On Wed, Jan 03, 2007 at 07:16:06AM +0100, Remco Barendse wrote:
On Wed, 3 Jan 2007, Tzafrir Cohen wrote:
P[ 0] -- mISDN Channel Driver Registred -- (BE AWARE THIS DRIVER IS
EXPERIMENTAL!)
..Jan 2 23:07:23 ERROR[25747]: chan_zap.c:10603 setup_zap: Unknown
signalling method
Thanks very much Chris.
I found usage for NoOp and verbose in Future of Telephony Appendix C
and it looks like they will do exactly what I need.
Larry
Chris Tooley wrote:
If you mean in the dialplan, you can use NoOp or verbose (verbose being
something that will get logged too), and if
Hello to all
When my SNOM (300 or 320) loses Internet connectivity, it loses its
Asterisk registration (ok, thats normal).
But when the phone is back online, he doesn't try to register in
Asterisk. I believe this happens to avoid flooding the private LANs when
the Internet link is lost but
Hi!
I'm trying to set up an asterisk based PBX with a TDM400P +2 FXS +2
FXO modules in UAE/Dubai for home switching / voicemailing. I am using
the card Asterisk/Zaptel 1.4.0. I want to include a special route when
a certain caller calls into via PSTN. The problem is that I cannot
detect the
Bill Gibbs wrote:
My next step is to connect the fax machine to a Wildcard X100P.
Check to see if there is Echo cancellation in the SPA-1001, and if so
turn it off. If there is an adaptive Jitter buffer on the SPA-1001, try
changing it to a fixed one (probably no more than 40ms).
Why would
Hi all,
I was having a similar issue, using TE110P from Digium all incoming faxes
were detected and correctly received.
When trying to send outbound faxes, they all get broken... I do believe it
may be related with Echo Cancel enabled on my Zapata.conf, any ways i've set
also fax detect for
Hi Joao,
I'm not very experienced with SNOM, but have you though about providing fix
IP for you VoIP hardphones?
That way you could avoid the registration problem. At least while you don't
get your final solution.
Hope it helps,
MoutaPT
On 1/3/07, Joao Pereira [EMAIL PROTECTED] wrote:
Hi all!
I am shore someone have writing about it bout I cant find it.
I have a extension that I need to block from making expansive mobil calls.
Everyone else should be aloud to do the calls.
I am shore it is possible to be done sens I had a
commercial asterisk based PBX that I did that on.
Hi!
I have my Trixbox running behind a ISA server.
However it works fine with Rix telecom (The service provider)
The same setup dos not allow my phone trow the ISA server.
It is seeing the phone as registering the public
adress of the firewall instead of port forwarding it.
anyone else had this
On 1/1/07, ram [EMAIL PROTECTED] wrote:
On 12/30/06, Rajkumar S [EMAIL PROTECTED] wrote:
On 12/29/06, Chris Earle [EMAIL PROTECTED] wrote:
anyone know the status of disconnect supervision on POTS lines in India?
Set up an asterisk box, TDM cards, in Mumbai, and doesn't seem to have
Hi I have been asked to ind out if there is a way to use asterisk to
answere a voice fax modem so it can provide an answering service and
record messages ?
--
Gregory Machin
[EMAIL PROTECTED]
www.linuxpro.co.za
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I have native music on hold setup to play ulaw encoded files. No
transcoding, caller is on a g.711u SIP channel. There is horrible
distortion and noise between files for 1 to 2 seconds.
Has anyone seen this? I check the files and trimmed silence from the
end, the source of the noise is not the
Hi - I just got a Sangoma A200 card with a single 2FXO module and
what appears to be an empty module. I put the card in my Dell GX260,
but the power light on the front of the box just blinks and won't
power up. I did take the power cable from the CDROM to put on the
card - I don't need
joe a. wrote:
Mark Greene[EMAIL PROTECTED] Wrote on: 1/2/2007 12:58 PM:
I believe I am going to start out with some refurbished Dell Poweredge
servers. They have had a high success rate with a friend.
I was going to go that route as well. But, depends on the model. I have
several of the
Colin Anderson wrote:
ASUS motherboards, in particular, have worked for me perfectly, everytime
with both Digium and Sangoma cards. They are also easy to work with and well
documented.
-Original Message-
From: Doug [mailto:[EMAIL PROTECTED]
Sent: Tuesday, January 02, 2007 1:04 PM
It sounds like a bad card. Call Sangoma and ask them to replace
it. You don't need to use the drive power cable for just a single
fxo module. You only need it for the fxs or if you go over 2 fxo cards.
In any case, it should not stop your computer from booting.
Tom
At 07:43 AM 1/3/2007,
Hi - I just got a Sangoma A200 card with a single 2FXO module and
what appears to be an empty module. I put the card in my Dell GX260,
but the power light on the front of the box just blinks and won't
power up.
Maybe your card is not properly seated.
seems to have a lack of documentation, but
If the light on the dell is blinking amber... that typically means you
have a power issue.
Rob
Time Bandit wrote:
Hi - I just got a Sangoma A200 card with a single 2FXO module and
what appears to be an empty module. I put the card in my Dell GX260,
but the power light on the front of the box
On Wed, 3 Jan 2007, Derek Whitten wrote:
no problems on my proliant DL580
Nothing but problems with my DL380's until I ran a non-SMP kernel.
Thanks in advance,
Steve Edwards [EMAIL PROTECTED] Voice:
When I try to record a call the console shows:
www*CLI
Starting recording of MeetMe Conference 1 into file
meetme-conf-rec-1-1167836078.0.wav.
www*CLI
The code being executed in extensions.conf is:
exten = s,n(record),MeetMe(,rDMpc) ;Make new Room and record call.
exten =
I've replaced 2XTE110 with an A102 with echo cancellation specifically to
deal with echo problems. However, user feedback has indicated to me that on
some calls (not a lot, but some) the call is unusable, with audio
artifiacts, described by one user, as: very bad phasing reverb feedback
(from my
Hello List
Does anybody have any experience with the FoneBridge line of products from
RedFone?
I think their HA implementation sounds interesting, and like the prospect of
having dedicated hardware for our PRI connections.
Kind Regards
Jon Leren Schøpzinsky
Steve Edwards wrote:
On Wed, 3 Jan 2007, Derek Whitten wrote:
no problems on my proliant DL580
Nothing but problems with my DL380's until I ran a non-SMP kernel.
Thanks in advance,
Steve Edwards [EMAIL
Hey guys,
I need to set up asterisk so that it sends the voicemail to the users email.
I understand that I need to say attatch=yes, but what else needs to be
done. I would think that somewhere I need to specify the server that it uses
to send the email, etc.
- Mark
lots of companies make scsi dvd drives -- g00gl3 is your friend...
http://www.google.com/search?hl=enq=scsi+dvdbtnG=Google+Search
Well, who'd have thought? All my ususal suppliers said no one makes them.
joe a.
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Aha, it just happened to me, so now I can characterize the audio: It
basically sounds like it's missing every other sample - fuzzy and distorted.
Timing?
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To
We tried them out early last year when we were looking at a large deployment
and they gave us a lot of the redundancy that we wanted. However we did run
into issues where calls seemed to get caught up in the system. It was as far as
we could tell rather random. No consistency to it at all.
Zaptel.conf:
loadzone = us
span=1,1,0,esf,b8zs
span=2,1,0,esf,b8zs
bchan=1-23
dchan=24
bchan=25-47
dchan=48
Just a quick thought in looking at the settings above, it appears that you
have set both spans as the primary timing source. I am pretty sure that
only one span should be the
Try switching the order of the blank module and the FXO or remove the blank,
I had a similar Dell do the same and after some experimenting found that the
removing the blank solved the problem.
On 1/3/07, Rob Schall [EMAIL PROTECTED] wrote:
If the light on the dell is blinking amber... that
Hi Mattias, add this to your dialplan:
exten= _/CALLERIDNUMBER,1,Hangup()
; Basically you are doing a pattern match with callerid match on your first
priority!
; You may keep your remaining dialplan, no changes needed
Pls Give me some feedback
Best Regards,
MoutaPT
On 1/3/07, Mattias
There should be an example in your voicemail.conf
Here is mine...
mail is tagged from [EMAIL PROTECTED] and sent to [EMAIL PROTECTED]
In voicemail.conf
mailcmd=/usr/sbin/sendmail -f [EMAIL PROTECTED] [EMAIL PROTECTED]
You of course would use the mailer that your system uses. I have
I think you are absolutely right. The audio I heard earlier sounds exactly
like a timing issue. So:
wanpipe1.conf:
TE_CLOCK= NORMAL
TE_REF_CLOCK= 0
wanpipe2.conf:
TE_CLOCK= MASTER
TE_REF_CLOCK= 1
zaptel.conf:
span=1,1,0,esf,b8zs
span=2,0,0,esf,b8zs
I'm going to make
Is there anyone with any experience of using the AMD app and the
settings that worked for them in the UK ?
Any help would be appreciated.
Julian
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To UNSUBSCRIBE or
Colin Anderson wrote:
I've replaced 2XTE110 with an A102 with echo cancellation specifically to
deal with echo problems. However, user feedback has indicated to me that on
some calls (not a lot, but some) the call is unusable, with audio
artifiacts, described by one user, as: very bad phasing
Hi,
I am having a problem that is a miracle to me: If I dial out via
voipstunt.com the call rings for a few seconds and then gives me a busy
sign.
- I do not have a timeout set in my dial command
- the remote station does not cause the busy either
- dialing the number with the voipstunt
Arik Raffael Funke wrote:
Hi,
I am having a problem that is a miracle to me: If I dial out via
voipstunt.com the call rings for a few seconds and then gives me a busy
sign.
Start out with not using the r option to the Dial line. That will
remove the faked ringing tone.
Does anybody happen to know the input power specs for the Polycom IP 500
and IP 600? We've mixed up our power supplies and we've got a whole box
of them and can't figure out which go to the Polycoms. I would rather
not kill the phones by trying random ones
(my pstn calls are coming in thru an upstream asterisk server, so the
called and calling phone number is passed as an extension.)
when caller comes in on 555, he will go to extension 1234 where he
will wait for the API to make a call to 999 for him. how do I
bridge the two calls?
501 - 12V, 1A and a power/data cable
601 - 24V, 0.5A
650 - 24V, 0.5A
- Dave
On Jan 3, 2007, at 11:48 AM, Peder @ NetworkOblivion wrote:
Does anybody happen to know the input power specs for the Polycom
IP 500 and IP 600? We've mixed up our power supplies and we've got
a whole box of them
The 501 is 12VDC, and the 601 is 24VDC, as I recall. There was a post a
few months ago that said that plugging the 24VDC into a IP501 will fry
the phone.
Peder @ NetworkOblivion wrote:
Does anybody happen to know the input power specs for the Polycom IP 500
and IP 600? We've mixed up our
Newbie question for sure... I'm unsure of how to tell if chan_zap.so is loaded.
Also, does autoload in modules.conf take care of it or is it done explicitly?
output of lsmod | grep zap:
zaptel208388 16
wcusb,wctdm,wcfxo,wctdm24xxp,wcte11xp,wct1xxp,w
FWIW, our Polycom IP601 phones use a transformer with output: 24VDC
500mA (center contact is positive).
A Polycom reseller (or Polycom sales) could probably give you
information on these other two models.
Alvin
Peder @ NetworkOblivion wrote:
Does anybody happen to know the input power specs
John French wrote:
Newbie question for sure... I'm unsure of how to tell if chan_zap.so is loaded.
Also, does autoload in modules.conf take care of it or is it done explicitly?
output of lsmod | grep zap:
zaptel208388 16
wcusb,wctdm,wcfxo,wctdm24xxp,wcte11xp,wct1xxp,w
We've been holding back on this release to coincide with
the Asterisk 1.4.0 release.
This is mostly a compatibility release, but there are a
few new features:
* No longer requires register_globals in PHP
* Separated code from configuration settings in
I have a strange issue going on with one system. If you park a call and
then do a page command, the parked call gets dropped. Both park and page
use meetme, and it appears that the page uses the same conference number
as parked call. So when the page is complete and hangs up, it drops the
parked
From: Eric \ManxPower\ Wieling [EMAIL PROTECTED]
John French wrote:
Newbie question for sure... I'm unsure of how to tell if chan_zap.so is
loaded. Also, does autoload in modules.conf take care of it or is it done
explicitly?
output of lsmod | grep zap:
zaptel208388 16
Just a thought, maybe it won't boot because there is no power to the CD ROM
drive
[EMAIL PROTECTED] 01/03/07 8:43 AM
Hi - I just got a Sangoma A200 card with a single 2FXO module and
what appears to be an empty module. I put the card in my Dell GX260,
but the power light on the front
I have uploaded a working patch for version 1.2.12.1, and other that
seems to work in Trunk, but few people is reporting results, you can
help to get this into Asterisk, go here:
http://bugs.digium.com/view.php?id=5841
The patch I ported to 1.2.12.1 is working fine, I have tested in my
servers,
I'm using a mix of Cisco 7960, Linksys SPA-942, Cisco 7961, Cisco 7970
phones in a paging group. I have all the phones set up with an extra
line that auto answers the dial from my paging extension when the
primary line is not in use. All of these are operating correctly however
the 7961/7970s
Hi,
I'm a voip service provider and i'm setting up a asterisk box to
register around 100 lines from my central softswitch. This asterisk
box will be placed inside a customer and has a digium card to be
interconected with customer's pabx.
My problem is that when asterisk send register message,
By the way, Chester, please report results to the bug I sent you, is
very imortant the users feedback to get this into Asterisk
Regards
On 1/3/07, Moises Silva [EMAIL PROTECTED] wrote:
I have uploaded a working patch for version 1.2.12.1, and other that
seems to work in Trunk, but few people
Yeah, I've played with both an older one (FoneBridge 1) and a FoneBridge 2
unit, and
they seem to work as advertised. The FoneBridge 2 is much nicer, sets up much
faster
and boots much faster. Besides, RedFone is great company to work with.
Bill Burdick
Hello List
Does anybody have any
I have a phone number for traditional phone lines through stana phone and a
working trixbox server. What do I need to do to connect the two so when
someone calls the number from a normal phone, they get my server?
Thanks!
___
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Hello list,
one of our clients is going to be deploying a system with over 200
differently composed queues and 100 agents. We are going to do a full test
of the viability of this solution before deployment, but I was wondering
if anyone has experience of such a setup and if there are any
I used these directions to get Stanaphone working on my FreePBX box:
http://www.voip-info.org/wiki/view/Asterisk%40Home+Handbook+Wiki+Chapter+6#614Stanaphone
Alex
On 1/3/07, blackwater dev [EMAIL PROTECTED] wrote:
I have a phone number for traditional phone lines through stana phone and
a
Yeah, get a Business Process specialist to analyze the client's
environment and develop a better solution. 200 queues with only 100
agents sounds pretty ludicrous to me!
On Wed, 2007-01-03 at 14:22 -0600, lenz wrote:
Hello list,
one of our clients is going to be deploying a system with over
From: blackwater dev [EMAIL PROTECTED]
I have a phone number for traditional phone lines through stana phone and a
working trixbox server. What do I need to do to connect the two so when
someone calls the number from a normal phone, they get my server?
Thanks!
Get a cheap X100P or a clone
The IP600 is 12v!!! I fried a 600 when I used power adapter from 601.
On 1/3/07, Alvin Austin [EMAIL PROTECTED] wrote:
FWIW, our Polycom IP601 phones use a transformer with output: 24VDC
500mA (center contact is positive).
A Polycom reseller (or Polycom sales) could probably give you
Peter,
I have 600's that are 12V 1.5A, + in the center. This differs from some
of the other answers, maybe those differences are regional (although
that would seem rather silly).
HTH
B
Peder @ NetworkOblivion wrote:
Does anybody happen to know the input power specs for the Polycom IP 500
And probably wont be as Steve Underwood explained to me that he is now
supporting openpbx and has stopped support for unicall on asterisk 1.4
Can anybody at digium confirm? Is unicall going to be left out of 1.4?
|-Original Message-
|From: [EMAIL PROTECTED] [mailto:asterisk-users-
I haven't read trough the thread well enough. The 600 is 12V 1.5A
indeed. Too bad they don't all have the same voltage.
LST wrote:
The IP600 is 12v!!! I fried a 600 when I used power adapter from 601.
On 1/3/07, *Alvin Austin* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote:
FWIW,
I set aside a couple of channels and removed echo cancellation on them. So
far, faxing outbound through an ATA is working fine now.
Bill
-Original Message-
From: [EMAIL PROTECTED] on behalf of Marco Mouta
Sent: Wed 1/3/2007 6:42 AM
To: Asterisk Users Mailing List - Non-Commercial
Not necessarily... The same agents could very well be providing support for
multiple companies. You wouldn't want an announcement from company A in
company B's queues.
Alex
On 1/3/07, Joe Dennick [EMAIL PROTECTED] wrote:
Yeah, get a Business Process specialist to analyze the client's
You just specify the users email address in the voicemail.conf file,
along with their mailbox number:
see the sample file:
[default]
; Define maximum number of messages per folder for a particular context.
;maxmsg=50
2506 = 2506,Grandstream,[EMAIL PROTECTED],,attach=yes|imapuser=fbeck
1234 =
I stand corrected, but it still seems excessive.
On Wed, 2007-01-03 at 15:06 -0600, Alex Robar wrote:
Not necessarily... The same agents could very well be providing
support for multiple companies. You wouldn't want an announcement from
company A in company B's queues.
Alex
On 1/3/07,
In my case it was never any confusion over what needs to be configured in
asterisk. I was wondering what mail program asterisk used and what needed to
be configured with it. In my case I had to set up sendmail on my system to
relay through our internal mail server.
sendmail.mc was the file I had
I am trying to use ARI for call monitoring. Recording conversations and
such. The problem is that I don't use AMP, and don't have any sort of a
database for CDR setup. It is all stored in the CSV file by default. When I
setup ARI I tell it to go into standalone mode, and I set the asterisk
Greetings list,
Does anyone know if there's a maintained 1.4 ebuild for Gentoo? Even with
the ~amd64 keyword, latest in the official Portage repository is 1.2.13.
Thanks in advance.
Regards,
Chris
--
C.M. Bagnall, Director, Minotaur I.T. Limited
This email is made from 100% recycled electrons
Kenneth Padgett wrote:
Bob,
It looks like the gnutls development package is called gnutls-devel:
'yum install gnutls-devel' should get the package installed.
Yah, I thought that would be it. I have that installed, as well as
gnutls. (I basically installed both packages you can find with
Any screenshots available?
I do not want to even test this without having any idea what it is or how it
works.
The brief description on sf.net is not enough.
--
--
Steven
http://www.glimasoutheast.org
Dan Austin [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED]
We've been holding
On Wed, 2007-01-03 at 16:55 -0500, Steven wrote:
Any screenshots available?
I do not want to even test this without having any idea what it is or how it
works.
The brief description on sf.net is not enough.
I'm testing the 2.0 version on asterisk 1.2 . What do you want to know
about
Thanks - that turned out to be the problem. Well- one of those
solutions. I removed the blank and swapped the FXO module to the
other port. I don't know if it was a bad port on the A200, but since
I don't plan on using it, I won't worry about it- just regret it in a
year when I get a
Any easy way to determine if IP connectivity before attempting a SIP call?
IP connectivity could be a timeout.
Yuan Liu
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Anyone seen this? It ocurred on a 'reload app_queue.so' command.
Asterisk version is 1.2.9.1.
Tried again, but it was not immediately reproducable.
Doug.
(gdb) bt
#0 reload_queues () at app_queue.c:3339
#1 0xb778a7a8 in reload () at app_queue.c:4012
#2 0x0805bb44 in ast_module_reload
From: Gregory Machin [EMAIL PROTECTED]
Hi I have been asked to ind out if there is a way to use asterisk to
answere a voice fax modem so it can provide an answering service and
record messages ?
Absolutely - if that MODEM happens to be an X100P clone - such as my
ENF656-PCIG-MOPR. There are
Thanks - that turned out to be the problem. Well- one of those solutions.
I removed the blank and swapped the FXO module to the other port. I don't
know if it was a bad port on the A200, but since I don't plan on using it, I
won't worry about it- just regret it in a year when I get a second FXO
Steven wrote:
Any screenshots available?
Sorry, not yet. I've been meaning to get the tools
together to capture some images, but coding, QA and
paid work have taken priority...
I do not want to even test this without having any
idea what it is or how it works.
It is a suite of tools to
So just when I thought I had caller ID going fine things seem to have
taken a turn for the worst. I'm now seeing lots of misses in picking up
the caller ID on a line I know provides it.
I know I've changed the TDM400 card and upgraded to the latest (1.2)
version of Zaptel but could this be
I have an upcoming install which places the switch close to some
employees in a quiet work environment. Can anyone recommend a quiet 24
port POE switch? The Linksys SRW224P behind me right now would be
objectionable, I'm sure.
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With the chanisavail command.
PaulH
On Wed, 2007-01-03 at 14:22 -0800, Yuan LIU wrote:
Any easy way to determine if IP connectivity before attempting a SIP call?
IP connectivity could be a timeout.
Yuan Liu
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lenz wrote:
Hello list,
one of our clients is going to be deploying a system with over 200
differently composed queues and 100 agents. We are going to do a full
test of the viability of this solution before deployment, but I was
wondering if anyone has experience of such a setup and if there
On 1/3/07, John French [EMAIL PROTECTED] wrote:
I have an upcoming install which places the switch close to some
employees in a quiet work environment. Can anyone recommend a quiet 24 port
POE switch?
The 8port Netgear switch on my desk doesn't have any fans. FS108p. Not sure
if they make
On 1/3/07, Anton Krall [EMAIL PROTECTED] wrote:
And probably wont be as Steve Underwood explained to me that he is now
supporting openpbx and has stopped support for unicall on asterisk 1.4
Can anybody at digium confirm? Is unicall going to be left out of 1.4?
This has nothing to do with
I suspect any 24port will have a fan. The Netgear FSM7326P are not
too bad and we have had good luck with them.
ps - I also load their open source software.
On Jan 3, 2007, at 4:51 PM, John French wrote:
I have an upcoming install which places the switch close to some
employees in a quiet
Older models, 500 and 600, are 12V, newer 601s are 24v
--
Chris Mason
(264) 497-5670 Fax: (264) 497-8463
Int: (305) 704-7249 Fax: (815)301-9759 UK 44.207.183.0271
Cell: 264-235-5670
Yahoo IM: [EMAIL PROTECTED]
--
This message has been scanned for viruses and
dangerous content by
The easiest way is thru using contexts.
On 1/3/07, Mattias Andersson [EMAIL PROTECTED] wrote:
Hi all!
I am shore someone have writing about it bout I cant find it.
I have a extension that I need to block from making expansive mobil calls.
Everyone else should be aloud to do the calls.
I am
i'm building asterisk v140 on osx 10.4.8.
openssl is installed in /usr/local/ssl,
which openssl
/usr/local/ssl/bin/openssl
openssl version
OpenSSL 0.9.8d 28 Sep 2006
asterisk is config'd with,
% ./configure \
While preping screenshots I found that a stupid little bug
had slipped past my QA, relating to CDR views. I've fixed it
and regenerated the tgz file and replaced the broken one
on SF.
If you have downloaded 3.0.0 today, please get a fresh copy.
The bug was small enough and I think I caught it
John French wrote:
I have an upcoming install which places the switch close to some
employees in a quiet work environment. Can anyone recommend a quiet
24 port POE switch? The Linksys SRW224P behind me right now would be
objectionable, I'm sure.
You will need a fanless switch like the
On 1/2/07, Kevin P. Fleming [EMAIL PROTECTED] wrote:
DiegoF wrote:
chan_vpb.o:chan_vpb.cc:(.text+0x4da6): first defined here
/usr/bin/ld: Warning: size of symbol `load_module' changed from 3274 in
chan_vpb.o to 3926 in chan_vpb.oo
collect2: ld devolvi el estado de salida 1
make[1]: ***
Am Mittwoch, den 03.01.2007, 16:51 -0600 schrieb John French:
I have an upcoming install which places the switch close to some
employees in a quiet work environment. Can anyone recommend a quiet
24 port POE switch? The Linksys SRW224P behind me right now would be
objectionable, I'm sure.
I
I was just trying astman with the latest svn trunk from Dec 31. It
connects, but if I attempt to make a call, asterisk segfaults, but
in pthread_kill in /lib/tls/libpthread.so not in the asterisk code.
Is this something others have seen? This is with glibc-2.3.4-2
I just upgraded to 2.3.6 (the
I'm using the latest svn trunk code. The app_cbmysql builds, installs, and
loads just fine. But I don't get any CBMysql application within asterisk.
The only message I get as the module loads is something about finding the
configuration file (successfully).
The database tables are created,
Moises
this sounds great!
three questions if you don't mind:
1. how is this fitting into 1.6?
2. are there some directions I can follow for downloading the right
source and applying your patches?
3. is there a central place for doc on your patches? (if not I would
be more than happy to write
does anyone know if ztdummy is requires under 1.6 or are they using
Linux' rtc?
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--Bandwidth and
Rob wrote:
I'm using the latest svn trunk code. The app_cbmysql builds,
installs, and loads just fine. But I don't get any CBMysql
application within asterisk. The only message I get as the
module loads is something about finding the configuration file
(successfully).
What version of
On Tue, Jan 02, 2007 at 03:17:35PM -0700, [EMAIL PROTECTED] wrote:
Has anyone made this combination work together? I've tried everything
and can't seem to get it work right. It all compiles fine, but when
rxfax is called, I get an unknown symbol error. From my reading,
everything points
Hi,
I am using these model from HP ProCurve
http://www.hp.com/rnd/products/switches/switch2600series/features.htm?jumpid=reg_R1002_USEN
http://www.hp.com/rnd/products/switches/ProCurve_Switch_3500yl-5400zl_Series/features.htm?jumpid=reg_R1002_USEN
Regards,
Angel
1. how is this fitting into 1.6?
1.6? do you mean 1.4? AFAIK the most advanced Asterisk development goes in 1.4
2. are there some directions I can follow for downloading the right
source and applying your patches?
Nope, but is not hard at all. All the patches include the version in
its name.
how is this fitting into 1.4?
- can it be compiled against 1.4 or only 1.2?
- if not, are there leanings in that direction?
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