subject:"\[OpenSIPS\-Users\] \- topology

Re: [OpenSIPS-Users] - topology_hiding and no ACK

2022-11-01 Thread Nitesh Divecha

Bogdan,

Ah haa! Found it and fixed it!

Much appreciated for your help... This was a good learning experience!

Next up is PSTN routing and local call routing.


Cheers,
Nitesh



On Tue, Nov 1, 2022 at 10:55 AM Bogdan-Andrei Iancu 
wrote:

> Hi Nitesh,
>
> Be sure all the remove_hf() are on the right execution path and BEFORE the
> t_relay() for the sequential requests.
>
> Regards,
>
> Bogdan-Andrei Iancu
>
> OpenSIPS Founder and Developer
>   https://www.opensips-solutions.com
> OpenSIPS Bootcamp 5-16 Dec 2022, online
>   https://www.opensips.org/training/OpenSIPS_eBootcamp_2022/
>
> On 11/1/22 2:15 PM, Nitesh Divecha wrote:
>
> Bogdan,
>
> Thanks for your feedback.
>
> After reading the docs and following this tutorial (
> https://opensips.org/Documentation/Tutorials-Topology-Hiding) I finally
> got it working. Thank you!
>
> INVITE is clean, topology is hidden and ACK is sent back and forth. Calls
> are holding good.
>
> While confirming each ACK and BYE, I noticed Originator's User-Agent is
> passed on again only on to ACK and BYE.
>
> I tried to define under:
>
> route{
>
> if (has_totag()) {
>
> if ( is_method("ACK") ) {
>
> if(remove_hf("User-Agent")){
> xlog("*** has-totag -> ACK
> User-Agent found and removed on $rm from $si (callid=$ci). ***\n");
> }
>
> }
>
> if (is_method("BYE")) {
>
> xlog("*** if is_method(BYE) on $rm from $si (callid=$ci) ***\n");
>
> if(remove_hf("User-Agent")){
>
> xlog("*** is_method BYE User-Agent found and removed on $rm from $si
> (callid=$ci). ***\n");
>
> }
>
> # do accounting even if the transaction fails
>
> do_accounting("db","failed");
>
> }
>
> }
>
> }
>
> But still Originator's User-Agent is passed on to ACK and BYE... Ignore
> the xlog(""), it is just my way of learning what each line is doing and if
> getting executed :-).
>
> Any suggestions?
>
> Cheers,
> Nitesh
>
>
>
>
>
>
>
>
>
> On Thu, Oct 27, 2022 at 11:34 AM Bogdan-Andrei Iancu 
> wrote:
>
>> Hi Nitesh,
>>
>> Where you have the loose_route() in your script right now, use
>> topology_hiding_match()
>>
>> https://opensips.org/html/docs/modules/3.2.x/topology_hiding.html#func_topology_hiding_match
>>
>> Regards,
>>
>> Bogdan-Andrei Iancu
>>
>> OpenSIPS Founder and Developer
>>   https://www.opensips-solutions.com
>> OpenSIPS Bootcamp 5-16 Dec 2022, online
>>   https://www.opensips.org/training/OpenSIPS_eBootcamp_2022/
>>
>> On 10/27/22 3:51 PM, Nitesh Divecha wrote:
>>
>> Hey Bogdan-Andrei,
>>
>> Thanks for your feedback.
>>
>> To be honest - I don't know!
>>
>> I followed some online tutorials and tried to implement
>> "topology_hiding()" after is_method("INVITE") since the first message
>> sent to Outbound Provider is INVITE.
>>
>> I never tried topology_hiding_match().
>>
>> What is the right way to implement topology_hiding module? My scenario is
>> simple, I'm using the default cfg and making ATA register to OpenSIPS to
>> make Outbound PSTN calls.
>>
>> I guess Bootcamp is the answer :-)
>>
>> Cheers,
>> Nitesh
>>
>>
>> On Thu, Oct 27, 2022 at 4:02 AM Bogdan-Andrei Iancu 
>> wrote:
>>
>>> Hi Nitesh,
>>>
>>> Do you do topology_hiding_match() for the sequential requests (instead
>>> of the typical loose_route()) ?  As it seems you cfg fails to properly
>>> handle the in-dialog / sequential requests.
>>>
>>> Regards,
>>>
>>> Bogdan-Andrei Iancu
>>>
>>> OpenSIPS Founder and Developer
>>>   https://www.opensips-solutions.com
>>> OpenSIPS Bootcamp 5-16 Dec 2022, online
>>>   https://www.opensips.org/training/OpenSIPS_eBootcamp_2022/
>>>
>>> On 10/26/22 9:05 PM, Nitesh Divecha wrote:
>>>
>>> Hello All,
>>>
>>> I don't know if this is by design or me not implementing correctly! I'm
>>> a newbie to OpenSIPS...
>>>
>>> So I generated a new opensips_residential.cfg file and only edited it
>>> with topology_hiding("UC"); under INVITE. Plus all the extras to make ATA
>>> register successfully and SIP trunk.
>>>
>>> Every time I make a call two INVITE dialogs are created. One from ATA to
>>> OpenSIPS and second one from OpenSIPS to Outbound Provider. When a PSTN
>>> call is answered, the Outbound Provider sends 200 OK back to OpenSIPS, and
>>> OpenSIPS sends back 200 OK to ATA. Then ATA acknowledges with ACK back to
>>> OpenSIPS BUT OpenSIPS fails to send ACK to the Outbound Provider. So
>>> Outbound Provider sends 200 OK again to OpenSIPS and OpenSIPS sends 200 OK
>>> to ATA then ATA acknowledges with ACK back to OpenSIPS BUT OpenSIPS fails
>>> to send back ACK... This dance goes on for 30 secs and until Outbound
>>> Provider drops the call due to no ACK.
>>>
>>> On the sngrep it shows that call from ATA to OpenSIPS "COMPLETED" while
>>> OpenSIPS to Outbound Provider "CALL SETUP".
>>>
>>> If I comment out topology_hiding("UC"); then everything works perfectly
>>> and sngrep shows only one INVITE dialog from ATA to OpenSIPS to Outbound
>>> Provider except the Outbound Provider can see everything that is

Re: [OpenSIPS-Users] - topology_hiding and no ACK

2022-11-01 Thread Bogdan-Andrei Iancu


Hi Nitesh,

Be sure all the remove_hf() are on the right execution path and BEFORE 
the t_relay() for the sequential requests.


Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
  https://www.opensips-solutions.com
OpenSIPS Bootcamp 5-16 Dec 2022, online
  https://www.opensips.org/training/OpenSIPS_eBootcamp_2022/

On 11/1/22 2:15 PM, Nitesh Divecha wrote:

Bogdan,

Thanks for your feedback.

After reading the docs and following this tutorial 
(https://opensips.org/Documentation/Tutorials-Topology-Hiding 
) I 
finally got it working. Thank you!


INVITE is clean, topology is hidden and ACK is sent back and forth. 
Calls are holding good.


While confirming each ACK and BYE, I noticed Originator's User-Agent 
is passed on again only on to ACK and BYE.


I tried to define under:

route{

if (has_totag()) {

if ( is_method("ACK") ) {

                              if(remove_hf("User-Agent")){
                                        xlog("*** has-totag -> ACK
User-Agent found and removed on $rm from $si (callid=$ci). ***\n");
                                }

}

if (is_method("BYE")) {

xlog("*** if is_method(BYE) on $rm from $si (callid=$ci)
***\n");

if(remove_hf("User-Agent")){

xlog("*** is_method BYE User-Agent found and removed
on $rm from $si (callid=$ci). ***\n");

}

# do accounting even if the transaction fails

do_accounting("db","failed");

}

}

}

But still Originator's User-Agent is passed on to ACK and BYE... 
Ignore the xlog(""), it is just my way of learning what each line is 
doing and if getting executed :-).


Any suggestions?

Cheers,
Nitesh









On Thu, Oct 27, 2022 at 11:34 AM Bogdan-Andrei Iancu 
mailto:bog...@opensips.org>> wrote:


Hi Nitesh,

Where you have the loose_route() in your script right now, use
topology_hiding_match()

https://opensips.org/html/docs/modules/3.2.x/topology_hiding.html#func_topology_hiding_match



Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
   https://www.opensips-solutions.com  
OpenSIPS Bootcamp 5-16 Dec 2022, online
   https://www.opensips.org/training/OpenSIPS_eBootcamp_2022/  


On 10/27/22 3:51 PM, Nitesh Divecha wrote:

Hey Bogdan-Andrei,

Thanks for your feedback.

To be honest - I don't know!

I followed some online tutorials and tried to implement
"topology_hiding()" after is_method("INVITE") since the first
message sent to Outbound Provider is INVITE.

I never tried topology_hiding_match().

What is the right way to implement topology_hiding module? My
scenario is simple, I'm using the default cfg and making ATA
register to OpenSIPS to make Outbound PSTN calls.

I guess Bootcamp is the answer :-)

Cheers,
Nitesh


On Thu, Oct 27, 2022 at 4:02 AM Bogdan-Andrei Iancu
mailto:bog...@opensips.org>> wrote:

Hi Nitesh,

Do you do topology_hiding_match() for the sequential requests
(instead of the typical loose_route()) ?  As it seems you cfg
fails to properly handle the in-dialog / sequential requests.

Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
   https://www.opensips-solutions.com  

OpenSIPS Bootcamp 5-16 Dec 2022, online
   https://www.opensips.org/training/OpenSIPS_eBootcamp_2022/  


On 10/26/22 9:05 PM, Nitesh Divecha wrote:

Hello All,

I don't know if this is by design or me not implementing
correctly! I'm a newbie to OpenSIPS...

So I generated a new opensips_residential.cfg file and only
edited it with topology_hiding("UC"); under INVITE. Plus all
the extras to make ATA register successfully and SIP trunk.

Every time I make a call two INVITE dialogs are created. One
from ATA to OpenSIPS and second one from OpenSIPS to
Outbound Provider. When a PSTN call is answered, the
Outbound Provider sends 200 OK back to OpenSIPS, and
OpenSIPS sends back 200 OK to ATA. Then ATA acknowledges
with ACK back to OpenSIPS BUT OpenSIPS fails to send ACK to
the Outbound Provider. So Outbound Provider sends 200 OK
again to OpenSIPS and OpenSIPS sends 200 OK to ATA then ATA
acknowledges with ACK back to OpenSIPS BUT OpenSIPS fails to
send back ACK... This dance goes on for 30 secs and until
Outbound Provider drops the call due to no ACK.

On the sngrep it shows that call from ATA to

Re: [OpenSIPS-Users] - topology_hiding and no ACK

2022-11-01 Thread Nitesh Divecha

Bogdan,

Thanks for your feedback.

After reading the docs and following this tutorial (
https://opensips.org/Documentation/Tutorials-Topology-Hiding) I finally got
it working. Thank you!

INVITE is clean, topology is hidden and ACK is sent back and forth. Calls
are holding good.

While confirming each ACK and BYE, I noticed Originator's User-Agent is
passed on again only on to ACK and BYE.

I tried to define under:

route{

if (has_totag()) {

if ( is_method("ACK") ) {

if(remove_hf("User-Agent")){
xlog("*** has-totag -> ACK
User-Agent found and removed on $rm from $si (callid=$ci). ***\n");
}

}

if (is_method("BYE")) {

xlog("*** if is_method(BYE) on $rm from $si (callid=$ci) ***\n");

if(remove_hf("User-Agent")){

xlog("*** is_method BYE User-Agent found and removed on $rm from $si
(callid=$ci). ***\n");

}

# do accounting even if the transaction fails

do_accounting("db","failed");

}

}

}

But still Originator's User-Agent is passed on to ACK and BYE... Ignore the
xlog(""), it is just my way of learning what each line is doing and if
getting executed :-).

Any suggestions?

Cheers,
Nitesh









On Thu, Oct 27, 2022 at 11:34 AM Bogdan-Andrei Iancu 
wrote:

> Hi Nitesh,
>
> Where you have the loose_route() in your script right now, use
> topology_hiding_match()
>
> https://opensips.org/html/docs/modules/3.2.x/topology_hiding.html#func_topology_hiding_match
>
> Regards,
>
> Bogdan-Andrei Iancu
>
> OpenSIPS Founder and Developer
>   https://www.opensips-solutions.com
> OpenSIPS Bootcamp 5-16 Dec 2022, online
>   https://www.opensips.org/training/OpenSIPS_eBootcamp_2022/
>
> On 10/27/22 3:51 PM, Nitesh Divecha wrote:
>
> Hey Bogdan-Andrei,
>
> Thanks for your feedback.
>
> To be honest - I don't know!
>
> I followed some online tutorials and tried to implement
> "topology_hiding()" after is_method("INVITE") since the first message
> sent to Outbound Provider is INVITE.
>
> I never tried topology_hiding_match().
>
> What is the right way to implement topology_hiding module? My scenario is
> simple, I'm using the default cfg and making ATA register to OpenSIPS to
> make Outbound PSTN calls.
>
> I guess Bootcamp is the answer :-)
>
> Cheers,
> Nitesh
>
>
> On Thu, Oct 27, 2022 at 4:02 AM Bogdan-Andrei Iancu 
> wrote:
>
>> Hi Nitesh,
>>
>> Do you do topology_hiding_match() for the sequential requests (instead of
>> the typical loose_route()) ?  As it seems you cfg fails to properly handle
>> the in-dialog / sequential requests.
>>
>> Regards,
>>
>> Bogdan-Andrei Iancu
>>
>> OpenSIPS Founder and Developer
>>   https://www.opensips-solutions.com
>> OpenSIPS Bootcamp 5-16 Dec 2022, online
>>   https://www.opensips.org/training/OpenSIPS_eBootcamp_2022/
>>
>> On 10/26/22 9:05 PM, Nitesh Divecha wrote:
>>
>> Hello All,
>>
>> I don't know if this is by design or me not implementing correctly! I'm a
>> newbie to OpenSIPS...
>>
>> So I generated a new opensips_residential.cfg file and only edited it
>> with topology_hiding("UC"); under INVITE. Plus all the extras to make ATA
>> register successfully and SIP trunk.
>>
>> Every time I make a call two INVITE dialogs are created. One from ATA to
>> OpenSIPS and second one from OpenSIPS to Outbound Provider. When a PSTN
>> call is answered, the Outbound Provider sends 200 OK back to OpenSIPS, and
>> OpenSIPS sends back 200 OK to ATA. Then ATA acknowledges with ACK back to
>> OpenSIPS BUT OpenSIPS fails to send ACK to the Outbound Provider. So
>> Outbound Provider sends 200 OK again to OpenSIPS and OpenSIPS sends 200 OK
>> to ATA then ATA acknowledges with ACK back to OpenSIPS BUT OpenSIPS fails
>> to send back ACK... This dance goes on for 30 secs and until Outbound
>> Provider drops the call due to no ACK.
>>
>> On the sngrep it shows that call from ATA to OpenSIPS "COMPLETED" while
>> OpenSIPS to Outbound Provider "CALL SETUP".
>>
>> If I comment out topology_hiding("UC"); then everything works perfectly
>> and sngrep shows only one INVITE dialog from ATA to OpenSIPS to Outbound
>> Provider except the Outbound Provider can see everything that is ATA's IP
>> info, etc...
>>
>> Here is my cfg:
>>
>> route {
>> ...
>> if (dp_translate(10 ,$rU ,$rU) ) {
>> xlog("*** Dial plan translate from source $avp(src) to
>> $rU ***\n");
>>
>> $avp(furi) = "sip:xxx...@gothamcity.com";
>> uac_replace_from( , "$avp(furi)");
>> #strip(1);
>> if (!do_routing(0)) {
>> send_reply(500,"No PSTN Route found");
>> exit;
>> }
>> # t_on_branch("change_from");
>> route(relay);
>> exit;
>> }
>> ...
>> }
>>
>> route[relay] {
>> # for INVITEs enable some additional helper routes
>> if (is_method("INVITE")) {
>> # create_dialog();
>>

Re: [OpenSIPS-Users] - topology_hiding and no ACK

2022-10-27 Thread Bogdan-Andrei Iancu


Hi Nitesh,

Where you have the loose_route() in your script right now, use 
topology_hiding_match()

https://opensips.org/html/docs/modules/3.2.x/topology_hiding.html#func_topology_hiding_match

Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
  https://www.opensips-solutions.com
OpenSIPS Bootcamp 5-16 Dec 2022, online
  https://www.opensips.org/training/OpenSIPS_eBootcamp_2022/

On 10/27/22 3:51 PM, Nitesh Divecha wrote:

Hey Bogdan-Andrei,

Thanks for your feedback.

To be honest - I don't know!

I followed some online tutorials and tried to implement 
"topology_hiding()" after is_method("INVITE") since the first message 
sent to Outbound Provider is INVITE.


I never tried topology_hiding_match().

What is the right way to implement topology_hiding module? My scenario 
is simple, I'm using the default cfg and making ATA register to 
OpenSIPS to make Outbound PSTN calls.


I guess Bootcamp is the answer :-)

Cheers,
Nitesh


On Thu, Oct 27, 2022 at 4:02 AM Bogdan-Andrei Iancu 
mailto:bog...@opensips.org>> wrote:


Hi Nitesh,

Do you do topology_hiding_match() for the sequential requests
(instead of the typical loose_route()) ?  As it seems you cfg
fails to properly handle the in-dialog / sequential requests.

Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
   https://www.opensips-solutions.com  
OpenSIPS Bootcamp 5-16 Dec 2022, online
   https://www.opensips.org/training/OpenSIPS_eBootcamp_2022/  


On 10/26/22 9:05 PM, Nitesh Divecha wrote:

Hello All,

I don't know if this is by design or me not implementing
correctly! I'm a newbie to OpenSIPS...

So I generated a new opensips_residential.cfg file and only
edited it with topology_hiding("UC"); under INVITE. Plus all the
extras to make ATA register successfully and SIP trunk.

Every time I make a call two INVITE dialogs are created. One from
ATA to OpenSIPS and second one from OpenSIPS to Outbound
Provider. When a PSTN call is answered, the Outbound Provider
sends 200 OK back to OpenSIPS, and OpenSIPS sends back 200 OK to
ATA. Then ATA acknowledges with ACK back to OpenSIPS BUT OpenSIPS
fails to send ACK to the Outbound Provider. So Outbound Provider
sends 200 OK again to OpenSIPS and OpenSIPS sends 200 OK to ATA
then ATA acknowledges with ACK back to OpenSIPS BUT OpenSIPS
fails to send back ACK... This dance goes on for 30 secs and
until Outbound Provider drops the call due to no ACK.

On the sngrep it shows that call from ATA to OpenSIPS "COMPLETED"
while OpenSIPS to Outbound Provider "CALL SETUP".

If I comment out topology_hiding("UC"); then everything works
perfectly and sngrep shows only one INVITE dialog from ATA to
OpenSIPS to Outbound Provider except the Outbound Provider can
see everything that is ATA's IP info, etc...

Here is my cfg:

route {
...
if (dp_translate(10 ,$rU ,$rU) ) {
              xlog("*** Dial plan translate from source $avp(src)
to $rU ***\n");

              $avp(furi) = "sip:xxx...@gothamcity.com
";
              uac_replace_from( , "$avp(furi)");
              #strip(1);
              if (!do_routing(0)) {
                      send_reply(500,"No PSTN Route found");
                      exit;
              }
              # t_on_branch("change_from");
              route(relay);
              exit;
      }
...
}

route[relay] {
      # for INVITEs enable some additional helper routes
      if (is_method("INVITE")) {
              # create_dialog();
              topology_hiding("UC");
              if(remove_hf("User-Agent")){
                      xlog("*** 4. User-Agent found and removed.
***\n");
              }

              if (isflagset("NAT") && has_body("application/sdp")) {
                      rtpproxy_offer("ro");
              }

              t_on_branch("per_branch_ops");
              t_on_reply("handle_nat");
              t_on_failure("missed_call");
      }

      if (isflagset("NAT")) {
              add_rr_param(";nat=yes");
      }

      if (!t_relay()) {
              send_reply(500,"Internal Error");
      }
      exit;
}

My eyes are sore and my head is spinning... Any help will be
highly appreciated... Thanks!

Cheers,
Nitesh


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Re: [OpenSIPS-Users] - topology_hiding and no ACK

2022-10-27 Thread Nitesh Divecha

Hey Bogdan-Andrei,

Thanks for your feedback.

To be honest - I don't know!

I followed some online tutorials and tried to implement "topology_hiding()"
after is_method("INVITE") since the first message sent to Outbound Provider
is INVITE.

I never tried topology_hiding_match().

What is the right way to implement topology_hiding module? My scenario is
simple, I'm using the default cfg and making ATA register to OpenSIPS to
make Outbound PSTN calls.

I guess Bootcamp is the answer :-)

Cheers,
Nitesh


On Thu, Oct 27, 2022 at 4:02 AM Bogdan-Andrei Iancu 
wrote:

> Hi Nitesh,
>
> Do you do topology_hiding_match() for the sequential requests (instead of
> the typical loose_route()) ?  As it seems you cfg fails to properly handle
> the in-dialog / sequential requests.
>
> Regards,
>
> Bogdan-Andrei Iancu
>
> OpenSIPS Founder and Developer
>   https://www.opensips-solutions.com
> OpenSIPS Bootcamp 5-16 Dec 2022, online
>   https://www.opensips.org/training/OpenSIPS_eBootcamp_2022/
>
> On 10/26/22 9:05 PM, Nitesh Divecha wrote:
>
> Hello All,
>
> I don't know if this is by design or me not implementing correctly! I'm a
> newbie to OpenSIPS...
>
> So I generated a new opensips_residential.cfg file and only edited it
> with topology_hiding("UC"); under INVITE. Plus all the extras to make ATA
> register successfully and SIP trunk.
>
> Every time I make a call two INVITE dialogs are created. One from ATA to
> OpenSIPS and second one from OpenSIPS to Outbound Provider. When a PSTN
> call is answered, the Outbound Provider sends 200 OK back to OpenSIPS, and
> OpenSIPS sends back 200 OK to ATA. Then ATA acknowledges with ACK back to
> OpenSIPS BUT OpenSIPS fails to send ACK to the Outbound Provider. So
> Outbound Provider sends 200 OK again to OpenSIPS and OpenSIPS sends 200 OK
> to ATA then ATA acknowledges with ACK back to OpenSIPS BUT OpenSIPS fails
> to send back ACK... This dance goes on for 30 secs and until Outbound
> Provider drops the call due to no ACK.
>
> On the sngrep it shows that call from ATA to OpenSIPS "COMPLETED" while
> OpenSIPS to Outbound Provider "CALL SETUP".
>
> If I comment out topology_hiding("UC"); then everything works perfectly
> and sngrep shows only one INVITE dialog from ATA to OpenSIPS to Outbound
> Provider except the Outbound Provider can see everything that is ATA's IP
> info, etc...
>
> Here is my cfg:
>
> route {
> ...
> if (dp_translate(10 ,$rU ,$rU) ) {
> xlog("*** Dial plan translate from source $avp(src) to $rU
> ***\n");
>
> $avp(furi) = "sip:xxx...@gothamcity.com";
> uac_replace_from( , "$avp(furi)");
> #strip(1);
> if (!do_routing(0)) {
> send_reply(500,"No PSTN Route found");
> exit;
> }
> # t_on_branch("change_from");
> route(relay);
> exit;
> }
> ...
> }
>
> route[relay] {
> # for INVITEs enable some additional helper routes
> if (is_method("INVITE")) {
> # create_dialog();
> topology_hiding("UC");
> if(remove_hf("User-Agent")){
> xlog("*** 4. User-Agent found and removed. ***\n");
> }
>
> if (isflagset("NAT") && has_body("application/sdp")) {
> rtpproxy_offer("ro");
> }
>
> t_on_branch("per_branch_ops");
> t_on_reply("handle_nat");
> t_on_failure("missed_call");
> }
>
> if (isflagset("NAT")) {
> add_rr_param(";nat=yes");
> }
>
> if (!t_relay()) {
> send_reply(500,"Internal Error");
> }
> exit;
> }
>
> My eyes are sore and my head is spinning... Any help will be highly
> appreciated... Thanks!
>
> Cheers,
> Nitesh
>
>
> ___
> Users mailing 
> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
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Re: [OpenSIPS-Users] - topology_hiding and no ACK

2022-10-27 Thread Bogdan-Andrei Iancu


Hi Nitesh,

Do you do topology_hiding_match() for the sequential requests (instead 
of the typical loose_route()) ?  As it seems you cfg fails to properly 
handle the in-dialog / sequential requests.


Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
  https://www.opensips-solutions.com
OpenSIPS Bootcamp 5-16 Dec 2022, online
  https://www.opensips.org/training/OpenSIPS_eBootcamp_2022/

On 10/26/22 9:05 PM, Nitesh Divecha wrote:

Hello All,

I don't know if this is by design or me not implementing correctly! 
I'm a newbie to OpenSIPS...


So I generated a new opensips_residential.cfg file and only edited it 
with topology_hiding("UC"); under INVITE. Plus all the extras to make 
ATA register successfully and SIP trunk.


Every time I make a call two INVITE dialogs are created. One from ATA 
to OpenSIPS and second one from OpenSIPS to Outbound Provider. When a 
PSTN call is answered, the Outbound Provider sends 200 OK back to 
OpenSIPS, and OpenSIPS sends back 200 OK to ATA. Then ATA acknowledges 
with ACK back to OpenSIPS BUT OpenSIPS fails to send ACK to the 
Outbound Provider. So Outbound Provider sends 200 OK again to OpenSIPS 
and OpenSIPS sends 200 OK to ATA then ATA acknowledges with ACK back 
to OpenSIPS BUT OpenSIPS fails to send back ACK... This dance goes on 
for 30 secs and until Outbound Provider drops the call due to no ACK.


On the sngrep it shows that call from ATA to OpenSIPS "COMPLETED" 
while OpenSIPS to Outbound Provider "CALL SETUP".


If I comment out topology_hiding("UC"); then everything works 
perfectly and sngrep shows only one INVITE dialog from ATA to OpenSIPS 
to Outbound Provider except the Outbound Provider can see everything 
that is ATA's IP info, etc...


Here is my cfg:

route {
...
if (dp_translate(10 ,$rU ,$rU) ) {
              xlog("*** Dial plan translate from source $avp(src) to 
$rU ***\n");


              $avp(furi) = "sip:xxx...@gothamcity.com 
";

              uac_replace_from( , "$avp(furi)");
              #strip(1);
              if (!do_routing(0)) {
                      send_reply(500,"No PSTN Route found");
                      exit;
              }
              # t_on_branch("change_from");
              route(relay);
              exit;
      }
...
}

route[relay] {
      # for INVITEs enable some additional helper routes
      if (is_method("INVITE")) {
              # create_dialog();
              topology_hiding("UC");
              if(remove_hf("User-Agent")){
                      xlog("*** 4. User-Agent found and removed. ***\n");
              }

              if (isflagset("NAT") && has_body("application/sdp")) {
                      rtpproxy_offer("ro");
              }

              t_on_branch("per_branch_ops");
              t_on_reply("handle_nat");
              t_on_failure("missed_call");
      }

      if (isflagset("NAT")) {
              add_rr_param(";nat=yes");
      }

      if (!t_relay()) {
              send_reply(500,"Internal Error");
      }
      exit;
}

My eyes are sore and my head is spinning... Any help will be highly 
appreciated... Thanks!


Cheers,
Nitesh


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[OpenSIPS-Users] - topology_hiding and no ACK

2022-10-26 Thread Nitesh Divecha

Hello All,

I don't know if this is by design or me not implementing correctly! I'm a
newbie to OpenSIPS...

So I generated a new opensips_residential.cfg file and only edited it
with topology_hiding("UC"); under INVITE. Plus all the extras to make ATA
register successfully and SIP trunk.

Every time I make a call two INVITE dialogs are created. One from ATA to
OpenSIPS and second one from OpenSIPS to Outbound Provider. When a PSTN
call is answered, the Outbound Provider sends 200 OK back to OpenSIPS, and
OpenSIPS sends back 200 OK to ATA. Then ATA acknowledges with ACK back to
OpenSIPS BUT OpenSIPS fails to send ACK to the Outbound Provider. So
Outbound Provider sends 200 OK again to OpenSIPS and OpenSIPS sends 200 OK
to ATA then ATA acknowledges with ACK back to OpenSIPS BUT OpenSIPS fails
to send back ACK... This dance goes on for 30 secs and until Outbound
Provider drops the call due to no ACK.

On the sngrep it shows that call from ATA to OpenSIPS "COMPLETED" while
OpenSIPS to Outbound Provider "CALL SETUP".

If I comment out topology_hiding("UC"); then everything works perfectly and
sngrep shows only one INVITE dialog from ATA to OpenSIPS to Outbound
Provider except the Outbound Provider can see everything that is ATA's IP
info, etc...

Here is my cfg:

route {
...
if (dp_translate(10 ,$rU ,$rU) ) {
xlog("*** Dial plan translate from source $avp(src) to $rU
***\n");

$avp(furi) = "sip:xxx...@gothamcity.com";
uac_replace_from( , "$avp(furi)");
#strip(1);
if (!do_routing(0)) {
send_reply(500,"No PSTN Route found");
exit;
}
# t_on_branch("change_from");
route(relay);
exit;
}
...
}

route[relay] {
# for INVITEs enable some additional helper routes
if (is_method("INVITE")) {
# create_dialog();
topology_hiding("UC");
if(remove_hf("User-Agent")){
xlog("*** 4. User-Agent found and removed. ***\n");
}

if (isflagset("NAT") && has_body("application/sdp")) {
rtpproxy_offer("ro");
}

t_on_branch("per_branch_ops");
t_on_reply("handle_nat");
t_on_failure("missed_call");
}

if (isflagset("NAT")) {
add_rr_param(";nat=yes");
}

if (!t_relay()) {
send_reply(500,"Internal Error");
}
exit;
}

My eyes are sore and my head is spinning... Any help will be highly
appreciated... Thanks!

Cheers,
Nitesh
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] - topology_hiding and no ACK

Re: [OpenSIPS-Users] - topology_hiding and no ACK

Re: [OpenSIPS-Users] - topology_hiding and no ACK

Re: [OpenSIPS-Users] - topology_hiding and no ACK

Re: [OpenSIPS-Users] - topology_hiding and no ACK

Re: [OpenSIPS-Users] - topology_hiding and no ACK

[OpenSIPS-Users] - topology_hiding and no ACK

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