[asterisk-users] chan_sip.c: Retransmission timeout reached on transmission

[Sam Basan]

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" <earohuanca at gmail.com> 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 at 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 at 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 at 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 at PROVIDER-IP>;tag=as27ef83ae
> To: <sip:dialed_number at PROVIDER-IP>
> Contact: <sip:outbound-trunk at PBX-PUBLIC_IP:5060>
> Call-ID: 6b9ad82d4673fdab722f9e53411a767d at 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 at 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 at 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
>
>
> --
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