<html><head></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; ">Take a look at the freeswitch debug log, it should tell you exactly why it hung up.<div><br></div><div>Mike</div><div><br><div><div>On Nov 12, 2009, at 10:01 AM, Lei Tang wrote:</div><br class="Apple-interchange-newline"><blockquote type="cite">Hi, I'm running a ivr script on FS, the call is from a softswitch to extenal sip endpoint of FS.<br>I added two dialplan in public dialplan xml file. as flow:<br><extension name="ivr_demo2"><br>
<condition field="destination_number" expression="^88888$"><br>
<action application="lua" data="../ivr/test.lua"/><br>
</condition><br>
</extension><br>
<br><extension name="ivr_demo2"><br> <condition field="destination_number" expression="^\*114$"><br> <action application="lua" data="../ivr/test.lua"/><br>
</condition><br> </extension><br><br>Every thing is ok when call to number 88888. but when I call the second number "*114", fs hangup after accept and answer the call, I captured the sip packets and found FS sent a bye packet after answer the call. the cause is "Reason: Q.850;cause=1;text="Unallocated (unassigned) number"". But as the fs console log show, the call is answered and the correct ivr script is runned. Why FS hangup the call? Does somebody have any idea about this problem?<br>
<br><br>============sip packets===================<br>********invite msg from softswitch<br>INVITE <a href="sip:*114@10.37.143.6:5060;user=phone">sip:*114@10.37.143.6:5060;user=phone</a> SIP/2.0<br>Contact: <<a href="http://sip:xxxxxxxxx@10.4.35.17:5061/">sip:xxxxxxxxx@10.4.35.17:5061</a>><br>
Content-Type: application/sdp<br>To: <<a href="sip:*114@10.37.143.6:5060;user=phone">sip:*114@10.37.143.6:5060;user=phone</a>><br>From: xxxxxxxxx<<a href="sip:xxxxxxxxx@10.4.35.17:5061;user=phone">sip:xxxxxxxxx@10.4.35.17:5061;user=phone</a>>;tag=949132463135364198E42500<br>P-Asserted-Identity: <<a href="sip:xxxxxxxxx@10.4.35.17:5061;user=phone">sip:xxxxxxxxx@10.4.35.17:5061;user=phone</a>><br>
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,COMET,UPDATE,PRACK,REFER,SUBSCRIBE,NOTIFY,MESSAGE<br>Supported: 100rel,timer,replaces,diversion<br>Expires: 155<br>Session-Expires: 1800<br>Min-SE: 90<br>Call-ID: 01FD10D1BD81400000010690@sip-3<br>
Max-Forwards: 70<br>CSeq: 1 INVITE<br>Timestamp: 58520<br>Via: SIP/2.0/UDP 10.4.35.17:5061;branch=z9hG4bK5C0F524645A70C943998751419749696<br>Content-Length: 150<br><br>v=0<br>o=- 54000602557 1258015146 IN IP4 10.4.35.59<br>
s=SDP Data<br>c=IN IP4 10.4.35.59<br>t=0 0<br>m=audio 30000 RTP/AVP 8<br>a=rtpmap:8 PCMA/8000<br>a=ptime:20<br><br><br>******FS ack<br>SIP/2.0 100 Trying<br>Via: SIP/2.0/UDP 10.4.35.17:5061;branch=z9hG4bK5C0F524645A70C943998751419749696<br>
From: xxxxxxxxx <<a href="sip:xxxxxxxxx@10.4.35.17:5061;user=phone">sip:xxxxxxxxx@10.4.35.17:5061;user=phone</a>>;tag=949132463135364198E42500<br>To: <<a href="sip:*114@10.37.143.6:5060;user=phone">sip:*114@10.37.143.6:5060;user=phone</a>><br>Call-ID: 01FD10D1BD81400000010690@sip-3<br>CSeq: 1 INVITE<br>Timestamp: 58520 0.000000<br>
User-Agent: FreeSWITCH-mod_sofia/1.0.4-14460<br>Content-Length: 0<br><br>*****FS answer the call (in lua script, I called session:answer() )<br>SIP/2.0 200 OK<br>Via: SIP/2.0/UDP 10.4.35.17:5061;branch=z9hG4bK5C0F524645A70C943998751419749696<br>
From: xxxxxxxxx <<a href="sip:xxxxxxxxx@10.4.35.17:5061;user=phone">sip:xxxxxxxxx@10.4.35.17:5061;user=phone</a>>;tag=949132463135364198E42500<br>To: <<a href="sip:*114@10.37.143.6:5060;user=phone">sip:*114@10.37.143.6:5060;user=phone</a>>;tag=UjZcZUKZXjHcQ<br>Call-ID: 01FD10D1BD81400000010690@sip-3<br>CSeq: 1 INVITE<br>
Contact: <<a href="sip:*114@10.37.143.6:5060;transport=udp">sip:*114@10.37.143.6:5060;transport=udp</a>><br>User-Agent: FreeSWITCH-mod_sofia/1.0.4-14460<br>Accept: application/sdp<br>Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO<br>
Require: timer<br>Supported: timer, precondition, path, replaces<br>Allow-Events: talk, refer<br>Session-Expires: 1800;refresher=uac<br>Min-SE: 120<br>Content-Type: application/sdp<br>Content-Disposition: session<br>Content-Length: 245<br>
<br>v=0<br>o=FreeSWITCH 1257988835 1257988836 IN IP4 10.37.143.6<br>s=FreeSWITCH<br>c=IN IP4 10.37.143.6<br>t=0 0<br>m=audio 24890 RTP/AVP 8 120<br>a=rtpmap:8 PCMA/8000<br>a=rtpmap:120 telephone-event/8000<br>a=fmtp:120 0-16<br>
a=silenceSupp:off - - - -<br>a=ptime:20<br>ACK <a href="sip:*114@10.37.143.6:5060;transport=udp">sip:*114@10.37.143.6:5060;transport=udp</a> SIP/2.0<br>CSeq: 1 ACK<br>To: <<a href="sip:*114@10.37.143.6:5060;user=phone">sip:*114@10.37.143.6:5060;user=phone</a>>;tag=UjZcZUKZXjHcQ<br>From: xxxxxxxxx<<a href="sip:xxxxxxxxx@10.4.35.17:5061;user=phone">sip:xxxxxxxxx@10.4.35.17:5061;user=phone</a>>;tag=949132463135364198E42500<br>
Call-ID: 01FD10D1BD81400000010690@sip-3<br>Max-Forwards: 70<br>Timestamp: 58520<br>Via: SIP/2.0/UDP 10.4.35.17:5061;branch=z9hG4bK0CC4AE6EE59CA15F69429CDB97848C21<br>Content-Length: 0<br><br>*******FS hangup the call<br>BYE <a href="sip:*114@10.37.143.6:5060;transport=udp">sip:*114@10.37.143.6:5060;transport=udp</a> SIP/2.0<br>
Reason: Q.850;cause=1;text="Unallocated (unassigned) number"<br>To: <<a href="sip:*114@10.37.143.6:5060;user=phone">sip:*114@10.37.143.6:5060;user=phone</a>>;tag=UjZcZUKZXjHcQ<br>From: xxxxxxxxx<<a href="sip:xxxxxxxxx@10.4.35.17:5061;user=phone">sip:xxxxxxxxx@10.4.35.17:5061;user=phone</a>>;tag=949132463135364198E42500<br>
Call-ID: 01FD10D1BD81400000010690@sip-3<br>Max-Forwards: 70<br>CSeq: 2 BYE<br>Timestamp: 58521<br>Via: SIP/2.0/UDP 10.4.35.17:5061;branch=z9hG4bKBE2D7D86B44CA171A5D374ECAA99A1DB<br>Content-Length: 0<br></blockquote></div><br></div></body></html>