Re: [OpenSIPS-Users] Removing Identity hdr
Thanks, yes I know the logic behind it, but nobody is going to reject calls because they have no caller id signed. That’s money. And this won’t stop caller id spoofing either. It’s very naive to think so, IMO. Hopefully it does , though! But telemarketers ingenuity never ceases to amaze me. Regards, David Villasmil email: david.villasmil.w...@gmail.com On Fri, 5 Jul 2024 at 13:36, Alain Bieuzent wrote: > Hi David, > > > > The implementation of stir-shaken in France is different from in the US. > > The text which requires French operators to implement stir-shaken aims to > stop the usurpation of caller-id. > > So, what is asked of the operator is to check their customer's caller-id > and sign outgoing calls, in the event of fraud, it will then be easy to > trace the malicious operator. > > For operators receiving unsigned or incorrectly signed traffic, the call > must be disconnected. > > > > for the case of call forwarding (A -> B then B-> C), there will therefore > be two signatures, a first issued by the operator of A and which will be > controlled by the operator of B. Then operator B will add his own signature > (in addition to that of A), both signatures will be controlled by C > > > > Regards > > *De : *Users au nom de David Villasmil > > *Répondre à : *OpenSIPS users mailling list > *Date : *vendredi 5 juillet 2024 à 11:39 > *À : *OpenSIPS users mailling list > *Objet : *Re: [OpenSIPS-Users] Removing Identity hdr > > > > this is really getting ridiculous... and they think they can stop > robocalls with this.. they never will. > > Regards, > > > > David Villasmil > > email: david.villasmil.w...@gmail.com > > > > > > > > On Tue, Jun 18, 2024 at 10:56 AM Alain Bieuzent > wrote: > > Hi, > > > > interesting question, because in future developments of stir/shaken in > France, for forwarded calls, it is planned that the identity field received > on the incoming call be forwarded to the outgoing leg but also to add a > signature (with the local certificate) on the outgoing call (so two > identity fields). > > > > Regards > > > > *De : *Users au nom de Srigo > Kanapathipillai > *Répondre à : *OpenSIPS users mailling list > *Date : *mardi 18 juin 2024 à 08:34 > *À : *OpenSIPS users mailling list > *Objet : *[OpenSIPS-Users] Removing Identity hdr > > > > Hi, > > > > I'm encountering an issue with removing an Identity header in OpenSIPS > 3.4. Here’s the situation: > > > > 1. An incoming call with an Identity header is received. > > 2. I perform a `stir_shaken_verify()` and remove the Identity header in a > request route. > > 3. The call is forwarded to an upstream server, but it fails. > > 4. In the `failure_route`, I need to forward the call to a PSTN number. > > > > 5. Before sending the call to the PSTN (in compliance with French > STIR/SHAKEN regulations), I need to sign it with my certificate. > > > > However, when I call `stir_shaken_auth()`, I receive an error -2 > indicating that the Identity header already exists. Despite running > `remove_hf(identity)` before calling this function, the header isn't > removed, and `$hdr(identity)` still returns the initial value of the > Identity header. > > > > What is the best way to remove the existing Identity header and re-sign > the call? > > > > Thank you, > > Srigo > > ___ Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > ___ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > ___ Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > ___ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Removing Identity hdr
this is really getting ridiculous... and they think they can stop robocalls with this.. they never will. Regards, David Villasmil email: david.villasmil.w...@gmail.com On Tue, Jun 18, 2024 at 10:56 AM Alain Bieuzent wrote: > Hi, > > > > interesting question, because in future developments of stir/shaken in > France, for forwarded calls, it is planned that the identity field received > on the incoming call be forwarded to the outgoing leg but also to add a > signature (with the local certificate) on the outgoing call (so two > identity fields). > > > > Regards > > > > *De : *Users au nom de Srigo > Kanapathipillai > *Répondre à : *OpenSIPS users mailling list > *Date : *mardi 18 juin 2024 à 08:34 > *À : *OpenSIPS users mailling list > *Objet : *[OpenSIPS-Users] Removing Identity hdr > > > > Hi, > > > > I'm encountering an issue with removing an Identity header in OpenSIPS > 3.4. Here’s the situation: > > > > 1. An incoming call with an Identity header is received. > > 2. I perform a `stir_shaken_verify()` and remove the Identity header in a > request route. > > 3. The call is forwarded to an upstream server, but it fails. > > 4. In the `failure_route`, I need to forward the call to a PSTN number. > > > > 5. Before sending the call to the PSTN (in compliance with French > STIR/SHAKEN regulations), I need to sign it with my certificate. > > > > However, when I call `stir_shaken_auth()`, I receive an error -2 > indicating that the Identity header already exists. Despite running > `remove_hf(identity)` before calling this function, the header isn't > removed, and `$hdr(identity)` still returns the initial value of the > Identity header. > > > > What is the best way to remove the existing Identity header and re-sign > the call? > > > > Thank you, > > Srigo > > ___ Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > ___ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Feedback about opensips-js
Hi everyone, Did some of you tried and succeeded to play with https://opensips-js.pages.dev<https://opensips-js.pages.dev/> ? or any piece of its ecosystem ? I know the project is quite new, is this mailing list available to discuss about it, or should I use any other media ? David ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Incoming Call
I don’t know, but from what you say, it sounds like a page thing, not opensips. Check the web page Regards, David Villasmil email: david.villasmil.w...@gmail.com phone: +34669448337 On Sun, 17 Mar 2024 at 14:19, Prathibha B wrote: > This happens once in a minutes. Once a minute, the reregistration happens > and the page reloads itself. > > On Sun, 17 Mar 2024 at 18:31, Prathibha B > wrote: > >> The call is generate once in a minute. >> >> On Sun, 17 Mar 2024 at 18:22, Prathibha B >> wrote: >> >>> Automatically the web page with browser phone url initiates the call >>> without me loading the webpage. >>> >>> Attachment: opensips.cfg >>> >>> -- >>> Regards, >>> B.Prathibha >>> >> >> >> -- >> Regards, >> B.Prathibha >> > > > -- > Regards, > B.Prathibha > ___ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Ringing time
Timeout? If so then https://opensips.org/html/docs/modules/2.3.x/tm.html#fr_timeout Regards, David Villasmil email: david.villasmil.w...@gmail.com phone: +34669448337 On Sun, 24 Dec 2023 at 10:15, Prathibha B wrote: > How to increase the ringing time? > > Sent from Outlook for Android <https://aka.ms/AAb9ysg> > ___ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Need some help in lookup flag "B" in opensips 3.2 version .
And what do you mean by “opensips creates a new invite as the caller sent”?
On Mon, 13 Nov 2023 at 08:57, Bogdan-Andrei Iancu
wrote:
> Hi,
>
> May I ask why using the "B" flag here ?
>
> Regards,
>
> Bogdan-Andrei Iancu
>
> OpenSIPS Founder and Developer
> https://www.opensips-solutions.com
> https://www.siphub.com
>
> On 11/7/23 9:00 AM, Sasmita Panda wrote:
>
> Hi All ,
>
> I am using opensips 3.2 version .
> route {
> ---
> if ($rm=="INVITE")
> {
>if(!lookup("location","B"))
> {
>if (!t_reply(404, "Not Found"))
> {
> sl_reply_error();
> }
> exit;
> }
> }
> if (!serialize_branches(1)){
> sl_send_reply(500,"Unable to load
> contacts");
> exit;
> }else{
># if (next_branches()){
> t_on_failure("1");
>}
> }
>
> Then of course the route(1) and failure_route(1) i have called . But what
> is happening in my case is . I have 2 branches , For Invite opensips tries
> both the branches but if both branch wont accept the call Opensips should
> reply error code to the caller . But Opensips create a new invite as the
> Caller sent and again follows the lookup logic .
>
> Why is this happening ? How will I manage this ?
>
>
> *Thanks & Regards*
> *Sasmita Panda*
> *Senior Network Testing and Software Engineer*
> *3CLogic , ph:07827611765*
>
> ___
> Users mailing
> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
___
Users mailing list
Users@lists.opensips.org
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Re: [OpenSIPS-Users] Too many Hops
Make you opensips finds a next hope. This usually happens when no next hope is set and opensips forwards to itself. On Sat, 23 Sep 2023 at 07:59, Prathibha B wrote: > While executing opensips using sipml5, I'm getting Too Mnay Hops error. > This occurs when I am using ip address instead of domain name for the > websocket url. How to resolve this? > > > -- > Regards, > B.Prathibha > ___ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Stir ans Shaken - number is not in E.164 format
damn... it seems there's a new law in France to do stir/shaken...
Regards,
David Villasmil
email: david.villasmil.w...@gmail.com
phone: +34669448337
On Wed, Sep 6, 2023 at 2:38 PM Mickael Hubert wrote:
> We are deploying it in France.
> In France on providers interconnections, we can see a format (made in
> France maybe ;) )
> prefix: +33
> portability prefix: 10200
> phonenumber national format without 0: 123456789
>
> ++
>
>
> Le mer. 6 sept. 2023 à 14:30, David Villasmil <
> david.villasmil.w...@gmail.com> a écrit :
>
>> Is ST/SH being used other than the US? AFAIK it only applies to US
>> numbers, thus 10 digits, no?
>>
>> On Wed, 6 Sep 2023 at 14:27, Mickael Hubert wrote:
>>
>>> yep I found...
>>>
>>> if (end - start < 2 || end - start > 15)
>>> return -1;
>>>
>>> I have to modify this code.
>>> I will propose a PR.
>>>
>>> Thanks a lot
>>> ++
>>>
>>> Le mer. 6 sept. 2023 à 14:25, Marcin Groszek a
>>> écrit :
>>>
>>>> Correction : maximum of 15 digits .
>>>> On 9/6/2023 7:21 AM, Marcin Groszek wrote:
>>>>
>>>> Your number is to long
>>>>
>>>> E.164 is + [1-9] and {1-14} digits for total of 15 digits NOT
>>>> starting with 0
>>>> On 9/6/2023 7:16 AM, Mickael Hubert wrote:
>>>>
>>>> Hi all,
>>>> I have an issue, when I verify a call with no E164 format (dest:
>>>> +3310200123456789)
>>>>
>>>> *logs:*
>>>> Sep 6 13:39:48 am-scr-001 /usr/local/sbin/opensips[622409]:
>>>> ERROR:stir_shaken:check_passport_phonenum: number is not in E.164 format:
>>>> 3310200123456789
>>>> Sep 6 13:39:48 am-scr-001 /usr/local/sbin/opensips[622409]:
>>>> ERROR:stir_shaken:w_stir_verify: failed to validate Destination number
>>>> (3310200123456789)
>>>>
>>>> *My configuration:*
>>>> # - module stir_shaken ---
>>>> loadmodule "stir_shaken.so"
>>>> #--- stir_shaken params -
>>>> modparam("stir_shaken", "ca_list", "/usr/local/etc/opensips/man_ca.pem")
>>>> modparam("stir_shaken", "require_date_hdr", 0)
>>>> modparam("stir_shaken", "verify_date_freshness", 60)
>>>>
>>>> According to the doc e164_strict_mode is disabled by default, so I
>>>> don't know why it doesn't work.
>>>>
>>>> *source of code: *
>>>> if (_is_e164(num, e164_strict_mode) == -1) {
>>>> LM_GEN(log_lev, "number is not in E.164 format:
>>>> %.*s\n", num->len, num->s);
>>>> return -1;
>>>> }
>>>>
>>>>
>>>> Do you have any help for me please ? I have to validate this format of
>>>> dest number.
>>>>
>>>> Thanks in advance
>>>>
>>>>
>>>> ___
>>>> Users mailing
>>>> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>
>>>> --
>>>> Best Regards:
>>>> Marcin Groszek
>>>> Business Phone Servicehttps://www.voipplus.net
>>>>
>>>>
>>>> ___
>>>> Users mailing
>>>> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>
>>>> --
>>>> Best Regards:
>>>> Marcin Groszek
>>>> Business Phone Servicehttps://www.voipplus.net
>>>>
>>>> ___
>>>> Users mailing list
>>>> Users@lists.opensips.org
>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>
>>> ___
>>> Users mailing list
>>> Users@lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>> ___
>> Users mailing list
>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Stir ans Shaken - number is not in E.164 format
Is ST/SH being used other than the US? AFAIK it only applies to US numbers,
thus 10 digits, no?
On Wed, 6 Sep 2023 at 14:27, Mickael Hubert wrote:
> yep I found...
>
> if (end - start < 2 || end - start > 15)
> return -1;
>
> I have to modify this code.
> I will propose a PR.
>
> Thanks a lot
> ++
>
> Le mer. 6 sept. 2023 à 14:25, Marcin Groszek a
> écrit :
>
>> Correction : maximum of 15 digits .
>> On 9/6/2023 7:21 AM, Marcin Groszek wrote:
>>
>> Your number is to long
>>
>> E.164 is + [1-9] and {1-14} digits for total of 15 digits NOT starting
>> with 0
>> On 9/6/2023 7:16 AM, Mickael Hubert wrote:
>>
>> Hi all,
>> I have an issue, when I verify a call with no E164 format (dest:
>> +3310200123456789)
>>
>> *logs:*
>> Sep 6 13:39:48 am-scr-001 /usr/local/sbin/opensips[622409]:
>> ERROR:stir_shaken:check_passport_phonenum: number is not in E.164 format:
>> 3310200123456789
>> Sep 6 13:39:48 am-scr-001 /usr/local/sbin/opensips[622409]:
>> ERROR:stir_shaken:w_stir_verify: failed to validate Destination number
>> (3310200123456789)
>>
>> *My configuration:*
>> # - module stir_shaken ---
>> loadmodule "stir_shaken.so"
>> #--- stir_shaken params -
>> modparam("stir_shaken", "ca_list", "/usr/local/etc/opensips/man_ca.pem")
>> modparam("stir_shaken", "require_date_hdr", 0)
>> modparam("stir_shaken", "verify_date_freshness", 60)
>>
>> According to the doc e164_strict_mode is disabled by default, so I don't
>> know why it doesn't work.
>>
>> *source of code: *
>> if (_is_e164(num, e164_strict_mode) == -1) {
>> LM_GEN(log_lev, "number is not in E.164 format: %.*s\n",
>> num->len, num->s);
>> return -1;
>> }
>>
>>
>> Do you have any help for me please ? I have to validate this format of
>> dest number.
>>
>> Thanks in advance
>>
>>
>> ___
>> Users mailing
>> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>> --
>> Best Regards:
>> Marcin Groszek
>> Business Phone Servicehttps://www.voipplus.net
>>
>>
>> ___
>> Users mailing
>> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>> --
>> Best Regards:
>> Marcin Groszek
>> Business Phone Servicehttps://www.voipplus.net
>>
>> ___
>> 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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Re: [OpenSIPS-Users] Issue with stir and shaken crl_list
; -BEGIN X509 CRL-
> > >
> > > -END X509 CRL-
> > >
> > > I configured opensips with this:
> > > modparam("stir_shaken", "crl_list",
> > "/etc/opensips/stir-shaken-ca/crl.pem")
> > >
> > > but I have an error:
> > > ul 19 12:39:07 [12] INFO:stir_shaken:verify_callback:
> > certificate
> > > validation failed: unable to get certificate CRL
> > > Jul 19 12:39:07 [12] INFO:stir_shaken:w_stir_verify: Invalid
> > certificate
> > >
> > > Can you tell me, what is exactly the correct format please ?
> > >
> > > Thanks in advance !
> > > ++
> > >
> > > ___
> > > Users mailing list
> > > Users@lists.opensips.org <mailto:Users@lists.opensips.org> Users@lists.opensips.org <mailto:Users@lists.opensips.org>> Users@lists.opensips.org <mailto:Users@lists.opensips.org> Users@lists.opensips.org <mailto:Users@lists.opensips.org>>>
> > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users <
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users> <
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users> <
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users;>
> > <http://lists.opensips.org/cgi-bin/mailman/listinfo/users> <
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users;> <
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users;> <
> http://lists.opensips.org/cgi-bin/mailman/listinfo/usersgt;;>
> >
> > ___
> > Users mailing list
> > Users@lists.opensips.org <mailto:Users@lists.opensips.org> Users@lists.opensips.org <mailto:Users@lists.opensips.org>> Users@lists.opensips.org <mailto:Users@lists.opensips.org> Users@lists.opensips.org <mailto:Users@lists.opensips.org>>>
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users <
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users> <
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users> <
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users;>
> > <http://lists.opensips.org/cgi-bin/mailman/listinfo/users> <
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users;> <
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users;> <
> http://lists.opensips.org/cgi-bin/mailman/listinfo/usersgt;;>
> >
> >
> > ___
> > Users mailing list
> > Users@lists.opensips.org <mailto:Users@lists.opensips.org> Users@lists.opensips.org <mailto:Users@lists.opensips.org>>
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users <
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users> <
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users> <
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users;>
>
>
>
>
> ___
> Users mailing list
> Users@lists.opensips.org <mailto:Users@lists.opensips.org> Users@lists.opensips.org <mailto:Users@lists.opensips.org>>
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users <
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users> <
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users> <
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users;>
>
>
>
>
>
>
>
>
>
>
> ___
> Users mailing list
> Users@lists.opensips.org <mailto:Users@lists.opensips.org>
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users <
> 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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Re: [OpenSIPS-Users] stir shaken verification
> I am having some issues with stir_shaken setup. I am sure this not an > issue with the module, but me. > > stir_shaken_auth works just fine and I am able to sign the calls, however > I was unable to find any document how to use a ca file available for > download at iconectiv/download-list as well as via API. They do come in as > jwt file, but after little manipulation individual certificates can be > extracted, and the first one is the root certificate; I think, and the rest > are trusted STI-CA. I guess my question is how do I use this file or any > other cert file as "ca_list" and/or "ca_dir" . > > After weeks and hundreds attempts I was unsuccessful, and I was unable to > locate any document explaining preparation/setup/steps to setup > verification. > > All I get is : > > ERROR:stir_shaken:load_cert: Failed to parse certificate > ERROR:stir_shaken:w_stir_verify: Failed to load certificate > on INVITE with valid identity header. > > When I remove or replace "ca_list" file with something bogus opensips > does not even start with errors: > > ERROR:stir_shaken:init_cert_validation: Failed to load trustefd CAs > ERROR:core:init_mod: failed to initialize module stir_shaken > > I would really appreciate some guidance on this one. > > > > ___ > Users mailing > listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -- > Best Regards: > Marcin Groszek > Business Phone Servicehttps://www.voipplus.net > > > ___ > Users mailing > listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -- > Best Regards: > Marcin Groszek > Business Phone Servicehttps://www.voipplus.net > > ___ > 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 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] OpenSIPS Equivalent to Kamailio's cfg_get
Great; thanks for confirming this On Thu, 5 Jan 2023 at 10:53, Bogdan-Andrei Iancu wrote: > OK, so they are changeable at runtime (via MI I guess). In this case there > is 99% similarity with the shvals from cfgutils. > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > OpenSIPS Bootcamp 5-16 Dec 2022, online > https://www.opensips.org/training/OpenSIPS_eBootcamp_2022/ > > On 1/5/23 11:47 AM, David Villasmil wrote: > > No, that’s the thing. I want to be able to change them without restarting > opensips > > On Thu, 5 Jan 2023 at 09:55, Bogdan-Andrei Iancu > wrote: > >> Hi David, >> >> Are these cfg values static during runtime? if so, you can simple use the >> defines provided by the template'ing support -> >> https://www.opensips.org/Documentation/Templating-Config-Files-3-2 >> >> Regards, >> >> Bogdan-Andrei Iancu >> >> OpenSIPS Founder and Developer >> https://www.opensips-solutions.com >> OpenSIPS Bootcamp 5-16 Dec 2022, online >> https://www.opensips.org/training/OpenSIPS_eBootcamp_2022/ >> >> On 1/4/23 12:07 AM, David Villasmil wrote: >> >> Hey Bodgan! >> >> It’s used to dynamically set constants, I.e.: a variable you use >> throughout the script. >> >> You can set it initially like >> >> Myvar = 3 >> >> Then via cli you can change it. >> >> I finally did it in opensips with cfgutils, if there’s a better way to do >> it, please let me know. >> >> Many thanks! >> >> David >> >> On Tue, 3 Jan 2023 at 20:12, Bogdan-Andrei Iancu >> wrote: >> >>> Hi David, >>> >>> not sure what this cfg_get is suppose to do :( >>> >>> Regards, >>> >>> Bogdan-Andrei Iancu >>> >>> OpenSIPS Founder and Developer >>> https://www.opensips-solutions.com >>> OpenSIPS Bootcamp 5-16 Dec 2022, online >>> https://www.opensips.org/training/OpenSIPS_eBootcamp_2022/ >>> >>> On 12/27/22 1:35 PM, David Villasmil wrote: >>> >>> Hello folks, >>> >>> Is there such a thing? cfgutils seems to be an alternative, but looks >>> like too much for such a simple thing. >>> >>> Regards, >>> >>> David Villasmil >>> email: david.villasmil.w...@gmail.com >>> phone: +34669448337 >>> >>> ___ >>> Users mailing >>> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >>> -- >> Regards, >> >> David Villasmil >> email: david.villasmil.w...@gmail.com >> phone: +34669448337 >> >> >> -- > Regards, > > David Villasmil > email: david.villasmil.w...@gmail.com > phone: +34669448337 > > > -- Regards, David Villasmil email: david.villasmil.w...@gmail.com phone: +34669448337 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] OpenSIPS Equivalent to Kamailio's cfg_get
Hello folks, Is there such a thing? cfgutils seems to be an alternative, but looks like too much for such a simple thing. Regards, David Villasmil email: david.villasmil.w...@gmail.com phone: +34669448337 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] RTPProxy Docker Image
Can you explain more? I.e: params and such? Thanks! On Tue, 20 Dec 2022 at 22:29, Saint Michael wrote: > Opensips+ RTPProxy only works fine with plain LXC containers, > privileged, which basically have access to all the resources of the > box. > That is the model I use with great success. > > On Tue, Dec 20, 2022 at 2:47 PM Brett Nemeroff wrote: > > > > Hello Terrance, > > I wouldn't really recommend this. RTPProxy is going to use a lot of > ports in a very large range. That just doesn't work great in docker, but > even worse in K8S. > > > > I personally would put the RTPProxy outside of K8S. While you might be > able to get it to work, you are likely going against some basic design > concepts in containerization. I feel like the tech should propel the > solution and not be a hindrance to it. In this case, I'm not sure that K8S > is buying you anything of value, but instead creating architectural > challenges. > > > > I'd love to hear feedback or experiences from others. There's always > something to learn :) > > -Brett > > > > On Tue, Dec 20, 2022 at 11:43 AM Terrance Devor > wrote: > >> > >> Was it something I said? > >> > >> Terrance > >> > >> On Sun, Dec 18, 2022 at 12:50 PM Terrance Devor > wrote: > >>> > >>> Hello Everyone, > >>> > >>> Wow! Blast from the past... I am a long time member of this list, been > a while. > >>> > >>> Question, anyone successful in deploying RTPProxy to a dockerized > environment? Preferably to a Kubernetes managed environment. > >>> > >>> Please Help Team :) > >>> > >>> Kind Regards, > >>> Terrance > >> > >> ___ > >> Users mailing list > >> Users@lists.opensips.org > >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > ___ > > Users mailing list > > Users@lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > ___ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- Regards, David Villasmil email: david.villasmil.w...@gmail.com phone: +34669448337 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] kamilio's htable equivalent?
Thanks Daniel, Amazingly enough I had! But then completely forgot about it. Thanks!! On Wed, 21 Dec 2022 at 13:26, Daniel Zanutti wrote: > 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 >> > ___ >> 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 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] kamilio's htable equivalent?
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 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Load_Balancing
Agreed, just go with dispatcher (I usually do random, which distributes the calls pretty well) On Fri, 9 Sep 2022 at 12:00, Bogdan-Andrei Iancu wrote: > Vadim, > > The 2 INVITE requests are not part of the same dialog, so you cannot use > dlg_val's - each initial INVITE is creating a different dialog. > > Now, if you really want, you can rely on the fact that the 2 INVITEs > have the same call-id and and use the cachedb_local to remember which > Ast box handled the first INVITE. But this somehow will invalidate the > whole idea of balancing calls. > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer >https://www.opensips-solutions.com > OpenSIPS Summit 27-30 Sept 2022, Athens >https://www.opensips.org/events/Summit-2022Athens/ > > On 9/9/22 10:44 AM, Vadim Dumalekov wrote: > > Thanks! > > > > I have one more question. Why can't the dlg_val be set in this case. > > This variable (dlg_val) is not passed to the second INVITE, although it > is a SIP Dialog (INVITE -> 401 -> ACK -> INVITE ...) > > > > > >> 9 сент. 2022 г., в 9:14, Bogdan-Andrei Iancu > написал(а): > >> > >> Hi, > >> > >> Considering the fact that Ast_2 cannot perform auth on a challenge done > by Ast_1, you should re-consider the routing logic in OpenSIPS, and not to > use LB, but rather dispatcher with hashing over call-id for example. > >> > >> Regards, > >> > >> Bogdan-Andrei Iancu > >> > >> OpenSIPS Founder and Developer > >> https://www.opensips-solutions.com > >> OpenSIPS Summit 27-30 Sept 2022, Athens > >> https://www.opensips.org/events/Summit-2022Athens/ > >> > >> On 9/8/22 1:43 PM, Vadim Dumalekov via Users wrote: > >>> Thank you for the answer! > >>> > >>> Yes, of cource. But there is this situation: > >>> > >>> UAC (INVITE w/o auth) -> OpenSIPS (LB: Ast_1) -> Ast_1 (401 Unauth) > >>> UAC (INVITE with auth) -> OpenSIPS (LB: Ast_2) -> Ast_2 (401 > Unauth) > >>> > >>> ... etc, until LB selects the same Asterisk for two INVITE`s (w/o auth > and with auth). > >>> > >>> > >>> Vadim > >>> > >>>> 8 сент. 2022 г., в 12:16, Bogdan-Andrei Iancu > написал(а): > >>>> > >>>> Hi Vadim, > >>>> > >>>> If you have a cluster of ASterisk servers, each box from the cluster > should be able to handle the auth response, even if the challenge was done > by a different one. Otherwise it is not a cluster, but a bunch of servers. > >>>> > >>>> Regards, > >>>> > >>>> Bogdan-Andrei Iancu > >>>> > >>>> OpenSIPS Founder and Developer > >>>> https://www.opensips-solutions.com > >>>> OpenSIPS Summit 27-30 Sept 2022, Athens > >>>> https://www.opensips.org/events/Summit-2022Athens/ > >>>> > >>>> On 9/7/22 3:52 PM, Vadim Dumalekov via Users wrote: > >>>>> Hello! > >>>>> > >>>>> Please help me. I'm using the Load_Balancer module for the incoming > calls to an Asterisk cluster. When an INVITE is sent to one of the > Asterisks, we receive the "401 Unauthorized" message from that server. But > when UAC sends the INVITE with the authorization, the LB-module sends it to > another Asterisk. Sometimes it happens multiple times. > >>>>> > >>>>> > >>>>> Thanks in advance! > >>>>> > >>>>> Vadim. > >>>>> > >>>>> > >>>>> ___ > >>>>> Users mailing list > >>>>> Users@lists.opensips.org > >>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > >>>> ___ > >>>> Users mailing list > >>>> Users@lists.opensips.org > >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > >>> ___ > >>> Users mailing list > >>> Users@lists.opensips.org > >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > ___ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- Regards, David Villasmil email: david.villasmil.w...@gmail.com phone: +34669448337 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] RTP proxy RE-INVITE with late SDP (ACK cannot have SDP body)
Hello Callum, Can you share how you are doing late negotiation? Regards, David Villasmil email: david.villasmil.w...@gmail.com phone: +34669448337 On Wed, May 15, 2019 at 6:37 PM Callum Guy wrote: > Hi All, > > I am working on a problem where for a few destinations my OpenSIPs is > receiving RE-INVITE messages with late SDP. This is causing a breakdown in > the rtpproxy engagement and causing the audio to fail mid call. > > The OpenSIPs deployment is acting as a SIP proxy which traverses NAT and > rtpproxy is used in bridging mode. I am using rtpproxy_engage to tie the > integration to the dialog session and this is for all other purposes > working as expected. > > My failure scenario is when the remote system sends a RE-INVITE message > which includes no SDP. This passes through to my FreeSWITCH server which > responds with a 200 including SDP. This message is processed fine and > interacts with rtpproxy as expected and provides the remote with the > correct public IP and port for RTP (the same as returned during call > setup). In response the remote system returns an ACK with SDP which > triggers an OpenSIPs error message (below) which results in the remotes > public IP being passed through in SDP which causes the FreeSWITCH to start > sending RTP direct resulting in one way audio as the media server is not > publicly accessible. > > *ERROR:rtpproxy:engage_force_rtpproxy: not a late negotiation - ACK cannot > have SDP body* > > As I understand it the FreeSWITCH behaviour is OK, although I am not clear > why it feels the need to resend the SDP. All I want to happen in this > scenario is for rtpproxy module to re-write the SDP in the way it has for > all previous messages. I am very interested to hear if there is any reason > for rtpproxy to disallow late negotiation in this scenario, if anyone can > point to a relevant RFC that would be interesting! > > Is there any way around this other than some sort of manual SDP re-write > (not helpful to me as I am using a pool of rtpproxy instances)? Might I > have more luck with offer/answer or indeed rtpengine? > > I've illustrated the scenario better on the following link (sngrep paste): > > > https://gist.githubusercontent.com/spacetourist/ef0478c0bf4e2d736f9b5663042087dd/raw/6f0a984a1a2838e7e2c4539f059fd68935a3b0b1/gistfile1.txt > > Thanks, looking forward to any advice! > > Best regards, > > Callum > > > > > *0333 332 | www.x-on.co.uk <http://www.x-on.co.uk> | ** > <https://www.linkedin.com/company/x-on> <https://www.facebook.com/XonTel> > <https://twitter.com/xonuk> * > > X-on is a trading name of Storacall Technology Ltd a limited company > registered in England and Wales. > Registered Office : Avaland House, 110 London Road, Apsley, Hemel > Hempstead, Herts, HP3 9SD. Company Registration No. 2578478. > The information in this e-mail is confidential and for use by the > addressee(s) only. If you are not the intended recipient, please notify > X-on immediately on +44(0)333 332 and delete the > message from your computer. If you are not a named addressee you must not > use, disclose, disseminate, distribute, copy, print or reply to this email. > Views > or opinions expressed by an individual > within this email may not necessarily reflect the views of X-on or its > associated companies. Although X-on routinely screens for viruses, > addressees should scan this email and any attachments > for viruses. X-on makes no representation or warranty as to the absence of > viruses in this email or any attachments. > > ___ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] dynamic routing failover ONLY ONCE on the provider and continue
Hey Bodgan, Sorry for the caps, was just trying to illustrate a very important point. That was a typo: it's provider. So what i mean is: - Provier1 - gw1 - gw2 - Provider2 - gw1 - gw2 and so on. The providers could have more than 2 gws, but i only want it to attempt the first 2. Is this possible? Regards, On Tue, 6 Sep 2022 at 16:41, Ben Newlin wrote: > If I’m not mistaken, the functionality David is describing is the default > behavior of the module and the use_next_gw function. All carriers are > loaded on the call to do_routing, and use_next_gw will go through each > gateway of each carrier in order, unless the flag Bogdan referenced below > is set. > > > > Ben Newlin > > > > *From: *Users on behalf of > Bogdan-Andrei Iancu > *Date: *Tuesday, September 6, 2022 at 10:06 AM > *To: *OpenSIPS users mailling list , David > Villasmil > *Subject: *Re: [OpenSIPS-Users] dynamic routing failover ONLY ONCE on the > provider and continue > > * EXTERNAL EMAIL - Please use caution with links and attachments * > > > -- > > David, > > Define the "provide" as carrier and set the "use only first gw from cr" > flag for it, see > https://www.opensips.org/Documentation/Install-DBSchema-3-2#GEN-DB-DR-CARRIERS > > PS: no need for caps ;) > > Regards, > > Bogdan-Andrei Iancu > > > > OpenSIPS Founder and Developer > > https://www.opensips-solutions.com > > OpenSIPS Summit 27-30 Sept 2022, Athens > > https://www.opensips.org/events/Summit-2022Athens/ > > On 9/6/22 4:57 PM, David Villasmil wrote: > > Hello folks, > > > > I'm trying to route to the first provider and if the first gw attempted > fails, try the next gw on that provider, and if that fails THEN failover to > the next provider. NOTE ALL PROVIDERS CAN HAVE MULTIPLE gws. > > > > Is this possible on 2.4.7? > > > > > > I really appreciate your help! > > > > > > David Villasmil > > email: david.villasmil.w...@gmail.com > > phone: +34669448337 > > > > ___ > > 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 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] dynamic routing failover ONLY ONCE on the provider and continue
Is there anything like “use_next_carrier”? I.e.: decide when I want to stop trying gws for the current carrier. On Tue, 6 Sep 2022 at 18:04, David Villasmil wrote: > I may not have been clear, I want to try the first _two_ (2) gws for each > carrier. > > Is this possible? > > On Tue, 6 Sep 2022 at 17:14, David Villasmil < > david.villasmil.w...@gmail.com> wrote: > >> Hey Bodgan, >> >> Sorry for the caps, was just trying to illustrate a very important point. >> >> That was a typo: it's provider. >> >> So what i mean is: >> >> - Provier1 >> - gw1 >> - gw2 >> - Provider2 >> - gw1 >> - gw2 >> >> and so on. >> >> The providers could have more than 2 gws, but i only want it to attempt >> the first 2. >> >> Is this possible? >> >> Regards, >> >> David Villasmil >> email: david.villasmil.w...@gmail.com >> phone: +34669448337 >> >> >> On Tue, Sep 6, 2022 at 4:05 PM Bogdan-Andrei Iancu >> wrote: >> >>> David, >>> >>> Define the "provide" as carrier and set the "use only first gw from cr" >>> flag for it, see >>> https://www.opensips.org/Documentation/Install-DBSchema-3-2#GEN-DB-DR-CARRIERS >>> >>> PS: no need for caps ;) >>> >>> Regards, >>> >>> Bogdan-Andrei Iancu >>> >>> OpenSIPS Founder and Developer >>> https://www.opensips-solutions.com >>> OpenSIPS Summit 27-30 Sept 2022, Athens >>> https://www.opensips.org/events/Summit-2022Athens/ >>> >>> On 9/6/22 4:57 PM, David Villasmil wrote: >>> >>> Hello folks, >>> >>> I'm trying to route to the first provider and if the first gw attempted >>> fails, try the next gw on that provider, and if that fails THEN failover to >>> the next provider. NOTE ALL PROVIDERS CAN HAVE MULTIPLE gws. >>> >>> Is this possible on 2.4.7? >>> >>> >>> I really appreciate your help! >>> >>> >>> David Villasmil >>> email: david.villasmil.w...@gmail.com >>> phone: +34669448337 >>> >>> ___ >>> Users mailing >>> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >>> -- > Regards, > > David Villasmil > email: david.villasmil.w...@gmail.com > phone: +34669448337 > -- Regards, David Villasmil email: david.villasmil.w...@gmail.com phone: +34669448337 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] dynamic routing failover ONLY ONCE on the provider and continue
Hello folks, I'm trying to route to the first provider and if the first gw attempted fails, try the next gw on that provider, and if that fails THEN failover to the next provider. NOTE ALL PROVIDERS CAN HAVE MULTIPLE gws. Is this possible on 2.4.7? I really appreciate your help! David Villasmil email: david.villasmil.w...@gmail.com phone: +34669448337 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Dispatcher within a K8s environment
> Thanks for the response!
>
>
>
> Jon
>
>
>
>
>
> Sent from Mail
> <https://nam12.safelinks.protection.outlook.com/?url=https%3A%2F%2Fgo.microsoft.com%2Ffwlink%2F%3FLinkId%3D550986=05%7C01%7C%7Cdb99d78253b542300c5308da8a5c90a7%7C84df9e7fe9f640afb435%7C1%7C0%7C637974427420952414%7CUnknown%7CTWFpbGZsb3d8eyJWIjoiMC4wLjAwMDAiLCJQIjoiV2luMzIiLCJBTiI6Ik1haWwiLCJXVCI6Mn0%3D%7C3000%7C%7C%7C=z9ZEj1kZZJOf%2BiBAEn8aC2H4cG%2B5Qc90poMgOQINcxA%3D=0>
> for Windows
>
>
>
> *From: *Bogdan-Andrei Iancu
> *Sent: *24 August 2022 12:29
> *To: *OpenSIPS users mailling list ; Jonathan
> Hunter
> *Subject: *Re: [OpenSIPS-Users] Dispatcher within a K8s environment
>
>
>
> Hi Jonathan,
>
> I guess this will be a good topic (DS and K8S) for the OpenSIPS Summit in
> Athens - I think this is the 3rd time in the last week coming across it :)
>
> Unfortunately there is no way to skip at the moment that DNS failure when
> loading the destinations :(even more, there some code that relies on
> the fact that there is an "IP" attached to any destination.And I just
> checked, a local error in sending the ping (like the DNS err) does not
> results in marking the destination as failed or so. so it is not so
> straight as ignoring the DNS error.
>
> Best regards,
>
>
> Bogdan-Andrei Iancu
>
>
>
> OpenSIPS Founder and Developer
>
> https://www.opensips-solutions.com
> <https://nam12.safelinks.protection.outlook.com/?url=https%3A%2F%2Fwww.opensips-solutions.com%2F=05%7C01%7C%7Cdb99d78253b542300c5308da8a5c90a7%7C84df9e7fe9f640afb435%7C1%7C0%7C637974427420952414%7CUnknown%7CTWFpbGZsb3d8eyJWIjoiMC4wLjAwMDAiLCJQIjoiV2luMzIiLCJBTiI6Ik1haWwiLCJXVCI6Mn0%3D%7C3000%7C%7C%7C=%2FMOAoJ8bvkJYB%2BO%2B6WjL5vq7zhzZdMxOhNJNl3NeT%2Fw%3D=0>
>
> OpenSIPS Summit 27-30 Sept 2022, Athens
>
> https://www.opensips.org/events/Summit-2022Athens/
> <https://nam12.safelinks.protection.outlook.com/?url=https%3A%2F%2Fwww.opensips.org%2Fevents%2FSummit-2022Athens%2F=05%7C01%7C%7Cdb99d78253b542300c5308da8a5c90a7%7C84df9e7fe9f640afb435%7C1%7C0%7C637974427420952414%7CUnknown%7CTWFpbGZsb3d8eyJWIjoiMC4wLjAwMDAiLCJQIjoiV2luMzIiLCJBTiI6Ik1haWwiLCJXVCI6Mn0%3D%7C3000%7C%7C%7C=7P9%2BRI%2BIxCfmddnUntpf5JooWm%2BXUtAFjb2xkZJZGBU%3D=0>
>
> On 8/24/22 12:24 AM, Jonathan Hunter wrote:
>
> Hi All,
>
>
>
> I have a query around dispatcher behaviour, I am running 3.2 in a k8s
> environment.
>
>
>
> I have 2 freeswitch instances defined in a destination set, both of which
> are pods.
>
>
>
> As people may be aware its fun implementing in k8s as pods can restart and
> disappear at times so I ideally want this reflected in the cache and output
> of opensips-cli -x mi ds_list where I was hoping the freeswitch entries
> would be defined but with a state of probing or inactive.
>
>
>
> With my current setup, when restarting opensips for example, I have the
> dispatcher table populated in postgres db , and if opensips cant resolve
> the URI it wont load it into cache, like wise if opensips is running and
> freeswitch pod drops, I see this in the logs;
>
>
>
> Aug 23 21:22:01 [55] ERROR:dispatcher:add_dest2list: could not resolve
> freeswitch-opensips-deployment-1.freeswitch-opensips, skipping it
>
> Aug 23 21:22:01 [55] WARNING:dispatcher:ds_load_data: failed to add
> destination
>
> in group 10
>
>
>
> I therefore don’t see it listed in cache when I run ds_list.
>
>
>
> Does anyone know if its possible to tweak dispatcher to always load the
> database entries into cache at startup, and also set their status to
> probing/inactive if not reachable due to a resolving issue as above?
>
>
>
> My dispatcher settings are;
>
>
>
> Dynamic routing
>
> loadmodule "dispatcher.so"
>
> modparam("dispatcher", "db_url", "postgres://x.x.x.x/opensips")
>
> modparam("dispatcher", "ds_probing_mode", 1)
>
> modparam("dispatcher", "ds_probing_threshhold", 1)
>
> modparam("dispatcher", "persistent_state", 0)
>
> modparam("dispatcher", "ds_ping_interval", 5)
>
> modparam("dispatcher", "table_name", "dispatcher")
>
> modparam("dispatcher", "cluster_id", 1)
>
>
>
> Hope that makes sense!
>
>
>
> Many thanks
>
>
>
> Jon
>
>
>
>
>
>
> ___
>
> Users mailing list
>
> Users@lists.opensips.org
>
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
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>
>
>
>
>
>
>
>
>
>
>
>
> ___
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> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
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Regards,
David Villasmil
email: david.villasmil.w...@gmail.com
phone: +34669448337
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[OpenSIPS-Users] $param?
Hello folks, It's been a while since i've worked with OpenSIPS, and I'm seeing a script using $var(addPrefix) = $param(1); I don't really know what that $param does, can anyone help me out here? Regards, David Villasmil email: david.villasmil.w...@gmail.com phone: +34669448337 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] CDR not generated on 302 redirect
Look very carefully at the config. There’s probably somewhere it’s enabled. Maybe this https://github.com/OpenSIPS/opensips/blob/master/examples/acc.cfg might give you some ideas… On Tue, 15 Mar 2022 at 14:14, Saint Michael wrote: > My new business is to provide 302 Redirect services and Opensips does not > genrate a CDR for those calls. Other type of calls do generate a record. Is > this by design or is it a bug? > Every call that goes through Opensips should generate a record. > Any idea about what is going on? > ___ > 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 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Forwarding ACK to 200 OK to itself.
Hey how are your Bogdan! Yes that was the issue, solved it already. Thanks!!! I appreciate your help! On Mon, 6 Dec 2021 at 17:07, Bogdan-Andrei Iancu wrote: > Hi David, > > IMHO, you do record_route() in request route (before sending to SS) and > once again later, in failure route, after SS, before the Termination. Could > you check this? > > Normally doing it in request route is more then enough, covering the whole > call processing; there is no need to do it per branch or so. > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > OpenSIPS eBootcamp 2021 > https://opensips.org/training/OpenSIPS_eBootcamp_2021/ > > On 11/26/21 3:48 AM, David Villasmil wrote: > > Hello folks, > > I'm facing this issue on this scenario: > > [ 1] USER---INVITE--->OPENSIPS > [ 2] OPENSIPS---INVITE-->STIR/SHAKEN (OPENSIPS) > [ 3] OPENSIPS<--- 302 ---STIR/SHAKEN > [ 4] OPENSIPS<--- ACK ---STIR/SHAKEN > [ 5] OPENSIPS---INVITE->TERMINATION > [ 6] OPENSIPS<-- 180 --TERMINATION > [ 7] OPENSIPS<-- 183 --TERMINATION > [ 8] OPENSIPS<-- 200 --TERMINATION > [ 9] USER<-- 200 -OPENSIPS > [10] USER--- ACK >OPENSIPS > [11] OPENSIPS--- ACK > [12] OPENSIPS<-- ACK > [13] OPENSIPS<-- 200 --TERMINATION > [14] USER<-- 200 -OPENSIPS > [15] USER--- ACK >OPENSIPS > [16] OPENSIPS--- ACK > [17] OPENSIPS<-- ACK > > > This only happens when I enable that STIR/SHAKEN functionality and use the > 302, when OpenSIPS sends out the INVITE [ > > The 200 OK has this as a record route: > Record-Route:, > , > , > , > > > what do i need to do to remove those duplicated record-routes after > receiving the 302? > > Thanks alot for your help! > > Regards, > > David Villasmil > email: david.villasmil.w...@gmail.com > phone: +34669448337 > > ___ > Users mailing > listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > -- Regards, David Villasmil email: david.villasmil.w...@gmail.com phone: +34669448337 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Parse P-Asserted-Identity
https://www.opensips.org/Documentation/Script-Tran-2-4#toc60
Regards,
David Villasmil
email: david.villasmil.w...@gmail.com
phone: +34669448337
On Mon, Nov 29, 2021 at 12:22 PM Mickael MONSIEUR <
mickael.monsi...@gmail.com> wrote:
> Hello,
>
> My provider add to my INVITE's :
>
> P-Asserted-Identity: "Anonymous"
> ;party=calling;privacy=yes;screen=no
>
> Whether the call should be Anonymized to end-users.
>
> How to get the value of "privacy" ?
>
> I try:
>
> if( $(ai{privacy}) == "yes" )
>
> But it does not work. (error when starting opensips)
>
> Thanks
>
> ___
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> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
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Re: [OpenSIPS-Users] Forwarding ACK to 200 OK to itself.
Hello folks, So, the way the script is, it's adding a record_route() before doing the STSH stuff, and that's adding that route for the rest of the flow... I wonder if there's any way of removing it. Regards, David Villasmil email: david.villasmil.w...@gmail.com phone: +34669448337 On Fri, Nov 26, 2021 at 1:48 AM David Villasmil < david.villasmil.w...@gmail.com> wrote: > Hello folks, > > I'm facing this issue on this scenario: > > [ 1] USER---INVITE--->OPENSIPS > [ 2] OPENSIPS---INVITE-->STIR/SHAKEN (OPENSIPS) > [ 3] OPENSIPS<--- 302 ---STIR/SHAKEN > [ 4] OPENSIPS<--- ACK ---STIR/SHAKEN > [ 5] OPENSIPS---INVITE->TERMINATION > [ 6] OPENSIPS<-- 180 --TERMINATION > [ 7] OPENSIPS<-- 183 --TERMINATION > [ 8] OPENSIPS<-- 200 --TERMINATION > [ 9] USER<-- 200 -OPENSIPS > [10] USER--- ACK >OPENSIPS > [11] OPENSIPS--- ACK > [12] OPENSIPS<-- ACK > [13] OPENSIPS<-- 200 --TERMINATION > [14] USER<-- 200 -OPENSIPS > [15] USER--- ACK >OPENSIPS > [16] OPENSIPS--- ACK > [17] OPENSIPS<-- ACK > > > This only happens when I enable that STIR/SHAKEN functionality and use the > 302, when OpenSIPS sends out the INVITE [ > > The 200 OK has this as a record route: > > Record-Route: > > what do i need to do to remove those duplicated record-routes after > receiving the 302? > > Thanks alot for your help! > > Regards, > > David Villasmil > email: david.villasmil.w...@gmail.com > phone: +34669448337 > ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Forwarding ACK to 200 OK to itself.
Hello folks, I'm facing this issue on this scenario: [ 1] USER---INVITE--->OPENSIPS [ 2] OPENSIPS---INVITE-->STIR/SHAKEN (OPENSIPS) [ 3] OPENSIPS<--- 302 ---STIR/SHAKEN [ 4] OPENSIPS<--- ACK ---STIR/SHAKEN [ 5] OPENSIPS---INVITE->TERMINATION [ 6] OPENSIPS<-- 180 --TERMINATION [ 7] OPENSIPS<-- 183 --TERMINATION [ 8] OPENSIPS<-- 200 --TERMINATION [ 9] USER<-- 200 -OPENSIPS [10] USER--- ACK >OPENSIPS [11] OPENSIPS--- ACK [12] OPENSIPS<-- ACK [13] OPENSIPS<-- 200 --TERMINATION [14] USER<-- 200 -OPENSIPS [15] USER--- ACK >OPENSIPS [16] OPENSIPS--- ACK [17] OPENSIPS<-- ACK This only happens when I enable that STIR/SHAKEN functionality and use the 302, when OpenSIPS sends out the INVITE [ The 200 OK has this as a record route: Record-Route: what do i need to do to remove those duplicated record-routes after receiving the 302? Thanks alot for your help! Regards, David Villasmil email: david.villasmil.w...@gmail.com phone: +34669448337 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] STIR/SHAKEN E.164 strict mode module parameter not working .
Hello,
What does the orig/dest look like in the payload?
On Wed, 17 Nov 2021 at 15:28, Devang Dhandhalya <
devang.dhandha...@ecosmob.com> wrote:
> hello all
>
>
> Above E.164 Error still getting .Right now I'm getting the below error .
> Can anyone tell me why I am getting this error ? as far as i know this
> error for x5u parameter in stir_shaken_auth function , this issue coming
> for certificate path or certificate file format .
>
>
> I check the certificate file with .der and .cer format also .
>
> Here is the code snippet used .
>
> $var(rc_auth)=stir_shaken_auth("A", "GWID-123456","$var(cert)",
> "$var(pkey)","http://localhost/certificate.pem","$var(orig)","$var(dest)");
>
>
> Below Error i am getting .
>
>
> ERROR:stir_shaken:add_identity_hf: Failed to convert from DER to internal
> format
>
> ERROR:stir_shaken:w_stir_auth: Failed to add Identity header
>
> STIR_SHAKEN AUTHENTICATION SERVICE return code : -1
>
>
> Kindly let me know if there is something wrong that I could be doing.
>
> Many Thanks
>
>
> Devang Dhandhalya
>
>
> On Wed, Nov 17, 2021 at 11:37 AM Devang Dhandhalya <
> devang.dhandha...@ecosmob.com> wrote:
>
>> Hi All
>>
>> I configured the e164 strict mode module parameter as 0 (disabled) . but
>> still i am getting errors related to its e164 format .While if orig/dest
>> number is not in e164 format then also opensips have to accept it but it is
>> not accepting . I have a user like extension123 for this function I have
>> to perform authentication service . if i have a user extension123 is it
>> possible to perform authenticate service for this kind of user ?
>>
>> I think this is a bug for the e164 strict mode module parameter . I am
>> getting the below error .
>>
>> opensips version : 3.2.2
>>
>> ERROR :
>> ERROR:stir_shaken:check_passport_phonenum: number is not in E.164
>> format: extension123
>> ERROR:stir_shaken:w_stir_auth: failed to validate Originator number
>> (extension123)
>>
>>
>> loadmodule "stir_shaken.so"
>> modparam("stir_shaken", "auth_date_freshness", 300)
>> modparam("stir_shaken", "verify_date_freshness", 300)
>> modparam("stir_shaken", "require_date_hdr", 0)
>> modparam("stir_shaken", "e164_strict_mode", 0)
>>
>> $var(orig) = $fU;
>> $var(dest) = $tU
>> $var(rc_auth)=stir_shaken_auth("A", "GWID-123456","$var(cert)",
>> "$var(pkey)","http://localhost/certificate.pem
>> ","$var(orig)","$var(dest)");
>>
>> Please suggest a solution to this .
>>
>> Many Thanks
>> Devang
>>
>
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email: david.villasmil.w...@gmail.com
phone: +34669448337
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Re: [OpenSIPS-Users] How to get all listening socket IP address as variable
You can pass it to opensips as a variable on startup. On Wed, 17 Nov 2021 at 00:19, Muhamad Putra Abdullah wrote: > Hi, > > Is there a way to get the IP address of DHCP interface to use in the > script? I can get the call go through if I set both the interface as static. > > Regards > > Get Outlook for Android <https://aka.ms/AAb9ysg> > > -- > *From:* Bogdan-Andrei Iancu > *Sent:* Tuesday, November 16, 2021, 6:45 PM > *To:* OpenSIPS users mailling list; Muhamad Putra Abdullah > *Subject:* Re: [OpenSIPS-Users] How to get all listening socket IP > address as variable > > Hi, > > via the socket_xx() vars you can get only the sockets relative to/used by > that call, you cannot iterate thru the listening sockets. > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > OpenSIPS eBootcamp 2021 > https://opensips.org/training/OpenSIPS_eBootcamp_2021/ > > On 11/15/21 9:39 AM, Muhamad Putra Abdullah wrote: > > Hi, > > > > I have 2 listening interface for opensips 3.2. How do I get both IP > address to be used as variable in opensips config file? I try to use > socket_in/ socket_out but failed to get the other interface IP address. > > > > Thanks > > > > Sent from Mail <https://go.microsoft.com/fwlink/?LinkId=550986> for > Windows > > > > ___ > Users mailing > listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > ___ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- Regards, David Villasmil email: david.villasmil.w...@gmail.com phone: +34669448337 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] drouting gw list
Thanks again Ben, I'm still not understanding: Setting it to (as is, nothing) would then only send to the first gw, and only to the second on failure? Regards, David Villasmil email: david.villasmil.w...@gmail.com phone: +34669448337 On Wed, Nov 10, 2021 at 2:06 PM Ben Newlin wrote: > Yes, in 2.4 it is controlled by the flags. If you set it to 2 it will only > ever use the first gateway and there will be no failover. If you want to > allow for failover and use the gateways in the order they are defined you > should not set any flags. > > > > https://opensips.org/docs/modules/2.4.x/drouting.html#idp163040 > > > > Ben Newlin > > > > *From: *Users on behalf of David > Villasmil > *Date: *Wednesday, November 10, 2021 at 7:50 AM > *To: *OpenSIPS users mailling list > *Subject: *Re: [OpenSIPS-Users] drouting gw list > > Sorry, > > > > it's defined in cr_carriers, that's the list of gws which are then defined > on dr_gateways. > > The "flags" field is "1", which i take to mean weight-based (this is > opensips 2.4) > > So i would need to set the flags to "2"? > > Regards, > > > > David Villasmil > > email: david.villasmil.w...@gmail.com > > phone: +34669448337 > > > > > > On Wed, Nov 10, 2021 at 2:56 AM Ben Newlin wrote: > > David, > > > > Is this gw_list defined on a dr_rule or a dr_carrier? What is the value of > the sort_alg column for that dr_rule/dr_carrier? If the sort_alg is set to > W, then the assigned values are interpreted as weights. So with this > configuration it would send 100 out of 110 calls to PrimaryGW and 10 out of > 110 calls to SecondaryGW. > > > > If you don’t want to use the SecondaryGW except for failover you should > not use weighted routing, so you should set the sort_alg column to N. > > > > https://www.opensips.org/Documentation/Install-DBSchema-3-2#GEN-DB-DR-RULES > > > > Ben Newlin > > > > *From: *Users on behalf of David > Villasmil > *Date: *Tuesday, November 9, 2021 at 7:37 PM > *To: *users@lists.opensips.org > *Subject: *[OpenSIPS-Users] drouting gw list > > Hello folks, > > > > i have the following gw_list > > > > PrimaryGW=100,SecondaryGW=10 > > > > I'm seeing calls going to the SecondaryGW. > > How should i set this if i only want the SecondaryGW to be used for > failover? > > > Regards, > > > > David Villasmil > > email: david.villasmil.w...@gmail.com > > phone: +34669448337 > > ___ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > ___ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] drouting gw list
Sorry, it's defined in cr_carriers, that's the list of gws which are then defined on dr_gateways. The "flags" field is "1", which i take to mean weight-based (this is opensips 2.4) So i would need to set the flags to "2"? Regards, David Villasmil email: david.villasmil.w...@gmail.com phone: +34669448337 On Wed, Nov 10, 2021 at 2:56 AM Ben Newlin wrote: > David, > > > > Is this gw_list defined on a dr_rule or a dr_carrier? What is the value of > the sort_alg column for that dr_rule/dr_carrier? If the sort_alg is set to > W, then the assigned values are interpreted as weights. So with this > configuration it would send 100 out of 110 calls to PrimaryGW and 10 out of > 110 calls to SecondaryGW. > > > > If you don’t want to use the SecondaryGW except for failover you should > not use weighted routing, so you should set the sort_alg column to N. > > > > https://www.opensips.org/Documentation/Install-DBSchema-3-2#GEN-DB-DR-RULES > > > > Ben Newlin > > > > *From: *Users on behalf of David > Villasmil > *Date: *Tuesday, November 9, 2021 at 7:37 PM > *To: *users@lists.opensips.org > *Subject: *[OpenSIPS-Users] drouting gw list > > Hello folks, > > > > i have the following gw_list > > > > PrimaryGW=100,SecondaryGW=10 > > > > I'm seeing calls going to the SecondaryGW. > > How should i set this if i only want the SecondaryGW to be used for > failover? > > > Regards, > > > > David Villasmil > > email: david.villasmil.w...@gmail.com > > phone: +34669448337 > ___ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] drouting gw list
Hello folks, i have the following gw_list PrimaryGW=100,SecondaryGW=10 I'm seeing calls going to the SecondaryGW. How should i set this if i only want the SecondaryGW to be used for failover? Regards, David Villasmil email: david.villasmil.w...@gmail.com phone: +34669448337 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Remove previously added header
Ok i stumbled on to your email from back 2009 asking just about the same
thing (
https://users.opensips.narkive.com/W78SWsO9/opensips-users-remove-hf-doesn-t-always-work
)
I ended up doing
t_on_branch("1")
# Add the header before sending it out the parameter server
branch_route[1] {
append_hf("X-MyHeader: 1\r\n");
}
then when actually sending it out to the termination provider:
t_on_branch("2");
# Remove the header before seding it out to the termination
branch_route[2] {
remove_hf("X-MyHeader");
}
That's working perfectly!
Is this right or is there a more efficient/easier way to do it?
Thanks!
Regards,
David Villasmil
email: david.villasmil.w...@gmail.com
phone: +34669448337
On Tue, Jun 15, 2021 at 12:10 AM David Villasmil <
david.villasmil.w...@gmail.com> wrote:
> Thanks Jeff,
>
> That's not going to work for me, I don't want to add new branches.
> What I'm doing is forwarding the call to a sip server that replies with a
> 302 and some extra headers.
> I then need to get these headers and continue routing normally, one single
> branch.
>
> Regards,
>
> David Villasmil
> email: david.villasmil.w...@gmail.com
> phone: +34669448337
>
>
> On Mon, Jun 14, 2021 at 11:12 PM Jeff Pyle wrote:
>
>> Add it in a branch_route. That way if you have to route advance, it'll
>> already be gone because you'll be on a new branch.
>>
>>
>> - Jeff
>>
>>
>> On Mon, Jun 14, 2021 at 5:52 PM David Villasmil <
>> david.villasmil.w...@gmail.com> wrote:
>>
>>> Hello guys,
>>>
>>> So, I'm appending a header (append_hf("header")) to a forward.
>>> That forward fails and I'm trying to remove it with remove_hf("header"),
>>> but it's not getting removed for some reason, what am I doing wrong?
>>>
>>> Thanks everyone!
>>>
>>> David Villasmil
>>> email: david.villasmil.w...@gmail.com
>>> phone: +34669448337
>>> ___
>>> 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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Re: [OpenSIPS-Users] Remove previously added header
Thanks Jeff,
That's not going to work for me, I don't want to add new branches.
What I'm doing is forwarding the call to a sip server that replies with a
302 and some extra headers.
I then need to get these headers and continue routing normally, one single
branch.
Regards,
David Villasmil
email: david.villasmil.w...@gmail.com
phone: +34669448337
On Mon, Jun 14, 2021 at 11:12 PM Jeff Pyle wrote:
> Add it in a branch_route. That way if you have to route advance, it'll
> already be gone because you'll be on a new branch.
>
>
> - Jeff
>
>
> On Mon, Jun 14, 2021 at 5:52 PM David Villasmil <
> david.villasmil.w...@gmail.com> wrote:
>
>> Hello guys,
>>
>> So, I'm appending a header (append_hf("header")) to a forward.
>> That forward fails and I'm trying to remove it with remove_hf("header"),
>> but it's not getting removed for some reason, what am I doing wrong?
>>
>> Thanks everyone!
>>
>> David Villasmil
>> email: david.villasmil.w...@gmail.com
>> phone: +34669448337
>> ___
>> 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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[OpenSIPS-Users] Remove previously added header
Hello guys,
So, I'm appending a header (append_hf("header")) to a forward.
That forward fails and I'm trying to remove it with remove_hf("header"),
but it's not getting removed for some reason, what am I doing wrong?
Thanks everyone!
David Villasmil
email: david.villasmil.w...@gmail.com
phone: +34669448337
___
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Re: [OpenSIPS-Users] Getting header from 302
Hello again,
So following up on this, I understood i could get the header from the 302
reply with:
$(hdr(X-MyHeader))
Which is good, that works perfectly, but i can't verify the existence of
that header with:
is_present_hf("X-MyHeader")
Is there a way of doing that or should i just check $(hdr(X-MyHeader))
is not $null?
Regards,
David Villasmil
email: david.villasmil.w...@gmail.com
phone: +34669448337
On Thu, Jun 3, 2021 at 2:27 PM David Villasmil <
david.villasmil.w...@gmail.com> wrote:
> Thanks Ben,
>
> I understand now. Perfect.
>
> David
>
> On Thu, 3 Jun 2021 at 14:24, Ben Newlin wrote:
>
>> Yes, if a reply route is armed or the global reply route exists, they are
>> triggered first for any reply. Failure route, if armed, is triggered after
>> that for any >=300 response codes.
>>
>>
>>
>> Ben Newlin
>>
>>
>>
>> *From: *Users on behalf of David
>> Villasmil
>> *Date: *Thursday, June 3, 2021 at 8:10 AM
>> *To: *OpenSIPS users mailling list
>> *Subject: *Re: [OpenSIPS-Users] Getting header from 302
>>
>> Oh I didn’t register that param. I’ll try with that.
>>
>> On the other hand, I may be confused about how opensips processes >299
>> replies. Does it process them on onreply route and then goes to the failure
>> route? I mean that’s what I’m doing right now, but is this intended by
>> design? I would’ve though it’d go straight to the failure route, but
>> actually going to the onreply route sounds smart.
>>
>>
>>
>> On Wed, 2 Jun 2021 at 22:08, Jeff Pyle wrote:
>>
>> I've been working on a proxy to sit between MS Teams and "normal" SIP
>> stacks. Teams sends way too many 180s and RTP-less 183s so I sanitize them
>> like this:
>>
>> onreply_route[relay_reply] {
>> if (t_check_status("180")) {
>> if (isflagset("GOT_180")) {
>> drop;
>> } else {
>> setflag("GOT_180");
>> }
>> }
>>
>> if (isflagset("GOT_180") && t_check_status("183")) {
>> drop;
>> }
>>
>> }
>>
>>
>>
>> With this I stop superfluous 18x messages from being relayed downstream.
>> The 'drop' here kills the message completely. You could include the drop
>> if you want to stop the message from being relayed (which you probably do)
>> and are finished processing it in the script (which you are probably not).
>>
>>
>>
>> If I understand your application correctly, I'd populate the AVP in the
>> reply route and do everything else in the failure route. Or, try Liviu's
>> suggestion of using $(hdr(Identity)) in the failure_route directly.
>> Either way, then continue in the failure_route to do whatever else needs to
>> happen.
>>
>>
>>
>>
>>
>> - Jeff
>>
>>
>>
>>
>>
>>
>>
>> On Wed, Jun 2, 2021 at 2:10 PM David Villasmil <
>> david.villasmil.w...@gmail.com> wrote:
>>
>> Hello Jeff,
>>
>>
>>
>> That's exactly what I'm doing:
>>
>>
>>
>> # Relay to REDIRECT server
>> route[relay_to_REDIRECT]
>> {
>> t_on_reply("reply_from_REDIRECT");
>> t_on_failure("failure_from_REDIRECT");
>>
>> xlog("L_ERR", "[$ci][$rm]: Relaying to REDIRECT");
>> if (!t_relay()) {
>> xlog("L_ERR", "[$ci][$rm]: unable to relay request $ru to $tU --
>> replying with error");
>> sl_reply_error();
>> }
>>
>> exit;
>> }
>>
>> # Response from REDIRECT will come in here.
>> failure_route[failure_from_REDIRECT]
>> {
>> xlog("L_ERR", "[$ci][$rm]: I'm in
>> failure_route[failover_from_REDIRECT]");
>> if (t_was_cancelled()) {
>> exit;
>> }
>>
>> if(is_avp_set("$avp(myheader)")) {
>> xlog("L_ERR", "[$ci][$rm]: Got Identity Header:
>> $(hdr(myheader))");
>> setflag(100);
>> route(invite);
>> }
>> }
>>
>> # Response 302 from REDIRECT will come in here.
>> onreply_route[reply_from_REDIRECT]
>> {
>> xlog("L_ERR", "[$ci][$rm]: I'm in
>> onreply_route[reply_from_REDIRECT]");
>> if (t_was_cancelled()) {
>> exit;
>> }
>>
>> # detect redirect, store the header and send to "invite" as normally
>> if (t_check_status("302") && is_present_hf("myheader")) {
>> $avp(identity_header) = $(hdr(myheader));
>> setflag(100);
>> drop();
>> }
>> }
>>
>>
>>
>> So I suppose i don't need the drop()?
>>
>>
>>
>> Regards,
>>
>>
>>
>> David Villasmil
>>
>> email: david.villasmil.w...@gmail.com
>>
>> phone: +34669448337
>>
>>
>>
>> ___
>> Users mailing list
>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
> --
> Regards,
>
> David Villasmil
> email: david.villasmil.w...@gmail.com
> phone: +34669448337
>
___
Users mailing list
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Re: [OpenSIPS-Users] Set flag at runtime?
Thanks Ovidiu, that was it! Didn't register one was a bitmask and the other
the position.
Regards,
David Villasmil
email: david.villasmil.w...@gmail.com
phone: +34669448337
On Thu, Jun 10, 2021 at 2:11 AM Ovidiu Sas wrote:
> Read the docs.
> In the script, the argument is the position of the flag.
> Via MI, the argument is a bitmask.
>
> -ovidiu
>
> On Wed, Jun 9, 2021 at 6:33 PM David Villasmil
> wrote:
> >
> > So I'm trying to use gflags.so in 2.4.7
> >
> > I set the module as:
> >
> > loadmodule "gflags.so"
> > modparam("gflags", "initial", 0)
> >
> > then i'm doing:
> >
> > if (is_gflag("1")) {
> > xlog("L_ERR", "[$ci][$rm]: gflag(1) is set\n");
> > } else {
> > xlog("L_ERR", "[$ci][$rm]: gflag(1) is not set\n");
> > }
> >
> > Then, via CLI I'm setting the flag like so:
> >
> > # opensipsctl fifo is_gflag "1"
> > FALSE
> >
> > And is set it with:
> >
> > # opensipsctl fifo set_gflag "1"
> >
> > and check it is actually set:
> >
> > # opensipsctl fifo is_gflag "1"
> > TRUE
> >
> > But no matter what i do, the script always sees the flag as not set.
> >
> > Did I misunderstand the purpose of the module? I want to be able to set
> the flag dynamically from the CLI to enable/disable a feature at runtime...
> is this not possible?
> >
> > thanks guys.
> >
> > Regards,
> >
> > David Villasmil
> > email: david.villasmil.w...@gmail.com
> > phone: +34669448337
> >
> >
> > On Wed, Jun 9, 2021 at 8:55 PM David Villasmil <
> david.villasmil.w...@gmail.com> wrote:
> >>
> >> gflags!
> >>
> >> Regards,
> >>
> >> David Villasmil
> >> email: david.villasmil.w...@gmail.com
> >> phone: +34669448337
> >>
> >>
> >> On Wed, Jun 9, 2021 at 5:05 PM David Villasmil <
> david.villasmil.w...@gmail.com> wrote:
> >>>
> >>> Hello guys,
> >>>
> >>> On 2.4 is it possible to set a flag at runtime? I want to set a
> feature toggle based on that.
> >>>
> >>> Regards,
> >>>
> >>> David Villasmil
> >>> email: david.villasmil.w...@gmail.com
> >>> phone: +34669448337
> >
> > ___
> > Users mailing list
> > Users@lists.opensips.org
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
> --
> VoIP Embedded, Inc.
> http://www.voipembedded.com
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
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Re: [OpenSIPS-Users] Set flag at runtime?
Point is on the CLI I get TRUE, but on the script no matter what I do it is
always false
On Thu, 10 Jun 2021 at 01:51, David Villasmil <
david.villasmil.w...@gmail.com> wrote:
> Thanks Ben, I tried both :(
>
> On Thu, 10 Jun 2021 at 01:26, Ben Newlin wrote:
>
>> Per the documentation [1], the CLI command wants the value as an integer,
>> not quoted.
>>
>>
>>
>> Have you tried
>>
>>
>>
>> # opensipsctl fifo set_gflag 1
>>
>>
>>
>> [1] -
>> https://opensips.org/docs/modules/2.4.x/gflags.html#exported_mi_functions
>>
>>
>>
>> Ben Newlin
>>
>>
>>
>> *From: *Users on behalf of David
>> Villasmil
>> *Date: *Wednesday, June 9, 2021 at 6:34 PM
>> *To: *users@lists.opensips.org
>> *Subject: *Re: [OpenSIPS-Users] Set flag at runtime?
>>
>> So I'm trying to use gflags.so in 2.4.7
>>
>>
>>
>> I set the module as:
>>
>>
>>
>> loadmodule "gflags.so"
>> modparam("gflags", "initial", 0)
>>
>>
>>
>> then i'm doing:
>>
>>
>>
>> if (is_gflag("1")) {
>> xlog("L_ERR", "[$ci][$rm]: gflag(1) is set\n");
>> } else {
>> xlog("L_ERR", "[$ci][$rm]: gflag(1) is not set\n");
>> }
>>
>>
>>
>> Then, via CLI I'm setting the flag like so:
>>
>>
>>
>> # opensipsctl fifo is_gflag "1"
>> FALSE
>>
>>
>>
>> And is set it with:
>>
>>
>>
>> # opensipsctl fifo set_gflag "1"
>>
>>
>>
>> and check it is actually set:
>>
>>
>>
>> # opensipsctl fifo is_gflag "1"
>> TRUE
>>
>>
>>
>> But no matter what i do, the script always sees the flag as not set.
>>
>>
>>
>> Did I misunderstand the purpose of the module? I want to be able to set
>> the flag dynamically from the CLI to enable/disable a feature at runtime...
>> is this not possible?
>>
>>
>>
>> thanks guys.
>>
>>
>> Regards,
>>
>>
>>
>> David Villasmil
>>
>> email: david.villasmil.w...@gmail.com
>>
>> phone: +34669448337
>>
>>
>>
>>
>>
>> On Wed, Jun 9, 2021 at 8:55 PM David Villasmil <
>> david.villasmil.w...@gmail.com> wrote:
>>
>> gflags!
>>
>>
>> Regards,
>>
>>
>>
>> David Villasmil
>>
>> email: david.villasmil.w...@gmail.com
>>
>> phone: +34669448337
>>
>>
>>
>>
>>
>> On Wed, Jun 9, 2021 at 5:05 PM David Villasmil <
>> david.villasmil.w...@gmail.com> wrote:
>>
>> Hello guys,
>>
>>
>>
>> On 2.4 is it possible to set a flag at runtime? I want to set a feature
>> toggle based on that.
>>
>>
>> Regards,
>>
>>
>>
>> David Villasmil
>>
>> email: david.villasmil.w...@gmail.com
>>
>> phone: +34669448337
>>
>> ___
>> 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
>
--
Regards,
David Villasmil
email: david.villasmil.w...@gmail.com
phone: +34669448337
___
Users mailing list
Users@lists.opensips.org
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Re: [OpenSIPS-Users] Set flag at runtime?
Thanks Ben, I tried both :(
On Thu, 10 Jun 2021 at 01:26, Ben Newlin wrote:
> Per the documentation [1], the CLI command wants the value as an integer,
> not quoted.
>
>
>
> Have you tried
>
>
>
> # opensipsctl fifo set_gflag 1
>
>
>
> [1] -
> https://opensips.org/docs/modules/2.4.x/gflags.html#exported_mi_functions
>
>
>
> Ben Newlin
>
>
>
> *From: *Users on behalf of David
> Villasmil
> *Date: *Wednesday, June 9, 2021 at 6:34 PM
> *To: *users@lists.opensips.org
> *Subject: *Re: [OpenSIPS-Users] Set flag at runtime?
>
> So I'm trying to use gflags.so in 2.4.7
>
>
>
> I set the module as:
>
>
>
> loadmodule "gflags.so"
> modparam("gflags", "initial", 0)
>
>
>
> then i'm doing:
>
>
>
> if (is_gflag("1")) {
> xlog("L_ERR", "[$ci][$rm]: gflag(1) is set\n");
> } else {
> xlog("L_ERR", "[$ci][$rm]: gflag(1) is not set\n");
> }
>
>
>
> Then, via CLI I'm setting the flag like so:
>
>
>
> # opensipsctl fifo is_gflag "1"
> FALSE
>
>
>
> And is set it with:
>
>
>
> # opensipsctl fifo set_gflag "1"
>
>
>
> and check it is actually set:
>
>
>
> # opensipsctl fifo is_gflag "1"
> TRUE
>
>
>
> But no matter what i do, the script always sees the flag as not set.
>
>
>
> Did I misunderstand the purpose of the module? I want to be able to set
> the flag dynamically from the CLI to enable/disable a feature at runtime...
> is this not possible?
>
>
>
> thanks guys.
>
>
> Regards,
>
>
>
> David Villasmil
>
> email: david.villasmil.w...@gmail.com
>
> phone: +34669448337
>
>
>
>
>
> On Wed, Jun 9, 2021 at 8:55 PM David Villasmil <
> david.villasmil.w...@gmail.com> wrote:
>
> gflags!
>
>
> Regards,
>
>
>
> David Villasmil
>
> email: david.villasmil.w...@gmail.com
>
> phone: +34669448337
>
>
>
>
>
> On Wed, Jun 9, 2021 at 5:05 PM David Villasmil <
> david.villasmil.w...@gmail.com> wrote:
>
> Hello guys,
>
>
>
> On 2.4 is it possible to set a flag at runtime? I want to set a feature
> toggle based on that.
>
>
> Regards,
>
>
>
> David Villasmil
>
> email: david.villasmil.w...@gmail.com
>
> phone: +34669448337
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
--
Regards,
David Villasmil
email: david.villasmil.w...@gmail.com
phone: +34669448337
___
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Users@lists.opensips.org
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Re: [OpenSIPS-Users] Set flag at runtime?
So I'm trying to use gflags.so in 2.4.7
I set the module as:
loadmodule "gflags.so"
modparam("gflags", "initial", 0)
then i'm doing:
if (is_gflag("1")) {
xlog("L_ERR", "[$ci][$rm]: gflag(1) is set\n");
} else {
xlog("L_ERR", "[$ci][$rm]: gflag(1) is not set\n");
}
Then, via CLI I'm setting the flag like so:
# opensipsctl fifo is_gflag "1"
FALSE
And is set it with:
# opensipsctl fifo set_gflag "1"
and check it is actually set:
# opensipsctl fifo is_gflag "1"
TRUE
But no matter what i do, the script always sees the flag as not set.
Did I misunderstand the purpose of the module? I want to be able to set the
flag dynamically from the CLI to enable/disable a feature at runtime... is
this not possible?
thanks guys.
Regards,
David Villasmil
email: david.villasmil.w...@gmail.com
phone: +34669448337
On Wed, Jun 9, 2021 at 8:55 PM David Villasmil <
david.villasmil.w...@gmail.com> wrote:
> gflags!
>
> Regards,
>
> David Villasmil
> email: david.villasmil.w...@gmail.com
> phone: +34669448337
>
>
> On Wed, Jun 9, 2021 at 5:05 PM David Villasmil <
> david.villasmil.w...@gmail.com> wrote:
>
>> Hello guys,
>>
>> On 2.4 is it possible to set a flag at runtime? I want to set a feature
>> toggle based on that.
>>
>> Regards,
>>
>> David Villasmil
>> email: david.villasmil.w...@gmail.com
>> phone: +34669448337
>>
>
___
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Re: [OpenSIPS-Users] Set flag at runtime?
gflags! Regards, David Villasmil email: david.villasmil.w...@gmail.com phone: +34669448337 On Wed, Jun 9, 2021 at 5:05 PM David Villasmil < david.villasmil.w...@gmail.com> wrote: > Hello guys, > > On 2.4 is it possible to set a flag at runtime? I want to set a feature > toggle based on that. > > Regards, > > David Villasmil > email: david.villasmil.w...@gmail.com > phone: +34669448337 > ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Set flag at runtime?
Hello guys, On 2.4 is it possible to set a flag at runtime? I want to set a feature toggle based on that. Regards, David Villasmil email: david.villasmil.w...@gmail.com phone: +34669448337 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Getting header from 302
Thanks Ben,
I understand now. Perfect.
David
On Thu, 3 Jun 2021 at 14:24, Ben Newlin wrote:
> Yes, if a reply route is armed or the global reply route exists, they are
> triggered first for any reply. Failure route, if armed, is triggered after
> that for any >=300 response codes.
>
>
>
> Ben Newlin
>
>
>
> *From: *Users on behalf of David
> Villasmil
> *Date: *Thursday, June 3, 2021 at 8:10 AM
> *To: *OpenSIPS users mailling list
> *Subject: *Re: [OpenSIPS-Users] Getting header from 302
>
> Oh I didn’t register that param. I’ll try with that.
>
> On the other hand, I may be confused about how opensips processes >299
> replies. Does it process them on onreply route and then goes to the failure
> route? I mean that’s what I’m doing right now, but is this intended by
> design? I would’ve though it’d go straight to the failure route, but
> actually going to the onreply route sounds smart.
>
>
>
> On Wed, 2 Jun 2021 at 22:08, Jeff Pyle wrote:
>
> I've been working on a proxy to sit between MS Teams and "normal" SIP
> stacks. Teams sends way too many 180s and RTP-less 183s so I sanitize them
> like this:
>
> onreply_route[relay_reply] {
> if (t_check_status("180")) {
> if (isflagset("GOT_180")) {
> drop;
> } else {
> setflag("GOT_180");
> }
> }
>
> if (isflagset("GOT_180") && t_check_status("183")) {
> drop;
> }
>
> }
>
>
>
> With this I stop superfluous 18x messages from being relayed downstream.
> The 'drop' here kills the message completely. You could include the drop
> if you want to stop the message from being relayed (which you probably do)
> and are finished processing it in the script (which you are probably not).
>
>
>
> If I understand your application correctly, I'd populate the AVP in the
> reply route and do everything else in the failure route. Or, try Liviu's
> suggestion of using $(hdr(Identity)) in the failure_route directly.
> Either way, then continue in the failure_route to do whatever else needs to
> happen.
>
>
>
>
>
> - Jeff
>
>
>
>
>
>
>
> On Wed, Jun 2, 2021 at 2:10 PM David Villasmil <
> david.villasmil.w...@gmail.com> wrote:
>
> Hello Jeff,
>
>
>
> That's exactly what I'm doing:
>
>
>
> # Relay to REDIRECT server
> route[relay_to_REDIRECT]
> {
> t_on_reply("reply_from_REDIRECT");
> t_on_failure("failure_from_REDIRECT");
>
> xlog("L_ERR", "[$ci][$rm]: Relaying to REDIRECT");
> if (!t_relay()) {
> xlog("L_ERR", "[$ci][$rm]: unable to relay request $ru to $tU --
> replying with error");
> sl_reply_error();
> }
>
> exit;
> }
>
> # Response from REDIRECT will come in here.
> failure_route[failure_from_REDIRECT]
> {
> xlog("L_ERR", "[$ci][$rm]: I'm in
> failure_route[failover_from_REDIRECT]");
> if (t_was_cancelled()) {
> exit;
> }
>
> if(is_avp_set("$avp(myheader)")) {
> xlog("L_ERR", "[$ci][$rm]: Got Identity Header: $(hdr(myheader))");
> setflag(100);
> route(invite);
> }
> }
>
> # Response 302 from REDIRECT will come in here.
> onreply_route[reply_from_REDIRECT]
> {
> xlog("L_ERR", "[$ci][$rm]: I'm in onreply_route[reply_from_REDIRECT]");
> if (t_was_cancelled()) {
> exit;
> }
>
> # detect redirect, store the header and send to "invite" as normally
> if (t_check_status("302") && is_present_hf("myheader")) {
> $avp(identity_header) = $(hdr(myheader));
> setflag(100);
> drop();
> }
> }
>
>
>
> So I suppose i don't need the drop()?
>
>
>
> Regards,
>
>
>
> David Villasmil
>
> email: david.villasmil.w...@gmail.com
>
> phone: +34669448337
>
>
>
> ___
> 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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Re: [OpenSIPS-Users] Getting header from 302
Oh I didn’t register that param. I’ll try with that.
On the other hand, I may be confused about how opensips processes >299
replies. Does it process them on onreply route and then goes to the failure
route? I mean that’s what I’m doing right now, but is this intended by
design? I would’ve though it’d go straight to the failure route, but
actually going to the onreply route sounds smart.
On Wed, 2 Jun 2021 at 22:08, Jeff Pyle wrote:
> I've been working on a proxy to sit between MS Teams and "normal" SIP
> stacks. Teams sends way too many 180s and RTP-less 183s so I sanitize them
> like this:
>
> onreply_route[relay_reply] {
> if (t_check_status("180")) {
> if (isflagset("GOT_180")) {
> drop;
> } else {
> setflag("GOT_180");
> }
> }
>
> if (isflagset("GOT_180") && t_check_status("183")) {
> drop;
> }
> }
>
> With this I stop superfluous 18x messages from being relayed downstream.
> The 'drop' here kills the message completely. You could include the drop
> if you want to stop the message from being relayed (which you probably do)
> and are finished processing it in the script (which you are probably not).
>
> If I understand your application correctly, I'd populate the AVP in the
> reply route and do everything else in the failure route. Or, try Liviu's
> suggestion of using $(hdr(Identity)) in the failure_route directly.
> Either way, then continue in the failure_route to do whatever else needs to
> happen.
>
>
> - Jeff
>
>
>
> On Wed, Jun 2, 2021 at 2:10 PM David Villasmil <
> david.villasmil.w...@gmail.com> wrote:
>
>> Hello Jeff,
>>
>> That's exactly what I'm doing:
>>
>> # Relay to REDIRECT server
>> route[relay_to_REDIRECT]
>> {
>> t_on_reply("reply_from_REDIRECT");
>> t_on_failure("failure_from_REDIRECT");
>>
>> xlog("L_ERR", "[$ci][$rm]: Relaying to REDIRECT");
>> if (!t_relay()) {
>> xlog("L_ERR", "[$ci][$rm]: unable to relay request $ru to $tU --
>> replying with error");
>> sl_reply_error();
>> }
>>
>> exit;
>> }
>>
>> # Response from REDIRECT will come in here.
>> failure_route[failure_from_REDIRECT]
>> {
>> xlog("L_ERR", "[$ci][$rm]: I'm in
>> failure_route[failover_from_REDIRECT]");
>> if (t_was_cancelled()) {
>> exit;
>> }
>>
>> if(is_avp_set("$avp(myheader)")) {
>> xlog("L_ERR", "[$ci][$rm]: Got Identity Header:
>> $(hdr(myheader))");
>> setflag(100);
>> route(invite);
>> }
>> }
>>
>> # Response 302 from REDIRECT will come in here.
>> onreply_route[reply_from_REDIRECT]
>> {
>> xlog("L_ERR", "[$ci][$rm]: I'm in
>> onreply_route[reply_from_REDIRECT]");
>> if (t_was_cancelled()) {
>> exit;
>> }
>>
>> # detect redirect, store the header and send to "invite" as normally
>> if (t_check_status("302") && is_present_hf("myheader")) {
>> $avp(identity_header) = $(hdr(myheader));
>> setflag(100);
>> drop();
>> }
>> }
>>
>> So I suppose i don't need the drop()?
>>
>> Regards,
>>
>> David Villasmil
>> email: david.villasmil.w...@gmail.com
>> phone: +34669448337
>>
>>
>> On Wed, Jun 2, 2021 at 4:32 PM Jeff Pyle wrote:
>>
>>> If I arm both t_on_failure() and t_on_reply(), do a t_relay(), and a 302
>>> comes back, I have access to the reply in the onreply_route, then the
>>> failure_route. From a SIP perspective, a 302 is a failure since it's not
>>> 2xx-series, no? I don't do a drop() in the onreply_route. It just
>>> naturally follows its course to the failure_route.
>>>
>>> David, in your case, since you're trying to drop any 302 that doesn't
>>> have an Identity header, I'd check for its presence in the onreply_route
>>> and set a flag if there accordingly. And, capture its value in an AVP if
>>> you need. Next, in the failure_route, if (t_check_status("302") &&
>>> !isflagset("302_HAS_ID_HEADER")) drop; or something similar. You could
>>> easily expand that block to route-advance to your next carrier,
>>> send_repl
Re: [OpenSIPS-Users] Getting header from 302
Hello Jeff,
That's exactly what I'm doing:
# Relay to REDIRECT server
route[relay_to_REDIRECT]
{
t_on_reply("reply_from_REDIRECT");
t_on_failure("failure_from_REDIRECT");
xlog("L_ERR", "[$ci][$rm]: Relaying to REDIRECT");
if (!t_relay()) {
xlog("L_ERR", "[$ci][$rm]: unable to relay request $ru to $tU --
replying with error");
sl_reply_error();
}
exit;
}
# Response from REDIRECT will come in here.
failure_route[failure_from_REDIRECT]
{
xlog("L_ERR", "[$ci][$rm]: I'm in
failure_route[failover_from_REDIRECT]");
if (t_was_cancelled()) {
exit;
}
if(is_avp_set("$avp(myheader)")) {
xlog("L_ERR", "[$ci][$rm]: Got Identity Header: $(hdr(myheader))");
setflag(100);
route(invite);
}
}
# Response 302 from REDIRECT will come in here.
onreply_route[reply_from_REDIRECT]
{
xlog("L_ERR", "[$ci][$rm]: I'm in onreply_route[reply_from_REDIRECT]");
if (t_was_cancelled()) {
exit;
}
# detect redirect, store the header and send to "invite" as normally
if (t_check_status("302") && is_present_hf("myheader")) {
$avp(identity_header) = $(hdr(myheader));
setflag(100);
drop();
}
}
So I suppose i don't need the drop()?
Regards,
David Villasmil
email: david.villasmil.w...@gmail.com
phone: +34669448337
On Wed, Jun 2, 2021 at 4:32 PM Jeff Pyle wrote:
> If I arm both t_on_failure() and t_on_reply(), do a t_relay(), and a 302
> comes back, I have access to the reply in the onreply_route, then the
> failure_route. From a SIP perspective, a 302 is a failure since it's not
> 2xx-series, no? I don't do a drop() in the onreply_route. It just
> naturally follows its course to the failure_route.
>
> David, in your case, since you're trying to drop any 302 that doesn't have
> an Identity header, I'd check for its presence in the onreply_route and set
> a flag if there accordingly. And, capture its value in an AVP if you
> need. Next, in the failure_route, if (t_check_status("302") &&
> !isflagset("302_HAS_ID_HEADER")) drop; or something similar. You could
> easily expand that block to route-advance to your next carrier,
> send_reply(499, "Something Else"), or whatever you makes sense for your
> application.
>
>
> - Jeff
>
> On Wed, Jun 2, 2021 at 10:19 AM Johan De Clercq wrote:
>
>> that's because 302 is not an error.
>> So I guess that drop() is the only way.
>>
>> Op wo 2 jun. 2021 om 15:42 schreef David Villasmil <
>> david.villasmil.w...@gmail.com>:
>>
>>> Thanks Ben,
>>>
>>> That’s a good point. But only way I’ve found to jump over from oneply to
>>> failure_route is by doing a drop(). If there’s another way, I’d love to
>>> know about it!
>>>
>>> David
>>>
>>> On Wed, 2 Jun 2021 at 08:29, Ben Newlin wrote:
>>>
>>>> You still don’t need to call drop() as long as you are handling the
>>>> request in failure_route. The 302 will not be sent back upstream as long as
>>>> failure_route either creates a new branch request or sends back a different
>>>> reply code. Only if failure_route exits without doing either of these
>>>> things would the downstream 302 be sent back upstream as-is.
>>>>
>>>>
>>>>
>>>> In fact, as far as I know drop() has no functionality for responses >=
>>>> 200.
>>>>
>>>>
>>>>
>>>> Ben Newlin
>>>>
>>>>
>>>> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Getting header from 302
Thanks Ben,
That’s a good point. But only way I’ve found to jump over from oneply to
failure_route is by doing a drop(). If there’s another way, I’d love to
know about it!
David
On Wed, 2 Jun 2021 at 08:29, Ben Newlin wrote:
> You still don’t need to call drop() as long as you are handling the
> request in failure_route. The 302 will not be sent back upstream as long as
> failure_route either creates a new branch request or sends back a different
> reply code. Only if failure_route exits without doing either of these
> things would the downstream 302 be sent back upstream as-is.
>
>
>
> In fact, as far as I know drop() has no functionality for responses >= 200.
>
>
>
> Ben Newlin
>
>
>
> *From: *Users on behalf of Jeff Pyle <
> j...@ugnd.org>
> *Date: *Tuesday, June 1, 2021 at 2:48 PM
> *To: *OpenSIPS users mailling list
> *Subject: *Re: [OpenSIPS-Users] Getting header from 302
>
> Oh! Understood.
>
>
>
>
>
> - Jeff
>
>
>
>
>
> On Tue, Jun 1, 2021 at 2:42 PM David Villasmil <
> david.villasmil.w...@gmail.com> wrote:
>
> The thing is I _want_ to drop the 302, I don't want to do anything else
> with it.
>
>
> Regards,
>
>
>
> David Villasmil
>
> email: david.villasmil.w...@gmail.com
>
> phone: +34669448337
>
>
>
>
>
> On Tue, Jun 1, 2021 at 6:46 PM Jeff Pyle wrote:
>
> In my experience you don't need drop() in the reply route. Just store
> the AVP and move on. Something like this:
>
>
>
> onreply_route[collect_identity] {
>
> if (is_present_hf("Identity")) {
>
> $avp(identity) := $hdr(Identity);
>
> setflag("GOT_IDENTITY");
>
> }
>
> }
>
>
>
> If you've armed both the reply and failure routes with t_on_reply() and
> t_on_failure(), the $avp(identity) variable set here will be available in
> the failure_route. The GOT_IDENTITY flag, too.
>
>
>
>
>
> - Jeff
>
>
>
>
>
>
>
> - Jeff
>
>
>
>
>
> On Tue, Jun 1, 2021 at 11:20 AM David Villasmil <
> david.villasmil.w...@gmail.com> wrote:
>
> Yes, I see it is documented.
>
>
>
> So the reply header is only availanble on the "onreply" route, not on the
> "failure" route. That was my problem. I do indeed use an avp to store the
> header.
>
> I ended up getting the header on the "onreply" and storing it in an avp,
> set a flag and then drop(). I noticed the "failure" route is then executed.
>
> From there I can send the processing to the invite route and by checking
> the flag, adding the header from the avp.
>
>
>
> Thanks for your help!
>
>
> Regards,
>
>
>
> David Villasmil
>
> email: david.villasmil.w...@gmail.com
>
> phone: +34669448337
>
>
>
>
>
> On Tue, Jun 1, 2021 at 3:52 PM Ben Newlin wrote:
>
> It’s documented that it works this way. The message being processed in
> failure_route is the original request; in reply_route it’s the reply. [1]
>
>
>
> You can use variable context to access the reply from failure_route [2].
> Another option would be to extract the header value into and AVP in
> reply_route and then reference the AVP from failure_route.
>
>
>
>
>
> [1] - https://www.opensips.org/Documentation/Script-Routes-3-2
>
> [2] - https://www.opensips.org/Documentation/Script-CoreVar-3-2
>
>
>
> Ben Newlin
>
>
>
> *From: *Users on behalf of David
> Villasmil
> *Date: *Tuesday, June 1, 2021 at 10:43 AM
> *To: *OpenSIPS users mailling list
> *Subject: *Re: [OpenSIPS-Users] Getting header from 302
>
> Yeah, my thing is when i use the failure route i can in theory grab the
> response header and ignore the 302 and send to the invite route again to
> actually send the call out via do_routing.
>
> What I'm trying to do is:
>
> - On receiving an invite: forward to an endpoint.
>
> - This endpoint will simply reply with 302 including a header.
>
> - I want to grab that header and continue routing normally (do_routing)
>
>
>
> I could do that with the failure route, but not so sure about the onreply
> route.
>
>
> Regards,
>
>
>
> David Villasmil
>
> email: david.villasmil.w...@gmail.com
>
> phone: +34669448337
>
>
>
>
>
> On Tue, Jun 1, 2021 at 2:34 PM Jeff Pyle wrote:
>
> I don't think you're doing anything wrong. I think I found the same
> thing, that headers on the reply were available only in a reply route and
> not in a failure route. If you know where to look for them to populate the
> AVP, I suppose it doesn't matter much.
>
>
>
>
Re: [OpenSIPS-Users] Getting header from 302
The thing is I _want_ to drop the 302, I don't want to do anything else
with it.
Regards,
David Villasmil
email: david.villasmil.w...@gmail.com
phone: +34669448337
On Tue, Jun 1, 2021 at 6:46 PM Jeff Pyle wrote:
> In my experience you don't need drop() in the reply route. Just store
> the AVP and move on. Something like this:
>
> onreply_route[collect_identity] {
> if (is_present_hf("Identity")) {
> $avp(identity) := $hdr(Identity);
> setflag("GOT_IDENTITY");
> }
> }
>
> If you've armed both the reply and failure routes with t_on_reply() and
> t_on_failure(), the $avp(identity) variable set here will be available in
> the failure_route. The GOT_IDENTITY flag, too.
>
>
> - Jeff
>
>
>
> - Jeff
>
>
> On Tue, Jun 1, 2021 at 11:20 AM David Villasmil <
> david.villasmil.w...@gmail.com> wrote:
>
>> Yes, I see it is documented.
>>
>> So the reply header is only availanble on the "onreply" route, not on the
>> "failure" route. That was my problem. I do indeed use an avp to store the
>> header.
>> I ended up getting the header on the "onreply" and storing it in an avp,
>> set a flag and then drop(). I noticed the "failure" route is then executed.
>> From there I can send the processing to the invite route and by checking
>> the flag, adding the header from the avp.
>>
>> Thanks for your help!
>>
>> Regards,
>>
>> David Villasmil
>> email: david.villasmil.w...@gmail.com
>> phone: +34669448337
>>
>>
>> On Tue, Jun 1, 2021 at 3:52 PM Ben Newlin wrote:
>>
>>> It’s documented that it works this way. The message being processed in
>>> failure_route is the original request; in reply_route it’s the reply. [1]
>>>
>>>
>>>
>>> You can use variable context to access the reply from failure_route [2].
>>> Another option would be to extract the header value into and AVP in
>>> reply_route and then reference the AVP from failure_route.
>>>
>>>
>>>
>>>
>>>
>>> [1] - https://www.opensips.org/Documentation/Script-Routes-3-2
>>>
>>> [2] - https://www.opensips.org/Documentation/Script-CoreVar-3-2
>>>
>>>
>>>
>>> Ben Newlin
>>>
>>>
>>>
>>> *From: *Users on behalf of David
>>> Villasmil
>>> *Date: *Tuesday, June 1, 2021 at 10:43 AM
>>> *To: *OpenSIPS users mailling list
>>> *Subject: *Re: [OpenSIPS-Users] Getting header from 302
>>>
>>> Yeah, my thing is when i use the failure route i can in theory grab the
>>> response header and ignore the 302 and send to the invite route again to
>>> actually send the call out via do_routing.
>>>
>>> What I'm trying to do is:
>>>
>>> - On receiving an invite: forward to an endpoint.
>>>
>>> - This endpoint will simply reply with 302 including a header.
>>>
>>> - I want to grab that header and continue routing normally (do_routing)
>>>
>>>
>>>
>>> I could do that with the failure route, but not so sure about the
>>> onreply route.
>>>
>>>
>>> Regards,
>>>
>>>
>>>
>>> David Villasmil
>>>
>>> email: david.villasmil.w...@gmail.com
>>>
>>> phone: +34669448337
>>>
>>>
>>>
>>>
>>>
>>> On Tue, Jun 1, 2021 at 2:34 PM Jeff Pyle wrote:
>>>
>>> I don't think you're doing anything wrong. I think I found the same
>>> thing, that headers on the reply were available only in a reply route and
>>> not in a failure route. If you know where to look for them to populate the
>>> AVP, I suppose it doesn't matter much.
>>>
>>>
>>>
>>> I haven't looked at the code but I suspect all the routes other than an
>>> onreply_route give you access to the requests headers, and onreply_route
>>> gives you access to the reply headers. Makes sense I guess.
>>>
>>>
>>>
>>>
>>>
>>> - Jeff
>>>
>>>
>>>
>>>
>>>
>>> On Tue, Jun 1, 2021 at 9:31 AM David Villasmil <
>>> david.villasmil.w...@gmail.com> wrote:
>>>
>>> Thanks Jeff,
>>>
>>>
>>>
>>> MMM, that's strange, I was using it on failure route and the route was
>>> being executed, but the data wasn't there. I put it on the onreply route
>>> and that o
Re: [OpenSIPS-Users] Getting header from 302
Yes, I see it is documented.
So the reply header is only availanble on the "onreply" route, not on the
"failure" route. That was my problem. I do indeed use an avp to store the
header.
I ended up getting the header on the "onreply" and storing it in an avp,
set a flag and then drop(). I noticed the "failure" route is then executed.
>From there I can send the processing to the invite route and by checking
the flag, adding the header from the avp.
Thanks for your help!
Regards,
David Villasmil
email: david.villasmil.w...@gmail.com
phone: +34669448337
On Tue, Jun 1, 2021 at 3:52 PM Ben Newlin wrote:
> It’s documented that it works this way. The message being processed in
> failure_route is the original request; in reply_route it’s the reply. [1]
>
>
>
> You can use variable context to access the reply from failure_route [2].
> Another option would be to extract the header value into and AVP in
> reply_route and then reference the AVP from failure_route.
>
>
>
>
>
> [1] - https://www.opensips.org/Documentation/Script-Routes-3-2
>
> [2] - https://www.opensips.org/Documentation/Script-CoreVar-3-2
>
>
>
> Ben Newlin
>
>
>
> *From: *Users on behalf of David
> Villasmil
> *Date: *Tuesday, June 1, 2021 at 10:43 AM
> *To: *OpenSIPS users mailling list
> *Subject: *Re: [OpenSIPS-Users] Getting header from 302
>
> Yeah, my thing is when i use the failure route i can in theory grab the
> response header and ignore the 302 and send to the invite route again to
> actually send the call out via do_routing.
>
> What I'm trying to do is:
>
> - On receiving an invite: forward to an endpoint.
>
> - This endpoint will simply reply with 302 including a header.
>
> - I want to grab that header and continue routing normally (do_routing)
>
>
>
> I could do that with the failure route, but not so sure about the onreply
> route.
>
>
> Regards,
>
>
>
> David Villasmil
>
> email: david.villasmil.w...@gmail.com
>
> phone: +34669448337
>
>
>
>
>
> On Tue, Jun 1, 2021 at 2:34 PM Jeff Pyle wrote:
>
> I don't think you're doing anything wrong. I think I found the same
> thing, that headers on the reply were available only in a reply route and
> not in a failure route. If you know where to look for them to populate the
> AVP, I suppose it doesn't matter much.
>
>
>
> I haven't looked at the code but I suspect all the routes other than an
> onreply_route give you access to the requests headers, and onreply_route
> gives you access to the reply headers. Makes sense I guess.
>
>
>
>
>
> - Jeff
>
>
>
>
>
> On Tue, Jun 1, 2021 at 9:31 AM David Villasmil <
> david.villasmil.w...@gmail.com> wrote:
>
> Thanks Jeff,
>
>
>
> MMM, that's strange, I was using it on failure route and the route was
> being executed, but the data wasn't there. I put it on the onreply route
> and that one is now executed with the data correctly there...
>
>
>
> I probably did something wrong.
>
>
>
> Thanks again Jeff!
>
>
> Regards,
>
>
>
> David Villasmil
>
> email: david.villasmil.w...@gmail.com
>
> phone: +34669448337
>
>
>
>
>
> On Tue, Jun 1, 2021 at 12:37 PM Jeff Pyle wrote:
>
> In which route are you trying to use if (is_present_hf("Identity"))?
> Since the 302 is both a reply and a "failure", I suggest seeing if it
> appears in either the armed onreply_route or failure_route.
>
>
>
> I think From is available because it was present in the original request
> route.
>
>
>
>
>
> - Jeff
>
>
>
> On Tue, Jun 1, 2021 at 5:39 AM David Villasmil <
> david.villasmil.w...@gmail.com> wrote:
>
> Anyone has any idea about this? Appreciate your help.
>
>
>
> On Mon, 31 May 2021 at 21:11, David Villasmil <
> david.villasmil.w...@gmail.com> wrote:
>
> So weird,
>
> I can get the From header, but not "Identity"...
>
>
> Regards,
>
>
>
> David Villasmil
>
> email: david.villasmil.w...@gmail.com
>
> phone: +34669448337
>
>
>
>
>
> On Mon, May 31, 2021 at 8:22 PM David Villasmil <
> david.villasmil.w...@gmail.com> wrote:
>
> This is really weird,
>
>
>
> if (is_present_hf("Identity"))
>
>
>
> says it is not present, but it is!
>
>
> Regards,
>
>
>
> David Villasmil
>
> email: david.villasmil.w...@gmail.com
>
> phone: +34669448337
>
>
>
>
>
> On Mon, May 31, 2021 at 7:47 PM David Villasmil <
> david.villasmil.w...@
Re: [OpenSIPS-Users] Getting header from 302
Yeah, my thing is when i use the failure route i can in theory grab the
response header and ignore the 302 and send to the invite route again to
actually send the call out via do_routing.
What I'm trying to do is:
- On receiving an invite: forward to an endpoint.
- This endpoint will simply reply with 302 including a header.
- I want to grab that header and continue routing normally (do_routing)
I could do that with the failure route, but not so sure about the onreply
route.
Regards,
David Villasmil
email: david.villasmil.w...@gmail.com
phone: +34669448337
On Tue, Jun 1, 2021 at 2:34 PM Jeff Pyle wrote:
> I don't think you're doing anything wrong. I think I found the same
> thing, that headers on the reply were available only in a reply route and
> not in a failure route. If you know where to look for them to populate the
> AVP, I suppose it doesn't matter much.
>
> I haven't looked at the code but I suspect all the routes other than an
> onreply_route give you access to the requests headers, and onreply_route
> gives you access to the reply headers. Makes sense I guess.
>
>
> - Jeff
>
>
> On Tue, Jun 1, 2021 at 9:31 AM David Villasmil <
> david.villasmil.w...@gmail.com> wrote:
>
>> Thanks Jeff,
>>
>> MMM, that's strange, I was using it on failure route and the route was
>> being executed, but the data wasn't there. I put it on the onreply route
>> and that one is now executed with the data correctly there...
>>
>> I probably did something wrong.
>>
>> Thanks again Jeff!
>>
>> Regards,
>>
>> David Villasmil
>> email: david.villasmil.w...@gmail.com
>> phone: +34669448337
>>
>>
>> On Tue, Jun 1, 2021 at 12:37 PM Jeff Pyle wrote:
>>
>>> In which route are you trying to use if (is_present_hf("Identity"))?
>>> Since the 302 is both a reply and a "failure", I suggest seeing if it
>>> appears in either the armed onreply_route or failure_route.
>>>
>>> I think From is available because it was present in the original request
>>> route.
>>>
>>>
>>> - Jeff
>>>
>>> On Tue, Jun 1, 2021 at 5:39 AM David Villasmil <
>>> david.villasmil.w...@gmail.com> wrote:
>>>
>>>> Anyone has any idea about this? Appreciate your help.
>>>>
>>>> On Mon, 31 May 2021 at 21:11, David Villasmil <
>>>> david.villasmil.w...@gmail.com> wrote:
>>>>
>>>>> So weird,
>>>>> I can get the From header, but not "Identity"...
>>>>>
>>>>> Regards,
>>>>>
>>>>> David Villasmil
>>>>> email: david.villasmil.w...@gmail.com
>>>>> phone: +34669448337
>>>>>
>>>>>
>>>>> On Mon, May 31, 2021 at 8:22 PM David Villasmil <
>>>>> david.villasmil.w...@gmail.com> wrote:
>>>>>
>>>>>> This is really weird,
>>>>>>
>>>>>> if (is_present_hf("Identity"))
>>>>>>
>>>>>> says it is not present, but it is!
>>>>>>
>>>>>> Regards,
>>>>>>
>>>>>> David Villasmil
>>>>>> email: david.villasmil.w...@gmail.com
>>>>>> phone: +34669448337
>>>>>>
>>>>>>
>>>>>> On Mon, May 31, 2021 at 7:47 PM David Villasmil <
>>>>>> david.villasmil.w...@gmail.com> wrote:
>>>>>>
>>>>>>> Hello Guys,
>>>>>>>
>>>>>>> I'm getting a header on a 302 which i'm trying to get, but for some
>>>>>>> reason I can't.
>>>>>>>
>>>>>>> This is an example 302:
>>>>>>>
>>>>>>> 2021/05/31 18:42:36.499157 10.231.32.237:5060 -> 10.231.57.11:6075
>>>>>>> SIP/2.0 302 Redirect
>>>>>>> Via: SIP/2.0/UDP 1.2.3.4:
>>>>>>> 6075;branch=z9hG4bKd8e8.50036ee6.0;received=10.231.57.11
>>>>>>> Via: SIP/2.0/UDP
>>>>>>> 10.231.33.135;branch=z9hG4bKd8e8.fe2738f41d26b2b68328691c326a077a.0
>>>>>>> v:SIP/2.0/UDP 10.231.49.211:6060
>>>>>>> ;received=10.231.49.211;rport=6060;branch=z9hG4bK68SgceeareaUa
>>>>>>> f:"+1888333";tag=ZBQ713X9pgD5S
>>>>>>> t:>>>>>> >;tag=9dd61ff61e802d8e2bef5f14621ef3c2.50cf6b6c
>>>>>>> i:cf649a38-3ce2-123a-eaad-122eaa5d9655
>
Re: [OpenSIPS-Users] Getting header from 302
Thanks Jeff,
MMM, that's strange, I was using it on failure route and the route was
being executed, but the data wasn't there. I put it on the onreply route
and that one is now executed with the data correctly there...
I probably did something wrong.
Thanks again Jeff!
Regards,
David Villasmil
email: david.villasmil.w...@gmail.com
phone: +34669448337
On Tue, Jun 1, 2021 at 12:37 PM Jeff Pyle wrote:
> In which route are you trying to use if (is_present_hf("Identity"))?
> Since the 302 is both a reply and a "failure", I suggest seeing if it
> appears in either the armed onreply_route or failure_route.
>
> I think From is available because it was present in the original request
> route.
>
>
> - Jeff
>
> On Tue, Jun 1, 2021 at 5:39 AM David Villasmil <
> david.villasmil.w...@gmail.com> wrote:
>
>> Anyone has any idea about this? Appreciate your help.
>>
>> On Mon, 31 May 2021 at 21:11, David Villasmil <
>> david.villasmil.w...@gmail.com> wrote:
>>
>>> So weird,
>>> I can get the From header, but not "Identity"...
>>>
>>> Regards,
>>>
>>> David Villasmil
>>> email: david.villasmil.w...@gmail.com
>>> phone: +34669448337
>>>
>>>
>>> On Mon, May 31, 2021 at 8:22 PM David Villasmil <
>>> david.villasmil.w...@gmail.com> wrote:
>>>
>>>> This is really weird,
>>>>
>>>> if (is_present_hf("Identity"))
>>>>
>>>> says it is not present, but it is!
>>>>
>>>> Regards,
>>>>
>>>> David Villasmil
>>>> email: david.villasmil.w...@gmail.com
>>>> phone: +34669448337
>>>>
>>>>
>>>> On Mon, May 31, 2021 at 7:47 PM David Villasmil <
>>>> david.villasmil.w...@gmail.com> wrote:
>>>>
>>>>> Hello Guys,
>>>>>
>>>>> I'm getting a header on a 302 which i'm trying to get, but for some
>>>>> reason I can't.
>>>>>
>>>>> This is an example 302:
>>>>>
>>>>> 2021/05/31 18:42:36.499157 10.231.32.237:5060 -> 10.231.57.11:6075
>>>>> SIP/2.0 302 Redirect
>>>>> Via: SIP/2.0/UDP 1.2.3.4:
>>>>> 6075;branch=z9hG4bKd8e8.50036ee6.0;received=10.231.57.11
>>>>> Via: SIP/2.0/UDP
>>>>> 10.231.33.135;branch=z9hG4bKd8e8.fe2738f41d26b2b68328691c326a077a.0
>>>>> v:SIP/2.0/UDP 10.231.49.211:6060
>>>>> ;received=10.231.49.211;rport=6060;branch=z9hG4bK68SgceeareaUa
>>>>> f:"+1888333";tag=ZBQ713X9pgD5S
>>>>> t:>>>> >;tag=9dd61ff61e802d8e2bef5f14621ef3c2.50cf6b6c
>>>>> i:cf649a38-3ce2-123a-eaad-122eaa5d9655
>>>>> CSeq:36689486 INVITE
>>>>> Identity:
>>>>> eyJhbGciOiJFUzI1NiIsInBwdCI6InNoYWtlbiIsInR5cCI6InBhc3Nwb3J0IiwieDV1IjoiaHR0cHM6Ly9vcHMtc3RhdGljLnMzLmFtYXpvbmF3cy5jb20vc3Rpci1zaGFrZW4vZWMyNTYtcHVibGljLnBlbSJ9.eyJhdHRlc3QiO
>>>>>
>>>>> BIiwiZGVzdCI6eyJ0biI6WyIxNzg2NDEwNzgzNyJdfSwiaWF0IjoxNjIyNDg2NTU2LCJvcmlnIjp7InRuIjoiKzEzMTU5ODUyNTk0In0sIm9yaWdpZCI6IjhlZGE4M2Q1LWY1MjEtNDQzZC1iNDI0LWIzNDQ3MDc4ZjYxZCJ9.cjIz9VwlS9_6qA
>>>>>
>>>>> 6mmDgottk41BLpQcA40HdvV_6jAPqQ1EIL3_jLWl25oHeVEWOzTMhcERp4Jn-JZ4vP_n3w;info=<
>>>>> https://somedomain.com/stir-shaken/ec256-public.pem
>>>>> >;alg=ES256;ppt=shaken
>>>>> Server: kamailio (5.5.0 (x86_64/linux))
>>>>> Content-Length: 0
>>>>>
>>>>> I'm trying to get the "Identity" header with:
>>>>>
>>>>> $avp(identity_header) = $(hdr(Identity));
>>>>>
>>>>> But It's coming up
>>>>>
>>>>> Any ideas of what I'm doing wrong?
>>>>>
>>>>> Regards,
>>>>>
>>>>> David Villasmil
>>>>> email: david.villasmil.w...@gmail.com
>>>>> phone: +34669448337
>>>>>
>>>> --
>> Regards,
>>
>> David Villasmil
>> email: david.villasmil.w...@gmail.com
>> phone: +34669448337
>> ___
>> 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] Getting header from 302
Anyone has any idea about this? Appreciate your help.
On Mon, 31 May 2021 at 21:11, David Villasmil <
david.villasmil.w...@gmail.com> wrote:
> So weird,
> I can get the From header, but not "Identity"...
>
> Regards,
>
> David Villasmil
> email: david.villasmil.w...@gmail.com
> phone: +34669448337
>
>
> On Mon, May 31, 2021 at 8:22 PM David Villasmil <
> david.villasmil.w...@gmail.com> wrote:
>
>> This is really weird,
>>
>> if (is_present_hf("Identity"))
>>
>> says it is not present, but it is!
>>
>> Regards,
>>
>> David Villasmil
>> email: david.villasmil.w...@gmail.com
>> phone: +34669448337
>>
>>
>> On Mon, May 31, 2021 at 7:47 PM David Villasmil <
>> david.villasmil.w...@gmail.com> wrote:
>>
>>> Hello Guys,
>>>
>>> I'm getting a header on a 302 which i'm trying to get, but for some
>>> reason I can't.
>>>
>>> This is an example 302:
>>>
>>> 2021/05/31 18:42:36.499157 10.231.32.237:5060 -> 10.231.57.11:6075
>>> SIP/2.0 302 Redirect
>>> Via: SIP/2.0/UDP 1.2.3.4:
>>> 6075;branch=z9hG4bKd8e8.50036ee6.0;received=10.231.57.11
>>> Via: SIP/2.0/UDP
>>> 10.231.33.135;branch=z9hG4bKd8e8.fe2738f41d26b2b68328691c326a077a.0
>>> v:SIP/2.0/UDP 10.231.49.211:6060
>>> ;received=10.231.49.211;rport=6060;branch=z9hG4bK68SgceeareaUa
>>> f:"+1888333";tag=ZBQ713X9pgD5S
>>> t:>> >;tag=9dd61ff61e802d8e2bef5f14621ef3c2.50cf6b6c
>>> i:cf649a38-3ce2-123a-eaad-122eaa5d9655
>>> CSeq:36689486 INVITE
>>> Identity:
>>> eyJhbGciOiJFUzI1NiIsInBwdCI6InNoYWtlbiIsInR5cCI6InBhc3Nwb3J0IiwieDV1IjoiaHR0cHM6Ly9vcHMtc3RhdGljLnMzLmFtYXpvbmF3cy5jb20vc3Rpci1zaGFrZW4vZWMyNTYtcHVibGljLnBlbSJ9.eyJhdHRlc3QiO
>>>
>>> BIiwiZGVzdCI6eyJ0biI6WyIxNzg2NDEwNzgzNyJdfSwiaWF0IjoxNjIyNDg2NTU2LCJvcmlnIjp7InRuIjoiKzEzMTU5ODUyNTk0In0sIm9yaWdpZCI6IjhlZGE4M2Q1LWY1MjEtNDQzZC1iNDI0LWIzNDQ3MDc4ZjYxZCJ9.cjIz9VwlS9_6qA
>>>
>>> 6mmDgottk41BLpQcA40HdvV_6jAPqQ1EIL3_jLWl25oHeVEWOzTMhcERp4Jn-JZ4vP_n3w;info=<
>>> https://somedomain.com/stir-shaken/ec256-public.pem
>>> >;alg=ES256;ppt=shaken
>>> Server: kamailio (5.5.0 (x86_64/linux))
>>> Content-Length: 0
>>>
>>> I'm trying to get the "Identity" header with:
>>>
>>> $avp(identity_header) = $(hdr(Identity));
>>>
>>> But It's coming up
>>>
>>> Any ideas of what I'm doing wrong?
>>>
>>> Regards,
>>>
>>> David Villasmil
>>> email: david.villasmil.w...@gmail.com
>>> phone: +34669448337
>>>
>> --
Regards,
David Villasmil
email: david.villasmil.w...@gmail.com
phone: +34669448337
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Re: [OpenSIPS-Users] Getting header from 302
So weird,
I can get the From header, but not "Identity"...
Regards,
David Villasmil
email: david.villasmil.w...@gmail.com
phone: +34669448337
On Mon, May 31, 2021 at 8:22 PM David Villasmil <
david.villasmil.w...@gmail.com> wrote:
> This is really weird,
>
> if (is_present_hf("Identity"))
>
> says it is not present, but it is!
>
> Regards,
>
> David Villasmil
> email: david.villasmil.w...@gmail.com
> phone: +34669448337
>
>
> On Mon, May 31, 2021 at 7:47 PM David Villasmil <
> david.villasmil.w...@gmail.com> wrote:
>
>> Hello Guys,
>>
>> I'm getting a header on a 302 which i'm trying to get, but for some
>> reason I can't.
>>
>> This is an example 302:
>>
>> 2021/05/31 18:42:36.499157 10.231.32.237:5060 -> 10.231.57.11:6075
>> SIP/2.0 302 Redirect
>> Via: SIP/2.0/UDP 1.2.3.4:
>> 6075;branch=z9hG4bKd8e8.50036ee6.0;received=10.231.57.11
>> Via: SIP/2.0/UDP
>> 10.231.33.135;branch=z9hG4bKd8e8.fe2738f41d26b2b68328691c326a077a.0
>> v:SIP/2.0/UDP 10.231.49.211:6060
>> ;received=10.231.49.211;rport=6060;branch=z9hG4bK68SgceeareaUa
>> f:"+1888333";tag=ZBQ713X9pgD5S
>> t:> >;tag=9dd61ff61e802d8e2bef5f14621ef3c2.50cf6b6c
>> i:cf649a38-3ce2-123a-eaad-122eaa5d9655
>> CSeq:36689486 INVITE
>> Identity:
>> eyJhbGciOiJFUzI1NiIsInBwdCI6InNoYWtlbiIsInR5cCI6InBhc3Nwb3J0IiwieDV1IjoiaHR0cHM6Ly9vcHMtc3RhdGljLnMzLmFtYXpvbmF3cy5jb20vc3Rpci1zaGFrZW4vZWMyNTYtcHVibGljLnBlbSJ9.eyJhdHRlc3QiO
>>
>> BIiwiZGVzdCI6eyJ0biI6WyIxNzg2NDEwNzgzNyJdfSwiaWF0IjoxNjIyNDg2NTU2LCJvcmlnIjp7InRuIjoiKzEzMTU5ODUyNTk0In0sIm9yaWdpZCI6IjhlZGE4M2Q1LWY1MjEtNDQzZC1iNDI0LWIzNDQ3MDc4ZjYxZCJ9.cjIz9VwlS9_6qA
>>
>> 6mmDgottk41BLpQcA40HdvV_6jAPqQ1EIL3_jLWl25oHeVEWOzTMhcERp4Jn-JZ4vP_n3w;info=<
>> https://somedomain.com/stir-shaken/ec256-public.pem>;alg=ES256;ppt=shaken
>> Server: kamailio (5.5.0 (x86_64/linux))
>> Content-Length: 0
>>
>> I'm trying to get the "Identity" header with:
>>
>> $avp(identity_header) = $(hdr(Identity));
>>
>> But It's coming up
>>
>> Any ideas of what I'm doing wrong?
>>
>> Regards,
>>
>> David Villasmil
>> email: david.villasmil.w...@gmail.com
>> phone: +34669448337
>>
>
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Re: [OpenSIPS-Users] Getting header from 302
This is really weird,
if (is_present_hf("Identity"))
says it is not present, but it is!
Regards,
David Villasmil
email: david.villasmil.w...@gmail.com
phone: +34669448337
On Mon, May 31, 2021 at 7:47 PM David Villasmil <
david.villasmil.w...@gmail.com> wrote:
> Hello Guys,
>
> I'm getting a header on a 302 which i'm trying to get, but for some reason
> I can't.
>
> This is an example 302:
>
> 2021/05/31 18:42:36.499157 10.231.32.237:5060 -> 10.231.57.11:6075
> SIP/2.0 302 Redirect
> Via: SIP/2.0/UDP 1.2.3.4:
> 6075;branch=z9hG4bKd8e8.50036ee6.0;received=10.231.57.11
> Via: SIP/2.0/UDP
> 10.231.33.135;branch=z9hG4bKd8e8.fe2738f41d26b2b68328691c326a077a.0
> v:SIP/2.0/UDP 10.231.49.211:6060
> ;received=10.231.49.211;rport=6060;branch=z9hG4bK68SgceeareaUa
> f:"+1888333";tag=ZBQ713X9pgD5S
> t: >;tag=9dd61ff61e802d8e2bef5f14621ef3c2.50cf6b6c
> i:cf649a38-3ce2-123a-eaad-122eaa5d9655
> CSeq:36689486 INVITE
> Identity:
> eyJhbGciOiJFUzI1NiIsInBwdCI6InNoYWtlbiIsInR5cCI6InBhc3Nwb3J0IiwieDV1IjoiaHR0cHM6Ly9vcHMtc3RhdGljLnMzLmFtYXpvbmF3cy5jb20vc3Rpci1zaGFrZW4vZWMyNTYtcHVibGljLnBlbSJ9.eyJhdHRlc3QiO
>
> BIiwiZGVzdCI6eyJ0biI6WyIxNzg2NDEwNzgzNyJdfSwiaWF0IjoxNjIyNDg2NTU2LCJvcmlnIjp7InRuIjoiKzEzMTU5ODUyNTk0In0sIm9yaWdpZCI6IjhlZGE4M2Q1LWY1MjEtNDQzZC1iNDI0LWIzNDQ3MDc4ZjYxZCJ9.cjIz9VwlS9_6qA
>
> 6mmDgottk41BLpQcA40HdvV_6jAPqQ1EIL3_jLWl25oHeVEWOzTMhcERp4Jn-JZ4vP_n3w;info=<
> https://somedomain.com/stir-shaken/ec256-public.pem>;alg=ES256;ppt=shaken
> Server: kamailio (5.5.0 (x86_64/linux))
> Content-Length: 0
>
> I'm trying to get the "Identity" header with:
>
> $avp(identity_header) = $(hdr(Identity));
>
> But It's coming up
>
> Any ideas of what I'm doing wrong?
>
> Regards,
>
> David Villasmil
> email: david.villasmil.w...@gmail.com
> phone: +34669448337
>
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[OpenSIPS-Users] Getting header from 302
Hello Guys, I'm getting a header on a 302 which i'm trying to get, but for some reason I can't. This is an example 302: 2021/05/31 18:42:36.499157 10.231.32.237:5060 -> 10.231.57.11:6075 SIP/2.0 302 Redirect Via: SIP/2.0/UDP 1.2.3.4: 6075;branch=z9hG4bKd8e8.50036ee6.0;received=10.231.57.11 Via: SIP/2.0/UDP 10.231.33.135;branch=z9hG4bKd8e8.fe2738f41d26b2b68328691c326a077a.0 v:SIP/2.0/UDP 10.231.49.211:6060 ;received=10.231.49.211;rport=6060;branch=z9hG4bK68SgceeareaUa f:"+1888333";tag=ZBQ713X9pgD5S t:;tag=9dd61ff61e802d8e2bef5f14621ef3c2.50cf6b6c i:cf649a38-3ce2-123a-eaad-122eaa5d9655 CSeq:36689486 INVITE Identity: eyJhbGciOiJFUzI1NiIsInBwdCI6InNoYWtlbiIsInR5cCI6InBhc3Nwb3J0IiwieDV1IjoiaHR0cHM6Ly9vcHMtc3RhdGljLnMzLmFtYXpvbmF3cy5jb20vc3Rpci1zaGFrZW4vZWMyNTYtcHVibGljLnBlbSJ9.eyJhdHRlc3QiO BIiwiZGVzdCI6eyJ0biI6WyIxNzg2NDEwNzgzNyJdfSwiaWF0IjoxNjIyNDg2NTU2LCJvcmlnIjp7InRuIjoiKzEzMTU5ODUyNTk0In0sIm9yaWdpZCI6IjhlZGE4M2Q1LWY1MjEtNDQzZC1iNDI0LWIzNDQ3MDc4ZjYxZCJ9.cjIz9VwlS9_6qA 6mmDgottk41BLpQcA40HdvV_6jAPqQ1EIL3_jLWl25oHeVEWOzTMhcERp4Jn-JZ4vP_n3w;info=< https://somedomain.com/stir-shaken/ec256-public.pem>;alg=ES256;ppt=shaken Server: kamailio (5.5.0 (x86_64/linux)) Content-Length: 0 I'm trying to get the "Identity" header with: $avp(identity_header) = $(hdr(Identity)); But It's coming up Any ideas of what I'm doing wrong? Regards, David Villasmil email: david.villasmil.w...@gmail.com phone: +34669448337 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Ignore redirect?
Many thanks Jeff! That’s exactly what I did :) Appreciate the help. David On Wed, 26 May 2021 at 11:36, Jeff Pyle wrote: > David, > > You catch the 302 in the failure_route, decide if it's what you want, and > if not just send to another route block for do_routing and such? > > > - Jeff > > On Tue, May 25, 2021 at 5:12 PM David Villasmil < > david.villasmil.w...@gmail.com> wrote: > >> Hello guys, >> >> I'm receiving an INVITE which I forward to a redirect server. When this >> redirect server's response is not what i need, i need to continue >> processing the call normally, i.e.: No forward to redirect. >> >> I'm able to forward to the redirect server and get the 302 properly. But >> when the 302 is not what I need, how can I continue processing the call >> normally? i.e. using do_routing. >> >> Thanks all, >> >> David Villasmil >> email: david.villasmil.w...@gmail.com >> phone: +34669448337 >> > ___ > 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 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Ignore redirect?
Hello guys, I'm receiving an INVITE which I forward to a redirect server. When this redirect server's response is not what i need, i need to continue processing the call normally, i.e.: No forward to redirect. I'm able to forward to the redirect server and get the 302 properly. But when the 302 is not what I need, how can I continue processing the call normally? i.e. using do_routing. Thanks all, David Villasmil email: david.villasmil.w...@gmail.com phone: +34669448337 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Doubt when receiving a CANCEL request in b2b mode
Dear all We are testing the module b2b in the devel version and when we receive a CANCEL request from A-side, we see that the 487 request terminated error is not sent from Opensips to client, and it relies on the 487 reception from B-side. If not received this causes a 408 to client after sometime. Would it be possible to generate a 487 from opensips to the client to terminate the cancelled transaction? We have tested this on top hiding default scenario, and using another custom scenario, but still the same result. Using b2b_send_reply(487, "Request Terminated"); b2b_delete_entity(); Would it work? This would continue processing the CANCEL request to the B-side normally? thanks a lot and regards David ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Help for Create Proxy Sip witch OpenSips
Hello Rafael, You might want to start by installing Opensips and going through the quick-start: https://www.opensips.org/Documentation/Tutorials-GettingStarted Once you've done that, and gone through the configuration file, you can ask more pointed questions. Regards, David Villasmil email: david.villasmil.w...@gmail.com phone: +34669448337 On Fri, Jan 29, 2021 at 11:31 AM Rafael Domingos wrote: > HI, > > > My name is Rafael and i have a multiple PBXIP in my structure. > I would like redirect traffic (register, options, invites etc..) > based proxy sip. I would lik remove NAT configurations, today I need creat > rules of redirect ports and IP's for PBX. > > I need this configuration: > I need only redirect sip connections for any asterisk. Exemplo: registre > UAC for domainA.com <http://domaina.com/> —->>> IP ASTeriskA domainB.com > <http://domainb.com/> —->>> IP ASTERISKB > > Anybody can help me? > > > Att.. > *Rafael Domingos /** 31-988485832* > *SKYPE: radib...@hotmail.com * > *EMAIL: rad...@gmail.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
Re: [OpenSIPS-Users] OpenSIPS 3.1 & NAT issues
Check out what IPs are offered in the SDPs in asterisk. Make sure they’re both public IPs. If you only have 1 asterisk, forwarding the rtp port range configured in asterisk from the firewall to asterisk should do it. On Thu, 14 Jan 2021 at 08:23, Mark Allen wrote: > Thanks Adrian > > The firewall has SIP-ALG disabled and just forwards ports from externally > to where they need to be internally - so ports 5060 and 1 - 65535 of > 46.x.x.x are mapped to 192.168.x.x (the OpenSIPS box) > > On Wed, 13 Jan 2021 at 17:32, Adrian Georgescu wrote: > >> Google search for SIP ALG problem to see if this is relevant for your >> case. >> >> Regards, >> Adrian >> >> >> On 13 Jan 2021, at 13:08, Mark Allen wrote: >> >> Hi all - I've been banging my head against this but not succeeding. >> >> Our setup... >> >> UAC 192.168.x.x >> | >> Router5.x.x.x >> | >> (internet) >> | >> Firewall 46.x.x.x maps >> | directly to >> OpenSIPS 192.168.x.x Mid-registrar >> | >> Asterisk 192.168.x.x >> >> >> Current situation: >> - UAC can register on Asterisk via OpenSIPS >> - UAC can call destination registered on Asterisk on local n/w to >> Asterisk box >> - Destination extension rings and can pick up call >> - There is no audio either way & call drops after about 30 secs (Asterisk >> kills call with "Requested channel not available" because not RTP >> traffic is reaching destination) >> >> I have tried passing audio through Mediaproxy on OpenSIPS box but with no >> success. Using Wireshark I can see RTP traffic initiated at both ends, but >> it doesn't reach the other end either way. >> >> Is there some definitive guide to setting this up correctly or are there >> specific steps that I need to follow? >> >> ___ >> Users mailing list >> Users@lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> ___ >> Users mailing list >> Users@lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > ___ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- Regards, David Villasmil email: david.villasmil.w...@gmail.com phone: +34669448337 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] mid_registrar load balance
You should be able to use the dispatcher module just before relaying. Have
you tried that?
On Fri, 13 Nov 2020 at 13:09, Andy Kama wrote:
> Hi Guys
>
> is it possible to load balance with mid_registrar?
> perhaps setting multiple $ru in this part
>
>if (is_method("REGISTER")) {
> mid_registrar_save("location");
> switch ($retcode) {
> case 1:
> xlog("forwarding REGISTER to main registrar ($$ci=$ci)\n");
> $ru = "sip:1.2.3.4:5060";
> t_relay();
> break;
> case 2:
> xlog("absorbing REGISTER! ($$ci=$ci)\n");
> break;
> default:
> xlog("failed to save registration! ($$ci=$ci)\n");
> }
>
> exit;
> }
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>
--
Regards,
David Villasmil
email: david.villasmil.w...@gmail.com
phone: +34669448337
___
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Re: [OpenSIPS-Users] store data locally
Db_text not being obsoleted: +1 On Wed, 23 Sep 2020 at 15:09, Ben Newlin wrote: > > > > > > > > > > > > > > > > > We use db_text and I agree with Ovidiu. We use it mostly for built-in > datastores on our instances. So we build the DB and then bake it onto our > AMIs before launch. The data is never modified during runtime, so we don’t > need full SQL functionality. > > Using db_text removes dependencies on sqlite in both the build and runtime > environments. We use Ansible/Jinja2 to template the db_text databases from > text files. It’s very straightforward. > > > > > > So if you need SQL features then sqlite is great, but db_text is a very > fast, easy way to get a DB running with no overhead. > > > > > > > > Ben Newlin > > > > > > > > > > > > *From:*Users > > > *Date: *Wednesday, September 23, 2020 at 9:50 AM > > > *To: *OpenSIPS users mailling list > > > *Subject: *Re: [OpenSIPS-Users] store data locally > > > > > > > Hello Liviu, > > > > > > SQLite is the way to go if you want something standard. > > > But having the ability to just use a text editor to add/remove/modify > > > records in db_text is golden for me. > > > I used db_text a lot on embedded platforms. > > > I would vote against obsolete-ing db_text. > > > For me, it's easier to use db_text instead of sqlite. For others, it > > > might be the other way around. > > > > > > Regards, > > > Ovidiu Sas > > > > > > On Wed, Sep 23, 2020 at 1:55 AM Liviu Chircu wrote: > > > > > > > > On 22.09.2020 18:42, Ovidiu Sas wrote: > > > > > If you don't want to run a full blown db, then you can use db_text > > > > > without cacheDB. > > > > > The data is cached into memory at startup. If you update the text > > > > > file, you can re-cache the data [1]. > > > > > > > > Hi, Ovidiu! > > > > > > > > May I segue into discussing some questions that have been puzzling me > > > > for a while now? > > > > > > > > "In 2020, why should developers choose DB TEXT over DB SQLITE? > > > > Shouldn't DB TEXT be obsoleted?" > > > > > > > > As far as history goes, it seems DB TEXT was created by Daniel in 2003. > > > > Which, in my opinion, even for that time, it seems like a "reinvent the > > > > wheel" kind of effort, since SQLite had already been GA'ed for almost 3 > > > > years [1]. > > > > > > > > In order to come up with an answer, we can break down my original > questions: > > > > > > > > * DB TEXT and SQLite both aim to be lightweight, serverless, file-based > > > > SQL databases with RAM caching. True or false? > > > > * does DB TEXT have any features that SQLite doesn't? > > > > * does DB TEXT support a richer SQL syntax than SQLite? > > > > * does DB TEXT have less bugs than SQLite? > > > > * does DB TEXT handle in-memory caching better than SQLite? > > > > * does DB TEXT handle disk files better than SQLite? > > > > > > > > Thank you in advance for the discussion! > > > > > > > > Best regards, > > > > > > > > > [1]: > > https://protect-us.mimecast.com/s/hqFICG6Q9Nir8YGJTKiBEi?domain=sqlite.org > <https://sqlite.org/changes.html> > > > > > > > > -- > > > > Liviu Chircu > > > > www.twitter.com/liviuchircu | > > www.opensips-solutions.com > > > > > > > > > > > > ___ > > > > Users mailing list > > > > Users@lists.opensips.org > > > > > > > https://protect-us.mimecast.com/s/QQVvCJ67Pkiknr58cGCME1?domain=lists.opensips.org > <http://lists.opensips.org/cgi-bin/mailman/listinfo/users> > > > > > > > > > > > > > > -- > > > VoIP Embedded, Inc. > > > http://www.voipembedded.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 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] opensips UDP retransmit on load
There's a lot of info here which applies to everything, really. https://freeswitch.org/confluence/display/FREESWITCH/Performance+Testing+and+Configurations Regards, David Villasmil email: david.villasmil.w...@gmail.com phone: +34669448337 On Tue, Sep 22, 2020 at 3:36 PM Kirill Galinurov wrote: > SIP UDP Interface #1 (udp:10.169.115.19:5073) > Receive Queue: 4.8KB > Avg. CPU usage: 0% (last 1 sec) > > Process 6 load: 12%, 10%, 3% (SIP receiver udp:10.169.115.19:5073) > Process 7 load: 13%, 10%, 4% (SIP receiver udp:10.169.115.19:5073) > Process 8 load: 13%, 10%, 4% (SIP receiver udp:10.169.115.19:5073) > Process 9 load: 15%, 13%, 5% (SIP receiver udp:10.169.115.19:5073) > Process 10 load: 13%, 12%, 4% (SIP receiver udp:10.169.115.19:5073) > Process 11 load: 12%, 11%, 4% (SIP receiver udp:10.169.115.19:5073) > Process 12 load: 11%, 10%, 4% (SIP receiver udp:10.169.115.19:5073) > Process 13 load: 18%, 13%, 6% (SIP receiver udp:10.169.115.19:5073) > > WARNING: the receive queue is NOT empty, SIP signaling may be slower! > > How to check why Receive Queue not null ? > ___ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Fine tuning high CPS and msyql queries
Keep in mundo the don’t make Ubuntu for SIP applications, which have their own idiosyncrasies. They make it general purpose. So finding a value that doesn’t work perfectly with what you need for this very specific application, is not a big deal. On Sat, 13 Jun 2020 at 00:38, Calvin Ellison wrote: > I doubt the system will be using all of that buffer. I also don't know if > the issue was in the receive buffer or send buffer since I changed both at > once. Many resources are available online from people who have already done > much more scientific testing that indicate the default values should be > increased for certain applications, which is the reason I changed it to > begin with. There's no one-size-fits all for server configurations, and > what works for this UDP application with a small number of clients might > not work well for a different application with many TCS connections. > > "absolutely terrible" may be too strong of a way to put it, but that the > before and after don't lie. > > On Fri, Jun 12, 2020 at 4:02 PM Alex Balashov > wrote: > >> But increasing the depth of the queue by 78x (if I'm not mistaken, >> 212992 is the default--at least, it is on all my CentOS 7.x and 8.x >> >> ___ > 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 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Fine tuning high CPS and msyql queries
Basically, the application is not processing the received packets as quickly as it should, so the kernel stores the packets in the buffer so it doesn’t have to throw them away. It’s not so difficult to understand. If this is happening all the time, you won’t solve this by making the buffer bigger. You solve this by figuring out why the application is not processing the packets fast enough. On Sat, 13 Jun 2020 at 00:28, Alex Balashov wrote: > On 6/12/20 7:20 PM, Calvin Ellison wrote: > > > I think the important point here is that the receive buffers are used to > > hold received data until it is read by the application. In fact, too > > small of a receive buffer would cause packets to be discarded outright, > > regardless of how fast the application can respond. Not knowing how > > large of a buffer is needed was the problem, not the raw processing > > power. It doesn't matter how fast I can eat if the server only has very > > small plates to bring the food every trip from the kitchen. > > In absolute terms, this is true. But if your kitchen is putting out so > much food that not even ~200,000 plates "in flight" will do, you've got > a bigger problem to address and adding more plates is just papering it > over. > > Monitor your receive queue scrupulously at a very high timing > resolution. If you found default values for rmem_max to be "absolutely > terrible", that means the backlog was increasing monotonically until you > ran out of space. If you increase the queue depth, you will be able to > prolong this effect for a while. > > The kernel's packet queue is a backstop--an emergency release valve, not > a main thoroughfare. It's there to help you deal with ephemeral > congestion caused by things like periodic big-lock background process > contention, scheduler hiccups, disk controller patrol reads, etc. But > the base load should result in a long-run queue backlog of zero. > Applications which properly cope with their workload don't cause > non-trivial packet or connection queueing on the OS side. > > -- Alex > > -- > 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/ > > ___ > 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 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Write acc_extra data to separate table
You could use https://opensips.org/html/docs/modules/1.7.x/avpops.html#id293960 Regards, David Villasmil email: david.villasmil.w...@gmail.com phone: +34669448337 On Thu, Jun 11, 2020 at 12:07 PM Mark Farmer wrote: > Hmm... Doesn't seem to exist in OpenSIPS. > > How do others do this kind of thing? > > > > On Thu, 11 Jun 2020 at 11:58, David Villasmil < > david.villasmil.w...@gmail.com> wrote: > >> You’re going to need to use sqlops for that, I think. >> >> On Thu, 11 Jun 2020 at 11:39, Mark Farmer wrote: >> >>> Thanks for the reply Johan, I'm not sure that's quite what I'm looking >>> for. >>> I'd like the usual acc data to be written to the normal acc table but my >>> extra data written to a different table, preferably in a different database. >>> >>> Many thanks >>> Mark. >>> >>> >>> On Thu, 11 Jun 2020 at 11:32, Johan De Clercq wrote: >>> >>>> If I am not wrong, you can specify the accounting table and db in the >>>> module params. >>>> Hence make a copy of the table, insert in a new db, add your extra >>>> column and adapt the module parameters. >>>> >>>> wkr, >>>> >>>> Op do 11 jun. 2020 om 12:18 schreef Mark Farmer : >>>> >>>>> Hi everyone >>>>> >>>>> I have a couple of extra fields in my acc database table which is >>>>> added by acc_extra. This causes issues when upgrading. >>>>> I would prefer to have that data stored in a different database/table. >>>>> >>>>> Is there a nice way to do that? >>>>> OpenSIPS 3.0 in this case. >>>>> >>>>> Many thanks >>>>> Mark. >>>>> >>>>> ___ >>>>> 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 >>>> >>> >>> >>> -- >>> Mark Farmer >>> farm...@gmail.com >>> ___ >>> 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 >> ___ >> Users mailing list >> Users@lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > > > -- > Mark Farmer > farm...@gmail.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
Re: [OpenSIPS-Users] Write acc_extra data to separate table
You’re going to need to use sqlops for that, I think. On Thu, 11 Jun 2020 at 11:39, Mark Farmer wrote: > Thanks for the reply Johan, I'm not sure that's quite what I'm looking for. > I'd like the usual acc data to be written to the normal acc table but my > extra data written to a different table, preferably in a different database. > > Many thanks > Mark. > > > On Thu, 11 Jun 2020 at 11:32, Johan De Clercq wrote: > >> If I am not wrong, you can specify the accounting table and db in the >> module params. >> Hence make a copy of the table, insert in a new db, add your extra column >> and adapt the module parameters. >> >> wkr, >> >> Op do 11 jun. 2020 om 12:18 schreef Mark Farmer : >> >>> Hi everyone >>> >>> I have a couple of extra fields in my acc database table which is >>> added by acc_extra. This causes issues when upgrading. >>> I would prefer to have that data stored in a different database/table. >>> >>> Is there a nice way to do that? >>> OpenSIPS 3.0 in this case. >>> >>> Many thanks >>> Mark. >>> >>> ___ >>> 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 >> > > > -- > Mark Farmer > farm...@gmail.com > ___ > 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 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Maybe it's a bug
+1 Always ALWAYS round up, never down (in terms of ms to secs) +1 Also do your billing were your application is, not on a proxy. On Tue, 9 Jun 2020 at 19:58, Liviu Chircu wrote: > On 09.06.2020 15:43, Saint Michael wrote: > > I talked to Vlad, who I believe wrote the code, and he does not think > > it is a bug and I should use the ms and not the seconds. But thousands > > of businessmen will not spot this and thus their billing will never > > match the carrier, and they will lose money. If anybody thinks for a > > second that a call with a 200 OK will be free, is dreaming. Not in > > America. > > Hi, SM! > > Opinion #1: I doubt that anyone who is serious about their billing & > revenue (e.g. your nitpicky carrier) would leave to randomness the > answer to the most basic question of: "does our platform correctly bill > each call?". No disrespect here, just maybe highlighting the fact that > your platform could benefit from a bit more testing. > > Opinion #2: we could definitely change the default of the > second-accurate precision to be _greedy_ instead of _generous_. I bet > most people (myself included) would be more happy with a ceil() [1] > behavior instead of a trunc() [2] one. That is: round _upwards_, not > _downwards_. More opinions would be useful here! > > Best regards, > > [1]: man ceil > [2]: man trunc > > -- > Liviu Chircu > www.twitter.com/liviuchircu | www.opensips-solutions.com > > > ___ > 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 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Fine tuning high CPS and msyql queries
No idea, but can always check with any of several utilities, I.e.: netstat On Fri, 5 Jun 2020 at 01:37, Calvin Ellison wrote: > On Thu, Jun 4, 2020 at 5:18 PM David Villasmil > wrote: > > > > Maybe you are hitting the max connections? How many connections are > there when it starts to show those errors? > > I'd definitely benefit from a monitor on this. Is this available from > within opensips? > > ___ > 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 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Fine tuning high CPS and msyql queries
ntermittent database connection errors. > We'll see what raising the max connections and ulimits on the server > does. I've also backed off on children and increased the async > connection pool size to result in the same number of total maximum > connections. Presumably this will reduce context switches and timer > delays. > > > E) Are your memcached processes using heavy cpu? If you are caching > multiple lists, I've found it helps to use unique memcached instance per > list. > > All of the various SIP dips are the same db stored procedure with many > fields in the response. Those fields are cached as a CSV string, so > any cached dip can be used by any other kind of dip. The same call is > likely to use multiple dips, so we should only hit the DB once per > call regardless of how many different dips we apply. > > > F) Look for memory related log messages. If the memory starts getting > exhausted you will see defrag messages. This will chew up available > computation cycles. > > Both opensips servers and the database have plenty of free memory. How > do I know how much shared and process memory to use? I see warnings > about the reactor size shrinking to a percentage of the process memory > but have no idea what that implies. > > ___ > 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 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Load testing
Sippy is really cool. Though I’ve heard sipp‘s media stack starts behaving funny on big load tests... I personally use freeswitch depending on what I’m trying to test (scenarios, etc) On Tue, 12 May 2020 at 09:27, Tomi Hakkarainen wrote: > Thanks, did not know of its existence before this :) > > Br, Tomi > > On 12. May 2020, at 11.14, Callum Guy wrote: > > Sippy cup has some good media generation capabilities, still using sipp > under the hood. > > https://mojolingo.github.io/sippy_cup/ > > > > On Mon, 11 May 2020 at 18:45, Tomi Hakkarainen wrote: > >> I agree >> >> BR, Tomi >> >> On 11. May 2020, at 19.48, johan wrote: >> >> hmmm sipp with your own rtp files. >> On 11/05/2020 18:19, miha- via Users wrote: >> >> Hi >> >> What is best tool for load testing that can generate also RTP? >> >> Tnx >> >> miha >> >> ___ >> Users mailing >> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> ___ >> Users mailing list >> Users@lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> ___ >> Users mailing list >> Users@lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > > <https://www.x-on.co.uk/service/surgery-connect/coronavirus.htm> > > *0333 332 | x-on.co.uk <https://www.x-on.co.uk/> | ** > <https://www.linkedin.com/company/x-on> <https://www.facebook.com/XonTel> > <https://twitter.com/xonuk> ** | Coronavirus > <https://www.x-on.co.uk/service/surgery-connect/coronavirus.htm>* > > THE ITSPA AWARDS 2020 <http://www.itspa.org.uk/itspa-awards> AND Best > ITSP - Mid Market, Best Software and Best Vertical Solution are trade marks > of the Internet Telephony Services Providers' Association, used under > licence. > > X-on is a trading name of Storacall Technology Ltd a limited company > registered in England and Wales. > Registered Office : Avaland House, 110 London Road, Apsley, Hemel > Hempstead, Herts, HP3 9SD > <https://www.google.com/maps/search/110+London+Road,+Apsley,+Hemel+Hempstead,%0D%0AHerts,+HP3+9SD?entry=gmail=g>. > Company Registration No. 2578478. > The information in this e-mail is confidential and for use by the > addressee(s) only. If you are not the intended recipient, please notify > X-on immediately on +44(0)333 332 and delete the > message from your computer. If you are not a named addressee you must not > use, disclose, disseminate, distribute, copy, print or reply to this email. > Views > or opinions expressed by an individual > within this email may not necessarily reflect the views of X-on or its > associated companies. Although X-on routinely screens for viruses, > addressees should scan this email and any attachments > for viruses. X-on makes no representation or warranty as to the absence of > viruses in this email or any attachments. > > ___ > 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 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Dockerize OpenSIPS
They all get their dialplan/config from an API backend. They’re DB agnostic. On Sat, 2 May 2020 at 13:28, H Yavari wrote: > Thank you David, > Interested to know, you are running a cluster/group of FS with centralized > DB, with same functionality? > > > Regards, > HYavari > > > On Saturday, May 2, 2020, 4:54:01 PM GMT+4:30, David Villasmil < > david.villasmil.w...@gmail.com> wrote: > > > Hello, > > I have never run into those problems. The people I’ve seen running into > them are doing thousands upon thousands of packets por seconds. Hardware is > cheap nowadays, I just spread the load. > > I wouldn’t dockerize rtpengine, although I know people who do. > > In a normal setup probably you won’t run into them, we run freeswitch on > containers for production without a hitch (I must stress I was very > reluctant to do that a few years ago, but docker has come a long way since > then). > > David > > On Sat, 2 May 2020 at 12:42, H Yavari wrote: > > Thank you David. > > What do you think about networking concerns? you mentioned to them but I > didn't get your point. > RTP restrictions, port proxy, iptables, fail2ban are top ones. > > PS: No matter which telephony platform (Asterisk/FS/OpenSIPS/Kamailio), > these concerns are in general. > PS: We have more challenges with projects like Freepbx. > > > Regards, > HY > > > > On Saturday, May 2, 2020, 3:48:11 PM GMT+4:30, David Villasmil < > david.villasmil.w...@gmail.com> wrote: > > > Not sure about OpenSIPS specifically, but I would assume it has been > implemented in docker just as much as kamailio and freeSWITCH/Asterisk. > > This is done all over the world. Docker is not an emulator or a virtual > machine host. When you run something on docker, its speed is (almost) > exactly the same as running it on the host itself, since there’s no OS > overhead, it works by separating processes via Cgroups, no by virtualizing > or emulating hardware. > > Quote: > > The Docker technology uses the Linux kernel > <https://www.redhat.com/en/topics/linux/what-is-the-linux-kernel> and > features of the kernel, like Cgroups > <https://access.redhat.com/documentation/en-US/Red_Hat_Enterprise_Linux/6/html/Resource_Management_Guide/ch01.html> > and namespaces <https://lwn.net/Articles/528078/>, to segregate > processes so they can run independently. This independence is the intention > of containers‐the ability to run multiple processes and apps separately > from one another to make better use of your infrastructure while retaining > the security <https://www.redhat.com/en/topics/security>you would have > with separate systems. > > So in simple terms, docker simply separates processes. > > There ARE, nonetheless, some problems with dockerizing everything. I have > read issues like If the network traffic is way way way too high, you may > encounter issues like dropped packets, etc. but this is a problem on the > networking side, I.e: the iptables rules. Also the natting related to using > docker can be cumbersome, but once you’re over that, it’s home free. > > So, as long as you manage your infrastructure well, you shouldn’t have > problems. > > In terms of troubleshooting a failing container. All logging should be > sent to some log server, and you can do your troubleshooting there. Also, > don’t kill a failing container so you can access it (via ssh or attach or > exec) and troubleshoot it. > > The pros of using docker/k8s greatly outweighs the cons, in my opinion. > > Hope this help. > > David > > On Sat, 2 May 2020 at 11:33, H Yavari via Users > wrote: > > Thank you Johan, > > When your infrastructure goes to run with k8s or other same platforms, > it's hard to make some exceptions. > Also softwares like opensips that are working just with DB, can run very > smoothly. > > Although I haven't seen any problem yet after moving it to containers, but > I am interested in hearing from others and developers team. > > > Regards, > HY > > > > On Saturday, May 2, 2020, 12:51:51 PM GMT+4:30, johan > wrote: > > > First of all, I am not aware of a production kubernetes cluster. > > Using containers has advantages : fast install, easy to move. The > annoying thing is that if it goes wrong, it is not easy to troubleshoot. > Secondly, you add an extra abstraction layer, abstraction (most of the > time) reduces speed and decreases capacity. > > In short : it all depends on the size of your system. In ip4 I don't see > the advantage. What could be a nice scalable system, is to deploy on ip6 > with anycast. > > Just my thoughts ... > On 2/05/2
Re: [OpenSIPS-Users] Dockerize OpenSIPS
Hello, I have never run into those problems. The people I’ve seen running into them are doing thousands upon thousands of packets por seconds. Hardware is cheap nowadays, I just spread the load. I wouldn’t dockerize rtpengine, although I know people who do. In a normal setup probably you won’t run into them, we run freeswitch on containers for production without a hitch (I must stress I was very reluctant to do that a few years ago, but docker has come a long way since then). David On Sat, 2 May 2020 at 12:42, H Yavari wrote: > Thank you David. > > What do you think about networking concerns? you mentioned to them but I > didn't get your point. > RTP restrictions, port proxy, iptables, fail2ban are top ones. > > PS: No matter which telephony platform (Asterisk/FS/OpenSIPS/Kamailio), > these concerns are in general. > PS: We have more challenges with projects like Freepbx. > > > Regards, > HY > > > > On Saturday, May 2, 2020, 3:48:11 PM GMT+4:30, David Villasmil < > david.villasmil.w...@gmail.com> wrote: > > > Not sure about OpenSIPS specifically, but I would assume it has been > implemented in docker just as much as kamailio and freeSWITCH/Asterisk. > > This is done all over the world. Docker is not an emulator or a virtual > machine host. When you run something on docker, its speed is (almost) > exactly the same as running it on the host itself, since there’s no OS > overhead, it works by separating processes via Cgroups, no by virtualizing > or emulating hardware. > > Quote: > > The Docker technology uses the Linux kernel > <https://www.redhat.com/en/topics/linux/what-is-the-linux-kernel> and > features of the kernel, like Cgroups > <https://access.redhat.com/documentation/en-US/Red_Hat_Enterprise_Linux/6/html/Resource_Management_Guide/ch01.html> > and namespaces <https://lwn.net/Articles/528078/>, to segregate > processes so they can run independently. This independence is the intention > of containers‐the ability to run multiple processes and apps separately > from one another to make better use of your infrastructure while retaining > the security <https://www.redhat.com/en/topics/security>you would have > with separate systems. > > So in simple terms, docker simply separates processes. > > There ARE, nonetheless, some problems with dockerizing everything. I have > read issues like If the network traffic is way way way too high, you may > encounter issues like dropped packets, etc. but this is a problem on the > networking side, I.e: the iptables rules. Also the natting related to using > docker can be cumbersome, but once you’re over that, it’s home free. > > So, as long as you manage your infrastructure well, you shouldn’t have > problems. > > In terms of troubleshooting a failing container. All logging should be > sent to some log server, and you can do your troubleshooting there. Also, > don’t kill a failing container so you can access it (via ssh or attach or > exec) and troubleshoot it. > > The pros of using docker/k8s greatly outweighs the cons, in my opinion. > > Hope this help. > > David > > On Sat, 2 May 2020 at 11:33, H Yavari via Users > wrote: > > Thank you Johan, > > When your infrastructure goes to run with k8s or other same platforms, > it's hard to make some exceptions. > Also softwares like opensips that are working just with DB, can run very > smoothly. > > Although I haven't seen any problem yet after moving it to containers, but > I am interested in hearing from others and developers team. > > > Regards, > HY > > > > On Saturday, May 2, 2020, 12:51:51 PM GMT+4:30, johan > wrote: > > > First of all, I am not aware of a production kubernetes cluster. > > Using containers has advantages : fast install, easy to move. The > annoying thing is that if it goes wrong, it is not easy to troubleshoot. > Secondly, you add an extra abstraction layer, abstraction (most of the > time) reduces speed and decreases capacity. > > In short : it all depends on the size of your system. In ip4 I don't see > the advantage. What could be a nice scalable system, is to deploy on ip6 > with anycast. > > Just my thoughts ... > On 2/05/2020 07:49, H Yavari via Users wrote: > > Hi to all, > > As you know docker and K8s, are growing quickly. So we dockerized Asterisk > and OpenSIPS also. > But I see some community members are against it. They have some reasons > like NAT, RTP ports and performance. > > Do you agree with them ? > Is there any successful large scale OpenSIPS cluster based on K8s ? > > > Thanks for sharing your experiences. > > > Regards, > HY > > ___ &
Re: [OpenSIPS-Users] Dockerize OpenSIPS
Not sure about OpenSIPS specifically, but I would assume it has been implemented in docker just as much as kamailio and freeSWITCH/Asterisk. This is done all over the world. Docker is not an emulator or a virtual machine host. When you run something on docker, its speed is (almost) exactly the same as running it on the host itself, since there’s no OS overhead, it works by separating processes via Cgroups, no by virtualizing or emulating hardware. Quote: The Docker technology uses the Linux kernel <https://www.redhat.com/en/topics/linux/what-is-the-linux-kernel> and features of the kernel, like Cgroups <https://access.redhat.com/documentation/en-US/Red_Hat_Enterprise_Linux/6/html/Resource_Management_Guide/ch01.html> and namespaces <https://lwn.net/Articles/528078/>, to segregate processes so they can run independently. This independence is the intention of containers‐the ability to run multiple processes and apps separately from one another to make better use of your infrastructure while retaining the security <https://www.redhat.com/en/topics/security>you would have with separate systems. So in simple terms, docker simply separates processes. There ARE, nonetheless, some problems with dockerizing everything. I have read issues like If the network traffic is way way way too high, you may encounter issues like dropped packets, etc. but this is a problem on the networking side, I.e: the iptables rules. Also the natting related to using docker can be cumbersome, but once you’re over that, it’s home free. So, as long as you manage your infrastructure well, you shouldn’t have problems. In terms of troubleshooting a failing container. All logging should be sent to some log server, and you can do your troubleshooting there. Also, don’t kill a failing container so you can access it (via ssh or attach or exec) and troubleshoot it. The pros of using docker/k8s greatly outweighs the cons, in my opinion. Hope this help. David On Sat, 2 May 2020 at 11:33, H Yavari via Users wrote: > Thank you Johan, > > When your infrastructure goes to run with k8s or other same platforms, > it's hard to make some exceptions. > Also softwares like opensips that are working just with DB, can run very > smoothly. > > Although I haven't seen any problem yet after moving it to containers, but > I am interested in hearing from others and developers team. > > > Regards, > HY > > > > On Saturday, May 2, 2020, 12:51:51 PM GMT+4:30, johan > wrote: > > > First of all, I am not aware of a production kubernetes cluster. > > Using containers has advantages : fast install, easy to move. The > annoying thing is that if it goes wrong, it is not easy to troubleshoot. > Secondly, you add an extra abstraction layer, abstraction (most of the > time) reduces speed and decreases capacity. > > In short : it all depends on the size of your system. In ip4 I don't see > the advantage. What could be a nice scalable system, is to deploy on ip6 > with anycast. > > Just my thoughts ... > On 2/05/2020 07:49, H Yavari via Users wrote: > > Hi to all, > > As you know docker and K8s, are growing quickly. So we dockerized Asterisk > and OpenSIPS also. > But I see some community members are against it. They have some reasons > like NAT, RTP ports and performance. > > Do you agree with them ? > Is there any successful large scale OpenSIPS cluster based on K8s ? > > > Thanks for sharing your experiences. > > > Regards, > HY > > ___ > Users mailing > listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > ___ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > ___ > 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 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Messages
Have you tried with 1 or 0? Regards, David Villasmil email: david.villasmil.w...@gmail.com phone: +34669448337 On Wed, Apr 15, 2020 at 12:09 AM Saint Michael wrote: > what is the lowest debug level that is safe to use? > > > On Tue, Apr 14, 2020 at 6:59 PM David Villasmil < > david.villasmil.w...@gmail.com> wrote: > >> Maybe lower the debug? >> >> On Tue, 14 Apr 2020 at 23:00, Saint Michael wrote: >> >>> I see thousands of the messages below. If they are not important, how do >>> I hide them >>> ARNING:dialog:dlg_onroute: tight matching failed for ACK with >>> callid='5063212c34ac27e23d25ea143ca8bf4f@64.140.166.100:5060'/52, >>> ftag='as480e6ada'/10, ttag='8e2e-a7407c352db2ab880c67b8cc744fbbfe'/37 and >>> direction=1 >>> Apr 14 21:56:06 brian opensips[22702]: Apr 14 21:56:06 [22787] >>> WARNING:dialog:dlg_onroute: dialog identification elements are >>> callid='5063212c34ac27e23d25ea143ca8bf4f@64.140.166.100:5060'/52, >>> caller tag='as480e6ada'/10, callee tag='KSQ5DeD148vBD'/13 >>> ___ >>> 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 >> ___ >> Users mailing list >> Users@lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > ___ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Messages
Maybe lower the debug? On Tue, 14 Apr 2020 at 23:00, Saint Michael wrote: > I see thousands of the messages below. If they are not important, how do I > hide them > ARNING:dialog:dlg_onroute: tight matching failed for ACK with > callid='5063212c34ac27e23d25ea143ca8bf4f@64.140.166.100:5060'/52, > ftag='as480e6ada'/10, ttag='8e2e-a7407c352db2ab880c67b8cc744fbbfe'/37 and > direction=1 > Apr 14 21:56:06 brian opensips[22702]: Apr 14 21:56:06 [22787] > WARNING:dialog:dlg_onroute: dialog identification elements are > callid='5063212c34ac27e23d25ea143ca8bf4f@64.140.166.100:5060'/52, caller > tag='as480e6ada'/10, callee tag='KSQ5DeD148vBD'/13 > ___ > 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 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] failed to load module siptrace.so
I just made a fresh install on a docker and indeed siptrace is not there...
going to compile it
Regards,
David Villasmil
email: david.villasmil.w...@gmail.com
phone: +34669448337
On Tue, Apr 7, 2020 at 10:41 AM David Villasmil <
david.villasmil.w...@gmail.com> wrote:
> or install?
>
> Regards,
>
> David Villasmil
> email: david.villasmil.w...@gmail.com
> phone: +34669448337
>
>
> On Tue, Apr 7, 2020 at 10:39 AM David Villasmil <
> david.villasmil.w...@gmail.com> wrote:
>
>> did you compile sipcapture?
>>
>> Regards,
>>
>> David Villasmil
>> email: david.villasmil.w...@gmail.com
>> phone: +34669448337
>>
>>
>> On Tue, Apr 7, 2020 at 10:32 AM Aleksandar Sosic wrote:
>>
>>> Hi Guys,
>>>
>>> I'm installing Opensips v3.0 via `deb http://apt.opensips.org buster
>>> 3.0-releases` with:
>>> ```
>>> apt-get install -y opensips opensips-json-module
>>> opensips-restclient-module opensips-http-modules
>>> ```
>>>
>>> When specifying in the conf `loadmodule "siptrace.so"` upon running
>>> opensips I get this error:
>>> ```
>>> Apr 7 09:21:49 [333] CRITICAL:core:yyerror: parse error in
>>> /etc/opensips/opensips.cfg:38:13-14: failed to load module siptrace.so
>>> ```
>>>
>>> The file indeed is not present in
>>> `/usr/lib/x86_64-linux-gnu/opensips/modules/`.
>>> I do have `sipcapture.so` and `proto_hep.so` although but no siptrace.
>>> What am I missing here?
>>>
>>> I'm trying to send all to a HEP agent like I do in kamailio with:
>>> ```
>>> loadmodule "siptrace.so"
>>>
>>> modparam("siptrace", "trace_on", 1)
>>> modparam("siptrace", "trace_to_database", 0)
>>> modparam("siptrace", "hep_mode_on", 1)
>>> modparam("siptrace", "hep_version", 3)
>>> modparam("siptrace", "hep_capture_id", 1)
>>>
>>> request_route {
>>> sip_trace("hep-agent.local", "$ci");
>>> ...
>>> }
>>> ```
>>>
>>> Any ideas or examples of how to do this with Opensips v3?
>>>
>>> Thanks,
>>> --
>>> Alex
>>>
>>> ___
>>> Users mailing list
>>> Users@lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] failed to load module siptrace.so
or install?
Regards,
David Villasmil
email: david.villasmil.w...@gmail.com
phone: +34669448337
On Tue, Apr 7, 2020 at 10:39 AM David Villasmil <
david.villasmil.w...@gmail.com> wrote:
> did you compile sipcapture?
>
> Regards,
>
> David Villasmil
> email: david.villasmil.w...@gmail.com
> phone: +34669448337
>
>
> On Tue, Apr 7, 2020 at 10:32 AM Aleksandar Sosic wrote:
>
>> Hi Guys,
>>
>> I'm installing Opensips v3.0 via `deb http://apt.opensips.org buster
>> 3.0-releases` with:
>> ```
>> apt-get install -y opensips opensips-json-module
>> opensips-restclient-module opensips-http-modules
>> ```
>>
>> When specifying in the conf `loadmodule "siptrace.so"` upon running
>> opensips I get this error:
>> ```
>> Apr 7 09:21:49 [333] CRITICAL:core:yyerror: parse error in
>> /etc/opensips/opensips.cfg:38:13-14: failed to load module siptrace.so
>> ```
>>
>> The file indeed is not present in
>> `/usr/lib/x86_64-linux-gnu/opensips/modules/`.
>> I do have `sipcapture.so` and `proto_hep.so` although but no siptrace.
>> What am I missing here?
>>
>> I'm trying to send all to a HEP agent like I do in kamailio with:
>> ```
>> loadmodule "siptrace.so"
>>
>> modparam("siptrace", "trace_on", 1)
>> modparam("siptrace", "trace_to_database", 0)
>> modparam("siptrace", "hep_mode_on", 1)
>> modparam("siptrace", "hep_version", 3)
>> modparam("siptrace", "hep_capture_id", 1)
>>
>> request_route {
>> sip_trace("hep-agent.local", "$ci");
>> ...
>> }
>> ```
>>
>> Any ideas or examples of how to do this with Opensips v3?
>>
>> Thanks,
>> --
>> Alex
>>
>> ___
>> Users mailing list
>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] failed to load module siptrace.so
did you compile sipcapture?
Regards,
David Villasmil
email: david.villasmil.w...@gmail.com
phone: +34669448337
On Tue, Apr 7, 2020 at 10:32 AM Aleksandar Sosic wrote:
> Hi Guys,
>
> I'm installing Opensips v3.0 via `deb http://apt.opensips.org buster
> 3.0-releases` with:
> ```
> apt-get install -y opensips opensips-json-module
> opensips-restclient-module opensips-http-modules
> ```
>
> When specifying in the conf `loadmodule "siptrace.so"` upon running
> opensips I get this error:
> ```
> Apr 7 09:21:49 [333] CRITICAL:core:yyerror: parse error in
> /etc/opensips/opensips.cfg:38:13-14: failed to load module siptrace.so
> ```
>
> The file indeed is not present in
> `/usr/lib/x86_64-linux-gnu/opensips/modules/`.
> I do have `sipcapture.so` and `proto_hep.so` although but no siptrace.
> What am I missing here?
>
> I'm trying to send all to a HEP agent like I do in kamailio with:
> ```
> loadmodule "siptrace.so"
>
> modparam("siptrace", "trace_on", 1)
> modparam("siptrace", "trace_to_database", 0)
> modparam("siptrace", "hep_mode_on", 1)
> modparam("siptrace", "hep_version", 3)
> modparam("siptrace", "hep_capture_id", 1)
>
> request_route {
> sip_trace("hep-agent.local", "$ci");
> ...
> }
> ```
>
> Any ideas or examples of how to do this with Opensips v3?
>
> Thanks,
> --
> Alex
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Access to listen/advertised IP Addresses
this is where the advertised address is set https://github.com/OpenSIPS/opensips/blob/628a126fe3523e800440855f10f9841d6a2c39eb/cfg.y#L2373 Regards, David Villasmil email: david.villasmil.w...@gmail.com phone: +34669448337 On Mon, Apr 6, 2020 at 4:12 PM David Villasmil < david.villasmil.w...@gmail.com> wrote: > i only see $rd which is the domain to which the sip message was sent, it > "should" have the advertised ip, or de actual domain, in which case if you > need the actual ip, it is useless > Regards, > > David Villasmil > email: david.villasmil.w...@gmail.com > phone: +34669448337 > > > On Mon, Apr 6, 2020 at 4:00 PM Johan De Clercq wrote: > >> It,s not exposed I think. I can’t find it back either >> >> Outlook voor iOS <https://aka.ms/o0ukef> downloaden >> -- >> *Van:* Users namens David Villasmil < >> david.villasmil.w...@gmail.com> >> *Verzonden:* Monday, April 6, 2020 4:49:36 PM >> *Aan:* OpenSIPS users mailling list >> *Onderwerp:* Re: [OpenSIPS-Users] Access to listen/advertised IP >> Addresses >> >> No, you’re right. It’s not in the core variables and I can’t find it >> either. Which makes me think it’s either not exposed or somewhere in a >> module (it’s not in proto_udp) >> >> I will research a little to try and find it.. >> >> On Mon, 6 Apr 2020 at 14:04, Mark Farmer wrote: >> >> Thanks David. But I see no reference to the same variable in OpenSIPS. >> >> https://www.opensips.org/Documentation/Script-CoreVar-2-4 >> >> Am I missing something? >> >> >> On Mon, 6 Apr 2020 at 13:45, David Villasmil < >> david.villasmil.w...@gmail.com> wrote: >> >> Right here: >> >> >> https://www.kamailio.org/wiki/cookbooks/5.2.x/pseudovariables#rai_-_received_advertised_ip_address >> >> Regards, >> >> David Villasmil >> email: david.villasmil.w...@gmail.com >> phone: +34669448337 >> >> >> On Mon, Apr 6, 2020 at 12:08 PM Mark Farmer wrote: >> >> Many thanks for the reply. >> >> $Ri is certainly useful when the request comes from a non-natted >> interface. Thanks for pointing that out :) >> >> Is there a way to reference the advertised IP address defined in the >> listen statement? >> >> listen=udp:xxx.xxx.xxx.xxx:5060 as xxx.xxx.xxx.xxx:5060 >> >> Thanks >> Mark. >> >> >> On Thu, 2 Apr 2020 at 17:32, Sharad Kumar via Users < >> users@lists.opensips.org> wrote: >> >> Hi Mark, >> >> If your initial goal is to get the interface IP where request is >> received then you can try these variables. >> >> *$Ri* - reference to IP address of the interface where the request has >> been received >> >> *$Rp* - reference to the port where the message was received >> ___ >> Users mailing list >> Users@lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> >> -- >> Mark Farmer >> farm...@gmail.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 >> >> >> >> -- >> Mark Farmer >> farm...@gmail.com >> ___ >> 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 >> ___ >> Users mailing list >> Users@lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Access to listen/advertised IP Addresses
i only see $rd which is the domain to which the sip message was sent, it "should" have the advertised ip, or de actual domain, in which case if you need the actual ip, it is useless Regards, David Villasmil email: david.villasmil.w...@gmail.com phone: +34669448337 On Mon, Apr 6, 2020 at 4:00 PM Johan De Clercq wrote: > It,s not exposed I think. I can’t find it back either > > Outlook voor iOS <https://aka.ms/o0ukef> downloaden > -- > *Van:* Users namens David Villasmil < > david.villasmil.w...@gmail.com> > *Verzonden:* Monday, April 6, 2020 4:49:36 PM > *Aan:* OpenSIPS users mailling list > *Onderwerp:* Re: [OpenSIPS-Users] Access to listen/advertised IP Addresses > > No, you’re right. It’s not in the core variables and I can’t find it > either. Which makes me think it’s either not exposed or somewhere in a > module (it’s not in proto_udp) > > I will research a little to try and find it.. > > On Mon, 6 Apr 2020 at 14:04, Mark Farmer wrote: > > Thanks David. But I see no reference to the same variable in OpenSIPS. > > https://www.opensips.org/Documentation/Script-CoreVar-2-4 > > Am I missing something? > > > On Mon, 6 Apr 2020 at 13:45, David Villasmil < > david.villasmil.w...@gmail.com> wrote: > > Right here: > > > https://www.kamailio.org/wiki/cookbooks/5.2.x/pseudovariables#rai_-_received_advertised_ip_address > > Regards, > > David Villasmil > email: david.villasmil.w...@gmail.com > phone: +34669448337 > > > On Mon, Apr 6, 2020 at 12:08 PM Mark Farmer wrote: > > Many thanks for the reply. > > $Ri is certainly useful when the request comes from a non-natted > interface. Thanks for pointing that out :) > > Is there a way to reference the advertised IP address defined in the > listen statement? > > listen=udp:xxx.xxx.xxx.xxx:5060 as xxx.xxx.xxx.xxx:5060 > > Thanks > Mark. > > > On Thu, 2 Apr 2020 at 17:32, Sharad Kumar via Users < > users@lists.opensips.org> wrote: > > Hi Mark, > > If your initial goal is to get the interface IP where request is received > then you can try these variables. > > *$Ri* - reference to IP address of the interface where the request has > been received > > *$Rp* - reference to the port where the message was received > ___ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > -- > Mark Farmer > farm...@gmail.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 > > > > -- > Mark Farmer > farm...@gmail.com > ___ > 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 > ___ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Access to listen/advertised IP Addresses
No, you’re right. It’s not in the core variables and I can’t find it either. Which makes me think it’s either not exposed or somewhere in a module (it’s not in proto_udp) I will research a little to try and find it.. On Mon, 6 Apr 2020 at 14:04, Mark Farmer wrote: > Thanks David. But I see no reference to the same variable in OpenSIPS. > > https://www.opensips.org/Documentation/Script-CoreVar-2-4 > > Am I missing something? > > > On Mon, 6 Apr 2020 at 13:45, David Villasmil < > david.villasmil.w...@gmail.com> wrote: > >> Right here: >> >> >> https://www.kamailio.org/wiki/cookbooks/5.2.x/pseudovariables#rai_-_received_advertised_ip_address >> >> Regards, >> >> David Villasmil >> email: david.villasmil.w...@gmail.com >> phone: +34669448337 >> >> >> On Mon, Apr 6, 2020 at 12:08 PM Mark Farmer wrote: >> >>> Many thanks for the reply. >>> >>> $Ri is certainly useful when the request comes from a non-natted >>> interface. Thanks for pointing that out :) >>> >>> Is there a way to reference the advertised IP address defined in the >>> listen statement? >>> >>> listen=udp:xxx.xxx.xxx.xxx:5060 as xxx.xxx.xxx.xxx:5060 >>> >>> Thanks >>> Mark. >>> >>> >>> On Thu, 2 Apr 2020 at 17:32, Sharad Kumar via Users < >>> users@lists.opensips.org> wrote: >>> >>>> Hi Mark, >>>> >>>> If your initial goal is to get the interface IP where request is >>>> received then you can try these variables. >>>> >>>> *$Ri* - reference to IP address of the interface where the request has >>>> been received >>>> >>>> *$Rp* - reference to the port where the message was received >>>> ___ >>>> Users mailing list >>>> Users@lists.opensips.org >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> >>> >>> >>> -- >>> Mark Farmer >>> farm...@gmail.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 >> > > > -- > Mark Farmer > farm...@gmail.com > ___ > 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 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Access to listen/advertised IP Addresses
Right here: https://www.kamailio.org/wiki/cookbooks/5.2.x/pseudovariables#rai_-_received_advertised_ip_address Regards, David Villasmil email: david.villasmil.w...@gmail.com phone: +34669448337 On Mon, Apr 6, 2020 at 12:08 PM Mark Farmer wrote: > Many thanks for the reply. > > $Ri is certainly useful when the request comes from a non-natted > interface. Thanks for pointing that out :) > > Is there a way to reference the advertised IP address defined in the > listen statement? > > listen=udp:xxx.xxx.xxx.xxx:5060 as xxx.xxx.xxx.xxx:5060 > > Thanks > Mark. > > > On Thu, 2 Apr 2020 at 17:32, Sharad Kumar via Users < > users@lists.opensips.org> wrote: > >> Hi Mark, >> >> If your initial goal is to get the interface IP where request is >> received then you can try these variables. >> >> *$Ri* - reference to IP address of the interface where the request has >> been received >> >> *$Rp* - reference to the port where the message was received >> ___ >> Users mailing list >> Users@lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > > > -- > Mark Farmer > farm...@gmail.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
Re: [OpenSIPS-Users] using load balancer and lookup together
Why are you trying to do all at once?
Why not first do the lookup
https://github.com/davidcsi/kamailio-private-public/blob/a81d7f777a8c5ee2dbb32311f7e6b5a3cf94bf32/kamailio.cfg#L771
and then start load balancing?
https://github.com/davidcsi/kamailio-private-public/blob/a81d7f777a8c5ee2dbb32311f7e6b5a3cf94bf32/kamailio.cfg#L1109
Do you have some special need to fulfill?
David
On Sun, 5 Apr 2020 at 06:34, Michael Vale via Users <
users@lists.opensips.org> wrote:
> hi,
>
> perhaps this can be solved with a failure route and or a check status
> but i dont know and it would be nice if i could do it without it.
>
> no matter how i write the script, either a uac to uac call goes to the
> load balancer or the load balancer is stuck with a 404 reply from the
> script or uac to uac works but when one end is not registered it goes
> to the load balancer instead of getting a 404.
>
> i've tried failure routes and get the same problem. here is a snippet.
>
>if (!lb_start(1,"pstn")) && (!lookup("location","m",)) {
> lb_disable_dst();
> #route(relay);
> #send_reply(404,"No user or gateway");
> if (lb_start(1,"pstn")) {
> send_reply(500,"SIPSIPSIPS");
> # t_relay();
> exit;
> }
> # exit;
> } else if (lookup("location","m")) &&
> (!lb_start(1,"pstn")) {
> lb_disable_dst();
> route(relay);
> exit;
> } else if (lb_start(1,"pstn")) &&
> (lookup("location","m")) {
> lb_disable_dst();
> route(relay);
> exit;
> } else if (!lookup("location","m")) &&
> (!lb_start(1,"pstn")) {
> send_reply(404,"Not Found");
> exit;
> } else if (lb_start(1,"pstn")) &&
> (!lookup("location","m")) {
> # #lb_disable_dst();
> if (!lookup("location","m")) {
> route(relay);
> exit;
> }
> if (lookup("location","m")) {
> lb_disable_dst();
> route(relay);
> exit;
> }
> }
>
> thanks in advance,
>
> michael.
>
>
> ___
> 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
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Issue with 'To' tag and t_reply
404 Not Found > Via: SIP/2.0/UDP > A.A.A.A:5060;received=A.A.A.A;rport=5060;branch=z9hG4bK773616538 > Call-ID: 469A5568-E092-4038-B1B8-13AC9B9571CA > From: "Y" ;tag=117583367 > To: "X" ; > *tag=0b49dc32-2c4b-413e-a349-c781a23d53b9* > CSeq: 1741310 INVITE > Server: PBX > Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, > CANCEL, UPDATE, PRACK, REFER > Reason: Q.850;cause=1 > Content-Length: 0 > > And at this point Server A can match this reply and responds with an ACK: > > ACK sip:X@B.B.B.B SIP/2.0 > Via: SIP/2.0/UDP A.A.A.A:5060;rport;branch=z9hG4bK773616538 > From: "Y" ;tag=117583367 > To: "X" ; > *tag=0b49dc32-2c4b-413e-a349-c781a23d53b9* > Call-ID: 469A5568-E092-4038-B1B8-13AC9B9571CA > CSeq: 1741310 ACK > Max-Forwards: 67 > Contact: > User-Agent: User Agent > Content-Length: 0 > > I think that t_reply is creating a new transaction instead of using > existing one, but I'm not sure why and how to fix this? > > Thanks! > > Best regards, > Yury. > ___ > 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 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] help on failover routing
I’ve never done it, but I understand there’s a route every time a TLS socket is disconnected. You can probably use that to remove the contact. On Fri, 20 Mar 2020 at 10:43, Liviu Chircu wrote: > On 20.03.2020 12:37, johan wrote: > > > > Hence, > > > > - when the softphone is registered, a call comes on that DID in udp > > (we do lookup_location) and we send it to the user in tls (this works) > > > > - when the softphone is off for a long time, there is no record in > > location so then I route the call via the provider to his real mobile > > number (this works also) > > > > - the problem is when the mobile looses his dataconnection, then I do > > have a record in location, I try to send the call, which will fail. > > Upon failure, I drop the record in subscriber. And here the problem > > begins. > > > > The invite is adapted at this point already for tls => provider > > doesn't want it as he is udp. > > > > > > So how can I have the original request back for routing to the real > > mobile number ? Or how can I check if the user is still connected (aka > > how can I send options to see if he's alive) before calling t_relay. > > Hi, Johan! > > 1. this solution of calling remove() after a routing failure is nice. > Alexey Vasilyev put together a feature request [1] related to this > problem, where he asks for an automated mechanism of deleting a contact > whenever its TLS connection is found to be dead. > > 2. Did you try to force the sending socket of the INVITE ($fs variable) > to your "udp:1.2.3.4:5060" listener? I think this should work inside a > failure_route and should properly route to your provider via UDP. Also, > I believe Bogdan fixed this recently [2] (but master branch only?!), > such that "$fs" is not set to the TLS listener inside failure_route - > might wanna check. > > 3. As a long-term solution to this problem, we are working on adding > RFC 8599 Push Notification support via SIP in OpenSIPS 3.1. The spec is > still rather new, but I'm curious if your app's SIP stack supports it > :) Basically, this will allow you to wake up the phone so it > re-registers whenever you need to deliver an INVITE to it, in a > standards-approved way. > > Best regards, > > [1]: https://github.com/OpenSIPS/opensips/issues/1769 > > [2]: https://github.com/OpenSIPS/opensips/commit/f73abff9 > > [3]: https://tools.ietf.org/html/rfc8599 > > -- > Liviu Chircu > www.twitter.com/liviuchircu | www.opensips-solutions.com > > OpenSIPS Summit, Amsterdam, May 2020 >www.opensips.org/events > > > ___ > 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 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] IMPORTANT: OpenSIPS Summit 2020 - what's next ?
I’d go for October. This situation is going to last months, not weeks. And then there’s some catch up to do everywhere. Cheers On Wed, 18 Mar 2020 at 19:47, Podrigal, Aron wrote: > I wouldn't be able to join if it is between September 13 and October 13 > > > On Wed, Mar 18, 2020, 2:33 PM Bogdan-Andrei Iancu > wrote: > >> May I ask why? :) >> >> Bogdan-Andrei Iancu >> >> OpenSIPS Founder and Developer >> https://www.opensips-solutions.com >> OpenSIPS Summit, Amsterdam, May 2020 >> https://www.opensips.org/events/Summit-2020Amsterdam/ >> >> On 3/18/20 7:17 PM, Johan De Clercq wrote: >> >> Please plan middle September. >> >> Outlook voor iOS <https://aka.ms/o0ukef> downloaden >> -- >> *Van:* Users >> namens Bogdan-Andrei Iancu >> >> *Verzonden:* Wednesday, March 18, 2020 6:13:10 PM >> *Aan:* OpenSIPS users mailling list >> ; developensips >> ; busin...@lists.opensips.org >> ; >> n...@lists.opensips.org >> >> *Onderwerp:* Re: [OpenSIPS-Users] IMPORTANT: OpenSIPS Summit 2020 - >> what's next ? >> >> Hello (again), >> >> First, many thanks to those who shared with us their opinions and took >> the community poll. >> >> What was the outcome ? >> >> >> https://blog.opensips.org/2020/03/18/opensips-summit-2020-the-september-replaning/ >> >> Best regards, >> >> Bogdan-Andrei Iancu >> >> OpenSIPS Founder and Developer >> https://www.opensips-solutions.com >> OpenSIPS Summit, Amsterdam, May 2020 >> https://www.opensips.org/events/Summit-2020Amsterdam/ >> >> On 3/13/20 12:11 PM, Bogdan-Andrei Iancu wrote: >> >> Hello all, >> >> In the context of the COVID-19 pandemic, we reach back to you, the >> community, to understand what people expect from the upcoming OpenSIPS >> Summit May 2020, Amsterdam. >> >> Safety of people is first and there is no trade off here. There are risks >> of infection and also risks of traveling with more and more borders being >> closed and flights canceled. Nevertheless life goes on, the community still >> needs the OpenSIPS value/knowledge and we are still committed to provide >> it. So, the dilemma is how should we balance both. >> >> By filling in the next form, you will let us know your opinion on this >> delicate matter, concerning both us and you, so please help us to make the >> right choice. >> >> http://bit.ly/2IKgmtm >> >> And do not forget, your safely and your opinion matter to us ! >> >> Best regards, >> >> -- >> Bogdan-Andrei Iancu >> >> OpenSIPS Founder and Developer >> https://www.opensips-solutions.com >> OpenSIPS Summit, Amsterdam, May 2020 >> https://www.opensips.org/events/Summit-2020Amsterdam/ >> >> >> ___ >> Users mailing >> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> >> ___ >> Users mailing >> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> ___ >> Users mailing list >> Users@lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > ___ > 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 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Public IP in REGISTER
Have you tried setting the bflag right before save()’ing during the
REGISTER?
On Sun, 8 Mar 2020 at 23:34, Jehanzaib Younis
wrote:
> Hi David,
>
> I have one contact in the Usrloc
> Do you think it could be a timeout issue? or i should use
> remove_on_timeout_bflag option?
>
> On Mon, Mar 9, 2020 at 12:31 PM David Villasmil <
> david.villasmil.w...@gmail.com> wrote:
>
>> Make sure you don’t have 2 contacts
>>
>> On Sun, 8 Mar 2020 at 23:22, Jehanzaib Younis
>> wrote:
>>
>>> but i have strange issue.
>>> nathelper keep on sending the OPTION to old IP:PORT. As soon as the new
>>> REGISTER comes in, it should only send the option to the latest one.
>>> For example, I see OPTION going to xx.xxx.xx.xx:25001 and opensips keep
>>> on sending to this after every 120 seconds (which is my setting)
>>> The new register comes in and the xx.xxx.xx.xx:25004 letsay and now
>>> opensips sends to 25001 as well as 25004. Obviously my cpe only replies to
>>> the latest one.
>>>
>>> Anyone have faced this issue ?
>>>
>>> Thank you
>>>
>>> On Fri, Mar 6, 2020 at 2:54 PM Jehanzaib Younis <
>>> jehanzaib.ki...@gmail.com> wrote:
>>>
>>>> Thank you for your suggestions Diptesh,
>>>>
>>>> Actually adding modparam("nathelper", "received_avp", "$avp(s:rcv)") &
>>>> modparam("registrar", "received_avp", "$avp(s:rcv)") did the trick.
>>>>
>>>>
>>>>
>>>> On Thu, Mar 5, 2020 at 7:25 PM Dipteshkumar Patel <
>>>> diptesh.pa...@ecosmob.com> wrote:
>>>>
>>>>> Hello Jehan,
>>>>>
>>>>> OpenSIPS handle NAT different way for INVITE and REGISTER packets. If
>>>>> we use fix_nated_contact(), it will get the actual source ip from network
>>>>> and create a lump for that and replace the headers(like Contact, Received
>>>>> in Via and c parameter in SDP packet) just before sent out or relay the
>>>>> packet.
>>>>>
>>>>> In your case, You need to manage the REGISTER and we are using
>>>>> OpenSIPS as a Registrar so we are not relay the packet so
>>>>> fix_nated_contact() will not help you. and your location table will have
>>>>> the private ip not public. So Let me guide how NAT can be managed in
>>>>> Registration.
>>>>>
>>>>> There are three modules are responsible for the registration with NAT
>>>>> handling.
>>>>> 1. registrar module
>>>>> 2. usrloc module
>>>>> 3. nathelper module
>>>>>
>>>>> nathelper module can check the packet source is behind nat or nat and
>>>>> get the public ip from source ip.
>>>>>
>>>>> We need to define a module parameter for netheper module with an
>>>>> avp variable so the module will store the received IP in that avp. and
>>>>> similar avp should be in registrar module so registrar module can read the
>>>>> avp and store it into location as received parameter.
>>>>>
>>>>> modparam("nathelper", "received_avp", "$avp(received)")
>>>>>
>>>>> modparam("registrar", "received_avp", "$avp(received)")
>>>>>
>>>>>
>>>>> Refer the following snippet.
>>>>>
>>>>> /*Other registrar Parameters*/
>>>>> modparam("registrar", "received_avp", "$avp(received)")
>>>>>
>>>>> /*Other nethelper Parameters*/
>>>>> modparam("nathelper", "received_avp", "$avp(received)") #keep in mind
>>>>> that this avp should be same in registrar module.
>>>>>
>>>>> /*Other usrloc Parameters*/
>>>>> modparam("usrloc", "nat_bflag", "NAT_FLAG")
>>>>>
>>>>> route(NAT_MANAGE);
>>>>> ---
>>>>> /*Some Authentication Stuff*/
>>>>> ---
>>>>> if(!save("location")) {
>>>>> sl_reply_error();
>>>>> }
>>>>>
>>>>> route[NAT_MANAGE] {
>>>>> if(nat_uac_test("19")){
>>>>> xlog("L_INFO","--- [NAT_MANAGE] UAC IS BEHIND NAT ---")
Re: [OpenSIPS-Users] Public IP in REGISTER
ich is wrong. It should go to its public IP:port >>>>>> >>>>>> but weird thing is, i see the OPTION is sent to its Public IP >>>>>> (OPTIONS sip:180.xx.xx.xx:1502). I also see the 200 OK which is perfect. >>>>>> >>>>>> I tried to use fix_nated_register(); but it does not change anything. >>>>>> >>>>>> Can anyone help please? >>>>>> >>>>>> Thank you >>>>>> >>>>>> >>>>>> Regards, >>>>>> Jehan >>>>>> ___ >>>>>> Users mailing list >>>>>> Users@lists.opensips.org >>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>>>> >>>>> >>>>> >>>>> *0333 332 | www.x-on.co.uk <http://www.x-on.co.uk> | ** >>>>> <https://www.linkedin.com/company/x-on> >>>>> <https://www.facebook.com/XonTel> >>>>> <https://twitter.com/xonuk> * >>>>> >>>>> X-on is a trading name of Storacall Technology Ltd a limited company >>>>> registered in England and Wales. >>>>> Registered Office : Avaland House, 110 London Road, Apsley, Hemel >>>>> Hempstead, Herts, HP3 9SD >>>>> <https://www.google.com/maps/search/110+London+Road,+Apsley,+Hemel+Hempstead,%0D%0AHerts,+HP3+9SD?entry=gmail=g>. >>>>> Company Registration No. 2578478. >>>>> The information in this e-mail is confidential and for use by the >>>>> addressee(s) only. If you are not the intended recipient, please notify >>>>> X-on immediately on +44(0)333 332 and delete the >>>>> message from your computer. If you are not a named addressee you must >>>>> not use, disclose, disseminate, distribute, copy, print or reply to this >>>>> email. Views or opinions expressed by an individual >>>>> within this email may not necessarily reflect the views of X-on or its >>>>> associated companies. Although X-on routinely screens for viruses, >>>>> addressees should scan this email and any attachments >>>>> for viruses. X-on makes no representation or warranty as to the >>>>> absence of viruses in this email or any attachments. >>>>> >>>>> ___ >>>>> Users mailing list >>>>> Users@lists.opensips.org >>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> >>>> >> >>> >>>>> A cloud hosted phone system <https://vndpbx.com> >>>> ___ >>>> Users mailing list >>>> Users@lists.opensips.org >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> >>> >>> *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 >>> of this communication is prohibited. Please inform originator if you have >>> received it in error. >>> >>> *Caution for viruses, malware etc.* >>> This communication, including any attachments, may not be free of >>> viruses, trojans, similar or new contaminants/malware, interceptions or >>> interference, and may not be compatible with your systems. You shall carry >>> out virus/malware scanning on your own before opening any attachment to >>> this e-mail. The sender of this e-mail and Company including its sister >>> concerns shall not be liable for any damage that may incur to you as a >>> result of viruses, incompleteness of this message, a delay in receipt of >>> this message or any other computer problems. >>> ___ >>> Users mailing list >>> Users@lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >> >> >> -- >> Regards, >> Jehanzaib >> > > > -- > Regards, > Jehanzaib > ___ > 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 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] ratelimit network algorithm traffic limit
There we go
On Mon, 2 Mar 2020 at 18:43, Ben Newlin wrote:
> The limit is set when the pipe is created the first time you call rl_check
> as mentioned earlier. OpenSIPS doesn’t use a modparam to create the pipes;
> instead they are created the first time they are referenced.
>
>
>
> https://opensips.org/docs/modules/3.0.x/ratelimit.html#func_rl_check
>
>
>
> “Check the current request against the pipe identified by name and
> changes/updates the limit. If no pipe is found, then a new one is created
> with the specified limit and algorithm, if specified.”
>
>
>
> Ben Newlin
>
>
>
> *From: *Users on behalf of David
> Villasmil
> *Reply-To: *OpenSIPS users mailling list
> *Date: *Monday, March 2, 2020 at 1:40 PM
> *To: *OpenSIPS users mailling list
> *Subject: *Re: [OpenSIPS-Users] ratelimit network algorithm traffic limit
>
>
>
> But that’s Kamaililio. He’s talking about openSIPS, and I can’t see that
> same param in opensips 2.4
>
>
>
> On Mon, 2 Mar 2020 at 18:24, Ovidiu Sas wrote:
>
> Take a look at the pipe parameter:
> https://kamailio.org/docs/modules/5.4.x/modules/ratelimit.html#idp49883756
>
> # define pipe 4 with a limit of 1 pending bytes in the rx_queue
> # using NETWORK algorithm
> modparam("ratelimit", "pipe", "4:NETWORK:1")
>
> Hope this helps!
>
> Regards,
> Ovidiu Sas
>
> On Mon, Mar 2, 2020 at 12:53 PM Jeff Pyle wrote:
> >
> > But how does it work for the NETWORK algorithm? The docs specifically
> mention a modparam, but even if that's not the case anymore, what unit is
> the limit one might specify with rl_check()?
> >
> > More generally, how does one implement the NETWORK algorithm?
> >
> >
> > - Jeff
> >
> >
> > On Mon, Mar 2, 2020 at 12:19 PM David Villasmil <
> david.villasmil.w...@gmail.com> wrote:
> >>
> >> Jeff,
> >>
> >> Yep, you’re totally right. The limit should be set when calling the
> check, I.e:
> >>
> >> if (!rl_check("$rU", "50", "TAILDROP")) {
> >> sl_send_reply("503", "Server Unavailable");
> >> exit;
> >> };
> >>
> >>
> >>
> >> On Mon, 2 Mar 2020 at 16:19, Jeff Pyle wrote:
> >>>
> >>> This doesn't appear to have anything to do with a any type of limit or
> the network algorithm. In fact, it says, "...and only affects the Taildrop
> and RED algorithms." ?
> >>>
> >>>
> >>> - Jeff
> >>>
> >>>
> >>> On Mon, Mar 2, 2020 at 11:05 AM David Villasmil <
> david.villasmil.w...@gmail.com> wrote:
> >>>>
> >>>>
> https://opensips.org/html/docs/modules/2.4.x/ratelimit.html#param_limit_per_interval
> >>>>
> >>>>
> >>>> On Mon, 2 Mar 2020 at 15:54, Jeff Pyle wrote:
> >>>>>
> >>>>> Hello,
> >>>>>
> >>>>> The ratelimit doc page (v2.4) section 1.3.4 says the following: "If
> the returned amount exceeds the limit specified in the modparam, rl_check
> returns an error." The problem is I don't see a modparam to define the
> limit.
> >>>>>
> >>>>>
> >>>>>
> >>>>> - Jeff
> >>>>>
> >>>>> ___
> >>>>> 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
> >>>>
> >>>>
> >>>
> >>> ___
> >>> 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
> >> ___
> >> 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
>
>
>
> --
> VoIP Embedded, Inc.
> http://www.voipembedded.com
>
> ___
> 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
> ___
> 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
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] ratelimit network algorithm traffic limit
But that’s Kamaililio. He’s talking about openSIPS, and I can’t see that
same param in opensips 2.4
On Mon, 2 Mar 2020 at 18:24, Ovidiu Sas wrote:
> Take a look at the pipe parameter:
> https://kamailio.org/docs/modules/5.4.x/modules/ratelimit.html#idp49883756
>
> # define pipe 4 with a limit of 1 pending bytes in the rx_queue
> # using NETWORK algorithm
> modparam("ratelimit", "pipe", "4:NETWORK:1")
>
> Hope this helps!
>
> Regards,
> Ovidiu Sas
>
> On Mon, Mar 2, 2020 at 12:53 PM Jeff Pyle wrote:
> >
> > But how does it work for the NETWORK algorithm? The docs specifically
> mention a modparam, but even if that's not the case anymore, what unit is
> the limit one might specify with rl_check()?
> >
> > More generally, how does one implement the NETWORK algorithm?
> >
> >
> > - Jeff
> >
> >
> > On Mon, Mar 2, 2020 at 12:19 PM David Villasmil <
> david.villasmil.w...@gmail.com> wrote:
> >>
> >> Jeff,
> >>
> >> Yep, you’re totally right. The limit should be set when calling the
> check, I.e:
> >>
> >> if (!rl_check("$rU", "50", "TAILDROP")) {
> >> sl_send_reply("503", "Server Unavailable");
> >> exit;
> >> };
> >>
> >>
> >>
> >> On Mon, 2 Mar 2020 at 16:19, Jeff Pyle wrote:
> >>>
> >>> This doesn't appear to have anything to do with a any type of limit or
> the network algorithm. In fact, it says, "...and only affects the Taildrop
> and RED algorithms." ?
> >>>
> >>>
> >>> - Jeff
> >>>
> >>>
> >>> On Mon, Mar 2, 2020 at 11:05 AM David Villasmil <
> david.villasmil.w...@gmail.com> wrote:
> >>>>
> >>>>
> https://opensips.org/html/docs/modules/2.4.x/ratelimit.html#param_limit_per_interval
> >>>>
> >>>>
> >>>> On Mon, 2 Mar 2020 at 15:54, Jeff Pyle wrote:
> >>>>>
> >>>>> Hello,
> >>>>>
> >>>>> The ratelimit doc page (v2.4) section 1.3.4 says the following: "If
> the returned amount exceeds the limit specified in the modparam, rl_check
> returns an error." The problem is I don't see a modparam to define the
> limit.
> >>>>>
> >>>>>
> >>>>>
> >>>>> - Jeff
> >>>>>
> >>>>> ___
> >>>>> 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
> >>>>
> >>>> ____
> >>>
> >>> ___
> >>> 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
> >> ___
> >> 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
>
>
>
> --
> VoIP Embedded, Inc.
> http://www.voipembedded.com
>
> ___
> 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
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] ratelimit network algorithm traffic limit
Jeff,
Yep, you’re totally right. The limit should be set when calling the check,
I.e:
if (!rl_check("$rU", "50", "TAILDROP")) {
sl_send_reply("503", "Server Unavailable");
exit;
};
On Mon, 2 Mar 2020 at 16:19, Jeff Pyle wrote:
> This doesn't appear to have anything to do with a any type of limit or the
> network algorithm. In fact, it says, "...and only affects the Taildrop and
> RED algorithms." ?
>
>
> - Jeff
>
>
> On Mon, Mar 2, 2020 at 11:05 AM David Villasmil <
> david.villasmil.w...@gmail.com> wrote:
>
>>
>> https://opensips.org/html/docs/modules/2.4.x/ratelimit.html#param_limit_per_interval
>>
>>
>> On Mon, 2 Mar 2020 at 15:54, Jeff Pyle wrote:
>>
>>> Hello,
>>>
>>> The ratelimit doc page (v2.4) section 1.3.4 says the following: "If the
>>> returned amount exceeds the limit specified in the modparam, rl_check
>>> returns an error." The problem is I don't see a modparam to define the
>>> limit.
>>>
>>>
>>>
>>> - Jeff
>>>
>>> ___
>>> 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
>>
>
>>
> ___
> 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
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] ratelimit network algorithm traffic limit
https://opensips.org/html/docs/modules/2.4.x/ratelimit.html#param_limit_per_interval On Mon, 2 Mar 2020 at 15:54, Jeff Pyle wrote: > Hello, > > The ratelimit doc page (v2.4) section 1.3.4 says the following: "If the > returned amount exceeds the limit specified in the modparam, rl_check > returns an error." The problem is I don't see a modparam to define the > limit. > > > > - Jeff > > ___ > 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 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Selecting from a variable with \n as separator
Not sure, try with \\n instead of \n
On Sun, 1 Mar 2020 at 12:08, Saint Michael wrote:
> Could you post an example? I have no idea?
>
> On Sun, Mar 1, 2020, 4:54 AM David Villasmil <
> david.villasmil.w...@gmail.com> wrote:
>
>>
>> Have you tried scaping it?
>>
>> On Sun, 1 Mar 2020 at 03:10, Saint Michael wrote:
>>
>>> I am executing an external application that sends back a '\n' and
>>> garbage after the end of the useful information. I am trying to select my
>>> piece using this:
>>> $var(new_ani)=$(var(new_anix){s.select,0,\n});
>>> but it does not work. How can I do this?
>>> Philip Orleans
>>> ___
>>> 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
>> ___
>> 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
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Selecting from a variable with \n as separator
Have you tried scaping it?
On Sun, 1 Mar 2020 at 03:10, Saint Michael wrote:
> I am executing an external application that sends back a '\n' and garbage
> after the end of the useful information. I am trying to select my piece
> using this:
> $var(new_ani)=$(var(new_anix){s.select,0,\n});
> but it does not work. How can I do this?
> Philip Orleans
> ___
> 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
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] help with error messages
Did you set the new IPs, etc? On Wed, 26 Feb 2020 at 06:24, Saint Michael wrote: > I moved a VM from one network to another, started opensips and I get > these errors. Could somebody shed light? > Feb 26 06:19:20 opensips systemd[1]: Started OpenSIPS is a very fast and > flexible SIP (RFC3261) server. > Feb 26 06:19:21 opensips opensips[4663]: Feb 26 06:19:21 [4744] > ERROR:core:pv_set_ruri_host: bad parameters > Feb 26 06:19:21 opensips opensips[4663]: Feb 26 06:19:21 [4744] > ERROR:core:do_assign: setting PV failed > Feb 26 06:19:21 opensips opensips[4663]: Feb 26 06:19:21 [4744] > ERROR:core:do_assign: error at /etc/opensips/opensips.cfg:169 > Feb 26 06:19:22 opensips opensips[4663]: Feb 26 06:19:22 [4744] > ERROR:core:pv_set_ruri_host: bad parameters > Feb 26 06:19:22 opensips opensips[4663]: Feb 26 06:19:22 [4744] > ERROR:core:do_assign: setting PV failed > Feb 26 06:19:22 opensips opensips[4663]: Feb 26 06:19:22 [4744] > ERROR:core:do_assign: error at /etc/opensips/opensips.cfg:169 > Feb 26 06:19:28 opensips opensips[4663]: Feb 26 06:19:28 [4744] > ERROR:core:pv_set_ruri_host: bad parameters > Feb 26 06:19:28 opensips opensips[4663]: Feb 26 06:19:28 [4744] > ERROR:core:do_assign: setting PV failed > Feb 26 06:19:28 opensips opensips[4663]: Feb 26 06:19:28 [4744] > ERROR:core:do_assign: error at /etc/opensips/opensips.cfg:169 > Feb 26 06:19:32 opensips opensips[4663]: Feb 26 06:19:32 [4744] > ERROR:core:pv_set_ruri_host: bad parameters > Feb 26 06:19:32 opensips opensips[4663]: Feb 26 06:19:32 [4744] > ERROR:core:do_assign: setting PV failed > Feb 26 06:19:32 opensips opensips[4663]: Feb 26 06:19:32 [4744] > ERROR:core:do_assign: error at /etc/opensips/opensips.cfg:169 > Feb 26 06:19:34 opensips opensips[4663]: Feb 26 06:19:34 [4744] > ERROR:core:pv_set_ruri_host: bad parameters > Feb 26 06:19:34 opensips opensips[4663]: Feb 26 06:19:34 [4744] > ERROR:core:do_assign: setting PV failed > Feb 26 06:19:34 opensips opensips[4663]: Feb 26 06:19:34 [4744] > ERROR:core:do_assign: error at /etc/opensips/opensips.cfg:169 > > > ___ > 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 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] How to start OpenSIPS 3.0 server
Try to find an ERR or CRITICAL in the output. After executing manually, if opensips is running (and you’re executing it while you’re toot), then it’s probably related to permissions for the using running it (opensips) On Thu, 13 Feb 2020 at 04:19, Tekin, Arda wrote: > It prints too many logs when I run below command manually > > opensips –DD –E > > > > I tried to share the output of command but mail server is blocking my > mail. That’s my 3rd try > > > > Let me share these notes, > > > > Opensips binary is under standard prefix > > [root@a3783871a39f ~]# which opensips > > /usr/local/sbin/opensips > > > > > > When I run opensips manually it works > > [root@a3783871a39f init.d]# opensips > > Listening on > > udp: 172.16.30.241 [172.16.30.241]:5060 > > Aliases: > > > > Verified that it listens udp:5060 > > [root@a3783871a39f run]# netstat -anp | grep opensips > > udp0 0 172.16.30.241:5060 0.0.0.0:* > 31542/opensips > > unix 2 [ ] DGRAM21043523 31542/opensips > > unix 2 [ ] DGRAM21043526 31542/opensips > > > > Opensips service start script looks like this, > > [root@a3783871a39f ~]# cat /etc/rc.d/init.d/opensips > > … > > … > > > > # Source function library. > > . /etc/rc.d/init.d/functions > > prog=opensips > > opensips=/usr/local/sbin/$prog > > cfgdir="/usr/local/etc/$prog" > > pidfile="/var/run/$prog.pid" > > lockfile="/var/lock/subsys/$prog" > > configfile="$cfgdir/$prog.cfg" > > m4configfile="$cfgdir/$prog.m4" > > m4archivedir="$cfgdir/archive" > > OPTIONS="" > > S_MEMORY=32 > > P_MEMORY=32 > > RETVAL=0 > > … > > … > > --- > > > > systemctl start opensips > > service opensips start > > commands gives the same result. > > Starting opensips (via systemctl): Job for opensips.service failed > because a configured resource limit was exceeded. See "systemctl status > opensips.service" and "journalctl -xe" for details. > >[FAILED] > > > > Any idea? > > Do you see any help document that explains how Opensips 3.0 should be > started by a command/service script? > > > > Regards, > > Arda > > > > > > *From:* Users *On Behalf Of *David > Villasmil > *Sent:* Wednesday, February 12, 2020 4:55 PM > *To:* OpenSIPS users mailling list > *Subject:* Re: [OpenSIPS-Users] How to start OpenSIPS 3.0 server > > > > *Attention: This email was sent from someone outside of Afiniti. Always > use caution when opening attachments, clicking links from unknown senders > or when receiving unexpected emails.* > > > > That’s your problem. Make sure the path exists and opensips can create the > pid file there. > > > > Other test you can do is start manually, I.e: opensips -DD -E > > Providing the proper config file; etc. > > > ___ > 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 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] How to start OpenSIPS 3.0 server
That’s your problem. Make sure the path exists and opensips can create the pid file there. Other test you can do is start manually, I.e: opensips -DD -E Providing the proper config file; etc. On Wed, 12 Feb 2020 at 05:23, Tekin, Arda wrote: > Hi David > > > > I just see this log entry when I call “journalctl -xe” > > > > -- Unit opensips.service has begun starting up. > > Feb 12 10:04:24 a3783871a39f opensips[15922]: Starting opensips: [FAILED] > > Feb 12 10:04:24 a3783871a39f systemd[1]: Can't open PID file > /var/run/opensips.pid (yet?) after start: No such file or directory > > Feb 12 10:04:24 a3783871a39f systemd[1]: Failed to start LSB: start, stop > OpenSIPS. > > -- Subject: Unit opensips.service has failed > > -- Defined-By: systemd > > -- Support: http://lists.freedesktop.org/mailman/listinfo/systemd-devel > > -- > > -- Unit opensips.service has failed. > > -- > > -- The result is failed. > > Feb 12 10:04:24 a3783871a39f systemd[1]: Unit opensips.service entered > failed state. > > Feb 12 10:04:24 a3783871a39f systemd[1]: opensips.service failed. > > > > > > *From:* Users *On Behalf Of *David > Villasmil > *Sent:* Wednesday, February 12, 2020 2:22 AM > *To:* OpenSIPS users mailling list > *Subject:* Re: [OpenSIPS-Users] How to start OpenSIPS 3.0 server > > > > *Attention: This email was sent from someone outside of Afiniti. Always > use caution when opening attachments, clicking links from unknown senders > or when receiving unexpected emails.* > > > > And the output of > > > > journalctl -xe > > > > ? > > > > On Tue, 11 Feb 2020 at 17:20, Tekin, Arda wrote: > > How can I run OpenSIPS 3.0? > > > > Previously we are running `opensipsctl start` command. > > > > How to set and use db connection configuration before starting OpenSIPS? > > > > I have compiled latest source code on CentOS 7 successfully. Copied init > script > > cp /root/src/opensips-3.0/packaging/redhat_fedora/opensips.init > /etc/rc.d/init.d/. > > > > Edited the init file > > prog=opensips > > opensips=/usr/local/sbin/$prog > > cfgdir="/usr/local/etc/$prog" > > pidfile="/var/run/$prog.pid" > > lockfile="/var/lock/subsys/$prog" > > configfile="$cfgdir/$prog.cfg" > > m4configfile="$cfgdir/$prog.m4" > > m4archivedir="$cfgdir/archive" > > OPTIONS="" > > S_MEMORY=32 > > P_MEMORY=32 > > RETVAL=0 > > > > > > When I run init script I get this error. > > Starting opensips (via systemctl): Job for opensips.service failed > because a configured resource limit was exceeded. See "systemctl status > opensips.service" and "journalctl -xe" for details. > >[FAILED] > > > > Where is doc page explaining how OpenSIPS 3 starts? > > > > Kind Regards, > > Arda > > ___ > 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 > ___ > 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 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] How to start OpenSIPS 3.0 server
And the output of journalctl -xe ? On Tue, 11 Feb 2020 at 17:20, Tekin, Arda wrote: > How can I run OpenSIPS 3.0? > > > > Previously we are running `opensipsctl start` command. > > > > How to set and use db connection configuration before starting OpenSIPS? > > > > I have compiled latest source code on CentOS 7 successfully. Copied init > script > > cp /root/src/opensips-3.0/packaging/redhat_fedora/opensips.init > /etc/rc.d/init.d/. > > > > Edited the init file > > prog=opensips > > opensips=/usr/local/sbin/$prog > > cfgdir="/usr/local/etc/$prog" > > pidfile="/var/run/$prog.pid" > > lockfile="/var/lock/subsys/$prog" > > configfile="$cfgdir/$prog.cfg" > > m4configfile="$cfgdir/$prog.m4" > > m4archivedir="$cfgdir/archive" > > OPTIONS="" > > S_MEMORY=32 > > P_MEMORY=32 > > RETVAL=0 > > > > > > When I run init script I get this error. > > Starting opensips (via systemctl): Job for opensips.service failed > because a configured resource limit was exceeded. See "systemctl status > opensips.service" and "journalctl -xe" for details. > >[FAILED] > > > > Where is doc page explaining how OpenSIPS 3 starts? > > > > Kind Regards, > > Arda > ___ > 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 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] uac_replace_from/to
Dlg_manage() maybe?
On Tue, 4 Feb 2020 at 14:40, Антон Ершов wrote:
> Hello friends
> Help me to understand. In request_route i make replace from and to fields.
> After an unsuccessful attempt to invite, I try to direct the call to
> another place and again I need to replace the from and to fields. But
> instead of replacing, new values are simply added
>
> route {
> ...
> route(rewrite_header);
> ...
> }
> failure_route[missed_call] {
> setflag(need_uac_restore);
> ...
> route(rewrite_header);
> }
> route[rewrite_header] {
> if (isflagset(need_uac_restore)) {
> xlog("L_INFO", "[ $ci ] - restore from and to fields\n");
> uac_restore_from();
> uac_restore_to();
> }
> uac_replace_from("","sip:$avp(fromuser)@$avp(fromdomain)");
> uac_replace_to("","sip:$avp(touser)@$avp(todomain)");
> }
>
> in the end I get such fields
>
> From: >;tag=fe654d87-7565-408f-9925-98a5bac99e1c
> To:
>
> it simply appends the new value to the old and does not replace it
> changing replace modes in uac module does not change behavior
> in debug log i see this
>
> DBG:uac:replace_uri: uri to replace [],
> replacement is []
> DBG:dialog:new_dlg_val: inserting <739823>=<>
> DBG:dialog:store_dlg_value_unsafe: var found-> < >>!
> DBG:uac:replace_uri: uri to replace [],
> replacement is []
> DBG:dialog:new_dlg_val: inserting <739824>=<>
> DBG:dialog:store_dlg_value_unsafe: var found-> < >>!
>
> 10.23.100.40 - call initiator
> 10.10.10.10 - where should the call go first
> 10.100.100.100 - opensips
> sipofon.loc - where should the call go after failure_route has worked
>
>
>
>
>
> ___
> 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
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
