- Original Message -
From: Chandra [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, December 10, 2003 10:37 PM
Subject: * with RADIUS
hi,
i have been looking for implementations of asterisk with RADIUS which
would
ease for accounting purposes. where can i find more
Chandra wrote:
hi,
i have been looking for implementations of asterisk with RADIUS which
would
ease for accounting purposes. where can i find more information on this?
help.
cm
Explain why you think you really need RADIUS Accounting? Why not talk
right to the database itself
On Wed, 2003-12-10 at 19:44, Rich Adamson wrote:
In a test system I can take out half the RAM, slow the CPU clock
or run the CPU without the cooling fan and just measure what
happens. Yes, stupid do do those things in a system people
are depending on.
Agreed 100%. If you want
On 10 Dec 2003 21:52 Leif Madsen wrote:
On Wed, 2003-12-10 at 10:47, Steve Underwood wrote:
Hi all,
Does anyone know of any work in progress on IAX based telephony for
PDAs? Putting iaxcomm on a Zaurus or iPAQ, for example.
If someone where doing this, I would be the most ecstatic,
This is great , thanks...
One more question...
How can I handle the return codes ? I mean , I want to see if the call was
successfull or not...
I can call my own application once the call is answered... There is no way
where I can call another application if the call was not answered ?
Regards
Hi,
I have a strange problem trying to update Asterik on one of my boxes.
I have done the following:
- delete all the old source files
- download the new file using:
cvs checkout zaptel zapata libpri asterisk
- compiling the new source file using:
make clean ; make install
Everything is
Hi,
I'm trying to install * with TE410P and Mandrake 9.0
for E1.
My first problem is that I can not find the
openssl-devel. Do anyone know from where I could
download it ?
(I have the openssl-0.9.6g-1mdk installed)
Also, is there any zaptel.conf sample for E1 ?
Thanks,
MarinBlu
Thinking about this problem I would like to point out the root cause of all this:
*** The Asterisk open source PBX is a success story ***
We are a growing crowd. New users keep joining the list all the time, experimenting,
installing, getting along.
Some of them are used to Open
Seems very strange. Check which asterisk you start.
Stop the asterisk and start the binary in the source directory with ./asterisk and the
connect
to that version and check CVS date.
If it's different, then you have several versions in your path.
My 2 cents...
/O
Rich Adamson [EMAIL PROTECTED] said:
[2.2ghz Celeron]
In an idle condition (no calls being processed), top is the heaviest
app. Placing a single asterisk demo call from a sip phone (forcing iax2
to Digium) causes asterisk to bump towards the top at about 0.3% cpu
utilization with an occasional
On Wed, 2003-12-10 at 16:44, Chris Albertson wrote:
--- Steven Critchfield [EMAIL PROTECTED] wrote:
On Wed, 2003-12-10 at 12:47, Trench Shoring wrote:
I have been reading asterisks and everything I can get my hands on
for the
past week. I want to know what class processor is the bare
Dear ALL:
Following Mark's request: if all those who want to meet in Paris on 20th
could please email me asap. Please confirm if there is a time you would not
be able to make it. I will then judge numbers and make a posting/reply to
you all on Sunday with time/venue etc.
[EMAIL PROTECTED]
Thank
Hi,
- Original Message -
From: Olle E. Johansson [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, December 11, 2003 10:44 AM
Subject: Re: [Asterisk-Users] * CVS checkout does not work on one box
Seems very strange. Check which asterisk you start.
Stop the asterisk and start
On Thu, 2003-12-11 at 03:23, Dan wrote:
Hi,
I have a strange problem trying to update Asterik on one of my boxes.
I have done the following:
- delete all the old source files
- download the new file using:
cvs checkout zaptel zapata libpri asterisk
- compiling the new source file
On Thu, 2003-12-11 at 02:26, marin blu wrote:
Hi,
I'm trying to install * with TE410P and Mandrake 9.0
for E1.
My first problem is that I can not find the
openssl-devel. Do anyone know from where I could
download it ?
(I have the openssl-0.9.6g-1mdk installed)
This is where you would
Stephen Wingfield wrote:
Dear ALL:
Following Mark's request: if all those who want to meet in Paris on
20th could please email me asap. Please confirm if there is a time
you would not be able to make it. I will then judge numbers and make
a posting/reply to you all on Sunday with time/venue
Maybe you need to take a step back and describe what you are trying to
accomplish. If you hit dead ends, there is possibly another whole route
to solve the problem that doesn't include the dead end.
On Thu, 2003-12-11 at 02:16, Alexandru Coseru wrote:
This is great , thanks...
One more
Ok , let me explain it more clearly..
I have a message to deliver on someone's phone..
Actually , I'm making the call , playing the sound and that's it...
But , if the person is not answering , or the network is busy , etc... , I
have to know it , in order to put the call back to queque and
Hello..
I'm having another problem right now...
There is a way to force the execution of an application into an extension ,
even if the user hangs up before reaching it ?
For instance:
exten = 500,1,Answer
exten=500,2,Play(prompt1)
exten=500,3,Play(prompt2)
exten=500,4,Play(prompt3)
On Wednesday 10 December 2003 05:44 pm, Chris Albertson wrote:
equipment but I'm still looking to reduce power, heat, noise and
space to the bare minimum. No need to buy a CPU that burns
120W of power if you can use a one that uses 45W and
lets you get rid of one of the fans. Same with
Hi,
- Original Message -
From: Leif Madsen [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, December 11, 2003 11:07 AM
Subject: Re: [Asterisk-Users] * CVS checkout does not work on one box
On Thu, 2003-12-11 at 03:23, Dan wrote:
Hi,
I have a strange problem trying to
WHY NOT use an AGI script and create a callback function !
That work fine for me!
-Areski
On Thu, 2003-12-11 at 10:49, Alexandru Coseru wrote:
Hello..
I'm having another problem right now...
There is a way to force the execution of an application into an extension ,
even if the user hangs
On Thu, 2003-12-11 at 03:39, Alexandru Coseru wrote:
Ok , let me explain it more clearly..
I have a message to deliver on someone's phone..
Actually , I'm making the call , playing the sound and that's it...
But , if the person is not answering , or the network is busy , etc... , I
have
On Thu, 2003-12-11 at 03:49, Alexandru Coseru wrote:
Hello..
I'm having another problem right now...
There is a way to force the execution of an application into an extension ,
even if the user hangs up before reaching it ?
For instance:
exten = 500,1,Answer
On Thu, 2003-12-11 at 03:53, Dan wrote:
Hi,
- Original Message -
From: Leif Madsen [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, December 11, 2003 11:07 AM
Subject: Re: [Asterisk-Users] * CVS checkout does not work on one box
On Thu, 2003-12-11 at 03:23, Dan wrote:
Ok..
Can u give some samples and/or some docs ?
Thanks a lot
Alex
- Original Message -
From: Areski [EMAIL PROTECTED]
To: Asterisk-Users Mailing-list [EMAIL PROTECTED]
Sent: Thursday, December 11, 2003 12:00 PM
Subject: Re: [Asterisk-Users] Forcing to exec an app into an extension
I thought that this is the best solution..
I've already made some small apps for * , and it is not so hard indeed...
The question was asked as a general thing , not related to playing
prompts I could have some voicemail apps there or any other
application...
Thanks a lot
Alex
-
Hi,
- Original Message -
From: Steven Critchfield [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, December 11, 2003 12:10 PM
Subject: Re: [Asterisk-Users] * CVS checkout does not work on one
box -solved now
..
It doesn't have to do with the directory, just the fact that the
Dan schrieb:
Hi,
- Original Message -
From: Steven Critchfield [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, December 11, 2003 12:10 PM
Subject: Re: [Asterisk-Users] * CVS checkout does not work on one
box -solved now
..
It doesn't have to do with the directory, just the
On Thursday 11 December 2003 04:53 am, Dan wrote:
Oops.. this is still the old one...
I have deleted the files using the following commands from the src
directory:
rm -r -f ./zaptel/*
rm -r -f ./zapata/*
rm -r -f ./libpri/*
rm -r -f ./asterisk/*
You need to start showing the output
On Thu, 2003-12-11 at 04:32, Dan wrote:
Hi,
- Original Message -
From: Steven Critchfield [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, December 11, 2003 12:10 PM
Subject: Re: [Asterisk-Users] * CVS checkout does not work on one
box -solved now
..
It doesn't have
Hi,
- Original Message -
From: Steven Critchfield [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, December 11, 2003 12:49 PM
Subject: Re: [Asterisk-Users] * CVS checkout does not work on one box-solved
now
On Thu, 2003-12-11 at 04:32, Dan wrote:
Hi,
- Original
Hi,
I'm trying to use iaxcomm. I can place a call from the softphone, but
when I place a call to it, when I answer I get ...
NOTICE[16401]: File channel.c, Line 1094 (ast_read): Dropping
incompatible voice frame on IAX2[paulohm]/3 of format GSM since our
native format has changed to ALAW
My
hi all
Using a vanilla 2.4.23, I try to setup the USB FXS interface to add an
old gray 1960 rotary dial phone for a demo. But it doesn't work :(
Can someone decipher the below messages and help me out here?
Please ask if more info is needed
roy
usb.c: kusbd: /sbin/hotplug add 4
usb.c: kusbd:
On 10/12/03 20:07, Steven Critchfield wrote:
postfix and exim should provide a sendmail link or binary that should be
command line compatible as the original for sending mail. I don't know
about ssmtp.
SSMTP does indeed provide command line sendmail compatibility (within
reason).
Regards,
Explain why you think you really need RADIUS Accounting? Why not talk
right to the database itself and save yourself that unneeded
complication and points of failure.
For the exact same reasons RADIUS exists in the first place? Consistency?
The ability to change authentication backends
A few months ago I discovered *. Recently have taken more interest
in it, since I have to install phones into my wife's small office.
Hope that my questions weren't discussed recently and are not to
dumb.
The current requirements for * consist of a single incoming ISDN
BRI line (currently
What I would like to do now is to refresh the asterisk.org website. We
need to add more visible pointers to where information can be found,
adopt it to the crowd of new users that join our community all of the
time. One special thing we have to add, is information on how an open
source
[EMAIL PROTECTED] wrote:
I've been running 0.5.0, which is dated sometime in September of this
year and I've noticed a couple of new features in more recent code
that
I'd like to use, but am hesitant to go w/ CVS code. My system is not
exactly a production system, it's mostly test, but I'm
Andrew Kohlsmith wrote:
Explain why you think you really need RADIUS Accounting? Why not talk
right to the database itself and save yourself that unneeded
complication and points of failure.
For the exact same reasons RADIUS exists in the first place?
RADIUS was created to authenticate
Hi,
I have just started using asterisk some time ago. I
am trying to write some u-law audio data to the
channel using ast_write(chan,myframe) but it gives me
a segmentation fault. I will be glad to receive any
help.
I am sending the application file as an attachment.
Actually I am interested in
I new to * but getting a grip on it (I hope)
On computer hardware that's a far different story I have been in the
computer business since 1984 I have built, sold and serviced a lot of PC's
in that time and think I am qualified to give hardware advice.
I have built three * boxes
1.
PIII 1000
Sorry to bother again, but what is the syntax of a dchannel? I'm trying
1, zap/1, ... without success
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark Spencer
Sent: quarta-feira, 10 de dezembro de 2003 19:10
To: [EMAIL PROTECTED]
Subject: Re:
Hi Folks
Back at the beginning of November Mark Spencer replied to a question
about using the zaptel drivers on 2.4.21 kernels or above in
combination with data, ie CONFIG_ZAPTEL_NET:
http://www.mail-archive.com/[EMAIL PROTECTED]/msg14869.html
Has anybody tried this patch? Does it work? Does
On Thu, 2003-12-11 at 03:26, Tilghman Lesher wrote:
On Wednesday 10 December 2003 23:15, Steven Thomas wrote:
Hi All,
I have add the below error ever since installing and running with *
for the past 6 months. It only occurs on calls from * to a H.323
gateway. I am using chan_h323.
Does anyone know what would be involved in making
Asterisk work as a voicemail system in a Centrex
environment? We have a Centrigram voicemail system
that belongs in the Smithsonian. There are analog
lines coming into the box and a 56KB data feed from
the phone company's switch.
Peter
On Wed, 2003-12-10 at 20:19, Dorian Gray wrote:
working fine here as well. was not able to install manually since the
sipura site had only a windows .exe last time I checked; however, it got
upgraded to 1.0.18 when I signed up for the free month of voicepulse
service and they provisioned
That would do the job
I have just been using Secure CRT for so long I have gotten use to it
James Schenck
Egraph Design Inc.
Arkansas Online Internet Services
(870) 857-3287
IAXTEL (700) 857-3287
[EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
Talking to myself ... ;-)
Solved this by ...
disallow=all
allow=gsm
;allow=ulaw
;allow=alaw
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paulo
Mannheimer
Sent: quinta-feira, 11 de dezembro de 2003 09:02
To: [EMAIL PROTECTED]
Subject:
Hi Dan,
this'll update the version number:
/usr/src/asterisk/make update
Cheers, Philipp
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Hi,
We'he bought 4 g729 licenses. By the first time i installed, a ran the
Registration program out of the /usr/src/asterisk directory.
This worked fine, but after a few minutes, I read some docs, and
reliaze that I need to run Registration from /usr/src/asterisk
FWIW,
Most Telcos (At least for us CLECs) it becomes more cost effective for
us to run a DS1 to a customers location if they have 5 CO lines or more.
I would hunt around for a company that would do that for you.
If you get the T100P Card @ $500 it's cheaper than 6 $100 FXO cards.
Telco can
The following is from zapata.conf.sample:
; Ring groups (a.k.a. call groups) and pickup groups.
If a phone is ringing
; and it is a member of a group which is one of your
pickup groups, then
; you can answer it by picking up and dialing *8#.
For simple offices, just
; make these both the same
;
Hi!
1. I plan to buy some el cheapo (relatively, since VoIP phones are
still quite expensive here) Planet VoIP phones for internal
extension: http://www.planet.com.tw/product/product_dm.php?
product_id=192menu_id=3
Anybody have experience with those and *?
Interesting, I've never seen
On Thu, 2003-12-11 at 16:02, Philipp von Klitzing wrote:
Hi!
1. I plan to buy some el cheapo (relatively, since VoIP phones are
still quite expensive here) Planet VoIP phones for internal
extension: http://www.planet.com.tw/product/product_dm.php?
product_id=192menu_id=3
Anybody
Hello--
For all those who inquired as to the gsm files for the sound prompts for
the telemarketing Torture menus I put on the wiki, and also all those
who had too much dignity to request them, I humbly submit that I have
added a link to my version of the sound prompts to the wiki page, and
/dev/zap/1
Martin
On Thu, 11 Dec 2003, Paulo Mannheimer wrote:
Sorry to bother again, but what is the syntax of a dchannel? I'm trying
1, zap/1, ... without success
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark Spencer
Sent: quarta-feira,
On Thu, 11 Dec 2003 07:30:27 -0500, Jeremy McNamara wrote:
Andrew Kohlsmith wrote:
The ability to change authentication backends without having to touch *?
You can do this already.
Passing off authentication to a third party?
You can do this already.
Ah, pardon my ignorance Jeremy,
Hi,
I have tried to install and configure the FAX app using the steps described
in several mails on this list.
When a FAX call arrives on X100P, the fax extension is executed, which looks
like:
exten = fax,1,RxFax(/var/spool/asterisk/incoming/${CALLERIDNUM}.tif)
exten = fax,2,Hangup
It tries to
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Jeremy McNamara
Sent: Thursday, December 11, 2003 2:19 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Re: * with RADIUS
[...]
Explain why you think you really need RADIUS Accounting?
A particularly analogy-adept person the other day described Asterisk
as the stem cell of VoIP, meaning that he had seen Asterisk grow
into pretty much whatever task was at hand as long as it was VoIP or
telephony-related. I think I like that phrase almost as much as the
apache of telecom,
On Dec 11, 2003, at 1:38 AM, Steven Critchfield wrote:
lots deleted
If you look over the list again, you will see that questions that get
ignored tend to get the entire list flamed for not being helpful.
No one here tries to run people off but you will rarely see a message
where we treat any one
snip
I know this has come up before, and in a perfect world, where * was the
primary app, you don't need RADIUS. In enterprise environments where
RADIUS accounting is already embedded into other aspects of the
workflow, it would be beneficial.
Understand* boxes are in real live actual
Hello,
We use RADIUS with a MySQL backend database server for dialup
authentication.
Because our accounting system is XML based, I would prefer to use one AAA
(i.e RADIUS)
server to provision and validate our VoIP UA's. LDAP is another AAA solution
we are looking at using.
Of course a direct SQL
Juan J. Sierralta P. wrote:
On Wed, 2003-12-10 at 20:19, Dorian Gray wrote:
working fine here as well. was not able to install manually since the
sipura site had only a windows .exe last time I checked; however, it got
upgraded to 1.0.18 when I signed up for the free month of voicepulse
I just got a new in box phone from ebay, i called cisco this morning to
get a service contract on it, they told me I would have to ship it to
them, for them to recertifiy it.
Does anyone have any idea how to go and get a service contract from cisco,
so that I can download the firmware?
(dammit, sent to jeremy directly last time, sorry)
For the exact same reasons RADIUS exists in the first place?
RADIUS was created to authenticate Dialup users
a common authentication and accounting system for dialup users, yes. It's
grown into a common auth/acct system for user-based
- Original Message -
From: Doug Shubert [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, December 11, 2003 11:48 AM
Subject: Re: [Asterisk-Users] Re: * with RADIUS
Hello,
We use RADIUS with a MySQL backend database server for dialup
authentication.
Because our accounting
Hello.
Is this normal. Or does it mean there is a problem ?
-
stop now
Beginning asterisk shutdown
Executing last minute cleanups
== Destroying any remaining musiconhold processes
Yuck! Error in buffer handling...: Connection reset by peer
Yuck! Error in buffer
- Original Message -
From: Doug Shubert [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, December 11, 2003 11:48 AM
Subject: Re: [Asterisk-Users] Re: * with RADIUS
Hello,
We use RADIUS with a MySQL backend database server for dialup
authentication.
Because our accounting
I've come to the same conclusion. There seems to be very little specific
info in the archives about what vendor's products work and what doesn't.
Could someone post what FXO sip gateways work great with * ?. It'll save
return shipping charges :-)
Thanx
John Breeden
Hawaii
Michael,
Can someone give me an idea exactly what things are intended to be tested
via RADIUS, or some other AAA system?
Are we talking about building SIP/IAX/H323 entries from RADIUS?
This is where the PAM system I developed for * comes into play. I've got
most of it working at the moment, but I'm
Good day,
I'm just finished installing Asterisk and I run the demo context (in fact, I did
not changed the extensions.conf file).
I call it in G.711 u-law.
The problem is that Asterisk does not seem to handle the digits I punch on the phone
to get to sub-menus. The file keeps on
That's all musiconhold mp3 stuff(mpg123). If you deactivate musiconhold then
it goes away.
The inherant problems with mpg123 can rarely cause crashes and very often
cause lots of mpg123 processes running on your machine. I got rid of it and
just use half-hour-long gsm files for on hold extensions
My system has been down for over 2 hours with this same type of message!
It's a major problem!
- Original Message -
From: marrandy [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, December 11, 2003 12:12 PM
Subject: [Asterisk-Users] Yuck! Error in buffer handling
Hello.
Is
I have gotten the MediaTrix 1204 with the Sip configuration working! It took
a while to get it going but it sure works after you set it up! It has 4 FXO
ports.
- Original Message -
From: John Breeden [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, December 11, 2003 12:20 PM
Just for a lark I typed in the error message below (from you logs) into
google:
http://www.mail-archive.com/[EMAIL PROTECTED]/msg13622.html
This might be of use to you.
Hope it helps - Jon Carnes
On Thu, 2003-12-11 at 11:03, Dan wrote:
Hi,
I have tried to install and configure the FAX app
Is the party at the Paris Hilton?
sorry, couldn't help it...
--
Bob Knight
[-w] the work option
[EMAIL PROTECTED]
925-449-9163
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Is there a way to increase the number of retries or the time to help
with this?
WARNING[40966]: File chan_sip.c, Line 462 (retrans_pkt): Maximum retries
exceeded on call [EMAIL PROTECTED] for
seqno 103 (Request)
WARNING[40966]: File chan_sip.c, Line 462 (retrans_pkt): Maximum retries
exceeded
- Original Message -
From: Doug Shubert [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, December 11, 2003 11:48 AM
Subject: Re: [Asterisk-Users] Re: * with RADIUS
Hello,
We use RADIUS with a MySQL backend database server for dialup
authentication.
Because our
Anyone use this provider before ?
Comments on service ?
I have emailed them for more info but never heard back.
(I think the owner occasionally posts on this list.)
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Hi,
- Original Message -
From: Jon Carnes [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, December 11, 2003 7:53 PM
Subject: Re: [Asterisk-Users] FAX application does not work for me
Just for a lark I typed in the error message below (from you logs) into
google:
It's a hardcoded value in channels/chan_sip.c:
#define MAX_RETRANS 5 /* Try only 5
times for retransmissions */
Change this value and then recompile.
Christian
On Thursday 11 December 2003 10:03, Scott England wrote:
Is there a way to
On Thursday 11 December 2003 12:53 pm, Bob Knight wrote:
Is the party at the Paris Hilton?
sorry, couldn't help it...
--
Bob Knight
Bob...I'm really surprised at you !!!
I thought you would have said, 'Is the party in Paris Hilton'
lol ;-)
--
Q: What's hard going in and soft
Does this card only work as PRI or can it be used like a standard T-1 wired
to a PSTN Switch?
TIA
-Seth
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On Thursday 11 December 2003 12:12 pm, marrandy wrote:
*CLI show version
Asterisk CVS-12/10/03-11:49:50
--
Just curious.
An addition.
When I'm picking up the extension on an incoming call, I get Three tones.
Am I dreaming, or has this started to happen after my
It can be either.
Does this card only work as PRI or can it be used like a standard T-1
wired
to a PSTN Switch?
TIA
-Seth
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Is there a way to increase the number of retries or the time to help
with this?
WARNING[40966]: File chan_sip.c, Line 462 (retrans_pkt): Maximum retries
exceeded on call [EMAIL PROTECTED] for
seqno 103 (Request)
WARNING[40966]: File chan_sip.c, Line 462 (retrans_pkt): Maximum retries
exceeded on
Hello,
I am in Argentina configuring an E1 in R2 and have some inconveniences.
When the call takes the salient line, I receive an error of signaling.
In * I have loaded the following:
- libr2 installed
- The [ar] in zonedata.c and indications.conf is configured
- zaptel.conf o:
Hi!
I am getting the following error message:
Got SIP response 403 That is ugly -- use From=id next
time (OB) back from 195.37.77.101
I'm not quite sure what that means. Does anybody know
what I might have done wrong?
Here is my configuration:
sip.conf
register = account:[EMAIL
I have asterisk running as a voicemail system off of our Merlin Legend
switch. We replaced our old Audix Voice Power (when the power supply fan
died and burned it up) with asterisk a week ago. Many thanks to those who
provided information about integrated VMI on the legend.
The Audix system
Hey,
--- Philipp von Klitzing
[EMAIL PROTECTED] wrote:
Hi!
I am getting the following error message:
Got SIP response 403 That is ugly -- use From=id
next
time (OB) back from 195.37.77.101
I'm not quite sure what that means. Does anybody
know
what I might have done wrong?
1. I plan to buy some el cheapo (relatively, since VoIP phones are
still quite expensive here) Planet VoIP phones for internal
extension: http://www.planet.com.tw/product/product_dm.php?
product_id=192menu_id=3
Anybody have experience with those and *?
Interesting, I've never seen
In an attempt to reduce bandwidth usage, I tried forcing my Asterisk to
use Speex. I did a disallow=all then an allow=speex. The crazy
thing is, it didn't reduce the bandwidth usage at all!
I can do an IAX2 show channels and it shows the call being in format 512
(Speex, right)?
Then I
It means that the username in From and the username in digest
credentials are different.
The reason for this test is that we do not want our users to pretend
that they are somebody else. Without this test it would be possible to
put [EMAIL PROTECTED] in From and all phones will display it,
Is it possible to play live streams of audio for the hold music instead
of just MP3 files? Like some online radio stations or something?
Jonathan
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[EMAIL PROTECTED]
Hello!
We are having an interesting problem with the queue. What is happening
is that no matter how many agents are logged into the queue, only one
phone will ring at one time. So, for example, if we have two agents in
the queue and two incoming calls. The first incoming call will ring on
one
strategy=ringall
in queues.conf
matteo
Il gio, 2003-12-11 alle 23:18, Derek Barber ha scritto:
Hello!
We are having an interesting problem with the queue. What is happening
is that no matter how many agents are logged into the queue, only one
phone will ring at one time. So, for example,
That does get both phones ringing, however that is not the solution
problem we are having. We need the queue to work in leastrecent mode.
If there is only one call in the queue only one agent's phone should
ring. However, if there is two calls in the queue then two agent's
phones should ring.
so, from queues.conf.sample
A strategy may be specified. Valid strategies include:
ringall - ring all available channels until one answers (default)
roundrobin - take turns ringing each available interface
leastrecent - ring interface which was least recently called by this
queue
fewestcalls -
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