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remote office

mario
Posts: 51
 Member Since:
2007-03-13
Submitted by mario on Sun, 04/01/2007 - 6:43am.

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 !!

‹ cannot repark calls or transfer after park pickup Changing CALLERID for Outbound calls based on IVR Choices? ›

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mario
Posts: 51
 Member Since:
2007-03-13
ooopppss > up one more
Sun, 04/01/2007 - 8:04am

ooopppss > up one more time ....

thank


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cosmicwombat
Posts: 1173
 Member Since:
2006-05-31
Is there draytek forum?
Sun, 04/01/2007 - 10:56am

This looks more related to the 2800vg than to trixbox. Is there a forum for Draytek? Cause I'd try that too.

Robert Keller - FtOCC
Isle of Lummi

--

Robert Keller - Chief Technologist at large
The VoIP Experience
Open Telephony Training Seminar


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mario
Posts: 51
 Member Since:
2007-03-13
draytek
Sun, 04/01/2007 - 2:51pm

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 ......


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cosmicwombat
Posts: 1173
 Member Since:
2006-05-31
What does the extension config look like?
Sun, 04/01/2007 - 7:37pm

I still think this has to do with NATing two networks on one device. Unless I have that wrong?

Can you try an IAX softphone from the remote?

Robert Keller - FtOCC
Isle of Lummi

--

Robert Keller - Chief Technologist at large
The VoIP Experience
Open Telephony Training Seminar


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mario
Posts: 51
 Member Since:
2007-03-13
IAX
Mon, 04/02/2007 - 3:39pm

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


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mario
Posts: 51
 Member Since:
2007-03-13
up for an help thank a lot
Wed, 04/04/2007 - 10:01am

up for an help thank a lot


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cosmicwombat
Posts: 1173
 Member Since:
2006-05-31
The IAX softphone worked?
Wed, 04/04/2007 - 11:28am

If so, then you need to look at your port forwarding rules. Can you put them here? Sanitized for public IP's of course....

Robert Keller - FtOCC
Isle of Lummi

--

Robert Keller - Chief Technologist at large
The VoIP Experience
Open Telephony Training Seminar


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mario
Posts: 51
 Member Since:
2007-03-13
ok then her the config of
Wed, 04/04/2007 - 4:03pm

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


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mario
Posts: 51
 Member Since:
2007-03-13
up one more time
Thu, 04/05/2007 - 6:11am

up one more time


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cosmicwombat
Posts: 1173
 Member Since:
2006-05-31
I am not sure, but....
Thu, 04/05/2007 - 7:00am

the fxs port still has to use IP to get to the trixbox. I would open those ports on the remote just to test.

Did you say if an IAX2 softphone did work?

Robert Keller - FtOCC
Isle of Lummi

--

Robert Keller - Chief Technologist at large
The VoIP Experience
Open Telephony Training Seminar


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mario
Posts: 51
 Member Since:
2007-03-13
yes it works i tried with a
Thu, 04/05/2007 - 8:40am

yes it works i tried with a softphone iax2. Now i'm going to try to open port on the remote side an then i'll repost the result.......


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mario
Posts: 51
 Member Since:
2007-03-13
ok opened port 5060 tcp and
Thu, 04/05/2007 - 8:46am

ok opened port 5060 tcp and 10001-20000 udp but never changed
i'm very frustrated


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mario
Posts: 51
 Member Since:
2007-03-13
here are the lines of the
Thu, 04/05/2007 - 8:55am

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.


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mario
Posts: 51
 Member Since:
2007-03-13
sorry up again !!!
Thu, 04/05/2007 - 1:45pm

sorry up again !!!


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mario
Posts: 51
 Member Since:
2007-03-13
RESOLVED !!! REMOTE OFFICE -> VPN and NAT
Thu, 04/05/2007 - 3:28pm

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 .....


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cosmicwombat
Posts: 1173
 Member Since:
2006-05-31
Please elaborate...
Thu, 04/05/2007 - 4:51pm

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

--

Robert Keller - Chief Technologist at large
The VoIP Experience
Open Telephony Training Seminar


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mario
Posts: 51
 Member Since:
2007-03-13
externip=xxxxxx.dyndns.org lo
Fri, 04/06/2007 - 8:48am

externip=xxxxxx.dyndns.org
localnet=10.97.4.160/255.255.255.224 ;Asterisk Side
localnet=192.168.222.192/255.255.255.240 ;Remote side
nat=yes

LIKE THIS !!!!

Thank again Robert !!!!


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