Traveling Man -- Can't call remote extension...

[dandy_don]

Fala,

OK -- Looks like there are at lease three of us with the same problem...

Tonight I put the server into the router's DMZ and it made absolutely NO difference whatsoever. I've since put it back behind the firewall.

Tomorrow I'm going to pull the router out and connect it directly to the cable modem and test using the cell phone and the laptop softphone.

So far, I'm less doubtful that it is a router issue. Can't rule the router out until I completely remove it from the network. I'll post results tomorrow.

Thanks,
Don

 

[dandy_don]

Well I shutdown my server, and connected it directly to the cable modem, and directly onto the internet and fired up the server again.

NOTHING changed. I can still dial into the server from traveling man and reach my cell phone, and I can still call into the server from my cell phone, but I can't call out to the softphone.

Can't be the router. Not sure if TimeWarner Roadrunner turbo blocks ports, but I doubt it. I'm pretty sure it is something in the server setup...

The server is back behind the firewall again.

Anyone have any ideas?

Thanks,
Don

 

[dandy_don]

Is there another forum/website that would be more appropriate for posting these questions (And for getting some support)?

I had thought that this would be the best venue for getting help, but I'm clearly getting nowhere.

Thanks,
Don

 

[wardmundy]

What you're really asking is for someone to troubleshoot your Linux box at a remote site. As I understand it, you have a softphone running on that box?? Something on the box is obviously blocking the inbound connections. It could be IPtables or some other firewall at the site. How we would know??

My best advice is START SIMPLE. Get a SIP phone that is not connected through your remote Liinux server and register it directly with the host machine at the primary site. Does it stay registered? If so, the problem is definitely something on your Linux box. If not, then I suspect it's a problem with either your router or provider. As I mentioned, there are some hosting providers that are not VoIP-friendly. We can't fix that. Sorry.

 

[dandy_don]

Ward,

Thanks. Yes -- I've done EXACTLY as you suggest. I can take the softphone inside the firewall and register, stay connected, make and receive multiple calls -- all flawlessly.

I can then take that same softphone and register remotely via traveling man, and make phone calls perfectly, all day long. It remains registered until I unregister it. One time I kept the laptop running for over 6 hours all the while being registered to the server, and still making phone calls into the server. Audio works fine both ways. I just can never dial out to the softphone when it is remote.

I've eliminated the router completely by connecting the pbx directly to the cable modem.

So -- I don't think it is the firewall because I removed it completely and the problems persisted. This is a "stock" install of piaf. I posted the details in one of the first posts.

I agree -- The problem is in the Linux box. This is what I'm asking help with.

I have 4 GXP2000 phones behind the firewall with the server. They all work perfectly. Skype into and out of the server works perfectly (Thanks!).

I've tried to eliminate everything. At this point it must either be something in the server, or else something to do with Time Warner roadrunner service. Certainly there must be others using traveling man with Time Warner cable modems?!?

Any help is appreciated.

Thanks,
Don

 

[dandy_don]

Ward,

Just one more question regarding possible port blocking...

As I understand it, when the extension is configured to "qualify=yes" that the server will query the existence of the phone. The server will continue to query every minute. The server uses this query to determine whether the extension is "reachable." When that extension is deemed to be unreachable, even though it may be registered, the server will not dial out to the extension.

Please correct me if I have this wrong...

What ports does the server use to qualify the (remote) phone. I think this would be the logical place for me to look since the remote phone works fine otherwise. If the port that is needed for qualifying the phone is blocked, that would cause this behavior (I think).

All of the other ports "seem" to be forwarded/routed/un-blocked as appropriate as the other signaling, and audio appear fine.

If you can correct me on the port used to qualify the extension, this might be the problem/solution.

Thanks,
Don

 

[wardmundy]

This tells me you've got network problems, not Linux problems...

[2011-08-10 23:11:32] NOTICE[6001] chan_sip.c: Peer '501' is now Reachable. (759ms / 2000ms)
[2011-08-10 23:18:01] VERBOSE[5645] asterisk.c: -- Remote UNIX connection
[2011-08-10 23:18:01] VERBOSE[21587] asterisk.c: -- Remote UNIX connection disconnected
[2011-08-10 23:18:42] NOTICE[6001] chan_sip.c: Peer '501' is now Lagged. (3385ms / 2000ms)
[2011-08-10 23:18:56] NOTICE[6001] chan_sip.c: Peer '501' is now UNREACHABLE! Last qualify: 3385

 

[dandy_don]

Ward,

I think I have resolved those timing problems... It was due to the way I had the softphones incorrectly configured.

 

[dandy_don]

Please,

Does anyone know what port(s) the server uses to qualify the (remote) phone?

Since everything else is working fine (except placing calls to the external phone) I thought this would be the logical place for me to look.



Thanks.

 

[wardmundy]

UDP 5060

 

[dandy_don]

OK -- Now I've really got this system screwed up!

I re-read this post again {freepbx.org/support/documentation/howtos/howto-setup-a-remote-sip-
extension}

I had followed previous posts regarding configuration of the sip_nat.conf, sip_custom.conf, etc. However, whenever I put either my real world IP address or the real world host into these files, it broke outbound skype as well as outbound calling, so I opted to leave the real-world IP stuff out of these files...

Tonight I followed the post above and went into FreePBX and chose "Asterisk SIP Settings" and entered my dynamic IP as well as the local network address. Then restarted everything.

This absolutely KILLED all outbound calling as well as outbound Skype. No matter what I try, I can't get outbound Skype to work again, and outbound calling from the hardwired SIP phones is flakey at best.

Why does setting "Asterisk SIP Settings" as described in the posting (above) kill what was working? How do I get back to a working system?

Thanks,
Don

 

[wardmundy]

Seriously...

Have you considered an Obi device? They're less than $50.

 

[dandy_don]

I had a working system -- albeit travling man didn't work correctly. Now I just need to get back to a working system again. Frankly suggesting the Obi device does absolutely nothing to help getting back to a working system.



Don

 

[dandy_don]

OK -- This is why this forum is so frustrating! Can someone please help enlighten me as to why setting the asterisk sip settings as suggested in the forums would BREAK the working system and more importantly, how do I get back to where I as before?!? I know that I'm an a noob here...

Don

 

[NetKatz]

Yes, Don.. your error is an Id-10t error most likely cause by operator error..

 

[rossiv]

NetKatz This is a three year old thread. Please try to refrain from waking the dead.