[Asterisk-Users] Newbie X100P Clone question
I've got an X100P, I'm brand new to Asterisk. I've been able to set up
SIP extensions and have them working, now I've added the X100P in so I
can drop a line in and eventually be my outside world connection.
I've downloaded the zaptel code via CVS, and configured it up pretty
much exactly like
http://www.digium.com/index.php?menu=configuration#X100P by adding items
into my configurations. My current extensions.conf looks like this:
[general]
static=yes
writeprotect=yes
[bogon-calls]
exten = _.,1,Congestion
[default]
exten = _XX,1,Dial,Zap/1/${EXTEN} ; Press any 7 digit number and
try to dial that number through Zap channel 1
exten = s,1,Wait(1)
exten = s,2,Answer
exten = s,2,Playback(demo-congrats) ; Plays the demo-congrats file
after answering the line
exten = s,3,Hangup
[from-sip]
exten = 3073,1,Dial(SIP/3073,20)
exten = 3073,2,Voicemail(u3073)
exten = 3073,102,Voicemail(b3073)
exten = 3073,103,Hangup
exten = 3087,1,Dial(SIP/3087,20)
exten = 3087,2,Voicemail(u3073)
exten = 3087,102,Voicemail(b3073)
exten = 3087,103,Hangup
exten = 3089,1,Dial(SIP/3089,20)
exten = 3089,2,Voicemail(u3089)
exten = 3089,102,Voicemail(b3089)
exten = 3089,103,Hangup
exten = 3123,1,VoicemailMain(${CALLERIDNUM})
Here's zapata.conf:
[trunkgroups]
[channels]
context=default
switchtype=national
signalling=fxs_ks
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
echotraining=400
rxgain=0.0
txgain=0.0
callgroup=1
pickupgroup=1
immediate=no
busydetect=no
callprogress=no
callerid=asreceived
group=1
context=default
channel = 1
An inbound call to the extension doesn't play back the congrats demo
gsm recording. Running asterisk with -gc I get the following upon
dial in:
*CLI -- Starting simple switch on 'Zap/1-1'
-- Executing Wait(Zap/1-1, 1) in new stack
-- Executing Answer(Zap/1-1, ) in new stack
-- Executing Hangup(Zap/1-1, ) in new stack
== Spawn extension (default, s, 3) exited non-zero on 'Zap/1-1'
-- Hungup 'Zap/1-1'
-- Starting simple switch on 'Zap/1-1'
Nov 5 14:54:59 WARNING[1967]: chan_zap.c:5466 ss_thread: CallerID
returned with error on channel 'Zap/1-1'
-- Executing Wait(Zap/1-1, 1) in new stack
-- Executing Answer(Zap/1-1, ) in new stack
-- Executing Hangup(Zap/1-1, ) in new stack
== Spawn extension (default, s, 3) exited non-zero on 'Zap/1-1'
-- Hungup 'Zap/1-1'
What's up with the exited non-zero on the spawn extension?
Also, whenever starting Asterisk I always get this about 10 seconds
after init:
Nov 5 14:54:45 NOTICE[1958]: pbx_dundi.c:2841 destroy_trans: Peer
'00:50:8b:f3:75:bb' has become UNREACHABLE!
What does that mean?
Thanks very much in advance. This setup is very very interesting when
compared to our current production Interactive Intelligence CIC
system
Steve
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Newbie X100P Clone question
On Fri, 2004-11-05 at 15:58, Steve Frank wrote:
I've got an X100P, I'm brand new to Asterisk. I've been able to set up
SIP extensions and have them working, now I've added the X100P in so I
can drop a line in and eventually be my outside world connection.
I've downloaded the zaptel code via CVS, and configured it up pretty
much exactly like
http://www.digium.com/index.php?menu=configuration#X100P by adding items
into my configurations. My current extensions.conf looks like this:
[general]
static=yes
writeprotect=yes
[bogon-calls]
exten = _.,1,Congestion
[default]
exten = _XX,1,Dial,Zap/1/${EXTEN} ; Press any 7 digit number and
try to dial that number through Zap channel 1
exten = s,1,Wait(1)
exten = s,2,Answer
exten = s,2,Playback(demo-congrats) ; Plays the demo-congrats file
after answering the line
You've numbered both lines as priority 2. Fix that and you'll be fine.
-Seth
exten = s,3,Hangup
[from-sip]
exten = 3073,1,Dial(SIP/3073,20)
exten = 3073,2,Voicemail(u3073)
exten = 3073,102,Voicemail(b3073)
exten = 3073,103,Hangup
exten = 3087,1,Dial(SIP/3087,20)
exten = 3087,2,Voicemail(u3073)
exten = 3087,102,Voicemail(b3073)
exten = 3087,103,Hangup
exten = 3089,1,Dial(SIP/3089,20)
exten = 3089,2,Voicemail(u3089)
exten = 3089,102,Voicemail(b3089)
exten = 3089,103,Hangup
exten = 3123,1,VoicemailMain(${CALLERIDNUM})
Here's zapata.conf:
[trunkgroups]
[channels]
context=default
switchtype=national
signalling=fxs_ks
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
echotraining=400
rxgain=0.0
txgain=0.0
callgroup=1
pickupgroup=1
immediate=no
busydetect=no
callprogress=no
callerid=asreceived
group=1
context=default
channel = 1
An inbound call to the extension doesn't play back the congrats demo
gsm recording. Running asterisk with -gc I get the following upon
dial in:
*CLI -- Starting simple switch on 'Zap/1-1'
-- Executing Wait(Zap/1-1, 1) in new stack
-- Executing Answer(Zap/1-1, ) in new stack
-- Executing Hangup(Zap/1-1, ) in new stack
== Spawn extension (default, s, 3) exited non-zero on 'Zap/1-1'
-- Hungup 'Zap/1-1'
-- Starting simple switch on 'Zap/1-1'
Nov 5 14:54:59 WARNING[1967]: chan_zap.c:5466 ss_thread: CallerID
returned with error on channel 'Zap/1-1'
-- Executing Wait(Zap/1-1, 1) in new stack
-- Executing Answer(Zap/1-1, ) in new stack
-- Executing Hangup(Zap/1-1, ) in new stack
== Spawn extension (default, s, 3) exited non-zero on 'Zap/1-1'
-- Hungup 'Zap/1-1'
What's up with the exited non-zero on the spawn extension?
Also, whenever starting Asterisk I always get this about 10 seconds
after init:
Nov 5 14:54:45 NOTICE[1958]: pbx_dundi.c:2841 destroy_trans: Peer
'00:50:8b:f3:75:bb' has become UNREACHABLE!
What does that mean?
Thanks very much in advance. This setup is very very interesting when
compared to our current production Interactive Intelligence CIC
system
Steve
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Seth Remington
SaberLogic, LLC
661-B Weber Drive
Wadsworth, Ohio 44281
Phone: (330)335-6442
Fax: (330)336-8559
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Newbie X100P Clone question
[default]
exten = _XX,1,Dial,Zap/1/${EXTEN} ; Press any 7 digit
number and
try to dial that number through Zap channel 1 exten = s,1,Wait(1)
exten = s,2,Answer exten = s,2,Playback(demo-congrats) ;
Plays the
demo-congrats file after answering the line
You've numbered both lines as priority 2. Fix that and you'll be fine.
-Seth
Sweet! Thanks Seth, that did it. I guess the Digium page needs to be
updated to fix this error.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
