Hey All,
Yeah yeah, bad form to reply to myself, but mrgoby on IRC helped me
out with the answer just as I sent me question. I'm following up for the
archives.
Looks like there is an option in the iax.conf file called
notransfer=yes. Seems to do the same thing as canreinvite=no does in
Reed
I had the same problem yesterday. I have a gentoo system here. I first used
gentoo's emerge to install festival 1.4.3-r1. This worked out that it needed
speech tools and downloaded that along with festival and built with no
problems. From the command line I could get festival working but I
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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
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-
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 associated
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 patch..
Hope this help. If so let it know.
Have fun !!!
On Sat, 2004-05-01 at 02:35, Reed Wade wrote:
Can someone tell
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:
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
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 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
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
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 (ATT), BroadVoice,
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
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 finding
snip
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
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
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 :)
So,
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
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 ** or
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
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
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|BYEXTENSION) in
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
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 now
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
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
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
PSTN
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
hears the
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
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 =
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:
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 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
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 Of
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 PM
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
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
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 simultaneous
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 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 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 flexable
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.
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
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 missing the
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
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
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
- 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 added. iLBC
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
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 23:57, 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 phone does
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