date:20070806

This is an option in the queue application. You could just create a
queue for that single user.

Thanks,
Steve

Jeng Yu wrote:
Hi All,

The Asterisk book (the pdf version) is excellent!! I want to thank all
the guys that put it together. I am most grateful for it.

There is something about writing a dialplan that I'm not clear about.
What I'm trying to figure out how to do is this: when I transfer a
call to the destination number, and the person called picks up the
call, I want to play a greeting message to the person called for a few
seconds before Asterisk bridges the two parties to talk. In other
words, only the person called should hear the greeting message, not
the person calling. How do I do this in Asterisk?

Why do you need something like this, you ask? Simple. I want to put
bulletin messages, reminder messages, corporate communication
snippets, etc for employees to hear - no more than 3 seconds.

Thanks for your suggestions.

Jeng

Yahoo! Answers - Get better answers from someone who knows. Try it now
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Re: [asterisk-users] asterisk always rining phone

If you setup your dialplan correctly and use priority jumping (off by 
default), you can play congestion or you could play the unavailable 
message for voicemail an then have the person who's phone it is record a 
message saying that they are on the phone, please leave a message.

Thanks,
Steve Totaro

satish patel wrote:
 is there any way to send busy tone to the calling party measn when i 
 call 2 some body and phone would be busy then i got busy tone so i can 
 guess party still talking 2 somebody...

 */Steve Totaro [EMAIL PROTECTED]/* wrote:

 Sounds like you have call waiting on the phones. You can disable this
 on the Asterisk side. To verify, make a call on your phone and then
 dial yourself from another phone. Depending on the phone, you will
 have
 some sort of indication that a second call is coming in.

 Thanks,
 Steve Totaro

 satish patel wrote:
  Dear all
 
  I have setup of asterisk 1.2.14 with 100 SIP phone
  and it is working fine but thing is that when i call to somebody on
  local extention my asterisk not give me notification like party
 phone
  is busy or busy tone alway it give me rining single how can i
 justify
  other party is not pickup the phone or he/she talking with
 somebody on
  phone caz my phone rining on both stages is there any special
  configuration for it ??
 
 
 
 
 
 
 
  Got a little couch potato?
  Check out fun summer activities for kids.
 
 
 
 
 
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Re: [asterisk-users] Difference between WaitExten and TIMEOUT (response)

Michiel van Baak wrote:
 On 05:27, Fri 03 Aug 07, bilal ghayyad wrote:
   
 Hi List;

 What is the difference between WaitExten function and
 TIMEOUT (response)? As I see that both are used to
 determine the allowed time to enter the digits, any
 one can advise?
 

 WaitExten is waiting for you to type an extension.
 TIMEOUT is to set the default timeout for promtps in IVR and
 stuff but is not actually waiting for you to provide an
 extension
   
More specifically, timeout is the time between dialing digits when using 
WaitExten or background for asterisk to decide you are done dialing an 
option or extension and place the call.

Anthony

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Re: [asterisk-users] Sangoma PRI

Stephen Bosch wrote:
 Steve Totaro wrote:
   
 Note to Digium

 I wish I could upgrade my wct4xxp drivers locally.  I still have the v1 
 firmware on my card.

 It is kind of hard (next to impossible) to pull it from a production 
 machine and ship it to Digium.  That might take a week if all goes well.
 

 The only way this will ever happen is if Digium completely redesigns the
 card, which is a long way of saying that you will buy a new card before
 you have that request filled.

 This is one of the great things about the Sangoma hardware -- it was
 designed to be fully field upgradeable (they use an FPGA
 architecture). The design approach is worth emulating.

 -Stephen-

   

I will go Sangoma from now on with maybe a few test systems running the 
Tormenta III boards which can be had for under $500 for four ports.  
They also fit 3.3v and 5v.

Thanks,
Steve Totaro


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[asterisk-users] Cant Play gsm file

2007-08-06 Thread atik

Hi,

i am having problem on playing asterisk sound file on my new installed
asterisk..
i have the following extension , if i call from any SIP / IAX  phone
playback or voicemail doesnt play anything  but when i dial 102, I
hear the MP3 music ..

exten = 99,1,Answer()
exten = 99,2,Playback(prepaid-welcome)
exten = 99,3,Hangup()

exten = 101,1,VoiceMailMain()

exten = 102,1,Answer()
exten = 102,2,MusicOnHold(default)

I have format_gsm.so, codec_gsm.so loaded and i am using
asterisk-sounds-1.2.1  , asterisk-1.2.23 on debian 4.0.

do i miss any audio library?

thanks
atik

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Re: [asterisk-users] Teliax Quality of Service

Tim Panton wrote:
 On 5 Aug 2007, at 06:54, Douglas Garstang wrote:

   
 I don't think creating a network without a single point of failure  
 is unreasonable.
 

 It's impossible. I can't think of a single example where this  
 actually exists.

 Getting even close is hideously expensive.

 Tim, speaking for himself :-)

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In fact, the only people who would say something like this are folks who 
have never PHYSICALLY implemented a network, they simply don't 
understand the limitations involved.

Anthony

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Re: [asterisk-users] Before Bridging Two Calls

On Mon, 2007-08-06 at 10:05 +0100, Jeng Yu wrote:
 The Asterisk book (the pdf version) is excellent!! I want to thank all
 the guys that put it together. I am most grateful for it.

I'm glad you enjoyed the book.

 I want to play a greeting message to the person called for a few
 seconds before Asterisk bridges the two parties to talk. 

One way to do this is with the A option to the Dial() application.  For
example, your extension might look like this:

exten = 123,1,Dial(SIP/some_phone,30,A(hello-world))

This would play the Hello World prompt to the called party before
bridging the two calls together.


-- 
Jared Smith
Community Relations Manager
Digium, Inc.


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[asterisk-users] Setting gain levels with mISDN

2007-08-06 Thread Andrea Spadaccini

Hello everybody,
I'm aware that I can try to balance gain levels with PSTN cards using the
ztmonitor tool, as described in 
http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current/docs-html/x1695.html
(Adjusting the rxgain/txgain Settings).

Is there a similar tool for mISDN? If not, what is your approach to gain
setting in mISDN?

Thanks in advance,

-- 
Dr. Andrea Spadaccini
Multimedia Technologies Institute - MTI S.r.l.
Web: www.x-voice.it - Tel: +39 (0) 95 7224945

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Re: [asterisk-users] IAX2 - DualServer Problem

On Sat, 2007-08-04 at 12:41 +0300, Mustafa Sakalsiz wrote:
 Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REJECT
Timestamp: 1ms  SCall: 4  DCall: 3 [10.10.10.73:4569]
CAUSE   : No authority found
CAUSE CODE  : 50

This no authority found message means that one server is rejecting the
other server's authentication.  This is most likely a username,
password, or authentication mismatch.

-- 
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Community Relations Manager
Digium, Inc.


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Re: [asterisk-users] asterisk always rining phone

On Mon, 2007-08-06 at 09:50 -0400, Steve Totaro wrote:
 If you setup your dialplan correctly and use priority jumping (off by 
 default), you can play congestion or you could play the unavailable 
 message for voicemail an then have the person who's phone it is record a 
 message saying that they are on the phone, please leave a message.

Or, you can leave priority jumping off, and simply look at the channel
variable named DIALSTATUS after you've called the Dial() application.
The DIALSTATUS variable will tell you why the Dial() application was
unable to bridge the calls.


-- 
Jared Smith
Community Relations Manager
Digium, Inc.


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[asterisk-users] ATA phones ring when they register

2007-08-06 Thread Vieri

Hi,

I have an 8-port Grandstream GXW-4008 V1.2A ATA
converter with analog phones connected to it.

They work fine except for just one feature I would
like to modify. Somehow, each time the ATA
re-registers the SIP clients or each time the device
has to be rebooted for maintenance, the phones ring
once. This feature can be useful as it notifies the
user of the re-registration.

However, it is quite of a problem to have 8+ phones
all ring at the same time and most users will get
confused and will pick them up.

So I would like to know how to disable this audio
notification and disable phone ringing for this
event.

What I don't know yet is if it's a purely ATA
config-related issue or if I also need to change
Asterisk's settings.

I tried qualify=no and the phones still ring.
I tried several combinations in the ATA config but the
phones still ring when they register.

I'm sure this is simple to solve but I just can't find
the right option.

Has anyone seen this behavior in the GXW-4008 ATA or
similar?

Thanks,

Vieri



   

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Sims Stories at Yahoo! Games.
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Re: [asterisk-users] asterisk always rining phone

2007-08-06 Thread satish patel

i want more example of extention.conf i have find many on google but it is 
documented i want live example if u have extention.conf can u send me working 
extention.conf i m new for asterisk so that send me one file 


Jared Smith [EMAIL PROTECTED] wrote: On Mon, 2007-08-06 at 09:50 -0400, Steve 
Totaro wrote:
 If you setup your dialplan correctly and use priority jumping (off by 
 default), you can play congestion or you could play the unavailable 
 message for voicemail an then have the person who's phone it is record a 
 message saying that they are on the phone, please leave a message.

Or, you can leave priority jumping off, and simply look at the channel
variable named DIALSTATUS after you've called the Dial() application.
The DIALSTATUS variable will tell you why the Dial() application was
unable to bridge the calls.


-- 
Jared Smith
Community Relations Manager
Digium, Inc.


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Re: [asterisk-users] asterisk always rining phone

2007-08-06 Thread satish patel

i am new for asterisk can u give me suggestion for my setup i m not expert for 
dialplan so can u send me example file which one u have i need help for 
extention.conf file options

Rgds

satish patel

Steve Totaro [EMAIL PROTECTED] wrote: If you setup your dialplan correctly 
and use priority jumping (off by 
default), you can play congestion or you could play the unavailable 
message for voicemail an then have the person who's phone it is record a 
message saying that they are on the phone, please leave a message.

Thanks,
Steve Totaro

satish patel wrote:
 is there any way to send busy tone to the calling party measn when i 
 call 2 some body and phone would be busy then i got busy tone so i can 
 guess party still talking 2 somebody...

 */Steve Totaro /* wrote:

 Sounds like you have call waiting on the phones. You can disable this
 on the Asterisk side. To verify, make a call on your phone and then
 dial yourself from another phone. Depending on the phone, you will
 have
 some sort of indication that a second call is coming in.

 Thanks,
 Steve Totaro

 satish patel wrote:
  Dear all
 
  I have setup of asterisk 1.2.14 with 100 SIP phone
  and it is working fine but thing is that when i call to somebody on
  local extention my asterisk not give me notification like party
 phone
  is busy or busy tone alway it give me rining single how can i
 justify
  other party is not pickup the phone or he/she talking with
 somebody on
  phone caz my phone rining on both stages is there any special
  configuration for it ??
 
 
 
 
 
 
 
  Got a little couch potato?
  Check out fun summer activities for kids.
 
 
 
 
 
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Re: [asterisk-users] Connecting two Asterisk servers with a frame relay connection

2007-08-06 Thread Eric \ManxPower\ Wieling

MOSBAH ABDELKADER wrote:
 Hello,
 
 Have i to buy an asterisk card like TDM400P to connect the two asterisk
 servers with frame relay.

I never do that.  I use a router that supports Frame Relay.  For me, 
installing a Digium card just to connect to a Frame Relay network is 
much more work, poorly documented, and just much more hassle than using 
a router with Frame Relay support.

You could not use any of the TDM cards, you would need a T-1/E-1 card 
from Digium or Sangoma if you wanted to go that route.  I would not 
recommend it.

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Re: [asterisk-users] Teliax Quality of Service

2007-08-06 Thread Eric \ManxPower\ Wieling

Douglas Garstang wrote:
 I don't think creating a network without a single point of failure is 
 unreasonable.

How often have you built a network without a single point of failure?

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Re: [asterisk-users] Teliax Quality of Service

2007-08-06 Thread Eric \ManxPower\ Wieling

Douglas Garstang wrote:
 Let's assume for a moment that it's impossible. That does not mean adding 
 additional servers and additional networking equipment does not add value, or 
 is a worthless endeavour.

I agree with that.  At least two people that I know run ITSPs.  Each 
time they have an outage (which is not very often) they DO learn from 
the experience and work to avoid a future outage cause by the same issue.

You would be surprised at how many little things can cause an outage.


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Re: [asterisk-users] Learn some terminalogy before mountingthistask.

2007-08-06 Thread James Collier

Flash Operator Panel would do it.

Also the Aastra 55i phones with the expansion module, which has 36 lines on
it should work, but you will need to cofigure your Asterisk for Shared Line
Appearances (also called Bridged Line Appearance) for the Busy Lamp Field
(BLF) to work.  The Aastra 55i would show you if they are talking or not.




-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] nombre de James R.
Stevens
Enviado el: lunes, 06 de agosto de 2007 5:39
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Asunto: Re: [asterisk-users] Learn some terminalogy before
mountingthistask.


All,

In the design of an Asterisk system using Cisco 7900 series SIP phones
we are struggling with giving the reception folks (3) hardware that can
tell them the status of everyone in the office (10 or so) (On the phone,
out of office etc) Something that would register each of the extensions
we choose and give status of that ext.

What hardware (Phone or other) could we give the receptionist to do
this?


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jerry
Jones
Sent: Monday, July 02, 2007 4:53 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Learn some terminalogy before mounting
thistask.


On Jul 2, 2007, at 4:31 PM, James R. Stevens wrote:

 All,

 It's been some time since this thread was alive but we are now seeing
 some progress in this project. Which I will document.
 We have ordered a T1 for the new building which we are moving (We are
 getting 14 channels of the T1.) and have a Sangoma A101 card for a 3U
 rack server.
 The T1 will have B8ZF decoding and ESF framing  which the sangoma card
 should handle.

 They asked me if we want NI1 or NI2 ?? Is this a reference to the  
 PRI ?
Yes. You want NI2.



 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Pierre
 Marceau
 Sent: Tuesday, April 10, 2007 11:25 PM
 To: asterisk-users@lists.digium.com
 Subject: Re: [asterisk-users] Learn some terminalogy before mounting
 this task.

 James,

 I'm sorry that I can't add anything but just wanted you to know that I
 am watching this thread with great interest and suspect that many  
 others
 will too.

 Thanks in advance for posting lots of details as you go thru the
 process.

 Pierre


 [EMAIL PROTECTED] 4/10/2007 10:41:36 PM 

 Hi James,

Admittedly, the terminological and conceptual barrier may present
 some
 impediments to the completeness and specificity of answers, so we  
 might
 have to work at this a bit, but let's see how we can help:

 On Tue, 10 Apr 2007, James R. Stevens said something to this effect:

 We have a T1 coming into the building(FYI-Our Voice and Data are on
 separate T's) terminating at the Smart Jack.

Are you implying that there are two T1 circuits -- one voice,  
 and one

 data?  Or do you mean that the T1 is channelised and some of the
 channels
 are used for voice and some for data?  That's kind of what it sounds
 like.
 Sounds like you can do 7 calls on voice channels and the rest are
 provisioned as a clear-channel data pipe.

That would mean that you have some equipment for breaking them  
 out on

 your premises.  The channel bank would break out the voice lines as  
 FXO
 analogue lines (if you set it to) and those probably feed into your  
 PBX.

 The rest of the channels used for data would probably be signaled  
 out on
 another T1 interface, but with some subrate DS0 channels missing.
 That's
 ust a guess.

But what you say below suggests that my theory is wrong, so perhaps
 it is
 the case that you have separate voice and data T1s after all, even
 though
 you refer to it in the singular.

Do be aware that under no circumstances does anyone generally refer
 to a
 T1 as a T.  :)

 I can tell you our current phone system can handle 7 phone calls at a
 time:

   Does this mean the T only has 7 channels provisioned out of the 24
 possible?

This is possible.  Do you happen to know what kind of signaling is
 used
 on it?  Is it an ISDN PRI, or an EM trunk?

  Does a channel (In terms of the T1) = a port?

A port on what?  The channel bank?

Channel banks generally do break the DS0s (subrate 64 kbps  
 channels,
 of
 which there are 24 on a T1) out, but some more sophisticated ones have
 the
 capability to do other things as well.

If so, the answer is yes.

  How many phone calls can one TDM400 support concurrently? (four ??)

If it has four FXO ports and four FXO modules, yes.  They come in
 different combinations.  Some come with 2 FXO (outside POTS lines  
 to CO)

 and 2 FXS (plain analogue POTS handsets) ports, etc.

  Would I be better off getting a Zapata T1 card and forgetting the
 Channel bank all together(Use the digital signal)?

You could do that.  Personally, the easiest approach I would say
 would be
 to order a PRI.  They've probably considerably gone down in prices,

[asterisk-users] SIP RegEvent - RFC3680

2007-08-06 Thread Olivier

Hi,

Does Asterisk support rfc3680 ?
This relates to registration event package.

This features seem to be convenient when implementing free sitting features

Regards
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[asterisk-users] iax2 registration being rejected

2007-08-06 Thread John covici

Hi.  I have a server which is trying to register with me using iax2
and asterisk 1.2.  My asterisk server is rejecting the registration
saying ip address of server is not dynamic.  What does this mean and
what do I need to change to accept the registration?

Thanks.

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

 John Covici
 [EMAIL PROTECTED]

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Re: [asterisk-users] Royalty for On Hold Music ?

2007-08-06 Thread Jay R. Ashworth

On Mon, Aug 06, 2007 at 12:18:36AM -0400, SIP wrote:
 Jay R. Ashworth wrote:
  ASCAP and BMI annual blankets aren't actually that expensive.  A live
  music venue run by some friends of mine had both, and for 535 fire-code
  seating and about 150 nights a year, I think they paid $500 a year to
  each of them.
 
  So, let's decide that songwriting is something worth paying people to
  do (that's who BMI and ASCAP royalties go to, people), and quit
  whining, ok?

 Oh I'm hardly whining. Songwriters usually JUST scrape by on their 
 music, and are often screwed over both by people who don't pay royalties 
 and by ASCAP, BMI, SESAC, NPMI, etc, who are perfectly happy to adjust 
 the royalties paid out in such ways as to maximise what they get to keep 
 and minimise what they have to share.

Ok, I retract 'whining'.  :-)  I'm not in a position to speak about
whether the PRI's are screwing their members or not, but I would assume
they're not screwing them as badly as record companies who own their
performers masters.

I'm merely explaining that, if you 
 WANT to tempt fate and not pay performance royalties for on-hold music 
 or music in the lobby of your office, then you have less of a chance of 
 having to worry about it if you're a tiny shop than if you're a big one. 

Sure.

 And also that there are ways around having to pay yearly fees by using 
 royalty-free music or writing your own.

Not always, which was the point of my original followup: not all buyout
music avoids BMI and ASCAP licensing; the buyout is on the *recording
and sync rights*, not always the performance licensing.

 However, if you get caught willfully performing copyrighted music 
 without paying ASCAP, BMI, et al, you're liable for a $100,000 fine 
 ($20,000 per song if it's not deemed willful) per song.

I wonder how much of *that* money goes to the songwriters.  ;-)

Cheers
-- jra
-- 
Jay R. Ashworth   Baylink  [EMAIL PROTECTED]
Designer The Things I Think   RFC 2100
Ashworth  Associates http://baylink.pitas.com '87 e24
St Petersburg FL USA  http://photo.imageinc.us +1 727 647 1274

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Re: [asterisk-users] help: H323 and SIP

2007-08-06 Thread Dino Anaclerio

Hi,

you must choose the h323 channel, install and configure it.
I've used ooh323 for a similar project and I have edited the
ooh323.conffile with the Gnugk's IP address (your h323 gatekeeper) and
a new context
for your test. I've also configured the file .ini in Gnugk (I've used the
Win version).  In the extensions.conf file I've created a dialplan for the
new context. I've registered the H.323 endpoint (in your case the Cisco GW)
with Gnugk.

This is what happens: H.323 Endpoint-GnuGK-Asterisk-SIP client. But the
ooh323 channel in Asterisk must be active and properly configured.

Regards

Dino
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Re: [asterisk-users] Royalty for On Hold Music ?

2007-08-06 Thread SIP

Jay R. Ashworth wrote:
 However, if you get caught willfully performing copyrighted music 
 without paying ASCAP, BMI, et al, you're liable for a $100,000 fine 
 ($20,000 per song if it's not deemed willful) per song.
 

 I wonder how much of *that* money goes to the songwriters.  ;-)

 Cheers
 -- jra
   
I actually tried to find that out (even something anecdotal), but so far 
no luck. I'm guessing not that much. The law allows for adjusting the 
percentages somewhat on the fly for various reasons (for instance, web 
radio performances give much less money to the songwriters than regular 
radio performances because they fall under a new category created 
specifically to handle web radio), and I imagine that any legal action 
that accrued hefty fines would likely be deemed to be mostly 
administrative and legal costs as opposed to damages to the songwriter.

Just a guess, though, borne from experience and my cynical nature. ;)


N.

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Re: [asterisk-users] iax2 registration being rejected

On Mon, 2007-08-06 at 11:00 -0400, John covici wrote:
 Hi.  I have a server which is trying to register with me using iax2
 and asterisk 1.2.  My asterisk server is rejecting the registration
 saying ip address of server is not dynamic.  What does this mean and
 what do I need to change to accept the registration?

You must change your peer section in iax.conf to say host=dynamic
instead of having an IP address or name for the host setting.  (If the
host is on a static IP, there's no reason for them to register to you,
right?)


-- 
Jared Smith
Community Relations Manager
Digium, Inc.


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Re: [asterisk-users] ATA phones ring when they register

On Mon, 2007-08-06 at 07:42 -0700, Vieri wrote:
 What I don't know yet is if it's a purely ATA
 config-related issue or if I also need to change
 Asterisk's settings.

As far as I know, this is a setting on the ATA, and nothing you change
in Asterisk would affect it.

-- 
Jared Smith
Community Relations Manager
Digium, Inc.


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[asterisk-users] Free sitting

2007-08-06 Thread Olivier

Hello,

How would you implement free sitting ?

The idea is to offer teachers the ability to share the same desk and
hardphone : for instance, Mr Foo is teaching mechanics on mondays while Mr
Bar is teaching english on wednesdays.
Each has his own extension but use the same hardphone.

1. Does a program check a calendar or database somewhere to allocate a phone
to a user (as teachers schedules are known in advance) ?
2. Every morning, users have to login (logoff is automatic during nighttime)
?
3. Users have to login/logoff themselves using a dedicated IVR ?
4. Users have to login/logoff themselves using a dedicated program on their
PC ?

Do you offer basic services (emergency and internals calls) between logins ?
Do you use any phone specific menu ?

Regards
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Re: [asterisk-users] Learn some terminalogy before mountingthistask.

2007-08-06 Thread James R. Stevens

Thank you for your reply as it is exactly what we would need. Sorry I
didn't find it myself. I do have a question about configuration within
Asterisk. 

 

I'm reading the PDF on the Cisco Expansion module and it says 'When used
as a DN key buttons are illuminated ...'

 

Is that what we are doing within Asterisk or Trixbox when we configure
an extension?  (A Directory Number??)

 

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of James
FitzGibbon
Sent: Monday, August 06, 2007 7:37 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Learn some terminalogy before
mountingthistask.

 

On 8/5/07, James R. Stevens [EMAIL PROTECTED] wrote:

In the design of an Asterisk system using Cisco 7900 series SIP
phones
we are struggling with giving the reception folks (3) hardware
that can
tell them the status of everyone in the office (10 or so) (On
the phone, 
out of office etc) Something that would register each of the
extensions
we choose and give status of that ext.

What hardware (Phone or other) could we give the receptionist to
do
this?


You're probably looking for something like this:

http://www.cisco.com/en/US/products/hw/phones/ps379/products_data_sheet0
9186a008008883d.html 

I have no experience integrating this specific piece of hardware with
Asterisk, but I've done what you're trying to do with the Grandstream
equivalent for our front reception:

http://www.grandstream.com/gxp2000.html

and

http://www.grandstream.com/gxp2000ext.html

As I understand it, so long as the device can do a SIP SUBSCRIBE for
each extension you want to monitor and you configure hints in your
Asterisk dialplan for those extensions, it should work.  You may need to
set 'subscribecontext' (in sip.conf) for the phone that will be watching
the extensions unless your hints are in the same context as the phone
uses for outbound dialing.

Of course, what the device does with the various payloads contained in
the SIP NOTIFY messages is going to be different for each phone.  On the
Grandstream I can see 'not in use' (solid green) and 'ringing' (flashing
red) distinctly, but 'unavailable' and 'in use' are both mapped to a
solid red, which makes it somewhat useless for transiently connected
user agents like softphones. 


Hopefully someone with experience will speak up and confirm that the
7900 series does interop properly with Asterisk for SUBSCRIBE and
NOTIFY.

If that doesn't work, you could always go with a software solution, like
the Flash Operator Panel.  voip-info has a list (look at the Operator
section on the page): http://www.voip-info.org/wiki/view/Asterisk+GUI

-- 
j. 


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Re: [asterisk-users] iax2 registration being rejected

2007-08-06 Thread John covici

OK, thanks  -- I guess I hadn't quite figured out the purpose of
registration.

on Monday 08/06/2007 Jared Smith([EMAIL PROTECTED]) wrote
  On Mon, 2007-08-06 at 11:00 -0400, John covici wrote:
   Hi.  I have a server which is trying to register with me using iax2
   and asterisk 1.2.  My asterisk server is rejecting the registration
   saying ip address of server is not dynamic.  What does this mean and
   what do I need to change to accept the registration?
  
  You must change your peer section in iax.conf to say host=dynamic
  instead of having an IP address or name for the host setting.  (If the
  host is on a static IP, there's no reason for them to register to you,
  right?)
  
  
  -- 
  Jared Smith
  Community Relations Manager
  Digium, Inc.
  
  
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-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

 John Covici
 [EMAIL PROTECTED]

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Re: [asterisk-users] Teliax Quality of Service

Anthony Francis wrote:
 Tim Panton wrote:
   
 On 5 Aug 2007, at 06:54, Douglas Garstang wrote:

   
 
 I don't think creating a network without a single point of failure  
 is unreasonable.
 
   
 It's impossible. I can't think of a single example where this  
 actually exists.

 Getting even close is hideously expensive.

 Tim, speaking for himself :-)

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 In fact, the only people who would say something like this are folks who 
 have never PHYSICALLY implemented a network, they simply don't 
 understand the limitations involved.

 Anthony

   

What if a train derails and slices through the main fiber connections.  
OK, so you have XO, Global Crossing, Verizon, and UCN all for 
redundancy.  Well guess what?  They are all most likely running over 
those strands of fiber.  You better have a VSAT connection too!

Thanks,
Steve


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Re: [asterisk-users] ATA phones ring when they register

2007-08-06 Thread Mr Shunz

On 8/6/07, Vieri [EMAIL PROTECTED] wrote:
 Hi,

 I have an 8-port Grandstream GXW-4008 V1.2A ATA
 converter with analog phones connected to it.

Hi,

we hava a GXW-4004 but i think it has the same sw ...

 They work fine except for just one feature I would
 like to modify. Somehow, each time the ATA
 re-registers the SIP clients or each time the device
 has to be rebooted for maintenance, the phones ring
 once.

we had the same problem and we came to this solution:

go under profile settings and set

Caller ID Scheme as

ETSI-FSK Prior to Ringing with DTAS...

best regards

-- 
Daniele Santi.o.
[EMAIL PROTECTED]  ..o
Linux User #415108   ooo

8N1:
8 bit di dati,
1 bit di stop,
Nessuna Pieta`

-
()  ascii ribbon campaign - against html mail
/\- against microsoft attachments
-

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Re: [asterisk-users] Free sitting

2007-08-06 Thread Julian J. M.

Freepbx has devices and users concept. It may be what you're looking for.
You can have your users log in in any phone with their extension
number and password. After that, all calls to his extension would ring
on that phone.

http://www.freepbx.org

Julian J. M.

On 8/6/07, Olivier [EMAIL PROTECTED] wrote:
 Hello,

 How would you implement free sitting ?

 The idea is to offer teachers the ability to share the same desk and
 hardphone : for instance, Mr Foo is teaching mechanics on mondays while Mr
 Bar is teaching english on wednesdays.
 Each has his own extension but use the same hardphone.

 1. Does a program check a calendar or database somewhere to allocate a phone
 to a user (as teachers schedules are known in advance) ?
 2. Every morning, users have to login (logoff is automatic during nighttime)
 ?
 3. Users have to login/logoff themselves using a dedicated IVR ?
 4. Users have to login/logoff themselves using a dedicated program on their
 PC ?

 Do you offer basic services (emergency and internals calls) between logins ?
 Do you use any phone specific menu ?

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Re: [asterisk-users] iax2 registration being rejected

On Mon, 2007-08-06 at 11:42 -0400, John covici wrote:
 OK, thanks  -- I guess I hadn't quite figured out the purpose of
 registration.

A device registers to Asterisk to tell Asterisk what it's current IP
address is, so that Asterisk knows where to send calls destined for that
device.  That's all there is to it.


-- 
Jared Smith
Community Relations Manager
Digium, Inc.


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Re: [asterisk-users] Teliax Quality of Service

2007-08-06 Thread SIP

Steve Totaro wrote:
 Anthony Francis wrote:
   
 Tim Panton wrote:
   
 
 On 5 Aug 2007, at 06:54, Douglas Garstang wrote:

   
 
   
 I don't think creating a network without a single point of failure  
 is unreasonable.
 
   
 
 It's impossible. I can't think of a single example where this  
 actually exists.

 Getting even close is hideously expensive.

 Tim, speaking for himself :-)

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 In fact, the only people who would say something like this are folks who 
 have never PHYSICALLY implemented a network, they simply don't 
 understand the limitations involved.

 Anthony

   
 

 What if a train derails and slices through the main fiber connections.  
 OK, so you have XO, Global Crossing, Verizon, and UCN all for 
 redundancy.  Well guess what?  They are all most likely running over 
 those strands of fiber.  You better have a VSAT connection too!

 Thanks,
 Steve


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And then your customers complain not about lack of availability, but 
about the hideous delay caused by the sat latency. ;)

All SIP deployments should come with emergency communications kits 
consisting of two cans and a spool of string.

N.

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Re: [asterisk-users] Learn some terminalogy before mountingthistask.

2007-08-06 Thread James FitzGibbon

On 8/6/07, James R. Stevens [EMAIL PROTECTED] wrote:

  I'm reading the PDF on the Cisco Expansion module and it says 'When used
 as a DN key buttons are illuminated …'



 Is that what we are doing within Asterisk or Trixbox when we configure an
 extension?  (A Directory Number??)


I suspect DN Key is just one way of describing a multi-function button
that can both display extension status and serve as a speed dial / transfer
destination.

On the Grandstream I have to configure the expansion car buttons as
Asterisk BLF buttons, even though BLF (busy lamp field) isn't an
Asterisk setting that I turn on.  To enable BLF functionality in Asterisk, I
have to set up hints in the dialplan and configure the user agent to
subscribe to status notitications for those extensions.

I'd search for asterisk user testimonials to be safe (assuming nobody steps
up and says I got that working).  Often times you'll find someone's blog
about how they got a feature working with a particular piece of hardware,
along with configuration samples.

-- 
j.
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[asterisk-users] TAE to RJ11 connector (hope not OT)

2007-08-06 Thread gincantalupo

Hi,
I'm trying to use a Detewe TA 33-clip but there is no rj11 connector on 
it...only a TAE connector.
I'd like to create an adapter so I need to know which TAE pins to 
connect to RJ 11 pins.
Is there anybody who knows where I can find a schema of that adapter?
Single connector pinout may help too.

TIA

Giorgio Incantalupo

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Re: [asterisk-users] Sangoma PRI

2007-08-06 Thread Kevin P. Fleming

Stephen Bosch wrote:

 The only way this will ever happen is if Digium completely redesigns the
 card, which is a long way of saying that you will buy a new card before
 you have that request filled.

That is incorrect. The TE4XXP cards with v2 or later firmware *can* be
upgraded in the field, but we have not released an upgrade for those
cards that warrants distributing it to end users (there is a v3 but it
is only necessary for the PCI Express variants). This may change soon,
though, as there is work to produce some improved firmware for all the
TE4XXP cards in process right now. Unfortunately cards with v1 firmware
will not be able to be upgraded in the field.

Steve Totaro: We regularly allow users to cross-ship (advance
replacement) cards for firmware upgrades; you should not be required to
have your system out of service for any length of time longer than what
it takes to swap cards.

 This is one of the great things about the Sangoma hardware -- it was
 designed to be fully field upgradeable (they use an FPGA
 architecture). The design approach is worth emulating.

Have you looked at the Sangoma cards and the Digium cards? Did you
notice that *both* of them are based on large Xilinx FPGA parts? They
both use an 'FPGA architecture', at least for the PCI interface and
TDM/data buffering (both cards use dedicated T1/E1/J1 framer chips,
because it would be silly to not do so G).

-- 
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - The Genuine Asterisk Experience (TM)

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Re: [asterisk-users] Sangoma PRI

Kevin P. Fleming wrote:
 Stephen Bosch wrote:

   
 The only way this will ever happen is if Digium completely redesigns the
 card, which is a long way of saying that you will buy a new card before
 you have that request filled.
 

 That is incorrect. The TE4XXP cards with v2 or later firmware *can* be
 upgraded in the field, but we have not released an upgrade for those
 cards that warrants distributing it to end users (there is a v3 but it
 is only necessary for the PCI Express variants). This may change soon,
 though, as there is work to produce some improved firmware for all the
 TE4XXP cards in process right now. Unfortunately cards with v1 firmware
 will not be able to be upgraded in the field.

 Steve Totaro: We regularly allow users to cross-ship (advance
 replacement) cards for firmware upgrades; you should not be required to
 have your system out of service for any length of time longer than what
 it takes to swap cards.

   

Who do I contact for this.  Is the firmware upgrade still free?  My last 
email to the lady responsible (forgot her name) never replied or her 
email went into /dev/spam/null.  Can you get the ball rolling or give me 
an email address please?

Thanks,
Steve

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Re: [asterisk-users] Learn some terminalogy before mountingthistask.

2007-08-06 Thread Ryan Amos

The 7914 only works under SCCP; the SIP firmware does not support it at
all (the expansion panel won't even power on fully.) The SCCP channel
driver under Asterisk doesn't really support the 7914 very well,
currently it will only show onhook/offhook state (though there has been
much discussion recently about changing this.) If you want to do this
with SIP then you're better off with something like the grandstream
mentioned, or just use the Flash Operator Panel (IMO it gives you more
flexibility at a much lower cost.)
 
I have personally found receptionist phone functionality handled much
better with FOP. I have a 7914 and its functionality (and usefulness) is
very limited under Asterisk.



From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of James R.
Stevens
Sent: Monday, August 06, 2007 10:41 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Learn some terminalogy before
mountingthistask.



Thank you for your reply as it is exactly what we would need. Sorry I
didn't find it myself. I do have a question about configuration within
Asterisk. 

 

I'm reading the PDF on the Cisco Expansion module and it says 'When used
as a DN key buttons are illuminated ...'

 

Is that what we are doing within Asterisk or Trixbox when we configure
an extension?  (A Directory Number??)

 

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of James
FitzGibbon
Sent: Monday, August 06, 2007 7:37 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Learn some terminalogy before
mountingthistask.

 

On 8/5/07, James R. Stevens [EMAIL PROTECTED] wrote:

In the design of an Asterisk system using Cisco 7900 series SIP
phones
we are struggling with giving the reception folks (3) hardware
that can
tell them the status of everyone in the office (10 or so) (On
the phone, 
out of office etc) Something that would register each of the
extensions
we choose and give status of that ext.

What hardware (Phone or other) could we give the receptionist to
do
this?


You're probably looking for something like this:

http://www.cisco.com/en/US/products/hw/phones/ps379/products_data_sheet0
9186a008008883d.html 

I have no experience integrating this specific piece of hardware with
Asterisk, but I've done what you're trying to do with the Grandstream
equivalent for our front reception:

http://www.grandstream.com/gxp2000.html

and

http://www.grandstream.com/gxp2000ext.html

As I understand it, so long as the device can do a SIP SUBSCRIBE for
each extension you want to monitor and you configure hints in your
Asterisk dialplan for those extensions, it should work.  You may need to
set 'subscribecontext' (in sip.conf) for the phone that will be watching
the extensions unless your hints are in the same context as the phone
uses for outbound dialing.

Of course, what the device does with the various payloads contained in
the SIP NOTIFY messages is going to be different for each phone.  On the
Grandstream I can see 'not in use' (solid green) and 'ringing' (flashing
red) distinctly, but 'unavailable' and 'in use' are both mapped to a
solid red, which makes it somewhat useless for transiently connected
user agents like softphones. 


Hopefully someone with experience will speak up and confirm that the
7900 series does interop properly with Asterisk for SUBSCRIBE and
NOTIFY.

If that doesn't work, you could always go with a software solution, like
the Flash Operator Panel.  voip-info has a list (look at the Operator
section on the page): http://www.voip-info.org/wiki/view/Asterisk+GUI

-- 
j. 


-- 
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Re: [asterisk-users] Sangoma PRI

2007-08-06 Thread Julian Lyndon-Smith

And what of all the folk that have a v1 card (I've got 2 quad-ports 
sitting here) ?

And can you cross-ship a v1 card for a v2 card replacement ?

Julian.

Steve Totaro wrote:
 Kevin P. Fleming wrote:
   
 Stephen Bosch wrote:

   
 
 The only way this will ever happen is if Digium completely redesigns the
 card, which is a long way of saying that you will buy a new card before
 you have that request filled.
 
   
 That is incorrect. The TE4XXP cards with v2 or later firmware *can* be
 upgraded in the field, but we have not released an upgrade for those
 cards that warrants distributing it to end users (there is a v3 but it
 is only necessary for the PCI Express variants). This may change soon,
 though, as there is work to produce some improved firmware for all the
 TE4XXP cards in process right now. Unfortunately cards with v1 firmware
 will not be able to be upgraded in the field.

 Steve Totaro: We regularly allow users to cross-ship (advance
 replacement) cards for firmware upgrades; you should not be required to
 have your system out of service for any length of time longer than what
 it takes to swap cards.

   
 

 Who do I contact for this.  Is the firmware upgrade still free?  My last 
 email to the lady responsible (forgot her name) never replied or her 
 email went into /dev/spam/null.  Can you get the ball rolling or give me 
 an email address please?

 Thanks,
 Steve

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Re: [asterisk-users] Teliax Quality of Service

2007-08-06 Thread Stephen Bosch

Eric ManxPower Wieling wrote:
 Douglas Garstang wrote:
 Let's assume for a moment that it's impossible. That does not mean adding 
 additional servers and additional networking equipment does not add value, 
 or is a worthless endeavour.
 
 I agree with that.  At least two people that I know run ITSPs.  Each 
 time they have an outage (which is not very often) they DO learn from 
 the experience and work to avoid a future outage cause by the same issue.
 
 You would be surprised at how many little things can cause an outage.

My own experience is that increasing failover redundancy, which adds
correspondingly increasing complexity, also increases the odds of an outage.

It is very rare that failover redundancy works as intended during an
actual failover, no matter how many times you simulate it.

I would rather have a simple network design where the cause of failure,
when it happens, is obvious and quickly corrected. For example, I would
rather have replacement parts on the shelf and be able to slap them in
quickly than be running hot standbys and paying for the electricity, and
then have the thing break anyway when there's a failure.

-Stephen-

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Re: [asterisk-users] Sangoma PRI

Might as well unless you have to pay shipping twice.

Wanna sell one of those quad port cards if it is just sitting there 
(after you get the firmware upgraded of course :-) )?

Thanks,
Steve

Julian Lyndon-Smith wrote:
 And what of all the folk that have a v1 card (I've got 2 quad-ports 
 sitting here) ?

 And can you cross-ship a v1 card for a v2 card replacement ?

 Julian.

 Steve Totaro wrote:
   
 Kevin P. Fleming wrote:
   
 
 Stephen Bosch wrote:

   
 
   
 The only way this will ever happen is if Digium completely redesigns the
 card, which is a long way of saying that you will buy a new card before
 you have that request filled.
 
   
 
 That is incorrect. The TE4XXP cards with v2 or later firmware *can* be
 upgraded in the field, but we have not released an upgrade for those
 cards that warrants distributing it to end users (there is a v3 but it
 is only necessary for the PCI Express variants). This may change soon,
 though, as there is work to produce some improved firmware for all the
 TE4XXP cards in process right now. Unfortunately cards with v1 firmware
 will not be able to be upgraded in the field.

 Steve Totaro: We regularly allow users to cross-ship (advance
 replacement) cards for firmware upgrades; you should not be required to
 have your system out of service for any length of time longer than what
 it takes to swap cards.

   
 
   
 Who do I contact for this.  Is the firmware upgrade still free?  My last 
 email to the lady responsible (forgot her name) never replied or her 
 email went into /dev/spam/null.  Can you get the ball rolling or give me 
 an email address please?

 Thanks,
 Steve

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Re: [asterisk-users] Sangoma PRI

2007-08-06 Thread Stephen Bosch

Kevin P. Fleming wrote:
 Have you looked at the Sangoma cards and the Digium cards? Did you
 notice that *both* of them are based on large Xilinx FPGA parts? They
 both use an 'FPGA architecture', at least for the PCI interface and
 TDM/data buffering (both cards use dedicated T1/E1/J1 framer chips,
 because it would be silly to not do so G).

No, I hadn't taken a close look at both cards; Thanks for correcting me.

What's noticeable about the Sangoma cards is that, when you look across
the product line, the cards have the same basic frame, and the modular
design is really elegant. I'm just admiring fine design.

-Stephen-

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Re: [asterisk-users] ATA phones ring when they register

2007-08-06 Thread Guillermo Salas M.

On Mon, 2007-08-06 at 17:46 +0200, Mr Shunz wrote:
 we had the same problem and we came to this solution:
 
 go under profile settings and set
 
 Caller ID Scheme as
 
 ETSI-FSK Prior to Ringing with DTAS...
 
 best regards 

I'm experiencing the same issue with linksys pap2. Any knows how to stop
the ringing when the ATA registers with my asterisk box.

Regards,

-- 
Guillermo Salas M.
Telconet S.A.
Calle 15 y Avenida 24 Esq
Edificio Barre #2 Primer Piso
Telefono : +593 5 262 8071
Celular  : +593 9 985 5138
e-mail   : [EMAIL PROTECTED]
www  : http://www.manta.telconet.net
   http://www.telcocarrier.net

Linux User: 255902

Beat me, whip me, make me use Windows!

Please avoid sending me Word or PowerPoint attachments.
See http://www.fsf.org/philosophy/no-word-attachments.html

Please avoid the Top Posting, see
http://es.wikipedia.org/wiki/Top-posting


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Re: [asterisk-users] Sangoma PRI

2007-08-06 Thread David Gomillion

(top-posting because Julian did, and I'm too lazy to fix it all)

Last I checked, the replacement with the new firmware is only for those who
bought the card in the last year (i.e. the card is still under warranty).
Those of us who were early adopters cannot enjoy the improvements of the
upgraded firmware without buying all new cards.

Hopefully, I'm wrong and someone will correct me.

On 8/6/07, Julian Lyndon-Smith [EMAIL PROTECTED] wrote:

 And what of all the folk that have a v1 card (I've got 2 quad-ports
 sitting here) ?

 And can you cross-ship a v1 card for a v2 card replacement ?

 Julian.

 Steve Totaro wrote:
  Kevin P. Fleming wrote:
 
  Stephen Bosch wrote:
 
 
 
  The only way this will ever happen is if Digium completely redesigns
 the
  card, which is a long way of saying that you will buy a new card
 before
  you have that request filled.
 
 
  That is incorrect. The TE4XXP cards with v2 or later firmware *can* be
  upgraded in the field, but we have not released an upgrade for those
  cards that warrants distributing it to end users (there is a v3 but it
  is only necessary for the PCI Express variants). This may change soon,
  though, as there is work to produce some improved firmware for all the
  TE4XXP cards in process right now. Unfortunately cards with v1 firmware
  will not be able to be upgraded in the field.
 
  Steve Totaro: We regularly allow users to cross-ship (advance
  replacement) cards for firmware upgrades; you should not be required to
  have your system out of service for any length of time longer than what
  it takes to swap cards.
 
 
 
 
  Who do I contact for this.  Is the firmware upgrade still free?  My last
  email to the lady responsible (forgot her name) never replied or her
  email went into /dev/spam/null.  Can you get the ball rolling or give me
  an email address please?
 
  Thanks,
  Steve
 
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Re: [asterisk-users] Digium|Asterisk World

Too bad it is August 6th

*P.S. Remember, as a member of the Digium Family we have secured a 
special discount of 50% off of the conference fee for you if you 
register by July 29, 2007. To take advantage of this limited time offer, 
please register here 
https://secure.pulver.com/cgi-bin/von?mode=gpurconf=dawfal07type=gpricode=billm!*



Digium, Inc wrote:

 If you are having trouble reading this email, read the online version 
 http://now.eloqua.com/es.asp?s=491e=C07203767EE74FA9B9F905466D01E083elq=6B2052B3407646EC89D85380A1014DE5.




 https://secure.pulver.com/digiumAsteriskWorld/2007/boston/web/attendRegister.htm
  


 Dear Steve,

 I am pleased to announce that the conference program 
 http://www.digiumasteriskworld.com/2007/boston/web/confSchedule.htm 
 for *Digium|Asterisk World 
 http://www.digiumasteriskworld.com/2007/boston/web/* is now posted 
 to the website!

 Click here 
 https://secure.pulver.com/cgi-bin/von?mode=gpurconf=dawfal07type=gpricode=billm
  
 to register today!

 Digium|Asterisk World is the premier open communication event for the 
 channel that encompasses the world of open source platforms and 
 applications in the realm of IP communications. Whether you are a 
 service provider, VAR, systems integrator or someone who is rolling 
 out IP communications internally, Digium Asterisk World is the place 
 to be.

 *Digium|Asterisk World will be held October 30 - 31, 2007 at the 
 Boston Conference and Convention Center in Boston, MA*. Please visit 
 http://www.digiumasteriskworld.com 
 http://www.digiumasteriskworld.com/ for complete details.

 As a member of the Digium Family, we have secured a special discount 
 of *50%* *off of the conference fee for you if you register by July 
 29, 2007*. To take advantage of this limited time offer, please 
 register here 
 https://secure.pulver.com/cgi-bin/von?mode=gpurconf=dawfal07type=gpricode=billm!
   
 Also, please feel free to pass this on to your colleagues who may be 
 interested as well.

 Sincerely,
 Bill Miller
 VP of Product Management  Marketing
 Digium

 P.S. Remember, as a member of the Digium Family we have secured a 
 special discount of 50% off of the conference fee for you if you 
 register by July 29, 2007. To take advantage of this limited time 
 offer, please register here 
 https://secure.pulver.com/cgi-bin/von?mode=gpurconf=dawfal07type=gpricode=billm!

   


 Save 50% off
 of the conference!

 Register by July 29, 2007

 https://secure.pulver.com/cgi-bin/von?mode=gpurconf=dawfal07type=gpricode=billm
  


 https://secure.pulver.com/cgi-bin/von?mode=gpurconf=dawfal07type=gpricode=billm
  

  


 Copyright © Digium, Inc. - The Asterisk Company, 150 West Park Loop, 
 Ste 100, Huntsville, AL 35806
 Visit our website: Digium.com http://www.digium.com | Unsubscribe 
 http://www.digium.com/en/mediacenter/subscriptions.php | Update 
 Subscriptions http://www.digium.com/en/mediacenter/subscriptions.php

 --=_NextPart_000_0E0D_01C7D827.25487C2 


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Re: [asterisk-users] Royalty for On Hold Music ?

2007-08-06 Thread Jay R. Ashworth

On Mon, Aug 06, 2007 at 11:26:25AM -0400, SIP wrote:
 I actually tried to find that out (even something anecdotal), but so far 
 no luck. I'm guessing not that much. The law allows for adjusting the 
 percentages somewhat on the fly for various reasons (for instance, web 
 radio performances give much less money to the songwriters than regular 
 radio performances because they fall under a new category created 
 specifically to handle web radio), and I imagine that any legal action 
 that accrued hefty fines would likely be deemed to be mostly 
 administrative and legal costs as opposed to damages to the songwriter.

Well, that there is actually a valid reason not to pass such fines
along to their members: if they're escrowing them to bankroll legal
action on behalf of such members.

Cheers,
-- jra
-- 
Jay R. Ashworth   Baylink  [EMAIL PROTECTED]
Designer The Things I Think   RFC 2100
Ashworth  Associates http://baylink.pitas.com '87 e24
St Petersburg FL USA  http://photo.imageinc.us +1 727 647 1274

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Re: [asterisk-users] iax2 registration being rejected

2007-08-06 Thread Tim Panton


On 6 Aug 2007, at 16:53, Jared Smith wrote:

 On Mon, 2007-08-06 at 11:42 -0400, John covici wrote:
 OK, thanks  -- I guess I hadn't quite figured out the purpose of
 registration.

 A device registers to Asterisk to tell Asterisk what it's current IP
 address is, so that Asterisk knows where to send calls destined for  
 that
 device.  That's all there is to it.

For IAX it also has the side benefit of setting up  path through nat  
and port
mapping  routers.


Tim Panton

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




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Re: [asterisk-users] iax2 registration being rejected

Tim Panton wrote:
 On 6 Aug 2007, at 16:53, Jared Smith wrote:

   
 On Mon, 2007-08-06 at 11:42 -0400, John covici wrote:
 
 OK, thanks  -- I guess I hadn't quite figured out the purpose of
 registration.
   
 A device registers to Asterisk to tell Asterisk what it's current IP
 address is, so that Asterisk knows where to send calls destined for  
 that
 device.  That's all there is to it.
 

 For IAX it also has the side benefit of setting up  path through nat  
 and port
 mapping  routers.


 Tim Panton

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

   
Yes, since IAX2 only uses one port, this is correct.  Another thing to 
keep in mind is to set a low qualify value in Asterisk since some 
routers will tear down the connection pretty quickly.  The qualify acts 
as a keep-alive and prevents the router from closing the port and losing 
the map.

Thanks,
Steve

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Re: [asterisk-users] Learn some terminology before mountingthistask.

2007-08-06 Thread James R. Stevens

All,(Ryan-Your response saved me lots of RD time- Thank you very much)

 

I have been on TixBox site all morning reading through the MANY posts as
recent as 6-4-2007 on SLA and the need or reason it is not needed.

 

1)  In our office we do not have a single receptionist, rather a
ring group (Consists of 4 customer support people) that answer inbound
calls and directs them to the right person/dept.

2)  As is stands they each have other responsibilities and the
phone(s) are a split responsibility (Yes I am trying to show them a
better way)

Because of this, FOP or HUD would get in the way of the other
applications they are working with on the desktop(Although it is the
PERFECT solution for a dedicated reception person IMHO)

3)  The ONLY necessity here is that they be able to look down at the
hardware at a glance to see if someone is on the phone

4)  I was sticking with Cisco as I am very versed in
deployment/support and didn't want to support many different phones
however,

If the grandstream is what it takesThat's what we will do.

5)  How do the grandstreams get their configuration info? TFTP
similar to the Cisco model?

 

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ryan Amos
Sent: Monday, August 06, 2007 11:25 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Learn some terminalogy before
mountingthistask.

 

The 7914 only works under SCCP; the SIP firmware does not support it at
all (the expansion panel won't even power on fully.) The SCCP channel
driver under Asterisk doesn't really support the 7914 very well,
currently it will only show onhook/offhook state (though there has been
much discussion recently about changing this.) If you want to do this
with SIP then you're better off with something like the grandstream
mentioned, or just use the Flash Operator Panel (IMO it gives you more
flexibility at a much lower cost.)

 

I have personally found receptionist phone functionality handled much
better with FOP. I have a 7914 and its functionality (and usefulness) is
very limited under Asterisk.

 



From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of James R.
Stevens
Sent: Monday, August 06, 2007 10:41 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Learn some terminalogy before
mountingthistask.

Thank you for your reply as it is exactly what we would need. Sorry I
didn't find it myself. I do have a question about configuration within
Asterisk. 

 

I'm reading the PDF on the Cisco Expansion module and it says 'When used
as a DN key buttons are illuminated ...'

 

Is that what we are doing within Asterisk or Trixbox when we configure
an extension?  (A Directory Number??)

 

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of James
FitzGibbon
Sent: Monday, August 06, 2007 7:37 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Learn some terminalogy before
mountingthistask.

 

On 8/5/07, James R. Stevens [EMAIL PROTECTED] wrote:

In the design of an Asterisk system using Cisco 7900 series SIP
phones
we are struggling with giving the reception folks (3) hardware
that can
tell them the status of everyone in the office (10 or so) (On
the phone, 
out of office etc) Something that would register each of the
extensions
we choose and give status of that ext.

What hardware (Phone or other) could we give the receptionist to
do
this?


You're probably looking for something like this:

http://www.cisco.com/en/US/products/hw/phones/ps379/products_data_sheet0
9186a008008883d.html 

I have no experience integrating this specific piece of hardware with
Asterisk, but I've done what you're trying to do with the Grandstream
equivalent for our front reception:

http://www.grandstream.com/gxp2000.html

and

http://www.grandstream.com/gxp2000ext.html

As I understand it, so long as the device can do a SIP SUBSCRIBE for
each extension you want to monitor and you configure hints in your
Asterisk dialplan for those extensions, it should work.  You may need to
set 'subscribecontext' (in sip.conf) for the phone that will be watching
the extensions unless your hints are in the same context as the phone
uses for outbound dialing.

Of course, what the device does with the various payloads contained in
the SIP NOTIFY messages is going to be different for each phone.  On the
Grandstream I can see 'not in use' (solid green) and 'ringing' (flashing
red) distinctly, but 'unavailable' and 'in use' are both mapped to a
solid red, which makes it somewhat useless for transiently connected
user agents like softphones. 


Hopefully someone with experience will speak up and confirm that the
7900 series does interop properly with Asterisk for SUBSCRIBE and
NOTIFY.

If that doesn't work, you could always go with a software solution, like
the Flash

Re: [asterisk-users] iax2 registration being rejected

2007-08-06 Thread Jaswinder Singh



 Yes, since IAX2 only uses one port, this is correct.  Another thing to
 keep in mind is to set a low qualify value in Asterisk since some
 routers will tear down the connection pretty quickly.  The qualify acts
 as a keep-alive and prevents the router from closing the port and losing
 the map.

 Thanks,
 Steve



But if you set timeout lower than actual latency to peer .. it will result
in asterisk not sending any calls to peer at all so keeping it too low will
create  more problem  .. however peer will be able to make outgoing calls .
I think asterisk doesnt rely on qualify= parameter to keep connection open .
Main purpose of qualify option is to make sure peer is not lagged then
specified timeout period else call quality will be pathetic .. qualify=200
seems ok  . Btw i have never seen a device losing registration when qualify
value is set huge ( i keep  qualify = 2000 for a very dirty connection
sometimes :D  so that asterisk will show latency when i do sip show peers
and iax2 show peers in cli  )


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[asterisk-users] low-level dump for PRI dchan debugging

I've been going back and forth with my telco for several days, trying
different configurations to get a new PRI to come up.  The bchannels
are all up and the T1 is not in alarm status.  The dchannel refuses to
come up however.  We've tried ni2, qsig, and now dms100 for the
switchtype.  The telco tech I've been working with says that he's been
sending reset all channels signals to my system, to which he's
getting an establish remote response from my asterisk box.  I've
been running a packet dump (wanpipemon -i w1g1 -c trd) of my d-channel
this whole time and have yet to see a single incoming packet.  I
believe I *should* be seeing an incoming packet when he sends the
reset, correct?  Is there any way to do a completely raw dump of the
d-channel?

Here are my specs:
linux-2.6.16
libpri-1.3.5
zaptel-1.2.19
asterisk-1.2.21.1

The PRI interface is a Sangoma A102...it's running the latest firmware
and I'm running wanpipe-2.3.4-12 for the sangoma drivers.

Any ideas?

-- 
Erik Anderson
http://andersonfam.org

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Re: [asterisk-users] Free sitting

2007-08-06 Thread Olivier

Thanks.

In fact, my questions are more about usage than about technical background.
For instance, I doubt a user will log his system off when leaving : some
don't even turn their PC off.


Does anyone has an experience to share about that ?
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Re: [asterisk-users] low-level dump for PRI dchan debugging

Call Sangoma and give them root if you can.  They will fix it quickly or 
at least give you ammunition that it is the telco's issue.

Thanks,
Steve

Erik Anderson wrote:
 I've been going back and forth with my telco for several days, trying
 different configurations to get a new PRI to come up.  The bchannels
 are all up and the T1 is not in alarm status.  The dchannel refuses to
 come up however.  We've tried ni2, qsig, and now dms100 for the
 switchtype.  The telco tech I've been working with says that he's been
 sending reset all channels signals to my system, to which he's
 getting an establish remote response from my asterisk box.  I've
 been running a packet dump (wanpipemon -i w1g1 -c trd) of my d-channel
 this whole time and have yet to see a single incoming packet.  I
 believe I *should* be seeing an incoming packet when he sends the
 reset, correct?  Is there any way to do a completely raw dump of the
 d-channel?

 Here are my specs:
 linux-2.6.16
 libpri-1.3.5
 zaptel-1.2.19
 asterisk-1.2.21.1

 The PRI interface is a Sangoma A102...it's running the latest firmware
 and I'm running wanpipe-2.3.4-12 for the sangoma drivers.

 Any ideas?

   


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Re: [asterisk-users] polycom custom ring tones (slightly OT)

2007-08-06 Thread Doug

At 08:16 8/6/2007, Rob Schall wrote:
With Polycom 501s, creating custom ringtones isn't hard at all.

First, grab your favorite mp3 or wav file and create a file that is
about 10 seconds long (max). If its an mp3, convert it to a wav file.
Next, use this command to ensure the wav file is properly formatted
for a Polycom phone:
sox mywave.wav -r 8000 -U -c1 mywave.wav resample -ql

Now, if you type file mywave.wav, it should report:
mywave.wav: RIFF (little-endian) data, WAVE audio, ITU G.711 mu-law,
mono 8000 Hz

Copy this file into your phone's ftp directory (same folder as your
sip.cfg, etc). Make sure it has similar permissions to the default
wav files in that folder.

Finally, you can add this new file either by editing the config
files, or by accessing the phone's config website and adding the
filename to the wav file. Then reboot the phone, and access the
phones settings and the new ringtone will appear if everything
worked correctly.

Rob

Thanks

Stephen Bosch wrote:

Doug wrote:

At 21:59 7/29/2007, Paul Hales wrote:

I even got a Polycom here saying I'll be back which was funny for
about an hour, then not funny at all.

PaulH

Kewwl! How do you get the .wav files into the Polycom?

If it's not obvious, I'd be interested in this information too.

Most people seem to think you can't change the ringtones on the Polycom
sets.

-Stephen-

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[asterisk-users] Friday Aug 10th Asterisk Users Conference at 12:30 PM EDT

2007-08-06 Thread randulo

This Friday, part II of TDM solutions including ATA that do IAX and
SIP without opening the box and installing a card. Your experience in
this area would be appreciated.

You can find us here:

 http://www.AsteriskUsersConference.org

Also, a Google group has been created for discussions and scheduling
of the conferences. If you feel like this is of interest, please join
us:

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I hope we can make this a good way for you to know if  topic of
interest to you comes up. In the future, we'd like to get people using
ENUM and DUNDI to contribute their experience.

Please consider joining us.

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Re: [asterisk-users] low-level dump for PRI dchan debugging

2007-08-06 Thread Darryl Dunkin

wanpipemon is the way to do it as far as I know.

For starters, what do your zaptel/zapata configs look like?

I would first verify that your D-channel is set properly, you can view
that in the console as follows:
asterisk pri show span 1/0
Primary D-channel: 24
Status: Provisioned, Up, Active
Switchtype: National ISDN

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Erik
Anderson
Sent: Monday, August 06, 2007 11:09
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] low-level dump for PRI dchan debugging

I've been going back and forth with my telco for several days, trying
different configurations to get a new PRI to come up.  The bchannels
are all up and the T1 is not in alarm status.  The dchannel refuses to
come up however.  We've tried ni2, qsig, and now dms100 for the
switchtype.  The telco tech I've been working with says that he's been
sending reset all channels signals to my system, to which he's
getting an establish remote response from my asterisk box.  I've
been running a packet dump (wanpipemon -i w1g1 -c trd) of my d-channel
this whole time and have yet to see a single incoming packet.  I
believe I *should* be seeing an incoming packet when he sends the
reset, correct?  Is there any way to do a completely raw dump of the
d-channel?

Here are my specs:
linux-2.6.16
libpri-1.3.5
zaptel-1.2.19
asterisk-1.2.21.1

The PRI interface is a Sangoma A102...it's running the latest firmware
and I'm running wanpipe-2.3.4-12 for the sangoma drivers.

Any ideas?

-- 
Erik Anderson
http://andersonfam.org

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Re: [asterisk-users] low-level dump for PRI dchan debugging

On 8/6/07, Steve Totaro [EMAIL PROTECTED] wrote:
 Call Sangoma and give them root if you can.  They will fix it quickly or
 at least give you ammunition that it is the telco's issue.

Good idea - I just emailed them. Hopefully they'll respond quickly. My
normal contact there (Jignesh) is either out of the office today or at
least he forgot to start up MSN this morning, as he's showing offline.
 Hopefully he's not the only tech support guy there.

-erik

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Re: [asterisk-users] low-level dump for PRI dchan debugging

Erik Anderson wrote:
 On 8/6/07, Steve Totaro [EMAIL PROTECTED] wrote:
   
 Call Sangoma and give them root if you can.  They will fix it quickly or
 at least give you ammunition that it is the telco's issue.
 

 Good idea - I just emailed them. Hopefully they'll respond quickly. My
 normal contact there (Jignesh) is either out of the office today or at
 least he forgot to start up MSN this morning, as he's showing offline.
  Hopefully he's not the only tech support guy there.

 -erik

   

I have done a conference call with the telco guy, myself, and a Sangoma 
tech at the same time.  I was just quite and let them battle it out.  It 
turned out to be a telco issue but the Global Crossing tech wanted to 
blame me and my equipment.  He ate a little humble pie on that one.

If I were you, I would call Sangoma, sometimes the French Canadian 
accent is tough but if you give them root, it shouldn't be that bad.  
They have several techs and any one of them should be able to help.

Thanks,
Steve


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Re: [asterisk-users] low-level dump for PRI dchan debugging

On 8/6/07, Darryl Dunkin [EMAIL PROTECTED] wrote:
 wanpipemon is the way to do it as far as I know.

 For starters, what do your zaptel/zapata configs look like?

lpdlnx04*CLI pri show span 1
Primary D-channel: 24
Status: Provisioned, Down, Active
Switchtype: Nortel DMS100
Type: Network

I know it's odd, but the telco instructed me to set my equipment as
the network end...hence pri_net:

/etc/zaptel.conf
loadzone=us
defaultzone=us

#Sangoma A102 port 1 [slot:10 bus:2 span: 1]
span=1,1,0,esf,b8zs
bchan=1-8
dchan=24


/etc/asterisk/zapata.conf
[trunkgroups]

[channels]
context=default
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
relaxdtmf=yes
rxgain=0.0
txgain=0.0
group=1
callgroup=1
pickupgroup=1

immediate=no

;;Sangoma A102 port 1 [slot:10 bus:2 span: 1]
switchtype=dms100
context=from-pstn
group=1
signalling=pri_net
channel = 1-8

There you go.

As an aside, turns out that it's a national holiday in CA, so the
Sangoma support guys are on vacation for the day.

-erik

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Re: [asterisk-users] low-level dump for PRI dchan debugging

2007-08-06 Thread Darren Nickerson

Erik Anderson [EMAIL PROTECTED] wrote:

 Good idea - I just emailed them. Hopefully they'll respond quickly. My
 normal contact there (Jignesh) is either out of the office today or at
 least he forgot to start up MSN this morning, as he's showing offline.
 Hopefully he's not the only tech support guy there.

Jignesh is by no means the only tech there, but I doubt any of them are 
doing much today - it's a holiday in Canada.

-Darren

-- 
Darren Nickerson
Telephony Depot
www.telephonydepot.com
+1.215.825.8710 ext 8106 (office)
+1.215.243.8335 (fax) 


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Re: [asterisk-users] IAX bat phone.

2007-08-06 Thread Facundo Ameal

Grandstream HT386 also has that feature. Into the configuration you
can find a field called 'Audial Off-hook', there you can set any
extension so the ATA will dial as soon as you pick up the handset.

On 8/6/07, James FitzGibbon [EMAIL PROTECTED] wrote:
 On 8/3/07, Michael Munger [EMAIL PROTECTED] wrote:
 
 
 
 
  Is there a way to setup an IAX bat phone (immediate=yes) or is this a
 privilege only reserved for ZAP channels?

 As I understand it, this would have to be supported by your specific
 hard/soft phone.

 It's the same with SIP - taking a handset off-hook doesn't cause any traffic
 to go to Asterisk.  The first packet from the user agent is sent when the
 phone tries to dial something.  Depending on the user agent, this could be
 as soon as someone presses a single key (so-called early dial with SIP 484
 responses), or more typically when an entire number has been dialed and a
 timeout has occurred or send button has been pressed.  Zap FXS ports can
 tell when a handset has gone off-hook and take some action based on that due
 to the change in electrical impedance.

 Some soft-phones support bat-phone operation, though you have to hunt
 through the docs to get it to work.  My Linksys SPA942 desk phone has a dial
 plan syntax that allows this:

 (:S0)

 Which means prefix whatever I type with  and match an empty string,
 dialing as soon as you have a match, which causes the phone to calll 
 as soon as I take it off hook.  But it's obviously device-specific, and has
 nothing to do with SIP or IAX or Asterisk for that matter.  When the call
 arrives at my server, it doesn't look any different than a call to  from
 a phone with a more traditional dialplan.

 --
 j.
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-- 
Facundo Ameal.
famealatgmaildotcom
Linux User #395088

Share your knowledge, use free software.

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Re: [asterisk-users] low-level dump for PRI dchan debugging

On 8/6/07, Steve Totaro [EMAIL PROTECTED] wrote:

 I have done a conference call with the telco guy, myself, and a Sangoma
 tech at the same time.  I was just quite and let them battle it out.  It
 turned out to be a telco issue but the Global Crossing tech wanted to
 blame me and my equipment.  He ate a little humble pie on that one.

 If I were you, I would call Sangoma, sometimes the French Canadian
 accent is tough but if you give them root, it shouldn't be that bad.
 They have several techs and any one of them should be able to help.

This sounds like a great idea - I'm going to try and get Sangoma and
the telco tech on the horn at the same time tomorrow.

-Erik

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Re: [asterisk-users] low-level dump for PRI dchan debugging

Darren Nickerson wrote:
 Erik Anderson [EMAIL PROTECTED] wrote:
   
 Good idea - I just emailed them. Hopefully they'll respond quickly. My
 normal contact there (Jignesh) is either out of the office today or at
 least he forgot to start up MSN this morning, as he's showing offline.
 Hopefully he's not the only tech support guy there.
 

 Jignesh is by no means the only tech there, but I doubt any of them are 
 doing much today - it's a holiday in Canada.

 -Darren

   
They should have an on-call tech for emergencies even if it is a holiday.

Thanks,
Steve

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Re: [asterisk-users] low-level dump for PRI dchan debugging

Erik Anderson wrote:
 On 8/6/07, Steve Totaro [EMAIL PROTECTED] wrote:
   
 Call Sangoma and give them root if you can.  They will fix it quickly or
 at least give you ammunition that it is the telco's issue.
 

 Good idea - I just emailed them. Hopefully they'll respond quickly. My
 normal contact there (Jignesh) is either out of the office today or at
 least he forgot to start up MSN this morning, as he's showing offline.
  Hopefully he's not the only tech support guy there.

 -erik

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also in asterisk do:
pri intense debug span 1
Then you should see UA's and SABME's, If you don't, your not talking to 
them.

Anthony

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Re: [asterisk-users] Free sitting

Olivier wrote:
 Thanks.

 In fact, my questions are more about usage than about technical 
 background.
 For instance, I doubt a user will log his system off when leaving : 
 some don't even turn their PC off.


 Does anyone has an experience to share about that ?

 

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You could use Gotoiftime() to do what you want, look it up on voip-info.org

Anthony

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Re: [asterisk-users] Sangoma PRI

2007-08-06 Thread Noah Miller

Hi David -

 Last I checked, the replacement with the new firmware is only for those who
 bought the card in the last year (i.e. the card is still under warranty).
 Those of us who were early adopters cannot enjoy the improvements of the
 upgraded firmware without buying all new cards.

 Hopefully, I'm wrong and someone will correct me.

I don't know if this is still the case, but I had a v1 TE410P card
that Digium replaced with a v2 at no cost.  Because I chose to do the
cross-ship option (so I could do an immediate swap), I had to give
them a credit card number, but I don't think they ever actually
charged the card.  Free upgrade to a field-upgradable TE410P v2.  Yay!

I'm glad that everyone is having great experiences with Sangoma
support.  I feel I should add that the few times I've dealt with
Digium support, my experiences were equally good.


- Noah

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Re: [asterisk-users] Telco is not detecting HangUp w/ TDM400P

2007-08-06 Thread Alex Pankratov

Thanks, Julian. 

I saw this explanation, and it does not apply. There is no 
hangup supervision with my carrier. I think they used to 
have it, when I had different number, but even then it was 
not 30 seconds, but more like 3 to 5.

I am now inclined to think that it has something to do with
a cell phone provider. I am going to run more tests today 
using regular land line and report what's up.

Alex 

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Julian J. M.
 Sent: Monday, August 06, 2007 2:07 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Telco is not detecting HangUp w/ TDM400P
 
 That's ok, and is expected behaviour. The telco will keep the line
 open for about 30 seconds. It's useful when there is no PBX, and just
 2 or 3 phones attached to the same line... you can hangup on one room,
 go to another, pickup and continue the conversation.
 
 Anyway, i guess the telco can reduce that timeout or remove it
 completely. Just tell them you have a PBX on that line.
 
 Julian J. M.
 
 On 8/6/07, Alex Pankratov [EMAIL PROTECTED] wrote:
  Hi guys,
 
  I spent a couple of hours in Google, but the problem
  appears to be uncommon, so I'd like to ask about it here.
 
  The problem is exactly the opposite to Asterisk does
  not detect FXO hangup. In my case it's the Telco who
  does not appear to be detecting Asterisk's hangups.
 
  Telco is Telus in Vancouver, Canada. The setup is very
  simple -
 
   Telco - FXO/TDM400p - * - softphone
 
  The log is -
 
  -- Starting simple switch on 'Zap/4-1'
  -- Executing [EMAIL PROTECTED]:1] Answer(Zap/4-1, ) in new stack
  -- Executing [EMAIL PROTECTED]:2] Dial(Zap/4-1, 
 IAX2/alex|5|r) in new
  stack
  -- Called alex
  -- Call accepted by 192.168.1.102 (format gsm)
  -- Format for call is gsm
  -- IAX2/alex-2 is ringing
  -- Nobody picked up in 5000 ms
  -- Hungup 'IAX2/alex-2'
  -- Executing [EMAIL PROTECTED]:3] Hangup(Zap/4-1, ) in new stack
  -- Hungup 'Zap/4-1'
 
  At this point the caller (say, me on my cell phone) still
  sits connected and enjoying the white noise. The longest I
  waited was about 20 seconds and then I hung up.
 
  Similar problem is described here (November 2006) -
 
  
 http://lists.digium.com/pipermail/asterisk-dev/2006-November/0
24768.html
 
  but there's no solution and the discussion is not very
  helpful.
 
  Any pointers and/or ideas are greatly appreciated.
 
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Re: [asterisk-users] low-level dump for PRI dchan debugging

On 8/6/07, Anthony Francis [EMAIL PROTECTED] wrote:
 also in asterisk do:
 pri intense debug span 1
 Then you should see UA's and SABME's, If you don't, your not talking to
 them.

I see plenty of SABMEs, but nothing else:

 [ 02 01 7f ]

 Unnumbered frame:
 SAPI: 00  C/R: 1 EA: 0
  TEI: 000EA: 1
   M3: 3   P/F: 1 M2: 3 11: 3  [ SABME (set asynchronous balanced mode 
 extended) ]
 0 bytes of data
Sending Set Asynchronous Balanced Mode Extended

 [ 02 01 7f ]

 Unnumbered frame:
 SAPI: 00  C/R: 1 EA: 0
  TEI: 000EA: 1
   M3: 3   P/F: 1 M2: 3 11: 3  [ SABME (set asynchronous balanced mode 
 extended) ]
 0 bytes of data
Sending Set Asynchronous Balanced Mode Extended
lpdlnx04*CLI pri
 [ 02 01 7f ]
lpdlnx04*CLI pri
 Unnumbered frame:
 SAPI: 00  C/R: 1 EA: 0
  TEI: 000EA: 1
   M3: 3   P/F: 1 M2: 3 11: 3  [ SABME (set asynchronous balanced mode 
 extended) ]
 0 bytes of data
Sending Set Asynchronous Balanced Mode Extended

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[asterisk-users] CDR/MySQL basic config

2007-08-06 Thread Adrian Marsh

Hi,

I'm trying to add mysql CDR onto a vanilla Asterisk 1.2 install.  The
add-ons pack has been installed for a while, so now I'm trying to add
the Mysql config.

I've created a mysql database, added the grants for a user acces, and
can run a mysql -u asteriskcdruser -p and can connect to the database.

I've been using this as a guide:
http://www.757.org/~joat/wiki/index.php/Asterisk#Viewing_CDR_Data_with_A
sterisk:_CDR_Analyzer

I've created cdr_mysql.conf:

[global]
hostname=localhost
dbname=asteriskcdrdb
table=cdr
password=password
user=asteriskcdruser
port=3306
sock=/tmp/mysql.sock
userfield=1

But when I start asterisk (1.4 on my test machine), I get:

  == Parsing '/etc/asterisk/cdr_mysql.conf': Found
[Aug  6 21:01:14] ERROR[32512]: cdr_addon_mysql.c:436 my_load_module:
Failed to connect to mysql database asteriskcdrdb on localhost.
cdr_addon_mysql.so = (MySQL CDR Backend)
[Aug  6 21:01:14] ERROR[32512]: res_config_mysql.c:627 mysql_reconnect:
MySQL RealTime: Failed to connect database server  on  (err 2002). Check
debug for more info.
[Aug  6 21:01:14] WARNING[32512]: res_config_mysql.c:474 load_module:
MySQL RealTime: Couldn't establish connection. Check debug.
[Aug  6 21:01:14] NOTICE[32512]: config.c:1171
ast_config_engine_register: Registered Config Engine mysql
MySQL RealTime driver loaded.
res_config_mysql.so = (MySQL RealTime Configuration Driver)


I'm also looking as to what CDR viewers there are available, and which
people think are best.  I want to view/report on the calls made within
A*k.

Thanks,

Adrian

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Re: [asterisk-users] Sangoma PRI

2007-08-06 Thread Kevin P. Fleming

David Gomillion wrote:

 Last I checked, the replacement with the new firmware is only for those
 who bought the card in the last year (i.e. the card is still under
 warranty). Those of us who were early adopters cannot enjoy the
 improvements of the upgraded firmware without buying all new cards.

The standard warranty on our cards is two years, and from what I
remember (but I am software geek so I could easily be wrong) the
firmware upgrades are free within the warranty period, not including
shipping costs.

To answer Steve and the other posters: normally what happens in a
cross-ship firmware upgrade situation is that we send you a card (from
our RMA stock) with v2 firmware on it, and you send your card back to
us. When your card arrives, we verify that it works, is under warranty,
upgrade it to v2 firmware, and put it into our RMA stock. You only ever
swap cards once (you don't get your original card back unless you really
want it).

If you have contacted our RMA department in the past and not gotten an
adequate response, I would encourage you to try again. In the past few
months we have begun using SalesForce to track incoming customer
requests and I'm pretty confident every request now gets a response as
it should G

-- 
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - The Genuine Asterisk Experience (TM)

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Re: [asterisk-users] Sangoma PRI

2007-08-06 Thread Julian Lyndon-Smith

Kevin P. Fleming wrote:
 David Gomillion wrote:
 
 Last I checked, the replacement with the new firmware is only for those
 who bought the card in the last year (i.e. the card is still under
 warranty). Those of us who were early adopters cannot enjoy the
 improvements of the upgraded firmware without buying all new cards.
 
 The standard warranty on our cards is two years, and from what I
 remember (but I am software geek so I could easily be wrong) the

Dang! One of mine (TE410P) is 2.5 years ... but the other one (TE405P) 
is definitely younger than that.

/me is going to contact the RMA department tomorrow ...

Thanks for the info.


 firmware upgrades are free within the warranty period, not including
 shipping costs.
 
 To answer Steve and the other posters: normally what happens in a
 cross-ship firmware upgrade situation is that we send you a card (from
 our RMA stock) with v2 firmware on it, and you send your card back to
 us. When your card arrives, we verify that it works, is under warranty,
 upgrade it to v2 firmware, and put it into our RMA stock. You only ever
 swap cards once (you don't get your original card back unless you really
 want it).
 
 If you have contacted our RMA department in the past and not gotten an
 adequate response, I would encourage you to try again. In the past few
 months we have begun using SalesForce to track incoming customer
 requests and I'm pretty confident every request now gets a response as
 it should G
 


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Re: [asterisk-users] low-level dump for PRI dchan debugging

2007-08-06 Thread Darryl Dunkin

Have you completely ignored the telco suggestion and attempted pri_cpe?
Sounds like a miscommunication in settings to me.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Erik
Anderson
Sent: Monday, August 06, 2007 12:02
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] low-level dump for PRI dchan debugging

lpdlnx04*CLI pri show span 1
Primary D-channel: 24
Status: Provisioned, Down, Active
Switchtype: Nortel DMS100
Type: Network

I know it's odd, but the telco instructed me to set my equipment as
the network end...hence pri_net:

/etc/zaptel.conf
loadzone=us
defaultzone=us

#Sangoma A102 port 1 [slot:10 bus:2 span: 1]
span=1,1,0,esf,b8zs
bchan=1-8
dchan=24


/etc/asterisk/zapata.conf
[trunkgroups]

[channels]
context=default
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
relaxdtmf=yes
rxgain=0.0
txgain=0.0
group=1
callgroup=1
pickupgroup=1

immediate=no

;;Sangoma A102 port 1 [slot:10 bus:2 span: 1]
switchtype=dms100
context=from-pstn
group=1
signalling=pri_net
channel = 1-8

There you go.

As an aside, turns out that it's a national holiday in CA, so the
Sangoma support guys are on vacation for the day.

-erik

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Re: [asterisk-users] low-level dump for PRI dchan debugging

Darryl Dunkin wrote:
 wanpipemon is the way to do it as far as I know.

 For starters, what do your zaptel/zapata configs look like?

 I would first verify that your D-channel is set properly, you can view
 that in the console as follows:
 asterisk pri show span 1/0
 Primary D-channel: 24
 Status: Provisioned, Up, Active
 Switchtype: National ISDN

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Erik
 Anderson
 Sent: Monday, August 06, 2007 11:09
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [asterisk-users] low-level dump for PRI dchan debugging

 I've been going back and forth with my telco for several days, trying
 different configurations to get a new PRI to come up.  The bchannels
 are all up and the T1 is not in alarm status.  The dchannel refuses to
 come up however.  We've tried ni2, qsig, and now dms100 for the
 switchtype.  The telco tech I've been working with says that he's been
 sending reset all channels signals to my system, to which he's
 getting an establish remote response from my asterisk box.  I've
 been running a packet dump (wanpipemon -i w1g1 -c trd) of my d-channel
 this whole time and have yet to see a single incoming packet.  I
 believe I *should* be seeing an incoming packet when he sends the
 reset, correct?  Is there any way to do a completely raw dump of the
 d-channel?

 Here are my specs:
 linux-2.6.16
 libpri-1.3.5
 zaptel-1.2.19
 asterisk-1.2.21.1

 The PRI interface is a Sangoma A102...it's running the latest firmware
 and I'm running wanpipe-2.3.4-12 for the sangoma drivers.

 Any ideas?

   
You should never be the signaling source, you are always a slave to the 
provider, go with pri_cpe and see if things go better.

Anthony

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Re: [asterisk-users] low-level dump for PRI dchan debugging

Erik Anderson wrote:
 On 8/6/07, Anthony Francis [EMAIL PROTECTED] wrote:
   
 also in asterisk do:
 pri intense debug span 1
 Then you should see UA's and SABME's, If you don't, your not talking to
 them.
 

 I see plenty of SABMEs, but nothing else:

   
 [ 02 01 7f ]
 

   
 Unnumbered frame:
 SAPI: 00  C/R: 1 EA: 0
  TEI: 000EA: 1
   M3: 3   P/F: 1 M2: 3 11: 3  [ SABME (set asynchronous balanced mode 
 extended) ]
 0 bytes of data
 
 Sending Set Asynchronous Balanced Mode Extended

   
 [ 02 01 7f ]
 

   
 Unnumbered frame:
 SAPI: 00  C/R: 1 EA: 0
  TEI: 000EA: 1
   M3: 3   P/F: 1 M2: 3 11: 3  [ SABME (set asynchronous balanced mode 
 extended) ]
 0 bytes of data
 
 Sending Set Asynchronous Balanced Mode Extended
 lpdlnx04*CLI pri
   
 [ 02 01 7f ]
 
 lpdlnx04*CLI pri
   
 Unnumbered frame:
 SAPI: 00  C/R: 1 EA: 0
  TEI: 000EA: 1
   M3: 3   P/F: 1 M2: 3 11: 3  [ SABME (set asynchronous balanced mode 
 extended) ]
 0 bytes of data
 
 Sending Set Asynchronous Balanced Mode Extended

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Yeah you are sending the SABME's because you think you are the master, 
they are not replaying with a UA because they think they are the master, 
you should def be pri_cpe.

There is one other potential cause here, you may not have had the 
sangoma install patch and rebuild zaptel. Not doing that can cause a D 
channel lockout on your end, but the provider should be able to see the 
the D is in lockout.

Anthony

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Re: [asterisk-users] low-level dump for PRI dchan debugging

On 8/6/07, Anthony Francis [EMAIL PROTECTED] wrote:
 You should never be the signaling source, you are always a slave to the
 provider, go with pri_cpe and see if things go better.

That's what I've experienced in the past, but they were adamant about
me being the network end.  I tried switching to cpe for the heck of
it, but that didn't help...

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Re: [asterisk-users] Teliax Quality of Service

2007-08-06 Thread Douglas Garstang

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of SIP
 Sent: Monday, August 06, 2007 8:56 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Teliax Quality of Service

 Steve Totaro wrote:
  Anthony Francis wrote:

  Tim Panton wrote:

  On 5 Aug 2007, at 06:54, Douglas Garstang wrote:

  I don't think creating a network without a single point of
failure
  is unreasonable.

  It's impossible. I can't think of a single example where this
  actually exists.

  Getting even close is hideously expensive.

  Tim, speaking for himself :-)

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  In fact, the only people who would say something like this are
folks
 who
  have never PHYSICALLY implemented a network, they simply don't
  understand the limitations involved.

  Anthony

  What if a train derails and slices through the main fiber
connections.
  OK, so you have XO, Global Crossing, Verizon, and UCN all for
  redundancy.  Well guess what?  They are all most likely running over
  those strands of fiber.  You better have a VSAT connection too!

Good grief. No, you have two physical collocations. One in say in Nevada
or Idaho (least likely states to suffer natural disasters) and one in
New York.

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Re: [asterisk-users] Teliax Quality of Service

2007-08-06 Thread Douglas Garstang

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Stephen Bosch
 Sent: Monday, August 06, 2007 9:42 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Teliax Quality of Service

 Eric ManxPower Wieling wrote:
  Douglas Garstang wrote:
  Let's assume for a moment that it's impossible. That does not mean
 adding additional servers and additional networking equipment does not
add
 value, or is a worthless endeavour.

  I agree with that.  At least two people that I know run ITSPs.  Each
  time they have an outage (which is not very often) they DO learn
from
  the experience and work to avoid a future outage cause by the same
 issue.

  You would be surprised at how many little things can cause an
outage.

 My own experience is that increasing failover redundancy, which adds
 correspondingly increasing complexity, also increases the odds of an
 outage.

 It is very rare that failover redundancy works as intended during an
 actual failover, no matter how many times you simulate it.

 I would rather have a simple network design where the cause of
failure,
 when it happens, is obvious and quickly corrected. For example, I
would
 rather have replacement parts on the shelf and be able to slap them in
 quickly than be running hot standbys and paying for the electricity,
and
 then have the thing break anyway when there's a failure.

This might work for a web service, but people have a zero tolerance for
no phone service. They expect to be able to pick up their handset, and
get a functional dialtone immediately.

Adding additional servers, additional network components, and some
smarts into your design saves being woken at 3am when a server fails.

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Re: [asterisk-users] low-level dump for PRI dchan debugging

On 8/6/07, Anthony Francis [EMAIL PROTECTED] wrote:
 Yeah you are sending the SABME's because you think you are the master,
 they are not replaying with a UA because they think they are the master,
 you should def be pri_cpe.

Tried it...no go.

 There is one other potential cause here, you may not have had the
 sangoma install patch and rebuild zaptel. Not doing that can cause a D
 channel lockout on your end, but the provider should be able to see the
 the D is in lockout.

I re-patched zaptel, compiled, and re-installed.  No difference.

I think I'm just going to have to wait until tomorrow when I can get
both Sangoma and the telco on the phone.

-erik

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[asterisk-users] Call Center SoftPhone with Auto Answer

2007-08-06 Thread Joao Pereira

Hello
I need a Softphone with auto answer where users can't turn it off.
Does someone knows a softphone where users can't turn the auto answer off?
Or is there any way Asterisk could force the clients to answer the phone?

Thanks
Regards
Joao Pereira

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Re: [asterisk-users] AgentCallBackLogin vsAddQueueMember

2007-08-06 Thread Delca

I'd like to know what alternative is available for those who run a
call centre with dynamic agent-queue allocation.

We have people monitoring the queues and assigning agents depending on
the queue demand.

cheers!
Santiago

On 7/5/07, Martin Schrott - thinking:systems [EMAIL PROTECTED] wrote:
 sorry, was only for users list...
 Hi Kevin,
 Hi list,

 you are right, acting now is not needed, when callbacklogin will be removed
 anywhere in future...
 But thinking how to realice alternatives can't be so wrong.

 Callbacklogin gives a very simple way to use more queues for one agent,
 which only has to logon to only one system.
 No need to make dbs or tables for saving, where the agent has to be logged
 in. No need to create your own login functions. No additional tables, which
 members are logged in.
  Just one entry in queues.conf and agents.conf
 This is simple.

 For sure, it would also be possible to use addqueuemembers functionality:
 -make own tables where you save, in which queues each member has to be
 logged in.
 -create a table, to see wich members exist and which are logged in. Do not
 forget the destination to call them.
 -create a login functionallity, to use your tables.
 -Then add the member to each queue by calling aqm once for each queue. (Our
 cpu will thank us) for using it so much.
 -do not think of logs. (there are patches helping you... and members-name,
 wich you can use... try how)
 It is as simple as callbacklogin ;-)

 Next difficulty is, using agent-groups... When we use aqm to call different
 groups, we only have to make groups in agents.conf and put them into the
 queues.
 That is it.

 But no problem, we also can create additional tables and script a little
 bit. We do not need to sleep at night.

 To summerice: using aqm we would have to make own tables of groups. Then we
 have to make tables of members, that are logged in. Then we have to read
 this tables, check who is logged in, then call aqm for each member that is
 logged in and put it into each queue, the third table has saved this member
 for...

 !!! Only to write it here is more work then using agent callbacklogin!
 scripting it would be crazy, when callbacklogin does it for us !!!

 So we can only hope, that there will be an alternative application, that
 works like callbacklogin.
 I am sure, a lot of cc designers will stop upgrading, if callbacklogin is
 removed and now new simmilar application is provided! Nobody can effort to
 do this additional work to change all dialplans. :-)

 Where is the problem keeping callbacklogin as additional feature in future
 versions. Nobody has to support or change it. Just keep it working. Or
 create a new application that does all the same, when you can't stand it.

 If you can tell me in thre lines how to use addqueuemember doing all things
 we need from callbacklogin app, then I will use it from today on.
 Othervise it is a reinventing of the wheel.

 Hope there will be a alternate application in newer versions of asterisk.

 Thanks

 Martin



 - Original Message -
 From: Kevin P. Fleming [EMAIL PROTECTED]
 To: Alan Ferrency [EMAIL PROTECTED]
 Cc: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 Sent: Wednesday, April 11, 2007 11:45 PM
 Subject: Re: [asterisk-users] AgentCallBackLogin vs AddQueueMember


 Alan Ferrency wrote:

  However, this is not what we need. This adds a phone channel to the
  queue, and does not track which person is using that phone. This means
  that all queue activity is associated with a SIP channel in the logs,
  which is not acceptable.

 Right. This is why we added the 'membername' argument to the
 AddQueueMember application, so that queue logs can reflect a logical
 name for the queue member, regardless of what channel/interface they
 logged in from.

  Using this map of people to phones, our dial plan would then need to
  ensure that:
  - a person cannot be logged into more than one phone
  - only one person at a time can be logged into a phone
  - queue activity logs are associated with a person, not a phone

 For points #1 and #2, you are correct that this logic will have to be
 built. Point #3 is already taken care of by the addition of the
 'membername' as I commented on above.

 However, I personally see this as a huge benefit; I much prefer Asterisk
 to provide mechanisms for users to do things, but not the policy on how
 they are to be used. When chan_agent is in use, you don't get to decide
 what to do if a second user tries to log in from the same channel, that
 has been decided for you. If instead you write that logic in the
 dialplan (or start from an example you find in the docs, on the wiki,
 etc.) you can completely control how the system behaves.

  Can the AddQueueMember solution handle the equivalent of autologoff if
  a queue member fails to answer a queued call in time?

 Absolutely; the example in doc/queues-with-callback-members.txt shows
 how to do it.

  To me, saying We

Re: [asterisk-users] Free sitting

2007-08-06 Thread Time Bandit

 In fact, my questions are more about usage than about technical background.
 For instance, I doubt a user will log his system off when leaving : some
 don't even turn their PC off.


 Does anyone has an experience to share about that ?
When I tried it, when a user login at a phone, it replaced any
previously logged one.

hope that help

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Re: [asterisk-users] Free sitting