Hello,
What are some solutions folks are using for faxes
(inbound)? I was considering the Stanafax option.
Regards,
---
LB
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-user
care and thanks!
Regards,
LB
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Lenny
Sent: Tuesday, September 05, 2006 10:26 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [asterisk-users] Keys pressed not register
ohn covici [mailto:[EMAIL PROTECTED]
Sent: Saturday, September 02, 2006 1:13 PM
To: Lenny
Subject: RE: [asterisk-users] Keys pressed not registering ...
The dtmf is in the peer details of the trunk which turns into the
sip.conf, however remember that if you change this, your provider has
to chan
John,
Ok .. I'm really under the idea that this is an ISP issue and a conflict
with trying to run VoIP on an already VoIP enabled line..
Thanks for the suggestions...
LB
-Original Message-
From: John covici [mailto:[EMAIL PROTECTED]
Sent: Saturday, September 02, 2006 1:13 PM
To:
.
In freepbx, its in the peer details of the trunk.
on Sunday 09/03/2006 Ronald Wiplinger([EMAIL PROTECTED]) wrote
> Lenny wrote:
> > Hello Ronald ..
> >
> > This is what I'm trying to learn of now ..
> >
> > Where in freepbx do I place these setting
PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ronald
Wiplinger
Sent: Saturday, September 02, 2006 12:20 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Keys pressed not registering ...
Lenny wrote:
> Hello Ronald ..
>
> This is what I'm
urday, September 02, 2006 11:52 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Keys pressed not registering ...
Lenny wrote:
>
> Hello all,
>
> For some reason when dialing in I get the IVR or if I forward to my
> conference line... any ke
CFLAGS ... looks like Gentoo ..
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen
Sent: Saturday, September 02, 2006 8:54 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Problems compil 1.2.11
On Thu, Aug 31, 2006 at 05:5
ers] Keys pressed not registering ...
Note that the other end also has to make the change as well, so you
need to talk to them unless its yours.
on Saturday 09/02/2006 Lenny([EMAIL PROTECTED]) wrote
> Hello,
>
> So make the changes to what part in FreePBX?
>
> Thanks..
>
ovici
Sent: Saturday, September 02, 2006 7:08 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Keys pressed not registering ...
Note that the other end also has to make the change as well, so you
need to talk to them unless its yours.
on Saturday 09/02/
[asterisk-users] Keys pressed not registering ...
Try changing the DTMF mode for that line, I've found that if rfc2833 doesn't
work, inband will
>-Original Message-
>From: [EMAIL PROTECTED]
>[mailto:[EMAIL PROTECTED] On Behalf Of Lenny
>Sent: Saturday, September 02,
Hello all,
For some reason when dialing in I get the IVR or if I forward
to my conference line... any keys pressed seem like they aren’t received
.. Like I’m pressing them, but they aren’t being registered with
the server .. Any ideas?
I’m using the vmware nerdvittles build, the lat
I can't seem to register properly with broadvoice servers. Looking at
tcpdump and log files I see registrations attemtps and traffic to
broadvoice, but no traffic or error messages of any kind from broadvoice.
Do my rules look ok ?
ACCEPT all -- anywhere anywhere
ACCEPT all
d
list price).
Looks interesting.
--
Lenny Tropiano E-mail: [EMAIL PROTECTED]
Partner, Networking Specialist Pager: [EMAIL PROTECTED]
VoIPing, LLCURL:http://www.voiping.com/
PO Box 867, Cedar Park, TX 78630-
VOIP pioneer predicts a roiling 2005 for IP telephony
Eetasia.com (subscription) - USA
Open source software communications will begin to influence the VoIP market
in a big way next year, according to VoIP pioneer Jeff Pulver. ...
http://www.eetasia.com/article_content.php3?article_id=8800354924
(
I have my Linux 2.6 kernel with the necessary HDLC config and also recompiled
zaptel accordingly. On modprobe, I get:
Found a Wildcard: Digium Wildcard T100P T1/PRI
Debug: sleeping function called from invalid context at mm/slab.c:2000
in_atomic():0[expected: 0], irqs_disabled():1
[<0211e605>]
So Asterisk gurus out there, is there a nice clean way in the dialplan
to determine if the caller is coming from a transferred call, and on
the unavailable context in the dial, instead of going to e-mail go
back to the transferee?
If anyone has this sort of logic or could spit out an extensions
Also, I imagine I could use
call queues but this is supposed to be a Reception phone and that
doesn't seem to fit here.
Does anyone have any suggestions? Any help will be greatly appreciated.
Thanks.
-- Lenny
Lenny Self
[EMAIL PROTECTED]
___
A
These taken tonight (9/22/2004) at the Expo and Reception
Enjoy. http://photos.tropiano.org/gallery/astricon-2004
Lenny
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update
>
> Does anyone know if the Marriott has Wi-Fi? LAN connection in the room?
According to the STSN (www.stsn.com) hotel locator, the Marriott does
have in "room" wired access. Wireless access and Meeting room access.
At $9.99/day (cheaper usually if you buy blocks of in multiple days)
locked to
> > Did I see something on here about using an AGI script to do reverse
> > lookups via anywho.com? I have a PRI that only gets caller-id number and
> > no Alpha.
[...]
I put a copy of it here...
http://www.voiping.com/calleridnamelookup.agi
It was written by James Golovich <[EMAIL PROTEC
Hello. I've seen several posts talking about line quality using Digium
cards that are sharing IRQs or on machines where X is running but after
trying all of those fixes I am still having a problem with line static
on outoing calls. BTW, calls that are from one extension to another
extension h
I have these phones working well with Asterisk, thanks to
http://www.freedomphones.net/polycom/files
The code there is not the most current (and my Polycom Extranet login hasn't gotten
updated yet to allow downloads). Been waiting a while for their support folks to
update the access.
If
We're doing some SIP development and have a question on "additional parameters"
supplied to the register (in this case maddr= and the non-standard clport= in
our example below).
What we're experiencing is the INVITE doesn't included these parameters
and they get dropped when the INVITE is sent to
I really like the functionality that Gastman provides, it would solve a problem
I currently have that "Secretaries" don't know when/who's on the phone before they
transfer the caller...
But I'm seeing some oddities, maybe just because the code line hasn't
been updated in a while. Take a look
I just read that Pingtel (www.pingtel.com) will be releasing it's IP PBX (which
runs under Linux) to open source (similar model to Redhat Linux, charging
for support, etc.). Read more about it at... http://www.pingtel.com/a_opensource.jsp
and http://www.tmcnet.com/usubmit/2004/Feb/1024036.htm
I
Looking for a current precompiled Win32 binary for gastman, don't
have a build environment for Windows. Also does gastman compile
under Linux and is there a current binary as well...
Thanks
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists
I've noticed I have an issue with my Dialplan ... apparently instead of a busy
signal when the caller is busy it falls through and gets a Congestion...
What's the proper syntax for this, reorder tone when there is a reorder and
busy when there is a busy...
SBC is a T1/PRI.
[macro-sbc-outdial]
ex
Here's a simple small expect script ...
I call it "phreboot", usage: phreboot IP
$ phreboot 10.99.1.1
-- cut here --
#!/usr/bin/expect -f
set timeout -1
spawn $env(SHELL)
match_max 1
send -- "telnet [lrange $argv 0 0]\r"
expect -exact "word :"
send -- "cisco\r"
expect -exact "Phone> "
send
t of recording a "mp3" of a ringing phone) for the person
to get a ringing sound instead of the MOH?
Thanks,
Lenny
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or updat
Folks --
I know this isn't directly an * issue, but I need to buy 14 7940s (preferably)
(or 7960s if the price is also reasonable) --- no power cubes, immediately. If
anyone has a good price, contact me offline at 512-427-1324 or
lenny @ rocksteady.com
Thanks,
I'd like to play a sound to a user I dial (via SIP) once
they answer play the sound and then allow me to talk to them.
The new Cisco 7960 SIP code allows to set lines to autoanswer
via the speaker phone, I'd like to play a "tone" after it rings
through and then talk...
Any thoughts on how to do t
y more interested in the performance of the codec (ie what kind of raw
power will I need to run it, how many can I run at once on a decently powered box
etc...)
Lenny
-Original Message-
From: James O. Sizemore III [mailto:[EMAIL PROTECTED]
Sent: Thursday, March 27, 2003 1:29 PM
To: [EMAI
What about G.723.1? Anybody have any experiece with this? I know I can order the
documentation (including C source) from the ITU, what does this entitle me to do? Are
there any licencing gotcha's to going with this approach?
Lenny
-Original Message-
From: Steven Critchfield [m
AIL PROTECTED]
Subject: Re: [Asterisk-Users] MWI behavior change?
How about we say "yes" only if there are new messages
cvs update and let me know if that worked.
Mark
On Wed, 26 Mar 2003, Lenny Tropiano / asterisk.org Mailing list wrote:
> Looking at the cvs rdiff from 1.5 to 1.
ow the message isn't
going out correctly. Try turning on sip debug with "sip debug" and see if
the "Notify" message is going out with the correct values:
Content-Type: application/simple-message-summary
Content-Length: xxx
Message-Waiting: no
Voicemail: 0/0
Mark
On We
Looking at the cvs rdiff from 1.5 to 1.6, apparently the behavior changed?
The old way it just checked "inbox" for new messages and turned on the MWI. Now it
seems
to add the new+old to turn on the Message Waiting Indicator. Personally after I've
listened
to the messages and saved them in a Ol
In the latest CVS build (today) my MWI indicator on my 7960 came on and stays on
without any messages in my mailbox. I cannot get it to go off? A bug that was
introduced today? Or some config change?
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
gcc -shared -Xlinker -x -o chan_phone.so chan_phone.o
gcc -c -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g
-Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i586
-DASTERISK_VERSION=\"CVS-03/25/03-10:49:30\" -DINSTALL_PREFIX=\"\"
-DASTETCDIR=\"/etc/
oFR technology working over IP
but the price is prohibitive)
Thanks in advance.
Lenny Post
Technical Manager, Engineering/Operations
Quick Link Communications Ltd.
(403) 537-5907
http://www.qlccom.com
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
h
ZT_SIG_SF undeclared?
make[1]: Entering directory `/usr/local/src/asterisk/channels'
gcc -c -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g
-Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i586
-DASTERISK_VERSION=\"CVS-03/12/03-21:24:47\" -DINSTALL_
-- Registered SIP '' at 192.70.239.2 port 5060 expires 3600
-- Executing Playback("SIP/lenny-4ee2", "transfer|skip") in new stack
-- Executing Macro("SIP/lenny-4ee2", "dial|7555|SIP/lenny-lap") in new stack
-- Executing Dial(&qu
This is two SIP/7960 phones... when one is put on hold, the
music on hold doesn't come to the other. I have the official
mpg123 code installed (not the mpg321...). I do have the zaptel
driver installed since I have a Wildcard FXO card in there
for PSTN access...
zapata.conf:
musiconhold=
I had updated CVS this morning and it broke me being able
to call the voicemail extension from my SIP/Cisco 7960 phone
it won't receive DTMF digits...
Restored back to Mar 10 2003 and it worked just fine...
___
Asterisk-Users mailing list
[EMAIL PROTEC
A bug that I've been meaning to report...
When you call someone and have the remote person put
you on hold (both are Cisco 7960/SIP recipients), when
they come back off of hold they can hear me, but I
cannot hear them... sounds like one of the audio
channels is not restored properly..
I'll be h
I can't login anymore... used to be able to. Timing doesn't seem to be working well
any ideas? Also what is this "NOTICE" I'm getting?
*CLI> == Accepting call on 'SIP/lenny-b19c' ("Lenny Tropiano" <>)
-- Executing VoiceMa
I have the following in extensions.conf:
[global]
MP3ROOT=/var/lib/asterisk/mohmp3
[default]
exten => ,1,Answer ; Answer the line
exten => ,2,DigitTimeout,5 ; Set Digit Timeout to 5 seconds
exten => ,3,MP3Player(${MP3ROOT}/sample-hold.mp3)
The command that
Can I get the voicemail application turn on / off the MWI (message waiting indicator)
on the Cisco 7960?
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
I have the following in my config files:
extensions.conf:
exten => ,1,Directory(employees)
[employees]
exten => ,1,Playback(transfer,skip)
exten => ,2,Macro(stdexten,,SIP/lenny)
;;
exten => 5556,1,Playback(transfer,skip)
exten => 5556,2,Macro(stdexten
49 matches
Mail list logo
