Re: [OpenSIPS-Users] 404 Not Here
Hi Brad,
Thinks are a bit more complicated, it seems
In the INVITE your opensips sends to 64.93 IP, you have the Contact
with 192.168.1.21 (priv IP of asterisk).
When you receive the BYE from 64.93 IP, the Route hdrs are ok (the 2
hdrs added by opensips to reflect the interface exchange), but the RURI
is wrong - it must be the contact from the INVITE you sent, but it seems
to be the IP of your opensips - this makes opensips to do act as strict
router and not like a loose routerand routing gets broken.
So, the 64.93 party or some other behind it, screw up the Contact in
the your INVITE and this alters the in-dialog requests - you should
check with the upstream guys.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 02/25/2013 04:36 PM, brad smith wrote:
I just tested an outbound call (Asterisk originate) without bridging
and get the same '404 not here' if that helps.
Thanks again,
Brad
On Mon, Feb 25, 2013 at 8:01 AM, Vlad Paiu vladp...@opensips.org
mailto:vladp...@opensips.org wrote:
Hello,
Seems the incoming BYE does not have any Route headers, and the
loose_route() function returns false.
Since you have dialog support in your script, try
if (has_totag()) {
# sequential request withing a dialog should
# take the path determined by record-routing
if (loose_route() || match_dialog()) {
This way you will force matching of dialog sequential requests
that have no Route headers.
Best Regards,
Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com
On 02/24/2013 02:57 AM, brad smith wrote:
Hello,
I am currently running opensips 1.8.1 no tls. It is
multi-homed with a public and private address.
I have a asterisk
1.8.19 in the lan that is connected to opensips via lan
address.
*issue*
A caller calls in
and then I place an outbound call and finally bridge the two
calls.
This works as
expected, except when the outbound caller hangs up first the
BYE never gets back to Asterisk.
I can see the BYE
reach OpenSips but a '404 not here' is returned to the ISP.
sip trace https://gist.github.com/5009662
opensips.cfg https://gist.github.com/5009704
thanks for your time.
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Re: [OpenSIPS-Users] 404 Not Here
Bogdan,
Thanks for responding.
I am using vitelity for my upstream; I will send them a ticket. If they
fail to act, do you have any suggestions...switch carriers? any
config change?
Thanks again,
Brad
On Tue, Feb 26, 2013 at 7:58 AM, Bogdan-Andrei Iancu bog...@opensips.orgwrote:
**
Hi Brad,
Thinks are a bit more complicated, it seems
In the INVITE your opensips sends to 64.93 IP, you have the Contact
with 192.168.1.21 (priv IP of asterisk).
When you receive the BYE from 64.93 IP, the Route hdrs are ok (the 2
hdrs added by opensips to reflect the interface exchange), but the RURI is
wrong - it must be the contact from the INVITE you sent, but it seems to be
the IP of your opensips - this makes opensips to do act as strict router
and not like a loose routerand routing gets broken.
So, the 64.93 party or some other behind it, screw up the Contact in
the your INVITE and this alters the in-dialog requests - you should check
with the upstream guys.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
On 02/25/2013 04:36 PM, brad smith wrote:
I just tested an outbound call (Asterisk originate) without bridging and
get the same '404 not here' if that helps.
Thanks again,
Brad
On Mon, Feb 25, 2013 at 8:01 AM, Vlad Paiu vladp...@opensips.org wrote:
Hello,
Seems the incoming BYE does not have any Route headers, and the
loose_route() function returns false.
Since you have dialog support in your script, try
if (has_totag()) {
# sequential request withing a dialog should
# take the path determined by record-routing
if (loose_route() || match_dialog()) {
This way you will force matching of dialog sequential requests that have
no Route headers.
Best Regards,
Vlad Paiu
OpenSIPS Developerhttp://www.opensips-solutions.com
On 02/24/2013 02:57 AM, brad smith wrote:
Hello,
I am currently running opensips 1.8.1 no tls. It is
multi-homed with a public and private address.
I have a asterisk
1.8.19 in the lan that is connected to opensips via lan
address.
*issue*
A caller calls in
and then I place an outbound call and finally bridge the two
calls.
This works as
expected, except when the outbound caller hangs up first the
BYE never gets back to Asterisk.
I can see the BYE
reach OpenSips but a '404 not here' is returned to the ISP.
sip trace https://gist.github.com/5009662
opensips.cfg https://gist.github.com/5009704
thanks for your time.
___
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___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
___
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Re: [OpenSIPS-Users] 404 Not Here
Well, escalating the problem will be the right thing to do.
As a workaround on your side, you could try to enable the topo-hiding on
the dialog module, for your calls - this will take care of the contact
issue.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 02/26/2013 04:00 PM, brad smith wrote:
Bogdan,
Thanks for responding.
I am using vitelity for my upstream; I will send them a ticket. If
they fail to act, do you have any suggestions...switch carriers? any
config change?
Thanks again,
Brad
On Tue, Feb 26, 2013 at 7:58 AM, Bogdan-Andrei Iancu
bog...@opensips.org mailto:bog...@opensips.org wrote:
Hi Brad,
Thinks are a bit more complicated, it seems
In the INVITE your opensips sends to 64.93 IP, you have the
Contact with 192.168.1.21 (priv IP of asterisk).
When you receive the BYE from 64.93 IP, the Route hdrs are ok
(the 2 hdrs added by opensips to reflect the interface exchange),
but the RURI is wrong - it must be the contact from the INVITE you
sent, but it seems to be the IP of your opensips - this makes
opensips to do act as strict router and not like a loose
routerand routing gets broken.
So, the 64.93 party or some other behind it, screw up the
Contact in the your INVITE and this alters the in-dialog requests
- you should check with the upstream guys.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 02/25/2013 04:36 PM, brad smith wrote:
I just tested an outbound call (Asterisk originate) without
bridging and get the same '404 not here' if that helps.
Thanks again,
Brad
On Mon, Feb 25, 2013 at 8:01 AM, Vlad Paiu vladp...@opensips.org
mailto:vladp...@opensips.org wrote:
Hello,
Seems the incoming BYE does not have any Route headers, and
the loose_route() function returns false.
Since you have dialog support in your script, try
if (has_totag()) {
# sequential request withing a dialog should
# take the path determined by record-routing
if (loose_route() || match_dialog()) {
This way you will force matching of dialog sequential
requests that have no Route headers.
Best Regards,
Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com
On 02/24/2013 02:57 AM, brad smith wrote:
Hello,
I am currently running opensips 1.8.1 no tls. It
is
multi-homed with a public and private address.
I have a asterisk
1.8.19 in the lan that is connected to
opensips via lan
address.
*issue*
A caller calls in
and then I place an outbound call and finally
bridge the two
calls.
This works as
expected, except when the outbound caller
hangs up first the
BYE never gets back to Asterisk.
I can see the BYE
reach OpenSips but a '404 not here' is
returned to the ISP.
sip trace https://gist.github.com/5009662
opensips.cfg https://gist.github.com/5009704
thanks for your time.
___
Users mailing list
Users@lists.opensips.org mailto:Users@lists.opensips.org
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___
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Users@lists.opensips.org mailto:Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
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Re: [OpenSIPS-Users] 404 Not Here
Hello,
Seems the incoming BYE does not have any Route headers, and the
loose_route() function returns false.
Since you have dialog support in your script, try
if (has_totag()) {
# sequential request withing a dialog should
# take the path determined by record-routing
if (loose_route() || match_dialog()) {
This way you will force matching of dialog sequential requests that have
no Route headers.
Best Regards,
Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com
On 02/24/2013 02:57 AM, brad smith wrote:
Hello,
I am currently running opensips 1.8.1 no tls. It is multi-homed with a
public and private address.
I have a asterisk 1.8.19 in the lan that is connected to opensips via
lan address.
*issue*
A caller calls in and then I place an outbound call and finally bridge
the two calls.
This works as expected, except when the outbound caller hangs up first
the BYE never gets back to Asterisk.
I can see the BYE reach OpenSips but a '404 not here' is returned to
the ISP.
sip trace https://gist.github.com/5009662
opensips.cfg https://gist.github.com/5009704
thanks for your time.
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
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Users@lists.opensips.org
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Re: [OpenSIPS-Users] 404 Not Here
Vlad,
Thanks for responding. Unfortunately, I made the suggested change and
still have the same results. Here is some more information. I have added
a new sip trace and opensips log.
Thanks,
Brad
The call is origianted from (7278516359)
The caller dials 8665551212
The call is sent to 63.246.156.XX (opensips 1.8.1)
There the call is forwarded to 192.168.1.21 from 192.168.1.22 (opensips
second interface)
Asterisk answers, dials 7275551212 and then Bridges the two calls.
7275551212 hangs up first and the problem arises.
If 7278516359 hangs up first, the call terminates correctly.
One thing I did notice, when 7278516359 hangs up the sip trace shows the
path:
ISP --- opensips (public IP) -- Astersik (private IP)
When 727551212 hangs up first, the path is as follows.
ISP --- opensips (public IP) -- Opensips (public IP) -- 404 not here
opensips log
https://gist.github.com/anonymous/5030013
sip trace
https://gist.github.com/5030094
On Mon, Feb 25, 2013 at 8:01 AM, Vlad Paiu vladp...@opensips.org wrote:
**
Hello,
Seems the incoming BYE does not have any Route headers, and the
loose_route() function returns false.
Since you have dialog support in your script, try
if (has_totag()) {
# sequential request withing a dialog should
# take the path determined by record-routing
if (loose_route() || match_dialog()) {
This way you will force matching of dialog sequential requests that have
no Route headers.
Best Regards,
Vlad Paiu
OpenSIPS Developerhttp://www.opensips-solutions.com
On 02/24/2013 02:57 AM, brad smith wrote:
Hello,
I am currently running opensips 1.8.1 no tls. It is multi-homed with a
public and private address.
I have a asterisk 1.8.19 in the lan that is connected to opensips via lan
address.
*issue*
A caller calls in and then I place an outbound call and finally bridge the
two calls.
This works as expected, except when the outbound caller hangs up first the
BYE never gets back to Asterisk.
I can see the BYE reach OpenSips but a '404 not here' is returned to the
ISP.
sip trace https://gist.github.com/5009662
opensips.cfg https://gist.github.com/5009704
thanks for your time.
___
Users mailing
listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
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Users@lists.opensips.org
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Re: [OpenSIPS-Users] 404 Not Here
I just tested an outbound call (Asterisk originate) without bridging and
get the same '404 not here' if that helps.
Thanks again,
Brad
On Mon, Feb 25, 2013 at 8:01 AM, Vlad Paiu vladp...@opensips.org wrote:
**
Hello,
Seems the incoming BYE does not have any Route headers, and the
loose_route() function returns false.
Since you have dialog support in your script, try
if (has_totag()) {
# sequential request withing a dialog should
# take the path determined by record-routing
if (loose_route() || match_dialog()) {
This way you will force matching of dialog sequential requests that have
no Route headers.
Best Regards,
Vlad Paiu
OpenSIPS Developerhttp://www.opensips-solutions.com
On 02/24/2013 02:57 AM, brad smith wrote:
Hello,
I am currently running opensips 1.8.1 no tls. It is multi-homed with a
public and private address.
I have a asterisk 1.8.19 in the lan that is connected to opensips via lan
address.
*issue*
A caller calls in and then I place an outbound call and finally bridge the
two calls.
This works as expected, except when the outbound caller hangs up first the
BYE never gets back to Asterisk.
I can see the BYE reach OpenSips but a '404 not here' is returned to the
ISP.
sip trace https://gist.github.com/5009662
opensips.cfg https://gist.github.com/5009704
thanks for your time.
___
Users mailing
listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
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[OpenSIPS-Users] 404 Not Here
Hello, I am currently running opensips 1.8.1 no tls. It is multi-homed with a public and private address. I have a asterisk 1.8.19 in the lan that is connected to opensips via lan address. *issue* A caller calls in and then I place an outbound call and finally bridge the two calls. This works as expected, except when the outbound caller hangs up first the BYE never gets back to Asterisk. I can see the BYE reach OpenSips but a '404 not here' is returned to the ISP. sip trace https://gist.github.com/5009662 opensips.cfg https://gist.github.com/5009704 thanks for your time. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] 404 Not Here error
All,I was reading the thread regarding the uac_replace_from issues Jeff brought up and was thinking my issue may be similar. I have a carrier who sends me BYE messages with a RURI that does NOT match the Contact header in the 200 OK. Of course, OpenSIPs replies with a 404 Not Here. The last message in Jeff's question regarding uac_replace_from suggested that the dialog module could be used to help identify the replies properly. I already use the dialog module in my flows, but I'm unsure how it would help with this particular problem (or if it would). Any ideas? Or am I just dealing with a broken UA on the other side? Thanks, Brett ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] 404 Not Here error
Hi Brett, The dialog module can be used to validate the sequential requests (based on the stored info like RR and contacts) - maybe some functions to do that will be useful :). Thinking in the future (but debatable), you can do fixing of the sequential requests (based on the stored info). Regards, Bogdan Brett Nemeroff wrote: All, I was reading the thread regarding the uac_replace_from issues Jeff brought up and was thinking my issue may be similar. I have a carrier who sends me BYE messages with a RURI that does NOT match the Contact header in the 200 OK. Of course, OpenSIPs replies with a 404 Not Here. The last message in Jeff's question regarding uac_replace_from suggested that the dialog module could be used to help identify the replies properly. I already use the dialog module in my flows, but I'm unsure how it would help with this particular problem (or if it would). Any ideas? Or am I just dealing with a broken UA on the other side? Thanks, Brett ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
