I've had a closer listen to 400*17 through the handpiece rather than just on
speaker phone, and I get the feeling that the Australian ringing tone must
have been tweaked slightly, perhaps with the introduction of the newer
Ericsson AXE exchanges?
400*17 sounds familiar, perhaps the older exchanges
I am also getting this warning. But i noticed that this happens only when i
issue a shell command from the Asterisk CLI. But it has never affected any
functionality. My linux box is a very slow one.
Regards...
Girish
From: Frankie Gravato <[EMAIL PROTECTED]>
Reply-To: [EMAIL PROTECTED]
To: Ast
I've been beating my head for 5 hours to figure out why my asterisk
server or sipura isn't passing my voice over to the caller. It seems i
can hear the caller but they can't hear me it seems either the
asterisk or the sipura isn't passing this information.
Here's my setup specs
aster
Christopher Lee wrote:
Hi Steve,
Interesting... I'm not sure! My copy of the original indications.conf had
400+17, and looking at the wiki it's the same there also
http://www.voip-info.org/tiki-index.php?page=Asterisk%20indications%20defaul
t
I tested 400*17 and it made a difference, but I still
I have seen bluetooth do nothing but grow. More and more devices are
getting bluetooth the headsets can be had for 64 or so at buy.com
bkw
On Sun, 25 Jan 2004, Steve Underwood wrote:
> Hi Don,
>
> A large number of GSM phones and PDAs now have bluetooth. It looks
> likely that through 2004 the m
Hi Steve,
Interesting... I'm not sure! My copy of the original indications.conf had
400+17, and looking at the wiki it's the same there also
http://www.voip-info.org/tiki-index.php?page=Asterisk%20indications%20defaul
t
I tested 400*17 and it made a difference, but I still think 420+400 sounds
mu
Hello list,
I've been experiencing choppy sound as well.
The version on Asterisk I was using originally was dated 10/24/03 (I
think), the problem appeared after I updated from that version.
My setup is a little different though. I'm having choppy sound only on
some incoming calls -- from PSTN->P
On Sun, 25 Jan 2004, Christopher Lee wrote:
> The original indications has 400+17/400, but I find that sounds more like
> two beeps (which could possibly be confused with the Australian
> congestion/busy tones).
Shouldn't it be 400*17?
Steve
___
Ast
For the benefit of anyone with the same questions or
searching the archives, I’ve solved my problem to the below.
The Cisco 7940 (and other SIP devices) generate their own
indication tones of ring etc., I found by placing an Answer before a dial, then
Asterisk will answer the call and b
Hi Don,
A large number of GSM phones and PDAs now have bluetooth. It looks
likely that through 2004 the majority of GSM phones anywhere above entry
level will have Bluetooth. My guess is that this will collapse in 2005,
and bluetooth will be dead soon after. In the meantime, I don't seem
many
We have one we are going to start selling in Feb.
Currently we use it with asterisk, an Excel running ADS and Max TNT voip MVAM
gatekeeper.
- Original Message -
From:
Don Feuer
To: [EMAIL PROTECTED]
Sent: Saturday, January 24, 2004 11:37
PM
Subject: [Asterisk-Use
Has anyone out there
written a good web based convergent billing system out there that they want to
sell??
The system needs to
work with both voip as well as TDM switches for provisioning and
presentment.
If anyone out there
knows of a good one that works along with anyone with one ple
Does anyone out there have any suggestions on a decent telecommunications
web based billing system for billing and provisioning multiple
applications??
Has anyone written one that they want to sell??
Please e-mail me at [EMAIL PROTECTED]
Thanks!
Don
-Original Message-
From: [EMAIL PRO
Hello,
I am seeing this error:
Jan 24 20:02:53 WARNING[1175660480]: Ring/Off-hook in strange state 6 on
channel 3
on one of my phone lines that comes into my channel bank. It only happens
on one line, and it still seems to work. Also, I have had two different
channel banks installed (ADIT 600 &
I have seen a number of phones being made by companies in Korea, but do not
know much about what has happened to them. I have seen a host of cellular
to Bluetooth phones at COMDEX two years ago, and I am a strong proponent of
Bluetooth.
It would be good if someone is from Korea or can read Korea
On Friday 23 January 2004 12:18, Paul Mahler wrote:
> On Friday, January 23, 2004 at 8:04 AM, Steven Critchfield wrote:
> > On Fri, 2004-01-23 at 09:30, Darren Martz wrote:
> > > I have 8 lines coming into an existing PBX system and am looking
> > > for a cost
> > > effective way to replace the exi
Try this: make your outbound call via a Local channel, and see if
that gets logged.
Channel: Local/[EMAIL PROTECTED]
MaxRetries: 0
RetryTime: 60
WaitTime: 60
Context: home
Extension: 10
Priority: 1
and then...
[callout]
exten => _X.,1,Dial(Zap/1/${EXTEN})
exten => _X.,2,Congestion
exten => _X.,1
"Chris Wilson" <[EMAIL PROTECTED]> wrote:
>Has anyone had this problem:
>
>(When calling to ext. 1010)
>
>Jan 24 10:50:27 WARNING[-1252262992]: file.c:446 ast_openstream: File digits/" does
>not exist in any format
>Jan 24 10:50:27 WARNING[-1252262992]: file.c:734 ast_streamfile: Unable to open
"Chris Wilson" <[EMAIL PROTECTED]> wrote:
>Hey,
>
>I'm getting an odd message in my logs, and have'nt been able to find much information
>on it:
>
>Jan 24 00:22:39 WARNING[-1137431632]: chan_sip.c:486 retrans_pkt: Maximum retries
>exceeded on call [EMAIL PROTECTED] for seqno 102 (Request)
Just
btw, i think i've found an (ugly) work-around. by dumping both callers
into a meetme conference, and only giving one of them the ability to drop
out with #, the other is able to press pound w/out exiting... as long as they
remember who's who, it should be possible to have one caller use # to call
o
Hi!
> > I've concluded that the Netgear router (FVS318) performing the NAT is
> > corrupting the outgoing RTP packets. Traces confirmed that the BudgeTone
> > is sending them out with a UDP checksum of 0 but the next hop after the
> > Netgear router they are set to a non-zero value (an incorrect
I think this is related to a device (GS in my case) that has an sip
entry but you physically removed it and switched it off. Somehow * still
thinks connected. Comment out the entry and reload or put the device back.
Mark Rizzo wrote:
I have seen similar error which coincided with my GS phone ta
hi philipp (et al). thanks for the suggestion - but i
can't seem to get it to work.
i am able to use a local channel to pass arguments to a dial application.
for example, this successfully enables transfer on one leg of the call:
[incoming]
exten => s,1, Answer
exten => s,2, Dial(Local/[EMAIL PRO
"Jess Magnaye" <[EMAIL PROTECTED]> wrote:
>Go for inter-fone products. it can both support sip and h323.
I took a look at their site, and some of the products look quite
interesting. The prices don't seem too bad either, if I am
interpreting them right. It is hard to tell if the prices include
Frankie,
Thanks for your response, and BKW too.
I am not 'thrashing' anybody here. This is my experience and I have seen
people posting their experiences (good, bad) with many other voip providers
on this list.
well, I subscribed to NuFone because I've seen your kind of postings on this
list. Ma
I don't think Sathya is ripping Nufone per-se,
just trying to figure out what is going on. I'm
sure you would be doing the same thing IF you did
not get a reply, and did not know where to reach
him. Maybe Sathya does NOT know about the
chatroom. Lighten up.
And I agree, if there is a problem, let
I want to hear about problems with VOIP vendors. Sweeping them under the
rug isn't going to help. If its a valid problem please post it.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Frankie
Gravato
Sent: Saturday, January 24, 2004 11:37 AM
To: Sathya
Hello All,
I am experiencing some intermittent problems with calls coming inbound on my DID
trunk. I have 12 DIDs that come into an Adtran 750. From there T-1 to a port on
T400P. The problem is that some calls that come in don't seem to bridge properly.
Heres what happens.
Call comes in on T
On Sat, 24 Jan 2004, WipeOut wrote:
> Happy Birthday Greg..
>
> Have a good one.. :)
Hehehe.. Within 20 minutes, I received a simple patch from [EMAIL PROTECTED]
that will allow the next release to build on Fedora Core 1 ;)
> Greg Boehnlein wrote:
>
> >Hello all,
> > It's my birthday tod
Happy Birthday Greg..
Have a good one.. :)
Greg Boehnlein wrote:
Hello all,
It's my birthday today, so as my present I would like everyone
possible to download and test my updated set of RPMS for Asterisk 0.7.1.
By popular request, I installed and built a set of RPMS for RedHat 9.0,
and in t
Hello Asterisk-List Folks
Jan 24 14:39:27 WARNING[1082805040]: asterisk.c:255 listener: Select retured error:
Interrupted system call
Jan 24 14:39:27 WARNING[1082805040]: asterisk.c:255 listener: Select retured error:
Interrupted system call
== Detected 4 licensed G.729 transcoders
Jan 24 14:3
Hello Sathya,
Saturday, January 24, 2004, 12:25:43 PM, you wrote:
S> Folks,
S> I've ordered a new account from Nufone last month. Transferred money to
S> Nufone through their paypal account. I had communication with Nufone sales
S> up until two weeks back. Since then there were no replies to my
Hello all,
It's my birthday today, so as my present I would like everyone
possible to download and test my updated set of RPMS for Asterisk 0.7.1.
By popular request, I installed and built a set of RPMS for RedHat 9.0,
and in the process fixed a bunch of issues from the initial build. I
This is similar to the last version and applies against the current cvs.
cd asterisk
patch -p0 < Parking.patch
Then the double has transfer should be back.
Iain
--On Friday, January 16, 2004 6:10 pm -0500 mattf <[EMAIL PROTECTED]>
wrote:
Hello,
I was using the doublehash.patch that Iain
Hi
I don´t know if this is CTP compatible but it uses Bluetooth:
http://www.olympia-it.de/cdp.htm (Sorry german only)
Sascha
> -Ursprüngliche Nachricht-
> Von: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] Im Auftrag von Senad Jordanovic
> Gesendet: Samstag, 24. Januar
Has anyone had this problem:
(When calling to ext. 1010)
Jan 24 10:50:27 WARNING[-1252262992]: file.c:446
ast_openstream: File digits/" does not exist in any formatJan 24 10:50:27
WARNING[-1252262992]: file.c:734 ast_streamfile: Unable to open digits/" (format
ULAW): No such file or dire
Hi,
Citeren Kannaiyan Natesan <[EMAIL PROTECTED]>:
> > BT broadband voice uses ATA-186s configured as MGCP devices.
>
> I think asterisk supports MGCP. I want to configure MGCP with asterisk to
> connect to my BT Broadband Voice.
> Do you have any idea relating to that.
Here's my view. AFAIK as
Also on a side note... I have noticed emails from Jeremy back to me
getting caught by spamassassin because of his ip addresses on his dynamic
dsl connection.
bkw
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/a
Have you tried to call them? Your emails could have been caught up in a
spam filer or such I use nufone daily for our 888 service. I talk to
Jermey daily. So I dont know what your beef is but your rant has no place
on this mailing list if you are having problems and have spent any time
tryin
It's NexTrieve for Linux, I wrote it in C. I don't know what a wintel box is
but it sounds windows-like. That won't work for it.
- Kim Hendrikse
> What technology is it written using? I have a wintel box set up (Linux
> coming soon) and would be pleased to host it.
>
> Regards
>
> Mike
> --
Hm, had that enabled since i set everything up. I
tried with nat=no as well, same problem.
Welp, I guess if anyone figgers it out i'd
appreciate any help that comes my way :).
Thanks!
Chris
- Original Message -
From:
Kannaiyan
Natesan
To: [EMAIL PROTECTED]
Sent
Folks,
I've ordered a new account from Nufone last month. Transferred money to
Nufone through their paypal account. I had communication with Nufone sales
up until two weeks back. Since then there were no replies to my emails.
I am afraid with this kind of unresponsiveness how one would run a reli
Linus Surguy wrote:
>> IRC channel chatter says that there are some new developments with a
>> cool presence trick that Mark has come up with for bluetooth devices.
>> I know a bit about it, but I think the general population here would
>> like to see some details if they're available.
>
> I don't
Exten h isn't needed at all to record CDR info. Also exten h won't run if
you park the call.
bkw
On Sat, 24 Jan 2004, Girish Gopinath wrote:
> Hi friends,
>
> I have the entry exten => h,Hangup in my extensions.conf, and I am trying to
> record the call details for billing. From the wiki i foun
> IRC channel chatter says that there are some new developments with a
> cool presence trick that Mark has come up with for bluetooth devices.
> I know a bit about it, but I think the general population here would
> like to see some details if they're available.
I don't know if this is what you ar
Hi,
Wanted to test Asterisk in safemode using safe_asterisk. Tried to add
the command to the bootup sequence and tried it via the command
line. but Asterisk refuses to start up in that mode (it died with
code 127)
[EMAIL PROTECTED] /usr/sbin/safe_asterisk
[EMAIL PROTECTED] Asterisk e
IRC channel chatter says that there are some new developments with a
cool presence trick that Mark has come up with for bluetooth devices.
I know a bit about it, but I think the general population here would
like to see some details if they're available.
Mark - care to give the list a rundown o
I have seen similar error which coincided
with my GS phone taking a call-waiting call while I was on the GS phone. I got
two of the errors (101 102 I think) and then the GS phone or Asterisk
terminated the call I was on (including the call-waiting call that was trying
to get through).
I notice the same issue, I think the 'sidetone' is so low that you are able
to hear faint far-end echo. If the sidetone was louder you would never hear
the far-end echo.
I am new to Grand Stream, has anyone directly asked them to help fix/support
this?
Mark
Perpetual Entertainment, INC.
-Or
Hi,
We have termination based on IAX and SIP at Brazil.
Daniel
[EMAIL PROTECTED] wrote:
Hi,
I am looking for voip
termination all over the world especially based on IAX or SIP.
Regards.
___
Asterisk-Users mailing list
Here's an example - placing a call to 271536 from local extension 10. The
call file is:
Channel: Zap/1/271536
MaxRetries: 0
RetryTime: 60
WaitTime: 60
Context: home
Extension: 10
Priority: 1
... and the cdr record generated by * on completion of the call is:
"","","10","home","","Zap/1-1","SIP/
If you are considering such a service, you need to develop a more
thorough understanding of VoIP protocols and methods for load
distribution. To echo what Stephen Critchfield said to someone else
just a few hours ago: it's not simple, and you'll probably need a
consultant. After you've spent
Hi,Key Aavoja,
Have you successfully registed to * with secret specificated?
Regards.
bfrac
- Original Message -
From: "Key Aavoja" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Friday, January 23, 2004 2:00 AM
Subject: [Asterisk-Users] SIP register/auth with Grandstream BudgeTon
Hi,
I am looking for voip
termination all over the world especially based on IAX or SIP.
Regards.
Of course unstable has 0.7.1 already in the main disribution.
On Fri, 2004-01-23 at 13:48, FastJack wrote:
> hi everybody...
>
> have you checked the asterisk backports from www.backports.org? I'm
> currently building my asterisk system and i think i will use these debs as
> I've successfully use
Zot O'Connor wrote:
I am getting (with older image):
RFC3389 support incomplete. Turn off on client if possible
How do I turn that off for the DG104s? Or if I can't how do I tweak
asterisk?
I see posts about ATA-186's having an audiomode, but the closet I came
to was inbanddtmf. I tried =0 an
Zot O'Connor wrote:
I am installing a used DG104S
I got it to ring from gnophone, but all I got was fast busies. so I
upgraded based on Pavel's link:
ftp://ftp.dlink.ru/pub/VoIP/DG-104S/Firmware/MGCPDG104.zip
So I now have:
PROM Version: 3.0B22-DRUNTIME Version: 3.0B44-D
But when
dkwok <[EMAIL PROTECTED]> wrote:
>For people who are using GS 101, what do you think the sidetone
>generated by the phone.
Seems fine on the two we have.
>I find mind a bit annoying. It has a delay and you notice it as an echo.
>The volume of the sidetone is also quite hight. I am distracted w
Hi,
Yep, I got the latest firmware (and the next-to-latest, and the
next-to-next-to-latest, and one earlier yet) for SIP. The first three
(firmware versions 1228, 1227, and 1226) all have that password protected
"Advanced Configuration" page. The fourth one I found (version ) is a
bit more o
Well, I like the features asterisk gives me such as voicemail and IVR,
Prompts, etc. I would like to offer an IP Centrex like service, but don't
believe that I can handle very large amounts of users on a box. The reason
I believe this is that the box would be doing all the media processing/DSP
wo
What technology is it written using? I have a wintel box set up (Linux
coming soon) and would be pleased to host it.
Regards
Mike
- Original Message -
From: "Girish Gopinath" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Sunday, January 25, 2004 12:07 AM
Subject: RE: [Asterisk-User
Hi,
Very good! I tried some of the features and they are really good, especially
the search within selected months. I have been reading this list for the
last 4 months. Answers for some(all) of my doubts are there in the mails
posted within these months. Normally when googling, I try different
Has any one seen or heard of the lastest developments fo the Farfon IAX
phone? the web site
Thanks
Robb
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
Title: RE: [Asterisk-Users] 8 lines - best approach
Vegastream is a good choice. And it is
tested to work with Primus' SIP platform. Only $2212 with 10
FXO.
David
- Original Message -
From:
Kostur,
Andre
To: '[EMAIL PROTECTED]'
Sent: Saturday, January 24, 2004
Hi,
I've placed a demo search engine of the asterisk users archive here:
http://asterisk.nextrieve.com/cgi-bin/asterisk
I know there are a number of other ways to search this list that have been
suggested and one person suggested that another wasn't necessary but this
engine will do some thi
John Todd said:
>
> Time to dump the Netgear router. That's an unacceptable answer for a
> router vendor to say "Oh, well, for this MAJOR protocol we're going
> to simply corrupt those packets so they're unusable." What!?
>
> JT
> __
OR get an older on
Hi Miklos,
listas iPfone wrote:
> Hi ! thanks for the answer..
>
> I use rh9...
Sorry, I am familiar with Linux From Scratch, Debian and Gentoo but not
with RH.
>
> > I think with an interrupt problem, any startup will fail, may it be
> > manual or automatic during startup.
>
> but.. you thin
hi all
I have a strange problem that started right after an upgrade from
1.0.3.81: Every now and then the display flashes 484 when the phone is
idle, on hook. Early Dial is disabled, and I don't understand anything.
Everything works fine apart from this annoying flashing...
Anyone that knows w
Thanks to Dave and Matt for your help. I now have Asterisk running on my
Mandrake 9.1 box - time to learn how to configure it :)
Regards
Mike
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To
Thanks Dave, I will give that a try!
Regards
Mike
- Original Message -
From: "Dave Cotton" <[EMAIL PROTECTED]>
To: "Asterisk List" <[EMAIL PROTECTED]>
Sent: Saturday, January 24, 2004 7:11 PM
Subject: Re: [Asterisk-Users] Problem installing Asterisk with Mandrake 9.1
> On Sat, 2004-01-
Hi,
Anybody has clue of what protocol is using in MSN 6.1? SIP?
Thanks
Regards,
Soragan
You are having Cisco 7960G behind NAT.
Try with nat=yes
I'm not sure any other settings will solve that in
asterisk.
I have tried but no luck.
Kannaiyan
- Original Message -
From:
Chris Wilson
To: [EMAIL PROTECTED]
Sent: Saturday, January 24, 2004 8:26
A
Yes, i know that there are many ISDN card on the market.
But when i spend money for ISDN card, i prefer to be Digiums, to get all
support and help Asterisk :).
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/as
Why don't we make RFC3389 support complete.
Is there is any progress around on that?
Kannaiyan
- Original Message -
From: "Brian West" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Saturday, January 24, 2004 1:49 AM
Subject: Re: [Asterisk-Users] RFC3389 support issue with DG104S
>
Hey,
I'm getting an odd message in my logs, and have'nt
been able to find much information on it:
Jan 24 00:22:39 WARNING[-1137431632]:
chan_sip.c:486 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED]
for seqno 102 (Request)
I'm running asterisk with a Cisco 7960G
If an
Kannaiyan Natesan wrote:
If you are not users from mysql database then you can disable in the
makefile.
For this,
USE_MYSQL_FRIENDS=1
change it to
USE_MYSQL_FRIENDS=0
You won't get that error.
Alternatively you can install mysqlclient library to compile it without
errors.
Kannaiyan
What i
On Sat, 2004-01-24 at 08:25, Mike Nash [Tall Emu] wrote:
> make: bison: Command not found
> make: *** [ast_expr.c] Error 127
IIRC Mandrake 9.1 does not have Bison it has Bison++ or something, I had
the same problems and took a real Bison rpm from an earlier release.
--
Dave Cotton <[EMAIL PRO
Title: RE: [Asterisk-Users] 8 lines - best approach
Hey neighbour! I'll be posting on here what sort of experience we'll have with 8 (actually 10) incoming FXO lines going to a Vegastream gateway...
> -Original Message-
> From: Darren Martz [mailto:[EMAIL PROTECTED]]
> Sent: Friday,
Thanks Matt,
Thanks for the very quick response. The reason I had thought it was an
error is because of the
"> > cc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g-c -o
"
lines, and my subsequent inability to compile Asterisk. Thanks for the
heads up on that, I know what to che
--- dkwok <[EMAIL PROTECTED]> wrote:
> Chris Albertson wrote:
>
> |What firmware version do you have?
>
> program version 1.0.4.39
I've got the same firmware version. So it appears that sidetone
volume is not dependent on the firmware version.
>
> --
> David Kwok
>
> Iaxtel/FWD # 170018
--- dkwok <[EMAIL PROTECTED]> wrote:
> Chris Albertson wrote:
>
> |What firmware version do you have?
>
> program version 1.0.4.39
I've got the same firmware version. So it appears that sidetone
volume is not dependent on the firmware version.
>
> --
> David Kwok
>
> Iaxtel/FWD # 170018
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