[asterisk-users] [SOLVED]PRI issue its BUSY

satish patel satish_lx at hotmail.com
Mon Jun 6 22:08:12 CDT 2011



Solution:
pridialplan=unknow 

From: satish_lx at hotmail.com
To: asterisk-users at lists.digium.com
Date: Tue, 7 Jun 2011 02:33:44 +0000
Subject: Re: [asterisk-users] PRI issue its BUSY








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 		 	   		  

--
_____________________________________________________________________
-- 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/20110607/54c9b7cf/attachment.htm>


More information about the asterisk-users mailing list