Dropped Calls after 20-ish Seconds
Submitted by farazm on Sat, 04/26/2008 - 12:21am.
Hi,
I am having trouble using my linksys voip phones remotely (through the internet) with my trixbox server. I have forwarded all of the relevant ports on the router to the server, and the phone registers fine. Every time I make a call everything goes as per normal, and then the call just drops after about 20 seconds. If I am watching it in the trixbox panel, both the ZAP and extension just hangup. On the phone it seems to think it is in an active call, but it is obviously just silence.
I have tried a fresh install of trixbox, as well as upgraded the routers firmware. If it was a problem with the router blocking ports, wouldn't the call just be blocked in the first place? Has anyone else experienced anything similar to this?
I have tried from a few different phone's (Including a softphone) and the same thing happens with each one when it's from outside the office.
Help? Anyone?
Cheers,
Faraz
I had this problem too. It
Mon, 04/28/2008 - 7:28am
I had this problem too. It seems the phones do not respond within a timely manner to the SIP server. My solution: install a SIP proxy. I use the Adtran TotalAccess 908e, though to save money you could probably get an older computer and use the Endian firewall SIP proxy to handle this for you, or you could buy one of their appliances.
I'm not sure if the Endian community version has this built in, but you can give it a try.
Also, I think the m0n0wall project has a sip proxy of some kind built into their software, and DD-WRT has a VoIP firmware, though I'm not quite sure if that is a mini asterisk install or if it does sip proxying.
NOTE: I'm only listing software that is easily configurable from a web interface. If you're feeling savvy, try OpenSER ( I dare you :-P )
The problem it seems doesn't
Mon, 04/28/2008 - 7:57am
The problem it seems doesn't lie with asterisk, but with the phones. In most cases you can't have more than one SIP phone on a remote network that is behind a NAT because the router can't properly forward the SIP packets to the right phone because it simply does not know which one to forward those packets to. the phone will not respond to something it never gets, therefore asterisk gives up on waiting for a response from the phone and disconnects the call.
A sip proxy will store the SIP registration information and IP address of each phone and when it receives an incoming sip message destined for a certain sip device, it will know exactly what to do with it.
Also, if anyone sees a mistake in the previous statement, please correct me.
Hi again,
Okay, so it is not
Mon, 04/28/2008 - 10:44am
Hi again,
Okay, so it is not the asterisk server that is the issue. The NAT issue does make some sense though, but I am still a bit confused.
If I am correct in understanding what has been said, only one phone should be able to connect through the router? However on several occasions had 3 or 4 phones connected remotely with the same issue. They all just 'time out' after 20 seconds or so. If it was specifically a NAT issue wouldn't the other phones be unable to connect/interfere?
The router I have is: http://www.linksys.com/servlet/Satellite?c=L_Product_C2&childpage...
Which is quite high up by Linksys standards, and claims to be designed for VoIP applications.
The VoIP server is set up as a DMZ if this helps, so all requests/ports are forwarded to the server.
Can someone help me clarify this?
Cheers,
Faraz
I've found on all Linksys
Tue, 04/29/2008 - 5:47am
I've found on all Linksys kit (albeit only on the internal network) that if the "gateway" address in Network settings is incorrect (normally this is your routers address) this exact problem will appear.
if this is happening intermittently however, this may indicate a problem with your internet link.
Solution Found
Wed, 04/30/2008 - 7:46am
Just an update to anyone experiencing similar problems.
Turns out the problem was not in fact with the router/firewall.
Asterisk was, as previously suggested, dropping the calls because of a lack of communication. There is some good information on http://nerdvittles.com/index.php?p=194 about registering phones from an external network...e.g internet.
I have copied their tips below:
Getting Rid of One-Way Audio. There are some settings you'll need to add to /etc/asterisk/sip_custom.conf if you want to have reliable, two-way communications with Asterisk: nano -w /etc/astersk/sip_custom.conf. The entries depend upon whether your Internet connection has a fixed IP address or a DHCP address issued by your provider. In the latter case, you also need to configure your router to support Dynamic DNS (DDNS) using a service such as dyndns.org. If you have a fixed IP address, then enter settings like the following using your actual public IP address and your private IP subnet:
externip=180.12.12.12
localnet=192.168.1.0/255.255.255.0 (NOTE: The first 3 octets need to match your private IP addresses!)
If you have a public address that changes and you're using DDNS, then the settings would look something like the following:
externhost=myserver.dyndns.org
localnet=192.168.0.0/255.255.255.0 (NOTE: The first 3 octets need to match your private IP addresses!)
Once you've made your entries, save the file: Ctrl-X, Y, then Enter. Reload Asterisk: amportal restart. If you assigned a permanent IP address, reboot your server: shutdown -r now.
Worked fine for me, now I can have mutliple phones working remotely with no issues.
Thanks for the support
Faraz
Member Since:
2006-11-17