vicidial.org

by **ruben23** » Wed Sep 17, 2008 11:05 am

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..

by **ruben23** » Thu Sep 18, 2008 11:17 am

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

by **mflorell** » Thu Sep 18, 2008 1:33 pm

Did you by chance change your IP address?

by **ruben23** » Thu Sep 18, 2008 1:40 pm

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...

by **mflorell** » Thu Sep 18, 2008 2:00 pm

What are the server_ips listed in your vicidial_conferences table?

by **ruben23** » Thu Sep 18, 2008 3:17 pm

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..

by **ruben23** » Fri Sep 19, 2008 10:26 am

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

by **mflorell** » Fri Sep 19, 2008 12:18 pm

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

by **ruben23** » Fri Sep 19, 2008 1:04 pm

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..

by **mflorell** » Fri Sep 19, 2008 6:44 pm

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

by **ruben23** » Mon Sep 22, 2008 3:10 am

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...

by **ruben23** » Mon Sep 22, 2008 2:28 pm

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...
:arrow:

by **mflorell** » Tue Sep 23, 2008 1:09 am

Is meetme loaded?

'show application meetme'

by **ruben23** » Tue Sep 23, 2008 2:17 pm

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....
:shock:

by **mflorell** » Tue Sep 23, 2008 5:19 pm

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

by **ruben23** » Wed Sep 24, 2008 9:32 am

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

by **ruben23** » Wed Sep 24, 2008 11:09 am

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.

by **mflorell** » Thu Sep 25, 2008 10:17 am

version and build of vicidial.php?

by **ruben23** » Thu Sep 25, 2008 12:16 pm

im using vicidialnow 1.1

Astguiclient release 2.0.4.1rc3

Phpsysinfo 2.5.4

Thats what i found....

by **mflorell** » Thu Sep 25, 2008 12:28 pm

Upgrade to at least 2.0.4rc4

by **ruben23** » Thu Sep 25, 2008 2:35 pm

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..

by **mflorell** » Thu Sep 25, 2008 5:02 pm

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

by **ruben23** » Fri Sep 26, 2008 9:56 am

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..

by **mflorell** » Fri Sep 26, 2008 7:59 pm

yes.

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

by **ruben23** » Mon Sep 29, 2008 11:24 am

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..

by **ruben23** » Mon Sep 29, 2008 1:02 pm

hi its ok i already found it..

by **ruben23** » Mon Sep 29, 2008 2:07 pm

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...

by **mflorell** » Mon Sep 29, 2008 2:44 pm

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

by **ruben23** » Mon Sep 29, 2008 3:43 pm

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..

by **ruben23** » Mon Sep 29, 2008 4:12 pm

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.

by **ruben23** » Mon Sep 29, 2008 5:13 pm

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..

by **ruben23** » Tue Sep 30, 2008 7:47 am

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

by **ruben23** » Tue Sep 30, 2008 8:15 am

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

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