[Asterisk-Users] NEWBIE HELP : chan_zap.c: Exception on 16, channel 1, call not being picked up on incoming X1-100P zap
I am running [EMAIL PROTECTED] ( asterisk 1.2beta1 with two X100P cards ) on
centos 4.1 box with a 2.6.12 kernel.
I ran genzaptelconf
and added two trunks for each of the devices however the incoming calls
when I ring just get ignored.
asterisk -r tells me that it just gets hangupcall, and in the the log
files I see exceptions.
I am running asterisk 1.2 beta. Can someone help as to how to debug this
I am new to the asterisk game so any hints would be greatfully received.
== Manager 'admin' logged on from 127.0.0.1
-- Starting simple switch on 'Zap/1-1'
-- Executing Macro("Zap/1-1", "hangupcall") in new stack
-- Executing ResetCDR("Zap/1-1", "w") in new stack
-- Executing NoCDR("Zap/1-1", "") in new stack
-- Executing Wait("Zap/1-1", "5") in new stack
-- Executing Hangup("Zap/1-1", "") in new stack
== Spawn extension (macro-hangupcall, s, 4) exited non-zero on
'Zap/1-1' in macro 'hangupcall'
== Spawn extension (from-internal, s, 1) exited non-zero on 'Zap/1-1'
-- Executing Macro("Zap/1-1", "hangupcall") in new stack
-- Executing ResetCDR("Zap/1-1", "w") in new stack
-- Executing NoCDR("Zap/1-1", "") in new stack
-- Executing Wait("Zap/1-1", "5") in new stack
-- Executing Hangup("Zap/1-1", "") in new stack
== Spawn extension (macro-hangupcall, s, 4) exited non-zero on
'Zap/1-1' in macro 'hangupcall'
== Spawn extension (from-internal, h, 1) exited non-zero on 'Zap/1-1'
Oct 19 19:01:58 VERBOSE[10782] logger.c: -- Starting simple switch
on 'Zap/1-1'
[EMAIL PROTECTED] ~]# Oct 19 19:02:03 NOTICE[10782] chan_zap.c: Got event
18 (Event 18)...
Oct 19 19:02:03 VERBOSE[10782] logger.c: -- Executing
Macro("Zap/1-1", "hangupcall") in new stack
Oct 19 19:02:03 VERBOSE[10782] logger.c: -- Executing
ResetCDR("Zap/1-1", "w") in new stack
Oct 19 19:02:03 DEBUG[10782] cdr_addon_mysql.c: cdr_mysql: inserting a
CDR record.
Oct 19 19:02:03 DEBUG[10782] cdr_addon_mysql.c: cdr_mysql: SQL command
as follows: INSERT INTO cdr
(calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode)
VALUES ('2005-10-19 19:02:03','\"device\"
<400>','400','s','from-internal', 'Zap/1-1','','ResetCDR','w',0,0,'NO
ANSWER',3,'')
Oct 19 19:02:03 VERBOSE[10782] logger.c: -- Executing
NoCDR("Zap/1-1", "") in new stack
Oct 19 19:02:03 WARNING[10782] cdr.c: CDR on channel 'Zap/1-1' not posted
Oct 19 19:02:03 WARNING[10782] cdr.c: CDR on channel 'Zap/1-1' lacks end
Oct 19 19:02:03 VERBOSE[10782] logger.c: -- Executing
Wait("Zap/1-1", "5") in new stack
Oct 19 19:02:03 DEBUG[10782] chan_zap.c: Exception on 16, channel 1
Oct 19 19:02:03 DEBUG[10782] chan_zap.c: Got event Ring/Answered(2) on
channel 1 (index 0)
Oct 19 19:02:03 DEBUG[10782] chan_zap.c: Setting IDLE polarity due to
ring. Old polarity was 0
Oct 19 19:02:08 DEBUG[10782] chan_zap.c: Exception on 16, channel 1
Oct 19 19:02:08 DEBUG[10782] chan_zap.c: Got event Event 18(18) on
channel 1 (index 0)
Oct 19 19:02:08 DEBUG[10782] chan_zap.c: Dunno what to do with event 18
on channel 1
Oct 19 19:02:08 DEBUG[10782] acl.c: # Testing 192.168.0.108 with
192.168.0.0
Oct 19 19:02:08 NOTICE[10782] chan_sip.c: Registration from 'new
' failed for '192.168.0.108' - Wrong password
Oct 19 19:02:08 VERBOSE[10782] logger.c: -- Executing
Hangup("Zap/1-1", "") in new stack
Oct 19 19:02:08 VERBOSE[10782] logger.c: == Spawn extension
(macro-hangupcall, s, 4) exited non-zero on 'Zap/1-1' in macro 'hangupcall'
Oct 19 19:02:08 VERBOSE[10782] logger.c: == Spawn extension
(from-internal, s, 1) exited non-zero on 'Zap/1-1'
Oct 19 19:02:08 VERBOSE[10782] logger.c: -- Executing
Macro("Zap/1-1", "hangupcall") in new stack
Oct 19 19:02:08 VERBOSE[10782] logger.c: -- Executing
ResetCDR("Zap/1-1", "w") in new stack
Oct 19 19:02:08 VERBOSE[10782] logger.c: -- Executing
NoCDR("Zap/1-1", "") in new stack
Oct 19 19:02:08 VERBOSE[10782] logger.c: -- Executing
Wait("Zap/1-1", "5") in new stack
Oct 19 19:02:08 DEBUG[10782] chan_zap.c: Exception on 16, channel 1
Oct 19 19:02:08 DEBUG[10782] chan_zap.c: Got event Ring/Answered(2) on
channel 1 (index 0)
Oct 19 19:02:08 DEBUG[10782] chan_zap.c: Setting IDLE polarity due to
ring. Old polarity was 0
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[Asterisk-Users] Newbie Help - Auto Fallthrough
I am a serious Asterisk newbie: just installed asterisk last week and it is
now running with our Voicetronix OpenLine4 hardware.
All is working as expected with one exception, in the following sequence
(extracted from my extensions.conf file):
[GetConfirmation]
exten => s,n,SetVar(TimeOut=0) ; if timeout and TimeOut=1 then user
already timed out once, so hangup
exten => s,n,SetVar(State=GetConfirmation) ; set up for
time-out return
exten => s,n,ResponseTimeout(10); Set
Response Timeout to 10 seconds
exten => s,n,Background(mymenu}) ; play menu
msg (press 1..., press 2... press 3...), wait for response
include => TimeOut ; include
timeout handler
exten => 1,1,Goto(DoTransaction,s,1); submit
transaction
exten => 2,1,Goto(GetFare,s,1) ; send user
back to re-enter fare, tip and cab #
exten => 3,1,Goto(s,1) ; replay
confirmation msg
[TimeOut]
; user timed out, so see if TimeOut flag set. If set, then this is the
second time in a row that user has timed out
; so hang up. If not set, then set TimeOut and let user try again
; Usage: include => TimeOut (placed as last entry in any context
(routine) where timeout handling required)
exten => t,1,NoOp(In TimeOut: ${TimeOut})
exten => t,2,Gotoif($[${TimeOut}]?:5:3)
exten => t,3,SetVar(TimeOut=1) ; users first time-out so
set flag
exten => t,4,Goto(${State},s,1)); start current sub-routine
all over again
exten => t,5,Hangup ; users second time-out, so
hang up
In the above sequence (context) the menu message plays as expected, and as
long as the user enters a DTMF digit BEFORE the message playback completes,
all words as it should. However, if the message playback completes, there
is no 10 second wait for the user's entry, instead the call hangs up
immediately and the following debug info is displayed on the console:
-- Executing SetVar("vpb/1-1", "State=GetConfirmation") in new stack
-- Executing ResponseTimeout("vpb/1-1", "10") in new stack
-- Set Response Timeout to 10
-- Executing BackGround("vpb/1-1", "1-1") in new stack
-- Playing '1-1' (language 'taxi')
== Auto fallthrough, channel 'vpb/1-1' status is 'UNKNOWN'
== vpb/1-1: Hangup requested
== vpb/1-1: Ending record mode (1/yes)
> vpb/1-1: stopped record thread on vpb/1-1
== vpb/1-1: Ending play mode on vpb/1-1
> vpb/1-1: Setting state down
== vpb/1-1: Hangup complete
> Restarting monitor
> Trying to reawake monitor
> Monitor restarted
> Monitor got null event
> vpb/1-4: Event [12=>[03] Loop Drop]
> vpb/1-4: handle_notowned: mode=3, event[12][[03] Loop Drop
]=[0]
> vpb/1-4: handle_notowned: mode=3, [12=>0]
Can anyone tell me:
(1) why there is no 10 second wait time?
(2) why the TimeOut code did not execute?
Any and all ideas, comments, suggestions appreciated!
Jonathan
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RE: [Asterisk-Users] newbie: help two cisco phones (sip)
Have you set your DNS SRV entry for SIP correctly??? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Andrew White Sent: Tuesday, February 15, 2005 7:04 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] newbie: help two cisco phones (sip) Hi, I have two cisco phones with sip images and I am trying to configure to work with asterisk. Both can call demo numbers and voicemail etc. but can't call each other. sip show registry and sip show users both indicate that asterisk doesn't know the phones ip addresses, and when u try to place a call, it forwards to unanswered voicemail immediately. I have tried user_info: ip and also phone, but can't seem to get the phones to register. sip.conf has host=dynamic for both phones SIP image is version 7 anyone able to tell me where i'm going wrong ? tks Andrew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] newbie: help two cisco phones (sip)
Hi, I have two cisco phones with sip images and I am trying to configure to work with asterisk. Both can call demo numbers and voicemail etc. but can't call each other. sip show registry and sip show users both indicate that asterisk doesn't know the phones ip addresses, and when u try to place a call, it forwards to unanswered voicemail immediately. I have tried user_info: ip and also phone, but can't seem to get the phones to register. sip.conf has host=dynamic for both phones SIP image is version 7 anyone able to tell me where i'm going wrong ? tks Andrew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 help/pointers required - configure xlite with asterisk
> I just want one of my incoming numbers to go to an IVR service that will > allow me to select what I want. IVR is a key word. Try this: http://asteriskdocs.org There is a movie theatre example that will give you this. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 help/pointers required - configure xlite with asterisk
> I just want one of my incoming numbers to go to an IVR service that will > allow me to select what I want. > > For example > > "Press 1 for Mike, 2 for Karen, 3 for other, 9 for voicemail" etc just put your incoming line in a context where you have a "s" extension Something like this (not really good one, just wrote it from the top of my head) [incoming-menu] exten => s,1,Answer exten => s,2,Playback(welcome-message) exten => s,3,... look in the sample config that came with asterisk, you have samples in there And remember, google is your friend ;) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Newbie help/pointers required - configure xlite with asterisk
I just want one of my incoming numbers to go to an IVR service that will allow me to select what I want. For example "Press 1 for Mike, 2 for Karen, 3 for other, 9 for voicemail" etc > Just need to learn how to configure services now so that I can put a menu on > one of my numbers! Elaborate please, I'm not clear on "put a menu on one of my numbers". Give an example of what you want to accomplish and I'm sure many people here will help you. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.8.6 - Release Date: 07/02/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 help/pointers required - configure xlitewith asterisk
> Just need to learn how to configure services now so that I can put a menu on > one of my numbers! Elaborate please, I'm not clear on "put a menu on one of my numbers". Give an example of what you want to accomplish and I'm sure many people here will help you. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 help/pointers required - configure xlitewith asterisk
Cheers for that. I seem to have I working fine now. Just need to learn how to configure services now so that I can put a menu on one of my numbers! > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Wilson Pickett > The answers to the questions you've been asking are probably here: > Starter articles: > > http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html > http://www.onlamp.com/pub/a/onlamp/2004/01/22/asterisk2.html > -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.8.5 - Release Date: 03/02/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 help/pointers required - configure xlite with asterisk
The answers to the questions you've been asking are probably here: Starter articles: http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html http://www.onlamp.com/pub/a/onlamp/2004/01/22/asterisk2.html Full install etc. http://automated.it/guidetoasterisk.htm And of course: http://www.asteriskdocs.org ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Newbie help/pointers required - configure xlite with asterisk
I could use a few pointers to get this working please? I have asterisk installed on my linux server. It is setup to register with sipgate and works for incoming calls. I have xlite installed on my windows pc and this connects fine with the asterisk server and can get the incoming calls fine. Now I want to be able to make outboun calls from xlite via sipgate. I also need to be able to dial extensions on the asterisk server (for voicemail etc) Any pointers appreciated as I cannot see how the later bit (outgoing and extensions) works! Cheers -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.8.5 - Release Date: 03/02/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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) help please?
Hi,
i have the same problem in the last week with i4l on linux.
The solution is, that you modify the Dial string in the extension.conf:
from
Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
to
Dial(${TRUNK}:${EXTEN:${TRUNKMSD}})
I hope, this helps.
Regards,
Andreas.
--On Freitag, April 16, 2004 23:58:20 +0200 Mark Elkins <[EMAIL PROTECTED]>
wrote:
What I've got...
Software:
Linux: Slackware 9.1
Asterisk: out of CVS - so its new.
isdn4k-utils: to test the ISDN Card
Hardware:
PII Pentium 400Mhz (Its a test of concept machine) with 320Kb RAM
1 x ISDN BRI Card - DIVA EICON (Installed + working)
2 x Grandstream (Barbie?) BT100 SIP Phones.
What Works..
I can call from one phone to the other... get read voicemail...
I can dial from a PSTN phone the BRI Number - and get the * demo
messages
Whats been read..
Lots.. Andy's Getting Started (www.automated.it/guidetoasterisk.htm)
and lots from http://www.voip-info.org/wiki-Asterisk+ISDN4Linux and
I've followed almost every link from www.asterisk.org...
All examples seem to include Digiums hardware :-(
I'm looking for clean, clear examples with a generic ISDN card - which
is my trunk line, and the two SIP phones.
The numbering plan in South Africa is pretty simple
7 digits for local calls
12 digits for long distance
Anyone in S.A. got some example configs to share with?
Currently - I'm stuck with the message..
-- Executing Dial("SIP/phone1-082a", "Modem/g1/8070590") in new stack
Apr 17 00:09:00 WARNING[507919]: chan_modem.c:181 modem_call:
Destination g1/8070590 requres a real destination (device:destination)
-- Couldn't call g1/8070590
-- Hungup 'Modem[i4l]/ttyI1'
... when I dial '98070590' (9 for outside - which I'll make '0' one
day!)
(its late, head hurts, wife is loosing patience)
help? hints?
--
. . ___. .__ Posix Systems - Sth Africa
/| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE
/ |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496
--
"Ich denke, man hat kein Recht, andere zu kontrollieren oder Ihnen etwas
aufzuzwingen, den eigenen Glauben oder die eigene Art zu leben."
- Dalai Lama "Begegnungen".
---
Andreas Czerniak <[EMAIL PROTECTED]> - Kiel - FRG - Fax:+49-431-2000447
PGPkey: http://wwwkeys.nl.pgp.net:11371/pks/lookup?op=get&search=0xEDB224EC
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[Asterisk-Users] (Newbie) help please?
What I've got...
Software:
Linux: Slackware 9.1
Asterisk: out of CVS - so its new.
isdn4k-utils: to test the ISDN Card
Hardware:
PII Pentium 400Mhz (Its a test of concept machine) with 320Kb RAM
1 x ISDN BRI Card - DIVA EICON (Installed + working)
2 x Grandstream (Barbie?) BT100 SIP Phones.
What Works..
I can call from one phone to the other... get read voicemail...
I can dial from a PSTN phone the BRI Number - and get the * demo
messages
Whats been read..
Lots.. Andy's Getting Started (www.automated.it/guidetoasterisk.htm)
and lots from http://www.voip-info.org/wiki-Asterisk+ISDN4Linux and
I've followed almost every link from www.asterisk.org...
All examples seem to include Digiums hardware :-(
I'm looking for clean, clear examples with a generic ISDN card - which
is my trunk line, and the two SIP phones.
The numbering plan in South Africa is pretty simple
7 digits for local calls
12 digits for long distance
Anyone in S.A. got some example configs to share with?
Currently - I'm stuck with the message..
-- Executing Dial("SIP/phone1-082a", "Modem/g1/8070590") in new stack
Apr 17 00:09:00 WARNING[507919]: chan_modem.c:181 modem_call:
Destination g1/8070590 requres a real destination (device:destination)
-- Couldn't call g1/8070590
-- Hungup 'Modem[i4l]/ttyI1'
... when I dial '98070590' (9 for outside - which I'll make '0' one
day!)
(its late, head hurts, wife is loosing patience)
help? hints?
--
. . ___. .__ Posix Systems - Sth Africa
/| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE
/ |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496
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Description: This is a digitally signed message part
[Asterisk-Users] Newbie - help
Hi, Is there a work around about Fax and Answering Machinedetection ?If not, where is the all process, at chan_zap.c ?Any site that could help ? Actually, how is this working? When we originate a call, * just recognize if the line is busy and then creates a record for that call at CDR ? or not ? Thanks, Marin Blu Do you Yahoo!? Yahoo! Finance: Get your refund fast by filing online
Re: [Asterisk-Users] newbie help.
On Wednesday 10 September 2003 09:47 am, Steve Bradwell wrote: > Hello All, > > I am a newbie looking to learn about Asterisk. I'm new to IVR and all > that goes with it. I would like to know if it is possible to grab the > number of an incoming call, have Asterisk, or third party software > return the call with an automated voice message allowing the original > placer of the call to select another person to call (eg "Select 1 for > Bob") then have Asterisk automatically place that call for the person. Yes, Asterisk can do this as I understand Asterisk. I may be worng on that, I am still learning about asterisk. > Is this a complex solution? That depends largely on your definition of "complex". By my standard, this would not be very complex to do. > Would anyone mind giving me some direction and possibly filling me in > on if this is possible? Sure. Read this article first: http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html Then read the Asterisk Handbook: http://www.digium.com/index.php?menu=asterisk_handbook and everything at: http://www.digium.com/index.php?menu=documentation Here is a resource about Asterisk scripting, links, tips, and other help: http://www.voip-info.org/tiki-index.php?page=Asterisk >Also how customizable is Asterisk and what > language do I use to customize or develop for it? Asterisk is extremely customizable. You can develop in nearly any language you want. A favorite on this list seems to be Perl. -- Thanks, Timothy Soos XQL, LLC ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] newbie help.
Hello All, I am a newbie looking to learn about Asterisk. I'm new to IVR and all that goes with it. I would like to know if it is possible to grab the number of an incoming call, have Asterisk, or third party software return the call with an automated voice message allowing the original placer of the call to select another person to call (eg "Select 1 for Bob") then have Asterisk automatically place that call for the person. Is this a complex solution? Would anyone mind giving me some direction and possibly filling me in on if this is possible? Also how customizable is Asterisk and what language do I use to customize or develop for it? Thanks in advance, and sorry for my ignorance =] Steve. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Newbie Help
Replying to myself - sorted now, so ignore me. I last cvs'd out about 10pm GMT - did it again just now and all is fine. Maybe I caught someone in the middle of an update or something. -- Tom [EMAIL PROTECTED] ICQ: 8018364 MSN: [EMAIL PROTECTED] Home# +44(0)1634309229 Fax# +44(0)1634710086 Mobile# +44(0)7764486175 US# +1(917)4386847 - Original Message - From: "Thomas Elliott" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Wednesday, July 23, 2003 11:16 AM Subject: [Asterisk-Users] Newbie Help > Hi - after hearing others rave about * I thought I'd have a go - extract > from a 'make' on a stock debian system as follows... (I tried to post the > whole make up to this point but it was too big for the list) > > make[1]: Leaving directory `/usr/src/asterisk/channels' > make[1]: Entering directory `/usr/src/asterisk/pbx' > gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarat > ions -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 > -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\"CVS-07/22/03-22:29:39\" -DINS TA > LL_PREFIX=\"\" -DASTETCDIR=\"/etc/asterisk\" -DASTLIBDIR=\"/usr/lib/asterisk > \" -DASTVARLIBDIR=\"/var/lib/asterisk\" -DASTVARRUNDIR=\"/var/run\" -DASTSPO > OLDIR=\"/var/spool/asterisk\" -DASTLOGDIR=\"/var/log/asterisk\" -DASTCONFPAT > H=\"/etc/asterisk/asterisk.conf\" -DASTMODDIR=\"/usr/lib/asterisk/modules\" > -DASTAGIDIR=\"/var/lib/asterisk/agi-bin\"-DBUSYDETECT -fPIC -c - o > pbx_config.o pbx_config.c > gcc -shared -Xlinker -x -o pbx_config.so pbx_config.o > gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarat > ions -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 > -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\"CVS-07/22/03-22:29:39\" -DINS TA > LL_PREFIX=\"\" -DASTETCDIR=\"/etc/asterisk\" -DASTLIBDIR=\"/usr/lib/asterisk > \" -DASTVARLIBDIR=\"/var/lib/asterisk\" -DASTVARRUNDIR=\"/var/run\" -DASTSPO > OLDIR=\"/var/spool/asterisk\" -DASTLOGDIR=\"/var/log/asterisk\" -DASTCONFPAT > H=\"/etc/asterisk/asterisk.conf\" -DASTMODDIR=\"/usr/lib/asterisk/modules\" > -DASTAGIDIR=\"/var/lib/asterisk/agi-bin\"-DBUSYDETECT -fPIC -c - o > pbx_wilcalu.o pbx_wilcalu.c > gcc -shared -Xlinker -x -o pbx_wilcalu.so pbx_wilcalu.o > gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarat > ions -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 > -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\"CVS-07/22/03-22:29:39\" -DINS TA > LL_PREFIX=\"\" -DASTETCDIR=\"/etc/asterisk\" -DASTLIBDIR=\"/usr/lib/asterisk > \" -DASTVARLIBDIR=\"/var/lib/asterisk\" -DASTVARRUNDIR=\"/var/run\" -DASTSPO > OLDIR=\"/var/spool/asterisk\" -DASTLOGDIR=\"/var/log/asterisk\" -DASTCONFPAT > H=\"/etc/asterisk/asterisk.conf\" -DASTMODDIR=\"/usr/lib/asterisk/modules\" > -DASTAGIDIR=\"/var/lib/asterisk/agi-bin\"-DBUSYDETECT -fPIC -c - o > pbx_spool.o pbx_spool.c > gcc -shared -Xlinker -x -o pbx_spool.so pbx_spool.o > make[1]: Leaving directory `/usr/src/asterisk/pbx' > make[1]: Entering directory `/usr/src/asterisk/apps' > gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarat > ions -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 > -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\"CVS-07/22/03-22:29:39\" -DINS TA > LL_PREFIX=\"\" -DASTETCDIR=\"/etc/asterisk\" -DASTLIBDIR=\"/usr/lib/asterisk > \" -DASTVARLIBDIR=\"/var/lib/asterisk\" -DASTVARRUNDIR=\"/var/run\" -DASTSPO > OLDIR=\"/var/spool/asterisk\" -DASTLOGDIR=\"/var/log/asterisk\" -DASTCONFPAT > H=\"/etc/asterisk/asterisk.conf\" -DASTMODDIR=\"/usr/lib/asterisk/modules\" > -DASTAGIDIR=\"/var/lib/asterisk/agi-bin\"-DBUSYDETECT -fPIC -c - o > app_dial.o app_dial.c > app_dial.c: In function `wait_for_answer': > app_dial.c:232: parse error before `o' > app_dial.c:242: parse error before `o' > app_dial.c:285: parse error before `o' > make[1]: *** [app_dial.o] Error 1 > make[1]: Leaving directory `/usr/src/asterisk/apps' > make: *** [subdirs] Error 1 > hermes:/usr/src/asterisk# > > I've googled around and asked a few other asterisk users and they've not > seen this before. > > Anyone seen this before? > Thanks > -- > Tom > [EMAIL PROTECTED] > ICQ: 8018364 MSN: [EMAIL PROTECTED] > Home# +44(0)1634309229 Fax# +44(0)1634710086 > Mobile# +44(0)7764486175 US# +1(917)4386847 > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Newbie Help
Hi - after hearing others rave about * I thought I'd have a go - extract from a 'make' on a stock debian system as follows... (I tried to post the whole make up to this point but it was too big for the list) make[1]: Leaving directory `/usr/src/asterisk/channels' make[1]: Entering directory `/usr/src/asterisk/pbx' gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarat ions -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\"CVS-07/22/03-22:29:39\" -DINSTA LL_PREFIX=\"\" -DASTETCDIR=\"/etc/asterisk\" -DASTLIBDIR=\"/usr/lib/asterisk \" -DASTVARLIBDIR=\"/var/lib/asterisk\" -DASTVARRUNDIR=\"/var/run\" -DASTSPO OLDIR=\"/var/spool/asterisk\" -DASTLOGDIR=\"/var/log/asterisk\" -DASTCONFPAT H=\"/etc/asterisk/asterisk.conf\" -DASTMODDIR=\"/usr/lib/asterisk/modules\" -DASTAGIDIR=\"/var/lib/asterisk/agi-bin\"-DBUSYDETECT -fPIC -c -o pbx_config.o pbx_config.c gcc -shared -Xlinker -x -o pbx_config.so pbx_config.o gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarat ions -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\"CVS-07/22/03-22:29:39\" -DINSTA LL_PREFIX=\"\" -DASTETCDIR=\"/etc/asterisk\" -DASTLIBDIR=\"/usr/lib/asterisk \" -DASTVARLIBDIR=\"/var/lib/asterisk\" -DASTVARRUNDIR=\"/var/run\" -DASTSPO OLDIR=\"/var/spool/asterisk\" -DASTLOGDIR=\"/var/log/asterisk\" -DASTCONFPAT H=\"/etc/asterisk/asterisk.conf\" -DASTMODDIR=\"/usr/lib/asterisk/modules\" -DASTAGIDIR=\"/var/lib/asterisk/agi-bin\"-DBUSYDETECT -fPIC -c -o pbx_wilcalu.o pbx_wilcalu.c gcc -shared -Xlinker -x -o pbx_wilcalu.so pbx_wilcalu.o gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarat ions -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\"CVS-07/22/03-22:29:39\" -DINSTA LL_PREFIX=\"\" -DASTETCDIR=\"/etc/asterisk\" -DASTLIBDIR=\"/usr/lib/asterisk \" -DASTVARLIBDIR=\"/var/lib/asterisk\" -DASTVARRUNDIR=\"/var/run\" -DASTSPO OLDIR=\"/var/spool/asterisk\" -DASTLOGDIR=\"/var/log/asterisk\" -DASTCONFPAT H=\"/etc/asterisk/asterisk.conf\" -DASTMODDIR=\"/usr/lib/asterisk/modules\" -DASTAGIDIR=\"/var/lib/asterisk/agi-bin\"-DBUSYDETECT -fPIC -c -o pbx_spool.o pbx_spool.c gcc -shared -Xlinker -x -o pbx_spool.so pbx_spool.o make[1]: Leaving directory `/usr/src/asterisk/pbx' make[1]: Entering directory `/usr/src/asterisk/apps' gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarat ions -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\"CVS-07/22/03-22:29:39\" -DINSTA LL_PREFIX=\"\" -DASTETCDIR=\"/etc/asterisk\" -DASTLIBDIR=\"/usr/lib/asterisk \" -DASTVARLIBDIR=\"/var/lib/asterisk\" -DASTVARRUNDIR=\"/var/run\" -DASTSPO OLDIR=\"/var/spool/asterisk\" -DASTLOGDIR=\"/var/log/asterisk\" -DASTCONFPAT H=\"/etc/asterisk/asterisk.conf\" -DASTMODDIR=\"/usr/lib/asterisk/modules\" -DASTAGIDIR=\"/var/lib/asterisk/agi-bin\"-DBUSYDETECT -fPIC -c -o app_dial.o app_dial.c app_dial.c: In function `wait_for_answer': app_dial.c:232: parse error before `o' app_dial.c:242: parse error before `o' app_dial.c:285: parse error before `o' make[1]: *** [app_dial.o] Error 1 make[1]: Leaving directory `/usr/src/asterisk/apps' make: *** [subdirs] Error 1 hermes:/usr/src/asterisk# I've googled around and asked a few other asterisk users and they've not seen this before. Anyone seen this before? Thanks -- Tom [EMAIL PROTECTED] ICQ: 8018364 MSN: [EMAIL PROTECTED] Home# +44(0)1634309229 Fax# +44(0)1634710086 Mobile# +44(0)7764486175 US# +1(917)4386847 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Newbie help getting started
Correct me if I am wrong, as I am a newbie too. When I installed my first * box, the kernel modules weren't loading at boot. This may be your problem. So how I made the sample configs and it install a file zaptel into my /etc/sysconfig folder. It also installed an rc script for boot. Remember to backup you base line. Also had to change the order in which modules for * started to make sure the correct channel order was in place. Hope this helps, if it doesn't please ignore. Matt -- Original Message -- From: "Joel Becker" <[EMAIL PROTECTED]> Reply-To: [EMAIL PROTECTED] Date: Wed, 4 Jun 2003 21:04:12 +0100 I'm running Redhat version 9 with a TDM400P and AVM B4 ISDN2 card. I am assuming the first step to get * working is to get a tone from the fxo ports, having looked and played with both the zapata and zaptel files, I still can't get anything.on the ports. When starting asterisk -vvvgc it does not appear to be loading Zapata file. Any guidance for a newbie would be greatly appreciated. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Newbie help getting started
I’m running Redhat version 9 with a TDM400P and AVM B4 ISDN2 card. I am assuming the first step to get * working is to get a tone from the fxo ports, having looked and played with both the zapata and zaptel files, I still can’t get anything.on the ports. When starting asterisk –vvvgc it does not appear to be loading Zapata file. Any guidance for a newbie would be greatly appreciated.
