On 21-Apr-09, at 1:06 PM, Anthony Francis wrote:
You are correct, not seeing that means that the signaling was
either in
the audio stream (which doesn't survive compression) or it was sent
in
the sip signaling. However one must also note that your ITSP's
gateway
may have been having
I've been using one of the popular asterisk ISO distributions for a
couple of years and DTMF had always worked.
I recently switched to another asterisk ISO distribution, and outbound
DTMF is no longer working.
After doing a bit of digging, I noticed that the new distribution
wasn't setting
This link
(http://blogs.digium.com/2008/10/13/asterisknow-15-beta-available-more-coming-soon/
) seems to indicate that in order to upgrade AsteriskNOW v1.5 from
Asterisk 1.4 to 1.6, it's as easy as installing an upgrade package.
Does anyone know where to find that upgrade package? If it
support. That would be a bug
in my
packaging. I'll see what I can do with it.
Jason Lixfeld wrote:
I'm having some issues getting app_voicemail_imapstorage to talk
to my
IMAP server. From imapstorage.txt, I've got the voicemail.conf
configured properly, but if I leave a voicemail
I'm having some issues getting app_voicemail_imapstorage to talk to my
IMAP server. From imapstorage.txt, I've got the voicemail.conf
configured properly, but if I leave a voicemail for extension , I
see no indication that the module is trying to reach the IMAP server.
What am I
Hi folks,
I've been a trixbox user for a few years now but I'm thinking about
jumping ship.
Trixbox is great, but it's missing two features out of the box which
are really important to me: outbound faxing (hylafax) and imap
voicemail. I see no indication that they will be included
I'm getting this really strange permissions error when I try to start
asterisk:
# /usr/sbin/asterisk -U asterisk -G asterisk -gc
== Parsing '/etc/asterisk/asterisk.conf': Found
Running as group 'asterisk'
Running as user 'asterisk'
Asterisk 1.2.26.1 svn rev 79171, Copyright (C) 1999
I realize this may be less of an Asterisk question and more of a...
well... everything but asterisk, but still relating to asterisk
question.
I was looking for a Click to Dial/Web Dial solution and I found
AsteriDex. I'm looking for something I can use on the road where I
can hit an
...if the question is ... how do you get asterisk voice mail to
show up on an iPhone...I am all ears. Groovy concept - if
anybody has a hack - I'd love to see it.
Elvis
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
] On Behalf Of Jason Lixfeld
Sent: Monday
it in and then send the email.
It does not appear that the iPhone is using a proprietary
format so just try the default recording format and see
what happens.
-baji.
ps : I don't have an iPhone, nor have I used * voicemail yet
caveat emptor :-)
--
On 10/24/07, Jason Lixfeld wrote
It plays wav, but as far as I understand, * encodes the wav using
something like ulaw which iPhone doesn't support. If I can switch the
codec to pcm, that may work - is that possible?
On 24-Oct-07, at 12:25 PM, Darrick Hartman (lists) wrote:
Jason Lixfeld wrote:
I guess what I'm asking
Seems the answer was simple enough - set format=wav and it works
fine. Mine was set at wav49.
On 24-Oct-07, at 1:02 PM, Jason Lixfeld wrote:
It plays wav, but as far as I understand, * encodes the wav using
something like ulaw which iPhone doesn't support. If I can switch the
codec to pcm
Anyone managed to get this to work? What's the recipe?
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I ran the gambit and eventually, against my better judgement, I
finally broke down and installed HylaFax+IAXModem and I have had
absolutely zero problems with it. I'm extremely impressed.
This is the how-to I followed. A small amount of the instructions
aren't extremely clear; there is a
On 12-Sep-06, at 3:14 PM, Richard Klingler wrote:
Hi Jason
Hi! Sorry for the delay! :/
loadInformation6 model=IP Phone 7970SIP70.8-0-4SR1S/
loadInformation6
1. Stick with the 8.0.2 SIP image as it works best with asterisk...
at least for me (o;
- Here are TFTP server logs to
I have a 7970G with 5.0.3.0S Skinny (Load File: TERM70.5-0-3-0S) on
it and I'd like to get it up to 8.x.
- With the SEPMAC.cnf.xml in place (which was taken from voip-info
(http://www.voip-info.org/wiki/view/Asterisk+phone+cisco+7970+SIP
under This worked for me...)), I get Load ID
Ok, this is the 3rd RMA replacement I've gotten from Cisco. None of
the phones were bad, but I have been trying to upgrade them to SIP
firmware from Skinny and I've always managed to turn them into
doorstops because I've factory reset them and afterwards realize I
can't do anything with
On 10-Sep-06, at 1:58 PM, Chris Jones wrote:
Jason Lixfeld wrote:
Ok, this is the 3rd RMA replacement I've gotten from Cisco. None
of the phones were bad, but I have been trying to upgrade them to
SIP firmware from Skinny and I've always managed to turn them into
doorstops because I've
Moore - Aspendora wrote:
Not sure where you got your SIP image, but my SIP files have that
particular file in it.
On 9/1/06, Jason Lixfeld [EMAIL PROTECTED] wrote:
On 1-Sep-06, at 3:41 AM, Tomislav Parčina wrote:
In article [EMAIL PROTECTED], jason
[EMAIL PROTECTED] says...
I've been
I've been having some problems with a couple of 7970G so I decided to
factory them. In doing so, it seems to have rendered the phones
unbootable as they (as I understand it) are factory'd with some
version of SCCP code, which makes it constantly looking for
term70.default.loads file which
On 1-Sep-06, at 3:41 AM, Tomislav Parčina wrote:
In article [EMAIL PROTECTED], jason
[EMAIL PROTECTED] says...
I've been having some problems with a couple of 7970G so I decided to
factory them. In doing so, it seems to have rendered the phones
unbootable as they (as I understand it) are
I've been able to use RxFax directly to receive faxes over iax from
my DID/PSTN provider but I can't seem to get NVDetectFax to work. I
set the timeout to something ridiculously long, like 10 seconds, send
a fax with my laptop fax modem, but it never gets punted to the fax
extension,
Let me preface this by saying that, I realize I should be asking this
in *-bsd, but I posted there last week and heard nothing so I thought
I would post here to see if anyone had any thoughts:
I just compiled all these from source:
asterisk-1.2.9.1
zaptel-bsd (from svn downloaded on jun 8)
A T1 or E1 or PRI are all made up of copper, at least the last mile
is; that is the last portion between the Telco Demark and the CPE. A
T1 or E1 or PRI can be multiplexed into higher capacity circuits such
as DS3, or OCx and run over transports like ATM and/or SONET. These
higher
I have a 7960 with P0S3-08-2-00 firmware. Call waiting is enabled on
the phone:
...
call_waiting : 1
...
When I call from the 7960 to the 7960, I get a 486 busy here on the
asterisk console.
I can call from the 7960 to a soft phone no problem and vice-verse.
I can also call from soft
On 9-Jun-06, at 2:52 PM, Gary Reuter wrote:
On 6/9/06, Jason Lixfeld [EMAIL PROTECTED] wrote:
Kinda confused by this... I have a Cisco 7960 configured with a
couple SIP extensions configured on the phone. Just trying to dial
one extension from the other on the same phone, but when I do, I
enabling all for testing ..then limit..
On 6/9/06, Jason Lixfeld [EMAIL PROTECTED]
wrote:Kinda confused by this... I have a Cisco 7960 configured with a
couple SIP extensions configured on the phone. Just trying to dial
one extension from the other on the same phone, but when I do, I get
:
On 6/9/06, Jason Lixfeld [EMAIL PROTECTED] wrote:
Kinda confused by this... I have a Cisco 7960 configured with a
couple SIP extensions configured on the phone. Just trying to dial
one extension from the other on the same phone, but when I do, I
get:
Could the phone be returning 'busy
Kinda confused by this... I have a Cisco 7960 configured with a
couple SIP extensions configured on the phone. Just trying to dial
one extension from the other on the same phone, but when I do, I get:
-- Remote UNIX connection
-- Executing Dial(SIP/2001-ffd4, SIP/2002) in new stack
If you have to use it, make sure you only use the mpg123 bundled with
the asterisk distribution. mpg123 from any other source (yes, evem
the developer's website) will yield major issues.
On 8-Jun-06, at 8:12 AM, Matt wrote:
I've noticed that native music on hold volume seems to be very
Hi,
Here's my setup:
(PSTN)--[ASTERISK1]--(IAX)--[ASTERISK2]--(IAX)--[ASTERISK3]
I don't run asterisk 1, but I do run asterisk 2 and asterisk 3. I
have a DID via PSTN on asterisk 1 that is directed at asterisk 2 via
IAX. On asterisk 2 I want to direct that DID at asterisk 3. I have
I need a simple system with MoH, Meetme and timing using a TDM400P
with an FXO.
Any user reports?
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I'm looking to install Asterisk on an EM64T Dell 1850. PERC raid 1,
1GB ram, single 3Ghz Xeon.
Any red flags or anything I should know? Should I bother installing
a 64 bit OS? (gentoo-amd64)? Does asterisk work in 64 bit mode?
Should I turn hyper threading off? Etc?
In my experience, ztdummy doesn't work anywhere near as well as the
$150 card, especially if you want to start doing conferencing, etc...
On 6-May-06, at 12:45 PM, Sean Cook wrote:
hm... why not just use ztdummy and save the $150 for the card?
Sean
Steve Totaro wrote:
I have a TDM4xxp
Ran into this myself. Portage has a depend for mpg123 but it
installs the one that's also in portage. The one in portage is
broken (read: it doesn't work well with Asterisk). I avoid portage
when it comes to anything asterisk related. I build from source.
The asterisk source has
Running Asterisk 1.2.1 on Suse 10.0 X86-64.
Tried to get mpg123 0.59r which came with the 1.2.1 dist running on
this box, but all I get is poop:
as -o decode_i586.o decode_i586.s
decode_i586.s: Assembler messages:
decode_i586.s:44: Error: suffix or operands invalid for `push'
Anything else funky I need to do to get the exit context in
queues.conf working? I have the exit context defined, but when I'm
in the queue, I press 1 or 2 but it keeps me in the queue. No break
in musiconhold or anything. It's like the queue is ignoring/not
recognizing my keypress.
Thought it had to be something simple, thanks.
Still wrapping my head around the n priority.
On 16-Dec-05, at 11:40 AM, Kevin P. Fleming wrote:
Jason Lixfeld wrote:
; extensions.conf
[supportq-emergency]
exten = 1,n,Wait(0.5)
exten = 1,n,Voicemail(u9000)
exten = 1,n,Hangup
exten = 2,n,Wait
Going try my best to explain this and hopefully it will make sense:
We're trying to come up with something that we can only refer to as a
context picker. The idea is that if someone dials 98625551212, the
context picker will direct the call to the proper context based on
the dialing
(${LDIAXOUT}/${LDEXT},,r)
exten = ldout,2,Playback(last-error-was)
exten = ldout,3,SayDigits(${CAUSECODE})
exten = ldout,4,Playback(tt-somethingwrong)
exten = ldout,5,Hangup
exten = ldout,102,NoOp(seq 102 check)
On 14-Dec-05, at 5:56 PM, Jason Lixfeld wrote:
Going try my best to explain this and hopefully
Here's what I'm trying to do.. We have a small system, there are
only two of us. We both do sales and we both do support. We like
Queues better than music on hold with a bunch of dials happening in
the background to try our phones, then cells, etc. Problem is, we
don't like the idea of
,AgentCallbackLogin(${EXTEN:1},,[EMAIL PROTECTED])
Thanks
Jason Lixfeld wrote:
Here's what I'm trying to do.. We have a small system, there are
only two of us. We both do sales and we both do support. We
like Queues better than music on hold with a bunch of dials
happening
Can you do that without using agents?
On 22-Nov-05, at 11:08 AM, Jeremy Kenney wrote:
why don't you just build your cells into the queues and setup the
queue to ringall.
Jason Lixfeld wrote:
Here's what I'm trying to do.. We have a small system, there are
only two of us. We both do
as an agent.
Not sure if I'm doing a very good job of expressing what I'm trying
to achieve so I apologize in advance if I seem to be going in circles.
On 22-Nov-05, at 12:01 PM, Nicolás Gudiño wrote:
On 11/22/05, Jason Lixfeld [EMAIL PROTECTED] wrote:
Here's what I'm trying to do.. We
If you have only one member configured in queues.conf, it will bridge
the call immediately. 1 member configured and it works as expected.
Thanks for everyone's help!
On 22-Nov-05, at 11:22 PM, Jason Lixfeld wrote:
With this method, I can't seem to get the expected queue
functionality
Hi,
Is there any difference in the amount of hardware timing
something like a Wildcard X100P can provide over something like a
Wildcard TE411P? If someone has a machine that pretty much just does
very low volume MeetMe, Voicemail, SIP + IAX and 2 or 3 channels
worth of codec
Is there a way to get answer confirmation via IAX and not only via
ZAP? We get our outbound service via an IAX trunk to our provider so
we aren't in control of their ZAP configs, but ideally we'd like to
be able to achieve the answer confirmation functionality regardless,
especially in
Was using this monitor line to get soxmix to mix test-in.wav and test-
out.wav into test.wav.
exten = 1200,1,Monitor(wav|/tmp/test|m)
When I start the conference, the * console shows this:
monitor executing ( nice -n 19 soxmix //tmp/test-in.wav //tmp/test-
out.wav //tmp/test.wav rm -f
I'd like to provide the ability for a friend to conduct interviews
using an asterisk conference and then email them to him when done.
Kinda like a voicemail. There doesn't seem to be one single hook to
be able to do this so I'm wondering what other people have used to
jam this together
I have a 7960 desk phone and I'm running x-lite on my laptop. They are
both behind a NAT box so they would appear to * as being from the same
IP. I'm trying to make them ring at the same time but appear to
everyone as one extension. Is it possible to have them both register
to * as the same
I'm using * 1.0.3 on Gentoo 2004.3, zaprtc from bri-stuff for timing.
When I put a caller on hold, the volume of the hold music in the
callers ear is extremely loud. I'm using the default entry from the
musiconhold.conf:
default = quietmp3:/var/lib/asterisk/mohmp3
Volumes with a called or
back through the NAT to the phone connected to the laptop.
That's what I'm seeing. My SIP phone cannot register to my asterisk
box through siproxd. I'm not sure if it's the phone or siproxd but
it's not asterisk -- asterisk doesn't care.
Thanks.
On Mon, 2005-01-24 at 13:54 -0500, Jason Lixfeld
On Jan 25, 2005, at 2:02 AM, Adam Goryachev wrote:
On Mon, 2005-01-24 at 23:51 -0700, Kim Lux wrote:
I'm trying to get similar working with a Grandstream. I'm getting a
lot
of echo. My laptop is crashing when the call terminates.
What are you using for the NAT setup on your laptop ?
Ok, I have a 7960 that's plugged into my laptop. my home network is
wireless so I don't have a switch anywhere to plug the phone into
directly. I'm running siproxd on my OS X laptop and I can make
outbound calls from the 7960 fine (I guess I don't have the phone
configured to register
I've got a simple MeetMe conference configured using Asterisk 1.0.3 on
Gentoo. I'm using zaprtc for timing from the bri-stuff package.
extensions.conf
exten = 37455,1,NoOp(Drill Squad Conference)
exten = 37455,2,Monitor(wav,drillsquad-37455,mb)
exten = 37455,3,MeetMe(37455,pMs)
Now, when I
I've recently switched my * server from FreeBSD to Gentoo using the
same configs from FreeBSD on my Linux machine, except the new Linux
machine is running 1.0.3 where the old machine was running 1.0.2.
Whenever I try to dial into one of my DIDs, I get this in the debugs
and the call gets
Anyone know what these messages mean? I see then scrolling about one
every 10 seconds while running asterisk -vcdg
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Had the same issue with 5.3 1.0.2. Looks like some screwed up
somewhere.
On Dec 16, 2004, at 7:55 PM, Andrea Riela wrote:
Hi folks,
when I try to compile the freebsd port (1.0.2) I see:
ast_h323.cpp: In method `void
MyH323Connection::SendUserInputTone(char, unsigned int)':
ast_h323.cpp:725:
New 5.3 install. Saw some stuff around about problems with pwlib but
those were in 0.9 and have long since been fixed and I haven't found
anything else out there to explain this stuff.
Anyone have any ideas?
gmake[2]: Entering directory
Hey folks,
Using FreeBSD 5.2.1 and I've got the current zaptel driver installed
from ports (0.8_1) and current ports asterisk (1.0.1). I've set
options HZ=1000 in my kernel config, recompiled and rebooted and as far
as I can tell, I've done everything right but when I try to use the
Hey folks,
Using FreeBSD 5.2.1 and I've got the current zaptel driver installed
from ports (0.8_1) and current ports asterisk (1.0.1). I've set
options HZ=1000 in my kernel config, recompiled and rebooted and as far
as I can tell, I've done everything right but when I try to use the
I've read about Fax to Email, but is there such a beast as email to
fax? If not, what do people use to take care of outbound faxing?
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Sorry for the waste of bandwidth. Problem was the ztdummy driver
wasn't actually being loaded. Seems it was omitted by the startup
script provided with the port. I added it and re-ran the startup
script and it works like a charm now.
On Nov 30, 2004, at 1:04 AM, Jason Lixfeld wrote:
Hello
Hello,
I'm trying to set up a conference room. When I dial it's extension, I
get an audible error saying Not a valid conference room, please try
again followed by a disconnect. I've got debug sip peer 1001 (my
X-Lite client) and I see this in the logs: (I'm pretty sure it has
something to do
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