<DIV>Hi guys still have the problem to setup the IVR correctly.</DIV>
<DIV> </DIV>
<DIV><STRONG>I am forwarding call from ser :</STRONG></DIV>
<DIV><BR>if (method == "INVITE") { <BR> if (uri =~ "sip:1[0-9]{10}@*"){ <BR> log(1, "Forwarding to Asterisk\n"); <BR> rewritehostport("xxx.xxx.xxx.xxx:5061"); <BR> t_relay(); <BR> break; <BR> } <BR> } </DIV>
<DIV> </DIV>
<DIV><STRONG>inside sip.conf</STRONG></DIV>
<DIV>-------------------------------------------------------------------------------------</DIV>
<DIV>port=5061 <BR>bindaddr=0.0.0.0 <BR>srvlookup=yes </DIV>
<DIV> </DIV>
<DIV>[ser]<BR>type=peer<BR>host=xxx.xxx.xxx.xxx<BR>context=ser1</DIV>
<DIV> </DIV>
<DIV><STRONG>inside extensions.conf</STRONG></DIV>
<DIV>-------------------------------------------------------------------------------------</DIV>
<DIV>[ser1]<BR>Exten => 40,1,Answer<BR>Exten => 40,2,SetMusicOnHold(default)<BR>Exten => 40,3,DigitTimeout,5<BR>Exten => 40,4,ResponseTimeout,10<BR>Exten => 40,5,Background(greeting)</DIV>
<DIV>Exten => 1,1,Playback(secr) ; if you press <91>1<92> playback message <93>secr<94><BR>Exten => 1,2,Dial(SIP/Phone1/20)</DIV>
<DIV>Exten => 2,1,Playback(studentservice)<BR>Exten => 2,2,Dial(SIP/Phone1/20)</DIV>
<DIV>Exten => 3,1,Playback(it)<BR>Exten => 3,2,Dial(SIP/Phone1/20)</DIV>
<DIV>Exten => 4,1,Playback(operator)<BR>Exten => 4,2,Dial(SIP/Phone1/20)</DIV>
<DIV> </DIV>
<DIV> </DIV>
<DIV>Inside asterisk debug i see what the forwarding of the call working :</DIV>
<DIV><STRONG>log of ASTERISK DEBUG</STRONG></DIV>
<DIV>----------------------------------------------------------------------------------------</DIV>
<DIV>Sip read: <BR>INVITE sip:1phoneiamcalling@xxx.xxx.xxx.xxx:5061 SIP/2.0<BR>Via: SIP/2.0/UDP xxx.xxx.xxx.xxx;branch=z9hG4bKb148.00624e85.0<BR>Via: SIP/2.0/UDP ipoftphone:5060;branch=z9hG4bK06ffef7d<BR>From: "Alexg" <sip:phonesoftphone@xxx.xxx.xxx.xxx>;tag=00036b09607e0047524bda98-4b96b81e<BR>To: <sip:1phoneiamcalling@xxx.xxx.xxx.xxx><BR>Call-ID: <A href="mailto:00036b09-607e0047-3dc1c568-1d31a410@ipoftphone">00036b09-607e0047-3dc1c568-1d31a410@ipoftphone</A><BR>CSeq: 101 INVITE<BR>User-Agent: CSCO/6<BR>Contact: <sip:phonesoftphone@ipoftphone:5060><BR>Expires: 180<BR>Content-Type: application/sdp<BR>Content-Length: 249<BR>Accept: application/sdp</DIV>
<DIV>v=0<BR>o=Cisco-SIPUA 28416 11732 IN IP4 ipoftphone<BR>s=SIP Call<BR>c=IN IP4 ipoftphone<BR>t=0 0<BR>m=audio 26298 RTP/AVP 0 8 18 101<BR>a=rtpmap:0 PCMU/8000<BR>a=rtpmap:8 PCMA/8000<BR>a=rtpmap:18 G729/8000<BR>a=rtpmap:101 telephone-event/8000<BR>a=fmtp:101 0-15</DIV>
<DIV>13 headers, 11 lines<BR>Using latest request as basis request<BR>Sending to xxx.xxx.xxx.xxx : 5060 (non-NAT)<BR>Found peer 'ser'<BR>Found RTP audio format 0<BR>Found RTP audio format 8<BR>Found RTP audio format 18<BR>Found RTP audio format 101<BR>Peer audio RTP is at port ipoftphone:26298<BR>Found description format PCMU<BR>Found description format PCMA<BR>Found description format G729<BR>Found description format telephone-event<BR>Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)<BR>Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723)<BR>Looking for 1phoneiamcalling in ser1<BR>Reliably Transmitting (no NAT):<BR>SIP/2.0 404 Not Found<BR>Via: SIP/2.0/UDP xxx.xxx.xxx.xxx;branch=z9hG4bKb148.00624e85.0<BR>Via: SIP/2.0/UDP ipoftphone:5060;branch=z9hG4bK06ffef7d<BR>From: "Alexg" <sip:phonesoftphone@xxx.xxx.xxx.xxx>;tag=00036b09607e0047524bda98-4b96b81e<BR>To:
<sip:1phoneiamcalling@xxx.xxx.xxx.xxx>;tag=as125ae8d3<BR>Call-ID: <A href="mailto:00036b09-607e0047-3dc1c568-1d31a410@ipoftphone">00036b09-607e0047-3dc1c568-1d31a410@ipoftphone</A><BR>CSeq: 101 INVITE<BR>User-Agent: Asterisk PBX<BR>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER<BR>Contact: <sip:1phoneiamcalling@xxx.xxx.xxx.xxx:5061><BR>Content-Length: 0</DIV>
<DIV><BR> to xxx.xxx.xxx.xxx:5060</DIV>
<DIV><BR>Sip read: <BR>ACK sip:1phoneiamcalling@xxx.xxx.xxx.xxx:5061 SIP/2.0<BR>Via: SIP/2.0/UDP xxx.xxx.xxx.xxx;branch=z9hG4bKb148.00624e85.0<BR>From: "Alexg" <sip:phonesoftphone@xxx.xxx.xxx.xxx>;tag=00036b09607e0047524bda98-4b96b81e<BR>Call-ID: <A href="mailto:00036b09-607e0047-3dc1c568-1d31a410@ipoftphone">00036b09-607e0047-3dc1c568-1d31a410@ipoftphone</A><BR>To: <sip:1phoneiamcalling@xxx.xxx.xxx.xxx>;tag=as125ae8d3<BR>CSeq: 101 ACK<BR>User-Agent: Sip EXpress router(0.8.14 (i386/linux))<BR>Content-Length: 0</DIV>
<DIV><BR>8 headers, 0 lines<BR>Destroying call <A href="mailto:'00036b09-607e0047-3dc1c568-1d31a410@ipoftphone'">'00036b09-607e0047-3dc1c568-1d31a410@ipoftphone'</A></DIV>
<DIV> </DIV>
<DIV> </DIV>
<DIV><STRONG>var/log/asterisk/messages</STRONG></DIV>
<DIV>---------------------------------------------------------------------------------------</DIV>
<DIV>Unable to open /dev/dsp: No such device</DIV>
<DIV> </DIV>
<DIV> </DIV>
<DIV><STRONG>I am calling to number 12222222222 and ser forwarding it to the asterisk (port 5061) (see configuration of sip.conf) to the ser1 context.</STRONG></DIV>
<DIV><STRONG>in extensions.conf i have ser1 context and extensions for ivr under ser1 context.</STRONG></DIV>
<DIV><STRONG>After the call i am hearing the busy line and that's it. i tried to play with extensions.conf with no success.</STRONG></DIV>
<DIV><STRONG>I need a help to setup the IVR system.</STRONG></DIV>
<DIV><STRONG></STRONG> </DIV>
<DIV><STRONG>Thanks.</STRONG></DIV><p>
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