:D pretty much true !
On Tue, Jan 24, 2012 at 12:23 PM, Alex Balashov
abalas...@evaristesys.comwrote:
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
Hi Shaun,
So you mean to say that if I will download the SIP firmware the phone will
directly work with Asterisk with no Avaya Call Manager (ACM) in between ?
My main requirement is to directly make an Avaya phone talk with the asterisk
server with no ACM in between.
Regards,
Aamir
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
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
Alex, I would hate for you to have to pen such a long email again using your
mobile and do appreciate the comments hence the change to appear more human.
--
Thanks, Phil
- Original Message -
I wasn't so much poking fun at the substance of your post as the fact
that you're the only
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
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
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
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
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 :
Great !
I'll test it ASAP and report back here (tomorrow, if possible).
2012/1/23, Alec Davis siva...@paradise.net.nz:
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
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
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
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
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
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
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.
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
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
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
On Tue, 24 Jan 2012 09:55:12 -0500, Jim DeVito
asterisk-users-mailing-l...@devito.cc 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
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
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
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,
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
On 01/24/2012 09:03 AM, Gilles wrote:
On Tue, 24 Jan 2012 09:55:12 -0500, Jim DeVito
asterisk-users-mailing-l...@devito.cc 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
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
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
On Tue, 24 Jan 2012 09:26:26 -0600, Kevin P. Fleming
kpflem...@digium.com 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
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
I found the fix; in my sip.cfg I changed the following line from
DTMF tone.dtmf.level=-15 tone.dtmf.onTime=50 tone.dtmf.offTime=50
tone.dtmf.chassis.masking=0
To
DTMF tone.dtmf.level=-9 tone.dtmf.onTime=50 tone.dtmf.offTime=50
tone.dtmf.chassis.masking=0
And it fixed my issue.
Thanks,
Motty
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
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 =
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
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
On 1/24/2012 5:32 PM, Kevin P. Fleming wrote:
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.
good stuff. thanks for the
36 matches
Mail list logo
