Voip
SIP:沒有匹配的編解碼器 / 480 暫時不可用
我正在 VoIP 環境中安裝 SIP 電話。有 2 部系統電話可以正常工作(與 PBX 相同的製造商),第三部電話可以呼叫,但不能呼叫其他 2 部電話。
PBX 顯示錯誤:“沒有匹配的編解碼器!呼叫被拒絕” 這是從第三部電話的角度來看的對話:
INVITE sip:20@192.168.0.250 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.14:5060;branch=z9hG4bK1040318360;rport From: <sip:192.168.0.250>;tag=1540961770 To: <sip:20@192.168.0.250> Call-ID: 54473404-5060-16@BJC.BGI.A.BE CSeq: 160 INVITE Contact: <sip:192.168.0.14:5060> X-Grandstream-PBX: true Max-Forwards: 70 User-Agent: Grandstream GXP2140 1.0.5.18 Privacy: none P-Preferred-Identity: <sip:192.168.0.250> Supported: replaces, path, timer Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Type: application/sdp Accept: application/sdp, application/dtmf-relay Content-Length: 306 v=0 o=- 8000 8000 IN IP4 192.168.0.14 s=SIP Call c=IN IP4 192.168.0.14 t=0 0 m=audio 5004 RTP/AVP 9 8 18 2 101 a=sendrecv a=rtpmap:9 G722/8000 a=ptime:20 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:2 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 SIP/2.0 480 Temporarily not available Via: SIP/2.0/UDP 192.168.0.14:5060;branch=z9hG4bK1040318360;rport=5060 From: <sip:192.168.0.250>;tag=1540961770 To: <sip:20@192.168.0.250>;tag=74C1BCB3775433109F0E49A014240025 Call-ID: 54473404-5060-16@BJC.BGI.A.BE CSeq: 160 INVITE Content-Length: 0 ACK sip:20@192.168.0.250 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.14:5060;branch=z9hG4bK1040318360;rport From: <sip:192.168.0.250>;tag=1540961770 To: <sip:20@192.168.0.250>;tag=74C1BCB3775433109F0E49A014240025 Call-ID: 54473404-5060-16@BJC.BGI.A.BE CSeq: 160 ACK Content-Length: 0
但是來自來電:
INVITE sip:21@192.168.0.14;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.0.250:5060;branch=z9hG4bKE85E5CC7F25533109F4949A014240025;rport From: "Sys Tel 20" <sip:20@192.168.0.250;user=phone>;tag=1E2DB6AE775433109F0C49A014240025 To: <sip:21@192.168.0.250;user=phone> Call-ID: 303F5CC7F25533109F4849A014240025 CSeq: 1 INVITE Contact: <sip:21@192.168.0.250:5060;transport=udp> Max-Forwards: 70 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, MESSAGE, SUBSCRIBE, UPDATE, PRACK, REFER Supported: 100rel, replaces, timer User-Agent: hybird_130j V.9.1 Rev. 10 (Patch 4) IPSec Alert-Info: <http://127.0.0.1>;info=alert-internal Allow-Events: refer, message-summary, dialog P-Asserted-Identity: "Sys Tel 20" <sip:20@192.168.0.250;user=phone> Session-Expires: 1800 Content-Type: application/sdp Content-Length: 328 v=0 o=- 71 1 IN IP4 192.168.0.250 s=SIP call c=IN IP4 192.168.0.250 t=0 0 m=audio 10848 RTP/AVP 0 8 18 2 9 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:2 G726-32/8000 a=rtpmap:9 G722/8000 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 192.168.0.250:5060;branch=z9hG4bKE85E5CC7F25533109F4949A014240025;rport=5060 From: "Sys Tel 20" <sip:20@192.168.0.250;user=phone>;tag=1E2DB6AE775433109F0C49A014240025 To: <sip:21@192.168.0.250;user=phone> Call-ID: 303F5CC7F25533109F4849A014240025 CSeq: 1 INVITE Supported: replaces, path, timer User-Agent: Grandstream GXP2140 1.0.5.18 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.0.250:5060;branch=z9hG4bKE85E5CC7F25533109F4949A014240025;rport=5060 From: "Sys Tel 20" <sip:20@192.168.0.250;user=phone>;tag=1E2DB6AE775433109F0C49A014240025 To: <sip:21@192.168.0.250;user=phone>;tag=471383942 Call-ID: 303F5CC7F25533109F4849A014240025 CSeq: 1 INVITE Contact: <sip:192.168.0.14:5060> Supported: replaces, path, timer User-Agent: Grandstream GXP2140 1.0.5.18 Allow-Events: talk, hold Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.250:5060;branch=z9hG4bKE85E5CC7F25533109F4949A014240025;rport=5060 From: "Sys Tel 20" <sip:20@192.168.0.250;user=phone>;tag=1E2DB6AE775433109F0C49A014240025 To: <sip:21@192.168.0.250;user=phone>;tag=471383942 Call-ID: 303F5CC7F25533109F4849A014240025 CSeq: 1 INVITE Contact: <sip:192.168.0.14:5060> Supported: replaces, path, timer User-Agent: Grandstream GXP2140 1.0.5.18 Session-Expires: 1800;refresher=uac Require: timer Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Type: application/sdp Content-Length: 306
請注意,兩者都提供 PCMU/PCMA 和其他一些編解碼器。為什麼通話失敗?
電話的 IP 是
192.168.0.12
和192.168.0.14
,PBX 有192.168.0.250
。
可悲的事實是,那些非 elmeg 系統的電話沒有正確註冊。我必須在 Hybird 130j 界面中為使用者分配 PIN,然後用於向系統進行身份驗證。
Elmeg 沒有在任何地方說明必須設置此 PIN,並將其用作非 elmeg 電話的 SIP 密碼。
除非您有許可證,否則請刪除 g729。
使用 711a、711u、722、GSM。
檢查電話是否都在註冊,並且它們使用相同的傳輸 (udp) 以及 nat 設置是否相同。