Hi!
I've been on this list for over 3 years and this is my first post.
We have a reporting application for Asterisk that is soon to be in beta.
It's a windows application that generates reports from log files (CDR and
queue).
It has a drag and drop approach to report creating. There is a pivot
Hi!
I have the same phone with the same problems:
1. Asterisk box does not have fixed IP address, but dyndns name.
2. Phone is at a different location, connected to a router/ADSL modem
Siemens Gigaset (with option not to disconnect from internet ever - set
on).
3. Inside asterisk LAN, phone
Hi!
I don't understand what you mean by : „configure voice part on it, but I
can give general guidelines:
First you setup SPA3000 web UI:
1) Line1 Tab:
Sip settings:
SIP port : 5060
Proxy and Registration:
Proxy: Asterisk IP
Subscriber Information:
Display Name:
Hi everyone!
I just want to thank everybody. My phone works now and just a little hint:
set qualify=no in sip.conf of your phone's extension.
Best regards
Mihaela MJ
On 1/21/07, Token PBX [EMAIL PROTECTED] wrote:
On 1/20/07, Pavel Jezek [EMAIL PROTECTED] wrote:
you have probably
Hi!
I did manage to load phone with SIP image : SIP70.8-0-3S, made
SEP-MAC.cnf.xml, but phone never read the configuration from it.
On the screen it's written Unprovisioned, and phone is not trying to
register with asterisk.
Please help!!
MihaelaMJ
-
MAC.cnf.xml.
Please help and thanks.
Mihaela MJ
- Original Message
From: Token PBX [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Saturday, 20 January, 2007 1:01:25 PM
Subject: [asterisk-users] Cisco 7970 Unprovisioned
Hi!
I did manage to load phone with SIP image : SIP70.8
On 1/20/07, Pavel Jezek [EMAIL PROTECTED] wrote:
you have probably something wron in config file and phone refuses to
configure,
here is my minimalistic file for 7941/61, you can try...
device
deviceProtocolSIP/deviceProtocol
sshUserIdadmin/sshUserId
sshPasswordadmin/sshPassword
devicePool
Hi!
I have several SPA3000 devices (older versions of SPA3102) and they are
working OK, sound quality is good. It is very configurable to the slightest
details. I use it whenever I need just one or two FXO ports, like for small
scale PSTN integration, or for connecting some other equipment that
The comment below makes me wonder could ttyIAX be configured to answer
mgetty?
I have made the mgetty talk to ttyIAX however, as soon as a ring comes
into th eextension , mgetty shuts down... so I cannot keep the signal up.
I tried to use the pppd daemon directly with ttyIAX and it said that the
mgetty dump:
10/25 11:23:17 IAX tio_get_rs232_lines: TIOCMGET failed: Invalid argument
10/25 11:23:17 # data dev=ttyIAX, pid=15158, caller='none',
conn='DIRECT', name='', cmd='/bin/login', user='Fedora Core release 3
(Heidelberg)'
--
10/25 11:23:36 IAX mgetty: experimental test release
I have downloaded SJPhone - and well.. it does connect to my system,
however popping audio is heard when i dial my music on hold extension...
the quality is really really bad..
i have a WLAN-SDIO utility, the signal strength is at 11mb/s. however. is
that sufficient? The codecs for sjphone are
2.6.12
On Fri, 2005-10-21 at 09:27 -0700, [EMAIL PROTECTED] wrote:
I received some postings back, as I was trying to do the same thing.
it' is a problem with Kernel 2.6... 2.4 works fine .. this is the
summary
I got from reading the posts before.
I hope that helps... I dont have the
I received some postings back, as I was trying to do the same thing.
it' is a problem with Kernel 2.6... 2.4 works fine .. this is the summary
I got from reading the posts before.
I hope that helps... I dont have the ability to go DOWn in kernel to 2.4..
I'm going to poll the group one more
Trixter:
Thanks for the guide to setting this up:... I have tried the below
configuration with my settings, and when I place /goiax-in after my
register command, my register statement fails.
If i remove it. I get a Rejected connect attempt from goiax's server IP,
trying to reach 's@'
I have put
That is What I stated in the email.. my GOIAX #. not the DID #.
That is not the issue.
for the incoming context put your goiax.com http://goiax.com phone
number
not the free DID number but the other one.
On 10/19/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Trixter:
Thanks for the
That did it... Thank you.
putting /goiax.com phone number after the register line caused it to not
register any more... and i would get error
server1.goiax.com/my goiax# could not be found.
anyways.. thanks for your help guys :)
Replace
[goiax]
with
[PHONENUMBER]
username= don't
I have 3 different SIP extensions in my DIAL string.
i.e. I have HOME_PHONES_TO_RING=SIP/2000SIP/2001SIP/2002
so in my Extensions file i have Dial(${HOME_PHONES_TO_RING},30,tTr)
So... when the home phone line rings, all three phones ring.
Anyways.. the problem is.. in the CDR log, sometimes
David:
Also port 1:2 is a good idea to forward to the server as well..
David,
Yes, I've also forwarded port 4569 to the server.
Since the router is forwarding to the server, I cannot
forward it to the client as well -- however, as the
client isn't going out past the LAN, it
I'm game for using them /and testing them.
Ben..
Roman:
I created two bash scripts called Mail2Fax and Fax2Mail for use with the
asterisk sever.
They leverage the app_txfax and app_rxfax scripts, along with ast_fax.
They
make using these apps a lot easier, including being able to mail
Has anyone succesfully used TDMoE on Fedora Core 2.6.12+ Kernel versions?
I'm having the issue that is in the Mantis bug database with badness with
the kernel.
My Story:
I can get the dynamic span to come up and show OK in the zttool on both
machines. However i get errors every second (Warning:
Right... I had seen the multiple issues, on the flip side, the only
solution was NOT to use Kernel 2.6.+...
So... I'm happy that the behaviour is reproduceable (from what I have seen
from my steps, to that of others, and other Distros..)
Anyone out there have the magic wand to make it work with
In an answer about the voicemail box extension, I have 3 different sipura
boxes linked to the same voicemail account?
Sipura 1 = Sipura 2000 - with line 1 being my home phone
Sipura841 - Extension 2 = also my home phone
Sipura 2 = Sipura 1001 = phone in the bedroom - being the home phone
so i
I would check your /etc/ld.so.conf file and make sure that you have the
library path for the IODBC libraries in there...
Then run ldconfig
and try reloading asterisk again.
Hello.
It's possible to use IODBC instead UNIXODBC with realtime?
As I see, the res Makefile load a odbcinst.h file,
Then did you do a make clean / make / make install?
Then do show applications at the CLI prompt after you have restarted
asterisk.
service asterisk stop
service asterisk start
...
I downloaded Cepstral to my Asterisk Box. I did the install and let it
install to /opt/swift.
I brought down
I have configured TDMoE sucessfully and I am able to make a Zap connection
from one box to the other.
The question I have is..
I'm getting repeated errors every second on both systems..
Oct 3 09:53:16 WARNING[4409]: chan_zap.c:6252 handle_init_event: Detected
alarm on channel 1: No Alarm
Oct
the app_cepstral.c file had a problem that it was trying use
#include ../asterisk.h
I had to force it to where asterisk.h was located... in my case it was in
/usr/src/asterisk/include
so i changed the #include to say
#include /usr/src/asterisk/include/asterisk.h and then it would compile
YUP!!!
I guess working on this at 11:30pm at night, really need your wits about
you... Thanks... did that and it worked like a champ..
One other question I have the app_dtmftotext.c file is located at the
root of spandsp... however, when I had this file in my Makefile to
compile, and then
Since the search engine on voip-info.org is not working correctly with old
links, etc..
I was curious if there is some hidden talent in the IAX2 outbound dialing?
What I'm asking about is:
Dial(IAX2/g1/${EXTEN})
Is there a way to set up groups like the above command using either SIP or
IAX2
I'm trying to put together a package of asterisk-head, spandsp, and
app_rx,tx fax.
I can get everything to compile:
spandsp-0.0.2pre20
asterisk-head (cvs co -r HEAD asterisk)
the app_rx/tx from soft-switch.org in the 1.1 folder
However, asterisk complains that there is unused symbols when
On Sat, Oct 01, 2005 at 08:32:34AM -0700, [EMAIL PROTECTED] wrote:
I'm trying to put together a package of asterisk-head, spandsp, and
app_rx,tx fax.
I can get everything to compile:
spandsp-0.0.2pre20
asterisk-head (cvs co -r HEAD asterisk)
the app_rx/tx from soft-switch.org in the 1.1
This is probably generic to any device... However..
Incoming callerid is working with number only.
If I try for any reason to use the function Set(Callerid(name)=blah
blah) it will then send only the outgoing extension as the callerid to
the phone that i s connected to the sipura device...
I
ACCESS supports ODBC driven connections..
Well guys here comes the fun part. I have a Microsoft access (VBA)
application that interfaces with my SQL database. This app pulls of info
from the SQL record and than picks up the phone and dials that locations
number. I have purchased a few
I have looked into this callerid problem now for a few hours.
1) Caller id on a sipura-2000 now shows:
cidname
2000
Where cidname is the new outputted formate from the cid_rewrite agi script
and 2000 is the exten number.
In looking at the Dial() application,
option o
'o' -- Original
Hi All.
After some input, I created a V1.1 version of my ODBC VM retrieval from
the ODBC_Storage
It now uses either Mysql or unixODBC drivers to connect to the database
I didn't have php compiled with unixODBC so i had to recompile it in
./configure --with-unixODBC --with-mysql
Ok.
I was sucessful in installing ODBC storage
I'm using MySQL in the backend as it is. but all my drivers are now ODBC.
I am running asterisk-cvs head as of last night 9/19/05
My question is this... the old voicemail.cgi script that allowed checking
voicemail no longer works etc, and never
Hi all..
After some further research I have come up with a quick and dirty way to
playback the longblob recordings from the ODBC database for those of you
that are running the ODBC storage for voicemail.
Have a look
http://www.itsngroup.com/software/asterisk/downloads/ODBC_VM_1.0.tar
A little
I thought that I would try this on iax.conf as well however I still get
asterisk asterisk as the callerid name and num.
I have the latest CVS as of 8/17/05
Has anyone have this working with iax incoming?
Thanks
Ben
That worked. The following line also got rid of asterisk without
entering
I thought that I would try this on iax.conf as well however I still get
asterisk asterisk as the callerid name and num.
I have the latest CVS as of 8/17/05
Has anyone have this working with iax incoming?
Thanks
Ben
That worked. The following line also got rid of asterisk without
entering
I would like to do the same thing, and the easiest would be to use MySql
and a web connector :..
I can help.
Ben
Does anybody have an example follow-me configuration web page code
written in either php or perl that can write out the follow-me config
into the asterisk files?
I'd like to
Remember in the good ol days when answering machines were smart enough to
know when there was a message on the machine, and it would pick up after 2
rings rather than 4? (that is, if you knew how to turn it on - that
required to know how to set the time on your VCR to avoid the flashing
12:00:00)
I thought that I would try this on iax.conf as well however I still get
asterisk asterisk as the callerid name and num.
I have the latest CVS as of 8/17/05
Has anyone have this working with iax incoming?
Thanks
Ben
That worked. The following line also got rid of asterisk without
entering
We will start installing TE411 next week, I'll keep the list informed !
jack
Eric Rees wrote:
Has anyone on the list tried one of these new cards with built-in echo
cancellation?
This electronic message transmission, including attachments, is for the
exclusive use of the individuals to
Hello all.
I am looking for the great callerID rewrite script that does the 411
lookup and then stores the information in a database.
If there is information in the Database for the callerid coming in, then
use that and pass it along to the phone.
I lost my entire system hard drive this week,
Hi all, my client wants to double his bandwidth by using 2 x DSL lines
into one Asterisk network
How can I do this ?
Thanks
Henry
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
perform as
advertised.
Best regards,
Bernard
*/VoIP-PBX [EMAIL PROTECTED]/* wrote:
Hi all, my client wants to double his bandwidth by using 2 x DSL
lines
into one Asterisk network
How can I do this ?
Thanks
Henry
This is not the problem
There is an example for faxing. you should set some other parms. ie.
unique filename etc., you also need permissions etc.
Look at NVBackground detect from Newman Telecom. Use the wiki and search
for NVFaxDetect
HTH
Hello i would like to receive incoming faxes thru'
Hello:
I have been using the asterisk system now for almost 5 months. I'm very
happy with it's performance, however I have been using the RXFAX on and
off for the last month or so. I gave up after a while and just had it
route to my analog fax machine in the fax context.
However, I have always
I have found that in the iax_buddies database, if a field doesn't exist...
then just create it :) then it works.
The schemas have been outdated it seems from the create table blah blah
blah
so if it is in the IAX.conf file, and it's not in the iax_buddies table
structure, add the field.
Tada!
I have a question for those of you out there using VoicePulseConnect for
incoming did
I have in my Realtime extensions Database
(the x's are replaced with my phone number)
context = voicepulse-in-01
exten = xx/
Priority=1
app=NoOp
appdata = Incoming call with no callerid on xx
Ditto That:)
Thats what i use!
What's wrong with :-
host W2K {
hardware ethernet 00:30:1B:AC:39:E3;
fixed-address 192.168.1.130;
}
this box always gets the same IP and I know who's got what.
___
You could simply put something in your dialplan / ivr that allows for
mysql insert into the cdr..
I've gotten my CDR working the way I like, but I am looking to customize
it a bit. I have set up an IVR menu, which works great. I would like to
be able to capture the prompted DTMF digits
Easy:
I just had to do the same thing recently.
in your inbound context of [voicepulse-in-01]
you have something like this in your extensions.conf
exten = 1234567890,1,NoOp(incoming call on ${EXTEN})
exten = 1234567890,2,Dial(blah blah blah)
exten = 9876543210,1,NoOp(IncomingCall on ${EXTEN})
Easy:
I just had to do the same thing recently.
in your inbound context of [voicepulse-in-01]
you have something like this in your extensions.conf
exten = 1234567890,1,NoOp(incoming call on ${EXTEN})
exten = 1234567890,2,Dial(blah blah blah)
exten = 9876543210,1,NoOp(IncomingCall on ${EXTEN})
the line
and then it runs NVFaxDectect from newman telecom, and if it's a fax, it
jumps to the fax priority and i then dial ${FAX} where FAX=SIP/exten of
fax
and walla, the fax machine answers and all is well.
I have installed two Sipura 3000's on my office pbx as a test, they work
well an have
, it jumps to the fax priority and i then dial
${FAX} where FAX=SIP/exten of
fax
and walla, the fax machine answers and all is well.
I have installed two Sipura 3000's on my office pbx as a test, they
work well an have some great features including fax detect,
but I was
hoping to allow
How does one process / capture a timeout that has happened in using an AGI
script.. Preferably PHP.
I know you can set the wait timeout for a certain time, but how does the
script continue?
Thanks
Ben
___
Asterisk-Users mailing list
The patch is not that much.
I just opened it up - and then typed in the patch information into the
Makefile manually.
I have to do this for another application, so it wasn't that hard to do.
Even if i tried to copy / paste it would complain (the make process) on
the app_txfax.so : app_txfax.c
I have an open ticket with them, and I have had better quality now that
they saw something wrong with my account on their end.
Anyways, the next question is in iax_peers / iax_buddies /
iax_friends, etc etc in realtime.
In the Codec order (allow) i have g729;ulaw;ilbc;gsm
However, LiveVOIP
Is anyone having problems with LiveVOIP for outbound calls since their
network upgrade a week ago?
Ever since the network upgrade, it takes 2-3 times MINIMUM in order for a
call to go from my system to theirs.
I haven't changed any configs on my side, it just says
call accepted by blah
I was just going to ask this same question
Is this the normal behavior that you have to do, jump back to the .conf file?
It is how I have it configured, but it's more a hybrid than a true
realtime system.
Thanks
Use the Goto statement with '|' instead of ','. And make tables for each
I have been trying to get this script to work as well
cid_rewrite
However this is what the CLI reports:
-- Executing EAGI(Zap/1-1, cid_rewrite/cid_rewrite.php|us)
-- Launched AGI Script /var/lib/asterisk/agi-bin/cid_rewrite.php
AGI Tx agi_request: cid_rewrite.php
AGI Tx agi_channel:
That was it!
Told me exactly what i needed.
in the php file it's looking for /usr/bin/php
mine was in /usr/local/bin/php
ARGH! so simple
Thanks for that though!!
it's working Great!!!
Run ./cid_rewrite.php from the the shell to see where it's failing.
-Original Message-
From:
Hello all:
I am trying to use the mysql command to retrieve information from a mysql
database.
my example here was formed from using the wiki reference to using the
mysql command.
The problem is with the fetch command.
Here is the macro code:
Mysql(QueryString=SELECT\ ivr-password\ from\ users\
A while back I converted back to static conf files from a database setup.
However I decided to tackle it again.
The problem that I was experiencing, was, there was no stutter tone on my
sipura 2000 or 3000 when there was a voicemail left at either extension
when I was using mysql setup for peers
Hi.. .and thank you for your help.
I just tried your example.. and yes it did return what i wanted (with my
information)
One thing I need to add here i'm using mysql-realtime configuration.
So when I'm running using mysql extensions I get this:
-- Executing MYSQL(IAX2/[EMAIL
The information that is below is from the CLI window.
the # for ${resultid} only appears if you using the .conf file. If you
realtime with mysql resultid has no value and thus is blank.
I have copied and pasted exact same line from the .conf file to the
appdata field in mysql and i get the same
I was searching for help on how to handle the errors that are returned
from the MeetMe application.
for instance.
1) if a user tries to join a conference that is locked, allison says that
the conference is locked and then comes back to the dialplan, however it
does not continue down the
There are plenty on the wiki...
Are there any BYOD providers out that that people have had positive
experiences with? I have broadvoice and they suck lately. Anyony have
anyone with a good amount of peers and not a lot of downtime?
--
Michael Lyszczek
New York, NY, 10282
NEW EMAIL :
It looks like from the ztcfg output that you are trying to reference port
3, which is if your looking at the card with modules up. the one on the
far right.
Move the module to the far left... and that will be channel 1
The far left, is the TOP port on the card.
HIH
Hi, I did not find any
I have had issues with using * in Realtime for dialplan, voicemail,
sip_buddies, sipfriends, etc... sip this and that.
The problem that I have is the VMI doesn't work on my Sipura ata's.. I
have a 1001, 2000, and 3000.
The VMI only works when I configure the sip registration in the sip.conf
Hi all,
I am thinking in implementing asterisk into my buisness. I heard all
sorts of good things about it. The question im asking my self is what
linux distribution is best to use? Do you know what distribution they
use for their asterisk training?
Thanks for reading me.
PBXER
Scott Stingel wrote:
Sid-
Try connecting one port to another. Note that one of the ports must
be set up as cpe and the other as net in zapata.conf when you loop
them together like this.
A suitable crossover cable for testing can be constructed by cutting
up a CAT 5 cable, and connecting:
Pin
Eric Bishop wrote:
Well it's clear now that this is not an isolated issue. Has anyone
been in touch with Digium about this issue? I have logged a support
issue with them, but thus far have not received a response. Anyone
had better luck with Digium support and the Compaq/HP G4 server
series?
On
Read Asterisk install,
You need to install libssl package
Wesley Jay Deypalan wrote:
Hi,
I tried to compiled Asterisk on a Mandrake 9.2 and I encountered this error
command: make
after compiling for sometime then this error appeared
gcc -g -o asterisk -Wl,-E io.o sched.o logger.o frame.o
[EMAIL PROTECTED] wrote:
Dear Users,
i have the following scnario.
1. Alcatel PBX with e1 module
2. asterisk server with 2 digium e100p cards. 1 connected to pstn e1, 1
connected to alcatel pbx.
i m having problem in outgoing from alcatel.
incoming from pstn - asterisk - alcatel working fine
Joost , I am running 2.6.8 SMP Sarge/Debian on a HP ml330 an have no
problem.
The Zaptel hardware is e T410P
Are you running without Zaptel Hardware ?
Jack
Joost Kraaijeveld wrote:
Hi all,
For some reason Music On Hold does not work. I have searched the internet for
solutions but found
Not realy sure, but it seems that you are missing Zaptel timing,
Just Google for ZTDUMY or viop-org, ZTDUMY take TDM timming from a
USB-DEVICE
this might be your problem...
I am running in under root!
Jack
Joost Kraaijeveld wrote:
Hi Jack,
[EMAIL PROTECTED] schreef:
Joost , I am running
the commande is make;make install
not make:make install
you typed : instead of ;
Rodney Acosta Coya wrote:
i found de file Makefile but i dont now what to do with it
look at this
inux:/inst/pbx/asterisk-1.0.0 # make:make install
bash: make:make: command not found
linux:/inst
Dear Khaled,
I thing you must read the documentation a little bit more deapely!
does zaptel compile ok ?
which kernel are you using ?
have you configure the zaptel.conf file
which parameters are you using for r2 signaling ?
refer to this page as guide for starting
Hello,
I just installed a Asterisk box with A
TDM400P and a TE405P
Installing the TE405P alone in the (connected
to Ericcson -E1 euro ISDN ) everythings works great!
installing the TDM400P alone ( connected to analog
bundle works great)!
Configuring both card gives problem.
if I
Hello ,
You need to check if the module relative to your card is loaded with lsmod
depending on the card you have will need to start the following command
modprobe wcfxs for tdm400p
or
modprobe wcfxo - for fxs card ( x100p)
or
modprobe wct1xxp for single E1/T1 or
modprob wct4xxp for quad
Dear Friends,
I'm trying to install the spandsp in my Asterisk box, in order to use
rxfax and txfax.
Well, following the instruccions i try to patch the Makefile within the
asterisk/apps source directory and found the following error message
[EMAIL PROTECTED] apps]# patch Makefile.patch
Dear Friends,I'm trying to install the spandsp in my Asterisk box,
in order to userxfax and txfax.Well, following the instruccions i
try to patch the Makefile within theasterisk/apps source directory and found
the following error message[EMAIL PROTECTED] apps]# patch
Makefile.patchpatching
I am looking for a provider that will provide a
phone number via IAX, IAX2 or SIP using numbers in Barcelona or Tarragona or
even Other City in Spain. May be a 902 number.
Thanks in advance.
Dear Sirs,
I'm running an Asterisk 0.9.1 in a Fedora Core 2 box.
I installed a X100P card on my box and when i try to load modules i am
rejected.
When i type modprobe zaptel my Fedora respond : FATAL: Module zaptel not
found. . The same uccurs when i type modprobe wcfxo
May you help me.
Thank
Title: Message
One
way maybe not the best, but can work.
If you
are pulling down the config from a tftp server. Turn on telnet to the
phone. Create a perl script to telnet and reboot the phone. When the
phone boots back up it will grab the new config from the tftp
server.
-gcc
Is there anway to do MOH on a FXS extension like what is done using SIP.
There has to be a way within manager or something, to send this call to
MOH and then retreive the call.
I need to set this up, so users are just hitting one button to put
callers on hold and one or another button to retrieve
I am new to the sip side of things and have a question regarding
ringback. I don't hear ringback when using the sjphone softphone when
dialing internal extensions. It's fine when dialing outside over the
pstn.
Is this a issue of the softphone, configuration or sip in general?
Thank you,
-gcc
Ok...
Let me give a better example.
A caller calls in and a user picks up the phone. Then the user needs to
put the caller on hold so he can go check on something. He would like
to press the hold button on the phone and hang the receiver up. He can
do this, but the caller never hears MOH.
I know this has been covered more times than to mention and this is
where I got most of my info from... But I am having issues with this. I
can't seem to get the phone to register with *. This is being tested on
a internal network right now.
Here is the setup -
sip.conf
[general]
port = 5060
Thompson
Posted At: Thursday, December 18, 2003 11:16 PM
Posted To: Asterisk User Group
Conversation: [Asterisk-Users] SIP / X-ten Softphone
Subject: Re: [Asterisk-Users] SIP / X-ten Softphone
As with any debugging, you should try steps seperately...
- Original Message -
From: PBX [EMAIL
Has anyone come up with any ideas on how to place a call on hold and
have them use MOH, with out having to park the call? This is using a
analog phone. I know you can hit flash or # but that just gives me dial
tone.
I need to come up with a solution that the user can place the caller on
hold,
I should have stated this. Is there any Analog phones that can do this.
-gcc
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ernest W.
Lessenger
Posted At: Tuesday, December 09, 2003 11:57 PM
Posted To: Asterisk User Group
Conversation: [Asterisk-Users]
Ok - Here is where I am at. I know this topic has been discussed
before, but never a solid answer was set in place. Is anyone aware of
any phones that can put a caller on hold and the caller hear MOH by the
user pressing the hold button. I understand most phones are only muting
the speaker and
Ok,
I have spent that past 4 - 5hrs working (trying to) figure some AGI
syntax out in perl. Maybe I'm Looney / slow or what I don't know, but I
am lost. I need to figure out how to send STDIN into the script. I
understand the concept of it, but lost when it comes down to it. No
matter what I
:06 AM
Posted To: Asterisk User Group
Conversation: [Asterisk-Users] AGI - Freakin Lost
Subject: Re: [Asterisk-Users] AGI - Freakin Lost
http://asterisk.gnuinter.net
Jeremy McNamara
PBX wrote:
Ok,
I have spent that past 4 - 5hrs working (trying to) figure some AGI
syntax out in perl. Maybe
Ok.. I was thinking about this.. It is not a very wise decsion to put
the user input in a loop.. So how could I do some error checking outside
of the loop?
-gcc
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of PBX
Posted At: Thursday, November 27, 2003 9
I have a client who needs an application for there field techs to call
in when they arrive on site and when they leave. The logic behind it
seems pretty simple. I am going to write something in AGI to capture
some DTMF tones and update this data into MySQL to run some reports
from.
But here's
I think I figured out my callerid issue... For some reason the callerid
is not getting passed anymore... This is controlled by the PhoneCompany
- Yes?
-gcc
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of PBX
Posted At: Wednesday, November 26, 2003 7:22
] On Behalf Of PBX
Sent: Wednesday, November 26, 2003 7:42 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] AGI - CallerID ??
pamAssassin 2.55 (1.174.2.19-2003-05-19-exp)
I think I figured out my callerid issue... For some reason the
callerid
is not getting passed anymore... This is controlled
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