2016-06-08 68 views
0

我有一個FreeSwitch服務器(Ubuntu上的1.4.26)。 將呼入重定向到外部服務器時,在呼叫連接30分鐘後,我從目標服務器收到RE-INVITE消息。我的FreeSwitch服務器響應「481呼叫不存在」,然後呼叫斷開,雖然它很好。在481通話不存在的通話結果中重新邀請

我承擔再邀請的一半時間後,發送「會話到期:3600;複習= UAC」已經過去。

我試圖告訴FreeSWITCH的忽略重新邀請,使用set sip_ignore_reinvites =前橋真。似乎沒有任何效果。也嘗試在橋的起源字符串。沒有幫助。

我怎樣才能防止這種情況發生?

這裏是SIP日誌(1111調用9999):在

send 1069 bytes to udp/[99.99.99.99]:5060 at 15:02:29.531004: 
    ------------------------------------------------------------------------ 
    INVITE sip:[email protected]:5060 SIP/2.0 
    Via: SIP/2.0/UDP 55.55.55.55;rport;branch=z9hG4bKvjUSX912pUc9Q 
    Max-Forwards: 67 
    From: "12121111111" <sip:[email protected]>;tag=p02B1veKSg0tS 
    To: <sip:[email protected]:5060> 
    Call-ID: [email protected]_01 
    CSeq: 92276610 INVITE 
    Contact: <sip:[email protected]:5060> 
    User-Agent: FreeSWITCH-mod_sofia/1.4.26+git~20151215T154940Z~5579a4ba46~64bit 
    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY 
    Supported: timer, path, replaces 
    Allow-Events: talk, hold, conference, refer 
    Content-Type: application/sdp 
    Content-Disposition: session 
    Content-Length: 223 
    X-FS-Support: update_display,send_info 
    Remote-Party-ID: "12121111111" <sip:[email protected]>;party=calling;screen=yes;privacy=off 

    v=0 
    o=FreeSWITCH 1465208343 1465208344 IN IP4 55.55.55.55 
    s=FreeSWITCH 
    c=IN IP4 55.55.55.55 
    t=0 0 
    m=audio 17006 RTP/AVP 0 101 13 
    a=rtpmap:0 PCMU/8000 
    a=rtpmap:101 telephone-event/8000 
    a=fmtp:101 0-16 
    a=ptime:20 
    ------------------------------------------------------------------------ 
recv 802 bytes from udp/[99.99.99.99]:5060 at 15:02:50.184437: 
    ------------------------------------------------------------------------ 
    SIP/2.0 200 OK 
    Via: SIP/2.0/UDP 55.55.55.55;received=55.55.55.55;branch=z9hG4bKvjUSX912pUc9Q;rport=5060 
    From: "12121111111" <sip:[email protected]>;tag=p02B1veKSg0tS 
    To: <sip:[email protected]:5060>;tag=9307555045152742154 
    Call-ID: [email protected]_01 
    CSeq: 92276610 INVITE 
    Content-Type: application/sdp 
    Session-Expires: 3600;refresher=uas 
    Contact: <sip:[email protected]:5060;user=phone;transport=udp> 
    Supported: timer,100rel 
    Content-Length: 288 

    v=0 
    o=MG4000|2.0 193121 196925 IN IP4 99.99.99.99 
    s=- 
    c=IN IP4 99.99.99.99 
    t=0 0 
    m=audio 61308 RTP/AVP 0 101 
    a=rtpmap:101 telephone-event/8000 
    a=fmtp:101 0-15 
    a=ptime:20 
    a=X-vrzcap:vbd Ver=0 Mode=FaxPr ModemRtpRed=0 
    a=X-vrzcap:identification bin=DSR2883 Prot=mgcp App=MG 
    ------------------------------------------------------------------------ 
recv 934 bytes from udp/[99.99.99.99]:5060 at 15:32:50.190171: 
    ------------------------------------------------------------------------ 
    INVITE sip:[email protected]:5060 SIP/2.0 
    Via: SIP/2.0/UDP 99.99.99.99:5060;branch=z9hG4bKi8jups3010shhe7um2g0sb0000g00.1 
    Call-ID: [email protected]_01 
    From: <sip:[email protected]:5060>;tag=9307555045152742154 
    To: "12121111111" <sip:[email protected]>;tag=p02B1veKSg0tS 
    Content-Type: application/sdp 
    Min-SE: 90 
    Session-Expires: 3600;refresher=uac 
    CSeq: 1 INVITE 
    Contact: <sip:[email protected]:5060;user=phone;transport=udp> 
    Allow: INVITE,CANCEL,BYE,ACK,REFER,UPDATE,INFO,PRACK 
    Supported: timer,100rel 
    Max-Forwards: 69 
    User-Agent: VCS 5.10.2.10-02 
    Content-Length: 288 

    v=0 
    o=MG4000|2.0 193121 196925 IN IP4 99.99.99.99 
    s=- 
    c=IN IP4 99.99.99.99 
    t=0 0 
    m=audio 61308 RTP/AVP 0 101 
    a=rtpmap:101 telephone-event/8000 
    a=fmtp:101 0-15 
    a=ptime:20 
    a=X-vrzcap:vbd Ver=0 Mode=FaxPr ModemRtpRed=0 
    a=X-vrzcap:identification bin=DSR2883 Prot=mgcp App=MG 
    ------------------------------------------------------------------------ 
send 513 bytes to udp/[99.99.99.99]:5060 at 15:32:50.190379: 
    ------------------------------------------------------------------------ 
    SIP/2.0 481 Call Does Not Exist 
    Via: SIP/2.0/UDP 99.99.99.99:5060;branch=z9hG4bKi8jups3010shhe7um2g0sb0000g00.1 
    From: <sip:[email protected]:5060>;tag=9307555045152742154 
    To: "12121111111" <sip:[email protected]>;tag=p02B1veKSg0tS 
    Call-ID: [email protected]_01 
    CSeq: 1 INVITE 
    User-Agent:FreeSWITCH-mod_sofia/1.4.26+git~20151215T154940Z~5579a4ba46~64bit 
    Allow:INVITE,ACK,BYE,CANCEL,OPTIONS,MESSAGE,INFO,UPDATE,REGISTER,REFER,NOTIFY 
    k:timer,path,replaces 
    l:0 

    ------------------------------------------------------------------------ 
recv 374 bytes from udp/[99.99.99.99]:5060 at 15:32:50.276999: 
    ------------------------------------------------------------------------ 
    ACK sip:[email protected]:5060 SIP/2.0 
    Via: SIP/2.0/UDP 99.99.99.99:5060;branch=z9hG4bKi8jups3010shhe7um2g0sb0000g00.1 
    CSeq: 1 ACK 
    Call-ID: [email protected]_01 
    From: <sip:[email protected]:5060>;tag=9307555045152742154 
    To: "12121111111" <sip:[email protected]>;tag=p02B1veKSg0tS 
    Max-Forwards: 69 
    Content-Length: 0 

    ------------------------------------------------------------------------ 
recv 477 bytes from udp/[99.99.99.99]:5060 at 15:32:50.290275: 
    ------------------------------------------------------------------------ 
    BYE sip:[email protected]:5060 SIP/2.0 
    Via: SIP/2.0/UDP 99.99.99.99:5060;branch=z9hG4bKi8jups3010shhe7um2g0sd0000010.1 
    Call-ID: [email protected]_01 
    From: <sip:[email protected]:5060>;tag=9307555045152742154 
    To: "12121111111" <sip:[email protected]>;tag=p02B1veKSg0tS 
    CSeq: 2 BYE 
    Supported: timer,100rel 
    Max-Forwards: 69 
    Reason: SIP;cause=0;iintcode=516;isubsystem=0 
    User-Agent: VCS 5.10.2.10-02 
    Content-Length: 0 

    ------------------------------------------------------------------------ 
send 510 bytes to udp/[99.99.99.99]:5060 at 15:32:50.290421: 
    ------------------------------------------------------------------------ 
    SIP/2.0 481 Call Does Not Exist 
    Via: SIP/2.0/UDP 99.99.99.99:5060;branch=z9hG4bKi8jups3010shhe7um2g0sd0000010.1 
    From: <sip:[email protected]:5060>;tag=9307555045152742154 
    To: "12121111111" <sip:[email protected]>;tag=p02B1veKSg0tS 
    Call-ID: [email protected]_01 
    CSeq: 2 BYE 
    User-Agent:FreeSWITCH-mod_sofia/1.4.26+git~20151215T154940Z~5579a4ba46~64bit 
    Allow:INVITE,ACK,BYE,CANCEL,OPTIONS,MESSAGE,INFO,UPDATE,REGISTER,REFER,NOTIFY 
    k:timer,path,replaces 
    l:0 

回答

0

您應該刪除標記(標記= p02B1veKSg0tS)重新邀請,因爲這是一個新的事務。

+0

重新邀請從遠程服務器發送到我的服務器。如果我不是發送此SIP消息的人,我該如何刪除標籤? 此外,這是相同的交易,具有相同的標籤。爲什麼FS服務器聲稱它不存在? – Eliram

+0

我在想這是你自己的客戶。否則,它不是同一個事務,對於新事務(由reINVITE啓動)不應該設置標記。檢查一下其他softhone,如果可行,那麼你可以確認這是造成這個問題。 – Istvan

1

如果上面是整個跟蹤奇怪的是,現在ACK從FS在200後重新邀請發送是像你說的這恰好相同的對話框上的會話刷新,但它是一個不同的事務。

望着呼叫Id從標籤/到標籤重新邀請看起來是正確的。

做一個tcpdump的/ Wireshark的,並確保重新邀請被髮送到正確的端口,並且有後200ok的初始邀請

1

你試過打開上FreeSWITCH的RFC 4028支持的ACK ?

https://wiki.freeswitch.org/wiki/Sofia.conf.xml#SIP_Related_options

在您的SIP模式:

<param name="enable-timer" value="true"/> 
+0

我的印象是這個參數的默認值是「true」,因爲我確實看到INVITE中支持「定時器」。 我曾嘗試禁用它,這在某些情況下實際上有所幫助。 「定時器」不再存在。 另一個似乎有用的技巧是:我設置了,這比我從遠程站點獲取時短。這導致我的服務器發送一個RE-INVITE,並且我從遠程登陸並且通話繼續。 – Eliram

+0

當您設置時,它可能會將enable-timer從「false」更改爲「true」,因爲如果禁用了功能,則沒有定時器。 – os11k