vicidial.org

When an Agent clicks on Transfer Conference and the Park Call on Agent Screen, it hungs up the soft phone.

9 agents logged in on all servers
System Load Average: 0.48

When normally agent presses Park Call, then all good... !!

Feb 6 04:20:16 NOTICE[2996]: chan_sip.c:11742 do_monitor: Disconnecting call 'SIP/cc130-08725da0' for lack of RTP activity in 61 seconds


Feb 6 04:22:17 WARNING[2996]: chan_sip.c:9983 handle_response_register: Got 200 OK on REGISTER that isn't a register
Feb 6 04:22:17 WARNING[2996]: chan_sip.c:9983 handle_response_register: Got 200 OK on REGISTER that isn't a register
Feb 6 04:22:17 WARNING[2996]: chan_sip.c:9983 handle_response_register: Got 200 OK on REGISTER that isn't a register

Any help ?

what is your asterisk version ?
do you have a probleme in your network.
i see here that SIP/cc130 is laged in rtp ,do you use rtp .
i recommend the use of dns primary and secondory in your resolv.conf then register your softphone .
verify your firewall.

"i see here that SIP/cc130 is laged in rtp ,do you use rtp .
i recommend the use of dns primary and secondory in your resolv.conf then register your softphone . "


How to change it from rtp to dns ?

real time protocol "rtp" is not the same thing as dns primary and secondy.
the rtp it used to run voice over ip and the rtp use too the the two protocol tcp and udp so when we speak about dns so here i mean your internet connection dns for it .
each connection internet have a dns primary and secondy here if you know them from your operator.
or try to use dmz.
but i'am sure you have a probleme in your network.
between server and your agent pc.

Sorry but I am confused on what step should I try to resolve this ?

I have

nameserver 208.67.222.222

in my resolve.conf file.

ok this is your dns primary maybe try to add your dns secondry here two
add
nameserver then the dns secondry
verify your sip show peers and the time of ping for each agent it there a diffirence between them
what version of asterisk do you have ?
verity if you hav open the rtp port 10000 and 2000 and 5060

I have added

202.56.215.54
202.56.215.55

I have

nameserver 202.56.215.54
nameserver 202.56.215.55

and Removed the earlier one.

How to check open the rtp port 10000 and 2000 and 5060?

5060 is open as Voice is going.10000 is open as I can access webmin

I am using asterisk Asterisk 1.2.30.2

first i recomend the upgrade of asterisk
then to check if your port is opening over the ip of server
try for exemple ntop tools in linux
then push "p" boutton to check port .

I use VNOW 1.3 and it has 1.2 version.

Like when we transfer one customer to IVR..soemtime customer ask question that time we kep the IVR on hold .and clear customer question

if IVR line kept on hold for more then 20 seconds it get droped automatically !!!