I found a possible solution to deal with sip connection problem in a network environment with Draytek router 2920 or 2930 and IP01. I understand there is always problems with this setup for sip client to connect to the IP01 despite one has open the port and done port redirection.
I figure our that in the draytek router, use port re-direction to direct an unused port (i.e. not UDP 5060) e.g. 15060 to the IP01 IP:5060. In this case, the IP01 does not need to change any port binding and allows internal client to connect via standard 5060 UDP port. For outside LAN or internet connection, the client has to use a different port (usually in a form of xxx.xxx.xxx.xxx:15060 to connect.
I try this with great success.
See if you need this as a workaround for draytek router.
The port forwarding issue using Dratek router has already been discussed with bubblestar few days ago. With this feature, the internal port 5060 does not need to be changed. We only change the port number from outside and transfrom it into another one such as 5060 for the use of internal Asterisk server.
Mine is Vigor 2930n. It should not have much difference but I failed to do it for dual servers in same lan environement. Hope you can make it and enlighten me the way to get it work.
2930VN is a different issue. It has VOIP as well. I don't know why it needs to change to different port but I believe that the router has something set up to default all traffic of sip to its ports first.
You may recall others using draytek has similar problem in sip connection.
Anyway it helps to fix some problems and at the same time enhance security i.e. not using standard 5060 port for SIP.
Basically, the internal/external using the default UDP5060 is no problem. If I use single server, I can still connect it from outside world with port 15060. However, when I set up another (2nd) Asterisk server in the same lan, I cannot use other ports like 25060 to re-direct to the internal 5060 on the second server. That's my existing situation.
Anyway, Draytek Vigor 2930n is a good router for me, so far. Maybe, I have to spend some time to get familiar with it further.
Additional tricks for the set up
1. since IP01 is not a very powerful CPU, when you update the file via GUI, you need to wait for a while to allow the write to complete
2. Switchfin GUI though is better is still limited in handling the conf file. Please bear this in mind and try to use SSH or FTP within the lan for file editing. I.e. encourage to do it via APL not GUI
3. if after changes, not able to sip login, try to reboot the IP01 and your sip client. Very often, it caches the previous information.
4. always save copies of working conf and it allows you to restore the working conf when you encounter problem.
FYI, my sip.conf. is corrupt but working, it is a result of previous GUI write with duplicate settings. I try to trim down and remove duplicates, then it does not work. Don'r know why!!
I have been puzzling about the question that when I set up new trunks using APL, there is no way for me to see the trunk in Outgoing Calling Rules and Incoming Calling Rules in the GUI interface. In this case, how can I create DP for the newly established trunks?
OK. If I set up all IN/OUT calling rules with APL as well. I'm still unable to see the trunk status from GUI. Is that normal? Are you just living with that and dial out by memorizing dialing prefix without the need to refer to GUI?
Another question is that I discover there is no crontab under the /persistent/etc/ directory. Shall I need to create one myself to do crob job?
發表於 2011-2-6 21:04
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