date:20060503

Hello List,
I know this has been brought up many times but I
wanted to know if anyone had any expirience in the
following. I setting up several voice mail systems.
Each one is going to have a TDM400P. Two FXO for
people to leave messages and two FXS for POTS phones
so people can listen. Anyone know if there are any
simple specific dell models that will handle this
without a problem ?

Thanks.

Dovid

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Re: [Asterisk-Users] Sangoma Card Question

Greetings,
There is no 'flashing' going on, though. Just hanging up.Perhaps
the Sangoma card is somehow creating a flash on the line? I guess I
could double check the configuration to make sure there is no
callwaiting, etc configured on it.But, to my knowledge they are
just hanging up the phone when they are done talking, and it
immediately rings back in... but the line is dead.

On 5/2/06, stevanus [EMAIL PROTECTED] wrote:

Hi Matt,

I guess this is the problem within asterisk which wrongly assume the
hangup as on-hold call.
Do you/your staffs/your customer hang up the phone so quickly that
asterisk mistakenly belief that the act is for call waiting?
As we know to do some call waiting we just flash the hook swiftly and
the other person will hear a music-on-hold.
Then if we put the handset down then the phone will ringing once each a
couple seconds to remind us that there is call waiting on the phone ;)
To avoid this behaviour, try to flash the hook a little longer when hang
up the phone (about 2 seconds will be enough)..

Regards,

Stevanus

Matt wrote:

By the system you mean the phone company? Or asterisk?

So what you are saying is I hang up... the sangoma hangups... but
the phone company sees it as a flash... then says.. HEY DUDE! YOU
JUST HUNG UP ON YOUR CALLER.. Here they are back *ring*.

On 5/2/06, Melcon Moraes [EMAIL PROTECTED] wrote:

Maybe some kind of callwaiting/threewaycalling activated on that? The
system is identifying the hang up as a flash.

-Original Message-
From: Matt [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Cc:
Sent: Tue, 2 May 2006 16:30:56 -0400
Delivered: Tue, 02 May 2006 17:28:33
Subject:[Asterisk-Users] Sangoma Card Question

Hi,
I have a Sangoma 200A (I think that's the model #) analog 4 port card.
It works great... however almost everytime after someone hangs up a
call they were on.. the system rings the call back in, as though it
were a new call coming in. When they pickup no one is there.

Can anyone suggest why this is happening, and how I can make it stop?
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RE: [Asterisk-Users] Simple Dell Computers

The problem with the Dell's is their incompatibility with the TigerJet
Chipset, I have had problems with the SC 4X0 line of machines, they are
known to have issues.

Alex


 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Dovid Bender
 Sent: Wednesday, May 03, 2006 8:10 AM
 To: asterisk-users@lists.digium.com
 Subject: [Asterisk-Users] Simple Dell Computers
 
 Hello List,
 I know this has been brought up many times but I
 wanted to know if anyone had any expirience in the
 following. I setting up several voice mail systems.
 Each one is going to have a TDM400P. Two FXO for
 people to leave messages and two FXS for POTS phones
 so people can listen. Anyone know if there are any
 simple specific dell models that will handle this
 without a problem ?
 
 Thanks.
 
 Dovid
 
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Re: [Asterisk-Users] asterisk with Dialogic BRI /2VFD

2006-05-03 Thread Nardis Dome


Hi Tom,

thx for the answer...

--- Tom [EMAIL PROTECTED] wrote:

 richard Coco wrote:
  Hi all,
 
  i have an Asterisk box with an Eicon 4BRI with
  chan_capi-cm and every thing works fine. We now
 plan
  to install a new Asterisk using a Dialogic
 BRI/2VFD.
  Is the Dialogic card supported and can i use
  chan_capi-cm? Has anyone managed to install this
 card?
  Unfortunately i was unable to find documentation
 about
  Asterisk with Dialogic?
 
  thx in advance for your input!!!
 

 Richard,
 
 i think the only dialogic cards that with work are
 the jct models.  then 
 i think you need to buy drivers from digium
 hope this helps
 
 Tom
 
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[Asterisk-Users] Limit on number of SIP channels?

2006-05-03 Thread Chris Hastie

Can anyone tell me what limits the total number of SIP channels available? I'm
just setting up an Asterisk system and have found that I seem to be limited to
about 5 - is there a configuration option somewhere? call-limit is not set in
sip.conf.

The symptoms:

If I have two incoming SIP connections, each connected to a local SIP phone, an
attempt to make a further inbound SIP call gets me to asterisk, but trying to
put the call through to a local SIP phone results in a channel unavailable
error. Similar things happen when I try to call out. An inbound IAX call can be
put through to the SIP phone. But having done that, further inbound SIP calls do
nothing at all - no apparent response from *. All the inbound calls are using
GSM and there is plenty of bandwidth left. It just seems like I can get to five
SIP connections and no more.

-- 
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Re: [Asterisk-Users] brittle IAX connections ?

2006-05-03 Thread Rich Adamson



I have several asterisk 1.2.7.1 servers connected through iax2 and often 
the local asterisk would no longer see the remote one, even thought the 
link is high quality and the ping is perfect.


Is there some issues to take into account about IAX2 connections? 

Is asterisk's DNS resolution too fragile and should I use raw IP's in my 
configs?


The DNS resolution is somewhat fragile and/or incomplete as has been 
stated previously on the list. Most notably, if asterisk cannot get dns 
resolution the system basically hangs. The workaround is to establish a 
dns caching server on the * box. Second, asterisk code does not properly 
handle dns records that contain more then one IP address; it uses only 
the first entry found within the dns response.


Using hard coded IP addresses is one way to address the issue if that 
works for you.


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[Asterisk-Users] Re: brittle IAX connections ?

2006-05-03 Thread Louis-David Mitterrand

On Wed, May 03, 2006 at 07:48:37AM -0500, Rich Adamson wrote:
 I have several asterisk 1.2.7.1 servers connected through iax2 and often 
 the local asterisk would no longer see the remote one, even thought the 
 link is high quality and the ping is perfect.
 
 Is there some issues to take into account about IAX2 connections? 
 
 Is asterisk's DNS resolution too fragile and should I use raw IP's in my 
 configs?
 
 The DNS resolution is somewhat fragile and/or incomplete as has been 
 stated previously on the list. Most notably, if asterisk cannot get dns 
 resolution the system basically hangs. The workaround is to establish a 
 dns caching server on the * box. Second, asterisk code does not properly 
 handle dns records that contain more then one IP address; it uses only 
 the first entry found within the dns response.
 
 Using hard coded IP addresses is one way to address the issue if that 
 works for you.

Thanks Rich,

It confirms what I suspected. Using hardcoded IP's indeed aleviates the 
problem.

Cheers,
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Re: [Asterisk-Users] Under which project , auto-dial feature comes

2006-05-03 Thread John Joseph


--- Eric \ManxPower\ Wieling [EMAIL PROTECTED]
wrote:

 
  Thanks Eric
 
 Asterisk does not have an auto-dial feature.  If
 you describe the 
 feature you are having problems with, perhaps we can
 tell you what term 
 we use for that feature.
 
 
Thanks Eric   
 I have problem , when keeping sample.call in
/var/spool/asterisk/outgoing/
my applications get executed before
the caller attend the calls , this happens only when I
call outside no , ie when I use something like  
Channel: ZAP/1/050745














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RE: [Asterisk-Users] Hi...Please help me

2006-05-03 Thread William Piper

Wouldn't it be easier to replace the callername to the exten.

example:

exten = _x.,1,SetCallerIDname(${EXTEN})
exten = _x.,2,SetCallerIDnum(${CALLERIDNUM})
exten = _x.,3,dial,SIP/number

That way, the Caller Name would show the extension it is ringing and the
callerid will still show the calling party.

Now you don't need the softphone to do it... just a phone with callerid.

bp

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew
Kohlsmith
Sent: Tuesday, May 02, 2006 5:00 PM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Hi...Please help me

On Tuesday 02 May 2006 16:42, hugolivude wrote:
 We share SIP phones at the office in a 1:4 ratio.  You're probably
 asking - how do you know when a ringing phone is for you?  Well,
 everyone in our office gets an XLite softphone, and I direct calls to
 make BOTH the SIP phone AND the XLite ring.  If your XLite pops up,
 you know that ring phone is for you.

That seems to be humongous overkill... why not just use any of the caller ID

popup apps instead of running that behemoth X-Lite?  If the popup comes up, 
the phone's for you.

-A.
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__ NOD32 1.1443 (20060314) Information __

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Re: [Asterisk-Users] mISDN: No DID/extension information returns busy to caller

2006-05-03 Thread Ralf Schlatterbeck

On Fri, Apr 28, 2006 at 02:10:10PM +0300, Dmitry Ivanov wrote:
 On Friday 28 April 2006 11:35, Ralf Schlatterbeck wrote:
  I don't see the call at all in asterisk.
 
 Maybe your telco does not route these calls with incomplete number to 
 you?
Oh yes, it does. With chan_capi (instead of chan_misdn) I'm seeing these
calls, but chan_capi doesn't work for me for other reasons.

-- 
Ralf Schlatterbeck
email: [EMAIL PROTECTED] FAX: +43/2243/26465/23

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Re: SV: [Asterisk-Users] How does asterisk behave when multiple phonesare logged in on a single SIP/account?

2006-05-03 Thread Jerry Jones


Perhaps a setgroup/checkgroup before your dial command?


On May 3, 2006, at 1:24 AM, Arne Morten Johansen wrote:

Yeah I do use ring groups at the moment. But the problem is that I  
can’t control “the flow”.


Let’s take your example.



dial(SIP/dev1SIP/dev2SIP/dev3)



If I dial these 3 numbers, and dev2 is already one the phone. How  
do I check for that? I only want one of the three phones active at  
the time. But if no telephone is busy, they all should ring until  
the call is picked up (this is what it always does now). Is this  
possible in asterisk?




Thanks



Fra: [EMAIL PROTECTED] [mailto:asterisk-users- 
[EMAIL PROTECTED] På vegne av Gary Richardson

Sendt: 2. mai 2006 16:51
Til: Asterisk Users Mailing List - Non-Commercial Discussion
Emne: Re: [Asterisk-Users] How does asterisk behave when multiple  
phonesare logged in on a single SIP/account?




The last sip device to register gets the call. The way around this  
is to have your sip devices register under different accounts and  
create a ring group (dial(SIP/dev1SIP/dev2SIP/devN))


AFAIK, there isn't a reliable method of determining if a sip device  
is busy other than calling it.


On 5/1/06, Arne Morten Johansen [EMAIL PROTECTED] wrote:

Hi.

How does this work?

What if this SIP/account was a member (agent) of a queue?

Ex: dial(SIP/account,20,tT). Would the dialstatus be set as busy  
when one of the phones is actively talking, or will the other  
phones continue to ring?


You may have seen my other submissions to this list. I'm looking  
for a way to make the other phones in a group unavailable when one  
of them is busy. Because one person will have multiple phones.


Thanks

Arne Morten Johansen.


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Re: [Asterisk-Users] Sangoma Card Question

Well I double checked, and we do not have any callwaiting or
three-way-calling on those lines.

On 5/3/06, Matt [EMAIL PROTECTED] wrote:

On 5/2/06, stevanus [EMAIL PROTECTED] wrote:
Hi Matt,

Regards,

Stevanus

Matt wrote:

By the system you mean the phone company? Or asterisk?

So what you are saying is I hang up... the sangoma hangups... but
the phone company sees it as a flash... then says.. HEY DUDE! YOU
JUST HUNG UP ON YOUR CALLER.. Here they are back *ring*.

On 5/2/06, Melcon Moraes [EMAIL PROTECTED] wrote:

Maybe some kind of callwaiting/threewaycalling activated on that? The
system is identifying the hang up as a flash.

Can anyone suggest why this is happening, and how I can make it stop?
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Re: [Asterisk-Users] Queue reporting seems broken.

2006-05-03 Thread Johann


The wiki page doesn't mention the other fields in the file.

TIMESTAMP:UNIQUEID:QUEUE:AGENT:ACTION:ARG1:ARG2:ARG3

Depending on the action some of the fields may contain 'NONE' instead.  You will 
need to use the AGENT field to match who the connect and 
completeagent/completecaller messages are for.


Keep in mind the UNIQUEID field will be the same for a caller as they go 
through the queue.  So the enterqueue, connect, complete actions will have the same.



--johann

Thermal Wetland wrote:

I am trying to figure out which one of our agents is answering the calls.

According to http://www.voip-info.org/wiki/view/Asterisk+log+queue_log 
the only time the queue_log puts the channel (agent) is during logoff  
logon.


There is the connect  completeagent message, but it doesn't show which 
channel (agent) answered the phone.


I can't even figure it our cross referencing the CDR records, the CDR 
record only has the queue number.


Is there a way around this?

Aloha,
Matt




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[Asterisk-Users] Can I recreate a Fax from a recorded file?









This is a very KGB / NSA / InterPOL / CIA type question, but
if I have a recorded file (G.711, no compression) can I feed it into standard
in of an application and have it recreate the fax that was send?





I dont know enough about the Fax handshaking to
understand this.








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[Asterisk-Users] RE: Huawei EP201S

2006-05-03 Thread bram kortleven

Did you check the Grandstream BudgeTone series?
They're really cost-effective, and have good sound quality, and are perfectly 
supported by Asterisk. We used them for over 2 years now, and no problems to be 
mentioned.
In belgium, they cost around EUR80, so that's about US$95-100, I think. (don't 
shoot me if I'm wrong!)
This is the link:
 
http://www.grandstream.com/y-bt100.htm

Let me know what you're going to choose and how it ends up

Bram

 

 


Has anybody used Huawei EP201S IP phone? I need 9 SIP phones that cost around 
100USD, and those phones are one of options.

Can anybody suggest anything else that costs around 100USD?



--
Tomislav Parhina
Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21)495148
SIP: [EMAIL PROTECTED]
e-mail: tparcina#lama.hr
http://www.lama.hr http://www.lama.hr/ 

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Re: [Asterisk-Users] asterisk intergration in third party web application

2006-05-03 Thread ChaosMedia WebDev


thx that's what i was looking for..

it looks nice, it seems selected parts of the config can be stored in 
the database for realtime purposes..


but on the other hand i've checked freepbx doc and it seems it doesn't 
use asterisk reatime feature and maybe won't work when realtime is 
enabled, so can you tell me if there are any web frontals that use 
asterisk RT so i can both manage asterisk from the web and have my web 
app do modifications on its own..


thx again

Webdev

Roman Yeryomin wrote:


On Wednesday 03 May 2006 13:19, ChaosMedia  WebDev wrote::
 


In fact i was expecting asterisk to have some kind of configuration that
would enable mysql but i can't seem to find anything like that, like it
   



hope this will help
http://www.voip-info.org/wiki/view/Asterisk+RealTime
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Re: [Asterisk-Users] Unicall MFC/R2 B3,B4 and clear back

2006-05-03 Thread Steve Underwood


Hi Denis,

This is a chan_unicall.c issue. The Unicall library provides for full 
control of the call, but at one time I had problems with quirky 
behaviour from Asterisk, and made chan_unicall.c treat the call is a 
simplistic way. It is fairly easy to change chan_unicall.c to have the 
call properly. I will send you a version of that file to try.


Steve


Dennis Nacino wrote:


Hi All,


I have an R2 installation still undergoing testings, during the test I notice 
that the Unicall
always respond B6 to a II-1 (from a forward switch). Except, for a DNIS that 
can't be found in the
dial plan, in this case it respond with B5. My real problem is, the call will 
be terminate on a
Cisco 7206 with ISDN/PRI thru SIP. If the Called number is busy or the Cisco 
7206 is busy or
congested, it seems there's no way for Unicall to issue B3 or B4 since its 
already on accepted
state. Please see the log below;

May  3 12:51:11 WARNING[11325]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/17  - 1 on 
[2/   2/Group B   /Go to grp II ]

May  3 12:51:11 WARNING[11325]: chan_unicall.c:2644 handle_uc_event: Unicall/17 
event Offered
May  3 12:51:11 WARNING[11325]: chan_unicall.c:2699 handle_uc_event: CRN 32782 
- Offered on
channel 0 (ANI: 09797280105, DNIS: 0015107973287, Cat: 0)
May  3 12:51:11 WARNING[11325]: chan_unicall.c:627 unicall_report: MFC/R2 
UniCall/17 Call
control(4)
May  3 12:51:11 WARNING[11325]: chan_unicall.c:627 unicall_report: MFC/R2 
UniCall/17 Accept call
May  3 12:51:11 WARNING[11325]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/17 6 on  - 
[2/   4/Group B   /Go to grp II ]

May  3 12:51:11 WARNING[11325]: chan_unicall.c:627 unicall_report: MFC/R2 
UniCall/17  - 1 off
[2/   4/Group B   /Accepted Paid]
May  3 12:51:11 WARNING[11325]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/17 6 off - 
[2/   4/Group B   /Accepted Paid]

May  3 12:51:11 WARNING[11325]: chan_unicall.c:627 unicall_report: MFC/R2 
UniCall/17 Answer guard
expired
May  3 12:51:11 WARNING[11325]: chan_unicall.c:2644 handle_uc_event: Unicall/17 
event Accepted
May  3 12:51:11 WARNING[11325]: chan_unicall.c:627 unicall_report: MFC/R2 
UniCall/17 Channel gains
   -- Executing Dial(UniCall/17-1, SIP/aaa.bbb.ccc.ddd/15107973287|45||) in 
new stack
   -- Called aaa.bbb.ccc.ddd/15107973287
   -- Got SIP response 486 Busy here back from aaa.bbb.ccc.ddd
   -- SIP/aaa.bbb.ccc.ddd-7bad is busy
 == Everyone is busy/congested at this time (1:1/0/0)
 == Auto fallthrough, channel 'UniCall/17-1' status is 'BUSY'
May  3 12:51:20 WARNING[12011]: chan_unicall.c:2441 unicall_indicate: 
unicall_indicate 5
May  3 12:51:31 WARNING[12011]: chan_unicall.c:627 unicall_report: MFC/R2 
UniCall/17 Channel gains
May  3 12:51:31 WARNING[12011]: chan_unicall.c:627 unicall_report: MFC/R2 
UniCall/17 Channel
switching
May  3 12:51:31 WARNING[12011]: chan_unicall.c:627 unicall_report: MFC/R2 
UniCall/17 Call
control(6)
May  3 12:51:31 WARNING[12011]: chan_unicall.c:627 unicall_report: MFC/R2 
UniCall/17 Drop
call(cause=User busy [17])
May  3 12:51:31 WARNING[12011]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/17 1101  - 
[1/  20/Group B   /Accepted Paid]

   -- Hungup 'UniCall/17-1'

The worst part of it, the forward switch, look lost and never respond to that 
clearback thus
never release the channel. 
As a another test I called an extension with Busy as an asterisk application, it still respond

with B6.
May  3 13:21:50 WARNING[11325]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/17  - 1 on 
[2/   2/Group B   /Go to grp II ]

May  3 13:21:50 WARNING[11325]: chan_unicall.c:2644 handle_uc_event: Unicall/17 
event Offered
May  3 13:21:50 WARNING[11325]: chan_unicall.c:2699 handle_uc_event: CRN 32783 
- Offered on
channel 0 (ANI: 09797280105, DNIS: 006321234569, Cat: 0)
May  3 13:21:50 WARNING[11325]: chan_unicall.c:627 unicall_report: MFC/R2 
UniCall/17 Call
control(4)
May  3 13:21:50 WARNING[11325]: chan_unicall.c:627 unicall_report: MFC/R2 
UniCall/17 Accept call
May  3 13:21:50 WARNING[11325]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/17 6 on  - 
[2/   4/Group B   /Go to grp II ]

May  3 13:21:50 WARNING[11325]: chan_unicall.c:627 unicall_report: MFC/R2 
UniCall/17  - 1 off
[2/   4/Group B   /Accepted Paid]
May  3 13:21:50 WARNING[11325]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/17 6 off - 
[2/   4/Group B   /Accepted Paid]

May  3 13:21:50 WARNING[11325]: chan_unicall.c:627 unicall_report: MFC/R2 
UniCall/17 Answer guard
expired
May  3 13:21:50 WARNING[11325]: chan_unicall.c:2644 handle_uc_event: Unicall/17 
event Accepted
May  3 13:21:50 WARNING[11325]: chan_unicall.c:627 unicall_report: MFC/R2 
UniCall/17 Channel gains
   -- Executing Busy(UniCall/17-1, 8) in new stack
May  3 13:21:50 WARNING[12259]: chan_unicall.c:2441 unicall_indicate: 
unicall_indicate 5
 == Spawn extension (nextel-r2, 006321234569, 1) exited

RE: [Asterisk-Users] Can I recreate a Fax from a recorded file?

2006-05-03 Thread Technical Support




Interesting question! If you have the audio in only 
(assuming it was a fax received) or audio out only (assuming it was a fax sent), 
and you pair with an identical fax machine to the original (assuming it responds 
exactly the same in terms of handshakes, speeds, ECM, etc) then it might 
work. I assume you have a lot of time on your hands :) 


MD


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Alexander 
LopezSent: Wednesday, May 03, 2006 9:33 AMTo: Asterisk 
Users Mailing List - Non-Commercial DiscussionSubject: 
[Asterisk-Users] Can I recreate a Fax from a recorded file?


This is a very KGB / NSA / InterPOL 
/ CIA type question, but if I have a recorded file (G.711, no compression) can I 
feed it into standard in of an application and have it recreate the fax that was 
send?


I dont know enough about the Fax 
handshaking to understand this.

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RE: RE: [Asterisk-Users] Dial Option wW picking up the *1 is a bit flaky

2006-05-03 Thread Guillaume de Lafontaine



 -Original Message-
 From: Mark Ackroyd [EMAIL PROTECTED]
 To: Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com
 CC: 
 Subject: RE: [Asterisk-Users] Dial Option wW picking up the *1 is a bit 
 flaky
 Sent: mer., 03 mai 2006 09:15:13 GMT
 Received: mer., 03 mai 2006 09:18:20 GMT
 Read: mer., 03 mai 2006 10:07:50 GMT
  When I hit *1 in my system, I got a beep to let me know that the
  recording started. Is this not happenning to you?
 
 No ! , it doesn't.  Most of the time it doesn't pick up the *1.
 

Hi!

Just make sure to hit *1 very quickly... I mean with minimum delay between * 
and 1.

Guillaume

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Re: [Asterisk-Users] Sangoma Card Question

2006-05-03 Thread Rich Adamson

I'll jump in here to suggest the problem is likely a dialplan issue and 
likely has something to do with how the hangup is being treated within 
the dialplan. The reasoning behind that is that I don't have any issues 
whatsoever with the A200D card, and I've not heard of anyone else with 
similar problems. They just work.



Well I double checked, and we do not have any callwaiting or
three-way-calling on those lines.

On 5/3/06, Matt [EMAIL PROTECTED] wrote:

Greetings,
There is no 'flashing' going on, though.   Just hanging up.Perhaps
the Sangoma card is somehow creating a flash on the line?  I guess I
could double check the configuration to make sure there is no
callwaiting, etc configured on it.But, to my knowledge they are
just hanging up the phone when they are done talking, and it
immediately rings back in... but the line is dead.

On 5/2/06, stevanus [EMAIL PROTECTED] wrote:
 Hi Matt,

 I guess this is the problem within asterisk which wrongly assume the
 hangup as on-hold call.
 Do you/your staffs/your customer  hang up the phone so quickly that
 asterisk mistakenly belief that the act is for call waiting?
 As we know to do some call waiting we just flash the hook swiftly and
 the other person will hear a music-on-hold.
 Then if we put the handset down then the phone will ringing once each a
 couple seconds to remind us that there is call waiting on the phone ;)
 To avoid this behaviour, try to flash the hook a little longer when 
hang

 up the phone (about 2 seconds will be enough)..

 Regards,

 Stevanus

 Matt wrote:

  By the system you mean the phone company?  Or asterisk?
 
  So what you are saying is I hang up... the sangoma hangups... but
  the phone company sees it as a flash... then says.. HEY DUDE!  YOU
  JUST HUNG UP ON YOUR CALLER.. Here they are back *ring*.
 
  ?
 
 
  On 5/2/06, Melcon Moraes [EMAIL PROTECTED] wrote:
 
  Maybe some kind of callwaiting/threewaycalling activated on that? 
The

  system is identifying the hang up as a flash.
 
 
   -Original Message-
  From:   Matt [EMAIL PROTECTED]
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  asterisk-users@lists.digium.com
  Cc:
  Sent:  Tue, 2 May 2006 16:30:56 -0400
  Delivered:  Tue,  02 May 2006 17:28:33
  Subject:[Asterisk-Users] Sangoma Card Question
 
  Hi,
  I have a Sangoma 200A (I think that's the model #) analog 4 port 
card.

   It works great... however almost everytime after someone hangs up a
  call they were on.. the system rings the call back in, as though it
  were a new call coming in.  When they pickup no one is there.
 
  Can anyone suggest why this is happening, and how I can make it 
stop?


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RE: [Asterisk-Users] Dial Option wW picking up the *1 is a bit flaky

2006-05-03 Thread Adam Hatia

Both the SIP phone, and asterisk (when the call is via our IAX line) use
alaw. If the external call is via the Zap interface, it's equally
unreliable.

Where can I see what DTMF mode is being used by * and the SIP phone?


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric
ManxPower Wieling
Sent: 03 May 2006 12:22
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Dial Option wW picking up the *1 is a bit
flaky

Sounds like you are using an inband codec and not using ulaw or alaw. 
If you are not using alaw or ulaw, then you need to use either INFO or 
RFC2844 DTMF.   Just remember that both Asterisk and the phone device 
MUST be using the same DTMF mode.

Adam Hatia wrote:
 I have exactly the same problem - attended transfer (*2) is the same -
 sometimes asterisk just generates the DTMF tome for * followed by the
number
 instead of interpreting the command. Are there any DTMF configuration
 settings that can be tweaked? How does Asterisk decide is a key sequence
is
 a command or needs to be transferred as DTMF on the line?
 
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Mark Ackroyd
 Sent: 03 May 2006 10:15
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: RE: [Asterisk-Users] Dial Option wW picking up the *1 is a bit
 flaky
 
 When I hit *1 in my system, I got a beep to let me know that the
 recording started. Is this not happenning to you?
 
 No ! , it doesn't.  Most of the time it doesn't pick up the *1.


-- 
Now accepting new clients in Birmingham, Atlanta, Huntsville, 
Chattanooga, and Montgomery.
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Re: [Asterisk-Users] Sangoma Card Question


Maybe this will help?   If a call comes in.. and hangs up before
someone picks up a phone... the phones will continue to ring, but then
you pick them up and they are dead.   Any thoughts on that one?   It's
like the person hung up... but asterisk continues to ring the lines.

On 5/3/06, Rich Adamson [EMAIL PROTECTED] wrote:

I'll jump in here to suggest the problem is likely a dialplan issue and
likely has something to do with how the hangup is being treated within
the dialplan. The reasoning behind that is that I don't have any issues
whatsoever with the A200D card, and I've not heard of anyone else with
similar problems. They just work.

 Well I double checked, and we do not have any callwaiting or
 three-way-calling on those lines.

 On 5/3/06, Matt [EMAIL PROTECTED] wrote:
 Greetings,
 There is no 'flashing' going on, though.   Just hanging up.Perhaps
 the Sangoma card is somehow creating a flash on the line?  I guess I
 could double check the configuration to make sure there is no
 callwaiting, etc configured on it.But, to my knowledge they are
 just hanging up the phone when they are done talking, and it
 immediately rings back in... but the line is dead.

 On 5/2/06, stevanus [EMAIL PROTECTED] wrote:
  Hi Matt,
 
  I guess this is the problem within asterisk which wrongly assume the
  hangup as on-hold call.
  Do you/your staffs/your customer  hang up the phone so quickly that
  asterisk mistakenly belief that the act is for call waiting?
  As we know to do some call waiting we just flash the hook swiftly and
  the other person will hear a music-on-hold.
  Then if we put the handset down then the phone will ringing once each a
  couple seconds to remind us that there is call waiting on the phone ;)
  To avoid this behaviour, try to flash the hook a little longer when
 hang
  up the phone (about 2 seconds will be enough)..
 
  Regards,
 
  Stevanus
 
  Matt wrote:
 
   By the system you mean the phone company?  Or asterisk?
  
   So what you are saying is I hang up... the sangoma hangups... but
   the phone company sees it as a flash... then says.. HEY DUDE!  YOU
   JUST HUNG UP ON YOUR CALLER.. Here they are back *ring*.
  
   ?
  
  
   On 5/2/06, Melcon Moraes [EMAIL PROTECTED] wrote:
  
   Maybe some kind of callwaiting/threewaycalling activated on that?
 The
   system is identifying the hang up as a flash.
  
  
-Original Message-
   From:   Matt [EMAIL PROTECTED]
   To: Asterisk Users Mailing List - Non-Commercial Discussion
   asterisk-users@lists.digium.com
   Cc:
   Sent:  Tue, 2 May 2006 16:30:56 -0400
   Delivered:  Tue,  02 May 2006 17:28:33
   Subject:[Asterisk-Users] Sangoma Card Question
  
   Hi,
   I have a Sangoma 200A (I think that's the model #) analog 4 port
 card.
It works great... however almost everytime after someone hangs up a
   call they were on.. the system rings the call back in, as though it
   were a new call coming in.  When they pickup no one is there.
  
   Can anyone suggest why this is happening, and how I can make it
 stop?

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[Asterisk-Users] LDAPget

2006-05-03 Thread Joao Pereira


Hello to all
Im using [EMAIL PROTECTED] 2.7 and I would like to do LDAP querys.
Can I simply use LDAPget or do I need to install Asterisk::LDAP from 
Alkaloid Networks?


Thanks
Joao Pereira
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Re: [Asterisk-Users] Sangoma Card Question

2006-05-03 Thread Rich Adamson

There is always a delay associated with the pstn line hangup, asterisk 
sensing the hangup (which is really a delay in the Central Office), and 
asterisk sending the sip packet to the sip phone indicating the call was 
hung up. Not unusual for a sip phone to ring one or two times after the 
pstn caller hung up. Whether its one ring or two rings is 100% dependent 
on how quickly the Central Office disconnects the line. In my case, the 
disconnect arrives about 5 to 8 seconds after a pstn caller hangs up.


Based only on your description of events, I would doubt the above issue 
has anything to do with the issue you described in earlier posts.


To help understand what might be happening, I'd suggest inserting some 
NoOp dialplan statements to see if you can track down the original 
hangup issues.  If you're using an 'exten = h, ' statement, try 
inserting a step something like 'exten = h,1,NoOp,step one' to help 
detect whether this is a dialplan issue.




Maybe this will help?   If a call comes in.. and hangs up before
someone picks up a phone... the phones will continue to ring, but then
you pick them up and they are dead.   Any thoughts on that one?   It's
like the person hung up... but asterisk continues to ring the lines.

On 5/3/06, Rich Adamson [EMAIL PROTECTED] wrote:

I'll jump in here to suggest the problem is likely a dialplan issue and
likely has something to do with how the hangup is being treated within
the dialplan. The reasoning behind that is that I don't have any issues
whatsoever with the A200D card, and I've not heard of anyone else with
similar problems. They just work.

 Well I double checked, and we do not have any callwaiting or
 three-way-calling on those lines.

 On 5/3/06, Matt [EMAIL PROTECTED] wrote:
 Greetings,
 There is no 'flashing' going on, though.   Just hanging up.Perhaps
 the Sangoma card is somehow creating a flash on the line?  I guess I
 could double check the configuration to make sure there is no
 callwaiting, etc configured on it.But, to my knowledge they are
 just hanging up the phone when they are done talking, and it
 immediately rings back in... but the line is dead.

 On 5/2/06, stevanus [EMAIL PROTECTED] wrote:
  Hi Matt,
 
  I guess this is the problem within asterisk which wrongly assume the
  hangup as on-hold call.
  Do you/your staffs/your customer  hang up the phone so quickly that
  asterisk mistakenly belief that the act is for call waiting?
  As we know to do some call waiting we just flash the hook swiftly 
and

  the other person will hear a music-on-hold.
  Then if we put the handset down then the phone will ringing once 
each a
  couple seconds to remind us that there is call waiting on the 
phone ;)

  To avoid this behaviour, try to flash the hook a little longer when
 hang
  up the phone (about 2 seconds will be enough)..
 
  Regards,
 
  Stevanus
 
  Matt wrote:
 
   By the system you mean the phone company?  Or asterisk?
  
   So what you are saying is I hang up... the sangoma 
hangups... but

   the phone company sees it as a flash... then says.. HEY DUDE!  YOU
   JUST HUNG UP ON YOUR CALLER.. Here they are back *ring*.
  
   ?
  
  
   On 5/2/06, Melcon Moraes [EMAIL PROTECTED] wrote:
  
   Maybe some kind of callwaiting/threewaycalling activated on that?
 The
   system is identifying the hang up as a flash.
  
  
-Original Message-
   From:   Matt [EMAIL PROTECTED]
   To: Asterisk Users Mailing List - Non-Commercial Discussion
   asterisk-users@lists.digium.com
   Cc:
   Sent:  Tue, 2 May 2006 16:30:56 -0400
   Delivered:  Tue,  02 May 2006 17:28:33
   Subject:[Asterisk-Users] Sangoma Card Question
  
   Hi,
   I have a Sangoma 200A (I think that's the model #) analog 4 port
 card.
It works great... however almost everytime after someone 
hangs up a
   call they were on.. the system rings the call back in, as 
though it

   were a new call coming in.  When they pickup no one is there.
  
   Can anyone suggest why this is happening, and how I can make it
 stop?

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[Asterisk-Users] Listening a conversation

2006-05-03 Thread Olivier Saulnier


Hello,

is it possible to listen a conversation in real time, without recording it?

Best regards,

--
Olivier Saulnier
STEGANUX
35 Quai Louis Blanc
03100 Montluçon
T: 04.70.02.80.55
F: 04.70.02.80.57
http://www.steganux.com

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[Asterisk-Users] SendURL

2006-05-03 Thread Scott Gifford

Hello,

I just started working with Asterisk about a month ago, and so far
I've had great luck with it!  Things I expected to be hard were easy,
and it's easy to customize.  Thanks!

I'm trying to send a URL with a queue; that is, when an agent picks up
the phone, I'd like a particular URL to be displayed on the agent's
screen, depending on the queue or the dialed number (DNIS).  The
Queue() application supports this via a URL parameter, which is
exactly what I want.  But I can't seem to find a client that will do
anything with the URL.  I tried creating an extension that just uses
SendURL, and nothing seemed to work there, either.  I'm testing on
Linux; the actual application will probably run on Windows initially,
then hopefully move to Linux over the next few months, so something
cross-platform would be ideal.  So, a few questions on this:

  * Does Asterisk support SendURL over SIP, or only over IAX?  Is
there support in the SIP protocol for sending URLs or similar?

  * Does anybody know of a softphone that works with Asterisk's
SendURL command?  Cross-platform would be nice, open source ideal.

  * If not, can anybody recommend a good open-source softphone that
I could add URL support to?

Thanks!

Scott.
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Re: [Asterisk-Users] Listening a conversation

2006-05-03 Thread Tom Vile


yes, with ChanSpy.

On 5/3/06, Olivier Saulnier [EMAIL PROTECTED] wrote:

Hello,

is it possible to listen a conversation in real time, without recording it?

Best regards,

--
Olivier Saulnier
STEGANUX
35 Quai Louis Blanc
03100 Montluçon
T: 04.70.02.80.55
F: 04.70.02.80.57
http://www.steganux.com

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--
Tom Vile
Baldwin Technology Solutions, Inc
Consulting - Web Design - VoIP Telephony
www.baldwintechsolutions.com
Phone: 518-631-2855 x205
Fax: 518-631-2856
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RE: [Asterisk-Users] Listening a conversation

2006-05-03 Thread Steve Totaro

Chanspy is my method of choice

-Original Message- 
From: Olivier Saulnier [mailto:[EMAIL PROTECTED] 
Sent: Wed 5/3/2006 10:52 AM 
To: Asterisk Users Mailing List - Non-Commercial Discussion 
Cc: 
Subject: [Asterisk-Users] Listening a conversation



Hello,

is it possible to listen a conversation in real time, without recording 
it?

Best regards,

--
Olivier Saulnier
STEGANUX
35 Quai Louis Blanc
03100 Montluçon
T: 04.70.02.80.55
F: 04.70.02.80.57
http://www.steganux.com

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Re: [Asterisk-Users] Listening a conversation

2006-05-03 Thread Gareth Blades

If its going over a zaptel interface then you certenly can. See
http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+ZapBarge

On Wed, 2006-05-03 at 15:52, Olivier Saulnier wrote:
 Hello,
 
 is it possible to listen a conversation in real time, without recording it?
 
 Best regards,

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RE: [Asterisk-Users] Can I recreate a Fax from a recorded file?

2006-05-03 Thread Steve Totaro

Maybe if you had the un-muxed sending side but I really have no idea.  
Interesting question though.

-Original Message- 
From: Alexander Lopez [mailto:[EMAIL PROTECTED] 
Sent: Wed 5/3/2006 9:32 AM 
To: Asterisk Users Mailing List - Non-Commercial Discussion 
Cc: 
Subject: [Asterisk-Users] Can I recreate a Fax from a recorded file?

This is a very KGB / NSA / InterPOL / CIA type question, but if I have 
a recorded file (G.711, no compression) can I feed it into standard in of an 
application and have it recreate the fax that was send?

I don’t know enough about the Fax handshaking to understand this.

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RE: [Asterisk-Users] Simple Dell Computers

2006-05-03 Thread Kerry Garrison

Most of the Dells will work fine with some minor workarounds. First off, go
into the BIOS and disable every possible device (USB, Floppy controller
serial, parallel, etc). Then if the card does not work correctly, move it to
a different slot. With most of the lower end Dells you will find that the
card will only function properly in one of the three PCI slots. If you get a
motherboard that has more than 3 PCI slots your chances of success are
dramatically higher.

Kerry Garrison
Publisher - http://VOIPSpeak.net
(949) 502-7819 x200 - [EMAIL PROTECTED]
http://www.techdatapros.com 
 

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Dovid Bender
 Sent: Wednesday, May 03, 2006 5:10 AM
 To: asterisk-users@lists.digium.com
 Subject: [Asterisk-Users] Simple Dell Computers
 
 Hello List,
 I know this has been brought up many times but I wanted to 
 know if anyone had any expirience in the following. I setting 
 up several voice mail systems.
 Each one is going to have a TDM400P. Two FXO for people to 
 leave messages and two FXS for POTS phones so people can 
 listen. Anyone know if there are any simple specific dell 
 models that will handle this without a problem ?
 
 Thanks.
 
 Dovid
 
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Re: [Asterisk-Users] Listening a conversation

2006-05-03 Thread Moises Silva


Look for ChanSpy Application in voip-info.org

Regards

On 5/3/06, Olivier Saulnier [EMAIL PROTECTED] wrote:

Hello,

is it possible to listen a conversation in real time, without recording it?

Best regards,

--
Olivier Saulnier
STEGANUX
35 Quai Louis Blanc
03100 Montluçon
T: 04.70.02.80.55
F: 04.70.02.80.57
http://www.steganux.com

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--
Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org;
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[Asterisk-Users] Running applications when a queued call is answered

2006-05-03 Thread Scott Gifford

Hello,

I'm experimenting with Asterisk for possible use in a call center.
I'm trying to figure out how to run applications when an agent answers
a call in the queue.  I see that the queue itself supports a very
limited range of applications; for example, I can give a URL to the
Queue() application to SendURL(), or an announcement to read to the
agent.  I'd like to do some slightly more sophisticated things, like
run an external application with System().

When I was using normal extensions and routing the call to one person,
I could do something like this:

exten = 3772,1,Ringing()
exten = 3772,2,System(/home/sgifford/ircsay sgifford Call for ${EXTEN} at 
${DATETIME})
exten = 3772,3,Wait(2)
exten = 3772,4,Dial(SIP/sgifford)

to run an external application and wait 2 seconds while the caller
still heard ringing.  Is there a way to do something similar when a
queued call is delivered?  Maybe with AGI?

I've seen some recommendations to tail the logfile, but that seems
kludgey...

I'm currently using the 1.0.7-BRIstuffed-0.2.0-RC7k Asterisk package
included with Debian 3.1 (Sarge), but I'd be happy to upgrade to a
newer version if that would help.

Thanks for any tips or ideas!

Scott.
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RE: [Asterisk-Users] LDAPget

2006-05-03 Thread Mimmus

Asterisk::LDAP is unrelated to app_ldap.
Just pre-install openldap-devel for your distro, download app_ldap.c, put it
under apps dir of Asterisk source tree and recompile Asterisk normally.

DV

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Joao Pereira
 Sent: Wednesday, May 03, 2006 4:37 PM
 To: asterisk-users@lists.digium.com
 Subject: [Asterisk-Users] LDAPget
 
 Hello to all
 Im using [EMAIL PROTECTED] 2.7 and I would like to do LDAP querys.
 Can I simply use LDAPget or do I need to install 
 Asterisk::LDAP from Alkaloid Networks?

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Re: [Asterisk-Users] User Defined VoiceMail announcement?

Well said. Or you can create an extension to which
people dial in to to check thier VM

Exten 8000,1,Voicemailman()

--- C F [EMAIL PROTECTED] wrote:

 RTFM
 
 On 4/24/06, Benoit Panizzon [EMAIL PROTECTED]
 wrote:
  Hi all
 
  I noticed that most caller are quite confused by
 the standard voicemail
  announcement text. Especialy as the number read is
 the 'internal' number.
  Callers often hang up because they think having
 called the wrong number when
  they hear the announcement.
 
  Is there a way (like in many other PBXes) that the
 VoiceMail user could record
  his own announcement? (like, hello, this is the
 Voicebox of John Smith,
  please leave a message after the tone).
 
  Mit freundlichen Grüssen
 
  Benoit Panizzon
  --
  I m p r o W a r e   A G-System Services
 

__
 
  Zurlindenstrasse 29 Tel  +41 61 826 93
 00
  CH-4133 PrattelnFax  +41 61 826 93
 01
  Schweiz Web 
 http://www.imp.ch
 

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Re: [Asterisk-Users] User Defined VoiceMail announcement?

Sorry for the typo

Exten 8000,1,Voicemailmain
--- C F [EMAIL PROTECTED] wrote:

 RTFM
 
 On 4/24/06, Benoit Panizzon [EMAIL PROTECTED]
 wrote:
  Hi all
 
  I noticed that most caller are quite confused by
 the standard voicemail
  announcement text. Especialy as the number read is
 the 'internal' number.
  Callers often hang up because they think having
 called the wrong number when
  they hear the announcement.
 
  Is there a way (like in many other PBXes) that the
 VoiceMail user could record
  his own announcement? (like, hello, this is the
 Voicebox of John Smith,
  please leave a message after the tone).
 
  Mit freundlichen Grüssen
 
  Benoit Panizzon
  --
  I m p r o W a r e   A G-System Services
 

__
 
  Zurlindenstrasse 29 Tel  +41 61 826 93
 00
  CH-4133 PrattelnFax  +41 61 826 93
 01
  Schweiz Web 
 http://www.imp.ch
 

__
 
 
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Re: [Asterisk-Users] Listening a conversation

2006-05-03 Thread Olivier Saulnier


Ok, thanks everybody :-)

--
Olivier Saulnier
STEGANUX
35 Quai Louis Blanc
03100 Montluçon
T: 04.70.02.80.55
F: 04.70.02.80.57
http://www.steganux.com

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Re: [Asterisk-Users] Can I recreate a Fax from a recorded file?

2006-05-03 Thread Steve Underwood


Alexander Lopez wrote:

This is a very KGB / NSA / InterPOL / CIA type question, but if I have 
a recorded file (G.711, no compression) can I feed it into standard in 
of an application and have it recreate the fax that was send?


I don’t know enough about the Fax handshaking to understand this.

In spandsp there is a program in the tests directory called fax_decode. 
It isn't very sophisticated, as it is intended for my test work, rather 
than general decoding. It is able to decode some FAX audio from a wave 
file, though.


There are some expensive commercial programs which do the job. Because 
FAX only sends one way at a time, the audio from the two directions is 
never jumbled up in recordings.


Steve

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RE: [Asterisk-Users] Running applications when a queued call is answered

Use the Local channel and add the agents using that IE:

Member Local/[EMAIL PROTECTED]

Snip

 Hello,
 
 I'm experimenting with Asterisk for possible use in a call center.
 I'm trying to figure out how to run applications when an agent answers
 a call in the queue.  I see that the queue itself supports a very
 limited range of applications; for example, I can give a URL to the
 Queue() application to SendURL(), or an announcement to read to the
 agent.  I'd like to do some slightly more sophisticated things, like
 run an external application with System().
 
 When I was using normal extensions and routing the call to one person,
 I could do something like this:
 
 exten = 3772,1,Ringing()
 exten = 3772,2,System(/home/sgifford/ircsay sgifford Call for
 ${EXTEN} at ${DATETIME})
 exten = 3772,3,Wait(2)
 exten = 3772,4,Dial(SIP/sgifford)
 
 to run an external application and wait 2 seconds while the caller
 still heard ringing.  Is there a way to do something similar when a
 queued call is delivered?  Maybe with AGI?
 
 I've seen some recommendations to tail the logfile, but that seems
 kludgey...
 
 I'm currently using the 1.0.7-BRIstuffed-0.2.0-RC7k Asterisk package
 included with Debian 3.1 (Sarge), but I'd be happy to upgrade to a
 newer version if that would help.
 
Snip


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RE: RE: [Asterisk-Users] Dial Option wW picking up the *1 is a bitflaky

2006-05-03 Thread Adam Hatia

Thanks Guillaume.
What's the maximum allowed delay? I there a way of setting it?

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Guillaume de
Lafontaine
Sent: 03 May 2006 14:59
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: RE: [Asterisk-Users] Dial Option wW picking up the *1 is a
bitflaky



 -Original Message-
 From: Mark Ackroyd [EMAIL PROTECTED]
 To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
 CC: 
 Subject: RE: [Asterisk-Users] Dial Option wW picking up the *1 is a bit
flaky
 Sent: mer., 03 mai 2006 09:15:13 GMT
 Received: mer., 03 mai 2006 09:18:20 GMT
 Read: mer., 03 mai 2006 10:07:50 GMT
  When I hit *1 in my system, I got a beep to let me know that the
  recording started. Is this not happenning to you?
 
 No ! , it doesn't.  Most of the time it doesn't pick up the *1.
 

Hi!

Just make sure to hit *1 very quickly... I mean with minimum delay between *
and 1.

Guillaume

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Re: [Asterisk-Users] User Defined VoiceMail announcement?

2006-05-03 Thread Steven Ringwald

I think that he meant recording the unavailable and busy messages for 
the mailbox.


To do this, log into the voicemail-box, and hit '0' for Mailbox options. 
The options that you are interested in are numbered 1-3.


Steve

Dovid Bender wrote:

Well said. Or you can create an extension to which
people dial in to to check thier VM

Exten 8000,1,Voicemailman()

--- C F [EMAIL PROTECTED] wrote:

  

RTFM

On 4/24/06, Benoit Panizzon [EMAIL PROTECTED]
wrote:


Hi all

I noticed that most caller are quite confused by
  

the standard voicemail


announcement text. Especialy as the number read is
  

the 'internal' number.


Callers often hang up because they think having
  

called the wrong number when


they hear the announcement.

Is there a way (like in many other PBXes) that the
  

VoiceMail user could record


his own announcement? (like, hello, this is the
  

Voicebox of John Smith,


please leave a message after the tone).

Mit freundlichen Grüssen

  



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Re: [Asterisk-Users] SendURL

2006-05-03 Thread Time Bandit


  * Does anybody know of a softphone that works with Asterisk's
SendURL command?  Cross-platform would be nice, open source ideal.

I'm currently working on an updated version of my MediaX phone and it
supports receiving URL. It works only on windows and is not open
source. But if you want to try it, email me directly and I will send
it to you.

hth
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Re: [Asterisk-Users] Queue reporting seems broken.

2006-05-03 Thread Joe Dennick

On the wiki, there is a little perl script that can be used to parse the queue 
log and insert the data into a database.  I've modified the script to use a 
MySQL database.  With that, I have a cron job that shuts down Asterisk, parses 
the queue-log into the MySQL database, and then restarts Asterisk (thus 
re-initializing a new (empty) queue log).  Once the data is in the database, 
its pretty easy to find the information you are seeking.  

Simply query the database for all records (COUNT(*)) where action = 
'ENTERQUEUE' and date is between the range you are searching for to find out 
how many calls entered the queue for that time-period.

If you do a similar query for action = 'COMPLETECALLER' or 'COMPLETEAGENT' you 
will see all of the answered calls.  You can further limit that query by 
specifying a particular agent.

I've wrapped all of these queries in PHP and provide web forms (html) to allow 
the Call Center Manager to select what data to report on (date ranges, queue, 
agent, etc.).  I can send you some of the PHP code if you are interested. 

In summary, once you've got the data in a database, you can extract it in any 
way that's meaningful to you.  The wiki provides really good information about 
what ACTIONS are recorded, and then what information is provide in the info1, 
info2, and info3 fields for each ACTION.  The database structure is very flat 
and easy to work with, so you really don't have to know very much about 
databases to achieve excellent reporting results.

Johann [EMAIL PROTECTED] wrote the May 3, 2006 8:26 AM:

 The wiki page doesn't mention the other fields in the file.
 
 TIMESTAMP:UNIQUEID:QUEUE:AGENT:ACTION:ARG1:ARG2:ARG3
 
 Depending on the action some of the fields may contain 'NONE' instead.  You 
 will 
 need to use the AGENT field to match who the connect and 
 completeagent/completecaller messages are for.
 
 Keep in mind the UNIQUEID field will be the same for a caller as they go 
 through the queue.  So the enterqueue, connect, complete actions will have 
 the same.
 
 
 --johann
 
 Thermal Wetland wrote:
  I am trying to figure out which one of our agents is answering the calls.
  
  According to http://www.voip-info.org/wiki/view/Asterisk+log+queue_log 
  the only time the queue_log puts the channel (agent) is during logoff  
  logon.
  
  There is the connect  completeagent message, but it doesn't show which 
  channel (agent) answered the phone.
  
  I can't even figure it our cross referencing the CDR records, the CDR 
  record only has the queue number.
  
  Is there a way around this?
  
  Aloha,
  Matt
  
  
  
  
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RE: [Asterisk-Users] Can I recreate a Fax from a recorded file?




Acually, I have no time on my hands, but this was the 
thought while in the shower this AM. Thought was the following. I needed to have 
one fax sent to me and a customer at the same time. I know that I can recieve 
and resend to both but I want to be able to 'snoop'.


  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Technical 
  SupportSent: Wednesday, May 03, 2006 9:57 AMTo: 
  'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: 
  RE: [Asterisk-Users] Can I recreate a Fax from a recorded 
  file?
  
  Interesting question! If you have the audio in only 
  (assuming it was a fax received) or audio out only (assuming it was a fax 
  sent), and you pair with an identical fax machine to the original (assuming it 
  responds exactly the same in terms of handshakes, speeds, ECM, etc) then it 
  might work. I assume you have a lot of time on your hands :) 
  
  
  MD
  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Alexander 
  LopezSent: Wednesday, May 03, 2006 9:33 AMTo: Asterisk 
  Users Mailing List - Non-Commercial DiscussionSubject: 
  [Asterisk-Users] Can I recreate a Fax from a recorded 
  file?
  
  
  This is a very KGB / NSA / 
  InterPOL / CIA type question, but if I have a recorded file (G.711, no 
  compression) can I feed it into standard in of an application and have it 
  recreate the fax that was send?
  
  
  I dont know enough about the Fax 
  handshaking to understand this.
  
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[Asterisk-Users] QSIG support in Asterisk

2006-05-03 Thread Asterisk User


I am looking to get the info about QSIG support in Asterisk. 
Does Asterisk have QSIG support?
Does Asterisk support QSIG SIP Tunneling or QSIG SIP Interworking?

If so, How to configure that?

Thanks
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RE: [Asterisk-Users] Can I recreate a Fax from a recorded file?

You da' Man!!!

I'll try this.
 
In spandsp there is a program in the tests directory called 
fax_decode. 
It isn't very sophisticated, as it is intended for my test 
work, rather than general decoding. It is able to decode some 
FAX audio from a wave file, though.

There are some expensive commercial programs which do the 
job. Because FAX only sends one way at a time, the audio from 
the two directions is never jumbled up in recordings.

Steve
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Re: [Asterisk-Users] SATA hard disk compatibility

Assaf I am going to argue with you and agree with John
ehre. FC has a very small shelf life and AFAIK it
dosent have what CentOS does. CentOS is RHLE without a
serial num or support. Try it out (www.centos.org).

--- Assaf Flatto [EMAIL PROTECTED] wrote:

 Fedora core stable version now is FC4 (which is to
 say Red hat 10 
 version 4 or even Red Hat 11 if we count in the old
 way RH did ).
 IAX and the configuration have changed a bit from
 1.0.3  so you'll need 
 to modify the file to match the new configuration
 but other then that it 
 should be no problem to move to a more Up-To-Date
 system.
 the current stable asterisk is 1.2.7.1 , and it
 works smoothly for me on 
 several machine i installed and used in several
 locations.
 
 As John suggested CentOs is a good idea to use ,
 however there are some 
 well documented problems with the zaptel ( that is
 if you are using 
 digium hardware ) compilation ,but all in all it
 shouldn't be a problem.
 
 Assaf
 
 
 amna saleem wrote:
  Thanks alot for the help.
  I have not worked on fedra core .Which version
 should I use
  Also can you tell me that if I am using Red hat
 Enterprise, which 
  asterisk version will be the best suited ? and
 will i be able to use 
  the same .conf files which i used earlier with
 aserisk 1.0.3.
  I only need to use IAX ,and the IAX soft phones
 ,don`t really have to 
  use SIP or H323.
  Also I want a stable asterisk version like 1.0.3
 which doesn`t need to 
  be upgraded continuously.
   
  I hope you will help me 
  Regards,
  Amna 
 
   
  On 4/27/06, *Assaf Flatto* [EMAIL PROTECTED] 
  mailto:[EMAIL PROTECTED] wrote:
 
  The Hardware support of SATA in RH9.0 is not
 fully integrated
  AFAIK , so
  moving to a SATA hard disk without an upgrade
 might not be the
  safest bet.
 
  on the other hand until you try you won't know
 for sure .
 
  have you thought of using the Fedora Core ?
 those have SATA
  support and
  they should be the closest thing to RH9 you
 can find.
 
 
  why don't you want to upgrade the asterisk ?
 1.0.3 is a very old
  version
  and many fixes and features where added to the
 software .
 
 
  Assaf
 
  amna saleem wrote:
   Hi!
   I have been using ASterisk 1.0.3 on Red hat
 Linux 9.0 for a long
  time
   now on my Home PC.
   I want to shift to a PC having SATA hard
 disk .Can I install Redhat
   9.0 on SATA hard disk ??some people are
 telling me that I have to go
   for Linux Enterprise 4.0.I don`t want to
 leave Linux 9.0 because I
   want to run Asterisk 1.0.3
  
   Can anyone help me??
   Amna
  
 


  
  
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  --
  Assaf Flatto
  Atelis IT Manager
  Cellular: +972-54-5679230
  e-mail: [EMAIL PROTECTED]
 mailto:[EMAIL PROTECTED]
 
 
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 -- 
 Assaf Flatto
 Atelis IT Manager
 Cellular: +972-54-5679230
 e-mail: [EMAIL PROTECTED]
 
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[Asterisk-Users] G.722 Softphone?

2006-05-03 Thread Rusty Dekema


Hi,

Does anyone know of a softphone that supports G.722; preferably one
that is available free of charge? Either IAX2 or SIP would be fine.

Thanks,
Rusty
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[Asterisk-Users] SIP Phones behind dynamic IPs

2006-05-03 Thread Chris Bagnall

Greetings list,

I'm coming across an issue with some of the GXP-2000 phones we have out in
the wild at clients' employees' homes. In most cases they're behind consumer
ADSL NAT routers on a dynamic IP from their ISP.

In a nutshell, the phone is unable to be called unless it's restarted first,
after which it's fine for a good few hours, then it stops working until
restarted again.

The problem doesn't seem to be anywhere near as regular with users that are
on cable connections (these tend to have much more sticky IP addresses -
they change only every few months rather than every time the ADSL router
connects), and non-existent on ADSL connections with static IPs.

I've tried various permutations - with STUN, without STUN, NAT keep-alives
down as low as 10 seconds, nat=yes in sip.conf, ports forwarded to the
phone, ports *not* forwarded to the phone, etc.

I think what's happening is that the ADSL router is reconnecting after a
break in the connection (as it should), getting a different IP, but the
phones don't seem to be recognising they've got a different IP and updating
the asterisk server with the good news.

Has anyone else encountered similar issues? Anything else I can try (bearing
in mind I have no control over the ADSL connections the users are subscribed
to)?

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] Phone Emergency - Need IAX Help

cant answer for your current problem. I had an
emergency a few weeks ago in the middle of the night.
Signed up with teliax.com and I was up in about 5
minutes.

Dovid

--- David Tillman [EMAIL PROTECTED]
wrote:

 SBC has an outage that is expected to last until
 tomorrow
 in our area. This has taken out our 5 POTS lines and
 our T1.
 
 I have signed up with EXGN for outbound calls and am
 using
 IAX. Calls ring through to the other party (my cell
 phone in
 this case) but Asterisk doesn't seem to think the
 call
 was answered. Ideas?
 
 -dave
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[Asterisk-Users] Voipjet Problem?

2006-05-03 Thread Mark Hulber

I started to have a problem today that all my calls through voipjet 
result in just timing out after my assigned timeout period.  I tried 
multiple of their servers with the same problem.  Anyone else having a 
problem?  I am running:


Asterisk SVN-branch-1.2-r24381M built by root @ asterisk.hulber.com on a 
i686 running Linux on 2006-05-03 14:14:07 UTC


I can connect with other IAX providers.

MARK.
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Re: [Asterisk-Users] PrivacyManager FastAGI: Rewrite or use?

snip
 I'm building an app that will do the following:
 
  1. Force the caller to record their name.
  2. Dial the party to call.
  3. Play a short menu:
  1 = Accept Call
  2 = Decline Call, go to VM if available
  3 = Accept Call forever, never ask again
  4 = Decline Call forever, block number, get
 rid of caller
 
  4. do things based on that choice.
/snip
When its done can you share it with the class ?

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Re: [Asterisk-Users] Problem with a TDM-400P

Call Digium support.

--- Nigel Smith [EMAIL PROTECTED] wrote:

 (Sorry of this appears in the list twice, but I
 wasn't sure if it was
 blocked or not)
 
 Hi there,
 
 I'm having a problem with my TDM-400P which has been
 working like a
 charm up until very recently. It started to fail
 last week, and so I was
 hoping someone could illuminate me with some
 information as to why.  Its
 configuration is as follows:
 
 
 FXS (green) module is in position 1, closest to the
 bracket. This
 appears to be the failing component.
 FXO (red) module is in position 4, furthest from the
 bracket
 
 
 From what I can gather it's just the FXS module
 that is failing, and not
 the card. What I was hoping someone could tell me is
 whether the whole
 card is U/S or just that module, based on the
 information I have
 provided below. Any help will be greatly
 appreciated.
 
 Thanks,
 Nigel
 
 
 The interesting parts of the zapata.conf are:
 --
 ; The FXS port, or the phone port
 signalling=fxo_ks
 echocancel=yes ; You can set this to 32, 64, or 128,
 tweak to your needs.
 echocancelwhenbridged=yes
 echotraining=400 ; Asterisk trains to the beginning
 of the call, number
 is in milliseconds
 [EMAIL PROTECTED]
 group=1
 context=outgoing ; Points to the default context of
 your extensions.conf
 channel = 1
 usecallerid=no
 ;cidsignalling=bell
 ;cidstart=ring
 ;hidecallerid=no
 callwaiting=yes
 
 
 ; The FXO port, or the line port
 signalling=fxs_ks
 group=2
 ;rxwink=300  ; Atlas seems to use long
 (250ms) winks
 ;hanguponpolarityswitch ; doesn't work on ADSL lines
 busydetect=yes
 busycount=3
 context=incoming-pstn
 channel= 4 ; Again change the 'X' to the number of
 FXO modules you have
 usedistinctiveringdetection=no
 usecallerid=no
 ;sendcalleridafter=2
 ;cidsignalling=bell
 ;cidstart=ring
 ;hidecallerid=no
 ;restrictcid=no
 callwaiting=yes
 
 
 The asterisk failure is
 -
 Apr 27 10:00:46 splash asterisk[14625]:
 WARNING[14625]: chan_zap.c:923
 in zt_open: Unable to specify channel 1: No such
 device
 Apr 27 10:00:46 splash asterisk[14625]:
 ERROR[14625]: chan_zap.c:6878 in
 mkintf: Unable to open channel 1: No such device
 here = 0, tmp-channel
 = 1, channel = 1
 Apr 27 10:00:46 splash asterisk[14625]:
 ERROR[14625]: chan_zap.c:10314
 in setup_zap: Unable to register channel '1'
 Apr 27 10:00:46 splash asterisk[14625]:
 WARNING[14625]: loader.c:414 in
 __load_resource: chan_zap.so: load_module failed,
 returning -1
 Apr 27 10:00:46 splash asterisk[14625]:
 WARNING[14625]: loader.c:554 in
 load_modules: Loading module chan_zap.so failed!
 
 And asterisk promptly dies.
 
 If I execute the following:
 
 #  ztcfg -vv
 
 Zaptel Configuration
 ==
 
 
 Channel map:
 
 Channel 01: FXO Kewlstart (Default) (Slaves: 01)
 Channel 04: FXS Kewlstart (Default) (Slaves: 04)
 
 2 channels configured.
 --
 You can see that the channels are configured OK.
 
 And the output from the following commands is a
 little strange as well:
 -
 # cat /dev/zap/1
 cat: /dev/zap/1: No such device
 # cat /dev/zap/2
 cat: /dev/zap/2: No such device or address
 # cat /dev/zap/3
 cat: /dev/zap/3: No such device or address
 # cat /dev/zap/4
 (And lots of stuff gets spewed out onto the screen)
 --
 
 
 
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Re: [Asterisk-Users] QSIG support in Asterisk

2006-05-03 Thread Marco Mouta

http://www.voip-info.org/wiki-Asterisk+config+zapata.confI've made some tests using this in Portugal and seems to work:---
switchtype=qsig ; you may try this in your zapata.conf--I suppose you are using PRI access.But i've also red that qsig is not fully compliant in Asterisk. I've found some PRI states not recognized in asterisk sending CAUSECODE=98 to my legaccy PBX when it seems to me that Asterisk should handle it...
Any ways it seems to work in a standardt architecture:PSTN--E1---LegacyPBX---QSIG---AsteriskI hope it helps,Marco MoutaOn 5/3/06, 
Asterisk User [EMAIL PROTECTED] wrote:

I am looking to get the info about QSIG support in Asterisk. 
Does Asterisk have QSIG support?
Does Asterisk support QSIG SIP Tunneling or QSIG SIP Interworking?

If so, How to configure that?

Thanks

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Re: [Asterisk-Users] SIP Phones behind dynamic IPs

2006-05-03 Thread Kevin P. Fleming

Chris Bagnall wrote:

 I think what's happening is that the ADSL router is reconnecting after a
 break in the connection (as it should), getting a different IP, but the
 phones don't seem to be recognising they've got a different IP and updating
 the asterisk server with the good news.

'recognize'? The phone cannot know that the external IP has been
changed, unless it is using a STUN server and periodically re-doing the
STUN queries (which I doubt any phones do).

Probably the best you are going to be able to do is to set the
registration interval on the phone to something short enough to not be
too painful for your Asterisk server but still enough to keep the phone
reachable.
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[Asterisk-Users] Selecting the outbound port from FXO device

2006-05-03 Thread Mustafa Sakalsiz


Hi,

I am using Wellgate 3806 FXO(SIP Proxy Mode) with Asterisk. I have 6 
lines registred as


SIP/2901 (line1)
.
.
SIP/2906 (line6)

to the asterisk. I have a dialplan configuration like

exten = _90NX,1,Dial(SIP/2902/${EXTEN:1})
or
exten = _90NX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])

When I call a number matching the above pattern, the first line (#1) of 
my FXO device is activated although I had chosen the second line (#2).


On the other way, with a configuration like below

exten = 72,1,Dial(SIP/2902)

I call 72, and the correct outbound line is activated and I get the dial 
tone from line.


I want to choose the outbound port of the FXO device while one stage 
dialing. How can I do that?


Your help is very appreciated.

Thanks,
Saki
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[Asterisk-Users] Forwarded Numbers and Timeouts

2006-05-03 Thread Douglas Garstang

I have a tricky situation. I have a polycom phone with number 3254103. I have 
configured the phone to forward to a new number, 1805999.

Here's my dialplan:
exten = 3254103,1,Dial(SIP/3254103,10,tr)
exten = 1805999,1,Dial(SIP/[EMAIL PROTECTED],40,tr)

When Asterisk dials 3254103, here's what comes up on the console:

hestia*CLI 
-- Executing Dial(SIP/2944093-6935, SIP/3254103|10|tr) in new stack
-- Called 3254103
-- Got SIP response 302 Moved Temporarily back from xxx.187.128.19
-- Now forwarding SIP/2944093-6935 to 'Local/[EMAIL PROTECTED]' (thanks to 
SIP/3254103-47ab)
-- Executing Dial(Local/[EMAIL PROTECTED],2, SIP/[EMAIL 
PROTECTED]|40|tr) in new stack
-- Called [EMAIL PROTECTED]
-- Local/[EMAIL PROTECTED],1 is ringing
-- SIP/proxy2-5adc is ringing
-- SIP/proxy2-5adc is making progress passing it to Local/[EMAIL 
PROTECTED],2
-- Nobody picked up in 1 ms
  == Spawn extension (betty_start, 1805999, 1) exited non-zero on 
'Local/[EMAIL PROTECTED],2'
  == Auto fallthrough, channel 'SIP/2944093-6935' status is 'NOANSWER'
hestia*CLI 

You can see that the phone tells Asterisk that the number has been forwarded. 
Asterisk re-enters the dialplan logic and tries to contact the forwarded 
number. That's all great...

We have a problem of timeouts here. In this situation, Asterisk drops all call 
flow at 10 seconds, which was the timeout set for the original number, 3254103, 
eventhough it has now re-entered the dialplan logic, dialling a new number with 
a timeout of 40 seconds. It's as if the timeout of the original number sets the 
timeout for the forwarded call.

Shouldn't the timeout used to dial 1805999 be 40 seconds? Why does Asterisk 
use the original timeout of 10s? This causes all sorts of problems.

Thanks,
Doug.





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[Asterisk-Users] QSIG support in Asterisk

2006-05-03 Thread Asterisk User

I am looking to get the info about QSIG support in Asterisk. 
Does Asterisk have QSIG support?
Does Asterisk support QSIG SIP Tunneling or QSIG SIP Interworking?

If so, How to configure that?

Thanks
--dp
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RE : [Asterisk-Users] Problem with a TDM-400P

2006-05-03 Thread f6hqz-m

Hello,

Check your gren module by moving it from slot to slot on the TDM400P card.
If the problem is following your module, it's the module itself the cause.
If not, and running well on other slot, it's the TDM400P itself.

Good Luck !

Best Regards,
Francois BERGERET,
France.


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Re: [Asterisk-Users] QSIG support in Asterisk

2006-05-03 Thread Daniel


Hello anon user. You can check voip wiki at http://www.voip-info.org


Asterisk User wrote:
 


 I am looking to get the info about QSIG support in Asterisk.

 Does Asterisk have QSIG support?

 Does Asterisk support QSIG SIP Tunneling or QSIG SIP Interworking?

If so, How to configure that?
 
Thanks





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Re: [Asterisk-Users] SendURL

2006-05-03 Thread Jean-Denis Girard


Scott Gifford a écrit :

Hello,

I just started working with Asterisk about a month ago, and so far
I've had great luck with it!  Things I expected to be hard were easy,
and it's easy to customize.  Thanks!

I'm trying to send a URL with a queue; that is, when an agent picks up
the phone, I'd like a particular URL to be displayed on the agent's
screen, depending on the queue or the dialed number (DNIS).  The
Queue() application supports this via a URL parameter, which is
exactly what I want.  But I can't seem to find a client that will do
anything with the URL.  I tried creating an extension that just uses
SendURL, and nothing seemed to work there, either.  I'm testing on
Linux; the actual application will probably run on Windows initially,
then hopefully move to Linux over the next few months, so something
cross-platform would be ideal.  So, a few questions on this:

  * Does Asterisk support SendURL over SIP, or only over IAX?  Is
there support in the SIP protocol for sending URLs or similar?

  * Does anybody know of a softphone that works with Asterisk's
SendURL command?  Cross-platform would be nice, open source ideal.


May I suggest MozIAX: it's a Mozilla / Firefox extension, so it does 
natively support receiving URL from Asterisk. It also adds tel: 
protocol to Firefox, so you can call from the web page. MozIAX also 
supports receiving / sending text messages to / from Asterisk, for chat 
sessions. It is open source, runs on windows and linux, and was reported 
to work on OS X. More info at: http://moziax.mozdev.org/.




Thanks,
--
Jean-Denis Girard

SysNux  Systèmes Linux en Polynésie française
http://www.sysnux.pf/   Tél: +689 483 527 / GSM: +689 797 527
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[Asterisk-Users] Colocation Denmark

2006-05-03 Thread Sahil Gupta


Hi there,
Is there anybody on the list that offers or can put me in touch with 
somebody that offers quality colocation services in Denmark?


Regards,


Sahil Gupta
VoiceValley
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[Asterisk-Users] RE: [asterisk-biz] Colocation Denmark

2006-05-03 Thread Bjorn Asmul

Try these guys: http://easyspeedy.com/

Haven't tried them, but when I was looking into a while back they
responded quickly.

-- Bjorn 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sahil Gupta
Sent: Wednesday, May 03, 2006 1:47 PM
To: asterisk-users@lists.digium.com
Cc: asterisk-biz@lists.digium.com
Subject: [asterisk-biz] Colocation Denmark

Hi there,
Is there anybody on the list that offers or can put me in touch with
somebody that offers quality colocation services in Denmark?

Regards,


Sahil Gupta
VoiceValley
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Re: [Asterisk-Users] Voipjet Problem?