I am trying to get a GSM gateway (Alpha Tech GSM Gateway Blue Gate Dual Band Analoog FXO) working with Asterisk. I had a working FXO configuration to a analog port of a small home 1/4 ISDN pbx. I used this same configuration to connect a GSM Gateway that is supposed to be connected to the external(FXO) analog port of a pbx.
I can get my configuration to dial the mobile number via the gateway, but Asterisk never detects that the remote side (the mobile phone) answered. When I use the r Dial option it will continue to signal a ringing tone to the local extension. When I leave out the r, I do get a connection but only for the duration of the configured number of seconds the Dial command is waiting for a connection. Because it did not detect the call being connected it will stop with a NOANSWER status effectively dropping the connection. I want to try two possible solutions: 1) I try to find out how to configure the GSM gateway behave in such a way that Asterisk correctly detects the call being connected. 2) I configure Asterisk to work with this device. The problem is that I looked thru the configuration options of the gateway and I could not find anything that made sense to me in relation to this prb. What (how) is the gateway supposed to signal back that the call is connected? If I know what is needed I can go back to my supplier with the right questions. If the gateway can not be configured properly, I want to know how I could configure Asterisk to work around this problem and make it work anyway. This is the zapata.conf for the FXO port: [channels] ; hardware channels ; default usecallerid=yes hidecallerid=no callwaiting=no threewaycalling=yes transfer=yes echocancel=yes echotraining=yes immediate=no flash=100 hanguponpolarityswitch=yes callprogress=yes progzone=nl busydetect=yes busycount=4 language=nl ; define channels context=binnenkomend signalling=fxs_ks channel => 1 ; pstn attached to port 1 This is the Dialplan fragment: exten => _87.,1,Wait(0.5) exten => _87.,n,Dial(Zap/1/${EXTEN:2},30,rtT) exten => _87.,n,Macro(fastbusy) exten => _87.,n,Hangup exten => _87.,102,Playback(tt-allbusy) [macro-fastbusy] exten => s,1,Answer exten => s,n,Wait,1 exten => s,n,Playback(vm-isunavail) exten => s,n,Wait(3) exten => s,n,Hangup -- Executing [EMAIL PROTECTED]:1] Wait("SIP/kimura1-08236550", "0.5") in new stack -- Executing [EMAIL PROTECTED]:2] Dial("SIP/kimura1-08236550", "Zap/1/0623027714|30|rtT") in new stack -- Called 1/0623027714 -- Nobody picked up in 30000 ms -- Hungup 'Zap/1-1' -- Executing [EMAIL PROTECTED]:3] Macro("SIP/kimura1-08236550", "fastbusy") in new stack -- Executing [EMAIL PROTECTED]:1] Answer("SIP/kimura1-08236550", "") in new stack -- Executing [EMAIL PROTECTED]:2] Wait("SIP/kimura1-08236550", "1") in new stack -- Executing [EMAIL PROTECTED]:3] Playback("SIP/kimura1-08236550", "vm-isunavail") in new stack -- <SIP/kimura1-08236550> Playing 'vm-isunavail' (language 'nl') -- Executing [EMAIL PROTECTED]:4] Wait("SIP/kimura1-08236550", "3") in new stack -- Executing [EMAIL PROTECTED]:5] Hangup("SIP/kimura1-08236550", "") in new stack == Spawn extension (macro-fastbusy, s, 5) exited non-zero on 'SIP/kimura1-08236550' in macro 'fastbusy' == Spawn extension (macro-fastbusy, s, 5) exited non-zero on 'SIP/kimura1-08236550' Any help would be appreciated. Hans Feringa _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users