Noone is in your session 8000051

[sberg]

Hi all

when i login in /server:ip/agc/vicidial.php for every less than 2mins i get this message in blue screen..

Noone is in your session 8000051
Go back
Call Agent again

I am getting this in all the users i have logged in..
any solution for this...

Regards,
Sberg

[gerski]

can you post your asterisk CLI

astguiclient version?

did the phone ring once it connected? and did you hear "your the only person in this conference"

[sberg]

HI
ya when i have loaded the lead i got my phone xlite phone rining..
i got the demo...

asterisk - 1.2.17
astguiclient - 2.0.3
OS- RHEL 4

[gerski]

can you post asterisk cli when you login..

[sberg]

Hi

astereisk CLI:

Connected to Asterisk 1.2.17 currently running on localhost (pid = 4937)
Verbosity is at least 3
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'listencron' logged on from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'updatecron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1

Regards,
Sberg
== Manager 'sendcron' logged off from 127.0.0.1

[gerski]

do you have 8600051 in your meetme.conf? also in extensions.conf

[sberg]

hi,

No there is no entry in the extension.conf or meetme.conf with the id 8600051
i think that is the id automatically generated for that session.

if any enteries has to be made plse do tell me

Regards,
Sberg

[gerski]

no, you must have both 8600051, 8600052, etc... conference in your meetme as well in your extensions.conf.. also you need it to input in your database... please follow scratch install..

[sberg]

Hi,

i have added both the 8600051 and 8600052 in database, meetme.conf and extension.conf as in scratch_installation, but it is coming up same as befor..for every 10-15 seconds.


Regards,
Sberg

[Op3r]

dont make this sticky.

people who doesnt follow the scratch_install have problems like this.

[Michael_N]

dont make this sticky.

people who doesnt follow the scratch_install have problems like this.

Yes thats why i make it sticky.. so they dont have to start a new tread
But can read the answer in this tread..

[sberg]

Thanks for the reply Opr3,

if u have any idea regarding the problem...plse tell me.
May b u dont have the solution to his..plse dont comment..


ThankQ

[gerski]

can you post your asterisk cli again when you login?

[aster1]

looking at your asterisk cli it seems your verbosity level is set to 0 . Type set verbose 99 in asterisk cli and then login from vicidial web interface and get the cli output .

[badboy2]

looking at your asterisk cli it seems your verbosity level is set to 0 . Type set verbose 99 in asterisk cli and then login from vicidial web interface and get the cli output .

I have the same problem. Wait i need two posts so i can reply. Ill post everything.

[badboy2]

Noone is in your session 8000051
Go back
Call Agent again


Im having the same problem. the message above keeps on popping up every 5-15 seconds on all the agents logged in. Whether they are on a call or not it pops up. 8600051 and 8600052 in database, meetme.conf and extension.conf as in scratch_installation. It just suddenly happen 2days go without any warning. What could possibly cause this? It is somehow irritating. Calls do not hangup when I click "Hangup Costumer". The call stays and no Dead AGI on the CLI. Dead AGI appears when I hangup the dial pad or when the client hangs up. It does not hangs up when I click on the hangup costumer button. Any idea on why it does not hang up properly? Below is my extensions.conf. We only do outbound calling. It happens on both Manual and Auto dial. It does not hang u when I press the Hangup Costumer button.

It never happened before. The phone rings when agent logs in and I hear "your the only person in this conference". I am using the packages below:

OS: debian
Protocol: SIP (through a 3rd party SIP provider)
vicidial version : 2.0.3
asterisk version: Asterisk 1.2.18

pbx2:~# screen -r
There are several suitable screens on:
2749.ASTlisten (Detached)
2747.ASTsend (Detached)
2757.ASTfastlog (Detached)
2751.ASTVDauto (Detached)
2755.ASTVDadapt (Detached)
2753.ASTVDremote (Detached)

[badboy2]

Connected to Asterisk 1.2.18 currently running on pbx2 (pid = 2655)
-- Remote UNIX connection
Verbosity is at least 3
pbx2*CLI> set verbose 99
Verbosity was 3 and is now 99
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
Jun 3 08:44:38 WARNING[3245]: chan_zap.c:7799 zt_request: Unknown option '-' in '10-1'
> Channel Zap/10-1 was answered.
-- Executing MeetMe("Zap/10-1", "8600052") in new stack
== Parsing '/etc/asterisk/meetme.conf': Found
-- Created MeetMe conference 1023 for conference '8600052'
-- Playing 'conf-onlyperson' (language 'en')
== Manager 'sendcron' logged off from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'updatecron' logged on from 127.0.0.1
== Manager 'updatecron' logged off from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1

[badboy2]

crontab entries:
pbx2:~# crontab -l
# m h dom mon dow command
1,6,11,16,21,26,31,36,41,46,51,56 * * * 1-6 /usr/share/astguiclient/AST_CRON_mix_recordings_MP3.pl
* * * * * /usr/share/astguiclient/ADMIN_keepalive_ALL.pl
* * * * * /usr/share/astguiclient/AST_manager_kill_hung_congested.pl
* * * * * /usr/share/astguiclient/AST_vm_update.pl
* * * * * /usr/share/astguiclient/AST_conf_update.pl
11 * * * * /usr/share/astguiclient/AST_flush_DBqueue.pl -q
33 * * * * /usr/share/astguiclient/AST_cleanup_agent_log.pl
* * * * * /usr/share/astguiclient/AST_VDhopper.pl -q
1 1 * * * /usr/share/astguiclient/ADMIN_adjust_GMTnow_on_leads.pl --debug --postal-code-gmt
2 1 * * * /usr/share/astguiclient/AST_reset_mysql_vars.pl
3 1 * * * /usr/share/astguiclient/AST_DB_optimize.pl
30 * * * * /usr/local/bin/ntpdate -u 18.145.0.30 2>/dev/null 1>&2
2 0 * * 0 /usr/share/astguiclient/AST_agent_week.pl
* * * * * perl /usr/share/astguiclient/AST_CLEAR_VICIDIAL_BLOCK.pl
30 * * * * ntpdate -u 18.145.0.30 2>/dev/null 1>&2

T1 card as timer: TE120p

[badboy2]

extensions.conf:
[default]
exten => _9XXXXXXXXXXX,1,AGI(****//127.0.0.1:4577/call_log)
exten => _9XXXXXXXXXXX,2,Dial(SIP/${EXTEN:1}@acl,65,tTo)
exten => _9XXXXXXXXXXX,3,Hangup

....


; FastAGI for VICIDIAL/astGUIclient call logging
exten => h,1,DeadAGI(****//127.0.0.1:4577/call_log)
exten => h,2,DeadAGI(****//127.0.0.1:4577/VD_hangup--HVcauses--PRI-----NODEBUG-----${HANGUPCAUSE}-----${DIALSTATUS}-----${DIALEDTIME}-----${ANSWEREDTIME}))


any ideas why it does not hangup the call? It suddenly appeared 2 days ago. Thanks!

[badboy2]

Hi! while browsing the forum I found the answer! Many thanks to ytcracker! My asterisk_live_channel got corrupted.. anyway everything works now. Cheers!


PostPosted: Wed May 02, 2007 4:24 pm Post subject: no one is in your session after power outage/power off .... by ytcracker

[bobby]

Hi All,

I have the same problem - When i login with agc/vicidial.php, after about 15 to 30 seconds the message comes "Noone is in your session: 8600051"

Also when I login I don't get the ring on my soft phone - Xlite.
I followed the instructions on scratch installation almost to the letter. I don't have any ZAP or IAX trunks - using only SIP. So left out those sections.

This is my first experience at setting up the VICIDIAL. My h/w is an Intel Dual Core 2 with SATA Harddisk, 2GB RAM. My OS is Ubuntu. Asterisk version 1.2.18 and AstGuiClient version 2.0.3. Codecs used are g729 and gsm. Only using VOIP.

I have configured another AsteriskNow Server for inbound calls - and thats working fine for last 2 months. I use XLite Soft Phones for that. Other than this experience, I am a newbie. My first time here with VICIDIAL. I have gone through the agent and manager manuals, gone through all the previous postings in this forum and seached the bugs in the VICIDIAL Issue Tracker. Some of you claim that following the scratch install should work - so i have gone back and examined it carefully thrice - still no luck yet. All my extensions are in the default context. I even changed the Meetme module to the app_conference hoping that it was to do with zt_dummy and zaptel not being present. But that didn't work.

I have only two screens when I do screen -r. Not sure why - but sometimes I see more but usually its just these 2 screens:
4434.ASTfastlog (Detached)
4432.ASTVDadapt (Detached)

I also find that I need to start manually start the Asterisk Console for my XLite phone to register with it. This was not mentioned anywhere in the installation doc. But I am doing it. I am able to make a SIP call out with my phone directly but not sure if VICIDIAL is able to do the same. I have enabled the cron jobs as mentioned in the scratch install and also can see them load at boot time but Asterisk Console is required for my XLite Soft Phone Client to be registered. Is this how it works? or am i missing something else here? But even with this running I don't get the ring when the agc/vicidial.php is logged in. Instead I get the error "Noone...."

I have enabled debugging in the logger.conf and am watching the asterisk messages in the log but don't find anything specific to this issue.

Would appreciate some urgent help here. Struggling with this for more than 3 days. I hope i have given sufficient info, if not please ask. Thanks.

[ramindia]

Hi

the suggestion here is

check all config files again
sip.conf
extension.conf
manager.conf
meetme.conf

and check again is there any steps missing from scratch_install.
regarding the database tables. main thing conference rooms.

make sure your crond running. and ntp sync.

ram

[bobby]

The problem was with the way NAT was configured. My server has 2 cards - I was using the external ip for all config (including the database). But my SIP clients were using the internal ips on my LAN. The moment I changed the ip of the server using the internal ip - it worked.

So looks like a silly problem - but I assumed the external ip was being discussed because of the examples given in the scratch_install doc.


Now I have another problem. Seems to be more asterisk related. I get the error message when vicidial dials out saying "an error has occurred, contact tech support".

I notice that asterisk console messages reveal a context called Local/xxxxx@default

The console messages are
Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
-- Executing Conference("Local/8600051@default-634b,2", "8600051") in new stack
Jul 15 23:09:45 NOTICE[8948]: member.c:415 member_exec: [ $Revision: 1.9 $ ] begin processing member thread, channel => Local/8600051@default-634b,2
> Channel Local/8600051@default-634b,1 was answered.
-- Executing Dial("Local/8600051@default-634b,1", "SIP/inphonex-out/11234567890|55|o") in new stack
Jul 15 23:09:45 NOTICE[8948]: member.c:742 create_member: attempting to parse passed params, stringp => 8600051
Jul 15 23:09:45 NOTICE[8948]: member.c:793 create_member: parsed data params, id => 8600051, flags => , priority => 0, vad_prob_start => 0.050000, vad_prob_continue => 0.020000
Jul 15 23:09:45 NOTICE[8948]: member.c:1077 create_member: created member, type => S, priority => 0, readformat => 64
-- Called inphonex-out/16093570153
Jul 15 23:09:45 NOTICE[8948]: member.c:451 member_exec: CHANNEL INFO, CHANNEL => Local/8600051@default-634b,2, DNID => (null), CALLER_ID => 1234567890, ANI => 1234567890
Jul 15 23:09:45 NOTICE[8948]: member.c:454 member_exec: CHANNEL CODECS, CHANNEL => Local/8600051@default-634b,2, NATIVE => 64, READ => 64, WRITE => 64
Jul 15 23:09:45 NOTICE[8948]: conference.c:504 start_conference: attempting to find requested conference
Jul 15 23:09:45 NOTICE[8948]: conference.c:563 find_conf: found conference in conflist, name => 8600051
Jul 15 23:09:45 NOTICE[8948]: conference.c:796 add_member: member added to conference, name => 8600051
Jul 15 23:09:45 NOTICE[8948]: member.c:514 member_exec: begin member event loop, channel => Local/8600051@default-634b,2
Jul 15 23:09:45 NOTICE[8948]: member.c:532 member_exec: Conference Members: 2
Jul 15 23:09:45 NOTICE[8948]: member.c:538 member_exec: Quiet debug 0 - 0
Jul 15 23:09:45 NOTICE[8948]: member.c:688 basic_play_sound: playing conference message enter
Jul 15 23:09:45 NOTICE[8948]: member.c:688 basic_play_sound: playing conference message enter
Jul 15 23:09:45 NOTICE[8948]: member.c:351 process_outgoing: unanticipated delivery time, delivery_diff => -889788173, delivery.tv_usec => 523500
-- SIP/inphonex-out-007e6510 answered Local/8600051@default-634b,1
== Manager 'sendcron' logged off from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1

== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Spawn extension (default, 11234567890, 1) exited non-zero on 'Local/8600051@default-634b,1'
-- Executing DeadAGI("Local/8600051@default-634b,1", "agi://127.0.0.1:4577/call_log") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing DeadAGI("Local/8600051@default-634b,1", "agi://127.0.0.1:4577/VD_hangup--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----20-----18)") in new stack
-- AGI Script agi://127.0.0.1:4577/VD_hangup--HVcause ... -20-----18) completed, returning 0
Jul 15 23:10:05 NOTICE[8948]: member.c:612 member_exec: unable to read from channel, channel => Local/8600051@default-634b,2
Jul 15 23:10:05 NOTICE[8948]: member.c:635 member_exec: end member event loop, time_entered => 1184521185
== Spawn extension (default, 8600051, 1) exited non-zero on 'Local/8600051@default-634b,2'
-- Executing DeadAGI("Local/8600051@default-634b,2", "agi://127.0.0.1:4577/call_log") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing DeadAGI("Local/8600051@default-634b,2", "agi://127.0.0.1:4577/VD_hangup--HVcauses--PRI-----NODEBUG-----0---------------)") in new stack
-- AGI Script agi://127.0.0.1:4577/VD_hangup--HVcause ... ----------) completed, returning 0
Jul 15 23:10:05 ERROR[8714]: conference.c:216 conference_exec: skipping leave message on Local/8600051@default-634b,2
Jul 15 23:10:05 NOTICE[8714]: member.c:688 basic_play_sound: playing conference message leave
Jul 15 23:10:05 NOTICE[8714]: conference.c:234 conference_exec: found member slated for removal, channel => Local/8600051@default-634b,2
Jul 15 23:10:05 NOTICE[8714]: conference.c:844 remove_member: member accounting, channel => Local/8600051@default-634b,2, te => 1184521185, fi => 891, fid => 53, fo => 1003, fod => 0, tt => 20
Jul 15 23:10:05 NOTICE[8714]: member.c:1104 delete_member: freeing member flags, name => Local/8600051@default-634b,2
Jul 15 23:10:05 NOTICE[8714]: member.c:1116 delete_member: deleting member input frames, name => Local/8600051@default-634b,2
Jul 15 23:10:05 NOTICE[8714]: member.c:1130 delete_member: deleting member output frames, name => Local/8600051@default-634b,2
Jul 15 23:10:05 NOTICE[8714]: member.c:1156 delete_member: freeing member translator paths, name => Local/8600051@default-634b,2
Jul 15 23:10:05 NOTICE[8714]: member.c:1168 delete_member: freeing member channel name, name => Local/8600051@default-634b,2
Jul 15 23:10:05 NOTICE[8714]: member.c:1175 delete_member: freeing member
Jul 15 23:10:05 NOTICE[8714]: conference.c:869 remove_member: removed member from conference, name => 8600051, remaining => 1
== Manager 'sendcron' logged off from 127.0.0.1


My extensions has a context called local that only has these lines:
[local]
ignorepat => 9
include => default

The default context has all the details as per the scratch_install doc.
I am able to make a normal SIP call out without using vicidial. Any ideas why this error when vicidial dials out?

thanks in advance,

[mflorell]

Any reason you are using app_conference instead of meetme?

meetme is much more stable.

[bobby]

Hi

I managed to get it working. The problem was elsewhere - to do with the service provider registrations and nothing to do with the Local contexts.

But yes the concern on the performance exists.
The only reason i switched to app_conference is because of the zt_dummy - I don't have any Zap lines, only using VOIP, so i read in some articles that Meetme would not perform well without a real Zap card in the system. Is this true? The app_conference documentation said this was faster and not a affected by lack of Zap card.

Appreciate your advice. Currently I am trying to run 10 channels with a single system with mysql etc., all on the same system. Have SATA hard drive with 7.2K rpm and have 2 GB RAM with Intel Dual Core 2.

What do you advice for performance?
thanks everyone for all help. These forums and others have been a great help.

[gerski]

meetme uses timing of zt_dummy or x100p even if you don't have zap lines. its true that ztdummy is not reliable but you can get cheap X100P in the internet.

app_conference is not very much stable.. i suggest using meetme conference and getting cheap X100P for timing.

i can do about 20 users, all recordings in one server and it works very well with your server specs.