Re: [asterisk-users] Outside lines are just not happening...

[Brian Capouch]

Stephen Bosch wrote:


If you get dumbfounded responses ask to speak to someone in the
programming group (unless they are a tiny little phone company, they
will have one). If you open a ticket, it usually means they will
escalate the problem, even if the agent you are speaking with has no
idea what you are talking about. Best to be friendly with them!



And what do you do when they say:

"We have a modern, relatively-new switch for which that sort of feature 
change is a trivial click on a GUI checkbox.  However, we do not have 
any tariffed requirement to provide disconnect supervision.  So we won't 
do it for you.  Goodbye."


And then the call ends.

B.

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RE: [asterisk-users] Call to an arbitrary outbound number by asterisk

[Yuan LIU]

From: "Arpit Mehta" <[EMAIL PROTECTED]>
Date: Fri, 18 May 2007 02:31:22 -0400

Hi,

I have a strange problem. I have a TE110p digium card.

I want to dial 19173995791 when any incoming call comes in.  What is
happening is that when I dial 19173-995791. Asterisk picks up the first 5
digits assuming it is the extension and appends 212-85 (here in the
university most numbers start with this) in front . Therefore I get
connected to some random number 212-85-(19173) (where the voicemail is
running).
I cannot understand why asterisk is doing this whereas my dialplan says it
needs to connect to other number
  "exten => _.,1,Dial(Zap/g1/19173995791)"

Also any idea if this is an Asterisk problem or a telco problem. Any
help/hints/suggestions would be most welcome


If you are sure that your university doesn't have a PBX, that's a telco 
problem.  Looks like that the switch has a dial plan that does not allow you 
to dial this sequence directly and interpret all dialed sequence as a local 
call. (This is usually the function of a PBX but ...)  What is this number 
19173995791, any way? (and what is 212-85?) If you attach a phone directly 
to a channel bank, would you be able to dial this sequence?


Yuan Liu


Here are my files.

zapata.conf
context=incoming
switchtype=national
signalling=pri_cpe
group=1
channel=>1-23

extension.conf
[incoming]
exten => _.,1,Dial(Zap/g1/19173995791)

# I have added this line in the dialplan is because I want it to
match the  last 5 digit and simply dial the number 19173995791 such that a
call leg is established between the calling party and the number 
19173995791




CLI debug information
-- Requested transfer capability: 0x00 - SPEECH
   -- Called g1/19173995791
   -- Zap/1-1 is proceeding passing it to Zap/23-1
   -- Zap/1-1 is making progress passing it to Zap/23-1

### The call keeps ringing for sometime then it goes to
voicemail. The message comes when the voicemail start. Note that I have not
setup any voice mail

   -- Zap/1-1 answered Zap/23-1

### Goes to the voicemail
   -- Native bridging Zap/23-1 and Zap/1-1

   -- Channel 0/23, span 1 got hangup request
   -- Hungup 'Zap/1-1'
 == Spawn extension (incoming, 17689, 1) exited non-zero on 'Zap/23-1'
   -- Hungup 'Zap/23-1'


Regards

--
Arpit Mehta
Graduate Student
Department of Computer Science
Columbia University

Tel: 1-646-387-5998



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Re: [asterisk-users] HPEC audio clipping

[Stephen Bosch]

Eric "ManxPower" Wieling wrote:
> What are the advantages of 9.x over the 8.x that I currently use?

I was about to ask the same question. What if my 8.x EC works just fine?

(Why expose yourself to the possibility that even the patched version
fails?)

-Stephen-
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[asterisk-users] Call to an arbitrary outbound number by asterisk

[Arpit Mehta]

Hi,

I have a strange problem. I have a TE110p digium card.

I want to dial 19173995791 when any incoming call comes in.  What is
happening is that when I dial 19173-995791. Asterisk picks up the first 5
digits assuming it is the extension and appends 212-85 (here in the
university most numbers start with this) in front . Therefore I get
connected to some random number 212-85-(19173) (where the voicemail is
running).
I cannot understand why asterisk is doing this whereas my dialplan says it
needs to connect to other number
  "exten => _.,1,Dial(Zap/g1/19173995791)"

Also any idea if this is an Asterisk problem or a telco problem. Any
help/hints/suggestions would be most welcome

Here are my files.

zapata.conf
context=incoming
switchtype=national
signalling=pri_cpe
group=1
channel=>1-23

extension.conf
[incoming]
exten => _.,1,Dial(Zap/g1/19173995791)

# I have added this line in the dialplan is because I want it to
match the  last 5 digit and simply dial the number 19173995791 such that a
call leg is established between the calling party and the number 19173995791



CLI debug information
-- Requested transfer capability: 0x00 - SPEECH
   -- Called g1/19173995791
   -- Zap/1-1 is proceeding passing it to Zap/23-1
   -- Zap/1-1 is making progress passing it to Zap/23-1

### The call keeps ringing for sometime then it goes to
voicemail. The message comes when the voicemail start. Note that I have not
setup any voice mail

   -- Zap/1-1 answered Zap/23-1

### Goes to the voicemail
   -- Native bridging Zap/23-1 and Zap/1-1

   -- Channel 0/23, span 1 got hangup request
   -- Hungup 'Zap/1-1'
 == Spawn extension (incoming, 17689, 1) exited non-zero on 'Zap/23-1'
   -- Hungup 'Zap/23-1'


Regards

--
Arpit Mehta
Graduate Student
Department of Computer Science
Columbia University

Tel: 1-646-387-5998
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[asterisk-users] astertisk for dummies

[clive.chan\(Alpha Trilogies Networks\)]

Hi guys, 
Do some one read about "Asterisk for Dummies" and wish to hear some feedback
about this book as compare to the TFOT.



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Re: [asterisk-users] Asterisk Queue MOH

[TienSen Chong]

Thank you very much for sharing this.

Chong

On 5/17/07, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:


>From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of TienSen
Chong
>Sent: 17. maí 2007 10:51
>To: Asterisk Users Mailing List - Non-Commercial Discussion
>Subject: [asterisk-users] Asterisk Queue MOH
>Is there any way if i want the caller to hear dial tone rather than the
MOH?

Perhaps you could use something like

Queue(yourqueuename|rt|||60);

in extensions.conf or extension.ael? The r is defined as "ring instead of
playing MOH".

Baldvin.



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[asterisk-users] Anyone tested the new Sony Ericsson P1 phones..

[Rosli Sukri]

Hi,
Has anyone on this list tested out the new SE P1 phones (
http://www.uncrate.com/men/gear/cell-phones/sony-ericsson-p1/). It says it
supports VOIP, wonder if it is working with asterisk.
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Re: [asterisk-users] Get sip response code

[Robert Lister]

On Thu, May 17, 2007 at 12:11:48AM -0600, Ken Williams wrote:
> It's funny Robert would come looking for this tonight, as I've been 
> spending a fair amount of time trying to track this down today.  I then 
> went to the source and found what Andreas had found below.
>  
> However, I'm not a real programmer, but just a hack of a hackI tried 
> to make my own variable but failed because I don't really know what the 
> hell I'm doing! :D
>  
> Here's one form of what I tried, though I did try lots of different ways, 
> but wasn't able to get it to compile without errors.  At best I got the 
> server to do nothing, at worse I crashed the server when trying to use it:

I asked the question on digium bugs, and I got back a response along the 
lines of: use ${HANGUPCAUSE}. They were not receptive to the idea of having 
a SIP response code variable, or willing to discuss it, or the fact that my 
original problem stems from the fact that "CONGESTION" is used for too many 
things, not just CONGESTION, so it makes it difficult. It should really have 
a FAIL response. (or just rename "CONGESTION" to "FAIL" since that's what it 
acually means.)

It does seem strange though that you can see every sip header with 
${SIP_HEADER()} but not the actul SIP response.

Hangupcause returns a value (including SIP channels) which is interpreted 
back into a "cause code" RFC3398.

I have since update the wiki docs, as this was all a bit non-obvious to me:

http://www.voip-info.org/wiki/view/Asterisk+Variable+DIALSTATUS
http://www.voip-info.org/wiki/view/Asterisk+Variable+HANGUPCAUSE


Rob

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Re: [asterisk-users] Re: FastAGI hangs up channel if server is not available

[Lee Jenkins]

JR Richardson wrote:

Running 1.2.14

When I call a FastAGI script such as this script for an incoming call:

[calldirect]
exten=>s,1,Answer()
exten=>s,2,AGI(agi://192.168.1.175/calldirect?check&${CALLERID(num)})
exten=>s,3,Goto(check_time,s,1)



Thanks for the hint.  I thought about doing this but then I would have 
to have another executable process just to shell out to the FastAGI 
which is what I use by choice to reduce load on the server ;)




--

Warm Regards,

Lee



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Re: [asterisk-users] Cascading Queues

[Robert Lister]

On Thu, May 17, 2007 at 03:50:52PM -0400, Jason Adams wrote:
> Scenario 1:
> We are working with a client that currently has one support queue with
> about 10 agents.  They are starting to get pretty long hold times for
> their customers and they have requested three queues.  Queue 1 will have
> a timeout of 4 minutes.  After that it will move to Queue 2 which has a
> default timeout of 3 minutes.  After that we will transfer the call to
> the receptionist who will either take a message or put them back in the
> queue with a higher priority if they want to continue to hold.  Queue 2
> will have more agents in that queue plus the agents that were in Queue
> 1.
>  
> Question:
> Can I have the same agents in multiple queues to work the way I'm
> thinking above?  So if the caller reaches Queue 2 the agents from Queue
> 1 will be available if they get off a call, plus new agents are added
> into Queue 2.

You can have the same agents in multiple queues, chan_agent only allows one 
call to happen per agent channel. As long as you are not mixing Agent/xxx 
and SIP/xxx destinations which are routing to the same people in the queues, 
it should work. (When an agent channel is busy, the SIP channel might not 
be, and vice-versa.)

> So I'm thinking something like:
> exten => s,1,Queue(support1)
> exten => s,2,Queue(support2)
> exten => s,3,Dial(SIP/${RECEPTIONIST})
>  
> Then the receptionist would just dial a special extension which would
> add priority=10 to the queue.

You might want to check the ${QUEUESTATUS} so you can work out why the call 
dropped out of the first queue, and if you want it to immediately drop out 
of the first queue if it is full, or sit there for 4 minutes waiting etc.
(In queues.conf, check the joinempty/leavewhenempty options for the queue, 
then check it like:- (where the numbers are priorites you want to jump to)

exten => _s,n,GotoIf($["${QUEUESTATUS}" = "UNKNOWN"]?500)
exten => _s,n,GotoIf($["${QUEUESTATUS}" = "BUSY"]?850)
exten => _s,n,GotoIf($["${QUEUESTATUS}" = "FULL"]?850)
exten => _s,n,GotoIf($["${QUEUESTATUS}" = "JOINUNAVAIL"]?650)
exten => _s,n,GotoIf($["${QUEUESTATUS}" = "LEAVEUNAVAIL"]?650)
exten => _s,n,GotoIf($["${QUEUESTATUS}" = "LEAVEEMPTY"]?650)
exten => _s,n,GotoIf($["${QUEUESTATUS}" = "TIMEOUT"]?700)

Your setup will need a bit of work, for example what will happen if the 
receptionist is not available, how to trap calls going round and round in 
loops etc. You could automate bits of this further. (Maybe with the 
voicemail app and a breakout 'o' extn, and record a greeting that says 
"leave us a message, or press 0 to continue to hold...")

> Scenario 2:
> This same customer is starting to sell their product internationally.
> They are purchasing VOIP DID's from various countries for local calls
> from that area.  Would this just be like setting up a regular VOIP line
> to register the account in sip.conf and then creating a context for
> those countries so we know where they are coming from?

Yes. The sip.conf entry for the peer will point it to the right context=, 
and/or the register= statement can point to a specific extension, so you can 
tell where the call is coming from either way.

If you want to place outbound calls via those numbers to return calls etc, 
then you will probably need to add a prefix as the call comes in (Hack the 
${CALLERID(num)} on the way in to add the prefix so the call goes back out 
the right way.)

Rob


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Re: [asterisk-users] Multiple lines on Linksys/Sipura phones

[Chris Mason (Lists)]

Kevin DeGraaf wrote:


What am I missing?  Thanks.

  
Nothing. Keep your money in your wallet. Your users will never need or 
understand using more than 2 different calls appearances atone time. 
Even I, with several PBXs to mointor, I never us all the call 
appearances I have on a 601.


--
Chris Mason
(264) 497-5670  Fax: (264) 497-8463 

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



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Re: [asterisk-users] HPEC audio clipping

[Eric \"ManxPower\" Wieling]

What are the advantages of 9.x over the 8.x that I currently use?

Matthew Fredrickson wrote:
There's a new version of HPEC released that fixes the bug.  Try updating 
now :-)


Matthew Fredrickson

On May 16, 2007, at 2:35 PM, Noah Miller wrote:


Hi Olivier -

Our last trial was so conclusive (every call was affected), we step 
back to

previous situation without HPEC.

We will do our best to help to solve this (gathering audio captures for
instance) though it will be very hard for me to convince our customer to
try.

Being able to to limit HPEC to given channels would have certainly 
helped to

create conditions for customer acceptance.

As far as I can tell, no bug is opened on this HPEC audio, though a long
thread, a couple of weeks ago, proved us we were not the ones affected.


Do you have an old machine and a spare TDM card you could use?  Maybe
you could connect just one POTS line to another machine and grab some
audio captures.  Or maybe you can test during off-hours?

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Re: [asterisk-users] zaptel huge irq problem

[Tzafrir Cohen]

On Thu, May 17, 2007 at 07:26:10PM +0200, François Delawarde wrote:
> Hi,
> >Why are you so determined to use Asterisk in a VM? You're asking for
> >trouble. Asterisk belongs on dedicated hardware.
> >  
> I actually want to use Asterisk in a machine HOSTING a VM (that's what I 
> implied with the Dom-0 thing I said earlier), sorry for the 
> misunderstanding. I agree with you that given the state of advancement 
> of just about any 'virtualizer', I would have to be totally stupid to 
> try running Asterisk inside a VM. (I also wouldn't have asked here in 
> the first place, as I would have been totally certain that problems came 
> from the virtualizer itself)

What kind of separation do you really need?

Xen, VMWare and such are "big cannons" here. Every virtual machine will 
consume fixed ammount of memory. There is a considerable overhead for 
hardware access.

It allows you things like running different OS/distribution on each 
guest. But for some reason I'm not sure you really need that?

Will the users have direct acces to the dialplan and the rest of the
configuration? If not: just run a single instance of Asterisk.

If you do need multiple asterisk instances, verver or openvz might
help you to give a separate "container" for that user's personal usage.
Stephan has mentioned in this thread he set up several Asterisk-es on a 
vserver system.

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
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Re: [asterisk-users] zaptel huge irq problem

[Stephen Bosch]

François Delawarde wrote:

> And I thank you for that (the helping part), you've found the deep cause
> of all my zaptel problems (Xen), so please don't leave me alone! ;-)
> 
> To be a bit more constructive, I'd like to ask you or anyone that dared
> to try using Asterisk on a non-dedicated hardware, specifically those
> that tried on a machine hosting VMs the following:
> 
> - If there is no way running Asterisk with Xen, what type of
> 'hypervisor' should I use in order not to have problems? KVM?, KQemu?,
> VMWare?

The only one I would bother with is VMWare Server. It is solid, proven
technology, and they have a big team of very talented engineers who have
worked years to get the virtualization to the point where it can be sold
as an enterprise grade product.

If I were to try virtualizing anything, it would be on VMWare Server.

> - What type of problems should I expect if I dare to do that? (of
> course, Asterisk will be realtime-niced to make it more important)

Well, in particular anything that expects unfettered access to hardware
(as most realtime applications which rely on interface cards do) is
going to be vulnerable to the proclivities of the hypervisor.

Virtualization is still mostly rocket science. I have no doubt that it
is the future and one day everything will run in virtualized
environments -- but we're still a bit away from that.

Virtualization makes financial sense when you have 20 database servers
running at 10% utilization; you can drop your hardware requirements by
at least a third... but for systems relying on dedicated hardware, I
would be very careful (again -- I speak from ugly experience here).

-Stephen-
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[asterisk-users] Re: DUNDi configuration problem

[JR Richardson]

[mappings]
priv => dundi-priv-canonical,0,SIP,${IPADDR}/${NUMBER},nopartial
priv => dundi-priv-customers,100,SIP,${IPADDR}/${NUMBER},nopartial
priv => dundi-priv-via-pstn,400,SIP,${IPADDR}/${NUMBER},nopartial



Your mappings are wrong, this is for IAX, for SIP to work, it should be:

priv => dundi-priv-canonical,0,SIP,${NUMBER}@"the real IP Address",nopartial

The rest looked ok I think.

Good luck.

JR
--
JR Richardson
Engineering for the Masses
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[asterisk-users] Re: FastAGI hangs up channel if server is not available

[JR Richardson]

Running 1.2.14

When I call a FastAGI script such as this script for an incoming call:

[calldirect]
exten=>s,1,Answer()
exten=>s,2,AGI(agi://192.168.1.175/calldirect?check&${CALLERID(num)})
exten=>s,3,Goto(check_time,s,1)

and the FastAGI server is not running (Asterisk gets "connection
refused" TCP error), Asterisk just terminates the call like so:

May 17 12:58:00 WARNING[10154]: res_agi.c:210 launch_netscript: Connect
to 'agi://192.168.1.175/calldirect?check&NN' failed: Connection
refused
  == Spawn extension (remove_caller, s, 2) exited non-zero on
'SIP/datatrak-0978b670'

The text of the relevant wiki article
(http://www.voip-info.org/wiki-Asterisk+FastAGI) infers that execution
should roll through to the next executable line in the context where it
was called in the case of an error in executing the FastAGI:

"Asterisk 1.2
Under Asterisk 1.2, if a request to a FastAGI service failed for any
reason, there was no way to determine this from the dialplan. The
recommended action is to set a channel variable before calling the
FastAGI, setting that variable to a known value within the FastAGI, and
then checking that variable once the AGI has returned..."


I had a similar issue, we resolved it by calling a local agi script on
the local pbx that in turn calls the fast agi on the remote host.  The
local agi was written in perl, which, if the remote agi does not
respond, then the call is given back to the dialplan to the next
priority.

So for your example, do something like this:

[calldirect]
exten=>s,1,Answer()
exten=>s,2,AGI(test.agi&${CALLERID(num)})
exten=>s,3,Goto(check_time,s,1)


Perl AGI File 'test.agi' in the local asterisk agi directory:

---start of
file-
#!/usr/bin/perl

#
#Modules to Use   ###
#

use Asterisk::AGI;
$AGI = new Asterisk::AGI;
my %input = $AGI->ReadParse();
# Set variables according to supplied arquments
$number = $ARGV[0];
$AGI->exec("agi","agi://192.168.1.175/calldirect?check&number=$number");

end of
file---


Also you need the asterisk perl agi modules at http://asterisk.gnuinter.net/

Good luck.

JR
--
JR Richardson
Engineering for the Masses
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Re: [asterisk-users] HPEC audio clipping

[Matthew Fredrickson]

There's a new version of HPEC released that fixes the bug.  Try 
updating now :-)


Matthew Fredrickson

On May 16, 2007, at 2:35 PM, Noah Miller wrote:


Hi Olivier -

Our last trial was so conclusive (every call was affected), we step 
back to

previous situation without HPEC.

We will do our best to help to solve this (gathering audio captures 
for
instance) though it will be very hard for me to convince our customer 
to

try.

Being able to to limit HPEC to given channels would have certainly 
helped to

create conditions for customer acceptance.

As far as I can tell, no bug is opened on this HPEC audio, though a 
long
thread, a couple of weeks ago, proved us we were not the ones 
affected.


Do you have an old machine and a spare TDM card you could use?  Maybe
you could connect just one POTS line to another machine and grab some
audio captures.  Or maybe you can test during off-hours?


- Noah
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[asterisk-users] Re: FastAGI hangs up channel if server is not available

[Tony Mountifield]

In article <[EMAIL PROTECTED]>,
Lee Jenkins <[EMAIL PROTECTED]> wrote:
> Tony Mountifield wrote:
> > 
> > No, the information is wrong. You need to make a small mod to res/res_agi.c
> > in order to trap this case. Look for the second occurrence of
> > LOCAL_USER_REMOVE, and add the lines just before it as show by this
> > patch extract (ignore the line numbers; I have other mods too):
> > 
> > @@ -2065,6 +2106,9 @@
> > close(fds[1]);
> > if (efd > -1)
> > close(efd);
> > +   } else {
> > +   pbx_builtin_setvar_helper(chan, "AGISTATUS", "NOTFOUND");
> > +   res = 0;
> > }
> > LOCAL_USER_REMOVE(u);
> > return res;
> > 
> > With this change, a refused connection will not hang up the channel, but
> > instead will set the AGISTATUS variable to "NOTFOUND".
> > 
> > Cheers
> > Tony
> 
> Nice.  Thanks, Tony.
> 
> Has this patched been accepted/committed in subsequent releases such as 
> a 1.2.17?

No, I have never got around to submitting it.

Cheers
Tony
-- 
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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[asterisk-users] Digium HPEC 9.00.003 released

[Kevin P. Fleming]

I have just placed HPEC version 9.00.003 onto the Digium FTP site; this
release cures the recently discovered bug where incoming audio was
choppy or lost completely under specific circumstances (did not affect
all users, but if a user's system had the problem it would have it on
all calls).
-- 
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The Genuine Asterisk Experience" (TM)
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Re: [asterisk-users] Re: FastAGI hangs up channel if server is not available

[Sean Bright]

Woops.  The documentation problem was my fault.  I was basing that on a
patched version of asterisk that jumped to another priority on FastAGI
failure.  I'll update the wiki.

Sean

On 5/17/07, Tony Mountifield <[EMAIL PROTECTED]> wrote:


In article <[EMAIL PROTECTED]>,
Lee Jenkins <[EMAIL PROTECTED]> wrote:
>
> Running 1.2.14
>
> When I call a FastAGI script such as this script for an incoming call:
>
> [calldirect]
> exten=>s,1,Answer()
> exten=>s,2,AGI(agi://192.168.1.175/calldirect?check&${CALLERID(num)})
> exten=>s,3,Goto(check_time,s,1)
>
> and the FastAGI server is not running (Asterisk gets "connection
> refused" TCP error), Asterisk just terminates the call like so:
>
> May 17 12:58:00 WARNING[10154]: res_agi.c:210 launch_netscript: Connect
> to 'agi://192.168.1.175/calldirect?check&NN' failed: Connection
> refused
>== Spawn extension (remove_caller, s, 2) exited non-zero on
> 'SIP/datatrak-0978b670'
>
> The text of the relevant wiki article
> (http://www.voip-info.org/wiki-Asterisk+FastAGI) infers that execution
> should roll through to the next executable line in the context where it
> was called in the case of an error in executing the FastAGI:
>
> "Asterisk 1.2
> Under Asterisk 1.2, if a request to a FastAGI service failed for any
> reason, there was no way to determine this from the dialplan. The
> recommended action is to set a channel variable before calling the
> FastAGI, setting that variable to a known value within the FastAGI, and
> then checking that variable once the AGI has returned..."
>
> Is there a setting/var that I am missing?

No, the information is wrong. You need to make a small mod to
res/res_agi.c
in order to trap this case. Look for the second occurrence of
LOCAL_USER_REMOVE, and add the lines just before it as show by this
patch extract (ignore the line numbers; I have other mods too):

@@ -2065,6 +2106,9 @@
close(fds[1]);
if (efd > -1)
close(efd);
+   } else {
+   pbx_builtin_setvar_helper(chan, "AGISTATUS", "NOTFOUND");
+   res = 0;
}
LOCAL_USER_REMOVE(u);
return res;

With this change, a refused connection will not hang up the channel, but
instead will set the AGISTATUS variable to "NOTFOUND".

Cheers
Tony
--
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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Re: [asterisk-users] Blacklist

[Jonathan Creasy]

Looks to me like you are saving it to the Asterisk DB.

http://www.asteriskguru.com/tutorials/dbget_function.html

-Jonathan

Nitesh Divecha wrote:

Hello All,

I was wondering where does Asterisk stores the blacklist numbers?

I looked into the dialplan and it shows that it 
*"Set(DB(blacklist/${blacknr})=1)"* the number... Does it save in 
MySQL DB?


hyperion*CLI> show dialplan app-blacklist-add
[ Context 'app-blacklist-add' created by 'pbx_config' ]
 '1' =>1. *Set(DB(blacklist/${blacknr})=1)*
[pbx_config]
   2. Playback(num-was-successfully) 
[pbx_config]
   3. Playback(added)
[pbx_config]
   4. Wait(1)
[pbx_config]
   5. Hangup()   
[pbx_config]
 's' =>1. Answer()   
[pbx_config]
   2. Wait(1)
[pbx_config]
   3. Playback(enter-num-blacklist)  
[pbx_config]
   4. Set(TIMEOUT(digit)=5)  
[pbx_config]
   5. Set(TIMEOUT(response)=60)  
[pbx_config]
   6. Read(blacknr|then-press-pound) 
[pbx_config]
   7. SayDigits(${blacknr})  
[pbx_config]
   8. Playback(if-correct-press) 
[pbx_config]
   9. Playback(digits/1) 
[pbx_config]
[end]  10. Noop(Waiting for input)   
[pbx_config]
 Include =>'app-blacklist-add-custom'
[pbx_config]

hyperion*CLI>


Thanks,
Nitesh


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[asterisk-users] OK to have Asterisk and clients behind firewalls?

[Vincent Delporte]

Hi

	To investigate the UNREACHABLE issue I'm having, I need to have 
confirmation that it's OK for the Asterisk server to be behind a NAT 
router, and also have clients elsewhere on the Net behind their own NAT router?


I know that clients must use STUN to resolve their public IP and punch UDP 
holes in their firewall, but is there something special that must be done 
in the configuration of Asterisk so it knows it's living in a private 
network, behind a NAT router?


And if someone knows of tools to investigate SIP issues, especially a 
text-based sniffer (no X available in the Asterisk live CD I'm using), I'm 
interested :-)


Thank you.

PS: FWIW, extension 203 (softphone) and 204 (IP phone) are both located on 
the same network and behind a NAT router, and both connect out to an 
Asterisk server somewhere on the Net behing its own NAT router:


slast*CLI> sip show peers
Name/username  HostDyn Nat ACL 
Port Status
204/20482.237.x.y D  5060 UNREACHABLE 


203/20382.237.x.y D   N  46838OK (925 ms)

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Re: [asterisk-users] asterisk setup for church / conference call/ speaker system integration

[Steve Prior]

Dean Collins wrote:


Worth looking at using Talkshoe for this application maybe?


My personal podcast authoring tool is vi (and you thought it was
just a website authoring tool...) so I don't have any user friendly
recommendations, but there's got to be a lot out there.

Steve
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Re: [asterisk-users] Re: FastAGI hangs up channel if server is not available

[Lee Jenkins]

Tony Mountifield wrote:

In article <[EMAIL PROTECTED]>,
Lee Jenkins <[EMAIL PROTECTED]> wrote:

Running 1.2.14

When I call a FastAGI script such as this script for an incoming call:

[calldirect]
exten=>s,1,Answer()
exten=>s,2,AGI(agi://192.168.1.175/calldirect?check&${CALLERID(num)})
exten=>s,3,Goto(check_time,s,1)

and the FastAGI server is not running (Asterisk gets "connection 
refused" TCP error), Asterisk just terminates the call like so:


May 17 12:58:00 WARNING[10154]: res_agi.c:210 launch_netscript: Connect 
to 'agi://192.168.1.175/calldirect?check&NN' failed: Connection 
refused
   == Spawn extension (remove_caller, s, 2) exited non-zero on 
'SIP/datatrak-0978b670'


The text of the relevant wiki article 
(http://www.voip-info.org/wiki-Asterisk+FastAGI) infers that execution 
should roll through to the next executable line in the context where it 
was called in the case of an error in executing the FastAGI:


"Asterisk 1.2
Under Asterisk 1.2, if a request to a FastAGI service failed for any 
reason, there was no way to determine this from the dialplan. The 
recommended action is to set a channel variable before calling the 
FastAGI, setting that variable to a known value within the FastAGI, and 
then checking that variable once the AGI has returned..."


Is there a setting/var that I am missing?


No, the information is wrong. You need to make a small mod to res/res_agi.c
in order to trap this case. Look for the second occurrence of
LOCAL_USER_REMOVE, and add the lines just before it as show by this
patch extract (ignore the line numbers; I have other mods too):

@@ -2065,6 +2106,9 @@
close(fds[1]);
if (efd > -1)
close(efd);
+   } else {
+   pbx_builtin_setvar_helper(chan, "AGISTATUS", "NOTFOUND");
+   res = 0;
}
LOCAL_USER_REMOVE(u);
return res;

With this change, a refused connection will not hang up the channel, but
instead will set the AGISTATUS variable to "NOTFOUND".

Cheers
Tony


Nice.  Thanks, Tony.

Has this patched been accepted/committed in subsequent releases such as 
a 1.2.17?


Thanks again,

Lee


--

Warm Regards,

Lee



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[asterisk-users] AGI and fork()

[GDrayer]

I'm writing a series of AGI scripts in C and I was curious; if I fork()
in my AGI script and the parent exits and returns to the dialplan, will
the child still have connection with asterisk, i.e. will I be able to
SET channel variables, EXEC applications, etc.?

Thanks once again.
George
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[asterisk-users] Multiple lines on Linksys/Sipura phones

[Kevin DeGraaf]

I'm going to be deploying around 30 IP phones with Asterisk in the near
future.  I've tentatively settled on the Linksys/Sipura SPA9xx family.

I am unclear on the notion of "lines" in the context of SIP phones like
these.  The SPA942 model has a 2-line-to-4-line upgrade available, but I
don't know why I'd need to purchase it.

I have tested a SPA942 with Asterisk, and even without the upgrade, I
can easily send/receive/hold four separate calls at a time, using the
four line keys.

What am I missing?  Thanks.

-- 
Kevin DeGraaf
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Re: [asterisk-users] Blacklist

[Derek Whitten]

Nitesh Divecha wrote:
> Hello All,
> 
> I was wondering where does Asterisk stores the blacklist numbers?
> 
> I looked into the dialplan and it shows that it
> *"Set(DB(blacklist/${blacknr})=1)"* the number... Does it save in MySQL DB?
> 
> hyperion*CLI> show dialplan app-blacklist-add
> [ Context 'app-blacklist-add' created by 'pbx_config' ]
>  '1' =>1. *Set(DB(blacklist/${blacknr})=1)*   
> [pbx_config]
>2. Playback(num-was-successfully)
> [pbx_config]
>3. Playback(added)   
> [pbx_config]
>4. Wait(1)   
> [pbx_config]
>5. Hangup()  
> [pbx_config]
>  's' =>1. Answer()  
> [pbx_config]
>2. Wait(1)   
> [pbx_config]
>3. Playback(enter-num-blacklist) 
> [pbx_config]
>4. Set(TIMEOUT(digit)=5) 
> [pbx_config]
>5. Set(TIMEOUT(response)=60) 
> [pbx_config]
>6. Read(blacknr|then-press-pound)
> [pbx_config]
>7. SayDigits(${blacknr}) 
> [pbx_config]
>8. Playback(if-correct-press)
> [pbx_config]
>9. Playback(digits/1)
> [pbx_config]
> [end]  10. Noop(Waiting for input)  
> [pbx_config]
>  Include =>'app-blacklist-add-custom'   
> [pbx_config]
> hyperion*CLI>
> 
> 
> Thanks,
> Nitesh
> 
> 
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it's Stored in the astdb





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Re: [asterisk-users] DTMF not working using *98, but OK on inbound routes?

[Anthony Francis]

You should really include the dialplan snippet that controls *98, so 
that people can formulate better responses, so that being said, I have 
to ask, did you do an Answer() step?


Doug wrote:

Has anyone seen anything like this:

I dial *98.  Asterisk says "Password?"   I punch in
the password, and the system doesn't recognize the
tones.

However, if I dial my own number and ignore the
incoming call, it goes to voicemail, and then
I can get into voicemail.

I have a sneaking suspicion that Asterisk is
somehow not recognizing the DTMF tones somewhere
along the way.

This happens intermittently with Linksys ATAs and
Polycom phones.  Using a Cisco 3640 VOIP router.

Any ideas on what to check?


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[asterisk-users] Cascading Queues

[Jason Adams]

Hey Everyone,
 
Have a couple of questions here..
 
Scenario 1:
We are working with a client that currently has one support queue with
about 10 agents.  They are starting to get pretty long hold times for
their customers and they have requested three queues.  Queue 1 will have
a timeout of 4 minutes.  After that it will move to Queue 2 which has a
default timeout of 3 minutes.  After that we will transfer the call to
the receptionist who will either take a message or put them back in the
queue with a higher priority if they want to continue to hold.  Queue 2
will have more agents in that queue plus the agents that were in Queue
1.
 
Question:
Can I have the same agents in multiple queues to work the way I'm
thinking above?  So if the caller reaches Queue 2 the agents from Queue
1 will be available if they get off a call, plus new agents are added
into Queue 2.
 
So I'm thinking something like:
exten => s,1,Queue(support1)
exten => s,2,Queue(support2)
exten => s,3,Dial(SIP/${RECEPTIONIST})
 
Then the receptionist would just dial a special extension which would
add priority=10 to the queue.
 
Scenario 2:
This same customer is starting to sell their product internationally.
They are purchasing VOIP DID's from various countries for local calls
from that area.  Would this just be like setting up a regular VOIP line
to register the account in sip.conf and then creating a context for
those countries so we know where they are coming from?

 Thanks for helping me think through this!

 - Jason

 
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Re: [asterisk-users] GUI: Not Found. Move along

[Tim Verscheure]

Still nothing. I'll give my config files:

manager.conf

;
; Asterisk Call Management support
;

; By default asterisk will listen on localhost only.
[general]
displaysystemname = yes
enabled = yes
webenabled = yes
port = 5038
httptimeout = 60
bindaddr = 0.0.0.0

; No access is allowed by default.
; To set a password, create a file in /etc/asterisk/manager.d
; use creative permission games to allow other serivces to create their own
; files
#include "manager.d/*.conf"

[admin]
secret = javali
deny=0.0.0.0/0.0.0.0
permit=127.0.0.1/255.255.255.0
permit=192.168.1.68/255.255.255.0
read = system,call,log,verbose,command,agent,user,config
write = system,call,log,verbose,command,agent,user,config

[panel]
secret = javali
deny=0.0.0.0/0.0.0.0
permit=192.168.1.68/255.255.255.0
permit=127.0.0.1/255.255.255.0
read = system,call,log,verbose,command,agent,user,config
write = system,call,log,verbose,command,agent,user,config


http.conf
---
;
; Asterisk Builtin mini-HTTP server
;
;
[general]
;
; Whether HTTP interface is enabled or not.  Default is no.
;
enabled=yes
;
; Whether Asterisk should serve static content from http-static
; Default is no.
;
enablestatic=yes
;
; Address to bind to.  Default is 0.0.0.0
;
bindaddr=0.0.0.0
;
; Port to bind to (default is 8088)
;
bindport=8088
;
; Prefix allows you to specify a prefix for all requests
; to the server.  The default is "asterisk" so that all
; requests must begin with /asterisk
;
;prefix=asterisk

; The post_mappings section maps URLs to real paths on the filesystem.  If a
; POST is done from within an authenticated manager session to one of the
; configured POST mappings, then any files in the POST will be placed in the
; configured directory.
;
;[post_mappings]
;
; In this example, if the prefix option is set to "asterisk", then using the
; POST URL: /asterisk/uploads will put files in /var/lib/asterisk/uploads/.
;uploads = /var/lib/asterisk/uploads/
;


thanks in advance, Tim


2007/5/17, Troy Ayers <[EMAIL PROTECTED]>:

Tim Verscheure wrote:
> Hi there,
>
> I just installed the GUI for Asterisk 1.4.4 and correctly set my
> settings but when I use my browser to access it, it gives me an error
> saying "Not Found. Nothing to see here, move along" with "asterisk" in
> the header and footer...
>
> anyone had this problemn before?
>
>
> greetz
Try https:// not http://
-Troy

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Re: [asterisk-users] asterisk setup for church / conference call / speaker system integration

[Tim Litwiller]

Drew Gibson wrote:

Tim Litwiller wrote:

David Gomillion wrote:
On 5/17/07, *Tim Litwiller* <[EMAIL PROTECTED] 
> wrote:


We have several people in our church that recently became
disabled. I am
thinking of setting up an asterisk server and several phone 
lined so

that they can call in to church during services to listen to the
service.

If it were me, I would:

1. create a conference room
2. create a .call file that dials into the speaker system
3. create .call files to dial the participants, muting them

This can obviously be scripted very easily. A simple cron job 
copying the .call files should do nicely. If the disabled persons 
wish to no longer participate, you can simply delete that .call 
file; conversely, if you need to add someone, you can just create 
another one. Then you won't have to worry about incoming phone 
numbers and coordinating with the private school, you can bring it 
in when you need it, etc.


That does simplify it a bit - but is probably not flexible enough. I 
don't want to have to have someone call and request to be added to 
the conference if they are sick sunday morning. We will publish the 
number for anyone in the congregation to use as they need it.  For 
those that always listen at home tho this would work fine.


Oh and we will also want to record the services so that if someone 
wants a copy or to listen later they can call in to listen or we can 
burn then a copy. So I'll need to program a recording menu system 
that is somewhat automated and lists the last few weeks of services 
by date or something.


We are currently using freeconferencecall.com but it is a long 
distance call from church and from all of our members.


Use GotoIfTime to divert incoming calls, only during regular service 
times, to a voice menu with a choice of listen live, or make a call to 
extension.
Use David's call file suggestion to connect the speakers to the 
conference and record (Monitor) that channel.
Have an extension that callers can dial at anytime for a voice menu 
with a choice of recordings.


regards,

Drew


Good suggestions everyone, thanks.
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[asterisk-users] Re: UK zaptel and zapata.conf for TDM400P

[Chris Earle]

2 TDM400P's in one machine can co-exist?  I thought this was near impossible
 anyone confirm?


--
Chris



"Gordon Henderson" <[EMAIL PROTECTED]> wrote in message
news:[EMAIL PROTECTED]
> On Sat, 21 Apr 2007, Steve Kennedy wrote:
>
> > Has anyone got a sensible zaptel.conf and zapata.conf for 2 TDM400P's
> > working with UK set-up.
> >
> > They're set-up with 7 analogue phones and 1 PSTN port.
> >
> > Currently zaptel.conf has
> > fxoks=1-7
> > fxsks=8
> > loadzone=uk
> > defaultzone=uk
> >
> > It's really zapata.conf that would be useful.
> >
> > Currently using the zaptel/asterisk that comes with Ubuntu (latest)
> > which needed a bit of tweaking (1.2.16), but could compile latest 1.4
> > release.
>
> See below for a system I'm using right now. 2 PSTN ports and 1 analogue
> port. This is a compile & build from scratch system, not supplied in a
> package or using trixbox, etc.
>
> Curious about your 2 x TDM400 cards though - I'm presuming you've got no
> interrupt issues, etc. ?
>
> ...
>
> Based on this zaptel.conf:
>
>fxoks=1
>fxsks=3
>fxsks=4
>loadzone=uk
>defaultzone=uk
>
> I have:
>
> [trunkgroups]
>
> [channels]
>
> ; Default settings applicable to all channels
>
> usecallerid=yes
> cidsignalling=v23
> cidstart=polarity
> hidecallerid=no
> callwaiting=no
> threewaycalling=yes
> transfer=yes
> echocancel=yes
> echotraining=yes
> echocancelwhenbridged=yes
> immediate=no
> faxdetect=no
>
> ; Channel 1: Local analogue line
> context=internal
> group=0
> signalling=fxo_ks
> sendcalleridafter=2
> rxgain=0
> txgain=0
> mailbox=103
> callerid=Analogue Port <103>
> channel => 1
>
> ; Channel 3: PSTN line
> context=incoming
> group=1
> usecallerid=yes
> faxdetect=none
> signalling=fxs_ks
> rxgain=7
> txgain=7
> callerid=asreceived
> channel => 3
>
> ; Channel 4: PSTN line
> context=incoming
> group=1
> usecallerid=yes
> faxdetect=none
> signalling=fxs_ks
> rxgain=8
> txgain=8
> callerid=asreceived
> channel => 4
>
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Re: [asterisk-users] FastAGI hangs up channel if server is not available

[Lee Jenkins]

Lee Jenkins wrote:


Hi all,

Running 1.2.14

When I call a FastAGI script such as this script for an incoming call:

[calldirect]
exten=>s,1,Answer()
exten=>s,2,AGI(agi://192.168.1.175/calldirect?check&${CALLERID(num)})
exten=>s,3,Goto(check_time,s,1)

and the FastAGI server is not running (Asterisk gets "connection 
refused" TCP error), Asterisk just terminates the call like so:


May 17 12:58:00 WARNING[10154]: res_agi.c:210 launch_netscript: Connect 
to 'agi://192.168.1.175/calldirect?check&NN' failed: Connection 
refused
  == Spawn extension (remove_caller, s, 2) exited non-zero on 
'SIP/datatrak-0978b670'


The text of the relevant wiki article 
(http://www.voip-info.org/wiki-Asterisk+FastAGI) infers that execution 
should roll through to the next executable line in the context where it 
was called in the case of an error in executing the FastAGI:


"Asterisk 1.2
Under Asterisk 1.2, if a request to a FastAGI service failed for any 
reason, there was no way to determine this from the dialplan. The 
recommended action is to set a channel variable before calling the 
FastAGI, setting that variable to a known value within the FastAGI, and 
then checking that variable once the AGI has returned..."





Apparently, this is a standing issue with FastAGI (at least ins 1.2 
branch).  I found a work around using the "h".


Bread Crumbs follow:

Work around using "h" extension:
http://lists.digium.com/pipermail/asterisk-users/2006-August/161459.html

Still a PIA to have to include pseudo exception handling each time a 
FastAGI is referenced.


There is also a bug report from '05 that I came across:

http://bugs.digium.com/view.php?id=4029

But it looks like it was stalled.

--

Warm Regards,

Lee



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Re: [asterisk-users] DTMF not working using *98, but OK on inbound routes?

[Doug]

At 21:40 5/16/2007, Doug wrote:
>Has anyone seen anything like this:
>
>I dial *98.  Asterisk says "Password?"   I punch in
>the password, and the system doesn't recognize the
>tones.
>
>However, if I dial my own number and ignore the
>incoming call, it goes to voicemail, and then
>I can get into voicemail.
>
>I have a sneaking suspicion that Asterisk is
>somehow not recognizing the DTMF tones somewhere
>along the way.
>
>This happens intermittently with Linksys ATAs and
>Polycom phones.  Using a Cisco 3640 VOIP router.
>
>Any ideas on what to check?

Looks like I found a solution.

Apparently, in Asterisk (1.2.18) the extension needs
to be configured as RFC2833, while in the device
itself, it needs to be set to SIP INFO.

It doesn't make sense, but it seems to work.

Developers, why would this be?  Is this a bug?


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[asterisk-users] Re: FastAGI hangs up channel if server is not available

[Tony Mountifield]

In article <[EMAIL PROTECTED]>,
Lee Jenkins <[EMAIL PROTECTED]> wrote:
> 
> Running 1.2.14
> 
> When I call a FastAGI script such as this script for an incoming call:
> 
> [calldirect]
> exten=>s,1,Answer()
> exten=>s,2,AGI(agi://192.168.1.175/calldirect?check&${CALLERID(num)})
> exten=>s,3,Goto(check_time,s,1)
> 
> and the FastAGI server is not running (Asterisk gets "connection 
> refused" TCP error), Asterisk just terminates the call like so:
> 
> May 17 12:58:00 WARNING[10154]: res_agi.c:210 launch_netscript: Connect 
> to 'agi://192.168.1.175/calldirect?check&NN' failed: Connection 
> refused
>== Spawn extension (remove_caller, s, 2) exited non-zero on 
> 'SIP/datatrak-0978b670'
> 
> The text of the relevant wiki article 
> (http://www.voip-info.org/wiki-Asterisk+FastAGI) infers that execution 
> should roll through to the next executable line in the context where it 
> was called in the case of an error in executing the FastAGI:
> 
> "Asterisk 1.2
> Under Asterisk 1.2, if a request to a FastAGI service failed for any 
> reason, there was no way to determine this from the dialplan. The 
> recommended action is to set a channel variable before calling the 
> FastAGI, setting that variable to a known value within the FastAGI, and 
> then checking that variable once the AGI has returned..."
> 
> Is there a setting/var that I am missing?

No, the information is wrong. You need to make a small mod to res/res_agi.c
in order to trap this case. Look for the second occurrence of
LOCAL_USER_REMOVE, and add the lines just before it as show by this
patch extract (ignore the line numbers; I have other mods too):

@@ -2065,6 +2106,9 @@
close(fds[1]);
if (efd > -1)
close(efd);
+   } else {
+   pbx_builtin_setvar_helper(chan, "AGISTATUS", "NOTFOUND");
+   res = 0;
}
LOCAL_USER_REMOVE(u);
return res;

With this change, a refused connection will not hang up the channel, but
instead will set the AGISTATUS variable to "NOTFOUND".

Cheers
Tony
-- 
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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[asterisk-users] Blacklist

[Nitesh Divecha]

Hello All,

I was wondering where does Asterisk stores the blacklist numbers?

I looked into the dialplan and it shows that it 
*"Set(DB(blacklist/${blacknr})=1)"* the number... Does it save in MySQL DB?


hyperion*CLI> show dialplan app-blacklist-add
[ Context 'app-blacklist-add' created by 'pbx_config' ]
 '1' =>1. *Set(DB(blacklist/${blacknr})=1)*
[pbx_config]
   2. Playback(num-was-successfully) 
[pbx_config]
   3. Playback(added)
[pbx_config]
   4. Wait(1)
[pbx_config]
   5. Hangup()   
[pbx_config]
 's' =>1. Answer()   
[pbx_config]
   2. Wait(1)
[pbx_config]
   3. Playback(enter-num-blacklist)  
[pbx_config]
   4. Set(TIMEOUT(digit)=5)  
[pbx_config]
   5. Set(TIMEOUT(response)=60)  
[pbx_config]
   6. Read(blacknr|then-press-pound) 
[pbx_config]
   7. SayDigits(${blacknr})  
[pbx_config]
   8. Playback(if-correct-press) 
[pbx_config]
   9. Playback(digits/1) 
[pbx_config]
[end]  10. Noop(Waiting for input)   
[pbx_config]
 Include =>'app-blacklist-add-custom'
[pbx_config]

hyperion*CLI>


Thanks,
Nitesh


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Re: [asterisk-users] zaptel huge irq problem

[Stephen Davies]

On 14/05/07, Salvatore Giudice
<[EMAIL PROTECTED]> wrote:

Try switching to a Sangoma card. You won't have anymore  IRQ issues once you
abandon Digium hardware.


Its not true, by the way.

I've assisted more than one person using a Sangoma who was having
issues caused by interrupt stuff.

And it was the same sort of things that might affect a Digium board-
motherboard raid disabling interrupts, sharing an IRQ with a
heavy-interrupting LAN card, etc.

Not suprising since its the same underlying problem - excessing
interrupt handling latency.

Steve
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Re: [asterisk-users] zaptel huge irq problem

[Stephen Davies]

Hi,

I want to quickly mention that I've had great success with running
Asterisk in the under-appreciated Linux-VServer environment.

This is not so much a virtualisation environment as a system partioner
on steroids.  Nothing to do with running windows on Linux and
suchlike, but a good way to run lots of Asterisk and other stuff
isolated from each other.

There is only one kernel, and hardware is not virtualised.  A particular guest
We run about 10 Asterisk instances, together with web servers, Mysql
and more.  All on a 1GB RAM Pentium D box.  CPU and memory to spare.

hildegard steve # vserver-stat
CTX   PROCVSZRSS  userTIME   sysTIMEUPTIME NAME
0   82 266.5M  20.6M  21h19m47   8h04m51  26d02h24 root server
8   63 710.6M   1.4G  14m10s98   5m31s37  26d02h20 ctel_web
9   13   1.1G  31.8M  16m28s43  14m02s46  25d05h57 ctel_pbx
10  19 694.8M 172.7M   2h01m43  19m59s65  20d07h58 voipconnect
11   8 927.9M  98.9M  40m43s42  10m09s30  26d02h21 ctel_admin
12   5 210.5M  21.5M   5h13m35  36m41s61  26d02h21 ctel_db
13   8 903.4M  55.2M   3m00s10   1m09s40  26d02h20 ctel_intranet
15   5   213M   1.2M   9m09s35  12m51s00  26d02h19 xconnect
33  29 261.2M  22.7M   0m07s65   0m09s13   3d08h43 testtrunk
56  13   1.1G10M   1m44s45   1m18s94  26d02h21 aaa
57  13   1.1G  13.9M   8m47s52   8m46s10  26d02h21 bbb
58  13   1.1G  16.1M  54m29s99  24m46s22  26d02h14 ccc
60   9 293.4M  31.9M  10h44m42   2h00m55  26d02h20 ddd
61  13   1.1G   6.7M  12m45s11  13m26s42  26d02h19 eee

(26 days uptime?  Our hosting provider had to do power maintenance.
We've never had a crash of the host system).

My zttest inside a guest machine:

voipconnect zaptel # ./zttest
Opened pseudo zap interface, measuring accuracy...
100.00% 99.987793% 100.00% 99.987793% 100.00% 100.00%
99.987793%
100.00% 100.00% 99.987793% 100.00% 99.987793% 100.00%
100.00% 99.987793%
100.00% 99.987793% 100.00% 99.987793% 100.00% 100.00%
99.987793% 100.00%
99.987793% 100.00% 100.00% 99.987793% 100.00% 99.987793%
100.00% 100.00%
--- Results after 31 passes ---
Best: 100.00 -- Worst: 99.987793 -- Average: 99.995277


Try that with Xen or VMWare.

http://www.linux-vserver.org/

(Our host is hardened gentoo with PaX and GRSecurity, plus vserver;
guests are gentoo too, though VServer does support guests being
different distributions).

Hope this helps someone,
Steve Davies
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RE: [asterisk-users] asterisk setup for church / conference call/ speaker system integration

[Dean Collins]

Worth looking at using Talkshoe for this application maybe?

 

Regards,

Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
+1-212-203-4357 Ph
+1-917-207-3420 Mb
+61-2-9016-5642 (Sydney in-dial).

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Steve Prior
> Sent: Thursday, 17 May 2007 12:53 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] asterisk setup for church / conference
call/ speaker
> system integration
> 
> Tim Litwiller wrote:
> > Oh and we will also want to record the services so that if someone
wants
> > a copy or to listen later they can call in to listen or we can burn
then
> > a copy. So I'll need to program a recording menu system that is
somewhat
> > automated and lists the last few weeks of services by date or
something.
> 
> This part sounds like a podcast to me.
> 
> Steve
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[asterisk-users] Compiling DBQuery

[Douglas Garstang]

Has anyone tried to compile the current version of MySQLPool from 
http://www.yosd.at   against Asterisk 1.4.4?

 

It seems to not compile...

 

[EMAIL PROTECTED] res_mysqlpool]# make

gcc -pipe  -pipe  -Wall -Wstrict-prototypes -Wmissing-prototypes 
-Wmissing-declarations -g  -Iinclude -I../include 
-I/usr/local/mysql/include/mysql -D_REENTRANT -D_GNU_SOURCE  -O6-DUSE_CVS   
 -fPIC -c -o res_mysqlpool.o res_mysqlpool.c

In file included from res_mysqlpool.c:35:

res_mysqlpool.h:54: warning: function declaration isnât a prototype

res_mysqlpool.c:87: warning: data definition has no type or storage class

res_mysqlpool.c:87: warning: type defaults to âintâ in declaration of 
âSTANDARD_LOCAL_USERâ

res_mysqlpool.c:89: warning: data definition has no type or storage class

res_mysqlpool.c:89: warning: type defaults to âintâ in declaration of 
âLOCAL_USER_DECLâ

res_mysqlpool.c:199: warning: function declaration isnât a prototype

res_mysqlpool.c: In function âget_host_listâ:

res_mysqlpool.c:648: warning: assignment discards qualifiers from pointer 
target type

res_mysqlpool.c:649: warning: assignment discards qualifiers from pointer 
target type

res_mysqlpool.c:656: warning: assignment discards qualifiers from pointer 
target type

res_mysqlpool.c:675: warning: assignment discards qualifiers from pointer 
target type

res_mysqlpool.c:680: warning: assignment discards qualifiers from pointer 
target type

res_mysqlpool.c:686: warning: assignment discards qualifiers from pointer 
target type

res_mysqlpool.c:695: warning: assignment discards qualifiers from pointer 
target type

res_mysqlpool.c:703: warning: assignment discards qualifiers from pointer 
target type

res_mysqlpool.c: At top level:

res_mysqlpool.c:946: warning: no previous prototype for âunload_moduleâ

res_mysqlpool.c: In function âunload_moduleâ:

res_mysqlpool.c:957: error: âSTANDARD_HANGUP_LOCALUSERSâ undeclared (first use 
in this function)

res_mysqlpool.c:957: error: (Each undeclared identifier is reported only once

res_mysqlpool.c:957: error: for each function it appears in.)

res_mysqlpool.c: At top level:

res_mysqlpool.c:964: warning: no previous prototype for âload_moduleâ

res_mysqlpool.c:985: warning: no previous prototype for âreloadâ

res_mysqlpool.c:992: warning: no previous prototype for âdescriptionâ

res_mysqlpool.c:997: warning: no previous prototype for âusecountâ

res_mysqlpool.c: In function âusecountâ:

res_mysqlpool.c:999: warning: implicit declaration of function 
âSTANDARD_USECOUNTâ

res_mysqlpool.c: At top level:

res_mysqlpool.c:1004: warning: function declaration isnât a prototype

make: *** [res_mysqlpool.o] Error 1

 

Doug.

 

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Re: [asterisk-users] GUI: Not Found. Move along

[Troy Ayers]

Tim Verscheure wrote:

Hi there,

I just installed the GUI for Asterisk 1.4.4 and correctly set my
settings but when I use my browser to access it, it gives me an error
saying "Not Found. Nothing to see here, move along" with "asterisk" in
the header and footer...

anyone had this problemn before?


greetz

Try https:// not http://
-Troy

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Re: [asterisk-users] CDR changes in 1.4.3?

[Bryan Laird]

I did a quick search through the online forums for reference to CALEA  
and didn't see much,  What is the stance with asterisk and CALEA  
compliance.  My assumption is and correct me if I'm off base
it's a pbx box not an ss7 so any did's / npa-nxx are being delivered  
form a LEC.  That mean the LEC is required to be complaint however  
asterisk in it's own guts is not.  That being said if you were trying
to be compliant the solution would be a external device that one of  
the clearing houses can talk to and setup the monitoring.  Am I  
entirely off base here or am I looking at this the correct way?






From a billing standpoint no whats the point? For statistical
purposes
I think its useful. For VoIP serviceprovider also very useful
customer
probably wants full call logs. I don't think your idea is too much
CALEA-compliant either.


Viva la CALEA! OK. Forget the filtering. The Gov shall have  
everything.


-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-
Bryan Laird, Sr. Manager CM Operations
Phone: 703-944-9909
   -+-
Cablemodems are the gateway to the Internet.
The Internet is a gateway to some things that are  better left un-seen.


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[asterisk-users] GUI: Not Found. Move along

[Tim Verscheure]

Hi there,

I just installed the GUI for Asterisk 1.4.4 and correctly set my
settings but when I use my browser to access it, it gives me an error
saying "Not Found. Nothing to see here, move along" with "asterisk" in
the header and footer...

anyone had this problemn before?


greetz
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Re: [asterisk-users] zaptel huge irq problem

[François Delawarde]

Hi,

Why are you so determined to use Asterisk in a VM? You're asking for
trouble. Asterisk belongs on dedicated hardware.
  
I actually want to use Asterisk in a machine HOSTING a VM (that's what I 
implied with the Dom-0 thing I said earlier), sorry for the 
misunderstanding. I agree with you that given the state of advancement 
of just about any 'virtualizer', I would have to be totally stupid to 
try running Asterisk inside a VM. (I also wouldn't have asked here in 
the first place, as I would have been totally certain that problems came 
from the virtualizer itself)


If you feel concerned with my reasons for doing that anyway:

- No one told me that Asterisk belonged on dedicated hardware before 
you, so I didn't know.
- I'm just not very rich and try to integrate some things I need in my 
machine (don't worry, I did not framebuffered or X.orged it yet) because 
I cannot afford to buy another one (yes, even the 200€ one)... The part 
you don't want to know is how many people I had to kill in order to get 
my TDM400 card, until I found out that other cheaper solutions existed. :-)



We're just trying to help -- but if you insist on running Asterisk in a
VM, then you're on your own.
  
And I thank you for that (the helping part), you've found the deep cause 
of all my zaptel problems (Xen), so please don't leave me alone! ;-)


To be a bit more constructive, I'd like to ask you or anyone that dared 
to try using Asterisk on a non-dedicated hardware, specifically those 
that tried on a machine hosting VMs the following:


- If there is no way running Asterisk with Xen, what type of 
'hypervisor' should I use in order not to have problems? KVM?, KQemu?, 
VMWare?
- What type of problems should I expect if I dare to do that? (of 
course, Asterisk will be realtime-niced to make it more important)



Thanks and sorry again for the misunderstandings,
François.
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[asterisk-users] Call waiting / hook flash on ZAP trunk from SIP phone?

[Sean M. Pappalardo]

Hello.

After doing much web searching and searching archives of this mailing 
list, I see that my question has been asked at least 6 separate times 
but no answers have been attached.


In a nutshell, is there a way for a SIP phone to easily hook flash a ZAP 
analog trunk mid-call? (This is important when trying to make use of 
features on a PSTN analog line such as call waiting, call forwarding, 
3-way calling, etc.)


I've seen the *3 trick (mentioned in the comments here 
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Flash ), but I 
don't know in which context to put the extensions.conf stuff:


exten => s,n,Set(DYNAMIC_FEATURES=zapflash)
exten => s,n,Dial(SIP/,15,tw)

...to make it take effect while on a call.

My configuration:

- 1 X100P card with one analog line attached with call waiting, caller 
ID, 3-way calling enabled on the line.

- 1 Snom 300 SIP desk phone

My Zapata.conf:

[trunkgroups]
; define any trunk groups

[channels]
; hardware channels
; default

usecallingpres=yes
usecallerid=yes
cidstart=ring
cidsignalling=bell
callwaiting=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=no
callreturn=no
immediate=no   ; for Caller ID, allows time for telco to send digits
hidecallerid=no
echocancel=yes
echotraining=yes
useincomingcalleridonzaptransfer=yes
context=from-pstn   ; Incoming calls go to [from-pstn]
signalling=fxs_ks   ; Use FXS signalling for an FXO channel
group=0 ; Use with Zap/g0
channel => 1; PSTN attached to port 1


Thank you very much for your time. Hopefully we can get an answer to 
this and put it on the Wiki for all to see!


Sincerely,
Sean M. Pappalardo

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[asterisk-users] FastAGI hangs up channel if server is not available

[Lee Jenkins]

Hi all,

Running 1.2.14

When I call a FastAGI script such as this script for an incoming call:

[calldirect]
exten=>s,1,Answer()
exten=>s,2,AGI(agi://192.168.1.175/calldirect?check&${CALLERID(num)})
exten=>s,3,Goto(check_time,s,1)

and the FastAGI server is not running (Asterisk gets "connection 
refused" TCP error), Asterisk just terminates the call like so:


May 17 12:58:00 WARNING[10154]: res_agi.c:210 launch_netscript: Connect 
to 'agi://192.168.1.175/calldirect?check&NN' failed: Connection 
refused
  == Spawn extension (remove_caller, s, 2) exited non-zero on 
'SIP/datatrak-0978b670'


The text of the relevant wiki article 
(http://www.voip-info.org/wiki-Asterisk+FastAGI) infers that execution 
should roll through to the next executable line in the context where it 
was called in the case of an error in executing the FastAGI:


"Asterisk 1.2
Under Asterisk 1.2, if a request to a FastAGI service failed for any 
reason, there was no way to determine this from the dialplan. The 
recommended action is to set a channel variable before calling the 
FastAGI, setting that variable to a known value within the FastAGI, and 
then checking that variable once the AGI has returned..."



Is there a setting/var that I am missing?

Thanks for taking the time.

--

Warm Regards,

Lee



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Re: [asterisk-users] asterisk setup for church / conference call / speaker system integration

[Steve Prior]

Tim Litwiller wrote:
Oh and we will also want to record the services so that if someone wants 
a copy or to listen later they can call in to listen or we can burn then 
a copy. So I'll need to program a recording menu system that is somewhat 
automated and lists the last few weeks of services by date or something.


This part sounds like a podcast to me.

Steve
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[asterisk-users] Busy tone with different length tone

[alaa fahham]

Hi,
I have a  problem with busy tone detection.
the problem is busy tone with different length tone and silence! Means:
Busy tone = 400/400,0/345,400/230,0/520 
400 on
345 off
230 on
520 off
Repeat 
I try in Zapata.conf to enable busy tone detection by this way
 
busydetect=yes
callprogress=no
busycount=3
busypattern=400,345
But the problem busypattern take only one on and one off. But I have now two on 
and two off.
So what I can do?
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[asterisk-users] Re: asterisk setup for church / conference call

[David Cook]

Quoting Tim Litwiller

> to connect to the speaker system I either need to trigger a ring on a
> analog line to the phone interface on our speaker system, it picks up
> on
> the first ring, or we can manully push a button that picks up the
> line.
> If we do the second we would have to have something in asterisk
> connect
> it to the conference when it picks up.

We just put a softphone on the PC that runs Easy Worship and plugged the
soundcard output into a mixer channel and the input to an Aux out.

Recording is fairly painless, just use mix monitor appl as part of the
dial plan for the secret extension that launches the conference from
the PC.

This is assuming you have a manned sound reinforcement system. If your
services are more low-key then the pastor will have to do it himself
before the service starts.

FYI: Churches are the _perfect_ example of a distributed business
environment where Asterisk shines. What other company do you know of
that has such a number of workers who don't have an office in the
building?

Our church is currently moving - possession June 30 - and we will have
deskphones for offices, deskphones at home for key ministry leaders,
softphones for minor ministry leaders and/or phantom VM's with email
attachments.

Don't for get aliases to job functions! People in ministry and volunteer
can come/go. Make sure your IVR has name and functions. Somebody will
call looking for "Children's Ministry", or for the "Pastor" without
knowing who that person would be.

dbc.
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Re: [asterisk-users] CDR is not written

[Steve Murphy]

On Thu, 2007-05-17 at 10:54 -0700, Khaled Chehab wrote:
> I created Master.csv in /var/log/asterisk/cdr-csv ,even did not work,do
> freepbx make this problem,and how can I trouble shoot it.
> 
> 

To get the CSV backend to pump CDRs into
the /var/log/asterisk/cdr-csv/Master.csv file, you need to:

a. in cdr.conf, in the general section, the enable=yes can optionally be
there.
   just make sure it does ***NOT*** say enable=no
b. in cdr.conf, the [csv] section must be there, and not commented out.

I don't know what freepbx does to cdr.conf, but that file is a good
place to start checking.

BTW, doesn't freepbx have a mailing list? Wouldn't it be better asking
there?

> Thanks
> 
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Ken Williams
> Sent: Wednesday, May 16, 2007 11:58 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [SPAM]RE: [asterisk-users] CDR is not written 
> 
> I fought this for a bit when I found if the file Master.csv didn't exist, it
> wouldn't create it on it's own.  I created an empty csv file, CDR started
> writing.
>  
> Ken
> 
> 
> 
> From: [EMAIL PROTECTED] on behalf of Khaled Chehab
> Sent: Thu 5/17/2007 10:50 AM
> To: [EMAIL PROTECTED]
> Cc: asterisk-users@lists.digium.com
> Subject: [asterisk-users] CDR is not written 
> 
> 
> 
> I installed asterisk 1.4.4 final ,but the cdr is not written any patch or
> tweaking can be done 
> 
>  
> 
> 
> 
> Regards
> 
>  
> 
> 
> 
> 
> 
> *
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> sender by electronic mail. You must not copy this message or attachment or 
> disclose its content to any other person.
> 
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> of its attachments, or that they are free from computer viruses or other 
> defects.
> *

Re: [asterisk-users] asterisk setup for church / conference call / speaker system integration

[Drew Gibson]

Tim Litwiller wrote:

David Gomillion wrote:
On 5/17/07, *Tim Litwiller* <[EMAIL PROTECTED] 
> wrote:


We have several people in our church that recently became
disabled. I am
thinking of setting up an asterisk server and several phone lined so
that they can call in to church during services to listen to the
service.

If it were me, I would:

1. create a conference room
2. create a .call file that dials into the speaker system
3. create .call files to dial the participants, muting them

This can obviously be scripted very easily. A simple cron job copying 
the .call files should do nicely. If the disabled persons wish to no 
longer participate, you can simply delete that .call file; 
conversely, if you need to add someone, you can just create another 
one. Then you won't have to worry about incoming phone numbers and 
coordinating with the private school, you can bring it in when you 
need it, etc.


That does simplify it a bit - but is probably not flexible enough. I 
don't want to have to have someone call and request to be added to the 
conference if they are sick sunday morning. We will publish the number 
for anyone in the congregation to use as they need it.  For those that 
always listen at home tho this would work fine.


Oh and we will also want to record the services so that if someone 
wants a copy or to listen later they can call in to listen or we can 
burn then a copy. So I'll need to program a recording menu system that 
is somewhat automated and lists the last few weeks of services by date 
or something.


We are currently using freeconferencecall.com but it is a long 
distance call from church and from all of our members.


Use GotoIfTime to divert incoming calls, only during regular service 
times, to a voice menu with a choice of listen live, or make a call to 
extension.
Use David's call file suggestion to connect the speakers to the 
conference and record (Monitor) that channel.
Have an extension that callers can dial at anytime for a voice menu with 
a choice of recordings.


regards,

Drew

--
Drew Gibson

Systems Administrator
OANDA Corporation
416-593-6767 x322
www.oanda.com

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RE: [asterisk-users] how to define a key to decline incoming call

[Steve Langstaff]

How are you proposing to seize the line to dial the feature code without
answering the call?

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> [EMAIL PROTECTED]
> Sent: 17 May 2007 09:41
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] how to define a key to decline incoming call
> 
> Hi all.
> We have Snom phones which do have a defined key in order to 
> drop incoming call WITHOUT answering.
> 
> Pressing that key, a "SIP/2.0 486 Busy Here" message is sent back.
> 
> We have other phones (I.E. DECT Siemens C450IP, or ATCOM 320 
> or other) which DO NOT have any key to do that (or the key 
> does not work, as is with Siemens C450 IP ): you have to 
> answer and immediatly after hangup the call.
> 
> Acting on feature.conf we succed in defining keys for blind 
> transfer or attended transfer: the last thing we need is the 
> ability to drop an incoming  call without answering it. Is 
> there any way to define a key (or double-key, i.e. "*4") to 
> send back a  "SIP/2.0 486 Busy Here" message ?
> 
> thanks in advance,
> 
> Andrea
> 
> Chi ricevesse questa mail per errore e' gentilmente pregato 
> di cancellarla.
> 
> Visitate il sito http://www.frameweb.it
> 
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[asterisk-users] DUNDi configuration problem

[Tim Verscheure]

Hi peeps,

I've been struggling with DUNDi for a few days now and I can't seem to
make call from Asterisk A to Asterisk B. If I do a "dundi show peers",
it finds the other peer but I can't seem to make any calls. Can
anybody help me out here.

Here's the situation:

Machine 1: Debian with Asterisk 1.4.4 --> 192.168.1.103
Machine 2: AsteriskNOW --> 192.168.1.69

The machines are like you can see on the same subnetwork! I'm using
SIP and for the configurations I based myself on
http://www.voip-info.org/wiki/view/DUNDi+Enterprise+Configuration+SIP,
but no luck!

here are my config files:

=
== MACHINE 1: 192.168.1.103 ==
=

DUNDI.CONF
---
[general]
bindaddr=0.0.0.0
port=4520
entityid=00:0E:A6:0B:E7:50
cachetime=60; 1 minute
ttl=5
autokill=yes
;secretpath=dundi
storehistory=yes

[mappings]
priv => dundi-priv-canonical,0,SIP,${IPADDR}/${NUMBER},nopartial
priv => dundi-priv-customers,100,SIP,${IPADDR}/${NUMBER},nopartial
priv => dundi-priv-via-pstn,400,SIP,${IPADDR}/${NUMBER},nopartial

; AsteriskNOW server
[00:07:95:BD:F1:FF]
model = symmetric
host = 192.168.1.69
inkey = asnow
outkey = asreg
include = priv  ;all
permit = priv
qualify = yes
order = primary


SIP.CONF

[priv]
type=user
dbsecret=dundi/secret
context=dundi-priv-local

[5000]
type=friend
username=tim
secret=1234
host=dynamic
context=testeke

[5001]
type=friend
username=greg
secret=1234
host=dynamic
context=testeke


EXTENSIONS.CONF

[general]
static=yes
writeprotect=no
;autofallthrough=no
clearglobalvars=no
;priorityjumping=yes
;userscontext=default

[dundi-e164-canonical]
;
; List canonical entries here
;
;exten => 12564286000,1,Macro(stdexten,6000,IAX2/foo)
;exten => _125642860XX,1,Dial(IAX2/otherbox/${EXTEN:7})

[dundi-e164-customers]
;
; If you are an ITSP or Reseller, list your customers here.
;
;exten => _12564286000,1,Dial(SIP/customer1)
;exten => _12564286001,1,Dial(IAX2/customer2)

[dundi-e164-via-pstn]
;
; If you are freely delivering calls to the PSTN, list them here
;
;exten => _1256428,1,Dial(Zap/g2/${EXTEN:7}) ; Expose all of 256-428
;exten => _1256325,1,Dial(Zap/g2/${EXTEN:7}) ; Ditto for 256-325

[dundi-e164-local]
;
; Context to put your dundi IAX2 or SIP user in for
; full access
;
include => dundi-e164-canonical
include => dundi-e164-customers
include => dundi-e164-via-pstn

[dundi-e164-switch]
;
; Just a wrapper for the switch
;
switch => DUNDi/e164

[dundi-e164-lookup]
;
; Locally to lookup, try looking for a local E.164 solution
; then try DUNDi if we don't have one.
;
include => dundi-e164-local
include => dundi-e164-switch
;
; DUNDi can also be implemented as a Macro instead of using
; the Local channel driver.
;
[macro-dundi-e164]
;
; ARG1 is the extension to Dial
;
; Extension "s" is not a wildcard extension that matches "anything".
; In macros, it is the start extension. In most other cases,
; you have to goto "s" to execute that extension.
;
; For wildcard matches, see above - all pattern matches start with
; an underscore.
exten => s,1,Goto(${ARG1},1)
include => dundi-e164-lookup

;DUNDI CONFIG;;

; Private DUNDi network
[dundi-priv-canonical]
; Direct numbers (dundi priority 0)

exten => 5000,1,Dial(SIP/5000)
exten => 5001,1,Dial(SIP/5001)

[dundi-priv-customers]
; If you are an ITSP or Reseller, list your customers here.  (dundi
priority 100)

[dundi-priv-via-pstn]
; If you are freely delivering calls to the PSTN, list them here
(dundi priority 400)

[dundi-priv-local]
include => dundi-priv-canonical
include => dundi-priv-customers
include => dundi-priv-via-pstn

[dundi-priv-switch]
switch => DUNDi/priv

[dundi-priv-lookup]
include => dundi-priv-local
include => dundi-priv-switch

[macro-dundi-priv]
exten => s,1,Goto(${ARG1},1)
include => dundi-priv-lookup
DUNDI CONFIG;

; Dundi
exten => _60XX,1,Macro(dundi-priv,${EXTEN})

[testeke]
exten => 5000,1,Dial(SIP/5000)
exten => 5001,1,Dial(SIP/5001)



=
== MACHINE 2: 192.168.1.69  ==
=

DUNDI.CONF
-
[general]
bindaddr=0.0.0.0
port=4520
entityid=00:07:95:BD:F1:FF
cachetime=60
ttl=5
autokill=yes
;secretpath=dundi
storehistory=yes

[mappings]
priv => dundi-priv-canonical,0,SIP,${IPADDR}/${NUMBER},nopartial
priv => dundi-priv-customers,100,SIP,${IPADDR}/${NUMBER},nopartial
priv => dundi-priv-via-pstn,400,SIP,${IPADDR}/${NUMBER},nopartial

; Asterisk Server (laptop)
[00:0E:A6:0B:E7:50]
model = symmetric
host = 192.168.1.103
inkey = asreg
outkey = asnow
include = priv  ;all
permit = priv
qualify = yes
order = primary


SIP.CONF

[priv]
type=user
dbsecret=dundi/secret
context=dundi-priv-local


USERS.CONF
---
[6000]
callwaiting = yes
cid_number = 6000
context =
email = [EMAIL PROTECTED]
fullname = Tim
group =
hasagent = yes
hasdirectory = yes
hasiax = no
hasmanager = no
hassip = ye

Re: [asterisk-users] zaptel huge irq problem

[Stephen Bosch]

François Delawarde wrote:
> I don't really know of other virtualization technology other than Xen,
> and I thank you for guiding me through this, but I have a few doubts
> related to the choice of a virtualization technology in a host with
> Asterisk:
> 
> - Isn't the fact that KVM is now included in the mainstream Linux kernel
> as of 2.6.20 a certain type of 'proof' that it could be stable enough
> compared to others (of course there could be licensing or other
> political/friendship issues)?

It is anything but proof. Kernel inclusions are often a matter of
convenience.

> - Even if the virtual guests aren't totally stable and 100% reliable
> yet, wouldn't the use of KVM be better with Zaptel compatible cards than
> Xen, in architecture point of vue, as it is only a kernel module that
> -as far as I know- don't appear to be changing fundamental issues like
> IRQ handling or I/O scheduling in the kernel, and from the fact that
> virtual machines are treated like simple processes?

Why are you so determined to use Asterisk in a VM? You're asking for
trouble. Asterisk belongs on dedicated hardware.

We're just trying to help -- but if you insist on running Asterisk in a
VM, then you're on your own.

That's not a risk I'd want to take.

-Stephen-
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Re: [asterisk-users] CDR changes in 1.4.3?

[Steve Murphy]

- "Andreas van dem Helge" <[EMAIL PROTECTED]> wrote:
> On 4/27/07, Steve Murphy <[EMAIL PROTECTED]> wrote:
> > I'm the guilty party. I've been trying to fix several CDR bugs,
> > involving stuff like missing times, missing changes in state (like
> > NO_ANSWER when the call was ANSWERED), etc.
> 
> Now that we are talking about CDRs, I must ask: in 1.2.x if the CDR
> is
> forked into two the uniqueid is the same for both CDR records. Is
> that
> the intended behavior? Does that remain in 1.4.x?
> 

It should be the same. The uniqueid is a channel attribute, and when the CDR is 
initialized,
that value is copied from the channel. That hasn't changed from 1.2 to 1.4, nor 
have I mucked
with it.

> > CDR's are complicated by the fact that they record 3 events:
> "start",
> > "Answer", and "end" events. Add to that the fact that in most cases
> at
> > least two channels are involved, sometimes 4 or 5, or even more,
> > involving bridging, maquerading, parking, transfers, local
> channels,
> > AGI, conferences, and more...
> >
> > Some cases were impossible to fix unless CDR's were attached to
> every
> > channel,
> > and merged to collect the bits and pieces that sometimes were on
> the
> > wrong side of the bridge.
> 
> It would be nice if the CDR engine could be configured to allow for
> these transactions either to be merged or not and what to do with the
> bits and pieces as you describe them. However it would seem logical
> that if various pieces are merged then ultimately they should not be
> logged as that would be redundant... However I'd rather see it be a
> configuration choice.
> 

I'll do what I can... but I don't see merging entries to be an option in the 
near future.
I think I'll be lucky to be able to give you accurate info that you can merge 
at the backend...!

> > The result is that several more cases are more accurate, but also,
> that
> > rather uninteresting CDR's can be generated. In contemplating what
> could
> > be done to get rid of some of these, I sometimes have to ask, "is
> this
> > truly something we have to get rid of?"... In the meantime,
> > uninteresting CDR's with NO_ANSWER and billsec=0, should be easy to
> > filter out, right?
> 
> I don't think CDRs with NO ANSWER disposition or billsec=0 should be
> discarded. Why not make it configurable?

Now, this may be option. 

> 
> > I will, in the coming days, look at some of the extraneous CDR's
> that
> > are generated, and see what I can do to get rid of them. It's not
> always
> > that simple.
> > If we ring a phone, for instance, and no-one answers it, is that
> truly,
> > really, something that no-one will ever be, could ever be,
> interested
> > in? (just a fer-instance).
> 
> 
> From a billing standpoint no whats the point? For statistical
> purposes
> I think its useful. For VoIP serviceprovider also very useful
> customer
> probably wants full call logs. I don't think your idea is too much
> CALEA-compliant either.

Viva la CALEA! OK. Forget the filtering. The Gov shall have everything.


> 
> > I welcome your input. Complain up a storm. I'll try my best to make
> you
> > happy.
> >
> 
> Make it configurable.

I'll try!


-- 
Steve Murphy
Software Developer
Digium

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Re: [asterisk-users] Anyone Installed a Digium TE110P or TE120P card in Canada?

[Stephen Bosch]

Klaverstyn, David C wrote:
> The Telco in Canada is been real painful.  I was wondering if anyone has
> installed a Digium TE1X0P card in Canada and if their Telco was so
> difficult.

Who is the telco? Where?

> The Telco will not provide us a service until they see a FCC or DOC
> number for the equipment ware are connecting to their service.
> 
> If have found “FCC Part 68, ANSI/ITA-968-A, Including Amendment A1 and
> A2 Industry Canada CS-03” thanks to Nabeel a forum users.
> 
> I now need to know if the statement above is what I need to tell the Telco.

No, that's the relevant regulation. The card has a registration number
with Industry Canada. That's what they want.
> 
> As I am not in Canada this make it a bit difficult for me.

The number is on a label on the card. You will need a person on-site to
get it for you.

-Stephen-

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RE: [asterisk-users] Asterisk Queue MOH

[asterisk-users]

>From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of TienSen Chong
>Sent: 17. maí 2007 10:51
>To: Asterisk Users Mailing List - Non-Commercial Discussion
>Subject: [asterisk-users] Asterisk Queue MOH
>Is there any way if i want the caller to hear dial tone rather than the
MOH?

Perhaps you could use something like

Queue(yourqueuename|rt|||60);

in extensions.conf or extension.ael? The r is defined as "ring instead of
playing MOH".

Baldvin.



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Re: [asterisk-users] Quadbri Cellular Issue

[Tzafrir Cohen]

On Thu, May 17, 2007 at 11:47:25AM +0200, Antonio Martínez Contreras wrote:
> Hello everybody, and first of all sorry for my poor English.
> 
> I'm having trouble with Quadbri installed on Asterisk 
> 1.2.17-BRIstuffed-0.3.0-PRE-1y-e. Everything is working fine, except 
> calling to switched off or "out of coverage" cell phones. In this case I 
> have to wait 40 seconds until Asterisk tell me that "all circuits are busy 
> now" instead of receive cell phone company message of "The cell phone you 
> are calling is unavailable". We tried a lot of configurations in 
> zapata.conf and zaptel.conf, but we still have the same problem. I show my 
> config:
> 
> zapata.conf:
> 
> =
> ;
> ; Zapata telephony interface
> ;
> ; Configuration file
> 
> [trunkgroups]
> 
> [channels]
> 
> language=es
> 
> usecallerid=yes
> hidecallerid=no
> callwaiting=yes
> usecallingpres=yes
> callwaitingcallerid=yes
> threewaycalling=yes
> transfer=yes
> cancallforward=yes
> callreturn=yes
> group=1
> echocancel=yes
> echocancelwhenbridged=yes
> echotraining=800
> immediate=no
> 
> switchtype=euroisdn
> signalling=bri_cpe_ptmp
> pridialplan=unknown
> prilocaldialplan=unknown
> priindication=outofband
> facilityenable=yes
> relaxdtmf=yes
> context=from-zaptel
> rxgain=0.0
> txgain=0.0
> channel => 1,2
> 
> faxdetect=incoming
> 
> ;Include genzaptelconf configs
> #include zapata-auto.conf
> 
> #group=1
> 
> ;Include AMP configs
> #include zapata_additional.conf

Could you please provide those two files (zapata_additional.conf and 
zapata-auto.conf )?

> 
> 
> 
> ;Include BRI-HFC configs
> #include zapata-BRI-HFC.conf
> 
> ===
> 
> zaptel.conf
> 
> ===
> 
> # Autogenerated by /usr/local/sbin/genzaptelconf -- do not hand edit
> # Zaptel Configuration File
> #
> # This file is parsed by the Zaptel Configurator, ztcfg
> #
> 
> # It must be in the module loading order
> 
> 
> # Global data
> 
> loadzone = es
> defaultzone = es
> 
> # HFC CARD NOTE:
> # When run for the first time the card numbers start from 1
> # When run subsequently the card numbers start from 0
> # Go figure!
> 
> 
> # Span 1: ztqoz/2/1 "quadBRI PCI ISDN Card 1 Span 1 [TE]"
> span=1,1,3,ccs,ami
> bchan=1-2
> dchan=3
> 
> 
> # Span 2: ztqoz/2/2 "quadBRI PCI ISDN Card 1 Span 2 [TE]"
> #span=2,0,3,ccs,ami
> #bchan=4-5
> #dchan=6
> 
> 
> # Span 3: ztqoz/2/3 "quadBRI PCI ISDN Card 1 Span 3 [TE]"
> #span=3,0,3,ccs,ami
> #bchan=7-8
> #dchan=9
> 
> 
> # Span 4: ztqoz/2/4 "quadBRI PCI ISDN Card 1 Span 4 [TE]"
> #span=4,0,3,ccs,ami
> #bchan=10-11
> #dchan=12
> 
> ===
> 
> In channel debug I have:
> 
> Debugging on new channels is enabled
> 
>-- Executing Macro("SIP/200-09fc1698", "dialout-trunk|2|637574972||") in 
> new stack
> 
>-- Executing Set("SIP/200-09fc1698", "DIAL_TRUNK=2") in new stack
> 
>-- Executing Set("SIP/200-09fc1698", "_NODEST=") in new stack
> 
>-- Executing Set("SIP/200-09fc1698", "DIAL_NUMBER=637574972") in new 
> stack
> 
>-- Executing Set("SIP/200-09fc1698", "ROUTE_PASSWD=") in new stack
> 
>-- Executing Set("SIP/200-09fc1698", "DIAL_TRUNK_OPTIONS=tr") in new 
> stack
> 
>-- Executing GotoIf("SIP/200-09fc1698", "1?noauth") in new stack
> 
>-- Goto (macro-dialout-trunk,s,8)
> 
>-- Executing Set("SIP/200-09fc1698", "GROUP()=OUT_2") in new stack
> 
>-- Executing Macro("SIP/200-09fc1698", "user-callerid|SKIPTTL") in new 
> stack
> 
>-- Executing NoOp("SIP/200-09fc1698", "user-callerid: device 200") in 
> new stack
> 
>-- Executing GotoIf("SIP/200-09fc1698", "0?report") in new stack
> 
>-- Executing GotoIf("SIP/200-09fc1698", "0?start") in new stack
> 
>-- Executing Set("SIP/200-09fc1698", "REALCALLERIDNUM=200") in new stack
> 
>-- Executing NoOp("SIP/200-09fc1698", "REALCALLERIDNUM is 200") in new 
> stack
> 
>-- Executing Set("SIP/200-09fc1698", "AMPUSER=200") in new stack
> 
>-- Executing Set("SIP/200-09fc1698", "AMPUSERCIDNAME=200") in new stack
> 
>-- Executing GotoIf("SIP/200-09fc1

Re: [asterisk-users] asterisk setup for church / conference call / speaker system integration

[Tim Litwiller]

David Gomillion wrote:
On 5/17/07, *Tim Litwiller* <[EMAIL PROTECTED] 
> wrote:


We have several people in our church that recently became
disabled. I am
thinking of setting up an asterisk server and several phone lined so
that they can call in to church during services to listen to the
service. 



If it were me, I would:

1. create a conference room
2. create a .call file that dials into the speaker system
3. create .call files to dial the participants, muting them

This can obviously be scripted very easily. A simple cron job copying 
the .call files should do nicely. If the disabled persons wish to no 
longer participate, you can simply delete that .call file; conversely, 
if you need to add someone, you can just create another one. Then you 
won't have to worry about incoming phone numbers and coordinating with 
the private school, you can bring it in when you need it, etc. 



That does simplify it a bit - but is probably not flexible enough. I 
don't want to have to have someone call and request to be added to the 
conference if they are sick sunday morning. We will publish the number 
for anyone in the congregation to use as they need it.  For those that 
always listen at home tho this would work fine.


Oh and we will also want to record the services so that if someone wants 
a copy or to listen later they can call in to listen or we can burn then 
a copy. So I'll need to program a recording menu system that is somewhat 
automated and lists the last few weeks of services by date or something.


We are currently using freeconferencecall.com but it is a long distance 
call from church and from all of our members. 



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Re: [asterisk-users] how to define a key to decline incoming call

[Alex Balashov]

On Thu, 17 May 2007, [EMAIL PROTECTED] said something to this effect:

Is there any way to define a key (or double-key, i.e. "*4") to send back 
a "SIP/2.0 486 Busy Here" message ?


  Can't you do something like Hangup() or direct to voicemail from the 
features.conf directives?


  Never actually tried, so I could be blowing smoke here...

-- Alex

--
Alex Balashov   <[EMAIL PROTECTED]>
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[asterisk-users] emergency call called party control

[Ray Chen]

Hi, For emergency calls only the callee, the emergency dispatch center
operator, can hand-up the call. If the caller hand-ups the call the
system needs to maintain the voice path and whenever the caller picks up
the phone the call should be resumed. Any body has an idea how to do
this? Thanks,
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RE: [asterisk-users] cpu usage for G.729 codec

[Race Vanderdecken]

 
G.729 is a compromise of bandwidth vs. CPU power. It takes more CPU but
less bandwidth.
 
It depends on what your want to do with the G.729. 
 
Pass through does not involve any transcoding, that I know of, so it is
just an RTP packet movement, no different than the cost of other pass
through codecs.
 
I did work on converting G.729 to G.711 to disk storage in real time and
that took about 3% of a Xeon CPU for full duplex.
 
Memory wise each convert call might have used 640KB in buffers and
trash, but not much.
 
 
 
Race Vanderdecken
Code Tyrant
 
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Khaled
Chehab
Sent: Tuesday, May 15, 2007 2:51 PM
To: [EMAIL PROTECTED]
Cc: asterisk-users@lists.digium.com
Subject: [asterisk-users] cpu usuage
 
 
Do any one knows the formula to  calculate memory and cpu usuage for
channel on g729 codec,to know the hardware required for 100 concurrent
call.
 
 
 
Regards
 
 
  _  

*
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Re: [asterisk-users] Microsoft's Move Into IP PBX Market

[Jessee J Holmes]

I want the "hot-dog stand" theme on my phone :)

On May 17, 2007, at 10:03 AM, Drew Gibson wrote:


George Pajari wrote:

From c|net News:
"On Monday,Microsoft and nine leading phone manufacturers--Asustek  
Computer, GN, LG-Nortel, NEC, Plantronics, Plycom, Samsung,  
Tatung, and Vitelix--announced the public beta program for  
Microsoft Office Communications Server 2007 and Microsoft Office  
Communicator 2007."


http://news.com.com/8301-10784_3-9719931-7.html? 
part=rss&subj=news&tag=2547-1_3-0-20



I'm told the main hardware requirements are
1.  Colour screens (well, _blue_ anyway)
2.  Support for the prompt "Error bridging call, (A)bort, (R)etry,  
(F)ail? _"


regards,

Drew

--
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com

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Re: [asterisk-users] Microsoft's Move Into IP PBX Market

[Drew Gibson]

George Pajari wrote:

From c|net News:
"On Monday,Microsoft and nine leading phone manufacturers--Asustek 
Computer, GN, LG-Nortel, NEC, Plantronics, Plycom, Samsung, Tatung, 
and Vitelix--announced the public beta program for Microsoft Office 
Communications Server 2007 and Microsoft Office Communicator 2007."


http://news.com.com/8301-10784_3-9719931-7.html?part=rss&subj=news&tag=2547-1_3-0-20 




I'm told the main hardware requirements are
1.  Colour screens (well, _blue_ anyway)
2.  Support for the prompt "Error bridging call, (A)bort, (R)etry, 
(F)ail? _"


regards,

Drew

--
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com

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Re: [asterisk-users] Outside lines are *STILL* just not happening...

[Noah Miller]

Hi David -


Sometimes (about half the time) the phone I'm calling (in this case, a
cell) will give part of one ring, then report a missed call.  The SIP
phone hangs up after about 5 seconds.  But not always.  The rest of the
time, the SIP phone just eventually (15 or 20 secs) hangs up on its'
own, and the cell never reports a missed call.


Have you tried using a regular analog phone on the PSTN lines (without
using asterisk at all)?  If you're getting partial connections that
are failing, it could be a line quality issue.  For testing, you might
also try turning off as many options as possible in zapata.conf
(particularly the echo cancel).


- Noah
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Re: [asterisk-users] dundi problem * 1.4.2

[Tim Verscheure]

I assume you use this configuration for interconnecting 2 asterisk
machines. Just IP based...?

greetz, Tim

2007/4/25, Asterisk [Submusic] <[EMAIL PROTECTED]>:

Hi,

My configuration:

SERVER 1: 192.168.1.1 => submusic
SERVER 2: 192.168.1.2 => vns

SERVER 1: Extension 32XX
SERVER 2: Extension 31XX

If you want, I can explain off list for more informations or Dundi concept

Tell me if you understand my configuration.

Fred


; DUNDI.conf SERVER 1 (Submusic)


[general]

bindaddr=0.0.0.0
port=4520

entityid=00:04:76:DB:54:7F

cachetime=1200

ttl=32
autokill=yes
storehistory=yes

[mappings]

asterisk-france =>
dundi-priv-canonical,0,IAX,asterisk-france:[EMAIL PROTECTED]/${NUMBER},n
opartial

; VNS
[00:00:F8:04:C4:51]
model = symmetric
host = 192.168.1.2
inkey = vns
include = all
outkey = submusic
permit = asterisk-france
qualify = 3000
order= primary



; DUNDI.conf SERVER 2 (VNS)


[general]

bindaddr=0.0.0.0
port=4520

entityid=00:00:F8:04:C4:51

cachetime=1200
ttl=32
autokill=yes
storehistory=yes

[mappings]

asterisk-france =>
dundi-priv-canonical,0,IAX,asterisk-france:[EMAIL PROTECTED]/${NUMBER},n
opartial

; SUBMUSIC
[00:04:76:DB:54:7F]
model = symmetric
host = 192.168.1.1
inkey = submusic
include = all
outkey = vns
permit = asterisk-france
qualify = yes
order= primary


; IAX.conf (Same for both)


[asterisk-france]
type=user
dbsecret=dundi/secret
context=dundi-priv-local



=
; Extension.conf Server 1 (Submusic)
=


; This macro is used to do the lookup and the match to the other host over
the Dundi Network

[macro-dundi-priv]
exten => s,1,Goto(${ARG1},1)
switch => DUNDi/asterisk-France


; This Context is where the Lookup function is looking for extension
matching, just put the priority 1 and a NoOP
This server is just responding for 3 Extension over the Dundi Network

[dundi-priv-canonical]
exten => 3202,1,NooP(DUNDI LOOKUP 3202)
exten => 3216,1,NooP(DUNDI LOOKUP 3216)
exten => 3220,1,NooP(DUNDI LOOKUP 3220)

; This context is used to receipt the IAX Call, it must match with the
iax.conf.

[dundi-priv-local]
exten => 3202,1,Dial(SIP/3202)
exten => 3216,1,Dial(SIP/3216)
exten => 3220,1,Dial(SIP/3220)


; This Extension is used for the lookup and the dial over the Dundi Network.
; You must put it in the context that allow tu dial over the Dundi Network

exten => _31XX,1,Macro(dundi-priv,${EXTEN})  ; VNS

=
; Extension.conf Server 2 (VNS)
=


; This macro is used to do the lookup and the match to the other host over
the Dundi Network

[macro-dundi-priv]
exten => s,1,Goto(${ARG1},1)
switch => DUNDi/asterisk-France


; This Context is where the Lookup function is looking for extension
matching, just put the priority 1 and a NoOP
This server is just responding for 3 Extension over the Dundi Network

[dundi-priv-canonical]
exten => 3101,1,NOOP(DUNDI)
exten => 3102,1,NOOP(DUNDI)
exten => 3103,1,NOOP(DUNDI)

; This context is used to receipt the IAX Call, it must match with the
iax.conf.

[dundi-priv-local]
; Direct numbers (dundi priority 0)
include => VNS

exten => 3101,1,Dial(SIP/3101)
exten => 3102,1,Dial(SIP/3102)
exten => 3103,1,Dial(SIP/3103)


===
End


-Message d'origine-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Remco Post
Envoyé: mercredi, 25. avril 2007 00:26
À: Asterisk Users Mailing List - Non-Commercial Discussion
Objet: Re: [asterisk-users] dundi problem * 1.4.2

Asterisk [Submusic] wrote:
> Hi,
>
> I'm not working with Asterisk 1.4.X, but i think your Dundi.conf is not
> correct.
>

well, things haven't changed in the dundi.conf going from 1.2 to 1.4, so
that should be ok.

> If you want i can send you my complete working exemple with Asterisk 1.2.x
> (I think the config is the same)
>

Please do. I've had a friend look at my dundi.conf, he couldn't find
anything wrong with it, but it is quite likely that there is.

> Fred
>
>
>
>


--

Remco Post

"I didn't write all this code, and I can't even pretend that all of it
makes sense." -- Glen Hattrup
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Re: [asterisk-users] asterisk setup for church / conference call / speaker system integration

[David Gomillion]

On 5/17/07, Tim Litwiller <[EMAIL PROTECTED]> wrote:


We have several people in our church that recently became disabled. I am
thinking of setting up an asterisk server and several phone lined so
that they can call in to church during services to listen to the service.



If it were me, I would:

1. create a conference room
2. create a .call file that dials into the speaker system
3. create .call files to dial the participants, muting them

This can obviously be scripted very easily. A simple cron job copying the
.call files should do nicely. If the disabled persons wish to no longer
participate, you can simply delete that .call file; conversely, if you need
to add someone, you can just create another one. Then you won't have to
worry about incoming phone numbers and coordinating with the private school,
you can bring it in when you need it, etc.
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[asterisk-users] asterisk setup for church / conference call / speaker system integration

[Tim Litwiller]

We have several people in our church that recently became disabled. I am 
thinking of setting up an asterisk server and several phone lined so 
that they can call in to church during services to listen to the service.


The phone lines at church are also used by our private school during the 
week.  So I am thinking that I need to setup a schedule so that the 
lines used for the service will automatically go to the conference 
during normal service times and ring the phones the rest of the time.


We have 2 analog lines currently,  DSL is available, and I'm checking 
now to see if we can get a voip provider that will provide 3 - 10 local 
numbers.


to connect to the speaker system I either need to trigger a ring on a 
analog line to the phone interface on our speaker system, it picks up on 
the first ring, or we can manully push a button that picks up the line. 
If we do the second we would have to have something in asterisk connect 
it to the conference when it picks up.


the next consideration is that sometimes we might want to connect out to 
a conference call from another church. so we need to be able to dial out 
from one of the phones and then somehow trigger the speaker systems 
incoming side to pickup. ( I just thought of it now - I suppose a call 
transfer to that line would be all thats needed for that)



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Re: [asterisk-users] Re: MeetMe and ChannelRedirect

[Andrew Furey]

On 17/05/07, Andrew Furey <[EMAIL PROTECTED]> wrote:


[nothing]


Ugh, what happened there? must have clicked the wrong button. Sorry
for the noise folks.

Andrew

--
Linux supports the notion of a command line or a shell for the same
reason that only children read books with only pictures in them.
Language, be it English or something else, is the only tool flexible
enough to accomplish a sufficiently broad range of tasks.
 -- Bill Garrett
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Re: [asterisk-users] Re: MeetMe and ChannelRedirect

[Andrew Furey]

On 17/05/07, Tony Mountifield <[EMAIL PROTECTED]> wrote:

In article <[EMAIL PROTECTED]>,
Rafael Vidal Aroca <[EMAIL PROTECTED]> wrote:
> i'm trying to implement the following scenario:
>
> - A user calls number 700
> - Asterisk then dials to extensions 100, 200, 300, 400 and 500
> - And then bridges all calls to a conference room
>
> I tried to use MeetMe and ChannelRedirect, but seems that after
> channel redirect nothing more is executed. So, this seem to work for the
> caller and first called, but the others stay outside.
>
> Could anyone help or give me a hint?

The way I did this kind of thing was like this:

1. Extension 700 calls an AGI script which generates a .call file in
/var/spool/asterisk/outgoing for each of the calls to the other extensions.
Extension 700 then drops into the Meetme room to wait for the others.

2. Each call file specifies a Local channel to make the call to the
extension,
and uses the Context, Extension and Priority fields to direct the answered
call into the Meetme room.

If any of the calls to the other extensions fails (e.g. busy), you don't get
any notification of that. If you want such notification, you will need to
get a lot more complex, probably involving a controlling process using the
Manager API.

Hope this helps.

Cheers
Tony
--
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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--
Linux supports the notion of a command line or a shell for the same
reason that only children read books with only pictures in them.
Language, be it English or something else, is the only tool flexible
enough to accomplish a sufficiently broad range of tasks.
 -- Bill Garrett
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Re: [asterisk-users] Anyone Installed a Digium TE110P or TE120P card in Canada?

[Jon Pounder]

Quoting "Klaverstyn, David C" <[EMAIL PROTECTED]>:


The Telco in Canada is been real painful.  I was wondering if anyone has
installed a Digium TE1X0P card in Canada and if their Telco was so
difficult.



which telco have you been dealing with ? I have had Telus service over  
Bell Canada lines for T1 and connected them to a linux box with an lmc  
card in it and other than Telus being shocked when their end was  
configured wrong after a long install process, no one really cared  
what I had on my end, other than they were convinced I had things  
setup wrong.


my experience is they come in, slap their pairgain hdsl modem on the  
line, run some tests at the t1 port on it themselves and leave. What  
you plug in after that is up to you, just like a regular phone jack.  
Just don't call them unless the problem is on their side of things or  
you get a bill.










The Telco will not provide us a service until they see a FCC or DOC
number for the equipment ware are connecting to their service.



If have found "FCC Part 68, ANSI/ITA-968-A, Including Amendment A1 and
A2 Industry Canada CS-03" thanks to Nabeel a forum users.



I now need to know if the statement above is what I need to tell the
Telco.



As I am not in Canada this make it a bit difficult for me.



Your help is greatly appreciated.












Jon Pounder

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Tools to Power Your e-Business Solutions
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Re: [asterisk-users] Which KDE editor to edit Asterisk config files ?

[Diego Iastrubni]

Well, this KDE user (if I have a working SVN account do I qualify as a KDE 
developer...?) uses only kate and hates VIM. I 
use "sftp://root:[EMAIL PROTECTED]/etc/asterisk/*.conf" to open files on the 
remote servers. 

Ok, so instead of bitching, I am attaching here my first draft of the XML file 
which represents the kate syntax hightlighing. Save in in 
~/.kde/share/katepart/diego-kicks-ass.xml, and you will see "Asterisk config 
files" under "Configuration".

It's not done yet, as I want the sample configuration file (also attached 
here) to be working as expected. 

Please comment and review.

PS:
Someone said something about bluefish. This is a GtkSourceView syntax 
highlighter which will work on GEdit as well no...? Can you share?

On Wednesday 16 May 2007 19:57, Andrew Kohlsmith wrote:
> On Wednesday 16 May 2007 11:47 am, Olivier wrote:
> > Do you mean nobody has ever done this before (as I thought before asking
> > this question to the list) ?
> > So which tool KDE users are using for this ?
>
> I am a KDE user, although on Slackware.  Have been for many, many years.
>
> Typically you will find that those who wish to use their GUIs to manipulate
> Asterisk will do so through one of the available GUIs.  Those who want to
> work on the text files will use vim or emacs.
>
> I develop embedded systems; I use kdevelop for coding for the most part,
> and once in a while I'll use Kate to edit config files, but 99% of my time
> manipulating text files is done in vim.
>
> Even as I type this I have kdevelop open for the source and html, but I
> have three konsole tabs open: one to a screen session to a server I IRC
> from, one to a screen session to my development box in the server room
> (which has two login sessions going), one to a telnet session to the board
> I'm developing for, and finally one to a serial port server which the
> serial console of the development box is connected to.
>
> Kate's open, but contains a little textfile I append to which has todo
> lists and notes for the development project.
; comment

; illegal content: not inside a context
include => context1
exten => s,1,Answer

illegal syntax

#include another_file
#include *more_files

[context1] ; a comment
exten => s,1,Answer ; more comments
exten => s,2,Hungup ; and even more

[context2] 
exten ;; should be an error
exten -> ;; should be an error
exten -> s ;; should be an error
exten -> s,n ;; should be an error

[context3] shuold be an error
include => context2 ; ok
include => "context2"   ; ok, but fishie

[context3] shuold be an error ; and this a comment
include => context2 ; ok
include => "context2"   ; ok, but fishie







 exten 
 include 




















































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Re: [asterisk-users] Which KDE editor to edit Asterisk config files ?

[Diego Iastrubni]

Well, this KDE user (if I have a working SVN account do I qualify as a KDE 
developer...?) uses only kate and hates VIM. I 
use "sftp://root:[EMAIL PROTECTED]/etc/asterisk/*.conf" to open files on the 
remote servers. 

Ok, so instead of bitching, I am attaching here my first draft of the XML file 
which represents the kate syntax hightlighing. Save in in 
~/.kde/share/katepart/diego-kicks-ass.xml, and you will see "Asterisk config 
files" under "Configuration".

It's not done yet, as I want the sample configuration file (also attached 
here) to be working as expected. 

Please comment and review.

PS:
Someone said something about bluefish. This is a GtkSourceView syntax 
highlighter which will work on GEdit as well no...? Can you share?

PS2:
The asterisk config file was blocked, how funny... I am attaching here inline:

-
; comment

; illegal content: not inside a context
include => context1
exten => s,1,Answer

illegal syntax

#include another_file
#include *more_files

[context1] ; a comment
exten => s,1,Answer ; more comments
exten => s,2,Hungup ; and even more

[context2] 
exten ;; should be an error
exten -> ;; should be an error
exten -> s ;; should be an error
exten -> s,n ;; should be an error

[context3] shuold be an error
include => context2 ; ok
include => "context2"   ; ok, but fishie

[context3] shuold be an error ; and this a comment
include => context2 ; ok
include => "context2"   ; ok, but fishie
---


On Wednesday 16 May 2007 19:57, Andrew Kohlsmith wrote:
> On Wednesday 16 May 2007 11:47 am, Olivier wrote:
> > Do you mean nobody has ever done this before (as I thought before asking
> > this question to the list) ?
> > So which tool KDE users are using for this ?
>
> I am a KDE user, although on Slackware.  Have been for many, many years.
>
> Typically you will find that those who wish to use their GUIs to manipulate
> Asterisk will do so through one of the available GUIs.  Those who want to
> work on the text files will use vim or emacs.
>
> I develop embedded systems; I use kdevelop for coding for the most part,
> and once in a while I'll use Kate to edit config files, but 99% of my time
> manipulating text files is done in vim.
>
> Even as I type this I have kdevelop open for the source and html, but I
> have three konsole tabs open: one to a screen session to a server I IRC
> from, one to a screen session to my development box in the server room
> (which has two login sessions going), one to a telnet session to the board
> I'm developing for, and finally one to a serial port server which the
> serial console of the development box is connected to.
>
> Kate's open, but contains a little textfile I append to which has todo
> lists and notes for the development project.




	
		
			 exten 
			 include 
		
		
		
			




			
			
			


			
			
			

			
			
			

			
			
			

			
			
			
			
			
			
			
		
		
		
			
			
			
			
			
			
		
	
	
	
		
			
		
		
	


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Re: [asterisk-users] Which KDE editor to edit Asterisk config files ?

[Diego Iastrubni]

Well, this KDE user (if I have a working SVN account do I qualify as a KDE 
developer...?) uses only kate and hates VIM. I 
use "sftp://root:[EMAIL PROTECTED]/etc/asterisk/*.conf" to open files on the 
remote servers. 

Ok, so instead of bitching, I am attaching here my first draft of the XML file 
which represents the kate syntax hightlighing. Save in in 
~/.kde/share/katepart/diego-kicks-ass.xml, and you will see "Asterisk config 
files" under "Configuration".

It's not done yet, as I want the sample configuration file (also attached 
here) to be working as expected. 

Please comment and review.

PS:
Someone said something about bluefish. This is a GtkSourceView syntax 
highlighter which will work on GEdit as well no...? Can you share?

On Wednesday 16 May 2007 19:57, Andrew Kohlsmith wrote:
> On Wednesday 16 May 2007 11:47 am, Olivier wrote:
> > Do you mean nobody has ever done this before (as I thought before asking
> > this question to the list) ?
> > So which tool KDE users are using for this ?
>
> I am a KDE user, although on Slackware.  Have been for many, many years.
>
> Typically you will find that those who wish to use their GUIs to manipulate
> Asterisk will do so through one of the available GUIs.  Those who want to
> work on the text files will use vim or emacs.
>
> I develop embedded systems; I use kdevelop for coding for the most part,
> and once in a while I'll use Kate to edit config files, but 99% of my time
> manipulating text files is done in vim.
>
> Even as I type this I have kdevelop open for the source and html, but I
> have three konsole tabs open: one to a screen session to a server I IRC
> from, one to a screen session to my development box in the server room
> (which has two login sessions going), one to a telnet session to the board
> I'm developing for, and finally one to a serial port server which the
> serial console of the development box is connected to.
>
> Kate's open, but contains a little textfile I append to which has todo
> lists and notes for the development project.




	
		
			 exten 
			 include 
		
		
		
			




			
			
			


			
			
			

			
			
			

			
			
			

			
			
			
			
			
			
			
		
		
		
			
			
			
			
			
			
		
	
	
	
		
			
		
		
	


; comment

; illegal content: not inside a context
include => context1
exten => s,1,Answer

illegal syntax

#include another_file
#include *more_files

[context1] ; a comment
exten => s,1,Answer ; more comments
exten => s,2,Hungup ; and even more

[context2] 
exten ;; should be an error
exten -> ;; should be an error
exten -> s ;; should be an error
exten -> s,n ;; should be an error

[context3] shuold be an error
include => context2	; ok
include => "context2"	; ok, but fishie

[context3] shuold be an error ; and this a comment
include => context2	; ok
include => "context2"	; ok, but fishie

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Re: [asterisk-users] HUDlite Server on Debian Etch/Asterisk 1.4

[Tzafrir Cohen]

On Wed, May 16, 2007 at 03:22:35PM +0200, Jack wrote:
> Hi,
> 
> has anyone managed to get hudlite server working on a Debian Etch
> based installation of Asterisk 1.4?
> 
> So far I managed to eliminate all error messages, but the process is
> killed directly after starting the hudlite server without showing any
> error messages.
> 
> I would be very happy if anyone can give me some hints or point me to
> a installation guide.

What I would do in such a situation, is run everything under strace.

However, recall that you're dealing with a proprietary program here. The 
only ones who have the full information to help you are Fonality.

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
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Re: [asterisk-users] queue_exec: Unable to join queue

[lenz]

I would try one of the two things:
1. adding a "hint" for the Local/[EMAIL PROTECTED] channels
2. using  the => for queue members

member => Agent/1001
member => Agent/1002
member => Agent/1003

Does this change anything?
l.


In data Tue, 15 May 2007 08:25:51 +0200, Per Jessen <[EMAIL PROTECTED]> ha  
scritto:



lenz wrote:


Is the queue "enidan" configured at all in queues.conf? and how is it
defined?
l.


Sorry, I should have added that:

from queues.conf:

[enidan]
strategy = ringall
;announce = enidan-queue
member = Local/[EMAIL PROTECTED]
member = Local/[EMAIL PROTECTED]
member = Local/[EMAIL PROTECTED]
member = Local/[EMAIL PROTECTED]

Also, what I discovered yesterday is the following:

just after an asterisk restart:
*CLI> show queue
enidan   has 0 calls (max unlimited) in 'ringall' strategy (0s
holdtime), W:0, C:0, A:0, SL:0.0% within 0s
   Members:
  Local/[EMAIL PROTECTED] (Invalid) has taken no calls yet
  Local/[EMAIL PROTECTED] (Invalid) has taken no calls yet
  Local/[EMAIL PROTECTED] (Invalid) has taken no calls yet
  Local/[EMAIL PROTECTED] (Invalid) has taken no calls yet
   No Callers

The "(Invalid)" bit is worrying, but after a reload of app_queue:

*CLI> show queue
enidan   has 0 calls (max unlimited) in 'ringall' strategy (0s
holdtime), W:0, C:0, A:0, SL:0.0% within 0s
   Members:
  Local/[EMAIL PROTECTED] (Not in use) has taken no calls yet
  Local/[EMAIL PROTECTED] (Not in use) has taken no calls yet
  Local/[EMAIL PROTECTED] (Not in use) has taken no calls yet
  Local/[EMAIL PROTECTED] (Not in use) has taken no calls yet
   No Callers



/Per Jessen, Zürich





--
Home of QueueMetrics - http://queuemetrics.com

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[asterisk-users] Asterisk Queue MOH

[TienSen Chong]

Hi,

I have a queue with a few members. When i tried to call the number directing
to the queue, one of the member is ringing and the music on hold is being
played at the caller. Is there any way if i want the caller to hear dial
tone rather than the MOH?

Regards,
Chong
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Re: [asterisk-users] socket_process: Received mini frame before first full voice frame

[Tim Panton]

On 15 May 2007, at 21:48, Sanjay Rajdev wrote:

Never received a response for this from anyone. This is being seen  
more frequently now.

Please Suggest.


The 'official' cause for this would be that your network has delayed  
or dropped the
initial Voice Fullframe in a call, so that the subsequent mini-frame  
has arrived first.
Asterisk needs to see the fullframe to be sure which codec the call  
will use.

The call should pick up fine once the fullframe arrives (or is retried).

On the other hand it could be a symptom of some other problem.

Do your calls go through ok when you get this message ?

Tim Panton

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



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[asterisk-users] Re: GSM Cards for Asterisk (UK)

[Benny Amorsen]

> "MB" == Matt Brown <[EMAIL PROTECTED]> writes:

MB> Does anyone have any experience with a GSM card, preferably Quad
MB> Span (4 GSM modules or higher) for use in the UK. I have seen the
MB> Junghanns* version but I am not keen on the limitation of having
MB> to use a BriStuffed version of Asterisk.

The Junghanns GSM cards are ok, but they have the IRQ issues which
used to plague Digium's cards as well. (Kudos to Digium for fixing
that by the way!). Junghanns has also been less than responsive about
issues.

Voismart is the only competitor I have found. We do not have any of
their cards yet, but they have been very responsive to my technical
questions so far.


/Benny


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[asterisk-users] Quadbri Cellular Issue

[Antonio Martínez Contreras]

Hello everybody, and first of all sorry for my poor English.

I'm having trouble with Quadbri installed on Asterisk 
1.2.17-BRIstuffed-0.3.0-PRE-1y-e. Everything is working fine, except calling 
to switched off or "out of coverage" cell phones. In this case I have to 
wait 40 seconds until Asterisk tell me that "all circuits are busy now" 
instead of receive cell phone company message of "The cell phone you are 
calling is unavailable". We tried a lot of configurations in zapata.conf and 
zaptel.conf, but we still have the same problem. I show my config:


zapata.conf:

=
;
; Zapata telephony interface
;
; Configuration file

[trunkgroups]

[channels]

language=es

usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
group=1
echocancel=yes
echocancelwhenbridged=yes
echotraining=800
immediate=no

switchtype=euroisdn
signalling=bri_cpe_ptmp
pridialplan=unknown
prilocaldialplan=unknown
priindication=outofband
facilityenable=yes
relaxdtmf=yes
context=from-zaptel
rxgain=0.0
txgain=0.0
channel => 1,2

faxdetect=incoming

;Include genzaptelconf configs
#include zapata-auto.conf

#group=1

;Include AMP configs
#include zapata_additional.conf



;Include BRI-HFC configs
#include zapata-BRI-HFC.conf

===

zaptel.conf

===

# Autogenerated by /usr/local/sbin/genzaptelconf -- do not hand edit
# Zaptel Configuration File
#
# This file is parsed by the Zaptel Configurator, ztcfg
#

# It must be in the module loading order


# Global data

loadzone = es
defaultzone = es

# HFC CARD NOTE:
# When run for the first time the card numbers start from 1
# When run subsequently the card numbers start from 0
# Go figure!


# Span 1: ztqoz/2/1 "quadBRI PCI ISDN Card 1 Span 1 [TE]"
span=1,1,3,ccs,ami
bchan=1-2
dchan=3


# Span 2: ztqoz/2/2 "quadBRI PCI ISDN Card 1 Span 2 [TE]"
#span=2,0,3,ccs,ami
#bchan=4-5
#dchan=6


# Span 3: ztqoz/2/3 "quadBRI PCI ISDN Card 1 Span 3 [TE]"
#span=3,0,3,ccs,ami
#bchan=7-8
#dchan=9


# Span 4: ztqoz/2/4 "quadBRI PCI ISDN Card 1 Span 4 [TE]"
#span=4,0,3,ccs,ami
#bchan=10-11
#dchan=12

===

In channel debug I have:

Debugging on new channels is enabled

   -- Executing Macro("SIP/200-09fc1698", "dialout-trunk|2|637574972||") in 
new stack


   -- Executing Set("SIP/200-09fc1698", "DIAL_TRUNK=2") in new stack

   -- Executing Set("SIP/200-09fc1698", "_NODEST=") in new stack

   -- Executing Set("SIP/200-09fc1698", "DIAL_NUMBER=637574972") in new 
stack


   -- Executing Set("SIP/200-09fc1698", "ROUTE_PASSWD=") in new stack

   -- Executing Set("SIP/200-09fc1698", "DIAL_TRUNK_OPTIONS=tr") in new 
stack


   -- Executing GotoIf("SIP/200-09fc1698", "1?noauth") in new stack

   -- Goto (macro-dialout-trunk,s,8)

   -- Executing Set("SIP/200-09fc1698", "GROUP()=OUT_2") in new stack

   -- Executing Macro("SIP/200-09fc1698", "user-callerid|SKIPTTL") in new 
stack


   -- Executing NoOp("SIP/200-09fc1698", "user-callerid: device 200") in 
new stack


   -- Executing GotoIf("SIP/200-09fc1698", "0?report") in new stack

   -- Executing GotoIf("SIP/200-09fc1698", "0?start") in new stack

   -- Executing Set("SIP/200-09fc1698", "REALCALLERIDNUM=200") in new stack

   -- Executing NoOp("SIP/200-09fc1698", "REALCALLERIDNUM is 200") in new 
stack


   -- Executing Set("SIP/200-09fc1698", "AMPUSER=200") in new stack

   -- Executing Set("SIP/200-09fc1698", "AMPUSERCIDNAME=200") in new stack

   -- Executing GotoIf("SIP/200-09fc1698", "0?report") in new stack

   -- Executing Set("SIP/200-09fc1698", "CALLERID(all)=200 <200>") in new 
stack


   -- Executing Set("SIP/200-09fc1698", "REALCALLERIDNUM=200") in new stack

   -- Executing NoOp("SIP/200-09fc1698", "TTL:  ARG1: SKIPTTL") in new 
stack


   -- Executing GotoIf("SIP/200-09fc1698", "1?continue") in new stack

   -- Goto (macro-user-callerid,s,21)

   -- Executing NoOp("SIP/200-09fc1698", "Using CallerID "200" <200>") in 
new stack


   -- Executing Macro("SIP/200-09fc1698", "record-enable|200|OUT") in new 
stack


   -- Executing GotoIf("SIP/200-09fc1698", "0?2:4") in new stack

   -- Goto (macro-record-enable,s,4)

   -- Executing DeadAGI("SIP/200-09fc1698", 
"recordingcheck|20070517-111347|asterisk-2589-1179393227.3") in new stack


   -- Launched AG

[asterisk-users] Re: MeetMe and ChannelRedirect

[Tony Mountifield]

In article <[EMAIL PROTECTED]>,
Rafael Vidal Aroca <[EMAIL PROTECTED]> wrote:
> i'm trying to implement the following scenario:
> 
> - A user calls number 700
> - Asterisk then dials to extensions 100, 200, 300, 400 and 500
> - And then bridges all calls to a conference room
> 
> I tried to use MeetMe and ChannelRedirect, but seems that after 
> channel redirect nothing more is executed. So, this seem to work for the 
> caller and first called, but the others stay outside.
> 
> Could anyone help or give me a hint?

The way I did this kind of thing was like this:

1. Extension 700 calls an AGI script which generates a .call file in
/var/spool/asterisk/outgoing for each of the calls to the other extensions.
Extension 700 then drops into the Meetme room to wait for the others.

2. Each call file specifies a Local channel to make the call to the extension,
and uses the Context, Extension and Priority fields to direct the answered
call into the Meetme room.

If any of the calls to the other extensions fails (e.g. busy), you don't get
any notification of that. If you want such notification, you will need to
get a lot more complex, probably involving a controlling process using the
Manager API.

Hope this helps.

Cheers
Tony
-- 
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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Re: [asterisk-users] zaptel huge irq problem

[François Delawarde]

I don't really know of other virtualization technology other than Xen, 
and I thank you for guiding me through this, but I have a few doubts 
related to the choice of a virtualization technology in a host with 
Asterisk:


- Isn't the fact that KVM is now included in the mainstream Linux kernel 
as of 2.6.20 a certain type of 'proof' that it could be stable enough 
compared to others (of course there could be licensing or other 
political/friendship issues)?


- Even if the virtual guests aren't totally stable and 100% reliable 
yet, wouldn't the use of KVM be better with Zaptel compatible cards than 
Xen, in architecture point of vue, as it is only a kernel module that 
-as far as I know- don't appear to be changing fundamental issues like 
IRQ handling or I/O scheduling in the kernel, and from the fact that 
virtual machines are treated like simple processes?


François.



Andrew Kohlsmith wrote:

On Wednesday 16 May 2007 1:00 pm, François Delawarde wrote:
  

Thanks again for your help, and sorry if I was not 'that' convinced on
your first answer and sent a mail to Xen user mailing list to check if
they knew that issue (no answer yet). Now I almost believe you a lot. If
I understand well I have two options, recode Xen or abandon it. I'll
probably go for the 2nd choice and start looking at other solutions, KVM
seems to be a good choice and shouldn't interfere much with Asterisk
(again: as far as mailing lists say).



Let me try to understand this:

Xen is a (far) more mature virtualization technology than KVM, and it's been 
said that it's commercialization was rushed.  So you're going to try KVM, 
which is still under heavy development, as a stable solution?


-A.
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Re: [asterisk-users] GSM Cards for Asterisk (UK)

[Steve Kennedy]

On Wed, May 16, 2007 at 09:15:49PM +0100, Matt Brown wrote:

[snip]
> No, this client has a number of engineers all over the UK and they  
> have a large mobile contract with several handsets - their current  
> tariff includes free calls to other mobiles under the contract
> so what they are trying to achieve is operators in the call centre  
> calling their own engineers get to make the call for free. The plan  
> was to get another 4 or so SIMS from the mobile company and
> slot them into a PCI based device, having a pre-defined list of  
> mobiles and then these calls are routed via the GSM card and network.
> I assume this would be acceptable ? In which case I presume that they  
> are unable to find a GSM gateway in the UK ?

Yup sounds fine. There are several vendors of "external" multi-SIM
gateways, prices vary considerably.

Steve

-- 
NetTek Ltd  UK mob +44-(0)7775 755503
UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455
Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED]
Euro Tech News Blog http://eurotechnews.blogspot.com
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[asterisk-users] how to define a key to decline incoming call

[asterisk]

Hi all.
We have Snom phones which do have a defined key in order to drop incoming
call WITHOUT answering.

Pressing that key, a "SIP/2.0 486 Busy Here" message is sent back.

We have other phones (I.E. DECT Siemens C450IP, or ATCOM 320 or other)
which DO NOT have any key to do that (or the key does not work, as is with
Siemens C450 IP ): you have to answer and immediatly after hangup the call.

Acting on feature.conf we succed in defining keys for blind transfer or
attended transfer: the last thing we need is the ability to drop an
incoming  call without answering it. Is there any way to define a key (or
double-key, i.e. "*4") to send back a  "SIP/2.0 486 Busy Here" message ?

thanks in advance,

Andrea

Chi ricevesse questa mail per errore e' gentilmente pregato di cancellarla.

Visitate il sito http://www.frameweb.it

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[asterisk-users] CDR is not written

[Khaled Chehab]

I created Master.csv in /var/log/asterisk/cdr-csv ,even did not work,do
freepbx make this problem,and how can I trouble shoot it.


Thanks


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ken Williams
Sent: Wednesday, May 16, 2007 11:58 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [SPAM]RE: [asterisk-users] CDR is not written 

I fought this for a bit when I found if the file Master.csv didn't exist, it
wouldn't create it on it's own.  I created an empty csv file, CDR started
writing.
 
Ken



From: [EMAIL PROTECTED] on behalf of Khaled Chehab
Sent: Thu 5/17/2007 10:50 AM
To: [EMAIL PROTECTED]
Cc: asterisk-users@lists.digium.com
Subject: [asterisk-users] CDR is not written 



I installed asterisk 1.4.4 final ,but the cdr is not written any patch or
tweaking can be done 

 

 

Regards

 





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RE: [asterisk-users] SIP Hardware Phone

[Per Jessen]

Chris Bagnall wrote:

> We've had good
> results with the following (in ascending price order, as per prices in
> the UK): Elmeg IP290 (almost identical to the old Snom 190) 
> Linksys SPA942 
> Aastra 57i 
> Linksys SPA962

We're using only Linksys - 921s and 941s.  When I was researching
prices, I found I could buy them off a local Swiss retailer, and sell
them on at ebay in the UK - for some reason the UK prices were
significantly higher.  This was about two months ago - today:

SPA921 - SFr124 - roughly GBP51  (incl.VAT)
SPA941 - SFr136 - roughly GBP56. (incl.VAT)

On ebay.co.uk, the SPA-921 is selling for almost twice the price above.
I just don't get it.  I don't know the street-price though.



/Per Jessen, Zürich

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Re: [asterisk-users] The downside of Asterisk and least cost routing...

[Per Jessen]

Alex Balashov wrote:

> On Wed, 16 May 2007, Stephen Bosch said something to this effect:
> 
>> The fax-to-e-mail services charge as much as the telco does for a
>> business line, sometimes more (at least, the ones I can deal with in
>> this area). Better to set-up hylafax, IMHO.
> 
>Not necessarily, except perhaps in cases of very high volumes.
> 

Actually, I think hylafax+iaxmodem are particularly useful for small
volumes - the kind of situation where you do need a fax, but you might
only receive or transmit something once a week at most.  Setting up
hylafax+iaxmodem takes about an hour.  I haven't found any drawbacks
yet.


/Per Jessen, Zürich

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RE: [SPAM]RE: [asterisk-users] CDR is not written

[Khaled Chehab]

I created Master.csv in /var/log/asterisk/cdr-csv ,even did not work,do
freepbx do this problem,and how can I trouble shoot it.


Thanks


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ken Williams
Sent: Wednesday, May 16, 2007 11:58 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [SPAM]RE: [asterisk-users] CDR is not written 

I fought this for a bit when I found if the file Master.csv didn't exist, it
wouldn't create it on it's own.  I created an empty csv file, CDR started
writing.
 
Ken



From: [EMAIL PROTECTED] on behalf of Khaled Chehab
Sent: Thu 5/17/2007 10:50 AM
To: [EMAIL PROTECTED]
Cc: asterisk-users@lists.digium.com
Subject: [asterisk-users] CDR is not written 



I installed asterisk 1.4.4 final ,but the cdr is not written any patch or
tweaking can be done 

 

 

Regards

 





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