我在FreePBX / Asterisk中设置了一个SIP TRUNK，可以完美地传入呼叫。 这是相关的configuration：
type=friend host=201.217.31.10 callerid=mynumber [email protected] [email protected] fromuser=595XXYYZZZZZZ fromdomain=prepago.com.py secret=****** dtmfmode=auto trunkname=covoip context=from-trunk hasexten=no hasiax=no hassip=yes registeriax=no registersip=yes trunkstyle=voip nat=force_rport,comedia insecure=port,invite disallow=all allow=alaw,ulaw,gsm qualify=yes 然而，每当我尝试发出呼叫（通过相同的主干），我有一个“所有线路忙”信号从星号。 如果我启用SIP DEBUG，这就是我得到的（显然我的电话被拒绝，因为在另一端的无效别名 ，我无法控制，因为它是我的VOIP提供商）：
<--- SIP read from UDP:201.217.31.10:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.16.50:5061;received=190.128.230.22;branch=z9hG4bK6a440fdb;rport=5061 From: <sip:[email protected]>;tag=as3a625f1c To: <sip:[email protected]> Call-ID: 59fbc0e25c141a603114ce2214c9d208@[::1] CSeq: 180 REGISTER Contact: <sip:[email protected]:5061>;expires=30 Expires: 30 User-Agent: FPBX-AsteriskNOW-12.0.33(13.0.1) Content-Length: 0 <-------------> --- (10 headers 0 lines) --- [2015-02-19 15:48:50] NOTICE[2015]: chan_sip.c:23725 handle_response_register: Outbound Registration: Expiry for 201.217.31.10 is 30 sec (Scheduling reregistration in 24 s) Really destroying SIP dialog '59fbc0e25c141a603114ce2214c9d208@[::1]' Method: REGISTER [2015-02-19 15:48:52] WARNING[1833]: func_cdr.c:349 cdr_write_callback: CDR requires a value (CDR(variable)=value) Audio is at 16688 Adding codec ulaw to SDP Adding codec alaw to SDP Adding codec gsm to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 201.217.31.10:5060: INVITE sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP 192.168.16.50:5061;branch=z9hG4bK61ad8aec;rport Max-Forwards: 70 From: <sip:[email protected]:5061>;tag=as23ae8214 To: <sip:[email protected]> Contact: <sip:[email protected]:5061> Call-ID: [email protected] CSeq: 102 INVITE User-Agent: FPBX-AsteriskNOW-12.0.33(13.0.1) Date: Thu, 19 Feb 2015 18:48:52 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 300 v=0 o=root 1709304421 1709304421 IN IP4 192.168.16.50 s=Asterisk PBX 13.0.1 c=IN IP4 192.168.16.50 t=0 0 m=audio 16688 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=maxptime:150 a=sendrecv --- <--- SIP read from UDP:201.217.31.10:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.16.50:5061;received=190.128.230.22;branch=z9hG4bK61ad8aec;rport=5061 From: <sip:[email protected]:5061>;tag=as23ae8214 To: <sip:[email protected]> Call-ID: [email protected] CSeq: 102 INVITE <-------------> --- (6 headers 0 lines) --- <--- SIP read from UDP:201.217.31.10:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.16.50:5061;received=190.128.230.22;branch=z9hG4bK61ad8aec;rport=5061 From: <sip:[email protected]:5061>;tag=as23ae8214 To: <sip:[email protected]>;tag=b72e12N2654e5f93c-504b Call-ID: [email protected] CSeq: 102 INVITE Reason: Q.850 ;cause=38 ;text="11017 - Invalid alias" Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Transmitting (NAT) to 201.217.31.10:5060: ACK sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP 192.168.16.50:5061;branch=z9hG4bK61ad8aec;rport Max-Forwards: 70 From: <sip:[email protected]:5061>;tag=as23ae8214 To: <sip:[email protected]>;tag=b72e12N2654e5f93c-504b Contact: <sip:[email protected]:5061> Call-ID: [email protected] CSeq: 102 ACK User-Agent: FPBX-AsteriskNOW-12.0.33(13.0.1) Content-Length: 0
任何想法可能是错误的东西我的一面？
如果我连接一个简单的软电话到我的VOIP提供商，它完美地工作（来电和去电）。
38 503 NETWORK_OUT_OF_ORDER network out of order [Q.850] This cause indicates that the network is not functioning correctly and that the condition is likely to last a relatively long period of time eg immediately re-attempting the call is not likely to be successful.
我的猜测是你的来电显示是冒犯他们的。 你是否设置了除实际分配的DID 之外的任何内容？
基于：