[Freeswitch-users] hangup incoming call by Reason: Q.850; cause=1; text="Unallocated (unassigned) number"

Lei Tang lei.tlfly at gmail.com
Thu Nov 12 07:01:02 PST 2009


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:*114 at 10.37.143.6:5060;user=phone SIP/2.0
Contact: <sip:xxxxxxxxx at 10.4.35.17:5061>
Content-Type: application/sdp
To: <sip:*114 at 10.37.143.6:5060;user=phone>
From: xxxxxxxxx<sip:xxxxxxxxx at 10.4.35.17:5061
;user=phone>;tag=949132463135364198E42500
P-Asserted-Identity: <sip:xxxxxxxxx at 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: 01FD10D1BD81400000010690 at 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:xxxxxxxxx at 10.4.35.17:5061
;user=phone>;tag=949132463135364198E42500
To: <sip:*114 at 10.37.143.6:5060;user=phone>
Call-ID: 01FD10D1BD81400000010690 at 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:xxxxxxxxx at 10.4.35.17:5061
;user=phone>;tag=949132463135364198E42500
To: <sip:*114 at 10.37.143.6:5060;user=phone>;tag=UjZcZUKZXjHcQ
Call-ID: 01FD10D1BD81400000010690 at sip-3
CSeq: 1 INVITE
Contact: <sip:*114 at 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:*114 at 10.37.143.6:5060;transport=udp SIP/2.0
CSeq: 1 ACK
To: <sip:*114 at 10.37.143.6:5060;user=phone>;tag=UjZcZUKZXjHcQ
From: xxxxxxxxx<sip:xxxxxxxxx at 10.4.35.17:5061
;user=phone>;tag=949132463135364198E42500
Call-ID: 01FD10D1BD81400000010690 at 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:*114 at 10.37.143.6:5060;transport=udp SIP/2.0
Reason: Q.850;cause=1;text="Unallocated (unassigned) number"
To: <sip:*114 at 10.37.143.6:5060;user=phone>;tag=UjZcZUKZXjHcQ
From: xxxxxxxxx<sip:xxxxxxxxx at 10.4.35.17:5061
;user=phone>;tag=949132463135364198E42500
Call-ID: 01FD10D1BD81400000010690 at 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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