[asterisk-users] How to get correct dial result for outgoing calls thru ISDN?

2008-11-12 Thread Daniel - Asterisk 
Hi everyone,

Currectly I'm having some troubles to get correct status of my calls throug
ISDN lines, when outbound calls don't get its destination I always receive
NO ANSWER as ${DIALSTATUS}  despite the fact I know the target number
doesn't exists or is busy at that time.

Maybe there is something I must change in my zaptel.conf or zapata.conf,
current configs follows:

 *zaptel.conf*
# Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 (MASTER)
span=1,1,0,ccs,hdb3
# termtype: te
bchan=1-15,17-31
dchan=16

 *zapata.conf*
 [channels]
language=en
context=default
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=no
threewaycalling=yes
transfer=yes
canpark=no
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.5
txgain=0.0
immediate=yes
busydetect=yes
busycount=4
busypattern=500,500
answeronpolarityswitch=yes
hanguponpolarityswitch=yes

;*groups*
group=2
context=incoming
immediate=no
switchtype=euroisdn
signalling=pri_cpe
channel=32-46,48-62

 *extensions.conf*

exten = _9.,n,Dial(Zap/g2/${EXTEN:1})

Asterisk version is: 1.4.21.1 and I'm using a Digium TE420P card to connect
to ISDN.

Thanks in advance,

Daniel Arohuanca
+51 1 994149553
___
-- 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

Re: [asterisk-users] How to get correct dial result for outgoing calls thru ISDN?

2008-11-12 Thread Eric ManxPower Wieling 
Use ${HANGUPCAUSE} which provides a Q.931 hangup cause.  chan_sip and 
chan_iax (maybe other channels) translate the protocol specific causes 
to a Q.931 hangup cause.

I would recommend you check the doc directory in the Asterisk source 
code for channelvariables.txt, but for some reason that I can't figure 
out most of those docs where translated into TeX format in 1.6.  Maybe 
the Wiki has a text version.

Daniel - Asterisk wrote:
 Hi everyone,
 
 Currectly I'm having some troubles to get correct status of my calls 
 throug ISDN lines, when outbound calls don't get its destination I 
 always receive NO ANSWER as ${DIALSTATUS}  despite the fact I know the 
 target number doesn't exists or is busy at that time.
 
 Maybe there is something I must change in my zaptel.conf or zapata.conf, 


-- 
Consulting and design services for LAN, WAN, voice and data.  Based near 
Birmingham, AL.  Now accepting clients worldwide. Contact me for Tellabs 
echo canceling systems.  Also see http://www.fnords.org/skillslist.html

___
-- 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

Re: [asterisk-users] How to get correct dial result for outgoing calls thru ISDN?

2008-11-12 Thread Eric ManxPower Wieling 


Tilghman Lesher wrote:
 On Wednesday 12 November 2008 23:15:01 Eric ManxPower Wieling wrote:
 Use ${HANGUPCAUSE} which provides a Q.931 hangup cause.  chan_sip and
 chan_iax (maybe other channels) translate the protocol specific causes
 to a Q.931 hangup cause.

 I would recommend you check the doc directory in the Asterisk source
 code for channelvariables.txt, but for some reason that I can't figure
 out most of those docs where translated into TeX format in 1.6.  Maybe
 the Wiki has a text version.
 
 If you 'make asterisk.pdf', the LaTeX files format nicely into a singly
 contained PDF document.

Don't you need LaTeX installed for that?

-- 
Consulting and design services for LAN, WAN, voice and data.  Based near 
Birmingham, AL.  Now accepting clients worldwide. Contact me for Tellabs 
echo canceling systems.  Also see http://www.fnords.org/skillslist.html

___
-- 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

Re: [asterisk-users] How to get correct dial result for outgoing calls thru ISDN?

2008-11-12 Thread Tilghman Lesher 
On Wednesday 12 November 2008 23:44:49 Eric ManxPower Wieling wrote:
 Tilghman Lesher wrote:
  On Wednesday 12 November 2008 23:15:01 Eric ManxPower Wieling wrote:
  Use ${HANGUPCAUSE} which provides a Q.931 hangup cause.  chan_sip and
  chan_iax (maybe other channels) translate the protocol specific causes
  to a Q.931 hangup cause.
 
  I would recommend you check the doc directory in the Asterisk source
  code for channelvariables.txt, but for some reason that I can't figure
  out most of those docs where translated into TeX format in 1.6.  Maybe
  the Wiki has a text version.
 
  If you 'make asterisk.pdf', the LaTeX files format nicely into a singly
  contained PDF document.

 Don't you need LaTeX installed for that?

You do, but the PDF is already built in the release tarballs.  Only people
running out of SVN would need the corresponding LaTeX libraries to build
the PDF.

-- 
Tilghman

___
-- 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