problem with VOIP call time

[georgem]

hi

asterisk 1.2.27
vici 2.0.4 rc2
VICIDIALNOW 1.1

i am facing some problem with my vici system. the agent disposes the call here within a minute.. but the CDR of teh voip gateway says the call lasted for over 1 hr or 2 hrs some times. i am billed for the whole of that time but i see the dialer disposing it at 2 minutes.. i pulled out master.csv file in /var/log/asterisk/cdr-csv and see the same 2 minutes there.

when i reported this to the VOIP provider, they told tht there is some improper config in ur SIP conf. here goes my sip conf.

[general]
context=default ; Default context for incoming calls
port=5060 ; UDP Port to bind to (SIP standard port is 5060)
bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)
allow=g729
allow=ulaw
allow=alaw
allow=gsm
canreinvite=no
rtpholdtimeout=300
rtptimeout=60




register=>xxxx:xxxx@:xxxxx5060


[SIPtrunk]
type=friend
username=xxxx
secret=xxxx
host=xxxx
fromdomain=xxxxx
fromuser=xxxxx
context=default
insecure=very
disallow=all
;allow=ulaw
allow=g729
canreinvite=no


[cc100]
disallow=all
allow=ulaw
allow=alaw
type=friend
username=cc100
secret=test
host=dynamic
dtmfmode=rfc2833
;defaultip=192.168.1.249
qualify=1000
mailbox=100

[cc101]
disallow=all
allow=ulaw
allow=alaw
type=friend
username=cc101
secret=test
host=dynamic
dtmfmode=inband
defaultip=192.168.1.249
qualify=1000
mailbox=101

[cc102]
disallow=all
allow=ulaw
allow=alaw
type=friend
username=cc102
secret=test
host=dynamic
dtmfmode=inband
defaultip=192.168.1.249
qualify=1000
mailbox=102

[cc103]
disallow=all
allow=ulaw
allow=alaw
type=friend
username=cc103
secret=test
host=dynamic
dtmfmode=inband
defaultip=192.168.1.249
qualify=1000
mailbox=103

[cc104]
disallow=all
allow=ulaw
allow=alaw
type=friend
username=cc104
secret=test
host=dynamic
dtmfmode=inband
defaultip=192.168.1.249
qualify=1000
mailbox=104

[cc105]
disallow=all
allow=ulaw
allow=alaw
type=friend
username=cc105
secret=test
host=dynamic
dtmfmode=inband
defaultip=192.168.1.249
qualify=1000
mailbox=105


they told me that the BYE signal(for SIP) is released from the dialer after 1 or 2 hrs for those calls.
i think that ther is some problem with RTP timeout.
i tried by best to sort it but failed. kindly help me out here.


regards
george m

[gmcust3]

I faced the same problem and believe it or not I lost more than 250$ before I considering upgrading to 2.0.5 and NOW its COOL !!1

[georgem]

hi

any tricks to solve the current system?
it is a production server and cant do a new installation as of now.

regards

george m

[gmcust3]

Simply go for Upgrade.

[georgem]

hi

i guess this is an asterisk bug!!
any other tricks to solve it.. i am afraid to go for an upgrade being a production server.

any RTP timout etc need to be adjusted !!
kindly suggest.

regards
george m

[gmcust3]

Let me tell u the secret , But don't share !!

Upgrade WORKS PERFECTLY !!