Here you go, Ian..-- Forwarded message --From: G3RIR [EMAIL PROTECTED]Date: 05-Apr-2006 20:54
Subject: [dmuars] Eh up - March 144 results alteredTo: [EMAIL PROTECTED]
What's going on here.
The results of the MArch 144 UKAC have been re-published and we have lost out
Oops! Fat fingers, sorry, all.
On 06/04/06, Peter Bowyer [EMAIL PROTECTED] wrote:
Here you go, Ian..
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Hi DovidActually I dont how to set up my DTMF. Anyway here is the setting :-/etc/vpb/vtcore.conf[general]name = vtcorechannels=12cards=2[card0]type=openpcichannels=4hwplaygain=12hwrecordgain=-12chan = 0/etc/asterisk/vpb.conf[general]type = v4pcicards = 1[interfaces]board = 1echocancel = oncontext
Hi group.
I have install chan sccp drivers following instructions on
http://chan-sccp.berlios.de/#build
I have setup two Cisco 7970 phones. They register fine. When I call from one
sccp phone another it rings, and when I pick up the phone asterisk dies.
This is what it shows on CLI:
--
how do you set two types of caller id one for internal calling and one
for external? Basically everyone calling out from asterisk from one
context I want to assign a single callerid. On all other contexts I
want to assign a caller ID specific to each line for all calls going out
to asterisk.
On 17:47, Wed 05 Apr 06, Bryan Mahin wrote:
Hello all,
I am looking for a way to restrict users from logging in two separate
phones with the same authorization name/password at the same time.
Meaning, I only want users to be able to place a call from one phone in
one location, but have the
JP Carballo wrote:
Ronald Wiplinger wrote:
I tried now many places to put these lines in. The system still
announces This card number is in use.
Can you give me a place where to put it in?
It's not receiving a card number.
Find the following 3 lines:
#
# At this point we have a valid card
maybe firewall tends to close iax connection, you can try to
decrease qualify check interval (maybe qualify=5000?) PJ
Peraphs. 'qualify = 1000' seems to alleviate the problem.
Thanks
Domenico
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how can I put fax server functionality on Asterisk? * as a
reliable fax server for 500-1000 fax/day (mostly incoming)?
Fax server should be like HylaFax, i.e. stable, low
maintenance and functionality like receiving fax as email
with PDF attachment, sending faxes per WHFC.
Asterisk doesn't
Hi,
I have same setup:
PSTN E1 PRI --- Asterisk --- Crossed E1 cable --- Alcatel 4400 PBX
with some IP phones directly connected to Asterisk and a lot of
analog/digital phones connected to 4400.
When I call from an IP phone to an Alcatel one, I'm able to see full
CallerIDName.
I set it using:
Hi,
I currently use Asterisk 1.0.7-BRIstuffed-0.2.0-RC7k on a debian sarge with a
kernel 2.4.27 on a P4 3Gig with 1Gig of memory
When i use i4l on any call, the called party ( on the telco operator side )
ear me with a delay of 1 sec after 1 minutes , 2
sec after 3 minutes and so on...
After
Mimmus napisał(a):
Hi,
I have same setup:
PSTN E1 PRI --- Asterisk --- Crossed E1 cable --- Alcatel 4400 PBX
I don't know if I'm using Q.Sig or EuroISDN!
1) it's in config file
2) Should be easy to check when you say what kind of PABX card you use:
PRA/PRA2/BRA2 - EuroISDN
DLT - qsig
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
On 04/04/06 19:20 Tomislav Pareina said the following:
Ooh323 channel driver from asterisk-addons-1.2.1 has same problem
have you managed to get this working ?
I certainly hope so, but I'm not sure. I have applied patch yesterday. Now
On 04/06/06 05:36 Avi Miller said the following:
If I dialled from a SIP phone on Asterisk 1 to the Phone on the Avaya,
it worked fine. If I dialled from a phone on the Avaya, the SIP phone
would ring, but the call would drop as soon as it was answered because
of codec negotiation failure.
I don't know if I'm using Q.Sig or EuroISDN!
1) it's in config file
2) Should be easy to check when you say what kind of PABX
card you use:
PRA/PRA2/BRA2 - EuroISDN
DLT - qsig
OK, I'm using EuroISDN.
Thanks
DV
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Hi there!I was asked to set up a led on a snom phone monitoring a conference room (lit when someone is in conference).I know that there is a patch for hinting parking lots, anyone has made something similiar for conferences ?
Tnx for the support!P.S.What about monitoring a global var ?It would be
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Hi,
I'm using Asterisk and a TE110P E1 PRI in Chile.
When I call to a disconnected number or any not operational number, the
telco sends the Hangupcause disconnection code and an audio message
notifying the disconnection cause to the
Hello,I've reccorded a voice message for the IVR. (.wav, 16 bits, 8kHz)The file is /var/lib/asterisk/sound/11ivrrecording.wav.When asterisk (1.2.5) starts this file i can't hear it on my phone.Here is the log :
Apr 6 17:00:16 VERBOSE[845] logger.c: -- Executing SetCallerID(SIP/11-97b9, Patrice 11)
Hi all,
I need help in disconnect supervision. Im running on AAH ver.2.5 at home
with TDM400P with 1 FXO and 1 FXS (TDM11B). I have implemented DISA on
AAH for origination (PSTN to VOIP bridging).
I'm facing problems with disconnection supervision. My calls are not
getting disconnected at times
Asterisk SVN-trunk-r7353M
I have a EuroISDN line. I am sometimes out of the office so I get my
extension to ring both my mobile and desk top (7960) phone at the same time.
This all works just peachy. However, I have a question regarding
callerid. Is there any way of setting the callerid so
The only thing registration does is inform Asterisk about what IP the
device is at. It has nothing at all to do with Device - Asterisk
calls. Registration only affects Asterisk - Device calls. In a Device
- Asterisk call, Asterisk does not care what IP the device is at as
long as the
Julian Lyndon-Smith wrote:
Asterisk SVN-trunk-r7353M
I have a EuroISDN line. I am sometimes out of the office so I get my
extension to ring both my mobile and desk top (7960) phone at the same
time.
This all works just peachy. However, I have a question regarding
callerid. Is there any way
Hi!
Is anyone managed to transfer an alredy bridged call, to a cell phone?
Some days ago, someone told me to look for the solution in features.conf,
but I still haven't found it. I tryied to use de default blindxfer, but
it only
accept internal extensions.
Thanks in advance,
Giuseppe
Hi,
I'm trying to test my dial out function so I did something like this in
extensions.conf
exten = 999,1,Dial(Zap/g1/02601591)
exten = 999,102,Congestion()
My Zapata.conf looks something like this
[channels]
context=from-pstn
group=0
switchtype=euroisdn
overlapdial=yes
faxdetect=no
;
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
I'm trying to set the Pickup feature. I'm setting my extensions.conf as:
I'm using pickup from features.conf. I don't need anything better (for now).
--
Tomislav Parcina
tparcina#lama.hr
___
Hi,
i was able to fix this problem when i added the line pridialplan=local in the
zapata.conf but it depends on your telco, i think.
marcus
-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
Gesendet: Donnerstag, 6. April 2006 11:50
An:
Hi!
I tried this in features.conf
testfeature = *9,callee,Dial,CAPI/ISDN4/my_phone_number/b,60,T
and it works... but... I would be able to transfer a call to any phone
number,
so I tried to use this line:
testfeature = _*9.,callee,Dial,CAPI/ISDN4/${EXTEN:2}/b,60,T
but... Asterisk crash! (it
Thanks Armin,
It works with rw-rw-rw permissions to /dev/capi20.
Amaury
-Message d'origine-
De : Armin Schindler
Envoyé : mercredi 5 avril 2006 19:49
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : Re: [Asterisk-Users] can't start chan_capi with asterisk group
It
Hi.
First, pardon my bad English.
I have * configured with one avm b1 and latest chan-capi. I can dial out and receive incoming calls from isdn.
The problem is that i do not know the way to get the ring tone (hear the ringing on the caller phone when i dial with capi)
For example if i dial 0
Okay, so your group settings/permissions are not correct then.
Armin
On Thu, 6 Apr 2006, amaury BOSSE wrote:
Thanks Armin,
It works with rw-rw-rw permissions to /dev/capi20.
Amaury
-Message d'origine-
De : Armin Schindler
Envoyé : mercredi 5 avril 2006 19:49
À : Asterisk
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
on a related note, we notice that if we've set atxfer = *1 in features.conf
and blindxfer=#1, then attended transfers dont work. somehow, the Queue app
captures the '*' and hangs up the call. is this the behaviour others have
observed
Is there a fix for these errors for the junghanns card ?
Apr 6 13:11:08 asterix qozap: CRC error for HDLC frame on card 1
(cardID 0) S/T port 1
Apr 6 13:11:35 asterix qozap: CRC error for HDLC frame on card 1
(cardID 0) S/T port 3
Apr 6 13:11:39 asterix qozap: CRC error for HDLC frame on card
Hi,
can i have a FXO/FXS card and a E1/T1 card in the same system. I have used them seperatly many
times before, but not together in one machine.
I usually have for the analogue card
signalling=fxs_ks
channel = 1
and for the e1 card
signalling=pri_net
group=1
callerid=asreceived
channel =
I apologize if this information is posted elsewhere. Unfortunately I
haven't found it yet if it is. I'm not familiar with the channel
counting features could you please explain? Also, how are you tagging
the phones to account codes?
You can limit calls using the set/check group commands.
Andrzej Wolski napisał(a):
Is there a fix for these errors for the junghanns card ?
Apr 6 13:11:08 asterix qozap: CRC error for HDLC frame on card 1
Witam,
Przepraszam za komercyjny charakter tego maila, ale jeśli byłby Pan
zainteresowany to za kilka tygodni otwieramy w pełni sprzedaż
hello
all
soembody can give me
an example config how can i let dial a asterisk server via SIP over another
asterisk server to a pstn gateway ip?!?!
asterisk1: x.x.x.x
have to dial over asterisk2: y.y.y.y and then the asterisk2 should forward the
call to the PSTN gateway.
What i have to set
in
Hi,
I have Wildcard TDM400P with 2 FXS y 2 FXO. After all work fine but
now do
it:
- load driver: wctdm y zaptel (zaptel-1.2.1)
Module 0: Installed -- AUTO FXS/DPO
Unable to do INITIAL ProSLIC powerup on module 1
Unable to do INITIAL ProSLIC powerup on module 1
Module 1: FAILED
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Hi
What causes deadlock?
Apr 5 14:02:43 WARNING[2413] channel.c: Avoided initial deadlock for
'0x82acb10', 10 retries!
Apr 5 14:02:43 WARNING[2413] channel.c: Avoided initial deadlock for
'0x8298160', 10 retries!
Does this
Tomislav Parčina wrote:
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Hi
What causes deadlock?
Apr 5 14:02:43 WARNING[2413] channel.c: Avoided initial deadlock for
'0x82acb10', 10 retries!
Apr 5 14:02:43 WARNING[2413] channel.c: Avoided initial deadlock for
'0x8298160', 10
I am having issues with a TDM2400P. It appears when the ZAP channel dials
out, it randomly chops the first digit off of the number. I have tried
relaxdtmf=yes, turning up and down the txgain, turned off and on the echo
cancellation, generated new zaptel (with updated spinlock.h)...
I am at a
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
how do you set two types of caller id one for internal calling and one
for external? Basically everyone calling out from asterisk from one
context I want to assign a single callerid. On all other contexts I
want to assign a caller ID
We have had this problem with the TDM400 and just about every thing we
have ever had... it isn't the card that is chopping off the first
digit. It is the fact that it picks up too quickly and starts to dial.
Change your dial to be Zap/g0/w${EXTEN} and see if that takes care of
the problem
HI all,
Any of you having experience with voice master? I tried using the
openh323 channel it doesn't give me voice at all. THere's no packet
coming in. There's no problem with any other equipment but voicemaster
doesn't send voice at all.
Funny thing, i have an old version of OpenPhone, it's
AFAIK, you can use database lookups to fetch the internal caller id
and external caller id depending on the channel that is placing the
call. Then, simply set the corresponding caller id before placing the
call. Alternatively, which is what I currently do, since I don't use
account codes,
I am having issues with a TDM2400P. It appears when the ZAP channel dials
out, it randomly chops the first digit off of the number. I have tried
relaxdtmf=yes, turning up and down the txgain, turned off and on the echo
cancellation, generated new zaptel (with updated spinlock.h)...
I am at
On Thu, 06 Apr 2006 13:17:29 +0200, Tomislav Parčina [EMAIL PROTECTED]
wrote:
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
on a related note, we notice that if we've set atxfer = *1 in
features.conf
and blindxfer=#1, then attended transfers dont work. somehow, the Queue
app
On 03/31/06 08:24 Wai Wu said the following:
In Asterisk, what happens to the files when both legs of the call hangs
up? Is there a way to create a conference room on the flight? i.e.
without pre-defining the conference ID in meetme.conf.
look at the 'd' option to MeetMe.
--
Regards,
I've had a similar problem with CentOS and yummed kernels. The problem
seems to be that the zaptel doesn't quite know where to put the modules.
If you check the directory for your current kernel version, you'll see
they're not there.
I have fixed this in two different ways:
1) Per the wiki -
I'm attempting to use callprogress in my system,
and I'm having trouble. Callprogress always can tell if the line is
busy or ringing, but when the line is answered, the call does not get
bridged. Messages showing that "line is ringing" stop in the console and
if the called party hangs up,
Hello All
I am wondering whether you can increase the volume on the trunk port when it
is running on pure VoIP with no channels involved.
Sam
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Hi,
I'm using IPSwitchboard v 1.8.10, a sort of Operator Panel, to monitor my
Asterisk's extensions.
Recently I noticed that on the official site
(http://ipswitchboard.thorben.dk/), where I downloaded the software some weeks
ago, this project is no longer supported.
Is there anyone that can
Is there any way to define call parking parameters for different
contexts?
For example, if I have a client in context 100 and another client in
context 200, can they both define parking positions, say, from
701-710, where 701 in context 100 is different from 701 in context 200?
Or even
On 04/06/06 19:17 Tomislav Parèina said the following:
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
on a related note, we notice that if we've set atxfer = *1 in features.conf
and blindxfer=#1, then attended transfers dont work. somehow, the Queue app
captures the '*' and hangs
First, I'm not sure is this Asterisk or ooh323 channel problem.
It seams that I have solved (I do hope so!) deadlock problem with ooh323
(thanks to Sean and his patch). Now I have another one. It seams that some
channels stay open even they should not. This is what I see from CLI:
pbx*CLI show
Look at hints for Local Channel. That may be what you
are looking for.
Alex
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alessio
FocardiSent: Thursday, April 06, 2006 4:34 AMTo:
asterisk-users@lists.digium.comSubject: [Asterisk-Users] Hinting a
Welcome to the painful world of analog phone lines. Unless
you are using a digital line, there really is no true call progress detection
available. In many situations this is not a problem, where we see this the most
is when you are trying to ring a zip device and a zap channel at the same
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Jonathan k. Creasy wrote:
I apologize if this information is posted elsewhere. Unfortunately I
haven't found it yet if it is. I'm not familiar with the channel
counting features could you please explain? Also, how are you tagging
the phones to
Hi,
It will work - it is just a matter of the order in which the zaptel
driver for the particular card is loaded.
Just insert your card, load necessary driver and see /proc/zaptel/* - it
is self explanatory.
Ondrej
yusuf wrote:
Hi,
can i have a FXO/FXS card and a E1/T1 card in the same system.
i am also getting this warning since upgrading to 1.2 when running asterisk with -p param (realtime priority)On 4/6/06, Raymond Chen
[EMAIL PROTECTED] wrote:Tomislav Parčina wrote: In article
[EMAIL PROTECTED], [EMAIL PROTECTED] says... Hi What causes deadlock? Apr5 14:02:43 WARNING[2413]
Once upon a time there was an app called app_valetparking, and its big
brother SUPERvaletparking.
They both addressed that very senario. However, the brothers proved to
be a expensive load on the PBX as searching within and moving throughout
the Multiple parking lots required much time and
Hello friends,
I am using SIP on Asterisk 1.2.4. All my configurations are working perfectly
on a Welltech fxo box. But today I changed to an audiocodes MP104 fxo box. All
the sip signalling works fine but the noise is something like an alien
invasion, I mean, its completely outrageous. I
Ok, so multiple people have said that hinting is possible with chan_sccp
on the 7940/7960's and such, has anyone got this working? How do you go
about getting this to work?
I'd use the wiki, but it's link to the mailing list topic on that doesn't
work anymore :(
--
Aaron Daniel
Computer
You sound very poetic. Thanks for the info.
- Waldo
On Apr 6, 2006, at 10:27 AM, Alexander Lopez wrote:
Once upon a time there was an app called app_valetparking, and its big
brother SUPERvaletparking.
They both addressed that very senario. However, the brothers proved to
be a expensive load
But is there a way of doing this without a prefix?
because people should dial without prefixes: [EMAIL PROTECTED] , not like:
[EMAIL PROTECTED]
How can we make this without a prefix? something like:
if( !uri=~@mydomain.pt ){
forward the all to the Internet
}
:)
Thanks
Joao Pereira
Shad
Hi,
I wanted to use the same extensions for Pausing and
UnPausing queue members.
Is that a variable that is set up with the agent
status (on call, available, not logged, paused) so that I could use it to make
some logic in this extension?
exten =
Eric Buruschkin wrote:
I'm attempting to use callprogress in my system, and I'm having
trouble. Callprogress always can tell if the line is busy or ringing,
but when the line is answered, the call does not get bridged.
If the call is not bridged as soon as * is done dialing, then you have
Sam Tam wrote:
Hello All
I am wondering whether you can increase the volume on the trunk port when it
is running on pure VoIP with no channels involved.
No, there isn't any such settings.
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Is the Cisco 7960 capable of monitoring other extensions (hint status)
with a SIP implementation? Seems like it could, just can't find any
info on it...
Sean
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To
Hello to all
Can we put Asterisk in a company that has an ADSL connection with just
one public IP address? Because with just one public IP, Asterisk must
have a private (NATed) IP... but the idea is to make him dial other SIP
domains.
Can Asterisk work behing NAT, and still route calls to
Sadly, no. The SIP firmware on the Cisco phones doesn't support
subscribing to other lines. I heard chan_sccp does though.. now to
figure out how.
Aaron
On Thu, 6 Apr 2006, Sean Cook wrote:
Is the Cisco 7960 capable of monitoring other extensions (hint status) with a
SIP
Hello,
I just received what seems to be a nice SIP-DECT gateway but can't
make it work with asterisk. The manual is very unclear (written in
chinese english) and the web configurator is ambiguous as well.
Has anyone succeeded in making one of these babies work with * ?
info:
Are you using chan_sccp for you cisco implementation?
Aaron Daniel wrote:
Sadly, no. The SIP firmware on the Cisco phones doesn't support
subscribing to other lines. I heard chan_sccp does though.. now
to figure out how.
Aaron
On Thu, 6 Apr 2006, Sean Cook wrote:
Is the Cisco 7960
Hi all
Hmm, often when my Asterisk tryes to register, it get's the answer back:
Got SIP response 302 Moved temporarely (and an IP).
But it looks like it's not respecting this redirection and tryes again and
again to register to the server configured in sip.conf instead of the one the
SIP
Julio Arruda escreveu:
Paulo,
He is mentioning E1/PRI, so I assume the well known collect call on
E1/R2 thingie doesn't apply to him.
Julio,
I have 1 E1 from telefonica and 1 from Embratel. Telefonica has a better
deal for incoming calls (gave us more DIDs) but Embratel has better
rates.
Can you post your iax.conf?
On 4/4/06, Marco Mouta [EMAIL PROTECTED] wrote:
Password and username are ok.
On 4/4/06, Joshua Colp [EMAIL PROTECTED] wrote:
Marco Mouta wrote:
Hi all,
I've 2 * tryning to connect each other
Server A is already registred on server B
But server
ali asma wrote:
I have recompiled my zaptel drivers but I still get
the same error
--- Derek Whitten [EMAIL PROTECTED] a écrit :
ali asma wrote:
I modified the configuration but I still have the
same
error.
Please tell me in whach directory should I
execute:
modprobe zaptel
modprobe
Password and username are ok.
On 4/4/06, Joshua Colp [EMAIL PROTECTED] wrote:
Marco Mouta wrote:
Hi all,
I've 2 * tryning to connect each other
Server A is already registred on server B
But server B never registers in server A
I always get this:
Tx-Frame Retry[000] --
On Wednesday 05 April 2006 07:26, Eric ManxPower Wieling wrote:
We reboot all our Asterisk servers once per week if they have a TDM400P
in them. If we don't do that, then the TDM400P modules stop working.
I have *never* rebooted an Asterisk system because of the TDM400. Granted,
the driver
Hi,
What kind of problem happens?
Show your dialplan.
Daniel
On 4/4/06, Toke [EMAIL PROTECTED] wrote:
Hi Antonio,
What problems are you having with it? Which operator give you E1
connectivity??
If you want mail me directly and we will try to have it working if it is
possible.
Regards
If you have a temporary message set up, it always uses the temporary
greeting. If you want it to use the regular busy/unavailable messages,
you have to remove the temporary greeting.
Aaron
On Tue, 4 Apr 2006, Dov Bigio wrote:
Hi,
I know this has already been discussed here, but I still
Pavel Jezek wrote:
Hello Jeremy,
do you think, that adding features to original h323 channel is
perspective? is still maintained or will be replaced eg. with ooh323
(from asterisk add-ons)?
anyway I'm currently using original h323, it working prety fine for me
(with ooh323/oh323 I had
Ing. Oscar Andrés Carriles
I got a CPU hog of 100% running asterisk 1.0.9
The problem is caused by a single process capturing all available CPU in
one call. When this call hang up seldom others continue in normal
service.
I have all 30 SIP softPhones eyebean, 1E1 AFT101 Sangoma card signalling
Dov Bigio wrote:
When I call a mailbox in a context company is doesn't play my busy
message... It goes directly to the temp message...
Am I doing something wrong?
If you have a temp message, it is supposed to override your other messages.
___
voipstunt: Forbidden - wrong password on authentication for INVITE to
I have paid, the password was not changed, ... I have no idea why.
Is there anything what I can do to get this failed call over to
another provider, so that the user can complete the call?
(Dialstatus was an idea, but
Is there any way to query the number of people in a call queue from the
dialplan?
Our freephone provider has a feature where if we busy a call they record
the voicemail and email it to us. This enables us to divert calls to
them if our incoming lines start to get full. In order to do this we
need
On Wed, 2006-04-05 at 08:52 +0200, René Enskat [Teamware GmbH] wrote:
hi
i updated asterisk today via svn no i can'T start asterisk i get core
dumps.
i have to comment some modules then i can start:
noload = format_au.so
noload = format_mp3.so
noload = format_pcm_alaw.so.so
noload =
JP Carballo wrote:
Ronald Wiplinger wrote:
Insert this in astcc.agi; anywhere after the calls for it to load
and connect to the db.
if ($phoneno eq RESET_INUSE) {
setinuse($carddata-{number}, 0);
exit(0);
}
Thanks!
I use it here:
elsif ($phoneno eq BALANCE) {
Matt, it is the first time I hear positive about Sphinx.
Do you have a menu for the installation you did?
He's just exceptionally easy to please. :-)
Is there a problem with Sphinx that I have missed? So far it really
seems to be hitting the words right on.
Hi all, I'm sure something similar has been discussed, but one can only
wade through the archives for so long.
I'm setting up a T1 and my telco has a bunch of questions it wants me to
answer. I know much of the TE110p is configurable to do any of this, but
I wanted to know if there is an optimal
On 14:36, Tue 04 Apr 06, Anton Krall wrote:
Guys, if you define recording on queues.conf and also define a
monitor_filename var on your dialplna, you can record a queue call but,
isthere a way to do something with the file after the call ends? I need to
move the file to some other place but I
Hi,
I have been listening to Blue Box: The VoIP Security Podcast -
http://www.blueboxpodcast.com, and thought that SPIT could pose a
problem if not already one. Like to know if there are any OSS solutions,
within Asterisk or can integrates well with it, that focus in this area?
Regards
Andy Tan
Title: [Asterisk-Users] Anyone have a definitive list of Manager eventsper category?
hm, I have to try that. I am using for third party control
so the need to know all the events.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Josh
McAllisterSent: Tuesday, April 04, 2006
Title: [Asterisk-Users] Anyone have a definitive list of Manager eventsper category?
Hi,
After
apply patch and make clean; make install. Do I have to do a make sample to have
new asterisk running?
___
--Bandwidth and Colocation provided by
Don Pobanz escreveu:
Frame slips are NOT motherboard related!
I had problems with some combinations of motherboards, memory sizes and
linux kernel versions.
There are timing problems that also causes frame slips, like buffer
overruns or underruns, but these are software related.
--
Paulo
I have a worst issue for you... If your fax solution is ever going to
receive fax in Brazil, how would you block collect calls?
I have made a fax server solution with cheap Digium hardware that works
in Brazil (2 E1s).
--
Paulo
Adolfo R. Brandes escreveu:
Greetings, All-Knowing Asterisk
Lorentz Hinrichsen wrote:
I've had very poor results with the Digium cards, I am using a couple of the
new Sangoma ones now (they are cheaper and have hardware echo cancellation).
Which boards are cheaper _and_ have hardware echo cancellation?
___
Kind of like DND, but some phones seem ok. They all give the message
even if it rings through that the person is on the phone even if they
are not. Normally it says that when they are on the phone, and it says
unavailable if they are not on the phone but never answer...
Almost like astericks
I'm using MixMonitor. Be aware that some people encounter a bug where MixMonitor stops recording at random (see http://bugs.digium.com/view.php?id=6457). There are a couple of working patches for it.
Thanks.On 4/3/06, Wai Wu [EMAIL PROTECTED] wrote:
Hi all,I am setting up a script to record all
Marco Mouta wrote:
Password and username are ok.
On 4/4/06, Joshua Colp [EMAIL PROTECTED] wrote:
Marco Mouta wrote:
Hi all,
I've 2 * tryning to connect each other
Server A is already registred on server B
But server B never registers in server A
I always get this:
Tx-Frame Retry[000] --
a mirror to soft-switch can be found at:
http://zarzamora.com.mx/mirror/www.soft-switch.org/
regards
On 4/3/06, Steve Underwood [EMAIL PROTECTED] wrote:
Hi Dennis,
Update to libmfcr2-0.0.3 pre9. I made a slip in pre8. Sorry.
Steve
Dennis Nacino wrote:
Hi,
I have three R2
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