[Asterisk-Users] redirecting incoming calls to external phone (cell)

[Jay Christopherson]

Hi-

I am attempting to setup Asterisk for the first time, and I think I am about 
99% there.   I am using vonage softphone, and want to use asterisk to 
redirect incoming calls to my cell phone primarily, and maybe other remote 
lines.

Right now, I am able to register with vonage, and trap incoming calls.  The 
only issue I have is that I don't think my syntax in extensions.conf is 
correct to dial out to my cell:

XXXXXXXXXX --> my vonage softphone number
YYYYYYYYYY-> my cell phone number


sip.conf:
[sphone.vopr.vonage.net]
username=1XXXXXXXXXX
port=5060
nat=yes
type=friend
secret=XXXXX
host=sphone.vopr.vonage.net
fromuser=1XXXXXXXXXX
fromdomain=sphone.vopr.vonage.net
dtmfmode=rfc2833
auth=md5
canreinvite=no
context=out
;
[vonage-in]
username=1XXXXXXXXXX
type=friend
port=5060
nat=yes
secret=21V9bkQ5MR
host=sphone.vopr.vonage.net
insecure=very
fromuser=1XXXXXXXXXX
fromdomain=sphone.vopr.vonage.net
context=in
canreinvite=no
auth=md5

extensions.conf
[in]
;exten => s,1,Dial(SIP/1YYYYYYYYYY,25,rt)
;exten => s,2,Ringing()
;exten => s,3,Wait(60)
exten => _1XXXXXXXXXX,1,dial(sip/1YYYYYYYYYY,20,r)

[out]
exten => _1NXXNXXXXXX,1,Dial(SIP/1XXXXXXXXXX at sphone.vopr.vonage.net,25,rt)

Right now, when I get an incoming, this is the message I get:

From: "ZZZ-ZZZ-ZZZZ" 
<sip:1XXXXXXXXXX at atlas4.atlas.vonage.net:5061;user=phone>;tag=1939305037
To: <sip:1XXXXXXXXXX at atlas4.atlas.vonage.net:5061;user=phone>
Call-ID: 
94530e40-5188-1130028473-379073938-128603415204566900000000-1 at 192.168.100.100
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:1XXXXXXXXXX at 192.168.100.135>
Content-Length: 0

to 216.115.25.198:5060
    -- Executing Dial("SIP/1XXXXXXXXXX-6e16", "sip/1YYYYYYYYYY|20|r") in new 
stack
Oct 23 00:58:20 WARNING[6933]: chan_sip.c:1401 create_addr: No such host: 
1YYYYYYYYYY
Destroying call '7c1a970f409070f07f8a6f03048e0f82 at 127.0.0.1'
Oct 23 00:58:20 NOTICE[6933]: app_dial.c:764 dial_exec: Unable to create 
channel of type 'sip'
  == Everyone is busy/congested at this time
Oct 23 00:58:30 WARNING[6933]: pbx.c:1948 ast_pbx_run: Timeout, but no rule 
't' in context 'in'

Thanks-
Jay