[OpenSIPS-Users] MOH / rtpengine

2021-08-08 Thread Miha via Users

Hi

when call is being trasfered to another number MS Teams sends new Invite 
with SDP as 'a=inactive'. How can I put ringback ton as MOH for this 
sitation?


I tried with:

if(is_audio_on_hold()) {
 xlog("L_INFO", "onHOLD");

rtpengine_play_media("file=/home/ringback.wav");
  }

From logs i can see that due to a=inactive rtpengine will not play 
media. I tried also to replace inactive with sendonly with function 
body_replace before I call rtpengine_play_media but it does not help.



thank you
miha
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Re: [OpenSIPS-Users] replace_body() issue

2021-06-18 Thread Miha via Users
I have without ^ and $ before, i have now tried again like you suggested 
but it does not work.
It goes in the if loop ("search_body") but it does not replace inactive 
with sendonly.

I have also look at return code from replace_body and it gives me "-1".

Stas Kobzar je 6/18/2021 ob 1:04 PM napisal:

Hello,
Just do not use ^ and $ in the search pattern. It is probably trying 
to match the whole SDP packet, not single line.


On Fri, Jun 18, 2021 at 5:09 AM Miha via Users 
mailto:users@lists.opensips.org>> wrote:


Hello

 have issue with replace_body as it does not change SDP.
My code looks like this:

if (has_body("application/sdp")){
            if(search_body("a=inactive")){
*replace_body("^a=inactive$", "a=sendonly");*

            }

                 $var(rtpengine_flags) ="trust-address
replace-origin replace-session-connection  ICE=remove RTP/AVP
rtcp-mux-demux";
                 rtpengine_offer("$var(rtpengine_flags)");

              if(is_audio_on_hold()) {

                    rtpengine_play_media("callee
file=/home/ringback.wav");
                  }

             t_on_reply("1");
}

What could be wrong that inactive is not replaced by sendonly?
On a leg I can see "a=inactive" and also on b leg "a=inactive".


thank you
miha
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[OpenSIPS-Users] replace_body() issue

2021-06-18 Thread Miha via Users

Hello

 have issue with replace_body as it does not change SDP.
My code looks like this:

if (has_body("application/sdp")){
            if(search_body("a=inactive")){
*replace_body("^a=inactive$", "a=sendonly");*

            }

                 $var(rtpengine_flags) ="trust-address replace-origin 
replace-session-connection  ICE=remove RTP/AVP rtcp-mux-demux";

                 rtpengine_offer("$var(rtpengine_flags)");

              if(is_audio_on_hold()) {

                    rtpengine_play_media("callee file=/home/ringback.wav");
                  }

             t_on_reply("1");
}

What could be wrong that inactive is not replaced by sendonly?
On a leg I can see "a=inactive" and also on b leg "a=inactive".


thank you
miha
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[OpenSIPS-Users] ACK with wrong RURI

2021-06-14 Thread Miha via Users

Hello

when call is not pickup on teams and just canceled I get 603 Declined 
which opensips send it further to SBC (our main sbc). When we get back 
ACK and from SBC, there is wrong RURI and opensips does not relay this 
ACK to MS teams.


Is anything can be done on opensips side in this situation so that i 
will not have this issue?


SIP/2.0 603 Decline
FROM: ;tag=3832669067-409068083
TO: 
;tag=7cb08507d0944d7a838dfdc418227564

CSEQ: 1 INVITE
CALL-ID: 901735-3832669067-641737...@sbc1.test.com
VIA: SIP/2.0/UDP SBC:5060;branch=z9hG4bK029c59d110eb800c4e70bac6d27181cb
REASON: 
Q.850;cause=21;text="84fbb35a-99e1-4ae7-b6cd-12d428c96190;CallEndReasonLocalUserInitiated"
RECORD-ROUTE: 
,,
CONTACT: 


CONTENT-LENGTH: 0
ALLOW: INVITE,ACK,OPTIONS,CANCEL,BYE,NOTIFY
SERVER: Microsoft.PSTNHub.SIPProxy v.2021.5.28.7 i.EUWE.6


ACK sip:87654321@OPENSIPS_IP SIP/2.0
Max-Forwards: 69
To: 
;tag=7cb08507d0944d7a838dfdc418227564

From: ;tag=3832669067-409068083
Call-ID: 901735-3832669067-641737...@sbc1.softnet.si
CSeq: 1 ACK
Via: SIP/2.0/UDP SBC:5060;branch=z9hG4bK029c59d110eb800c4e70bac6d27181cb
Contact: 
Content-Length: 0



thank you
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Re: [OpenSIPS-Users] MS teams, reinvite after ACK

2021-06-02 Thread Miha via Users

yes. Thank you.

is there any way to get also attended transfer working?

Johan De Clercq je 6/2/2021 ob 11:21 AM napisal:

remove Refer from your supported methods.
Do note that attended transfer will not work in this case.

wkr,

Op wo 2 jun. 2021 om 10:15 schreef Miha via Users 
mailto:users@lists.opensips.org>>:


Hello

i manage to fix this. I did not do t_relay() also seq Invites,
after this everything works ok.

Just on question, regarding transfers, i see that MS Teams send
REFER in which trafter is defined. How do you deal with this? You
do not allow REFER from MS teams and hope that MS teams will send
new INVITE?


thank you
    miha

Jeff Pyle je 6/1/2021 ob 3:26 PM napisal:

    Miha,

First, do you need to use "mtsbc.test.com:5060
<http://mtsbc.test.com:5060>" in the first record_route_preset()
param?  Can you use the IP address of your proxy instead?  FQDNs
are legal of course, but outside of MS Teams' implementation,
they're rarely required.  It's just another thing to go wrong. 
Especially while testing.

The ACK to the 200 OK is a sequential (in-dialog) request.  It's
not part of the original INVITE transaction.  Your script will
have a section like

if (has_totag()) {
if (loose_route()) {
t_relay();
}
}

for sequential requests through a loose-routing proxy.  This is
very oversimplified and yours will have more.  In this section,
however, is where you'll process the ACK because it has a to-tag
(line 293) and a route header (line 298) so the conditions match.

Use xlogs or the debug tool of your choice to diagnose what's
happening in this section with the ACK.  In my scripts, I use
global flag 0 to indicate when I want logging.  So, I might have
something like this:

   if (has_totag()) {
   if (is_gflag(0)) xlog("L_NOTICE", "...in-dialog
$rm request\n");
# ...do all the things...maybe more logging like the line above...


- Jeff


On Tue, Jun 1, 2021 at 4:57 AM Miha via Users
mailto:users@lists.opensips.org>> wrote:

Hello


I have an issue and I am unable to find out what is wrong.
Incoming calls are working but when doing outbound call after
200OK, which is send to Teams I get back ACK and after that
Teams do again initial. I guess this is not ok.

I am doing this for outband calls:


xlog("L_INFO", "rtp rtps record route");
record_route_preset("mtsbc.test.com:5060
<http://mtsbc.test.com:5060>","mtsbc.test.com
<http://mtsbc.test.com>:5061;transport=tls");
add_rr_param(";r2=on");

I am pasting here trace. Opensips is in the middle.

Thank you for help!

https://pastebin.com/qM0dMiCc <https://pastebin.com/qM0dMiCc>
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Re: [OpenSIPS-Users] MS teams, reinvite after ACK

2021-06-02 Thread Miha via Users

ok, it does new seq invite, so not is is working.


thank you for help.
miha

Miha via Users je 6/2/2021 ob 10:11 AM napisal:

Hello

i manage to fix this. I did not do t_relay() also seq Invites, after 
this everything works ok.


Just on question, regarding transfers, i see that MS Teams send REFER 
in which trafter is defined. How do you deal with this? You do not 
allow REFER from MS teams and hope that MS teams will send new INVITE?



thank you
miha

Jeff Pyle je 6/1/2021 ob 3:26 PM napisal:

Miha,

First, do you need to use "mtsbc.test.com:5060 
<http://mtsbc.test.com:5060>" in the first record_route_preset() 
param?  Can you use the IP address of your proxy instead?  FQDNs are 
legal of course, but outside of MS Teams' implementation, they're 
rarely required.  It's just another thing to go wrong. Especially 
while testing.


The ACK to the 200 OK is a sequential (in-dialog) request.  It's not 
part of the original INVITE transaction. Your script will have a 
section like


if (has_totag()) {
if (loose_route()) {
t_relay();
}
}

for sequential requests through a loose-routing proxy.  This is very 
oversimplified and yours will have more.  In this section, however, 
is where you'll process the ACK because it has a to-tag (line 293) 
and a route header (line 298) so the conditions match.


Use xlogs or the debug tool of your choice to diagnose what's 
happening in this section with the ACK.  In my scripts, I use global 
flag 0 to indicate when I want logging.  So, I might have something 
like this:


   if (has_totag()) {
   if (is_gflag(0)) xlog("L_NOTICE", "...in-dialog $rm 
request\n");

# ...do all the things...maybe more logging like the line above...


- Jeff


On Tue, Jun 1, 2021 at 4:57 AM Miha via Users 
mailto:users@lists.opensips.org>> wrote:


Hello


I have an issue and I am unable to find out what is wrong.
Incoming calls are working but when doing outbound call after
200OK, which is send to Teams I get back ACK and after that Teams
do again initial. I guess this is not ok.

I am doing this for outband calls:


xlog("L_INFO", "rtp rtps record route");
record_route_preset("mtsbc.test.com:5060
<http://mtsbc.test.com:5060>","mtsbc.test.com
<http://mtsbc.test.com>:5061;transport=tls");
add_rr_param(";r2=on");

I am pasting here trace. Opensips is in the middle.

Thank you for help!

https://pastebin.com/qM0dMiCc <https://pastebin.com/qM0dMiCc>
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Re: [OpenSIPS-Users] MS teams, reinvite after ACK

2021-06-02 Thread Miha via Users

Hello

i manage to fix this. I did not do t_relay() also seq Invites, after 
this everything works ok.


Just on question, regarding transfers, i see that MS Teams send REFER in 
which trafter is defined. How do you deal with this? You do not allow 
REFER from MS teams and hope that MS teams will send new INVITE?



thank you
miha

Jeff Pyle je 6/1/2021 ob 3:26 PM napisal:

Miha,

First, do you need to use "mtsbc.test.com:5060 
<http://mtsbc.test.com:5060>" in the first record_route_preset() 
param?  Can you use the IP address of your proxy instead?  FQDNs are 
legal of course, but outside of MS Teams' implementation, they're 
rarely required. It's just another thing to go wrong.  Especially 
while testing.


The ACK to the 200 OK is a sequential (in-dialog) request. It's not 
part of the original INVITE transaction.  Your script will have a 
section like


if (has_totag()) {
if (loose_route()) {
t_relay();
}
}

for sequential requests through a loose-routing proxy.  This is very 
oversimplified and yours will have more.  In this section, however, is 
where you'll process the ACK because it has a to-tag (line 293) and a 
route header (line 298) so the conditions match.


Use xlogs or the debug tool of your choice to diagnose what's 
happening in this section with the ACK.  In my scripts, I use global 
flag 0 to indicate when I want logging.  So, I might have something 
like this:


   if (has_totag()) {
   if (is_gflag(0)) xlog("L_NOTICE", "...in-dialog $rm 
request\n");

# ...do all the things...maybe more logging like the line above...


- Jeff


On Tue, Jun 1, 2021 at 4:57 AM Miha via Users 
mailto:users@lists.opensips.org>> wrote:


Hello


I have an issue and I am unable to find out what is wrong.
Incoming calls are working but when doing outbound call after
200OK, which is send to Teams I get back ACK and after that Teams
do again initial. I guess this is not ok.

I am doing this for outband calls:


xlog("L_INFO", "rtp rtps record route");
record_route_preset("mtsbc.test.com:5060
<http://mtsbc.test.com:5060>","mtsbc.test.com
<http://mtsbc.test.com>:5061;transport=tls");
add_rr_param(";r2=on");

I am pasting here trace. Opensips is in the middle.

Thank you for help!

https://pastebin.com/qM0dMiCc <https://pastebin.com/qM0dMiCc>
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[OpenSIPS-Users] MS teams, reinvite after ACK

2021-06-01 Thread Miha via Users

Hello


I have an issue and I am unable to find out what is wrong. Incoming 
calls are working but when doing outbound call after 200OK, which is 
send to Teams I get back ACK and after that Teams do again initial. I 
guess this is not ok.


I am doing this for outband calls:


xlog("L_INFO", "rtp rtps record route");

record_route_preset("mtsbc.test.com:5060","mtsbc.test.com:5061;transport=tls");
add_rr_param(";r2=on");

I am pasting here trace. Opensips is in the middle.

Thank you for help!

https://pastebin.com/qM0dMiCc
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Re: [OpenSIPS-Users] TLS to UDP, record route

2021-05-18 Thread miha- via Users
Thank you
I will check.


Br

miha
On 18 May 2021, 13:08 +0200, John Quick , wrote:
> The client I was working with used this:
> https://docs.microsoft.com/en-us/microsoftteams/direct-routing-sbc-multiple-tenants
>
> I touched on the topic in my article about MS Teams, under the heading 
> "Terminology and multi-tenant solutions"
>
> John Quick
> Smartvox Limited
>
> > From: Miha 
> > Sent: 18 May 2021 11:59
> > To: john.qu...@smartvox.co.uk; users@lists.opensips.org
> > Subject: Re: [OpenSIPS-Users] TLS to UDP, record route
> >
> > btw what is the trick if you have multiple trunks to sbc teams (inbound, 
> > outbound)? multiple companies?
> >
> > miha
>
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Re: [OpenSIPS-Users] TLS to UDP, record route

2021-05-18 Thread Miha via Users
btw what is the trick if you have multiple trunks to sbc teams (inbound, 
outbound)? multiple companies?



br
miha

John Quick je 5/18/2021 ob 11:15 AM napisal:

Miha

Altering the text in the Record-Route headers with subst() function is not
the correct approach.
I believe the problem is that you are not inserting the correct RR headers
in the first place.
If you get the RR headers right then it will also fix the problem with ACK
using the wrong protocol.
This pseudo-code snippet illustrates what is required when adding RR headers
to the initial INVITE request:

if (INVITE-from-Teams-Proxy-to-us) {
 record_route_preset("IP:port", "SBC_FQDN:5061;transport=tls");
 add_rr_param(";r2=on");
} else if (INVITE-from-us-to-Teams-Proxy) {
 record_route_preset( "SBC_FQDN:5061;transport=tls", "IP:port");
 add_rr_param(";r2=on");
} else
 record_route();

Don't insert any RR headers when handling loose-routed requests.

I tried to explain all this stuff in a number of articles. Here are the
links:
https://kb.smartvox.co.uk/opensips/opensips-as-ms-teams-sbc/
https://kb.smartvox.co.uk/opensips/nat-contact-and-via-fixing-in-sip-part-4/
https://kb.smartvox.co.uk/opensips/contact-and-record-route-headers-explaine
d/

John Quick
Smartvox Limited



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Re: [OpenSIPS-Users] TLS to UDP, record route

2021-05-18 Thread Miha via Users

hello John

i have found what was causing the issue. is was topology hiding when ACK 
was received by opensips.



thank you for all your help and time :)

br
miha

John Quick je 5/18/2021 ob 11:15 AM napisal:

Miha

Altering the text in the Record-Route headers with subst() function is not
the correct approach.
I believe the problem is that you are not inserting the correct RR headers
in the first place.
If you get the RR headers right then it will also fix the problem with ACK
using the wrong protocol.
This pseudo-code snippet illustrates what is required when adding RR headers
to the initial INVITE request:

if (INVITE-from-Teams-Proxy-to-us) {
 record_route_preset("IP:port", "SBC_FQDN:5061;transport=tls");
 add_rr_param(";r2=on");
} else if (INVITE-from-us-to-Teams-Proxy) {
 record_route_preset( "SBC_FQDN:5061;transport=tls", "IP:port");
 add_rr_param(";r2=on");
} else
 record_route();

Don't insert any RR headers when handling loose-routed requests.

I tried to explain all this stuff in a number of articles. Here are the
links:
https://kb.smartvox.co.uk/opensips/opensips-as-ms-teams-sbc/
https://kb.smartvox.co.uk/opensips/nat-contact-and-via-fixing-in-sip-part-4/
https://kb.smartvox.co.uk/opensips/contact-and-record-route-headers-explaine
d/

John Quick
Smartvox Limited



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Re: [OpenSIPS-Users] TLS to UDP, record route

2021-05-17 Thread Miha via Users

thank you.

Anther thing, when you got ACK, how opensips route it, how it will 
choose right interface?
Initial invite is send via tls to ms teams by opensips. But ACK which I 
get from another SBC is routed via UDP and not TLS? Where i should look 
for this issue?


thank you
miha


U Opensips:5060 -> Another_SBC(Not opensips):5060 #5
SIP/2.0 200 OK.
FROM: opensips)>;tag=3830244708-1542214291.
TO: 
;tag=797d0c0c74c94df5abf29cc5ba182311.

CSEQ: 1 INVITE.
CALL-ID: 16826581-3830244708-1607573...@sbc2.test.com.
VIA: SIP/2.0/UDP Another_SBC(Not 
opensips):5060;branch=z9hG4bK4c6433227c52d1863c051b23a170706e.
RECORD-ROUTE: 
,,.
CONTACT: 
.

Content-Length: 309.
SUPPORTED: timer.
CONTENT-TYPE: application/sdp.
ALLOW: INVITE,ACK,OPTIONS,CANCEL,BYE,NOTIFY.
REQUIRE: timer.
SESSION-EXPIRES: 3600;refresher=uac.
SERVER: Microsoft.PSTNHub.SIPProxy v.2021.5.5.15 i.EUWE.3.


U  Another_SBC(Not opensips):5060 -> Opensips:5060 #6
ACK 
sip:52.114.75.24:5061;x-i=d1df8c6c-2e6c-44b8-b670-de5b94e4b93e;x-c=e6e92b7127765ed3843b370d59046d54/s/1/6cff2c6f65024a6b86a5c5f3794202a5 
SIP/2.0.

Max-Forwards: 68.
Route: .
Route: .
Route: .
To: 
;tag=797d0c0c74c94df5abf29cc5ba182311.
From: opensips)>;tag=3830244708-1542214291.

Call-ID: 16826581-3830244708-1607573...@sbc2.test.com.
CSeq: 1 ACK.
Via: SIP/2.0/UDP  Another_SBC(Not 
opensips):5060;branch=z9hG4bK58b13e244cbca78925b45e9bcc69d3a2.

Contact: .
Content-Length: 0.
.


U Opensips:5060 -> 52.114.75.24:5061 #7
ACK 
sip:52.114.75.24:5061;x-i=d1df8c6c-2e6c-44b8-b670-de5b94e4b93e;x-c=e6e92b7127765ed3843b370d59046d54/s/1/6cff2c6f65024a6b86a5c5f3794202a5 
SIP/2.0.

Max-Forwards: 67.
Route: .
Route: .
Route: .
To: 
;tag=797d0c0c74c94df5abf29cc5ba182311.
From: opensips)>;tag=3830244708-1542214291.

Call-ID: 16826581-3830244708-1607573...@sbc2.test.com.
CSeq: 1 ACK.
Via: SIP/2.0/UDP Opensips:5060;branch=z9hG4bKed06.0b1dc487.2.
Via: SIP/2.0/UDP  Another_SBC(Not 
opensips):5060;branch=z9hG4bK58b13e244cbca78925b45e9bcc69d3a2.

Contact: .
Content-Length: 0.



Callum Guy je 5/17/2021 ob 10:15 AM napisal:

subst_uri only works on the request uri, try again with subst()!


On Mon, 17 May 2021 at 08:58, Miha via Users <mailto:users@lists.opensips.org>> wrote:


Hello

i need a little help how to chnage RR in responses to UDP GW
(requestes goes via TLS to MS teams).

So in reply i have like this:
 RECORD-ROUTE:

,.

But i should have like this: RECORD-ROUTE:

,.

I tried to do it like:  subst_uri('/mtsbc.test.com
<http://mtsbc.test.com>:5061;transport=tls/xxx.xxx.xxx.:5060/i');
but it does not match.


thank you
miha

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[OpenSIPS-Users] TLS to UDP, record route

2021-05-17 Thread Miha via Users

Hello

i need a little help how to chnage RR in responses to UDP GW (requestes 
goes via TLS to MS teams).


So in reply i have like this:
 RECORD-ROUTE: 
,.


But i should have like this: RECORD-ROUTE: 
,.


I tried to do it like:  
subst_uri('/mtsbc.test.com:5061;transport=tls/xxx.xxx.xxx.:5060/i'); 
but it does not match.



thank you
miha

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Re: [OpenSIPS-Users] MS team issue

2021-05-11 Thread Miha via Users
hello

i tried to put this in address table:
"*.pstnhub.microsoft.com" but it does not work.

On Tue, 11 May 2021 09:13:37 +0200
 Johan De Clercq  wrote:
> the pstnhub's can change their ip address.
> Therefore you need to use the fqdn.
> 
> Op ma 10 mei 2021 om 21:33 schreef Miha via Users
>  >:
> 
> > found an issue. It was missing ip in addresses. Is
> there
> > any easier way to put all servers from Ms to addresses,
> > maybe just domain with "*."?
> >
> >
> > thank you
> >
> >
> > On Mon, 10 May 2021 19:23:20 +0200
> >  Miha via Users  wrote:
> > > Hello
> > >
> > > it seems for me that this works now. I only do not
> know
> > > why
> > > after 200 ok, opensips sends OPTIONS also to it self,
> > > which
> > > is quite wird.
> > >
> > > pasting logs.
> > >
> > > y 10 19:20:26 mtsbc opensips[7582]:
> > > DBG:tm:print_request_uri:
> sip:sip.pstnhub.microsoft.com
> > > May 10 19:20:26 mtsbc opensips[7582]:
> > > DBG:tm:run_local_route: building sip_msg from buffer
> > > May 10 19:20:26 mtsbc opensips[7582]:
> DBG:core:parse_msg:
> > > SIP Request:
> > > May 10 19:20:26 mtsbc opensips[7582]:
> DBG:core:parse_msg:
> > >  method:  
> > > May 10 19:20:26 mtsbc opensips[7582]:
> DBG:core:parse_msg:
> > >  uri: 
> > > May 10 19:20:26 mtsbc opensips[7582]:
> DBG:core:parse_msg:
> > >  version: 
> > > May 10 19:20:26 mtsbc opensips[7582]:
> > > DBG:core:parse_headers: flags=
> > > May 10 19:20:26 mtsbc opensips[7582]:
> > > DBG:core:parse_via_param: found param type 232,
> 
> > > =
> > > ; state=16
> > > May 10 19:20:26 mtsbc opensips[7582]:
> DBG:core:parse_via:
> > > end of header reached, state=5
> > > May 10 19:20:26 mtsbc opensips[7582]:
> > > DBG:core:parse_headers: via found,
> flags=
> > > May 10 19:20:26 mtsbc opensips[7582]:
> > > DBG:core:parse_headers: this is the first via
> > > May 10 19:20:26 mtsbc opensips[7582]:
> DBG:core:_parse_to:
> > > end of header reached, state=9
> > > May 10 19:20:26 mtsbc opensips[7582]:
> DBG:core:_parse_to:
> > > display={}, ruri={sip:sip.pstnhub.microsoft.com}
> > > May 10 19:20:26 mtsbc opensips[7582]:
> > > DBG:core:get_hdr_field:  [31];
> > > uri=[sip:sip.pstnhub.microsoft.com]
> > > May 10 19:20:26 mtsbc opensips[7582]:
> > > DBG:core:get_hdr_field: to body
> > > [sip:sip.pstnhub.microsoft.com#015#012
> > <http://sip.pstnhub.microsoft.com#015%23012>]
> > > May 10 19:20:26 mtsbc opensips[7582]:
> > > DBG:core:get_hdr_field: cseq : <14> 
> > > May 10 19:20:26 mtsbc opensips[7582]:
> > > DBG:core:get_hdr_field: content_length=0
> > > May 10 19:20:26 mtsbc opensips[7582]:
> > > DBG:core:get_hdr_field: found end of header
> > > May 10 19:20:26 mtsbc opensips[7582]:
> > > DBG:core:parse_headers: flags=
> > > May 10 19:20:26 mtsbc opensips[7582]:
> > > DBG:core:parse_headers: flags=78
> > > May 10 19:20:26 mtsbc opensips[7582]:
> > > DBG:core:parse_headers: flags=
> > > May 10 19:20:26 mtsbc opensips[7582]:
> > > DBG:core:parse_headers: flags=
> > > May 10 19:20:26 mtsbc opensips[7582]:
> > > DBG:core:parse_to_param:
> > > tag=a665d66adab06c7308a33b8567de92d6-7c10
> > > May 10 19:20:26 mtsbc opensips[7582]:
> DBG:core:_parse_to:
> > > end of header reached, state=29
> > > May 10 19:20:26 mtsbc opensips[7582]:
> DBG:core:_parse_to:
> > > display={}, ruri={sip:prober@localhost}
> > > May 10 19:20:26 mtsbc opensips[7582]:
> > > DBG:sipcapture:w_sip_capture: src_ip:
> [xxx.xxx.xxx.xxx]
> > > May 10 19:20:26 mtsbc opensips[7582]:
> > > DBG:sipcapture:w_sip_capture: dst_ip:
> [xxx.xxx.xxx.xxx]
> > > May 10 19:20:26 mtsbc opensips[7582]:
> > > DBG:sipcapture:w_sip_capture: dst_port: [5061]
> > > May 10 19:20:26 mtsbc opensips[7582]:
> > > DBG:sipcapture:w_sip_capture: src_port: [5061]
> > > May 10 19:20:26 mtsbc opensips[7582]:
> > > DBG:sipcapture:w_sip_capture: DONE
> > > May 10 19:20:26 mtsbc opensips[7582]:
> > > DBG:sipcapture:db_sync_store: storing info...
> > > May 10 19:20:26 mtsbc opensips[7582]:
> > > DBG:db_mysql:db_mysql_do_prepared_query:
> > > conn=0x7f60225e6108 (tail=140050870197640)
> > > MC=

Re: [OpenSIPS-Users] MS team issue

2021-05-10 Thread Miha via Users
found an issue. It was missing ip in addresses. Is there
any easier way to put all servers from Ms to addresses,
maybe just domain with "*."?


thank you


On Mon, 10 May 2021 19:23:20 +0200
 Miha via Users  wrote:
> Hello
> 
> it seems for me that this works now. I only do not know
> why
> after 200 ok, opensips sends OPTIONS also to it self,
> which
> is quite wird. 
> 
> pasting logs.
> 
> y 10 19:20:26 mtsbc opensips[7582]:
> DBG:tm:print_request_uri: sip:sip.pstnhub.microsoft.com
> May 10 19:20:26 mtsbc opensips[7582]:
> DBG:tm:run_local_route: building sip_msg from buffer
> May 10 19:20:26 mtsbc opensips[7582]: DBG:core:parse_msg:
> SIP Request:
> May 10 19:20:26 mtsbc opensips[7582]: DBG:core:parse_msg:
>  method:  
> May 10 19:20:26 mtsbc opensips[7582]: DBG:core:parse_msg:
>  uri: 
> May 10 19:20:26 mtsbc opensips[7582]: DBG:core:parse_msg:
>  version: 
> May 10 19:20:26 mtsbc opensips[7582]:
> DBG:core:parse_headers: flags=
> May 10 19:20:26 mtsbc opensips[7582]:
> DBG:core:parse_via_param: found param type 232, 
> =
> ; state=16
> May 10 19:20:26 mtsbc opensips[7582]: DBG:core:parse_via:
> end of header reached, state=5
> May 10 19:20:26 mtsbc opensips[7582]:
> DBG:core:parse_headers: via found, flags=
> May 10 19:20:26 mtsbc opensips[7582]:
> DBG:core:parse_headers: this is the first via
> May 10 19:20:26 mtsbc opensips[7582]: DBG:core:_parse_to:
> end of header reached, state=9
> May 10 19:20:26 mtsbc opensips[7582]: DBG:core:_parse_to:
> display={}, ruri={sip:sip.pstnhub.microsoft.com}
> May 10 19:20:26 mtsbc opensips[7582]:
> DBG:core:get_hdr_field:  [31];
> uri=[sip:sip.pstnhub.microsoft.com] 
> May 10 19:20:26 mtsbc opensips[7582]:
> DBG:core:get_hdr_field: to body
> [sip:sip.pstnhub.microsoft.com#015#012]
> May 10 19:20:26 mtsbc opensips[7582]:
> DBG:core:get_hdr_field: cseq : <14> 
> May 10 19:20:26 mtsbc opensips[7582]:
> DBG:core:get_hdr_field: content_length=0
> May 10 19:20:26 mtsbc opensips[7582]:
> DBG:core:get_hdr_field: found end of header
> May 10 19:20:26 mtsbc opensips[7582]:
> DBG:core:parse_headers: flags=
> May 10 19:20:26 mtsbc opensips[7582]:
> DBG:core:parse_headers: flags=78
> May 10 19:20:26 mtsbc opensips[7582]:
> DBG:core:parse_headers: flags=
> May 10 19:20:26 mtsbc opensips[7582]:
> DBG:core:parse_headers: flags=
> May 10 19:20:26 mtsbc opensips[7582]:
> DBG:core:parse_to_param:
> tag=a665d66adab06c7308a33b8567de92d6-7c10
> May 10 19:20:26 mtsbc opensips[7582]: DBG:core:_parse_to:
> end of header reached, state=29
> May 10 19:20:26 mtsbc opensips[7582]: DBG:core:_parse_to:
> display={}, ruri={sip:prober@localhost}
> May 10 19:20:26 mtsbc opensips[7582]:
> DBG:sipcapture:w_sip_capture: src_ip: [xxx.xxx.xxx.xxx]
> May 10 19:20:26 mtsbc opensips[7582]:
> DBG:sipcapture:w_sip_capture: dst_ip: [xxx.xxx.xxx.xxx]
> May 10 19:20:26 mtsbc opensips[7582]:
> DBG:sipcapture:w_sip_capture: dst_port: [5061]
> May 10 19:20:26 mtsbc opensips[7582]:
> DBG:sipcapture:w_sip_capture: src_port: [5061]
> May 10 19:20:26 mtsbc opensips[7582]:
> DBG:sipcapture:w_sip_capture: DONE
> May 10 19:20:26 mtsbc opensips[7582]:
> DBG:sipcapture:db_sync_store: storing info...
> May 10 19:20:26 mtsbc opensips[7582]:
> DBG:db_mysql:db_mysql_do_prepared_query:
> conn=0x7f60225e6108 (tail=140050870197640)
> MC=0x7f60225e6218
> May 10 19:20:26 mtsbc opensips[7582]:
> DBG:db_mysql:db_mysql_do_prepared_query: new
> query=|insert
> into sip_capture
>
(date,micro_ts,method,reply_reason,ruri,ruri_user,from_user,from_tag,to_user,to_tag,pid_user,contact_user,auth_user,callid,callid_aleg,via_1,via_1_branch,cseq,reason,content_type,auth,user_agent,source_ip,source_port,destination_ip,destination_port,contact_ip,contact_port,originator_ip,originator_port,proto,family,rtp_stat,type,node,correlation_id,from_domain,to_domain,ruri_domain,msg,custom_field1,custom_field2,custom_field3
> ) values
>
(?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?)|
> May 10 19:20:26 mtsbc opensips[7582]:
> DBG:db_mysql:re_init_statement:  query  is  sip_capture
>
(date,micro_ts,method,reply_reason,ruri,ruri_user,from_user,from_tag,to_user,to_tag,pid_user,contact_user,auth_user,callid,callid_aleg,via_1,via_1_branch,cseq,reason,content_type,auth,user_agent,source_ip,source_port,destination_ip,destination_port,contact_ip,contact_port,originator_ip,originator_port,proto,family,rtp_stat,type,node,correlation_id,from_domain,to_domain,ruri_domain,msg,custom_field1,custom_field2,custom_field3
> ) values
>
(?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?)>,
> ptr=(nil)
> May 10 19:

Re: [OpenSIPS-Users] MS team issue

2021-05-10 Thread Miha via Users
te#015#012FROM:
;tag=dd7d4e13-28ca-405e-9108-d83d4ea81f40#015#012TO:
;tag=bd3a.8217c8258ec89ff4f5427ff311954104#015#012CSEQ:
1 OPTIONS#015#012CALL-ID:
bf155f68-8a4c-4f18-b5f5-1340da0d618d#015#012VIA:
SIP/2.0/TLS
52.114.75.24:5061;branch=z9hG4bK9fe9fb55#015#012Server:
OpenSIPS (3.1.1 (x86_64/linux))#015#012Content-Length:
0#015#012#015#012
May 10 19:20:26 mtsbc opensips[7575]:
DBG:core:destroy_avp_list: destroying list (nil)


On Mon, 10 May 2021 11:45:32 -0300
 Carlos Eduardo  wrote:
> Thank you Nick.
> 
> I've read these docs lots of times and didn't pay
> attention on it.
> 
> 
> Em seg., 10 de mai. de 2021 às 11:44, Nick Altmann
> 
> escreveu:
> 
> > Yes. You can use avp for this.
> >
>
https://opensips.org/docs/modules/3.1.x/tls_mgm.html#param_client_sip_domain_avp
> >
> > --
> > Nick
> >
> > пн, 10 мая 2021 г. в 16:09, Carlos Eduardo
> :
> >
> >> Hey all,
> >>
> >> About using the right certificate, is it possible to
> ensure opensips is
> >> going to use the right one when multiple are set in
> tls_mgm?
> >>
> >> Em seg., 10 de mai. de 2021 às 04:41, Răzvan Crainea
> 
> >> escreveu:
> >>
> >>> Hi, Miha!
> >>>
> >>> According to your logs, opensips is 100% sending the
> OPTIONS through
> >>> tls, but I am not sure it is using the right
> certificate.
> >>> You can try to setup sip trace and see the
> communication between
> >>> opensips and MSTeams.
> >>>
> >>> Best regards,
> >>>
> >>> Răzvan Crainea
> >>> OpenSIPS Core Developer
> >>> http://www.opensips-solutions.com
> >>>
> >>> On 5/10/21 9:54 AM, Miha via Users wrote:
> >>> > Hello
> >>> >
> >>> > I have used letsenrypt for generating certs for
> Opensips.
> >>> >
> >>> > Regarding configuration i have fallowed your
> configuration steps on
> >>> > OpenSips blog.
> >>> >
> >>> > socket=udp:xxx.xxx.xxx.xxx:5060   # CUSTOMIZE ME
> >>> > socket=tls:xxx.xxx.xxx.xxx:5061
> >>> >
> >>> >
> >>> >
> >>> >
> >>> > ### Proto TLS
> >>> > loadmodule "proto_tls.so"
> >>> > modparam("proto_tls", "tls_handshake_timeout", 300)
> >>> >  TLS module
> >>> > loadmodule "tls_mgm.so"
> >>> > #modparam("tls_mgm", "db_url",
> "mysql://root:@localhost/opensips")
> >>> > modparam("tls_mgm", "client_sip_domain_avp",
> "mtsbcs.test.com")
> >>> > modparam("tls_mgm", "server_domain", "mt")
> >>> > #modparam("tls_mgm", "match_ip_address",
> "[mt]xxx.xxx.xxx.xxx:5061")
> >>> > #modparam("tls_mgm", "match_sip_domain",
> "[mt]mtsbcs.test.com")
> >>> > modparam("tls_mgm", "certificate",
> >>> >
> "[mt]/etc/letsencrypt/live/mtsbcs.test.com/cert.pem")
> >>> > modparam("tls_mgm", "private_key",
> >>> >
> "[mt]/etc/letsencrypt/live/mtsbcs.test.com/privkey.pem")
> >>> > modparam("tls_mgm", "ca_list",
> >>> "[mt]/etc/ssl/certs/ca-certificates.crt")
> >>> > modparam("tls_mgm", "ca_dir",
> "[mt]/etc/ssl/certs/")
> >>> > modparam("tls_mgm","verify_cert", "[mt]1")
> >>> > modparam("tls_mgm","require_cert", "[mt]1")
> >>> > modparam("tls_mgm","tls_method", "[mt]TLSv1_2")
> >>> > modparam("proto_tls", "tls_max_msg_chunks", 8)
> >>> > #modparam("tls_mgm", "tls_handshake_timeout", 300)
> >>> >
> >>> >  if(is_method("OPTIONS") &&
> is_domain_local("$rd") &&
> >>> > check_source_address(0)) {
> >>> >  xlog("L_INFO", "[MS TEAMS] OPTIONS
> In");
> >>> >  send_reply(200, "OK");
> >>> >  exit;
> >>> >  }
> >>> >
> >>> >
> >>> > local_route {
> >>> >$var(dst) = "pstnhub.microsoft.com"

[OpenSIPS-Users] MS team issue

2021-05-09 Thread Miha via Users
f
May 10 08:53:10 mtsbc opensips[1020]: DBG:tm:run_local_route: Change in 
local route -> rebuilding buffer

May 10 08:53:10 mtsbc opensips[1020]: DBG:core:parse_headers: flags=2000
May 10 08:53:10 mtsbc opensips[1020]: DBG:core:parse_headers: 
flags=

May 10 08:53:10 mtsbc opensips[1020]: DBG:core:extract_ftc_hdrs: flags = 15
May 10 08:53:10 mtsbc opensips[1020]: DBG:core:extract_ftc_hdrs: hdr 2 
extracted as 
May 10 08:53:10 mtsbc opensips[1020]: DBG:core:extract_ftc_hdrs: hdr 1 
extracted as ;tag=a665d66adab06c7308a33b8567de92d6-f627#015#012>
May 10 08:53:10 mtsbc opensips[1020]: DBG:core:extract_ftc_hdrs: hdr 8 
extracted as 
May 10 08:53:10 mtsbc opensips[1020]: DBG:proto_tls:proto_tls_send: no 
open tcp connection found, opening new one
May 10 08:53:10 mtsbc opensips[1020]: DBG:core:probe_max_sock_buff: 
getsockopt: snd is initially 16384
May 10 08:53:10 mtsbc opensips[1020]: DBG:core:probe_max_sock_buff: 
using snd buffer of 416 kb
May 10 08:53:10 mtsbc opensips[1020]: DBG:core:init_sock_keepalive: TCP 
keepalive enabled on socket 5
May 10 08:53:10 mtsbc opensips[1020]: DBG:core:print_ip: tcpconn_new: 
new tcp connection to: 52.114.75.24
May 10 08:53:10 mtsbc opensips[1020]: DBG:core:tcpconn_new: on port 
5061, proto 3
May 10 08:53:10 mtsbc opensips[1020]: DBG:proto_tls:tls_conn_init: 
Creating a whole new ssl connection
May 10 08:53:10 mtsbc opensips[1020]: DBG:core:tcpconn_destroy: 
destroying connection 0x7f45d7e08078, flags 0018
May 10 08:53:10 mtsbc opensips[1020]: DBG:tm:insert_timer_unsafe: [0]: 
0x7f45d7e066b0 (1625)
May 10 08:53:15 mtsbc opensips[1020]: DBG:tm:timer_routine: timer 
routine:0,tl=0x7f45d7e066b0 next=(nil), timeout=1625
May 10 08:53:15 mtsbc opensips[1020]: DBG:tm:final_response_handler: 
Cancel sent out, sending 408 (0x7f45d7e06460)
May 10 08:53:15 mtsbc opensips[1020]: DBG:tm:t_should_relay_response: 
T_code=0, new_code=408
May 10 08:53:15 mtsbc opensips[1020]: DBG:tm:t_pick_branch: picked 
branch 0, code 408 (prio=800)
May 10 08:53:15 mtsbc opensips[1020]: DBG:tm:is_3263_failure: 
dns-failover test: branch=0, last_recv=408, flags=0
May 10 08:53:15 mtsbc opensips[1020]: DBG:tm:t_should_relay_response: 
trying DNS-based failover
May 10 08:53:15 mtsbc opensips[1020]: DBG:tm:do_dns_failover: new 
destination available

May 10 08:53:15 mtsbc opensips[1020]: DBG:core:parse_headers: flags=2000
May 10 08:53:15 mtsbc opensips[1020]: 
DBG:core:build_req_buf_from_sip_req: id added: <;i=0>, rcv proto=3
May 10 08:53:15 mtsbc opensips[1020]: DBG:core:parse_headers: 
flags=
May 10 08:53:15 mtsbc opensips[1020]: DBG:proto_tls:proto_tls_send: no 
open tcp connection found, opening new one
May 10 08:53:15 mtsbc opensips[1020]: DBG:core:probe_max_sock_buff: 
getsockopt: snd is initially 16384
May 10 08:53:15 mtsbc opensips[1020]: DBG:core:probe_max_sock_buff: 
using snd buffer of 416 kb
May 10 08:53:15 mtsbc opensips[1020]: DBG:core:init_sock_keepalive: TCP 
keepalive enabled on socket 5
May 10 08:53:15 mtsbc opensips[1020]: DBG:core:print_ip: tcpconn_new: 
new tcp connection to: 52.114.132.46
May 10 08:53:15 mtsbc opensips[1020]: DBG:core:tcpconn_new: on port 
5061, proto 3
May 10 08:53:15 mtsbc opensips[1020]: DBG:proto_tls:tls_conn_init: 
Creating a whole new ssl connection
May 10 08:53:15 mtsbc opensips[1020]: DBG:core:tcpconn_destroy: 
destroying connection 0x7f45d7e08078, flags 0018
May 10 08:53:15 mtsbc opensips[1020]: DBG:proto_tls:proto_tls_send: no 
open tcp connection found, opening new one
May 10 08:53:15 mtsbc opensips[1020]: DBG:core:probe_max_sock_buff: 
getsockopt: snd is initially 16384
May 10 08:53:15 mtsbc opensips[1020]: DBG:core:probe_max_sock_buff: 
using snd buffer of 416 kb
May 10 08:53:15 mtsbc opensips[1020]: DBG:core:init_sock_keepalive: TCP 
keepalive enabled on socket 5
May 10 08:53:15 mtsbc opensips[1020]: DBG:core:print_ip: tcpconn_new: 
new tcp connection to: 52.114.14.70
May 10 08:53:15 mtsbc opensips[1020]: DBG:core:tcpconn_new: on port 
5061, proto 3
May 10 08:53:15 mtsbc opensips[1020]: DBG:proto_tls:tls_conn_init: 
Creating a whole new ssl connection
May 10 08:53:15 mtsbc opensips[1020]: DBG:core:tcpconn_destroy: 
destroying connection 0x7f45d7e08078, flags 0018
May 10 08:53:15 mtsbc opensips[1020]: DBG:tm:local_reply: branch=0, 
save=0, winner=0
May 10 08:53:15 mtsbc opensips[1020]: DBG:tm:local_reply: local 
transaction completed
May 10 08:53:15 mtsbc opensips[1020]: DBG:tm:run_trans_callbacks: 
trans=0x7f45d7e06460, callback type 256, id 0 entered
May 10 08:53:15 mtsbc opensips[1020]: DBG:tm:insert_timer_unsafe: [2]: 
0x7f45d7e064e0 (1630)

May 10 08:53:15 mtsbc opensips[1020]: DBG:tm:final_response_handler: done



Thank you
miha

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[OpenSIPS-Users] utimer task already scheduled

2021-05-07 Thread Miha via Users

Hi

I have falow opensips configuration from your blog regarding MsTeams. 
Version that I am using is:


(opensips-cli): mi  version
{
    "Server": "OpenSIPS (3.1.1 (x86_64/linux))"
}

ay  7 13:15:06 mtsbc opensips[1966]: WARNING:core:utimer_ticker: utimer 
task  already scheduled 100 ms ago (now 155250 ms), delaying 
execution
May  7 13:15:06 mtsbc opensips[1966]: WARNING:core:utimer_ticker: utimer 
task  already scheduled 200 ms ago (now 155350 ms), delaying 
execution
May  7 13:15:06 mtsbc opensips[1966]: WARNING:core:utimer_ticker: utimer 
task  already scheduled 290 ms ago (now 155440 ms), delaying 
execution



What could cause this behaviour? On opensips for now nothing is running, 
this starts from the beginning when opensips start.



thank you
miha
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Re: [OpenSIPS-Users] opensips-cli issue with DB creation

2021-05-03 Thread Miha via Users

hello

i changed config to like this and now it works:

[default]
#database_modules: acc clusterer dialog dialplan dispatcher domain 
rtpproxy usrloc

database_modules: ALL


log_level: WARNING
#prompt_name: opensips-cli
#prompt_intro: Welcome to OpenSIPS
#prompt_emptyline_repeat_cmd: False
#history_file: ~/.opensips-cli.history
#history_file_size: 1000
#output_type: pretty-print
communication_type: fifo
fifo_file: /tmp/opensips_fifo
#[mysql]
database_path: /tmp/opensips-3.1/scripts
database_name: opensips
database_admin_url: mysql://root:xxx@localhost
database_url: mysql://opensips:opensipsrw@localhost
database_schema_path: /tmp/opensips-3.1/scripts

Miha via Users je 5/3/2021 ob 12:00 PM napisal:

Pasting also log:

 database create
DEBUG: running command 'create' '[]'
DEBUG: db_name: 'opensips'
Password for admin MySQL user (root):
DEBUG: read password: 'xxx'
DEBUG: admin DB URL: 'mysql://root:xxx@localhost'
DEBUG: connecting to mysql://root:xxx@localhost
DEBUG: check database URL 'mysql://root:xxx@localhost/opensips'
DEBUG: DB does not exist
DEBUG: Create Database 'opensips' for dialect 'mysql' ...
DEBUG: success
DEBUG: DB URL: 'mysql://opensips:opensipsrw@localhost'
DEBUG: connecting to mysql://opensips:opensipsrw@localhost/opensips
INFO: creating access user for opensips ...
/usr/local/lib/python3.7/dist-packages/SQLAlchemy-1.3.3-py3.7-linux-x86_64.egg/sqlalchemy/engine/default.py:552: 
Warning: (3163, "Authorization ID 'opensips'@'%' already exists.")

  cursor.execute(statement, parameters)
INFO: created user 'opensips'
INFO: set password 'ow' for 'opensips' (MySQL)
INFO: granted access to user 'opensips' on DB 'opensips'
INFO: flushed privileges
DEBUG: connecting to mysql://opensips:opensipsrw@localhost/opensips
ERROR: failed to connect to DB as opensips, please provide or fix the 
'database_url'


Miha via Users je 5/3/2021 ob 10:50 AM napisal:

Hello

I have config for opensips-cli like this in /etc/opensips-cli.cfg

database_modules: ALL

log_level: WARNING
prompt_name: opensips-cli
prompt_intro: Welcome to OpenSIPS at SECUREVOIP
prompt_emptyline_repeat_cmd: False
history_file: ~/.opensips-cli.history
history_file_size: 1000
output_type: pretty-print
communication_type: fifo
fifo_file: /tmp/opensips_fifo
database_path: /tmp/opensips-3.1/scripts
database_name: opensips
database_admin_url: mysql://root:@localhost:3306/opensips
database_url: mysql://opensips:opensipsrw@localhost:3306/opensips

When i run in opensips-cli shell "database create" i get:

ERROR: failed to connect to DB as opensips, please provide or fix the 
'database_url'


I can see data DB opensips was created and also opensips user was 
created.



thank you
miha

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Re: [OpenSIPS-Users] opensips-cli issue with DB creation

2021-05-03 Thread Miha via Users

Pasting also log:

 database create
DEBUG: running command 'create' '[]'
DEBUG: db_name: 'opensips'
Password for admin MySQL user (root):
DEBUG: read password: 'xxx'
DEBUG: admin DB URL: 'mysql://root:xxx@localhost'
DEBUG: connecting to mysql://root:xxx@localhost
DEBUG: check database URL 'mysql://root:xxx@localhost/opensips'
DEBUG: DB does not exist
DEBUG: Create Database 'opensips' for dialect 'mysql' ...
DEBUG: success
DEBUG: DB URL: 'mysql://opensips:opensipsrw@localhost'
DEBUG: connecting to mysql://opensips:opensipsrw@localhost/opensips
INFO: creating access user for opensips ...
/usr/local/lib/python3.7/dist-packages/SQLAlchemy-1.3.3-py3.7-linux-x86_64.egg/sqlalchemy/engine/default.py:552: 
Warning: (3163, "Authorization ID 'opensips'@'%' already exists.")

  cursor.execute(statement, parameters)
INFO: created user 'opensips'
INFO: set password 'ow' for 'opensips' (MySQL)
INFO: granted access to user 'opensips' on DB 'opensips'
INFO: flushed privileges
DEBUG: connecting to mysql://opensips:opensipsrw@localhost/opensips
ERROR: failed to connect to DB as opensips, please provide or fix the 
'database_url'


Miha via Users je 5/3/2021 ob 10:50 AM napisal:

Hello

I have config for opensips-cli like this in /etc/opensips-cli.cfg

database_modules: ALL

log_level: WARNING
prompt_name: opensips-cli
prompt_intro: Welcome to OpenSIPS at SECUREVOIP
prompt_emptyline_repeat_cmd: False
history_file: ~/.opensips-cli.history
history_file_size: 1000
output_type: pretty-print
communication_type: fifo
fifo_file: /tmp/opensips_fifo
database_path: /tmp/opensips-3.1/scripts
database_name: opensips
database_admin_url: mysql://root:@localhost:3306/opensips
database_url: mysql://opensips:opensipsrw@localhost:3306/opensips

When i run in opensips-cli shell "database create" i get:

ERROR: failed to connect to DB as opensips, please provide or fix the 
'database_url'


I can see data DB opensips was created and also opensips user was created.


thank you
miha

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[OpenSIPS-Users] opensips-cli issue with DB creation

2021-05-03 Thread Miha via Users

Hello

I have config for opensips-cli like this in /etc/opensips-cli.cfg

database_modules: ALL

log_level: WARNING
prompt_name: opensips-cli
prompt_intro: Welcome to OpenSIPS at SECUREVOIP
prompt_emptyline_repeat_cmd: False
history_file: ~/.opensips-cli.history
history_file_size: 1000
output_type: pretty-print
communication_type: fifo
fifo_file: /tmp/opensips_fifo
database_path: /tmp/opensips-3.1/scripts
database_name: opensips
database_admin_url: mysql://root:@localhost:3306/opensips
database_url: mysql://opensips:opensipsrw@localhost:3306/opensips

When i run in opensips-cli shell "database create" i get:

ERROR: failed to connect to DB as opensips, please provide or fix the 
'database_url'


I can see data DB opensips was created and also opensips user was created.


thank you
miha
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[OpenSIPS-Users] how can i combine signaling and RTP from rtpproxy

2021-04-20 Thread Miha via Users

Hello

due to debugging i would like to combine cap from opensips and also cap 
from rtpproxy (they are on different servers) so that I can check if RTP 
is missing for certain call.



Can you help me with solving this issue :)

thank you
miha
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[OpenSIPS-Users] Load testing

2020-05-11 Thread miha- via Users
Hi

What is best tool for load testing that can generate also RTP?

Tnx

miha
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[OpenSIPS-Users] migration to 2.4

2020-05-07 Thread Miha via Users

Hello

we are running still on 2.1. Due to some other things I would like first 
to migrate to version 2.4. I went over documentation for version 
migration from 2.1 to 2.2 and from 2.2 to 2.3 and from 2.3. to 2.4.


What I would like to know is what exactly is wrong in my config in where 
i should be looking for.
the main issue is that I do not see this in logs. Log level is 4 (i 
tried aslo with 7 and other leves.)


This are logs:
Is there any other way to find this issue?

  6 21:26:24 debian opensips[6423]: NOTICE:core:main: version: opensips 
2.4.7 (x86_64/linux)
May  6 21:26:24 debian opensips[6423]: INFO:core:main: using 32 Mb of 
shared memory
May  6 21:26:24 debian opensips[6423]: INFO:core:main: using 2 Mb of 
private process memory
May  6 21:26:24 debian opensips[6423]: INFO:core:init_reactor_size: 
reactor size 1024 (using up to 0.03Mb of memory per process)
May  6 21:26:24 debian opensips[6423]: INFO:core:evi_publish_event: 
Registered event 
May  6 21:26:24 debian opensips[6423]: INFO:core:evi_publish_event: 
Registered event 
May  6 21:26:24 debian opensips[6423]: INFO:core:evi_publish_event: 
Registered event 
May  6 21:26:24 debian opensips[6423]: INFO:tm:mod_init: TM - 
initializing...
May  6 21:26:24 debian opensips[6423]: INFO:sl:mod_init: Initializing 
StateLess engine
May  6 21:26:24 debian opensips[6423]: NOTICE:signaling:mod_init: 
initializing module ...

May  6 21:26:24 debian opensips[6423]: INFO:rr:mod_init: rr - initializing
May  6 21:26:24 debian opensips[6423]: INFO:maxfwd:mod_init: initializing...
May  6 21:26:24 debian opensips[6423]: INFO:sipmsgops:mod_init: 
initializing...
May  6 21:26:24 debian opensips[6423]: INFO:usrloc:ul_init_locks: locks 
array size 512
May  6 21:26:24 debian opensips[6423]: INFO:core:evi_publish_event: 
Registered event 
May  6 21:26:24 debian opensips[6423]: INFO:core:evi_publish_event: 
Registered event 
May  6 21:26:24 debian opensips[6423]: INFO:core:evi_publish_event: 
Registered event 
May  6 21:26:24 debian opensips[6423]: INFO:core:evi_publish_event: 
Registered event 
May  6 21:26:24 debian opensips[6423]: INFO:core:evi_publish_event: 
Registered event 
May  6 21:26:24 debian opensips[6423]: INFO:core:evi_publish_event: 
Registered event 
May  6 21:26:24 debian opensips[6423]: INFO:registrar:mod_init: 
initializing...

May  6 21:26:24 debian opensips[6423]: INFO:acc:mod_init: initializing...
May  6 21:26:24 debian opensips[6423]: INFO:core:evi_publish_event: 
Registered event 
May  6 21:26:24 debian opensips[6423]: INFO:core:evi_publish_event: 
Registered event 
May  6 21:26:24 debian opensips[6423]: INFO:core:evi_publish_event: 
Registered event 
May  6 21:26:24 debian opensips[6423]: INFO:core:mod_init: initializing 
UDP-plain protocol
May  6 21:26:24 debian opensips[6423]: INFO:core:probe_max_sock_buff: 
using rcv buffer of 416 kb
May  6 21:26:24 debian opensips: INFO:core:daemonize: pre-daemon process 
exiting with 0

May  7 09:15:33 debian opensips: NOTICE:core:main: Exiting
May  7 09:15:57 debian opensips: NOTICE:core:main: Exiting
May  7 09:16:56 debian opensips: NOTICE:core:main: Exiting
May  7 09:17:23 debian opensips: NOTICE:core:main: Exiting


thank you
miha
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Re: [OpenSIPS-Users] Opensips + rtpproxy issue

2020-05-06 Thread Miha via Users
Hello Maxim

rtpproxy -V
2.2.alpha.aa26b45

it did not. Now rtpproxy is running as sing process, could
it be lunched like opensips (multiple instances)?

Is there any other thing that can be done?


thank you
miha

On Wed, 6 May 2020 07:22:07 -0700
 Maxim Sobolev  wrote:
> Oh, sorry, looks like the proper way to check that is via
> using -V option.
> 
> [ssp-root@macmini2 /home/ssp]$ rtpproxy -V
> 2.2.alpha.3794729c-dirty
> 
> Let us know if that upgrade helped. Thanks!
> 
> -Max
> 
> On Wed, May 6, 2020 at 2:09 AM Miha 
> wrote:
> 
> > Hello Maxim
> >
> > I removed package via apt and then install it via
> source/git.
> >
> > But from what I see version is still the same:
> > How would I know for sure that the version is something
> like 2.1?
> >
> > ./rtpproxy -version
> > Basic version: 20040107
> > Extension 20040107: Basic RTP proxy functionality
> > Extension 20050322: Support for multiple RTP streams
> and MOH
> > Extension 20060704: Support for extra parameter in the
> V command
> > Extension 20071116: Support for RTP re-packetization
> > Extension 20071218: Support for forking (copying) RTP
> stream
> > Extension 20080403: Support for RTP statistics querying
> > Extension 20081102: Support for setting codecs in the
> update/lookup command
> > Extension 20081224: Support for session timeout
> notifications
> > Extension 20090810: Support for automatic bridging
> > Extension 20140323: Support for tracking/reporting load
> > Extension 20140617: Support for anchoring session
> connect time
> > Extension 20141004: Support for extendable performance
> counters
> > Extension 20150330: Support for allocating a new port
> ("Un"/"Ln" commands)
> > Extension 20150420: Support for SEQ tracking and new
> rtpa_ counters; Q
> > command extended
> > Extension 20150617: Support for the wildcard
> %%CC_SELF%% as a disconnect
> > notify target
> > Extension 20191015: Support for the && sub-command
> specifier
> > Extension 20200226: Support for the N command to stop
> recording
> >
> >
> >
> > Maxim Sobolev je 5/6/2020 ob 2:22 AM napisal:
> >
> > Hi Miha, sorry to hear about your issues. In order to
> troubleshoot it
> > further could you please also provide rtpproxy package
> version as reported
> > by the system package manager (apt, rpm etc) if the
> software has been
> > installed via that channel or branch name if it's been
> built from sources?
> > Unfortunately version reporting of the --version
> command has been bit
> > crippled until recently, already improved in latest
> master and 2.1 I
> > believe.
> >
> > In general performance under virtual environment has
> not been terrific,
> > due to some design choices made early in our work.
> Hovewer I believe it
> > should be much better in 2.0 and 2.1 vs. 1.x series.
> Some of it is
> > inherently due to VM scheduling jitter, some is because
> we are unwilling to
> > put it into unsafe domain (i.e. kernel mode). As a rule
> of thumb, you might
> > expect 3-5x drop in max pps until jitter becomes an
> issue as compared to
> > running on comparable bare metal. Spinning multiple
> instances might help to
> > mitigate some of it though, but it also depends on
> hypervisor version and
> > even particular CPU generation.
> >
> > -Max
> >
> > On Tue., May 5, 2020, 6:10 a.m. Miha via Users,
> 
> > wrote:
> >
> >> Hello
> >>
> >> we have virtualized opensips and rtpproxy running on
> the same server
> >> which is virtualized in vmware infrastructure. Servers
> are not old, also
> >> traffic is not so big (cca 50 simultaneous calls).
> when there is a peak cca
> >> 80 simultaneous calls RTP starts to break.
> >>
> >> is there any special setting/flag to be set, so that I
> can optimze this?
> >> load on VM is very low.
> >>
> >> rtpproxy -version
> >> Basic version: 20040107
> >>
> >> Opensips is 2.1
> >>
> >>
> >> thank you for help.
> >> Miha
> >> ___
> >> Users mailing list
> >> Users@lists.opensips.org
> >>
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >>
> >
> >
> 
> -- 
> Maksym Sobolyev
> Sippy Software, Inc.
> Internet Telephony (VoIP) Experts
> Tel (Canada): +1-778-783-0474
> Tel (Toll-Free): +1-855-747-7779
> Fax: +1-866-857-6942
> Web: http://www.sippysoft.com
> MSN: sa...@sippysoft.com
> Skype: SippySoft


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Re: [OpenSIPS-Users] Opensips + rtpproxy issue

2020-05-06 Thread Miha via Users

Hello Maxim

I removed package via apt and then install it via source/git.

But from what I see version is still the same:
How would I know for sure that the version is something like 2.1?

./rtpproxy -version
Basic version: 20040107
Extension 20040107: Basic RTP proxy functionality
Extension 20050322: Support for multiple RTP streams and MOH
Extension 20060704: Support for extra parameter in the V command
Extension 20071116: Support for RTP re-packetization
Extension 20071218: Support for forking (copying) RTP stream
Extension 20080403: Support for RTP statistics querying
Extension 20081102: Support for setting codecs in the update/lookup command
Extension 20081224: Support for session timeout notifications
Extension 20090810: Support for automatic bridging
Extension 20140323: Support for tracking/reporting load
Extension 20140617: Support for anchoring session connect time
Extension 20141004: Support for extendable performance counters
Extension 20150330: Support for allocating a new port ("Un"/"Ln" commands)
Extension 20150420: Support for SEQ tracking and new rtpa_ counters; Q 
command extended
Extension 20150617: Support for the wildcard %%CC_SELF%% as a disconnect 
notify target

Extension 20191015: Support for the && sub-command specifier
Extension 20200226: Support for the N command to stop recording



Maxim Sobolev je 5/6/2020 ob 2:22 AM napisal:
Hi Miha, sorry to hear about your issues. In order to troubleshoot it 
further could you please also provide rtpproxy package version as 
reported by the system package manager (apt, rpm etc) if the software 
has been installed via that channel or branch name if it's been built 
from sources? Unfortunately version reporting of the --version command 
has been bit crippled until recently, already improved in latest 
master and 2.1 I believe.


In general performance under virtual environment has not been 
terrific, due to some design choices made early in our work. Hovewer I 
believe it should be much better in 2.0 and 2.1 vs. 1.x series. Some 
of it is inherently due to VM scheduling jitter, some is because we 
are unwilling to put it into unsafe domain (i.e. kernel mode). As a 
rule of thumb, you might expect 3-5x drop in max pps until jitter 
becomes an issue as compared to running on comparable bare metal. 
Spinning multiple instances might help to mitigate some of it though, 
but it also depends on hypervisor version and even particular CPU 
generation.


-Max

On Tue., May 5, 2020, 6:10 a.m. Miha via Users, 
mailto:users@lists.opensips.org>> wrote:


Hello

we have virtualized opensips and rtpproxy running on the same
server which is virtualized in vmware infrastructure. Servers are
not old, also traffic is not so big (cca 50 simultaneous calls).
when there is a peak cca 80 simultaneous calls RTP starts to break.

is there any special setting/flag to be set, so that I can optimze
this? load on VM is very low.

rtpproxy -version
Basic version: 20040107

Opensips is 2.1


thank you for help.
Miha
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Re: [OpenSIPS-Users] Opensips + rtpproxy issue

2020-05-05 Thread miha- via Users
Hello Danies

• what do you mean by enough max open do files? I do no linit or set anything
• I traced with tshark and i can see issue with A and B leg


Thank you for help!
Br
Miha

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

2020-05-05 Thread Miha via Users
Hi,

no CPU usage is around 1% to 5%, basically nothing.
In sound there is big distortion it is impossibly to
comunicate with each other.

We have two cors deticated to it. Do you have any special
thing set on it?

tnx
miha

On Tue, 5 May 2020 10:27:22 -0300
 Daniel Zanutti  wrote:
> Hi Miha
> 
> Could you explaining how does it break? We use it in
> virtual machines and
> our safe limit is around 500 simultaneous calls, on
> dedicated single core
> VPS. Does CPU usage reach 100%?
> 
> 
> 
> On Tue, May 5, 2020 at 10:11 AM Miha via Users
> 
> wrote:
> 
> > Hello
> >
> > we have virtualized opensips and rtpproxy running on
> the same server which
> > is virtualized in vmware infrastructure. Servers are
> not old, also traffic
> > is not so big (cca 50 simultaneous calls). when there
> is a peak cca 80
> > simultaneous calls RTP starts to break.
> >
> > is there any special setting/flag to be set, so that I
> can optimze this?
> > load on VM is very low.
> >
> > rtpproxy -version
> > Basic version: 20040107
> >
> > Opensips is 2.1
> >
> >
> > thank you for help.
> > Miha
> > ___
> > Users mailing list
> > Users@lists.opensips.org
> >
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >


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[OpenSIPS-Users] Opensips + rtpproxy issue

2020-05-05 Thread Miha via Users

Hello

we have virtualized opensips and rtpproxy running on the same server 
which is virtualized in vmware infrastructure. Servers are not old, also 
traffic is not so big (cca 50 simultaneous calls). when there is a peak 
cca 80 simultaneous calls RTP starts to break.


is there any special setting/flag to be set, so that I can optimze this? 
load on VM is very low.


rtpproxy -version
Basic version: 20040107

Opensips is 2.1


thank you for help.
Miha
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Re: [OpenSIPS-Users] Alias domain / dns srv

2020-04-15 Thread Miha via Users

Hello Bogdan

not mixing, just maybe wrong discribing :) That is what i did.

Thank you for your help and explenation!


br
miha

Bogdan-Andrei Iancu je 4/15/2020 ob 7:31 PM napisal:

Hi Miha,

You are mixing the SIP domains with the SIP server location. The SIP 
domains have nothing to do with SRV, while for SIP server location you 
can use it.


The idea is to set as SIP user u...@sip.test.com (and 'sip.test.com' 
is the SIP domain all the time).


If the domain does not support SRV, it will do an A lookup on 
sip.test.com, and you can point it , as IP, to proxy1.test.com.


If the domain supports SRVyou know the drill .

But in both cases the SIP domain in SIP messages will be 'sip.test.com'

Regards,
Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
   https://www.opensips-solutions.com

On 4/14/20 11:57 AM, Miha via Users wrote:

Hello

we have dns srv record for failover. In dns srv we have two record.
So, one version of our devices does not support dns srv records. Is 
it possible to register device directly to one A record which is 
wirtten in DNS SRV record and then use ALIAS in opensips to right domain?


DNS SRV.

sip.test.com  (proxy1.test.com, proxy2.test.com)
Devices that do not support will register to proxy1.test.com 
(opensips will have alias which will point to sip.test.com)?





thank you
miha

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[OpenSIPS-Users] Alias domain / dns srv

2020-04-14 Thread Miha via Users

Hello

we have dns srv record for failover. In dns srv we have two record.
So, one version of our devices does not support dns srv records. Is it 
possible to register device directly to one A record which is wirtten in 
DNS SRV record and then use ALIAS in opensips to right domain?


DNS SRV.

sip.test.com  (proxy1.test.com, proxy2.test.com)
Devices that do not support will register to proxy1.test.com (opensips 
will have alias which will point to sip.test.com)?





thank you
miha
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Re: [OpenSIPS-Users] different ip in from as initial invite

2020-01-28 Thread Miha via Users

Liviu

thank you very much for your quick answer! I will try then to stick as 
it is as it is the right way. If there will be no other choise that 
maybe i try this.


thank you again!
miha

Liviu Chircu je 1/28/2020 ob 1:52 PM napisal:

On 28.01.2020 14:43, Miha via Users wrote:
Costumer is saying that he expects from like it was send in 200ok 
(not in inital invite, tag and CALLERID stays always the same) and we 
should confirm with ACK that has from same as in 200 ok from them.


Hi miha,

That is complete nonsense, RFC 3261 is on your side, section § 8.2.6.2:

   The From field of the response MUST equal the From header field of
   the request.  The Call-ID header field of the response MUST equal the
   Call-ID header field of the request.  The CSeq header field of the
   response MUST equal the CSeq field of the request. The Via header
   field values in the response MUST equal the Via header field values
   in the request and MUST maintain the same ordering.

However, if you are really keen to help them out... maybe you could 
store their
200 OK From header in a $dlg_val, then fix the ACK's From header to 
use this val.


But how will you handle the From header for other sequential 
requests?  And if these
requests are initiated by the downstream side, you will have to change 
the To instead
of the From, as the UAC must swap them!  We are basically opening 
Pandora's Box by
doing down this route.  It's not impossible to get right, but it will 
take some work.


Regards,

--
Liviu Chircu
www.twitter.com/liviuchircu  |www.opensips-solutions.com

OpenSIPS Summit, Amsterdam, May 2020
   www.opensips.org/events
OpenSIPS Bootcamp, Miami, March 2020
   www.opensips.org/training


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[OpenSIPS-Users] different ip in from as initial invite

2020-01-28 Thread Miha via Users

Hi

first call flow.
1. Invite with FROM 12345@1.2.3.4
2. 200 ok with FROM 1.2.3.4@1.2.3.5
3. ACK, FROM is like in initial invite 12345@1.2.3.4

Costumer is saying that he expects from like it was send in 200ok (not 
in inital invite, tag and CALLERID stays always the same) and we should 
confirm with ACK that has from same as in 200 ok from them.


Problem is that in my case opensips adds FROM from initial invite (ip 
1.2.3.4, it should be 1.2.3.5). IN onreply route a can not use 
uac_change_from.



Can this be change and how?

thank you
miha
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[OpenSIPS-Users] Remove doubled connection information in SDP

2019-10-29 Thread Miha via Users

Hello

I get two connection infomrmation in SDP (doubled), which are the same. 
How to remove one?


ps.: i am using rtpproxy.

thank you.
Miha
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Re: [OpenSIPS-Users] opensips/rtpporxy and early media

2019-08-23 Thread Miha via Users

Hi Razvan

just one thing to be clear as I do not see anything in my debugging mode.

So after i get beck 180 ringing, must call:

rtpproxy_answer("rfo",,"2");
and
rtpproxy_stream2uas("/tmp/wav.wav", "-1"); (I did conversion and files 
are like wav.wav.3, .0, .8)


Must I do something else? This is done on on_replay route. I guess I 
must change to 183 session in progress?



thank you for help!
Miha





On 8/19/2019 8:57 PM, Miha via Users wrote:

Hi, Răzvan!

thank you, so i was thinking right :)


br
miha

On Mon, 19 Aug 2019 17:28:07 +0300
  Răzvan Crainea  wrote:

Hi, Miha!

You first need to convert the wav file you want to stream
to a RTP payload, one for each codec you support. To do
that, you can use the makeann tool that rtpproxy
provides[1].
Once you have those files (named file.3 for GSM, file.0
for PCMU. file.8 for PCMA), you need to call the
rtpproxy_stream2uac("file"). This will automatically do
the codec selection and choose the right file.

[1] https://github.com/sippy/rtpproxy/tree/master/makeann

Best regards,
Răzvan

On 8/19/19 4:07 PM, Miha via Users wrote:

Hello guys

first time doing this, normally I use freeswitch... Se

in combination with rtpproxy how to enable ringback tone.
I need to call rtpproxy_stream2() i add it as file? Or
there is some other option for this if I would like that
is played by UAS?


thank you for help!
miha

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Re: [OpenSIPS-Users] opensips/rtpporxy and early media

2019-08-19 Thread Miha via Users
Hi, Răzvan!

thank you, so i was thinking right :)


br
miha

On Mon, 19 Aug 2019 17:28:07 +0300
 Răzvan Crainea  wrote:
> Hi, Miha!
> 
> You first need to convert the wav file you want to stream
> to a RTP payload, one for each codec you support. To do
> that, you can use the makeann tool that rtpproxy
> provides[1].
> Once you have those files (named file.3 for GSM, file.0
> for PCMU. file.8 for PCMA), you need to call the
> rtpproxy_stream2uac("file"). This will automatically do
> the codec selection and choose the right file.
> 
> [1] https://github.com/sippy/rtpproxy/tree/master/makeann
> 
> Best regards,
> Răzvan
> 
> On 8/19/19 4:07 PM, Miha via Users wrote:
> > Hello guys
> > 
> > first time doing this, normally I use freeswitch... Se
> in combination with rtpproxy how to enable ringback tone.
> I need to call rtpproxy_stream2() i add it as file? Or
> there is some other option for this if I would like that
> is played by UAS?
> > 
> > 
> > thank you for help!
> > miha
> > 
> > ___
> > Users mailing list
> > Users@lists.opensips.org
> >
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> > 
> 
> -- 
> Răzvan Crainea
> OpenSIPS Core Developer
>http://www.opensips-solutions.com
> 
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[OpenSIPS-Users] opensips/rtpporxy and early media

2019-08-19 Thread Miha via Users

Hello guys

first time doing this, normally I use freeswitch... Se in combination 
with rtpproxy how to enable ringback tone. I need to call 
rtpproxy_stream2() i add it as file? Or there is some other option for 
this if I would like that is played by UAS?



thank you for help!
miha
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Re: [OpenSIPS-Users] issue with compiling opensips

2017-11-20 Thread Miha

hello

is there on www where written which libraries are all mandatory (debian)?


tnx
miha

On 17/11/2017 08:42, Ketan Kothari wrote:

Hello Miha,

Please verify below dependency installed on your server or not.

libjson-c-dev
libjson-c-doc
libjson-c2
libjson-c2-dbg
libjson0
libjson0-dev



On Fri, Nov 17, 2017 at 12:58 PM, Miha <mailto:m...@softnet.si>> wrote:


hey

this is what i have:

cat /etc/issue
Debian GNU/Linux 9 \n \


apt list libjson*
Listing... Done
libjson-any-perl/stable 1.39-1 all
libjson-c-dev/stable,now 0.12.1-1.1 amd64 [installed]
libjson-c-doc/stable 0.12.1-1.1 all
libjson-c3/stable,now 0.12.1-1.1 amd64 [installed,automatic]
libjson-glib-1.0-0/stable,now 1.2.6-1 amd64 [installed,automatic]
libjson-glib-1.0-common/stable,now 1.2.6-1 all [installed,automatic]
libjson-glib-dev/stable 1.2.6-1 amd64
libjson-glib-doc/stable 1.2.6-1 all
libjson-java/stable 2.4-3 all
libjson-maybexs-perl/stable 1.003008-1 all
libjson-multivalueordered-perl/stable 0.005-1 all
libjson-perl/stable 2.90-1 all
libjson-pointer-perl/stable 0.07-1 all
libjson-pp-perl/stable 2.27400-1 all
libjson-rpc-perl/stable 1.06-2 all
libjson-simple-doc/stable 1.1.1-4 all
libjson-simple-java/stable 1.1.1-4 all
libjson-smart-java/stable 2.2-1 all
libjson-types-perl/stable 0.05-1 all
libjson-validator-perl/stable 0.92+dfsg-1 all
libjson-webtoken-perl/stable 0.10-2 all
libjson-xs-perl/stable 3.030-1 amd64
libjsoncpp-dev/stable,now 1.7.4-3 amd64 [installed]
libjsoncpp-doc/stable 1.7.4-3 all
libjsoncpp1/stable,now 1.7.4-3 amd64 [installed,automatic]
libjsonm-ocaml/stable 0.9.1-2 amd64
libjsonm-ocaml-dev/stable 0.9.1-2 amd64
libjsonm-ocaml-doc/stable 0.9.1-2 all
libjsonp-java/stable 1.0.4-1 all
libjsonp-java-doc/stable 1.0.4-1 all
libjsonpath-java/stable 2.0.0-3 all
libjsonrpccpp-client0/stable 0.7.0-1+b2 amd64
libjsonrpccpp-client0-dbg/stable 0.7.0-1+b2 amd64
libjsonrpccpp-common0/stable 0.7.0-1+b2 amd64
libjsonrpccpp-common0-dbg/stable 0.7.0-1+b2 amd64
libjsonrpccpp-dev/stable 0.7.0-1+b2 amd64
libjsonrpccpp-server0/stable 0.7.0-1+b2 amd64
libjsonrpccpp-server0-dbg/stable 0.7.0-1+b2 amd64
libjsonrpccpp-stub0/stable 0.7.0-1+b2 amd64
libjsonrpccpp-stub0-dbg/stable 0.7.0-1+b2 amd64
libjsonrpccpp-tools/stable 0.7.0-1+b2 amd64




On 16/11/2017 16:47, Răzvan Crainea wrote:

What version of libjson are you using, and what OS are you running?
Răzvan Crainea
OpenSIPS Developer
www.opensips-solutions.com <http://www.opensips-solutions.com>
On 11/16/2017 01:49 PM, Miha wrote:

hello Razvan

no, i downloaded it from opensips on git, i did this yesterday (2.3)


br
miha

On 16/11/2017 11:23, Răzvan Crainea wrote:

Are you using an older version of OpenSIPS? This should have
been fixed in all supported versions.

Best regards,
Răzvan Crainea
OpenSIPS Developer
www.opensips-solutions.com <http://www.opensips-solutions.com>
On 11/15/2017 01:52 PM, Miha wrote:

Hello

which deb pack should I install regarding this error:
https://pastebin.com/c4RHMbcT

I installed this package " libjson-c-dev" but it is not ok.


tnx
miha


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Re: [OpenSIPS-Users] issue with compiling opensips

2017-11-16 Thread Miha

hey

this is what i have:

cat /etc/issue
Debian GNU/Linux 9 \n \


apt list libjson*
Listing... Done
libjson-any-perl/stable 1.39-1 all
libjson-c-dev/stable,now 0.12.1-1.1 amd64 [installed]
libjson-c-doc/stable 0.12.1-1.1 all
libjson-c3/stable,now 0.12.1-1.1 amd64 [installed,automatic]
libjson-glib-1.0-0/stable,now 1.2.6-1 amd64 [installed,automatic]
libjson-glib-1.0-common/stable,now 1.2.6-1 all [installed,automatic]
libjson-glib-dev/stable 1.2.6-1 amd64
libjson-glib-doc/stable 1.2.6-1 all
libjson-java/stable 2.4-3 all
libjson-maybexs-perl/stable 1.003008-1 all
libjson-multivalueordered-perl/stable 0.005-1 all
libjson-perl/stable 2.90-1 all
libjson-pointer-perl/stable 0.07-1 all
libjson-pp-perl/stable 2.27400-1 all
libjson-rpc-perl/stable 1.06-2 all
libjson-simple-doc/stable 1.1.1-4 all
libjson-simple-java/stable 1.1.1-4 all
libjson-smart-java/stable 2.2-1 all
libjson-types-perl/stable 0.05-1 all
libjson-validator-perl/stable 0.92+dfsg-1 all
libjson-webtoken-perl/stable 0.10-2 all
libjson-xs-perl/stable 3.030-1 amd64
libjsoncpp-dev/stable,now 1.7.4-3 amd64 [installed]
libjsoncpp-doc/stable 1.7.4-3 all
libjsoncpp1/stable,now 1.7.4-3 amd64 [installed,automatic]
libjsonm-ocaml/stable 0.9.1-2 amd64
libjsonm-ocaml-dev/stable 0.9.1-2 amd64
libjsonm-ocaml-doc/stable 0.9.1-2 all
libjsonp-java/stable 1.0.4-1 all
libjsonp-java-doc/stable 1.0.4-1 all
libjsonpath-java/stable 2.0.0-3 all
libjsonrpccpp-client0/stable 0.7.0-1+b2 amd64
libjsonrpccpp-client0-dbg/stable 0.7.0-1+b2 amd64
libjsonrpccpp-common0/stable 0.7.0-1+b2 amd64
libjsonrpccpp-common0-dbg/stable 0.7.0-1+b2 amd64
libjsonrpccpp-dev/stable 0.7.0-1+b2 amd64
libjsonrpccpp-server0/stable 0.7.0-1+b2 amd64
libjsonrpccpp-server0-dbg/stable 0.7.0-1+b2 amd64
libjsonrpccpp-stub0/stable 0.7.0-1+b2 amd64
libjsonrpccpp-stub0-dbg/stable 0.7.0-1+b2 amd64
libjsonrpccpp-tools/stable 0.7.0-1+b2 amd64




On 16/11/2017 16:47, Răzvan Crainea wrote:

What version of libjson are you using, and what OS are you running?
Răzvan Crainea
OpenSIPS Developer
www.opensips-solutions.com
On 11/16/2017 01:49 PM, Miha wrote:

hello Razvan

no, i downloaded it from opensips on git, i did this yesterday (2.3)


br
miha

On 16/11/2017 11:23, Răzvan Crainea wrote:
Are you using an older version of OpenSIPS? This should have been 
fixed in all supported versions.


Best regards,
Răzvan Crainea
OpenSIPS Developer
www.opensips-solutions.com
On 11/15/2017 01:52 PM, Miha wrote:

Hello

which deb pack should I install regarding this error: 
https://pastebin.com/c4RHMbcT


I installed this package " libjson-c-dev" but it is not ok.


tnx
miha


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Re: [OpenSIPS-Users] issue with compiling opensips

2017-11-16 Thread Miha

hello Razvan

no, i downloaded it from opensips on git, i did this yesterday (2.3)


br
miha

On 16/11/2017 11:23, Răzvan Crainea wrote:
Are you using an older version of OpenSIPS? This should have been 
fixed in all supported versions.


Best regards,
Răzvan Crainea
OpenSIPS Developer
www.opensips-solutions.com
On 11/15/2017 01:52 PM, Miha wrote:

Hello

which deb pack should I install regarding this error: 
https://pastebin.com/c4RHMbcT


I installed this package " libjson-c-dev" but it is not ok.


tnx
miha


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[OpenSIPS-Users] issue with compiling opensips

2017-11-15 Thread Miha

Hello

which deb pack should I install regarding this error: 
https://pastebin.com/c4RHMbcT


I installed this package " libjson-c-dev" but it is not ok.


tnx
miha
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Re: [OpenSIPS-Users] radius accounting, Acct-Status-Type = 0

2017-01-17 Thread Miha

Tnx:)

working!


br
miha



On 17/01/2017 12:30, Ionut Ionita wrote:
Also if you'll check latest commit in master[0], modified radius 
code to throw error and not start if any attribute/
value/vendor is not found in order to not get to similar situations in 
the future. Backported the commit back to 2.1.


[0] 
https://github.com/OpenSIPS/opensips/commit/75573b41c5453b495a3fa9ad1bdf2df3ee0f4c2f


Regards,

Ionut Ionita
OpenSIPS Developer
On 01/17/2017 12:06 PM, Ionut Ionita wrote:

Hello again,

Found a fix for your problem. You should use *dictionary.rfc2866* 
(freeradius has it). This dictionary has value defined

for *Failed*.

VALUE›  Acct-Status-Type›   ›   Start›  ›   › 1
VALUE›  Acct-Status-Type›   ›   Stop›   ›   ›   2
VALUE›  Acct-Status-Type›   ›   Alive›  ›   ›   3   # dup
VALUE›  Acct-Status-Type›   ›   Interim-Update› ›   3
VALUE›  Acct-Status-Type›   ›   Accounting-On›  ›   7
VALUE›  Acct-Status-Type›   ›   Accounting-Off› ›   8
VALUE›  Acct-Status-Type›   ›   Failed› ›   ›   15


Regards,

Ionut Ionita
OpenSIPS Developer
On 01/17/2017 10:33 AM, Miha wrote:
ok, then i will use this value and on radius side i will catch it 
and set to internally to STOP if this is only soluton.


The thing is when 486/busy comes, opensips sends this broken radius 
accouting request and radius server does not replay to it and that 
is why the opensips waits almost 40s to send back ACK on 486.



Or is there any other solution?


br
miha

On 17/01/2017 09:27, Ionut Ionita wrote:
No, thank you it's fine now. That value was for testing purposes 
only, you can remove it now.
OpenSIPS it's using *Failed* value for *Acct-Status-Type* which is 
not defined anywhere (nor in
our dictionary, nor in any RFC or somewhere else). Not finding that 
value results in having *0
*for *Acct-Status-Type*, the value you were seeing before. Will let 
you know when we'll decide

how we should fix this issue.

Regards,
Ionut Ionita
OpenSIPS Developer
On 01/17/2017 09:38 AM, Miha wrote:

Hi Ionut

do I need on bouth sides or opensips side?

I can see that now i get:  Acct-Status-Type = Modem-Start in radius.


br
miha

On 16/01/2017 16:45, Ionut Ionita wrote:


Hi Miha,

Can you set in your radius dictionary file where the 
*Acct-Status-Type* values are defined


VALUEAcct-Status-TypeStart›  ›   ›   1
VALUEAcct-Status-TypeStop›   ›   ›   2
VALUEAcct-Status-TypeAlive›  ›   ›   3
VALUEAcct-Status-TypeAccounting-On›  › 7
VALUEAcct-Status-TypeAccounting-Off› ›   8

the following line, just below the others:

VALUEAcct-Status-TypeFailed4

and then check if you'll see *4 *instead of *0* for 
*Acct-Status-Type*?


It seems that opensips it's using a value that's not in the 
radius dictionary, *Failed* value.



Ionut Ionita
OpenSIPS Developer
On 01/16/2017 04:05 PM, Miha wrote:

Hello

how can i define that for 486/busy opensips will send 
Acct-Status-Type = 2 to radius server?
Acct-Status-Type = 0 it not like standard thing and it should 
not be send :)



I imported dictinary.opensips and .sip.


tnx
miha


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Re: [OpenSIPS-Users] radius accounting, Acct-Status-Type = 0

2017-01-17 Thread Miha
ok, then i will use this value and on radius side i will catch it and 
set to internally to STOP if this is only soluton.


The thing is when 486/busy comes, opensips sends this broken radius 
accouting request and radius server does not replay to it and that is 
why the opensips waits almost 40s to send back ACK on 486.



Or is there any other solution?


br
miha

On 17/01/2017 09:27, Ionut Ionita wrote:
No, thank you it's fine now. That value was for testing purposes only, 
you can remove it now.
OpenSIPS it's using *Failed* value for *Acct-Status-Type* which is not 
defined anywhere (nor in
our dictionary, nor in any RFC or somewhere else). Not finding that 
value results in having *0
*for *Acct-Status-Type*, the value you were seeing before. Will let 
you know when we'll decide

how we should fix this issue.

Regards,
Ionut Ionita
OpenSIPS Developer
On 01/17/2017 09:38 AM, Miha wrote:

Hi Ionut

do I need on bouth sides or opensips side?

I can see that now i get:  Acct-Status-Type = Modem-Start in radius.


br
miha

On 16/01/2017 16:45, Ionut Ionita wrote:


Hi Miha,

Can you set in your radius dictionary file where the 
*Acct-Status-Type* values are defined


VALUEAcct-Status-TypeStart›  ›   ›   1
VALUEAcct-Status-TypeStop›   ›   ›   2
VALUEAcct-Status-TypeAlive›  ›   ›   3
VALUEAcct-Status-TypeAccounting-On›  › 7
VALUEAcct-Status-TypeAccounting-Off› ›   8

the following line, just below the others:

VALUEAcct-Status-TypeFailed4

and then check if you'll see *4 *instead of *0* for *Acct-Status-Type*?

It seems that opensips it's using a value that's not in the radius 
dictionary, *Failed* value.



Ionut Ionita
OpenSIPS Developer
On 01/16/2017 04:05 PM, Miha wrote:

Hello

how can i define that for 486/busy opensips will send 
Acct-Status-Type = 2 to radius server?
Acct-Status-Type = 0 it not like standard thing and it should not 
be send :)



I imported dictinary.opensips and .sip.


tnx
miha


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Re: [OpenSIPS-Users] radius accounting, Acct-Status-Type = 0

2017-01-16 Thread Miha

Hi Ionut

do I need on bouth sides or opensips side?

I can see that now i get:  Acct-Status-Type = Modem-Start in radius.


br
miha

On 16/01/2017 16:45, Ionut Ionita wrote:


Hi Miha,

Can you set in your radius dictionary file where the 
*Acct-Status-Type* values are defined


VALUEAcct-Status-TypeStart›  ›   ›   1
VALUEAcct-Status-TypeStop›   ›   ›   2
VALUEAcct-Status-TypeAlive›  ›   ›   3
VALUEAcct-Status-TypeAccounting-On›  › 7
VALUEAcct-Status-TypeAccounting-Off› ›   8

the following line, just below the others:

VALUEAcct-Status-TypeFailed4

and then check if you'll see *4 *instead of *0* for *Acct-Status-Type*?

It seems that opensips it's using a value that's not in the radius 
dictionary, *Failed* value.



Ionut Ionita
OpenSIPS Developer
On 01/16/2017 04:05 PM, Miha wrote:

Hello

how can i define that for 486/busy opensips will send 
Acct-Status-Type = 2 to radius server?
Acct-Status-Type = 0 it not like standard thing and it should not be 
send :)



I imported dictinary.opensips and .sip.


tnx
miha


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[OpenSIPS-Users] radius accounting, Acct-Status-Type = 0

2017-01-16 Thread Miha

Hello

how can i define that for 486/busy opensips will send Acct-Status-Type = 
2 to radius server?

Acct-Status-Type = 0 it not like standard thing and it should not be send :)


I imported dictinary.opensips and .sip.


tnx
miha
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Re: [OpenSIPS-Users] Opensips crash

2016-12-23 Thread Miha

Hey Denis

https://www.opensips.org/Documentation/TroubleShooting-Crash

it will create core.


miha

On 23/12/2016 10:35, Denis wrote:
One question. If in the destination directory of the core file will be 
located another "core" file, what will be? Would "old core" file be 
replaced by a new one, or Opensips makes another core file with a 
fresh data?

Thank you.
--
С уважением, Денис.
Best regards, Denis
23.12.2016, 12:23, "Răzvan Crainea" :
Please update to the latest 2.2.2. If you still have problems, try to 
make sure opensips can generate a corefile[1].


[1] http://www.opensips.org/Documentation/TroubleShooting-Crash

Best regards,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com <http://www.opensips-solutions.com/>
On 12/23/2016 11:16 AM, Denis wrote:

Hello!

Server:: OpenSIPS (2.2.1 (x86_64/linux))
Today i had a crash of Opensips.
Everything that i could collect is here 
https://yadi.sk/i/dyNnXpBr34YJQ3
Unfortunately, i could not find any fresh core file, despite of the 
fact that Opensips starts with -w /opensipscore option.

In opensipscore i found only core file at 29 Nov.
Thank you for any help.
--
С уважением, Денис.
Best regards, Denis

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Re: [OpenSIPS-Users] nat issue

2016-11-21 Thread Miha

Hello Bogdan

i think it is no need to do that if this client is broken. You already 
doing so much good with opensips ;)


Tnx so much with all explanation and all you work!
Miha

On 21/11/2016 11:13, Bogdan-Andrei Iancu wrote:

Hi Miha,

According the SIP grammar, that parameter is perfectly legitimate. The 
client is broken as it is not able to cope with it (in the worst case, 
to simply ignore it).


There is no out of the box way to disable it, but I may provide you a 
patch for that - just to see if that fixes your problem.


Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 18.11.2016 19:54, Miha wrote:

Hello bogdan

I guess, but it looks like so. Is it possible to remove it?


tnx
miha

On 18/11/2016 15:39, Bogdan-Andrei Iancu wrote:
I guess your UAC freezes when receiving back in the 200 OK REGISTER 
the "received" hdr param in Contact ??


Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 18.11.2016 16:33, Bogdan-Andrei Iancu wrote:

HI Miha,

Sorry, but I'm not able to follow the case you mentioned with 
Innovaphone PBX - maybe you can post (to see the differences) the 
sent and returned contact hdrs in the REGISTER request + reply for 
the 2 cases (OpenSIPS and Innovaphone PBX).


Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 18.11.2016 11:20, Miha wrote:
I do not know if this is the case. But from what I can see what I 
register it on some Innovaphone PBX, innovaphone sends back in 
contact (200 ok) just ip of IPBX and also when INVITE is send in 
contact there is URI of PBX and only and it works.


i tried this but did not have any luck.

br
miha

On 18/11/2016 09:48, Bogdan-Andrei Iancu wrote:

Hi Miha,

You mean the UAC does not like the multi-URI Contact header in 
the 200 OK  If so, that UAC is really broken as 1) breaks the 
SIP syntax (which allows it) and 2) breaks the the SIP 
Registration as per RFC3261.


What about the second contact (the one already existing in usrloc 
when this registration comes) ? can it be discarded ? If YES, you 
can try passing the "c1f" flags to save() :

http://www.opensips.org/html/docs/modules/2.2.x/registrar.html#id294033

That will make OpenSIPS to accept only 1 contact per AOR/user and 
any new contact will override the existing one.


Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 18.11.2016 10:15, Miha wrote:

Hi Bogdan

I did few more test. This contact bothers UAC. Is there anything 
i can do in this case in OpenSIPS so that it will only reply 
with one URI in contact?


Contact:;expires=1518
;received="sip:84.41.125.21:5060",;
expires=180.


tnx so much!
MIha

On 17/11/2016 12:11, Bogdan-Andrei Iancu wrote:

Hi Miha,

yes, that is parallel forking (you may have more than 2 
contacts only).


Are you sure your DB was sync'ed? OpenSIPS is periodically 
flushing the memory cache into the location table (see the 
"state" of the contact (as per "ul show") if marked as DIRTY).


In regards to RFC, I think you quote the wrong section (maybe 
about callings?) - for REGISTERs, any number of URIs are 
allowed AFAIK.


Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 17.11.2016 12:35, Miha wrote:

Bodan

so this is dual forking...?
So if you have one account and you have two phones on it and 
first will try  to register, 200 ok will will have contact of 
both phones?
In location table I can see only one registration for this 
user but for "opensipsctl ul show" it shows me two contacts, 
which is strange? (When i do trace only one invite is send) 
and UAC replay with Busy all the time due to two contacts 
(this what i have been told).


Ok, but if you look at rfc there is only one URI allowed in 
contact if I understand this right?



The Contact header field MUST be present and contain exactly one SIP
or SIPS URI in any request that can result in the establishment of a
dialog

Please correct me if I am wrong.


tnx so much!
Miha

On 17/11/2016 11:22, Bogdan-Andrei Iancu wrote:

Hi Miha,

OpenSIPS returns in the 200 OK for a REGISTER all the valid 
registrations for that user (for all the devices the user may 
have).


I guess your user has 2 registrations, so the 200 OK will 
report back both of them. You can check via "opensipsctl ul show"


Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 17.11.2016 12:13, Miha wrote:

Hello Bogdan

i changed this and it works in all cases, only in one I 
noticed today this (Opensips reply only in this case with 
two URI on contact):


 UAC:5060 ->OpenSIPS:5060
REGISTER sip:opsp.test.net:5060 SIP/2.0.
Via: SIP/2.0/UDP opsp.test.net;branch=z9hG4bKa40225bd7495297c6.
Max-Forwards: 70.
From: 042335040 ;tag=1f62205074.
To: 042335040 .
Cal

Re: [OpenSIPS-Users] nat issue

2016-11-18 Thread Miha

Hello bogdan

I guess, but it looks like so. Is it possible to remove it?


tnx
miha

On 18/11/2016 15:39, Bogdan-Andrei Iancu wrote:
I guess your UAC freezes when receiving back in the 200 OK REGISTER 
the "received" hdr param in Contact ??


Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 18.11.2016 16:33, Bogdan-Andrei Iancu wrote:

HI Miha,

Sorry, but I'm not able to follow the case you mentioned with 
Innovaphone PBX - maybe you can post (to see the differences) the 
sent and returned contact hdrs in the REGISTER request + reply for 
the 2 cases (OpenSIPS and Innovaphone PBX).


Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 18.11.2016 11:20, Miha wrote:
I do not know if this is the case. But from what I can see what I 
register it on some Innovaphone PBX, innovaphone sends back in 
contact (200 ok) just ip of IPBX and also when INVITE is send in 
contact there is URI of PBX and only and it works.


i tried this but did not have any luck.

br
miha

On 18/11/2016 09:48, Bogdan-Andrei Iancu wrote:

Hi Miha,

You mean the UAC does not like the multi-URI Contact header in the 
200 OK  If so, that UAC is really broken as 1) breaks the SIP 
syntax (which allows it) and 2) breaks the the SIP Registration as 
per RFC3261.


What about the second contact (the one already existing in usrloc 
when this registration comes) ? can it be discarded ? If YES, you 
can try passing the "c1f" flags to save() :

http://www.opensips.org/html/docs/modules/2.2.x/registrar.html#id294033

That will make OpenSIPS to accept only 1 contact per AOR/user and 
any new contact will override the existing one.


Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 18.11.2016 10:15, Miha wrote:

Hi Bogdan

I did few more test. This contact bothers UAC. Is there anything i 
can do in this case in OpenSIPS so that it will only reply with 
one URI in contact?


Contact:;expires=1518
;received="sip:84.41.125.21:5060",;
expires=180.


tnx so much!
MIha

On 17/11/2016 12:11, Bogdan-Andrei Iancu wrote:

Hi Miha,

yes, that is parallel forking (you may have more than 2 contacts 
only).


Are you sure your DB was sync'ed? OpenSIPS is periodically 
flushing the memory cache into the location table (see the 
"state" of the contact (as per "ul show") if marked as DIRTY).


In regards to RFC, I think you quote the wrong section (maybe 
about callings?) - for REGISTERs, any number of URIs are allowed 
AFAIK.


Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 17.11.2016 12:35, Miha wrote:

Bodan

so this is dual forking...?
So if you have one account and you have two phones on it and 
first will try  to register, 200 ok will will have contact of 
both phones?
In location table I can see only one registration for this user 
but for "opensipsctl ul show" it shows me two contacts, which is 
strange? (When i do trace only one invite is send) and UAC 
replay with Busy all the time due to two contacts (this what i 
have been told).


Ok, but if you look at rfc there is only one URI allowed in 
contact if I understand this right?



The Contact header field MUST be present and contain exactly one SIP
or SIPS URI in any request that can result in the establishment of a
dialog

Please correct me if I am wrong.


tnx so much!
Miha

On 17/11/2016 11:22, Bogdan-Andrei Iancu wrote:

Hi Miha,

OpenSIPS returns in the 200 OK for a REGISTER all the valid 
registrations for that user (for all the devices the user may 
have).


I guess your user has 2 registrations, so the 200 OK will 
report back both of them. You can check via "opensipsctl ul show"


Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 17.11.2016 12:13, Miha wrote:

Hello Bogdan

i changed this and it works in all cases, only in one I 
noticed today this (Opensips reply only in this case with two 
URI on contact):


 UAC:5060 ->OpenSIPS:5060
REGISTER sip:opsp.test.net:5060 SIP/2.0.
Via: SIP/2.0/UDP opsp.test.net;branch=z9hG4bKa40225bd7495297c6.
Max-Forwards: 70.
From: 042335040 ;tag=1f62205074.
To: 042335040 .
Call-ID: 61c67f739bef5a2e.
CSeq: 1804289391 REGISTER.
Allow:  INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE,
PRACK, INFO.
Authorization: Digest
username="99942335040",realm="opsp.test.net",nonce="582d810c58b
d73adccf0d455c2a2159b3a3403c1f7a3",uri="sip:opsp.test.net:5060",res
ponse="bc0c757c17f9b0976af35ec633dd83ca".
Contact: 042335040 
;ex

pires=3600.
Privacy: none.
Supported: path.
User-Agent: Brcm-Callctrl/v1.10.3 M5T SIP Stack/4.1.2.2.
Content-Length: 0.

UOpenSIPS:5060 -> UAC:5060
SIP/2.0 401 Unauthorized.
Via: SIP/2.0/UDP
opsp.test.net;received=UAC;rport=5060;branch=z9hG4bKa4022
5bd7495297c6.
F

Re: [OpenSIPS-Users] nat issue

2016-11-18 Thread Miha
I do not know if this is the case. But from what I can see what I 
register it on some Innovaphone PBX, innovaphone sends back in contact 
(200 ok) just ip of IPBX and also when INVITE is send in contact there 
is URI of PBX and only and it works.


i tried this but did not have any luck.

br
miha

On 18/11/2016 09:48, Bogdan-Andrei Iancu wrote:

Hi Miha,

You mean the UAC does not like the multi-URI Contact header in the 200 
OK  If so, that UAC is really broken as 1) breaks the SIP syntax 
(which allows it) and 2) breaks the the SIP Registration as per RFC3261.


What about the second contact (the one already existing in usrloc when 
this registration comes) ? can it be discarded ? If YES, you can try 
passing the "c1f" flags to save() :

http://www.opensips.org/html/docs/modules/2.2.x/registrar.html#id294033

That will make OpenSIPS to accept only 1 contact per AOR/user and any 
new contact will override the existing one.


Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 18.11.2016 10:15, Miha wrote:

Hi Bogdan

I did few more test. This contact bothers UAC. Is there anything i 
can do in this case in OpenSIPS so that it will only reply with one 
URI in contact?


Contact:;expires=1518
;received="sip:84.41.125.21:5060",;
expires=180.


tnx so much!
MIha

On 17/11/2016 12:11, Bogdan-Andrei Iancu wrote:

Hi Miha,

yes, that is parallel forking (you may have more than 2 contacts only).

Are you sure your DB was sync'ed? OpenSIPS is periodically flushing 
the memory cache into the location table (see the "state" of the 
contact (as per "ul show") if marked as DIRTY).


In regards to RFC, I think you quote the wrong section (maybe about 
callings?) - for REGISTERs, any number of URIs are allowed AFAIK.


Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 17.11.2016 12:35, Miha wrote:

Bodan

so this is dual forking...?
So if you have one account and you have two phones on it and first 
will try  to register, 200 ok will will have contact of both phones?
In location table I can see only one registration for this user but 
for "opensipsctl ul show" it shows me two contacts, which is 
strange? (When i do trace only one invite is send) and UAC replay 
with Busy all the time due to two contacts (this what i have been 
told).


Ok, but if you look at rfc there is only one URI allowed in contact 
if I understand this right?



The Contact header field MUST be present and contain exactly one SIP
or SIPS URI in any request that can result in the establishment of a
    dialog

Please correct me if I am wrong. tnx so much! Miha

On 17/11/2016 11:22, Bogdan-Andrei Iancu wrote:

Hi Miha,

OpenSIPS returns in the 200 OK for a REGISTER all the valid 
registrations for that user (for all the devices the user may have).


I guess your user has 2 registrations, so the 200 OK will report 
back both of them. You can check via "opensipsctl ul show"


Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 17.11.2016 12:13, Miha wrote:

Hello Bogdan

i changed this and it works in all cases, only in one I noticed 
today this (Opensips reply only in this case with two URI on 
contact):


 UAC:5060 ->OpenSIPS:5060
REGISTER sip:opsp.test.net:5060 SIP/2.0.
Via: SIP/2.0/UDP opsp.test.net;branch=z9hG4bKa40225bd7495297c6.
Max-Forwards: 70.
From: 042335040 ;tag=1f62205074.
To: 042335040 .
Call-ID: 61c67f739bef5a2e.
CSeq: 1804289391 REGISTER.
Allow:  INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE,
PRACK, INFO.
Authorization: Digest
username="99942335040",realm="opsp.test.net",nonce="582d810c58b
d73adccf0d455c2a2159b3a3403c1f7a3",uri="sip:opsp.test.net:5060",res
ponse="bc0c757c17f9b0976af35ec633dd83ca".
Contact: 042335040 ;ex
pires=3600.
Privacy: none.
Supported: path.
User-Agent: Brcm-Callctrl/v1.10.3 M5T SIP Stack/4.1.2.2.
Content-Length: 0.

UOpenSIPS:5060 -> UAC:5060
SIP/2.0 401 Unauthorized.
Via: SIP/2.0/UDP
opsp.test.net;received=UAC;rport=5060;branch=z9hG4bKa4022
5bd7495297c6.
From: 042335040 ;tag=1f62205074.
To: 042335040 ;tag=0c7ff67d927afc274
b272138ce65100a.ac4d.
Call-ID: 61c67f739bef5a2e.
CSeq: 1804289391 REGISTER.
WWW-Authenticate: Digest realm="opsp.test.net",
nonce="582d81135a88b92d0287a7460acce0a84e5d2a200b33", stale=true.
Server: OpenSIPS (1.10.0beta-tls (x86_64/linux)).
Content-Length: 0.


U UAC:5060 ->OpenSIPS:5060
REGISTER sip:opsp.test.net:5060 SIP/2.0.
Via: SIP/2.0/UDP opsp.test.net;branch=z9hG4bKb5f2bbbf80e346f48.
Max-Forwards: 70.
From: 042335040 ;tag=1f62205074.
To: 042335040 .
Call-ID: 61c67f739bef5a2e.
CSeq: 1804289392 REGISTER.
Allow:  INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE,
PRACK, INFO.
Authorization: Digest
username="99942335040",realm="opsp.test.net",nonce="582d8

Re: [OpenSIPS-Users] nat issue

2016-11-18 Thread Miha

Hi Bogdan

I did few more test. This contact bothers UAC. Is there anything i can 
do in this case in OpenSIPS so that it will only reply with one URI in 
contact?


Contact:;expires=1518
;received="sip:84.41.125.21:5060",;
expires=180.


tnx so much!
MIha


On 17/11/2016 12:11, Bogdan-Andrei Iancu wrote:

Hi Miha,

yes, that is parallel forking (you may have more than 2 contacts only).

Are you sure your DB was sync'ed? OpenSIPS is periodically flushing 
the memory cache into the location table (see the "state" of the 
contact (as per "ul show") if marked as DIRTY).


In regards to RFC, I think you quote the wrong section (maybe about 
callings?) - for REGISTERs, any number of URIs are allowed AFAIK.


Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 17.11.2016 12:35, Miha wrote:

Bodan

so this is dual forking...?
So if you have one account and you have two phones on it and first 
will try  to register, 200 ok will will have contact of both phones?
In location table I can see only one registration for this user but 
for "opensipsctl ul show" it shows me two contacts, which is strange? 
(When i do trace only one invite is send) and UAC replay with Busy 
all the time due to two contacts (this what i have been told).


Ok, but if you look at rfc there is only one URI allowed in contact 
if I understand this right?



The Contact header field MUST be present and contain exactly one SIP
or SIPS URI in any request that can result in the establishment of a
dialog

Please correct me if I am wrong. tnx so much! Miha

On 17/11/2016 11:22, Bogdan-Andrei Iancu wrote:

Hi Miha,

OpenSIPS returns in the 200 OK for a REGISTER all the valid 
registrations for that user (for all the devices the user may have).


I guess your user has 2 registrations, so the 200 OK will report 
back both of them. You can check via "opensipsctl ul show"


Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 17.11.2016 12:13, Miha wrote:

Hello Bogdan

i changed this and it works in all cases, only in one I noticed 
today this (Opensips reply only in this case with two URI on contact):


 UAC:5060 ->OpenSIPS:5060
REGISTER sip:opsp.test.net:5060 SIP/2.0.
Via: SIP/2.0/UDP opsp.test.net;branch=z9hG4bKa40225bd7495297c6.
Max-Forwards: 70.
From: 042335040 ;tag=1f62205074.
To: 042335040 .
Call-ID: 61c67f739bef5a2e.
CSeq: 1804289391 REGISTER.
Allow:  INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE,
PRACK, INFO.
Authorization: Digest
username="99942335040",realm="opsp.test.net",nonce="582d810c58b
d73adccf0d455c2a2159b3a3403c1f7a3",uri="sip:opsp.test.net:5060",res
ponse="bc0c757c17f9b0976af35ec633dd83ca".
Contact: 042335040 ;ex
pires=3600.
Privacy: none.
Supported: path.
User-Agent: Brcm-Callctrl/v1.10.3 M5T SIP Stack/4.1.2.2.
Content-Length: 0.

UOpenSIPS:5060 -> UAC:5060
SIP/2.0 401 Unauthorized.
Via: SIP/2.0/UDP
opsp.test.net;received=UAC;rport=5060;branch=z9hG4bKa4022
5bd7495297c6.
From: 042335040 ;tag=1f62205074.
To: 042335040 ;tag=0c7ff67d927afc274
b272138ce65100a.ac4d.
Call-ID: 61c67f739bef5a2e.
CSeq: 1804289391 REGISTER.
WWW-Authenticate: Digest realm="opsp.test.net",
nonce="582d81135a88b92d0287a7460acce0a84e5d2a200b33", stale=true.
Server: OpenSIPS (1.10.0beta-tls (x86_64/linux)).
Content-Length: 0.


U UAC:5060 ->OpenSIPS:5060
REGISTER sip:opsp.test.net:5060 SIP/2.0.
Via: SIP/2.0/UDP opsp.test.net;branch=z9hG4bKb5f2bbbf80e346f48.
Max-Forwards: 70.
From: 042335040 ;tag=1f62205074.
To: 042335040 .
Call-ID: 61c67f739bef5a2e.
CSeq: 1804289392 REGISTER.
Allow:  INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE,
PRACK, INFO.
Authorization: Digest
username="99942335040",realm="opsp.test.net",nonce="582d81135a8
8b92d0287a7460acce0a84e5d2a200b33",uri="sip:opsp.test.net:5060",res
ponse="9ce3622addeedf74622a23697e6f3728".
Contact: 042335040 ;ex
pires=3600.
Privacy: none.
Supported: path.
User-Agent: Brcm-Callctrl/v1.10.3 M5T SIP Stack/4.1.2.2.
Content-Length: 0.
.


UOpenSIPS:5060 -> UAC:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP
opsp.test.net;received=UAC;rport=5060;branch=z9hG4bKb5f2b
bbf80e346f48.
From: 042335040 ;tag=1f62205074.
To: 042335040 ;tag=766e4f757c55b3450
c9992a50fb64799-9163.
Call-ID: 61c67f739bef5a2e.
CSeq: 1804289392 REGISTER.
Contact: ;expires=3600
;received="sip:UAC:5060",  ;expires=119.
Server: OpenSIPS (1.10.0beta-tls (x86_64/linux)).
Content-Length: 0.

Do you see where could be an issue?


tnx
miha


On 16/11/2016 08:11, Miha wrote:

Hello Bogdan

yes this was the case...

thank you!


br
miha

On 15/11/2016 18:35, Bogdan-Andrei Iancu wrote:

Hi Miha,

When you handle REGISTER requests (from behind NAT) most probably 
you use fix_nated_contact() instead of fix_nated_register().


Regards,
Bogdan-Andrei Iancu

Re: [OpenSIPS-Users] nat issue

2016-11-17 Thread Miha

Hello Bogdan

how would I know that is marked as DIRTY? how this will look like?


tnx
miha

On 17/11/2016 12:11, Bogdan-Andrei Iancu wrote:

Hi Miha,

yes, that is parallel forking (you may have more than 2 contacts only).

Are you sure your DB was sync'ed? OpenSIPS is periodically flushing 
the memory cache into the location table (see the "state" of the 
contact (as per "ul show") if marked as DIRTY).


In regards to RFC, I think you quote the wrong section (maybe about 
callings?) - for REGISTERs, any number of URIs are allowed AFAIK.


Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 17.11.2016 12:35, Miha wrote:

Bodan

so this is dual forking...?
So if you have one account and you have two phones on it and first 
will try  to register, 200 ok will will have contact of both phones?
In location table I can see only one registration for this user but 
for "opensipsctl ul show" it shows me two contacts, which is strange? 
(When i do trace only one invite is send) and UAC replay with Busy 
all the time due to two contacts (this what i have been told).


Ok, but if you look at rfc there is only one URI allowed in contact 
if I understand this right?



The Contact header field MUST be present and contain exactly one SIP
or SIPS URI in any request that can result in the establishment of a
dialog

Please correct me if I am wrong. tnx so much! Miha

On 17/11/2016 11:22, Bogdan-Andrei Iancu wrote:

Hi Miha,

OpenSIPS returns in the 200 OK for a REGISTER all the valid 
registrations for that user (for all the devices the user may have).


I guess your user has 2 registrations, so the 200 OK will report 
back both of them. You can check via "opensipsctl ul show"


Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 17.11.2016 12:13, Miha wrote:

Hello Bogdan

i changed this and it works in all cases, only in one I noticed 
today this (Opensips reply only in this case with two URI on contact):


 UAC:5060 ->OpenSIPS:5060
REGISTER sip:opsp.test.net:5060 SIP/2.0.
Via: SIP/2.0/UDP opsp.test.net;branch=z9hG4bKa40225bd7495297c6.
Max-Forwards: 70.
From: 042335040 ;tag=1f62205074.
To: 042335040 .
Call-ID: 61c67f739bef5a2e.
CSeq: 1804289391 REGISTER.
Allow:  INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE,
PRACK, INFO.
Authorization: Digest
username="99942335040",realm="opsp.test.net",nonce="582d810c58b
d73adccf0d455c2a2159b3a3403c1f7a3",uri="sip:opsp.test.net:5060",res
ponse="bc0c757c17f9b0976af35ec633dd83ca".
Contact: 042335040 ;ex
pires=3600.
Privacy: none.
Supported: path.
User-Agent: Brcm-Callctrl/v1.10.3 M5T SIP Stack/4.1.2.2.
Content-Length: 0.

UOpenSIPS:5060 -> UAC:5060
SIP/2.0 401 Unauthorized.
Via: SIP/2.0/UDP
opsp.test.net;received=UAC;rport=5060;branch=z9hG4bKa4022
5bd7495297c6.
From: 042335040 ;tag=1f62205074.
To: 042335040 ;tag=0c7ff67d927afc274
b272138ce65100a.ac4d.
Call-ID: 61c67f739bef5a2e.
CSeq: 1804289391 REGISTER.
WWW-Authenticate: Digest realm="opsp.test.net",
nonce="582d81135a88b92d0287a7460acce0a84e5d2a200b33", stale=true.
Server: OpenSIPS (1.10.0beta-tls (x86_64/linux)).
Content-Length: 0.


U UAC:5060 ->OpenSIPS:5060
REGISTER sip:opsp.test.net:5060 SIP/2.0.
Via: SIP/2.0/UDP opsp.test.net;branch=z9hG4bKb5f2bbbf80e346f48.
Max-Forwards: 70.
From: 042335040 ;tag=1f62205074.
To: 042335040 .
Call-ID: 61c67f739bef5a2e.
CSeq: 1804289392 REGISTER.
Allow:  INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE,
PRACK, INFO.
Authorization: Digest
username="99942335040",realm="opsp.test.net",nonce="582d81135a8
8b92d0287a7460acce0a84e5d2a200b33",uri="sip:opsp.test.net:5060",res
ponse="9ce3622addeedf74622a23697e6f3728".
Contact: 042335040 ;ex
pires=3600.
Privacy: none.
Supported: path.
User-Agent: Brcm-Callctrl/v1.10.3 M5T SIP Stack/4.1.2.2.
Content-Length: 0.
.


UOpenSIPS:5060 -> UAC:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP
opsp.test.net;received=UAC;rport=5060;branch=z9hG4bKb5f2b
bbf80e346f48.
From: 042335040 ;tag=1f62205074.
To: 042335040 ;tag=766e4f757c55b3450
c9992a50fb64799-9163.
Call-ID: 61c67f739bef5a2e.
CSeq: 1804289392 REGISTER.
Contact: ;expires=3600
;received="sip:UAC:5060",  ;expires=119.
Server: OpenSIPS (1.10.0beta-tls (x86_64/linux)).
Content-Length: 0.

Do you see where could be an issue?


tnx
miha


On 16/11/2016 08:11, Miha wrote:

Hello Bogdan

yes this was the case...

thank you!


br
miha

On 15/11/2016 18:35, Bogdan-Andrei Iancu wrote:

Hi Miha,

When you handle REGISTER requests (from behind NAT) most probably 
you use fix_nated_contact() instead of fix_nated_register().


Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 15.11.2016 09:11, Miha wrote:

Hello

i need one info.
I have one phone behind NAT and it is registe

Re: [OpenSIPS-Users] nat issue

2016-11-17 Thread Miha

Bodan

so this is dual forking...?
So if you have one account and you have two phones on it and first will 
try  to register, 200 ok will will have contact of both phones?
In location table I can see only one registration for this user but for 
"opensipsctl ul show" it shows me two contacts, which is strange? (When 
i do trace only one invite is send) and UAC replay with Busy all the 
time due to two contacts (this what i have been told).


Ok, but if you look at rfc there is only one URI allowed in contact if I 
understand this right?



The Contact header field MUST be present and contain exactly one SIP
   or SIPS URI in any request that can result in the establishment of a
   dialog

Please correct me if I am wrong. tnx so much! Miha


On 17/11/2016 11:22, Bogdan-Andrei Iancu wrote:

Hi Miha,

OpenSIPS returns in the 200 OK for a REGISTER all the valid 
registrations for that user (for all the devices the user may have).


I guess your user has 2 registrations, so the 200 OK will report back 
both of them. You can check via "opensipsctl ul show"


Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 17.11.2016 12:13, Miha wrote:

Hello Bogdan

i changed this and it works in all cases, only in one I noticed today 
this (Opensips reply only in this case with two URI on contact):


 UAC:5060 ->OpenSIPS:5060
REGISTER sip:opsp.test.net:5060 SIP/2.0.
Via: SIP/2.0/UDP opsp.test.net;branch=z9hG4bKa40225bd7495297c6.
Max-Forwards: 70.
From: 042335040 ;tag=1f62205074.
To: 042335040 .
Call-ID: 61c67f739bef5a2e.
CSeq: 1804289391 REGISTER.
Allow:  INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE,
PRACK, INFO.
Authorization: Digest
username="99942335040",realm="opsp.test.net",nonce="582d810c58b
d73adccf0d455c2a2159b3a3403c1f7a3",uri="sip:opsp.test.net:5060",res
ponse="bc0c757c17f9b0976af35ec633dd83ca".
Contact: 042335040 ;ex
pires=3600.
Privacy: none.
Supported: path.
User-Agent: Brcm-Callctrl/v1.10.3 M5T SIP Stack/4.1.2.2.
Content-Length: 0.

UOpenSIPS:5060 -> UAC:5060
SIP/2.0 401 Unauthorized.
Via: SIP/2.0/UDP
opsp.test.net;received=UAC;rport=5060;branch=z9hG4bKa4022
5bd7495297c6.
From: 042335040 ;tag=1f62205074.
To: 042335040 ;tag=0c7ff67d927afc274
b272138ce65100a.ac4d.
Call-ID: 61c67f739bef5a2e.
CSeq: 1804289391 REGISTER.
WWW-Authenticate: Digest realm="opsp.test.net",
nonce="582d81135a88b92d0287a7460acce0a84e5d2a200b33", stale=true.
Server: OpenSIPS (1.10.0beta-tls (x86_64/linux)).
Content-Length: 0.


U UAC:5060 ->OpenSIPS:5060
REGISTER sip:opsp.test.net:5060 SIP/2.0.
Via: SIP/2.0/UDP opsp.test.net;branch=z9hG4bKb5f2bbbf80e346f48.
Max-Forwards: 70.
From: 042335040 ;tag=1f62205074.
To: 042335040 .
Call-ID: 61c67f739bef5a2e.
CSeq: 1804289392 REGISTER.
Allow:  INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE,
PRACK, INFO.
Authorization: Digest
username="99942335040",realm="opsp.test.net",nonce="582d81135a8
8b92d0287a7460acce0a84e5d2a200b33",uri="sip:opsp.test.net:5060",res
ponse="9ce3622addeedf74622a23697e6f3728".
Contact: 042335040 ;ex
pires=3600.
Privacy: none.
Supported: path.
User-Agent: Brcm-Callctrl/v1.10.3 M5T SIP Stack/4.1.2.2.
Content-Length: 0.
.


UOpenSIPS:5060 -> UAC:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP
opsp.test.net;received=UAC;rport=5060;branch=z9hG4bKb5f2b
bbf80e346f48.
From: 042335040 ;tag=1f62205074.
To: 042335040 ;tag=766e4f757c55b3450
c9992a50fb64799-9163.
Call-ID: 61c67f739bef5a2e.
CSeq: 1804289392 REGISTER.
Contact: ;expires=3600
;received="sip:UAC:5060",  ;expires=119.
Server: OpenSIPS (1.10.0beta-tls (x86_64/linux)).
Content-Length: 0.

Do you see where could be an issue?


tnx
miha


On 16/11/2016 08:11, Miha wrote:

Hello Bogdan

yes this was the case...

thank you!


br
miha

On 15/11/2016 18:35, Bogdan-Andrei Iancu wrote:

Hi Miha,

When you handle REGISTER requests (from behind NAT) most probably 
you use fix_nated_contact() instead of fix_nated_register().


Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 15.11.2016 09:11, Miha wrote:

Hello

i need one info.
I have one phone behind NAT and it is registered on OpenSIPS. IN 
register packet, which is send to OpenSIPS I can see contact: 
"sip:11181600519@192.168.0.101:5060;transport=UDP"


and let says that the public ip for this device is xxx.xxx.xxx.xxx.


When opensips sends INVITE it send to right public ip and right 
port (source ip and source port generated by router). The issue is 
this:
Invite is like:  
"sip:11181600...@xxx.xxx.xxx.xxx:5060;transport=UDP" and this 
request is then fw to this UAC behind router. The UAC replays to 
this INVITE with 404 Not found as it is waiting to receive the 
same URI which was written in contact (the userpart is ok, put the 
ip is public, not private and this is the issue).From what I can 
s

Re: [OpenSIPS-Users] nat issue

2016-11-17 Thread Miha

Hello Bogdan

i changed this and it works in all cases, only in one I noticed today 
this (Opensips reply only in this case with two URI on contact):


 UAC:5060 ->OpenSIPS:5060
REGISTER sip:opsp.test.net:5060 SIP/2.0.
Via: SIP/2.0/UDP opsp.test.net;branch=z9hG4bKa40225bd7495297c6.
Max-Forwards: 70.
From: 042335040 ;tag=1f62205074.
To: 042335040 .
Call-ID: 61c67f739bef5a2e.
CSeq: 1804289391 REGISTER.
Allow:  INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE,
PRACK, INFO.
Authorization: Digest
username="99942335040",realm="opsp.test.net",nonce="582d810c58b
d73adccf0d455c2a2159b3a3403c1f7a3",uri="sip:opsp.test.net:5060",res
ponse="bc0c757c17f9b0976af35ec633dd83ca".
Contact: 042335040 ;ex
pires=3600.
Privacy: none.
Supported: path.
User-Agent: Brcm-Callctrl/v1.10.3 M5T SIP Stack/4.1.2.2.
Content-Length: 0.

UOpenSIPS:5060 -> UAC:5060
SIP/2.0 401 Unauthorized.
Via: SIP/2.0/UDP
opsp.test.net;received=UAC;rport=5060;branch=z9hG4bKa4022
5bd7495297c6.
From: 042335040 ;tag=1f62205074.
To: 042335040 ;tag=0c7ff67d927afc274
b272138ce65100a.ac4d.
Call-ID: 61c67f739bef5a2e.
CSeq: 1804289391 REGISTER.
WWW-Authenticate: Digest realm="opsp.test.net",
nonce="582d81135a88b92d0287a7460acce0a84e5d2a200b33", stale=true.
Server: OpenSIPS (1.10.0beta-tls (x86_64/linux)).
Content-Length: 0.


U UAC:5060 ->OpenSIPS:5060
REGISTER sip:opsp.test.net:5060 SIP/2.0.
Via: SIP/2.0/UDP opsp.test.net;branch=z9hG4bKb5f2bbbf80e346f48.
Max-Forwards: 70.
From: 042335040 ;tag=1f62205074.
To: 042335040 .
Call-ID: 61c67f739bef5a2e.
CSeq: 1804289392 REGISTER.
Allow:  INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE,
PRACK, INFO.
Authorization: Digest
username="99942335040",realm="opsp.test.net",nonce="582d81135a8
8b92d0287a7460acce0a84e5d2a200b33",uri="sip:opsp.test.net:5060",res
ponse="9ce3622addeedf74622a23697e6f3728".
Contact: 042335040 ;ex
pires=3600.
Privacy: none.
Supported: path.
User-Agent: Brcm-Callctrl/v1.10.3 M5T SIP Stack/4.1.2.2.
Content-Length: 0.
.


UOpenSIPS:5060 -> UAC:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP
opsp.test.net;received=UAC;rport=5060;branch=z9hG4bKb5f2b
bbf80e346f48.
From: 042335040 ;tag=1f62205074.
To: 042335040 ;tag=766e4f757c55b3450
c9992a50fb64799-9163.
Call-ID: 61c67f739bef5a2e.
CSeq: 1804289392 REGISTER.
Contact: ;expires=3600
;received="sip:UAC:5060",  ;expires=119.
Server: OpenSIPS (1.10.0beta-tls (x86_64/linux)).
Content-Length: 0.

Do you see where could be an issue?


tnx
miha


On 16/11/2016 08:11, Miha wrote:

Hello Bogdan

yes this was the case...

thank you!


br
miha

On 15/11/2016 18:35, Bogdan-Andrei Iancu wrote:

Hi Miha,

When you handle REGISTER requests (from behind NAT) most probably you 
use fix_nated_contact() instead of fix_nated_register().


Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 15.11.2016 09:11, Miha wrote:

Hello

i need one info.
I have one phone behind NAT and it is registered on OpenSIPS. IN 
register packet, which is send to OpenSIPS I can see contact: 
"sip:11181600519@192.168.0.101:5060;transport=UDP"


and let says that the public ip for this device is xxx.xxx.xxx.xxx.


When opensips sends INVITE it send to right public ip and right port 
(source ip and source port generated by router). The issue is this:
Invite is like:  
"sip:11181600...@xxx.xxx.xxx.xxx:5060;transport=UDP" and this 
request is then fw to this UAC behind router. The UAC replays to 
this INVITE with 404 Not found as it is waiting to receive the same 
URI which was written in contact (the userpart is ok, put the ip is 
public, not private and this is the issue).From what I can see in 
RFC this is the case.



Till now Idid not have any issues with this, but now I found first 
phone which replays with 404 and from RFC point of view there should 
be private ip request :) . So is there anything I can do :)?



tnx
miha


___
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] nat issue

2016-11-15 Thread Miha

Hello Bogdan

yes this was the case...

thank you!


br
miha

On 15/11/2016 18:35, Bogdan-Andrei Iancu wrote:

Hi Miha,

When you handle REGISTER requests (from behind NAT) most probably you 
use fix_nated_contact() instead of fix_nated_register().


Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 15.11.2016 09:11, Miha wrote:

Hello

i need one info.
I have one phone behind NAT and it is registered on OpenSIPS. IN 
register packet, which is send to OpenSIPS I can see contact: 
"sip:11181600519@192.168.0.101:5060;transport=UDP"


and let says that the public ip for this device is xxx.xxx.xxx.xxx.


When opensips sends INVITE it send to right public ip and right port 
(source ip and source port generated by router). The issue is this:
Invite is like:  "sip:11181600...@xxx.xxx.xxx.xxx:5060;transport=UDP" 
and this request is then fw to this UAC behind router. The UAC 
replays to this INVITE with 404 Not found as it is waiting to receive 
the same URI which was written in contact (the userpart is ok, put 
the ip is public, not private and this is the issue).From what I can 
see in RFC this is the case.



Till now Idid not have any issues with this, but now I found first 
phone which replays with 404 and from RFC point of view there should 
be private ip request :) . So is there anything I can do :)?



tnx
miha


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




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


[OpenSIPS-Users] nat issue

2016-11-14 Thread Miha

Hello

i need one info.
I have one phone behind NAT and it is registered on OpenSIPS. IN 
register packet, which is send to OpenSIPS I can see contact: 
"sip:11181600519@192.168.0.101:5060;transport=UDP"


and let says that the public ip for this device is xxx.xxx.xxx.xxx.


When opensips sends INVITE it send to right public ip and right port 
(source ip and source port generated by router). The issue is this:
Invite is like:  "sip:11181600...@xxx.xxx.xxx.xxx:5060;transport=UDP" 
and this request is then fw to this UAC behind router. The UAC replays 
to this INVITE with 404 Not found as it is waiting to receive the same 
URI which was written in contact (the userpart is ok, put the ip is 
public, not private and this is the issue).From what I can see in RFC 
this is the case.



Till now Idid not have any issues with this, but now I found first phone 
which replays with 404 and from RFC point of view there should be 
private ip request :) . So is there anything I can do :)?



tnx
miha
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Re: [OpenSIPS-Users] Help with big amount of data for routing

2016-10-12 Thread Miha

ok tnx so much Bogdan!

br
miha
On 12/10/2016 14:39, Bogdan-Andrei Iancu wrote:
Give it a try. Just be sure you properly adjust the shared memory and 
note that the loading will take a bit of a time.


Regards,

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

On 12.10.2016 15:37, Miha wrote:
this is great! tnx :) I was thinking that but as it was so much data 
i did not even want to try it :)


I will try and let you know!


br

miha



On 12/10/2016 14:33, Bogdan-Andrei Iancu wrote:
Why don't you use DR for that translation. Make a routing group 
where you put all DIDs (as prefixes) in dr_rules and have the NET_ID 
as attribute for the rule. And when looking it up:

do_routing("group","LC");
See :
http://www.opensips.org/html/docs/modules/2.2.x/drouting.html#id295067

DR is more memory efficient and much much faster in the lookup.

Regards,

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

On 12.10.2016 15:30, Miha wrote:
yes. I have in redis like: KEY (DID) and VALUE (NETID), than I am 
doing lookup in opensips script.


 I was looking what is the most a appropriate way to do this. As 
redis is quite good in this cases I choose it but the issue is 
memory :(



tnx

miha


On 12/10/2016 14:25, Bogdan-Andrei Iancu wrote:
So basically you need to determine the NET_ID based on the DID 
number ? this is what you do in REDIS now ?


Regards,

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

On 12.10.2016 15:09, Miha wrote:

Hi Bogdan

i missed your email, sorry...

Operator need to have number like NET_ID + DID. So routing is 
based on NET_ID. I manage to get all number in redis (quite a big 
amount of memory is used :) ).So first I do lookup in redis, get 
NET_ID and then I am using d_routing based od NET_IDs.


br
miha
On 05/10/2016 15:17, Bogdan-Andrei Iancu wrote:

Hi Miha,

so you have 120M records (NET_ID + DID) - how do you use them 
from OpenSIPS ? As I fail to understand what are the operations 
you want to perform over this data.


Regards,

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

On 05.10.2016 09:44, Miha wrote:

HI

the is not really opensips issue:) I need somehow to store big 
amount of data for routing.


To every telephone operator I must send RURI like 
Net_ID+Telephone_number (value indicates to who number belongs 
to). In this country they have around 120 millions of numbers.


After i have all NET_IDs with numbers I will use drouting for 
routing numbers to right operator based on NET_ID.


Here is the issue:
- I tried this with redis (lookup must be quick) but this takes 
so much memory that basically redis brakes everytime in between 
50 millions and 70 millions entries

- I tried with hash (hset) in redis but did not do any good


Do you have any suggestion how to deal with this, what would be 
the best thing to use?




tnx

miha


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Re: [OpenSIPS-Users] Help with big amount of data for routing

2016-10-12 Thread Miha
this is great! tnx :) I was thinking that but as it was so much data i 
did not even want to try it :)


I will try and let you know!


br

miha



On 12/10/2016 14:33, Bogdan-Andrei Iancu wrote:
Why don't you use DR for that translation. Make a routing group where 
you put all DIDs (as prefixes) in dr_rules and have the NET_ID as 
attribute for the rule. And when looking it up:

do_routing("group","LC");
See :
http://www.opensips.org/html/docs/modules/2.2.x/drouting.html#id295067

DR is more memory efficient and much much faster in the lookup.

Regards,

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

On 12.10.2016 15:30, Miha wrote:
yes. I have in redis like: KEY (DID) and VALUE (NETID), than I am 
doing lookup in opensips script.


 I was looking what is the most a appropriate way to do this. As 
redis is quite good in this cases I choose it but the issue is memory :(



tnx

miha


On 12/10/2016 14:25, Bogdan-Andrei Iancu wrote:
So basically you need to determine the NET_ID based on the DID 
number ? this is what you do in REDIS now ?


Regards,

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

On 12.10.2016 15:09, Miha wrote:

Hi Bogdan

i missed your email, sorry...

Operator need to have number like NET_ID + DID. So routing is based 
on NET_ID. I manage to get all number in redis (quite a big amount 
of memory is used :) ).So first I do lookup in redis, get NET_ID 
and then I am using d_routing based od NET_IDs.


br
miha
On 05/10/2016 15:17, Bogdan-Andrei Iancu wrote:

Hi Miha,

so you have 120M records (NET_ID + DID) - how do you use them from 
OpenSIPS ? As I fail to understand what are the operations you 
want to perform over this data.


Regards,

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

On 05.10.2016 09:44, Miha wrote:

HI

the is not really opensips issue:) I need somehow to store big 
amount of data for routing.


To every telephone operator I must send RURI like 
Net_ID+Telephone_number (value indicates to who number belongs 
to). In this country they have around 120 millions of numbers.


After i have all NET_IDs with numbers I will use drouting for 
routing numbers to right operator based on NET_ID.


Here is the issue:
- I tried this with redis (lookup must be quick) but this takes 
so much memory that basically redis brakes everytime in between 
50 millions and 70 millions entries

- I tried with hash (hset) in redis but did not do any good


Do you have any suggestion how to deal with this, what would be 
the best thing to use?




tnx

miha


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Re: [OpenSIPS-Users] Help with big amount of data for routing

2016-10-12 Thread Miha
yes. I have in redis like: KEY (DID) and VALUE (NETID), than I am doing 
lookup in opensips script.


 I was looking what is the most a appropriate way to do this. As redis 
is quite good in this cases I choose it but the issue is memory :(



tnx

miha


On 12/10/2016 14:25, Bogdan-Andrei Iancu wrote:
So basically you need to determine the NET_ID based on the DID number 
? this is what you do in REDIS now ?


Regards,

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

On 12.10.2016 15:09, Miha wrote:

Hi Bogdan

i missed your email, sorry...

Operator need to have number like NET_ID + DID. So routing is based 
on NET_ID. I manage to get all number in redis (quite a big amount of 
memory is used :) ).So first I do lookup in redis, get NET_ID and 
then I am using d_routing based od NET_IDs.


br
miha
On 05/10/2016 15:17, Bogdan-Andrei Iancu wrote:

Hi Miha,

so you have 120M records (NET_ID + DID) - how do you use them from 
OpenSIPS ? As I fail to understand what are the operations you want 
to perform over this data.


Regards,

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

On 05.10.2016 09:44, Miha wrote:

HI

the is not really opensips issue:) I need somehow to store big 
amount of data for routing.


To every telephone operator I must send RURI like 
Net_ID+Telephone_number (value indicates to who number belongs to). 
In this country they have around 120 millions of numbers.


After i have all NET_IDs with numbers I will use drouting for 
routing numbers to right operator based on NET_ID.


Here is the issue:
- I tried this with redis (lookup must be quick) but this takes so 
much memory that basically redis brakes everytime in between 50 
millions and 70 millions entries

- I tried with hash (hset) in redis but did not do any good


Do you have any suggestion how to deal with this, what would be the 
best thing to use?




tnx

miha


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Re: [OpenSIPS-Users] Help with big amount of data for routing

2016-10-12 Thread Miha

Hi Bogdan

i missed your email, sorry...

Operator need to have number like NET_ID + DID. So routing is based on 
NET_ID. I manage to get all number in redis (quite a big amount of 
memory is used :) ).So first I do lookup in redis, get NET_ID and then I 
am using d_routing based od NET_IDs.


br
miha
On 05/10/2016 15:17, Bogdan-Andrei Iancu wrote:

Hi Miha,

so you have 120M records (NET_ID + DID) - how do you use them from 
OpenSIPS ? As I fail to understand what are the operations you want to 
perform over this data.


Regards,

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

On 05.10.2016 09:44, Miha wrote:

HI

the is not really opensips issue:) I need somehow to store big amount 
of data for routing.


To every telephone operator I must send RURI like 
Net_ID+Telephone_number (value indicates to who number belongs to). 
In this country they have around 120 millions of numbers.


After i have all NET_IDs with numbers I will use drouting for routing 
numbers to right operator based on NET_ID.


Here is the issue:
- I tried this with redis (lookup must be quick) but this takes so 
much memory that basically redis brakes everytime in between 50 
millions and 70 millions entries

- I tried with hash (hset) in redis but did not do any good


Do you have any suggestion how to deal with this, what would be the 
best thing to use?




tnx

miha


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Re: [OpenSIPS-Users] Help with big amount of data for routing

2016-10-05 Thread Miha

Tnx Daniel for this!

I will try.


br

miha


On 05/10/2016 15:09, Daniel Zanutti wrote:

Hi Miha

I have a similar situation, but around 20 M routes.

The native routing mecanims wasn't performing well, so I developed a 
custom mecanism using Opensips scripting. Everything is stored on 
MySQL database.


The best approach was use avp_db_query to get the route, the primary 
key (and index) of the table is the route prefix and stored as BIG 
INT, so you have up to 19 digits of routes, which is OK to me. I could 
achieve more than 100 cps with this method.


You have to find the longest route "by hand", so I developed this 
procedure:


DELIMITER $$

CREATE DEFINER=`root`@`localhost` PROCEDURE `getLongestRoute`(IN route 
VARCHAR(50), OUT bestroute BIGINT, OUT regionid INT)

BEGIN
DECLARE rotatemp VARCHAR(50);
DECLARE tempprefix BIGINT;
CREATE TEMPORARY TABLE IF NOT EXISTS temptabrotas ( prefix BIGINT 
UNSIGNED) ENGINE=HEAP;

SET rotatemp = SUBSTRING(route, 1, LENGTH(route));
INSERT INTO temptabrotas (prefix) VALUES (rotatemp);
WHILE (LENGTH(rotatemp) > 1) DO
 SET rotatemp = SUBSTRING(route, 1, LENGTH(rotatemp)-1);
 INSERT INTO temptabrotas (prefix) VALUES (rotatemp);
END WHILE;
SELECT routes.prefix, routes.regionid FROM routes
 INNER JOIN temptabrotas
   ON routes.prefix = temptabrotas.prefix
ORDER BY routes.prefix DESC
LIMIT 1
INTO bestroute, regionid;
DROP TABLE temptabrotas;
END$$

DELIMITER ;

Hope it helps.

Regards


On Wed, Oct 5, 2016 at 4:16 AM, Miha <mailto:m...@softnet.si>> wrote:


Hi Alex

i tried, but mysql takes so long time for every select. What do u
have in mind?


tnx

miha



On 05/10/2016 08:46, Alex Balashov wrote:

Why do you believe that using a traditional RDBM necessarily
means slow lookups?

    On 10/05/2016 02:44 AM, Miha wrote:

HI

the is not really opensips issue:) I need somehow to store
big amount of
data for routing.

To every telephone operator I must send RURI like
Net_ID+Telephone_number (value indicates to who number
belongs to). In
this country they have around 120 millions of numbers.

After i have all NET_IDs with numbers I will use drouting
for routing
numbers to right operator based on NET_ID.

Here is the issue:
- I tried this with redis (lookup must be quick) but this
takes so much
memory that basically redis brakes everytime in between 50
millions and
70 millions entries
- I tried with hash (hset) in redis but did not do any good


Do you have any suggestion how to deal with this, what
would be the best
thing to use?



    tnx

miha


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Re: [OpenSIPS-Users] Help with big amount of data for routing

2016-10-05 Thread Miha

Hi Alex

i tried, but mysql takes so long time for every select. What do u have 
in mind?



tnx

miha


On 05/10/2016 08:46, Alex Balashov wrote:
Why do you believe that using a traditional RDBM necessarily means 
slow lookups?


On 10/05/2016 02:44 AM, Miha wrote:


HI

the is not really opensips issue:) I need somehow to store big amount of
data for routing.

To every telephone operator I must send RURI like
Net_ID+Telephone_number (value indicates to who number belongs to). In
this country they have around 120 millions of numbers.

After i have all NET_IDs with numbers I will use drouting for routing
numbers to right operator based on NET_ID.

Here is the issue:
- I tried this with redis (lookup must be quick) but this takes so much
memory that basically redis brakes everytime in between 50 millions and
70 millions entries
- I tried with hash (hset) in redis but did not do any good


Do you have any suggestion how to deal with this, what would be the best
thing to use?



tnx

miha


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[OpenSIPS-Users] Help with big amount of data for routing

2016-10-04 Thread Miha

HI

the is not really opensips issue:) I need somehow to store big amount of 
data for routing.


To every telephone operator I must send RURI like 
Net_ID+Telephone_number (value indicates to who number belongs to). In 
this country they have around 120 millions of numbers.


After i have all NET_IDs with numbers I will use drouting for routing 
numbers to right operator based on NET_ID.


Here is the issue:
- I tried this with redis (lookup must be quick) but this takes so much 
memory that basically redis brakes everytime in between 50 millions and 
70 millions entries

- I tried with hash (hset) in redis but did not do any good


Do you have any suggestion how to deal with this, what would be the best 
thing to use?




tnx

miha


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Re: [OpenSIPS-Users] PRACK, 404 not here

2016-09-15 Thread Miha

Hi Razvan

i was on trip... so sorry for my late response.

I tried with match_dialog() but then I got loop.

59.878504 SBC_2 -> SBC_1 SIP 524 Request: PRACK 
sip:38111422@SBC_1:5060;transport=udp |
 59.878742 SBC_1 -> SBC_1 SIP 530 Request: PRACK 
sip:SBC_1;lr;ftag=1875648965;did=b4a.460f9d7 |
 59.878959 SBC_1 -> SBC_1 SIP 596 Request: PRACK 
sip:SBC_1;lr;ftag=1875648965;did=b4a.460f9d7 |
 59.879128 SBC_1 -> SBC_1 SIP 662 Request: PRACK 
sip:SBC_1;lr;ftag=1875648965;did=b4a.460f9d7 |
 59.879299 SBC_1 -> SBC_1 SIP 728 Request: PRACK 
sip:SBC_1;lr;ftag=1875648965;did=b4a.460f9d7 |
 59.879892 SBC_1 -> SBC_1 SIP 794 Request: PRACK 
sip:SBC_1;lr;ftag=1875648965;did=b4a.460f9d7 |
 59.880214 SBC_1 -> SBC_1 SIP 860 Request: PRACK 
sip:SBC_1;lr;ftag=1875648965;did=b4a.460f9d7 |
 59.880417 SBC_1 -> SBC_1 SIP 926 Request: PRACK 
sip:SBC_1;lr;ftag=1875648965;did=b4a.460f9d7 |
 59.880945 SBC_1 -> SBC_1 SIP 992 Request: PRACK 
sip:SBC_1;lr;ftag=1875648965;did=b4a.460f9d7 |


Yes in my script I have:  listen=udp:SBC_1:5060

Do you have any other recommendation what I should try?


tnx
miha
On 07/09/2016 10:12, Răzvan Crainea wrote:

Hi, Miha!

It looks like loose_route() fails - did you try to look into the logs 
and see if it indicates something? Is the SBC_1 IP advertised in the 
Route header a listener of OpenSIPS?


Also, if loose_route() fails, you should still try to match the PRACK 
against the dialog. So your scripting logic should look like this:


if (has_totag()) {
if (loose_route() || match_dialog()) {
...
}
}

Let us know how that goes.

Best regards,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com
On 09/07/2016 10:38 AM, Miha wrote:

Hi

i have one issue and do not know how to solve it...

Initial invite:

U SBC_2:5060 -> SBC_1:5060 INVITE 
sip:7422@SBC_1:5060;user=phone SIP/2.0. Via: SIP/2.0/UDP 
SBC_2:5060;branch=z9hG4bK57fa.67ccbb16.0. From: 
;tag=*1875283502*. To: 
. Call-ID: 
*fb9e258ae909d311a85a0090332e03ed@PBX.* CSeq: 1193456 INVITE. 
Contact: . Alert-Info: 
. Allow: 
REGISTER,SUBSCRIBE,NOTIFY,INVITE,ACK,P
SeqU PBX:5060 -> SBC_2:5060 PRACK sip:SBC_2;did=8d9.43418513 SIP/2.0. 
Via: SIP/2.0/UDP PBX:5060;branch=z9hG4bK-002AF6E3;rport. From: 
;tag=*1875283502*. To: 
;tag=*FamBBcayZeKgF*. 
Call-ID:*fb9e258ae909d311a85a0090332e03ed@PBX.* CSeq: 1193457 PRACK. 
Content-Length: 0. Max-Forwards: 70. RAck: 1601153264 1193456 INVITE. 
. Seq U SBC_2:5060 -> SBC_1:5060 PRACK 
sip:7422@SBC_1:5060;transport=udp SIP/2.0. Route: 
. Via: SIP/2.0/UDP 
SBC_2:5060;branch=z9hG4bK67fa.e6a41de2.0. From: 
;tag=*1875283502*. To: 
;tag=*FamBBcayZeKgF*. Call-ID: 
*fb9e258ae909d311a85a0090332e03ed@PBX.* CSeq: 1193457 PRACK. 
Content-Length: 0. Max-Forwards: 69. RAck: 1601153264 1193456 INVITE.


Seq
U SBC_1:5060 -> SBC_2:5060 SIP/2.0 404 Not here. Via: SIP/2.0/UDP 
SBC_2:5060;branch=z9hG4bK67fa.e6a41de2.0. From: 
;tag=1875283502. To: 
;tag=FamBBcayZeKgF. Call-ID: 
fb9e258ae909d311a85a0090332e03ed@PBX. CSeq: 1193457 PRACK. Server: 
OpenSIPS (2.1.1 (x86_64/linux)). Content-Length: 0.


Why I am getting 404 from Opensips. is should be routed like seq request, right?





if (has_totag()) { # sequential requests within a dialog should # 
take the path determined by record-routing if (loose_route()) { 
xlog("loose_route"); #if ($DLG_status!=NULL) xlog("dlg_status"); if 
(!validate_dialog()){ fix_route_dialog(); xlog("fix_route_dialog"); } 
if (is_method("BYE")) { setflag(1); # do accounting ... 
#setflag(ACC_FAILED); # ... even if the transaction fails } else if 
(is_method("INVITE")) { # even if in most of the cases is useless, do 
RR for # re-INVITEs alos, as some buggy clients do change route set # 
during the dialog. record_route(); xlog("check_fraud"); } # route it 
out to whatever destination was set by loose_route() # in $du 
(destination URI). route(relay); } else { if ( is_method("ACK") ) { 
if ( t_check_trans() ) { # non loose-route, but stateful ACK; must be 
an ACK after # a 487 or e.g. 404 from upstream server t_relay(); 
exit; } else { # ACK without matching transaction -> # ignore and 
discard exit; } } sl_send_reply("404","Not here"); } exit; tnx miha



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Re: [OpenSIPS-Users] PRACK, 404 not here

2016-09-07 Thread Miha

Hi Razvan

will take a look and let you know.

I tried with match_dialog() but then got some loop.


tnx

miha


On 07/09/2016 10:12, Răzvan Crainea wrote:

Hi, Miha!

It looks like loose_route() fails - did you try to look into the logs 
and see if it indicates something? Is the SBC_1 IP advertised in the 
Route header a listener of OpenSIPS?


Also, if loose_route() fails, you should still try to match the PRACK 
against the dialog. So your scripting logic should look like this:


if (has_totag()) {
if (loose_route() || match_dialog()) {
...
}
}

Let us know how that goes.

Best regards,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com
On 09/07/2016 10:38 AM, Miha wrote:

Hi

i have one issue and do not know how to solve it...

Initial invite:

U SBC_2:5060 -> SBC_1:5060 INVITE 
sip:7422@SBC_1:5060;user=phone SIP/2.0. Via: SIP/2.0/UDP 
SBC_2:5060;branch=z9hG4bK57fa.67ccbb16.0. From: 
;tag=*1875283502*. To: 
. Call-ID: 
*fb9e258ae909d311a85a0090332e03ed@PBX.* CSeq: 1193456 INVITE. 
Contact: . Alert-Info: 
. Allow: 
REGISTER,SUBSCRIBE,NOTIFY,INVITE,ACK,P
SeqU PBX:5060 -> SBC_2:5060 PRACK sip:SBC_2;did=8d9.43418513 SIP/2.0. 
Via: SIP/2.0/UDP PBX:5060;branch=z9hG4bK-002AF6E3;rport. From: 
;tag=*1875283502*. To: 
;tag=*FamBBcayZeKgF*. 
Call-ID:*fb9e258ae909d311a85a0090332e03ed@PBX.* CSeq: 1193457 PRACK. 
Content-Length: 0. Max-Forwards: 70. RAck: 1601153264 1193456 INVITE. 
. Seq U SBC_2:5060 -> SBC_1:5060 PRACK 
sip:7422@SBC_1:5060;transport=udp SIP/2.0. Route: 
. Via: SIP/2.0/UDP 
SBC_2:5060;branch=z9hG4bK67fa.e6a41de2.0. From: 
;tag=*1875283502*. To: 
;tag=*FamBBcayZeKgF*. Call-ID: 
*fb9e258ae909d311a85a0090332e03ed@PBX.* CSeq: 1193457 PRACK. 
Content-Length: 0. Max-Forwards: 69. RAck: 1601153264 1193456 INVITE.


Seq
U SBC_1:5060 -> SBC_2:5060 SIP/2.0 404 Not here. Via: SIP/2.0/UDP 
SBC_2:5060;branch=z9hG4bK67fa.e6a41de2.0. From: 
;tag=1875283502. To: 
;tag=FamBBcayZeKgF. Call-ID: 
fb9e258ae909d311a85a0090332e03ed@PBX. CSeq: 1193457 PRACK. Server: 
OpenSIPS (2.1.1 (x86_64/linux)). Content-Length: 0.


Why I am getting 404 from Opensips. is should be routed like seq request, right?





if (has_totag()) { # sequential requests within a dialog should # 
take the path determined by record-routing if (loose_route()) { 
xlog("loose_route"); #if ($DLG_status!=NULL) xlog("dlg_status"); if 
(!validate_dialog()){ fix_route_dialog(); xlog("fix_route_dialog"); } 
if (is_method("BYE")) { setflag(1); # do accounting ... 
#setflag(ACC_FAILED); # ... even if the transaction fails } else if 
(is_method("INVITE")) { # even if in most of the cases is useless, do 
RR for # re-INVITEs alos, as some buggy clients do change route set # 
during the dialog. record_route(); xlog("check_fraud"); } # route it 
out to whatever destination was set by loose_route() # in $du 
(destination URI). route(relay); } else { if ( is_method("ACK") ) { 
if ( t_check_trans() ) { # non loose-route, but stateful ACK; must be 
an ACK after # a 487 or e.g. 404 from upstream server t_relay(); 
exit; } else { # ACK without matching transaction -> # ignore and 
discard exit; } } sl_send_reply("404","Not here"); } exit; tnx miha



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[OpenSIPS-Users] PRACK, 404 not here

2016-09-07 Thread Miha

Hi

i have one issue and do not know how to solve it...

Initial invite:

U SBC_2:5060 -> SBC_1:5060 INVITE sip:7422@SBC_1:5060;user=phone 
SIP/2.0. Via: SIP/2.0/UDP SBC_2:5060;branch=z9hG4bK57fa.67ccbb16.0. 
From: ;tag=*1875283502*. To: 
. Call-ID: 
*fb9e258ae909d311a85a0090332e03ed@PBX.* CSeq: 1193456 INVITE. Contact: 
. Alert-Info: . 
Allow: REGISTER,SUBSCRIBE,NOTIFY,INVITE,ACK,P
SeqU PBX:5060 -> SBC_2:5060 PRACK sip:SBC_2;did=8d9.43418513 SIP/2.0. Via: 
SIP/2.0/UDP PBX:5060;branch=z9hG4bK-002AF6E3;rport. From: 
;tag=*1875283502*. To: 
;tag=*FamBBcayZeKgF*. 
Call-ID:*fb9e258ae909d311a85a0090332e03ed@PBX.* CSeq: 1193457 PRACK. 
Content-Length: 0. Max-Forwards: 70. RAck: 1601153264 1193456 INVITE. . 
Seq U SBC_2:5060 -> SBC_1:5060 PRACK 
sip:7422@SBC_1:5060;transport=udp SIP/2.0. Route: 
. Via: SIP/2.0/UDP 
SBC_2:5060;branch=z9hG4bK67fa.e6a41de2.0. From: 
;tag=*1875283502*. To: 
;tag=*FamBBcayZeKgF*. Call-ID: 
*fb9e258ae909d311a85a0090332e03ed@PBX.* CSeq: 1193457 PRACK. 
Content-Length: 0. Max-Forwards: 69. RAck: 1601153264 1193456 INVITE.


Seq
U SBC_1:5060 -> SBC_2:5060 SIP/2.0 404 Not here. Via: SIP/2.0/UDP 
SBC_2:5060;branch=z9hG4bK67fa.e6a41de2.0. From: 
;tag=1875283502. To: 
;tag=FamBBcayZeKgF. Call-ID: 
fb9e258ae909d311a85a0090332e03ed@PBX. CSeq: 1193457 PRACK. Server: 
OpenSIPS (2.1.1 (x86_64/linux)). Content-Length: 0.


Why I am getting 404 from Opensips. is should be routed like seq request, right?





if (has_totag()) { # sequential requests within a dialog should # take 
the path determined by record-routing if (loose_route()) { 
xlog("loose_route"); #if ($DLG_status!=NULL) xlog("dlg_status"); if 
(!validate_dialog()){ fix_route_dialog(); xlog("fix_route_dialog"); } if 
(is_method("BYE")) { setflag(1); # do accounting ... 
#setflag(ACC_FAILED); # ... even if the transaction fails } else if 
(is_method("INVITE")) { # even if in most of the cases is useless, do RR 
for # re-INVITEs alos, as some buggy clients do change route set # 
during the dialog. record_route(); xlog("check_fraud"); } # route it out 
to whatever destination was set by loose_route() # in $du (destination 
URI). route(relay); } else { if ( is_method("ACK") ) { if ( 
t_check_trans() ) { # non loose-route, but stateful ACK; must be an ACK 
after # a 487 or e.g. 404 from upstream server t_relay(); exit; } else { 
# ACK without matching transaction -> # ignore and discard exit; } } 
sl_send_reply("404","Not here"); } exit; tnx miha

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Re: [OpenSIPS-Users] Issue with ACK and rtpproxy, setID

2016-06-30 Thread Miha

Ben tnx.

make sense :)



br

miha

On 30/06/2016 15:17, Newlin, Ben wrote:

AVPs are tied to a transaction, so the transaction must be matched before they 
will be available. You should use t_check_trans() to do this.

However, I think this will not work for you because ACKs are their own 
transactions and I don’t believe they will have access to the AVPs from the 
INVITE transaction. If you need to store state information across multiple 
transactions, you will need to use the dialog module and the $dlg_val 
variables. These persist across the entire SIP call.


Ben Newlin

On 6/30/16, 8:06 AM, "users-boun...@lists.opensips.org on behalf of Miha" 
 wrote:

HI

I have two RTPproxies and doing spiral so that I can put them in chain.

Before t_relay() I am setting avps (setID) so that I can do
rtpproxy_answer latter if there is SDP in ACK. The issue is that avp is
all the time null for ACK (for Initial invite avp was set).

In TM module i set onreply_avp_mode to 1. Is there anything else I must
do or do you suggest some other approche?


tnx

miha


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[OpenSIPS-Users] Issue with ACK and rtpproxy, setID

2016-06-30 Thread Miha

HI

I have two RTPproxies and doing spiral so that I can put them in chain.

Before t_relay() I am setting avps (setID) so that I can do 
rtpproxy_answer latter if there is SDP in ACK. The issue is that avp is 
all the time null for ACK (for Initial invite avp was set).


In TM module i set onreply_avp_mode to 1. Is there anything else I must 
do or do you suggest some other approche?



tnx

miha


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Re: [OpenSIPS-Users] engage_rtp_proxy()

2016-06-01 Thread Miha

HI

engage_rtp_proxy() work ok. I was having some other issue with dialog. 
Tnx to @Bogdan I figure it out.



br

miha


On 31/05/2016 15:54, Sasmita Panda wrote:
In my case its working great . So I haven't done such experiments to 
know what is happening with dialog module  . We are using this form 
years .


 If you got to know then let me know also . That may help me in 
future .


*/Thanks & Regards/*
/Sasmita Panda/
/Network Testing and Software Engineer/
/3CLogic , ph:07827611765/

On Tue, May 31, 2016 at 6:56 PM, Miha <mailto:m...@softnet.si>> wrote:


@Sasmita I had writen cfg script like this and it works. I tried
than with engage_rtp_proxy() but did not work automatically, that
is why i asked :)

So, can be some issue with dialog module? Not configured properly?


tnx

miha


On 31/05/2016 15:21, Sasmita Panda wrote:

Yes . This should happen . But I don't know the exact problem .
What I explain is the way we are using rtpproxy .
This is clearly mention in the document also .. You can go
through opensips.org <http://opensips.org>

This is what we are doing .  Rest I am not an expertise in
opensips .
route {
...
 if (is_method("INVITE")) {
 if (has_body("application/sdp")) {
 if (rtpproxy_offer())
 t_on_reply("1");
 } else {
 t_on_reply("2");
 }
 }
 if (is_method("ACK") && has_body("application/sdp"))
 rtpproxy_answer();
...
}

onreply_route[1]
{
...
 if (has_body("application/sdp"))
 rtpproxy_answer();
...
}

onreply_route[2]
{
...
 if (has_body("application/sdp"))
 rtpproxy_offer();
...
}

*/Thanks & Regards/*
/Sasmita Panda/
/Network Testing and Software Engineer/
/3CLogic , ph:07827611765/

On Tue, May 31, 2016 at 6:32 PM, Max Mühlbronner mailto:m...@42com.com>> wrote:

Hi,


@Miha: Are you sure that it does not automatically set the
rtpproxies for 200OK & ACK?

@Sasmita: According to the documentation it is not necessary
to invoke engage_rtp_proxy() for replies as this is handled
by the dialog module.


"Function must only be called for the initial INVITE and
internally takes care of rewriting the body of 200 OKs and
ACKs. "



Best Regards

Max M.


On 31.05.2016 14:42, Miha wrote:

@Sasmita, totally clear :)

I asked wrong question :)


What is the difference between using engage_rtp_proxy() or
using rtpproxy_offer(), rtpproxy_answer()?


tnx

miha


On 31/05/2016 14:39, Sasmita Panda wrote:

If you are using in INVITE , then it should be
offer . Because firstly we are offering media to someone .
If its 200 Ok then it will be answer because the 2nd party
is answering the call .

  If there is no sdp in INVITE but in ACK , then it
will get reversed . In 200 OK you should offer and in ACK
you have to answer .
This can be done in loop .

 I hope I make you understand .

*/Thanks & Regards/*
/Sasmita Panda/
/Network Testing and Software Engineer/
/3CLogic , ph:07827611765/

On Tue, May 31, 2016 at 6:02 PM, Miha mailto:m...@softnet.si> <mailto:m...@softnet.si>
<mailto:m...@softnet.si>> wrote:

ok tnx. I understand documentation on wrong way.

But then, what is the difference with  using rtpproxy
offer, answer ?


br

mia


On 31/05/2016 14:17, Sasmita Panda wrote:

If there is sdp in ACK and u wanted to engage rtp
proxy , the
 you have to write it inside ACK also ... By writing
for INVITE
cant help you to update ACK also . For 200 OK , you
must write it
in reply route .

*/Thanks & Regards/*
/Sasmita Panda/
/Network Testing and Software Engineer/
/3CLogic , ph:07827611765/

On Tue, May 31, 2016 at 5:35 PM, Johan De Clercq
mailto:jo...@democon.be>
<mailto:jo...@democon.be> <mailto:jo...@democon.be>> wrote:

put it also in reply route.

2016-05-31 13:42 GMT+02:00 Miha mailto:m...@softnet.si>
<mailto:m...@softnet.si> <mailto:m...@softnet.si>>:

HI

if I use engage_rtp_proxy(), I can use it only
on initial
INVITE and opensips should automatically
rewritten also
  

Re: [OpenSIPS-Users] engage_rtp_proxy()

2016-05-31 Thread Miha
@Sasmita I had writen cfg script like this and it works. I tried than 
with engage_rtp_proxy() but did not work automatically, that is why i 
asked :)


So, can be some issue with dialog module? Not configured properly?


tnx

miha


On 31/05/2016 15:21, Sasmita Panda wrote:
Yes . This should happen . But I don't know the exact problem . What I 
explain is the way we are using rtpproxy .
This is clearly mention in the document also .. You can go through 
opensips.org <http://opensips.org>


This is what we are doing .  Rest I am not an expertise in opensips .
route {
...
 if (is_method("INVITE")) {
 if (has_body("application/sdp")) {
 if (rtpproxy_offer())
 t_on_reply("1");
 } else {
 t_on_reply("2");
 }
 }
 if (is_method("ACK") && has_body("application/sdp"))
 rtpproxy_answer();
...
}

onreply_route[1]
{
...
 if (has_body("application/sdp"))
 rtpproxy_answer();
...
}

onreply_route[2]
{
...
 if (has_body("application/sdp"))
 rtpproxy_offer();
...
}

*/Thanks & Regards/*
/Sasmita Panda/
/Network Testing and Software Engineer/
/3CLogic , ph:07827611765/

On Tue, May 31, 2016 at 6:32 PM, Max Mühlbronner <mailto:m...@42com.com>> wrote:


Hi,


@Miha: Are you sure that it does not automatically set the
rtpproxies for 200OK & ACK?

@Sasmita: According to the documentation it is not necessary to
invoke engage_rtp_proxy() for replies as this is handled by the
dialog module.


"Function must only be called for the initial INVITE and
internally takes care of rewriting the body of 200 OKs and ACKs. "



Best Regards

Max M.


On 31.05.2016 14:42, Miha wrote:

@Sasmita, totally clear :)

I asked wrong question :)


    What is the difference between using engage_rtp_proxy() or using
rtpproxy_offer(), rtpproxy_answer()?


tnx

miha


On 31/05/2016 14:39, Sasmita Panda wrote:

If you are using in INVITE , then it should be offer .
Because firstly we are offering media to someone . If its 200 Ok
then it will be answer because the 2nd party is answering the
call .

  If there is no sdp in INVITE but in ACK , then it will get
reversed . In 200 OK you should offer and in ACK you have to
answer .
This can be done in loop .

 I hope I make you understand .

*/Thanks & Regards/*
    /Sasmita Panda/
/Network Testing and Software Engineer/
/3CLogic , ph:07827611765/

On Tue, May 31, 2016 at 6:02 PM, Miha mailto:m...@softnet.si> <mailto:m...@softnet.si>
<mailto:m...@softnet.si>> wrote:

ok tnx. I understand documentation on wrong way.

But then, what is the difference with  using rtpproxy offer,
answer ?


br

mia


On 31/05/2016 14:17, Sasmita Panda wrote:

If there is sdp in ACK and u wanted to engage rtp proxy , the
 you have to write it inside ACK also ... By writing for
INVITE
cant help you to update ACK also . For 200 OK , you must
write it
in reply route .

*/Thanks & Regards/*
/Sasmita Panda/
/Network Testing and Software Engineer/
/3CLogic , ph:07827611765/

On Tue, May 31, 2016 at 5:35 PM, Johan De Clercq
mailto:jo...@democon.be>
<mailto:jo...@democon.be> <mailto:jo...@democon.be>> wrote:

put it also in reply route.

2016-05-31 13:42 GMT+02:00 Miha mailto:m...@softnet.si>
<mailto:m...@softnet.si> <mailto:m...@softnet.si>>:

HI

if I use engage_rtp_proxy(), I can use it only on
initial
INVITE and opensips should automatically rewritten
also
200 OK and ACK with SDP, right?
But when I am using this function, I can see from
trace
that only SDP for initial invite is rewritten, 200 ok
with sdp is not changed. Must I do something else?

Rtpproxy is not running in bridge mode.


tnx
miha



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Re: [OpenSIPS-Users] engage_rtp_proxy()

2016-05-31 Thread Miha

@Max, yes I am sure. Did some traces...

That is way I asked what I could be doing wrong. I read that there is an 
issue doing it with rtpproxy in bridge, but in my case it is not in bridge.



tnx

miha


On 31/05/2016 15:02, Max Mühlbronner wrote:

Hi,


@Miha: Are you sure that it does not automatically set the rtpproxies 
for 200OK & ACK?


@Sasmita: According to the documentation it is not necessary to invoke 
engage_rtp_proxy() for replies as this is handled by the dialog module.



"Function must only be called for the initial INVITE and internally 
takes care of rewriting the body of 200 OKs and ACKs. "




Best Regards

Max M.


On 31.05.2016 14:42, Miha wrote:

@Sasmita, totally clear :)

I asked wrong question :)


What is the difference between using engage_rtp_proxy() or using 
rtpproxy_offer(), rtpproxy_answer()?



tnx

miha


On 31/05/2016 14:39, Sasmita Panda wrote:
If you are using in INVITE , then it should be offer . 
Because firstly we are offering media to someone . If its 200 Ok 
then it will be answer because the 2nd party is answering the call .


  If there is no sdp in INVITE but in ACK , then it will get 
reversed . In 200 OK you should offer and in ACK you have to answer .

This can be done in loop .

 I hope I make you understand .

*/Thanks & Regards/*
/Sasmita Panda/
/Network Testing and Software Engineer/
/3CLogic , ph:07827611765/

On Tue, May 31, 2016 at 6:02 PM, Miha <mailto:m...@softnet.si>> wrote:


ok tnx. I understand documentation on wrong way.

But then, what is the difference with  using rtpproxy offer, 
answer ?



br

mia


On 31/05/2016 14:17, Sasmita Panda wrote:

If there is sdp in ACK and u wanted to engage rtp proxy , the
 you have to write it inside ACK also ... By writing for INVITE
cant help you to update ACK also . For 200 OK , you must write it
in reply route .

*/Thanks & Regards/*
/Sasmita Panda/
/Network Testing and Software Engineer/
/3CLogic , ph:07827611765/

On Tue, May 31, 2016 at 5:35 PM, Johan De Clercq
mailto:jo...@democon.be>> wrote:

put it also in reply route.

    2016-05-31 13:42 GMT+02:00 Miha mailto:m...@softnet.si>>:

HI

if I use engage_rtp_proxy(), I can use it only on initial
INVITE and opensips should automatically rewritten also
200 OK and ACK with SDP, right?
But when I am using this function, I can see from trace
that only SDP for initial invite is rewritten, 200 ok
with sdp is not changed. Must I do something else?

Rtpproxy is not running in bridge mode.


tnx
miha



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Re: [OpenSIPS-Users] engage_rtp_proxy()

2016-05-31 Thread Miha

@Sasmita, totally clear :)

I asked wrong question :)


What is the difference between using engage_rtp_proxy() or using 
rtpproxy_offer(), rtpproxy_answer()?



tnx

miha


On 31/05/2016 14:39, Sasmita Panda wrote:
If you are using in INVITE , then it should be offer . Because 
firstly we are offering media to someone . If its 200 Ok then it will 
be answer because the 2nd party is answering the call .


  If there is no sdp in INVITE but in ACK , then it will get 
reversed . In 200 OK you should offer and in ACK you have to answer .

This can be done in loop .

 I hope I make you understand .

*/Thanks & Regards/*
/Sasmita Panda/
/Network Testing and Software Engineer/
/3CLogic , ph:07827611765/

On Tue, May 31, 2016 at 6:02 PM, Miha <mailto:m...@softnet.si>> wrote:


ok tnx. I understand documentation on wrong way.

But then, what is the difference with  using rtpproxy offer, answer ?


br

mia


On 31/05/2016 14:17, Sasmita Panda wrote:

If there is sdp in ACK and u wanted to engage rtp proxy , the
 you have to write it inside ACK also ... By writing for INVITE
cant help you to update ACK also . For 200 OK , you must write it
in reply route .

*/Thanks & Regards/*
/Sasmita Panda/
/Network Testing and Software Engineer/
/3CLogic , ph:07827611765/

On Tue, May 31, 2016 at 5:35 PM, Johan De Clercq
mailto:jo...@democon.be>> wrote:

put it also in reply route.

2016-05-31 13:42 GMT+02:00 Miha mailto:m...@softnet.si>>:

HI

if I use engage_rtp_proxy(), I can use it only on initial
INVITE and opensips should automatically rewritten also
200 OK and ACK with SDP, right?
But when I am using this function, I can see from trace
that only SDP for initial invite is rewritten, 200 ok
with sdp is not changed. Must I do something else?

Rtpproxy is not running in bridge mode.


    tnx
miha



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Re: [OpenSIPS-Users] engage_rtp_proxy()

2016-05-31 Thread Miha

ok tnx. I understand documentation on wrong way.

But then, what is the difference with  using rtpproxy offer, answer ?


br

mia


On 31/05/2016 14:17, Sasmita Panda wrote:
If there is sdp in ACK and u wanted to engage rtp proxy , the  you 
have to write it inside ACK also ... By writing for INVITE cant help 
you to update ACK also . For 200 OK , you must write it in reply route .


*/Thanks & Regards/*
/Sasmita Panda/
/Network Testing and Software Engineer/
/3CLogic , ph:07827611765/

On Tue, May 31, 2016 at 5:35 PM, Johan De Clercq <mailto:jo...@democon.be>> wrote:


put it also in reply route.

2016-05-31 13:42 GMT+02:00 Miha mailto:m...@softnet.si>>:

HI

if I use engage_rtp_proxy(), I can use it only on initial
INVITE and opensips should automatically rewritten also 200 OK
and ACK with SDP, right?
But when I am using this function, I can see from trace that
only SDP for initial invite is rewritten, 200 ok with sdp is
not changed. Must I do something else?

Rtpproxy is not running in bridge mode.


    tnx
miha



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[OpenSIPS-Users] engage_rtp_proxy()

2016-05-31 Thread Miha

HI

if I use engage_rtp_proxy(), I can use it only on initial INVITE and 
opensips should automatically rewritten also 200 OK and ACK with SDP, right?
But when I am using this function, I can see from trace that only SDP 
for initial invite is rewritten, 200 ok with sdp is not changed. Must I 
do something else?


Rtpproxy is not running in bridge mode.


tnx
miha



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[OpenSIPS-Users] Compiling issue, recipe for target 'sipmemcache.o' failed

2016-03-09 Thread Miha

Hi,

need a little help. I have installed memcache-dev, what else could i be 
missing?


make[2]: Entering directory '/usr/src/opensips_2_1/modules/lua'
Compiling sipmemcache.c
sipmemcache.c:25:22: fatal error: memcache.h: No such file or directory
 #include 
  ^
compilation terminated.
../../Makefile.rules:25: recipe for target 'sipmemcache.o' failed
make[2]: *** [sipmemcache.o] Error 1
make[2]: Leaving directory '/usr/src/opensips_2_1/modules/lua'
Makefile:194: recipe for target 'modules' failed
make[1]: *** [modules] Error 2
make[1]: Leaving directory '/usr/src/opensips_2_1'


tnx
miha
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Re: [OpenSIPS-Users] Listening ips and sending call from them

2016-03-01 Thread Miha
Tnx Sammy,

let me try :)


br
miha
On Tue, 1 Mar 2016 16:49:29 -0500
 SamyGo  wrote:
> Hey Miha,
> 
> See if this thread helps you:
>
http://lists.opensips.org/pipermail/users/2010-October/015150.html
> 
> Regards,
> Sammy
> 
> On Tue, Mar 1, 2016 at 9:35 AM, Miha 
> wrote:
> 
> > HI.
> >
> > If you have two ips on your server, let say X in Y.
> > When you route calls to providers is it possible to use
> ip
> > X for one provider and ip Y for other provider?
> >
> >
> > tnx
> > miah
> >
> >
> > ___
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> >
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[OpenSIPS-Users] Listening ips and sending call from them

2016-03-01 Thread Miha
HI.

If you have two ips on your server, let say X in Y.
When you route calls to providers is it possible to use ip
X for one provider and ip Y for other provider?


tnx
miah


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Re: [OpenSIPS-Users] websocket t_relay()

2016-02-23 Thread Miha

tnx Răzvan :)

br
miha

On 23/02/2016 15:24, Răzvan Crainea wrote:

Hi, Miha!

Make sure you are calling fix_nated_register() function for all 
REGISTERS coming from a Websocket client.


Best regards,
Răzvan

On 02/23/2016 04:12 PM, Miha wrote:

Hi,

first time looking at module for websocket (opensips 2.1).

Reciving calls from SipJs is working, but how to send call beck to 
websocket? If i just do lookup location and do t_relay() I am getting 
"Unresolvable destination"?


In location table I have: 
"sip:a4tn9tda@iibhaol8elto.invalid;transport=ws"



tnx
miha

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[OpenSIPS-Users] websocket t_relay()

2016-02-23 Thread Miha

Hi,

first time looking at module for websocket (opensips 2.1).

Reciving calls from SipJs is working, but how to send call beck to 
websocket? If i just do lookup location and do t_relay() I am getting 
"Unresolvable destination"?


In location table I have: "sip:a4tn9tda@iibhaol8elto.invalid;transport=ws"


tnx
miha

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[OpenSIPS-Users] enum asynchronous

2015-12-16 Thread Miha

Hi,

is there any news when enum could be done asynchronously?


tnx
miha

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[OpenSIPS-Users] 2.1.1 segfault

2015-11-02 Thread Miha

Hi,

could some look at this crash, what could be wrong:

Nov  2 09:20:17 sbc-adria kernel: opensips[3791]: segfault at 2 ip 
0002 sp 7fff0ca4d638 error 14 in opensips[40+1e2000]
Nov  2 09:20:17 sbc-adria abrtd: Directory 
'ccpp-2015-11-02-09:20:17-3791' creation detected
Nov  2 09:20:17 sbc-adria abrt[3811]: Saved core dump of pid 3791 
(/usr/local/src/opensips_2_1/opensips) to 
/var/spool/abrt/ccpp-2015-11-02-09:20:17-3791 (38289408 bytes)
Nov  2 09:20:17 sbc-adria ./opensips[3776]: INFO:core:handle_sigs: child 
process 3791 exited by a signal 11
Nov  2 09:20:17 sbc-adria ./opensips[3776]: INFO:core:handle_sigs: core 
was generated
Nov  2 09:20:17 sbc-adria ./opensips[3776]: INFO:core:handle_sigs: 
terminating due to SIGCHLD
Nov  2 09:20:18 sbc-adria ./opensips[3803]: INFO:core:sig_usr: signal 15 
received
Nov  2 09:20:18 sbc-adria ./opensips[3802]: INFO:core:sig_usr: signal 15 
received
Nov  2 09:20:18 sbc-adria ./opensips[3801]: INFO:core:sig_usr: signal 15 
received
Nov  2 09:20:18 sbc-adria ./opensips[3800]: INFO:core:sig_usr: signal 15 
received
Nov  2 09:20:18 sbc-adria ./opensips[3799]: INFO:core:sig_usr: signal 15 
received
Nov  2 09:20:18 sbc-adria ./opensips[3793]: INFO:core:sig_usr: signal 15 
received
Nov  2 09:20:18 sbc-adria ./opensips[3792]: INFO:core:sig_usr: signal 15 
received
Nov  2 09:20:18 sbc-adria ./opensips[3790]: INFO:core:sig_usr: signal 15 
received
Nov  2 09:20:18 sbc-adria ./opensips[3789]: INFO:core:sig_usr: signal 15 
received
Nov  2 09:20:18 sbc-adria ./opensips[3788]: INFO:core:sig_usr: signal 15 
received
Nov  2 09:20:18 sbc-adria ./opensips[3787]: INFO:core:sig_usr: signal 15 
received

Nov  2 09:20:18 sbc-adria ./opensips[3776]: INFO:core:cleanup: cleanup
Nov  2 09:20:18 sbc-adria abrtd: Executable 
'/usr/local/src/opensips_2_1/opensips' doesn't belong to any package and 
ProcessUnpackaged is set to 'no'
Nov  2 09:20:18 sbc-adria abrtd: 'post-create' on 
'/var/spool/abrt/ccpp-2015-11-02-09:20:17-3791' exited with 1
Nov  2 09:20:18 sbc-adria abrtd: Deleting problem directory 
'/var/spool/abrt/ccpp-2015-11-02-09:20:17-3791'



http://pastebin.com/p5827h1e


tnx
miha

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[OpenSIPS-Users] from 2.1.0 to 2.1.1 version upgrade issue

2015-11-01 Thread Miha

Hi,

I had installed opensips 2.1.0 from git. Now I have try to upgrade (to 
2.1.1) it like this:

git pull --rebase
make all
make install

After "make all" I got:

make[1]: Entering directory `/usr/local/src/opensips_2_1/modules/pdt'
make[1]: *** No targets specified and no makefile found.  Stop.
make[1]: Leaving directory `/usr/local/src/opensips_2_1/modules/pdt'
make: *** [modules] Error 2

And after I tried with running it I am getting in logs:

Nov  2 08:42:03 sip-adria opensips: ERROR:core:version_control: module 
version mismatch for regex; core: opensips 2.2-dev (x86_64/linux); 
module: opensips 2.1.0 (x86_64/linux)


The same steps I have done on other server and it worked.
Could you please help me with this upgrade from 2.1.0 to 2.1.1.


tnx
miha

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[OpenSIPS-Users] NAT issue

2015-10-27 Thread Miha

Hi,

I have one big issue and I do not know how to fix it.

Invite is recived, which has:
- User Datagram Protocol, Src Port: 1334 (1334), Dst Port: 5060 (5060)
- Via: SIP/2.0/UDP 192.168.131.120:5072;branch=z9hG4bK-a7b4d998

Than 407 is replied back on wrong port (5072) and UAC send again same 
INVITE with not credentials in it.

- User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5072 (5072)


This does not work as reply should be send to port 1334.

Please let me know if I can fix this and how :)


tnx
miha

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Re: [OpenSIPS-Users] parse_uri: bad uri

2015-09-15 Thread Miha

Hi Razvan,

you are right, it was wrong URI. Regarding drouting, i can no to routing 
with drouting module if in Ruri is "E", right?


So the best case would be to transform "E" with dialplan in some number 
and then do drouting or do you have any better preposition?


br
miha

On 15/09/2015 12:06, Răzvan Crainea wrote:

Hi, Miha!

That does not look like an URI at all: it contains only the username, 
not the scheme, host, port, etc. Do you have the original message to 
track this down?


Best regards,

Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com

On 09/15/2015 11:41 AM, Miha wrote:

Hi,

one of our costumer is sending as uri with "E" like "and opensips return:


Sep 15 10:36:30 sip-adria ./opensips[18583]: ERROR:core:parse_uri: 
bad uri, state 0 parsed:  (4) /  (14)
Sep 15 10:36:30 sip-adria ./opensips[18583]: 
ERROR:core:parse_sip_msg_uri: bad uri 


how to deal with this as we need also to send him with this kind of 
prefix?


tnx
miha

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[OpenSIPS-Users] parse_uri: bad uri

2015-09-15 Thread Miha

Hi,

one of our costumer is sending as uri with "E" like "and opensips return:


Sep 15 10:36:30 sip-adria ./opensips[18583]: ERROR:core:parse_uri: bad 
uri, state 0 parsed:  (4) /  (14)
Sep 15 10:36:30 sip-adria ./opensips[18583]: 
ERROR:core:parse_sip_msg_uri: bad uri 


how to deal with this as we need also to send him with this kind of prefix?

tnx
miha

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[OpenSIPS-Users] Group module problem / db_get_user_group

2015-06-16 Thread Miha

Hi,

if I use db_is_user_in() this function returns me ok if user is in this 
group.
if I use db_get_user_group() and do group lookup for exactly the same 
user (URI) which was used for db_is_user_in() I do not get anything.


Could some help we figure it out what I am doing wrong?

I am using $fu for lookup.

tnx
miha


 |
 |


 |
 |

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Re: [OpenSIPS-Users] closeddial, migration from 1.10 to 2.1

2015-05-27 Thread Miha

Hi,

Razvan helped me on irc.

Group and dialplan module.

br
miha

On 27/05/2015 09:06, Miha wrote:

Hi,

i noticed that module closeddial was removed. What can be used instead?

tnx
miha

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[OpenSIPS-Users] closeddial, migration from 1.10 to 2.1

2015-05-27 Thread Miha

Hi,

i noticed that module closeddial was removed. What can be used instead?

tnx
miha

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Re: [OpenSIPS-Users] Enum modul

2015-01-15 Thread Miha
Hi Bogdan,

Tnx for info. In future this future would be good to
implement;)

Ne
Miha

On Thu, 15 Jan 2015 14:30:59 +0200
 Bogdan-Andrei Iancu  wrote:
> Hi Miha,
> 
> In OpenSIPS, for any DNS lookup, the res_search()
> frunction from glib is used, function that uses the
> /etc/resolv.conf file for fetching the default DNS
> server.
> I guess this function does not have the ability to figure
> out what's the direct DNS server and go directly there.
> 
> Best regards,
> 
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developer
> http://www.opensips-solutions.com
> 
> On 14.01.2015 08:17, Miha wrote:
> > Hi,
> >
> > I am using enum module for ported numbers. Everything
> works fine just one thing is that is bothering me.
> >
> > If I use dig to see who is authoritative dns server for
> one enum zone I can see that this not not dns server
> which is set in resolve.conf, so opensips should be doing
> this dns request for enum to authoritative dns server.
> >
> > I have exactly the same config on nextone sbc and it is
> sending request to authoritative dns server for this
> zone.
> >
> > How to fix this as now opensips sends request to non
> authoritative server for this zone, dns server send it to
> enum server and then in oposite direction for resonses
> which is not ok.
> >
> > Thx
> > Miha
> >
> > trace:
> >
> >  7.476490 opensips -> DNS SERVER DNS 102 Standard query
> 0x0687 NAPTR 0.5.1.3.0.3.8.1.6.8.3.enumzone.domain.com
> >   7.478184 DNS SERVER -> opensips DNS 143 Standard
> query response 0x0687 No such name
> >
> > [root@sip2 ~]# dig enumzone.domain.com NS
> >
> > ; <<>> DiG 9.8.2rc1-RedHat-9.8.2-0.17.rc1.el6_4.6 <<>>
> enumzone.domain.com NS
> > ;; global options: +cmd
> > ;; Got answer:
> > ;; ->>HEADER<<- opcode: QUERY, status: NOERROR, id:
> 42310
> > ;; flags: qr rd ra; QUERY: 1, ANSWER: 1, AUTHORITY: 0,
> ADDITIONAL: 1
> >
> > ;; QUESTION SECTION:
> > ;enumzone.domain.com.IN  NS
> >
> > ;; ANSWER SECTION:
> > enumzone.domain.com. 239 IN  NS
> ns1.enum.domain.com.
> >
> > ;; ADDITIONAL SECTION:
> > ns1.enum.domain.com.251 IN  A
>   ENUM_SERVER
> >
> > ;; Query time: 0 msec
> > ;; SERVER: DNS SERVER#53(DNS SERVER)
> > ;; WHEN: Wed Jan 14 07:05:28 2015
> > ;; MSG SIZE  rcvd: 75
> >
> >
> >
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> >
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[OpenSIPS-Users] Enum modul

2015-01-13 Thread Miha

Hi,

I am using enum module for ported numbers. Everything works fine just 
one thing is that is bothering me.


If I use dig to see who is authoritative dns server for one enum zone I 
can see that this not not dns server which is set in resolve.conf, so 
opensips should be doing this dns request for enum to authoritative dns 
server.


I have exactly the same config on nextone sbc and it is sending request 
to authoritative dns server for this zone.


How to fix this as now opensips sends request to non authoritative 
server for this zone, dns server send it to enum server and then in 
oposite direction for resonses which is not ok.


Thx
Miha

trace:

 7.476490 opensips -> DNS SERVER DNS 102 Standard query 0x0687 NAPTR 
0.5.1.3.0.3.8.1.6.8.3.enumzone.domain.com
  7.478184 DNS SERVER -> opensips DNS 143 Standard query response 
0x0687 No such name


[root@sip2 ~]# dig enumzone.domain.com NS

; <<>> DiG 9.8.2rc1-RedHat-9.8.2-0.17.rc1.el6_4.6 <<>> 
enumzone.domain.com NS

;; global options: +cmd
;; Got answer:
;; ->>HEADER<<- opcode: QUERY, status: NOERROR, id: 42310
;; flags: qr rd ra; QUERY: 1, ANSWER: 1, AUTHORITY: 0, ADDITIONAL: 1

;; QUESTION SECTION:
;enumzone.domain.com.IN  NS

;; ANSWER SECTION:
enumzone.domain.com. 239 IN  NS ns1.enum.domain.com.

;; ADDITIONAL SECTION:
ns1.enum.domain.com.251 IN  A   ENUM_SERVER

;; Query time: 0 msec
;; SERVER: DNS SERVER#53(DNS SERVER)
;; WHEN: Wed Jan 14 07:05:28 2015
;; MSG SIZE  rcvd: 75



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Re: [OpenSIPS-Users] table_version: invalid version

2014-10-20 Thread Miha

Tnx Liviu,

I have imported mysql shama of 1.10 and now it is ok.

tnx
miha

On 20/10/2014 12:48, Liviu Chircu wrote:
It looks like you're using a pre-1.11 OpenSIPS and you are pointing it 
to a future (1.11+) dr_gateways table structure. There are several 
ways to fix this specific problem:


- simply change version table entry from 6 -> 5 only if no other 
OpenSIPS instance uses it

- create a new database for your current version with opensipsdbctl

Liviu

On 10/20/2014 12:55 PM, Miha wrote:

Liviu,

this is the same thing: ERROR:core:db_check_table_version: invalid 
version 6 for table dr_gateways found, expected 5


as i do not find any syntax error.

tnx
miha

On 20/10/2014 11:17, Miha wrote:

Hi Liviu,

tnx for quick reponse. Yes, you were right, I was looking at logs 
and did not know that this could mean also invalid syntax.


tnx
miha

On 20/10/2014 11:05, Liviu Chircu wrote:

Hi miha,

It looks like you have a typo in your script. Probably something 
like "www_authorize("...", "*suscriber*")" !!


Best regards,
Liviu Chircu
OpenSIPS Developer
http://www.opensips-solutions.com
On 10/20/2014 11:51 AM, Miha wrote:

Hi,

i have installed opensips from repository. I am using centos os. 
From what i can see in logs my opensips does not start do to wrong 
db version


Oct 20 10:48:26 sbc-adria /usr/sbin/opensips[22872]: 
ERROR:core:db_check_table_version: invalid version 0 for table 
suscriber found, expected 7
Oct 20 10:48:26 sbc-adria /usr/sbin/opensips[22872]: 
ERROR:auth_db:auth_fixup: error during table version check.



I tried to import db also from source file on web but still the same.

tnx for help
miha


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Re: [OpenSIPS-Users] table_version: invalid version

2014-10-20 Thread Miha

Liviu,

this is the same thing: ERROR:core:db_check_table_version: invalid 
version 6 for table dr_gateways found, expected 5


as i do not find any syntax error.

tnx
miha

On 20/10/2014 11:17, Miha wrote:

Hi Liviu,

tnx for quick reponse. Yes, you were right, I was looking at logs and 
did not know that this could mean also invalid syntax.


tnx
miha

On 20/10/2014 11:05, Liviu Chircu wrote:

Hi miha,

It looks like you have a typo in your script. Probably something like 
"www_authorize("...", "*suscriber*")" !!


Best regards,
Liviu Chircu
OpenSIPS Developer
http://www.opensips-solutions.com
On 10/20/2014 11:51 AM, Miha wrote:

Hi,

i have installed opensips from repository. I am using centos os. 
From what i can see in logs my opensips does not start do to wrong 
db version


Oct 20 10:48:26 sbc-adria /usr/sbin/opensips[22872]: 
ERROR:core:db_check_table_version: invalid version 0 for table 
suscriber found, expected 7
Oct 20 10:48:26 sbc-adria /usr/sbin/opensips[22872]: 
ERROR:auth_db:auth_fixup: error during table version check.



I tried to import db also from source file on web but still the same.

tnx for help
miha


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Re: [OpenSIPS-Users] table_version: invalid version

2014-10-20 Thread Miha

Hi Liviu,

tnx for quick reponse. Yes, you were right, I was looking at logs and 
did not know that this could mean also invalid syntax.


tnx
miha

On 20/10/2014 11:05, Liviu Chircu wrote:

Hi miha,

It looks like you have a typo in your script. Probably something like 
"www_authorize("...", "*suscriber*")" !!


Best regards,
Liviu Chircu
OpenSIPS Developer
http://www.opensips-solutions.com
On 10/20/2014 11:51 AM, Miha wrote:

Hi,

i have installed opensips from repository. I am using centos os. From 
what i can see in logs my opensips does not start do to wrong db version


Oct 20 10:48:26 sbc-adria /usr/sbin/opensips[22872]: 
ERROR:core:db_check_table_version: invalid version 0 for table 
suscriber found, expected 7
Oct 20 10:48:26 sbc-adria /usr/sbin/opensips[22872]: 
ERROR:auth_db:auth_fixup: error during table version check.



I tried to import db also from source file on web but still the same.

tnx for help
miha


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[OpenSIPS-Users] table_version: invalid version

2014-10-20 Thread Miha

Hi,

i have installed opensips from repository. I am using centos os. From 
what i can see in logs my opensips does not start do to wrong db version


Oct 20 10:48:26 sbc-adria /usr/sbin/opensips[22872]: 
ERROR:core:db_check_table_version: invalid version 0 for table suscriber 
found, expected 7
Oct 20 10:48:26 sbc-adria /usr/sbin/opensips[22872]: 
ERROR:auth_db:auth_fixup: error during table version check.



I tried to import db also from source file on web but still the same.

tnx for help
miha


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Re: [OpenSIPS-Users] NAT, UPDATE problem

2014-10-17 Thread Miha

hi,

I have solved this by adding onreply route and it it I put 
fix_nated_contact();


Now contact in 200ok is fixed and media is ok.

tnx
miha

On 16/10/2014 15:53, Miha wrote:

Hi,

in my cfg file i have this:

if (nat_uac_test("18")) {
xlog("fixing nat");
if (method=="REGISTER") {

fix_nated_register();
fix_nated_contact();
} else {
fix_nated_contact();
}
force_rport();
}

But when 200ok with sdp is send this part of script does not execute 
as in contact is still private ip. I have 18 in nat_uac_test so if src 
port port in via are different this should be triggered but it is not 
that is why proxy sends media to private ip.


Here is my sip trace:

http://pastebin.com/KvNdttj9

Tnx

miha

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[OpenSIPS-Users] NAT, UPDATE problem

2014-10-16 Thread Miha

Hi,

in my cfg file i have this:

if (nat_uac_test("18")) {
xlog("fixing nat");
if (method=="REGISTER") {

fix_nated_register();
fix_nated_contact();
} else {
fix_nated_contact();
}
force_rport();
}

But when 200ok with sdp is send this part of script does not execute as 
in contact is still private ip. I have 18 in nat_uac_test so if src port 
port in via are different this should be triggered but it is not that is 
why proxy sends media to private ip.


Here is my sip trace:

http://pastebin.com/KvNdttj9

Tnx

miha

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Re: [OpenSIPS-Users] About "Not enough free memory, no more shm memory and out of mem" Error

2014-10-15 Thread Miha
how do you start opensips, how much private mem and shared mam do you 
asigne?


brM

On 15/10/2014 14:25, wilddra...@sina.com wrote:

Hi, My friends,
  When I do a stress testing for opensips(version: 1.11.2TLS) , I 
received the following errors.  How to tuning it to solve these problem.

  anyone can help me, thanks.

opensips[1362]: ERROR:dialog:dialog_update_db: could not add another 
dialog to db
opensips[1358]: WARNING:core:fm_malloc: Not enough free memory, will 
attempt defragmentation

opensips[1358]: ERROR:usrloc:new_ucontact: no more shm memory
opensips[1358]: ERROR:usrloc:mem_insert_ucontact: failed to create new 
contact

opensips[1358]: ERROR:usrloc:insert_ucontact: failed to insert contact
opensips[1358]: ERROR:registrar:insert_contacts: failed to insert contact
opensips[1359]: WARNING:core:fm_malloc: Not enough free memory, will 
attempt defragmentation

opensips[1359]: ERROR:tm:new_t: out of mem
opensips[1359]: ERROR:tm:t_newtran: new_t failed
opensips[1360]: WARNING:core:fm_malloc: Not enough free memory, will 
attempt defragmentation

opensips[1360]: ERROR:tm:new_t: out of mem
opensips[1360]: ERROR:tm:t_newtran: new_t failed
opensips[1359]: WARNING:core:fm_malloc: Not enough free memory, will 
attempt defragmentation

opensips[1359]: ERROR:usrloc:new_ucontact: no more shm memory
opensips[1359]: ERROR:usrloc:mem_insert_ucontact: failed to create new 
contact

opensips[1359]: ERROR:usrloc:insert_ucontact: failed to insert contact
opensips[1359]: ERROR:registrar:insert_contacts: failed to insert contact
opensips[1362]: CRITICAL:db_mysql:wrapper_single_mysql_stmt_execute: 
driver error (1048): Column 'callee_contact' cannot be null



wilddra...@sina.com


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Re: [OpenSIPS-Users] enum waiting

2014-10-14 Thread Miha

Hi Bogdan,

tnx for this info.

Will try with this dns_retr_time and dns_retr_no.

And yes, this is very painful as whole voip system is not working not 
just in this exp portable numbers ;)



Br
Miha

On 14/10/2014 18:28, Bogdan-Andrei Iancu wrote:

Hi Miha,

Indeed, these blocking I/O's are painful and the 2.x release will 
start addressing this problem. The 2.1 code is already making first 
steps in solving it.


For radius, OpenSIPS cannot control (via the libradiusclient lib) the 
timeout on how long to wait for the responses.

For enum/dns, see dns_retr_time  and dns_retr_no :
http://www.opensips.org/Documentation/Script-CoreParameters-1-11#toc44

Regards,

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

On 14.10.2014 09:20, Miha wrote:

Hi,

last time radius server was not sending back requests, so opensips 
was waiting and waiting and voip was not working.


Today our enum server broke and opensips was agin waiting and waiting 
for responses and our voip did not work for the time that enum server 
was down.


Is it possible to define some time for how long opensips should wait 
so that all other call will work and not that the whole system drops?


tnx
miha

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[OpenSIPS-Users] enum waiting

2014-10-13 Thread Miha

Hi,

last time radius server was not sending back requests, so opensips was 
waiting and waiting and voip was not working.


Today our enum server broke and opensips was agin waiting and waiting 
for responses and our voip did not work for the time that enum server 
was down.


Is it possible to define some time for how long opensips should wait so 
that all other call will work and not that the whole system drops?


tnx
miha

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Re: [OpenSIPS-Users] strange behaviour, UAC needs to be rebooted

2014-06-05 Thread Miha

Hi Bogdan,

it is not related with NAT as modems are on public ip's.

From what I have understand is that modem memory become fully used as 
some memory leak thing. I also thruly do not know when exacly happens as 
I am waiting for some explenation feedback from vendor (about memory 
they told me throught the phone).


I will try to get some info from vendor as quick as possible to describe 
things more clear.


br
miha


Dne 6/5/2014 1:49 PM, piše Bogdan-Andrei Iancu:

Hi Miha,

I do not understand the explanation from the modem vendor (about the 
buffer) - what data is kept in that buffer ? is it something SIP 
related ? Do you do NAT pinging to those devices ?


Regards,

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

On 05.06.2014 14:15, Miha wrote:

Hi,

we are experiancing strange thing on UAC's that are registered on 
opensips (not all). After I while they just stop responding, call 
reaches UAC, you can hear rinback tone but the UAC does not send this 
call to heandset, also outside calls do not work. When you restart 
UAC everything is back to normal, for 1 week, maybe 2 but than again 
UAC must be restarted. This we are experiancing on cable modems and 
also on gigaset ip phones.


Company that we are buying modems from, send as fw fix (they said 
that the modem buffer is full and that is why this happens on modem).


Ok on first look this is not related with opensips but it is UAC 
thing, yes i know that but interesting thing is that when we have 
this modems on three other softswithes this did not happen for about 
6 years (this is how long we are having this modems).


If someone has experianced the same thing i have a clue what could be 
wrong or could halp in this case please help me:)


p.s.: company send as non production fw, which has hallped on this 
issue but still they are refusing to do official releas as some of 
this modem are out of support (not supported any more).


tnx
miha

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[OpenSIPS-Users] strange behaviour, UAC needs to be rebooted

2014-06-05 Thread Miha

Hi,

we are experiancing strange thing on UAC's that are registered on 
opensips (not all). After I while they just stop responding, call 
reaches UAC, you can hear rinback tone but the UAC does not send this 
call to heandset, also outside calls do not work. When you restart UAC 
everything is back to normal, for 1 week, maybe 2 but than again UAC 
must be restarted. This we are experiancing on cable modems and also on 
gigaset ip phones.


Company that we are buying modems from, send as fw fix (they said that 
the modem buffer is full and that is why this happens on modem).


Ok on first look this is not related with opensips but it is UAC thing, 
yes i know that but interesting thing is that when we have this modems 
on three other softswithes this did not happen for about 6 years (this 
is how long we are having this modems).


If someone has experianced the same thing i have a clue what could be 
wrong or could halp in this case please help me:)


p.s.: company send as non production fw, which has hallped on this issue 
but still they are refusing to do official releas as some of this modem 
are out of support (not supported any more).


tnx
miha

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