Greetings to all,
I am trying to compile 1.2 rc2 on a SUSE 9.3 box, and the compile fails with
this error:
make[1]: Entering directory `/usr/src/asterisk-1.2.0-rc2/channels'
gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes
-Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT
Can anyone tell me how to recover an Asterisk
password if it is forgotten or do I have to do a complete
re-install.
Kris
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Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
Update:
Here is the full error:
make[1]: Entering directory `/usr/src/asterisk-1.2.0-rc2/channels'
gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes
-Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE
-O6 -march=i686 -DZAPTEL_OPTIMIZATIONS
Hi!
While the Asterisk feature list (http://www.asterisk.org/features) and
many other articles in the web state asterisk has a predictive dialer, I
cannot find any information about it.
It seems, I have to use a third party product like GnuDialer or VICIDAL
(of astGUIclient).
Can anyone confirm
kchase wrote:
Can anyone tell me how to recover an Asterisk password if it is
forgotten or do I have to do a complete re-install.
Which password?
Do you have the SSH password for the [EMAIL PROTECTED] server?
David
Kris
Hi,
Is there a way so that I can record the voice messages in mp3 format
instead of wav? I think it is much smaller in size compare to wav. It
is also easier to send small sized file as an attachment. Currently
when my users record voice messages the format is wav. Where can I
configure it
You'll find the GSM codec renders smaller filesizes.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ryan Pagquil
Sent: Wednesday, November 16, 2005 12:19 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Recording voice messages in mp3
Hi
I have octo bri card connected to 4 telco lines and 4 alcatel PBX lines.
After few hours of calling i get message
Ring requested on unconfigured channel 255/255 span 1
this message occurs immediate when call get to asterisk and it is
denied immediately.
If i restart asterisk it is working
FXO ports are for dialling outunless you use them for dialling in.
PaulH
- Original Message -
From: Dulmandakh Sukhbaatar [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, November 16, 2005 5:57 PM
My Asterisk box is installed in the DMZ of an IPCop firewall.
The RED interface of IPCop has a static public IP address, and all traffic
directed to that address is forwarded to the PBX in the DMZ.
The IPCop also routes traffic from LAN (192.168.2.0) to DMZ (172.16.0.0),
so Asterisk is reachable
On Tue, 2005-11-15 at 16:00 -0500, Jerry Geis wrote:
I found the chan_bluetooth app and downloaded it but it does not
compile and have not been successful in reaching the author.
There's a version which should work in my CVS tree:
cvs -d :pserver:[EMAIL PROTECTED]:/home/cvs co chan_bluetooth
10 Million is ca 10,000 boards/licenses if you assume 1000 USD in
average. I know the Digium boards are some of the cheapest around, but
the actual production cost should be below 150 USD for a 4xE1/T1 of this
type, meaning that they still should have very decent margins.
jan
[EMAIL
The question is really un-important for this list, it is ONLY
important
between the person who thinks that they can use the g729
codec ignoring
the patent or considering that it is not legally enforcable
for them and
their lawyer who will give them concise information about the legal
[EMAIL PROTECTED] wrote:
FXO ports are for dialling outunless you use them for dialling in.
PaulH
- Original Message -
From: Dulmandakh Sukhbaatar [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday,
Are you using wav or wav49? You can check in
/etc/asterisk/voicemail.conf under the format option... wav49 creates
much smaller files than normal wav and doesn't need a special player
like gsm files would and as far as using mp3, I'm not sure how to go
about that.
-Gerard
Ryan Pagquil wrote:
On Tue, 2005-11-15 at 15:06 -0800, trixter aka Bret McDanel wrote:
on automatic proximity detection, no BT device it forwards to your
mobile. The BT headset can be used as your 'soft phone' (more like
using chan_oss than a soft phone though).
It's not _quite_ as bad as chan_oss; you do
On Wed, 2005-11-16 at 08:45 +, David Woodhouse wrote:
On Tue, 2005-11-15 at 15:06 -0800, trixter aka Bret McDanel wrote:
on automatic proximity detection, no BT device it forwards to your
mobile. The BT headset can be used as your 'soft phone' (more like
using chan_oss than a soft
how can you play .gsm files what program can you use both in windows and linux system?On 11/16/05, Gerard Dupont III
[EMAIL PROTECTED] wrote:Are you using wav or wav49? You can check in/etc/asterisk/voicemail.conf under the format option... wav49 creates
much smaller files than normal wav and
AFAIK there's no working predictive dialer embedded in asterisk. We are
trying to implement VICIDIAL[1] for our predictive dialer. Anyone have
a working server for VICIDIAL? or even GnuDialer[2].
[1] astguiclient.sf.net[2] gnudialer.org
On 11/16/05, Marcus Deluigi (intern) [EMAIL PROTECTED]
Hi,
I have
set-up two asterisk servers with an IAX trunk between them. There is a
queue-system and callagents configured on one of them
Agents on
both servers logon to the one queuesystemI have set up, which works fine.
But autologoff (agents.conf) only seems to work with agents connected
the password to login into asterisk
- Original Message -
From: David Uzzell [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, November 16, 2005 3:11 AM
Subject: Re: [Asterisk-Users] Asterisk @ Home password
Kevin P. Fleming wrote:
trixter aka Bret McDanel wrote:
Now for the ITUs site, I found the data set that has the '3
diskettes' (their words) in the zip file (pdf or word) however that is
82 swiss francs, does anyone have a copy that doesnt require such
payment (legally of course)?
The
Well I normally do this for lost passwords on linux boxen
Boot from your favourite rescue cd
mkdir /target
mount /dev/hdXX /target ;where XX is eg a1 a2 b3 etc...
chroot /target passwd
type a new root passwd
now you have a root passwd reboot the system and use passwd to change
whatever
I thought I'd just document a problem I've been having in case anyone
else comes across it.
I have an AVM-C2 card which I'm using with chan_capi. capiinit fails to
load the driver when run under 2.4 or 2.6 kernels on AMD Sempron
processors with 64 bit cores, and at least some 64 bit Athlons,
On Wed, 2005-11-16 at 05:51 -0500, Paul wrote:
TU is not involved in licensing or patent indemnification; they
are non-profit standards body. The G.729 patent holders have given
Sipro the task of managing their patent portfolio licensing, so that
is who you would need to contact. Sipro
kchase wrote:
the password to login into asterisk
If you still have the SSH password you can log into the box and change
the maint password.
[EMAIL PROTECTED] ~]# help-aah
[EMAIL PROTECTED] - HELP
CommandsDescriptions
Allison Smith wrote:
Love you all! You're all perfect lambs!
Allison
Allison, you are a star!
How about tossing in a few extra recordings with the next batch:
Go to sleep now, my poor tired baby. I will be here for you tomorrow.
Rise and shine you sweet wonderful geek
[EMAIL PROTECTED] wrote:
10 Million is ca 10,000 boards/licenses if you assume 1000 USD in
average. I know the Digium boards are some of the cheapest around, but
the actual production cost should be below 150 USD for a 4xE1/T1 of
this type, meaning that they still should have very decent
On Tue, 2005-11-15 at 23:11 +, Julian Lyndon-Smith wrote:
Done. Done. Done :)
http://bugs.digium.com/view.php?id=5762
Julian.
In case anyone is interested, this has worked like this for at least 18
months, I had the same problem, so I just changed to using Monitor
before dropping the
On Wed, 2005-11-16 at 05:59 -0500, Paul wrote:
Allison Smith wrote:
Love you all! You're all perfect lambs!
Allison
Allison, you are a star!
How about tossing in a few extra recordings with the next batch:
Go to sleep now, my poor tired baby. I will be here for you
Bugger. I had hoped that it was a recent bug ..
As a matter of interest, how do you know which agents have answered the
call ? I liked having the agent number as part of the monitored filename.
Julian.
Adam Goryachev wrote:
On Tue, 2005-11-15 at 23:11 +, Julian Lyndon-Smith wrote:
On Tue, 2005-11-15 at 09:35 -0800, trixter aka Bret McDanel wrote:
On Tue, 2005-11-15 at 11:56 -0500, Jason Pyeron wrote:
a unicode document comes in two flavors UTF8 and UCS2 in windows UTF8
may work, but UCS2 cannot work, as it is 2 bytes per character.
UTF8 will not work if wordpad
Francesco Angi ha scritto:
Two
simple questions about Cisco 7905 on Asterisk using chan_sccp.
1) using both sccp firmware 5.0 and 6.1 I
cannot put a call in hold,
because there's no Hold Button at all! Is
there a way to configure
The 7905 has
an hard button for the hold stuff, the
Chris Bagnall wrote:
The question is really un-important for this list, it is ONLY
important
between the person who thinks that they can use the g729
codec ignoring
the patent or considering that it is not legally enforcable
for them and
their lawyer who will give them concise information
On Wed, 2005-11-16 at 11:26 +, Julian Lyndon-Smith wrote:
Bugger. I had hoped that it was a recent bug ..
As a matter of interest, how do you know which agents have answered the
call ? I liked having the agent number as part of the monitored filename.
Julian.
Well, in my case, there
Hi all,
I have 1 a-z carrier i want to forward all calls to
that carrier, can any one hint me where i should add
this carrier information?
I will be appricate if any one give me direction way?
Yours,
Abdul Lateef
Computer Programmer
HATIF COM
Mob: +974 - 5405022
Tel: +974 - 4883068
Hi all!
I'm trying to play some music from asterisk, and when I call to the PBX
from a GSM mobile phone, the more I speak while hearing the music, the
worst is the quality of the music I hear... My audio is at 8Khz,
16bits/sample.
I've tried different codecs for asterisk, but results are the
Hi all,
In case you have a number of trunks there is a software named
astbill (www.astbill.com) in which you can configure the trunks and decide
their costs and it will automatically choose the most suitable trunk.
Thx
MAG
Pikoro wrote:
By "trunk" I mean each trunk is a different account
on
There are actually 8 add-on predictive dialer solutions listed on the wiki:
http://www.voip-info.org/wiki/view/Predictive+dialer
GnuDialer and VICIDIAL are the only two that are fully GPL.
As for an integrated predictive dialer in Asterisk, I can assure you
that the core developers of Asterisk
Hello,
I am a newbie here. Want to learn the source code of Asterisk. Can anybody give me some clues for how to learn?
Thanks beforehand!
Best Regards
___
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Asterisk-Users mailing list
Hello,
I am a newbie here, wants to learn the source code of Asterisk. Can anybody give me some hints? Thanks beforehand!
Best Regards
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
Hello,
I'm looking for some solution how to use Asterisk as a Fax
server. I have an ISDN line terminated on Patton SmartNode 1400. SN1400
is then connected to the LAN with Asterisk PBX using SIP.
I want to make Asterisk send faxes to/receive faxes from PSTN
via Patton.
As
This has come up a bunch of times on this list..
Take a look at http://www.voip-info.org/wiki-Asterisk+sound+files
Hope that helps
-Gerard
Mark Quitoriano wrote:
how can you play .gsm files what program can you use both in windows and
linux system?
I am monitoring via my queues.conf.
[310]
wrapuptime=30
timeout=15
strategy=ringall
retry=5
queue-youarenext=
queue-thereare=
queue-thankyou=custom/aa_6
queue-callswaiting=
music=Support
monitor-join=yes
monitor-format=gsm
maxlen=0
leavewhenempty=no
joinempty=no
context=aa_6
announce-holdtime=no
hi all,
I just configured a junghanns card like described on beronet. (using
mISDN) All seems to work, but I'm more concerned about the 'differences'
in echo cancellation.
If one uses the junghanns drivers, you're stuck with asterisk1.0.9 at
the moment; using chan_mISDN I can use asterisk
Hi everyone!
Okay. I was reading on the voip-info.org about FXO and FXS. Is it
possible just to get a card with FXO and FXS together? I know Digium
sells them, but as I've said, I'm looking to spend too much.
Thanks for everyone's input!
Logan.
___
On Wed, 2005-11-16 at 19:31 +1300, Matt Riddell wrote:
Sergey Okhapkin wrote:
Already supported (simple patch exists).
http://bugs.digium.com/view.php?id=5374
Which?
silence-suppression-2.diff
It's already supported in Asterisk or you can patch Asterisk to add it?
You can patch
Hi,
the best thing to do is get a Sipura 3000 that has 1 FXO and 1 FXS port.
You won't need to bother with IRQs and echo problems that at least here
in UK we have with FXO cards.
Yiannis
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Logan
Sent:
How can I set up a group of outbound trunks which will rotate use
dependant on how many outbound calls need to be made.
You could do this by writing a simple (but laboriously long) macro to
try the accounts in order, dialing via the first available one.
There would be a dial() command followed
Hello Abdul,
Yes,
Using a port forwarding
in a files config oh323.conf change port 1720 for your new port
2005/11/15, Abdul Lateef [EMAIL PROTECTED]:
Hello Reli,
If i am going to install chan_h323 with different port
instead of 1719 and 1720, is it will work? Becuase
already i have MVTS
Dear All,
I'm in the process of writing a voice recording application for a
customer using an Inter-Tel PBX.
I was planning on using the T1 interface on the Inter-Tel to route calls
to an Asterisk server that would do the voice recording. I was going to
use the Caller ID or DNIS to help with
Hello,
I have a problem for asterisk-oh323.
when I apelle with a phone sip and that the termination and in h323.
I have this error message
cleared, reason 24 (Call ended with Q.931 cause [28 - Invalid number
format])
I using the latest version for asterisk, openh323 and pwlib.
my config is.
Allison Smith wrote:
Love you all! You're all perfect lambs!
Allison
Allison, you are a star!
How about tossing in a few extra recordings with the next batch:
Go to sleep now, my poor tired baby. I will be here for you tomorrow.
Rise and shine you sweet wonderful geek
I
I don't understand exactly what you're telling me, but I'm currently using
TxFAX with an already generated TIFF file to send a fax to another machine
that uses RxFAX to receive it. RxFAX and TxFAX are not running on the same
machine. The only strange thing is that I'm using a SIP connection
Hello gurus,
When asterisk is reloaded it displays single line
output with below notice.
What does it mean?How can I get rid of
it?
Nov 16 16:48:14 NOTICE[2654]: indications.c:397
ast_unregister_indication_country: Removed default indication country
'za'
Thank you in advance for
On 11/16/05, Marcus Deluigi (intern) [EMAIL PROTECTED] wrote:
A stupid question, but is it possible to use the PauseQueueMember
function with AgendLogin?
Whenever I use AgendLogin, I have a connection and the agend cannot dial
another extension to pause.
Creating a new line does not help,
On 11/16/05, Joseph Rothstein [EMAIL PROTECTED] wrote:
Update:
Here is the full error:
make[1]: Entering directory `/usr/src/asterisk-1.2.0-rc2/channels'
gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes
-Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE
Klaus Darilion wrote:
Hi!
I read in the archive a lot of problems using the Dell 1850 servers
and digium TE4xxP and TE2xxP hardware. I wonder if ever anybody has
tried the Dell Poweredge 850 series and can report some experiences?
btw: Does somebody knows why there are problems with 1850
[EMAIL PROTECTED] wrote:
Here's a question: why are you building your hobby box? To gain
practical experience? Then forget the X100: it's like learning Windows
NT: yeah, the information might be somewhat valid today, but it's way
obselete. The X100 is dead, and very unlamented. If
FXO ports are an interface between your system and a phone carrier. FXS
ports are an interface between your system and a phone station (or
handset).
You can send outbound calls on an FXO port as well as receive them.
To dial on the TDM card you would dial : DIAL(Zap/g1/${number}) or
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Russ Price
Sent: Wednesday, November 16, 2005 2:11 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk hobby box
Funny you would say that - I have a box running with a pair
Juan Jose Comellas wrote:
I don't understand exactly what you're telling me, but I'm currently using
TxFAX with an already generated TIFF file to send a fax to another machine
that uses RxFAX to receive it. RxFAX and TxFAX are not running on the same
machine. The only strange thing is that
Logan, I have a hobby box set up running asterisk at home with two winmodems, not digium clones, and it works just fine. As long as debian has drivers for it there shouldn't be any problem. Granted I have only set this up in at home. As for FXS cards the best way to go there is to either use a
On Tue, 2005-11-15 at 16:00 -0500, Jerry Geis wrote:
/ I found the chan_bluetooth app and downloaded it but it does not
// compile and have not been successful in reaching the author.
/
There's a version which should work in my CVS tree:
cvs -d :pserver:anoncvs at cvs.infradead.org
Steven Ringwald wrote:
I apologize if this question has been asked before. Did something
change with the behaviour of the 'sip show inuse' command between
1.0.9 and 1.2-rc1? I used to be able to see a list of extensions and
the number of in/out calls. Now it just reports:
asterisk*CLI sip
Hi all,
Can i find somewhere a list of tested motherboards or server that works
fine with Asterisk and Digium boards? Digium has a page that mention
some models that are already known don't work fine with the Digium
cards, but i am looking for something more updated. Any clue?
Thanks
George
I'm not sure I
follow. As I understand it, app_rxfax and app_txfax are designed to
transfer fax data, over any asterisk channel. Since the fax is converted
to audio over the channel, and the channels are IP, isn't this fax over
VOIP?
I am successfully
using app_rxfax to receive
On Wed, 2005-11-16 at 00:56 -0800, trixter aka Bret McDanel wrote:
It's not _quite_ as bad as chan_oss; you do generally have at least
No but its more like that than a softphone in terms of how it accesses
asterisk. Which was what I originally said :)
True.
--
dwmw2
I'm using the 850 series. Works well. Only major problem is having to use
a third party PCI-e sata raid controller, well thats if you want HW raid in
your system.
Craig
- Original Message -
From: Kevin Hanson [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial
LIU.ANDY wrote:
Hello,
I am a newbie here. Want to learn the source code of Asterisk. Can
anybody give me some clues for how to learn?
The under the hood docs are still up:
http://zapteldoc.blogspot.com/
--
Cheers,
Matt Riddell
___
Paul wrote:
You assume that all sales are direct to end users with highest prices
and margins. Most companies increase volume by selling quantity at
discounts to resellers and distributors.
Also, maybe you assume they pay extremely low wages.
I've seen Mark scrounging around rubbish bins
BJ Weschke wrote:
On 11/16/05, Marcus Deluigi (intern) [EMAIL PROTECTED] wrote:
A stupid question, but is it possible to use the PauseQueueMember
function with AgendLogin?
Whenever I use AgendLogin, I have a connection and the agend cannot dial
another extension to pause.
Creating a new line
Technical Support wrote:
I'm not sure I follow. As I understand it, app_rxfax and app_txfax
are designed to transfer fax data, over any asterisk channel. Since
the fax is converted to audio over the channel, and the channels are
IP, isn't this fax over VOIP?
On 11/16/05, Matt Riddell [EMAIL PROTECTED] wrote:
BJ Weschke wrote:
On 11/16/05, Marcus Deluigi (intern) [EMAIL PROTECTED] wrote:
A stupid question, but is it possible to use the PauseQueueMember
function with AgendLogin?
Whenever I use AgendLogin, I have a connection and the agend
Hi all,
I have Asterisk 1.2rc2, TE205P and spandsp 0.0.2pre21. Everything work
fine but when there is a lot of traffic (more than 40-50 channels
active) i receive or send broken faxes (the pages are cut short). Is
that because of frame slip problem? The PCI slot the the card is on
always
Kevin P. Fleming wrote:
Michael Welter wrote:
Nov 15 20:09:15 NOTICE[27290]: chan_zap.c:7395 pri_dchannel: PRI got
event: HDLC Abort (6) on Primary D-channel of span 1
I ran top during that time, and there was no significant cpu usage.
Probably interrupt starvation... are there any
Hello Kevin ,
On Wed, 16 Nov 2005, Kevin Hanson wrote:
Steven Ringwald wrote:
I apologize if this question has been asked before. Did something change
with the behaviour of the 'sip show inuse' command between 1.0.9 and
1.2-rc1? I used to be able to see a list of extensions and the
Joseph Rothstein wrote:
make[1]: *** Deleting file `chan_sip.o'
make[1]: *** [chan_sip.o] Interrupt
make: *** [subdirs] Interrupt
'Interrupt' means you pressed Ctrl-C or killed the compile process in
some other way.
___
--Bandwidth and Colocation
Paul wrote:
Kevin, I gave those policy changes a quick read a few months ago. I get
the impression that Digium has a definite grandfathered advantage. It
looks to me like they don't want such general purpose licensing in the
future.
That is correct, it appears that Sipro wants to directly
On Tue, 15 Nov 2005 16:00:46 -0500, Jerry Geis [EMAIL PROTECTED] wrote:
Anyone doing anything with bluetooth headsets?
Both DIAX and iaxComm will use the bluetooth headset for audio.
I think that DIAX has some additional support for the hook button, but I'm not
sure.
I'd love to add such a
Rise and shine you sweet wonderful geek
That prompt would be perfect for a wake-up call.
___
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Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
Steve, is there a new txfax / rxfax to try and compile with 1.2rc2? 0.0.2c
is compiling but at load time it gives out resources problems and crashes
asterisk. This has been tested in 3 systems with no luck so far..
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED]
- Original Message -
From: Sergio Chersovani [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, November 15, 2005 2:25 PM
Subject: Re: [Asterisk-Users] Cisco 7960 Multiple Line Appearance
Matt Hoskins ha
I have the TE210p card, with two distinct PRI lines connected to each
of the spans. For each span, there are 23 B channels, numbered 1-23,
and one D channel, numbered 24. One set of 23 channels is already set
up, in group 1 (channels = 1-23 in zapata.conf). How do set up a
different group with
spandsp-0.0.2pre21c does work with 1.2rc2. If you are having problems
you will need to be more specific about the errors you get.
Steve
Anton Krall wrote:
Steve, is there a new txfax / rxfax to try and compile with 1.2rc2? 0.0.2c
is compiling but at load time it gives out resources problems
As I have not seen my post show up in my mail-client, I am resending it to
be sure it arrives on the list. My apologies if it is a double-post!...
I am trying to set up the following:
Asterisk Server (1.0.9-stable) with BriStuff (0.2.0.RC8j) and florz
From my config on a 4 port card...
;PRI on channel 1 of DS3 #2
group=1
callgroup=1
pickupgroup=1
channel = 1-23
;
;PRI on channel 3 of DS3 #2
group=2
echocancel=no
callgroup=2
pickupgroup=2
channel = 25-47
Use Dial(zap/g1/xx) or Dial(zap/g2/xx) to pick a PRI
good Luck
On
On Tue, 2005-11-15 at 19:37 -0600, Kevin P. Fleming wrote:
Kenneth Shaw wrote:
Currently I have automon enabled, but I have absolutely no idea how to
get it to work. I have the latest CVS HEAD release.
This was just talked about in Mantis... set DYNAMIC_FEATURES in the
dialplan to include
Hi,
I have been reading your comments in the digium Asterisk
users list regarding HA of asterisk servers.
I am looking into doing the same with my asterisk boxes but
am coming up against the same issues as you. I would be greatful if you could inform
me if you came to some conclusion
1.2rc2 and spandsp 0.0.2pre21c compile perfectly but when I load asterisk I
get this:
[Nov 12 10:14:16] [app_rxfax.so][Nov 12 10:14:16] WARNING[12188]:
loader.c:325 __load_resource: /usr/lib/asterisk/modules/app_rxfax.so:
undefined symbol: fax_set_phase_d_handler [Nov 12 10:14:16]
hi,
I have installed 1 TE110P on a decent computer and generate 30 calls
towards it. Asterisk do answer, but it uses a heck of a long time doing
so. It takes ca 5 sec before the first call is answered and when ca 12
before all is answered. Is there any reason for this? Config etc. on my
Anton Krall wrote:
1.2rc2 and spandsp 0.0.2pre21c compile perfectly but when I load asterisk I
get this:
[Nov 12 10:14:16] [app_rxfax.so][Nov 12 10:14:16] WARNING[12188]:
loader.c:325 __load_resource: /usr/lib/asterisk/modules/app_rxfax.so:
undefined symbol: fax_set_phase_d_handler [Nov 12
Thanks. I was just unsure about referring to the second span as
channels 25-47, since my provider refers to them as separate banks of
1-23. It seems that ztcfg agrees with you--it choked when I tried to
have two settings for 1-23, but came up OK when the new one was set to
25-47.
On 11/16/05,
Which digium card do you use? 1 port or 2/4 port E1/T1? or TDM?
klaus
Craig Guy wrote:
I'm using the 850 series. Works well. Only major problem is having to
use a third party PCI-e sata raid controller, well thats if you want HW
raid in your system.
Craig
- Original Message -
On 11/16/05 06:25 Carlos said the following:
Well I have 3 405p cards in one machine a p4 2.4 with a gig of ram. Running
good all 12x t1's are connected to channel banks.
are you able to sustain a fully loaded 12x24 channels on this box ? it does
seem that a P4 would be able to handle at
If I understand this thread correctly, they are discussing monitoring
the status of other agents. This cannot be done with the SIP firmware
on the CP-79x0 phones.
We are currently working on testing the SCCP firmware with the
revisions made to the chan-sccp module, but I am stuck on getting the
On Wed, 2005-11-16 at 08:50 -0800, Kenneth Shaw wrote:
On Tue, 2005-11-15 at 19:37 -0600, Kevin P. Fleming wrote:
Kenneth Shaw wrote:
Currently I have automon enabled, but I have absolutely no idea how to
get it to work. I have the latest CVS HEAD release.
This was just talked about
Hey dinesh,
Have never had that type of load on any given system they all go to fax
machines and ext's so the probabilty of everyone picking up there phone and
calling out at once is very low.
Carlos Alcantar
Race Technologies, Inc.
101 Haskins Way
South San Francisco, CA 94080
P: 650.246.8900
Kenneth Shaw wrote:
Ok, fair enough. But I have a couple more questions. What would the
Set(DYNAMIC_FEATURES= ... ) look like?
Set it to 1 to enable.
--
Cheers,
Matt Riddell
___
http://www.sineapps.com/news.php (Daily Asterisk News - html)
Kenneth Shaw wrote:
Ok, fair enough. But I have a couple more questions. What would the
Set(DYNAMIC_FEATURES= ... ) look like?
Hmmm or maybe its one of:
caller
callee
both
http://bugs.digium.com/file_download.php?file_id=7321type=bug
--
Cheers,
Matt Riddell
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