0.3.0-PRE-1u asterisk 1.2.13
0.3.0-PRE-1t
- fixed "removed "callstate fix" fix
- added experimental support to display the correct callerid on a SNOM phone
after making an attended transfer to that phone
- introducing res_esel (Extension State Export Logic)
- updated to asterisk 1.2.12.1, zaptel 1.2.9.1
0.3.0-PRE-1s
- added "FASTBUSYONBUSY" Makefile option to libpri
- fixed "BAD CRC" error on layer 1 activation in TE mode
- added hardware bridging support to qozap
- moved debugging messages in qozap to debug > 0
- updated to asterisk 1.2.10
- libgsmat fix for "network registration rejected"
- removed "callstate fix"
- autoanswer "exception flag" fix
- MG2 is now the default echo canceler (zaptel)
- audio will not pass through userspace if transfercapablity is digital (app_dial)
- hangup cause fix when using priindication=passthrough
0.3.0-PRE-1r
- fixed devicestates for peer register/unregister
- moved extensionstates to AMI privilege "extensionstatus"
- added "Hint:" to extensionstatus event (AMI)
- added "gsm [no] debug channel <channel>" to chan_zap
- modified autoanswer to re-register dynamic extensions on reload
- fixed drivers to compile with latest 2.6 kernels
- improved libpri/chan_zap communication for BRI_CPE and BRI_CPE_PTMPbristuff-0.3.0-PRE-1v
- fixed segfault introduced by 0.3.0-PRE-1u in chan_sipfli4l 3.1.1 # cat /proc/version
Linux version 2.6.16.27 (jw5@eisler) (gcc version 3.4.4 20050314 (prerelease) (Debian 3.4.3-13)) #1 Tue Aug 29 13:59:47 CEST 2006
li4l 3.1.1 # lsmod
zaphfc 12172 3 - Live 0xf0992000
zaptel 187204 11 zaphfc, Live 0xf0b41000
crc_ccitt 1552 1 zaptel, Live 0xf0990000
fli4l 3.1.1 # ztcfg -vvvv
Zaptel Configuration
======================
SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1)
Channel map:
Channel 01: Clear channel (Default) (Slaves: 01)
Channel 02: Clear channel (Default) (Slaves: 02)
Channel 03: D-channel (Default) (Slaves: 03)
3 channels configured.
fli4l 3.1.1 # asterisk -dddvvvr
Parsing /etc/asterisk/asterisk.conf
Parsing /data/share/asterisk/etc/extconfig.conf
== Parsing '/etc/asterisk/asterisk.conf': Found
== Parsing '/data/share/asterisk/etc/extconfig.conf': Found
Asterisk 1.2.13-BRIstuffed-0.3.0-PRE-1v, Copyright (C) 1999 - 2006 Digium, Inc. and others.
Created by Mark Spencer <[email protected]>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'show license' for details.
=========================================================================
Connected to Asterisk 1.2.13-BRIstuffed-0.3.0-PRE-1v currently running on fli4l (pid = 4642)
Verbosity is at least 3
-- Remote UNIX connection
fli4l*CLI> zap show status
Description Alarms IRQ bpviol CRC4
HFC-S PCI A ISDN card 0 [NT] layer 1 DE· OK 0 0 0
fli4l*CLI> zap show channels
Chan Extension Context Language MusicOnHold
pseudo outbound de
1 outbound de
2 outbound de
0.3.0-PRE-1w
- fixed line sync source selection in qozap
- fixed hardware bridging "one-way-audio" bug in qozap
- fixed stuck gsm channels
- improved gsm sms support (manager commands, pbx_spool, PDU deccoding)
- fixed sim hotplug changer support
to replace a sim, shutdown the gsm module using:
ztcfg -s<SPAN> (e.g. "ztcfg -s1")
replace the sim and start the gsm module using:
ztcfg -a<SPAN> (e.g. "ztcfg -a1")
This can be done while asterisk is running. Sometimes this can be useful
to reset individual PRI/BRI spans0.3.0-PRE-1x
- added software hdlc option for cwain driver ("modprobe cwain hw_hdlc=0")
- added hardware audio bridging to cwain driver ("modprobe cwain dacs=1")
- updated to asterisk 1.2.14
- fixes for ztgsm/chan_zap ("..unable to dial in state..")
0.3.0-PRE-1w
- fixed line sync source selection in qozap
- fixed hardware bridging "one-way-audio" bug in qozap
- fixed stuck gsm channels
- improved gsm sms support (manager commands, pbx_spool, PDU deccoding)
- fixed sim hotplug changer support
to replace a sim, shutdown the gsm module using:
ztcfg -s<SPAN> (e.g. "ztcfg -s1")
replace the sim and start the gsm module using:
ztcfg -a<SPAN> (e.g. "ztcfg -a1")
This can be done while asterisk is running. Sometimes this can be useful
to reset individual PRI/BRI spans.
0.3.0-PRE-1v
- fixed segfault introduced by 0.3.0-PRE-1u in chan_sip
0.3.0-PRE-1u asterisk 1.2.13
0.3.0-PRE-1t
- fixed "removed "callstate fix" fix
- added experimental support to display the correct callerid on a SNOM phone
after making an attended transfer to that phone
- introducing res_esel (Extension State Export Logic)
- updated to asterisk 1.2.12.1, zaptel 1.2.9.1
0.3.0-PRE-1s
- added "FASTBUSYONBUSY" Makefile option to libpri
- fixed "BAD CRC" error on layer 1 activation in TE mode
- added hardware bridging support to qozap
- moved debugging messages in qozap to debug > 0
- updated to asterisk 1.2.10
- libgsmat fix for "network registration rejected"
- removed "callstate fix"
- autoanswer "exception flag" fix
- MG2 is now the default echo canceler (zaptel)
- audio will not pass through userspace if transfercapablity is digital (app_dial)
- hangup cause fix when using priindication=passthrough
0.3.0-PRE-1r
- fixed devicestates for peer register/unregister
- moved extensionstates to AMI privilege "extensionstatus"
- added "Hint:" to extensionstatus event (AMI)
- added "gsm [no] debug channel <channel>" to chan_zap
- modified autoanswer to re-register dynamic extensions on reload
- fixed drivers to compile with latest 2.6 kernels
- improved libpri/chan_zap communication for BRI_CPE and BRI_CPE_PTMPfli4l 3.1.0 # asterisk -r
Asterisk 1.2.14-BRIstuffed-0.3.0-PRE-1x, Copyright (C) 1999 - 2006 Digium, Inc. and others.
Created by Mark Spencer <[email protected]>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'show license' for details.
=========================================================================
Connected to Asterisk 1.2.14-BRIstuffed-0.3.0-PRE-1x currently running on fli4l (pid = 7482)
fli4l*CLI>[channels]
language=de
switchtype=euroisdn
signalling=bri_net_ptmp
pridialplan=local
echocancel=yes
immediate=no
;setcallerid(""<${CALLERIDNUM}>)
overlapdial=yes
group=1
context=default
channel=>1-2
;
;signalling = bri_cpe_ptmp
;group = 2
;channel => 4-5
