.so.preload is also missing on
the FC4 laptop which works, so I concluded that that was not the problem.)
Any help much appreciated.
Regards
Roger
[EMAIL PROTECTED] sbin]$ sudo strace ./asterisk
execve("./asterisk", ["./asterisk"], [/* 27 vars */]) = 0
uname({sys
Stefan Reuter schrieb:
...
So having a look at Asterisk 1.2-beta2 is probably the way to go.
Great! Yes, this will solve my problem.
Let's upgrade ...
Thanks!
Roger.
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-
.
Now I'm looking for mean, to get the ringing event (of the outgoing
channel) related to the incoming channel name.
Roger.
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
dot? Is it possible, that
the outgoing channels gets there a higher number than
the one from the incoming incremented by 1? May e.g. another
incoming channel at almost exact the same time get that
number instead?
Thanks for some hints about that!
Roger
simple way are welcome!
Does anybody think about a new state in ${DIALSTATUS}:
FAST_HANGUP (for a hangup before ringing)?
Would anybody else appreciate a dialplan
variable ${RINGTIME} like DIALEDTIME and ANSWEREDTIME,
but indicating, when we received the ringing-notification?
Roger
em.
Essentially I'd like the call to ring through immediately and not
perform this one ring click until the call is routed to the correct
line. Has anyone else seen this and is there a way to fix the problem?
Please let me know if I can provide any additi
answers.
Roger.
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http
0 msec.
Is there an explanation for this behaviour?
Thanks for any hints!
Roger.
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-us
As far as I see, there are more users faxing without observing
quality differences to ISDN than users with problems with fax
over VoIP.
This is, what various partners of ours do report after having
replaced BRI connections by VoIP in some small and middle sized
com
just impossible (without T.38).
Roger.
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options
erage
load on our machines. Interconnections to telcos are costy,
and thus some more off peak traffic would help.
So we have to check, whether it is possible, which solutions
are suitable, and what each solution will cost.
Unfortunately our budget is currently very limited.
der.
His setups are success stories.
Thus, I assume, IAX and the jitter won't be the problem, when
the internet connectivity is good enough.
Well, we'll proceed our tests and see.
Roger.
___
--Bandwidth and Colocation sponsored by Easynew
he IAXy, a Sipura ATA and the FritzboxFon. The latter
is one of the best ATAs I ever saw. All those three enable
faxing under the condition, the ping round trip to the gateway
to PSTN is not too long. Below 20ms should be reasonable.
The quality and reliability of this setup is excellen
asterisk just started to support T.38 pass through,
and probably will do it reliably in some weeks. Maybe this will
enable T.38 support for asterisk - though unfortunately not for free nor
open source.
Roger.
___
--Bandwidth and Colocation sponsored by
both parties, who are already speaking,
because they get interrupted.
In the cdr I can find the call with 120 secs duration, 0 secs billsec.
Has anyone had a similar problem so far?
Or any ideas?
Roger.
___
--Bandwidth and Colocation sponsored by Easynew
debug this?
How can I force Zap to data mode. The d option seem to be
something different.
Did anybody try sending SMS to german Vodafone or other
SMSC mentioned in the smsclient package?
Thanks for hints!
Roger.
___
--Bandwidth and Colocation sponsored by
faxed with a
"Sample.Call" file.(For lack of a better method thus far)
I don't understand, what you are telling or asking us
with this information.
Has it something to do with your question? If not, please
avoid confusing with additional infos which are
haven't yet understood the whole souce, so I don't know,
whether my solution was silly or not or whether my solution
is reliable or not.
For me it is working great, but use with care!
Roger.
My proposal for a fax-send-success-feedback:
1. Look in txfax.c (approx line 60) for th
understanding the content)?
Please tell me, if you have knowledges or experiences on this
topic!
Othervice, and if I won't find further reliable information saying
it cannot work, I'll try it. And of course I will report the results
later her
via a digital line is complicate,
and I shouldn't complain, but I would like to know, whether
these are typical values or whether one could increase
the max fax number by any means? Maybe force to a slower,
but more error proof modulation?
Regards,
Roger.
P.S.
After faxing approx 100 faxes,
hints!
Roger.
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digiu
the called phone is busy)
Congestion (which I use, if our gateway or one of the used carriers
are busy)
Thanks for any hints, if there further means!
Roger.
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
Asterisk-Us
your experience!
Roger.
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
sterisk is just stdin and stdout and a little
parsing. So don't worry too much whether your preferred programming
language is suitable for AGI programming!
Roger.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.dig
On Thu, 2005-08-18 at 09:41 -0600, Damon Estep wrote:
> I am using realtime mysql for extensions, sip, and voicemail.
>
> Outbound call routing does not really perform well in realtime
> extensions due to the high number of rows in the database (300k), so I
> can not use it. It appears with my lim
or a reliable way to determine, whether TxFax did send
a fax completely. I also tried the option "debug", but never saw
such a trace.
Which version of spandsp are you using?
Roger.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
h
B)
v
RxFax (Box A)
TxFax and RxFax ran on Box A. The PSTN call was accepted
at Box B and then forwarded via IAX2 to Box A.
RxFax and TxFax did nothing, and were never terminated, and
thus needed an expicit "Hangup" command.
Rega
e, is, when the other party hangs
up. Any other case called the next priority in the dialplan.
Is there any reliable mean, to check, whether a fax is really sent
successfully and complete?
Thanks for any hints!
Roger.
___
Asterisk-Users mailing list
Ast
enable asterisk to fax to itsself?
Well, it won't be the normal operation, but when allowing clients
to fax, it can happen by chance, that someone faxes to another
user on the same machine without knowing it.
Thanks for any hints!
Roger.
___
Ast
solution, maybe a foreign
service provider doing it, or a working (asterisk independent)
software?
Thanks for any hints!
Roger.
P.S.
Currently I'm trying to understand, what ionidea's T.38 software is
already able to do, but I'm
Hi,
asterisk compiled fine and is running very stable on
our dual opteron in 64 bit mode.
When loading G.729 library we have to peload libz manually for any
reason, but besides that minor issue, everthing is fine.
We didn't yet test the limits of that machine.
omeone providing GSM-A stacks for asterisk.
For the second option, it might be interesting for you,
that we are currently also working on asterisk support
for a GSM-modem.
Roger.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.co
Response: Success
CALLERID: (null)
Any hints?
Roger.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailma
hears the busy sound by
his network operator.
Are there any files like busy-tone.gsm with german (aka french
or dutch) busy sounds?
Roger.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-
Marco Parmeggiani wrote:
> ...
i had no problems receiving faxes with version 3.7.2.
on the other hand i have big problems in sending multipage faxes. only
Hi,
where did you get that version?
On libtiff.org, 3.6.1 is the most recent one.
Ro
k download page. Maybe it'll help
someone.
It works fine together with the other asterisk stuff (around version
1.0.7) located in that directory:
http://planinternet.net/download/voip/asterisk
Roger.
___
Asterisk-Users mailing list
Asterisk-Users@l
btiff 3.5.7: Pay attention to
my subsequent email!
Roger.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/l
2
cable?
Thanks for any hints!
Roger.
P.S.
Somewhere I read the advice, that I should connect the (analogue) audio
connector to the PC's soundcard, which is supported by miax. But
transferring the GSM voice data analogious and digitizing again
afterwords is not
report errors (wrong line lengths) and finaly display
a picture, which has nothing to do with the original.
Did anyone encounter the same problem? Is it a version
problem of libtiff? (libtiff-3.6.1 seems to be the most
recent one, besides CVS.)
Thanks for any hints!
Roger
nts!
Roger.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
canreinvite=no
Thanks for any hints!
Roger.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo
On Wed, 2005-05-04 at 17:22 -0600, Rich Adamson wrote:
> > > > > TDM & X100P card users:
> >
> > > > I get average numbers very close to 1.024 (especially if I take some
> > > > rounding error into account).
> > >
> > > That's a very good point. Now I'm not sure since the only thing I've
> > > go
On Wed, 2005-05-04 at 06:48 -0600, Rich Adamson wrote:
> > On Tue, 2005-05-03 at 09:48 -0600, Rich Adamson wrote:
> > > TDM & X100P card users:
> > I get average numbers very close to 1.024 (especially if I take some
> > rounding error into account).
>
> That's a very good point. Now I'm not sure
On Tue, 2005-05-03 at 09:48 -0600, Rich Adamson wrote:
> TDM & X100P card users:
>
> Attached is a modified zaptel/zttest.c app called "attest-mod.c". It
> has been modified to report the "delay" in receiving 8,192 bytes
> from the TDM card (instead of reporting a percentage). It works with
> the
Bill Ford wrote:
Since all the asterisk program needs to do is send mail through smtp,
and since using sendmail for this purpose is a bit like using Jeff
Gordon's racing engine on a bicycle we opted to scrap sendmail and use
msmtp. This is basically just an smtp engine. To our mail server, it
looks
Altus Snyman wrote:
Good day all
I have a Fedora core 3 installation
Is there any hassles with asterisk?
Thanks
Altus
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or
See subject:
Does [EMAIL PROTECTED] support Dual-Processor installations? I didn't see
anything on the sourceforge page clarifying that. I suppose they could
leave out the SMP version of the Linux kernel to save space on the .iso?
I had trouble some time ago installing version .5 of [EMAIL PRO
On Mon, 2005-03-21 at 19:03 +1100, Adam Goryachev wrote:
> In case you need it, there are X servers available for MS Windows
> platforms as well. Used to be one called exceed, but that was about 10
> years ago, I just use linux on my desktop now instead :)
CygWin (http://www.cygwin.com/) has X se
On Mon, 2005-03-21 at 08:57 -0700, Tom wrote:
> We don't want to have to spend an extra 3 grand for another
> server just to take up more space when we have this box that is sitting here
> idle 99% of the time, and as it has worked spectacularly well with the wctdm
> cards, I don't see why it can'
Whenever I try to install [EMAIL PROTECTED], I get this error at about 43%
There was an error installing rpmdb-redhat-3.4-0.20050105. This
can indicate media failure, lack of disk space, and/or hardware
problems. This is a fatal error and your install will be aborted.
Please verify your media a
- Original Message -
From: "Stuart Ford" <[EMAIL PROTECTED]>
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
Sent: Thursday, March 10, 2005 8:02 AM
Subject: RE: [Asterisk-Users] Re: Grandstream Message button
>Indeed - the bug was fixed in 1.0.5.18 (Nov 14 2004)
accord
Original Message -
From: "dean collins" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>
Sent: Friday, February 18, 2005 6:04 PM
Subject: [Asterisk-Users] [EMAIL PROTECTED] festival weather report
This script was developed by Mark Johnson.
All I did (
My broadvoice works perfectly. I am using a standard registration
string, however. Not the funky one broadvoice says to use. I can make
outbound and receive inbound calls over broadvoice.
I'm using AMP also.
register=phonenumber:[EMAIL PROTECTED]
sip.conf:
[952XX]
username=952XX
type
- Original Message -
From: "Dinesh" <[EMAIL PROTECTED]>
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
Sent: Tuesday, March 08, 2005 4:51 AM
Subject: RE: [Asterisk-Users] BroadVoice configuration changes for
Outbound
Hello roger,
Y
- Original Message -
From: "Dinesh" <[EMAIL PROTECTED]>
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
Sent: Tuesday, March 08, 2005 3:39 AM
Subject: RE: [Asterisk-Users] BroadVoice configuration changes for
Outbound
Hello all,
I am sorry for posting again, but when a
- Original Message -
From: "Gabriel Gunderson" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>; "Asterisk Users Mailing List - Non-Commercial
Discussion"
Sent: Monday, February 28, 2005 4:49 PM
Subject: Re: [Asterisk-Users] Dial out through Broadvoice
Am i not providing some helpfull info?
see bottom
- Original Message -
From: "John Millican" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Monday, February 28, 2005 10:21 AM
Subject: Re: [Asterisk-Users] Dial out through Broadvoice
On Saturday February 26 2005 4:45 pm, John Millican
- Original Message -
From: <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Friday, February 25, 2005 9:20 AM
Subject: Re: [Asterisk-Users] Asterisk With Broadvoice
Great! It works now!! Thanks so much.
What was it that made it work? Share the
ove type=peer from [Broadvoice] in sip.conf incoming calls
work great but outgoing calls don't work. If i leave type=peer in
there, outgoing calls work great but incoming calls get routed to
Broadvoice's Voicemail . . .
Roger Hanson wrote:
- Original Message - From: <[EM
- Original Message -
From: <[EMAIL PROTECTED]>
To:
Sent: Thursday, February 24, 2005 10:12 PM
Subject: [Asterisk-Users] Asterisk With Broadvoice
I have configured asterisk with the AMP php configuration utility. I
am able to make outgoing calls through broadvoice but incoming calls
a
ify hereby the advances of my SS7 interest and the
status reports you mentioned?
Roger.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options vis
Hi,
it is the same hardware, but with a firmware by Brian F. G. Bidulock.
It has nothing to do with the libisup project, Steve Underwood wrote
several times within this mailing list and soon will be made public
as SS7 support for asterisk with that Digium card.
Roger
Dragos Ungureanu schrieb:
> ...
> The redirection itself it's working but nothing is written in the CDR
Hi,
include
amaFlags=billing
and maybe
accountCode=AN_APPROPRIATE_ACCOUNTNAME
into your oh323.conf!
Roger.
___
Asterisk-Users m
lpful having a list of all the known codes
somewhere in the internet, but found only very partial lists.
Now, I started a list on my own. If you think it may help you,
please use it:
http://voip-providers.org
Roger.
___
Asterisk-Users mailing list
Asterisk-
05 10:20 AM
Subject: Re: [Asterisk-Users] [EMAIL PROTECTED] .05 release questions on
setup.
Roger Hanson wrote:
I've downloaded 2x and burned 2 cds and get an error "invalid
compressed format (err=2) system halted" message both times.
It'd be nice to have a MD5 to verify my do
;Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Saturday, February 12, 2005 4:50 PM
Subject: Re: [Asterisk-Users] [EMAIL PROTECTED] .05 release questions on
setup.
Roger Hanson wrote:
I've downloaded 2x and burned 2 cds and get an error "invalid
compressed format (
I've downloaded 2x and burned 2 cds and get an error "invalid compressed
format (err=2) system halted" message both times.
It'd be nice to have a MD5 to verify my download is OK. It'd narrow
down the problem to either the download or the burn, wouldn't it?
- Original Message -
From:
Michael Manousos schrieb:
...
Use Asterisk-1.0.3 with asterisk-oh323-0.6.5.
Hi,
may I ask, whether that combination runs really stable
at your machine?
I have now those versions installed.
I have asterisk crashes at least once every hour, when
several simultanious calls take place.
Roger
No problems here - works fine.
- Original Message -
From: "Joseph" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Wednesday, February 09, 2005 8:17 AM
Subject: [Asterisk-Users] IAX <=> FWD down again?
Can anybody confirm if IAX on FWD is down ag
Hi
I am trying to get app_pppd to make an outgoing call to my ISP.
Has anybody got this to work yet?
Thanks
Roger
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or
ry it right now.
Thanks!
Roger.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Hi,
which is currently a stable combination of asterisk and
asterisk-oh?
The combination of asterisk-1.0.3 and asterisk-oh-0.7.0 is
not stable at all and crashes approx once the hour when
having approx 3 simultanious calls.
Thanks for telling me your experience!
Roger
- Original Message -
From: "Siju George" <[EMAIL PROTECTED]>
To:
Sent: Monday, February 07, 2005 11:34 PM
Subject: [Asterisk-Users] Best OS for Asterisk--newbie!!!
Hi all,
Could some one tell me which OS is best suited for installing Asterisk
at present??
I had planned to install it on
the client and sends an error message. (Though in the log
files those "Too late messages" are mentioned.)
Does anyone know a solution for the below shown scenery?
Thanks for any hints!
Roger.
By the way:
Registration succeeds without any problems.
Following is the invite scenery:
Client s
Username Refresh State
sip.broadvoice.com:5060 952nnn 15 Registered
asterisk*CLI>
I hope that helps.
Roger
BTW, I use AMP as well. I have a DID set up for my broadvoice number to
ring my digital assistant.
___
Asterisk-Us
y
HostUsername Refresh State
sip.broadvoice.com:5060 952nnn 15 Registered
asterisk*CLI>
I hope that helps.
Roger
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.c
asterisk gateways without problems.
Thanks,
Roger.
full-log:
Feb 2 21:06:49 DEBUG[30130]: Auto destroying call '[EMAIL PROTECTED]'
Feb 2 21:06:50 DEBUG[30130]: Setting NAT on RTP to 4
Feb 2 21:06:50 NOTICE[30130]: Unable to create/find channel
Feb 2 21:06:50 DEBUG[30130]: Stopping retra
e any difference, but I have no more idea where to
look.
Thanks for any hints, how to get more informations about that error!
Roger.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRI
On Tue, 2005-02-01 at 16:26 +0100, Robert Rozman wrote:
> Hi,
>
> I'm looking for adressbook that could easily integrate into Asterisk, so it
> should:
> - have agi script to search for caller id name
> - have fields for notes on previous contacts (for CRM spawning of FOP)
> - have web interface t
from wrapper_misc.hxx:35,
from wrapper_misc.cxx:34:
Hi,
I had a similar problems when using a too old compiler.
Use gcc>=3.3
By the way: use asterisk-oh-0.7.x!
Regards,
Roger.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digi
id", $new_callerid);
Hope, that helps!
Roger.
P.S.
Read the docs about your ISDN channel driver, e.g. chan_capi!
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE
_callerid = $input{'callerid'};
... (some code to manipulte the callerid)
$AGI->set_callerid($new_callerid);
^^^^
Roger.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mai
asily done with an AGI-script and
maybe a (mySQL) database.
Alternatively look for a provider able to accept H.323 and
passing callerids unchanged to PSTN. Of course that provider
will bind you to some regulations in order not to abuse
this feature to spoof c
;Janus-patched" version from
http://www.inaccessnetworks.com
and applied patch mentioned in the asterisk-oh323-0.7.0-README
file.
Hope that helps!
Roger.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/l
Thus setting this value in sipfriends to 0 lets this var
to 0
=> callerid is send to the remote sip end.
Roger.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or up
When call comes from PSTN, those values are displayed in the
outgoing SIP header (and finally the called sip phone).
When call comes from SIP, there is still "unknown" displayed
in the outgoing SIP header (and nothing in the called sip phone).
How can I force chan_sip to use the values from
channel variable.
Both, ${CALLERID} and ${CALLERIDNAME} do contain just
the number.
I wonder, if the mysql-authentication overwrites
the calleridname.
Is there any mean, to get the SIP "real name", which
is presented in the "from"-header as shown above?
Thanks for any hints!
Roger.
__
Sorry, I don't know what happened to my line returns. Here's that
message readable.
On January 24, 2005 08:38 am, Roger Hanson wrote:
I did see the wiki items: "asterisk auto-dial out deliver message"
and
"Asterisk Auto-dial out" and think I may be able to mudd
Carlos Chavez schrieb:
I want to buy a new server to run Asterisk and after looking at prices
for the Athlon XP 3000+ it costs the same as an Athlon 64 at the same speed
rating. I was wondering if Zaptel/Asterisk will compile/work on an Athlon 64?
...
Yes, it will compile and work.
Roger
On January 24, 2005 08:38 am, Roger Hanson wrote:
I did see the wiki items: "asterisk auto-dial out deliver message"
and
"Asterisk Auto-dial out" and think I may be able to muddle my way
through getting that working (although that may be questionable) but
is
it feasable to
On January 24, 2005 08:38 am, Roger Hanson wrote:
I did see the wiki items: "asterisk auto-dial out deliver message"
and
"Asterisk Auto-dial out" and think I may be able to muddle my way
through getting that working (although that may be questionable) but
is
it feasable to
sip.c:7321 handle_request: Nothing
to pick up
-- SIP/collegue-92e5 is ringing
while the other phone kept ringing.
I'm using asterisk-1.0.3
What went wrong? Thanks for any hints!
Roger.
___
Asterisk-Users mailing list
Asterisk-Users@lists.
e asterisk tarball... edit the file, move it to
/var/spool/asterisk/outgoing and it'll dial and connect de callee with
the extension of your choice...
Greetings
On Mon, 24 Jan 2005 02:57:13 -0600, Roger Hanson <[EMAIL PROTECTED]>
wrote:
I'm trying to get a script working on a websit
I'm trying to get a script working on a website to send out automatic
email reminders to customers reminding them monthly to change furnace
filters. I haven't got one running successfully, yet.
That made me think - could it be done with a phone call using Asterisk?
A monthly automated phone ca
I'm using CentOS - which is another Red Hat Enterprise clone, like WBEL
www.centos.org
I've had no problems of any kind with the OS
- Original Message -
From: "Imran Sadiq" <[EMAIL PROTECTED]>
To:
Sent: Tuesday, January 11, 2005 9:58 PM
Subject: [Asterisk-Users] What is the best and easie
ard. I assume the line is total
potential free, so it would not interfere.
Are there any experiences?
Roger.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or upd
code only for foreign calls.
Now I'm looking for any mean to decide, whether the
received callerid begins with a country code and thus
comes from another country or is domestic.
Is there maybe any variable indicating this?
Roger.
___
Asterisk-Users ma
Is the meeting still on for Saturday 1/8/05?
11:30am at 2375 University Av W STE120, Saint Paul.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit
On Wed, 2005-01-05 at 15:52, Mike Dent wrote:
> That sounds like it might just be the ticket Roger.
> I like the web page idea too.
> Would you be willing to share it please?
I've attached the agi script.
My web site is written in Mason which probably doesn't interest many
I saw the post on the wiki a last month stating the meeting was this
Saturday. Is that confirmed? Still on for 1/8?
Roger
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To
201 - 300 of 451 matches
Mail list logo
