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[earohuanca at gmail.com]

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

[sbasan at bluebe.net]

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@gmail.com (earohuanca@gmail.com)> כתב:

[earohuanca at gmail.com]

Hello Sam,

Do you have any recommendation to overcome these NAT issues?

On 8/14/15, Sam Basan <sbasan@bluebe.net> wrote:

[sbasan at bluebe.net]

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@gmail.com (earohuanca@gmail.com)> כתב: