Displaying 20 results from an estimated 2000 matches similar to: "[Fwd: voicemail locked up Asterisk 1.4.13]"
2008 Mar 16
0
Telemarketer Torture.... (was: Re: asterisk-users Digest, Vol 44, Issue 49)
You could accept as the "passcode" the caller punching in their own
phone#, then checking that against your whitelist. Lets associates get
past the challenge when using someone else's phone, without their
remembering some arbitrary passcode.
And strangers or barred old associates who abuse it can get an earful
about how you're suing them for wire fraud. Preferably after you
2009 Nov 06
2
Routing incoming call based on caller id
I am not that good at regex and it's use in Asterisk. I am running
Asterisk 1.4.13
Currently I have this in my extensions.conf for incoming calls on our
house phone line:
[housemenu]
exten => s,1,GotoIF($["${CALLERID(num)}" = "815xxxxxxx"]?s|12);
815xxxxxxx is our home phone number, when caller id fails or is missing
that is what is recorded.
I want to expand this
2007 Nov 27
1
Voice mail & Uniden UIP-200 phones
I had a working system using * 1.0 and then 1.2 and now Asterisk 1.4.13
with addons 1.4.4, zaptel 1.4.6, libpri 1.4.2. I have a mix of
Grandstream (GXP2000), Uniden uip-200, Linksys Wireless G, and analog
phones via Adtran chan bank. When I went to * 1.4.13, the Uniden phones
stopped being able to login to voicemail. All phones are on same lan
with Asterisk.
I get 'Login incorrect'
2011 Jul 23
9
Securing Asterisk
I beg to differ. Digium is hiding from the real world and somebody is
going take the software and run with it. My customers lost in excess
of $50.000 and cut my pay in half, because of hackers. The hackers
figured out how to scan every asterisk for weak passwords or open
ports, and bang them real good. We need two things: a) disable in
sip.conf the reply for INVITES that have wrong user
2004 Aug 08
2
pbx answers after answering from analog phone
I am setting up my * for at home office and still have analog phones
attached and answer from those analog phones and not necessarily through the
pbx. I found that with the X100P cards, they see the 2nd ring and will be
ready to answer the line. I used a Wait to pause and allow another 2 rings
before * answers. But found that if we answer the line after the 2nd ring
and before the 4th, * still
2004 Aug 06
1
oem x100p undefined symbol ast_get_txt
I am putting together my first *. I had it running with two other pc's
running xlite and setup voicemail and a couple of menus and submenus and had
that running well. I had order a couple of oem x100p cards from
digitnetworks.
I installed them as they said with their voicepet2.2.zip drivers and did the
modprobe on zaptel and wcfxo and then ran ztcfg -vv and got this:
Zaptel Configuration
2004 Dec 24
1
Uniden UIP200 firmware v4.63
I just spent the last hour or so trying to get this firmware to work across
a NAT with no success. I have a GS BT101 working through the same NAT, so I
don't think it's the NAT itself.
I have a STUN setup in * and pointed the UIP200 to it and I tryed several
combinations of nat= in the sip.conf and in the config files for this phone.
No luck(yes, I did a reload now with each change in
2008 Sep 30
1
OT: real 2 line phone vs. 1 line and call waiting
I'm looking into getting a new phone and wondering what the difference
in functionality is between a single line phone with call waiting and a
real 2 line phone (either a real SIP phone or an analog 2 line phone and
a 2 port ATA) is. Why would I want the real 2 lines vs. just being able
to take an incoming call via call-waiting?
Cheers,
b.
-------------- next part --------------
A non-text
2012 Jun 18
1
TDM410 PTSN line setup with 1 analog phone
Hello, I have a current asterisk 1.8.13.0 asterisk-addons 1.6.24 asterisk-sounds
1.2.1 dahdi-linux-complete 2.6.1+2.6.1 libpri 1.4.12 and asterisk-gui 2.1.0.rc1
(not trying to use the gui, want to do everything by hand) with a TDM410 with
2FXO and 2FXS. I have my POTS (PTNS) line plugged into port 1 (FXO) and a
analog phone connected to port 3 (FXS). I compiled asterisk with asterisk
2011 Apr 07
2
Asterisk Avaya SIP Trunking One Way Audio
I am facing one way audio problem in sip trunking between asterisk and
avaya.
+-------------+ +----+
| avaya sip |-------| P1 |
+-------------+ +----+
|
|
|
+-------------+
| Asterisk | WAN
2007 Oct 27
2
Uniden UIP200 phones
I am trying to get distinctive ringing going again with these phones,
depending on the outside line the call comes in on.
I had a working 1.0.x Asterisk setup using:
SetVar(ALERT_INFO=<http://127.0.0.1/Bellcore-dr2>)
Which used the short quick rings.
In Asterisk 1.4, I have tried several things, but I think the correct
syntax is:
Set(_ALERT_INFO=<http://127.0.0.1/Bellcore-dr2>)
But
2004 Sep 21
3
Uniden uip200
I got a Uniden UIP200 and started to configure it and I am lost....
I have a tftp server setup on my * server and have the files unidencom.txt
and uniden<mac>.txt there. But it doesn't quite work yet. It registers as
a sip phone (sip show peers), but I cann't dial it and the display shows #1
disconnected all the time. It has firmware version BS4.59a in it.
I have no idea if I
2004 Jul 16
7
7960 Dynamic DNS?
Hello everyone....
Searching the archives and google always comes up with entries regarding
the "dyn" dns option in the 7960, but I can't find answers to my
specific question....
My 7960 is connected via cable modem and is NAT'ed (everything is
working fine). On the 7960 under SIP configuration\NAT Address I have
the public IP of my cable connection. Comcast gives me a
2004 Jul 25
1
Can not make progdocs
Not even sure how important this is considering the state of many of the
online docs...
I have doxygen installed as is noted for the requirements for 'make
progdocs', but the make doesn't find dot. I have no idea where dot went, is
or should have been...
I am installing und Suse 9.0 and it's rough. If you forget something
duringthe initial install, adding the package later
2004 Dec 30
4
Voicemail and Zapatel
My PSTN line doesn't allways hang up properly after it goes to voicemail.
The problem occurs when a caller hangs up during the initial greeting.
Even though the hangup accured, voicemail continues to record, usually a
fast busy and/or a teleco generated "please hangup now" message. After the
voicemail.conf 'maxmessage=180' expires the line simply stays offhook.
The hardware
2009 Feb 17
2
install.rb:655:in `command'': system("make") failed (RuntimeError) (GatoLinux)
I am using ruby *1.8.4* (2005-12-24).
I added this code:
#define RARRAY_LEN(a) \
((RBASIC(a)->flags & RARRAY_EMBED_FLAG) ? \
(long)((RBASIC(a)->flags >> RARRAY_EMBED_LEN_SHIFT) & \
(RARRAY_EMBED_LEN_MASK >> RARRAY_EMBED_LEN_SHIFT)) : \
RARRAY(a)->as.heap.len)
#define RARRAY_PTR(a) \
((RBASIC(a)->flags & RARRAY_EMBED_FLAG) ? \
2015 Jun 24
0
Re: "connect at power-on" NIC feature equivalent to that found in ESX
As a work-around, you can remove the NIC from the VM before starting it,
edit the network configuration files and then re-add the hardware. This
has worked for me.
On Wed, Jun 24, 2015 at 1:49 PM, Ryan, Lyle <lyle.ryan@cubic.com> wrote:
> With ESX and the vSphere Client, I can enable or disable a VM’s NIC to
> “connect at power-on”.
>
> This is very useful when cloning, since
2015 Aug 27
0
Column name expansion in data.frame()
Dear R developers:
I am trying to add a column to a data.frame. The following does the
trick by expanding the name of the first data frame with the prefix foo:
> data.frame(foo = as.list(data.frame(items = 1:3, bar=1:3)), items =
1:3) foo.items foo.bar items
1 1 1 1
2 2 2 2
3 3 3 3
However, the following special case produces an
2014 Nov 30
1
Using FPP preprocessor for Fortran Code
Dear R Developers,
For package seriation I use Fortran code. I recently got a request to add
#if defined(__ICC) || defined(__INTEL_COMPILER)
USE IFPORT
#endif
to the code since the Intel Fortran compiler otherwise has problems with
rand(). However, to enable the FPP preprocessor I have to either add a
compiler flag (-cpp for gFortran) which is possibly not portable or
change the
2009 Sep 08
0
[LLVMdev] 2.6 request - Bug 4879
On Sep 7, 2009, at 2:27 PM, Michael Lyle wrote:
First-- thanks to Daniel Dunbar for reporting this issue from my
> earlier coarse report on IRC and to Devang Patel for fixing it.
>
> I'm writing to request that this fix (r81058) find its way into the
> 2.6 release. Code compiled with clang that uses VLAs is horribly
> broken without r81058 (at least on x86-64). I don't