[asterisk-users] Asterisk - e164 (enum) lookup confused

[Joseph]

I would like to implement enum lookup in my dial plan but searching for
solution / implementation I'm getting confused what is current
standard. 

On some pages I read that the ENUMLOOKUP is not in development anymore
and suggesting on using Enumlookup.agi scrip , some are saying that
Asterisk 1.2.0 comes with a new powerful ENUMLOOKUP. So there is
probably no need to use this script anymore; so I'm confused as to what
should I use. 

Where can I find good Howto (with good explanation)?

-- 
#Joseph
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RE: [asterisk-users] An ISDN ISPBX to Voip Gateway??

[Ahsan Masood]

We are using these gateways with asterisk. These are easy to configure using 
WEB or Telnet and worked well for us.

http://www.voiptalk.org/products/Parlay+VoXIP+104+ISDN+BRI+Gateway


-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Hector Rivas 
Gandara
Sent: 22 February 2007 11:28
To: Asterisk Users Mailing List
Subject: [asterisk-users] An ISDN ISPBX to Voip Gateway??

Hello,


I have 2 ISDN BRI connections, configured by my telephony provider as a ISPBX
calling group. This allows me to have 4 concurrent calls.

I want to use this connection with my VoIP network, with an Asterisk PBX, so I
need a ISPBX to Voip (SIP) gateway. The problem is that I can't find any valid
solution.
The most of the gateways and cards use S0 simple BRI, and can't work with ISPBX.
This is the case of the FritzBox.

Does anybody known any gateway/card that I can use with this configuration and
with asterisk?

I rather prefer gateways than cards, but a card is ok if there is not any better
solution.

Thank you!

-- 
Atentamente, LambdaStream
Héctor Rivas www.lambdastream.com
 +34 981 17 33 44

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Re: [asterisk-users] Help Voicemail to SMS using asterisk

[Yuan LIU]

From: Goke Aruna <[EMAIL PROTECTED]>
Date: Sat, 03 Mar 2007 06:30:31 +0100

Dean Collins wrote:
> Now if only Tellme would agree to a pre-paid sip gateway for the
> asterisk community you could have pretty much everything you wanted :)
> http://www.voip-info.org/wiki/view/tellme
>
> Regards,
>
> Dean Collins
> Cognation Pty Ltd
> [EMAIL PROTECTED]
> +1-212-203-4357 Ph
> +1-917-207-3420 Mb
> +61-2-9016-5642 (Sydney in-dial).
>
>
>
>> -Original Message-
>> From: [EMAIL PROTECTED] [mailto:asterisk-users-
>> [EMAIL PROTECTED] On Behalf Of Steve Totaro
>> Sent: Friday, 2 March 2007 1:02 PM
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> Subject: Re: [asterisk-users] Help Voicemail to SMS using asterisk
>>
>> Goke Aruna wrote:
>>
>>> Hello all,
>>>
>>> I will be glad, if someone can throw light on Voicemail to SMS using
>>> asterisk.
>>>
>>> 1. I want my users to dial certain number.
>>>
>>> 2.  Record a voicemail with destination number.
>>>
>>> 3.   Convert this Voicemail to Text.
>>>
>>> 4.   Send the text with sms apps.
>>>
>>> and I wish i connect my asterisk to smsc directly. Is it possible
>>> without kannel?
>>>
>>> I will be glad, if someone could explain how i can get this done in
>>> asterisk.
>>>
>>> Goksie
>>>
>>>
>> That is a tall order (well the speech to text part anyways).  Maybe
>>
> you
>
>> could (mis)use a TTY service for the deaf.  Other than that, voice
>> recognition is not very good (unless I am looking in the wrong places,
>> something new has come out, or you want to use some outrageously
>>
> priced
>
>> solution).  I think the best you can do with what is out there in the
>> "free" arena is a very limited vocab like, numbers, letters, yes and
>> no.  Text to speech is getting to the decent point.
>>
>> Thanks,
>> Steve
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>
thanks Dean,

I am looking for open source material.

goksie


Is there something called Sphinex? (Read from the Oreilley book.)

Yuan Liu


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Re: [asterisk-users] Help Voicemail to SMS using asterisk

[Goke Aruna]

Dean Collins wrote:
> Now if only Tellme would agree to a pre-paid sip gateway for the
> asterisk community you could have pretty much everything you wanted :)
> http://www.voip-info.org/wiki/view/tellme
>
>
>  
>
> Regards,
>
> Dean Collins
> Cognation Pty Ltd
> [EMAIL PROTECTED]
> +1-212-203-4357 Ph
> +1-917-207-3420 Mb
> +61-2-9016-5642 (Sydney in-dial).
>
>
>   
>> -Original Message-
>> From: [EMAIL PROTECTED] [mailto:asterisk-users-
>> [EMAIL PROTECTED] On Behalf Of Steve Totaro
>> Sent: Friday, 2 March 2007 1:02 PM
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> Subject: Re: [asterisk-users] Help Voicemail to SMS using asterisk
>>
>> Goke Aruna wrote:
>> 
>>> Hello all,
>>>
>>> I will be glad, if someone can throw light on Voicemail to SMS using
>>> asterisk.
>>>
>>> 1. I want my users to dial certain number.
>>>
>>> 2.  Record a voicemail with destination number.
>>>
>>> 3.   Convert this Voicemail to Text.
>>>
>>> 4.   Send the text with sms apps.
>>>
>>> and I wish i connect my asterisk to smsc directly. Is it possible
>>> without kannel?
>>>
>>> I will be glad, if someone could explain how i can get this done in
>>> asterisk.
>>>
>>> Goksie
>>>
>>>   
>> That is a tall order (well the speech to text part anyways).  Maybe
>> 
> you
>   
>> could (mis)use a TTY service for the deaf.  Other than that, voice
>> recognition is not very good (unless I am looking in the wrong places,
>> something new has come out, or you want to use some outrageously
>> 
> priced
>   
>> solution).  I think the best you can do with what is out there in the
>> "free" arena is a very limited vocab like, numbers, letters, yes and
>> no.  Text to speech is getting to the decent point.
>>
>> Thanks,
>> Steve
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>>http://lists.digium.com/mailman/listinfo/asterisk-users
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>   
thanks Dean,

I am looking for open source material.

goksie

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[asterisk-users] How to fail an AGI

[Yuan LIU]

I mean how do I set failure condition in AGI?  My script exits with code 0 
upon success, and non-zero when problems occur - the standard *nix way.  But 
Asterisk always report "AGI Script completed, returning 0", and AGISTATUS is 
always SUCCESS.


Yuan Liu


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Re: [asterisk-users] Re: Sending SMS

[Al Bochter]

I don't see why the cost to send SMS is around .15 each. What does the 
gateway know that I don't know about sending the SMS.
I just think .15 for each SMS send is high.  Or am I just over looking 
something?


Best regards,

Al Bochter
Bochter Services
http://www.BochterServices.com/?t=Email



Tomislav Parcina wrote:


Supa wrote:


Try this:
http://www.bayhamsystems.com/asterisk.html

Works for me just fine, and it is very easy to get up and running, 
even with older version 1.2.3



I don't see a point of using providers as Bayhamsystems. First, it's 
unpractical to send SMS from phone. If I'm going to use web interface, 
then is better to use some provider that has web interface just for 
that (or maybe they will provide application to send messages to 
groups or in certain time).


Only reason why I would like to do it true Asterisk is if I could use 
my VoIP or E1 provider so that I get only one bill. But using 
Bayhamsystems that isn't a case. So, why people use such providers?




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Re: [asterisk-users] Need comparison between PBXtra, Trixbox, Thirdlane, Druid, Aheeva etc.

[Mailing Lists]

On Mar 2, 2007, at 9:14 PM, Zeeshan Zakaria wrote:


Hi,

For a customer, I am looking for a good and reliable Asterisk based  
system. Five servers will be installed at different locations and  
will be linked together with each other. This system will work as a  
call center as well. It has to be a stable and reliable. Customer  
also needs GUIs for system administration and agents call activities.


He also wants video conferencing

Please help me select a good system.

Thanks
--
Zeeshan A Zakaria
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Running Asterisk with FreePBX on CentOS works great.  I started with  
Trixbox and used it for a few weeks before I simply downloaded and  
compiled my own.  Honestly, it is not that hard.  Just follow the  
instructions.  I created a series of scripts to run against a fresh  
CentOS install which deal with compiling and installing everything,  
including the FreePBX dependencies.  I'd be happy to share it.  I  
don't do any video conferencing, and I don't patch Asterisk for faxing.


FreePBX works quite well.  Combined with all the features of modern  
SIP phones, there is nothing you can't do.


I run my systems using the Intel 975XBX2 motherboard (975 chipset),  
which I assembled by buying components from New Egg.  Very stable -  
no issues with CentOS.


-Joe
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[asterisk-users] Need comparison between PBXtra, Trixbox, Thirdlane, Druid, Aheeva etc.

[Zeeshan Zakaria]

Hi,

For a customer, I am looking for a good and reliable Asterisk based system.
Five servers will be installed at different locations and will be linked
together with each other. This system will work as a call center as well. It
has to be a stable and reliable. Customer also needs GUIs for system
administration and agents call activities.

He also wants video conferencing

Please help me select a good system.

Thanks
--
Zeeshan A Zakaria
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Re: [asterisk-users] IP addresses

[Steve Totaro]

Mike Hammett wrote:


I have multiple IP addresses on my box. My provider just changed my 
eth0 IP off to another interface (lo:9) and a new IP on eth0. Nothing 
works anymore because calls to the old IP address are being answered 
by the new IP address. How do I straighten this out?


  
Your SIP configuration should allow you to bind to a particular IP. 


Thanks,
Steve

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Re: [asterisk-users] svn 1.4 - mp3 support and changing the installation directory

[Kevin P. Fleming]

tzieleniewski wrote:
> 1. In many docs on the web there is an info to make asterisk by invoking 
> #make mpg123 in order to have the mp3 support, when I try to do it this way 
> make informs that there are no such make rules.

Those documents are referring to versions prior to 1.4. Asterisk 1.4 no
longer provides any assistance for building MP3 support to be used as
music-on-hold, as we have more efficient and reliable ways to use native
format files now.

> 2. Also in docs on the web there an info about Debian zaptel installation 
> that the easyiest way is to install zaptel-modules as the package. but at the 
> moment the there is only 1.2 verion available. So I need to install it from 
> sources. What do I need to do to load the zaptel modules.
> Will it be enough just to make the following?:
> # cd /.../zaptel_source
> # make
> # make install
> # modprobe zaptel

Yes.

> 3. I would like to have my all asterisk stuff (asterisk,addons,core-sounds) 
> installed in the asterisk system user home directory.
> Is it sufficient to set the INSTALL_PREFIX= in the make file to the desired 
> value??

Not really, no. That will be a good start, but there are configuration
files that will need hand-editing to ensure that all parts of Asterisk
know that the installation has been done to that location. If I remember
correctly, there are recipes for doing so on the voip-info wiki.

> 4. is it possible to get a packet of core-sounds directly from the svn?

No, they are not in Subversion, they are on the FTP server. You are
welcome to download them from there.
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Re: [asterisk-users] FAX using T38

[Kevin P. Fleming]

Andrew Joakimsen wrote:
> That's like saying a pinto is fast when you upgrade the engine. Well
> Asterisk also supports T.38 for free... if you backport OpenPBX.org
> fixes.
> 
> But realisticly ASTERISK DOES NOT HAVE FAX SUPPORT STOP CLUTTERING MY
> INBOX WITH DISCUSSION OF FEATURE THAT DOES NOT EXSIST.

It seems to me that a simple solution for you, since you clearly prefer
to use OpenPBX instead of Asterisk, is to just unsubscribe from this
mailing list. Nobody is directly sending these messages about FAX
support to your inbox; you subscribed to the mailing list, so you get to
see the messages.
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Re: [asterisk-users] Asterisk 1.4.1

[Kevin P. Fleming]

Paul wrote:
> Forrest Beck wrote:
> 
>> Any idea when 1.4.1 will be available.  There is a bug fix in the cvs
>> head that I need, and I don't want to run the cvs build on a
>> production machine.
>>
>> Thanks...
>>
> I wouldn't be building anything at all on a production machine without
> doing some testing on another machine first.

And on top of that, Asterisk 1.4.1 is only a snapshot of the SVN 1.4
branch at the time we choose to release it anyway. There is no 'quality'
difference between building from the Asterisk 1.4.1 tarball or an SVN
checkout taken fifteen minutes before the tarball was produced.
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[asterisk-users] Asterisk 1.4.1 Released

[Asterisk Development Team]

The Asterisk and Zaptel development teams have released Asterisk 1.4.1.

This release contains a very large number of bug fixes, including a fix
for the recently discovered security vulnerability.

It also contains a complete rewrite of the Shared Line Appearance (SLA)
support that was first released as part of Asterisk 1.4.0. The new
version of this functionality has been tested against a variety of
phones and provides much more flexibility and configurability (along
with actually working properly in most scenarios, which the original
implementation failed to do). Users who are interested in SLA
functionality should update to this version and try it out; we welcome
bug reports and test reports.

Because of the security vulnerability fix present in this version, all
users of Asterisk 1.4 are urged to update as soon as they can schedule it.

Thanks for your support of Asterisk and Zaptel!
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[asterisk-users] Asterisk 1.2.16 Released

[Asterisk Development Team]

The Asterisk and Zaptel development teams have released Asterisk 1.2.16.

This release contains a number of bug fixes, including a fix for a
recently discovered security vulnerability. All Asterisk 1.2 users are
urged to update to this release as soon as possible.

Thanks for your support of Asterisk and Zaptel!
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[asterisk-users] Zaptel 1.2.15 Released

[Asterisk Development Team]

The Asterisk and Zaptel development teams have released Zaptel 1.2.15.

This release contains a significant Astribank (XPP) driver update,
support for Digium's TE120P card, and various bug fixes.

Thanks for your support of Asterisk and Zaptel!

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Re: [asterisk-users] 1.4 - SLA

[Russell Bryant]

Lacy Moore - Aspendora wrote:

Russell, I don't have any specifics at this time.  I need to dig a
little further.  I'm thinking the autocontext is what is giving me
fits.  I can receive calls and place calls, but the hint status is not
working.  It currently registers as a hint showing not in use.  It
does not show in use.


If you aren't seeing any lights change on the phones when calls are 
going on, check "sip show subscriptions" at the CLI.  If the phones have 
not properly subscribed to the right extensions, you won't see anything.



I ended up using some of the config from the bottom of the sla.txt
file.  The sample file may be missing the template section.  The
sample config does not match the config in the sla.txt.  I couldn't
get the sample config to work at all.  Again, hopefully over the
weekend I'll be able to get more information.


You are correct.  The sample configuration is missing the template.  I 
will add it now.  However, I just made the tarballs for 1.4.1, so this 
config fix didn't make it in.



Using the config in the sample file, the hint status was working.  I
could see the line ringing, but I could not answer the lines or place
calls.  Using the config from the sla.txt file, I could place calls
and receive calls, but the hints never showed any activity, just
always not in use.


As I noted earlier, check your "sip show subscriptions" to make sure the 
phones are subscribed to the right thing.


Another helpful thing that you can use for debugging is to look at the 
output of "sla show stations".  You can see the state of each line 
appearance on each station.  This should correspond with what you see on 
the phone  ... unless there is a problem, of course.



If possible, could you provide the config that you've used for
testing?  I'm testing using Polycom phones to try to keep things
simple.  I'm assuming you are using a Polycom.


I have been testing with a variety of different phones.  I have not 
tested all of the Polycom models, yet.  This is one of the things we're 
going to have to work through.  I would like to document issues with 
specific phones in sla.txt as we come across them.


The configuration I'm using for testing looks just like the stuff in 
configs/sla.conf.sample.  Essentially, it is:



[line1]
type=trunk
device=Zap/3
autocontext=line1

[line2]
type=trunk
device=Zap/4
autocontext=line2

[station](!)
type=station
autocontext=sla_stations
trunk=line1
trunk=line2

[station1] (station)
device=SIP/station1

[station2](station)
device=SIP/station2

[station3](station)
device=SIP/station3


Thanks for providing some feedback on this.  You are the first one to 
say anything about it.  :)  I am very eager to get everything working 
well so that everyone is happy.  Just please be patient as I work 
through the initial flood of reports since it is just now getting out in 
the field.


--
Russell Bryant
Software Engineer
Digium, Inc.
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Re: [asterisk-users] REMOTE CRASH FIX

[Mike Lynchfield]

nope

http://svn.digium.com/view/asterisk/branches/1.2/channels/chan_sip.c?r1=56230&r2=57475

is avail for free..

6574a6575,6579

if (uri == NULL) {
 ast_log(LOG_WARNING, "register_verify: URI is NULL!\n");
 transmit_response_with_date(p, "503 Bad Request", req);
 return -3;
}



is my patch

We just offering to support people that don't know how to mod stuff and
recompile..



On 3/2/07, BJ Weschke <[EMAIL PROTECTED]> wrote:


On 3/2/07, Mike Lynchfield <[EMAIL PROTECTED]> wrote:
> Please note that we are available to fix the current REMOTE crash that
> affects Asterisk/openpbx/trixbox and crashes these systems via a
malformed
> packet
>
> please contacts use if you need a hand to patch your systems.
>
>

And you'll do it for free too? How gracious of you! If you were
charging money, I'd say you belong on the -biz list, but while you're
being so gracious, maybe your resources would like to volunteer for
some bug marshalling tasks too?

--
Bird's The Word Technologies, Inc.
http://www.btwtech.com/
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--
Mike
Sales Manager
http://www.voicemeup.com
Making it happen
1.877.807.VOIP (8647)
1.514.312.7030
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[asterisk-users] How to log VERBOSE statement to a file?

[Larry Alkoff]

I would like to log a verbose statement in my 900/976 extens to a 
special file called 'attacks'.


These are not standard messages like debug, notice, warning, error, 
vebose or dtmf that could be logged to /var/log/asterisk/messages.


Does the 'verbose' in VERBOSE commands have anything to do with the 
'verbose' in error messages?



I tried >> redirection of a VERBOSE statement - did not work.

Larry

--
Larry Alkoff N2LA - Austin TX
Using Thunderbird on Linux
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Re: [asterisk-users] Asterisk Java w/ Threads

[Stefan Reuter]

Doug Garstang wrote:
> Can the Asterisk Java API be written with threads?
sure.

> Ie, I need to connect
> to multiple Asterisk systems from the one java application. I tried to
> make my  class which implements ManagerEventListener, also implement
> Runnable, but got errors because the Runnable interface doesn't throw
> InterruptedException.
It would certainly help if you provided some example of what you tried.

> Anywho...
A better place to ask questions regarding Asterisk-Java is the
asterisk-java-users list:
http://asterisk-java.org/development/mail-lists.html

=Stefan

-- 
reuter network consulting
Neusser Str. 110
50760 Koeln
Germany
Telefon: +49 221 1305699-0
Telefax: +49 221 1305699-90
E-Mail:  [EMAIL PROTECTED]
Jabber:  [EMAIL PROTECTED]

Steuernummern 215/5140/1791 USt-IdNr. DE220701760



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Re: [asterisk-users] DTMF from TDM400P and X100P

[Tzafrir Cohen]

On Fri, Mar 02, 2007 at 01:48:21PM -0800, Yuan LIU wrote:
> >From: Tzafrir Cohen <[EMAIL PROTECTED]>
> >Date: Fri, 2 Mar 2007 22:14:09 +0200
> >
> >On Fri, Mar 02, 2007 at 09:36:28AM -0800, Yuan LIU wrote:
> >> With one IVR payment system, I noticed quite a difference in DTMF
> >> transmission between these two cards.  The IVR missed nearly all digits
> >> from X100P, while receiving digits from TDM fine.
> >>
> >> Since neither card process or synthesize audio, what can the difference be?
> >>
> >> (This particular IVR has problem with some regular phone devices, too.)
> >
> >audio quality?
> 
> You mean from DAC?  That could make sense - altough the MODEM function 
> X100P is designed for would require it to be fairly accurate.  DTMF itself 
> is one of basic MODEM functions.

No. I mean: bad distortion of the analog audio that breaks the DTMF
detection (which is done in Asterisk).

> 
> >listen to the audio with e.g. ztmonitor.
> 
> Thanks for the input.  Have yet to put a sound card with TDM. (The IVR in 
> question is not my machine.)  I suspect that ztmonitor listens to the 
> digital output, though.

Record it to a sound file.

A simplistic but useful test is to play that very same sound file to a
phone and see if it "dials" the digits you have expected it to dial.

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
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Re: [asterisk-users] blieve i my TE110P or My teleco provider ??

[younss azzayani]

1) You think your configuration is correct but you are wrong and it
indeed is incorrect
evenif with my config i get a green led and with the telco config the
lid is red ?
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Re: [asterisk-users] REMOTE CRASH FIX

[Kevin P. Fleming]

Mike Lynchfield wrote:
> Please note that we are available to fix the current REMOTE crash that
> affects Asterisk/openpbx/trixbox and crashes these systems via a
> malformed packet
> 
> please contacts use if you need a hand to patch your systems.

This list is for non-commercial discussion, as is clearly stated in the
list name and description. If you want to advertise your services, use
the asterisk-biz list.
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Re: [asterisk-users] REMOTE CRASH FIX

[BJ Weschke]

On 3/2/07, Mike Lynchfield <[EMAIL PROTECTED]> wrote:

Please note that we are available to fix the current REMOTE crash that
affects Asterisk/openpbx/trixbox and crashes these systems via a malformed
packet

please contacts use if you need a hand to patch your systems.




And you'll do it for free too? How gracious of you! If you were
charging money, I'd say you belong on the -biz list, but while you're
being so gracious, maybe your resources would like to volunteer for
some bug marshalling tasks too?

--
Bird's The Word Technologies, Inc.
http://www.btwtech.com/
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[asterisk-users] Asterisk Java w/ Threads

[Doug Garstang]

Ok, so I ain't much of a Java programmer, but...

Can the Asterisk Java API be written with threads? Ie, I need to connect 
to multiple Asterisk systems from the one java application. I tried to 
make my  class which implements ManagerEventListener, also implement 
Runnable, but got errors because the Runnable interface doesn't throw 
InterruptedException.


Anywho...

Doug.

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Re: [asterisk-users] DTMF from TDM400P and X100P

[Yuan LIU]

From: Tzafrir Cohen <[EMAIL PROTECTED]>
Date: Fri, 2 Mar 2007 22:14:09 +0200

On Fri, Mar 02, 2007 at 09:36:28AM -0800, Yuan LIU wrote:
> With one IVR payment system, I noticed quite a difference in DTMF
> transmission between these two cards.  The IVR missed nearly all digits
> from X100P, while receiving digits from TDM fine.
>
> Since neither card process or synthesize audio, what can the difference 
be?


(At least the TDM400P actually has a hardware TDMF detector, but it is
not used, AFAIK)

>
> (This particular IVR has problem with some regular phone devices, too.)

audio quality?


You mean from DAC?  That could make sense - altough the MODEM function X100P 
is designed for would require it to be fairly accurate.  DTMF itself is one 
of basic MODEM functions.



listen to the audio with e.g. ztmonitor.


Thanks for the input.  Have yet to put a sound card with TDM. (The IVR in 
question is not my machine.)  I suspect that ztmonitor listens to the 
digital output, though.


Yuan Liu

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



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[asterisk-users] IP addresses

[Mike Hammett]

I have multiple IP addresses on my box.  My provider just changed my eth0 IP
off to another interface (lo:9) and a new IP on eth0.  Nothing works anymore
because calls to the old IP address are being answered by the new IP
address.  How do I straighten this out?

 

 

 

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[asterisk-users] svn 1.4 - mp3 support and changing the installation directory

[tzieleniewski]

Hi ALL!!
This is my first time with asterisk and my first post:)
Please be so kind and give me few clues for the good beginning:)

I am using version 1.4 from svn on the Debian etch OS kernel 2.6.

I have four question:

1. In many docs on the web there is an info to make asterisk by invoking #make 
mpg123 in order to have the mp3 support, when I try to do it this way make 
informs that there are no such make rules.

2. Also in docs on the web there an info about Debian zaptel installation that 
the easyiest way is to install zaptel-modules as the package. but at the moment 
the there is only 1.2 verion available. So I need to install it from sources. 
What do I need to do to load the zaptel modules.
Will it be enough just to make the following?:
# cd /.../zaptel_source
# make
# make install
# modprobe zaptel

3. I would like to have my all asterisk stuff (asterisk,addons,core-sounds) 
installed in the asterisk system user home directory.
Is it sufficient to set the INSTALL_PREFIX= in the make file to the desired 
value??

4. is it possible to get a packet of core-sounds directly from the svn?

Thanks in advance
tomasz
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[asterisk-users] Asterisk Java and Astmanproxy

[Doug Garstang]

Has anyone used talked to astmanproxy with the Asterisk Java Manager 
interface? First suspiscions are that it will not work.


Astmanproxy sends a connection banner of 'Asterisk Call Manager 
Proxy/1.21' which is not what Asterisk Java is expecting. Also, 
astmanproxy preprends the name of the host to the front of each line of 
output, and that will break Asterisk Java... I think... anyone done this?


Doug.

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Re: [asterisk-users] Help Voicemail to SMS using asterisk

[Goke Aruna]

Steve Totaro wrote:
> Goke Aruna wrote:
>> Hello all,
>>
>> I will be glad, if someone can throw light on Voicemail to SMS using
>> asterisk.
>>
>> 1. I want my users to dial certain number.
>>
>> 2.  Record a voicemail with destination number.
>>
>> 3.   Convert this Voicemail to Text.
>>
>> 4.   Send the text with sms apps.
>>
>> and I wish i connect my asterisk to smsc directly. Is it possible
>> without kannel?
>>
>> I will be glad, if someone could explain how i can get this done in
>> asterisk.
>>
>> Goksie
>>   
>
> That is a tall order (well the speech to text part anyways).  Maybe
> you could (mis)use a TTY service for the deaf.  Other than that, voice
> recognition is not very good (unless I am looking in the wrong places,
> something new has come out, or you want to use some outrageously
> priced solution).  I think the best you can do with what is out there
> in the "free" arena is a very limited vocab like, numbers, letters,
> yes and no.  Text to speech is getting to the decent point.
>
> Thanks,
> Steve
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>
Thanks steve,

could this be done with asterisk?

goksie
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Re: [asterisk-users] rtsavesysname not working in 1.4

[Bruce Reeves]

Try renaming you column in the peers table to regserver

On 3/2/07, David Thomas <[EMAIL PROTECTED]> wrote:


I am trying to have asterisk update the system name in my realtime
peers, but it does not seem to be working. Here is what I've done so
far.

- added systemname => mysystemname in asterisk.conf
- set rtsavesysname=yes in sip.conf.
- created a table called "sysname" in my peers table in mysql
- restarted asterisk
- rebooted my phone to force a re-register

Is there something I'm missing?

Thanks!
David
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--
Bruce
Nortex Networks
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[asterisk-users] REMOTE CRASH FIX

[Mike Lynchfield]

Please note that we are available to fix the current REMOTE crash that
affects Asterisk/openpbx/trixbox and crashes these systems via a malformed
packet

please contacts use if you need a hand to patch your systems.



--
Mike
Sales Manager
http://www.voicemeup.com
Making it happen
1.877.807.VOIP (8647)
1.514.312.7030
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Re: [asterisk-users] PRI progress codes.

[Eric \"ManxPower\" Wieling]

John Bittner wrote:

Anyone know how to let asterisk deal with the progress codes coming
from the carrier? The problem I am having is when a customer calls an
invalid number the carrier tells me the call is invalid via a progress
code but doesn't route me to a recording (this number is invalid).
Instead they hang up on me causing a fast busy or sometimes hold up
the call with dead air for 15 to 30 seconds then a fast busy. I am
working with the carrier to get this fixed but its not going easy.
Is there anyway when asterisk sees the progress code to cancel the
dial and playback a message mapped to the progress code type.


[macro-dial-result]
;
; Handle Disconnect Cause Codes
;
exten => s,1,Noop(HANGUPCAUSE is ${HANGUPCAUSE} and DIALSTATUS is 
${DIALSTATUS})

exten => s,2,Noop
exten => s,3,GotoIf($[${HANGUPCAUSE} = 0]?s,5)
exten => s,4,Goto(cause-${HANGUPCAUSE},1)
exten => s,5,GotoIf($[${DIALSTATUS} = NOANSWER]?cause-16,1)
exten => s,6,GotoIf($[${DIALSTATUS} = BUSY]?cause-17,1)
exten => s,7,GotoIf($[${DIALSTATUS} = CHANUNAVAIL]?cause-3,1)
exten => s,8,GotoIf($[${DIALSTATUS} = ANSWER]?cause-16,1)
exten => s,9,Goto(cause-0,1)

exten => cause-0,1,NoOp(UNKNOWN)
exten => cause-0,2,Goto(redial,1)

exten => cause-1,1,NoOp(AST_CAUSE_UNALLOCATED)
exten => cause-1,2,Macro(number-disconnected,${MACRO_EXTEN})

exten => cause-2,1,NoOp(AST_CAUSE_NO_ROUTE_TRANSIT_NET)
exten => cause-2,2,Goto(redial,1)

exten => cause-3,1,NoOp(AST_CAUSE_NO_ROUTE_DESTINATION)
exten => cause-3,2,Goto(redial,1)

exten => cause-6,1,NoOp(AST_CAUSE_CHANNEL_UNACCEPTABLE)
exten => cause-6,2,Goto(redial,1)

exten => cause-7,1,NoOp(AST_CAUSE_CALL_AWARDED_DELIVERED)
exten => cause-7,2,Goto(redial,1)

exten => cause-16,1,NoOp(AST_CAUSE_NORMAL_CLEARING)
exten => cause-16,2,Hangup

exten => cause-17,1,NoOp(AST_CAUSE_USER_BUSY)
exten => cause-17,2,Busy

exten => cause-18,1,NoOp(AST_CAUSE_NO_USER_RESPONSE)
exten => cause-18,2,Goto(redial,1)

exten => cause-19,1,NoOp(AST_CAUSE_NO_ANSWER)
exten => cause-19,2,Goto(redial,1)

exten => cause-21,1,NoOp(AST_CAUSE_CALL_REJECTED)
exten => cause-21,2,Goto(redial,1)

exten => cause-22,1,NoOp(AST_CAUSE_NUMBER_CHANGED)
exten => cause-22,2,Goto(redial,1)

exten => cause-27,1,NoOp(AST_CAUSE_DESTINATION_OUT_OF_ORDER)
exten => cause-27,2,Goto(redial,1)

exten => cause-28,1,NoOp(AST_CAUSE_INVALID_NUMBER_FORMAT)
exten => cause-28,2,Goto(redial,1)

exten => cause-29,1,NoOp(AST_CAUSE_FACILITY_REJECTED)
exten => cause-29,2,Goto(redial,1)

exten => cause-30,1,NoOp(AST_CAUSE_RESPONSE_TO_STATUS_ENQUIRY)
exten => cause-30,2,Goto(redial,1)

exten => cause-31,1,NoOp(AST_CAUSE_NORMAL_UNSPECIFIED)
exten => cause-31,2,Hangup

exten => cause-34,1,NoOp(AST_CAUSE_NORMAL_CIRCUIT_CONGESTION)
exten => cause-34,2,Goto(redial,1)

exten => cause-38,1,NoOp(AST_CAUSE_NETWORK_OUT_OF_ORDER)
exten => cause-38,2,Goto(redial,1)

exten => cause-41,1,NoOp(AST_CAUSE_NORMAL_TEMPORARY_FAILURE)
exten => cause-41,2,Goto(redial,1)

exten => cause-42,1,NoOp(AST_CAUSE_SWITCH_CONGESTION)
exten => cause-42,2,Goto(redial,1)

exten => cause-43,1,NoOp(AST_CAUSE_ACCESS_INFO_DISCARDED)
exten => cause-43,2,Goto(redial,1)

exten => cause-44,1,NoOp(AST_CAUSE_REQUESTED_CHAN_UNAVAIL)
exten => cause-44,2,Goto(redial,1)

exten => cause-45,1,NoOp(AST_CAUSE_PRE_EMPTED)
exten => cause-45,2,Goto(redial,1)

exten => cause-50,1,NoOp(AST_CAUSE_FACILITY_NOT_SUBSCRIBED)
exten => cause-50,2,Goto(error,1)

exten => cause-52,1,NoOp(AST_CAUSE_OUTGOING_CALL_BARRED)
exten => cause-52,2,Goto(redial,1)

exten => cause-54,1,NoOp(AST_CAUSE_INCOMING_CALL_BARRED)
exten => cause-54,2,Goto(redial,1)

exten => cause-57,1,NoOp(AST_CAUSE_BEARERCAPABILITY_NOTAUTH)
exten => cause-57,2,Goto(redial,1)

exten => cause-58,1,NoOp(AST_CAUSE_BEARERCAPABILITY_NOTAVAIL)
exten => cause-58,2,Goto(redial,1)

exten => cause-65,1,NoOp(AST_CAUSE_BEARERCAPABILITY_NOTIMPL)
exten => cause-65,2,Goto(redial,1)

exten => cause-66,1,NoOp(AST_CAUSE_CHAN_NOT_IMPLEMENTED)
exten => cause-66,2,Goto(redial,1)

exten => cause-69,1,NoOp(AST_CAUSE_FACILITY_NOT_IMPLEMENTED)
exten => cause-69,2,Goto(redial,1)

exten => cause-81,1,NoOp(AST_CAUSE_INVALID_CALL_REFERENCE)
exten => cause-81,2,Goto(redial,1)

exten => cause-88,1,NoOp(AST_CAUSE_INCOMPATIBLE_DESTINATION)
exten => cause-88,2,Goto(redial,1)

exten => cause-95,1,NoOp(AST_CAUSE_INVALID_MSG_UNSPECIFIED)
exten => cause-95,2,Goto(redial,1)

exten => cause-96,1,NoOp(AST_CAUSE_MANDATORY_IE_MISSING)
exten => cause-96,2,Goto(redial,1)

exten => cause-97,1,NoOp(AST_CAUSE_MESSAGE_TYPE_NONEXIST)
exten => cause-97,2,Goto(redial,1)

exten => cause-98,1,NoOp(AST_CAUSE_WRONG_MESSAGE)
exten => cause-98,2,Goto(redial,1)

exten => cause-99,1,NoOp(AST_CAUSE_IE_NONEXIST)
exten => cause-99,2,Goto(redial,1)

exten => cause-100,1,NoOp(AST_CAUSE_INVALID_IE_CONTENTS)
exten => cause-100,2,Goto(redial,1)

exten => cause-101,1,NoOp(AST_CAUSE_WRONG_CALL_STATE)
exten => cause-101,2,Goto(redial,1)

exten => cause-102,1,NoOp(AST_CAUSE_RECOVERY_ON_TIMER_EXPIRE)
exten => cause-102,2,Goto(redial,1)

exten => cau

Re: [asterisk-users] multiple phones registered for the same user

[Andrew Joakimsen]

Maybe.. if you dont expect to recieve calls to any device, then I just
wouldnt bother to register.

On 2/28/07, Ricardo Carvalho <[EMAIL PROTECTED]> wrote:

Can't I register multiple phones with the same user/password? That's
what I pretend to do, not ring groups...

Thanks,
Ricardo.




Azfhasterisk wrote:
> Create a different user for each phone and create a ring group with the
> phones that you want to ring.
>
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Ricardo
> Carvalho
> Sent: Wednesday, February 28, 2007 9:15 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] multiple phones registered for the same user
>
> Dear all,
>
> I've noticed that when I have a phone registered in Asterisk, and then I
> register another phone with the same user, the "sip show peers" in the
> CLI shows that Asterisk replaced the IP of the first phone by the IP of
> the last one registered for that user. Consequently, if someone calls
> that user, only the last phone rings!!
> How may I configure Asterisk to be able to fork all incoming calls to
> every phones registered for each user, so that every phone ring until
> someone answers the call in one of them?
>
> Thanks in advance,
> Ricardo.
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Re: [asterisk-users] FAX using T38

[Andrew Joakimsen]

That's like saying a pinto is fast when you upgrade the engine. Well
Asterisk also supports T.38 for free... if you backport OpenPBX.org
fixes.

But realisticly ASTERISK DOES NOT HAVE FAX SUPPORT STOP CLUTTERING MY
INBOX WITH DISCUSSION OF FEATURE THAT DOES NOT EXSIST.

On 3/1/07, Zoa <[EMAIL PROTECTED]> wrote:

So does asterisk (Albeit with a commercial package)

http://www.attractel.com/t38.html


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Re: [asterisk-users] blieve i my TE110P or My teleco provider ??

[Andrew Joakimsen]

If the remote end isnt up that could explain why the correct settings
dont work. If they told you to use those settings thats how it should
be provisioned, either:

1) You think your configuration is correct but you are wrong and it
indeed is incorrect

2) Your telco didn't correctly provision or otherwise activate the line.

I would go back to the "correct" settings and ring your telco's
helpdesk for further assistance.

On 3/1/07, younss azzayani <[EMAIL PROTECTED]> wrote:

hi eveybody,
after many test with your help and the irc channels help, i get the
led on TE110P green
with this config:
span=1,1,0,ccs,ami
=> alarms OK  Green Led

but the provider  say that i have to set my span to this
span=1,1,0,ccs,hdb3,crc4

=> alarms: YEL/RED

i can't make call's yet to test because they have to sync the
Modulator in the other side
so any remark?
is my card TE110P get crazy?
is the TELECO are crazy?
any idea
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Re: [asterisk-users] T38

[Andrew Joakimsen]

No there is no fax support in Asterisk.

On 3/2/07, Khaled <[EMAIL PROTECTED]> wrote:





Dears

Any one know how to let t38 works on asterisk 1.2 or an distribution like
trixbox have asterisk 1.4



Regards





Khaled Chehab

System Integration Engineer

Xplorium Offshore.

Sakiet Al Janzir

Postal Code: 1102-2080

Tel: (961) 1- 868 686

Fax :(961) 1-808 810

GSM: (961) 3-979 343



 
 *
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Re: [asterisk-users] Asterisk and Fax

[Andrew Joakimsen]

No. Asterisk does not have fax support and there are no plans to add it.

You can however send the faxes as voice calls, however there is no
assurance as to its reliability. Maybe most will work but some will
without a doubt be predestine to fail.

On 3/2/07, --[ UxBoD ]-- <[EMAIL PROTECTED]> wrote:

Hi,

I have a requirement for sending and receiving faxes and was wondering the best 
way to achieve it with Asterisk as I only have one phone line.

I currently have a TDM11B in my server (1 x FXO, 1 x FXS), so I was thinking 
that I would need to get a additional FXS module, connect that to a Eicon Fax 
card, and then when receiving a call detect the fax tone and bridge the call to 
the FXS channel.

Would that work okay ? Just seems fax cards are very expensive for what they do.
--
--[ UxBoD ]--
// PGP Key: "curl -s http://www.splatnix.net/uxbod.asc | gpg --import"
// Fingerprint: 543A E778 7F2D 98F1 3E50 9C1F F190 93E0 E8E8 0CF8
// Keyserver: www.keyserver.net Key-ID: 0xE8E80CF8
// SIP Phone: [EMAIL PROTECTED]

--
This message has been scanned for viruses and dangerous content by MailScanner, 
and is
believed to be clean.

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RE: [asterisk-users] Alec Saunders post about Mashable Telco's

[Dean Collins]

Then install it out in Iowa and get your money back at 2c a minute :)

http://gigaom.com/2007/02/26/iowa-telcos-att-owes-12-million 
http://saunderslog.com/2006/10/11/whats-with-the-712-area-code 

I posted this question the other day but didn't get an answer, are there
any other toll zones outside of Iowa worth looking at?

 

Regards,

Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
+1-212-203-4357 Ph
+1-917-207-3420 Mb
+61-2-9016-5642 (Sydney in-dial).


> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Cory Andrews
> Sent: Friday, 2 March 2007 2:12 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [asterisk-users] Alec Saunders post about Mashable
Telco's
> 
> Somebody make an application where I can browse to a web front end,
set
> up some RSS feeds, and the system assigns me an access number or PIN.
> Later, when I am stuck in the airport, I call the app, and a decent
TTS
> engine reads me my RSS feeds.
> 
> 
> Cory Andrews
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Steve
> Totaro
> Sent: Friday, March 02, 2007 12:55 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Alec Saunders post about Mashable
Telco's
> 
> Nerd Vittles has some pretty cool apps for news and having your email
> read to you.  I am downloading now, hopefully it is halfway decent.
> 
> Thanks,
> Steve
> 
> Dean Collins wrote:
> > Lol - I didn't mean it that way besides there is a big difference
> > between discussing an application and replicating it.
> >
> > Besides if you are out there selling something the general public is
> > going to know about it.
> >
> > Or should I be checking your public website for the super duper cool
> > ones :)
> >
> > Seriously though why aren't we releasing cooler applications? (and
> > when the hell is someone going to do something more with the text to
> > speech weather app then just weather?)
> >
> >
> >
> > Regards,
> >
> > Dean Collins
> > Cognation Pty Ltd
> > [EMAIL PROTECTED]
> > +1-212-203-4357 Ph
> > +1-917-207-3420 Mb
> > +61-2-9016-5642 (Sydney in-dial).
> >
> >
> >
> >> -Original Message-
> >> From: [EMAIL PROTECTED]
[mailto:asterisk-users-
> 
> >> [EMAIL PROTECTED] On Behalf Of Steve Totaro
> >> Sent: Friday, 2 March 2007 11:19 AM
> >> To: Asterisk Users Mailing List - Non-Commercial Discussion
> >> Subject: Re: [asterisk-users] Alec Saunders post about Mashable
> >>
> > Telco's
> >
> >>> Where are the really interesting asterisk applications for hosted
> >>>
> > ITSP's?
> >
> >>> Can anyone on this list name a Gee Wizz application they cant live
> >>> without? (or are we still just coding ex-girlfriend routines?)
> >>>
> >>> Regards,
> >>>
> >>> Dean Collins
> >>> Cognation Pty Ltd
> >>>
> >>>
> >> Yes, please give me the next "Big Thing", "Gee Wizz" application
too!
> >> Free business ideas are great! I really like the ad for inventors
to
> >> call a toll free number to give the company your ideas.
> >>
> >> Anytime I see someone rallying enthusiasm and then asking for free
> >> ideas, I ask myself, "what is in it for me?". Call me selfish, but
> >> certainly my ideas are worth something, especially my "Gee Wizz"
> >>
> > ideas.
> >
> >> Thanks,
> >> Steve
> >>
> > ___
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> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
> >
> >
> 
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Re: [asterisk-users] Problem with TE212P

[Ioan Indreias]

Hello Benito,

From http://www.beronet.com/download/card_installation_guide.pdf  we 
could find that:


/After loading the driver and the executing "ztcfg", status LEDs for 
each port should be flashing red, unless the port is connected to a 
device. If the LED does not light up, the driver did not load properly 
and execution of "ztcfg" did not complete. When you connect a port to 
your NT PRI, the LED should go green. Green means that the circuit is 
connected physically and the link is active (Layer 1)

/
From your description it looks that for second span ztcfg did not 
complete. What is the output of ztcfg -vv?


Also:
1. It is strange that cat /proc/zaptel/3 do not show HDB3/CCS/CRC4 RED 
NOTOPEN as we could see from cat /proc/zaptel/2. Did you missed that 
info when you past in your first message?


2. Please double check if there is not am empty space after the 
configuration of span #2. Better, rewrite from scratch zaptel.conf file 
to be 100% sure there are no strange characters.


3. What I do not understand is that you mention:
"Before I do load the modules the leds are ligthing" - I think the leds 
should be off before loading the module.


4. Do you have a cross E1 cable in order to inter-connect span #1 with 
span #2? Try to see what is the result - in normal situations the leds 
should turn from "flashing red" to "green".


Best regards,
## nini @ www.modulo.ro ##



Benito Camelas wrote:

thanks Ioan

I've try this too but problem continues

I put :

span=1,1,0,ccs,hdb3,crc4
span=2,2,0,ccs,hdb3,crc4

span=1,1,0,ccs,hdb3,crc4
span=2,0,0,ccs,hdb3,crc4

span=1,0,0,ccs,hdb3,crc4
span=2,0,0,ccs,hdb3,crc4

In all this configurations the problem is the same
I'm desperate
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RE: [asterisk-users] Help Voicemail to SMS using asterisk

[Dean Collins]

Now if only Tellme would agree to a pre-paid sip gateway for the
asterisk community you could have pretty much everything you wanted :)
http://www.voip-info.org/wiki/view/tellme


 

Regards,

Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
+1-212-203-4357 Ph
+1-917-207-3420 Mb
+61-2-9016-5642 (Sydney in-dial).


> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Steve Totaro
> Sent: Friday, 2 March 2007 1:02 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Help Voicemail to SMS using asterisk
> 
> Goke Aruna wrote:
> > Hello all,
> >
> > I will be glad, if someone can throw light on Voicemail to SMS using
> > asterisk.
> >
> > 1. I want my users to dial certain number.
> >
> > 2.  Record a voicemail with destination number.
> >
> > 3.   Convert this Voicemail to Text.
> >
> > 4.   Send the text with sms apps.
> >
> > and I wish i connect my asterisk to smsc directly. Is it possible
> > without kannel?
> >
> > I will be glad, if someone could explain how i can get this done in
> > asterisk.
> >
> > Goksie
> >
> 
> That is a tall order (well the speech to text part anyways).  Maybe
you
> could (mis)use a TTY service for the deaf.  Other than that, voice
> recognition is not very good (unless I am looking in the wrong places,
> something new has come out, or you want to use some outrageously
priced
> solution).  I think the best you can do with what is out there in the
> "free" arena is a very limited vocab like, numbers, letters, yes and
> no.  Text to speech is getting to the decent point.
> 
> Thanks,
> Steve
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RE: Spam? Re: [asterisk-users] cmd page crashes Asterisk SVN-branch-1.4-r57207

[Hall, Eric M.]

Did that. No change



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Lee
Jenkins
Sent: Friday, March 02, 2007 10:24 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Spam? Re: [asterisk-users] cmd page crashes Asterisk
SVN-branch-1.4-r57207

Hall, Eric M. wrote:
> Group
> 
> I'm having some trouble with asterisk and the page cmd.
> Any help would be great!
> 
> This is what's in my extensions.conf
> 
> exten => _**2,1,SIPAddHeader(Call-Info: answer-after=0)
> 
> exten => _**2,2,Page(SIP/36651)|d
> 
> exten => _**2,3,Hangup
> 

Looks like you have at least a syntax error.  You have:

_**2,2,Page(SIP/36651)|d

And it should be

_**2,2,Page(SIP/36651|d)

Try fixing the "d" option by placing it within the right parenthesis and

try it again.

-- 

Warm Regards,

Lee

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Re: [asterisk-users] 1.4 - SLA

[Lacy Moore - Aspendora]

On 3/2/07, Russell Bryant <[EMAIL PROTECTED]> wrote:

If you are interested in beginning to look at it now, just pull the code
from the 1.4 branch.



Russell, I don't have any specifics at this time.  I need to dig a
little further.  I'm thinking the autocontext is what is giving me
fits.  I can receive calls and place calls, but the hint status is not
working.  It currently registers as a hint showing not in use.  It
does not show in use.

I ended up using some of the config from the bottom of the sla.txt
file.  The sample file may be missing the template section.  The
sample config does not match the config in the sla.txt.  I couldn't
get the sample config to work at all.  Again, hopefully over the
weekend I'll be able to get more information.

Using the config in the sample file, the hint status was working.  I
could see the line ringing, but I could not answer the lines or place
calls.  Using the config from the sla.txt file, I could place calls
and receive calls, but the hints never showed any activity, just
always not in use.

If possible, could you provide the config that you've used for
testing?  I'm testing using Polycom phones to try to keep things
simple.  I'm assuming you are using a Polycom.
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Re: [asterisk-users] IAX best practices

[Tzafrir Cohen]

On Fri, Mar 02, 2007 at 10:20:07AM -0800, Tim Panton wrote:
> 
> On 2 Mar 2007, at 08:04, Steve Totaro wrote:
> 
> >Asterisk wrote:
> >>Thanks Steve!
> >>
> >>What are usually the best approaches in troubleshooting the audio
> >>quality issues and QoS related stuff when putting two Asterisk boxes
> >>together via IAX?
> >>
> >>Have you ever tried connecting Asterisk boxes in the same VPN (but  
> >>still
> >>in different countries)?
> >>
> >>Regards,
> >>Alex
> >>
> >>
> >Alex,
> >
> >VPN will only help in keeping your data more secure.  It will  
> >actually hurt your latency and throughput since there is processor  
> >overhead involved in encrypting/decrypting and and bandwidth in  
> >encapsulating the data.
> 
> I've heard rumors that some networks (e.g. 3g mobile carriers) put a  
> QoS on VPN packets to improve
> the response time for their business customers. So in some  
> exceptional cases the IAX over VPN could
> show lower latency than straight IAX, but it is pretty unlikely.

Tunnel it over http, then?

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
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Re: [asterisk-users] DTMF from TDM400P and X100P

[Tzafrir Cohen]

On Fri, Mar 02, 2007 at 09:36:28AM -0800, Yuan LIU wrote:
> With one IVR payment system, I noticed quite a difference in DTMF 
> transmission between these two cards.  The IVR missed nearly all digits 
> from X100P, while receiving digits from TDM fine.
> 
> Since neither card process or synthesize audio, what can the difference be?

(At least the TDM400P actually has a hardware TDMF detector, but it is
not used, AFAIK)

> 
> (This particular IVR has problem with some regular phone devices, too.)

audio quality?

listen to the audio with e.g. ztmonitor.

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

[Mojo with Horan & Company, LLC]

Possibly the called party is not sending their DTMF properly?  maybe 
experiment with inband/rfc2833/etc in the CALLED party's peer definition


Denis V. Gudtsov wrote:

Hello!

I'm using asterisk 1.2.13, in extension.ael is set Dial(SIP/${EXTEN},12,tT)

but only calling party can do forward. How to configure '*' to take this
possibility to called party?

ps
both calling/called use sip


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Re: [asterisk-users] RE: Polycom reject button

[Mojo with Horan & Company, LLC]

Another option is to have the user hit the forward button on their phone 
and manually type in their cellphone number when they're going to be out 
of the office.


Jason Walker wrote:

exten => 111,1,Wait(1)
exten => 111,2,Playback(Randy)
exten => 111,3,Dial(Sip/Randy,20)
exten => 111,4,Goto(111-${DIALSTATUS},1)
exten => 111-BUSY,1,Voicemail([EMAIL PROTECTED],u)
exten => 111-NOANSWER,1,Dial(IAX2/${TELIAX_OUT}/212551212)


works GREAT

Thanks a lot
Jason

Doug Lytle wrote:

Mike wrote:

Jason,

If you do test if JR's tip works, please share your finding with us.  
I am

interested in this as well.
  


It'll work fine, the Polycom responds with BUSY when the DND button is 
pressed.  Using DIALSTATUS, it'll drop to voicemail and play the busy 
message if recorded if that's what you have it programmed to do.

Doug




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[asterisk-users] rtsavesysname not working in 1.4

[David Thomas]

I am trying to have asterisk update the system name in my realtime
peers, but it does not seem to be working. Here is what I've done so
far.

- added systemname => mysystemname in asterisk.conf
- set rtsavesysname=yes in sip.conf.
- created a table called "sysname" in my peers table in mysql
- restarted asterisk
- rebooted my phone to force a re-register

Is there something I'm missing?

Thanks!
David
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Re: [asterisk-users] Test

[Mojo with Horan & Company, LLC]

I DID receive it.  Please don't re-send it.

C F wrote:

Hi, I'm the admin on the list your test didin't work you should resend it.

















































































































Well I am not the admin, just wanted you to realized that you
shouldn't annoy thousands of people just because you don't want to
wait around till the next message arrives to see that you are really
subscribed.



On 3/1/07, Wai Wu <[EMAIL PROTECTED]> wrote:


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[asterisk-users] PRI progress codes.

[John Bittner]

Anyone know how to let asterisk deal with the progress codes coming
from the carrier? The problem I am having is when a customer calls an
invalid number the carrier tells me the call is invalid via a progress
code but doesn't route me to a recording (this number is invalid).
Instead they hang up on me causing a fast busy or sometimes hold up
the call with dead air for 15 to 30 seconds then a fast busy. I am
working with the carrier to get this fixed but its not going easy.
Is there anyway when asterisk sees the progress code to cancel the
dial and playback a message mapped to the progress code type.

Any help on this would be appreciated.

John Bittner
Simlab.net
9734333011



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RE: [asterisk-users] Alec Saunders post about Mashable Telco's

[Dean Collins]

How about mashing text to speech and Avantgo together.

Using dtmf you could have your avantgo selections read to you while you
drive to work in the morning? (or even better use lumenvox/tell me
speech recognition)

 

Regards,

Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
+1-212-203-4357 Ph
+1-917-207-3420 Mb
+61-2-9016-5642 (Sydney in-dial).


> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Steve Totaro
> Sent: Friday, 2 March 2007 12:55 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Alec Saunders post about Mashable
Telco's
> 
> Nerd Vittles has some pretty cool apps for news and having your email
> read to you.  I am downloading now, hopefully it is halfway decent.
> 
> Thanks,
> Steve
> 
> Dean Collins wrote:
> > Lol - I didn't mean it that way besides there is a big difference
> > between discussing an application and replicating it.
> >
> > Besides if you are out there selling something the general public is
> > going to know about it.
> >
> > Or should I be checking your public website for the super duper cool
> > ones :)
> >
> > Seriously though why aren't we releasing cooler applications? (and
when
> > the hell is someone going to do something more with the text to
speech
> > weather app then just weather?)
> >
> >
> >
> > Regards,
> >
> > Dean Collins
> > Cognation Pty Ltd
> > [EMAIL PROTECTED]
> > +1-212-203-4357 Ph
> > +1-917-207-3420 Mb
> > +61-2-9016-5642 (Sydney in-dial).
> >
> >
> >
> >> -Original Message-
> >> From: [EMAIL PROTECTED]
[mailto:asterisk-users-
> >> [EMAIL PROTECTED] On Behalf Of Steve Totaro
> >> Sent: Friday, 2 March 2007 11:19 AM
> >> To: Asterisk Users Mailing List - Non-Commercial Discussion
> >> Subject: Re: [asterisk-users] Alec Saunders post about Mashable
> >>
> > Telco's
> >
> >>> Where are the really interesting asterisk applications for hosted
> >>>
> > ITSP's?
> >
> >>> Can anyone on this list name a Gee Wizz application they cant live
> >>> without? (or are we still just coding ex-girlfriend routines?)
> >>>
> >>> Regards,
> >>>
> >>> Dean Collins
> >>> Cognation Pty Ltd
> >>>
> >>>
> >> Yes, please give me the next "Big Thing", "Gee Wizz" application
too!
> >> Free business ideas are great! I really like the ad for inventors
to
> >> call a toll free number to give the company your ideas.
> >>
> >> Anytime I see someone rallying enthusiasm and then asking for free
> >> ideas, I ask myself, "what is in it for me?". Call me selfish, but
> >> certainly my ideas are worth something, especially my "Gee Wizz"
> >>
> > ideas.
> >
> >> Thanks,
> >> Steve
> >>
> > ___
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> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
> >
> >
> 
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RE: [asterisk-users] Alec Saunders post about Mashable Telco's

[Cory Andrews]

Somebody make an application where I can browse to a web front end, set
up some RSS feeds, and the system assigns me an access number or PIN.
Later, when I am stuck in the airport, I call the app, and a decent TTS
engine reads me my RSS feeds. 


Cory Andrews

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve
Totaro
Sent: Friday, March 02, 2007 12:55 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Alec Saunders post about Mashable Telco's

Nerd Vittles has some pretty cool apps for news and having your email
read to you.  I am downloading now, hopefully it is halfway decent. 

Thanks,
Steve

Dean Collins wrote:
> Lol - I didn't mean it that way besides there is a big difference 
> between discussing an application and replicating it.
>
> Besides if you are out there selling something the general public is 
> going to know about it.
>
> Or should I be checking your public website for the super duper cool 
> ones :)
>
> Seriously though why aren't we releasing cooler applications? (and 
> when the hell is someone going to do something more with the text to 
> speech weather app then just weather?)
>
>  
>
> Regards,
>
> Dean Collins
> Cognation Pty Ltd
> [EMAIL PROTECTED]
> +1-212-203-4357 Ph
> +1-917-207-3420 Mb
> +61-2-9016-5642 (Sydney in-dial).
>
>
>   
>> -Original Message-
>> From: [EMAIL PROTECTED] [mailto:asterisk-users-

>> [EMAIL PROTECTED] On Behalf Of Steve Totaro
>> Sent: Friday, 2 March 2007 11:19 AM
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> Subject: Re: [asterisk-users] Alec Saunders post about Mashable
>> 
> Telco's
>   
>>> Where are the really interesting asterisk applications for hosted
>>>   
> ITSP's?
>   
>>> Can anyone on this list name a Gee Wizz application they cant live 
>>> without? (or are we still just coding ex-girlfriend routines?)
>>>
>>> Regards,
>>>
>>> Dean Collins
>>> Cognation Pty Ltd
>>>
>>>   
>> Yes, please give me the next "Big Thing", "Gee Wizz" application too!
>> Free business ideas are great! I really like the ad for inventors to 
>> call a toll free number to give the company your ideas.
>>
>> Anytime I see someone rallying enthusiasm and then asking for free 
>> ideas, I ask myself, "what is in it for me?". Call me selfish, but 
>> certainly my ideas are worth something, especially my "Gee Wizz"
>> 
> ideas.
>   
>> Thanks,
>> Steve
>> 
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>   

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Re: [asterisk-users] Double DTMF digits sent on IAX native bridge

[Russell Bryant]

Remi Quezada wrote:

I have two asterisk servers one is connected to the PSTN and the other
one is connected to SIP users.  The two servers connect with each other
using IAX.  When I have an incoming call from PSTN to the asterisk
servers and have a forward to go back out to the PSTN the two IAX
channel bridge together.  Now every time I dial a DTMF digit, the
asterisk is sending two DTMF digits.  I enable debugging for iax and I
do see it sending the DTMF digits two times.  Here is what I see:


The IAX debug that you show below only shows one of each digit.  For 
each one, it shows Receiving the digit from one leg of the call, and 
then transmitting it out the other.  I have spaced out your debug to 
separate each digit.


Each one shows ...

   <- digit 
- ACK ->
- digit --->
< ACK --

which is exactly what is supposed to happen.



Rx-Frame Retry[ No] -- OSeqno: 018 ISeqno: 021 Type: DTMFSubclass: 1
   Timestamp: 51523ms  SCall: 3  DCall: 2 [192.168.15.201:4569]
Tx-Frame Retry[-01] -- OSeqno: 021 ISeqno: 019 Type: IAX Subclass:
ACK   
   Timestamp: 51523ms  SCall: 2  DCall: 3 [192.168.15.201:4569]

Tx-Frame Retry[000] -- OSeqno: 019 ISeqno: 022 Type: DTMFSubclass: 1
   Timestamp: 51543ms  SCall: 16385  DCall: 4 [192.168.15.201:4569]
Rx-Frame Retry[ No] -- OSeqno: 022 ISeqno: 020 Type: IAX Subclass:
ACK   
   Timestamp: 51543ms  SCall: 4  DCall: 16385 [192.168.15.201:4569]




Rx-Frame Retry[ No] -- OSeqno: 019 ISeqno: 021 Type: DTMFSubclass: 2
   Timestamp: 52083ms  SCall: 3  DCall: 2 [192.168.15.201:4569]
Tx-Frame Retry[-01] -- OSeqno: 021 ISeqno: 020 Type: IAX Subclass:
ACK   
   Timestamp: 52083ms  SCall: 2  DCall: 3 [192.168.15.201:4569]

Tx-Frame Retry[000] -- OSeqno: 020 ISeqno: 022 Type: DTMFSubclass: 2
   Timestamp: 52103ms  SCall: 16385  DCall: 4 [192.168.15.201:4569]
Rx-Frame Retry[ No] -- OSeqno: 022 ISeqno: 021 Type: IAX Subclass:
ACK   
   Timestamp: 52103ms  SCall: 4  DCall: 16385 [192.168.15.201:4569]




Rx-Frame Retry[ No] -- OSeqno: 020 ISeqno: 021 Type: DTMFSubclass: 3
   Timestamp: 52663ms  SCall: 3  DCall: 2 [192.168.15.201:4569]
Tx-Frame Retry[-01] -- OSeqno: 021 ISeqno: 021 Type: IAX Subclass:
ACK   
   Timestamp: 52663ms  SCall: 2  DCall: 3 [192.168.15.201:4569]

Tx-Frame Retry[000] -- OSeqno: 021 ISeqno: 022 Type: DTMFSubclass: 3
   Timestamp: 52683ms  SCall: 16385  DCall: 4 [192.168.15.201:4569]
Rx-Frame Retry[ No] -- OSeqno: 022 ISeqno: 022 Type: IAX Subclass:
ACK   
   Timestamp: 52683ms  SCall: 4  DCall: 16385 [192.168.15.201:4569]




Rx-Frame Retry[ No] -- OSeqno: 021 ISeqno: 021 Type: DTMFSubclass: 4
   Timestamp: 53223ms  SCall: 3  DCall: 2 [192.168.15.201:4569]
Tx-Frame Retry[-01] -- OSeqno: 021 ISeqno: 022 Type: IAX Subclass:
ACK   
   Timestamp: 53223ms  SCall: 2  DCall: 3 [192.168.15.201:4569]

Tx-Frame Retry[000] -- OSeqno: 022 ISeqno: 022 Type: DTMFSubclass: 4
   Timestamp: 53243ms  SCall: 16385  DCall: 4 [192.168.15.201:4569]
Rx-Frame Retry[ No] -- OSeqno: 022 ISeqno: 023 Type: IAX Subclass:
ACK   
   Timestamp: 53243ms  SCall: 4  DCall: 16385 [192.168.15.201:4569]




Rx-Frame Retry[ No] -- OSeqno: 022 ISeqno: 021 Type: DTMFSubclass: 5
   Timestamp: 53703ms  SCall: 3  DCall: 2 [192.168.15.201:4569]
Tx-Frame Retry[-01] -- OSeqno: 021 ISeqno: 023 Type: IAX Subclass:
ACK   
   Timestamp: 53703ms  SCall: 2  DCall: 3 [192.168.15.201:4569]

Tx-Frame Retry[000] -- OSeqno: 023 ISeqno: 022 Type: DTMFSubclass: 5
   Timestamp: 53723ms  SCall: 16385  DCall: 4 [192.168.15.201:4569]
Rx-Frame Retry[ No] -- OSeqno: 022 ISeqno: 024 Type: IAX Subclass:
ACK   
   Timestamp: 53723ms  SCall: 4  DCall: 16385 [192.168.15.201:4569]




Rx-Frame Retry[ No] -- OSeqno: 023 ISeqno: 021 Type: DTMFSubclass: 6
   Timestamp: 54163ms  SCall: 3  DCall: 2 [192.168.15.201:4569]
Tx-Frame Retry[-01] -- OSeqno: 021 ISeqno: 024 Type: IAX Subclass:
ACK   
   Timestamp: 54163ms  SCall: 2  DCall: 3 [192.168.15.201:4569]

Tx-Frame Retry[000] -- OSeqno: 024 ISeqno: 022 Type: DTMFSubclass: 6
   Timestamp: 54183ms  SCall: 16385  DCall: 4 [192.168.15.201:4569]
Rx-Frame Retry[ No] -- OSeqno: 022 ISeqno: 025 Type: IAX Subclass:
ACK   
   Timestamp: 54183ms  SCall: 4  DCall: 16385 [192.168.15.201:4569]



--
Russell Bryant
Software Engineer
Digium, Inc.
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Re: [asterisk-users] 1.4 - SLA

[Russell Bryant]

Bruce Reeves wrote:
There was talk last week that SLA in 1.4 was not working correctly and 
was being rewritten for a 1.4.1 release.


The re-write of SLA support in Asterisk 1.4 is pretty much complete. 
Asterisk 1.4.1 will be released very soon with it included.  I don't 
expect to make any more changes until people start testing it out and 
submitting bug reports.


If you are interested in beginning to look at it now, just pull the code 
from the 1.4 branch.


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

Once you have it, take a look at these two files for information on how 
to set it up:


doc/sla.txt
configs/sla.conf.sample

It supports any channel type on the trunk side, but only SIP on the 
station side.  This is not a shortcoming of the SLA implementation 
itself, but just that chan_sip is the only channel driver that supports 
all of the required features.


However, we are already working on chan_skinny in the trunk to get it up 
to where it can support SLA as well.


--
Russell Bryant
Software Engineer
Digium, Inc.
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Re: [asterisk-users] IAX best practices

[Tim Panton]

On 2 Mar 2007, at 08:04, Steve Totaro wrote:


Asterisk wrote:

Thanks Steve!

What are usually the best approaches in troubleshooting the audio
quality issues and QoS related stuff when putting two Asterisk boxes
together via IAX?

Have you ever tried connecting Asterisk boxes in the same VPN (but  
still

in different countries)?

Regards,
Alex



Alex,

VPN will only help in keeping your data more secure.  It will  
actually hurt your latency and throughput since there is processor  
overhead involved in encrypting/decrypting and and bandwidth in  
encapsulating the data.


I've heard rumors that some networks (e.g. 3g mobile carriers) put a  
QoS on VPN packets to improve
the response time for their business customers. So in some  
exceptional cases the IAX over VPN could

show lower latency than straight IAX, but it is pretty unlikely.


Tim Panton

www.mexuar.net
www.westhawk.co.uk/



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Re: [asterisk-users] Help Voicemail to SMS using asterisk

[Steve Totaro]

Goke Aruna wrote:

Hello all,

I will be glad, if someone can throw light on Voicemail to SMS using
asterisk.

1. I want my users to dial certain number.

2.  Record a voicemail with destination number.

3.   Convert this Voicemail to Text.

4.   Send the text with sms apps.

and I wish i connect my asterisk to smsc directly. Is it possible
without kannel?

I will be glad, if someone could explain how i can get this done in
asterisk.

Goksie
  


That is a tall order (well the speech to text part anyways).  Maybe you 
could (mis)use a TTY service for the deaf.  Other than that, voice 
recognition is not very good (unless I am looking in the wrong places, 
something new has come out, or you want to use some outrageously priced 
solution).  I think the best you can do with what is out there in the 
"free" arena is a very limited vocab like, numbers, letters, yes and 
no.  Text to speech is getting to the decent point.


Thanks,
Steve
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FW: [asterisk-users] Alec Saunders post about Mashable Telco's

[Dean Collins]

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Tim Panton
> Sent: Friday, 2 March 2007 11:36 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Alec Saunders post about Mashable
Telco's
> 
> On the otherhand there were a number of weird and wonderful asterisk
> based devices here at eTel,
> like the dtmf steered vacuum cleaner or the front door you can open
> with a phonecall.
> 
> Tim.
> 


WTF Source please :) 



Regards,

Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
+1-212-203-4357 Ph
+1-917-207-3420 Mb
+61-2-9016-5642 (Sydney in-dial).
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Re: [asterisk-users] Alec Saunders post about Mashable Telco's

[Steve Totaro]

Nerd Vittles has some pretty cool apps for news and having your email 
read to you.  I am downloading now, hopefully it is halfway decent. 


Thanks,
Steve

Dean Collins wrote:

Lol - I didn't mean it that way besides there is a big difference
between discussing an application and replicating it.

Besides if you are out there selling something the general public is
going to know about it.

Or should I be checking your public website for the super duper cool
ones :)

Seriously though why aren't we releasing cooler applications? (and when
the hell is someone going to do something more with the text to speech
weather app then just weather?)

 


Regards,

Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
+1-212-203-4357 Ph
+1-917-207-3420 Mb
+61-2-9016-5642 (Sydney in-dial).


  

-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Steve Totaro
Sent: Friday, 2 March 2007 11:19 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Alec Saunders post about Mashable


Telco's
  

Where are the really interesting asterisk applications for hosted
  

ITSP's?
  

Can anyone on this list name a Gee Wizz application they cant live
without? (or are we still just coding ex-girlfriend routines?)

Regards,

Dean Collins
Cognation Pty Ltd

  

Yes, please give me the next "Big Thing", "Gee Wizz" application too!
Free business ideas are great! I really like the ad for inventors to
call a toll free number to give the company your ideas.

Anytime I see someone rallying enthusiasm and then asking for free
ideas, I ask myself, "what is in it for me?". Call me selfish, but
certainly my ideas are worth something, especially my "Gee Wizz"


ideas.
  

Thanks,
Steve


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[asterisk-users] DTMF from TDM400P and X100P

[Yuan LIU]

With one IVR payment system, I noticed quite a difference in DTMF 
transmission between these two cards.  The IVR missed nearly all digits from 
X100P, while receiving digits from TDM fine.


Since neither card process or synthesize audio, what can the difference be?

(This particular IVR has problem with some regular phone devices, too.)

Yuan Liu


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[asterisk-users] Double DTMF digits sent on IAX native bridge

[Remi Quezada]

Hi,

I have two asterisk servers one is connected to the PSTN and the other
one is connected to SIP users.  The two servers connect with each other
using IAX.  When I have an incoming call from PSTN to the asterisk
servers and have a forward to go back out to the PSTN the two IAX
channel bridge together.  Now every time I dial a DTMF digit, the
asterisk is sending two DTMF digits.  I enable debugging for iax and I
do see it sending the DTMF digits two times.  Here is what I see:


Rx-Frame Retry[ No] -- OSeqno: 022 ISeqno: 019 Type: IAX Subclass:
ACK   
   Timestamp: 50018ms  SCall: 4  DCall: 16385 [192.168.15.201:4569]
Rx-Frame Retry[ No] -- OSeqno: 018 ISeqno: 021 Type: DTMFSubclass: 1
   Timestamp: 51523ms  SCall: 3  DCall: 2 [192.168.15.201:4569]
Tx-Frame Retry[-01] -- OSeqno: 021 ISeqno: 019 Type: IAX Subclass:
ACK   
   Timestamp: 51523ms  SCall: 2  DCall: 3 [192.168.15.201:4569]
Tx-Frame Retry[000] -- OSeqno: 019 ISeqno: 022 Type: DTMFSubclass: 1
   Timestamp: 51543ms  SCall: 16385  DCall: 4 [192.168.15.201:4569]
Rx-Frame Retry[ No] -- OSeqno: 022 ISeqno: 020 Type: IAX Subclass:
ACK   
   Timestamp: 51543ms  SCall: 4  DCall: 16385 [192.168.15.201:4569]
Rx-Frame Retry[ No] -- OSeqno: 019 ISeqno: 021 Type: DTMFSubclass: 2
   Timestamp: 52083ms  SCall: 3  DCall: 2 [192.168.15.201:4569]
Tx-Frame Retry[-01] -- OSeqno: 021 ISeqno: 020 Type: IAX Subclass:
ACK   
   Timestamp: 52083ms  SCall: 2  DCall: 3 [192.168.15.201:4569]
Tx-Frame Retry[000] -- OSeqno: 020 ISeqno: 022 Type: DTMFSubclass: 2
   Timestamp: 52103ms  SCall: 16385  DCall: 4 [192.168.15.201:4569]
Rx-Frame Retry[ No] -- OSeqno: 022 ISeqno: 021 Type: IAX Subclass:
ACK   
   Timestamp: 52103ms  SCall: 4  DCall: 16385 [192.168.15.201:4569]
Rx-Frame Retry[ No] -- OSeqno: 020 ISeqno: 021 Type: DTMFSubclass: 3
   Timestamp: 52663ms  SCall: 3  DCall: 2 [192.168.15.201:4569]
Tx-Frame Retry[-01] -- OSeqno: 021 ISeqno: 021 Type: IAX Subclass:
ACK   
   Timestamp: 52663ms  SCall: 2  DCall: 3 [192.168.15.201:4569]
Tx-Frame Retry[000] -- OSeqno: 021 ISeqno: 022 Type: DTMFSubclass: 3
   Timestamp: 52683ms  SCall: 16385  DCall: 4 [192.168.15.201:4569]
Rx-Frame Retry[ No] -- OSeqno: 022 ISeqno: 022 Type: IAX Subclass:
ACK   
   Timestamp: 52683ms  SCall: 4  DCall: 16385 [192.168.15.201:4569]
Rx-Frame Retry[ No] -- OSeqno: 021 ISeqno: 021 Type: DTMFSubclass: 4
   Timestamp: 53223ms  SCall: 3  DCall: 2 [192.168.15.201:4569]
Tx-Frame Retry[-01] -- OSeqno: 021 ISeqno: 022 Type: IAX Subclass:
ACK   
   Timestamp: 53223ms  SCall: 2  DCall: 3 [192.168.15.201:4569]
Tx-Frame Retry[000] -- OSeqno: 022 ISeqno: 022 Type: DTMFSubclass: 4
   Timestamp: 53243ms  SCall: 16385  DCall: 4 [192.168.15.201:4569]
Rx-Frame Retry[ No] -- OSeqno: 022 ISeqno: 023 Type: IAX Subclass:
ACK   
   Timestamp: 53243ms  SCall: 4  DCall: 16385 [192.168.15.201:4569]
Rx-Frame Retry[ No] -- OSeqno: 022 ISeqno: 021 Type: DTMFSubclass: 5
   Timestamp: 53703ms  SCall: 3  DCall: 2 [192.168.15.201:4569]
Tx-Frame Retry[-01] -- OSeqno: 021 ISeqno: 023 Type: IAX Subclass:
ACK   
   Timestamp: 53703ms  SCall: 2  DCall: 3 [192.168.15.201:4569]
Tx-Frame Retry[000] -- OSeqno: 023 ISeqno: 022 Type: DTMFSubclass: 5
   Timestamp: 53723ms  SCall: 16385  DCall: 4 [192.168.15.201:4569]
Rx-Frame Retry[ No] -- OSeqno: 022 ISeqno: 024 Type: IAX Subclass:
ACK   
   Timestamp: 53723ms  SCall: 4  DCall: 16385 [192.168.15.201:4569]
Rx-Frame Retry[ No] -- OSeqno: 023 ISeqno: 021 Type: DTMFSubclass: 6
   Timestamp: 54163ms  SCall: 3  DCall: 2 [192.168.15.201:4569]
Tx-Frame Retry[-01] -- OSeqno: 021 ISeqno: 024 Type: IAX Subclass:
ACK   
   Timestamp: 54163ms  SCall: 2  DCall: 3 [192.168.15.201:4569]
Tx-Frame Retry[000] -- OSeqno: 024 ISeqno: 022 Type: DTMFSubclass: 6
   Timestamp: 54183ms  SCall: 16385  DCall: 4 [192.168.15.201:4569]
Rx-Frame Retry[ No] -- OSeqno: 022 ISeqno: 025 Type: IAX Subclass:
ACK   
   Timestamp: 54183ms  SCall: 4  DCall: 16385 [192.168.15.201:4569]

Any ideas on how I can fix this so it only detects one?  I've tried
relaxing the DTMF detect, but that didn't seem to work.  This is only
happening when the two IAX channel bridge together.  If I have a call
terminating to a SIP user or the PSTN and dial any digit it detects it
only once.   I verified this by turning on iax debugging also. 

Thanks

Remi
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Re: [asterisk-users] Re: Sending SMS

[Michiel van Baak]

On 08:42, Fri 02 Mar 07, Steve Totaro wrote:
> Do they let you specify what number the SMS is coming from or does it 
> just come from one of their number pool?  Is it possible for the person 
> reply to the SMS and have it come straight to my phone?

Yes. Both CIDname and CIDnumber can be specified at will.
Makes it real nice for our setup so we know wether the SMS
is from monitoring or something else ;)

-- 

Michiel van Baak
[EMAIL PROTECTED]
http://michiel.vanbaak.eu
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x71C946BD

"Why is it drug addicts and computer afficionados are both called users?"

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[asterisk-users] Voicemail to SMS using asterisk

[Goke Aruna]

Hello all,

I will be glad, if someone can throw light on Voicemail to SMS using
asterisk.

1. I want my users to dial certain number.

2.  Record a voicemail with destination number.

3.   Convert this Voicemail to Text.

4.   Send the text with sms apps.

and I wish i connect my asterisk to smsc directly. Is it possible
without kannel?

I will be glad, if someone could explain how i can get this done in
asterisk.

Goksie
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Re: [asterisk-users] gtalktovoip and Asteirsk

[Cosmin Prund]

I don't think it works. I tried calling my own yahoo messenger ID with 
no success: it rings a number of times and then it goes to some sort of 
voice mail.
And I did "invite" the user they specified to my yahoo list, I also 
entered my yahoo id into the registration form on the site.

I used a extensions.conf command like this for the try:

exten => 641,1,Dial(SIP/[EMAIL PROTECTED])

(and yes, that's one of the yahoo ID I tryed with, and I don't think it 
exists! )


Klaverstyn, David C wrote:


Has anyone managed to get gtalktovoip working at all?  If so please 
explain.


 


http://www.gtalk2voip.com/faq.shtml

 

 


*2. Q: Ok, how can I call Google Talk, MSN or Yahoo users from SIP ?*

A: This is a major feature of our gateway and it is very easy.

oGTalk: [EMAIL PROTECTED] can be reached by calling to 
sip:[EMAIL PROTECTED]


oMSN: [EMAIL PROTECTED] can be reached by calling to 
sip:[EMAIL PROTECTED]


oYahoo: [EMAIL PROTECTED] can be reached by calling to 
sip:[EMAIL PROTECTED]


 




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Re: [asterisk-users] Asterisk and Fax

[--[ UxBoD ]--]

On Fri, 02 Mar 2007 10:59:25 -0500
"Robert A. Rawlinson" <[EMAIL PROTECTED]> wrote:

> I would just put the phone line into the fax  machine and the phone jack 
> from it into the fxs card. It is a simple solution.
> Bob
> 
> --[ UxBoD ]-- wrote:
> > Hi,
> >
> > I have a requirement for sending and receiving faxes and was wondering the 
> > best way to achieve it with Asterisk as
> > I only have one phone line.
> >
> > I currently have a TDM11B in my server (1 x FXO, 1 x FXS), so I was 
> > thinking that I would need to get a additional
> > FXS module, connect that to a Eicon Fax card, and then when receiving a 
> > call detect the fax tone and bridge the
> > call to the FXS channel.
> >
> > Would that work okay ? Just seems fax cards are very expensive for what 
> > they do.
> >   
> 
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> 
The reason for going down the route I proposed was so that faxes would end up 
as attachments in my partners mailbox.

I am trying to centralise emails, voicemail and faxes on our home server for 
her business.

Regards,

-- 
--[ UxBoD ]--
// PGP Key: "curl -s http://www.splatnix.net/uxbod.asc | gpg --import"
// Fingerprint: 543A E778 7F2D 98F1 3E50 9C1F F190 93E0 E8E8 0CF8
// Keyserver: www.keyserver.net Key-ID: 0xE8E80CF8
// SIP:[EMAIL PROTECTED]

-- 
This message has been scanned for viruses and dangerous content by MailScanner, 
and is
believed to be clean.

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[asterisk-users] Help Voicemail to SMS using asterisk

[Goke Aruna]

Hello all,

I will be glad, if someone can throw light on Voicemail to SMS using
asterisk.

1. I want my users to dial certain number.

2.  Record a voicemail with destination number.

3.   Convert this Voicemail to Text.

4.   Send the text with sms apps.

and I wish i connect my asterisk to smsc directly. Is it possible
without kannel?

I will be glad, if someone could explain how i can get this done in
asterisk.

Goksie
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RE: [asterisk-users] Alec Saunders post about Mashable Telco's

[Dean Collins]

Lol - I didn't mean it that way besides there is a big difference
between discussing an application and replicating it.

Besides if you are out there selling something the general public is
going to know about it.

Or should I be checking your public website for the super duper cool
ones :)

Seriously though why aren't we releasing cooler applications? (and when
the hell is someone going to do something more with the text to speech
weather app then just weather?)

 

Regards,

Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
+1-212-203-4357 Ph
+1-917-207-3420 Mb
+61-2-9016-5642 (Sydney in-dial).


> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Steve Totaro
> Sent: Friday, 2 March 2007 11:19 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Alec Saunders post about Mashable
Telco's
> 
> > Where are the really interesting asterisk applications for hosted
ITSP's?
> >
> > Can anyone on this list name a Gee Wizz application they cant live
> > without? (or are we still just coding ex-girlfriend routines?)
> >
> > Regards,
> >
> > Dean Collins
> > Cognation Pty Ltd
> >
> Yes, please give me the next "Big Thing", "Gee Wizz" application too!
> Free business ideas are great! I really like the ad for inventors to
> call a toll free number to give the company your ideas.
> 
> Anytime I see someone rallying enthusiasm and then asking for free
> ideas, I ask myself, "what is in it for me?". Call me selfish, but
> certainly my ideas are worth something, especially my "Gee Wizz"
ideas.
> 
> Thanks,
> Steve
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Re: [asterisk-users] IAX best practices

[Steve Totaro]

Henry J. Cobb wrote:

You will likely have latency issues - causing choppiness.  Start with a
traceroute to validate latency.



Anybody tried IAX trunking on G.729 with jitter buffer internationaly?

-HJC
  
Not G.729 but SPEEX 8 variable.  Sounded great but lots of delay where 
GSM was breaking up so badly, there could be no conversation at all. 


Thanks,
Steve
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Re: [asterisk-users] Alec Saunders post about Mashable Telco's

[Tim Panton]

I don't think it is a matter of GeeWhizz, for me it is about simple  
things that make my life easier.


1) I get an SMS from meetme when the first joiner arrives , the sms  
has the DID of the conference room in it.
2) I get calls from asterisk when one of our servers are down. The  
calls are placed by a box on the PSTN so

it doesn't require any network resources.
3) If I want content delivered to mobiles I can use asterisk to spot  
a  UK mobile number and have it send an

SMS to the caller containing a link to the content I want to push.

On the otherhand there were a number of weird and wonderful asterisk  
based devices here at eTel,
like the dtmf steered vacuum cleaner or the front door you can open  
with a phonecall.


Tim.


On 2 Mar 2007, at 07:35, Dean Collins wrote:

Interesting read in Alec Saunders blog today. http:// 
saunderslog.com/2007/03/01/mashable-telcos/




Thought it may interest some people on this list.



As food for thought, why it is that ITSP’s haven’t come up with  
more ‘interesting’ voice applications? When asterisk first became  
available I thought it was the beginning of seeing really neat  
applications, think Verzion’s iobi but for everyone.




I mean I love Angel.com’s (www.angel.com) 1800 speech enabled  
services on demand but I want it available via a SIP gateway to  
hook up with my own phone numbers, and I cant wait for more people  
to implement Iotum so when I try to make calls I wont continually  
be hitting voicemails (www.iotum.com) but I still think it is  
missing ‘pieces’ for widespread adoption.




Where are the really interesting asterisk applications for hosted  
ITSP’s?




Can anyone on this list name a Gee Wizz application they cant live  
without? (or are we still just coding ex-girlfriend routines?)






Regards,

Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
+1-212-203-4357 Ph
+1-917-207-3420 Mb
+61-2-9016-5642 (Sydney in-dial).



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Tim Panton

www.mexuar.net
www.westhawk.co.uk/



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Re: [asterisk-users] IAX best practices

[Zoa]

Some tricks:

If you have a high latency link, watch out i've experienced problems 
with it in the past. (high latency = > 300ms)
Stay away from trunking unless you have a lot of time to spend, if you 
do use trunks, do not use a jitter buffer, make sure it works in both 
directions and don't make it too big (although that might be fixed in 
more recent asterisk versions, there was a patch included for it).


If you use a VPN, make sure it is UDP based, and check it for low latency.
Make sure both asterisk's are on a public ip. (Although you should be 
able to connect to an asterisk B registered to Asterisk A, it - at least 
used to - not work very well in production).


If you do a lot of simultaneous calls, make sure your vpn servers can 
handle the load.


Zoa
www.asteriskguru.com

Michelle Dupuis wrote:

You will likely have latency issues - causing choppiness.  Start with a
traceroute to validate latency.

Michelle Dupuis
Technical Support Specialist

Generation Software - Linux and Asterisk solutions and support
Visit us at www.generationd.com
 
 


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Asterisk
Sent: Friday, March 02, 2007 10:26 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] IAX best practices

Thanks Steve!

What are usually the best approaches in troubleshooting the audio quality
issues and QoS related stuff when putting two Asterisk boxes together via
IAX?

Have you ever tried connecting Asterisk boxes in the same VPN (but still in
different countries)?

Regards,
Alex


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro
Sent: Thursday, March 01, 2007 4:22 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] IAX best practices

Asterisk wrote:
  

Hi guys,

I am planning to connect two Asterisk boxes that are currently running 
in two different countries, using IAX.


I was wondering if anyone could provide me with some links or


suggestion
  

regarding best practices in connecting two Asterisk in such way. I


guess
  
many of you have already tried this, and already have some know-how 
(what I should be careful about, what to avoid, etc...)?


Regards,
Alex
  


Bandwidth and latency.  IAX2 is remarakably good at traversing NAT and even
double NATs.  It should just work.  The issues that I ran into are low
bandwidth and latency.  Not much you can do about latency besides getting a
better route and putting QoS on your equipment and hoping that

your provider either observes your tagging or is not very latent to begin
with.  The other is bandwidth which I found SPEEX works wonders (but adds to
latency).

In my experience, bandwidth issues result in choppy audio and latency
results in delays which cause people to talk on top of each other and can be
extremely annoying.

Try pinging a router or device at the remote side to get an idea of how
latent your connection will be. 


Thanks,
Steve
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Re: [asterisk-users] RE: Polycom reject button

[Jason Walker]

exten => 111,1,Wait(1)
exten => 111,2,Playback(Randy)
exten => 111,3,Dial(Sip/Randy,20)
exten => 111,4,Goto(111-${DIALSTATUS},1)
exten => 111-BUSY,1,Voicemail([EMAIL PROTECTED],u)
exten => 111-NOANSWER,1,Dial(IAX2/${TELIAX_OUT}/212551212)


works GREAT

Thanks a lot
Jason

Doug Lytle wrote:

Mike wrote:

Jason,

If you do test if JR's tip works, please share your finding with us.  
I am

interested in this as well.
  


It'll work fine, the Polycom responds with BUSY when the DND button is 
pressed.  Using DIALSTATUS, it'll drop to voicemail and play the busy 
message if recorded if that's what you have it programmed to do.

Doug




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RE: [asterisk-users] IAX best practices

[Asterisk]

Thanks for the tip.

Would that apply to connections via VPN, or is this likely to happen in
any scenario where two Asterisk boxes are far away from each other?

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michelle
Dupuis
Sent: Friday, March 02, 2007 4:54 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [asterisk-users] IAX best practices

You will likely have latency issues - causing choppiness.  Start with a
traceroute to validate latency.

Michelle Dupuis
Technical Support Specialist

Generation Software - Linux and Asterisk solutions and support
Visit us at www.generationd.com
 
 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Asterisk
Sent: Friday, March 02, 2007 10:26 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] IAX best practices

Thanks Steve!

What are usually the best approaches in troubleshooting the audio
quality
issues and QoS related stuff when putting two Asterisk boxes together
via
IAX?

Have you ever tried connecting Asterisk boxes in the same VPN (but still
in
different countries)?

Regards,
Alex


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve
Totaro
Sent: Thursday, March 01, 2007 4:22 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] IAX best practices

Asterisk wrote:
> Hi guys,
>
> I am planning to connect two Asterisk boxes that are currently running

> in two different countries, using IAX.
>
> I was wondering if anyone could provide me with some links or
suggestion
> regarding best practices in connecting two Asterisk in such way. I
guess
> many of you have already tried this, and already have some know-how 
> (what I should be careful about, what to avoid, etc...)?
>
> Regards,
> Alex
>   
Bandwidth and latency.  IAX2 is remarakably good at traversing NAT and
even
double NATs.  It should just work.  The issues that I ran into are low
bandwidth and latency.  Not much you can do about latency besides
getting a
better route and putting QoS on your equipment and hoping that

your provider either observes your tagging or is not very latent to
begin
with.  The other is bandwidth which I found SPEEX works wonders (but
adds to
latency).

In my experience, bandwidth issues result in choppy audio and latency
results in delays which cause people to talk on top of each other and
can be
extremely annoying.

Try pinging a router or device at the remote side to get an idea of how
latent your connection will be. 

Thanks,
Steve
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RE: [asterisk-users] DTMF detection problems on PRI channels?

[Michelle Dupuis]

Sounds like the DTMF tones are too far from spec, or noisy.  Is the DTMF
being transcoded somewhere along the way?

If you have time to killtry to separate the two frequencies in your
software (I don't know goldwave) - are both present and clean and same
amplitude and on freq?  Remove the two frequencies and what's left?  If
there's a lot of noise, then the other party is doing a bad job encoding the
DTMF.  Otherwise we can start to chase your machine causes

Michelle Dupuis
Technical Support Specialist

Generation Software - Linux and Asterisk solutions and support.  Visit us at
www.generationd.com
 



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tony
Mountifield
Sent: Friday, March 02, 2007 10:56 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] DTMF detection problems on PRI channels?

I am using Asterisk 1.2 with a TE410P connected to E1 PRI trunks.
The application relies on a DTMF digit string sent by the phone after the
call has connected. This DTMF is detected by Asterisk under the control of
WAIT FOR DIGIT commands send from an AGI processor over a FastAGI
connection.

Usually the DTMF is detected without error, but on a significant minority of
calls, Asterisk is missing digits.

In order to diagnose this, I modified chan_zap to save the received Alaw
audio direct to a file, BEFORE the dsp is called for DTMF detection.
I needed to do this because the detection routines do not pass the DTMF
audio on, so using the standard recording or monitoring commands from the
dialplan does not actually capture the tones as received from the wire.
This capturing is turned on and off by an AGI command, so that my AGI
program can turn it on before waiting for the DTMF string and off again
afterwards.

Examining this captured audio in an audio editor such as Goldwave does not
provide any clue why the digits might have been missed. On most occasions
the digits are clear, long enough and well spaced. Yet Asterisk still misses
them.

The system does not seem to have been heavily loaded at the time either.

Can anyone offer any clues as to why this might be the case, and what I
could do to solve it? Hacking the code doesn't bother me, although I know
very little about DSP.

Last I knew, the TE411P board could do on-board DTMF detection, but that the
newer TE412P could not. Is that still the case?

Cheers
Tony
--
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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[asterisk-users] WMI from Asterisk to Cisco Call Manager

[Frédéric Marti]

Hi all,
 
We want to put an Asterisk Voicemail Server behind a Cisco Call Manager.
The idea is to have Cisco Phones (SCCP) registred to the CCM and the voicemail 
in the Asterisk Box.
The trunk inter PBX is in SIP.
 
My question is:
Is it possible to activate MWI LED from the Asterisk to the Cisco Phones 
registred to the CCM when they receive a new voicemail ?
 
Thanks in advance
Fred
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Re: [asterisk-users] Alec Saunders post about Mashable Telco's

[Steve Totaro]

Dean Collins wrote:


Interesting read in Alec Saunders blog today. 
http://saunderslog.com/2007/03/01/mashable-telcos/


Thought it may interest some people on this list.

As food for thought, why it is that ITSP’s haven’t come up with more 
‘interesting’ voice applications? When asterisk first became available 
I thought it was the beginning of seeing really neat applications, 
think Verzion’s iobi but for everyone.


I mean I love Angel.com’s (www.angel.com ) 1800 
speech enabled services on demand but I want it available via a SIP 
gateway to hook up with my own phone numbers, and I cant wait for more 
people to implement Iotum so when I try to make calls I wont 
continually be hitting voicemails (www.iotum.com 
) but I still think it is missing ‘pieces’ for 
widespread adoption.


Where are the really interesting asterisk applications for hosted ITSP’s?

Can anyone on this list name a Gee Wizz application they cant live 
without? (or are we still just coding ex-girlfriend routines?)


Regards,

Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
+1-212-203-4357 Ph
+1-917-207-3420 Mb
+61-2-9016-5642 (Sydney in-dial).

Yes, please give me the next "Big Thing", "Gee Wizz" application too! 
Free business ideas are great! I really like the ad for inventors to 
call a toll free number to give the company your ideas.


Anytime I see someone rallying enthusiasm and then asking for free 
ideas, I ask myself, "what is in it for me?". Call me selfish, but 
certainly my ideas are worth something, especially my "Gee Wizz" ideas.


Thanks,
Steve
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RE: [asterisk-users] IAX best practices

[Henry J. Cobb]

> You will likely have latency issues - causing choppiness.  Start with a
> traceroute to validate latency.

Anybody tried IAX trunking on G.729 with jitter buffer internationaly?

-HJC

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Re: [asterisk-users] Asterisk and Fax

[Steve Totaro]

--[ UxBoD ]-- wrote:

Hi,

I have a requirement for sending and receiving faxes and was wondering the best 
way to achieve it with Asterisk as I only have one phone line.

I currently have a TDM11B in my server (1 x FXO, 1 x FXS), so I was thinking 
that I would need to get a additional FXS module, connect that to a Eicon Fax 
card, and then when receiving a call detect the fax tone and bridge the call to 
the FXS channel.

Would that work okay ? Just seems fax cards are very expensive for what they do.
  

Setup a separate server with the modem and run Hylafax.

Thanks,
Steve
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Re: [asterisk-users] IAX best practices

[Steve Totaro]

Asterisk wrote:

Thanks Steve!

What are usually the best approaches in troubleshooting the audio
quality issues and QoS related stuff when putting two Asterisk boxes
together via IAX?

Have you ever tried connecting Asterisk boxes in the same VPN (but still
in different countries)?

Regards,
Alex

  

Alex,

VPN will only help in keeping your data more secure.  It will actually 
hurt your latency and throughput since there is processor overhead 
involved in encrypting/decrypting and and bandwidth in encapsulating the 
data.


As I said before, audio issues with voice breaking up are usually 
indicative of not enough bandwidth and delay (sometimes echo) is cause 
by latency.  Unless your carrier offers QoS end to end, there is very 
little you can do about it except on your little piece of of the 
network.  You can always buy more bandwidth though.


How big is the pipe and what are your ping times (through the VPN and 
over the public internet)?


Thanks,
Steve
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Re: [asterisk-users] Asterisk and Fax

[Robert A. Rawlinson]

I would just put the phone line into the fax  machine and the phone jack 
from it into the fxs card. It is a simple solution.

Bob

--[ UxBoD ]-- wrote:

Hi,

I have a requirement for sending and receiving faxes and was wondering the best 
way to achieve it with Asterisk as I only have one phone line.

I currently have a TDM11B in my server (1 x FXO, 1 x FXS), so I was thinking 
that I would need to get a additional FXS module, connect that to a Eicon Fax 
card, and then when receiving a call detect the fax tone and bridge the call to 
the FXS channel.

Would that work okay ? Just seems fax cards are very expensive for what they do.
  


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Re: [asterisk-users] Asterisk and Fax

[Tzafrir Cohen]

On Fri, Mar 02, 2007 at 03:34:19PM +, --[ UxBoD ]-- wrote:
> Hi,
> 
> I have a requirement for sending and receiving faxes and was 
> wondering the best way to achieve it with Asterisk as I only have one 
> phone line.
> 
> I currently have a TDM11B in my server (1 x FXO, 1 x FXS), so I was 
> thinking that I would need to get a additional FXS module, connect 
> that to a Eicon Fax card, and then when receiving a call detect the 
> fax tone and bridge the call to the FXS channel.

Why not put a fax machine on the FXS port?

Or route those calls to a local iaxmodem/hylafax? or simply recieve
faxes to a rxfax?

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
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[asterisk-users] DTMF detection problems on PRI channels?

[Tony Mountifield]

I am using Asterisk 1.2 with a TE410P connected to E1 PRI trunks.
The application relies on a DTMF digit string sent by the phone
after the call has connected. This DTMF is detected by Asterisk
under the control of WAIT FOR DIGIT commands send from an AGI
processor over a FastAGI connection.

Usually the DTMF is detected without error, but on a significant minority
of calls, Asterisk is missing digits.

In order to diagnose this, I modified chan_zap to save the received
Alaw audio direct to a file, BEFORE the dsp is called for DTMF detection.
I needed to do this because the detection routines do not pass the DTMF
audio on, so using the standard recording or monitoring commands from
the dialplan does not actually capture the tones as received from the wire.
This capturing is turned on and off by an AGI command, so that my AGI
program can turn it on before waiting for the DTMF string and off again
afterwards.

Examining this captured audio in an audio editor such as Goldwave does
not provide any clue why the digits might have been missed. On most
occasions the digits are clear, long enough and well spaced. Yet Asterisk
still misses them.

The system does not seem to have been heavily loaded at the time either.

Can anyone offer any clues as to why this might be the case, and what I
could do to solve it? Hacking the code doesn't bother me, although I know
very little about DSP.

Last I knew, the TE411P board could do on-board DTMF detection, but that
the newer TE412P could not. Is that still the case?

Cheers
Tony
-- 
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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RE: [asterisk-users] IAX best practices

[Michelle Dupuis]

You will likely have latency issues - causing choppiness.  Start with a
traceroute to validate latency.

Michelle Dupuis
Technical Support Specialist

Generation Software - Linux and Asterisk solutions and support
Visit us at www.generationd.com
 
 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Asterisk
Sent: Friday, March 02, 2007 10:26 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] IAX best practices

Thanks Steve!

What are usually the best approaches in troubleshooting the audio quality
issues and QoS related stuff when putting two Asterisk boxes together via
IAX?

Have you ever tried connecting Asterisk boxes in the same VPN (but still in
different countries)?

Regards,
Alex


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro
Sent: Thursday, March 01, 2007 4:22 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] IAX best practices

Asterisk wrote:
> Hi guys,
>
> I am planning to connect two Asterisk boxes that are currently running 
> in two different countries, using IAX.
>
> I was wondering if anyone could provide me with some links or
suggestion
> regarding best practices in connecting two Asterisk in such way. I
guess
> many of you have already tried this, and already have some know-how 
> (what I should be careful about, what to avoid, etc...)?
>
> Regards,
> Alex
>   
Bandwidth and latency.  IAX2 is remarakably good at traversing NAT and even
double NATs.  It should just work.  The issues that I ran into are low
bandwidth and latency.  Not much you can do about latency besides getting a
better route and putting QoS on your equipment and hoping that

your provider either observes your tagging or is not very latent to begin
with.  The other is bandwidth which I found SPEEX works wonders (but adds to
latency).

In my experience, bandwidth issues result in choppy audio and latency
results in delays which cause people to talk on top of each other and can be
extremely annoying.

Try pinging a router or device at the remote side to get an idea of how
latent your connection will be. 

Thanks,
Steve
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RE: [asterisk-users] Asterisk and Fax

[Michelle Dupuis]

For an all electronic solution, use fax2mail and mail2fax (from
www.generationd.com).  For a fancier all VOIP solution consider hylafax.

For analog only you can plug your fax machine in as you suggest.  For a step
up, buy an ATA with T.38 capability and plug your fax machine into that.

MD

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of --[ UxBoD ]--
Sent: Friday, March 02, 2007 10:34 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Asterisk and Fax

Hi,

I have a requirement for sending and receiving faxes and was wondering the
best way to achieve it with Asterisk as I only have one phone line.

I currently have a TDM11B in my server (1 x FXO, 1 x FXS), so I was thinking
that I would need to get a additional FXS module, connect that to a Eicon
Fax card, and then when receiving a call detect the fax tone and bridge the
call to the FXS channel.

Would that work okay ? Just seems fax cards are very expensive for what they
do.
--
--[ UxBoD ]--
// PGP Key: "curl -s http://www.splatnix.net/uxbod.asc | gpg --import"
// Fingerprint: 543A E778 7F2D 98F1 3E50 9C1F F190 93E0 E8E8 0CF8 //
Keyserver: www.keyserver.net Key-ID: 0xE8E80CF8 // SIP Phone:
[EMAIL PROTECTED]

--
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MailScanner, and is believed to be clean.

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[asterisk-users] Alec Saunders post about Mashable Telco's

[Dean Collins]

Interesting read in Alec Saunders blog today.
http://saunderslog.com/2007/03/01/mashable-telcos/

 

Thought it may interest some people on this list.

 

As food for thought, why it is that ITSP's haven't come up with more
'interesting' voice applications? When asterisk first became available I
thought it was the beginning of seeing really neat applications, think
Verzion's iobi but for everyone.

 

I mean I love Angel.com's (www.angel.com  ) 1800
speech enabled services on demand but I want it available via a SIP
gateway to hook up with my own phone numbers, and I cant wait for more
people to implement Iotum so when I try to make calls I wont continually
be hitting voicemails (www.iotum.com  ) but I
still think it is missing 'pieces' for widespread adoption.

 

Where are the really interesting asterisk applications for hosted
ITSP's?

 

Can anyone on this list name a Gee Wizz application they cant live
without? (or are we still just coding ex-girlfriend routines?) 

 

 

Regards,

Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
 +1-212-203-4357 Ph
+1-917-207-3420 Mb
+61-2-9016-5642 (Sydney in-dial).

 

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[asterisk-users] Asterisk and Fax

[--[ UxBoD ]--]

Hi,

I have a requirement for sending and receiving faxes and was wondering the best 
way to achieve it with Asterisk as I only have one phone line.

I currently have a TDM11B in my server (1 x FXO, 1 x FXS), so I was thinking 
that I would need to get a additional FXS module, connect that to a Eicon Fax 
card, and then when receiving a call detect the fax tone and bridge the call to 
the FXS channel.

Would that work okay ? Just seems fax cards are very expensive for what they do.
-- 
--[ UxBoD ]--
// PGP Key: "curl -s http://www.splatnix.net/uxbod.asc | gpg --import"
// Fingerprint: 543A E778 7F2D 98F1 3E50 9C1F F190 93E0 E8E8 0CF8
// Keyserver: www.keyserver.net Key-ID: 0xE8E80CF8
// SIP Phone: [EMAIL PROTECTED]

-- 
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and is
believed to be clean.

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RE: [asterisk-users] IAX best practices

[Asterisk]

Thanks Steve!

What are usually the best approaches in troubleshooting the audio
quality issues and QoS related stuff when putting two Asterisk boxes
together via IAX?

Have you ever tried connecting Asterisk boxes in the same VPN (but still
in different countries)?

Regards,
Alex


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve
Totaro
Sent: Thursday, March 01, 2007 4:22 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] IAX best practices

Asterisk wrote:
> Hi guys,
>
> I am planning to connect two Asterisk boxes that are currently running
> in two different countries, using IAX.
>
> I was wondering if anyone could provide me with some links or
suggestion
> regarding best practices in connecting two Asterisk in such way. I
guess
> many of you have already tried this, and already have some know-how
> (what I should be careful about, what to avoid, etc...)?
>
> Regards,
> Alex
>   
Bandwidth and latency.  IAX2 is remarakably good at traversing NAT and 
even double NATs.  It should just work.  The issues that I ran into are 
low bandwidth and latency.  Not much you can do about latency besides 
getting a better route and putting QoS on your equipment and hoping that

your provider either observes your tagging or is not very latent to 
begin with.  The other is bandwidth which I found SPEEX works wonders 
(but adds to latency).

In my experience, bandwidth issues result in choppy audio and latency 
results in delays which cause people to talk on top of each other and 
can be extremely annoying.

Try pinging a router or device at the remote side to get an idea of how 
latent your connection will be. 

Thanks,
Steve
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Re: [asterisk-users] cmd page crashes Asterisk SVN-branch-1.4-r57207

[Lee Jenkins]

Hall, Eric M. wrote:

Group

I’m having some trouble with asterisk and the page cmd.
Any help would be great!

This is what’s in my extensions.conf

exten => _**2,1,SIPAddHeader(Call-Info: answer-after=0)

exten => _**2,2,Page(SIP/36651)|d

exten => _**2,3,Hangup



Looks like you have at least a syntax error.  You have:

_**2,2,Page(SIP/36651)|d

And it should be

_**2,2,Page(SIP/36651|d)

Try fixing the "d" option by placing it within the right parenthesis and 
try it again.


--

Warm Regards,

Lee

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Re: [asterisk-users] Test

[C F]

Hi, I'm the admin on the list your test didin't work you should resend it.

















































































































Well I am not the admin, just wanted you to realized that you
shouldn't annoy thousands of people just because you don't want to
wait around till the next message arrives to see that you are really
subscribed.



On 3/1/07, Wai Wu <[EMAIL PROTECTED]> wrote:


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Re: [asterisk-users] No Caller ID Name PRI NI2

[C F]

On 3/2/07, Webster, Andrew <[EMAIL PROTECTED]> wrote:

>
> What do you mean by outbound CallerID Name? So that when calling a
> POTS with CallerID service from telco the Name should show up as you
> send it?
> If the answer to the above is yes, then stop trying to do that. It
> won't work, as the name that the POTS subscriber sees is NOT the one
> you send, but what the provider of that POTS line sees when the do the
> lookup on the name that is listed (usualy) with the number received as
> callerid.
>

Ah, that clarifies things a bit.  So I have to tell my telco what I want
to have show up for each callerID number that I send?


Yes you do, and with most telcos it's what you tell them should show
up for directory listing, and not all telcos will allow you to set
different directory listings for different DIDs.




> On 2/28/07, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
> > I there,
> >
> > I have some trouble to do working caller id name for outgoing calls
on
> > the PRI we just installed. Caller id name work on incoming calls
only.
> > Caller id number work on incoming and outgoing calls.
> >
> >
> > The provider, Goup Telecom, said that's in what i'm sending. They
said
> > that I send the cid name in ascii code and to do it working, I need
to
> > send it in hex.
> >
> > So I take some traces but i'm unable to figure where is the problem.
> >
> > What I see In case that work: incoming call:
> > < [1c 1c 9f 8b 01 00 a1 16 02 01 01 02 01 00 80 0e 49 4e 46 4f 46 4f
52
> > 54 49 4e 20 46 41 58]
> > < Facility (len=30, codeset=0) [ 0x9F, 0x8B, 0x01, 0x00, 0xA1, 0x16,
> > 0x02, 0x01, 0x01, 0x02, 0x01, 0x00, 0x80, 0x0E, 'INFOFORTIN FAX' ]
> > PROTOCOL 1F
> >
> > What I see in case that doesn't work: outgoing call:
> > > [28 05 b1 69 6e 66 6f]
> > > Display (len= 5) Charset: 31 [ info ]
> >
> >
> > completes traces:
> >
> > working:
> > < [ 02 01 da d6 08 02 02 34 05 04 03 80 90 a2 18 03 a9 83 81 1c 1c
9f
> > 8b 01 00 a1 16 02 01 01 02 01 00 80 0e 49 4e 46 4f 46 4f 52 54 49 4e
20
> > 46 41 58 1e 02 82 83 6c 0c 21 83 38 31 39 37 38 30 31 32 37 33 70 0b
a1
> > 38 31 39 33 34 30 30 39 37 37 ]
> > < Informational frame:
> > < SAPI: 00  C/R: 1 EA: 0
> > <  TEI: 000EA: 1
> > < N(S): 109   0: 0
> > < N(R): 107   P: 0
> > < 76 bytes of data
> > -- ACKing all packets from 106 to (but not including) 107
> > -- Since there was nothing left, stopping T200 counter
> > -- Stopping T203 counter since we got an ACK
> > -- Nothing left, starting T203 counter
> > < Protocol Discriminator: Q.931 (8)  len=76
> > < Call Ref: len= 2 (reference 564/0x234) (Originator)
> > < Message type: SETUP (5)
> > < [04 03 80 90 a2]
> > < Bearer Capability (len= 5) [ Ext: 1  Q.931 Std: 0  Info transfer
> > capability: Speech (0)
> > <  Ext: 1  Trans mode/rate: 64kbps,
> > circuit-mode (16)
> > <  Ext: 1  User information layer 1:
u-Law
> (34)
> > < [18 03 a9 83 81]
> > < Channel ID (len= 5) [ Ext: 1  IntID: Implicit  PRI  Spare: 0
> > Exclusive  Dchan: 0
> >  > <   Ext: 1  Coding: 0  Number Specified  Channel
> Type: 3
> > <   Ext: 1  Channel: 1 ]
> > < [1c 1c 9f 8b 01 00 a1 16 02 01 01 02 01 00 80 0e 49 4e 46 4f 46 4f
52
> > 54 49 4e 20 46 41 58]
> > < Facility (len=30, codeset=0) [ 0x9F, 0x8B, 0x01, 0x00, 0xA1, 0x16,
> > 0x02, 0x01, 0x01, 0x02, 0x01, 0x00, 0x80, 0x0E, 'INFOFORTIN FAX' ]
> > PROTOCOL 1F
> > 8B 0001 00 (CONTEXT SPECIFIC [11])
> > A1 0016 (CONTEXT SPECIFIC [1])
> >   02 0001 01 (INTEGER: 1)
> >   02 0001 00 (INTEGER: 0)
> >   80 000E 49 4E 46 4F 46 4F 52 54 49 4E 20 46 41 58 (CONTEXT
SPECIFIC
> [0])
> > < [1e 02 82 83]
> > < Progress Indicator (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard
> > (0)  0: 0  Location: Public network serving the local user (2)
> > <   Ext: 1  Progress Description:
Calling
> > equipment is non-ISDN. (3) ]
> > < [6c 0c 21 83 38 31 39 37 38 30 31 32 37 33]
> > < Calling Number (len=14) [ Ext: 0  TON: National Number (2)  NPI:
> > ISDN/Telephony Numbering Plan (E.164/E.163) (1)
> > <   Presentation: Presentation allowed of
> > network provided number (3)  '8197801273' ]
> > < [70 0b a1 38 31 39 33 34 30 30 39 37 37]
> > < Called Number (len=13) [ Ext: 1  TON: National Number (2)  NPI:
> > ISDN/Telephony Numbering Plan (E.164/E.163) (1)  '8193400977' ]
> > -- Making new call for cr 564
> > -- Processing Q.931 Call Setup
> > -- Processing IE 4 (cs0, Bearer Capability)
> > -- Processing IE 24 (cs0, Channel Identification)
> > -- Processing IE 28 (cs0, Facility)
> > Q.932 Interpretation component is not handled
> > Handle Q.932 ROSE Invoke component
> >   [ Handling operation 0 ]
> >   Handle Name display operation
> > Received caller name 'INFOFORTIN FAX'
> > -- Processing IE 30 (cs0, Progress Indicator)
> > -- Processing IE 108 (cs0, Calling Party Number)
> > -- Processing IE 112 (cs0, Called Par

Re: [asterisk-users] RE: Polycom reject button

[Doug Lytle]

Mike wrote:

Jason,

If you do test if JR's tip works, please share your finding with us.  I am
interested in this as well.
  


It'll work fine, the Polycom responds with BUSY when the DND button is 
pressed.  Using DIALSTATUS, it'll drop to voicemail and play the busy 
message if recorded if that's what you have it programmed to do. 


Doug



--

Ben Franklin quote:

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deserve neither Liberty nor Safety."


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[asterisk-users] Multiple simultaneous calls

[Stefano Totaro]

Quite surprising, yes! :-)
I am from north east Italy, now I live in Verona (Romeo and Juliet's city :).
I cannot do it connecting amp to the PBX. I have quite a long distance to cover 
and a network is already there.
My "phones" are have a quite smart processor so we may probably run the ices2 
that you are suggesting or something similar.
I will check the links that you sent me.

Thanks,
Stefano


>Stefano Totaro,


>
>Off topic.  I just noticed your name and was a little surprised!? ;-)   

>Are you in Italy / Sicily?


>
>Anyways, you can achieve overhead paging through a sound card hooked to 

>an Amp and speakers from your PBX.  I have yet to do it but have read 

>about it.  I think this may be the better solution for you unless you 

>are set on doing it over IP. 

>
>Check here for several options 

>http://www.voip-info.org/wiki/view/Asterisk+Paging+and+Intercom 

>
>I am pretty sure if you use a ring group or meetme, there is no way 

>around each phone having it's own stream. 

>
>Interestingly, 3Com systems do conferencing and paging through multicast 

>which is a nice idea but in practice can be a real pain to configure 

>network components to work properly (especially if you do not control 

>the network or you are trying to implement paging between remote 

>offices).  I have spent hours on this exact problem in the past.


>
>If it were me, I would probably not want all that traffic on the PBX 

>unless that is all that it will be doing or if you go the sound card 

>route.  I would use ices2 and let everyone stream from a different 

>server than the PBX. 


>
>Since you are using phones, I do not know that ices2 would work for you, 

>something must initiate the call.  I would probably have a second 

>Asterisk box to just handle the paging, setup an extension the dialplan 

>of the main PBX to dial the paging machine via SIP (and possibly include 

>Authenticate) that would drop the call into something like this:  

>http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+page 

>
>Thanks,

>Steve Totaro


>

>

>
>>stefano.totaro at transport.alstom.com wrote:

>>

>> Hello Steve,

>> thanks for your anwer.

>> Yes, you are right we want to do VoIP telephone system capable also of 

>> "public address" (overhead paging) service.

>> So synchronization is a key issue if we want to avoid unpleasant effects.

>> We are designing our phones and they will have also onboard amplifiers.

>> What I am trying to understand is whether we may use the phone system 

>> also for this service or if it is better

>> to go for a specific streaming technology (Ices2 is a good suggestion 

>> thanks).

>>

>> What happen if I put all the phones in a ring? Do they join the same 

>> multicast stream or a single stream for

>> each phone will be created?

>>

>> Thanks again.

>> Stefano

>>

>>

>> Inactive hide details for Steve Totaro Steve 

>> Totaro 

>>

>>

>>

>> 

>> 
>> *Steve Totaro *

>> Inviato da:

>> asterisk-users-bounces at lists.digium.com

>>

>>

>> Phone:

>> 01/03/2007 18.33

>> Per favore, rispondere a Asterisk Users

>> Mailing List - Non-Commercial Discussion

>>

>> 

>>

>> Per: Asterisk Users Mailing List - Non-Commercial Discussion 

>> 

>> Cc: (ccr: Stefano TOTARO/ITVRN01/Transport/ALSTOM)

>>Oggetto: Re: [asterisk-users] Multiple simultaneous calls

>>

>>
>>
>>stefano.totaro at transport.alstom.com wrote:

>>>

>>> Hi Guys,

>>> I am a novice of Asterisk and I need some experts help to understand

>>> what I can get out of it.

>>> I need to make multiple calls (let say 50) at once to autoanswering

>>> softphones on a LAN and send all of them the same message that they

>>> will repeat with loudspeakers in the same environment.

>>> I am a little concerned about synchronization of the phones and

>>> moreover it is not much clear to me if I have to open 50 connections

>>> and send 50 times the same packets or if can use in some way the

>>> multicast.

>>> Is there anybody that may give me some idea.

>>> Thanks in advance,

>>> Stefano

>>>

>> I suppose you could do that although, I am unclear on the auto-answering

>> softphone and the loudspeaker thing. Is this just for overhead paging

>> or something?

>>

>> You could put all the phones in a ring group with ringall and use the

>> computer's sound card to connect to an amplified speaker setup.

>>

>> You could also look at ices2 to stream audio or some other streaming

>> technology.

>>

>> Thanks,

>> Steve

>






 

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RE: [asterisk-users] RE: Polycom reject button

[Mike]

Jason,

If you do test if JR's tip works, please share your finding with us.  I am
interested in this as well.

Mike  

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of JR Richardson
Sent: Thursday, March 01, 2007 21:11
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] RE: Polycom reject button 

> I have users in my dialplan that go from SIP to Cell When they are at 
> their desk and they hit reject call, it goes to the next thing in the 
> dialplan, thus transferring to their cell.  Not what they want.  Is it 
> possible to change the reject button to make it go to voice mail or a 
> new ext?

I don't think so, only options for the polycoms is to move the buttons
around on the phone, not change what they actually do.  In my dialplan I
have the next priority after the dial cmd going to voicemail and the reject
button works as expected there.

Maybe use the dial-status variables to send a reject to voicemail and a
no-answer to send call to cell phone.  You would have to test, but I'm sure
the dial-status is different between a reject and a no-answer.

Good luck.

JR

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Re: [asterisk-users] Problem with TE212P

[Benito Camelas]

thanks Ioan

I've try this too but problem continues

I put :

span=1,1,0,ccs,hdb3,crc4
span=2,2,0,ccs,hdb3,crc4

span=1,1,0,ccs,hdb3,crc4
span=2,0,0,ccs,hdb3,crc4

span=1,0,0,ccs,hdb3,crc4
span=2,0,0,ccs,hdb3,crc4

In all this configurations the problem is the same
I'm desperate
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[asterisk-users] T38

[Khaled]

Dears 

Any one know how to let t38 works on asterisk 1.2 or an distribution like
trixbox have asterisk 1.4 

 

Regards

 

 

Khaled Chehab

System Integration Engineer

Xplorium Offshore.

Sakiet Al Janzir

Postal Code: 1102-2080

Tel: (961) 1- 868 686

Fax :(961) 1-808 810

GSM: (961) 3-979 343

 




*
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[asterisk-users] cmd page crashes Asterisk SVN-branch-1.4-r57207

[Hall, Eric M.]

Group

 I'm having some trouble with asterisk and the page cmd.

 

Any help would be great!

 

 

 

This is what's in my extensions.conf

 

exten => _**2,1,SIPAddHeader(Call-Info: answer-after=0)

exten => _**2,2,Page(SIP/36651)|d

exten => _**2,3,Hangup

 

 

CLI output



Connected to Asterisk SVN-branch-1.4-r57207 currently running on
VoIP-PBX (pid = 11317)

-- Remote UNIX connection

Verbosity is at least 3

 Extension Changed 36652 new state InUse for Notify User 36653

-- Executing [EMAIL PROTECTED]:1] SIPAddHeader("SIP/36652-b7d0c1f0",
"Call-Info: answer-after=0") in new stack

-- Executing [EMAIL PROTECTED]:2] Page("SIP/36652-b7d0c1f0", "SIP/36651")
in new stack

-- Called 36651

--  Playing 'beep' (language 'en')

 Extension Changed 36651 new state Ringing for Notify User 36653

-- SIP/36651-09eb3648 is ringing

-- SIP/36651-09eb3648 answered

 Extension Changed 36651 new state InUse for Notify User 36653

-- Created MeetMe conference 1023 for conference '10382980d'

[Mar  2 09:14:58] WARNING[11449]: channel.c:1686 ast_hangup: Hard hangup
called by thread 29141936 on SIP/36651-09eb3648, while fd is blocked by
thread 20036528 in procedure ast_waitfor_nandfds!  Expect a failure

  == Spawn extension (amaxx, **2, 2) exited non-zero on
'SIP/36652-b7d0c1f0'

 Extension Changed 36651 new state Idle for Notify User 36653

 Extension Changed 36652 new state Idle for Notify User 36653

VoIP-PBX*CLI> 

Disconnected from Asterisk server

 

 

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Re: [asterisk-users] Help Needed: Can't make "local" calls on a brand new PRI

[Andrew Latham]

Telco Switching could be waiting for a ten digit number.  I know that
Sprint and some others expect ten digit local calls


On 2/28/07, Mark Engelhardt <[EMAIL PROTECTED]> wrote:

Hello,

I just installed a PRI and when I make a local (seven digit) call, I
get Code 28 back from the telco, (I believe code 28 means "Invalid
Number") and I hear a fast busy on the phone.

Here is the output:
 -- Executing Dial("SIP/marke-17b1", "ZAP/G1/4967171") in new stack
 -- Requested transfer capability: 0x00 - SPEECH
 -- Called G1/4967171
 -- Zap/23-1 is proceeding passing it to SIP/marke-17b1
 -- PROGRESS with cause code 28 received
 -- Zap/23-1 is making progress passing it to SIP/marke-17b1

As you can see, asterisk is reporting 4967171 as the dialed number
(which is valid)

When I dial long distance, everything works fine.

Here is the output from long distance...

 -- Executing Set("SIP/marke-80f8", "CALLERID(all)=Small Dog
Electronics<8005116277>") in new stack
 -- Executing Dial("SIP/marke-80f8", "ZAP/G1/17077510895") in new
stack
 -- Requested transfer capability: 0x00 - SPEECH
 -- Called G1/17077510895
 -- Zap/23-1 is proceeding passing it to SIP/marke-80f8
 -- Zap/23-1 is ringing

If I just send the full 18024967171 to the telco, I get a voice from
the telco saying it is not necessary to dial 1 or the area code when
calling this number.

So the questions:  Is there anyway to further verify that asterisk is
not sending any extra digits or filler digits to the telco on the PRI?
If the problem is not in asterisk or zaptel, what do I say to the
Telco to get them to believe the problem is on their end?

We are running:
Asterisk 1.2.6
Zaptel 1.2.14
TE110P Card

Mark Engelhardt




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--
---
Andrew Latham - AKA: LATHAMA (lay-th-ham-eh)
[EMAIL PROTECTED] - [EMAIL PROTECTED]
If any of the above are down we have bigger problems than my email!
Hind sight is most always 20/20 or better.
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RE: [asterisk-users] BLF not working with Asterisk 1.4.0

[Steve Langstaff]

Looks like it's a problem already logged on Mantis:
http://bugs.digium.com/view.php?id=8800
 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Andrey Solovjov
> Sent: 02 March 2007 14:01
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] BLF not working with Asterisk 1.4.0
> 
> Hi
> The same is for me. BLF doesn't work with 1.4. I've added 
> notifyringing = yes and this doesn't help.
> Show hints doesn't show any status changes so asterisk 
> doesn't send any NOTIFY messages to grandstream. Message is 
> only sent when extension unregisters.
> Andrew.
> 
> Ricardo Carvalho:
> > Dear all,
> >
> > I've implemented BLF for use with some Grandstream GXP-2000 
> phones and 
> > it works fine in 1.2.x versions of Asterisk, although I 
> tested it with 
> > version 1.4.0 and it doesn't work! Has the needed syntax 
> changed for 
> > configure BLF for this version of Asterisk? It it a bug of this 
> > version? Or should it be misconfiguration that I'm doing?
> >
> > Thanks,
> > Ricardo.
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> >
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Re: [asterisk-users] BLF not working with Asterisk 1.4.0

[Andrey Solovjov]

Hi
The same is for me. BLF doesn't work with 1.4. I've added notifyringing 
= yes and this doesn't help.
Show hints doesn't show any status changes so asterisk doesn't send any 
NOTIFY messages to grandstream. Message is only sent when extension 
unregisters.

Andrew.

Ricardo Carvalho:

Dear all,

I've implemented BLF for use with some Grandstream GXP-2000 phones and 
it works fine in 1.2.x versions of Asterisk, although I tested it with 
version 1.4.0 and it doesn't work! Has the needed syntax changed for 
configure BLF for this version of Asterisk? It it a bug of this 
version? Or should it be misconfiguration that I'm doing?


Thanks,
Ricardo.
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Re: [asterisk-users] Cannot hear ringback music from telco

[Steve Totaro]

Vincent Tam wrote:

Hi Trevor,
Thanks for your suggestion, it works by adding a Answer() in between!
However it will make everycall in the CDR become "Answered".
Later on I found that setting progressinband=no in sip.conf finally 
fixed this problem!

Best Regards,
Vincent
So does that mean that the call is not answered yet as far as the telco 
goes?  If that is the case, how long can a call stay in this state?


I ask for inbound 800 billing.

Thanks
Steve
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Re: [asterisk-users] Re: Sending SMS

[Steve Totaro]

Tomislav Parcina wrote:

Supa wrote:

Try this:
http://www.bayhamsystems.com/asterisk.html

Works for me just fine, and it is very easy to get up and running, 
even with older version 1.2.3


I don't see a point of using providers as Bayhamsystems. First, it's 
unpractical to send SMS from phone. If I'm going to use web interface, 
then is better to use some provider that has web interface just for 
that (or maybe they will provide application to send messages to 
groups or in certain time).


Only reason why I would like to do it true Asterisk is if I could use 
my VoIP or E1 provider so that I get only one bill. But using 
Bayhamsystems that isn't a case. So, why people use such providers?


Do they let you specify what number the SMS is coming from or does it 
just come from one of their number pool?  Is it possible for the person 
reply to the SMS and have it come straight to my phone?


Thanks,
Steve
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Re: [asterisk-users] Multiple simultaneous calls

[Steve Totaro]

Stefano Totaro,

Off topic.  I just noticed your name and was a little surprised!? ;-)   
Are you in Italy / Sicily?


Anyways, you can achieve overhead paging through a sound card hooked to 
an Amp and speakers from your PBX.  I have yet to do it but have read 
about it.  I think this may be the better solution for you unless you 
are set on doing it over IP.


Check here for several options 
http://www.voip-info.org/wiki/view/Asterisk+Paging+and+Intercom


I am pretty sure if you use a ring group or meetme, there is no way 
around each phone having it's own stream. 

Interestingly, 3Com systems do conferencing and paging through multicast 
which is a nice idea but in practice can be a real pain to configure 
network components to work properly (especially if you do not control 
the network or you are trying to implement paging between remote 
offices).  I have spent hours on this exact problem in the past.


If it were me, I would probably not want all that traffic on the PBX 
unless that is all that it will be doing or if you go the sound card 
route.  I would use ices2 and let everyone stream from a different 
server than the PBX. 

Since you are using phones, I do not know that ices2 would work for you, 
something must initiate the call.  I would probably have a second 
Asterisk box to just handle the paging, setup an extension the dialplan 
of the main PBX to dial the paging machine via SIP (and possibly include 
Authenticate) that would drop the call into something like this:  
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+page


Thanks,
Steve Totaro



[EMAIL PROTECTED] wrote:


Hello Steve,
thanks for your anwer.
Yes, you are right we want to do VoIP telephone system capable also of 
"public address" (overhead paging) service.

So synchronization is a key issue if we want to avoid unpleasant effects.
We are designing our phones and they will have also onboard amplifiers.
What I am trying to understand is whether we may use the phone system 
also for this service or if it is better
to go for a specific streaming technology (Ices2 is a good suggestion 
thanks).


What happen if I put all the phones in a ring? Do they join the same 
multicast stream or a single stream for

each phone will be created?

Thanks again.
Stefano


Inactive hide details for Steve Totaro <[EMAIL PROTECTED]>Steve 
Totaro <[EMAIL PROTECTED]>






*Steve Totaro <[EMAIL PROTECTED]>*
Inviato da:
[EMAIL PROTECTED]


Phone:
01/03/2007 18.33
Per favore, rispondere a Asterisk Users
Mailing List - Non-Commercial Discussion



Per: Asterisk Users Mailing List - Non-Commercial Discussion 


Cc: (ccr: Stefano TOTARO/ITVRN01/Transport/ALSTOM)
Oggetto: Re: [asterisk-users] Multiple simultaneous calls



[EMAIL PROTECTED] wrote:
>
> Hi Guys,
> I am a novice of Asterisk and I need some experts help to understand
> what I can get out of it.
> I need to make multiple calls (let say 50) at once to autoanswering
> softphones on a LAN and send all of them the same message that they
> will repeat with loudspeakers in the same environment.
> I am a little concerned about synchronization of the phones and
> moreover it is not much clear to me if I have to open 50 connections
> and send 50 times the same packets or if can use in some way the
> multicast.
> Is there anybody that may give me some idea.
> Thanks in advance,
> Stefano
>
I suppose you could do that although, I am unclear on the auto-answering
softphone and the loudspeaker thing. Is this just for overhead paging
or something?

You could put all the phones in a ring group with ringall and use the
computer's sound card to connect to an amplified speaker setup.

You could also look at ices2 to stream audio or some other streaming
technology.

Thanks,
Steve


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Re: [asterisk-users] Running Fax on E1 line

[Doug Lytle]

younss azzayani wrote:

Hi everybody :)
Can i configure My E1 line te recive & send Fax?


Yes, via iaxmodem and HylaFAX+

http://iaxmodem.sourceforgenet
http://hylafax.sourceforge.net

Doug

--

Ben Franklin quote:

"Those who would give up Essential Liberty to purchase a little Temporary Safety, 
deserve neither Liberty nor Safety."


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