Hi folks, This is what I am doing at this time :
exten => _XXXX,1,TrySystem(..command that sends a jabber message..)
exten => _XXXX,2,Set(calling=${EXTEN:0:4})
exten => _XXXX,3,ChanIsAvail(SIP/[EMAIL PROTECTED])
exten => _XXXX,4,Dial(SIP/[EMAIL PROTECTED],15,tr)
exten => _XXXX,5,Goto(_XXXX-${DIALSTATUS},1)
exten => _XXXX,104,Dial(Zap/g2/${calling},15,tr)
exten => _XXXX,105,Goto(_XXXX-${DIALSTATUS},1)
exten => _XXXX-NOANSWER,1,Voicemail(u${calling})
exten => _XXXX-NOANSWER,2,Hangup
exten => _XXXX-CHANUNAVAIL,1,Voicemail(u${calling})
exten => _XXXX-CHANUNAVAIL,2,Hangup
exten => _XXXX-CONGESTION,1,Voicemail(u${calling})
exten => _XXXX-CONGESTION,2,Hangup
exten => _XXXX-BUSY,1,Voicemail(b${calling})
exten => _XXXX-BUSY,2,Hangup
exten => _XXXX-CANCEL,1,Voicemail(u${calling})
exten => _XXXX-CANCEL,2,Hangup
Long story short, Asterisk lets me know if the SIP users has their SIP
phone on. If it is, I call it. Otherwise I call their POTS.
Probably should consider calling the POTS if SIP does not answer in timeout time, but that's for another day :).
Thanks everyone for helping out along the way, JES C F wrote:
You can try one more thing, and that is the M option, and create a macro that announces to the user to accept the call..... as documented at: http://www.voip-info.org/wiki-asterisk+cmd+dial On 11/23/05, James MacLean <[EMAIL PROTECTED]> wrote:Oh boy :(. As Roman politely explained in a private email... I was using ports 1 and 2 thinking they were the outbound fxs ports :(. That's it, these glasses are going, and no more testing from home :). When I switched to testing with ports 3 and 4, everything worked the same as G2. Not of course as cute as what I had hoped for when I see the local telco can do something like "Dial(ZAP/g2/8888&SIP/[EMAIL PROTECTED])" and have it wait 'til the correct phone is answered :(. Thanks to C F for the "c" option but my goal was to just have the 4 digit number call folks with and without SIP. I would not expect users to know to press #. I don't think dvlinedetect will quite cut it either. callprogress looked promising, but, alas, as many others have found, it hangs up after timeout seconds. I'll keep digging :). Thanks again everyone, JES James B. MacLean wrote:Hi C F, I am not well versed in this level of telephony or Asterisk, so please bare with me :). My setup is really typical. Bought the digium card with 4 ports. 2 fxs / 2 fxo. The 2 fxo's are connected directly to phones, belong to group 1 according to zapata.conf, and exist as "fxoks=1-2" in /etc/zaptel.conf. The 2 fxs ports are connected to the telco, belong to group 2 according to zapata.conf, and are setup as "fxsks=3-4" in zaptel.conf. Dial(Zap/1/8888&SIP/[EMAIL PROTECTED],15,r) works as expected, Dial(Zap/2/8888&SIP/[EMAIL PROTECTED],15,r) works as expected but: Dial(Zap/g2/8888&SIP/[EMAIL PROTECTED],15,r) Rings once and reports answered to Asterisk. Does this support what you are explaining? I'm honestly confused by how an fxs module operates as an fxo module? Thanks for any more direction you might have, JES
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