vicidial.org

by **ssin14** » Mon Aug 11, 2008 4:11 pm

Hi
I have steup 3 servers

1 for asterisk
1 for MySQL
1 for Apahe/vicidial

in manager.con in server 1 do I have to do anything to the "bindaddress 0.0.0.0"

in crontab -e in server for apache/vicidial

do I have to put server Ip address for asterisk server instead of ****

My sofphone does not ring and I cant see anything happening in asterisk -r

I use 1.4.21.22asterisk
and everything else using from scratch install.

by **mflorell** » Mon Aug 11, 2008 4:24 pm

It would be better to have:

1 x MySQL/Apache/PHP server
2 x Asterisk/VICIDIAL servers

take a look at the LOAD_BALANCE document for some more info.

by **ssin14** » Mon Aug 11, 2008 4:55 pm

thanks for that advice.

But at present I just want to have one asterisk and I can get more we need more.

I wanted to keep MYSQL apache and asterisk separately

I wanted to use other applications that uses MySQL.

Would you know what my problem is why software is not ringing

by **mflorell** » Mon Aug 11, 2008 5:53 pm

output of 'screen -r' on your VICIDIAL server?

by **ssin14** » Mon Aug 11, 2008 6:48 pm

I think nothing is happening when I log in

because it does not show anything when I do asterisk -r or screen -r

1 server is for asterisk (11.0.0.4)/vicidial
1 server is for MYSQL (11.0.0.3)
1 server for Apache/php (11.0.0.2)

now when I do perl install.pl I in all servers have choose my asterisk server )11.0.0.4) as my server

and database 11.0.0.3

and I use 11.0.0.2 to access my vicidial and agc web pages.

I have placed crontab -e only in 11.0.0.2

in phases 6.1

I used 11.0.0.4 for my server updater values and other such as conference values and serverip

But there is manager.conf in 11.0.0.4 that ask for bindaddress.

(do I have to use 11.0.0.2/11.0.0.3 do bind this) I have left it like that.

When you install 3 servers where SQL and apache is separate
do you do anything else.

I normally change things by either running perl install.pl or /etc/astguiclient.conf

I guess this updates dbconnect.php files and we dont have to do anything in here.

I think scripts not running right. How can I know this and find out where this is so.

Thanks

by **ssin14** » Mon Aug 11, 2008 6:54 pm

I get few messages in 11.0.0.2 server where the crontab -e is placed

Subject: cron: /usr/share/astguiclient/AST_conf_update.pl
problem connecting to "localhost", port 5038: Connection refused at /usr/share/astguiclient/AST_conf_update.pl line 188

by **mflorell** » Mon Aug 11, 2008 7:02 pm

You should run the crontab scripts on your asterisk VICIDIAL server

by **ssin14** » Mon Aug 11, 2008 7:14 pm

o.k thanks

now delete crontabs from 11.0.0.2

and now placed in asterisk vicidial server(11.0.0.4)

I get this message coming now

Subject: cron: /usr/share/astguiclient/ADMIN_keepalive_ALL.pl
/bin/sh: /usr/share/astguiclient/ADMIN_keepalive_ALL.pl: No such file or directory

by **ssin14** » Mon Aug 11, 2008 7:32 pm

figured it out

ADMIN_keepalive.pl was not the director so by reinstalling astguiclient it worked

now I can receive calls into my software by another problem

when I answer the phone this happens

Manager 'sendcron' logged off from 127.0.0.1
== Starting SIP/101-081c8560 at default,8600051,1 failed so falling back to exten 's'
== Starting SIP/101-081c8560 at default,s,1 still failed so falling back to context 'default'
[Aug 12 12:28:30] WARNING[21438]: pbx.c:2483 __ast_pbx_run: Channel 'SIP/101-081c8560' sent into invalid extension 's' in context 'default', but no invalid handler

by **ssin14** » Mon Aug 11, 2008 7:47 pm

what I can notice is

when it throws calls upon login..... before the call is answered I can see in asterisk -r
that

it gets log off
==Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1

This happend same time before I answer a phone call thats coming from
asterisk

by **mflorell** » Mon Aug 11, 2008 8:07 pm

check your meetme.conf file

by **ssin14** » Mon Aug 11, 2008 8:56 pm

[rooms]
conf => 8600
conf => 8601,1234

conf => 8600001
conf => 8600002
conf => 8600003
conf => 8600004
conf => 8600005
conf => 8600006
conf => 8600007
conf => 8600008
conf => 8600009
conf => 8600010
conf => 8600011
conf => 8600012
conf => 8600013
conf => 8600014
conf => 8600015
conf => 8600016
conf => 8600017
conf => 8600018
conf => 8600019
......
till 8600200

I dont think anything wrong

by **ssin14** » Mon Aug 11, 2008 9:17 pm

I again noticed another big things that I have no idea.

# vi /usr/src/astguiclient/docs/conf_examples/extensions.conf.sample
and

# vi /usr/src/astguiclient/docs/conf_examples/extensions.conf.sample-1.4

I use 1.4.21.22 asterisk which means that

I have put that "F" and the end of conference entries???

thats the difference I found?

by **ssin14** » Mon Aug 11, 2008 9:20 pm

sorry

its clearly written there in scratch install...

cheers! will see if this works now

by **ssin14** » Mon Aug 11, 2008 9:49 pm

still not workinggg

what may be the cause

my meetme is o.k now

extensions o.k
manager.conf o.k

by **ssin14** » Mon Aug 11, 2008 9:58 pm

[Aug 12 14:53:29] VERBOSE[28260] logger.c: == Starting SIP/101-08200498 at default,8600051,1 failed so falling back to exte
n 's'

this is what I can get from /var/log/asterisk/messages

somehow it cant handle call at 8600051,1

Any ideas or what can I do so we can have more information about the problme.

thanks

by **mflorell** » Mon Aug 11, 2008 10:37 pm

post results of "show application meetme" in the asterisk CLI

by **ssin14** » Mon Aug 11, 2008 10:55 pm

it says application not registerd

Your application(s) is (are) not registered

what do I need to do?

(I had to put "core show application meetme", I think coz of 1.4.21.22 version)

Thanks

by **mflorell** » Tue Aug 12, 2008 1:35 am

You need to recompile zaptel and ensure that ztdummy is being built, then AFTER compiling zaptel you need to compile Asterisk.

by **ssin14** » Tue Aug 12, 2008 2:21 am

"ensure that ztdummy is being built"

How will ensure this...... I infact been trying to figure this out

by **mflorell** » Tue Aug 12, 2008 2:43 am

What version of zaptel are you compiling?

after compiling zaptel do you get any errors?

can you run the following commands without errors?
modprobe zaptel
modprobe ztdummy

by **ssin14** » Tue Aug 12, 2008 2:54 am

using zaptel-1.4.11

/lib/modules/2.4.31/kernel/drivers/usb/host/usb-uhci.o.gz: init_module: No such device
/lib/modules/2.4.31/kernel/drivers/usb/host/usb-uhci.o.gz: Hint: insmod errors c an be caused by incorrect module parameters, including invalid IO or IRQ paramet ers.
You may find more information in syslog or the output from dmesg
/lib/modules/2.4.31/kernel/drivers/usb/host/usb-uhci.o.gz: insmod /lib/modules/2 .4.31/kernel/drivers/usb/host/usb-uhci.o.gz failed
/lib/modules/2.4.31/kernel/drivers/usb/host/usb-uhci.o.gz: insmod ztdummy failed

Thanks matt...

by **mflorell** » Tue Aug 12, 2008 7:37 am

Do you have the usb-uhci Linux module enabled in your kernel?

What Linux kernel version are you using?

by **ssin14** » Tue Aug 12, 2008 3:29 pm

I am using 2.4.31
I think its enabled since when I do show modules it shows.

and i do lsmod this all comes: here can see uhci there and even x100p

Module Size Used by Not tainted
snd-pcm-oss 36736 0 (unused)
snd-mixer-oss 12376 0 [snd-pcm-oss]
uhci 24284 0 (unused)
ehci-hcd 17516 0 (unused)
usbcore 59148 1 [uhci ehci-hcd]
snd-intel8x0 18304 0 (unused)
snd-ac97-codec 58556 0 [snd-intel8x0]
snd-pcm 54344 0 [snd-pcm-oss snd-intel8x0 snd-ac97-codec]
snd-timer 13764 0 [snd-pcm]
snd 32772 0 [snd-pcm-oss snd-mixer-oss snd-intel8x0 snd-ac97-codec snd-pcm snd-timer]
soundcore 3396 4 [snd]
snd-page-alloc 4712 0 [snd-mixer-oss snd-intel8x0 snd-pcm snd-timer snd]
eepro100 18836 1
mii 2272 0 [eepro100]
hisax 501264 0 (unused)
isdn 115820 0 [hisax]
slhc 4592 0 [isdn]
isa-pnp 29968 0 [hisax]
wcfxo 7712 0 (unused)
zaptel 219424 0 [wcfxo]
pcmcia_core 39172 0
ide-scsi 9392 0
agpgart 45508 0 (unused)

by **ssin14** » Tue Aug 12, 2008 3:37 pm

I did a "zttest"

and got this

Best: 99.996 -- Worst: 99.992 -- Average: 99.995032, Difference: 99.995032

which should mean my zaptel interface is working (x100p)

but on other hand ztdummy is not showing...

by **ssin14** » Tue Aug 12, 2008 3:48 pm

the Makefile below shows:

TOPDIR_MODULES:=pciradio tor2 torisa wcfxo wct1xxp wctdm wcte11xp wcusb zaptel ztd-eth ztd-loc ztdummy ztdynamic zttranscode
SUBDIR_MODULES:=wct4xxp wctc4xxp xpp wctdm24xxp wcte12xp

there is no "#" in front of ztdummy.

by **mflorell** » Tue Aug 12, 2008 7:28 pm

I have not tried using a 2.4 Linux kernel on a new install in a couple years. This may be your problem.

by **ssin14** » Wed Aug 13, 2008 9:27 pm

But do we really need 2.6 and ztdummy

when x100p is installed

I manage to get meetme application running

show modules shoe:

app_meetme.so MeetMe conference bridge 0

show application meetme show:

-= Info about application 'MeetMe' =-

[Synopsis]
MeetMe conference bridge

[Description]
MeetMe([confno][,[options][,pin]]): Enters the user into a specified MeetMe
conference. If the conference number is omitted, the user will be prompted
to enter one. User can exit the conference by hangup, or if the 'p' option
is specified, by pressing '#'.
Please note: The Zaptel kernel modules and at least one hardware driver (or ztdummy)
must be present for conferencing to operate properly. In addition, the chan_zap
channel driver must be loaded for the 'i' and 'r' options to operate at all.

The option string may contain zero or more of the following characters:
'a' -- set admin mode
'A' -- set marked mode
'b' -- run AGI script specified in ${MEETME_AGI_BACKGROUND}
Default: conf-background.agi (Note: This does not work with
non-Zap channels in the same conference)
'c' -- announce user(s) count on joining a conference
'd' -- dynamically add conference
'D' -- dynamically add conference, prompting for a PIN
'e' -- select an empty conference
'E' -- select an empty pinless conference
'F' -- Pass DTMF through the conference.
'i' -- announce user join/leave with review
'I' -- announce user join/leave without review
'l' -- set listen only mode (Listen only, no talking)
'm' -- set initially muted
'M' -- enable music on hold when the conference has a single caller
'o' -- set talker optimization - treats talkers who aren't speaking as
being muted, meaning (a) No encode is done on transmission and
(b) Received audio that is not registered as talking is omitted
causing no buildup in background noise. Note that this option
will be removed in 1.6 and enabled by default.
'p' -- allow user to exit the conference by pressing '#'
'P' -- always prompt for the pin even if it is specified
'q' -- quiet mode (don't play enter/leave sounds)
'r' -- Record conference (records as ${MEETME_RECORDINGFILE}
using format ${MEETME_RECORDINGFORMAT}). Default filename is
meetme-conf-rec-${CONFNO}-${UNIQUEID} and the default format is
wav.
's' -- Present menu (user or admin) when '*' is received ('send' to menu)
't' -- set talk only mode. (Talk only, no listening)
'T' -- set talker detection (sent to manager interface and meetme list)
'w[(<secs>)]'
-- wait until the marked user enters the conference
'x' -- close the conference when last marked user exits
'X' -- allow user to exit the conference by entering a valid single
digit extension ${MEETME_EXIT_CONTEXT} or the current context
if that variable is not defined.
'1' -- do not play message when first person enters

by **mflorell** » Wed Aug 13, 2008 11:05 pm

ztdummy relies on the kernel to get timing. It is much easier to get ztdummy working on 2.6 Linux kernels.

by **ssin14** » Thu Aug 14, 2008 3:58 am

Hi If I am going to install 2.6 kernal, after its done do I have to reinstall evrytjhing again.

thanks

by **mflorell** » Thu Aug 14, 2008 8:05 am

I don't think you will need to install everything, but certainly zaptel/asterisk will need to be reinstalled.

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