Topic: Asterisk dropping out every so often

[David Harper]

What version of Asterisk are you using?

rpm -q asterisk

[d6hq]

asterisk-1.4.18.1-59.el4

All servers are SME7.3 loaded from the Sark 7.3 iso then updated via yum localinstall to Sail-2.2.1-716 or higher. The servers are connected on broadband lines and all other services - very few actually being used - are working without issue. Confguration changes have been minimal. The two analogue cards are from OpenVox.

[David Harper]

I would upgrade Asterisk (& addons etc.) to the version in my SARK archive. This is the last version that works with Zaptel.

http://sme.david-harper.com/data/sark/

See if that fixes your issue.

[d6hq]

Thanks David - we will give it a go this evening.

Is the zaptel driver in there suitable for 7.4 ?

[David Harper]

Only 7.4 Gold (-78.0.8 ). The driver for the updated kernel that was released a few days ago (-78.0.13) is in the smecontribs and/or atrpms repository.

[d6hq]

We did the update as suggested and are now monitoring the situation to establish whether or not we still get the dropouts. I will report back.

[SARK devs]

What does 'sip show peers' give if you type it at the asterisk console?

S

[d6hq]

Returning to this issue. We have yummed the test box to 7.4 with 7.8.0.13 kernel. Upgraded Asterisk and Zaptel turned on full logging and left the thing to run untouched since Monday afternoon. Asterisk has failed at some stage overnight on Tuesday / Wesdnesday. It ran for 24 + hours OK but is displaying exactly the same symptoms as before. A sudden unexplained stop of Asterisk for no apparent reason.

Log extract.

[Mar 24 09:58:28] WARNING[4350] chan_sip.c: Got 200 OK on REGISTER that isn't a register
[Mar 24 10:23:18] NOTICE[4350] chan_sip.c: Peer '8232498' is now UNREACHABLE!  Last qualify: 39
[Mar 24 10:23:45] NOTICE[4350] chan_sip.c: Peer '8232498' is now Reachable. (1517ms / 3000ms)
[Mar 24 11:04:53] NOTICE[4350] chan_sip.c: Peer '8232498' is now UNREACHABLE!  Last qualify: 38
[Mar 24 11:05:03] NOTICE[4350] chan_sip.c: Peer '8232498' is now Reachable. (62ms / 3000ms)
[Mar 24 11:42:09] NOTICE[4350] chan_sip.c: Peer '8232498' is now Lagged. (4059ms / 3000ms)
[Mar 24 11:42:19] NOTICE[4350] chan_sip.c: Peer '8232498' is now Reachable. (45ms / 3000ms)
[Mar 24 13:07:39] NOTICE[4350] chan_sip.c: Peer '8232498' is now UNREACHABLE!  Last qualify: 36
[Mar 24 13:07:49] NOTICE[4350] chan_sip.c: Peer '8232498' is now Reachable. (36ms / 3000ms)
[Mar 24 13:55:42] WARNING[4350] chan_sip.c: Got 200 OK on REGISTER that isn't a register
[Mar 24 14:39:19] WARNING[4350] chan_sip.c: Got 200 OK on REGISTER that isn't a register
[Mar 24 14:53:57] WARNING[4350] chan_sip.c: Got 200 OK on REGISTER that isn't a register
[Mar 24 15:03:44] WARNING[4350] chan_sip.c: Got 200 OK on REGISTER that isn't a register
[Mar 24 15:18:15] WARNING[4350] chan_sip.c: Got 200 OK on REGISTER that isn't a register
[Mar 24 16:20:22] NOTICE[4350] chan_sip.c: Peer '8232498' is now Lagged. (4396ms / 3000ms)
[Mar 24 16:20:32] NOTICE[4350] chan_sip.c: Peer '8232498' is now Reachable. (37ms / 3000ms)
[Mar 24 21:13:14] VERBOSE[4278] logger.c:   == Starting post polarity CID detection on channel 1
[Mar 24 21:13:15] NOTICE[5539] chan_zap.c: Got event 17 (Polarity Reversal)...


Test box hardware is as follows.
Dell SC600 - 640mb RAM, 2 x 80GB HD.
Openvox 400P card with 2 x FXO modules from Voipon.co.uk

My money is on the PCI card as the box was originally built without and did not start failing until the card and the necessary directives for UK caller ID were added.

[David Harper]

So let's confirm that suspicion by removing the card (over the weekend perhaps) and seeing how Asterisk copes.

[d6hq]

Thanks David - I have added the log extract from asterisk/full above. As you will see the last event is a polarity reversal on the card then nothing. I shall remove the card this afternoon and run it overnight without.

Can anyone advise whether they have experienced similar issues with true Digium cards in UK with BT caller ID on.

The zapata.conf modifications as advised by Selintra are follows

usecallerid=yes
callerid=asreceived
cidsignalling=v23 ; Added for UK CLI detection
cidstart=polarity ; Added for UK CLI detection
;internationalprefix: 00
;nationalprefix:0

[SARK devs]

We will check the various BB's to see what we can see.  We ourselves have only one customer who uses analog (4 sites), but they don't use CLIP and we've never had a failure of this nature with them (their oldest site is just coming up for its second birthday).

For us, analog isn't worth the hassel so we only ever install BRI and PRI (mostly PRI).   Digium Analog is very Ma Bell orientated and it shows in the zaptel code, which is bodged up to hell and back for BT and other networks.

Sorry we can't help more but this really does look like an Asterisk or Digium card issue.

Best

S

[d6hq]

Selintra

Unfortunately ISDN in any form isn't an option here. We have two SME customers where they wanted to keep their BT line and number but expand their ability to make outbound calls.

In one case we have managed to work around the drop out with a simple reload of sail at 4am whether the system is down or not. This works but has an annoying bi-product of making the extensions and trunklines appear to be unregistered in the GUI even though they are properly registered. We can live with that Ok.

In the other case the drop outs are more frequent and more severe in their effect. We applied a reload at 4am but found further drop outs before they open for business at 9am so changed to 8am only to find them occurring between 8am and 9am. They make the system too unreliable. In this case they need to keep the analogue BT circuit as the number is in circulation. They do however have 4 x Siemens S450's on a single base station so I guess I could remove the OpenVox card and reimplement the Siemen's landline support but then they don't get the voicemail etc etc etc.

Your customer with analogue cards - do you recall what manufacturer ?

[SARK devs]

All Digium - we don't use anything else (except for single and twin BRI, which Digium don't do so we use OpenVox). 

Try running without CLIP and see if it cures it.  It looks to be failing on the polarity reversal.

Best

S

[d6hq]

We will try without CLIP as this does look like it could be the cause.

Have you any experience of the Cisco SPA3102 used simply as an FXO gateway ?

[SARK devs]

Yes, we have a site in Scotland which we upgraded from Tbox and they have either 2 or 3 of them (can't remember which).  They seem to work OK but I've heard a lot of horror stories regarding echo and such like.

Best

S