呼叫gxw410x sip中继失败

Question

我有一个问题，使出席传输到fxo网关(grand stream gxw4108)。

我正在使用特征码(*2)提交呼叫参与转移。

先发起呼叫，然后在pstn外振铃时终止呼叫。
盲转工作正常，参与转内部工作正常，但这个问题只在转到gxw4108网关时出现。

here my configuration(sip.conf):

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

我使用的是elastix 2.4版本这是嗅探流量:(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

Answer 1

刚刚找到这个问题的解决方案，分享它可能会帮助别人:
原因:
attend transfer超时，默认值= 15秒，该超时时间不足以建立对gxw4108的呼叫，再由gxw4108建立对PSTN的呼叫。因此，15秒后，星号发送取消请求以终止传输。

解决方案:
通过在/etc/asterisk/features.conf中设置值atxfernoanswertimeout = 60 来增加超时时间