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multi callcenter(vicidial) with one asterisk in same machine

PostPosted: Thu May 24, 2007 8:24 am
by smth
hi,
who can give me a hand?
what I am doing is that I am managing to run multi callcenters with multi web portal and only one asterisk in same machinne. that meant to service a couple of company respectively,like each company will have a completely virtual callcenter.

First of all.only one vicidial running at my machine has got worked well.
then I run the second one. still seem not be ok.

my system FC4
kernel:2.6.11
asterisk:1.2.14
vicidial:2.0.2

Code: Select all
There are several suitable screens on:
        2725.ASTupdate  (Detached)
        2468.ASTVDadapt (Detached)
        23039.ASTVDauto (Detached)
        2731.ASTlisten  (Detached)
        23055.ASTfastlog        (Detached)
        2472.ASTfastlog (Detached)
        23050.ASTVDadapt        (Detached)
        2734.ASTVDauto  (Detached)
        23047.ASTVDremote       (Detached)
        2507.asterisk   (Detached)
        25790.ASTupdate (Detached)
        2737.ASTVDremote        (Detached)
        2728.ASTsend    (Detached)


as if second listener and sender dosn't work yet. If I run listener manualy got message like:
Code: Select all
 input buffer: 82     lines: 0     partial: 0
|Event: PeerStatus
Privilege: system,all
Peer: SIP/qta1001
PeerStatus: Registered

|
input lines: 3|848600|50|
+++++++++++++++++++++++++++++++sending keepalive transmit line 848600|2007-05-23 11:14:04|2007-05-23 11:14:04|
input buffer: 82     lines: 1     partial: 0
|

Event: PeerStatus
Privilege: system,all
Peer: SIP/qta1001
PeerStatus: Registered

|
input buffer: 82     lines: 0     partial: 0
|Event: PeerStatus
Privilege: system,all
Peer: SIP/qta1001
PeerStatus: Registered

|
input buffer: 82     lines: 0     partial: 0
|Event: PeerStatus
Privilege: system,all
Peer: SIP/qta1001
PeerStatus: Registered

|
input buffer: 82     lines: 0     partial: 0
|Event: PeerStatus
Privilege: system,all
Peer: SIP/qta1001
PeerStatus: Registered


When agent logged in the web portal of second vicidial.the agent phone dosn't get ring.As for the first vicidial basically get working ok simultaneously!

By the way I have set different context in extension.conf for each callcenter.

Dose everyone can help me figure out? I really really appreciate.

PostPosted: Thu May 24, 2007 8:43 am
by smth
Even though I have started the listerner manually, the sender can not run.When try to make it work manually,got message as follow:
Code: Select all
checking to see if listener is dead |||
checking to see if listener is dead ||1|
checking to see if listener is dead ||1|
checking to see if listener is dead ||1|
checking to see if listener is dead ||1|
checking to see if listener is dead ||1|
checking to see if listener is dead ||1|
checking to see if listener is dead ||1|
checking to see if listener is dead ||1|
checking to see if listener is dead ||1|
checking to see if listener is dead ||1|
loop counter: |863882|

PostPosted: Fri May 25, 2007 12:59 pm
by smth
got ok basically after restart asterisk

Code: Select all
There are several suitable screens on:
        22854.ASTupdate (Detached)
        2604.ASTfastlog (Detached)
        2602.ASTVDadapt (Detached)
        22863.ASTVDauto (Detached)
        22853.ASTVDremote       (Detached)
        22850.ASTVDauto (Detached)
        22857.ASTsend   (Detached)
        22847.ASTlisten (Detached)
        22844.ASTsend   (Detached)
        22866.ASTVDremote       (Detached)
        22841.ASTupdate (Detached)
        2611.ASTfastlog (Detached)
        2609.ASTVDadapt (Detached)