Can you give any help on debugging AGI. The CLI just shows the following:
May 8 13:23:54 VERBOSE[18102]: -- Executing
EAGI(IAX2/[EMAIL PROTECTED]/1, cid_rewrite.php) in new stack
May 8 13:23:54 VERBOSE[18102]: -- Launched AGI Script
/var/lib/asterisk/agi-bin/cid_rewrite.php
May 8 13:23:54
When I run the script from the command line, I get the following error:
[EMAIL PROTECTED] agi-bin]# ./cid_rewrite.php
br /
bParse error/b: parse error, expecting `T_OLD_FUNCTION' or
`T_FUNCTION' or `T_VAR' or `'}'' in
b/var/lib/asterisk/agi-bin/astlib_jm.php/b on line b73/bbr /
br /
bFatal
Can anybody maybe help me a little here. I made the changes below...
but I don't think I did them exactly correct. When I get a fax now, it
is named:
_var_spool_asterisk_fax_1114530379.5.tif.pdf
I think the problem is with the second line.
$p .= cont Content-Type: $type;\n name= .
Jim,
Thanks for sharing this. I am currently using cidlookup.agi written by
James Golovich. http://asterisk.gnuinter.net/
However the problem I have with that script and probably this one also
is that my provider sends the number as +16105551212 so I need a way
to strip out the leading two
Yes and yes.
On Apr 9, 2005 6:03 PM, Bellows, Jared [EMAIL PROTECTED] wrote:
I'm looking into the TelIAX pay-as-you-go plan. I'm assuming that they
charge incoming calls minutes as well? Is there the $0.02 connection fee for
the incoming call as well?
Thanks,
Jared
A free solution would be to use YAC in conjunction with netcat. A
guide is on the wiki.
On Sun, 20 Mar 2005 15:42:07 -0600, Anton Krall
[EMAIL PROTECTED] wrote:
OK, the outbound problem is fixed... Now, my other question is, anybody
using identapop for popup CID on your screen?
The FWD - Vonage interconnect has been down for some time now. Vonage
claimed there was a secuity issue and pulled the plug. No word when/if
it will ever be working again.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of cmisip
Sent: Domingo, 20 de
Found this info on their website:
http://www.livevoip.com/index.php?subject=2content=networkStatus
LiveVoip Operations Staff
DTMF - Ringback Issues
Currently, Asterisk is using the timing of the input stream to
reproduce the output stream. This means that when no RTP streams are
being
You get this when you lose registration. Try qualify=100 or
qualify=yes, to see if that alleviates the problem.
I can make and receive calls for about 30 seconds before this happens.
On Mon, 14 Mar 2005 15:26:19 -0500, Randy Johnson
[EMAIL PROTECTED] wrote:
I have my broadvoice asterisk
. Too bad you must have a regular account first :(
On Thu, 17 Feb 2005 19:03:39 -0500, Brian Dingman [EMAIL PROTECTED] wrote:
I can't seem to dial out with Voicepulse Open Access service using *.
Incoming works fine. Another user posted a few weeks back that they
were having problems
No. But I would be interested in seeing how well it handles fax
detection over SIP/IAX.
On Mon, 14 Mar 2005 11:56:20 -0700, Joseph [EMAIL PROTECTED] wrote:
Has anybody tried NVFaxDetect Fax detection for sip SIP/IAX channel?
There is a new application from Newman Telecom for fax detection.
Did you ever get arounnd this issue? I am seeing the same thing,
On Sun, 13 Mar 2005 00:04:54 +0545, Vicky Shrestha
[EMAIL PROTECTED] wrote:
Thanks,
I have that already in my /etc/hosts
But it's still not working :(
On Saturday 12 March 2005 03:48, Rich Adamson wrote:
For everyone
I thought this patch was added into the 1.04 and later source code?
On Mon, 14 Mar 2005 07:24:44 +0200, Dimitris Kounalakis
[EMAIL PROTECTED] wrote:
I never managed to make outgoing calls to broadvoice without the
following patch to the file channels/chan_sip.c
it comes from
I doubt that was the problem. I would be interested in hearing what
else you did besides that to get it working.
On Sat, 12 Mar 2005 17:46:58 -0500, Jay Carter [EMAIL PROTECTED] wrote:
... I just tried again after removing my hosts file entry (again) and
outbound is now working! I had taken it
You have to wait till you get an email from them saying your account
is setup. I had the same problem where my DID was setup before my
outgoing account.
On Sat, 19 Feb 2005 13:16:15 -0800, Ed Greenberg [EMAIL PROTECTED] wrote:
I have several DIDs (working well) with LiveVoip and I just signed
Chris,
Did you ever get this working?
On Sat, 15 Jan 2005 03:18:01 -0500, Chris Wallace
[EMAIL PROTECTED] wrote:
I have researched my issue a little more and this is what I have come up
with. Here a examples of my configurations so far and the error I get when
I try to dial an external
I can't seem to dial out with Voicepulse Open Access service using *.
Incoming works fine. Another user posted a few weeks back that they
were having problems and there are some threads at dslreports.com
about this as well. Maybe someone here can figure out what the issue
is from the sip debug
Sam thing here. Waiting 10+ business days for my DID. Can't get
through to them by phone and email responses take days.
These guys are worthless.
On Tue, 15 Feb 2005 10:40:44 -0500, Mark Eissler [EMAIL PROTECTED] wrote:
On Feb 15, 2005, at 10:26 AM, BJ Weschke wrote:
I've had the same
I hate to beat a dead horse, but it looks like their toll free
offerings are completely gone. If you try and assign a random 800
number to your account, it says they are out of stock.
I dont't think any of us we being seeing a number from them any time
soon. Their IVR for customer support is a
Wow. I posted that a long time ago. Thanks. Festival doesn't seem very
stable to me though.
On Tue, 15 Feb 2005 15:14:47 +1100, Rod Bacon
[EMAIL PROTECTED] wrote:
SIOD ERROR: wrong type of argument to car : wholeutt
Try changing your festival.scm to the following:
(Note the extra () on
I am running * 1.0.5 and have been having lots of problems with
outgoing calls and their sound quality. I am using ULAW for the codec
and sixtel for termination. Basically the problem is that portions of
the call seem to be lost and replaced with silence. Sometimes I can't
hear the person talking
Silence Suppression is off or set to no of the SPA. I changed
jitterbuffer=no and things seem better. Will need to do some more
testing.
On Wed, 09 Feb 2005 13:03:06 -0500, Andres [EMAIL PROTECTED] wrote:
Just as further info, I am using a SPA-2000 to connect to * with G711u
as the
Checkout http://www.voip-info.org/wiki-NVBackgroundDetect
I haven't had a chance to try it yet, but supposedly it works on SIP,
ZAP, and IAX.
On Tue, 8 Feb 2005 21:26:28 +1100, Mike Sander
[EMAIL PROTECTED] wrote:
That's all very well, but what do you do if you only have SIP extensions and
Was does your sip.conf look like for this Sipura?
On Tue, 8 Feb 2005 22:56:15 -0500 (EST), Paul Dugas
[EMAIL PROTECTED] wrote:
Running * CVS with a SPA-841 (0.9.5) and can't seem to get the message
waiting light to come on automatically. There is a control in the web
interface to turn it on
Just for others edification. The problem here was that I was not
performing an Answer before issuing the DIAL command to the forwarded
number. Once I did that the calls natively bridged and left my system
completely.
On Sun, 6 Feb 2005 15:31:43 -0500, Brian Dingman [EMAIL PROTECTED] wrote:
I am
I am having the same problems. No matter what I try, * won't detect
faxes. I have faxdetect=both in zaptel.conf and my extensions.conf
looks like this:
[fromPSTN]
exten = s,1,Answer
exten = s,2,DigitTimeout(2)
exten = s,3,ResponseTimeout(10)
exten = s,4,Wait(3)
exten =
Yes. It just never gets there. I have tested spandsp and associated
libs by using:
exten = s,2,Goto(fax,s,1)
and it works fine. * just won't auto detect the fax call.
On Tue, 08 Feb 2005 10:16:26 +1100, Howard Lowndes [EMAIL PROTECTED] wrote:
On Tue, 2005-02-08 at 09:49, Brian Dingman wrote
=Incoming 000-000-
;callerid=asreceived
channel = 1
faxdetect=both
On Mon, 7 Feb 2005 19:16:18 -0500, Brian Dingman [EMAIL PROTECTED] wrote:
No, the fax extension position is at the end... Does it matter?
show dialplan is as follows for the fax extension:
[ Context 'fax' created
I ran across this on the wiki
http://www.voip-info.org/wiki-NVBackgroundDetect
Is anyone using this? Seems a little more robust.
On Mon, 7 Feb 2005 19:39:05 -0500, Brian Dingman [EMAIL PROTECTED] wrote:
Here is my zapata.conf file for grins
[channels]
;
; X100P plugged into PSTN
Thanks.
I noticed this today also. And was curious what was causing this.
Might want to add it to the wiki.
On Mon, 07 Feb 2005 18:36:21 +, Asterisk [EMAIL PROTECTED] wrote:
We've managed to setup spandsp to receive faxes and email them to the
appropriate person.
We did all of our
No.
My answer was assuming you didn't have one in originally.
On Tue, 08 Feb 2005 13:46:26 +1100, David Uzzell
[EMAIL PROTECTED] wrote:
Brian Dingman wrote:
This is just a guess, but try an Answer before sending it to VM.
Hmm ok not sure what that would do but I am willing to try anything
Thank you. Worked like a charm. I never would have caught that on my own.
On Mon, 07 Feb 2005 23:06:54 -0500, Roger Gulbranson
[EMAIL PROTECTED] wrote:
On Mon, 2005-02-07 at 19:39 -0500, Brian Dingman wrote:
Here is my zapata.conf file for grins
[channels]
;
; X100P plugged into PSTN
I haven't tried identapop, but an alternative is to use netcat along
with YAC listener on the windows PC.
See the wiki:
http://www.voip-info.org/tiki-index.php?page=Asterisk%20call%20notification
Works well for me.
On Sat, 5 Feb 2005 11:49:03 +, John Middleton
[EMAIL PROTECTED] wrote:
Hi,
I am trying to do the following:
1. Call comes in to my * box over IAX (VP Connect DID)
2. Check to see if call should be forwarded to my cell
3. Forward the call to my cell phone and take * out of the media path.
I am able to do all of the above except * is not able to natively
bridge the call.
Maybe I am missing your exact point, but what about handling this in
your extensions.conf
[voicepulse-incoming]
exten = 2124007999,1,Goto(nyc,s,1)
exten = 2124007998,1,Goto(nyc2,s,1)
That will put calls to 2124007999 into context nyc and calls to
2124007998 into context nyc2.
I guess the real
Who is your DID provider?
On Sun, 6 Feb 2005 15:03:19 -0500, Scott Simpson [EMAIL PROTECTED] wrote:
Hi,
I setup asterisk as an autoattendant. When I call using IAX I get the
autoattendent okay, but when I dial one of the extensions, there is no
ringing sound passed back to the caller.
It
A lot of times we all overlook the obvious or easiest way to do things :)
On Sun, 6 Feb 2005 13:26:25 -0800 (PST), Tim Burt
[EMAIL PROTECTED] wrote:
Ah.. the obvious. I don't know why I missed it.
I am just a newbie at this PBX stuff.
Thanks for the pointer. It worked. First off.
I have contacted VP regarding this issue and have included links to
this thread. My ticket number is [Incident: 050120-92] for
reference. Might want to fire off an email referencing it
On Fri, 4 Feb 2005 18:27:18 -0500, Daryl G. Jurbala
[EMAIL PROTECTED] wrote:
-Original Message-
2, 2005, at 1:25 PM, Brian Dingman wrote:
Finally got a reply from LV support. Not what I was hoping for.
Hopefully they will file a bug with Digium since they investigated the
issue not holding my breath.
Since this is such basic * functionality that they can't seem to
accomplish
Finally got a reply from LV support. Not what I was hoping for.
Hopefully they will file a bug with Digium since they investigated the
issue not holding my breath.
Since this is such basic * functionality that they can't seem to
accomplish I would think twice before aquiring DID's from them.
I have been looking around for Outlook Integration for Asterisk. Saw
the Asterisk TAPI wiki page and also ran across this:
http://www.fonality.com/pop.cgi?page=pop_pbxtray.tt (PBXtray)
It looks like Fonality has managed to make an app that does screen
pops and allows click to dial. Has anyone
:
Curiousity question, do you know how this would work, maybe as operating
as a softphone in windows?
Dan
On Tue, 1 Feb 2005, Brian Dingman wrote:
I have been looking around for Outlook Integration for Asterisk. Saw
the Asterisk TAPI wiki page and also ran across this:
http
So I guess that begs the question... Does anybody know where to get
the PBXtray app that Fonality uses?
What if they are using IdentaPoP for window Pop functionality.
Probably can't get that under the GPL.
On Mon, 31 Jan 2005 12:53:23 -0800, Manjit Riat [EMAIL PROTECTED] wrote:
The partner list
:05:15 -0500, Brian Dingman [EMAIL PROTECTED] wrote:
So I guess that begs the question... Does anybody know where to get
the PBXtray app that Fonality uses?
What if they are using IdentaPoP for window Pop functionality.
Probably can't get that under the GPL.
On Mon, 31 Jan 2005 12:53:23
Placing a call from Outlook is trivial with AST TAPI. How does one
accomplish a screen pop in Outlook?
Here is some more info from fonality:
PBXtray is not developed as part of Asterisk source. It is written in
an entirely different language (C++) and is a Windows app. It
integrates with our
I have also had issues with VP Connect ONLY on incoming calls also. It
doesn't happen all the time and has cleared up in recent weeks. But
when it happens, it would sound like I was listening to the caller
through a blown speaker.
Have you reported this problem to them?
Some things to try would
http://www.iax.cc has Vegas numbers.
On Sun, 30 Jan 2005 17:05:57 -0800, Manjit Riat [EMAIL PROTECTED] wrote:
I am thinking of dumping broadvoice so I need some other VoIP providers that
have a las vegas DID and a service better than broadvoice.
There is the little problem of having to switch numbers and then
communicating to everyone that the number has changed. This also only
seems to be a problem on inbound calls.
On Sat, 29 Jan 2005 23:34:49 -0500, Andrew Kohlsmith
[EMAIL PROTECTED] wrote:
On January 29, 2005 11:29 pm, Brian
This is driving me crazy. I have resorted to using the m option in the
Dial command just so folks don't hang up. I can't believe nobody else
is having this issue.
Any ideas to work around this?
On Wed, 26 Jan 2005 12:11:42 -0500, Brian Dingman [EMAIL PROTECTED] wrote:
Some more info. Using
is
full of bugs and not their only supported platform.
On Fri, 28 Jan 2005 02:21:36 -0500, Andrew Kohlsmith
[EMAIL PROTECTED] wrote:
On January 27, 2005 11:20 pm, Brian Dingman wrote:
To combat this problem you will want to change the following line to
actually do something:
exten = dial
From Support:
Asterisk is full of bugs and in many cases you fix one thing only to
have another show up.
We suggested users move to 1.0.3
Our team will look at more things in the software as a part of our ongoing
support to clients. We are looking at this version as well as 1.0.3
for some other
They definitely have capacity issues as well. It is not uncommon to
get a busy signal when placing a call. i.e. Making Progess - Nobody is
available.
This is a real pain too since the Dial Command won't rollover to the
next step once it starts making progress.
On Thu, 27 Jan 2005 12:59:17
PCMU is g711 ULAW and PCMA is G711 ALAW. ALAW is more common in
Europe. Not sure about VAD.
On Thu, 27 Jan 2005 18:44:09 -0700, Kim Lux [EMAIL PROTECTED] wrote:
Thanks for the tips.
The Grandstream doesn't have a G711 or uLaw option for codecs. It has
PCMU, PCMA and iLBC. Are any of
Just as an fyi.. one of the problems I am having with LiveVoip and my
guess is that some of you are also is that the LiveVoip call starts
making progress but for whatever reason it comes back and says nobody
available.
To combat this problem you will want to change the following line to
actually
Here is the call flow:
[ivr-incoming]
exten = s,1,LookupCIDName
exten = s,2,DigitTimeout(2)
exten = s,3,ResponseTimeout(10)
exten = s,4,Wait(1)
exten = s,5,Background(custom/ivr-incoming)
exten = 1,1,Background(pls-wait-connect-call)
exten = 1,2,Dial(${RINGPHONENUMBERS},20,r)
exten =
Some more info. Using this exact call flow, ringback works for PSTN
callers over WIldcard, IAX Callers over VP Connect, but NOT IAX
callers over LiveVoip. Could this possibly be a bug with their new
patch?
On Wed, 26 Jan 2005 11:43:59 -0500, Brian Dingman [EMAIL PROTECTED] wrote:
Here
Is anyone else having issues with ringback (see my other post to this
list) since the patch last night.
On Wed, 26 Jan 2005 09:33:50 -0600, Tim Lewis [EMAIL PROTECTED] wrote:
LiveVoIP did not issue any end user patches last night. They had a
problem connecting to Level 3's network. LiveVoIP
There was discussion of this before... I thought:
cvs checkout -r v1-0
would get you the latest stable version 1.0.X code
On Tue, 25 Jan 2005 08:58:46 -0500, [EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
Follow these simple steps to update you tree :
# cd /usr/src
# export
Mark,
I don't know what to tell you. With my DID's from VP Connect, DTMF
works fine over IAX. Even one of the lines I have with LiveVoip seems
OK over IAX. The other well... it really doesn't work at all.
So what does this say about * and DTMF recognition over IAX? Or the
service providers?
On
Keith,
VP Connect is having issues right now with callerid being
transmitted... as much as they don't want to believe it. Sometimes it
works, sometimes it doesn't. Maybe this is part of the problem. Does
PM not work 100% of the time for you?
On Mon, 24 Jan 2005 21:29:37 -0500, Keith O'Brien
Mark,
I have the same exact settings except I moved to 1.0.5. DTMF
recognition is fundamental to using *. Problems like this shouldn't
happen. As for the LiveVoip DID's, the two of them I have are down and
out. They were rendering fast busy signals - totally different problem
than DTMF, so support
So are you saying that * does not see the callerid but it should. Is
this a possible bug in the callerid application. RIght now I am seeing
that callerid isn't recognized 100% of the time (or possibly not
transmitted) when I receive calls from VP Connect. If I do a
NoOp(${CALLERIDNUM}) on incoming
LiveVoip has a problem with Asterisk users on versions less than 1.0.3 If
you are not using that version you need to upgrade now.
We have a problem with two of our carriers at their gateway related to the
Asterisk users. Our staff has developed a patch that is
being tested at this time. Once the
I have a couple of DID's with LiveVoip and am having major DTMF issues
on incoming calls. I am connecting to them through IAX using ULAW.
When someone dials one of these DD's (from a landline) they are for
the most part unable to navigate the IVR menu successfuly. I would say
the failure rate is
The ironic thing here is that dtmf works fine for my VP Connect DID's.
Go Figure. Also outgoing call quality is perfectly fine in and out for
me. The ONLY issue I have is with incoming call's incoming audio not
being very crisp throughout the call. I had there regualr service with
this SAME number
Out of the two DID's I have with LiveVoip, one works OK. My toll-free
DID is horrific. DTMF accuracy is less than 10%.
On Mon, 24 Jan 2005 14:29:12 -0700, Brandon Patterson
[EMAIL PROTECTED] wrote:
Our people are looking at this right now and have been for the past few
days.
Use Asterisk
Did you ever get DTMF to work reliably with LiveVoip. I am having the
exact same problems.
On Mon, 17 Jan 2005 20:22:30 -0500, Jess Coburn [EMAIL PROTECTED] wrote:
Hello all,
What my app does is accepts a call in on a Dial-In Number (DID) via
IAX, and then prompts the caller for the top
Did you ever figure a way around this? It would be a good time to test
since LiveVoip is having some issues today.
On Sat, 8 Jan 2005 14:44:23 -0500, Nabeel Jafferali
[EMAIL PROTECTED] wrote:
Hello.
I am having an issue where sometimes the cheapest provider for certain
international
Any thoughts? Could this be a jitterbuffer problem?
On Fri, 21 Jan 2005 19:08:59 -0500, Brian Dingman [EMAIL PROTECTED] wrote:
I have a couple of DID's through VP Connect and have been having sound
quality issues on incoming calls. During the call, the calling parties
voice sometimes sound
I am seeing the following
asterisk*CLI iax2 show registry
217.160.244.186:4569 usernamexx.xx.xxx.xx:4569 60 Registered
66.234.228.170:4569 username xx.xx.xxx.xx:4569 60 Registered
65.39.205.121:4569username xx.xx.xxx.xx:4569 60 Registered
On Sat,
I have a couple of DID's through VP Connect and have been having sound
quality issues on incoming calls. During the call, the calling parties
voice sometimes sound like it is crackling, in other words it is not
very crisp. I would liken it to listening to a radio with a blown
speaker. This sound
Kurt,
Here is a real basic setup of how the a extension can be used in
context with the rest of the dialplan. The a extension must call
VoiceMailMain NOT Voicemail or you will get your voicemail again and
not the voicemail system.
[fromPSTN]
exten = s,1,Answer
exten =
If you put the following in your Dialplan, pressing * should break you
out of voicemail and call VoiceMailMain
exten = a,1,VoicemailMain,EXTEN
exten = a,2,Hangup
On Wed, 19 Jan 2005 11:33:23 -0500, kurt x [EMAIL PROTECTED] wrote:
I want to know if there is way to break out of the voicemail
Put /usr/local/lib in /etc/ld.so.conf then run ldconfig.
[app_rxfax.so]Jan 18 15:46:05 WARN
ING[7952]: loader.c:258 as
t_load_resource: libspandsp.so.0: cannot open shared
object file: No su
ch file or directory
Jan 18 15:46:05 WARNING[7952]: loader.c:
440 load_modules: Loading module
It has to do with spandsp and receiving incoming faxes. This should
probably be updated in the documentation.
On Tue, 18 Jan 2005 17:09:04 -0800 (PST), beonice [EMAIL PROTECTED] wrote:
--- Brian Dingman [EMAIL PROTECTED] wrote:
Put /usr/local/lib in /etc/ld.so.conf then run
ldconfig
http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20DISA
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This link might help:
http://www.dslreports.com/forum/remark,11775216~mode=flat
On Fri, 14 Jan 2005 23:29:34 -0500, Randy [EMAIL PROTECTED] wrote:
Chris,
I do not have VoicePulse Open Access, but I do have an incoming number through
VoicePulse Connect. You might want to give the following
Can you show us the CLI output of what is happening?
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I grabbed the latest sources from CVS yesteday and am having problems
compiling. * v1.0.3 was running previously without issue. I tried
checking out the older source but get the same make errors.
The box is running RH 9. I am getting the following errors. Any
thoughts on what is wrong?
gcc
I edited the makefile and asterisk builds properly, but when I go to
start it, I get the following error:
[app_rxfax.so]Jan 11 18:44:12 WARNING[13877]: loader.c:258
ast_load_resource: libspandsp.so.0: cannot open shared object file: No
such file or directory
Jan 11 18:44:12 WARNING[13877]:
the latest sources seem to be
problematic.
The machine is an AMD 1700+ with 512MB RAM.
On Wed, 12 Jan 2005 14:53:00 +1300, Matt Riddell
[EMAIL PROTECTED] wrote:
Brian Dingman wrote:
I grabbed the latest sources from CVS yesteday and am having problems
compiling. * v1.0.3 was running
At the time I didn't realize it was a common error. I thought it was a
problem with the Makefile. I promise to google before I post :)
Anyway it works now... somewhat. tiff's are incomplete but I will have
to troubleshoot more.
On Wed, 12 Jan 2005 14:37:48 +1300, Matt Riddell
[EMAIL PROTECTED]
Asterisk v1.0 is running on RH 9. I installed festival RPM
(festival-1.4.2-16.i386.rpm) and edited the festival.scm file to add:
(define (tts_textasterisk string mode)
(tts_textasterisk STRING MODE)
Apply tts to STRING. This function is specifically designed for
use in server mode so a single
I changed all the text to lower case and removed the quotes. After
doing so, I got the following error:
SIOD ERROR: wrong type of argument to car : wholeutt
Strange thing is sometimes I get the error and sometimes I don't
On Mon, 10 Jan 2005 13:54:25 -0500, Brian Dingman [EMAIL PROTECTED
Anyone care to pass on a makefile that works. This is what my
makefile.rej looks like:
***
*** 71,76
rm -f $(DESTDIR)$(MODULES_DIR)/app_datetime.so
rm -f $(DESTDIR)$(MODULES_DIR)/app_qcall.so
app_curl.so: app_curl.o
$(CC) $(SOLINK) -o $@ $ $(CURLLIBS)
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