参加转移到gxw410x sip trunk失败

Question

我有问题转发到fxo网关（大流gxw4108）。

我正在使用功能代码（* 2）提交呼叫转移。

首先启动呼叫，然后在外部pstn电话响铃时终止传输 盲转移工作正常，有人转移在内部工作正常但这个问题只在转移到gxw4108网关时出现。

这里是我的配置（sip.conf）：

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

我正在使用elastix版本2.4 这是在嗅探流量：（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

刚刚找到解决这个问题的方法，分享它可能有助于某人：
原因：
参加传输超时默认= 15秒，此时间不足以建立对gxw4108的呼叫，然后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atxfernoanswertimeout = 60

中设置值/etc/asterisk/features.conf来增加超时时间