在 Asterisk 中响应“488 Not Acceptable here”
Response "488 Not Acceptable here" in Asterisk
我在这里和 google 上阅读了很多关于此的帖子，但我仍然无法解决问题。我在服务器上安装了 Asterisk 并从 GSM 调用它。跟踪显示 488 Not Acceptable Here。这是日志
<--- SIP read from UDP:xxx.xxx.xxx.xxx:5078 --->
INVITE sip:1002@xxx.xx.x.xx;user=phone SIP/2.0
Via: SIP/2.0/UDP xxx.xx.x.xx:5078;branch=z9hG4bKiectcmpi5pjew7vw7etticvmv;X-DispMsg=1401
Route: <sip:xxx.xx.x.xx:5060;transport=udp;lr>
Call-ID: t7mjcpnsmcc668tsnwjijwnmiucvjsuv@xxx.xx.x.xx
From: "1003"<sip:1003@xxx.xx.x.xx;transport=udp;user=phone>;tag=vww8u6mn-CC-1005-OFC-64
To: "1002"<sip:1002@xxx.xx.x.xx;transport=udp;user=phone>
CSeq: 1 INVITE
P-Charging-Vector: icid-value=A621B143ED238320161219141053;orig-ioi=xxx.xx.x.xx
Max-Forwards: 70
P-Access-Network-Info: GEN-ACCESS;"area-number=+xxx"
Contact: <sip:xxx.xx.x.xx:5060>
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,NOTIFY,MESSAGE,REFER,UPDATE
P-Asserted-Identity: <tel:878010200>
Supported: 100rel,timer,histinfo,precondition
Min-SE: 90
Session-Expires: 1800;refresher=uac
P-Early-Media: supported
Content-Length: 335
Content-Type: application/sdp
v=0
o=HuaweiSoftx3000 1073786885 1073786886 IN IP4 xxx.xx.x.xx
s=SipCall
c=IN IP4 xxx.xx.x.xx
t=0 0
m=audio 41908 RTP/AVP 8 116
a=rtpmap:8 PCMA/8000
a=rtpmap:116 telephone-event/8000
a=ptime:5
a=curr:qos local sendrecv
a=curr:qos remote none
a=des:qos optional local sendrecv
a=des:qos optional remote sendrecv
a=3gOoBTC
<------------->
--- (19 headers 14 lines) ---
Sending to xxx.xx.x.xx:5078 (NAT)
Sending to xxx.xx.x.xx:5078 (NAT)
Using INVITE request as basis request - t7mjcpnsmcc668tsnwjijwnmiucvjsuv@xxx.xx.x.xx
Found peer '1003' for '1003' from xxx.xx.x.xx:5078
== Using SIP RTP CoS mark 5
Found RTP audio format 8
Found RTP audio format 116
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 116
[Dec 19 09:10:00] NOTICE[4051][C-0000004a]: chan_sip.c:10563 process_sdp: No compatible codecs, not accepting this offer!
注意：IP 是虚拟的，因为信息是敏感的。我相信这是关于
的部分
m=audio 41908 RTP/AVP 8 116
a=rtpmap:8 PCMA/8000
a=rtpmap:116 telephone-event/8000
如您所见，错误与编解码器有关。
[Dec 19 09:10:00] NOTICE[4051][C-0000004a]: chan_sip.c:10563 process_sdp: No compatible codecs, not accepting this offer!
我在服务器端添加了这个编解码器
sip.conf
[general]
regcontext=dundiextens
srvlookup=no
nat=force_rport
bindport=5060
allowguest=yes
canreinvite=no
rtcachefriends=yes
disallow=all
allow=alaw
allow=ulaw
allow=g729
allow=gsm
有人可以帮我解决这个问题吗？
CLI 中的编解码器：
*CLI> core show codecs
Disclaimer: this command is for informational purposes only.
It does not indicate anything about your configuration.
ID TYPE NAME DESCRIPTION
-----------------------------------------------------------------------------------
30 image png (PNG Image)
5 audio g726 (G.726 RFC3551)
3 audio alaw (G.711 a-law)
1 audio g723 (G.723.1)
19 audio speex (SpeeX)
20 audio speex (SpeeX 16khz)
21 audio speex (SpeeX 32khz)
23 audio g722 (G722)
31 video h261 (H.261 video)
32 video h263 (H.263 video)
7 audio adpcm (Dialogic ADPCM)
24 audio siren7 (ITU G.722.1 (Siren7, licensed from Polycom))
27 audio g719 (ITU G.719)
33 video h263p (H.263+ video)
34 video h264 (H.264 video)
18 audio g729 (G.729A)
8 audio slin (16 bit Signed Linear PCM)
9 audio slin (16 bit Signed Linear PCM (12kHz))
10 audio slin (16 bit Signed Linear PCM (16kHz))
11 audio slin (16 bit Signed Linear PCM (24kHz))
12 audio slin (16 bit Signed Linear PCM (32kHz))
13 audio slin (16 bit Signed Linear PCM (44kHz))
14 audio slin (16 bit Signed Linear PCM (48kHz))
15 audio slin (16 bit Signed Linear PCM (96kHz))
16 audio slin (16 bit Signed Linear PCM (192kHz))
2 audio ulaw (G.711 u-law)
17 audio lpc10 (LPC10)
26 audio testlaw (G.711 test-law)
39 audio none (<Null> codec)
25 audio siren14 (ITU G.722.1 Annex C, (Siren14, licensed from Polycom))
6 audio g726aal2 (G.726 AAL2)
36 video vp8 (VP8 video)
4 audio gsm (GSM)
35 video mpeg4 (MPEG4 video)
22 audio ilbc (iLBC)
37 text red (T.140 Realtime Text with redundancy)
38 text t140 (Passthrough T.140 Realtime Text)
28 audio opus (Opus Codec)
29 image jpeg (JPEG image)
我看到您在 [general] 中允许了一些编解码器，但您是否在对等配置中禁止了任何编解码器？对等配置将覆盖 [general] 中的任何内容。
执行 sip show peer PEERNAME 检查允许特定对等点的编解码器。如果缺少编解码器，则更新配置。
检查至少一种来自 sip show peer PEERNAME 的编解码器在您使用的软电话上可用。
我在这里和 google 上阅读了很多关于此的帖子，但我仍然无法解决问题。我在服务器上安装了 Asterisk 并从 GSM 调用它。跟踪显示 488 Not Acceptable Here。这是日志
<--- SIP read from UDP:xxx.xxx.xxx.xxx:5078 --->
INVITE sip:1002@xxx.xx.x.xx;user=phone SIP/2.0
Via: SIP/2.0/UDP xxx.xx.x.xx:5078;branch=z9hG4bKiectcmpi5pjew7vw7etticvmv;X-DispMsg=1401
Route: <sip:xxx.xx.x.xx:5060;transport=udp;lr>
Call-ID: t7mjcpnsmcc668tsnwjijwnmiucvjsuv@xxx.xx.x.xx
From: "1003"<sip:1003@xxx.xx.x.xx;transport=udp;user=phone>;tag=vww8u6mn-CC-1005-OFC-64
To: "1002"<sip:1002@xxx.xx.x.xx;transport=udp;user=phone>
CSeq: 1 INVITE
P-Charging-Vector: icid-value=A621B143ED238320161219141053;orig-ioi=xxx.xx.x.xx
Max-Forwards: 70
P-Access-Network-Info: GEN-ACCESS;"area-number=+xxx"
Contact: <sip:xxx.xx.x.xx:5060>
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,NOTIFY,MESSAGE,REFER,UPDATE
P-Asserted-Identity: <tel:878010200>
Supported: 100rel,timer,histinfo,precondition
Min-SE: 90
Session-Expires: 1800;refresher=uac
P-Early-Media: supported
Content-Length: 335
Content-Type: application/sdp
v=0
o=HuaweiSoftx3000 1073786885 1073786886 IN IP4 xxx.xx.x.xx
s=SipCall
c=IN IP4 xxx.xx.x.xx
t=0 0
m=audio 41908 RTP/AVP 8 116
a=rtpmap:8 PCMA/8000
a=rtpmap:116 telephone-event/8000
a=ptime:5
a=curr:qos local sendrecv
a=curr:qos remote none
a=des:qos optional local sendrecv
a=des:qos optional remote sendrecv
a=3gOoBTC
<------------->
--- (19 headers 14 lines) ---
Sending to xxx.xx.x.xx:5078 (NAT)
Sending to xxx.xx.x.xx:5078 (NAT)
Using INVITE request as basis request - t7mjcpnsmcc668tsnwjijwnmiucvjsuv@xxx.xx.x.xx
Found peer '1003' for '1003' from xxx.xx.x.xx:5078
== Using SIP RTP CoS mark 5
Found RTP audio format 8
Found RTP audio format 116
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 116
[Dec 19 09:10:00] NOTICE[4051][C-0000004a]: chan_sip.c:10563 process_sdp: No compatible codecs, not accepting this offer!
注意：IP 是虚拟的，因为信息是敏感的。我相信这是关于
的部分m=audio 41908 RTP/AVP 8 116
a=rtpmap:8 PCMA/8000
a=rtpmap:116 telephone-event/8000
如您所见，错误与编解码器有关。
[Dec 19 09:10:00] NOTICE[4051][C-0000004a]: chan_sip.c:10563 process_sdp: No compatible codecs, not accepting this offer!
我在服务器端添加了这个编解码器
sip.conf
[general]
regcontext=dundiextens
srvlookup=no
nat=force_rport
bindport=5060
allowguest=yes
canreinvite=no
rtcachefriends=yes
disallow=all
allow=alaw
allow=ulaw
allow=g729
allow=gsm
有人可以帮我解决这个问题吗？
CLI 中的编解码器：
*CLI> core show codecs
Disclaimer: this command is for informational purposes only.
It does not indicate anything about your configuration.
ID TYPE NAME DESCRIPTION
-----------------------------------------------------------------------------------
30 image png (PNG Image)
5 audio g726 (G.726 RFC3551)
3 audio alaw (G.711 a-law)
1 audio g723 (G.723.1)
19 audio speex (SpeeX)
20 audio speex (SpeeX 16khz)
21 audio speex (SpeeX 32khz)
23 audio g722 (G722)
31 video h261 (H.261 video)
32 video h263 (H.263 video)
7 audio adpcm (Dialogic ADPCM)
24 audio siren7 (ITU G.722.1 (Siren7, licensed from Polycom))
27 audio g719 (ITU G.719)
33 video h263p (H.263+ video)
34 video h264 (H.264 video)
18 audio g729 (G.729A)
8 audio slin (16 bit Signed Linear PCM)
9 audio slin (16 bit Signed Linear PCM (12kHz))
10 audio slin (16 bit Signed Linear PCM (16kHz))
11 audio slin (16 bit Signed Linear PCM (24kHz))
12 audio slin (16 bit Signed Linear PCM (32kHz))
13 audio slin (16 bit Signed Linear PCM (44kHz))
14 audio slin (16 bit Signed Linear PCM (48kHz))
15 audio slin (16 bit Signed Linear PCM (96kHz))
16 audio slin (16 bit Signed Linear PCM (192kHz))
2 audio ulaw (G.711 u-law)
17 audio lpc10 (LPC10)
26 audio testlaw (G.711 test-law)
39 audio none (<Null> codec)
25 audio siren14 (ITU G.722.1 Annex C, (Siren14, licensed from Polycom))
6 audio g726aal2 (G.726 AAL2)
36 video vp8 (VP8 video)
4 audio gsm (GSM)
35 video mpeg4 (MPEG4 video)
22 audio ilbc (iLBC)
37 text red (T.140 Realtime Text with redundancy)
38 text t140 (Passthrough T.140 Realtime Text)
28 audio opus (Opus Codec)
29 image jpeg (JPEG image)
我看到您在 [general] 中允许了一些编解码器，但您是否在对等配置中禁止了任何编解码器？对等配置将覆盖 [general] 中的任何内容。
执行 sip show peer PEERNAME 检查允许特定对等点的编解码器。如果缺少编解码器，则更新配置。
检查至少一种来自 sip show peer PEERNAME 的编解码器在您使用的软电话上可用。