[asterisk-users] Calls Disconnecting out of the blue .. [Renamed]

[Tarek Sawah]

Greetings again List.
I'm facing a strange case with one of the productive Asterisk servers..
i have 3 providers sending traffic to the call center where agents pickup the calls.
calls come into the server>> Queue>> Agents

Last October .. an undersea cable got disconnected placing Egypt and the countries in the region offline.. when internet came back .. the call center located in Egypt had no SIP protocol working.. and we shifted to IAX.. 26 days later SIP started to work again .. but since then calls started to disconnect out of the blue.. we get calls that may last for 45 minutes.. and end normaly .. and we get calls that ring and disconnect the moment the agent picks up
been facing a problem with my client as they use the Flash Operator Panel to monitor the call flow through the server and the regualr setup Queue>> Local users  won't work for them as the Flash operator flash offline static agents as online so the client won't know who is on and who is off.. and it's impossible to teach the agents to Login and Logoff the Queue.. so the only solution is the following..

Caller>> Queue>> FindMeFollowMe Extension>> Local SIP extensions

this way .. my client is able to monitor the calls and things won't get complicated.. (this is the setup we have been using for 6 months before the problem with the internet occures)
since the internet problem and calls are getting disconnected .. out of the blue.. nothing has changed.. and to make sure things are going well .. we moved the server to a Hosting company in California with 10 mb/s connection speed.. (Same Setup that was working well)
and still calls get disconnected.. 
after a lot of problems with the client .. i asked them to change the ISP (my prime suspect was the internet)
and finaly they managed to change the ISP .. but the problem is still there.. 

my server informations are the following

Asterisk 1.4.22-3
Uname -a:  Linux 2.6.18-92.1.18.el5

sip.conf
;;Agent Sample from Sip.conf
[3000]
type=friend
secret=3000
qualify=yes
port=5060
disallow=all
allow=g729
nat=yes
host=dynamic
dtmfmode=rfc2833
dial=SIP/3000
context=from-internal
canreinvite=no
call-limit=1
busy-limit=1

;;Provider's Sample from Sip.conf
[50011]
type=peer
qualify=yes
port=5060
pickupgroup=
nat=no
host=XXX.YYY.ZZZ.NNN
disallow=all
allaw=alaw
allaw=ulaw
allow=g729
dial=SIP/50011
context=from-internal
canreinvite=no
deny=0.0.0.0/0.0.0.0
permit=XXX.YYY.ZZZ.NNN/255.255.255.255

#########
extensions.conf 

;;the provider sends calls to Virtual DIDs (Extensions) in my system which is 8000

exten => 8000,1,GotoIfTime(07:00-16:00|sat-fri|1-31|jan-dec?ext-queues,*8000,1)
exten => 8000,n,Answer
exten => 8000,n,Queue(8000,t,,,10)
exten => 8000,n,Dial(IAX2/6005:6005 at BackupServer/100001) ;; sends the call to a backup server.
exten => *8000,1,Answer
exten => *8000,n,Dial(IAX2/6005:6005 at BackUpServer/100001)

########

the Providers strictly send calls with codec G.729
my agents get best voice quality with G.711u 

I need your advice .. am i missing anything in this setup?? it used to work .. and it STILL works on another hosted server with Agents located in Morocco.. with a different version of Asterisk 1.4.20-1 and better hold time for the calls.. 


-- AHD Tarek Sawah

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