Re: [asterisk-users] Attended transfer and dialplan

[Steve Murphy]

On Fri, May 29, 2009 at 11:35 PM, Olivier  wrote:

>
>
> 2009/5/29 Danny Nicholas 
>
>>  I’m pretty sure that attended transfer is a “features” function, not a
>> dialplan one.
>>
>
> Yes, you're right but do you think there's such a big difference between
> both that it shouldn't be easy or even possible to add support of attended
> transfer in dialplan ?
>
> What I have in my mind is this :
>
> Today, Dial application M or U options allows macro execution when caller
> and callee are connected.
> What if this same macro could be also launched during some later events
> (like attended transfer) ?
> With features.conf, you could then specify :
> - how lo launch an attended transfer (which key to type as today),
> - if a given "feature" (attended transfer, parking, ...) should be
> supported by Dial macro option (for compatibilty, default could be set to
> none)
>
> and with extension.conf, you could specify :
> - which specific treatment (sending UserEvents, launching an external
> program, ...) to apply
>
> In this puzzle, if Asterisk could support a few more standard variables
> like ATTENDED_TRANSFERER ATTENDED_TRANSFER_TARGET, you would everything to
> define and run attended transfers specific logic :
>
> exten => 123,1,Dial(SIP/123,M(mymacro^arg1^arg2)) ; mymacro is launched
> upon connection and specified (in features.conf) events
>
> [macro-mymacro]
> GotoIf("x${ATTENDED_TRANSFERER}", 
>
> What about that ?
>

Olivier--

This is actually not a bad idea.Why single out just the Attended xfer? Why
would you treat
attended xfers differently than blind xfers? Just curious.

Also, calling just one macro for all features seems a bit restricted. Why
not allow the features.conf
to specify which macro/gosub to call, for each feature?  Dial is already
overloaded with options,
anything that could be offloaded would probably be desirable. Plus, calls
that were not initiated by
a dialplan "Dial()" invocation might not be able to provide that option.

Another question: what do you need this functionality to *do*? It could be
that there is an already
existing functionality that you could exploit to get the same results?

murf


>
>
>>
>> On my system I do *2 and asterisk says transfer, then I punch in the new
>> extension.
>>
>>
>>
>>
>>  --
>>
>> *From:* asterisk-users-boun...@lists.digium.com [mailto:
>> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Olivier
>> *Sent:* Friday, May 29, 2009 10:29 AM
>> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
>> *Subject:* [asterisk-users] Attended transfer and dialplan
>>
>>
>>
>> Hi,
>>
>> How can you add specific statements into Asterisk dialplan (extension.ael,
>> ...) for attented transfers ?
>>
>> I can see Asterisk sending Transfer or Masquerade events through AMI (in
>> 1.6.1) but I could use an external program to catch those events but I would
>> prefer to use dialplan instead.
>>
>> Any idea ?
>>
>> Regards
>>
>> --
Steve Murphy
ParseTree Corp
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[asterisk-users] Achoring MEdia

[Jay Ray]

I have 2 hosts that Asterisk is in between of...and for both I have 
canreinvite=no - but asterisk still sends re-invite to get out of the media 
path.
 
Proxy 1 --> Asterisk--> Proxy 2
 
I want asterisk to anshor media..
 
In extenstions.conf I have an entry to send calls for say 5551000 to proxy 2 
and if  I suffix that entry with ||t , asterisk does anchor media
 
However, it is rejecting the REFER sip method by sending a 6xx messageproxy 
2 is trying to send the call back out as a transfer and hence it sends a 
REFERhow can I make it such that media is anchored as wel as the REFER 
method is accepted by asterisk


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Re: [asterisk-users] Digium Fax Driver

[Lee Howard]

Tilghman Lesher wrote:
> On Sunday 07 June 2009 19:39:50 Lee Howard wrote:
>   
>> Tilghman Lesher wrote:
>> 
 What's the use case for the Digium
 driver? Am I missing something by not using it?
 
>>> While they accomplish the same goal, the commercial driver is based upon
>>> a different codebase,
>>>   
>> Ok.
>>
>> 
>>> provides support for patented fax protocols,
>>>   
>> Really?  V.34-fax (33,600 bps) is supported?  I had understood differently.
>> 
>
> I would research the patents involved, but I am prohibited by employment
> contract from exploring patents granted.

Due to said employment contract prohibitions you can't tell me whether 
or not Digium's Fax Application supports V.34-fax (33,600 bps)?

> My understanding is that there are
> certain aspects of fax that are still under patent,

Yes.  Specifically V.34.  If my understanding is correct the relevant 
patents expire in a few years.

> and those are provided
> (along with indemnification) by the commercial driver.
>   

Understood.  But it was my understanding that V.34-fax was not supported 
by Digium's Fax Application.  And if that's correct, then there are no 
patents for which indemnification is necessary.  That's not to say that 
a commercial fax driver does not have its place with some customers.  I 
only want to clear up any misrepresentations about possible patent 
infringements by spandsp to which you alluded.

> I'm not suggesting that the commercial driver is more reliable,
> only that it enjoys far more testing.
>   

Again, regardless of your knowledge of how much testing goes into your 
employer's product, I question your ability to know with any degree of 
certainty as to how much testing has been involved with competing 
products.  I certainly know that *I* have no clue with regards to 
spandsp other than the testing to which I've been witness.  So I am 
curious to know how you are able to make such assertions.

> That said, hours of use in production do not speak to the amount of testing
> done.

Scrutiny of production use exposure does not constitute testing?  Well, 
I would argue that you cannot possibly test real-world conditions 
without actually placing the test system into the real-world with 
real-world use (thus, production).  I cannot think of a better way to 
test software than to eventually put it into real-world production use 
and then have the developers monitor those systems closely.

> IAXmodem is a completely different ball of wax, and I think you would agree
> that if the builtin FAX support in spandsp provided excellent support, there
> never would have been a reason for IAXmodem to be developed.

I'm interested to know how you understand my intent in developing 
IAXmodem differs from what I recall.  I developed IAXmodem because I 
needed to interface HylaFAX through an Asterisk PBX without purchasing 
additional hardware (other than the T1 cards that were already involved).

Thanks,

Lee.


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Re: [asterisk-users] Digium Fax Driver

[Tilghman Lesher]

On Sunday 07 June 2009 19:39:50 Lee Howard wrote:
> Tilghman Lesher wrote:
> >> What's the use case for the Digium
> >> driver? Am I missing something by not using it?
> >
> > While they accomplish the same goal, the commercial driver is based upon
> > a different codebase,
>
> Ok.
>
> > provides support for patented fax protocols,
>
> Really?  V.34-fax (33,600 bps) is supported?  I had understood differently.

I would research the patents involved, but I am prohibited by employment
contract from exploring patents granted.  My understanding is that there are
certain aspects of fax that are still under patent, and those are provided
(along with indemnification) by the commercial driver.

> > and enjoys
> > far more testing than Steve can reasonably do for an unpaid side project
>
> I can't speak for how much testing the T.38 or T.30 sides to spandsp
> have had.  However, the T.31 and V.17, V.29, V.27ter, and V.21 aspects
> have undergone a *tremendous* amount of testing and development scrutiny
> (as these are used in IAXmodem).  I am aware of single installations
> that communicate successfully with very arbitrary kinds of fax machines
> in the USA which alone have communicated several millions of pages of
> fax in the last two years.  If there ever is a problem (and it is
> extremely rare) I hear about it.

And I agree that IAXmodem is by far more reliable than the builtin fax support
in spandsp.  I'm not suggesting that the commercial driver is more reliable,
only that it enjoys far more testing.  I don't know what part of my knowledge
is considered under NDA, so I'm hesitant to say more than that.

That said, hours of use in production do not speak to the amount of testing
done.  I myself have software that has been in production for millions of
hours (unrelated to Asterisk), and the amount of testing it enjoyed was less
than one hour.

> So I don't know of what kind of testing you speak, but I would caution
> you to reserve judgment simply on the basis that it is "an unpaid side
> project".

IAXmodem is a completely different ball of wax, and I think you would agree
that if the builtin FAX support in spandsp provided excellent support, there
never would have been a reason for IAXmodem to be developed.

-- 
Tilghman

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Re: [asterisk-users] How run AsyncAGI commands in background

[Moises Silva]

On Sun, Jun 7, 2009 at 4:37 PM, Jose Arias wrote:
> Hi Moy,
>
> I'll do it so, but for your answer, it seems you are thinking about it as it
> could be a bug. I don't think so. I mean: the redirect action on a channel
> in AsyncAGI stops the current agi execution. It's the normal behavior. It's
> the way to stop a playfile on a channel if it was previously launched from
> AsyncAGI: making a redirect out of the AsyncAGI loop.
>
> Therefore, when I realized the previously launched EXE MixMonitor AsyncAGI
> execution was stopping after doing a redirect to meetme, I didn't think it
> was a bug. I though what I was needing it was a way to tell AsyncAGI, "hey,
> don't stop this agi execution on the channel, even it will be redirected out
> of AGI" on an individual basis for each AsyncAGI EXEC command launched.
>
> Thanks
> Jose
>
The way I see it if you make EXEC MixMonitor inside AsyncAGI loop and
then redirect to MeetMe and you don't get the audio recorded, then
it's not a normal behavior, MixMonitor is an application that should
passively monitor the channel audio independently of where the channel
is (regardless of whether the command was executed in Async AGI or
dial plan or whatever). However you are also using an old asterisk
version and is not likely you can report a bug unless you upgrade to
the latest Asterisk and reproduce without a patched Asterisk (for
example executing EXEC MixMonitor inside a regular AGI script and then
redirect to MeetMe).

-- 
Moises Silva
Software Developer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON
L3R 9T3 Canada
t. 1 905 474 1990 x 128 | e. m...@sangoma.com

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Re: [asterisk-users] CDR question

[Steve Murphy]

Jim--


On Thu, Jun 4, 2009 at 1:40 AM, Jim Boykin  wrote:

> Hi,
>
> Asterisk does not post CDR when dial status is CHANUNAVAIL.


CDR's are, at the current time, and always have been attached to the channel
struct;
so, if you don't create a channel, then there is nowhere to attach a CDR,
and no way
to process that.


>
>
> Can someone tell me what are the conditions under which CDR is not posted?


We try to filter the CDR if a channel were created, but did nothing; an
example is
where a Dahdi device is taken off hook, and then hung up again. But getting
all the
conditions right has been tricky to filter this sort of event sequence.

I think you'll find that CDR's are one of the least solid parts of Asterisk
at the moment.
There's brave and creative folks working on fixing the current
implementation, but
as far as I'm concerned, it's got some fundamental problems, and needs to be

overhauled. If you are interested, you can read my spec for a new approach
by:

svn co http://svn.digium.com/svn/asterisk/team/murf/RFCs

and then looking at the pdf in that dir, for my spec for the CEL->CDR
proposal.

While I have abominated the complexity of the ForkCDR/NoCDR/etc mechanisms
of the current solution, I have considered making the spec include them for
backward
compatibility... Current implementations based on the current mechanisms
shouldn't
have to be made obsolete, although they usually do depend on a great deal of
undocumented
behavior, that may be tricky to imitate.

murf


>
>
> Thanks
> Jim
>
>
-- 
Steve Murphy
ParseTree Corp
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[asterisk-users] remote queue members

[Gabriel Ortiz Lour]

Hi all,

  I'd like to know the best way to deal with queue member that are reached
trough a SIP trunk. Let me explain:

1) "Master" asterisk box with my call queue
2) "Slave" asterisk box with a channel bank interface

the two boxes are connected trough a SIP trunk, and the dialplan in the 1st
box connect (for eg.) Local/0...@gw to the 30rd channel on the 2nd box. All
my 60 channels share this SIP trunk and they are used for queue member.

How should I add this to the queues? I was adding Local/0...@gw in my
QueueAddMember Interface, but I dont see how this hould know the state of
the member on the other side of the SIP trunk. Is there any way to get this
states correctly?

Thanks,
Gabriel
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Re: [asterisk-users] Digium Fax Driver

[Lee Howard]

Tilghman Lesher wrote:
>> What's the use case for the Digium
>> driver? Am I missing something by not using it?
>> 
>
> While they accomplish the same goal, the commercial driver is based upon
> a different codebase,

Ok.

> provides support for patented fax protocols,

Really?  V.34-fax (33,600 bps) is supported?  I had understood differently.

> and enjoys
> far more testing than Steve can reasonably do for an unpaid side project

I can't speak for how much testing the T.38 or T.30 sides to spandsp 
have had.  However, the T.31 and V.17, V.29, V.27ter, and V.21 aspects 
have undergone a *tremendous* amount of testing and development scrutiny 
(as these are used in IAXmodem).  I am aware of single installations 
that communicate successfully with very arbitrary kinds of fax machines 
in the USA which alone have communicated several millions of pages of 
fax in the last two years.  If there ever is a problem (and it is 
extremely rare) I hear about it.

So I don't know of what kind of testing you speak, but I would caution 
you to reserve judgment simply on the basis that it is "an unpaid side 
project".

Thanks,

Lee.

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Re: [asterisk-users] Digium Fax Driver

[Tilghman Lesher]

On Sunday 07 June 2009 16:28:53 sean darcy wrote:
> Steve Underwood wrote:
> > Elliot Murdock wrote:
> >> Hello!
> >> I have a 64 bit Asterisk system and am wondering how to use Digium's
> >> 32 bit fax driver.  Is there some kind of emulation that can be used?
> >> Thanks!
> >> Elliot
> >
> > Use the FAX support built into Asterisk 1.6 and you won't have that
> > limitation.
>
> What IS the difference between Digium fax driver and the FAX support in
> 1.6? When should you use one, when the other? Do they have different use
> cases, or are they just competitors?
>
> I'm using ReceiveFax() in 1.6, together with fax2mail, for all our
> incoming faxes. It works great. What's the use case for the Digium
> driver? Am I missing something by not using it?

While they accomplish the same goal, the commercial driver is based upon
a different codebase, provides support for patented fax protocols, and enjoys
far more testing than Steve can reasonably do for an unpaid side project.

-- 
Tilghman

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Re: [asterisk-users] How run AsyncAGI commands in background

[Jose Arias]

Hi Moy, 

I'll do it so, but for your answer, it seems you are thinking about it 
as it could be a bug. I don't think so. I mean: the redirect action on a 
channel in AsyncAGI stops the current agi execution. It's the normal 
behavior. It's the way to stop a playfile on a channel if it was 
previously launched from AsyncAGI: making a redirect out of the AsyncAGI 
loop.


Therefore, when I realized the previously launched EXE MixMonitor 
AsyncAGI execution was stopping after doing a redirect to meetme, I 
didn't think it was a bug. I though what I was needing it was a way to 
tell AsyncAGI, "hey, don't stop this agi execution on the channel, even 
it will be redirected out of AGI" on an individual basis for each 
AsyncAGI EXEC command launched.


Thanks
Jose

Moises Silva escribió:

> then it should work, create a *simple* extensions.conf and pastebin it
> along with instructions so I can try to reproduce.
>
> > On Sat, Jun 6, 2009 at 5:02 PM, Jose Arias> wrote:

> >/ Hi,
/> >/ Asterisk 1.4.18
/> >/ AsyncAGI patch from http://moythreads.com/testasync2.diff
/> >/ Regards
/> >/ Jose

/
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Re: [asterisk-users] Digium Fax Driver

[sean darcy]

Steve Underwood wrote:
> Elliot Murdock wrote:
>> Hello!
>> I have a 64 bit Asterisk system and am wondering how to use Digium's 
>> 32 bit fax driver.  Is there some kind of emulation that can be used?
>> Thanks!
>> Elliot
> Use the FAX support built into Asterisk 1.6 and you won't have that 
> limitation.
> 
> Steve
> 

What IS the difference between Digium fax driver and the FAX support in 
1.6? When should you use one, when the other? Do they have different use 
cases, or are they just competitors?

I'm using ReceiveFax() in 1.6, together with fax2mail, for all our 
incoming faxes. It works great. What's the use case for the Digium 
driver? Am I missing something by not using it?

sean


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Re: [asterisk-users] Call recording in - out

[Joao Gomes Pereira]

Hello
I did as you told me, but the problem remains.
Im using Asterisk 1.2.x

and this is my config:

queues.conf -

[general]
persistentmembers = no


[queue_1]

persistentmembers = no
monitor-format=wav
monitor-join=yes
monitor-type=MixMonitor

wrapuptime=3
timeout=15
strategy=roundrobin
retry=5
queue-youarenext=
queue-thereare=
queue-thankyou=
queue-callswaiting=
member => Agent/600
member => Agent/601


agents.conf -

[general]
persistentagents=no

[agents]

updatecdr=no


custom_beep=beep
group=1
wrapuptime=19
ackcall=no
musiconhold => music
group=1

agent => 600,1234,Jose
agent => 601,1234,Maria


The calls are recordedbut always produces 2 separated files, with 
"in" and "out".
What could be missing?
Do I need to create a line in crontab to mix the 2 files?
Thanks
regards
Joao Pereira



Kurian Thayil wrote:
> Hi,
>
> I had similar issue which happened when record option was mentioned in
> both agents.conf and queues.conf. When I commented the recordagentcalls
> option in agents.conf, it started to work. Mention the monitor option
> only in the queues.conf file. Do try.
>
> Regards,
>
> Kurian Thayil.
>
> On Sun, 2009-06-07 at 17:51 +0100, Joao Gomes Pereira wrote:
>   
>> Hello to all
>> I'm trying to record the calls going to my queues, but asterisk creates 
>> 2 files, one with the inbound and another with the outbound sound.
>> I know Sox should mix the 2 files automatically in the end, but this 
>> isn't happening.
>> I have sox installed in my server.
>>
>> How can I force Sox to mix the files?
>> Here is my config:
>>
>>
>> queues.conf-
>>
>> [general]
>> persistentmembers = no
>> monitor-format=wav
>> monitor-join=yes
>> monitor-type=mixmonitor
>>
>>
>>
>> [queue_1]
>>
>> persistentmembers = no
>> monitor-format=wav
>> monitor-join=yes
>> monitor-type=mixmonitor
>>
>>
>> wrapuptime=3
>> timeout=15
>> strategy=roundrobin
>> retry=5
>> member => Agent/600
>> member => Agent/601
>>
>> agents.conf-
>>
>>
>> [general]
>> persistentagents=no
>>
>> [agents]
>>
>> updatecdr=no
>>
>> recordagentcalls=yes
>> recordformat=wav
>> monitor-join=yes
>> savecallsin=/var/www/html/recordings/
>>
>> custom_beep=beep
>> group=1
>> wrapuptime=19
>> ackcall=no
>> group=1
>>
>> agent => 600,1234,Jose
>> agent => 601,1234,Maria
>>
>>
>>
>> Thanks
>> Regards
>> Joao Pereira
>>
>> 


-- 
StarTel - A Rede Livre
Joao Gomes Pereira
www.startel.pt
+351 304500650
sip: gomespere...@startel.pt


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Re: [asterisk-users] Ekiga, Twinkle and from where to start with open source

[Philipp Kempgen]

César Sequeira schrieb:

> I try to connect Qutecom in my Asterisk Server but without success.
> 
> What field I need to complete?
> 
> Username;
> Password;
> Realm (asterisk IP Address);

Default: "asterisk"

> Server (asterisk IP Address);
> Proxy (asterisk IP address);
> 
> It's correct?


Philipp Kempgen
-- 
AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied  ->  http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
Videos of the AMOOCON VoIP conference 2009 ->  http://www.amoocon.de
-- 

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Re: [asterisk-users] Asterisk Clustering

[Torintino T]

Thanks Noah for your helpful reply.
 
My setup will be 2 Asterisk (Trixbox) servers, Active/Passive, 2 PRIs through 2 
Vega 400 gateways.
 
i tried to follow some threats, like as the below:
 
http://www.trixbox.org/forums/trixbox-forums/open-discussion/ha-cluster
 
But i think something wrong in this or my setup,
as from time to time the floating IP is timing out, and some services on both 
servers can't operate perfectly.
 
As i was collecting different parts from different threats,
And unfortunately i couldn't get a complete guide to follow to finalize my 
clustering setup successfully.
 
So your help will be highly appreciated, if you can post your successful setup 
steps somewhere.
 
Thanks a lot for your help and time.
 
Torintino


 
> Date: Fri, 29 May 2009 22:52:06 -0400
> From: noahisaacmil...@gmail.com
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] Asterisk Clustering
> 
> > Please, does anybody have a good document describes well
> > the optimum method to achieve Asterisk Redundancy/Clustering on 2 servers.
> 
> Documentation?!... well... there's not much.
> 
> It depends on what you're trying to achieve with your cluster. If you
> want a simple active/passive failover cluster, I'd suggest
> heartbeat/pacemaker for clusterizing the services coupled with drbd
> for replicating files. I recently set up a cluster like this that's
> now in production. This particular system connects to the PSTN via
> PRIs, and a specialized piece of hardware detects which system is the
> active node and physically routes the PRIs to that node.
> 
> I should probably write something up and post it somewhere, but time
> is always an issue. If you need specific help with this kind of
> setup, though, feel free to ask, and I may be able to assist.
> 
> If you want an active/active setup, I think you'll have to look into
> using dundi.
> 
> 
> - Noah
> 
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Re: [asterisk-users] Teliax: where's the space in CALLERID(num) from?

[sean darcy]

Philipp Kempgen wrote:
> sean darcy schrieb:
>> I'm having trouble setting callerid with teliax. I use a simple dial-out 
>> subroutine to set the callerid depending on the calling extension, and 
>> then dial out. Teliax is saying they're not seeing any callerid info.
> 
>> exten => s,n,Set(CALLERID(num)=${IF($[0${CALLERID(num)} > 140] ? 
>> ${MAINSTUB}${CALLERID(num)}:${MAINNUMBER} )} )
>^  ^
>   remove the trailing spaces

I'll try that tomorrow.

BTW, how can I test this remotely - that when no one is in the office.

I tried at the CLI:

originate SIP/178 extension 2024532...@longdistance

but that just generated the response:

-- Got SIP response 486 "Busy Here" back from 10.10.10.148

sean


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Re: [asterisk-users] Call recording in - out

[Kurian Thayil]

Hi,

I had similar issue which happened when record option was mentioned in
both agents.conf and queues.conf. When I commented the recordagentcalls
option in agents.conf, it started to work. Mention the monitor option
only in the queues.conf file. Do try.

Regards,

Kurian Thayil.

On Sun, 2009-06-07 at 17:51 +0100, Joao Gomes Pereira wrote:
> Hello to all
> I'm trying to record the calls going to my queues, but asterisk creates 
> 2 files, one with the inbound and another with the outbound sound.
> I know Sox should mix the 2 files automatically in the end, but this 
> isn't happening.
> I have sox installed in my server.
> 
> How can I force Sox to mix the files?
> Here is my config:
> 
> 
> queues.conf-
> 
> [general]
> persistentmembers = no
> monitor-format=wav
> monitor-join=yes
> monitor-type=mixmonitor
> 
> 
> 
> [queue_1]
> 
> persistentmembers = no
> monitor-format=wav
> monitor-join=yes
> monitor-type=mixmonitor
> 
> 
> wrapuptime=3
> timeout=15
> strategy=roundrobin
> retry=5
> member => Agent/600
> member => Agent/601
> 
> agents.conf-
> 
> 
> [general]
> persistentagents=no
> 
> [agents]
> 
> updatecdr=no
> 
> recordagentcalls=yes
> recordformat=wav
> monitor-join=yes
> savecallsin=/var/www/html/recordings/
> 
> custom_beep=beep
> group=1
> wrapuptime=19
> ackcall=no
> group=1
> 
> agent => 600,1234,Jose
> agent => 601,1234,Maria
> 
> 
> 
> Thanks
> Regards
> Joao Pereira
> 
-- 
Kurian Mathew Thayil.
(GPG KeyID: E232394F)


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[asterisk-users] Call recording in - out

[Joao Gomes Pereira]

Hello to all
I'm trying to record the calls going to my queues, but asterisk creates 
2 files, one with the inbound and another with the outbound sound.
I know Sox should mix the 2 files automatically in the end, but this 
isn't happening.
I have sox installed in my server.

How can I force Sox to mix the files?
Here is my config:


queues.conf-

[general]
persistentmembers = no
monitor-format=wav
monitor-join=yes
monitor-type=mixmonitor



[queue_1]

persistentmembers = no
monitor-format=wav
monitor-join=yes
monitor-type=mixmonitor


wrapuptime=3
timeout=15
strategy=roundrobin
retry=5
member => Agent/600
member => Agent/601

agents.conf-


[general]
persistentagents=no

[agents]

updatecdr=no

recordagentcalls=yes
recordformat=wav
monitor-join=yes
savecallsin=/var/www/html/recordings/

custom_beep=beep
group=1
wrapuptime=19
ackcall=no
group=1

agent => 600,1234,Jose
agent => 601,1234,Maria



Thanks
Regards
Joao Pereira

-- 
StarTel - A Rede Livre
Joao Gomes Pereira
www.startel.pt
+351 304500650
sip: gomespere...@startel.pt


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[asterisk-users] BUSYDETECT_* flags

[Phillip Neumann]

Hello all,

I have asterisk-1.4.25, and having some problems with analog lines  
becouse my Telco does not have "disconnect supervision".

So, i realize there are some switches on main/dsp.c:

BUSYDETECT_MARTIN, BUSYDETECT_TONEONLY, and  
BUSYDETECT_COMPARE_TONE_AND_SILENCE.

As far as i could read on the web, some people has use this to solve  
the "asteriks not hanging up" problem, and want to try them.

I dont see anything about this on any Makefile, how do i enable thouse  
flags?
Or is it that they are obsolete now?

Thanks!

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[asterisk-users] ANI

[Cary Fitch]

When Asterisk sends a call to "a phone company" via a PRI/Dahdi, does it
actually send CALLERID(ANI), or only CALLERID(NUM)?

Cary Fitch


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Re: [asterisk-users] How run AsyncAGI commands in background

[Jose Arias]

Never mind, it was my mistake. I had some problems with my email client.
Regards
Jose
2009/6/7 Philipp Kempgen 

> Moises Silva schrieb:
> > On Sat, Jun 6, 2009 at 7:18 PM, Philipp
> > Kempgen wrote:
> >> Jose Arias schrieb:
>  >>> Hi,
> >>> Asterisk 1.4.18
> >>> AsyncAGI patch from //http://moythreads.com/testasync2.diff
> >>> //
> >>> Regards
> >>
> >> So what?
> >>
> > What do you mean with "so what?", if you have not been involved in the
> > conversation you would not understand.
> >
> > http://lists.digium.com/pipermail/asterisk-users/2009-June/232995.html
>
> Sorry for the noise. I didn't realize this was a discussion. The
> message didn't quote anything and the subject didn't start with
> "Re: " so it appeared as if Jose was just posting his version of
> Asterisk without any context.
>
>
>Philipp Kempgen
> --
> AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied  ->  http://www.amooma.de
> Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
> Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
> Videos of the AMOOCON VoIP conference 2009 ->  http://www.amoocon.de
> --
>
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Re: [asterisk-users] Called party name with Cisco-2,811 gateway

[David Backeberg]

On Sun, Jun 7, 2009 at 4:20 AM, Yehavi
Bourvine wrote:
> Hello,
>
>   I am using a Cisco 2,811 gateway to connect Asterisk over PRI to our
> Nortel TX-1 PBX. Up to now I had only the calling party names passed both
> ways. After upgrading the Cisco to the latest release (12.4.24T) it began
> honoring the "remote-part-ID" field sent by Asterisk and sends the called
> name to the Nortel. However, I still do not get the called name from the
> Nortel to Asterisk.
>
> Has anyone managed to make this working?

There are a lot of choices when you set up your interconnectivity
between your Cisco gear and the T1/PRI channels. If you post your
dialpeer for this I'll take a look, although strictly speaking this
isn't really an asterisk question.

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Re: [asterisk-users] chan_dahdi missing in * 1.6.1.1

[David Backeberg]

On Sun, Jun 7, 2009 at 6:19 AM, Steve Repo wrote:
> I have a Sangoma A200 analog card with 2 FXO ports. It's working well
> with asterisk 1.4.22 and Zaptel. I decided to upgrade to asterisk
> 1.6/dahdi.
>
> I compiled and installed,
>
> dahdi-linux-2.1.0.4
> dahdi-tools-2.1.0.2
> libpri-1.4.10
> wanpipe-3.4.1
> asterisk 1.6.1.1
>
> My analog card is recognized in dahdi_hardware. However, asterisk
> cannot compile chan_dahdi.so. I've tried passing --with-dahdi to
> dahdi-linux-2.1.0.4 and dahdi-tools-2.1.0.2 src yet no luck. I tried
> passing --with-dahdi to dahdi-tools install dir yet no luck.
>
> What is the magic to compile chan_dahdi.so in asterisk 1.6.1.x?  I'm
> on centos 5.3.
>
> Also, asterisk 1.4.25 cannot compile chan_dahdi as well while the
> previous versions do. What changed or what am i missing?

There probably isn't magic. If you post the errors you got during the
compile we'll be more likely to be able to tell you what's going
wrong.

Specifically the stuff you got when you said you cannot compile
chan_dahdi.so would be important to post.

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Re: [asterisk-users] Meetme timeout

[David Backeberg]

On Sun, Jun 7, 2009 at 5:19 AM, Kurian Thayil wrote:
> extension is hit by channel. I wasn't aware that we can scan a channel
> continuously using an AGI. If so, how could we do that?

A somewhat ugly solution that gets the job done:
1) crontab, set up a periodic job that does something like
2) asterisk -rx "agi you want to kick off"

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[asterisk-users] Callback with a2billing

[abdelkader]

Hello,

Can anyone give me a sample configuration of Callback feature on a2billing.

Thanks.
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Re: [asterisk-users] How run AsyncAGI commands in background

[Philipp Kempgen]

Moises Silva schrieb:
> On Sat, Jun 6, 2009 at 7:18 PM, Philipp
> Kempgen wrote:
>> Jose Arias schrieb:
>>> Hi,
>>> Asterisk 1.4.18
>>> AsyncAGI patch from //http://moythreads.com/testasync2.diff
>>> //
>>> Regards
>>
>> So what?
>>
> What do you mean with "so what?", if you have not been involved in the
> conversation you would not understand.
> 
> http://lists.digium.com/pipermail/asterisk-users/2009-June/232995.html

Sorry for the noise. I didn't realize this was a discussion. The
message didn't quote anything and the subject didn't start with
"Re: " so it appeared as if Jose was just posting his version of
Asterisk without any context.


Philipp Kempgen
-- 
AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied  ->  http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
Videos of the AMOOCON VoIP conference 2009 ->  http://www.amoocon.de
-- 

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Re: [asterisk-users] Meetme timeout

[Kurian Thayil]

Hello Danni,

As you said, I went through the post and found that is applicable
everytime no matter how many members are there in a conference. And I
understand that I cannot completely rely on that. I need to do some
logical tweaks with some other application like an AGI to crack this
issue. I was under a notion that I could activate an AGI only when an
extension is hit by channel. I wasn't aware that we can scan a channel
continuously using an AGI. If so, how could we do that?

Regards,

Kurian Thayil.

On Thu, 2009-06-04 at 08:35 -0500, Danny Nicholas wrote:
> There was a nice post earlier this week about timing out a meetme
> conference.   You could combine that information with an AGI to monitor the
> meetme room and kick out after the timeout if the user count did not change.
> 
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kurian Thayil
> Sent: Wednesday, June 03, 2009 11:22 PM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] Meetme timeout
> 
> Hi All,
> 
> I am looking for an option in Meetme or similar which will enable to
> skip to next priority (a voicemail) if the person in Meetme conference
> is alone and if he is there for some time (say 3 minutes)? Any hints on
> this? Thanks in advance.
> 
> Regards,
-- 
Kurian Mathew Thayil.
(GPG KeyID: E232394F)


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[asterisk-users] chan_dahdi missing in * 1.6.1.1

[Steve Repo]

Hello,

I have a Sangoma A200 analog card with 2 FXO ports. It's working well
with asterisk 1.4.22 and Zaptel. I decided to upgrade to asterisk
1.6/dahdi.

I compiled and installed,

dahdi-linux-2.1.0.4
dahdi-tools-2.1.0.2
libpri-1.4.10
wanpipe-3.4.1
asterisk 1.6.1.1

My analog card is recognized in dahdi_hardware. However, asterisk
cannot compile chan_dahdi.so. I've tried passing --with-dahdi to
dahdi-linux-2.1.0.4 and dahdi-tools-2.1.0.2 src yet no luck. I tried
passing --with-dahdi to dahdi-tools install dir yet no luck.

What is the magic to compile chan_dahdi.so in asterisk 1.6.1.x?  I'm
on centos 5.3.

Also, asterisk 1.4.25 cannot compile chan_dahdi as well while the
previous versions do. What changed or what am i missing?

Thanks,
Steve

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[asterisk-users] Called party name with Cisco-2,811 gateway

[Yehavi Bourvine]

Hello,

  I am using a Cisco 2,811 gateway to connect Asterisk over PRI to our
Nortel TX-1 PBX. Up to now I had only the calling party names passed both
ways. After upgrading the Cisco to the latest release (12.4.24T) it began
honoring the "remote-part-ID" field sent by Asterisk and sends the
*called*name to the Nortel. However, I still do not get the called
name from the
Nortel to Asterisk.

Has anyone managed to make this working?

  Thanks! __Yehavi:
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Re: [asterisk-users] Sangoma A200 wt HW echo on 64bit Centos , 4 fxo and crosstalk!

[Gavin Henry]

That is correct. That is the first test we did.

On 07/06/2009, Moises Silva  wrote:
> On Sat, Jun 6, 2009 at 3:18 PM, Gavin Henry wrote:
>> Every call as soon as the sangoma card is live.
>>
>> Speak to Konrad on your techdesk for more info.
>>
>> Thanks.
>>
>
> I'll speak with him on Monday.
>
> However if you can provide more information before Monday I will be
> able to think beforehand on this matter.
>
> So please confirm this. If you get an incoming call and send it to
> Playback(demo-congrats) and then receive a second call and send it to
> Playback(tt-monkeys), both callers will listen both demo-congrats and
> tt-monkeys sounds?
>
> --
> Moises Silva
> Software Developer
> Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON
> L3R 9T3 Canada
> t. 1 905 474 1990 x 128 | e. m...@sangoma.com
>
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-- 
Sent from my mobile device

http://www.suretecsystems.com/services/openldap/
http://www.suretectelecom.com

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