VoIP phone which cannot be called (no incoming calls)

Asterisk Asterisk Support
1

Hello,

I have a problem with one phone (snom 300) in a voip network based on freePBX 2.1.1.
I added 8 phones to the network. Settings were changed for each one of them in a same way. I add extension number (1604) and inbound route for it as well, but…i cannot call this number, i get a message: service 1604 unavailable. But i can make calls from that number(1604) to any other phones! everything seems to be fine, except of that you can call that number(1604) from other phone.
Settings on a phone are corrrect. I set up that extension number(1604) on another phone and… it’s the same problem! so it has to be sth with freePBX settings, sth had to be added to the freepbx settings, sth wasnt cleared and there are results…

here are few lines from asterisk log file (freepbx -> tools)

Sep 25 11:43:11 VERBOSE[4848] logger.c: -- SIP Seeding peer from astdb: '1604' at 

1604@x.x.x.x:x for 3600 
(...) 
i've found this line in: /var/log/asterisk/full 
(...) 
Sep 25 11:43:16 DEBUG[13275] db.c: Unable to find key 'SIP/1604' in family 'cfb' 
(...) 
Sep 25 11:43:28 VERBOSE[4848] logger.c: -- Unregistered SIP '1604' 
Sep 25 11:43:28 DEBUG[4836] chan_sip.c: Checking device state for peer 1604 
Sep 25 11:43:28 DEBUG[4836] devicestate.c: Changing state for SIP/1604 - state 5 

(Unavailable) 
Sep 25 11:43:28 DEBUG[4836] chan_sip.c: Checking device state for peer 1604 
Sep 25 11:43:28 DEBUG[13276] app_queue.c: Device 'SIP/1604' changed to state '5' 

(Unavailable) 
Sep 25 11:43:28 DEBUG[4848] chan_sip.c: **** Received REGISTER (2) - Command in 

SIP REGISTER 
Sep 25 11:43:28 VERBOSE[4848] logger.c: -- Registered SIP '1604' at x.x.x.x port x 

expires 3600 
(...)

do you know where can be a problem?
thx

2

Besides for setting up a sip account for the phone you have to set up an exten statement so asterisk knows where to send the call. Something like
exten => 1234,1,Dial(SIP/1234).

3

jack, does the problem temporarily disappear if you force asterisk to reload it’s config? (i.e. by saving some form in freepbx and submitting the change)

maybe we’ve got the same problem over here. after some time of normal operation (1-3 days), asterisk seems to lose some extensions (not always the same) – the ‘lost’ extensions can call but cannot be called. rebooting the phone or restarting the softphone does not solve this, only an asterisk reload does.

we have this problem with softphones as well as hardware phones (see signature).

we started about two months ago with asterisk 1.2.10 and upgraded to asterisk 1.2.12.1 a few days ago, but this did not help.

our SIP provider suggested downgrading to asterisk 1.0.x – i am forced to consider this but have yet to check if it’s compatible with freepbx 2.1.1 and if we would lose some needed functionality…

so any advice is very welcome.

michael

4

i’ve one client who reported this issue … running Asterisk 1.2.4 -> 1.2.12.1 the same symptoms showed … got rid of the GXP-2000s and the problems gone.

5

we have this problem since switching to asterisk. the GXPs have been added recently to replace most of our softphones (and btw solved most of the problems we had with audio/conn quality and usability). we used only softphones (twinkle, idefisk, x-lite, kphone) before and had the same problem with those.

michael