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我在进行有人值守转移到 fxo 网关(大流 gxw4108)时遇到问题。

我正在使用功能代码 (*2) 提交呼叫参与转移。

当外部 pstn 电话响铃时,首先启动呼叫,然后终止转移。
盲转工作正常,有人参与的转移在内部工作正常,但只有在转移到 gxw4108 网关时才会出现此问题。

这是我的配置(sip.conf):

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

我正在使用 elastix 2.4 版,这是对流量的嗅探:(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
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1 回答 1

0

刚刚找到此问题的解决方案,分享它可能对某人有所帮助:
原因:
参与转接超时,默认情况下 = 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:通过设置值来
增加超时atxfernoanswertimeout = 60 /etc/asterisk/features.conf

于 2014-05-29T22:07:22.907 回答