Scenario, I got some very good help earlier from Joseph getting me up and started but I have a couple of small problems still. Setup: FC2 Asterisk 1.0 Zaptel 1.0, TMP400P, FXS Port 1 FXO Port 4 Analog dialout line and Analog handset plugged in. Problems: 1. Incoming calls work and the phone rings and can be answered no problems, (although I wouldn't mind being able to adjust the ring but that's not important), I can't ring out, I just get a busy signal and nothing comes up on the console. I am pretty sure its just a simple line missing from extensions.conf. 2. I am based in australia and when I have an incoming call with callerid turned on then I get the following error on console. -- Zap/1-1 is ringing Sep 25 22:49:14 WARNING[-203428944]: chan_zap.c:3413 zt_handle_event: Didn't finish Caller-ID spill. Cancelling. ----------------------------------------------- /etc/zaptel.conf fxols=1 fxsls=4 loadzone=au /etc/asterisk/extensions.conf [pstn] exten => s,1,NoOp(Comment Only: Call from ${CALLERIDNUM}) ; Just put a comment in the CLI for info. exten => s,2,Dial(Zap/g1,45,t) ;Dial the group=1 zap card mod above #exten => s,3,VoiceMail(u100) ;Whatever box you want. [internal] exten => i,1,Playback(invalid) exten => i,2,Hangup exten => t,1,Hangup exten => 099,1,Echo ;simple echo test /etc/asterisk/zapata.conf [channels] context=default switchtype=national usecallerid=yes hidecallerid=no callwaiting=no usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 group=1 signalling=fxo_ls callgroup=1 pickupgroup=1 immediate=no context=internal busydetect=yes callerid="James Bean<690>" ;assuming extension 690 mailbox=690 ;stutter tone for voicemail - you can use an optional context here transfer=yes channel=>1 group=2 signalling=fxs_ls context=pstn channel=>4 ----------------------------------------------- Any help would be very much appreciated. James
> Incoming calls work and the phone rings and can be answered > no problems, (although I wouldn't mind being able to adjust > the ring but that's not important), I can't ring out, I just > get a busy signal and nothing comes up on the console. I am > pretty sure its just a simple line missing from extensions.conf.In your [internal] context try something like.. exten => _0.,1,Answer exten => _0.,2,Dial(Zap/g1/${EXTEN:1}) exten => _0.,3,Hangup This way Asterisk will send all the digits dialled after the 0 to the zaptel card and you should be dialing out. You may not need the answer/hangup lines for your setup.> 2. > > I am based in australia and when I have an incoming call with > callerid turned on then I get the following error on console. > > -- Zap/1-1 is ringing > Sep 25 22:49:14 WARNING[-203428944]: chan_zap.c:3413 zt_handle_event: > Didn't finish Caller-ID spill. Cancelling.I'm not sure if this is related with inbound CallerID on an FXO, but to get Caller ID working on an FXS port I had to make this change to the chan_zap.c file and recompile:- http://lists.digium.com/pipermail/asterisk-users/2004-August/057349.html In /usr/src/asterisk/channels/chan_zap.c #define DEFAULT_CIDRINGS 2 The default is 1.. Seems we need this set to 2 in Australia, I dare say making this change might get the inbound caller ID working for you also. Hope this helps, Chris Lee
I don't see anything posted here in extensions.conf to allow dialing out on group 2. You need something like this: [outgoing] exten => _9X.,1,Dial(Zap/g2/${EXTEN:1}) exten => _9X.,2,Congestion() And add the context outgoing to those extensions that you allow to dial out to the PSTN. Lyle ----- Original Message ----- From: "James Bean" <james@hdcs.com.au> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Sent: Saturday, September 25, 2004 8:28 AM Subject: [Asterisk-Users] Help with dialing out with TDM400P Scenario, I got some very good help earlier from Joseph getting me up and started but I have a couple of small problems still. Setup: FC2 Asterisk 1.0 Zaptel 1.0, TMP400P, FXS Port 1 FXO Port 4 Analog dialout line and Analog handset plugged in. Problems: 1. Incoming calls work and the phone rings and can be answered no problems, (although I wouldn't mind being able to adjust the ring but that's not important), I can't ring out, I just get a busy signal and nothing comes up on the console. I am pretty sure its just a simple line missing from extensions.conf. 2. I am based in australia and when I have an incoming call with callerid turned on then I get the following error on console. -- Zap/1-1 is ringing Sep 25 22:49:14 WARNING[-203428944]: chan_zap.c:3413 zt_handle_event: Didn't finish Caller-ID spill. Cancelling. ----------------------------------------------- /etc/zaptel.conf fxols=1 fxsls=4 loadzone=au /etc/asterisk/extensions.conf [pstn] exten => s,1,NoOp(Comment Only: Call from ${CALLERIDNUM}) ; Just put a comment in the CLI for info. exten => s,2,Dial(Zap/g1,45,t) ;Dial the group=1 zap card mod above #exten => s,3,VoiceMail(u100) ;Whatever box you want. [internal] exten => i,1,Playback(invalid) exten => i,2,Hangup exten => t,1,Hangup exten => 099,1,Echo ;simple echo test /etc/asterisk/zapata.conf [channels] context=default switchtype=national usecallerid=yes hidecallerid=no callwaiting=no usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 group=1 signalling=fxo_ls callgroup=1 pickupgroup=1 immediate=no context=internal busydetect=yes callerid="James Bean<690>" ;assuming extension 690 mailbox=690 ;stutter tone for voicemail - you can use an optional context here transfer=yes channel=>1 group=2 signalling=fxs_ls context=pstn channel=>4 ----------------------------------------------- Any help would be very much appreciated. James _______________________________________________ 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
--On Saturday, September 25, 2004 23:28 +1000 James Bean <james@hdcs.com.au> wrote:> 1. > Incoming calls work and the phone rings and can be answered no problems, > (although I wouldn't mind being able to adjust the ring but that's not > important), I can't ring out, I just get a busy signal and nothing comesindications.conf -- you can adjust that and then set your country setting to the au country....i can't remember how to do the latter right now, but that should get you going in the right direction methinks.