[Asterisk-Users] Getting asterisk to work with callthroughs?

[MF Hulber]

Ok, first I'll tell you some of the things I'm ignoring because you said 
you are having trouble receiving the inbound call.  First, why aren't 
you using DISA?  Ok, so you want to try this out, that's fine.  Second, 
it appears you set the variable NR to be empty so I don't know why you 
are using it in the first place.  Third, your extension to capture the 
outgoing number is _x so you will only capture 1 digit input.  You need 
_x. to catch anything 1 digit or longer. 

So with that all said, why don't you isolate the problem to why you are 
aren't receiving inbound calls from your SIP provider?  Do you have a 
registration statement?  Without that you aren't going to get a thing.

Why don't use browse through these pages:

http://www.sipgate.co.uk/faq/index.php

Can you make a call through Sipgate?

MARK.

Huw Morgan wrote:

> Hi,
>
> Firstly, what I'm trying to do is:
> * Get asterisk to pick up a SIP call via a DID
> * Prompt the user
> * When the user puts in a number, go to IAX.conf and route it 
> according to what I've specified there, i.e Least Cost Routing, etc.
>
> I've set-up something similar to what I've found online, but it 
> doesn't work! Asterisk doesn't pick up the call at all..... :(
>
> The files I used:
>
> sip.conf (for the DID)
>
> [general]
> context=default
> recordhistory=yes
> port=5060
> bindaddr=0.0.0.0
> srvlookup=yes
> tos=lowdelay
> maxexpirey=3600
> defaultexpirey=120
> allow=ulaw
> allow=alaw
> musicclass=default
> language=en
> relaxdtmf=yes
> rtptimeout=60
> trustrpid = no
> progressinband=yes
> useragent=Asterisk PBX
> promiscredir = no
>
> [incoming]
> ; For incoming calls only.
> type=user
> username=xxxxxx
> secret=xxxxxxxx
> host=sipgate.co.uk
> fromuser=xxxxxx
> fromdomain=sipgate.co.uk
> authuser=xxxxxxx
> dtmfmode=info
> context=from-sip
> insecure=very
> disallow=all
> allow=ulaw
> allow=alaw
>
>
> iax.conf (for the peers/terminating services)
> Can paste this in if it is relevant, although I THINK it's working as 
> it shows them registered ok on the CLI.
>
>
> extensions.conf extract - how I'm routing the calls
>
> [globals]
> ${OUTGOING-NUM}=XXXX
>
> [general]
> static=yes
> writeprotect=no
> [from-sip]
> exten => _NXXNXXXXXX,1,Answer
> exten => _NXXNXXXXXX,2,Background(vm-password)
> exten => _NXXNXXXXXX,3,Authenticate(123)
> exten => _NXXNXXXXXX,4,Playback(beep)
> exten => _NXXNXXXXXX,5,SetVar(NR=)
> exten => _NXXNXXXXXX,6,Goto(testdtmf|s|1)
>
> [testdtmf]
> exten => s,1,SetVar(NR=)
> exten => s,2,Background(pls-entr-num-uwish2-call)
> exten => s,3,Background(and-prs-pound-whn-finished)
> exten => s,4,Background(beep)
> exten => s,5,WaitExten(10)
> exten => _x,1,SetVar(NR=${NR}${EXTEN})
> exten => _x,2,NoOp(${NR})
> exten => _x,3,Goto(testdtmf|s|5)
> exten => _#,1,Goto(verifynumber|s|1)
> exten => i,1,Goto(testdtmf|s|1)
> exten => t,1,Hangup
>
> [verifynumber]
> exten => s,1,Background(you-dialed)
> exten => s,2,SayDigits(${NR})
> exten => s,3,Background(if-correct-press)
> exten => s,4,Background(pound)
> exten => s,5,Background(otherwise-press)
> exten => s,6,Background(star)
> exten => _#,1,Background(pls-wait-connect-call)
> exten => _#,2,Dial(IAX2/${OUTGOING-NUM}@voxee/${NR},30)
> exten => _#,3,Background(something-terribly-wrong);
> exten => _#,4,Background(goodbye)
> exten => _#,5,Hangup
> exten => _*,1,Goto(testdtmf|s|1)
>
> -- 
>
> Any ideas why Asterisk is NOT picking up the SIP call.... And any 
> pointers where I've gone wrong?
>
> Thanks in advance!
>
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