[asterisk-users] PRI issue its BUSY
satish patel
satish_lx at hotmail.com
Mon Jun 6 21:33:44 CDT 2011
This is wired..
If i connect my old asterisk 1.2 box my PRI working great! all inbound outbound calls.. But its not working with asterisk 1.8 :( ( i can call in but not out)
From: satish_lx at hotmail.com
To: asterisk-users at lists.digium.com
Date: Tue, 7 Jun 2011 02:11:28 +0000
Subject: Re: [asterisk-users] PRI issue its BUSY
sometime i am getting Span 1: Channel 0/23 got hangup request, cause 16 but my call doesn't get completed
== Primary D-Channel on span 1 up
-- Restart requested on entire span 1
== Using SIP RTP CoS mark 5
-- Executing [7076941815 at from-sip:1] Dial("SIP/7328-00000004", "DAHDI/G1/17076941815") in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called DAHDI/G1/17076941815
-- DAHDI/i1/17076941815-4 is proceeding passing it to SIP/7328-00000004
-- DAHDI/i1/17076941815-4 is ringing
-- DAHDI/i1/17076941815-4 is making progress passing it to SIP/7328-00000004
-- DAHDI/i1/17076941815-4 answered SIP/7328-00000004
-- Span 1: Channel 0/23 got hangup request, cause 16
-- Executing [h at from-sip:1] Hangup("SIP/7328-00000004", "") in new stack
== Spawn extension (from-sip, h, 1) exited non-zero on 'SIP/7328-00000004'
From: caryf at usawide.net
To: asterisk-users at lists.digium.com
Date: Mon, 6 Jun 2011 20:24:06 -0500
Subject: Re: [asterisk-users] PRI issue its BUSY
From:
asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of satish patel
Sent: Monday, June 06, 2011 8:20
PM
To: asterisk-users
Subject: [asterisk-users] PRI
issue its BUSY
Hi all,
I just configures my PRI and incoming calls are working fine but outside
calling giving error PRI is BUSY :( any idea ? I have same setup on
other box and that boxes works perfect.
-- DAHDI/i1/6463279153-2 is proceeding passing it to SIP/7328-00000002
-- DAHDI/i1/6463279153-2 is making progress passing it to
SIP/7328-00000002
-- DAHDI/i1/6463279153-2 is busy
-- Hungup 'DAHDI/i1/6463279153-2'
== Everyone is busy/congested at this time (1:1/0/0)
-- Auto fallthrough, channel 'SIP/7328-00000002' status is
'BUSY'
Maybe
the problem is external to the box.
Try
swapping PRIs briefly for testing.
C.
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110607/96cd37fd/attachment.htm>
More information about the asterisk-users
mailing list