RE: [Asterisk-Users] RTP session between two end users

[Erdem HAKİ]

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling
aka ManxPower
Sent: Tuesday, June 28, 2005 6:19 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] RTP session between two end users

Erdem HAKİ wrote:

> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling
> aka ManxPower
> Sent: Monday, June 27, 2005 8:32 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] RTP session between two end users
> 
> Erdem HAKİ wrote:
> 
> 
>>Is it possible that a RTP session between two end users  (so i want to use
>>asterisk as a signaling proxy and bypass RTP sessions)?
>>
>> 
>>
>>I used "canreinvite=yes" but it didn't work. 
>>
>>
>>Description from asterisk conf. File;
>>
>>(canreinvite=yes; allow RTP voice traffic to bypass
>>Asterisk)
> 
> 
> 
> It's sip.conf.  reinvites only work if the codec is the same for the 
> two endpoints and Asterisk does NOT have to listen for DTMF (no t or T 
> on the dial line, no meetme, etc.)
> 
> ***
> We use same codec and don't use meetme etc...  So what else should i do?
>
>How are you determining if RTP audio is going thru Asterisk? 
>Remember, SIP signaling will always go thru Asterisk.

>Also do a "sip show channels" during a call to confirm that the codecs 
>are the same.

-- 
>Always do right. This will gratify some people and astonish the rest.
Mark Twain
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Hi,

I determine signaling with ethereal and i am sure that both sides use the
same codec.

By the way, i searched forum again and i read something below;

> In wiki pages it is stated that The audio channels (RTP) may go directly 
> from phone to phone or may go through Asterisk's media bridge.
> 
> Currently with my settings, I notice that all rtps are passing through
>  my asterisk. How could I achieve that they go directly from phone to
> phone?  I assume this way, my machine will have less load and therefore 
> could handle more calls.

As bkw pointed out, use canreinvite=yes for each sip phone definition.
But, that will only work if the phones can reach each other directly
(the phones and/or asterisk can't be behind a nat/firewall box).



Thanks 

Erdem HAKI [EMAIL PROTECTED]



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RE: [Asterisk-Users] Linksys WRT54GP2-NA settings for performanceandlow bandwidth?

[Marcel van Kaam, Fonetica]

You can set, in the linksys, the codec G729 for your line. In the Linksys
also set only to use that codec. This can be done at the admin page of the
line you use in the linksys. Also do that in the asterisk for your device. 
First buy the license from Digium.

Then you will use less bandwidth and have a better sound upstream. 

Marcel
 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paul Fielding
Sent: woensdag 29 juni 2005 1:24
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Linksys WRT54GP2-NA settings for
performanceandlow bandwidth?

Hmm... Except that if I bring my Vonage ATA for my Vonage line with me to 
the same hotel, I can get reliable connectivity.   Assuming the hotel isn't 
helping me on the QOS front, and the Hotel's connectivity is the last word, 
then my Vonage ATA should be choppy, as well, no?  This is what leads me to 
think I can do some tweaking

later,

Paul
- Original Message - 
From: "Greg Oliver" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Sent: Tuesday, June 28, 2005 2:17 PM
Subject: Re: [Asterisk-Users] Linksys WRT54GP2-NA settings for 
performanceand low bandwidth?


> Nothing you can do on this one..  Without the provider accepting your
> QoS settings, you are at their mercy.  And yes, you are correct, most
> multi-tenant dwellings use xDSL for their connectivity due to it's
> price, and the upstream is usually less bandwidth than the downstream..
>
> -Greg
>
> On Tue, 2005-06-28 at 13:00 -0600, Paul Fielding wrote:
>> So I'm using a WRT54GP2-NA when I travel, as I travel alot, to give me
>> a phone at my hotel rooms, etc.   During the day or late at night the
>> thing works great - best ATA I've ever used.
>>
>> However, in the mid-evening (when many business travellers are at the
>> hotel room doing work), the outgoing audio channel gets so choppy that
>> the person on the other end can't make me out clearly.
>> Interestingly, I can usually hear them just fine - I attribute that to
>> larger incoming bandwidth than outgoing on the hotel's part.
>>
>> This device has a *lot* of settings that one can tweak.   Anyone have
>> any suggestions on tuning this thing (or tuning Asterisk or both) to
>> improve the SIP performance of the audio from the Linksys to the
>> server to try to reduce choppiness?   I note that Vonage, who also
>> uses these devices, seems to have got it down - it doesn't seem to
>> matter where I use my Vonage Linksys device, I can get pretty
>> reasonable performance.   So I figure I should be able to do similar
>> tweaks to mine... *shrug*
>>
>> regards,
>>
>> Paul
>>
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[Asterisk-Users] Hop-On WIFI Phone MSRP $40

[Cory Andrews]

I have a lot of folks asking me about an auto-negotiating WLAN phone
supposedly being brought to market by Hop-On, which is touted to carry an
MSRP of $40  Press photos (stock art) of the device shows it looks almost
identical to devices from Zyxel and UTStarCom.

I am trying to explain to folks there is no way in hell you are going to be
able to buy these phones, hardware only, for $40  Hop-On's press releases
are somewhat ambiguous, but at this price point, they would have to bundle
the phone with a length service contract in order to subsidize the hardware
cost on the phone.

Anyone have any inside info regarding Hop-On?

Cory Andrews
Partner / Purchasing
VOIPSupply.com
++
454 Sonwil Drive
Buffalo, NY 14225
++
v - 800.398.VOIP Ext 22
f - 716.630.1548
e - [EMAIL PROTECTED]


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Fw: [Asterisk-Users] Shoutcast Music On Hold problems?

[hank]

- Original Message - 
From: "hank" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 


Sent: Tuesday, June 28, 2005 10:52 PM
Subject: Re: [Asterisk-Users] Shoutcast Music On Hold problems?



I am using [EMAIL PROTECTED] 1.0
my mp3 is called
mp3
it has nothing before it
it is 0 bytes
does my mp3 of 0 bytes need to have a .mp3 or does it need to be called 
anything?

thanks
hank

- Original Message - 
From: "Huddleston, Robert" <[EMAIL PROTECTED]>
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" 


Sent: Tuesday, June 28, 2005 11:52 AM
Subject: RE: [Asterisk-Users] Shoutcast Music On Hold problems?


Worked for me with a different stream... I ran into this same problem 
before - but it was my own fault for not RTM... Both the manual and ast 
install advised of verifying correct version of mpg123... I had wrong 
version and thus got no noise...
If you follow the directions explicitly laid out on the wiki you should 
have no problems.

I use christianrock.net's shoutcast stream
Like this in musiconhold.conf
default => 
quietmp3:/var/lib/asterisk/mohmp3-empty,http://209.51.128.160:5112/




-Original Message-
From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of hank

Sent: Tuesday, June 28, 2005 1:41 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Shoutcast Music On Hold problems?

I tried that the stream i tried to use orriginally was
http://209.97.198.50:30518
all I get is silence when I put the person on hold thanks hank
- Original Message -
From: "Patrick" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"

Sent: Tuesday, June 28, 2005 2:50 AM
Subject: Re: [Asterisk-Users] Shoutcast Music On Hold problems?



On Mon, 2005-06-27 at 22:51 -0700, hank wrote:

 mp3:/var/lib/asterisk/mohmp3-empty,http://www.waixwave.com:8000/


I haven't tried this myself but if I put www.waixwave.com:8000 in
Firefox I get "connection refused". Try another site that actually
streams music. Shoutcast.org should have a nice list.

Regards,
Patrick

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[Asterisk-Users] audiocodes

[Joe Murray]

Is anyone on this list using and audiocodes FXO gateway? I have  
Asterisk(1.07 on OS X) setup and working fine, including SIP phones  
and IAX2 phones - I can make outbound calls just fine and receive  
inbound calls just fine. However, I can't seem to find the right  
series of DTMF settings on the AudioCodes to allow DTMF tones to be  
sent after an outbound call is connected(phone banking, long distance  
provider etc...) while still allow the client devices(phones) to  
access Asterisk voicemail. It seems I can either have the phones use  
inband DTMF and work with the Audiocodes PSTN's or outband and work  
with Asterisk, but not both? Any help/thoughts/experiences would be  
appreciated...


-joe
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Re: [Asterisk-Users] ASTCC not billing

[Juan Luis Moyano]

Also while looking at the cards table I noticed that the creation field
on a card is always changing to the date I made the last call from that
card. Is this the way it has to be?

-- 
Juan Luis Moyano
[EMAIL PROTECTED]

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RE: [Asterisk-Users] AMP/A@H (asterisk at home) custom incoming routing

[David Phelan]

You would be better using extensions_custom only because of the fact that
when you restart ampportal, it will overwrite extensions_additional with
what ever it has stored in the Database.


 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tom Rymes
Sent: Wednesday, 29 June 2005 2:47 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] AMP/[EMAIL PROTECTED] (asterisk at home) custom 
incoming routing

Folks,

First off, this is messy, and I hope someone will be kind enough to help me
clean this up (the part added to extensions_additional.conf).  
You've been warned!

For those of your using AMP or [EMAIL PROTECTED], there has been a lot of talk 
about how
to route incoming calls to different places based on which trunk is ringing.
The standard answer is that you can only do this by using DIDs, which is all
fine and good, unless you are using a plain old copper line that doesn't
support DID. Anyhow, I have figured out how to make a call that comes in on
a specific ZAP channel ring at a specific extension (not that it was brain
surgery). I'm not certain if it would be better to use the file
extensions_custom.conf instead of extensions_additional.conf, does anyone
know?

I have an [EMAIL PROTECTED] box with an unused TDM11P card in it at home in my 
basement.
The [EMAIL PROTECTED] box normally handles incoming calls for my small 
business, but I
wanted to plug my home phone line into the FXO port, and all of my phones
into the FXS port (They're cordless, so no worries about ringer equivalence,
etc.). That way I can route outgoing calls over VOIP, but my incoming calls
will still ring my home phones. The hitch with [EMAIL PROTECTED] was that AMP 
doesn't
allow you to differentiate between incoming calls from Broadvoice (or
wherever) and incoming calls on the FXO port. I wanted incoming calls on my
home line to ring to ring the extension associated with the FXS port on my
TDM card. All other incoming calls should still follow whatever I set up in
AMP, since that is how I control where my incoming business calls go.

Here's what I did. I know that you aren't supposed to use the extension "_."
but that is the only way I got it to work. Please let me know if there is a
better way.

1.) Edit /etc/asterisk/zapata-channels.conf and change the context for your
incoming port to something new. I used "tdm-in".
2.) Edit pico /etc/asterisk/extensions_additional.conf and add this at the
bottom:

[tdm-in]
exten => _.,1,Goto(ext-local,200,1);
3.) If you haven't already, add the ZAP channel as a trunk in AMP so you can
make outgoing calls on this channel.

That's it,

Tom
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Version: 7.0.323 / Virus Database: 267.8.6/33 - Release Date: 28/06/2005
 

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Re: [Asterisk-Users] Re: ERROR[22927]: Failed to create socketpair for player(24, Too many open files).

[Wai-Sun Chia]

On 6/29/05, Yap Teong Eng <[EMAIL PROTECTED]> wrote:
> Hi All,
> 
> I am running * v1.0.6 on a P4 2.8 Ghz server with 512 Mb.
> 
> My * is configured to handle h323 calls using inaccessnetworks OH323.
> 
> I have a problem of * terminating with this error after a few days.
> 
> ERROR[22927]: Failed to create socketpair for player(24, Too many open
> files).

Thi s may not be the root cause but the symptom is that you're running
out of  file descriptors (FD). This typically caused by a rogue
application which didn't close all of the files which it opened
causing it to leak FDs.

p.s. Greetings from a fellow Malaysian. :-)
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Re: [Asterisk-Users] ASTCC not billing

[Juan Luis Moyano]

Has anyone noticed that the primary key in the cdrs table is cardnum? so
it won't record more than the first call made by different cards.
Perhaps I'm not understanding the purpose of de cdrs table. Maybe one
solution is to add an auto_increment uniqueid field like in the
asteriskcdrdb cdr table. Can anyone point me in the right direction on
this one?

-- 
Juan Luis Moyano
[EMAIL PROTECTED]

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[Asterisk-Users] AMP/A@H (asterisk at home) custom incoming routing

[Tom Rymes]

Folks,

First off, this is messy, and I hope someone will be kind enough to  
help me clean this up (the part added to extensions_additional.conf).  
You've been warned!


For those of your using AMP or [EMAIL PROTECTED], there has been a lot of talk  
about how to route incoming calls to different places based on which  
trunk is ringing. The standard answer is that you can only do this by  
using DIDs, which is all fine and good, unless you are using a plain  
old copper line that doesn't support DID. Anyhow, I have figured out  
how to make a call that comes in on a specific ZAP channel ring at a  
specific extension (not that it was brain surgery). I'm not certain  
if it would be better to use the file extensions_custom.conf instead  
of extensions_additional.conf, does anyone know?


I have an [EMAIL PROTECTED] box with an unused TDM11P card in it at home in my  
basement. The [EMAIL PROTECTED] box normally handles incoming calls for my small  
business, but I wanted to plug my home phone line into the FXO port,  
and all of my phones into the FXS port (They're cordless, so no  
worries about ringer equivalence, etc.). That way I can route  
outgoing calls over VOIP, but my incoming calls will still ring my  
home phones. The hitch with [EMAIL PROTECTED] was that AMP doesn't allow you to  
differentiate between incoming calls from Broadvoice (or wherever)  
and incoming calls on the FXO port. I wanted incoming calls on my  
home line to ring to ring the extension associated with the FXS port  
on my TDM card. All other incoming calls should still follow whatever  
I set up in AMP, since that is how I control where my incoming  
business calls go.


Here's what I did. I know that you aren't supposed to use the  
extension "_." but that is the only way I got it to work. Please let  
me know if there is a better way.


1.) Edit /etc/asterisk/zapata-channels.conf and change the context  
for your incoming port to something new. I used "tdm-in".
2.) Edit pico /etc/asterisk/extensions_additional.conf and add this  
at the bottom:


[tdm-in]
exten => _.,1,Goto(ext-local,200,1);
3.) If you haven't already, add the ZAP channel as a trunk in AMP so  
you can make outgoing calls on this channel.


That's it,

Tom
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[Asterisk-Users] H323 on ovislink

[[EMAIL PROTECTED]]

Hi,
I have this:

  FXO  h323
Ext1 --> PBX Panasonic -->Ovislink --> Asterisk ---> ATA

But how register ovislink in h323.conf?

i put this h323.conf

[mypeer1]
type=peer
context=context2
ip=192.168.5.240   ; IP Ovislink
port=1720
outgoinglimit=4
e164=101

i put this extensions.conf

exten => 101, 1, Dial(H323/mypeer1)
exten => 101, 2, Hangup


The problem is, when i dial 101 nobody answer.. 


Thanks

alguien de Jalisco México para hacer contacto..??


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Re: [Asterisk-Users] ASTCC not billing

[Juan Luis Moyano]

Bernard Cresencia wrote:
> sorry, I meant my.cnf, not my.conf.
>
> Once logging is enabled, I would do tail -f
> /var/log/myslqd.log and watch as the database is being
> accessed during a call.


I've done what Bernard suggested and this is my output from mysql.log on
a successful call to number 612 on FWD. I'd like to know if any of you
see something wrong or rare. Thanks a lot.

Time Id CommandArgument
050629  1:02:02   1 Connect [EMAIL PROTECTED] on astcc
050629  1:02:04   1 Query   SELECT * FROM cards WHERE number='21'
  1 Query   SELECT * FROM cards WHERE number='21'
  1 Query   SELECT * FROM cards WHERE number='21'
  1 Query   SELECT * FROM cards WHERE number='21'
  1 Query   UPDATE cards SET used='1801' WHERE
number='21'
  1 Query   UPDATE cards SET inuse='1' WHERE
number='21'
050629  1:02:10   1 Query   SELECT * FROM routes WHERE '612'
RLIKE pattern ORDER BY LENGTH(pattern) DESC
050629  1:02:25   1 Query   SELECT * FROM cards WHERE number='21'
  1 Query   SELECT * FROM trunks WHERE name='FWD'
050629  1:02:37   1 Query   INSERT INTO cdrs
(cardnum,callerid,callednum,trunk,disposition,billseconds,billcost,callstart)
VALUES ('21', '\"Coco\" <21>', '612', 'FWD', 'ANSWER', '9', '150', 'Wed
Jun 29 01:02:37 2005')
  1 Query   UPDATE cards SET used='1951' WHERE
number='21'
  1 Query   UPDATE cards SET inuse='0' WHERE
number='21'
  1 Query   SELECT * FROM cards WHERE number='21'
  1 Query   UPDATE cards SET used='1951' WHERE
number='21'
  1 Query   UPDATE cards SET inuse='0' WHERE
number='21'
  1 Quit

-- 
Juan Luis Moyano
[EMAIL PROTECTED]

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Re: [Asterisk-Users] FXO/FXS cpu spikes, data loss and ztclock.

[qrss]

>Probably you are only looking for results of FXO cards, but maybe it's
>usefull:

Any results that folks can supply are useful. Thankyou for giving ztclock
a try. This information is certainly helpful.

>483328 samples in 60.416004 sec. (483329 sample intervals) 99.999794%
>483328 samples in 60.416000 sec. (483328 sample intervals) 100.00%
>483328 samples in 60.416004 sec. (483329 sample intervals) 99.999794%
>Estimate 8 frame slips every 483.328003 seconds.

I assume that your system is clocking from the BRI rather than from the
TDM card. Is that correct?  It looks to me like you have a perfect clock
source that is 100% accurate. I would guess that the slight deviation of 4
microseconds in two of the results is likely accounted for as a percentage
of error in the test itself. In reality, I'm sure that you are not seeing
any slips at all.  I'll be compiling any results that I receive in order
to try to improve the accuracy of the test.

>
>No CPU spikes afaics.
>
>Card:
>Module 0: Installed -- AUTO FXS/DPO
>Module 1: Installed -- AUTO FXS/DPO
>Module 2: Installed -- AUTO FXS/DPO
>Module 3: Installed -- AUTO FXS/DPO
>Found a Wildcard TDM: Wildcard TDM400P REV E/F (4 modules)

>I run a fax on it, this works. But this is FXS. I have ISDN lines
>outgoing with the junghanns quadBRI.

Do you have fax machines connected to the FXS ports, or are you running
something like spandsp?


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Re: [Asterisk-Users] Asterisk rss list feeder ready

[Frank Mayhar]

On Wed, 2005-06-29 at 00:45 +0200, Sjaak Nabuurs wrote:
> http://asterisk.voipexco.com

It's a nice idea but not very useful (at least for me) as the RSS feed
only has the subject and the first few words.  I would need the whole
article...
-- 
Frank Mayhar [EMAIL PROTECTED] http://www.exit.com/
Exit Consulting http://www.gpsclock.com/
http://www.exit.com/blog/frank/

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[Asterisk-Users] Anyone using SipP to produce RTP load?

[Matthew Boehm]

Hey gang,
 I've been able to use sipp to produce some call volume on our asterisk
server. The server has no problems handling 50 simul calls. But then again,
no RTP is being done. I tried to use the rtp echo ability of sipp but that
doesn't seem to work right.
 I also setup a fake number in asterisk that when called by sipp, would dial
another number via PRI, hoping that some 729 conversion would occur.
Nothing. I was able to pump 10 simul calls that went this path:

  sipp -> asterisk -> pri -> telco ->pri ->asterisk

..and still no 729 usage or any other discernable load on the server.

Can anyone offer suggestion on how to really simulate calls (using sipp or
other tester) to asterisk to verify its ability to process X calls?

I know someone out there has done this, but forget who it was.

Thanks,
Matthew


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[Asterisk-Users] Re: [Asterisk-Dev] ClueCon, Vote?

[Greg Boehnlein]

On Tue, 28 Jun 2005, Brian West wrote:

> Ok I have to get a vote of all the people that are going to come to  
> Cluecon so we order the beer keg's for the developers board room.
> 
> Anyone have any preference? (if you haven't registered for ClueCon  
> now is the time to register!)
> 
> Choices... choices... choices... I want Red Bull on tap!

I would suggest that you investigate Great Lakes Brewing 
(http://www.greatlakesbrewing.com), specifically their Dortmunder Gold. It 
is the only beer that I drink when given the choice. A second, cheaper 
option is Killians (AKA Coors).

Dortmunder Gold Lager

ABV:  5.8% 
ABW:  4.3% 
IBU:  30 

TYPE/STYLE
Dortmunder: During the mid-19th century, seven breweries within the city 
of Dortmund, Germany began brewing beers in the same manner, resulting in 
what has come to be known as the "Dortmunder" style. 

FOOD COMPLEMENTS
Because neither malt nor hops dominate this beer, it complements most 
foods, especially salads, fish and chicken. 

BRAND NAME
The name of our flagship beer reflects the unprecedented accolades and 
recognition it has earned in major worldwide beer tasting competitions. 

AWARDS
Silver Medal, 2004 World Beer Championships

Gold Medal, 2002 World Beer Championships

Gold Medal, 2001 World Beer Championships

Gold Medal, 2000 World Beer Championships

Gold Medal, 1999 World Beer Championships

Gold Medal, 1998 World Beer Championships

World Champion, Gold Medal, 1997 World Beer Championships

World Champion, Gold Medal, 1996 World Beer Championships

World Champion, Gold Medal, 1995 World Beer Championships

World Champion, Gold Medal, 1994 World Beer Championships

Gold Medal, 1990 Great American Beer Festival

-- 
Vice President of N2Net, a New Age Consulting Service, Inc. Company
 http://www.n2net.net Where everything clicks into place!
 KP-216-121-ST



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Re: [Asterisk-Users] Asterisk server with remote monitoring capabilities

[Marie]

I've yet to run into a Co-Lo facility that didn't offer a reboot
service. Yes, some charged me for it, but it always seemed to be an
option.

Check your machine before purchasing the remote control power strips.
I bought one for a troublesome server that was not capable of
automatically repowering itself. Made for a rather expensive "D'oh".

Most of my Asterisk boxes are running full Linux installs (gui, etc --
not stripped down -- vnc, webserver, and Oracle one) and go without
trouble for months and months -- the only time I've had to reboot them
is when I do something stupid.

For what the remote power strips cost, you're better off determine
what the problem is to begin with and investing in new hardware (if
necessary). Any of these solutions worth having are not cheap. It's
not going to help you much if you buy a crap remote power solution
that ends up needing to be monitored itself.
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[Asterisk-Users] Re: ERROR[22927]: Failed to create socketpair for player(24, Too many open files).

[Yap Teong Eng]

Hi All,

I am running * v1.0.6 on a P4 2.8 Ghz server with 512 Mb.

My * is configured to handle h323 calls using inaccessnetworks OH323.

I have a problem of * terminating with this error after a few days.

ERROR[22927]: Failed to create socketpair for player(24, Too many open 
files).


I have searched google for the above error message but couldn't find any and 
I would
appreciate any help rendered. Is the "player" in the message referring to 
the moh player ?
I am currently not using moh. Below is the musiconholf.conf file. I have 
hashed out everything.

;
; Music on hold class definitions
;
[classes]
;default => quietmp3:/var/lib/asterisk/mohmp3
;loud => mp3:/var/lib/asterisk/mohmp3
;random => quietmp3:/var/lib/asterisk/mohmp3,-z
;unbuffered => mp3nb:/var/lib/asterisk/mohmp3
;quietunbuf => quietmp3nb:/var/lib/asterisk/mohmp3
; Note that the custom mode cannot handle escaped parameters (specifically 
embedded spaces)
;manual => custom:/var/lib/asterisk/mohmp3,/usr/bin/mpg123 -q -r 8000 -f 
8192 -b 2048 --mono -s


Thank you.

Warmest Regards
T.E.Yap 


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Re: [Asterisk-Users] How do you handle NAT?

[Eric Wieling aka ManxPower]

Sebastian Silva wrote:


Hi everyone.

1.  Asterisk as a SIP client behind nat, connecting to outside SIP Proxies:
#1 works with a NAT-supporting proxy as SIP Express router as the 
outside proxy. (Get an account at IPtel.org and try!). Fails with Free 
World Dialup.


This will also work without any outside proxy by using externip= and 
localnet= in sip.conf, as well as doing the correct port forwarding on 
your NAT router.



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Mark Twain
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[Asterisk-Users] Red Hat Enterprise 3.0 issue

[Federico Alves]

I use this code:
cd /usr/src/asterisk
make config
but the automatic startup for Red Hat does not work.
My Red Hat is the 3.0 update 4. Has anybody made this work in the licensed
version of Red Hat?
It says "Asterisk ended with exit status 0" "Asterisk shutdown normally"
Any ideas? 
Federico Alves

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Re: [Asterisk-Users] Revision I Board TDM04b

[Rich Adamson]

> I cannot get this thing to work.  Anyone know of any tricks?

Call digium support; its free.


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[Asterisk-Users] H323 on ovislink

[[EMAIL PROTECTED]]

Hi,
I have this:

  FXO  h323
Ext1 --> PBX Panasonic -->Ovislink --> Asterisk ---> ATA

But how register ovislink in h323.conf?

i put this h323.conf

[mypeer1]
type=peer
context=context2
ip=192.168.5.240   ; IP Ovislink
port=1720
outgoinglimit=4
e164=101

i put this extensions.conf

exten => 101, 1, Dial(H323/mypeer1)
exten => 101, 2, Hangup


The problem is, when i dial 101 nobody answer.. 


Thanks

alguien de Jalisco México para hacer contacto..??


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[Asterisk-Users] manager api call number, pause, dial ext

[Thomas Miller]

Hi,
I have a program that sends commands to the Manager
API and places outgoing calls. 

I want to make it call a number, then pause, and dial
an office extension. Is this possible?  How?

 "Action: Originate\r\nChannel: " +
asteriskVoipChannel + "/" + PhoneNumber +
"\r\nContext: " + context + "\r\nCallerID: " +
callerId + "\r\nExtension: " + context +
"\r\nPriority: 1\r\nActionId: 1\r\n\r\nAction:
Logoff\r\n\r\n";


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RE: [Asterisk-Users] Livevoip 800 Choppy Audio

[Rich Adamson]

Yes, I've used them in the past. The issue is that you sent no less
then a dozen posts (more like two or three dozen if you count the
last few months) blasting a vendor. No problem expressing an opinion
at all, but not very professional (for anyone) to keep blasting away
forever. We all got your point a month or two ago. Lets move on.


> You must be a LiveVoip crony.  Regardless, there is nothing wrong with
> my post.  Expressing that we should all learn from their mistakes is
> hardly an objective.  So whatever your issue, I think you can take your
> own advise.

> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Rich
> Adamson
> Sent: Monday, June 27, 2005 7:31 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] Livevoip 800 Choppy Audio
> 
> Wiley, its about time to grow up and stop this personal BS. We all
> certainly
> understand your objective.
> 
> 
>   From: Wiley Siler <[EMAIL PROTECTED]>
>   Subject: RE: [Asterisk-Users] Livevoip 800 Choppy Audio
>   Date: Mon, 27 Jun 2005 08:30:41 -0700 
>   To: Asterisk Users Mailing List - Non-Commercial Discussion
> 
> 
> 
> > LiveVoip has been a learning experience for anyone who purchased from
> > them.  With any luck, it was a learning experience in "what not to do"
> > for anyone out there that provides similar services.  At least I hope
> so
> > since it seems obvious that LiveVoip never learned a thing during
> their
> > interaction with the community.  That is a real shame too considering
> > how people will embrace a company.
> > 
> > I like someone else's post "survival of the fittest".
> > Your damn skippy...
> > 
> > W
> > 
> > 
> > 
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] On Behalf Of John
> > Kington
> > Sent: Sunday, June 26, 2005 7:54 AM
> > To: asterisk-users@lists.digium.com
> > Subject: RE: [Asterisk-Users] Livevoip 800 Choppy Audio
> > 
> > Wiley, thanks for pointing me to NuFone for tollfree DID. I was
> planning
> > to 
> > report
> > on results between LiveVOIP and NuFone. The apparent bankruptsy of
> > LiveVOIP
> > means that my choppy audio will probably never be resolved. I set up
> > both 
> > DID to
> > go through DISA and I could then use the echo test application.
> > Everytime I 
> > tested
> > LiveVOIP, the audio was choppy. I have not experienced any choppiness
> > with
> > NuFone but the echo seemed to take longer to get back to me compared
> to
> > LiveVOIP.
> > I now get a message that my call can not be completed when I call the
> > LiveVOIP
> > DID and I see that I can not register my asterisk to them. I am glad I
> > did 
> > not have
> > big dollars invested in them.
> > Regards,
> > John
> > 
> > 
> > ___
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> > http://lists.digium.com/mailman/listinfo/asterisk-users
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> > ___
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> >http://lists.digium.com/mailman/listinfo/asterisk-users
> > 
> 
> ---End of Original Message-
> 
> 
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---End of Original Message-


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[Asterisk-Users] ASDI Programming through an ATA/SIP device?

[C. Hatton Humphrey]

As I've said before, I have a number of phones that I've converted
from Packet8's Virtual Office to work with my Asterisk stetup.  They
are basically Leadtek 85XX SIP devices with Astra 390 phones
connected.

I've been investigating reprogramming the Astras to say something
other than "Virtual Office powered by Packet8" but when I try ADSIProg
I get something about CPE not being available.

I'm afraid to say but I think that I'm going to have to get a Zaptel
device to get the phones programmed (a single port that I can plug the
phones into and then have them call the programming extension, maybe)
unless there's something that I'm missing --- can I program an ADSI
device using a SIP interface?

Thanks!
Hatto n
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Re: [Asterisk-Users] Linksys WRT54GP2-NA settings for performanceand low bandwidth?

[Paul Fielding]

Hmm... Except that if I bring my Vonage ATA for my Vonage line with me to 
the same hotel, I can get reliable connectivity.   Assuming the hotel isn't 
helping me on the QOS front, and the Hotel's connectivity is the last word, 
then my Vonage ATA should be choppy, as well, no?  This is what leads me to 
think I can do some tweaking


later,

Paul
- Original Message - 
From: "Greg Oliver" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 


Sent: Tuesday, June 28, 2005 2:17 PM
Subject: Re: [Asterisk-Users] Linksys WRT54GP2-NA settings for 
performanceand low bandwidth?




Nothing you can do on this one..  Without the provider accepting your
QoS settings, you are at their mercy.  And yes, you are correct, most
multi-tenant dwellings use xDSL for their connectivity due to it's
price, and the upstream is usually less bandwidth than the downstream..

-Greg

On Tue, 2005-06-28 at 13:00 -0600, Paul Fielding wrote:

So I'm using a WRT54GP2-NA when I travel, as I travel alot, to give me
a phone at my hotel rooms, etc.   During the day or late at night the
thing works great - best ATA I've ever used.

However, in the mid-evening (when many business travellers are at the
hotel room doing work), the outgoing audio channel gets so choppy that
the person on the other end can't make me out clearly.
Interestingly, I can usually hear them just fine - I attribute that to
larger incoming bandwidth than outgoing on the hotel's part.

This device has a *lot* of settings that one can tweak.   Anyone have
any suggestions on tuning this thing (or tuning Asterisk or both) to
improve the SIP performance of the audio from the Linksys to the
server to try to reduce choppiness?   I note that Vonage, who also
uses these devices, seems to have got it down - it doesn't seem to
matter where I use my Vonage Linksys device, I can get pretty
reasonable performance.   So I figure I should be able to do similar
tweaks to mine... *shrug*

regards,

Paul

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Re: [Asterisk-Users] Unable to connect to remote asterisk

[beonice]

Jason, when I've seen this, it's usually meant that I
was logged in as a user who didn't have the right
permissions. :)

Check your notes to figure out which user you gave the
permission to (root, maybe?) and log in as that user,
then try to connect using "asterisk -r".

Hope that helps,
Maya


--- Jason Greene <[EMAIL PROTECTED]> wrote:

> Hello,
>   I'm trying to figure out why the asterisk service
> starts fine, but 
> when i try to connect by typing asterisk -r I get:
> Unable to connect to remote asterisk
> 
> The service is running and lists under ps -ef as:
> asterisk -vvvg -c
> 
> any help is appreciated
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Re: [Asterisk-Users] How do you handle NAT?

[Matthew Boehm]

Where is "Asterisk as a SIP server on the outside, SIP clients behind NAT
connecting to Asterisk" ?

Our Asterisk server is on public IP, no NAT. Right now we have 4 separate
customers, all of which are on T1s with some sort of firewall/NAT.

We have access to all 4 FW/NAT's. Would allowing full, unrestricted access
to our Asterisk's IP remove the need for qualify ?

-Matthew


> From: hank <[EMAIL PROTECTED]>
> Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
> 
> Date: Tue, 28 Jun 2005 15:46:29 -0700
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> 
> Subject: Re: [Asterisk-Users] How do you handle NAT?
> 
> I think my problem is numbrer 3 cause basicly my friend who is not on my
> router is trying to get connected to me but can't and I am the 1 that is
> behind a nat.
> thanks
> hank
> - Original Message -
> From: "Sebastian Silva" <[EMAIL PROTECTED]>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> 
> Sent: Tuesday, June 28, 2005 12:45 PM
> Subject: Re: [Asterisk-Users] How do you handle NAT?
> 
> 
>> Hi everyone.
>> 
>> 1.  Asterisk as a SIP client behind nat, connecting to outside SIP
>> Proxies:
>> #1 works with a NAT-supporting proxy as SIP Express router as the outside
>> proxy. (Get an account at IPtel.org and try!). Fails with Free World
>> Dialup.
>> 
>> 2. Asterisk as a SIP client behind nat, connecting to inside SIP proxies:
>> #2 Works- no NAT in between
>> 
>> 3. Asterisk as a SIP server behind nat, clients on the outside connecting
>> to Asterisk:
>> #3 Works with port forwarding and some header mangling magic
>> 
>> 4. Asterisk as a SIP server behind nat, clients on the inside connecting
>> to Asterisk:
>> #4 Works - no NAT in between
>> 
>> 5. Asterisk as a SIP client outside nat, connecting to outside SIP
>> proxies:
>> #5 is no problem. No NAT in the middle
>> 
>> 6. Asterisk as a SIP client outside nat, connecting to inside SIP proxies:
>> #6 is a problem if no port forwarding is done, similar to 3 above.
>> 
>> 7. Asterisk as a SIP server outside nat, clients on the outside connecting
>> to Asterisk:
>> #7 is no problem. No NAT in the middle
>> 
>> 8. Asterisk as a SIP server outside nat, clients on the inside connecting
>> to Asterisk:
>> #8 is solved with nat=yes and qualify=xxx in sip.conf for the client in
>> most cases. Some clients (X-lite) assist themselves by using STUN and
>> sending UDP keep-alive packets. Qualify sends keep-alive packets from
>> Asterisk to the client on the inside.
>> 
>> from wiki
>> 
>> Now, if you net to define a NAT, you have to set asterisk to
>> "canreinvite=no", "qualify=yes" and "nat=1".
>> 
>> Also, INSTEAD of NAT, you can use a STUN server. To use a STUN server you
>> should set asterisk to "canreinvite=no", "qualify=no" and "nat=0" (the
>> STUN configuration is in your agents).
>> 
>> Sebas
>> 
>> hank wrote:
>>> how easy is it to set up a stun server? with asterisk amd will this fix
>>> part of the nat problem?
>>> - Original Message - From: "Ray Van Dolson"
>>> <[EMAIL PROTECTED]>
>>> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>>> 
>>> Sent: Tuesday, June 28, 2005 8:14 AM
>>> Subject: Re: [Asterisk-Users] How do you handle NAT?
>>> 
>>> 
 We've been feeling our way along with the NAT stuff (using SIP) as well.
 
 At this point we are fairly small, so the keep-alive packets are not too
 bad.
 What type of user load are you at and what are the specs on your
 Asterisk box?
 I'm concerned we may run into this as well.
 
 We do have the luxury that each Sipura device we use is sitting behind
 its own
 NAT (a customer CPE).  So we can do port-forwarding and in combination
 with a
 STUN server (MyStun), things work quite well.  The only issues left to
 deal
 with are a lingering problem with ip_conntrack entries staying cached
 because
 of the "keep alive" packets due to qualify=yes after the CPE's IP
 address
 changes.
 
 Curious to hear other's setups as well.  I would *love* to start using
 the
 IAXy instead, but it has a couple shortcomings over the Sipura 2002's
 we're
 using now:
 
 - About $10/more
 - Only has one line (apparently two lines is a bit more of a selling
 point).
 
 Still trying to figure out a good way to make a case for the IAXy
 though.
 
 Ray
 
 On Tue, Jun 28, 2005 at 09:59:49AM -0500, Matthew Boehm wrote:
 
> We are interested in how other people are handling NAT problems. We
> have
> several customers all of which have some sort of firewall/NAT device at
> their location. For simplicity sake, all customers' internal networks
> are 192.168.*.*.
> 
> Our asterisk box is on public IP not blocked by any FW/NAT.
> 
> I use QUALIFY=yes on all our customers' phones and I feel that sending
> out 80-something keep-alive packets is causing our box to crawl and
>

RE: [Asterisk-Users] SIP Phone Config Generator

[Tarpo, Louie]

We use a mix of Cisco/Polycom phones. I just keep generic template files and 
make a copy of them for each new phone.  It would be fairly trivial to put a 
webserver on the same server with the tftpd.  In php or perl it would be fairly 
trivial to make a webpage that copied template files, replaced default values, 
and wrote out SIPmac.cnf or mac.cfg and mac-ext.cfg.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Max Clark
Sent: Tuesday, June 28, 2005 5:33 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] SIP Phone Config Generator


Hi all,

Cisco/Polycom phones will pull their configuration via a tftp server to 
help manage mass deployments of phone systems. Are there any config 
generators available that will create the file for the tftp server?

TIA,
Max
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Re: [Asterisk-Users] SIP Phone Config Generator

[Steve Totaro]

I believe that [EMAIL PROTECTED] will create cisco files.

- Original Message - 
From: "Max Clark" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"

Sent: Tuesday, June 28, 2005 4:32 PM
Subject: [Asterisk-Users] SIP Phone Config Generator


> Hi all,
>
> Cisco/Polycom phones will pull their configuration via a tftp server to
> help manage mass deployments of phone systems. Are there any config
> generators available that will create the file for the tftp server?
>
> TIA,
> Max
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Re: [Asterisk-Users] MeetMe application in Asterisk V1.07

[Moises Silva]

the line 187 of app_meetme.c of the CVS version says:

#define CONFFLAG_INTROUSER (1 << 14)/* If set, user will be ask
record name on entry of conference */

in Asterisk 1.0.7 from the sources that Debian download, this flag
does not exists, actually there are many others differences in the
sources, this is just one of them. So, yoy have to use CVS also, or
ask your friend to use 1.0.7 if you want to behave the same way.
Other option is hack the code :p but i guess is not fully recommended.

Best Regards

On 6/28/05, Monty <[EMAIL PROTECTED]> wrote:
> Hello,
> 
> Thanks for the message.  The exact version numbers and conf info follow:
> 
> My Asterisk:
> 
> /usr/sbin/asterisk -V = Asterisk 1.0.7-BRIstuffed-0.2.0-RC7k
> 
> Debian/AMD64 package file name/version:
>   pool/main/a/asterisk/asterisk_1.0.7.dfsg.1-2_amd64.deb
> 
> Extension.conf:
> 
> exten => 8800,1,Meetme(8801|aciMps)
> 
> meetme.conf:
> 
> [rooms]
> conf => 8801
> 
> Friends Asterisk:
> 
> /usr/sbin/asterisk -V = Asterisk-CVS-head
> 
> Note:  Above version seams strange but that's what it returned.  It was
> downloaded via CVS on June 27th 2005.
> 
> extension.conf
> 
> exten => 2003,1,Meetme(2003|aciMps)
> 
> meetme.conf:
> 
> [rooms]
> conf => 2003
> 
> 
> Does that help at all?
> 
> Thanks,
> Monty
> 
> 
> On Tue, 28 Jun 2005, Moises Silva wrote:
> 
> > i think that the thing that really matters here is wich version of
> > Asterisk are you using exactly. I dont know wich version the latest
> > debian package is using, and i dont know wich version from CVS your
> > friend has compiled.
> >
> > Also, its needed to show the extensions.conf configuration of your
> > friend. Specially the options passed to meetme
> >
> > best regards
> >
> > On 6/28/05, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
> >> Hello list,
> >>
> >> I wonder if someone might be able to clear up something for me.
> >>
> >> I recently set up asterisk and have now managed to get the MeetMe
> >> application up and running.
> >>
> >>
> >> When I dial the extension to access the conference/MeetMe application, the
> >> only prompt I hear is:"You are currently the only person in this
> >> conference."  When I use a friend's newly installed asterisk, I hear:
> >> "After the tone, say your name and then press the pound key."  We both
> >> have used virtually the same Meetme configurations.(The FWD 514 extension
> >> works the same way)  I believe I have all the necessary sounds but am
> >> really quite stuck here.  Please help! I am using the latest Debian/AMD64
> >> package and my friend compiled from cvs source I believe.
> >>
> >> My settings follow:
> >>
> >> extension.conf:
> >>
> >> exten => 8800,1,Meetme(8801|aciMps)  ; ext-8800 accesses conf-room  8801
> >>
> >> meetme.conf
> >>
> >> [rooms]
> >> conf => 8801
> >>
> >>
> >> Thanks for any help,
> >>Monty
> >> ___
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> >> Asterisk-Users@lists.digium.com
> >> http://lists.digium.com/mailman/listinfo/asterisk-users
> >> To UNSUBSCRIBE or update options visit:
> >>http://lists.digium.com/mailman/listinfo/asterisk-users
> >>
> >
> >
> >
> ___
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-- 
"Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org";
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[Asterisk-Users] Newbie Confusion on Call Forward and DBput/DBdel

[Jeffrey Starin]

I forgot to mention in my previous post that as well as:

/CFIM/2000 :12125553434

showing up in the database, the phone is registered properly and also showing 
up in the database show.  For some reason it
seems that Asterisk is not reading the database before connecting the call.  
Does that sound right?

Using CVS-HEAD

Thanks,

B.


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[Asterisk-Users] SIP Phone Config Generator

[Max Clark]

Hi all,

Cisco/Polycom phones will pull their configuration via a tftp server to 
help manage mass deployments of phone systems. Are there any config 
generators available that will create the file for the tftp server?


TIA,
Max
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Re: [Asterisk-Users] Asterisk RSS list feeder ready

[Denis Galvão - iSolve]

Where?

Denis.

On 28 de jun de 2005, at 19:42, Sjaak Nabuurs wrote:


Hello


Just for fun a rss newsreader for the asterisk users and biz list.
Easy to use and now with the complete history to search.

Just use it if you like

Thanks


Sjaak

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[Asterisk-Users] Asterisk rss list feeder ready

[Sjaak Nabuurs]

Sorry URL is needed


http://asterisk.voipexco.com


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Re: [Asterisk-Users] How do you handle NAT?

[hank]

I think my problem is numbrer 3 cause basicly my friend who is not on my 
router is trying to get connected to me but can't and I am the 1 that is 
behind a nat.

thanks
hank
- Original Message - 
From: "Sebastian Silva" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 


Sent: Tuesday, June 28, 2005 12:45 PM
Subject: Re: [Asterisk-Users] How do you handle NAT?



Hi everyone.

1.  Asterisk as a SIP client behind nat, connecting to outside SIP 
Proxies:
#1 works with a NAT-supporting proxy as SIP Express router as the outside 
proxy. (Get an account at IPtel.org and try!). Fails with Free World 
Dialup.


2. Asterisk as a SIP client behind nat, connecting to inside SIP proxies:
#2 Works- no NAT in between

3. Asterisk as a SIP server behind nat, clients on the outside connecting 
to Asterisk:

#3 Works with port forwarding and some header mangling magic

4. Asterisk as a SIP server behind nat, clients on the inside connecting 
to Asterisk:

#4 Works - no NAT in between

5. Asterisk as a SIP client outside nat, connecting to outside SIP 
proxies:

#5 is no problem. No NAT in the middle

6. Asterisk as a SIP client outside nat, connecting to inside SIP proxies:
#6 is a problem if no port forwarding is done, similar to 3 above.

7. Asterisk as a SIP server outside nat, clients on the outside connecting 
to Asterisk:

#7 is no problem. No NAT in the middle

8. Asterisk as a SIP server outside nat, clients on the inside connecting 
to Asterisk:
#8 is solved with nat=yes and qualify=xxx in sip.conf for the client in 
most cases. Some clients (X-lite) assist themselves by using STUN and 
sending UDP keep-alive packets. Qualify sends keep-alive packets from 
Asterisk to the client on the inside.


from wiki

Now, if you net to define a NAT, you have to set asterisk to 
"canreinvite=no", "qualify=yes" and "nat=1".


Also, INSTEAD of NAT, you can use a STUN server. To use a STUN server you 
should set asterisk to "canreinvite=no", "qualify=no" and "nat=0" (the 
STUN configuration is in your agents).


Sebas

hank wrote:
how easy is it to set up a stun server? with asterisk amd will this fix 
part of the nat problem?
- Original Message - From: "Ray Van Dolson" 
<[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 


Sent: Tuesday, June 28, 2005 8:14 AM
Subject: Re: [Asterisk-Users] How do you handle NAT?



We've been feeling our way along with the NAT stuff (using SIP) as well.

At this point we are fairly small, so the keep-alive packets are not too 
bad.
What type of user load are you at and what are the specs on your 
Asterisk box?

I'm concerned we may run into this as well.

We do have the luxury that each Sipura device we use is sitting behind 
its own
NAT (a customer CPE).  So we can do port-forwarding and in combination 
with a
STUN server (MyStun), things work quite well.  The only issues left to 
deal
with are a lingering problem with ip_conntrack entries staying cached 
because
of the "keep alive" packets due to qualify=yes after the CPE's IP 
address

changes.

Curious to hear other's setups as well.  I would *love* to start using 
the
IAXy instead, but it has a couple shortcomings over the Sipura 2002's 
we're

using now:

- About $10/more
- Only has one line (apparently two lines is a bit more of a selling 
point).


Still trying to figure out a good way to make a case for the IAXy 
though.


Ray

On Tue, Jun 28, 2005 at 09:59:49AM -0500, Matthew Boehm wrote:

We are interested in how other people are handling NAT problems. We 
have

several customers all of which have some sort of firewall/NAT device at
their location. For simplicity sake, all customers' internal networks
are 192.168.*.*.

Our asterisk box is on public IP not blocked by any FW/NAT.

I use QUALIFY=yes on all our customers' phones and I feel that sending
out 80-something keep-alive packets is causing our box to crawl and
cause bad calls.

Would SER be better in this case? Should I have phones register with 
SER

instead of with Asterisk?

Thanks,
Matthew

P.S. Yes, I have read stuff on NAT on the wiki. I'm more interested in
other real world, working, solutions.


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--
Sebastian Silva
G R U P O  G A U S S
Depto. Sistemas
Av. Libertador 6250 4 piso
Tl.: 4 706- (int. 121)
[EMAIL PROTECTED]
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Re: [Asterisk-Users] How do you handle NAT?

[OMS]

Does MyStun take care of RTP issues?

> The STUN server was extremely easy to set up.  Just check ou the MyStun
> sources (you have to use CVS), compile and run the server executable.
That's
> about all there is to it.

People have been posting  this question before, but did not get any clear
cut response, or probably there is'nt one?
How Vonage and Road Runner is doing it then?

I am able to get my inside Sonicwall SIP phones registered to the Asterisk
(on public IP).

The scenario is
SIP ATA (NAT) <---> Asterisk <> AS5400 <---> PSTN

ATA to PSTN calls are working great.
But when I call from PSTN to SIP ATA I am getting 

a) no audio when I have canreinvte=no in SIP.conf
b) one way audio from SIP to PSTN only if I have canreinvte=yes

Again do you think  MyStun will take care of RTP issues in this case?  Is ok
to install it on same machine.

I know this can be resolve using SIP enabled firewall or VPN, but it is not
practical for us.

Obaid.

- Original Message - 
From: "Ray Van Dolson" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"

Sent: Tuesday, June 28, 2005 4:23 PM
Subject: Re: [Asterisk-Users] How do you handle NAT?


> The STUN server was extremely easy to set up.  Just check ou the MyStun
> sources (you have to use CVS), compile and run the server executable.
That's
> about all there is to it.
>
> Ray
>
> On Tue, Jun 28, 2005 at 11:07:48AM -0700, hank wrote:
> > how easy is it to set up a stun server? with asterisk amd will this fix
> > part of the nat problem?
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Re: [Asterisk-Users] Shoutcast Music On Hold problems?

[hank]

how is your mp3 named? in the empty directory?
I think I may have mine screwed up
can you email me the mp3 that you have so I can just drop in to the 
directory and then add the line to my musiconhold.conf?

thanks
hank
- Original Message - 
From: "Huddleston, Robert" <[EMAIL PROTECTED]>
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" 


Sent: Tuesday, June 28, 2005 11:52 AM
Subject: RE: [Asterisk-Users] Shoutcast Music On Hold problems?


Worked for me with a different stream... I ran into this same problem 
before - but it was my own fault for not RTM... Both the manual and ast 
install advised of verifying correct version of mpg123... I had wrong 
version and thus got no noise...
If you follow the directions explicitly laid out on the wiki you should 
have no problems.

I use christianrock.net's shoutcast stream
Like this in musiconhold.conf
default => 
quietmp3:/var/lib/asterisk/mohmp3-empty,http://209.51.128.160:5112/




-Original Message-
From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of hank

Sent: Tuesday, June 28, 2005 1:41 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Shoutcast Music On Hold problems?

I tried that the stream i tried to use orriginally was
http://209.97.198.50:30518
all I get is silence when I put the person on hold thanks hank
- Original Message -
From: "Patrick" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"

Sent: Tuesday, June 28, 2005 2:50 AM
Subject: Re: [Asterisk-Users] Shoutcast Music On Hold problems?



On Mon, 2005-06-27 at 22:51 -0700, hank wrote:

 mp3:/var/lib/asterisk/mohmp3-empty,http://www.waixwave.com:8000/


I haven't tried this myself but if I put www.waixwave.com:8000 in
Firefox I get "connection refused". Try another site that actually
streams music. Shoutcast.org should have a nice list.

Regards,
Patrick

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RE: [Asterisk-Users] Asterisk with Lucent TNT echo

[Carlos]

Hey jeremiah,

Do you hear a click click click sound I remember getting that with the
licent tnt with the asterisk server main reason we stopped using the tnt.

Carlos Alcantar
Race Technologies, Inc.
101 Haskins Way
South San Francisco, CA 94080
P: 650.246.8900
F: 650.246.8901
E: carlos at race.com 

-Original Message-
From: Jeremiah Millay [mailto:[EMAIL PROTECTED] 
Sent: Tuesday, June 28, 2005 2:50 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Asterisk with Lucent TNT echo

I'm running SIP between my Lucent TNT acting as a gateway, and an asterisk
server. We have a PRI coming into the Lucent. Basically the problem I'm
having is mostly on inbound calls but some outbound calls as well. I hear
echo and sometimes some weird artifacting on calls coming in from the
lucent. Everything routed over IAX to VoIP Jet or Nufone sounds fine. It
seems like every 3 calls I get is a bad one.
Does anyone on the list run asterisk with this configuration? Does anyone
have any tips to solve this issue?
I've tried modifying the gains at the lucent, as well as turn off and on
jitter buffers on asterisk. Tweaking these seems to help but I'm looking for
something more solid. Any help would be appreciated.
Regards,
Jeremiah

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Re: [Asterisk-Users] Asterisk Realtime and ODBC

[Tim Lyakhovetskiy]

Yes, and there are no errors.

Thanks,
-Tim


Tim Lyakhovetskiy wrote:

Hello all!

My basic problem is that we haven't been able to get realtime to use ODBC 
to store configuration data. Here are the details:


Are you watching the asterisk debug log for any errors?

-Matthew

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Re: [Asterisk-Users] How do you handle NAT?

[Geert Nijpels]

On 6/28/05, Sebastian Silva <[EMAIL PROTECTED]> wrote:
> Hi everyone.
> 
> 1.  Asterisk as a SIP client behind nat, connecting to outside SIP Proxies:
> #1 works with a NAT-supporting proxy as SIP Express router as the
> outside proxy. (Get an account at IPtel.org and try!). Fails with Free
> World Dialup.
> 
> 2. Asterisk as a SIP client behind nat, connecting to inside SIP proxies:
> #2 Works- no NAT in between
> 
> 3. Asterisk as a SIP server behind nat, clients on the outside
> connecting to Asterisk:
> #3 Works with port forwarding and some header mangling magic
> 
> 4. Asterisk as a SIP server behind nat, clients on the inside connecting
> to Asterisk:
> #4 Works - no NAT in between
> 
> 5. Asterisk as a SIP client outside nat, connecting to outside SIP proxies:
> #5 is no problem. No NAT in the middle
> 
> 6. Asterisk as a SIP client outside nat, connecting to inside SIP proxies:
> #6 is a problem if no port forwarding is done, similar to 3 above.
> 
> 7. Asterisk as a SIP server outside nat, clients on the outside
> connecting to Asterisk:
> #7 is no problem. No NAT in the middle
> 
> 8. Asterisk as a SIP server outside nat, clients on the inside
> connecting to Asterisk:
> #8 is solved with nat=yes and qualify=xxx in sip.conf for the client in
> most cases. Some clients (X-lite) assist themselves by using STUN and
> sending UDP keep-alive packets. Qualify sends keep-alive packets from
> Asterisk to the client on the inside.
> 
> from wiki
> 
> Now, if you net to define a NAT, you have to set asterisk to
> "canreinvite=no", "qualify=yes" and "nat=1".
> 
> Also, INSTEAD of NAT, you can use a STUN server. To use a STUN server
> you should set asterisk to "canreinvite=no", "qualify=no" and "nat=0"
> (the STUN configuration is in your agents).
> 

You can use STUN instead of nat=yes (if the phone supports STUN
properly). However, our experience is that we also need qualify=yes to
prevent the phones becoming unreachable.

Geert

> hank wrote:
> > how easy is it to set up a stun server? with asterisk amd will this fix
> > part of the nat problem?
> > - Original Message - From: "Ray Van Dolson" <[EMAIL PROTECTED]>
> > To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> > 
> > Sent: Tuesday, June 28, 2005 8:14 AM
> > Subject: Re: [Asterisk-Users] How do you handle NAT?
> >
> >
> >> We've been feeling our way along with the NAT stuff (using SIP) as well.
> >>
> >> At this point we are fairly small, so the keep-alive packets are not
> >> too bad.
> >> What type of user load are you at and what are the specs on your
> >> Asterisk box?
> >> I'm concerned we may run into this as well.
> >>
> >> We do have the luxury that each Sipura device we use is sitting behind
> >> its own
> >> NAT (a customer CPE).  So we can do port-forwarding and in combination
> >> with a
> >> STUN server (MyStun), things work quite well.  The only issues left to
> >> deal
> >> with are a lingering problem with ip_conntrack entries staying cached
> >> because
> >> of the "keep alive" packets due to qualify=yes after the CPE's IP address
> >> changes.
> >>
> >> Curious to hear other's setups as well.  I would *love* to start using
> >> the
> >> IAXy instead, but it has a couple shortcomings over the Sipura 2002's
> >> we're
> >> using now:
> >>
> >> - About $10/more
> >> - Only has one line (apparently two lines is a bit more of a selling
> >> point).
> >>
> >> Still trying to figure out a good way to make a case for the IAXy though.
> >>
> >> Ray
> >>
> >> On Tue, Jun 28, 2005 at 09:59:49AM -0500, Matthew Boehm wrote:
> >>
> >>> We are interested in how other people are handling NAT problems. We have
> >>> several customers all of which have some sort of firewall/NAT device at
> >>> their location. For simplicity sake, all customers' internal networks
> >>> are 192.168.*.*.
> >>>
> >>> Our asterisk box is on public IP not blocked by any FW/NAT.
> >>>
> >>> I use QUALIFY=yes on all our customers' phones and I feel that sending
> >>> out 80-something keep-alive packets is causing our box to crawl and
> >>> cause bad calls.
> >>>
> >>> Would SER be better in this case? Should I have phones register with SER
> >>> instead of with Asterisk?
> >>>
> >>> Thanks,
> >>> Matthew
> >>>
> >>> P.S. Yes, I have read stuff on NAT on the wiki. I'm more interested in
> >>> other real world, working, solutions.
> >>
> >> ___
> >> Asterisk-Users mailing list
> >> Asterisk-Users@lists.digium.com
> >> http://lists.digium.com/mailman/listinfo/asterisk-users
> >> To UNSUBSCRIBE or update options visit:
> >>   http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
> > ___
> > Asterisk-Users mailing list
> > Asterisk-Users@lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
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> >   http://lists.digium.com/mailman/listinfo/asterisk-users
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> 
> --
> Sebastian Silva
> G R U P O  G

[Asterisk-Users] Asterisk RSS list feeder ready

[Sjaak Nabuurs]

Hello


Just for fun a rss newsreader for the asterisk users and biz list.
Easy to use and now with the complete history to search.

Just use it if you like

Thanks


Sjaak

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Re: RE : [Asterisk-Users] HowTo start DIAL by a sillent training as Wfor modems

[Andrew Latham]

exten=>101,1,Wait(2)
exten=>101,2,Dial(123456789)

On 6/28/05, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
> Hello the list, hello Doug,
> 
> Thank you, but I don't see any correct reply in this page.
> I want to have a "silent header" of 1 or 2 seconds between to pick up the
> line and before to start to sned the DTMF numbering, because my RTC provider
> doesn't give the prompt tone or listen the DTMF before this time.
> If I place a "normal" call, it fails, because the first tones of my Asterisk
> numbering sequence are missed.
> 
> If any idea...
> 
> Best Regards,
> Francois BERGERET,
> France.
> 
> 
> -Message d'origine-
> De : [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] De la part de Doug Lytle
> Envoyé : mardi 28 juin 2005 12:25
> À : Asterisk Users Mailing List - Non-Commercial Discussion
> Objet : Re: [Asterisk-Users] HowTo start DIAL by a sillent training as Wfor
> modems
> 
> 
> 
> [EMAIL PROTECTED] wrote:
> 
> >Hi all the list,
> >
> >I am searching how to insert few seconds of silence just before to send
> >the DTMF sequence via a FXO WildCard X101P to PSTN.
> >
> >I remember that Hayes compatible modems knows a special character "W"
> >that do a 1 sec pause. Is it possible to do something like this in DIAL
> >line sequence ?
> >
> >
> >
> This should help:
> 
> http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+Dial
> 
> 
> Doug
> 
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-- 

Andrew Latham - AKA: LATHAMA (lay-th-ham-eh)
WWW: http://lathama.com
Email: [EMAIL PROTECTED] - [EMAIL PROTECTED] - [EMAIL PROTECTED]
If any of the above are down we have bigger problems than my email!

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RE: [Asterisk-Users] OT: Good soft-phone on Linux

[John Mylchreest]

Title: RE: [Asterisk-Users] OT: Good soft-phone on Linux







sjphone by sjlabs do a linux binary.
its actually quite excellent, but you need oss emu/oss for sound output/input

-Original Message-
From: [EMAIL PROTECTED] on behalf of Jesus Mogollon
Sent: Tue 28/06/2005 22:02
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] OT: Good soft-phone on Linux

How about Xlite for Linux? It's already out... www.xten.com

2005/6/27, Hamish Whittal <[EMAIL PROTECTED]>:
>
> Hi Folks,
>
> I am wanting advise on a good soft-phone on Linux. I have looked at
> Gnophone but cannot seem to get it to compile under debian sarge. I am
> now looing at sipXphone seem to be picking up that it is not that
> stable, but perhaps someone here can advise on what softphone I can use
> on Linux.
>
> Thanks in advance,
>
> Hamish
> ---
> | Hamish Whittal | Mobile: +27 82 803 5533 |
> | QED Technologies cc | landline: +27 21 671 7710 |
> | 21 Marne Avenue, Claremont, Cape Town | fax: +27 21 674 9184 |
> |fortune cookie below autogenerated_|
> You are confused; but this is your normal state.
>
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Re: [Asterisk-Users] VoipJet TOS (was Teliax and also LiveVoip)

[steve szmidt]

On Tuesday 28 June 2005 15:02, Wiley Siler wrote:
> One would assume they have better things to do as they are quite busy.
> I think this is just a proactive measure meaning they say you cannot do
> it upfront so that in the event of a problem, it was predeclared.  As to
> the rest of the TOS, I could be wrong but I got the impression that the
> owner of VoipJet speaks English as a second language due to some email
> exchanges.  If that is the case, the TOS issue can just be one of
> cultural and language variation.  I could be totally wrong though...
>
> Regardless, one can assume that the ability to listen to calls is there.

And quite frankly is usually illegal. AOL used to get around that by having 
their EULA say that AOL owns all the emails etc.

-- 

Steve Szmidt

"They that would give up essential liberty for temporary safety 
deserve neither liberty nor safety."
Benjamin Franklin
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Re: [Asterisk-Users] Asterisk Realtime and ODBC

[Tim Lyakhovetskiy]

Tried all sorts of different combinations; loading just a config file, or 
loading just peers, or loading just extensions, etc... None of it produces 
any query packets on the network.


Thanks,
-Tim



try putting your sip.conf on disk and load peers/users from the  database.

/b
---
Anakin: “You’re either with me, or you’re my enemy.”
Obi-Wan: “Only a Sith could be an absolutist.”

On Jun 28, 2005, at 1:36 PM, Tim Lyakhovetskiy wrote:


Hello all!

My basic problem is that we haven't been able to get realtime to  use ODBC 
to store configuration data. Here are the details:


We've installed Asterisk on a CentOS machine as follows:
1. Downloaded, compiled, and installed FreeTDS 0.63
2. Downloaded, compiled, and installed  unixODBC 2.2.11
3. Downloaded, compiled, and installed Asterisk, Asterisk-Addons,  and 
Zaptel from CVS HEAD


We checked the FreeTDS and ODBC connections several ways, using the  
"tsql" and "isql" utilities, and Ethereal for network monitoring.  
Everything worked perfectly. Then we set up asterisk as follows:




[settings]

sip.conf => odbc,asterisk,ast_config
sipusers => odbc,asterisk,sip_buddies
sippeers => odbc,asterisk,sip_buddies
extensions_table => odbc,asterisk,extensions_table



[default]
switch => Realtime/[EMAIL PROTECTED]



[asterisk]
dsn => "our DSN name"
username => "our username"
password => "our password"
pre-connect => yes



[modules]
autoload=yes

noload => pbx_gtkconsole.so
noload => pbx_kdeconsole.so
noload => app_intercom.so

load => chan_modem.so
load => res_musiconhold.so

noload => chan_alsa.so

noload => res_config_mysql.so
noload => cdr_addon_mysql.so
load => app_realtime.so
load => pbx_realtime.so

After that, we loaded up Asterisk, and there were no error  messages. Not 
only that, the res_odbc and res_config_odbc modules  loaded up just fine, 
AND Ethereal showed that a login to our MS SQL  database went through. 
However, when we tried to add extensions or  SIP friends, nothing worked. 
The strange thing was that Asterisk  didn't even try to query the database 
(observed through Ethereal)  and just tried to use the conf files.


Just to make sure we weren't crazy, we set up Asterisk using MySQL  and 
that setup worked perfectly right from the start. We would  still like to 
use ODBC->MS SQL though, because we would like to  keep the amount of 
different databases minimal.


voip-info isn't much help on this, and we've looked far and wide  for any 
more information regarding Asterisk ODBC setups, finding  very little.


Anything we missed?

Thanks,
-Tim Lyakhovetskiy

_
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[Asterisk-Users] Asterisk with Lucent TNT echo

[Jeremiah Millay]

I'm running SIP between my Lucent TNT acting as a gateway, and an 
asterisk server. We have a PRI coming into the Lucent. Basically the 
problem I'm having is mostly on inbound calls but some outbound calls as 
well. I hear echo and sometimes some weird artifacting on calls coming 
in from the lucent. Everything routed over IAX to VoIP Jet or Nufone 
sounds fine. It seems like every 3 calls I get is a bad one.
Does anyone on the list run asterisk with this configuration? Does 
anyone have any tips to solve this issue?
I've tried modifying the gains at the lucent, as well as turn off and on 
jitter buffers on asterisk. Tweaking these seems to help but I'm looking 
for something more solid. Any help would be appreciated.

Regards,
Jeremiah

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Re: [Asterisk-Users] TDM04B Echo on Only One Channel

[Steve Totaro]

Good thought and after replacing the module on port four, everything is
great.


- Original Message - 
From: "qrss" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"

Sent: Tuesday, June 28, 2005 1:49 PM
Subject: Re: [Asterisk-Users] TDM04B Echo on Only One Channel


I have not seen the problem, but here's a thought...

You might try moving the daughterboard for that port.  If the trouble
still follows the port, then I'd suspect it's a hardware problem with that
daughterboard. If it does not, then perhaps a software/configuration
issue.

-Original Message-
From: Steve Totaro
Sent: Tue, June 28, 2005 6:45 pm

>I have a system that is experiencing profound echo on channel 4.  I
>switched POTS lines around to see if it was the line or the port and it is
>still a problem on port four.  Has anyone ever seen this before?


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Re: [Asterisk-Users] How do you handle NAT?

[Ray Van Dolson]

The STUN server was extremely easy to set up.  Just check ou the MyStun
sources (you have to use CVS), compile and run the server executable.  That's
about all there is to it.

Ray

On Tue, Jun 28, 2005 at 11:07:48AM -0700, hank wrote:
> how easy is it to set up a stun server? with asterisk amd will this fix 
> part of the nat problem?
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[Asterisk-Users] BRIstuff/OctoBRI problem: Ring requested on unconfigured channel 255/255 span 5

[Bruno . Voigt]

Hi all,

I just posted this question before last week.
Meanwhile after upgrading Asterisk 1.0.7-BRIstuffed-0.2.0-RCg to 
1.0.8-BRIstuffed-0.2.0-RCh
the same problem occurs, but seems to be more seldom.
Attached is now the output of "zap show channel" .
-

I'm running a stable Asterisk on a HP DL380G2 1.4Ghz 0,5GB RAM
equipped with 1x TE410P and 2xJunghanns OctoBRI running in NT-mode.

Connected to the BRI-Ports are 12 Fax-Modems (Elsa MicroLink ISDN/TL V.34)
which are only operating in dial out analog mode to deliver fax messages.

After a while of running fine (50-200 dial out connections)
on some S0 spans the following message occurs over and over again:

chan_zap.c:8009 pri_dchannel: Ring requested on unconfigured channel 
255/255 span 5

The Modems connected to this span get NO DIALTONE for every ATD.
Modems on other spans continue to operate.

After restarting asterisk everything is okay again for a while.

Any hints about what is going on here are greatly appreciated ;-)

TIA, Bruno - bruno @ ic3s.de

Connected to Asterisk 1.0.8-BRIstuffed-0.2.0-RC8h currently running on pbx 
(pid = 20305)
Verbosity is at least 5
pbx*CLI> show channels
Channel  (ContextExtensionPri )   State Appl. Data
   Zap/25-1  (pri1   s1   )  Up Bridged Call 
Zap/128-1
  Zap/128-1  (from-s0-faxmodems 00711xxx 5   )  Up Dial 
Zap/r1/0711xxx
   Zap/24-1  (pri1   s1   )  Up Bridged Call 
Zap/132-1
  Zap/132-1  (from-s0-faxmodems 00242xxx  5   )  Up Dial 
Zap/r1/0242xxx
4 active channel(s)
Jun 24 11:49:19 WARNING[20305]: chan_zap.c:8009 pri_dchannel: Ring 
requested on unconfigured channel 255/255 span 5
  == Primary D-Channel on span 6 down for TEI 64
  == Primary D-Channel on span 6 up for TEI 64
-- Accepting overlap voice call from '' to '00394' on channel 0/2, 
span 6
-- Starting simple switch on 'Zap/129-1'
-- Channel 0/2, span 7 got hangup
-- Hungup 'Zap/24-1'
  == Spawn extension (from-s0-faxmodems, 00242xxx, 5) exited non-zero on 
'Zap/132-1'
-- Hungup 'Zap/132-1'
-- Executing SetCallerPres("Zap/129-1", "prohib") in new stack
-- Executing NoOp("Zap/129-1", "") in new stack
-- Executing SetTransferCapability("Zap/129-1", "3K1AUDIO") in new 
stack
-- Setting transfer capability to: 0x10 - 3K1AUDIO.
-- Executing SetCIDNum("Zap/129-1", "0410612345") in new stack
-- Executing Dial("Zap/129-1", "Zap/r1/039") in new stack
-- Requested transfer capability: 0x10 - 3K1AUDIO
-- Called r1/039
-- Zap/26-1 is ringing
-- Zap/26-1 answered Zap/129-1
-- Attempting native bridge of Zap/129-1 and Zap/26-1
Jun 24 11:49:35 WARNING[20305]: chan_zap.c:8009 pri_dchannel: Ring 
requested on unconfigured channel 255/255 span 5
  == Primary D-Channel on span 7 down for TEI 65
  == Primary D-Channel on span 7 up for TEI 65

--
Output of "zap show channel":

  == Primary D-Channel on span 6 up for TEI 65
Jun 28 17:56:24 WARNING[23107]: chan_zap.c:8039 pri_dchannel: Ring 
requested on unconfigured channel 255/255 span 6
  == Primary D-Channel on span 5 down for TEI 64
Jun 28 17:56:27 WARNING[23107]: chan_zap.c:8039 pri_dchannel: Ring 
requested on unconfigured channel 255/255 span 7

pbx*CLI> zap show channel 131
Channel: 131
File Descriptor: 140
Span: 7
Extension:
Dialing: no
Context: from-s0-faxmodems
Caller ID string:
Destroy: 0
InAlarm: 0
Signalling Type: PRI Signalling
Owner: 
Real: 
Callwait: 
Threeway: 
Confno: -1
Propagated Conference: -1
Real in conference: 0
DSP: no
Relax DTMF: no
Dialing/CallwaitCAS: 0/0
Default law: alaw
Fax Handled: no
Pulse phone: no
Echo Cancellation: 128 taps unless TDM bridged, currently OFF
PRI Flags: Call
PRI Logical Span: Implicit
Actual Hookstate: Onhook

pbx*CLI> zap show channel 132
Channel: 132
File Descriptor: 141
Span: 7
Extension: 00569xxx
Dialing: no
Context: from-s0-faxmodems
Caller ID string:
Destroy: 0
InAlarm: 0
Signalling Type: PRI Signalling
Owner: Zap/132-1
Real: Zap/132-1
Callwait: 
Threeway: 
Confno: -1
Propagated Conference: -1
Real in conference: 1
DSP: yes
Relax DTMF: no
Dialing/CallwaitCAS: 0/0
Default law: alaw
Fax Handled: no
Pulse phone: no
Echo Cancellation: 128 taps unless TDM bridged, currently OFF
Slave Channel: 1
PRI Flags: Call
PRI Logical Span: Implicit
Actual Confinfo: Num/1, Mode/0x0009
Actual Confmute: No
Actual Hookstate: Onhook
Jun 28 17:57:11 WARNING[23107]: chan_zap.c:8039 pri_dchannel: Ring 
requested on unconfigured channel 255/255 span 6
  == Primary D-Channel on span 6 down for TEI 65
  == Primary D-Channel on span 6 up for TEI 65
Jun 28 17:57:12 WARNING[23107]: chan_zap.c:8039 pri_dchannel: Ring 
requested on unconfigured channel 255/255 span 6
Jun 28 17:57:16 WARNING[23107]: chan_zap.c:8039 pri_dchannel: Ring 
requested on unconfigured channel 255/255 span 7
-- Channel 0/2, span 5 got hangup
  == Primary D-Channel on span 6 up for TEI 65
Jun 28 17:56:24 WARNING[23107]: chan_zap.c:8039 pri_dchan

Re: [Asterisk-Users] Group/Broadcast Voicemail

[Robert Goodyear]

On Jun 28, 2005, at 2:06 PM, Chris Stinson wrote:


Were you guys able to figure this out?


Robert Goodyear <[EMAIL PROTECTED]> wrote  :



On Jun 22, 2005, at 1:50 PM, Zen Kato wrote:


Hi Robert,


Let me guess... mailbox 5103 or 5203 were the last in the list to
receive it?


Every trials(1-6) I got only 51 mailboxes copied. My quick guess is
256/5(u0103 and &xx03s)=51...1, so changing tmp[256] to tmp[4096]
does not work. 'Pseudo-diagram' as you mentioned before(6/8/05)
is desirable for expandability, but it also did not work.




So what about the variable BASEMAXINLINE? Did you change that and
recompile yet?

Yes, I changed #define BASEMAXLINE on step 5(line 80) and step 6(line
82)
and recomiled each case.


I haven't had time to play with this. I posted over to DEV hoping 
someone who had their hands on that source had something to say, but 
nothing yet.



--
Robert Goodyear
Brand Up LLC
http://www.brand-up.com

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Re: [Asterisk-Users] initial setup: problem

[Moises Silva]

the error is pretty clear, the 404 is HTTP not found response. Not
sure what the problem may be, but i think the most common error is
that you have typed incorrectly the address, try addin a final slash
http://10.10.10.10/admin/

best regards

On 6/28/05, Sean Browne (Cardiff IT Support Ltd)
<[EMAIL PROTECTED]> wrote:
> hi,
> 
> i just installed AAH. from the bootable CDROM disk after i downloaded iso
> image.
> 
> AAH box is at http://10.10.10.10
> 
> I use browser to access the box, and can get to main menu and voicemail
> login and sugar login.
> 
> However, the asterisk management http://10.10.10.10/admin gives a 404 error,
> page unavailable.
> 
> Any ideas? should I re-install?
> 
> thanks,
> 
> sean
> 
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[Asterisk-Users] Inter-Tel 8662 configuration problem.

[Alejandro Sosa]

Hi,

 

I’m trying to use the Inter-Tel 8662 endpoint (SIP
phone), but it’s giving me problems with the dial plan configuration.

I get the phone to register with Asterisk. I can place and
receive internal calls (to/from extensions within the PBX), but when I try to
dial out thru a trunk (9x), the phone doesn’t let me enter all the
digits. Instead it goes back to dial tone after I enter like 7 digits.

I suspect it has something to do with some dial plan
settings that the phone itself has (not the dial plan in asterisk, but actual
entries in the web interface for the phone configuration)... but I haven’t
been able to find administrator’s documentation for this phone (only a
users guide in the website, but it has no information on how to configure it).

If anyone has worked with these devices or has any
documentation on them, please contact me.

Thanks a lot.

 

Alejandro Sosa.

 






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Re: [Asterisk-Users] Voice Mail hangup on not messages

[Moises Silva]

actually i think that notification exists, may be the problem is that
the sound is not found. Try checking the verbose output in the console

best regards

On 6/28/05, David Romero <[EMAIL PROTECTED]> wrote:
> i have a problem whit voice mail  on [EMAIL PROTECTED] whit CVS HEAD
>  
>  when try to use voice mail  and i have at least 1 message all work ok...
>  
>  when try to use voice mail  and not have messages asterisk hangup after
> type my voicemail password  whitout any notification.
>  
>  how i can do a "not messages found " notification and enable the  others
> voice mail options like record not available message.
>  
>  
>  thanks
> 
> -- 
> David Romero
> ## 
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Re: [Asterisk-Users] TE110P Card

[Juan Pablo Abuyeres]

I guess you've solved this already.. but just in case, you should rmmod zaptel and modprobe it again.. I'm pretty sure you had a different version of zaptel loaded and you should be using the new one.


On Wed, 2005-06-22 at 15:58 +, MSEYE wrote:

Hi everybody !
I'm trying to setup a TE110P card. But i'm facind this error... and * cannot start.
 
 [chan_zap.so]Jun 22 11:50:33 WARNING[16384]: loader.c:248 ast_load_resource: /u
sr/lib/asterisk/modules/chan_zap.so: undefined symbol: pri_dump_info
Jun 22 11:50:33 WARNING[16384]: loader.c:429 load_modules: Loading module chan_z
ap.so failed!
 
Does someone already experience this case...
Waiting for u'r help
 
THANX
 
MSEYE

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Re: [Asterisk-Users] OT: Good soft-phone on Linux

[Jesus Mogollon]

How about Xlite for Linux? It's already out... www.xten.com2005/6/27, Hamish Whittal <[EMAIL PROTECTED]>:
Hi Folks,I am wanting advise on a good soft-phone on Linux. I have looked at
Gnophone but cannot seem to get it to compile under debian sarge. I amnow looing at sipXphone seem to be picking up that it is not thatstable, but perhaps someone here can advise on what softphone I can useon Linux.
Thanks in advance,Hamish ---|
Hamish
Whittal|
Mobile:   +27 82 803 5533 || QED Technologies
cc  
| landline: +27 21 671 7710 || 21 Marne Avenue, Claremont, Cape Town | fax:  +27 21 674 9184 ||fortune cookie below autogenerated_|You are confused; but this is your normal state.
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RE : RE : [Asterisk-Users] HowTo start DIAL by a sillent training as W for modems

[f6hqz-m]

Eureka ! It's working now !

The solution was in my question, in fact. So stupid I am !  :-)))
Many thanks to you guys, working around this fantastic project, and specialy
to Eric Wieling for his nice help.

Have a Nice Night !
Merci camarade !

Best Regards,
Francois BERGERET,
France.

http://www.hamwlan.net


-Message d'origine-
De : Eric Wieling aka ManxPower [mailto:[EMAIL PROTECTED] 
Envoyé : mardi 28 juin 2005 21:54
À : [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
Objet : Re: RE : [Asterisk-Users] HowTo start DIAL by a sillent training as
Wfor modems


If you are dialing out an analog Zap FXO port then you use the "w" 
option.  Dial(Zap/1/ww5551212)

[EMAIL PROTECTED] wrote:

> Hello the list, hello Doug,
> 
> Thank you, but I don't see any correct reply in this page.
> I want to have a "silent header" of 1 or 2 seconds between to pick up 
> the line and before to start to sned the DTMF numbering, because my 
> RTC provider doesn't give the prompt tone or listen the DTMF before 
> this time. If I place a "normal" call, it fails, because the first 
> tones of my Asterisk numbering sequence are missed.
> 
> If any idea...
> 
> Best Regards,
> Francois BERGERET,
> France.
> 
> 
> -Message d'origine-
> De : [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] De la part de Doug 
> Lytle Envoyé : mardi 28 juin 2005 12:25 À : Asterisk Users Mailing 
> List - Non-Commercial Discussion Objet : Re: [Asterisk-Users] HowTo 
> start DIAL by a sillent training as Wfor modems
> 
> 
> 
> [EMAIL PROTECTED] wrote:
> 
> 
>>Hi all the list,
>>
>>I am searching how to insert few seconds of silence just before to 
>>send
>>the DTMF sequence via a FXO WildCard X101P to PSTN.
>>
>>I remember that Hayes compatible modems knows a special character "W"
>>that do a 1 sec pause. Is it possible to do something like this in DIAL 
>>line sequence ?
>>
>> 
>>
> 
> This should help:
> 
> http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+Dial
> 
> 
> Doug
> 
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-- 
Always do right. This will gratify some people and astonish the rest. Mark
Twain

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[Asterisk-Users] Dial application timeout

[Rob Fugina]

For a *long* time now, I've noticed that the timeout passed to the
Dial application doesn't really work as documented.  When I
Dial(Zap/1&Zap/2&Zap/g3,15,tP) the Dial doesn't timeout for 25 to 30
seconds.  I found this problem when I had a 20 or 30 second timeout
specified, and it wasn't actually timing out for 40 - 60 seconds.  I
switched to a 15 second timeout specifically to get a the 20-30 second
timeout I wanted.

I've inserted some debug code in app_dial.c to verify that my timeout
parameter is being parsed out correctly (around line 1500 in the
current app_dial.c), but that's about as far as I can decipher at the
moment.

Is there anyone else out there that can duplicate this problem?

Thanks,
Rob
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[Asterisk-Users] Net2Phone equipment and different VOIP providers

[bph]

Hello we are a small call center with only 8 lines 
we use max4 and the 2-2 port gateways from net2phone . There equipment is good 
but we are getting hit by lower cost competition. We need to be able to compete. 
We have a couple of providers who are 50% less in some cases even more. So it 
makes sense that we would like to be able to compete . Since we have spent quite 
a bit of money on existing equipment that is in good operating condition we 
don't want to just throw it away but rather use it with the new provider. Has 
anyone out there been able to use the Max or 2 port gateway on other service 
providers. If so we would greatly appreciate you letting us know how. We are not 
experts by any means but we will try anything that works . I hope someone outher 
can help.
 
Julio Periz
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Re: [Asterisk-Users] How do you handle NAT?

[Andres]

Matthew Boehm wrote:

We are interested in how other people are handling NAT problems. We 
have several customers all of which have some sort of firewall/NAT 
device at their location. For simplicity sake, all customers' internal 
networks are 192.168.*.*.


Our asterisk box is on public IP not blocked by any FW/NAT.

I use QUALIFY=yes on all our customers' phones and I feel that sending 
out 80-something keep-alive packets is causing our box to crawl and 
cause bad calls.


Would SER be better in this case? Should I have phones register with 
SER instead of with Asterisk?


Matthew,

We use SER for user registration and NAT Keep Alives.  It can handle 
thousands of messages per second without skipping a beat.   I am pretty 
sure that Asterisk is not too happy about receiving all your keep alives:)




Thanks,
Matthew





--
Andres
Network Admin
http://www.telesip.net


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Re: [Asterisk-Users] TDM04B Echo on Only One Channel

[qrss]

I have not seen the problem, but here's a thought...

You might try moving the daughterboard for that port.  If the trouble
still follows the port, then I'd suspect it's a hardware problem with that
daughterboard. If it does not, then perhaps a software/configuration
issue.

-Original Message-
From: Steve Totaro
Sent: Tue, June 28, 2005 6:45 pm

>I have a system that is experiencing profound echo on channel 4.  I
>switched POTS lines around to see if it was the line or the port and it is
>still a problem on port four.  Has anyone ever seen this before?


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Re: [Asterisk-Users] MeetMe application in Asterisk V1.07

[Monty]

Hello,

Thanks for the message.  The exact version numbers and conf info follow:

My Asterisk:

/usr/sbin/asterisk -V = Asterisk 1.0.7-BRIstuffed-0.2.0-RC7k

Debian/AMD64 package file name/version:
 pool/main/a/asterisk/asterisk_1.0.7.dfsg.1-2_amd64.deb

Extension.conf:

exten => 8800,1,Meetme(8801|aciMps)

meetme.conf:

[rooms]
conf => 8801

Friends Asterisk:

/usr/sbin/asterisk -V = Asterisk-CVS-head

Note:  Above version seams strange but that's what it returned.  It was 
downloaded via CVS on June 27th 2005.


extension.conf

exten => 2003,1,Meetme(2003|aciMps)

meetme.conf:

[rooms]
conf => 2003


Does that help at all?

Thanks,
   Monty


On Tue, 28 Jun 2005, Moises Silva wrote:


i think that the thing that really matters here is wich version of
Asterisk are you using exactly. I dont know wich version the latest
debian package is using, and i dont know wich version from CVS your
friend has compiled.

Also, its needed to show the extensions.conf configuration of your
friend. Specially the options passed to meetme

best regards

On 6/28/05, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:

Hello list,

I wonder if someone might be able to clear up something for me.

I recently set up asterisk and have now managed to get the MeetMe
application up and running.


When I dial the extension to access the conference/MeetMe application, the
only prompt I hear is:"You are currently the only person in this
conference."  When I use a friend's newly installed asterisk, I hear:
"After the tone, say your name and then press the pound key."  We both
have used virtually the same Meetme configurations.(The FWD 514 extension
works the same way)  I believe I have all the necessary sounds but am
really quite stuck here.  Please help! I am using the latest Debian/AMD64
package and my friend compiled from cvs source I believe.

My settings follow:

extension.conf:

exten => 8800,1,Meetme(8801|aciMps)  ; ext-8800 accesses conf-room  8801

meetme.conf

[rooms]
conf => 8801


Thanks for any help,
   Monty
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[Asterisk-Users] Trying to get *8 call pickup to work

[Robert Woodcock]

I'm using the Debian Sarge package of Asterisk - 1.0.7 + bristuff. When
I call from a zap channel or a SIP phone to another SIP phone, then dial
*8 from a third SIP phone, I get 503 Service Unavailable on the
third phone and I get this at the Asterisk console:

Jun 28 09:01:24 DEBUG[16774]: res_features.c:1709 ast_pickup_call: No call 
pickup possible...
Jun 28 09:01:24 NOTICE[16774]: chan_sip.c:7402 handle_request: Nothing to pick 
up

I'd appreciate hearing from anyone that has this working. 

Here's my sip.conf, features.conf, and zapata.conf:

# < zapata.conf sed 's/;.*//g' | grep -v ^$
[trunkgroups]
[channels]
context=default
switchtype=national
signalling=em_w
rxwink=300
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
group=1
callgroup=1
pickupgroup=1
immediate=no
callerid=asreceived
callprogress=yes
musiconhold=default
channel => 1-24

# < features.conf sed 's/;.*//g' | grep -v ^$
[general]
parkext => 700
parkpos => 701-720
context => parkedcalls
pickupexten = *8

# < sip.conf sed 's/;.*//g' | grep -v ^$ | grep -v '^[  ]' | sed 
s/secret=.*/secret=donttell/g
[general]
context=default
callgroup=1
pickupgroup=1
port=5060
bindaddr=0.0.0.0
srvlookup=yes
disallow=all
allow=ulaw
allow=alaw
allow=g723.1
allow=g729
callgroup=1
pickupgroup=1
context=default
nat=no
canreinvite=yes
dtmfmode=rfc2833
incominglimit=4
[1310]
username=1310
secret=donttell
type=friend
host=dynamic
callerid=Grandstream SIP <1310>
[EMAIL PROTECTED]
[i1310]
username=i1310
secret=donttell
type=friend
host=dynamic
callerid=Grandstream SIP <1310>
[1311]
username=1311
secret=donttell
type=friend
host=dynamic
callerid=John Jacob Jingleheime <1311>
[EMAIL PROTECTED]
[1312]
username=1312
secret=donttell
type=friend
host=dynamic
callerid=Cisco 7960G Test <1312>
[EMAIL PROTECTED]

FWIW, I get identical behavior with callgroup=/pickupgroup= specified
for each extension. Here's some sanitized verbose output with SIP
debugging enabled:

-- Starting simple switch on 'Zap/24-1'
Jun 28 10:43:18 DEBUG[16774]: chan_sip.c:771 __sip_autodestruct: Auto 
destroying call 'a01052a-13c4-42c104ea-371e-1957'
Destroying call 'a01052a-13c4-42c104ea-371e-1957'
Jun 28 10:43:19 DEBUG[17450]: chan_zap.c:4242 zt_read: DTMF digit: 1 on Zap/24-1
Jun 28 10:43:19 DEBUG[17450]: chan_zap.c:4242 zt_read: DTMF digit: 3 on Zap/24-1
Jun 28 10:43:19 DEBUG[17450]: chan_zap.c:4242 zt_read: DTMF digit: 1 on Zap/24-1
Jun 28 10:43:20 DEBUG[17450]: chan_zap.c:4242 zt_read: DTMF digit: 2 on Zap/24-1
Jun 28 10:43:20 DEBUG[17450]: chan_zap.c:1381 zt_enable_ec: Enabled echo 
cancellation on channel 24
-- Executing Macro("Zap/24-1", "stdexten|1312|SIP/1312") in new stack
-- Executing Dial("Zap/24-1", "SIP/1312|20") in new stack
Jun 28 10:43:20 DEBUG[17450]: chan_sip.c:1309 create_addr: Setting NAT on RTP 
to 0
Jun 28 10:43:20 DEBUG[17450]: chan_sip.c:1487 sip_call: Outgoing Call for 1312
Jun 28 10:43:20 DEBUG[17450]: chan_sip.c:1620 update_user_counter: Call from 
user '1312' is 1 out of 0
We're at asterisk.server.ip.addr port 19630
Answering/Requesting with root capability 0x4 (ulaw)
Answering with preferred capability 0x8 (alaw)
Answering with preferred capability 0x1 (g723)
Answering with preferred capability 0x100 (g729)
Answering with non-codec capability 0x1 (telephone-event)
12 headers, 13 lines
Reliably Transmitting:
INVITE sip:[EMAIL PROTECTED]:5061 SIP/2.0
Via: SIP/2.0/UDP asterisk.server.ip.addr:5060;branch=z9hG4bK359ec760
From: "asterisk" ;tag=as61d8a13d
To: 
Contact: 
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Tue, 28 Jun 2005 17:43:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 17450 17450 IN IP4 asterisk.server.ip.addr
s=session
c=IN IP4 asterisk.server.ip.addr
t=0 0
m=audio 19630 RTP/AVP 0 8 4 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
 (no NAT) to called.phone.ip.addr:5061
-- Called 1312


Sip read:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP asterisk.server.ip.addr:5060;branch=z9hG4bK359ec760
From: "asterisk" ;tag=as61d8a13d
To: 
Call-ID: [EMAIL PROTECTED]
Date: Tue, 28 Jun 2005 17:43:20 GMT
CSeq: 102 INVITE
Server: CSCO/7
Contact: 
Content-Length: 0


10 headers, 0 lines
Jun 28 10:43:20 DEBUG[16774]: chan_sip.c:872 __sip_semi_ack: (Provisional) 
Stopping retransmission (but retaining packet) on '[EMAIL PROTECTED]' Request 
102: Found


Sip read:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP asterisk.server.ip.addr:5060;branch=z9hG4bK359ec760
From: "asterisk" ;tag=as61d8a13d
To: ;tag=001280b9cebf00025bfd45ed-7102ff29
Call-ID: [EMAIL PROTECTED]
Date: Tue, 28 Jun 2005 17:43:20 GMT
CSeq: 102 INVITE
Server: CSCO/7
Contact: 
Content-Length: 0


10 headers, 0 lines
Jun 28 10:43:20 DEBUG

RE: [Asterisk-Users] Group/Broadcast Voicemail

[Chris Stinson]

Were you guys able to figure this out?

-

Chris Stinson
Network Operations Center
ISDN-Net, Inc.
615-221-4200 x103
[EMAIL PROTECTED]

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Zen Kato
Sent: Wednesday, June 22, 2005 4:45 PM
To: asterisk-users@lists.digium.com; [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Group/Broadcast Voicemail

Hi,

Please see inline:

In Message-ID: <[EMAIL PROTECTED]>
Robert Goodyear <[EMAIL PROTECTED]> wrote  :

> 
> On Jun 22, 2005, at 1:50 PM, Zen Kato wrote:
> 
> > Hi Robert,
> >
> >> Let me guess... mailbox 5103 or 5203 were the last in the list to
> >> receive it?
> >
> > Every trials(1-6) I got only 51 mailboxes copied. My quick guess is
> > 256/5(u0103 and &xx03s)=51...1, so changing tmp[256] to tmp[4096]
> > does not work. 'Pseudo-diagram' as you mentioned before(6/8/05)
> > is desirable for expandability, but it also did not work.
> >
> >
> 
> So what about the variable BASEMAXINLINE? Did you change that and 
> recompile yet?
Yes, I changed #define BASEMAXLINE on step 5(line 80) and step 6(line
82) 
and recomiled each case.

Regards,

Zen Kato
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Re: RE : [Asterisk-Users] HowTo start DIAL by a sillent training as

[Doug Lytle]

[EMAIL PROTECTED] wrote:


Hello the list, hello Doug,

doesn't give the prompt tone or listen the DTMF before this time. 
If I place a "normal" call, it fails, because the first tones of my Asterisk

numbering sequence are missed.

 



The w option will cause a wait.  The more Ws, the longer the way.  The 
Ws can be intermixed within the number.  Hence, Dial(www555www1212) 
would result in a bit of a wait, dial 555, wait some more then dial 1212.


Doug

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Re: [Asterisk-Users] FXO/FXS cpu spikes, data loss and ztclock.

[Michael Welter]

qrss wrote:

Hey,  That's great news!  Are you running the TDM400P? Are you able to
successfully run FAX data through it?

Haven't tried yet.  The machine is about to be installed at a customer 
site.  It has a TDM04b.




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Re: [Asterisk-Users] Linksys WRT54GP2-NA settings for performance and low bandwidth?

[Greg Oliver]

Nothing you can do on this one..  Without the provider accepting your
QoS settings, you are at their mercy.  And yes, you are correct, most
multi-tenant dwellings use xDSL for their connectivity due to it's
price, and the upstream is usually less bandwidth than the downstream..

-Greg

On Tue, 2005-06-28 at 13:00 -0600, Paul Fielding wrote:
> So I'm using a WRT54GP2-NA when I travel, as I travel alot, to give me
> a phone at my hotel rooms, etc.   During the day or late at night the
> thing works great - best ATA I've ever used.
>  
> However, in the mid-evening (when many business travellers are at the
> hotel room doing work), the outgoing audio channel gets so choppy that
> the person on the other end can't make me out clearly.
> Interestingly, I can usually hear them just fine - I attribute that to
> larger incoming bandwidth than outgoing on the hotel's part.
>  
> This device has a *lot* of settings that one can tweak.   Anyone have
> any suggestions on tuning this thing (or tuning Asterisk or both) to
> improve the SIP performance of the audio from the Linksys to the
> server to try to reduce choppiness?   I note that Vonage, who also
> uses these devices, seems to have got it down - it doesn't seem to
> matter where I use my Vonage Linksys device, I can get pretty
> reasonable performance.   So I figure I should be able to do similar
> tweaks to mine... *shrug*
>  
> regards,
>  
> Paul
>  
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Re: RE : [Asterisk-Users] HowTo start DIAL by a sillent training as Wfor modems

[Steve Clark]

[EMAIL PROTECTED] wrote:

Hello the list, hello Doug,

Thank you, but I don't see any correct reply in this page.
I want to have a "silent header" of 1 or 2 seconds between to pick up the
line and before to start to sned the DTMF numbering, because my RTC provider
doesn't give the prompt tone or listen the DTMF before this time. 
If I place a "normal" call, it fails, because the first tones of my Asterisk

numbering sequence are missed.

If any idea...

Best Regards,
Francois BERGERET,
France.


-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Doug Lytle
Envoyé : mardi 28 juin 2005 12:25
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : Re: [Asterisk-Users] HowTo start DIAL by a sillent training as Wfor
modems



[EMAIL PROTECTED] wrote:



Hi all the list,

I am searching how to insert few seconds of silence just before to send 
the DTMF sequence via a FXO WildCard X101P to PSTN.


I remember that Hayes compatible modems knows a special character "W" 
that do a 1 sec pause. Is it possible to do something like this in DIAL 
line sequence ?






This should help:

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


Doug


This worked for me calling into an analog pbx that had an Audiocodes MP105-FXS 
connected to 4 incomming

lines on the pbx.

exten => _2XX,1,Dial(SIP/xxx.xxx.xxx.xxx,30,tD(w${EXTEN}))

^
HTH,
Steve

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Re: [Asterisk-Users] enabling stun on asterisk?

[Wojciech Tryc]

Asterisk doesn't have a STUN client. Most likely 
you wany to configure your clients (phones) which are behind non-symmetrical 
NATs to use STUN while connecting to Asterisk acting as your proxy.
Thanks,
Wojtek

  - Original Message - 
  From: 
  hank 
  
  To: Asterisk Users Mailing List - 
  Non-Commercial Discussion 
  Sent: Tuesday, June 28, 2005 2:34 
PM
  Subject: [Asterisk-Users] enabling stun 
  on asterisk?
  
  hello I am going to be setting up a stun server 
  on windows how do I enable it to work withasterisk?
  thanks
  hank
  
  

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Re: [Asterisk-Users] LiveVoip is Bankrupt - Why this thread

[trixter http://www.0xdecafbad.com]

On Tue, 2005-06-28 at 14:24 -0400, Andrew Kohlsmith wrote:
> On Tuesday 28 June 2005 13:49, hank wrote:
> > a man loose his mind?
> > what was the archive posting on that one? I want to read that :)
> 
> http://lists.digium.com/pipermail/asterisk-users/2005-April/098768.html
> 
> The whole thread's kind of funny...  He started off really calm and rational, 
> offering to donate time and space on his "modern, progressive" web forum with 
> all his experience and so on, but when we (the community) ultimately turned 
> him down he quite literally lost his mind, I think.
> 
> -A.
gmane.org appears to have the group
gmane.comp.telephony.pbx.asterisk.users (and biz and dev).  Gmane is a
free service that does smtp->nntp and back again.  Has a web interface
to read the groups, etc.  For those that dont want email they should be
able to use that, or at least use that as a feeder if they dont want to
deal with email delivery.

It also provides group searching for people who want to read archives
for more help.  

Perhaps the easiest way would be to have a mini-faq that explains how to
properly ask a question (what information should be provided when
asking) along with places to search for information (lists.digium.com,
gmane.org, google queries if you want to search, etc).  That way some of
the complaints about the list can be mitigated.  

To be honest I dont know if the list sent something like this on
subscription, but it may have.  Some of the other lists I am on have a
faq maintainer (so 100 people dont all do it at the same time) that
sends such a faq once per month, but that may be extreme.

http://dir.gmane.org/index.php?prefix=gmane.comp.telephony.pbx.asterisk

There are 10 lists with posts in them that can all be read that way.

Note: I have no affiliation with gmane, and from using it, it appears to
be sponsored by donations not banner ads, etc, so it is a clean
interface (with multiple interfaces, rss, blog style, threaded, etc).



-- 
Trixter http://www.0xdecafbad.com Bret McDanel
UK +44 870 340 4605   Germany +49 801 777 555 3402
US +1 360 207 0479 or +1 516 687 5200
FreeWorldDialup: 635378


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[Asterisk-Users] TDM04B Echo on Only One Channel

[Steve Totaro]

Hi,
 
I have a system that is experiencing profound echo 
on channel 4.  I switched POTS lines around to see if it was the line or 
the port and it is still a problem on port four.  Has anyone ever seen 
this before?
 
Thanks,
Steve
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Re: RE : [Asterisk-Users] HowTo start DIAL by a sillent training as Wfor modems

[Eric Wieling aka ManxPower]

If you are dialing out an analog Zap FXO port then you use the "w" 
option.  Dial(Zap/1/ww5551212)


[EMAIL PROTECTED] wrote:


Hello the list, hello Doug,

Thank you, but I don't see any correct reply in this page.
I want to have a "silent header" of 1 or 2 seconds between to pick up the
line and before to start to sned the DTMF numbering, because my RTC provider
doesn't give the prompt tone or listen the DTMF before this time. 
If I place a "normal" call, it fails, because the first tones of my Asterisk

numbering sequence are missed.

If any idea...

Best Regards,
Francois BERGERET,
France.


-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Doug Lytle
Envoyé : mardi 28 juin 2005 12:25
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : Re: [Asterisk-Users] HowTo start DIAL by a sillent training as Wfor
modems



[EMAIL PROTECTED] wrote:



Hi all the list,

I am searching how to insert few seconds of silence just before to send 
the DTMF sequence via a FXO WildCard X101P to PSTN.


I remember that Hayes compatible modems knows a special character "W" 
that do a 1 sec pause. Is it possible to do something like this in DIAL 
line sequence ?






This should help:

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


Doug

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RE: [Asterisk-Users] astcc no billed cost

[Ade Agbero]

Did anyone find a solution to this problem, I have tried all I know, but the problem persists
Nabeel Jafferali <[EMAIL PROTECTED]> wrote:
> Can anyone please help with an astcc problem. I just got it going, but> "billed cost" stays 0.> The test route is setup with "Inc. Seconds" = 6 and "Cost per additional> minute" = 1.> What can the problem be?If you're using local channels, search for a posting by me a week or two agoregarding this problem.--Nabeel JafferaliX2 Networkswww.x2n.caT: 1.647.722.69001.877.VOIP.X2NF: 1.866.655.6698FWD: 46990___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users
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calling worldwide with voicemail
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Re: [Asterisk-Users] Asterisk Realtime and ODBC

[Matthew Boehm]

Tim Lyakhovetskiy wrote:

Hello all!

My basic problem is that we haven't been able to get realtime to use 
ODBC to store configuration data. Here are the details:


Are you watching the asterisk debug log for any errors?

-Matthew

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Re: [Asterisk-Users] Voicemail max time length

[Johann]

Disregard.  I missed the obvious config file options :)

--johann

Johann wrote:

Is it possible to limit how long a voicemail can be?  Can it be done 
on a based on the length of the vm or the size of the file?


--johann
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Re: [Asterisk-Users] How do you handle NAT?

[Sebastian Silva]

Hi everyone.

1.  Asterisk as a SIP client behind nat, connecting to outside SIP Proxies:
#1 works with a NAT-supporting proxy as SIP Express router as the 
outside proxy. (Get an account at IPtel.org and try!). Fails with Free 
World Dialup.


2. Asterisk as a SIP client behind nat, connecting to inside SIP proxies:
#2 Works- no NAT in between

3. Asterisk as a SIP server behind nat, clients on the outside 
connecting to Asterisk:

#3 Works with port forwarding and some header mangling magic

4. Asterisk as a SIP server behind nat, clients on the inside connecting 
to Asterisk:

#4 Works - no NAT in between

5. Asterisk as a SIP client outside nat, connecting to outside SIP proxies:
#5 is no problem. No NAT in the middle

6. Asterisk as a SIP client outside nat, connecting to inside SIP proxies:
#6 is a problem if no port forwarding is done, similar to 3 above.

7. Asterisk as a SIP server outside nat, clients on the outside 
connecting to Asterisk:

#7 is no problem. No NAT in the middle

8. Asterisk as a SIP server outside nat, clients on the inside 
connecting to Asterisk:
#8 is solved with nat=yes and qualify=xxx in sip.conf for the client in 
most cases. Some clients (X-lite) assist themselves by using STUN and 
sending UDP keep-alive packets. Qualify sends keep-alive packets from 
Asterisk to the client on the inside.


from wiki

Now, if you net to define a NAT, you have to set asterisk to 
"canreinvite=no", "qualify=yes" and "nat=1".


Also, INSTEAD of NAT, you can use a STUN server. To use a STUN server 
you should set asterisk to "canreinvite=no", "qualify=no" and "nat=0" 
(the STUN configuration is in your agents).


Sebas

hank wrote:
how easy is it to set up a stun server? with asterisk amd will this fix 
part of the nat problem?

- Original Message - From: "Ray Van Dolson" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 


Sent: Tuesday, June 28, 2005 8:14 AM
Subject: Re: [Asterisk-Users] How do you handle NAT?



We've been feeling our way along with the NAT stuff (using SIP) as well.

At this point we are fairly small, so the keep-alive packets are not 
too bad.
What type of user load are you at and what are the specs on your 
Asterisk box?

I'm concerned we may run into this as well.

We do have the luxury that each Sipura device we use is sitting behind 
its own
NAT (a customer CPE).  So we can do port-forwarding and in combination 
with a
STUN server (MyStun), things work quite well.  The only issues left to 
deal
with are a lingering problem with ip_conntrack entries staying cached 
because

of the "keep alive" packets due to qualify=yes after the CPE's IP address
changes.

Curious to hear other's setups as well.  I would *love* to start using 
the
IAXy instead, but it has a couple shortcomings over the Sipura 2002's 
we're

using now:

- About $10/more
- Only has one line (apparently two lines is a bit more of a selling 
point).


Still trying to figure out a good way to make a case for the IAXy though.

Ray

On Tue, Jun 28, 2005 at 09:59:49AM -0500, Matthew Boehm wrote:


We are interested in how other people are handling NAT problems. We have
several customers all of which have some sort of firewall/NAT device at
their location. For simplicity sake, all customers' internal networks
are 192.168.*.*.

Our asterisk box is on public IP not blocked by any FW/NAT.

I use QUALIFY=yes on all our customers' phones and I feel that sending
out 80-something keep-alive packets is causing our box to crawl and
cause bad calls.

Would SER be better in this case? Should I have phones register with SER
instead of with Asterisk?

Thanks,
Matthew

P.S. Yes, I have read stuff on NAT on the wiki. I'm more interested in
other real world, working, solutions.


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--
Sebastian Silva
G R U P O  G A U S S
Depto. Sistemas
Av. Libertador 6250 4 piso
Tl.: 4 706- (int. 121)
[EMAIL PROTECTED]
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Re: [Asterisk-Users] Asterisk Realtime and ODBC

[Brian West]

try putting your sip.conf on disk and load peers/users from the  
database.


/b
---
Anakin: “You’re either with me, or you’re my enemy.”
Obi-Wan: “Only a Sith could be an absolutist.”

On Jun 28, 2005, at 1:36 PM, Tim Lyakhovetskiy wrote:


Hello all!

My basic problem is that we haven't been able to get realtime to  
use ODBC to store configuration data. Here are the details:


We've installed Asterisk on a CentOS machine as follows:
1. Downloaded, compiled, and installed FreeTDS 0.63
2. Downloaded, compiled, and installed  unixODBC 2.2.11
3. Downloaded, compiled, and installed Asterisk, Asterisk-Addons,  
and Zaptel from CVS HEAD


We checked the FreeTDS and ODBC connections several ways, using the  
"tsql" and "isql" utilities, and Ethereal for network monitoring.  
Everything worked perfectly. Then we set up asterisk as follows:




[settings]

sip.conf => odbc,asterisk,ast_config
sipusers => odbc,asterisk,sip_buddies
sippeers => odbc,asterisk,sip_buddies
extensions_table => odbc,asterisk,extensions_table



[default]
switch => Realtime/[EMAIL PROTECTED]



[asterisk]
dsn => "our DSN name"
username => "our username"
password => "our password"
pre-connect => yes



[modules]
autoload=yes

noload => pbx_gtkconsole.so
noload => pbx_kdeconsole.so
noload => app_intercom.so

load => chan_modem.so
load => res_musiconhold.so

noload => chan_alsa.so

noload => res_config_mysql.so
noload => cdr_addon_mysql.so
load => app_realtime.so
load => pbx_realtime.so

After that, we loaded up Asterisk, and there were no error  
messages. Not only that, the res_odbc and res_config_odbc modules  
loaded up just fine, AND Ethereal showed that a login to our MS SQL  
database went through. However, when we tried to add extensions or  
SIP friends, nothing worked. The strange thing was that Asterisk  
didn't even try to query the database (observed through Ethereal)  
and just tried to use the conf files.


Just to make sure we weren't crazy, we set up Asterisk using MySQL  
and that setup worked perfectly right from the start. We would  
still like to use ODBC->MS SQL though, because we would like to  
keep the amount of different databases minimal.


voip-info isn't much help on this, and we've looked far and wide  
for any more information regarding Asterisk ODBC setups, finding  
very little.


Anything we missed?

Thanks,
-Tim Lyakhovetskiy

_
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[Asterisk-Users] Linksys WRT54GP2-NA settings for performance and low bandwidth?

[Paul Fielding]

So I'm using a WRT54GP2-NA when I travel, as I 
travel alot, to give me a phone at my hotel rooms, etc.   During the 
day or late at night the thing works great - best ATA I've ever 
used.
 
However, in the mid-evening (when many business 
travellers are at the hotel room doing work), the outgoing audio channel gets so 
choppy that the person on the other end can't make me out clearly.   
Interestingly, I can usually hear them just fine - I attribute that to larger 
incoming bandwidth than outgoing on the hotel's part.
 
This device has a *lot* of settings that one can 
tweak.   Anyone have any suggestions on tuning this thing (or tuning 
Asterisk or both) to improve the SIP performance of the audio from the Linksys 
to the server to try to reduce choppiness?   I note that Vonage, who 
also uses these devices, seems to have got it down - it doesn't seem to matter 
where I use my Vonage Linksys device, I can get pretty reasonable 
performance.   So I figure I should be able to do similar tweaks to 
mine... *shrug*
 
regards,
 
Paul
 
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Re: [Asterisk-Users] pbx_extension_helper: No application 'agi'

[Tom Fielding]

Moises!

That fixed it!! Thanks very much!

Maybe I'm blind but nowhere in the installation instructions or even
on the AGI Wiki did I see reference to loading res_agi.so in
modules.conf.

Thanks,
Tom

On 6/28/05, Moises Silva <[EMAIL PROTECTED]> wrote:
> i could not follow this message from the beginning, but by the subject
> i may have an idea of whats going on.
> 
> does the command 'show applications' in asterisk console shows up AGI
> in its output?
> do you have load => res_agi.so in /etc/asterisk/modules.conf ???
> 
> best regards
> 
> On 6/28/05, Jay Milk <[EMAIL PROTECTED]> wrote:
> > That's incorrect, both versions represent valid syntax.
> >
> > To the OP -- show us the contents of test.agi
> >
> > > -Original Message-
> > > From: John Millican [mailto:[EMAIL PROTECTED]
> > > Sent: Tuesday, June 28, 2005 8:23 AM
> > > To: Tom Fielding; Asterisk Users Mailing List -
> > > Non-Commercial Discussion
> > > Subject: Re: [Asterisk-Users] pbx_extension_helper: No
> > > application 'agi'
> > >
> > >
> > > exten => s,1,agi(test.agi)
> > > should be
> > > exten => s,1,agi,test.agi
> > > If there any arguments to send the script use
> > > exten => s,1,agi,test.agi|args_to_pass
> > >
> > > John M
> > > ___
> > > Asterisk-Users mailing list
> > > Asterisk-Users@lists.digium.com
> > > http://lists.digium.com/mailman/listinfo/aster> isk-users
> > > To
> > > UNSUBSCRIBE or update options visit:
> > >
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
> >
> > ___
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> > http://lists.digium.com/mailman/listinfo/asterisk-users
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> >
> 
> 
> --
> "Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org";
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RE : [Asterisk-Users] HowTo start DIAL by a sillent training as Wfor modems

[f6hqz-m]

Hello the list, hello Doug,

Thank you, but I don't see any correct reply in this page.
I want to have a "silent header" of 1 or 2 seconds between to pick up the
line and before to start to sned the DTMF numbering, because my RTC provider
doesn't give the prompt tone or listen the DTMF before this time. 
If I place a "normal" call, it fails, because the first tones of my Asterisk
numbering sequence are missed.

If any idea...

Best Regards,
Francois BERGERET,
France.


-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Doug Lytle
Envoyé : mardi 28 juin 2005 12:25
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : Re: [Asterisk-Users] HowTo start DIAL by a sillent training as Wfor
modems



[EMAIL PROTECTED] wrote:

>Hi all the list,
>
>I am searching how to insert few seconds of silence just before to send 
>the DTMF sequence via a FXO WildCard X101P to PSTN.
>
>I remember that Hayes compatible modems knows a special character "W" 
>that do a 1 sec pause. Is it possible to do something like this in DIAL 
>line sequence ?
>
>  
>
This should help:

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


Doug

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Re: [Asterisk-Users] FXO/FXS cpu spikes, data loss and ztclock.

[qrss]

Hey,  That's great news!  Are you running the TDM400P? Are you able to
successfully run FAX data through it?

-Original Message-
From: Michael Welter
Sent: Tue, June 28, 2005 12:57 pm

FYI, I'm not seeing CPU spikes on the Tyan S2850 mobo (single Opteron
> CPU).



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RE: [Asterisk-Users] VoipJet TOS (was Teliax and also LiveVoip)

[Wiley Siler]

One would assume they have better things to do as they are quite busy.
I think this is just a proactive measure meaning they say you cannot do
it upfront so that in the event of a problem, it was predeclared.  As to
the rest of the TOS, I could be wrong but I got the impression that the
owner of VoipJet speaks English as a second language due to some email
exchanges.  If that is the case, the TOS issue can just be one of
cultural and language variation.  I could be totally wrong though...

Regardless, one can assume that the ability to listen to calls is there.
It just could not possibly be worth anyone's time to do so.

Unless you have nuclear secrets or something I would not worry about
listeners...

Cheers,
Wiley


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of hank
Sent: Tuesday, June 28, 2005 10:57 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Re: teliax [Was: LiveVoip is Bankrupt]

how do they know if your calling your tax dude or something? what do
they do 
monitor the calls or something?

- Original Message - 
From: "John Goerzen" <[EMAIL PROTECTED]>
To: 
Sent: Tuesday, June 28, 2005 6:46 AM
Subject: [Asterisk-Users] Re: teliax [Was: LiveVoip is Bankrupt]


> On 2005-06-28, r00t <[EMAIL PROTECTED]> wrote:
>> I'll second voipjet for outbound only. While many reported problems
to
>
> VoipJet bothers me for two reasons.  First, their terms of service are
> absolutely insane.  Users are specifically forbidden to place calls
> regarding medical or financial matters over VoipJet.  So I couldn't
call
> my tax preparer or schedule a doctor appointment under their contract.
> There are many other insane things about it; see the thread at
>
http://lists.digium.com/pipermail/asterisk-users/2005-March/094251.html.
> One of the little gems is that if you tell anyone you use VoipJet, you
> violate your contract.  So all of you that have been praising your
> VoipJet service here: prepare to be disconnected! :-)
>
> Secondly, they are not honest about what they are doing.  They clearly
> are aiming some services at small self-sufficient end-users, yet they
> claim to provide services to commercial carriers only (and their ToS
> tries to enforce that.)  Got to love little statements like "Emerging
> VoIP service providers can make payments through PayPal".
>
> -- John
>
>
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[Asterisk-Users] Asterisk Realtime and ODBC

[Tim Lyakhovetskiy]

Hello all!

My basic problem is that we haven't been able to get realtime to use ODBC to 
store configuration data. Here are the details:


We've installed Asterisk on a CentOS machine as follows:
1. Downloaded, compiled, and installed FreeTDS 0.63
2. Downloaded, compiled, and installed  unixODBC 2.2.11
3. Downloaded, compiled, and installed Asterisk, Asterisk-Addons, and Zaptel 
from CVS HEAD


We checked the FreeTDS and ODBC connections several ways, using the "tsql" 
and "isql" utilities, and Ethereal for network monitoring. Everything worked 
perfectly. Then we set up asterisk as follows:




[settings]

sip.conf => odbc,asterisk,ast_config
sipusers => odbc,asterisk,sip_buddies
sippeers => odbc,asterisk,sip_buddies
extensions_table => odbc,asterisk,extensions_table



[default]
switch => Realtime/[EMAIL PROTECTED]



[asterisk]
dsn => "our DSN name"
username => "our username"
password => "our password"
pre-connect => yes



[modules]
autoload=yes

noload => pbx_gtkconsole.so
noload => pbx_kdeconsole.so
noload => app_intercom.so

load => chan_modem.so
load => res_musiconhold.so

noload => chan_alsa.so

noload => res_config_mysql.so
noload => cdr_addon_mysql.so
load => app_realtime.so
load => pbx_realtime.so

After that, we loaded up Asterisk, and there were no error messages. Not 
only that, the res_odbc and res_config_odbc modules loaded up just fine, AND 
Ethereal showed that a login to our MS SQL database went through. However, 
when we tried to add extensions or SIP friends, nothing worked. The strange 
thing was that Asterisk didn't even try to query the database (observed 
through Ethereal) and just tried to use the conf files.


Just to make sure we weren't crazy, we set up Asterisk using MySQL and that 
setup worked perfectly right from the start. We would still like to use 
ODBC->MS SQL though, because we would like to keep the amount of different 
databases minimal.


voip-info isn't much help on this, and we've looked far and wide for any 
more information regarding Asterisk ODBC setups, finding very little.


Anything we missed?

Thanks,
-Tim Lyakhovetskiy

_
Express yourself instantly with MSN Messenger! Download today - it's FREE! 
http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/


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RE: [Asterisk-Users] Shoutcast Music On Hold problems?

[Huddleston, Robert]

Worked for me with a different stream... I ran into this same problem before - 
but it was my own fault for not RTM... Both the manual and ast install advised 
of verifying correct version of mpg123... I had wrong version and thus got no 
noise...
If you follow the directions explicitly laid out on the wiki you should have no 
problems.
I use christianrock.net's shoutcast stream
Like this in musiconhold.conf
default => quietmp3:/var/lib/asterisk/mohmp3-empty,http://209.51.128.160:5112/



-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of hank
Sent: Tuesday, June 28, 2005 1:41 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Shoutcast Music On Hold problems?

I tried that the stream i tried to use orriginally was
http://209.97.198.50:30518
all I get is silence when I put the person on hold thanks hank
- Original Message -
From: "Patrick" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Sent: Tuesday, June 28, 2005 2:50 AM
Subject: Re: [Asterisk-Users] Shoutcast Music On Hold problems?


> On Mon, 2005-06-27 at 22:51 -0700, hank wrote:
>>  mp3:/var/lib/asterisk/mohmp3-empty,http://www.waixwave.com:8000/
>
> I haven't tried this myself but if I put www.waixwave.com:8000 in
> Firefox I get "connection refused". Try another site that actually
> streams music. Shoutcast.org should have a nice list.
>
> Regards,
> Patrick
>
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[Asterisk-Users] enabling stun on asterisk?

[hank]

hello I am going to be setting up a stun server on 
windows how do I enable it to work withasterisk?
thanks
hank
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Re: [Asterisk-Users] How do you handle NAT?

[hank]

I am fighting this as we speak
I have a friend who can't connect to me cause of a damn nat frankly its 
irritating me

so any recommendations are welcome
- Original Message - 
From: "Matthew Boehm" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 


Sent: Tuesday, June 28, 2005 7:59 AM
Subject: [Asterisk-Users] How do you handle NAT?


We are interested in how other people are handling NAT problems. We have 
several customers all of which have some sort of firewall/NAT device at 
their location. For simplicity sake, all customers' internal networks are 
192.168.*.*.


Our asterisk box is on public IP not blocked by any FW/NAT.

I use QUALIFY=yes on all our customers' phones and I feel that sending out 
80-something keep-alive packets is causing our box to crawl and cause bad 
calls.


Would SER be better in this case? Should I have phones register with SER 
instead of with Asterisk?


Thanks,
Matthew

P.S. Yes, I have read stuff on NAT on the wiki. I'm more interested in 
other real world, working, solutions.


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Re: [Asterisk-Users] LiveVoip is Bankrupt - Why this thread

[Tzafrir Cohen]

Hi

On Tue, Jun 28, 2005 at 10:49:29AM -0700, hank wrote:
> a man loose his mind?
> what was the archive posting on that one? I want to read that :)

Thanks for the kind, tactful and on-topic post. Also thank you for 
trimming irrelevant quoted text from your message.

Shall we drop this?

-- 
Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
http://tzafrir.org.il |   | a Mutt's  
[EMAIL PROTECTED] |   |  best
ICQ# 16849755 |   | friend
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Re: [Asterisk-Users] LiveVoip is Bankrupt - Why this thread

[Andrew Kohlsmith]

On Tuesday 28 June 2005 13:49, hank wrote:
> a man loose his mind?
> what was the archive posting on that one? I want to read that :)

http://lists.digium.com/pipermail/asterisk-users/2005-April/098768.html

The whole thread's kind of funny...  He started off really calm and rational, 
offering to donate time and space on his "modern, progressive" web forum with 
all his experience and so on, but when we (the community) ultimately turned 
him down he quite literally lost his mind, I think.

-A.
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[Asterisk-Users] Mitel SX2000 Integration

[David Sampson]

I have a Mitel SX2000 with no voicemail.  I’m
wondering if it would be possible to use Asterisk to meet the need.  This is a
hotel property with mostly analog extensions.

Dave

 






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[Asterisk-Users] Voicemail max time length

[Johann]

Is it possible to limit how long a voicemail can be?  Can it be done on 
a based on the length of the vm or the size of the file?


--johann
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Re: [Asterisk-Users] Re: teliax [Was: LiveVoip is Bankrupt]

[hank]

how do they know if your calling your tax dude or something? what do they do 
monitor the calls or something?


- Original Message - 
From: "John Goerzen" <[EMAIL PROTECTED]>

To: 
Sent: Tuesday, June 28, 2005 6:46 AM
Subject: [Asterisk-Users] Re: teliax [Was: LiveVoip is Bankrupt]



On 2005-06-28, r00t <[EMAIL PROTECTED]> wrote:

I'll second voipjet for outbound only. While many reported problems to


VoipJet bothers me for two reasons.  First, their terms of service are
absolutely insane.  Users are specifically forbidden to place calls
regarding medical or financial matters over VoipJet.  So I couldn't call
my tax preparer or schedule a doctor appointment under their contract.
There are many other insane things about it; see the thread at
http://lists.digium.com/pipermail/asterisk-users/2005-March/094251.html.
One of the little gems is that if you tell anyone you use VoipJet, you
violate your contract.  So all of you that have been praising your
VoipJet service here: prepare to be disconnected! :-)

Secondly, they are not honest about what they are doing.  They clearly
are aiming some services at small self-sufficient end-users, yet they
claim to provide services to commercial carriers only (and their ToS
tries to enforce that.)  Got to love little statements like "Emerging
VoIP service providers can make payments through PayPal".

-- John


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Re: [Asterisk-Users] How do you handle NAT?

[hank]

how easy is it to set up a stun server? with asterisk amd will this fix part 
of the nat problem?
- Original Message - 
From: "Ray Van Dolson" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 


Sent: Tuesday, June 28, 2005 8:14 AM
Subject: Re: [Asterisk-Users] How do you handle NAT?



We've been feeling our way along with the NAT stuff (using SIP) as well.

At this point we are fairly small, so the keep-alive packets are not too 
bad.
What type of user load are you at and what are the specs on your Asterisk 
box?

I'm concerned we may run into this as well.

We do have the luxury that each Sipura device we use is sitting behind its 
own
NAT (a customer CPE).  So we can do port-forwarding and in combination 
with a
STUN server (MyStun), things work quite well.  The only issues left to 
deal
with are a lingering problem with ip_conntrack entries staying cached 
because

of the "keep alive" packets due to qualify=yes after the CPE's IP address
changes.

Curious to hear other's setups as well.  I would *love* to start using the
IAXy instead, but it has a couple shortcomings over the Sipura 2002's 
we're

using now:

- About $10/more
- Only has one line (apparently two lines is a bit more of a selling 
point).


Still trying to figure out a good way to make a case for the IAXy though.

Ray

On Tue, Jun 28, 2005 at 09:59:49AM -0500, Matthew Boehm wrote:

We are interested in how other people are handling NAT problems. We have
several customers all of which have some sort of firewall/NAT device at
their location. For simplicity sake, all customers' internal networks
are 192.168.*.*.

Our asterisk box is on public IP not blocked by any FW/NAT.

I use QUALIFY=yes on all our customers' phones and I feel that sending
out 80-something keep-alive packets is causing our box to crawl and
cause bad calls.

Would SER be better in this case? Should I have phones register with SER
instead of with Asterisk?

Thanks,
Matthew

P.S. Yes, I have read stuff on NAT on the wiki. I'm more interested in
other real world, working, solutions.

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Re: [Asterisk-Users] Revision I Board TDM04b

[Steve Totaro]

So does the Digium mailing list :0

- Original Message - 
From: "Tim Pushor" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"

Sent: Tuesday, June 28, 2005 10:06 AM
Subject: Re: [Asterisk-Users] Revision I Board TDM04b


> Call Digium? They do provide installation support 
>
> Steve Totaro wrote:
>
> > I cannot get this thing to work.  Anyone know of any tricks?
>
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[Asterisk-Users] initial setup: problem

[Sean Browne (Cardiff IT Support Ltd)]

hi,

i just installed AAH. from the bootable CDROM disk after i downloaded iso 
image.


AAH box is at http://10.10.10.10

I use browser to access the box, and can get to main menu and voicemail 
login and sugar login.


However, the asterisk management http://10.10.10.10/admin gives a 404 error, 
page unavailable.


Any ideas? should I re-install?

thanks,

sean 


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RE: [Asterisk-Users] This weeks Developer meeting

[Dean Collins]

Hi Brian,
I didn't know there was a developers call in existence.

How often does it occur, is there an agenda? Homepage? How many people
dial in for this?

Cheers,
Dean


> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Brian West
> Sent: Tuesday, 28 June 2005 12:21 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Cc: Asterisk Developers Mailing List
> Subject: [Asterisk-Users] This weeks Developer meeting
> 
> IAX2/[EMAIL PROTECTED]/996 at 1pm CDT on Thurday the 30th.
> 
> If you have any topics that need to be covered please direct them to
me.
> 
> Thanks,
> /b
> ---
> Anakin: "You're either with me, or you're my enemy."
> Obi-Wan: "Only a Sith could be an absolutist."
> 
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Re: [Asterisk-Users] LiveVoip is Bankrupt

[Walt Reed]

On Tue, Jun 28, 2005 at 10:59:31AM -0400, steve szmidt said:
> On Monday 27 June 2005 20:04, Robert Webb wrote:
> > > I agree with that fact the same questions get posted, but
> > > that problem is compounded by the fact the archives are not
> > > really searchable. If the were as lease some users would search.
> > > The archives need to be fully indexed.
> >
> > In a Google search box: site:lists.digium.com "What you are searching
> > for"
> 
> The problem many newbies faces is TOO MUCH information. Not being able to see 
> the trees because of the forest basically.
> 
> It does not matter either if it has been discussed until someone went crazy 
> or 
> died. The reason it keeps coming up is because it has not been solved.

The problem is NOT that the archives are not searchable or indexed. The
problem is that we are dealing with a very complex subject /
application. Telephony is Very different from general computing. The
terminology, technology, typical problem set is unique to the industry.
Now we add all the typical computer and network issues on top such as
IRQ's, QoS, firewalls, NAT, etc. 

Newbies don't have a chance unless they are willing to spend the time it
takes to learn about the technology they are trying to implement.

Maybe some day this will change as hardware gets better, easier to
configure, and the software matures. People that want something that
"just works" without "tinkering" NOW should either be NOT using VoIP
technology, or should be purchasing a complete solution (or consulting
services) from one of the many vendors available.

* and VoIP are still very young, about where Apache was as a web
server (and the internet in general) back in 1996 - pretty stable,
but still immature.


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[Asterisk-Users] Voice Mail hangup on not messages

[David Romero]

i have a problem whit voice mail  on [EMAIL PROTECTED] whit CVS HEAD

when try to use voice mail  and i have at least 1 message all work ok...

when try to use voice mail  and not have messages asterisk hangup
after type my voicemail password  whitout any notification.

how i can do a "not messages found " notification and enable the 
others voice mail options like record not available message.


thanks-- David Romero##
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Re: [Asterisk-Users] Junghanns 4 port BRI problem

[Kristof Hardy]

Doug Reid - Stormcorp wrote:

I have a Junghanns BRI 4 port installed where only the first channel
of each line is working i.e. channels 1 and 4 work but 2 and 5 don't.
Our config is the same on this box as 15 other similar installations
where all works well. the only error I see is in /var/log/messages:


Not sure if it's related, but I had the same error once when I used a 
wrong parameter in zapata.conf. For signalling I had something 
incorrect, but after i changed it to the following, it was okay:


signalling = bri_cpe


Cheers
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Re: [Asterisk-Users] ASTCC not billing

[Bernard Cresencia]

sorry, I meant my.cnf, not my.conf.

Once logging is enabled, I would do tail -f
/var/log/myslqd.log and watch as the database is being
accessed during a call.


--- Ade Agbero <[EMAIL PROTECTED]> wrote:

> I have just installed [EMAIL PROTECTED] version 1.1, I
> have made a number of successful calls, but the bill
> cost remains "0".
>  
> I have kept everything simple:
>  
> PatternCommentTrunksConnect FeeInc. SecondsCost per
> additional minute [input] DANSAM0010
> 
>  
> Trunk NameTechnologyPeer/Trunk [input]
> TESTSIP213.45.62.117
>  
> 
> Darren Wiebe <[EMAIL PROTECTED]> wrote:
> We are using it on [EMAIL PROTECTED] version 1.1
> 
> Darren Wiebe
> [EMAIL PROTECTED]
> 
> 
> Darren Wiebe wrote:
> 
> > Ade Agbero wrote:
> >
> >> I set "notransfer=yes" and on SIP you set
> "canreinvite=no", but 
> >> ASTCC is still not billing.
> >> 
> >> I formated and reinstalled [EMAIL PROTECTED] and 
> >> got the latest CVS of Astcc, but ASTCC is still
> not billing.
> >> 
> >> What version of [EMAIL PROTECTED] can be confirmed 
> >> working with Astcc.
> >
> >
> > I can try to find that out tomorrow. I don't know
> off the top of my 
> > head. Did you try the latest stable version of
> asterisk? That is 
> > what I did to resolve the issue.
> >
> > Darren Wiebe
> > [EMAIL PROTECTED]
> >
> >>
> >> */Darren Wiebe /* wrote:
> >>
> >> On IAX you set "notransfer=yes" and on SIP you
> set "canreinvite=no"
> >>
> >> Darren
> >>
> >> Juan Luis Moyano wrote:
> >>
> >> >On Sab, 25 de Junio de 2005, 5:20 pm, Darren
> Wiebe dijo:
> >> >
> >> >
> >> >>Do you have the notransfer and reinvite lines
> set properly? I
> >> had this
> >> >>same problem with ASTCC but found that if I
> removed asterisk
> >> including
> >> >>the source and did a clean reinstall it worked
> suddenly.
> >> >>
> >> >>Darren
> >> >>
> >> >>
> >> >>
> >> >Darren, how is the proper way of setting
> notransfer and
> >> canreinvite lines
> >> >on IAX. TIA.
> >> >
> >> >
> >> >
> >>
> >> ___
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> >>
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> >>
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> >>
> >>
>

> >> How much free photo storage do you get? Store
> your holiday snaps for 
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Re: [Asterisk-Users] FXO/FXS cpu spikes, data loss and ztclock.

[Michael Welter]

FYI, I'm not seeing CPU spikes on the Tyan S2850 mobo (single Opteron CPU).


Rich Adamson wrote:

I have not really tried any other values for N1, M1 or CGM.  I actually
used the formulas from the Si3035 data sheet to calculate what they
"should be" for 8Khz.  There's a lot of math in there, but it looks like
there may be several ways to arrive at the same output values.  Not sure
if using a different calculation for the different dividers might give
better results using the same crystal or not.  This was my first shot at
it but your idea seems like a good one.



It might be possible to change the values slightly to judge their impact.
I've not done the math, so not sure if changing the values has any real
merit.



I'm not sure what profiling tools might be useful, but would be delighted
to hear any suggestions that anyone can contribute.  It really appears
that things are choking up somewhere in the interrupt handling routines
and I'm guessing somewhere in the zaptel driver.



I'm not a proficient programmer at all, but some experienced programmers
use various profiling tools to help understand which routines are consuming
cycles. It would seem like that could be used to help isolate the 
repetitive cpu spikes.
 


If the problem turns out to be a timing sync problem due to oversampling a
sample or so per second, then the best solution may be a hardware one. 



Its my understanding (which could be incorrect) the clock on the TDM card
is used for two purposes. First to drive the onboard chipset and second
to generate an interrupt on a recurring basis. And, that same interrupt is
used to "time" or "sync" other functions within asterisk. At least that
has been the argument behind "do you have a zaptel timing device". Each
of the digium cards seem to use that same architecture, however it also
seems the TDM card is the only card that leaves something on the table.

So, is the missed data resulting from:
 1. pcm data arriving to fast/slow on the card for the pci controller to
cause an interrupt and transfer the data across the bus reliably?
 2. to much time spent handling the interrupt within asterisk drivers
causing an interrupt to be missed (or delayed service)?
 3. timing design conflicts between clocking the 3050 (pcm conversation)
verses interrupt requirements?
 4. potential problems in the pci controller design?

I would have to believe the clock is driving the pcm encoding function
within the 3050 chip, and the design objective is to cause the chip to
encode exactly 8,000 samples per second. Therefore, changing that 
clocking mechanism is likely to generate 7,990 or 8,010 samples (or

some other non-standard rate) that is likely to negatively impact other
asterisk functions (due to the reliance on the interrupts as a timing
source). But, the flip side of that would suggest the existing design
is running at some rate other then 8,000 samples/sec now.

For the TDM card, there is no such thing as syncing its clock to anything
since its handling incoming analog audio that contains no such info.



I'm still trying to get a handle on exactly how the overall system timing
works with the zaptel driver.  It does not seem like even multiple
(non-t1) cards of the same type in an asterisk system sync their clocks. 
For example, each seems to bring data into the system according to the
timing of it's own internal oscillator.  



I believe that is correct and was very likely one of the driving forces
in the design of the TDM card (e.g., one interrupt handling four pstn
lines as opposed to multiple x100p cards each with their own interrupt
servicing requirements.



That's my assessment of the wcfxo
style cards at least.  The TDM400 seems to derive it's clock a little
differently.  Perhaps somebody could jump in and shed a little light on
how the hardware clocking works for that card.  It seems that overall the
basic theory of operation is quite similar - Tiger Jet 320 PCI controller,
DAA (or SLIC for FXS) etc.  As far as I know, the problems of CPU spikes
and data loss are not apparent on a properly configured T1 setup.



I don't believe anyone has confirmed the cpu spikes are actually 
responsible for missed frames. At least I won't assume that for now.


The T1 card is different since a properly configured card will sync its
onboard clock with an external source that is considered highly accurate.
When the clock is in sync, there is no such thing as missed pcm frames
on a T1 card. But, I'm sure you're read the various postings from folks
that did not properly define the card sync and those postings generally
relate to audio clicks (and other disturbances) that are essentially the
same apparent issues as a free-wheeling TDM clock.



I think that any data that we can gain from others running vmstat 1
(looking for cpu spikes) in combination with running ztclock would be
useful.  Especially on differing hardware including the various T1 cards. 
ztclock is looking pretty good to me on my hardware,