503 Service Unavailable bug

[Dain]

eyeBeam Enhanced - Build 47739

I have local Asterisk which routes calls to different trunks, eyeBeam registered only at local Asterisk. Here's the problem with latest release.

After call to extension with currently not working trunk (after Asterisk returned 503 Service Unavailable message) eyeBeam stops working for about 95-100 seconds - all calls to all extensions, no matter local or remote returned by eyeBeam as Service Unavailable, however there's no mention of these calls in Asterisk logs at all, even no attempts to place a call.

I've tried number of softphones including X-Lite, X-Pro and number of others - all of them work and eyeBeam accepts calls, but can't place calls within these 95-100 seconds. Then everything returns to normal until next 503 error from Asterisk.

Could you please look into that ?

Update - just asked my pal with Bria to test same setup - exactly the same behaviour. Hmm... same codebase and different skins ?

[dontosh]

Newbie with VERY limited computer skill,searched this forum re Error 503
somehow I clicked on "speakerphone" Error disappeared. If this comment is too elementary, Please excuse.

[juvs]

Any solution for this issue, we are experience the same problem, and is urgent to get fix! we are in production right now and get this error.

Here's our scenario:

When we try to call using our E1 connection from our telco and for some reason the channel is busy or in congestion we get the following message from AsterisK

[Oct 1 14:56:40] NOTICE[14571]: chan_unicall.c:2790 handle_uc_event: CRN 32773 - far disconnected cause=Switching equipment congestion [42]
-- Channel 0 got hangup
-- UniCall/1-1 is circuit-busy
-- Hungup 'UniCall/1-1'
== Everyone is busy/congested at this time (1:0/1/0)
== Auto fallthrough, channel 'SIP/6000-b6403a50' status is 'CONGESTION'

Then the SIP Phone lost their registration an only works once we restart eyeBeam.

Regards!

[juvs]

Any solution for this issue, we are experience the same problem, and is urgent to get fix! we are in production right now and get this error.

Here's our scenario:

When we try to call using our E1 connection from our telco and for some reason the channel is busy or in congestion we get the following message from AsterisK

[Oct 1 14:56:40] NOTICE[14571]: chan_unicall.c:2790 handle_uc_event: CRN 32773 - far disconnected cause=Switching equipment congestion [42]
-- Channel 0 got hangup
-- UniCall/1-1 is circuit-busy
-- Hungup 'UniCall/1-1'
== Everyone is busy/congested at this time (1:0/1/0)
== Auto fallthrough, channel 'SIP/6000-b6403a50' status is 'CONGESTION'

The message we get on eyeBean is "Call failed: Service Unavailable" and everything gets disabled (I mean buttons).

Then the SIP Phone lost their registration an only works once we restart eyeBeam.


Regards!

[kegler]

I am seeing the same problem. A 503 Service Unavailable results in the phone being unable to dial out until a new registration takes place. If you manually re-register the phone, it will work again until the 503 occurs again.

The worst part about this behavior is that the phone will not even dial out to other known working extensions during this period.

[Derek Jacobs]

This is a known issue.
The workaround at this time is to set your server to not send a 503.

[jackryan]

[quote="Derek Jacobs"]This is a known issue.
The workaround at this time is to set your server to not send a 503.[/quote]

1.5.19.4 installed, this problem is still there. How do I tell Asterisk to not send a 503?

[Derek Jacobs]

I am not aware of how to make changes to asterisk. I'm sorry.

[squiffy]

Hi,
any idea when this will be fixed?

If I upgrade, Is it fixed in the pay version (Bria?)

Thanks.

[squiffy]

One way to control what Asterisk returns is to ensure any wrong numbers are handled by your dial plan instead of letting Asterisk deal with it internally.

You'll need a "catch-all" entry, such as

exten=>_X.,1,Busy

This will return busy instead of unavailable, or you could do a Hangup(x) where x is the clear cause you require (see www dot voip-info dot org/wiki/view/Asterisk+cmd+hangup for a list of clear causes you can use - I know it is for PRI but they seem to work on VoIP, too).

Just a thought ...

[pnaves]

Witch file should I put this command to change return error? I have the same issue in production environment .