Re: [OpenSIPS-Users] Iterate headers

[Bogdan-Andrei Iancu]

Hi Nick,

no rush as you are heading towards a "no" response - right now there is 
no way to iterate through all the headers in a msg. You can iterate 
through the all the headers with the same name.


Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
  http://www.opensips-solutions.com
OpenSIPS Summit 2018
  http://www.opensips.org/events/Summit-2018Amsterdam

On 03/29/2018 04:48 PM, Nick Altmann wrote:

Nobody knows? OpenSIPS Team?

2018-03-28 22:50 GMT+03:00 Nick Altmann >:


Hi,

Is there any way to iterate all headers? For example to remove all
headers except list?

--
Nick




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Re: [OpenSIPS-Users] Iterate headers

[Nick Altmann]

Nobody knows? OpenSIPS Team?

2018-03-28 22:50 GMT+03:00 Nick Altmann :

> Hi,
>
> Is there any way to iterate all headers? For example to remove all headers
> except list?
>
> --
> Nick
>
>
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Re: [OpenSIPS-Users] rtpengine

[Nick Altmann]

It's ready for transcoding in 2.4.
http://www.opensips.org/html/docs/modules/2.4.x/rtpengine.html#idp5666752

2018-03-29 15:06 GMT+03:00 :

> Hello Everyone,
>
> Is rtpengine module will be updated with Transcoding and repacketization
> flags ?
>
> volga629
>
>
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[OpenSIPS-Users] rtpengine

[volga629]

Hello Everyone,

Is rtpengine module will be updated with Transcoding and 
repacketization flags ?


volga629


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Re: [OpenSIPS-Users] How to extract SIP-I ISUP 'Redirecting number' parameter

[Eugene Prokopiev]

Btw, is it possible to extract ISUP part as is in binary form and read
by some external script? Can anybody advice me any library to parse
ISUP binary data?

2018-03-29 12:50 GMT+03:00 Eugene Prokopiev :
> Thank you!
>
> I submitted https://github.com/OpenSIPS/opensips/issues/1323 - maybe
> your patch can be applied?
>
> 2018-03-29 10:53 GMT+03:00 Arto Kuiri :
>> Hello,
>>
>> I had same problem and made my own patch. See attached diff, it adds support 
>> for:
>> - redirection info (reason in same format as in SIP diversion header, 
>> commented ITU Q.763 version)
>> - original called number
>> - redirection number
>>
>> Meant to make feature request/patch, but somehow forgot it.
>>
>> Best regards,
>> Arto Kuiri
>>
>>
>>
>>
>> Lähettäjä: Users  käyttäjän Vlad Patrascu 
>>  puolesta
>> Lähetetty: 28. maaliskuuta 2018 20:48
>> Vastaanottaja: users@lists.opensips.org
>> Aihe: Re: [OpenSIPS-Users] How to extract SIP-I ISUP 'Redirecting number' 
>> parameter
>>
>>
>> Hi,
>>
>> Unfortunately "Redirecting Number" is not currently supported, but you
>> could open up a feature request regarding this, as it would be not very
>> difficult to implement.
>>
>> Regards,
>>
>> Vlad Patrascu
>> OpenSIPS Developer
>> http://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 28.03.2018 18:32, Eugene Prokopiev wrote:
>>> Hi,
>>>
>>> I need to extract ISUP parameter 'Redirecting number' from SIP-I
>>> INVITE request and add it as SIP header with 'X-' prefix to process in
>>> FreeSWITCH. Can't found any about 'Redirecting number' in
>>> http://www.opensips.org/html/docs/modules/2.3.x/sip_i.html, so how to
>>> extract it? 'Redirecting number' parameter value is visible in
>>> Wireshark now as ISUP parameter with t=11 and l=7
>>>
>>
>>
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>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
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>>
>
>
>
> --
> WBR,
> Eugene Prokopiev



-- 
WBR,
Eugene Prokopiev

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Re: [OpenSIPS-Users] How to extract SIP-I ISUP 'Redirecting number' parameter

[Eugene Prokopiev]

Thank you!

I submitted https://github.com/OpenSIPS/opensips/issues/1323 - maybe
your patch can be applied?

2018-03-29 10:53 GMT+03:00 Arto Kuiri :
> Hello,
>
> I had same problem and made my own patch. See attached diff, it adds support 
> for:
> - redirection info (reason in same format as in SIP diversion header, 
> commented ITU Q.763 version)
> - original called number
> - redirection number
>
> Meant to make feature request/patch, but somehow forgot it.
>
> Best regards,
> Arto Kuiri
>
>
>
>
> Lähettäjä: Users  käyttäjän Vlad Patrascu 
>  puolesta
> Lähetetty: 28. maaliskuuta 2018 20:48
> Vastaanottaja: users@lists.opensips.org
> Aihe: Re: [OpenSIPS-Users] How to extract SIP-I ISUP 'Redirecting number' 
> parameter
>
>
> Hi,
>
> Unfortunately "Redirecting Number" is not currently supported, but you
> could open up a feature request regarding this, as it would be not very
> difficult to implement.
>
> Regards,
>
> Vlad Patrascu
> OpenSIPS Developer
> http://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 28.03.2018 18:32, Eugene Prokopiev wrote:
>> Hi,
>>
>> I need to extract ISUP parameter 'Redirecting number' from SIP-I
>> INVITE request and add it as SIP header with 'X-' prefix to process in
>> FreeSWITCH. Can't found any about 'Redirecting number' in
>> http://www.opensips.org/html/docs/modules/2.3.x/sip_i.html, so how to
>> extract it? 'Redirecting number' parameter value is visible in
>> Wireshark now as ISUP parameter with t=11 and l=7
>>
>
>
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>
>
> ___
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>



-- 
WBR,
Eugene Prokopiev

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Re: [OpenSIPS-Users] RTPEngine equivalent to rtpproxy_sockets table

[Pat Burke]

Thanks Razvan.  We are looking forward to the release of 2.4.

 

Thanks again,

Pat Burke

 

From: Users  On Behalf Of Bogdan-Andrei Iancu
Sent: Wednesday, March 28, 2018 11:43 AM
To: OpenSIPS users mailling list ; Pat Burke 

Subject: Re: [OpenSIPS-Users] RTPEngine equivalent to rtpproxy_sockets table

 

Thanks to Razvan, there is now DB support for provisioning rtpengine (in 
opensips 2.4):

https://github.com/OpenSIPS/opensips/commit/094c850dd91286ff75bf3463dafbf83a398aaf8f

Regards,



Bogdan-Andrei Iancu
 
OpenSIPS Founder and Developer
  http://www.opensips-solutions.com
OpenSIPS Summit 2018
  http://www.opensips.org/events/Summit-2018Amsterdam

On 03/09/2018 04:26 PM, Pat Burke wrote:

Răzvan,

 

What is the process of contributing the change?  I will look at copying the 
rtpproxy code.

 

Thanks, 

Pat Burke 

 

 

 

 

 

 

 Original message 

From: Răzvan Crainea    

Date: 3/9/18 2:59 AM (GMT-06:00) 

To: users@lists.opensips.org   

Subject: Re: [OpenSIPS-Users] RTPEngine equivalent to rtpproxy_sockets table 

 

Hi, Pat!
 
There currently is no way to dynamically adjust the rtpengine sets over 
the database. There's already a feature request for this[1], but there's 
not yet implemented. You can track that ticket to keep up with the 
progress for this.
 
[1] https://github.com/OpenSIPS/opensips/issues/1092
 
Best regards,
Răzvan
 
On 03/08/2018 07:00 PM, Pat Burke wrote:
> We were looking at using RTPEngine for SRTP and wondering if there is a 
> table equivalent to RTProxy's rtpproxy_sockets?
> 
> 
> If not, is there a way to dynamically adjust the rtpengine sets?  With 
> RTPProxy we can update the rtpproxy_sockets table and then issue a 
> rtpproxy_reload command.
> 
> 
> Thanks for your input.
> 
> 
> Regards,
> *Pat Burke*
> 
> 
> 
> ___
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> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> 
 
-- 
Răzvan Crainea
OpenSIPS Core Developer
   http://www.opensips-solutions.com
OpenSIPS Summit 2018
   http://www.opensips.org/events/Summit-2018Amsterdam
 
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Re: [OpenSIPS-Users] How to terminate a call, via Opensips, before it being answered?

[Rodrigo Pimenta Carvalho]

Ok.


Thank you very much!


Best regards.


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



De: Users  em nome de Ben Newlin 

Enviado: terça-feira, 27 de março de 2018 17:10
Para: OpenSIPS users mailling list
Assunto: Re: [OpenSIPS-Users] How to terminate a call, via Opensips, before it 
being answered?


Yes, when the ACK is lost there will be retransmissions of the 200 OK. But if 
the ACK is being misrouted or the connectivity issue persists for too long then 
the ACK will never be received. Now the endpoint that did not receive the ACK 
*should* then send a BYE to disconnect. However, not all endpoints operate as 
they should at all times and we have seen this sometimes does not occur. Also, 
if the network connectivity issue affected both sides of the call, then the BYE 
will not be received either.



So you are right that the problem scenario requires both the ACK and BYE to be 
lost/misrouted/not sent. But as I said, it doesn’t happen often and even if it 
does many times the “stuck” calls cause no issues. But if billing or some other 
reporting/analytics are being done, the stuck calls can negatively affect those 
results.



The INVITE refresh mechanism is part of the Dialog module and can be enabled 
when the dialog is created [1].



[1] http://www.opensips.org/html/docs/modules/2.3.x/dialog.html#idp5828384

dialog Module - 
opensips.org
www.opensips.org
The dialog module provides dialog awareness to the OpenSIPS proxy. Its 
functionality is to keep trace of the current dialogs, to offer information 
about them (like ...




Thanks,

Ben Newlin



From: Users  on behalf of Rodrigo Pimenta 
Carvalho 
Reply-To: OpenSIPS users mailling list 
Date: Tuesday, March 27, 2018 at 1:55 PM
To: OpenSIPS users mailling list 
Subject: Re: [OpenSIPS-Users] How to terminate a call, via Opensips, before it 
being answered?



Hi Ben.



Thank you very much!

I didn't realized such problems, until you explain that.

I will check if my project will need the same procedure.

In that case, I will study about INVITE refreshes.

What I have observed in my OpenSIPS is that when a ACK is lost for a SIP OK, 
the callee sends SIP OK again and again.



Could you point the OpenSIPS web page (from OpenSIPS documentation) that 
explain about INVITE refresh, please?



Best regards.



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





De: Users  em nome de Ben Newlin 

Enviado: terça-feira, 27 de março de 2018 14:15
Para: OpenSIPS users mailling list
Assunto: Re: [OpenSIPS-Users] How to terminate a call, via Opensips, before it 
being answered?



You don’t have to read very far back in the mailing list archives to see that 
misrouted ACKs are a fairly common problem when implementing SIP proxies. ☺



Mishandling of the Record-Route headers is the common problem, but loss of 
connectivity with the far end server can occur as well. Because the INVITE 
transaction is completed, the TM timers will not catch this and the dialog will 
stay in the CONFIRMED but not ACKed state until the $DLG_timeout expires.



It doesn’t happen very often at all, but if it does and the timeout is set very 
high then you end up with a stuck call until the timer pops. If you are doing 
billing on the same endpoint then you potentially end up with a very long call 
being billed.



There are also other ways to accomplish similar safeguards as this, including 
OPTIONS or INVITE refreshes using the Dialog module. We are still running 1.11 
in production so the INVITE refreshes were not available to us and some of our 
partners do not accept OPTIONS refreshes. We plan to implement the INVITE 
refreshes once we have completed the upgrade to 2.X.



Thanks,

Ben Newlin



From: Users  on behalf of Rodrigo Pimenta 
Carvalho 
Reply-To: OpenSIPS users mailling list 
Date: Tuesday, March 27, 2018 at 12:57 PM
To: OpenSIPS users mailling list 
Subject: Re: [OpenSIPS-Users] How to terminate a call, via Opensips, before it 
being answered?



Hi.



Just as curiosity, what would  cause an ACK lost in your system?



Best regards.



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





De: Users  em nome de Ben Newlin 

Enviado: terça-feira, 27 de março de 2018 11:18
Para: OpenSIPS users mailling list
Assunto: Re: [OpenSIPS-Users] How to terminate a call, via Opensips, before it 
being answered?



Rodrigo,



Yes, they do. I am using them to do exactly what you describe. The final reply 
(fr) timer is how long a transaction will wait to receive a final reply 
(>=200). If the timer expires without receiving a final reply the transaction 
will be canceled and failure route will be triggered with, I think, a local 408 
response.



As for $DLG_timeout, you can set that

[OpenSIPS-Users] Send rtp on other server.

[Kavin Chauhan]

Hi all,
I am having some query in bifurcation of sip and rtp via opensips.

I have one provider which requires rtp on different server which ip address
he is sending me in INVITE.

My current configuration is sending rtp to same server which address is in
FROM of INVITE.
Is there any way to achieve this in opensips? Let me know if you want any
extra details.

Thanks in advance.
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Re: [OpenSIPS-Users] [OpenSIPS-Devel] [Release] OpenSIPS 2.4.0 major release, beta version

[Bogdan-Andrei Iancu]

Hi Maxim,

The fixes are welcome all the time ;)

Best regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
  http://www.opensips-solutions.com
OpenSIPS Summit 2018
  http://www.opensips.org/events/Summit-2018Amsterdam

On 03/29/2018 04:33 AM, Maxim Sobolev wrote:
Hice! I'll add new branch to my voiptests build. We have some patches 
to fix memory leak in the python module, I will post them soon. 
Hopefully they can make it into the release.


-Max

On Wed, Mar 28, 2018, 4:19 PM > wrote:


Great work !!!

volga629

On Wed, Mar 28, 2018 at 3:12 PM, Bogdan-Andrei Iancu
mailto:bog...@opensips.org>> wrote:
> Hi All !!
>
> I guess everybody knows the drill by now - it is March, so it's time
> for a new major OpenSIPS release. Almost 4 months ago we were
> announcing our ambitious roadmap for OpenSIPS 2.4 .
>
> Well, this has just come reality !!
>
> I’m happy to announce the beta release of the OpenSIPS 2.4.0 major
> version. Curios to find out more about this release? See the
> philosophy behind this release by reading the overview of OpenSIPS
> 2.4.o, code name The Cluster Maker.
>
> And keep in mind that 2.4 is still a beta release, targeting 30th of
> April to become fully stable. So, we have one month of testing ahead
> of us :).
>
> Many thanks to our awesome community for contributing with ideas,
> code, patches, tests and reports!
>
> With the occasion of the OpenSIPS Summit 2018, the stable OpenSIPS
> 2.4 will be our star, as we will present and demo all its
> capabilities!
>
> Looking for downloading it? See the tarball or the GIT repo.
Packages
> will be available soon.
>
> Enjoy it !!
>
> --
> Bogdan-Andrei Iancu
>
> OpenSIPS Founder and Developer
> http://www.opensips-solutions.com
> OpenSIPS Summit 2018
> http://www.opensips.org/events/Summit-2018Amsterdam


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Re: [OpenSIPS-Users] How to extract SIP-I ISUP 'Redirecting number' parameter

[Arto Kuiri]

Hello,

I had same problem and made my own patch. See attached diff, it adds support 
for:
- redirection info (reason in same format as in SIP diversion header, commented 
ITU Q.763 version)
- original called number
- redirection number

Meant to make feature request/patch, but somehow forgot it.

Best regards,
Arto Kuiri




Lähettäjä: Users  käyttäjän Vlad Patrascu 
 puolesta
Lähetetty: 28. maaliskuuta 2018 20:48
Vastaanottaja: users@lists.opensips.org
Aihe: Re: [OpenSIPS-Users] How to extract SIP-I ISUP 'Redirecting number' 
parameter
  

Hi,

Unfortunately "Redirecting Number" is not currently supported, but you 
could open up a feature request regarding this, as it would be not very 
difficult to implement.

Regards,

Vlad Patrascu
OpenSIPS Developer
http://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 28.03.2018 18:32, Eugene Prokopiev wrote:
> Hi,
>
> I need to extract ISUP parameter 'Redirecting number' from SIP-I
> INVITE request and add it as SIP header with 'X-' prefix to process in
> FreeSWITCH. Can't found any about 'Redirecting number' in
> http://www.opensips.org/html/docs/modules/2.3.x/sip_i.html, so how to
> extract it? 'Redirecting number' parameter value is visible in
> Wireshark now as ISUP parameter with t=11 and l=7
>


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diff --git a/modules/sip_i/isup.c b/modules/sip_i/isup.c
index 76f8cf0..9dad30f 100644
--- a/modules/sip_i/isup.c
+++ b/modules/sip_i/isup.c
@@ -109,6 +112,43 @@ static struct isup_subfield forward_call_ind_subf[] = {
 		{str_init("no indication"), str_init("connectionless"), str_init("connection"),
 		 str_init("connectionless and connection")}, {0,1,2,3}}},
 	SUBF_INIT_EMPTY};
+/*
+static struct isup_subfield redirection_info_subf[] = {
+{str_init("Redirecting indicator"), {7,
+{str_init("No Redirection"),
+		str_init("Call rerouted"),
+		str_init("Call rerouted, all rediection information presentation restricted"),
+		str_init("Call diverted"),
+		str_init("Call diverted, all redirection information presentation restricted"),
+		str_init("Call rerouted, redirection number presentation restricted"),
+		str_init("Call diversion, redirection number presentation restricted")}, {0,1,2,3,4,5,6}}},
+{str_init("Original redirection reason"), {4,
+{str_init("Unknown/not available"), str_init("User busy"), str_init("No reply"), str_init("Unconditional")}, {0,1,2,3}}},
+{str_init("Redirection counter"), {5,
+{str_init("1"), str_init("2"), str_init("3"), str_init("4"), str_init("5")}, {1,2,3,4,5}}},
+{str_init("Redirection reason"), {4,
+{str_init("Unknown/not available"), str_init("User busy"), str_init("No reply"), str_init("Unconditional")}, {0,1,2,3}}},
+SUBF_INIT_EMPTY};
+*/
+
+static struct isup_subfield redirection_info_subf[] = {
+{str_init("Redirecting indicator"), {7,
+{str_init("No Redirection"),
+str_init("Call rerouted"),
+str_init("Call rerouted, all redirection information presentation restricted"),
+str_init("Call diverted"),
+str_init("Call diverted, all redirection information presentation restricted"),
+str_init("Call rerouted, redirection number presentation restricted"),
+str_init("Call diversion, redirection number presentation restricted")}, {0,1,2,3,4,5,6}}},
+{str_init("Original redirection reason"), {4,
+{str_init("unavailable"), str_init("user-busy"), str_init("no-answer"), str_init("unconditional")}, {0,1,2,3}}},
+{str_init("Redirection counter"), {5,
+{str_init("1"), str_init("2"), str_init("3"), str_init("4"), str_init("5")}, {1,2,3,4,5}}},
+{str_init("Redirection reason"), {4,
+{str_init("unavailable"), str_init("user-busy"), str_init("no-answer"), str_init("unconditional")}, {0,1,2,3}}},
+SUBF_INIT_EMPTY};
+
+
 
 static struct isup_subfield opt_forward_call_ind_subf[] = {
 	{str_init("Closed user group call indicator"), {3,
@@ -205,6 +245,19 @@ static struct isup_subfield connected_num_subf[] = {
 	{str_init("Address signal"), {0, {{0, 0}}, {0}}},
 	SUBF_INIT_EMPTY};
 
+static struct isup_subfield original_called_num_subf[] = {
+{str_init("Odd/even indicator"), {2,
+{str_init("even"), str_init("odd")}, {0,1}}},
+{str_init("Nature of address indicator"), {4,
+{str_init("subscriber"), str_init("unknown"), str_init("national"),
+ str_init("international")}, {1,2,3,4}}},
+{str_init("Numbering plan indicator"), {3,
+{str_in