Re: [asterisk-users] Question on Priorities

[Philipp Kempgen]

Yuan LIU wrote:

>> From: "Rizwan Hisham" <[EMAIL PROTECTED]>
>> Date: Sat, 31 Mar 2007 17:01:51 +0500
>>
>> [inbound-sip]
>> exten => uxbod,1,Dial(sip/1001,20,jt)
>> exten => uxbod,n,Hangup
>>
>> exten => uxbod,102,PlayBack(uxbod)
>> exten => uxbod,103,VoiceMail([EMAIL PROTECTED],s)
>> exten => uxbod,104,Hangup()
>>
>> here if dial fails then n+101 =102 extension will get executed unless you
>> use j option in dial application and priority jumping has to be set to
>> priorityjumping=yes in the general section of your extensions.conf file.
>>
>> In your dialplan i dont know y you r forcing the caller to goto voicemail
>> even if the call  has already answered. I hope you understand my
>> modification in your dialplan.
>>
>> On 3/31/07, --[ UxBoD ]-- <[EMAIL PROTECTED]> wrote:
>>> Hi,
>>>
>>> I am attempting to change my dialplan to use 'n' priorities and labels
>>> for easier reading, and less re-numbering :) but how do you handle the
>>> plus 101 ? In my extensions.conf I have a simple plan for testing :-
> 
> There is a simpler way, by using label.

Priority jumping is deprecated anyways. Better use something
like Goto(s-${DIALSTATUS},1). See extensions.conf for examples.

Regards,
  Philipp

-- 
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Re: [asterisk-users] Question on Priorities

[--[ UxBoD ]--]

Cool. That is nice and clean :) Many thanks.

On Sat, 31 Mar 2007 23:32:45 -0700
"Yuan LIU" <[EMAIL PROTECTED]> wrote:

> >From: "Rizwan Hisham" <[EMAIL PROTECTED]>
> >Date: Sat, 31 Mar 2007 17:01:51 +0500
> >
> >[inbound-sip]
> >exten => uxbod,1,Dial(sip/1001,20,jt)
> >exten => uxbod,n,Hangup
> >
> >exten => uxbod,102,PlayBack(uxbod)
> >exten => uxbod,103,VoiceMail([EMAIL PROTECTED],s)
> >exten => uxbod,104,Hangup()
> >
> >here if dial fails then n+101 =102 extension will get executed
> >unless you use j option in dial application and priority jumping has
> >to be set to priorityjumping=yes in the general section of your
> >extensions.conf file.
> >
> >In your dialplan i dont know y you r forcing the caller to goto
> >voicemail even if the call  has already answered. I hope you
> >understand my modification in your dialplan.
> >
> >On 3/31/07, --[ UxBoD ]-- <[EMAIL PROTECTED]> wrote:
> >>
> >>Hi,
> >>
> >>I am attempting to change my dialplan to use 'n' priorities and
> >>labels for easier reading, and less re-numbering :) but how do you
> >>handle the plus 101 ? In my extensions.conf I have a simple plan
> >>for testing :-
> 
> There is a simpler way, by using label.
> 
> [inbound-sip]
> exten => uxbod(ntest),1,Dial(sip/1001,20,t)
> exten => uxbod,n,PlayBack(uxbod)
> exten => uxbod,n,Hangup()
> exten => uxbod,ntest+101,PlayBack(uxbod)
> exten => uxbod,n,VoiceMail([EMAIL PROTECTED],s)
> exten => uxbod,n,Hangup()
> 
> Yuan Liu
> 
> >>[inbound-sip]
> >>exten => uxbod,1,Dial(sip/1001,20,t)
> >>exten => uxbod,n,PlayBack(uxbod)
> >>exten => uxbod,n,VoiceMail([EMAIL PROTECTED],s)
> >>exten => uxbod,n,Hangup()
> >>exten => uxbod,103,PlayBack(uxbod)
> >>exten => uxbod,104,VoiceMail([EMAIL PROTECTED],s)
> >>exten => uxbod,105,Hangup()
> >>
> >>So when the extension has to add 101 do I just do n+101 ?
> >>
> >>TIA
> >>
> >>--
> >>This message has been scanned for viruses and dangerous content by
> >>MailScanner, and is
> >>believed to be clean.
> >>
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> >>
> >
> >
> >
> >--
> >Regards
> >Rizwan Hisham
> >Software Engineer
> 
> 
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Re: [asterisk-users] Question on Priorities

[--[ UxBoD ]--]

Okay, I have changed it too :-

[inbound-sip]
exten => uxbod(u1),1,Dial(sip/1001,20,t)
exten => uxbod,n,PlayBack(uxbod)
exten => uxbod,n,Hangup()
exten => uxbod,u1+101,PlayBack(uxbod)
exten => uxbod,n,VoiceMail([EMAIL PROTECTED],s)
exten => uxbod,n,Hangup()

but when I do a extensions reload I get :-

[Apr  1 10:12:14] WARNING[20191]: pbx_config.c:2256 pbx_load_config:
Invalid priority/label 'u1' at line 75

On Sat, 31 Mar 2007 23:32:45 -0700
"Yuan LIU" <[EMAIL PROTECTED]> wrote:

> >From: "Rizwan Hisham" <[EMAIL PROTECTED]>
> >Date: Sat, 31 Mar 2007 17:01:51 +0500
> >
> >[inbound-sip]
> >exten => uxbod,1,Dial(sip/1001,20,jt)
> >exten => uxbod,n,Hangup
> >
> >exten => uxbod,102,PlayBack(uxbod)
> >exten => uxbod,103,VoiceMail([EMAIL PROTECTED],s)
> >exten => uxbod,104,Hangup()
> >
> >here if dial fails then n+101 =102 extension will get executed
> >unless you use j option in dial application and priority jumping has
> >to be set to priorityjumping=yes in the general section of your
> >extensions.conf file.
> >
> >In your dialplan i dont know y you r forcing the caller to goto
> >voicemail even if the call  has already answered. I hope you
> >understand my modification in your dialplan.
> >
> >On 3/31/07, --[ UxBoD ]-- <[EMAIL PROTECTED]> wrote:
> >>
> >>Hi,
> >>
> >>I am attempting to change my dialplan to use 'n' priorities and
> >>labels for easier reading, and less re-numbering :) but how do you
> >>handle the plus 101 ? In my extensions.conf I have a simple plan
> >>for testing :-
> 
> There is a simpler way, by using label.
> 
> [inbound-sip]
> exten => uxbod(ntest),1,Dial(sip/1001,20,t)
> exten => uxbod,n,PlayBack(uxbod)
> exten => uxbod,n,Hangup()
> exten => uxbod,ntest+101,PlayBack(uxbod)
> exten => uxbod,n,VoiceMail([EMAIL PROTECTED],s)
> exten => uxbod,n,Hangup()
> 
> Yuan Liu
> 
> >>[inbound-sip]
> >>exten => uxbod,1,Dial(sip/1001,20,t)
> >>exten => uxbod,n,PlayBack(uxbod)
> >>exten => uxbod,n,VoiceMail([EMAIL PROTECTED],s)
> >>exten => uxbod,n,Hangup()
> >>exten => uxbod,103,PlayBack(uxbod)
> >>exten => uxbod,104,VoiceMail([EMAIL PROTECTED],s)
> >>exten => uxbod,105,Hangup()
> >>
> >>So when the extension has to add 101 do I just do n+101 ?
> >>
> >>TIA
> >>
> >>--
> >>This message has been scanned for viruses and dangerous content by
> >>MailScanner, and is
> >>believed to be clean.
> >>
> >>___
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> >>
> >>asterisk-users mailing list
> >>To UNSUBSCRIBE or update options visit:
> >>http://lists.digium.com/mailman/listinfo/asterisk-users
> >>
> >
> >
> >
> >--
> >Regards
> >Rizwan Hisham
> >Software Engineer
> 
> 
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> 
> 
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Re: [asterisk-users] CallerID + Name

[Ola Lidholm]

On 29 mar 2007, at 23.01, Rob Schall wrote:


We have the caller id with name option enabled with our provider,
however, our polycom 501 phones will only display the number of the
incoming call. Is there a way to see the callerid name from the cli  
when

the call is coming in (like a print in the dial plan)? I'm not sure if
the problem is with asterisk or our phones. I did turn on the
calleridpres option in zapata, but I'm unsure what else needs to be  
set.


We use Polycom Phones at my work, and there is nothing special you  
need to do to make them display the name.


Anyway, you can use NoOp(${CALLERID}) to print the callerid in the  
dial plan. You should se something like this if everything is  
correct: "Joe Schmoo <123456789>".


/Ola Lidholm
[EMAIL PROTECTED]

"Whatever one man is capable of conceiving, other men are able to  
achieve." - Jules Verne.


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Re: [asterisk-users] Speed Dial Application in *

[Ola Lidholm]

On 30 mar 2007, at 14.19, Chris Nighswonger wrote:


Hi all,
 Is there a "speed dial" type application in *? The NEC PBX we
currently use has a feature which allows any phone to access a
system-wide speed dail database simply by keying the speed-dial number
and pressing the 'redial' key from any extension. Even using a vinella
phone on an sli the user can dial 77+speedial# and access this
directory.
 Does * have a similar feature?


You can create this in your dialplan (extensions.conf).

/Ola Lidholm
[EMAIL PROTECTED]

"Whatever one man is capable of conceiving, other men are able to  
achieve." - Jules Verne.


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Re: [asterisk-users] Sponsored development - Monodirectional audiohandling

[Dovid B]

Try posting this on the developers list.

- Original Message - 
From: "Edoardo Serra" <[EMAIL PROTECTED]>

To: 
Sent: Saturday, March 31, 2007 6:42 PM
Subject: [asterisk-users] Sponsored development - Monodirectional 
audiohandling




Hi Guys,
we're needing a special implementation on Asterisk
Our intention is to contribute the development and share back the code to 
Asterisk community


Here is what we need:

- An option to Asterisk Dial command which, if used, when calls is 
answered gives monodirectional audio

(Caller should hear the called party but not vice-versa)

- A DTMF sequence (maybe handled in features.conf) for the Caller to start 
to have bidirectional audio


- When the Callers makes the audio 'bidirectional' an Event should be 
generated so that we can see it from the manager API


The purpose of thisi implementation is to deal with some carriers that 
give us the call as ANSWERED when the called party is still ringing.
Our billing software is billing the user (and the carrier is billing us) 
even with unsuccessful calls.


This way we can start billing when the user press the DTMF sequence to 
unlock audio (even if carriers bill us wrongly)


Someone wants to help ??

Regards

Edoardo Serra
WeBRainstorm S.r.l.

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Re: [asterisk-users] Speed Dial Application in *

[Dovid B]

You can try putting out a bounty to make a new app in asterisk or set a 
variable in the asterisk db.

Exten => _*XX,1,nswer
exten => _*XX,2,Set(speeddial=${EXTEN:1})
exten => _*XX,3,Goto(setspeeddial,s,1)
[speeddial]
eexten => s,1,Set(TIMEOUT(digit)=3)
exten => s,2,Set(TIMEOUT(response)=3)
exten => s,3,BackGround(enter-number-to-set) ; You have to record the sound 
file

exten => _X.,1,Set(DB(speeddial/${speedial})=${EXTEN})
exten => _X.,2,Playback(the-speed-dial) ; You have to record the sound file
exten => _X.,3,SayNumber(${speeddial},f)
exten => _X.,4,Playback(has-been-set-to); You have to record the sound 
file

exten => _X.,5,SayDigits(${EXTEN})
exten => _X.,6,Hangup

And in Dial Plan
exten => _77.,1,Dial(SIP/${DB(speeddial/${EXTEN:2})[EMAIL PROTECTED]) ; Dials 
the speed dial


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


Sent: Friday, March 30, 2007 3:19 PM
Subject: [asterisk-users] Speed Dial Application in *



Hi all,
 Is there a "speed dial" type application in *? The NEC PBX we
currently use has a feature which allows any phone to access a
system-wide speed dail database simply by keying the speed-dial number
and pressing the 'redial' key from any extension. Even using a vinella
phone on an sli the user can dial 77+speedial# and access this
directory.
 Does * have a similar feature?

Thanks,
Chris
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Re: [asterisk-users] Security on long distance calls

[Dovid B]

Or you can do

exten => _011.,1,Authenticate(1234)
exten => _011.,2,Dial(SIP/[EMAIL PROTECTED])

Also this is a bit more complicated but you can do it by sip extension. If CID 
of phone = phone that is allowed then let it go out. This will be hard 
considering you will have to make a gotoif for every extension that you want to 
allow to call intl.
  - Original Message - 
  From: Rizwan Hisham 
  To: Asterisk Users Mailing List - Non-Commercial Discussion 
  Sent: Friday, March 30, 2007 11:18 AM
  Subject: Re: [asterisk-users] Security on long distance calls


  For operator:

  [longdistance]
  include=> local
  include=> International

  for extensions:

  [localcalls]
  include=> local

  now assign longdistance context to operator and localcalls context to every 
other user for whom you want to restrict intl calls 

  [local]
  should include all local extension codes

  [International]
  should include all international extension codes

  you get my point?


  On 3/30/07, Stefano Corsi <[EMAIL PROTECTED]> wrote:
Hello, 

which kind of method could you use to inhibit long distance calls to
_some_ extensions?
Is there a way to do it with freepbx or you have to do it manually in
the config files? I wouldn't like to set a route password, because 
that is not confortable for the pbx operator. I just would like the
operator being able to call whatever number, while the extensions
should only be able to make local calls.

Thanks
Stefano


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  -- 
  Regards
  Rizwan Hisham
  Software Engineer 


--


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Re: [asterisk-users] REG : H.323 Configurations with Asterisk

[Dovid B]

1) Good luck with H.323 on asterisk.
2) Should be handled by Asterisk it's self.

- Original Message - 
From: "Anisha Kumar" <[EMAIL PROTECTED]>

To: 
Sent: Friday, March 30, 2007 7:03 AM
Subject: FW: [asterisk-users] REG : H.323 Configurations with Asterisk



  Hi ,
I am new to Asterisk community. I have some queries. Please
guide me on the following :

1)I want to configure H.323 softphones, How do I do that ? I am
using the Asterisk windows versio 0.60.There is no chan_h.323.so file
.Also there are no help files or documents for configuring h.323
softphones. Can someone guide me inthis regard ?

2)And I want to know how the supplementary features for the
H.323 like Transfer,hold and forward are handled in Asterisk. Is that
part of the OpenH323 library or the Asterisk takes care of them.

  Regards,
  Anisha 
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[asterisk-users] Re: asterisk-users Digest, Vol 33, Issue 1

[fb]

Je suis absent du  2/04/2007 au 11/04/2007.

Je répondrai à votre message dès mon retour. Pour toute urgence, contacter
Emmanuelle Parache Moga ou Cédric Buzay.


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Re: [asterisk-users] Question on Priorities

[Philipp Kempgen]

--[ UxBoD ]-- wrote:

> [inbound-sip]
> exten => uxbod(u1),1,Dial(sip/1001,20,t)
> exten => uxbod,n,PlayBack(uxbod)
> exten => uxbod,n,Hangup()
> exten => uxbod,u1+101,PlayBack(uxbod)
> exten => uxbod,n,VoiceMail([EMAIL PROTECTED],s)
> exten => uxbod,n,Hangup()
> 
> but when I do a extensions reload I get :-
> 
> [Apr  1 10:12:14] WARNING[20191]: pbx_config.c:2256 pbx_load_config:
> Invalid priority/label 'u1' at line 75

The label belongs to the priority not to the extension.


Regards,
  Philipp

-- 
amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
 Let's use IT to solve problems and not to create new ones.
   Asterisk? -> http://www.das-asterisk-buch.de

Geschäftsführer: Stefan Wintermeyer
Handelsregister: Neuwied B 14998
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Re: [asterisk-users] Meetme question

[Dovid B]

Create two seperate extensions. One for the admin and one for the regular 
users that go to the same room. The issue you will have then is that the 
admin will have to call in first to create the dynamic room.


- Original Message - 
From: "Adrian Marsh" <[EMAIL PROTECTED]>

To: 
Sent: Saturday, March 31, 2007 3:58 PM
Subject: [asterisk-users] Meetme question


Hi,

I'm experimenting with the Meetme feature of Asterisk 1.2,

exten => 2095,1,MeetMe(|Ds)

This almost gives me what I want, where each employee can create their own 
on-the-fly conferences with a personal Conference Number and PIN.  However, 
as the PIN is actually set by the first callee, then its subject to problems 
(first callee might enter the wrong PIN, and then no-one else can join).


What I really want is something that covers the below:

- One call-in number
- Employees get their own unique conference # (this could be their own 
extension), and can set a public PIN that only they can change.
- I don't really want a www-based system, as most of my users are usually 
mobile, and might not have access to the corporate intranet.



Thanks,

Adrian Marsh


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Re: [asterisk-users] h323

[Dovid B]

Did you compile H.323 for asterisk and then make install asterisk ?

- Original Message - 
From: "Pezhman Lali" <[EMAIL PROTECTED]>

To: 
Sent: Wednesday, March 28, 2007 4:30 PM
Subject: [asterisk-users] h323


hi
After compiling and installing pwlib and openh323 ,
the asterisk, give the folloing error.
please tell me where the problem is ?
Best
Mani


*CLI> -- Executing Dial("SIP/2.2.2.2-086f5ac0",
"H323/[EMAIL PROTECTED]|60") in new stack
Mar 28 14:17:23 WARNING[11985]: channel.c:2576
ast_request: No translator path exists for channel
type H323 (native 4) to 256
Mar 28 14:17:23 NOTICE[11985]: app_dial.c:1059
dial_exec_full: Unable to create channel of type
'H323' (cause 0 - Unknown)
 == Everyone is busy/congested at this time (1:0/0/1)
 == Auto fallthrough, channel 'SIP/2.2.2.2-086f5ac0'
status is 'CHANUNAVAIL'





Don't get soaked.  Take a quick peek at the forecast
with the Yahoo! Search weather shortcut.
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[asterisk-users] Weird extension behavior

[Mark Hennessy]

Hi, I'm using Asterisk with two Cisco 7960 phones using SIP.
I'm seeing the following weird behavior:
SIP Phome 1 is extension 4002
SIP Phone 2 is extension 4003

I call 4002 from 4003 and that works fine.
I call 4003 from 4002, and it rings locally to 4002, never gets to 4003.

I'm able to send a config query packet to 4003 from the asterisk  
console and get a response, when I send one to 4002 there is no respone.


I know that both phones pull down their config via TFTP properly, I  
look in the network settings and see that 4002 has been given an IP of  
x.y.z.201 and 4003 has been given an IP of x.y.z.202 and the asterisk  
box is running on x.y.z.74.


I combed through all of the config files in both Asterisk's config and  
the TFTP-downloaded configs for the phones looking for any possible  
instance of 4003 being transposed for 4002 or vice versa and was not  
able to find any.


What additional information is necessary to provide to trace down and  
resolve this issue?


AFAICT, the server is using Asterisk 1.2.x and beyond the 7960 phones,  
no other specialized hardware is in use.


--
Mark P. Hennessy


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RE: [asterisk-users] Multi-Level Queue

[Chris Bagnall]

How about using one queue to provide the caller with music and position info, 
then delayed dialling on the mobiles:

[queue]
timeout=30
retry=0
joinempty=yes
member => SIP/201
member => SIP/202
member => SIP/203
member => SIP/204
member => Local/@delaydial
member => Local/@delaydial

[delaydial]
exten => _X.,1,Wait(10)
exten => _X.,1,Dial(Local/[EMAIL PROTECTED])

[outbound]
exten => _X.,1,Dial(IAX2/provider/${EXTEN},,gM(acceptcall))
exten => _X.,t,Playback(toolate)

macro acceptcall {
Read(ACCEPT,press1,1);
if (${ACCEPT} != 1)
MACRO_RESULT=CONTINUE;
};
[queue] goes in queues.conf, obviously. [delaydial] and [outbound] go in 
extensions.conf. acceptcall is an AEL macro. Modify as required. The sound file 
"toolate" is played back to the caller who doesn't press 1. You can remove it 
if you wish. I think this fits most of your requirements:

The SIP clients won't have to ack to receive the call in this scenario, only 
the mobile clients. Personally, I think that's preferred, since SIP clients 
will only go to VM if you tell them to in your dialplan.

In a nutshell, an incoming call flows as follows:

1) call comes into the queue, receives announcements, etc. as you specify.
2) all queue members are dialled, SIP extensions ring immediately, but since 
mobiles are going to the [delaydial] context which gives them a 10 second 
delay, they won't ring immediately.
3) if none of the SIP members pick up the call, the mobiles are dialled. When 
they're connected, the sound file "press1" is played to the other end.
4) Mobile user presses 1 to be connected to the caller. If you end up with VM, 
it obviously doesn't press 1, so call is dropped as soon as one of the others 
picks up.
5) If nothing picks up, queue retries after 30 seconds.

To achieve your 20 second SIP / 20 second mobile back-and-forth, change the 
Wait() time in delaydial to 20, and the queue timeout to 40.

You'll need to have a play with it to ensure the timings work out for your 
setup (e.g. it takes a good 5-6 seconds for mobiles to start ringing after the 
Dial() command is sent here - that might be different where you are).

Hope that helps.

Regards,

Chris
-- 
C.M. Bagnall, Director, Minotaur I.T. Limited
For full contact details visit http://www.minotaur.it/chris.html
This email is made from 100% recycled electrons

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Re: [asterisk-users] DTMF Corruption Problem in 1.4.2 for SIP RFC2833 plz halp

[Justin Tunney]

I actually meant to take that out before copy and pasting.  It was
just a zany option I tried to see if things would get better but they
didn't.

I will post a bug later this week about this.

On 3/30/07, Kevin P. Fleming <[EMAIL PROTECTED]> wrote:

Justin Tunney wrote:

> rfc2833compensate=yes

Why do you have this turned on? This setting is _ONLY_ for receiving
RFC2833 DTMF from pre-1.4 Asterisk servers, it should never be used for
any other SIP endpoint.
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[asterisk-users] Anyone here have opinion on the Linksys SPA-400?

[Michael Graves]

I may have need for a small multi-port FXO i/f. Anyone have hands-on experience 
with this linksys device?

On past projects I've used SPA-3000 & Digium TDM400 and found them 
unsatisfactory.

Michael



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On Topic: Cheapest Asterisk USB Key? (was: Re: [asterisk-users] Off Topic: Open Source USB Softphone)

[Matthew Rubenstein]

Here's a flipside of this subject: what is the absolute cheapest Linux
device that can be connected to a PC's USB port? That has just enough
power for a minimal Asterisk server running on it. The Asterisk just
maintains a CDR database on its Flash memory, which it periodically
submits over the PC's network connection with an HTTP hit on a remote
full-service Asterisk server? No call handling, DSP or anything really
number crunching, no telephony terminal or other services. The
lowest-performance device that plugs into the USB, with its own Linux
instance. In OEM quantity, under $50? Under $100?


On Sun, 2007-04-01 at 02:51 -0700,
[EMAIL PROTECTED] wrote:
> Date: Sat, 31 Mar 2007 16:02:06 -0500
> From: "Mike Lynchfield" <[EMAIL PROTECTED]>
> Subject: Re: [asterisk-users] Off Topic: Open Source USB Softphone
> To: [EMAIL PROTECTED],   "Asterisk Users Mailing List -
> Non-Commercial
> Discussion"  
> Message-ID:
> <[EMAIL PROTECTED]>
> Content-Type: text/plain; charset="iso-8859-1"
> 
> sip would be the required one as iax..well..
> 
> also openwengo wont work.. to much overhead .. broswrer needed.. ie
> component + flash + css+js etc.. not viable..
> 
> so im also asking anyone have one ? since ihave a supply of around
> 2000 of
> the vonage usb stick OEM..
> 
> On 3/30/07, Michael Van Donselaar <[EMAIL PROTECTED]>
> wrote:
> >
> > Which USB Phone?  I have written custom versions of iaxcomm for
> various
> > people,
> > and have a version that works with the Yealink phone.
> >
> > On Thu, 29 Mar 2007 11:33:07 -0300, "Luis Claudio Santos" <
> > [EMAIL PROTECTED]>
> > wrote:
> >
> > >I need a softphone - for usb phone devices - that I can alter
> (insert
> > logo,
> > >menu, etc).
> > >
> > >Does somebody know such one?
> > >
> > >[]s
> >
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> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> 
> 
> 
> -- 
> Mike
> Sales Manager
> http://www.voicemeup.com
> Making it happen
> 1.877.807.VOIP (8647)
> 1.514.312.7030 
-- 

(C) Matthew Rubenstein

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Re: On Topic: Cheapest Asterisk USB Key? (was: Re: [asterisk-users] Off Topic: Open Source USB Softphone)

[Michael Graves]

Probably not exactly what you're looking for but Astlinux runs on Gumstix. 
Would be suitable for prototyping.

Michael

On Sun, 01 Apr 2007 09:08:17 -0400, Matthew Rubenstein wrote:

>   Here's a flipside of this subject: what is the absolute cheapest Linux
>device that can be connected to a PC's USB port? That has just enough
>power for a minimal Asterisk server running on it. The Asterisk just
>maintains a CDR database on its Flash memory, which it periodically
>submits over the PC's network connection with an HTTP hit on a remote
>full-service Asterisk server? No call handling, DSP or anything really
>number crunching, no telephony terminal or other services. The
>lowest-performance device that plugs into the USB, with its own Linux
>instance. In OEM quantity, under $50? Under $100?


>On Sun, 2007-04-01 at 02:51 -0700,
>[EMAIL PROTECTED] wrote:
>> Date: Sat, 31 Mar 2007 16:02:06 -0500
>> From: "Mike Lynchfield" <[EMAIL PROTECTED]>
>> Subject: Re: [asterisk-users] Off Topic: Open Source USB Softphone
>> To: [EMAIL PROTECTED],   "Asterisk Users Mailing List -
>> Non-Commercial
>> Discussion"  
>> Message-ID:
>> <[EMAIL PROTECTED]>
>> Content-Type: text/plain; charset="iso-8859-1"
>> 
>> sip would be the required one as iax..well..
>> 
>> also openwengo wont work.. to much overhead .. broswrer needed.. ie
>> component + flash + css+js etc.. not viable..
>> 
>> so im also asking anyone have one ? since ihave a supply of around
>> 2000 of
>> the vonage usb stick OEM..
>> 
>> On 3/30/07, Michael Van Donselaar <[EMAIL PROTECTED]>
>> wrote:
>> >
>> > Which USB Phone?  I have written custom versions of iaxcomm for
>> various
>> > people,
>> > and have a version that works with the Yealink phone.
>> >
>> > On Thu, 29 Mar 2007 11:33:07 -0300, "Luis Claudio Santos" <
>> > [EMAIL PROTECTED]>
>> > wrote:
>> >
>> > >I need a softphone - for usb phone devices - that I can alter
>> (insert
>> > logo,
>> > >menu, etc).
>> > >
>> > >Does somebody know such one?
>> > >
>> > >[]s
>> >
>> > ___
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>> >
>> > asterisk-users mailing list
>> > To UNSUBSCRIBE or update options visit:
>> >http://lists.digium.com/mailman/listinfo/asterisk-users
>> >
>> 
>> 
>> 
>> -- 
>> Mike
>> Sales Manager
>> http://www.voicemeup.com
>> Making it happen
>> 1.877.807.VOIP (8647)
>> 1.514.312.7030 
>-- 

>(C) Matthew Rubenstein

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Re: [asterisk-users] Anyone here have opinion on the Linksys SPA-400?

[Dovid B]

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


Sent: Sunday, April 01, 2007 3:59 PM
Subject: [asterisk-users] Anyone here have opinion on the Linksys SPA-400?


I may have need for a small multi-port FXO i/f. Anyone have hands-on 
experience with this linksys device?


On past projects I've used SPA-3000 & Digium TDM400 and found them 
unsatisfactory.


Michael


Try Sangoma. It has worked great for me. What didnt you like about the 
TDM400 ? If you had issues you could try contacting Digium support. 



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Re: [asterisk-users] Question on Priorities

[--[ UxBoD ]--]

Thank you - Got it now. Makes everything look a lot cleaner :)

On Sun, 01 Apr 2007 11:58:20 +0200
Philipp Kempgen <[EMAIL PROTECTED]> wrote:

> --[ UxBoD ]-- wrote:
> 
> > [inbound-sip]
> > exten => uxbod(u1),1,Dial(sip/1001,20,t)
> > exten => uxbod,n,PlayBack(uxbod)
> > exten => uxbod,n,Hangup()
> > exten => uxbod,u1+101,PlayBack(uxbod)
> > exten => uxbod,n,VoiceMail([EMAIL PROTECTED],s)
> > exten => uxbod,n,Hangup()
> > 
> > but when I do a extensions reload I get :-
> > 
> > [Apr  1 10:12:14] WARNING[20191]: pbx_config.c:2256 pbx_load_config:
> > Invalid priority/label 'u1' at line 75
> 
> The label belongs to the priority not to the extension.
> 
> 
> Regards,
>   Philipp
> 

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

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[asterisk-users] Linking incoming calls

[Ronaldo Zacarias Afonso]

Hi all,

I just want to know how I can make sure that incoming calls to my
asterisk server are being treated by [incoming] section of
extension.conf file.
Thanks in advance.

Ronaldo.
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Re: [asterisk-users] Anyone here have opinion on the Linksys SPA-400?

[Michael Graves]

On Sun, 1 Apr 2007 16:14:32 +0300, Dovid B wrote:


>- Original Message - 
>From: "Michael Graves" <[EMAIL PROTECTED]>
>To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
>
>Sent: Sunday, April 01, 2007 3:59 PM
>Subject: [asterisk-users] Anyone here have opinion on the Linksys SPA-400?


>>I may have need for a small multi-port FXO i/f. Anyone have hands-on 
>>experience with this linksys device?
>>
>> On past projects I've used SPA-3000 & Digium TDM400 and found them 
>> unsatisfactory.
>>
>> Michael

>Try Sangoma. It has worked great for me. What didnt you like about the 
>TDM400 ? If you had issues you could try contacting Digium support. 

My asterisk install is an embedded system so a freestanding FXO i/f is prefered.

I had one of the earliest TDM400s and it was frought with problems. Noisey, 
poor disconnection monitoring, sensitive to IRQ/PCI issues. It was a pain.

Michael

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Re: [asterisk-users] Linking incoming calls

[Dovid B]

You can put in something like
exten => s,x,Noop("Yup it's working right")

and look at the cli to see if it comes up.

- Original Message - 
From: "Ronaldo Zacarias Afonso" <[EMAIL PROTECTED]>

To: 
Sent: Sunday, April 01, 2007 4:28 PM
Subject: [asterisk-users] Linking incoming calls



Hi all,

I just want to know how I can make sure that incoming calls to my
asterisk server are being treated by [incoming] section of
extension.conf file.
Thanks in advance.

Ronaldo.
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[asterisk-users] jiaxclient run error

[Mitko Georgiev]

Hi,

I have a problem when compile jiaxclient-0.0.6.

Mine enviroment is:

WindowsXP
Cygwin



I compile iaxclient library and afterwards I compile jiaxclient:

./configure
make
make sign
make install

Everything is ok with the compile and install - generated files:
jiaxc__x86.jar
jiaxc__x86_md5.jar
libjiaxc.jar
inifile.jar
jiaxclient.jar
jiaxtest.html
jiaxtest.jar

but when I start the .html file applet is not initialized error occured.

Do you have any ideas what is the problem?

Mitko
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Re: [asterisk-users] Linking incoming calls

[Lee Jenkins]

Ronaldo Zacarias Afonso wrote:

Hi all,

I just want to know how I can make sure that incoming calls to my
asterisk server are being treated by [incoming] section of
extension.conf file.
Thanks in advance.



You want to do this kind of check on a consistent basis or are you just 
checking to make sure that calls are coming in through that context to 
test your dialplan?


If the latter, you can use noop

[incoming]
exten=>s,1,Noop(Hey it is in the incoming context)


--

Warm Regards,

Lee


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[asterisk-users] Announcement: Asterisk Service Provider Edition v1.0 Beta

[Olle E Johansson]

The Asterisk Developer Team is proud to announce the Asterisk SPE  
v1.0 Beta Release

for immediate download on tftp.digium.com.
The SPE has been developed as a joint project between Digium, the  
Asterisk Company,
Voop, the European Asterisk Dialtone provider and the Asterisk  
community.


The Asterisk Service Provider Edition is focused on the needs for the  
new breed
of Telecom companies - the Voice over IP Service Providers.  It will  
be available

both as a free download in Open Source and as a commercial product
called Asterisk Commercial Service Provider Edition, ACSPE.

- "We felt the need to focus on being an enabler for this new kind of  
telco,
   making sure that Asterisk fits into their network as well as  
business models
   in a professional way" says Matt Penser, Asterisk innovator. "The  
previous versions

   was more targeted to the needs of the business user, a market where
   Asterisk already is stronger than any other offering on the market".

The Asterisk SPE has a number of new features, that makes it the most  
powerful

platform for these companies. No other Open Source package can deliver a
matching feature set:

- All the features from Asterisk 1.4 and the business edition
- Asterisk VoipRoute(R) technology for SmartRTP(R) bridging
- Asterisk RateRoute(TM) technology for route selection
- Asterisk SpitWall(R) core for SPIT filtering

These new solutions will enhance Asterisk and will help the VSP's to
leap lightyears ahead of their competion.

* Asterisk VoipRoute(R) SmartRTP(R) Bridging


The VoipRoute SmartRTP bridging technology enhances the Asterisk RTP
bridge with a new scheme. In addition to the current RTP bridges -  
the native bridge,
the remote bridge and the hybrid RTP-direct bridge, SmartRTP uses a  
combination
of the BGP IP routing protocols and the TRIP VoIP routing system to  
find the
best and fastest way to route calls between IP nodes on the Internet  
or local network.


- "The SmartRTP bridge system, based on our patented VoipRoute core,
   makes sure that call latency is minimal. We also enhanced it with a
   MediaRescue solution that will capture lost media frames and re- 
insert
   them in the audio or video stream before it reaches the  
destination." says

   Josua Polk, the Asterisk RTP developer.
   "This system implements an Asterisk VoipRoute layer on top of the  
Internet
   and uses Dundi(TM) to automatically discover new SmartRTP relays  
and their
   properties. It practically erases packet loss, jitter and latency  
from the list of
   issues for the provider's support department. We call it SPEake- 
friendly calls!"



* Asterisk RateRoute(TM) Least Cost Routing
-

The RateRoute(TM) solution is only available in the ACSPE due to  
licenses from
other vendors, soon to be disclosed. The RateRoute system analyze the  
call
from fifteen distinct properties and use an external hardware  
accelerator to
find the best route to forward the call, be it PSTN or VoIP channels.  
By using

the hardware accelerator RR520P PCI express card, LCR decisions is now
down to microseconds without accessing external databases.

- "We've implemented this in our commercial VoIP network during  
development,
   and cut our costs by at least 75% and enhanced call quality.  
Billing and CDR
   mediation is much easier, since the RateRoute system always  
picked one
   outbound service provider that always matched the fifteen  
criteria for
   carrier selection" says Anders Runnstam at PulseVoip in Bergen,  
Norway.



* Asterisk SPITwall(R) - filtering away tomorrows VoIP spam today!
 
--


The SPITwall(R) technology is developed by Olle E. Johansson, a
member of the Asterisk developer team and Senior Technical Advisor
for Voop in Bergen, Norway - the Asterisk Dialtone provider.

- "I got more and more annoying calls during development, which  
disturbed

  me a lot and caused me to loose concentration. On the other hand,
  it inspired me to develop SPITwall to be able to filter them out.
  I have measured up to 95% success rate on call filtering,
  which is far beyond any similar products on the market. By not  
bothering

  with answering the final 5%, I could concentrate on development again
  and succesfully finish my development projects." says Olle.

The SPITwall is built on a shared database and use bayesian techniques
to analyze the content of the call. It requires Asterisk ChanSpy to  
be able
to listen in and warn the callee about ongoing unsolicited calls. The  
callee

can also press certain DTMF sequences during the call to mark the call
as SPIT. The voice pattern, SPITwall checksums and call properties will
then immediately be stored in the Digium SPITcore repository to be
available for all other users.

- "Using the community to build a SPIT-f

Re: [asterisk-users] Announcement: Asterisk Service Provider Edition v1.0 Beta

[Jaswinder Singh]

Wow i need a tftp client to download it now .
Nice April  1 joke  :P .
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[asterisk-users] Re: SIP/IAX peers UNREACHABLE and audio loss

[Edoardo Serra]

Hi guys,
I think I got the point of the problem.

I guess it's related to a lock in res_perl (which we use to do lcr, 
billing, ecc...)


I'll open another thread for that

Tnx for hep

Regards

Edoardo Serra
WeBRainstorm S.r.l.

Edoardo Serra ha scritto:

Hi all,
I'm having a problem with some Asterisk servers interconnected with 
each other using IAX (I also tried with SIP without solving the problem)


Sometimes, with apparently no reason, some peers become UNREACHABLE
(I have qualify=yes in iax.conf) and REACHABLE again as soon as
another qualify test is made.

Our users are also complaining about audio loss during their calls,
apparently randomly, everything goes ok for days and bad for another few 
days.


I strongly believe the 2 problems are strictly related because in the 
logs I see REACHABLE / UNREACHABLE messages only for certains days

without regularity.
The days in wich i see a lot of messages are exactly the days with
most of complaint about audio loss

I just noticed that timestamps of the logs (REACHABLE / UNREACHABLE)
are quite always during business hours, this makes me think at somewhat 
related to load (cpu load, badwidth load, calls load, etc...)


But, looking at hardware specs of our lan, servers and average load I 
don't think they are over-stressed.


Our servers are all:
2 x Intel(R) Xeon(TM) CPU 3.20GHz
1 GB RAM
2 x IDE HDDs Software RAID 1
Asterisk 1.2.13 with res_perl
Gentoo Linux
Some of them has a Sangoma card connected with an E1

Most ot these are on the same LAN, interconnected with a 1 GB switch
(I don't think it should be a bandwidth problem).

Load averages of these server is varying from 0.5 to 1.0
(I guess it should be ok)

On each server we don't have more than 50 concurrent calls
(bridged SIP <-> IAX2 or IAX2 <-> ZAP)

Used codec is mostly G729

Sometimes on asterisk cli i see some messages like
"Avoided initial deadlock for '0x9fd130', 10 retries!"
I don't know if it could be somehow related.

Someone of you can point me in the right direction ?

Tnx in advance

Regards

Ing. Edoardo Serra
WeBRainstorm S.r.l.

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[asterisk-users] Re: asterisk-users Digest, Vol 33, Issue 2

[fb]

Je suis absent du  2/04/2007 au 11/04/2007.

Je répondrai à votre message dès mon retour. Pour toute urgence, contacter
Emmanuelle Parache Moga ou Cédric Buzay.


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Re: [asterisk-users] Announcement: Asterisk Service Provider Editionv1.0 Beta

[Dovid B]

For immediate release, April 1st 2007
On behalf of the Asterisk development team and project 0401



Project 0401. You don't need to make it an abvious that its an April Fool's 
joke. 



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Re: [asterisk-users] Announcement: Asterisk Service Provider Edition v1.0 Beta

[Philipp Kempgen]

Olle E Johansson wrote:

> Asterisk SPE  

Nice. ;)
Was SpitShare developed by project 0401? Didn't read carefully.


Regards,
  Philipp

-- 
amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
 Let's use IT to solve problems and not to create new ones.
   Asterisk? -> http://www.das-asterisk-buch.de

Geschäftsführer: Stefan Wintermeyer
Handelsregister: Neuwied B 14998
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Re: [asterisk-users] Re: SIP/IAX peers UNREACHABLE and audio loss

[Lukas]

Hi guys. 

I have the same problem and donnow why. 
I set up Nat=yes and Qulify=yes ... and notthin' happens.

So..do you think it's related to a perl function?
If u find the sol , tell us please. THANKS.

El dom, 01-04-2007 a las 18:02 +0200, Edoardo Serra escribió:

> Hi guys,
>   I think I got the point of the problem.
> 
> I guess it's related to a lock in res_perl (which we use to do lcr, 
> billing, ecc...)
> 
> I'll open another thread for that
> 
> Tnx for hep
> 
> Regards
> 
> Edoardo Serra
> WeBRainstorm S.r.l.
> 
> Edoardo Serra ha scritto:
> > Hi all,
> > I'm having a problem with some Asterisk servers interconnected with 
> > each other using IAX (I also tried with SIP without solving the problem)
> > 
> > Sometimes, with apparently no reason, some peers become UNREACHABLE
> > (I have qualify=yes in iax.conf) and REACHABLE again as soon as
> > another qualify test is made.
> > 
> > Our users are also complaining about audio loss during their calls,
> > apparently randomly, everything goes ok for days and bad for another few 
> > days.
> > 
> > I strongly believe the 2 problems are strictly related because in the 
> > logs I see REACHABLE / UNREACHABLE messages only for certains days
> > without regularity.
> > The days in wich i see a lot of messages are exactly the days with
> > most of complaint about audio loss
> > 
> > I just noticed that timestamps of the logs (REACHABLE / UNREACHABLE)
> > are quite always during business hours, this makes me think at somewhat 
> > related to load (cpu load, badwidth load, calls load, etc...)
> > 
> > But, looking at hardware specs of our lan, servers and average load I 
> > don't think they are over-stressed.
> > 
> > Our servers are all:
> > 2 x Intel(R) Xeon(TM) CPU 3.20GHz
> > 1 GB RAM
> > 2 x IDE HDDs Software RAID 1
> > Asterisk 1.2.13 with res_perl
> > Gentoo Linux
> > Some of them has a Sangoma card connected with an E1
> > 
> > Most ot these are on the same LAN, interconnected with a 1 GB switch
> > (I don't think it should be a bandwidth problem).
> > 
> > Load averages of these server is varying from 0.5 to 1.0
> > (I guess it should be ok)
> > 
> > On each server we don't have more than 50 concurrent calls
> > (bridged SIP <-> IAX2 or IAX2 <-> ZAP)
> > 
> > Used codec is mostly G729
> > 
> > Sometimes on asterisk cli i see some messages like
> > "Avoided initial deadlock for '0x9fd130', 10 retries!"
> > I don't know if it could be somehow related.
> > 
> > Someone of you can point me in the right direction ?
> > 
> > Tnx in advance
> > 
> > Regards
> > 
> > Ing. Edoardo Serra
> > WeBRainstorm S.r.l.
> > 
> > ___
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> > 
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >   http://lists.digium.com/mailman/listinfo/asterisk-users
> > 
> 
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RE: [asterisk-users] Security on long distance calls

[Salvatore Giudice]

Using caller id to authenticate anyone is asking for a toll fraud problem.
4-digit pins really are not a good idea either.

 

Try putting your operators and your users in different contexts. If you have
specific numbers you don't want the users to be able to dial, then create
patterns for those numbers. Play a sound byte that says the call is not
allowed and then hang up.

 

 

 

--
Salvatore Giudice
[EMAIL PROTECTED]

VoIP Security Training, LLC
http://VoIPSecurityTraining.com

848 N. Rainbow Blvd. #1676
Las Vegas, NV 89107
Phone: (702) 979-2906
Fax: (212) 279-2906

 

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dovid B
Sent: Sunday, April 01, 2007 5:54 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Security on long distance calls

 

Or you can do

 

exten => _011.,1,Authenticate(1234)

exten => _011.,2,Dial(SIP/[EMAIL PROTECTED]
 )

 

Also this is a bit more complicated but you can do it by sip extension. If
CID of phone = phone that is allowed then let it go out. This will be
hard considering you will have to make a gotoif for every extension that you
want to allow to call intl.

- Original Message - 

From: Rizwan Hisham   

To: Asterisk Users Mailing List - 
Non-Commercial Discussion 

Sent: Friday, March 30, 2007 11:18 AM

Subject: Re: [asterisk-users] Security on long distance calls

 

For operator:

[longdistance]
include=> local
include=> International

for extensions:

[localcalls]
include=> local

now assign longdistance context to operator and localcalls context to every
other user for whom you want to restrict intl calls 

[local]
should include all local extension codes

[International]
should include all international extension codes

you get my point?

On 3/30/07, Stefano Corsi <[EMAIL PROTECTED]> wrote: 

Hello, 

which kind of method could you use to inhibit long distance calls to
_some_ extensions?
Is there a way to do it with freepbx or you have to do it manually in
the config files? I wouldn't like to set a route password, because 
that is not confortable for the pbx operator. I just would like the
operator being able to call whatever number, while the extensions
should only be able to make local calls.

Thanks
Stefano


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-- 
Regards
Rizwan Hisham
Software Engineer 

  _  

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RE: On Topic: Cheapest Asterisk USB Key? (was: Re: [asterisk-users] Off Topic: Open Source USB Softphone)

[Salvatore Giudice]

Try installing Monte Vista http://www.mvista.com/ on the usb stick. It will
be a lot cleaner than taking a standard server distribution of linux and
stripping out all the unwanted kernel modules.

Monte Vista is an embedded linux that should be able to boot your server off
a 128mb usb stick with Asterisk installed. You should probably strip
asterisk down to the bare essentials for your project as well.

You should be aware that flash memory is generally not the best medium to
store data when you have a high number of read/writes. Flash memory will
fail much more quickly under these conditions. You might want to consider
using a usb microdrive instead of a flash stick. Pick a microdrive that
generates as little heat as possible.

BTW, what exactly is the motivation for running linux off of a usb stick? If
you would like cdr's, you could likely do so with ngrep and a perl script.

Good luck, SG

--
Salvatore Giudice
[EMAIL PROTECTED]

VoIP Security Training, LLC
http://VoIPSecurityTraining.com

848 N. Rainbow Blvd. #1676
Las Vegas, NV 89107
Phone: (702) 979-2906
Fax: (212) 279-2906

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matthew
Rubenstein
Sent: Sunday, April 01, 2007 9:08 AM
To: Asterisk-Users
Subject: On Topic: Cheapest Asterisk USB Key? (was: Re: [asterisk-users] Off
Topic: Open Source USB Softphone)

Here's a flipside of this subject: what is the absolute cheapest
Linux
device that can be connected to a PC's USB port? That has just enough
power for a minimal Asterisk server running on it. The Asterisk just
maintains a CDR database on its Flash memory, which it periodically
submits over the PC's network connection with an HTTP hit on a remote
full-service Asterisk server? No call handling, DSP or anything really
number crunching, no telephony terminal or other services. The
lowest-performance device that plugs into the USB, with its own Linux
instance. In OEM quantity, under $50? Under $100?


On Sun, 2007-04-01 at 02:51 -0700,
[EMAIL PROTECTED] wrote:
> Date: Sat, 31 Mar 2007 16:02:06 -0500
> From: "Mike Lynchfield" <[EMAIL PROTECTED]>
> Subject: Re: [asterisk-users] Off Topic: Open Source USB Softphone
> To: [EMAIL PROTECTED],   "Asterisk Users Mailing List -
> Non-Commercial
> Discussion"  
> Message-ID:
> <[EMAIL PROTECTED]>
> Content-Type: text/plain; charset="iso-8859-1"
> 
> sip would be the required one as iax..well..
> 
> also openwengo wont work.. to much overhead .. broswrer needed.. ie
> component + flash + css+js etc.. not viable..
> 
> so im also asking anyone have one ? since ihave a supply of around
> 2000 of
> the vonage usb stick OEM..
> 
> On 3/30/07, Michael Van Donselaar <[EMAIL PROTECTED]>
> wrote:
> >
> > Which USB Phone?  I have written custom versions of iaxcomm for
> various
> > people,
> > and have a version that works with the Yealink phone.
> >
> > On Thu, 29 Mar 2007 11:33:07 -0300, "Luis Claudio Santos" <
> > [EMAIL PROTECTED]>
> > wrote:
> >
> > >I need a softphone - for usb phone devices - that I can alter
> (insert
> > logo,
> > >menu, etc).
> > >
> > >Does somebody know such one?
> > >
> > >[]s
> >
> > ___
> > --Bandwidth and Colocation provided by Easynews.com --
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> 
> 
> 
> -- 
> Mike
> Sales Manager
> http://www.voicemeup.com
> Making it happen
> 1.877.807.VOIP (8647)
> 1.514.312.7030 
-- 

(C) Matthew Rubenstein

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[asterisk-users] Asterisk 1.2 and res_perl - lock that leads to weird behaviour

[Edoardo Serra]

Hi guys,
	as I wrote in a previous thread I was experiencing dropped audio 
(apparently randomly) and SIP + IAX peers getting REACHABLE / 
UNREACHABLE without reason, servers were in the same LAN.


Investingating deeply in the problem I also noticed that 'show channels' 
command on the CLI, sometimes were returning strange results, for 
example it wasn0t showing some channels I was sure were active.


Looking at our DB's log, I also notited there were a race condition 
which could lock a query for a long time (up to 30 secs)

I don't want to annoy you wit DB issues explaining why it happened

My Asterisk dialplan, when a user try to place a call, make a query to 
our db (through res_perl) to check some parameters, among them the 
user's credit to set the maximum duration of a call.


If that perl script does not end in a reasonable time (I cannot tell how 
much is reasonable, but the 30secs due to the lock were surely too many)
and other users try to call (and also their queries get locked) Asterisk 
begins causing weird problems.
(I saw on res_perl documentation that it acquires some lock in asterisk 
during scripts execution but I didn't imagine that locks could affect 
the whole Asterisk box)


Common problems in these cases are peers qualified as UNREACHABLE, 
dropped audio (sometimes in both directions, sometimes in just one 
direction), channels missing in 'show channels', etc


I solved the race condition at db level and problem have magically 
disappeared but I'd like to go deep in the problem, I wouldn't like that 
o happen again because of a slow query or sloq execution of a perl 
script (it could happen for a lot of reasons)


Someone can help with that ?
Sorry for the crosspost but I think also asterisk-devel could be 
involved in it.


Tnn in advance for help

Regards

Edoardo Serra
WeBRainstorm S.r.l.

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[asterisk-users] Asterisk 1.2 and res_perl - lock that leads to weird behaviour

[Edoardo Serra]

Hi guys,
as I wrote in a previous thread I was experiencing dropped audio 
(apparently randomly) and SIP + IAX peers getting REACHABLE / 
UNREACHABLE without reason, servers were in the same LAN.


Investingating deeply in the problem I also noticed that 'show channels' 
command on the CLI, sometimes were returning strange results, for 
example it wasn0t showing some channels I was sure were active.


Looking at our DB's log, I also notited there were a race condition 
which could lock a query for a long time (up to 30 secs)

I don't want to annoy you wit DB issues explaining why it happened

My Asterisk dialplan, when a user try to place a call, make a query to 
our db (through res_perl) to check some parameters, among them the 
user's credit to set the maximum duration of a call.


If that perl script does not end in a reasonable time (I cannot tell how 
much is reasonable, but the 30secs due to the lock were surely too many)
and other users try to call (and also their queries get locked) Asterisk 
begins causing weird problems.
(I saw on res_perl documentation that it acquires some lock in asterisk 
during scripts execution but I didn't imagine that locks could affect 
the whole Asterisk box)


Common problems in these cases are peers qualified as UNREACHABLE, 
dropped audio (sometimes in both directions, sometimes in just one 
direction), channels missing in 'show channels', etc


I solved the race condition at db level and problem have magically 
disappeared but I'd like to go deep in the problem, I wouldn't like that 
o happen again because of a slow query or sloq execution of a perl 
script (it could happen for a lot of reasons)


Someone can help with that ?
Sorry for the crosspost but I think also asterisk-devel could be 
involved in it.


Tnn in advance for help

Regards

Edoardo Serra
WeBRainstorm S.r.l.


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RE: [asterisk-users] Weird extension behavior

[Yuan LIU]

From: Mark Hennessy <[EMAIL PROTECTED]>
Date: Sun, 01 Apr 2007 06:15:40 -0400

Hi, I'm using Asterisk with two Cisco 7960 phones using SIP.
I'm seeing the following weird behavior:
SIP Phome 1 is extension 4002
SIP Phone 2 is extension 4003

I call 4002 from 4003 and that works fine.
I call 4003 from 4002, and it rings locally to 4002, never gets to 4003.

I'm able to send a config query packet to 4003 from the asterisk  console 
and get a response, when I send one to 4002 there is no respone.


I know that both phones pull down their config via TFTP properly, I  look 
in the network settings and see that 4002 has been given an IP of  
x.y.z.201 and 4003 has been given an IP of x.y.z.202 and the asterisk  box 
is running on x.y.z.74.


I combed through all of the config files in both Asterisk's config and  the 
TFTP-downloaded configs for the phones looking for any possible  instance 
of 4003 being transposed for 4002 or vice versa and was not  able to find 
any.


What additional information is necessary to provide to trace down and  
resolve this issue?


Corresponding entries in sip.conf may help.

Yuan Liu

AFAICT, the server is using Asterisk 1.2.x and beyond the 7960 phones,  no 
other specialized hardware is in use.


--
Mark P. Hennessy



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[asterisk-users] RE: On Topic: Cheapest Asterisk USB Key?

[Matthew Rubenstein]

I need a USB microprocessor *device* on which the Linux and Asterisk
will run (even if very slowly), not just a storage drive from which to
run it on the PC. MonteVista is a good distro, though there are other
"minimal" embedded distros, of which I've already got one selected. The
CDR usage of a single user's PC is just fine in performance and total
lifetime read/writes (usually upwards of 100K) for the CDR data that
needs to persist, as opposed to the device's RAM for executing the
Asterisk. I'm looking for a device under $100 or $50 in OEM quantity,
which is where just microdrives start. I want to run Asterisk itself,
even if stripped down, for easy sync and single platform maintenance
across all the Asterisk instances I've got, as well as guaranteed
compatibility between data/network formats/protocols.


On Sun, 2007-04-01 at 13:08 -0400, Salvatore Giudice wrote:
> Try installing Monte Vista http://www.mvista.com/ on the usb stick. It will
> be a lot cleaner than taking a standard server distribution of linux and
> stripping out all the unwanted kernel modules.
> 
> Monte Vista is an embedded linux that should be able to boot your server off
> a 128mb usb stick with Asterisk installed. You should probably strip
> asterisk down to the bare essentials for your project as well.
> 
> You should be aware that flash memory is generally not the best medium to
> store data when you have a high number of read/writes. Flash memory will
> fail much more quickly under these conditions. You might want to conside
> using a usb microdrive instead of a flash stick. Pick a microdrive that
> generates as little heat as possible.
> 
> BTW, what exactly is the motivation for running linux off of a usb stick? If
> you would like cdr's, you could likely do so with ngrep and a perl script.
> 
> Good luck, SG
> 
> --
> Salvatore Giudice
> [EMAIL PROTECTED]
> 
> VoIP Security Training, LLC
> http://VoIPSecurityTraining.com
> 
> 848 N. Rainbow Blvd. #1676
> Las Vegas, NV 89107
> Phone: (702) 979-2906
> Fax: (212) 279-2906
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Matthew
> Rubenstein
> Sent: Sunday, April 01, 2007 9:08 AM
> To: Asterisk-Users
> Subject: On Topic: Cheapest Asterisk USB Key? (was: Re: [asterisk-users] Off
> Topic: Open Source USB Softphone)
> 
>   Here's a flipside of this subject: what is the absolute cheapest
> Linux
> device that can be connected to a PC's USB port? That has just enough
> power for a minimal Asterisk server running on it. The Asterisk just
> maintains a CDR database on its Flash memory, which it periodically
> submits over the PC's network connection with an HTTP hit on a remote
> full-service Asterisk server? No call handling, DSP or anything really
> number crunching, no telephony terminal or other services. The
> lowest-performance device that plugs into the USB, with its own Linux
> instance. In OEM quantity, under $50? Under $100?
> 
> 
> On Sun, 2007-04-01 at 02:51 -0700,
> [EMAIL PROTECTED] wrote:
> > Date: Sat, 31 Mar 2007 16:02:06 -0500
> > From: "Mike Lynchfield" <[EMAIL PROTECTED]>
> > Subject: Re: [asterisk-users] Off Topic: Open Source USB Softphone
> > To: [EMAIL PROTECTED],   "Asterisk Users Mailing List -
> > Non-Commercial
> > Discussion"  
> > Message-ID:
> > <[EMAIL PROTECTED]>
> > Content-Type: text/plain; charset="iso-8859-1"
> > 
> > sip would be the required one as iax..well..
> > 
> > also openwengo wont work.. to much overhead .. broswrer needed.. ie
> > component + flash + css+js etc.. not viable..
> > 
> > so im also asking anyone have one ? since ihave a supply of around
> > 2000 of
> > the vonage usb stick OEM..
> > 
> > On 3/30/07, Michael Van Donselaar <[EMAIL PROTECTED]>
> > wrote:
> > >
> > > Which USB Phone?  I have written custom versions of iaxcomm for
> > various
> > > people,
> > > and have a version that works with the Yealink phone.
> > >
> > > On Thu, 29 Mar 2007 11:33:07 -0300, "Luis Claudio Santos" <
> > > [EMAIL PROTECTED]>
> > > wrote:
> > >
> > > >I need a softphone - for usb phone devices - that I can alter
> > (insert
> > > logo,
> > > >menu, etc).
> > > >
> > > >Does somebody know such one?
> > > >
> > > >[]s
> > >
> > > ___
> > > --Bandwidth and Colocation provided by Easynews.com --
> > >
> > > asterisk-users mailing list
> > > To UNSUBSCRIBE or update options visit:
> > >http://lists.digium.com/mailman/listinfo/asterisk-users
> > >
> > 
> > 
> > 
> > -- 
> > Mike
> > Sales Manager
> > http://www.voicemeup.com
> > Making it happen
> > 1.877.807.VOIP (8647)
> > 1.514.312.7030 
-- 

(C) Matthew Rubenstein

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RE: On Topic: Cheapest Asterisk USB Key? (was: Re: [asterisk-users]Off Topic: Open Source USB Softphone)

[Stelios Koroneos]

>
>   Here's a flipside of this subject: what is the absolute
> cheapest Linux
> device that can be connected to a PC's USB port? That has just enough
> power for a minimal Asterisk server running on it. The Asterisk just
> maintains a CDR database on its Flash memory, which it periodically
> submits over the PC's network connection with an HTTP hit on a remote
> full-service Asterisk server? No call handling, DSP or anything really
> number crunching, no telephony terminal or other services. The
> lowest-performance device that plugs into the USB, with its own Linux
> instance. In OEM quantity, under $50? Under $100?
>

When you say "devices" do you mean an off the self device or a module you
can use to build a custom device ?
In the first case there are a lot of "fisrt generation" routers coming into
the market at very low prices
for example
http://www.wirelesslan.gr/product_info.php?cPath=127&products_id=866
http://www.wirelesslan.gr/product_info.php?products_id=670

If you are looking for a SoC type device there are several although, the
100$ range looks more realistic

There are several devices that could be used.
DimmPC comes in my mind - http://www.amctechcorp.com/dimmpci/index.html
Digi's Connectcore
http://www.digi.com/products/embeddedsolutions/connectcore9u.jsp

Check Linux devices for a larger list
http://www.linuxdevices.com/articles/AT8498487406.html


Hope it helps

Stelios



Stelios S. Koroneos
Digital OPSiS - Embedded Inteligence
http://www.digital-opsis.com



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[asterisk-users] Add/remove international prefix

[Paolo Prandini]

I have 2 asterisk servers located in different countries.
When there is an outside call incoming in either country
the phones ring in both; but the problem is understanding
where the call origin is! The incoming number has always
no international prefix if it is from the some country
where the called number is located, and so it is impossibile
to see if the caller is in the first or in the second country,
while if it is an international call from a third country
there is no problem at all of course.
What I would like is to receive caller identification without
international prefix as usual if the phone that is ringing
is in the same country as the calling one ( e.g. a call from
Germany to our office in Germany ) while adding the international
prefix if in the other country ( e.g. the same call from Germany
but ringing in Italy, received by the second server via IAX2).
Can it be done?
Thanks to anyone for your help.
Paolo
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[asterisk-users] Re: asterisk-users Digest, Vol 33, Issue 3

[fb]

Je suis absent du  2/04/2007 au 11/04/2007.

Je répondrai à votre message dès mon retour. Pour toute urgence, contacter
Emmanuelle Parache Moga ou Cédric Buzay.


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Re: [asterisk-users] Speed Dial Application in *

[Gordon Henderson]

On Sun, 1 Apr 2007, Dovid B wrote:


You can try putting out a bounty to make a new app in asterisk or set a
variable in the asterisk db.


Why a new app? That way you get "forced" into doing it the app way, (like 
FollowMe in 1.4 which, if I understand the wiki page on it, doesn't do 
anything remotely like what I want out of a FollowMe, which I've written 
for my own system entirly in dialplan) or write your own - and if you're 
writing your own anyway then ...


And as there appears to be sounds-files in the sound add-ons package for 
speed-dialling, I'm guessing that more than one person has done this in 
the past ...


So here's a quick & dirty personal (per extension) speed-dial system, 
independant of what phone you are using (which I'm not claiming any 
originality for as I got the idea from somewhere else, just altered it for 
my own system) It's trivial to alter for a company-wide system too - (I 
use *1x and *2x to implement 20 system-wide speed-dials) slightly harder 
to put a web front-end on it (but not impossible as I'm in the process of 
testing that for me), and with that, then you can use incoming caller-id 
to associate a name to a number, if they are in the speed-dial list (just 
like most phones do with their own personal phone-book)


;==
; Personal speed-dial:
;   Let the extension user assign a personal speed dial code
;   From *00 through *09
;==

;Setup: *0 then a single digit 0-9 then the phone number to store at that 
location

exten => _*0X.,1,Answer()
exten => _*0X.,n,Set(me=${CALLERID(num)})
exten => _*0X.,n,Set(pos=${EXTEN:2:1})
exten => _*0X.,n,Set(num=${EXTEN:3})
exten => _*0X.,n,Set(DB(${me}/sd${pos})=${num})
exten => _*0X.,n,SayDigits(${num})
exten => _*0X.,n,Playback(at)
exten => _*0X.,n,Playback(position)
exten => _*0X.,n,SendText(*0${pos}=${num})
exten => _*0X.,n,SayDigits(${pos})
exten => _*0X.,n,Macro(starAck)

;List:  *0X*

exten => _*0X*,1,Answer()
exten => _*0X*,n,Set(me=${CALLERID(num)})
exten => _*0X*,n,Set(pos=${EXTEN:2:1})
exten => _*0X*,n,Set(num=${DB(${me}/sd${pos})})
exten => _*0X*,n,GotoIf(${num}?:noNum)
exten => _*0X*,n,SendText(*0${pos}=${num})
exten => _*0X*,n,SayDigits(${num})
exten => _*0X*,n,Playback(at)
exten => _*0X*,n,Playback(position)
exten => _*0X*,n,SayDigits(0${pos})
exten => _*0X*,n,Hangup()
exten => _*0X*,n(noNum),Playback(that-number)
exten => _*0X*,n,Playback(is-not-in-the)
exten => _*0X*,n,Playback(speed-dial)
exten => _*0X*,n,Playback(system)
exten => _*0X*,n,Hangup()

;Recall and dial:   *0X

exten => _*0X,1,Answer()
exten => _*0X,n,Set(me=${CALLERID(num)})
exten => _*0X,n,Set(pos=${EXTEN:2:1})
exten => _*0X,n,Set(num=${DB(${me}/sd${pos})})
exten => _*0X,n,GotoIf(${num}?:noNum)
exten => _*0X,n,SendText(Calling: ${num})
exten => _*0X,n,Goto(${num},1)
exten => _*0X,n(noNum),Playback(that-number)
exten => _*0X,n,Playback(is-not-in-the)
exten => _*0X,n,Playback(speed-dial)
exten => _*0X,n,Playback(system)
exten => _*0X,n,Hangup()



And in Dial Plan
exten => _77.,1,Dial(SIP/${DB(speeddial/${EXTEN:2})[EMAIL PROTECTED]) ; Dials 
the speed dial


If you just "Goto" the speed-dial, then, assuming you have dial-out in the 
same context it will dial then number for you - which would be an internal 
extension, or be dialled out by whatever out-dialling rules you have - eg. 
least cost routing, etc.


However with the Goto system, it's possible to really screw up your system 
with a cleverly dialled speed-dial, but I'll leave that as an excercise to 
the reader ;-)


Enjoy!
Gordon
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[asterisk-users] [MACRO-SCREEN] and MACRO_RESULT

[Andy Hester]

I am following the example at 
http://www.voip-info.org/wiki/view/Asterisk+tips+findme but I find that no 
matter what, the call is connected.  Can anyone confirm that config is working 
for them?  Any suggestions appreciated.

I need to transfer calls to a list of cell phones, ring all of them, allow them 
to screen the call, connect the call to the first number that accepts the call, 
and allow others to reject the call.

Thanks,
Andy


[macro-screen]
exten => s,1,Wait(1)
exten => s,n,Background(csp_ackshort-male)
exten => s,n,Set(TIMEOUT(response=10))
exten => 1,1,NoOp(Call Accepted)
exten => 2,1,Set(MACRO_RESULT=CONTINUE)
exten => t,1,Set(MACRO_RESULT=CONTINUE)

;original macro
;exten => 1,1,NoOp(Caller accepted) ; Do not set MACRO_RESULT to anything to 
connect the caller
;exten => i,1,Set(MACRO_RESULT=CONTINUE)
;exten => t,1,Set(MACRO_RESULT=CONTINUE)

[default]
include => architel
include => local
include => trunkintl
exten => _6XX,1,Dial(ZAP/G1/${EXTEN:1},40,M(screen))
exten => _6XX,2,Hangup


[findme]
exten => s,1,Playback(transfer_csp-male)
exten => s,n,Dial(LOCAL/6${findme1}&LOCAL/6${findme2},40,m)
exten => s,n,Hangup()

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Re: [asterisk-users] [MACRO-SCREEN] and MACRO_RESULT

[Philipp Kempgen]

Andy Hester wrote:

> exten => s,n,Set(TIMEOUT(response=10))

Should be
exten => s,n,Set(TIMEOUT(response)=10)


Regards,
  Philipp

-- 
amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
 Let's use IT to solve problems and not to create new ones.
   Asterisk? -> http://www.das-asterisk-buch.de

Geschäftsführer: Stefan Wintermeyer
Handelsregister: Neuwied B 14998
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RE: [asterisk-users] [MACRO-SCREEN] and MACRO_RESULT

[Andy Hester]

-Original Message-
From: [EMAIL PROTECTED] on behalf of Philipp Kempgen
Sent: Sun 4/1/2007 4:02 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] [MACRO-SCREEN] and MACRO_RESULT
 
Andy Hester wrote:

> exten => s,n,Set(TIMEOUT(response=10))

Should be
exten => s,n,Set(TIMEOUT(response)=10)


Regards,
  Philipp
--

Thanks Philipp,

I fixed this, but I still have the problem that the call is connected 
regardless...

-Andy

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[asterisk-users] RE: On Topic: Cheapest Asterisk USB Key?

[Salvatore Giudice]

That's quite interesting. You can get the microdrives cheaper than $50. We
recently purchased 2 GB microdrives for $17.

Try contacting this company:

IPMedia Asia Co. Ltd 
PO Box 2074
Northbrook
Illinois
United States 60065
Tel:  (886 2) 85227000  Ext : 107  (1 847) 6565759 
Fax:  (886 2) 66021000 /  (1 847) 5560164 

IPMedia Asia Co. Ltd
10F-3, No. 107
Jhongshan Road, Sec. 1
Sinjhuang City
Taipei
Taiwan 24250
Tel:  (886 2) 85227000  Ext : 107 
Fax:  (886 2) 66021000

One of our partner firms in Japan purchases USB sticks from them for
promotional distribution at security conferences. They also have a line of
Microdrives, I think you will find quite affordable.


--
Salvatore Giudice
[EMAIL PROTECTED]

VoIP Security Training, LLC
http://VoIPSecurityTraining.com

848 N. Rainbow Blvd. #1676
Las Vegas, NV 89107
Phone: (702) 979-2906
Fax: (212) 279-2906


-Original Message-
From: Matthew Rubenstein [mailto:[EMAIL PROTECTED] 
Sent: Sunday, April 01, 2007 1:47 PM
To: Salvatore Giudice
Cc: Asterisk-Users
Subject: RE: On Topic: Cheapest Asterisk USB Key?

I need a USB microprocessor *device* on which the Linux and Asterisk
will run (even if very slowly), not just a storage drive from which to
run it on the PC. MonteVista is a good distro, though there are other
"minimal" embedded distros, of which I've already got one selected. The
CDR usage of a single user's PC is just fine in performance and total
lifetime read/writes (usually upwards of 100K) for the CDR data that
needs to persist, as opposed to the device's RAM for executing the
Asterisk. I'm looking for a device under $100 or $50 in OEM quantity,
which is where just microdrives start. I want to run Asterisk itself,
even if stripped down, for easy sync and single platform maintenance
across all the Asterisk instances I've got, as well as guaranteed
compatibility between data/network formats/protocols.


On Sun, 2007-04-01 at 13:08 -0400, Salvatore Giudice wrote:
> Try installing Monte Vista http://www.mvista.com/ on the usb stick. It
will
> be a lot cleaner than taking a standard server distribution of linux and
> stripping out all the unwanted kernel modules.
> 
> Monte Vista is an embedded linux that should be able to boot your server
off
> a 128mb usb stick with Asterisk installed. You should probably strip
> asterisk down to the bare essentials for your project as well.
> 
> You should be aware that flash memory is generally not the best medium to
> store data when you have a high number of read/writes. Flash memory will
> fail much more quickly under these conditions. You might want to conside
> using a usb microdrive instead of a flash stick. Pick a microdrive that
> generates as little heat as possible.
> 
> BTW, what exactly is the motivation for running linux off of a usb stick?
If
> you would like cdr's, you could likely do so with ngrep and a perl script.
> 
> Good luck, SG
> 
> --
> Salvatore Giudice
> [EMAIL PROTECTED]
> 
> VoIP Security Training, LLC
> http://VoIPSecurityTraining.com
> 
> 848 N. Rainbow Blvd. #1676
> Las Vegas, NV 89107
> Phone: (702) 979-2906
> Fax: (212) 279-2906
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Matthew
> Rubenstein
> Sent: Sunday, April 01, 2007 9:08 AM
> To: Asterisk-Users
> Subject: On Topic: Cheapest Asterisk USB Key? (was: Re: [asterisk-users]
Off
> Topic: Open Source USB Softphone)
> 
>   Here's a flipside of this subject: what is the absolute cheapest
> Linux
> device that can be connected to a PC's USB port? That has just enough
> power for a minimal Asterisk server running on it. The Asterisk just
> maintains a CDR database on its Flash memory, which it periodically
> submits over the PC's network connection with an HTTP hit on a remote
> full-service Asterisk server? No call handling, DSP or anything really
> number crunching, no telephony terminal or other services. The
> lowest-performance device that plugs into the USB, with its own Linux
> instance. In OEM quantity, under $50? Under $100?
> 
> 
> On Sun, 2007-04-01 at 02:51 -0700,
> [EMAIL PROTECTED] wrote:
> > Date: Sat, 31 Mar 2007 16:02:06 -0500
> > From: "Mike Lynchfield" <[EMAIL PROTECTED]>
> > Subject: Re: [asterisk-users] Off Topic: Open Source USB Softphone
> > To: [EMAIL PROTECTED],   "Asterisk Users Mailing List -
> > Non-Commercial
> > Discussion"  
> > Message-ID:
> > <[EMAIL PROTECTED]>
> > Content-Type: text/plain; charset="iso-8859-1"
> > 
> > sip would be the required one as iax..well..
> > 
> > also openwengo wont work.. to much overhead .. broswrer needed.. ie
> > component + flash + css+js etc.. not viable..
> > 
> > so im also asking anyone have one ? since ihave a supply of around
> > 2000 of
> > the vonage usb stick OEM..
> > 
> > On 3/30/07, Michael Van Donselaar <[EMAIL PROTECTED]>
> > wrote:
> > >
> > > Which USB Phone?  I have written cus

Re: [asterisk-users] Re: Sponsored development - Monodirectional audio handling

[Andrew Joakimsen]

And also you should post it on the voip-info wiki there is a page just
with bounties.

On 3/31/07, Edoardo Serra <[EMAIL PROTECTED]> wrote:

Salvatore Giudice ha scritto:
> You could put a bounty on this. You may find someone who will be willing to
> write this for money.

My Bounty for that feature is 500 USD

>
> --
> Salvatore Giudice
> [EMAIL PROTECTED]
>
> VoIP Security Training, LLC
> http://VoIPSecurityTraining.com
>
> 848 N. Rainbow Blvd. #1676
> Las Vegas, NV 89107
> Phone: (702) 979-2906
> Fax: (212) 279-2906
>
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Edoardo Serra
> Sent: Saturday, March 31, 2007 11:42 AM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] Sponsored development - Monodirectional audio
> handling
>
> Hi Guys,
>   we're needing a special implementation on Asterisk
> Our intention is to contribute the development and share back the code
> to Asterisk community
>
> Here is what we need:
>
> - An option to Asterisk Dial command which, if used, when calls is
> answered gives monodirectional audio
> (Caller should hear the called party but not vice-versa)
>
> - A DTMF sequence (maybe handled in features.conf) for the Caller to
> start to have bidirectional audio
>
> - When the Callers makes the audio 'bidirectional' an Event should be
> generated so that we can see it from the manager API
>
> The purpose of thisi implementation is to deal with some carriers that
> give us the call as ANSWERED when the called party is still ringing.
> Our billing software is billing the user (and the carrier is billing us)
> even with unsuccessful calls.
>
> This way we can start billing when the user press the DTMF sequence to
> unlock audio (even if carriers bill us wrongly)
>
> Someone wants to help ??
>
> Regards
>
> Edoardo Serra
> WeBRainstorm S.r.l.
>
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Re: [asterisk-users] [MACRO-SCREEN] and MACRO_RESULT

[Andrew Joakimsen]

The logic of the macro is totally opposite of what it should be. I do
recall sending a corrected version of the script to someone a while
back, it might be on the mailing lists archive.

However, there is an option for the Dial() command to do exactly what you wish

p: This option enables screening mode. This is basically Privacy mode
without memory of how to handle the caller. It looks for the file
sounds/priv-callerintros/${IF($[ "${CALLERID(num)}" != ""
]?${CALLERID(num)}:NOCALLERID_${EXTEN}${CUT(CHANNEL,/,1)}=${CUT(CHANNEL,/,2)})}.gsm
and if it is not found, prompts the caller to say his name. It then
rings the called party and plays sounds/priv-callpending,
sounds/priv-callerintros/, and
sounds/screen-callee-options. If the called party enters 1, the call
is accepted, 2, the DIAL command exits with ${DIALSTATUS} set to
NOANSWER, 3, set to TORTURE and 4, set to DONTCALL. If no valid entry
is made, the DIAL command exits with ${DIALSTATUS} set to ANSWER. The
check for pre-existence of the name recording may not be what you
want. For example, everyone from the same number is not necessarily
the same person, especially if the number is OUTOFAREA, but if the
file is there, no new name will be recorded. Since the files are never
removed, you may wish to remove them with a System(rm
/var/lib/asterisk/sounds/priv-callerintros/${IF($[ "${CALLERID(num)}"
!= "" 
]?${CALLERID(num)}:NOCALLERID_${EXTEN}${CUT(CHANNEL,/,1)}=${CUT(CHANNEL,/,2)})}.*)
right before the Dial command and clean up old ones with a cron job.

On 4/1/07, Andy Hester <[EMAIL PROTECTED]> wrote:

-Original Message-
From: [EMAIL PROTECTED] on behalf of Philipp Kempgen
Sent: Sun 4/1/2007 4:02 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] [MACRO-SCREEN] and MACRO_RESULT

Andy Hester wrote:

> exten => s,n,Set(TIMEOUT(response=10))

Should be
exten => s,n,Set(TIMEOUT(response)=10)


Regards,
 Philipp
--

Thanks Philipp,

I fixed this, but I still have the problem that the call is connected 
regardless...

-Andy


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Re: [asterisk-users] Sponsored development - Monodirectional audio handling

[Philipp Kempgen]

Edoardo Serra wrote:

> The purpose of thisi implementation is to deal with some carriers that 
> give us the call as ANSWERED when the called party is still ringing.
> Our billing software is billing the user (and the carrier is billing us) 
> even with unsuccessful calls.

How about you try a different carrier or send your lawyer?


Regards,
  Philipp

-- 
amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
 Let's use IT to solve problems and not to create new ones.
   Asterisk? -> http://www.das-asterisk-buch.de

Geschäftsführer: Stefan Wintermeyer
Handelsregister: Neuwied B 14998
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[asterisk-users] Problems with TE110P

[Klaverstyn, David C]

I have a new server using Zaptel 1.2.16

 

Issuing a ztcfg gives the following error:

ZT_CHANCONFIG failed on channel 1: No such device or address (6)

 

Issuing the command

lsmod | grep -i wct

 

results in:

wcte11xp   26016  0

wct4xxp   221120  0

zaptel184996  3 ztdummy,wcte11xp,wct4xxp

 

 

In my zaptel Makefile I have:

MODULES:=zaptel wcte11xp

 

Please help. 

 

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Re: [asterisk-users] Dialplan Streaming

[Steve Totaro]

Madplay

Doug Garstang wrote:

Oh poo. No one seems to know. :(

Doug Garstang wrote:

All,

Is there a dial plan command that can stream uncompressed audio from 
another source? I see there's an MP3Player command that can stream, 
but I assume that plays MP3's, which means it has to decode them. I'm 
looking for something that could play .wav or .ulaw (g711) streams.


Doug.

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Re: [asterisk-users] Call dies when I press *

[Steve Totaro]

Is this while using queues?

Mike Diehl wrote:

Hi all,

I've trying to fix a problem.  If I'm in a call and I press the * key, the 
call goes silent but doesn't hang up.  I need to be able to send the * key 
for various IVR's that I interact with.


Since I thought this was related to the features.conf file, you can view it 
at:  http://www.diehlnet.com/features.conf


Any ideas are welcome.

TIA,
  


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Re: [asterisk-users] Polycom and Asterisk

[Steve Totaro]

Flawless as far as I know.

Bruce Reeves wrote:

Matt,

I am running Polycom 2.1 on both 1.4 and 1.2 svn releases without any 
problems. What kind of issues did you experience?


On 3/28/07, *Mike Hammett * <[EMAIL PROTECTED] 
> wrote:


I was previously having an issue with a Polycom phone and Polycom
support said that Asterisk didn't play well with Polycom firmware
versions 1.6.7 and newer due to SIP compatibility issues. I
believe I heard a lot of things were fixed\adjusted in 1.4 and was
wondering if anyone has had success with Asterisk 1.4 and the
latest Polycom firmware releases.


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--
Bruce Reeves
Nortex Networks


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RE: [asterisk-users] Problems with TE110P

[Klaverstyn, David C]

I forgot to mention that my /etc/zaptel.conf file contains:

 

loadzone=au

defaultzone=au

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

bchan=1-10

unused=11-15,17-31

dchan=16

 

 

 

 

I have a new server using Zaptel 1.2.16

 

Issuing a ztcfg gives the following error:

ZT_CHANCONFIG failed on channel 1: No such device or address (6)

 

Issuing the command

lsmod | grep -i wct

 

results in:

wcte11xp   26016  0

wct4xxp   221120  0

zaptel184996  3 ztdummy,wcte11xp,wct4xxp

 

 

In my zaptel Makefile I have:

MODULES:=zaptel wcte11xp

 

Please help. 

 

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Re: [asterisk-users] Problems with TE110P

[C F]

Looks like a UDEV problem

On 4/1/07, Klaverstyn, David C <[EMAIL PROTECTED]> wrote:

I have a new server using Zaptel 1.2.16



Issuing a ztcfg gives the following error:

ZT_CHANCONFIG failed on channel 1: No such device or address (6)



Issuing the command

lsmod | grep -i wct



results in:

wcte11xp   26016  0

wct4xxp   221120  0

zaptel184996  3 ztdummy,wcte11xp,wct4xxp





In my zaptel Makefile I have:

MODULES:=zaptel wcte11xp



Please help.





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[asterisk-users] Trigger and Email in Dial Plan

[Robert DeVries]

I have a friend traveling overseas.  I want to allow him to call a number
which will give him a busy signal (so no charge), but will then send me an
email that he has called.

I know how to use a call file to trigger a call (I created a callback system
for myself when I traveled overseas a few months ago), but I don't know how
to trigger an email.  All I want is a simple message like "Smitty called."

Any ideas?
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Re: [asterisk-users] Asterisk hangs up SIP call after 6 200 retransmits

[kjcsb]

>One potential reason could be that the ACK request being sent to Asterisk is 
>malformed. Notice >"branch=0" in the top Via. This should start with "z9hG4bK" 
>magic cookie since the INVITE was an RFC >3261 transaction. 

>While "branch=0" is valid in RFC 2543, I don't think an INVITE can start-off 
>as RFC 3261 and then the >ACK can switch over to RFC 2543 in the middle of the 
>transaction. Clearly, Asterisk is dropping this ACK >on the floor. 

OK. But in the calls that don't get dropped, the "branch=0" is present also. 
See below for an example:

<-- SIP read from 147.202.nnn.nnn:5060: 
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Record-Route: 
Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=z9hG4bK22ab.697375a4.0
Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK4cf2bb78;rport=5060
From: "649444" ;tag=as1370b1ab
To: 
Contact: 
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 69
Date: Mon, 02 Apr 2007 03:37:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 338
v=0
o=root 11402 11402 IN IP4 202.180.nnn.nnn
s=session
c=IN IP4 202.180.nnn.nnn
t=0 0
m=audio 39686 RTP/AVP 18 97 3 0 8 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:97 iLBC/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
--- (15 headers 15 lines) ---
Using INVITE request as basis request - [EMAIL PROTECTED]
Sending to 147.202.nnn.nnn : 5060 (non-NAT)
Found peer 'DLS'
Found RTP audio format 18
Found RTP audio format 97
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 202.180.nnn.nnn:39686
Found description format G729
Found description format iLBC
Found description format GSM
Found description format PCMU
Found description format PCMA
Found description format telephone-event
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x50e 
(gsm|ulaw|alaw|g729|ilbc)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 
(telephone-event), combined - 0x1 (telephone-event)
Looking for 649977 in from-trunk (domain 203.89.nnn.nnn)
list_route: hop: 
Transmitting (no NAT) to 147.202.nnn.nnn:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 
147.202.nnn.nnn;branch=z9hG4bK22ab.697375a4.0;received=147.202.nnn.nnn
Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK4cf2bb78;rport=5060
From: "649444" ;tag=as1370b1ab
To: 
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: 
Content-Length: 0

---
-- Goto (ivr-3,s,1)
-- Executing Set("SIP/649977-b7908550", "LOOPCOUNT=0") in new stack
-- Executing Set("SIP/649977-b7908550", "__DIR-CONTEXT=11000111000") in 
new stack
-- Executing Answer("SIP/649977-b7908550", "") in new stack
We're at 203.89.nnn.nnn port 15804
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 147.202.nnn.nnn:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 
147.202.nnn.nnn;branch=z9hG4bK22ab.697375a4.0;received=147.202.nnn.nnn
Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK4cf2bb78;rport=5060
Record-Route: 
From: "649444" ;tag=as1370b1ab
To: ;tag=as7ecf44d1
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: 
Content-Type: application/sdp
Content-Length: 244
v=0
o=root 16300 16300 IN IP4 203.89.nnn.nnn
s=session
c=IN IP4 203.89.nnn.nnn
t=0 0
m=audio 15804 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
-- Executing Wait("SIP/649977-b7908550", "1") in new stack
capetown*CLI> 
<-- SIP read from 147.202.nnn.nnn:5060: 
ACK sip:[EMAIL PROTECTED] SIP/2.0
Record-Route: 
Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=0
Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK5ba4f251;rport=5060
From: "649444" ;tag=as1370b1ab
To: ;tag=as7ecf44d1
Contact: 
Call-ID: [EMAIL PROTECTED]
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 69
Content-Length: 0

--- (12 headers 0 lines) ---
-- Executing Set("SIP/649977-b7908550", "TIMEOUT(digit)=3") in new stack
-- Digit timeout set to 3
-- Executing Set("SIP/649977-b7908550", "TIMEOUT(response)=10") in new 
stack
-- Response timeout set to 10
-- Executing BackGround("SIP/649977-b7908550", 
"custom/11000111000-welcome") in new stack
-- Playing 'custom/11000111000-welcome' (language 'nz')
capetown*CLI> 
<-- SIP read from 147.202.nnn.nnn:5060: 
BYE sip:[EMAIL PROTECTED] SIP/2.0
Record-Route: 
Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=z9hG4bK32ab.feee6b67.0
Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK7916f637;rport=5060
From: "649444" ;tag=as1370b1ab
To: ;tag=as

Re: [asterisk-users] Trigger and Email in Dial Plan

[Tzafrir Cohen]

On Sun, Apr 01, 2007 at 08:35:23PM -0700, Robert DeVries wrote:
> I have a friend traveling overseas.  I want to allow him to call a number
> which will give him a busy signal (so no charge), but will then send me an
> email that he has called.
> 
> I know how to use a call file to trigger a call (I created a callback system
> for myself when I traveled overseas a few months ago), but I don't know how
> to trigger an email.  All I want is a simple message like "Smitty called."

Running something from inside Asterisk:

  System(mail -s "Smitty called" your_alias)

(stdi is redirected from /dev/null, so this will not hang, right).

But it is even simpler from the script that generates the call file.

-- 
   Tzafrir Cohen   
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RE: [asterisk-users] Trigger and Email in Dial Plan

[Alexander Lopez]

Put this in the incoming context for that number called.

 

Exten => s,1,Wait(1)

Exten => s.2.System("mail -s 'Smitty called from ${CALLERID(all)'
[EMAIL PROTECTED]")

Exten => s,3,Congestion

 

 



From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Robert
DeVries
Sent: Sunday, April 01, 2007 11:35 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Trigger and Email in Dial Plan

 

I have a friend traveling overseas.  I want to allow him to call a
number which will give him a busy signal (so no charge), but will then
send me an email that he has called.

I know how to use a call file to trigger a call (I created a callback
system for myself when I traveled overseas a few months ago), but I
don't know how to trigger an email.  All I want is a simple message like
"Smitty called." 

Any ideas?

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Re: [asterisk-users] Problems with TE110P

[Tzafrir Cohen]

On Sun, Apr 01, 2007 at 11:34:20PM -0400, C F wrote:
> 
> On 4/1/07, Klaverstyn, David C <[EMAIL PROTECTED]> wrote:
> >I have a new server using Zaptel 1.2.16
> >
> >
> >
> >Issuing a ztcfg gives the following error:
> >
> >ZT_CHANCONFIG failed on channel 1: No such device or address (6)
>
> Looks like a UDEV problem

No, it is not. If it were a uudev problem (that is: missing device file)
you would get the error "no such file or directory" from ztcfg.

-- 
   Tzafrir Cohen   
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Re: [asterisk-users] Problems with TE110P

[Tzafrir Cohen]

On Mon, Apr 02, 2007 at 12:50:18PM +1000, Klaverstyn, David C wrote:
> I have a new server using Zaptel 1.2.16
> 
> Issuing a ztcfg gives the following error:
> 
> ZT_CHANCONFIG failed on channel 1: No such device or address (6)
>  
> 
> loadzone=au
> 
> defaultzone=au
> 
> span=1,1,0,ccs,hdb3,crc4
> 
> bchan=1-10
> 
> unused=11-15,17-31
> 
> dchan=16
> 

Now what do you actually have loaded?

cat /proc/zaptel/*

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[asterisk-users] Re: asterisk-users Digest, Vol 33, Issue 4

[fb]

Je suis absent du  2/04/2007 au 11/04/2007.

Je répondrai à votre message dès mon retour. Pour toute urgence, contacter
Emmanuelle Parache Moga ou Cédric Buzay.


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Re: [asterisk-users] OT - IP Network Call Recording

[Tom Lynn]

Check out Oreka at sourceforge, too.(aka OrkAudio)

On 2/15/07, Kristian Kielhofner <[EMAIL PROTECTED]> wrote:


On 2/15/07, Cory Andrews <[EMAIL PROTECTED]> wrote:
> Apologies in advance as this is not directly Asterisk related, however I
> thought I might be able to leverage the experience of particiapants on
> this listserv for some advice.
>
> I have a client who is utilizing Talkswith PBX appliances, which have no
> native call monitoring/call recording capabilities.  They are looking
> for an additional application, service or appliance that can sit on the
> LAN, and allow an administrator to monitor or recording inbound/outbound
> calls.  If anyone is aware of a mechanism or solution that would provide
> this capability, please shoot me an email.
>
> Thanks
>
> Cory Andrews

Cory,

  From their website it appears they are using SIP.  With any luck it
will be SIP + ulaw (without re-invites).  If so, do this:

1)  Get a decent managed switch that can setup monitor ports.
Configure one port to monitor the port connected to the Talkswitch.

2)  Get a decent dual-homed machine.

3)  Connect one interface of the dual-homed machine to the monitor
port.  Running Linux, do an ifconfig up [interface name] (no IP
address).  Configure the other interface to connect to a network for
management, copying files, etc.

4)  Start up tcpdump on the interface, writing to a file.

5)  Use something like Cain + Abel to read the RTP and dump the audio to a
file.

6)  Convert files to desired format using sox.

  The only step I left out was "Profit!".  Seriously though, this
depends on a few key assumptions about the Talkswitch:

1)  That it is standard SIP.

2)  It uses ulaw.

3)  It doesn't do re-invites.

  Not any one of these is a show stopper for this type of sollution,
but any one of them (or all of them) could make life a bit harder for
you...

  Good luck!

--
Kristian Kielhofner
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RE: [asterisk-users] Problems with TE110P

[Klaverstyn, David C]

Type in cat /proc/zaptel/* displays

Span 1: ZTDUMMY/1 "ZTDUMMY/1 1"


But if I type in 
lsmod | grep -i wct

I get
wcte11xp   26016  0
wct4xxp   221120  0
zaptel184996  3 ztdummy,wcte11xp,wct4xxp

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir
Cohen
Sent: Monday, 2 April 2007 2:22 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Problems with TE110P

On Mon, Apr 02, 2007 at 12:50:18PM +1000, Klaverstyn, David C wrote:
> I have a new server using Zaptel 1.2.16
> 
> Issuing a ztcfg gives the following error:
> 
> ZT_CHANCONFIG failed on channel 1: No such device or address (6)
>  
> 
> loadzone=au
> 
> defaultzone=au
> 
> span=1,1,0,ccs,hdb3,crc4
> 
> bchan=1-10
> 
> unused=11-15,17-31
> 
> dchan=16
> 

Now what do you actually have loaded?

cat /proc/zaptel/*

-- 
   Tzafrir Cohen   
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+972-50-7952406   mailto:[EMAIL PROTECTED]   
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RE: [asterisk-users] Problems with TE110P

[Joel Hill]

Give this a try it fixes a problem we have had with a couple of Via
boxes.

modprobe wcte11xp
modprobe wcte11xp
ztfcg -vv
zttool

We found that probing the card twice before running ztcfg helped alot.

Cheers,


Joel.



Joel Hill
Support Engineer
Asterisk IT
03 8320 8100

On Mon, 2007-04-02 at 14:56 +1000, Klaverstyn, David C wrote:
> Type in cat /proc/zaptel/* displays
> 
> Span 1: ZTDUMMY/1 "ZTDUMMY/1 1"
> 
> 
> But if I type in 
>   lsmod | grep -i wct
> 
> I get
> wcte11xp   26016  0
> wct4xxp   221120  0
> zaptel184996  3 ztdummy,wcte11xp,wct4xxp
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir
> Cohen
> Sent: Monday, 2 April 2007 2:22 PM
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] Problems with TE110P
> 
> On Mon, Apr 02, 2007 at 12:50:18PM +1000, Klaverstyn, David C wrote:
> > I have a new server using Zaptel 1.2.16
> > 
> > Issuing a ztcfg gives the following error:
> > 
> > ZT_CHANCONFIG failed on channel 1: No such device or address (6)
> >  
> > 
> > loadzone=au
> > 
> > defaultzone=au
> > 
> > span=1,1,0,ccs,hdb3,crc4
> > 
> > bchan=1-10
> > 
> > unused=11-15,17-31
> > 
> > dchan=16
> > 
> 
> Now what do you actually have loaded?
> 
> cat /proc/zaptel/*
> 

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Re: [asterisk-users] Problems with TE110P

[Tzafrir Cohen]

On Mon, Apr 02, 2007 at 02:56:30PM +1000, Klaverstyn, David C wrote:
> Type in cat /proc/zaptel/* displays
> 
> Span 1: ZTDUMMY/1 "ZTDUMMY/1 1"

The driver has not picked up your card.

> 
> 
> But if I type in 
>   lsmod | grep -i wct
> 
> I get
> wcte11xp   26016  0
> wct4xxp   221120  0
> zaptel184996  3 ztdummy,wcte11xp,wct4xxp

Two things to do:

1. What is the output of:   lspci

2. What really happens when the module loads?

  rmmod wcte11xp
  modprobe wcte11xp
  dmesg | tail

What linux distribution is it?

-- 
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Re: [asterisk-users] Problems with TE110P

[Tzafrir Cohen]

On Mon, Apr 02, 2007 at 03:38:18PM +1000, Joel Hill wrote:
> Give this a try it fixes a problem we have had with a couple of Via
> boxes.
> 
> modprobe wcte11xp
> modprobe wcte11xp
> ztfcg -vv
> zttool
> 
> We found that probing the card twice before running ztcfg helped alot.

Are you sure that this is not an issue of timing? Specifically, I would
expect an error message of "no such file or directory" in such a case,
because udev files are not yet there.

What the current debian init.d script does is "wait_for_zapctl" before
running ztcfg.

wait_for_zapctl is a loop that checks if /dev/zap/ctl is there, and if
not, waits a second. 

It seems that once you have /dev/zap/ctl all the others are there as
well.

-- 
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[asterisk-users] Best Hardphone (Subjective?)

[Corporate IT Solutions - Michael Dunne]

After working with the Grandstream GXP 2000 series phones, I have
decided that I am quite unhappy with their problems, both voice quality,
volume, features and others. For their price now, there are plenty of
phones to choose from as  well.

 

So subjectively what would be the best Hardphone for a small/medium
business with multiple line support, BLF, etc.

 

Are the Cisco 7960 the best of the bunch, or Aastra etc?

 

I have just had a GXP start its daily lockup where you come in in the
morning and it requires a hard reset, and this is getting a little
boring having to keep flashing and carrying on.

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RE: [asterisk-users] Problems with TE110P

[Klaverstyn, David C]

Lspci does show:
03:02.0 Ethernet controller: Digium, Inc.: Unknown device 0120 (rev 11)

dmesg | tail
ip_tables: (C) 2000-2002 Netfilter core team
ip_tables: (C) 2000-2002 Netfilter core team
tg3: eth0: Link is up at 100 Mbps, full duplex.
tg3: eth0: Flow control is on for TX and on for RX.
lp: driver loaded but no devices found
NET: Registered protocol family 10
Disabled Privacy Extensions on device c0344160(lo)
IPv6 over IPv4 tunneling driver
divert: not allocating divert_blk for non-ethernet device sit0
eth0: no IPv6 routers present


Using CentOS 4.4

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir
Cohen
Sent: Monday, 2 April 2007 3:43 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Problems with TE110P

On Mon, Apr 02, 2007 at 02:56:30PM +1000, Klaverstyn, David C wrote:
> Type in cat /proc/zaptel/* displays
> 
> Span 1: ZTDUMMY/1 "ZTDUMMY/1 1"

The driver has not picked up your card.

> 
> 
> But if I type in 
>   lsmod | grep -i wct
> 
> I get
> wcte11xp   26016  0
> wct4xxp   221120  0
> zaptel184996  3 ztdummy,wcte11xp,wct4xxp

Two things to do:

1. What is the output of:   lspci

2. What really happens when the module loads?

  rmmod wcte11xp
  modprobe wcte11xp
  dmesg | tail

What linux distribution is it?

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
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Re: [asterisk-users] Best Hardphone (Subjective?)

[Bill Hackensack]

On 4/2/07, Corporate IT Solutions - Michael Dunne <
[EMAIL PROTECTED]> wrote:


 So subjectively what would be the best Hardphone for a small/medium
business with multiple line support, BLF, etc.



Does _anyone_ read the archives anymore?  This is like a weekly question or
something.
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Re: [asterisk-users] Problems with TE110P

[Tzafrir Cohen]

On Mon, Apr 02, 2007 at 04:21:45PM +1000, Klaverstyn, David C wrote:
> Lspci does show:
> 03:02.0 Ethernet controller: Digium, Inc.: Unknown device 0120 (rev 11)
> 
> dmesg | tail
> ip_tables: (C) 2000-2002 Netfilter core team
> ip_tables: (C) 2000-2002 Netfilter core team
> tg3: eth0: Link is up at 100 Mbps, full duplex.
> tg3: eth0: Flow control is on for TX and on for RX.
> lp: driver loaded but no devices found
> NET: Registered protocol family 10
> Disabled Privacy Extensions on device c0344160(lo)
> IPv6 over IPv4 tunneling driver
> divert: not allocating divert_blk for non-ethernet device sit0
> eth0: no IPv6 routers present
> 

But please run it immedietly after running: 

  rmmod wcte11xp; modprobe wcte11xp

as it shows the last messages fromt he kernel.

> Using CentOS 4.4

Alternatively, look in /var/log/messages .

-- 
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