search for: incoming_pstn

...OK 0 0 0 T2XXP (PCI) Card 0 Span 2 RED 0 0 0 Span 2 is expected to be down as we don't have it connected to anything. asterisk*CLI> zap show channels Chan Extension Context Language MOH Interpret 1 incoming_pstn default 2 incoming_pstn default 3 incoming_pstn default 4 incoming_pstn default 5 incoming_pstn default 6 incoming_pstn default 7...

...fxo card or zaptel cards in general? I'm running Asterisk 1.0.2-2 on Debain Linux (with zaptel 1.0.0-2) and i've got the modem connected to an NTL phone line in the UK. Here is my config's: /etc/zaptel.conf: fxsls=1 loadzone=uk defaultzone=uk /etc/zapata.conf: [channels] context=incoming_pstn signalling=fxs_ls usecallerid=yes echocancel=yes echocancelwhenbridged=yes echotraining=yes rxgain=0.0 txgain=0.0 channel => 1 ztcfg -vvv: Zaptel Configuration ====================== Channel map: Channel 01: FXS Loopstart (Default) (Slaves: 01) 1 channels configured. Any...

...something else is on the line before picking up on this channel? For example, I dont want to pick up and dial out on the line if someone is on it using another phone (which is connected directly to the line, rather than through Asterisk). Also, when an incoming call comes in, i've got this: [incoming_pstn] exten => s,1,NoOp(${CALLERID} calling on Zap/1) exten => s,2,Wait,15 exten => s,3,Answer exten => s,4,Voicemail(su1) exten => s,5,Hangup I really want to wrap the Answer, Voicemail and Hangup commands in some kind of if statement so if Asterisk didn't detect that the...

...er.com A tcpdump clearly shows the INVITE coming in, but asterisk seems to be ignoring it (theres no reply outbound packet). All the source/dest IPs and ports look good. A "sip set debug trace ip <sourceip>" is blank, showing nothing at all. The sip.conf default context is incoming_pstn. The incoming_pstn context is: [incomming_pstn] include => local-UK include => local-US include => test_numbers and [test_numbers] includes: exten => 555,1,Answer(0) ; Pick up phone instantly exten => 555,n,Playback(vq51) ; Let them know...

Hi All, I was made aware today that some of my calls coming in are not going to voicemail... Below are some logs, and the macro that should run on the incoming_pstn context for that extension. I can see that theres a non-zero exit before it gets to voicemail, but I've no idea why. In this case theres 2 SIP clients to sim-call. On other occasions it works fine. In the CDR logs, I can see "NO ANSWER" and "ANSWERED" - what would be the...