Re: [asterisk-users] Core Dump - Asterisk 1.4.24 - Elastix

[Tzafrir Cohen]

On Tue, Nov 03, 2009 at 01:18:14AM -0300, Fernando Berretta wrote:
> Hi,
> 
> Yesterday I've got a core dump from Asterisk, other times I was able to 
> discover what this core dump was related with through gdb Ouput info,, 
> but this time.. I'm really lost. Could some one please help me
> 
> GDB output is at
> 
> http://pastebin.com/m603e6a74

When Asterisk is installed from a binary package, you will normally get
binaries without debug symbols to save space. You sypically don't need
the debug information.

However for the cases you do need them, the debug symbols are available
in a separate package (with rpm: the *-debuginfo packages , with debs:
-dbg packages or several similar things).

Debug symbols are not needed at core dumping time. They are only needed
when you try to get something useful from a core dump (e.g. with gdb).
So you can install them after dumping the core. However they must be of
exacatly the same version of the Asterisk package installed on your
system (that dumped the core in the first place).

So basically: just install the package asterisk-debuginfo and try
aagain.

There's also a similar package for glibc for the libc stuff, though this
typically is less useful.

-- 
   Tzafrir Cohen
icq#16849755  jabber:tzafrir.co...@xorcom.com
+972-50-7952406   mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

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[asterisk-users] Exchange 2007 UM issues with Asterisk 1.6

[Kumeil Juma]

Hi Guys,

Does anyone know how to make the custom build of Asterisk 1.6.1 work with 
Exchange 2007 UM? It always times out with system unavailable and won't go 
through. I have TCP enabled and all the trunk and outgoing settings configured. 

 

When I use TrixBox CE 2.8.0.2, it works. But custom install, won't work, 
everything else works except for Exchange 2007 UM. 

 

Any help would be appreciated.

 

Thanks,

KJ
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[asterisk-users] Core Dump - Asterisk 1.4.24 - Elastix

[Fernando Berretta]

Hi,

Yesterday I've got a core dump from Asterisk, other times I was able to 
discover what this core dump was related with through gdb Ouput info,, 
but this time.. I'm really lost. Could some one please help me

GDB output is at

http://pastebin.com/m603e6a74

Any help would be appreciated.

Best Regards,
Fernando

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Re: [asterisk-users] Asterisk as Outbound Proxy ?

[Alex Balashov]

"Proxy" is not the correct term to describe this scenario, but yes, it 
is possible in principle.

Kristijan Vrban wrote:

> Hello, short question: is there a possibility to use asterisk as an 
> outbound proxy? iam open for any suggestions, use asterisk trunk, dirty 
> patches, ugly workarounds, everything.
> 
> What is want to build is:
> 
> SIP Phone -> via TLS/SRTP -> Asterisk as outbound proxy -> via UDP/RTP 
> -> VoIP-Provider
> 
> So Asterisk should just forward any incoming SIP messages (INVITE, 
> REGISTER) to the VoIP-Provider and do SIP TLS-> SIP UDP and SRTP -> RTP 
> translation (via *1.6.2 and the SRTP patch)
> 
> Kristijan
> 
> 
> 
> 
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-- 
Alex Balashov - Principal
Evariste Systems
Web : http://www.evaristesys.com/
Tel : (+1) (678) 954-0670
Direct  : (+1) (678) 954-0671

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Re: [asterisk-users] turn the ring tone OFF during dialing

[Joseph]

On 11/02/09 19:56, Joseph wrote:
>Is the a way to turn the ring tone OFF during dialing?
>When I'm in a macro mode I have to listen to ring the tone for 20sec before 
>macro finish and I get connected.

I've found a better solution, setting musing on hold before calling party 
answers: 
m: Provide Music on Hold to the calling party until the called channel answers. 

but it doesn't seems to work, there in musing during dialing:

[goto-dialout]
exten => 51,1,SetMusicOnHold(default)
exten => 51,n,Dial(SIP/18775898...@pstn-5665,20,m(default)M(atb))

Music on Hold is working OK, so why this one isn't working? 

-- 
Joseph

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[asterisk-users] turn the ring tone OFF during dialing

[Joseph]

Is the a way to turn the ring tone OFF during dialing?
When I'm in a macro mode I have to listen to ring the tone for 20sec before 
macro finish and I get connected.

-- 
Joseph

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Re: [asterisk-users] Real replacement for AgentCallBackLogin() on Asterisk 1.6

[Mariano Lecuona]

My mental plan orginilly was:

1.- Creating a macro that acceps ARGs like.
a.- agent
b.- queue/s

In the macro we could have the voice respose for the loging. I am using on
1.4 the following procedure.

* the agents call to 21 to loging, and it is promt just for the
passwd
* the agents call to 22 to logoff

using the same philosofy we could implement some easy marco that only ask
for the password and:

1.- sets the astdb
2.- sets the globals "AGENTBYCALLERID_X="
3.- adds the agent to the queues.


Let me work deeper on this idea and see what comes up.

ML

2009/11/2 Lenz Emilitri 

> We were thinking about doing something similar as well. A lot of people are
> asking for this. If there is anybody else interested, we could share the
> load
>
> I was thinking about creating a context like @agents, so that when you do
> the log-on you basically add Local/1...@agents as a member of the queue.
> When you ring it, it basically looks up for the actual device in AstDB and
> dials it like:
>
> Queue -> (member) Local/1...@agents -> (astdb) SIP/234
>
> I think that we should be able to forward channel state as well (using
> hints? I've never done it)  so that app_queue does not try dialling agents
> that are busy.
>
> I was thinking about storing queue-agent associations into config strings,
> and/or AstDB, and/or http over curl. And yes, ideally it should work fine on
> 1.4's as well
>
> Things that should be working from version one:
> - logging compatible with older asterisk's
> - authentication using Voicemail
> -.plug and play on most systems
> - channel states
> - pause/unpause with pause codes
> - ...you tell me
>
> Anybody interested?
> l.
>
>
> 2009/10/30 Mariano Lecuona 
>
>> Hi all,
>>
>> I would like to know if there is any application replacement for the
>> AgentCallBackLogin() from asterisk 1.4 on asterisk 1.6. I know, from what
>> I've read that the call back login agent can be done using a smart dialplan
>> as showed on the docs. But I cannot thinks a flexible dialplan for a dinamic
>> reassignation of agents to different queues every day.
>>
>> Thanks in advance.
>>
>> Mariano
>>
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>
>
>
> --
> Loway - home of QueueMetrics - http://queuemetrics.com
>
>
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Re: [asterisk-users] Asterisk 1.4 and Fax

[Steve Underwood]

On 11/03/2009 04:25 AM, Thomas Kenyon wrote:
> Kevin P. Fleming wrote:
>
>> Lee Howard wrote:
>>  
>>> I've heard of people who go to casinos and come home with a couple
>>> thousand bucks winnings, too.  But the truth is that invariably the vast
>>> majority of people who gamble don't win.
>>>
>>> http://hylafax.sourceforge.net/docs/fax-over-voip.pdf
>>>
>>> Everyone wants to see if they're lucky.  The smart ones, however, don't
>>> trust luck.
>>>
>> FAX over SIP does not necessarily mean FAX over VOIP. FAX over SIP can
>> also describe T.38, which is not as much of a gamble as FAX over VOIP :-)
>>
>>  
> True, although I've yet to find a provider in this country (UK) that
> supports T.38.
>
> He may be better off porting the number to a fax2email service (although
> ime they are worth play testing first before you put any real work on
> them, eg. recently I've found one that doesn't support Fine Print or
> higher res faxes).
>
> AFAICT, to get a (real) fax machine using T.38, you either need to buy
> one that already supports it (never seen one, but I am assured they
> exist), Buy an ATA that supports it, or move to callweaver.
>
>
T.38 FAX machines do exist, although they are rare. A number of high end 
office machines support T.38, or have a T.38 option. There are small FAX 
machines from Sagem which support T.38.

Steve


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Re: [asterisk-users] Asterisk 1.4 and Fax

[Christian Victor]

2009/11/2 Doug Lytle 

> Christian Victor wrote:
> > 2009/11/2 Doug Lytle mailto:supp...@drdos.info>>
> >
> >
> > Faxing over SIP is never a good idea.
> >
> > And why would that be? I think that faxing over SIP using T.38 is a
> > fantastic idea.
>
> As far as I know, T.38 isn't supported under 1.4
>

That would be "Faxing using Asterisk 1.4 is never a good idea". Sorry for
being such a bean counter. ;-)

To stay on-topic: Terminating fax over PSTN works quite well in 1.4 but the
original poster should be warned of trying to terminate fax over a SIP
trunk. Using SIP/G.711 to connect the fax machine to Asterisk over LAN works
quite well in my experience but others had worse results.

Chris
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Re: [asterisk-users] Asterisk as Outbound Proxy ?

[Kevin P. Fleming]

Kristijan Vrban wrote:
> Hello, short question: is there a possibility to use asterisk as an
> outbound proxy? iam open for any suggestions, use asterisk trunk, dirty
> patches, ugly workarounds, everything.
> 
> What is want to build is:
> 
> SIP Phone -> via TLS/SRTP -> Asterisk as outbound proxy -> via UDP/RTP
> -> VoIP-Provider
> 
> So Asterisk should just forward any incoming SIP messages (INVITE,
> REGISTER) to the VoIP-Provider and do SIP TLS-> SIP UDP and SRTP -> RTP
> translation (via *1.6.2 and the SRTP patch)

It is highly unlikely that you'll be able to get Asterisk configured in
a transparent enough fashion to appear as a proxy in this scenario.
You'd be far better off to use an actual proxy, if that's the
functionality you need.

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kpflem...@digium.com
Check us out at www.digium.com & www.asterisk.org

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[asterisk-users] Asterisk as Outbound Proxy ?

[Kristijan Vrban]

Hello, short question: is there a possibility to use asterisk as an outbound
proxy? iam open for any suggestions, use asterisk trunk, dirty patches, ugly
workarounds, everything.

What is want to build is:

SIP Phone -> via TLS/SRTP -> Asterisk as outbound proxy -> via UDP/RTP ->
VoIP-Provider

So Asterisk should just forward any incoming SIP messages (INVITE, REGISTER)
to the VoIP-Provider and do SIP TLS-> SIP UDP and SRTP -> RTP translation
(via *1.6.2 and the SRTP patch)

Kristijan
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Re: [asterisk-users] Asterisk 1.4 and Fax

[Thomas Kenyon]

Dan Journo wrote:
> How do these fax2email providers run their service?
> 
I've not the faintest Idea, the provider I use afaict outsource it.

> Do they all use physical lines rather than use the internet?
> 
> Thanks
> Dan
> 

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Re: [asterisk-users] Anyone seen this before

[David Backeberg]

On Mon, Nov 2, 2009 at 4:26 PM, Robert Grignon  wrote:
> Testing a new gateway and have a Rhino Channel Bank... Sending a test fax
> and everything works fine (Receive the fax fine) But I notice this in the
> log
>
> Google search didn't return much of anything...
>
>
> DAHDI hook failed returned -1 (trying 1): Device or resource busy

I've never used the channel bank, but different types of phones do
different things to signal a hangup, including perhaps temporarily
reversing polarity. Whichever way your card is configured to detect
hangups, try switching the line signaling and see if that fixes your
problems.

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[asterisk-users] Anyone seen this before

[Robert Grignon]

Testing a new gateway and have a Rhino Channel Bank... Sending a test
fax and everything works fine (Receive the fax fine) But I notice this
in the log
 
Google search didn't return much of anything...
 
 
DAHDI hook failed returned -1 (trying 1): Device or resource busy

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Re: [asterisk-users] Tutorial for SIP user

[giancarlo lombardo]

Ciao,
I installed Xlite on Windows Vista, the IP connection (ping)  is working,
shall I check something else ?

Thanks in advance.

2009/11/1 Farooq Hussain 

> Dear Giancarlo,
>
> On which OS your are installing XLITE. If you are trying to connect XLITE
> using Winodws XP please make a entry in your firewall. I think that would
> solve your problem
>
>   On Sun, Nov 1, 2009 at 10:27 AM, giancarlo lombardo <
> gianclomba...@gmail.com> wrote:
>
>>   Dear all,
>> I'm trying to setup a SIP call with XLITE using my asterisk PBX, but I
>> have trouble, I see on XLITE console:
>>
>> Registration Error: 503 - Service unavailable.
>> Someone have a tutorial or a step by step description how to do that ?
>>
>> Thanks in advance
>>
>> --
>> Giancarlo Lombardo
>>
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>
>
>
> --
> Thanks
>
> Farooq Hussain
>
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-- 
Giancarlo Lombardo
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[asterisk-users] DTMF Timing and Fujitsu F9600 Switch

[Will Szopko]

We're in the process of replacing an ancient Centigram-based, but
Fujitsu-labelled, voicemail system with an Asterisk solution. The system
will interface with a Fujitsu F9600 switch and use the SMDI module in
Asterisk 1.6.1.x to communicate the calling information needed to make the
interplay between the two systems work smoothly. This portion of the
project is configured and working fine, as is the programming in
extensions.conf. The SMDI work done by Russell at Digium came out really
well (thanks).

We are having one problem, though. That is, the DTMF interactions are not
working consistently from the Fujitsu to Asterisk (particularly the
sending of extension number and passcode). Sometimes the system will get
just a portion of the extension number or the password and hangup the call
before the entering is finished. Sometimes it will work fine, especially
if one "dials" slowly.

I have to figure that these problems are resulting from a discordance
between the DTMF timing settings on the Fujitsu and those on the Asterisk
box. However, I am unsure of how to figure out what settings to use. I can
certainly view the settings on the Fujitsu, but do not know of how to
"tune" the DAHDI settings on the Asterisk side. The options looked
limited. Or, conversely, what settings does Asterisk expect to see? If I
knew that, I could tune on the Fujitsu side of the equation.

BTW, the telephony hardware we are using is a Digium Wildcard AEX800 Board
1.

Thanks for any help any of you can provide.

- Will


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Re: [asterisk-users] Asterisk 1.4 and Fax

[David Backeberg]

On Mon, Nov 2, 2009 at 3:38 PM, Dan Journo  
wrote:
> How do these fax2email providers run their service?
>
> Do they all use physical lines rather than use the internet?

If you read far enough back in the archives, you'll find somebody who
claimed they used
asterisk-1.4
(I think hylafax)
and voip

But that they did so in a colo, one-hop and almost no RTT away from
their provider. Again, at which point, you're not saving money
compared to an analogue fax over PSTN unless you have a really large
volume, and even then you can often get better bulk pricing for PSTN.

You know your usage and you know your budget.

If you don't have time to fight broken faxes, learn asterisk-1.6, and
provision a voip provider, just stick with analogue fax over PSTN.

My business situation:
channelized DS3, that's 28x 23 voice channels -> Cisco voice routers
-> SIP -> asterisk-1.6 app_fax()

Working very well for us, but I don't know whether your budget or
usage is going to justify something like that. As for what a
commercial service uses, they use whatever was the cheapest wherever
they host their services.

Real modem pools, or real brooktrout modem boards are common. That
would have been a better idea for my situation if I wasn't sharing the
circuits with other voice services.

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Re: [asterisk-users] Asterisk 1.4 and Fax

[Doug Lytle]

David Backeberg wrote:
> On Mon, Nov 2, 2009 at 12:22 PM, Dan Journo 
> mailto:d...@keshercommunications.com>> 
> wrote:asterisk app_fax (which depends on SpanDSP from Lee Howard).

SpanDSP was written by Steve Underwood.

Doug

-- 

Ben Franklin quote:

"Those who would give up Essential Liberty to purchase a little Temporary 
Safety, deserve neither Liberty nor Safety."


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Re: [asterisk-users] Asterisk 1.4 and Fax

[David Backeberg]

On Mon, Nov 2, 2009 at 12:22 PM, Dan Journo 
wrote:

>  Hi,
>
> Does anyone have an up to date guide for setting up fax 2 email with
> asterisk?
>

You can buy this shrink-wrapped from Cisco if you're willing to pay what
they're asking. There are probably other vendors who sell that too.

If you insist on doing this yourself, and using asterisk, start by moving to
1.6. The fax support is night and day better in 1.6 than 1.4, using native
asterisk app_fax (which depends on SpanDSP from Lee Howard).

If you want to go SIP as part of the deployment, I recommend either:
1) terminate PSTN at your premise, and only use SIP internally inside your
PSTN gateway
2) if you're going to go with a SIP provider, tunnel them on a dedicated
circuit so you're not fighting bandwidth limit in addition to the various
problems you'll inevitably face with their implementation of fax over voip.

Once you price #2 you'll probably discover that #1 is cheaper, and I've
already said it's more likely to be reliable when you can control as much of
the voip as possible.
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Re: [asterisk-users] Asterisk 1.4 and Fax

[Dan Journo]

How do these fax2email providers run their service?

Do they all use physical lines rather than use the internet?

Thanks
Dan

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Thomas
Kenyon
Sent: 02 November 2009 20:25
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 1.4 and Fax

Kevin P. Fleming wrote:
> Lee Howard wrote:
>> I've heard of people who go to casinos and come home with a couple 
>> thousand bucks winnings, too.  But the truth is that invariably the
vast 
>> majority of people who gamble don't win.
>>
>> http://hylafax.sourceforge.net/docs/fax-over-voip.pdf
>>
>> Everyone wants to see if they're lucky.  The smart ones, however,
don't 
>> trust luck.
> 
> FAX over SIP does not necessarily mean FAX over VOIP. FAX over SIP can
> also describe T.38, which is not as much of a gamble as FAX over VOIP
:-)
> 
True, although I've yet to find a provider in this country (UK) that 
supports T.38.

He may be better off porting the number to a fax2email service (although

ime they are worth play testing first before you put any real work on 
them, eg. recently I've found one that doesn't support Fine Print or 
higher res faxes).

AFAICT, to get a (real) fax machine using T.38, you either need to buy 
one that already supports it (never seen one, but I am assured they 
exist), Buy an ATA that supports it, or move to callweaver.

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Re: [asterisk-users] Asterisk 1.4 and Fax

[Thomas Kenyon]

Kevin P. Fleming wrote:
> Lee Howard wrote:
>> I've heard of people who go to casinos and come home with a couple 
>> thousand bucks winnings, too.  But the truth is that invariably the vast 
>> majority of people who gamble don't win.
>>
>> http://hylafax.sourceforge.net/docs/fax-over-voip.pdf
>>
>> Everyone wants to see if they're lucky.  The smart ones, however, don't 
>> trust luck.
> 
> FAX over SIP does not necessarily mean FAX over VOIP. FAX over SIP can
> also describe T.38, which is not as much of a gamble as FAX over VOIP :-)
> 
True, although I've yet to find a provider in this country (UK) that 
supports T.38.

He may be better off porting the number to a fax2email service (although 
ime they are worth play testing first before you put any real work on 
them, eg. recently I've found one that doesn't support Fine Print or 
higher res faxes).

AFAICT, to get a (real) fax machine using T.38, you either need to buy 
one that already supports it (never seen one, but I am assured they 
exist), Buy an ATA that supports it, or move to callweaver.

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Re: [asterisk-users] Asterisk 1.4 and Fax

[Doug Lytle]

Christian Victor wrote:
> 2009/11/2 Doug Lytle mailto:supp...@drdos.info>>
>
>
> Faxing over SIP is never a good idea.
>
> And why would that be? I think that faxing over SIP using T.38 is a 
> fantastic idea.

As far as I know, T.38 isn't supported under 1.4

Doug



-- 

Ben Franklin quote:

"Those who would give up Essential Liberty to purchase a little Temporary 
Safety, deserve neither Liberty nor Safety."


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Re: [asterisk-users] Asterisk 1.4 and Fax

[Kevin P. Fleming]

Lee Howard wrote:

>> FAX over SIP does not necessarily mean FAX over VOIP. FAX over SIP can
>> also describe T.38, which is not as much of a gamble as FAX over VOIP :-)
> 
> Does Asterisk 1.4 support T.38?

Only for passthrough between SIP channels; Asterisk 1.6.0 and later also
support T.38 termination and origination.

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kpflem...@digium.com
Check us out at www.digium.com & www.asterisk.org

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Re: [asterisk-users] Asterisk 1.4 and Fax

[Lee Howard]

Kevin P. Fleming wrote:
> Lee Howard wrote:
>   
>> I've heard of people who go to casinos and come home with a couple 
>> thousand bucks winnings, too.  But the truth is that invariably the vast 
>> majority of people who gamble don't win.
>>
>> http://hylafax.sourceforge.net/docs/fax-over-voip.pdf
>>
>> Everyone wants to see if they're lucky.  The smart ones, however, don't 
>> trust luck.
>> 
>
> FAX over SIP does not necessarily mean FAX over VOIP. FAX over SIP can
> also describe T.38, which is not as much of a gamble as FAX over VOIP :-)

Does Asterisk 1.4 support T.38?

Thanks,

Lee.

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Re: [asterisk-users] Asterisk 1.4 and Fax

[Kevin P. Fleming]

Lee Howard wrote:
> I've heard of people who go to casinos and come home with a couple 
> thousand bucks winnings, too.  But the truth is that invariably the vast 
> majority of people who gamble don't win.
> 
> http://hylafax.sourceforge.net/docs/fax-over-voip.pdf
> 
> Everyone wants to see if they're lucky.  The smart ones, however, don't 
> trust luck.

FAX over SIP does not necessarily mean FAX over VOIP. FAX over SIP can
also describe T.38, which is not as much of a gamble as FAX over VOIP :-)

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kpflem...@digium.com
Check us out at www.digium.com & www.asterisk.org

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[asterisk-users] Unexpected control subclass '-1'

[Carlos Chavez]

I have been getting the following message every time I make a call for
the past few months:

[Nov  2 13:08:18] WARNING[9859]: file.c:1273 waitstream_core: Unexpected
control subclass '-1'

Everything seems to be working so I do not know if this is important.
I am using Asterisk 1.4.26.1 (upgrading today to .2) with Asterisk
Addons 1.4.9, Zaptel 1.4.12.1, Libpri 1.4.10 (upgrading to 1.4.10.1
today).  What does this message mean?

-- 
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez Prats
Director de Tecnología
+52-55-91169161 ext 2001


signature.asc
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Re: [asterisk-users] Asterisk 1.4 and Fax

[Christian Victor]

2009/11/2 Doug Lytle 

> Dan Journo wrote:
> >
> > I need to get it up and running before we can put in the order to
> > transfer the fixed line number over to SIP.
> >
>
> Faxing over SIP is never a good idea.
>

And why would that be? I think that faxing over SIP using T.38 is a
fantastic idea.

Chris
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Re: [asterisk-users] Asterisk 1.4 and Fax

[Lee Howard]

I've heard of people who go to casinos and come home with a couple 
thousand bucks winnings, too.  But the truth is that invariably the vast 
majority of people who gamble don't win.

http://hylafax.sourceforge.net/docs/fax-over-voip.pdf

Everyone wants to see if they're lucky.  The smart ones, however, don't 
trust luck.

Lee.


Dan Journo wrote:
> I've heard mixed reports.
>
> Some say they've had no problems, some say that faxes fail most of the
> time.
>
> I want to try it out and see how it goes.
>
>
>
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle
> Sent: 02 November 2009 18:34
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Asterisk 1.4 and Fax
>
> Dan Journo wrote:
>   
>> I need to get it up and running before we can put in the order to
>> transfer the fixed line number over to SIP.
>>
>> 
>
> Faxing over SIP is never a good idea.
>
> Doug
>
>   


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Re: [asterisk-users] Asterisk 1.4 and Fax

[Dan Journo]

I've heard mixed reports.

Some say they've had no problems, some say that faxes fail most of the
time.

I want to try it out and see how it goes.




-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle
Sent: 02 November 2009 18:34
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 1.4 and Fax

Dan Journo wrote:
>
> I need to get it up and running before we can put in the order to
> transfer the fixed line number over to SIP.
>

Faxing over SIP is never a good idea.

Doug

-- 

Ben Franklin quote:

"Those who would give up Essential Liberty to purchase a little
Temporary Safety, deserve neither Liberty nor Safety."


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Re: [asterisk-users] Cannot make calls

[Cliconnect]

Thank you ,

I did that 
sip set debug ip 198.163.0.103
SIP Debugging Enabled for IP: 198.163.0.103

I checked the /var/log/asterisk files and there is no information
there. 

Could you please inform where am I suppose to see the debug information
?

tks

Jair



Ott Rose wrote:

  you
can get debug info a couple of ways from the asterisk CLI. I like this
command the best. sip set debug ip xxx.xxx.xx.xxx where xxx.xxx.xxx.xxx
is the of the x-lite phone. It will give you a lot of info. I haven't
figured out how to redirect output yet.
  
  Date: Fri, 30 Oct 2009 13:05:35 -0700
From: cliconn...@cliconnect.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Cannot make calls
  
  
No change on it.
  
 Do I have to enter a command ?
  
I have changed the port to 5060 in both clients. Still the same problem.
  
thanks
  
Jair Santos
  
Warren Selby wrote:
  You're
attempting to connect on ports 5061-5062 but are
bound to port 5060...?

What does your CLI look like during a failed call attempt?

Thanks,
--Warren Selby

On Fri, Oct 30, 2009 at 2:18 PM,
Cliconnect 
wrote:

   
  
  
Thank you,
  
  
>>How are you setting up xlite and the ata?
  
  
Xlite
  
User name : 1000
Domain: IP of the server running Asterisk
Register with domain and receive incoming calls: clear
Port used in local computer : manually specify range : 5061-5062
  
ATA
SIP server address: IP of the server running Asterisk
Outbond Proxy : IP of the server running Asterisk
SIP User id : 1001
Accoount ID : 1001
Use DNS SRV : yes
User id is phone number : yes
SIP registration : no
Local sip port : 5062
  
  
>>Which version of Asterisk are you using?
  
Asterisk 1.6.1.6, Copyright (C) 1999 - 2009 Digium,
  
  
>> What does the general section of your sip.conf look like?
  
  
[general]
context=default
allowoverlap=no
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
  
When I 
  
sip show peers
  
Name/username  Host    Dyn Nat ACL Port
Status 
1000   (Unspecified)    D  5060
Unmonitored 
1001   (Unspecified)    D  5060
Unmonitored 
2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 0
offline]
  
regards
  
Jair Santos
  
  
  
   
  
  
  
Warren Selby wrote:
  

How are you setting up xlite and the ata? 
Which
version
of Asterisk are you using?  What does the general section of your
sip.conf look like?

On Fri, Oct 30, 2009 at 1:01 PM,
Cliconnect 
wrote:

  Hi all,
  
I can only get a line signal when  I set the phones to not register
with domain . 
  
All phones are in the same NAT and I cannot make calls.
  
I am getting "Call failed : Proxy Authentication Required" in Xlite
and  a busy signal when using an ATA.
  
Here is my extensions.conf
[internal]
exten => 1000,1,Verbose(1|Extension 1000)
;exten => 1000,n,Echo()
;exten => 1000,n,Hangup()
exten => 1000,n,Dial(SIP/1000,30)
exten => 1000,n,Hangup()
  
exten => 1001,1,Verbose(1|Extension 1001)
exten => 1001,n,Dial(SIP/1001,30)
exten => 1001,n,Hangup()
  
[phones]
include => internal
  
  
and sip.conf
[1000]
type=friend
context=phones
host=dynamic
[1001]
type=friend
context=phones
host=dynamic
  
  
I am not setting a password .
  
Any help will be appreciated.
  
TIA
  
Jair Santos
  -- 
  
  
  
  
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No virus found in this incoming message.
Checked by AVG - www.avg.com 
Version: 9.0.698 / Virus Database: 270.14.39/2469 - Release Date: 10/30/09 00:52:00

  
  
  
  
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No virus found in this incoming message.
Checked

Re: [asterisk-users] DAHDI/ZAP overlap dialing

[Vieri]

--- On Mon, 11/2/09, Martin  wrote:

> I'd double check that you really have overlapdial=yes for
> those
> channels ... it should be declared
> before channel => keyword in
> zapata.conf/chan_dahdi.conf

I declared overlapdial in zapata.conf:

switchtype = euroisdn
signalling = pri_cpe
overlapdial=yes
context=from-alcatel-custom
group = 1
callgroup = 1
pickupgroup = 1
immediate=no
echocancel=yes
echotraining=yes
echocancelwhenbridged=yes
rxgain=2.0
txgain=1.0
busydetect=no
facilityenable = yes
; pritransfer = ect ; either no, ect, or hangup
channel => 1-15,17-31

Will try to change libpri versions or move to another 1.4 * server.

Thanks for your time.

Vieri



  

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Re: [asterisk-users] Asterisk 1.4 and Fax

[Doug Lytle]

Dan Journo wrote:
>
> I need to get it up and running before we can put in the order to
> transfer the fixed line number over to SIP.
>

Faxing over SIP is never a good idea.

Doug

-- 

Ben Franklin quote:

"Those who would give up Essential Liberty to purchase a little Temporary 
Safety, deserve neither Liberty nor Safety."


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[asterisk-users] Execute Macro AFTER connecting to a channel

[Joseph]

Is the a way to execute the macro AFTER connecting to the channel:

M(x[^arg]) - Execute the Macro for the *called* channel before connecting
to the calling channel.

doesn't work for me as I need to listen to the macro progress as it is sending 
DTMF tone and respond from the connected channel.

-- 
Joseph

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Re: [asterisk-users] Remote IP Phone's

[Tilghman Lesher]

On Monday 02 November 2009 11:46:49 am Steve Edwards wrote:
> Un-top-posting...
>
> > On Mon, 2 Nov 2009, Danny Nicholas wrote:
> >> You can restrict on IP address, MAC address and port type and that's
> >> just what I know.  If someone want's through bad enough you're going to
> >> have a problem, but you can at least slow down or stop casual hackers.
> >
> > Aren't MAC addresses only available to devices on the local network?
> >
> > Don't you mean "port number" instead of "port type?"
>
> On Mon, 2 Nov 2009, Danny Nicholas wrote:
> > Actually, both.  You can (AFAIK) specify 5060, 1 etc and UDP/TCP,
> > etc. Of course, I have been wrong at least once before :)
>
> I thought SIP on Asterisk was still limited to UDP. I must have missed the
> memo...

No, you probably missed that some people are using 1.6, in which TCP and TLS
are now available for SIP.  If you aren't using 1.6, then TCP and TLS are not
available options for SIP.

-- 
Tilghman

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Re: [asterisk-users] Asterisk 1.4 and Fax

[Dan Journo]

Sorry Steve,

Forgot to remove it before sending the email.


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve
Howes
Sent: 02 November 2009 17:35
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 1.4 and Fax

On 2 Nov 2009, at 17:22, Dan Journo wrote:
> Does anyone have an up to date guide for setting up fax 2 email with  
> asterisk?

So you can fax them obnoxiously long signatures?

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Re: [asterisk-users] Asterisk 1.4 and Fax

[Dan Journo]

We want to disconnect our PSTN fax line and transfer the number over to
our asterisk server.

I need to get it up and running before we can put in the order to
transfer the fixed line number over to SIP.

Thanks
Dan

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve
Howes
Sent: 02 November 2009 17:35
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 1.4 and Fax

On 2 Nov 2009, at 17:22, Dan Journo wrote:
> Does anyone have an up to date guide for setting up fax 2 email with  
> asterisk?

So you can fax them obnoxiously long signatures?

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Re: [asterisk-users] Remote IP Phone's

[Steve Edwards]

Un-top-posting...

> On Mon, 2 Nov 2009, Danny Nicholas wrote:
>
>> You can restrict on IP address, MAC address and port type and that's 
>> just what I know.  If someone want's through bad enough you're going to 
>> have a problem, but you can at least slow down or stop casual hackers.
>
> Aren't MAC addresses only available to devices on the local network?
>
> Don't you mean "port number" instead of "port type?"

On Mon, 2 Nov 2009, Danny Nicholas wrote:

> Actually, both.  You can (AFAIK) specify 5060, 1 etc and UDP/TCP, 
> etc. Of course, I have been wrong at least once before :)

I thought SIP on Asterisk was still limited to UDP. I must have missed the 
memo...

-- 
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000

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Re: [asterisk-users] Asterisk 1.4 and Fax

[Steve Howes]

On 2 Nov 2009, at 17:22, Dan Journo wrote:
> Does anyone have an up to date guide for setting up fax 2 email with  
> asterisk?

So you can fax them obnoxiously long signatures?

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Re: [asterisk-users] Remote Party ID

[Tzafrir Cohen]

On Mon, Nov 02, 2009 at 03:18:03PM -, Dan Journo wrote:
> 
> Does anyone know how to set the remote party id?

I guess someone does.

If you provide more details you'll have better chances of getting a good
answer.

Version of Asterisk?

Why do you want to set the remote party ID?

How have you tried setting it so far?

Reading the following may give you some hints:

  http://www.catb.org/~esr/faqs/smart-questions.html

-- 
   Tzafrir Cohen
icq#16849755  jabber:tzafrir.co...@xorcom.com
+972-50-7952406   mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

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Re: [asterisk-users] Remote IP Phone's

[Danny Nicholas]

Actually, both.  You can (AFAIK) specify 5060, 1 etc and UDP/TCP, etc.
Of course, I have been wrong at least once before :)

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: Monday, November 02, 2009 11:22 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Remote IP Phone's

Un-top-posting...

> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Connor 
> Spiess
>
> I am wondering what people are doing for security when registering IP 
> phone's remotely if you do not have the equipment to do a VPN tunnel at 
> the remote site. The phone I would be working with mainly is the Polycom 
> lineup.

On Mon, 2 Nov 2009, Danny Nicholas wrote:

> You can restrict on IP address, MAC address and port type and that's 
> just what I know.  If someone want's through bad enough you're going to 
> have a problem, but you can at least slow down or stop casual hackers.

Aren't MAC addresses only available to devices on the local network?

Don't you mean "port number" instead of "port type?"

-- 
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000

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Re: [asterisk-users] Remote IP Phone's

[Steve Edwards]

Un-top-posting...

> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Connor 
> Spiess
>
> I am wondering what people are doing for security when registering IP 
> phone's remotely if you do not have the equipment to do a VPN tunnel at 
> the remote site. The phone I would be working with mainly is the Polycom 
> lineup.

On Mon, 2 Nov 2009, Danny Nicholas wrote:

> You can restrict on IP address, MAC address and port type and that's 
> just what I know.  If someone want's through bad enough you're going to 
> have a problem, but you can at least slow down or stop casual hackers.

Aren't MAC addresses only available to devices on the local network?

Don't you mean "port number" instead of "port type?"

-- 
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000

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Re: [asterisk-users] DAHDI/ZAP overlap dialing

[Martin]

I can only tell you that it worked before...

> Nov  2 17:22:28 VERBOSE[11329] logger.c: -- Extension '1004' in context 
> 'from-pstn-deviate-custom' from '7034' does not

when you have overlapdial turned on it should have checked if there's
a potential matching extension
which you have it right there and asterisk should have sent SETUP_ACK
message back.

if you won't find the solution for this I might fix that as a bounty
if you're interested

I'd double check that you really have overlapdial=yes for those
channels ... it should be declared
before channel => keyword in zapata.conf/chan_dahdi.conf

Martin

On Mon, Nov 2, 2009 at 10:28 AM, Vieri  wrote:
>
> --- On Sat, 10/31/09, Martin  wrote:
>
>> On Sat, Oct 31, 2009 at 5:27 AM,
>> Tzafrir Cohen 
>> wrote:
>> > I'm not sure if handling of overlap hasn't changed
>> since.
>> >
>> > But can you provide a trace of how Asterisk sees
>> things? e.g. 'pri
>> > intense debug span 1'
>> >
>>
>> the intense debug is overkill we only need messages of
>> layer 3 ...
>> just do "pri debug span 1"
>>
>> Martin
>
> Here's the pri trace:
>
> Nov  2 17:22:28 VERBOSE[11329] logger.c: < Protocol Discriminator: Q.931 (8)  
> len=38
> Nov  2 17:22:28 VERBOSE[11329] logger.c: < Call Ref: len= 2 (reference 
> 16976/0x4250) (Originator)
> Nov  2 17:22:28 VERBOSE[11329] logger.c: < Message type: SETUP (5)
> Nov  2 17:22:28 VERBOSE[11329] logger.c: < [04 03 80 90 a3]
> Nov  2 17:22:28 VERBOSE[11329] logger.c: < Bearer Capability (len= 5) [ Ext: 
> 1  Q.931 Std: 0  Info transfer capability: Speech (0)
> Nov  2 17:22:28 VERBOSE[11329] logger.c: <                              Ext: 
> 1  Trans mode/rate: 64kbps, circuit-mode (16)
> Nov  2 17:22:28 VERBOSE[11329] logger.c: <                                
> User information layer 1: A-Law (35)
> Nov  2 17:22:28 VERBOSE[11329] logger.c: < [18 03 a9 83 8b]
> Nov  2 17:22:28 VERBOSE[11329] logger.c: < Channel ID (len= 5) [ Ext: 1  
> IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0
> Nov  2 17:22:28 VERBOSE[11329] logger.c: <                        ChanSel: As 
> indicated in following octets
> Nov  2 17:22:28 VERBOSE[11329] logger.c: <                       Ext: 1  
> Coding: 0   Number Specified   Channel Type: 3
> Nov  2 17:22:28 VERBOSE[11329] logger.c: <                       Ext: 1  
> Channel: 11 ]
> Nov  2 17:22:28 VERBOSE[11329] logger.c: < [1e 02 80 83]
> Nov  2 17:22:28 VERBOSE[11329] logger.c: < Progress Indicator (len= 4) [ Ext: 
> 1  Coding: CCITT (ITU) standard (0) 0: 0   Location: User (0)
> Nov  2 17:22:28 VERBOSE[11329] logger.c: <                               Ext: 
> 1  Progress Description: Calling equipment is non-ISDN. (3) ]
> Nov  2 17:22:28 VERBOSE[11329] logger.c: < [6c 06 00 81 37 30 33 34]
> Nov  2 17:22:28 VERBOSE[11329] logger.c: < Calling Number (len= 8) [ Ext: 0  
> TON: Unknown Number Type (0)  NPI: Unknown Number Plan (0)
> Nov  2 17:22:28 VERBOSE[11329] logger.c: <                           
> Presentation: Presentation permitted, user number passed network screening 
> (1) '7034' ]
> Nov  2 17:22:28 VERBOSE[11329] logger.c: < [70 05 80 31 30 30 34]
> Nov  2 17:22:28 VERBOSE[11329] logger.c: < Called Number (len= 7) [ Ext: 1  
> TON: Unknown Number Type (0)  NPI: Unknown Number Plan (0) '1004' ]
> Nov  2 17:22:28 VERBOSE[11329] logger.c: < [7d 02 91 81]
> Nov  2 17:22:28 VERBOSE[11329] logger.c: < IE: High-layer Compatibility (len 
> = 4)
> Nov  2 17:22:28 VERBOSE[11329] logger.c: -- Making new call for cr 16976
> Nov  2 17:22:28 VERBOSE[11329] logger.c: -- Processing Q.931 Call Setup
> Nov  2 17:22:28 VERBOSE[11329] logger.c: -- Processing IE 4 (cs0, Bearer 
> Capability)
> Nov  2 17:22:28 VERBOSE[11329] logger.c: -- Processing IE 24 (cs0, Channel 
> Identification)
> Nov  2 17:22:28 VERBOSE[11329] logger.c: -- Processing IE 30 (cs0, Progress 
> Indicator)
> Nov  2 17:22:28 VERBOSE[11329] logger.c: -- Processing IE 108 (cs0, Calling 
> Party Number)
> Nov  2 17:22:28 VERBOSE[11329] logger.c: -- Processing IE 112 (cs0, Called 
> Party Number)
> Nov  2 17:22:28 VERBOSE[11329] logger.c: -- Processing IE 125 (cs0, 
> High-layer Compatibility)
> Nov  2 17:22:28 VERBOSE[11329] logger.c:     -- Extension '1004' in context 
> 'from-pstn-deviate-custom' from '7034' does not exist.  Rejecting call on 
> channel 1/11, span 1
> Nov  2 17:22:28 VERBOSE[11329] logger.c: NEW_HANGUP DEBUG: Calling 
> q931_hangup, ourstate Call Present, peerstate Call Initiated
> Nov  2 17:22:28 VERBOSE[11329] logger.c: > Protocol Discriminator: Q.931 (8)  
> len=9
> Nov  2 17:22:28 VERBOSE[11329] logger.c: > Call Ref: len= 2 (reference 
> 16976/0x4250) (Terminator)
> Nov  2 17:22:28 VERBOSE[11329] logger.c: > Message type: RELEASE COMPLETE (90)
> Nov  2 17:22:28 VERBOSE[11329] logger.c: > [08 02 81 81]
> Nov  2 17:22:28 VERBOSE[11329] logger.c: > Cause (len= 4) [ Ext: 1  Coding: 
> CCITT (ITU) standard (0) 0: 0   Location: Private network serving the local 
> user (1)
> Nov  2 17:22:28 VERBOSE[11329] logger.c: >                 

[asterisk-users] Asterisk 1.4 and Fax

[Dan Journo]

Hi,

 

Does anyone have an up to date guide for setting up fax 2 email with asterisk?

 

Thanks

Dan

 

 

 

 



 

IT Maintenance Contract Clients can now access our Instant Chat Service to 
receive immediate remote IT support. Click the chat link below for support.
For more information on receiving IT support from £150 per month, please 
contact Kesher Communications.



 

 

Dan Journo
IT Business Consultant
Kesher Communications Ltd

Tel: 07957 233 599
Web: http://www.KesherCommunications.com  
Live Chat/Instant Support: Click Here 

  

This email and any files transmitted with it are confidential and intended 
solely for the recipient(s). If you are not the named addressee you should not 
disseminate, copy or alter this email. Under no circumstances may this email be 
distributed without written permission from the sender. Warning: Although the 
Company has taken reasonable precautions to ensure no viruses are present in 
this email, the company cannot accept responsibility for any loss or damage 
arising from the use of this email or attachments. All prices exclude VAT 
unless otherwise stated. No responsibility is taken for any recommendations 
made by a sender or by Kesher Communications Ltd. Recipient(s) takes 
responsibility for any actions taken as a result of advice and recommendations 
given by Kesher Communications Ltd.

 

 

 

 

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Re: [asterisk-users] Real replacement for AgentCallBackLogin() on Asterisk 1.6

[Jim Dickenson]

I have sent this before but here is how I do agent login and queue:

;  Agent login logout 
exten => *20,1,Verbose(2,Doing agent login/logout)
exten => *20,n,Answer()
exten => *20,n,wait(.0.5)
exten => *20,n,Read(AgentNumber,agent-user)
exten => *20,n,Set(UserID=${DB(ExtenToUser/${AgentNumber})})
exten => *20,n,GotoIf($["${UserID}"=""]?NOUSER)
exten => *20,n,Set(AgentStatus=${DB(users/${UserID}/AgentStatus)})
exten => *20,n,GotoIf($["${AgentStatus}"="1"]?VERIFY)
exten => *20,n,GotoIf($["${AgentStatus}"="2"]?VERIFY)
exten => *20,n(NOUSER),Playback(cfmc/bad-agent)
exten => *20,n,Playback(vm-goodbye)
exten => *20,n,Hangup()
exten => *20,n(VERIFY),VMAuthenticate(${agentnumb...@ourvm)
exten => *20,n,GotoIf($["${AgentStatus}"="2"]?AGENTOFF)
exten => *20,n,Set(DB(users/${UserID}/AgentStatus)=2)
exten => *20,n,Set(DB(users/${UserID}/AgentDevice)=${CUT(CHANNEL,-,1)})
exten => *20,n,AddQueueMember(support,Local/Queue${AgentNumber} 
@ansqueue${C$

;   AQMSTATUS can be  ADDED | MEMBERALREADY | NOSUCHQUEUE
exten => *20,n,Playback(agent-loginok)
exten => *20,n,Verbose(2,Agent ${AgentNumber} added ${DB(users/$ 
{UserID}/AgentD$

exten => *20,n,Hangup()
exten => *20,n(AGENTOFF),Set(DB(users/${UserID}/AgentStatus)=1)
exten => *20,n,Set(OldVal=${DB_DELETE(users/${UserID}/AgentDevice)})
exten => *20,n,RemoveQueueMember(support,Local/Queue${AgentNumber} 
@ansqueue)

exten => *20,n,Playback(agent-loggedoff)
exten => *20,n,Verbose(2,Agent ${AgentNumber} removed)
exten => *20,n,Hangup()


exten => 201,1,Verbose(2,Doing support call)
exten => 201,n,Answer()
exten => 201,n,Wait(0.5)
;exten => 201,n,Set(qac=${QUEUE_MEMBER(support,free)})
exten => 201,n,Set(qac=${QUEUE_MEMBER_COUNT(support)})
exten => 201,n,GotoIf($[${qac} > 0]?HAVEAGNT)
exten => 201,n,Verbose(2,No agents free in support queue)
exten => 201,n,Playback(cfmc/support-no-agent)
exten => 201,n,Voicemail(2...@ourvm,u)
exten => 201,n,Playback(goodbye)
exten => 201,n,Hangup()
exten => 201,n(HAVEAGNT),Playback(cfmc/support-intro)
exten => 201,n,Verbose(2,Queuing caller for support agent)
exten => 201,n,Queue(support,nrt,,,120)
exten => 201,n,Verbose(2,Support agent did not answer call)
exten => 201,n,Voicemail(2...@ourvm,b)
exten => 201,n,Playback(goodbye)
exten => 201,n,Hangup()


[ansqueue]
exten => _Queue.,1,Set(AgentNumber=${EXTEN:5})
exten => _Queue.,n,Set(UserID=${DB(ExtenToUser/${AgentNumber})})
exten => _Queue.,n,Set(AgentDevice=${DB(users/${UserID}/AgentDevice)})
;exten => _Queue.,n,Verbose(2,Agent ${AgentNumber} status is $ 
{DEVICE_STATE(${A$
exten => _Queue.,n,Verbose(2,Agent ${AgentNumber} status is ${DEVSTATE 
(${AgentD$
;exten => _Queue.,n,GotoIf($["${DEVICE_STATE($ 
{AgentDevice})}"="NOT_INUSE"]?DIA$
exten => _Queue.,n,GotoIf($["${DEVSTATE(${AgentDevice})}"="NOT_INUSE"]? 
DIALIT)

exten => _Queue.,n,Busy()
exten => _Queue.,n,Hangup()
exten => _Queue.,n(DIALIT),Dial(${AgentDevice},,g)
exten => _Queue.,n,Hangup()

--
Jim Dickenson
mailto:dicken...@cfmc.com

CfMC
http://www.cfmc.com/



On Nov 2, 2009, at 8:44 AM, Lenz Emilitri wrote:

We were thinking about doing something similar as well. A lot of  
people are asking for this. If there is anybody else interested, we  
could share the load


I was thinking about creating a context like @agents, so that when  
you do the log-on you basically add Local/1...@agents as a member of  
the queue. When you ring it, it basically looks up for the actual  
device in AstDB and dials it like:


Queue -> (member) Local/1...@agents -> (astdb) SIP/234

I think that we should be able to forward channel state as well  
(using hints? I've never done it)  so that app_queue does not try  
dialling agents that are busy.


I was thinking about storing queue-agent associations into config  
strings, and/or AstDB, and/or http over curl. And yes, ideally it  
should work fine on 1.4's as well


Things that should be working from version one:
- logging compatible with older asterisk's
- authentication using Voicemail
-.plug and play on most systems
- channel states
- pause/unpause with pause codes
- ...you tell me

Anybody interested?
l.


2009/10/30 Mariano Lecuona 
Hi all,

I would like to know if there is any application replacement for the  
AgentCallBackLogin() from asterisk 1.4 on asterisk 1.6. I know, from  
what I've read that the call back login agent can be done using a  
smart dialplan as showed on the docs. But I cannot thinks a flexible  
dialplan for a dinamic reassignation of agents to different queues  
every day.


Thanks in advance.

Mariano

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Re: [asterisk-users] Real replacement for AgentCallBackLogin() on Asterisk 1.6

[Lenz Emilitri]

To avoid boring everybody else to death with the discussion, I created a
mailing list for that on Google Groups - see  http://tinyurl.com/yjtf62s
Thanks
l.


2009/11/2 Lenz Emilitri 

> We were thinking about doing something similar as well. A lot of people are
> asking for this. If there is anybody else interested, we could share the
> load
>
> I was thinking about creating a context like @agents, so that when you do
> the log-on you basically add Local/1...@agents as a member of the queue.
> When you ring it, it basically looks up for the actual device in AstDB and
> dials it like:
>
> Queue -> (member) Local/1...@agents -> (astdb) SIP/234
>
> I think that we should be able to forward channel state as well (using
> hints? I've never done it)  so that app_queue does not try dialling agents
> that are busy.
>
> I was thinking about storing queue-agent associations into config strings,
> and/or AstDB, and/or http over curl. And yes, ideally it should work fine on
> 1.4's as well
>
> Things that should be working from version one:
> - logging compatible with older asterisk's
> - authentication using Voicemail
> -.plug and play on most systems
> - channel states
> - pause/unpause with pause codes
> - ...you tell me
>
> Anybody interested?
> l.
>
>
> 2009/10/30 Mariano Lecuona 
>
>> Hi all,
>>
>> I would like to know if there is any application replacement for the
>> AgentCallBackLogin() from asterisk 1.4 on asterisk 1.6. I know, from what
>> I've read that the call back login agent can be done using a smart dialplan
>> as showed on the docs. But I cannot thinks a flexible dialplan for a dinamic
>> reassignation of agents to different queues every day.
>>
>> Thanks in advance.
>>
>> Mariano
>>
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>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
>
> --
> Loway - home of QueueMetrics - http://queuemetrics.com
>
>


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Re: [asterisk-users] Real replacement for AgentCallBackLogin() on Asterisk 1.6

[Lenz Emilitri]

We were thinking about doing something similar as well. A lot of people are
asking for this. If there is anybody else interested, we could share the
load

I was thinking about creating a context like @agents, so that when you do
the log-on you basically add Local/1...@agents as a member of the queue. When
you ring it, it basically looks up for the actual device in AstDB and dials
it like:

Queue -> (member) Local/1...@agents -> (astdb) SIP/234

I think that we should be able to forward channel state as well (using
hints? I've never done it)  so that app_queue does not try dialling agents
that are busy.

I was thinking about storing queue-agent associations into config strings,
and/or AstDB, and/or http over curl. And yes, ideally it should work fine on
1.4's as well

Things that should be working from version one:
- logging compatible with older asterisk's
- authentication using Voicemail
-.plug and play on most systems
- channel states
- pause/unpause with pause codes
- ...you tell me

Anybody interested?
l.


2009/10/30 Mariano Lecuona 

> Hi all,
>
> I would like to know if there is any application replacement for the
> AgentCallBackLogin() from asterisk 1.4 on asterisk 1.6. I know, from what
> I've read that the call back login agent can be done using a smart dialplan
> as showed on the docs. But I cannot thinks a flexible dialplan for a dinamic
> reassignation of agents to different queues every day.
>
> Thanks in advance.
>
> Mariano
>
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>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>



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Re: [asterisk-users] Dialstatus

[Joseph]

On 11/02/09 07:28, Steve Edwards wrote:
>On Mon, 2 Nov 2009, Patrick Plattes wrote:
>
>> you can do print the dialstatus to the console e.g.:
>> exten => s,n,NoOp(${DIALSTATUS})
>>
>> More info:
>> http://www.voip-info.org/wiki/view/Asterisk+cmd+NoOp
>
>A "better practice" would be to use verbose() -- an application with
>greater functionality written specifically for this purpose.
>
>--
>Thanks in advance,
>-
>Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
>Newline  Fax: +1-760-731-3000

What was looking to do something in macro after the channel gets connected 
"dialstatus=Answer" but it doesn't work.

Running the macro I don't hear anything (only a dial tone) until the macro is 
finished.

-- 
Joseph

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[asterisk-users] Nagios check_asterisk_peers needs rights to question the Asterisk-server

[jonas kellens]

When executing the following command :

[r...@nagios ~]# /usr/local/nagios/libexec/check_nrpe -H ip_address -c
check_asterisk_peers

I get the following output :

NRPE: Unable to read output

Somewhere Nagios does not have enough rights to question Asterisk about
the sip peers.

These are the rights of the check_nrpe on the Nagios Server :

[r...@nagios ~]# ls -l /usr/local/nagios/libexec/check_nrpe
-rwxrwxr-x 1 nagios nagios 58017 Oct 31
11:40 /usr/local/nagios/libexec/check_nrpe

These are the rights of the plugin that questions about the SIP-peers on
the Asterisk-server :

bash-3.2# ls -l /usr/local/nagios/libexec/nagisk.pl
-rwxr-x--- 1 nagios nagios 4163 Nov  2
17:12 /usr/local/nagios/libexec/nagisk.pl

The NRPE-plugin on the Asterisk-server is part of the Xinetd-proces.

Asterisk himself is currently running as the root-user.

Question : I'm confused about which proces/plugin I need to give more
rights so that the Asterisk-server can be questioned about its
sip-peers. 

Jonas.
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Re: [asterisk-users] DAHDI/ZAP overlap dialing

[Vieri]

--- On Sat, 10/31/09, Martin  wrote:

> On Sat, Oct 31, 2009 at 5:27 AM,
> Tzafrir Cohen 
> wrote:
> > I'm not sure if handling of overlap hasn't changed
> since.
> >
> > But can you provide a trace of how Asterisk sees
> things? e.g. 'pri
> > intense debug span 1'
> >
> 
> the intense debug is overkill we only need messages of
> layer 3 ...
> just do "pri debug span 1"
> 
> Martin

Here's the pri trace:

Nov  2 17:22:28 VERBOSE[11329] logger.c: < Protocol Discriminator: Q.931 (8)  
len=38
Nov  2 17:22:28 VERBOSE[11329] logger.c: < Call Ref: len= 2 (reference 
16976/0x4250) (Originator)
Nov  2 17:22:28 VERBOSE[11329] logger.c: < Message type: SETUP (5)
Nov  2 17:22:28 VERBOSE[11329] logger.c: < [04 03 80 90 a3]
Nov  2 17:22:28 VERBOSE[11329] logger.c: < Bearer Capability (len= 5) [ Ext: 1  
Q.931 Std: 0  Info transfer capability: Speech (0)
Nov  2 17:22:28 VERBOSE[11329] logger.c: <  Ext: 1  
Trans mode/rate: 64kbps, circuit-mode (16)
Nov  2 17:22:28 VERBOSE[11329] logger.c:  Protocol Discriminator: Q.931 (8)  
len=9
Nov  2 17:22:28 VERBOSE[11329] logger.c: > Call Ref: len= 2 (reference 
16976/0x4250) (Terminator)
Nov  2 17:22:28 VERBOSE[11329] logger.c: > Message type: RELEASE COMPLETE (90)
Nov  2 17:22:28 VERBOSE[11329] logger.c: > [08 02 81 81]
Nov  2 17:22:28 VERBOSE[11329] logger.c: > Cause (len= 4) [ Ext: 1  Coding: 
CCITT (ITU) standard (0) 0: 0   Location: Private network serving the local 
user (1)
Nov  2 17:22:28 VERBOSE[11329] logger.c: >  Ext: 1  Cause: 
Unallocated (unassigned) number (1), class = Normal Event (0) ]
Nov  2 17:22:28 VERBOSE[11329] logger.c: NEW_HANGUP DEBUG: Calling q931_hangup, 
ourstate Null, peerstate Null
Nov  2 17:22:28 VERBOSE[11329] logger.c: NEW_HANGUP DEBUG: Destroying the call, 
ourstate Null, peerstate Null

The 'from-pstn-deviate-custom' context has lines such as:
exten => _100[14567]XXX,1,...
exten => _100[14567]XXX,n,...

Any ideas?

Vieri



  

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Re: [asterisk-users] SNOM 870

[Steve Howes]

>> The mortality rate on power supplies, diplays and the number or  
>> broken
>> receiver hook swicthes on the lot of Snom 360's i bought 3 years  
>> ago is
>> outright embarrassing.
>>
> That's odd. We've had Snom 190s, 320s, and 360s running day in day out
> for years with not a single issue.

Would say about 1 in 10 PSUs fail in 2 years. About 1 in 20 in 2 years  
for screens. Never had a hook switch fail.

Steve

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Re: [asterisk-users] SNOM 870

[SIP]

Remco Barendse wrote:
> On Fri, 30 Oct 2009, hbk wrote:
>
>   
>> Hi,
>>
>> I have played with the 820 for some weeks, mostly love it excellent speech 
>> quality. Even got the "mini" browser running
>> showing my favorite webcam, this could be put to real use too:)
>>
>> Issues so far:
>> Some embarrassing crashes while speaking, was able to speak but all freezed. 
>> Still a little fresh firmware I guess.
>> Error 404 after showing webcam picture, but it works!
>> Have to use *1 to start recording, record soft button does not seem to work 
>> with *.
>>
>> Still I recommend it, best IP phone I have tried!
>> Not sure 870 is worth the extra money, not tested that yet.
>> 
>
> How is the build quality of the 870?
>
> The mortality rate on power supplies, diplays and the number or broken 
> receiver hook swicthes on the lot of Snom 360's i bought 3 years ago is 
> outright embarrassing.
>
>
>   
That's odd. We've had Snom 190s, 320s, and 360s running day in day out
for years with not a single issue. Maybe we got all the good ones from
your batch. If that's the case, I thank you for 'taking one for the
team' as it were. ;)

N.

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Re: [asterisk-users] Skyp SIP? - what is free for a home *

[Alejandro Lengua Vega]

There is something called Opensky that claims to allow making and receiving
skype calls
on your SIP device (including Asterisk). However I haven´t tested it yet.

http://latestgeeknews.blogspot.com/2009/02/opensky-skype-interface-gateway-who.html

2009/11/1 hbk 

> Hi,
>
> I get confused about all solutions for Skype!
>
> I want to connect as simple as possible out home *  to be able to at
> least answer Skype calls.
> Now I use a PC USB box and a FXO, works ok both call directions but uses
> a PC.
>
> Any good and free idea ?
>
> Thank you!
> HB
>
>
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Re: [asterisk-users] Asterisk Fax Module

[Danny Nicholas]

Just my .02;  You shouldn't use outlying features like fax on rc releases -
these aren't usually but can be (b)leading edge stuff.

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Warren Selby
Sent: Monday, November 02, 2009 9:56 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk Fax Module

 

Probably after 1.6.2 has been officially released beyond the release
candidate stage.

Thanks,
--Warren Selby

On Mon, Nov 2, 2009 at 4:14 AM, Khaled W Chehab 
wrote:

When we can expect to have a res_fax and res_fax_degium module for  asterisk
V 1.6.2

 

Regards

 

 

Khaled  Chehab

   NGN Eng.

 

 Untitled

 Operations Office - Lebanon

 Office : +961 1 868686 ext 115

 Mobile: +961 3 045212

 E-mail:  kche...@xplorium.com  

 MSN ID :khalidche...@hotmail.com  

 Web Site: http://www.Xplorium.com

 

 

 

  _  

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Re: [asterisk-users] MySQL CDR

[Warren Selby]

What version of asterisk are you installing?

Thanks,
--Warren Selby

On Mon, Nov 2, 2009 at 5:59 AM, Dan Journo wrote:

>  Hello,
>
>
>
> Does anyone know where I can get an up to date guide on installing
> CDR_MSQL?
>
>
>
> VOIP-Info has old information.
>
>
>
> Many thanks
>
> Dan
>
>
>
>
>
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Re: [asterisk-users] Asterisk Fax Module

[Warren Selby]

Probably after 1.6.2 has been officially released beyond the release
candidate stage.

Thanks,
--Warren Selby

On Mon, Nov 2, 2009 at 4:14 AM, Khaled W Chehab wrote:

>  When we can expect to have a res_fax and res_fax_degium module for
> asterisk V 1.6.2
>
>
>
> Regards
>
>
>
>
>
> *Khaled  Chehab*
>
> *   NGN Eng.*
>
>
>
>  [image: Untitled]
>
> * Operations Office - Lebanon*
>
>  Office : +961 1 868686 ext 115
>
>  Mobile: +961 3 045212
>
>  E-mail:  kche...@xplorium.com 
>
>  MSN ID :khalidche...@hotmail.com
>
>  Web Site: http://www.Xplorium.com
>
>
>
>
>
>
> --
> *
>
> No employee or agent is authorized to conclude any binding agreement on 
> behalf of Xplorium with another party by e-mail without express written 
> confirmation by an officer of Xplorium. Any views expressed by an individual 
> in this electronic message do not necessarily reflect views of Xplorium or 
> its subsidiaries and associates.
>
>
> This electronic message and its attachments are solely addressed to the 
> addressee(s), and contain confidential information protected from disclosure 
> belonging to Xplorium.
>
>
> If you are not the intended addressee of this electronic message and its 
> attachments, kindly delete it immediately from your system and notify the 
> sender by electronic mail. You must not copy this message or attachment or 
> disclose its content to any other person.
>
>
> Xplorium does not guarantee the integrity of this electronic message and any 
> of its attachments, or that they are free from computer viruses or other 
> defects.
> *
>
>
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Re: [asterisk-users] SNOM 870

[Remco Barendse]

On Fri, 30 Oct 2009, hbk wrote:

> Hi,
> 
> I have played with the 820 for some weeks, mostly love it excellent speech 
> quality. Even got the "mini" browser running
> showing my favorite webcam, this could be put to real use too:)
> 
> Issues so far:
> Some embarrassing crashes while speaking, was able to speak but all freezed. 
> Still a little fresh firmware I guess.
> Error 404 after showing webcam picture, but it works!
> Have to use *1 to start recording, record soft button does not seem to work 
> with *.
> 
> Still I recommend it, best IP phone I have tried!
> Not sure 870 is worth the extra money, not tested that yet.

How is the build quality of the 870?

The mortality rate on power supplies, diplays and the number or broken 
receiver hook swicthes on the lot of Snom 360's i bought 3 years ago is 
outright embarrassing.



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Re: [asterisk-users] Remote IP Phone's

[Dean Hoover]

Hello Connor,

You might be able to start with this link:

http://www.voip-info.org/wiki/view/Asterisk+sip+md5secret

You can also go even further if you know the IP address of where the 
phones are coming from by using the permit/deny options:

http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+permit-deny-mask

Hope that helps.

Dean Hoover


Connor Spiess wrote:
> Hi all,
> 
>  
> 
> I am wondering what people are doing for security when registering IP 
> phone’s remotely if you do not have the equipment to do a VPN tunnel at 
> the remote site. The phone I would be working with mainly is the Polycom 
> lineup.
> 
>  
> 
> Thanks,
> 
> Connor Spiess
> 
>  
> 
> 
> 
> 
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Re: [asterisk-users] supermicro hardware + sangoma

[Jacek Blaschke]

Hi, 
 
Try Berofix / beronet - tested with Tylersburg Supermicro mb's - works perfectly well.
 
Jacek
 
-Original Message-From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of marek cervenkaSent: Monday, November 02, 2009 1:38 PMTo: asterisk-users@lists.digium.comSubject: [asterisk-users] supermicro hardware + sangoma
 
hi,
 
i have new supermicro server (centos5, 2.6.30.9/2.6.27.34/2.6.18-distro 
kernels, wanpipe 3.5.6)
card is:
1 . AFT-A101-SH : SLOT=4 : BUS=8 : IRQ=11 : CPU=A : PORT=1 : HWEC=32 : V=36
 
and i have this in log
 
irq 17: nobody cared (try booting with the "irqpoll" option)
Pid: 0, comm: swapper Not tainted 2.6.30.9rh #1
Call Trace:
  [] __report_bad_irq+0x27/0x90
  [] note_interrupt+0x13a/0x180
  [] handle_fasteoi_irq+0x9f/0xd0
  [] ? handle_fasteoi_irq+0x0/0xd0
  [] ? do_IRQ+0x46/0xb0
  [] ? acpi_hw_write_port+0x27/0x71
  [] ? common_interrupt+0x29/0x30
  [] ? acpi_idle_enter_bm+0x218/0x241
  [] ? cpuidle_idle_call+0x6e/0xc0
  [] ? cpu_idle+0x35/0x60
  [] ? start_secondary+0x182/0x1e0
handlers:
[] (sdla_isr+0x0/0x310 [wanpipe])
Disabling IRQ #17
 
dou you have idea what is the problem? 
irqpoll doesnt help
 
i have tried this supermicro motherboards
http://www.supermicro.com/products/motherboard/QPI/5500/X8DTU-F.cfm
http://www.supermicro.com/products/motherboard/QPI/5500/X8DTi-F.cfm
http://www.supermicro.com/products/motherboard/QPI/5500/X8DTL-iF.cfm
 
do you have someone working sangoma card with Tylersburg(intel 5520/5500) 
chipset?
 
thanks
 
p.s. sorry for offtopic :(
 
---
Marek Cervenka
===
 
 
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Re: [asterisk-users] Remote IP Phone's

[Danny Nicholas]

You can restrict on IP address, MAC address and port type and that's just
what I know.  If someone want's through bad enough you're going to have a
problem, but you can at least slow down or stop casual hackers.

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Connor Spiess
Sent: Monday, November 02, 2009 9:38 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Remote IP Phone's

 

Hi all,

 

I am wondering what people are doing for security when registering IP
phone's remotely if you do not have the equipment to do a VPN tunnel at the
remote site. The phone I would be working with mainly is the Polycom lineup.

 

Thanks,

Connor Spiess

 

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Re: [asterisk-users] Dynamic DNS trunk --- SOLVED

[B.Masoud @ SH]

dnsmgr.conf:
enable=yes  
refreshinterval=300

regards.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of B.Masoud @ SH
Sent: Friday, October 30, 2009 3:28 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Dynamic DNS trunk

Thanks
I did this

dnsmgr.conf:
enable=yes  
refreshinterval=300

I did "dnsmgr refresh", the DNS in the trunk did not got the new ip, also I
waited 5 min.

do I have to add an entry to dnsmgr??

Thanks!

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matt Riddell
Sent: Friday, October 30, 2009 1:53 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Dynamic DNS trunk

On 30/10/09 6:42 AM, B.Masoud @ SH wrote:
> Hi
>
> I tried with registration, it did not update the IP address
>
> I can only see it updated if I typed:
>
> Sip reload
>
> I have few questions:
>
> Is there any way Asterisk automatically updates the DNS?

Yep /etc/asterisk/dnsmgr.conf

-- 
Cheers,

Matt Riddell
Director
___

http://www.venturevoip.com/news.php (Daily Asterisk News)
http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer)
http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems)

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[asterisk-users] Remote IP Phone's

[Connor Spiess]

Hi all,

I am wondering what people are doing for security when registering IP phone's 
remotely if you do not have the equipment to do a VPN tunnel at the remote 
site. The phone I would be working with mainly is the Polycom lineup.

Thanks,
Connor Spiess

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Re: [asterisk-users] Async Agi problem

[Moises Silva]

because read=agi lets you read agi events, not send agi actions, agi in
write= must be set too if you want to send agi commands.

On Mon, Nov 2, 2009 at 3:25 AM, Robert Bielik wrote:

> Robert Bielik skrev:
> > Ok, now pretty much everything is up 'n running, however when I try to
> send an ANSWER (or any) command to *, it replies with
> > org.asteriskjava.manager.response.ManagerError "Permission Denied". In
> manager.conf for the *-java client, I have
> >
> > read =
> system,call,log,verbose,agent,user,config,dtmf,reporting,cdr,dialplan,agi
> > write = system,call,agent,user,config,command,reporting,originate
> >
> > * is 1.6.1.4 and *-java is 1.0.0
> >
>
> Hmm... setting "write = all" makes it work...
>
> /Rob
>
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-- 
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] Dialstatus

[Steve Edwards]

On Mon, 2 Nov 2009, Patrick Plattes wrote:

> you can do print the dialstatus to the console e.g.:
> exten => s,n,NoOp(${DIALSTATUS})
>
> More info:
> http://www.voip-info.org/wiki/view/Asterisk+cmd+NoOp

A "better practice" would be to use verbose() -- an application with 
greater functionality written specifically for this purpose.

-- 
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000

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[asterisk-users] Remote Party ID

[Dan Journo]

Hello,

 

Does anyone know how to set the remote party id?

 

Thanks

Dan Journo

 

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[asterisk-users] change L(x[:y][:z]) parameter of DIAL command after call is bridged

[Thomas Winter]

Hi,

is there any way from outside change x,y an z after a call is bridged?

maybe with AMI interface?

best regards 
Thomas

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Re: [asterisk-users] IVR

[Danny Nicholas]

As I understand this thread, you want two different contexts based on the
number you dial.  If you dial 1703... the big10 context should be executed.
If 1567... then Cleveland is executed.  Is this correct?  If so
Then this is what the two lines in [default] should read:
> exten => _1703.,1,Goto(big10-IVR,s,1)
> exten => _1567.,1,Goto(cleveland-IVR,s,1)

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Samuel Nair
Sent: Monday, November 02, 2009 12:08 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] IVR

Try running your asterisk service with the -vvvc option or connect to it 
via the -r option, and then try making a call that would cause it to 
land in the default context, you will see the way asterisk traverses the 
dial plan, this will give you good debug info.

sam!!

Thomas Perron wrote:
> Hi Juan,
> I have this:
>
> [default]
> ;include => stdexten
> include => big10-IVR
> include => cleveland-IVR
> exten => _1703XXX,1,Goto(big10-IVR,s,1)
> exten => _1567XXX,1,Goto(cleveland-IVR,s,1)
>
> You recommend I have this:
>
> [default]
> exten => _1703XXX,1,Goto(big10-IVR,s,1)
> exten => _1567XXX,1,Goto(cleveland-IVR,s,1)
>
> I tried this and it does not seem to work.
> Other thoughts?
> Where located please?
>
>
>
> 2009/11/1 "Juan E. Rodríguez"  >
>
> As I see here, you do not have to include the big10 context inside
> the default context, as you have an extension defined to reach
> that context and its extention is start extension.
> If the cleveland-IVR is based on the start extension too, the same
> applies.
>
> Besides that, it would work...(maybe not the way you expect... :-) )
>
> Regards,
> Juan
>
> Thomas Perron wrote:
>> Is this going to work:
>>
>> [default]
>> include => stdexten
>> include => big10-IVR
>> include => cleveland-IVR
>> exten => _17035745353,1,Goto(big10-IVR,s,1)
>> exten => _15672528431,1,Goto(cleveland-IVR,s,1)
>>
>>
>> [big10-IVR]
>> exten => s,1,Answer()
>> exten => s,n,Background(dir-welcome)
>> ;exten => s,n,WaitExten(1)
>> ;exten => s,n,Background(astcc-please-enter-your)
>>

>>
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>
> 
>
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Re: [asterisk-users] GSM and Wav format

[Danny Nicholas]

FWIW, I convert all of my files to WAV for Web reading using SOX.  Sox will
let you put all your files into the compressed gsm format for storage (sox
file.wav file.gsm), then you can just reverse the process for presentation
(sox file.gsm /tmp/file.wav)

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of ABBAS SHAKEEL
Sent: Monday, November 02, 2009 6:50 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] GSM and Wav format

 

After conversion from .wav to .mp3 the size remains almost the same. 

On Mon, Nov 2, 2009 at 5:46 PM, ABBAS SHAKEEL 
wrote:

Thanks Patrick.

 

First: I dont do that intentionally.

 

Thanks for suggestion. Let me investigate it.

 

On Mon, Nov 2, 2009 at 5:34 PM, Patrick Plattes 
wrote:

Hi,

at first: why do you use capitals for your name? Don't do that if you
don't have a very good reason.

You can convert wav to mp3 on the recording server and then send it to
the central system.

Bye,
 Patrick


On Mon, Nov 2, 2009 at 1:11 PM, ABBAS SHAKEEL
 wrote:
>
> Hello,
>
> Let me explain a scenario
>
> There are different Asterisk Servers at different Remote locations.
> Recording in different formats for FIVE seconds reveals that
>
> Format : Size
> wav : 84 KB
> gsm : 8.3 KB
> sln : 84 KB
>
> It can be recorded in any format. This is size for five seconds only. We
> need to transfer these files from different remote servers to a
centralized
> server.
> We need to play these recorded files on WEB.
>
> We have following options
>
> 1. Record in GSM and send to central Server. Which will convert to it to
WAV
> format using some code / any other thing. The issue in this is that CPU
will
> get very busy in this case. Because GSM Files can be very frequent.
> 2. Recored in Wav and send to central server. In this case we may face
> Network Bandwidth problem.(Even we create VPN).
>
>
> QUESTION IS: Is there any other format in which we can record using the
> record application provided its is small in size and directly playable on
> WEB.
>
>
>
>
> Best Regards
> Shakeel Abbas
>

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--
Niemann + Frey GmbH
Bischofstraße 80
47809 Krefeld
Tel. +49 2151 5554-263
Geschäftsführer: Gerd Frey
Sitz und Registergericht: Krefeld HRB 10851


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-- 
Best Regards
Shakeel Abbas




-- 
Best Regards
Shakeel Abbas

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Re: [asterisk-users] pattern matching DID

[Jared Smith]

On Sun, 2009-11-01 at 18:50 -0500, Thomas Perron wrote:
> Where is everyone located?  I am in Virginia, USA

There are literally thousands of people on this mailing list, so I doubt
it's worth having everyone tell you where they're from.  That being
said, I'm also in Virginia (near Fredericksburg), and there's enough
interest in the area that we might start up a local Asterisk users group
in the area.  What part of Virginia are you from?



-- 
Jared Smith
Training Manager
Digium, Inc.


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Re: [asterisk-users] asterisk 1.6.0 seems to have improper dial status when dialing dahdi extension

[covici]

svn 6466  from trunk.

Tzafrir Cohen  wrote:

> On Sun, Nov 01, 2009 at 04:13:22PM -0500, cov...@ccs.covici.com wrote:
> > Hi.  When I dial a Dahdi extension using asterisk 1.6.0, and there is no
> > answer, the extension hangs up, but the dial status is busy instead of
> > no answer.  How do I get this to work -- do I need to update dahdi?  The
> > card is an X400p using its FXS module.
> 
> What version of DAHDI?
> 
> -- 
>Tzafrir Cohen
> icq#16849755  jabber:tzafrir.co...@xorcom.com
> +972-50-7952406   mailto:tzafrir.co...@xorcom.com
> http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir
> 
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-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

 John Covici
 cov...@ccs.covici.com

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Re: [asterisk-users] Xorcom device not showing up in /proc

[Tzafrir Cohen]

On Mon, Nov 02, 2009 at 02:28:36PM +0100, Loic Didelot wrote:
> Hello,
> I now have this which looks better.
> r...@pbx1:~# lsusb 
> Bus 002 Device 010: ID e4e4:1162  
> Bus 002 Device 001: ID :  
> Bus 001 Device 004: ID 0403:e6c8 Future Technology Devices 
> Bus 001 Device 003: ID 0403:6001 Future Technology Devices 
> Bus 001 Device 001: ID :  
> 
> r...@pbx1:~# zaptel_hardware 
> usb:002/010  xpp_usb- e4e4:1162 Astribank-modular
> FPGA-firmware
> pci::01:05.0 wct4xxp+ d161:0210 Wildcard TE210P 

That is odd. Now is the time to look at /var/log/kern.log .

-- 
   Tzafrir Cohen
icq#16849755  jabber:tzafrir.co...@xorcom.com
+972-50-7952406   mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

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Re: [asterisk-users] Xorcom device not showing up in /proc

[Loic Didelot]

Hello,
I now have this which looks better.
r...@pbx1:~# lsusb 
Bus 002 Device 010: ID e4e4:1162  
Bus 002 Device 001: ID :  
Bus 001 Device 004: ID 0403:e6c8 Future Technology Devices 
Bus 001 Device 003: ID 0403:6001 Future Technology Devices 
Bus 001 Device 001: ID :  

r...@pbx1:~# zaptel_hardware 
usb:002/010  xpp_usb- e4e4:1162 Astribank-modular
FPGA-firmware
pci::01:05.0 wct4xxp+ d161:0210 Wildcard TE210P 


But the device is still not listed in /proc

Best regards,
Loïc.


On Mon, 2009-11-02 at 15:10 +0200, Tzafrir Cohen wrote:
> On Mon, Nov 02, 2009 at 10:38:45AM +0100, Loic Didelot wrote:
> > Hi,
> > can I reset and reload the firmware while the kernel modules are loaded?
> 
> Yes, there shouldn't be a problem with that.
> 
> > 
> > Should the files appear in /proc once the firmware has been loaded
> > correctly or do I need to unload/reload the kernel modules?
> 
> Kernel loading is purely userspace.
> 
-- 
Loïc DIDELOT
MIXvoip S.a.
Tel: +352 20  20
Fax: +352 20  90
ldide...@mixvoip.com
http://www.mixvoip.com


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Re: [asterisk-users] GSM and Wav format

[Tzafrir Cohen]

On Mon, Nov 02, 2009 at 05:11:53PM +0500, ABBAS SHAKEEL wrote:
> Hello,
> 
> Let me explain a scenario
> 
> There are different Asterisk Servers at different Remote locations.
> Recording in different formats for FIVE seconds reveals that

".WAV" (wav49, wav/gsm) should be playable by most systems.

-- 
   Tzafrir Cohen
icq#16849755  jabber:tzafrir.co...@xorcom.com
+972-50-7952406   mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

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Re: [asterisk-users] Xorcom device not showing up in /proc

[Tzafrir Cohen]

On Mon, Nov 02, 2009 at 10:38:45AM +0100, Loic Didelot wrote:
> Hi,
> can I reset and reload the firmware while the kernel modules are loaded?

Yes, there shouldn't be a problem with that.

> 
> Should the files appear in /proc once the firmware has been loaded
> correctly or do I need to unload/reload the kernel modules?

Kernel loading is purely userspace.

-- 
   Tzafrir Cohen
icq#16849755  jabber:tzafrir.co...@xorcom.com
+972-50-7952406   mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

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Re: [asterisk-users] GSM and Wav format

[ABBAS SHAKEEL]

After conversion from .wav to .mp3 the size remains almost the same.

On Mon, Nov 2, 2009 at 5:46 PM, ABBAS SHAKEEL
wrote:

> Thanks Patrick.
>
> First: I dont do that intentionally.
>
> Thanks for suggestion. Let me investigate it.
>
>
> On Mon, Nov 2, 2009 at 5:34 PM, Patrick Plattes wrote:
>
>> Hi,
>>
>> at first: why do you use capitals for your name? Don't do that if you
>> don't have a very good reason.
>>
>> You can convert wav to mp3 on the recording server and then send it to
>> the central system.
>>
>> Bye,
>>  Patrick
>>
>> On Mon, Nov 2, 2009 at 1:11 PM, ABBAS SHAKEEL
>>  wrote:
>> >
>> > Hello,
>> >
>> > Let me explain a scenario
>> >
>> > There are different Asterisk Servers at different Remote locations.
>> > Recording in different formats for FIVE seconds reveals that
>> >
>> > Format : Size
>> > wav : 84 KB
>> > gsm : 8.3 KB
>> > sln : 84 KB
>> >
>> > It can be recorded in any format. This is size for five seconds only. We
>> > need to transfer these files from different remote servers to a
>> centralized
>> > server.
>> > We need to play these recorded files on WEB.
>> >
>> > We have following options
>> >
>> > 1. Record in GSM and send to central Server. Which will convert to it to
>> WAV
>> > format using some code / any other thing. The issue in this is that CPU
>> will
>> > get very busy in this case. Because GSM Files can be very frequent.
>> > 2. Recored in Wav and send to central server. In this case we may face
>> > Network Bandwidth problem.(Even we create VPN).
>> >
>> >
>> > QUESTION IS: Is there any other format in which we can record using the
>> > record application provided its is small in size and directly playable
>> on
>> > WEB.
>> >
>> >
>> >
>> >
>> > Best Regards
>> > Shakeel Abbas
>> >
>> > ___
>> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> >
>> > asterisk-users mailing list
>> > To UNSUBSCRIBE or update options visit:
>> >   http://lists.digium.com/mailman/listinfo/asterisk-users
>> >
>>
>>
>>
>> --
>> Niemann + Frey GmbH
>> Bischofstraße 80
>> 47809 Krefeld
>> Tel. +49 2151 5554-263
>> Geschäftsführer: Gerd Frey
>> Sitz und Registergericht: Krefeld HRB 10851
>>
>> ___
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>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
>
> --
> Best Regards
> Shakeel Abbas
>
>


-- 
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Shakeel Abbas
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Re: [asterisk-users] GSM and Wav format

[ABBAS SHAKEEL]

Thanks Patrick.

First: I dont do that intentionally.

Thanks for suggestion. Let me investigate it.

On Mon, Nov 2, 2009 at 5:34 PM, Patrick Plattes wrote:

> Hi,
>
> at first: why do you use capitals for your name? Don't do that if you
> don't have a very good reason.
>
> You can convert wav to mp3 on the recording server and then send it to
> the central system.
>
> Bye,
>  Patrick
>
> On Mon, Nov 2, 2009 at 1:11 PM, ABBAS SHAKEEL
>  wrote:
> >
> > Hello,
> >
> > Let me explain a scenario
> >
> > There are different Asterisk Servers at different Remote locations.
> > Recording in different formats for FIVE seconds reveals that
> >
> > Format : Size
> > wav : 84 KB
> > gsm : 8.3 KB
> > sln : 84 KB
> >
> > It can be recorded in any format. This is size for five seconds only. We
> > need to transfer these files from different remote servers to a
> centralized
> > server.
> > We need to play these recorded files on WEB.
> >
> > We have following options
> >
> > 1. Record in GSM and send to central Server. Which will convert to it to
> WAV
> > format using some code / any other thing. The issue in this is that CPU
> will
> > get very busy in this case. Because GSM Files can be very frequent.
> > 2. Recored in Wav and send to central server. In this case we may face
> > Network Bandwidth problem.(Even we create VPN).
> >
> >
> > QUESTION IS: Is there any other format in which we can record using the
> > record application provided its is small in size and directly playable on
> > WEB.
> >
> >
> >
> >
> > Best Regards
> > Shakeel Abbas
> >
> > ___
> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >   http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
>
>
> --
> Niemann + Frey GmbH
> Bischofstraße 80
> 47809 Krefeld
> Tel. +49 2151 5554-263
> Geschäftsführer: Gerd Frey
> Sitz und Registergericht: Krefeld HRB 10851
>
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>



-- 
Best Regards
Shakeel Abbas
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[asterisk-users] supermicro hardware + sangoma

[marek cervenka]

hi,

i have new supermicro server (centos5, 2.6.30.9/2.6.27.34/2.6.18-distro 
kernels, wanpipe 3.5.6)
card is:
1 . AFT-A101-SH : SLOT=4 : BUS=8 : IRQ=11 : CPU=A : PORT=1 : HWEC=32 : V=36

and i have this in log

irq 17: nobody cared (try booting with the "irqpoll" option)
Pid: 0, comm: swapper Not tainted 2.6.30.9rh #1
Call Trace:
  [] __report_bad_irq+0x27/0x90
  [] note_interrupt+0x13a/0x180
  [] handle_fasteoi_irq+0x9f/0xd0
  [] ? handle_fasteoi_irq+0x0/0xd0
  [] ? do_IRQ+0x46/0xb0
  [] ? acpi_hw_write_port+0x27/0x71
  [] ? common_interrupt+0x29/0x30
  [] ? acpi_idle_enter_bm+0x218/0x241
  [] ? cpuidle_idle_call+0x6e/0xc0
  [] ? cpu_idle+0x35/0x60
  [] ? start_secondary+0x182/0x1e0
handlers:
[] (sdla_isr+0x0/0x310 [wanpipe])
Disabling IRQ #17

dou you have idea what is the problem? 
irqpoll doesnt help

i have tried this supermicro motherboards
http://www.supermicro.com/products/motherboard/QPI/5500/X8DTU-F.cfm
http://www.supermicro.com/products/motherboard/QPI/5500/X8DTi-F.cfm
http://www.supermicro.com/products/motherboard/QPI/5500/X8DTL-iF.cfm

do you have someone working sangoma card with Tylersburg(intel 5520/5500) 
chipset?

thanks

p.s. sorry for offtopic :(

---
Marek Cervenka
===


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Re: [asterisk-users] GSM and Wav format

[Patrick Plattes]

Hi,

at first: why do you use capitals for your name? Don't do that if you
don't have a very good reason.

You can convert wav to mp3 on the recording server and then send it to
the central system.

Bye,
 Patrick

On Mon, Nov 2, 2009 at 1:11 PM, ABBAS SHAKEEL
 wrote:
>
> Hello,
>
> Let me explain a scenario
>
> There are different Asterisk Servers at different Remote locations.
> Recording in different formats for FIVE seconds reveals that
>
> Format : Size
> wav : 84 KB
> gsm : 8.3 KB
> sln : 84 KB
>
> It can be recorded in any format. This is size for five seconds only. We
> need to transfer these files from different remote servers to a centralized
> server.
> We need to play these recorded files on WEB.
>
> We have following options
>
> 1. Record in GSM and send to central Server. Which will convert to it to WAV
> format using some code / any other thing. The issue in this is that CPU will
> get very busy in this case. Because GSM Files can be very frequent.
> 2. Recored in Wav and send to central server. In this case we may face
> Network Bandwidth problem.(Even we create VPN).
>
>
> QUESTION IS: Is there any other format in which we can record using the
> record application provided its is small in size and directly playable on
> WEB.
>
>
>
>
> Best Regards
> Shakeel Abbas
>
> ___
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>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 
Niemann + Frey GmbH
Bischofstraße 80
47809 Krefeld
Tel. +49 2151 5554-263
Geschäftsführer: Gerd Frey
Sitz und Registergericht: Krefeld HRB 10851

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[asterisk-users] GSM and Wav format

[ABBAS SHAKEEL]

Hello,

Let me explain a scenario

There are different Asterisk Servers at different Remote locations.
Recording in different formats for FIVE seconds reveals that

Format : Size
wav : 84 KB
gsm : 8.3 KB
sln : 84 KB

It can be recorded in any format. This is size for five seconds only. We
need to transfer these files from different remote servers to a centralized
server.
We need to play these recorded files on WEB.

We have following options

1. Record in GSM and send to central Server. Which will convert to it to WAV
format using some code / any other thing. The issue in this is that CPU will
get very busy in this case. Because GSM Files can be very frequent.
2. Recored in Wav and send to central server. In this case we may face
Network Bandwidth problem.(Even we create VPN).


QUESTION IS: Is there any other format in which we can record using the
record application provided its is small in size and directly playable on
WEB.





Best Regards
Shakeel Abbas
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[asterisk-users] MySQL CDR

[Dan Journo]

Hello,

 

Does anyone know where I can get an up to date guide on installing
CDR_MSQL?

 

VOIP-Info has old information.

 

Many thanks

Dan

 

 

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Re: [asterisk-users] Dialstatus

[Patrick Plattes]

Hi,

you can do print the dialstatus to the console e.g.:
exten => s,n,NoOp(${DIALSTATUS})

More info:
http://www.voip-info.org/wiki/view/Asterisk+cmd+NoOp

Bye,
 Patrick

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Re: [asterisk-users] SNOM 870

[--[ UxBoD ]--]

- "Garth van Sittert"  wrote:

| We have the 870 working great in our test environment so far.
| 
| 
| Garth van Sittert
| BSC (Physics & Comp Sci)
| Technical Director
| BitCo
| 08600 24826
| www.bitco.co.za
| 
| 
| 
| --[ UxBoD ]-- wrote:
| > Anybody tried one with Asterisk yet ? Views ?
| >
| > Best Regards,
| >
| >
| >   
| 
How responsive is the touch screen ?

Best Regards,


-- 
This message has been scanned for viruses and
dangerous content and is believed to be clean.

SplatNIX IT Services :: Innovation through collaboration


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Re: [asterisk-users] SNOM 870

[Garth van Sittert]

We have the 870 working great in our test environment so far.


Garth van Sittert
BSC (Physics & Comp Sci)
Technical Director
BitCo
08600 24826
www.bitco.co.za



--[ UxBoD ]-- wrote:
> Anybody tried one with Asterisk yet ? Views ?
>
> Best Regards,
>
>
>   

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Re: [asterisk-users] hardware requirements for asterisk

[Hans Witvliet]

On Mon, 2009-11-02 at 09:37 +, aster...@opensourcesolution.in wrote:
> hello friends
>  friend i had just finished my chapters of asterisk. ill be
> configuring asterisk in for home for r/d purpose.  i am having p4
> machine with 1 GB RAM, ill be configuring asterisk on centos 5.3, the
> only doubt which i am having is  which hardware ill have to buy to
> configure asterisk. i think analog card ? plz clear my doubt. n be
> with me from beginning till end, of the journey of asterisk. 
> Regards,
> Pawan

Hi Pawan,

It vey much depend on what you expect the box to be handling
As you wrote: soho + R&D, i presume it will be anoccasional call.

Personally, i would recommend to leave the analogue stuff out of your
PC. (no hassle with pci-slots, shared-IRQ's, PSU, )
Leave the handling of analogue-parts to an ATA-box.
Linksys (and others) are making those at reasonable prices (Cheaper than
an analogue card)

hw

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[asterisk-users] Asterisk Fax Module

[Khaled W Chehab]

When we can expect to have a res_fax and res_fax_degium module for  asterisk
V 1.6.2

 

Regards

 

 

Khaled  Chehab

   NGN Eng.

 

 Untitled

 Operations Office - Lebanon

 Office : +961 1 868686 ext 115

 Mobile: +961 3 045212

 E-mail:  kche...@xplorium.com  

 MSN ID :khalidche...@hotmail.com  

 Web Site: http://www.Xplorium.com

 

 



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Re: [asterisk-users] hardware requirements for asterisk

[Alex Balashov]

aster...@opensourcesolution.in wrote:

> hello friends
>  friend i had just finished my chapters of asterisk. ill be configuring 
> asterisk in for home for r/d purpose.  i am having p4 machine with 1 GB 
> RAM, ill be configuring asterisk on centos 5.3, the only doubt which i 
> am having is  which hardware ill have to buy to configure asterisk. i 
> think analog card ? plz clear my doubt. n be with me from beginning till 
> end, of the journey of asterisk.

Depending on what you intend to accomplish, you may not need any 
additional hardware;  you do not need PSTN connectivity to use 
Asterisk.  If you want it anyway, you can get PSTN origination (calls 
from the PSTN->VoIP) and termination (VoIP->PSTN) over IP without any 
need for physical lines.

If you have a fixed analog line and are determined to interface it 
with Asterisk, you would need an FXO card.  TDM hardware that 
interfaces with T1/E1 circuits (ISDN PRI, typically) is also available.

-- Alex

-- 
Alex Balashov - Principal
Evariste Systems
Web : http://www.evaristesys.com/
Tel : (+1) (678) 954-0670
Direct  : (+1) (678) 954-0671

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Re: [asterisk-users] Xorcom device not showing up in /proc

[Loic Didelot]

Hi,
can I reset and reload the firmware while the kernel modules are loaded?

Should the files appear in /proc once the firmware has been loaded
correctly or do I need to unload/reload the kernel modules?

Best regards,
Loic.


On Mon, 2009-11-02 at 10:38 +0200, Tzafrir Cohen wrote:
> On Mon, Nov 02, 2009 at 09:33:21AM +0100, Loic Didelot wrote:
> > Hello,
> > I remember having a similar problem sometime back but I do not remember
> > the solutions. My Xorcom FXS bank is not showing up in /proc.
> > 
> > Here some helpful output:
> > r...@pbx1:~# zaptel_hardware 
> > usb:002/006  xpp_usb- e4e4:1161 Astribank-modular USB-firmware
> 
> 1161: the FPGA firmware is not loaded.
> 
> http://docs.tzafrir.org.il/dahdi-tools/README.Astribank.html#_lsusb_test
> 
> What version of Zaptel is it?
> 


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[asterisk-users] hardware requirements for asterisk

[asterisk]

hello friends
 friend i had just finished my chapters of asterisk. ill be
configuring asterisk in for home for r/d purpose. i am having p4 machine
with 1 GB RAM, ill be configuring asterisk on centos 5.3, the only doubt
which i am having is which hardware ill have to buy to configure asterisk.
i think analog card ? plz clear my doubt. n be with me from beginning till
end, of the journey of asterisk. 
Regards,
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Re: [asterisk-users] Xorcom device not showing up in /proc

[Loic Didelot]

Hi,
I am using this version zaptel-1.4.12.9.svn.r4653.

Best regards,
Loïc.


On Mon, 2009-11-02 at 10:38 +0200, Tzafrir Cohen wrote:
> On Mon, Nov 02, 2009 at 09:33:21AM +0100, Loic Didelot wrote:
> > Hello,
> > I remember having a similar problem sometime back but I do not remember
> > the solutions. My Xorcom FXS bank is not showing up in /proc.
> > 
> > Here some helpful output:
> > r...@pbx1:~# zaptel_hardware 
> > usb:002/006  xpp_usb- e4e4:1161 Astribank-modular USB-firmware
> 
> 1161: the FPGA firmware is not loaded.
> 
> http://docs.tzafrir.org.il/dahdi-tools/README.Astribank.html#_lsusb_test
> 
> What version of Zaptel is it?
> 


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Re: [asterisk-users] asterisk 1.6.0 seems to have improper dial status when dialing dahdi extension

[Tzafrir Cohen]

On Sun, Nov 01, 2009 at 04:13:22PM -0500, cov...@ccs.covici.com wrote:
> Hi.  When I dial a Dahdi extension using asterisk 1.6.0, and there is no
> answer, the extension hangs up, but the dial status is busy instead of
> no answer.  How do I get this to work -- do I need to update dahdi?  The
> card is an X400p using its FXS module.

What version of DAHDI?

-- 
   Tzafrir Cohen
icq#16849755  jabber:tzafrir.co...@xorcom.com
+972-50-7952406   mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

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Re: [asterisk-users] SIP Peers still ping with SIP OPTIONS on a reload

[Marc Leurent]

I have the same result with Asterisk 1.4.21 on a Debian Lenny server
-- 
-- --
Marc LEURENT
lf...@leurent.eu

Le mercredi, 28 octobre 2009 12.27:59, Marc Leurent a écrit :
> Hello, when I remove a peer from my sip.conf and just do a reload, the peer
>  is still ping with SIP OPTIONS until I restart Asterisk, I use Asterisk
>  1.4.27-rc2. Is it normal? Thanks
> 
> As an example, I have added and after removed this lines and
> 
> ;[sip_trk_vm]
> ;host=88.191.80.8
> ;type=peer
> ;context=default
> ;dtmfmode=info
> ;insecure=port,invite
> ;nat=never
> ;sendrpid=yes
> ;disallow=all
> ;allow=alaw
> 

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Re: [asterisk-users] Forward DID to another server

[Randy R]

Alex,

You forgot to clip the extra > from the quote, shame on you!

On Mon, Nov 2, 2009 at 9:47 AM, Alex Balashov  wrote:
> Tzafrir Cohen wrote:
>
>> Top-posting, on top of your other sins.
>>
>> Please spare us this capital punishment.
>
> An entirely fair point.
>
> Nevertheless, I eagerly await your similarly convicted petitions aimed
> at curbing illiterate, obnoxious and indolent attempts to get others
> to do extensive work on one's behalf to fix a problem one has not done
> the due diligence to rudimentarily understand.

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Re: [asterisk-users] Forward DID to another server

[Alex Balashov]

Tzafrir Cohen wrote:

> Top-posting, on top of your other sins.
> 
> Please spare us this capital punishment.

An entirely fair point.

Nevertheless, I eagerly await your similarly convicted petitions aimed 
at curbing illiterate, obnoxious and indolent attempts to get others 
to do extensive work on one's behalf to fix a problem one has not done 
the due diligence to rudimentarily understand.

-- 
Alex Balashov - Principal
Evariste Systems
Web : http://www.evaristesys.com/
Tel : (+1) (678) 954-0670
Direct  : (+1) (678) 954-0671

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Re: [asterisk-users] Xorcom device not showing up in /proc

[Tzafrir Cohen]

On Mon, Nov 02, 2009 at 09:33:21AM +0100, Loic Didelot wrote:
> Hello,
> I remember having a similar problem sometime back but I do not remember
> the solutions. My Xorcom FXS bank is not showing up in /proc.
> 
> Here some helpful output:
> r...@pbx1:~# zaptel_hardware 
> usb:002/006  xpp_usb- e4e4:1161 Astribank-modular USB-firmware

1161: the FPGA firmware is not loaded.

http://docs.tzafrir.org.il/dahdi-tools/README.Astribank.html#_lsusb_test

What version of Zaptel is it?

-- 
   Tzafrir Cohen
icq#16849755  jabber:tzafrir.co...@xorcom.com
+972-50-7952406   mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

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[asterisk-users] PSTN Line Parameters Checking

[Torintino T]

I have Asterisk and Digium AEX808B

What are please the commands that i can run on Asterisk to get the information 
about the connected lines from PSTN to see the parameters of them 
and as well the corresponding files in Asterisk that i can change into, to tune 
these parameters to be matched together.

Thanks a lot.
  
_
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Re: [asterisk-users] Forward DID to another server

[Tzafrir Cohen]

On Mon, Nov 02, 2009 at 02:49:05AM -0500, ALEX BALASHOV wrote:
> In a manner of speaking.

Top-posting, on top of your other sins.

Please spare us this capital punishment.

-- 
   Tzafrir Cohen
icq#16849755  jabber:tzafrir.co...@xorcom.com
+972-50-7952406   mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

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Re: [asterisk-users] Forward DID to another server

[Tzafrir Cohen]

On Mon, Nov 02, 2009 at 12:19:41PM +0530, DHAVAL INDRODIYA wrote:
> hello all,
> 
> i have 2 asterisk boxes on that 1 have public IP Address and another is only
> have local IP address
> 
> now on public IP there are some 7 DID  forwarded , now i want to forward 3
> DID out of 7 DID to local machine we called server B , I know there are 
> DIal , and Switch statement in asterisk ,
> 
> but is there any other convenient way to do this. because if call ratio is
> high then my call legs become very long.

You have just one public IP address? Just one IP connection? If both
servers are connected on the same network connection to the internet,
the lag wouldn't matter anyway.

If server B is on a different connection, the provider can technically
forward calls to it once it has been registered. Though this may not
work well with the way this provider works, so this may or may not work.
So you'll have to provide some more specific details if you want to be
more specific.

Proper puctuation and proper separation of paragraphs (rather than
randomly pressing Enter) will also help making your messages look more
readable.

-- 
   Tzafrir Cohen
icq#16849755  jabber:tzafrir.co...@xorcom.com
+972-50-7952406   mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

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[asterisk-users] Xorcom device not showing up in /proc

[Loic Didelot]

Hello,
I remember having a similar problem sometime back but I do not remember
the solutions. My Xorcom FXS bank is not showing up in /proc.

Here some helpful output:
r...@pbx1:~# zaptel_hardware 
usb:002/006  xpp_usb- e4e4:1161 Astribank-modular
USB-firmware
pci::01:05.0 wct4xxp+ d161:0210 Wildcard TE210P 

r...@pbx1:~# ls /proc/zaptel/
1  2

r...@pbx1:~# ls /proc/bus/usb/
001  002  devices


lsmod:
pp   156092  2 xpd_fxs,xpp_usb
ftdi_sio   37640  2 
usbserial  35688  6 ftdi_sio
wmi_acer9644  0 
wct4xxp   353920  52 
zaptel200068  109 xpd_fxs,xpp,wct4xxp
crc_ccitt   3072  1 zaptel
button  9232  0 


The strange thing is that my configuration worked until I rebooted the
server. 


Best regards,
Loïc.



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Re: [asterisk-users] Async Agi problem

[Robert Bielik]

Robert Bielik skrev:
> Ok, now pretty much everything is up 'n running, however when I try to send 
> an ANSWER (or any) command to *, it replies with
> org.asteriskjava.manager.response.ManagerError "Permission Denied". In 
> manager.conf for the *-java client, I have
> 
> read = 
> system,call,log,verbose,agent,user,config,dtmf,reporting,cdr,dialplan,agi
> write = system,call,agent,user,config,command,reporting,originate
> 
> * is 1.6.1.4 and *-java is 1.0.0
> 

Hmm... setting "write = all" makes it work... 

/Rob

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