This evening I posted a new set of Digium G.729 codec modules to our FTP
server and web site, for Linux x86 and x86-64 processors. They were
built using GCC 4.0.1, and they now report the processor they were
optimized for when they are loaded.
The previous x86-64 module required a non-standard
I have an office up and running with 40 SIP handsets. Currently when an
incoming call is parked, they then dial ext 9876 which I have setup to
do a page (it does this through an agi script that uses the Polycom
autoanswer to page via the phones). What I would like to be able to do
is transfer
Paul,
the only stuff really differente here in Brazil is
the ring frequency. It's a 4-seconds-pause-1-second-25-Hz-ring. I had some
problems with telephone devices made in Brazil that did not recognize a 20Hz
ring (US standard). Also, the dialtone is a little differente (check asterisk's
i
Asterisk does all its DSP work in software, using the host's CPU(s).
Since the Cell Processor is likely a mediocre CPU but with a monstrous
stream processing capability, wouldn't it make sense to run very large
Asterisk installations on the Cell Processor? Or on dual Cell
Processors? The Signal p
Well, The Polycom & Cisco are high-end. I have
used others like a pingtel, sipura 841, etc. Nothing has the 'feel' of the
cisco, and nothing has the functionality of the polycom (like call drop from
conferences). Next I will try the aastra wireless combo phone for the
office. Looks nice
On Fri, Oct 07, 2005 at 09:45:53PM -0400, Paul wrote:
> Doug Meredith wrote:
> Also consider that there are situations where 100% open source is never
> allowed. Check out visa/mastercard processor certification for a good
> example. Digium dual licensing availability means I could actually stand
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir
Cohen
Sent: Friday, October 07, 2005 8:17 AM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Asterisk PBX in Debian
On Fri, Oct 07, 2005 at 07:37:26AM -0400, [EMAIL PROTECTED] wrote
Hi!
If you are interested in Comfort Noise Generation for Asterisk or would
like to experience the beauty of white noise, have a look at:
http://bugs.digium.com/view.php?id=5409
Comments and suggestions are welcome on Mantis.
Thanks!
Carlos
-- "We hold [...] that all men are created equal; that
Has anyone used the DS3 card from Sangoma with Asterisk?
I have read many posts from users that the Sangoma cards have better echo
canceling and so forth. I guess I am just wondering if there are more
benefits to using this brand.
I currently am responsible for multiple Asterisk servers all wit
On 7-Oct-05, at 9:45 PM, Paul wrote:
The thing to remember is that the digium folks are not going to
spend months slaving over a new hardware product and then put the
device driver source under a closed license only. The gpl code can
be used in an asterisk fork like openpbx or in somethin
Andy - The Sangoma echo can has a longer MS duration of tail
(buffer/compensation) than the Digium.
Cory Andrews
Senior Partner
+++
VOIPSupply.com
454 Sonwil Drive
Buffalo, NY 14225
+++
voice - 716.630.1555 X22
email - [EMAIL PROTECTED]
fax - 716.630.1548
Andy Goss wrote:
I
> I wouldn't think anyone would consider Sprint a dying company. They just
> acquired Nextel so they've got money to spend.
>
> Maybe as an ILEC (which they are here in Ohio) they are viewing Vonage
> and Voiceglo as a force that needs to be stopped to prevent further
> eroding of their POTS netw
hi:
When our user call cell phones, they are unable to
hear the cell phone user's music. anyone know how to fix this?
Best Regards
Matt
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Doug Meredith wrote:
gincantalupo <[EMAIL PROTECTED]> wrote:
why a fork???
I don't know any of the people involved, or what their motivation
might be, but I will make a guess:
Digium's model tends to stifle innovation. Look at eclipse.org for a
much better model. Eclipse is truly
On Sat, 8 Oct 2005 00:15, Michael Graves wrote:
Try "Strix" as the access points, It has cell handover and roaming features.
Very important for telephony wireless.
> Jim,
>
> I have one of the Hitachi WIP-5000 wifi phones. Been using it about 6
> months. I've even travelled with it and tried
Michael Coburn wrote:
Or why not create all sound files under /usr/share/asterisk/sounds and
then subdirs from there for your own touched files i.e.
/usr/share/asterisk/sounds/custom ?
--
Michael Coburn
Short answer is because we live by rules.
Any good linux distro follows standards regard
Hello,
Please, give me any advice how to speech a text file with festival.
This is for a simple IVR to give the user a response based on his
numeric ID.
Extensions.conf:
exten => 307,1,Festival(dial your numeric id)
exten => 307,2,DigitTimeout(4)
exten => 307,3,ResponseTimeout(9)
exten => 30
Can anyone explain this one?
Sometime when I make a call and the line is busy I get:
Oct 7 15:52:00 VERBOSE[20249] logger.c: -- Called g1/3235838
Oct 7 15:52:00 VERBOSE[20249] logger.c: -- Channel 0/1, span 1 got hangup
Oct 7 15:52:00 VERBOSE[20249] logger.c: -- Zap/1-1 is busy
(whi
Sangoma did announce a quad T1/E1 card with hardware echo-cancellation
a couple months ago. I believe they will start shipping it in the next
few weeks.
There should be more info on their website including contact information:
http://www.sangoma.com
MATT---
On 10/7/05, Matt <[EMAIL PROTECTED]> wr
I just bought a Pingtel Xpressa from VoipSupply for use with
Asterisk. I know that Pingtel has sold off their hardphone line and
discontinued support for their phones, but I'd like to track down a
few of the Java applications that they distributed before they went
away, specifically their
That is true. It's just one of those things that is easier to leave
alone to avoid breakage in upgrades. It would be nice to get fixed
though
Darren Wiebe
[EMAIL PROTECTED]
Eric Lyons wrote:
Looking at the code, it would appear that the 'callstart' column of
the cdrs table should reall
Looking at the code, it would appear that the 'callstart' column of the cdrs table should really be
called 'callend':
$dialstr = "IAX2/$res->{path}/$phone|30|HL(" . ($maxtime * 60 * 1000) .
":6:3)";
$res = $AGI->exec("DIAL $dialstr");
$answe
We are using sangoma quad card with 4 E1 hooked, no echo problems.
Best Regards
Matt
- Original Message -
From: "Andy Goss" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Friday, October 07, 2005 2:45 PM
Subject: [Asterisk-Users] hardware echo ca
Matt wrote:
Well this is all good in practice and I do have a /custom directory..
but, to my knowledge, there is no way to get things like the voicemail
module to read out of the /custom directory..
Not true... at least two people have already posted here saying that if
you set the language to
Hi,
Could some one give me help please, i have a Digium card TE405P and i want to use it with this serveur:" IBM eServer BladeCenter HS20 Xe 3.2 GHz 1 Go", does some one know if this two elements could be used together?
thks!!!
//Vador
___
--Bandwi
I'm not sure where the noise is coming from, but you can change the timing source in zaptel.conf
in zaptel.conf:
span=1,0,0,esf,b8zs --- Asterisk is using external timing source
span=1,1,0,esf,b8zs -Asterisk is providing timing to the channel bank
AK
On 10/7/05, Eddie <[EMAIL
I have been reassigned from my normal duties to figure out the asterisk
echo problems we have been experiencing. We currently use a TE110P card
(I think.) I know that the problem is the worst when calling from our
office to a residential analog line or a analog PBX. Occasionally the
problem will
At 16:39 10/7/2005, Obelix wrote:
>
>If you have configured Asterisk to remote to a SIP provider, how do you
verify
>that the registration has been successful?
sip show peers
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Asterisk-Users
If you have configured Asterisk to remote to a SIP provider, how do you verify
that the registration has been successful?
This message was sent using IMP, the Internet Messaging Program.
__
Thanks for the reply. Forgive me for being naïve, however have jumped in to
this asterisk project at work due to some circumstances beyond my control and I
don't know a lot about carriers and how this all works. I am figuring it out,
but it's a lot of trial by fire.
As far as I know, we onl
On Friday 07 October 2005 15:24, Troy Settle wrote:
> Licence changes can be made... look at Cistron Radius. They started
> with Livingston's code, which was under the BSD license. Once their
> code had been completely rewritten, they did an audit and found that
> they were no longer using the or
Hello,
I bought a copy of this software in the hope of bridging Skype into my
* box. It sort of works but the audio is all distorted and there's huge
latency. Does someone have this working well with their * server?
According to the wiki someone does, but I don't know who. I'm needing
advise on wh
On Fri, 2005-10-07 at 11:17 +0200, Kib Eki wrote:
> Hello,
>
> can anybody tell me where to get the latetest SIP Firmware 1.6.2 for the
> Polycom
> phones?
>
http://www.freedomphones.net/polycom/files/
--
Jesse Keating
GameHouse -- Systems Engineer
_
In article <[EMAIL PROTECTED]>,
Nathan C. Smith <[EMAIL PROTECTED]> wrote:
> How can they be a great loss if their ideas and work never made it into the
> codebase?
But a lot of it did
Cheers
Tony
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
> Sent: Frida
This one drove me crazy for a while too. I found out that some
companies don't exactly play fair and don't pass answer supervision on a
call until you are actually speaking with a live person. The person I
spoke to about this wasn't sure if that was even legal, but he said it
happens quite a
Andy Kuo wrote:
Hi Andrew,
I'm using a TE406P too, and I have "echocancel=yes" in zapata.conf.
Is this redundant? Should I take the line out?
Please advice.
No, if you don't put 'echocancel=yes' in zapata.conf, Asterisk will not
request echo cancellation on the channels. If it doesn't reques
Is there an easy (or even a hard) way to save to
the CDR a userfield value with the call's codec in it?
Chris Coulthurst
[EMAIL PROTECTED]
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On Fri, Oct 07, 2005 at 03:41:43PM -0400, Min Qiu wrote:
>
> Hi all,
>
> I just installed an TDM02B. My system is a dell pc with
> linux 2.6.12-1.1456_FC4
> asterisk-1.2.0-beta1
> zaptel-1.2.0-beta1
> libpri-1.2.0-beta1
>
> in /etc/zaptel.conf I have (all others are default):
> fxsks=3-4
At the Boston VON last month Linksys was showing an
add-on for their SPA-941 to make it wireless.
See:
http://www.voip-info.org/tiki-index.php?page=Linksys
- Original Message -
From:
Will Glass-Husain
To: asterisk-users@lists.digium.com
Sent: Friday, October
Does anyone have the lates firmware for the
AudioCodes MP-104 FXS?
If so, please send me a link or e-mail
directly.
Thanks
Michael
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I would appreciate seeing the scripts as well. Nice job!
Desktophero at gmail.com
Thank you
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Troy Swaine
Sent: Friday, October 07, 2005 12:42 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
S
Before I go to the trouble myself, has anyone else created a BBEdit
Language Module for the syntax in asterisk config files?
Regards,
--
Anthony Rodgers
Business Systems Analyst
District of North Vancouver
Web: http://www.dnv.org
RSS Feed: http://www.dnv.org/rss.asp
smime.p7s
Description: S/MI
Hi there,
We are experiencing an issue on RHEL 4 (2.6.9-22.EL) with our TDM110P -
whenever we enter 'ztcfg -s' to stop the span, the entire system
crashes, requiring a reset. I have seen this
(http://lists.digium.com/pipermail/asterisk-users/2005-June/
112097.html) and thought it might be
I am trying to setup Asterisk as the voicemail server for Cisco Call Manager. I
have just about everything working except for the message waiting indicator.
I have the following setup in context [ccm] in my extensions.conf file:
;MWI
exten => _2807XXX,1,SetCallerID(${EXTEN:3})
exten => _2807XXX,
Whenever we call IBM, the call counter on the phone never starts and in
the CLI the zap channel never gets the answered signal from the PRI.
See below.
-- Executing Dial("SIP/5933-645d", "Zap/g1/18004267378") in new
stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called g1/180
This post is exactly my problem:
http://lists.digium.com/pipermail/asterisk-users/2005-July/117988.html
Has anybody encountered this and been able to solve it and use g729
successfully? Are there other g729 implementations available as a codec
for asterisk?
Mojo
Mojo with Horan & Company, L
Can you please also send to phpkidhotmail.com.
Rplace the of course.
Thanks.
TRoy
Rajesh kumar wrote:
Please send them to me at [EMAIL PROTECTED]
regards,
Rajesh
- Original Message -
From: Technical Support
To: asterisk-users@lists.digium.com ; 'Roman'
Sent: Friday, October
Hi all,
I just installed an TDM02B. My system is a dell pc with
linux 2.6.12-1.1456_FC4
asterisk-1.2.0-beta1
zaptel-1.2.0-beta1
libpri-1.2.0-beta1
in /etc/zaptel.conf I have (all others are default):
fxsks=3-4 <--- I saw light in the ports
channels=1-2<
On Fri, Oct 07, 2005 at 09:15:05AM -0700, Chris Jones wrote:
> Hey all,
>
> I just tried running a 'make rpm' on a fresh install of Fedora Core 4
> and ran into an error near the end of the build process. This is the
> output of the build when the error occurs:
>
> done
> rm -f /tmp/asterisk/var/
With verbose and debug both on 255, here's all I get at the CLI. The X
is during the call, at the instant the Zap leg seems to drop, almost
concurrently with the 'Hungup Zap/1-1'.
-- Executing Macro("SIP/112-a88a", "internaldialout|7476011") in new
stack
-- Executing ChanIsAvail("SIP/1
I don't know what to look for in my sip debug logs, can anybody suggest
what sorts of messages my phones might unexpectedly give asterisk
causing it to drop the zap leg?
Mojo with Horan & Company, LLC wrote:
Hello - I have 8 polycom 501s all setup great using ulaw. We have put
them through a
Rod Bacon wrote:
Do the echo cancellation settings in zapata.conf have any effect when
hardware echo cancellation is installed on a 406p/411p?
The only setting that has any effect is enabled/disabled.
How can I tell if the echo is being cancelled by hardware or software?
The software echo c
Jean-Michel Hiver wrote:
IMO, there's absolutely nothing wrong with a fork. In fact, were I
someone with some seroius coding skills and/or the resources to make
it happen, I'd have forked the damned thing 2 years ago, and likely
would have been able to migrate it over to a true OSS license (B
Or why not create all sound files under /usr/share/asterisk/sounds and
then subdirs from there for your own touched files i.e.
/usr/share/asterisk/sounds/custom ?
--
Michael Coburn
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir
Cohen
Sent: Friday,
try compiling with 586 and change the makefile to disable mmx codes (if
any). I remember to have this working on a few different processors, but
forgot how I did it.
Not necessary to disable mmx codes. The C3 processors have the full mmx
set. They are missing some of the SSE instructio
How can they be a great loss if their ideas and work never made it into the
codebase?
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
Sent: Friday, October 07, 2005 12:04 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Re: www.openpbx.org
In article
Good Afternoon,
The next Asterisk Users Group meeting has been scheduled for tomorrow,
October 8th at 11:30am.
Meetings are held monthly on the second Saturday of each month, excluding
July and December.
!! NEW ADDRESS !!
Sound Choice Communcations has moved to Bloomington Minnesota, jus
Linksys makes the WRT54GP2-NA that has ATA functionality.
Garrett Smith
<[EMAIL PROTECTED]>
716-250-3408 Direct
716-903-9495 Cell
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Garth Summey
Sent: Friday, October 07, 2005 1:45 PM
To: Asterisk Users Maili
IMO, there's absolutely nothing wrong with a fork. In fact, were I
someone with some seroius coding skills and/or the resources to make
it happen, I'd have forked the damned thing 2 years ago, and likely
would have been able to migrate it over to a true OSS license (BSD) by
now.
Tss, tss.
Hello,
I'm installing asterisk 1.0.9 in a colinux 0.6.2
I have download asterisk with cvs and when i have do the command
make install in /usr/src/asterisk, i have this error:
checking for gcc... gcc
checking whether the C compiler (gcc ) works... no
configure: error: installation or confi
>My setup is: 1 HFC card with bristuff -> ZAP/g1 (2B + 1D
>channels), SIP phones (I just removed TDM400P with 4 FXS)
>
>I created test extension 222 which goes directly to g1. In
>Zapata.conf overlapdial is set to yes.
>
>First I created this extension:
>
>exten => 222,1,Dial(zap/g1,100,tc)
>
>an
I've got a Polycom 501 that I run off a Wifi "Game Adapter" in my home. It works fine.
On 10/7/05, John Reynolds <[EMAIL PROTECTED]> wrote:
On 10/7/05, Will Glass-Husain <[EMAIL PROTECTED]> wrote:> Hi,
>> I'm provisioning an office with limited cabling. I'm looking for a desk> based wifi phone.
I've also only heard of the Clipcomm
Along the same lines...
Why doesn't anyone make a wireless ATA? Am I the only one with a need
for such a thing? By the time I plug in a wireless bridge, an ata and a
cordless phone, I need a five outlet powerstrip and shoebox to hide all
the components.
I wouldn't think anyone would consider Sprint a dying company. They just
acquired Nextel so they've got money to spend.
Maybe as an ILEC (which they are here in Ohio) they are viewing Vonage
and Voiceglo as a force that needs to be stopped to prevent further
eroding of their POTS network. I know t
On 10/7/05, Will Glass-Husain <[EMAIL PROTECTED]> wrote:
> Hi,
>
> I'm provisioning an office with limited cabling. I'm looking for a desk
> based wifi phone. Most of the ones I've seen are handsets. Any ideas?
>
> Thanks, WILL
>
Will,
I don't know of a specific wifi deskphone... but I have ru
In article <[EMAIL PROTECTED]>,
Doug Meredith <[EMAIL PROTECTED]> wrote:
> Further info. The domain is registered to Marc Olivier Chouinard. He
> has posted in the dev list.
Yes, it looks like the main people behind it are bkw, anthm and moc.
They will be a great loss to the Asterisk community i
Doesn't asterisk cache the IP? So even if your IP changes and
Dynamic DNS has updated your IP to point to the new IP, asterisk will
still see the old IP until you reload asterisk?On 10/7/05, Wilson Pickett <[EMAIL PROTECTED]> wrote:
> Is there no way to have asterisk determine its IP either via up
I hope this does take off as I am starting to feel a bit uncomfortable with
the Digium model and where it is headed. Mark Spencer and Digium deserve
full credit for creating this beautiful thing called Asterisk. They did it
knowing full well these sorts of possibilities existed in the future. M
Please send them to me at [EMAIL PROTECTED]
regards,
Rajesh
- Original Message -
From:
Technical Support
To: asterisk-users@lists.digium.com
; 'Roman'
Sent: Friday, October 07, 2005 9:54
AM
Subject: [Asterisk-Users] RE: faxing
to/from asterisk - new scripts
> Is there no way to have asterisk determine its IP either via upnp or else
> resolve a dyndns hostname rather then having an entry in the config file?
For SIP you need to change the externip= variable and then "sip
reload". If you detect the ip change using a cron script (dyndns.org
has info on t
Dear Group,
I have been able to configure my Asterisk BOX to receive calls from
Mediatrix 1204.
I'm having problems sending calls out via my Mediatrix unit.
The SIP Invite is sent to the Mediatrix but the Mediatrix unit sends
back a Status : 480 Temporarily Unavailable.
This is my configuratio
Personally, I believe it's a good thing. It gives more choice.
Look at other products: IPCop (Linux based firewall) is a fork derived from
Smoothwall. They made such a nice job that Smoothwall were playing catch-up
with IPCop for quite some time. I don't know the current situation.
GPL allows for
Hi All,
With spandsp.0.0.2 pre20 installed I can't seem to send faxes with
tx_fax over a Zap channel (POTS). rx_fax works just fine so no issues
with libtiff and (presumably) libxml2.
Basically I get 'slow carrier up' and 'slow carrier down' together with
accompanying beeping noises until tx
Clipcomm has a WIFI extensible SIP desk phone that we have successfully
integrated with Asterisk.
Cory J Andrews
Partner / Purchasing
+++
VOIPSupply.com - Everything you need for VOIP
454 Sonwil Drive
Buffalo, NY 14225
+++
tf voice - 800-398-VOIP X22
l voice - 716.630.155
> >Two asterisk boxes 150 miles apart, both cvs-head as of this morning
> >(and since Sept 27th), connected via iax2 with low-utilized ds3 internet,
> >C7960 calls exten on remote system (also C7960), and call goes to VM.
> >No other calls in either system (eg, no load).
> >
> >Both boxes have iax
> On Friday 07 October 2005 11:28, Jon Pounder wrote:
>> contributors more choice. As long as the two streams stay compatible
>> (which they likely will) it should be better for everyone.
>
> Don't count on it, the rumblings in the IRC channel sound like it will be
> totally INcompatible except to
Hi
While loading asterisk I am getting this error can any one guide me resolve
this
[cdr_addon_mysql.so]Ouch ... error while writing audio data: : Broken pipe
Shaikh
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Hey all,
I just tried running a 'make rpm' on a fresh install of Fedora Core 4
and ran into an error near the end of the build process. This is the
output of the build when the error occurs:
done
rm -f /tmp/asterisk/var/lib/asterisk/mohmp3/sample-hold.mp3
mkdir -p /tmp/asterisk/var/spool/asterisk
Hi,
I'm provisioning an office with limited
cabling. I'm looking for a desk based wifi phone. Most of the ones
I've seen are handsets. Any ideas?
Thanks, WILL
___
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Asterisk-Users mailin
> sigh.. meaning take the fork
if you want a ford, buy a ford, if you want a gm buy a gm, they are both
cars. if no one wanted a gmc, they would not be around much longer. No one
is going to question your reasons for wanting one or the other, you are
free to choose. There is room for both and if
On Friday 07 October 2005 11:28, Jon Pounder wrote:
> contributors more choice. As long as the two streams stay compatible
> (which they likely will) it should be better for everyone.
Don't count on it, the rumblings in the IRC channel sound like it will be
totally INcompatible except to pass cal
Nice smartass remark... of course anyone can register a domain name.
Is forking asterisk legal? Of course it is! Asterisk is under the GPL,
which means that anyone can fork it at any time for any reason.
Look at this in a positive light... many open source projects have
forked, and the bra
I would be interested as well...
Why not post them somewhere?
Regards,
Marc
[EMAIL PROTECTED] wrote:
I'm game for using them /and testing them.
Ben..
Roman:
I created two bash scripts called Mail2Fax and Fax2Mail for use with the
asterisk sever.
They leverage the app_txfax and app_rxfax
sigh.. meaning take the fork
..o---o..
Brian Fertig
Network/Systems Engineer
IT Administrator
Planet Telecom, Inc.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kevin
Walsh
Sent: Friday, October 07, 20
Jon Pounder [EMAIL PROTECTED] wrote:
> There are people out there who wish to contribute, and not have their work
> lost on an individual project website since they do not choose to accept
> digium's terms to contribute to asterisk. This gives them an opportunity
> to do so, and have their work agg
I'm game for using them /and testing them.
Ben..
> Roman:
>
> I created two bash scripts called Mail2Fax and Fax2Mail for use with the
> asterisk sever.
>
> They leverage the app_txfax and app_rxfax scripts, along with ast_fax.
> They
> make using these apps a lot easier, including being able to
Brian C. Fertig [EMAIL PROTECTED] wrote:
> > Further info. The domain is registered to Marc Olivier Chouinard. He
> > has posted in the dev list.
> >
> Can they do this? Is this legal?
>
Yes - anyone can register a domain name.
--
_/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/
_/_/_/
You could use a modem/router with a DynDNS server in it, that would take
care of finding the Asterisk address from the outside.
On the Asterisk server you can can (I think by default it is anyway) tell it
to bind to all IP addresses.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMA
On Fri, October 7, 2005 16:26, John Crowhurst said:
> Does anyone know how to fix this error message?
>
> Is it a fault with the card?
>
> The symptoms are excessive disk access and then Asterisk stops responding,
> a powerdown and restart is generally required to resolve the issue.
>
> I'm running
Hi,
In fact, a S0 is like a T0 interface except the fact that it is
internal. Normally, the S0 should be powered by a PBX or something.
So, normally, you should be ablt to connect to it in TE mode.
At our office, I have a PBX (Nortel) with a S0 bus on which I have
connected Asterisk...it works
Does anyone know how to fix this error message?
Is it a fault with the card?
The symptoms are excessive disk access and then Asterisk stops responding,
a powerdown and restart is generally required to resolve the issue.
I'm running 2.6.13.1 with a P4 processor, Slackware Linux.
--
John
___
There are people out there who wish to contribute, and not have their work
lost on an individual project website since they do not choose to accept
digium's terms to contribute to asterisk. This gives them an opportunity
to do so, and have their work aggregated with everyone else in the same
categ
Have you ever read the GPL?
-bill
On 7-Oct-05, at 10:51 AM, Brian C. Fertig wrote:
Can they do this? Is this legal?
..o---o..
Brian Fertig
Network/Systems Engineer
IT Administrator
Planet Telecom, Inc.
-Original Message-
From: [EM
Rich Adamson wrote:
Two asterisk boxes 150 miles apart, both cvs-head as of this morning
(and since Sept 27th), connected via iax2 with low-utilized ds3 internet,
C7960 calls exten on remote system (also C7960), and call goes to VM.
No other calls in either system (eg, no load).
Both boxes have
Looks like Terracall
has not only reduced the rates but also reduced their ability to connect the
calls to India.
Today we are not
able to make even one call, but the CDRs are still coming as connected and we
are being charged.
Please note the
request I sent below for the credit.
Ade
On Fri, Oct 07, 2005 at 10:51:45AM -0400, Brian C. Fertig wrote:
> Can they do this? Is this legal?
Google "fork open source".
--
Mike
___
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Asterisk-Users mailing list
Asterisk-Users@lists.di
On Fri, Oct 07, 2005 at 02:02:06PM +1000, Rod Bacon wrote:
> Upon closer inspection, I don't think my system ever tries to establish a
> zaptel native bridge. Is there somewhere where this function is
> enabled/disabled?
Yeah, if you have echocancelwhenbridged or any other options that would
mak
Roman:
I created two bash
scripts called Mail2Fax and Fax2Mail for use with the asterisk
sever.
They leverage the
app_txfax and app_rxfax scripts, along with ast_fax. They make using these
apps a lot easier, including being able to mail to [EMAIL PROTECTED] for outgoing faxes and then
gincantalupo <[EMAIL PROTECTED]> wrote:
>why a fork???
I don't know any of the people involved, or what their motivation
might be, but I will make a guess:
Digium's model tends to stifle innovation. Look at eclipse.org for a
much better model. Eclipse is truly open source. IBM's commercial
pr
I am using Teliax to terminate my calls, and I have 3 licenses' for
g729 from Digium. "show translations" verifies that the registration
took place.
When I place a call, having "allow=g729" as the only allow option in
iax.conf, I get the following error:
WARNING[361]: chan_iax2.c:6017 socket_read
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