kevin,
make menuselect - creates an xml file... let me look to see where it is
[EMAIL PROTECTED] asterisk-1.4.5]# ls -l menu*
Current Directory is /usr/local/src/asterisk-1.4.5
-rw-r--r-- 1 root 2065 Jun 25 18:36 menuselect.makedeps
-rw-r--r-- 1 root 1654 Jun 25 18:36
jim,
the asterisk gui doesn't interact with apache or apache2... it has it's
own httpd... perhaps you can move the vmail.cgi script to the apache2
directory structure cgi-bin. I haven't tried that as of yet so I don't
know how that would work.
daveC
Jim Archer wrote:
Hi Everyone...
I am
-0400 dave cantera
[EMAIL PROTECTED] wrote:
the asterisk gui doesn't interact with apache or apache2... it has it's
own httpd... perhaps you can move the vmail.cgi script to the apache2
directory structure cgi-bin. I haven't tried that as of yet so I don't
know how that would work
on the CLI type this command:
dialplan show [EMAIL PROTECTED]
-and-
dialplan show [EMAIL PROTECTED]
you should see a dialplan returned to you. if not, which is what I
expect, you have to include the section [where6009is] in [local] or
[default]... i.e.
[local]
include = where6009is
hi,
can anyone point me to answering machine beep detection methods or writeups for
*?
thanks,
daveC
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satish,
please clarify...
do you want people to dial 1171 on the avaya system to get to you?
do you want people to dial 1171 on the * box to get to you?
do you want people to dial 71 on either box to get to you?
daveC
satish patel wrote:
Dear all
I have asterisk 1.2
ed,
do you positively have to have 1.4.0?
just download 1.4.9 or 1.4.8... 1.4.0 is too old...
I can email you 1.4.8, 1.4.5, 1.4.9...
I just downloaded 1.4.9 from:
http://www.digium.com/elqNow/elqRedir.htm?ref=http://ftp.digium.com/pub/asterisk/releases/asterisk-1.4.9.tar.gz
daveC
EdPimentl
matt,
I just had the same problem... does your CLI report 'unable to
create channel Zap/#'
post the CLI output to help us determine the problem.
daveC
Matt Scott wrote:
Dear All
The
setup is te110p with an 8 channels PRI to make and receive all calls.
SIP phones throughout
eric
try this...
sox foo.wav -r 8000 foo.gsm resample -ql
# add -c1 to write the file in mono
I can't remember if you have to do something special in the recording
too depends on your recorder.. oh, now I remember. you have set
the recording to 16bit 14400 hz or something like that...
baji, mhoppes,
remember, if you have Only the g729 codec allowed or if this is the
only allow= entry in the sip.conf file, callers requesting any other
codec will be rejected
daveC
Baji Panchumarti wrote:
On 7/27/07, Matt [EMAIL PROTECTED] wrote:
Can someone comfirm my logic
randulo,
I could not get into the conference today... the SIP line was busy, no
matter what I do, the website thinks I'm not logged in and gives me the
login page. after I login, anything I want to do brings me back to the
login page... so I tried to re-setup the account thinking I wasn't
aryjunior,
is your dialplan and registration configured to connect to another *
server?...include your config so we can analyze it...
daveC
Carlos Rojas wrote:
Hello,
Do you have porf forwardin for SIP protocol in your firewall?
SIP: 5060 udp
rtp 1 - 2 udp (default)
and
michael,
this is what I use for centOS 4, but I think its too loose... let me
know if you don't know where to put it...
daveC
# for asterisk
-A RH-Firewall-1-INPUT -p udp -m udp --dport 5060 -j ACCEPT
-A RH-Firewall-1-INPUT -p udp -m udp --dport 4569 -j ACCEPT IAX
-A RH-Firewall-1-INPUT -p
my shared webhosting is going strong...
daveC
Asterisk guy wrote:
1and1 dedicated server's service has been down for a few hours ,
unable to reach them by phone or email. do anyone know what is going
on there ?
Mario
picturephone -dot- com/
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matt,
are you looking for unit testing of the * components or systems testing,
testing the finished product? or both?
I think you are onto something here... I hope it takes root. I would
say put it in the addons. it would be Great if digium takes it up. it
is a smart move for them to
nick,
I am actually playing with skills based routing right now...
how would you propose to send multiple calls requiring different skills
into a single queue and have agents w/o that particular skill in the
same queue?
daveC
Nick Brown wrote:
Morning All,
Has anyone here successfully
for those of you who have not joined the conference call yet, I highly
recommend it. there is always several interesting tidbits that will
help you in your * implementations...
see you at 12:30p today!
daveC
randulo wrote:
As usual, we'll be jawing about any and all asterisk-related
for those of you who have not joined the conference call yet, I highly
recommend it. there is always several interesting tidbits that will
help you in your * implementations...
see you at 12:30p today!
daveC
randulo wrote:
As usual, we'll be jawing about any and all asterisk-related
dean,
I am an active member of AUG NYC... you can email me off list for any
info you need.
also, I am preparing to start a south jersey * UG. the phila group is
waning...
thanks,
daveC
Dean Collins wrote:
This is an email to all New
York
based Asterisk users.
hello,
I use both * 1.4 and *NOW... because the *gui is incomplete in *NOW, I
loaded 1.4 over *NOW because the gui regenerates files that, well, don't
seem to work very well. it seems to me the gui creates the users.conf
file, and then a script creates or uses the users.conf to create the
donovan,
by wake up call, I am assuming you have some condition that will trigger
a call not an actual 'wake me (a human) up call'...
here is what I set up to remind me to remind my son to take his
singulair pill. at 5;30pm.. I created a cron job to kick this shell
script off... thankfully, he
cf,
I haven' t used the * manager... but from research, that is how I would
expect to do it... I would have a cron job fire off every 5 minutes
(or so, probably configurable) and connect to * via the manager, request
the status, then send an email based on the result... would be pretty
ango,
can you provide some sip.conf and extens.conf info?
daveC
Rilawich Ango wrote:
hi,
I have 2 asterisks with the following configuration.
asterisk server 1 (S1) has an user 9002
asterisk server 2 (S2) has an user 9003
Both users can make call to each other without problem.
Now I add both
0xception,
yes, I suspect this is the reason. the users.conf file may not used as
you expect. the users.conf file in 1.4 is a source file for generating
the dialplan on-the-fly by the gui... if you hand edit it, the gui
doesn't regenerate it... one way to verify this is to, at the CLI,
robert,
I might be interested depending on cost, message, and quality...
keep me in the loop.
daveC
Robert Augustyn wrote:
Hi,
I am working on developing a professional Marketing Materials for my
systems.
I plan on using a very good(expensive) company to do that so splitting
the costs with
I got from the above case. Do you have such configuration?
I have no idea to solve the problem
On 4/20/07, dave cantera [EMAIL PROTECTED] wrote:
ango,
can you provide some sip.conf and extens.conf info?
daveC
Rilawich Ango wrote:
hi,
I have 2
]; Asterisk Users Mailing List
- Non-Commercial Discussion
Subject: Re: [asterisk-users] Developing Marketing materials ...
I am also working on this, and have a
marketing/communications background. I may be able to help
cheaper than the "big agency" :)
thanks,
matt
On 20/04/07, da
...
daveC
Robert Augustyn wrote:
Dave,
Agreed, is there anything you
have in mind?
robert
From: dave
cantera [mailto:[EMAIL PROTECTED]]
Sent: Saturday, April 21, 2007 5:39 PM
To: Robert Augustyn
Cc: 'Asterisk Users Mailing List - Non-Commercial
Discussion'; [EMAIL
khaled,
you might check lamppix or knoppix... they have a remastering scheme
http://lamppix.tinowagner.com/
http://www.knoppix.org/
http://www.wifi.com.ar/english/cdrouter.html
haven't played with it in a while but I did create an iso... that
worked! :)
daveC
Khaled Chehab wrote:
shadowym,
best thing to do is talk to a lot of consultants, coaches, and marketing
people... take the approach you do with learning open source only
reverse it... instead of reading source (internal) ask people
(external)... it is a big undertaking and the most important task you
have...
. With IT it's usually a combination of
cold call/networking/word of mouth. I'm hoping that Telco is the same but I
never see any telco guys at networking events so I am thinking they cold
call and advertise targeted at business owners. I'm not sure though.
-Original Message-
From: dave
oliver,
what gateway provider are you referring to?doesn't your sip phone
connect directly to * as your diagram indicated?
DSL providers should not be doing any codec anything!
daveC
Oliver Brandt wrote:
Hi!
As the upstream of my DSL-connection is very slow, I'd like my
sip-phones to use
oliver,
ugh, it is too obvious... why did it take me so long to figure it
out...
both phones have to have to negotiate the same codec for audio... as
far as I know, * is supposed to do automatic translation and your
gateway should be doing translations only on the below codecs. I
haven't had
mitch,
not that I can answer your problem but is this ver 1.4.1? I had a
similiar problem in that zapscan was updating the zaptel.conf and
nothing would work until I mucked with zaptel.conf.zapscan... I might
have the filename wrong as I have multiple files now :(... it has
zapscan in the
here is a way that I solved a similar problem... have a shell script
that runs and indexes all the files in the directory into an ascii flat
file with a format of
filename
0001 directory/tt-weasels
0002 directory/tt-monkeys
in your dialplan use the rand() to pick a number, pass it to the
. Is there
any way
to do this?
I do this with an AGI.
On Wed, 2 May 2007, dave cantera wrote:
here is a way that I solved a similar problem... have a shell script
that
runs and indexes all the files in the directory into an ascii flat
file with
a format of
filename
0001 directory/tt-weasels
has anyone run into this message? for some reason, which I can not
determine, this script stop working and now gives this error. I googled
'outgoingspoolfailed' but not too much turned up... only questions, no
answers... :(
I am mv'ng a .call file to the ./outgoing directory. the call
nitesh,
you are correct. you need 1.4.x...
daveC
Nitesh Divecha wrote:
Hello All,
I just received some test units of Grandstream GXV-3000 IP Video Phone.
I did some research and looks like Asterisk 1.2 does not support video
H.264 but Asterisk 1.4 does. Is it correct?
Actually I did try
shawn,
you can set an archive variable in the .call file to 'yes' and it will
save it in ./outgoing_done... if there is now outbound line availible,
the .call file is updated (appended to) as per the status... * will keep
trying till it completes the calls or the number of retries is reached.
morgan,
I've seen some info on additional variables in the CDR... but haven't
tried it... look to these pages:
daveC
http://www.asterisk.org/doxygen/1.2/AstCDR.html
In addition, you can set your own extra variables by using Set(CDR(name)=value).
These variables can be output into a
remco, et al,
could I use dundi where I could use an area code to determine the
connecting server or dial string? just like we would use 88XXX to dial
a 3 digit extension on another server at location 88? or dial 84XXX for
a 3 digit extension on a server located at 84?...
thanks,
daveC
benedikt,
* 1.4 does no video codec translation... it is just a pass through so
using the same unit on both ends is a plus. you might try adding this
codec too.
allow=h264
I assume that audio is ok, just no video, right!?
there may be a nat problem, try
nat=no
I have some video experience
benedikt,
try putting these (or your version of these) in the sip.conf [general]
heading. it was suggested to me before, that what is general should
go in the general section, what is specific to a particular extension
should go in the specific extension section. also, I put a ton of
options
jason,
shouldn't there be an answer in there somewhere?... like...
[inbound-sip]
exten = 300,1,Wait(1)
exten = 300,n,Answer()
exten = 300,n,NoOp(${EXTEN})
exten = 300,n,NoOp(${CALLERID})
exten = 300,n,Dial(SIP/300,15)
exten = 300,n,VoiceMailMain
exten = 300,n,Hangup()
daveC
tim, patrick,
SSO is a hot button for large orgs/corps... have heard it bantered for
years but no solution. I have seen a product that had a small utility
on windows that transmitted login info to a linux box. it was in every
users profile so upon login, the transaction was completed. I have
here! here! they are different beasts...
Giorgio Incantalupo wrote:
Hi
all,
since Asterisk 1.4 seems to have too many differences from previous
versions, wouldn't be nice to have a new mailing list?
Giorgio Incantalupo
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jeronimo
there is no difference...
Jeronimo Romero wrote:
Is there any technical difference between a T1 cable and a cat5e patch
cable as far as using them with Digium T1/E1 cards? Can PRI circuits
terminating at a smart jack connect successfully to Digium cards using
straight through
george,
is this a production system you are upgrading to 1.4?
daveC
George C. Attopany wrote:
Hi,
I run ASTERISK 1.2 with a Wildcard TE410P-Xilinx on Redhat Linux
8.0(Psyche) with Kernel 2.4.18-14 on an i686 with a Channel bank to
enable me use anlog handsets.
I would like to
carlos,
this is coming from a linux admin perspective but here is something to
get started...
active directory transfers info to-from windows domain controllers via
the network. there are probably api frameworks available as open
source, although they may be incomplete. I have seen another
thomas,
the dialplan is quite different in 1.4.x... they use a users.conf file
for, I think, all endpoints (phones not providers)... there is no
documentation, that I have seen yet, on this... if someone has seen it
please post on the list so we can get more info on the configuration...
I just
bryan,
can it be that polycom is the best?
my headset doesn't work with the 600 volume resets to default on
every hangup,
speakerphone resets intermittenly (haven't figured out why),
my 301 has a speaker phone but no mic (very useful!), every
config changes reboots the phone taking
:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]] On Behalf Of dave cantera
Sent: Thursday, 22
March 2007 6:51
PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
[asterisk-users]
Linksys/Sipura SPA-942 phones in largerdeployments
bryan,
can it be that polycom
office:
678:248:2637 x:1500
direct:
678:229:1809
iaxtel:
700:248:2637 x:1500
http://www.sheltonjohns.com
On Mar 22, 2007, at 6:51 PM, dave cantera wrote:
bryan,
can it be that polycom is the best?
my headset doesn't work with the 600 volume resets to default
2007
now I am wondering... since I started with asteriskNow, I wonder if
1.4.x has a users.conf...
let me check... no, there is no users.conf in a 1.4.1 release...
must be unique to asteriskNow!... hmmm... the mystery ensues...
daveC
Lacy Moore - Aspendora wrote:
On 3/22/07, dave cantera [EMAIL
I have an interesting task for my son's lacrosse team... it is the
time-old telephone tree...
I am pretty sure someone has already done this w/*, why re-invent the
wheel?...
a) coach calls in leaves a msg, others call in retrieve the msg
b) coach calls in leaves a msg, kicks of a call to
hi,
I'm getting registration errors I can't debug...
[Mar 23 11:07:20] NOTICE[2952]: chan_iax2.c:7344 socket_process:
Registration of 'host2' rejected: 'Registration Refused' from:
'10.10.10.82'
I was getting a 'Cause Code: 29' INV,POKE,...,REJ but I can't
duplicate that level of debugging
olivier,
soft phones on a PC require a port to connect to the server... haven't
tried multiple soft phones, simultaneously, connecting to one server or
multiple servers but if you can configure the outgoing port, it should
be possible... NAT might get quite confusing so I would try it before
nathan,
can you post your extensions.conf file [to-sip], and your sip.conf
section for extension 201... ie [201]?
it looks like, perhaps, it is a dialplan problem...
daveC
Nathan Bell wrote:
I tried to add a couple of SIP phones (polycom 601s) to my existing
asterisk installation. I can
nathan,
try dial() directly to the extension
[to-sip]
exten = _X.,1,Dial(SIP/[EMAIL PROTECTED],120)
try
exten = _X.,1,Dial(SIP/${EXTEN},20)
where ${EXTEN} = 201
and
[201] in /etc/sip.conf is
[201]
type=friend; Friends place calls and receive calls
context=from-sip
and
sip show users
Noah Miller wrote:
Hi Nathan -
No loop now, but instead I get this:
Mar 26 15:42:18 NOTICE[1854] app_dial.c: Unable to create channel of
type 'SIP' (cause 3 - No route to destination)
Mar 26 15:42:18 VERBOSE[1854] logger.c: == Everyone is
khaled,
I successfull remaster a router CD, and lamppix CD both using knoppix
or debian as the base, I am pretty sure...
try
http://lamppix.tinowagner.com/
http://www.wifi.com.ar/english/cdrouter/
daveC
Khaled Chehab wrote:
Anyone have an idea to re master centos,in other
joe,
when I have problems with audio and other connections seem to work, I
always look for a codec incompatibility... use 'sip set debug peer
extension' and look for the codec handshaking... make sure both
extensions have a compatible codec choice...
daveC
Using INVITE request as basis
to all,
I have a cell interface that hooks up to a standard pots handset... can
use a cingular, tmobile, or SIM card provider.. hookup is about 30
seconds has a remote antenna so you can locate the unit about 10 ft from
the antenna... quality is good... well, as good as cingular anyway... :)
vijay,
I had a similar problem with a pots line and 1.4.1... zap wasn't
loading. from the CLI check that zap is loaded with 'zap show channels'
pbx15*CLI zap show channels
Chan Extension Context Language MOH Interpret
pseudodefaultdefault
25
Subscription-State: terminated;reason=timeout
Content-Length: 0
-
Does this imply anyting to anyone?
Call can be made, after this.
joe a.
**
dave cantera [EMAIL PROTECTED] Wrote: 4/7/2007 3:53 PM:
joe,
when I have problems with audio and other connections seem to work, I
always
brandon,
engineers are BIB on monitoring... count me as one who wants to monitor!!!
daveC
Brandon Kruse wrote:
Yes,
I have actually written a resource module for asterisk and the gui to
use rrdtool to make REAL pretty gradient shaded graphs based on asterisk
data.
So, if you want the cacti
I had a similar problem... I forget exactly how I resolved it because it
happened twice... here is the solution from memory.
the sequence of the zaptel, libpri, and asterisk is important. if you
compile zap before libpri, zap doesn't know how to make it and then
asterisk doesn't include it
alex,
I have been considering linux clustering for *... am not ready today and
expect it in about 6-9 months... was wondering if anyone had put * on a
cluster and what experiences they gained? have you done this?
daveC
Alex Balashov wrote:
On Wed, 11 Apr 2007, Andrew Joakimsen said
alex,
thanks for the note... oh well, fun times ahead! :)
daveC
Alex Balashov wrote:
Dave,
On Fri, 13 Apr 2007, dave cantera said something to this effect:
alex,
I have been considering linux clustering for *... am not ready today
and expect it in about 6-9 months... was wondering
at each stage
any help would be appreciated...
thanks,
dave cantera
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II Chronicles 16:9 NAB
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hi,
is there anything going with VoiceXML in asterisk??? is this the list
to query regarding this or should I put this on the dev list?
thanks,
dave cantera
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does anyone know how to stop the clicking sound that happens at the end
of a playback() command?
is it something I can do in the recording?
I looked in the 'book' but there was only a 'j' option...
thanks,
daveC
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rafael,
it should work. both systems are auto configurable...
daveC
Rafael Canchola wrote:
Hi all
I sold new TDM800P card with 8 FXO ports, someone know if can be use
this card on AsteriskNOW or trixbox?
What can i do for use this card?
Thanks.
to all,
I am available for work either US or Non-US for * consulting,
configuring, integration with other business applications. have been
working with * for about three years on and off and would like to do
this full time. am available for on-site or remote project work.
have 20+ years
steve,
oops, you are right... sorry.. wrong list...
daveC
Steve Edwards wrote:
On Sat, 1 Dec 2007, dave cantera wrote:
[snip]
You forgot "i don't know what the shift key is" and "i don't understand
what Non-Commercial Discussion means."
vieri,
you can get sip status with the following shell script... I named it
'sipshowpeer'... to execute, chmod 755 sipshowpeers
daveC
-- cut here -
#!/bin/sh
# sipshowpeers
#
# show current asterisk SIP peers
asterisk -r -x 'sip show
carlos,
you got further than I did... AMD didn't work at all on my release.. I
think I was using 1.4.11 at the time...
I ended up using the below
daveC
;--- amdtest (ext 13) starts here
;
; restructure this for the following conditions:
; 13 using
artifex,
if you want call recording transparently, check out orecX.com they
have a commercial and an open source SIP call recording package... no
zap recording but if you are forwarding to sip exensions, you should be
golden! saw them at VON 2007 boston... they have a recorded calls
rilawich,
in the CLI type the following:
CLI dialplan show [EMAIL PROTECTED]
then
CLI dialplan show [EMAIL PROTECTED]
-or-
CLI dialplan show [EMAIL PROTECTED]
and see if * recognizes the x100 in either of those...
daveC
Rilawich Ango wrote:
HI,
I have tried to add the context but it
bart,
one way is to write the recorded files to a known directory, then
launch an AGI script to use sox to combine/concatenate the two...
if you cat them into a known filename, just use the playback() cmd to
play it.
do you need specifics?
daveC
Bart Fisher wrote:
Does anyone know
speaking of multi-casting voice. since it isn't likely to get the ip
phones changed, could an app_multicast do the job?
has anyone thought of doing that?
daveC
Kristian Kielhofner wrote:
On Dec 10, 2007 1:17 PM, Jerry Geis [EMAIL PROTECTED] wrote:
Using asterisk 1.4 with 100M or 1000M
silvia,
I don't know how to pickup the message but if it is getting into the
dailplan as a variable, you can send it to an AGI() script as a
parameter...
AGI(my_script.php,${IM_TEXT})
if you give me an example of what you have already, perhaps I can think
on it more...
daveC
cimsi wrote:
tim,
sounds like a problem I had with bandwidth... too many devices
communicating on the same network connection to the internet...
have you tcpdump'd or used a bandwidth tool to see what the usage is?
nat=yes or nat=no? should be yes..
did you change the router between upgrades?
just some
google app_pickup2, i just found it myself...
oh, still have the URL up... here it is..
http://www.thorsten-knabe.de/linux/asterisk/pickup.jsp
Lukassky wrote:
Hi everybody again.
A week ago I started a new Term about Pickup group over IAX or mISDN. I've
set all the config up with callgroup
I did some research on spam filter about a year ago. there are image
analyzers that can detect human skin tones in images detecting porn. I
have seen some examples of how the porn guys speckle the images to
obscure, somewhat, the naked bodies.
the OCR idea would be useful but the OCR engine
oliver,
portsip.com has an sdk with a softphone applet... you might try googling
'softphone applet'
there was another java softphone promoted somewhere too, so try
'softphone sdk java'
could get you closer to a solution
daveC
Olivier wrote:
Hello,
From a previous thread, I learned Callto://
marco,
I use 1.4 exclusively but I would think a minor version would go pretty
easy if you are installing from sources for the current version as well
as the upgrade...
I would note (not a mental note, a written note) which source versions
you are using for libpri, zaptel, *, and addons. you
here are some snippets from previous posts... let us know what you like
the best...
CrossPlatform Linux, Windows, Mac OpenSource WebHuddle at
http://sourceforge.net/projects/webhuddle
I've tried dimdim and it was ok, but not as good as WiredRed.
take a look at
steve,
FYI: randy randulo already has a voip group at
http://food4wine.ning.com/
try that, it is already established...
daveC
BerkHolz, Steven wrote:
asterisk
linkedin group
I
have created an asterisk linkedin group for anyone interested.
bilal,
flash operator panel (fop) or any of the asterisk gui does this...
asteriskNow for example...
http://www.asternic.org/
daveC
bilal ghayyad wrote:
Hi All;
Is there an GUI for Asterisk that can help in showing
the call flow (who is in progress, who is connected,
called number, ...)?
rilawich,
can you post the CLI output so we can see what is going on?
from the exten, it is doing exactly what you tell it to do... dial then
hangup
daveC
Rilawich Ango wrote:
Hi all,
I will a TDM card with FXO modules on it. Below is the dial plan.
When someone can 9123456, CLI will
phil,
I think you are on to it... the best path is to load a new system up
with 1.4.x and port your existing dialplan over, test it out, lock it
down and then roll it out...
I've worked as a UNIX system integrator for 20+ years, worked with open
source and custom developed C/C++ code, Ada, and
lolu,
sounds more like a telco/itsp problem then *.
I would
tcpdump -i eth0 port 5060
to make sure it is actually going out... change 5060 if you have changed
your port to your itsp, of course.
to see what is going on as well as the other debugging notes mentioned
in this thread.
daveC
Lolu
, stopping a running asterisk, getting the
current release, untar'ng it and compiling it...
enjoy,
daveC
#!/bin/sh
#
#get_latest_rel.sh
#
# Dave Cantera: [EMAIL PROTECTED]
#
#get the current asterisk release components, put them in our REPOSITORY
#and unpack them in SRC_ROOT
tzafrir,
thanks for the note. btw, Great docs!
asciidocs looks cool too!
thanks!
daveC
Tzafrir Cohen wrote:
Hi
On Wed, Dec 19, 2007 at 12:19:08AM -0500, dave cantera wrote:
ok, here is my $0.02... I created a script since I had to
install/update so often and for various
my client purchased a couple of shoreline ip-100 phones... I managed
to get them to Not boot up... shows the polycom logo then goes
blank... looks like the want mcgp... oh, mgcp...
is there a solution for this? besides sending it back to polycom?
daveC
dovid...
while this seems like a good idea to have both sip show channels and
show channels sip having two, three or even four ways to do the same
thing would confuse/cripple the learning curve... * would turn into a
microsoft mentality where there are dozens of ways to
mojo,
nice suggestion.
daveC
Mojo with Horan Company, LLC wrote:
So I'm guessing this is what you're doing:
--
[ids]
exten = s,1,playback(enter your id number)
exten = s,2,WaitExten(10)
exten = s,3,Goto(1)
exten
0-09e062e8 is ringing
== Spawn extension (local-sip, 300, 8) exited non-zero on
'SIP/202-b753da18'
daveC
Lolu Gbenga wrote:
Thanks
Please am using putty to again access to my Linux asterisk box.
How can i use tcpdump to get your request on the exact Ethernet port
and port number.
I will apprec
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