You could always run another T100P into a HylaFAX-run T1 fax modem.
That way you can use your T1 for faxing.
Could you explain a litter further? Thanks.
Well, you can do something like this:
T1 -- TE405P(1) -- Asterisk -- TE405P(2) -- Patton 2977 -- HylaFAX
So you've got a TE405P with 4 ports on
Keith O'Brien wrote:
Essentially its because * has been architected to send an rtp packet
after receiving a packet. If * never see's and incoming rtp
packet, then it won't send an rtp packet (which usually contains some
amount of audio). Thus choppy audio in one direction.
So why cant *
- Original Message -
From: Race Vanderdecken [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Wednesday, February 16, 2005 6:57 PM
Subject: RE: [Asterisk-Users] speech recognition V 2.0
Greetings David,
PerlBox
Alex Zarubin [EMAIL PROTECTED] wrote:
We are having a significant ( 1 sec) delay in a multi-asterisk conference,
with IAX2 legs
connecting meetme on different boxes.
All the other legs are PSTN (TE410P). The example configuration
Slave box 1 meetme --- IAX2 --- Master box meetme --- IAX2
Has anybody succeeded in getting IAXy to work with Verizons BroadBandAccess?
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Thanks for the suggestion. Changing the RTP Packet Size in the Sipura
to 40ms did improve the call quality slightly, but still well below
par compared to the Cisco 7960.
In my ethereal captures, I did notice something interesting. While
the RTP stream from the Cisco to asterisk seemed to have a
Hi there,
I am having a problem. It looks like this:
Feb 16 15:01:10 WARNING[11122]: chan_iax2.c:5546 socket_read: Call
rejected by XXX.XXX.XXX.XXX: No authority found
Feb 16 15:01:10 NOTICE[11122]: chan_iax2.c:1375 iax2_destroy: Avoiding
IAX destroy deadlock
-- Hungup 'IAX2/user/1'
Even I
Hi,
I need to make call recgnition with Asterisk (external calls). Which TDM
card i would can to buy for make this?
Thanks in advace
Pablo Fernandes
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Forgot to mention that when I set the RTP Packet Size to 20ms that the
difference was 160 (like the Cisco) but call quality was much worse.
On Wed, 16 Feb 2005 15:36:44 -0500, Pedro [EMAIL PROTECTED] wrote:
Thanks for the suggestion. Changing the RTP Packet Size in the Sipura
to 40ms did
On Wed, 16 Feb 2005 15:40:19 -0500, Sergey Kuznetsov
[EMAIL PROTECTED] wrote:
Hi there,
I am having a problem. It looks like this:
Feb 16 15:01:10 WARNING[11122]: chan_iax2.c:5546 socket_read: Call
rejected by XXX.XXX.XXX.XXX: No authority found
Is there any solution?
The log is telling
I have two BV accounts. For host= I used sip.broadvoice.com at first.
Then I changed it to the faster proxy. My sip.conf looks just like this
without all the extensions.
sip.conf
[general]
port=5060 ; Port to bind to
bindaddr=0.0.0.0 ; Address to bind SIP channel
I new to this as will. But add more info like your sip.conf file.
David
On Wed, 2005-02-16 at 18:04 +0300, Julius Kidubuka wrote:
Hi,
I have installed two X-Lite phones and theyre able to login
successfully. The two phones plus the Asterisk system are all on the
same LAN with private
I am having problems loading the zaphfc from bristuff and wcte11xp
drivers at the same time.
If I load zaphfc then all works fine.
If I then load wcte11xp, the card using the zaphfc doesn't pick up calls
anymore.
I am using bristuff 0.2.0-RC5.
Anyone else seen this problem, know of a fix, or can
Here is what I use for outbound calls.
exten = _1NXXNXX,1,SetVar(CALLFILENAME=${EXTEN:1}-${TIMESTAMP})
exten = _1NXXNXX,2,Monitor(wav,${CALLFILENAME},m)
exten = _1NXXNXX,3,Dial(${TRUNKL3}/${EXTEN})
David
On Wed, 2005-02-16 at 07:57 -0800, Jason Goecke wrote:
Hello,
I have been
not a permanent solution according to many on the list but try type=friend
in your iax.conf
- Original Message -
From: Sergey Kuznetsov [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, February 16, 2005 3:40
I have been playing with a Clipcom that is pretty cool.
- Original Message -
From: Olaf Klein [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Wednesday, February 16, 2005 2:03 PM
Subject: [Asterisk-Users] WLAN-Voip phones anyone?
Hello,
Does anyone here use any WLAN
Is there a
way to increase the volume for the voicemail? Whenever someone leaves a
message, the volume is so low its hard to hear.
-Dave
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I have the following configuration:
CLEC - T-1 - Asterisk - Adtran Channel Bank - (analog) - Nortel
Don't complain that it's ugly. I've already done plenty of that.
The CLEC manages their Adtran remotely and needs to be able to
continue to do so. I assume they use FDL to do the management. We
They are the same. That's what I've checked first.
Peter Bowyer wrote:
On Wed, 16 Feb 2005 15:40:19 -0500, Sergey Kuznetsov
[EMAIL PROTECTED] wrote:
Hi there,
I am having a problem. It looks like this:
Feb 16 15:01:10 WARNING[11122]: chan_iax2.c:5546 socket_read: Call
rejected
Can you add ez-ipupdate to the web interface and maybe PPPOE
configuration
http://ez-ipupdate.com/
Thanks, David
You ask why?? I run this from home..
On Wed, 2005-02-16 at 10:59 -0800, [EMAIL PROTECTED] wrote:
New features include Festival text to speech and a new
Web Conferencing GUI.
Thats what I already have.
Here is the entry:
[user]
type=friend
accountcode=XX
amaflags=billing
host=dynamic
secret=mostsecret
auth=md5,plaintext
context=iax_out
disallow=all
allow=gsm
allow=ulaw
allow=alaw
allow=adpcm
callerid="User" 416XXX
trunk=no
jitterbuffer=yes
dropcount=5
On Wed, 2005-02-16 at 14:13 -0600, Chris Wade wrote:
Keith O'Brien wrote:
Essentially its because * has been architected to send an rtp packet
after receiving a packet. If * never see's and incoming rtp
packet, then it won't send an rtp packet (which usually contains some
amount
I would eliminate everything that is not necessary,
like the amaflags and the auth=, account code stuff. I would also use IP
address rather than domain and get it working. Then I would start adding
the extras back in.
- Original Message -
From:
Sergey Kuznetsov
To:
On Tue, 2005-02-15 at 18:55 -0500, Brian M. Arlinghaus wrote:
I am using a T100P for a 23-channel voice T1. Is it possible to create an
extension that would allow sending a fax to HylaFax? Would I have the same
problems as faxing through a TDM card? Can HylaFax send faxes through the
On Wed, 16 Feb 2005, David Ishmael wrote:
Is there a way to increase the volume for the voicemail? Whenever someone
leaves a message, the volume is so low it's hard to hear.
This is a known bug - see bug number 2023:
http://bugs.digium.com/bug_view_advanced_page.php?bug_id=0002023
Peter
Hi all,
Just a quick reminder: If you're in Melbourne and want to talk
Asterisk or VOIP in general, tonight's the night. Come out come out!
It ought to be a fun evening. Details below:
-- Forwarded message --
From: jurgen [EMAIL PROTECTED]
Date: Thu, 10 Feb 2005 12:54:43 +1100
I've been trying to resolve some quality issues and I was hoping
someone might be able to provide some insight.
To give you an idea the calls are coming in via a SIP DID and sent out
via an IAX2 connection. Latency to both the SIP equipment and IAX
equipment are around 80ms with 0 packet loss
Are there any settings that I need to change on the phone to match this?
Thank you!
--
Jason A. Crome
Senior Software Engineer, DEVNET, Inc.
E-Mail: [EMAIL PROTECTED]
http://www.devnetinc.com
-Original Message-
From: [EMAIL PROTECTED]
In article [EMAIL PROTECTED],
Steven Critchfield [EMAIL PROTECTED] wrote:
Simple answer would be to just get a proper timing source. Barring
timing from a piece of hardware, asterisk falls back to triggering sends
because something was received.
I thought Asterisk *always* used the incoming
David Ishmael wrote:
Is there a way to increase the volume for the voicemail? Whenever
someone leaves a message, the volume is so low its hard to hear.
-Dave
___
Asterisk-Users
On Wed, 16 Feb 2005, Deti Fliegl wrote:
I tried to use Voicemail from a PRI interface but it didn't work because
pressing a key on the ISDN phone just caused Q931_IE_KEYPAD_FACILITY
messages which are normally handled by a bri-stuffed libpri.
Unfortunately a wrong if condition stops
Hi Everyone -
I just got my hands on a Cisco 7970 and was told that I should do a
factory reset before trying to configure it to work with Asterisk.
After the factory reset, it will not boot at all, instead sits with the
line button lights flashing one at a time in sequence.
I have had no luck
Your approach is really great, I love it,
questions:
do you have a roadmap ( sort of ) ?
which exiting features can we expect ?
when do you think you will reach 1.00 ?
success,
jl
On Wed, 16 Feb 2005 10:59:29 -0800 (PST), [EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
New features include
Be there and be square!
PaulH
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of jurgen
Sent: Thursday, 17 February 2005 8:54 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Melbourne Asterisk Users meet TONIGHT
Hi
FYI - Seems the latest firmware in conjunction with changing the
packet size to 10ms improved the call quality to usable. The Cisco
7960 is stell superior, but now at least the SPA-2100 is acceptable
(and with 2 working g729 channels including 3-way calling).
On Wed, 16 Feb 2005 15:44:58 -0500,
Hi all,
For the interested people you can download the new DIAX (0.9.10d) from the
following
location only:
http://www.laser.com/dante/diax/diax0910d.zip
What's new comparing with the official 0.9.10a:
' - faster language change inside application
' - full Eutectics USB phone support (including
thanks to all for the responce
the idea is to use a grunt system someone say david and asterisk tranfer to
me i dont care if some stupid cannt say david i will put the option to press
the number if no valid option is selected in the speech recognition
please if someone can say to me how to put
On Wed, 2005-02-16 at 21:59 +, Tony Mountifield wrote:
In article [EMAIL PROTECTED],
Steven Critchfield [EMAIL PROTECTED] wrote:
Simple answer would be to just get a proper timing source. Barring
timing from a piece of hardware, asterisk falls back to triggering sends
because
On Feb 16, 2005, at 3:07 PM, Eric Wieling wrote:
I have the following configuration:
CLEC - T-1 - Asterisk - Adtran Channel Bank - (analog) - Nortel
Don't complain that it's ugly. I've already done plenty of that.
The CLEC manages their Adtran remotely and needs to be able to
continue to do so.
I'm getting a lot of this too :-( my fax stuff worked great under 1.0
but after upgrading to 1.0.5 i've been broken..
Fax3Decode2D: Warning, (FakeInput): Premature EOL at scanline 496 (got
912, expected 1728).
Fax3Decode2D: (FakeInput): Bad code word at scanline 497 (x 470).
Fax3Decode2D:
Title: RE: [Asterisk-Users] Sipura g729 call quality to PSTN
Next thing I would check are the de-jitter buffers if possible on the Sipura, or jitter in general.
Do you have control of the PSTN gateway ? Measure the jitter on ingress to the gateway. You can do this crudely by using Ethereal
Essentially its because * has been architected to send an rtp packet
after receiving a packet. If * never see's and incoming rtp
packet, then it won't send an rtp packet (which usually contains some
amount of audio). Thus choppy audio in one direction.
So why cant * just play
You could always run another T100P into a HylaFAX-run T1 fax modem.
That way you can use your T1 for faxing.
Could you explain a litter further? Thanks.
Well, you can do something like this:
T1 -- TE405P(1) -- Asterisk -- TE405P(2) -- Patton 2977 -- HylaFAX
So you've got a
I am running asterisk 1.0.1 with OH323 compiled in.
We have a 323 trunk to CallManager with a mgcp controlled pri router.
When using sip phones (directly registered with asterisk) to call out
the 323 trunnk to PSTN, calls timeout after 3 rings. If I answer b4 3
rings - no problem, otherwise I
Is there a way to increase the volume for the voicemail? Whenever someone
leaves a message,
the volume is so low its hard to hear.
Add your comments to bug #2023.
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C F [EMAIL PROTECTED] wrote:
Use the latest stable or CVS HEAD and modify features.conf. You can
change it there.
FYI, only CVS HEAD (not stable) supports the new features.conf options.
--
Robert L Mathews, Tiger Technologieshttp://www.tigertech.net/
It is trying to download its firmware. You need to setup a TFTP Server. Also be aware that the 7970 only supports SCCP not SIP. Further, the * implementation of SCCP doesnt support the latest version of SCCP which is required for the 7970. I dont see how it would work at all with *.Hi
It's
more than that, from what I know a *missing*
RTP packet could be
'silence'
(vad) or it could be 'network related' (jitter). * not seeing
a packet
doesn't always mean it was vad, it might mean your network had
a split
second (subsecond) hiccup that caused the packet to disappear
Jerry wrote:
On Feb 16, 2005, at 3:07 PM, Eric Wieling wrote:
I have the following configuration:
CLEC - T-1 - Asterisk - Adtran Channel Bank - (analog) - Nortel
Don't complain that it's ugly. I've already done plenty of that.
The CLEC manages their Adtran remotely and needs to be able to
Title: Message
I have benn having
trouble with the Monitor Command.
Basically any time
that I send a call into a MeetMe room I am only able to monitor half of the
conversation.
File-in is recorded
with the incoming voice but file-out does NOT record
anything.
I have tried this
with both
Greetings Mr. Weber,
Remember the rule in mathematics that is much easier to solve for one
variable.
You stateed you are having a problem with the 1088 extension. If look
like you are trying to make a call from the 404 extension to the 1088
extension.
1.
If you have 6 ATA's running shut 5 of
Hmmm, that worked?
Interesting that you can change the sample size to 10ms since the standard
is 20ms that most people don't go below. I know you *can* do below 20 but if
you are doubt the technical ability of the box it seems strange they are
capable of that.
This seems to smack of bad
Are you both using Digium cards?
Do you know if you are using G3 (standard) or SuperG3 (like a modem) fax
machines?
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Justin Richards
Sent: Wednesday, February 16, 2005 4:25 PM
To:
Rich Adamson [EMAIL PROTECTED] wrote:
Any analog modem (fax or pc) is going to be limited to 9600 baud or
slower,
and will only achieve that speed if g711 is used through the entire path
(including asterisk). If a modem call comes in one T1 (or PRI) and goes
out another, asterisk is still
Use the manager API to send a call from the meetme room to an extension that
does Monitor for a specified period of time. That is how we do it in the
astGUIclient suite and it works great.
; extensions.conf entry:
; this is used for recording conference calls, the client app sends the
filename
;
Matt,
How do you stop the recording if it is set for a period of time? Eg if
set the period as 30 minutes and the call finishes early will it cease
recording or hold up the line for 30 mins
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of mattf
Sent:
On Wed, 16 Feb 2005 17:42:02 +0100 (CET), Peter Svensson
[EMAIL PROTECTED] wrote:
Asterisk clocks outgoing rtp data to a device from the incoming rtp
stream from the same device. This is a known limitation and there has been
some talk about implementing an internal clocking system.
In
http://www.srh.noaa.gov/fwd/productviewnation.php?pil=OKXZFPOKXversion=0
can anyone suggest how I could set up [EMAIL PROTECTED] to read
out allowed the following text when I dial extension 850?
815 PM EST WED FEB 16 2005.OVERNIGHT...MOSTLY CLEAR. LOWS 30 TO 35. NORTHWEST WINDS 15 TO
My sip.conf file;
[luke]
type=friend
host=dynamic
username=luke
secret=luke
;dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info
dtmfmode=rfc2833
mailbox=202 ; Mailbox for message waiting indicator
allow=all
context=sip
callerid=luke 2123
nat=yes
[mike]
type=friend
host=dynamic
But then how do we get past the problem of silence when there is only
one person in the room or recording a bunch of music when there is only
one person in the MeetMe room?
Also, just for clarification on the Channel: section below.
What is the break down of the value and what do they mean?
we don't have a reoad map. i thinks most of the
features that people want are in [EMAIL PROTECTED] now.
1.0 shuld be soon and it will be just bug fixes untill
then.
Any other features you want?
--- Jean-Louis curty [EMAIL PROTECTED] wrote:
Your approach is really great, I love it,
Is there a way to do an inplace upgrade from v.0.5 to v.0.6?
Thanks
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Wednesday, February 16, 2005 11:34 PM
To: Jean-Louis curty; Asterisk Users Mailing List -
On Thu, 2005-02-17 at 15:24, dean collins wrote:
http://www.srh.noaa.gov/fwd/productviewnation.php?pil=OKXZFPOKXversion=0
can anyone suggest how I could set up [EMAIL PROTECTED] to read out
allowed the following text when I dial extension 850?
815 PM EST WED FEB 16 2005
I've installed a TDM400. Having a go with AMP.
I would like incoming calls to be put throuhg to an extension (at my desk)
and a mobile (cell phone) at the same time. Whichever picks up, gets the
call..
This should be possible with AMP (call groups, 200|201|0*0408xx), but it
didn't work, so
Is there anyway to destroy a sip channel ? I get hung up channels like this
in sip show channels:
67.153.9.20 2145558260 33ae28f6088 00102/0 unknow
67.153.9.20 2145558260 52be8085005 00102/0 unknow
67.153.9.20 2145558260 4653d937578 00102/0 unknow
67.153.9.20
I've been using my voicepulse connect number for over
a month now, but today it simply won't connect. My
partner and I each have a number, both are mapped in
my iax.conf and extensions.conf files. This has been
working fine.
Today, either number gives this message:
Feb 16 21:53:14 NOTICE[4330]:
Hi all,
Does anyone know of a method of sending a raw G711 stream to
an address in Asterisk.
For example, an application that takes a argument of a phone
and a port.
The reason? I have found a method to paging on Zultys ZIP2
and ZIP4x4 handsets. Basically it involves sending a
I downloaded asterisk to use cvs to checkout the release version.
After installing, I would like to load module chan_h323.so but there is some
error :
*CLI load chan_h323.so
Feb 17 15:22:38 WARNING[2865]: loader.c:258 ast_load_resource:
/usr/lib/asterisk/m
odules/chan_h323.so: undefined symbol:
When I load asterisk there are following messages
on the concole:
== Creating H.323
Endpoint
== Adding alias "gated" to endpoint
== Adding Prefix "1981" to endpoint
== H.323 listener
started
Error registration with gatekeeper
"X.X.X.X".
Feb 17 11:10:23 ERROR{[1650]: chan_h323.c:2118
In our client apps, we track the call placed and Hangup that call when
conference is over. All you need to do is either have a function that hangs
up those recording channels if they are the only one in the conference(perl
script running periodically parsing Show Channels) Or you could link a
You are using illegal characters in your file name.
See this line in your output?
ast_writefile: No such format 'wav|rec_to_448704386865_at_16022005-16'
It can't get past it because the colon is not a valid filename
character.
[EMAIL PROTECTED] wrote:
Hello,
I have been attempting to
Thanks Steven,
You are correct in assuming that I meant a PRI. I thought perhaps Asterisk
would receive without problems. I should have done a bit more research. I
think I saw a script somewhere that will print the fax after it is received.
So, I guess you answered my question. And since
I was doing some testing and it seems to be related to
my extensions.conf.
I have a #include extensions_from_mysql.conf that
was working fine yesterday:
[voicepulse_connect_context2]
exten = s,1,Answer
exten = s,2,NoOp,${CALLERID}
#include extensions_from_mysql.conf
and
Any analog modem (fax or pc) is going to be limited to 9600 baud or
slower,
and will only achieve that speed if g711 is used through the entire path
(including asterisk). If a modem call comes in one T1 (or PRI) and goes
out another, asterisk is still handling the pcm packets. The packets
don't
On Thu, 2005-02-17 at 15:51, [EMAIL PROTECTED] wrote:
I've installed a TDM400. Having a go with AMP.
I would like incoming calls to be put throuhg to an extension (at my desk)
and a mobile (cell phone) at the same time. Whichever picks up, gets the
call..
This should be possible with AMP
Hello,
There is no colon the filename below.
Jason
--- Jim Van Meggelen [EMAIL PROTECTED] wrote:
You are using illegal characters in your file name.
See this line in your output?
ast_writefile: No such format
'wav|rec_to_448704386865_at_16022005-16'
It can't get past it because the
OK, forget I said that. Wrong side of my brain.
Still, it is funny that it truncates the filename at the colon.
These lines are suspicious:
CLIast_writefile: No such format
'wav|rec_to_448704386865_at_16022005-16'
Where'd the :56:35 on the end go? Also, why is is trying to set the
format to
[EMAIL PROTECTED] wrote:
Hello,
There is no colon the filename below.
Exactly. But there *is* (or rather *was*) in the filename you told it
you wanted to write.
Where'd it go?
--- Jim Van Meggelen [EMAIL PROTECTED] wrote:
You are using illegal characters in your file name.
See this
On Thu, 17 Feb 2005, Jim Van Meggelen wrote:
You are using illegal characters in your file name.
See this line in your output?
ast_writefile: No such format 'wav|rec_to_448704386865_at_16022005-16'
It can't get past it because the colon is not a valid filename
character.
In what way
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