Incoming calls terminated after 15secs. ACK not sent.

neurodot · Brekeke Member Joined: 21 Mar 2012 Posts: 15

1. Brekeke Product Name and version:
Brekeke PBX, Version 2.4.9.0 , Basic
2. Java version:
Java6 Update31
3. OS type and the version:
Windows Server 2008 R2 SP1 Enterprise
4. UA (phone), gateway or other hardware/software involved:
lync, x-lite, pstn
5. Select your network pattern from http://www.brekeke-sip.com/bbs/network/networkpatterns.html :
Patern 9
6. Your problem:

Every call which is routed through ARS (from PSTN to SIP - incoming call) has following behavior:
1) call is terminated after 15 secs (approx)
2) i can hear only voice only in one direction

I believe that #1 and #2 are caused by same misconfiguration.
According to Wireshark, PSTN is requesting ACK, but Brekeke is not responding. RTP packets seems to be sent correctly to both directions.

In the Dial plan I have:
$request=^INVITE
From=sip:(.+)@188.175.113.182
To=sip:(.+)@188.175.113.182
-----------
To=sip:%2@127.0.0.1:15060
$auth=false
$b2bua=false
$transport=udp

ARS rules are:
IN:
From: sip:(.+)@188.175.113.182
To: sip:312316600@127.0.0.1
-------------
To: 1000

OUT:
To: sip:(.+)@voip.neurodot-consulting.com
-----
nothing

If I make a call from registered UA using dial plan only, everything is fine.

Any suggestions?

Many thanks.

hope · Brekeke Master Guru Joined: 15 Jan 2008 Posts: 862

how about add following line in dial plan rule deploy pattern for the call

&net.sip.fixed.addrport.uac = true

neurodot · Brekeke Member Joined: 21 Mar 2012 Posts: 15

neurodot · Brekeke Member Joined: 21 Mar 2012 Posts: 15

I can see this in the Brekeke log. It seems strange to me, because of sending ACK to sip:312316600@127.0.0.1:15060 .... at localhost??? Is this correct?

(To better understand. PSTN UA with number 603780198 is calling to 312316600. )
==============================================

session.20: pkt=7 dp=1 st=0 sip:603780198@188.175.113.182(188.175.113.182:5060) --> sip:312316600@localhost:15060(127.0.0.1:15060)
send="ACK sip:312316600@127.0.0.1:15060 SIP/2.0"

session.20: processtime=3

session.20: send: to=UAS:127.0.0.1:15060(UDP) at 03/29/12 20:42:29.208
==============================================
ACK sip:312316600@127.0.0.1:15060 SIP/2.0
Call-ID: afde3591-669c4942-7733e98d-800c04fb@188.175.113.182
CSeq: 102 ACK
From: "603780198" <sip:603780198@188.175.113.182>;tag=a06c5ba07c31a09c24a0bc1f
To: <sip:312316600@127.0.0.1:5060>;tag=b2f30b2b3p
Via: SIP/2.0/UDP 127.0.0.1:5060;rport;branch=z9hG4bK874bbf691b4bbf1b6834b-67332436-c84497b9
Via: SIP/2.0/UDP 188.175.113.182;branch=z9hG4bK-4f7591584f74babe3e6
Record-Route: <sip:127.0.0.1:5060;lr>
Content-Length: 0

Please advice. Thank you.

neurodot · Brekeke Member Joined: 21 Mar 2012 Posts: 15

I have just found one interesting thing ... if push some number on the phone (from which I can't hear a thing) while call is established (within those 15secs), I can hear that tone in caller's phone. Does this mean anything to my issue?
Maybe it is just a proof that audio packets are flowing fine and problem is really ACK message not delivered to somewhere?

Please suggest. Im trying to make this work for long days now.

hope · Brekeke Master Guru Joined: 15 Jan 2008 Posts: 862

what status shown in sip server side active session page about call?
is it talking or accepted or anything else?

if capture packets at callee side, can you see ACK sent to callee?

add both following lines to the dial plan rules "deploy patterns" applied to the call
&net.sip.fixed.addrport.uas = true
&net.sip.fixed.addrport.uac = true

neurodot · Brekeke Member Joined: 21 Mar 2012 Posts: 15

voipwell.com · Posted: Tue Apr 03, 2012 10:23 am Post subject:

You don't need two patterns in ARS for a call going from an ip phone thru the pbx to a provider. You only need an ARS OUT pattern.

1st leg of call
ip phone to pbx will be received by "to pbx" sip dial plan and passed to PBX.

2nd leg of call

PBX will check for matching ARS pattern to route the call. All is needed is an ARS OUT pattern. PBX sends the processed ARS route to SIP Server "from pbx" dialplan which sends it out to provider.

You need an ARS IN pattern to be able to route an incoming call to the pbx from a provider.

davi · Brekeke Addict Joined: 26 Jan 2011 Posts: 34

Set "net.sip.fixed.addrport.uas = true" in the SIP Server's [Config] -> [Advanced] page, and reboot the SIP server.

With this setting, the ACK packet must be sent to the callee (UAS).

neurodot · Brekeke Member Joined: 21 Mar 2012 Posts: 15

Thanks for the tips. ACK is created now without problems.

Still Im not able to get ARS work only with OUT pattern. I simply don't know what else can be possibly wrong. If I have only OUT pattern (and no IN pattern), the call won't get connected (lync client gets busy error). If I only add one IN pattern with the very same settings as existing OUT pattern has, it starts working.

My OUT pattern is very simple:
Matching: From=sip:(.+)_lync@neurodot-consulting.com
nothing more, all other fields in ARS are empty or default values.

So, as I wrote at the begining, when I add IN pattern which has also From=sip:(.+)_lync@neurodot-consulting.com, it starts working.

I believe you guys and documentation says the same, but OUT pattern alone is not working for me. So what else can possibly affect this behavior?

Many thanks for helping.

hope · Brekeke Master Guru Joined: 15 Jan 2008 Posts: 862

from wiki page below
http://wiki.brekeke.com/wiki/Security
From PBX v2.2.7.7 and later, the calls will be rejected by PBX when no ARS rule can be applied to the calls from/to non-registered users.
The ARS pattern-in is needed, because caller(lync user) is not pbx user and not registered at pbx/sip server

neurodot · Brekeke Member Joined: 21 Mar 2012 Posts: 15