I have a number of servers with TE405P cards. The servers are DELL 1850's (which I _NOW_
see are listed on the digium not recommended page because
of the ethernet interface).
The problem I have is only during bridged calls.
If I place a call into a service hosted on the box, or out to a VOIP
Can anyone tell me if Asterisk would speficically benefit from the reduced latency of a preemptible Linux Kernel? I know it was recommended against in
the early days, but I'm wondering if there are any updated opinions?
--
==
Rod Bacon
Empowered
http://www.crazygreek.co.uk/content/chan_bluetooth
maybe we should tell the author about the bounty :P
On Mon, 2005-06-20 at 00:32 -0500, Jay Milk wrote:
This is what you want:
http://www.voip-info.org/wiki-Asterisk+bounty+bluetooth+cell-phone+suppo
rt
Add money to the bounty -- maybe
Will the CVS HEAD version of the Zaptel drivers work with the STABLE branch of
*?
--
==
Rod Bacon
Empowered Communications
Ground Floor, 102 York St. South Melbourne
Victoria, Australia. 3205
Phone: +613 99401600Fax: +613 99401650
FWD: 512237
tos 16 = 0x10 (lowdelay).. that part is ok, but is a workaround, NOT a
fix-up...
On Sun, 2005-06-19 at 18:48 -0400, Doug Lytle wrote:
Calin Serbanescu wrote:
static int tos=16;
I think it is tos=0x16
Doug
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Hi,
I have installed latest stable version of Asterisk. I
registered 1 Quintum Gateway and 5 eyeBeam SoftPhone. But I can make just one
call at the same time, if i try to make calls from 2 softphones to anotherone,
second caller listens the person have extension is on the phone
Hi,
Im new to
asterisk, so, hi!
Finally got it
installed and working (lots of troubles with oh323), but i cant seem to get it
to record the callrecordsin any CDR files (or in
mysql).
I had one version
of asterisk on a test system which did seem to record some cdr records, but ive
since
Hi,
I wouldlike use webvmail on my asterisk, I use debain Sarge with
asterisk 1.0.7 package.
I have installed package asterisk-web-vmail but when i go to
http://MyAsteriskBOx, i have a page of presentation of Apache.
Could you help me please?
Thanks
Does anyone have any comments about using Debian stable release Vs Fedora
for running Asterisk?
Syed Akbar
Alico Systems Inc
www.alicosystems.com
Tel: 562-436-1510
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You should choose your distrib if you konw Fedora or Debian.
I use Debian because I prefer debian system package in order to update
and security patch?
Syed Akbar a écrit :
Does anyone have any comments about using Debian stable release Vs Fedora
for running Asterisk?
Syed Akbar
Alico
On Fri, Jun 17, 2005 at 10:34:25PM +0200, Conrad Beckert wrote:
... probably one of those RTFM kind of questions (while I'd be happy to know
where a good reference FM is :-) )
Has anyone an idea on how to disable the console sound driver. My problem is
that a running asterisk is muting
hi
is there anybody using g723 with oh323 and sending call by asterisk. if so
please let me know how i can use this same, i need to call quintum by g723 .
Thanks
Bashir
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On Sun, 19 Jun 2005, Dan Morin wrote:
If anyone has any experience with a Panasonic KX-TD1232 phone system, I
would really like to talk to you for a few minutes.
I have asterisk connected to a Panasonic system via FXS - CO ports.
I'm trying to get the Panasonic configured so that if someone
Hi,
Visit http://aussievoip.com.au/wiki-G723-1-Install you'll find how to
install g723, but first you have to install g729
http://aussievoip.com.au/wiki-G729-Install
I have tested it with Quintum, it works
Enjoy :)
Erdem HAKI - [EMAIL PROTECTED]
-Original Message-
From: [EMAIL
Hi all,
I want to build a central call diverter via asterisk (http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20SIPredirect). Calls would come in via SIP from Cisco Callmanager Asterisk would do some searching an diverts the call to an extension, which is also located at Callmanager.
remove the default index.html from /var/www/htdocs and/or be sure the
apache default DocumentRoot is pointing to the web space and not the
default debian pages
On Mon, 2005-06-20 at 09:25 +0200, sylvain garcia wrote:
Hi,
I wouldlike use webvmail on my asterisk, I use debain Sarge with
Axel Schemberg wrote:
I use Asterisk on Debian via: ap-get install asterisk, which is Version
1.07.
The page you linked says: new in Asterisk 1.2.x. I guess that this
pretty much explains why it does not work in your case :)
BTW, this looks like a -users question to me, so I've moved it
ok but i don't know where is the web space with cgi scripts.
the package "asterisk-web-mail" don't install cgi script, i think
Scott Kamp a crit:
remove the default index.html from /var/www/htdocs and/or be sure the
apache default DocumentRoot is pointing to the web space and not the
So, I've been able to receive faxes quite reliably through teliax with
g711 so far; I think I can live with it.
For outbound faxing, I'd really like to get a service that lets me
send faxes, but doesn't charge me a monthly fee (I don't send enough
faxes to justify it). T.38 is a requirement; I
Hi,
After upgrading to 1.5.2 I no longer can directly access to my voicemail
by pressing the Message button, I have to go through the
urgent,new,old report first. The oneTouchVoicemail parameter is set to
1 but not taken into account apparently.
Anyone noticed that problem?
Roger Schreiter ha scritto:
Hi,
package tiff-v3.5.7 contains the currently recommended version
of libtiff in order to run spandsp (fax support for asterisk).
i had no problems receiving faxes with version 3.7.2.
on the other hand i have big problems in sending multipage faxes. only
the
Hi,
Does anybody have an idea on how to realise ad-hoc conferencing with
Asterisk ? Although Asterisk MeetMe and maybe a procedure with Call
Holding could somehow come close to ad-hoc conferencing, it doesn't seem to
be the right way to do it. Any experience with ad-hoc conferencing using
Hi,I'm running CVS-head for quite some time now, and util last saturday without serous problems.Last saturday however I updated * again, and now the cpu load goes to a 100%.It seems that the sip register is the problem.when I comment out all sip registers there is no problem, but enabling just one
Hi list!
I'm trying to use call files to place outgoing calls.
I want to schedule an outbound call and want it to ring on my sip phone.
My sip phone is SIP/228 and the call should go out according to the LCR
rules as defined in the dialplan. I don't mind waiting for the call to be
answered
Title: Message
hi
all,
im trying to get a
php file to run thru AGI...but when i run it through *this is what i
get
i have the latest
php installed ver. 5...had ver 4.3.11...but had this problem so i figured an
upgrade to the highest would resolve it...
im running CVS Head
and
Am looking for everyones advise/recommendations.
I have am setting up a network of both office and home based workers.
The office workers will be on the same network as the Asterisk box so no
NAT hassles there. However, the home workers are on their own DSL
connections so I imagine that
It seems that my * does not react to tos=whatever field in iax.conf. I
am using latest CVS HEAD code.
Can anybody help me with this issue?
ps:
if i go to chan_iax2.c and modify the initial definition of tos
variable, it works fine marking packets with the value specified there:
static
I don't quite know what you mean but usually the conferencing portion of the
call is actually done by the phone. If you're using a phone that is
incapable of this and want asterisk to take over, yes meetme is the only way
to do it... There's no other way to do the audio mixing easily.
- Joshua
Make sure you're not using asterisk or you will have no T.38 support, not
even passthrough.
- Joshua Colp
On 6/20/05 6:46 AM, Adam Megacz [EMAIL PROTECTED] wrote:
So, I've been able to receive faxes quite reliably through teliax with
g711 so far; I think I can live with it.
For outbound
I have an asterisk installation connected to 2 isdn lines via an AVM C2
card.
modules seems to load well, lsmod gives :
c4 19588 4
b1 24192 1 c4
capidrv28468 2
isdn 134604 9 capidrv
slhc7552 1 isdn
Hello,
I just tried it, and it worked fine for me. Of course the context and the
Extension where different. Is the Channel correct?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Remco Barende
Sent: Monday, June 20, 2005 2:12 PM
To: Asterisk Users List
On 06:11, Mon 20 Jun 05, Jeromy Grimmett wrote:
hi all,
im trying to get a php file to run thru AGI...but when i run it through
*this is what i get
i have the latest php installed ver. 5...had ver 4.3.11...but had this
problem so i figured an upgrade to the highest would resolve
Jeromy Grimmett wrote:
hi all,
im trying to get a php file to run thru AGI...but when i run it through
*this is what i get
i have the latest php installed ver. 5...had ver 4.3.11...but had this
problem so i figured an upgrade to the highest would resolve it...
im running CVS Head
Yes, the channel should be correct.
I'm using AMP and from-internal is the context the sip phones are normally
in.
Do you see anything on the console even if you dial a number that isn't
answered?
Thanks!
On Mon, 20 Jun 2005, jurczak wrote:
Hello,
I just tried it, and it worked fine
Hi folks
I seted up the asterisk with an active ISDN B1 AVM Card (german vendor)
and it works fine, various SIP clients (IP fon snom, xlite under
MacOSX) and also incoming and outgoing connectins. Ok. No problem.
After that I configured a CP7940G with a MGCP IOS. It's connected to the
asterisk
I have a Asterisk @home ver 1.0 running with a TDMB11 card. Several sip
extensions and a regular phone connected to the box. All routing works fine
from the regular phone connected to the box, whether its going to FWD,
broadvoice or the PSTN. The problem I am experiencing comes from making
I tried also with wrong channel, and after a while the file was disappeared
and asterisk said that he was unable to call that channel
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Remco Barende
Sent: Monday, June 20, 2005 4:37 PM
To: Asterisk Users
He knows -- see the bounty-page.
-Original Message-
From: trixter http://www.0xdecafbad.com
[mailto:[EMAIL PROTECTED]
Sent: Monday, June 20, 2005 1:33 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] bluetooth audio and asterisk
Remco Barende ha scritto:
Do you see anything on the console even if you dial a number that isn't
answered?
i see this for a non existant number:
Attempting call on Zap/g1/12345 for [EMAIL PROTECTED]:1 (Retry 1)
i guess it prints out for every call originated by a call file.
asterisk
I installed Asterisk Voicemail in an office and now most of the
employees are complaining that when they're listening to the
messages, it takes forever to listen to their messages. The reason
being is that before the message is played, the voicemail says the
full date and time when the
Hi Joseph, did you try iaxping?
http://www.bpvn.com/asterisk/iaxping.zip
Invaluable for IAX troubleshooting. hth
-Original Message-
From: Joseph [mailto:[EMAIL PROTECTED]
Sent: Saturday, June 18, 2005 2:16 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
Since www.bkw.org seems not to exist anymore (getting response from some
hosting provider), does anyone happend to have a copy of app_valetparking.c
from www.bkw.org - the one that should work with * stable 1.0.X ? If so
please contact me.
One that can be downloaded from www.loligo.com dosn't
hello there,
can somebody please comment which one of these channel drivers will give
best output doing g729|g723 pass-thru. only pass-thru is needed no
transcoding.
please share your experience. if somebody has some figures (simultanous
calls using a certain channel driver) it will be
I have fooled the SurfBoard on Shaw with the MAC address issue by
resetting it 3 times sucessively, then it will accept a new MAC address
immediately. hth.
-Original Message-
From: Joseph [mailto:[EMAIL PROTECTED]
Sent: Saturday, June 18, 2005 3:19 PM
To: Asterisk Users Mailing List -
Turn off the envelope announcement (envelope=no) in voicemail.conf.
- Original Message -
From: Waldo Rubinstein [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, June 20, 2005 9:55 AM
Subject: [Asterisk-Users]
Use Voicemail with mail of your employees in order to send with attachment.
Waldo Rubinstein a écrit :
I installed Asterisk Voicemail in an office and now most of the
employees are complaining that when they're listening to the
messages, it takes forever to listen to their messages. The
I'm trying to use LookupCIDName to tag outgoing calls on my CDRs but it
seems that application only tags incoming calls?
Any sugestions?
Leandro
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I really don't know why but after restarting asterisk it just works all of
a sudden, no change to the call file
I guess that doesn't offer a real solution but it works
Thanks guys!
On Mon, 20 Jun 2005, Marco Parmeggiani wrote:
Remco Barende ha scritto:
Do you see anything on the console
Marco Parmeggiani ha scritto:
on the other hand i have big problems in sending multipage faxes. only
the first page goes through.
uhm, no, neither the first page is received. i was optimistic.
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Can someone explain me what's going on and why the receiver of this fax
guives up saying communication error?
Slow carrier up
Slow carrier down
Slow carrier up
CSI: 40 20 20 20 20 20 20 20 34 39 34 35 36 34 39 35 30 20 39 33 2b
CSI without final frame tag
Remote fax gave CSI as: +39 059465494
In your voicemail.conf file you should see global options.
Uncomment out envelope=no and nextaftercmd=yes
This should help
http://www.voip-info.org/tiki-index.php?page=Asterisk+config
+voicemail.conf
John B
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
Check your voicemail.conf !! you can do your own date prompt (or no date
prompt)
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] la part de Waldo
Rubinstein
Envoye : lundi 20 juin 2005 15:56
A : Asterisk Users Mailing List - Non-Commercial Discussion
Objet :
Can't get a Digium
TDM04B working. Asterisk is running. I seem to have setup the trunks
OK. But whenever I make an outgoing call get the 'all circuits are busy
now' message. If I call in nothing happens at all!
Here is my zapata.conf file:
;; Zapata telephony interface;;
Configuration
Andrew,
I presume you mean in the Cisco 7940/7960 SIP Phone Administrator's Guide?
When you say mapped, dou mean that it needs an explicit entry in the
dialplan.xml like:
TEMPLATE MATCH=# Timeout=0 User=Phone/ !--
Explicit # for Asterisk --
Mike
- Original Message
Yes,
I'm running it right now, CVS as of a few days ago, and * 1.0.7 on 2.6.x
kernel and FC2.
Chad
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rod Bacon
Sent: June 20, 2005 2:44 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users]
On 6/18/05, Balaji NJL [EMAIL PROTECTED] wrote:
Hi All,
I am a new bee to *. I just installed [EMAIL PROTECTED] on
FC3. I hv a FXO card. I hv configured two extensions
one x-lite and other iaxComm. I configured * using
AMP. The following setup works
- x-lite (x 200) to iaxComm (x 201)
-
Please,
send us zapata.conf. It's possible that you don't have well
configure zapata.conf, because in your trace you try to dial through g0
group and your Zap/4(I understand is your Zap connected to PSTN) must be
into the 0 group.
Regards,
srsergio
-Mensaje original-
De:
If the lines are put in seperrate groups according
to destination then each group can be accessed buy dialing 8+group number. Eg.
81 for trunks in group 1, 82 for group 2, etc
This applies to uk systems but I suspect
the same feature exists in other variants.
Neil
From:
Hi,
-Original Message-
Will the CVS HEAD version of the Zaptel drivers work with the STABLE
branch of *?
Err, why specifically would you want that ?
Florian
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Marco Parmeggiani wrote:
Can someone explain me what's going on and why the receiver of this
fax guives up saying communication error?
Start tx page 1
EOP: 2f
RTN: 4c
DCN: fb
Disconnecting
The receiver says communication error because txfax's response to the
receiver's RTN signal was a
Try this
Since www.bkw.org seems not to exist anymore (getting response from
some hosting provider), does anyone happend to have a copy of
app_valetparking.c from www.bkw.org - the one that should work with *
stable 1.0.X ? If so please contact me.
One that can be downloaded from
Florian Overkamp wrote:
Hi,
-Original Message-
Will the CVS HEAD version of the Zaptel drivers work with the STABLE
branch of *?
Err, why specifically would you want that ?
Florian,
In our case, the CVS drivers (At the time that I did it) showed enhanced
information
There is another one that has to be set now. I don't remember it but
look in the documentation it's in there.
On 6/20/05, Louis-David Mitterrand [EMAIL PROTECTED] wrote:
Hi,
After upgrading to 1.5.2 I no longer can directly access to my voicemail
by pressing the Message button, I have to go
On Mon, 2005-06-20 at 08:52 -0500, Jay Milk wrote:
He knows -- see the bounty-page.
I did and it looked as if it was written by a 3rd party not himself.
On the authors page he was asking for donations to help him finish and
such. Why I think that the two resources needed to be put together.
It looks like a NAT issue.
On 6/20/05, Matt [EMAIL PROTECTED] wrote:
Hi all,
I've been messing around with the g729 codec in some phones I use and had
made all phones use the codec for all calls for testing purposes. The
problem is when I attempt to dial out on my Polycom IP 500
On 6/19/05, Bob Goddard [EMAIL PROTECTED] wrote:
On Friday 17 Jun 2005 18:05, Manuel Casal wrote:
Marco Parmeggiani escribió:
Manuel Casal ha scritto:
I made the make menuconfig and make dep in the kernel sources.
i do not remember well how i solved that problem but i'm sure that
So far you always dodged the point, which is that you pay more for
calling cell phones in Australia, if that's not the case say it, don't
just say that YOU don't pay for YOUR cell phones. The fact is that the
biggest portion of you phone bill goes towards calling cell phones in
Australia, you have
which zap channel or group goes to the FXO?
On 6/19/05, Jose Vicente Ortega [EMAIL PROTECTED] wrote:
I have a Asterisk @home ver 1.0 running with a TDMB11 card. Several sip
extensions and a regular phone connected to the box. All routing works fine
from the regular phone connected to the box,
On 6/17/05, David Hajek [EMAIL PROTECTED] wrote:
Do you have analog TDM in it?
-David
Oswaldo Arratia wrote:
I bought a Dell SC1425 and installed a T1/E1 card from Digium and I tried to
configure it using [EMAIL PROTECTED] scripts and did not work, so I went the
long way and
What DTMF mode are you using?
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Nope ! This is the one that tries to include PRE 1.0.X header file
parking.h.
It cannot compile on * 1.0.X (I have tried also to include features.h
instead of parking.h (as far as I know features.h is successor to
parking.h), but still without results).
Thanks anyway.
Nenad
Try this
On 6/20/05, Waldo Rubinstein [EMAIL PROTECTED] wrote:
snip
Is there anyway to shorten that or even give users the option to not play
that?
Voicemail as some options that can help you with that.
See http://www.voip-info.org/tiki-index.php?page=Asterisk+config+voicemail.conf
It can be
As far as I remember, Nate Carlson worked with ZZ to document the
progress on the wiki page. Let's shoot them all emails and see what
they have to say.
-Original Message-
From: trixter http://www.0xdecafbad.com
[mailto:[EMAIL PROTECTED]
Sent: Monday, June 20, 2005 11:25 AM
To:
eval { $cost = int($adjcost * $adjtime / 60) };
#cost = 253
Corrected, this would be 250.
Viewed another way, using a 6 second increment, 147 seconds
represents
25 such increments (actually 24.5, but we get all of the last
increment, so it's 25).
25 * 10 (the cost of
On Mon, 2005-06-20 at 12:28 -0400, Doug Lytle wrote:
Florian Overkamp wrote:
Hi,
-Original Message-
Will the CVS HEAD version of the Zaptel drivers work with the STABLE
branch of *?
Err, why specifically would you want that ?
Florian,
In our case,
Looks familiar. Since AGI files are hash-banged, try to run it from
shell. If it still fails, check the file for windows-style LF/CR and
correct the file-format.
-Original Message-
From: Jeromy Grimmett [mailto:[EMAIL PROTECTED]
Sent: Monday, June 20, 2005 6:11 AM
To: 'Asterisk Users
Hi,
I am considering these two devices and would like to get your opionion on
which one to choose for the best voice quality.
I undestand that for two lines I would need two sipuras and one tdm with two
modules.
One limitation could be my asterisk box which is only 333 P2 with 512Ram.
Thanks in
On Sun, 19 Jun 2005, Stefan Gofferje wrote:
Armin Schindler schrieb:
I would like to announce the first release of the chan_capi
channel driver on sourceforge.net
The package is available for download with name chan_capi-cm-0.5
and is the current CVS HEAD.
It is derived from
Hello,
I'm running asterisk 1.0.7 on a kernel 2.4 linux box. I have my SIP
users database on MySQL, also I'm running asterisk from static .conf
files. What I want to get done is that after a pre defined period of
time, my SIP users table must be dumped into a .conf file. Perhaps with
some script
Oops, I sent the wrong one. Here's one I modified to work with 1.0.X
Try again
Nope ! This is the one that tries to include PRE 1.0.X header file
parking.h.
It cannot compile on * 1.0.X (I have tried also to include
features.h instead of parking.h (as far as I know features.h is
successor
I have rfc2833 listed as my dtmfmode in sip.conf... sorry if this wasn't what
you meant -
I'm pretty new to this stuff. Just let me know if you meant to look/ask
somewhere else
for that
On Mon, 20 Jun 2005 13:04:12 -0400, Bryan M. Johns [EMAIL PROTECTED]
wrote :
What DTMF mode are
Hello all,
Recently I purchased an QuadBRI card from junghanns.net after some
playing around, reconfiguring dialplans etc with the exception of 1
thing everything seems to work:
I seem to be unable to set the outbound callerid. The dutch telecom
operator (KPN) provided me with 4 MSN's on 1 BRI
Hi,
-Original Message-
-Original Message-
Will the CVS HEAD version of the Zaptel drivers work with the STABLE
branch of *?
Err, why specifically would you want that ?
In our case, the CVS drivers (At the time that I did it)
showed enhanced
information coming across
June 20th, 2005
PBXfreeware.org Open for business!
Anthony Minessale II, CTO and implementer of Asterlink, has announced
the grand opening of PBXfreeware.org, a new site designed to supply
the open source PBX community with access to contributed open source
applications. To kick start the
Here is is.
; Zapata telephony interface
;
; Configuration file
[trunkgroups]
[channels]
language=en
context=from-pstn
signalling=fxs_ks
rxwink=300 ; Atlas seems to use long (250ms) winks
;
; Whether or not to do distinctive ring detection on FXO lines
;
Are you sure that for the BRI outgoing callerid is allowed?
I have several BRI lines where outgoing callerid is blocked (on my
request) by KPN. No matter what you pass to the BRI line, KPN will never
pass callerid.
On Mon, 20 Jun 2005, Stijn Jonker wrote:
Hello all,
Recently I purchased
Hello.
It seems very very
strange to me that nobody talks about connecting ISDN lines to asterisk using a
device with RCAPI Support.
There are some
devices (most of them isdn routers, like the famous Cisco 800 family) that
support this protocol. It basically is an ISDN to IP gateway.
Hmm,
Replying to my own post, forgot one important detail:
Asterisk 1.0.7-BRIstuffed-0.2.0-RC8g
Stijn
On 20-Jun-2005 20:20, Stijn Jonker wrote:
Hello all,
Recently I purchased an QuadBRI card from junghanns.net after some
playing around, reconfiguring dialplans etc with the
I just discovered an 18 hour call to Brazil that was 60 seconds of
an employee calling a customer, then 18 hours and 47 minutes of background
noise in their office. The Cisco 7960's have an issue where you sometime
don't realize the phone is still off hook as was the case for this call.
On 21:11, Mon 20 Jun 05, Remco Barende wrote:
Are you sure that for the BRI outgoing callerid is allowed?
I have several BRI lines where outgoing callerid is blocked (on my
request) by KPN. No matter what you pass to the BRI line, KPN will never
pass callerid.
Isn't KPN allowing it by
On Mon, 20 Jun 2005 [EMAIL PROTECTED] wrote:
Hello.
It seems very very strange to me that nobody talks about connecting ISDN
lines to asterisk using a device with RCAPI Support.
There are some devices (most of them isdn routers, like the famous Cisco 800
family) that support this
it works well on sip to me too, the problem is in chan_iax
On Mon, 2005-06-20 at 06:38 -0600, Rich Adamson wrote:
It seems that my * does not react to tos=whatever field in iax.conf. I
am using latest CVS HEAD code.
Can anybody help me with this issue?
ps:
if i go to
Digging further into the FXO cpu spike vs clock issue, I
removed the 18.432 MHZ crystal from an FXO card and replaced
it with a 20.000 MHZ crystal. This of course forced the zaptel
timing way off ~ 93% accurate using ztclock. I then proceeded to
modify the wcfxo.c driver source code to set the
Hello Remco Michiel,
First of all thanks for the replies.
On 20-Jun-2005 21:47, Michiel van Baak wrote:
On 21:11, Mon 20 Jun 05, Remco Barende wrote:
Are you sure that for the BRI outgoing callerid is allowed?
Yep, with pain in the heart i reconnected the KPN ISDN phone directly to
the NT1
Is there an easy way to automatically log agents in?
We are using the queuing function to front end a main number
without really using multiple agents. The downside is during a restart,
or system reboot someone must remember to log in the agent. If I could
incorporate it into a startup
Another odd thing I noticed was I will dial my cell phone, I will pick up and
just for
that half a second right after i pick up the phone I can hear audio both
ways...it then
cuts out and only audio is heard talking to the poly and heard on the cell but
not the
other way around. I was
Ya, going with whichever you know best is not a bad idea
Although I think more ppl have used Debian with success/little hassle than
with Fedora.
There are always little bugs with installing on the Fedora releases...
I ended up abandoning my fedora core 2 install in favour of sticking
Hi Stijn,
On 22:18, Mon 20 Jun 05, Stijn Jonker wrote:
Hello Remco Michiel,
First of all thanks for the replies.
On 20-Jun-2005 21:47, Michiel van Baak wrote:
On 21:11, Mon 20 Jun 05, Remco Barende wrote:
Are you sure that for the BRI outgoing callerid is allowed?
Yep, with pain
This one has compiled cleanly ! Thank you very much.
Nenad
Oops, I sent the wrong one. Here's one I modified to work with 1.0.X
Try again
Nope ! This is the one that tries to include PRE 1.0.X header file
parking.h.
It cannot compile on * 1.0.X (I have tried also to include
features.h
Try with this zapata.conf
[channels]
language=en
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=no
echotraining=800
rxgain=0.0
txgain=0.0
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