Hi,
By the sip trace is very difficult to tell because the SIP messages are fine.
Try to enable all codec, and if possible copy and paste your asterisk sip
configuration for this peer.
Enviado do meu telefone Android usando o Symantec TouchDown (www.symantec.com)
-----Original Message-----
From: Technical Support [support at telium.ca]
Received: sexta-feira, 21 ago 2015, 19:46
To: asterisk-users at lists.digium.com [asterisk-users at lists.digium.com]
Subject: [asterisk-users] Incoming calls get 488 error
I got a new SNOM M65 which works fine for outgoing calls, but incoming
calls never ring at the handset. I captured the SIP traffic and see
that my M65 is replying with an "488 not acceptable here". From what
I
read this is usually codec related but both asterisk and the M65 are set
for ulaw as first choice.
I have a SIP trace below. Can someone suggest why the 488 is being
generated?
-----------------------------------
Received from udp:192.168.253.4:5060 at 21/08/2015 18:37:00 (1198 bytes)
INVITE sip:290006 at 192.168.253.20;line=14994 SIP/2.0
Via: SIP/2.0/UDP 192.168.253.4:5060;branch=z9hG4bK4990095a
Max-Forwards: 70
From: "test user" <sip:230 at 192.168.253.4>;tag=as7b616c8d
To: <sip:290006 at 192.168.253.20;line=14994>
Contact: <sip:230 at 192.168.253.4:5060>
Call-ID: 36334383058109cd2325341a0f18ac79 at 192.168.253.4:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.11.0(11.10.2)
Date: Fri, 21 Aug 2015 22:37:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 606
v=0
o=root 1678280845 1678280845 IN IP4 192.168.253.4
s=Asterisk PBX 11.10.2
c=IN IP4 192.168.253.4
b=CT:384
t=0 0
m=audio 18090 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
m=video 12226 RTP/AVP 99 98 34 31
a=rtpmap:99 H264/90000
a=fmtp:99
redundant-pic-cap=0;parameter-add=0;packetization-mode=0;level-asymmetry-allowed=0
a=rtpmap:98 H263-1998/90000
a=fmtp:98 F=0;I=0;J=0;T=0;K=0;N=0;BPP=0;HRD=0
a=rtpmap:34 H263/90000
a=fmtp:34 F=0;I=0;J=0;T=0;K=0;N=0;BPP=0;HRD=0
a=rtpmap:31 H261/90000
a=sendrecv
Sent to udp:192.168.253.4:5060 at 21/08/2015 18:37:00 (280 bytes)
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.253.4:5060;branch=z9hG4bK4990095a
From: "test user" <sip:230 at 192.168.253.4>;tag=as7b616c8d
To: <sip:290006 at 192.168.253.20;line=14994>
Call-ID: 36334383058109cd2325341a0f18ac79 at 192.168.253.4:5060
CSeq: 102 INVITE
Content-Length: 0
Sent to udp:192.168.253.4:5060 at 21/08/2015 18:37:00 (441 bytes)
SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/UDP 192.168.253.4:5060;branch=z9hG4bK4990095a
Max-Forwards: 70
From: "test user" <sip:230 at 192.168.253.4>;tag=as7b616c8d
To: <sip:290006 at 192.168.253.20;line=14994>;tag=ld65q
Call-ID: 36334383058109cd2325341a0f18ac79 at 192.168.253.4:5060
CSeq: 102 INVITE
Contact: <sip:290006 at 192.168.253.20;line=14994>
User-Agent: snomM700/03.24.0007 (MAC=0004136103FB; SER= 00000; HW=255)
Content-Length: 0
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users