Thanks.
On 4/13/2010 3:07 AM, Ioan Indreias wrote:
> On Mon, Apr 12, 2010 at 8:19 PM, Mark Hulber
> wrote:
>
>> I want to use a voip speaker phone as a room monitor. Requirements:
>>
>> A phone that I can set to auto answer in speaker mode.
>> A phone with
I want to use a voip speaker phone as a room monitor. Requirements:
A phone that I can set to auto answer in speaker mode.
A phone with a good speaker phone.
Ability to make the audio one way. I want to monitor the room but not
have my voice heard in the room. Yes, the mute button can accompli
I have the same problem. I have asterisk on the public internet and
other ips on the private lan. When the internet goes down my private
asterisk network is compromised. My thought is that it has something to
do with the ports/ips on which asterisk is trying to communicate. It
may be a conf
This is how I understand it. The other end is trying to set up comfort
noise and asterisk is letting you know that it's trying to do so and
maybe you can turn this off on the other end. I have a particular voip
provider where I get this message. I think if you get it turned off
there's a lit
ten over.
On 1/26/2010 11:15 AM, Tilghman Lesher wrote:
> On Tuesday 26 January 2010 10:08:39 Mark Hulber wrote:
>
>> On 1/25/2010 7:06 PM, Tilghman Lesher wrote:
>>
>>> On Monday 25 January 2010 08:52:45 Mark Hulber wrote:
>>>
>>>>
When I run "make install" I don't see this file getting overwritten. Do
I have to delete it to get this to happen?
On 1/25/2010 7:06 PM, Tilghman Lesher wrote:
> On Monday 25 January 2010 08:52:45 Mark Hulber wrote:
>
>> Recently safe_asterisk is failing to pick
Recently safe_asterisk is failing to pick up ASTSBINDIR. I've never had
this problem before and even when I move to back versions I have the
issue. I did upgrade safe_asterisk and the init.d scripts a version or
so ago but even when I try older ones I still have the problem. When I
hard code
Looks like the Makefile is broken and putting SLN16 instead of sln16.
Mark Hulber wrote:
> It looks like there's a problem with the location or naming of the Extra
> SLN16 sounds:
>
> --14:11:43--
>
> http://downloads.digium.com/pub/telephony/sounds/releases/a
It looks like there's a problem with the location or naming of the Extra
SLN16 sounds:
--14:11:43--
http://downloads.digium.com/pub/telephony/sounds/releases/asterisk-extra-sounds-fr-SLN16-1.4.9.tar.gz
Resolving downloads.digium.com... 76.164.171.232
Connecting to downloads.digi
It might help to show your Support context in outbound.conf.
MARK.
Alexander Topolanek wrote:
> Hi,
>
> recently I changend a few things in the configuration of the Asterisk
> 1.2.17-BRIstuffed-0.3.0-PRE-1y-d of a customer. One demand was that
> different groups of SIP-Phones are using different
I've had a good ongoing experience using http://www.unlimitel.ca. They
are responsive and reliable.
MARK.
Asterisk guy wrote:
> hi
>
> Can anyone recommend a good DID provider offering numbers in Toronto ?
>
> ( 1 very stable 2 support porting numbers from Bell, primus, telus.. )
>
>
Have you tried setting the CALLERID variables? If the provider is
ignoring those then I guess they are asking you to set per call
blocking? I don't know how to do that.
exten => s,1,Set(CALLERID(number)=3025551212|a)
exten => s,n,Set(CALLERID(name)=Joe Smith|a)
MARK.
Emil Thelin wrote:
Hi!
Before I open a bug I'll ask again if anyone else is having trouble with
receiving MWI on SIP devices in 1.4. My configuration was working fine
in 1.2 but as soon as I change to any build of 1.4 I don't get
notification on any of several SIP devices. I can post my configuration
but since it w
I set up mine with the web interface but I notice that some settings can
only be made by config files. Do you know how to extract the current
config file from the phone?
Here's how I set up the web interface:
Authentication Name: aastra480_1
Password: password
BLA Number:
Line Mode: Generic
Anyone else having trouble with MWI on 1.4 Beta? The messages are
getting stored and I'm getting the emails but no stutter tone or MWI as
far as I can tell.
MARK.
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Yes, it worked. I didn't get the announcement of 1.2.9.1.
MARK.
Tzafrir Cohen wrote:
On Wed, Sep 13, 2006 at 12:00:27PM -0400, Mark Hulber wrote:
Any pointers about on how to get around this build problem in Zaptel 1.2.9?
Get 1.2.9.1, that has fixed exactly that.
(and im
Any pointers about on how to get around this build problem in Zaptel 1.2.9?
/usr/src/zaptel-1.2.9/wct4xxp/fw2h
/usr/src/zaptel-1.2.9/wct4xxp/OCT6114-128D.ima
/usr/src/zaptel-1.2.9/wct4xxp/vpm450m_fw.h
make[3]: *** No rule to make target
`/usr/src/zaptel-1.2.9/wct4xxp/../oct612x/inc
I was using zap but I ditched the PSTN for now. Try taking a look at:
CALLERID(name) or CALLERID(number) instead.
MARK.
Greg Delgado wrote:
Has anyone got a working analog connection to POTS
wherein DTMF, *not* FSK is used to send caller id by
the telco switch towards asterisk?
I've tried A
Hey Dean,
Maybe it would be easier if you would describe what you would like to
happen as a result of doing what you are asking. When you have an
incoming call from this provider do you know what number was dialed? Are
you expecting this number to be displayed somewhere or are you looking
to
I started to have a problem today that all my calls through voipjet
result in just timing out after my assigned timeout period. I tried
multiple of their servers with the same problem. Anyone else having a
problem? I am running:
Asterisk SVN-branch-1.2-r24381M built by root @ asterisk.hulbe
Have you tried dialing an 800 number? Does that work? This extension:
exten => _9XX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})
seems to be missing one X since it's only 10 digits long. Your PSTN
probably requires a 1 to be dialed also. On the other hand,
exten => _91[1234567]XXNXXX
> show function STRFTIME
-= Info about function 'STRFTIME' =-
[Syntax]
STRFTIME([][,[timezone][,format]])
[Synopsis]
Returns the current date/time in a specified format.
[Description]
Not available
Example:
exten => s,n,NoOp(TIME=${STRFTIME(,EST5EDT,%d%b%Y-%H:%M:%S)})
The only time I see recorded in your log is that of the recording check
-- Executing AGI("Zap/1-1", "recordingcheck|20060227-131600|1141046151.2")
in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20060227-131600|1141046151.2: Inbound recording no
ge-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Mark Hulber
Sent: Sunday, 5 February 2006 10:21 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] DTMF Sporadicaly Being Generated
I've been having horrible DTM
I've been having horrible DTMF problems lately on from Sipura ATAs to
ZAP and IAX. It's primarily with repeated digits. I'm starting to move
my connections to SIP until I can get it all figured out. Other than
updating to the newest SVN trunk I haven't made changes on my end that
should have
It sounds like you both need a Zap card. You can ring the analog phone
and/or the Sip phones when a call comes in on the POTS line that is
connected to the card.
MARK.
Brian J. Murrell wrote:
On Fri, 2006-02-03 at 07:37 -0700, Bromont Quebec wrote:
Well in my setup I have a few IP phone
My experience is that when an iax or sip channel is unavailable for some
reason it fails right away despite whatever timeout I have set for the
call. In these cases the caller doesn't even realize that the call has
failed over to the next carrier.
exten => s,n(dial1),Dial(${VOIPJET}/${ARG1}
e
but the callerid of the original incoming caller.
MARK.
Kevin P. Fleming wrote:
Mark Hulber wrote:
exten => s,n,Set(CALLERID(name)=${CALLERIDNAME})
This could never have accomplished anything, since those two
references affect the exact same variable internally.
because I want the outg
Previously, when I wanted to forward to incoming callerid when I
forwarded a call to another number I had to set the callerid on the
outgoing call to be that of the incoming number. So today I do this:
exten => s,n,Set(CALLERID(name)=${CALLERIDNAME})
because I want the outgoing callerid that
The paper is definitely interesting and I commend them for their effort
but it doesn't represent a complete understanding of the Skype protocol
to the extent that an Asterisk server could speak the Skype protocol.
They say that much of the Skype protocol is encrypted and needs to be
inferred t
I'm all for criticism where it's due but I'm sure for all the bashing of
Voipjet going on in this thread I'm sure there are just as many
"non-users" who are generally happy with the service they provide and
the price at which they provide it.
I for one am "also" a customer of Verizon, a fact I
specifies Bellcore-dr3. Sending the number would be so
much more reliable...
MARK.
Mark Hulber wrote:
Ok, your solution does work but in looking at my console output I saw
that SIPGetHeader was deprecated for the new dialplan function
SIP_HEADER. Below is my modification. You don't n
Ok, your solution does work but in looking at my console output I saw that
SIPGetHeader was deprecated for the new dialplan function SIP_HEADER. Below is
my modification. You don't need a priority+101.
exten => 212999,1,Set(Var_Alert=${SIP_HEADER(Alert-Info)})
exten => 212999,n,GotoIf
I have a single Broadvoice account with more than one number. I am
trying to distinguish between the numbers on an inbound call. I have
already tried using different incoming extensions that match each number
but it always defaults to the primary. Someone earlier mentioned
SIPGetHeader as a
You can pass a context to DISA for dialing outbound. Do you have a
dialplan that works like this for non-DISA calls? You can use the same
one. Otherwise, I do this with nested dialplans by putting the most
specific and longest rules first. By nesting, you only enter an
included context if n
I'm not sure if a failed qualify will affect your active call but you
might want to try to use the qualifysmoothing variable in iax.conf.
This won't "disqualify" a peer for a single bad sample.
;qualify=yes; Make sure this peer is alive
;qualifysmoothing = yes
It probably makes no difference to your problem but it's "canreinvite"
not "canreinvete". You'll want to include dialout extensions in
[siptest]. For instance, maybe include your default context.
MARK.
Wagner Nunes wrote:
Hi all!!!
I have an asterisk compiled and started in one computer he
As I understand it, you can initiate the call by having one of the
dynamic endpoints call the other through the fixed ip host and then the
fixed host can allow the two endpoints to create a native bridge.
Otherwise, I think you'll have to somehow cache the registration at the
dynamics hosts and try
This is an old issue on which you can seach and find info. Some info
indicates that you need a timing source such as a zaptel card or
ztdummy. Other suggests that if you are using native music on hold and
mpg123 is in the path you might run into this error.
MARK.
Daniel Corbe wrote:
Has a
I think for SIP the control channel can still go through the proxy while
the data is bridged natively allowing you to still account for the
call. I'm not sure of the details on how Asterisk does it.
MARK.
David Bandel wrote:
On 11/2/05, Mark Hulber <[EMAIL PROTECTED]> wrote:
I think this means that it attempted to create a native bridge, which is
that it was trying to have the call go directly between the two
endpoints instead of going through the asterisk server but that process
failed. So in that case, Asterisk continued to proxy the call data. If
that's the ca
data, no one knows what those other scheduled tasks might be,
so there's no way to answer the question of 'what's going on'.
Just ignore the item for now.
Rich
----
From: Mark Hulber <[EMAIL PROTECTED]>
Subject: Re: [Asterisk-Users] Warning -- c
I can get it to repeat. It happens in a condition that has had no problem
normally in the past. I have an incoming call, SIP or ZAP it doesn't really
matter and then I'm dialing out on IAX to try and connect the call to one or
more numbers on an IAX channel. For each IAX channel I create I se
Not busy at all. Like these were the only calls in progress at the time.
MARK.
Rich Adamson wrote:
I've started to see this lately on outgoing IAX calls using CVS Head:
-- IAX2/voipjet-out-5 is making progress passing it to
SIP/64.26.157.252-094e4058
-- IAX2/voipjet-out-6 is making p
I've started to see this lately on outgoing IAX calls using CVS Head:
-- IAX2/voipjet-out-5 is making progress passing it to
SIP/64.26.157.252-094e4058
-- IAX2/voipjet-out-6 is making progress passing it to
SIP/64.26.157.252-094e4058
Nov 2 16:56:15 WARNING[22566]: chan_iax2.c:7948 networ
I am in the US, NYC using a TDM400 card. I never have never seen this
issue until now. I see some code has been changed in this area recently.
MARK.
Rich Adamson wrote:
Previously I would get two events on my Zap channel which indicated
ringing and answered. Now I am getting polarity revers
Previously I would get two events on my Zap channel which indicated
ringing and answered. Now I am getting polarity reversal events:
Nov 2 07:01:25 NOTICE[18246]: chan_zap.c:6014 ss_thread: Got event 17
(Polarity Reversal)...
Nov 2 07:01:28 NOTICE[18246]: chan_zap.c:6014 ss_thread: Got event
I don't use quotes on either if that makes a difference. When are you
setting it? Maybe you are losing the incoming number by the time you
set it the number has been changed to the extension. I use Sipuras and
have no problem with this. Here's an example of a macro I use when
forwarding an
Yes, or this for example:
[macro-rhangup]
exten => s,1,NoOp(DIALSTATUS=${DIALSTATUS})
exten => s,n,NoOp(TIME=${DATETIME})
exten => s,n,Hangup
I also output the date and time prior to dialing out.
MARK.
Sherwood McGowan wrote:
You could always just add some
exten = NUM,PRIO,VERB
I noted this on Friday. I don't think I had problems using the devices
but "sip show peers" took some time to show the registrations.
MARK.
Kevin P. Fleming wrote:
Rich Adamson wrote:
Once the phones register again, they can be called, but not until then.
Not sure what's going on yet... an
In recent CVS Head build when I run: "sip show peers" my dynamic peers show:
Name/username HostDyn Nat ACL Port Status
sipura2_2/sipura2_2(Unspecified)D N 0UNKNOWN
sipura2_1/sipura2_1(Unspecified)D N 0UNKNOWN
It might be helpful to show what's using the CPU (the rest of Top).
MARK.
Julian Lyndon-Smith wrote:
We had to move from a old * server to a new one in a hurry (hardware
failure). The old server was a dual pentium 700 with 512MB ram running
fedora core 2, the new one is a single 3GHz Pentium w
/usr/src/asterisk/apps/app_meetme.so
/usr/lib/asterisk/modules/app_meetme.so
Fabio Montemaggiore wrote:
I haven't app_meetme.so file...
Where I can search?
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I had some complaints today that one of my incoming SIP numbers was
failing for several hours. I looked at my console and didn't see
anything unusual but SIP show registry confirmed that my registrations
were in a failed state. I did a SIP reload and saw this in the output:
Sep 22 18:54:48 N
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