Network Configurations
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Hi List.
I'm in need of something that will allow me to analyze cdr details
either via .csv or mysql that will give me call durations as well as
call
DumpChan (it's there in 1.2 as well) would be great, if it were a manager
command where you can choose the channel to dump and not a diaplan
function that outputs the current channel config to the CLI.
l.
In data Wed, 18 Apr 2007 02:30:09 +0200, Philipp von Klitzing
[EMAIL
Well, the larger the better :)
l.
In data Wed, 18 Apr 2007 04:15:28 +0200, Melcon Moraes
[EMAIL PROTECTED] ha scritto:
How large is large for you?
[]'s
MM
-Original Message-
From: Lenz [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
Manolet Gmail wrote:
Hi to all! i have installed asterisk 1.4.2 and asterisknow from the
digium svn repository, when i was installing i select using menuselect
utility the spanish voice lenguage pack. everythink is ok but i dont
know how or where to tell asterisk to use the spanish as the
callmanager can also be running in ios firmware in router (callmanager
express), with near all funcionality as server version...
Adam KOSA wrote:
Antonopoulos Angelos wrote:
Thanks for your help..But i dont know yet whether is CCM embeded on
cisco 2851 or it is an extra element?
yes and it is still set to speech
I've even tried to port the old patch here
http://bugs.digium.com/view.php?id=6251 to the system with no luck
robb
Melcon Moraes wrote:
Have you tried:
exten = s,n,SetTransferCapability(DIGITAL)
?
[]'s
MM
-Original Message-
From: robert
Yuan LIU wrote:
My dialplan looks like this:
exten = 0900900941,1,Goto(smsmotx,${CALLERIDNUM},1)
exten = 0794998990,1,Goto(smsmotx,${CALLERIDNUM},1)
How do callers get into these extensions?
They're specified on the smsq command, e.g.:
smsq --concurrent=3 --mo
Hi all,
lets say I've registered at several Sip-Providers. Provider A offers
best rates but is often too busy to get a line. Sip Provider B is stable
(but more expensive). The asterisk box has a high call volume therefore
problems at provider A will get obvious after a few calls stalled. In
On Wed, 18 Apr 2007 09:04:22 +0200, Knud Müller wrote:
I
think it can be done by using the dialplan and the database to store the
statistical information but maybe there is an easier way that integrates
better with asterisk!?
i dont think you'd even need a database with statistics. just
On 2007-04-17 00:53:56 -0700, Dinesh Nair [EMAIL PROTECTED] said:
On Mon, 16 Apr 2007 20:14:40 -0700, Martin Joseph wrote:
The phone no longer registers with asterisk, although it displays the
little icon as though it has, and it doesn't even seem to try to pass
calls to asterisk...
So, I
Hello Dan,
What version of Asterisk are you using? I've had recording
working with SVN before 1.4, the 1.4 betas and currently 1.4.1.
*** Update ***
Recordings are tied to a moderator joining the conference at this
time. I may need to change that based on feedback/requests to
do so.
***
Hi Moises,
the Asterisk SVN-branch-1.4-r60989 make a change in the
ast_channel_alloc function:
This is a big improvement over the current CDR fixes. It may still
need refinement, but this won't have as many folks bothered.
here the patch for chan_unicall.c ;p
--
Humberto Figuera - Using Linux
Tnaks for your answer. Sorry, if I'm missing something obvious here.
Yes, it's asterisk 1.4. I've configured a user entry in users.conf. One
of the lines is context = numberplan-custom-1. I suppose that should
make that user use the dialplan context [numberplan-custom-1]. I have
Hi Moises,
the Asterisk SVN-branch-1.4-r60989 make a change in the
ast_channel_alloc function:
This is a big improvement over the current CDR fixes. It may still
need refinement, but this won't have as many folks bothered.
here the patch for chan_unicall.c ;p
--- chan_unicall.c.orig
Hi,
have you tried different values of callerid? Maybe setting
*useincomingcalleridonzaptransfer* to yes can help you.
Giorgio Incantalupo
OCOSA ListAcc wrote:
Hello,
When I upgraded a while back the caller ID stop working I have tried
everything and searched the lists no answer. Please
Hello
I'm about to order a GrandStream GXP-2000 and a Linksys SPA-921
I'd like to have some user feedback about how those phones perform, and
whether their LCD screen displays both the caller ID name and number (The
GrandStream BT-100 only displays numbers, which isn't very helpful).
Thank
While I can't speak for the Linksys SPA-921, I /can/ comment on the
Grandstream GXP-2000.
We're running half a dozen of these at the moment, primarily for
testing. I can confirm that the LCD display /does/ display both caller
name and number - assuming of course that both are presented.
Hi,
are you using PoE or power supplies?
As power supllies usually are not grounded it could be that it's comming
from the power source.
You could try using a grounded PoE switch or probably a power backup to test
if this is the case.
Cheers
Tim
On 3/30/07, Louis-David Mitterrand [EMAIL
Dinesh Nair wrote:
On Wed, 18 Apr 2007 09:04:22 +0200, Knud Müller wrote:
I
think it can be done by using the dialplan and the database to store the
statistical information but maybe there is an easier way that integrates
better with asterisk!?
i dont think you'd even need a
I have experience with both. GXP is a great phone for its low price and it
has all the features of the IP phones. It doesn't have any considerable
issues with it. On the other hand Linksys 921 is superior in voice quality,
look, and TFTP support but limited in features, like limited line
Gilles Ganault wrote:
Hello
I'm about to order a GrandStream GXP-2000 and a Linksys SPA-921
I'd like to have some user feedback about how those phones perform,
and whether their LCD screen displays both the caller ID name and
number (The GrandStream BT-100 only displays numbers, which isn't
spa-922/942 has backlighted display, inline power (PoE), internal switch,
audio gain/attenuation can be tunned,
works great in bussines environment (voice vlan negotiation through cdp
from ci$co switch), solid design, robust chassis
lack of features like programable buttons for pickup or busy
On Wed, 18 Apr 2007, Rob Hillis wrote:
While I can't speak for the Linksys SPA-921, I /can/ comment on the
Grandstream GXP-2000.
We're running half a dozen of these at the moment, primarily for testing. I
can confirm that the LCD display /does/ display both caller name and number -
Per Jessen wrote:
Yuan LIU wrote:
My dialplan looks like this:
exten = 0900900941,1,Goto(smsmotx,${CALLERIDNUM},1)
exten = 0794998990,1,Goto(smsmotx,${CALLERIDNUM},1)
How do callers get into these extensions?
They're specified on the smsq command, e.g.:
smsq --concurrent=3 --mo
On Apr 18, 2007, at 6:50 AM, Rob Hillis wrote:
We've had the very occasional problem with the phone locking up,
but nothing overly serious.
Are you using DHCP on the GXPs that are locking up?
I have one and it would lock up almost every night requiring the
power to be pulled in the
On Wed, Apr 18, 2007 at 01:04:31PM +0200, Tim Koehler wrote:
Hi,
are you using PoE or power supplies?
As power supllies usually are not grounded it could be that it's comming
from the power source.
We are using PoE
You could try using a grounded PoE switch or probably a power backup to
Greetings from sunny Malaysia! This is a reminder that the Call for
Papers for the upcoming HITBSecConf2007 - Malaysia is closing on the 1st
of May.
HITBSecConf2007 - Malaysia is set to take place from the 3rd till the
6th of September in Kuala Lumpur. Our event last year attracted over 600
Salvatore, most, if not all VoIP providers support LNP. We do.
On 4/17/07, Salvatore Giudice [EMAIL PROTECTED]
wrote:
Can anyone recommend a VoIP provider who supports LNP? I need to move to
a new provider for inbound calling and I want to keep my current numbers. My
current provider is a
Hello,
I've got various phones (mostly SPA-922) behind NAT registered to
Asterisk. I've set nat=yes and canreinvite=no, and everything seemed to
work great with 1.2.17. After upgrading to 1.4.2 using users.conf and
macro-stdexten my spa-922 can't call other extensions.
-- Executing [EMAIL
Per Jessen wrote:
Per Jessen wrote:
OK, part of the confusion is now clearing up. But I'm not getting
much further. When I try to send an SMS, I see the call going
through, but no SMS is ever sent.
This is a bit of what I see in the debug output: (this is sending a
longer message,
Hello list,
QueueMetrics 1.3.4 has been released today. Among other features, it
provides realtime cluster monitoring through the manager API and, by
popular demand, user defined time intervals in the daily call breakdown.
You can find the latest version at http://queuemetrics.com and
hello,
im having trouble with asterisk with medium load, it seems im running out of
files, here is a chunk of the logs with grep \(file\|pipe\):
Apr 18 15:40:46 WARNING[11644] res_agi.c: Unable to create toast pipe: Too
many open files
Apr 18 15:40:46 WARNING[11574] channel.c: Channel
Am Mittwoch, den 18.04.2007, 13:18 +0200 schrieb Knud Müller:
Dinesh Nair wrote:
On Wed, 18 Apr 2007 09:04:22 +0200, Knud Müller wrote:
I
think it can be done by using the dialplan and the database to store the
statistical information but maybe there is an easier way that integrates
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi robb,
Have you just seen the bearer capability in asterisk or is the call nat
working? I've seen that a digital call shows up as speech.
You are using Zap? Or are you using mISDN? Cause there you have to set
an extra parameter in the dial
Try adding userscontext = numberplan-custom-1 to the [general]
section of extensions.conf to see if that helps
regards,
Drew
dima wrote:
Tnaks for your answer. Sorry, if I'm missing something obvious here.
Yes, it's asterisk 1.4. I've configured a user entry in users.conf. One
of the lines
Gilles Ganault wrote:
I'm about to order a GrandStream GXP-2000 and a Linksys SPA-921
I'd like to have some user feedback about how those phones perform,
and whether their LCD screen displays both the caller ID name and
number (The GrandStream BT-100 only displays numbers, which isn't very
Asterisk 1.4
I have strategy= leastrecent and autofill = yes
I have 2 agents, one is answering a call and the other is free and have some
calls waiting in the queue.
Only when the first agent hangup the second agent receive the first call in
the queue.
It happends some times.
This behavior
Theo Band wrote:
Some points to consider for the SPA-921:
Very complex web interface (yes you have the freedom to tweak
everything, I prefer a simpler interface)
But the SPA-921 can also be remote provisioned/configured over TFTP,
which is just perfect. IMHO.
The display has no backlight
Hi,
On Wed, 2007-04-18 at 15:56 +0300, Maysara A. Abdulhaq wrote:
hello,
i tried to increase the number in /proc/sys/fs/file-max , which was:
203511
and file-nr was
21120 203511
so i did :
echo 400176 /proc/sys/fs/file-max
but it didn't help, what could possibly make this
Hi Maysara,
I have your same problem.
are you using mISDN? If yes update your driver.
Giorgio Incantalupo
Maysara A. Abdulhaq wrote:
hello,
im having trouble with asterisk with medium load, it seems im running
out of files, here is a chunk of the logs with grep \(file\|pipe\):
Apr 18
On 17 Apr 2007, at 22:32, Lenz wrote:
Hello list,
we are developing a new application that uses the Manager API in
order to find a set of channels where variables are set in a
predefined way. To do this, we currently send a Status command to
obtain all available channels and then query
Kenneth Padgett wrote:
I have learned the hard way that using old configs with new firmware is
asking for trouble. It is much better to keep your custom configurations
in a MAC specific overrides file and replace the sip.cfg and phone1.cfg
files completely.
This doesn't guarantee that you
Try adding userscontext = numberplan-custom-1 to the [general]
section of extensions.conf
Done that. No change happened. Extesions are still executed in default
context. One strange thing I've noticed is that in lines like
SIP/80.1.61.21-092c23b0 before I used to see a number of extension
that
Try ringall or roundrobbin. You only have two agents.
Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com
KB3OPB
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of equis
software
Sent: Wednesday, April 18, 2007 9:21 AM
To: asterisk-users@lists.digium.com
Subject:
the cdr analyzer should work for most of what you need. The call costs
will be the hard part. If you know how much each type of call should
cost (based on destination number, location, etc), then you could do the
math on your own. But if you don't, then you'll have to wait for your
provider to
Hi all!!
I have downloaded the asterisk from svn checkout
http://svn.digium.com/svn/asterisk/trunk asterisk-trunk (is the asterisk 1.4
subversion). I also downloaded the patch for cellphone and make it work
fine. Then I bought the tdm11b board to have phone connection in my
computer.
I
Anselm Martin Hoffmeister wrote:
Am Mittwoch, den 18.04.2007, 13:18 +0200 schrieb Knud Müller:
Dinesh Nair wrote:
On Wed, 18 Apr 2007 09:04:22 +0200, Knud Müller wrote:
I
think it can be done by using the dialplan and the database to store the
statistical information but
Feedback on the GXP2000 - we have around 10 of them:
1) Great if the firmware's recent (but not too recent - see GS info over at
http://www.voip-info.org/wiki/view/GXP-2000)
2) Good caller ID
3) Speakerphone OK
4) Good features - Asterisk friendly and they support paging/announcements
5) BLF
On Wed, 2007-04-18 at 14:58 +, Iban Lopetegi Zinkunegi wrote:
I go to my asterisk recompile it but I realize there is no
chan_zap.so! When I recompile it, i check the make menuselect and the
channel zapata is not appearing there. Does any body know any patch for
that? Or how to
do you have also compiled latest svn-trunk zaptel?
Iban Lopetegi Zinkunegi wrote:
Hi all!!
I have downloaded the asterisk from svn checkout
http://svn.digium.com/svn/asterisk/trunk asterisk-trunk (is the
asterisk 1.4 subversion). I also downloaded the patch for cellphone
and make it work
Hi, sometimes I have only two agents, but most of time I have four or five.
On 4/18/07, Steve Totaro [EMAIL PROTECTED] wrote:
Try ringall or roundrobbin. You only have two agents.
Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com
KB3OPB
--
*From:* [EMAIL
I preffer not dialing 9 and have set up my server like this. One thing that
does puzzle me is would it be possible to dial +441232345634 I come accross
this problem as all my cell phone contacts are preffixed + I then sync these
contacts with my laptop and sometimes cut / past the number into a
On Wed, 18 Apr 2007, Knud Müller said something to this effect:
Hi all,
lets say I've registered at several Sip-Providers. Provider A offers best
rates but is often too busy to get a line. Sip Provider B is stable (but
more expensive). The asterisk box has a high call volume therefore
1)i downloaded the zaptel drivers from svn checkout
http://svn.digium.com/svn/zaptel/trunk.
2) I did make distclean, ./configure while my zaptel is already running.
However now i check in make menuselect and still can not see the zaptel
module.
Any other idea?
Thanks
iban
From: Pavel
Wireless wrote:
I preffer not dialing 9 and have set up my server like this. One thing that
does puzzle me is would it be possible to dial +441232345634 I come accross
this problem as all my cell phone contacts are preffixed + I then sync these
contacts with my laptop and sometimes cut / past
Ondrej wrote:
What version of Asterisk are you using? I've had recording
working with SVN before 1.4, the 1.4 betas and currently 1.4.1.
*** Update ***
Recordings are tied to a moderator joining the conference at this
time. I may need to change that based on feedback/requests to
do so.
Hello all,
I'm having a quite simple configuration like:
SIP provider = asterisk SIP = lan
Everythings works fine but sometime I can't get incoming call.
here are some of the logs from set debug 25 set verbosity 25 sip show
debug and sip.conf and a part of extension.conf
thanks in advance
Kevin P. Fleming wrote:
Eric ManxPower Wieling wrote:
I'll be sending Digium support the info they requested later today. I
hope it helps.
We have a developer working on extending Zaptel to support pre-echo
audio capture right now, so that we can work on debugging these issues
with real data
Also - this is probably again a problem of the missing documentation,
but let me clarify my problem in detail:
If I create a conference, there is a button email participants. If I
click that button, nothing happens (). How does the whole email
procedure works? How does the web-meetme gather
Also - this is probably again a problem of the missing documentation,
but let me clarify my problem in detail:
If I create a conference, there is a button email participants. If I
click that button, nothing happens (). How does the whole email
procedure works? How does the web-meetme gather
Sorry about that!!!
IS WORKING!! you were right, i had to make distclean!! I was confused
because i could not see zaptel channel in make menuselect, but i can not
even see sip channel. I just followed normally with make and make install
and is working fine for me!!
Thank you
Iban
From:
I didn't do anything special, I just used the command to split the
resources into four equal nodes, I think its called vzsplit.
The only possible extra step I remember was I had to play around with
the tty variable and how its used in safe_asterisk but I don't
remember what I actually did or
Giorgio,
That does not work it just shows up as useincomingcalleridonzaptransfer
I set the following: callerid=useincomingcalleridonzaptransfer. Are you
referring to something else like useincomingcalleridonzaptransfer=yes
Otis Surratt Jr. /
CallWeaver is the new name for OpenPBX
-Original Message-
From: Carlos Jerónimo [mailto:[EMAIL PROTECTED]
Sent: Tuesday, April 17, 2007 3:45 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] internal sounds of asterisk / freePBX
i use xlite
Hi,
I would just like to know if any work was ever done on COLP or its
related cousins? The last evidence of it seems to be about 2 years old
when K.Flemming and Olle both showed some mild interest. I am not sure
how well that code would apply to today's Asterisk.
(I realise that this is sort
Eric ManxPower Wieling wrote:
Any updates on this?
The code is done and initially tested; it is being reviewed internally
and should be available on Friday or Monday.
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing
Hello Dan,
The links to download a recording are already on the past
conference page IF the conference was recorded.
Aha, I see, intelligent. I will give it a try.
OK, I get it now. This is a side effect of offering too much
flexibility. I use and prefer the client-side mailer, and my
Hi there,
I'm converting a dialplan callback type application to fastagi as I'm
hitting the buffers with respects to getting useful results from CDRs.
It works by a spool call file triggering a Local extension, that extension
then does the first dial to a client. I dial to a local context
Did you have any E1/T1 cards in your server?
On 4/18/07, shadowym [EMAIL PROTECTED] wrote:
CallWeaver is the new name for OpenPBX
-Original Message-
From: Carlos Jerónimo [mailto:[EMAIL PROTECTED]
Sent: Tuesday, April 17, 2007 3:45 PM
To: Asterisk Users Mailing List - Non-Commercial
I don't know where he got the bizarre useincomingcalleridonzaptransfer
option, but it does not exist as you can see below:
[EMAIL PROTECTED] ~]# grep useincomingcalleridonzaptransfer
/home/software/asterisk/asterisk-1.2.17/configs/zapata.conf.sample
[EMAIL PROTECTED] ~]#
Maybe the option is
Eric ManxPower Wieling wrote:
I don't know where he got the bizarre
useincomingcalleridonzaptransfer option, but it does not exist as
you can see below:
*snipped
just a note, not sure if it is still in 1.4 tree, but it used to be in
CVS-TRUNK as an option for chan_zap
Gilles Ganault wrote:
Hello
I'm about to order a GrandStream GXP-2000 and a Linksys SPA-921
I don't know where you live, but I've seen significant price-differences
on the SPA-921 across Europe. Very pricey in the UK, less so in
Germany, but absolutely rock-bottom in Switzerland at SFr124.
no i don't have any card.
2007/4/18, Leonardo Kamache (Gmail) [EMAIL PROTECTED]:
Did you have any E1/T1 cards in your server?
On 4/18/07, shadowym [EMAIL PROTECTED] wrote:
CallWeaver is the new name for OpenPBX
-Original Message-
From: Carlos Jerónimo [mailto:[EMAIL PROTECTED]
Richard Lyman wrote:
Eric ManxPower Wieling wrote:
I don't know where he got the bizarre
useincomingcalleridonzaptransfer option, but it does not exist as
you can see below:
*snipped
just a note, not sure if it is still in 1.4 tree, but it used to be in
CVS-TRUNK as an option for chan_zap
Eric ManxPower Wieling wrote:
Richard Lyman wrote:
Eric ManxPower Wieling wrote:
I don't know where he got the bizarre
useincomingcalleridonzaptransfer option, but it does not exist as
you can see below:
*snipped
just a note, not sure if it is still in 1.4 tree, but it used to be
in
Ondrej wrote:
Ok, I understand that now as well - you click that button
and thunderbird should popup with the mail composer open,
right?
Yes.
Does not happen to me - most likely problem w/ my firefox
settings.
Browser security settings most likely
Now it all make a sense, sorry for
You could buy one of those X100P clones for ~$20 shipped and use that
for timing (and also an added FXO port), or a bare TDM400P with no
modules for ~$100 and have the option of adding modules for future upgrades.
Thanks,
Steve
Bryan M. Johns wrote:
Install zaptel and only enable the ztdummy
I've installed zaptel on FreeBSD and when I try to load ztdummy module I get
this error kldload: can't load ztdummy.ko No such file or directory. and
when I do
ztcfg:-
Notice: Configuration file is /usr/local/etc/zaptel.conf
line 0: Unable to open master device '/dev/zap/ctl'
Keyword:
Eric,
I have watched the CLI before and it said nothing although I did change
the position of the callerid = asreceived to right below and nothing it
still shows up on the phones asterisk and in voice mail sent via
e-mail unknown caller:
Here is an output from a while back but it stopped so
Today a 56-button expansion module for the GXP2000 came in.
When I program the buttons+leds on the expansion module for BLF, then
speed-dial works fine: when I press the button the programmed ext number
is called properly.
However the LEDs are always off: neither green nor red They are
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of dima
Sent: Tuesday, April 17, 2007 10:39 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] peers are using wrong contexts
Hello, everyone.
Today I've installed an
OCOSA ListAcc wrote:
Eric,
I have watched the CLI before and it said nothing although I did change
the position of the callerid = asreceived to right below and nothing it
still shows up on the phones asterisk and in voice mail sent via
e-mail unknown caller:
Here is an output from a while
On 4/18/07, Gilles Ganault [EMAIL PROTECTED] wrote:
Hello
I'm about to order a GrandStream GXP-2000 and a Linksys SPA-921
I'd like to have some user feedback about how those phones perform, and
whether their LCD screen displays both the caller ID name and number (The
GrandStream BT-100 only
On the GXP-2000 press the Mute/DEL button while the phone is ringing,
and it will return 486 (Busy).
This works to bounce new incoming calls while already in a call as well
(call waiting).
- Anthony Kepler
Andrew Joakimsen wrote:
My main complaint about both phones is there is no way to
Hello..I own a server running Slackware 10.2 with kernel 6.1.13 and I tried
unsuccessfully to install recently Asterisk 1.4.0. I install all packages but
when I execute the command asterisk -vc in order to start asterisk, I get a
message Segmentation Fault and the debugging stops suddenly.
Hi guys,
I know it's a little off topic but..Wondering if you can help.
My wife has been asked to find a writer to produce a story on The
dramatic ramifications of IPV6 on commercial businesses and how it will
change the product designs for ordinary household/commercial use in a
5-10 year
so to fix the no caller id thing will need to adjust the rx gain and tx
gain?
Otis Surratt Jr. / [EMAIL PROTECTED]
OCOSA Communications, LLC
PH: 918.585.9882 x 205 Fax: 918.585.5857
Visit Tulsa's Best Internet Data Center:
I'm chasing down some issues at a call center. Today I received a complaint
that audio file playback
ceased after they upgraded the system from FC4 to FC6, Asterisk 1.2.14 to
1.2.17. Zaptel is at
1.2.16. The system in question takes inbound calls via IAX2 and has a TE410P
with a couple of
On Wed, 18 Apr 2007, Anthony Kepler wrote:
On the GXP-2000 press the Mute/DEL button while the phone is ringing, and
it will return 486 (Busy).
This works to bounce new incoming calls while already in a call as well (call
waiting).
And push it when the phone isn't ringing and it set Do Not
Eric,
Thanks when I took the rx and tx to 0.0 on both the caller id showed up
I guess I will play with. My main reasoning for adjusting the rx and tx
was to get rid of the echo...What other tips do you suggest or anyone
out there? Thank you!
Otis
Hi all,
I've recently upgraded to Asterisk 1.4.2, coming from 1.2.14. Using my
existing Cisco 7960G handset(s). I've tried multiple installs of
asterisk 1.4.2 with multiple handsets and SIP will not authorize my
phone. I'll include some verbose log messages below to show a VALID
registration and
Hi! i have an error using the meetme aplication, and just dont work..
my meetme.conf is:
[rooms]
conf = 700
i calling from a sip phone, the extension number is 600. there is the error:
Executing [EMAIL PROTECTED]:1] MeetMe(SIP/600-09111e58,
700|MI) in new stack
WARNING[20055]: channel.c:3024
Manolet Gmail wrote:
Hi! i have an error using the meetme aplication, and just dont work..
my meetme.conf is:
[rooms]
conf = 700
i calling from a sip phone, the extension number is 600. there is the
error:
Executing [EMAIL PROTECTED]:1] MeetMe(SIP/600-09111e58,
700|MI) in new stack
hi, i donwload XLITE and see there is a fuction to send Instant Messages.
when i try to use it i get this error:
Error: Method Not Allowed.
there is anyway to enable IM on asterisk 1.4.2?
___
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2007/4/18, Rodrigo Gonzalez [EMAIL PROTECTED]:
Manolet Gmail wrote:
Hi! i have an error using the meetme aplication, and just dont work..
my meetme.conf is:
[rooms]
conf = 700
i calling from a sip phone, the extension number is 600. there is the
error:
Executing [EMAIL PROTECTED]:1]
Hi,
I'm having high load, choppy sound and slow responsives with an asterisk server
(version 1.2.12.1) that make a peak of 90 channels (around 60 phones calling at
max, isn't necessary to reach this peak to get the problem). All the traffic is
SIP, with recording for every call. The server
Had an appointment for these schmoes to come out and install another
line. Was supposed to be 8-12. Its now 6PM and not even call. Missed
3 sales calls waiting on these jerks.
No wonder customers were jumping ship to Vonage.
--
Warm Regards,
Lee
Hi Manolet,
You have to install zaptel in order to make MeetMe application to work.
MeetMe needs a kind of timer device that is provided by zaptel package.
Eventhough you don't have a zaptel card you need to install its package.
Search for MeetMe application in http://www.voip-info.org/ and
From: Per Jessen [EMAIL PROTECTED]
Date: Wed, 18 Apr 2007 14:48:45 +0200
Per Jessen wrote:
Per Jessen wrote:
OK, part of the confusion is now clearing up. But I'm not getting
much further. When I try to send an SMS, I see the call going
through, but no SMS is ever sent.
This is a bit of
Hi, I need to add the timestamp to the recorded call filename, I use this
variable ${TIMESTAMP} in the Monitor() function, but when I look for this
call, the TIMESTAMP is missing in the filename.
I try to export this as a environment variable but nothing changes.
Any help is welcome, thanks.
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