[Asterisk-Users] 1.2.1 can´t register with SIP-Provider, 1.0. 9 could
Hy List, i´m having a big problem, with my new Asterisk 1.2.1 Server i cannot register with my SIP-Providers. With my old Asterisk Server i hadn´t such problems. Here is the relevant part of my new sip.conf: [general] context=default bindport=5060 bindaddr=0.0.0.0 srvlookup=yes language=de ; Default language setting for all users/peers srvlookup=yes tos=0x18 insecure=very dtmfmode=rfc2833 canreinvite=yes ;localnet=172.22.20.195/255.255.0.0 register = 5550109:[EMAIL PROTECTED]/5550109 register = 08130884:[EMAIL PROTECTED]/08130884 register = 08130091:[EMAIL PROTECTED]/08130091 ;SIPGATE incoming [sipgate_in] type=peer host=sipgate.de disallow=all allow=alaw allow=ulaw allow=gsm nat=yes ;STANPHONE incoming# [stanaphone_in] type=peer fromdomain=sip.stanaphone.com host=sip.stanaphone.com disallow=all allow=alaw allow=ulaw allow=gsm nat=yes And here is the relevant part of my old sip.conf: [general] context=default ; Default context for incoming calls port=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to all) srvlookup=yes ; Enable DNS SRV lookups on outbound calls disallow=all; First disallow all codecs allow=alaw allow=ulaw allow=gsm ; This may also be set for individual users/peers language=de ; Default language setting for all users/peers srvlookup=yes tos=0x18 insecure=very dtmfmode=rfc2833 canreinvite=yes localnet=172.22.20.194/255.255.0.0 ; register = 08130884:[EMAIL PROTECTED]/08130884 register = 5550109:[EMAIL PROTECTED]/5550109 [stanaphone4]; type=friend username=08130884 fromuser=08130884 secret=XX nat=yes canreinvite=no qualify=yes fromdomain=sip.stanaphone.com host=sip.stanaphone.com auth=md5,plaintext [sipgate] type=friend nat=yes username=5550109 fromuser=5550109 secret=XX host=sipgate.de qualify=yes disallow=all allow=alaw allow=gsm I´m sitting with my Asterisk boxes behind a nat (Checkpoint NG Firewall), but i don´t think, the solution for my problem is somewhere in my Asterisk Configuration, because with my old Asterisk box and the Configuration above i had no problems behind this firewall. -- Mit freundlichen Grüßen With kind regards Klaus Peras begin:vcard fn:Klaus Peras n:Peras;Klaus org:HOB;Netzwerk Support adr;quoted-printable:;;Schwaderm=C3=BChlstrasse 3;Cadolzburg;Bayern;90556;Germany email;internet:[EMAIL PROTECTED] tel;work:09103 / 715 - 329 url:http://www.hob.de version:2.1 end:vcard ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] new in asterisk world
the oreilly Asterik book can be found at: http://www.nufone.net/downloads/asteriskdocs/AsteriskTFOT.zip Mit freundlichen Gren With kind regards Klaus Peras Dovid Bender schrieb: in Sip.conf make sure to add NAT=yes. Also I reccomend reading the new book that came out. (Dont have the URL if some one else can please post it). The book will help you learn the basics of asterisk and also answer many of your questions. Also check out the wiki. Dovid --- Ever Zalazar [EMAIL PROTECTED] wrote: Hi, I'm new in asterisk world. I have questions. For example I have my server with public IP address, but two customer with softphone in a private network. How can I do to make them work with the asterisk server? Best Regards -- Ever Zalazar ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users begin:vcard fn:Klaus Peras n:Peras;Klaus org:HOB;Netzwerk Support adr;quoted-printable:;;Schwaderm=C3=BChlstrasse 3;Cadolzburg;Bayern;90556;Germany email;internet:[EMAIL PROTECTED] tel;work:09103 / 715 - 329 url:http://www.hob.de version:2.1 end:vcard ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk won´t load module codec_g729a.so
Hello List, my Asterisk will not load the module codec_g729a.so asterisk3*CLI load codec_g729a.so Unable to load module codec_g729a.so What did i do wrong? I followed the README File from Digium step by step. cheers klaus begin:vcard fn:Klaus Peras n:Peras;Klaus org:HOB;Netzwerk Support adr;quoted-printable:;;Schwaderm=C3=BChlstrasse 3;Cadolzburg;Bayern;90556;Germany email;internet:[EMAIL PROTECTED] tel;work:09103 / 715 - 329 url:http://www.hob.de version:2.1 end:vcard ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Music On Hold
Wich player do you use?
I use the one that is coming with Asterisk. Just cd to the Asterisk
Sources, make mpg123, cd mpg..., make make install
and im done. It worked fine all the time.
cheers
klaus
Bud Bach schrieb:
Help! No Music on Hold.
Probably a novice
mistake but I cant figure it out. Here are the details:
CentOS 4.2
Asterisk 1.2.1 (Do I need
to do something to get MOH to
build?)
Ztdummy loaded
(conference works fine)
musiconhold.conf:
[default]
mode=quietmp3
directory=/var/lib/asterisk/mohmp3
Sip device (x-lite also
tried with an ATA) with canreinvite=no:
sip.conf:
[7211]
username=7211
secret=
host=dynamic
type=friend
context=standardphone
disallow=all
allow=gsm
allow=ulaw
allow=alaw
allow=g723.1
allow=g729
canreinvite=no
Extensions.conf:
exten =
8702,1,Answer()
exten =
8702,n,MusicOnHold(default)
exten =
8702,n,Hangup()
# asterisk -r
Asterisk 1.2.1, Copyright
(C) 1999 - 2005 Digium.
Written by Mark Spencer
[EMAIL PROTECTED]
=
Connected to Asterisk
1.2.1 currently running on ccsip (pid
= 4782)
Verbosity is at least 3
== Spawn extension
(standardphone, 8702, 2) exited
non-zero on 'SIP/7211-be01'
-- Executing
Answer("SIP/7211-cedb",
"") in new stack
-- Executing
MusicOnHold("SIP/7211-cedb",
"default") in new stack
-- Started music on
hold, class
'default', on channel 'SIP/7211-cedb'
-- Stopped music on
hold on SIP/7211-cedb
== Spawn extension
(standardphone, 8702, 2) exited
non-zero on 'SIP/7211-cedb'
The Stopped music on
hold happens immediately
like it cant find something. Should I give up and use madplay?
-- Bud
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n:Peras;Klaus
org:HOB;Netzwerk Support
adr;quoted-printable:;;Schwaderm=C3=BChlstrasse 3;Cadolzburg;Bayern;90556;Germany
email;internet:[EMAIL PROTECTED]
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Re: [Asterisk-Users] g729 translation to zap (ISDN) doesn´t work
I thougt i have some problems with ztdummy and removed that # in front of ztdummy in the zaptel Makefile before compiling. But still no change. I even tried it with another Phone, a Planet VIP-150T. Still the same Problem, i don´t hear anything from the SIP Phone on the ISDN Phone, but i hear everything fine the other way. Any Ideas? Thanks a lot for help. regards Klaus Peras Klaus Peras schrieb: Hi, i just figured out, that there is also a problem by going in a conference with the sip phone that runs the g729a codec. Could it be, that i have timing problems? I don´t have digium hardware installed, but i have ztdummy: asterisk3:/etc/asterisk# lsmod | grep ztdummy ztdummy 3748 0 zaptel225540 24 ztdummy,qozap Does anybody have a advice for me? Mit freundlichen Grüßen With kind regards Klaus Peras Klaus Peras schrieb: Hi Asterisk Users, i have a bristuffed-0.2.0-RC8q Asterisk 1.0.9 System running on a Debian 3.1. With a quadbri card installad, wich is running on the bristuff drivers. Everything seems to be fine so far. but now i wanted to use the g.729A Codec. I bought 5 licences and installed them: asterisk3*CLI show g729 0/0 encoders/decoders of 5 licensed channels are currently in use When i do sip to sip calls, everything is working fine (from a snom 190 wich is running with that codec to a sip phone with g.711a), asterisk is translating correct. the output on the CLI is: asterisk3*CLI show g729 1/0 encoders/decoders of 5 licensed channels are currently in use But if i try to call a zap channel from that sip phone (snom 190) wich runs that g729 Codec, i don´t hear anything on the ISDN Phone. the output on the CLI: asterisk3*CLI show g729 1/1 encoders/decoders of 5 licensed channels are currently in use Here is the output of the show channel command for the SIP Channel and the ZAP Channel: asterisk3*CLI show channel SIP/71-d293 -- General -- Name: SIP/71-d293 Type: SIP UniqueID: asterisk-2204-1134137006.49 Caller ID: 30071 DNID Digits: 329 State: Up (6) Rings: 0 NativeFormat: 256 WriteFormat: 256 ReadFormat: 64 1st File Descriptor: 31 Frames in: 7949 Frames out: 7956 Time to Hangup: 0 Elapsed Time: 0h2m39s -- PBX -- Context: default Extension: 329 Priority: 2 Call Group: 0 Pickup Group: 0 Application: Dial Data: Zap/g1/329 Stack: 0 Blocking in: ast_waitfor_nandfds asterisk3*CLI show channel Zap/1-1 -- General -- Name: Zap/1-1 Type: Zap UniqueID: asterisk-2204-1134137006.50 Caller ID: 30071 DNID Digits: 329 State: Up (6) Rings: 0 NativeFormat: 72 WriteFormat: 64 ReadFormat: 256 1st File Descriptor: 13 Frames in: 8255 Frames out: 8246 Time to Hangup: 0 Elapsed Time: 0h0m0s -- PBX -- Context: default Extension: s Priority: 1 Call Group: 0 Pickup Group: 0 Application: Bridged Call Data: SIP/71-d293 Stack: -1 Blocking in: ast_waitfor_nandfds I don´t know what i can do on this problem and would be pleased to get some help. Thank you very much! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users begin:vcard fn:Klaus Peras n:Peras;Klaus org:HOB;Netzwerk Support adr;quoted-printable:;;Schwaderm=C3=BChlstrasse 3;Cadolzburg;Bayern;90556;Germany email;internet:[EMAIL PROTECTED] tel;work:09103 / 715 - 329 url:http://www.hob.de version:2.1 end:vcard ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] WIFI Phones
and that ipaq can do iax2 ?? Guess not cheers klaus trixter aka Bret McDanel schrieb: On Wed, 2005-12-14 at 17:33 +0100, Matt Riddell wrote: rossi.tek wrote: I'm looking for iax2 wifi phones, do you know where i can buy them? Yes. Nowhere. :) not entirely true if you expand your definition :P I have an ipaq which is capable of acting like a soft phone (and it does, although I use a sip client) and it has integrated wifi. There are some really cheap pdas out there now with integrated wifi, in some cases cheaper than some of the wifi phones sold. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users begin:vcard fn:Klaus Peras n:Peras;Klaus org:HOB;Netzwerk Support adr;quoted-printable:;;Schwaderm=C3=BChlstrasse 3;Cadolzburg;Bayern;90556;Germany email;internet:[EMAIL PROTECTED] tel;work:09103 / 715 - 329 url:http://www.hob.de version:2.1 end:vcard ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] g729 translation to zap (ISDN) doesn´t work
Hi Asterisk Users, i have a bristuffed-0.2.0-RC8q Asterisk 1.0.9 System running on a Debian 3.1. With a quadbri card installad, wich is running on the bristuff drivers. Everything seems to be fine so far. but now i wanted to use the g.729A Codec. I bought 5 licences and installed them: asterisk3*CLI show g729 0/0 encoders/decoders of 5 licensed channels are currently in use When i do sip to sip calls, everything is working fine (from a snom 190 wich is running with that codec to a sip phone with g.711a), asterisk is translating correct. the output on the CLI is: asterisk3*CLI show g729 1/0 encoders/decoders of 5 licensed channels are currently in use But if i try to call a zap channel from that sip phone (snom 190) wich runs that g729 Codec, i don´t hear anything on the ISDN Phone. the output on the CLI: asterisk3*CLI show g729 1/1 encoders/decoders of 5 licensed channels are currently in use Here is the output of the show channel command for the SIP Channel and the ZAP Channel: asterisk3*CLI show channel SIP/71-d293 -- General -- Name: SIP/71-d293 Type: SIP UniqueID: asterisk-2204-1134137006.49 Caller ID: 30071 DNID Digits: 329 State: Up (6) Rings: 0 NativeFormat: 256 WriteFormat: 256 ReadFormat: 64 1st File Descriptor: 31 Frames in: 7949 Frames out: 7956 Time to Hangup: 0 Elapsed Time: 0h2m39s -- PBX -- Context: default Extension: 329 Priority: 2 Call Group: 0 Pickup Group: 0 Application: Dial Data: Zap/g1/329 Stack: 0 Blocking in: ast_waitfor_nandfds asterisk3*CLI show channel Zap/1-1 -- General -- Name: Zap/1-1 Type: Zap UniqueID: asterisk-2204-1134137006.50 Caller ID: 30071 DNID Digits: 329 State: Up (6) Rings: 0 NativeFormat: 72 WriteFormat: 64 ReadFormat: 256 1st File Descriptor: 13 Frames in: 8255 Frames out: 8246 Time to Hangup: 0 Elapsed Time: 0h0m0s -- PBX -- Context: default Extension: s Priority: 1 Call Group: 0 Pickup Group: 0 Application: Bridged Call Data: SIP/71-d293 Stack: -1 Blocking in: ast_waitfor_nandfds I don´t know what i can do on this problem and would be pleased to get some help. Thank you very much! -- Mit freundlichen Grüßen With kind regards Klaus Peras begin:vcard fn:Klaus Peras n:Peras;Klaus org:HOB;Netzwerk Support adr;quoted-printable:;;Schwaderm=C3=BChlstrasse 3;Cadolzburg;Bayern;90556;Germany email;internet:[EMAIL PROTECTED] tel;work:09103 / 715 - 329 url:http://www.hob.de version:2.1 end:vcard ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] g729 translation to zap (ISDN) doesn´t work
Hi, i just figured out, that there is also a problem by going in a conference with the sip phone that runs the g729a codec. Could it be, that i have timing problems? I don´t have digium hardware installed, but i have ztdummy: asterisk3:/etc/asterisk# lsmod | grep ztdummy ztdummy 3748 0 zaptel225540 24 ztdummy,qozap Does anybody have a advice for me? Mit freundlichen Grüßen With kind regards Klaus Peras Klaus Peras schrieb: Hi Asterisk Users, i have a bristuffed-0.2.0-RC8q Asterisk 1.0.9 System running on a Debian 3.1. With a quadbri card installad, wich is running on the bristuff drivers. Everything seems to be fine so far. but now i wanted to use the g.729A Codec. I bought 5 licences and installed them: asterisk3*CLI show g729 0/0 encoders/decoders of 5 licensed channels are currently in use When i do sip to sip calls, everything is working fine (from a snom 190 wich is running with that codec to a sip phone with g.711a), asterisk is translating correct. the output on the CLI is: asterisk3*CLI show g729 1/0 encoders/decoders of 5 licensed channels are currently in use But if i try to call a zap channel from that sip phone (snom 190) wich runs that g729 Codec, i don´t hear anything on the ISDN Phone. the output on the CLI: asterisk3*CLI show g729 1/1 encoders/decoders of 5 licensed channels are currently in use Here is the output of the show channel command for the SIP Channel and the ZAP Channel: asterisk3*CLI show channel SIP/71-d293 -- General -- Name: SIP/71-d293 Type: SIP UniqueID: asterisk-2204-1134137006.49 Caller ID: 30071 DNID Digits: 329 State: Up (6) Rings: 0 NativeFormat: 256 WriteFormat: 256 ReadFormat: 64 1st File Descriptor: 31 Frames in: 7949 Frames out: 7956 Time to Hangup: 0 Elapsed Time: 0h2m39s -- PBX -- Context: default Extension: 329 Priority: 2 Call Group: 0 Pickup Group: 0 Application: Dial Data: Zap/g1/329 Stack: 0 Blocking in: ast_waitfor_nandfds asterisk3*CLI show channel Zap/1-1 -- General -- Name: Zap/1-1 Type: Zap UniqueID: asterisk-2204-1134137006.50 Caller ID: 30071 DNID Digits: 329 State: Up (6) Rings: 0 NativeFormat: 72 WriteFormat: 64 ReadFormat: 256 1st File Descriptor: 13 Frames in: 8255 Frames out: 8246 Time to Hangup: 0 Elapsed Time: 0h0m0s -- PBX -- Context: default Extension: s Priority: 1 Call Group: 0 Pickup Group: 0 Application: Bridged Call Data: SIP/71-d293 Stack: -1 Blocking in: ast_waitfor_nandfds I don´t know what i can do on this problem and would be pleased to get some help. Thank you very much! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users begin:vcard fn:Klaus Peras n:Peras;Klaus org:HOB;Netzwerk Support adr;quoted-printable:;;Schwaderm=C3=BChlstrasse 3;Cadolzburg;Bayern;90556;Germany email;internet:[EMAIL PROTECTED] tel;work:09103 / 715 - 329 url:http://www.hob.de version:2.1 end:vcard ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP Phone binary
Hey there, does anybody know a SIP-Client that I only have to unpack and can run it on Linux just like SJPhone, except SJPhone?? I need a Softphone for a Levigo Thin-Client, wich is not having a compiler. regards Klaus Peras begin:vcard fn:Klaus Peras n:Peras;Klaus org:HOB;Netzwerk Support adr;quoted-printable:;;Schwaderm=C3=BChlstrasse 3;Cadolzburg;Bayern;90556;Germany email;internet:[EMAIL PROTECTED] tel;work:09103 / 715 - 329 url:http://www.hob.de version:2.1 end:vcard ___ 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] CLI SIP Client
Hey there, does anybody know a CLI SIP Client für Linux? -- Mit freundlichen Grüßen With kind regards Klaus Peras Support Networks/Networkmanagement HOB GmbH Co KG Schwadermühlstrasse 3 D-90556 Cadolzburg Tel: 0 9103 - 715 -329 Fax: 0 9103 - 715 -299 Mobil: 0 175 63 78 911 URLs: http://www.hob.de http://www.hob.de/produkte/netz/netz.htm begin:vcard fn:Klaus Peras n:Peras;Klaus org:HOB;Netzwerk Support adr;quoted-printable:;;Schwaderm=C3=BChlstrasse 3;Cadolzburg;Bayern;90556;Germany email;internet:[EMAIL PROTECTED] tel;work:09103 / 715 - 329 url:http://www.hob.de version:2.1 end:vcard ___ 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] problem in compiling chan_mISDN
Hi List, Im having problems compiling chan_misdn:
asterisk:/usr/src/chan_misdn-beta-0.0.3-rc4 # make install
cc -ggdb -Wall -D_GNU_SOURCE -Wno-missing-prototypes
-Wno-missing-declarations -fPIC -I/usr/src/asterisk/include
-DAST_CONFIG_DIR=\/etc/asterisk/\ -I/usr/src/mISDNuser/include
-I/usr/src/linux-2.6/include -I/usr/src/mISDNuser/i4lnet/ -Wall -c -o
chan_misdn.o chan_misdn.c
chan_misdn.c:30:34: asterisk/channel_pvt.h: No such file or directory
chan_misdn.c: In function `misdn_call':
chan_misdn.c:664: error: dereferencing pointer to incomplete type
chan_misdn.c: In function `misdn_answer':
chan_misdn.c:695: error: dereferencing pointer to incomplete type
chan_misdn.c: In function `misdn_digit':
chan_misdn.c:724: error: dereferencing pointer to incomplete type
chan_misdn.c: In function `misdn_fixup':
chan_misdn.c:767: error: dereferencing pointer to incomplete type
chan_misdn.c:768: error: dereferencing pointer to incomplete type
chan_misdn.c: In function `misdn_soption':
chan_misdn.c:782: error: dereferencing pointer to incomplete type
chan_misdn.c: In function `misdn_qoption':
chan_misdn.c:790: error: dereferencing pointer to incomplete type
chan_misdn.c: In function `misdn_transfer':
chan_misdn.c:798: error: dereferencing pointer to incomplete type
chan_misdn.c: In function `misdn_indication':
chan_misdn.c:808: error: dereferencing pointer to incomplete type
chan_misdn.c: In function `misdn_hangup':
chan_misdn.c:916: error: dereferencing pointer to incomplete type
chan_misdn.c:919: error: dereferencing pointer to incomplete type
chan_misdn.c:925: error: dereferencing pointer to incomplete type
chan_misdn.c: In function `misdn_read':
chan_misdn.c:1022: error: dereferencing pointer to incomplete type
chan_misdn.c: In function `misdn_write':
chan_misdn.c:1037: error: dereferencing pointer to incomplete type
chan_misdn.c: In function `misdn_new':
chan_misdn.c:1114: error: dereferencing pointer to incomplete type
chan_misdn.c:1115: error: dereferencing pointer to incomplete type
chan_misdn.c:1125: error: dereferencing pointer to incomplete type
chan_misdn.c:1128: error: dereferencing pointer to incomplete type
chan_misdn.c:1129: error: dereferencing pointer to incomplete type
chan_misdn.c:1130: error: dereferencing pointer to incomplete type
chan_misdn.c:1131: error: dereferencing pointer to incomplete type
chan_misdn.c:1132: error: dereferencing pointer to incomplete type
chan_misdn.c:1133: error: dereferencing pointer to incomplete type
chan_misdn.c:1137: error: dereferencing pointer to incomplete type
chan_misdn.c:1138: error: dereferencing pointer to incomplete type
chan_misdn.c:1139: error: dereferencing pointer to incomplete type
chan_misdn.c:1140: error: dereferencing pointer to incomplete type
chan_misdn.c:1143: error: dereferencing pointer to incomplete type
chan_misdn.c: In function `release_chan':
chan_misdn.c:1387: error: dereferencing pointer to incomplete type
chan_misdn.c: In function `load_module':
chan_misdn.c:2342: warning: passing arg 1 of `ast_channel_register' from
incompatible pointer type
chan_misdn.c:2342: error: too many arguments to function
`ast_channel_register'
chan_misdn.c: In function `unload_module':
chan_misdn.c:2381: warning: passing arg 1 of `ast_channel_unregister'
from incompatible pointer type
make: *** [chan_misdn.o] Error 1
Does anybody have an idea where i got wrong? My Makefile:
asterisk:/usr/src/chan_misdn-beta-0.0.3-rc4 # cat Makefile
#debugscript
DEBUGSCRIPT=sed -e s/{/{\nchan_misdn_log(\SEGFAULT_DEBUG:
%d\\\n\,__LINE__)/g
UNDEBUGSCRIPT=sed -e s/{/{\nchan_misdn_log(\SEGFAULT_DEBUG:
%d\\\n\,__LINE__)/g
#
# The following line tells the makefile where to install the module,
# if necessary, please edit it
#
INSTALL_MODPATH=/usr/lib/asterisk/modules
#
# The following line tells the makefile where to find the asterisk src, so
# please edit this one if necessary
#
ASTERISKSRC=/usr/src/asterisk
#
# The following line tells the makefile where to put in the configfile
# of this module
#
AST_CONFIG_DIR=/etc/asterisk/
#
# The Includes are Set appropriatly
#
ASTERISKINC=$(ASTERISKSRC)/include
#
# mISDNuser PATHS
#
MISDNUSER=/usr/src/mISDNuser
MISDNUSERINC=$(MISDNUSER)/include
MISDNUSERLIB=$(MISDNUSER)/lib
#
# mISDNuser Version
#
# If you dont use the Jolly mISDNuser version above 2.7 then comment this
#
#CFLAGS+=-DMISDNUSER_JOLLY
#
# ASTERISK Version
# If you are using a asterisk version above from stable (v1-0)
# then comment the following line out (good luck)
#
#CFLAGS+=-DASTERISK_STABLE
LINUXROOT=/usr/src/linux-2.6
#
# Linux Includes (must be patched with mISDN!)
#
LINUXINC=$(LINUXROOT)/include
CFLAGS+=-ggdb -Wall -D_GNU_SOURCE
CFLAGS+=-Wno-missing-prototypes -Wno-missing-declarations
all: chan_misdn.so
CFLAGS+=-fPIC -I$(ASTERISKINC) -DAST_CONFIG_DIR=\$(AST_CONFIG_DIR)\
-I$(MISDNUSERINC) -I$(LINUXINC) -I$(MISDNUSER)/i4lnet/ -Wall
ADDOBJS+=$(MISDNUSER)/i4lnet/libisdnnet.a $(MISDNUSER)/lib/libmISDN.a
chan_misdn.so: chan_misdn.o te_lib.o $(ADDOBJS)
$(CC) -shared
