Summary:ASTERISK-18029: Aastra 480i does not work with Asterisk 1.8
Reporter:David Brillert (aragon)Labels:
Date Opened:2011-06-16 15:49:50Date Closed:2011-06-17 10:23:07
Versions:1.8.4 Frequency of
Environment:Attachments:( 0) aastra480i_fw-1.4.3-SIPdebug_ast1.8.4.2-no-audio.txt
( 1) aastra480i_fw-1.4.3-SIPdebug_ast1.8.4.2-no-audio.txt
( 2) jira18029_SIP-debug.txt
Description:480i phones cannot make or receive any calls after upgrading from *1.4 to *
Looks like a call limit problem during CODEC negotiation.
Aastra 480i phones are discontinued so no further support from Aastra but since they used to work on *1.4 I think there must be a solution in 1.8.x
Comments:By: David Brillert (aragon) 2011-06-16 15:52:48.764-0500

SIP debug info

By: David Brillert (aragon) 2011-06-17 09:34:50.236-0500

SIP debug

By: David Brillert (aragon) 2011-06-17 09:39:13.460-0500

jira18029_SIP-debug.txt shows failure due to too many calls.
However this is not the source of the actual failure to produce signalling or audio.
The 480i testing involved simply did not set up or tear down and SIP channels properly and left lots of open events in sip show channels.
8 test calls is enough to busy up the extension in Asterisk.

sip show channels
Peer             User/ANR         Call ID          Format           Hold     Last Message    Expiry     Peer      (None)           30b4d7d0-c0a801  0x0 (nothing)    No       Rx: OPTIONS                <guest>   6001             20a5ab74a96c624  0x0 (nothing)    No       Rx: SUBSCRIBE              <guest>   6009             86448ce012421a8  0x4 (ulaw)       No       Rx: BYE                    6009      (None)           30b4d988-c0a801  0x0 (nothing)    No       Rx: OPTIONS                <guest>   (None)           00890b55-3a47cb  0x0 (nothing)    No       Rx: OPTIONS                <guest>   6009             edf1d00162b2f9c  0x4 (ulaw)       No       Rx: BYE                    6009   6009             cc8976622c745d0  0x4 (ulaw)       No       Rx: BYE                    6009
7 active SIP dialogs

The real problem I think appears in aastra480i_fw-1.4.3-SIPdebug_ast1.8.4.2-no-audio.txt with a 422 client response Session Interval Too Small.
What to do next?

By: David Brillert (aragon) 2011-06-17 10:14:27.265-0500

I see that Aastra 480i has a setting to define SIP session timer for SIP keep alives which I had previously defined at 30 seconds when using ast1.4 and some Googling shows that Asterisk 1.8 has default session timer 1800 seconds.  I tested Aastra with sip session timer = 1800 seconds and could finally make calls.
I also found an open bug report that may be related https://issues.asterisk.org/jira/browse/15945
I'm not sure if I have a config problem or if Asterisk 1.8 should be dealing with the 422 sip session timer too small error more gracefully?
Is this a bug or not?
sip show channels shows the effected extension 6009 in constant bye status and I thought this was supposed to clear so there are no lingering SIP channels which in my testing causing Aastra 480i firmware 1.4.x to fail with too many calls.

My sip.conf has no entry for:
session-timers=["accept", "originate", "refuse"]
session-refresher=["uas", "uac"]

So given that I would like the Aastra to support session timer value of 30 seconds for keep alives what would those recommended settings be?

By: David Brillert (aragon) 2011-06-17 10:22:41.365-0500

I did some reading and I think this is a configuration issue:
Asterisk has some default values which the Aastra configs must respect.

My Aastra config for SIP session timers=30 which is not supported by Asterisk.
This ticket can be closed

By: David Brillert (aragon) 2011-06-17 10:23:07.732-0500

config issue