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 >
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