Hello Sam, Do you have any recommendation to overcome these NAT issues?
On 8/14/15, Sam Basan <[email protected]> wrote: > Hi, > > It's looks like you are having NAT problem. > Packets from the provider fail reaching your box. > > נשלח מטלפון נייד > בתאריך 14 באוג' 2015 15:56, "Daniel - Asterisk" <[email protected]> > כתב: > >> Hello friends: >> >> I am facing cutoffs randomly when negotiating calls. >> >> The PBX dials the destination, the provider (softswitch) receives the >> request *[1]* and sudenly the PBX hangs up the call* [2]* while the >> provider is still dialing it, as a consequence the remote peer receives a >> ghost call. Along the atempt I could see six times a messages regarding >> NAT >> isuues *[3]* >> >> I hope anyone can give me an idea to solve this issue. Softswitch is >> using >> an implementation of RFC 3264 and the PBX being used is Elastix 2.3 with >> Asterisk 1.8.11.0 >> >> Thanks in advance >> >> Elder D. Arohuanca >> Lima - Peru >> >> >> *[1]* >> [Aug 12 19:21:05] VERBOSE[17115] app_dial.c: -- Called >> SIP/SIP-PROVIDER/965034648 >> >> >> *[2]* >> [Aug 12 19:21:14] WARNING[3477] chan_sip.c: Retransmission timeout >> reached >> on transmission 0e51f669152c660b3c97de1876d9e971@*PROVIDER-IP* for seqno >> 103 (Critical Request) -- See >> https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions >> Packet timed out after 8832ms with no response >> [Aug 12 19:21:14] WARNING[3477] chan_sip.c: Hanging up call >> 0e51f669152c660b3c97de1876d9e971@*PROVIDER-IP* - no reply to our critical >> packet (see >> https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions >> ). >> [Aug 12 19:21:14] VERBOSE[17115] app_dial.c: == Everyone is >> busy/congested at this time (1:0/0/1) >> [Aug 12 19:21:14] VERBOSE[17115] pbx.c: -- Executing >> [s@macro-dialout-trunk:20] NoOp("SIP/143-000001d8", "Dial failed for some >> reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 111") in new stack >> [Aug 12 19:21:14] VERBOSE[17115] pbx.c: -- Executing >> [s@macro-dialout-trunk:21] Goto("SIP/143-000001d8", "s-CHANUNAVAIL,1") in >> new stack >> >> *[3]* >> Retransmitting #3 (no NAT) to PROVIDER-IP:5060: >> INVITE sip:dialed_number@PROVIDER-IP SIP/2.0 >> Via: SIP/2.0/UDP PBX-PUBLIC_IP:5060;branch=z9hG4bK06c2c701 >> Max-Forwards: 70 >> From: "PBX-DID" <sip:outbound-trunk@PROVIDER-IP>;tag=as27ef83ae >> To: <sip:dialed_number@PROVIDER-IP> >> Contact: <sip:outbound-trunk@PBX-PUBLIC_IP:5060> >> Call-ID: 6b9ad82d4673fdab722f9e53411a767d@PROVIDER-IP >> CSeq: 103 INVITE >> User-Agent: FPBX-2.8.1(1.8.11.0) >> Proxy-Authorization: Digest username="outbound-trunk", >> realm="SoftSwitch", >> algorithm=MD5, uri="sip:dialed_number@PROVIDER-IP", >> nonce="d1b5806808a0888112190722408572932332", >> response="40c94f3c04e87e3382c7652d1f012dc9" >> Date: Thu, 13 Aug 2015 00:56:40 GMT >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, >> PUBLISH >> Supported: replaces, timer >> Remote-Party-ID: "PBX-DID" <sip:PBX-DID@PROVIDER-IP >> >;party=calling;privacy=off;screen=no >> Content-Type: application/sdp >> Content-Length: 260 >> >> v=0 >> o=root 502733417 502733418 IN IP4 PBX-PUBLIC_IP >> s=Asterisk PBX 1.8.11.0 >> c=IN IP4 PBX-PUBLIC_IP >> t=0 0 >> m=audio 13042 RTP/AVP 18 101 >> a=rtpmap:18 G729/8000 >> a=fmtp:18 annexb=no >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> a=ptime:20 >> a=sendrecv >> >> >> -- >> _____________________________________________________________________ >> -- 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 >> > -- _____________________________________________________________________ -- 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
