I have an issue making an attended transfer to fxo gateway (grand stream gxw4108).
I am using feature code (*2) to commit in call attended transfer.
Call first is initiated and then transfer terminated just when the external pstn phone ring.
Blind transfer is working fine , attended transfer is working fine internally but this issue appears only when transferring to the gxw4108 gateway.
here my configuration(sip.conf):
[gxw410x]
host= 192.168.10.239
type=peer
insecure=very
i am using elastix version 2.4 and this is sniffing for the 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 the Solution to this issue sharing it may help somebody:
Cause:
Attended transfer time out which is by default = 15 secs and this time is not enough to establish call to gxw4108 and then gxw4108 establish call to PSTN.
So after 15 secs asterisk sends Cancel Request to terminate the transfer.
Solution:
Increase Time out by setting value atxfernoanswertimeout = 60
in /etc/asterisk/features.conf