This is wonderful work, thank you.
Unfortunately I've been unable to get ztdummy to work for IAX2 trunking (haven't tried meetme, as I'm using G729 codecs). This is on a WL500g Deluxe running WhiteRussian RC6.
When Asterisk starts I get the following message:
Nov 20 12:00:39 WARNING[1396]: chan_iax2.c:8293 build_peer: Unable to support trunking on peer 'upstream' without zaptel timing
I have loaded the ztdummy module, and it loads without errors, but it doesn't seem to be used: here's lsmod output taken while Asterisk is running in the background.
root@OpenWrt:/etc/asterisk# lsmod
Module Size Used by Tainted: P
ztdummy 1832 0 (unused)
zaptel 212512 0 [ztdummy]
usb-uhci 27180 0 [ztdummy]
usbcore 74808 1 [usb-uhci]
wlcompat 15520 0 (unused)
wl 423640 0 (unused)
switch-robo 4460 0 (unused)
switch-core 4896 0 [switch-robo]
diag 18176 0 (unused)
root@OpenWrt:~# ls -l /dev/zap
crw-rw-rw- 1 root root 196, 254 Jan 1 1970 channel
crw-rw-rw- 1 root root 196, 0 Jan 1 1970 ctl
crw-rw-rw- 1 root root 196, 255 Jan 1 1970 pseudo
drwxr-xr-x 1 root root 0 Jan 1 1970 span1
crw-rw-rw- 1 root root 196, 253 Jan 1 1970 timer
I'm using the asterisk-mini_1.2.13-1_mipsel.ipk installation, and apart from IAX2 trunking it's working fine. (Actually it's so minimalist that it doesn't even have app_dial.so, so I copied that out of the full-blown asterisk ipkg)
# ls /usr/lib/asterisk/modules/
app_dial.so codec_gsm.so format_wav_gsm.so
app_meetme.so codec_ulaw.so pbx_config.so
chan_iax2.so format_gsm.so res_features.so
chan_local.so format_pcm.so res_musiconhold.so
chan_sip.so format_wav.so
OpenWrt*CLI> show modules
Module Description Use
Count
res_musiconhold.so Music On Hold Resource 1
res_features.so Call Features Resource 1
pbx_config.so Text Extension Configuration 0
chan_iax2.so Inter Asterisk eXchange (Ver 2) 0
chan_local.so Local Proxy Channel 0
chan_sip.so Session Initiation Protocol (SIP) 0
format_wav_gsm.so Microsoft WAV format (Proprietary GSM) 0
codec_ulaw.so Mu-law Coder/Decoder 0
codec_gsm.so GSM/PCM16 (signed linear) Codec Translat 0
format_gsm.so Raw GSM data 0
app_dial.so Dialing Application 0
app_meetme.so MeetMe conference bridge 0
format_pcm.so Raw uLaw 8khz Audio support (PCM) 0
format_wav.so Microsoft WAV format (8000hz Signed Line 0
14 modules loaded
So I'm wondering if there's something else I need to do, or some other module I need to load, in order to activate ztdummy.
I generated a dummy meetme.conf but that doesn't seem to help:
In case they're any use, here are my config files. I emptied out /etc/asterisk except for asterisk.conf, logger.conf and modules.conf, and then created the following:
--- iax.conf ---
[general]
bandwidth=low
allow=g729
tos=ef
autokill=yes
jitterbuffer=no
forcejitterbuffer=no
trunktimestamps=yes
[upstream]
type=peer
username=venue
secret=XXXXXXXX
auth=md5
host=X.X.X.X
qualify=yes
trunk=yes
--- sip.conf ---
; This is purposely an "insecure" configuration which allows any SIP user
; to place outbound calls through us without authentication!
[general]
context=outbound
allowexternaldomains=yes
bindport=5060
bindaddr=192.168.1.1
disallow=all
allow=g729
allow=gsm
--- extensions.conf ---
[general]
autofallthrough=no
[outbound]
exten => _X.,1,Dial(iax2/upstream/${EXTEN},45)
Many thanks, Brian.
