remote office
ok writing from italy and sorry for my english!!
here is the problem....
ok upgrade from trixbox 1.2.3 to 2.0.
I have i little office with 6 ip phones (GXP2000) in lan 10.x.x.x and a firewall router draytek 2800vg that keep up a VPN with a remote office where another lan (192.x.x.x) has two phones one x-lite and another analogic phone connected to a fxs port of the other draytek 2800vg. Well before upgrading all works great now when i call from the remote office to an internal of my office the line goes down after 10/15 seconds without any reasons. If i call from my office to the remote all goes well. Ok i don't have modify nothing into the routers configuration and the VPN works great also for other services.
For the test i did it seems there is a sort of timeout when i call from remote office but i did not have found any config file that show this flag.
Could anyone help me.
Thank in advance !!
thank Robert i thought this too but the problem there is only when the caller is the remote party. In fact whe the caller is the local party the call works fine. I have found this line on debug log for the local extension during the dropped call :
channel.c : Didn't get frame from channel : SIP/210 (local phone)
and then asterisk hangup.
It seems that asterisk search if the local phone is up and then not receiving answer drop the call every time after 20 seconds.
Help please ......
Tried it works perhaps you are right but... why with trixbox 1.2.3 all works fine ?
in what point nat now cause this problem. I have debugged a little bit and i found that asterisk says that he don't get frame from remote sip phone. What mean that he don't get frame ? port forward problem ? if yes what port have i to open and also need to open port over VPN ? I'm very confuse ... please help me with every idea .. thank
ok then her the config of the connection lan to lan VPN:
asterisk server side:
lan 10.97.4.160/255.255.255.224
server asterisk 10.97.4.177
local pc and GXP2000 in the subnet
router address 10.97.4.172
open ports and redirections
port 5060 TCP -> 10.97.4.177
port 10001-20000 UDP -> 10.97.4.177
remote lan:
lan 192.168.222.192/255.255.255.240
router address 192.168.222.193
phone connected to the fxs port of the router
no port redirection
encrypted VPN between the two router is always up
the config is the same that was with tb 1.2.3 and it worked.
thank Robert
here are the lines of the full log when the line drop:
Apr 5 17:50:48 WARNING[5627] chan_sip.c: Maximum retries exceeded on transmission cVO-30145@192.168.222.193 for seqno 3 (Critical Response)
Apr 5 17:50:48 WARNING[5627] chan_sip.c: Hanging up call cVO-30145@192.168.222.193 - no reply to our critical packet.
Apr 5 17:50:48 DEBUG[11638] channel.c: Didn't get a frame from channel: SIP/208-08ee5458
Apr 5 17:50:48 DEBUG[11638] channel.c: Bridge stops bridging channels SIP/208-08ee5458 and SIP/207-08f1ef48
Apr 5 17:50:48 DEBUG[11638] chan_sip.c: update_call_counter(207) - decrement call limit counter
Apr 5 17:50:48 DEBUG[11638] app_dial.c: Exiting with DIALSTATUS=ANSWER.
OK !
Perhaps it works !!!!
i saw i have two private networks one 10.97.4.x and the other 192.168.222.x
well i MUST insert in sip_nat.conf two localnet keywords
infact if i don't insert the second subnet, Asterisk, i think, tried to nat the ip address and
nothing worked. Now it take the ip of the remote office like a local ip and all goes fine.
Thank a lot!!
bye .....
So, your sip_nat.conf looks like:
A)
externhost=asterisk.xxx.com
localnet=10.97.4.0/255.255.255.0, 192.168.222.0/255.255.255.0
nat=yes
--or--
B)
externhost=asterisk.xxx.com
localnet=10.97.4.0/255.255.255.0
localnet=192.168.222.0/255.255.255.0
nat=yes
Robert Keller - FtOCC
Isle of Lummi


Member Since:
2007-03-13