Hi there
recently we change two asterisk 1.8 version to version 11.4.0 connected to a SBC acme packet to our internet employees softphone and teleworkers. Since the change, no call can be made to softphone. The SBC sends back a “513 Message Too Large” but, all softphones can make calls to company extensions (reverse path).
is this a bug or a bad configuration ?? any help will appreciated
tnks
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using SIP VIDEO TOS bits 136
== Using SIP VIDEO CoS mark 6
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Audio is at 12390
Video is at 192.168.20.10:13306
Adding codec 100010 (ilbc) to SDP
Adding codec 100002 (gsm) to SDP
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding video codec 200002 (h263) to SDP
Adding video codec 200003 (h263p) to SDP
Adding video codec 200001 (h261) to SDP
Adding video codec 200004 (h264) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 192.168.20.3:5060:
INVITE sip:99730-jsi8noipr627b@192.168.20.3:5060;rinstance=92F8DB5F;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.20.10:5060;branch=z9hG4bK148a4dc0;rport
Max-Forwards: 70
From: “Test” sip:730@192.168.20.10;tag=as439fc085
To: sip:99730-jsi8noipr627b@192.168.20.3:5060;rinstance=92F8DB5F;transport=udp
Contact: sip:730@192.168.20.10:5060
Call-ID: 60d7e37217f05be478657e8e711411a0@192.168.20.10:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.8.1(11.4.0)
Date: Fri, 21 Jun 2013 16:26:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Content-Type: application/sdp
Content-Length: 675
v=0
o=root 1704125188 1704125188 IN IP4 192.168.20.10
s=Asterisk PBX 11.4.0
c=IN IP4 192.168.20.10
b=CT:384
t=0 0
m=audio 12390 RTP/AVP 97 3 0 8 101
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:3 GSM/8000
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 13306 RTP/AVP 34 98 31 99
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: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:31 H261/90000
a=rtpmap:99 H264/90000
a=fmtp:99 redundant-pic-cap=0;parameter-add=0;packetization-mode=0;level-asymmetry-allowed=0
a=sendrecv
<— SIP read from UDP:192.168.20.3:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.20.10:5060;received=192.168.20.10;branch=z9hG4bK148a4dc0;rport=5060
From: “Test” sip:730@192.168.20.10;tag=as439fc085
To: sip:99730-jsi8noipr627b@192.168.20.3:5060;rinstance=92F8DB5F;transport=udp
Call-ID: 60d7e37217f05be478657e8e711411a0@192.168.20.10:5060
CSeq: 102 INVITE
<------------->
— (6 headers 0 lines) —
<— SIP read from UDP:192.168.20.3:5060 —>
SIP/2.0 513 Message Too Large
Via: SIP/2.0/UDP 192.168.20.10:5060;received=192.168.20.10;branch=z9hG4bK148a4dc0;rport=5060
From: “Test” sip:730@192.168.20.10;tag=as439fc085
To: sip:99730-jsi8noipr627b@192.168.20.3:5060;rinstance=92F8DB5F;transport=udp;tag=aprqngfrt-ta7p9h30000c6
Call-ID: 60d7e37217f05be478657e8e711411a0@192.168.20.10:5060
CSeq: 102 INVITE
Content-Length: 0
<------------->
— (7 headers 0 lines) —
– Got SIP response 513 “Message Too Large” back from 192.168.20.3:5060
Transmitting (NAT) to 192.168.20.3:5060:
ACK sip:99730-jsi8noipr627b@192.168.20.3:5060;rinstance=92F8DB5F;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.20.10:5060;branch=z9hG4bK148a4dc0;rport
Max-Forwards: 70
From: “Test” sip:730@192.168.20.10;tag=as439fc085
To: sip:99730-jsi8noipr627b@192.168.20.3:5060;rinstance=92F8DB5F;transport=udp;tag=aprqngfrt-ta7p9h30000c6
Contact: sip:730@192.168.20.10:5060
Call-ID: 60d7e37217f05be478657e8e711411a0@192.168.20.10:5060
CSeq: 102 ACK
User-Agent: FPBX-2.8.1(11.4.0)
Content-Length: 0