On Sat, 2004-05-01 at 23:57, Roger Gulbranson wrote:
> On Sat, 2004-05-01 at 23:28, Scott Weis wrote:
> > >
>
> > > 4. Incoming pstn calls that either go to IVR menues or VM do not properly
> > >sense disconnect supervision. Again, monitoring the pstn line via the
> > >LEDs on an analog ph
Here is the bug ID I posted: Add comments I guess
http://bugs.digium.com/bug_view_page.php?bug_id=0001522
- Original Message -
From: "Roger Gulbranson" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Cc: "Roger Gulbranson" <[EMAIL PROTECTED]>
Sent: Saturday, May 01, 2004 11:57 PM
Subject: Re:
On Sat, 2004-05-01 at 23:28, Scott Weis wrote:
> >
> > 4. Incoming pstn calls that either go to IVR menues or VM do not properly
> >sense disconnect supervision. Again, monitoring the pstn line via the
> >LEDs on an analog phone "does" indicate approximately .5 second of
> >no-battery
- Original Message -
From: "Brian Capouch" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Saturday, May 01, 2004 7:11 PM
Subject: Re: [Asterisk-Users] Grandstream Ringtones
> brian wrote:
> > Brian,
> > You forgot to mention the most important "hell is gonna freeze over"
> > feature
>
> Just installed the new 4-port FXO card and moved two pstn lines from
> existing x100p cards to ports on the FXO card. All zapata.conf entries
> that were functional on the x100p's were copied to the new FXO channels
> (including callprogress=no).
>
> Observations thus far:
> 1. asterisk will sp
Just installed the new 4-port FXO card and moved two pstn lines from
existing x100p cards to ports on the FXO card. All zapata.conf entries
that were functional on the x100p's were copied to the new FXO channels
(including callprogress=no).
Observations thus far:
1. asterisk will spontanously dec
All, Just thought I would post again in hopes of someone being able to
advise or give their thoughts on what may be causing this, basic problem
comes down to clicks while bridging SIP phone to SIP PSTN Gateway through
asterisk.
More detail, the problem below seems to happen when the two channels
On Sun, May 02, 2004 at 12:37:14AM +0100, Gavin Hamill spake thusly:
> On Sunday 02 May 2004 00:32, you wrote:
>
> > How about setting up a bounty?
> > http://voip-info.org/wiki-Asterisk+bounty
>
> If I had the money to rent-a-coder, would I have begged on a public mailing
> list?
You are missi
no "real" one, just do a google search with "site:lists.digium.com"
(without quotes) with your search query, and it will search only the
digium mailing lists.
DH
J Poz wrote:
I'm new to Asterisk and have been attempting various configurations.
I'm having problems with the basics of SIP to SIP
Think that has to do with how the ata does an attended transfer. It drops
and rings back which causes it to look like a new call to the exten.
Doesn't do that on my 7960.
bkw
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Juan J. S
I'm new to Asterisk and have been attempting various configurations. I'm having problems with the basics of SIP to SIP phone communications within my own network. I've configured two phones ( Xten X-Lite) and whenever I dial either one I get errors as follows:
*auto-congestion SIP/Phone 1
*SIP
On Sat, 2004-05-01 at 18:35, brian k. west wrote:
> Well to use the parking in asterisk you have to enable the 't' or 'T'
> depending on the call direction. But if you're like me that is such a hack.
>
> Check out www.bkw.org
>
> and http://www.bkw.org/app_valetparking.c
>
> Which is very flexa
On Sunday 02 May 2004 00:32, you wrote:
> How about setting up a bounty?
> http://voip-info.org/wiki-Asterisk+bounty
If I had the money to rent-a-coder, would I have begged on a public mailing
list?
Besides, I find the taste of all those X100Ps bought outside the USA not
having funded such a *
On Sat, 2004-05-01 at 16:42, Gavin Hamill wrote:
> Ah I've only noticed this thread has forked to CLID-in-non-Bellcore areas :)
> PLEASE PLEASE PLEASE PLEASE PLEASE PLEASE PLEASE can some Kind and Worthy soul
> spend a little time in getting this really important feature implemented? You
> would
On Sat, 2004-05-01 at 01:02, [EMAIL PROTECTED] wrote:
-SNIP-
> With a IAX
> trunk, I have already observed (at the house) serious
> call/voice deterioration due to channel overload. How does
> one stop this? I.e. it would be very desirable to specify
> channel capacity (say xx number of simultaneo
brian wrote:
Brian,
You forgot to mention the most important "hell is gonna freeze over"
feature added. iLBC
Yep, and it's working. The other asterisk clients with which I'm
interconnecting are reporting codec 1024 all right. Sound quality is
fine, and I'm working up a method of assess
On Fri, 30 Apr 2004, Mark Elkins wrote:
> Looking at pbx.c - I'm not sure if I should change the end time (ie
> midnight) to either 23:59 -or- 00:00.
it is 23:59
> 23:59 will work - but what happens to calls then between 23:59 and
> midnight?
23:59'59" is still 23:59 mainly because you are not h
Brian,
You forgot to mention the most important "hell is gonna freeze over"
feature added. iLBC
/me watches ice form now
bkw
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Brian Capouch
> Sent: Saturday, May 01, 2004 4:36
Good to hear that yours doesn't seg and it's the same code. From what I
could see in the BT it wasn't related to app_dbodbc in any way. Looked more
like a chan_sip bug or something.
bkw
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf O
On May 1, 2004, at 2:04 PM, Ryan Courtnage wrote:
I'm trying to get fax detection to work.
Hi Ryan, in stable FAX_DETECT is turned off by default in the code
(dsp.c).
I'm personally using it with spandsp and having no problems, but YMMV.
If you want to enable it, goto line 60 of dsp.c and uncom
On Fri, 2004-04-30 at 20:05, Mike Machado wrote:
> Is anyone out there using app_dbodbc
> (http://www.bkw.org/~brian/app_dbodbc.c)? Any problems with it?
I use it & works for me, no segfaults.
I checked the version of the file & it's identical to yours (although I
used a different URL: http://aste
yes
- Original Message -
From: "Steven Kalcevich" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Saturday, May 01, 2004 7:30 PM
Subject: [Asterisk-Users] Multiple music's on hold?
> Hey there,
>
> Is it possible to have multiple music on holds when you run asterisk?
>
>
> Steven kalce
Hey there,
Is it possible to have multiple music on holds when you run asterisk?
Steven kalcevich
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On Fri, 30 Apr 2004, Mark Elkins waxed:
> Playing with time ranges - using the examples found in one of the
> asterisk cook books... (pdf - page 17)
> ; After Hours
> include => night_menu|00:00-08:00|Tue-Fri|*|*
> include => night_menu|17:00-24:00|Mon-Thu|*|*
>
> this gives...
> ... pbx.c:2962 g
Duane is working on this, there are some problems with asterisk at the
moment in delivering this back to the 'caller' and the status ap is the
first step in delivering this awareness information.
Does anyone want to help duane and matthew write this as they both have
like a million things going on
On Fri, 30 Apr 2004, Dean Collins waxed:
> Ian, I'd love to see an example of this.
>
> Cheers,
> Dean
>
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Iain
> Stevenson
> Sent: Friday, 30 April 2004 1:47 AM
> To: [EMAIL PROTECTED]
> Subject: Re
okay, will use ${EXTEN}.
it all seems to be working now. I think my problem was
understanding the flow of control using contexts, but
i also needed to do some reading on syntax and variables
-- and more to come.
the working commands that we ended up using are:
[trunklocal]
exten => _9NX,1
The about-to-be-released Grandstream firmware now supports multiple
ringtones, but (so far) I haven't been able to unearth any documentation
as to how one uses them.
Anyone out there know anything about this? I've googled, read the
firmware READMEs and combed the GS site without any luck.
Thx
Well to use the parking in asterisk you have to enable the 't' or 'T'
depending on the call direction. But if you're like me that is such a hack.
Check out www.bkw.org
and http://www.bkw.org/app_valetparking.c
Which is very flexable parking that works with sip native transfers, blind
and attend
On Sat, 2004-05-01 at 15:30, Paul Tyreman wrote:
> Hi,
>
> I have just enabled call parking on my Asterisk system and it's working
> well.
>
> However, I am running some Grandstream phones on my system and when you
> press # on them to transfer the call, the user on the other end of the line
> he
> What is the equivalent decibel level of the RTP stream being transported
> through *. This may sound like a funny question but I will try to show what
> I mean by a diagram below:
>
> PSTN Receive level into FXO card Center point level after FXO or FXS
> Gain
> adjustment
>
> PS
On 30 Apr 2004, at 16:08, Gustavo García Bernardo wrote:
.
Skype - What do you know about it?
Unfortunately I don't know too much of its technical details - only
what I've seen on the Web (lost the address sorry) and the fact that
I've just been a user of Skype for quite a while and they seem to
Hi,
I have just enabled call parking on my Asterisk system and it's working
well.
However, I am running some Grandstream phones on my system and when you
press # on them to transfer the call, the user on the other end of the line
hears the tone produced by the phone, rather then Asterisk recognis
As you all know, one of the biggest criticisms of Asterisk has been the
lack of documentation.
Paul Mahler of Signate has taken the initiative and is writing an
introductory guide to Asterisk that Digium plans to help publish.
This is a guide for beginners, not for gurus. I would like to see the
Paul,
The reboot corrected the problem either due to you not stopping and
restarting * after making "certain" parameter changes, or, modules
(listed in /etc/modules.conf) were not actually running initially.
Several of the * config file changes are not actually implemented
unless you do a "stop n
Just FYI stop using BYEXTENSION because it will be going away soon.
use ${EXTEN} or ${EXTEN:x}
bkw
- Original Message -
From: "Tom Scott" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Saturday, May 01, 2004 12:29 PM
Subject: Re: [Asterisk-Users] dialing out to PSTN from SIP phones
thanks for responding.
the changed the include commands and they are now at least
causing the extension to match using one of the local
10-digit numbers. this is what shows up on the console:
Executing StripMSD("SIP/1008-32df", "1") in new stack
-- Executing Dial("SIP/1008-32df", "Zap/1|BYEXTENSIO
Hi All,
I'm using an X100P to connect to PSTN ( context=from-pstn ).
I'm trying to get fax detection to work.
Using the simplest dialplan, I cannot get * to detect fax tones:
[from-pstn]
exten => s,1,Answer
exten => fax,1,Goto(ext-fax,999,1)
The fax is never detected (ie: Goto never executed)
All
I have managed to get the two cards going eventually, I was sure everything
was set up right, so I rebooted and it all started working !!
The only problem I have now is that there is a bad echo on the line for a
while at the start of a call
Paul.
-Original Message-
From: [EMAIL PRO
Rich,
please send me a note pointing out the errors and we'll get them fixed.
Thanks!
/ed guy
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Rich Adamson
Sent: Saturday, May 01, 2004 12:01 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Using IAXTel
Dear All,
Is any way I can monitor digits pressed
by caller during active call ?i.e.
For calling card plateform normally
a caller callthroguh one channel and then after entering PINS etc he dial
to STD or IIDD via another channel(out-going). If during call he press
certain keys such ## or ** o
On Apr 30, 2004, at 9:20 PM, mitchel wrote:
The IP addresses assigned by my telco/isp to use for the serial (ppp)
interface is 67.153.163.202 with a gateway of 67.153.163.201 and a
netmask of 255.255.255.252. Thanks to Shido I have a nice firewall
script which routes all the network traffic beh
On Saturday 01 May 2004 03:50, Steve Underwood wrote:
> The compute load to simply monitor for energy appearing on the line is
> so low it hardly matters. What matters is someone actually bothers to
> implement it :-)
Ah I've only noticed this thread has forked to CLID-in-non-Bellcore areas :)
S
show dialplan will show the asterisk view of the dialplan.
show channels will display channels in use and
sip debug will show what the sip phones are doing.
Also, have a console open as this often provides clues, especially if
started with some verboseness -vv
You may try making a more ge
> I found a posting by J Todd where he gives BW utilization
> for various IAX2 codecs with trunking on. Now, the number of
> calls I can sustain over an IAX channel, obviously is going
> to be determined by the capacity and state of the physical
> pipe. Typically, here in the office I have a burst
On Fri, 30 Apr 2004, Jay Milk wrote:
> The term fee is in the current terms of service -- I'm pretty certain
> that it wasn't there two years ago, or I wouldn't have signed up. I
> don't "do" early termination fees unless I purchase a subsidized
> cell-phone or something. What are the odds of fi
> now. But if you have a look at this page ->
> http://www.freeworlddialup.com/advanced/iax you will find that you can now
> use FWD with IAX2 along with SIP :)
FWIW, I just moved our FWD account to iax2, and it works rather well
with *. The referenced web page does have a couple of configuration
>if too many of these services get up its just as bad a space
>as we were in before.
Agreed, since they have signed up Packet8 and Net2Phone they have a pretty
good head start as far as US VOIP providers. It will be interesting to see
if/what Vonage, VoicePulse, CallVantage (AT&T), BroadVoice,
Hi,
I would like to define a h323 user with serveral
ip address, like:
[roger1]
type=user
host=217.94.99.216,217.94.99.217,217.94.99.218
context=default
accountcode=schreiter
Ok, the above sample does not work.
Is it possible in any way to define a h323 user with
seveval (but not undefined) ip add
I installed Asterisk and a digium wildcard (X100P). Using
the extensions.conf with a few changes and a sip.conf file
that includes two extensions, I can place calls between the
SIP phones. I also can call in to the SIP phones from the
PSTN using the X100P. On incoming calls I can hear the
default d
Mark Elkins wrote:
I have * with i4l installed and working - on a dumb eicon card.
It seems in order to get DTMF out of the BRI (for business banking -
etc) - I should change from i4l drivers to capi drivers.
wiki help seems to be for the Fritz card only...???
I have ticked the suggested boxes i
Yes but no information about how this will operate, what regulation or
restrictions on joining, what connection protocols will be used etc etc
Cheers,
Dean
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Reid A.
Forrest
Sent: Saturday, 1 May 2004 8:21 PM
>From http://www.thevpf.com/
To join, please e-mail [EMAIL PROTECTED] or telephone 1-212-232-2020 (Mon-Fri
9AM-5PM EST).
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of jimfl
Sent: Saturday, May 01, 2004 5:11 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asteris
Hi,
had the same problem... and we wrote a patch.
This patch's are for speech_tools 1.2.3 and festival 1.4.3.
to use in the corresponding directory with:
#patch -p1 Can someone tell me how to build festival on a machine with gcc 3.3.2?
>
> I've searched all around and even found a reference o
>Jim/frank,
>Can you give us more information about how to access this enum? I've
>been to the stealth web site and there is no information about access.
>
>I look forward with interest to what you have up and running today for
>asterisk users to benefit from.
>
>Cheers,
>Dean
Sorry, I am not asso
Jim/frank,
Can you give us more information about how to access this enum? I've
been to the stealth web site and there is no information about access.
I look forward with interest to what you have up and running today for
asterisk users to benefit from.
Cheers,
Dean
-Original Message-
F
Stealth Communications Announces Registry to Avoid Access Fees
Posted on: 04/23/2004
Stealth Communications Inc. today announced the official launch of a registry that
allows service providers routing calls over the
Internet to avoid paying local phone companies access charges.
The VPF ENUM Reg
Hi
I have been playing around with asterisk for a few days now. I have asterisk
running with a single x100p card and a couple of x-lite "extensions". Here's
where I am at:
I can make calls between the extensions. Voice mail seems to work OK. I can
use the x100p card to dial out to the PSTN over t
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