Response "488 Not Acceptable here" in Asterisk

Question

I've readed quite a lot of posts here and on google about this but still I'm unable to resolve the issue. I have installed Asterisk on the server and calling to it from GSM. The trace show 488 Not Acceptable Here. This is the log

<--- SIP read from UDP:xxx.xxx.xxx.xxx:5078 --->
INVITE sip:[email protected];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: [email protected]
From: "1003"<sip:[email protected];transport=udp;user=phone>;tag=vww8u6mn-CC-1005-OFC-64
To: "1002"<sip:[email protected];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 - [email protected]
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!

Note: IP's are dummy since the information is sensitive. I believe this is the part which is about

m=audio 41908 RTP/AVP 8 116
a=rtpmap:8 PCMA/8000
a=rtpmap:116 telephone-event/8000

And as you can see the error is about codecs.

[Dec 19 09:10:00] NOTICE[4051][C-0000004a]: chan_sip.c:10563 process_sdp: No compatible codecs, not accepting this offer!

I have added this codecs on server side

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

Can someone help me on this?

Codecs in CLI:

    *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)

you can check the codecs enabled in asterisk using asterisk cli and the following command: show codecs **, show translation and show translation recalc 10 if I consider voip-info.org/wiki/view/Asterisk+codecs probably not the best advice, but did you try to restart your asterisk server or make it reload its config files? — yohann.martineau
Thanks. I see lot's of codecs when I type above commands. But which one is this a=rtpmap:8 PCMA/8000 — VLS
I've made core reload, sip reload, iax2 reload .. all kinds of reloads.. — VLS

Dan_J Dan_J · Accepted Answer · 2016-12-21T10:15:30

I see that you've allowed some codecs in [general], but have you disallowed any in the peer config? The peer config will override anything in [general].

Do sip show peer PEERNAME to check which codecs that specific peer is allowed. Then update the config if a codec is missing.

Check at least one of the codecs from sip show peer PEERNAME is available on the softphone you are using.

Response "488 Not Acceptable here" in Asterisk

1 Answers