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如何将FreeSwitch连接到FreeSwitch？

[英]How to connect FreeSwitch to FreeSwitch?

 原文  2014-10-06 08:55:09  8   2    voip/ freeswitch

问题描述

i have a problem with FreeSwitch. 我有FreeSwitch的问题。 I tried for hours to connect a FreeSwitch server on my system with a FreeSwitch server on another system. 我试了几个小时，将我系统上的FreeSwitch服务器与另一个系统上的FreeSwitch服务器连接起来。 However, what i want is to call with user "abc@myip" the user "123@buddysIp". 但是，我想要的是用户“abc @ myip”用户“123 @ buddysIp”。 What i tried is to add a new "list" item to the acl.conf.xml 我尝试的是在acl.conf.xml中添加一个新的“list”项 

<list name="buddy" default="deny">
      <node type="allow" cidr="hisip/32"/>
  </list>

also i tried to add an extension in the conf/dialplan/default directory 我也尝试在conf / dialplan / default目录中添加扩展名 

<include>
<extension name="outbound_calls">
    <condition field="destination_number" expression="^(.*)$">
        <action application="bridge" data="sofia/gateway/buddy/$1"/>
    </condition>
</extension>

and his gateway which is stored in conf/sip_profiles/buddy.xml and looks like this 和他的网关存储在conf / sip_profiles / buddy.xml中，看起来像这样 

<include>
<gateway name="buddy">
    <param name="realm" value="hisip"/>
    <param name="username" value="myuser"/>
    <param name="password" value="mypw"/>
</gateway>

I hope somebody can help me. 我希望有人可以帮助我。 Maybe i forgot something. 也许我忘记了什么。 We are in the same network. 我们在同一个网络中。 Please tell me if you need more informations, thanks. 如果您需要更多信息，请告诉我，谢谢。

Here are my call log: 这是我的通话记录： 

 INVITE sip:666@myip SIP/2.0
   Via: SIP/2.0/UDP myip:51155;rport;branch=z9hG4bKPjyv0EW.k04poUhm7kdxHae5kheAypVEBc
   Max-Forwards: 70
   From: "me" <sip:1001@myip>;tag=Hd47q6ldQvMGLJO5RcMsO4AJQwWmRU28
   To: <sip:666@myip>
   Contact: "me" <sip:1001@myip:51155;ob>
   Call-ID: WW8ZDSQvHKuKTSjTJTX.hiefO6liAh7W
   CSeq: 24074 INVITE
   Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
   Supported: replaces, 100rel, timer, norefersub
   Session-Expires: 1800
   Min-SE: 90
   User-Agent: Telephone 1.1.4
   Content-Type: application/sdp
   Content-Length:   479

   v=0
   o=- 3621578544 3621578544 IN IP4 myip
   s=pjmedia
   b=AS:84
   t=0 0
   a=X-nat:0
   m=audio 4032 RTP/AVP 103 102 104 109 3 0 8 9 101
   c=IN IP4 myip
   b=TIAS:64000
   a=rtcp:4033 IN IP4 myip
   a=sendrecv
   a=rtpmap:103 speex/16000
   a=rtpmap:102 speex/8000
   a=rtpmap:104 speex/32000
   a=rtpmap:109 iLBC/8000
   a=fmtp:109 mode=30
   a=rtpmap:3 GSM/8000
   a=rtpmap:0 PCMU/8000
   a=rtpmap:8 PCMA/8000
   a=rtpmap:9 G722/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-15
   ------------------------------------------------------------------------
send 382 bytes to udp/[myip]:51155 at 12:02:24.444389:
   ------------------------------------------------------------------------
   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP myip:51155;rport=51155;branch=z9hG4bKPjyv0EW.k04poUhm7kdxHae5kheAypVEBc
   From: "me" <sip:1001@myip>;tag=Hd47q6ldQvMGLJO5RcMsO4AJQwWmRU28
   To: <sip:666@myip>
   Call-ID: WW8ZDSQvHKuKTSjTJTX.hiefO6liAh7W
   CSeq: 24074 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.4.9+git~20140929T194948Z~ae069dcca7~64bit
   Content-Length: 0

   ------------------------------------------------------------------------
2014-10-06 12:02:24.435021 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/1001@myip [2b0eddac-0ad5-41b3-a9f4-eebf1a81565e]
send 884 bytes to udp/[myip]:51155 at 12:02:24.450259:
   ------------------------------------------------------------------------
   SIP/2.0 407 Proxy Authentication Required
   Via: SIP/2.0/UDP myip:51155;rport=51155;branch=z9hG4bKPjyv0EW.k04poUhm7kdxHae5kheAypVEBc
   From: "me" <sip:1001@myip>;tag=Hd47q6ldQvMGLJO5RcMsO4AJQwWmRU28
   To: <sip:666@myip>;tag=9Q0rj5B37FS7H
   Call-ID: WW8ZDSQvHKuKTSjTJTX.hiefO6liAh7W
   CSeq: 24074 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.4.9+git~20140929T194948Z~ae069dcca7~64bit
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: timer, path, replaces
   Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
   Proxy-Authenticate: Digest realm="myip", nonce="72eb4930-85ce-4590-b2a2-b3420109fb4e", algorithm=MD5, qop="auth"
   Content-Length: 0

   ------------------------------------------------------------------------
recv 345 bytes from udp/[myip]:51155 at 12:02:24.450538:
   ------------------------------------------------------------------------
   ACK sip:666@myip SIP/2.0
   Via: SIP/2.0/UDP myip:51155;rport;branch=z9hG4bKPjyv0EW.k04poUhm7kdxHae5kheAypVEBc
   Max-Forwards: 70
   From: "me" <sip:1001@myip>;tag=Hd47q6ldQvMGLJO5RcMsO4AJQwWmRU28
   To: <sip:666@myip>;tag=9Q0rj5B37FS7H
   Call-ID: WW8ZDSQvHKuKTSjTJTX.hiefO6liAh7W
   CSeq: 24074 ACK
   Content-Length:  0

   ------------------------------------------------------------------------
recv 1372 bytes from udp/[myip]:51155 at 12:02:24.450608:
   ------------------------------------------------------------------------
   INVITE sip:666@myip SIP/2.0
   Via: SIP/2.0/UDP myip:51155;rport;branch=z9hG4bKPjdQcDT3A-9MHF6CkHL-i9ocmvZjxVxGnD
   Max-Forwards: 70
   From: "me" <sip:1001@myip>;tag=Hd47q6ldQvMGLJO5RcMsO4AJQwWmRU28
   To: <sip:666@myip>
   Contact: "me" <sip:1001@myip:51155;ob>
   Call-ID: WW8ZDSQvHKuKTSjTJTX.hiefO6liAh7W
   CSeq: 24075 INVITE
   Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
   Supported: replaces, 100rel, timer, norefersub
   Session-Expires: 1800
   Min-SE: 90
   User-Agent: Telephone 1.1.4
   Proxy-Authorization: Digest username="1001", realm="myip", nonce="72eb4930-85ce-4590-b2a2-b3420109fb4e", uri="sip:666@myip", response="1220921ada500668b903722228634e1a", algorithm=MD5, cnonce="P1gGhwXHyrEHo1zMDrBEk3ryp1uKHbaK", qop=auth, nc=00000001
   Content-Type: application/sdp
   Content-Length:   479

   v=0
   o=- 3621578544 3621578544 IN IP4 myip
   s=pjmedia
   b=AS:84
   t=0 0
   a=X-nat:0
   m=audio 4032 RTP/AVP 103 102 104 109 3 0 8 9 101
   c=IN IP4 myip
   b=TIAS:64000
   a=rtcp:4033 IN IP4 myip
   a=sendrecv
   a=rtpmap:103 speex/16000
   a=rtpmap:102 speex/8000
   a=rtpmap:104 speex/32000
   a=rtpmap:109 iLBC/8000
   a=fmtp:109 mode=30
   a=rtpmap:3 GSM/8000
   a=rtpmap:0 PCMU/8000
   a=rtpmap:8 PCMA/8000
   a=rtpmap:9 G722/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-15
   ------------------------------------------------------------------------
send 382 bytes to udp/[myip]:51155 at 12:02:24.450830:
   ------------------------------------------------------------------------
   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP myip:51155;rport=51155;branch=z9hG4bKPjdQcDT3A-9MHF6CkHL-i9ocmvZjxVxGnD
   From: "me" <sip:1001@myip>;tag=Hd47q6ldQvMGLJO5RcMsO4AJQwWmRU28
   To: <sip:666@myip>
   Call-ID: WW8ZDSQvHKuKTSjTJTX.hiefO6liAh7W
   CSeq: 24075 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.4.9+git~20140929T194948Z~ae069dcca7~64bit
   Content-Length: 0

   ------------------------------------------------------------------------
2014-10-06 12:02:24.484827 [INFO] mod_dialplan_xml.c:558 Processing me <1001>->666 in context default
2014-10-06 12:02:24.484827 [CRIT] mod_dptools.c:1628 WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING
2014-10-06 12:02:24.484827 [CRIT] mod_dptools.c:1628 Open /usr/local/freeswitch/conf/vars.xml and change the default_password.
2014-10-06 12:02:24.484827 [CRIT] mod_dptools.c:1628 Once changed type 'reloadxml' at the console.
2014-10-06 12:02:24.495177 [CRIT] mod_dptools.c:1628 WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING
2014-10-06 12:02:34.854471 [NOTICE] switch_channel.c:1055 New Channel sofia/external/666 [3bc1a418-d55d-4dab-863b-9742ec0ae187]
send 1113 bytes to udp/[buddyip]:5060 at 12:02:34.865554:
   ------------------------------------------------------------------------
   INVITE sip:666@buddyip SIP/2.0
   Via: SIP/2.0/UDP myip:5080;rport;branch=z9hG4bK94rXNDv8U6KDg
   Max-Forwards: 69
   From: "Extension 1001" <sip:FreeSWITCH@buddyip>;tag=2N77KS1Ke6gpN
   To: <sip:666@buddyip>
   Call-ID: bc64e628-c7e2-1232-0e89-4d7ce0605bb2
   CSeq: 65958813 INVITE
   Contact: <sip:gw+amr@myip:5080;transport=udp;gw=amr>
   User-Agent: FreeSWITCH-mod_sofia/1.4.9+git~20140929T194948Z~ae069dcca7~64bit
   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: 292
   X-FS-Support: update_display,send_info
   Remote-Party-ID: "Extension 1001" <sip:2706446@buddyip>;party=calling;screen=yes;privacy=off

   v=0
   o=FreeSWITCH 1412565842 1412565843 IN IP4 myip
   s=FreeSWITCH
   c=IN IP4 myip
   t=0 0
   m=audio 23912 RTP/AVP 3 0 8 9 101 13
   a=rtpmap:3 GSM/8000
   a=rtpmap:0 PCMU/8000
   a=rtpmap:8 PCMA/8000
   a=rtpmap:9 G722/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=ptime:20
   ------------------------------------------------------------------------
recv 363 bytes from udp/[buddyip]:5060 at 12:02:34.867644:
   ------------------------------------------------------------------------
   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP myip:5080;rport=5080;branch=z9hG4bK94rXNDv8U6KDg
   From: "Extension 1001" <sip:FreeSWITCH@buddyip>;tag=2N77KS1Ke6gpN
   To: <sip:666@buddyip>
   Call-ID: bc64e628-c7e2-1232-0e89-4d7ce0605bb2
   CSeq: 65958813 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.5.14b+git~20141001T023048Z~a39db86863~64bit
   Content-Length: 0

   ------------------------------------------------------------------------
recv 865 bytes from udp/[buddyip]:5060 at 12:02:34.869949:
   ------------------------------------------------------------------------
   SIP/2.0 407 Proxy Authentication Required
   Via: SIP/2.0/UDP myip:5080;rport=5080;branch=z9hG4bK94rXNDv8U6KDg
   From: "Extension 1001" <sip:FreeSWITCH@buddyip>;tag=2N77KS1Ke6gpN
   To: <sip:666@buddyip>;tag=ytX68BX46p4Kp
   Call-ID: bc64e628-c7e2-1232-0e89-4d7ce0605bb2
   CSeq: 65958813 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.5.14b+git~20141001T023048Z~a39db86863~64bit
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: timer, path, replaces
   Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
   Proxy-Authenticate: Digest realm="buddyip", nonce="56db4231-66c0-4603-bec6-7d6fb163a011", algorithm=MD5, qop="auth"
   Content-Length: 0

   ------------------------------------------------------------------------
send 324 bytes to udp/[buddyip]:5060 at 12:02:34.870119:
   ------------------------------------------------------------------------
   ACK sip:666@buddyip SIP/2.0
   Via: SIP/2.0/UDP myip:5080;rport;branch=z9hG4bK94rXNDv8U6KDg
   Max-Forwards: 69
   From: "Extension 1001" <sip:FreeSWITCH@buddyip>;tag=2N77KS1Ke6gpN
   To: <sip:666@buddyip>;tag=ytX68BX46p4Kp
   Call-ID: bc64e628-c7e2-1232-0e89-4d7ce0605bb2
   CSeq: 65958813 ACK
   Content-Length: 0

   ------------------------------------------------------------------------
send 1389 bytes to udp/[buddyip]:5060 at 12:02:34.882192:
   ------------------------------------------------------------------------
   INVITE sip:666@buddyip SIP/2.0
   Via: SIP/2.0/UDP myip:5080;rport;branch=z9hG4bKaejpQ8ccSFa0B
   Max-Forwards: 69
   From: "Extension 1001" <sip:FreeSWITCH@buddyip>;tag=2N77KS1Ke6gpN
   To: <sip:666@buddyip>
   Call-ID: bc64e628-c7e2-1232-0e89-4d7ce0605bb2
   CSeq: 65958814 INVITE
   Contact: <sip:gw+amr@myip:5080;transport=udp;gw=amr>
   Expires: 3600
   User-Agent: FreeSWITCH-mod_sofia/1.4.9+git~20140929T194948Z~ae069dcca7~64bit
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
   Supported: timer, path, replaces
   Allow-Events: talk, hold, conference, refer
   Proxy-Authorization: Digest username="FreeSWITCH", realm="buddyip", nonce="56db4231-66c0-4603-bec6-7d6fb163a011", cnonce="vGWirMfiEjKJDk184GBbsg", algorithm=MD5, uri="sip:666@buddyip", response="9455fd51fcf3ce264eb85a35fd311f23", qop=auth, nc=00000001
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 292
   X-FS-Support: update_display,send_info
   Remote-Party-ID: "Extension 1001" <sip:2706446@buddyip>;party=calling;screen=yes;privacy=off

   v=0
   o=FreeSWITCH 1412565842 1412565843 IN IP4 myip
   s=FreeSWITCH
   c=IN IP4 myip
   t=0 0
   m=audio 23912 RTP/AVP 3 0 8 9 101 13
   a=rtpmap:3 GSM/8000
   a=rtpmap:0 PCMU/8000
   a=rtpmap:8 PCMA/8000
   a=rtpmap:9 G722/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=ptime:20
   ------------------------------------------------------------------------
recv 363 bytes from udp/[buddyip]:5060 at 12:02:34.883681:
   ------------------------------------------------------------------------
   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP myip:5080;rport=5080;branch=z9hG4bKaejpQ8ccSFa0B
   From: "Extension 1001" <sip:FreeSWITCH@buddyip>;tag=2N77KS1Ke6gpN
   To: <sip:666@buddyip>
   Call-ID: bc64e628-c7e2-1232-0e89-4d7ce0605bb2
   CSeq: 65958814 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.5.14b+git~20141001T023048Z~a39db86863~64bit
   Content-Length: 0

   ------------------------------------------------------------------------
recv 724 bytes from udp/[buddyip]:5060 at 12:02:34.891400:
   ------------------------------------------------------------------------
   SIP/2.0 403 Forbidden
   Via: SIP/2.0/UDP myip:5080;rport=5080;branch=z9hG4bKaejpQ8ccSFa0B
   From: "Extension 1001" <sip:FreeSWITCH@buddyip>;tag=2N77KS1Ke6gpN
   To: <sip:666@buddyip>;tag=Z3pZa7D83Zt6H
   Call-ID: bc64e628-c7e2-1232-0e89-4d7ce0605bb2
   CSeq: 65958814 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.5.14b+git~20141001T023048Z~a39db86863~64bit
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: timer, path, replaces
   Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
   Content-Length: 0

   ------------------------------------------------------------------------
send 324 bytes to udp/[buddyip]:5060 at 12:02:34.891596:
   ------------------------------------------------------------------------
   ACK sip:666@buddyip SIP/2.0
   Via: SIP/2.0/UDP myip:5080;rport;branch=z9hG4bKaejpQ8ccSFa0B
   Max-Forwards: 69
   From: "Extension 1001" <sip:FreeSWITCH@buddyip>;tag=2N77KS1Ke6gpN
   To: <sip:666@buddyip>;tag=Z3pZa7D83Zt6H
   Call-ID: bc64e628-c7e2-1232-0e89-4d7ce0605bb2
   CSeq: 65958814 ACK
   Content-Length: 0

   ------------------------------------------------------------------------
2014-10-06 12:02:34.884694 [NOTICE] sofia.c:7306 Hangup sofia/external/666 [CS_CONSUME_MEDIA] [CALL_REJECTED]
2014-10-06 12:02:34.899162 [INFO] mod_dptools.c:3234 Originate Failed.  Cause: CALL_REJECTED
2014-10-06 12:02:34.899162 [NOTICE] switch_channel.c:4685 Hangup sofia/internal/1001@myip [CS_EXECUTE] [CALL_REJECTED]
send 888 bytes to udp/[myip]:51155 at 12:02:34.910248:
   ------------------------------------------------------------------------
   SIP/2.0 403 Forbidden
   Via: SIP/2.0/UDP myip:51155;rport=51155;branch=z9hG4bKPjdQcDT3A-9MHF6CkHL-i9ocmvZjxVxGnD
   Max-Forwards: 70
   From: "me" <sip:1001@myip>;tag=Hd47q6ldQvMGLJO5RcMsO4AJQwWmRU28
   To: <sip:666@myip>;tag=a1SHm0v64rFtD
   Call-ID: WW8ZDSQvHKuKTSjTJTX.hiefO6liAh7W
   CSeq: 24075 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.4.9+git~20140929T194948Z~ae069dcca7~64bit
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: timer, path, replaces
   Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
   Reason: Q.850;cause=21;text="CALL_REJECTED"
   Content-Length: 0
   Remote-Party-ID: "666" <sip:666@myip>;party=calling;privacy=off;screen=no

   ------------------------------------------------------------------------
recv 345 bytes from udp/[myip]:51155 at 12:02:34.910602:
   ------------------------------------------------------------------------
   ACK sip:666@myip SIP/2.0
   Via: SIP/2.0/UDP myip:51155;rport;branch=z9hG4bKPjdQcDT3A-9MHF6CkHL-i9ocmvZjxVxGnD
   Max-Forwards: 70
   From: "me" <sip:1001@myip>;tag=Hd47q6ldQvMGLJO5RcMsO4AJQwWmRU28
   To: <sip:666@myip>;tag=a1SHm0v64rFtD
   Call-ID: WW8ZDSQvHKuKTSjTJTX.hiefO6liAh7W
   CSeq: 24075 ACK
   Content-Length:  0

   ------------------------------------------------------------------------
2014-10-06 12:02:34.914727 [NOTICE] switch_core_session.c:1633 Session 28 (sofia/external/666) Ended
2014-10-06 12:02:34.914727 [NOTICE] switch_core_session.c:1637 Close Channel sofia/external/666 [CS_DESTROY]
2014-10-06 12:02:34.945164 [NOTICE] switch_core_session.c:1633 Session 27 (sofia/internal/1001@myip) Ended
2014-10-06 12:02:34.945164 [NOTICE] switch_core_session.c:1637 Close Channel sofia/internal/1001@myip [CS_DESTROY]

2 个解决方案

解决方案1
 4 已采纳  2014-10-06 14:39:58

It works know. 它的作品知道。 I created a gateway in sip_profiles/external/mygateway.xml looks like 我在sip_profiles / external / mygateway.xml中创建了一个网关 

<include>
    <gateway name="buddygateway">
        <param name="proxy" value="buddyip"/>
        <param name="register" value="false"/>
        <param name="caller-id-in-from" value="true"/> <!--Most gateways seem to want this-->
    </gateway>
</include>

and created an outbound extension in dialplan/default/outbound_calls.xml looks like this: 并在dialplan / default / outbound_calls.xml中创建了一个出站扩展，如下所示： 

<include>
    <extension name="outbound_calls">
        <condition field="destination_number" expression="^BUDDYPREFIX(\d*)$">
            <action application="bridge" data="sofia/gateway/buddygateway/$1"/>
        </condition>
    </extension>
</include>

so every number i call now starts with BUDDYPREFIX calls the number on the remote server. 所以我现在调用的每个号码都以BUDDYPREFIX调用远程服务器上的号码开头。

Thanks for the answers :) Hope this will help somebody. 谢谢你的答案:)希望这会有所帮助。

解决方案2
 0  2014-10-06 10:46:26

you can send the call to "external" SIP profile (port 5080), then it won't be authenticated. 您可以将呼叫发送到“外部”SIP配置文件（端口5080），然后它将不会被验证。 Then you need the dialplan rules in the public context to process such calls. 然后，您需要在公共上下文中使用dialplan规则来处理此类调用。 Alternatively, you can create a new profile and attach it to a different UDP port, or even to a specific Ethernet port by explicitly specifying the bind IP address. 或者，您可以通过显式指定绑定IP地址来创建新的配置文件并将其附加到不同的UDP端口，甚至连接到特定的以太网端口。

In the piece of log that you provided, it looks like the call is landed on port 51155. Is there a special SIP profile attached to that port? 在您提供的日志中，看起来呼叫已落在端口51155上。是否有特殊的SIP配置文件连接到该端口？ If so, probably you need to disable authentication on that profile. 如果是这样，可能需要在该配置文件上禁​​用身份验证。

But in general, you need to order a couple of hours of consultancy, so that a specialist explains you how things work. 但总的来说，您需要订购几个小时的咨询，以便专家解释您的工作方式。 I can do that if needed too. 如果需要，我也可以这样做。

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