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

[Daniel - Asterisk]

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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>