Re: [asterisk-users] SIP calls dropping at 15 minutes
On 11/21/15 3:10 PM, Steve Edwards wrote: On 11/20/15 11:13 AM, Steve Edwards wrote: 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. 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? On Sat, 21 Nov 2015, Andres wrote: Looks like session timers are kicking in and a Re-Invite is being sent. I would disable them in sip.conf and try again: session-timers=refuse http://doxygen.asterisk.org/trunk/sip_session_timers.html 3) Should OpenSIPS be responding differently to the INVITE at 15 minutes? This appears to work, but it feels wrong. Shouldn't I be configuring Asterisk or OpenSIPS to respond or receive the re-invite correctly? Maybe but I would not lose sleep over having session timers disabled if it fixes your problem. -- Technical Support http://www.cellroute.net -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP calls dropping at 15 minutes
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 wrote: > On 11/20/15 11:13 AM, Steve Edwards wrote: >> > > 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. >>> >> > 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? >>> >> > On Sat, 21 Nov 2015, Andres wrote: > > Looks like session timers are kicking in and a Re-Invite is being sent. I >> would disable them in sip.conf and try again: >> >> session-timers=refuse >> >> http://doxygen.asterisk.org/trunk/sip_session_timers.html >> > > 3) Should OpenSIPS be responding differently to the INVITE at 15 minutes? >>> >> > This appears to work, but it feels wrong. Shouldn't I be configuring > Asterisk or OpenSIPS to respond or receive the re-invite correctly? > > -- > Thanks in advance, > - > Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP calls dropping at 15 minutes
On 11/20/15 11:13 AM, Steve Edwards wrote: 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. 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? On Sat, 21 Nov 2015, Andres wrote: Looks like session timers are kicking in and a Re-Invite is being sent. I would disable them in sip.conf and try again: session-timers=refuse http://doxygen.asterisk.org/trunk/sip_session_timers.html 3) Should OpenSIPS be responding differently to the INVITE at 15 minutes? This appears to work, but it feels wrong. Shouldn't I be configuring Asterisk or OpenSIPS to respond or receive the re-invite correctly? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP calls dropping at 15 minutes
On 11/20/15 11:13 AM, Steve Edwards wrote:
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:
"760xxx"
| |(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: "760xxx"
| |(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
| |
[asterisk-users] SIP calls dropping at 15 minutes
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: "760xxx"
| |(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:
"760xxx" (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|
