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.
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
When the INVITE is coming from my OpenSIPS/Proxy to the Callee I did
if ( is_method("INVITE") ) {
remove_hf("Record-Route");
insert_hf("Record-Route: $(hdr(Record-Route)[2])\r\n", "Via");
insert_hf("Record-Route: $(hdr(Record-Route)[1])\r\n", "Via");
insert_hf("Record-Route: $(hdr(Record-Route)[0])\r\n", "Via");
}
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
if (t_check_status("180")){
remove_hf("Record-Route");
insert_hf("Record-Route: $(hdr(Record-Route)[2])\r\n", "Via");
insert_hf("Record-Route: $(hdr(Record-Route)[1])\r\n", "Via");
insert_hf("Record-Route: $(hdr(Record-Route)[0])\r\n", "Via");
}
if (t_check_status("200")){
remove_hf("Record-Route");
insert_hf("Record-Route: $(hdr(Record-Route)[2])\r\n", "Via");
insert_hf("Record-Route: $(hdr(Record-Route)[1])\r\n", "Via");
insert_hf("Record-Route: $(hdr(Record-Route)[0])\r\n", "Via");
}
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.
Any feedback?
On , [email protected] wrote:
> I think I have multiple issues going on but I might be getting
closer to the issue.
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> I am wondering if this might be part of the issue.
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> If you look at the the following,
http://www.tech-invite.com/Ti-sip-dialog.html#inv , 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
>
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> INVITE sip:[email protected]:3072;line=9zx0whnm SIP/2.0
>
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> Record-Route:
4aoni525hc;nat=yes;vst=AAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAA;did=598.b8b26331>
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> Record-Route:
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> Record-Route:
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> 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 , [email protected] wrote:
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> > 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 [email protected]> 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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> > > 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
[email protected]> 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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> > > --
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> > > View this message in context:
http://opensips-open-sip-server.1449251.n2.nabble.com/Two-OpenSIPS-proxies-issue-tp7580685p7580743.html
>
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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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> > >
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> > > [email protected]
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> > >
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> > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
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> > >
_______________________________________________
Users mailing list
[email protected]
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
