[SOLVED] Response “488 Not Acceptable here” in Asterisk

Asterisk Asterisk SIP
1

'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: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!

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

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
;allow = g729
;disallow=all
;allow=alaw
;allow=gsm
allow=ulaw

I’ve tried all kind of combinations in sip.conf and everytime this error came.

2

What is the configuration of the peer ‘1003’ that the INVITE has matched against?

3

It is using Adding RealTime with mysql database and reading the extension directly from there from sipfriends and voicemail_users table. In this case I’m simulating it and calling to voicemail from the gsm which number I have in database saved.

4

The question is still applicable, what is the configuration? If the peer has differently configured codecs then you would see exactly the behavior you are seeing - a 488.

5

under column allow is gsm, under column disallow is all. I’ve tried to change allow to ulaw, alaw but got same error - 488. Is that what you mean by 'configuration` for the number/?

6

Yes. Set allow to alaw. You are also caching so you will need to restart Asterisk, do a “sip reload”, or use “sip prune realtime 1003” to have the change take effect.

1 Like
7

I’ve set in database allow to alaw changed it in sip.conf and currently looks like:

[general]
regcontext=dundiextens
srvlookup=no
nat=force_rport
bindport=5060
allowguest=yes
canreinvite=no
rtcachefriends=yes
disallow=all
allow=alaw

Then I’ve made sip prune realtime 1003

*CLI> sip prune realtime 1003
Peer '1003' pruned.

sip reload

*CLI> sip reload
 Reloading SIP
  == Parsing '/etc/asterisk/sip.conf': Found
  == Parsing '/etc/asterisk/users.conf': Found
  == Using SIP CoS mark 4
  == Parsing '/etc/asterisk/sip_notify.conf': Found

And this seems to fixed the issue. I didn’t knew that it is caching it and I need to prune realtime. Thank you.

1 Like
9

How can I mark the thread as Solved?