[OpenSIPS-Users] 401 and 407 in REPLY ROUTE

[Ali Pey]

Hello,

After I send out an Invite message to a carrier, the carrier responds with
407 and then 401 for authentication. I need to use the UAC_AUTH module to
respond with another Invite with authentication to resend it to the carrier
since my gateways don't have the trunk's username and password.

My first problem is that 407 and 401 go to the REPLY Route and not the
FAILURE Route. Should they not go to FAILURE route? What am I missing?

Thanks,
Ali Pey
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Re: [OpenSIPS-Users] TLS - Certificate Validation Failure error on SIP Phones - OpenSIPS version 1.11.5

[Ali Pey]

Hello Rodrigo,

Thank you for your response. I set verify_cert and require_cert to zero and
that fixes my problem. After that I was getting "Certificate Name Mismatch"
error on the eyeBeam and Zoiper phones and after some investigation, I
realized that it was due to wild cards in my certificate. Apparently,
eyeBeam and Zoiper cannot or do not handle wild cards (*) in a certificate.

Best regards,
Ali Pey


On Fri, Apr 8, 2016 at 10:48 AM, Rodrigo Pimenta Carvalho  wrote:

> Hi.
>
>
> I got the same problem in softphone ZOIPER.
>
> I just let my ZOIPER ignore the file received from OpenSIPS and then the
> problem was solved. Otherwise I should had to install the client party on
> the phone. It was possible for me because in my project I didn't have to
> use certificates, just cryptographic messages with TLS.
>
>
> See below the configuration in my OpenSIPS.cfg file (my proxy is version
> 2.2 from 2015):
>
>
> loadmodule
> "proto_tls.so"
>
>
>
>  modparam("proto_tls","verify_cert",
> "0")
>
>  modparam("proto_tls","require_cert", "0")  #0 means  *do not* force the
> client to present a certificate where as 1 means *do* ask the client to
> present a cert.
>  modparam("proto_tls","tls_method", "TLSv1")  #If you want RFC3261
> conformance and all your clients support TLSv1 (or you are planning to use
> encrypted "tunnels" only between differe
>
>
>
>
> modparam("proto_tls", "certificate",
> "/usr/local/etc/opensips/tls/rootCA/certs/cert.pem")
>
> modparam("proto_tls", "private_key",
> "/usr/local/etc/opensips/tls/rootCA/private/key.pem")
>
> modparam("proto_tls", "ca_list",
> "/usr/local/etc/opensips/tls/rootCA/cacert.pem")
>
> modparam("proto_tls", "ca_dir", "/usr/local/etc/opensips/tls/rootCA/")
>
>
> # Sets the TLS protocol. The first parameter, if set, represents the id of
> the domain. TLS method which can
> be:
> #
>
> #TLSv1_2 - means OpenSIPS will accept only TLSv1.2 connections
> (rfc3261
> conformant).
>
> #
>
> #TLSv1 - means OpenSIPS will accept only TLSv1 connections (rfc3261
> conformant).
>
> #
>
> #SSLv3 - means OpenSIPS will accept only SSLv3
> connections
>
> #
>
> #SSLv2 - means OpenSIPS will accept only SSLv2 connections (almost all
> old clients support
> this).
> #
>
> #SSLv23 - means OpenSIPS will accept any of the above methods, but the
> initial SSL hello must be v2 (in the initial hello all the supported
> protocols are advertised enabling swit
> #
>
> #Default value is SSLv23.
>
>
> Tell me if I'm wrongly, please.
>
>
> Best regards.
>
>
>
> RODRIGO PIMENTA CARVALHO
> Inatel Competence Center
> Software
> Ph: +55 35 3471 9200 RAMAL 979
>
>
> --
> *De:* users-boun...@lists.opensips.org 
> em nome de Ali Pey 
> *Enviado:* sexta-feira, 8 de abril de 2016 10:25
> *Para:* OpenSIPS users mailling list
> *Assunto:* Re: [OpenSIPS-Users] TLS - Certificate Validation Failure
> error on SIP Phones - OpenSIPS version 1.11.5
>
> Hello Hamid,
>
> The parameters below don't have any effects. In my scenario, the sip
> phones are rejecting the tls connection by saying "Certificate Validation
> Failure".
>
> Neither of parameters below had any effects.
>
>
> Anyone else has any idea what I need to look for?
>
> Regards,
> Ali Pey
>
>
> On Fri, Apr 8, 2016 at 4:00 AM, Hamid Hashmi 
> wrote:
>
>> Please define  following values
>>
>> tls_ca_list = "/path/to/file"
>> tls_method  = tlsv1
>>
>> for details please consult
>> http://www.opensips.org/html/docs/tutorials/tls-1.4.x.html
>> <https://contactmonkey.com/api/v1/tracker?cm_session=fe1ad39b-b209-487a-ae7d-5dc3874a3f4b&cm_type=link&cm_link=4c658b68-ff08-42fc-abc9-b28ade77429a&cm_destination=http://www.opensips.org/html/docs/tutorials/tls-1.4.x.html>
>>
>> Regards
>> Hamid R. Hashmi
>>
>> --
>> Date: Thu, 7 Apr 2016 13:14:28 -0400
>> From: ali...@gmail.com
>> To: users@lists.opensips.org
>> Subject: [OpenSIPS-Users] TLS - Certificate Validation Failure error on
>> SIP Phones - OpenSIPS version 1.11.5
>>
>>
>> Hello,
>>
>> My opensips server is just a registrar server and I have enabled tls with
>> the following settings:
>>
>> listen=tls:xx.xx.xx.xx:5061
>> disable_tls=no

Re: [OpenSIPS-Users] TLS - Certificate Validation Failure error on SIP Phones - OpenSIPS version 1.11.5

[Ali Pey]

Hello Hamid,

The parameters below don't have any effects. In my scenario, the sip phones
are rejecting the tls connection by saying "Certificate Validation Failure".

Neither of parameters below had any effects.


Anyone else has any idea what I need to look for?

Regards,
Ali Pey


On Fri, Apr 8, 2016 at 4:00 AM, Hamid Hashmi 
wrote:

> Please define  following values
>
> tls_ca_list = "/path/to/file"
> tls_method  = tlsv1
>
> for details please consult
> http://www.opensips.org/html/docs/tutorials/tls-1.4.x.html
> <https://contactmonkey.com/api/v1/tracker?cm_session=fe1ad39b-b209-487a-ae7d-5dc3874a3f4b&cm_type=link&cm_link=4c658b68-ff08-42fc-abc9-b28ade77429a&cm_destination=http://www.opensips.org/html/docs/tutorials/tls-1.4.x.html>
>
> Regards
> Hamid R. Hashmi
>
> --
> Date: Thu, 7 Apr 2016 13:14:28 -0400
> From: ali...@gmail.com
> To: users@lists.opensips.org
> Subject: [OpenSIPS-Users] TLS - Certificate Validation Failure error on
> SIP Phones - OpenSIPS version 1.11.5
>
>
> Hello,
>
> My opensips server is just a registrar server and I have enabled tls with
> the following settings:
>
> listen=tls:xx.xx.xx.xx:5061
> disable_tls=no
> tls_certificate="/etc/opensips/pbx-bundle.crt"
> tls_private_key="/etc/opensips/pbx.key"
>
>
> When my sip phones try to open tls connection, they reject the connection
> saying "Certificate Validation Failure". My certificate is valid and works
> fine on the https website.
>
> What am I missing? What should I look for?
>
> Regards,
> Ali Pey
>
>
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[OpenSIPS-Users] TLS - Certificate Validation Failure error on SIP Phones - OpenSIPS version 1.11.5

[Ali Pey]

Hello,

My opensips server is just a registrar server and I have enabled tls with
the following settings:

listen=tls:xx.xx.xx.xx:5061
disable_tls=no
tls_certificate="/etc/opensips/pbx-bundle.crt"
tls_private_key="/etc/opensips/pbx.key"


When my sip phones try to open tls connection, they reject the connection
saying "Certificate Validation Failure". My certificate is valid and works
fine on the https website.

What am I missing? What should I look for?

Regards,
Ali Pey
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Re: [OpenSIPS-Users] Connection IP in SDP is messed up after RTPPRoxy Offer

[Ali Pey]

Hello Aqs,

Yes, that was my problem. I didn't need fix_nated_sdp.

Thanks,
Ali


On Fri, Feb 26, 2016 at 4:23 PM, Aqs Younas  wrote:

> Calling fix_nated_sdp() and rtpproxy_offer() after one another does not
> make sense since both do same things in some cases. Make sure you have just
> one in route.
> On 27-Feb-2016 1:39 am, "Alex Balashov"  wrote:
>
>> Ali,
>>
>> Is there any danger that you are calling rtpproxy_offer() twice, or using
>> rtpproxy_offer() in combination with fix_nated_sdp()[1]?
>>
>> -- Alex
>>
>> [1]
>> http://www.opensips.org/html/docs/modules/2.1.x/nathelper.html#id293899
>> 
>>
>> --
>> Alex Balashov | Principal | Evariste Systems LLC
>> 303 Perimeter Center North, Suite 300
>> Atlanta, GA 30346
>> United States
>>
>> Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct)
>> Web:
>> 
>> http://www.evaristesys.com/
>> ,
>>
>> 
>> http://www.csrpswitch.com/
>> 
>>
>> ___
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>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>> 
>>
>
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>
>
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Re: [OpenSIPS-Users] Connection IP in SDP is messed up after RTPPRoxy Offer

[Ali Pey]

Hello Alex,

Good to hear from you.

Yes, I did have  fix_nated sdp and that was causing conflict. Copy and
paste issues:)

Thanks,
Ali


On Fri, Feb 26, 2016 at 3:39 PM, Alex Balashov 
wrote:

> Ali,
>
> Is there any danger that you are calling rtpproxy_offer() twice, or using
> rtpproxy_offer() in combination with fix_nated_sdp()[1]?
>
> -- Alex
>
> [1]
> http://www.opensips.org/html/docs/modules/2.1.x/nathelper.html#id293899
> 
>
> --
> Alex Balashov | Principal | Evariste Systems LLC
> 303 Perimeter Center North, Suite 300
> Atlanta, GA 30346
> United States
>
> Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct)
> Web:
> 
> http://www.evaristesys.com/
> ,
>
> 
> http://www.csrpswitch.com/
> 
>
>
> ___
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> 
>
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[OpenSIPS-Users] Connection IP in SDP is messed up after RTPPRoxy Offer

[Ali Pey]

Hello,

Sometime with certain clients the connection IP in the SDP in the Invite
message get's messed up (corrupted) after rtpproxy_offer:

Incoming Invite from public interface:
c=IN IP4 xx.xx.xx.xx

Outgoing Invite from opensips after rtpproxy_offer is:
c=IN IP4 xx.xx.xx.xxyy.yy.yy.yy

Basically instead of replacing the external IP, it adds the internal IP to
the end of connection line in the SDP. I'm using opensips version 1.11.5.
It works with most clients but this happens time to time.


Has anyone experienced this problem? How can I fix it?


Regards,
Ali Pey
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Re: [OpenSIPS-Users] apt.opensips.org not reachable

[Ali Pey]

Is there an alternative way to get the debian packages?

Thanks,
Ali


On Tue, Dec 15, 2015 at 4:28 AM, Frederik Bjerggaard Nielsen <
f...@firstcom.dk> wrote:

> Hello,
>
>
>
> It has been down for over a week now.. We are interested in hosting the
> Debian/Ubuntu repository for some period, so if somebody could contact me
> about that, that would be great.
>
>
>
> *---*
>
> *Best regards*
>
>
>
> Frederik Bjerggard Nielsen
>
> Technical Specialist
>
>
>
> *Firstcom A/S*
>
>
>
> *Fra:* users-boun...@lists.opensips.org [mailto:
> users-boun...@lists.opensips.org] *På vegne af *Ali Pey
> *Sendt:* 14. december 2015 20:16
> *Til:* OpenSIPS users mailling list 
> *Emne:* [OpenSIPS-Users]
> <https://contactmonkey.com/api/v1/tracker?cm_session=02685dfe-75b5-4dfe-b26e-98617454a764&cm_type=link&cm_link=41b5639c-acbb-4768-88a3-2c6a0e15e47b&cm_destination=http://apt.opensips.org>
> apt.opensips.org
> <https://contactmonkey.com/api/v1/tracker?cm_session=02685dfe-75b5-4dfe-b26e-98617454a764&cm_type=link&cm_link=884324f6-7457-4eca-a382-481fbbeedbef&cm_destination=http://apt.opensips.org>
> not reachable
>
>
>
> Hello,
>
>
>
> Official OpenSIPS Debian/Ubuntu repository is not reachable. Can someone
> please have a look? Is there a work around?
>
>
>
>
> <https://contactmonkey.com/api/v1/tracker?cm_session=77e8c2b6-04e1-4b8a-adab-352f7984cb88&cm_type=link&cm_link=1d66d86e-2591-4d04-9f0a-cf07e4310a99&cm_destination=http://apt.opensips.org>
> http://apt.opensips.org
> <https://contactmonkey.com/api/v1/tracker?cm_session=02685dfe-75b5-4dfe-b26e-98617454a764&cm_type=link&cm_link=ee51b14f-83f2-4184-a2f0-e48451c3ed01&cm_destination=http://apt.opensips.org>
>
>
>
> Regards,
>
> Ali Pey
>
>
>
> [image:
> https://contactmonkey.com/api/v1/tracker?cm_session=77e8c2b6-04e1-4b8a-adab-352f7984cb88&cm_type=open&cm_user_email=ali...@gmail.com][image:
> https://contactmonkey.com/api/v1/tracker?cm_session=77e8c2b6-04e1-4b8a-adab-352f7984cb88&cm_type=open&cm_user_email=ali...@gmail.com]
>
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[OpenSIPS-Users] apt.opensips.org not reachable

[Ali Pey]

Hello,

Official OpenSIPS Debian/Ubuntu repository is not reachable. Can someone
please have a look? Is there a work around?

<https://contactmonkey.com/api/v1/tracker?cm_session=77e8c2b6-04e1-4b8a-adab-352f7984cb88&cm_type=link&cm_link=3228142e-2bca-412d-8846-9ccd46d5c0e6&cm_destination=http://apt.opensips.org/>
http://apt.opensips.org
<https://contactmonkey.com/api/v1/tracker?cm_session=77e8c2b6-04e1-4b8a-adab-352f7984cb88&cm_type=link&cm_link=1d66d86e-2591-4d04-9f0a-cf07e4310a99&cm_destination=http://apt.opensips.org>

Regards,
Ali Pey
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Re: [OpenSIPS-Users] Dialplan module and priority

[Ali Pey]

Thank you for the responses.

I have alternated priorities and every other thing I could think of and it
would not change the behavior.

It always picks the first match if it's a regex type (match_op=1). It works
as expected if it's equal type (match_op=0)

Priority doesn't seem to kick in when it is a regex.



On Fri, Jul 17, 2015 at 7:19 AM, Muhammad Danish Moosa <
danishmo...@gmail.com> wrote:

> Hi.
>
> Higher the priority value means 888444's selection should be prioritized.
> Did you reload the config , have used similar rules and worked.
>
> On Fri, Jul 17, 2015 at 3:01 AM, SamyGo  wrote:
>
>> Hi,
>> Swapping the priority works the way you want ? I have a feeling this
>> makes sense (just like an ACL or firewall rules) ^888444* should get called
>> before ^888* .
>>
>>
>>
>> On Thu, Jul 16, 2015 at 5:47 PM, Ali Pey  wrote:
>>
>>> Hello,
>>>
>>> Let's say I have the two following rules in my dialplan table:
>>>
>>> Rule 1:
>>> - match_op = 1 (regexp)
>>> - Priority=1
>>> - match_exp = "^888[0-9]{7}"
>>>
>>> Rule 2:
>>> - match_op = 1 (regexp)
>>> - Priority=2
>>> - match_exp = "^888444[0-9]{4}"
>>>
>>>
>>> If I use dp_translate for 8882221234, it matches rule 1 and that's good.
>>> If I use dp_translate for 8884441234, it still matches rule 1 and
>>> that's not good. It should match rule 2.
>>>
>>>
>>> Is this a bug or expected behaviour?
>>>
>>> Is there a way I can work around this?
>>>
>>> Thanks,
>>> Ali Pey
>>>
>>>
>>>
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>>>
>>>
>>
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>>
>>
>
>
> --
> Muhammad Danish Moosa
>
> " The core of mans' spirit comes from new experiences. "___ Christopher
> McCandless
>
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[OpenSIPS-Users] Dialplan module and priority

[Ali Pey]

Hello,

Let's say I have the two following rules in my dialplan table:

Rule 1:
- match_op = 1 (regexp)
- Priority=1
- match_exp = "^888[0-9]{7}"

Rule 2:
- match_op = 1 (regexp)
- Priority=2
- match_exp = "^888444[0-9]{4}"


If I use dp_translate for 8882221234, it matches rule 1 and that's good.
If I use dp_translate for 8884441234, it still matches rule 1 and that's
not good. It should match rule 2.


Is this a bug or expected behaviour?

Is there a way I can work around this?

Thanks,
Ali Pey
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Re: [OpenSIPS-Users] DRouting and Range of numbers

[Ali Pey]

Hi Razvan.

Thank you for the response.

Regards,
Ali Pey


On Fri, Mar 27, 2015 at 4:29 AM, Răzvan Crainea  wrote:

>  Hi, Ali!
>
> Unfortunately drouting works only prefix based, so unless you write
> specific rules for the 5 numbers (i.e. add rules for each number from 
> 8881231231
> to 8881231235 in the table), you cannot achieve this.
>
> Best regards,
>
> Răzvan Crainea
> OpenSIPS Solutions 
> <https://contactmonkey.com/api/v1/tracker?cm_session=40f28dbf-89e8-4f38-a459-74a6eda3cceb&cm_type=link&cm_link=03e3d0aa-2062-4b6e-8fdd-39912818fef5&cm_destination=http://www.opensips-solutions.com>www.opensips-solutions.com
>  
> <https://contactmonkey.com/api/v1/tracker?cm_session=40f28dbf-89e8-4f38-a459-74a6eda3cceb&cm_type=link&cm_link=639b844a-b05b-4c11-a1de-ce2953ee3d5c&cm_destination=http://www.opensips-solutions.com>
>
> On 03/26/2015 08:29 PM, Ali Pey wrote:
>
> Hello,
>
>  Is it possible to have a rule with a range of numbers in Dynamic routing?
>
>  For instance I want 8881231231 to 5 to be routed to a specific gw. Can I
> do this with one rule only?
> I don't want 8881231236 to 9 to be routed to that gateway.
>
>  Thanks,
> Ali Pey
>
>
>
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[OpenSIPS-Users] DRouting and Range of numbers

[Ali Pey]

Hello,

Is it possible to have a rule with a range of numbers in Dynamic routing?

For instance I want 8881231231 to 5 to be routed to a specific gw. Can I do
this with one rule only?
I don't want 8881231236 to 9 to be routed to that gateway.

Thanks,
Ali Pey
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Re: [OpenSIPS-Users] Block user from registration

[Ali Pey]

You can also consider using the permissions module. If the src IP is there,
then you can accept the request, otherwise, drop the message.

Regards,
Ali Pey


On Wed, Dec 31, 2014 at 1:30 PM, Duane Larson 
wrote:

> My logic saves the user that is registering into the location table
> without challenging them for a password or checking that the user or domain
> is local to the OpenSIPs instance.  If you are looking for something more
> you might want to provide more detail.
>
> This would allow fake accounts to register if they are from a friendly IP.
>
>
> On Wednesday, December 31, 2014, Satish Patel 
> wrote:
>
>> How it will help if i want to allow only IP auth for specific user but
>> not registration auth? How your logic deal with User level?
>>
>>
>> On Wed, Dec 31, 2014 at 12:22 PM, Duane Larson 
>> wrote:
>>
>>> Would you not just do something like this?
>>>
>>> If(FriendlyIP && is_method("REGISTER"))
>>> {
>>> if (t_newtran()) {
>>> save("location");
>>> }
>>>
>>> exit;
>>> }
>>>
>>> On Wed, Dec 31, 2014 at 10:22 AM, Satish Patel 
>>> wrote:
>>>
>>>> Hi,
>>>>
>>>> We have many users using both registration method and IP auth method to
>>>> send calls but i wants if they use IP Auth method then we can disable
>>>> registration method ( just prevention from hacking attack).
>>>>
>>>> I believe registration is only required for incoming calls to find user
>>>> location, right? How do i tell opensips don't accept user registration
>>>> method even opensips challenge for proxy auth. any suggestion?
>>>>
>>>> ___
>>>> Users mailing list
>>>> Users@lists.opensips.org
>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>> <https://contactmonkey.com/api/v1/tracker?cm_session=2fbf233d-c6f2-4110-af3e-5cf568144048&cm_type=link&cm_link=fe5ffb6c-5456-4123-ac5e-ec6209f257ac&cm_destination=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://contactmonkey.com/api/v1/tracker?cm_session=2fbf233d-c6f2-4110-af3e-5cf568144048&cm_type=link&cm_link=b28e5084-e043-4bf9-916b-9dab1b052364&cm_destination=http://lists.opensips.org/cgi-bin/mailman/listinfo/users>
>>>
>>>
>>
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Re: [OpenSIPS-Users] RTP Proxy and Re-Invites

[Ali Pey]

Hi Razvan,

Thank you for your response and it makes sense.

I will search for a work around for media re-negotiation rejection and will
post my results here.

Best regards,
Ali Pey


On Fri, Nov 14, 2014 at 4:45 AM, Răzvan Crainea  wrote:

>  Hi, Ali!
>
> The reINVITES should be handled similar to INVITEs, with an offer for the
> request and an answer for the reply. But indeed, there's no way to recover
> from a rejected reINVITE.
> Actually the problem is a bit deeper and I don't really know how this
> should be handled. It depends on whether the client sends the re-invite to
> create a new media stream, or update the old one. In the first case,
> RTPProxy should create a new media stream and if that one fails, only
> delete the latter one - this is not supported by RTPProxy. In the second
> case, the entire media should be seized, and this is how it works now :).
>
> Best regards,
>
> Răzvan Crainea
> OpenSIPS Solutions 
> <https://contactmonkey.com/api/v1/tracker?cm_session=e1c3197b-a7bb-42f0-9b6f-814623288ae7&cm_type=link&cm_link=65f31eb4-983e-4f92-bfef-1acb9dfcef3c&cm_destination=http://www.opensips-solutions.com>www.opensips-solutions.com
>  
> <https://contactmonkey.com/api/v1/tracker?cm_session=e1c3197b-a7bb-42f0-9b6f-814623288ae7&cm_type=link&cm_link=655752ea-cb3a-434b-b241-a5c486c33b11&cm_destination=http://www.opensips-solutions.com>
>
> On 11/14/2014 03:30 AM, Jeff Pyle wrote:
>
> Ali,
>
>  This is what I use within loose_route() to handle rtpproxy.  In my
> particular case I'm bridging between two interfaces.
>
>
>if (has_body("application/sdp"))  {
># Begin rtp session update gyrations
>   if (method == "INVITE") {
>   # INVITE w/ SDP, so early neg
>   # This is offer, reply is answer
>   rtpproxy_offer("frocl");
>   t_on_reply("1");
>   } else if (method == "ACK") {
>   # ACK w/ SDP, so late neg (done now)
>   # This is answer
>   rtpproxy_answer("frocl");
>   }
>   } else {
>   if (method == "INVITE") {
>   # INVITE w/o SDP, so late neg
>   # This is nothing, reply is offer, ACK is answer (~5
> lines up)
>   t_on_reply("2");
>   } else {
>   t_on_reply("1");
>   }
>   }
>
>
>
>  And then:
>
>
>   onreply_route[1] {  # Manage content as necessary - early
> negotiation
>  if (has_body("application/sdp")) {
>  if ($DLG_status) {  # Update
>  rtpproxy_answer("frocl");
>  } else {# New
>  rtpproxy_answer("froc");
>  }
>  }
>  }
>
>
>   onreply_route[2] {  # Manage content as necessary -
> late negotiation - update
>  if (has_body("application/sdp")) {
>  rtpproxy_offer("frocl");
>  }
>  }
>
>
>
>
>  This does not handle rejected reinvites.  I don't know how to solve that
> one.  Otherwise, it works well, although in some spots it may be overkill.
>
>
>  - Jeff
>
>
>
> On Thu, Nov 13, 2014 at 5:13 PM, Ali Pey  wrote:
>
>> Hello,
>>
>>  What's the best way of handling rtpproxy with re-invites?
>>
>>  Should I do unforce and then offer/answer? What if the re-invite gets
>> rejected?
>>
>>  Any help appreciated.
>>
>>  Thanks,
>> Ali Pey
>>
>>
>> ___
>> Users mailing list
>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>> <https://contactmonkey.com/api/v1/tracker?cm_session=e1c3197b-a7bb-42f0-9b6f-814623288ae7&cm_type=link&cm_link=8cb3bf89-22b0-41ce-b966-be6d39cbf9c8&cm_destination=http://lists.opensips.org/cgi-bin/mailman/listinfo/users>
>>
>>
>
>
> ___
> Users mailing 
> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>  
> <https://contactmonkey.com/api/v1/tracker?cm_session=e1c3197b-a7bb-42f0-9b6f-814623288ae7&cm_type=link&cm_link=a8b09eef-0bbb-47f8-a71e-ad17e0e94859&cm_destination=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://contactmonkey.com/api/v1/tracker?cm_session=e1c3197b-a7bb-42f0-9b6f-814623288ae7&cm_type=link&cm_link=46daedd8-b7b3-451e-9d4b-adafaf98c6ed&cm_destination=http://lists.opensips.org/cgi-bin/mailman/listinfo/users>
>
>
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Re: [OpenSIPS-Users] RTP Proxy and Re-Invites

[Ali Pey]

Thank you Jeff.

Don't I need to do unforce before doing a new offer? Why?

Regards,
Ali Pey


On Thu, Nov 13, 2014 at 8:30 PM, Jeff Pyle  wrote:

> Ali,
>
> This is what I use within loose_route() to handle rtpproxy.  In my
> particular case I'm bridging between two interfaces.
>
>
> if (has_body("application/sdp"))  {
>  # Begin rtp session update gyrations
>  if (method == "INVITE") {
>  # INVITE w/ SDP, so early neg
>  # This is offer, reply is answer
>  rtpproxy_offer("frocl");
>  t_on_reply("1");
>  } else if (method == "ACK") {
>  # ACK w/ SDP, so late neg (done now)
>  # This is answer
>  rtpproxy_answer("frocl");
>  }
>  } else {
>  if (method == "INVITE") {
>  # INVITE w/o SDP, so late neg
>  # This is nothing, reply is offer, ACK is answer (~5
> lines up)
>  t_on_reply("2");
>  } else {
>  t_on_reply("1");
>  }
>  }
>
>
>
> And then:
>
>
> onreply_route[1] {  # Manage content as necessary - early negotiation
> if (has_body("application/sdp")) {
> if ($DLG_status) {  # Update
> rtpproxy_answer("frocl");
> } else {# New
> rtpproxy_answer("froc");
> }
> }
> }
>
>
> onreply_route[2] {  # Manage content as necessary - late negotiation -
> update
> if (has_body("application/sdp")) {
> rtpproxy_offer("frocl");
> }
> }
>
>
>
>
> This does not handle rejected reinvites.  I don't know how to solve that
> one.  Otherwise, it works well, although in some spots it may be overkill.
>
>
> - Jeff
>
>
>
> On Thu, Nov 13, 2014 at 5:13 PM, Ali Pey  wrote:
>
>> Hello,
>>
>> What's the best way of handling rtpproxy with re-invites?
>>
>> Should I do unforce and then offer/answer? What if the re-invite gets
>> rejected?
>>
>> Any help appreciated.
>>
>> Thanks,
>> Ali Pey
>>
>>
>> ___
>> Users mailing list
>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>> <https://contactmonkey.com/api/v1/tracker?cm_session=a99d3828-b6f0-4ff4-96a7-8f72caed13fe&cm_type=link&cm_link=6bd7525a-e72b-498e-8d0a-c7de613ceb37&cm_destination=http://lists.opensips.org/cgi-bin/mailman/listinfo/users>
>>
>>
>
> ___
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> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> <https://contactmonkey.com/api/v1/tracker?cm_session=a99d3828-b6f0-4ff4-96a7-8f72caed13fe&cm_type=link&cm_link=9c8bdfc7-56bf-4fc5-ae29-d8fa49edc371&cm_destination=http://lists.opensips.org/cgi-bin/mailman/listinfo/users>
>
>
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[OpenSIPS-Users] RTP Proxy and Re-Invites

[Ali Pey]

Hello,

What's the best way of handling rtpproxy with re-invites?

Should I do unforce and then offer/answer? What if the re-invite gets
rejected?

Any help appreciated.

Thanks,
Ali Pey
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Re: [OpenSIPS-Users] OpenSIPS/RTpProxy BridgeMode after failure route

[Ali Pey]

Hello Răzvan & Salman,

Thank you for your responses.

I was able to fix it by moving rtpproxy_offer to branch route instead of
having it in the main route. In failure route, I only needed to do unforce.

Regards,
Ali Pey


On Thu, Oct 9, 2014 at 3:31 AM, Răzvan Crainea  wrote:

>  Hi, Ali!
>
> For the initial branch (in request route) are you using engage_rtpproxy()?
> If so, try to use rtpproxy_offer().
>
> Best regards,
>
> Răzvan Crainea
> OpenSIPS Solutions 
> <https://contactmonkey.com/api/v1/tracker?cm_session=f72d879b-e536-4227-a59d-8a6c4b8ba6e9&cm_type=link&cm_link=1443c182-b295-4010-9767-59bc588bef8a&cm_destination=http://www.opensips-solutions.com>www.opensips-solutions.com
>  
> <https://contactmonkey.com/api/v1/tracker?cm_session=f72d879b-e536-4227-a59d-8a6c4b8ba6e9&cm_type=link&cm_link=4fe71057-5c3d-4ff5-89ac-a63e63631bb1&cm_destination=http://www.opensips-solutions.com>
>
> On 10/09/2014 12:06 AM, Ali Pey wrote:
>
>  Hello Salman,
>
>  Can you please elaborate on how you got this working? I have the same
> problem and can't get it to work.
>
>  In failure route, I do:
> unforce_rtp_proxy()
>  Then when I have a new destination, I do:
> rtpproxy_offer("rocie");
>
>  However, I end up with messed up SDP, in my second invite. It doesn't
> remove the old IP addresses and only adds the IP addresses again:
>  o=Sonus_UAC 9216 20203 IN IP4 10.160.11.16210.160.11.162a Capabilities
> c=IN IP4 10.160.11.16210.160.11.162udio 2311822970AVP 0 8 100
>
>
>  Please let me know how I can fix this.
>
>  Thanks.
>
>
> On Mon, Jan 6, 2014 at 10:26 AM, Salman Zafar 
> wrote:
>
>>  Hi Razvan,
>>  I got it working without branching, after banging head a lot I
>> got to learn unforcing drops the media ports for previous rtpproxy
>> offer/answer and after that directing the new flow though rtpproxy flags,IP
>> media works. I am able to traverse from eternal to internal play media and
>> then on failure do external to external with media flowing between public
>> interfaces. Just wondering if you know this method or certify.
>>
>>
>>
>> On Mon, Jan 6, 2014 at 4:35 PM, Răzvan Crainea 
>> wrote:
>>
>>> Hi, Salman!
>>>
>>> The sockets used by RTPProxy are created when the session is started
>>> (the first offer) and cannot be updated afterwards. Therefore the only
>>> solution I can see is to configure a per branch scenario, as you mentioned.
>>>
>>> Best regards,
>>>
>>> Razvan Crainea
>>> OpenSIPS Core Developer
>>> http://www.opensips-solutions.com
>>> <https://contactmonkey.com/api/v1/tracker?cm_session=66f49ebf-f052-47c3-adee-bf8dd17afa5d&cm_type=link&cm_link=c1574c01-908b-4910-aaff-83f9f8f63efd&cm_destination=http://www.opensips-solutions.com>
>>>
>>>
>>> On 12/30/2013 01:11 PM, Salman Zafar wrote:
>>>
>>>>  Hi,
>>>> I have a scenario of playing media at a private-ip media server and
>>>> send BUSY, next in failure route bridge call to a public IP. (SIP to
>>>> SIP).
>>>>
>>>> So the scenario is as follows:
>>>>
>>>> UA(Phone1) -> OpenSIPS/RTpProxy(ei) -> Media-Server (Private IP) -> BUSY
>>>> -> OpenSIPS(failure route) -> RTpProxy(ee) -> lookup -> (UA Phone2)
>>>>
>>>> Now the problem is RtpProxy is being offered (EI flags) in first case
>>>> where routing to Media sever at private IP, after failure it is again
>>>> used with (EE flags), also in corresponding replies.
>>>>
>>>> The second time RTpProxy does not effect SDP c= and ports in a way to
>>>> build media communication. SDP fix directly does not effect rtp ports.
>>>>
>>>> Is there any way of using RtpProxy differently in fail-over, or I have
>>>> to go for rtpproxy per branch?.
>>>>
>>>>
>>>> Thanks in advance.
>>>>
>>>> --
>>>> Regards
>>>>
>>>> Salman
>>>>
>>>>
>>>>
>>>>  ___
>>>> Users mailing list
>>>> Users@lists.opensips.org
>>>>
>>>> <https://contactmonkey.com/api/v1/tracker?cm_session=66f49ebf-f052-47c3-adee-bf8dd17afa5d&cm_type=link&cm_link=0138242a-38a8-4160-8e04-9e778d8a3ff2&cm_destination=http://lists.opensips.org/cgi->
>>>> http://lists.opensips.org/cgi-
>

Re: [OpenSIPS-Users] OpenSIPS/RTpProxy BridgeMode after failure route

[Ali Pey]

Hello Salman,

Can you please elaborate on how you got this working? I have the same
problem and can't get it to work.

In failure route, I do:
unforce_rtp_proxy()
Then when I have a new destination, I do:
rtpproxy_offer("rocie");

However, I end up with messed up SDP, in my second invite. It doesn't
remove the old IP addresses and only adds the IP addresses again:
o=Sonus_UAC 9216 20203 IN IP4 10.160.11.16210.160.11.162a Capabilities
c=IN IP4 10.160.11.16210.160.11.162udio 2311822970AVP 0 8 100


Please let me know how I can fix this.

Thanks.


On Mon, Jan 6, 2014 at 10:26 AM, Salman Zafar  wrote:

> Hi Razvan,
> I got it working without branching, after banging head a lot I got
> to learn unforcing drops the media ports for previous rtpproxy offer/answer
> and after that directing the new flow though rtpproxy flags,IP media works.
> I am able to traverse from eternal to internal play media and then on
> failure do external to external with media flowing between public
> interfaces. Just wondering if you know this method or certify.
>
>
>
> On Mon, Jan 6, 2014 at 4:35 PM, Răzvan Crainea 
> wrote:
>
>> Hi, Salman!
>>
>> The sockets used by RTPProxy are created when the session is started (the
>> first offer) and cannot be updated afterwards. Therefore the only solution
>> I can see is to configure a per branch scenario, as you mentioned.
>>
>> Best regards,
>>
>> Razvan Crainea
>> OpenSIPS Core Developer
>> http://www.opensips-solutions.com
>> 
>>
>>
>> On 12/30/2013 01:11 PM, Salman Zafar wrote:
>>
>>> Hi,
>>> I have a scenario of playing media at a private-ip media server and
>>> send BUSY, next in failure route bridge call to a public IP. (SIP to
>>> SIP).
>>>
>>> So the scenario is as follows:
>>>
>>> UA(Phone1) -> OpenSIPS/RTpProxy(ei) -> Media-Server (Private IP) -> BUSY
>>> -> OpenSIPS(failure route) -> RTpProxy(ee) -> lookup -> (UA Phone2)
>>>
>>> Now the problem is RtpProxy is being offered (EI flags) in first case
>>> where routing to Media sever at private IP, after failure it is again
>>> used with (EE flags), also in corresponding replies.
>>>
>>> The second time RTpProxy does not effect SDP c= and ports in a way to
>>> build media communication. SDP fix directly does not effect rtp ports.
>>>
>>> Is there any way of using RtpProxy differently in fail-over, or I have
>>> to go for rtpproxy per branch?.
>>>
>>>
>>> Thanks in advance.
>>>
>>> --
>>> Regards
>>>
>>> Salman
>>>
>>>
>>>
>>> ___
>>> 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
>
> M. Salman Zafar
>
> VoIP Professional
>
>
> ___
> Users mailing list
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> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> 
>
>
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Re: [OpenSIPS-Users] engage_rtp_proxy and unforce_rtp_proxy

[Ali Pey]

Thank you Razvan. Great info.

Regards,
Ali Pey


On Wed, Jul 16, 2014 at 3:28 AM, Răzvan Crainea  wrote:

> Hi, Ali!
>
> Rtpproxy offers an interface to communicate with it over network, through
> the communication socket (the -s parameter or the RTPProxy server). You can
> send over UDP a message similar to "COOKIE I", and RTPProxy will dump all
> the ongoing sessions.
>
> Best regards,
>
> Razvan Crainea
> OpenSIPS Core Developer
> http://www.opensips-solutions.com
> <https://contactmonkey.com/api/tracker?cm_session=5e5176d0-8684-4129-bfe8-d542b635dc35&cm_type=link&cm_link=d1713ac3-6a63-4069-9850-38035bec249d&cm_destination=http://www.opensips-solutions.com>
>
>
> On 07/15/2014 08:59 PM, Bogdan-Andrei Iancu wrote:
>
>> Ali,
>>
>> In rtpproxy, maybe Razvan can help you.
>>
>> Otherwise you can mark the dialogs in OpenSIPS (with profiles ) and
>> check it via MI.
>>
>> Regards,
>>
>> Bogdan-Andrei Iancu
>> OpenSIPS Founder and Developer
>> http://www.opensips-solutions.com
>> <https://contactmonkey.com/api/tracker?cm_session=5e5176d0-8684-4129-bfe8-d542b635dc35&cm_type=link&cm_link=cb51d41f-0a04-4285-bef4-32be1a20fc67&cm_destination=http://www.opensips-solutions.com>
>>
>> On 15.07.2014 20:51, Ali Pey wrote:
>>
>>> Thanks Bogdan.
>>>
>>> Is there a way to see active sessions on the rtpproxy instances? I'm
>>> running rtpproxy on a different server on multiple interfaces in
>>> bridge mode.
>>>
>>> Regards,
>>> Ali
>>>
>>>
>>> On Tue, Jul 15, 2014 at 1:46 PM, Bogdan-Andrei Iancu
>>> mailto:bog...@opensips.org>> wrote:
>>>
>>> Hi Ali,
>>>
>>> Nope...when using engage, the re-arming (at re-invites) and
>>> termination (when dialog is ended) are automatically done.
>>>
>>> Regards,
>>>
>>> Bogdan-Andrei Iancu
>>> OpenSIPS Founder and Developer
>>> http://www.opensips-solutions.com
>>> <https://contactmonkey.com/api/tracker?cm_session=5e5176d0-8684-4129-bfe8-d542b635dc35&cm_type=link&cm_link=c248a9e7-ef93-4cf2-a1ca-03ca2a48c618&cm_destination=http://www.opensips-solutions.com>
>>>  <
>>> <https://contactmonkey.com/api/tracker?cm_session=e8bf4d0b-e9f9-42da-8832-6272fe80eeb4&cm_type=link&cm_link=dd652bed-9e0d-4b92-a52b-d42fb12badce&cm_destination=http://www.opensips-solutions.com>
>>> https://contactmonkey.com/
>>> <https://contactmonkey.com/api/tracker?cm_session=5e5176d0-8684-4129-bfe8-d542b635dc35&cm_type=link&cm_link=292cfe57-d72d-406e-82d3-8cdd90bd5736&cm_destination=https://contactmonkey.com/>
>>> api/tracker?cm_session=e8bf4d0b-e9f9-42da-8832-
>>> 6272fe80eeb4&cm_type=link&cm_link=dd652bed-9e0d-4b92-a52b-
>>> d42fb12badce&cm_destination=http://www.opensips-solutions.com>
>>>
>>>
>>> On 15.07.2014 20:42, Ali Pey wrote:
>>>
>>>> Hello,
>>>>
>>>> If I use engage_rtp_proxy, do I still need to
>>>> do unforce_rtp_proxy when the call ends?
>>>>
>>>> Also, what are the differences between using engage_rtp_proxy
>>>> vs rtpproxy_offer and rtpproxy_answer?
>>>>
>>>> Thanks,
>>>> Ali Pey
>>>>
>>>>
>>>>
>>>> ___
>>>> Users mailing list
>>>> Users@lists.opensips.org  <mailto:Users@lists.opensips.org>
>>>>
>>>> <https://contactmonkey.com/api/tracker?cm_session=5e5176d0-8684-4129-bfe8-d542b635dc35&cm_type=link&cm_link=ef5a3694-9bbd-4a42-a5f2-641f305a09a5&cm_destination=http://lists.opensips.org/cgi-bin/mailman/listinfo/users>
>>>> http://lists.opensips.org/cgi-
>>>> <https://contactmonkey.com/api/tracker?cm_session=5e5176d0-8684-4129-bfe8-d542b635dc35&cm_type=link&cm_link=2fab34c6-0680-4fab-8c7a-551bf88fcfea&cm_destination=http://lists.opensips.org/cgi->bin/mailman/listinfo/users
>>>>  <
>>>> <https://contactmonkey.com/api/tracker?cm_session=e8bf4d0b-e9f9-42da-8832-6272fe80eeb4&cm_type=link&cm_link=e9696622-d31d-43ef-9051-78c4e36b193a&cm_destination=http://lists.opensips.org/cgi-bin/mailman/listinfo/users>
>>>> https://contactmonkey.com/
>>>> <https://contactmonkey.com/api/tracker?cm_session=5e5176d0-8684-4129-bfe8-d542b635dc35&cm_type=link&cm_link=bed42b95-f0dc-46b5-b2df-31920f40e93d&cm_destination=https://contactmonkey.com/>
>>>> api/tracker?cm_session=e8bf4d0b-e9f9-42da-8832-
>>>> 6272fe80eeb4&cm_type=link&cm_link=e9696622-d31d-43ef-9051-
>>>> 78c4e36b193a&cm_destination=http://lists.opensips.org/cgi-
>>>> bin/mailman/listinfo/users>
>>>>
>>>
>>>
>>>
>>
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[OpenSIPS-Users] engage_rtp_proxy and unforce_rtp_proxy

[Ali Pey]

Hello,

If I use engage_rtp_proxy, do I still need to do unforce_rtp_proxy when the
call ends?

Also, what are the differences between using engage_rtp_proxy
vs rtpproxy_offer and rtpproxy_answer?

Thanks,
Ali Pey
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[OpenSIPS-Users] Load balancer module: lb_count_call - what if I don't know the group id

[Ali Pey]

Hello,

In load balancer module, I can use lb_count_call(ip,port,grp,resources) to
add the current call to the resource of the specified server.

What if I don't know the group id, that this server belongs to, if I pass
-1, would it add the call to the right group?

Thanks,
Ali
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Re: [OpenSIPS-Users] Issue in VIA Header

[Ali Pey]

Are you running on a VM environment? If so, try your script on a bare metal
hardware. VMs mess with OS quite a bit.


Regards,
Ali Pey



On Tue, Apr 15, 2014 at 1:33 PM, B. Buitenhuis wrote:

>  Hi ,
>
>  I'm not using any aliases in my config and I've tested the config with
> different machines in different networks.
>
>  About the messages
> They are set properly before they reach open sips. Weird thing is that
> after running the open sips deamon (about a hour) the problem looks solved.
>
>
>   Van: Ali Pey 
>
> Beantwoorden - Aan: OpenSIPS users mailling list  >
> Datum: dinsdag 15 april 2014 19:29
>
> Aan: OpenSIPS users mailling list 
> Onderwerp: Re: [OpenSIPS-Users] Issue in VIA Header
>
>   Have a look at your system network configuration and aliases in
> opensips?
>
>  Where do these messages come from and where do they go? Have you checked
> if they are set properly before they reach opensips.
>
>
>  Regards,
> Ali Pey
>
>
> On Tue, Apr 15, 2014 at 7:14 AM, B. Buitenhuis wrote:
>
>>  Anyone has any clue where to look?
>>
>>
>>
>>   Van: MT 
>> Beantwoorden - Aan: OpenSIPS users mailling list <
>> users@lists.opensips.org>
>>  Datum: maandag 14 april 2014 12:35
>>
>> Aan: OpenSIPS users mailling list 
>>  Onderwerp: Re: [OpenSIPS-Users] Issue in VIA Header
>>
>>   Also with the latest master branch I've got this problem.
>> After running opensips deamon for an hour the problems seems te be solved.
>>
>>  Is this a known bug?
>>
>>   Van: MT 
>> Beantwoorden - Aan: OpenSIPS users mailling list <
>> users@lists.opensips.org>
>> Datum: zaterdag 12 april 2014 00:09
>> Aan: "users@lists.opensips.org" 
>> Onderwerp: [OpenSIPS-Users] Issue in VIA Header
>>
>>   Hello,
>>
>>  I'm facing the same problem as:
>> https://github.com/OpenSIPS/opensips/issues/190
>>
>>  When I look at a wireshark, I see the following in the VIA header:
>> Via: SIP/2.0/UDP
>> \000\000\000\000\000\000\000\000\000\000\000\000\000:5060;branch=z9hG4bK9414.22d66c76.0
>>
>>  As far as I can see it's only with the CANCEL method, in the config it
>> looks like:
>>
>>  if (method=="CANCEL")
>> {
>>  set_advertised_address(185.XX.XX.XX);
>>
>>  route("PROCESS_RTP_PROXY");
>> unforce_rtp_proxy();
>> if (t_check_trans())
>> t_relay();
>> exit;
>> }
>>
>>
>>  I've tested it with 1.8.3 and 1.10.1
>>
>>  Does anyone has this same problem?
>>
>>
>>  Regards,
>>
>>  Bernard Buitenhuis
>>
>>
>> ___
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>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>
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Re: [OpenSIPS-Users] dlg_end_dlg ALL?

[Ali Pey]

I second this as well. +1

Thanks,
Ali Pey



On Tue, Apr 15, 2014 at 10:48 AM, Jeff Pyle  wrote:

> Very useful indeed.  +1
>
>
>
> - Jeff
>
>
> On Tue, Apr 15, 2014 at 10:28 AM, Brett Nemeroff wrote:
>
>> Hi Liviu,
>> I think this would be a very useful feature as well. Specifically I'd
>> like to be able to say something like
>> :dlg_end_profile
>> 
>>
>> and it'd hang up all calls (dlg_end_dlg method) all calls in that
>> profile. Given that you already know what calls are in a profile, this
>> should be a very efficient way to hangup an entire account, gateway,
>> customer, etc rather than iterating any kind of dialog list.
>>
>> Just my $0.02. Thanks!
>> -Brett
>>
>>
>>
>> On Fri, Mar 28, 2014 at 12:35 PM, Liviu Chircu wrote:
>>
>>> Hello Kneeoh,
>>>
>>> Point taken. I just added this feature on the list!
>>>
>>>
>>> Best regards,
>>>
>>> Liviu Chircu
>>> OpenSIPS Developer
>>> http://www.opensips-solutions.com
>>>
>>> On 03/28/2014 07:30 PM, Kneeoh wrote:
>>>
>>>> mmm. I really don't want to use the fifo. Thats the problem I have now.
>>>> I already have an iterative script to do what you describe. Instead I want
>>>> to use the mi_xmlrpc_ng interface to interact with the dialog module.
>>>> Instead of listing dialogs via the xmlrpc interface and using an external
>>>> parser to get all the dialog ids, I just want to say "kill all dialogs" or
>>>> more specifically, "kill all dialogs with these attributes".
>>>>
>>>> ___
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>>>>
>>>
>>>
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>>>
>>
>>
>> ___
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>>
>>
>
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Re: [OpenSIPS-Users] Issue in VIA Header

[Ali Pey]

Have a look at your system network configuration and aliases in opensips?

Where do these messages come from and where do they go? Have you checked if
they are set properly before they reach opensips.


Regards,
Ali Pey


On Tue, Apr 15, 2014 at 7:14 AM, B. Buitenhuis wrote:

>  Anyone has any clue where to look?
>
>
>
>   Van: MT 
> Beantwoorden - Aan: OpenSIPS users mailling list  >
> Datum: maandag 14 april 2014 12:35
>
> Aan: OpenSIPS users mailling list 
> Onderwerp: Re: [OpenSIPS-Users] Issue in VIA Header
>
>   Also with the latest master branch I've got this problem.
> After running opensips deamon for an hour the problems seems te be solved.
>
>  Is this a known bug?
>
>   Van: MT 
> Beantwoorden - Aan: OpenSIPS users mailling list  >
> Datum: zaterdag 12 april 2014 00:09
> Aan: "users@lists.opensips.org" 
> Onderwerp: [OpenSIPS-Users] Issue in VIA Header
>
>   Hello,
>
>  I'm facing the same problem as:
> https://github.com/OpenSIPS/opensips/issues/190
>
>  When I look at a wireshark, I see the following in the VIA header:
> Via: SIP/2.0/UDP
> \000\000\000\000\000\000\000\000\000\000\000\000\000:5060;branch=z9hG4bK9414.22d66c76.0
>
>  As far as I can see it's only with the CANCEL method, in the config it
> looks like:
>
>  if (method=="CANCEL")
> {
>  set_advertised_address(185.XX.XX.XX);
>
>  route("PROCESS_RTP_PROXY");
> unforce_rtp_proxy();
> if (t_check_trans())
> t_relay();
> exit;
> }
>
>
>  I've tested it with 1.8.3 and 1.10.1
>
>  Does anyone has this same problem?
>
>
>  Regards,
>
>  Bernard Buitenhuis
>
>
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>
>
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Re: [OpenSIPS-Users] OPTIONS over TCP

[Ali Pey]

You don't really need Options for TCP. You use Options to keep UDP
ports/connection open. TCP is a stateful connection. You only need to make
sure that opensips keeps the connection open for the period of the time it
is needed.

is_from_gw should work since it works based on the src_ip.


Regards,
Ali Pey



On Tue, Apr 8, 2014 at 5:40 AM, Vincent DOCQUOIS <
v.docquois.netvi...@gmail.com> wrote:

> Hello all,
>
> I am using Opensips 1.10 for SIP trunking purposes.
> I use DR module in order to route calls to external gateways. One of
> destination gateways is only handling SIP over TCP. By adding
> "transport=tcp" header, I have no problem to use Opensips as a UDP to TCP
> gateway but I encounter 2 problems with keepalive process (using OPTIONS
> method) :
>
> 1) Is there a way to make DR module send OPTIONS over TCP instead of
> default UDP ?
> 2) In the opposite direction, when an incoming TCP OPTIONS is received by
> Opensips from the external gateway, the 'is_from_gw' function seems not to
> be able to identify my external gateway. Note that it is OK in UDP with
> another gateway.
>
> Is it just a limitation or is there any solution to handle OPTIONS over
> TCP ?
>
> Thanks in advance
>
> Vincent
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Re: [OpenSIPS-Users] [RFC] Deprecating mi_xmlrpc

[Ali Pey]

Will this affect OpenSIPS-CP?

Regards,
Ali Pey



On Wed, Mar 19, 2014 at 10:18 AM, Kneeoh  wrote:

> I'm all for the deprecation as long as the documentation on the
> mi_xmlrpc_ng module is updated to a usable level. I find myself referencing
> the documentation for xmlrpc and hoping that it holds true for xmlrpc_ng.
>
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Re: [OpenSIPS-Users] Query about RTP flow

[Ali Pey]

Hello Nandini,

RTP session is negotiated through SDP. If the Invite message has the
OpenSIPS IP address in the SDP, then the client will send the packets to
opensips server and you need to run rtp proxy. If the other client IP
address is in the SDP, then the RTP packets don't come to opensips and go
directly to the other client. The same goes for 200 OK.

As for switches and routers, that's how IP packets are routed based on your
cabling, IP addresses and mac addresses. These are two separate things.
Think of them completely separately. RTP packets have voice and video data
but how they are routed to their destination IP is all in IP protocol and
has nothing to do with SIP. The destination IP address in your RTP packets,
that comes from SDP negotiation through SIP. A different story.

Regards,
Ali Pey


On Tue, Feb 25, 2014 at 2:20 PM, Nandini  wrote:

> Dear Ali Pey,
>
> Sorry for creating kind of mess in my question.
> No i am not talking about any IP routing, But  my question is totally
> related only to RTP packet flow between client -to- client. I mean when SIP
> establihes a call session between two SIP clients (Which are registered to
> OpenSIPS server), in that point of time How RTP packets (which carries
> voice/video) flows from one client to another client ?
>
> Here below link is my test bed set-up :
>
> My_test_bed_set-up.png
> <
> http://opensips-open-sip-server.1449251.n2.nabble.com/file/n7589774/My_test_bed_set-up.png
> >
>
> And now My question is:
> What is RTP session client-to-client ?
> Is this RTP packets in voice/video call from caller will pass through
> router
> 1 ---> switch > router 2---> destination SIP client ? or just Caller to
> Destination client ? (without pass through any nodes in between).
>
> (If suppose i run media proxy server with OpenSIPs server then RTP packets
> has to pass through all the nodes (routers and switch) in between to get
> relay to other end client right ?)
>
> So in this context if i run only openSIPS server without any Media-proxy
> servers, how the RTP packets are achieving Client-to-client transfer ?
>
> Please help me in clarifying this questions.
>
> Regards,
> Nandini
>
>
>
>
> --
> View this message in context:
> http://opensips-open-sip-server.1449251.n2.nabble.com/Query-about-RTP-flow-tp7589763p7589774.html
> Sent from the OpenSIPS - Users mailing list archive at Nabble.com.
>
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Re: [OpenSIPS-Users] Query about RTP flow

[Ali Pey]

Nandini,

I guess you are asking about IP routing. IP routing is based on IP
addresses, Mac Addresses and ARP tables. The routers and switches receive
the Packets, examine the destination IP address and then send it the packet
to the Mac address that can re-route the packet. Is this the kind of
information you are looking for? I suggest you do a bit of reading on IP
routing and the 7 networking layers.

Regards,
Ali Pey


On Tue, Feb 25, 2014 at 12:49 PM, Nandini madhu  wrote:

> Dear Ali Pey,
>
> Thank you for the reply,
>
> I am agreeing with your answer, but I am specifically wanted to know how
> physically the RTP packets flow from one client to other client in the
> instance like in my above mentioned set-up.
> As clients registers with SIP server, once signalling path establihed by
> SIP, then RTP will take another path to transfer media between the clients.
> so in this context what is the path this RTP packets follow to reach other
> end client (to achieve client-client connection), when the SIP clients
> (running on Andriod smart phones) are connected two diffrent Wi-Fi Access
> Point ?
> How RTP packets transfer from one client to another client if both the
> clients connected to  Single WAP ?
>
> Please help me in clarifying this issues.
>
> I hope you could help me.
>
> Regards,
> Nandini
>
>
> On Tue, Feb 25, 2014 at 6:11 PM, Ali Pey  wrote:
>
>> Terms Direct RTP or Proxy RTP through your SIP server has nothing to do
>> with your network routers and switches. It basically means if Client A is
>> sending RTP packets to the IP address of Client B or if Client A is sending
>> RTP packets to the IP address of your SIP server and the SIP server needs
>> to re-send the packets to Client B and vise versa.
>>
>> So if the SIP server is in the middle of RTP communications that's proxy
>> RTP. If RTP packets don't reach your SIP server and is only between the
>> Clients, that's Direct speech path.
>>
>> Regards,
>> Ali Pey
>>
>>
>> On Tue, Feb 25, 2014 at 3:46 AM, Nandini madhu wrote:
>>
>>> Dear All,
>>>
>>> I have read about SIP and RTP packet flows but ended with a question
>>> about RTP packets flow logic between client-to client when the SIP session
>>> starts.
>>> I have a running Opensips server on ubuntu platform which is also
>>> configured fro DNS and DHCP servers.
>>> And following is my test bed set-up:
>>>
>>>
>>> Opensips server
>>>  (DNS+
>>> DHCP+Opensips)
>>>
>>>  |
>>>   Router <>
>>> Switch<> Router
>>>   |
>>>|
>>>   clientA <-
>>> RTP(audio/video) ? > client B
>>>
>>> Now when the SIP session established (SIP Packets have to pass through
>>> all the routers and switch to reach Opensips server), RTP session will
>>> start between Client A and B. But What is meant by client to client here ?
>>> what is the RTP packets path between the clients to reach their
>>> destinations ?
>>> Is the RTP packets have to pass through Routers and switch to reach its
>>> destination client ? (i.e Client A--->Router 1>switch -> Router
>>> 2-> client B)  ?
>>> What if i connect both client A and client B to the same router ?
>>>
>>> PS: I have not configured any media-relays (RTPproxy/Media-proxy)
>>>
>>> Please help me in clarifying all this doubts.
>>>
>>> Anybody's help will greatly appreciate.
>>>
>>> Regards,
>>> Nandini
>>>
>>> ___
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>>>
>>>
>>
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Re: [OpenSIPS-Users] Query about RTP flow

[Ali Pey]

Terms Direct RTP or Proxy RTP through your SIP server has nothing to do
with your network routers and switches. It basically means if Client A is
sending RTP packets to the IP address of Client B or if Client A is sending
RTP packets to the IP address of your SIP server and the SIP server needs
to re-send the packets to Client B and vise versa.

So if the SIP server is in the middle of RTP communications that's proxy
RTP. If RTP packets don't reach your SIP server and is only between the
Clients, that's Direct speech path.

Regards,
Ali Pey


On Tue, Feb 25, 2014 at 3:46 AM, Nandini madhu  wrote:

> Dear All,
>
> I have read about SIP and RTP packet flows but ended with a question about
> RTP packets flow logic between client-to client when the SIP session starts.
> I have a running Opensips server on ubuntu platform which is also
> configured fro DNS and DHCP servers.
> And following is my test bed set-up:
>
>
> Opensips server
>  (DNS+
> DHCP+Opensips)
>
>|
>   Router <>
> Switch<> Router
>   |
>|
>   clientA <-
> RTP(audio/video) ? > client B
>
> Now when the SIP session established (SIP Packets have to pass through all
> the routers and switch to reach Opensips server), RTP session will start
> between Client A and B. But What is meant by client to client here ?
> what is the RTP packets path between the clients to reach their
> destinations ?
> Is the RTP packets have to pass through Routers and switch to reach its
> destination client ? (i.e Client A--->Router 1>switch -> Router
> 2-> client B)  ?
> What if i connect both client A and client B to the same router ?
>
> PS: I have not configured any media-relays (RTPproxy/Media-proxy)
>
> Please help me in clarifying all this doubts.
>
> Anybody's help will greatly appreciate.
>
> Regards,
> Nandini
>
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Re: [OpenSIPS-Users] STUN

[Ali Pey]

Hi Nick,

I don't think such a thing is possible in opensips. There can only be one
advertised_address as far as I know. I think you need to do your own
message manipulation in your route script based on the route it's going to.

I'm interested in this scenario as well and look forward to more replies
from more knowledgeable people.

Please post updates when you find a solution.


Regards,
Ali Pey


On Wed, Feb 19, 2014 at 5:47 AM, Nick Altmann wrote:

> Hi!
>
> How to correctly configure STUN module if we have two interfaces with
> internal addresses where NATed two external IPs? (ports are the same)
> I mean something like advertised_address option.
>
> --
> Nick
>
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Re: [OpenSIPS-Users] 200 OK retransmission

[Ali Pey]

Hello,

Yes, 200 OK must be retransmitted if Ack is not received. opensips can't do
that on behalf of the SIP trunk. Retransmission is the very basic
requirement and I've never heard that a sip trunk to not be able to do so.
Discuss with your provider.

Regards,
Ali Pey


On Wed, Feb 19, 2014 at 4:42 AM, M.Khaled W Chehab wrote:

> Hi,
>
>
>
> In the below scenario
>
> UAC-à Opensipsàtrunk
>
> 200 Ok SDP   ßopensips ß200 SDP  from trunk
>
> UAC didn't reply by ACK ,as he complain the 200 OK with SDP was not
> received and it's a packet loos
>
> Is there a part on RFC ask to resend the 200OK with SDP if no ACk received
> from UAC, or a way in opensips to resend the 200 Ok with SDp if no ACK
> received
>
>
>
> Regards
>
>
>
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Re: [OpenSIPS-Users] 407 Proxy authorization required

[Ali Pey]

This seems to be a very slow network connectivity. Your Bria is
retransmitting Invites before it receives any responses from OpenSIPS. Then
it sends Invite we authentication with invalid nounce. You can disable
nounce check in opensips and that would fix it but would make your opensips
vulnerable.

This seems to be Bria on a mobile with a very slow network connectivity.

Regards,
Ali Pey


On Mon, Feb 17, 2014 at 10:34 AM, Chandra Prakash <
chandraprak...@virtualemployee.com> wrote:

> Hi,
>
> I cannot establish a call, Can someone pls help to identify the problem?
>
>
>
> This is the log..
> XX.XX.XX.243=>Bria
> XX.XX.XX8.250=>opensips
>
>
> U XX.XX.XX.243:41485 -> XX.XX.XX8.250:5060
> INVITE sip:7447962...@xx.xx.XX8.250 SIP/2.0.
> Via: SIP/2.0/UDP
> XX.XX.XX.243:41485;rport;branch=z9hG4bKPjkX1h2zP7jg3VQFpH2M5OeYz.sYXk1tjq.
> Max-Forwards: 70.
> From: "4457675066"
> ;tag=.btzkWF2ELYjfFGj0rlE1aHH5bu.9LQv.
> To: sip:7447962...@xx.xx.XX8.250.
> Contact: "4457675066" .
> Call-ID: bqOk1trJJW1FZGjXv-Ubua4FtVp3Cas9.
> CSeq: 5398 INVITE.
> Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY,
> REFER, MESSAGE, OPTIONS.
> Supported: replaces, 100rel, timer, norefersub.
> User-Agent: Bria Android 3.0.2.
> Content-Type: application/sdp.
> Content-Length:   315.
> .
> v=0.
> o=- 3601639408 3601639408 IN IP4 XX.XX.XX.243.
> s=cpc_med.
> c=IN IP4 XX.XX.XX.243.
> t=0 0.
> m=audio 4004 RTP/AVP 3 18 8 0 101.
> a=sendrecv.
> a=rtpmap:3 GSM/8000.
> a=rtpmap:18 G729/8000.
> a=fmtp:18 annexb=no.
> a=rtpmap:8 PCMA/8000.
> a=rtpmap:0 PCMU/8000.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-15.
> a=ptime:20.
>
>
> U XX.XX.XX8.250:5060 -> XX.XX.XX.243:41485
> SIP/2.0 407 Proxy Authentication Required.
> Via: SIP/2.0/UDP
>
> XX.XX.XX.243:41485;received=XX.XX.XX.243;rport=41485;branch=z9hG4bKPjkX1h2zP
> 7jg3VQFpH2M5OeYz.sYXk1tjq.
> From: "4457675066"
> ;tag=.btzkWF2ELYjfFGj0rlE1aHH5bu.9LQv.
> To: sip:7447962...@xx.xx.XX8.250
> ;tag=f0b4f841c180728ee357d262d8a5cefc.b77f.
> Call-ID: bqOk1trJJW1FZGjXv-Ubua4FtVp3Cas9.
> CSeq: 5398 INVITE.
> Proxy-Authenticate: Digest realm="XX.XX.XX8.250",
> nonce="52f83b07000a516fa9ae3bd8e965e6ef9e1d7552df86".
> Server: OpenSIPS (1.11.0dev-notls (x86_64/linux)).
> Content-Length: 0.
> .
>
>
> U XX.XX.XX.243:41485 -> XX.XX.XX8.250:5060
> INVITE sip:7447962...@xx.xx.XX8.250 SIP/2.0.
> Via: SIP/2.0/UDP
> XX.XX.XX.243:41485;rport;branch=z9hG4bKPjkX1h2zP7jg3VQFpH2M5OeYz.sYXk1tjq.
> Max-Forwards: 70.
> From: "4457675066"
> ;tag=.btzkWF2ELYjfFGj0rlE1aHH5bu.9LQv.
> To: sip:7447962...@xx.xx.XX8.250.
> Contact: "4457675066" .
> Call-ID: bqOk1trJJW1FZGjXv-Ubua4FtVp3Cas9.
> CSeq: 5398 INVITE.
> Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY,
> REFER, MESSAGE, OPTIONS.
> Supported: replaces, 100rel, timer, norefersub.
> User-Agent: Bria Android 3.0.2.
> Content-Type: application/sdp.
> Content-Length:   315.
> .
> v=0.
> o=- 3601639408 3601639408 IN IP4 XX.XX.XX.243.
> s=cpc_med.
> c=IN IP4 XX.XX.XX.243.
> t=0 0.
> m=audio 4004 RTP/AVP 3 18 8 0 101.
> a=sendrecv.
> a=rtpmap:3 GSM/8000.
> a=rtpmap:18 G729/8000.
> a=fmtp:18 annexb=no.
> a=rtpmap:8 PCMA/8000.
> a=rtpmap:0 PCMU/8000.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-15.
> a=ptime:20.
>
>
> U XX.XX.XX8.250:5060 -> XX.XX.XX.243:41485
> SIP/2.0 407 Proxy Authentication Required.
> Via: SIP/2.0/UDP
>
> XX.XX.XX.243:41485;received=XX.XX.XX.243;rport=41485;branch=z9hG4bKPjkX1h2zP
> 7jg3VQFpH2M5OeYz.sYXk1tjq.
> From: "4457675066"
> ;tag=.btzkWF2ELYjfFGj0rlE1aHH5bu.9LQv.
> To: sip:7447962...@xx.xx.XX8.250
> ;tag=f0b4f841c180728ee357d262d8a5cefc.b77f.
> Call-ID: bqOk1trJJW1FZGjXv-Ubua4FtVp3Cas9.
> CSeq: 5398 INVITE.
> Proxy-Authenticate: Digest realm="XX.XX.XX8.250",
> nonce="52f83b07000b8d78328f27a7f9dbc9dbdcdd8522f3dc".
> Server: OpenSIPS (1.11.0dev-notls (x86_64/linux)).
> Content-Length: 0.
> .
>
>
> U XX.XX.XX.243:41485 -> XX.XX.XX8.250:5060
> INVITE sip:7447962...@xx.xx.XX8.250 SIP/2.0.
> Via: SIP/2.0/UDP
> XX.XX.XX.243:41485;rport;branch=z9hG4bKPjkX1h2zP7jg3VQFpH2M5OeYz.sYXk1tjq.
> Max-Forwards: 70.
> From: "4457675066"
> ;tag=.btzkWF2ELYjfFGj0rlE1aHH5bu.9LQv.
> To: sip:7447962...@xx.xx.XX8.250.
> Contact: "4457675066" .
> Call-ID: bqOk1trJJW1FZGjXv-Ubua4FtVp3Cas9.
> CSeq: 5398 INVITE.
> Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY,
> REFER, MESSAGE, OPTIONS.
> Supported: replaces, 100rel, timer, norefersub.
> User-Agent: Bria

Re: [OpenSIPS-Users] [Freeswitch-users] Brainstorming load balancing feature - dispatcher & FS

[Ali Pey]

This would be a pretty cool feature and I know that OpenSIPS guys have
discussed such a feature with FreeSWITCH dev. This would be a great
addition for load balancing.

Adding OpenSIPS user mailing list.


Regards,
Ali Pey


On Mon, Feb 17, 2014 at 8:15 AM, Karsten Horsmann wrote:

> Hello List,
>
> i read too the nice 
> http://wiki.freeswitch.org/wiki/Enterprise_deploymentEnterprise Deployment 
> Setups.
> Personaly i run FS 1.2.x as Session Border Controller where sip clients
> could register and internal (3rd party) IVRs could reachs the clients.
>
> I plan to use DNS SRV on the to scale up the FS Servers and have fault
> talerance.
>
> For that i have to diffrent uplinks that i could use for the DNS SRV
> Records and AFAIK i could use track-calls=true in sofia profiles
> to track the right incoming/outgoing profile.
>
> The only thing i missed is that my 3rd party IVRs used FS as outbound
> proxy and therefore only one IP is spezified on the internal network.
>
> With an virtual IP on the FS side i could solve that but then the FS
> servers is IMHO limited to be a active/passiv duo.
>
> Any ideas how to handle that?
>
>
>
> 2014-02-17 2:45 GMT+01:00 jay binks :
>
>> So I want to bash out a feature I was thinking of, please can anyone
>> chime in some ideas on this.
>>
>> So its easy to have OpenSIP / Kamailio ( dispatcher ) send SIP Options to
>> FS to see if its responding ..  this is great ...
>>
>> My request would be to have some kind of control over the response FS
>> Sends back.
>> What I would love is to respond with idle CPU , current SPS / Max SPS or
>> channels / Max Channels.
>>
>> This is only a high level "this would be nice" , so I havnt totally
>> thought this through but...
>> we would have to provide an ACL for who to provide this back to,  coz I
>> wouldnt really want any old guy on the internet sending me options and me
>> respond going ... oh yea bro, Im at 90% ... dont have to try hard to DDOS
>> Me.. :)
>>
>> but yea... if we could give this back to the load balancers, then we
>> could easily instruct them to back traffic off to this gateway.
>>
>> the next part is , how do you back off the traffic.
>> one option is to have FS respond differently if any of these are close to
>> thresholds.
>> or to provide these values in custom headers, but that would require
>> changes in dispatcher ( which is ok I Guess )
>>
>> anyways... I wanted some input on peoples thoughts here..
>> let me know what you think.
>>
>>
>> Jay
>>
>> _
>> Professional FreeSWITCH Consulting Services:
>> consult...@freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server
>> http://www.cudatel.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> freeswitch-us...@lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Mit freundlichen Grüßen
> *Karsten Horsmann*
>
> _
> Professional FreeSWITCH Consulting Services:
> consult...@freeswitch.org
> http://www.freeswitchsolutions.com
>
> FreeSWITCH-powered IP PBX: The CudaTel Communication Server
> http://www.cudatel.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> freeswitch-us...@lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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Re: [OpenSIPS-Users] Users Digest, Vol 64, Issue 70

[Ali Pey]

Please always respond on a proper thread with a proper subject line. No one
can tell what you are talking about and to what thread you are responding
when you reply to a Users Digest email.

Regards,
Ali Pey


On Tue, Dec 3, 2013 at 4:47 AM, Chandra Prakash <
chandraprak...@virtualemployee.com> wrote:

> Thanks Jeff,
>
> And Sorry for the delayed response.
>
> The endpoints are Bria.
> Network :- 5 Mbps leased line which is directly connected to the server.
>
> I tried to reconfigure the RTPproxy but in vain.
>
> Thanks
>
> -Original Message-
> From: users-boun...@lists.opensips.org
> [mailto:users-boun...@lists.opensips.org] On Behalf Of
> users-requ...@lists.opensips.org
> Sent: 26 November 2013 23:05
> To: users@lists.opensips.org
> Subject: Users Digest, Vol 64, Issue 70
>
> Send Users mailing list submissions to
> users@lists.opensips.org
>
> To subscribe or unsubscribe via the World Wide Web, visit
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> or, via email, send a message with subject or body 'help' to
> users-requ...@lists.opensips.org
>
> You can reach the person managing the list at
> users-ow...@lists.opensips.org
>
> When replying, please edit your Subject line so it is more specific than
> "Re: Contents of Users digest..."
>
>
> Today's Topics:
>
>1. Delay with RTPproxy (Chandra Prakash)
>2. DNS SRV failover not working for B2BUA (1.10) (Jeff Pyle)
>3. Re: opensips sends CANCEL (Miha)
>4. Re: Delay with RTPproxy (Jeff Pyle)
>5. Re: DNS SRV failover not working for B2BUA (1.10) (Jeff Pyle)
>
>
> --
>
> Message: 1
> Date: Tue, 26 Nov 2013 19:46:51 +0530
> From: "Chandra Prakash" 
> Subject: [OpenSIPS-Users] Delay with RTPproxy
> To: 
> Message-ID: <01ceeab2$29cc5ba0$7d6512e0$@virtualemployee.com>
> Content-Type: text/plain;   charset="us-ascii"
>
>
> Hi,
>
> I've configured opensip 1.8 with RTPproxy 1.2.
>
> All work fine except there is delay in RTP sessions. Is there any fix for
> this problem ?
>
> Pls help
>
> Thanks
>
>
>
>
> --
>
> Message: 2
> Date: Tue, 26 Nov 2013 09:23:01 -0500
> From: Jeff Pyle 
> Subject: [OpenSIPS-Users] DNS SRV failover not working for B2BUA
> (1.10)
> To: OpenSIPS users mailling list 
> Message-ID:
> <
> cacyjg3ld_usgp10wgqu4lfemmfh5mppvgwhkcizhqxn4wef...@mail.gmail.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Hello,
>
> I have an SRV name as follows:
>
> $ dig +short -tSRV _sip._udp.proxyname.domain.com
> 1 10 5060 host1.domain.com.
> 2 10 5060 host2.domain.com.
>
> If I t_relay() a request with proxyname.domain.com as the request domain,
> and host1 does not answer within fr_timer seconds, the request is
> re-relayed
> to host2.  This is good.
>
> If I b2b_init_request("top hiding") instead of t_relay(), the 408 is
> forwarded back towards the UAC after host1 doesn't answer.  The request is
> never re-relayed to host2.
>
> There is a difference in the debug output after the 408 is generated for
> host1.  With the B2BUA it looks like is_3263_failure() never happens.
>
> t_relay:
> Nov 26 09:03:38 [16296] DBG:tm:timer_routine: timer
> routine:0,tl=0x7fbe6c7c0060 next=(nil), timeout=61 Nov 26 09:03:38 [16296]
> DBG:tm:final_response_handler: Cancel sent out, sending 408
> (0x7fbe6c7bfe10)
> Nov 26 09:03:38 [16296] DBG:tm:t_should_relay_response: T_code=100,
> new_code=408
> Nov 26 09:03:38 [16296] DBG:tm:t_pick_branch: picked branch 0, code 408
> (prio=800)
> Nov 26 09:03:38 [16296] DBG:tm:is_3263_failure: dns-failover test:
> branch=0, last_recv=408, flags=1
> Nov 26 09:03:38 [16296] DBG:tm:t_should_relay_response: trying DNS-based
> failover Nov 26 09:03:38 [16296] DBG:tm:do_dns_failover: new destination
> available Nov 26 09:03:38 [16296] DBG:core:check_ip_address: params
> 127.0.0.1, 127.0.0.1, 0 Nov 26 09:03:38 [16296] DBG:tm:set_timer: relative
> timeout is 50
>
> b2b_init_request:
> Nov 26 09:12:21 [16401] DBG:tm:timer_routine: timer
> routine:0,tl=0x7f09ddcbf7e8 next=(nil), timeout=11 Nov 26 09:12:21 [16401]
> DBG:tm:final_response_handler: Cancel sent out, sending 408
> (0x7f09ddcbf598)
> Nov 26 09:12:21 [16401] DBG:tm:t_should_relay_response: T_code=0,
> new_code=408
> Nov 26 09:12:21 [16401] DBG:tm:t_pick_branch: picked branch 0, code 408
> (prio=800)
> Nov 26 09:12:21 [16401] DBG:tm:local_reply: branch=0, save=0, winner=0 Nov
> 26 09:12:21 [16401] DBG:tm:local_reply:

Re: [OpenSIPS-Users] Firing gwlist in specific order with failover

[Ali Pey]

In do_routing, you can specify a list of gateways:

http://www.opensips.org/html/docs/modules/devel/drouting.html#id294454

- gw_whitelist - a comma separated white list of gateways. This will force
routing over this list of carriers or gateways (a subset of what found in
the prefix rules).

Or you can have multiple groups and have your perl script return a specific
group id for a particular call.

There are many ways you can do this. You can do this through your Perl
script as well. Just route the call manually to one destination at a time!!


Regards,
Ali Pey


On Wed, Jan 8, 2014 at 8:49 PM, Nick Cameo  wrote:

> Oh hey Ali! What I am asking is for is a function that I can pass a gwlist
> too at runtime, basically retiring the do_routing since
> I was told it does not handle gwllists sequentially, if I understood
> correctly. I have a perl module that orders the gwlist as needed,
> now I just need a function that will fire the INVITE in order with
> failover and all of that is provided by do_routing.
>
> I hope this is clear?
>
> ___
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> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
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Re: [OpenSIPS-Users] Firing gwlist in specific order with failover

[Ali Pey]

Hello Nick,

What is the question then?

In rules, you can list gateways, carriers and assign wait. How is that
different from what you want?

Regards,
Ali Pey


On Wed, Jan 8, 2014 at 1:09 PM, Nick Cameo  wrote:

> Hello Everyone,
>
> We are currently using the dr module with dr_rules/gateways populated, and
> it works
> great.What we are looking for a is a do_routing like method that accepts a
> gwlist, terminates
> to the gatway, and failover to the next one on the list in sequential
> order.
> Exactly as the do_routing method, only with the provided gwlist, and
> strict chronological order.
>
> I did ask this question before but it went stale.
>
> Thank in Advance,
>
> Nick.
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
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Re: [OpenSIPS-Users] BYE not reaching UAC

[Ali Pey]

I see the Bye has a route for sip:700@54.204.19.238:5060.

54.204.19.238:5060 has to be your opensips server. You probably need an
alias for this IP.

This is only my guess. In situations like this you always need to look at
your via, record-route and route headers, plus the source IPs. This is how
things are routed in SIP.

Here is a good reference: http://www.in2eps.com/fo-sip/tk-fo-sip-ex3261.html

Regards,
Ali Pey



On Fri, Dec 20, 2013 at 1:23 PM, Zoho Junk  wrote:

> Greetings,
>
> I have OpenSips installed on an Amazon EC2 with an elastic IP
> (54.242.85.140). All non REGISTER methods are passed through
> rewritehostport("mytwiliodomain.sip.twilio.com") to another SIP proxy.
>
> Everything is working except when BYE is sent from a callee that is
> registered to my OpenSips server. The caller never receives the BYE.
> I've changed my config file at least 100 times and still have no success.
> Please let me know if I need to include anything from my config.
>
> *My First Invite:*
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
> *U 2013/12/20 12:44:21.298060 107.21.231.147:5060
> <http://107.21.231.147:5060> -> 10.167.11.171:5060
> <http://10.167.11.171:5060>INVITE sip:7...@sip.opensipsserver.com
> <7...@sip.opensipsserver.com> SIP/2.0.Record-Route:
> .From:
> "17175554895 <17175554895>"  <17175554...@sip.twilio.com>>;tag=31945052_6772d868_3b1eb7f5-c41b-4da5-9a97-c756d7cd8fe3.To:
> >.CSeq: 102
> INVITE.Max-Forwards: 68.Date: Fri, 20 Dec 2013 17:42:59 GMT.Call-ID:
> 6fa66f8ea2ce4b2dae83149f8b34b2e2@0.0.0.0
> <6fa66f8ea2ce4b2dae83149f8b34b2e2@0.0.0.0>.Contact: "17175554895
> <17175554895>"  <17175554895>@10.70.227.82:5060;transport=udp>.Allow:
> INVITE,ACK,CANCEL,OPTIONS,BYE.User-Agent: Twilio
> Gateway.X-Twilio-ApiVersion: 2010-04-01.X-Twilio-AccountSid:
> AC7ae5d84004d4269fgfhfhfjghgjghgbb.Content-Type:
> application/sdp.X-Twilio-CallSid: CA79e7a3c2c1c418af0777467467474747.Via:
> SIP/2.0/UDP 107.21.231.147:5060;branch=z9hG4bKaeaf.02902511.0.Via:
> SIP/2.0/UDP
> 10.70.227.82:5060;branch=z9hG4bK3b1eb7f5-c41b-4da5-9a97-c756d7cd8fe3_6772d868_425297635397116.Content-Length:
> 251..v=0.o=- 1274389856 1274389856 IN IP4 54.196.144.208.s=session.c=IN IP4
> 54.196.144.208.t=0 0.m=audio 18800 RTP/AVP 0 101.a=rtpmap:0
> PCMU/8000.a=rtpmap:101 telephone-event/8000.a=fmtp:101
> 0-16.a=silenceSupp:off - - - -.a=ptime:20.a=sendrecv.*
>
> *My First Bye:*
>
>
>
>
>
>
> *U 2013/12/20 12:44:24.194615 74.106.255.63:5907
> <http://74.106.255.63:5907> -> 10.167.11.171:5060
> <http://10.167.11.171:5060>BYE sip:17175554895
> <17175554895>@107.21.231.147:5060;transport=udp;nat=yes SIP/2.0.Via:
> SIP/2.0/UDP 74.106.255.63:5907;branch=z9hG4bK-41214dcd.From:
>  <7...@sip.opensipsserver.com>>;tag=5ed7c43daadb6f7fi0.To: "17175554895
> <17175554895>"  <17175554...@sip.twilio.com>>;tag=31945052_6772d868_3b1eb7f5-c41b-4da5-9a97-c756d7cd8fe3.Call-ID:
> *
>
>
>
>
> *6fa66f8ea2ce4b2dae83149f8b34b2e2@0.0.0.0 <http://0.0.0.0>.CSeq: 101
> BYE.Max-Forwards: 70.Route:
> ,
> .User-Agent:
> Linksys/SPA942-6.1.3(a).Content-Length: 0.*
>
> *My Debug Log:*
> http://pastebin.com/0pAAZ68J
>
>
> *NGrep:*http://pastebin.com/TVYUmM58
>
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>
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Re: [OpenSIPS-Users] Checking for Initial INVITE in branch route

[Ali Pey]

has_totag can be used in request route:

http://opensips.org/html/docs/modules/devel/uri.html#id293942

You can set appropriate flags if you need to.

As a general comment, if things seem to be complicated, the chances are
that you are not looking at it the right way, try to reconsider your
design. Things should not be complicated.

I'm not sure what you are trying to do, but when an invite reaches
opensips, it's easy to tell if it's an initial invite and route it
accordingly.

Regards,
Ali Pey


On Mon, Dec 16, 2013 at 10:26 AM, Nick Cameo  wrote:

> Are we able to use has_totag() in branch route to check for initial
> INVITES? After looking further into the problem, it seem like the
> error "ERROR:uac:replace_uri: decline FROM replacing in sequential
> request in auto mode (has TO tag)" is happening on a failover and when
> opensips branches out to the next available gateway.
>
> Should I be unchanging the to tag in the failover? We are using:
> restore_mode auto as the error would suggest.
>
> Kind Regards,
>
> Nick.
>
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Re: [OpenSIPS-Users] Checking for Initial INVITE in branch route

[Ali Pey]

Examine to the to tag. If there is a to tag, that's not a initial
invite: has_totag()

Regards,
Ali Pey


On Fri, Dec 13, 2013 at 4:08 PM, Nick Cameo  wrote:

> Hello Everyone,
>
> I am changing the TO Header once in the branch route, and I seeing the
> following errors:
>
> ERROR:uac:replace_uri: decline FROM replacing in sequential request in
> auto mode (has TO tag)
>
> I am sure that it's being done only once in the branch where I check
> if it's an INVITE request:
>
> if (is_method("INVITE")) {
> ..
> }
>
> Is there any way to check for initial INVITES in branches? I suspect
> it's happening
> on the failover, though did not take a closer look.
>
> Kind Regards,
>
> Nick.
>
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Re: [OpenSIPS-Users] Nat_keepalive help

[Ali Pey]

Yes, the best thing is to do it on the client side and most (almost all)
phones support many nat pinging features. You should be able to add it to
the phones' config file and push it out.

Regards,
Ali Pey


On Thu, Dec 12, 2013 at 4:35 PM, Miha  wrote:

> Ignor this question:)
>
> Ok tnx! So the best thing is to do that on uac side.
>
> Br
> Miha
>
> On Thu, 12 Dec 2013 14:58:45 -0500
>  Ali Pey  wrote:
> > What do you mean by "what i just do that opensips will
> > triger uac"?
> >
> > Trigger what?
> >
> > Regards,
> > Ali
> >
> >
> >
> > On Thu, Dec 12, 2013 at 2:17 PM, Miha 
> > wrote:
> >
> > > Ali how can i tell opensips or what i just do that
> > opensips
> > > will triger uac?
> > >
> > > I have registration time set to 3600, as on this server
> > > will be around 20k users registration time 30s is not
> > > possible:)
> > >
> > > On Thu, 12 Dec 2013 12:22:00 -0500
> > >  Ali Pey  wrote:
> > > > The udp packets should come from inside the firewall
> > for
> > > > nat binding to
> > > > stay open so it should come from the client side.
> > > > Opensips can send Options
> > > > to trigger a response from the client to keep the nat
> > > > binding open.
> > > >
> > > > If you don't have lots of clients and your clients
> > don't
> > > > have any other
> > > > pinging capability, you can decrease the
> > re-registration
> > > > time on the client
> > > > side to 30 seconds or so. That will also keep the nat
> > > > binding open.
> > > >
> > > > Regards,
> > > > Ali Pey
> > > >
> > > >
> > > >
> > > > On Thu, Dec 12, 2013 at 12:13 PM, Miha
> > 
> > > > wrote:
> > > >
> > > > > Ali tnx.
> > > > >
> > > > > Opensips is on public ip, uac's are behind nat.
> > > > >
> > > > > I tried to use info, options but stikal the same
> > > > error:(
> > > > >
> > > > > What abou that opensips sends udp packets, so that
> > i
> > > > would
> > > > > not have to enable this on uac?
> > > > >
> > > > > We have One softswitch and there is no problem with
> > > > expired
> > > > > nat session and arso does not send
> > info/notify/options,
> > > > i
> > > > > guess it is sending some udp packets.
> > > > >
> > > > > Tnx
> > > > > Miha
> > > > >
> > > > > On Thu, 12 Dec 2013 11:22:05 -0500
> > > > >  Ali Pey  wrote:
> > > > > > BTW. the error you are getting is probably for
> > the
> > > > fact
> > > > > > that your client
> > > > > > doesn't like the Notify pinging. Change it to
> > Options
> > > > > > message and it will
> > > > > > work.
> > > > > >
> > > > > > Regards,
> > > > > > Ali Pey
> > > > > >
> > > > > >
> > > > > >
> > > > > > On Thu, Dec 12, 2013 at 11:19 AM, Ali Pey
> > > > > >  wrote:
> > > > > >
> > > > > > > Hello Miha,
> > > > > > >
> > > > > > > The best way to handle nat keep alive is from
> > your
> > > > sip
> > > > > > clients. They
> > > > > > > usually offer a variety of option and the most
> > > > > > efficient one is the udp
> > > > > > > pinging, in which it keeps sending opensips a
> > udp
> > > > > > packet with 4 zeros in it
> > > > > > > to keep the nat binding open. That would work
> > > > perfectly
> > > > > > if you have static
> > > > > > > nat binding on the opensips server which means
> > > > doing
> > > > > > port forwarding on the
> > > > > > > firewall the opensips sits behind.
> > > > > > >
> > > > > > > You can also enable options pinging on your
> > client
> > > > as
> > > > > > well and that will
> > > > > > > keep nats open on both side as opensips will
> > > > res

Re: [OpenSIPS-Users] Nat_keepalive help

[Ali Pey]

What do you mean by "what i just do that opensips will triger uac"?

Trigger what?

Regards,
Ali



On Thu, Dec 12, 2013 at 2:17 PM, Miha  wrote:

> Ali how can i tell opensips or what i just do that opensips
> will triger uac?
>
> I have registration time set to 3600, as on this server
> will be around 20k users registration time 30s is not
> possible:)
>
> On Thu, 12 Dec 2013 12:22:00 -0500
>  Ali Pey  wrote:
> > The udp packets should come from inside the firewall for
> > nat binding to
> > stay open so it should come from the client side.
> > Opensips can send Options
> > to trigger a response from the client to keep the nat
> > binding open.
> >
> > If you don't have lots of clients and your clients don't
> > have any other
> > pinging capability, you can decrease the re-registration
> > time on the client
> > side to 30 seconds or so. That will also keep the nat
> > binding open.
> >
> > Regards,
> > Ali Pey
> >
> >
> >
> > On Thu, Dec 12, 2013 at 12:13 PM, Miha 
> > wrote:
> >
> > > Ali tnx.
> > >
> > > Opensips is on public ip, uac's are behind nat.
> > >
> > > I tried to use info, options but stikal the same
> > error:(
> > >
> > > What abou that opensips sends udp packets, so that i
> > would
> > > not have to enable this on uac?
> > >
> > > We have One softswitch and there is no problem with
> > expired
> > > nat session and arso does not send info/notify/options,
> > i
> > > guess it is sending some udp packets.
> > >
> > > Tnx
> > > Miha
> > >
> > > On Thu, 12 Dec 2013 11:22:05 -0500
> > >  Ali Pey  wrote:
> > > > BTW. the error you are getting is probably for the
> > fact
> > > > that your client
> > > > doesn't like the Notify pinging. Change it to Options
> > > > message and it will
> > > > work.
> > > >
> > > > Regards,
> > > > Ali Pey
> > > >
> > > >
> > > >
> > > > On Thu, Dec 12, 2013 at 11:19 AM, Ali Pey
> > > >  wrote:
> > > >
> > > > > Hello Miha,
> > > > >
> > > > > The best way to handle nat keep alive is from your
> > sip
> > > > clients. They
> > > > > usually offer a variety of option and the most
> > > > efficient one is the udp
> > > > > pinging, in which it keeps sending opensips a udp
> > > > packet with 4 zeros in it
> > > > > to keep the nat binding open. That would work
> > perfectly
> > > > if you have static
> > > > > nat binding on the opensips server which means
> > doing
> > > > port forwarding on the
> > > > > firewall the opensips sits behind.
> > > > >
> > > > > You can also enable options pinging on your client
> > as
> > > > well and that will
> > > > > keep nats open on both side as opensips will
> > respond
> > > > with 200 OK.
> > > > >
> > > > > Enabling nat keep alive from opensips is more
> > expensive
> > > > in terms of
> > > > > resources but that is also a possible way to do.
> > > > >
> > > > > Regards,
> > > > > Ali Pey
> > > > >
> > > > >
> > > > >
> > > > > On Thu, Dec 12, 2013 at 7:16 AM, Miha
> > 
> > > > wrote:
> > > > >
> > > > >>  HI,
> > > > >>
> > > > >> I need a little help with nat. UAC register's ok
> > and
> > > > it is reachable but
> > > > >> after a while it become unreachable due to nat
> > issue.
> > > > If I use
> > > > >> nat_keepalive on uac deivce everything is ok, but
> > I
> > > > would like that this is
> > > > >> handled by opensips.
> > > > >>
> > > > >> I tried to use  nat_keepalive but getting all the
> > time
> > > > this:
> > > > >>
> > > > >>  54.820695 opensips -> uac_public_ip SIP 308
> > Request:
> > > > NOTIFY
> > > > >> sip:uac_public_ip:13401 |
> > > > >>  54.829847 uac_public_ip -> opensips SIP 

Re: [OpenSIPS-Users] Nat_keepalive help

[Ali Pey]

BTW. the error you are getting is probably for the fact that your client
doesn't like the Notify pinging. Change it to Options message and it will
work.

Regards,
Ali Pey



On Thu, Dec 12, 2013 at 11:19 AM, Ali Pey  wrote:

> Hello Miha,
>
> The best way to handle nat keep alive is from your sip clients. They
> usually offer a variety of option and the most efficient one is the udp
> pinging, in which it keeps sending opensips a udp packet with 4 zeros in it
> to keep the nat binding open. That would work perfectly if you have static
> nat binding on the opensips server which means doing port forwarding on the
> firewall the opensips sits behind.
>
> You can also enable options pinging on your client as well and that will
> keep nats open on both side as opensips will respond with 200 OK.
>
> Enabling nat keep alive from opensips is more expensive in terms of
> resources but that is also a possible way to do.
>
> Regards,
> Ali Pey
>
>
>
> On Thu, Dec 12, 2013 at 7:16 AM, Miha  wrote:
>
>>  HI,
>>
>> I need a little help with nat. UAC register's ok and it is reachable but
>> after a while it become unreachable due to nat issue. If I use
>> nat_keepalive on uac deivce everything is ok, but I would like that this is
>> handled by opensips.
>>
>> I tried to use  nat_keepalive but getting all the time this:
>>
>>  54.820695 opensips -> uac_public_ip SIP 308 Request: NOTIFY
>> sip:uac_public_ip:13401 |
>>  54.829847 uac_public_ip -> opensips SIP 297 Status: *481 Call
>> Leg/Transaction Does Not Exis*t |
>>
>> What is the best way to implement this?
>>
>>
>>
>> modparam("registrar", "received_avp", "$avp(received_avp)")
>> modparam("registrar", "tcp_persistent_flag", 10)
>> modparam("nathelper", "received_avp", "$avp(received_avp)")
>> modparam("nathelper", "natping_interval", 10)
>> modparam("nathelper", "ping_nated_only", 1)
>>
>>
>>   if (nat_uac_test("18")) {
>> xlog("fixing nat");
>>  if (method=="REGISTER") {
>> nat_keepalive();
>>  fix_nated_register();
>>
>> } else {
>> fix_nated_contact();
>> }
>> force_rport();
>> }
>>
>> tnx!
>>
>> miha
>>
>>
>>
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>>
>>
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Re: [OpenSIPS-Users] Nat_keepalive help

[Ali Pey]

Hello Miha,

The best way to handle nat keep alive is from your sip clients. They
usually offer a variety of option and the most efficient one is the udp
pinging, in which it keeps sending opensips a udp packet with 4 zeros in it
to keep the nat binding open. That would work perfectly if you have static
nat binding on the opensips server which means doing port forwarding on the
firewall the opensips sits behind.

You can also enable options pinging on your client as well and that will
keep nats open on both side as opensips will respond with 200 OK.

Enabling nat keep alive from opensips is more expensive in terms of
resources but that is also a possible way to do.

Regards,
Ali Pey



On Thu, Dec 12, 2013 at 7:16 AM, Miha  wrote:

>  HI,
>
> I need a little help with nat. UAC register's ok and it is reachable but
> after a while it become unreachable due to nat issue. If I use
> nat_keepalive on uac deivce everything is ok, but I would like that this is
> handled by opensips.
>
> I tried to use  nat_keepalive but getting all the time this:
>
>  54.820695 opensips -> uac_public_ip SIP 308 Request: NOTIFY
> sip:uac_public_ip:13401 |
>  54.829847 uac_public_ip -> opensips SIP 297 Status: *481 Call
> Leg/Transaction Does Not Exis*t |
>
> What is the best way to implement this?
>
>
>
> modparam("registrar", "received_avp", "$avp(received_avp)")
> modparam("registrar", "tcp_persistent_flag", 10)
> modparam("nathelper", "received_avp", "$avp(received_avp)")
> modparam("nathelper", "natping_interval", 10)
> modparam("nathelper", "ping_nated_only", 1)
>
>
>   if (nat_uac_test("18")) {
> xlog("fixing nat");
>  if (method=="REGISTER") {
> nat_keepalive();
>  fix_nated_register();
>
> } else {
> fix_nated_contact();
> }
> force_rport();
> }
>
> tnx!
>
> miha
>
>
>
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Re: [OpenSIPS-Users] convert UPDATE into re-INVITE

[Ali Pey]

Who is sending you the Update sip messages and for what purpose?

If it is for pinging, you can simply drop the update messages in opensips
and ping your cpe (and I don't know what you mean by cpe) by other means if
you need to.

The solution is not how to convert an Update message to Invite. You need to
look for ways to provide the same functionality and design with different
means and modules in opensips.

Regards,
Ali Pey


On Thu, Dec 5, 2013 at 3:52 AM, matrix  wrote:

> We have opensips as a voip switch.
> We are getting UPDATE sip message but some of our cpe is not supporting the
> UPDATE.
> Is there any way that we can handle UPDATE from opensips itself without
> passing it to cpe?
>
>
>
>
> --
> View this message in context:
> http://opensips-open-sip-server.1449251.n2.nabble.com/Re-convert-UPDATE-into-re-INVITE-tp7588822p7588848.html
> Sent from the OpenSIPS - Users mailing list archive at Nabble.com.
>
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Re: [OpenSIPS-Users] Users Digest, Vol 64, Issue 70

[Ali Pey]

Please always respond on a proper thread with a proper subject line. No one
can tell what you are talking about and to what thread you are responding
when you reply to a Users Digest email.

Regards,
Ali Pey


On Tue, Dec 3, 2013 at 4:47 AM, Chandra Prakash <
chandraprak...@virtualemployee.com> wrote:

> Thanks Jeff,
>
> And Sorry for the delayed response.
>
> The endpoints are Bria.
> Network :- 5 Mbps leased line which is directly connected to the server.
>
> I tried to reconfigure the RTPproxy but in vain.
>
> Thanks
>
> -Original Message-
> From: users-boun...@lists.opensips.org
> [mailto:users-boun...@lists.opensips.org] On Behalf Of
> users-requ...@lists.opensips.org
> Sent: 26 November 2013 23:05
> To: users@lists.opensips.org
> Subject: Users Digest, Vol 64, Issue 70
>
> Send Users mailing list submissions to
> users@lists.opensips.org
>
> To subscribe or unsubscribe via the World Wide Web, visit
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> or, via email, send a message with subject or body 'help' to
> users-requ...@lists.opensips.org
>
> You can reach the person managing the list at
> users-ow...@lists.opensips.org
>
> When replying, please edit your Subject line so it is more specific than
> "Re: Contents of Users digest..."
>
>
> Today's Topics:
>
>1. Delay with RTPproxy (Chandra Prakash)
>2. DNS SRV failover not working for B2BUA (1.10) (Jeff Pyle)
>3. Re: opensips sends CANCEL (Miha)
>4. Re: Delay with RTPproxy (Jeff Pyle)
>5. Re: DNS SRV failover not working for B2BUA (1.10) (Jeff Pyle)
>
>
>
>
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Re: [OpenSIPS-Users] Implementing routing and failure using perl

[Ali Pey]

Hello Nick,

(Since you are from Toronto, I looked that up for you. I'm from Ottawa)

It's fairly simple. In your perl code, you do lookup $rU and then set some
flag. Then back in your opensips route, you can examine this flag and route
accordingly. I don't think you can return a result set to your opensips
route.

sub check_number {
my $m = shift;
my $snumber = get_snumber( $m );
my $called = $m->pseudoVar( '$rU' );

if ( $called ne '' ) {
my $p = select_row( 'select number from numbers where number=' . quote(
$called ) . ' limit 1' );
if ( defined $p && $p->{ 'number' } ne '' ) {
$m->setFlag( 7 );
return 1;
}
}

$m->resetFlag( 7 );
return 1;
}


Regards,
Ali Pey


On Tue, Dec 3, 2013 at 12:45 PM, Nick Cameo  wrote:

> Hello Ali,
>
> Thank your for your response. Yes I was inquiring about a OpenSIPS
> perl API (use OpenSIPS). The only thing is I cannot find any API
> documentation in terms of what implementation is available. Only
> complex examples that I am looking into right now. With that said,
> what I am working on initially is not that complicated.
>
> For every INVITE:
>
> 1) Extract the TO info (substring first 3/4 digits)
> 2) Query the databases for all prefixes that LIKE the substring of the
> TO number order by price
> 3) do_routing (or whatever) for the first record in the result set.
> Failing over to the next etc..
>
> Kind Regards,
>
> Nick from Toronto.
>
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Re: [OpenSIPS-Users] Implementing routing and failure using perl

[Ali Pey]

Hello,

When you use perl, you would have to implement your own logic and you can't
use the dynamic routing or load balancer module from inside your perl
script. You must be thinking of something similar to ESL in freeswitch or
AGI in asterisk and that's not here.

Regards,
Ali


On Tue, Dec 3, 2013 at 12:17 PM, Nick Cameo  wrote:

> Hello Everyone,
>
> We use dr (ie, do_routing), and failure (ie, use_next_gw()) in our
> opensips configuration. Now we are trying to migrate certain
> functionality to our perl script where we do some processing on the
> INVITEs.
>
> My question is which predefined OpenSIPS perl functions do we have the
> will allow us to implement the same routing and failover functionality
> that we currently have using OpenSIPS script. More specifically, what
> Perl functions do we have to implement relaying and failover. As
> mentioned earlier, we do use DR so our dr_rules/gateways tables are
> populated.
>
> Kind Regards,
>
> Nick.
>
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Re: [OpenSIPS-Users] convert UPDATE into re-INVITE

[Ali Pey]

Can you elaborate more on what you are trying to do? Who is sending who an
update and for what purpose?

You can't convert a message to another message. They are different nature,
there is no such a thing and sip message converter.

Regards,
Ali Pey


On Tue, Dec 3, 2013 at 4:36 AM, M.D.Patel  wrote:

> I need some solution to over come situation in which cpe is not supporting
> UPDATE. Something that I can manage from opensips.
>
>
> On Tue, Dec 3, 2013 at 12:40 PM, M.D.Patel  wrote:
>
>> Hello,
>>
>> I have some of cpe which are not supporting UPDATE sip message.
>> How can I convert UPDATE into re-INVITE from opensips?
>>
>> Thanks.
>>
>
>
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Re: [OpenSIPS-Users] opensips for route traffic only

[Ali Pey]

Hello,

There are many ways you can do this. It depends on how many soft clients
you have.

Here is my recommendation:
1- Have all the phones register to your opensips. Asterisk does not need to
know about the registrations and can send/receive all calls only to/from
opensips.
2- Use the load balancer or dispatcher module to equally balance the
incoming load to both your asterisk servers. The load balancer would also
know if one asterisk server is down and will direct all calls to the other
or you can add more asterisk servers in the future. You could disable a
server to go to drain mode or assign resources.
3- For outbound, you could use drouting module to define rules and have
alternate routes.
4- You could also use the dial plan module for additional manipulation or
rules.

Regards,
Ali Pey


On Thu, Nov 28, 2013 at 4:36 AM, driver  wrote:

> Hello,
> I have such infrastructure:
>
> Asterisk1 (provider 1) - IP1
> Asterisk2 (provider 2) - IP2
>
> Softphones on client computers with configured two IPs to two Asterisk
> servers. User has to manually switch between each server.
>
> I need to change this that user will have only one account configured (eg.
> Opensips) and Opensips will be used only to forward to Ast1 or Ast2 by
> previously defined rules. How is it possible?
>
> comp1 > Opensips ---> Ast1
> comp2 > Opensips ---> Ast1
> comp3 > Opensips ---> Ast2
> comp4 > Opensips ---> Ast2
>
> Thank you,
> Marcin
>
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Re: [OpenSIPS-Users] Opensips as presence server + OpenIMS = 407 Status

[Ali Pey]

Hello Estefania,

Who sends the 407 message? OpenSIPS?

In opensips.cfg, you can examine the source IP and port and if it is from
you OpenIMSCore, then don't not authenticate it.

RTPProxy wouldn't have anything to do with this.

Regards,
Ali Pey


On Wed, Nov 27, 2013 at 2:05 PM, Estefania Figueroa Buitrago <
kaoru_chan0...@hotmail.com> wrote:

> Hello,
>
> I was just looking around and I found an example when they use a RTPProxy.
> I'm no implementing a RTPProxy. Has this something to do with this error?
>
> Regards,
> Estefania Figueroa
>
>
>
> --
> View this message in context:
> http://opensips-open-sip-server.1449251.n2.nabble.com/Opensips-as-presence-server-OpenIMS-407-Status-tp7588692p7588738.html
> Sent from the OpenSIPS - Users mailing list archive at Nabble.com.
>
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Re: [OpenSIPS-Users] Opensips 1.9.1 and NAT

[Ali Pey]

Why don't you use "nat_uac_test()" when you receive a request to examine
NAT? Using nat_uac_tes() you can tell if the message is coming from a
client behind the nat or not. With lookup and the nat_bflg, you can tell if
the destination is behind a nat. They are two different things.

Regards,
Ali Pey


On Mon, Nov 25, 2013 at 11:40 PM, dpa  wrote:

> “What is it that you are trying to do?”
>
>
>
> I try to check if message came from behind NAT or not and make some
> actions depending on result of checking.
>
> To speak the truth I tried to include in opensips.cfg logic about
> direction of the call.
>
>
>
> What I thought. I thought that nat_bflag activated only after lookup()
> function processing, i.e. when call goes TO UAC behind NAT and NOT
> activated when call came FROM UAC behind NAT. So in onreply route I make
> the test about this flag and waiting (in my current case) that the test of
> natb_flag will be failed.
>
>
>
>
>
> *From:* users-boun...@lists.opensips.org [mailto:
> users-boun...@lists.opensips.org] *On Behalf Of *Ali Pey
> *Sent:* Tuesday, November 26, 2013 8:16 AM
>
> *To:* OpenSIPS users mailling list
> *Subject:* Re: [OpenSIPS-Users] Opensips 1.9.1 and NAT
>
>
>
> As I said earlier, there is no such a thing as FROM or TO (i.e. direction)
> unless it's implemented in your logic. When a SIP request reaches the IP
> address of your opensips server, it goes through your routing script in
> opensips.cfg and then it's routed out. What you have there, it would apply
> to the request message. Put yourself in opensips shoes and follow the logic
> - that's how we debugged code while at uni :)
>
>
>
> If you have a concept of direction and you need different behavior based
> on your concept of direction, you need to implement it in your logic. For
> instance you can examine the source IP or subnet to decide if this message
> is from internal or external and then apply different logic to it - or
> whatever else that is specific to your environment.
>
>
>
> What is it that you are trying to do?
>
>
>
> Regards,
>
> Ali Pey
>
>
>
>
>
> On Mon, Nov 25, 2013 at 10:52 PM, dpa  wrote:
>
> OK, but in the current case another question.
>
> Is nat_bflag appear in transaction when call goes TO uac behind nat (by
> using lookup(), for example), or it appears  in transaction when call goes
> FROM uac behind nat?
>
>
>
> Thank you.
>
>
>
> *From:* users-boun...@lists.opensips.org [mailto:
> users-boun...@lists.opensips.org] *On Behalf Of *Ali Pey
> *Sent:* Monday, November 25, 2013 8:38 PM
>
>
> *To:* OpenSIPS users mailling list
> *Subject:* Re: [OpenSIPS-Users] Opensips 1.9.1 and NAT
>
>
>
> You also examining isbflagset(10). Isn't that set?
>
>
>
> Regards,
>
> Ali Pey
>
>
>
> On Mon, Nov 25, 2013 at 3:09 AM, dpa  wrote:
>
> Hello
>
>
>
> I understand but in onreply route I make a test: nat_uac_test("55") and
> only if it successful I make “fix_nated_contact()”.
>
> In my case nat_uac_test("55") must be fail after checking 183 ringing and
> 200 OK.
>
>
>
> *From:* users-boun...@lists.opensips.org [mailto:
> users-boun...@lists.opensips.org] *On Behalf Of *Ali Pey
> *Sent:* Friday, November 22, 2013 5:37 PM
> *To:* OpenSIPS users mailling list
> *Subject:* Re: [OpenSIPS-Users] Opensips 1.9.1 and NAT
>
>
>
> Hello,
>
>
>
> The question is not quite clear. In your opensips.cfg you call 
> "fix_nated_contact()"
> on both route and reply route. that's why it changes the route.
>
>
>
> There is no such a thing as direction in opensips unless it's implemented
> in your logic. A message goes through your route or reply route scripts
> when it reaches opensips.
>
>
>
> Regards,
>
> Ali
>
>
>
> On Fri, Nov 22, 2013 at 1:10 AM, dpa  wrote:
>
> Hello
>
>
>
> I have a one question about nat processing of Opensips.
>
>
>
> There is such scheme
>
>
>
> UAC1 (softphone behind nat) à Opensips -> UAC2 (another softswitch), i.e.
> UAC1 initiates a call to Opensips and a signaling port = 5068.
>
>
>
> So UAC2 becomes “ringing” by sending 183 message.
>
>
>
> In attachment 183 message from UAC2 and some parts of opensips.cfg
>
>
>
> After 183 processing by Opensips port in Contact header of 183 (and later
> 200 OK) messages become 5060, i.e. Opensips detects NAT and changes Contact
> header.
>
>
>
> So my question is, Why does Opensips changes Contact header?
>
> Once Opensips detects nat transaction (setting setflag(21)) is it check

Re: [OpenSIPS-Users] Opensips 1.9.1 and NAT

[Ali Pey]

As I said earlier, there is no such a thing as FROM or TO (i.e. direction)
unless it's implemented in your logic. When a SIP request reaches the IP
address of your opensips server, it goes through your routing script in
opensips.cfg and then it's routed out. What you have there, it would apply
to the request message. Put yourself in opensips shoes and follow the logic
- that's how we debugged code while at uni :)

If you have a concept of direction and you need different behavior based on
your concept of direction, you need to implement it in your logic. For
instance you can examine the source IP or subnet to decide if this message
is from internal or external and then apply different logic to it - or
whatever else that is specific to your environment.

What is it that you are trying to do?

Regards,
Ali Pey


On Mon, Nov 25, 2013 at 10:52 PM, dpa  wrote:

> OK, but in the current case another question.
>
> Is nat_bflag appear in transaction when call goes TO uac behind nat (by
> using lookup(), for example), or it appears  in transaction when call goes
> FROM uac behind nat?
>
>
>
> Thank you.
>
>
>
> *From:* users-boun...@lists.opensips.org [mailto:
> users-boun...@lists.opensips.org] *On Behalf Of *Ali Pey
> *Sent:* Monday, November 25, 2013 8:38 PM
>
> *To:* OpenSIPS users mailling list
> *Subject:* Re: [OpenSIPS-Users] Opensips 1.9.1 and NAT
>
>
>
> You also examining isbflagset(10). Isn't that set?
>
>
>
> Regards,
>
> Ali Pey
>
>
>
> On Mon, Nov 25, 2013 at 3:09 AM, dpa  wrote:
>
> Hello
>
>
>
> I understand but in onreply route I make a test: nat_uac_test("55") and
> only if it successful I make “fix_nated_contact()”.
>
> In my case nat_uac_test("55") must be fail after checking 183 ringing and
> 200 OK.
>
>
>
> *From:* users-boun...@lists.opensips.org [mailto:
> users-boun...@lists.opensips.org] *On Behalf Of *Ali Pey
> *Sent:* Friday, November 22, 2013 5:37 PM
> *To:* OpenSIPS users mailling list
> *Subject:* Re: [OpenSIPS-Users] Opensips 1.9.1 and NAT
>
>
>
> Hello,
>
>
>
> The question is not quite clear. In your opensips.cfg you call 
> "fix_nated_contact()"
> on both route and reply route. that's why it changes the route.
>
>
>
> There is no such a thing as direction in opensips unless it's implemented
> in your logic. A message goes through your route or reply route scripts
> when it reaches opensips.
>
>
>
> Regards,
>
> Ali
>
>
>
> On Fri, Nov 22, 2013 at 1:10 AM, dpa  wrote:
>
> Hello
>
>
>
> I have a one question about nat processing of Opensips.
>
>
>
> There is such scheme
>
>
>
> UAC1 (softphone behind nat) à Opensips -> UAC2 (another softswitch), i.e.
> UAC1 initiates a call to Opensips and a signaling port = 5068.
>
>
>
> So UAC2 becomes “ringing” by sending 183 message.
>
>
>
> In attachment 183 message from UAC2 and some parts of opensips.cfg
>
>
>
> After 183 processing by Opensips port in Contact header of 183 (and later
> 200 OK) messages become 5060, i.e. Opensips detects NAT and changes Contact
> header.
>
>
>
> So my question is, Why does Opensips changes Contact header?
>
> Once Opensips detects nat transaction (setting setflag(21)) is it check
> all reply messages (and doesn`t matter from which UAC they have been
> received) or Opensips can detect direction and makes decision about nat
> process?
>
>
>
> Thank you  for any help.
>
>
>
>
>
>
>
>
>
>
>
>
> ___
> 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
>
>
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Re: [OpenSIPS-Users] Opensips 1.9.1 and NAT

[Ali Pey]

You also examining isbflagset(10). Isn't that set?

Regards,
Ali Pey


On Mon, Nov 25, 2013 at 3:09 AM, dpa  wrote:

> Hello
>
>
>
> I understand but in onreply route I make a test: nat_uac_test("55") and
> only if it successful I make “fix_nated_contact()”.
>
> In my case nat_uac_test("55") must be fail after checking 183 ringing and
> 200 OK.
>
>
>
> *From:* users-boun...@lists.opensips.org [mailto:
> users-boun...@lists.opensips.org] *On Behalf Of *Ali Pey
> *Sent:* Friday, November 22, 2013 5:37 PM
> *To:* OpenSIPS users mailling list
> *Subject:* Re: [OpenSIPS-Users] Opensips 1.9.1 and NAT
>
>
>
> Hello,
>
>
>
> The question is not quite clear. In your opensips.cfg you call 
> "fix_nated_contact()"
> on both route and reply route. that's why it changes the route.
>
>
>
> There is no such a thing as direction in opensips unless it's implemented
> in your logic. A message goes through your route or reply route scripts
> when it reaches opensips.
>
>
>
> Regards,
>
> Ali
>
>
>
> On Fri, Nov 22, 2013 at 1:10 AM, dpa  wrote:
>
> Hello
>
>
>
> I have a one question about nat processing of Opensips.
>
>
>
> There is such scheme
>
>
>
> UAC1 (softphone behind nat) à Opensips -> UAC2 (another softswitch), i.e.
> UAC1 initiates a call to Opensips and a signaling port = 5068.
>
>
>
> So UAC2 becomes “ringing” by sending 183 message.
>
>
>
> In attachment 183 message from UAC2 and some parts of opensips.cfg
>
>
>
> After 183 processing by Opensips port in Contact header of 183 (and later
> 200 OK) messages become 5060, i.e. Opensips detects NAT and changes Contact
> header.
>
>
>
> So my question is, Why does Opensips changes Contact header?
>
> Once Opensips detects nat transaction (setting setflag(21)) is it check
> all reply messages (and doesn`t matter from which UAC they have been
> received) or Opensips can detect direction and makes decision about nat
> process?
>
>
>
> Thank you  for any help.
>
>
>
>
>
>
>
>
>
>
>
>
> ___
> 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] Opensips 1.9.1 and NAT

[Ali Pey]

Hello,

The question is not quite clear. In your opensips.cfg you call
"fix_nated_contact()"
on both route and reply route. that's why it changes the route.

There is no such a thing as direction in opensips unless it's implemented
in your logic. A message goes through your route or reply route scripts
when it reaches opensips.

Regards,
Ali


On Fri, Nov 22, 2013 at 1:10 AM, dpa  wrote:

> Hello
>
>
>
> I have a one question about nat processing of Opensips.
>
>
>
> There is such scheme
>
>
>
> UAC1 (softphone behind nat) à Opensips -> UAC2 (another softswitch), i.e.
> UAC1 initiates a call to Opensips and a signaling port = 5068.
>
>
>
> So UAC2 becomes “ringing” by sending 183 message.
>
>
>
> In attachment 183 message from UAC2 and some parts of opensips.cfg
>
>
>
> After 183 processing by Opensips port in Contact header of 183 (and later
> 200 OK) messages become 5060, i.e. Opensips detects NAT and changes Contact
> header.
>
>
>
> So my question is, Why does Opensips changes Contact header?
>
> Once Opensips detects nat transaction (setting setflag(21)) is it check
> all reply messages (and doesn`t matter from which UAC they have been
> received) or Opensips can detect direction and makes decision about nat
> process?
>
>
>
> Thank you  for any help.
>
>
>
>
>
>
>
>
>
>
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
<>___
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Re: [OpenSIPS-Users] OpenSIPS + ingress profit protectioin

[Ali Pey]

Yes, I would also recommend drouting module as well. It's quite powerful
for handling scenarios like this. On the egress, try the load balancer
module.

Regards,
Ali Pey

On Tue, Nov 19, 2013 at 10:40 PM, Nick Cameo  wrote:

> I sent that email too fast from the handheld. I am currently working
> on the same thing. What I will end up doing is using Dynamic Routing
> and perl script per INVITE. Or you can do DR pre-processing of the
> routes.
>
> Kind Regards,
>
> Nick from Toronto.
>
> ___
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>
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Re: [OpenSIPS-Users] Opensips Users Registartion Problem

[Ali Pey]

Hello Vishnu,

You can look at the opensips logs or network traces to see what's
happening. No one can tell you what's wrong with your system because we
don't know what you have done. You need to do your debugging and post
specific question that can be answered.

Regards,
Ali Pey


On Thu, Nov 14, 2013 at 1:38 AM, Vishnu Vardhan wrote:

> Hi,
>
> I done the installation of opensips 1.8 with opensips cp-5.0 in centos
> 6.4.Every thing should be fine,and i intigrated the asterisk11.0  to
> opensips server.I add the users in opensips UI page but when i try to
> register the users in Softphone i am getting error,the users are not
> registering in softphone.I structed here please give the suggetion to get
> out of this problem and how to check asterisk in integrated to opensips or
> not.Please give the suggestion to get out of this problem.
>
> Regards,
> Vishnu
>
>
> ___
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>
>
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Re: [OpenSIPS-Users] perl script abnormal termination handling

[Ali Pey]

Don't do die in the perl script. Try return 0 or 1.

Regards,
Ali Pey


On Fri, Nov 8, 2013 at 10:32 AM, Jeff Pyle  wrote:

> Hello,
>
> I run a perl script from an Opensips 1.6 config.  I'm having trouble
> handling a perl "die" condition in the Opensips script.  For example:
>
> if !(perl_exec("dietest1")) {
> xlog("L_INFO", "Perl script failed.\n"
> } else {
> xlog("L_INFO", "Perl script succeeded.\n"
> }
>
> where test1.pl ends with:
> die "Perl script dying"
>
> When the perl script fails, I see only:
>
> ERROR:core:XS_OpenSIPS__Message_log: perl error: Perl script dying at
> /etc/opensips/perl/perltest.pl line 165.#012
>
> I never see the "Perl script failed" xlog from the script.  It seems that
> a dying perl script stops processing on the current message.  Is there a
> way around this, to handle the perl error?
>
>
> Regards,
> Jeff
>
>
>
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Re: [OpenSIPS-Users] Dialog module sql server 2008 r2 database backend

[Ali Pey]

Is the problem the create table command?

Connect to your db using the isql command and run the create table command
and see if it works that way? Or create the table separately on your DB
directly.

There seems to be an incompatibility issue. Try to find what command or
field is causing it.

Regards,
Ali Pey


On Thu, Nov 7, 2013 at 5:14 PM, bluerain  wrote:

> As I stated in my reply to dave, yes I have that setup, I have all module
> running on sql server 2008 r2 via freetds, the only module does not load
> correctly is the dialog module.  That is why I am wondering what is wrong.
>
>
>
> --
> View this message in context:
> http://opensips-open-sip-server.1449251.n2.nabble.com/Dialog-module-sql-server-2008-r2-database-backend-tp7588430p7588436.html
> Sent from the OpenSIPS - Users mailing list archive at Nabble.com.
>
> ___
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Re: [OpenSIPS-Users] uac_registrant, how to force registration port?

[Ali Pey]

Hello,

This has nothing to do with opensips. This is your router that is picking a
random port when it does NAT.

How you can fix it: it all depends on what kind of router you have. Some
allow static NAT binding. The cheap routers don't have that flexibility.

Some routers, if you do a port forwarding from your public IP:port to
opensips, they use that binding for outgoing NAT as well - just another
thing to try.

It also depends on how many devices you have on your LAN side of the router
and if any other device is using that port.

Also, sometimes when you reboot the router, it gives you the same port as
opensips and will keep it as long as it's up.

So, just do some research on your router.

Regards,
Ali Pey


On Thu, Nov 7, 2013 at 2:48 PM, Max E. Reyes Vera J. wrote:

> Hi,
>
> I was testing the uac_registrant module to register opensips with my sip
> provider, but I have one issue, when the opensips is registered it use a
> random port, so when I go to my account details(on the sip provider web
> page) I can see the status as Registered with my Public_IP:RandomPort, i.e:
> 189.256.36.12:10623. I tried to use the forced_socket field but with no
> luck, every time it register with random port.
>
> My opensips is behind NAT
>
> opensips-->router-->sip provider.
>
> Is there a way to force the port on registration just like asterisk does
> with the 5060 port?
>
> Thanks & best regards.
>
> PD: How can I reply future answers. I mean, I just click on the owner's
> thread email and that open my mail client with the RE:subject, but when I
> check the list my response appear like a new thread. :(
>
> --
> Saludos Cordiales
>
> *dCAA*. Max E. Reyes Vera Juárez
> Desarrollo y Soporte Asterisk  / Hardware
> navai...@gmail.com
> mreyesv...@digital-merge.com
> (+52) 55 4170-3703 xt 5000
> www.Digital-Merge.com
>
> ___
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> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
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Re: [OpenSIPS-Users] Dialog module sql server 2008 r2 database backend

[Ali Pey]

You need to use ODBC and freetds. You can't directly connect to MS SQL.

It works quite well with ODBC and freetds.

Regards,
Ali Pey


On Thu, Nov 7, 2013 at 3:38 PM, David J  wrote:

> I hope this does not sound silly but what driver are you using to connect
> to mssql? Is it just odbc? I am not sure opensips will support mssql.
> On Nov 7, 2013 3:07 PM, "bluerain"  wrote:
>
>> Anybody tried to use the dialog module with sql server 2008 r2 back end?
>>
>> Opensips will start ok if I use mysql as db for dialog module, but as soon
>> as I switch to sql server it give me this error in the log
>>
>> Nov  7 19:59:50 OSIPIBD-2 /usr/local/sbin/opensips[16842]:
>> INFO:dialog:mod_init: Dialog module - initializing
>> Nov  7 19:59:50 OSIPIBD-2 kernel: [ 3628.711987] opensips[16842]: segfault
>> at 0 ip   (null) sp 71abbf78 error 14 in
>> opensips[40+21f000]
>>
>>
>> Here is the script for my sql server dialog table:
>>
>> CREATE TABLE [dbo].[dialog](
>> [dlg_id] [bigint] IDENTITY(1,1) NOT NULL,
>> [callid] [char](255) NOT NULL,
>> [from_uri] [char](128) NOT NULL,
>> [from_tag] [char](64) NOT NULL,
>> [to_uri] [char](128) NOT NULL,
>> [to_tag] [char](64) NOT NULL,
>> [mangled_from_uri] [char](64) NULL,
>> [mangled_to_uri] [char](64) NULL,
>> [caller_cseq] [char](11) NOT NULL,
>> [callee_cseq] [char](11) NOT NULL,
>> [caller_ping_cseq] [int] NOT NULL,
>> [callee_ping_cseq] [int] NOT NULL,
>> [caller_route_set] [varchar](max) NULL,
>> [callee_route_set] [varchar](max) NULL,
>> [caller_contact] [char](128) NOT NULL,
>> [callee_contact] [char](128) NOT NULL,
>> [caller_sock] [char](64) NOT NULL,
>> [callee_sock] [char](64) NOT NULL,
>> [state] [int] NOT NULL,
>> [start_time] [int] NOT NULL,
>> [timeout] [int] NOT NULL,
>> [vars] [varchar](max) NULL,
>> [profiles] [varchar](max) NULL,
>> [script_flags] [int] NOT NULL,
>> [flags] [int] NOT NULL
>> ) ON [PRIMARY] TEXTIMAGE_ON [PRIMARY]
>>
>> any thoughts?  Thank you!
>>
>>
>>
>> --
>> View this message in context:
>> http://opensips-open-sip-server.1449251.n2.nabble.com/Dialog-module-sql-server-2008-r2-database-backend-tp7588430.html
>> Sent from the OpenSIPS - Users mailing list archive at Nabble.com.
>>
>> ___
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>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>
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>
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Re: [OpenSIPS-Users] Request For Opensips Information

[Ali Pey]

Hi Vishnu,

Yes, those versions are supported and you can install on one server or
multiple servers. OpenSIPS can not be explained in an email. You need to
read the documentation: http://www.opensips.org/Documentation/Manuals

Try the installation process and ask specific questions if you don't
understand something.

Regards,
Ali Pey


On Thu, Nov 7, 2013 at 2:46 PM, Vishnu Vardhan  wrote:

> Hi,
>
> I am new to opensips.Opensips will support Centos 6.4 and asterisk
> 11/12.And i am getting confusion how to install openSips and asterisk
> is that are both to install in different systems or we manage with in
> same system.And please guide me for this installation of both Opensips
> and Asterisk,how to integrate this both .Please share any installed
> steps of this both.
>
> Regards,
> Vishnu
>
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Re: [OpenSIPS-Users] TCP to UDP proxy

[Ali Pey]

Add this in your route:

subst_uri('/(sip:.*);transport=tcp/\1/');

To remove transport=tcp from request-uri.

This should fix it.

Regards,
Ali Pey


On Wed, Nov 6, 2013 at 2:15 PM, Simon Quigley wrote:

> Hi,
>
> I've done some more testing, and it seems the issue I'm having with the
> NATed contact is related to the UA I'm using.
>
> If I use empathy/telepathy-sofiasip, then the contact isn't being mangled
> properly, which causes it to send the 200 OK packet after answering,
> receive no ACK, and continue resending the 200 OK, until it gives up and
> sends a BYE.
>
> The same behaviour occurs if I use sipdroid, however it doesn't send a BYE
> when it doesn't receive an ACK, but the call still isn't up properly, and
> when I do hang up, it doesn't send a BYE, so the originating UA (and
> switch) thinks the call is still up.
>
> However, if I use the built in Android SIP client, it's contact is mangled
> properly, and so the OK / ACK process works properly.
>
> How can I debug to see what the fix nated contact function is doing, to
> see if I can work out what the differences between these UA's packets are?
>
> Thanks
>
>
> On Fri, Nov 1, 2013 at 5:44 PM, Simon Quigley wrote:
>
>> Hi Bogdan,
>>
>> Thanks for your response, your suggestion was much appreciated, and gave
>> me a good direction to work in.
>>
>> You are correct, I was trying to mix stateless and stateful routing
>> together, I realised after watching the webinar.
>>
>> I've made some changes to my configuration file to use record_route() and
>> loose_route(), in a few different ways, but it hasn't made any difference.
>>
>> The 200 OK is always sent back to the switch containing the non routable
>> IP in the contact of the device, rather than fix_nated_contact changing it
>> to the IP the packet arrives from, and it's also failing to match it and
>> translate it for the ACK response to get to the device.
>>
>> I've enabled debugging, and looking through the output, I think it's an
>> issue with matching the dialogue.
>>
>> Here's the config I'm using now:
>>
>>
>> listen=udp:[internal VM IP]:5060
>> listen=tcp:[internal VM IP]:5060
>>
>> disable_tcp=no
>>
>> advertised_address="[dns name]"
>> alias="[dns name]"
>>
>>
>> route{
>> if (src_ip != "[switch ip]"){
>> # from device
>> if (!is_method("REGISTER|MESSAGE")) {
>>
>> if (is_method("INVITE")) {
>> fix_nated_sdp("7");
>> record_route();
>>
>> }
>> else if (is_method("OPTIONS")){
>> exit;
>> }
>> else {
>> loose_route();
>> fix_nated_contact();
>> if (has_totag()){
>> fix_nated_sdp("7");
>> }
>> t_relay("udp:[switch ip]:5060");
>>
>> }
>> }
>> }
>>
>> else if (src_ip == "[switch ip]"){
>> # from switch
>> if (is_method("INVITE")) {
>>  fix_nated_contact();
>> fix_nated_sdp("7");
>> record_route();
>> }
>> else {
>> loose_route();
>>
>> }
>>
>> # mangle to TCP
>> $du = $ru + ";transport=tcp";
>>
>> t_relay();
>> }
>> exit;
>> }
>>
>> Is there something obvious that I'm doing wrong?
>>
>> Thanks again for your help.
>>
>>
>> On Wed, Oct 30, 2013 at 4:52 AM, Bogdan-Andrei Iancu > > wrote:
>>
>>>  Hello Simon,
>>>
>>> Most probably it is a mistake on the way you do record_routing (for
>>> initial INVITE) and loose_route (for sequential requests). All sequential
>>> requests are driven by this mechanism. If the case, check how it suppose to
>>> work :
>>> http://www.opensips.org/Documentation/Webinars#toc12  (5.5)
>>>
>>> Basically you need to do record_route() for INVITE and loose_route() for
>>> ACK - OpenSIPS will take care of changing the interface for the ACK after
>>> the loose_route() call.
>>>
>>> Regards,
>>>
>>> Bogdan-Andrei Iancu
>>> OpenSIPS Founder and Developerhtt

[OpenSIPS-Users] Astricon Discount Code

[Ali Pey]

Hello Everyone,

Digium has graciously offered a %20 registration discount for OpenSIPS
community. Discount Code: AC13DIGI


Don't forget to also register for OpenSIPS Summit following Astricon on
Friday  Oct 11:

http://www.opensips.org/Community/Summit-2013Atlanta

Email me if you need a discount code for this event.


Regards,
Ali Pey
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Re: [OpenSIPS-Users] OpenSIPS Documentation re-Factory

[Ali Pey]

Hello,

The new documentation lay out is much much better and it is much easier to
find the info you need faster. Thanks for the great work.

I found a broken link though: Database schema

Overall, well done!!

Regards,
Ali Pey



On Mon, Jun 3, 2013 at 5:01 AM, Bogdan-Andrei Iancu wrote:

> **
> Hi Pete,
>
> We avoided to purge the old URLs (actually we reconfigured all of them as
> redirects to the new corresponding pages) to avoid external broken links
> (there are external pages linking parts of the docs by the old pages).
>
> The Indexing of the functions is not yet addressed - we try to find a way
> to automatize the way the Index is generating. The 1.4 one is manually
> done, but this simply does not work on regular bases :(.
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
>
> On 06/02/2013 10:42 AM, Pete Kelly wrote:
>
> The main menus at least are much better than before.
>
>  One of the problems with the old docs was that some of the links still
> pointed to v1.4 versions etc and you really had to click a lot of links to
> get through to the modules documentation.
>
>  Now it's in a tree structure and contains the version at the end it's
> much cleaner.
>
>  Is there a way to purge the old URLs so that any search results won't
> bring them up? The function index is very useful but it seems to be stuck
> on 1.4 still:
>
>  http://www.opensips.org/Documentation/Script-FunctionsIdx-1-4
>
>
>  On 31 May 2013 13:43, Bogdan-Andrei Iancu  wrote:
>
>> Hello,
>>
>> In the past weeks, we undertook a complex process of re-factoring the
>> OpenSIPS documentation on the OpenSIPS web site.
>>
>> The main targets of this re-factoring were:
>>
>> - enhancing docs by covering new areas like installing, configuring,
>> db managing
>>
>> - creating an OpenSIPS Manual with a logic flow to help you with
>> understanding OpenSIPS from downloading to scripting.
>>
>> - re-structuring in a per-version approach (you can jump across
>> versions for the same topic)  and  with a "next" and "prev" linking (to
>> go through documentation in a more fluid way).
>>
>>
>> The root of everything is http://opensips.org/Documentation/Manuals ,
>> please give it ride and let us know your opinion on this:
>>  - is it better than before ?
>>  - missing topics in manual ?
>>  - broken links ?
>>
>> Feeback is welcome and it we will make it matter ;)
>>
>>
>> Best regards,
>>
>> --
>> Bogdan-Andrei Iancu
>> OpenSIPS Founder and Developer
>> http://www.opensips-solutions.com
>>
>>
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>>
>
>
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>
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Re: [OpenSIPS-Users] [OpenSIPS-Devel] OpenSIPS Tracker Migration on GitHub

[Ali Pey]

I would vote for the second option. It would take a bit to clean up but it
would be cleaner and much less work. Usually when you move things, there
are errors, mistakes, etc. It would be cleaner and less work with option 2.

Regards,
Ali



On Tue, Jun 4, 2013 at 9:18 AM, Răzvan Crainea  wrote:

> Hello all!
>
> After migrating all the OpenSIPS repositories from SourceForge to
> GitHub[1], the next step is to proceed with the tracker migration. We wrote
> down two options here:
>
> 1. Move all tickets (patches, feature requests, bugs) to the GitHub
> tracker. There are several problems with this approach: first is that we
> won't be able to preserve the submitter information, therefore it will no
> longer be updated about further updates (comments, fixes, etc). Moreover
> this has to be done manually, so it requires a lot of time and effort.
>
> 2. Keep all the tickets already opened on SourceForge, but open new
> tickets on GitHub. In this case you will no longer be able to open tickets
> (this will be blocked from SourceForge), but only to update the new ones
> (comment, close, etc). All new tickets will have to be opened on the GitHub
> tracker.
>
> We prefer the second option because it overcomes all the issues of the
> first one. The downside is that for a certain period of time we will have
> to follow two different tracks (both SourceForge and GitHub). However, we
> believe that this is an acceptable compromise.
>
> Let us know what you think about this plan. Our intention is to do the
> migration process as smoothly as possible. So if you have a different idea,
> don't hesitate to share it with us!
>
> [1] 
> http://www.opensips.org/About/**GitHub-Migration
>
> Best regards,
> --
> Razvan Crainea
> OpenSIPS Core Developer
> http://www.opensips-solutions.**com 
>
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> de...@lists.opensips.org
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[OpenSIPS-Users] Send a SIP Notify message to a registered SIP Phone (config resync)

[Ali Pey]

Hello,

What's the best way to generate and send a custom Notify SIP message to a
registered sip phone (for configuration resync)?

Thanks,
Ali Pey
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Re: [OpenSIPS-Users] [OpenSIPS-Devel] [RFC] New Release Policy for OpenSIPS project

[Ali Pey]

I agree with this. Not every release needs to be supported for a full 2
years.

Regards,
Ali Pey



On Fri, Nov 30, 2012 at 12:24 PM, Ryan Bullock  wrote:

> On Mon, Nov 26, 2012 at 1:57 PM, Bogdan-Andrei Iancu 
> wrote:
>
>> **
>> Hi Ali,
>>
>> Thanks for feedback - regarding the support for previous releases, Saul
>> raised the same point as you, and I have to admit I didn't do the math - 2
>> release ~= 1 year, which indeed is too short - I mean this will force an
>> upgrade each year.
>>
>> So, we need to somehow get to ~ 2 year lifetime for a release. My
>> suggestion is to actually set a life span for 2 years.
>>
>> Regards,
>>
>> Bogdan-Andrei Iancu
>> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>>
>> What about adding a long term support branch that is released every two
> years and supported for 2 years, and then a release every 6 months for
> 'standard' releases. Each standard release would be supported for 1 year.
>
> Something like this, assuming 1.10 is the first long term support:
> 1.10 - Long term support (2 years)
> 1.11 - Standard release (1 year)
> 1.12 - Standard release (1 year)
> 1.13 - Standard release (1 year)
> 1.14 - Long term support (2 years)
>
> Those wanting new features can go for the standard releases, and those
> looking for stability and better support can stick with the long term
> support releases. It should strike a decent balance between getting
> features out the door and support.
>
> Just a thought.
>
> Regards,
>
> Ryan
>
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Re: [OpenSIPS-Users] [RFC] New Release Policy for OpenSIPS project

[Ali Pey]

Hi Bogdan,

This is great to see and I quite like the more open, predictable and
transparent approach. I also agree that a time driven release cycle is more
practical than feature driven. There are always grey areas and exceptions
but in practice a time driven release cycle is much better manageable for
real world deployments.

In terms of support for previous releases, as much as it would allow a
deployment to go on for two years before it requires an upgrade I agree.
It's not practical to have more frequent upgrade cycles. Also most people
don't usually upgrade to the latest version. For instance when 1.9 comes
out, we probably will upgrade to 1.8.2.

Again, this is a great step forward for opensips development and thank you
for the great work.

Regards,
Ali Pey



On Fri, Nov 23, 2012 at 4:36 AM, Saúl Ibarra Corretgé
wrote:

> Hi,
>
> > The problem I see with the features-based release cycle is that they are
> unpredictable as time - some features may not be properly (or impossible)
> time evaluated -> it may stretch the interval between releases ; IMHO, for
> a project to reliable it is a must to be predictable. The best examples are
> what is happening now with OpenSIPS (the interval between releases is keep
> growing) and Debian (lack of predictability and huge intervals between
> release ended up in the Ubuntu alternative).
> > Being able to predict the releases (as time) without huge differences
> between versions (to make an upgrade something easy you are not scared like
> hell to do it) should be some key-feature of the project.
> >
> > The time-based releases should not be affected by how long a feature
> takes to be implemented - 6 months of development for a feature is really
> more than enough, IMHO.
> >
>
> I agree that is good for bugfix releases. However, when planning the next
> release (lets say 1.10) I guess you'd plan what features are to be
> implemented. Then of course time needs to be weighted in the equation, but
> IMHO the time constraint should be a bit more loose for major releases.
>
> >
> > PS: let me ask you: how many OpenSIPS installations do you still have
> running old versions because upgrade is really painful ? ;)
>
> Fortunately not many :-) I had to migrate from 1.4 to 1.8 and, well,
> things can get complicated. Of course the gap is big and wouldn't be so big
> between 1.7 and 1.8 or between 1.8 and 1.9, but updating requires time and
> a reason. If a customer has a working OpenSIPS version and I update it just
> for the sake of it, new bugs can be introduced, and he'll probably not see
> any of the new features because he doesn't need them, for example. This is
> what I mean by not taking it lightly.
>
> [snip]
>
> >> What about security fixes? I can understand that when 1.9 is released
> 1.7 goes to EOL (sort of), but what if there is a bug in the parser (for
> example) which can cause a crash just by using a stupid script? IMHO there
> should be a security-fixes-only period, since migrating to a new OpenSIPS
> version is not  a task to be taken lightly.
> > [bogdan]
> > That is true problem that may have as solutions:
> >1) simply upgrade (most common way to go in open source world) ,
> considering that upgrades should become easier.
>
> New versions have new features and new bugs. So updating may get you
> trouble for little gain, in case you are not using any of the new stuff.
>
> >2) try to define what is really critical (based on what??) and still
> do backporting - but at the end of the day we need to encourage people to
> use the new versions - keep patching and supporting really old versions
> (consider 1.6 at this point) is a waist of effort. Taking your example:
> debian is not supporting something older than 1 release :D
> >
>
> Not 100% accurate: -) "The security team tries to support a stable
> distribution for about one year after the next stable distribution has been
> released". So Debian "oldstable" still gets security updates a year after
> "stable" has been out.
>
> We can try to see how a similar approach can work out for us. Instead of a
> year, say 6 months. What's important is to define what a security fix is
> and what it's not. An error in the software that can be consistently
> triggered from the outside (ie, with SIP traffic) and cause any kind of
> outage could be considered a security fix. This is from the top of my head,
> it would need to be refined :-)
>
>
> Regards,
>
> --
> Saúl Ibarra Corretgé
> AG Projects
>
>
>
>
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Re: [OpenSIPS-Users] OpenSIPS Control Panel limitation with table dialplan for more than 40k rows.

[Ali Pey]

I think this was fixed in opensips-cp last week (or the week before). Make
sure you get the latest fixes from trunk and try again.

Regards,
Ali Pey


2012/11/21 Muhammad Shahzad 

> Its most like your web server issue, you need to increase output buffer
> size. If using Apache check error_log file, it will tell you exactly what
> is happening.
>
> Thank you.
>
>
> 2012/11/7 Miguel J. 
>
>> **
>> Dear Opensips lists:
>>
>> I've found a trouble with the OpenSips Control Panel tool when the
>> dialplan table has more than 4 rows.
>>
>> If this table contains up to 40,000 rows, the content is provided in
>> the web tool and everything works fine. If this table contains a record
>> more, the tool leaves the panel empty, as I show in the attached document.
>>
>> Can I configure or adapt the tool to avoid this limitation?
>>
>> Thank you very much and best regards
>>   --
>>
>>
>>
>> -
>> Sus datos de carácter personal (nombre, apellidos, dirección postal y de
>> correo electrónico, etc.) son tratados para la gestión de su relación con
>> la Entidad, así como para el envío de información sobre nuestra actividad y
>> la de terceros relacionadas con la actividad de Consulting Smartic
>> Solutions, S.L., CIF: B85130037, C/Pº de la Castellana, 135, 7ª planta,
>> 28046 Madrid. Usted puede ejercer sus derechos de acceso, rectificación,
>> cancelación y oposición dirigiéndose por escrito, con copia de un documento
>> que acredite su identidad, a la dirección info (arroba) smartic.es.
>> Este mensaje puede contener información confidencial. Si usted no es su
>> destinatario, no debe leerlo, copiarlo, distribuirlo, ni hacer uso de la
>> información que contiene. En este caso, por favor, llámenos o
>> comuníquenoslo por escrito y borre este mensaje de su sistema.
>> -
>>
>>
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Re: [OpenSIPS-Users] how to Register with OpenSIPS?: help needed!!

[Ali Pey]

Yes, all the account info would be in opensips. Your pbx's configuration
will be quite simpler.
There are many ways that opensips can distribute the calls to PBXs. You can
use the load balancer, drouting or dispatcher modules. If you don't have
many pbxs, you can simply include the ip addresses in your opensips.cfg
file. The call will be forwarded to those ip addresses.

Have a look at registrar, authentication, load balancer, drouting,
dispatcher and dial plan modules. Also there are three sample opensips.cfg
files created by opensips installation that you can use as a reference.

Regards,
Ali Pey


On Thu, Nov 15, 2012 at 9:46 AM, Christian Cambier  wrote:

>  Hello.
>
> >"Your sip phones only register to the opensips servers. Your pbx dosen't*
> ***
>
> >need to sip registrations."
>
> But what do you do then with the account-settings that were created on the 
> PBX?
>
> Account-settings like username and password, do you need to redefine them on 
> the proxy then and not anymore on the PBX?
>
> >Your sip phones only talk to the opensips servers, opensips then will
>
> >forward the calls to the pbx.
>
> But how does the opensips know to which PBX to forward the call to.
> Is has to be defined somewhere? But where and how please?
>
> 
>
>
> regards
>
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Re: [OpenSIPS-Users] how to Register with OpenSIPS?: help needed!!

[Ali Pey]

Your sip phones only register to the opensips servers. Your pbx dosen't
need to sip registrations.
Your sip phones only talk to the opensips servers, opensips then will
forward the calls to the pbx.
The pbx only talks to the opensips servers, then opensips will send the
call to the sip phones based on the registration server.
Basically opensips would be your proxy server. It keeps and digests all the
registration info and will handle the routing between your sip
clients/trunks and your pbxs.
You need some reading and some help. This is the general idea.

Regards,
Ali Pey



On Thu, Nov 15, 2012 at 7:59 AM, Christian Cambier  wrote:

>  Hi.
>
> Here's what i try to do: I have a SipPhone with extension 5006
> Without proxy i register at a PBX with address (10.1.2.3). Ok, no problem
>
> What I'm struggling with is, how can i have this phone register at that
> PBX when every SIP request has to pass via a OpenSIPS proxy first.
> Somehow the proxy must know how to forward REGISTER messages to the PBX
> no? I'd say, It must know the existence of the PBX.
> Or don't I need the PBX for REGISTER?
> But surely, INVITE must pass the PBX no?
>
> Anyway, how do I have a cooporate OpenSIPS proxy with a PBX?
> Where to configure what?
>
> Please help!!!
>
> thx
> Chris
>
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Re: [OpenSIPS-Users] how to Register with OpenSIPS?

[Ali Pey]

What do you mean by 'Registrar Changes'?

You can basically have your sip phones register with the opensips
server(s). Your pbx would only need to know the opensips ip address. You
can have redundant opensips servers with a virtual ip address and
heartbeat. When one goes down, the other one would take over. All the
registration info would be in a database such as mysql.

Regards,
Ali Pey


On Thu, Nov 15, 2012 at 3:13 AM, Christian Cambier  wrote:

>  Hi.
>
> ** **
>
> I'm new to using SIP proxies (OpenSIPS) so maybe it is a basic question
> but anyway...
>
> ** **
>
> I have a few SIP phones that register at some PBX (say 10.2.3.4)
>
> Registration without a proxy works fine
>
> ** **
>
> What I'd like to do now is to use OpenSIPS as proxy and define the
> registrar for all my clients there.
>
> So that when the registrar changes i only have to change it in the proxy
> instead of in all the clients
>
> ** **
>
> Is that possible? if so, how do you implement it?
>
> By specifying a route in some config file? 
>
> ** **
>
> thx
>
> Chris
>
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Re: [OpenSIPS-Users] 404 Not Found error with Sipp UAC

[Ali Pey]

Hi Steve,

The request-uri in your invite is to "service". In your opensips script, it
does a lookup on user and I don't think you have the user "service"
registered.

Either have sipp to register first, or in your opensips.cfg, put a
condition for calls from sipp to go to route one directly with no lookup.

Hope this is clear enough.

Regards,
Ali Pey


On Thu, Nov 8, 2012 at 8:05 PM, Steve Mitchell  wrote:

> Hi,
>
> I'm trying to get a simple scenario working to generate CDRs in bulk and
> am using a basic configuration (generated with osipconfig) with the sipp
> UAC. However, I continue to get a 404 Not Found response to the INVITE. My
> config and UAC file are below.
>
> Any thoughts?
>
> Thanks much!
>
> Steve
>
> #
> # $Id: opensips_residential.m4 9042 2012-05-17 13:57:10Z vladut-paiu $
> #
> # OpenSIPS residential configuration script
> # by OpenSIPS Solutions 
> #
> # This script was generated via "make menuconfig", from
> #   the "Residential" scenario.
> # You can enable / disable more features / functionalities by
> #   re-generating the scenario with different options.#
> #
> # Please refer to the Core CookBook at:
> #  http://www.opensips.org/Resources/DocsCookbooks
> # for a explanation of possible statements, functions and parameters.
> #
>
>
> ### Global Parameters #
>
> debug=6
> fork=no
> log_stderror=yes
> log_facility=LOG_LOCAL1
>
> #fork=yes
> #children=4
>
> /* uncomment the following lines to enable debugging */
> #debug=6
> #fork=no
> #log_stderror=yes
>
> /* uncomment the next line to enable the auto temporary blacklisting of
>not available destinations (default disabled) */
> #disable_dns_blacklist=no
>
> /* uncomment the next line to enable IPv6 lookup after IPv4 dns
>lookup failures (default disabled) */
> #dns_try_ipv6=yes
>
> /* comment the next line to enable the auto discovery of local aliases
>based on revers DNS on IPs */
> auto_aliases=no
>
>
> listen=udp:10.145.185.49:5060   # CUSTOMIZE ME
>
> disable_tcp=no
> listen=tcp:10.145.185.49:5060   # CUSTOMIZE ME
>
> disable_tls=yes
>
>
> ### Modules Section 
>
> #set module path
> mpath="/usr/local/opensips_proxy/lib64/opensips/modules/"
>
>  SIGNALING module
> loadmodule "signaling.so"
>
>  StateLess module
> loadmodule "sl.so"
>
>  Transaction Module
> loadmodule "tm.so"
> modparam("tm", "fr_timer", 5)
> modparam("tm", "fr_inv_timer", 30)
> modparam("tm", "restart_fr_on_each_reply", 0)
> modparam("tm", "onreply_avp_mode", 1)
>
>  Record Route Module
> loadmodule "rr.so"
> /* do not append from tag to the RR (no need for this script) */
> modparam("rr", "append_fromtag", 0)
>
>  MAX ForWarD module
> loadmodule "maxfwd.so"
>
>  SIP MSG OPerationS module
> loadmodule "sipmsgops.so"
>
>  FIFO Management Interface
> loadmodule "mi_fifo.so"
> modparam("mi_fifo", "fifo_name", "/tmp/opensips_fifo")
> modparam("mi_fifo", "fifo_mode", 0666)
>
>
>  URI module
> loadmodule "uri.so"
> modparam("uri", "use_uri_table", 0)
>
>
>
>
>  MYSQL module
> loadmodule "db_mysql.so"
>
>  USeR LOCation module
> loadmodule "usrloc.so"
> modparam("usrloc", "nat_bflag", 10)
> modparam("usrloc", "db_mode",   2)
> modparam("usrloc", "db_url",
> "mysql://opensips:opensipsrw@localhost/opensips") # CUSTOMIZE ME
>
>
>  REGISTRAR module
> loadmodule "registrar.so"
> modparam("registrar", "tcp_persistent_flag", 7)
>
> /* uncomment the next line not to allow more than 10 contacts per AOR */
> #modparam("registrar", "max_contacts", 10)
>
>  ACCounting module
> loadmodule "acc.so"
> /* what special events should be accounted ? */
> modparam("acc", "early_media", 0)
> modparam("acc", "report_cancels", 0)
> /* by default we do not adjust the direct of the sequential requests.
>if you enable this parameter, be sure the enable "append_fromtag"
>in "rr" module */
> modparam("acc", "detect_direction", 0)
> modparam("acc", "failed_transaction_flag", 3)
> /* account triggers (flags) */
> modparam("acc",

Re: [OpenSIPS-Users] Feature request for OpenSIPS 1.9: support for macro definitions

[Ali Pey]

Thank you Bogdan...


On Thu, Nov 8, 2012 at 12:14 PM, Bogdan-Andrei Iancu wrote:

> **
> Hi Ali,
>
> Added on the list http://www.opensips.org/Main/Ver190#toc9 with a quite
> high priority ;).
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
>
> On 11/08/2012 07:06 PM, Ali Pey wrote:
>
> I second this as well. Named flags would make debugging and scripting
> quite simpler.
>
>  Regards,
> Ali
>
>
>
> On Thu, Nov 8, 2012 at 11:23 AM, Bogdan-Andrei Iancu 
> wrote:
>
>> Hi Michael,
>>
>> You can already use names for the route, not only numerical IDs (without
>> the need of defining).
>>
>> For flags, you can use only numbers, indeed - usually I use M4 as text
>> preprocessor to do different complex ops over the script (like defines,
>> ifdefs, etc).
>> But maybe a built in support for names of flags will not be a big issue,
>> especially it is not a big deal.
>>
>> Regards,
>>
>> Bogdan-Andrei Iancu
>> OpenSIPS Founder and Developer
>> http://www.opensips-solutions.com
>>
>>
>>
>> On 11/08/2012 01:22 PM, Michael Renzmann wrote:
>>
>>> Hi all.
>>>
>>> I'm relatively new to OpenSIPS and still am at the very beginning of
>>> learning how to tame this wonderful beast ;-).
>>>
>>> One thing that I, personally, find pretty uncomfortable is that in
>>> various
>>> places in the opensips.cfg one needs to use plain numbers to define or
>>> modify the behaviour. That easily results in constructs like:
>>>
>>> === cut ===
>>> route {
>>>  ...
>>>  route(42);
>>>  ...
>>> }
>>>
>>> route[42] {
>>>  ...
>>>  setflag(8);
>>>  ...
>>> }
>>> === cut ===
>>>
>>> Obviously, it is possible to memorize the meaning of the different
>>> numbers
>>> depending on the context they are used in - this is what people do now
>>> when they work with OpenSIPS. But I guess things may become pretty
>>> painful
>>> when you have a complex configuration and need to urgently debug an issue
>>> in it which just has brought down a heavily loaded, productively used
>>> OpenSIPS instance...
>>>
>>> Using comments certainly helps to relieve the problem a bit, but they can
>>> not solve it. However, things would become much easier if one could use
>>> "speaking names" instead of plain numbers:
>>>
>>> === cut ===
>>> define(ROUTE_NAT_DETECTION, 42);
>>> define(FLAG_NAT_DETECTED, 8);
>>>
>>> route {
>>> ...
>>> route(ROUTE_NAT_DETECTION);
>>> ...
>>> }
>>>
>>> route[ROUTE_NAT_DETECTION] {
>>> ...
>>> setflag(FLAG_NAT_DETECTED);
>>> ...
>>> }
>>> === cut ===
>>>
>>> Yes, I am aware that this could also be achieved by using an external
>>> macro preprocessor such as M4. But that would add another dependency -
>>> one
>>> which could be avoided if OpenSIPS had built-in support for simple macro
>>> definitions like those shown in the example above.
>>>
>>> Bye, Mike
>>>
>>> ___
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>>> Users@lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>>
>> ___
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>>
>
>
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>
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Re: [OpenSIPS-Users] Errors in OpenSIPS

[Ali Pey]

I think it's telling you it can't add another dialog to db. I am almost
certain that's what's happening.

Regards,
Ali Pey



On Thu, Nov 8, 2012 at 4:22 AM, Jorge Ortea  wrote:

> Hi all,
>
> I am getting the following errors:
>
>
> Nov  7 11:47:19 hgt-tero45 /usr/local/opensips/sbin/opensips[12582]:
> ERROR:dialog:update_dialog_dbinfo: could not add another dialog to db
> Nov  7 11:47:19 hgt-tero45 /usr/local/opensips/sbin/opensips[12548]:
> ERROR:dialog:update_dialog_dbinfo: could not add another dialog to db
> Nov  7 11:47:20 hgt-tero45 /usr/local/opensips/sbin/opensips[12558]:
> ERROR:dialog:update_dialog_dbinfo: could not add another dialog to db
> Nov  7 11:57:32 hgt-tero45 /usr/local/opensips/sbin/opensips[12554]:
> ERROR:dialog:update_dialog_dbinfo: could not add another dialog to db
> Nov  7 11:57:32 hgt-tero45 /usr/local/opensips/sbin/opensips[12578]:
> ERROR:dialog:update_dialog_dbinfo: could not add another dialog to db
> Nov  7 11:59:07 hgt-tero45 /usr/local/opensips/sbin/opensips[12556]:
> ERROR:dialog:update_dialog_dbinfo: could not add another dialog to db
> Nov  7 11:59:07 hgt-tero45 /usr/local/opensips/sbin/opensips[12574]:
> ERROR:dialog:update_dialog_dbinfo: could not add another dialog to db
> Nov  7 12:23:23 hgt-tero45 /usr/local/opensips/sbin/opensips[12580]:
> ERROR:dialog:update_dialog_dbinfo: could not add another dialog to db
> Nov  7 12:23:24 hgt-tero45 /usr/local/opensips/sbin/opensips[12544]:
> ERROR:dialog:update_dialog_dbinfo: could not add another dialog to db
> Nov  7 12:23:25 hgt-tero45 /usr/local/opensips/sbin/opensips[12572]:
> ERROR:dialog:update_dialog_dbinfo: could not add another dialog to db
> Nov  7 12:24:47 hgt-tero45 /usr/local/opensips/sbin/opensips[12558]:
> ERROR:dialog:update_dialog_dbinfo: could not add another dialog to db
> Nov  7 12:24:47 hgt-tero45 /usr/local/opensips/sbin/opensips[12562]:
> ERROR:dialog:update_dialog_dbinfo: could not add another dialog to db
> Nov  7 12:26:45 hgt-tero45 /usr/local/opensips/sbin/opensips[12550]:
> ERROR:dialog:update_dialog_dbinfo: could not add another dialog to db
> Nov  7 12:26:45 hgt-tero45 /usr/local/opensips/sbin/opensips[12556]:
> ERROR:dialog:update_dialog_dbinfo: could not add another dialog to db
> Nov  7 12:26:46 hgt-tero45 /usr/local/opensips/sbin/opensips[12570]:
> ERROR:dialog:update_dialog_dbinfo: could not add another dialog to db
> Nov  7 12:28:20 hgt-tero45 /usr/local/opensips/sbin/opensips[12558]:
> ERROR:dialog:update_dialog_dbinfo: could not add another dialog to db
> Nov  7 12:28:20 hgt-tero45 /usr/local/opensips/sbin/opensips[12564]:
> ERROR:dialog:update_dialog_dbinfo: could not add another dialog to db
> Nov  7 12:28:21 hgt-tero45 /usr/local/opensips/sbin/opensips[12568]:
> ERROR:dialog:update_dialog_dbinfo: could not add another dialog to db
> Nov  7 12:30:23 hgt-tero45 /usr/local/opensips/sbin/opensips[12550]:
> ERROR:dialog:update_dialog_dbinfo: could not add another dialog to db
> Nov  7 12:30:23 hgt-tero45 /usr/local/opensips/sbin/opensips[12572]:
> ERROR:dialog:update_dialog_dbinfo: could not add another dialog to db
> Nov  7 12:30:25 hgt-tero45 /usr/local/opensips/sbin/opensips[12576]:
> ERROR:dialog:update_dialog_dbinfo: could not add another dialog to db
> Nov  7 13:12:13 hgt-tero45 /usr/local/opensips/sbin/opensips[12548]:
> ERROR:dialog:update_dialog_dbinfo: could not add another dialog to db
> Nov  7 13:12:13 hgt-tero45 /usr/local/opensips/sbin/opensips[12546]:
> ERROR:dialog:update_dialog_dbinfo: could not add another dialog to db
> Nov  7 13:12:13 hgt-tero45 /usr/local/opensips/sbin/opensips[12576]:
> ERROR:dialog:update_dialog_dbinfo: could not add another dialog to db
> Nov  7 13:23:34 hgt-tero45 /usr/local/opensips/sbin/opensips[12582]:
> ERROR:dialog:update_dialog_dbinfo: could not add another dialog to db
> Nov  7 13:23:34 hgt-tero45 /usr/local/opensips/sbin/opensips[12566]:
> ERROR:dialog:update_dialog_dbinfo: could not add another dialog to db
> Nov  7 13:23:34 hgt-tero45 /usr/local/opensips/sbin/opensips[12580]:
> ERROR:dialog:update_dialog_dbinfo: could not add another dialog to db
> Nov  7 13:33:09 hgt-tero45 /usr/local/opensips/sbin/opensips[12572]:
> ERROR:dialog:update_dialog_dbinfo: could not add another dialog to db
> Nov  7 13:33:09 hgt-tero45 /usr/local/opensips/sbin/opensips[12578]:
> ERROR:dialog:update_dialog_dbinfo: could not add another dialog to db
> Nov  7 13:33:09 hgt-tero45 /usr/local/opensips/sbin/opensips[12574]:
> ERROR:dialog:update_dialog_dbinfo: could not add another dialog to db
> Nov  7 14:10:19 hgt-tero45 /usr/local/opensips/sbin/opensips[12556]:
> ERROR:dialog:update_dialog_dbinfo: could not add another dialog to db
> Nov  7 14:10:19 hgt-tero45 /usr/local/opensips/sbin/opensips[12564]:
> ERROR:dialog:update_dialo

Re: [OpenSIPS-Users] Feature request for OpenSIPS 1.9: support for macro definitions

[Ali Pey]

I second this as well. Named flags would make debugging and scripting quite
simpler.

Regards,
Ali



On Thu, Nov 8, 2012 at 11:23 AM, Bogdan-Andrei Iancu wrote:

> Hi Michael,
>
> You can already use names for the route, not only numerical IDs (without
> the need of defining).
>
> For flags, you can use only numbers, indeed - usually I use M4 as text
> preprocessor to do different complex ops over the script (like defines,
> ifdefs, etc).
> But maybe a built in support for names of flags will not be a big issue,
> especially it is not a big deal.
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developer
> http://www.opensips-solutions.**com 
>
>
>
> On 11/08/2012 01:22 PM, Michael Renzmann wrote:
>
>> Hi all.
>>
>> I'm relatively new to OpenSIPS and still am at the very beginning of
>> learning how to tame this wonderful beast ;-).
>>
>> One thing that I, personally, find pretty uncomfortable is that in various
>> places in the opensips.cfg one needs to use plain numbers to define or
>> modify the behaviour. That easily results in constructs like:
>>
>> === cut ===
>> route {
>>  ...
>>  route(42);
>>  ...
>> }
>>
>> route[42] {
>>  ...
>>  setflag(8);
>>  ...
>> }
>> === cut ===
>>
>> Obviously, it is possible to memorize the meaning of the different numbers
>> depending on the context they are used in - this is what people do now
>> when they work with OpenSIPS. But I guess things may become pretty painful
>> when you have a complex configuration and need to urgently debug an issue
>> in it which just has brought down a heavily loaded, productively used
>> OpenSIPS instance...
>>
>> Using comments certainly helps to relieve the problem a bit, but they can
>> not solve it. However, things would become much easier if one could use
>> "speaking names" instead of plain numbers:
>>
>> === cut ===
>> define(ROUTE_NAT_DETECTION, 42);
>> define(FLAG_NAT_DETECTED, 8);
>>
>> route {
>> ...
>> route(ROUTE_NAT_DETECTION);
>> ...
>> }
>>
>> route[ROUTE_NAT_DETECTION] {
>> ...
>> setflag(FLAG_NAT_DETECTED);
>> ...
>> }
>> === cut ===
>>
>> Yes, I am aware that this could also be achieved by using an external
>> macro preprocessor such as M4. But that would add another dependency - one
>> which could be avoided if OpenSIPS had built-in support for simple macro
>> definitions like those shown in the example above.
>>
>> Bye, Mike
>>
>> __**_
>> 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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Re: [OpenSIPS-Users] [OpenSIPS-Devel] [RELEASES] Planing OpenSIPS 1.9.0 major release

[Ali Pey]

Hello All,

Not sure if this is the right place to suggest changes/new features. I have
a request for a new function for the load_balancer module.

Load_balancer knows how many resources are available for each gateway and
keeps track of active calls for each resource to distribute the calls
accordingly. This works well if the gateways's calls always go through the
load_balancer module. However, this is not always the case.

Let's say I have a number of gateways for termination and I use the
load_balancer module to distribute the calls. If there happen to be some
outgoing calls from these gateways, load balancer would not know about them
and the number of active calls and available resources would be inaccurate.

This can be fixed if there was a function to add a call to a gateway's
resource (maybe based on the IP address) without calling the load_balance()
function.

This functionality can also be used when you are using resources of one of
your gateways but outside the load balancer module such as routing a test
number to a particular gateway and so on.

Regards,
Ali Pey
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Re: [OpenSIPS-Users] OPTIONs flood with large numbers of registered users

[Ali Pey]

Another way to tackle this is to enable pinging from your sip clients.
Makes things quite simpler.

Regards,
Ali Pey


On Sat, Nov 3, 2012 at 3:47 AM, Bogdan-Andrei Iancu wrote:

> Hi Brett,
>
> If you are using nathelper module, use multipe timer processes for pinging
> (it is a module parameter). This will partition and distribute the pings
> between multiple procs and different moments.
>
> Regards,
> Bogdan
>
>
> Sent from Samsung Mobile
>
> Brett Nemeroff  wrote:
> Hello List!
> I having a problem with a registrar server that has a large number of
> registered users. Actually, it's not really even that large, just a few
> thousand.
>
> When the server sends out OPTIONs pings, it is creating a huge spike in
> bandwidth that is making the edge router very unhappy. Network interface is
> gigabit ethernet and it's essentially sending it out as fast as it can.
> Granted it's not a whole lot of data, but it's a very brief spike that the
> router doesn't like.
>
> Is there anyway to spread out the requests a little bit? Has anyone else
> seen anything like this?
>
> Thanks,
> Brett
>
>
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>
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Re: [OpenSIPS-Users] [OpenSIPS-Devel] [RELEASES] Planing OpenSIPS 1.9.0 major release

[Ali Pey]

Which one sounds simpler? Having a new layer of proxies and extra hardware
on different software packages with their own set of configurations,
limitations and bugs than having WebSocket enabled on opensips and control
your routing logic all in one place off of same DB.

Regards,
Ali Pey


On Wed, Oct 31, 2012 at 12:48 PM, Iñaki Baz Castillo  wrote:

> 2012/10/31 Ali Pey 
>
>> Using OverSIP and the OpenSIPS would make things just more complex
>> specially for larger deployments.
>
>
> Really? IMHO it makes things MUCH MORE simple. Use OverSIP as an Outbound
> EDGE Proxy and you are done. Otherwise wait for a proper TCP/TLS and
> WebSocket support in OpenSIPS.
>
> In a large deployment run N OverSIP proxies in front of your OpenSIPS
> registrar. Just it.
>
> --
> Iñaki Baz Castillo
> 
>
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>
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Re: [OpenSIPS-Users] [OpenSIPS-Devel] [RELEASES] Planing OpenSIPS 1.9.0 major release

[Ali Pey]

Hi Bogdan,

Saul is correct. The key thing is to support WebSocket transport. WebRTC is
becoming quite popular and seems to be the thing of future. It is already
added in asterisk (version 11 released last week) and it is being added to
Kamalio. Using OverSIP and the OpenSIPS would make things just more complex
specially for larger deployments.

Also, deployments that are already using opensips would want to provide
WebRTC based apps and it makes a lot of sense to natively support it with
opensips. Asterisk's implementation seemed quite complete. Maybe you can
have a look at that.

Regards,
Ali Pey

On Wed, Oct 31, 2012 at 8:21 AM, Saúl Ibarra Corretgé
wrote:

>
> On Oct 31, 2012, at 12:52 PM, Bogdan-Andrei Iancu wrote:
>
> > Hi Saul,
> >
> > OK, aside the TCP part (which anyhow is scheduled for fixing) and some
> extra parsing, does supporting WebRTC imply something more on the OpenSIPS
> side ?
> >
>
> It requires that OpenSIPS is able to use SIP over a WebSocket transport.
> So OpenSIPS would need support for WebSocket. The transport behaves roughly
> the same as TCP, with the difference that you may only get a single SIP
> packet in each WebSocket segment. There are other couple of minor things,
> but the core of it is supporting the WebSocket transport.
>
> I didn't have the time to test OpenSIPS behind a OverSIP instance, to
> verify if there is anything to be fixed beforehand, but according to
> https://sourceforge.net/tracker/?func=detail&aid=3545859&group_id=232389&atid=1086412Via
>  parsing will fail because if doesn't recognize ws and wss as valid
> transports. I had a look at the attached patch and it seems to solve the
> problem by accepting any Via transport parameter. I guess that's a good
> idea, so OpenSIPS would not care about any transport used in the path, as
> long as it doesn't need to use it, but I guess some checks would need to be
> done to validate if the topmost Via has a transport OpenSIPS understands.
> Not sure if this check is already done though.
>
>
> Regards,
>
> --
> Saúl Ibarra Corretgé
> AG Projects
>
>
>
>
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Re: [OpenSIPS-Users] sip proxy and nat traversal

[Ali Pey]

Hello,

No you don't need two servers. One server is more than enough.

There are a few ways you can achieve this. The easiest and most efficient
way is to have RTP to go directly to asterisk instead of using RTP proxy.
This would require proper port forwarding on your firewall.

You need proper configuration on both asterisk and opensips to negotiate
with proper public IP addresses for both media and sip to your far end
devices. You need to know NAT and SIP to really be able to set this up.

Regards,
Ali Pey


On Sat, Oct 27, 2012 at 6:53 PM, spencer  wrote:

> Hi,
> I am trying to setup a sip proxy and I also want to do NAT traversal.  So
> I setup the residential config file using osipsconfig script.
> I started to try and modify the routing logic in the config file and also
> setup rtpproxy.  I have not had too much luck with this yet.
> Here are my question.
> - Should I really be setting up 2 servers?  One sip proxy and one nat
> traversal server with rtp proxy?
> - Could I use asterisk as the rtp proxy? And if so would i need opensips
> for nat traversal?
>
> Also, any general advice or guidance would be welcome.
>
> Thanks!
>
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Re: [OpenSIPS-Users] [OpenSIPS-Devel] [RELEASES] Planing OpenSIPS 1.9.0 major release

[Ali Pey]

I do also see a lot of value in sip over websocket. WebRTC is pretty much
here and it makes much more sense to be able to support it on one proxy
server rather than having to use OverSIP and then OpenSIPS. WebRTC seems to
be very popular and the thing of tomorrow and it will be very important for
a sip proxy server to support it.

Regards,
Ali Pey

On Fri, Oct 26, 2012 at 11:54 AM, Saúl Ibarra Corretgé  wrote:

>
> On Oct 26, 2012, at 5:46 PM, Duane Larson wrote:
>
> > Is there any roadmap for "SIP over Websocket"?  I know there is now
> OverSIP but wasn't sure if OpenSIPS had any plans to implement a module.
>  Just asking since WebRTC is still evolving.
> >
>
> I personally don't see the need to do it in the 1.x series. You can use
> OverSIP to do protocol translation to TCP/TLS and send the call to
> OpenSIPS. Now, IIRC there was some problem in parsing the Via headers,
> because they now have a different transport parameter (ws and wss), but I
> think someone posted a patch already.
>
> --
> Saúl Ibarra Corretgé
> AG Projects
>
>
>
>
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Re: [OpenSIPS-Users] ACK looping issue with end-point co-located with OpenSIPS

[Ali Pey]

What do you have for alias in your opensips.cfg? Opensips seems to think
192.168.117.4:5070 is itself.

Regards,
Ali Pey


On Mon, Oct 15, 2012 at 6:33 PM, Daniel Eiland wrote:

> Hi folks,
>
> I've got an issue with ACK messages being looped when they are sent to an
> endpoint that is co-located with my OpenSIPS proxy.
>
> I've got OpenSIPS located on server A @ port 5060 and two conference
> endpoints: C01 on server A @ port 5070 and C02 on a separate server.
>
> When my client calls into C01, the ACK message is continually routed by
> OpenSIPS back into itself instead of the conference endpoint listening on
> port 5070.  This doesn't happened when I call into C02, OpenSIPS properly
> routes it to the right destination.
>
> When I compare the two ACKs, they are fairly similar and when I looked at
> some OpenSIPS logs both messages are being routed in a similar fashion
> namely loose_route is true.  The only difference (which I'm sure if the
> problem, I'm just not sure why) is that the ACK to C01 has Destination User
> ($du) of 192.168.117.4 while the ACK to C02 has a $du of NULL.
>
> If anyone has any suggestions, I'd be grateful.
>
> Thanks,
> Daniel
>
> Also here are the two ACKs for comparison:
>
> ACK sip:C01@192.168.117.4:5070;transport=udp SIP/2.0
> Via: SIP/2.0/UDP
> QWE.RTY.XYZ.ABC:2453;rport;branch=z9hG4bKPjca04290517874935af64a839e6bf9701
> Max-Forwards: 70
> From: "deiland"  >;tag=d02a111d4a064b05a7cf987b006bd001
> To: ;tag=ma94e0688avFj
> Call-ID: 5c35c0a5dcb442a9afd324c988bd0a3c
> CSeq: 24503 ACK
> Route: 
> User-Agent: Blink 0.2.7 (Windows)
> Content-Length:  0
>
>
> ACK sip:C02@192.168.155.211:5070;transport=udp SIP/2.0
> Via: SIP/2.0/UDP
> QWE.RTY.XYZ.ABC:2122;rport;branch=z9hG4bKPj5318936898454066944aadf64ad846d0
> Max-Forwards: 70
> From: "deiland"  >;tag=ff6fb0856e264e7f96601529148bb206
> To: ;tag=54aZ48KZ7a9Xj
> Call-ID: afddb9bea6714f95bd84dff2159e2b14
> CSeq: 11988 ACK
> Route: 
> User-Agent: Blink 0.2.7 (Windows)
> Content-Length:  0
>
>
>
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>
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Re: [OpenSIPS-Users] multiple headers

[Ali Pey]

Did you try sipmsgops module? You can maybe remove the header and the add
the ones you'd like:

http://www.opensips.org/html/docs/modules/1.8.x/sipmsgops.html

You can also check out the textops module. There are some utilities there
as well.

Regards,
Ali Pey


On Thu, Oct 11, 2012 at 1:26 PM, DM  wrote:

> One of my clients is sending a buggy SIP invite which contains Privacy
> headers multiple times.
>
>
> Supported: replaces, timer
> Privacy: id
> Privacy: id
> Privacy: id
> Content-Type: application/sdp
> Content-Length: 298
>
>
> How can I strip out the last 2 Privacy headers and keep only the first
> before sending it to the destination via usrloc or to pstn?
>
> Thanks,
>
> Dmitri
>
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Re: [OpenSIPS-Users] duplicate information in sip_trace table

[Ali Pey]

Did you try to add logs to make sure sip_trace() is not called twice?

Regards,
Ali Pey

On Wed, Oct 10, 2012 at 10:29 AM, Dragomir Haralambiev
wrote:

> Hi,
>
> Thanks for your replay.
> The problem is not in IF operator.
>
> When use sip_trace() Opnesips make two records in sip_trace.
>
> Best regards,
> PlayMen
> 2012/10/10
>
>> Hi,
>>  Do it like this:
>> if (is_avp_set("$avp(trace_id)"){
>> # Do tracing
>> }
>> or like this :
>> If (!avp_check("$avp(trace_id)","re/^$/")){
>>
>> # Do tracing
>> }
>> where "/^$/" regex pattern for empty string.
>>
>> // Binan
>>
>>   --
>> *Från:* Dragomir Haralambiev 
>> *Till:* OpenSIPS users mailling list 
>> *Skickat:* tisdag, 9 oktober 2012 22:55
>> *Ämne:* [OpenSIPS-Users] duplicate information in sip_trace table
>>
>> Hello ,
>>
>> I see duplicate information in sip_trace table (Opensips: 1.8.1):
>> one line with traced_user=empty;
>> other traced_user=$avp(trace_id).
>>
>> How to setup Opensips to remove traced_user=empty ?
>> Here is part of opensips.cfg:
>>
>> modparam("siptrace", "db_url", "mysql://user:password@localhost
>> /opensips")
>> modparam("siptrace", "trace_on", 1)
>> modparam("siptrace", "enable_ack_trace", 1)
>> modparam("siptrace", "traced_user_avp", "$avp(trace_id)")
>> ..
>> if (is_method("INVITE")) {
>> . receive $avp(trace_id) from radius_send_auth ...
>>if ( $avp(trace_id) ) {
>> xlog("L_ERR", "Trace ON. Trace_id=$avp(trace_id)\n");
>> sip_trace();
>> }
>> .
>> }
>>
>> Best regards,
>> PlayMen
>>
>>
>> ___
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>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>>
>>
>>
>> ___
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>>
>>
>
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Re: [OpenSIPS-Users] Proxying - Hiding Caller From Callee

[Ali Pey]

Also check out these to places. You can use the uac module or directly
change the from uri username:

http://www.opensips.org/html/docs/modules/devel/uac.html
http://www.opensips.org/Resources/DocsCoreVar18#toc45

Regards,
Ali Pey

On Tue, Oct 9, 2012 at 6:56 PM, Ali Pey  wrote:

> Hi David,
>
> As I take it you need to manipulate your from header. There are a few easy
> ways you can achieve this. Check out the textops module:
>
> http://www.opensips.org/html/docs/modules/1.8.x/textops.html
>
> Regards,
> Ali Pey
>
>
> On Tue, Oct 9, 2012 at 1:24 PM, David Wylie wrote:
>
>> Not sure if this will reply properly?!
>>
>> Apologies for that question, it is one of the most vague and unhelpful
>> I've ever committed to paper.
>> I'll try and formulate a better one with examples on what I've done.
>>
>> In the mean time, feel free to ignore it (or if you can shed light with
>> such little info then great).
>>
>>
>>
>> ___
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>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>
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Re: [OpenSIPS-Users] Proxying - Hiding Caller From Callee

[Ali Pey]

Hi David,

As I take it you need to manipulate your from header. There are a few easy
ways you can achieve this. Check out the textops module:

http://www.opensips.org/html/docs/modules/1.8.x/textops.html

Regards,
Ali Pey


On Tue, Oct 9, 2012 at 1:24 PM, David Wylie wrote:

> Not sure if this will reply properly?!
>
> Apologies for that question, it is one of the most vague and unhelpful
> I've ever committed to paper.
> I'll try and formulate a better one with examples on what I've done.
>
> In the mean time, feel free to ignore it (or if you can shed light with
> such little info then great).
>
>
>
> ___
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> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
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Re: [OpenSIPS-Users] bash shell variable not kept when run in opensips.cfg

[Ali Pey]

Hello Xavier,

Try the avops module. This is your best option in this scenario:

http://www.opensips.org/html/docs/modules/1.8.x/avpops.html#id292750

Regards,
Ali Pey

On Sat, Oct 6, 2012 at 7:38 PM, Xavier Herlindo wrote:

> Hello all,
>
> this is my first post, so please bare with me if i commit any protocol
> error. All comments are welcome. Thanks all in advance for any help i can
> get.
>
> I am running Opensips 8.1 with a Centos 5.8 install. I am setting up a
> redirect server. Only using the slo.so, mi_fifo.so, rr.so (not sure if i
> really need this one) and exec.so modules.
>
> I am trying to run a simple bash script that checks if the number in the
> URI exists in a mysq table and if so, the script rewrites the prefix and ip
> address to redirect. If not, there is a default address to redirect to.
>
> The sh script works fine as i have tested it directly from the shell,
> running without issues. If i input an URI with an existing number in my
> mysql table, the script connects to the db, validates it, an the output is
> the modified URI as expected. Now if the script is called from within
> opensips.cfg file, the script is not validating in mysql. The opensips log
> file show no errors when running the sh file. If i try to set an incorrect
> connection param string there is no change either, as if the line is
> ignored.
>
> This is my unrecognized variable:
>
> exec_query=`mysql -u USER --password=TEST -s -D MY_DB -e "SELECT numB FROM
> valid_numb WHERE numB = '123';"`
>
>  If i echo the value for exec_query it is empty when running within
> opensips.cfg
> Running in the bash prompt the value of exec_query is set correctly.
>
> This same shell script was tested in an old SER box without not issues, so
> i think it is something out of the script directly related to opensips
> configuration.
>
> Do you know what may be the problem?
>
> Regards,
>
> Xavier
>
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Re: [OpenSIPS-Users] High Volume MediaProxy

[Ali Pey]

Also Media proxy can not run on a virtual environment:
http://mediaproxy.ag-projects.com/

I never done much testing myself but I have talked to people that achieved
close to 10,000 simultaneous calls. It is also depends on the call activity
as well. It's one thing to have 10,000 simultaneous calls and it's another
to have calls setup and tear down in a normal telephony environment.


On Mon, Oct 1, 2012 at 3:48 PM, Jeff Pyle  wrote:

> Muhammad,
>
> I don't have the requirements that Afshin does, but I've often wondered
> which server resource (ignoring bandwidth) would run out first.  Around 650
> calls gives a load average of about 0.15.  This particular relay has one
> processor with 8 cores at 2 GHz, Mediaproxy 2.4.4, no virtualization.  RAM
> utilization is negligible.
>
> If CPU load would become the bottleneck, can I truly run 650 / 0.15 * 8
> cores = 35,000 calls on one relay?  Continuing to ignore bandwidth that
> still doesn't seem right.
>
> Once a relay fails due to over-utilization, which actual resource tends to
> fill first?
>
>
> - Jeff
>
>
> On Mon, Oct 1, 2012 at 3:11 PM, Muhammad Shahzad <
> shaherya...@googlemail.com> wrote:
>
>> OR, if you can arrange an ESX with really good specs (2 x quad core 3.2
>> GHz processor with at least 32 GB RAM) then you can create about 8 virtual
>> servers running media proxy, the capacity of each media proxy in this case
>> will be reduced to around 750~900 concurrent calls but total output of this
>> single physical machine would be 750 x 8 = 6000 calls, you will need at
>> least 4 gigabit ethernets of course.
>>
>> Thank you.
>>
>>
>> On Mon, Oct 1, 2012 at 9:04 PM, Muhammad Shahzad <
>> shaherya...@googlemail.com> wrote:
>>
>>> If you really have this many calls, then you should probably consider
>>> distributed media proxies perhaps over different geographical locations. If
>>> you have done everything perfect then one media proxy would approx. do
>>> 1000-1250 concurrent calls over 100 mbps link this may get double if you
>>> have have 1 gbps link but not more then that. So you will need around 6 to
>>> 8 media proxies to handle 10K concurrent calls.
>>>
>>> Thank you.
>>>
>>>
>>> On Mon, Oct 1, 2012 at 6:47 PM, Ali Pey  wrote:
>>>
>>>> Hi Afshin,
>>>>
>>>> I take it that you need to proxy both the sip signalling and media. I
>>>> don't think you can do that on one server. One opensips server can do the
>>>> signalling but you would need additional servers for media proxy.
>>>>
>>>> Consider direct media path so you won't have to use a media proxy.
>>>>
>>>> Regards,
>>>> Ali Pey
>>>>
>>>> On Mon, Oct 1, 2012 at 3:19 AM, afshin afzali 
>>>> wrote:
>>>>
>>>>> Hi Guys,
>>>>>
>>>>> I'm looking for a solution for traversing 10,000 concurrent sessions
>>>>> (no encryption).
>>>>> Is it possible this on a single modern box by MediaProxy ?
>>>>>
>>>>> BEST,
>>>>> -- afshin
>>>>> ___
>>>>> 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
>>>>
>>>>
>>>
>>>
>>> --
>>> Muhammad Shahzad
>>> ---
>>> CISCO Rich Media Communication Specialist (CRMCS)
>>> CISCO Certified Network Associate (CCNA)
>>> Cell: +92 334 422 40 88
>>> MSN: shari_78...@hotmail.com
>>> Email: shaherya...@googlemail.com
>>>
>>
>>
>>
>> --
>> Muhammad Shahzad
>> ---
>> CISCO Rich Media Communication Specialist (CRMCS)
>> CISCO Certified Network Associate (CCNA)
>> Cell: +92 334 422 40 88
>> MSN: shari_78...@hotmail.com
>> Email: shaherya...@googlemail.com
>>
>> ___
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>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>
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Re: [OpenSIPS-Users] High Volume MediaProxy

[Ali Pey]

Hi Afshin,

I take it that you need to proxy both the sip signalling and media. I don't
think you can do that on one server. One opensips server can do the
signalling but you would need additional servers for media proxy.

Consider direct media path so you won't have to use a media proxy.

Regards,
Ali Pey

On Mon, Oct 1, 2012 at 3:19 AM, afshin afzali wrote:

> Hi Guys,
>
> I'm looking for a solution for traversing 10,000 concurrent sessions (no
> encryption).
> Is it possible this on a single modern box by MediaProxy ?
>
> BEST,
> -- afshin
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>
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Re: [OpenSIPS-Users] Max TCP connections

[Ali Pey]

Thanks Muhammad. These are great information. Thank you sharing it with us.

Two follow up questions:
1- What tool did you use to send registers over TCP and keep connections
open?
2- What did you set tcp_max_connections to in opensips?

Regards,
Ali Pey


On Sat, Sep 29, 2012 at 10:59 AM, Muhammad Shahzad <
shaherya...@googlemail.com> wrote:

> Every thing is a file in UNIX / Linux, so every TCP connection is a file
> too. Also persistent connects are sometime memory intensive too. So make
> sure adjust following things carefully.
>
> 1. Number of Open Files Limit, use either ulimit -n  in opensips
> startup script or set open_files_limit in global section on opensips dial
> plan. I usually use second options and set open_files_limit to 1024000.
> 2. Increase TCP children, tcp_children, default is i think 8, increase it
> to 16 or even 32 if you have very large deployment, this will speed up
> connection processing significantly, however consume much more memory. Also
> make sure to set all other TCP related opensips variables to reasonable
> values, such as tcp_connect_timeout, tcp_connection_lifetime etc. etc. Look
> here for details,
>
> http://www.opensips.org/Resources/DocsCoreFcn18#toc75
>
> 3. If you have a dedicated SIP server, running only OpenSIPs then reduce
> stack size as well. I found setting it to 1024KB or even less gives good
> results.
>
> Now some practical results. On Dell server R510 with 8GB RAM and 2.4GHz
> quad core Intel processor I was able to test 104,981 concurrent SIP
> registrations with load average between 1.0 to 1.5.
>
> Two important things to remember.
>
> 1. What is estimated / expected number of requests per second?
> 2. What is SIP re-register time?
>
> For me, i tested up to 350 requests per second with re-register time as
> 300 seconds this gives theoretically possible concurrent SIP connections to
> 350 x 300 = 105,000. Which means you can't have more then this number of
> concurrent connections. So i can safely say per my testing that opensips
> can practically handle up to ~100K SIP concurrent connections.
>
> Thank you.
>
>
> On Fri, Sep 28, 2012 at 7:21 PM, Ali Pey  wrote:
>
>> I am also interested in this. Any responds/updates?
>>
>> Regards,
>> Ali Pey
>>
>>
>> On Wed, Sep 12, 2012 at 9:34 AM, John Quick wrote:
>>
>>> Does anyone know what the practical limit is for the maximum number of
>>> TCP
>>> connections to OpenSIPS?
>>> It is a question that often comes up with clients who are developing
>>> SIP-based apps for smart phones. Persistent TCP seems to be strongly
>>> favoured over UDP for connections with mobile devices such as iPhone,
>>> iPad,
>>> etc. However, I cannot even get OpenSIPS to start if I set
>>> tcp_max_connections to a number greater than about 55000. Have tried it
>>> with
>>> v1.7 and 1.8. Is there a way to overcome this ceiling? Has anyone tested
>>> it
>>> with 10's of thousands of TCP connections?
>>>
>>> John Quick
>>> Smartvox Limited
>>>
>>>
>>>
>>>
>>> ___
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>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>
>>
>> ___
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>>
>>
>
>
> --
> Muhammad Shahzad
> ---
> CISCO Rich Media Communication Specialist (CRMCS)
> CISCO Certified Network Associate (CCNA)
> Cell: +92 334 422 40 88
> MSN: shari_78...@hotmail.com
> Email: shaherya...@googlemail.com
>
>
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Re: [OpenSIPS-Users] Max TCP connections

[Ali Pey]

I am also interested in this. Any responds/updates?

Regards,
Ali Pey

On Wed, Sep 12, 2012 at 9:34 AM, John Quick wrote:

> Does anyone know what the practical limit is for the maximum number of TCP
> connections to OpenSIPS?
> It is a question that often comes up with clients who are developing
> SIP-based apps for smart phones. Persistent TCP seems to be strongly
> favoured over UDP for connections with mobile devices such as iPhone, iPad,
> etc. However, I cannot even get OpenSIPS to start if I set
> tcp_max_connections to a number greater than about 55000. Have tried it
> with
> v1.7 and 1.8. Is there a way to overcome this ceiling? Has anyone tested it
> with 10's of thousands of TCP connections?
>
> John Quick
> Smartvox Limited
>
>
>
>
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Re: [OpenSIPS-Users] Problem with 408 Request Timeout

[Ali Pey]

So you don't want to send 408 request timeout??? What do you expect to
happen here? What is your desired behaviour?

Look at your routing script: /etc/opensips/opensips.cfg
The behaviour is configured there and can be change to your liking.

Regards,
Ali Pey

On Thu, Sep 27, 2012 at 4:16 AM, nguyen khue  wrote:

> Hi all,
>
> I have a problem with 408 request timeout. My model is following:
>
> X-Lite -- OpenSIPS ---
> Asterisk.
> X-Lite registered on OpenSIPS. Now, I make call from X-Lite to Asterisk
> via OpenSIPS, if Asterisk was died or not process INVITE message, OpenSIPS
> will generate 408 Request Timeout and send to X-Lite. I want capture
> message 408 from scenarino, how I can do it?
>
> Brs,
> Khue Nguyen.
>
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Re: [OpenSIPS-Users] authentication of authorized user agents

[Ali Pey]

Hi Dave,

First of all, this is opensips mailing list not Kamailio.

Secondly, you can pass the users sip credentials (sip username and
password) through your http secure connection and pass it to your sip stack
(user agent) to register with that credentials.

Regards,
Ali Pey


On Tue, Sep 18, 2012 at 9:53 PM, David Thomson wrote:

>  Hi,
>
> I am working on a project where a custom sip client (user agent) will be
> integrated into a suite of applications to provide voip.  The sip client
> will be working with Kamailio.  The goal is to ensure that the user agent
> is authorized for communication with kamailio before allowing any calls to
> be made.  Conventional username/password authentication for individual
> users will also be used once the user agent has been authenticated.
>
> Currently other applications in the suite use a digital signature in the
> http headers when communicating with server processes.  If the signature is
> validated by the server process then the applications identity is validated
> and communication with the server process is allowed.
>
> Is it possible to include a public key and digital signature in the
> register events and have kamailio perform the transformation to validate
> the user agent's identity?  If so which module provides such functionality?
>  Has something like this been implemented in the past?  Thanks for any
> input.
>
> ttyl,
> Dave
>
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Re: [OpenSIPS-Users] contol panel installation problems

[Ali Pey]

There shouldn't be a '>' at the end of the path. Re-examine your httpd.conf
again. You might have a '>' somewhere accidentally. Make sure you restart
the service anytime you make a change.

Regards,
Ali Pey

On Thu, Sep 13, 2012 at 7:16 PM, spencer  wrote:

> Thanks for the suggestions.
>
> I have determined that the sql statement in login.php is working fine.
> The messages for undefined variable resultset1 are going to happen when
> you use
> $config->admin_passwd_mode=1;
>
> So the problem is related to this error
> File does not exist: /var/www/html/opensips-cp/web/<
>
> I have restarted the httpd daemon with
> $ service httpd restart
>
> I am not really a php expert.  What is the < character at the end of the
> path?   Is this tag indicate some sort of template?
>
> Is there a way to show more details in the log?
>
> Thanks again.
>
>   --
> *From:* Ali Pey 
> *To:* OpenSIPS users mailling list 
> *Sent:* Thursday, September 13, 2012 11:22 AM
> *Subject:* Re: [OpenSIPS-Users] contol panel installation problems
>
> Also make sure you have restarted the service after changing httpd.conf
> service httpd restart
>
> Regards,
> Ali Pey
>
> On Thu, Sep 13, 2012 at 2:08 AM, Salman Zafar wrote:
>
> It means you are getting error due to this query not being executed in
> code.
>
> $sql = "select * from ocp_admin_privileges where username = ? and password
> = ?";
>
> I hope you have inserted user/password (e.g admin/admin) in the DB for
> authentication. Recheck and debug by echoing the $sql.
>
>
> --
> Regards
>
> Muhammad Salman Zafar
>
> On Thu, Sep 13, 2012 at 6:10 AM, spencer  wrote:
>
> Hi,
> I am trying to setup opensips and opensips-cp and have run into some
> problems with the control panel.
> I have using centos and followed the install guide. I am using mysql
>
> I get this error on the web page after logging in as admin
> Not Found The requested URL /cp/< was not found on this server.
>
> I get these errors in the error_log
>
> PHP Notice:  Undefined variable: resultset1 in
> /var/www/html/opensips-cp/web/
>  login.php on line 80, referer:
> http://mywebserver.com/cp/<http://ec2-184-72-158-150.compute-1.amazonaws.com/cp/>
>
> PHP Notice:  Undefined variable: resultset1 in
> /var/www/html/opensips-cp/web/**login.php on line 86, referer:
> http://mywebserver.com/cp/<http://ec2-184-72-158-150.compute-1.amazonaws.com/cp/>
>
> File does not exist: /var/www/html/opensips-cp/web/**<, referer:
> http://mywebserver.com/cp/**main.php <http://mywebserver.com/cp/main.php>
>
> I have this in httpd.conf
> Alias /cp "/var/www/html/opensips-cp/**web"
>
> Does anyone have an idea what may be causing this problem?
>
> Thanks!
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
>
>
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
> ___
> 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] contol panel installation problems

[Ali Pey]

Also make sure you have restarted the service after changing httpd.conf
service httpd restart

Regards,
Ali Pey

On Thu, Sep 13, 2012 at 2:08 AM, Salman Zafar  wrote:

> It means you are getting error due to this query not being executed in
> code.
>
> $sql = "select * from ocp_admin_privileges where username = ? and password
> = ?";
>
> I hope you have inserted user/password (e.g admin/admin) in the DB for
> authentication. Recheck and debug by echoing the $sql.
>
>
> --
> Regards
>
> Muhammad Salman Zafar
>
> On Thu, Sep 13, 2012 at 6:10 AM, spencer  wrote:
>
>> Hi,
>> I am trying to setup opensips and opensips-cp and have run into some
>> problems with the control panel.
>> I have using centos and followed the install guide. I am using mysql
>>
>> I get this error on the web page after logging in as admin
>> Not Found The requested URL /cp/< was not found on this server.
>>
>> I get these errors in the error_log
>>
>> PHP Notice:  Undefined variable: resultset1 in
>> /var/www/html/opensips-cp/web/
>>  login.php on line 80, referer:
>> http://mywebserver.com/cp/<http://ec2-184-72-158-150.compute-1.amazonaws.com/cp/>
>>
>> PHP Notice:  Undefined variable: resultset1 in
>> /var/www/html/opensips-cp/web/**login.php on line 86, referer:
>> http://mywebserver.com/cp/<http://ec2-184-72-158-150.compute-1.amazonaws.com/cp/>
>>
>> File does not exist: /var/www/html/opensips-cp/web/**<, referer:
>> http://mywebserver.com/cp/**main.php <http://mywebserver.com/cp/main.php>
>>
>> I have this in httpd.conf
>> Alias /cp "/var/www/html/opensips-cp/**web"
>>
>> Does anyone have an idea what may be causing this problem?
>>
>> Thanks!
>>
>> ___
>> Users mailing list
>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>
>
>
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] Can load balancer show total number of call for a gateway

[Ali Pey]

Hi Sammy,

Thank you for your response and it is a very good suggestion. I'm sorry if
you felt I was criticizing it.

I know there is a way to increase the number of active call for the load
balancer module:

set_dlg_profile("lbXchannel","server_id)");

I just don't know how I can get the server_id which is the load_balancer
table index without doing a DB query.

Thanks again,
Ali


On Fri, Sep 7, 2012 at 12:56 PM, SamyGo  wrote:

> Hi,
>
> Didn't understand you clearly.
>
> When you define a size i.e calls=100 per gateway and you send a call from
> opensips to that gateway its load becomes 1, and capacity obviously becomes
> calls=99. Now, When you gateway sends call back and you increment its call
> count variable $avp(IC_GW1) to 1 AND immediately resize the capacity like
> calls = calls - 1; The capacity of your gateway become calls=98.
>
> On hangup of that Incoming call from gateway you resize the capacity back.
>
> Using Dialog module profiling you can do it easily.
>
> And as per my understanding of your question this is what you wanted. I am
> not sure if there is any clean way of doing this w/o going into the
> load_balancer module code and editing it as per your requirement which will
> be something like : Link the load_balancer module with the dialog module.
> Create a modparam like "gw_incoming_call_monitor"= 1 | 0. if its set to 1,
> internally tell dialog module to count the active incoming calls of each
> load-balancer gateway and adjust the gateway's "load". I don't think this
> is much of a big task.
>
> Do spare me sir if you didn't like it.
>
> Thanks,
> Sammy
>
>
> On Fri, Sep 7, 2012 at 9:37 PM, Ali Pey  wrote:
>
>> Hi Sammy,
>>
>> I rather to be able to see the through number of active calls on a server
>> than re-sizing the capacity of each server twice for each call. That will
>> create a mess.
>>
>> Regards,
>> Ali Pey
>>
>>
>> On Fri, Sep 7, 2012 at 12:02 PM, SamyGo  wrote:
>>
>>> Hi Ali,
>>>
>>> No , I said it dialog based profiling of each gateway's incoming calls
>>> to opensips. Store the call counter for each gateway in any variable, I
>>> named it $avp(IC_GW1).
>>> Next is use this:
>>> http://www.opensips.org/html/docs/modules/1.8.x/load_balancer.html#id250348
>>> Resize a particular gateway's capacity using simple maths.
>>>
>>>
>>> Thanks,
>>> Sammy
>>>
>>>
>>> On Fri, Sep 7, 2012 at 8:30 PM, Muhammad Shahzad <
>>> shaherya...@googlemail.com> wrote:
>>>
>>>> Can you give more info about your topology, I think everybody is
>>>> considering following topology,
>>>>
>>>> caller -> opensips -> gateway -> opensips -> destination
>>>>
>>>> Which means call enter opensips twice, once for leg A and once for leg
>>>> B. Is that what you have? If yes, then do you want to count leg A and leg B
>>>> for each call?
>>>>
>>>> Thank you.
>>>>
>>>>
>>>> On Fri, Sep 7, 2012 at 5:20 PM, Ali Pey  wrote:
>>>>
>>>>> Hi Binan,
>>>>>
>>>>> Sometimes I wish you would read the question before trying to answer.
>>>>> If you don't have an answer, you don't need to come up with one. No body 
>>>>> is
>>>>> forcing you.
>>>>>
>>>>> 1- First of all, how is this related to load balancer modules? As you
>>>>> can see the title of email is talking about the load balancer module.
>>>>> 2- All my GW IP addresses are in the database, I don't want to list
>>>>> them in my config file and evey time these is s change I would have to
>>>>> change the config file and restart the service.
>>>>> 3- I need a solution that would integrate with load balacer module
>>>>> and/or the dialog module so I can use the MI commands and it would be
>>>>> automatic. What you are saying here doesn't have a real world practice.
>>>>>
>>>>> Thanks,
>>>>> Ali
>>>>>
>>>>>
>>>>> On Fri, Sep 7, 2012 at 5:09 AM, Binan AL Halabi <
>>>>> binanalhal...@yahoo.com> wrote:
>>>>>
>>>>>> Hi,*
>>>>>>
>>>>>> $si* - reference to IP source address of the message
>>>>>>
>>>>>>  if ($si==GW1-IP-ADDRESS) {
>>>>&

Re: [OpenSIPS-Users] Can load balancer show total number of call for a gateway

[Ali Pey]

Hi Sammy,

I rather to be able to see the through number of active calls on a server
than re-sizing the capacity of each server twice for each call. That will
create a mess.

Regards,
Ali Pey


On Fri, Sep 7, 2012 at 12:02 PM, SamyGo  wrote:

> Hi Ali,
>
> No , I said it dialog based profiling of each gateway's incoming calls to
> opensips. Store the call counter for each gateway in any variable, I named
> it $avp(IC_GW1).
> Next is use this:
> http://www.opensips.org/html/docs/modules/1.8.x/load_balancer.html#id250348
> Resize a particular gateway's capacity using simple maths.
>
>
> Thanks,
> Sammy
>
>
> On Fri, Sep 7, 2012 at 8:30 PM, Muhammad Shahzad <
> shaherya...@googlemail.com> wrote:
>
>> Can you give more info about your topology, I think everybody is
>> considering following topology,
>>
>> caller -> opensips -> gateway -> opensips -> destination
>>
>> Which means call enter opensips twice, once for leg A and once for leg B.
>> Is that what you have? If yes, then do you want to count leg A and leg B
>> for each call?
>>
>> Thank you.
>>
>>
>> On Fri, Sep 7, 2012 at 5:20 PM, Ali Pey  wrote:
>>
>>> Hi Binan,
>>>
>>> Sometimes I wish you would read the question before trying to answer. If
>>> you don't have an answer, you don't need to come up with one. No body is
>>> forcing you.
>>>
>>> 1- First of all, how is this related to load balancer modules? As you
>>> can see the title of email is talking about the load balancer module.
>>> 2- All my GW IP addresses are in the database, I don't want to list them
>>> in my config file and evey time these is s change I would have to change
>>> the config file and restart the service.
>>> 3- I need a solution that would integrate with load balacer module
>>> and/or the dialog module so I can use the MI commands and it would be
>>> automatic. What you are saying here doesn't have a real world practice.
>>>
>>> Thanks,
>>> Ali
>>>
>>>
>>> On Fri, Sep 7, 2012 at 5:09 AM, Binan AL Halabi >> > wrote:
>>>
>>>> Hi,*
>>>>
>>>> $si* - reference to IP source address of the message
>>>>
>>>>  if ($si==GW1-IP-ADDRESS) {
>>>>  # Incoming call
>>>>  # Increment $avp(IC_GW1)
>>>>
>>>>  }
>>>>
>>>> //Binan
>>>>
>>>> --- On *Thu, 9/6/12, SamyGo * wrote:
>>>>
>>>>
>>>> From: SamyGo 
>>>> Subject: Re: [OpenSIPS-Users] Can load balancer show total number of
>>>> call for a gateway
>>>> To: "OpenSIPS users mailling list" 
>>>> Date: Thursday, September 6, 2012, 10:25 PM
>>>>
>>>>
>>>> Hi Ali,
>>>>
>>>> AFAIK you can't do this from Load-balancer, but the possible solution
>>>> is use variable $avp(IC_GW1) for each gateway to count the incoming calls
>>>> from particular gateways (Dialogue module based profiling) and then see how
>>>> many calls a gateway has Incoming/Outgoing from/to a particular gateway.
>>>>
>>>> The real deal is to communicate the populated incoming calls variable
>>>> to the load-balancer module. (Dynamically increment/decrement a g/w
>>>> capacity based on the $avp(IC_GW1) )
>>>>
>>>> BR
>>>> Sammy
>>>>
>>>>
>>>> On Fri, Sep 7, 2012 at 1:58 AM, Ali Pey 
>>>> http://mc/compose?to=ali...@gmail.com>
>>>> > wrote:
>>>>
>>>> Hello,
>>>>
>>>> Let's say I have a sip trunk with 100 channels. The calls come in and
>>>> are distributed to 4 gateways through the load_balancer module. Load
>>>> balancer module can show all active call going to the gateways but not the
>>>> call originated from the gateways to a sip trunk.
>>>>
>>>> If there are outgoing calls from the gateways to the sip trunks, those
>>>> will not be accounted for on the load balancer module. Basically load
>>>> balancer can keep account for number of active calls on one direction.
>>>>
>>>> Is there a way that I can see the total number of active calls on a
>>>> gateway (both incoming and outgoing) through the load balancer module?
>>>>
>>>> Regards,
>>>> Ali Pey
>>>>
>>>>
>>>

Re: [OpenSIPS-Users] Can load balancer show total number of call for a gateway

[Ali Pey]

Yes, that's almost the topology. Thank you for clarifying it.

Yes, I would want to be able to account for both legs. It's not always two
legs though, sometimes it can only terminate on the gateway, e.g. voice
mail, etc.

Regards,
Ali Pey


On Fri, Sep 7, 2012 at 11:30 AM, Muhammad Shahzad <
shaherya...@googlemail.com> wrote:

> Can you give more info about your topology, I think everybody is
> considering following topology,
>
> caller -> opensips -> gateway -> opensips -> destination
>
> Which means call enter opensips twice, once for leg A and once for leg B.
> Is that what you have? If yes, then do you want to count leg A and leg B
> for each call?
>
> Thank you.
>
>
> On Fri, Sep 7, 2012 at 5:20 PM, Ali Pey  wrote:
>
>> Hi Binan,
>>
>> Sometimes I wish you would read the question before trying to answer. If
>> you don't have an answer, you don't need to come up with one. No body is
>> forcing you.
>>
>> 1- First of all, how is this related to load balancer modules? As you can
>> see the title of email is talking about the load balancer module.
>> 2- All my GW IP addresses are in the database, I don't want to list them
>> in my config file and evey time these is s change I would have to change
>> the config file and restart the service.
>> 3- I need a solution that would integrate with load balacer module and/or
>> the dialog module so I can use the MI commands and it would be automatic.
>> What you are saying here doesn't have a real world practice.
>>
>> Thanks,
>> Ali
>>
>>
>> On Fri, Sep 7, 2012 at 5:09 AM, Binan AL Halabi 
>> wrote:
>>
>>> Hi,*
>>>
>>> $si* - reference to IP source address of the message
>>>
>>>  if ($si==GW1-IP-ADDRESS) {
>>>  # Incoming call
>>>  # Increment $avp(IC_GW1)
>>>
>>>  }
>>>
>>> //Binan
>>>
>>> --- On *Thu, 9/6/12, SamyGo * wrote:
>>>
>>>
>>> From: SamyGo 
>>> Subject: Re: [OpenSIPS-Users] Can load balancer show total number of
>>> call for a gateway
>>> To: "OpenSIPS users mailling list" 
>>> Date: Thursday, September 6, 2012, 10:25 PM
>>>
>>>
>>> Hi Ali,
>>>
>>> AFAIK you can't do this from Load-balancer, but the possible solution is
>>> use variable $avp(IC_GW1) for each gateway to count the incoming calls from
>>> particular gateways (Dialogue module based profiling) and then see how many
>>> calls a gateway has Incoming/Outgoing from/to a particular gateway.
>>>
>>> The real deal is to communicate the populated incoming calls variable to
>>> the load-balancer module. (Dynamically increment/decrement a g/w capacity
>>> based on the $avp(IC_GW1) )
>>>
>>> BR
>>> Sammy
>>>
>>>
>>> On Fri, Sep 7, 2012 at 1:58 AM, Ali Pey 
>>> http://mc/compose?to=ali...@gmail.com>
>>> > wrote:
>>>
>>> Hello,
>>>
>>> Let's say I have a sip trunk with 100 channels. The calls come in and
>>> are distributed to 4 gateways through the load_balancer module. Load
>>> balancer module can show all active call going to the gateways but not the
>>> call originated from the gateways to a sip trunk.
>>>
>>> If there are outgoing calls from the gateways to the sip trunks, those
>>> will not be accounted for on the load balancer module. Basically load
>>> balancer can keep account for number of active calls on one direction.
>>>
>>> Is there a way that I can see the total number of active calls on a
>>> gateway (both incoming and outgoing) through the load balancer module?
>>>
>>> Regards,
>>> Ali Pey
>>>
>>>
>>> ___
>>> Users mailing list
>>> Users@lists.opensips.org <http://mc/compose?to=Users@lists.opensips.org>
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>>
>>>
>>> -Inline Attachment Follows-
>>>
>>>
>>> ___
>>> Users mailing list
>>> Users@lists.opensips.org <http://mc/compose?to=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
>>
>>
>
>
> --
> Muhammad Shahzad
> ---
> CISCO Rich Media Communication Specialist (CRMCS)
> CISCO Certified Network Associate (CCNA)
> Cell: +92 334 422 40 88
> MSN: shari_78...@hotmail.com
> Email: shaherya...@googlemail.com
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
___
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