Hi,
Recently upgraded freepbx and thus Asterisk to 11.6.0. Since doing this, outbound calls no longer work. The only output on the console (for any valid number dialed) is:
[2014-01-24 20:04:15] WARNING[1594][C-00000006]: chan_sip.c:24960 handle_request_invite: Received SIP INVITE with unsupported required extension: required:sdp-anat unsupported:sdp-anat
Anyone know which setting and where I need to disable it, to get rid of the SDP-ANAT requirement?
Thanks,
Matt.
Managed to get some real debug to match this issue. All phones in the system can dial outbound, except the 2 cisco 9971’s. They are getting a BAD EXTENSION error.
Hoping someone has come across this before.
Thanks,
Matt.
<--- SIP read from UDP:192.168.0.222:5060 --->
INVITE sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.0.222:5060;branch=z9hG4bK6ed72c22
From: "Study" ;tag=18339d14faf522c963fdc80f-62e302eb
To:
Call-ID: [email protected]
Max-Forwards: 70
Date: Wed, 29 Jan 2014 00:08:51 GMT
CSeq: 101 INVITE
User-Agent: Cisco-CP9971/9.4.1
Contact: ;video
Expires: 180
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE,INFO
Require: sdp-anat
Supported: replaces,join,sdp-anat,norefersub,resource-priority,extended-refer,X-cisco-callinfo,X-cisco-serviceuri,X-cisco-escapecodes,X-cisco-service-control,X-cisco-srtp-fallback,X-cisco-monrec,X-cisco-config,X-cisco-sis-6.0.2,X-cisco-xsi-8.0.1
Allow-Events: kpml,dialog
Authorization: Digest username="103",realm="asterisk",uri="sip:[email protected];user=phone",response="e908c8fef71e4def981ed1513ed1377e",nonce="38acc7b8",algorithm=MD5
Content-Length: 631
Content-Type: application/sdp
Content-Disposition: session;handling=optional
v=0
o=Cisco-SIPUA 1170 0 IN IP4 192.168.0.222
s=SIP Call
t=0 0
m=audio 16476 RTP/AVP 0 8 18 102 9 116 124 101
c=IN IP4 192.168.0.222
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:102 L16/16000
a=rtpmap:9 G722/8000
a=rtpmap:116 iLBC/8000
a=fmtp:116 mode=20
a=rtpmap:124 ISAC/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
m=video 21384 RTP/AVP 97
c=IN IP4 192.168.0.222
b=TIAS:1000000
a=rtpmap:97 H264/90000
a=fmtp:97 profile-level-id=42801E;packetization-mode=0;level-asymmetry-allowed=1
a=imageattr:* recv [x=640,y=480,q=0.50]
a=sendrecv
<------------->
— (20 headers 25 lines) —
Sending to 192.168.0.222:5060 (NAT)
<— Transmitting (NAT) to 192.168.0.222:5060 —>
SIP/2.0 420 Bad extension (unsupported)
Via: SIP/2.0/UDP 192.168.0.222:5060;branch=z9hG4bK6ed72c22;received=192.168.0.222;rport=5060
From: “Study” sip:[email protected];tag=18339d14faf522c963fdc80f-62e302eb
To: sip:[email protected];tag=as7de9e264
Call-ID: [email protected]
CSeq: 101 INVITE
Server: FPBX-2.11.0(11.6.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Date: Wed, 29 Jan 2014 00:09:00 GMT
Unsupported: sdp-anat
Content-Length: 0
<------------>
[2014-01-29 10:09:00] WARNING[1668][C-00000003]: chan_sip.c:24960 handle_request_invite: Received SIP INVITE with unsupported required extension: required:sdp-anat unsupported:sdp-anat
Scheduling destruction of SIP dialog ‘[email protected]’ in 32000 ms (Method: INVITE)
<— SIP read from UDP:192.168.0.222:5060 —>
ACK sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.0.222:5060;branch=z9hG4bK6ed72c22
From: “Study” sip:[email protected];tag=18339d14faf522c963fdc80f-62e302eb
To: sip:[email protected];tag=as7de9e264
Call-ID: [email protected]
Max-Forwards: 70
Date: Wed, 29 Jan 2014 00:08:51 GMT
CSeq: 101 ACK
Content-Length: 0
<------------->
— (9 headers 0 lines) —
Really destroying SIP dialog ‘[email protected]’ Method: INVITE
CODEC issue?
Managed to fix the issue. The INVITE request had a required field called SDP-ANAT that was causing the issue.
Within the extension setup in FreePBX there’s a field called “allow”. Just needed to add “sdp-anat” to the field and everything now works as expected.