http://sip.m2j.com.br/chan_ss7-trunk.tgz
I'll remove it from there next Monday.
DON'T USE THIS UNLESS YOU KNOW WHAT YOU'RE DOING.
DIFF THE SOURCE AND ANALYZE THE CHANGES.
I make a living out of providing paid support for linux/internet/tdm
stuff. So I won't provide any free support for anything.
On 07/12/12 07:58, Abdul Basit wrote:
These are good options. chan_ss7 need to be more optimized. I can test
in my environment if you can share some test cases.
What is the plan of including this patch in main stream?
On Thu, Jul 12, 2012 at 2:34 PM, Marcelo Pacheco <marc...@m2j.com.br
<mailto:marc...@m2j.com.br>> wrote:
A few bugs / features I found with chan_ss7 2.1.0, and the fixed I
applied in order to use it successfully:
1 - Even with dahdi mtp2 channel, chan_ss7 still sends FISU and
LSSU non-stop. When in mtp2 mode, it should conserve CPU by
sending SUs only when something has changed. Since I only plan to
use chan_ss7 with dahdi mtp2, I increased the select interval to
200ms to reduce the CPU intensity of the mtp3 sender thread. The
proper solution is to properly support mtp2 dahdi mode, not
sending extra FISU and LSSU never.
2 - ss7 link status - shows sent bytes as 16 always, this is
caused by writecount initialized with ZAP_BUF_SIZE and then never
incremented in dahdi mtp2 mode. I changed it so it gets
incremented only for successful writes with len > 6, only counting
sends with LI > 1 (skip FISU and LSSU). I also reduced writecount
and readcount to unsigned long, since 4GB is a LOT of sinalling
data, let it wrap.
3 - Nature of address 1 shouldn't add 00 prefix do ANI and DNI.
Disabled the intentional fall through in isup.c -
decode_isup_phonenum, this is needed for proper operations in Brazil.
Now to the bugs without a fix:
4 - seq_lth / seq_htl is buggy, after about 1000 calls in seq_lth,
I get:
[2012-07-04 12:34:04] WARNING[26854] l4isup.c: No idle circuit
found, linkset=XXX.
[2012-07-04 12:34:04] WARNING[26854] l4isup.c: SS7 requester: No
idle circuit available, linkset=XXX.
[2012-07-04 12:34:04] WARNING[26854] app_dial.c: Unable to create
channel of type 'SS7' (cause 34 - Circuit/channel congestion)
[2012-07-04 12:34:04] VERBOSE[26854] app_dial.c: == Everyone is
busy/congested at this time (1:0/1/0)
[2012-07-04 12:34:04] VERBOSE[26854] pbx.c: -- Auto
fallthrough, channel 'SIP/XXXXXX-00000001' status is 'CONGESTION'
even_mru works fine (processed about 2000 calls without issue).
The scenario is a single OPC/DPC pair with a single E1, CIC
1-15,17-31, signalling channel on 16, internal signalling link.
5 - ss7 reset crashes asterisk when in mtp3d mode.
6 - ss7 cluster crashes (it could be configuration dependent)
7 - chan_ss7 periodically suffers from some race condition and cpu
consumption spikes to 100% of one of the systems cpu threads. This
happens even with no active calls and no new call setups. I
confirmed this is caused by chan_ss7 by monitoring per thread cpu
consumption with top + H option and then executing strace on those
threads and the two threads that suffer from this are the
read/write to the sigchan threads.
I'm sending this as my contribution, however I gave up on chan_ss7
due to difficulty implementing the signalling network layout I
need. The same layout works beautifully with libss7 (on a single
system layout).
This is somehow not good. chan_ss7 should have more simplified layout.
Marcelo Pacheco
M2J Communications - Brazil
--
Regards,
Abdul Basit
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-ss7 mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-ss7
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-ss7 mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-ss7
