date:20120124

Question 1 - no 

The format is this

#define FORMAT2 "%-25.25s %-39.39s %-3.3s %-10.10s %-3.3s %-8s %-11s
%-32.32s %s\n"

 

Question 2

debsphone2*CLI> core show channels concise

SIP/1104-051b!default!99!2!Up!Playback!tt-monkeys!1104!!!3!3!(None)!1327
35.1307

debsphone2*CLI> core show channels verbose

Channel  Context  ExtensionPrio State
Application  Data  CallerIDDuration Accountcode
PeerAccount BridgedTo

SIP/1104-051bdefault  99  2 Up
Playback tt-monkeys110400:00:08
(None)

1 active channel

1 active call

 

>From cli.c (asterisk 10)

e->command = "core show channels [concise|verbose|count]";

e->usage =

"Usage: core show channels
[concise|verbose|count]\n"

"   Lists currently defined channels and some
information about them. If\n"

"   'concise' is specified, the format is
abridged and in a more easily\n"

"   machine parsable format. If 'verbose' is
specified, the output includes\n"

"   more and longer fields. If 'count' is
specified only the channel and call\n"

"   count is output.\n"

"   The 'concise' option is deprecated and will
be removed from future versions\n"

"   of Asterisk.\n";

return NULL;

 

case CLI_GENERATE:

return NULL;

}

 

if (a->argc == e->args) {

if (!strcasecmp(a->argv[e->args-1],"concise"))

concise = 1;

else if (!strcasecmp(a->argv[e->args-1],"verbose"))

verbose = 1;

else if (!strcasecmp(a->argv[e->args-1],"count"))

count = 1;

else

return CLI_SHOWUSAGE;

} else if (a->argc != e->args - 1)

return CLI_SHOWUSAGE;

 

if (!count) {

if (!concise && !verbose)

ast_cli(a->fd, FORMAT_STRING2, "Channel",
"Location", "State", "Application(Data)");

else if (verbose)

ast_cli(a->fd, VERBOSE_FORMAT_STRING2, "Channel",
"Context", "Extension", "Priority", "State", "Application", "Data",

"CallerID", "Duration", "Accountcode",
"PeerAccount", "BridgedTo");

}

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bryant
Zimmerman
Sent: Tuesday, January 24, 2012 4:29 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Is there a sip show equivelant.

 

Is there a way to get a parsable concise feed back from the "sip show peers"
command that is more like the "core show channels concise" command  The
issue is the sip show peers uses space delimiter to display the the list but
some feilds have values some times and not others.  If not what is the exact
character count for each column so I can pull back the data that way. 

Also I have been looking through the code trying to figure out exactly wich
fileds are present in the "core show channels concise" command.  It appears
that they are different from the "core show channels verbose" command. What
are the exact fields in the "core show channels concise" and why is there no
duration?

Thanks

Bryant

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Re: [asterisk-users] Is there a sip show equivelant.

2012-01-24 Thread Bryant Zimmerman

Is there a way to get a parsable concise feed back from the "sip show 
peers" command that is more like the "core show channels concise" command  
The issue is the sip show peers uses space delimiter to display the the 
list but some feilds have values some times and not others.  If not what is 
the exact character count for each column so I can pull back the data that 
way. 

Also I have been looking through the code trying to figure out exactly wich 
fileds are present in the "core show channels concise" command.  It appears 
that they are different from the "core show channels verbose" command. What 
are the exact fields in the "core show channels concise" and why is there 
no duration?

Thanks

Bryant
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Re: [asterisk-users] ConfBridge details

2012-01-24 Thread Kevin P. Fleming


On 01/24/2012 04:28 PM, Jeremy Kister wrote:

On 1/23/2012 3:53 PM, Jeremy Kister wrote:

What I'm trying to do is keep track of conferences that are used.


this seems to work:
[macro-confbridge-setup]
exten => s,1,Set(NUM=$[0${NUM} + 1]);
exten => s,n,Set(CONFNO=99${NUM})
exten => s,n,Set(CONFS=${SHELL(asterisk -rx "core show channels" | awk
'/ConfBridge/ { print $2 }' | awk -F@ '{ print $1 }' | sort | uniq |
grep ${CONFNO} )})
exten => s,n,GotoIf($["${CONFS}" = "${CONFNO}"]?1)
exten => s,n,Noop(got a new conference# ${CONFNO})

but there's got to be a better way than spawning X shell commands for
'asterisk -rx', right ?


Note that ConfBridge in Asterisk 1.8 is nearly 'experimental', and it 
was almost completely rewritten for Asterisk 10. There is gained a lot 
more functionality, and much more flexible configuration system, and 
dialplan functions that allow access to the sorts of information you are 
looking for.


In essence, I would suggest not spending too much time trying to work 
the Asterisk 1.8 version of ConfBridge into your dialplan/repertoire, 
unless you really need it. The version in Asterisk 10 is much, much better.


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

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Re: [asterisk-users] ConfBridge details

2012-01-24 Thread Jeremy Kister


On 1/23/2012 3:53 PM, Jeremy Kister wrote:

What I'm trying to do is keep track of conferences that are used.


this seems to work:
[macro-confbridge-setup]
exten => s,1,Set(NUM=$[0${NUM} + 1]);
exten => s,n,Set(CONFNO=99${NUM})
exten => s,n,Set(CONFS=${SHELL(asterisk -rx "core show channels" | awk 
'/ConfBridge/ { print $2 }' | awk -F@ '{ print $1 }' | sort | uniq | 
grep ${CONFNO} )})

exten => s,n,GotoIf($["${CONFS}" = "${CONFNO}"]?1)
exten => s,n,Noop(got a new conference# ${CONFNO})

but there's got to be a better way than spawning X shell commands for 
'asterisk -rx', right ?



--

Jeremy Kister
http://jeremy.kister.net./

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Re: [asterisk-users] Cordless SIP phone

2012-01-24 Thread Andreas Sikkema

On 1/23/12 4:39 PM, eherr wrote:
> Where I want to put the new on is outside the range.
> 
> I thought SIP cordless phones would be better on the range.

If you want to extend the range of a DECT basestation you can use
repeaters, but you then lose DECT encryption and you can only add up to
6 repeaters around one basestation.

Extending your range beyond that requires a proper DECT network and
brings you into a whole new cost level. But that can go up to 256x12
handsets and 256x8 (IIRC) simultaneous calls...

-- 
Andreas Sikkema

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Re: [asterisk-users] ChanSpy : how to know channel name ?

Did a little research on this using my Asterisk 10.0.  This should work for
you. 

exten => 1246,1,answer()

exten => 1246,n,set(inuse=${CHANNELS(miq8)})

exten => 1246,n,extenspy(${inuse}@default)

exten => 1246,n,hangup()

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Tuesday, January 24, 2012 9:52 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] ChanSpy : how to know channel name ?

 

Hello,

there is very little information about the function CHANNELS().

If I know the peer name (that I always know for sure), do you see a way of
using the function CHANNELS() to get the right channel ??

If CHANNELS() gives a "space-delimited" list of active channels, and I know
miq8... how can I get SIP/miq8-2419 ?

Thanks !


On 01/24/2012 04:46 PM, Danny Nicholas wrote: 

Extenspy(miq8@default) for miq8.  I would either proceed under the
assumption that I'm going to be listening to my extensions in the default
context or set up an AGI or something to load my needed ext@context
information.

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Tuesday, January 24, 2012 9:41 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] ChanSpy : how to know channel name ?

 

Hello,

how to use ExtenSpy(extension@context) when conversations are named like
this ? :

SIP/378680644-2 default  
SIP/rs4-2445 sub-uitinternation
SIP/3715320168-2 default
SIP/ibenla2-244 sub-uit789  
SIP/372083610-2 default  
SIP/cedhou0-24 sub-uit789  
SIP/travel3-2 pbx-routing 
SIP/INTELin-2 pbx-routing 
SIP/375382280-2 default   
SIP/miq8-2419  sub-uitGSM  
SIP/3749378004- default  
SIP/instlpr0-2 sub-uitinternation

Can you tell me what is the extension ? How will I know the context ? The
context is not always the same...



On 01/24/2012 04:32 PM, Danny Nicholas wrote: 

You are either going to be able to listen to SIP/miq8 or you are going to
have to know the sequence number like SIP/miq8-1.  Maybe you should just
use ExtenSpy instead?

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Tuesday, January 24, 2012 9:26 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] ChanSpy : how to know channel name ?

 

Of course I can control the name of my SIP-peer. Why do you tell me this ?!

Please answer my question : how do I know the channel name so I can ChanSpy
the correct channel ?



On 01/24/2012 04:13 PM, Danny Nicholas wrote: 

It's not random.  The "Channel Name" is Tech/peer-sequence (sequence is in
hex).  You can control (to a degree) the peer portion in
sip.conf/users.conf.

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Tuesday, January 24, 2012 9:07 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] ChanSpy : how to know channel name ?

 

Hello,

thanks. miq8 is the name of the SIP peer account.

So when I know the SIP peer name, and I strip of the numbers of the channel,
then I can use ChanSpy. So this answers my original question.

The only problem I see : it is Asterisk that gives the channel its name. How
do I change this ??

As far as I know, Asterisk randomly gives a channel name which consists of
the technology (SIP), the peername (miq8) and some numbers...

How to change the channel name ?



On 01/24/2012 03:53 PM, Danny Nicholas wrote: 

I would try chanspy(sip/miq8,b) - the b flag denotes to only listen to a
bridged call which (it seems to me) should pick up both sides.

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Tuesday, January 24, 2012 8:46 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] ChanSpy : how to know channel name ?

 

Hello,

OK thanks. But, I want to listen to the conversation (not just 1 channel out
of 2 channels). How then do I use ChanSpy ?



On 01/24/2012 03:41 PM, Danny Nicholas wrote: 

Strip off the -x.  Just listen to SIP/miq8 and SIP/375382280 in your
example.

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Tuesday, January 24, 2012 7:47 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] ChanSpy : how to know channel name ?

 

Hello list,

to use ChanSpy, one needs to know the name of the channel.

But on an incoming call from the provider, or an outgoing call to the
provider there are always numbers added. How can one then know the channel
name ??

core show channels verbo

Re: [asterisk-users] RFE idea for VM application

In the "classic" environment, you can move a "read" email from INBOX to OLD.
If the user thinks the content should be saved, that's the route I would
recommend.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Phil Daws
Sent: Tuesday, January 24, 2012 10:15 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] RFE idea for VM application

- Original Message -
> The RFE was to fix Asterisk, not the user who can't be taught.  If the 
> recipient can't/won't answer their phone and can't/won't read their 
> email, they are beyond help.
> 
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of A J 
> Stiles
> Sent: Tuesday, January 24, 2012 9:54 AM
> To: Phil Daws; Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] RFE idea for VM application
> 
> On Tuesday 24 January 2012, Phil Daws wrote:
> > This is in version 1.8 and 10.0 from what I can see. The problem is 
> > not that the caller is unaware of the recipients mailbox being full, 
> > as they do hear the message, but it is the recipient whom may be 
> > completely unaware. If a more verbose warning message was written 
> > out we could at least alert the user to the issue. They could then 
> > perform some timely and necessary mailbox clean up:)
> 
> Ah, but what you have here is a classic "break glass to release 
> hammer"
> situation.  The intended recipient doesn't know that their voicemail 
> box is full; but, evidently from the fact that they are accruing 
> voicemails at all, they must, for some unspecified reason, not be 
> contactable by telephone!
> 
> All you can do in that situation is mention in your "mailbox full"
> message
> an alternative means of contacting the intended recipient .
> 
> --
> AJS
> 

See my follow up email as they are reading the emails; just not deleting
them :)
--
Thanks, Phil

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Re: [asterisk-users] RFE idea for VM application

- Original Message -
> The RFE was to fix Asterisk, not the user who can't be taught.  If
> the
> recipient can't/won't answer their phone and can't/won't read their
> email,
> they are beyond help.
> 
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of A J
> Stiles
> Sent: Tuesday, January 24, 2012 9:54 AM
> To: Phil Daws; Asterisk Users Mailing List - Non-Commercial
> Discussion
> Subject: Re: [asterisk-users] RFE idea for VM application
> 
> On Tuesday 24 January 2012, Phil Daws wrote:
> > This is in version 1.8 and 10.0 from what I can see. The problem is
> > not that the caller is unaware of the recipients mailbox being
> > full,
> > as they do hear the message, but it is the recipient whom may be
> > completely unaware. If a more verbose warning message was written
> > out
> > we could at least alert the user to the issue. They could then
> > perform
> > some timely and necessary mailbox clean up:)
> 
> Ah, but what you have here is a classic "break glass to release
> hammer"
> situation.  The intended recipient doesn't know that their voicemail
> box is
> full; but, evidently from the fact that they are accruing voicemails
> at all,
> they must, for some unspecified reason, not be contactable by
> telephone!
> 
> All you can do in that situation is mention in your "mailbox full"
> message
> an alternative means of contacting the intended recipient .
> 
> --
> AJS
> 

See my follow up email as they are reading the emails; just not deleting them :)
-- 
Thanks, Phil

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Re: [asterisk-users] RFE idea for VM application

- Original Message -
> On Tuesday 24 January 2012, Phil Daws wrote:
> > This is in version 1.8 and 10.0 from what I can see. The problem is
> > not
> > that the caller is unaware of the recipients mailbox being full, as
> > they
> > do hear the message, but it is the recipient whom may be completely
> > unaware. If a more verbose warning message was written out we could
> > at
> > least alert the user to the issue. They could then perform some
> > timely and
> > necessary mailbox clean up:)
> 
> Ah, but what you have here is a classic "break glass to release
> hammer"
> situation.  The intended recipient doesn't know that their voicemail
> box is
> full; but, evidently from the fact that they are accruing voicemails
> at all,
> they must, for some unspecified reason, not be contactable by
> telephone!
> 
> All you can do in that situation is mention in your "mailbox full"
> message an
> alternative means of contacting the intended recipient .
> 
> --
> AJS
> 
> Answers come *after* questions.
> 

In the original post I mentioned that we are using IMAP (Zimbra) as the 
back-end VM storage, and what is happening is that users listen to their 
messages (marked as read) but then hang on to them for later use; or tardy 
mailbox management ;) Asterisk does not care about that and just counts up how 
many it can find, which I am guessing is by inspecting the inserted header in 
the message.

Yes there is an element of user education that needs to take place but as 
always not everybody remembers and we would be prefer to be alerted to that 
fact instead of the recipient not get their messages.

One thing I have not checked though is what happens if one user forwards a VM, 
from their IMAP mailbox, to another user ? Does it count against the recipient 
message count tally ?
-- 
Thanks, Phil


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Re: [asterisk-users] RFE idea for VM application

The RFE was to fix Asterisk, not the user who can't be taught.  If the
recipient can't/won't answer their phone and can't/won't read their email,
they are beyond help.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of A J Stiles
Sent: Tuesday, January 24, 2012 9:54 AM
To: Phil Daws; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] RFE idea for VM application

On Tuesday 24 January 2012, Phil Daws wrote:
> This is in version 1.8 and 10.0 from what I can see. The problem is 
> not that the caller is unaware of the recipients mailbox being full, 
> as they do hear the message, but it is the recipient whom may be 
> completely unaware. If a more verbose warning message was written out 
> we could at least alert the user to the issue. They could then perform 
> some timely and necessary mailbox clean up:)

Ah, but what you have here is a classic "break glass to release hammer" 
situation.  The intended recipient doesn't know that their voicemail box is
full; but, evidently from the fact that they are accruing voicemails at all,
they must, for some unspecified reason, not be contactable by telephone!

All you can do in that situation is mention in your "mailbox full" message
an alternative means of contacting the intended recipient .

--
AJS

Answers come *after* questions.

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Re: [asterisk-users] asterisk does not detect menus

2012-01-24 Thread motty.cruz

I found the fix; in my sip.cfg I changed the following line from 
mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling
Sent: Monday, January 23, 2012 11:03 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] asterisk does not detect menus

We had similar problems, updating to the latest 1.8.x seems to have solved
the issue for at least one number we were having issues with.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of motty.cruz
Sent: Monday, January 23, 2012 2:00 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] asterisk does not detect menus

Hello,
When I called companies with auto animate menus my system does not seem to
detect menus on ther other side. For instance I called this number (407)
886-3338 when I input the ext. number of any option on the list I don't get
a response however if I called the same number from my google account or my
cell phone number it works fine meaning I can select any option or input ext
number. 

I'm using Asterisk 1.8.4 on Centos 5, 

Any suggestions are welcome. 

Thanks,
Motty,

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Re: [asterisk-users] ChanSpy : how to know channel name ?


Hello,

there is very little information about the function CHANNELS().

If I know the peer name (that I always know for sure), do you see a way 
of using the function CHANNELS() to get the right channel ??


If CHANNELS() gives a "space-delimited" list of active channels, and I 
know miq8... how can I get /SIP/miq8-2419/ ?


Thanks !


On 01/24/2012 04:46 PM, Danny Nicholas wrote:


Extenspy(miq8@default) for miq8.  I would either proceed under the 
assumption that I'm going to be listening to my extensions in the 
default context or set up an AGI or something to load my needed 
ext@context information.


*From:*asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas 
Kellens

*Sent:* Tuesday, January 24, 2012 9:41 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] ChanSpy : how to know channel name ?

Hello,

how to use ExtenSpy(extension@context) when conversations are named 
like this ? :


/SIP/*378680644-2* default
SIP/*rs4-2445* sub-uitinternation
SIP/*3715320168-2* default
SIP*/ibenla2-244* sub-uit789
SIP/*372083610-2* default
SIP/*cedhou0-24* sub-uit789
SIP*/travel3-2* pbx-routing
SIP/*INTELin-2* pbx-routing
SIP/*375382280-2* default
SIP/*miq8-2419*  sub-uitGSM
SIP/*3749378004-* default
SIP*/instlpr0-2* sub-uitinternation
/
Can you tell me what is the extension ? How will I know the context ? 
The context is not always the same...




On 01/24/2012 04:32 PM, Danny Nicholas wrote:

You are either going to be able to listen to SIP/miq8 or you are going 
to have to know the sequence number like SIP/miq8-1.  Maybe you 
should just use ExtenSpy instead?


*From:*asterisk-users-boun...@lists.digium.com 
 
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas 
Kellens

*Sent:* Tuesday, January 24, 2012 9:26 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] ChanSpy : how to know channel name ?

Of course I can control the name of my SIP-peer. Why do you tell me 
this ?!


Please answer my question : how do I know the channel name so I can 
ChanSpy the correct channel ?




On 01/24/2012 04:13 PM, Danny Nicholas wrote:

It's not random.  The "Channel Name" is Tech/peer-sequence (sequence 
is in hex).  You can control (to a degree) the peer portion in 
sip.conf/users.conf.


*From:*asterisk-users-boun...@lists.digium.com 
 
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas 
Kellens

*Sent:* Tuesday, January 24, 2012 9:07 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] ChanSpy : how to know channel name ?

Hello,

thanks. miq8 is the name of the SIP peer account.

So when I know the SIP peer name, and I strip of the numbers of the 
channel, then I can use ChanSpy. So this answers my original question.


The only problem I see : it is Asterisk that gives the channel its 
name. How do I change this ??


As far as I know, Asterisk randomly gives a channel name which 
consists of the technology (SIP), the peername (miq8) and some numbers...


How to change the channel name ?



On 01/24/2012 03:53 PM, Danny Nicholas wrote:

I would try chanspy(sip/miq8,b) -- the b flag denotes to only listen 
to a bridged call which (it seems to me) should pick up both sides.


*From:*asterisk-users-boun...@lists.digium.com 
 
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas 
Kellens

*Sent:* Tuesday, January 24, 2012 8:46 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] ChanSpy : how to know channel name ?

Hello,

OK thanks. But, I want to listen to the conversation (not just 1 
channel out of 2 channels). How then do I use ChanSpy ?




On 01/24/2012 03:41 PM, Danny Nicholas wrote:

Strip off the --x.  Just listen to SIP/miq8 and SIP/375382280 in 
your example.


*From:*asterisk-users-boun...@lists.digium.com 
 
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas 
Kellens

*Sent:* Tuesday, January 24, 2012 7:47 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] ChanSpy : how to know channel name ?

Hello list,

to use ChanSpy, one needs to know the name of the channel.

But on an incoming call from the provider, or an outgoing call to the 
provider there are always numbers added. How can one then know the 
channel name ??


/core show channels verbose/ shows me for example :

/SIP/*378680644-2* default
SIP/*rs4-2445* sub-uitinternation
SIP/*3715320168-2* default
SIP*/ibenla2-244* sub-uit789
SIP/*372083610-2* default
SIP/*cedhou0-24* sub-uit789
SIP*/travel3-2* pbx-routing
SIP/*INTELin-2* pbx-

Re: [asterisk-users] RFE idea for VM application

2012-01-24 Thread A J Stiles

On Tuesday 24 January 2012, Phil Daws wrote:
> This is in version 1.8 and 10.0 from what I can see. The problem is not
> that the caller is unaware of the recipients mailbox being full, as they
> do hear the message, but it is the recipient whom may be completely
> unaware. If a more verbose warning message was written out we could at
> least alert the user to the issue. They could then perform some timely and
> necessary mailbox clean up:)

Ah, but what you have here is a classic "break glass to release hammer" 
situation.  The intended recipient doesn't know that their voicemail box is 
full; but, evidently from the fact that they are accruing voicemails at all, 
they must, for some unspecified reason, not be contactable by telephone!

All you can do in that situation is mention in your "mailbox full" message an 
alternative means of contacting the intended recipient .

-- 
AJS

Answers come *after* questions.

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Re: [asterisk-users] ChanSpy : how to know channel name ?

Extenspy(miq8@default) for miq8.  I would either proceed under the
assumption that I'm going to be listening to my extensions in the default
context or set up an AGI or something to load my needed ext@context
information.

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Tuesday, January 24, 2012 9:41 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] ChanSpy : how to know channel name ?

 

Hello,

how to use ExtenSpy(extension@context) when conversations are named like
this ? :

SIP/378680644-2 default  
SIP/rs4-2445 sub-uitinternation
SIP/3715320168-2 default
SIP/ibenla2-244 sub-uit789  
SIP/372083610-2 default  
SIP/cedhou0-24 sub-uit789  
SIP/travel3-2 pbx-routing 
SIP/INTELin-2 pbx-routing 
SIP/375382280-2 default   
SIP/miq8-2419  sub-uitGSM  
SIP/3749378004- default  
SIP/instlpr0-2 sub-uitinternation

Can you tell me what is the extension ? How will I know the context ? The
context is not always the same...



On 01/24/2012 04:32 PM, Danny Nicholas wrote: 

You are either going to be able to listen to SIP/miq8 or you are going to
have to know the sequence number like SIP/miq8-1.  Maybe you should just
use ExtenSpy instead?

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Tuesday, January 24, 2012 9:26 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] ChanSpy : how to know channel name ?

 

Of course I can control the name of my SIP-peer. Why do you tell me this ?!

Please answer my question : how do I know the channel name so I can ChanSpy
the correct channel ?



On 01/24/2012 04:13 PM, Danny Nicholas wrote: 

It's not random.  The "Channel Name" is Tech/peer-sequence (sequence is in
hex).  You can control (to a degree) the peer portion in
sip.conf/users.conf.

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Tuesday, January 24, 2012 9:07 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] ChanSpy : how to know channel name ?

 

Hello,

thanks. miq8 is the name of the SIP peer account.

So when I know the SIP peer name, and I strip of the numbers of the channel,
then I can use ChanSpy. So this answers my original question.

The only problem I see : it is Asterisk that gives the channel its name. How
do I change this ??

As far as I know, Asterisk randomly gives a channel name which consists of
the technology (SIP), the peername (miq8) and some numbers...

How to change the channel name ?



On 01/24/2012 03:53 PM, Danny Nicholas wrote: 

I would try chanspy(sip/miq8,b) - the b flag denotes to only listen to a
bridged call which (it seems to me) should pick up both sides.

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Tuesday, January 24, 2012 8:46 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] ChanSpy : how to know channel name ?

 

Hello,

OK thanks. But, I want to listen to the conversation (not just 1 channel out
of 2 channels). How then do I use ChanSpy ?



On 01/24/2012 03:41 PM, Danny Nicholas wrote: 

Strip off the -x.  Just listen to SIP/miq8 and SIP/375382280 in your
example.

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Tuesday, January 24, 2012 7:47 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] ChanSpy : how to know channel name ?

 

Hello list,

to use ChanSpy, one needs to know the name of the channel.

But on an incoming call from the provider, or an outgoing call to the
provider there are always numbers added. How can one then know the channel
name ??

core show channels verbose shows me for example :

SIP/378680644-2 default  
SIP/rs4-2445 sub-uitinternation
SIP/3715320168-2 default
SIP/ibenla2-244 sub-uit789  
SIP/372083610-2 default  
SIP/cedhou0-24 sub-uit789  
SIP/travel3-2 pbx-routing 
SIP/INTELin-2 pbx-routing 
SIP/375382280-2 default   
SIP/miq8-2419  sub-uitGSM  
SIP/3749378004- default  
SIP/instlpr0-2 sub-uitinternation
SIP/372089170-2 default  
SIP/v9q9uLT- from-GFATRUNK 
46 active channels
24 active calls


If I want to listen to the conversation of SIP/miq8-2419 and
SIP/375382280-2 (these 2 channels have been connected to 1
conversation), how do I use ChanSpy ??



Kind regards;
Jonas.

 
 
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New to Asterisk? Join

Re: [asterisk-users] ChanSpy : how to know channel name ?


Hello,

how to use ExtenSpy(extension@context) when conversations are named like 
this ? :


/SIP/*378680644-2* default
SIP/*rs4-2445* sub-uitinternation
SIP/*3715320168-2* default
SIP*/ibenla2-244* sub-uit789
SIP/*372083610-2* default
SIP/*cedhou0-24* sub-uit789
SIP*/travel3-2* pbx-routing
SIP/*INTELin-2* pbx-routing
SIP/*375382280-2* default
SIP/*miq8-2419*  sub-uitGSM
SIP/*3749378004-* default
SIP*/instlpr0-2* sub-uitinternation
/
Can you tell me what is the extension ? How will I know the context ? 
The context is not always the same...




On 01/24/2012 04:32 PM, Danny Nicholas wrote:


You are either going to be able to listen to SIP/miq8 or you are going 
to have to know the sequence number like SIP/miq8-1.  Maybe you 
should just use ExtenSpy instead?


*From:*asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas 
Kellens

*Sent:* Tuesday, January 24, 2012 9:26 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] ChanSpy : how to know channel name ?

Of course I can control the name of my SIP-peer. Why do you tell me 
this ?!


Please answer my question : how do I know the channel name so I can 
ChanSpy the correct channel ?




On 01/24/2012 04:13 PM, Danny Nicholas wrote:

It's not random.  The "Channel Name" is Tech/peer-sequence (sequence 
is in hex).  You can control (to a degree) the peer portion in 
sip.conf/users.conf.


*From:*asterisk-users-boun...@lists.digium.com 
 
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas 
Kellens

*Sent:* Tuesday, January 24, 2012 9:07 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] ChanSpy : how to know channel name ?

Hello,

thanks. miq8 is the name of the SIP peer account.

So when I know the SIP peer name, and I strip of the numbers of the 
channel, then I can use ChanSpy. So this answers my original question.


The only problem I see : it is Asterisk that gives the channel its 
name. How do I change this ??


As far as I know, Asterisk randomly gives a channel name which 
consists of the technology (SIP), the peername (miq8) and some numbers...


How to change the channel name ?



On 01/24/2012 03:53 PM, Danny Nicholas wrote:

I would try chanspy(sip/miq8,b) -- the b flag denotes to only listen 
to a bridged call which (it seems to me) should pick up both sides.


*From:*asterisk-users-boun...@lists.digium.com 
 
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas 
Kellens

*Sent:* Tuesday, January 24, 2012 8:46 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] ChanSpy : how to know channel name ?

Hello,

OK thanks. But, I want to listen to the conversation (not just 1 
channel out of 2 channels). How then do I use ChanSpy ?




On 01/24/2012 03:41 PM, Danny Nicholas wrote:

Strip off the --x.  Just listen to SIP/miq8 and SIP/375382280 in 
your example.


*From:*asterisk-users-boun...@lists.digium.com 
 
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas 
Kellens

*Sent:* Tuesday, January 24, 2012 7:47 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] ChanSpy : how to know channel name ?

Hello list,

to use ChanSpy, one needs to know the name of the channel.

But on an incoming call from the provider, or an outgoing call to the 
provider there are always numbers added. How can one then know the 
channel name ??


/core show channels verbose/ shows me for example :

/SIP/*378680644-2* default
SIP/*rs4-2445* sub-uitinternation
SIP/*3715320168-2* default
SIP*/ibenla2-244* sub-uit789
SIP/*372083610-2* default
SIP/*cedhou0-24* sub-uit789
SIP*/travel3-2* pbx-routing
SIP/*INTELin-2* pbx-routing
SIP/*375382280-2* default
SIP/*miq8-2419*  sub-uitGSM
SIP/*3749378004-* default
SIP*/instlpr0-2* sub-uitinternation
SIP/*372089170-2* default
SIP/*v9q9uLT-* from-GFATRUNK
46 active channels
24 active calls/


If I want to listen to the conversation of /SIP/*miq8-2419*/ and 
/SIP/*375382280-2*/ (these 2 channels have been connected to 1 
conversation), how do I use ChanSpy ??




Kind regards;
Jonas.

  
  
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Re: [asterisk-users] allowguest = yes? no?

2012-01-24 Thread Gilles

On Tue, 24 Jan 2012 09:26:26 -0600, "Kevin P. Fleming"
 wrote:
>By definition this is impossible. If the caller is a 'stranger', that 
>means you have no knowledge of them prior to their INVITE request 
>arriving at your server. If you have no knowledge of them, then you 
>don't have any 'shared secret', and thus they cannot authenticate to 
>your server.

Mmm, so if I want to allow strangers to call us over the Net, I must
1. allowgues=yes
2. make sure the context they enter will not allow them to make calls
through the PSTN, either directly (through our plug in the wall) or
indirectly (through an ITSP).

Thank you.


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Re: [asterisk-users] ChanSpy : how to know channel name ?

You are either going to be able to listen to SIP/miq8 or you are going to
have to know the sequence number like SIP/miq8-1.  Maybe you should just
use ExtenSpy instead?

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Tuesday, January 24, 2012 9:26 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] ChanSpy : how to know channel name ?

Of course I can control the name of my SIP-peer. Why do you tell me this ?!

Please answer my question : how do I know the channel name so I can ChanSpy
the correct channel ?

On 01/24/2012 04:13 PM, Danny Nicholas wrote: 

It's not random.  The "Channel Name" is Tech/peer-sequence (sequence is in
hex).  You can control (to a degree) the peer portion in
sip.conf/users.conf.

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Tuesday, January 24, 2012 9:07 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] ChanSpy : how to know channel name ?

Hello,

thanks. miq8 is the name of the SIP peer account.

So when I know the SIP peer name, and I strip of the numbers of the channel,
then I can use ChanSpy. So this answers my original question.

The only problem I see : it is Asterisk that gives the channel its name. How
do I change this ??

As far as I know, Asterisk randomly gives a channel name which consists of
the technology (SIP), the peername (miq8) and some numbers...

How to change the channel name ?

On 01/24/2012 03:53 PM, Danny Nicholas wrote: 

I would try chanspy(sip/miq8,b) - the b flag denotes to only listen to a
bridged call which (it seems to me) should pick up both sides.

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Tuesday, January 24, 2012 8:46 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] ChanSpy : how to know channel name ?

Hello,

OK thanks. But, I want to listen to the conversation (not just 1 channel out
of 2 channels). How then do I use ChanSpy ?

On 01/24/2012 03:41 PM, Danny Nicholas wrote: 

Strip off the -x.  Just listen to SIP/miq8 and SIP/375382280 in your
example.

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Tuesday, January 24, 2012 7:47 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] ChanSpy : how to know channel name ?

Hello list,

to use ChanSpy, one needs to know the name of the channel.

But on an incoming call from the provider, or an outgoing call to the
provider there are always numbers added. How can one then know the channel
name ??

core show channels verbose shows me for example :

SIP/378680644-2 default  
SIP/rs4-2445 sub-uitinternation
SIP/3715320168-2 default
SIP/ibenla2-244 sub-uit789  
SIP/372083610-2 default  
SIP/cedhou0-24 sub-uit789  
SIP/travel3-2 pbx-routing 
SIP/INTELin-2 pbx-routing 
SIP/375382280-2 default   
SIP/miq8-2419  sub-uitGSM  
SIP/3749378004- default  
SIP/instlpr0-2 sub-uitinternation
SIP/372089170-2 default  
SIP/v9q9uLT- from-GFATRUNK 
46 active channels
24 active calls

If I want to listen to the conversation of SIP/miq8-2419 and
SIP/375382280-2 (these 2 channels have been connected to 1
conversation), how do I use ChanSpy ??

Kind regards;
Jonas.

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Re: [asterisk-users] RFE idea for VM application

Hi Danny, 

Yes I did think about a stand-daemon to do exactly that; and on a side note was 
even considering using Node.JS :) Though the rationale for considering making 
the change in app_voicemail.c is that for it to alert on maxmsg it must have 
already calculated the number of messages in the first place so no real 
overhead should be introduced. Your suggestion makes perfect sense, and one I 
had discussed with a colleague, so will have a hack this evening on V10 and see 
if it would be easy to implement and back port. 

-- 
Thanks, Phil 

- Original Message -

> I would personally rather use a stand-alone daemon to query the
> mailboxes and send an email to the box owner when he or she reaches
> a tolerance level rather than depend on an overloaded application
> that is running God-only-knows what modifications to the original
> intent (IMAP, Real-Time, Active Directory, any other monkey wrench
> someone might throw at the original idea of a text file and wav on
> the actual host). Voicemail.conf presumably has the email of the box
> owner and the number of messages they can receive. I would suggest
> adding a warnmsg= to voicemail.conf to work in tandem with maxmsg=.
> Warnmsg could be a count of messages to send warning at or a
> percentage of maxmsg. If I ever get a free day, I might try this.

> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Phil
> Daws
> Sent: Tuesday, January 24, 2012 8:56 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] RFE idea for VM application

> This is in version 1.8 and 10.0 from what I can see. The problem is
> not that the caller is unaware of the recipients mailbox being full,
> as they do hear the message, but it is the recipient whom may be
> completely unaware. If a more verbose warning message was written
> out we could at least alert the user to the issue. They could then
> perform some timely and necessary mailbox clean up:)

> --
> Thanks, Phil

> - Original Message -

> > You don’t state which version this is for, but it seems like a
> > simple
> > patch would be for voicemail to play sorry-mailbox-full.wav
> > (standard sound). In lieu of all that, you could do a
> > quick-and-dirty AGI to read /v/l/a/m and play the message back
> > since
> > voicemail is one of the larger modules and therefore prone to a
> > better chance of error introduction on an RFE. PSOT – the Unix Bot
> > thing was pretty cool.
> 

> > From: asterisk-users-boun...@lists.digium.com
> > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Phil
> > Daws
> 
> > Sent: Tuesday, January 24, 2012 8:30 AM
> 
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> 
> > Subject: [asterisk-users] RFE idea for VM application
> 

> > Hello all,
> 

> > we are using IMAP for the storage of VMs and had a user yesterday
> > his
> > their maxmsg limit (default 100) and was wondering why nobody could
> > leave them messages. I see in /var/log/asterisk/messages that it
> > does write out a warning message of:
> 

> > ast_log(LOG_WARNING, "Unable to leave message since we will exceed
> > the maximum number of messages allowed (%u >= %u)\n", msgnum,
> > vmu->maxmsg);
> 

> > but I was wondering how feasible it would be to modify the code to
> > add:
> 

> > 1) the mailbox name of the user whom has hit the limit
> 
> > 2) a warning/critical threshold that the user is getting close to
> > the
> > limit
> 

> > using whatever monitoring tool one has available eg. OSSEC the
> > alert
> > could be trapped and the user notified.
> 

> > Is this worthy of an RFE? and possible help from people if I try
> > and
> > create the patch myself ?
> 

> > Thoughts or insults ?
> 

> > --
> 
> > Thanks, Phil
> 

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Re: [asterisk-users] allowguest = yes? no?

2012-01-24 Thread Kevin P. Fleming


On 01/24/2012 09:03 AM, Gilles wrote:

On Tue, 24 Jan 2012 09:55:12 -0500, Jim DeVito
  wrote:

What they are talking about is SIP URI dialling. Let say you have
extension 1000 the rings a phone on your system. With allowguest=yes I
would be allowed to dial SIP:/1...@yourdomain.com and assuming the
context defined in your [General] section had access to exten 1000 I
would connect to that phone. With alloweguest=no my call would be rejected.


Thanks for the clarification.

Provided I do want strangers to call extensions through an SIP URI
instead of using the PSTN, how can I raise security by requiring that
they authenticate?


By definition this is impossible. If the caller is a 'stranger', that 
means you have no knowledge of them prior to their INVITE request 
arriving at your server. If you have no knowledge of them, then you 
don't have any 'shared secret', and thus they cannot authenticate to 
your server.


--
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Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com & www.asterisk.org

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Re: [asterisk-users] ChanSpy : how to know channel name ?


Of course I can control the name of my SIP-peer. Why do you tell me this ?!

Please answer my question : how do I know the channel name so I can 
ChanSpy the correct channel ?




On 01/24/2012 04:13 PM, Danny Nicholas wrote:


It's not random.  The "Channel Name" is Tech/peer-sequence (sequence 
is in hex).  You can control (to a degree) the peer portion in 
sip.conf/users.conf.


*From:*asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas 
Kellens

*Sent:* Tuesday, January 24, 2012 9:07 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] ChanSpy : how to know channel name ?

Hello,

thanks. miq8 is the name of the SIP peer account.

So when I know the SIP peer name, and I strip of the numbers of the 
channel, then I can use ChanSpy. So this answers my original question.


The only problem I see : it is Asterisk that gives the channel its 
name. How do I change this ??


As far as I know, Asterisk randomly gives a channel name which 
consists of the technology (SIP), the peername (miq8) and some numbers...


How to change the channel name ?



On 01/24/2012 03:53 PM, Danny Nicholas wrote:

I would try chanspy(sip/miq8,b) -- the b flag denotes to only listen 
to a bridged call which (it seems to me) should pick up both sides.


*From:*asterisk-users-boun...@lists.digium.com 
 
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas 
Kellens

*Sent:* Tuesday, January 24, 2012 8:46 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] ChanSpy : how to know channel name ?

Hello,

OK thanks. But, I want to listen to the conversation (not just 1 
channel out of 2 channels). How then do I use ChanSpy ?




On 01/24/2012 03:41 PM, Danny Nicholas wrote:

Strip off the --x.  Just listen to SIP/miq8 and SIP/375382280 in 
your example.


*From:*asterisk-users-boun...@lists.digium.com 
 
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas 
Kellens

*Sent:* Tuesday, January 24, 2012 7:47 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] ChanSpy : how to know channel name ?

Hello list,

to use ChanSpy, one needs to know the name of the channel.

But on an incoming call from the provider, or an outgoing call to the 
provider there are always numbers added. How can one then know the 
channel name ??


/core show channels verbose/ shows me for example :

/SIP/*378680644-2* default
SIP/*rs4-2445* sub-uitinternation
SIP/*3715320168-2* default
SIP*/ibenla2-244* sub-uit789
SIP/*372083610-2* default
SIP/*cedhou0-24* sub-uit789
SIP*/travel3-2* pbx-routing
SIP/*INTELin-2* pbx-routing
SIP/*375382280-2* default
SIP/*miq8-2419*  sub-uitGSM
SIP/*3749378004-* default
SIP*/instlpr0-2* sub-uitinternation
SIP/*372089170-2* default
SIP/*v9q9uLT-* from-GFATRUNK
46 active channels
24 active calls/


If I want to listen to the conversation of /SIP/*miq8-2419*/ and 
/SIP/*375382280-2*/ (these 2 channels have been connected to 1 
conversation), how do I use ChanSpy ??




Kind regards;
Jonas.

  
  
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Re: [asterisk-users] ChanSpy : how to know channel name ?

It's not random.  The "Channel Name" is Tech/peer-sequence (sequence is in
hex).  You can control (to a degree) the peer portion in
sip.conf/users.conf.

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Tuesday, January 24, 2012 9:07 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] ChanSpy : how to know channel name ?

Hello,

thanks. miq8 is the name of the SIP peer account.

So when I know the SIP peer name, and I strip of the numbers of the channel,
then I can use ChanSpy. So this answers my original question.

The only problem I see : it is Asterisk that gives the channel its name. How
do I change this ??

As far as I know, Asterisk randomly gives a channel name which consists of
the technology (SIP), the peername (miq8) and some numbers...

How to change the channel name ?

On 01/24/2012 03:53 PM, Danny Nicholas wrote: 

I would try chanspy(sip/miq8,b) - the b flag denotes to only listen to a
bridged call which (it seems to me) should pick up both sides.

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Tuesday, January 24, 2012 8:46 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] ChanSpy : how to know channel name ?

Hello,

OK thanks. But, I want to listen to the conversation (not just 1 channel out
of 2 channels). How then do I use ChanSpy ?

On 01/24/2012 03:41 PM, Danny Nicholas wrote: 

Strip off the -x.  Just listen to SIP/miq8 and SIP/375382280 in your
example.

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Tuesday, January 24, 2012 7:47 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] ChanSpy : how to know channel name ?

Hello list,

to use ChanSpy, one needs to know the name of the channel.

But on an incoming call from the provider, or an outgoing call to the
provider there are always numbers added. How can one then know the channel
name ??

core show channels verbose shows me for example :

SIP/378680644-2 default  
SIP/rs4-2445 sub-uitinternation
SIP/3715320168-2 default
SIP/ibenla2-244 sub-uit789  
SIP/372083610-2 default  
SIP/cedhou0-24 sub-uit789  
SIP/travel3-2 pbx-routing 
SIP/INTELin-2 pbx-routing 
SIP/375382280-2 default   
SIP/miq8-2419  sub-uitGSM  
SIP/3749378004- default  
SIP/instlpr0-2 sub-uitinternation
SIP/372089170-2 default  
SIP/v9q9uLT- from-GFATRUNK 
46 active channels
24 active calls

If I want to listen to the conversation of SIP/miq8-2419 and
SIP/375382280-2 (these 2 channels have been connected to 1
conversation), how do I use ChanSpy ??

Kind regards;
Jonas.

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Re: [asterisk-users] RFE idea for VM application

I would personally rather use a stand-alone daemon to query the mailboxes and 
send an email to the box owner when he or she reaches a tolerance level rather 
than depend on an overloaded application that is running God-only-knows what 
modifications to the original intent (IMAP, Real-Time, Active Directory, any 
other monkey wrench someone might throw at the original idea of a text file and 
wav on the actual host).  Voicemail.conf presumably has the email of the box 
owner and the number of messages they can receive.  I would suggest adding a 
warnmsg= to voicemail.conf to work in tandem with maxmsg=.   Warnmsg could be a 
count of messages to send warning at or a percentage of maxmsg.  If I ever get 
a free day, I might try this.

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Phil Daws
Sent: Tuesday, January 24, 2012 8:56 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] RFE idea for VM application

 

This is in version 1.8 and 10.0 from what I can see. The problem is not that 
the caller is unaware of the recipients mailbox being full, as they do hear the 
message, but it is the recipient whom may be completely unaware.  If a more 
verbose warning message was written out we could at least alert the user to the 
issue.  They could then perform some timely and necessary mailbox clean up:)

-- 
Thanks, Phil

 

  _  

You don’t state which version this is for, but it seems like a simple patch 
would be for voicemail to play sorry-mailbox-full.wav (standard sound).  In 
lieu of all that, you could do a quick-and-dirty AGI to read /v/l/a/m and play 
the message back since voicemail is one of the larger modules and therefore 
prone to a better chance of error introduction on an RFE.  PSOT – the Unix Bot 
thing was pretty cool.

 

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Phil Daws
Sent: Tuesday, January 24, 2012 8:30 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] RFE idea for VM application

 

Hello all,

we are using IMAP for the storage of VMs and had a user yesterday his their 
maxmsg limit (default 100) and was wondering why nobody could leave them 
messages.  I see in /var/log/asterisk/messages that it does write out a warning 
message of:

ast_log(LOG_WARNING, "Unable to leave message since we will exceed the maximum 
number of messages allowed (%u >= %u)\n", msgnum, vmu->maxmsg);

but I was wondering how feasible it would be to modify the code to add:

1) the mailbox name of the user whom has hit the limit
2) a warning/critical threshold that the user is getting close to the limit

using whatever monitoring tool one has available eg. OSSEC the alert could be 
trapped and the user notified.

Is this worthy of an RFE? and possible help from people if I try and create the 
patch myself ?

Thoughts or insults ?

-- 
Thanks, Phil

 


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Re: [asterisk-users] ChanSpy : how to know channel name ?


Hello,

thanks. miq8 is the name of the SIP peer account.

So when I know the SIP peer name, and I strip of the numbers of the 
channel, then I can use ChanSpy. So this answers my original question.


The only problem I see : it is Asterisk that gives the channel its name. 
How do I change this ??


As far as I know, Asterisk randomly gives a channel name which consists 
of the technology (SIP), the peername (miq8) and some numbers...


How to change the channel name ?



On 01/24/2012 03:53 PM, Danny Nicholas wrote:


I would try chanspy(sip/miq8,b) -- the b flag denotes to only listen 
to a bridged call which (it seems to me) should pick up both sides.


*From:*asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas 
Kellens

*Sent:* Tuesday, January 24, 2012 8:46 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] ChanSpy : how to know channel name ?

Hello,

OK thanks. But, I want to listen to the conversation (not just 1 
channel out of 2 channels). How then do I use ChanSpy ?




On 01/24/2012 03:41 PM, Danny Nicholas wrote:

Strip off the --x.  Just listen to SIP/miq8 and SIP/375382280 in 
your example.


*From:*asterisk-users-boun...@lists.digium.com 
 
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas 
Kellens

*Sent:* Tuesday, January 24, 2012 7:47 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] ChanSpy : how to know channel name ?

Hello list,

to use ChanSpy, one needs to know the name of the channel.

But on an incoming call from the provider, or an outgoing call to the 
provider there are always numbers added. How can one then know the 
channel name ??


/core show channels verbose/ shows me for example :

/SIP/*378680644-2* default
SIP/*rs4-2445* sub-uitinternation
SIP/*3715320168-2* default
SIP*/ibenla2-244* sub-uit789
SIP/*372083610-2* default
SIP/*cedhou0-24* sub-uit789
SIP*/travel3-2* pbx-routing
SIP/*INTELin-2* pbx-routing
SIP/*375382280-2* default
SIP/*miq8-2419*  sub-uitGSM
SIP/*3749378004-* default
SIP*/instlpr0-2* sub-uitinternation
SIP/*372089170-2* default
SIP/*v9q9uLT-* from-GFATRUNK
46 active channels
24 active calls/


If I want to listen to the conversation of /SIP/*miq8-2419*/ and 
/SIP/*375382280-2*/ (these 2 channels have been connected to 1 
conversation), how do I use ChanSpy ??




Kind regards;
Jonas.

  
  
--

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Re: [asterisk-users] allowguest = yes? no?

2012-01-24 Thread Gilles

On Tue, 24 Jan 2012 09:55:12 -0500, Jim DeVito
 wrote:
>What they are talking about is SIP URI dialling. Let say you have 
>extension 1000 the rings a phone on your system. With allowguest=yes I 
>would be allowed to dial SIP:/1...@yourdomain.com and assuming the 
>context defined in your [General] section had access to exten 1000 I 
>would connect to that phone. With alloweguest=no my call would be rejected.

Thanks for the clarification.

Provided I do want strangers to call extensions through an SIP URI
instead of using the PSTN, how can I raise security by requiring that
they authenticate?

Of do you mean that the choice is between
- don't allow SIP URI at all (allowguest=no), so strangers can reach
extensions only through the PSTN (but it's a waste of money)
- allow SIP URI (allowguess=yes) and make sure the context doesn't
allow making calls to the PSTN?


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Re: [asterisk-users] RFE idea for VM application

This is in version 1.8 and 10.0 from what I can see. The problem is not that 
the caller is unaware of the recipients mailbox being full, as they do hear the 
message, but it is the recipient whom may be completely unaware. If a more 
verbose warning message was written out we could at least alert the user to the 
issue. They could then perform some timely and necessary mailbox clean up:) 

-- 
Thanks, Phil 

- Original Message -

> You don’t state which version this is for, but it seems like a simple
> patch would be for voicemail to play sorry-mailbox-full.wav
> (standard sound). In lieu of all that, you could do a
> quick-and-dirty AGI to read /v/l/a/m and play the message back since
> voicemail is one of the larger modules and therefore prone to a
> better chance of error introduction on an RFE. PSOT – the Unix Bot
> thing was pretty cool.

> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Phil
> Daws
> Sent: Tuesday, January 24, 2012 8:30 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] RFE idea for VM application

> Hello all,

> we are using IMAP for the storage of VMs and had a user yesterday his
> their maxmsg limit (default 100) and was wondering why nobody could
> leave them messages. I see in /var/log/asterisk/messages that it
> does write out a warning message of:

> ast_log(LOG_WARNING, "Unable to leave message since we will exceed
> the maximum number of messages allowed (%u >= %u)\n", msgnum,
> vmu->maxmsg);

> but I was wondering how feasible it would be to modify the code to
> add:

> 1) the mailbox name of the user whom has hit the limit
> 2) a warning/critical threshold that the user is getting close to the
> limit

> using whatever monitoring tool one has available eg. OSSEC the alert
> could be trapped and the user notified.

> Is this worthy of an RFE? and possible help from people if I try and
> create the patch myself ?

> Thoughts or insults ?

> --
> Thanks, Phil

> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Re: [asterisk-users] allowguest = yes? no?

2012-01-24 Thread Jim DeVito

What they are talking about is SIP URI dialling. Let say you have 
extension 1000 the rings a phone on your system. With allowguest=yes I 
would be allowed to dial SIP:/1...@yourdomain.com and assuming the 
context defined in your [General] section had access to exten 1000 I 
would connect to that phone. With alloweguest=no my call would be rejected.


That does not mean that strangers can not call an IVR and get to your 
1000 extension or even a DID that point right to it.


If you are going to allowguest=yes you need to take carfule note of your 
contexts so as not to allow strangers access to parts of your dial plan 
that have, lets say long distance routes.


Does that help?

Thanks!!

Jim

On 01/24/2012 09:34 AM, Gilles wrote:

Hello

I don't understand how I should use the "allowguest" item: If set to
"yes", callers from the Net should authenticate, but then, how can I
allow strangers to call extensions in my system?

"allowguest

If set to no, this disallows guest SIP connections. The default is to
allow guest connections. SIP normally requires authentication, but you
can accept calls from users who do not support authentication (i.e.,
do not have a secret field defined).Certain SIP appliances (such as
the Cisco Call Manager v4.1) do not support authentication, so they
will not be able to connect if you set allowguest=no:
allowguest=no|yes"

(from "Asterisk – The future of Telephony")

Thank you.


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Re: [asterisk-users] ChanSpy : how to know channel name ?

I would try chanspy(sip/miq8,b) - the b flag denotes to only listen to a
bridged call which (it seems to me) should pick up both sides.

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Tuesday, January 24, 2012 8:46 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] ChanSpy : how to know channel name ?

Hello,

OK thanks. But, I want to listen to the conversation (not just 1 channel out
of 2 channels). How then do I use ChanSpy ?

On 01/24/2012 03:41 PM, Danny Nicholas wrote: 

Strip off the -x.  Just listen to SIP/miq8 and SIP/375382280 in your
example.

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Tuesday, January 24, 2012 7:47 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] ChanSpy : how to know channel name ?

Hello list,

to use ChanSpy, one needs to know the name of the channel.

But on an incoming call from the provider, or an outgoing call to the
provider there are always numbers added. How can one then know the channel
name ??

core show channels verbose shows me for example :

SIP/378680644-2 default  
SIP/rs4-2445 sub-uitinternation
SIP/3715320168-2 default
SIP/ibenla2-244 sub-uit789  
SIP/372083610-2 default  
SIP/cedhou0-24 sub-uit789  
SIP/travel3-2 pbx-routing 
SIP/INTELin-2 pbx-routing 
SIP/375382280-2 default   
SIP/miq8-2419  sub-uitGSM  
SIP/3749378004- default  
SIP/instlpr0-2 sub-uitinternation
SIP/372089170-2 default  
SIP/v9q9uLT- from-GFATRUNK 
46 active channels
24 active calls

If I want to listen to the conversation of SIP/miq8-2419 and
SIP/375382280-2 (these 2 channels have been connected to 1
conversation), how do I use ChanSpy ??

Kind regards;
Jonas.

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Re: [asterisk-users] ChanSpy : how to know channel name ?


Hello,

OK thanks. But, I want to listen to the conversation (not just 1 channel 
out of 2 channels). How then do I use ChanSpy ?




On 01/24/2012 03:41 PM, Danny Nicholas wrote:


Strip off the --x.  Just listen to SIP/miq8 and SIP/375382280 in 
your example.


*From:*asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas 
Kellens

*Sent:* Tuesday, January 24, 2012 7:47 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] ChanSpy : how to know channel name ?

Hello list,

to use ChanSpy, one needs to know the name of the channel.

But on an incoming call from the provider, or an outgoing call to the 
provider there are always numbers added. How can one then know the 
channel name ??


/core show channels verbose/ shows me for example :

/SIP/*378680644-2* default
SIP/*rs4-2445* sub-uitinternation
SIP/*3715320168-2* default
SIP*/ibenla2-244* sub-uit789
SIP/*372083610-2* default
SIP/*cedhou0-24* sub-uit789
SIP*/travel3-2* pbx-routing
SIP/*INTELin-2* pbx-routing
SIP/*375382280-2* default
SIP/*miq8-2419*  sub-uitGSM
SIP/*3749378004-* default
SIP*/instlpr0-2* sub-uitinternation
SIP/*372089170-2* default
SIP/*v9q9uLT-* from-GFATRUNK
46 active channels
24 active calls/


If I want to listen to the conversation of /SIP/*miq8-2419*/ and 
/SIP/*375382280-2*/ (these 2 channels have been connected to 1 
conversation), how do I use ChanSpy ??




Kind regards;
Jonas.


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Re: [asterisk-users] ChanSpy : how to know channel name ?

Strip off the -x.  Just listen to SIP/miq8 and SIP/375382280 in your
example.

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Tuesday, January 24, 2012 7:47 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] ChanSpy : how to know channel name ?

Hello list,

to use ChanSpy, one needs to know the name of the channel.

But on an incoming call from the provider, or an outgoing call to the
provider there are always numbers added. How can one then know the channel
name ??

core show channels verbose shows me for example :

SIP/378680644-2 default  
SIP/rs4-2445 sub-uitinternation
SIP/3715320168-2 default
SIP/ibenla2-244 sub-uit789  
SIP/372083610-2 default  
SIP/cedhou0-24 sub-uit789  
SIP/travel3-2 pbx-routing 
SIP/INTELin-2 pbx-routing 
SIP/375382280-2 default   
SIP/miq8-2419  sub-uitGSM  
SIP/3749378004- default  
SIP/instlpr0-2 sub-uitinternation
SIP/372089170-2 default  
SIP/v9q9uLT- from-GFATRUNK 
46 active channels
24 active calls

If I want to listen to the conversation of SIP/miq8-2419 and
SIP/375382280-2 (these 2 channels have been connected to 1
conversation), how do I use ChanSpy ??

Kind regards;
Jonas.

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Re: [asterisk-users] RFE idea for VM application

You don’t state which version this is for, but it seems like a simple patch 
would be for voicemail to play sorry-mailbox-full.wav (standard sound).  In 
lieu of all that, you could do a quick-and-dirty AGI to read /v/l/a/m and play 
the message back since voicemail is one of the larger modules and therefore 
prone to a better chance of error introduction on an RFE.  PSOT – the Unix Bot 
thing was pretty cool.

 

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Phil Daws
Sent: Tuesday, January 24, 2012 8:30 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] RFE idea for VM application

 

Hello all,

we are using IMAP for the storage of VMs and had a user yesterday his their 
maxmsg limit (default 100) and was wondering why nobody could leave them 
messages.  I see in /var/log/asterisk/messages that it does write out a warning 
message of:

ast_log(LOG_WARNING, "Unable to leave message since we will exceed the maximum 
number of messages allowed (%u >= %u)\n", msgnum, vmu->maxmsg);

but I was wondering how feasible it would be to modify the code to add:

1) the mailbox name of the user whom has hit the limit
2) a warning/critical threshold that the user is getting close to the limit

using whatever monitoring tool one has available eg. OSSEC the alert could be 
trapped and the user notified.

Is this worthy of an RFE? and possible help from people if I try and create the 
patch myself ?

Thoughts or insults ?

-- 
Thanks, Phil

 

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Re: [asterisk-users] RFE idea for VM application

You don’t state which version this is for, but it seems like a simple patch 
would be for voicemail to play sorry-mailbox-full.wav (standard sound).  In 
lieu of all that, you could do a quick-and-dirty AGI to read /v/l/a/m and play 
the message back since voicemail is one of the larger modules and therefore 
prone to a better chance of error introduction on an RFE.  PSOT – the Unix Bot 
thing was pretty cool.

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Phil Daws
Sent: Tuesday, January 24, 2012 8:30 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] RFE idea for VM application

 

Hello all,

we are using IMAP for the storage of VMs and had a user yesterday his their 
maxmsg limit (default 100) and was wondering why nobody could leave them 
messages.  I see in /var/log/asterisk/messages that it does write out a warning 
message of:

ast_log(LOG_WARNING, "Unable to leave message since we will exceed the maximum 
number of messages allowed (%u >= %u)\n", msgnum, vmu->maxmsg);

but I was wondering how feasible it would be to modify the code to add:

1) the mailbox name of the user whom has hit the limit
2) a warning/critical threshold that the user is getting close to the limit

using whatever monitoring tool one has available eg. OSSEC the alert could be 
trapped and the user notified.

Is this worthy of an RFE? and possible help from people if I try and create the 
patch myself ?

Thoughts or insults ?

-- 
Thanks, Phil

 

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[asterisk-users] allowguest = yes? no?

2012-01-24 Thread Gilles

Hello

I don't understand how I should use the "allowguest" item: If set to
"yes", callers from the Net should authenticate, but then, how can I
allow strangers to call extensions in my system?

"allowguest

If set to no, this disallows guest SIP connections. The default is to
allow guest connections. SIP normally requires authentication, but you
can accept calls from users who do not support authentication (i.e.,
do not have a secret field defined).Certain SIP appliances (such as
the Cisco Call Manager v4.1) do not support authentication, so they
will not be able to connect if you set allowguest=no:
allowguest=no|yes"

(from "Asterisk  The future of Telephony")

Thank you.


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[asterisk-users] RFE idea for VM application

Hello all, 

we are using IMAP for the storage of VMs and had a user yesterday his their 
maxmsg limit (default 100) and was wondering why nobody could leave them 
messages. I see in /var/log/asterisk/messages that it does write out a warning 
message of: 

ast_log(LOG_WARNING, "Unable to leave message since we will exceed the maximum 
number of messages allowed (%u >= %u)\n", msgnum, vmu->maxmsg); 

but I was wondering how feasible it would be to modify the code to add: 

1) the mailbox name of the user whom has hit the limit 
2) a warning/critical threshold that the user is getting close to the limit 

using whatever monitoring tool one has available eg. OSSEC the alert could be 
trapped and the user notified. 

Is this worthy of an RFE? and possible help from people if I try and create the 
patch myself ? 

Thoughts or insults ? 

-- 
Thanks, Phil 

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Re: [asterisk-users] Pickup calls coming from queues

2012-01-24 Thread Olivier

Great !
I'll test it ASAP and report back here (tomorrow, if possible).

2012/1/23, Alec Davis :
>
>> How can I test this solution on a 1.8.8.1 system ?
>> If I'm not mistaken, diff
>> https://reviewboard.asterisk.org/r/1619 do not apply to 1.8.8.1.
>
> I've just checked out 1.8.8.1 and download my patch from
> https://reviewboard.asterisk.org/r/1619/diff/raw/ and it applied clean,
> using the following on a debian lenny box:
>
> svn co http://svn.digium.com/svn/asterisk/tags/1.8.8.1 asterisk-1.8.8.1
> cd asterisk-1.8.8.1
> wget --no-check-certificate
> https://reviewboard.asterisk.org/r/1619/diff/raw/
> mv index.html r1619.diff.txt
> patch -p0 < r1619.diff.txt
>
> Alec
>
>
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[asterisk-users] ChanSpy : how to know channel name ?


Hello list,

to use ChanSpy, one needs to know the name of the channel.

But on an incoming call from the provider, or an outgoing call to the 
provider there are always numbers added. How can one then know the 
channel name ??


/core show channels verbose/ shows me for example :

/SIP/*378680644-2* default
SIP/*rs4-2445* sub-uitinternation
SIP/*3715320168-2* default
SIP*/ibenla2-244* sub-uit789
SIP/*372083610-2* default
SIP/*cedhou0-24* sub-uit789
SIP*/travel3-2* pbx-routing
SIP/*INTELin-2* pbx-routing
SIP/*375382280-2* default
SIP/*miq8-2419*  sub-uitGSM
SIP/*3749378004-* default
SIP*/instlpr0-2* sub-uitinternation
SIP/*372089170-2* default
SIP/*v9q9uLT-* from-GFATRUNK
46 active channels
24 active calls/


If I want to listen to the conversation of /SIP/*miq8-2419*/and 
/SIP/*375382280-2*/(these 2 channels have been connected to 1 
conversation), how do I use ChanSpy ??




Kind regards;
Jonas.
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Re: [asterisk-users] SDP Issue

2012-01-24 Thread Alex Balashov

Phil, I applaud both the diplomacy of your responses and your 
willingness to consider the critique.  It was very gentlemanly of you.


For the interlopers cashing in cheap shots, my enthusiasm is more 
restrained.


--
Alex Balashov - Principal
Evariste Systems LLC
260 Peachtree Street NW
Suite 2200
Atlanta, GA 30303
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/

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Re: [asterisk-users] SDP Issue

Doubt really care TBH and the whole reason behind it was a nickname, Unix Bod, 
given to me by a very knowledgeable friend.  I am here to learn from the 
community and give back where I can.
-- 
Thanks, Phil

- Original Message -
> On 01/24/2012 07:34 AM, John Novack wrote:
> 
> > Phil has been using his pseudonym for years, and Alex and his
> > painful/painstaking posting is the only one I have seen even
> > raising
> > the issue.
> >
> > Says even more about Alex than Phil
> 
> Guilty as charged.  :-)
> 
> --
> Alex Balashov - Principal
> Evariste Systems LLC
> 260 Peachtree Street NW
> Suite 2200
> Atlanta, GA 30303
> Tel: +1-678-954-0670
> Fax: +1-404-961-1892
> Web: http://www.evaristesys.com/
> 
> --
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Re: [asterisk-users] SDP Issue

2012-01-24 Thread Alex Balashov


On 01/24/2012 07:34 AM, John Novack wrote:


Phil has been using his pseudonym for years, and Alex and his
painful/painstaking posting is the only one I have seen even raising
the issue.

Says even more about Alex than Phil


Guilty as charged.  :-)

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Evariste Systems LLC
260 Peachtree Street NW
Suite 2200
Atlanta, GA 30303
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/

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Re: [asterisk-users] SDP Issue

2012-01-24 Thread John Novack

Phil has been using his pseudonym for years, and Alex and his
painful/painstaking posting is the only one I have seen even raising the
issue.

Says even more about Alex than Phil

Peg Leg O'Brien

Alex Balashov wrote:

I wasn't so much poking fun at the substance of your post as the fact that
you're the only person on this mailing list that posts with a pseudonym, and at
that, one evocative of online gaming or forum environments. It just doesn't
fit with the culture or the relatively serious, substantive and adult-oriented
tenor of this type of list. Do you not notice that?

At the risk of being rude, "--[ UxBoD ]--" is something that belongs in WoW or
a phpBB board full of spotty adolescents.

If your real name is Phil, why not post as such? Okay, so maybe you don't want to give out your
surname for one reason or another--fair enough. So, post as "Phil", or "Phil
D.", if your full name were Phil Deleterious.

There's no rule saying you have to. However, the survival of most human social
institutions, including those devoted to the exchange of knowledge, is upheld
in part by adherence to some conventions of self-presentation and deportment.
These conventions help delineate the identity and character of the venue to
outsiders, and assist in self-knowledge and affirmation of that character
internally.

Everyone else here posts with their full name because it communicates: "I am a real,
adult person solving real-world technical problems related to Asterisk." It is, at
least in part, an affirmation of the fact that real personalities--real humans, real
identities--underlie participation in Internet forums, especially specialised ones. It
is also a nod to the benificent academic origins of the Internet. There are reasons for
these conventions. They encapsulate our creation mythos, and they tell us what kind of
people we are, as a community.

Quite frankly, your From: display name spits on the pedigree, on the storied
heritage of how this open-source community came to be. It is not deferential
to the accrued wisdom of Internet-focused technical specialists in areas such
as Asterisk or IP telephony, and it does not hallow the ground on which we
tread. It says that the ROFLcopter has landed!!!111 and lol p0wned teh n00bs.

Except, you're being the n00b. Come on, Phil. Self-awareness is important. I know I am
being a self-important ass pontificating on this to you. Are you okay with an ASCII art
pseudonym that says, "I'm a 14 year old playing WoW on a delapidated, slightly
yellowed Windows tower draped in dirty underwear"? If not for you, why not for us?
Please post with a real name.

--
This message was painstakingly thumbed out on my mobile, so apologies for
brevity, errors, and general sloppiness.

Alex Balashov - Principal
Evariste Systems LLC
260 Peachtree Street NW
Suite 2200
Atlanta, GA 30303
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/

On Jan 24, 2012, at 6:02 AM, "--[ UxBoD ]--" wrote:

LOL :) that really made me chuckle this morning; and very apt for the fact I
did not post any fundamental details about the issue. All points duly noted!
--
Thanks, Phil

- Original Message -

Is --[ UxBoD ]-- a first-person shooter gang--er, clan--name? Like,
one of those who rocket-jumps onto the platform and camps with the
grenade launcher, trying to stop the reds from capturing the blue
flag? I hate how the health and the ammo takes so long to respawn.
Is there any way to fix that in deathmatch?

--
This message was painstakingly thumbed out on my mobile, so apologies
for brevity, errors, and general sloppiness.

Alex Balashov - Principal
Evariste Systems LLC
260 Peachtree Street NW
Suite 2200
Atlanta, GA 30303
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/

On Jan 24, 2012, at 2:10 AM, "--[ UxBoD ]--"< ux...@splatnix.net>
wrote:

--[ UxBoD ]--

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Re: [asterisk-users] SDP Issue