488 Not acceptable here

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Asterisk Users 3 Comments

I am helping a friend on one of his sip trunk and couldn’t find the way
to resolve his problem.

His asterisk’s problem is like this:
0. When incoming call to one of his sip trunk, Asterisk reply with “488
Not acceptable here”. So the call get dropped.
1. Recently upgraded Elastix with Asterisk 1.4.33
2. Was working fine before the upgrade
3. There are total 4 SIP trunks connected to different providers. All
others works fine.
4. All codecs are allowed.
5. I setup his account on my Asterisk as a SIP trunk, both incoming and
outgoing call work fine. (So it is not his provider’s problem)
6. I checked his FreePBX style multi sip*.conf files and all seem correct.

So what can I do to find out where went wrong on this sip trunk?

Thanks.

Jian

Hers is the debug out put:
============================

<--- SIP read from 208.65.xxx.xxx:5060 --->
INVITE sip:160428xxxxx@192.168.1.83:5060 SIP/2.0
Via: SIP/2.0/UDP
208.65.xxx.xxx:5060;branch=z9hG4bK-d8754z-ad239907e0915d0b-1—d8754z-;rport
Via: SIP/2.0/UDP
208.65.xxx.xxx:5061;branch=z9hG4bK-pcerhxpz5hr4addh;rport=5061
Max-Forwards: 69
Record-Route:
Contact: “Anonymous”
To:
From: “CID NAME”;tag=kvspovbxperbwmfk.o
Call-ID: href=”mailto:12904465@208.xx.xx.xx”>12904465@208.xx.xx.xx~o
CSeq: 493 INVITE
Expires: 300
Content-Disposition: session
Content-Type: application/sdp
User-Agent: Sippy
cisco-GUID: 4084071434-3712422367-2859401243-560159692
h323-conf-id: 4084071434-3712422367-2859401243-560159692
Content-Length: 109

v=0
o=Sippy 153068680 0 IN IP4 74.205.xxx.xxx
s=-
t=0 0
m=audio 34772 RTP/AVP 0
c=IN IP4 74.205.xxx.xxx

<------------->

3 thoughts on - 488 Not acceptable here

  • Hello List;
    I am facing a trouble with a sip trunk on asterisk 1.4 and asterisk 1.8 and I am getting the following debug, can someone advise me about the solution:
    < --- SIP read from Provider_IP_Address:5083 --->INVITE sip:22021782@Asterisk_IP_Address:5060 SIP/2.0 Via: SIP/2.0/UDP Provider_IP_Address:5083;branch=z9hG4bKn1va9h109091cms8h5a0.1 From: “1828444” ;tag=rrZpHF51Z7a6D To:  Call-ID: 6bba7f72-3874-1234-0b95-0090fb3d96e0-UASession-CX9lUh8LWc-UASession-kOSBFqloi5 CSeq: 1 INVITE Max-Forwards: 68 Supported: timer Unsupported: refer Allow: INVITE,ACK,CANCEL,BYE,INFO,REGISTER,NOTIFY Contact:  Content-Length: 729 Content-Type: application/sdp User-Agent: Netborder SS7 to VoIP Media Gateway 5.1 Allow-Events: talk Accept: application/sdp Privacy: none X-IP-Info: 10.11.11.3  v=0 o=FreeSWITCH 1453083377 1453083378 IN IP4 Provider_IP_Address s=FreeSWITCH c=IN IP4 Provider_IP_Address t=0 0 m=audio 28388 RTP/AVP 8 0 98 9 99 100 18 3 102 101 13 a=rtpmap:98 AMR/8000 a=rtpmap:99 G7221/16000 a=fmtp:99 bitrate=32000 a=rtpmap:100 G726-32/8000 a=rtpmap:102 iLBC/8000 a=fmtp:102 mode=30 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 m=audio 29684 RTP/AVP 4 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:30 m=audio 21364 RTP/AVP 8 0 98 9 99 100 18 3 102 101 13 a=rtpmap:98 AMR/8000 a=rtpmap:99 G7221/16000 a=fmtp:99 bitrate=32000 a=rtpmap:100 G726-32/8000 a=rtpmap:102 iLBC/8000 a=fmtp:102 mode=30 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 
    < ------------->[Jan 18 10:52:37] VERBOSE[2421] logger.c: [Jan 18 10:52:37] — (18 headers 29 lines) —[Jan 18 10:52:37] VERBOSE[2421] logger.c: [Jan 18 10:52:37] Sending to Provider_IP_Address : 5083 (no NAT)[Jan 18 10:52:37] VERBOSE[2421] logger.c: [Jan 18 10:52:37] Using INVITE request as basis request – 6bba7f72-3874-1234-0b95-0090fb3d96e0-UASession-CX9lUh8LWc-UASession-kOSBFqloi5[Jan 18 10:52:37] VERBOSE[2421] logger.c: [Jan 18 10:52:37] Found peer ‘gulfnet'[Jan 18 10:52:37] VERBOSE[2421] logger.c: [Jan 18 10:52:37] Found RTP audio format 8[Jan 18 10:52:37] VERBOSE[2421] logger.c: [Jan 18 10:52:37] Found RTP audio format 0[Jan 18 10:52:37] VERBOSE[2421] logger.c: [Jan 18 10:52:37] Found RTP audio format 98[Jan 18 10:52:37] VERBOSE[2421] logger.c: [Jan 18 10:52:37] Found RTP audio format 9[Jan 18 10:52:37] VERBOSE[2421] logger.c: [Jan 18 10:52:37] Found RTP audio format 99[Jan 18 10:52:37] VERBOSE[2421] logger.c: [Jan 18 10:52:37] Found RTP audio format 100[Jan 18 10:52:37] VERBOSE[2421] logger.c: [Jan 18 10:52:37] Found RTP audio format 18[Jan 18 10:52:37] VERBOSE[2421] logger.c: [Jan 18 10:52:37] Found RTP audio format 3[Jan 18 10:52:37] VERBOSE[2421] logger.c: [Jan 18 10:52:37] Found RTP audio format 102[Jan 18 10:52:37] VERBOSE[2421] logger.c: [Jan 18 10:52:37] Found RTP audio format 101[Jan 18 10:52:37] VERBOSE[2421] logger.c: [Jan 18 10:52:37] Found RTP audio format 13[Jan 18 10:52:37] VERBOSE[2421] logger.c: [Jan 18 10:52:37] Found unknown media description format AMR for ID 98[Jan 18 10:52:37] VERBOSE[2421] logger.c: [Jan 18 10:52:37] Found unknown media description format G7221 for ID 99[Jan 18 10:52:37] VERBOSE[2421] logger.c: [Jan 18 10:52:37] Found audio description format G726-32 for ID 100[Jan 18 10:52:37] VERBOSE[2421] logger.c: [Jan 18 10:52:37] Found audio description format iLBC for ID 102[Jan 18 10:52:37] VERBOSE[2421] logger.c: [Jan 18 10:52:37] Found audio description format telephone-event for ID 101[Jan 18 10:52:37] VERBOSE[2421] logger.c: [Jan 18 10:52:37] Found RTP audio format 4[Jan 18 10:52:37] VERBOSE[2421] logger.c: [Jan 18 10:52:37] Found RTP audio format 101[Jan 18 10:52:37] VERBOSE[2421] logger.c: [Jan 18 10:52:37] Found RTP audio format 13[Jan 18 10:52:37] VERBOSE[2421] logger.c: [Jan 18 10:52:37] Found audio description format telephone-event for ID 101[Jan 18 10:52:37] VERBOSE[2421] logger.c: [Jan 18 10:52:37] Found RTP audio format 8[Jan 18 10:52:37] VERBOSE[2421] logger.c: [Jan 18 10:52:37] Found RTP audio format 0[Jan 18 10:52:37] VERBOSE[2421] logger.c: [Jan 18 10:52:37] Found RTP audio format 98[Jan 18 10:52:37] VERBOSE[2421] logger.c: [Jan 18 10:52:37] Found RTP audio format 9[Jan 18 10:52:37] VERBOSE[2421] logger.c: [Jan 18 10:52:37] Found RTP audio format 99[Jan 18 10:52:37] VERBOSE[2421] logger.c: [Jan 18 10:52:37] Found RTP audio format 100[Jan 18 10:52:37] VERBOSE[2421] logger.c: [Jan 18 10:52:37] Found RTP audio format 18[Jan 18 10:52:37] VERBOSE[2421] logger.c: [Jan 18 10:52:37] Found RTP audio format 3[Jan 18 10:52:37] VERBOSE[2421] logger.c: [Jan 18 10:52:37] Found RTP audio format 102[Jan 18 10:52:37] VERBOSE[2421] logger.c: [Jan 18 10:52:37] Found RTP audio format 101[Jan 18 10:52:37] VERBOSE[2421] logger.c: [Jan 18 10:52:37] Found RTP audio format 13[Jan 18 10:52:37] VERBOSE[2421] logger.c: [Jan 18 10:52:37] Found unknown media description format AMR for ID 98[Jan 18 10:52:37] VERBOSE[2421] logger.c: [Jan 18 10:52:37] Found unknown media description format G7221 for ID 99[Jan 18 10:52:37] VERBOSE[2421] logger.c: [Jan 18 10:52:37] Found audio description format G726-32 for ID 100[Jan 18 10:52:37] VERBOSE[2421] logger.c: [Jan 18 10:52:37] Found audio description format iLBC for ID 102[Jan 18 10:52:37] VERBOSE[2421] logger.c: [Jan 18 10:52:37] Found audio description format telephone-event for ID 101[Jan 18 10:52:37] VERBOSE[2421] logger.c: [Jan 18 10:52:37]< --- Reliably Transmitting (no NAT) to Provider_IP_Address:5083 --->SIP/2.0 488 Not acceptable here Via: SIP/2.0/UDP Provider_IP_Address:5083;branch=z9hG4bKn1va9h109091cms8h5a0.1;received=Provider_IP_Address From: “1828444”
    ;tag=rrZpHF51Z7a6D To: ;tag=as5d16dbaf Call-ID: 6bba7f72-3874-1234-0b95-0090fb3d96e0-UASession-CX9lUh8LWc-UASession-kOSBFqloi5 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0  
    < ------------>

    RegardsBilal

  • “488 Not acceptable here” usually means that negotiation failed for want of any mutually-supported codec. Make sure that you have “alaw”, which is the native format used by the PSTN in civilised countries (and therefore, there is little need to use anything else unless you know you will never want PSTN
    connectivity), enabled at your end.

    Can you run this command and post the output? (It should all be on one line, but my mail client or yours may have eaten it)

    $ awk ‘/[[]|allow/&&!/^[ \t]*;/{printf “%6d:%s\n”,NR, $0}’
    /etc/asterisk/sip.conf

    This will look for [section headers] in square brackets and lines containing
    “allow” (which also will catch “disallow”) that are not commented out, in your SIP configuration, and print them out with line numbers.

  • Hello;
    Thanks a lot for your kindly reply.Actually the alaw is enabled at asterisk but what I got to know from the other side that they only enabled ulaw. Below is my asterisk sip configuration for the sip trunk. Please advise.
    [user_name]type=peerhost=Provider_IP_Addressport=5083context=trunkinbounddisallow=allallow = ulaw,alaw,gsmcall-limit = 256  insecure = port,invitetrunkstyle = providertransport = udp  dtmfmode = rfc2833remoteregister = yescbcallerid = 22021782qualify = yessrtpcapable = no RegardsBilal

    “488 Not acceptable here” usually means that negotiation failed for want of any mutually-supported codec.  Make sure that you have “alaw”, which is the native format used by the PSTN in civilised countries  (and therefore, there is little need to use anything else unless you know you will never want PSTN
    connectivity),  enabled at your end.

    Can you run this command and post the output?  (It should all be on one line, but my mail client or yours may have eaten it)

    $ awk ‘/[[]|allow/&&!/^[ \t]*;/{printf “%6d:%s\n”,NR, $0}’
    /etc/asterisk/sip.conf

    This will look for [section headers] in square brackets and lines containing
    “allow” (which also will catch “disallow”) that are not commented out, in your SIP configuration, and print them out with line numbers.