Hi,
While compiling destival 1.4.2 on redhat 8.0 i got the final error ...
Making in directory intonation/tilt ...
Making in directory ./lib ...
Making in directory ./main ...
gcc -O3 -Wall -o ch_lab ch_lab_main.o -L../lib -lestools -L../lib -lestbase
-L../lib -leststring -ltermcap -ldl -lm -lstd
this is what it printed
(gdb) bt
#0 0x40120cb3 in chunk_free () from /lib/libc.so.6
#1 0x40120c53 in free () from /lib/libc.so.6
#2 0x403e7977 in __builtin_delete (ptr=0x80e41f8) from
/usr/local/lib/liboh323wrap.so
#3 0x4103a27a in PContainer::Destruct () from
/usr/local/lib/libpt_linux_x86_
A sip debug shows me the sip messages in their entirety.
Previously, when the "set verbose " setting was used, it would show a single
line describing the registration.
I was told that 3 is the highest verbose setting but this doesn't seem to
work either.
T
- Original Message -
From: "M
Well, someone is attempting to delete a object that has never been
allocated.
You might give chan_h323 a try, you might have better luck.
http://asterisk.nufone.net
Jeremy McNamara
Ben Clark wrote:
this is what it printed
(gdb) bt
#0 0x40120cb3 in chunk_free () from /lib/libc.so.6
#1 0x4
I'm hoping that someone else has seen this
problem.
Running CVS version from 03/08/03
20:00
T100P -> Zplex-10
All internal call routing seems to work fine.
Atempts to make calls on the FXO interface fail. Asterisk picks up the
correct number dialed but there seems to be a problem with
At 22:39 8-3-2003 +, you wrote:
Will it display caller ID on an analog phones display?
Sorry - don't know.
Yes it will - and if your phones support it - CallerIDname too :-)
Florian
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I've to project and build a fresh new box with * on.
Basically, i'll have this situation:
The office is connected to the phone carrier with a PRI.
I need to let users continue to use their analog phones in a office, and an
IP-phone based solution on the remote office, that will call using outside
On Sun, 2003-03-09 at 03:50, Raymond McKay wrote:
> I'm hoping that someone else has seen this problem.
>
> Running CVS version from 03/08/03 20:00
> T100P -> Zplex-10
>
> All internal call routing seems to work fine. Atempts to make calls
> on the FXO interface fail. Asterisk picks up the co
Ok, so apparently you compiled it from source? I have no problem with
doing that for gnophone and IAX. The builds to both of them seem pretty
noninvasive. They require only ./configure, make, and make install.
However the gsm source requires the actual make file be edited and not
being a cod
What did you change in the gsm make file? I've downloaded 3 source
packages: gnophone, IAX, and gsm. The IAX and gnophone appear to be
straightforward builds however the gsm requires editing of the make file
at least from the way I read it.
Thanks for all assistance.
AJ
On Sat, 8 Mar 2003, B
Are you building it from source or are you using the rpms?
AJ
On 8 Mar 2003, Karl Putland wrote:
> On Sat, 2003-03-08 at 07:02, William X Walsh wrote:
> > On Fri, 2003-03-07 at 19:32, [EMAIL PROTECTED] wrote:
> > > Has anyone been able to successfully get gnophone to work on a RedHat 8.0
> > >
Does kphone work with asterisk as well? If so, does it do all the same
things as gnophone?
AJ
On 8 Mar 2003, William X Walsh wrote:
>
> I had it working on redhat 7.3, but had nothing but problems with Redhat
> 8, like the original poster.
>
> I gave up and moved to using KPhone on linux.
>
I guess I have to figure out how to use CVS today! I will try to figure
out how to check things out of CVS.
AJ
On Sat, 8 Mar 2003, David T Hollis wrote:
> William X Walsh wrote:
>
> >On Fri, 2003-03-07 at 19:32, [EMAIL PROTECTED] wrote:
> >
> >
> >>Has anyone been able to successfully get gno
On Sun, 2003-03-09 at 06:32, Stefano Finetti wrote:
> I've to project and build a fresh new box with * on.
>
> Basically, i'll have this situation:
>
> The office is connected to the phone carrier with a PRI.
> I need to let users continue to use their analog phones in a office, and an
> IP-phone
It uses SIP instead of IAX, but yeah, it works.
On Sun, 2003-03-09 at 07:08, [EMAIL PROTECTED] wrote:
> Does kphone work with asterisk as well? If so, does it do all the same
> things as gnophone?
> AJ
>
>
> On 8 Mar 2003, William X Walsh wrote:
>
> >
> > I had it working on redhat 7.3, but
I did not compile IAX or GSM. I just installed the binary rpms (iax,
iax-develop, gsm, gsm develop. Since they do not depend on Mozilla, the
libraries just work.
I configured gnophone using the disable-mozilla option to configure and
compiled and installed it.
Hope that helps.
Gregg
On Sun,
Hello,
first of all I have to say: I read the handbook but there are some questions
left I couldn't answer. It's very basicaly. Sorry about that.
I have a AVM B1 PCI - Card. I think there is a I4L and a Capi-Port installed.
1. How can I define the right channels, how are they defined. I found
Friend of mine in the UK setting up an asterisk box is looking for an
ATA186 source in the UK. Anyone who knows of one, please email me
offlist.
--
William Walsh <[EMAIL PROTECTED]>
Jabber: [EMAIL PROTECTED]
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On Sun, 2003-03-09 at 08:07, [EMAIL PROTECTED] wrote:
Here's what I did to build gnophone.
dnld and install gsm rpms from
ftp://ftp.asterisk.org/pub/telephony/gnophone/gsm-1.0.10-2.i386.rpm
ftp://ftp.asterisk.org/pub/telephony/gnophone/gsm-devel-1.0.10-2.i386.rpm
mkdir -p ~/src/gnophone
cd ~/sr
Ok, for a total idiot on this end where do I configure the disable-mozilla
option? Do I pass that as a command line arguement or do I have to edit a
file? A second question there, what is mozilla's relationship to asterisk
or gnophone anyway? I merely want gnophone to be able to make calls and
Hi Cristoph,
the B1 has linux capi 2.0 drivers, so you can use chan_capi.
Grab it at http://www.junghanns.net/chan_capi.html
read INSTALL and README, take a look at the sample configuration
file capi.conf.
regards
kapejod
--
Klaus-Peter Junghanns
CEO,CTO
Junghanns.NET Internet-Services & Softw
Anyone having trouble parking calls? I haven't tried it in a while,
but it seems to have stopped working. If I dial 700, I get a invalid
extension. I have "include => parkedcalls" in the correct context, and
I can dial 701, which tells me no call is parked there.
Any ideas? Parking.conf is
dude,
cd to your gnophone source directory (in my case this is
/usr/src/gnophone-0.2.4).
type the command: ./configure --disable-mozilla
type the command: make install
gnophone uses mozilla to browse phone related information (web pages).
You don't need it to make calls.
asterisk doesn't use mo
When you park the call, does it say the number it parked it to?
Could you paste the parking.conf, even if it is stock?
denon
At 10:47 AM 3/9/2003 -0800, you wrote:
Anyone having trouble parking calls? I haven't tried it in a while, but
it seems to have stopped working. If I dial 700, I get a
Nope, doesn't say anything, just get an invalid extension.
Here is parking.conf
;
; Sample Parking configuration
;
[general]
parkext => #700 ; What ext. to dial to park
parkpos => 701-720 ; What extensions to park calls on
context => parkedcalls
tried this parking.conf too...
;
; Sample Parking configuration
;
[general]
parkext => 700 ; What ext. to dial to park
parkpos => 701-720 ; What extensions to park calls
on
context => parkedcalls ; Which context parked calls
are in
par
>Anyone having trouble parking calls? I haven't tried it in a while,
>but it seems to have stopped working. If I dial 700, I get a invalid
works fine here, make sure you have 't' parm on the dial app,
ie. exten => 1,2,Dial(Zap/1,20,t)
& make sure your doing #700 to park after receiving the cal
You do have an extensions.conf entry for it (700), right?
At 11:18 AM 3/9/2003 -0800, you wrote:
tried this parking.conf too...
;
; Sample Parking configuration
;
[general]
parkext => 700 ; What ext. to dial to park
parkpos => 701-720 ; What extensions
Hi..
I just wondering why DTMF are not recognized by aterisk on incoming calls
from my SIP provider ...
ANy suggesteions ?`
/Mike
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you're not supposed to have an extension conf entry for 700, nor th
eothers. the res_parking creats them as needed.
Mark
On Sun, 9 Mar 2003, denon wrote:
> You do have an extensions.conf entry for it (700), right?
>
> At 11:18 AM 3/9/2003 -0800, you wrote:
> >tried this parking.conf too...
> >;
try the new "dtmfmode" parameters on the user or peer. Note they are not
currently valid in the "[general]" section. you can set dtmfmode=inband or
dtmfmode=rfc2833
Mark
On Sun, 9 Mar 2003, Mikael Andersson wrote:
>
> Hi..
>
> I just wondering why DTMF are not recognized by aterisk on incoming
Didn't know you needed the t. Is that a new thing?
On Sunday, March 9, 2003, at 11:38 AM, TC wrote:
Anyone having trouble parking calls? I haven't tried it in a while,
but it seems to have stopped working. If I dial 700, I get a invalid
works fine here, make sure you have 't' parm on the dial
No, I just include parkedcalls in to the correct context. My
understanding was 700 was internal to the pbx and there is no need or
for that matter ability to create it by hand.
Thanks,
Mike
On Sunday, March 9, 2003, at 11:30 AM, denon wrote:
You do have an extensions.conf entry for it (700), r
doh, whoops ..
*hides in shame* .. been too long since I configured parking, sorry.
At 02:38 PM 3/9/2003 -0600, you wrote:
you're not supposed to have an extension conf entry for 700, nor th
eothers. the res_parking creats them as needed.
Mark
On Sun, 9 Mar 2003, denon wrote:
> You do have an ex
only for # transfer, not for flash-hook transfer.
Mark
On Sun, 9 Mar 2003, Mike Reiling wrote:
> Didn't know you needed the t. Is that a new thing?
>
> On Sunday, March 9, 2003, at 11:38 AM, TC wrote:
>
> >> Anyone having trouble parking calls? I haven't tried it in a while,
> >> but it seems
well parking is realy just a transfer, and lower case 't' always indicated
that the Called channel can transfer, it'd be nice to see orginating channel
could transfer if the uppper case 'T' flag was present :)
-Original Message-
From: Mike Reiling <[EMAIL PROTECTED]>
To: [EMAIL PROTECTED]
>Are you stripping the digit you use to specify it is goin to be an
> external call?
I don't use a digit for that purpose. I have always used
exten=>_NXX,1,Dial(Zap/g2/BYEXTENSION)
Was using this just fine on the Adtran 750 on the same lines but the
Adtran was just on loan so I'm stuck wi
Sorry I took you through that. A moment after I sent your reply I sat
down and thought about it, took a look at the config.in and executed the
./configure --disable-mozilla. Once I did that the configure script
coughed at me about the absense of gdk and imlib so I shot over to my
local mirro
I think you need to test it with asterisk since it only talks iax. Maybe
this is a good time to install asterisk and get it going.
Gregg
On Sun, 2003-03-09 at 17:37, [EMAIL PROTECTED] wrote:
> Sorry I took you through that. A moment after I sent your reply I sat
> down and thought about it, too
On Sun, 2003-03-09 at 16:04, Raymond McKay wrote:
> >Are you stripping the digit you use to specify it is goin to be an
> > external call?
>
> I don't use a digit for that purpose. I have always used
>
> exten=>_NXX,1,Dial(Zap/g2/BYEXTENSION)
>
> Was using this just fine on the Adtran 750
Could you please list codecs that are supported with this channel driver?
The case is that, voip provider I wanted to use for routing international
calls
supports only those 723, 729.
Is it possible to use it with asterisk?
Best regards,
Krzysztof Bujak
- Original Message -
From: "Jeremy
(note: this has only been tested with my Cisco ATA-186 v2.15 and SIP)
Update: inbound calls now work, again thanks to Mark's banging on the
code. I am able to receive calls on my ATA-186 with the settings
below, behind an Apple Airport NAT/PAT translator, as of CVS updates
late afternoon yest
At 14:39 2003-03-09 -0600, Mark Spencer wrote:
try the new "dtmfmode" parameters on the user or peer. Note they are not
currently valid in the "[general]" section. you can set dtmfmode=inband or
dtmfmode=rfc2833
Mark
On Sun, 9 Mar 2003, Mikael Andersson wrote:
Exactly where shoud I enter that val
Look into sip.conf.sample
[general]
port = 5060 ; Port to bind to
bindaddr = 0.0.0.0 ; Address to bind to
context = default ; Default for incoming calls
;tos=lowdelay
;tos=184
;maxexpirey=3600; Max length of incoming registration we
al
chan_h323 works with any codec that Asterisk can deal with. chan_h323
lets Asterisk deal with the audio directly, instead of attempting to
emulate a sound card.
If you want G.729 you are going to need to contact Greg Vance at Digium
to purchase a license.
Jeremy McNamara
Krzysztof Bujak
At 00:50 2003-03-10 +0100, Andre Bierwirth wrote:
Look into sip.conf.sample
[general]
port = 5060 ; Port to bind to
bindaddr = 0.0.0.0 ; Address to bind to
context = default ; Default for incoming calls
;tos=lowdelay
;tos=184
;maxexpirey=3600
Hmm... haven't been able to get this to work on my Cisco ATA-186.
Perhaps I'm trying the incorrect knobs? I'm making outbound calls
ATA-186->*->iconnecthere->PSTN.
I've set my ATA-186 to these various settings:
AudioMode: 0x00150015
AudioMode: 0x00250025
AudioMode: 0x00050005
(per the settings
Hi all,
Some problems again.
I am testing a setup like this:
phone-usb-*SIP-eth-SIP*SIP-NAT-Internet-SIP*
i.e. just trying to call outside my network.
When I reach asterisk on the enge of the internal net everything works fine
but when I make further connection to * in the internet I get connected
So I understand that G.723 is included in asterisk
distribution?
Sorry for lame:-(
Best regards,
KRzysztof Bujak
- Original Message -
From:
Jeremy McNamara
To: [EMAIL PROTECTED]
Sent: Sunday, March 09, 2003 11:43
PM
Subject: Re: [Asterisk-Users] H323 on and
o
What does blacklisting a call with *80 do? I tried it by dialing from my
cell, which presents caller id. I then blacklisted and the console debug
reported that it did it. But it seemed to have no effect. I could still
call again.
Thanks...
Jim
_
Is there a way to configure a channel to ethernet, such that we could use
one of the VoIP long distance carriers, like Vonage?
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It's because you're not using lookupblacklist
application on your callflow.
regards
Martin
On Sun, 9 Mar 2003, Jim Archer wrote:
> What does blacklisting a call with *80 do? I tried it by dialing from my
> cell, which presents caller id. I then blacklisted and the console debug
> reported that
Make sure you have app_lookblacklist loaded. and configured in your
incoming context for your incoming lines. I can send you me ext.conf
if you want an example
On Sunday, March 9, 2003, at 06:29 PM, Jim Archer wrote:
What does blacklisting a call with *80 do? I tried it by dialing from
my ce
Hey all,
Just a quick question.
I've managed to configure my generic rockwell voice modem with asterisk and
can make calls out to the PSTN through it. Thing is I get one way voice.
Is this a problem with the voice modem not being capable of full-duplex
voice, or is it the voice modem channel dri
Is there a way to configure a channel to ethernet, such that we
could use one of the VoIP long distance carriers, like Vonage?
Yes.
See http://www.loligo.com/asterisk/
for some sample configs using iconnect.
vonage does not (supposedly) allow people to use the
username/password they give yo
On Sun, 2003-03-09 at 21:23, Jamie Carl wrote:
> Hey all,
>
> Just a quick question.
>
> I've managed to configure my generic rockwell voice modem with asterisk and
> can make calls out to the PSTN through it. Thing is I get one way voice.
> Is this a problem with the voice modem not being capab
Check your zapata.conf too to be sure stripmsd=0 or is undefined.
Mark
On Sun, 9 Mar 2003, Raymond McKay wrote:
>
> >Are you stripping the digit you use to specify it is goin to be an
> > external call?
>
> I don't use a digit for that purpose. I have always used
>
> exten=>_NXX,1,Dial(Zap/
Just check-out asterisk from cvs, It compile but
crashes right off with?
# Ouch ... error while writing audio data: : Broken pipe
Ouch ... error while writing audio data: : Broken pipe
Any ideal how far back I need to go to get a working
build?
___
Asteri
> > I've managed to configure my generic rockwell voice modem with asterisk
and
> > can make calls out to the PSTN through it. Thing is I get one way
voice.
> > Is this a problem with the voice modem not being capable of full-duplex
> > voice, or is it the voice modem channel driver for asterisk?
make clean ; make install?
Mark
On Sun, 9 Mar 2003, James Sizemore wrote:
> Just check-out asterisk from cvs, It compile but
> crashes right off with?
>
> # Ouch ... error while writing audio data: : Broken pipe
> Ouch ... error while writing audio data: : Broken pipe
>
> Any ideal how far back
Thanks!
Now, I am a bit confused...
The example has (in sip.conf):
[iconnect]
type=friend
secret=stillnotmypassword
username=12691220
host=sipauth.deltathree.com
and in extensions.conf:
[macro-dialiconnect]
exten => s,1,SetCallerID(${ICONNECT1})
exten => s,2,SetCIDName(${MYNAME})
exten => s,3
Hell
rm -rf asterisk
cvs checkout asterisk
make samples
Same thing!
Mark Spencer wrote:
make clean ; make install?
Mark
On Sun, 9 Mar 2003, James Sizemore wrote:
Just check-out asterisk from cvs, It compile but
crashes right off with?
# Ouch ... error while writing audio data: : Broken p
No. G.723.1is encumbered with international patents and their
associated royalty fee's, so no, G.723.1 is not part of the standard
Asterisk distribution.
Jeremy McNamara
Krzysztof Bujak wrote:
So I understand that G.723 is
included in asterisk distribution?
Sorry for lame:
My buuld is just a week old. We see that error all the time, but its never
a problem...
--On Monday, March 10, 2003 12:03 AM -0600 James Sizemore <[EMAIL PROTECTED]>
wrote:
Hell
rm -rf asterisk
cvs checkout asterisk
make samples
Same thing!
Mark Spencer wrote:
make clean ; make install?
My example had a Macro to do the dialing towards iconnect, since I
got really tired of typing in the same dial routines (obviousl,
that's what Macros are intended to replace.)
So let's simplify my example and your configs. You have some extra
stuff in there that probably you should not have on
Thanks
I'll try this right now...
--On Sunday, March 09, 2003 10:09 PM -0800 John Todd <[EMAIL PROTECTED]>
wrote:
My example had a Macro to do the dialing towards iconnect, since I got
really tired of typing in the same dial routines (obviousl, that's what
Macros are intended to replace.)
S
Asterisk users:
Okay, I am thinking about getting ready to start an asterisk
server to use for a home phone system.
(a) Does anyone have a T1 card for sale used?
(b) Can a Carrier Access channel bank work okay with Asterisk?
(c) Anyone have an ADTRON 750 for sale
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