Re: [OpenSIPS-Users] Call fetch_dlg_value inside a timer route

[Daniel Zanutti]

That's a hack! Thanks, i'll try out.

On Tue, Jun 13, 2023 at 3:31 AM Bogdan-Andrei Iancu 
wrote:

> What Alberto said still applies.
>
> The timer route is outside the context of any SIP message, while the
> function you want to use (get_profile_size) is part of the dialog module,
> where most of the functions do requires the context of a SIP dialog. I
> agree get_profile_size() does not need the context and could be more
> flexible.
> So, try this hack of using a intermediary route[], like:
> timer_route[]
>{ route(foo); }
> route[foo]
>{ get_profile_size(); }
>
> Regards,
>
> Bogdan-Andrei Iancu
>
> OpenSIPS Founder and Developer
>   https://www.opensips-solutions.com
>   https://www.siphub.com
>
> On 5/18/23 3:13 PM, Daniel Zanutti wrote:
>
> Hi Alberto
>
> In fact, i need to use: get_profile_size()
>
> I want to count some type of calls. I've done using statistics but not
> very precise.
>
>
> On Wed, May 17, 2023 at 7:48 PM Alberto  wrote:
>
>> This reminded me of something else, maybe it can be used here too:
>> In a event route, before you can use dialog variables, you have to load
>> the dialog context by using func_load_dialog_ctx. Maybe it's the same with
>> timer routes.
>>
>> https://opensips.org/docs/modules/3.2.x/dialog#func_load_dialog_ctx
>>
>> On Wed, 17 May 2023, 21:07 Daniel Zanutti, 
>> wrote:
>>
>>> Hi folks
>>>
>>> Why is it not possible to call *fetch_dlg_value *inside a timer route?
>>> Is there any other alternative to it?
>>>
>>> I wanted to generate some statistics every X seconds.
>>>
>>> Thanks
>>> ___
>>> Users mailing list
>>> Users@lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>> ___
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>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>
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>
>
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Re: [OpenSIPS-Users] Access "Identity" header after calling stir_shaken_auth() on 3.2.x

[Daniel Zanutti]

Hi Bogdan

Yes, that's exactly how I ended. The original problem is I was using 3.2.x
version and this version hasn't out param. I updated just the module to
3.3.x and it worked.

On Tue, Jun 13, 2023 at 3:45 AM Bogdan-Andrei Iancu 
wrote:

> Daniel,
>
> The auth function has the option to return you into a variable the
> identity value . See the `out` param here:
>
> https://opensips.org/docs/modules/3.3.x/stir_shaken.html#func_stir_shaken_auth
>
> Regards,
>
> Bogdan-Andrei Iancu
>
> OpenSIPS Founder and Developer
>   https://www.opensips-solutions.com
>   https://www.siphub.com
>
> On 5/15/23 12:08 AM, Daniel Zanutti wrote:
>
> Hi Brett
>
> Just to respond, no it doesn't. This field is only visible when we are
> authenticating headers, not generating.
>
> At the end, I copied the module from the 3.3.x version, to my 3.2.11
> version. It worked flawlessly.
>
> Thanks
>
> On Fri, Apr 28, 2023 at 12:10 PM Brett Nemeroff 
> wrote:
>
>> Hello Daniel,
>> See if the $identity peudovariable offered by that module suits your
>> need:
>>
>> https://opensips.org/docs/modules/3.2.x/stir_shaken.html#pv_identity
>>
>> -Brett
>>
>>
>> On Fri, Apr 28, 2023 at 9:03 AM Daniel Zanutti 
>> wrote:
>>
>>> Hi
>>>
>>> How can I access the generated Identity header, after calling function
>>> stir_shaken_auth(), on opensips 3.2.x? On 3.3.x there is a new "out"
>>> parameter, is there a way on 3.2.x?
>>>
>>> It's just to store on DB.
>>>
>>> Thanks
>>> ___
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>>> Users@lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>> ___
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>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>
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>
>
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Re: [OpenSIPS-Users] Registration passthrough

[Daniel Zanutti]

Did you check SIP messages inside your opensips box? I saw home routers
changing the Contact some time ago.

I could help if you send the whole script, cannot help further without
seeing the whole picture.

Regards



On Fri, Jun 2, 2023 at 7:51 AM nutxase  wrote:

> So i now have only
> if (is_method("REGISTER")) {
> forward("myasterisk.server.com:5060 <http://node3.sipalto.com:5080>");
> exit;
> }
>
> but the contact shows as
> sip:zyq5d7rf@opensips-ip:5060^...so i need it to be
> sip:extensionnumber@deviceip:5060xxx
>
>
> Sent with Proton Mail <https://proton.me/> secure email.
>
> --- Original Message ---
> On Thursday, June 1st, 2023 at 9:31 PM, Daniel Zanutti <
> daniel.zanu...@gmail.com> wrote:
>
> Check if you are manipulating contact with some function like
> fix_contact() or usage of media proxy solutions like rtpproxy, rtpengine or
> mediaproxy.
>
> Opensips doesn't change Contact if you don't tell to do it.
>
> On Thu, Jun 1, 2023 at 12:57 PM nutxase  wrote:
>
>> even with a stateless forward asterisk see's this
>>
>>
>> Contact: 1000/sip:609q2255@opensipsip:5060 ee32101394 NonQual nan
>>
>> but i wanted it to be
>> Contact: 1000/sip:609q2255@uacip:5060 ee32101394 NonQual nan
>>
>> Sent with Proton Mail <https://proton.me/> secure email.
>>
>> --- Original Message ---
>> On Thursday, June 1st, 2023 at 3:00 PM, nutxase via Users <
>> users@lists.opensips.org> wrote:
>>
>> Hi Daniel
>>
>> Thanks for the reply
>>
>> So it seems asterisk pjsip shows the contact as the ip of opensips not
>> the uac
>> i will try with stateless as i just want to forward it via opensips and
>> asterisk to not know opensips
>>
>>
>> Sent with Proton Mail <https://proton.me/> secure email.
>>
>> --- Original Message ---
>> On Thursday, June 1st, 2023 at 1:59 PM, Daniel Zanutti <
>> daniel.zanu...@gmail.com> wrote:
>>
>> Hi
>>
>> By standard, opensips does not change the Contact and your asterisk box
>> should receive the original Contact, sent by UAC. Are you sure the contact
>> is being changed by Opensips? I saw asterisk ignoring the contact and
>> putting source IP and origin some times. Long time I don't work with
>> asterisk...
>>
>> Also, you can check the stateless forward, it may be enough for you (not
>> sure):
>> https://www.opensips.org/Documentation/Script-CoreFunctions-3-1#toc17
>>
>> Suggestion:
>>
>> Monitore SIP trace and confirm if Opensips is changing the contact
>> (probably not)
>>
>>
>>
>>
>> On Wed, May 31, 2023 at 10:54 AM nutxase via Users <
>> users@lists.opensips.org> wrote:
>>
>>> Hey guys
>>>
>>> I have set opensips to take registrations and pass them through to
>>> asterisk using the below
>>> but my question is how can i make it set the contact to the ip of the
>>> uac and not the ip of opensips ?
>>>
>>> if (is_method("REGISTER")) {
>>> sethostport("sip.asterisk.xyz:50 <http://node3.sipalto.com:5080>60");
>>> route(toasterisk);
>>> }
>>>
>>> route[toasterisk] {
>>> if (!t_relay()) {
>>> xlog("(Rewriting) t_relay 1 - M=$rm RURI=$ru F=$fu T=$tu IP=$si
>>> ID=$ci\n");
>>> sl_reply_error();
>>> }
>>>
>>> Sent with Proton Mail <https://proton.me/> secure email.
>>> ___
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>>> Users@lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>
>>
>>
>
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Re: [OpenSIPS-Users] Registration passthrough

[Daniel Zanutti]

Check if you are manipulating contact with some function like fix_contact()
or usage of media proxy solutions like rtpproxy, rtpengine or mediaproxy.

Opensips doesn't change Contact if you don't tell to do it.

On Thu, Jun 1, 2023 at 12:57 PM nutxase  wrote:

> even with a stateless forward asterisk see's this
>
>
>   Contact:  1000/sip:609q2255@opensipsip:5060 ee32101394 NonQual
> nan
>
> but i wanted it to be
>   Contact:  1000/sip:609q2255@uacip:5060 ee32101394 NonQual
> nan
>
> Sent with Proton Mail <https://proton.me/> secure email.
>
> --- Original Message ---
> On Thursday, June 1st, 2023 at 3:00 PM, nutxase via Users <
> users@lists.opensips.org> wrote:
>
> Hi Daniel
>
> Thanks for the reply
>
> So it seems asterisk pjsip shows the contact as the ip of opensips not the
> uac
> i will try with stateless as i just want to forward it via opensips and
> asterisk to not know opensips
>
>
> Sent with Proton Mail <https://proton.me/> secure email.
>
> --- Original Message ---
> On Thursday, June 1st, 2023 at 1:59 PM, Daniel Zanutti <
> daniel.zanu...@gmail.com> wrote:
>
> Hi
>
> By standard, opensips does not change the Contact and your asterisk box
> should receive the original Contact, sent by UAC. Are you sure the contact
> is being changed by Opensips? I saw asterisk ignoring the contact and
> putting source IP and origin some times. Long time I don't work with
> asterisk...
>
> Also, you can check the stateless forward, it may be enough for you (not
> sure):
> https://www.opensips.org/Documentation/Script-CoreFunctions-3-1#toc17
>
> Suggestion:
>
> Monitore SIP trace and confirm if Opensips is changing the contact
> (probably not)
>
>
>
>
> On Wed, May 31, 2023 at 10:54 AM nutxase via Users <
> users@lists.opensips.org> wrote:
>
>> Hey guys
>>
>> I have set opensips to take registrations and pass them through to
>> asterisk using the below
>> but my question is how can i make it set the contact to the ip of the uac
>> and not the ip of opensips ?
>>
>> if (is_method("REGISTER")) {
>> sethostport("sip.asterisk.xyz:50 <http://node3.sipalto.com:5080>60");
>> route(toasterisk);
>> }
>>
>> route[toasterisk] {
>> if (!t_relay()) {
>> xlog("(Rewriting) t_relay 1 - M=$rm RURI=$ru F=$fu T=$tu IP=$si
>> ID=$ci\n");
>> sl_reply_error();
>> }
>>
>> Sent with Proton Mail <https://proton.me/> secure email.
>> ___
>> Users mailing list
>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>
>
>
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Re: [OpenSIPS-Users] Registration passthrough

[Daniel Zanutti]

Hi

By standard, opensips does not change the Contact and your asterisk box
should receive the original Contact, sent by UAC. Are you sure the contact
is being changed by Opensips? I saw asterisk ignoring the contact and
putting source IP and origin some times. Long time I don't work with
asterisk...

Also, you can check the stateless forward, it may be enough for you (not
sure):
https://www.opensips.org/Documentation/Script-CoreFunctions-3-1#toc17

Suggestion:

Monitore SIP trace and confirm if Opensips is changing the contact
(probably not)




On Wed, May 31, 2023 at 10:54 AM nutxase via Users 
wrote:

> Hey guys
>
> I have set opensips to take registrations and pass them through to
> asterisk using the below
> but my question is how can i make it set the contact to the ip of the uac
> and not the ip of opensips ?
>
> if (is_method("REGISTER")) {
> sethostport("sip.asterisk.xyz:50 
> 60");
> route(toasterisk);
> }
>
> route[toasterisk] {
>   if (!t_relay()) {
>xlog("(Rewriting) t_relay 1 - M=$rm RURI=$ru F=$fu T=$tu
> IP=$si ID=$ci\n");
>   sl_reply_error();
>}
>
> Sent with Proton Mail  secure email.
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
___
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Re: [OpenSIPS-Users] Call fetch_dlg_value inside a timer route

[Daniel Zanutti]

Hi Alberto

In fact, i need to use: get_profile_size()

I want to count some type of calls. I've done using statistics but not very
precise.


On Wed, May 17, 2023 at 7:48 PM Alberto  wrote:

> This reminded me of something else, maybe it can be used here too:
> In a event route, before you can use dialog variables, you have to load
> the dialog context by using func_load_dialog_ctx. Maybe it's the same with
> timer routes.
>
> https://opensips.org/docs/modules/3.2.x/dialog#func_load_dialog_ctx
>
> On Wed, 17 May 2023, 21:07 Daniel Zanutti, 
> wrote:
>
>> Hi folks
>>
>> Why is it not possible to call *fetch_dlg_value *inside a timer route?
>> Is there any other alternative to it?
>>
>> I wanted to generate some statistics every X seconds.
>>
>> Thanks
>> ___
>> 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
>
___
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[OpenSIPS-Users] Call fetch_dlg_value inside a timer route

[Daniel Zanutti]

Hi folks

Why is it not possible to call *fetch_dlg_value *inside a timer route? Is
there any other alternative to it?

I wanted to generate some statistics every X seconds.

Thanks
___
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Re: [OpenSIPS-Users] X-header extraction

[Daniel Zanutti]

Just get the variable $hdr(X-notice)

On Wed, May 17, 2023 at 2:25 PM nutxase via Users 
wrote:

> Hi All,
>
> What is the best way to extract a custom X header from a sip message and
> log it as a variable
> example i receive X-notice:200 and i want to create a variable that with
> 200
>
> is that possible
>
> Sent with Proton Mail  secure email.
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>
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Re: [OpenSIPS-Users] Access "Identity" header after calling stir_shaken_auth() on 3.2.x

[Daniel Zanutti]

Hi Brett

Just to respond, no it doesn't. This field is only visible when we are
authenticating headers, not generating.

At the end, I copied the module from the 3.3.x version, to my 3.2.11
version. It worked flawlessly.

Thanks

On Fri, Apr 28, 2023 at 12:10 PM Brett Nemeroff  wrote:

> Hello Daniel,
> See if the $identity peudovariable offered by that module suits your need:
>
> https://opensips.org/docs/modules/3.2.x/stir_shaken.html#pv_identity
>
> -Brett
>
>
> On Fri, Apr 28, 2023 at 9:03 AM Daniel Zanutti 
> wrote:
>
>> Hi
>>
>> How can I access the generated Identity header, after calling function
>> stir_shaken_auth(), on opensips 3.2.x? On 3.3.x there is a new "out"
>> parameter, is there a way on 3.2.x?
>>
>> It's just to store on DB.
>>
>> Thanks
>> ___
>> 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
>
___
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[OpenSIPS-Users] Access "Identity" header after calling stir_shaken_auth() on 3.2.x

[Daniel Zanutti]

Hi

How can I access the generated Identity header, after calling function
stir_shaken_auth(), on opensips 3.2.x? On 3.3.x there is a new "out"
parameter, is there a way on 3.2.x?

It's just to store on DB.

Thanks
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Re: [OpenSIPS-Users] ACC module for rejected calls

[Daniel Zanutti]

Hi Alberto

You are correct, this is the line you need.

I think you need a created transaction. Since you are responding in
stateless, you may be missing the cdr.
Try changing this and let me know if solves:

sl_send_reply(488, "Not Acceptable Here");  ->  t_reply(488, "Not
Acceptable Here");

Regards



On Mon, Apr 3, 2023 at 6:09 PM Alberto  wrote:

> Hi,
>
> I'm trying to understand the ACC module, using this minimal configuration
> below.
> My question is: why is there no cdr created?
> I would have assumed that `do_accounting("log", "cdr|missed|failed");`
> always saves cdrs, but it doesn't when the call is rejected from inside the
> configuration script.
> Hope my question is clear. Thanks
>
>
> debug_mode=no
>
> log_level=3
> xlog_level=3
> log_stdout=yes
> log_stderror=yes
> log_facility=LOG_LOCAL0
>
> auto_aliases=no
>
> server_signature=yes
>
> socket=udp:0.0.0.0:5060
> socket=tcp:0.0.0.0:5060
>
> tcp_connect_timeout=300
>
> mpath="/usr/lib64/opensips/modules/"
>
> loadmodule "signaling.so"
>
> loadmodule "sl.so"
>
> loadmodule "tm.so"
> modparam("tm", "fr_inv_timeout", 120)
> modparam("tm", "fr_timeout", 30)
> modparam("tm", "onreply_avp_mode", 1)
> modparam("tm", "restart_fr_on_each_reply", 0)
>
> loadmodule "rr.so"
> modparam("rr", "append_fromtag", 1)
>
> loadmodule "acc.so"
> modparam("acc", "early_media", 1)
> modparam("acc", "report_cancels", 1)
> modparam("acc", "detect_direction", 1)
>
> loadmodule "dialog.so"
> modparam("dialog", "default_timeout", 14400) # 4 hours
> modparam("dialog", "dlg_match_mode", 1)
> modparam("dialog", "enable_stats", 0)
> modparam("dialog", "profiles_with_value", "caller")
>
> loadmodule "sipmsgops.so"
>
> loadmodule "proto_udp.so"
> loadmodule "proto_tcp.so"
>
> route {
>   if (has_totag()) {
> if (loose_route()) {
>   if ($DLG_status != NULL && !validate_dialog()) {
> exit;
>   }
> } else {
>   if (is_method("ACK")) {
> if (t_check_trans()) {
>   t_relay();
> }
> exit;
>   }
>   sl_send_reply(404, "Not Found");
>   exit;
> }
> t_relay();
> exit;
>   }
>   t_check_trans();
>   if (is_myself($si)) {
> send_reply(406, "Not Acceptable");
> exit;
>   }
>   if (is_method("INVITE")) {
> if (!create_dialog("B")) {
>   send_reply(500, "Internal Server Error");
>   exit;
> }
>   }
>   if (loose_route()) {
> if (!is_method("ACK")) {
>   sl_send_reply(403, "Preload Route denied");
> }
> exit;
>   }
>   if ($rU == NULL) {
> send_reply(484, "Address Incomplete");
> exit;
>   }
>   route(relay);
> }
>
> route[relay] {
>   if (is_method("INVITE")) {
> do_accounting("log", "cdr|missed|failed");
> sl_send_reply(488, "Not Acceptable Here");
> exit;
>   }
>   if (!t_relay()) {
> send_reply(500, "Internal Error");
>   }
>   exit;
> }
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Re: [OpenSIPS-Users] SIP Call ID duplicated on 3.1.14

[Daniel Zanutti]

Yes, since 1.x it's there.

On Sat, Mar 25, 2023 at 11:38 AM Saint Michael  wrote:

> I use 3.1, is that applicable?
>
>
> On Sat, Mar 25, 2023 at 9:49 AM Daniel Zanutti 
> wrote:
>
>> Hi  Federico
>>
>> Yes it does, need to create the transaction inside your script:
>> https://opensips.org/html/docs/modules/3.2.x/tm.html#func_t_newtran
>>
>> This will avoid opensips handling the duplicated invite as a new call.
>>
>>
>> On Fri, Mar 24, 2023 at 11:40 PM Saint Michael  wrote:
>>
>>> I have on a typical day many calls that arrive twice, same  SIP CALL ID,
>>> almost the same time, maybe the next millisecond. This happens because the
>>> dialer does not wait for a confirmation, and resends the same call multiple
>>> times. But it confuses my CDR.
>>> Is there a way to check for the CALL ID and discard the INVITE if there
>>> is another INVITE already in progress?
>>>
>>> ___
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>>>
>> ___
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>>
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Re: [OpenSIPS-Users] SIP Call ID duplicated on 3.1.14

[Daniel Zanutti]

Hi  Federico

Yes it does, need to create the transaction inside your script:
https://opensips.org/html/docs/modules/3.2.x/tm.html#func_t_newtran

This will avoid opensips handling the duplicated invite as a new call.


On Fri, Mar 24, 2023 at 11:40 PM Saint Michael  wrote:

> I have on a typical day many calls that arrive twice, same  SIP CALL ID,
> almost the same time, maybe the next millisecond. This happens because the
> dialer does not wait for a confirmation, and resends the same call multiple
> times. But it confuses my CDR.
> Is there a way to check for the CALL ID and discard the INVITE if there is
> another INVITE already in progress?
>
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Re: [OpenSIPS-Users] Check status of routes loading - drouting

[Daniel Zanutti]

Exactly!

Thanks

On Fri, Mar 24, 2023 at 11:25 AM Callum Guy  wrote:

> Hi Daniel,
>
> I believe you're looking for this feature as included since 3.3
>
> https://www.opensips.org/Documentation/Interface-StatusReport-3-3
>
> Enjoy,
>
> Callum
>
> On Fri, 24 Mar 2023 at 14:19, Daniel Zanutti 
> wrote:
>
>> Hi
>>
>> Is there a way to check the status of initial loading of routes, on the
>> drouting module?
>>
>> If routes are being loaded after a cold start, I want to do some
>> alternate routing.
>>
>> Thanks
>>
>> ___
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>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>
>
>
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[OpenSIPS-Users] Check status of routes loading - drouting

[Daniel Zanutti]

Hi

Is there a way to check the status of initial loading of routes, on the
drouting module?

If routes are being loaded after a cold start, I want to do some alternate
routing.

Thanks
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Re: [OpenSIPS-Users] Carrier_ID not writing on DRouting when using partitions

[Daniel Zanutti]

That solved! Don't know how I missed it.

Thank you Ben

On Wed, Feb 22, 2023 at 12:50 PM Ben Newlin  wrote:

> Daniel,
>
>
>
> I haven’t used the drouting module with partitions, but it looks like when
> you do that the name of the AVP no longer comes from the modparam setting
> but from the dr_partitions table. Do you have the carrier_id_avp value set
> for this partition in that table?
>
>
>
>
> https://opensips.org/docs/modules/3.2.x/drouting.html#param_db_partitions_table
>
>
> https://www.opensips.org/Documentation/Install-DBSchema-3-2#GEN-DB-DR-PARTITIONS
>
>
>
>
>
> Ben Newlin
>
>
>
> *From: *Users  on behalf of Daniel
> Zanutti 
> *Date: *Tuesday, February 21, 2023 at 9:47 PM
> *To: *OpenSIPS users mailling list 
> *Subject: *[OpenSIPS-Users] Carrier_ID not writing on DRouting when using
> partitions
>
> * EXTERNAL EMAIL - Please use caution with links and attachments *
>
>
> --
>
> Hey
>
>
>
> I'm having a weird issue, possibly a BUG, using opensips 3.2.8.
>
>
>
> The carrier_id_avp is not being written, when I enabled partitions or
> drouting module. Everything else works, just this value is not written to
> the AVP setted. Routing works fine using carriers, just the AVP is not
> written.
>
>
>
> When I disable partitions, it works fine writting to carrier_id_avp. =|
>
>
>
> Configuration:
>
> modparam("drouting", "carrier_id_avp", "$avp(carrier_id)")
>
>
>
> Example of call
>
> My dr_rule has gwlist = #651
>
>
>
> Carrier 651
>
> carrierid=651
>
> gwlist=651 (yes, same id)
>
>
>
> Applied routing with partition:
>
> do_routing(,"F",,,$avp(gw_attrs),,"partition1")
>
>
>
> Log:
>
> Feb 22 01:14:51 sbc4 /usr/local/sbin/opensips[856]:
> DBG:drouting:push_gw_for_usage: adding gw [651] as "sip:x...@yy.yy.yy.yy"
> in order 0
> Feb 22 01:14:51 sbc4 /usr/local/sbin/opensips[856]:
> DBG:drouting:push_gw_for_usage: setting GW id [651] as avp
> Feb 22 01:14:51 sbc4 /usr/local/sbin/opensips[856]:
> DBG:drouting:push_gw_for_usage: setting GW attr [1] as avp
>
>
>
> The gateway 651 does exist, it's the same name of Carrier but it shouldn't
> be a problem.
>
>
>
> Checking logs, $avp(carrier_id) is null:
>
> Carrier:yy.yy.yy.yy()
>
>
>
> Do you guys have any clues on how to solve it? Maybe a bug?
>
>
>
> This works fine when not using partitions.
>
>
>
> Thanks
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[OpenSIPS-Users] Carrier_ID not writing on DRouting when using partitions

[Daniel Zanutti]

Hey

I'm having a weird issue, possibly a BUG, using opensips 3.2.8.

The carrier_id_avp is not being written, when I enabled partitions or
drouting module. Everything else works, just this value is not written to
the AVP setted. Routing works fine using carriers, just the AVP is not
written.

When I disable partitions, it works fine writting to carrier_id_avp. =|

Configuration:
modparam("drouting", "carrier_id_avp", "$avp(carrier_id)")

Example of call
My dr_rule has gwlist = #651

Carrier 651
carrierid=651
gwlist=651 (yes, same id)

Applied routing with partition:
do_routing(,"F",,,$avp(gw_attrs),,"partition1")

Log:
Feb 22 01:14:51 sbc4 /usr/local/sbin/opensips[856]:
DBG:drouting:push_gw_for_usage: adding gw [651] as "sip:x...@yy.yy.yy.yy"
in order 0
Feb 22 01:14:51 sbc4 /usr/local/sbin/opensips[856]:
DBG:drouting:push_gw_for_usage: setting GW id [651] as avp
Feb 22 01:14:51 sbc4 /usr/local/sbin/opensips[856]:
DBG:drouting:push_gw_for_usage: setting GW attr [1] as avp

The gateway 651 does exist, it's the same name of Carrier but it shouldn't
be a problem.

Checking logs, $avp(carrier_id) is null:
Carrier:yy.yy.yy.yy()

Do you guys have any clues on how to solve it? Maybe a bug?

This works fine when not using partitions.

Thanks
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Re: [OpenSIPS-Users] Version 3.1 it stops writing cdrs

[Daniel Zanutti]

Virtual Router Redundancy Protocol (VRRP)

https://www.techopedia.com/definition/13483/virtual-router-redundancy-protocol-vrrp


On Wed, Feb 15, 2023 at 3:21 PM Saint Michael  wrote:
>
> what is VRRP ?
>
> On Wed, Feb 15, 2023 at 1:16 PM Kingsley Tart 
wrote:
>>
>> FWIW, I set up OpenSIPS here in a couple of clusters with VRRP managing
>> the service "floating" IP addresses. If I want to restart a node, I
>> mark it as disabled, which then means that
>>
>> * it stops replying to OPTIONS
>> * it responds 503 to INVITE if other nodes are still up
>> * when all early dialogs have ended, the IP moves to another node
>>
>> I can then restart OpenSIPS on this node without losing anything.
>>
>> Cheers,
>> Kingsley.
>>
>> On Thu, 2023-01-12 at 23:29 -0500, Saint Michael wrote:
>> > Is there a command that I may run without restarting opensips that
>> > restarts the process, internally?
>> > if I restart opensips I lose all the records.
>>
>>
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Re: [OpenSIPS-Users] kamilio's htable equivalent?

[Daniel Zanutti]

Hey David

Did you take a look at core functions of cache? ->
https://www.opensips.org/Documentation/Script-CoreFunctions-3-1#toc4


On Wed, Dec 21, 2022 at 9:14 AM David Villasmil <
david.villasmil.w...@gmail.com> wrote:

> Hello folks,
>
> I'm trying to find in opensips an equivalent to kamailio's htable module.
> Opensips' cachedb_local doesn't have autoexpires...
>
> Regards,
>
> David Villasmil
> email: david.villasmil.w...@gmail.com
> phone: +34669448337
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Re: [OpenSIPS-Users] - Not sending ACK back!

[Daniel Zanutti]

Hi Nitesh

I think the default OPENSIPS.CFG script has a Registrar Server and the
Lookup function. Please take a look there. To call then, you have to use
lookup function.

The ACK missing feature maybe related to NAT problems. Where is the ACK
lost? Did origin sent to Opensips? Did opensips forward to destination?
Please take a deep look at default opensips.cfg, I tested it in the past
and worked fine.


On Fri, Oct 21, 2022 at 3:05 PM Nitesh Divecha 
wrote:

> Daniel,
>
> Thanks for your detailed email. Much appreciated!
>
> Yes I totally understand about OpenSIPS and I'm going over all the
> documentations and tutorials I can find! My background is from Asterisk, so
> with that mentality I'm tackling OpenSIPS.
>
> Asterisk (Context) vs OpenSIPS (C-style script), it is a huge challenge
> but getting the hang of it!
>
> Lets pick option (3) Routing local calls (ATA to ATA)... Do you have any
> examples I can follow and set up a local route? I set up two extensions
> (8883456 and 7773456) via OpenSIPS Control Panel and I was able to register
> two ATA's. I do have the Registrar module loaded... How can I call each
> other?
>
>  REGISTRAR module
>
> loadmodule "registrar.so"
>
> modparam("registrar", "default_expires", 3600)
>
> modparam("registrar", "min_expires", 60)
>
> modparam("registrar", "max_expires", 120)
>
> # modparam("registrar", "tcp_persistent_flag", "TCP_PERSISTENT")
>
> # modparam("registrar", "received_avp", "$avp(received_nh)")/* uncomment
> the next line not to allow more than 10 contacts per AOR */
>
> modparam("registrar", "max_contacts", 10)
>
> modparam("registrar", "received_avp", "$avp(rcv)")
>
> modparam("registrar", "retry_after", 30)
>
>
> Regarding option (4) - I have both options. IP to IP and User/Pass
> authentication provider. Using User/Pass I was able to register OpenSIPS as
> UAC to a remote server and I was able to make outbound calls but call keeps
> on dropping due to no ACK.
>
> Cheers,
> Nitesh
>
>
>
>
>
> On Fri, Oct 21, 2022 at 1:26 PM Daniel Zanutti 
> wrote:
>
>> Hi Nitesh
>>
>> As you already know, opensips is a low level software. You have to
>> understand several aspects of SIP, network, RTP, DNS that when you use
>> Asterisk, most you don't need to understand deep.
>>
>> Trying to help you, your script is way simple for you achievements. You
>> need:
>> 1) Check NAT on all request + all replies. This is to fix your SIP
>> messages.
>>
>> 2) Check if you need to apply RTPPROXY on the call. You can use the
>> "engage" function on INVITE then forget about it OR you can use manual way
>> with "offer" function and handle all scenarios manually. For example, call
>> the "answer" function on the 200 OK. Then delete on BYE.
>>
>> 3) Routing local calls (ATA to ATA) you need to handle the Register first
>> with "save" function, then later handle the INVITE with the "lookup"
>> function, both of Registrar module.
>>
>> 4) PSTN can be used as a direct route or some dynamic routing solution.
>> Make it work first with direct routing. Need to check how authenticate
>> works on your carrier. If IP based will be fine, if user/pass you need to
>> make your opensips authenticate, it's a little harder.
>>
>> 5) DID - You have to create some specific INBOUND rules. Calls will be
>> anonymous or authenticated?
>>
>> 6) Fax - Better solve other issues first.
>>
>> Hope this gives you some direction. Look for some tutorials.
>>
>> Regards
>>
>> On Fri, Oct 21, 2022 at 11:11 AM Nitesh Divecha <
>> aviator.nites...@gmail.com> wrote:
>>
>>> Hello All,
>>>
>>> I have been scratching my head for a few days now... Just to recap:
>>>
>>> I'm a newbie with OpenSIPS so bear with me... I got OpenSIPS 3.3.1
>>> (residential) running on Debian 11 with OpenSIPS Control Panel 9.3.2 and
>>> MySQL.
>>>
>>> My goal is to:
>>> 1) Make two ATA's register and call each other (locally)... *Stopped
>>> working, I think routing logic is missing.*
>>> 2) Make ATA to call PSTN via an outbound SIP trunk or DID provider... *No
>>> ACK sent to Outbound provider.*
>>> 3) Receive inbound calls from PSTN or SIP trunk and forward it to
>>> registered ATA... *Getting rejected.*
>>> 4) Able to send and receive faxes from and to PST

Re: [OpenSIPS-Users] - Not sending ACK back!

[Daniel Zanutti]

Hi Nitesh

As you already know, opensips is a low level software. You have to
understand several aspects of SIP, network, RTP, DNS that when you use
Asterisk, most you don't need to understand deep.

Trying to help you, your script is way simple for you achievements. You
need:
1) Check NAT on all request + all replies. This is to fix your SIP messages.

2) Check if you need to apply RTPPROXY on the call. You can use the
"engage" function on INVITE then forget about it OR you can use manual way
with "offer" function and handle all scenarios manually. For example, call
the "answer" function on the 200 OK. Then delete on BYE.

3) Routing local calls (ATA to ATA) you need to handle the Register first
with "save" function, then later handle the INVITE with the "lookup"
function, both of Registrar module.

4) PSTN can be used as a direct route or some dynamic routing solution.
Make it work first with direct routing. Need to check how authenticate
works on your carrier. If IP based will be fine, if user/pass you need to
make your opensips authenticate, it's a little harder.

5) DID - You have to create some specific INBOUND rules. Calls will be
anonymous or authenticated?

6) Fax - Better solve other issues first.

Hope this gives you some direction. Look for some tutorials.

Regards

On Fri, Oct 21, 2022 at 11:11 AM Nitesh Divecha 
wrote:

> Hello All,
>
> I have been scratching my head for a few days now... Just to recap:
>
> I'm a newbie with OpenSIPS so bear with me... I got OpenSIPS 3.3.1
> (residential) running on Debian 11 with OpenSIPS Control Panel 9.3.2 and
> MySQL.
>
> My goal is to:
> 1) Make two ATA's register and call each other (locally)... *Stopped
> working, I think routing logic is missing.*
> 2) Make ATA to call PSTN via an outbound SIP trunk or DID provider... *No
> ACK sent to Outbound provider.*
> 3) Receive inbound calls from PSTN or SIP trunk and forward it to
> registered ATA... *Getting rejected.*
> 4) Able to send and receive faxes from and to PSTN... *Haven't even
> touched.*
>
> Fast forward... I did achieve a few of my goals but they stopped
> working... You fix one thing and you break others...
>
> My current issue is OpenSIPS is not sending ACK back to the Outbound
> provider when I make calls to PSTN thus calls are getting dropped from the
> Outbound provider due to no ACK. This issue started when I implemented
> topology_hiding('C"), rtpproxy_offer("ro"), uac_replace_from( ,
> "$avp(furi)").
>
> Here is my code snippet:
>
> ### Routing Logic 
>
>
> # main request routing logic
>
>
> route{
>
>
> #if ($rU=~"^\+[1-9][0-9]+$") {
>
> if (dp_translate(10 ,$rU ,$rU) ) {
>
> xlog("*** 2. Dial plan translate from source $avp(src) to
> $rU ***\n");
>
>
> $avp(furi) = "sip:aaabbbc...@gothamcity.com";
>
> uac_replace_from( , "$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] {
>
> xlog("*** 3. Entering route relay ***\n");
>
> # for INVITEs enable some additional helper routes
>
> if (is_method("INVITE")) {
>
> topology_hiding("C");
>
> if(remove_hf("User-Agent")){
>
> xlog("*** 4. User-Agent found and removed.
> ***\n");
>
> }
>
>
> if (isflagset("NAT") && has_body("application/sdp")) {
>
> rtpproxy_offer("ro");
>
> #rtpproxy_offer();
>
> }
>
>
> 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;
>
> }
>
> Any thoughts or suggestions on what to check for ACK?
>
> Cheers,
> Nitesh
>
>
>
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Re: [OpenSIPS-Users] Question about cache_store and lifetime of variables

[Daniel Zanutti]

No, works same way.

Just look at docs of 3.1

On Mon, Sep 26, 2022 at 11:58 AM Saint Michael  wrote:

> I use opensips 3.1, does it matter?
>
>
> On Mon, Sep 26, 2022 at 10:20 AM Daniel Zanutti 
> wrote:
>
>> can you write your own functions with opensips?
>> Yes -> using routes
>>
>> Can you point me to an example?
>> Storing->
>> https://opensips.org/html/docs/modules/2.2.x/dialog.html#idp5880336
>> Retrieving ->
>> https://opensips.org/html/docs/modules/2.2.x/dialog.html#idp5887712
>>
>> Or work with flags, if just true or false value
>> https://opensips.org/html/docs/modules/2.2.x/dialog.html#idp341408
>>
>> Regards
>> On Sun, Sep 25, 2022 at 1:45 PM Saint Michael  wrote:
>>
>>> Question:
>>> can you write your own functions with opensips?
>>>
>>>
>>> On Sun, Sep 25, 2022 at 12:05 PM Saint Michael 
>>> wrote:
>>>
>>>> Dear Daniel
>>>> Can you point me to an example?
>>>> Right now Opensios will get a clogged memory.
>>>> Many thanks.
>>>>
>>>>
>>>> On Sun, Sep 25, 2022, 11:45 AM Daniel Zanutti 
>>>> wrote:
>>>>
>>>>> You have to use dialog variable storing.
>>>>> Take a look at dialog module.
>>>>>
>>>>> Em dom., 25 de set. de 2022 10:42, Saint Michael 
>>>>> escreveu:
>>>>>
>>>>>> I noticed that the variable
>>>>>> $avp(lineid)
>>>>>> set in the section of the code handling the original INVITE, is null
>>>>>> when I need to close the call.
>>>>>> Is there a way to store a variable that will be available
>>>>>> throughout the call, everywhere?
>>>>>> I am trying:
>>>>>> cache_store("local","lineid_$ci","$avp(lineid)",0);
>>>>>> but I need this value to disappear when this call is closed. I cannot
>>>>>> set an expiration because the call may last for 2 hours or 2 seconds.
>>>>>>
>>>>>> many thanks for your help and guidance
>>>>>>
>>>>>> Philip
>>>>>>
>>>>>>
>>>>>> ___
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>>>>>>
>>>>> ___
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>>>>>
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Re: [OpenSIPS-Users] Question about cache_store and lifetime of variables

[Daniel Zanutti]

can you write your own functions with opensips?
Yes -> using routes

Can you point me to an example?
Storing->
https://opensips.org/html/docs/modules/2.2.x/dialog.html#idp5880336
Retrieving ->
https://opensips.org/html/docs/modules/2.2.x/dialog.html#idp5887712

Or work with flags, if just true or false value
https://opensips.org/html/docs/modules/2.2.x/dialog.html#idp341408

Regards
On Sun, Sep 25, 2022 at 1:45 PM Saint Michael  wrote:

> Question:
> can you write your own functions with opensips?
>
>
> On Sun, Sep 25, 2022 at 12:05 PM Saint Michael  wrote:
>
>> Dear Daniel
>> Can you point me to an example?
>> Right now Opensios will get a clogged memory.
>> Many thanks.
>>
>>
>> On Sun, Sep 25, 2022, 11:45 AM Daniel Zanutti 
>> wrote:
>>
>>> You have to use dialog variable storing.
>>> Take a look at dialog module.
>>>
>>> Em dom., 25 de set. de 2022 10:42, Saint Michael 
>>> escreveu:
>>>
>>>> I noticed that the variable
>>>> $avp(lineid)
>>>> set in the section of the code handling the original INVITE, is null
>>>> when I need to close the call.
>>>> Is there a way to store a variable that will be available
>>>> throughout the call, everywhere?
>>>> I am trying:
>>>> cache_store("local","lineid_$ci","$avp(lineid)",0);
>>>> but I need this value to disappear when this call is closed. I cannot
>>>> set an expiration because the call may last for 2 hours or 2 seconds.
>>>>
>>>> many thanks for your help and guidance
>>>>
>>>> Philip
>>>>
>>>>
>>>> ___
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>>>>
>>> ___
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>>>
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Re: [OpenSIPS-Users] Question about cache_store and lifetime of variables

[Daniel Zanutti]

You have to use dialog variable storing.
Take a look at dialog module.

Em dom., 25 de set. de 2022 10:42, Saint Michael 
escreveu:

> I noticed that the variable
> $avp(lineid)
> set in the section of the code handling the original INVITE, is null when
> I need to close the call.
> Is there a way to store a variable that will be available throughout the
> call, everywhere?
> I am trying:
> cache_store("local","lineid_$ci","$avp(lineid)",0);
> but I need this value to disappear when this call is closed. I cannot set
> an expiration because the call may last for 2 hours or 2 seconds.
>
> many thanks for your help and guidance
>
> Philip
>
>
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Re: [OpenSIPS-Users] The update from yesterday makes all calls fail after 20 seconds, how do I go back?

[Daniel Zanutti]

Hi Federico

You said "but my calls do not use the rtpproxy, so why is this affecting my
traffic and killing my calls?"

Surelly Opensips is trying to use rtpproxy in the calls, the log is showing
this. So the problem probably is related to the rtpproxy service. It's hard
to say anything without logs of the rtpproxy service, config seems to be
fine.

How many simultaneous calls are you handling? Why so many rtpproxy
services? Take a look at rtpproxy log to see what is happening.

If you have some urgency to solve this, I could help in a faster way
through consultancy. Let me know

Daniel

On Thu, Sep 15, 2022 at 11:36 AM Saint Michael  wrote:

> I added this code to the end of my script
> local_route
> {
>   if (is_method("BYE"))
>   {
>   xlog("L_ERR", "LOCAL_ROUTE - BYE - $DLG_end_reason - $ru - $ci");
>   }
> }
> and now my system is filed with these errors:
> Sep 15 14:25:36 node5 opensips[238750]: Sep 15 14:25:35 [238750]
> LOCAL_ROUTE - BYE - RTPProxy Timeout - sip:7867830513@208.73.234.96:5060
> - 0e87838610bfe6e1670023aa6bbe9590@208.73.234.96Sep 15 14:25:35 [238750]
> LOCAL_ROUTE - BYE - RTPProxy Timeout - sip:192.69.217.154;did=8fc1.e2444961
> - 0e87838610bfe6e1670023aa6bbe9590@208.73.234.96
> but my calls do not use the rtpproxy, so why is this affecting my traffic
> and killing my calls?
> I have 10 rttpproxy services:
> grep rtpproxy
>   rtpproxy1.serviceloaded active running
> RTPProxy1
>   rtpproxy10.service   loaded active running
> RTPProxy10
>   rtpproxy2.serviceloaded active running
> RTPProxy2
>   rtpproxy3.serviceloaded active running
> RTPProxy3
>   rtpproxy4.serviceloaded active running
> RTPProxy4
>   rtpproxy5.serviceloaded active running
> RTPProxy5
>   rtpproxy6.serviceloaded active running
> RTPProxy6
>   rtpproxy7.serviceloaded active running
> RTPProxy7
>   rtpproxy8.serviceloaded active running
> RTPProxy8
>   rtpproxy9.serviceloaded active running
> RTPProxy9
>
> here is number 1
> [Unit]
> Description=RTPProxy1
> After=network.target
> Requires=network.target
>
> [Service]
> Type=forking
> PIDFile=/var/run/rtpproxy1.pid
> #Environment='OPTIONS= -F -L 10240 -m 2 -M 3 -T 20 -d
> INFO:LOG_LOCAL5'
>
> Restart=on-failure
> RestartSec=5
>
>
> ExecStart=/usr/local/bin/rtpproxy -p /var/run/rtpproxy1.pid -l
> Public.Ip.Address \
>-s udp:127.0.0.1:7890 -F -L 10240 -m 1 -M 15000 -T 20 -d
> WARN:LOG_LOCAL5 -n tcp:127.0.0.1:7889
> ExecStop=/usr/bin/pkill -F /var/run/rtpproxy1.pid
>
>
> StandardOutput=syslog
> StandardError=syslog
> SyslogIdentifier=rtpproxy1
> SyslogFacility=local5
>
> TimeoutStartSec=10
> TimeoutStopSec=10
>
> [Install]
> WantedBy=multi-user.target
>
>
> Here is number 2
>
> [Unit]
> Description=RTPProxy2
> After=network.target
> Requires=network.target
>
> [Service]
> Type=forking
> PIDFile=/var/run/rtpproxy2.pid
> #Environment='OPTIONS= -F -L 10240 -m 2 -M 3 -T 20 -d
> INFO:LOG_LOCAL5'
>
> Restart=on-failure
> RestartSec=5
>
>
> ExecStart=/usr/local/bin/rtpproxy -p /var/run/rtpproxy2.pid -l
> Public.IP.address \
>-s udp:127.0.0.1:7891 -F -L 10240 -m 15000 -M 2 -T 20 -d
> WARN:LOG_LOCAL5 -n tcp:127.0.0.1:7889
> ExecStop=/usr/bin/pkill -F /var/run/rtpproxy2.pid
>
>
> StandardOutput=syslog
> StandardError=syslog
> SyslogIdentifier=rtpproxy2
> SyslogFacility=local5
>
> TimeoutStartSec=10
> TimeoutStopSec=10
>
> [Install]
> WantedBy=multi-user.target
>
>
> what am I doing wrong?
>
> Federico
>
>
>
>
>
> On Wed, Sep 14, 2022 at 4:55 PM Daniel Zanutti 
> wrote:
>
>> Hi
>>
>> Everytime opensips sends the BYE, it's generated inside local_route:
>> https://www.opensips.org/Documentation/Script-Routes-3-1#toc6
>>
>> So put a xlog there to see why. Something like this:
>> local_route
>> {
>>   if (is_method("BYE"))
>>   {
>>   xlog("L_ERR", "LOCAL_ROUTE - BYE - $DLG_end_reason - $ru - $ci");
>>   }
>> }
>>
>>
>> On Wed, Sep 14, 2022 at 5:04 PM Johan De Clercq  wrote:
>>
>>> Xlog(….);
>>>
>>> Outlook voor iOS <https://aka.ms/o0ukef> downloaden
>>> --
>>> *Van:* Users  namens Saint Michael <
>>> vene...@gmail.com>
>>> *Verzonden:* Wednesday, September 14, 2022 9:56:41 PM
>&g

Re: [OpenSIPS-Users] The update from yesterday makes all calls fail after 20 seconds, how do I go back?

[Daniel Zanutti]

Hi

Everytime opensips sends the BYE, it's generated inside local_route:
https://www.opensips.org/Documentation/Script-Routes-3-1#toc6

So put a xlog there to see why. Something like this:
local_route
{
  if (is_method("BYE"))
  {
  xlog("L_ERR", "LOCAL_ROUTE - BYE - $DLG_end_reason - $ru - $ci");
  }
}


On Wed, Sep 14, 2022 at 5:04 PM Johan De Clercq  wrote:

> Xlog(….);
>
> Outlook voor iOS <https://aka.ms/o0ukef> downloaden
> --
> *Van:* Users  namens Saint Michael <
> vene...@gmail.com>
> *Verzonden:* Wednesday, September 14, 2022 9:56:41 PM
> *Aan:* OpenSIPS users mailling list 
> *Onderwerp:* Re: [OpenSIPS-Users] The update from yesterday makes all
> calls fail after 20 seconds, how do I go back?
>
> how do I do this:
> " put some log on local_route"
> Sorry I am learning
>
>
> On Wed, Sep 14, 2022 at 3:55 PM Daniel Zanutti 
> wrote:
>
> So your Opensips is hanging up the call.
>
> Do you see any log on it? Try put some log on local_route if you don't see
> anything.
>
>
>
> On Wed, Sep 14, 2022 at 4:40 PM Saint Michael  wrote:
>
> This is a trace showing a BYE from Opensips, but none of the sides did
> actually hangup.
>
>
> On Wed, Sep 14, 2022 at 3:33 PM Saint Michael  wrote:
>
> I use opensips 3.1, and I did an update yesterday. in all the boxes that I
> upgraded all calls fail after 20 seconds.
>
> cd /usr/src/opensips-3.1/
> git pull
> make clean;make proper;make all
> make modules
> make install
> clearlog.sh
> systemctl restart opensips
> opensips -V
>
>
>
> How do I go back?
>
>
>
> ___
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> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
> ___
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>
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>
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Re: [OpenSIPS-Users] The update from yesterday makes all calls fail after 20 seconds, how do I go back?

[Daniel Zanutti]

So your Opensips is hanging up the call.

Do you see any log on it? Try put some log on local_route if you don't see
anything.



On Wed, Sep 14, 2022 at 4:40 PM Saint Michael  wrote:

> This is a trace showing a BYE from Opensips, but none of the sides did
> actually hangup.
>
>
> On Wed, Sep 14, 2022 at 3:33 PM Saint Michael  wrote:
>
>> I use opensips 3.1, and I did an update yesterday. in all the boxes that
>>> I upgraded all calls fail after 20 seconds.
>>
>> cd /usr/src/opensips-3.1/
>> git pull
>> make clean;make proper;make all
>> make modules
>> make install
>> clearlog.sh
>> systemctl restart opensips
>> opensips -V
>>
>>
>>
>> How do I go back?
>>
>>
>>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
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Re: [OpenSIPS-Users] Question about error 500 only via WIFI

[Daniel Zanutti]

Olá Rodrigo

The problem seems to be the IPV6 addresses. Did you look at the logs to
know exactly why opensips refused the call?

IPV4 is almost out in Brazil, mobile carriers are using ipv6 addresses on
mobile devices. This is the issue, but in theory opensips should handle
this seamlessly. Do you have IPV6 in your server or the message came on
ipv4 address but SIP has ipv6 addresses?

I recommend you check logs, the issue should be there. Also, if you plan to
allow mobile devices, you should be prepared to use ipv6 on your net.

On Thu, May 12, 2022 at 4:57 PM Rodrigo Pimenta Carvalho 
wrote:

> Hi.  I found the error cause.  But I still don't know why I have such
> issue.
>
>
> When I use my Internet Link (WIFI in my home office), the SIP register
> message is sent correctly. Like this:
>
> Session Initiation Protocol (REGISTER)
> Request-Line: REGISTER sip:54.233.189.46:5060;transport=UDP SIP/2.0
> Method: REGISTER
> Request-URI: sip:54.233.189.46:5060;transport=UDP
> [Resent Packet: False]
> Message Header
> Via: SIP/2.0/UDP 192.168.1.103:5060
> ;branch=z9hG4bK-524287-1---6dbfa766cffddeee;rport
> Max-Forwards: 70
> Contact:  ;rinstance=1afa98b2b6d17a34;transport=UDP>
> To: ;transport=UDP>
> From: ;tag=98bfc34c
> Call-ID: H1E0jkwiMniiyT5az1BT7g..
> CSeq: 1 REGISTER
> Expires: 60
> Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS,
> INFO, SUBSCRIBE
> User-Agent: Zoiper v2.10.18.1-mod
> Allow-Events: presence, kpml, talk
> Content-Length: 0
>
> Opensips got the message above.
>
> However, when I use the GSM mobile network (from VIVO) , some service
> changes the content of the SIP Register message. Like this:
>
> Session Initiation Protocol (REGISTER)
> Request-Line: REGISTER sip:*[64:ff9b::36e9:bd2e]*:5060;transport=UDP
> SIP/2.0
> Method: REGISTER
> Request-URI: sip:*[64:ff9b::36e9:bd2e]*:5060;transport=UDP
> [Resent Packet: False]
> Message Header
> Via: SIP/2.0/UDP *[64:ff9b::c000:4]*
> ;branch=z9hG4bK-524287-1---0a8189adf6c3449a
> Max-Forwards: 70
> Contact:  :5060;transport=UDP;rinstance=ffaac43f13178e89>
> To: 
> From: ;tag=f19aea4d
> Call-ID: VICBinZsDk5_ZhpHGd__CQ..
> CSeq: 1 REGISTER
> Expires: 60
> Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS,
> INFO, SUBSCRIBE
> User-Agent: Zoiper v2.10.18.1-mod
> Allow-Events: presence, kpml, talk
> Content-Length: 0
>
>
> That is why Opensips returns error 500. I guess some service changed IPv4
> to something IPv6.
> Could it be caused by the GSM operator (VIVO) ?
> What should I investigage to solve this problem?
>
> Any hint will be very helpful !
>
> Thanks alot.
>
> RODRIGO PIMENTA CARVALHO
> Inatel Competence Center
> Software
> Ph: +55 35 3471 9200 RAMAL: 979
>
> --
> *De:* Users  em nome de Daniel Zanutti <
> daniel.zanu...@gmail.com>
> *Enviado:* quinta-feira, 12 de maio de 2022 11:57
> *Para:* OpenSIPS users mailling list 
> *Assunto:* Re: [OpenSIPS-Users] Question about error 500 only via WIFI
>
> Olá Rodrigo, tudo bem? Saudações de São Paulo!
>
> Opensips doesn't differentiate the network, it will look just to the sip
> packet. I recommend you sniff through your packets and check what's
> different. Probably there's somenthing on opensips log you didn't get yet,
> recommend you take a look there first btw.
>
> About push, I think you're enable push notifications on your device, take
> a look:
> https://www.zoiper.com/en/support/home/article/205/Zoiper%20Push%20Proxy
>
> Regards
>
>
> On Wed, May 11, 2022 at 4:13 PM Rodrigo Pimenta Carvalho <
> pime...@inatel.br> wrote:
>
> Hi.
>
> My SIP proxy is an OpenSIPS version 2.4.7. It was 'off' for a while and
> today I turned it on again for some tests.
>
> I usually use my home office local WIFI to connect my softphones to the
> network and it can be all connected (online) to this SIP proxy.
>
> However, if I use the mobile network (LTE/4G) to connect the softphones to
> the SIP proxy, the OpenSIP returns an error 500. That is: "SIP/2.0 500
> Server error occurred (7/TM)".
>
> One of the softphones (Zoiper) allows me to use a kind of "proxy PUSH". If
> I use it, the problem is solved even while using the mobile network.
>
> What is a proxy PUSH?  Why OpenSIPs return error in a case, but not in the
> other one?
> What could I do to avoid using a 'proxy PUSH'?
>
> Local WIFI and mobile network come from different

Re: [OpenSIPS-Users] Question about error 500 only via WIFI

[Daniel Zanutti]

Olá Rodrigo, tudo bem? Saudações de São Paulo!

Opensips doesn't differentiate the network, it will look just to the sip
packet. I recommend you sniff through your packets and check what's
different. Probably there's somenthing on opensips log you didn't get yet,
recommend you take a look there first btw.

About push, I think you're enable push notifications on your device, take a
look:
https://www.zoiper.com/en/support/home/article/205/Zoiper%20Push%20Proxy

Regards


On Wed, May 11, 2022 at 4:13 PM Rodrigo Pimenta Carvalho 
wrote:

> Hi.
>
> My SIP proxy is an OpenSIPS version 2.4.7. It was 'off' for a while and
> today I turned it on again for some tests.
>
> I usually use my home office local WIFI to connect my softphones to the
> network and it can be all connected (online) to this SIP proxy.
>
> However, if I use the mobile network (LTE/4G) to connect the softphones to
> the SIP proxy, the OpenSIP returns an error 500. That is: "SIP/2.0 500
> Server error occurred (7/TM)".
>
> One of the softphones (Zoiper) allows me to use a kind of "proxy PUSH". If
> I use it, the problem is solved even while using the mobile network.
>
> What is a proxy PUSH?  Why OpenSIPs return error in a case, but not in the
> other one?
> What could I do to avoid using a 'proxy PUSH'?
>
> Local WIFI and mobile network come from different carriers.
>
> 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
>
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Re: [OpenSIPS-Users] Topology Hiding

[Daniel Zanutti]

https://www.opensips.org/Documentation/Tutorials-Topology-Hiding

On Tue, May 10, 2022 at 2:15 PM Saint Michael  wrote:

> Dear friends
> I am using opensips 3.1.9, with rtp proxy, and without topology hiding it
> would not talk to any carrier who has a Sonus box. I need to add topology
> hiding urgently and my support provider is missing in action. Can somebody
> provide instructions and code samples?
>
> Federico
>
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Re: [OpenSIPS-Users] incompatibility leads to massive CDR loss

[Daniel Zanutti]

I think this is your problem: branch=z9hG4bK-524287-1---b8aced18b4075aa3
*=49972*

You have char "=" inside a string, which is a reserved character and not
allowed on a string:
https://datatracker.ietf.org/doc/html/rfc3261#section-25.1

Should be something on client of your customer, since you received on 180
ringing, but i'm not sure if you can just solve it. It's violating RFC.


On Fri, May 6, 2022 at 4:42 PM Saint Michael  wrote:

> Dear friends
> Kindly look at the file attached. I am losing 10% of my CDR because some
> messages cannot be parsed by Opensips
>  opensips -V
> version: opensips 3.1.9 (x86_64/linux)
> flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC,
> Q_MALLOC, F_MALLOC, HP_MALLOC, DBG_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
> ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
> MAX_URI_SIZE 1024, BUF_SIZE 65535
> poll method support: poll, epoll, sigio_rt, select.
> git revision: 1a71fded7
> main.c compiled on 13:37:30 May  2 2022 with gcc 9
>
> I need some paid help generating a patch or fixing this somehow.
> I normally have a consultant but he may be tied up with the war and is not
> responding.
>
> Yours
> Federico
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Re: [OpenSIPS-Users] OpenSIPS : no 407 retransmission

[Daniel Zanutti]

Well, I don't have your script, cannot help further. If you are proxying,
you should proxy every message.

Maybe you are mixing stateful and stateless forwarding and are not handling
all scenarios?

It could be an Opensips problem, but again, don't know what you are doing
internally.



On Wed, May 4, 2022 at 2:19 PM Yannick LE COENT 
wrote:

> Hi Daniel,
>
> I do not think the ACK is sent by my script. It is sent by the TM module
> since it is a negative response.
> Am I wrong ?
>
> Thanks,
> Yannick
>
> Le 04/05/2022 à 18:48, Daniel Zanutti a écrit :
>
> Hi Yannick
>
> I think you should not reply with ACK to the 407 from destination. Just
> forward 407 to origin and wait for ACK. As soon you receive ACK from
> origin, you forward to destination.
>
> It's more like a stateless but I believe it's the only way.
>
> Regards
>
> On Tue, May 3, 2022 at 12:16 PM Yannick LE COENT <
> yannick.leco...@nexcom.fr> wrote:
>
>> Hello Daniel,
>>
>> This is not what I looking for.
>> My OpenSIPS instance is working as a relay between the softphone and
>> another proxy (proxy#2 in the call).
>> So it does not handle authentication.
>>
>> Alice   OpenSIPS  Proxy#2
>>| INVITE ||
>>|--->| INVITE |
>>|  100 Tring |--->|
>>|<---|407 |
>>||<---|
>>|| ACK|
>>||--->|
>>|407 ||
>>| X<-||
>>|  (no retrans.) |        |
>>
>> When the 407 is lost between OpenSIPS and Alice, it is not retransmitted
>> by OpenSIPS.
>>
>> I would like to force retransmission.
>>
>> Thanks,
>> Yannick
>>
>> Le 03/05/2022 à 15:16, Daniel Zanutti a écrit :
>>
>> Generate in Stateful -> www_challenge or proxy_challenge?
>> https://opensips.org/html/docs/modules/3.2.x/auth.html
>>
>> Is this what you are looking for?
>>
>>
>> On Tue, May 3, 2022 at 3:50 AM Yannick LE COENT <
>> yannick.leco...@nexcom.fr> wrote:
>>
>>> Hello all,
>>>
>>> Could you tell if there is a way to enable 407 in stateful mode ?
>>>
>>> Thanks,
>>> Yannick
>>>
>>> Le 30/04/2022 à 18:14, Yannick LE COENT a écrit :
>>>
>>> Hello Ben,
>>>
>>> Thanks for your answer.
>>>
>>> This problem occurs when OpenSIPS is not in charge of authenticating the
>>> INVITE request, but this is done downstream.
>>> I've sent this question to know if somebody has already solved this kind
>>> of problem.
>>>
>>> Best regards,
>>> Yannick
>>>
>>> Le 30/04/2022 à 16:15, Ben Newlin a écrit :
>>>
>>> I see. Apologies, I misunderstood the problem scenario.
>>>
>>>
>>>
>>> Ben Newlin
>>>
>>>
>>>
>>> *From: *Users 
>>>  on behalf of Yannick LE COENT
>>>  
>>> *Date: *Saturday, April 30, 2022 at 5:46 AM
>>> *To: *OpenSIPS users mailling list 
>>> 
>>> *Subject: *Re: [OpenSIPS-Users] OpenSIPS : no 407 retransmission
>>>
>>> *EXTERNAL EMAIL - Please use caution with links and attachments *
>>>
>>>
>>> --
>>>
>>> Hello Ben,
>>>
>>> The 407 is sent upstream, but when it is lost, it is not retransmitted
>>> by OpenSIPS.
>>> I do not have this problem with other negative status codes (e.g. 486).
>>>
>>> This is clearly explained in
>>> https://opensips.org/pub/opensips/1.8.6/src/ChangeLog
>>>
>>> 2012-03-21 18:36:58  Bogdan-Andrei Iancu, 
>>> * [8811] :
>>>
>>> TM will no longer do retransmission for the 407/401 replies (if no
>>> ACK is received) for both local or proxied replies.
>>>
>>> According to RFC 3261, retransmitting 407s/401s is probably a bad
>>> idea:
>>>
>>> 26.3.2.4 DoS Protection
>>>
>>> At the moment, my only solution is to use forward() instead of t_relay()
>>> in order to use the stateless mode.
>>>
>>> Yannick
>>>
>>>
>>> Yannick,
>>>
>>>
>>>
>>> The default behavior of OpenSIPS is to relay any received responses back 
>>> upstream

Re: [OpenSIPS-Users] OpenSIPS : no 407 retransmission

[Daniel Zanutti]

Hi Yannick

I think you should not reply with ACK to the 407 from destination. Just
forward 407 to origin and wait for ACK. As soon you receive ACK from
origin, you forward to destination.

It's more like a stateless but I believe it's the only way.

Regards

On Tue, May 3, 2022 at 12:16 PM Yannick LE COENT 
wrote:

> Hello Daniel,
>
> This is not what I looking for.
> My OpenSIPS instance is working as a relay between the softphone and
> another proxy (proxy#2 in the call).
> So it does not handle authentication.
>
> Alice   OpenSIPS  Proxy#2
>| INVITE ||
>|--->| INVITE |
>|  100 Tring |--->|
>|<---|407 |
>||<---|
>|| ACK|
>||--->|
>|407 ||
>| X<-||
>|  (no retrans.) ||
>
> When the 407 is lost between OpenSIPS and Alice, it is not retransmitted
> by OpenSIPS.
>
> I would like to force retransmission.
>
> Thanks,
> Yannick
>
> Le 03/05/2022 à 15:16, Daniel Zanutti a écrit :
>
> Generate in Stateful -> www_challenge or proxy_challenge?
> https://opensips.org/html/docs/modules/3.2.x/auth.html
>
> Is this what you are looking for?
>
>
> On Tue, May 3, 2022 at 3:50 AM Yannick LE COENT 
> wrote:
>
>> Hello all,
>>
>> Could you tell if there is a way to enable 407 in stateful mode ?
>>
>> Thanks,
>> Yannick
>>
>> Le 30/04/2022 à 18:14, Yannick LE COENT a écrit :
>>
>> Hello Ben,
>>
>> Thanks for your answer.
>>
>> This problem occurs when OpenSIPS is not in charge of authenticating the
>> INVITE request, but this is done downstream.
>> I've sent this question to know if somebody has already solved this kind
>> of problem.
>>
>> Best regards,
>> Yannick
>>
>> Le 30/04/2022 à 16:15, Ben Newlin a écrit :
>>
>> I see. Apologies, I misunderstood the problem scenario.
>>
>>
>>
>> Ben Newlin
>>
>>
>>
>> *From: *Users 
>>  on behalf of Yannick LE COENT
>>  
>> *Date: *Saturday, April 30, 2022 at 5:46 AM
>> *To: *OpenSIPS users mailling list 
>> 
>> *Subject: *Re: [OpenSIPS-Users] OpenSIPS : no 407 retransmission
>>
>> *EXTERNAL EMAIL - Please use caution with links and attachments *
>>
>>
>> --
>>
>> Hello Ben,
>>
>> The 407 is sent upstream, but when it is lost, it is not retransmitted by
>> OpenSIPS.
>> I do not have this problem with other negative status codes (e.g. 486).
>>
>> This is clearly explained in
>> https://opensips.org/pub/opensips/1.8.6/src/ChangeLog
>>
>> 2012-03-21 18:36:58  Bogdan-Andrei Iancu, 
>> * [8811] :
>>
>> TM will no longer do retransmission for the 407/401 replies (if no
>> ACK is received) for both local or proxied replies.
>>
>> According to RFC 3261, retransmitting 407s/401s is probably a bad
>> idea:
>>
>> 26.3.2.4 DoS Protection
>>
>> At the moment, my only solution is to use forward() instead of t_relay()
>> in order to use the stateless mode.
>>
>> Yannick
>>
>>
>> Yannick,
>>
>>
>>
>> The default behavior of OpenSIPS is to relay any received responses back 
>> upstream. If it is not doing that it would have to be because you are 
>> stopping it in the script. Take a look at the documentation for 
>> failure_route [1] which explains this. Check your own failure_route in your 
>> script; you must be doing something there that is telling OpenSIPS not to 
>> relay the 401/407 back upstream.
>>
>>
>>
>> [1] https://www.opensips.org/Documentation/Script-Routes-2-4#toc3
>>
>>
>>
>> Ben Newlin
>>
>>
>>
>> From: Users  
>>  on behalf of Yannick LE COENT 
>>  
>>
>> Date: Friday, April 29, 2022 at 6:44 PM
>>
>> To: users@lists.opensips.org  
>> 
>>
>> Subject: [OpenSIPS-Users] OpenSIPS : no 407 retransmission
>>
>> EXTERNAL EMAIL - Please use caution with links and attachments
>>
>>
>>
>> Hello,
>>
>>
>>
>> I'm using OpenSIPS as a proxy in front of another proxy. The 2nd proxy
>>
>> is in charge of authenticating the request.
>>
>>
>>
>> This is the callflow:
>>
>>
>>
>> Alice   OpenSIPS  Proxy#2

Re: [OpenSIPS-Users] OpenSIPS : no 407 retransmission

[Daniel Zanutti]

Generate in Stateful -> www_challenge or proxy_challenge?
https://opensips.org/html/docs/modules/3.2.x/auth.html

Is this what you are looking for?


On Tue, May 3, 2022 at 3:50 AM Yannick LE COENT 
wrote:

> Hello all,
>
> Could you tell if there is a way to enable 407 in stateful mode ?
>
> Thanks,
> Yannick
>
> Le 30/04/2022 à 18:14, Yannick LE COENT a écrit :
>
> Hello Ben,
>
> Thanks for your answer.
>
> This problem occurs when OpenSIPS is not in charge of authenticating the
> INVITE request, but this is done downstream.
> I've sent this question to know if somebody has already solved this kind
> of problem.
>
> Best regards,
> Yannick
>
> Le 30/04/2022 à 16:15, Ben Newlin a écrit :
>
> I see. Apologies, I misunderstood the problem scenario.
>
>
>
> Ben Newlin
>
>
>
> *From: *Users 
>  on behalf of Yannick LE COENT
>  
> *Date: *Saturday, April 30, 2022 at 5:46 AM
> *To: *OpenSIPS users mailling list 
> 
> *Subject: *Re: [OpenSIPS-Users] OpenSIPS : no 407 retransmission
>
> *EXTERNAL EMAIL - Please use caution with links and attachments *
>
>
> --
>
> Hello Ben,
>
> The 407 is sent upstream, but when it is lost, it is not retransmitted by
> OpenSIPS.
> I do not have this problem with other negative status codes (e.g. 486).
>
> This is clearly explained in
> https://opensips.org/pub/opensips/1.8.6/src/ChangeLog
>
> 2012-03-21 18:36:58  Bogdan-Andrei Iancu, 
> * [8811] :
>
> TM will no longer do retransmission for the 407/401 replies (if no ACK
> is received) for both local or proxied replies.
>
> According to RFC 3261, retransmitting 407s/401s is probably a bad idea:
>
> 26.3.2.4 DoS Protection
>
> At the moment, my only solution is to use forward() instead of t_relay()
> in order to use the stateless mode.
>
> Yannick
>
>
> Yannick,
>
>
>
> The default behavior of OpenSIPS is to relay any received responses back 
> upstream. If it is not doing that it would have to be because you are 
> stopping it in the script. Take a look at the documentation for failure_route 
> [1] which explains this. Check your own failure_route in your script; you 
> must be doing something there that is telling OpenSIPS not to relay the 
> 401/407 back upstream.
>
>
>
> [1] https://www.opensips.org/Documentation/Script-Routes-2-4#toc3
>
>
>
> Ben Newlin
>
>
>
> From: Users  
>  on behalf of Yannick LE COENT 
>  
>
> Date: Friday, April 29, 2022 at 6:44 PM
>
> To: users@lists.opensips.org  
> 
>
> Subject: [OpenSIPS-Users] OpenSIPS : no 407 retransmission
>
> EXTERNAL EMAIL - Please use caution with links and attachments
>
>
>
> Hello,
>
>
>
> I'm using OpenSIPS as a proxy in front of another proxy. The 2nd proxy
>
> is in charge of authenticating the request.
>
>
>
> This is the callflow:
>
>
>
> Alice   OpenSIPS  Proxy#2
>
>| INVITE ||
>
>|--->| INVITE |
>
>|  100 Tring |--->|
>
>|<---|407 |
>
>||<---|
>
>|| ACK|
>
>||--->|
>
>|407 ||
>
>| X<-||
>
>|||
>
>
>
> Since OpenSIPS does not retransmit 401/407, the call setup gets stuck.
>
>
>
> What can I do ?
>
> If I set auto_100trying=1, that works, but this increases the number of
>
> INVITE retransmissions since 180Ringing are not received instantly.
>
>
>
> Do you have any suggestion ?
>
>
>
> Thanks,
>
> Yannick
>
>
> ___
> Users mailing 
> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
> ___
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>
>
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Re: [OpenSIPS-Users] strange INVITE transmission

[Daniel Zanutti]

Hi Kiwon

You applied at beginning, before loose_route. Not sure it gonna work this
way, i sent you exactly line where to put the code.
Move this:
 if (nat_uac_test(23)) {
if (is_method("REGISTER")) {
fix_nated_register();
setbflag("NAT");
} else {
fix_nated_contact();
setflag("NAT");
}
}

After:
t_check_trans();

Anyway, I suggest you add some log to confirm messages are coming. Put this
line right after main route:
xlog("L_ERR","MESSAGE RECEIVED $rm [$fu/$tu/$ru/$ci/$si]");

There's no big deal on Opensips, everything comes to main route, internal
transaction responses comes from specific routes. (
https://www.opensips.org/Documentation/Script-Routes-3-1)

I have some spare time tomorrow, send me a direct message if you need help.


On Tue, Mar 29, 2022 at 12:43 AM 이기원  wrote:

> Hi Daniel and opensips users goup,
> Thank you for answering my questions.
>
> I missed to inform you about the opensips version I'm trying. It is 3.2.5.
>
> After modifying config, opensips does not respond for the REGISTER
> requests from phones.
> Actually I already tried the following
> https://kb.smartvox.co.uk/opensips/nat-contact-and-via-fixing-in-sip-part-3/
> article but I have the same problem - no response for REGISTERs.
>
> Is there any way to know why opensips ignores or does not respond for
> REGISTERs?
> Please find my new opensips.cfg that Diniel's advice is applied.
>
>
> Thank you
>
> Regards
> Kiwon
>
> 2022년 3월 28일 (월) 오후 10:47, Daniel Zanutti 님이 작성:
>
>> Hi Kiwon
>>
>> You need to handle NAT scenarios. Try putting this code on line 254,
>> right after "t_check_trans()":
>>
>> if (nat_uac_test("7"))
>> {
>>   #nathelper
>>   if(is_method("REGISTER"))
>> fix_nated_register();
>>   else
>> fix_nated_contact();
>>   xlog("L_NOTICE", "Fix contact - M=$rm RURI=$ru F=$fu T=$tu IP=$si
>> ID=$ci\n");
>> }
>>
>> You also need to enable nathelper module. The reason is that you need to
>> use the public IP/Port that sent data to opensips and ignore the Contact.
>>
>>
>> On Mon, Mar 28, 2022 at 5:20 AM 이기원  wrote:
>>
>>> Hi guys, I'm trying to introduce opensips into my company these days.
>>>
>>> I'm testing with many other phones. During the test, I had a very
>>> strange symptom.
>>>
>>> Please, take look at the picture below first. (Please find my screenshot
>>> from the link below)
>>> A. The most left one 14.52.252.236 is a phone (with hardware).
>>> B. 10.0.0.177 is proxy which is woking on cloud and its external IP is
>>> 58.79.209.75
>>> C. 175.223.34.31 is a zoiper softphone which is working on iphone
>>> D. 192.168.10.187 is the private IP address of A - a phone which is
>>> the most left one (14.52.252.236)
>>>
>>>
>>>
>>>
>>> https://drive.google.com/file/d/14zAREWLsluIa1TcU7tZJLSgff-iPqBDA/view?usp=sharing
>>>
>>> As you can see, C is calling A but opensips transmits the INVITE rqeuset
>>> to A's private IP address.
>>> INVITE request should be transmitted A's public IP address - In this
>>> scenario 14.52.252.236 (A)
>>>
>>> I also attach my opensips.cfg. What's wrong with me?
>>>
>>> Thank you
>>>
>>>
>>> Regards
>>> Kiwon
>>> ___
>>> Users mailing list
>>> Users@lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>> ___
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>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
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Re: [OpenSIPS-Users] strange INVITE transmission

[Daniel Zanutti]

Hi Kiwon

You need to handle NAT scenarios. Try putting this code on line 254, right
after "t_check_trans()":

if (nat_uac_test("7"))
{
  #nathelper
  if(is_method("REGISTER"))
fix_nated_register();
  else
fix_nated_contact();
  xlog("L_NOTICE", "Fix contact - M=$rm RURI=$ru F=$fu T=$tu IP=$si
ID=$ci\n");
}

You also need to enable nathelper module. The reason is that you need to
use the public IP/Port that sent data to opensips and ignore the Contact.


On Mon, Mar 28, 2022 at 5:20 AM 이기원  wrote:

> Hi guys, I'm trying to introduce opensips into my company these days.
>
> I'm testing with many other phones. During the test, I had a very strange
> symptom.
>
> Please, take look at the picture below first. (Please find my screenshot
> from the link below)
> A. The most left one 14.52.252.236 is a phone (with hardware).
> B. 10.0.0.177 is proxy which is woking on cloud and its external IP is
> 58.79.209.75
> C. 175.223.34.31 is a zoiper softphone which is working on iphone
> D. 192.168.10.187 is the private IP address of A - a phone which is
> the most left one (14.52.252.236)
>
>
>
>
> https://drive.google.com/file/d/14zAREWLsluIa1TcU7tZJLSgff-iPqBDA/view?usp=sharing
>
> As you can see, C is calling A but opensips transmits the INVITE rqeuset
> to A's private IP address.
> INVITE request should be transmitted A's public IP address - In this
> scenario 14.52.252.236 (A)
>
> I also attach my opensips.cfg. What's wrong with me?
>
> Thank you
>
>
> Regards
> Kiwon
> ___
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>
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Re: [OpenSIPS-Users] SIP Reg with radius

[Daniel Zanutti]

Take a look here: https://www.opensips.org/Documentation/Tutorials-Radius


On Sat, Feb 12, 2022 at 1:16 PM Vishal Pai  wrote:

> Hello Team
>
> I am new to Opensips. Can we have the sip registration to lookup for auth
> in Radius if yes then we can forward the sip invite to PBX with a unique
> header.
>
> Is there any document in opensips for reference with radius for
> authentication.
>
> Thanks
> ___
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>
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Re: [OpenSIPS-Users] BYE from UAC bypasses OpenSIPS

[Daniel Zanutti]

John

I highly recommend using the topology hiding module instead of inserting
routes and forwarding the SIP message. Several IP devices have problems
when you have a lot of routes. Even the SIP message size can be a problem
if your call flows through several proxies.

When you use topology hiding, your origin/destination will never know
what's on the other side, so current problem can never occur anymore.

Regards


On Wed, Sep 1, 2021 at 10:12 AM John Burke via Users <
users@lists.opensips.org> wrote:

> Hey Alexey,
>
> From your screenshots, it's not only the BYE but also the ACK that
> bypasses your proxy (both of which being sequential requests).  To get your
> proxy in the path of sequential reqs, you need to either (1) call
> record_route, or (2) use topology hiding module.  The UAC otherwise doesn't
> know about your proxy, as the 200 won't contain any Record-Route headers to
> build the route set for sequential reqs nor will the 200 contain a URI
> referencing your proxy in the Contact header.
>
> Thanks,
> John Burke
>
> On 9/1/21 6:48 AM, Alexey Kazantsev via Users wrote:
>
> Hi list,
>
> feeling shamefully but still can not fix the problem, trying different
> configurations.
>
> I’m setting drouting (INVITEs are routed well).
> But BYEs from UAC go directly to dr_gateway, not through OpenSIPS.
>
> I understand that this all is about in-dialog request, having to_tag,
> loose_route.
> I compared my config with config files of previously configured servers
> (not for drouting)
> which work well (I tested BYE routing).
>
> But still can not fix this issue.
>
> Here’re 2 screenshots of the same call, captured on UAC:
> [1] demonstrating INVITE from UAC,
> [2] demonstrating BYE from UAC (which goes directly, bypassing OpenSIPS,
> and I need to fix this behaviour).
>
> [1] https://ibb.co/NNNvHhM
> [2] https://ibb.co/bKd4c0r
>
> What should I pay attention to, to route BYE requests from UAC via
> OpenSIPS?
>
> ---
> BR, Alexey
> http://alexeyka.zantsev.com/
>
> ___
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>
>
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Re: [OpenSIPS-Users] learning the realm from authentication challenges

[Daniel Zanutti]

Don't forget to deal with CSEQ increment on the authenticated INVITE.

Also we had problems when any in-dialog message is received, we have to
deal with CSEQ on all of them. =(

On Fri, Sep 25, 2020 at 12:30 PM johan  wrote:

> Jeff, be warned that the datafill for registrar is not obvious.
> On 25/09/2020 16:40, Jeff Pyle wrote:
>
> I am not route-advancing in a typical way, so my application of
> credentials is a bit different perhaps.
>
> The environment I'm in has a variety of customer-facing platforms, over a
> dozen at last count.  Some are for trunking, some hosted, some hybrid.  The
> platform I'm writing on OpenSIPS is a testing one that will allow us to
> send and receive test calls to and from all of them.  So, rather than
> having a bunch of registrations on every test phone for every person who
> might want to test, this allows each person to have one appearance to this
> platform and select which upstream platform they want to send a call to via
> dialed prefixes.
>
> I use the uac_registrant module, and its registrant table, to handle the
> platforms that require registrations and it works excellently.  At call
> time, I'm working on the scripting right now that will query the registrant
> table for the appropriate credentials based on where we've sent the call
> and apply them in the failure_route upon receiving a 401 or 407.
>
> Think of it this way:  when you configure a gateway in FreeSWITCH or a SIP
> peer in Asterisk's chan_sip, do you need to define the realm ahead of
> time?  No, you don't care; it's just a mechanism under the hood that's
> necessary to complete the transaction.  That's where I'm at in OpenSIPS.
> With Johan's parsing it looks like I'm about there, too.  Friggin' regex
> gets me every time.
>
>
> - Jeff
>
> On Fri, Sep 25, 2020 at 10:25 AM Ben Newlin 
> wrote:
>
>> I think you do need to have credentials associated with the different
>> routes you have and load those properly. From your description, however, I
>> don’t understand why it is dependent on identifying the realm in the
>> response. If multiple downstream servers are all using the same realm (but
>> have different credentials?) then how are you differentiating based on the
>> realm value?
>>
>>
>>
>> The idea with uac_auth is that when you send, for example, to server
>> broadworks1 you would load all the possible valid credentials for
>> broadworks1, including the realm it will challenge with. When you then call
>> uac_auth() from failure route, it will look through all the loaded
>> credentials for one with a matching realm to the broadworks1 challenge and
>> use that. If the call fails for any reason to broadworks1 and then you
>> decide to route to server asterisk1, you would load all the possible
>> credentials for that server into the auth AVPs the same way and failure
>> route handling is the same.
>>
>>
>>
>> You could very well have a use case for verifying the realm in
>> failure_route; I’m not saying you don’t. I don’t see it from what you’ve
>> described, but I may be missing something. I think the reason there is no
>> variable for pulling the challenge realm value directly is because normally
>> with this mechanism it shouldn’t be needed.
>>
>>
>>
>> I would appreciate if someone could confirm that uac_auth() will match
>> the realm as I’m asserting. I’m 95% sure this is how it worked in my
>> testing, but that was a while ago and as I said the realm matching doesn’t
>> appear to be documented. I’d hate to be steering you down a wrong path.
>>
>>
>>
>> Ben Newlin
>>
>>
>>
>> *From: *Users 
>> *Date: *Friday, September 25, 2020 at 10:15 AM
>> *To: *OpenSIPS users mailling list 
>> *Subject: *Re: [OpenSIPS-Users] learning the realm from authentication
>> challenges
>>
>> Johan,
>>
>>   I will definitely try that.  Thank you!
>>
>>
>>
>> Ben,
>>
>>   The problem is I have multiple destinations with the same realm.  In my
>> case, several different Broadworks app servers.  I haven't checked them
>> exhaustively but I think they all reply with realm="BroadWorks" in their
>> authentication headers.  I've got some Asterisk boxes in here, and I think
>> they're all the domain of the SIP request URI in the case of an INVITE.  I
>> think I'll have to choose ahead of time which credentials go with which
>> route, no?  Unless I'm still not wrapping my head around how this is
>> supposed to work.
>>
>>
>>
>>
>>
>> - Jeff
>>
>>
>>
>>
>>
>>
>>
>>
>>
>> On Fri, Sep 25, 2020 at 9:22 AM Ben Newlin 
>> wrote:
>>
>> Jeff,
>>
>>
>>
>> My point was that the uac_auth() is supposed to handle the realm matching
>> for you. If you simply load all of the auth data based on the call target
>> as you already plan to do, uac_auth() should look through that data for you
>> to find credentials with a matching realm. You don’t need to do that part
>> yourself in the script.
>>
>>
>>
>> Ben Newlin
>>
>>
>>
>> *From: *Users 
>> *Date: *Thursday, September 24, 2020 at 11:14 PM
>> *To: *OpenSIPS users mailling list 
>> 

Re: [OpenSIPS-Users] Maybe it's a bug

[Daniel Zanutti]

Hi folks

We implemented millisecond billing in our platform, so no need to round on
the Opensips layer, the rounding is done in our business billing layer.
This way customers can have a different rounding than VoIP providers. It's
not a way to penalize customers, but some providers just work differently
than others and you cannot demand them to work "your way".

I suggest store a float (instead of integer) with milliseconds included OR
an option to store milliseconds on a separated column.

Regards


On Tue, Jun 9, 2020 at 4:45 PM Calvin Ellison 
wrote:

> +1 to ceil() rounding, and stating in the documentation that this is
> the method used. Alternatively, some new option to specify ceiling,
> floor, round, truncate, etc.
>
> I can back up SM's claim that a single billing interval discrepancy
> will cost people real money. Clarification in the documentation will
> help people avoid that pitfall. I also concur with Vlad that the
> duration in milliseconds is preferable. The millisecond data can help
> to settle any billing disputes from clients or vendors, and it
> demystifies CDRs for everyone: one field for actual call duration in
> ms, another field for call duration after rounding, and/or one for the
> final charged call duration after rounding and billing interval.
>
>
>
> Regards,
>
> Calvin Ellison
> Senior Voice Operations Engineer
> calvin.elli...@voxox.com
>
>
> On Tue, Jun 9, 2020 at 12:14 PM Johan De Clercq  wrote:
> >
> > Upwards seems best.
>
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Re: [OpenSIPS-Users] Opensips + rtpproxy issue

[Daniel Zanutti]

Hi Miha

If you are seeing issues on A leg, RTPPROXY didn't even processed the call,
right? Probably you have bandwidth limitation or maybe I/O problems on your
server.

Start checking bandwidth.

Regards

On Tue, May 5, 2020 at 2:13 PM  wrote:

> Hello Danies
>
>- what do you mean by enough max open do files? I do no linit or set
>anything
>- I traced with tshark and i can see issue with A and B leg
>
>
> Thank you for help!
> Br
> Miha
>
> Miha
> On 5 May 2020, 16:07 +0200, Daniel Zanutti ,
> wrote:
>
> No special configuration, we just set IP's and ports.
>
> Since CPU is not your problem, I believe you have some kind of bandwidth
> limitation in your network.
>
> I suggest you confirm:
> 1) You have enough max open files in your rtpproxy process ->
> /proc/PID/limits
> 2) Where the bottleneck is: CPU, IO or bandwidth. You can record some
> packets in wireshark inside RTPPROXY machine and confirm audio is
> distorted before and after rtpproxy.
>
> Regards
>
>
> On Tue, May 5, 2020 at 10:35 AM Miha  wrote:
>
>> Hi,
>>
>> no CPU usage is around 1% to 5%, basically nothing.
>> In sound there is big distortion it is impossibly to
>> comunicate with each other.
>>
>> We have two cors deticated to it. Do you have any special
>> thing set on it?
>>
>> tnx
>> miha
>>
>> On Tue, 5 May 2020 10:27:22 -0300
>>  Daniel Zanutti  wrote:
>> > Hi Miha
>> >
>> > Could you explaining how does it break? We use it in
>> > virtual machines and
>> > our safe limit is around 500 simultaneous calls, on
>> > dedicated single core
>> > VPS. Does CPU usage reach 100%?
>> >
>> >
>> >
>> > On Tue, May 5, 2020 at 10:11 AM Miha via Users
>> > 
>> > wrote:
>> >
>> > > Hello
>> > >
>> > > we have virtualized opensips and rtpproxy running on
>> > the same server which
>> > > is virtualized in vmware infrastructure. Servers are
>> > not old, also traffic
>> > > is not so big (cca 50 simultaneous calls). when there
>> > is a peak cca 80
>> > > simultaneous calls RTP starts to break.
>> > >
>> > > is there any special setting/flag to be set, so that I
>> > can optimze this?
>> > > load on VM is very low.
>> > >
>> > > rtpproxy -version
>> > > Basic version: 20040107
>> > >
>> > > Opensips is 2.1
>> > >
>> > >
>> > > thank you for help.
>> > > Miha
>> > > ___
>> > > Users mailing list
>> > > Users@lists.opensips.org
>> > >
>> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>> > >
>>
>>
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Re: [OpenSIPS-Users] Opensips + rtpproxy issue

[Daniel Zanutti]

No special configuration, we just set IP's and ports.

Since CPU is not your problem, I believe you have some kind of bandwidth
limitation in your network.

I suggest you confirm:
1) You have enough max open files in your rtpproxy process ->
/proc/PID/limits
2) Where the bottleneck is: CPU, IO or bandwidth. You can record some
packets in wireshark inside RTPPROXY machine and confirm audio is
distorted before and after rtpproxy.

Regards


On Tue, May 5, 2020 at 10:35 AM Miha  wrote:

> Hi,
>
> no CPU usage is around 1% to 5%, basically nothing.
> In sound there is big distortion it is impossibly to
> comunicate with each other.
>
> We have two cors deticated to it. Do you have any special
> thing set on it?
>
> tnx
> miha
>
> On Tue, 5 May 2020 10:27:22 -0300
>  Daniel Zanutti  wrote:
> > Hi Miha
> >
> > Could you explaining how does it break? We use it in
> > virtual machines and
> > our safe limit is around 500 simultaneous calls, on
> > dedicated single core
> > VPS. Does CPU usage reach 100%?
> >
> >
> >
> > On Tue, May 5, 2020 at 10:11 AM Miha via Users
> > 
> > wrote:
> >
> > > Hello
> > >
> > > we have virtualized opensips and rtpproxy running on
> > the same server which
> > > is virtualized in vmware infrastructure. Servers are
> > not old, also traffic
> > > is not so big (cca 50 simultaneous calls). when there
> > is a peak cca 80
> > > simultaneous calls RTP starts to break.
> > >
> > > is there any special setting/flag to be set, so that I
> > can optimze this?
> > > load on VM is very low.
> > >
> > > rtpproxy -version
> > > Basic version: 20040107
> > >
> > > Opensips is 2.1
> > >
> > >
> > > thank you for help.
> > > Miha
> > > ___
> > > Users mailing list
> > > Users@lists.opensips.org
> > >
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> > >
>
>
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Re: [OpenSIPS-Users] Opensips + rtpproxy issue

[Daniel Zanutti]

Hi Miha

Could you explaining how does it break? We use it in virtual machines and
our safe limit is around 500 simultaneous calls, on dedicated single core
VPS. Does CPU usage reach 100%?



On Tue, May 5, 2020 at 10:11 AM Miha via Users 
wrote:

> Hello
>
> we have virtualized opensips and rtpproxy running on the same server which
> is virtualized in vmware infrastructure. Servers are not old, also traffic
> is not so big (cca 50 simultaneous calls). when there is a peak cca 80
> simultaneous calls RTP starts to break.
>
> is there any special setting/flag to be set, so that I can optimze this?
> load on VM is very low.
>
> rtpproxy -version
> Basic version: 20040107
>
> Opensips is 2.1
>
>
> thank you for help.
> Miha
> ___
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> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
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Re: [OpenSIPS-Users] How to check the transport of the RTP sessions?

[Daniel Zanutti]

He didn't said SDP, he said RTP Sessions.
Opensips cannot inspect rtp sessions.

On Fri, Jan 31, 2020 at 11:09 AM David Villasmil <
david.villasmil.w...@gmail.com> wrote:

> You can also use the textops’ search function.
>
>
> On Fri, 31 Jan 2020 at 13:43, Daniel Zanutti 
> wrote:
>
>> Hi
>>
>> Are you using just Opensips or some RTP proxy solution? If you are using
>> just Opensips, the RTP traffic will be Peer-to-peer and you have to
>> monitore origin ou destination.
>>
>> If you are using some RTP proxy solution, just check on this machine.
>>
>> Regards
>>
>> On Fri, Jan 31, 2020 at 7:33 AM Abdoul Osséni 
>> wrote:
>>
>>> Hello,
>>>
>>> I try to detect the transport of the RTP session.
>>>
>>> Examples:detect if
>>> - RTP/SAVP
>>> - RTP/SAVPF
>>> - RTP/AVP
>>> - RTP/AVPF
>>>
>>> Can you help me?
>>> Regards
>>>
>>> Abdoul
>>> AfriCallShop
>>> https://www.africallshop.com/
>>> ___
>>> 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
>>
> --
> Regards,
>
> David Villasmil
> email: david.villasmil.w...@gmail.com
> phone: +34669448337
> ___
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>
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Re: [OpenSIPS-Users] How to check the transport of the RTP sessions?

[Daniel Zanutti]

Hi

Are you using just Opensips or some RTP proxy solution? If you are using
just Opensips, the RTP traffic will be Peer-to-peer and you have to
monitore origin ou destination.

If you are using some RTP proxy solution, just check on this machine.

Regards

On Fri, Jan 31, 2020 at 7:33 AM Abdoul Osséni 
wrote:

> Hello,
>
> I try to detect the transport of the RTP session.
>
> Examples:detect if
> - RTP/SAVP
> - RTP/SAVPF
> - RTP/AVP
> - RTP/AVPF
>
> Can you help me?
> Regards
>
> Abdoul
> AfriCallShop
> https://www.africallshop.com/
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
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Re: [OpenSIPS-Users] How to limit parallel calls duration of prepaid customers?

[Daniel Zanutti]

Hi  Diptesh

We tried to implement a native prepaid system on Opensips but didn't found
a way to do this natively, so we developed a custom prepaid mechanism to
our solution.

Our company (http://dazsoft.com) is focused on complete systems but we can
negotiate this specific part if you want. Let me know.

Regards


On Sat, Nov 16, 2019 at 1:50 PM Dipteshkumar Patel <
diptesh.pa...@ecosmob.com> wrote:

> Hello Team,
>
> I want to use opensips as a pbx system. I have prepaid customers so how
> can i manage prepaid calls scheduling based on customers' balance(dialog
> timeout). As specially in case we have parallel calls of a user. I found
> call-control from ag-projects for that.
>
> Can you please suggest any other possible way to implement this feature?
>
> Thanks & Regards
> *Diptesh Patel*
> Software Developer
> Ecosmob Technologies Ltd,
> Ahmedabad
> Mo:*+919898962659*
>
> *Disclaimer*
> In addition to generic Disclaimer which you have agreed on our website,
> any views or opinions presented in this email are solely those of the
> originator and do not necessarily represent those of the Company or its
> sister concerns. Any liability (in negligence, contract or otherwise)
> arising from any third party taking any action, or refraining from taking
> any action on the basis of any of the information contained in this email
> is hereby excluded.
>
> *Confidentiality*
> This communication (including any attachment/s) is intended only for the
> use of the addressee(s) and contains information that is PRIVILEGED AND
> CONFIDENTIAL. Unauthorized reading, dissemination, distribution, or copying
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Re: [OpenSIPS-Users] SIP Proxy and Paid Support

[Daniel Zanutti]

Hi Rick

I have a lot of experience on Opensips, maybe I can take a look at your
project.

Let me know if interested.

Thanks


On Mon, Sep 24, 2018 at 1:06 AM Alexander Jankowsky 
wrote:

>
>
> Hello Rick,
>
>
>
> There are some books around with the fundamentals so you can experiment
> and learn with the basics.
>
> There are formal courses some free and further higher levels for a fee so
> you can learn more formally.
>
>
>
> What is the time frame within which all of this has to happen or at least
> be started as work in progress.
>
> How many exactly do you mean by multiple, tens of, hundreds of, it would
> help to be a little more specific.
>
>
>
> There is a fair amount of documentation available to search out. Sometimes
> things move very quickly.
>
> In some cases the documentation is version specific and may not have
> caught up with what is being used.
>
> What version have you tried on what platform where have you specifically
> been running into problems.
>
>
>
> Alex
>
>
>
> *From:* Users [mailto:users-boun...@lists.opensips.org] *On Behalf Of *Rick
> Pearce
> *Sent:* Monday, 24 September 2018 8:45 AM
> *To:* users@lists.opensips.org
> *Subject:* [OpenSIPS-Users] SIP Proxy and Paid Support
>
>
>
> Hi All,
>
>
>
> I'm trying to contact someone at OpenSIPS to assist in setting up a
> carrier solution using the OpenSIPS proxy. I need OpenSIPS sitting behind a
> firewall on a 1to1 NAT with multiple individual customer FreePBXs on the
> same private subnet
>
>
>
> Is there a modern setup tutorial available for this type of solution using
> the newer versions?
>
>
>
> I would also like to engage paid support however the contact form doesn't
> appear to respond
>
>
>
> The form at http://www.opensips-solutions.com/contact.html freezes and
> there doesn't appear to be another contact address other than the mailing
> list, IRC or facebook. Unless of course I have read this completely wrong :/
>
>
>
> Please advise the support email so I can engage paid support, discuss an
> ongoing support contract plus training - sorry for the using the list
>
>
>
> Thanks and regards
>
> Rick
> ___
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Re: [OpenSIPS-Users] Doubts on call-center scenario

[Daniel Zanutti]

Got it!

Thanks

On Fri, Sep 21, 2018 at 6:12 AM Bogdan-Andrei Iancu 
wrote:

> Hi Daniel,
>
> OK, so you want to get the parameter into the local route (when the calls
> are sent out). Right now, the out of the box solution is what you already
> did - put the scenario parameters into RURI so you can extract them in the
> local route.
> This is ugly, let me think of a better solution, correlated with the b2b
> rework (in order to replace the xml control with the scripting control).
>
> Best regards,
>
> Bogdan-Andrei Iancu
>
> OpenSIPS Founder and Developer
>   http://www.opensips-solutions.com
> OpenSIPS Bootcamp 2018
>   http://opensips.org/training/OpenSIPS_Bootcamp_2018/
>
> On 09/20/2018 09:32 PM, Daniel Zanutti wrote:
>
> Hi Bogdan
>
> I'm triggering the script via MI. The idea is to send some parameter with
> the MI parameters and use this as a variable on localroute.
>
> For example, a dialer place call to B destination, after answer, place
> call to A destination, then bridge.
>
> On B side, i need to store on DB the gateway_id that will process the
> call. On A side, I'll call the Queue number, but I need to send and store
> the campaign ID. These 2 values must be stored on CDR.
>
> I didn't find a way to send, for example, the gateway_id to the opensips
> script when using B2B.
>
> Thanks
>
> On Thu, Sep 20, 2018 at 12:48 PM Bogdan-Andrei Iancu 
> wrote:
>
>> Hi Daniel,
>>
>> Could you detail a bit on the flow of this customer related value ? where
>> does is originate (when triggering the b2b script via MI?) and where do you
>> want to get it ?
>>
>> Best regards,
>>
>> Bogdan-Andrei Iancu
>>
>> OpenSIPS Founder and Developer
>>   http://www.opensips-solutions.com
>> OpenSIPS Bootcamp 2018
>>   http://opensips.org/training/OpenSIPS_Bootcamp_2018/
>>
>> On 09/12/2018 12:32 AM, Daniel Zanutti wrote:
>>
>> Hi everyone,
>>
>> I'm using opensips to originate a call to 2 destinations then bridge
>> then, using B2B scenario.
>>
>> How to send some custom parameters to help accounting?
>> I need to identify that this specific call, is related to some customer.
>> Didn't find in docs a proper way to do it, so my idea is to set r-uri with
>> some kind of csv data then fix it before it's sent out on local_route. But
>> I'm unsure this is a good idea.
>>
>> Is there a proper way to send custom parameter to be handled on
>> local_route?
>>
>> Thanks
>>
>>
>>
>> ___
>> Users mailing 
>> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>>
>
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Re: [OpenSIPS-Users] Doubts on call-center scenario

[Daniel Zanutti]

Hi Bogdan

I'm triggering the script via MI. The idea is to send some parameter with
the MI parameters and use this as a variable on localroute.

For example, a dialer place call to B destination, after answer, place call
to A destination, then bridge.

On B side, i need to store on DB the gateway_id that will process the call.
On A side, I'll call the Queue number, but I need to send and store the
campaign ID. These 2 values must be stored on CDR.

I didn't find a way to send, for example, the gateway_id to the opensips
script when using B2B.

Thanks

On Thu, Sep 20, 2018 at 12:48 PM Bogdan-Andrei Iancu 
wrote:

> Hi Daniel,
>
> Could you detail a bit on the flow of this customer related value ? where
> does is originate (when triggering the b2b script via MI?) and where do you
> want to get it ?
>
> Best regards,
>
> Bogdan-Andrei Iancu
>
> OpenSIPS Founder and Developer
>   http://www.opensips-solutions.com
> OpenSIPS Bootcamp 2018
>   http://opensips.org/training/OpenSIPS_Bootcamp_2018/
>
> On 09/12/2018 12:32 AM, Daniel Zanutti wrote:
>
> Hi everyone,
>
> I'm using opensips to originate a call to 2 destinations then bridge then,
> using B2B scenario.
>
> How to send some custom parameters to help accounting?
> I need to identify that this specific call, is related to some customer.
> Didn't find in docs a proper way to do it, so my idea is to set r-uri with
> some kind of csv data then fix it before it's sent out on local_route. But
> I'm unsure this is a good idea.
>
> Is there a proper way to send custom parameter to be handled on
> local_route?
>
> Thanks
>
>
>
> ___
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> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
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[OpenSIPS-Users] Doubts on call-center scenario

[Daniel Zanutti]

Hi everyone,

I'm using opensips to originate a call to 2 destinations then bridge then,
using B2B scenario.

How to send some custom parameters to help accounting?
I need to identify that this specific call, is related to some customer.
Didn't find in docs a proper way to do it, so my idea is to set r-uri with
some kind of csv data then fix it before it's sent out on local_route. But
I'm unsure this is a good idea.

Is there a proper way to send custom parameter to be handled on local_route?

Thanks
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Re: [OpenSIPS-Users] Doubt about call center module

[Daniel Zanutti]

You are correct, sorry.

I'll fix and start testing again.

Thanks

On Fri, Aug 31, 2018 at 10:10 AM Bogdan-Andrei Iancu 
wrote:

> As I said, in the cc_flows, you have no value for the "message_queue"
> column - this is a must, it has to be an URL to provide playback for the
> call queuing.
>
> Regards,
>
> Bogdan-Andrei Iancu
>
> OpenSIPS Founder and Developer
>   http://www.opensips-solutions.com
> OpenSIPS Bootcamp 2018
>   http://opensips.org/training/OpenSIPS_Bootcamp_2018/
>
> On 08/31/2018 04:06 PM, Daniel Zanutti wrote:
>
> Hi Bogdan
>
> Here it is table cc_flows:
> id  flowid  priority  skillprependcid  message_welcome
> message_queue
> --  --    ---  --  ---
> ---
>  1  fila-1   256  suporte  fila-1
>
>
> Also table agents:
> id  agentid location logstate
> skills   last_call_end
> --  --  ---  
> ---  ---
>  1  1...@plat5.domain.com  sip:1...@plat5.domain.com:5060 1
> suporte   1535650312
>
> Thanks
>
> On Fri, Aug 31, 2018 at 5:02 AM Bogdan-Andrei Iancu 
> wrote:
>
>> Hi Daniel,
>>
>> It is not about the B2B scenario, but about how you provisioned the flow
>> in DB. Could you simply dump the output of "select * from cc_flows" ?
>>
>> Regards,
>>
>> Bogdan-Andrei Iancu
>>
>> OpenSIPS Founder and Developer
>>   http://www.opensips-solutions.com
>> OpenSIPS Bootcamp 2018
>>   http://opensips.org/training/OpenSIPS_Bootcamp_2018/
>>
>> On 08/30/2018 08:34 PM, Daniel Zanutti wrote:
>>
>> Hi Bogdan
>>
>> Yes, It's the same scenario and same message. The call flow is:
>>
>> Asterisk Dials(port 5070) -> Opensips (port 5060) forward to Queue ->
>> Calls local user
>>
>> I'm using standard Queue scenario:
>> 
>> 
>> 
>> 
>> 
>> server1
>> 
>> 
>> client1
>> message
>> 
>> 1
>> 
>> 
>> 
>> 1
>> 
>> 
>>
>> And SIP message is the same on all calls, just changed Call-id/tags:
>>
>> U 10.10.10.10:5070 -> 10.10.10.10:5060
>> INVITE sip:fila-1@10.10.10.10:5060 SIP/2.0.
>> Via: SIP/2.0/UDP 10.10.10.10:5070;branch=z9hG4bK2abb2acc;rport.
>> Max-Forwards: 70.
>> From: ;tag=as6440e239.
>> To: .
>> Contact: .
>> Call-ID: 357cf76348e4e68325d065e85282320a@10.10.10.10:5070.
>> CSeq: 102 INVITE.
>> User-Agent: PBX SIPTEK.
>> Date: Thu, 30 Aug 2018 17:30:30 GMT.
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
>> PUBLISH, MESSAGE.
>> Supported: replaces, timer.
>> P-Asserted-Identity: "5511" .
>> Content-Type: application/sdp.
>> Content-Length: 353.
>> [SDP OMMITED]
>>
>> I updated to latest 2.4.2 GIT version (commit
>> 8b6830cdd96298682fcc298095ad1b718c54c77d), same problem is happening.
>>
>> Also you can access the server if you want, it's dedicated to this test.
>>
>> Thanks
>>
>>
>>
>>
>> On Thu, Aug 30, 2018 at 1:04 PM Bogdan-Andrei Iancu 
>> wrote:
>>
>>> Hi Daniel,
>>>
>>> Are you sure you configured a proper SIP URI as "message_queue" in the
>>> flow description ? My impression is you have an empty string there - and
>>> OpenSIPS is trying to put the call on the queue (as there is no agent), but
>>> the SIP URI is not valid.
>>>
>>> Regards,
>>>
>>> Bogdan-Andrei Iancu
>>>
>>> OpenSIPS Founder and Developer
>>>   http://www.opensips-solutions.com
>>> OpenSIPS Bootcamp 2018
>>>   http://opensips.org/training/OpenSIPS_Bootcamp_2018/
>>>
>>> On 08/29/2018 10:26 PM, Daniel Zanutti wrote:
>>>
>>> Got some more info.
>>>
>>> *This is the first call that worked fine:*
>>> ..
>>>
>>> *This is the second call that had the problem:*
>>> .
>>> Aug 29 16:04:38 plat5 /sbin/opensips[24890]:
>>> DBG:call_center:cc_call_state_machine: selecting QUEUE
>>> Aug 29 1

Re: [OpenSIPS-Users] Doubt about call center module

[Daniel Zanutti]

Hi Bogdan

Here it is table cc_flows:
id  flowid  priority  skillprependcid  message_welcome
message_queue
--  --    ---  --  ---
---
 1  fila-1   256  suporte  fila-1


Also table agents:
id  agentid location logstate
skills   last_call_end
--  --  ---  
---  ---
 1  1...@plat5.domain.com  sip:1...@plat5.domain.com:5060 1
suporte   1535650312

Thanks

On Fri, Aug 31, 2018 at 5:02 AM Bogdan-Andrei Iancu 
wrote:

> Hi Daniel,
>
> It is not about the B2B scenario, but about how you provisioned the flow
> in DB. Could you simply dump the output of "select * from cc_flows" ?
>
> Regards,
>
> Bogdan-Andrei Iancu
>
> OpenSIPS Founder and Developer
>   http://www.opensips-solutions.com
> OpenSIPS Bootcamp 2018
>   http://opensips.org/training/OpenSIPS_Bootcamp_2018/
>
> On 08/30/2018 08:34 PM, Daniel Zanutti wrote:
>
> Hi Bogdan
>
> Yes, It's the same scenario and same message. The call flow is:
>
> Asterisk Dials(port 5070) -> Opensips (port 5060) forward to Queue ->
> Calls local user
>
> I'm using standard Queue scenario:
> 
> 
> 
> 
> 
> server1
> 
> 
> client1
> message
> 
> 1
> 
> 
> 
> 1
> 
> 
>
> And SIP message is the same on all calls, just changed Call-id/tags:
>
> U 10.10.10.10:5070 -> 10.10.10.10:5060
> INVITE sip:fila-1@10.10.10.10:5060 SIP/2.0.
> Via: SIP/2.0/UDP 10.10.10.10:5070;branch=z9hG4bK2abb2acc;rport.
> Max-Forwards: 70.
> From: ;tag=as6440e239.
> To: .
> Contact: .
> Call-ID: 357cf76348e4e68325d065e85282320a@10.10.10.10:5070.
> CSeq: 102 INVITE.
> User-Agent: PBX SIPTEK.
> Date: Thu, 30 Aug 2018 17:30:30 GMT.
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
> PUBLISH, MESSAGE.
> Supported: replaces, timer.
> P-Asserted-Identity: "5511" .
> Content-Type: application/sdp.
> Content-Length: 353.
> [SDP OMMITED]
>
> I updated to latest 2.4.2 GIT version (commit
> 8b6830cdd96298682fcc298095ad1b718c54c77d), same problem is happening.
>
> Also you can access the server if you want, it's dedicated to this test.
>
> Thanks
>
>
>
>
> On Thu, Aug 30, 2018 at 1:04 PM Bogdan-Andrei Iancu 
> wrote:
>
>> Hi Daniel,
>>
>> Are you sure you configured a proper SIP URI as "message_queue" in the
>> flow description ? My impression is you have an empty string there - and
>> OpenSIPS is trying to put the call on the queue (as there is no agent), but
>> the SIP URI is not valid.
>>
>> Regards,
>>
>> Bogdan-Andrei Iancu
>>
>> OpenSIPS Founder and Developer
>>   http://www.opensips-solutions.com
>> OpenSIPS Bootcamp 2018
>>   http://opensips.org/training/OpenSIPS_Bootcamp_2018/
>>
>> On 08/29/2018 10:26 PM, Daniel Zanutti wrote:
>>
>> Got some more info.
>>
>> *This is the first call that worked fine:*
>> ..
>>
>> *This is the second call that had the problem:*
>> .
>> Aug 29 16:04:38 plat5 /sbin/opensips[24890]:
>> DBG:call_center:cc_call_state_machine: selecting QUEUE
>> Aug 29 16:04:38 plat5 /sbin/opensips[24890]:
>> DBG:call_center:cc_queue_push_call:  QUEUE - adding call 0x7fd8510524a8
>> Aug 29 16:04:38 plat5 /sbin/opensips[24890]:
>> DBG:call_center:cc_queue_push_call: adding call on pos 0 (already 1 calls),
>> l=(nil) h=(nil)
>> Aug 29 16:04:38 plat5 /sbin/opensips[24890]:
>> DBG:call_center:w_handle_call: new destination for call(0x7fd8510524a8) is
>> (state=2)
>> .
>>
>>
>> On Mon, Aug 27, 2018 at 6:15 PM Daniel Zanutti 
>> wrote:
>>
>>> Trying to configure the call center modules, but found a problem when
>>> there is no agents available.
>>>
>>> If there is 1 agent available, call is sent to him with no problem:
>>>
>>> Aug 27 18:11:00 plat5 /sbin/opensips[23575]: Recebida asterisk -
>>> Tentando entrar na fila fila-1
>>> Aug 27 18:11:00 plat5 /sbin/opensips[23575]: Entrou na fila com sucesso
>>> (fila-1)!
>>> Aug 27 18:11:01 plat5 /sbin/opensips[23569]: incoming reply
>>>
>>> But whe

Re: [OpenSIPS-Users] Doubt about call center module

[Daniel Zanutti]

Hi Bogdan

Yes, It's the same scenario and same message. The call flow is:

Asterisk Dials(port 5070) -> Opensips (port 5060) forward to Queue -> Calls
local user

I'm using standard Queue scenario:





server1


client1
message

1



1



And SIP message is the same on all calls, just changed Call-id/tags:

U 10.10.10.10:5070 -> 10.10.10.10:5060
INVITE sip:fila-1@10.10.10.10:5060 SIP/2.0.
Via: SIP/2.0/UDP 10.10.10.10:5070;branch=z9hG4bK2abb2acc;rport.
Max-Forwards: 70.
From: ;tag=as6440e239.
To: .
Contact: .
Call-ID: 357cf76348e4e68325d065e85282320a@10.10.10.10:5070.
CSeq: 102 INVITE.
User-Agent: PBX SIPTEK.
Date: Thu, 30 Aug 2018 17:30:30 GMT.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE.
Supported: replaces, timer.
P-Asserted-Identity: "5511" .
Content-Type: application/sdp.
Content-Length: 353.
[SDP OMMITED]

I updated to latest 2.4.2 GIT version (commit
8b6830cdd96298682fcc298095ad1b718c54c77d), same problem is happening.

Also you can access the server if you want, it's dedicated to this test.

Thanks




On Thu, Aug 30, 2018 at 1:04 PM Bogdan-Andrei Iancu 
wrote:

> Hi Daniel,
>
> Are you sure you configured a proper SIP URI as "message_queue" in the
> flow description ? My impression is you have an empty string there - and
> OpenSIPS is trying to put the call on the queue (as there is no agent), but
> the SIP URI is not valid.
>
> Regards,
>
> Bogdan-Andrei Iancu
>
> OpenSIPS Founder and Developer
>   http://www.opensips-solutions.com
> OpenSIPS Bootcamp 2018
>   http://opensips.org/training/OpenSIPS_Bootcamp_2018/
>
> On 08/29/2018 10:26 PM, Daniel Zanutti wrote:
>
> Got some more info.
>
> *This is the first call that worked fine:*
> ..
>
> *This is the second call that had the problem:*
> .
> Aug 29 16:04:38 plat5 /sbin/opensips[24890]:
> DBG:call_center:cc_call_state_machine: selecting QUEUE
> Aug 29 16:04:38 plat5 /sbin/opensips[24890]:
> DBG:call_center:cc_queue_push_call:  QUEUE - adding call 0x7fd8510524a8
> Aug 29 16:04:38 plat5 /sbin/opensips[24890]:
> DBG:call_center:cc_queue_push_call: adding call on pos 0 (already 1 calls),
> l=(nil) h=(nil)
> Aug 29 16:04:38 plat5 /sbin/opensips[24890]:
> DBG:call_center:w_handle_call: new destination for call(0x7fd8510524a8) is
> (state=2)
> .
>
>
> On Mon, Aug 27, 2018 at 6:15 PM Daniel Zanutti 
> wrote:
>
>> Trying to configure the call center modules, but found a problem when
>> there is no agents available.
>>
>> If there is 1 agent available, call is sent to him with no problem:
>>
>> Aug 27 18:11:00 plat5 /sbin/opensips[23575]: Recebida asterisk - Tentando
>> entrar na fila fila-1
>> Aug 27 18:11:00 plat5 /sbin/opensips[23575]: Entrou na fila com sucesso
>> (fila-1)!
>> Aug 27 18:11:01 plat5 /sbin/opensips[23569]: incoming reply
>>
>> But when there is no agent available, opensips refuses:
>> Aug 27 18:11:07 plat5 /sbin/opensips[23569]: Recebida asterisk - Tentando
>> entrar na fila fila-1
>> Aug 27 18:11:07 plat5 /sbin/opensips[23569]:
>> ERROR:b2b_logic:b2b_process_scenario_init: Failed to get the value for the
>> b2b client ruri
>> Aug 27 18:11:07 plat5 /sbin/opensips[23569]:
>> ERROR:call_center:set_call_leg: failed to init new b2bua call (empty ID
>> received)
>> Aug 27 18:11:07 plat5 /sbin/opensips[23569]:
>> ERROR:call_center:w_handle_call: failed to set new destination for call
>> Aug 27 18:11:07 plat5 /sbin/opensips[23569]: errnum: -1
>>
>> Error -1 means flowID is invalid, but I sent the same value on both calls.
>>
>> This is the call:
>>
>> cc_handle_call("$rU")
>>
>> I'm using Opensips 2.4.2 with Debian 8.11.
>>
>> Am I missing something or found a bug?
>>
>> Thanks
>>
>
>
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> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
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Re: [OpenSIPS-Users] Doubt about call center module

[Daniel Zanutti]

0.10.10.10:5070]
Aug 29 16:04:38 plat5 /sbin/opensips[24890]:
DBG:b2b_entities:b2b_search_htable_next_dlg: searching   totag
20[B2B.71.64.1535569477]
Aug 29 16:04:38 plat5 /sbin/opensips[24890]:
DBG:b2b_entities:b2b_search_htable_next_dlg: searching fromtag
10[as19cfceed]
Aug 29 16:04:38 plat5 /sbin/opensips[24890]:
DBG:b2b_entities:b2b_search_htable_next_dlg: Match for server dlg
[0x7fd851056830] dlg->uas_tran=[0x7fd851046cb0]
Aug 29 16:04:38 plat5 /sbin/opensips[24890]:
DBG:b2b_entities:b2b_entity_delete: Deleted dlg
[0x7fd851056830]->[B2B.71.64.1535569477] with dlginfo [0x7fd851055f80]
Aug 29 16:04:38 plat5 /sbin/opensips[24890]: DBG:tm:t_unref_cell:
UNREF_UNSAFE: [0x7fd851046cb0] after is 1
Aug 29 16:04:38 plat5 /sbin/opensips[24890]:
DBG:db_mysql:mysql_raise_event: MySQL status has not changed: connected
Aug 29 16:04:38 plat5 /sbin/opensips[24890]:
ERROR:call_center:set_call_leg: failed to init new b2bua call (empty ID
received)
Aug 29 16:04:38 plat5 /sbin/opensips[24890]:
ERROR:call_center:w_handle_call: failed to set new destination for call
Aug 29 16:04:38 plat5 /sbin/opensips[24890]: DBG:call_center:w_handle_call:
** onhold-- Error [0x7fd8510524a8]
Aug 29 16:04:38 plat5 /sbin/opensips[24890]:
DBG:call_center:print_call_list: [205.0] - 0x7fd85104c8b0
Aug 29 16:04:38 plat5 /sbin/opensips[24890]: DBG:call_center:free_cc_call:
free call 0x7fd8510524a8, []
Aug 29 16:04:38 plat5 /sbin/opensips[24890]: Falha entrando na fila -
erronum: -1



On Mon, Aug 27, 2018 at 6:15 PM Daniel Zanutti 
wrote:

> Trying to configure the call center modules, but found a problem when
> there is no agents available.
>
> If there is 1 agent available, call is sent to him with no problem:
>
> Aug 27 18:11:00 plat5 /sbin/opensips[23575]: Recebida asterisk - Tentando
> entrar na fila fila-1
> Aug 27 18:11:00 plat5 /sbin/opensips[23575]: Entrou na fila com sucesso
> (fila-1)!
> Aug 27 18:11:01 plat5 /sbin/opensips[23569]: incoming reply
>
> But when there is no agent available, opensips refuses:
> Aug 27 18:11:07 plat5 /sbin/opensips[23569]: Recebida asterisk - Tentando
> entrar na fila fila-1
> Aug 27 18:11:07 plat5 /sbin/opensips[23569]:
> ERROR:b2b_logic:b2b_process_scenario_init: Failed to get the value for the
> b2b client ruri
> Aug 27 18:11:07 plat5 /sbin/opensips[23569]:
> ERROR:call_center:set_call_leg: failed to init new b2bua call (empty ID
> received)
> Aug 27 18:11:07 plat5 /sbin/opensips[23569]:
> ERROR:call_center:w_handle_call: failed to set new destination for call
> Aug 27 18:11:07 plat5 /sbin/opensips[23569]: errnum: -1
>
> Error -1 means flowID is invalid, but I sent the same value on both calls.
>
> This is the call:
>
> cc_handle_call("$rU")
>
> I'm using Opensips 2.4.2 with Debian 8.11.
>
> Am I missing something or found a bug?
>
> Thanks
>
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[OpenSIPS-Users] Doubt about call center module

[Daniel Zanutti]

Trying to configure the call center modules, but found a problem when there
is no agents available.

If there is 1 agent available, call is sent to him with no problem:

Aug 27 18:11:00 plat5 /sbin/opensips[23575]: Recebida asterisk - Tentando
entrar na fila fila-1
Aug 27 18:11:00 plat5 /sbin/opensips[23575]: Entrou na fila com sucesso
(fila-1)!
Aug 27 18:11:01 plat5 /sbin/opensips[23569]: incoming reply

But when there is no agent available, opensips refuses:
Aug 27 18:11:07 plat5 /sbin/opensips[23569]: Recebida asterisk - Tentando
entrar na fila fila-1
Aug 27 18:11:07 plat5 /sbin/opensips[23569]:
ERROR:b2b_logic:b2b_process_scenario_init: Failed to get the value for the
b2b client ruri
Aug 27 18:11:07 plat5 /sbin/opensips[23569]:
ERROR:call_center:set_call_leg: failed to init new b2bua call (empty ID
received)
Aug 27 18:11:07 plat5 /sbin/opensips[23569]:
ERROR:call_center:w_handle_call: failed to set new destination for call
Aug 27 18:11:07 plat5 /sbin/opensips[23569]: errnum: -1

Error -1 means flowID is invalid, but I sent the same value on both calls.

This is the call:

cc_handle_call("$rU")

I'm using Opensips 2.4.2 with Debian 8.11.

Am I missing something or found a bug?

Thanks
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Re: [OpenSIPS-Users] Use RTPPROXY to bridge ipv4/ipv6

[Daniel Zanutti]

It's working.

Thanks Maxim

On Thu, Aug 2, 2018 at 8:33 PM Maxim Sobolev  wrote:

> Daniel, you can find some v4-v6 examples here:
>
> https://github.com/sippy/voiptests/blob/master/test_run.sh
>
> ${RTPPROXY} -p "${RTPP_PIDF}" -d dbug -f -s stdio: -s "${RTPP_SOCK_UDP}" \
>   -s "${RTPP_SOCK_CUNIX}" -s "${RTPP_SOCK_UNIX}" -s "${RTPP_SOCK_UDP6}" -s
> "${RTPP_SOCK_TCP}" \
>   -s "${RTPP_SOCK_TCP6}" -m 12000 -M 15000 -6 '/::' -l '0.0.0.0'
> ${RTPP_NOTIFY_ARG}
>
> In your case that would be (note "/" in front of IPv6 addr):
>
> /bin/rtpproxy -F -l "200.200.200.200" -6 "/2607:3f00:2
> <http://200.200.200.200/2607:3f00:2>"
>
> -Max
>
> On Thu, Aug 2, 2018 at 1:50 PM Daniel Zanutti 
> wrote:
>
>> Hi
>>
>> I'm trying to configure RTPPROXY to bridge ipv4 and ipv6 networks, but
>> didn't find the proper way.
>> Supposing IPs "200.200.200.200" and  "2607:3f00:2 " both on ETH0
>> interface.
>>
>> Tried:
>> /bin/rtpproxy -F -l 200.200.200.200/2607:3f00:2
>>
>> Got this error: Restarting rtpproxy: rtpproxy: host2bindaddr: Address
>> family for hostname not supported
>>
>> Then used -6 option and got same error:
>> /bin/rtpproxy -F -6 200.200.200.200/2607:3f00:2
>>
>> What is the right way?
>>
>> Thanks
>> ___
>> Users mailing list
>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>
>
>
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[OpenSIPS-Users] Use RTPPROXY to bridge ipv4/ipv6

[Daniel Zanutti]

Hi

I'm trying to configure RTPPROXY to bridge ipv4 and ipv6 networks, but
didn't find the proper way.
Supposing IPs "200.200.200.200" and  "2607:3f00:2 " both on ETH0
interface.

Tried:
/bin/rtpproxy -F -l 200.200.200.200/2607:3f00:2

Got this error: Restarting rtpproxy: rtpproxy: host2bindaddr: Address
family for hostname not supported

Then used -6 option and got same error:
/bin/rtpproxy -F -6 200.200.200.200/2607:3f00:2

What is the right way?

Thanks
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Re: [OpenSIPS-Users] Using Sipp stress tool with Opensips

[Daniel Zanutti]

I think the problem is related to configuring SIPP properly.

If I'm not wrong, SIPP standard scenario for UAC/UAS is configured to work
with a gateway (B2B), but Opensips is a proxy. You have to use Routes to
properly handle the incoming call and respond it.

Take a lookt at "rrs" param of recv command.

This is the code I use on UAC.xml, after receiving 200 OK from initial
invite:


  

  
  

  

  

  

  
  
  

  
  

  

  




On Wed, Jun 6, 2018 at 7:32 AM, David Villasmil <
david.villasmil.w...@gmail.com> wrote:

> Are the 2 sipp instances behind nat? If so, you need to set the appropiate
> switch on sipp, i believe it is
> -bind_local the actual local ip
> -i the public ip
>
> Take a look at sipp's documentation
>
> Hope it helps
>
> David
>
> On Wed, Jun 6, 2018, 10:33 Jan Skórczyński 
> wrote:
>
>> Hello,
>>
>> I'm a beginner in VOIP world so maybe this question have obvious
>> answer...
>> I would like to perform series of stress tests of my Opensips server.
>>
>> Opensips instance is located on VPS at adres XXX.XXX.XXX.XXX (public IP)
>> and it is listening on 5060 port. I'm using custom client apps, everything
>> works fine, however I have some issue when I try to test with Sipp.
>>
>> I'm setting UAS on my VPS, with the following command:
>>
>> sipp -sn uas -rsa XXX.XXX.XXX.XXX:5060 -i XXX.XXX.XXX.XXX
>>
>> and UAC on my PC:
>>
>> sipp -sn uac XXX.XXX.XXX.XXX:5061  -i 192.168.0.103
>>
>> I can see on UAS that it receives messages from my UAC, but it cannot
>> reply to them.
>>
>> -- Scenario Screen  [1-9]: Change
>> Screen --
>>   Port   Total-time  Total-calls  Transport
>>   5061  19.07 s   30  UDP
>>
>>   0 new calls during 1.004 s period  1 ms scheduler resolution
>>   0 callsPeak was 6 calls, after 6 s
>>   0 Running, 31 Paused, 4 Woken up
>>   90 dead call msg (discarded)
>>   4 open sockets
>>
>>  Messages  Retrans   Timeout
>>  Unexpected-Msg
>>   --> INVITE 300 0 0
>>
>>   <-- 180300
>>   <-- 20030140
>>   --> ACK E-RTD1 0 0 0 30
>>
>>   --> BYE0 0 0 0
>>   <-- 2000 0
>>   [   4000ms] Pause  0 0
>> -- Sipp Server Mode
>> ---
>>
>>
>> Logs from UAC:
>>
>> -- Scenario Screen  [1-9]: Change
>> Screen --
>>   Call-rate(length)   Port   Total-time  Total-calls  Remote-host
>>   10.0(0 ms)/1.000s   5060  73.29 s   90
>> XXX.XXX.XXX.XXX:5061(UDP)
>>
>>   0 new calls during 1.004 s period  1 ms scheduler resolution
>>   30 calls (limit 30)Peak was 30 calls, after 3 s
>>   0 Running, 62 Paused, 24 Woken up
>>   0 dead call msg (discarded)0 out-of-call msg (discarded)
>>
>>   3 open sockets
>>
>>  Messages  Retrans   Timeout
>>  Unexpected-Msg
>>   INVITE --> 90415   60
>>  100 <-- 0 0 0 0
>>  180 <-- 0 0 0 0
>>  183 <-- 0 0 0 0
>>  200 <--  E-RTD1 0 0 0 0
>>  ACK --> 0 0
>>Pause [  0ms] 0 0
>>  BYE --> 0 0 0
>>  200 <-- 0 0 0 0
>>
>> -- [+|-|*|/]: Adjust rate  [q]: Soft exit  [p]: Pause traffic
>> -
>>
>> And here is my opensips.cfg routing script:
>>
>> route{
>> xlog("Begin route");
>> if (!mf_process_maxfwd_header("10")) {
>> xlog("Too many hops");
>> send_reply("483","Too Many Hops");
>> exit;
>> }
>>
>> if (has_totag() && ($fU!="sipp")) {
>> xlog("Has totag");
>> # handle hop-by-hop ACK (no routing required)
>> if ( is_method("ACK") && t_check_trans() ) {
>> xlog("Method: ACK && t_check_trans. Relaying...");
>> t_relay();
>> exit;
>> }
>>
>> # sequential request within a dialog should
>> # take the path determined by record-routing
>> if ( !loose_route() ) {
>> # we do record-routing for all our traffic, so we should not
>> # receive any sequential requests without Route hdr.
>> send_reply("404","Not here");
>> exit;
>> }
>>
>> if (is_method("BYE")) {
>> # do accounting even if the transaction fails
>> #do_accounting("log","failed");
>> }
>>
>> # route it out to whatever destination was set by loose_route()
>> # in $du (destination URI).
>> route(relay);
>> exit;
>> }
>>
>> # CANCEL processing
>> if (is_method("CANCEL")) {
>> if (t_check_trans())
>> t_relay();
>> exit;
>> }
>>
>> # absorb 

Re: [OpenSIPS-Users] Problem with simultaneous CANCEL + 200 OK

[Daniel Zanutti]

Hi Bogdan

I could reproduce the problem on lab. This is the scenario:

A INVITE -> OPENSIPS
OPENSIPS INVITE -> B

OPENSIPS <- B 200 OK

A <- 200 OK OPENSIPS
This 200 OK is lost or discarded on A, not sure if was on network or
problem at client. Then:

A CANCEL -> OPENSIPS
A <- 200 OK (of Cancel) OPENSIPS

Then the call is up for ever.

Here is the log at the moment the CANCEL is received:
https://pastebin.com/79qanD2H

And PCAP:
https://drive.google.com/file/d/1lpHWN1uRD1SOwBTZdijSodXBWhUX-rM8/view?usp=sharing

Can you help?

Thanks





On Thu, Feb 22, 2018 at 1:24 PM, Bogdan-Andrei Iancu <bog...@opensips.org>
wrote:

> Daniel,
>
> The pcap shows only one leg of the the communication (I guess caller
> versus OpenSIPS). So I cannot see what happens on the callee side. Still, I
> see the first cancel is rejected with 400 (I guess by OpenSIPS - check the
> logs to see the reason) and the second one is accepted by OpenSIPS (no idea
> if relayed to callee or not). Still, if there is a race between the cancel
> from caller and the answer from callee, it is up to the caller to sort it
> outaccording to RFC, the caller must ACK the received 200 OK (even if
> CANCEL was sent) and if it really wants to terminate the call, it has to
> fire a BYE.
>
> Regards,
>
> Bogdan-Andrei Iancu
>
> OpenSIPS Founder and Developer
>   http://www.opensips-solutions.com
> OpenSIPS Summit 2018
>   http://www.opensips.org/events/Summit-2018Amsterdam
>
> On 02/22/2018 04:22 PM, Daniel Zanutti wrote:
>
> Hi Bogdan
>
> Thanks for replying.
>
> Here is the PCAP, please take a look: https://drive.google.
> com/file/d/1e7SKjxDtdVYmN-7fCHSEqNEsNHjPsaKo/view?usp=sharing
>
> Thanks
>
> On Thu, Feb 22, 2018 at 7:30 AM, Bogdan-Andrei Iancu <bog...@opensips.org>
> wrote:
>
>> Hi Daniel,
>>
>> Without a pcap showing the signaling is hard to understand (not to
>> mention helping) your scenario. Please provide a link to the pcap or ngrep.
>>
>> Regards,
>>
>> Bogdan-Andrei Iancu
>>
>> OpenSIPS Founder and Developer
>>   http://www.opensips-solutions.com
>> OpenSIPS Summit 2018
>>   http://www.opensips.org/events/Summit-2018Amsterdam
>>
>> On 02/20/2018 06:43 PM, Daniel Zanutti wrote:
>>
>> Hey
>>
>> I had a problem when receiving simultaneous CANCEL from customer and 200
>> OK from gateway.
>>
>> Seems that the first CANCEL was rejected, but the second CANCEL was
>> accepted. This second CANCEL did NOT go to the gateway, just Opensips
>> received and replied with 200 OK.
>>
>> This is the log of the first CANCEL:
>> Feb 15 18:39:22 /sbin/opensips[28845]: SCRIPT:TRAFFIC:WARNING: method
>> CANCEL ( 7Qbq3O3CReMfPflAtl8NY3ddTqPVBHO2785126@2.2.2.2/ XAeG2xj278512T2
>> / 1839212581509953 ) not validated and not fixed ( code=-1 )
>>
>> code -1 is the return of validate_dialog()
>>
>> Second CANCEL didn't generated a log.
>>
>> Shouldn't all CANCELs be rejected? On this case, just the first one was
>> rejected.
>>
>> I'm using version 1.9.11.
>>
>>
>>
>>
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>>
>>
>>
>
>
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Re: [OpenSIPS-Users] Wrong number of children

[Daniel Zanutti]

Hi

You should have 5 per interface + some internal control threads. I'm not
sure exactly.

Regards

On Fri, Feb 23, 2018 at 9:25 AM, xaled  wrote:

> Hi
>
>
>
> I have configured 5 children in  opensips.cfg, but 16 get logged and 18
> initiated what could be the cause of it?
>
> I search for word children in opensips directory and it only comes ones in
> opensips.conf.
>
>
>
> children=5
>
>
>
> Feb 23 13:07:09 srv01 opensips: DBG:core:daemonize: waiting for status
> code from children
>
> Feb 23 13:07:09 srv01 opensips: DBG:core:count_init_children: 16 children
> are going to be inited
>
>
>
> root@dus-appsrv01:/var/log/opensips# ps ax|grep sips
>
> 14854 ?S  0:00 /usr/sbin/opensips -P
> /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4
>
> 14856 ?S  0:00 /usr/sbin/opensips -P
> /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4
>
> 14857 ?S  0:00 /usr/sbin/opensips -P
> /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4
>
> 14858 ?S  0:00 /usr/sbin/opensips -P
> /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4
>
> 14861 ?S  0:00 /usr/sbin/opensips -P
> /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4
>
> 14862 ?S  0:00 /usr/sbin/opensips -P
> /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4
>
> 14863 ?S  0:00 /usr/sbin/opensips -P
> /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4
>
> 14864 ?S  0:00 /usr/sbin/opensips -P
> /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4
>
> 14865 ?S  0:00 /usr/sbin/opensips -P
> /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4
>
> 14867 ?S  0:00 /usr/sbin/opensips -P
> /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4
>
> 14869 ?S  0:00 /usr/sbin/opensips -P
> /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4
>
> 14870 ?S  0:00 /usr/sbin/opensips -P
> /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4
>
> 14871 ?S  0:00 /usr/sbin/opensips -P
> /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4
>
> 14872 ?S  0:00 /usr/sbin/opensips -P
> /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4
>
> 14874 ?S  0:00 /usr/sbin/opensips -P
> /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4
>
> 14875 ?S  0:00 /usr/sbin/opensips -P
> /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4
>
> 14876 ?S  0:00 /usr/sbin/opensips -P
> /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4
>
> 14877 ?S  0:00 /usr/sbin/opensips -P
> /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4
>
>
>
> With commended children parameter I get 19 children in the log and 21 in
> reality:
>
>
>
> Feb 23 13:13:24 dus-appsrv01 opensips: DBG:core:daemonize: waiting for
> status code from children
>
> Feb 23 13:13:24 dus-appsrv01 opensips: DBG:core:count_init_children: 19
> children are going to be inited
>
>
>
> root@srv01:/var/log/opensips# ps ax|grep sips
>
> 14723 ?S  0:00 /usr/sbin/opensips -P
> /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4
>
> 14725 ?S  0:00 /usr/sbin/opensips -P
> /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4
>
> 14726 ?S  0:00 /usr/sbin/opensips -P
> /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4
>
> 14727 ?S  0:00 /usr/sbin/opensips -P
> /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4
>
> 14730 ?S  0:00 /usr/sbin/opensips -P
> /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4
>
> 14731 ?S  0:00 /usr/sbin/opensips -P
> /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4
>
> 14732 ?S  0:00 /usr/sbin/opensips -P
> /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4
>
> 14733 ?S  0:00 /usr/sbin/opensips -P
> /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4
>
> 14734 ?S  0:00 /usr/sbin/opensips -P
> /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4
>
> 14736 ?S  0:00 /usr/sbin/opensips -P
> /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4
>
> 14737 ?S  0:00 /usr/sbin/opensips -P
> /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4
>
> 14738 ?S  0:00 /usr/sbin/opensips -P
> /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4
>
> 14739 ?S  0:00 /usr/sbin/opensips -P
> /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4
>
> 14741 ?S  0:00 /usr/sbin/opensips -P
> /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4
>
> 14742 ?S  

Re: [OpenSIPS-Users] Problem with simultaneous CANCEL + 200 OK

[Daniel Zanutti]

Hi Bogdan

Thanks for replying.

Here is the PCAP, please take a look:
https://drive.google.com/file/d/1e7SKjxDtdVYmN-7fCHSEqNEsNHjPsaKo/view?usp=sharing

Thanks

On Thu, Feb 22, 2018 at 7:30 AM, Bogdan-Andrei Iancu <bog...@opensips.org>
wrote:

> Hi Daniel,
>
> Without a pcap showing the signaling is hard to understand (not to mention
> helping) your scenario. Please provide a link to the pcap or ngrep.
>
> Regards,
>
> Bogdan-Andrei Iancu
>
> OpenSIPS Founder and Developer
>   http://www.opensips-solutions.com
> OpenSIPS Summit 2018
>   http://www.opensips.org/events/Summit-2018Amsterdam
>
> On 02/20/2018 06:43 PM, Daniel Zanutti wrote:
>
> Hey
>
> I had a problem when receiving simultaneous CANCEL from customer and 200
> OK from gateway.
>
> Seems that the first CANCEL was rejected, but the second CANCEL was
> accepted. This second CANCEL did NOT go to the gateway, just Opensips
> received and replied with 200 OK.
>
> This is the log of the first CANCEL:
> Feb 15 18:39:22 /sbin/opensips[28845]: SCRIPT:TRAFFIC:WARNING: method
> CANCEL ( 7Qbq3O3CReMfPflAtl8NY3ddTqPVBHO2785126@2.2.2.2/ XAeG2xj278512T2
> / 1839212581509953 ) not validated and not fixed ( code=-1 )
>
> code -1 is the return of validate_dialog()
>
> Second CANCEL didn't generated a log.
>
> Shouldn't all CANCELs be rejected? On this case, just the first one was
> rejected.
>
> I'm using version 1.9.11.
>
>
>
>
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[OpenSIPS-Users] Problem with simultaneous CANCEL + 200 OK

[Daniel Zanutti]

 Hey

I had a problem when receiving simultaneous CANCEL from customer and 200 OK
from gateway.

Seems that the first CANCEL was rejected, but the second CANCEL was
accepted. This second CANCEL did NOT go to the gateway, just Opensips
received and replied with 200 OK.

This is the log of the first CANCEL:
Feb 15 18:39:22 /sbin/opensips[28845]: SCRIPT:TRAFFIC:WARNING: method
CANCEL ( 7Qbq3O3CReMfPflAtl8NY3ddTqPVBHO2785126@2.2.2.2/ XAeG2xj278512T2 /
1839212581509953 ) not validated and not fixed ( code=-1 )

code -1 is the return of validate_dialog()

Second CANCEL didn't generated a log.

Shouldn't all CANCELs be rejected? On this case, just the first one was
rejected.

I'm using version 1.9.11.
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[OpenSIPS-Users] Solution to storing a lot of siptrace

[Daniel Zanutti]

Hi

I have around 2000 simultaneous calls, 50 CPS and would like to store sip
trace for all of them.

Storing on MySQL is not working. If you have some indexes on the table,
after 1M register it starts to slow down the whole server. If no indexes,
it's not searchable.

Do you guys have a good solution to store a lot of sip trace on Opensips?
Like 200 GB.

It needs to be searchable almost real-time (a few minutes is fine).

I was thinking on store on a flat text file and move to a NOSQL solution,
what do you guys think?

Thanks
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Re: [OpenSIPS-Users] Weird freeze/crash

[Daniel Zanutti]

Hi Razvan

Version in use is 1.11.9. The problem is that it freezes, so there's no
automatic crash dump being generated.

I'll try to take a look when it happens again.

Thanks

On Tue, Sep 19, 2017 at 10:20 AM, Răzvan Crainea <raz...@opensips.org>
wrote:

> Hi, Daniel!
>
> Looks like a crash.
> What version of OpenSIPS are you using? If you could extract a coredump,
> it would be really helpful.
>
> Best regards,
>
> Răzvan Crainea
> OpenSIPS Core Developer
> http://www.opensips-solutions.com
>
>
> On 09/19/2017 04:07 PM, Daniel Zanutti wrote:
>
>> Hi
>>
>> Do you guys have an idea of what happened?
>>
>> Sep 10 09:35:13 /sbin/opensips[13579]: NOTICE:core:io_wait_loop_epoll:
>> EPOLLIN(read) event: epoll_wait() set event EPOLLHUP - connection closed by
>> the remote peer!
>> Sep 10 09:35:13 /sbin/opensips[13579]: CRITICAL:core:receive_fd: EOF on 42
>> Sep 10 09:35:14 /sbin/opensips[13439]: NOTICE:presence:destroy: destroy
>> module ...
>> Sep 10 09:35:14 /sbin/opensips[13439]: 
>> ERROR:nat_traversal:save_keepalive_state:
>> failed to open keepalive state file for writing: Permission denied
>>
>> At this point, opensips froze and no calls were being processed. I had to
>> kill it to restart. Unfortunatelly had no time to gdb on it.
>>
>> Thanks
>>
>>
>>
>>
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>>
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[OpenSIPS-Users] Weird freeze/crash

[Daniel Zanutti]

Hi

Do you guys have an idea of what happened?

Sep 10 09:35:13 /sbin/opensips[13579]: NOTICE:core:io_wait_loop_epoll:
EPOLLIN(read) event: epoll_wait() set event EPOLLHUP - connection closed by
the remote peer!
Sep 10 09:35:13 /sbin/opensips[13579]: CRITICAL:core:receive_fd: EOF on 42
Sep 10 09:35:14 /sbin/opensips[13439]: NOTICE:presence:destroy: destroy
module ...
Sep 10 09:35:14 /sbin/opensips[13439]:
ERROR:nat_traversal:save_keepalive_state: failed to open keepalive state
file for writing: Permission denied

At this point, opensips froze and no calls were being processed. I had to
kill it to restart. Unfortunatelly had no time to gdb on it.

Thanks
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Re: [OpenSIPS-Users] Accounting of 200 OK and BYE

[Daniel Zanutti]

Hi Bogdan

So on transaction accounting, the record is written after transaction
receives final response. So it's not written as soon I receive BYE, but
when I receive the 200OK of the BYE.

My customer is complaining that the call is taking 200ms more and our
system is charging 1 sec more than on his billing. On million calls, this
is generating a some thousands difference.

Thanks for the information!

Regards

On Tue, Jul 25, 2017 at 6:36 AM, Bogdan-Andrei Iancu <bog...@opensips.org>
wrote:

> Hi Daniel,
>
> There are 3 types of accountings in OpenSIPS - per message, per
> transaction, per dialog.
>
> For the per message, it is clear :) . When doing per-transaction
> accounting, the ACC record is written when the transaction is completed
> with a final response (>=200) on the UAS side (towards caller). For the
> dialog based accounting, the time reference (for ending the call) is the
> reception of BYE request; still the CDR is written on the BYE final reply
> (as OpenSIPS allows you to collect CDR info from the BYE replies too).
>
> Best regards,
>
> Bogdan-Andrei Iancu
>   OpenSIPS Founder and Developer
>   http://www.opensips-solutions.com
>
> OpenSIPS Bootcamp 2017, Houston, US
>   http://opensips.org/training/OpenSIPS_Bootcamp_2017.html
>
> On 07/20/2017 10:49 PM, Daniel Zanutti wrote:
>
> Hi Alex
>
> I'm having a billing problem from receiving BYE to 200 OK is taking more
> than 500ms. If BYE is accounted when it's received, great!
>
> Are you absolutely sure it works this way?
>
> Thanks
>
> On Thu, Jul 20, 2017 at 4:26 PM, Alex Balashov <abalas...@evaristesys.com>
> wrote:
>
>> My understanding is that this is a rather simple module without
>> sophisticated state componentry, and that it logs things immediately as
>> received, in the same iteration of message processing.
>>
>> -- Alex
>>
>> --
>> Principal, Evariste Systems LLC (www.evaristesys.com)
>>
>> Sent from my Google Nexus.
>>
>> ___
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>>
>
>
>
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>
>
>
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Re: [OpenSIPS-Users] Accounting of 200 OK and BYE

[Daniel Zanutti]

Hi Alex

I'm having a billing problem from receiving BYE to 200 OK is taking more
than 500ms. If BYE is accounted when it's received, great!

Are you absolutely sure it works this way?

Thanks

On Thu, Jul 20, 2017 at 4:26 PM, Alex Balashov 
wrote:

> My understanding is that this is a rather simple module without
> sophisticated state componentry, and that it logs things immediately as
> received, in the same iteration of message processing.
>
> -- Alex
>
> --
> Principal, Evariste Systems LLC (www.evaristesys.com)
>
> Sent from my Google Nexus.
>
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Re: [OpenSIPS-Users] Accounting of 200 OK and BYE

[Daniel Zanutti]

Yes, ACC module.

On Thu, Jul 20, 2017 at 3:45 PM, Alex Balashov <abalas...@evaristesys.com>
wrote:

> On Thu, Jul 20, 2017 at 03:40:29PM -0300, Daniel Zanutti wrote:
>
> > In what exactly moment the 200OK and BYE messages are accounted and
> written
> > to the database?
> >
> > At the moment Opensips receive the 200 OK or after receive ACK of 200 OK?
> >
> > Also on BYE, when receive BYE or on 200 OK of BYE?
>
> Are you referring to the ACC module, or some other method of accounting?
> :-)
>
> --
> Alex Balashov | Principal | Evariste Systems LLC
>
> Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free)
> Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
>
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[OpenSIPS-Users] Accounting of 200 OK and BYE

[Daniel Zanutti]

In what exactly moment the 200OK and BYE messages are accounted and written
to the database?

At the moment Opensips receive the 200 OK or after receive ACK of 200 OK?

Also on BYE, when receive BYE or on 200 OK of BYE?

Thanks
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[OpenSIPS-Users] Looking for Mediaproxy developer

[Daniel Zanutti]

I'm looking for help to customize some things on mediaproxy software.

Need to:
1) Fix some bugs
2) Implement new features

Please contact me for details.

Thanks
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[OpenSIPS-Users] RTPENGINE (sipwise) working with opensips?

[Daniel Zanutti]

Question:

Is RTPENGINE (sipwise) working with opensips? I saw only Kamailio on
rtpengine page at github, but there is a module for opensips 2.1 (
http://www.opensips.org/html/docs/modules/2.1.x/rtpengine).

Thanks
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Re: [OpenSIPS-Users] Which SSL Certificate for SIP+TLS

[Daniel Zanutti]

That's great!

I'll take a look.

Thanks William

On Wed, Jun 7, 2017 at 12:19 PM, William Simon <wsi...@stratusvideo.com>
wrote:

> Web site certificates are fine. Do not purchase anything more than simple
> name validation because SIP clients won't care and you will waste your
> money. You can even use free certificates from LetsEncrypt.org
>
>
> On Jun 7, 2017, at 11:14 AM, Daniel Zanutti <daniel.zanu...@gmail.com>
> wrote:
>
> Hi Alex
>
> I have tried with self-generated certificate and it is working fine.
>
> The problem is that this server will be "public" to any kind of sip
> client, so I need a Certificate Authority behind it.
>
> I couldn't find the right certificate, can anyone point the right one to
> buy?
>
> Looked at instantssl, they are all website certificates:
>
> https://www.instantssl.com/ssl-certificate.html?track=8183
>
> Thanks in advance!
>
> On Wed, Jun 7, 2017 at 11:20 AM, Alexander Jankowsky <
> e75a4...@exemail.com.au> wrote:
>
>>
>>
>> It is fairly straight forward to generate your own TLS certificates,
>>
>> if it is for your own Opensips server or for your own Voip phones.
>>
>>
>>
>> It is putting the certificates to use that may require a little trial and
>> error.
>>
>> You really might first want and need to test everything and make sure
>>
>> it is all sorted out and working properly with your own certificates.
>>
>>
>>
>> Do you really just only need a certificate to get things running
>>
>> for now or does it have to be from a recognised authority.
>>
>>
>>
>> Alex
>>
>>
>>
>> *From:* Users [mailto:users-boun...@lists.opensips.org] *On Behalf Of *Daniel
>> Zanutti
>> *Sent:* Wednesday, 7 June 2017 11:31 PM
>> *To:* OpenSIPS users mailling list
>> *Subject:* [OpenSIPS-Users] Which SSL Certificate for SIP+TLS
>>
>>
>>
>> I need to install an TLS certificate for secure SIP communication.
>>
>>
>>
>> Could you guys please point a valid certificate so I can buy it?
>>
>>
>>
>> Thanks
>>
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>>
>>
>
>
> “The information transmitted is intended only for the person or entity to
> which it is addressed and may contain proprietary, business-confidential
> and/or privileged material. If you are not the intended recipient of this
> message you are hereby notified that any use, review, retransmission,
> dissemination, distribution, reproduction or any action taken in reliance
> upon this message is prohibited. If you received this in error, please
> contact the sender and delete the material from any computer.”
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>
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Re: [OpenSIPS-Users] Which SSL Certificate for SIP+TLS

[Daniel Zanutti]

Hi Alex

I have tried with self-generated certificate and it is working fine.

The problem is that this server will be "public" to any kind of sip client,
so I need a Certificate Authority behind it.

I couldn't find the right certificate, can anyone point the right one to
buy?

Looked at instantssl, they are all website certificates:

https://www.instantssl.com/ssl-certificate.html?track=8183

Thanks in advance!

On Wed, Jun 7, 2017 at 11:20 AM, Alexander Jankowsky <
e75a4...@exemail.com.au> wrote:

>
>
> It is fairly straight forward to generate your own TLS certificates,
>
> if it is for your own Opensips server or for your own Voip phones.
>
>
>
> It is putting the certificates to use that may require a little trial and
> error.
>
> You really might first want and need to test everything and make sure
>
> it is all sorted out and working properly with your own certificates.
>
>
>
> Do you really just only need a certificate to get things running
>
> for now or does it have to be from a recognised authority.
>
>
>
> Alex
>
>
>
> *From:* Users [mailto:users-boun...@lists.opensips.org] *On Behalf Of *Daniel
> Zanutti
> *Sent:* Wednesday, 7 June 2017 11:31 PM
> *To:* OpenSIPS users mailling list
> *Subject:* [OpenSIPS-Users] Which SSL Certificate for SIP+TLS
>
>
>
> I need to install an TLS certificate for secure SIP communication.
>
>
>
> Could you guys please point a valid certificate so I can buy it?
>
>
>
> Thanks
>
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[OpenSIPS-Users] Which SSL Certificate for SIP+TLS

[Daniel Zanutti]

I need to install an TLS certificate for secure SIP communication.

Could you guys please point a valid certificate so I can buy it?

Thanks
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Re: [OpenSIPS-Users] Opensips Late reply.

[Daniel Zanutti]

Hi Qasim

How did you limit CPS? Because i have a similar scenario (150cps) but i set
children to 20 or 24, never 200. You don't need 1 children per request.

On Wed, Apr 5, 2017 at 9:44 AM, qasimak...@gmail.com 
wrote:

> Hi,
>
> I have this scenario where i originate calls from mi_datagram and the
> calls are cancelled as soon as it starts ringing. The problem i am facing
> is that are running for a few minutes the response from opensips becomes
> slow i.e. it send packets back to far end after a few seconds. Keeping it
> running for a few hours and it crashes. I have currently limited the calls
> per second to 200 and max call session to 600, but still there are calls
> where opensips responds slow to SIP packets.
>
> Here are a few configurations that i am using:
>
> fork=yes
> children=200
>
> loadmodule "mi_datagram.so"
> modparam("mi_datagram", "socket_name", "/tmp/opensips.sock")
> modparam("mi_datagram", "socket_name", "udp:localhost:2000")
> modparam("mi_datagram", "children_count", 200)
>
>
> Do you have any pointers where i should start looking at? I have also
> generated core dump files let me know if you need some more info on this.
>
> Regards,
> Qasim
>
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Re: [OpenSIPS-Users] Mediaproxy speed calculations

[Daniel Zanutti]

Hi Dan

Ok about not being able to force calculation, it's done periodically. But
for some reason, it's not calculating =(

You misunderstood me about the machine. On this scenario, I have complete
control of the machines, all physical machines are ours. When I said the
system is not overloaded during the night, I mean that both physical and
virtual machines are not being used at all (load 0%).

I just checked that at least one of these machines are physical and not
virtual, this is the config:
CPU0: Intel(R) Xeon(R) CPU   X5560  @ 2.80GHz (fam: 06, model: 1a,
stepping: 05)
smpboot: Total of 8 processors activated (44687.68 BogoMIPS)

Based on your explanation, my physical machine with 2.8GHz is computing at
5MHz, which is surely wrong.

I have a similar scenario deployed on more than 50 machine and almost every
time Mediaproxy is started at linux boot, it doesn't calculate and show
speed. After restarting the process, speed is calculated fine.

Could you please consider that the software may have a bug? Are you
interested on fixing it? Can I help?

Thanks


On Tue, Mar 28, 2017 at 5:32 PM, Dan Pascu <d...@ag-projects.com> wrote:

>
> On 28 Mar 2017, at 20:55, Daniel Zanutti wrote:
>
> > Hi Dan
> >
> > Thanks for answering.
> >
> > The machine is not overloaded, actually i have the same problem with 10
> calls or 1000 calls.
> >
> > Syslog:
> > Mar 28 14:51:45 MP-104 media-relay[782]: warning: Aggregate speed
> calculation time exceeded 10ms: 15214 us for 418 sessions
> >
> >
> > TOP:
> > load average: 0.56, 0.61, 0.63
>
> You misunderstood me. I was not talking about your virtual machine, I was
> talking about the real hardware being overloaded, probably running too many
> virtual machines.
>
> From inside the virtual machine you cannot assess how loaded the real
> hardware is. You can have 0% CPU load inside your virtual machine, that
> doesn't mean things are OK. The fact that inside your virtual machine an
> operation takes 600 times longer than on 5 years old real hardware, means
> that your system is unable to perform as it should. If the real hardware
> CPU runs at let's say 3GHz, this is equivalent to saying that your virtual
> machine has a CPU running at 3000MHz/600 = 5MHz. You try to run a media
> relay that has to react in real time inside a virtual machine that behaves
> as if it has a 5MHz CPU!
>
> You may prefer to run things on virtual machines for reasons related to
> costs, but at the end of the day one thing is clear: a media relay requires
> a RTOS. Linux running on real hardware is not an RTOS, but if the machine
> doesn't run other things that can influence the resource allocation, it can
> approximate one pretty well. A virtual machine running 600 times slower
> than real hardware, with resources shared between multiple virtual
> machines, is as far removed from the idea of a RTOS as it can possibly be.
>
> > You are right about being virtual, but I'm sure the server is not
> overloaded because I have the same problem during the night, with almost no
> traffic. During the day, it MAY be overloaded but surely not during the
> night and this information never shows up on these relays.
> >
> > Is there any way to force it? Could you give some directions?
>
> Force what? As I said the traffic calculations are done periodically at an
> interval specified in the configuration, with the default being 15 seconds.
> You can disable them by setting the sampling interval to 0. The warning
> doesn't mean they are skipped, it only means the relay took too long to
> compute them and was unresponsive for other requests during that time.
>
> >
> > Thanks
> >
> >
> > On Tue, Mar 28, 2017 at 2:27 PM, Dan Pascu <d...@ag-projects.com> wrote:
> >
> > On 24 Mar 2017, at 19:51, Daniel Zanutti wrote:
> >
> > > Hi
> > >
> > > Looks like i'm diving deep on mediaproxy.
> > >
> > > Some of our relays are not calculating the speed on the network. If I
> restart a couple times it starts calculating fine.
> > >
> > > I found this log:
> > > media-relay[4100]: warning: Aggregate speed calculation time exceeded
> 10ms: 11644 us for 222 sessions
> > >
> > > Is there any solution to always calculate?
> >
> > The relay always calculates. That is just a warning when it takes too
> long, but the calculation still took place.
> >
> > The reasons why you might not see traffic:
> >
> > 1. There is no actual traffic, despite having sessions setup, the
> devices do not send media
> > 2. There is traffic but for some reason reading the traffic information
> from the kernel fails (I have no idea why that could hap

Re: [OpenSIPS-Users] Mediaproxy speed calculations

[Daniel Zanutti]

Hi Dan

Thanks for answering.

The machine is not overloaded, actually i have the same problem with 10
calls or 1000 calls. I can confirm there is a lot of traffic on it, for
instance:

1 1.1.1.1 2.6.1 108h09'30" 6.71Mbps 413 audio 413 Active
2 2.2.2.2 2.6.5 95h50'12" 0bps 435 audio 435 Active
3 3.3.3.3 2.6.5 95h50'16" 0bps 382 audio 382 Active
4 4.4.4.4 2.6.5 108h08'38" 7.29Mbps 402 audio 402 Active
5 5.5.5.5 2.6.5 107h59'38" 6.41Mbps 375 audio 375 Active
(fake IPs)

IPTRAF:
eth0  304104304104 00  20903.40 kbits/sec


Syslog:
Mar 28 14:51:45 MP-104 media-relay[782]: warning: Aggregate speed
calculation time exceeded 10ms: 15214 us for 418 sessions


TOP:
load average: 0.56, 0.61, 0.63


Kernel:
Linux MP-104 3.16.0-4-amd64 #1 SMP Debian 3.16.39-1+deb8u2 (2017-03-07)
x86_64 GNU/Linux


You are right about being virtual, but I'm sure the server is not
overloaded because I have the same problem during the night, with almost no
traffic. During the day, it MAY be overloaded but surely not during the
night and this information never shows up on these relays.

Is there any way to force it? Could you give some directions?

Thanks


On Tue, Mar 28, 2017 at 2:27 PM, Dan Pascu <d...@ag-projects.com> wrote:

>
> On 24 Mar 2017, at 19:51, Daniel Zanutti wrote:
>
> > Hi
> >
> > Looks like i'm diving deep on mediaproxy.
> >
> > Some of our relays are not calculating the speed on the network. If I
> restart a couple times it starts calculating fine.
> >
> > I found this log:
> > media-relay[4100]: warning: Aggregate speed calculation time exceeded
> 10ms: 11644 us for 222 sessions
> >
> > Is there any solution to always calculate?
>
> The relay always calculates. That is just a warning when it takes too
> long, but the calculation still took place.
>
> The reasons why you might not see traffic:
>
> 1. There is no actual traffic, despite having sessions setup, the devices
> do not send media
> 2. There is traffic but for some reason reading the traffic information
> from the kernel fails (I have no idea why that could happen, except maybe a
> severely overloaded virtual machine - see below)
>
> I noticed something very wrong with that warning. On a machine running on
> a Core I7 from 2012 (Sandy Bridge architecture, so not the latest hardware,
> but something from 5 years ago), the calculation for 222 sessions, takes 20
> us (that is micro seconds). You got 11644 us, which is approximately 600
> times slower. Which means your virtual machine is severely overloaded, or
> the amount of resources it has allocated from the real hardware is abysmal.
>
> On the same machine I mentioned before, having 2000 active sessions
> results in the speed calculations taking 170 us, which is well below the
> warning limit of 10 ms. Which means, the relay can drive thousands of
> sessions and you'll never see the warning.
>
> In conclusion, unless you run on a severely overloaded system, or a very
> underpowered virtual machine, you should never see that warning and seeing
> the warning doesn't mean that calculations didn't take place.
>
> --
> Dan
>
>
>
>
>
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Re: [OpenSIPS-Users] ACC Db Duration

[Daniel Zanutti]

Hi

Did you check "cdr_flag" on Acc module?

Otherwise it will generate 2 registers, 1 for INVITE and 1 for BYE.

Regards

On Tue, Mar 28, 2017 at 10:29 AM, qasimak...@gmail.com  wrote:

> Hi,
>
> Sorry for the spam last email i miss-clicked on send amidst writing the
> email.
>
> Anyways the problem i am facing is that my ACC module is configured with
> MySQL DB backend and the CDR's are being written. However the problem i am
> facing is that it is not logging duration into DB or syslog. Here are debug
> logs where query is being prepared and inserted
>
>
>- db_mysql_do_prepared_query: new query=|insert into acc(
>
> *method,from_tag,to_tag,callid,sip_code,sip_reason,time,caller_id,callee_id,serverid,info,billResponse,balance*
>) values (?,?,?,?,?,?,?,?,?,?,?,?,?)|
>- DBG:db_mysql:re_init_statement:  query  is 
> *method,from_tag,to_tag,callid,sip_code,sip_reason,time,caller_id,callee_id,serverid,info,billResponse,balance*
>) values(?,?,?,?,?,?,?,?,?,?,?,?,?)>, ptr=(nil)
>
> The problem is that i dont see duration in this query. Am i missing some
> flag or something that needs to be set for duration to be logged in DB?
>
> *P.S. I am using opensips 1.11*
>
> Regards,
>
> Qasim
>
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Re: [OpenSIPS-Users] ACC db duration

[Daniel Zanutti]

Hi

Did you check "cdr_flag" on Acc module?

Otherwise it will generate 2 registers, 1 for INVITE and 1 for BYE.

Regards


On Tue, Mar 28, 2017 at 10:09 AM, qasimak...@gmail.com  wrote:

> Hi,
>
> I have enabled acc module in my opensips installation with db, My CDR's
> are being written in MySQL backend but for every call the duration remains
> 0, I have checked but according to documentation duration is automatically
> logged in ACC module. PLease note following debug filtered where query is
> made and executed
>
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[OpenSIPS-Users] Mediaproxy speed calculations

[Daniel Zanutti]

Hi

Looks like i'm diving deep on mediaproxy.

Some of our relays are not calculating the speed on the network. If I
restart a couple times it starts calculating fine.

I found this log:
media-relay[4100]: warning: Aggregate speed calculation time exceeded 10ms:
11644 us for 222 sessions

Is there any solution to always calculate?

Thanks
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[OpenSIPS-Users] Which internal module did the hangup?

[Daniel Zanutti]

I have 2 modules that may hangup the call:
 - Dialog - duration timeout or sip ping with sip options
 - Mediaproxy - RTP timeout

On local_route, is there any way to know which module did the hangup?

Thanks
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Re: [OpenSIPS-Users] Monitoring Mediaproxy

[Daniel Zanutti]

Just counting how many is probably a good solution and very very simple.

Thanks =)

On Mon, Mar 20, 2017 at 7:00 AM, Carlos Oliva <carlos.ol...@numintec.com>
wrote:

> Hi Daniel!
>
> I made a little php script for Nagios some years ago that may help you.
>
> It is intented to use on opensips machine (where media dispatcher is
> running)
>
> You pass as argument the number of mediaproxy relay machines you expected
> to have and it returnks Ok and the list of your relays or error if the
> number of relays is not what you expect.
>
> It's very simple but works well, we've been using it for years. This is
> the script:
>
>
> #!/usr/bin/php
>  $errno="";
> $errstr="";
> $fp=fsockopen('127.0.0.1','25061',$errno,$errstr,'2');
> fputs($fp, "summary\r\n");
> $line  = fgets($fp);
> fclose($fp);
> $decoded=json_decode($line);
> $num_relays=0;
> $str_salida="";
> foreach($decoded as $relay)
> {
> if($relay->status=="active")
> {
> $num_relays++;
> $str_salida.=" ".$relay->ip;
> }
> }
> if($num_relays==$argv[1])
> {
> echo "OK IPs de Relays RTP: ".$str_salida."\n";
> exit(0);
> }
> else
> {
> echo "ERROR faltan Relays. IPs de Relays RTP: ".$str_salida."\n";
> exit(2);
> }
>
> ?>
>
> Best regards,
>
> Carlos Oliva
>
>
>
>
>
>
>
>
>
> 2017-03-17 18:57 GMT+01:00 Daniel Zanutti <daniel.zanu...@gmail.com>:
>
>> Understood.
>>
>> Thanks for explanation.
>>
>> Regards
>>
>> On Fri, Mar 17, 2017 at 2:47 PM, Dan Pascu <d...@ag-projects.com> wrote:
>>
>>>
>>> On 16 Mar 2017, at 15:58, Daniel Zanutti wrote:
>>>
>>> > Hi Dan
>>> >
>>> > This is exactly how I'm monitoring but looking to the dispatcher it's
>>> kind hard on a Nagios like system, because I'm monitoring Relay A, B and C,
>>> but the status will be on dispatcher machine D. But ok, if it's the only
>>> way.
>>>
>>> The relay doesn't listen for network connections, you cannot connect
>>> directly to a relay. A relay will only connect to all configured
>>> dispatchers. The dispatcher on the other hand has a control port where you
>>> can connect and give commands, including fetching statistics from relays
>>> over the connections the relay already established with the dispatcher.
>>>
>>> --
>>> Dan
>>>
>>>
>>>
>>>
>>>
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>>>
>>
>>
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>>
>>
>
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Re: [OpenSIPS-Users] Mediaproxy hanging sessions on high load

[Daniel Zanutti]

Hi Dan

Thanks for replying.

I understand that best case scenario is to run on a physical dedicated
server, but unfortunately this is impossible on all cases and virtual
machines is the only viable ($$) solution.

About the "frozen" sessions, as as workaround, I'll test dropping these
sessions directly using dispatcher interface. If it works, a console will
solve the problem.

The problem is that I cannot reproduce this scenario, on some clients it
happens, on some I have same quantity of calls and never happens. I'll try
to find a workaround on the timer.

Thanks for all help, you guys are great.


On Fri, Mar 17, 2017 at 3:08 PM, Dan Pascu <d...@ag-projects.com> wrote:

>
> On 17 Mar 2017, at 3:54, Daniel Zanutti wrote:
>
> > Adrian
> >
> > You may be correct, overload can be the problem. But since the call is
> already finished, how can I remove from the relay?
>
> One way is to issue commands to the dispatcher to end certain sessions, in
> the same way that opensips issues them when it receives a BYE. But this is
> easier said than done, because you will need to find out the call-id ,
> from-tag and to-tag of the call in order to do that.
>
> At some point we had the idea to add this kind of functionality to the
> monitoring web page allowing you to click a button next to a call to
> forcefully end it, but it never come to fruition.
>
> The only thing you can do for now is make sure you have at least another
> relay connected to the dispatcher, so it can absorb new calls, the run
> /etc/init.d/media-relay stop-gracefully and wait until this relay has no
> more active traffic, then you can run /etc/init.d/media-relay restart to
> restart it.
>
> --
> Dan
>
>
>
>
>
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Re: [OpenSIPS-Users] Monitoring Mediaproxy

[Daniel Zanutti]

Understood.

Thanks for explanation.

Regards

On Fri, Mar 17, 2017 at 2:47 PM, Dan Pascu <d...@ag-projects.com> wrote:

>
> On 16 Mar 2017, at 15:58, Daniel Zanutti wrote:
>
> > Hi Dan
> >
> > This is exactly how I'm monitoring but looking to the dispatcher it's
> kind hard on a Nagios like system, because I'm monitoring Relay A, B and C,
> but the status will be on dispatcher machine D. But ok, if it's the only
> way.
>
> The relay doesn't listen for network connections, you cannot connect
> directly to a relay. A relay will only connect to all configured
> dispatchers. The dispatcher on the other hand has a control port where you
> can connect and give commands, including fetching statistics from relays
> over the connections the relay already established with the dispatcher.
>
> --
> Dan
>
>
>
>
>
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Re: [OpenSIPS-Users] Mediaproxy hanging sessions on high load

[Daniel Zanutti]

Adrian

You may be correct, overload can be the problem. But since the call is
already finished, how can I remove from the relay? The final problem is the
relay processing freezing, i need to avoid this.

Thanks

On Thu, Mar 16, 2017 at 10:40 PM, Adrian Georgescu <a...@ag-projects.com>
wrote:

> Perhaps your virtual machine simply cannot handle the load. The commands
> to close sessions may also be dropped or lost under such environment.
>
> Adrian
>
>
>
> On 16 Mar 2017, at 11:22, Daniel Zanutti <daniel.zanu...@gmail.com> wrote:
>
> Hi Dan
>
> Looks like this problem is only happening on virtual machines, not on
> physical machines. And only while they are on high load.
>
> But i'm not sure about the kernel rule, is there any way to check it?
>
> Please take a look at this case, this Relay will never halt because there
> are more than 3k sessions that will never finish internally (the call has
> already hangup hours ago):
>
> 8 2.2.2.2 2.6.1 44h01'05"
> 112.03kbps 3045
> audio 3045 Halting
>
> Some of these calls:
>
>
>
>
>
>
>
>
>
>
>
>
> 728 *From:* 22@4.4.4.4
> *To:* 3...@sip.aaa.com.br
> [image: unknown agent] [image: HG4000/1.0] 6.6.6.6:55632 2.2.2.2:46640
> 2.2.2.2:46866 7.7.7.7:4170 active G729 audio 21h35'34" 0 0
> 729 *From:* 222@4.4.4.4:5064
> *To:* 3...@sip.aaa.com.br
> [image: TS-v4.6.0-11eW] [image: Agitel GSM Bridge v2.0] 6.6.6.6:34908
> 2.2.2.2:58158 2.2.2.2:54372 7.7.7.7:16846 active G729 audio 16h11'51" 0 0
> 730 *From:* @4.4.4.4
> *To:* 3...@sip.aaa.com.br
> [image: Mediant 2000/v.6.60A.328.003] [image: unknown agent] 6.6.6.6:46324
> 2.2.2.2:50156 2.2.2.2:48182 7.7.7.7:18516 active G729 audio 19h45'38" 0 0
> 731 *From:* 22@4.4.4.4:5061
> *To:* ...@sip.aaa.com.br
> [image: TS-v4.6.0-14b] [image: gsm-gw-3.4.1] 6.6.6.6:54800 2.2.2.2:43998
> 2.2.2.2:46144 7.7.7.7:12360 active G729 audio 19h09'41" 0 0
> 732 *From:* 222@4.4.4.4
> *To:* 3...@sip.aaa.com.br
> [image: Trinit IVR] [image: HG4000/1.0] 6.6.6.6:18854 2.2.2.2:51924
> 2.2.2.2:40512 7.7.7.7:4200 active G729 audio 19h37'59" 0 0
>
> Is there any way to drop these sessions? Maybe using the internal timeout
> system of mediaproxy?
>
> If you could take a look personally, we could negotiate an hourly rate.
>
> Thanks again
>
>
>
> On Thu, Mar 16, 2017 at 10:54 AM, Dan Pascu <d...@ag-projects.com> wrote:
>
>>
>> One thing came to mind. A case when the relay could get overloaded is if
>> a lot of clients start sessions and only one endpoint sends media. That is
>> the only case where the relay would have to deal with the media traffic
>> itself and having hundreds of such sessions at the same time could overload
>> the relay.
>>
>> The way the relay works is for each call it starts listening on 4 ports
>> (2 for RTP and 2 for RTCP). Each endpoint will send 2 streams (1 RTP one
>> RTCP) and initially the relay will just listen on these ports and when it
>> receives data it learns the endpoint's address. After it learns both
>> endpoint's addresses, it adds a conntrack rule in the kernel to allow the
>> kernel to directly relay the media streams between the endpoints and it
>> will never see a media packet from the endpoints again until the call ends.
>> This allows for very efficient data forwarding because it's done entirely
>> in the kernel with no data being transferred from kernel to user-space and
>> back like traditional solutions. We have seen media relays handling
>> hundreds of calls at a time with 0% CPU load on the relay.
>>
>> So the only thing I can think of causing something like what you describe
>> (even though I'm still not sure what you meant by hanging up sessions), is
>> that somehow this process didn't finish setting up completely and the relay
>> directly receives media streams from hundreds of devices because only one
>> endpoint sends data (or the other endpoint's data gets filtered at some
>> firewall), and because it cannot learn both endpoint's addresses it cannot
>> setup the kernel conntrack rule to move data forwarding to the kernel.
>>
>> On 14 Mar 2017, at 13:38, Dan Pascu wrote:
>>
>> >
>> > On 13 Mar 2017, at 18:58, Daniel Zanutti wrote:
>> >
>> >> Hi guys
>> >>
>> >> I sent this email a few days ago, anyone from Mediaproxy team could
>> take a look? I could debug it, just need some directions on where to look.
>> >
>> > We have never encountered this problem, so I', not sure what to
>> suggest, even more considering that the description is not v

Re: [OpenSIPS-Users] Monitoring Mediaproxy

[Daniel Zanutti]

Hi Robert

Acording to mediaproxy website:

MediaProxy is a media relay for RTP/RTCP and UDP streams that works in
tandem with OpenSIPS to provide NAT traversal capability for media streams
from SIP user agents located behind NAT. MediaProxy supports ICE
negotiation by behaving like a TURN relay candidate and the policy can be
controlled from OpenSIPS configuration.

Take a look at its website: http://mediaproxy.ag-projects.com/

Regards

On Thu, Mar 16, 2017 at 11:22 AM, Mundkowsky, Robert <rmundkow...@ets.org>
wrote:

> Sorry, I don't know much about telephony.
>
> Curious what I had wrong?
>
> Is it that only media is sent to Mediaproxy without SIP?
>
> Robert
>
>
> -Original Message-
> From: Users [mailto:users-boun...@lists.opensips.org] On Behalf Of Dan
> Pascu
> Sent: Thursday, March 16, 2017 9:41 AM
> To: OpenSIPS users mailling list <users@lists.opensips.org>
> Subject: Re: [OpenSIPS-Users] Monitoring Mediaproxy
>
> Lol, what?!?
>
> On 15 Mar 2017, at 23:03, Mundkowsky, Robert wrote:
>
> > SIP Options is used as a “SIP ping”. You likely can have an time event
> trigger route that can send one and then based on that disable/enable
> accordingly.  Hopefully mediaproxy will not respond to the “SIP ping” if
> frozen.
> >
> > Robert
> >
> > From: Users [mailto:users-boun...@lists.opensips.org] On Behalf Of
> Daniel Zanutti
> > Sent: Wednesday, March 15, 2017 4:55 PM
> > To: OpenSIPS users mailling list <users@lists.opensips.org>
> > Subject: Re: [OpenSIPS-Users] Monitoring Mediaproxy
> >
> > How can this be done?
> >
> > Or do you mean SIP options?
> >
> > On Wed, Mar 15, 2017 at 5:45 PM, Johan De Clercq <jo...@democon.be>
> wrote:
> > Send options.
> >
> > On 15 Mar 2017 11:48 PM, "Daniel Zanutti" <daniel.zanu...@gmail.com>
> wrote:
> > Hi
> >
> > What's the best way to check if a mediaproxy is running fine? Monit is
> monitoring PID but how can I check the process has is not frozen?
> >
> > Thanks
> >
> > ___
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> >
> >
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> >
> >
> >
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> in error, please notify the sender; do not disclose, copy, distribute, or
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> it from your system. Any other use of this e-mail is prohibited.
> >
> >
> > Thank you for your compliance.
> >
> > ___
> > Users mailing list
> > Users@lists.opensips.org
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
> --
> Dan
>
>
>
>
>
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> 
>
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Re: [OpenSIPS-Users] Mediaproxy hanging sessions on high load

[Daniel Zanutti]

Hi Dan

Looks like this problem is only happening on virtual machines, not on
physical machines. And only while they are on high load.

But i'm not sure about the kernel rule, is there any way to check it?

Please take a look at this case, this Relay will never halt because there
are more than 3k sessions that will never finish internally (the call has
already hangup hours ago):

8 2.2.2.2 2.6.1 44h01'05"
112.03kbps 3045
audio 3045 Halting

Some of these calls:












728 *From:* 22@4.4.4.4
*To:* 3...@sip.aaa.com.br
[image: unknown agent] [image: HG4000/1.0] 6.6.6.6:55632 2.2.2.2:46640
2.2.2.2:46866 7.7.7.7:4170 active G729 audio 21h35'34" 0 0
729 *From:* 222@4.4.4.4:5064
*To:* 3...@sip.aaa.com.br
[image: TS-v4.6.0-11eW] [image: Agitel GSM Bridge v2.0] 6.6.6.6:34908
2.2.2.2:58158 2.2.2.2:54372 7.7.7.7:16846 active G729 audio 16h11'51" 0 0
730 *From:* @4.4.4.4
*To:* 3...@sip.aaa.com.br
[image: Mediant 2000/v.6.60A.328.003] [image: unknown agent] 6.6.6.6:46324
2.2.2.2:50156 2.2.2.2:48182 7.7.7.7:18516 active G729 audio 19h45'38" 0 0
731 *From:* 22@4.4.4.4:5061
*To:* ...@sip.aaa.com.br
[image: TS-v4.6.0-14b] [image: gsm-gw-3.4.1] 6.6.6.6:54800 2.2.2.2:43998
2.2.2.2:46144 7.7.7.7:12360 active G729 audio 19h09'41" 0 0
732 *From:* 222@4.4.4.4
*To:* 3...@sip.aaa.com.br
[image: Trinit IVR] [image: HG4000/1.0] 6.6.6.6:18854 2.2.2.2:51924
2.2.2.2:40512 7.7.7.7:4200 active G729 audio 19h37'59" 0 0

Is there any way to drop these sessions? Maybe using the internal timeout
system of mediaproxy?

If you could take a look personally, we could negotiate an hourly rate.

Thanks again



On Thu, Mar 16, 2017 at 10:54 AM, Dan Pascu <d...@ag-projects.com> wrote:

>
> One thing came to mind. A case when the relay could get overloaded is if a
> lot of clients start sessions and only one endpoint sends media. That is
> the only case where the relay would have to deal with the media traffic
> itself and having hundreds of such sessions at the same time could overload
> the relay.
>
> The way the relay works is for each call it starts listening on 4 ports (2
> for RTP and 2 for RTCP). Each endpoint will send 2 streams (1 RTP one RTCP)
> and initially the relay will just listen on these ports and when it
> receives data it learns the endpoint's address. After it learns both
> endpoint's addresses, it adds a conntrack rule in the kernel to allow the
> kernel to directly relay the media streams between the endpoints and it
> will never see a media packet from the endpoints again until the call ends.
> This allows for very efficient data forwarding because it's done entirely
> in the kernel with no data being transferred from kernel to user-space and
> back like traditional solutions. We have seen media relays handling
> hundreds of calls at a time with 0% CPU load on the relay.
>
> So the only thing I can think of causing something like what you describe
> (even though I'm still not sure what you meant by hanging up sessions), is
> that somehow this process didn't finish setting up completely and the relay
> directly receives media streams from hundreds of devices because only one
> endpoint sends data (or the other endpoint's data gets filtered at some
> firewall), and because it cannot learn both endpoint's addresses it cannot
> setup the kernel conntrack rule to move data forwarding to the kernel.
>
> On 14 Mar 2017, at 13:38, Dan Pascu wrote:
>
> >
> > On 13 Mar 2017, at 18:58, Daniel Zanutti wrote:
> >
> >> Hi guys
> >>
> >> I sent this email a few days ago, anyone from Mediaproxy team could
> take a look? I could debug it, just need some directions on where to look.
> >
> > We have never encountered this problem, so I', not sure what to suggest,
> even more considering that the description is not very clear. What do you
> mean when you say the relay starts to hang some sessions? That it timeouts
> on them not having traffic and initiates a BYE for those sessions? Because
> in the next paragraph you imply that they never timeout.
> >
> >>
> >> Thanks
> >>
> >> On Tue, Mar 7, 2017 at 11:10 AM, Daniel Zanutti <
> daniel.zanu...@gmail.com> wrote:
> >> I'm using mediaproxy on several installations and have noticed that
> when the machine is on high load (> 700 sessions), the media-relay process
> starts to hang some sessions.
> >>
> >> These sessions doesn't have any RTP being sent/received anymore and
> they never hangup. After some hours of frozen sessions, the media-relay
> process doesn't connect to the dispatcher anymore, but keep using high CPU
> on the machine. Maybe it's on loop internally, not sure.
> >>
> >> Is ther

Re: [OpenSIPS-Users] Monitoring Mediaproxy

[Daniel Zanutti]

Hi Dan

This is exactly how I'm monitoring but looking to the dispatcher it's kind
hard on a Nagios like system, because I'm monitoring Relay A, B and C, but
the status will be on dispatcher machine D. But ok, if it's the only way.

Thanks

---

"Lol, what?!?"

I preferred to drop this =)

On Thu, Mar 16, 2017 at 10:41 AM, Dan Pascu <d...@ag-projects.com> wrote:

> Lol, what?!?
>
> On 15 Mar 2017, at 23:03, Mundkowsky, Robert wrote:
>
> > SIP Options is used as a “SIP ping”. You likely can have an time event
> trigger route that can send one and then based on that disable/enable
> accordingly.  Hopefully mediaproxy will not respond to the “SIP ping” if
> frozen.
> >
> > Robert
> >
> > From: Users [mailto:users-boun...@lists.opensips.org] On Behalf Of
> Daniel Zanutti
> > Sent: Wednesday, March 15, 2017 4:55 PM
> > To: OpenSIPS users mailling list <users@lists.opensips.org>
> > Subject: Re: [OpenSIPS-Users] Monitoring Mediaproxy
> >
> > How can this be done?
> >
> > Or do you mean SIP options?
> >
> > On Wed, Mar 15, 2017 at 5:45 PM, Johan De Clercq <jo...@democon.be>
> wrote:
> > Send options.
> >
> > On 15 Mar 2017 11:48 PM, "Daniel Zanutti" <daniel.zanu...@gmail.com>
> wrote:
> > Hi
> >
> > What's the best way to check if a mediaproxy is running fine? Monit is
> monitoring PID but how can I check the process has is not frozen?
> >
> > Thanks
> >
> > ___
> > 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
> >
> >
> >
> > This e-mail and any files transmitted with it may contain privileged or
> confidential information. It is solely for use by the individual for whom
> it is intended, even if addressed incorrectly. If you received this e-mail
> in error, please notify the sender; do not disclose, copy, distribute, or
> take any action in reliance on the contents of this information; and delete
> it from your system. Any other use of this e-mail is prohibited.
> >
> >
> > Thank you for your compliance.
> >
> > ___
> > Users mailing list
> > Users@lists.opensips.org
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
> --
> Dan
>
>
>
>
>
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Re: [OpenSIPS-Users] Monitoring Mediaproxy

[Daniel Zanutti]

How can this be done?

Or do you mean SIP options?

On Wed, Mar 15, 2017 at 5:45 PM, Johan De Clercq <jo...@democon.be> wrote:

> Send options.
>
> On 15 Mar 2017 11:48 PM, "Daniel Zanutti" <daniel.zanu...@gmail.com>
> wrote:
>
>> Hi
>>
>> What's the best way to check if a mediaproxy is running fine? Monit is
>> monitoring PID but how can I check the process has is not frozen?
>>
>> Thanks
>>
>> ___
>> Users mailing list
>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
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[OpenSIPS-Users] Monitoring Mediaproxy

[Daniel Zanutti]

Hi

What's the best way to check if a mediaproxy is running fine? Monit is
monitoring PID but how can I check the process has is not frozen?

Thanks
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Re: [OpenSIPS-Users] Mediaproxy hanging sessions on high load

[Daniel Zanutti]

Hi guys

I sent this email a few days ago, anyone from Mediaproxy team could take a
look? I could debug it, just need some directions on where to look.

Thanks

On Tue, Mar 7, 2017 at 11:10 AM, Daniel Zanutti <daniel.zanu...@gmail.com>
wrote:

> I'm using mediaproxy on several installations and have noticed that when
> the machine is on high load (> 700 sessions), the media-relay process
> starts to hang some sessions.
>
> These sessions doesn't have any RTP being sent/received anymore and they
> never hangup. After some hours of frozen sessions, the media-relay process
> doesn't connect to the dispatcher anymore, but keep using high CPU on the
> machine. Maybe it's on loop internally, not sure.
>
> Is there any solution for this? Maybe a timer to cleanup old sessions (2
> or 4+ hours old).
>
> Thanks
>
>
>
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Re: [OpenSIPS-Users] Mediaproxy hanging sessions on high load

[Daniel Zanutti]

Any idea guys?

On Tue, Mar 7, 2017 at 11:10 AM, Daniel Zanutti <daniel.zanu...@gmail.com>
wrote:

> I'm using mediaproxy on several installations and have noticed that when
> the machine is on high load (> 700 sessions), the media-relay process
> starts to hang some sessions.
>
> These sessions doesn't have any RTP being sent/received anymore and they
> never hangup. After some hours of frozen sessions, the media-relay process
> doesn't connect to the dispatcher anymore, but keep using high CPU on the
> machine. Maybe it's on loop internally, not sure.
>
> Is there any solution for this? Maybe a timer to cleanup old sessions (2
> or 4+ hours old).
>
> Thanks
>
>
>
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[OpenSIPS-Users] Mediaproxy hanging sessions on high load

[Daniel Zanutti]

I'm using mediaproxy on several installations and have noticed that when
the machine is on high load (> 700 sessions), the media-relay process
starts to hang some sessions.

These sessions doesn't have any RTP being sent/received anymore and they
never hangup. After some hours of frozen sessions, the media-relay process
doesn't connect to the dispatcher anymore, but keep using high CPU on the
machine. Maybe it's on loop internally, not sure.

Is there any solution for this? Maybe a timer to cleanup old sessions (2 or
4+ hours old).

Thanks
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Re: [OpenSIPS-Users] OpenSIPS Features

[Daniel Zanutti]

I think you have a wrong answer:

2) Yes, you can. But the SIP Signaling part is only half part of the
problem. Checkout this sip client: http://icanblink.com/

It implements desktop sharing over SIP and yes, they use Opensips.

Regards

On Sat, Feb 25, 2017 at 2:04 PM, Voice TAC  wrote:

> Thank you for your email :)
>
>
> Sent from my iPhone
>
> On Feb 25, 2017, at 12:23 PM, M. Salman  wrote:
>
> Hi,
>
> 1) Yes
> 2) No - It is more of a client side feature, would be better if you
> understand OpenSIPS is a SIP Signaling server, you can build various
> products using it in combination with other technologies.
>
>
> Regards,
> M. Salman
>
> On Sat, Feb 25, 2017 at 2:06 PM, Voice TAC  wrote:
>
>> Hello,
>>
>> I read about OpenSIPS and it looks a very powerful product. However,
>> before digging and learning it I would like to know if the following are
>> achievable. Could you please help me by adding Yes or No :)
>>
>> 1- Can I have instant messaging and presence status in OpenSIPS?
>>
>> 2- Can I implement desktop sharing between the clients?
>>
>>  Regards,
>>
>> ___
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>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>
>
> --
> Regards
>
> M. Salman
> VoIP Professional
>
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Re: [OpenSIPS-Users] Change 200 OK of BYE message

[Daniel Zanutti]

Hi

No, i didn't set it. After set the onreply, it worked fine. I didn't know
that the BYE is a new request internally.

Thanks Bogdan

On Tue, Feb 7, 2017 at 10:37 AM, Bogdan-Andrei Iancu <bog...@opensips.org>
wrote:

> Hi,
>
> As the 200OK belongs to the BYE request, are you sure you arm the onreply
> route when handling the _BYE_ request ?
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
> On 02/07/2017 01:48 PM, Daniel Zanutti wrote:
>
> The sip client is working fine, I can confirm that both BYE and 200 OK are
> flowing through the proxy, the problem is I cannot modify the 200 OK
> message.
>
> I need to modify a header on the 200 OK of the BYE but I cannot since the
> message doesn't pass to internal on_reply_route or main Route inside
> Opensips. It's being routed internally by Opensips directly to the
> destination.
>
> Do you have any idea?
>
> On Tue, Feb 7, 2017 at 5:40 AM, Johan De Clercq <jo...@democon.be> wrote:
>
>> Any chance to have a small capture ?
>> I tend to agree with Robert, if you use rr consistently there should be
>> no problem. It can of course also be that the client does something strange
>> ...
>>
>> 2017-02-06 21:43 GMT+01:00 Daniel Zanutti < <daniel.zanu...@gmail.com>
>> daniel.zanu...@gmail.com>:
>>
>>> Hi Robert
>>>
>>> Yes, all messages are passing through the proxy, but when I receive the
>>> 200 OK of the BYE message, it doesn't go to Main Route or Reply Route. It
>>> just go to the destination and I cannot change anything on it.
>>>
>>> Any idea?
>>>
>>>
>>>
>>> On Mon, Feb 6, 2017 at 6:14 PM, Mundkowsky, Robert <
>>> <rmundkow...@ets.org>rmundkow...@ets.org> wrote:
>>>
>>>> Did you use “record_route”?
>>>>
>>>>
>>>>
>>>> For reference:
>>>>
>>>> http://www.iptel.org/sip/intro/scenarios/rr
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> Robert Mundkowsky
>>>>
>>>>
>>>>
>>>> *From:* Users [mailto: <users-boun...@lists.opensips.org>
>>>> users-boun...@lists.opensips.org] *On Behalf Of *Daniel Zanutti
>>>> *Sent:* Monday, February 6, 2017 3:01 PM
>>>> *To:* OpenSIPS users mailling list <users@lists.opensips.org>
>>>> *Subject:* [OpenSIPS-Users] Change 200 OK of BYE message
>>>>
>>>>
>>>>
>>>> Hi
>>>>
>>>>
>>>>
>>>> I need to change something on the 200 OK of BYE message. Tried
>>>> everything on Opensips but looks like this message doesn't follow standard
>>>> message path. Neither Main Route or Reply route pass this message.
>>>>
>>>>
>>>>
>>>> Is there any way to do it?
>>>>
>>>>
>>>>
>>>> Thanks
>>>>
>>>> --
>>>>
>>>> This e-mail and any files transmitted with it may contain privileged or
>>>> confidential information. It is solely for use by the individual for whom
>>>> it is intended, even if addressed incorrectly. If you received this e-mail
>>>> in error, please notify the sender; do not disclose, copy, distribute, or
>>>> take any action in reliance on the contents of this information; and delete
>>>> it from your system. Any other use of this e-mail is prohibited.
>>>>
>>>> Thank you for your compliance.
>>>> --
>>>>
>>>> ___
>>>> 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
>>>
>>>
>>
>> ___
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>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>
>
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>
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Re: [OpenSIPS-Users] Change 200 OK of BYE message

[Daniel Zanutti]

The sip client is working fine, I can confirm that both BYE and 200 OK are
flowing through the proxy, the problem is I cannot modify the 200 OK
message.

I need to modify a header on the 200 OK of the BYE but I cannot since the
message doesn't pass to internal on_reply_route or main Route inside
Opensips. It's being routed internally by Opensips directly to the
destination.

Do you have any idea?

On Tue, Feb 7, 2017 at 5:40 AM, Johan De Clercq <jo...@democon.be> wrote:

> Any chance to have a small capture ?
> I tend to agree with Robert, if you use rr consistently there should be no
> problem. It can of course also be that the client does something strange
> ...
>
> 2017-02-06 21:43 GMT+01:00 Daniel Zanutti <daniel.zanu...@gmail.com>:
>
>> Hi Robert
>>
>> Yes, all messages are passing through the proxy, but when I receive the
>> 200 OK of the BYE message, it doesn't go to Main Route or Reply Route. It
>> just go to the destination and I cannot change anything on it.
>>
>> Any idea?
>>
>>
>>
>> On Mon, Feb 6, 2017 at 6:14 PM, Mundkowsky, Robert <rmundkow...@ets.org>
>> wrote:
>>
>>> Did you use “record_route”?
>>>
>>>
>>>
>>> For reference:
>>>
>>>     http://www.iptel.org/sip/intro/scenarios/rr
>>>
>>>
>>>
>>>
>>>
>>> Robert Mundkowsky
>>>
>>>
>>>
>>> *From:* Users [mailto:users-boun...@lists.opensips.org] *On Behalf Of 
>>> *Daniel
>>> Zanutti
>>> *Sent:* Monday, February 6, 2017 3:01 PM
>>> *To:* OpenSIPS users mailling list <users@lists.opensips.org>
>>> *Subject:* [OpenSIPS-Users] Change 200 OK of BYE message
>>>
>>>
>>>
>>> Hi
>>>
>>>
>>>
>>> I need to change something on the 200 OK of BYE message. Tried
>>> everything on Opensips but looks like this message doesn't follow standard
>>> message path. Neither Main Route or Reply route pass this message.
>>>
>>>
>>>
>>> Is there any way to do it?
>>>
>>>
>>>
>>> Thanks
>>>
>>> --
>>>
>>> This e-mail and any files transmitted with it may contain privileged or
>>> confidential information. It is solely for use by the individual for whom
>>> it is intended, even if addressed incorrectly. If you received this e-mail
>>> in error, please notify the sender; do not disclose, copy, distribute, or
>>> take any action in reliance on the contents of this information; and delete
>>> it from your system. Any other use of this e-mail is prohibited.
>>>
>>> Thank you for your compliance.
>>> --
>>>
>>> ___
>>> Users mailing list
>>> Users@lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>>
>>
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>>
>>
>
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Re: [OpenSIPS-Users] Change 200 OK of BYE message

[Daniel Zanutti]

Hi Robert

Yes, all messages are passing through the proxy, but when I receive the 200
OK of the BYE message, it doesn't go to Main Route or Reply Route. It just
go to the destination and I cannot change anything on it.

Any idea?



On Mon, Feb 6, 2017 at 6:14 PM, Mundkowsky, Robert <rmundkow...@ets.org>
wrote:

> Did you use “record_route”?
>
>
>
> For reference:
>
> http://www.iptel.org/sip/intro/scenarios/rr
>
>
>
>
>
> Robert Mundkowsky
>
>
>
> *From:* Users [mailto:users-boun...@lists.opensips.org] *On Behalf Of *Daniel
> Zanutti
> *Sent:* Monday, February 6, 2017 3:01 PM
> *To:* OpenSIPS users mailling list <users@lists.opensips.org>
> *Subject:* [OpenSIPS-Users] Change 200 OK of BYE message
>
>
>
> Hi
>
>
>
> I need to change something on the 200 OK of BYE message. Tried everything
> on Opensips but looks like this message doesn't follow standard message
> path. Neither Main Route or Reply route pass this message.
>
>
>
> Is there any way to do it?
>
>
>
> Thanks
>
> --
>
> This e-mail and any files transmitted with it may contain privileged or
> confidential information. It is solely for use by the individual for whom
> it is intended, even if addressed incorrectly. If you received this e-mail
> in error, please notify the sender; do not disclose, copy, distribute, or
> take any action in reliance on the contents of this information; and delete
> it from your system. Any other use of this e-mail is prohibited.
>
> Thank you for your compliance.
> --
>
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> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
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[OpenSIPS-Users] Change 200 OK of BYE message

[Daniel Zanutti]

Hi

I need to change something on the 200 OK of BYE message. Tried everything
on Opensips but looks like this message doesn't follow standard message
path. Neither Main Route or Reply route pass this message.

Is there any way to do it?

Thanks
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Re: [OpenSIPS-Users] How to Keep OpenSIPS in reverse path

[Daniel Zanutti]

I also suggest use topology_hiding(), i saw some equipments ignoring
Via/record-route order.

On Wed, Dec 14, 2016 at 3:53 PM, Muhammad Naseer Bhatti 
wrote:

>
> Hi Razvan,
> I am not using REGISTER, but I guess add_path() wont’ work for me, I am
> using record_route() for the INVITE though.
>
> …
> if (is_method("INVITE")) {
> record_route();
> }
> …
>
> On the media server I see the Via header,
>
>INVITE sip:6054775550@64.58.228.102: SIP/2.0
>Record-Route:  BRAp6IAQSKwtAN3Mjj2bPKL>
>Via: SIP/2.0/UDP 64.58.228.102:;branch=z9hG4bK31d2.
> b8fe57fe9c7545e354a93f391a1d0704.0
>Via: SIP/2.0/UDP 172.16.0.101:52207;received=172.16.0.101;rport=52207;
> branch=z9hG4bKPjyGufepGSH2zMGfp6J.CIKvEgl87YIPFN
>
> but when the media server sends
>
> recv 1432 bytes from udp/[64.58.228.89]: at 12:44:12.709234:
>---
> -
>INVITE sip:6054775550@23.29.112.144:15080 SIP/2.0
>Via: SIP/2.0/UDP 64.58.228.89:;rport;branch=z9hG4bKF8jFHt4maUBeQ
>Max-Forwards: 68
>From: "Naseer" ;tag=45yZ9S13S47HF
>
> Not sure what I am doing wrong. Here is my script, http://pastebin.com/
> Cmnxnf4c
>
>
> --
>
> Sent with Airmail
>
> From: Răzvan Crainea  
> Reply: OpenSIPS users mailling list 
> 
> Date: December 14, 2016 at 8:34:22 PM
> To: users@lists.opensips.org 
> 
> Subject:  Re: [OpenSIPS-Users] How to Keep OpenSIPS in reverse path
>
> Hi, Muhammad!
>
> The add_path() function should only be called on REGISTER messages, and it
> adds a Path header (not a Via). Do you see this header in your REGISTER
> message?
> For sequential requests, you should use the record_route() mechanism.
>
> Best regards,
>
> Răzvan Crainea
> OpenSIPS Solutionswww.opensips-solutions.com
>
> On 12/14/2016 06:39 PM, Muhammad Naseer Bhatti wrote:
>
>
> I am using dispatcher to distribute calls to multiple media servers, but
> also want the reply to go through OpenSIPS. So far I have tried add_path()
> function which add the Via header but FreeSWITCH sends the call directly to
> the gateway not sending the call back through OpenSIPS. I am not sure if I
> am doing something wrong, how can I achieve that?
>
>
> --
>
> Sent with Airmail
>
>
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[OpenSIPS-Users] Github x gzip version

[Daniel Zanutti]

Hi

Just a question about which version to use:

Is it safe to use the latest Github version of 1.11.x or is safe to use the
.tar.gz version?
My point is: Can I trust github version to use in production?

Thanks
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