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Juan Cardoza | 1 Oct 2009 01:03

Asterisk over CentOS the module for Digium TE121 is not in the zaptel file

Hello I have a CentOS OS that have asterisk installed, also zaptel, but when I use the:

lspci command

I have the next asnwer:
03:80.0 Ethernet controller: Unknown device d161:8000 (rev 11)

I also check the zaptel file that contain the modules that can support and the wcte12xp module is not in the file, so I think the problem is that the driver is not install into the OS.

I know that we can migrate to dahdi, but at this time I need a zaptel file that can support this card, does anyone can help me with this issue?

Thanks a lot for your help.
Jhon


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Steve Edwards | 1 Oct 2009 01:19

Re: How to finish a Meetme

>> On Wed, 30 Sep 2009, covici <at> ccs.covici.com wrote:
>>
>>> there is an undocumented feature in meetme using the kick option called
>>> all, which kicks everyone off if you want to be sure and end the
>>> conference.

> Steve Edwards <asterisk.org <at> sedwards.com> wrote:
>
>> Are you referring to the documented 'K' option for the meetmeadmin()
>> dialplan application or the inadequately documented "meetme kick <confno>
>> <usernumber>" CLI command -- which doesn't (1.2) document that
>> "<usernumber>" can be "all?" (Or that <confno> does not have to be
>> numeric.)

On Wed, 30 Sep 2009, covici <at> ccs.covici.com wrote:
>
> The cli command.  I wish you could some of this from the phone, but 
> you'd almost have to have an audio display of user numbers and caller 
> ids to have it make sense.

I did this a couple of months ago. An "admin," wanting to kick a user from 
a conference would execute an AGI that would map an index to a meetme user 
id via AMI so the admin could mute or un-mute a user (to identify the 
abusive user) or kick the user.

--

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

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David @ULC | 1 Oct 2009 01:40
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Calls at 2 different locations

I want to use IPKall with Asterisk.

Now, I want my calls to land at 2 different locations , not connected with each other.

If I want to configure IPKall DID number in Asterisk , I need to specify IP on IPKall.

How can I make it enable so that calls can land up at both locations ?

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Kirill 'Big K' Katsnelson | 1 Oct 2009 01:57

Choose IAX or SIP trunking?

Asterisk 1.6 behind a firewall and NAT. We are terminating multiple DID 
calls, originating and transferring.

A provider offers both SIP and IAX trunking. Cateris paribus, what is 
the preferred solution to choose? What points to consider?

I can name the provider if this is not against this list policy--is it?

Thanks,

  -kkm

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Steve Edwards | 1 Oct 2009 02:41

Re: Choose IAX or SIP trunking?

On Wed, 30 Sep 2009, Kirill 'Big K' Katsnelson wrote:

> Asterisk 1.6 behind a firewall and NAT. We are terminating multiple DID 
> calls, originating and transferring.
>
> A provider offers both SIP and IAX trunking. Cateris paribus, what is 
> the preferred solution to choose? What points to consider?

Ceteris paribus, I prefer IAX. It tends to "just work" and it has a lot 
fewer "knobs" to turn.

Some say the audio quality is better with SIP. My experience has been with 
"low volume" (xx) calls across the internet and "high volume" (xxx) within 
the same cabinet.

I'd try IAX since it is so simple to configure. If you are not satisfied, 
try SIP.

--

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

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Moises Silva | 1 Oct 2009 02:47
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Re: E1/T1 Tapping call recording in Asterisk - Testing needed


Is your code vendor locked to Sangoma ???


Hello Martin, not at all. The code is intended to be part of chan_dahdi Asterisk channel driver and as such any card capable of using the dahdi interface can benefit from it.
 
--
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. moy <at> sangoma.com
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John Millican | 1 Oct 2009 03:08

Re: chanspy and DISA

Steve Edwards wrote:
>> Steve Edwards wrote:
>>> Is the manager or are the agents using disa()?
>>>
>>> How about:
>>>
>>>          exten = *,n,                    set(SPYGROUP=ALLOW-SPYING)
>>>
>>> for the agents and:
>>>
>>>          exten = *,n,                    chanspy(,g(ALLOW-SPYING))
>>>
>>> the manager?
> 
> On Tue, 29 Sep 2009, John Millican wrote:
>> The manager wants to be able to spy on agents who dial through the PBX 
>> from their homes.  Currently the agents dial the main number, use the 
>> "secret" code to get to authenticate and DISA, and then dial back out 
>> for their sales calls. I have chanspy working great on all internal 
>> phones/extensions use group to limit who can spy and who can not. It not 
>> so much to allow spying it is finding the correct channel to spy on for 
>> the remote users.
> 
> How about something like these snippets:
> 
> [i](!)
>          exten = i,1,                    goto(${CONTEXT},s,1)
> [s](!)
>          exten = s,1,                    verbose(1,[${CONTEXT}:${EXTEN}])
> 
> [home-agent-login](i,s)
>          exten = s,n,                    read(AGENT-ID,enter-agent-number)
>          exten = s,n,                    set(SPYGROUP=${AGENT-ID})
>          .
>          .
>          .
> 
> [supervisor-login](i,s)
>          exten = s,n,                    read(AGENT-ID,enter-agent-number)
>          exten = s,n,                    chanspy(,g(${AGENT-ID}))
>          exten = s,n,                    goto(s,1)
>          .
>          .
>          .
> 

Thank you very much for this.
With a little tweaking it worked great, since each remote workers
callerid is matched before going to authenticate I just set the spy
group so the remote guys don't have a choice and now the manager has a
known group of one for each remote worker.
Thanks again for the help
JohnM

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Kirill 'Big K' Katsnelson | 1 Oct 2009 03:13

Re: Choose IAX or SIP trunking?

Steve Edwards wrote:
> Some say the audio quality is better with SIP. My experience has been with 
> "low volume" (xx) calls across the internet and "high volume" (xxx) within 
> the same cabinet.

My understanding was that IAX encapsulates the same RTP traffic, or, and 
the very least, same stream of data encoded by a codec. Is that not true 
in case of IAX? How can a transport protocol affect volume--or quality 
(lest it is dropping packets)?

  -kkm, now puzzled.

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Martin | 1 Oct 2009 04:07

Re: E1/T1 Tapping call recording in Asterisk - Testing needed

That's nice. At least now peopel that want to do call recording can do
so without having to keep Asterisk in between the circuits.
However all other applications like added voicemail, conferencing,
followme etc ... still needs Asterisk in between unless "they" have a
spare port on the PBX and do the routing...

Martin

On Wed, Sep 30, 2009 at 7:47 PM, Moises Silva <moises.silva <at> gmail.com> wrote:
>>
>> Is your code vendor locked to Sangoma ???
>>
>
> Hello Martin, not at all. The code is intended to be part of chan_dahdi
> Asterisk channel driver and as such any card capable of using the dahdi
> interface can benefit from it.
>
> --
> 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. moy <at> sangoma.com
>
> _______________________________________________
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> asterisk-users mailing list
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>   http://lists.digium.com/mailman/listinfo/asterisk-users
>

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Steve Edwards | 1 Oct 2009 04:10

Re: Choose IAX or SIP trunking?

> Steve Edwards wrote:

>> Some say the audio quality is better with SIP. My experience has been 
>> with "low volume" (xx) calls across the internet and "high volume" 
>> (xxx) within the same cabinet.

On Wed, 30 Sep 2009, Kirill 'Big K' Katsnelson wrote:

> My understanding was that IAX encapsulates the same RTP traffic, or, and 
> the very least, same stream of data encoded by a codec. Is that not true 
> in case of IAX? How can a transport protocol affect volume--or quality 
> (lest it is dropping packets)?

My (limited) understanding is that IAX sends all call control and RTP to 
port 4569. Thus, a busy pipe can adversely affect timing if the single 
thread reading from the socket can't process the packets fast enough.

Whether this manifests itself as dropped packets or jitter or whatever is 
beyond my experience. I've never had a client complain, but most of my 
traffic is within the same cabinet.

--

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

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Gmane