this is an interesting project, SIP protocol is easy to find, writing a php
script, perl script, or python would probably work. it would probably work
better if it was a daemon. what would be connecting to it that you would need a
SIP connection for?... interesting...
Dave Cantera
(856)813
for thought,
Dave Cantera
(856)813-7098 mobile/txt
david.cant...@ibsonecall.com
Sent from my ASUS Pad
Steve Totaro stot...@totarotechnologies.com wrote:
On Wed, Sep 25, 2013 at 3:22 AM, Endri Stefani endri.stef...@plus.alwrote:
Hi
** **
Greeting to all you out there.
** **
I
dan, elder,
I have played with scripts to generate calls and track their
completion, email me off-list if you have questions.
daveC
Daniel - Asterisk wrote:
Hello Everyone,
I wonder if someone could share a manual about using SIPp for
Asterisk's testing.
I'll be gratefull
Regards,
danny,
not that it matters, but I agree. if the design is a good design, it
would not have to be redesigned on every release. in fact, the modules
template should also follow this philosophy that way you can concentrate
on adding functions and not the design...
sometimes, it is smarter to
doug,
why are you shaking!?!?... do you have a better recommendation?
daveC
Doug Lytle wrote:
C F wrote:
model name : AMD-K6(tm) 3D processor
*shudder*
Doug
--
SJREIA South Jersey Real Estate Investors Association
Want to invest in Real Estate?
come out and join over 450 real
paul, doug,
I had several AMD athlons 64bit... no problems running centos, suse.
they seem solid on 1.4.xx... had a few intel celerons and P4s. they
were good as well. guess I was Lucky back then!
thanks for supporting the list!
daveC
Paul Hayes wrote:
On 04/05/11 17:10, || dave cantera
I've been away from asterisk for a while since 1.4.16 and only installed
1.6 once to run a test... can someone recommend what the best version to
install is and the recommended CPU/motherboard for an * box these days?
I'm just running about 20 handsets and 4-8 lines with POTS SIP mix.
I
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.
jaap,
found this some time ago... might do the trick...
daveC
http://www.venturevoip.com/vm.pdf
Jaap Winius wrote:
Hi list,
After wrestling with the voicemail system for a while (Asterisk
1.4.14, Debian etch), I got it to work, but I still have lots of
questions, like:
* Why
steve,
thanks for posting this tidbit!
daveC
Steve Johnson wrote:
Sorry to answer my own post, but I have found a solution which perhaps
others can use too...
In the .call file, instead of specifying a channel line as:
chan: SIP/140 (for example)
use instead:
chan:
ed,
this may be somewhat liberal but should do the trick...
daveC
-A RH-Firewall-1-INPUT -p tcp -m tcp --dport 69 -j ACCEPT
-A RH-Firewall-1-INPUT -p udp -m udp --dport 69 -j ACCEPT
-A RH-Firewall-1-INPUT -p udp -m udp --dport 5060 -j ACCEPT
-A RH-Firewall-1-INPUT -p udp -m udp --dport 5061 -j
mayur,
did you try inband? with sip?
daveC
;dtmfmode=inband ; Choices are inband, rfc2833, or
info
;allow=ulaw ; dtmfmode=inband only works with ulaw
or alaw!
;dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info
;allow=ulaw ; dtmfmode=inband only works with ulaw
or alaw!
Mayur wrote:
nhadie,
meetme requires a zaptel timing device... ztdummy is unreliable when
using meetme conferencing... I suggest you spend time elsewhere in *
until you get a digium tdm400 w/ or w/o any daughter modules... you
just need the board for the timing device you don't actually need any
glenn,
what an interesting way to use GotoIf() and 9. didn't know you
could do that in GotoIf()!
you could have used (broken out) the individual services
[trunklocal]
[trunkld]
[trunktollfree]
and just included the above individual context in with the groups that
you allowed a particular
robert,
with limited info below, are you port forwarding on the router with the
public IP, ports 10,000-20,000, 5004, along with 5060? and the other
router (internal, I assume)???
how do you have two firewalls configured with one * box?
do you have captures on both sides of the internal (I
no experience with sangoma cards.
daveC
Steve Edwards wrote:
dave cantera wrote:
nhadie,
meetme requires a zaptel timing device... ztdummy is unreliable when
using meetme conferencing.
On Wed, 9 Jan 2008, Nhadie wrote:
hi dave thank you
rilawich,
do you have the pickup group defined?
http://www.voip-info.org/wiki/view/Asterisk+callgroups+and+pickupgroups
daveC
Rilawich Ango wrote:
Below is what I got from CLI
[Jan 7 23:02:46] NOTICE[3450]: app_directed_pickup.c:159 pickup_exec:
No target channel found for 111.
On Jan 7,
I brick'd two of my polycom phones trying to get the shoretel phones
working with *...
does anyone have the equipment to unbrick them?
there is a jtag serial cable that is needed along with the knowledge of
embedded systems..
that is all I currently know.
polycom wants $180 and 30 days to fix
to all,
I had a similar thought... what I came up with was, not my idea just saw
it done somewhere else, a small windows binary that was exectued on
login. registered your login name with a server (content filter in that
case)... any browser requests were logged for filtering and tracking
shane, et. al.
shouldn't the console/dsp work? I have a handset, let me get it no
markings on it, has jacks to plug into your sound card for audio in and
audio out.. it worked on my laptop with asterisk or maybe a softphone...
if your sound board has an audio in/out that might work
brian,
cool, I attended one of you tutorials in baltimore... it was Great!
would go again because I know I would learn even more this time after
being exposed to it in greater detail... I could absorb more this time...
daveC
Brian Capouch wrote:
Martin Smith wrote:
Hey folks,
Is
dean, fredrik,
when I installed skype, ugh, it asked me if I wanted to link phone
numbers on the web page to be click2dial... I did it and every phone
number on a web page was a link... I ended up turning it off... it was
too annoying... so there are some plug-ins out there that can do that
gilberto,
check your config files, extensions.conf, to see if autofallthrough is
set or not
you don't have any extensions in [ura] either... only choice is 1 or
2... anything else is invalid...
daveC
Gilberto Nunes Ferreira wrote:
Hi all
First I want to wish for everone a happy new
bilal,
you are right. you need to add port forwarding (UDP) to your router...
should work nicely then for iax. also, don't forget you iptables or
firewall port config to accept iax on your * box.
daveC
bilal ghayyad wrote:
Hi List;
I heared that IAX is good for NATing issues, but I do
not
erik,
you can start here:
http://www.voip-info.org/wiki-Asterisk+cmd+GotoIfTime
http://www.asteriskguru.com/tutorials/gotoiftime.html
daveC
Erik Wartusch wrote:
Hi all,
Im using Asterisk 1.4.11 and I want to proceed some time and date operations
in my dial plan. (for a time shifted
jared,
Happy New Year too!
excellent point! two brains are always better than one!
daveC
Jared Smith wrote:
On Wed, 2008-01-02 at 08:27 -0300, Gilberto Nunes Ferreira wrote:
First I want to wish for everone a happy new year...
Happy New Year to you as well!
2008 16:02 schrieb dave cantera:
erik,
you can start here:
http://www.voip-info.org/wiki-Asterisk+cmd+GotoIfTime
http://www.asteriskguru.com/tutorials/gotoiftime.html
daveC
Erik Wartusch wrote:
Hi all,
Im using Asterisk 1.4.11 and I want to proceed some time and date
operations
tim,
I found exactly the same thing... hit or miss... I'm using php.
daveC
Timothy Legge wrote:
Hi
I have created a rudimentary perl script that does most of what I want
but occasionally in seems that a file will not play. I see the
message getting sent to Asterisk but no reply to say that
vincent,
here is a script that I used to convert a single wav file or the entire
directory... no file specified on launch, converts all files in the
current directory...
creates a logfile, although trivial...
daveC
#!/bin/sh
#
# convert-all.sh
#
# convert all *.wav files to .gsm .au formats
tzafrir,
thanks for the note... yep, it is useless...
daveC
Tzafrir Cohen wrote:
On Tue, Jan 01, 2008 at 11:27:54AM -0500, dave cantera wrote:
vincent,
here is a script that I used to convert a single wav file or the entire
directory... no file specified on launch, converts all files
glenn,
check your handset cord... it might be plugged into the wrong port in
the back of the phone. perhaps the headset jack...
daveC
Glenn Gillen wrote:
Hey all,
I've setup my asterisk install on a CentOS5 server, I've got a few
IAX2 and SIP softphone clients connected on the same subnet
in menuselect
file.(xml file)
so no need to add entry in module.conf
Bhrugu mehta
On Dec 27, 2007 7:37 PM, dave cantera [EMAIL PROTECTED] wrote:
bhrugu,
did you try and load it manually?
Modules are compiled in to shared object (.so) files. They are installed
to /usr/lib/asterisk/modules
william,
post your dialplan section for outbound, sip.conf minus passwords, and
CLI output so we can see what is going on... make sure you set
debugging higher or set sip debugging on for iDCS, assuming iDCS is a
sip provider, of course
daveC
William Stillwell (Ki4swy) wrote:
I'm
keshaw,
did you set your sip.conf to only allow g729?
disallow=all
allow=g729
I don't use g729 so the allow= may not be the correct syntax...
here is the config I uise:
disallow=all
allow=ulaw
allow=gsm
allow=alaw
daveC
Keshav K. wrote:
Hi,
I'm using Grandstream IP
bhrugu,
did you try and load it manually?
Modules are compiled in to shared object (.so) files. They are installed
to /usr/lib/asterisk/modules and can be turned on and off from loading
by editing /etc/asterisk/modules.conf. Modules must include
asterisk/modules.h. Modules must also export
this sounds like it might be:
1) a permissions problem...
did you install * as root?
is there a .gsm, .ulaw, .alaw file?
2) sip.conf problem
codec not allowed or not specified, did you allow the proper codec?
daveC
1) permissions
drwxr-xr-x 17 root root 221184 Dec
bob,
look on p20 of 'the book' edition 2, or p16 edition 1
this explains the 3.3v vs 5.0v issue with motherboard slots
http://www.oreilly.com/catalog/9780596510480/
daveC
Bob Smither wrote:
On Wed, 2007-12-26 at 15:46 +1000, Mattt wrote:
Sounds like a PCI bus version issue ;-)
dominik,
along with steve's note /var/log/messages, check
/var/log/asterisk/fullas well
daveC
Steve Totaro wrote:
Dominik Zalewski wrote:
Dear All,
I'm using Asterisk 1.4.16.2 with Zaptel 1.4.7 on Debian with kernel
2.6.18.
I have two analog lines Zap/1 and Zap/2 as group
the /tmp sometimes gets its sticky bit set...
# ls -ld /tmp
will tell you what permissions are set at
/tmp should read:
[EMAIL PROTECTED] ~]# ls -ld /tmp
drwxrwxrwx 7 root root 4096 Dec 26 08:13 /tmp
it may read, which would not allow file/directory creation (notice the
't' in the
russell,
I breezed through the document on 1.6 releases... it looks like a Great
move...
are you thinking, in the future, to moving to a development/release
schedule like linux, a 1.7.x.y?
daveC
Russell Bryant wrote:
Olivier wrote:
Is this going to be included in 1.6 ?
Any commitment
remco,
I just had the same problem/error on my CLI when I added a polycom
shoretel IP-100 phone to my network and enabled mgcp... couldn't
figure out how to get that working yet...
I don't think it is related to 1.4 as I have been running 1.4 has been
running for over a year now without that
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
made.
I attached two files.
One of the attached file is for the external call,which replied with
the PROBLEM all trunks are busy now,please try your call again later.
The second attachment is when i made internal calls and the phone rang.
Please,i will be expecting you
also, you want to think about transcoding... if you have different
technologies, the system load for transcoding would increase...
dean, cool, I didn't know you could hang a few * boxes together with
meetme...
daveC
Dean Collins wrote:
as far as I know it's unlimited and only tied to the
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
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
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
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
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, ...)?
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.
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
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
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
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."
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.
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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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
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
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
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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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
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
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
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
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...
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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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
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
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
. 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.
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
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
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
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
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