[Asterisk-Users] TDM-400 and dialplan -- how to ring a SIP extension *before* answering the PSTN line?
Hi, folks: Okay, so here's an idea. I have a TDM-400 card with an FXO card in it connected to the PSTN and a Polycom IP 501 phone. Observe the following simple dialplan for illustration: [incoming] ; incoming calls from the FXO port are directed to this context from zapata.conf exten = s,1,Answer() exten = s,2,Dial(SIP/polycom) And zapata.conf: [trunkgroups] ; define any trunk groups [channels] ; hardware channels ; default usecallerid=yes hidecallerid=no callwaiting=no threewaycalling=yes transfer=yes echocancel=yes echotraining=yes callprogress=yes ; define channels context=incoming signalling=fxs_ks channel = 4 Pretty straightforward stuff -- a call comes in on the PSTN line, the Asterisk answers the call, then rings the extension. The caller hears a ring tone throughout the entire process. The rub is that Asterisk has, in reality, taken the PSTN line off hook. Not great if the caller is at a payphone. What if nobody answers the extension? The caller is out his money (50 cents in most of the US, 35 cents in Alberta and 25 cents in the rest of Canada ;) ) So I had the idea of doing things a bit differently, like so: [incoming] ; incoming calls from the FXO port are directed to this context from zapata.conf exten = s,1,Dial(SIP/polycom) exten = s,2,Answer() This way, Asterisk dials the extension first, the idea being that when the SIP extension is answered, Asterisk answers the PSTN line and connects the channels. This did not have the expected result -- when I tried this, my SIP extension rang, but answering the extension did not result in Asterisk picking up the PSTN line. There is a way of doing this, isn't there? How can I make this work? Cheers, -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM-400 and dialplan -- how to ring a SIP extension *before* answering the PSTN line?
[incoming] ; incoming calls from the FXO port are directed to this context from zapata.conf exten = s,1,Answer() exten = s,2,Dial(SIP/polycom) Try this exten = s,1,Dial(SIP/polycom,20) exten = s,2,Hangup() I think this way, * won't answer the line until your SIP phone answers. If you don't pickup the phone after 20 seconds it will just ignore this incoming call hth ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM-400 and dialplan -- how to ring a SIP extension *before* answering the PSTN line?
Time Bandit wrote: [incoming] ; incoming calls from the FXO port are directed to this context from zapata.conf exten = s,1,Answer() exten = s,2,Dial(SIP/polycom) Try this exten = s,1,Dial(SIP/polycom,20) exten = s,2,Hangup() I think this way, * won't answer the line until your SIP phone answers. If you don't pickup the phone after 20 seconds it will just ignore this incoming call. Hmn. Very nice! It works! On the matter of timing -- Asterisk appears to wait two full PSTN rings before it dials the SIP extension. Is there any way we can tighten up this interval? Is that done in the Zap configuration? The driver? The dialplan? Cheers, -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM-400 and dialplan -- how to ring a SIP extension *before* answering the PSTN line?
Stephen Bosch wrote: Hi, folks: Okay, so here's an idea. I have a TDM-400 card with an FXO card in it connected to the PSTN and a Polycom IP 501 phone. Observe the following simple dialplan for illustration: [incoming] ; incoming calls from the FXO port are directed to this context from zapata.conf exten = s,1,Answer() exten = s,2,Dial(SIP/polycom) And zapata.conf: [trunkgroups] ; define any trunk groups [channels] ; hardware channels ; default usecallerid=yes hidecallerid=no callwaiting=no threewaycalling=yes transfer=yes echocancel=yes echotraining=yes callprogress=yes ; define channels context=incoming signalling=fxs_ks channel = 4 Pretty straightforward stuff -- a call comes in on the PSTN line, the Asterisk answers the call, then rings the extension. The caller hears a ring tone throughout the entire process. The rub is that Asterisk has, in reality, taken the PSTN line off hook. Not great if the caller is at a payphone. What if nobody answers the extension? The caller is out his money (50 cents in most of the US, 35 cents in Alberta and 25 cents in the rest of Canada ;) ) So I had the idea of doing things a bit differently, like so: [incoming] ; incoming calls from the FXO port are directed to this context from zapata.conf exten = s,1,Dial(SIP/polycom) exten = s,2,Answer() This way, Asterisk dials the extension first, the idea being that when the SIP extension is answered, Asterisk answers the PSTN line and connects the channels. This did not have the expected result -- when I tried this, my SIP extension rang, but answering the extension did not result in Asterisk picking up the PSTN line. There is a way of doing this, isn't there? How can I make this work? There is no need to include the answer in your dialplan. Without it, the call is processed by ringing the sip phone, and when that person answsers, an implied answer will occur back through the pstn. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM-400 and dialplan -- how to ring a SIP extension *before* answering the PSTN line?
Hmn. Very nice! It works! On the matter of timing -- Asterisk appears to wait two full PSTN rings before it dials the SIP extension. Is there any way we can tighten up this interval? Is that done in the Zap configuration? The driver? The dialplan? Asterisk is waiting for the CallerID, which is sent between the first and second ring. If you disable CallerID you will shorten the interval, but you will loose CallerID hth ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
