vicidial.org

hi to all...please help with this=====> i already installed vicidialnow and configure it, im using eyebeam as a SIP phone my user ID can register to the asterisk server but the problem is when i try to login in the vicidial webgui inputing the agent user ID and password i got error:

Sorry, there are no available sessions===>

anyone have idea with this?....is it with my extension.conf and SIP.conf?

here are my configuration:

sip.conf

[general]
port = 5060
bindaddr = 0.0.0.0
context = default
; register SIP account on remote machine if using SIP trunks
; register => testSIPtrunk:test@10.10.10.16:5060
;
; setup account for SIP trunking:
; [SIPtrunk]
; disallow=all
; allow=ulaw
; allow=alaw
; type=friend
; username=testSIPtrunk
; secret=test
; host=10.10.10.16
; dtmfmode=inband
; qualify=1000

[VoIP]
disallow=all
allow=g729
type=friend
host=208.64.253.203
dtmfmode=rfc2833
qualify=1000

[1001]
disallow=all
allow=g729
type=friend
username=1001
secret=1001
host=dynamic
dtmfmode=rfc2833
qualify=1000

[1002]
disallow=all
allow=g729
type=friend
username=1002
secret=1002
host=dynamic
dtmfmode=rfc2833
qualify=1000

and my extension.conf

[general]
static=yes
writeprotect=no

[globals]
;CONSOLE=Console/dsp ; Console interface for demo
;TRUNK=Zap/g1 ; Trunk interface
;TRUNKX=Zap/g2 ; 2nd trunk interface
;TRUNKIAX=IAX2/ASTtest1:test@10.10.10.16:4569 ; IAX trunk interface
;TRUNKIAX1=IAX2/ASTtest1:test@10.10.10.16:4569 ; IAX trunk interface
;TRUNKBINFONE=IAX2/1112223333:PASSWORD@iax.binfone.com ; IAX trunk interface
;SIPtrunk=SIP/1234:PASSWORD@sip.provider.net ; SIP trunk
;TRUNKloop = IAX2/ASTloop:test@127.0.0.1:40569 ; used for blind monitoring
;TRUNKblind = IAX2/ASTblind:test@127.0.0.1:41569 ; used for testing


[default]

; BE SURE TO CHANGE THIS LINE FOR YOUR IP ADDRESS!
exten => _192*168*001*002*.,1,Goto(default,${EXTEN:16},1)
exten => _8600XXX*.,1,AGI(agi-VDADfixCXFER.agi)
exten => _78600XXX*.,1,AGI(agi-VDADfixCXFER.agi)
; Local blind monitoring
exten => _08600XXX,1,Dial(${TRUNKblind}/6${EXTEN:1},55,To)


; Example phone extensions

; Extension 2000 Sipura/Linksys ATA line 1
exten => 2000,1,Dial(sip/spa2000,30,to) ; Ring, 30 secs max
exten => 2000,2,Voicemail,u2000 ; Send to voicemail...
; Extension 2001 Sipura/Linksys ATA line 2
exten => 2001,1,Dial(sip/spa2001,30,to) ; Ring, 30 secs max
exten => 2001,2,Voicemail,u2001 ; Send to voicemail...
; Extension 2102 rings Grandstream phone
exten => 2102,1,Dial(sip/gs102,30,to) ; Ring, 30 secs max
exten => 2102,2,Voicemail,u2102 ; Send to voicemail...
; Extension 401 rings the firefly softphone
exten => 401,1,Dial((IAX2/firefly01@firefly01/s||t)
exten => 401,2,Hangup

; extensions for other SIP and IAX call center phones
; cc100-cc150 SIP Phones
exten => _1[0-5]X,1,Dial(sip/cc${EXTEN},20,to)
; cc300-cc350 IAX Phones
exten => _3[0-5]X,1,Dial(IAX2/cc${EXTEN},20,to)

; extensions if using a T1 channelbank
exten => _19XX,1,Dial(Zap/${EXTEN:2},30,o)
exten => _19XX,2,Hangup


Hope anyone can suggest....thanks a lot for your time..

please anyone who have an idea.....really need help on this...

Did you by chance change your IP address?

i never change my IP address..its 192.168.2.8

i can view the vicidial php page..but cannot login coz it a no session error...

What are the server_ips listed in your vicidial_conferences table?

hi...i never done inputing my IP address to the vicidial_conference. this was new to me- maybe this is the problem....i found a post to input my IP add to the vicidial mysql database but had encounter another problem...

when i try to input this:

insert into vicidial_conferences values('8600051','10.10.10.15','');
insert into vicidial_conferences values('8600052','10.10.10.15','');

then hit return...

Error no databse selected...but i already login to mysql..

did i missed something..

hi- anyone who have an idea...on this

I think you need to run the ADMIN_update_server_ip.pl script and use 10.10.10.15 as the old server_ip.

I have know idea on how im going to do that...can you give some little hint or guide just to start that process....it would be deeply appreciated..

on the command line type this:
/usr/share/astguiclient/ADMIN_update_server_ip.pl

then when it prompts you for the old server IP address enter this:
10.10.10.15

ok i got it...hope it wpuld solve this no session error on my vicidialnow....thanks for your time...ill update you if what will happen...

hi to you i already input the script on the astguiclient...same problem sorry no session....& on my logs a found this error:


-- Registered SIP '1001' at 192.168.2.6 port 44622 expires 3600
-- Saved useragent "eyeBeam release 1100z stamp 47737" for peer 1001
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
> Channel SIP/1001-0a043138 was answered.
== Manager 'sendcron' logged off from 127.0.0.1
== Starting SIP/1001-0a043138 at default,,1 failed so falling back to exten 's'
== Starting SIP/1001-0a043138 at default,s,1 still failed so falling back to context 'default'
-- Sent into invalid extension 's' in context 'default' on SIP/1001-0a043138
-- Executing Playback("SIP/1001-0a043138", "invalid") in new stack
-- Playing 'invalid' (language

i ahve a welcome ring then when i answer it...it says that im sorry that is not a valid extensions...

thats what i got...

Is meetme loaded?

'show application meetme'

hi i already solve my no session error.. it with my conference extension ip add= not the same with my asterisk server.....

i test a call now and its already working...

but during my dialing process there is STATUS prompt that me on screen

like The page http://_._._. says;

the time interval on ringing are display..
i cannot proceed without clicking ok...

how can i remove that...it really hard if there a prompt on dialing no control on call....

I am not familiar with this error, could you post more information on it?

hi is it allowed here to post for a screenshot of my error...i mean its a graphical screenshot to let you see the error

ok perhaps i just have to explain it detailed....here are the steps:

==> first @ hand i run my eyebeam softphone together with my firefox browser ( w/ vicidialnow login page) as my SIP sofphone register to my asterisk server

==>now the agent login to the vici page by its user name & pass, and answer the welcome ring..ready to dial

==> the agent dial now on the vici page and stablish a call but a small message page appears on the vici page: saying that http://my asterisk ip add says: then the STATUS plus the number dialed then number of seconds are displayed but the problem is it will not proceed when the OK button be press.

==> the problem about it is when its answering machine it will continue counting the seconds even the phone is ended and the page cannot be close..it just says ringing 2 seconds-i click ok then ringing 3 sec==>4 secs==> and so on.


i hope this sounds clear. how im going to remove that mesage prompt.

version and build of vicidial.php?

im using vicidialnow 1.1

Astguiclient release 2.0.4.1rc3

Phpsysinfo 2.5.4

Thats what i found....

Upgrade to at least 2.0.4rc4

ok...so i need to re-install it all from scratch again? just for that new release

vicidial 2.0.4.1 rc4

is there no other workaround for this..im just about to deploy this to production...ive spend a month just to build it out from this forum and it means a lot-all the effort being done..

thats the only problem now so far....and its good to go..without that error..

A reinstall should take about 5 minutes, you are only installing the vicidial code, not the entire operating system.

Thanks for the idea... so i just need to overwrite the old version with the new one....all the configuration i made will be not affected....if i may ask how im goin to start with the upgrades..just a start idea form you..

yes.

You should read the UPGRADE document included in the newer releases for more information.

hi..., i view the attach docs on the new astguiclient release i downloaded for my upgrades....the only thing i found are how to install the vici from scratch and the upgrade _2.0.4sql.....is that the document what you min for upgrading the astguiclient.. its all about the sql database..

So im thinking that i just install the new release astguiclient just as like the installation on scratch and perform the upgrade_2.0.4sql

Am i on the right track..

hi its ok i already found it..

Also_ can i ask few things about upgrading astguiclient..i have the directory /usr/src/astguiclient/. so i need to download the astguiclient new release to that same directory...or it would be ok if a make a new one for the mew release...would that be a start for my upgrade...

To upgrade or install you can stick the source anywhere on your system.

i already installed and run the upgrade...test a call...during login to my SIP phone_registered on asterisk server-ok then when i enter the agent user login and pass...no welcome call happen which indicates your connected...after that no activity...i cant dial including manual..

and also i see that my SIP softphones are not logging on the asterisk...as i view it on the CLI command asterisk -r & reload..no indication of login, thats why i cannot hear a welcome call form asterisk....

What may had happen to my asterisk box after i run the upgrade..i just follow the procedure...og upgrading astguiclient 2.0.4.1r3 to 2.0.4.1rc4.

i have the error here..when im logging to vicidial...and no welcome call for asterisk.

== Using TOS bits 0
-- Registered SIP '1001' at 192.168.2.6 port 9796 expires 3600
-- Saved useragent "eyeBeam release 1100z stamp 47737" for peer 1001
== 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
-- Got SIP response 488 "Not Acceptable Here" back from 192.168.2.6
> Channel SIP/1001-09610ac8 was never answered.
Sep 29 18:10:42 WARNING[11217]: cdr.c:566 ast_cdr_disposition: Cause not handled
== Manager 'sendcron' logged off from 127.0.0.1

what could had happen..

Please any help from the admin people....yhanks

hi admin===> through persistent ive found the problem..it with the codec of my SIP softphones..im using speex which is not supported with my voip provider....thanks for all the support...specially to Mr. mflorell