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[asterisk.org at sedwar...]

I have a problem where SIP calls from some providers are dropping at 15
minutes.

The topology is: Client sends calls to a host running OpenSIPS, OpenSIPS
sends calls to an Asterisk server.

Below,

'Client' is the IP address of the client's host (running
FPBX-2.8.1(1.8.20.0)

'OpenSIPS' is the IP address of my host running OpenSIPS 1.7.2-tls

'Asterisk' is the IP address of my host running Asterisk 11.17.1.

The relevant snippet of opensips.cfg is:

# 317
if ($rU =~ '317*')
{
ds_select_dst(
'02' # set-id (in dispatcher.list)
, '4' # algorithm (4 = round-robin)
);
forward();
return;
}

where set-id 02 is 'sip:Asterisk:5061'

The 'Flow' diagram from Wireshark from a tcpdump on the OpenSIPS host
follows, hopefully the email clients will not mung it too much.

|Time | Client | Asterisk |
| | | OpenSIPS |
|7.158764 | INVITE SDP (g711U g7 | |SIP From: "760xxxxxxx" <sip:760xxxxxxx@client To:<sip:317xxxxxxx@OpenSIPS
| |(5060) ------------------> (5060) | |
|7.159003 | | INVITE SDP (g711U g7 |SIP Request
| | |(5060) ------------------> (5061) |
|7.161857 | | 100 Trying| |SIP Status
| | |(5060) <------------------ (5061) |
|7.161958 | 100 Trying| | |SIP Status
| |(5060) <------------------ (5060) | |
|7.538268 | | 200 OK SDP (g711U te |SIP Status
| | |(5060) <------------------ (5061) |
|7.538411 | 200 OK SDP (g711U te | |SIP Status
| |(5060) <------------------ (5060) | |
|7.585703 | ACK | | |SIP Request
| |(5060) ------------------> (5060) | |
|7.585941 | | ACK | |SIP Request
| | |(5060) ------------------> (5061) |
|7.586548 | INVITE SDP (g711U te | |SIP From: "760xxxxxxx" <sip:760xxxxxxx@client To:<sip:317xxxxxxx@OpenSIPS
| |(5060) ------------------> (5060) | |
|7.586726 | | INVITE SDP (g711U te |SIP Request
| | |(5060) ------------------> (5061) |
|7.587792 | | 100 Trying| |SIP Status
| | |(5060) <------------------ (5061) |
|7.587922 | 100 Trying| | |SIP Status
| |(5060) <------------------ (5060) | |
|7.588003 | | 200 OK SDP (g711U te |SIP Status
| | |(5060) <------------------ (5061) |
|7.588081 | 200 OK SDP (g711U te | |SIP Status
| |(5060) <------------------ (5060) | |
|7.635401 | ACK | | |SIP Request
| |(5060) ------------------> (5060) | |
|7.635674 | | ACK | |SIP Request
| | |(5060) ------------------> (5061) |
|907.588019| | INVITE SDP (g711U te |SIP Request
| | |(5060) <------------------ (5061) |
|907.590138| | 100 Giving a try |SIP Status
| | |(5060) ------------------> (5061) |
|907.590261| | INVITE SDP (g711U te |SIP Request
| | |(5060) ------------------> (5061) |
|907.591294| | 481 Call/Transaction |SIP Status
| | |(5060) <------------------ (5061) |
|907.591420| | ACK | |SIP Request
| | |(5060) ------------------> (5061) |
|907.591467| | 481 Call/Transaction |SIP Status
| | |(5060) ------------------> (5061) |
|907.592140| | ACK | |SIP Request
| | |(5060) <------------------ (5061) |
|907.867923| | BYE | |SIP Request
| | |(5060) <------------------ (5061) |
|907.868231| | BYE | |SIP Request
| | |(5060) ------------------> (5061) |
|907.869337| | 481 Call leg/transac |SIP Status
| | |(5060) <------------------ (5061) |
|907.869412| | 481 Call leg/transac |SIP Status
| | |(5060) ------------------> (5061) |
|1140.290782| INVITE SDP (g711U te | |SIP From: "760xxxxxxx" <sip:760xxxxxxx@client To:<sip:317xxxxxxx@OpenSIPS
| |(5060) ------------------> (5060) | |
|1140.291032| | INVITE SDP (g711U te |SIP Request
| | |(5060) ------------------> (5061) |
|1140.292338| | 481 Call/Transaction |SIP Status
| | |(5060) <------------------ (5061) |
|1140.292445| 481 Call/Transaction | |SIP Status
| |(5060) <------------------ (5060) | |
|1140.339890| ACK | | |SIP Request
| |(5060) ------------------> (5060) | |
|1140.340011| | ACK | |SIP Request
| | |(5060) ------------------> (5061) |
|1140.452758| BYE | | |SIP Request
| |(5060) ------------------> (5060) | |
|1140.452893| | BYE | |SIP Request
| | |(5060) ------------------> (5061) |
|1140.453470| | 481 Call leg/transac |SIP Status
| | |(5060) <------------------ (5061) |
|1140.453541| 481 Call leg/transac | |SIP Status
| |(5060) <------------------ (5060) | |

My knowledge of SIP is limited, but it appears that Asterisk is sending an
INVITE at 907.588019, OpenSIPS responds with an INVITE at 907.590261, but
Asterisk thinks the call doesn't exist and sends a BYE.

1) Is a 'ds_select_dst()' followed by a 'forward()' the right way to route
calls in OpenSIPS? It works most of the time.

2) Can (or should) I configure Asterisk to not send the INVITE at 15
minutes?

3) Should OpenSIPS be responding differently to the INVITE at 15 minutes?

--
Thanks in advance,
-------------------------------------------------------------------------
Steve Edwards sedwards@sedwards.com Voice: +1-760-468-3867 PST

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[andres at telesip.net]

On 11/20/15 11:13 AM, Steve Edwards wrote:

[asterisk.org at sedwar...]

[bc at iptel.co]

probably opensips isn't forwarding the re-invite to the endpoint.. set re-invites up and run sip tracing on your opensips and asterisk box and see what happens when the reinvites arrive.

On Sat, Nov 21, 2015 at 8:10 PM, Steve Edwards <asterisk.org@sedwards.com (asterisk.org@sedwards.com)> wrote:

[andres at telesip.net]

On 11/21/15 3:10 PM, Steve Edwards wrote: