[Asterisk-Users] Dial ZAP with group (g2) erroneously says call answered when it is still ringing - workaround

[James B. MacLean]

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=${EXTEN:0:4})
exten => _XXXX,3,ChanIsAvail(SIP/${calling}@${calling})
exten => _XXXX,4,Dial(SIP/${calling}@${calling},15,tr)
exten => _XXXX,5,Goto(_XXXX-${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 <macleajb at ednet.ns.ca> 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/8888 at 8888)" 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/8888 at 8888,15,r) works as expected,
>>>Dial(Zap/2/8888&SIP/8888 at 8888,15,r) works as expected
>>>
>>>but:
>>>
>>>Dial(Zap/g2/8888&SIP/8888 at 8888,15,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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