local SIP 7940 to Hosted VPS Elastix

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datagroove

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I used to have elastix on a vm locally, now i have it hosted on a server in the cloud. All I did was use the backup/restore option to retain my settings.

After making the move, and changing the TFTP on my Cisco 7940 SIP phone, I was able to make a few calls with no tweaks to any settings. This made me happy. But then, without changing anything, my phone no longer works. I try to make a call and nothing. Any ideas what happened, or what I have to do to get the phone on my desk to play nice with elastix in the wild?

Thanks
 
datagroove,

Have you tried another phone to see if it works and can place calls properly? Maybe check to see if something like IP address or passwords changed? Have you tried to reload/reboot phone?

amphibian
 
Thanks for the reply, I will try another phone, but i don't think that is it. I have done several factory resets on this phone, and again, it did work initially. I think it's a NAT or DNS issue. Even now, after a factory reset, i edit the TFTP and the phone reads the configs from the remote server.

I did notice that the extension info it is receiving, is outdated. For instance the name on the top right is name I used to use, i changed the name in elastix extension page and yet it still shows old name. Wondering if maybe there is a file somewhere that is defaulting all the settings, and as a result my changes to name, or more specifically the NAT settings, are not taking place.

where is that file on the server that the phone loads when it hits the TFTP?
 
If you don't mind me asking, where are you hosting your VPS?

As for the file, it is under /tftpboot/ directory. There you should see a file with the MAC address of your handset.
 
Hi. hinzinho, that's right, the tftp dir is at root.

I Got in there and saw the cnf file for the phone I am using. Changed the phone label just to ensure that the phone is reading from the tftp server, and it is. The label on my phone changed. But I still can't make calls. Even after putting asterisk into -vvvvr on the command line, I see nothing when I try to dial out. It's like my phone can get to the TFTP part of my server, but not the actual asterisk/elastix ssytem.

If anyone has thoughts on the following, please let me know:
1) NAT for this setup, on a cisco 7940, yes or no or never?
2) Port #, should i be opening for forwarding anything on my local router?
3) Endpoint discovery is what I used to use when the server was local and the phone was on the same network, but that feature of elastix seems to only work when local, right? Because the ip of the phone and ip of the server are on different networks.
4) should i be changing anything in the sip default cnf file or any other file in the TFTP folder?

thanks again for any help.


@hinzinho, I have a server in the wild the runs VMWARE ESXi, i just mounted the elastix iso and installed on that.
 
I would first check if your phone is registered under Elastix. Run "sip show peers" to see if your extension is registered. If Elastix is on a remote server, make sure you have ports 5060 and 10000-20000 open to your IP. Also Cisco 79XXs have issue which require messing with the NAT under the config to make it register to Asterisk. I would also check your files under the TFPT to ensure the ip address of the Elastix box is correct.
 
I am one step closer, the phone now shows date and time too.
the default.cnf file did have local IP info, i just changed every line that had a 192.168.x.x to the elastix server IP.


Here is the output of sip show peers (111 is my extension):
Name/username Host Dyn Forcerport ACL Port Status
101 (Unspecified) D N A 0 UNKNOWN
111 (Unspecified) D N A 0 Unmonitored
Simon-GV-BFF/GV1754XXXXXXX (Unspecified) N 0 UNKNOWN
Simon-GV-CB/GV1954XXXXXXX (Unspecified) N 0 UNKNOWN
Simon-GV-Tt/GV175XXXXXXX (Unspecified) N 0 UNKNOWN


I will now mess with the nat (i presume under the default.cnf ? or the extensions in the GUI..) Thanks again for your help.
 
Your extension is not registered with the Elastix, therefore you won't see anything under console when making a call.

As for changing the file, make the changes under SIPDefault.cnf:
nat_enable: "1"

if that doesn't work, try "0". It just depends on your network setup. Also you might want to turn off the autocomplete too. autocomplete: "0"
 
i will try that. Quick question, each time I make a change to the default.cnf file, how do i ensure the phone receives the changes? Right now i am factory resetting, then going in there and changing TFTP. But i imagine there is a better way.
 
reboot the phone and it will pull files from tftp.
 
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