Re: [asterisk-users] log incoming calls without answering

[Fabio Moretti]

Thank for all the replies, a lot of input and information!

Sorry for this useless mail, but I really wanted to say thank you.


Il 20/04/2017 17:26, Fabio Moretti ha scritto:
> Hi,
>
> I've some analogic lines and I'm asked if it's possible to program an 
> asterisk for "checking" the inbound calls without answering them, doing 
> something like this:
>
> analog line 1 -+-- asterisk
>|
>\__ analog phone
>
> when a call enter, asterisk sense it and store its values (callerid, date and 
> time, etc) somewhere, but nothing more, people will answer using the old 
> analog phone.
> The goal is to have a log of the inbound calls without touching the old 
> analog system (it's shared between different subjects).
>
> I'm pretty sure it's something possible, but how to tell asterisk: "ok, call 
> this AGI, and then don't answer and do nothing more".
>
> Any idea?
>
> Thanks
>
>
>
>   
>
>

-- 
Fabio Moretti
Gerente de Sistemas
www.tecytal.com <http://www.tecytal.com>
0800 8780
(+598) 248 77921


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Re: [asterisk-users] log incoming calls without answering

[Fabio Moretti]

Il 20/04/2017 18:09, kevin.lar...@pioneerballoon.com ha scritto:
>
> I honestly don't know if you can do what you want without some piece
> of equipment picking up the line. What I would do is get an analog
> line, an analog phone, an analog to sip device (there are many to
> choose from) and a basic asterisk instance. I would then make a small
> test setup where the analog line goes to a splitter. One side of the
> splitter goes to your analog phone. One side goes to your analog to
> SIP converter and then into your asterisk instance via your ethernet
> network. Use your cell phone to call the number of your analog line
> and see if it works. You would have to code a basic dialplan on the
> asterisk side and set up the trunk to your converter, which I am
> assuming you know how to do.

Yes, I'll definitely do the test before set up the whole proyect, but
the point basically is: it is possibile for asterisk to log a call
without answering it? How to do it in the dialplan? Or I'm wasting time
because an analog line who enter asterisk is always answered?







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Re: [asterisk-users] log incoming calls without answering

[Fabio Moretti]

Il 20/04/2017 17:32, kevin.lar...@pioneerballoon.com ha scritto:
>
> This gets kinda Rube Golberg-ish, but convert the incoming analog line
> to sip, route it through asterisk and have asterisk do its thing
> before converting it back to analog to send to the phone. Only problem
> is you get a lot of extra hardware involved in the mix to make it
> work. It will be a lot of expense and trouble, so you need to make
> sure that whatever part you want asterisk to play is worth that
> effort. Also, I wouldn't touch a fax line in this manner.
>
> If you could give a bit more info on what you want asterisk to do, we
> could maybe give better advice on how to solve your problem.

Hi Kevin,

I've already proposed your solution (is the most reasonable) but they
have more than 60 analogs lines (no faxes) and some of them terminate in
appliances like alarms, etc, so the solution must not touch in any way
the connection between the line and his termination: doing a analog to
digital conversion, passing it to asterisk and the convert it back to
analog is prone to problems (what if asterisk crashes? or if a gateway
fail?).
I can split the existing lines (there are no complex things like adsl or
digital signaling), convert the branches to digital and terminate then
into an asterisk machine, so any failure will not affect the old
circuit, but of course I've to configure asterisk to ONLY LOG calls and
nothing more.

This is what they want:
- line 1 ring
- line 1 is splitted in two, the first branch (let's say the "analog"
branch) go to an analog phone, that rings
- the second branch go through a gateway and then to asterisk
- asterisk log (with an AGI for example) "line 1 rings at  from "
no more is required from asterisk, if someone answer the analog phone or
not is not my business.







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[asterisk-users] log incoming calls without answering

[Fabio Moretti]

Hi,

I've some analogic lines and I'm asked if it's possible to program an asterisk 
for "checking" the inbound calls without answering them, doing something like 
this:

analog line 1 -+-- asterisk
   |
   \__ analog phone

when a call enter, asterisk sense it and store its values (callerid, date and 
time, etc) somewhere, but nothing more, people will answer using the old analog 
phone.
The goal is to have a log of the inbound calls without touching the old analog 
system (it's shared between different subjects).

I'm pretty sure it's something possible, but how to tell asterisk: "ok, call 
this AGI, and then don't answer and do nothing more".

Any idea?

Thanks






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[asterisk-users] strange warnings no samples for alawtolin

[Fabio Moretti]

[Aug 11 21:57:14] WARNING[1992] translate.c: no samples for alawtolin
[Aug 11 21:57:14] WARNING[2005] translate.c: no samples for alawtolin
[Aug 11 21:57:15] WARNING[2027] translate.c: no samples for alawtolin
Hi to all,

I have an elastix box running asterisk 1.8.20 without problem. It's
about four days I've started seen in log a warning message saying
translate.c: no samples for alawtolin, and now the frequency of this
message is about 6 times a second.

There's no other clue, everything is running smoothly and googling for
it doesn't help.

Here's an excerpt:

[Aug 11 21:57:15] WARNING[2029] translate.c: no samples for alawtolin
[Aug 11 21:57:15] WARNING[2038] translate.c: no samples for alawtolin
[Aug 11 21:57:15] WARNING[2045] translate.c: no samples for alawtolin
[Aug 11 21:57:15] WARNING[2055] translate.c: no samples for alawtolin
[Aug 11 21:57:15] WARNING[2059] translate.c: no samples for alawtolin
[Aug 11 21:57:15] WARNING[2078] translate.c: no samples for alawtolin
[Aug 11 21:57:19] WARNING[2093] translate.c: no samples for alawtolin
[Aug 11 21:57:19] WARNING[2095] translate.c: no samples for alawtolin
[Aug 11 21:57:19] WARNING[2110] translate.c: no samples for alawtolin
[Aug 11 21:57:19] WARNING[2120] translate.c: no samples for alawtolin
[Aug 11 21:57:19] WARNING[2125] translate.c: no samples for alawtolin
[Aug 11 21:57:19] WARNING[2132] translate.c: no samples for alawtolin
[Aug 11 21:57:19] WARNING[2139] translate.c: no samples for alawtolin
[Aug 11 21:57:19] WARNING[2141] translate.c: no samples for alawtolin
[Aug 11 21:57:20] WARNING[2152] translate.c: no samples for alawtolin
[Aug 11 21:57:20] WARNING[2174] translate.c: no samples for alawtolin
[Aug 11 21:57:20] WARNING[2177] translate.c: no samples for alawtolin
[Aug 11 21:57:20] WARNING[2208] translate.c: no samples for alawtolin
[Aug 11 21:57:20] WARNING[2210] translate.c: no samples for alawtolin
[Aug 11 21:57:20] WARNING[] translate.c: no samples for alawtolin

Does anyone have an idea of what is means and how I can get rid of it?

Thanks
-- 




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Re: [asterisk-users] CEL logging and queue APP_START/END, maybe an issue?

[Fabio Moretti]

  
  

Il 01/07/2013 15:17, Matthew Jordan
  scrisse:


  Nope, this is entirely expected.

  


  

  

[...snip...]

  

  On a side note, the fact that
masquerades are hard and tend to require people to do lots
of updates was a driving factor in the development efforts
that went on in 12. Masquerades are now an implementation
detail, so in the future, you won't have to deal with
BRIDGE_UPDATE.
  

  


ok matthew, thank you.
I understand, but now I'm getting a little confused: I think that
"linkedid" was the "long waited field" useful to follow a call
during its entire history (in fact I've made modification to my old
asterisk 1.4 dialplan to have something similar in the cdr using
accountcode field). Following what you say I should not only follow
the linkedid but, in case of a masquerading, I've to follow the peer
channel.
So, shoud I've to find and follow all the likedids related to every
BRIDGE_UPDATE? What if, for example, I've two ingoing call from
DAHDI that get bridged at some point? 
Is there a "correct" way to get all the records of a call in a way
that I can use to show the "history" of a call in a human readable
way? 
What I'm doing is experimenting, studying the CEL of an inbound
call-center, and due to the lack of documentation (and my lack and
experience) I can't understand how to follow correctly a call and,
for example, why rarely I get a BLINDTRANSFER, sometimes an
ATTENDEDTRANSFER and sometimes a FORWARD (I'm sure operators only
use the "TR" button on the phone). 
I think the most complete documentation is on wiki.asterisk.org, but
it's more like "XXX is when a channel is XXXed" than an explanation,
and there's no list of what apps can generate CEL events, when and
why.
I appreciate if you can point me to some document I've to study :)

Here I paste the events counts of my two-months CEL, maybe someone
can find it interesting:
ANSWER			166599
APP_END			42424
APP_START		42434
ATTENDEDTRANSFER	712
BLINDTRANSFER		15
BRIDGE_END		73575
BRIDGE_START		74325
BRIDGE_UPDATE		538
CHAN_END		1124624
CHAN_START		1124711
FORWARD			72
HANGUP			1124626
LINKEDID_END		54784

Thank you,
-- 
  

  
  
  Fabio
Moretti  
  Gerente de Sistemas 
   www.tecytal.com  
   0800 8780 
(+598) 248 77921  
  

  

  


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Re: [asterisk-users] Executing Script after MixMonitor is called

[Fabio Moretti]

.

  
  --Satish
Barot
  
  Ahmedabad,
India
  


  


  

  


  

  
  Some observations,
  
  (1) You are missing ^ in command
in Mixmonitor.In your case, It should be something like
MixMonitor(IND_PRI/${CDR(accountcode)}/OUT/${STRFTIME(${EPOCH},,%Y-%m)}/${STRFTIME(${EPOCH},,%d)}/${MIXMONITOR_FILENAME}.wav,m,/root/flac.sh
^${MIXMONITOR_FILENAME}.wav) 
  
  (2) You are passing just file
name as a parameter in your script and not a full path
for file. (Do you handle full path in a script?)


  --Satish Barot

Ahmedabad, India
  


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-- 
  

  
  
  Fabio
Moretti  
  Gerente de Sistemas 
   www.tecytal.com  
   0800 8780 
(+598) 248 77921  
  

  

  


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Re: [asterisk-users] Problem with CEL logging and channel bridging

[Fabio Moretti]

  
  

Il 13/06/2013 11:31, Fabio Moretti
  scrisse:


  Hi, I've already post this to the forum three days ago, sorry if it's
sounds like a crosspost, but I've got no replies, so I'm trying other
channels :)


ok, definitely CEL is a big question mark for most of us.

can someone point me to in deep CEL documentation or to an open
source code that use it so I can study more? not asterisk code,
please, I tried but I find really hard find how and then events are
generated.

thanks

-- 
  

  
  
  Fabio
Moretti  
  Gerente de Sistemas 
   www.tecytal.com  
   0800 8780 
(+598) 248 77921  
  

  

  


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Re: [asterisk-users] Problem with CEL logging and channel bridging

[Fabio Moretti]

  
  
Il 17/06/2013 11:13, Matthew Jordan scrisse:

  

  
Since you know that DAHDI/i1/96034296-30a3 is
  in a bridge with Local/1004@from-queue-00019c34;1 and
  Local/1004@from-queue-00019c34;2is in a bridge
  withIAX2/issuegroup-17175, you automatically know that
  DAHDI/i1/96034296-30a3 and IAX2/issuegroup-17175 can
  communicate (at least once everyone has Answered). The
  system you build on top of CEL has to understand the
  semantics of Local channels and tie the two together.
  
  
  
  Matt

  

matt, thank you very much. in fact I was wondering if
local-channel;1 and local-channel;2 have to be considered as "one"
channel or not.
Can I ask you if there's a in deep documentation of how channel and
events are generated/destroyed? I'm trying to find the time to
study, I'd like to generate a billing script based on CEL and a
graphical interface for visualizing calls history.

really thank you

-- 
  

  
  
  Fabio
Moretti  
  Gerente de Sistemas 
   www.tecytal.com  
   0800 8780 
(+598) 248 77921  
  

  

  


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[asterisk-users] Problem with CEL logging and channel bridging

[Fabio Moretti]

Hi, I've already post this to the forum three days ago, sorry if it's
sounds like a crosspost, but I've got no replies, so I'm trying other
channels :)

This is the link to the forum post if someone prefer to reply here:
http://forums.asterisk.org/viewtopic.php?f=1t=86985

I'm using Asterisk 1.8.20.0 (the freepbx build) with CEL logging
activated. I'm using CEL because in our pbx we have different queues and
trunks serving different customers (we are an inbound call center) and
we need to detect when and how we have to bill our customers.
I'm facing an issue with the call transfer, for example I have:
- call entering a queue
- operator answer the call
- operator make an outgoing call to reach the customer
- operator put in communication the ingoing call with the outgoing
this result in various channel to be created/destroyed, and I'm using
bridge events to detect what is going on with the call. In this case I
have (I've hidden CHAN_START,ANSWER and HANGUP events because they have
no useful information in this case):

++---+-+---+-+--+-+-+--+

| id | eventtype | eventtime   | exten | context | 
channame | appname | appdata | peer 
|

++---+-+---+-+--+-+-+--+

| 965224 | BRIDGE_START  | 2013-06-10 10:15:18 | 20| ext-queues  | 
DAHDI/i1/96034296-30a3   | Queue   | 20,t,,  | 
Local/1004@from-queue-00019c34;1 |

| 965226 | BRIDGE_START  | 2013-06-10 10:15:18 | s | macro-dial-one  | 
Local/1004@from-queue-00019c34;2 | Dial| SIP/1004,,trM(auto-blkvm) | 
SIP/1004-40ce|

| 965340 | BRIDGE_UPDATE | 2013-06-10 10:16:08 | s | macro-dialout-trunk | 
Local/1004@from-queue-00019c34;2 | Dial| IAX2/issuegroup/110,300,| 
IAX2/issuegroup-17175|

| 965513 | BRIDGE_END| 2013-06-10 10:18:15 | 20| ext-queues  | 
DAHDI/i1/96034296-30a3   | Queue   | 20,t,,  | 
Local/1004@from-queue-00019c34;1 |

| 965515 | BRIDGE_END| 2013-06-10 10:18:15 | s | macro-dialout-trunk | 
Local/1004@from-queue-00019c34;2 | Dial| IAX2/issuegroup/110,300,| 
IAX2/issuegroup-17175|

++---+-+---+-+--+-+-+--+


The first BRIDGE_START is the connection between the inbound call
(DAHDI/i1/96034296-30a3) and the local phone
(Local/1004@from-queue-00019c34;1), the second BRIDGE_START is the
connection between the local phone (Local/1004@from-queue-00019c34;2)
and the outgoing call (SIP/1004-40ce) that is going out by a IAX trunk.
After that I have a BRIGDE_UPDATE event where no field make me know
which channel is being updated, I only have the channame
(Local/1004@from-queue-00019c34;2) that is the channel being bridged out
and the outgoing channel (IAX2/issuegroup-17175), but I have no
information that in fact the ingoing call (DAHDI/i1/96034296-30a3) is
being bridged to the outgoing channel.
I have no other event (TRANSFER or something like that) to know what is
going on.

In my cel.conf I have:

apps=queue
events=CHAN_START,CHAN_END, APP_START,APP_END, ANSWER,HANGUP,
BRIDGE_START,BRIDGE_END,BRIDGE_UPDATE,
BLINDTRANSFER,ATTENDEDTRANSFER,TRANSFER, PICKUP, FORWARD,
PARK_START,PARK_END, LINKEDID_END

Should I change something in my configuration or it's wrong to rely on
bridges to follow a call? What kind of event should I follow to be sure
to catch where the call is going?

Thank you for any suggestion!


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Re: [asterisk-users] A problem with IAX2

[Fabio Moretti]

  
  
hi,

I've solved various iax2 problem mentioning calltoken when I put
these lines in the iax configuration:

requirecalltoken=no
calltokenoptional=0.0.0.0/0.0.0.0

bye

Il 11/06/2013 19:25, Mordechay Kaganer
  scrisse:


  B.H.
  On Jun 11, 2013 5:15 PM, "Steve Totaro" stot...@totarotechnologies.com
wrote:




 On Tue, Jun 11, 2013 at 8:32 AM, Mordechay Kaganer mkaga...@gmail.com
wrote:

 B.H.

 Hello!

 We have several Asterik boxes that are connected to
PSTN using PRI cards and they are interconnected using IAX2
trunks so that incoming calls are delivered from PSTN to the
servers they belong to.

 In past we were using asterisk 1.4 on the server that
is receiving IAX connections and everything worked as expected.
Recently, we have switched to a newer box with asterisk 1.8.22
and then we began to experience sometimes a strange problem:

 At some point of time, incoming IAX connections begin
to get refused by the server and we get the following messages
in the logs:

 WARNING[] chan_iax2.c: Too much delay in IAX2
calltoken timestamp from address X.X.X.X

 where X.X.X.X is the IP of the PSTN-IAX gateways
and all the incoming calls start to be rejected.

 Direct PSTN calls (both incoming and outgoing) to the
same server work OK. The only solution that helps is to kill the
asterisk and restart it.

 All the servers are connected to the same LAN segment,
with gigabit switch, there is no problems with the network. No
packet loss.

 There's already bug report present with very similar
issue, but it is "suspended" and, like stated there, the problem
is very hard to reproduce.

 See:https://issues.asterisk.org/jira/browse/ASTERISK-21762


 -- 
  NOW!


 Use SIP and never look back.

 Thanks,
 Steve Totaro

 --

  Thanks, that's what i actually going to do.
  But does this mean that IAX is obsolete? Actually i have
selected IAX in the first place because it looks like more
"native" for asterisk, so i thought it would be more suitable as
a protocol to interconnect asterisk boxes...
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-- 
  

  
  
  Fabio
Moretti  
  Gerente de Sistemas 
   www.tecytal.com  
   0800 8780 
(+598) 248 77921  
  

  

  


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[asterisk-users] (no subject)

[fabio alves]

Good morning gentlemen, is my first post in the list, now I'm starting asterisk 
wanted to have your help for some questions.



Well the first function is as follow me. Here
 I will demonstrate how this configuration follow me on my 
extensions.conf but it is not working, and do not know why, but 
something is missing?

You must set up followme.conf ?



What
 I want is that the follow-me is enabled for any of the extensions 
within the same context, like if I am absent from my table and go to 
extension 2801 DataCenter where I need to spend all afternoon and I will
 have the extension 2820 which enabled me to follow this extension and after 
back to my desk withdraw follow me.
; Ativa Siga-me incondicional



[sigame-on]exten  = _*71*.,1,NoCDR()

exten =  _*71*.,2,Set(DB(CF/${CALLERID(num)})=${EXTEN:4})

exten = _*71*.,3,Playback(call-fwd-unconditionalforextension)

exten = _*71*.,4,SayDigits(${CALLERID(num)}) 

exten = _*71*.,5,Playback(is-set-to)

exten =  _*71*.,6,SayDigits(${EXTEN:4}) 

exten = _*71*.,7,Playback(vm-saved)

exten =  _*71*.,8,Playback(beep)

exten = _*71*.,9,Hangup



; Desativa o siga-me incondicional



[sigame-off]exten  = _*72*,1,NoCDR()

exten = _*72*,2,DBdel(CF/${CALLERID(num)})

exten = _*72*,3,Playback(cancelled) exten = _*72*,4,Playback(beep)

exten = _*72*,5,Hangup







Bom, agora vamos ao pulo do gato, esse passo é muito importante pois é  
ele quem verifica se existe ou não o siga-me para o ramal.



Vamos ao contexto:



[disca]

exten = _3XXX,1,Noop(CF/${EXTEN})

exten =  _3XXX,2,Set(siga=${DB(CF/${EXTEN})})

exten = _3XXX,3,Dial(SIP/${siga},30,Ttw)

exten = _3XXX,4,Dial(SIP/${EXTEN}) ;  Unconditional forward

exten = _3XXX,5,Hangup

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[asterisk-users] TLS re-negotiation attack on SIP/TLS of Asterisk?

[Fabio Pietrosanti (naif)]

Hi all,

i read about the TLS-RENEGOTIATION vulnerability:

http://www.educatedguesswork.org/2009/11/understanding_the_tls_renegoti.html
http://www.sslshopper.com/article-ssl-and-tls-renegotiation-vulnerability-discovered.html
www.phonefactor.com/sslgapdocs/Renegotiating_TLS.pdf

Does the Asterisk 1.6/1.8 SIP/TLS implementation suffer from the TLS
Renegotiation vulnerability or the TLS-renegotiation it's disabled by
default, in how OpenSSL is used?

Fabio Pietrosanti

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[asterisk-users] Faxing with asterisk

[Fabio Mosti]

Hi All,

I need to setup asterisk to receive fax.

I'm try Spandsp (opensource) and Attrafax (commercial) both on
asterisk 1.4.23) but the results are disappointing.
with spandsp many times the fax arrives cut.
with Attrafax i have some problem.

Anyone have any idea or solution (Opensource or commercial) to suggest me ?

Best Regards

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Re: [asterisk-users] Faxing with asterisk

[Fabio Mosti]

2009/2/16 Steve Underwood ste...@coppice.org:

 You don't indicate the kind of setup you are using.

I use asterisk (Spandsp)  with a IAX2 trunk (ethernet connection) to
another asterisk (zap).

client-asterisk (Spandsp)-asterisk (zap)-fax

 Regards,
 Steve

Best Regards,

Fabio



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[asterisk-users] Incoming URL handling Problem (Asterisk problem ?)

[Fabio Mosti]

Hello,

I use an Asterisk box with the following configuration:

Operating System : linux Fedora Core 4  (2.6.17-1.2142_FC4smp #1)

Asterisk 1.4.18

I use the following asterisk command to send url to client :
Dial(IAX2/ciwww/[EMAIL 
PROTECTED],,,https://xx..it/es/crm/dashboard.php?codice_ordine=xxx-xx-xxx;)

I've a problem using the Incoming URL handling feature with my IAX2
client softphone.


I've dumped my lan traffic and I've filtered the correct URL:

(this is part of my dump (libcap/ASCII)

Begin
..pU..U...U.UU.U.UU...UT.U..T..TUU.U...U.U...UU.UU..UUU.UU.UU.TU.UU.U.UU.UUUU.UU..U.TUTT.UUUT.U.U.U..U...UU.UU.U.UU.T.UU..U

..U...AD.UUUU.U.U...UUU...U.UU.UU.UU..U..U.U...UU.U..UUU..UU.U.....U.UUUU..UUU.U..U..U..UUU...UUU.U...U.U.UU.T.UU.UUU.U
...%.
.p...UU.UUU...UU...UUTTTUUU......UU...UU...UUU.

..
...(...
...shttps://xx..it/es/crm/dashboard.php?codice_ordine=xxx-xx-xxx;..
.s...
...u..
dd
.I*P..
End


the incoming call works fine, but I can't see the url.


When the client (Zoiper Biz softphone 2.16 on  Windows Vista  Windows
2000) receives the call, it does not open any browser
and it does not generate any warning.

Possible Asterisk Problem ?

Can you help me ?

Best Regards,

Fabio

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[asterisk-users] problem about TDM400P ringback detection

[Fabio Antonini]

Hi to all
I'm a new user of TDM400P card. The configuration is OK and I have no problem 
for incoming call. When I try to place a outgoing call towards a PSTN number 
the call is not always answered. In other words TDM400P seems to not understand 
that the call has been accepted from the remote party. In other cases 
(different extension) the call is accepted succesfully. In my opinion TDM400P 
DSP can detect if the RIngBack tone is present or not. The call is declared 
answered if the ringback tone stops to occur. So my feeling is that the 
ringback ton detection fails form my country (italy) in most cases even if 
sometime it works fine. On Asterisk cli I can read
1) when the call is succesfully accepted

host529*CLI 
Channel Zap/4-1 was answered.
Launching Wait(100) on Zap/4-1
host529*CLI 

2) when the call is not succesfully accepted (after the timeout the call is 
properly terminated) even if the call has been accepted by the remote party and 
both the users can talk and listen each other.

host529*CLI 
-- Hungup 'Zap/4-1'
host529*CLI 

Please help me to understand what I miss in the configuration. Does the TDM400P 
support the italian tone sequences? I think yes. 
Which parameters can I tune in order to increase the tone detection performance?

Every help will greatly appreciated.


Regards
Fabio Antonini
SW Designer, Ph.D 
Kasko Networks S.r.l. 
Piazza Regina Margherita 7, 67100 L'Aquila (ITALY) 
ph/fax: +39 862200460 mob:+39 3939261941, +39 3280451965 
email: [EMAIL PROTECTED], [EMAIL PROTECTED]
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[asterisk-users] Conference rooms

[Fabio Cappelletti]

I all,
I have a question about the  use of conference rooms: can I, with a Voip
telephone or softphone call some other telephone and invite them in a
conference room? I read a lot of documentations about asterisk, but i
can't find any example !

Thanks, best regard

Fabio


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[asterisk-users] IVR and MySQL

[Fabio Ardeola]

Hi

Does somebody know if I can save the answers made by
the caller person on the IVR menu in a MySQL Database?
If yes, can I save the CallerID as well?

Thanks,
Fabio


  

Luggage? GPS? Comic books? 
Check out fitting gifts for grads at Yahoo! Search
http://search.yahoo.com/search?fr=oni_on_mailp=graduation+giftscs=bz

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Re: [asterisk-users] IVR and MySQL

[Fabio Ardeola]

James / Atis / Thiago

Let say that the user entry during the call is a
reference number of a house to rent. Would be possible
to check if the reference number is a valid entry on
the MySQL database and then base on its answer define
the next menu item on the IVR menu.

Thanks,
Fabio


--- James FitzGibbon [EMAIL PROTECTED]
wrote:

 On 8/14/07, Atis [EMAIL PROTECTED] wrote:
 
  That's possible, but i wouldn't recommend on large
 production system.
  Using MySQL you would need to connect and
 disconnect all the time, and
  it takes resources.. I would suggest to append
 that info to CDR
  userfield (if you are storing your CDR in MySQL),
 and run periodically
  some script that extracts them. Of course it's
 more complex, but that
  would be my way.
 
 
 If the data you wish to store is more complex than
 stuffing in the CDR
 userfield would allow, you can always call out to an
 AGI which can write the
 data to whatever file format you want for later
 loading into a database.
 
 If you used FastAGI and a pre-forking AGI server
 model, you could even take
 the database connection hit when the AGI server
 starts.  The per-call cost
 would then be the cost to establish the socket
 connection to the AGI server
 from Asterisk, the cost to perform the SQL inserts
 over an established
 database connection, plus whatever other calculation
 or transformation you
 needed to do before doing the insert.
 
 That architecture would hold up under a fairly large
 load.  Perl's
 Asterisk::FastAGI framework lets you specify the
 number of pre-forked
 children to launch, plus you can tell each child to
 exit (spawning a
 replacement for the pool) after processing a certain
 number of
 transactions.  It's very similar to the Apache
 prefork model.
 
 -- 
 j.
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Park yourself in front of a world of choices in alternative vehicles. Visit the 
Yahoo! Auto Green Center.
http://autos.yahoo.com/green_center/ 

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RE: [asterisk-users] Blind transfer 3/4 digits

[Fabio]

Hi all,

Ronald, if you are using #, try adjusting the featuredigittimeout
parameter in features.conf.This is the max time between digits for feature
activation. If is small, * could dial the wrong number, in your case 601
instead of 6014.

I think that you are not using # while your are using snom, because you said
that you needed to dial # in order to finish the transfer (this it's no
necessary for *). Or snom is catching the # and driving the transfer.

fabay


-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] nombre de Ronald
Wiplinger
Enviado el: Sábado, 02 de Septiembre de 2006 10:40 p.m.
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Asunto: Re: [asterisk-users] Blind transfer 3/4 digits


Tim St. Pierre wrote:
 Are you using # to transfer?  If so, it's not sending it as a new call,
it's
 just sending asterisk digits using whatever DTMF mode.  Asterisk parses
these
 based on a first match in the dialplan.  Make sure that the longer
 extension numbers are loaded first in the dialplan.



That is a good thought. I can remember that the docs said that you
cannot force the order of the dialplan, except with includes. I will try
that way.
However, I have doubts as well. If you are right, than why snom phone
does not have this problem? Would not here also the first match count?

bye

Ronald
 -Tim

 On September 2, 2006 20:12, Ronald Wiplinger wrote:

 Kevin Smith wrote:

 Dialing a number and transferring a number are two different things.
 And no offense, you are not really providing a lot of details along
 with your problem. So you can dial the numbers but not transfer from
 one to the other.

 I was not thinking that it would be too much difference. Therefore I
 also do not know what more info could help to distinguish the problem. I
 hardly can post my entire configuration.


 What does the CLI say when you try the transfer? That would provide a
 lot of information that could clue you in to what is going on.

 You hit another problem with that. I hardly see here anything anymore.
 The messages fly by so fast,  Especially annoying messages:
  chan_sip.c:10888 handle_request_register: Registration from
 'sip:192.168.250.20' failed for '192.168.250.244' - Username/auth name
 mismatch
  -- Got SIP response 486 Busy Here back from 192.168.250.244
  -- Got SIP response 400 Bad Request back from xx.xx.xx.126
 NOTICE[5936]: chan_sip.c:9600 handle_response_register: Failed to
 authenticate on REGISTER to '[EMAIL PROTECTED]' (Tries 3)
 .

 It would be nice to filter the CLI for such investigation for a moment.


 What type of phones are you using? Some phones have the ability to
 pattern match and wait for a certain number of seconds before sending
 the number to asterisk. For example. On our Polycom phones a user has
 3 seconds (between digits) to enter in 10 digits. This could be where
 most of your problem is.

 That is a very good point and I will contact the manufacturer of these
 no-name phones.


 My guess the problem lies with the Phones, not Asterisk form the
 information you provided.

 I disagree with that! Why Asterisk treats dialing and transfer
 different. That makes not really sense, does it?

 bye

 Ronald


 Kevin

 Ronald Wiplinger wrote:

 David Gagnon wrote:

 Ronald,

 You seem to be a little bit angry about VoIP. If so, I could give
 you my old Nortel system. Does this would make you happy?

 David

 David,

 I am not angry about VoIP, but please send my your old Nortel system
 !

 I just do not understand why I can DIAL 601 and 6014, but not use
 blind transfer. Is the question too difficult?

 I am sure there is somewhere a switch to say, wait two seconds (as
 for dialing) before you assume it is a complete number.
 It is also strange that snom phone can do it correct, because it uses
 the ok key.


 -Message d'origine-
 De : [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] De la part de Ronald
 Wiplinger
 Envoyé : 2 septembre 2006 04:20
 À : Asterisk Users Mailing List - Non-Commercial Discussion
 Objet : Re: [asterisk-users] Blind transfer 3/4 digits

 Anthony Rodgers wrote:

 With respect, the problem is with your numbering plan..

 This answer is therefore totally nonsense !!! (With all respect!!!)


 Both answers have actually not lead to any step further, but to more
 messages. I use to refer to such answers as NON-ANSWERS.
 Please only reply if and really only if you know a solution for the
 problem! Thanks for your understanding.

 bye

 Ronald - again, I am not angry at all.


 WHERE do you see a problem in the numbering plan?
 I see the problem in ASTERISK, because it does not wait for the last
 digit!!!
 Where can I set that it waits for it?

 The beauty on voip IS that you can have different length and
 overlapping, 

 bye

 Ronald


 CP

 On 1-Sep-06, at 10:37 PM, Ronald Wiplinger wrote:

 I found a problem in blind transfer:

 I have an extension number 601 and I have an extension 6014 

 If I get 

RE: [asterisk-users] Asterisk with PABX

[Fabio]

Hi Wederson,

- start reading about ivr and see the examples at voip-info.org.

- If you plans to use the analog extension on you pbx you need to use a fxo
adapter or a pc card (for example tdm400p) on * side, then you could finish
your ivr script with Dial application and call your old pbx. (very trivial
example). For digital (ex. T1/E1 links) the idea its the same, but you need
another hardware.

cheers,

Fabay


-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] nombre de Tux Wi-FI
Enviado el: Lunes, 28 de Agosto de 2006 02:10 p.m.
Para: asterisk-users@lists.digium.com
Asunto: [asterisk-users] Asterisk with PABX


Hi,

It would like to know if it is possible to establish connection
asterisk (IVR) with traditional PABX.

My company possesss a common structure of PABX currently and is
needing to implement IVR with ASTERISK, but for the time being she
would like to keep the structure of the normal PABX.

It pressures 1 for Support, asterisk directs the linking for branch
1867 of the common PABX.It is possible and as it would be?

Thanks.

Wederson R.
CeBoLINhA

--
[]´s

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http://www.tuxwifi.com.br
msn: [EMAIL PROTECTED]
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RE: [asterisk-users] intel vs amd motherboards

[Fabio]

I think it's the same,
10 calls in 200ms = 50 calls in 1s
because 1s = 5 x 200ms

IMHO, is better to use seconds as period, because is more ease to compare
rate speeds of each codec that are in bits per second.

fabio

-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] nombre de C F
Enviado el: Domingo, 09 de Julio de 2006 06:57 p.m.
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Asunto: Re: [asterisk-users] intel vs amd motherboards


Thanks for that Tzafrir. Why does it ignore the secend CPU?

BTW, on a side note on this topic, how can one calculate simultaneous
transcoded channels using show transalation?

In the case where it tells me 17 ms for encoding and 4 for decoding,
that gives me 21ms per channel, in what time frame can I squeeze in
how many channels before the calls start becoming  intolerable? In
other words should I aim for a 200ms time frame which means that I
will get around 10 channels? or can I aim for a full second? which
will give me around 50 channels?

Thank You

On 7/9/06, Tzafrir Cohen [EMAIL PROTECTED] wrote:
 On Sun, Jul 09, 2006 at 05:07:16AM -0400, C F wrote:
  Tzafrir, are you trying to tell me that I can realy do double on the
  intel becuase the second CPU will do it?

 In the ideal case you'll get double performance with two CPUs. In
 theory.

 A case of many concurrent calls is basically something that can be
 easily parallelized. So in theory nothing stops you from getting
 something closer to double performance. I don't know how close reality
 is to that nice theory.

 I only remarked that 'show translations' totally ignores the second CPU.

 --
 Tzafrir Cohen  sip:[EMAIL PROTECTED]
 icq#16849755   iax:[EMAIL PROTECTED]
 +972-50-7952406
 [EMAIL PROTECTED]  http://www.xorcom.com
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RE: [asterisk-users] intel vs amd motherboards

[Fabio]

looking on the code, I think that the time on the table is the time that
needed the CPU in order to translate 1 second of media.

So, on the case of calls with ulaw - alaw translation (1 ms in each
translation), this CPU could sustain, theoretically, 500 calls without delay
(two simultaneous translations, ulaw to alaw, and alaw to ulaw are needed,
to maintain the conversation full duplex).

please, somebody could confirm this ?.

fabio


-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] nombre de C F
Enviado el: Lunes, 10 de Julio de 2006 03:46 p.m.
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Asunto: Re: [asterisk-users] intel vs amd motherboards


But my question is, those that mean that it will take 1 second to
convert 50 channels? if so do I get a 1 second latency when coverting
50 channels?

On 7/10/06, Fabio [EMAIL PROTECTED] wrote:
 I think it's the same,
 10 calls in 200ms = 50 calls in 1s
 because 1s = 5 x 200ms

 IMHO, is better to use seconds as period, because is more ease to compare
 rate speeds of each codec that are in bits per second.

 fabio

 -Mensaje original-
 De: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] nombre de C F
 Enviado el: Domingo, 09 de Julio de 2006 06:57 p.m.
 Para: Asterisk Users Mailing List - Non-Commercial Discussion
 Asunto: Re: [asterisk-users] intel vs amd motherboards


 Thanks for that Tzafrir. Why does it ignore the secend CPU?

 BTW, on a side note on this topic, how can one calculate simultaneous
 transcoded channels using show transalation?

 In the case where it tells me 17 ms for encoding and 4 for decoding,
 that gives me 21ms per channel, in what time frame can I squeeze in
 how many channels before the calls start becoming  intolerable? In
 other words should I aim for a 200ms time frame which means that I
 will get around 10 channels? or can I aim for a full second? which
 will give me around 50 channels?

 Thank You

 On 7/9/06, Tzafrir Cohen [EMAIL PROTECTED] wrote:
  On Sun, Jul 09, 2006 at 05:07:16AM -0400, C F wrote:
   Tzafrir, are you trying to tell me that I can realy do double on the
   intel becuase the second CPU will do it?
 
  In the ideal case you'll get double performance with two CPUs. In
  theory.
 
  A case of many concurrent calls is basically something that can be
  easily parallelized. So in theory nothing stops you from getting
  something closer to double performance. I don't know how close reality
  is to that nice theory.
 
  I only remarked that 'show translations' totally ignores the second CPU.
 
  --
  Tzafrir Cohen  sip:[EMAIL PROTECTED]
  icq#16849755   iax:[EMAIL PROTECTED]
  +972-50-7952406
  [EMAIL PROTECTED]  http://www.xorcom.com
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RE: [asterisk-users] intel vs amd motherboards

[Fabio]

as tzafrir already said, you have the power of two processors in one .. .
and you dont need more stoves ;))


-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] nombre de C F
Enviado el: Viernes, 07 de Julio de 2006 12:29 a.m.
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Asunto: Re: [asterisk-users] intel vs amd motherboards


cat /proc/cpuinfo on amd:
 cat /proc/cpuinfo
processor   : 0
vendor_id   : AuthenticAMD
cpu family  : 15
model   : 47
model name  : AMD Athlon(tm) 64 Processor 3200+
stepping: 2
cpu MHz : 2000.000
cache size  : 512 KB
fpu : yes
fpu_exception   : yes
cpuid level : 1
wp  : yes
flags   : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca
cmov
pat pse36 clflush mmx fxsr sse sse2 syscall nx mmxext fxsr_opt lm 3dnowext
3dnow
 pni lahf_lm
bogomips: 4025.55
TLB size: 1024 4K pages
clflush size: 64
cache_alignment : 64
address sizes   : 40 bits physical, 48 bits virtual
power management: ts fid vid ttp tm stc

cat /proc/cpuinfo on intel:
cat /proc/cpuinfo
processor   : 0
vendor_id   : GenuineIntel
cpu family  : 15
model   : 6
model name  : Intel(R) Pentium(R) D CPU 2.80GHz
stepping: 2
cpu MHz : 2800.353
cache size  : 2048 KB
physical id : 0
siblings: 2
core id : 0
cpu cores   : 2
fdiv_bug: no
hlt_bug : no
f00f_bug: no
coma_bug: no
fpu : yes
fpu_exception   : yes
cpuid level : 6
wp  : yes
flags   : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge
mca cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe nx
lm pni monitor ds_cpl vmx cid cx16 xtpr lahf_lm
bogomips: 5609.03

processor   : 1
vendor_id   : GenuineIntel
cpu family  : 15
model   : 6
model name  : Intel(R) Pentium(R) D CPU 2.80GHz
stepping: 2
cpu MHz : 2800.353
cache size  : 2048 KB
physical id : 0
siblings: 2
core id : 1
cpu cores   : 2
fdiv_bug: no
hlt_bug : no
f00f_bug: no
coma_bug: no
fpu : yes
fpu_exception   : yes
cpuid level : 6
wp  : yes
flags   : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge
mca cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe nx
lm pni monitor ds_cpl vmx cid cx16 xtpr lahf_lm
bogomips: 5600.89


On 7/6/06, Fabio [EMAIL PROTECTED] wrote:
 Hi CF,

 please could you to include CPUs specs, thanks in advance.

 Fabio


 -Mensaje original-
 De: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] nombre de C F
 Enviado el: Jueves, 06 de Julio de 2006 04:32 p.m.
 Para: Asterisk Users Mailing List - Non-Commercial Discussion
 Asunto: Re: [asterisk-users] intel vs amd motherboards


 I have recently build 2 machines, one with an Intel Pentium Dual Core
 CPU, and one SATA HDD, and the other with a single AMD 64 bit CPU, and
 a RAID 1 w 2 SATA HDD. Both costed the same even though the AMD had 2
 HDDs. Here are the show translations from both:

 Intel Dual Core machine:
 pbx*CLI show translation
  Translation times between formats (in milliseconds)
   Source Format (Rows) Destination Format(Columns)

  g723   gsm  ulaw  alaw  g726 adpcm  slin lpc10  g729 speex  ilbc
g723 - - - - - - - - - - -
 gsm - - 2 2 2 2 1 517 -17
ulaw - 2 - 1 2 2 1 517 -17
alaw - 2 1 - 2 2 1 517 -17
g726 - 2 2 2 - 2 1 517 -17
   adpcm - 2 2 2 2 - 1 517 -17
slin - 1 1 1 1 1 - 416 -16
   lpc10 - 3 3 3 3 3 2 -18 -18
g729 - 4 4 4 4 4 3 7 - -19
   speex - - - - - - - - - - -
ilbc - 3 3 3 3 3 2 618 - -

 AMD 64 bit machine:
 pbx*CLI show translation
  Translation times between formats (in milliseconds)
   Source Format (Rows) Destination Format(Columns)

  g723   gsm  ulaw  alaw  g726 adpcm  slin lpc10  g729 speex  ilbc
g723 - - - - - - - - - - -
 gsm - - 2 2 2 2 1 313 -12
ulaw - 3 - 1 2 2 1 313 -12
alaw - 3 1 - 2 2 1 313 -12
g726 - 3 2 2 - 2 1 313 -12
   adpcm - 3 2 2 2 - 1 313 -12
slin - 2 1 1 1 1 - 212 -11

RE: [asterisk-users] IVR - Automatic Attendant database query

[Fabio]

one example using realtime:

on your database:

table: users
field: iduser id
fiel: enabled1 if user enabled

extconfig.conf

mydatabase = mysql,asterisk,users

extensions.conf

exten = s,n,Read(userid,please-your-id,3)  
exten = s,n,Realtime(mydatabase,id,${userid},user_)
exten = s,n,GotoIf($[${user_enabled} = 1]?s-login,1)
exten = s,n,PlayBack(ByBy)
exten = s,n,HangUp

exten s-login,1,PlayBack(HiMan)
...

-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] nombre de Dirk
Enrique Seiffert
Enviado el: Viernes, 07 de Julio de 2006 10:16 a.m.
Para: asterisk-users@lists.digium.com
Asunto: [asterisk-users] IVR - Automatic Attendant database query


Hello,

I am new to Asterisk, looking for a PBX solution doing and automatic
response based on a database query. I spent some hours googling, reading
manuals without too much luck. (Maybe I am just blind ...)

What we want to reach:

Caller dials in, gets prompted to dial an extension depending on the
information he is looking for. Lets say Dial 1 if you want to know if
your account is active. After dialing the number caller gets prompted to
dial his personal code. Based on this code we need do to a query to an
external Sybase database. Depending on the query result we will play a
recording to the caller. Yes, your account is active, Sorry, your
account has been disabled...

I couldn't find any examples on the auto attendent reading responses from
a database, though it looks like a common task to me. Can anybody provide
some hints, links, directions or experiences for this kind of
configuration?

Thanks a lot

Dirk


-- 
Dirk Enrique Seiffert - Lintec S.A.
Ed. Torre del Reloj - Of. 401
Plaza de los Coches, Centro
Cartagena - Colombia
http://www.lintecsa.com

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RE: [asterisk-users] intel vs amd motherboards

[Fabio]

Hi CF, 

please could you to include CPUs specs, thanks in advance.

Fabio


-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] nombre de C F
Enviado el: Jueves, 06 de Julio de 2006 04:32 p.m.
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Asunto: Re: [asterisk-users] intel vs amd motherboards


I have recently build 2 machines, one with an Intel Pentium Dual Core
CPU, and one SATA HDD, and the other with a single AMD 64 bit CPU, and
a RAID 1 w 2 SATA HDD. Both costed the same even though the AMD had 2
HDDs. Here are the show translations from both:

Intel Dual Core machine:
pbx*CLI show translation
 Translation times between formats (in milliseconds)
  Source Format (Rows) Destination Format(Columns)

 g723   gsm  ulaw  alaw  g726 adpcm  slin lpc10  g729 speex  ilbc
   g723 - - - - - - - - - - -
gsm - - 2 2 2 2 1 517 -17
   ulaw - 2 - 1 2 2 1 517 -17
   alaw - 2 1 - 2 2 1 517 -17
   g726 - 2 2 2 - 2 1 517 -17
  adpcm - 2 2 2 2 - 1 517 -17
   slin - 1 1 1 1 1 - 416 -16
  lpc10 - 3 3 3 3 3 2 -18 -18
   g729 - 4 4 4 4 4 3 7 - -19
  speex - - - - - - - - - - -
   ilbc - 3 3 3 3 3 2 618 - -

AMD 64 bit machine:
pbx*CLI show translation
 Translation times between formats (in milliseconds)
  Source Format (Rows) Destination Format(Columns)

 g723   gsm  ulaw  alaw  g726 adpcm  slin lpc10  g729 speex  ilbc
   g723 - - - - - - - - - - -
gsm - - 2 2 2 2 1 313 -12
   ulaw - 3 - 1 2 2 1 313 -12
   alaw - 3 1 - 2 2 1 313 -12
   g726 - 3 2 2 - 2 1 313 -12
  adpcm - 3 2 2 2 - 1 313 -12
   slin - 2 1 1 1 1 - 212 -11
  lpc10 - 3 2 2 2 2 1 -13 -12
   g729 - 4 3 3 3 3 2 4 - -13
  speex - - - - - - - - - - -
   ilbc - 4 3 3 3 3 2 414 - -


This shows that the AMD 64 bit is worth much more than just the price
difference.


On 7/6/06, Andrew Kirch [EMAIL PROTECTED] wrote:


  -Original Message-
  From: [EMAIL PROTECTED] [mailto:asterisk-users-
  [EMAIL PROTECTED] On Behalf Of Don
  Sent: Wednesday, July 05, 2006 11:00 AM
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  Subject: Re: [asterisk-users] intel vs amd motherboards
 
  If you want to handle, lets say 1000 calls or more at the same time,
 you
  should of course use a better processor. In my opinion, it doesn't
 matter
  whether you use Intel or AMD, because you said it will be a small
  Asterisk.
 
  In the world of asterisk...Intel or AMD really doesn't make a
 difference
  However AMD can do more for less money...
 
  I think you should concentrate more on a descent mainboard for
 whichever
  powerplant you chose to shove in it...


 Due to the recent nightmares I've had with the Asus K8N and the Dell
 PowerEdge 830, recommendations here would be greatly appreciated.  What
 board, what interfaces (digium/sangoma) and results as well as caveats.


 Andrew
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RE: [asterisk-users] for you guys setting up customer offices...

[Fabio]

Polycom 300, SPA 841/941 (841 is out of market...), pap2 (2 x fxs ata)

Fabio

-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] nombre de Shaun
Enviado el: Jueves, 06 de Julio de 2006 04:45 p.m.
Para: asterisk-users@lists.digium.com
Asunto: [asterisk-users] for you guys setting up customer offices...


What brand/model phones are you using.

-- 

~Shaun 



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RE: [asterisk-users] audio session start delay

[Fabio]

Hi Luca, 
are you using SIP reinvite ?

post a bit mor information (sip.conf)


Fabio

-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] nombre de Luca Corti
Enviado el: Jueves, 06 de Julio de 2006 01:59 p.m.
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Asunto: [asterisk-users] audio session start delay


Hello everyone,

I've set up an asterisk box with basic PBX features (DiD, MoH, MoT,
Blind and Attended Call Transfer, PickUp, ecc.) for 10 Cisco 79xx (7912
and 7960) with SIP image (8.0). PSTN gateway is done using a Cisco
AS5350 with two ISDN PRIs connected to Asterisk via SIP. Between the
phones and the PBX I have a router doing NAT and a 4mbit synchronous
line.

Sometimes when calling between extensions, after successful signaling,
there is a delay of 10 seconds before any audio is heard by both
parties.

Do you know what can cause this behaviour? Is this more likely to be a
phone or an Asterisk issue?

thanks

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RE: [Asterisk-Users] Help with IVR menu.

[Fabio]

Hi Zerthimon

take a look at http://www.voip-info.org/wiki/view/Asterisk+tips+ivr+menu

regards

Fabio


-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] nombre de Lior Goikhburg
Enviado el: Lunes, 03 de Julio de 2006 07:48 a.m.
Para: asterisk-users@lists.digium.com
Asunto: [Asterisk-Users] Help with IVR menu.


Greetings,

I'm an asterisk newbie. I have no problem making flat IVR trees but this
one is too confusing.

I'm trying to set up a dial plan with a multi-layered IVR:
I've been asked to create a menu for my company where one could dial a menu
entry like: for sales press 1, for management 2, etc..., then if the
extension is busy the system would tell The extension is busy, press 0 for
the operator, 1 to wait on the line or 2 to leave a message. Then the
caller will have to press either 0 for operator, 1 to wait or 2 to leave
voice mail. In case the extension is unavailable the caller would get The
extension is unavailable, press 0 for the operator or 1 to leave a message.

I'm having problem with making a menu where a caller has to make number of
choices, one after another.
Can you guys help me to set this up or show me an example if one has done
something similar?


Thanks in advance,
Zerthimon.


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RE: [Asterisk-Users] TDM400P bad echo problem, tried lots of things

[Fabio]

Hi All,

Also check that TDM400 not share interrups (yes, it sounds silly, but in
some cases it were the answer for me).

Fabio

-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] nombre de Matthew
Fredrickson
Enviado el: Jueves, 29 de Junio de 2006 09:41 p.m.
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Asunto: Re: [Asterisk-Users] TDM400P bad echo problem, tried lots of
things


Try fxotune.  That's the first thing you should have used.

Matthew Fredrickson

On Jun 20, 2006, at 11:55 AM, Carey O'Shea wrote:

 I have a bad echo problem on my TDM400P with one FXO module installed.

 I have tried a few things, such as:

 * setting rxgain and txgain to 0
 * setting echocancelwhenbridged to no / yes
 * settting echocancel to 64 / no / yes
 * setting echocanceltraining to 800 / no / yes
 * MG2 echo cancellation
 * MARK2 echo cancellation
 * KB1 echo cancellation
 * AGGRESSIVE_SUPPRESSOR option of MARK2

 Each time restarting Asterisk, then opening the Zap channel, and then
 speaking...only to hear my self played back almost instantly.

 None of these options changed the echo for me, it always sounded the
 same -- except for the AGGRESSIVE_SUPPRESSOR option, in which every
 time
 I spoke it made the other end a very low volume, so much that I
 couldn't
 hear the other end (ie: not useful).

 I don't have this problem with pure IP calls, it's only with my TDM400P
 and FXO that I have this echo problem. This means my headset and IP
 phones are fine (of course).

 So, what else can I try? :-)

 Any ideas why this is so consistent and persistent? Maybe it's
 something
 to do with my phone cable or something of that nature (hmm?)?

 Any input appreciated.

 Thanks,
 Carey O'Shea.


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RE: [Asterisk-Users] (no subject)

[Fabio]

Hi Tj,

yes, you can run two TDM400s (or more) on the same cpu, and the channels are
1 to 4, and 5 to 8. (also, you can set one or more groups for outgoing
calls).

Interrupts are the main issue. As far as possible avoids that the cards
share interruptions.

cheers

Fabio


-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] nombre de Ninneman, Tj
Enviado el: Miercoles, 28 de Junio de 2006 12:54 p.m.
Para: asterisk-users@lists.digium.com
Asunto: [Asterisk-Users] (no subject)


Hey everybody,

Is it alright to run two TDM400s on the same machine?  If it is, how would
one differentiate between the channels on each card?  So, if I'm running
strait FXS and my first card is fxsks 1-4, would the second be fxsks 5-8?
Would there be any interrupt problems?

Any help would be great!

Thanks!

Tj


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RE: [Asterisk-Users] Low volume/ audio problems on TDM400 card

[Fabio]

Hi 
Carlos,

- send us your interrupts configuration: cat 
/proc/interrupts
 
- tryadjusting line impedancy; see fxotune on zaptel source 
directory.

cheers,

Fabay
-Mensaje original-De: 
[EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED]En nombre de 
news.asterisk.usersEnviado el: Jueves, 15 de Junio de 2006 08:19 
a.m.Para: Asterisk Users Mailing List - Non-Commercial 
DiscussionAsunto: Re: [Asterisk-Users] Low volume/ audio problems on 
TDM400 card
That is the first thing I did.. rxgain seems to help to make it 
  louder when the two PSTN lines are bridged but also affectsthe audio 
  quality adversely. It's very touchy.. it also causes horrible feedback 
  (loud screeching when lines bridge). I've upgraded zapata and libpri 
  too.JDCarlos Rojas wrote: 
  Hi, I solved this problem in zapata.conf 
file;rxgain=0.0 
;Volume 
RXrxgain=8.0 
;Volume 
RX;txgain=0.0 
;Volume 
TXtxgain=1.0 
;Volume TXRegards
On 6/9/06, news.asterisk.users  
[EMAIL PROTECTED] wrote: 

  
  I've been attempting to get this 
  TDM400 card to work since March.While it sort of works, I'm having a 
  lot of audio issues with it.I am on Asterisk 1.2.9.1 svn rev 32797 and Zaptel 1.2.6. 

I've tried moving it to beefier hardware, upgrading asterisk/zaptel/etc, 

moving the X100 cards around on the TDM400 card, nothing seems to fix it.
The card says it's a revision J card. 

I get low volume on the receiving end (too quiet to hear when channels bridge) and
adjusting rx/txgain to comfortable levels causes popping and clicking when 

calls are made that cause bridged channels.  Upgrading to the newest zaptel fixed the horrible echo
and feedback issues I was having.

Before I send it in for a replacement I thought I'd ask about it in case there is a setting

I can make to fix it.


  


  

  
  

  


  

  
  


  


  
Kernel 
Version
2.6.9-34.0.1.ELsmp 
(SMP)
  
Distro Name
CentOS release 4.3 
(Final)
  
Uptime
2hours57minutes
  
Current 
Users
0
  
Load 
Averages
0.08 0.02 
  0.01

  
  

  


  Network Usage

  

  
  
Device
Received
Sent
Err/Drop
  
lo
2.02MB
2.02MB
0/0
  
eth0
2.49MB
4.67MB
0/0
  
sit0
0.00KB
0.00KB
0/0
  

  
  

  


  Hardware Information

  

  
  
Processors
2
  
Model
Intel(R) Pentium(R) D CPU 
  2.66GHz
  
CPU Speed
2.66 GHz
  
Cache Size
1024 KB
  
System 
Bogomips
10646.51
  
PCI Devices

  


-

RE: [Asterisk-Users] pickup problem

[Fabio]

Hi Denis,

are you using canreinvite=yes on your SIP endpoints definition ?

also check your features.conf, do you have pickupexten = *8 ?

fabay

-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] nombre de Denis
Shaposhnikov
Enviado el: Miércoles, 07 de Junio de 2006 03:42 a.m.
Para: asterisk-users@lists.digium.com
Asunto: [Asterisk-Users] pickup problem


Hi!

Could somebody help me with pickup feature? I've set

  callgroup = 1
  pickupgroup = 1

for my phones in sip.conf, but if I try to pickup call with *8
asterisk output to console

  Jun  6 15:04:44 WARNING[11857]: pbx.c:2401 __ast_pbx_run: Invalid
extension '*', but no rule 'i' in context 'office'

Thanks!

--
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RE: [Asterisk-Users] TDM does not disconnect

[Fabio]

Hi Vinicius,

on this link you have an explanation about your problem.
http://www.cisco.com/en/US/tech/tk652/tk653/technologies_tech_note09186a00800ae2d1.shtml

Basically, you need to know what method your pbx is using for signaling the end 
of the call, and then, you must to setup the zaptel driver and zapata channel 
according to that.

chers

Fabay

-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] nombre de Vinícius
Fontes - CANALL
Enviado el: Miércoles, 17 de Mayo de 2006 08:55 a.m.
Para: asterisk-users@lists.digium.com
Asunto: [Asterisk-Users] TDM does not disconnect


Hello all.

This is my very first message to the list. I have a TDM400P card, It  
has 2 FXO channels which are connected to extensions of my PBX  
(Ericsson BP250), so I can dial from any SIP softphone directly to  
physical (analog and digital) extensions on my company.

My PBX is configured so when I dial 8 on any extension, it will  
redirect to the first free FXO channel on my TDM400P card. Then I use  
the Asterisk's DISA application to get a dial tone, like this:


exten = s,1,disa(no-password,tdm-disa)

[tdm-disa]
exten = _XXX.,1,ChanIsAvail(Zap/3Zap/4) ; Checks for a free channel to dial
exten = _XXX.,2,Dial(${AVAILORIGCHAN}/${EXTEN}) ; Dials the number on  
the first channel available


But if the person I'm calling does not answer the phone and I hangup  
(fisically) the extension, the Zap channels doesn't hangup! They stay  
connected, and the line I called keeps on ringing.


So, this is the entire process:

1. I pickup a physical extension, and dial 8
2. The PBX redirects the call to the first FXO channel available
3. Asterisk answers the call and gives a dial tone using the DISA application
4. I dial the number I want
5. Asterisk dials using an available Zap channel
6. If the person I called does not answer the phone, I hangup my  
extension but the FXO channels doesn't hangup!


This is the logs I got running asterisk -vvv on the  
situation above. My comments on it are rounded with []:

[I pickup my physical extension and dial 8]

 -- Starting simple switch on 'Zap/3-1'
May 17 08:48:52 NOTICE[16335]: chan_zap.c:6184 ss_thread: Got event 18  
(Ring Begin)...
May 17 08:48:52 NOTICE[16335]: chan_zap.c:6184 ss_thread: Got event 2  
(Ring/Answered)...
May 17 08:48:56 NOTICE[16335]: chan_zap.c:6184 ss_thread: Got event 18  
(Ring Begin)...
 -- Executing DISA(Zap/3-1, no-password|tdm-disa) in new stack

[Asterisk gives me dial tone and I dial 081168345 - 0 + my cell phone number]

 -- Executing ChanIsAvail(Zap/3-1, Zap/3Zap/4) in new stack
 -- Hungup 'Zap/4-1'
 -- Executing NoOp(Zap/3-1, Canal: Zap/4) in new stack
 -- Executing Dial(Zap/3-1, Zap/4/081168345) in new stack
 -- Called 4/081168345

[My cell phone starts to ring, I hangup my extension. Cell phone keeps  
on ringing.]

[After a while (about one minute) the following shows up]

 -- Zap/4-1 is busy
 -- Hungup 'Zap/4-1'
   == Everyone is busy/congested at this time (1:1/0/0)
 -- Hungup 'Zap/3-1'







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RE: [Asterisk-Users] features.conf *1 Call Recording

[Fabio]

if you ar using SIP clients, try changing DTMF transfer mode.
For test use
 sip debug
on your * console, then place a call and watch the output. In INFO or
rfc2833 mode you must see the codes like SIP messages. If you are using
inband transfer mode (DTMF codes are  transferred like sounds) you don't see
the codes.

Also, try adjusting featuredigittimeout in features.conf:

[general]
featuredigittimeout = 2000 ; 2 seconds

because the default 500ms is a very short time.

Fabio Olaechea

3Tech SRL
Calle 48 Nro 632, Of. 67.
La Plata, CP B1900AMZ
Buenos Aires, Argentina.
Tel. +54 221 445 0244 Ext. 301
Fax. +54 221 445 0245
www.trestech.com.ar


-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] nombre de Dave Morrow
Enviado el: Miercoles, 10 de Mayo de 2006 02:48 p.m.
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Asunto: RE: [Asterisk-Users] features.conf *1 Call Recording


OK. You lost me.


David Morrow
Technical Systems Lead
Autodata Solutions Company
[EMAIL PROTECTED]
http://www.autodatasolutions.com

Tel: (519) 963-3020
Fax: (519) 451-6615

 Lead, follow or get out of the way! 

This message has originated from Autodata Solutions. The attached
material is the Confidential and Proprietary Information of Autodata
Solutions. This email and any files transmitted with it are confidential
and intended solely for the use of the individual or entity to whom they
are addressed. If you have received this email in error please delete
this message and notify the Autodata system administrator at
[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alejandro
Vargas
Sent: Wednesday, May 10, 2006 10:29 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] features.conf *1 Call Recording

2006/5/10, Dave Morrow [EMAIL PROTECTED]:
 I am attempting to setup Asterisk to allow me to press *1 while in a
 call to use automon to record the call but have had absolutely no
 success.  Is there a trick to this?

May be a problem with the way you are sending the dialtones. Try sending
as data.

--
Alejandro Vargas
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[Asterisk-Users] [Asterisk-User] Estension s don't start

[Fabio Montemaggiore]

Why the estension s dont' start?

In extensions.conf
  [default]
  exten = s,1,Answer
  exten = s,2,Playback(invalid)
  exten = s,3,Hangup

In sip.conf
  [general]
  context=default







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Re: [Asterisk-Users] [Asterisk-User] Estension s don't start

[Fabio Montemaggiore]

Asterisk in console don't show not all


--- José Luis Gómez [EMAIL PROTECTED] ha scritto:


 What do you see on asterisk console? 
 (asterisk -vc)
 
 
 El mar, 08-11-2005 a las 15:38 +0100, Fabio
 Montemaggiore escribió:
  Why the estension s dont' start?
  
  In extensions.conf
[default]
exten = s,1,Answer
exten = s,2,Playback(invalid)
exten = s,3,Hangup
  
  In sip.conf
[general]
context=default
  
  
  
  
  
  
  
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 -- 
 
 José Luis Gómez
 Qualis Information Technology
 Av. Rivadavia 2553, PB Of. 43 EP
 0342-4565684 int 102
 Bs. As. 011-51990896
 www.qualis.com.ar
 Soporte 0810-8880022
 Santa Fe - Argentina
 
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[Asterisk-users] VoiceMail help

[Fabio Montemaggiore]

I don't receveid e-mail with voicemail.
When I dial 2 with telephone, Asterisk record message
but don't send a e-mail at the mailbox. Why?
I have configuration this file.



In the voicemail.conf
[general]
attach=yes
format=wav
skipms=3000
maxsilence=10
silencethreshold=128
maxlogins=3
sendvoicemail=yes

[zonemessages]
italia=Europe/Rome|'vm-received' Q 'digit/at' HMP

[101]
100 = 100,100,[EMAIL PROTECTED],,|attach=yes


In the dialplan:
exten = 2,1,Answer
exten = 2,2,Wait(1)
exten = 2,3,VoiceMail(u100)
exten = 2,4,Playback(vm-goodbye)
exten = 2,5,Hangup







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[Asterisk-Users] chan.iax2.c errore

[Fabio Montemaggiore]

Why Asterisk show this message?

WARNING[14792]: chan_iax2.c:9355 load_module: Unable
to open IAX timing interface: No such device




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Re: [Asterisk-users] VoiceMail help

[Fabio Montemaggiore]

I don't set the mailserver.
What can I do?
I use Debian

Thanks






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[Asterisk-Users] Reset telephone IP PHONE 106

[Fabio Montemaggiore]

I have a telephone Voismart IP PHONE 106.
I have lost the password of the telephone and
therefore I am not able to set up it. How can I do to
do a reset of the telephone?



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[Asterisk-Users] Reset IP PHONE 106

[Fabio Montemaggiore]

 I have lost the password of the telephone, so I must
do a reset of the telephone. How can I do?
I have a VOISMART telehone: IP PHONE 106

Thanks






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[Asterisk-Users] Incoming call

[Fabio Montemaggiore]

When I receive a call, only one telephone ring...
Can I receive a call in much telephones, therefore
more telephones rings?






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[Asterisk-Users] Snom 220

[Fabio Montemaggiore]

I have a Snom 220 and I have a problem when more calls
arrive in the meantime. The receptionist holds on with
a customer when she is trasferring the call and press
a button meantime another call arrives, automatically
the two calls conference together; the hold on
customer speaks with the new other customer. What can
I do, please?






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[Asterisk-Users] more calls

[Fabio Montemaggiore]

I have a Snom 220 and I have a problem when more calls
arrive in the meantime. The receptionist holds on with
a customer when she is trasferring the call and press
a button meantime another call arrives, automatically
the two calls conference together; the hold on
customer speaks with the new other customer. What can
I do, please?









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[Asterisk-users] Configuration QuadBRI Junghanns

[Fabio Montemaggiore]

What I can configuration my card Junghanns QuadBri?
Where I can download drivers?

Thanks?






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[Asterisk-Users] Don't call

[Fabio Montemaggiore]

I receive a call, but don't call...
Asterisk show this message.
Are codecs the problem?

Sep 30 11:25:54 WARNING[4475]: chan_sip.c:1899
create_addr: No such host: sip.uni.it,r
Sep 30 11:25:54 NOTICE[4475]: app_dial.c:1109
dial_exec_full: Unable to create channel of type 'SIP'
(cause 3 - No route to destination)
  == Everyone is busy/congested at this time (1:0/0/1)






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[Asterisk-Users] Not Authenticate

[Fabio Montemaggiore]

Why Asterisk show this message?
What I can do?

Sep 30 15:45:18 NOTICE[3608]: chan_sip.c:9096
handle_response_invite: Failed to authenticate on
INVITE to '100
sip:[EMAIL PROTECTED];tag=as413bd6a8'
-- SIP/sip.uni.it-df15 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)

Thanks!!



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[Asterisk-Users] Not authenticate

[Fabio Montemaggiore]

Why Asterisk show this message?
What I can do?

Sep 30 15:45:18 NOTICE[3608]: chan_sip.c:9096
handle_response_invite: Failed to authenticate on
INVITE to '100
sip:[EMAIL PROTECTED];tag=as413bd6a8'
-- SIP/sip.uni.it-df15 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)

Thanks






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Re: [Asterisk-Users] No Incoming Calls on Asterisk

[Fabio Montemaggiore]

I use UNIVOICE provider, therefore you change
sip.uni.it with your provider.

View files






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extensions.conf
Description: 3949034846-extensions.conf


sip.conf
Description: 3455877249-sip.conf
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[Asterisk-users]

[Fabio Montemaggiore]

I would integrated my Asterisk PBX with CRM software,
and I tell you if you prefer Asterisk or [EMAIL PROTECTED]
for programming to.

Thanks






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[Asterisk-Users] Don't call

[Fabio Montemaggiore]

I have set up extension.conf and sip.con with default
parameter of UNIVOICE server, but Asterisk show this
message when I call a number:

Sep 29 11:34:52 WARNING[4179]: chan_sip.c:1899
create_addr: No such host: univoice,Ttr
Sep 29 11:34:52 NOTICE[4179]: app_dial.c:1109
dial_exec_full: Unable to create channel of type 'SIP'
(cause 3 - No route to destination)
  == Everyone is busy/congested at this time (1:0/0/1)
-- Executing Hangup(SIP/100-2331, ) in new
stack







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[Asterisk-Users] MeetMe error

[Fabio Montemaggiore]

I have install Flash Operator Panel but Asterisk show
this message:

WARNING[3564]: pbx.c:1650 pbx_extension_helper: No
application 'Meetme' for extension (conferences, 101, 1)






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[Asterisk-Users] Where MeetMe application

[Fabio Montemaggiore]

I haven't app_meetme.so file...
Where I can search?



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[Asterisk-Users] Re: chan_unicall.c compile error

[Fabio Vasco]

Hector,
This is my Linux Fedora Core 3 version info
[EMAIL PROTECTED] proc]# cat version
Linux version 2.6.9-1.667 ([EMAIL PROTECTED]) (gcc version 
3.4.2 20041017 (Red Hat 3.4.2-6.fc3)) #1 Tue Nov 2 14:41:25 EST 2004

I am get Asterisk with the cvs -r stable, i supose the version is 1.0.6.
Thanks for your help.
Fabio
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[Asterisk-Users] Re: Libunicall Make Error

[Fabio Vasco]

Angelo,
Try to make all packages (spandsp, libsupertone) with ./configure 
--prefix=/usr

In my case this working fine...
Regards,
Fabio Vasco
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[Asterisk-Users] chan_unicall.c compile error

[Fabio Vasco]

I have a error when try to compile de chan_unicall.c with Asterisk. Others 
modules like spandsp, libsupertone, libunicall  libmfc2 is sucessfully 
compiled (using --prefix=/usr) with the last version avaliable at 
ftp.soft-switch.org

I am using the stable_version from CVS, with zaptel  libpri... a TE110P is 
installed. The R2 variant is from Argentina, i am in Ecuador.

This is a output error... thank you so much por any help.
gcc -shared -Xlinker -x -o chan_mgcp.so chan_mgcp.o
gcc -pipe  -Wall -Wstrict-prototypes -Wmissing-prototypes 
-Wmissing-declarations -g  -Iinclude -I../include  -D_REENTRANT 
-D_GNU_SOURCE  -O6 -march=i686  -DZAPTEL_OPTIMIZATIONS 
-DASTERISK_VERSION=\CVS-04/20/05-21:22:27\ -DINSTALL_PREFIX=\\ 
-DASTETCDIR=\/etc/asterisk\ -DASTLIBDIR=\/usr/lib/asterisk\ 
-DASTVARLIBDIR=\/var/lib/asterisk\ -DASTVARRUNDIR=\/var/run\ 
-DASTSPOOLDIR=\/var/spool/asterisk\ -DASTLOGDIR=\/var/log/asterisk\ 
-DASTCONFPATH=\/etc/asterisk/asterisk.conf\ 
-DASTMODDIR=\/usr/lib/asterisk/modules\ 
-DASTAGIDIR=\/var/lib/asterisk/agi-bin\ -DBUSYDETECT_MARTIN 
-DNEW_PRI_HANGUP  -Wno-missing-prototypes -Wno-missing-declarations   
-DZAPATA_PRI  -DIAX_TRUNKING  -DCRYPTO -fPIC-c -o chan_iax2.o 
chan_iax2.c
chan_iax2.c: In function `__send_command':
chan_iax2.c:2875: warning: assignment discards qualifiers from pointer 
target type
gcc -pipe  -Wall -Wstrict-prototypes -Wmissing-prototypes 
-Wmissing-declarations -g  -Iinclude -I../include  -D_REENTRANT 
-D_GNU_SOURCE  -O6 -march=i686  -DZAPTEL_OPTIMIZATIONS 
-DASTERISK_VERSION=\CVS-04/20/05-21:22:27\ -DINSTALL_PREFIX=\\ 
-DASTETCDIR=\/etc/asterisk\ -DASTLIBDIR=\/usr/lib/asterisk\ 
-DASTVARLIBDIR=\/var/lib/asterisk\ -DASTVARRUNDIR=\/var/run\ 
-DASTSPOOLDIR=\/var/spool/asterisk\ -DASTLOGDIR=\/var/log/asterisk\ 
-DASTCONFPATH=\/etc/asterisk/asterisk.conf\ 
-DASTMODDIR=\/usr/lib/asterisk/modules\ 
-DASTAGIDIR=\/var/lib/asterisk/agi-bin\ -DBUSYDETECT_MARTIN 
-DNEW_PRI_HANGUP  -Wno-missing-prototypes -Wno-missing-declarations   
-DZAPATA_PRI  -DIAX_TRUNKING  -DCRYPTO -fPIC-c -o iax2-parser.o 
iax2-parser.c
gcc -shared -Xlinker -x -o chan_iax2.so chan_iax2.o iax2-parser.o
gcc -pipe  -Wall -Wstrict-prototypes -Wmissing-prototypes 
-Wmissing-declarations -g  -Iinclude -I../include  -D_REENTRANT 
-D_GNU_SOURCE  -O6 -march=i686  -DZAPTEL_OPTIMIZATIONS 
-DASTERISK_VERSION=\CVS-04/20/05-21:22:27\ -DINSTALL_PREFIX=\\ 
-DASTETCDIR=\/etc/asterisk\ -DASTLIBDIR=\/usr/lib/asterisk\ 
-DASTVARLIBDIR=\/var/lib/asterisk\ -DASTVARRUNDIR=\/var/run\ 
-DASTSPOOLDIR=\/var/spool/asterisk\ -DASTLOGDIR=\/var/log/asterisk\ 
-DASTCONFPATH=\/etc/asterisk/asterisk.conf\ 
-DASTMODDIR=\/usr/lib/asterisk/modules\ 
-DASTAGIDIR=\/var/lib/asterisk/agi-bin\ -DBUSYDETECT_MARTIN 
-DNEW_PRI_HANGUP  -Wno-missing-prototypes -Wno-missing-declarations   
-DZAPATA_PRI  -DIAX_TRUNKING  -DCRYPTO -fPIC-c -o chan_local.o 
chan_local.c
gcc -shared -Xlinker -x -o chan_local.so chan_local.o
gcc -pipe  -Wall -Wstrict-prototypes -Wmissing-prototypes 
-Wmissing-declarations -g  -Iinclude -I../include  -D_REENTRANT 
-D_GNU_SOURCE  -O6 -march=i686  -DZAPTEL_OPTIMIZATIONS 
-DASTERISK_VERSION=\CVS-04/20/05-21:22:27\ -DINSTALL_PREFIX=\\ 
-DASTETCDIR=\/etc/asterisk\ -DASTLIBDIR=\/usr/lib/asterisk\ 
-DASTVARLIBDIR=\/var/lib/asterisk\ -DASTVARRUNDIR=\/var/run\ 
-DASTSPOOLDIR=\/var/spool/asterisk\ -DASTLOGDIR=\/var/log/asterisk\ 
-DASTCONFPATH=\/etc/asterisk/asterisk.conf\ 
-DASTMODDIR=\/usr/lib/asterisk/modules\ 
-DASTAGIDIR=\/var/lib/asterisk/agi-bin\ -DBUSYDETECT_MARTIN 
-DNEW_PRI_HANGUP  -Wno-missing-prototypes -Wno-missing-declarations   
-DZAPATA_PRI  -DIAX_TRUNKING  -DCRYPTO -fPIC-c -o chan_skinny.o 
chan_skinny.c
gcc -shared -Xlinker -x -o chan_skinny.so chan_skinny.o
gcc -o gentone gentone.c -lm
./gentone busy 480 620
Wavelength 1 (in samples):   16.7
Minimum samples (1): 50 (3.00.3 wavelengths)
Wavelength 1 (in samples):   12.90323
Minimum samples (1): 400 (31.00.3 wavelengths)
Need 400 samples
Wrote busy.h
./gentone ringtone 440 480
Wavelength 1 (in samples):   18.18182
Minimum samples (1): 200 (11.00.3 wavelengths)
Wavelength 1 (in samples):   16.7
Minimum samples (1): 50 (3.00.3 wavelengths)
Need 200 samples
Wrote ringtone.h
gcc -pipe  -Wall -Wstrict-prototypes -Wmissing-prototypes 
-Wmissing-declarations -g  -Iinclude -I../include  -D_REENTRANT 
-D_GNU_SOURCE  -O6 -march=i686  -DZAPTEL_OPTIMIZATIONS 
-DASTERISK_VERSION=\CVS-04/20/05-21:22:27\ -DINSTALL_PREFIX=\\ 
-DASTETCDIR=\/etc/asterisk\ -DASTLIBDIR=\/usr/lib/asterisk\ 
-DASTVARLIBDIR=\/var/lib/asterisk\ -DASTVARRUNDIR=\/var/run\ 
-DASTSPOOLDIR=\/var/spool/asterisk\ -DASTLOGDIR=\/var/log/asterisk\ 
-DASTCONFPATH=\/etc/asterisk/asterisk.conf\ 
-DASTMODDIR=\/usr/lib/asterisk/modules\ 
-DASTAGIDIR=\/var/lib/asterisk/agi-bin\ -DBUSYDETECT_MARTIN 
-DNEW_PRI_HANGUP  -Wno-missing-prototypes -Wno-missing-declarations   
-DZAPATA_PRI  -DIAX_TRUNKING  -DCRYPTO -fPIC-c -o chan_oss.o chan_oss.c
gcc 

[Asterisk-Users] Edit MGCP response

[Fabio Margarido]

Hi there,

I'd like to know if there's any way I can edit the fields asterisk
sends in an MGCP response to my devices, without having to mess with
the source code. What happens is that asterisk sends an F parameter in
an audit endpoint message I don't want it to send. Does anyone know I
can solve this?
Thanks
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[Asterisk-Users] MGCP howto

[Fabio Margarido]

Hey there,

I'm an asterisk newbie and have just joined this mailing list. I have
to use asterisk as a call agent that supports MGCP requests. I'm
reading the documentation from asteriskdocs and voip-info.org but
those cover more specifically only IAX and SIP configuration. I'd
really appreciate it if someone can tell me where to find more
detailed documentation on how to configure asterisk to work with MGCP.
Thanks
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[Asterisk-Users] MGCP howto

[Fabio Margarido]

Hey there,

I'm an asterisk newbie and have just joined this mailing list. I have
to use asterisk as a call agent that supports MGCP requests. I'm
reading the documentation from asteriskdocs and voip-info.org but
those cover more specifically only IAX and SIP configuration. I'd
really appreciate it if someone can tell me where to find more
detailed documentation on how to configure asterisk to work with MGCP.
Thanks
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[Asterisk-Users] Meetme2

[Fabio Donaggio]

Hi!!

I try to install meetme2i follow instructions that i found in 

http://www.areski.net/asterisk-meetme/about.php?s=0

but, when i modify the "Asterisk/apps/Makefile" and i run the "make" 
command,
I have this type of error:

[EMAIL PROTECTED] apps]# makecc -pipe -fPIC 
-DUSEMYSQLVM -c -o app_meetme2.o app_meetme2.capp_meetme2.c:31:22: 
libpq-fe.h: No such file or directoryapp_meetme2.c:32:19: mysql.h: No such 
file or directoryapp_meetme2.c:36:26: linux/zaptel.h: No such file or 
directoryapp_meetme2.c: In function `launch_query':app_meetme2.c:139: 
`PGconn' undeclared (first use in this function)app_meetme2.c:139: (Each 
undeclared identifier is reported only onceapp_meetme2.c:139: for each 
function it appears in.)app_meetme2.c:139: `conn' undeclared (first use in 
this function)app_meetme2.c:140: `PGresult' undeclared (first use in this 
function)app_meetme2.c:140: `res' undeclared (first use in this 
function)app_meetme2.c:142: `MYSQL' undeclared (first use in this 
function)app_meetme2.c:142: parse error before 
"myconn"app_meetme2.c:143: `MYSQL_RES' undeclared (first use in this 
function)app_meetme2.c:143: `result' undeclared (first use in this 
function)app_meetme2.c:144: `MYSQL_ROW' undeclared (first use in this 
function)app_meetme2.c:144: parse error before "row"app_meetme2.c:145: 
`my_ulonglong' undeclared (first use in this function)app_meetme2.c:153: 
`CONNECTION_BAD' undeclared (first use in this function)app_meetme2.c:164: 
`PGRES_TUPLES_OK' undeclared (first use in this function)app_meetme2.c:180: 
warning: passing arg 1 of `atoi' makes pointer from integer without a 
castapp_meetme2.c:181: warning: passing arg 1 of `atoi' makes pointer from 
integer without a castapp_meetme2.c:182: warning: passing arg 1 of `atoi' 
makes pointer from integer without a castapp_meetme2.c:183: warning: passing 
arg 1 of `atoi' makes pointer from integer without a castapp_meetme2.c:184: 
warning: passing arg 1 of `atoi' makes pointer from integer without a 
castapp_meetme2.c:185: warning: passing arg 1 of `atoi' makes pointer from 
integer without a castapp_meetme2.c:186: warning: passing arg 2 of `strcpy' 
makes pointer from integer without a castapp_meetme2.c:197: `myconn' 
undeclared (first use in this function)app_meetme2.c:198: 
`MYSQL_OPT_COMPRESS' undeclared (first use in this 
function)app_meetme2.c:217: `num_row' undeclared (first use in this 
function)app_meetme2.c:224: `row' undeclared (first use in this 
function)app_meetme2.c: In function 
`launch_query_onefield':app_meetme2.c:255: `PGconn' undeclared (first use in 
this function)app_meetme2.c:255: `conn' undeclared (first use in this 
function)app_meetme2.c:256: `PGresult' undeclared (first use in this 
function)app_meetme2.c:256: `res' undeclared (first use in this 
function)app_meetme2.c:260: `MYSQL' undeclared (first use in this 
function)app_meetme2.c:260: parse error before 
"myconn"app_meetme2.c:262: `MYSQL_RES' undeclared (first use in this 
function)app_meetme2.c:262: `myresult' undeclared (first use in this 
function)app_meetme2.c:263: `MYSQL_ROW' undeclared (first use in this 
function)app_meetme2.c:263: parse error before "row"app_meetme2.c:264: 
`my_ulonglong' undeclared (first use in this function)app_meetme2.c:269: 
`CONNECTION_BAD' undeclared (first use in this function)app_meetme2.c:278: 
`PGRES_COMMAND_OK' undeclared (first use in this function)app_meetme2.c:296: 
`PGRES_TUPLES_OK' undeclared (first use in this function)app_meetme2.c:303: 
warning: passing arg 1 of `strlen' makes pointer from integer without a 
castapp_meetme2.c:332: warning: passing arg 1 of `strlen' makes pointer from 
integer without a castapp_meetme2.c:348: `myconn' undeclared (first use in 
this function)app_meetme2.c:350: `MYSQL_OPT_COMPRESS' undeclared (first use 
in this function)app_meetme2.c:380: `num_row' undeclared (first use in this 
function)app_meetme2.c:387: `row' undeclared (first use in this 
function)app_meetme2.c: In function `give_voice_next':app_meetme2.c:606: 
`ZT_CONF_CONFMON' undeclared (first use in this function)app_meetme2.c:606: 
`ZT_CONF_LISTENER' undeclared (first use in this function)app_meetme2.c:621: 
`ZT_CONF_CONF' undeclared (first use in this function)app_meetme2.c:621: 
`ZT_CONF_TALKER' undeclared (first use in this function)app_meetme2.c: In 
function `build_conf':app_meetme2.c:713: storage size of `ztc' isn't 
knownapp_meetme2.c:741: `ZT_CONF_CONF' undeclared (first use in this 
function)app_meetme2.c:741: `ZT_CONF_TALKER' undeclared (first use in this 
function)app_meetme2.c:741: `ZT_CONF_LISTENER' undeclared (first use in this 
function)app_meetme2.c:743: `ZT_SETCONF' undeclared (first use in this 
function)app_meetme2.c: In function `conf_run':app_meetme2.c:843: 
storage size of `ztc' isn't knownapp_meetme2.c:843: storage size of 
`ztc_tmp' isn't knownapp_meetme2.c:876: `ZT_BUFFERINFO' undeclared (first 
use in this function)app_meetme2.c:876: parse error before 
"bi"app_meetme2.c:939: `bi' undeclared 

[Asterisk-Users] Fw: DynExtenDB

[Fabio Donaggio]

Hi!! I have a problem with DynExtenDB.
This is the message:

ERROR[245776]: app_dynextendb.c:76 dynamic_extension: No DNID in channel
found - not possible to query extension. Skipping.

Can you help me? Thanks...

Fabio Donaggio
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[Asterisk-Users] DynExtenDB

[Fabio Donaggio]

Hi!! I have a problem with DynExtenDB.
This is the message:

ERROR[245776]: app_dynextendb.c:76 dynamic_extension: No DNID in channel
found - not possible to query extension. Skipping.

Anyone can help me? Thanks...

F

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[Asterisk-Users] Asterisk and MySQL

[Fabio Donaggio]

Hi to all!!!

Here's my problem:

-- Executing Dial("SIP/2002-ba7c", 
"SIP/2000|30|tr") in new stackMay 31 16:26:11 NOTICE[262161]: app_dial.c:536 
dial_exec: Unable to create channel of type 'SIP' == Everyone is busy 
at this time -- Executing VoiceMail("SIP/2002-ba7c", 
"b2000") in new stackMay 31 16:26:11 WARNING[262161]: app_voicemail.c:1517 
leave_voicemail: No entry in voicemail config file for 
'2000' -- Executing Hangup("SIP/2002-ba7c", "") in new 
stack == Spawn extension (from-sip, 2000, 103) exited non-zero on 
'SIP/2002-ba7c'
I followinstructions that I found 
in

http://www.voip-info.org/wiki-Asterisk+voicemail+database

but voicemail not work with my MySql 
database

I'm in your hands

Thanks

[Asterisk-Users] Asterisk addons

[Fabio Donaggio]

Hi to all!! 

Is there another method to download asterisk 
addons???

Thanks
F

[Asterisk-Users] Asterisk and MySQL

[Fabio Donaggio]

Hi to all!!
I'm successful to connect Asterisk to MySQL database...
Can anyone learn me how to store sip user in 
MySQL database and how to configure voicemail??

Thanks for all!!!
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[Asterisk-Users] Fw: Asterisk and MySQL

[Fabio Donaggio]

Hi!

It's all ok with CVS login...I download 
asterisk-addons.
I would try to store sip friends in MySQL database 
and also the voicemailcan you help me???
Thanks

[Asterisk-Users] Astersik and PostgreSQL

[Fabio Donaggio]

Hi to all!!
I'm successful to connect Asterisk to PostgreSQL database...
If it's possible, can anyone learn me how to store sip user in 
PostgreSQL database and how to configure voicemail??

Thanks for all!!!
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[Asterisk-Users] Asterisk and PostgreSQL

[Fabio Donaggio]

Hi to all!!
I'm successful to connect Asterisk to PostgreSQL database...
If it's possible, can anyone learn me how to store sip user in 
PostgreSQL database and how to configure voicemail??
  
Thanks for all!!!
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[Asterisk-Users] CVS login

[Fabio Donaggio]

Hi to all!!

Here is my problem:
[EMAIL PROTECTED] root]# cd /usr/src
[EMAIL PROTECTED] src]# export
CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot
[EMAIL PROTECTED] src]# cvs login
-bash: cvs: command not found
[EMAIL PROTECTED] src]#

Anyone can help me??

Thanks for all!!!

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[Asterisk-Users] PostgreSQL

[Fabio Donaggio]

Hi to all!!

Here's my problem:

[cdr_pgsql.so] = (PostgreSQL CDR Backend)
  == Parsing '/etc/asterisk/cdr_pgsql.conf': Found
May 26 17:21:35 ERROR[16384]: cdr_pgsql.c:298 my_load_module: cdr_pgsql:
Unable to connect to database server localhost.
Calls will not be logged!
May 26 17:21:35 ERROR[16384]: cdr_pgsql.c:299 my_load_module: cdr_pgsql:
Reason: could not connect to server:
Connection refused
Is the server running on localhost and accepting
TCP/IP connections on port 5432?

Anyone can help me??? Anyone have some suggest about this or about how to
connect PostgreSQL to Asterisk???
Thanks!

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[Asterisk-Users] Re: PostgreSQL

[Fabio Donaggio]

Thaks to all!!! Now it works! Thanks
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[Asterisk-Users] Some problems with download Asterisk-addons

[Fabio Donaggio]

Hi!

I have some problems with the download of Asterisk-addons.
I try to follow instructions that I found in
http://www.voip-info.org/tiki-index.php?page=Asterisk%20cdr%20mysql , but
nothing to do.

This is my shell:
[EMAIL PROTECTED] root]# cd /usr/src
[EMAIL PROTECTED] src]# export
CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot
[EMAIL PROTECTED] src]# cvs login
Logging in to :pserver:[EMAIL PROTECTED]:2401/usr/cvsroot
CVS password:
cvs checkout asterisk-addons
cvs [login aborted]: connect to cvs.digium.com(65.38.23.22):2401 failed:
Connection timed out
[EMAIL PROTECTED] src]# cvs checkout asterisk-addons
cvs [checkout aborted]: connect to cvs.digium.com(216.234.176.92):2401
failed: Connection timed out
[EMAIL PROTECTED] src]#

Can anyone help me?? Thanks.

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[Asterisk-Users] Mysql

[Fabio Donaggio]

Hi, to all!!!

I can't download asterisk-addons...I try with CVS, 
but i can't.
How can I do???

Thank you

Fabio