Take a look at the freeswitch debug log, it should tell you exactly why it hung up.
Mike On Nov 12, 2009, at 10:01 AM, Lei Tang wrote: > Hi, I'm running a ivr script on FS, the call is from a softswitch to extenal > sip endpoint of FS. > I added two dialplan in public dialplan xml file. as flow: > <extension name="ivr_demo2"> > <condition field="destination_number" expression="^88888$"> > <action application="lua" data="../ivr/test.lua"/> > </condition> > </extension> > > <extension name="ivr_demo2"> > <condition field="destination_number" expression="^\*114$"> > <action application="lua" data="../ivr/test.lua"/> > </condition> > </extension> > > 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? > > > ============sip packets=================== > ********invite msg from softswitch > INVITE sip:*...@10.37.143.6:5060;user=phone SIP/2.0 > Contact: <sip:xxxxxx...@10.4.35.17:5061> > Content-Type: application/sdp > To: <sip:*...@10.37.143.6:5060;user=phone> > From: > xxxxxxxxx<sip:xxxxxx...@10.4.35.17:5061;user=phone>;tag=949132463135364198E42500 > P-Asserted-Identity: <sip:xxxxxx...@10.4.35.17:5061;user=phone> > Allow: > INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,COMET,UPDATE,PRACK,REFER,SUBSCRIBE,NOTIFY,MESSAGE > Supported: 100rel,timer,replaces,diversion > Expires: 155 > Session-Expires: 1800 > Min-SE: 90 > Call-ID: 01fd10d1bd81400000010...@sip-3 > Max-Forwards: 70 > CSeq: 1 INVITE > Timestamp: 58520 > Via: SIP/2.0/UDP > 10.4.35.17:5061;branch=z9hG4bK5C0F524645A70C943998751419749696 > Content-Length: 150 > > v=0 > o=- 54000602557 1258015146 IN IP4 10.4.35.59 > s=SDP Data > c=IN IP4 10.4.35.59 > t=0 0 > m=audio 30000 RTP/AVP 8 > a=rtpmap:8 PCMA/8000 > a=ptime:20 > > > ******FS ack > SIP/2.0 100 Trying > Via: SIP/2.0/UDP > 10.4.35.17:5061;branch=z9hG4bK5C0F524645A70C943998751419749696 > From: xxxxxxxxx > <sip:xxxxxx...@10.4.35.17:5061;user=phone>;tag=949132463135364198E42500 > To: <sip:*...@10.37.143.6:5060;user=phone> > Call-ID: 01fd10d1bd81400000010...@sip-3 > CSeq: 1 INVITE > Timestamp: 58520 0.000000 > User-Agent: FreeSWITCH-mod_sofia/1.0.4-14460 > Content-Length: 0 > > *****FS answer the call (in lua script, I called session:answer() ) > SIP/2.0 200 OK > Via: SIP/2.0/UDP > 10.4.35.17:5061;branch=z9hG4bK5C0F524645A70C943998751419749696 > From: xxxxxxxxx > <sip:xxxxxx...@10.4.35.17:5061;user=phone>;tag=949132463135364198E42500 > To: <sip:*...@10.37.143.6:5060;user=phone>;tag=UjZcZUKZXjHcQ > Call-ID: 01fd10d1bd81400000010...@sip-3 > CSeq: 1 INVITE > Contact: <sip:*...@10.37.143.6:5060;transport=udp> > User-Agent: FreeSWITCH-mod_sofia/1.0.4-14460 > Accept: application/sdp > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, > REFER, UPDATE, REGISTER, INFO > Require: timer > Supported: timer, precondition, path, replaces > Allow-Events: talk, refer > Session-Expires: 1800;refresher=uac > Min-SE: 120 > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 245 > > v=0 > o=FreeSWITCH 1257988835 1257988836 IN IP4 10.37.143.6 > s=FreeSWITCH > c=IN IP4 10.37.143.6 > t=0 0 > m=audio 24890 RTP/AVP 8 120 > a=rtpmap:8 PCMA/8000 > a=rtpmap:120 telephone-event/8000 > a=fmtp:120 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > ACK sip:*...@10.37.143.6:5060;transport=udp SIP/2.0 > CSeq: 1 ACK > To: <sip:*...@10.37.143.6:5060;user=phone>;tag=UjZcZUKZXjHcQ > From: > xxxxxxxxx<sip:xxxxxx...@10.4.35.17:5061;user=phone>;tag=949132463135364198E42500 > Call-ID: 01fd10d1bd81400000010...@sip-3 > Max-Forwards: 70 > Timestamp: 58520 > Via: SIP/2.0/UDP > 10.4.35.17:5061;branch=z9hG4bK0CC4AE6EE59CA15F69429CDB97848C21 > Content-Length: 0 > > *******FS hangup the call > BYE sip:*...@10.37.143.6:5060;transport=udp SIP/2.0 > Reason: Q.850;cause=1;text="Unallocated (unassigned) number" > To: <sip:*...@10.37.143.6:5060;user=phone>;tag=UjZcZUKZXjHcQ > From: > xxxxxxxxx<sip:xxxxxx...@10.4.35.17:5061;user=phone>;tag=949132463135364198E42500 > Call-ID: 01fd10d1bd81400000010...@sip-3 > Max-Forwards: 70 > CSeq: 2 BYE > Timestamp: 58521 > Via: SIP/2.0/UDP > 10.4.35.17:5061;branch=z9hG4bKBE2D7D86B44CA171A5D374ECAA99A1DB > Content-Length: 0
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