date:20070103

(my pstn calls are coming in thru an upstream asterisk server, so the
called and calling phone number is passed as an extension.)

when caller comes in on 555, he will go to extension 1234 where he
will wait for the API to make a call to 999 for him.  how do I
bridge the two calls?

extensions.conf:

;context where caller comes in
[caller]
555,s,1 Answer()
555,s,n UserEvent(Init) ;this lets me know the connection for
555
555,1234,1 Noop(caller waits to be bridged)
555,1234,2 Background(soothingmusic)

;context for connection - is this needed?
[connect]


from the API:

(do I need to create a new context/extension first?)

Action: Originate
Channel: IAX2/upstream/999  -- calls 999222 thru upsteam IAX
Context: ??
Exten: ??
Priority: ??



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Re: [asterisk-users] Polycom Power Specs

2007-01-03 Thread Dave Schardin


501 - 12V, 1A  and a power/data cable
601 - 24V, 0.5A
650 - 24V, 0.5A

- Dave

On Jan 3, 2007, at 11:48 AM, Peder @ NetworkOblivion wrote:

Does anybody happen to know the input power specs for the Polycom  
IP 500 and IP 600?  We've mixed up our power supplies and we've got  
a whole box of them and can't figure out which go to the Polycoms.   
I would rather not kill the phones by trying random ones


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Looking for voice over IP products?  Visit our VoIP store at http:// 
voipstore.atacomm.com



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Re: [asterisk-users] Polycom Power Specs

2007-01-03 Thread Michael Welter

The 501 is 12VDC, and the 601 is 24VDC, as I recall.  There was a post a 
few months ago that said that plugging the 24VDC into a IP501 will fry 
the phone.


Peder @ NetworkOblivion wrote:
Does anybody happen to know the input power specs for the Polycom IP 500 
and IP 600?  We've mixed up our power supplies and we've got a whole box 
of them and can't figure out which go to the Polycoms.  I would rather 
not kill the phones by trying random ones


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[asterisk-users] Is chan_zap.so loaded?

2007-01-03 Thread John French

Newbie question for sure... I'm unsure of how to tell if chan_zap.so is loaded. 
Also, does autoload in modules.conf take care of it or is it done explicitly?
 
output of lsmod | grep zap:
zaptel208388  16 
wcusb,wctdm,wcfxo,wctdm24xxp,wcte11xp,wct1xxp,w 
ct4xxp,tor2
crc_ccitt   6465  1 zaptel
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Re: [asterisk-users] Polycom Power Specs

2007-01-03 Thread Alvin Austin

FWIW, our Polycom IP601 phones use a transformer with output: 24VDC 
500mA (center contact is positive).


A Polycom reseller (or Polycom sales) could probably give you 
information on these other two models.


Alvin

Peder @ NetworkOblivion wrote:
Does anybody happen to know the input power specs for the Polycom IP 
500 and IP 600?  We've mixed up our power supplies and we've got a 
whole box of them and can't figure out which go to the Polycoms.  I 
would rather not kill the phones by trying random ones


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Re: [asterisk-users] Is chan_zap.so loaded?

2007-01-03 Thread Eric \ManxPower\ Wieling


John French wrote:

Newbie question for sure... I'm unsure of how to tell if chan_zap.so is loaded. 
Also, does autoload in modules.conf take care of it or is it done explicitly?
 
output of lsmod | grep zap:

zaptel208388  16 
wcusb,wctdm,wcfxo,wctdm24xxp,wcte11xp,wct1xxp,w 
ct4xxp,tor2
crc_ccitt   6465  1 zaptel


asterisk -rx show modules | grep -a zap
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[asterisk-users] [Announce] Web-MeetMe 3.0.0 released

We've been holding back on this release to coincide with
the Asterisk 1.4.0 release.  

This is mostly a compatibility release, but there are a
few new features:
*  No longer requires register_globals in PHP
*  Separated code from configuration settings in
./lib/defines.php  (hopefully this will make
future upgrades easier)
*  Migrated all database interfaces to PEAR::DB
which simplifies the code a bit and opens
up the possibility of using other databases
to host the scheduling DB (app_cbmysql is
still only MySQL, but ODBC is planned/hoped for)
*  The conference monitoring code now uses the
concise output from meetme list, improving
the parsing of participant details.
*  Minor tweaks to improve the cbEnd.php script that
enforces the conference duration, plays announcements
and populates the conferencing CDRs.
*  Conference CDR records now store participant duration
in seconds instead of a formatted string, allowing
for further analysis (the web interface still
formats the duration for display purposes)
*  App_cbmysql is updated to work with Asterisk 1.4.0 
*  App_cbmysql has it's own build environment now, no
longer requiring a Makefile patch, etc...

The new release can be found at:  
http://sourceforge.net/projects/web-meetme/

We do have a volunteer developer who will be maintaining the
2.X.X chain for Asterisk 1.2.X compatibility, so bug fixes and
features that are not Asterisk version dependant will still be
made available for older installations.

Thanks,
The Web-MeetMe development team...
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[asterisk-users] Park and Page

2007-01-03 Thread Mike Clark

I have a strange issue going on with one system. If you park a call and
then do a page command, the parked call gets dropped. Both park and page
use meetme, and it appears that the page uses the same conference number
 as parked call. So when the page is complete and hangs up, it drops the
parked call. This is using Asterisk 1.2.13, Zaptel 1.2.11 and
libpri-1.2.4. The system has a Sangoma A101 with wanpipe 2.3.4-2.

Mike

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Re: [asterisk-users] Is chan_zap.so loaded?


From: Eric \ManxPower\ Wieling [EMAIL PROTECTED]

John French wrote:
Newbie question for sure... I'm unsure of how to tell if chan_zap.so is 
loaded. Also, does autoload in modules.conf take care of it or is it done 
explicitly?

 output of lsmod | grep zap:
zaptel208388  16 
wcusb,wctdm,wcfxo,wctdm24xxp,wcte11xp,wct1xxp,w
 ct4xxp,tor2

crc_ccitt   6465  1 zaptel


asterisk -rx show modules | grep -a zap


That's right - chan_zap is not a kernal module (as the .so name suggested).  
As such you won't see it with lsmod.  You must use an Asterisk command.  An 
alternative to grep is to use Asterisk's powerful command, e.g.,

asterisk -rx show modules like zap
asterisk -rx show modules like chan_
or simply,
asterisk -rx show modules like chan_zap

Generally modules.conf should work.

Yuan Liu


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Re: [asterisk-users] Sangoma Remora A202

2007-01-03 Thread Pierre Marceau

Just a thought, maybe it won't boot because there is no power to the CD ROM 
drive

 [EMAIL PROTECTED] 01/03/07 8:43 AM 
Hi - I just got a Sangoma A200 card with a single 2FXO module and  
what appears to be an empty module. I put the card in my Dell GX260,  
but the power light on the front of the box just blinks and won't  
power up.  I did take the power cable from the CDROM to put on the  
card - I don't need the CDROM right now..

I'm looking for direction in getting this card working - I currently  
have a new Trixbox, hoping it'll have the software for this card  
already.  If not, I'll be back asking what drivers I need.  Sangoma  
seems to have a lack of documentation, but it may just be me
  thanks
 Todd
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Re: [asterisk-users] API: how to bridge originated call?


I have uploaded a working patch for version 1.2.12.1, and other that
seems to work in Trunk, but few people is reporting results, you can
help to get this into Asterisk, go here:

http://bugs.digium.com/view.php?id=5841

The patch I ported to 1.2.12.1 is working fine, I have tested in my
servers, is the one called bridge-1.2.12.1.patch, there are other
ones that say trunk, obviously only work with the trunk version of
Asterisk.

Kind Regards

On 1/3/07, chester c young [EMAIL PROTECTED] wrote:

(my pstn calls are coming in thru an upstream asterisk server, so the
called and calling phone number is passed as an extension.)

when caller comes in on 555, he will go to extension 1234 where he
will wait for the API to make a call to 999 for him.  how do I
bridge the two calls?

extensions.conf:

;context where caller comes in
[caller]
555,s,1 Answer()
555,s,n UserEvent(Init) ;this lets me know the connection for
555
555,1234,1 Noop(caller waits to be bridged)
555,1234,2 Background(soothingmusic)

;context for connection - is this needed?
[connect]


from the API:

(do I need to create a new context/extension first?)

Action: Originate
Channel: IAX2/upstream/999  -- calls 999222 thru upsteam IAX
Context: ??
Exten: ??
Priority: ??



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[asterisk-users] Cisco 79x1 Auto-Answer

2007-01-03 Thread Jeremiah Millay

I'm using a mix of Cisco 7960, Linksys SPA-942, Cisco 7961, Cisco 7970 
phones in a paging group. I have all the phones set up with an extra 
line that auto answers the dial from my paging extension when the 
primary line is not in use. All of these are operating correctly however 
the 7961/7970s all ring once and then auto answer so the person paging 
all the phones has the first part of his dictation clipped. This only 
happens with 7961/7970. The linksys and the older 7960 (running 7.4 sip 
firmware) all answer right away and they hear everything. Anyone have 
the newer 7961s and 7970s running 8.X SIP firmware auto-answering 
without any initial ring?


Here are some configuration snippets:

extensions.conf

[globals]
INTERCOM=Local/[EMAIL PROTECTED]Local/[EMAIL PROTECTED]Local/[EMAIL PROTECTED]

[macro-page]
; Paging macro:
; Check to see if SIP device is in use and DO NOT PAGE if they are
; ${ARG1} - Device to page
; ${ARG2} - Other line (not paging line...we don't want to disturb 
their other line)

;

exten = s,1,ChanIsAvail(${ARG2}|js) ; j is for dump and s is for ANY 
call to indicate busy

exten = s,n,NoOp(${AVAILSTATUS})
exten = s,n,SIPAddHeader(Alert-Info: Bellcore-dr5) ; This is the 
shortest ringer for the Cisco phones

exten = s,n,SIPAddHeader(Call-Info:\;answer-after=0) ;  For Linksys SPA
exten = s,n,NoOp() ; Add others here
exten = s,n,Dial(${ARG1}||A())
exten = s,n,Hangup
exten = s,102,Hangup()

[page] ; Paging context

exten = 2201_com,1,Macro(page,SIP/2201_com,SIP/2201) ; 7970
exten = 2202_com,1,Macro(page,SIP/2202_com,SIP/2202) ; 7961
exten = 2203_com,1,Macro(page,SIP/2203_com,SIP/2203) ; 7960
exten = 2204_com,1,Macro(page,SIP/2204_com,SIP/2204) ; 7960
exten = 2205_com,1,Macro(page,SIP/2205_com,SIP/2205) ; 7960
exten = 2207,1,Macro(page,SIP/2207,SIP/2207) ; Linksys SPA-942




Here are the configuration lines relevant to auto-answer in the 
79x1/7970 configuration files:


autoAnswerTimer0/autoAnswerTimer
autoAnswerAltBehaviorfalse/autoAnswerAltBehavior
autoAnswerOverridetrue/autoAnswerOverride

This is configured under sipLines  line in the XML file

autoAnswer
 autoAnswerEnabled3/autoAnswerEnabled
/autoAnswer



The above phone configuration will allow the phone to auto-answer but 
after one ring. I'd like it to just work immediately. Any help would be 
appreciated.

Jeremiah

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[asterisk-users] Error on answer a SIP 401 message

2007-01-03 Thread Frederico Madeira


Hi,

I'm a voip service provider and i'm setting up a asterisk box to
register around 100 lines from my  central softswitch. This asterisk
box will be placed inside a customer and has a digium card to be
interconected with customer's pabx.

My problem is that when asterisk send register message, my softswitch
return with sip 401 and asterisk should send a register message with
Authorization in header.

Only after 10, or 15 or 30 or 43 or 2 messages 401 asterisk start to
send Authorization in header. This is a random time, don't follow any
rule.

This problem cause lines disregistration some times during a day.

How can i solve this problem ?

I use this parameters to register an account:

register=number:[EMAIL PROTECTED]/number
[fonar-number]
type=friend
context=default
secret=pass
username=number
host=sip.provider.com
fromuser=number
fromdomain=sip.provider.com
;nat=yes
;insecure=very
canreinvite=no
;qualify=1
dtmfmode=rfc2833

Thanks.

--
Frederico Madeira
[EMAIL PROTECTED]
www.madeira.eng.br
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Re: [asterisk-users] API: how to bridge originated call?


By the way, Chester, please report results to the bug I sent you, is
very imortant the users feedback to get this into Asterisk

Regards

On 1/3/07, Moises Silva [EMAIL PROTECTED] wrote:

I have uploaded a working patch for version 1.2.12.1, and other that
seems to work in Trunk, but few people is reporting results, you can
help to get this into Asterisk, go here:

http://bugs.digium.com/view.php?id=5841

The patch I ported to 1.2.12.1 is working fine, I have tested in my
servers, is the one called bridge-1.2.12.1.patch, there are other
ones that say trunk, obviously only work with the trunk version of
Asterisk.

Kind Regards

On 1/3/07, chester c young [EMAIL PROTECTED] wrote:
 (my pstn calls are coming in thru an upstream asterisk server, so the
 called and calling phone number is passed as an extension.)

 when caller comes in on 555, he will go to extension 1234 where he
 will wait for the API to make a call to 999 for him.  how do I
 bridge the two calls?

 extensions.conf:

 ;context where caller comes in
 [caller]
 555,s,1 Answer()
 555,s,n UserEvent(Init) ;this lets me know the connection for
 555
 555,1234,1 Noop(caller waits to be bridged)
 555,1234,2 Background(soothingmusic)

 ;context for connection - is this needed?
 [connect]


 from the API:

 (do I need to create a new context/extension first?)

 Action: Originate
 Channel: IAX2/upstream/999  -- calls 999222 thru upsteam IAX
 Context: ??
 Exten: ??
 Priority: ??



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Re: [asterisk-users] Fonebridge2

2007-01-03 Thread Bill Burdick

Yeah, I've played with both an older one (FoneBridge 1) and a FoneBridge 2 
unit, and
they seem to work as advertised.  The FoneBridge 2 is much nicer, sets up much 
faster
and boots much faster.  Besides, RedFone is great company to work with.

Bill Burdick

 Hello List

 Does anybody have any experience with the FoneBridge line of products from 
 RedFone?
 I think their HA implementation sounds interesting, and like the prospect of 
 having
 dedicated hardware for our PRI connections.

 Kind Regards

 Jon Leren Schøpzinsky
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[asterisk-users] have a phone number from stanaphone and a working trixbox, how do I connect them?

2007-01-03 Thread blackwater dev


I have a phone number for traditional phone lines through stana phone and a
working trixbox server.  What do I need to do to connect the two so when
someone calls the number from a normal phone, they get my server?

Thanks!
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[asterisk-users] over 200 queues, anyone?

2007-01-03 Thread lenz


Hello list,
one of our clients is going to be deploying a system with over 200  
differently composed queues and 100 agents. We are going to do a full test  
of the viability of this solution before deployment, but I was wondering  
if anyone has experience of such a setup and if there are any obvious  
problems or no-nos.

Any suggestion welcomed,
l.

--
Home of QueueMetrics - http://queuemetrics.loway.it

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Re: [asterisk-users] have a phone number from stanaphone and a working trixbox, how do I connect them?

2007-01-03 Thread Alex Robar


I used these directions to get Stanaphone working on my FreePBX box:

http://www.voip-info.org/wiki/view/Asterisk%40Home+Handbook+Wiki+Chapter+6#614Stanaphone

Alex

On 1/3/07, blackwater dev [EMAIL PROTECTED] wrote:


I have a phone number for traditional phone lines through stana phone and
a working trixbox server.  What do I need to do to connect the two so when
someone calls the number from a normal phone, they get my server?

Thanks!

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--
Alex Robar
[EMAIL PROTECTED]
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Re: [asterisk-users] over 200 queues, anyone?

2007-01-03 Thread Joe Dennick

Yeah, get a Business Process specialist to analyze the client's
environment and develop a better solution.  200 queues with only 100
agents sounds pretty ludicrous to me!

On Wed, 2007-01-03 at 14:22 -0600, lenz wrote:
 Hello list,
 one of our clients is going to be deploying a system with over 200  
 differently composed queues and 100 agents. We are going to do a full test  
 of the viability of this solution before deployment, but I was wondering  
 if anyone has experience of such a setup and if there are any obvious  
 problems or no-nos.
 Any suggestion welcomed,
 l.
 

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RE: [asterisk-users] have a phone number from stanaphone and a workingtrixbox, h


From: blackwater dev [EMAIL PROTECTED]

I have a phone number for traditional phone lines through stana phone and a
working trixbox server.  What do I need to do to connect the two so when
someone calls the number from a normal phone, they get my server?

Thanks!


Get a cheap X100P or a clone card.  I had a Motorola SM56 winMODEM that 
worked fine, then bought an X100P from Digitnetworks (not affiliated with 
Digium).  Clone cards can be found on eBay.


Yuan Liu


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Re: [asterisk-users] Polycom Power Specs

2007-01-03 Thread LST


The IP600 is 12v!!!  I fried a 600 when I used power adapter from 601.


On 1/3/07, Alvin Austin [EMAIL PROTECTED] wrote:


FWIW, our Polycom IP601 phones use a transformer with output: 24VDC
500mA (center contact is positive).

A Polycom reseller (or Polycom sales) could probably give you
information on these other two models.

Alvin

Peder @ NetworkOblivion wrote:
 Does anybody happen to know the input power specs for the Polycom IP
 500 and IP 600?  We've mixed up our power supplies and we've got a
 whole box of them and can't figure out which go to the Polycoms.  I
 would rather not kill the phones by trying random ones

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Re: [asterisk-users] Polycom Power Specs

2007-01-03 Thread Bas van der Veen


Peter,

I have 600's that are 12V 1.5A, + in the center. This differs from some 
of the other answers, maybe those differences are regional (although 
that would seem rather silly).


HTH

B

Peder @ NetworkOblivion wrote:
Does anybody happen to know the input power specs for the Polycom IP 500 
and IP 600?  We've mixed up our power supplies and we've got a whole box 
of them and can't figure out which go to the Polycoms.  I would rather 
not kill the phones by trying random ones


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[asterisk-users] no unicall on 1.4 (was: OnHook Call Announcement...)

2007-01-03 Thread Anton Krall

And probably wont be as Steve Underwood explained to me that he is now 
supporting openpbx and has stopped support for unicall on asterisk 1.4

Can anybody at digium confirm? Is unicall going to be left out of 1.4?
 


|-Original Message-
|From: [EMAIL PROTECTED] [mailto:asterisk-users-
|[EMAIL PROTECTED] On Behalf Of Carlos Chavez
|Sent: Tuesday, January 02, 2007 6:02 PM
|To: Asterisk
|Subject: [asterisk-users] OnHook Call Announcement...
|
|   I have a customer that is asking for a feature called On Hook Call
|Announcement.  The way he explains it is that when someone is on another call 
you can
|sort of break in into their conversation but only the local person hears you 
and not the
|external caller.
|
|   Basically he wants to use this function so he can call anyone in the 
company
|even if they are already on a call (he is the big boss).  I saw that there is 
a feature
|coming in 1.4 called Whisper paging that may do something like this but I need 
to know
|if it is possible to do it in 1.2 because there is still no support for 
Unicall on 1.4
|
|--
|Telecomunicaciones Abiertas de Mexico S.A. de C.V.
|Carlos Chvez Prats
|Director de Tecnologa
|+52-55-91169161 ext 2001


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Re: [asterisk-users] Polycom Power Specs

2007-01-03 Thread Bas van der Veen

I haven't read trough the thread well enough. The 600 is 12V 1.5A 
indeed. Too bad they don't all have the same voltage.


LST wrote:

The IP600 is 12v!!!  I fried a 600 when I used power adapter from 601.


On 1/3/07, *Alvin Austin* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]  wrote:

FWIW, our Polycom IP601 phones use a transformer with output: 24VDC
500mA (center contact is positive).

A Polycom reseller (or Polycom sales) could probably give you
information on these other two models.

Alvin

Peder @ NetworkOblivion wrote:
  Does anybody happen to know the input power specs for the Polycom IP
  500 and IP 600?  We've mixed up our power supplies and we've got a
  whole box of them and can't figure out which go to the Polycoms.  I
  would rather not kill the phones by trying random ones
 





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RE: [asterisk-users] yet another faxing issue (outbound only, via ATA)

2007-01-03 Thread Bill Gibbs

I set aside a couple of channels and removed echo cancellation on them.  So 
far, faxing outbound through an ATA is working fine now.

Bill

-Original Message-
From: [EMAIL PROTECTED] on behalf of Marco Mouta
Sent: Wed 1/3/2007 6:42 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] yet another faxing issue (outbound only, via ATA)

Hi all,

I was having a similar issue, using TE110P from Digium  all incoming faxes
were detected and correctly received.

When trying to send outbound faxes, they all get broken... I do believe it
may be related with Echo Cancel enabled on my Zapata.conf, any ways i've set
also fax detect for inbound and outbound on zapata, but that stills diferent
from the receiving model as it relies on NVfaxdetect to detect.

After many trials, i setup an architecture with another Server Running
Hylafax and IAXmodem registring on my * Box and i just get out of troubles.

It's perfect sending and receiving faxes with notifications and everything
else, Hylafax + IAXmodem and Asterisk are working like a charm.

I must say that we don't send too many faxes per day, but until now no
problems! And yes didn't change anything on Zapata config or something else
on Asterisk Box, i just added IAX account registred my Hylafax IAXmodem
there and Voilá :)

On 1/3/07, Thomas Kenyon [EMAIL PROTECTED] wrote:

 Bill Gibbs wrote:

  My next step is to connect the fax machine to a Wildcard X100P.

 Check to see if there is Echo cancellation in the SPA-1001, and if so
 turn it off. If there is an adaptive Jitter buffer on the SPA-1001, try
 changing it to a fixed one (probably no more than 40ms).

 Why would you connect a fax machine to an X100P, aren't they FXO cards?

 Have you tried terminating to a VOIP provider? (to see if the problem is
 with the ATA).

 Here I use a fax machine connected to a CS6220 which is connected to the
 asterisk box and terminates with a TDM400P card (so a completely
 different arrangement).

  Any other suggestions?  Black magic?  Voodoo?

  Bill

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Re: [asterisk-users] over 200 queues, anyone?

2007-01-03 Thread Alex Robar


Not necessarily... The same agents could very well be providing support for
multiple companies. You wouldn't want an announcement from company A in
company B's queues.

Alex

On 1/3/07, Joe Dennick [EMAIL PROTECTED] wrote:


Yeah, get a Business Process specialist to analyze the client's
environment and develop a better solution.  200 queues with only 100
agents sounds pretty ludicrous to me!

On Wed, 2007-01-03 at 14:22 -0600, lenz wrote:
 Hello list,
 one of our clients is going to be deploying a system with over 200
 differently composed queues and 100 agents. We are going to do a full
test
 of the viability of this solution before deployment, but I was wondering
 if anyone has experience of such a setup and if there are any obvious
 problems or no-nos.
 Any suggestion welcomed,
 l.


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--
Alex Robar
[EMAIL PROTECTED]
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Re: [asterisk-users] Voicemail to email

2007-01-03 Thread Forrest Beck


You just specify the users email address in the voicemail.conf file,
along with their mailbox number:

see the sample file:

[default]
; Define maximum number of messages per folder for a particular context.
;maxmsg=50
2506 = 2506,Grandstream,[EMAIL PROTECTED],,attach=yes|imapuser=fbeck
1234 = 4242,Example Mailbox,[EMAIL PROTECTED]
;4200 = 9855,Mark
Spencer,[EMAIL PROTECTED],[EMAIL PROTECTED],attach=no|[EMAIL 
PROTECTED]|tz=central|maxmsg=10
;4300 = 3456,Ben Rigas,[EMAIL PROTECTED]
;4310 = -5432,Sales,[EMAIL PROTECTED]
;4069 = 6522,Matt
Brooks,[EMAIL 
PROTECTED],,|tz=central|attach=yes|saycid=yes|dialout=fromvm|callback=fromvm|review=yes|operator=yes|envelope=yes|sayduration=yes|saydurationm=1
;4073 = 1099,Bianca Paige,[EMAIL PROTECTED],,delete=1
;4110 = 3443,Rob Flynn,[EMAIL PROTECTED]
;4235 = 1234,Jim Holmes,[EMAIL PROTECTED],,Tz=european


2503 = 2503,Forrest Beck,[EMAIL PROTECTED]

On 1/3/07, Doug Crompton [EMAIL PROTECTED] wrote:

There should be an example in your voicemail.conf

Here is mine...

mail is tagged from [EMAIL PROTECTED] and sent to [EMAIL PROTECTED]

In voicemail.conf

mailcmd=/usr/sbin/sendmail -f [EMAIL PROTECTED] [EMAIL PROTECTED]

You of course would use the mailer that your system uses. I have sendmail
on the same system as Asterisk.

There are many other things you can define for mail but all should be in
your example  voicemail.conf

Doug

On Wed, 3 Jan 2007, Mark Greene wrote:

 Hey guys,

 I need to set up asterisk so that it sends the voicemail to the users email.
 I understand that I need to say attatch=yes, but what else needs to be
 done. I would think that somewhere I need to specify the server that it uses
 to send the email, etc.

 - Mark



Those that sacrifice essential liberty to obtain a little temporary safety
 deserve neither liberty nor safety.  -- Ben Franklin (1759)


*  Doug Crompton   *
*  Richboro, PA 18954  *
*  215-431-6307*
*  *
* [EMAIL PROTECTED]*
* http://www.crompton.com  *



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Re: [asterisk-users] over 200 queues, anyone?

2007-01-03 Thread Joe Dennick

I stand corrected, but it still seems excessive.

On Wed, 2007-01-03 at 15:06 -0600, Alex Robar wrote:
 Not necessarily... The same agents could very well be providing
 support for multiple companies. You wouldn't want an announcement from
 company A in company B's queues. 
 
 Alex
 
 On 1/3/07, Joe Dennick [EMAIL PROTECTED] wrote:
 Yeah, get a Business Process specialist to analyze the
 client's
 environment and develop a better solution.  200 queues with
 only 100
 agents sounds pretty ludicrous to me!
 
 On Wed, 2007-01-03 at 14:22 -0600, lenz wrote: 
  Hello list,
  one of our clients is going to be deploying a system with
 over 200
  differently composed queues and 100 agents. We are going to
 do a full test
  of the viability of this solution before deployment, but I
 was wondering 
  if anyone has experience of such a setup and if there are
 any obvious
  problems or no-nos.
  Any suggestion welcomed,
  l.
 
 
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 -- 
 Alex Robar
 [EMAIL PROTECTED] 
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Re: [asterisk-users] Voicemail to email

2007-01-03 Thread Mark Greene


In my case it was never any confusion over what needs to be configured in
asterisk. I was wondering what mail program asterisk used and what needed to
be configured with it. In my case I had to set up sendmail on my system to
relay through our internal mail server.

sendmail.mc was the file I had to modify and SmartHost was what I had to
append to.
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[asterisk-users] ARI help

2007-01-03 Thread Mark Greene


I am trying to use ARI for call monitoring. Recording conversations and
such. The problem is that I don't use AMP, and don't have any sort of a
database for CDR setup. It is all stored in the CSV file by default. When I
setup ARI I tell it to go into standalone mode, and I set the asterisk
manager username and password that was defined in manager.conf, but it also
wants a cdr username and password that I don't know exists.

Also, EVERYTIME I leave the callmonitor module active, it tells me that it
could not find the DB extension and to check AMP, asterisk, and main.conf.

So do I NEED to have AMP installed for call monitor to work? How can I setup
ARI with JUST ARI and a STANDARD asterisk install. No AMP or SQL.

- Mark
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[asterisk-users] Gentoo ebuild for 1.4?

2007-01-03 Thread Chris Bagnall

Greetings list,

Does anyone know if there's a maintained 1.4 ebuild for Gentoo? Even with
the ~amd64 keyword, latest in the official Portage repository is 1.2.13.

Thanks in advance.

Regards,

Chris
-- 
C.M. Bagnall, Director, Minotaur I.T. Limited
This email is made from 100% recycled electrons


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Re: [asterisk-users] Problem with centos 4.4 and jabber/gtalk (really iksemel)

2007-01-03 Thread Bob Chiodini




Kenneth Padgett wrote:

Bob,


It looks like the gnutls development package is called gnutls-devel:
'yum install gnutls-devel' should get the package installed.


Yah, I thought that would be it. I have that installed, as well as
gnutls. (I basically installed both packages you can find with yum
search gnutls). Any other thoughts, can I just d/l the libs and
uncompress them somewhere?

-Kenneth 


Kenneth,

I don't have a Centos machine at home, but under Fedora Core 6 
autogen.sh and ./configure work after installing gnutls-devel.


I followed the instructions at:

http://voip-info.org/wiki/view/Asterisk+Speaks+with+Google+Talk


as you suggested in your first post.  Without gnutls-devel installed the 
autogen.sh step fails.


I would have thought that FC6 and Centos 4.4 would be pretty close as 
far as directory hierarchy.


Here is a list of the pertinent files from the FC6 gnutls-devel package:

rpm -q --filesbypkg gnutls-devel
gnutls-devel  /usr/bin/libgnutls-config
gnutls-devel  /usr/bin/libgnutls-extra-config
gnutls-devel  /usr/include/gnutls
gnutls-devel  /usr/include/gnutls/compat.h
gnutls-devel  /usr/include/gnutls/extra.h
gnutls-devel  /usr/include/gnutls/gnutls.h
gnutls-devel  /usr/include/gnutls/openpgp.h
gnutls-devel  /usr/include/gnutls/openssl.h
gnutls-devel  /usr/include/gnutls/pkcs12.h
gnutls-devel  /usr/include/gnutls/x509.h
gnutls-devel  /usr/lib/libgnutls-extra.a
gnutls-devel  /usr/lib/libgnutls-extra.so
gnutls-devel  /usr/lib/libgnutls-openssl.a
gnutls-devel  /usr/lib/libgnutls-openssl.so
gnutls-devel  /usr/lib/libgnutls.a
gnutls-devel  /usr/lib/libgnutls.so

You might want to check it against your Centos installation.  If it's 
different try:


./configure --prefix=/usr --with-libgnutls-prefix=PFX

Where PFX is the where libgnutls is installed (from ./configure --help).

Bob...

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[asterisk-users] Re: [Announce] Web-MeetMe 3.0.0 released

2007-01-03 Thread Steven

Any screenshots available?

I do not want to even test this without having any idea what it is or how it 
works.

The brief description on sf.net is not enough.

-- 
-- 
Steven

http://www.glimasoutheast.org



Dan Austin [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED]
We've been holding back on this release to coincide with
the Asterisk 1.4.0 release.

This is mostly a compatibility release, but there are a
few new features:
*  No longer requires register_globals in PHP
*  Separated code from configuration settings in
./lib/defines.php  (hopefully this will make
future upgrades easier)
*  Migrated all database interfaces to PEAR::DB
which simplifies the code a bit and opens
up the possibility of using other databases
to host the scheduling DB (app_cbmysql is
still only MySQL, but ODBC is planned/hoped for)
*  The conference monitoring code now uses the
concise output from meetme list, improving
the parsing of participant details.
*  Minor tweaks to improve the cbEnd.php script that
enforces the conference duration, plays announcements
and populates the conferencing CDRs.
*  Conference CDR records now store participant duration
in seconds instead of a formatted string, allowing
for further analysis (the web interface still
formats the duration for display purposes)
*  App_cbmysql is updated to work with Asterisk 1.4.0
*  App_cbmysql has it's own build environment now, no
longer requiring a Makefile patch, etc...

The new release can be found at:
http://sourceforge.net/projects/web-meetme/

We do have a volunteer developer who will be maintaining the
2.X.X chain for Asterisk 1.2.X compatibility, so bug fixes and
features that are not Asterisk version dependant will still be
made available for older installations.

Thanks,
The Web-MeetMe development team...
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Re: [asterisk-users] Re: [Announce] Web-MeetMe 3.0.0 released

2007-01-03 Thread Guillermo Salas M.

On Wed, 2007-01-03 at 16:55 -0500, Steven wrote:
 Any screenshots available?
 
 I do not want to even test this without having any idea what it is or how it 
 works.
 
 The brief description on sf.net is not enough.
 

I'm testing the 2.0 version on asterisk 1.2 . What do you want to know
about the application?

Best regards,


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

Linux User: 255902

Beat me, whip me, make me use Windows!

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

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

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Re: [asterisk-users] Sangoma Remora A202

2007-01-03 Thread Todd H

Thanks - that turned out to be the problem.  Well- one of those  
solutions.  I removed the blank and swapped the FXO module to the  
other port.  I don't know if it was a bad port on the A200, but since  
I don't plan on using it, I won't worry about it- just regret it in a  
year when I get a second FXO module ;)


As for documentation, I did find the info on WanPipe, but am not sure  
what Wanpipe is..   I'll do some more reading tonight.  Thanks for  
the info.

  Todd

On Jan 3, 2007, at 11:40 AM, Bruce Reeves wrote:

Try switching the order of the blank module and the FXO or remove  
the blank, I had a similar Dell do the same and after some  
experimenting found that the removing the blank solved the problem.



On 1/3/07, Rob Schall [EMAIL PROTECTED] wrote:
If the light on the dell is blinking amber... that typically means you
have a power issue.

Rob


Time Bandit wrote:
 Hi - I just got a Sangoma A200 card with a single 2FXO module and
 what appears to be an empty module. I put the card in my Dell  
GX260,

 but the power light on the front of the box just blinks and won't
 power up.
 Maybe your card is not properly seated.

 seems to have a lack of documentation, but it may just be me
 It is just you ;)

 http://wiki.sangoma.com/

 If you still have problems with the card, contact Sangoma, they have
 very good customer support : http://www.sangoma.com/main/contact

 hth
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--
Bruce
Nortex Networks
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[asterisk-users] Detect IP path before calling

Any easy way to determine if IP connectivity before attempting a SIP call?  
IP connectivity could be a timeout.


Yuan Liu


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[asterisk-users] Asterisk Core Dump in app_queue - Anyone seen?

2007-01-03 Thread Douglas Garstang

Anyone seen this? It ocurred on a 'reload app_queue.so' command.
Asterisk version is 1.2.9.1.

Tried again, but it was not immediately reproducable.

Doug.

(gdb) bt
#0  reload_queues () at app_queue.c:3339
#1  0xb778a7a8 in reload () at app_queue.c:4012
#2  0x0805bb44 in ast_module_reload (name=0x8137cc7 app_queue.so) at 
loader.c:257
#3  0x08092b3f in handle_reload (fd=33, argc=2, argv=0xbddfa470) at cli.c:147
#4  0x0809283e in ast_cli_command (fd=33, s=0x6d6f7250 Address 0x6d6f7250 out 
of bounds) at cli.c:1364
#5  0x080aef0f in action_command (s=0x81ead18, m=0xbddfaac0) at manager.c:927
#6  0x080b3ee4 in process_message (s=0x81ead18, m=0xbddfaac0) at manager.c:1305
#7  0x080b2ac5 in session_do (data=0x81ead18) at manager.c:1401
#8  0xb7f15ed8 in pthread_start_thread () from /lib/libpthread.so.0
#9  0xb7e147ea in clone () from /lib/libc.so.6
(gdb) bt full
#0  reload_queues () at app_queue.c:3339
q = (struct ast_call_queue *) 0x81adca8
ql = (struct ast_call_queue *) 0xbddfaec0
qn = (struct ast_call_queue *) 0xb7dc03b3
cfg = (struct ast_config *) 0x81aca30
cat = 0x81507e0 mcao_QMain
tmp = 0x6d6f7250 Address 0x6d6f7250 out of bounds
var = (struct ast_variable *) 0x811e340
prev = (struct member *) 0x8101b79
cur = (struct member *) 0x2854554f
newm = (struct member *) 0x0
new = 0
general_val = 0x2854554f Address 0x2854554f out of bounds
interface = '\0' repeats 79 times
penalty = 900
#1  0xb778a7a8 in reload () at app_queue.c:4012
No locals.
#2  0x0805bb44 in ast_module_reload (name=0x8137cc7 app_queue.so) at 
loader.c:257
m = (struct module *) 0x81f3b10
reloaded = 2
oldversion = 863401873
reload = (int (*)(void)) 0xb778a7a0 reload
#3  0x08092b3f in handle_reload (fd=33, argc=2, argv=0xbddfa470) at cli.c:147
x = 1
res = 1836020304
#4  0x0809283e in ast_cli_command (fd=33, s=0x6d6f7250 Address 0x6d6f7250 out 
of bounds) at cli.c:1364
argv = {0x8137cc0 reload, 0x8137cc7 app_queue.so, 0x0, 0x8227d68  
;\\b¬úæ·: 2007-01-03 15:17:39.165755\r\n, 
  0xbddfa49c h¥ß½ïÀÛ·h}\\bh}\\b, 
  0xb7dc3fea 
ë\234\211$ÿÐëÔ\213]ô\213uø\213}ü\211ì]é¿üÿÿU\211å\203ì(\211]ô\211uø\211}üè°Cûÿ\201ÃÜ\237\n,
 0xb7e6fa00 , 
  0x8227d68  ;\\b¬úæ·: 2007-01-03 15:17:39.165755\r\n, 0xb7e6dff4 
\034\020, 0x26 Address 0x26 out of bounds, 
  0x27 Address 0x27 out of bounds, 0xbddfa568 \200, 0xb7dbc0ef 
\213U\b\213\002\205Àu\b\213\205pÿÿÿ\211\002ÆD\aÿ, 
  0x8227d68  ;\\b¬úæ·: 2007-01-03 15:17:39.165755\r\n, 0x8227d68  
;\\b¬úæ·: 2007-01-03 15:17:39.165755\r\n, 
  0x26 Address 0x26 out of bounds, 0x8227d68  ;\\b¬úæ·: 2007-01-03 
15:17:39.165755\r\n, 0x0, 
  0x26 Address 0x26 out of bounds, 0xfbad8000 Address 0xfbad8000 out of 
bounds, 
  0x8227d68  ;\\b¬úæ·: 2007-01-03 15:17:39.165755\r\n, 0x8227d68  
;\\b¬úæ·: 2007-01-03 15:17:39.165755\r\n, 
  0x8227d68  ;\\b¬úæ·: 2007-01-03 15:17:39.165755\r\n, 0x8227d68  
;\\b¬úæ·: 2007-01-03 15:17:39.165755\r\n, 0x8227d8e , 
  0x8227dcc  , 0x8227d68  ;\\b¬úæ·: 2007-01-03 15:17:39.165755\r\n, 
0x8227dcc  , 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 
  0xbddfa544 ôßæ·\020\234 \bh¥ß½ê?Ü·, 0xb700 Address 0xb700 out of 
bounds, 0x0, 0xbddfa544 ôßæ·\020\234 \bh¥ß½ê?Ü·, 
  0xb7e6dff4 \034\020, 0x0, 0xb7e6da00 , 0x0, 0xb7f1a756 \201Ã\236H, 
0xb7f1eff4 tî, 0xb7e6fa00 , 0xb7e6fa00 , 
  0xbddfa54c h¥ß½ê?Ü·, 0xb7f170eb 
ëÃ\213\203pÿÿÿ;(r\022\213\203Ðÿÿÿ;(s\b\213\203¤ÿÿÿë½\213\203 ÿÿÿ\213, 
0xb7e6fa10 , 
  0x8209c10 È\017\025\bèÃ\035\b: 2007-01-03 15:17:39.165755\r\n, 0xb7e6fa00 
, 0xb7e6dff4 \034\020, 0xb7e6dff4 \034\020, 
  0x8209c10 È\017\025\bèÃ\035\b: 2007-01-03 15:17:39.165755\r\n, 0xbddfa568 
\200, 
  0xb7dc3fea 
ë\234\211$ÿÐëÔ\213]ô\213uø\213}ü\211ì]é¿üÿÿU\211å\203ì(\211]ô\211uø\211}üè°Cûÿ\201ÃÜ\237\n,
 0xb7e6fa00 , 
  0x8209c10 È\017\025\bèÃ\035\b: 2007-01-03 15:17:39.165755\r\n, 0x21 
Address 0x21 out of bounds, 
  0x21 Address 0x21 out of bounds, 0x81ead18 \017, 0x80 Address 0x80 out 
of bounds, 
  0x8091ffb \213\\$\030\203Ä\034ÃÇ\004$\004}
e = (struct ast_cli_entry *) 0x81197a0
x = 2
dup = 0x8137cc0 reload
tws = 0
#5  0x080aef0f in action_command (s=0x81ead18, m=0xbddfaac0) at manager.c:927
No locals.
#6  0x080b3ee4 in process_message (s=0x81ead18, m=0xbddfaac0) at manager.c:1305
ret = 0
eqe = (struct eventqent *) 0x0
action = Command, '\0' repeats 72 times
tmp = (struct manager_action *) 0x8144818
idText = ActionID: 2007-01-03 15:17:39.165755\r\n, '\0' repeats 217 
times
iabuf = 216.187.141.250
#7  0x080b2ac5 in session_do (data=0x81ead18) at manager.c:1401
m = {hdrcount = 3, headers = {Action: Command\000\n, '\0' repeats 
238 times, 
Command: reload app_queue.so\000\n, '\0' repeats 225 times, 
ActionID: 2007-01-03 15:17:39.165755\000\n, '\0' repeats 217 times, 
\000\n, '\0' repeats 253 times, 
'\0' repeats 255 times repeats 76 times}}

RE: [asterisk-users] voice fax modem and asterisk


From: Gregory Machin [EMAIL PROTECTED]

Hi I have been asked to ind out if there is a way to use asterisk to
answere a voice fax modem so it can provide an answering service and
record messages ?


Absolutely - if that MODEM happens to be an X100P clone - such as my 
ENF656-PCIG-MOPR.  There are quite a few - and you can buy one still.  It 
can do a lot more than answering and recording.


Yuan Liu


--
Gregory Machin
[EMAIL PROTECTED]
www.linuxpro.co.za



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Re: [asterisk-users] Sangoma Remora A202

2007-01-03 Thread Time Bandit


Thanks - that turned out to be the problem.  Well- one of those solutions.
I removed the blank and swapped the FXO module to the other port.  I don't
know if it was a bad port on the A200, but since I don't plan on using it, I
won't worry about it- just regret it in a year when I get a second FXO
module ;)


No you won't, since Sangoma cards come with a 5 year warranty ;)

Glad you fixed it
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RE: [asterisk-users] Re: [Announce] Web-MeetMe 3.0.0 released

Steven wrote:
 Any screenshots available?
Sorry, not yet.  I've been meaning to get the tools 
together to capture some images, but coding, QA and
paid work have taken priority...

 I do not want to even test this without having any 
 idea what it is or how it works.
It is a suite of tools to permit the scheduling of 
conferences.

 The brief description on sf.net is not enough.

There are four components to the suite:
1.  A collection of web pages (PHP and javascript)
that provide the interface to add, update, delete
conferences.  There is also a page to monitor
active conferences, with the ability to mute,
unmute and eject participants, as well as to
have Asterisk place a call to a participant.
2.  An Asterisk application that validates the
conference id, start time, participant count
and any user/moderator pins
3.  A small php script that logs the conference 
participants to a CDR-like database and
enforces a conference's scheduled endtime.
4.  A small number of sound files to convey 
conference events/status.  The files
were recorded by Alison, so they are a nice
match for Asterisk's standard sound files.

Key features:
1.  Enforcable start and end times.
Optionaly alert callers if they are too
early.  Also optionally allow callers to
join a conference if they are too early.  
2.  Configurable conference id, user and moderator pins.
Pins are optional.  If a moderator pin is set, then
the user pin is required.
3.  Recurring conferences.
Dialy, weekly and bi-weekly
4.  Future conferences can be edited.
The entire series, or one at a time
5.  A conference endtime can be extended once it has
started, but this is the only configuration
setting that can be changed once the conference
start time has been reached.
6.  Simple branding.  Reasonably easy to change logos
and page headings to refect your company (helps
get past management objections)
7.  Optional authentication (LDAP or Database) to
permit users to schedule and manage their own
conferences.

I'll work on screenshots soon.

Dan
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[asterisk-users] TDM400 UK Caller ID problems ...

2007-01-03 Thread Gordon Henderson



So just when I thought I had caller ID going fine things seem to have 
taken a turn for the worst. I'm now seeing lots of misses in picking up 
the caller ID on a line I know provides it.


I know I've changed the TDM400 card and upgraded to the latest (1.2) 
version of Zaptel but could this be connected?


This is what I'm seeing in the output:

  == Starting post polarity CID detection on channel 3
-- Starting simple switch on 'Zap/3-1'
Jan  3 21:21:27 NOTICE[11667]: chan_zap.c:5888 ss_thread: Got event 17 
(Polarity Reversal)...
-- Executing NoOp(Zap/3-1, Look whos calling: ) in new stack

(The NoOp line just dumps the caller ID, or not in this case)

Sometimes it does work - which is what makes it more preplexing.

Config files haven't changed and they have the right runes in them:

usecallerid=yes
cidsignalling=v23
cidstart=polarity


One thing I have started doing is running fxotune on the lines - do you 
think this might make a difference?


And has anyone any experience of CID on Telewest lines in the UK? (as 
opposed to BT which I'm currently using, but have a prospect who has some 
Telewest lines)



Thanks,

Gordon
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[asterisk-users] Any quiet 24 port POE switches out there?

2007-01-03 Thread John French

I have an upcoming install which places the switch close to some
employees in a quiet work environment.  Can anyone recommend a quiet 24
port POE switch?  The Linksys SRW224P behind me right now would be
objectionable, I'm sure.

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Re: [asterisk-users] Detect IP path before calling

2007-01-03 Thread Paul Hales


With the chanisavail command.

PaulH

On Wed, 2007-01-03 at 14:22 -0800, Yuan LIU wrote:
 Any easy way to determine if IP connectivity before attempting a SIP call?  
 IP connectivity could be a timeout.
 
 Yuan Liu
 
 
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Re: [asterisk-users] over 200 queues, anyone?

2007-01-03 Thread Richard Lyman


lenz wrote:

Hello list,
one of our clients is going to be deploying a system with over 200 
differently composed queues and 100 agents. We are going to do a full 
test of the viability of this solution before deployment, but I was 
wondering if anyone has experience of such a setup and if there are 
any obvious problems or no-nos.

Any suggestion welcomed,
one of our sites likes to micro manage things to the point of 38 queues. 

the thing you will find is that if your agents are members of various 
queues


each member instance of each queue will get QueueMemberStatus events. 
(i've commented mine out because of flooding that occurs of the manager 
interface)


deadlocks would be the thing to watch for.

just food for thought.

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Re: [asterisk-users] Any quiet 24 port POE switches out there?

2007-01-03 Thread Brian Roy


On 1/3/07, John French [EMAIL PROTECTED] wrote:


 I have an upcoming install which places the switch close to some
employees in a quiet work environment.  Can anyone recommend a quiet 24 port
POE switch?




The 8port Netgear switch on my desk doesn't have any fans. FS108p. Not sure
if they make a 24port switch or not.

-Brian
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Re: [asterisk-users] no unicall on 1.4 (was: OnHook Call Announcement...)


On 1/3/07, Anton Krall [EMAIL PROTECTED] wrote:

And probably wont be as Steve Underwood explained to me that he is now 
supporting openpbx and has stopped support for unicall on asterisk 1.4

Can anybody at digium confirm? Is unicall going to be left out of 1.4?


This has nothing to do with Digium, it has to do with anybody wanting
to code the version for 1.4, AFAIK Steve never worked for Digium and
Digium never distributed Unicall driver.

Porting Unicall to 1.4 is in my TODO since 1 month ago, may be this
month I will have the time to give a look at the code and try to make
it work on 1.4, if somebody else cant do it before.

Regards.

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Re: [asterisk-users] Any quiet 24 port POE switches out there?

2007-01-03 Thread Jerry Jones

I suspect any 24port will have a fan. The Netgear FSM7326P are not  
too bad and we have had good luck with them.


ps - I also load their open source software.


On Jan 3, 2007, at 4:51 PM, John French wrote:

I have an upcoming install which places the switch close to some  
employees in a quiet work environment.  Can anyone recommend a  
quiet 24 port POE switch?  The Linksys SRW224P behind me right now  
would be objectionable, I'm sure.

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Re: [asterisk-users] Polycom Power Specs

2007-01-03 Thread Chris Mason (Lists)


Older models, 500 and 600, are 12V, newer 601s are 24v

--
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(264) 497-5670 Fax: (264) 497-8463
Int:  (305) 704-7249 Fax: (815)301-9759 UK 44.207.183.0271
Cell: 264-235-5670
Yahoo IM: [EMAIL PROTECTED] 



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Re: [asterisk-users] Block some number outgoing from joust one extention

2007-01-03 Thread C F


The easiest way is thru using contexts.

On 1/3/07, Mattias Andersson [EMAIL PROTECTED] wrote:

Hi all!
I am shore someone have writing about it bout I cant find it.
I have a extension that I need to block from making expansive mobil calls.
Everyone else should be aloud to do the calls.

I am shore it is possible to be done sens I had a
commercial asterisk based PBX that I did that on.
However I have switch to Trixbox because I need
some custom functions not supported by the commercial product.
I would appreciate all help.
Regards
Mattias







Adress:
Mattias Andersson
Storskiftesvägen 6
S-145 60 Norsborg

Mobil: +46-70-799 44 41
Email: [EMAIL PROTECTED]
Skype: eskes1


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[asterisk-users] v140 ./configure not finding installed ssl

2007-01-03 Thread snowcrash+asterisk


i'm building asterisk v140 on osx 10.4.8.

openssl is installed in /usr/local/ssl,

which openssl
/usr/local/ssl/bin/openssl
openssl version
OpenSSL 0.9.8d 28 Sep 2006

asterisk is config'd with,

% ./configure  \
  --prefix=/usr/local/asterisk \
  --enable-shared  \
  --enable-static  \
  --with-ssl=/usr/local/ssl

./configure fails @,

(...)
checking for ssl2_connect in -lssl... yes
checking /usr/local/ssl/include/openssl/ssl.h usability... no
checking /usr/local/ssl/include/openssl/ssl.h presence... no
checking for /usr/local/ssl/include/openssl/ssl.h... no
configure: ***
configure: *** It appears that you do not have the ssl development
package installed.
configure: *** Please install it to include OpenSSL support, or
re-run configure
configure: *** without explicitly specifying --with-ssl

despite, checking,

% ls -al /usr/local/ssl/include/openssl/ssl.h
-rw-r--r-- 1 root wheel 79373 Sep 29 09:25
/usr/local/ssl/include/openssl/ssl.h

fwiw, ssl is used widely/successfully elsewhere.

suggestions as to what the issue is?

thanks.
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[asterisk-users] [Announce] Web-MeetMe 3.0.0 RE-released

While preping screenshots I found that a stupid little bug
had slipped past my QA, relating to CDR views.  I've fixed it
and regenerated the tgz file and replaced the broken one
on SF.

If you have downloaded 3.0.0 today, please get a fresh copy.
The bug was small enough and I think I caught it quick enough
to avoid bumping the version up to 3.0.1

Thanks,
Dan
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Re: [asterisk-users] Any quiet 24 port POE switches out there?

2007-01-03 Thread Andres


John French wrote:

I have an upcoming install which places the switch close to some 
employees in a quiet work environment.  Can anyone recommend a quiet 
24 port POE switch?  The Linksys SRW224P behind me right now would be 
objectionable, I'm sure.


 

You will need a fanless switch like the 16 Port Netgear FS116P (8 port 
PoE and the rest are normal)
http://www.tigerdirect.com/applications/searchtools/item-Details.asp?EdpNo=1697260sku=N100-2058 






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Re: [asterisk-users] Error compiling chan_vpb

2007-01-03 Thread DiegoF


On 1/2/07, Kevin P. Fleming [EMAIL PROTECTED] wrote:


DiegoF wrote:
 chan_vpb.o:chan_vpb.cc:(.text+0x4da6): first defined here
 /usr/bin/ld: Warning: size of symbol `load_module' changed from 3274 in
 chan_vpb.o to 3926 in chan_vpb.oo
 collect2: ld devolvi el estado de salida 1
 make[1]: *** [chan_vpb.so] Error 1
 rm chan_vpb.o
 make: *** [channels] Error 2


 hello, if somebody knows like solving this error, to him it will be been
 thankful.

This has been fixed in Subversion branch-1.4; the fix will be included
in the Asterisk 1.4.1 release.
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Thank you very much by the answer. And you know when you will leave this
version?. And if he already left, in where I can find it?.
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Re: [asterisk-users] Any quiet 24 port POE switches out there?

2007-01-03 Thread Anselm Martin Hoffmeister

Am Mittwoch, den 03.01.2007, 16:51 -0600 schrieb John French:
 I have an upcoming install which places the switch close to some
 employees in a quiet work environment.  Can anyone recommend a quiet
 24 port POE switch?  The Linksys SRW224P behind me right now would be
 objectionable, I'm sure.

I had to browse through the list of switches on the market recently for
different features.

Most switches do not feature an acoustic entry in their description.
Even those described as desktop devices... and just with it being named
a desktop device does not necessarily give you a silent device, au
contraire.

All I found was the Nortel BAS220 48T (with 24 out of 48 ports PoE
compliant), nominal 43.8 dB on the datasheet. I do not know that device,
but noise information on PoE switches seems not to be a thing that
manufacturers are proud of.

I guess building a 1u-switch with an included 300W++ power adaptor
requires active cooling, and the smaller the fans, the noisier the
whirl.

Maybe using several, smaller switches could do the trick for you. Brian
Roy mentioned the Netgear FS108p (with external power adaptor,
noiseless) as 8-port device. There is also a larger brother of it, the
FS116P, which also comes with an external power supply, does PoE on
eight of its 16 ports. I have no idea of your overall bandwidth
requirements, but if it is only about phones, 100 MBit should be by far
sufficient for those 20 devices, so you could cascade switches (like
plugging two FS108p into non-PoE ports on a single FS116P, for
instance). This is of course the cheapo way of doing it. Getting a
proper multi-port switch, perhaps even a real brand one would be (ask
the drooling sales droids out there) would be the real deal.

rant
Talking about NetGear switches, I once bought a 24port Gigabit Netgear
switch, noiseless, external PSU. It was meant to be screwed to a table
from below (in a classroom environment) with four metal brackets. The
switch kept crashing (not letting any data through) in that environment,
situation only changed when mounting that switch to a wall (with the
CAT6 cables hanging straight down from the plugs) - temperature problem
(which was not bad enough to go into warranty exchange. Just do not
use the switch in a hot environment. 20°C in a boring computer lab) On
a non-PoE device, with far less than 300W power to go through.
/rant

I personally do not trust wall-wart (a.k.a. external power supply)
switches too much. I do not think it is a problem in principle, but
those devices with internal power supply just tend to be better for me.
YMMV. 

If you find something worthy, with a decent sound, please report back to
the list so others can share a good experience.

BR
Anselm

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[asterisk-users] 1.4 segfaulting when manager client is connected

2007-01-03 Thread Brad Templeton


I was just trying astman with the latest svn trunk from Dec 31.  It
connects, but if I attempt to make a call, asterisk segfaults, but
in pthread_kill in /lib/tls/libpthread.so not in the asterisk code.

Is this something others have seen?  This is with glibc-2.3.4-2
I just upgraded to 2.3.6 (the lastest for Fedora core 3) and it's
the same.

Not much of a traceback, it's happening here:

static struct eventqent *unref_event(struct eventqent *e)
{
struct eventqent *ret = AST_LIST_NEXT(e, eq_next);
if (ast_atomic_dec_and_test(e-usecount)  ret)
pthread_kill(accept_thread_ptr, SIGURG);
return ret;
}


Should I file a bug on this?  I would presume if it's as trivial
to duplicate as it is for me that others would have seen it.

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Re: [asterisk-users] Re: [Announce] Web-MeetMe 3.0.0 released

2007-01-03 Thread Rob Fugina


I'm using the latest svn trunk code.  The app_cbmysql builds, installs, and
loads just fine.  But I don't get any CBMysql application within asterisk.
The only message I get as the module loads is something about finding the
configuration file (successfully).

The database  tables are created, though the DDL scripts provided seem to
require MySQL 5.  The configuration file (cbmysql.conf) is in place and has
been modified to reflect reality (username/pass and socket location).

Any hints about what I must be doing wrong?

Thanks,
Rob
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Re: [asterisk-users] API: how to bridge originated call?

Moises

this sounds great!

three questions if you don't mind:

1. how is this fitting into 1.6?

2. are there some directions I can follow for downloading the right
source and applying your patches?

3. is there a central place for doc on your patches?  (if not I would
be more than happy to write it)

thanks
cy



--- Moises Silva [EMAIL PROTECTED] wrote:

 By the way, Chester, please report results to the bug I sent you, is
 very imortant the users feedback to get this into Asterisk
 
 Regards
 
 On 1/3/07, Moises Silva [EMAIL PROTECTED] wrote:
  I have uploaded a working patch for version 1.2.12.1, and other
 that
  seems to work in Trunk, but few people is reporting results, you
 can
  help to get this into Asterisk, go here:
 
  http://bugs.digium.com/view.php?id=5841
 
  The patch I ported to 1.2.12.1 is working fine, I have tested in my
  servers, is the one called bridge-1.2.12.1.patch, there are other
  ones that say trunk, obviously only work with the trunk version of
  Asterisk.
 
  Kind Regards
 
  On 1/3/07, chester c young [EMAIL PROTECTED] wrote:
   (my pstn calls are coming in thru an upstream asterisk server, so
 the
   called and calling phone number is passed as an extension.)
  
   when caller comes in on 555, he will go to extension 1234
 where he
   will wait for the API to make a call to 999 for him.  how do
 I
   bridge the two calls?
  
   extensions.conf:
  
   ;context where caller comes in
   [caller]
   555,s,1 Answer()
   555,s,n UserEvent(Init) ;this lets me know the connection for
   555
   555,1234,1 Noop(caller waits to be bridged)
   555,1234,2 Background(soothingmusic)
  
   ;context for connection - is this needed?
   [connect]
  
  
   from the API:
  
   (do I need to create a new context/extension first?)
  
   Action: Originate
   Channel: IAX2/upstream/999  -- calls 999222 thru upsteam IAX
   Context: ??
   Exten: ??
   Priority: ??
  
  
  
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 http://www.gnu.org;
 
 
 
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[asterisk-users] ztdummy on 1.6

does anyone know if ztdummy is requires under 1.6 or are they using
Linux' rtc?

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RE: [asterisk-users] [Announce] Web-MeetMe 3.0.0 released

Rob wrote:
 I'm using the latest svn trunk code.  The app_cbmysql builds,
 installs, and loads just fine.  But I don't get any CBMysql 
 application within asterisk.  The only message I get as the 
 module loads is something about finding the configuration file
 (successfully). 
What version of Asterisk?  What does this command return:
*CLI cb mysql status


 The database  tables are created, though the DDL scripts 
 provided seem to require MySQL 5.  The configuration file 
 (cbmysql.conf) is in place and has been modified to reflect 
 reality (username/pass and socket location). 
I would not expect the scripts to be MySQL 5 dependant, but
they were contributed by another developer, so perhaps they are.

How are you determining that you do not have the application?

What Does this command report:
*CLI show application CBMySQL

Dan
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[asterisk-users] Re: asterisk-users Digest, Vol 30, Issue 4

2007-01-03 Thread Edwin Groothuis

On Tue, Jan 02, 2007 at 03:17:35PM -0700, [EMAIL PROTECTED] wrote:
 Has anyone made this combination work together?  I've tried everything 
 and can't seem to get it work right.  It all compiles fine, but when 
 rxfax is called, I get an unknown symbol error.  From my reading, 
 everything points to me having multiple copies of spandsp and it's maybe 
 calling the wrong one.

After the complete compile of asterisk, I jump into the apps/
directory and do this:

[~/asterisk/1.4/apps] [EMAIL PROTECTED]gcc -o app_rxfax.so -shared 
-Xlinker -x app_rxfax.o -lspandsp

After that, with ldd on app_rxfax.so you can confirm that is is
being linked.

Edwin

-- 
Edwin Groothuis  |Personal website: http://www.mavetju.org
[EMAIL PROTECTED]|  Weblog: http://weblog.barnet.com.au/edwin/
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Re: [asterisk-users] Any quiet 24 port POE switches out there?

2007-01-03 Thread Angel Heart

Hi,

I am using these model from HP ProCurve

http://www.hp.com/rnd/products/switches/switch2600series/features.htm?jumpid=reg_R1002_USEN

http://www.hp.com/rnd/products/switches/ProCurve_Switch_3500yl-5400zl_Series/features.htm?jumpid=reg_R1002_USEN

Regards,

Angel

Anselm Martin Hoffmeister [EMAIL PROTECTED] wrote:
Am Mittwoch, den 03.01.2007, 16:51 -0600 schrieb John French:
I have an upcoming install which places the switch close to some
employees in a quiet work environment. Can anyone recommend a quiet
24 port POE switch? The Linksys SRW224P behind me right now would be
objectionable, I'm sure.

I had to browse through the list of switches on the market recently for
different features.

Most switches do not feature an acoustic entry in their description.
Even those described as desktop devices... and just with it being named
a desktop device does not necessarily give you a silent device, au
contraire.

All I found was the Nortel BAS220 48T (with 24 out of 48 ports PoE
compliant), nominal 43.8 dB on the datasheet. I do not know that device,
but noise information on PoE switches seems not to be a thing that
manufacturers are proud of.

I guess building a 1u-switch with an included 300W++ power adaptor
requires active cooling, and the smaller the fans, the noisier the
whirl.

Maybe using several, smaller switches could do the trick for you. Brian
Roy mentioned the Netgear FS108p (with external power adaptor,
noiseless) as 8-port device. There is also a larger brother of it, the
FS116P, which also comes with an external power supply, does PoE on
eight of its 16 ports. I have no idea of your overall bandwidth
requirements, but if it is only about phones, 100 MBit should be by far
sufficient for those 20 devices, so you could cascade switches (like
plugging two FS108p into non-PoE ports on a single FS116P, for
instance). This is of course the cheapo way of doing it. Getting a
proper multi-port switch, perhaps even a real brand one would be (ask
the drooling sales droids out there) would be the real deal.

Talking about NetGear switches, I once bought a 24port Gigabit Netgear
switch, noiseless, external PSU. It was meant to be screwed to a table
from below (in a classroom environment) with four metal brackets. The
switch kept crashing (not letting any data through) in that environment,
situation only changed when mounting that switch to a wall (with the
CAT6 cables hanging straight down from the plugs) - temperature problem
(which was not bad enough to go into warranty exchange. Just do not
use the switch in a hot environment. 20°C in a boring computer lab) On
a non-PoE device, with far less than 300W power to go through.

I personally do not trust wall-wart (a.k.a. external power supply)
switches too much. I do not think it is a problem in principle, but
those devices with internal power supply just tend to be better for me.
YMMV.

If you find something worthy, with a decent sound, please report back to
the list so others can share a good experience.

BR
Anselm

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Re: [asterisk-users] API: how to bridge originated call?


1. how is this fitting into 1.6?

1.6? do you mean 1.4? AFAIK the most advanced Asterisk development goes in 1.4


2. are there some directions I can follow for downloading the right
source and applying your patches?

Nope, but is not hard at all. All the patches include the version in
its name. But you need to learn how to use the patch command in
Linux ( man patch ) and probably how to download code using SVN.


3. is there a central place for doc on your patches?  (if not I would
be more than happy to write it)

Not really, since they are still under development and approval of
Digium in bugtracker, how they work can change. But in the bugtracker
there are some places where is explained how is supposed to work and
be used. Check the link I sent you in my last email.

Regards

Moises
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Re: [asterisk-users] API: how to bridge originated call?