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

Dana Harding dharding at nucleus.com
Mon Apr 24 07:03:47 MST 2006 

Thank you.

It SEEMS to be working fine now as-is with the cranked-up registration time. 
When the time comes to tinker with it in the future - I will probably try 
working with groups again, or even work something out with astdb.  (and, 
most likely, end up breaking something that seems to work already)

I did have something similar to the problem outlined in the chain:   "RE: 
[Asterisk-Users] Polycom Asterisk 1.2.1 and Sipura SPA3000 strangeproblem" 
Where:
    - Call is in progress from USER1 -> Asterisk -> SPA -> PSTN_Person
    - SPA indicates an incoming call  (existing call is still in progress). 
Asterisk rings phones,  USER2 answers
    - USER2 answers, and ends up talking to PSTN_Person.        USER1 is 
disconnected.
I was playing with the spa3k's settings at the time, and attributed that 
instance to my actions.

Another instance:
    - Call is in progress USER1 -> Asterisk -> SPA -> PSTN_Person
    - SPA indicates an incoming call (existing call is still in progress). 
Asterisk rings phones, USER2 (me - in this case) answers
    - USER1 and USER2 and PSTN_Person end up in a 3-way call.
    - USER2 hangs up,    PSTN_Person is disconnected.
THIS occurred very close to the same time as the unit re-registering (I made 
some configuration changes and reset the box an hour earlier, with the 
registration time set to expire at 3600 seconds), and is what started me 
looking at it.

My test of making it hammer on the registrations isn't really a fair 
comparison to production use, and doesn't help in reproducing the two 
scenarios above - but it does seem to indicate that there is an issue in the 
registering code. (A dropped call is reproducible on 2 of my spa3k's - 
haven't tested the other two).

I guess what I am suggesting: as part of diagnosing an erratic behaviour 
problem with an spa3k,  look at the registering time(s).     I would be 
really interested if there is a correlation.



----- Original Message ----- 
From: "Moises Silva" <moises.silva at gmail.com>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
<asterisk-users at lists.digium.com>
Sent: Thursday, April 20, 2006 7:34 AM
Subject: Re: [Asterisk-Users] SPA-3000 Bug? Dropped calls while registering.


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 at 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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