Topic: Calls to DiD causing some sort of loop....

[gippsweb]

The last couple of days I'm seeing some strange stuff happening on incoming calls to my DiD number.
Incoming calls on the Did have asterisk calling the DiD number again and again until the 4 lines are used.
It seems to be throwing itself into some kind of loop.
the box is running sail 719 and asterisk 1.4.21

  == Spawn extension (default, h, 1) exited non-zero on 'SIP/5006-08a29948'
    -- Executing [0341024269@default:1] AGI("SIP/0094*00002-08a20520", "selintra|OutRoute|National") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/selintra
    -- AGI Script Executing Application: (Dial) Options: (SIP/0341024269@peer0094*00002)
    -- Called 0341024269@peer0094*00002
    -- Executing [0341024269@default:1] AGI("SIP/0094*00002-08a1eaf8", "selintra|OutRoute|National") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/selintra
    -- AGI Script Executing Application: (Dial) Options: (SIP/0341024269@peer0094*00002)
    -- Called 0341024269@peer0094*00002
    -- Executing [0341024269@default:1] AGI("SIP/0094*00002-08a30cb8", "selintra|OutRoute|National") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/selintra
    -- AGI Script Executing Application: (Dial) Options: (SIP/0341024269@peer0094*00002)
    -- Called 0341024269@peer0094*00002
    -- SIP/peer0094*00002-08a38b88 answered SIP/0094*00002-08a30cb8
    -- Packet2Packet bridging SIP/0094*00002-08a30cb8 and SIP/peer0094*00002-08a38b88
    -- SIP/peer0094*00002-b7d00578 answered SIP/0094*00002-08a1eaf8
    -- Packet2Packet bridging SIP/0094*00002-08a1eaf8 and SIP/peer0094*00002-b7d00578
    -- SIP/peer0094*00002-08a253d8 answered SIP/0094*00002-08a20520
    -- Packet2Packet bridging SIP/0094*00002-08a20520 and SIP/peer0094*00002-08a253d8
  == Spawn extension (default, 0341024269, 1) exited non-zero on 'SIP/0094*00002-08a20520'
    -- Executing [h@default:1] Hangup("SIP/0094*00002-08a20520", "") in new stack
  == Spawn extension (default, h, 1) exited non-zero on 'SIP/0094*00002-08a20520'
  == Spawn extension (default, 0341024269, 1) exited non-zero on 'SIP/0094*00002-08a1eaf8'
    -- Executing [h@default:1] Hangup("SIP/0094*00002-08a1eaf8", "") in new stack
  == Spawn extension (default, h, 1) exited non-zero on 'SIP/0094*00002-08a1eaf8'
  == Spawn extension (default, 0341024269, 1) exited non-zero on 'SIP/0094*00002-08a30cb8'
    -- Executing [h@default:1] Hangup("SIP/0094*00002-08a30cb8", "") in new stack
  == Spawn extension (default, h, 1) exited non-zero on 'SIP/0094*00002-08a30cb8'

Anyone have any ideas, or maybe have seen something like this happen before??

 

[David Harper]

I have had something like this happen, and the cause turned out to be my user error. Has anything been changed at all recently?

[SARK devs]

A routing dial plan which is too "loose" can cause this kind of behavioue.

Usually this


Best

S

[gippsweb]

I thought it would be something I'd done, but the only changes have been the SME updates last week, had to reboot back to the old kernel as I couldn't find one for a kmdl for the current kernel.

I do have some weird stuff happening with sip not loading when asterisk starts, for some reason it wasn't able to parse the sip.conf file as the [general] line got dropped from the file...

I hadn't been overly worried about it as we've been putting together a new box as this ones getting a tad slow....

[gippsweb]

Going on what you said Jeff, I've deleted the route that the call was looping out on.

It now gives the following;
chan_sip.c:14035 handle_request_invite: Call from '0094*00002' to extension '0341024269' rejected because extension not found.

I have trunk_Did_groups created for both the numbers and it has been working until the last couple of days.

When I recreated the did group for each of them I noticed a new option 'smartlink to extension' what does that do Jeff?

[SARK devs]

It's for big DiD ranges.  If you turn it on when you create a DiD range - say for example we had a range - 453600-453699, it will attempt to map the last 3 or 4 digits  (depending upon the number of digits you have chosen for your extensions) of each DiD it creates to an extension number.  If it gets a match it will automatically set up the routing to the extension for you.  Saves a lot of time on big site rollouts where the DiDs match the extensions.

Coming back to your problem...

Have you changed anything on your phone?   It genuinely looks like it is placing the same call over and over again. 

Also, the sequence in dalplan doesn't look quite correct.  It should start at internal.1 and then go to default.  Yours appears to go direct to default.  Check the stanzas on the SIP peers and make sure that a context=default hasn't crept in somewhere in one of your definitions.

Best

S   



 

[gippsweb]

 It genuinely looks like it is placing the same call over and over again.

Thats basically what is happening. A call is made to my DiD, sail is then trying to call the DiD from inside instead of passing it to an extension. This keeps going until it runs out of sip lines to use.

I see what you mean about the default context popping up in there. Also doesn't the 'h' signal a hangup? I've checked all the contexts in the trunks and can't find anything astray.

We're only running 2 sip trunks and they both have context=mainmenu in the right side box.

[SARK devs]

can you send a copy of sip,conf to admin@aelintra.com please?

Best

S

[gippsweb]

on it's way.

Mark

[SARK devs]

traced to sip.conf header record missing from selintra database.  Record replaced, system receiving calls normally.

S

[gippsweb]

Where would we be without you Jeff...

Don't know how that would have happened, but am glad it's now sorted.

Cheers

Mark

PS. I think I owe you a beer or 2

[David Harper]

Ha, if Jeff got all the beers he was owed he'd be drunk for a year!