Re: [OpenSIPS-Users] web sockets (wss) error

[John Nash]

Git version (2.2) is OK. May be in tar download latest files are not there
no biggi.

On Tue, Jun 21, 2016 at 10:08 PM, John Nash  wrote:

> I downloaded opensips 2.2 stable tar file and upgraded my existing
> opensips.cfg to use wss as per document
> http://www.opensips.org/Documentation/Tutorials-WebSocket-2-2
>
> but when I start opensips I get error "cannot handle protocol " right
> at the line where I have listener wss:127.0.0.1:443
>
> Do I need to clone current git version in order to test wss?
>
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

[OpenSIPS-Users] Leak AVPOS + SQLITE

[Rodrigo Pimenta Carvalho]

Hi.


Does someone here is getting/handling memory leaks with OpenSIPS 2.2 and last 
version of SQLite?

I'm using newest commit from OpenSIPS 2.2 and newest version of SQLite.


My query is :


avp_db_query("select Value from GeneralConfigurations where Attribute = 
'CONFIGURATION_INTERCOM_A_NAME'");


Valgrind shows:


==16087== ERROR SUMMARY: 0 errors from 0 contexts (suppressed: 2 from 2)

==16088== Searching for pointers to 296,489 not-freed blocks

==16088== Checked 103,297,688 bytes

==16088==

==16088== 1,024 bytes in 1 blocks are possibly lost in loss record 184 of 246

==16088==at 0x4C2745D: malloc (in 
/usr/lib64/valgrind/vgpreload_memcheck-amd64-linux.so)

==16088==by 0x8F8B05F: sqlite3MemMalloc (sqlite3.c:20167)

==16088==by 0x8F701C7: sqlite3Malloc (sqlite3.c:23846)

==16088==by 0x8F75459: pcache1Alloc (sqlite3.c:44312)

==16088==by 0x8F8019F: sqlite3BtreeCursor (sqlite3.c:44455)

==16088==by 0x8FD0FDD: sqlite3VdbeExec (sqlite3.c:80098)

==16088==by 0x8FDB89F: sqlite3_step (sqlite3.c:75131)

==16088==by 0x8FDC9A1: sqlite3_exec (sqlite3.c:108122)

==16088==by 0x8D20736: db_sqlite_raw_query (dbase.c:358)

==16088==by 0x9464DB8: db_query_avp (avpops_db.c:436)

==16088==by 0x946943E: ops_dbquery_avps (avpops_impl.c:840)

==16088==by 0x9459A61: w_dbquery_avps (avpops.c:1395)

==16088==

==16088== LEAK SUMMARY:

==16088==definitely lost: 0 bytes in 0 blocks

==16088==indirectly lost: 0 bytes in 0 blocks

==16088==  possibly lost: 1,024 bytes in 1 blocks

==16088==still reachable: 47,457,573 bytes in 296,488 blocks

==16088== suppressed: 0 bytes in 0 blocks

==16088== Reachable blocks (those to which a pointer was found) are not shown.

==16088== To see them, rerun with: --leak-check=full --show-leak-kinds=all



After some time running that query, I can see, via command top, that the 
available memory is decreasing.

In fact, the memory is not freed even after stop running the query for a time.


Best regards.


RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979





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

Re: [OpenSIPS-Users] DNS-SRV query in opensips

[Ramachandran, Agalya (Contractor)]

Hi Bogdan,

I have a question regarding seturi and setdsturi function calls.
As far as my understanding, when append_branch() is called, seturi () is called 
to set the URI where to fork the call.

I tried by calling only seturi () function call, after append_branch it was 
working same behavior as when I used setdsturi() as well.
My question is do we really need setdsturi or when is the case when setdsturi() 
is used.?

Regards,
Agalya

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org]
Sent: Tuesday, June 21, 2016 3:41 AM
To: OpenSIPS users mailling list 
>; Ramachandran, 
Agalya (Contractor) 
>; 
Ramachandran, Agalya (Contractor) 
>
Subject: Re: [OpenSIPS-Users] DNS-SRV query in opensips

Hi Agalya,

OpenSIPS does full flavor DNS lookup (with NATPR and SRV), but this is 
internal, and not accessible from script. OpenSIPS implements auto DNS-based 
failover :
http://www.opensips.org/html/docs/modules/2.2.x/tm.html#id293694

My understanding is you want the DNS resolving to be done at script level and 
to have access to the results ?

Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer

http://www.opensips-solutions.com
On 20.06.2016 20:35, Ramachandran, Agalya (Contractor) wrote:
Hi team,

We are using opensips for our project requirements.
I have a scenario where we need DNS-SRV query and the result of this should be 
placed as the desturi to send request out.(In case of forking call)
As far as I went through opensips documentation, there are some core parameters 
for dns related config such as 
"dns_retr_time"
 
,"dns_retr_time"
 
,"dns_servers_no"
 etc...
According to my understanding these config variables can be declared and used 
in the opensips.config file to control the settings of DNS query.

Is there any available function where I can use and pass the DNS server domain 
name, so that it fetches the IP address of the host ?
Please let us know what is the best way to achieve this?

Regards,
Agalya



___

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

[OpenSIPS-Users] web sockets (wss) error

[John Nash]

I downloaded opensips 2.2 stable tar file and upgraded my existing
opensips.cfg to use wss as per document
http://www.opensips.org/Documentation/Tutorials-WebSocket-2-2

but when I start opensips I get error "cannot handle protocol " right
at the line where I have listener wss:127.0.0.1:443

Do I need to clone current git version in order to test wss?
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] How to invok lookup() and get attr from the caller, without side effects?

[Rodrigo Pimenta Carvalho]

Hi Sevpal.


Yes. That is what I was doing. It worked very well.

But, nowadays I'm using db_mode = 0 for usrloc. So, the information is always 
only in RAM. In this case, the query will return no result. That is why I'm 
trying to read the attr column from table location, from RAM, and get specific 
information for the caller.


For the callee, everything is all right.


Regards.


RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979



De: users-boun...@lists.opensips.org  em nome 
de sevpal 
Enviado: terça-feira, 21 de junho de 2016 12:20
Para: OpenSIPS users mailling list
Assunto: Re: [OpenSIPS-Users] How to invok lookup() and get attr from the 
caller, without side effects?

Hi, have you tried/considered running a simple query on the database and 
parsing for the information you need?

From: Rodrigo Pimenta Carvalho
Sent: Tuesday, June 21, 2016 10:39 AM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] How to invok lookup() and get attr from the 
caller, without side effects?


Hi Răzvan.



I have tried that idea. But that didn't work. The SIP INVITE message is being 
changed by the OpenSIPS in a wrong way, in my point of view.

Do you know some way to save the entire SIP INVITE message before calling 
lookup() and then make the saved message take place after the lookup() 
execution?



My original message is:



INVITE sip:6...@mydomain.com.br SIP/2.0
Via: SIP/2.0/TCP 192.168.21.40:5090;rport;branch=z9hG4bK876727215
From: ;tag=179920819
To: 
Call-ID: 1410250893
CSeq: 21 INVITE
Contact: 
Proxy-Authorization: Digest username="crdphmacl_SPnuV5xqtnSX", 
realm="localhost", nonce="5769458c01cc263a7c0d6995dc48d42288ec6f8e4048", 
uri="sip:6...@mydomain.com.br", response="0f4c122d2a0a28dea6194c235cd77430", 
algorithm=MD5
Content-Type: application/sdp
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, 
INFO
Max-Forwards: 70
User-Agent: Linphone/3.6.1 (eXosip2/4.0.0)
Subject: Phone call
Content-Length:   227


This is being changed to:



INVITE 
sip:crdphmacl_SPnuV5xqtnSX@131.221.240.204:60672;transport=tcp;line=c6356a7d87d6f81
 SIP/2.0
Record-Route: 
Via: SIP/2.0/TCP myDomain.com.br:5060;branch=z9hG4bKe2db.49d54587.0;i=1
Via: SIP/2.0/TCP 
192.168.21.40:5090;received=xxx.yyy.240.204;rport=60672;branch=z9hG4bK716249970
From: ;tag=12586028
To: 
Call-ID: 1106771604
CSeq: 21 INVITE
Contact: 
Content-Type: application/sdp
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, 
INFO
Max-Forwards: 70
User-Agent: Linphone/3.6.1 (eXosip2/4.0.0)
Subject: Phone call
Content-Length:   224



So, the caller is receiving its own SIP INVITE.

That is why when A calls B, is A that rings, not B.



It is becoming a bit complicated. So, I suspect I'm going to the incorrect 
direction



Best regards.



RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979



De: users-boun...@lists.opensips.org  em nome 
de Răzvan Crainea 
Enviado: terça-feira, 21 de junho de 2016 04:24
Para: users@lists.opensips.org
Assunto: Re: [OpenSIPS-Users] How to invok lookup() and get attr from the 
caller, without side effects?


Hi, Rodrigo!


Have you tried restoring the R-URI after the caller lookup? Something like:


$var(ru) = $ru;

lookup("location", "", "$fu"); # this takes the caller from FROM uri, which I 
think is more suitable than from contact uri

$ru = $var(ru);

# continue your processing here



# now do the real lookup for the callee

lookup("location");


Don't do the lookups in the reversed way, because you might loose some contacts.


Best regards,

Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com

Home - OpenSIPS Solutions
www.opensips-solutions.com
OpenSIPS is a mature Open Source implementation of a SIP server. OpenSIPS is 
more than a SIP proxy/router as it includes application-level functionalities.

On 06/20/2016 09:02 PM, Rodrigo Pimenta Carvalho wrote:

Dear OpenSIPS-users,



The table location has the column attr where I use to store specific additional 
information for each registration.

Whenever A calls B, I have to read this specific information from the A record 
and from the B record. That is, I need to get and handle specific information 
about the caller and callee.



For the callee, I use to invoke the lookup("location") function that put 

Re: [OpenSIPS-Users] How to invok lookup() and get attr from the caller, without side effects?

[sevpal]

Hi, have you tried/considered running a simple query on the database and 
parsing for the information you need?

From: Rodrigo Pimenta Carvalho 
Sent: Tuesday, June 21, 2016 10:39 AM
To: OpenSIPS users mailling list 
Subject: Re: [OpenSIPS-Users] How to invok lookup() and get attr from the 
caller, without side effects?

Hi Răzvan.



I have tried that idea. But that didn't work. The SIP INVITE message is being 
changed by the OpenSIPS in a wrong way, in my point of view.

Do you know some way to save the entire SIP INVITE message before calling 
lookup() and then make the saved message take place after the lookup() 
execution?



My original message is:



INVITE sip:6...@mydomain.com.br SIP/2.0
Via: SIP/2.0/TCP 192.168.21.40:5090;rport;branch=z9hG4bK876727215
From: ;tag=179920819
To: 
Call-ID: 1410250893
CSeq: 21 INVITE
Contact: 
Proxy-Authorization: Digest username="crdphmacl_SPnuV5xqtnSX", 
realm="localhost", nonce="5769458c01cc263a7c0d6995dc48d42288ec6f8e4048", 
uri="sip:6...@mydomain.com.br", response="0f4c122d2a0a28dea6194c235cd77430", 
algorithm=MD5
Content-Type: application/sdp
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, 
INFO
Max-Forwards: 70
User-Agent: Linphone/3.6.1 (eXosip2/4.0.0)
Subject: Phone call
Content-Length:   227


This is being changed to:



INVITE 
sip:crdphmacl_SPnuV5xqtnSX@131.221.240.204:60672;transport=tcp;line=c6356a7d87d6f81
 SIP/2.0
Record-Route: 
Via: SIP/2.0/TCP myDomain.com.br:5060;branch=z9hG4bKe2db.49d54587.0;i=1
Via: SIP/2.0/TCP 
192.168.21.40:5090;received=xxx.yyy.240.204;rport=60672;branch=z9hG4bK716249970
From: ;tag=12586028
To: 
Call-ID: 1106771604
CSeq: 21 INVITE
Contact: 
Content-Type: application/sdp
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, 
INFO
Max-Forwards: 70
User-Agent: Linphone/3.6.1 (eXosip2/4.0.0)
Subject: Phone call
Content-Length:   224


So, the caller is receiving its own SIP INVITE.


That is why when A calls B, is A that rings, not B.



It is becoming a bit complicated. So, I suspect I'm going to the incorrect 
direction 




Best regards.



RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979







De: users-boun...@lists.opensips.org  em nome 
de Răzvan Crainea 
Enviado: terça-feira, 21 de junho de 2016 04:24
Para: users@lists.opensips.org
Assunto: Re: [OpenSIPS-Users] How to invok lookup() and get attr from the 
caller, without side effects? 

Hi, Rodrigo!




Have you tried restoring the R-URI after the caller lookup? Something like:




$var(ru) = $ru;

lookup("location", "", "$fu"); # this takes the caller from FROM uri, which I 
think is more suitable than from contact uri

$ru = $var(ru);

# continue your processing here




# now do the real lookup for the callee

lookup("location");




Don't do the lookups in the reversed way, because you might loose some contacts.




Best regards,

Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.comHome — OpenSIPS Solutions
  www.opensips-solutions.com
  OpenSIPS is a mature Open Source implementation of a SIP server. OpenSIPS 
is more than a SIP proxy/router as it includes application-level 
functionalities. 

On 06/20/2016 09:02 PM, Rodrigo Pimenta Carvalho wrote:

  Dear OpenSIPS-users,



  The table location has the column attr where I use to store specific 
additional information for each registration.

  Whenever A calls B, I have to read this specific information from the A 
record and from the B record. That is, I need to get and handle specific 
information about the caller and callee.



  For the callee, I use to invoke the lookup("location") function that put the 
needed information in the attr_avp. That is good and works very well. Then, I 
just have to read the attr_avp to get such specific information.



  For the caller, I use to invoke:



  $var(aorChamador) = $(ct.fields(uri));

  lookup("location","","$var(aorChamador)");




  However it causes amazing side effect in the SIP signaling. Ex: When A calls 
B, B stays quiet and A rings. So A can answer A. Crazy!

  According to the documentation, lookup will overwritten the Request-URI. I 
guess that is why the SIP signaling become incoherent.




  How could I get the caller attr specific information without side effects?




  Any hint will be very helpful!!




  Best regards.




  RODRIGO PIMENTA CARVALHO
  Inatel Competence Center
  Software
  Ph: +55 35 3471 9200 RAMAL 979


   

___
Users 

Re: [OpenSIPS-Users] How to invok lookup() and get attr from the caller, without side effects?

[Rodrigo Pimenta Carvalho]

Hi Răzvan.


I have tried that idea. But that didn't work. The SIP INVITE message is being 
changed by the OpenSIPS in a wrong way, in my point of view.

Do you know some way to save the entire SIP INVITE message before calling 
lookup() and then make the saved message take place after the lookup() 
execution?


My original message is:


INVITE sip:6...@mydomain.com.br SIP/2.0
Via: SIP/2.0/TCP 192.168.21.40:5090;rport;branch=z9hG4bK876727215
From: ;tag=179920819
To: 
Call-ID: 1410250893
CSeq: 21 INVITE
Contact: 
Proxy-Authorization: Digest username="crdphmacl_SPnuV5xqtnSX", 
realm="localhost", nonce="5769458c01cc263a7c0d6995dc48d42288ec6f8e4048", 
uri="sip:6...@mydomain.com.br", response="0f4c122d2a0a28dea6194c235cd77430", 
algorithm=MD5
Content-Type: application/sdp
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, 
INFO
Max-Forwards: 70
User-Agent: Linphone/3.6.1 (eXosip2/4.0.0)
Subject: Phone call
Content-Length:   227


This is being changed to:


INVITE 
sip:crdphmacl_SPnuV5xqtnSX@131.221.240.204:60672;transport=tcp;line=c6356a7d87d6f81
 SIP/2.0
Record-Route: 
Via: SIP/2.0/TCP myDomain.com.br:5060;branch=z9hG4bKe2db.49d54587.0;i=1
Via: SIP/2.0/TCP 
192.168.21.40:5090;received=xxx.yyy.240.204;rport=60672;branch=z9hG4bK716249970
From: ;tag=12586028
To: 
Call-ID: 1106771604
CSeq: 21 INVITE
Contact: 
Content-Type: application/sdp
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, 
INFO
Max-Forwards: 70
User-Agent: Linphone/3.6.1 (eXosip2/4.0.0)
Subject: Phone call
Content-Length:   224


So, the caller is receiving its own SIP INVITE.

That is why when A calls B, is A that rings, not B.


It is becoming a bit complicated. So, I suspect I'm going to the incorrect 
direction


Best regards.


RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979



De: users-boun...@lists.opensips.org  em nome 
de Răzvan Crainea 
Enviado: terça-feira, 21 de junho de 2016 04:24
Para: users@lists.opensips.org
Assunto: Re: [OpenSIPS-Users] How to invok lookup() and get attr from the 
caller, without side effects?


Hi, Rodrigo!


Have you tried restoring the R-URI after the caller lookup? Something like:


$var(ru) = $ru;

lookup("location", "", "$fu"); # this takes the caller from FROM uri, which I 
think is more suitable than from contact uri

$ru = $var(ru);

# continue your processing here



# now do the real lookup for the callee

lookup("location");


Don't do the lookups in the reversed way, because you might loose some contacts.


Best regards,

Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com

Home — OpenSIPS Solutions
www.opensips-solutions.com
OpenSIPS is a mature Open Source implementation of a SIP server. OpenSIPS is 
more than a SIP proxy/router as it includes application-level functionalities.

On 06/20/2016 09:02 PM, Rodrigo Pimenta Carvalho wrote:

Dear OpenSIPS-users,


The table location has the column attr where I use to store specific additional 
information for each registration.

Whenever A calls B, I have to read this specific information from the A record 
and from the B record. That is, I need to get and handle specific information 
about the caller and callee.


For the callee, I use to invoke the lookup("location") function that put the 
needed information in the attr_avp. That is good and works very well. Then, I 
just have to read the attr_avp to get such specific information.


For the caller, I use to invoke:


$var(aorChamador) = $(ct.fields(uri));

lookup("location","","$var(aorChamador)");


However it causes amazing side effect in the SIP signaling. Ex: When A calls B, 
B stays quiet and A rings. So A can answer A. Crazy!

According to the documentation, lookup will overwritten the Request-URI. I 
guess that is why the SIP signaling become incoherent.


How could I get the caller attr specific information without side effects?


Any hint will be very helpful!!


Best regards.


RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979



___
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

Re: [OpenSIPS-Users] How do I get the latest commit in the yum repo.

[Jim DeVito]

Super good. Thanks!!

---
Jim DeVito
Mobile 440.941.3860

On 2016-06-21 07:04, Bogdan-Andrei Iancu wrote:

Hi Jim,

 Currently we generate packages based on GIT tags only. So far there
is no TAG covering this commit - the commit is on 1st of June, while
2.2.0 was released on 27th of May.

 Still, you can use the nightly build RPMs:
  yum install
http://yum.opensips.org/2.2/nightly/el/7/x86_64/opensips-yum-nightly-2.2-3.el7.noarch.rpm
[3]

 Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com [2]

On 21.06.2016 16:21, Jim DeVito wrote:


Hi All,

Just submitted https://github.com/OpenSIPS/opensips/issues/914 [1]
and am planning to close it because it looks like the problem has
been fixed in a commit from 20 days ago. My question is how I get
the fix in the YUM release package with out running a nightly
release in production.

Thanks!!




Links:
--
[1] https://github.com/OpenSIPS/opensips/issues/914
[2] http://www.opensips-solutions.com
[3]
http://yum.opensips.org/2.2/nightly/el/7/x86_64/opensips-yum-nightly-2.2-3.el7.noarch.rpm


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

Re: [OpenSIPS-Users] How do I get the latest commit in the yum repo.

[Bogdan-Andrei Iancu]

Hi Jim,

Currently we generate packages based on GIT tags only. So far there is 
no TAG covering this commit - the commit is on 1st of June, while 2.2.0 
was released on 27th of May.


Still, you can use the nightly build RPMs:
|yum install 
http://yum.opensips.org/2.2/nightly/el/7/x86_64/opensips-yum-nightly-2.2-3.el7.noarch.rpm


Regards,
|

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 21.06.2016 16:21, Jim DeVito wrote:

Hi All,

Just submitted https://github.com/OpenSIPS/opensips/issues/914 and am 
planning to close it because it looks like the problem has been fixed 
in a commit from 20 days ago. My question is how I get the fix in the 
YUM release package with out running a nightly release in production.


Thanks!!



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

Re: [OpenSIPS-Users] TM flag 0x01 not working in 1.11.7

[Bogdan-Andrei Iancu]

Yes, or grab the 1.11 GIT branch

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 21.06.2016 16:46, Newlin, Ben wrote:

Re: [OpenSIPS-Users] TM flag 0x01 not working in 1.11.7

Bogdan,

Thanks. I assume the parameter will be available in 1.11.9?

Ben Newlin

*From: *Bogdan-Andrei Iancu 
*Date: *Tuesday, June 21, 2016 at 9:42 AM
*To: *OpenSIPS users mailling list , 
"Newlin, Ben" , "Newlin, Ben" 

*Subject: *Re: [OpenSIPS-Users] TM flag 0x01 not working in 1.11.7

Hi Ben,

My bad, I forgot to backport this param to 1.11 too (while the change 
affecting the 100 Trying generation was backported).


http://www.opensips.org/html/docs/modules/1.11.x/tm.html#id294521

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 21.06.2016 15:14, Newlin, Ben wrote:

Bogdan,

That is great for versions 2.1+, but I was asking about 1.11. This
sounds like a new feature to me and I don’t understand why it was
ported back to break existing functionality. The new TM parameter
you mention is not available in 1.11 according to the
documentation, so I have no way to get the previous functionality.
There is also no documentation in 1.11 or in the release notes
that indicates that the 0x01 flag no longer works.

Ben Newlin

*From: *Bogdan-Andrei Iancu 

*Date: *Tuesday, June 21, 2016 at 3:35 AM
*To: *OpenSIPS users mailling list 
, "Newlin, Ben"
 ,
"denis7...@mail.ru  >> Denis Putyato"
 
*Subject: *Re: [OpenSIPS-Users] TM flag 0x01 not working in 1.11.7

Hi

Please refer to https://github.com/OpenSIPS/opensips/issues/833 .

As per documentation, the 0x1 flag became obsolete, as sending the
100Trying is no longer linked to the t_relay() - the 100Trying is
now sent when the transaction is created.

If you want to disable the auto 100Trying, see the new TM flag
auto_100trying:
http://www.opensips.org/html/docs/modules/2.2.x/tm.html#id294523

Regards,



Bogdan-Andrei Iancu

OpenSIPS Founder and Developer

http://www.opensips-solutions.com

On 18.06.2016 16:49, Denis wrote:

Hello!

2.1 has the same problem

//mailto:denis7...@mail.ru




In my script I send all “100 Trying” responses manually* and
use the 0x01 flag when calling t_relay so that it will not
send its own “100 Giving a try” response, as per the
documentation [1].

Unfortunately, in 1.11.7 and 1.11.8 this flag seems to have
stopped working which results in multiple 100 responses.

This should be easily reproducible with a script that simply
calls t_relay with flag 0x01. You will see that a “100 Giving
a try” response is still sent.

[1]
http://www.opensips.org/html/docs/modules/1.11.x/tm.html#id294528

* I do this because I don’t like that the default is to send
the non-standard response “100 Giving a try” instead of “100
Trying”. I’ve always wondered why this is that way.
Additionally, the automatic response to CANCEL requests is
“200 canceling” instead of the standard “200 OK”.
Unfortunately, I have yet to find a way to workaround the
behavior for CANCEL.

Ben Newlin





___

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



___
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

Re: [OpenSIPS-Users] TM flag 0x01 not working in 1.11.7

[Newlin, Ben]

Bogdan,

Thanks. I assume the parameter will be available in 1.11.9?

Ben Newlin

From: Bogdan-Andrei Iancu 
Date: Tuesday, June 21, 2016 at 9:42 AM
To: OpenSIPS users mailling list , "Newlin, Ben" 
, "Newlin, Ben" 
Subject: Re: [OpenSIPS-Users] TM flag 0x01 not working in 1.11.7

Hi Ben,

My bad, I forgot to backport this param to 1.11 too (while the change affecting 
the 100 Trying generation was backported).

http://www.opensips.org/html/docs/modules/1.11.x/tm.html#id294521

Regards,


Bogdan-Andrei Iancu

OpenSIPS Founder and Developer

http://www.opensips-solutions.com
On 21.06.2016 15:14, Newlin, Ben wrote:
Bogdan,

That is great for versions 2.1+, but I was asking about 1.11. This sounds like 
a new feature to me and I don’t understand why it was ported back to break 
existing functionality. The new TM parameter you mention is not available in 
1.11 according to the documentation, so I have no way to get the previous 
functionality. There is also no documentation in 1.11 or in the release notes 
that indicates that the 0x01 flag no longer works.


Ben Newlin

From: Bogdan-Andrei Iancu 
Date: Tuesday, June 21, 2016 at 3:35 AM
To: OpenSIPS users mailling list 
, "Newlin, Ben" 
, 
"denis7...@mail.ru >> Denis Putyato" 

Subject: Re: [OpenSIPS-Users] TM flag 0x01 not working in 1.11.7

Hi

Please refer to https://github.com/OpenSIPS/opensips/issues/833 .

As per documentation, the 0x1 flag became obsolete, as sending the 100Trying is 
no longer linked to the t_relay() - the 100Trying is now sent when the 
transaction is created.

If you want to disable the auto 100Trying, see the new TM flag auto_100trying:
 http://www.opensips.org/html/docs/modules/2.2.x/tm.html#id294523

Regards,




Bogdan-Andrei Iancu

OpenSIPS Founder and Developer

http://www.opensips-solutions.com
On 18.06.2016 16:49, Denis wrote:
Hello!

2.1 has the same problem

  mailto:denis7...@mail.ru

In my script I send all “100 Trying” responses manually* and use the 0x01 flag 
when calling t_relay so that it will not send its own “100 Giving a try” 
response, as per the documentation [1].

Unfortunately, in 1.11.7 and 1.11.8 this flag seems to have stopped working 
which results in multiple 100 responses.

This should be easily reproducible with a script that simply calls t_relay with 
flag 0x01. You will see that a “100 Giving a try” response is still sent.

[1] http://www.opensips.org/html/docs/modules/1.11.x/tm.html#id294528

* I do this because I don’t like that the default is to send the non-standard 
response “100 Giving a try” instead of “100 Trying”. I’ve always wondered why 
this is that way. Additionally, the automatic response to CANCEL requests is 
“200 canceling” instead of the standard “200 OK”. Unfortunately, I have yet to 
find a way to workaround the behavior for CANCEL.

Ben Newlin






___

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

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

Re: [OpenSIPS-Users] TM flag 0x01 not working in 1.11.7

[Bogdan-Andrei Iancu]

Hi Ben,

My bad, I forgot to backport this param to 1.11 too (while the change 
affecting the 100 Trying generation was backported).


http://www.opensips.org/html/docs/modules/1.11.x/tm.html#id294521

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 21.06.2016 15:14, Newlin, Ben wrote:

Re: [OpenSIPS-Users] TM flag 0x01 not working in 1.11.7

Bogdan,

That is great for versions 2.1+, but I was asking about 1.11. This 
sounds like a new feature to me and I don’t understand why it was 
ported back to break existing functionality. The new TM parameter you 
mention is not available in 1.11 according to the documentation, so I 
have no way to get the previous functionality. There is also no 
documentation in 1.11 or in the release notes that indicates that the 
0x01 flag no longer works.


Ben Newlin

*From: *Bogdan-Andrei Iancu 
*Date: *Tuesday, June 21, 2016 at 3:35 AM
*To: *OpenSIPS users mailling list , 
"Newlin, Ben" , "denis7...@mail.ru >> Denis 
Putyato" 

*Subject: *Re: [OpenSIPS-Users] TM flag 0x01 not working in 1.11.7

Hi

Please refer to https://github.com/OpenSIPS/opensips/issues/833 .

As per documentation, the 0x1 flag became obsolete, as sending the 
100Trying is no longer linked to the t_relay() - the 100Trying is now 
sent when the transaction is created.


If you want to disable the auto 100Trying, see the new TM flag 
auto_100trying:

http://www.opensips.org/html/docs/modules/2.2.x/tm.html#id294523

Regards,


Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 18.06.2016 16:49, Denis wrote:

Hello!

2.1 has the same problem

//mailto:denis7...@mail.ru




In my script I send all “100 Trying” responses manually* and use
the 0x01 flag when calling t_relay so that it will not send its
own “100 Giving a try” response, as per the documentation [1].

Unfortunately, in 1.11.7 and 1.11.8 this flag seems to have
stopped working which results in multiple 100 responses.

This should be easily reproducible with a script that simply calls
t_relay with flag 0x01. You will see that a “100 Giving a try”
response is still sent.

[1] http://www.opensips.org/html/docs/modules/1.11.x/tm.html#id294528

* I do this because I don’t like that the default is to send the
non-standard response “100 Giving a try” instead of “100 Trying”.
I’ve always wondered why this is that way. Additionally, the
automatic response to CANCEL requests is “200 canceling” instead
of the standard “200 OK”. Unfortunately, I have yet to find a way
to workaround the behavior for CANCEL.

Ben Newlin




___

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


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

[OpenSIPS-Users] How do I get the latest commit in the yum repo.

[Jim DeVito]

Hi All,

Just submitted https://github.com/OpenSIPS/opensips/issues/914 and am 
planning to close it because it looks like the problem has been fixed in 
a commit from 20 days ago. My question is how I get the fix in the YUM 
release package with out running a nightly release in production.


Thanks!!

--
Jim DeVito

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

Re: [OpenSIPS-Users] How to invok lookup() and get attr from the caller, without side effects?

[Rodrigo Pimenta Carvalho]

Hi Razvan Crainea.


I didn't know about this possibility.


I will try this idea now.


Thank you very much!!


Best regards.



RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979



De: users-boun...@lists.opensips.org  em nome 
de Razvan Crainea 
Enviado: terça-feira, 21 de junho de 2016 04:24
Para: users@lists.opensips.org
Assunto: Re: [OpenSIPS-Users] How to invok lookup() and get attr from the 
caller, without side effects?


Hi, Rodrigo!


Have you tried restoring the R-URI after the caller lookup? Something like:


$var(ru) = $ru;

lookup("location", "", "$fu"); # this takes the caller from FROM uri, which I 
think is more suitable than from contact uri

$ru = $var(ru);

# continue your processing here



# now do the real lookup for the callee

lookup("location");


Don't do the lookups in the reversed way, because you might loose some contacts.


Best regards,

Razvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com

On 06/20/2016 09:02 PM, Rodrigo Pimenta Carvalho wrote:

Dear OpenSIPS-users,


The table location has the column attr where I use to store specific additional 
information for each registration.

Whenever A calls B, I have to read this specific information from the A record 
and from the B record. That is, I need to get and handle specific information 
about the caller and callee.


For the callee, I use to invoke the lookup("location") function that put the 
needed information in the attr_avp. That is good and works very well. Then, I 
just have to read the attr_avp to get such specific information.


For the caller, I use to invoke:


$var(aorChamador) = $(ct.fields(uri));

lookup("location","","$var(aorChamador)");


However it causes amazing side effect in the SIP signaling. Ex: When A calls B, 
B stays quiet and A rings. So A can answer A. Crazy!

According to the documentation, lookup will overwritten the Request-URI. I 
guess that is why the SIP signaling become incoherent.


How could I get the caller attr specific information without side effects?


Any hint will be very helpful!!


Best regards.


RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979



___
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

Re: [OpenSIPS-Users] Some calls fail with tcp_connect_blocking error on version 2.1.1

[Owais Ahmad]

Hi
​
Răzvan,

I have tested with your recommendations. Here are my findings:

i) Disabling tcp_accept_aliases increases frequency of this error.
ii) Upgrading to 2.1.3 does not resolve the issue. On this new version, I
also tested tcp_accept_aliases=0/1. I get the exact same error I posted
previously.

Is there a way we can control the number of retries opensips should make if
there is a UAC connect failure? Saying that because I know for sure that B
party has a working network connection.

I have configured the following, in case it helps you point me to the
actual issue:

maxbuffer=65536
tcp_children=100
tcp_accept_aliases=0
tcp_connect_timeout=3
tcp_keepalive=0
modparam("proto_tls", "tls_max_msg_chunks", 8)
modparam("proto_tls", "tls_handshake_timeout", 3)
modparam("proto_tls", "tls_send_timeout", 7)

Regards,
Owais

On Tue, Jun 21, 2016 at 1:16 PM, Răzvan Crainea  wrote:

> Hi, Owais!
>
> You should consider upgrade OpenSIPS to a newer version (at least 2.1.3),
> because there were some fixes done related to this issue. If you cannot
> upgrade right now, try to disable the auto TCP aliasing[1] (set it to 0).
>
> [1] http://www.opensips.org/Documentation/Script-CoreParameters-2-1#toc90
>
> Best regards,
>
> ​​
> Răzvan Crainea
> OpenSIPS Solutionswww.opensips-solutions.com
>
>
>
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] Some calls fail with tcp_connect_blocking error on version 2.1.1

[Răzvan Crainea]

Hi, Owais!

You should consider upgrade OpenSIPS to a newer version (at least 
2.1.3), because there were some fixes done related to this issue. If you 
cannot upgrade right now, try to disable the auto TCP aliasing[1] (set 
it to 0).


[1] http://www.opensips.org/Documentation/Script-CoreParameters-2-1#toc90

Best regards,

Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com

On 06/21/2016 10:05 AM, Owais Ahmad wrote:
I am getting the following error on opensips version 2.1.1. The 
scenario is that I have a load balancer listening on TLS and it 
dispatches requests to backend registrar servers listening on UDP.
90% of the calls are successful, but some calls fail and the load 
balancer throws the following error when INVITE is being relayed to B 
party.


DBG:core:parse_headers: flags=2000
DBG:core:check_ip_address: params 10.0.0.7, 10.0.0.7, 0
DBG:core:tcp_conn_get: 0  port 56415
DBG:core:print_ip: tcpconn_find: ip 2.4.6.8
DBG:proto_tls:proto_tls_send: no open tcp connection found, opening 
new one

DBG:core:probe_max_sock_buff: getsockopt: snd is initially 16384
DBG:core:probe_max_sock_buff: trying : 32768
DBG:core:probe_max_sock_buff: setting snd: set=32768,verify=65536
DBG:core:probe_max_sock_buff: trying : 65536
DBG:core:probe_max_sock_buff: setting snd: set=65536,verify=131072
INFO:core:probe_max_sock_buff: using snd buffer of 128 kb
INFO:core:init_sock_keepalive: TCP keepalive enabled on socket 6
ERROR:core:tcp_connect_blocking: timeout 2132 ms elapsed from 3000 s
ERROR:proto_tls:tls_sync_connect: tcp_blocking_connect failed
ERROR:proto_tls:proto_tls_send: connect failed
ERROR:tm:msg_send: send() for proto 3 failed
ERROR:tm:t_forward_nonack: sending request failed
DBG:tm:t_relay_to: t_forward_nonack returned error
DBG:core:parse_headers: flags=
DBG:core:check_ip_address: params 10.0.0.7, 10.0.0.7, 0
DBG:tm:reset_timer: (group 3, tl=0x7fcb6a490970)
DBG:tm:reset_timer: (group 0, tl=0x7fcb6a4909a0)
DBG:tm:cleanup_uac_timers: RETR/FR timers reset
DBG:tm:set_timer: relative timeout is 50
DBG:tm:insert_timer_unsafe: [4]: 0x7fcb6a490898 (60930)
DBG:tm:insert_timer_unsafe: [0]: 0x7fcb6a4908c8 (618)


Any hints as to what might be going wrong?

Regards,
Owais




___
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

Re: [OpenSIPS-Users] DNS-SRV query in opensips

[Bogdan-Andrei Iancu]

Hi Agalya,

OpenSIPS does full flavor DNS lookup (with NATPR and SRV), but this is 
internal, and not accessible from script. OpenSIPS implements auto 
DNS-based failover :

http://www.opensips.org/html/docs/modules/2.2.x/tm.html#id293694

My understanding is you want the DNS resolving to be done at script 
level and to have access to the results ?


Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 20.06.2016 20:35, Ramachandran, Agalya (Contractor) wrote:


Hi team,

We are using opensips for our project requirements.

I have a scenario where we need DNS-SRV query and the result of this 
should be placed as the desturi to send request out.(In case of 
forking call)


As far as I went through opensips documentation, there are some core 
parameters for dns related config such as “dns_retr_time 
” 
,”dns_retr_time 
” ,”dns_servers_no 
” 
etc…


According to my understanding these config variables can be declared 
and used in the opensips.config file to control the settings of DNS query.


Is there any available function where I can use and pass the DNS 
server domain name, so that it fetches the IP address of the host ?


Please let us know what is the best way to achieve this?

Regards,

Agalya



___
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

Re: [OpenSIPS-Users] Async event_route and cachdb_redis

[Răzvan Crainea]

Hi, Sammy!

Could you try this patch:

https://gist.github.com/razvancrainea/9d239c82474bb0f1c403b6459dbdb647

Thanks,

Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com

On 06/19/2016 08:56 PM, SamyGo wrote:

Hi,
I'm seeing errors from cachedb_redis module when called in an event 
route in async mode.


event_route[E_UL_CONTACT_INSERT,async] {
...
cache_raw_query("redis:group1","SET ABC");
..

}

OpenSIPS throws error stating that redis group1 unavailable

DBG:core:cachedb_raw_query: from script [redis] - with grp [group1]
ERROR:core:cachedb_raw_query: failed to get connection for grp name 
[group1]


I tried same command in main route of reply route, all works normal. 
if I remove the "async" from the event_route definition it works in 
event route.


Any logical reason why async route don't recognize the connections ?

Tried with OpenSIPS 2.2 and 2.1 as well, same behavior.


Regards,
Sammy



___
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

Re: [OpenSIPS-Users] TM flag 0x01 not working in 1.11.7

[Bogdan-Andrei Iancu]

Hi

Please refer to https://github.com/OpenSIPS/opensips/issues/833 .

As per documentation, the 0x1 flag became obsolete, as sending the 
100Trying is no longer linked to the t_relay() - the 100Trying is now 
sent when the transaction is created.


If you want to disable the auto 100Trying, see the new TM flag 
auto_100trying:

http://www.opensips.org/html/docs/modules/2.2.x/tm.html#id294523

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 18.06.2016 16:49, Denis wrote:

Re: [OpenSIPS-Users] TM flag 0x01 not working in 1.11.7 Hello!

2.1 has the same problem

//mailto:denis7...@mail.ru


	In my script I send all “100 Trying” responses manually* and use the 
0x01 flag when calling t_relay so that it will not send its own “100 
Giving a try” response, as per the documentation [1].


Unfortunately, in 1.11.7 and 1.11.8 this flag seems to have stopped 
working which results in multiple 100 responses.


This should be easily reproducible with a script that simply calls 
t_relay with flag 0x01. You will see that a “100 Giving a try” 
response is still sent.


[1] http://www.opensips.org/html/docs/modules/1.11.x/tm.html#id294528

* I do this because I don’t like that the default is to send the 
non-standard response “100 Giving a try” instead of “100 Trying”. I’ve 
always wondered why this is that way. Additionally, the automatic 
response to CANCEL requests is “200 canceling” instead of the standard 
“200 OK”. Unfortunately, I have yet to find a way to workaround the 
behavior for CANCEL.


Ben Newlin



___
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

[OpenSIPS-Users] Some calls fail with tcp_connect_blocking error on version 2.1.1

[Owais Ahmad]

I am getting the following error on opensips version 2.1.1. The scenario is
that I have a load balancer listening on TLS and it dispatches requests to
backend registrar servers listening on UDP.
90% of the calls are successful, but some calls fail and the load balancer
throws the following error when INVITE is being relayed to B party.

DBG:core:parse_headers: flags=2000
DBG:core:check_ip_address: params 10.0.0.7, 10.0.0.7, 0
DBG:core:tcp_conn_get: 0  port 56415
DBG:core:print_ip: tcpconn_find: ip 2.4.6.8
DBG:proto_tls:proto_tls_send: no open tcp connection found, opening new one
DBG:core:probe_max_sock_buff: getsockopt: snd is initially 16384
DBG:core:probe_max_sock_buff: trying : 32768
DBG:core:probe_max_sock_buff: setting snd: set=32768,verify=65536
DBG:core:probe_max_sock_buff: trying : 65536
DBG:core:probe_max_sock_buff: setting snd: set=65536,verify=131072
INFO:core:probe_max_sock_buff: using snd buffer of 128 kb
INFO:core:init_sock_keepalive: TCP keepalive enabled on socket 6
ERROR:core:tcp_connect_blocking: timeout 2132 ms elapsed from 3000 s
ERROR:proto_tls:tls_sync_connect: tcp_blocking_connect failed
ERROR:proto_tls:proto_tls_send: connect failed
ERROR:tm:msg_send: send() for proto 3 failed
ERROR:tm:t_forward_nonack: sending request failed
DBG:tm:t_relay_to: t_forward_nonack returned error
DBG:core:parse_headers: flags=
DBG:core:check_ip_address: params 10.0.0.7, 10.0.0.7, 0
DBG:tm:reset_timer: (group 3, tl=0x7fcb6a490970)
DBG:tm:reset_timer: (group 0, tl=0x7fcb6a4909a0)
DBG:tm:cleanup_uac_timers: RETR/FR timers reset
DBG:tm:set_timer: relative timeout is 50
DBG:tm:insert_timer_unsafe: [4]: 0x7fcb6a490898 (60930)
DBG:tm:insert_timer_unsafe: [0]: 0x7fcb6a4908c8 (618)


Any hints as to what might be going wrong?

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