<p dir="ltr">Hi, <br>
You must have two thing for start:<br>
1. Set your FW to allow sip port (by default 5060) to your asterisk IP address.<br>
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)</p>
<p dir="ltr">Google for asterisk NAT configuration parameters.<br><br></p>
<p dir="ltr">נשלח מטלפון נייד</p>
<div class="gmail_quote">בתאריך 14 באוג' 2015 22:12, "Daniel - Asterisk" <<a href="mailto:earohuanca@gmail.com">earohuanca@gmail.com</a>> כתב:<br type="attribution"><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Hello Sam,<br>
<br>
Do you have any recommendation to overcome these NAT issues?<br>
<br>
On 8/14/15, Sam Basan <<a href="mailto:sbasan@bluebe.net">sbasan@bluebe.net</a>> wrote:<br>
> Hi,<br>
><br>
> It's looks like you are having NAT problem.<br>
> Packets from the provider fail reaching your box.<br>
><br>
> נשלח מטלפון נייד<br>
> בתאריך 14 באוג' 2015 15:56, "Daniel - Asterisk" <<a href="mailto:earohuanca@gmail.com">earohuanca@gmail.com</a>><br>
> כתב:<br>
><br>
>> Hello friends:<br>
>><br>
>> I am facing cutoffs randomly when negotiating calls.<br>
>><br>
>> The PBX dials the destination, the provider (softswitch) receives the<br>
>> request *[1]* and sudenly the PBX hangs up the call* [2]* while the<br>
>> provider is still dialing it, as a consequence the remote peer receives a<br>
>> ghost call. Along the atempt I could see six times a messages regarding<br>
>> NAT<br>
>> isuues *[3]*<br>
>><br>
>> I hope anyone can give me an idea to solve this issue. Softswitch is<br>
>> using<br>
>> an implementation of RFC 3264 and the PBX being used is Elastix 2.3 with<br>
>> Asterisk 1.8.11.0<br>
>><br>
>> Thanks in advance<br>
>><br>
>> Elder D. Arohuanca<br>
>> Lima - Peru<br>
>><br>
>><br>
>> *[1]*<br>
>> [Aug 12 19:21:05] VERBOSE[17115] app_dial.c: -- Called<br>
>> SIP/SIP-PROVIDER/965034648<br>
>><br>
>><br>
>> *[2]*<br>
>> [Aug 12 19:21:14] WARNING[3477] chan_sip.c: Retransmission timeout<br>
>> reached<br>
>> on transmission 0e51f669152c660b3c97de1876d9e971@*PROVIDER-IP* for seqno<br>
>> 103 (Critical Request) -- See<br>
>> <a href="https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions" rel="noreferrer" target="_blank">https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions</a><br>
>> Packet timed out after 8832ms with no response<br>
>> [Aug 12 19:21:14] WARNING[3477] chan_sip.c: Hanging up call<br>
>> 0e51f669152c660b3c97de1876d9e971@*PROVIDER-IP* - no reply to our critical<br>
>> packet (see<br>
>> <a href="https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions" rel="noreferrer" target="_blank">https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions</a><br>
>> ).<br>
>> [Aug 12 19:21:14] VERBOSE[17115] app_dial.c: == Everyone is<br>
>> busy/congested at this time (1:0/0/1)<br>
>> [Aug 12 19:21:14] VERBOSE[17115] pbx.c: -- Executing<br>
>> [s@macro-dialout-trunk:20] NoOp("SIP/143-000001d8", "Dial failed for some<br>
>> reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 111") in new stack<br>
>> [Aug 12 19:21:14] VERBOSE[17115] pbx.c: -- Executing<br>
>> [s@macro-dialout-trunk:21] Goto("SIP/143-000001d8", "s-CHANUNAVAIL,1") in<br>
>> new stack<br>
>><br>
>> *[3]*<br>
>> Retransmitting #3 (no NAT) to PROVIDER-IP:5060:<br>
>> INVITE sip:dialed_number@PROVIDER-IP SIP/2.0<br>
>> Via: SIP/2.0/UDP PBX-PUBLIC_IP:5060;branch=z9hG4bK06c2c701<br>
>> Max-Forwards: 70<br>
>> From: "PBX-DID" <sip:outbound-trunk@PROVIDER-IP>;tag=as27ef83ae<br>
>> To: <sip:dialed_number@PROVIDER-IP><br>
>> Contact: <sip:outbound-trunk@PBX-PUBLIC_IP:5060><br>
>> Call-ID: 6b9ad82d4673fdab722f9e53411a767d@PROVIDER-IP<br>
>> CSeq: 103 INVITE<br>
>> User-Agent: FPBX-2.8.1(1.8.11.0)<br>
>> Proxy-Authorization: Digest username="outbound-trunk",<br>
>> realm="SoftSwitch",<br>
>> algorithm=MD5, uri="sip:dialed_number@PROVIDER-IP",<br>
>> nonce="d1b5806808a0888112190722408572932332",<br>
>> response="40c94f3c04e87e3382c7652d1f012dc9"<br>
>> Date: Thu, 13 Aug 2015 00:56:40 GMT<br>
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,<br>
>> PUBLISH<br>
>> Supported: replaces, timer<br>
>> Remote-Party-ID: "PBX-DID" <sip:PBX-DID@PROVIDER-IP<br>
>> >;party=calling;privacy=off;screen=no<br>
>> Content-Type: application/sdp<br>
>> Content-Length: 260<br>
>><br>
>> v=0<br>
>> o=root 502733417 502733418 IN IP4 PBX-PUBLIC_IP<br>
>> s=Asterisk PBX 1.8.11.0<br>
>> c=IN IP4 PBX-PUBLIC_IP<br>
>> t=0 0<br>
>> m=audio 13042 RTP/AVP 18 101<br>
>> a=rtpmap:18 G729/8000<br>
>> a=fmtp:18 annexb=no<br>
>> a=rtpmap:101 telephone-event/8000<br>
>> a=fmtp:101 0-16<br>
>> a=ptime:20<br>
>> a=sendrecv<br>
>><br>
>><br>
>> --<br>
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