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Hello,<br>
<br>
This is quite strange, can you please also post a full OpenSIPS
debug for the call where that ACK got relayed out like<br>
<br>
<div class="de2"> ACK
sip:50.xx.xx.156;lr;ftag=d4xut7i3jx;nat=yes;vst=AAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAA;did=0f9.1ddb82a6
SIP/2.0.</div>
<div class="de1"> Record-Route:
<sip:99.xx.xx.161;r2=on;lr>.</div>
<div class="de2"> Record-Route:
<sip:192.168.88.1;r2=on;lr>.</div>
<br>
<br>
Regards,<br>
<pre class="moz-signature" cols="72">Vlad Paiu
OpenSIPS Developer
<a class="moz-txt-link-freetext" href="http://www.opensips-solutions.com">http://www.opensips-solutions.com</a> </pre>
<br>
On 07/09/2012 07:09 AM, <a class="moz-txt-link-abbreviated" href="mailto:duane.larson@gmail.com">duane.larson@gmail.com</a> wrote:
<blockquote cite="mid:20cf30563f436675ea04c45dc9d5@google.com"
type="cite">I just got my calls working by removing the
Record-Route's and then reinserting then in an order that would
according to my topology.
<br>
<br>
I will need to go back and start from scratch to see if a lot of
the other stuff I did was really needed or not and then update but
here is were I edited the Record-Routes
<br>
<br>
When the INVITE is coming from my OpenSIPS/Proxy to the Callee I
did
<br>
<br>
if ( is_method("INVITE") ) {
<br>
remove_hf("Record-Route");
<br>
insert_hf("Record-Route: $(hdr(Record-Route)[2])\r\n", "Via");
<br>
insert_hf("Record-Route: $(hdr(Record-Route)[1])\r\n", "Via");
<br>
insert_hf("Record-Route: $(hdr(Record-Route)[0])\r\n", "Via");
<br>
}
<br>
<br>
Then when the 180 and 200 are coming from the Callee to the Caller
before the 180 and 200 go to the Caller I did the following
<br>
<br>
<br>
if (t_check_status("180")){
<br>
remove_hf("Record-Route");
<br>
insert_hf("Record-Route: $(hdr(Record-Route)[2])\r\n", "Via");
<br>
insert_hf("Record-Route: $(hdr(Record-Route)[1])\r\n", "Via");
<br>
insert_hf("Record-Route: $(hdr(Record-Route)[0])\r\n", "Via");
<br>
<br>
}
<br>
<br>
<br>
if (t_check_status("200")){
<br>
remove_hf("Record-Route");
<br>
insert_hf("Record-Route: $(hdr(Record-Route)[2])\r\n", "Via");
<br>
insert_hf("Record-Route: $(hdr(Record-Route)[1])\r\n", "Via");
<br>
insert_hf("Record-Route: $(hdr(Record-Route)[0])\r\n", "Via");
<br>
<br>
}
<br>
<br>
<br>
So not sure if there is something wrong with the way OpenSIPS
places the Record-Route ordering when OpenSIPS has multiple
interfaces. I am not 100% sure if what I have done here is right
or not but calls are working now.
<br>
<br>
Any feedback?
<br>
<br>
<br>
On , <a class="moz-txt-link-abbreviated" href="mailto:duane.larson@gmail.com">duane.larson@gmail.com</a> wrote:
<br>
> I think I have multiple issues going on but I might be
getting closer to the issue.
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> <br>
> I am wondering if this might be part of the issue.
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> <br>
> If you look at the the following,
<a class="moz-txt-link-freetext" href="http://www.tech-invite.com/Ti-sip-dialog.html#inv">http://www.tech-invite.com/Ti-sip-dialog.html#inv</a> , for the first
INVITE message that the Callee receives the first Proxy that the
callee needs to take in its Record-Route is first in the list of
Record-Routes on the INVITE message. As for the Caller the
Record-Route set gets flipped (whatever Record-Route is on the top
will be its last route hop). So if this is the case then why is
the OpenSIPS/SBC device sending my Callee device an INVITE message
with the far end proxy, OpenSIPS/Proxy, on the top of the
Record-Route list? Here is the INVITE that my callee is getting
<br>
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> <br>
> INVITE <a class="moz-txt-link-abbreviated" href="mailto:sip:9013XX3XX6@192.168.88.14:3072;line=9zx0whnm">sip:9013XX3XX6@192.168.88.14:3072;line=9zx0whnm</a>
SIP/2.0
<br>
> <br>
> <br>
> Record-Route:
4aoni525hc;nat=yes;vst=AAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAA;did=598.b8b26331><br>
> <br>
> <br>
> Record-Route: <br>
> <br>
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> Record-Route: <br>
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> <br>
> The Record-Route with 50.XX.XX.156 should be at the bottom of
the list I think because that is the OpenSIPS/Proxy that is on the
Internet. Am I wrong on this? On the SIP trace I posted on
pastebin this INVITE to the Callee starts on line 299.
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> On , <a class="moz-txt-link-abbreviated" href="mailto:duane.larson@gmail.com">duane.larson@gmail.com</a> wrote:
<br>
> <br>
> <br>
> > I'm really not sure if I am just duck taping the issue
but I was able to make most of the call work. The only problem now
is when the Callee hangs up the BYE is sent directly to the
OpenSIPS/Proxy IP instead of going to the OpenSIPS/SBC. This will
not work due to firewall issues.
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> > My ACKs are no longer not showing up as Non-Loose Route
messages, but the BYEs are.
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> > So if the Caller hangs up the Callee sees the BYE
message (GOOD!), but if the Callee hangs up the Caller never sees
the BYE message (Bad).
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> > I will send a PCAP trace to Ali directly.
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> > On , Ali Pey <a class="moz-txt-link-abbreviated" href="mailto:alipey@gmail.com">alipey@gmail.com</a>> wrote:
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> > > Duane,
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> > > The Ack should not have any request-route headers.
Only Route headers. If you see request-route headers, then you
need to find how they got there and fix that first.
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> > > I believe it is ok if the Ack doesn't go through
loose route, in that case it should be sent to the request-uri
destination ip and that IP should be your client IP.
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> <br>
> > > Let me know if this help. If not, can you attach
here a wireshark trace and I will go through your signalling for
you. Going thought a text trace can be quit time consuming. In
wireshark it's a lot easier to jump from a message to another
through the call flow. You can use tcpdump to capture to .cap file
for wireshark.
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> > > Regards,
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> > > Ali Pey
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> > > On Sat, Jul 7, 2012 at 3:35 PM, osiris123d
<a class="moz-txt-link-abbreviated" href="mailto:duane.larson@gmail.com">duane.larson@gmail.com</a>> wrote:
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> > > This is driving me crazy. I was right the first
time when I said that one of
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> > > the ACKs was not showing up as a loose route. It
is the third ACK that is
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> > > coming from the OpenSIPS/Proxy. When it reaches
the OpenSIPS/SBC device the
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> > > ACK fails as a loose route.
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> > > It would make sense that this would not be a loose
route because there are
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> > > no Route headers so the loose_route() function
would return FALSE.
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> > > The issue still remains that when the ACK reaches
the OpenSIPS/SBC it still
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> > > isn't routed to the Callee, instead it is looped
and routed to the same
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> > > interface it came from because that is whats in the
RURI.
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> > > View this message in context:
<a class="moz-txt-link-freetext" href="http://opensips-open-sip-server.1449251.n2.nabble.com/Two-OpenSIPS-proxies-issue-tp7580685p7580743.html">http://opensips-open-sip-server.1449251.n2.nabble.com/Two-OpenSIPS-proxies-issue-tp7580685p7580743.html</a><br>
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> > > Sent from the OpenSIPS - Users mailing list archive
at Nabble.com.
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> > > _______________________________________________
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> > > Users mailing list
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> > > <a class="moz-txt-link-abbreviated" href="mailto:Users@lists.opensips.org">Users@lists.opensips.org</a>
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_______________________________________________
Users mailing list
<a class="moz-txt-link-abbreviated" href="mailto:Users@lists.opensips.org">Users@lists.opensips.org</a>
<a class="moz-txt-link-freetext" href="http://lists.opensips.org/cgi-bin/mailman/listinfo/users">http://lists.opensips.org/cgi-bin/mailman/listinfo/users</a>
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