Voip hardphone Outgoing ok but Incoming does nothing


I’m fairly new to asterisk and I am trying to get a F1000 wireless voip phone to work with the sever. Currently the phone connects with the correct extension and can make outgoing calls but whenever it’s extension is dialed it does not ring. I also have multiple corded hardphones connected to the asterisk box which work flawlessly Incoming and Outoing.

In the output of the following command ext. 505 is the hard phone and 504 is the wireless one.

bluelava-aib*CLI> sip show peers
Name/username Host Dyn Nat ACL Mask Port Status
505/505 D N 5060 OK (27 ms)
504/504 (Unspecified) D N 0 UNKNOWN

Any ideas about what the issue could be? I’ve head if asterisk displayed Unspecified it could cause issues but what type of issues and how do I fix these, or is this even an asterisk problem?

FYI, both phones use publicly routable ip addresses.

Thanks in advance,

Just guessing: could it be a simple registration issue? What does sip show registry say?

Strangely enough “sip show registry” shows nothing. BUT there have been 20 phones running on this sever for over 2 years calling eachother and calling out and recieving calls.

I just turned the phone off and on while monitoring asterisk, the flollowing is the output. Right after the phone registerd I tried to contact it from another extension that I know works (205)

bluelava-aib*CLI> sip show registry
Host Username Refresh State
– Registered SIP ‘503’ at port 5060 expires 6
– Saved useragent “UTSTARCOM F1000/Device ID-0007ba2520cb” for peer 503
– Executing Macro(“SIP/205-f9bd”, “stdexten|503|SIP/503”) in new stack
– Executing Dial(“SIP/205-f9bd”, “SIP/503|20|t”) in new stack
Feb 21 16:51:07 NOTICE[589839]: app_dial.c:696 dial_exec: Unable to create channel of type ‘SIP’
== Everyone is busy/congested at this time

Here is me calling 205(corded phone) from the 503(wireless phone)

== Spawn extension (internal-origination, h, 1) exited non-zero on ‘SIP/503-316c’
– Executing Macro(“SIP/503-890d”, “stdexten|205|SIP/205”) in new stack
– Executing Dial(“SIP/503-890d”, “SIP/205|20|t”) in new stack
– Called 205
– SIP/205-bef6 is ringing

You may want to turn up SIP debug (and other debug).

Just want to add that we need to check REGISTER messages in debug.
Probably the phone cannot register itself to * and, as a result, * cannot send the calls to it. One of the reasons may be very short expiration value or simply the authentication issue.

Thanks, the register was set for 6 seconds I changed that, rebooted the phone and it works! Hope the same fix will work on the other ones too.