asterisk - Attended Transfer to gxw410x sip trunk Failed -

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i have issue making attended transfer fxo gateway (grand stream gxw4108).

i using feature code (*2) commit in call attended transfer.

call first initiated , transfer terminated when external pstn phone ring.
blind transfer working fine , attended transfer working fine internally issue appears when transferring gxw4108 gateway.

here configuration(sip.conf):

[gxw410x] host= 192.168.10.239 type=peer insecure=very

i using elastix version 2.4 , sniffing traffic: (192.168.10.231: asterisk , 192.168.10.239: gxw4108)

invite sip:991xxxxxxxxxxx@192.168.10.239 sip/2.0  via: sip/2.0/udp 192.168.10.231:5060;branch=z9hg4bk5c0ae243;rport  max-forwards: 70  from: "100" <sip:100@192.168.10.231>;tag=as1973acc2  to: <sip:991xxxxxxxxxxx@192.168.10.239>  contact: <sip:100@192.168.10.231:5060>  call-id: 21f5e75c5c575af45b939d0f349a40fc@192.168.10.231:5060  cseq: 102 invite  user-agent: fpbx-2.8.1(1.8.20.0)  date: sat, 10 may 2014 20:52:01 gmt  allow: invite, ack, cancel, options, bye, refer, subscribe, notify, info, publish  supported: replaces, timer  content-type: application/sdp  content-length: 288    v=0  o=root 2108910474 2108910474 in ip4 192.168.10.231  s=asterisk pbx 1.8.20.0  c=in ip4 192.168.10.231  t=0 0  m=audio 15580 rtp/avp 0 8 3 101  a=rtpmap:0 pcmu/8000  a=rtpmap:8 pcma/8000  a=rtpmap:3 gsm/8000  a=rtpmap:101 telephone-event/8000  a=fmtp:101 0-16  a=ptime:20  a=sendrecv  sip/2.0 100 trying  via: sip/2.0/udp 192.168.10.231:5060;branch=z9hg4bk5c0ae243;rport  from: "100" <sip:100@192.168.10.231>;tag=as1973acc2  to: <sip:991xxxxxxxxxxx@192.168.10.239>  call-id: 21f5e75c5c575af45b939d0f349a40fc@192.168.10.231:5060  cseq: 102 invite  user-agent: grandstream gxw4108 (hw 2.0, ch:7) 1.3.4.13  content-length: 0    sip/2.0 180 ringing  via: sip/2.0/udp 192.168.10.231:5060;branch=z9hg4bk5c0ae243;rport  from: "100" <sip:100@192.168.10.231>;tag=as1973acc2  to: <sip:991xxxxxxxxxxx@192.168.10.239>;tag=27454245bd077ea3  call-id: 21f5e75c5c575af45b939d0f349a40fc@192.168.10.231:5060  cseq: 102 invite  user-agent: grandstream gxw4108 (hw 2.0, ch:7) 1.3.4.13  contact: <sip:gxw410x@192.168.10.239:5074;transport=udp>  allow: invite,ack,cancel,bye,notify,refer,options,info,subscribe,update,prack  content-length: 0    cancel sip:991xxxxxxxxxxx@192.168.10.239 sip/2.0  via: sip/2.0/udp 192.168.10.231:5060;branch=z9hg4bk5c0ae243;rport  max-forwards: 70  from: "100" <sip:100@192.168.10.231>;tag=as1973acc2  to: <sip:991xxxxxxxxxxx@192.168.10.239>  call-id: 21f5e75c5c575af45b939d0f349a40fc@192.168.10.231:5060  cseq: 102 cancel  user-agent: fpbx-2.8.1(1.8.20.0)  content-length: 0    sip/2.0 200 ok  via: sip/2.0/udp 192.168.10.231:5060;branch=z9hg4bk5c0ae243;rport  from: "100" <sip:100@192.168.10.231>;tag=as1973acc2  to: <sip:991xxxxxxxxxxx@192.168.10.239>;tag=27454245bd077ea3  call-id: 21f5e75c5c575af45b939d0f349a40fc@192.168.10.231:5060  cseq: 102 cancel  user-agent: grandstream gxw4108 (hw 2.0, ch:7) 1.3.4.13  supported: replaces, timer, 100rel, path  content-length: 0    sip/2.0 487 request cancelled  via: sip/2.0/udp 192.168.10.231:5060;branch=z9hg4bk5c0ae243;rport  from: "100" <sip:100@192.168.10.231>;tag=as1973acc2  to: <sip:991xxxxxxxxxxx@192.168.10.239>;tag=27454245bd077ea3  call-id: 21f5e75c5c575af45b939d0f349a40fc@192.168.10.231:5060  cseq: 102 invite  user-agent: grandstream gxw4108 (hw 2.0, ch:7) 1.3.4.13  content-length: 0    ack sip:gxw410x@192.168.10.239:5074;transport=udp sip/2.0  via: sip/2.0/udp 192.168.10.231:5060;branch=z9hg4bk5c0ae243;rport  max-forwards: 70  from: "100" <sip:100@192.168.10.231>;tag=as1973acc2  to: <sip:991xxxxxxxxxxx@192.168.10.239>;tag=27454245bd077ea3  contact: <sip:100@192.168.10.231:5060>  call-id: 21f5e75c5c575af45b939d0f349a40fc@192.168.10.231:5060  cseq: 102 ack  user-agent: fpbx-2.8.1(1.8.20.0)  content-length: 0    options sip:gxw410x@192.168.10.239:5074;transport=udp sip/2.0  via: sip/2.0/udp 192.168.10.231:5060;branch=z9hg4bk4b3e2af1;rport  max-forwards: 70  from: "unknown" <sip:unknown@192.168.10.231>;tag=as7aaf1080  to: <sip:gxw410x@192.168.10.239:5074;transport=udp>  contact: <sip:unknown@192.168.10.231:5060>  call-id: 12a9092b47984994709a95bd75d8c60b@192.168.10.231:5060  cseq: 102 options  user-agent: fpbx-2.8.1(1.8.20.0)  date: sat, 10 may 2014 20:52:18 gmt  allow: invite, ack, cancel, options, bye, refer, subscribe, notify, info, publish  supported: replaces, timer  content-length: 0    sip/2.0 200 ok  via: sip/2.0/udp 192.168.10.231:5060;branch=z9hg4bk4b3e2af1;rport  from: "unknown" <sip:unknown@192.168.10.231>;tag=as7aaf1080  to: <sip:gxw410x@192.168.10.239:5074;transport=udp>;tag=as2cee3cf7  call-id: 12a9092b47984994709a95bd75d8c60b@192.168.10.231:5060  cseq: 102 options  user-agent: grandstream gxw4108 (hw 2.0, ch:15) 1.3.4.13  contact: <sip:gxw410x@192.168.10.239:5074;transport=udp>  allow: invite,ack,cancel,bye,notify,refer,options,info,subscribe,update,prack  supported: replaces, timer, 100rel, path  content-length: 0

just found solution issue sharing may somebody:
cause:
attended transfer time out default = 15 secs , time not enough establish call gxw4108 , gxw4108 establish call pstn. after 15 secs asterisk sends cancel request terminate transfer.

solution:
increase time out setting value atxfernoanswertimeout = 60 in /etc/asterisk/features.conf


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