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[support at telium.ca]

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@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@192.168.253.4>;tag=as7b616c8d
To: <sip:290006@192.168.253.20;line=14994>
Contact: <sip:230@192.168.253.4:5060>
Call-ID: 36334383058109cd2325341a0f18ac79@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@192.168.253.4>;tag=as7b616c8d
To: <sip:290006@192.168.253.20;line=14994>
Call-ID: 36334383058109cd2325341a0f18ac79@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@192.168.253.4>;tag=as7b616c8d
To: <sip:290006@192.168.253.20;line=14994>;tag=ld65q
Call-ID: 36334383058109cd2325341a0f18ac79@192.168.253.4:5060
CSeq: 102 INVITE
Contact: <sip:290006@192.168.253.20;line=14994>
User-Agent: snomM700/03.24.0007 (MAC=0004136103FB; SER= 00000; HW=255)
Content-Length: 0



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[prado at practis.com.br]

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@telium.ca]
Received: sexta-feira, 21 ago 2015, 19:46
To: asterisk-users@lists.digium.com [asterisk-users@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@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@192.168.253.4>;tag=as7b616c8d
To: <sip:290006@192.168.253.20;line=14994>
Contact: <sip:230@192.168.253.4:5060>
Call-ID: 36334383058109cd2325341a0f18ac79@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@192.168.253.4>;tag=as7b616c8d
To: <sip:290006@192.168.253.20;line=14994>
Call-ID: 36334383058109cd2325341a0f18ac79@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@192.168.253.4>;tag=as7b616c8d
To: <sip:290006@192.168.253.20;line=14994>;tag=ld65q
Call-ID: 36334383058109cd2325341a0f18ac79@192.168.253.4:5060
CSeq: 102 INVITE
Contact: <sip:290006@192.168.253.20;line=14994>
User-Agent: snomM700/03.24.0007 (MAC=0004136103FB; SER= 00000; HW=255)
Content-Length: 0



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--
_____________________________________________________________________
-- 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
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[andres at telesip.net]

On 8/21/15 6:45 PM, Technical Support wrote: