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Hi there,
I'm now the proud owner of an X100P and am struggling to set up a
CVS-compiled
Asterisk to do my bidding. I checked zaptel/zapata/asterisk out today and
pretty much did a straight make install on all packages.
So far the only consistent trick I can make it perform is calling from one SIP
phone to another. Could I get a bit of feedback on my configuration files to
see where I'm going wrong? I'm focusing on simply trying to make our
two
internal Grandstream phones call out correctly through the Zaptel interface
or Voicepulse depending on a prefix digit.
I've had it route calls over Voicepulse with an alternatively bodged
configuration, but can't get it to repeat the trick. The call now seem to
time out despite the fact that I can see plenty of IP traffic flow to and
from Voicepulse's server when I try a US number with the prefix.
Likewise I've managed to dial out on our office line through the Zap/1
channel
which makes my mobile ring for a split second but it cuts out almost
immediately. Any ideas why it won't let the phone ring and connect the
call?
The final mystery is when I get it to execute Background() to play a gsm
sonud, the asterisk debug output says that it can't find the file but from
an
strace report, it hasn't tried to even open a file! i.e.
WARNING[229391]: File file.c, Line 446 (ast_openstream): File bytemark/welcome
does not exist in any format
WARNING[229391]: File file.c, Line 734 (ast_streamfile): Unable to open
bytemark/welcome (format ULAW): No such file or directory
Why does it think it can't find a file when the process doesn't seem to
have
tried? This happens for the demo sounds too.
My extensions.conf reads like this:
[general]
static=yes
writeprotect=no
[globals]
PSTN=Zap/1
IAXCOM_AUTHVOICEPULSE_AUTHVOICEPULSE_LOGIN
[direct_extensions]
; these seem to work 100% of the time, hooray!
exten => 201,1,Dial(SIP/matthew)
exten => 202,1,Dial(SIP/pete)
[incoming]
;
; Initial answer and welcome, set default values for incoming call
;
include => navigation
exten => s,1,Answer
exten => s,2,DigitTimeout,5
exten => s,3,ResponseTimeout,20
exten => s,4,BackGround(bytemark/welcome)
exten => s,5,Goto(mainmenu,s,1)
[mainmenu]
;
; Follows on from cheery welcome, or navigated back to from exiting
; other parts of the system.
;
include => navigation
exten => s,1,BackGround(bytemark/mainmenu)
[navigation]
;
; This is included from most context to allow people to jump anywhere they
; want any time they want.
;
; 0 for inquiries
; 1 for support message
; 2 for sales
; 3 for support person
;
exten => 0,1,Goto(801,1)
exten => 1,1,Goto(support,s,1)
exten => 2,1,Goto(801,1)
exten => 3,1,Goto(801,1)
include => direct_extensions
[support]
; play caller the current support information, let them jump out at
; any point
include => navigation
exten => s,1,BackGround(bytemark/support_current)
exten => s,2,BackGround(bytemark/support_next)
exten => s,3,Goto(mainmenu,s,1)
[internal]
;
; Office user has picked up phone
;
; * tests incoming call context
;
exten => *,1,Goto(incoming,s,1)
include => direct_extensions
include => dialout_all
[dialout_all]
include => dialout_voicepulse
include => dialout_ptsn
include => dialout_iaxtel
[dialout_ptsn]
exten => _9X.,1,Dial,Zap/1/${EXTEN:1}
[dialout_iaxtel]
exten => _8X.,1,Dial,IAX2/${IAXINFO}@iaxtel/${EXTEN:1}@iaxtel
[dialout_voicepulse]
exten => _7X.,1,Dial,IAX2/${VOICEPULSE_LOGIN}@voicepulse/${EXTEN:1}
My minimal zapata.conf reads like this:
[channels]
signalling=fxs_ks
context=incoming
Is this correct for a UK phone line? ztcfg -vv completes without error, so I
believe the driver module is set up correctly.
Finally is there a way for me to see Asterisk's train of thought as it
considers and follows/rejects each extension?
I've scoured docs on digium's site and the voip-info.org wiki but not
found
any easy answers the above questions!
thanks,
- --
Matthew Bloch Bytemark Hosting
tel. +44 (0) 8707 455026
http://www.bytemark-hosting.co.uk/
Dedicated Linux hosts from 15ukp ($26) per month
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