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

Sat Aug 15 11:12:11 CDT 2015

Hi,
You must have two thing for start:
1. Set your FW to allow sip port (by default 5060) to your asterisk IP
address.
2. Set your asterisk configuration with the external public IP and your
local subnet address (so asterisk will put his public address for outside
the networks calls)

Google for asterisk NAT configuration parameters.

נשלח מטלפון נייד
בתאריך 14 באוג' 2015 22:12,‏ "Daniel - Asterisk" <earohuanca at gmail.com> כתב:

> Hello Sam,
>
> Do you have any recommendation to overcome these NAT issues?
>
> On 8/14/15, Sam Basan <sbasan at bluebe.net> 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" <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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> >>
> >
>
> --
> _____________________________________________________________________
> -- 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
>
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