a client and a 
specific server. So long as you have
qualify=yes in your configuration files, each client will continue to 
talk to the one server until they are turned off/
deactivated for at least 15 minutes (or whatever time period you 
configure into it). I've not tested LVS with
Asterisk, but it may be the right direction for you to take.

Cheers,
-Shaun

Matthew Boehm wrote:

>I've read several other emails and pages on the wiki but none give any
>deffinate answers. if you have 20 asterisk servers each with 4 pri's, all
>running RealTime Extensions and RealTime SIPBuddies from the same MySQL
>server, what prevents you from putting all 20 servers behind a single load
>balancer? That way all of your UA's can use the same IP to register to; vs
>maintaining which customer is assigned to which machine.
>
>perhaps its just that i am not that familiar with load balancers. i was
>under the impression that a load balancer could/would send each recieved
>packet to a different server.
>this doesn't matter in the case of register requests since all asterisk
>boxes share same SIP registry database.
>
>but what about invite requests and the rtp stream? you would have a majorly
>broken conversation if each packet in the rtp stream went to a different
>asterisk box.
>
>or are load balancers SIP aware? or is there some sort of session control
>that the balancer is aware of and will send all packets in a "sip session"
>to the same asterisk box?
>
>and then what about meet me conferences? if 10 UA's all dial a conference
>DID number and all 10 get balanced to 10 different servers then they are all
>sitting in seperate rooms right?
>
>hints, opinions, facts...all welcome and appreciated.
>
>-Matthew
>
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[Asterisk-Users] Re: Terrible inbound call quality vs. outbound

[Miguel Ruiz Velasco Sobrino]

I've had similar problems but with dial-up modems.

ISP's mantain large queues in the inbound side of your connection to maximize 
download
speed, but that same hurts latency on your side. You may be saturating the BW 
and thus
the queue makes it's job.
Use the bw conditioner that is described in the advanced linux routing howto, 
in the
cookbook, that is named a thing like "the ultimate bw conditioner, fast 
downloads and
uploads and blablabla". Modify it by putting the ports that the RTP or IAX 
stream pases,
assigning them with a filter to the interactive class.
Also don't forget to put the correct uplink and downlink values, or you will be 
putting a
bw restrictor.

The thing that is very weird is that only inbound calls are affected, I would 
think that
both inbound and outbound calls were affected.

> 
> --- [EMAIL PROTECTED] wrote:
> 
> Hi. I'm having a terrible time with call quality coming into my * box.
> 
> I'm using VoicePulse over a 1.5/1.5 mbit line. Outbound calls are 
> crystal clear on both the RX/TX sides of the conversation. Inbound 
> calls, though, are HORRIBLY garbled on the RX side. I can barely hear 
> the caller, but they report my quality is fine. Getting loads of 
> garbled sounds and weird echoes. (Could just be jumbled up voice 
> packets?)
> 

Miguel Ruiz Velasco



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[Asterisk-Users] Re: Terrible inbound call quality vs. outbound

[Miguel Ruiz Velasco Sobrino]

I've had similar problems but with dial-up modems.

ISP's mantain large queues in the inbound side of your connection to maximize 
download
speed, but that same hurts latency on your side. You may be saturating the BW 
and thus
the queue makes it's job.
Use the bw conditioner that is described in the advanced linux routing howto, 
in the
cookbook, that is named a thing like "the ultimate bw conditioner, fast 
downloads and
uploads and blablabla". Modify it by putting the ports that the RTP or IAX 
stream pases,
assigning them with a filter to the interactive class.
Also don't forget to put the correct uplink and downlink values, or you will be 
putting a
bw restrictor.

The thing that is very weird is that only inbound calls are affected, I would 
think that
both inbound and outbound calls were affected.

--- [EMAIL PROTECTED] wrote:

Hi. I'm having a terrible time with call quality coming into my * box.

I'm using VoicePulse over a 1.5/1.5 mbit line. Outbound calls are 
crystal clear on both the RX/TX sides of the conversation. Inbound 
calls, though, are HORRIBLY garbled on the RX side. I can barely hear 
the caller, but they report my quality is fine. Getting loads of 
garbled sounds and weird echoes. (Could just be jumbled up voice 
packets?)


Miguel Ruiz Velasco



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[Asterisk-Users] Re: Making digital/data calls through asterisk

[Miguel Ruiz Velasco Sobrino]

There is the ITU V.150 that defines a framework for transmiting modem signals 
using VoIP,
in a efficient way, but there is no implementation other that R&D in big 
companies.

If you only want to forward an analog modem call over a PRI to the PSTN, you 
may need a
channel bank connected to the TE410E card. The use of a TDM400 from many post 
on the
lists seems unreliable for low speed faxes, so i guess it would be a nightmare 
to use
with a modem.

And if you are using an IDSN modem, well, i don't know if anybody has managed 
to put that
to work. If you manage to make it work, please add an entry in the wiki for 
that, i think
it would be "a little step for a man, but a great leap to asterisk".

Or you may be reconsider this setup and if the thing you want is to use some 
channels for
voice and others for data over de PRI, take a look in the hdlc setup for the 
TE410E, that
works link any NIC in /dev/hdlcX.


"A. Valentin" <[EMAIL PROTECTED]> wrote:
>Hi!
>
>We planing to by some PRI/BRI equipment to replace our existing telephone 
>system.
>So I am going to try this:
>ISDN Card Outgoing Digital Call /
>   Capability: Unrestricted digital information
> -> octobri -> asterisk -> TE410E
> -> Internet Provider / Receiver for Capability Unrestricted digital
>information
>
>The question is: Is asterisk possible of transmit this digital call to the
>destination and what is needed to achieve this?
>I  found out that a variable called CALLTYPE exists, but I could not find
>out, if the Dial applications set the capability for the outgoing call correct?
>
>Thanks in advance for our help,
>
> André


=
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Version: OpenKeyServer v1.2
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[Asterisk-Users] Re: bellster.net - GREATadvance

[Miguel Ruiz Velasco Sobrino]

>Shoval Tomer wrote:
>> As far as I know it's not legal to join bellster in Israel.
>> 
>> It means that you're reselling the minutes you buy from the telco
>> company.
>
>Wouldn't you need to be selling them to be reselling?
>
>Does that make DISA illegal, and VoIP connections between offices if you 
>dial out the other end?

Well, a thing like bellster is illegal in any country with a telecomm law (to 
be read:
all countries).
If you do it in a low volume fashion, it's very difficult for the telco or the 
regulators
to catch up, also you can argue that you are not doing any economic profit of 
that. Let's
say, even if bellster uses +240 min/day of your phone, that hardly is 
noticeable for
them.

To the private intra-company VoIP, you can make a VoIP call from one office to 
another
and then hop to the PSTN without any problem, it's company only traffic. I do 
that, and
the telco or the regulators will never come up. Also, if they come they will 
have an
incredibly hard time proving you that.

BUT, if you start making economic profit (selling) a toll bypassing service 
without a
gov't permit, the regulators will come to nock your door in a very short time, 
and you
WILL have a very big fat problem. As example, in Mexico, last month the 
goverment closed
and seized the equipment of (the major -almost a monopoly- telco complained 
about having
competition, and the gov't quickly served them as allways) 13 of 14 established 
VoIP biz
for not having the legal permits in place (the only left already had a permit).



=
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Version: OpenKeyServer v1.2
Comment: Extracted from belgium.keyserver.net
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[Asterisk-Users] ANNOUNCEMENT : NEW CallingCard Application forAsterisk

[Miguel Ruiz Velasco Sobrino]

>> It is better to stay with Postgres. If you don't want to loose your
>> business stay away from MySQL.
>> If you are from Toronto ( I suppose you are ), you can check my posts to
>> TLUG (Toronto Linux User Group)
>> regarding MySQL and Postgres. I would say Postgres is a Open Source
>> Oracle. It's very stable, very scalable
>> and it's perfectly works under serious workload. MySQL is dying at the
>> same configuration.
>> I have client of mine who having issue with MySQL. Under some workload (
>> 10 users inserting at the same time )
>> it corrupts the index. Even MySQL 4.0.X is still corrupts the indexes
>> under heavy load.
>> I never saw it with Postgres. At the same time Postgres provides you a
>> very flexible SQL language and features,
>> as well as you can make stored procedures on Perl and many-many more.

> RIIIGHT it sounds like someone doesn't know what they are doing.  I have
> NEVER EVER had anything bad happen to mysql under heavy load.  

I think you don't know what is HEAVY load!! or had put a mysql under that.
And also, mysql lacks lots of very useful features (in v5, the list is a bit 
shorter, but
not a big difference anyway), like triggers, good procedural lang support, 
custom
types...  almost all of the missing features are the "enterprise" ones.
A experienced DBA with commercial DBMS's will laugh on MySQL list of features 
when
compared with almost any other *decent* DBMS.

Also, please do not turn this thread into a flame war or a holy DB war, like 
the one of
linux vs windoze in november.


=
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Version: OpenKeyServer v1.2
Comment: Extracted from belgium.keyserver.net
Signature: 0x59831109



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[Asterisk-Users] Re: Asterisk bandwidth tuning?

[Miguel Ruiz Velasco Sobrino]

Well, I don't know how to tune it more, it connects at about that rate in a 
mediocre
rural landline.

ILBC uses samples of 30ms, so if you set the trunkfreq set to 20 you will be 
using more
of the necesary scarce bandwidth AND dropping sample info in each frame, thus 
making
audio choppy and unclear.

Make shure to disallow all codecs and then allow only ILBC or lpc10 (search for 
it in
iax.conf), use the one that gives better results in your case.

lpc10 uses 20ms samples, so you can not allow at the same time both!!.

if you cannot connect at least at about 20kbps, your audio will suffer, and 
from mi
experience, at 16kbps the audio becomes so distored that is very difficult to 
understand
the other party.



--- [EMAIL PROTECTED] wrote:
Message: 1
Date: Wed, 19 Jan 2005 15:05:53 -0200
From: "alexandre::aldeia digital" <[EMAIL PROTECTED]>
Subject: Re: [Asterisk-Users] Re: Asterisk bandwidth tuning?
To: Asterisk Users Mailing List - Non-Commercial Discussion

Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed

18 - 22 Kbps  my dream!

I have asterisk -> INTERNET -> asterisk connection with IAX2 and I try 
iLBC, gsm, g729 and speex and the minimun bandwidth was 38 Kbps for 1 
channel.

What the parameters do you set to have this rate ???

Thank you.


Miguel Ruiz Velasco Sobrino wrote:
> I have an installation that connects in a [very] good day at 22kbps, but the 
> normal is
> about 18kbps.
> I use de ILBC codec, and also change in iax.conf the
> trunkfreq = 20
> to 
> trunkfreq = 30
> 
> It works, you can understand well the other person, but don't expect miracles 
> or an
> outstanding sound quality.
> 
> 
> 
>>Dear Dan;
>>
>>Thanks alot for your kindly reply.
>>
>>Well, what u advise us to use if the bandwidth is about 22kbps (dial up
>>connection in very old countries)?
>>
>>>GSM Codec is 13k plus overhead. That may work?
> 
> 
>>No way. GSM is 13.2kbps, and with Asterisk's hardcoded 20ms 
>>packetization, this gives 29200bps with RTP-based protocols or 26000bps 
>>with IAX, and as long as Asterisk doesn't support silence suppression, 
>>this needs to be full duplex, and I doubt you don't get that from a 
>>modem. If we could add silence suppression, we could do with half 
>>duplex, effectively saving half the bandwidth. in addition, if we could 
>>increase the packetization to something like 50 or 100ms for extreme 
>>use (like dial-up), we could end up with a lot less. Say, using 100ms 
>>slicing with IAX and speex at 6.3kbps, we'll end up with effective 
>>bandwidth use of 8860bps plus ethernet. now that's something you can 
>>brag about
> 
> 
>>roy
> 
> 
> Miguel Ruiz Velasco
> 

=
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Version: OpenKeyServer v1.2
Comment: Extracted from belgium.keyserver.net
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[Asterisk-Users] Re: Asterisk bandwidth tuning?

[Miguel Ruiz Velasco Sobrino]

I have an installation that connects in a [very] good day at 22kbps, but the 
normal is
about 18kbps.
I use de ILBC codec, and also change in iax.conf the
trunkfreq = 20
to 
trunkfreq = 30

It works, you can understand well the other person, but don't expect miracles 
or an
outstanding sound quality.


> Dear Dan;
>
> Thanks alot for your kindly reply.
>
> Well, what u advise us to use if the bandwidth is about 22kbps (dial up
> connection in very old countries)?
>>
>> GSM Codec is 13k plus overhead. That may work?

>No way. GSM is 13.2kbps, and with Asterisk's hardcoded 20ms 
>packetization, this gives 29200bps with RTP-based protocols or 26000bps 
>with IAX, and as long as Asterisk doesn't support silence suppression, 
>this needs to be full duplex, and I doubt you don't get that from a 
>modem. If we could add silence suppression, we could do with half 
>duplex, effectively saving half the bandwidth. in addition, if we could 
>increase the packetization to something like 50 or 100ms for extreme 
>use (like dial-up), we could end up with a lot less. Say, using 100ms 
>slicing with IAX and speex at 6.3kbps, we'll end up with effective 
>bandwidth use of 8860bps plus ethernet. now that's something you can 
>brag about

>roy

Miguel Ruiz Velasco



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[Asterisk-Users] RE: R2/MFC Mexico FREE calls to test chan_unicall

[Miguel Ruiz Velasco Sobrino]

if you dial this to reach the airport (using international long distance): 
9-011-52-5-571-3600

in extensions.conf

exten => _90115255.,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:8},90,Tt)
or
exten => _901152.,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:8},90,Tt)

you should configure the extension in this way (adding the 55 in the exten), 
AND (this
the important part!) the ${EXTEN:8}, because in mexico city the numbers are 8 
digits
long, not 10, if you leave the 10 digit numbers you will call all sort of wrong 
places.


Message: 8
Date: Thu, 13 Jan 2005 11:05:30 -0600
From: "Greg Blakely" <[EMAIL PROTECTED]>
Subject: RE: [Asterisk-Users] R2/MFC Mexico FREE calls to test  chan_unicall
To: "Asterisk Users Mailing List - Non-Commercial Discussion"

Message-ID:
<[EMAIL PROTECTED]>
Content-Type: text/plain; charset="us-ascii"

Works for me, too.  But I found that the Benito Juarez International
airport was reachable by 9-011-52-5-571-3600.

 

To get this from my PBX-like setup, I have the following in
extensions.conf:

 

exten => _901152.,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:6},90,Tt
 )

and the following in iax.conf

disallow=all

allow=GSM


=
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[Asterisk-Users] IAX2 keep alive?

[Miguel Ruiz Velasco Sobrino]

In a setup I've made i have a problem in the two way origination of the call.

Asterisk 1 <==> Public internet <==> NAT <==> Asterisk 2

note the two way pointing arrows.
When you call from the *2 to the *1, the call passes, but if you try to call 
from the *1
to the *2, it will not connect, unless you call from *1 to *2, and inmediately 
after
hang-up, you call in the other way.
I'm pretty sure it's a NAT loosing state too fast, and i can do nothing to fix 
the NAT.
Is there a way to have a keep-alive between the two * boxes?
I read about a config in iax.conf called  qualify=yes  to do that? does the 
qualify
parameter activates a sort of keep-alive?

Miguel Ruiz Velasco



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[Asterisk-Users] Re: Best gateway to use for *?

[Miguel Ruiz Velasco Sobrino]

Also, i've used it for a couple of months and i have had a great experience 
since the
begining; I think if you have a too cheap MoBo, your chances of getting that 
work, will
be lower.

>
>I have to date NOT had a problem with the Digium HW. You just got to pick the 
>right
Mobo. 
>
>-Original Message-
>From: [EMAIL PROTECTED] <[EMAIL PROTECTED]>
>To: [EMAIL PROTECTED] <[EMAIL PROTECTED]>; Asterisk Users Mailing List - 
>Non-Commercial Discussion 
>Sent: Sat Jan 08 14:57:53 2005
>Subject: RE: [Asterisk-Users] Best gateway to use for *?
>
>Sipura spa-3000, about $100 each, so 4 will take care of your 4 fxo need
>and you will also have 4 fxs ports, there is one fxo and one fxo per
>unit. Digium cards would be the ideal solutions, but there does seem to
>be some issues that Digium is not resolving quickly. 
>
>> -Original Message-
>> From: [EMAIL PROTECTED] 
>> [mailto:[EMAIL PROTECTED] On Behalf Of 
>> Joel Duffield
>> Sent: Saturday, January 08, 2005 10:08 AM
>> To: asterisk-users@lists.digium.com
>> Subject: [Asterisk-Users] Best gateway to use for *?
>> 
>> Hi All
>> 
>> I am working on setting up a * system to replace our current 
>> voicemail box. I may also end up using it for a few Voip 
>> calls. Anyway, I have heard some people complaining about the 
>> new Digium Fxo cards and having problems with them. I do not 
>> yet have the computer so if certain issues are caused by 
>> other hardware I could work around them. Does anyone trust 
>> these cards enough to use them? I hant to make sure this 
>> setup is clean so I can convince my boss to move to asterisk 
>> for the whole system. If these cards are no good, what other 
>> hardware is out there in the same kind of price range 
>> ($300-400) that can handle 4 Fxo's.
>> 
>> Thanks
>> Joel Duffield



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[Asterisk-Users] Re: RFC3389 support incomplete

[Miguel Ruiz Velasco Sobrino]

I'm not sure but I think it's not about DMTF, but about the silence supression 
or VAD
(voice activity detection) that * doesn't support. Try unabling it in the 
client.


> We try a X-Lite client from remote to connect to my *
>
> I can call X-Lite and X-Lite can call me. However, X-Lite can hear my 
> voice, while I cannot hear him.
> *CLI> shows
> *CLI> (date) NOTICE[4637] rtp.c:317 process_rfc3389: RFC3389 support 
> incomplete.  Turn off on client if possible
> RFC3389: 5 bytes, level 4 ...
>
> I tried in my sip.conf to change dtmfmode from inband to info and to 
> rfc2833  without success.
>
> Can anybody give me a hint?
>
> bye
>
> Ronald

Miguel Ruiz Velasco



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[Asterisk-Users] Re: IAX hardware

[Miguel Ruiz Velasco Sobrino]

> Hi,
> 
> I've been loosing my mind with NAT and read that IAX doesn't have problems 
> about nat.
>
> Does anyone knows about hadware (routers and etc) support IAX?
>
> Best regards
> helder

Well, in fact, IAX doesn't needs an ALG (application level gateway) unlike SIP, 
IRC or
FTP.
It uses a one normal socket, so there is no more thinks to worry about, except 
maybe you
will want HW that honors QoS prioritization (or better DiffServ) to make things 
run
smoother.

Miguel Ruiz Velasco



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RE: [Asterisk-Users] Virtual Modems

[Miguel Ruiz Velasco Sobrino]

I may be missing something, but...
why do you want to relay a modem over an VoIP network? isn't it no-sense to 
warp digital
data inside analog signals to be warped over digital data?.
Isn't it easier (and MUCH MORE efficent in terms of bandwidth) to transmit the 
digital
data directly from one end to other? maybe with the use of some security layer 
or
protocol, so you can get rid yourself of the great overhead that those implies?


> I think for now you are stuck with modem pass through. Our company does
> a lot of Fax over IP applications and T.38 is used exclusively for
> faxing.
> 
> There has been some work on modem relay but it is still in the white
> paper stage at this point. There is an ITU protocol V.150 designed for
> modem relay over IP but I have yet to see any devices out there that
> support it.
>
> Here's a good albeit dated article on the topic:
>
> http://www.commsdesign.com/design_corner/OEG20030312S0017
>
>
>
> -Original Message-
> From: Nathan Goodwin [mailto:[EMAIL PROTECTED] 
> Sent: Tuesday, December 14, 2004 10:57 AM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] Virtual Modems
>
> After searching the archives, I came acrross a few people mentioning 
> this, but I never saw anything about what became of it.
>
> Has anyone tried to make a virtual modem that could be directly handled 
> by astrisk, I saw a while ago that someone was going to try and make one
>
> using the same DSP libaries that the WinModems use, but then nothing.
>
> Would do this even be possible, and if so, what kind of connection 
> speeds could one hope to achive, with the compression and such?
> 
> I don't need a 56k connect (though it would be nice), but be able todo 
> 28.8 would be fine. :)
> 
> Also, would using the new T.38 (I think it is), that was designed for 
> faxes help modem calls any?

=
Miguel Ruiz Velasco

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Re: [Asterisk-Users] Budgetone 101 phones ? SIP through NAT ?

[Miguel Ruiz Velasco Sobrino]

>I'm new to VOIP.  We are thinking of setting up a VOIP system between a
>couple remote offices.  I've been lurking on this group for a while. 
>
>What is the consensus on these phones:
>
>http://www.netvoice.ca/grandstream/budgetone101.htm
>
>
>I'm confused about the SIP protocol... can a SIP phone be located behind
>a NATing firewall ?
>When people use asterisk on a broadband connection used for data and
>VOIP, do they put the asterisk machine behind a firewall or do they put
>the firewall on the asterisk machine ?

You will have a rather big problem doing that, you will likely end in the 
one-way audio
scenario; to overcome that you may use STUN or TURN or an application level 
gateway
(example: *). STUN is rather complicated to configure and I advise that only if 
you
REALLY need it; never used TURN, but is almost the same. Now, you can put a 
multihomed
machine (with an IP and posibly a NIC in each side of de fw), to do the passing 
of the
calls to an external provider.

But if you will link different offices each one with an * server, use IAX2 in 
the middle,
it doesn't suffer the on-way-audio-problem, it's NAT friendly (if you have one 
or many in
the middle), and if you enable trunking, saves a good bunch of bandwidth for 
each
additional conversation.

>Is anyone using QOS throttling when sharing VOIP and data on the same
>broadband connection ?  Is it necessary ?  

I use QOS to prevent de data traffic eating all the available bandwidth and 
render VoIP
unusable. You can use a HTB with SFQ in each bucket, or a pfifo, or if you are 
brave and
know what are about to do, use diffserv.

>Thanks.
Miguel Ruiz Velasco

=
Miguel Ruiz Velasco

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Re: RE: [Asterisk-Users] Unable to open master device

[Miguel Ruiz Velasco Sobrino]

>rom: "Jose Hernandez" <[EMAIL PROTECTED]>
>Subject: RE: [Asterisk-Users] Unable to open master device
>   '/dev/zap/ctl'
>
>If you are using udev you will get a message during compile suggesting to
>read README.udev, I did not pay attention to this message.
>
>Anyways.. I did not get the error after creating a udev rules file in
>/etc/udev/rules.d with the rules in /usr/src/zaptel/README.udev
>
>/etc/udev/rules.d/zaptel.rules
># section for zaptel device
>KERNEL="zapctl", NAME="zap/ctl"
>KERNEL="zaptimer",   NAME="zap/timer"
>KERNEL="zapchannel", NAME="zap/channel"
>KERNEL="zappseudo",  NAME="zap/pseudo"
>KERNEL="zap[0-9]*",  NAME="zap/%n"
>

>Now I get this error, I will troubleshoot it later tonight.
>[EMAIL PROTECTED] zaptel]# ztcfg -vv
>Zaptel Configuration
>==
>Channel map:
>Channel 01: FXS Kewlstart (Default) (Slaves: 01)
>Channel 02: FXO Kewlstart (Default) (Slaves: 02)
>2 channels configured.
>ZT_CHANCONFIG failed on channel 2: No such device or address (6)
>

If you watch de output of dmesg or /var/log/messages, you will find that a 
module of the
digium card failed to inicialize. Fix that, maybe you are trying to use a 
TDM400 with de
wcfxo module or the other way arround. 
For de TDM400 use allways (and only) the wcfxs and for the TDM100 the wcfxo; if 
you don't
have the other card don't load the module because you may end with an errors 
one of them
like this one:
unable to use trunking on channel IAX2 because there is no timing information 
from HW
(the error is not a transcription only a hint), also the error could be in the 
voicemail
or the music-on-hold.
Miguel



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Re: Re: [Asterisk-Users] How to encript SIP comunications?

[Miguel Ruiz Velasco Sobrino]

Hello Fach,
I have used openvpn for a while and in the new release thereis a feature called 
"server
mode" that makes posible to have a full network of vpn links besides a single 
TUN/TAP
adaptor (a pure software NIC) in the server. I haven't used that feature, but I 
think
this is what you need. Also openvpn runs on linux, *bsd, solaris, windows, and 
maybe in
other OS. 

Miguel

>
>Hello Gregory
>
>Thanks for your tip, but this looks like a point to point encription,
>but how about between extensions registered in a Asterisk server.
>
>Let's say I got a building 200 users registered and a given set of
>extensions, any of the users can be out of town or in another building
>in another city but for the matter of their job their communications
>have to be encripted. I can do your suggestion, but is group of users
>move from place to place then how would I do?
>
>I would appreciate to have a clear solutions for a more flexible
>scenario of encription
>
>All suggestions are highly appreciated
>
>Bye
>
>Fach

=
Miguel Ruiz Velasco

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