I need Help Please.Cannot Recieve Calls on remote

[rollinsolo]

I am having issues receiving calls on my Remote IP phones, they are the Atcom 510 phones, I know not the most popular, but I gave them a shot, anyways I have all of the ports forwarded to the phone from the remote site udp 5060 and 10001 to 20000, as well those ports are forwarded to the PBX site also. I am using the stable 1.3 Elastix and have edited my sip_nat.conf file two ways, one I used the static IP address attached the the router

externip=XX.xx.xx.xx
localnet=192.168.1.0/255.255.255.0

and the other for dydns which I do use even though he has a static ip address (easy to remember)

externhost=mydomain.dyndns.org
externrefresh=120
localnet=192.168.1.0/255.255.255.0

I have not used them both at the same time I have just replaced one with the other and did amportal restart.

I can make calls from all of the remote ip Extensions, but I cannot receive calls. The only way I was able to accomplish a received call is setting up and additional IAX2 extension that the ip phone accepts and then I can make and receive calls but after a couple of hours or so, this breaks so I decided to dive in head first and I have scoured the forums to find an answer. I am sure it will be something simple. But I really need to fix this so I can move on with my life. LOL. Anyways here is the pastebin from the CLI of me calling into the system and trying to dial the remote extension to test. http://pastebin.com/m1c1e2a39 .

I am using a PRI and zaptel. The phones register for both SIP and IAX, but I know I am doing something wrong. He has Linksys WRTG54 routers at each site. I know they are not built for VOIP but I would think as long as the ports are forwarded then I should be fine.


Any help would be a God Send. Thanks.

 

[blackgecko]

use this in the CLI
CLI> sip set debug ip xxx.xxx.xxx.xxx

where xxx.xxx.xxx.xxx is the ip of the remote extension.

Make a new call and paste the result.

Are you using qualify=yes in this phones ?
maybe they time out and thats why you cant call them.

 

[rollinsolo]

Ok here is the new one, Thanks again for your help, I have had a few people try and not successful.

http://pastebin.com/m601df6ea

I ran the sip set debug ip xx.xx.xx.xx ( I assume you mean the external (what is my IP) of the network that the phone is attached).

Thanks. Solo.

 

[rollinsolo]

To answer the other question yes I am using qualify=yes on all of the extensions.

 

[blackgecko]

can you paste a
CLI>sip show peers

based on your trace it seems is having troubles to register, what kind of ipphone are you using ?

 

[rollinsolo]

Before I can do that how do you stop this debugging once it starts
it is just going crazy.

 

[rollinsolo]

Ok I got sip show peers to work and its gone now, but it says the remote phones are unreachable they are Atcom 510 phones my x partner said they would be great, well he is my ex partner for a reason.

 

[rollinsolo]

Name/username Host Dyn Nat ACL Port Status
804 (Unspecified) D N 0 UNKNOWN
803/803 xxx.xx.xx.xxx D N 5060 UNREACHABLE
802/802 xxx.xx.xx.xxx D N 5060 UNREACHABLE
801/801 xxx.xx.xx.xxx D N 55301 UNREACHABLE
800/800 xxx.xx.xx.xxx D N 49155 UNREACHABLE
7447/7447 192.168.1.10 D N 5060 OK (12 ms)
7446/7446 192.168.1.9 D N 5060 OK (11 ms)
7445/7445 192.168.1.2 D N 5060 OK (13 ms)
7444/7444 192.168.1.3 D N 5060 OK (11 ms)
7443/7443 192.168.1.5 D N 5060 OK (10 ms)
7442/7442 192.168.1.7 D N 5060 OK (11 ms)
7441/7441 192.168.1.8 D N 5060 OK (12 ms)
7440/7440 192.168.1.11 D N 5060 OK (44 ms)
6901/6901 xxx.xx.xx.xxx D N 5060 UNREACHABLE

 

[rollinsolo]

Dont know why they all show Nat as N it should be yes, I have checked my settings they are all set to yes from the web gui

 

[blackgecko]

sip set debug off

to stop debugging.

They cant receive calls because they are unreachable, ive ran across this issue with linksys, there should be an option in the ipphone to send nat keepalives.

ill check the atcom manual to see what i can get, ill let you know

 

[blackgecko]

its on the advanced sip settings

Register Expire Time

Nat keepalive interval

play with those 2 parameters

 

[rollinsolo]

I have been playing with those parameters for over a week now, I am stuck at a loss

 

[rollinsolo]

http://pastebin.com/d57d44ea4

no I have two peers showing up with a crazy port #, I am not sure why yet.

 

[blackgecko]

can you try using qualify=no in those remote extensiones.

also enable the debug and look if the extension is actually sending the keepalives

 

[rollinsolo]

By golly I think that did it. We are looking good, also I was forwarding ports 5060 and 10001 to 20000 to each phone, it was not necessary so I removed it and some of them came up now I will enter qualify as no and Test for a day or two. Thank you so much, I hope your on here more often I see you post as much as me but it seems you know what your talking about. Where are you located, I'm in Tampa, FL

 

[rollinsolo]

Its weird that some of them did not need to put qualify = no they had yes and worked. I think it was all router issues.

 

[rollinsolo]

OK so I thought everything was working, but I think qualify might be important, I can ring the remote extensions but they cannot actually pick the phone up and hear the calling party, all they hear are the phone ringing and then it goes to VM. Hmmm back to debugging, I think we are close though. Any thoughts.

 

[rollinsolo]

Sorry Correction they do not hear the phone ringing if they try to pick up the call they hear nothing.

 

[blackgecko]

sorry i was out of the office, the qualify is a parameter that test the latency that exist between the endpoint and the pbx, when you set it to no, it doesnt care how much latency is there, what this test probes is that you have to much latency between this two points.

 

[rollinsolo]

why dont you jump in the new thread someone started and I answered with a similar problem, I think you could help there, you know what your talking about for sure.
here is the link

http://www.elastix.org/index.php?option ... mitstart=0