[Asterisk-Users] SPA-3000 Bug? Dropped calls while registering.

[Dana Harding]

Hello All!

I am in the process of assembling an asterisk-based phone system for my office -   using SPA-3000s to connect the network to the PSTN.   I am wondering if anybody else can get (or has already seen) the same behaviour out of their 3000.

The short version:   Send multiple Calls to the SPA's FXO port at the same time it is re-registering with Asterisk.
    SPA HTTP Configuration:      PSTN Line -> Register Expires:      5      (to ensure it is registering all the time)
    Dial one number through the SPA's FXO port - establish a conversation
    Dial another number through the same FXO port (SPA3000/NXXXXXY).

What SHOULD happen is the second caller receives a '504 - Service Unavailable' error while the first caller happily continues the established conversation.     What happens here:  the already established call gets dropped, AND the second caller gets a 504 error.

I did send a note to Linksys - and will see what kind of reponse they have.

With longer "Register Expires:" times (10, 30, 60 seconds) it took more attempts to make the call drop.        
I have my Register Expires time cranked up to 86400 (1 day) now - and am hoping I don't see another repeat.  

-------------------
There are three SPA-3000s in the system.       I looked at some more complicated dialplan layouts,  and decided to keep it simple:

exten => s,1,Dial(${PSTN2}/${ARG1},,n)
exten => s,2,Dial(${PSTN3}/${ARG1},,n)
exten => s,3,Dial(${PSTN1}/${ARG1},,n)
exten => s,4,Wait(1)
exten => s,5,Playback(all-circuits-busy-now)
exten => s,6,Congestion()

PSTN1,2,3 are 3 SPA-3000s registered with Asterisk.
This approach relies on the SPA denying a call if it is already in use.

Looking through the logs,  the SIP packets seem to be in order.     INVITE, 100-Trying, 504-Service Unavailable, ACK.

I'm at the end of my technical limit (ever increasing as I play in the open-source world) - but my best guess is:
During the Register process,  something is temporarily reset  (such as a variable indicating that the line is in use) such that when the second call comes in - it is actually connected to the existing conversation for a brief period before the SPA realizes that the line is actually already in use.      As part of a cleanup procedure - a hangup procedure is run:  disconnecting the call. 

The Equipment my trials were done on:
SPA3000 Hardware Version: 2.0.1(7376),     Software Version: 3.1.10(GWd),  and also tried Software 3.1.7.  
Nothing plugged into the FXS port.    
Asterisk 1.2.4 running on FreeBSD 5.4 (i386),  AMD Athlon 64 3200+, 1GB RAM.
SNOM 320.  Application-Version: snom320-SIP 5.3.6     Rootfs: snom320 jffs2 v3.36
Polycom IP501  <don't have access to the software/hardware version from where I am right now>
Cellphone

All SIP equipment is running on a dedicated LAN.  Network "splitters" were used to run two parallel LANs through the existing cabling.  (cat5e has 4 twisted pairs,  only 2 twisted pairs are needed for a 100BASET connection) The only computers on the LAN are the asterisk box,  and my workstation (2 NICs each). 


Regards,
Dana Harding
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