Re: [asterisk-users] Asterisk DIAL() premature timeout on a PRI trunk to legacy PBX

--- Tzafrir Cohen <tzafrir.cohen@xorcom.com> wrote:

> You can probably get an answer to that if you enable
> and log debug
> messages of Asterisk .

Thanks, I thought that a pri debug was enough but now
I have the missing information:

Sep 16 12:37:28 VERBOSE[19175] logger.c: --
Zap/1-1 is ringing
Sep 16 12:37:32 DEBUG[9060] chan_sip.c: Auto
destroying call
'1457599B12DC407D91479E2D1EEB51960xc0a8fe05'
Sep 16 12:37:46 DEBUG[19175] dsp.c: ast_dsp_busydetect
detected busy, avgtone: 1525, avgsilence 3000
Sep 16 12:37:46 DEBUG[19175] dsp.c: Requesting Hangup
because the busy tone was detected on channel Zap/1-1
Sep 16 12:37:46 VERBOSE[19175] logger.c: --
Zap/1-1 is busy

So I guess it's clear enough: the Alcatel is the first
party to force the connection to be dropped because it
issues a busy tone. Is that right?

Can I honestly say that it's an Alcatel issue and that
nothing can be done on the Asterisk side?

-Vieri

____________________________________________________________________________________
Take the Internet to Go: Yahoo!Go puts the Internet in your pocket: mail, news, photos & more.

http://mobile.yahoo.com/go?refer=1GNXIC

_______________________________________________

Sign up now for AstriCon 2007! September 25-28th.

http://www.astricon.net/

--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:

http://lists.digium.com/mailman/listinfo/asterisk-users