Another "Noone is in your session: 8600051"

[willbordoni]

hi,
I'v already read lots of topics about my problem but none of them solved it, so I'm posting a new one:

I'v already installed vicidial from scratch (with asterisk running properly), but when an agent logs in he get the error "Noone is in your session: 8600051" every 10s.
Moreover, it seems that vicidial is not dialing because i get nothing in my asterisk console when i clic on "Dial next number".

There is NO problem with the hopper (it's filled with leads)

I think ntpd is OK:

Code: Select all

# ntpq -p
     remote           refid      st t when poll reach   delay   offset  jitter
==============================================================================
+xxxxxxxxxxx  195.83.222.27    2 u    7   64  377   54.888   99.246   2.958
*xxxxxxxxxx     192.93.2.20      2 u    7   64  377   55.675  100.320   2.565
+xxxxxxxxxxx   195.83.180.3     2 u   12   64  377   61.506   95.192   2.032

# screen -ls
There are screens on:
        12161.ASTupdate (Detached)
        12167.ASTlisten (Detached)
        12163.ASTsend   (Detached)
        12170.ASTVDauto (Detached)
        8590.ASTVDadapt (Detached)
5 Sockets in /var/run/screen/S-root.


I think the problem comes from AST_update.pl because the tables "live_channels" and "live_sip_channels" are always empty


I'm working on it for 3 days, I really need some help or ideas.

W.B

[kwatson]

Hi, do you have vicidial conferences defined on the server with the correct IP address?

Also do you have the dial plan entry relating to that IP address and to the agents phone. Either the dialler cant dial back to the phone or conference the phone to the conference room.

Thanks

Kenny

[mflorell]

Asterisk version?

[willbordoni]

Thanks a lot for your answer.

Asterisk version: 1.2.31.1
Vicidial version: 2.0.4-119

I think my .conf are OK since they are the sample of vicidial. I replaced the IP with mine of course(127.0.0.1)
I also gave my 2 sip phones the correct login/pass defined in these .conf files (cc100/test and cc101/test)

When i make a direct call without vicidial, it's OK

What is annoying me is that Asterisk console gets nothing when vicidial tries to dial something... and always this message "noone is in your session"
Is there a way to be sure that vicidial is able to communicate with asterisk?

[mflorell]

Are you using the vicidial manager.conf file?

Is there a reason you are not starting Asterisk with the start_asterisk_boot.pl script?

[willbordoni]

i get this message in the console now, when an agent logs in:

Code: Select all

Mar 25 15:06:38 WARNING[23191]: chan_sip.c:2018 create_addr: No such host: 100
Mar 25 15:06:38 NOTICE[23191]: channel.c:2514 __ast_request_and_dial: Unable to request channel SIP/100

I'm going to investigate, your help is still more than welcomed

[willbordoni]

yes i'm using the vicidial manager.conf file
and yes i'm starting asterisk with the perl script.

i found out a problem : i registered my phone extension as "100" instead of "cc100" in the admin/phone section !

now when i log in, i get the call from vicidial ! but i always have the "noone in your session" problem.

here is the console output:

Code: Select all

  == Parsing '/etc/asterisk/manager.conf': Found
  == Manager 'sendcron' logged on from 127.0.0.1
       > Channel SIP/cc100-01459450 was answered.
Mar 25 15:20:28 WARNING[25981]: pbx.c:1720 pbx_extension_helper: No application 'Meetme' for extension (default, 8600051, 1)
  == Spawn extension (default, 8600051, 1) exited non-zero on 'SIP/cc100-01459450'
    -- Executing DeadAGI("SIP/cc100-01459450", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------)") in new stack
Mar 25 15:20:28 WARNING[25981]: res_agi.c:210 launch_netscript: Connect to 'agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------)' failed: Connection refused
  == Spawn extension (default, h, 1) exited non-zero on 'SIP/cc100-01459450'

I'm going to look out for the 2 WARNINGS now.

[kwatson]

check your meetme.conf to make sure that the conference 8600051 is defined.

Also not sure if it has any bearing you are using the loopback address for your server I.P you might be better using that eth0 address (again not sure if that will matter if the dialplan is correct for 127.0.0.1)

Thanks

Kenny

[mflorell]

no application meetme usually means no zaptel timer.

[ykhan]

What is the poutput of "lsmod" in Linux console? Do you see anything with Zaptel or ztdummy? If not then recompile Asterisk including zaptel.

[willbordoni]

meetme conferences are correctly defined in meetme.conf
I'v already checked the loopback adress, it's OK for its references in the mysql DB and for the dialplan.

In fact i have no zaptel nor ztdummy modules loaded, and asterisk was compiled without zaptel flag (i'm using gentoo)
I tried to recompile asterisk with zaptel but an error occured :

Code: Select all

* Could not find a usable .config in the kernel source directory.
 * Please ensure that /usr/src/linux points to a configured set of Linux sources.
 * If you are using KBUILD_OUTPUT, please set the environment var so that
 * it points to the necessary object directory so that it might find .config.
 *
 * ERROR: net-misc/zaptel-1.2.26-r1 failed.
 * Call stack:
 *                 ebuild.sh, line   48:  Called pkg_setup
 *   zaptel-1.2.26-r1.ebuild, line   82:  Called linux-mod_pkg_setup
 *          linux-mod.eclass, line  579:  Called require_configured_kernel
 *         linux-info.eclass, line  192:  Called die
 * The specific snippet of code:
 *              die "Kernel not configured; no .config found in ${KV_OUT_DIR}"
 *  The die message:
 *   Kernel not configured; no .config found in /usr/src/linux

I think i need a kernel support for zaptel, or something like that...
Do i need to load both zaptel and ztdummy modules? Do Zaptel module requires some specific hardware ?

EDIT :

ok i fixed this easily, but now i get this error message :

Code: Select all

 * ERROR: net-misc/zaptel-1.2.26-r1 failed.
 * Call stack:
 *               ebuild.sh, line   48:  Called src_compile
 *             environment, line 3672:  Called die
 * The specific snippet of code:
 *       make KVERS=${KV_FULL} KSRC=${KV_DIR} ARCH=$(tc-arch-kernel) || die;
 *  The die message:
 *   (no error message)

What's wrong with this :/

[mflorell]

I'm not that familiar with Gentoo packages, we always zompile zaptel from source. Have you tried that?

[willbordoni]

I compiled zaptel from source and it works (had a hard time with recompiliing my kernel :s)

Code: Select all

# lsmod
Module                  Size  Used by
ztdummy                 3968  0
zaptel                193928  1 ztdummy

But it didn't solve my problem... I always have this "noone is in your session" F.... message

Asterisk console output when i try to login as an agent :

Code: Select all

== Manager 'sendcron' logged on from 127.0.0.1
       > Channel SIP/cc100-0081f330 was answered.
Mar 26 14:26:11 WARNING[9230]: pbx.c:1720 pbx_extension_helper: No application 'Meetme' for extension (default, 8600051, 1)
  == Spawn extension (default, 8600051, 1) exited non-zero on 'SIP/cc100-0081f330'
    -- Executing DeadAGI("SIP/cc100-0081f330", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------)") in new stack
Mar 26 14:26:11 WARNING[9230]: res_agi.c:210 launch_netscript: Connect to 'agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------)' failed: Connection refused
  == Spawn extension (default, h, 1) exited non-zero on 'SIP/cc100-0081f330'
  == Manager 'sendcron' logged on from 127.0.0.1
Mar 26 14:27:21 NOTICE[9445]: chan_local.c:526 local_alloc: No such extension/context @default creating local channel
Mar 26 14:27:21 NOTICE[9445]: channel.c:2514 __ast_request_and_dial: Unable to request channel Local/@default



I'm working on it .. all your help is appreciated !

[mflorell]

you have to recompile asterisk AFTER you have compiled zaptel, so just recompile Asterisk and you should be fine.

[willbordoni]

I compiled asterisk, but the problem is still occuring.

I don't understand why is says "No application 'Meetme' for extension (default, 8600051, 1)"

Here is my manager.conf file:

Code: Select all

[general]
enabled = yes
port = 5038
bindaddr = 0.0.0.0

[cron]
secret = 1234
read = system,call,log,verbose,command,agent,user
write = system,call,log,verbose,command,agent,user

[updatecron]
secret = 1234
read = command
write = command

[listencron]
secret = 1234
read = system,call,log,verbose,command,agent,user
write = command

[sendcron]
secret = 1234
read = command
write = system,call,log,verbose,command,agent,user


in extensions.conf i have this line :
exten => 8600001,1,Meetme,8600001|q

in meetme.conf, i have this line :
conf => 8600051

any ideas ?

EDIT:

ok it seems that the module app_meetme.so is not loaded.
when i try to load it i have an error :

Code: Select all

xxxxxxx*CLI> load app_meetme.so
Unable to load module app_meetme.so
Mar 26 16:41:17 WARNING[2791]: loader.c:326 __load_resource: /usr/lib64/asterisk/modules/app_meetme.so: undefined symbol: ast_null_frame

sos

[mflorell]

Not sure what to suggest now, maybe posting on the asterisk forums or asterisk users list or IRC.

[mcargile]

I hate to say this but you are probably going to need to wipe and rebuild your system. I have installed Vicidial successfully on Gentoo in the past and I have never used their Asterisk packages. The reason being is that they are based on 1.2 and do not include the patches that are required for Vicidial to work properly. Mainly the cli concise delimiter patch. You should wipe the system and when it comes time to install Asterisk and Zaptel download the versions from Digiums website and patch Asterisk with the various patches.

[willbordoni]

OK.
I started all over from scratch install...
Now I don't have the connection refused for AGI script but still having this Meetme error :

Code: Select all

== Manager 'sendcron' logged on from 127.0.0.1
       > Channel SIP/cc100-0224f370 was answered.
Mar 27 12:25:47 WARNING[29322]: pbx.c:1720 pbx_extension_helper: No application 'Meetme' for extension (default, 8600051, 1)
  == Spawn extension (default, 8600051, 1) exited non-zero on 'SIP/cc100-0224f370'
    -- Executing DeadAGI("SIP/cc100-0224f370", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------)") in new stack
    -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------) completed, returning 0

I really don't know what to do ! The Meetme is configured in extensions.conf and meetme.conf.

[mflorell]

That is not a meetme.conf issue, that is a problem with when you compiled Asterisk.

[willbordoni]

thanks a lot, but I can't compile asterisk with app_meetme

with make menuselect I can't enable app_meetme (it's marked XXX)
I don't have app_meetme.so in /usr/lib/asterisk/modules

I don't understand why i can't install this meetme module, zaptel is installed with ztdummy module.

[williamconley]

I have installed vicidial on a lot of gentoo boxes and written custom solutions for weird little problems like this that have come up. One of the easiest solutions has been to download and install asterisk/zaptel/libpri right on top of the emerged asterisk. emerge does a lot of the work for you and the asterisk installs nicely (most of the time!)

Here's a sample extracted from our scripts regarding just the asterisk portion of the install: (beware, it's just an extraction! use your head while you're doing it)

Code: Select all

emerge asterisk asterisk-addons asterisk-sounds
mkdir /usr/src/asterisk
cd /usr/src/asterisk
wget http://ftp.digium.com/pub/asterisk/releases/asterisk-1.2.26.tar.gz
wget http://ftp.digium.com/pub/asterisk/releases/asterisk-1.2.26.tar.gz.sha1
wget http://ftp.digium.com/pub/zaptel/releases/zaptel-1.2.22.tar.gz
wget http://ftp.digium.com/pub/zaptel/releases/zaptel-1.2.22.tar.gz.sha1
wget http://ftp.digium.com/pub/libpri/releases/libpri-1.2.4.tar.gz
wget http://ftp.digium.com/pub/libpri/releases/libpri-1.2.4.tar.gz.sha1
sha1sum -c *.sha1 > checksums
cd /usr/src/asterisk
gunzip asterisk-1.2.26.tar.gz
tar xvf asterisk-1.2.26.tar
gunzip zaptel-1.2.22.tar.gz
tar xvf zaptel-1.2.22.tar
gunzip libpri-1.2.4.tar.gz
tar xvf libpri-1.2.4.tar
nano checksums
perl -pi -e 's/define MAXMSG 100/define MAXMSG 500/' /usr/src/asterisk/asterisk-1.2.26/apps/app_voicemail.c
cd /usr/src/asterisk/zaptel-1.2.22
make clean && make && make install && make config
cd ../libpri-1.2.4
make clean && make && make install
cd ../asterisk-1.2.26
make clean && make && make install
make samples
echo 'modprobe zaptel' >> /etc/conf.d/local.start
echo 'modprobe ztdummy' >> /etc/conf.d/local.start
echo 'sleep 5' >> /etc/conf.d/local.start
echo 'chmod 0666 /dev/zap/*' >> /etc/conf.d/local.start
nano /etc/asterisk/asterisk.conf
         ****************
         *change astrundir to '/var/run/asterisk'
nano /etc/udev/rules.d/zaptel.rules
verify that this file contains all these lines:
***********************
# zaptel devices with ownership/permissions for running as non-root
KERNEL=="zapctl", NAME="zap/ctl", OWNER="asterisk", GROUP="asterisk", MODE="0660"
KERNEL=="zaptimer", NAME="zap/timer", OWNER="asterisk", GROUP="asterisk", MODE="0660"
KERNEL=="zapchannel", NAME="zap/channel", OWNER="asterisk", GROUP="asterisk", MODE="0660"
KERNEL=="zappseudo", NAME="zap/pseudo", OWNER="asterisk", GROUP="asterisk", MODE="0660"
KERNEL=="zaptranscode", NAME="zap/transcode", OWNER="asterisk", GROUP="asterisk", MODE="0660"
KERNEL=="zap[0-9]*", NAME="zap/%n", OWNER="asterisk", GROUP="asterisk", MODE="0660"


Hope it helps.

[mcargile]

Which version of zaptel did you install?

Which version of asterisk did you install?

Did you compile zaptel before asterisk?

[willbordoni]

Yes I compiled zaptel before asterisk

I tried asterisk version 1.2.24, 1.4.23 and 1.6.0

I'm using zaptel version 1.4.12 My kernel version is 2.6.27, what's yours williamconley please?

the "undefined symbol: ast_null_frame" error made me feel it's a kernel configuration probleme. But i configured it properly (as the scratch install said) I'm going to try with a 2.6.17 kernel... and after on a slackware (it's the only differences between my install and the scratch one)

[williamconley]

if you send me your .config from your kernel build, i can look at it to see if there's anything i find missing.

The versions are in the file names of the download.

wget http://ftp.digium.com/pub/asterisk/rele ... .26.tar.gz
wget http://ftp.digium.com/pub/zaptel/releas ... .22.tar.gz
wget http://ftp.digium.com/pub/libpri/releas ... 2.4.tar.gz

[willbordoni]

yes, i was just asking for your kernel version.
Anyway i'm gonna work on it this weekend and ask again for help on monday if things are not better.
thanks a lot for your advices.

[williamconley]

! Oh, sorry, read it wrong. Actually I have many kernel versions ... here's one: 2.6.27-gentoo-r8

[cristian]

I love gentoo.

If you promise to fully document the solution for the forums I will give you a bit of my time. I emailed you my phone number, be ready with ssh if you can. If its an odd time you can email me, I get them quick.

[cristian]

Alternately, put

>=sys-kernel/gentoo-sources-2.6.26

in /etc/portage/package.mask and run

emerge --sync
emerge gentoo-sources

This gives you the last kernel version, known good. Check your /usr/src/linux symlink and rebuild.

williamconley, didn't we discuss this earlier this month?

[mcargile]

if you are using zaptel 1.4.12 then you should be using asterisk 1.4.21.2 because unless you know some tricks (making symlinks to the correct headers and using the right version of zaptel) asterisk 1.2 will not compile with zaptel 1.4. Also Asterisk 1.6 does not support zaptel at all. It uses DAHDI which Vicidial does not support at all.

[cristian]

Per mcargile zaptel 1.2 will not be pathed to kernels higher than 2.6.26, hence my version freeze.

[willbordoni]

In facts i gave up with asterisk and vicidial.

I installed vicidialnow on a new server and it works perfectly without doing anything !

thanks all.

[qadir52786]

dear all plz help me with this error

i had install vicidialnow .i also get call frm srv but after 1 min i got this error
Noone is in your session: 8600051 plz plz help me


thankx

[williamconley]

i had install vicidialnow .i also get call frm srv but after 1 min i got this error
Noone is in your session: 8600051

please read the stickies (the listings at the top of the forum that say "sticky").

Also be sure you read the manager's manual (available free for download at eflo.net and you can buy the full version if you want to but it is not necessary to "get it working").

Then when you post, start your own "topic" and include your configuration in your post.

We don't know enough about your problem, but we'd love to help when you give us enough information!

[qadir52786]

i am sorry sir for not posting details

[root@vici asterisk]# screen -r
There are several suitable screens on:
2610.ASTupdate (Detached)
2616.ASTlisten (Detached)
2625.ASTVDadapt (Detached)
2613.ASTsend (Detached)
2628.ASTfastlog (Detached)
2469.astshell (Detached)
2474.asterisk (Detached)
2622.ASTVDremote (Detached)
2619.ASTVDauto (Detached)
Type "screen [-d] -r [pid.]tty.host" to resume one of them.

[qadir52786]

i am sorry i cant post other details be its showing me error that your account must have 5 days old

[williamconley]

you can still post your hardware/software configuration. without that we have no idea what you have. also what other software is in your box besides vicidial (if any). also how did you install? vicidial scratch install? what OS? or vicidialNOW? or viciBox?

and you can DEFINITELY read the manager's manual and do the exercises starting at the beginning.

have you successfully logged in as an agent? (this requires going to the agent screen, putting in a Phone and password, then an Agent and password, then choosing a campaign, then hitting submit, then having your phone ring, then answering the phone and then hearing "you are the only person in this conference).

If you have NOT successfully logged in as an agent, you need to go back to the manager's manual and go through the exercises one line at a time and tell us when you get your first error. That's what it's for, to get you past that first login.