![SIP:沒有匹配的編解碼器/480 暫時不可用](https://rvso.com/image/668803/SIP%EF%BC%9A%E6%B2%92%E6%9C%89%E5%8C%B9%E9%85%8D%E7%9A%84%E7%B7%A8%E8%A7%A3%E7%A2%BC%E5%99%A8%2F480%20%E6%9A%AB%E6%99%82%E4%B8%8D%E5%8F%AF%E7%94%A8.png)
我正在 VoIP 環境中安裝 SIP 電話。有 2 部系統電話正常運作(與 PBX 同一製造商),第三部電話可以呼叫,但無法呼叫其他 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進行的對話:
INVITE sip:[email protected] 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:[email protected]>
Call-ID: [email protected]
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:[email protected]>;tag=74C1BCB3775433109F0E49A014240025
Call-ID: [email protected]
CSeq: 160 INVITE
Content-Length: 0
ACK sip:[email protected] 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:[email protected]>;tag=74C1BCB3775433109F0E49A014240025
Call-ID: [email protected]
CSeq: 160 ACK
Content-Length: 0
但從來電:
INVITE sip:[email protected];transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.0.250:5060;branch=z9hG4bKE85E5CC7F25533109F4949A014240025;rport
From: "Sys Tel 20" <sip:[email protected];user=phone>;tag=1E2DB6AE775433109F0C49A014240025
To: <sip:[email protected];user=phone>
Call-ID: 303F5CC7F25533109F4849A014240025
CSeq: 1 INVITE
Contact: <sip:[email protected]: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:[email protected];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:[email protected];user=phone>;tag=1E2DB6AE775433109F0C49A014240025
To: <sip:[email protected];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:[email protected];user=phone>;tag=1E2DB6AE775433109F0C49A014240025
To: <sip:[email protected];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:[email protected];user=phone>;tag=1E2DB6AE775433109F0C49A014240025
To: <sip:[email protected];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 和一些其他編解碼器。為什麼呼叫失敗?
電話的 IP 為192.168.0.12
和192.168.0.14
,PBX 的 IP 為192.168.0.250
。
答案1
可悲的事實是,那些非elmeg系統的手機沒有正確註冊。我必須在 Hybird 130j 介面中為使用者指派 PIN,然後用於透過系統進行身份驗證。
Elmeg 沒有在任何地方聲明必須設定此 PIN,並將其用作非 Elmeg 電話的 SIP 密碼。
答案2
除非您有許可證,否則刪除 g729。
使用 711a、711u、722、GSM。
檢查電話是否全部註冊,它們使用相同的傳輸 (udp) 以及 nat 設定是否全部相同。