Sys. Concept Inc. wrote:
I want * to answer the phone when call comes-in.
I've enabled /dev/phone0 in phone.conf and created /dev/phone0 with a
command:
mknod /dev/phone0 c 100 0
Though, when I start * I get:
Parsing '/etc/asterisk/phone.conf': Found
Sep 12 00:18:42 WARNING[16384]: chan_phone.c:9
is it in ebook format at all?
I am a blind computer user and have no way of getting it scanned in to my
computer even if I were to purchase it.
thanks
hank
- Original Message -
From: "Sys. Concept Inc." <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Saturday, September 11, 2004 10:08
I want * to answer the phone when call comes-in.
I've enabled /dev/phone0 in phone.conf and created /dev/phone0 with a
command:
mknod /dev/phone0 c 100 0
Though, when I start * I get:
Parsing '/etc/asterisk/phone.conf': Found
Sep 12 00:18:42 WARNING[16384]: chan_phone.c:950 mkif: Unable to open
'
I am hooked up with broadvoice and have been having no problems that are
major there voice mail system went on the blits for about 30 minutes
yesterday but that was about it.
what kind of problems you expierencing?
- Original Message -
From: "Joel Gathercole" <[EMAIL PROTECTED]>
To: <[EM
On Sun, 12 Sep 2004 00:10:51 +0530, Renu Rangnekar
<[EMAIL PROTECTED]> wrote:
> Can i make asterisk to work as a softphone by itself and with the help of sound card
> can i route the call to a GSM circuit. Basically i want to connect the incoming
> landline call to a GSM circuit through a PBX.
W
David wrote:
It sounds like my lockups may be related since my TDM422b card has the FXS FXS FXO
FXO configuration and doesn't have an FXO in position 1 either.
My card is identified in software as Rev E/F and has the wire jumper on the back.
Further investigation shows that my TDM cards have the w
On Saturday 11 September 2004 14:39, Brian Cuthie wrote:
> Anybody know an easy way to adjust audio level of recordings made in
> Asterisk (using the 'record' application)? I've noticed that recordings
> using the "wav" format are about twice the level of those made using
> "WAV" or "wav49". Unfor
Beginners.
Paul
Paul Mahler
[EMAIL PROTECTED]
Signate, LLC
665 Third Street
Suite 100
San Francisco, CA
94107-1901
Asterisk Services and Training
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Sys. Concept Inc.
> Sent: Saturday
On Sat, 11 Sep 2004 18:23:07 -0400, Ed Guy <[EMAIL PROTECTED]> wrote:
> Your problem appears fixed; I provisioned your account manually.
> I'm not sure what caused the problem, but, I'll make sure that
> the FWD support and Operations team is aware of the issue as it may
> affect other FWD features
Does anybody have the book: VoIP Telephony with Asterisk by Paul
Mahler.
Is it for beginners or advanced users?
--
#Joseph
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The only reason I had problems before that was because of my braindead ISP
who kept dropping my connection and causing BV to unregister... They have
fixed their stupidity and I haven't so much as restarted * in about 3 weeks
and it's been working great!
-Chris
I've been using it for 3 months now, it's been up rock solid for about 3
weeks now, the only major complaint I have is the VoiceMail dropping problem
and that's *'s fault, not BroadVoice's
-Chris
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htt
Hello,
I am just curious how many people are hooked up with BroadVoice and have
recently been experiencing a lot of dificulty.
Joel
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Forgot to mention I have the sip.conf and phone set up to use ulaw,
alaw, libc in that order.
Guys,
I have a problem that I can seem to run down. I'm running
CVS-HEAD-09/10/04-18:14:15
on a Celeron 500 with an Intel chipset motherboard. The audio from my
GS phone to *
is sometimes decoded at a
Guys,
I have a problem that I can seem to run down. I'm running
CVS-HEAD-09/10/04-18:14:15
on a Celeron 500 with an Intel chipset motherboard. The audio from my
GS phone to *
is sometimes decoded at about twice the normal speed on some outgoing
calls. Also.when
recordind a message to * from the
On Sat, 2004-09-11 at 21:41, Marc Storck wrote:
> Hello,
>
> I want to link several * boxes together. Some of them are dedicated as
> "user" servers (SIP and IAX clients connect to them) and some are used
> as PRI servers (where the PRIs are hooked onto).
>
> I think TDMoE is the only channel t
Title: Help!
DEAN RENU
LET ME KNOW ONCE YOUR GSM INTERFACE IS DONE, WE
WILL BE INTERESTED TO BUY THESE
THANKS AND KIND REGARDS
ANDY SINGH
+1 416 856 7047
- Original Message -
From:
Renu
Rangnekar
To: [EMAIL PROTECTED]
Sent: Saturday, September 11, 2004 2:4
Hello,
I have been running some tests to get a better understanding of PRIs and the
HANGUPCAUSE variable and I'm not having any luck. I have tried calling
disconnected numbers and the call is connected through to my extension and I
hear the tri-tones. And it looks like HANGUPCAUSE is always 16
(
Hi,
Unless you have a very large configuration, the bandwidth of a 100Mbps
ethernet will not be the issue. Theoretically you could have 10 E1's worth of
TDMoE traffic on a single 100Mbps wire. I have sucessfully used EuroISDN with
one 31 channel TDMoE (E1 is 32 64Kbps channels, where one is use
Hello,
I want to link several * boxes together. Some of them are dedicated as
"user" servers (SIP and IAX clients connect to them) and some are used
as PRI servers (where the PRIs are hooked onto).
I think TDMoE is the only channel type where you can group different
Interfaces into a single gro
Ahh, I just read about this somewhere. There are two versions of IAX, IAX1
and IAX2. IAX1 uses udp port 5036 and IAX2 uses udp port 4569. For what
ever reason, I guess to avoid conflict & confusion until IAX2 becomes
standard, the port is hardcoded into the source.
About the IAX working flawlessl
First, I am surprised at IAX... My asterisk server is behind a
firewall, I am behind a firewall, and my iax client connected...
Voodoo??? :)
But, the firewall has one unblocked port, which I think I should set
to the IAX, so the magic won't be necessary, and it makes sense to
think that there will
Hi.
I've succesfuly installed Asterisk (at least I think
so since compilations were cool). I havent installed
my x100p yet and I wish to make a call from the
command line to test my configuration is it
possible?
I've seen there is a "sample.call" file at the
asterisk source dir, it says you
I have somewhat miraculously got my server to stay up for over 24 hours now. I was
at my remote location, however, and I can't make calls that used to work find. I
get the following messages. I get a brief bit of good sound and about the time I
see the message "Ooh, voice format changed to 4" al
Raul, will you learn how to post correctly a message to a mailing list!
Hitting reply on any message and changing the subject is not the correct
way of doing it.
--
#Joseph
On Sat, 2004-09-11 at 18:59, Raul Elizondo (wizardteam) wrote:
> Hi,
>
> My linuxbox has 2 eth's, one with pppoe for dsl, a
Nick,
I too battled a similar problem with my TDM400p. I solved it by putting the
following in the channel descriptions in zapata.conf:
stripmsd=0
Clearly this is not the default which I think should be obvious...
David
Nick Barnes said:
>
> Hi all,
>
> I've been batting my head against a bri
On Sat, 2004-09-11 at 15:40, Clif Jones wrote:
> I have a friend with a PRI coming into a modem bank that is receiving
> 56K modem calls and some ISDN data calls. He wants to dump his analog
> office phone lines and use some of the capacity on the PRI. I have been
> digging through the mail arc
Hi,
My linuxbox has 2 eth's, one with pppoe for dsl, and also i got an ip_gre
tunnel. At the time i run asterisk, even i got bindaddr=0.0.0.0, it does
not show any port open for sip (5060), if i change 0.0.0.0 for any ip, next
time i reload, it opens the specific ip, changing back to 0.0.0.0 and
We have purchased a couple of Uniden UIP300 phones (h.323 only). We love them all features work except hold and transfer. The hold function is local unless you define a DTMF hold sequence... Does Asterisk support this? I had heard that # Was transfer when I hit the pound key I get... " I am sorry t
I'm staring with Asterisk and want to setup it up (at the moment) as
simple answering machine different message will be plaid between 8am -
5pm and different after hours. I'll add extensions later on.
I've Supira 3000.
I was reading wiki page and to my understanding I have to place
device=/dev/
Hi all,
I've been batting my head against a brick wall for the best part of the day
and still haven't got any further (apart from getting a big headache, that
is). I've searched the Wiki and Googled the hours away, but I still can't
find supportive documentation.
I've just replaced my ISDN Fritz
Sirs/Ladies,
Not sure if anyone saw anything like that before...
I was playing with an Asterisk setup with a Grandstream BT101 (1.0.5.11)
and www.voipjet.com (IAX2).
The other devices I have home (Sipura 3k and DTA-310) seem to work just
fine, but the Grandstream seems to suffer from one-way voic
Thanks so much for the info. It's very appreciated.
Bill
- Original Message -
From: "David Cook" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Saturday, September 11, 2004 5:13 PM
Subject: [Asterisk-Users] Re: Distinctive Ring
> Quoting [EMAIL PROTECTED]:
>
> > If I have a Wildca
I created a small patch to make this happen. You enter an l for the
extension when you call AgentCallbackLogin. It sees that (ell) and
sets the extension to "" as if the person on the phone simply hit #.
This needs to be properly implemented as an AgentCallbackLogout()
application, but this is w
Steve,
Your problem appears fixed; I provisioned your account manually.
I'm not sure what caused the problem, but, I'll make sure that
the FWD support and Operations team is aware of the issue as it may
affect other FWD features.
Note: the best way to reach the FWD support team is via the
web for
Quoting [EMAIL PROTECTED]:
> If I have a Wildcard X100P and Asterisk, it is possible to make it
> answer only the distinctive ring call of two short rings and ignore
> the regular incoming ring?
>
> Bill Lohr
Absolutely.
[zapata.conf]
dring1=0,0,0
dring1context=distring1
dring2=326,0,0
dring2con
Same problem here
Steve Maroney a écrit :
Ahhh, I didn't fix my "The User you are calling is offline" problem. It
seemed that when I added Answer() as my first priority, my problem went
away. Well it didn't. The problem is inconsistient (spelling??). Sometimes
my asterisk server handles the call ok
I found the issue. I had linked /usr/src/linux-2.6 to /usr/src/linux
The correct link is to the linux-obj folder
cd /usr/src
ln -s linux-obj/i386/default linux-2.6
The answer was in
/usr/src/linux/README.SUSE
I am now able to compile successfully. Granted I don't have the hardware
yet, so I don'
I have a friend with a PRI coming into a modem bank that is receiving
56K modem calls and some ISDN data calls. He wants to dump his analog
office phone lines and use some of the capacity on the PRI. I have been
digging through the mail archives and Wiki site on this subject but the
informatio
Ahhh, I didn't fix my "The User you are calling is offline" problem. It
seemed that when I added Answer() as my first priority, my problem went
away. Well it didn't. The problem is inconsistient (spelling??). Sometimes
my asterisk server handles the call ok (my sip phones rings) or I get the
offli
I am trying to compile zapata under a 2.6 kernel (Suse 9.1 all patches
installed).
I am getting the error bellow:
Any ideas?
Anybody able to successfully compile this in Suse 9.1?
[EMAIL PROTECTED]:~/dl/pbx/zaptel-1.0-RC2> make linux26
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_Z
LoopStart or KewlStart pstn lines(at least here in the US) do not provide
answer supervision to the orginating caller, so for CDR purposes any call
placed this way is considered answered and logged as such.
Lyle
- Original Message -
From: "Jose Maria Guisasola" <[EMAIL PROTECTED]>
To: <[E
John Stegenga wrote:
[sarcasm on]
Thank you ALL for your warm welcome to this list. I posted this message
yesterday, and since I'm only getting Digest I figured I'd see a response in
a day...
[sarcasm off]
C'mon. This is the Asterisk Users mail list, isn't it? This is where the
Voip WIKI tells m
Hey guys,
I have most of my problem fixed. It was a really simple fix ... Answer().
But I have an excuse for overlooking that. When an incoming call came in
from FWD, the only priority i had was to dial one of my extensions. When
that happend, I heard a half of a ring.
I did enjoy having all thos
Oleg,
Thanks very much for your help. That fixed it.
Patrick
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Oleg A.
Arkhangelsky
Sent: Saturday, September 11, 2004 3:36 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Can't get ChanSpy to work
Hel
Brian
Take a look at sox (type man sox at the command prompt if sox it installed
for details on the options available). There is a "vol" argument that
allows you to adjust the gain. If this is what you need, you can call sox
after record (using the system command) to adjust the gain.
Bill Seddo
Hello to all:
When I make a call of extension (FXS) to extension (FXS) (TDM40B) status of the
call is the following one:
Case call answered
--- Log init
-- Starting simple switch on 'Zap/4-1'
-- Executing Macro("Zap/4-1", "stdexten|103|Za
Title: Help!
Hello all,
I am an MTech student and currently working on a project on building GSM air interface to fixed line. I am making use of Asterisk soft PBX. I am stuck at a point regarding this. As far as I understood from the available Asterisk documentation that Asterisk can e
Anybody know an easy way to adjust audio level of recordings made in
Asterisk (using the 'record' application)? I've noticed that recordings
using the "wav" format are about twice the level of those made using
"WAV" or "wav49". Unfortunately, the "wav" recordings are uncompressed
and about 10
Steve
Try re-registering the IAX option from your FWD account. I had exactly the
same problem until Thursday when, out of frustration I re-registered. When
I reported this resolution here in response to another list member's
question, Ed Guy, the FWD support person, responded to my post to me to
- Original Message -
>
> [sarcasm on]
> Thank you ALL for your warm welcome to this list. I posted this message
> yesterday, and since I'm only getting Digest I figured I'd see a response
in
> a day...
> [sarcasm off]
>
> C'mon. This is the Asterisk Users mail list, isn't it? This is w
Good evening all,
to complete test that I made in august (and which where discussed here)
about garbage when receiving fax, I went more down in my tests.
At first, my setup is an * CVS-HEAD-09/08/04, spandsp-0.0.1k, X100P
(Tiger Jet Network Model 300 128k), kernel 2.4.26 on an SMP SoftRaid
mach
> Im trying to get IAX to work between my * and FWD. I
> activated my iax2 account on iax.fwdnet.net and I get the output:
>
> "Registered to '65.39.205.121', who sees us as 68.14.203.254:4569"
>
> when I start asterisk. I tried used the Call Me tool on
> fwdnet.net but I dont get any calls ev
the user you are calling is currently offline
is what I get when calling fwd number
hth
hank
- Original Message -
From: "Steve Maroney" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Saturday, September 11, 2004 9:29 AM
Subject: [Asterisk-Users] FWD
Im trying to get IAX to work betwee
mjr,
Satellite links can be pretty tough to troubleshoot. It sounds like
you are running into a uplink buffer issue. On heavily loaded
uplinks, the input buffers can get quite large, and if the satellite
provider isn't using some form of buffer handling that prioritizes udp
traffic, it may be th
Im trying to get IAX to work between my * and FWD. I activated my iax2
account on iax.fwdnet.net and I get the output:
"Registered to '65.39.205.121', who sees us as 68.14.203.254:4569"
when I start asterisk. I tried used the Call Me tool on fwdnet.net but I
dont get any calls even though the Ca
Jan,
Quite easy, do something like this
zapata.conf:
context=inbound-e1
channel => 1
context=reject
channel => 2-30
extensions.conf
[inbound-e1]
[reject]
exten => i,1,Answer
exten => i,2,Playback(
exten => i,3,Hangup
On 08 Sep 2004 20:06:00 +0200, jan terje tønnessen <[EMAIL PROTECTED
Hi everybody,
I've got a problem concerning the DTMF detection. I have a ISDN phone
connected to a HFC-ISDN-card using the bri-stuff-drivers. I can make
calls via a SIP provider. In the sip.conf the DTMF mode is set to
RFC2833. However, when I make a call using the GSM codec, the DTMF
signals a
Thank you very much for your help, I think is working now
Erick
- Original Message -
From: "Darren Wiebe" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<[EMAIL PROTECTED]>
Sent: Friday, September 10, 2004 8:27 PM
Subject: Re: [Asterisk-Users] ASTCC Hel
If I have a Wildcard X100P and Asterisk, it is
possible to make it answer only the distinctive ring call of two short rings and
ignore the regular incoming ring?
Bill Lohr
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On Sat, 11 Sep 2004 10:01:27 -0400, John Stegenga <[EMAIL PROTECTED]> wrote:
> yesterday, and since I'm only getting Digest I figured I'd see a response in
> a day...
> [sarcasm off]
Some people may have a filter in their inbox that has "newbie" in it
going directly to trash. Just kidding, it's b
On Sat, 11 Sep 2004, John Stegenga wrote:
> [sarcasm on]
> Thank you ALL for your warm welcome to this list. I posted this message
> yesterday, and since I'm only getting Digest I figured I'd see a response in
> a day...
> [sarcasm off]
>
> C'mon. This is the Asterisk Users mail list, isn't it?
[sarcasm on]
Thank you ALL for your warm welcome to this list. I posted this message
yesterday, and since I'm only getting Digest I figured I'd see a response in
a day...
[sarcasm off]
C'mon. This is the Asterisk Users mail list, isn't it? This is where the
Voip WIKI tells me to go for informat
Hi everybody,
What´s necessary to do the voicemail work fine with Brazilian Portuguese?
Look my log files:
-- Playing 'vm-login' (language 'pt')
-- Playing 'vm-password' (language 'pt')
-- Playing 'vm-youhave' (language 'pt')
-- Playing 'digits/1F' (language 'pt')
-- Playing
John Morris <[EMAIL PROTECTED]> writes:
> Running FC1, ThinkPad T22, headset thru the soundcard. Asterisk is
> asterisk-1.0_RC1. No NAT. The phones I've tried so far are as follows.
I've got NetBSD-current on a Thinkpad X31, ear plugs connected to the
built-in sound card, using integral microp
[EMAIL PROTECTED] wrote:
> HI all.
>
> I was wondering... Isn't it a good thing to store the IP of the
> client making the call ?
>
> Or does asterisk store that in some other place ?
>
> /Mike
I'm gonna reply to myself here and add another quiestion.
When an incoming call is being transfere
I can part a call (dial #700 it is parked on 701)
but if I dial 701 I am told it is not a valid extension?
I have include => parkedcalls in my local
extension context. I have Ttr on all extensions and the incoming pots
line.
It parks, plays MOH but I can't retrieve
it.
--john
___
Hi!
On Sat, Sep 11, 2004 at 10:38:48AM +0200, Maurizio Marini wrote:
> my machine did hangup as growing logs fullfilled partition
hmm I see, mine is 8G, but it has gronwn from 0.5G to 2.0G
> it does apply to asterisk, not to zaphfc :(
> it was a misleading suggestion, so
> i solved it installin
What was the symptom of the sound problem?
Echo cancellation on the * box should make the call sound good for the standard * user
side I would think. Does the sound quality suffer on the iPaq side then?
I wonder if the encoding process on the iPaq is to blame for bad sound at the * side.
Hello,
Scottstuff.net has an example to "forward calls when
busy" to a nufone 800. My telco will allow me to do
this for a 5$/month charge. :( No luck duplicating
Scott's example, or playing with the
ChanIsAvail/cut/forward. Since everything can be done
with asterisk, can anyone slap me upside
Hello,
Current moment,
I've successfully put the incoming calls into Queues and dial to
an idle agents. When the agents answer the calls, the
agents can hear the pre-recorded message to incidate what's the service that the
call is calling.
But there one problem that I'm not able
On Saturday 11 September 2004 10:09, Hartmut Wahl wrote:
> Hello,
>
> i have exactly the same problem.
> > Aug 24 04:47:57 weblogin kernel: zaphfc: sync lost, pci performance too
> > low. you might have some cpu throtteling enabled.
>
> I am running the card in NT-Mode, it happens some hours after
Hello,
i have exactly the same problem.
> Hi,
> debian sid
> kernel 2.6.7
> cpu: AMD Duron(tm) Processor
same dist, same kernel, same CPU, also bri-stuff RC4a
> kernel.log:
> Aug 23 17:33:40 weblogin kernel: Zapata Telephony Interface Registered on major 196
> Aug 23 17:33:40 weblogin kernel: z
We've installed and used PPC X-Lite on an iPAQ with 802.11b. While the
sound quality of the iPAQ user was OK (not great but OK) the sound quality
as heard by the other caller was very poor.
If you try using a softphone on a PPC, I'd be interested to hear of your
experience.
Bill Seddon
-Ori
HI all.
I was wondering... Isn't it a good thing to store the IP of the client
making the call ?
Or does asterisk store that in some other place ?
/Mike
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I don't really see how that's possible with the current Queue setup. I
don't see why you couldn't use AGI or an app to query a callback table
and orginate the call back and connect it to an available agent.
I'm curious on this as well so feel free to contact me offlist. I'm
going to add it to 1 of
Hello Patrick,
Saturday, September 11, 2004, 4:03:45 AM, you wrote:
PJC> When I dial this extension from a SIP phone, and then make a call (which I
PJC> am trying to monitor) from a Zap channel, I get the following error:
PJC> Sep 10 20:03:35 WARNING[270359]: file.c:475 ast_openstream: File zap d
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