Question

我已经在这里和谷歌上发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

正如您所看到的，错误与编解码器有关。

[Dec 19 09:10:00]注意[4051] [C-0000004a]：chan_sip.c：10563 process_sdp：没有兼容的编解码器，不接受此优惠！

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)

Answer 1

我看到你在[general]中允许了一些编解码器，但是你在同伴配置中禁止了吗？对等配置将覆盖[general]中的任何内容。

执行sip show peer PEERNAME以检查允许特定对等方的编解码器。如果缺少编解码器，则更新配置。

检查您正在使用的软电话上是否至少有sip show peer PEERNAME中的一个编解码器。

在Asterisk中回复“488 Not Acceptable here”

1 个答案: