[Asterisk-Users] forward incoming analog call to SIP?

[Mark Phillips]

Try changing [analog] to this

[analog]
include => test
include => local

exten => s,1,Answer ; Answer the call so we know its getting into *
exten => s,2,Playback(transfer) ; Tell caller pbx is working
exten => s,3,Dial(SIP/1237) ; transfer call to extension 1237

You have not allowed the ZAp software to dump the call into a dialplan 
it can deal with. In your setup the call comes in and is passed to the 
[analog] context but cannot be dealt with because the ZAP software 
doesn't know how to give the call to a particular extension. Adding the 
above line should cure that.

Mark

Dave Williams wrote:
> I'm trying to setup a demo where my Asterisk box with a TDM01B (FXO) 
> answers an incoming call and forwards that call to a SIP softphone 
> (X-lite.)
> 
> Seems all is built/installed okay:
> 
> # ztcfg -vv
> Zaptel Configuration
> ======================
> Channel map:
> Channel 01: FXS Kewlstart (Default) (Slaves: 01)
> 1 channels configured.
> 
> I'm pretty new at this and the extensions.conf file is eating my lunch. 
> Here are my various config files - maybe someone will take pitty on me 
> and point me in the right direction. Needless to say, Asterisk pukes on 
> my dialplan when I try and startup. .
> 
> (zapata.conf)
> context=analog
> signalling=fxs_ks
> language=en
> channel => 1
> 
> (sip.conf)
> [sip_proxy]
> For incoming calls only. Example: FWD (Free World Dialup)
> type=user
> context=sip
> 
> [xlite1]
> "Transmit Silence"=YES
> type=friend
> regexten=1234                 ; When they register, create extension 1234
> username=xlite1
> callerid="Jane Smith" <5678>
> host=dynamic
> allow=ulaw
> allow=alaw
> 
> 
> (extensions.conf)
> [general]
> static=yes
> writeprotect=no
> 
> [analog]
> include=>test
> include=>local
> 
> [sip]
> include=>test
> include=>local
> 
> [test]
> 611,1,echo_test
> 
> [local]
> exten => 1237,1,Dial(SIP/xlite1,10,t)
> 
> 
> 
> 
> 
> 
> 
> 
> 
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-- 

Mark, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com