TUTORIAL U Need Travelin' Man

dandy_don

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Ward,

I already have this set just as you suggested. Not sure why this doesn't work.

I can call in (make phones ring), but no audio either way and I can't call out to the remote extension...
 

dandy_don

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sip_nat.conf?

Ward, et al.,

In looking around for clues, I've found several posts that mention that the ip address or host address should be in sip_nat.conf.

The sip_nat.conf is completely empty in my incredible pbx setup. Is it likely this is the problem?

Thanks,
Don
 

jrglass

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I had the same problem and I followed this.

Solving One-Way Audio Problems. If you experience one-way audio on some of your phone calls, you may need to adjust the settings in /etc/asterisk/sip_custom.conf. Just uncomment the first two lines by removing the semicolons. Then replace 173.15.238.123 with your public IP address, and replace 192.168.0.0 with the subnet address of your private network. There are similar settings in gtalk.conf that can be activated although we've never had to use them. In fact, we've never had to use any of these settings. After making these changes, save the file(s) and restart Asterisk with the command: amportal restart.

Jeff

Ward, et al.,

In looking around for clues, I've found several posts that mention that the ip address or host address should be in sip_nat.conf.

The sip_nat.conf is completely empty in my incredible pbx setup. Is it likely this is the problem?

Thanks,
Don
 

dandy_don

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Jeff,

I tried as you suggested but still no luck. Anyone in the Dayton Ohio area that might be able to help me figure this out?

Thanks,
Don
 

vanDivX

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Shouldn't you guys first find out if you remote phone registers and gets voice when you don't use Travelin' Man?
Simply 'premit' in FPBX the IP (or IP range if you are not certain of particular IP) of your remote location and see if you register and if voice works. You could also permit all IPs, nobody is going eat your PIAF right away LOL

Because it seem to me as if your problems have nothing to do with Travelin' Man but simply problems with remote extension as such which is altogether a separate thing, isn't it?

BTW with Incredible (and I think really with any kind of PIAF) the sip_nat.conf is empty because the audio problems that were handled in it are now handled in the FPBX in Tools - Asterisk SIP Settings menu. At the very top is the Nat Settings where you select NAT Yes and then if you got Static or Dynamic IP (if your IP is dynamic but actually doesn't change for years on end you can use 'static' setting. The AutoConfigure button actually detects your router IP and fills it in, neat. If dynamic you have to type in your FQDN
 

dandy_don

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remote extension registers... Just NO audio -- Ever...

The remote extension does in fact register. With it I can ring the phones inside of the firewall, but there is NEVER any audio passed. Never.

I don't know where the problem is. When I use that same netbook and connect via the wifi connection on the inside of my firewall, I can use the softphone perfectly. When I try to use traveling man outside the firewall, it registers, can ring phones, and I can ring the remote extension from phones internal to the firewall, but there is NEVER any audio.

I have tried everything I have found by searching the web, this forum, etc., but still no luck.

Does anyone have any idea how to fix this?

Thanks,
Don
 

wardmundy

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Replace your router with a dLink that handles NAT properly. :wink5:
 

vanDivX

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You should check if the netbook phone uses SIP RTP ports 10000-20000 range, some phones have settings for it and its set somewhere in the range of 3000 -6000 in which case it won't work since you have forwarded ports 10k to 20k on your firewall to PIAF - on second thought perhaps it shouldn't matter how its set on the phone, still its worth looking and changing it.

I used to have remote SIP extensions working, then I upgraded FPBX to newer version and I lost voice on them and like you I tried this and that and still no voice, then I installed new version of PIAF in VMWare and with it the phones have voice and I am reinstalling my hardware based PIAF and trust it will work on it too...

I would suggest if your PIAF is not latest version to use this as opportunity to reinstall or maybe experiment with it in VMWare - once installed and configured with basic extensions, shutdown your hardware PIAF and give the VMWare PIAF the same IP so you don't have to change router/firewall forwarding and see if it works.

Its not like I changed settings freely and messed up and then my voice on remotes didn't work, perhaps the FPBX upgrade resulted in some glitch or a bug and voice on remotes just wouldn't work no matter how I fiddled with PIAF, sometimes its worthwhile to try new install and VMWare is easy to do.

If the reinstall didn't work I'd look at the router as suggested. In my case I didn't because I had remotes working before.

@Ward: I feel like I should ask for rank demotion, guru is overblown in my case, I don't want people think I am some expert because I am still bumbling
 

wardmundy

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Sorry, but you've earned your stripes. You have to keep 'em. :gnorsi:
 

graybans

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SIP Port really high

I am also having the audio issue with the Bria iphone edition. It will ring the phones but there is no audio either way. I verified the phone registers and can see it in Asterisk Info, but the SIP port is really high, 61816, 58432, etc. Can't figure it out. I have tried using several STUN servers. and reconfigured darn near everything...no luck. Any ideas?
 
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I am also having the audio issue with the Bria iphone edition. It will ring the phones but there is no audio either way. I verified the phone registers and can see it in Asterisk Info, but the SIP port is really high, 61816, 58432, etc. Can't figure it out. I have tried using several STUN servers. and reconfigured darn near everything...no luck. Any ideas?

What type of router are you using? You need to enable "consistant NAT" so that your router will map consistantly to port 5060.
 

graybans

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Where I am I have no control over the router. I have also tried to use the 3G on the iPad with the same result. My router at the office is a RV-042 with 1:1 NAT. Is there a stun server or other setting that can help?
 

dandy_don

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Ward,

I am using a WBR-2310 but I'm not convinced it is a good router. I need to reboot it periodically... Perhaps I got a lemon. I'll buy something tomorrow during the sales.

Thanks,
Don
 

chemcat9

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Replace your router with a dLink that handles NAT properly. :wink5:

I've finally had some time to work on access using the Travel'nman script and have been successful using the DLG-4100. It might be worth noting that trying to connect with various soft clients may be necessary. I still can not connect with the 3CXPhone, but can with the Zoiper and QuteCom softphones.

For reference I created the following:
Virtual Servers:
Name IP Address Protocol Private/Public Port
SIP 192.168.x.xxx UDP 5060/5060
Travelnman 192.168.x.xxx TCP 83/83

Game Rules List
Name IP Address Ports Inbound
SIP 192.168.x.xxx UDP 5060 Allow All
HTTP 192.168.x.xxx TCP 83 Allow All
RTP 192.168.x.xxx UDP 10000-20000 Allow All

x.xxx represents the final local ip of the Incredible server

Can others with the Travel'nman working comment on their configurations. I'd like to ensure I've not opened the network too much and have preserved the inherent security of the Incredible PBX.

Thanks in advance,

Ralph.
 

chemcat9

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Well ... so much for that. My Travel'nman quit on me. Pulls the ip but doesn't allow me any access. A trouble shooting matrix would be nice. Best I can tell is that the ports in are being blocked (http://www.canyouseeme.org/).

No changes were made to the network. I have to question as to whether or not it was setup properly to begin with.
 

chemcat9

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Continuing to troubleshoot using DynDNS.com, it appears that port 5060 isn't open even though in the DGL4100 I have forwarded UDP 5060 to the pbx server.

Webmin has the following in the iptables (I have not added/deleted any entries, this is how it appears as loaded?):

Accept If input interface is not eth0
Accept If protocol is TCP and TCP flags ACK (of ACK) are set
Accept If state of connection is ESTABLISHED
Accept If state of connection is RELATED
Accept If protocol is UDP and destination port is 1024:65535 and source port is 53
Accept If protocol is ICMP and ICMP type is echo-reply
Accept If protocol is ICMP and ICMP type is destination-unreachable
Accept If protocol is ICMP and ICMP type is source-quench
Accept If protocol is ICMP and ICMP type is time-exceeded
Accept If protocol is ICMP and ICMP type is parameter-problem
Accept If protocol is TCP and destination port is ssh
Accept If protocol is TCP and destination port is auth
Accept If protocol is TCP and destination port is 80
Accept If protocol is TCP and destination port is 83
Accept If protocol is TCP and destination port is 443
Accept If protocol is TCP and destination port is 21
Accept If protocol is TCP and destination port is 9001
Accept If protocol is TCP and destination port is 9080Accept If protocol is UDP and destination port is 5222
Accept If protocol is UDP and source is 64.27.1.153 and destination port is 4569
Accept If protocol is UDP and source is 66.54.140.46 and destination port is 4569
Accept If protocol is UDP and source is 66.54.140.47 and destination port is 4569
Accept If protocol is UDP and destination port is 5000:5082
Accept If protocol is UDP and destination port is 10000:20000
Accept If protocol is TCP and destination port is 4445
Accept If protocol is TCP and destination port is 5038
Accept If protocol is TCP and destination port is 88
Accept If protocol is TCP and destination port is 5060
Accept If protocol is UDP and destination port is 123
Accept If protocol is UDP and destination port is 69
Accept If protocol is TCP and destination port is 9022
I'm not certain where it is pulling these settings from, but the forth to the last entry shows 5060 as TCP, shouldn't 5060 be UDP? Can someone post their IP table of a working system for comparison?

Thanks.
 

tm1000

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Continuing to troubleshoot using DynDNS.com, it appears that port 5060 isn't open even though in the DGL4100 I have forwarded UDP 5060 to the pbx server.

Webmin has the following in the iptables (I have not added/deleted any entries, this is how it appears as loaded?):

Accept If protocol is UDP and destination port is 5000:5082


You have a range open from 5000 - 5082 UDP. So yes UDP 5060 is open

Also TCP 5060 is supported in Asterisk as well as UDP 5060 (SIP TCP & SIP UDP)
 

chemcat9

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You have a range open from 5000 - 5082 UDP. So yes UDP 5060 is open)

Thanks, I totally missed that even though I looked straight at it when I spotted the rtp ports.

This leaves things a mystery as DynDNS sees some of the other ports opened in the router but not necessarily 5060 (DynDNS reports: An attempted connection to 165.xx.xx.xxx:5060 was refused. This typically indicates that there are no services available on that port, but that it is NOT being blocked by a firewall or your ISP), nor rtp. This appears to be more of a router issue.
 

vanDivX

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the SIP port is really high, 61816, 58432,

high range of sip ports sounds like funky router, I had at one place remote sip phone behind Huawei modem/router and it did that and I ended getting rid of it and the WRT54GL with custom firmware (Tomato) right away kept 5060 port...

Never quite understood that router, all setting on it was terribly non-standard including port forwarding, don't touch that brand again.

---------------------------------

I've just tested and for me Travelin' man works just fine, after setting it up my remote phone was keeping registered for more than 24h but then I rebooted PIAF and the phone no longer could register and on the display I got message - Registration failed, 401 not authorized access (or was it unauthorized access?), then I just went to the Travelin' Man webpage and as the phone didn't register automatically (my new Nortel1535) I went to its menu and registered no problem and made calls and it worked (I have two remote sip extensions actually at the remote location and I tested by making call from one phone to the other).

Those for whom it doesn't work I strongly suggest what I said before, troubleshoot remote extension only and only bring in Travelin' Man after you made your phones working without it - make sure you configure correctly that Asterisk SIP Setting in FPBX "NAT Setting" section at the top of the page (under Tools tab) as I said in my previous post here, it really does work, initially when I was setting up the two remote extensions I also didn't have any voice and configuring this section fixed it right away.

Thank you Ward for such terrific way of securing remote SIP extensions. I have been wondering if Travelin' Man might be made to work for remote IAX extensions as well?
 

wardmundy

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The remote IP addresses activated in IPtables using Travelin' Man are never saved to your IPtables configuration, i.e. they live only in RAM. So either a reboot or service iptables restart will wipe them out and force them to be reactivated. :aureola:
 

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