[Asterisk-Users] IVR setup problems

[Alex]

Hi guys still have the problem to setup the IVR correctly.
 
I am forwarding call from ser :

if (method == "INVITE") { 
    if (uri =~ "sip:1[0-9]{10}@*"){ 
        log(1, "Forwarding to Asterisk\n"); 
        rewritehostport("xxx.xxx.xxx.xxx:5061"); 
        t_relay(); 
        break; 
    } 
 } 
 
inside sip.conf
-------------------------------------------------------------------------------------
port=5061                 
bindaddr=0.0.0.0               
srvlookup=yes 
 
[ser]
type=peer
host=xxx.xxx.xxx.xxx
context=ser1
 
inside extensions.conf
-------------------------------------------------------------------------------------
[ser1]
Exten => 40,1,Answer
Exten => 40,2,SetMusicOnHold(default)
Exten => 40,3,DigitTimeout,5
Exten => 40,4,ResponseTimeout,10
Exten => 40,5,Background(greeting)
Exten => 1,1,Playback(secr) ; if you press <91>1<92> playback message <93>secr<94>
Exten => 1,2,Dial(SIP/Phone1/20)
Exten => 2,1,Playback(studentservice)
Exten => 2,2,Dial(SIP/Phone1/20)
Exten => 3,1,Playback(it)
Exten => 3,2,Dial(SIP/Phone1/20)
Exten => 4,1,Playback(operator)
Exten => 4,2,Dial(SIP/Phone1/20)
 
 
Inside asterisk debug i see what the forwarding of the call working :
log of ASTERISK DEBUG
----------------------------------------------------------------------------------------
Sip read: 
INVITE sip:1phoneiamcalling at xxx.xxx.xxx.xxx:5061 SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx;branch=z9hG4bKb148.00624e85.0
Via: SIP/2.0/UDP ipoftphone:5060;branch=z9hG4bK06ffef7d
From: "Alexg" <sip:phonesoftphone at xxx.xxx.xxx.xxx>;tag=00036b09607e0047524bda98-4b96b81e
To: <sip:1phoneiamcalling at xxx.xxx.xxx.xxx>
Call-ID: 00036b09-607e0047-3dc1c568-1d31a410 at ipoftphone
CSeq: 101 INVITE
User-Agent: CSCO/6
Contact: <sip:phonesoftphone at ipoftphone:5060>
Expires: 180
Content-Type: application/sdp
Content-Length: 249
Accept: application/sdp
v=0
o=Cisco-SIPUA 28416 11732 IN IP4 ipoftphone
s=SIP Call
c=IN IP4 ipoftphone
t=0 0
m=audio 26298 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
13 headers, 11 lines
Using latest request as basis request
Sending to xxx.xxx.xxx.xxx : 5060 (non-NAT)
Found peer 'ser'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port ipoftphone:26298
Found description format PCMU
Found description format PCMA
Found description format G729
Found description format telephone-event
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723)
Looking for 1phoneiamcalling in ser1
Reliably Transmitting (no NAT):
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx;branch=z9hG4bKb148.00624e85.0
Via: SIP/2.0/UDP ipoftphone:5060;branch=z9hG4bK06ffef7d
From: "Alexg" <sip:phonesoftphone at xxx.xxx.xxx.xxx>;tag=00036b09607e0047524bda98-4b96b81e
To: <sip:1phoneiamcalling at xxx.xxx.xxx.xxx>;tag=as125ae8d3
Call-ID: 00036b09-607e0047-3dc1c568-1d31a410 at ipoftphone
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:1phoneiamcalling at xxx.xxx.xxx.xxx:5061>
Content-Length: 0

 to xxx.xxx.xxx.xxx:5060

Sip read: 
ACK sip:1phoneiamcalling at xxx.xxx.xxx.xxx:5061 SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx;branch=z9hG4bKb148.00624e85.0
From: "Alexg" <sip:phonesoftphone at xxx.xxx.xxx.xxx>;tag=00036b09607e0047524bda98-4b96b81e
Call-ID: 00036b09-607e0047-3dc1c568-1d31a410 at ipoftphone
To: <sip:1phoneiamcalling at xxx.xxx.xxx.xxx>;tag=as125ae8d3
CSeq: 101 ACK
User-Agent: Sip EXpress router(0.8.14 (i386/linux))
Content-Length: 0

8 headers, 0 lines
Destroying call '00036b09-607e0047-3dc1c568-1d31a410 at ipoftphone'
 
 
var/log/asterisk/messages
---------------------------------------------------------------------------------------
Unable to open /dev/dsp: No such device
 
 
I am calling to number 12222222222 and ser forwarding it to the asterisk (port 5061) (see configuration of sip.conf) to the ser1 context.
in extensions.conf i have ser1 context and extensions for ivr under ser1 context.
After the call i am hearing the busy line and that's it. i tried to play with extensions.conf with no success.
I need a help to setup the IVR system.
 
Thanks.

		
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