Since I started using 1.4 I'm also not getting MWI. I am not using
realtime.
MARK.
Benjamin Jacob wrote:
Hello ppl,
Am using realtime odbc storage for voicemail, sip users/peers, static
for extensions and so on.
My issue is I am not getting MWI for any fones, even tho I've got
Did you try mM()?
extensions.conf: exten =
s,n(dial),Dial(SIP/sipura2_1SIP/sipura2_2|30|mtTM(screen))
MARK.
Ed Greenberg wrote:
If I dial with an M() option to run a macro after answer, (how) can I
get music on hold for the caller?
___
I can't see anything immediately wrong here. Maybe it's something in
your dialplan? It sounds like Asterisk is dialing out but the PSTN
doesn't like the number you are dialing. Are you in an area that
requires 10 digits or does not like if you dial 11 for local calls?
From what I can tell,
Set(CALLERID(name)=VOICE MAIL)
Andy Vega wrote:
Is it possible to show CallerID names for dialplan applications? When
I call from phone-to-phone, it shows the CallerID from sip.conf or
iax.conf, but I don't know of any way to show CallerID Name when I
call the extension for an application
Although I'm convinced that Broadvoice doesn't have the most stable of
ping times, it seems like I get ping results that are approximately the
ping time +2000ms at times. Has anyone experienced this problem with
qualify on a SIP connection before?
So here, was the ping 20ms or 2020ms as
Specify a different context for each Zap channel (context=homephone)
(context=workline) in zapata.conf instead of just inbound-analog. Then
in your extensions define a context for each that includes a different
dialplan.
On the second problem, you could remove the forward from the Verizon
for the reply. my box is Intel based. and there is no USB
conflicts at all. I ran FC3 well, but, I think new kernel (in the FC4) may
be the place to see.
Thank you
Kumara
- Original Message -
From: MF Hulber [EMAIL PROTECTED]
To: Kumara Jayaweera [EMAIL PROTECTED]; Asterisk Users Mailing
List
based. and there is no USB
conflicts at all. I ran FC3 well, but, I think new kernel (in the FC4) may
be the place to see.
Thank you
Kumara
- Original Message -
From: MF Hulber [EMAIL PROTECTED]
To: Kumara Jayaweera [EMAIL PROTECTED]; Asterisk Users Mailing
List - Non-Commercial
It sounds like you need to adjust your txgain or possibly rxgain.
MARK.
jonny hashem wrote:
my customers complain that when they make a call they
hear the another side very well but the another side
hears the first side well but in low sound.what is the
ptoblem here and i have to change?
I don't use Fedora but I do use RHEL AS 4 without any problem. Do you
have any USB conflicts?
MARK.
Kumara Jayaweera wrote:
Hi all,
Does anyone run Asterisk on FC4? with Digium's TDM40B cards. any success
stories? my Intel 865 M'd+ Intel 3.0GHz freezee during installation (FC4).
Please any
Without having said why you want to connect the phone through the
computers I have a hard time understanding why you want to do that. As
someone else has suggested, either buy IP phones and connect them
directly to your LAN or buy analog adapters and use your existing phones.
I don't see the
The next question is, was your call successful? I see you dialed an 8
digit number. Is that what's required on your line?
MARK.
Eric Wieling aka ManxPower wrote:
VoIP Newbie wrote:
Hi all,
When I was making calls from an IP phone, through a X100P, to PSTN,
the following error was
Ok, first I'll tell you some of the things I'm ignoring because you said
you are having trouble receiving the inbound call. First, why aren't
you using DISA? Ok, so you want to try this out, that's fine. Second,
it appears you set the variable NR to be empty so I don't know why you
are
It's not just him. The list was majorly down from sometime on the 29th
until the 1st.
MARK.
Derek Whitten wrote:
must be just you.. get messages all day every day here..
:-)
On Mon, 2005-08-01 at 05:49, Howard Leadmon wrote:
This is usually a very active list, but looking
I don't think anyone was getting mail from any of the lists (or at least
a very few were). I know that I wasn't but I have started to receive it
now.
MARK.
Michel Koenen wrote:
Same here, nothing is coming in anymore on my gmail address neither. I
read your posting by going to the web
Just a thought: do you have DSL on the PSTN line and are you using a
line filter?
MARK.
Jochen Witte wrote:
Hello,
I have an IVR with Intel HMP SIP stack, which is a peer behind my Asterisk
box (Asterisk 1.0.7, Digium PRI). When dialing in via PSTN, there are
klicking sounds in the
Yes, I always have two.
MARK.
Billy Dunn wrote:
Does everyone have two processes running for mpg123? I always have
them when I'm running an idle Asterisk box. No calls going in or out
and nothing off hook. Is this normal? Thanks!
5008 ?S 0:00 mpg123 -q -s --mono -r 8000 -b
And is there some bit of information I get at verbose level 255 that I
don't get at 254? It just seems like a lot of levels.
MARK.
John Novack wrote:
MF Hulber wrote:
It's a little odd. Something like asterisk -v4 seems more
appropriate. You can also use set verbose level so that you
, Jul 10, 2005 at 09:49:37PM -0400, MF Hulber wrote:
Maybe it shows up after a certain verbosity level. Try asterisk -r
When I do that NoOps always show up.
Looks like you're right. Guess I never used enough v's ;)
___
Asterisk-Users
Maybe it shows up after a certain verbosity level. Try asterisk -r
When I do that NoOps always show up.
MARK.
George Garvey wrote:
I believed from reading that NoOp would display something on the
console. I assume the console is * in the foreground. During testing,
I've often been
Take a look here:
http://www.digium.com/index.php?menu=product_detailcategory=hardwareproduct=TDM400P
MARK.
Dan Adams wrote:
Hi, I am sorta a newbie to the asterisk community at least in the realm of
hardware types. I was wondering, what type of card is used to allow asterisk,
on a
My apologies for any bounced mail from me today. My mail server was
having a bit of a fit.
MARK.
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Try two different entries:
sip.conf:
[general]
;disallow=gsm
;allow=ulaw
port = 5060 ; Port to bind to
bindaddr = 0.0.0.0 ; Address to bind to
context = default ; Default for incoming calls
callerid=No CallID
register =
I have two different inbound DID providers and on each network I have
had reports in the last several days from callers that when they try to
call me from a cell phone their provider reports that they are trying to
make an international call and thus doesn't let them complete the
calls. In
Has anyone had any luck configuring the UTStarCom F1000 with asterisk? I
get the wireless to work but the sip registration is a problem. Below is
my SIP Debug. The server is 192.168.0.80 and the phone is 192.168.0.166.
Sip.conf:
[f1000_1]
type=friend
host=dynamic
I have Canada DIDs from the companies below and both have been reliable:
http://www.unlimitel.ca
http://www.livevoip.com
MARK.
Adrian A wrote:
Does anyone have any recommendations for a SIP/IAX provider I can use
for inbound callls? The plan is to have a 1800 number people can call
Unfortunately I believe there is a lot of truth to it. The speed in
which 911 legislation took effect is no coincidence and you don't see
the big telcos complaining about it. He's right about the price issue
too. Do you see how much big providers charge for VoIP service?
MARK.
Colin
In making calls outbound through various VoIP origination providers
people on the PSTN network have been complaining of poor voice quality.
Diagnosing has been difficult with limited feedback but I tried using
different originators and seem to have the same problem. I used
TestYourVoip.com
Although you may not see it displayed on a reload it may actually be
loaded. Try show dialplan and its alternatives to be sure that what
you are looking for is not loaded.
MARK.
John Melody wrote:
Is there a limit to the number of extensions that can be defined in
extensions.conf ? I just
I can't disagree with you on the customer service aspects. I have found
the new online ticket reporting system a bit better but I often get the
feeling that they feel customers are a nuisance. As far as the VoIP
service, I haven't had any problems to speak of. I'm still waiting for
a DID
Anyone know when the new IAXy will be available and what changes there
are besides the form factor?
MARK.
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I've tried it by changing the MOH context after I have identified the
caller but I find that mpg123 doesn't properly switch over to the new
context. Once it starts playing a stream it appears to be stuck with it.
exten = s,1,SetMusicOnHold(default)
MARK.
[EMAIL PROTECTED] wrote:
Is it
Have you seen this story? Cisco definitely wants to own the VoIP
market. I wonder what effect this will have on Sipura products.
http://story.news.yahoo.com/news?tmpl=storyu=/nf/20050427/bs_nf/33554
MARK.
___
Asterisk-Users mailing list
These things are dirt cheap. Are they any good?
MARK.
Iassen Hristov wrote:
Maybe this fits the bill.
http://www.gigafast.com/products/product_detail/EE2400-SS.htm
It retails for less than $100
Message: 9
Date: Fri, 22 Apr 2005 10:42:20 -0700
From: Max Clark [EMAIL PROTECTED]
Subject:
I don't think it's correct to put dashes in the CIDNum.
MARK.
Paul Fielding wrote:
Hmmm... I still can't get name, though number works. Perhaps I'm
missing something?
context livevoip in iax.conf that hooks me to livevoip
dial 9 in front of long distance number to dial livevoip instead of
Try terminating the GotoIf statement with a ')'
MARK.
Mark Halverson wrote:
I have unlimited local calling on my cell phone provider but not long
distance; so I wanted to create authentication based on me calling in and
authenticating based on the callerid of my cell phone.
Here is what I tried
BTW, you hijacked someone's thread...
Try this:
-- sip.conf --
register = 5595551212:[EMAIL PROTECTED]/5595551212
[broadvoice-in]
type=peer
host=sip.broadvoice.com
context=broadvoice-inbound
canreinvite=no
qualify=yes
nat=no
-- extensions.conf --
[broadvoice-inbound]
exten =
I am able to set name and number with Livevoip. Make sure your
variables are actually being set.
exten = s,1,SetCIDNum(xx|a)
exten = s,n,SetCIDName(first last|a)
exten = s,n,Noop(Caller Name: ${CALLERIDNAME}, Number: ${CALLERIDNUM})
MARK.
Cameron Schaus wrote:
Is there any way I
asterisk/sounds.txt
Josiah Bryan wrote:
On Tuesday 05 April 2005 1:24 pm, Dov Bigio wrote:
Hello all,
I am looking for a list of all available sound files for asterisk and a
transcription of their content, so that I can have someone translate them
into portuguese.
I vaguely remeber reading
I believe there is a problem with LiveVoip ringback that is in the
process of being resolved.
MARK.
Jon Califf wrote:
If I call my 800 number, I do correctly go to the queue, the agents
are dialed and the call can be picked up but I don't hear any ring
tone while I'm waiting. If I pick up one
You might try adding:
exten = h,1,Hangup
Chris Blake wrote:
Greetings *`s,
I have set up a call which constantly loops a pre-recorded message
waiting for the user to press a digit on their phone. At this point the
call is sent elsewhere in the dialplan.
But if the called party doesn`t press any
I don't have any difficulty with DTMF with LiveVoip incoming or outgoing.
MARK.
Brian Litzinger wrote:
I read in the archives a number of discussions about livevoip, DID,
and DTMF not working.
However, no resolutions.
I just setup a livevoip DID and indeed the DTMF does not work.
The same asterisk
asterisk*CLI show application dial
snip
'D([digits])' -- Send DTMF digit string *after* called party has
answered
but before the bridge. (w=500ms sec pause)
snip
MARK.
Shadow Roldan wrote:
Hi Everyone
I'm trying to pass a call to a outbound SIP peer(broadvoice) and pass
DTMF
The way it works with my provider is that although both numbers enter
the same context, each number will match its own extension. If I have
two numbers: 11 and 22 it works as follows:
[sip-in]
exten = 11,1,Noop(First number dialed)
exten = 22,1,Noop(Second
iax.conf
[livevoip-out]
type=peer
host=217.160.244.186
auth=md5
context=livevoip-dialout
callerid=aaa bbb (xxx) xxx-
username=username
secret=secret
qualify=yes
notransfer=yes
exten = s,n,SetVar(LIVEVOIP=IAX2/[EMAIL PROTECTED])
exten = s,n(dial1),Dial(${LIVEVOIP}/dialed-number|30)
Take a look at the Random() command.
MARK.
Ronald Wiplinger wrote:
I want to change the below lines:
exten = _011.,1,SetGroup(line1); set current group to
line
exten = _011.,2,CheckGroup(1); check line1 does
not have more than 1
exten = _011.,3,Dial,SIP/[EMAIL
voicemail.conf
;format=gsm|wav49|wav
;format=wav|wav49|gsm
format=wav
Jim Sturtevant wrote:
Ive recently installed asterisk and am working with the email a
voicemail function. When a voice msg is left 4 files are created in
the /var/spool directory. They are .gsm, .txt, .wav and .WAV. The
.wav
For fun, try changing your context from [sip.broadvoice.com] to your
phone number [55]
MARK.
Courtney Couch wrote:
The asterisk config that i have is:
[sip.broadvoice.com]
type=peer
authname=55
canreinvite=no
context=test
dtmf=inband
dtmfmode=inband
fromdomain=sip.broadvoice.com
Me too.
MARK.
Tim Connolly wrote:
Anyone else seeing problems trying to browse the wiki?
Like no response on port 80?
___
Asterisk-Users mailing list
Is it correct to have the same context (202) listed twice in sip.conf?
Courtney Couch wrote:
I have Polycom ip-300 phones that worked yesterday but dont seem to
work today (at least dtmf signalling once connected to the asterisk box)
The current configuration is:
[general]
port = 5060
bindaddr =
Haven't used it since I don't think this is available yet. The only
thing that bothers me about this unit is that it's single line. I
currently have a similar analog Panasonic 2.4GHz system. In general I
like the Panasonic except fir a couple things:
They never seem to support the mute
All of my sounds are under /var/lib/asterisk/sounds. I don't have a
directory /usr/share/asterisk. None of my configuration files have a
pointer to a sounds directory so I'm assuming it's looking in
/var/lib/asterisk/sounds by default.
MARK.
G.Marshall wrote:
Hello,
I have moved from
rfc3261
http://www.faqs.org/rfcs/rfc3261.html
Ronald Wiplinger wrote:
Where can I get a list of all possible SIP ... response numbers and
their meaning?
bye
Ronald
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Asterisk-Users@lists.digium.com
I'm getting the same problem running Asterisk CVS-HEAD-03/21/05-03:24:01
built by [EMAIL PROTECTED] on a i686 running Linux which is the
code from yesterday.
Robert Goodyear wrote:
FWIW I get the same exact error at the end of every VM session as
well, thus:
-- Playing 'vm-intro'
I have the same motherboard. I put the card in the 2nd slot from the
bottom. In this slot, if you look at the manual, it will possibly be in
conflict with some USB channels. I believe I may have disabled one of
them but in any event I'm not using any USB devices. Otherwise, I
didn't have
I started with Broadvoice recently but I am constantly having problems.
I got everything configured and now it is dropping outgoing calls after
40 seconds and incoming calls are going direct to voicemail. Getting
customer service is proving very difficult.
I've started to look at some of
that's pretty reliable and only charge 1.27
cents per minute. You can add a toll-free number for an extra $1 per
month. Setup was a breeze!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]] On Behalf Of MF Hulber
Sent: Sunday, March 20, 2005 9:45 AM
To: [EMAIL
I took a look at teliax. The pay as you go plan appears not to include
international dialing and the commercial plan is fixed price of $44.99
per month capped at 500 international minutes a month. Are you aware
if they have international rates based on usage?
MARK.
Rich Adamson wrote:
I have the same problem but not with X-Lite. I was using Broadvoice
all day today and then I changed rate plans because I thought
everything was working well. Now my calls get dropped within 2 minutes
and my incoming calls go direct to broadvoice voicemail.
MARK.
Scott Wolfe wrote:
Try changing the extension from Broadvoice1 to the actual phone number
(and don't send your secret in a public email or maybe that's Chris'):
[*8475100139*]
type=peer
;user=phone
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
fromuser=8475100139
secret=XXX
username=8475100139
I concur. I rebuilt today and now I seem to be able to dial out.
MARK.
Chris Nibeck wrote:
thank you everyone!
It does not seen that it was configuration problems at all.
It appears it was the CVS that I was using from yesterday.
I decided to start over, downloaded the latest CVS, recompiled, and
Original Message
Subject:
Re: [Asterisk-Users] BroadVoice configuration changes for
Outbound
Date:
Sun, 06 Mar 2005 19:11:22 -0500
From:
MF Hulber [EMAIL PROTECTED]
To:
Asterisk Users
I'll give that a try. Do you make a symbolic link to Madplay as mpg123
or is there a way to configure * to use a different executable? An
issue I have with MOH at all is that if I'm on a conference call, I
don't want MOH to play.
MARK.
Ken Godee wrote:
MF Hulber wrote:
I'm looking
Sorry, in my haste I didn't read your musiconhold.conf that answers my
question about setting up the executable.
MARK.
Ken Godee wrote:
MF Hulber wrote:
I'm looking for a simple way to disable MusicOnHold in my
environment. I'm not really interested in having it and it causes
too many
I'm looking for a simple way to disable MusicOnHold in my environment.
I'm not really interested in having it and it causes too many problems
with hanging mpg123 processes and memory management errors. The problem
is, so many other modules seem to depend on it. I can't just cause a
noload
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