All,
I’m wondering if anyone can help me figure out why I am seeing this SIP response error. I know there have been postings about this in the past (like lists.digium.com/pipermail/aster … 42627.html), but none of them have seemed to help. Here’s our situation:
We are testing a new SIP client which is calling an Asterisk server and whenever we send an INVITE message we get a “488 Not Acceptable Here (codec error)” response. The strange thing is that we only seem to get this response when using this particular SIP client. However the SIP Client is able to talk to other SIP servers without any problem. Seems to only happen on Asterisk. Making a call with X-Pro or other UAs seem to work fine. We’ve taken a look at the SIP traces and can’t seem to see how the INVITE messages are all that different from each other. We’re assuming it’s not a config issue because it works with some SIP Clients and not others.
Following is an INVITE message that works fine (I’ve xxx’d some stuff out):
INVITE sip:5555678@test.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.1:5060;rport;branch=z9hG4bK411E7113C26B46C1A13A3AF469E9ACC4
From: Jim Heising sip:5551234@test.com;tag=1393924468
To: sip:5555678@test.com
Contact: sip:5551234@test.com:5060
Call-ID: E5332061-67A5-41FF-B445-31BDC72075AF@192.168.1.1
CSeq: 15091 INVITE
Proxy-Authorization: Digest username=“test”,realm=“asterisk”,nonce=“5d0e4b80”,response=“a044bad7c38f3de5a77240c5d82ae075”,uri="sip:5555678@test.com"
Max-Forwards: 70
Content-Type: application/sdp
User-Agent: X-Lite release 1103m
Content-Length: 276
v=0
o=test 254083140 254083156 IN IP4 192.168.1.1
s=X-Lite
c=IN IP4 192.168.1.1
t=0 0
m=audio 8000 RTP/AVP 0 8 3 98 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:3 gsm/8000
a=rtpmap:98 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
Following is the INVITE Message that receives the 488 Error:
INVITE sip:5555678@test.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.1:5060;rport;branch=z9hG4bK0EBF15A1CC814A2D9AC88ACDE999F076
From: “Jim Heising” sip:5551234@test.com;tag=C6FA3F8314C24D3AAA4AB84D2060F3F9
To: sip:sip:5555678@test.com
Contact: sip:5551234@test.com:5060
Call-ID: 5B05D0877FE04ABD8B3A13360B53A3B7@192.168.1.1
CSeq: 5079 INVITE
Max-Forwards: 70
Content-Type: application/sdp
User-Agent: WOSISIP 1.0
Proxy-Authorization: Digest username=“test”,realm=“asterisk”,nonce=“6714cfdc”,response=“3ee314cc98524dba0608ebddafd6980c”,uri="sip:5555678@test.com",algorithm=MD5
Content-Length: 349
v=0
o=test 632623981138593750 632623981138593750 IN IP4 192.168.1.1
s=WOSI SIP 0.5
c=IN IP4 192.168.1.1
t=0 0
m=audio 8000 RTP/AVP 0 8 3 98 7 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:98 iLBC/8000
a=rtpmap:7 LPC/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
I know there are some extra codecs listed in the SDP message body but I wouldn’t think that would cause a problem since the first codecs are the same and should match up with a codec on the server.
Anyone have any ideas? I’m beginning to think that this (codec error) message may not really be a codec error message and may be an indicator of something else wrong with the message?
Any help would be greatly appreciated!
-alphamacaroon