Hi Duane,<div><br></div><div>I looked at the pcap file you sent me yesterday and I see quite a few problems. I don't have your routing script and am not quite sure what IP address is what device but here are the things I can say you need to look into:</div>
<div><br></div><div>1- When the callee sends a 100 Trying, it is routed back to the callee. That doesn't make sense. Seems like a alias problem. </div><div><br></div><div>2- Then you receive another 100 Trying and 180 Ringing messages from 192.168.88.14 that is sent back to the callee. That is just weired.</div>
<div><br></div><div>3- Note that your clients should ONLY talk to their respective servers, if they are sending messages directly to the other server or clients, your request-route headers or aliases/domains in your opensips.cfg script is messed up. First double check your aliases and configured domains. OpenSIPS needs to know what IP addresses and domains are his.</div>
<div><br></div><div>4- In scenarios like this, I use <span style="background-color:rgb(255,255,255);font-family:Helvetica,Arial">record_route_preset() function to force the signalling path. You need to know what you are doing with that though.</span></div>
<div><font face="Helvetica, Arial"><br></font></div><div><font face="Helvetica, Arial">5- Have the signalling path set on a piece of paper from the original invite and make sure every message is routed the way it should be. By the time you get to Ack and Bye, things will work by itself. </font></div>
<div><font face="Helvetica, Arial"><br></font></div><div><font face="Helvetica, Arial">6- Never try to patch a fix or cut corners to make it work. Then it will keep breaking with different scenarios and will get harder and harder to fix. Make sure the original invites are properly set and everything should fall in place.</font></div>
<div><font face="Helvetica, Arial"><br></font></div><div><font face="Helvetica, Arial">7- Use the sip trace module to capture and that gives you better view of what opensips doing. tcpdump can only capture what's on the wire but sometimes messages loop inside that you can only see in the logs or with the sip trace module.</font></div>
<div><font face="Helvetica, Arial"><br></font></div><div><font face="Helvetica, Arial">Hope these helps. I suspect you have some alias/domain problems. Let us know what you find.</font></div><div><font face="Helvetica, Arial"><br>
</font></div><div><font face="Helvetica, Arial">Regards,</font></div><div><font face="Helvetica, Arial">Ali Pey</font></div><div><font face="Helvetica, Arial"><br></font><br><div class="gmail_quote">On Mon, Jul 9, 2012 at 4:45 AM, Vlad Paiu <span dir="ltr"><<a href="mailto:vladpaiu@opensips.org" target="_blank">vladpaiu@opensips.org</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><u></u>
<div bgcolor="#ffffff" text="#000000">
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> ACK
sip:50.xx.xx.156;lr;ftag=d4xut7i3jx;nat=yes;vst=AAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAA;did=0f9.1ddb82a6
SIP/2.0.</div>
<div> Record-Route:
<sip:99.xx.xx.161;r2=on;lr>.</div>
<div> Record-Route:
<sip:192.168.88.1;r2=on;lr>.</div>
<br>
<br>
Regards,<br>
<pre cols="72">Vlad Paiu
OpenSIPS Developer
<a href="http://www.opensips-solutions.com" target="_blank">http://www.opensips-solutions.com</a> </pre><div><div class="h5">
<br>
On 07/09/2012 07:09 AM, <a href="mailto:duane.larson@gmail.com" target="_blank">duane.larson@gmail.com</a> wrote:
<blockquote 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 href="mailto:duane.larson@gmail.com" target="_blank">duane.larson@gmail.com</a> wrote:
<br>
> I think I have multiple issues going on but I might be
getting closer to the issue.
<br>
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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 href="http://www.tech-invite.com/Ti-sip-dialog.html#inv" target="_blank">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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> INVITE <a href="mailto:sip:9013XX3XX6@192.168.88.14:3072;line=9zx0whnm" target="_blank">sip:9013XX3XX6@192.168.88.14:3072;line=9zx0whnm</a>
SIP/2.0
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> Record-Route:
4aoni525hc;nat=yes;vst=AAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAA;did=598.b8b26331><br>
> <br>
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> Record-Route: <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 href="mailto:duane.larson@gmail.com" target="_blank">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 href="mailto:alipey@gmail.com" target="_blank">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 href="mailto:duane.larson@gmail.com" target="_blank">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 href="http://opensips-open-sip-server.1449251.n2.nabble.com/Two-OpenSIPS-proxies-issue-tp7580685p7580743.html" target="_blank">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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<pre><fieldset></fieldset>
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</pre>
</blockquote>
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<br></blockquote></div><br></div>