我安装了一个星号，它一次又一次地重复这个语句，我的服务器因为这些错误而超载。
[Sep 19 18:34:23] NOTICE[1641][C-00000398]: chan_sip.c:10995 process_sdp: No compatible codecs, not accepting this offer!
请有人帮助我。
--- (10 headers 0 lines) ---
Really destroying SIP dialog '[email protected]' Method: ACK
<--- SIP read from UDP:111.111.128.46:5060 --->
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 111.111.128.46:5060;branch=z9hG4bK5814af823fb470c7
From: <sip:[email protected]>;tag=77b347fa5b331553
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 21703 INVITE
Contact: <sip:[email protected]:5060>
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, UPDATE, PRACK
Supported: timer, linknat
Original-Info: hIEtgm6x4Rq4wFcwf0ECXFlXQQceAk8GX+Ku
Max-Forwards: 70
User-Agent: VOS3000 V2.1.7.03
Session-Expires: 600
Original-ID: 17b43a2f-11b45755-7b663-7fc95ce87268-8a534841-13c4-7225
Content-Type: application/sdp
Content-Length: 223
v=0
o=- 36535 36535 IN IP4 111.111.128.46
s=VOS3000
c=IN IP4 111.111.128.46
t=0 0
m=audio 37366 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
<------------->
--- (16 headers 11 lines) ---
Sending to 111.111.128.46:5060 (no NAT)
Sending to 111.111.128.46:5060 (no NAT)
Using INVITE request as basis request - [email protected]
No matching peer for '442075304284' from '111.111.128.46:5060'
== Using SIP RTP CoS mark 5
Got SDP version 36535 and unique parts [- 36535 IN IP4 94.130.128.46]
Found RTP audio format 18
Found RTP audio format 101
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
[Sep 19 18:45:59] NOTICE[1641][C-0000114a]: chan_sip.c:10995 process_sdp: No compatible codecs, not accepting this offer!
<--- Reliably Transmitting (no NAT) to 111.111.128.46:5060 --->
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP 94.130.128.46:5060;branch=z9hG4bK5814af823fb470c7;received=94.130.128.46
From: <sip:[email protected]>;tag=77b347fa5b331553
To: <sip:[email protected]>;tag=as7a56c326
Call-ID: [email protected]
CSeq: 21703 INVITE
Server: Asterisk PBX 16.28.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0```
答案1
您的设备（VOS3000 V2.1.7.03）位于 111.111.128.46，配置为仅提供单一语音编解码器 G729：
m=audio 37366 RTP/AVP 18 101
a=rtpmap:18 G729/8000
但是我怀疑您的 Asterisk 盒子没有 G729 模块和/或许可证。您可以检查：
*CLI> core show translation paths g729
--- Translation paths SRC Codec "g729" sample rate 8000 ---
还要确保 SIP 对等体allow=g729也已配置。
但我建议配置更常见的设备提供 G711。