Fast busy on T1, can't connect via web

[ksalter]

I just installed a PIAF with a TE110P T1 card. Inbound calls to the T1 run a custom context where I send the call to out a SIP trunk.

This will work fine for a while. Then, out of the blue, I will dial in and get fast busy.

When I connect via ssh to asterisk, it indicates the T1 is OK. However, a zap show channels shows a "stuck" call on one of the channels. There really isn't a call. I also notice that I can't connect using FreePBX web app. Also, when I'm in the CLI, I cannot stop or shutdown asterisk.

In order to clear this up, I have to kill the asterisk process, then unload zaptel, reload zaptel, then restart asterisk.

Anyone with ideas? Right now I'm having to check every hour to make sure this is working...

 

[ednunnemaker]

If i were to guess

If I were to make a guess, this sounds very similar to a no disconnect issue. We used to have all kinds of problems with VSR voice mails not disconnecting calls when the customer had a T-1, I'm not sure in your case, it may be the sip provider not providing the right disconnect signal so your box knows the call has been disconnected, since the sip call is connected to the T-1 call it may be locking that channel out. there is setting somewhere that lets you set the disconnect timer. I don't know if that really helps but it may lead you down the right road.

 

[ksalter]

If I were to make a guess, this sounds very similar to a no disconnect issue. We used to have all kinds of problems with VSR voice mails not disconnecting calls when the customer had a T-1, I'm not sure in your case, it may be the sip provider not providing the right disconnect signal so your box knows the call has been disconnected, since the sip call is connected to the T-1 call it may be locking that channel out. there is setting somewhere that lets you set the disconnect timer. I don't know if that really helps but it may lead you down the right road.

Thanks for the reply. I'll check out the sip issue, but this doesn't happen all the time. Once every several hours or so. The sip provider in this case is an application I wrote. It works with other sip providers (like Vitelity) with no problem, so I couldn't imagine what would be different with Asterisk.

Even if a call was left in limbo, why would that lock up the whole T1 driver and Asterisk to the point I have to kill everything to get it back up?

I am using DMS100 switch type on my Digium single T1 card. Any known issues?

I googled for this disconnect timer but couldn't find any info. Does anyone know what this may be?

 

[ksalter]

More info:

It appears to hang here:

Accepting call from 'xxxxxxxxxx' to 'xxxxxxx' on channel 0/1, span 1
-- Executing [xxxxxxx@from-zaptel:1] Set("Zap/1-1", "DID=xxxxxxx") in new stack
-- Executing [xxxxxxx@from-zaptel:2] Goto("Zap/1-1", "s|1") in new stack
-- Goto (from-zaptel,s,1)
-- Executing [s@from-zaptel:1] NoOp("Zap/1-1", "Entering from-zaptel with DID == xxxxxxx") in new stack
-- Executing [s@from-zaptel:2] Ringing("Zap/1-1", "") in new stack

Like I said, this only happens occasionally. The last time this happened, there were no other calls in progress.
I have three other asterisk pbx (all trixbox) and I don't have this problem at all.

 

[ksalter]

Nevermind, it just hung up somewhere else.

I am running on a Dell 1650. Is there an issue with running Asterisk on that computer?

 

[ednunnemaker]

Sorry

I was just throwing that out there, we use a PiaF server in our office with a PRI and 4 SIP trunks from Les.net, we do trunk to trunk transfers all the time with no issues at all, our server has been up since January 3rd with no issues, it could be a hardware issue.