asterisk users - Apr 2006 - SPA-3000 Bug? Dropped calls while registering.

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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i just got a SPA3000 but still not using it on production, and i
havent tested deeply. However, have you tried using "incominglimit=1"
in the register context of the SPA?? i guess that would limit in the
PBX rather that sending the call to the SPA.

Regards

On 4/20/06, Dana Harding <dharding@nucleus.com>
wrote:
>
> 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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