Re: [OpenSIPS-Users] Jitsi sent REGISTER, Opensips received nothing

[jacky]

I disableb the firewall, and they can ping each other through.
Now, I rewrited the content of opensips.cfg, changed
"modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:2") " to
"modparam("rtpproxy", "rtpproxy_sock", "udp:42.123.76.60:2") "
"42.123.76.60" is a public network IP.
And now i got errors showed in opensips.log:
localhost /usr/local/opensips_proxy/sbin/opensips[5179]:
ERROR:rtpproxy:send_rtpp_command: timeout waiting reply from a RTP proxy
localhost /usr/local/opensips_proxy/sbin/opensips[5179]:
ERROR:rtpproxy:send_rtpp_command: timeout waiting reply from a RTP proxy
localhost /usr/local/opensips_proxy/sbin/opensips[5179]:
ERROR:rtpproxy:send_rtpp_command: proxy  does not
respond, disable it
localhost /usr/local/opensips_proxy/sbin/opensips[5179]:
ERROR:rtpproxy:send_rtpp_command: proxy  does not
respond, disable it
localhost /usr/local/opensips_proxy/sbin/opensips[5179]:
WARNING:rtpproxy:rtpp_test: can't get version of the RTP proxy
localhost /usr/local/opensips_proxy/sbin/opensips[5179]:
WARNING:rtpproxy:rtpp_test: can't get version of the RTP proxy
localhost /usr/local/opensips_proxy/sbin/opensips[5179]:
WARNING:rtpproxy:rtpp_test: support for RTP proxy 
has been disabled temporarily
localhost /usr/local/opensips_proxy/sbin/opensips[5179]:
WARNING:rtpproxy:rtpp_test: support for RTP proxy 
has been disabled temporarily
localhost opensips: INFO:core:daemonize: pre-daemon process exiting with 0

thanks for your attention!



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Re: [OpenSIPS-Users] error 483

[Carlos Cruz]

this was it

 

" Maybe the SIP phone sends the IP of the server instead of the domain that
you have configured, and your script is configured to route out such
requests" 

 

this was it thanks!!!

 

 

Carlos

 

 

From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Vlad Paiu
Sent: Thursday, March 19, 2015 12:27 PM
To: users@lists.opensips.org
Subject: Re: [OpenSIPS-Users] error 483

 

Hello,

483 usually means 'Too Many Hops'. If you do a SIP trace on the server, do
you see OpenSIPS looping the request to itself ? Maybe the SIP phone sends
the IP of the server instead of the domain that you have configured, and
your script is configured to route out such requests.

Best Regards,



Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com 

On 19.03.2015 04:15, Carlos Cruz wrote:

Hi;

 

Can someone tell me why  or where I may find the  info I need; I'm able to
register external remote phones (hard phones), but the internal phone (soft
phones) within the same network as the OpenSIPS test server report error
483. 

 

Thanks!!

Carlos

 

 






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Re: [OpenSIPS-Users] WebSocket Support in OpenSIPS 2.1

[Eric Tamme]

yes - a tutorial was linked in the announcement which includes setting 
up rtpengine with opensips+websockets to do rtp<->dtls-srtp interop.


http://www.opensips.org/Documentation/Tutorials-WebSocket-2-1
-Eric

On 03/19/2015 02:32 PM, Tito Cumpen wrote:

Great news,


Are there any media engines that can be used in conjunction with 
opensips that would allow the interop between webrtc sip clients and 
standard sip? I am aware that freeswitch will currently do this.


Thanks,
Tito

On Wed, Mar 18, 2015 at 3:36 AM, Saúl Ibarra Corretgé 
mailto:s...@ag-projects.com>> wrote:


Congrats to the team! Great work!

On 17 Mar 2015, at 20:00, Răzvan Crainea mailto:raz...@opensips.org>> wrote:

> Hello, all!
>
> Aaand, we're finally making it official: OpenSIPS 2.1 will have
*WebSockets* support!
>
> Are you planning to use (or perhaps you're already using) WebRTC
based SIP clients, but you are having hard time setting up the
platform? Starting from now, it has never been simpler - based on
your needs and feedback[1] we decided to implement a WebSocket
server directly in OpenSIPS.
>
> And we're doing it now! Starting with the new OpenSIPS 2.1
release you will be able to plug your web-based SIP clients
directly in your OpenSIPS server using the new WebSocket transport
protocol[2].
>
> We've also setup a short tutorial[3] for you to integrate this
feature in your platform easier. Many thanks to Eric Tamme
(lirakis) for all his help with the tutorial as well as for the
intensive tests.
>
> [1] http://www.opensips.org/Community/IRCmeeting20141029
> [2] http://www.opensips.org/html/docs/modules/2.1.x/proto_ws
> [3] http://www.opensips.org/Documentation/Tutorials-WebSocket-2-1
>
> Best regards,
> --
> Răzvan Crainea
> OpenSIPS Core Developer
> http://www.opensips-solutions.com
>
> ___
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> Users@lists.opensips.org 
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

--
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AG Projects




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Re: [OpenSIPS-Users] WebSocket Support in OpenSIPS 2.1

[Tito Cumpen]

Great news,


Are there any media engines that can be used in conjunction with opensips
that would allow the interop between webrtc sip clients and standard sip? I
am aware that freeswitch will currently do this.

Thanks,
Tito

On Wed, Mar 18, 2015 at 3:36 AM, Saúl Ibarra Corretgé 
wrote:

> Congrats to the team! Great work!
>
> On 17 Mar 2015, at 20:00, Răzvan Crainea  wrote:
>
> > Hello, all!
> >
> > Aaand, we're finally making it official: OpenSIPS 2.1 will have
> *WebSockets* support!
> >
> > Are you planning to use (or perhaps you're already using) WebRTC based
> SIP clients, but you are having hard time setting up the platform? Starting
> from now, it has never been simpler - based on your needs and feedback[1]
> we decided to implement a WebSocket server directly in OpenSIPS.
> >
> > And we're doing it now! Starting with the new OpenSIPS 2.1 release you
> will be able to plug your web-based SIP clients directly in your OpenSIPS
> server using the new WebSocket transport protocol[2].
> >
> > We've also setup a short tutorial[3] for you to integrate this feature
> in your platform easier. Many thanks to Eric Tamme (lirakis) for all his
> help with the tutorial as well as for the intensive tests.
> >
> > [1] http://www.opensips.org/Community/IRCmeeting20141029
> > [2] http://www.opensips.org/html/docs/modules/2.1.x/proto_ws
> > [3] http://www.opensips.org/Documentation/Tutorials-WebSocket-2-1
> >
> > Best regards,
> > --
> > Răzvan Crainea
> > OpenSIPS Core Developer
> > http://www.opensips-solutions.com
> >
> > ___
> > Users mailing list
> > Users@lists.opensips.org
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
> --
> Saúl Ibarra Corretgé
> AG Projects
>
>
>
>
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>
>
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Re: [OpenSIPS-Users] SIP/2.0 477 Send failed (477/TM) - Route

[Bogdan-Andrei Iancu]

Hi Leo,

If you look in your logs, you should see some errors where OpenSIPS 
complains about not being able to open some TCP connection. Basically 
OpenSIPS tried to forward the call by TCP but failed for some reasons 
(TCP related). Check the logs.


Regards,

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

On 17.03.2015 18:37, leo wrote:

Hello,

I'm receiving the following message when a try to place a call:
SIP/2.0 477 Send failed (477/TM)

This is desired issue because the callee UA is not online and in userloc it
is not expired yet.

My question is, which would be the process or the route this event (477 Send
failed) is processed? I've tried to log on failure_route, onreply_route and
even on branch_route but it was unsuccessful.

Thanks a lot,

Leo




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Re: [OpenSIPS-Users] Dispatcher user specific route question - 2.1

[Satish Patel]

Thanks! for quick answer!!

On Thu, Mar 19, 2015 at 12:41 PM, Vlad Paiu  wrote:

>  Hello,
>
> It will do fail-over.
>
> Best Regards,
>
> Vlad Paiu
> OpenSIPS Developerhttp://www.opensips-solutions.com
>
> On 19.03.2015 18:39, Satish Patel wrote:
>
> Thanks Vlad,
>
>  Superb! so it will do round-robin? or fail-over?
>
> On Thu, Mar 19, 2015 at 12:30 PM, Vlad Paiu  wrote:
>
>>  Hello,
>>
>> If you want to do dispatching between multiple setids, ds_select_dst()
>> allows that. See the docs at [1] , you can provide a comma separated list
>> of setids - so your $avp(zone) can contain '1,2' and OpenSIPS will try to
>> first send to the servers in setid 1, and then, if those fail, to the
>> servers in setid 2.
>>
>> [1] http://www.opensips.org/html/docs/modules/1.11.x/dispatcher#id294368
>>
>> Best Regards,
>>
>> Vlad Paiu
>> OpenSIPS Developerhttp://www.opensips-solutions.com
>>
>>  On 19.03.2015 06:17, Satish Patel wrote:
>>
>>  I have add extra "zone" column in subscriber table,
>>
>> +--+-+
>> | username |  zone |
>> +--+-+
>> |1001 |1|
>> |1002 |2|
>> +--+-+
>>
>>
>>  In dispatcher table I have following two Freeswitch in two groups.
>>
>> +---+-++
>> | setid | destination  | description|
>> +---+--+---+
>> | 1 | sip:fs1.example.com | Freeswitch-1 |
>> | 2 | sip:fs2.example.com | Freeswitch-2 |
>> +---+--+---+
>>
>>
>>  in opensips.cfg script i am query subscriber table base on incoming
>> username and storing zone in avp(zone) variable, and calling same variable
>> in following code
>>
>> if ( !ds_select_dst("$avp(zone)", "4", "FM10"))
>>
>>  Question: now either user belongs to zone 1 or 2, so it is *NOT* going
>> to do load-balancing between two. But if I want to allow some user to do
>> load-balancing then how it will be possible in above scenario?
>>
>>  Can i set "setid" on fly so i can pass request along with user request
>> and set same group for both switch and user call load-balance on both
>> switch?
>>
>>  Any other idea?
>>
>>
>>  ___
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>> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>>
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>>
>
>
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Re: [OpenSIPS-Users] Dispatcher user specific route question - 2.1

[Vlad Paiu]

Hello,

It will do fail-over.

Best Regards,

Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com

On 19.03.2015 18:39, Satish Patel wrote:

Thanks Vlad,

Superb! so it will do round-robin? or fail-over?

On Thu, Mar 19, 2015 at 12:30 PM, Vlad Paiu > wrote:


Hello,

If you want to do dispatching between multiple setids,
ds_select_dst() allows that. See the docs at [1] , you can provide
a comma separated list of setids - so your $avp(zone) can contain
'1,2' and OpenSIPS will try to first send to the servers in setid
1, and then, if those fail, to the servers in setid 2.

[1]
http://www.opensips.org/html/docs/modules/1.11.x/dispatcher#id294368

Best Regards,

Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com  


On 19.03.2015 06:17, Satish Patel wrote:

I have add extra "zone" column in subscriber table,

+--+-+
| username |  zone |
+--+-+
|1001 |1|
|1002 |2|
+--+-+


In dispatcher table I have following two Freeswitch in two groups.

+---+-++
| setid | destination  | description|
+---+--+---+
| 1 | sip:fs1.example.com  |
Freeswitch-1 |
| 2 | sip:fs2.example.com  |
Freeswitch-2 |
+---+--+---+


in opensips.cfg script i am query subscriber table base on
incoming username and storing zone in avp(zone) variable, and
calling same variable in following code

if ( !ds_select_dst("$avp(zone)", "4", "FM10"))

Question: now either user belongs to zone 1 or 2, so it is *NOT*
going to do load-balancing between two. But if I want to allow
some user to do load-balancing then how it will be possible in
above scenario?

Can i set "setid" on fly so i can pass request along with user
request and set same group for both switch and user call
load-balance on both switch?

Any other idea?


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Re: [OpenSIPS-Users] Dispatcher user specific route question - 2.1

[Satish Patel]

Thanks Vlad,

Superb! so it will do round-robin? or fail-over?

On Thu, Mar 19, 2015 at 12:30 PM, Vlad Paiu  wrote:

>  Hello,
>
> If you want to do dispatching between multiple setids, ds_select_dst()
> allows that. See the docs at [1] , you can provide a comma separated list
> of setids - so your $avp(zone) can contain '1,2' and OpenSIPS will try to
> first send to the servers in setid 1, and then, if those fail, to the
> servers in setid 2.
>
> [1] http://www.opensips.org/html/docs/modules/1.11.x/dispatcher#id294368
>
> Best Regards,
>
> Vlad Paiu
> OpenSIPS Developerhttp://www.opensips-solutions.com
>
> On 19.03.2015 06:17, Satish Patel wrote:
>
> I have add extra "zone" column in subscriber table,
>
> +--+-+
> | username |  zone |
> +--+-+
> |1001 |1|
> |1002 |2|
> +--+-+
>
>
>  In dispatcher table I have following two Freeswitch in two groups.
>
> +---+-++
> | setid | destination  | description|
> +---+--+---+
> | 1 | sip:fs1.example.com | Freeswitch-1 |
> | 2 | sip:fs2.example.com | Freeswitch-2 |
> +---+--+---+
>
>
>  in opensips.cfg script i am query subscriber table base on incoming
> username and storing zone in avp(zone) variable, and calling same variable
> in following code
>
> if ( !ds_select_dst("$avp(zone)", "4", "FM10"))
>
>  Question: now either user belongs to zone 1 or 2, so it is *NOT* going
> to do load-balancing between two. But if I want to allow some user to do
> load-balancing then how it will be possible in above scenario?
>
>  Can i set "setid" on fly so i can pass request along with user request
> and set same group for both switch and user call load-balance on both
> switch?
>
>  Any other idea?
>
>
> ___
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>
>
>
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Re: [OpenSIPS-Users] Jitsi sent REGISTER, Opensips received nothing

[Vlad Paiu]

Hello,

Well, if you did a tcpdump on the OpenSIPS box and saw nothing, then it 
means the packages aren't actually reaching the box. Please check that 
there are no firewalls in between the client and OpenSIPS that are 
blocking the traffic.


Best Regards,

Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com

On 19.03.2015 08:23, jacky wrote:

I have a test with one Jitsi using Opensips on the Internet
Wireshark showed me that Jitsi sent several "REGISTER" packages, in the same
time I used the command "tcpdump" to listen on the Opensips Server , but got
nothing.
Anyway, on the server, opensips, rtpproxy, media-dispatcher, media-relay are
running!
what happened to opensips server? why it won't response to distanced
request?

I appreciate your opinion, thanks a lot!

Best regards




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Re: [OpenSIPS-Users] Dispatcher user specific route question - 2.1

[Vlad Paiu]

Hello,

If you want to do dispatching between multiple setids, ds_select_dst() 
allows that. See the docs at [1] , you can provide a comma separated 
list of setids - so your $avp(zone) can contain '1,2' and OpenSIPS will 
try to first send to the servers in setid 1, and then, if those fail, to 
the servers in setid 2.


[1] http://www.opensips.org/html/docs/modules/1.11.x/dispatcher#id294368

Best Regards,

Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com

On 19.03.2015 06:17, Satish Patel wrote:

I have add extra "zone" column in subscriber table,

+--+-+
| username |  zone |
+--+-+
|1001 |1|
|1002 |2|
+--+-+


In dispatcher table I have following two Freeswitch in two groups.

+---+-++
| setid | destination  | description|
+---+--+---+
| 1 | sip:fs1.example.com  | Freeswitch-1 |
| 2 | sip:fs2.example.com  | Freeswitch-2 |
+---+--+---+


in opensips.cfg script i am query subscriber table base on incoming 
username and storing zone in avp(zone) variable, and calling same 
variable in following code


if ( !ds_select_dst("$avp(zone)", "4", "FM10"))

Question: now either user belongs to zone 1 or 2, so it is *NOT* going 
to do load-balancing between two. But if I want to allow some user to 
do load-balancing then how it will be possible in above scenario?


Can i set "setid" on fly so i can pass request along with user request 
and set same group for both switch and user call load-balance on both 
switch?


Any other idea?


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Re: [OpenSIPS-Users] ERROR:dialog:dlg_validate_dialog:

[Satish Patel]

Great! will give it a shot!

Just surprised why it is not matching both dlg and req? does
fix_route_dialog();
 has any impact on system when you have very high CPS etc?

It would be good if fix issue from root, instead of external resources
which eat CPU ticks :)

 dlg=[sip:16463737221@188.178.235.222:5061;transport=udp] ,
req=[sip:188.178.235.222;lr;ftag=840e2e35;did=1f4.ca6a6956]

On Thu, Mar 19, 2015 at 12:24 PM, Vlad Paiu  wrote:

>  Hello,
>
> Just to recap, you are saying that the Contact the user agent is sending
> is broken and you are happy that OpenSIPS is properly fixing the message,
> but you want to get rid of the ERRORs in the log ? If this is the case, you
> can use setdebug [1] for this.
>
> Try something like
>
> setdebug(-3)
> if ($DLG_status!=NULL && !validate_dialog() ) {
> xlog(" in-dialog bogus request \n");
> fix_route_dialog();
> }
> setdebug()
>
> http://www.opensips.org/Documentation/Script-CoreFunctions-2-1#toc48
>
> Best Regards,
>
> Vlad Paiu
> OpenSIPS Developerhttp://www.opensips-solutions.com
>
> On 18.03.2015 22:47, Satish Patel wrote:
>
> I know you guys are super busy in OpenSIPS 2.1 release, but any suggestion
> on above issue?
>
> On Wed, Mar 18, 2015 at 12:17 AM, Satish Patel 
> wrote:
>
>>  I am getting following error in log, I can understand my contact: and
>> Route: values mismatching here. why it is happening? is there a way to get
>> rid on this error?
>>
>>  Following is scenario. Only getting error in BYE message.
>>
>>  [UA][OpenSIP]---[Freeswitch]-[Opensip]-[SIP
>> Provide]
>>
>>
>> ERROR:dialog:dlg_validate_dialog: failed to validate remote contact:
>> dlg=[sip:16463737221@188.178.235.222:5061;transport=udp] ,
>> req=[sip:188.178.235.222;lr;ftag=840e2e35;did=1f4.ca6a6956]
>>
>>  I am using fix_route_dialog() in loose_route()
>>
>> if (has_totag()) {
>> # sequential request withing a dialog should
>> # take the path determined by record-routing
>> if (loose_route() || match_dialog())  {
>> if ($DLG_status!=NULL && !validate_dialog() ) {
>> xlog(" in-dialog bogus request \n");
>> fix_route_dialog();
>>  }
>>
>> xlog("L_INFO", "Loose route failed on
>> $hdr(route)\n");
>> if (is_method("BYE")) {
>> #setflag(ACC_DO); # 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();
>> }
>>
>> if (check_route_param("nat=yes"))
>> setflag(NAT);
>>
>> # route it out to whatever destination was set by
>> loose_route()
>> # in $du (destination URI).
>> route(relay);
>>  }  else {
>>
>> if ( is_method("ACK") ) {
>> if ( t_check_trans() ) {
>> # non loose-route, but stateful
>> ACK; must be an ACK after
>> # a 487 or e.g. 404 from upstream
>> server
>> xlog("non loose-route section\n");
>> t_relay();
>> exit;
>> } else {
>> # ACK without matching
>> transaction ->
>> # ignore and discard
>> xlog("ACK without matching
>> transaction\n");
>> exit;
>> }
>> }
>> xlog("L_INFO", "destination uri after
>> loose_route: <$du>\n");
>> sl_send_reply("404","Not here");
>> }
>> exit;
>> }
>>
>>
>>
>>
>>
>>
>>
>
>
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Re: [OpenSIPS-Users] error 483

[Vlad Paiu]

Hello,

483 usually means 'Too Many Hops'. If you do a SIP trace on the server, 
do you see OpenSIPS looping the request to itself ? Maybe the SIP phone 
sends the IP of the server instead of the domain that you have 
configured, and your script is configured to route out such requests.


Best Regards,

Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com

On 19.03.2015 04:15, Carlos Cruz wrote:


Hi;

Can someone tell me why  or where I may find the  info I need; I'm 
able to register external remote phones (hard phones), but the 
internal phone (soft phones) within the same network as the OpenSIPS 
test server report error 483.


Thanks!!

Carlos



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Re: [OpenSIPS-Users] ERROR:dialog:dlg_validate_dialog:

[Vlad Paiu]

Hello,

Just to recap, you are saying that the Contact the user agent is sending 
is broken and you are happy that OpenSIPS is properly fixing the 
message, but you want to get rid of the ERRORs in the log ? If this is 
the case, you can use setdebug [1] for this.


Try something like

setdebug(-3)
if ($DLG_status!=NULL && !validate_dialog() ) {
xlog(" in-dialog bogus request \n");
fix_route_dialog();
}
setdebug()

http://www.opensips.org/Documentation/Script-CoreFunctions-2-1#toc48

Best Regards,

Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com

On 18.03.2015 22:47, Satish Patel wrote:
I know you guys are super busy in OpenSIPS 2.1 release, but any 
suggestion on above issue?


On Wed, Mar 18, 2015 at 12:17 AM, Satish Patel > wrote:


I am getting following error in log, I can understand my contact:
and Route: values mismatching here. why it is happening? is there
a way to get rid on this error?

Following is scenario. Only getting error in BYE message.

[UA][OpenSIP]---[Freeswitch]-[Opensip]-[SIP
Provide]


ERROR:dialog:dlg_validate_dialog: failed to validate remote
contact: dlg=[sip:16463737221
@188.178.235.222:5061;transport=udp] ,
req=[sip:188.178.235.222;lr;ftag=840e2e35;did=1f4.ca6a6956]

I am using fix_route_dialog() in loose_route()

if (has_totag()) {
# sequential request withing a dialog should
# take the path determined by record-routing
if (loose_route() || match_dialog())  {
if ($DLG_status!=NULL &&
!validate_dialog() ) {
xlog(" in-dialog bogus request \n");
fix_route_dialog();
 }

xlog("L_INFO", "Loose route failed on
$hdr(route)\n");
if (is_method("BYE")) {
#setflag(ACC_DO); # 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();
}

if (check_route_param("nat=yes"))
setflag(NAT);

# 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
xlog("non loose-route
section\n");
t_relay();
exit;
} else {
# ACK without matching
transaction ->
# ignore and discard
xlog("ACK without matching
transaction\n");
exit;
}
}
xlog("L_INFO", "destination uri after
loose_route: <$du>\n");
sl_send_reply("404","Not here");
}
exit;
}









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Re: [OpenSIPS-Users] Again BLF and Presence with Snom 7xx phones and OpenSips

[sevpal]

You need to handle the in-dialog SUBSCRIBE requests. eg:

if has_totag() {
...
if (loose_route()) {
...
} else {
...
if (is_method("SUBSCRIBE")) { 
  route(2);
  exit; 
}
...
  }
  ...
}

From: Bogdan-Andrei Iancu 
Sent: Thursday, February 26, 2015 7:56 AM
To: OpenSIPS users mailling list ; mailto:michele.pina...@unisi.it 
Subject: Re: [OpenSIPS-Users] Again BLF and Presence with Snom 7xx phones and 
OpenSips

Hi Michele,

The problem in your script is that you do not handle the sequential (in-dialog) 
SUBSCRIBE requests (as you have the second one in your trace, ending with 404 
and terminating the subscription).

In the " if ( has_totag() ) " block, you have:
} else { 
if (is_method("SUBSCRIBE") && $rd == "127.0.0.1:5060") { # CUSTOMIZE ME

The $rd detection does not cover all your cases, as you configure the presence 
module to advertise as SIP contact "sip:prese...@voip.unisi.it:5060". So, the 
test fails.

You can adapt the test like:
if (is_method("SUBSCRIBE") && $rd == "voip.unisi.it") { # CUSTOMIZE ME

Or set the contact in presence with the real IP:
modparam("presence", "server_address", 
mailto:sip:presence@127.0.0.1:5060)

Best regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.comOn 24.02.2015 12:04, Michele Pinassi wrote:

  Hi all,

  I'm still stuck on this issue: BLF not working. For example, on my SNOM 760 
(ext 5002) i activated BLF for some ext, like 5020. Using SIPGREP i saw:

  SUBSCRIBE sip:5...@voip.unisi.it;user=phone SIP/2.0.
  Via: SIP/2.0/UDP 172.20.1.10:57286;branch=z9hG4bK-nprg3gvnk4q1;rport.
  From: mailto:sip:5...@voip.unisi.it:5060;tag=nyux2omhly.
  To: mailto:sip:5...@voip.unisi.it;user=phone.
  Call-ID: 3944ec54dc20-pfzjpjhrpm6p.
  CSeq: 2 SUBSCRIBE.
  Max-Forwards: 70.
  Contact: mailto:sip:5002@172.20.1.10:57286;reg-id=1.
  Event: dialog.
  Accept: application/dialog-info+xml.
  User-Agent: snom760/8.7.3.25.9.
  Proxy-Authorization: Digest 
  Expires: 3600.
  Content-Length: 0.

  SIP/2.0 200 OK.
  Via: SIP/2.0/UDP 
172.20.1.10:57286;received=172.20.1.10;branch=z9hG4bK-nprg3gvnk4q1;rport=57286.
  From: mailto:sip:5...@voip.unisi.it:5060;tag=nyux2omhly.
  To: 
mailto:sip:5...@voip.unisi.it;user=phone;tag=f315b2d58ae8829149b784764c5a40e3-163d.
  Call-ID: 3944ec54dc20-pfzjpjhrpm6p.
  CSeq: 2 SUBSCRIBE.
  Expires: 3600.
  Contact: mailto:sip:prese...@voip.unisi.it:5060.
  Server: OpenSIPS (1.11.3-tls (i386/linux)).
  Content-Length: 0.

  NOTIFY sip:5002@172.20.1.10:57286 SIP/2.0.
  Via: SIP/2.0/UDP 172.20.1.2:5060;branch=z9hG4bKdb02.83d58916.0.
  To: mailto:sip:5...@voip.unisi.it;tag=nyux2omhly.
  From: mailto:sip:5...@voip.unisi.it;tag=f315b2d58ae8829149b784764c5a40e3-163d.
  CSeq: 1 NOTIFY.
  Call-ID: 3944ec54dc20-pfzjpjhrpm6p.
  Max-Forwards: 70.
  Content-Length: 147.
  User-Agent: OpenSIPS (1.11.3-tls (i386/linux)).
  Event: dialog.
  Contact: mailto:sip:prese...@voip.unisi.it:5060.
  Subscription-State: active;expires=3600.
  Content-Type: application/dialog-info+xml.
  .
  
  mailto:sip:5...@voip.unisi.it/>

  SIP/2.0 200 Ok.
  Via: SIP/2.0/UDP 172.20.1.2:5060;branch=z9hG4bKdb02.83d58916.0.
  From: mailto:sip:5...@voip.unisi.it;tag=f315b2d58ae8829149b784764c5a40e3-163d.
  To: mailto:sip:5...@voip.unisi.it;tag=nyux2omhly.
  Call-ID: 3944ec54dc20-pfzjpjhrpm6p.
  CSeq: 1 NOTIFY.
  Content-Length: 0.

  SUBSCRIBE sip:prese...@voip.unisi.it:5060 SIP/2.0.
  Via: SIP/2.0/UDP 172.20.1.25:32768;branch=z9hG4bK-lbgnea3kuorq;rport.
  From: mailto:sip:5...@voip.unisi.it:5060;tag=w8vp9q5iyn.
  To: 
mailto:sip:5...@voip.unisi.it;user=phone;tag=f315b2d58ae8829149b784764c5a40e3-29cc.
  Call-ID: 54ec3a578c9e-klgn0s3i32zo.
  CSeq: 75 SUBSCRIBE.
  Max-Forwards: 70.
  Contact: mailto:sip:5007@172.20.1.25:32768;reg-id=1.
  Event: dialog.
  Accept: application/dialog-info+xml.
  User-Agent: snom710/8.7.3.25.9.
  Expires: 3600.
  Content-Length: 0.

  SIP/2.0 404 Not here.
  Via: SIP/2.0/UDP 
172.20.1.25:32768;received=172.20.1.25;branch=z9hG4bK-lbgnea3kuorq;rport=32768.
  From: mailto:sip:5...@voip.unisi.it:5060;tag=w8vp9q5iyn.
  To: 
mailto:sip:5...@voip.unisi.it;user=phone;tag=f315b2d58ae8829149b784764c5a40e3-29cc.
  Call-ID: 54ec3a578c9e-klgn0s3i32zo.
  CSeq: 75 SUBSCRIBE.
  Server: OpenSIPS (1.11.3-tls (i386/linux)).
  Content-Length: 0.

  NOTIFY sip:5002@172.20.1.10:57286 SIP/2.0.
  Via: SIP/2.0/UDP 172.20.1.2:5060;branch=z9hG4bKdbe9.7966c706.0.
  To: mailto:sip:5...@voip.unisi.it;tag=iklb1qjh1v.
  From: mailto:sip:5...@voip.unisi.it;tag=f315b2d58ae8829149b784764c5a40e3-b571.
  CSeq: 2 NOTIFY.
  Call-ID: ee35ec54a72b-draf1nwo4qn7.
  Max-Forwards: 70.
  Content-Length: 0.
  User-Agent: OpenSIPS (1.11.3-tls (i386/linux)).
  Event: dialog.
  Contact: mailto:sip:prese...@voip.unisi.it:5060.
  Subscription-State: terminated;reason=timeout.

  SIP/2.0 200 Ok.
  Via: SIP/2.0/UDP 172.20.1.2:5060;branch=z9hG4bKdbe9.7966c706.0.
  From: mailto:sip:5...@voip.unisi.i

[OpenSIPS-Users] Presence Error messages

[Peter Kust]

I am attempting to troubleshoot what I think is a presence/b2b_sca issue.

I keep getting an error message from presence as follows:
ERROR:presence:update_presentity: No E_Tag match 
[ff5ad69c9be06cffaa136492f4fb3b50]

At some point during the day, I will see this error message:
ERROR:presence:handle_subscribe: in event specific subscription handling

At this point, an outbound call from a line appearance provisioned to the 
b2b_sca module fails.  The outbound call is being attempted from a Cisco 
SPA525G2, and the message on the phone screen shows "no line"-despite happening 
at a time when it is known there are no calls on the system that I can see.  
The observed behavior of the phone is to show "No line", and the phone itself 
gets a

What do these error messages mean?  What is the system trying to tell me?  I am 
trying to get my brain around what the error messages are saying so I can 
figure out where to look next in my troubleshooting.  The proxy is functioning 
well in all other respects.

Cordially,

Peter Nayland Kust
Director of Technologies
BusinesSuites
24624 Interstate 45 North, Suite 200
Houston, TX 77386
Tel:  281.378.8051
Fax:  855.287.6961
peter.k...@businessuites.com
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Re: [OpenSIPS-Users] SIps as SBC

[Varadhan Work]

Checkout Blox.org

Thanks
Varadhan M

On Thu, Mar 19, 2015 at 7:33 PM, malik sherif  wrote:

>
>
> Can SIPS can be used as an SBC?
> Thanks
> Abdul
>
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Re: [OpenSIPS-Users] SIps as SBC

[Terrance Devor]

Who?​

T
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[OpenSIPS-Users] SIps as SBC

[malik sherif]

Can SIPS can be used as an SBC?
Thanks
Abdul
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Re: [OpenSIPS-Users] Which port router open

[Satish Patel]

Default SIP port 5060 UDP also you need media port call RTP

--
Sent from my iPhone

> On Mar 19, 2015, at 7:29 AM, Mattia Adducchio  wrote:
> 
> Hello Everyone,
> 
> I'm trying to setup my personal sip server. In this moment it works only in 
> my network, but now I want to open the router port for external access.
> 
> I have open the port 5060 but maybe it's not enough.
> 
> Thank you,
> Mattia
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Re: [OpenSIPS-Users] 500 Server error in REGISTER message

[Babil (Golam Sarwar)]

There's a mismatched curly-brace issue in your configuration.

Brace at line 2 matches with brace at line 18. Nothing matches with the
closing curly-brace at line 19. I think we are missing a curly-brace at
line 15.

My two-cents to the OpenSIPS team would be consider verbose curly-braces
for the configuration script. Python/C like tab-based indenting might
seem to improve readability and keep the code concise, but it introduces
these unintended errors.


```
  1 if (is_method("REGISTER"))
  2 {
  3 # authenticate the REGISTER requests (uncomment to enable auth)
  4 ##if (!www_authorize("", "subscriber"))
  5 ##{
  6 ## www_challenge("", "0");
  7 ## exit;
  8 ##}
  9 ##
 10 ##if (!db_check_to())
 11 ##{
 12 ## sl_send_reply("403","Forbidden auth ID");
 13 ## exit;
 14 ##}
 15 if (!save("location"))
 16 sl_reply_error();
 17 exit;
 18 }
 19 }
 20
```

On 3/17/15 10:52 AM, Satish Patel wrote:
> I got those code from Book Building Telephony System with OpenSIPS 1.6
> 
> Here is the code from book
> 
> if (is_method("REGISTER"))
> {
> # authenticate the REGISTER requests (uncomment to enable auth)
> ##if (!www_authorize("", "subscriber"))
> ##{
> ## www_challenge("", "0");
> ## exit;
> ##}
> ##
> ##if (!db_check_to())
> ##{
> ## sl_send_reply("403","Forbidden auth ID");
> ## exit;
> ##}
> if (!save("location"))
> sl_reply_error();
> exit;
> }
> }
> 
> 
> 
> On Tue, Mar 17, 2015 at 1:48 PM, Satish Patel  > wrote:
> 
> Eric,
> 
> I found what was the issue, I sent you REGISTER method snippet
> before, if you look at it, If remove/comment out "sl_reply_error();"
>  line in following code, it stopped sending 500 Error. Very
> interesting..  Do you think i need to put that in "curly braces" { } ?  
> 
>  if (!save("location"))
> xlog("L_ERR", "Saving contact failed - M=$rm
> RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");
> sl_reply_error();
> 
> exit;
> }
> 
> 
> On Tue, Mar 17, 2015 at 1:27 PM, Satish Patel  > wrote:
> 
> Even after disabled "siptrace" it is happening. no luck :(
> 
> On Tue, Mar 17, 2015 at 1:20 PM, Eric Tamme  > wrote:
> 
> Turn of your sip tracing and see if the issue occurs.  Its
> running some sl_callbacks (which i assume are realated to
> siptrace).
> 
> 
> 
> On 03/17/2015 11:19 AM, Satish Patel wrote:
>> I haven't done anything related "stateless".  also in my
>> config, i haven't manually specify that 500 error anywhere
>> where i can doubt.  I don't know from where it is coming.
>> must be internally from opensips. 
>>
>> On Tue, Mar 17, 2015 at 1:14 PM, Eric Tamme
>> mailto:e...@uphreak.com>> wrote:
>>
>> Ah - nm, i see it in an sl callback
>>
>> Mar 17 22:19:01 sip2 
>> /usr/local/opensips-2-head/sbin/opensips[31285]: DBG:sl:sl_reply_error: 
>> error text is Server error occurred (1/SL)
>>
>> ... so are you doing anything statless in your config?  This 
>> looks like it might be siptrace related.
>>
>>
>>
>> On 03/17/2015 11:11 AM, Eric Tamme wrote:
>>> I do not see the 500 from opensips in this log.
>>>
>>> On 03/17/2015 11:07 AM, Satish Patel wrote:
 Here is the debug 4 logs  http://pastebin.com/CdPxFrNp

 173.48.111.111  - UA 
 188.79.242.164  - OpenSIPs

 On Tue, Mar 17, 2015 at 12:45 PM, Eric Tamme
 mailto:e...@uphreak.com>> wrote:

 This is a ladder diagram, not a sip trace.  A
 ladder diagram is not useful in this case.

 Turn your debug up to 4, capture the log of the
 register/500 happening and submit a link to the
 pastebin.  DO NOT paste the contents into an email.


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[OpenSIPS-Users] Which port router open

[Mattia Adducchio]

Hello Everyone,

I'm trying to setup my personal sip server. In this moment it works only 
in my network, but now I want to open the router port for external access.


I have open the port 5060 but maybe it's not enough.

Thank you,
Mattia

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Re: [OpenSIPS-Users] [Release] OpenSIPS 2.1 Release Candidate is out !

[Răzvan Crainea]

On 03/18/2015 08:26 PM, Bogdan-Andrei Iancu wrote:

Hi all,

I'm really excited to announce that after almost one year of hard work,
the OpenSIPS 2.1 RC is now available for download.
  http://opensips.org/pub/opensips/latest/src/

The 2.1 version is a major step in OpenSIPS evolution, encapsulating
major redesign (protocols, timers, reactors, async support) and valuable
additions (websockets, Fraud Detection, SIP Compression, Async DB
queries, Topology Hiding and more).
  http://www.opensips.org/About/Version-2-1-0-Notes

OpenSIPS 2.1 is the first in line benefiting of the New Design (NG)
brainstorming and work - where many radical and innovative concepts have
been introduced, to put OpenSIPS on the top of the SIP server engines -
in terms of efficiency, scalability, features set and flexibility.

OpenSIPS 2.1 is the result of the whole community effort - there were
many people involved in the design, in implementation, testing and
reporting - and we want to thank you all for making this release possible:
  https://github.com/OpenSIPS/opensips/blob/2.1/CREDITS

Version 2.1 is a release candidate - the stable GA version is to be
release in the next 1.5 month, after more in depth testing .

In the mean while, enjoy the fantastic ride with OpenSIPS 2.1 :)


For a better experience with the system provisioning and user 
management, we recommend you to use the new OpenSIPS Control Panel 
6.1[1], which is now fully compatible with OpenSIPS 2.1.


Many greetings from the OpenSIPS team [2]!

[1] https://sourceforge.net/projects/opensips-cp
[2] http://opensips.org/images/team/opensips-team.jpg

Best regards,

Răzvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com

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Re: [OpenSIPS-Users] Jitsi sent REGISTER, Opensips received nothing

[Gordon E. Sims, Jr.]

I would take a look at the network in between.  If Jitsi is sending out 
packets, and opensips is not receiving, then look at routers, switches and 
firewall in between.  Are you able to ping the opensips device from the same 
network that Jitsi is on?  This should verify basic routing, then start looking 
at any potential firewall rules.  If you are not getting ping requests, then 
you will want to check your routing with opensips. Make sure can ping outside 
of your network that Jitsi is on as well.

Just my thoughts,

Gordon

Sent from my iPhone6

> On Mar 19, 2015, at 1:23 AM, jacky <542590...@qq.com> wrote:
>
> I have a test with one Jitsi using Opensips on the Internet
> Wireshark showed me that Jitsi sent several "REGISTER" packages, in the same
> time I used the command "tcpdump" to listen on the Opensips Server , but got
> nothing.
> Anyway, on the server, opensips, rtpproxy, media-dispatcher, media-relay are
> running!
> what happened to opensips server? why it won't response to distanced
> request?
>
> I appreciate your opinion, thanks a lot!
>
> Best regards
>
>
>
>
> --
> View this message in context: 
> http://opensips-open-sip-server.1449251.n2.nabble.com/Jitsi-sent-REGISTER-Opensips-received-nothing-tp7596019.html
> Sent from the OpenSIPS - Users mailing list archive at Nabble.com.
>
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