Re: [OpenSIPS-Users] B2B BYEs

[Bogdan-Andrei Iancu]

Hi Søren,

It seems that the callee route set is stored in db (see the b2b_entities 
table, the route1 field 
="")


Can you run the b2be_list MI command before and after the restart, to 
see if indeed, the callee route set is not properly restored ?


Regards,

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

On 15.01.2016 12:21, Søren Andersen wrote:


Yup.

SELECT * FROM b2b.b2b_logic;

# id, si_key, scenario, sstate, next_sstate, sparam0, sparam1, 
sparam2, sparam3, sparam4, sdp, lifetime, e1_type, e1_sid, e1_from, 
e1_to, e1_key, e2_type, e2_sid, e2_from, e2_to, e2_key, e3_type, 
e3_sid, e3_from, e3_to, e3_key


'226', '360.0', 'StofaB2B', '1', '1', 'sip:...@xx.xx.xx.120', '', 
'', '', '', '', '1452896043', '0', 'server1', 
'sip:...@oxd3-sxw.xx.txxx.xxx;user=phone', 
'sip:x...@xx.xx.xx.117:5060', 'B2B.39.160', '1', 'client1', 
'sip:x...@oxx3-sx.xx.xxx.xxx;user=phone', 
'sip:...@xx.xx.xx.120', 'B2B.448.7695796', NULL, NULL, NULL, NULL, 
NULL


SELECT * FROM b2b.b2b_entities;

# id, type, state, ruri, from_uri, to_uri, from_dname, to_dname, tag0, 
tag1, callid, cseq0, cseq1, contact0, contact1, route0, route1, 
sockinfo_srv, param, lm, lrc, lic, leg_cseq, leg_route, leg_tag, 
leg_contact, leg_sockinfo


'316', '0', '5', 'sip:...@xx.xx.xx.117:5060;user=phone', 
'sip:x...@osx3-sxw.xx..xxx;user=phone', 
'sip:X04@10.250.224.22;user=phone', '', '', 'mvgh6xvkp7z', 
'B2B.39.160', '1q2rqu6-3232128...@osdx-ssw.xx.xxx.xx', '32736', '1', 
'sip:x...@xx.xx.xx.36:5060;transport=udp', 
'sip:XX.XX.XX.117:5060', '', '', 
'udp:XX.XX.XX.117:5060', '360.0', '0', '200', '32736', NULL, NULL, 
NULL, NULL, NULL


'317', '1', '5', '', 'sip:x...@osdx-ssw.xx..xx;user=phone', 
'sip:x...@xx.xx.xx.120', '', '', 
'a9ff31f61f2601c916f6044c2e20e16a-3c47', '2761431845', 
'B2B.448.7695796', '32737', '1', 'sip:XX.XXX.XXX.117:5060', 
'sip:XX04@10.101.16.179:5060;transport=udp', 
'', 
'', 'udp:XX.XX.XX.117:5060', '360.0', '4', '0', '32737', '32737', 
'', 
'2761431845', 'sip:XX04@10.101.16.179:5060;transport=udp', NULL


/Søren

*Fra:*Bogdan-Andrei Iancu [mailto:bog...@opensips.org]
*Sendt:* 15. januar 2016 11:07
*Til:* Søren Andersen ; users@lists.opensips.org
*Emne:* Re: SV: SV: [OpenSIPS-Users] B2B BYEs

Hi Søren,

In the scenario with the restart, after the restart, the sequential 
requests generated by b2b is missing the Route header - and it is 
bypassing the proxy - can you ppst the DB records for b2b you have 
during the restart?


Regards,

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

On 15.01.2016 10:57, Søren Andersen wrote:

Hi Bogdan-Andrei,

Sure. I’ve attached the two files.

 (B2B server)

 (OpenSIPS Proxy)

 (SIP Client)

My scenario:











server1





client1

message



1





 

1





I’m doing this because my SIP provider can’t handle multi dialogs,
so if I first send a call to the sip client and if the call is
unanswered I can’t redirect the call to voicemail since this will
generate a new to_tag. And this can’t my SIP provider not handle.
So therefore I need some B2B logic to make this happen for me.

/Søren

*Fra:*Bogdan-Andrei Iancu [mailto:bog...@opensips.org]
*Sendt:* 14. januar 2016 16:41
*Til:* Søren Andersen  ;
OpenSIPS users mailling list 

*Emne:* Re: SV: [OpenSIPS-Users] B2B BYEs

Hi Søren,

Do you have an ngrep capture (taken from opensips b2b, covering
all calls/legs involved in the scenario) showing the call with the
B2B restarted ?

Regards,


Bogdan-Andrei Iancu

OpenSIPS Founder and Developer

http://www.opensips-solutions.com

On 13.01.2016 14:39, Søren Andersen wrote:

Hi Bogdan-Andrei,

Only if I reload the B2B the BYEs is sent directly to the
client. – But the funny thing is this only happens if the
client is receives a call.  If the client initialize the call
everything works fine.

/Søren

*Fra:*Bogdan-Andrei Iancu [mailto:bog...@opensips.org]
*Sendt:* 13. januar 2016 11:39
*Til:* OpenSIPS users mailling list 
; Søren Andersen
 
*Emne:* Re: [OpenSIPS-Users] B2B BYEs

Hi Søren,

With or without restarting the B2B isn;t the BYE sent by
client to the B2B instance ?

Regards,

Re: [OpenSIPS-Users] YUM repository downtime

[Bogdan-Andrei Iancu]

Hey, it seems to be up again !!! :))

Regards,

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

On 18.01.2016 10:24, Nick Altmann wrote:
Thank you. Yes, you’re right. It’s still down. New repository will be 
up a bit later today.


--
Nick

2016-01-18 11:03 GMT+03:00 Markus Rydh >:


Nick, the yum repo seems to be down for the last few days. Any
updates on
when it might be available? Appreciate your work on hosting the repo!

// Markus



--
View this message in context:

http://opensips-open-sip-server.1449251.n2.nabble.com/YUM-repository-downtime-tp7600572p7600780.html
Sent from the OpenSIPS - Users mailing list archive at Nabble.com.

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Re: [OpenSIPS-Users] Extra Tables Creation Error

[Bogdan-Andrei Iancu]

Hi Nathaniel,

Do you see the other "emergency" module related table (like 
emergency_report or emergency_service_provider) created ?


Regards,

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

On 18.01.2016 21:38, Nathaniel L. Keeling III wrote:

Hello,

I was running the "opensipsdbctl create" to create the tables in 
postgresql 8.4 and received this error when creating the extra tables 
for Opensips 2.1:


ERROR:  relation "emergency_routing" does not exist
ERROR:  relation "emergency_routing_id_seq" does not exist
ERROR: Grant privileges to extra tables failed!

I checked the database and neither the emergency_routing or the 
emergency_report tables were created. All other table creations seem 
to be ok.


Thanks

Nathaniel L Keeling

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Re: [OpenSIPS-Users] WSS Client did not present a TLS certificate

[Bogdan-Andrei Iancu]

Hi Sebastian,

That message is just an INFO (not an error) - you say TLS handshake 
fails on opensips side as it expects a certificate from the end point ?


Regards,

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

On 18.01.2016 06:32, Sebastian Sastre wrote:

I’ve been trying to setup WSS using 2.2 latest branch.

When trying to open the web socket i get “ Client did not present a 
TLS certificate” . Im using the included default ssl certs for 
the server to avoid mistakes . What certificate is the user supposed 
to present?


I tried using sip.js and jssip to connect without any luck. i also 
tried disabling cert requirement but didn’t work.


—— Config ——-

listen=wss:123.456.789.987:5060
listen=tls:123.456.789.987:5061
listen=wss:123.456.789.987:443

load module "proto_udp.so"
load module “proto_tls.so”
loadmodule "proto_wss.so"

loadmodule "tls_mgm.so"
modparam("tls_mgm", "certificate", "/etc/opensips/tls/rootCA/cacert.pem")
modparam("tls_mgm", "private_key", 
"/etc/opensips/tls/rootCA/private/cakey.pem")

modparam("tls_mgm", "ca_list", "/etc/opensips/tls/rootCA/cacert.pem")
modparam("tls_mgm", "ca_dir", "/etc/opensips/tls/rootCA/")
modparam("tls_mgm", "require_cert", "0")
modparam(“tls_mgm", "verify_cert", "0")


——- Logs ——-
/sbin/opensips[12468]: INFO:core:probe_max_sock_buff: using snd buffer 
of 416 kb
/sbin/opensips[12468]: INFO:core:init_sock_keepalive: TCP keepalive 
enabled on socket 37
/sbin/opensips[12460]: INFO:proto_wss:ls_accept: New TLS connection 
from xx.xx.xx.xx:50815 accepted
/sbin/opensips[12460]: INFO:proto_wss:tls_accept: Client did not 
present a TLS certificate
/sbin/opensips[12460]: INFO:proto_wss:ls_dump_cert_info: tls_accept: 
local TLS server certificate subject: 
/CN=OpenSIPS/ST=opensips.org/C=IP/emailAddress=t...@opensips.org/O=opensips.org 
, 
issuer: 
/CN=OpenSIPS/ST=opensips.org/C=IP/emailAddress=t...@opensips.org/O=opensips.org 




Thanks !



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Re: [OpenSIPS-Users] registrar contact delete

[Bogdan-Andrei Iancu]

Hi Tito,

What do you understand by socket ?? The TCP connection ? or ?

Best regards,

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

On 15.01.2016 22:56, Tito Cumpen wrote:

Group,

I am currently using a max contact param within the registrar module. 
I am using if (!save("location","f")) to force the new 
registration the case that the max has been reached.It appears that 
the oldest registration is deleted. Will the respective socket to the 
deleted contact be closed after the contact has been deleted ? Or will 
it linger until the previous registration expiry time concludes?



Thanks,
Tito


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Re: [OpenSIPS-Users] Request Timeout on INVITE After Idle Registration

[Bogdan-Andrei Iancu]

Indeed, for UDP, you can clearly see the INVITEs going to callee device 
10.45.155.206  (and the retransmissions), followed by the opensips  
generated 408 timeout.


So this is clear, I will take a look at the logs for the TLS scenario.

Regards,

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

On 13.01.2016 08:46, Nabeel wrote:

and here is the more detailed SIP trace for UDP:

http://pastebin.com/UfQJJz3Y



On 13 January 2016 at 06:19, Nabeel > wrote:


Hi Bogdan,

I changed log_stderror=yes and log_facility=LOG_DAEMON.  Now I see
some more in the log.  Do you see anything obviously wrong?

http://pastebin.com/MzJW1P1S

On 12 January 2016 at 09:10, Bogdan-Andrei Iancu
> wrote:

Hi Nabeel,

Be sure you are looking into the right log file - maybe the
debug level is redirected by your syslog to another log
file... Debug level 4 is the most verbose one in opensips.

Regards,

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

On 08.01.2016 21:04, Nabeel wrote:


Hi Bogdan,

I have the following near the top of my config file:

## Global Parameters #

debug=4
log_stderror=no
log_facility=LOG_LOCAL1

The log I posted earlier is from opensips running with these
configurations.

On 8 Jan 2016 3:49 pm, "Bogdan-Andrei Iancu"
> wrote:

Hi Nabeel,

have you tried running opensips is debug mode (level 4)
to see what it is doing with the request ?

Regards,

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

On 07.01.2016 11:37, Nabeel wrote:


Hi Bogdan,

I used the tshark command as explained here on page 14:

http://opensips.org/pub/events/2015-05-12_OpenSIPS-Summit_Amsterdam/Lorenzo_

Mangani-OpenSIPS

_

Summit2015-SIPCapture.pdf



tshark -o "ssl.desegment_ssl_records: TRUE"  -o
"ssl.desegment_ssl_application_data: TRUE"  -o
"ssl.keys_list:
162.249.6.110,5061,sip,/install/tls/domain.com-key.pem"
-i eth0  -f "tcp port 5061"

I'm using a command line version of Linux without a
graphic UI, so I could not "configure Wireshark to
decide TLS" as mentioned in that document, however I did
pass the private key in the command as shown above.

Does tshark require configuring to decode TLS, other
than passing the private key in the command?

Hi Nabeel,

Indeed, the 408 seems generated by OpenSIPS (after 5
seconds). Such reply is generated only if the the
request was actually sent out (if no request sent, there
is no timeout). But the network capture does not show
anything :( ... maybe wrong capturing ?

So you see anything in the logs ? have you tried to run
with debug level 4 ?

Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 06.01.2016 23:07, Nabeel wrote:
I managed to capture the SIP traffic with Wireshark. 
It seems that the party generating the 408 reply is

OpenSIPS, not the callee.  OpenSIPS does not seem to
forward the call to the callee at all.

Below are traces showing a successful call and a call
with Request Timeout.
The server IP is 162.249.6.110, the caller IP is
92.40.249.9, and the callee IP is 188.29.165.24.

Trace for a successful call:

http://pastebin.com/2xn0bkEU

Trace for a call with Request Timeout:

http://pastebin.com/WR7BA6pj

Please advise what may be causing this.
















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Re: [OpenSIPS-Users] registrar contact delete

[Tito Cumpen]

Bogdan,

Yea will the server close tcp connection to the client if the record is
replace as I mentioned ?

Thanks,
Tito
On Jan 19, 2016 5:24 AM, "Bogdan-Andrei Iancu"  wrote:

> Hi Tito,
>
> What do you understand by socket ?? The TCP connection ? or ?
>
> Best regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
> On 15.01.2016 22:56, Tito Cumpen wrote:
>
> Group,
>
> I am currently using a max contact param within the registrar module. I am
> using  if (!save("location","f")) to force the new
> registration the case that the max has been reached.It appears that the
> oldest registration is deleted. Will the respective socket to the deleted
> contact be closed after the contact has been deleted ? Or will it linger
> until the previous registration expiry time concludes?
>
>
> Thanks,
> Tito
>
>
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Re: [OpenSIPS-Users] Opensips 1.8 ds_list behavior in dispatcher

[Bogdan-Andrei Iancu]

Hi,

This was fixed starting with 1.11, so you should upgrade your version as 
1.8 is deprecated.


Regards,

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

On 18.01.2016 20:22, Gunjan Korlekar wrote:

Hi Bogdan,

Thanks for the tip! I checked the opensips logs and I saw the 
following message when we executed a 'dispatcher reload' command -


WARNING:dispatcher:ds_load_db: no dispatching data in the db -- empty 
destination set


Digging into the code for the ds_load_db function in dispatch.c I 
found the following -


if(nr_rows == 0)
{
LM_WARN("no dispatching data in the db -- empty destination set\n");
ds_dbf.free_result(ds_db_handle, res);
return 0;
}

So, essentially when the dispatcher table is empty, the reload 
function does not sync the secondary memory with the primary and 
returns, which causes stale data to remain in the primary, which is 
why 'fifo ds_list' command never updated after the 'dispatcher rmgw'.


Is there a patch or a fix you can provide for the above in the 
opensips 1.8 version of the source?


Thanks,
Gunjan

On Fri, Jan 15, 2016 at 1:31 AM, Bogdan-Andrei Iancu 
> wrote:


Hi Gunjan,

have you checked the opensips logs for any potential errors ?
maybe the reload fails.

Regards,

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

On 14.01.2016 19:10, Gunjan Korlekar wrote:

Hi Bogdan,

Thanks for the clarification regarding the memory and the ds_list
command.
From what I see in the opensipsctl source, the 'opensipsctl
dispatcher rmgw' command also runs a 'ds_reload' after running
the database query to remove the entry. Hence in my above
sequence of operations, the primary storage should have synced
with the secondary storage before I ran 'fifo ds_list' and should
not have expected data in the ds_list output -

./opensipsctl dispatcher show
dispatcher gateways

++---+--++---++---+-+
| id | setid | destination  | socket | flags | weight | attrs
| description |

++---+--++---++---+-+
|  5 |40 | sip:2.2.2.6:5060  | NULL |  
  0 |  1 |   | test1   |


++---+--++---++---+-+

 ./opensipsctl dispatcher rmgw 5 -- ds_reload here
 ./opensipsctl fifo ds_list <<<- should show synced output ?
SET_NO:: 1
SET:: 40
URI:: sip:2.2.2.6:5060  flag=A
 ./opensipsctl dispatcher show
dispatcher gateways

I tried the above by running 'fifo ds_reload' explicitly after
'dispatcher rmgw' as well. However I did not see a difference in
the result. Can you please help in understanding the issue?

Thanks,
Gunjan

On Thu, Jan 14, 2016 at 1:39 AM, Bogdan-Andrei Iancu
> wrote:

Hi Gunjan,

In dispatcher there are 2 storages:
- primary storage which is in memory
- secondary storage (read only at startup or reload)
which is the DB

First is about synchronizing the 2 storages (from secondary
to primary via a ds_reload command). AFter that is about
which storage you are querying for data. Like the
"opensipsctl fifo ds_list" (via MI command) gets data from
memory cache while the "opensipsctl dispatcher show" simply
dumps db content.

Regards,

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

On 14.01.2016 03:44, Gunjan Korlekar wrote:

Hello,

I am observing some unusual behavior of the ds_list command
when adding and removing gateways in the dispatcher table
for opensips 1.8

I am running the following sequence of commands -

./opensipsctl dispatcher addgw 40 sip:2.2.2.4:5060
 0 'test1'
 ./opensipsctl dispatcher addgw 40 sip:2.2.2.5:5060
 0 'test1'
 ./opensipsctl dispatcher addgw 40 sip:2.2.2.6:5060
 0 'test1'

When I view the dispatcher list using ds_list I see-
 ./opensipsctl fifo ds_list
SET_NO:: 1
SET:: 40
URI:: sip:2.2.2.4:5060  flag=A
URI:: sip:2.2.2.5:5060  flag=A
URI:: sip:2.2.2.6:5060  flag=A
and the ./opensipsctl dispatcher show output is as follows -
dispatcher gateways

++---+--++---++---+-+
| id | setid | destination  | socket | flags | weight |
attrs | description |

Re: [OpenSIPS-Users] Request Timeout on INVITE After Idle Registration

[Bogdan-Andrei Iancu]

According to the logs, the INVITE is sent out :

Jan 13 06:12:08 server1 opensips: Jan 13 06:12:08 [17457] 
DBG:core:_shm_resize: resize(0) called
Jan 13 06:12:08 server1 opensips: Jan 13 06:12:08 [17457] 
DBG:core:mk_proxy: doing DNS lookup...
Jan 13 06:12:08 server1 opensips: Jan 13 06:12:08 [17457] 
DBG:tm:set_timer: relative timeout is 50
Jan 13 06:12:08 server1 opensips: Jan 13 06:12:08 [17457] 
DBG:tm:insert_timer_unsafe: [4]: 0x7f8518e24938 (709250)
Jan 13 06:12:08 server1 opensips: Jan 13 06:12:08 [17457] 
DBG:tm:insert_timer_unsafe: [0]: 0x7f8518e24968 (7097)
Jan 13 06:12:08 server1 opensips: Jan 13 06:12:08 [17457] 
DBG:tm:t_relay_to: new transaction fwd'ed



there, are several retransmissions:

Jan 13 06:12:09 server1 opensips: Jan 13 06:12:09 [17458] 
DBG:tm:retransmission_handler: retransmission_handler : request 
resending (t=0x7f8518e24718, INVITE si ... )
Jan 13 06:12:10 server1 opensips: Jan 13 06:12:10 [17467] 
DBG:tm:retransmission_handler: retransmission_handler : request 
resending (t=0x7f8518e24718, INVITE si ... )
Jan 13 06:12:12 server1 opensips: Jan 13 06:12:12 [17458] 
DBG:tm:retransmission_handler: retransmission_handler : request 
resending (t=0x7f8518e24718, INVITE si ... )


and finally, on timeout, it is canceled (toward UAS) and 408 reply 
generated:


Jan 13 06:12:14 server1 opensips: Jan 13 06:12:14 [17457] 
DBG:tm:final_response_handler: Cancel sent out, sending 408 (0x7f8518e24718)
Jan 13 06:12:14 server1 opensips: Jan 13 06:12:14 [17457] 
DBG:tm:t_should_relay_response: T_code=100, new_code=408
Jan 13 06:12:14 server1 opensips: Jan 13 06:12:14 [17457] 
DBG:tm:t_pick_branch: picked branch 0, code 408 (prio=800)
Jan 13 06:12:14 server1 opensips: Jan 13 06:12:14 [17457] 
DBG:tm:relay_reply: sent buf=0x7f851f8e9518: SIP/2.0 4..., 
shmem=0x7f8518e6f618: SIP/2.0 4


I see even a ACC log is created:

Jan 13 06:12:14 server1 opensips: ACC: call missed: 
timestamp=1452665534;method=INVITE;from_tag=z9hG4bK17850112;to_tag=;call_id=369319113465@10.121.239.28;code=408;reason=Request 
Timeout



So, everything looks good here.

Regards,

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

On 13.01.2016 08:19, Nabeel wrote:

Hi Bogdan,

I changed log_stderror=yes and log_facility=LOG_DAEMON. Now I see some 
more in the log.  Do you see anything obviously wrong?


http://pastebin.com/MzJW1P1S

On 12 January 2016 at 09:10, Bogdan-Andrei Iancu > wrote:


Hi Nabeel,

Be sure you are looking into the right log file - maybe the debug
level is redirected by your syslog to another log file... Debug
level 4 is the most verbose one in opensips.

Regards,

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

On 08.01.2016 21:04, Nabeel wrote:


Hi Bogdan,

I have the following near the top of my config file:

## Global Parameters #

debug=4
log_stderror=no
log_facility=LOG_LOCAL1

The log I posted earlier is from opensips running with these
configurations.

On 8 Jan 2016 3:49 pm, "Bogdan-Andrei Iancu" > wrote:

Hi Nabeel,

have you tried running opensips is debug mode (level 4) to
see what it is doing with the request ?

Regards,

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

On 07.01.2016 11:37, Nabeel wrote:


Hi Bogdan,

I used the tshark command as explained here on page 14:

http://opensips.org/pub/events/2015-05-12_OpenSIPS-Summit_Amsterdam/Lorenzo_

Mangani-OpenSIPS

_

Summit2015-SIPCapture.pdf



tshark -o "ssl.desegment_ssl_records: TRUE"  -o
"ssl.desegment_ssl_application_data: TRUE"  -o
"ssl.keys_list:
162.249.6.110,5061,sip,/install/tls/domain.com-key.pem" -i
eth0  -f "tcp port 5061"

I'm using a command line version of Linux without a graphic
UI, so I could not "configure Wireshark to decide TLS" as
mentioned in that document, however I did pass the private
key in the command as shown above.

Does tshark require configuring to decode TLS, other than
passing the private key in the command?

Hi Nabeel,

Indeed, the 408 seems generated by OpenSIPS (after 5
seconds). Such reply is 

Re: [OpenSIPS-Users] registrar contact delete

[Bogdan-Andrei Iancu]

There is no 1 to 1 relation between calls/registration (or generic SIP 
elements) and the TCP connections - a TCP conn may be used by multiple 
such SIP elements.


A TCP conn is closed by opensips if there is no traffic on it for 
certain amount of time.


Do you use the "tcp_persistent_flag" ?
http://www.opensips.org/html/docs/modules/1.11.x/registrar.html#id293556

Regards,

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

On 19.01.2016 13:02, Tito Cumpen wrote:


Bogdan,

Yea will the server close tcp connection to the client if the record 
is replace as I mentioned ?


Thanks,
Tito

On Jan 19, 2016 5:24 AM, "Bogdan-Andrei Iancu" > wrote:


Hi Tito,

What do you understand by socket ?? The TCP connection ? or ?

Best regards,

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

On 15.01.2016 22:56, Tito Cumpen wrote:

Group,

I am currently using a max contact param within the registrar
module. I am using   if (!save("location","f")) to force
the new registration the case that the max has been reached.It
appears that the oldest registration is deleted. Will the
respective socket to the deleted contact be closed after the
contact has been deleted ? Or will it linger until the previous
registration expiry time concludes?


Thanks,
Tito


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Re: [OpenSIPS-Users] B2B BYEs

[Søren Andersen]

Hi Bogdan-Andrei,

Hmm.. Actually it look likes "route_se" is missing in the LEGS:: after a 
restart?

# Before restart #
dlg:: 227897 param=774.0 state=5 last_invite_cseq=29551 last_method=4 
last_reply_code=200 db_flag=2
 callid:: B2B.348.227897
 from::  uri=sip:xxx...@xx.xx.xx.36;user=phone 
tag=e33f629b154d16f5f57a07bf5d17124e-7644
 to::  uri=sip:xxx...@xx.xx.xx.120 tag=2075842316
 cseq::  caller=29551 callee=1
 route_set::  
caller=
 contact::  caller=sip:xx.xx.xx.117:5060 
callee=sip:xx04@10.101.16.179:5060;transport=udp
 send_sock:: xx.xx.xx.117
 LEGS::
  leg:: 0 
tag=2075842316 cseq=29551 contact=sip:xx04@10.101.16.179:5060;transport=udp 
route_se=

# After restart #
dlg:: 227897 param=774.0 state=5 last_invite_cseq=29551 last_method=0 db_flag=0
 callid:: B2B.348.227897
 from::  uri=sip:x...@xx.xx.xx.36;user=phone 
tag=e33f629b154d16f5f57a07bf5d17124e-7644
 to::  uri=sip:x...@xx.xx.xxx.120 tag=2075842316
 cseq::  caller=29551 callee=1
 route_set::  
caller=
 contact::  caller=sip:xx.xx.xx.117:5060 
callee=sip:xx04@10.101.16.179:5060;transport=udp
 send_sock:: xx.xx.xx.117
 LEGS::
  leg:: 0 
tag=2075842316 cseq=29551 contact=sip:xx04@10.101.16.179:5060;transport=udp


/Søren
Fra: Bogdan-Andrei Iancu [mailto:bog...@opensips.org]
Sendt: 19. januar 2016 11:13
Til: Søren Andersen ; users@lists.opensips.org
Emne: Re: SV: SV: SV: [OpenSIPS-Users] B2B BYEs

Hi Søren,

It seems that the callee route set is stored in db (see the b2b_entities table, 
the route1 field 
="")

Can you run the b2be_list MI command before and after the restart, to see if 
indeed, the callee route set is not properly restored ?

Regards,


Bogdan-Andrei Iancu

OpenSIPS Founder and Developer

http://www.opensips-solutions.com
On 15.01.2016 12:21, Søren Andersen wrote:
Yup.

SELECT * FROM b2b.b2b_logic;
# id, si_key, scenario, sstate, next_sstate, sparam0, sparam1, sparam2, 
sparam3, sparam4, sdp, lifetime, e1_type, e1_sid, e1_from, e1_to, e1_key, 
e2_type, e2_sid, e2_from, e2_to, e2_key, e3_type, e3_sid, e3_from, e3_to, e3_key
'226', '360.0', 'StofaB2B', '1', '1', 
'sip:...@xx.xx.xx.120', '', '', '', '', 
'', '1452896043', '0', 'server1', 
'sip:...@oxd3-sxw.xx.txxx.xxx;user=phone',
 'sip:x...@xx.xx.xx.117:5060', 
'B2B.39.160', '1', 'client1', 
'sip:x...@oxx3-sx.xx.xxx.xxx;user=phone',
 'sip:...@xx.xx.xx.120', 
'B2B.448.7695796', NULL, NULL, NULL, NULL, NULL

SELECT * FROM b2b.b2b_entities;
# id, type, state, ruri, from_uri, to_uri, from_dname, to_dname, tag0, tag1, 
callid, cseq0, cseq1, contact0, contact1, route0, route1, sockinfo_srv, param, 
lm, lrc, lic, leg_cseq, leg_route, leg_tag, leg_contact, leg_sockinfo
'316', '0', '5', 
'sip:...@xx.xx.xx.117:5060;user=phone',
 
'sip:x...@osx3-sxw.xx..xxx;user=phone',
 
'sip:X04@10.250.224.22;user=phone',
 '', '', 'mvgh6xvkp7z', 'B2B.39.160', 
'1q2rqu6-3232128...@osdx-ssw.xx.xxx.xx',
 '32736', '1', 
'sip:x...@xx.xx.xx.36:5060;transport=udp',
 'sip:XX.XX.XX.117:5060', '', '', 
'udp:XX.XX.XX.117:5060', '360.0', '0', '200', '32736', NULL, NULL, NULL, NULL, 
NULL
'317', '1', '5', '', 
'sip:x...@osdx-ssw.xx..xx;user=phone',
 'sip:x...@xx.xx.xx.120', '', '', 
'a9ff31f61f2601c916f6044c2e20e16a-3c47', '2761431845', 'B2B.448.7695796', 
'32737', '1', 'sip:XX.XXX.XXX.117:5060', 
'sip:XX04@10.101.16.179:5060;transport=udp',
 
'',
 '', 'udp:XX.XX.XX.117:5060', 

Re: [OpenSIPS-Users] why tls not enable

[Bogdan-Andrei Iancu]

Hi ,

if you do "netstat -lnp | grep opensips" do you see the other 2 UDP 
listeners ?


Regards,

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

On 13.01.2016 03:30, chiu ching cheng wrote:

Hello :Bordon

Now "opensipsctl start " is ok , and

opensips -f /usr/etc/opensips/opensips.cfg

Listening on
 udp: 125.227.130.1 [125.227.130.1]:5060
 udp: 125.227.130.1 [125.227.130.1]:3478
 tls: 125.227.130.1 [125.227.130.1]:5061
Aliases:
 *: 125.227.130.1:*

But :~# netstat -tuln | grep 5061
tcp0  0 0.0.0.0:25061  0.0.0.0:* 
  LISTEN


tls service not start

On Tue, Jan 12, 2016 at 4:45 PM, Bogdan-Andrei Iancu 
> wrote:


Hi,

Try to manually start "opensips -f /path/to/opensips.cfg"

Regards,

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

On 12.01.2016 08:56, chiu ching cheng wrote:

Hello Bogdan:

I am start up opensips with "opensipsctl start " , and no "-f"
parameter . In my impression , it would inspect opensips.cfg
configuration for opensipsctl script .

On Mon, Jan 11, 2016 at 6:06 PM, Bogdan-Andrei Iancu
> wrote:

Hi,

I do not contest the correctness of your cfg, but I'm simply
asking if you are 100% sure that your opensips is using the
correct opensips.cfg file ( be sure by explicitly pointing
the file via "-f" startup option).

Regards,

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

On 11.01.2016 03:46, chiu ching cheng wrote:


openssips.cfg

listen=tls:X.X.X.X:5061   # CUSTOMIZE ME
rev_dns=no

advertised_address="X.X.X.X"
alias=X.X.X.X


loadmodule "proto_udp.so"


loadmodule "proto_tls.so"
modparam("proto_tls","verify_cert", "0")
modparam("proto_tls","require_cert", "0")
modparam("proto_tls","tls_method", "TLSv1")
modparam("proto_tls", "ciphers_list", "NULL")
modparam("proto_tls","certificate",
"/usr/local/etc/opensips/tls/user/user-cert.pem")
modparam("proto_tls","private_key",
"/usr/local/etc/opensips/tls/user/user-privkey.pem")
modparam("proto_tls","ca_list",
"/usr/local/etc/opensips/tls/user/user-calist.pem")


On Fri, Jan 8, 2016 at 11:39 PM, Bogdan-Andrei Iancu
> wrote:

Hi,

Are you sure your opensips is using the proper config
file (where the TLS interface is defined) ?

Regards,

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

On 08.01.2016 12:06, chiu ching cheng wrote:

Dear Bordan :

netstat -lnp | grep opensips
udp00 127.0.0.1:5060
  0.0.0.0:* 1365/opensips

you can find opensips is run , but just enable udp ,
not tls

On Wed, Jan 6, 2016 at 5:24 PM, Bogdan-Andrei Iancu
> wrote:

Try

"netstat -lnp | grep opensips"

Regards,

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

On 06.01.2016 04:26, chiu ching cheng wrote:

Dear Bogdan:

thanks your reply , but as following , the sip udp
is enable , opensips is enable ,and log no error .
but tls not enable

I ever install another opensips server which is
behind the firewall and tls work fine , but this
server is one interface public ip , one interface
private , I just config one public ip . I don't
know if this caused failure ( opensips at lease
need two ip ?) .

thanks !

root@125-227-130-1:~# netstat -tuln | grep 5061
tcp  0  0 0.0.0.0:25061 
0.0.0.0:* LISTEN
root@125-227-130-1:~# netstat -tuln | grep 5060
tcp  0  0 0.0.0.0:25060 
0.0.0.0:* LISTEN
udp  0  0 127.0.0.1:5060
  0.0.0.0:*
root@125-227-130-1:~# ps auxw | grep opensips
root  1466  0.0  0.0  70052  4248 ?  S10:12
0:00 /usr/local/sbin/opensips -P /var/run/opensips.pid
root  1467  0.0  0.0  

Re: [OpenSIPS-Users] tomcat external app listening to OpenSIPS events

[Bogdan-Andrei Iancu]

Hi Julian,

I see your problem and I agree with you - the path should be be missing 
in the HTTP URL. I will open a bug report and have it fixed asap.


Thanks and regards,

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

On 14.01.2016 00:28, Julian Kay wrote:

HI,

I really appreciate you taking the time!!

yes OpenSIPS is connecting to Tomcat, the problem I see (I think) is when the 
event is being raised OpenSIPS is NOT sending the complete URI. Tomcat server 
returns an error of 404.
the Tomcat logs seem to indicate  OpenSIPS is only be sending 192.168.3.132:8080 
-> when it should be sending 192.168.3.132:8080/ccurbiz/xmlrpc

Thanks!!
Juls

-Original Message-
From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu
Sent: Wednesday, January 13, 2016 5:44 AM
To: OpenSIPS users mailling list; juli...@vazycomm.com
Subject: Re: [OpenSIPS-Users] tomcat external app listening to OpenSIPS events

Hi Julian,

So, you say OpenSIPS is actually connecting via HTTP to tomcat in order to 
deliver the event, right ?

Regards,

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

On 13.01.2016 03:02, Julian Kay wrote:

Hi;

Thanks for the help!

the event subscription seems to be successful as  show from the output
of a test xmlrpc test client
Event:: E_SIP_MESSAGE id=11Subscriber::
socket=xmlrpc:192.168.3.132:8080:ccradius/xmlrpc/OpenSIPSInterface.msg
tests expire=never

the parameters I'm using for the for subscribe_event:
subscribe_event("E_SIP_MESSAGE","xmlrpc:192.168.3.132:8080:ccradius/xm
lrpc/OpenSIPSInterface.msgtests");

for testing purposes I raise the event with ->
raise_event("E_SIP_MESSAGE")

and it seems OpenSIPs attempts to deliver the event because this is
what I see in the Tomcat logs:  192.168.3.167 - -
[06/Jan/2016:23:21:19 -0500] "POST /RPC2 HTTP/1.1" 404 959

I'm able to successfully call the Java class from a test xmlrpc test client.

I would really appreciate ideas how I can resolve this as this is critical for 
me. I'm even willing to consider some other methodology as long as I can 
interface with Tomcat.

Thanks for your input!!
Juls

-Original Message-
From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei
Iancu
Sent: Tuesday, January 12, 2016 4:14 AM
To: OpenSIPS users mailling list; juli...@vazycomm.com
Subject: Re: [OpenSIPS-Users] tomcat external app listening to
OpenSIPS events

Hi Julian,

First check if your event subscription was successful (and still valid).
Use the "subscriber_list" MI command :
http://www.opensips.org/Documentation/Interface-CoreMI-1-11#toc18

After that, when events happens, check at network level if there is any attempt 
from opensips side to deliver the event via XMLRPC to the indicated URL.

Regards,

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

On 12.01.2016 05:15, Julian Kay wrote:

Hi;

Has anyone successfully interfaced OpenSIPS event_xmlrpc with a Tombat xmlrpc 
server servlet? If you yes can you share any tips, because I've been trying it 
for a while without any success.

Thanks for any help!!

Juls

-Original Message-
From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Julian Kay
Sent: Thursday, January 07, 2016 9:33 AM
To: 'OpenSIPS users mailling list'
Subject: Re: [OpenSIPS-Users] external app listening to OpenSIPS/SIP
events

Thanks for your help!!

yes the event is reaching the xmlrpc server I'm including 2 entries in the 
Tomcat log file:

call using xmlrpc-test-tool:192.168.3.167 - - [06/Jan/2016:16:26:11 -0500] "POST 
/ccurbiz/xmlrpc/RPC2 HTTP/1.0" 200 156

call using event_xmlrpc:192.168.3.167 - - [06/Jan/2016:23:21:19 -0500] "POST 
/RPC2 HTTP/1.1" 404 959

(192.168.3.167 is the OpenSIPs server)

I've tried a few variations for the host name parameter including:
subscribe_event("E_SIP_MESSAGE","xmlrpc:192.168.3.132:8080:ccurbiz.xm
l
rpc.OpenSIPSInterface.msgtests");

subscribe_event("E_SIP_MESSAGE","xmlrpc:192.168.3.132:8080/ccurbiz/xmlrpc:OpenSIPSInterface.msgtests");
 -> this is the most similar to other xmlrpc clients I've tested.

Best regards!
Juls


-Original Message-
From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Razvan Crainea
Sent: Thursday, January 07, 2016 4:34 AM
To: users@lists.opensips.org
Subject: Re: [OpenSIPS-Users] external app listening to OpenSIPS/SIP
events

Hi, Julian!

First of all, is the event reaching the xmlrpc server? If not sure, try to make 
a tcpdump to capture the communication.
If it does reach the server, what are the errors you are seeing in the Tomcat 
server?

Best regards,

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

On 01/07/2016 12:32 AM, Julian Kay wrote:

THANKS!!

_raising the event in the script got rid of the 

Re: [OpenSIPS-Users] X-Auth-IP How?

[Jason Bedward]

I see your point, in that case I'm not sure. Head over to the freeswitch
irc they should be able to clear that up. I would say its probably
technically possible but not sure why you would want that anyway?

On Tue, Jan 19, 2016 at 9:18 PM, Tim King  wrote:

> It is working now and I am doing things the same way you described. The
> part I was expecting that I don't see was that I was going to say hey
> freeswitch for this proxy check the stuff received in X-AUTH-IP against
> this specific ACL. It appears that it just checks all of the ACL's
>
>
> On 1/19/2016 4:13 PM, Jason Bedward wrote:
>
> Sorry I'm confused is it working now?
>
> The way I have things is opensips ip is the proxy. When freeswitch see it
> its like hey, I know you, I need to look out for the other ip you have
> X-Auth and process that. Freeswitch then looks up the X-Auth ip as it that
> was the IP that initiated the session
>
> On Tue, Jan 19, 2016 at 9:10 PM, Tim King  wrote:
>
>>
>> This is the intended behavior I was requesting. The purpose of the ACL
>> setup is to only allow calls through that are in the list of IP's set to
>> allow. The same way you would with Freeswitch without a proxy. The
>> relationship I was looking for that apparently does not exist is where you
>> tell freeswitch which list to use for the proxy-acl. Apparently there is no
>> relationship between them it just searches all the lists to see if ti finds
>> a match? I'm not really sure all I know is at this point if I put the IP's
>> of the addresses I wish to allow in the last list in my acl.conf which
>> happens to be names opensips the calls are allowed. If I take the IP out
>> they are rejected... Thank you very much for your help.
>>
>>
>>
>> ___
>> Users mailing list
>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>
>
> ___
> Users mailing 
> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] YUM repository downtime

[Chris Stone]

Now seeing connections refused:

  

[root@gliese ~]# yum update  
Loaded plugins: fastestmirror  
http://yum.opensips.org/2.1/releases/el/7/x86_64/repodata/repomd.xml: [Errno
14] curl#7 - "Failed connect to yum.opensips.org:80; Connection refused"  
Trying other mirror.  
http://yum.opensips.org/2.1/releases/el/7/x86_64/repodata/repomd.xml: [Errno
14] curl#7 - "Failed connect to yum.opensips.org:80; Connection refused"  
Trying other mirror.  
http://yum.opensips.org/2.1/releases/el/7/x86_64/repodata/repomd.xml: [Errno
14] curl#7 - "Failed connect to yum.opensips.org:80; Connection refused"  
Trying other mirror.  
http://yum.opensips.org/2.1/releases/el/7/x86_64/repodata/repomd.xml: [Errno
14] curl#7 - "Failed connect to yum.opensips.org:80; Connection refused"  
Trying other mirror.  
http://yum.opensips.org/2.1/releases/el/7/x86_64/repodata/repomd.xml: [Errno
14] curl#7 - "Failed connect to yum.opensips.org:80; Connection refused"  
Trying other mirror.  
http://yum.opensips.org/2.1/releases/el/7/x86_64/repodata/repomd.xml: [Errno
14] curl#7 - "Failed connect to yum.opensips.org:80; Connection refused"  
Trying other mirror.  
http://yum.opensips.org/2.1/releases/el/7/x86_64/repodata/repomd.xml: [Errno
14] curl#7 - "Failed connect to yum.opensips.org:80; Connection refused"  
Trying other mirror.  
http://yum.opensips.org/2.1/releases/el/7/x86_64/repodata/repomd.xml: [Errno
14] curl#7 - "Failed connect to yum.opensips.org:80; Connection refused"  
Trying other mirror.  
http://yum.opensips.org/2.1/releases/el/7/x86_64/repodata/repomd.xml: [Errno
14] curl#7 - "Failed connect to yum.opensips.org:80; Connection refused"  
Trying other mirror.  
http://yum.opensips.org/2.1/releases/el/7/x86_64/repodata/repomd.xml: [Errno
14] curl#7 - "Failed connect to yum.opensips.org:80; Connection refused"  
Trying other mirror.  

  

  

Chris

> On Jan 19 2016, at 11:05 am, Chris Stone axi...@gmail.com wrote:  
I'll give it another trythanks

>

>  

>

> Chris

>

>> On Jan 19 2016, at 9:59 am, Nick Altmann nick.altm...@gmail.com
wrote:  

>>

>> The repo is in testing mode.  

>>

>> Have you tried today? Try again, all rpms are signed.

>>

>>  

>>

>> \--  
Nick

>>

>>  

>>

>> 2016-01-19 19:43 GMT+03:00 Chris Stone
[axi...@gmail.com](mailto:axi...@gmail.com):  

>>

>>> It is - but the packages (rpms) are all unsigned

>>>

>>>  

>>>

>>>  

>>>

>>> Chris

>>>

 On Jan 19 2016, at 3:15 am, Bogdan-Andrei Iancu
[bog...@opensips.org](mailto:bog...@opensips.org) wrote:  
Hey, it seems to be up again !!! :))  
  
Regards,  

  
  
 Bogdan-Andrei Iancu

 OpenSIPS Founder and Developer

 



 On 18.01.2016 10:24, Nick Altmann wrote:  



> Thank you. Yes, you’re right. It’s still down. New repository will be up
a bit later today.

>

>  

>

> \--  
Nick

>

>  

>

> 2016-01-18 11:03 GMT+03:00 Markus Rydh
[markus.r...@gmail.com](mailto:markus.r...@gmail.com):  

>

>> Nick, the yum repo seems to be down for the last few days. Any updates
on  
when it might be available? Appreciate your work on hosting the repo!  
  
// Markus  
  
  
  
\--  
View this message in context:   
Sent from the OpenSIPS - Users mailing list archive at Nabble.com.  
  
___  
Users mailing list  
[Users@lists.opensips.org](mailto:Users@lists.opensips.org)  
  

>

>  

>

>  
  

>  
>  
> ___

> Users mailing list

> [Users@lists.opensips.org](mailto:Users@lists.opensips.org)

> 

>  


  

>>>

>>>  
___  
Users mailing list  
[Users@lists.opensips.org](mailto:Users@lists.opensips.org)  
  
  

>>

>>  

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

Re: [OpenSIPS-Users] X-Auth-IP How?

[Tim King]

I have a name in there do I need to change that to 20.172.0.46/32 ? If 
yes do I need my list to be named for that IP? Sorry for all the 
questions, I just have not found a working documented example and it 
seems like it is very close now.










On 1/19/2016 3:42 PM, Jason Bedward wrote:

Also edit the "apply-proxy-acl" IP you are missing the /32

On Wed, Jan 13, 2016 at 6:14 PM, Tim King > wrote:


Thank You. do you have an example as to how to have the Freeswitch
ACL work with he via headers you discussed? I have followed many
guides for setting up OpenSIPs as a load balancer for Freeswitch,
but none of them addressed making the Freeswitch ACL functional.


On 1/13/2016 12:49 PM, Stefano Pisani wrote:

to be precise:

append_hf("X-Auth-IP: $i\r\n");

according to documentation

Il 13/01/2016 18.43, Tim King ha scritto:

I have read countless articles now talking about using x-auth-ip
as a method for using OpenSIPs as a load balancer serving to a
cluster of Freeswtich servers and having a method to maintain
the original IP address.

Direct from the Freeswitch wiki it states:

apply-proxy-acl

Use the IP specified in X-AUTH-IP header sent from proxy for
apply-inbound-acl Note: You'll need to configure your proxy to
add this header


However I am not able to find any example of how to make
OpenSIPs add this header. Can anyone provide me some guidance as
to how to accomplish this?


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Re: [OpenSIPS-Users] X-Auth-IP How?

[Jason Bedward]

Also edit the "apply-proxy-acl" IP you are missing the /32

On Wed, Jan 13, 2016 at 6:14 PM, Tim King  wrote:

> Thank You. do you have an example as to how to have the Freeswitch ACL
> work with he via headers you discussed? I have followed many guides for
> setting up OpenSIPs as a load balancer for Freeswitch, but none of them
> addressed making the Freeswitch ACL functional.
>
>
> On 1/13/2016 12:49 PM, Stefano Pisani wrote:
>
> to be precise:
>
> append_hf("X-Auth-IP: $i\r\n");
>
> according to documentation
>
> Il 13/01/2016 18.43, Tim King ha scritto:
>
> I have read countless articles now talking about using x-auth-ip as a
> method for using OpenSIPs as a load balancer serving to a cluster of
> Freeswtich servers and having a method to maintain the original IP address.
>
> Direct from the Freeswitch wiki it states:
>
> apply-proxy-acl
>
> Use the IP specified in X-AUTH-IP header sent from proxy for
> apply-inbound-acl Note: You'll need to configure your proxy to add this
> header
>
>
> However I am not able to find any example of how to make OpenSIPs add this
> header. Can anyone provide me some guidance as to how to accomplish this?
>
>
> ___
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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] X-Auth-IP How?

[Tim King]

It is working now and I am doing things the same way you described. The 
part I was expecting that I don't see was that I was going to say hey 
freeswitch for this proxy check the stuff received in X-AUTH-IP against 
this specific ACL. It appears that it just checks all of the ACL's


On 1/19/2016 4:13 PM, Jason Bedward wrote:

Sorry I'm confused is it working now?

The way I have things is opensips ip is the proxy. When freeswitch see 
it its like hey, I know you, I need to look out for the other ip you 
have X-Auth and process that. Freeswitch then looks up the X-Auth ip 
as it that was the IP that initiated the session


On Tue, Jan 19, 2016 at 9:10 PM, Tim King > wrote:



This is the intended behavior I was requesting. The purpose of the
ACL setup is to only allow calls through that are in the list of
IP's set to allow. The same way you would with Freeswitch without
a proxy. The relationship I was looking for that apparently does
not exist is where you tell freeswitch which list to use for the
proxy-acl. Apparently there is no relationship between them it
just searches all the lists to see if ti finds a match? I'm not
really sure all I know is at this point if I put the IP's of the
addresses I wish to allow in the last list in my acl.conf which
happens to be names opensips the calls are allowed. If I take the
IP out they are rejected... Thank you very much for your help.



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Re: [OpenSIPS-Users] X-Auth-IP How?

[Jason Bedward]

Sorry I'm confused is it working now?

The way I have things is opensips ip is the proxy. When freeswitch see it
its like hey, I know you, I need to look out for the other ip you have
X-Auth and process that. Freeswitch then looks up the X-Auth ip as it that
was the IP that initiated the session

On Tue, Jan 19, 2016 at 9:10 PM, Tim King  wrote:

>
> This is the intended behavior I was requesting. The purpose of the ACL
> setup is to only allow calls through that are in the list of IP's set to
> allow. The same way you would with Freeswitch without a proxy. The
> relationship I was looking for that apparently does not exist is where you
> tell freeswitch which list to use for the proxy-acl. Apparently there is no
> relationship between them it just searches all the lists to see if ti finds
> a match? I'm not really sure all I know is at this point if I put the IP's
> of the addresses I wish to allow in the last list in my acl.conf which
> happens to be names opensips the calls are allowed. If I take the IP out
> they are rejected... Thank you very much for your help.
>
>
>
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Re: [OpenSIPS-Users] X-Auth-IP How?

[Jason Bedward]

Look and think, you are saying acl = test. But in test you deny everything
aside from you sip phone!...lol

On Tue, Jan 19, 2016 at 8:47 PM, Tim King  wrote:

> I have a name in there do I need to change that to 20.172.0.46/32 ? If
> yes do I need my list to be named for that IP? Sorry for all the questions,
> I just have not found a working documented example and it seems like it is
> very close now.
>
> 
> 
> 
> 
> 
>
>
>
>
> On 1/19/2016 3:42 PM, Jason Bedward wrote:
>
> Also edit the "apply-proxy-acl" IP you are missing the /32
>
> On Wed, Jan 13, 2016 at 6:14 PM, Tim King  wrote:
>
>> Thank You. do you have an example as to how to have the Freeswitch ACL
>> work with he via headers you discussed? I have followed many guides for
>> setting up OpenSIPs as a load balancer for Freeswitch, but none of them
>> addressed making the Freeswitch ACL functional.
>>
>>
>> On 1/13/2016 12:49 PM, Stefano Pisani wrote:
>>
>> to be precise:
>>
>> append_hf("X-Auth-IP: $i\r\n");
>>
>> according to documentation
>>
>> Il 13/01/2016 18.43, Tim King ha scritto:
>>
>> I have read countless articles now talking about using x-auth-ip as a
>> method for using OpenSIPs as a load balancer serving to a cluster of
>> Freeswtich servers and having a method to maintain the original IP address.
>>
>> Direct from the Freeswitch wiki it states:
>>
>> apply-proxy-acl
>>
>> Use the IP specified in X-AUTH-IP header sent from proxy for
>> apply-inbound-acl Note: You'll need to configure your proxy to add this
>> header
>>
>>
>> However I am not able to find any example of how to make OpenSIPs add
>> this header. Can anyone provide me some guidance as to how to accomplish
>> this?
>>
>>
>> ___
>> Users mailing 
>> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>>
>>
>> ___
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>> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>>
>> ___
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>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>
>
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>
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Re: [OpenSIPS-Users] X-Auth-IP How?

[Tim King]

This is the intended behavior I was requesting. The purpose of the ACL 
setup is to only allow calls through that are in the list of IP's set to 
allow. The same way you would with Freeswitch without a proxy. The 
relationship I was looking for that apparently does not exist is where 
you tell freeswitch which list to use for the proxy-acl. Apparently 
there is no relationship between them it just searches all the lists to 
see if ti finds a match? I'm not really sure all I know is at this point 
if I put the IP's of the addresses I wish to allow in the last list in 
my acl.conf which happens to be names opensips the calls are allowed. If 
I take the IP out they are rejected... Thank you very much for your help.



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Re: [OpenSIPS-Users] X-Auth-IP How?

[Jason Bedward]

change to IP as I originally said about 2 hours ago :)

On Tue, Jan 19, 2016 at 8:47 PM, Tim King  wrote:

> I have a name in there do I need to change that to 20.172.0.46/32 ? If
> yes do I need my list to be named for that IP? Sorry for all the questions,
> I just have not found a working documented example and it seems like it is
> very close now.
>
> 
> 
> 
> 
> 
>
>
>
>
> On 1/19/2016 3:42 PM, Jason Bedward wrote:
>
> Also edit the "apply-proxy-acl" IP you are missing the /32
>
> On Wed, Jan 13, 2016 at 6:14 PM, Tim King  wrote:
>
>> Thank You. do you have an example as to how to have the Freeswitch ACL
>> work with he via headers you discussed? I have followed many guides for
>> setting up OpenSIPs as a load balancer for Freeswitch, but none of them
>> addressed making the Freeswitch ACL functional.
>>
>>
>> On 1/13/2016 12:49 PM, Stefano Pisani wrote:
>>
>> to be precise:
>>
>> append_hf("X-Auth-IP: $i\r\n");
>>
>> according to documentation
>>
>> Il 13/01/2016 18.43, Tim King ha scritto:
>>
>> I have read countless articles now talking about using x-auth-ip as a
>> method for using OpenSIPs as a load balancer serving to a cluster of
>> Freeswtich servers and having a method to maintain the original IP address.
>>
>> Direct from the Freeswitch wiki it states:
>>
>> apply-proxy-acl
>>
>> Use the IP specified in X-AUTH-IP header sent from proxy for
>> apply-inbound-acl Note: You'll need to configure your proxy to add this
>> header
>>
>>
>> However I am not able to find any example of how to make OpenSIPs add
>> this header. Can anyone provide me some guidance as to how to accomplish
>> this?
>>
>>
>> ___
>> Users mailing 
>> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>>
>>
>> ___
>> Users mailing 
>> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>>
>> ___
>> Users mailing list
>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>
>
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[OpenSIPS-Users] Is there some way to set Alias at execution time (dinamically)?

[Rodrigo Pimenta Carvalho]

Dear OpenSIPS-users,


I have used the global parameter Alias successful. The same for 
advertised_address.


Now, I have to configure these values at execution time. Such values will be 
determined just after starting my system. So, I need a way to configure these 
values in a dynamic way.


Then, I will use the core function set_advertised_address.

But, I did not find a core function to set the Alias at execution time?


Is there a way to set Alias at execution time? Is there a function for that or 
can I set Alias with a database query. Ex: Alias = select from


Any hint will be very helpful!


Best regards.


RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979
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Re: [OpenSIPS-Users] YUM repository downtime

[Chris Stone]

I'll give it another trythanks

  

Chris

> On Jan 19 2016, at 9:59 am, Nick Altmann nick.altm...@gmail.com
wrote:  

>

> The repo is in testing mode.  

>

> Have you tried today? Try again, all rpms are signed.

>

>  

>

> \--  
Nick

>

>  

>

> 2016-01-19 19:43 GMT+03:00 Chris Stone
[axi...@gmail.com](mailto:axi...@gmail.com):  

>

>> It is - but the packages (rpms) are all unsigned

>>

>>  

>>

>>  

>>

>> Chris

>>

>>> On Jan 19 2016, at 3:15 am, Bogdan-Andrei Iancu
[bog...@opensips.org](mailto:bog...@opensips.org) wrote:  
Hey, it seems to be up again !!! :))  
  
Regards,  

>>>  
>>>  
>>> Bogdan-Andrei Iancu

>>> OpenSIPS Founder and Developer

>>> 

>>>

>>> On 18.01.2016 10:24, Nick Altmann wrote:  

>>>

 Thank you. Yes, you’re right. It’s still down. New repository will be up
a bit later today.



  



 \--  
Nick



  



 2016-01-18 11:03 GMT+03:00 Markus Rydh
[markus.r...@gmail.com](mailto:markus.r...@gmail.com):  



> Nick, the yum repo seems to be down for the last few days. Any updates
on  
when it might be available? Appreciate your work on hosting the repo!  
  
// Markus  
  
  
  
\--  
View this message in context:   
Sent from the OpenSIPS - Users mailing list archive at Nabble.com.  
  
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 [Users@lists.opensips.org](mailto:Users@lists.opensips.org)

 

  
>>>

>>>  

>>

>>  
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Re: [OpenSIPS-Users] YUM repository downtime

[Nick Altmann]

No repository – no problem with signing. ;-)
Fixed now.

--
Nick

2016-01-20 0:11 GMT+03:00 Chris Stone :

> Now seeing connections refused:
>
> [root@gliese ~]# yum update
> Loaded plugins: fastestmirror
> http://yum.opensips.org/2.1/releases/el/7/x86_64/repodata/repomd.xml:
> [Errno 14] curl#7 - "Failed connect to yum.opensips.org:80; Connection
> refused"
> Trying other mirror.
> http://yum.opensips.org/2.1/releases/el/7/x86_64/repodata/repomd.xml:
> [Errno 14] curl#7 - "Failed connect to yum.opensips.org:80; Connection
> refused"
> Trying other mirror.
> http://yum.opensips.org/2.1/releases/el/7/x86_64/repodata/repomd.xml:
> [Errno 14] curl#7 - "Failed connect to yum.opensips.org:80; Connection
> refused"
> Trying other mirror.
> http://yum.opensips.org/2.1/releases/el/7/x86_64/repodata/repomd.xml:
> [Errno 14] curl#7 - "Failed connect to yum.opensips.org:80; Connection
> refused"
> Trying other mirror.
> http://yum.opensips.org/2.1/releases/el/7/x86_64/repodata/repomd.xml:
> [Errno 14] curl#7 - "Failed connect to yum.opensips.org:80; Connection
> refused"
> Trying other mirror.
> http://yum.opensips.org/2.1/releases/el/7/x86_64/repodata/repomd.xml:
> [Errno 14] curl#7 - "Failed connect to yum.opensips.org:80; Connection
> refused"
> Trying other mirror.
> http://yum.opensips.org/2.1/releases/el/7/x86_64/repodata/repomd.xml:
> [Errno 14] curl#7 - "Failed connect to yum.opensips.org:80; Connection
> refused"
> Trying other mirror.
> http://yum.opensips.org/2.1/releases/el/7/x86_64/repodata/repomd.xml:
> [Errno 14] curl#7 - "Failed connect to yum.opensips.org:80; Connection
> refused"
> Trying other mirror.
> http://yum.opensips.org/2.1/releases/el/7/x86_64/repodata/repomd.xml:
> [Errno 14] curl#7 - "Failed connect to yum.opensips.org:80; Connection
> refused"
> Trying other mirror.
> http://yum.opensips.org/2.1/releases/el/7/x86_64/repodata/repomd.xml:
> [Errno 14] curl#7 - "Failed connect to yum.opensips.org:80; Connection
> refused"
> Trying other mirror.
>
>
> Chris
>
>> On Jan 19 2016, at 11:05 am, Chris Stone  wrote:
>> I'll give it another trythanks
>>
>> Chris
>>
>> On Jan 19 2016, at 9:59 am, Nick Altmann  wrote:
>> The repo is in testing mode.
>> Have you tried today? Try again, all rpms are signed.
>>
>> --
>> Nick
>>
>> 2016-01-19 19:43 GMT+03:00 Chris Stone :
>>
>> It is - but the packages (rpms) are all unsigned
>>
>>
>> Chris
>>
>> On Jan 19 2016, at 3:15 am, Bogdan-Andrei Iancu 
>> wrote:
>> Hey, it seems to be up again !!! :))
>>
>> Regards,
>>
>> Bogdan-Andrei Iancu
>> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>>
>> On 18.01.2016 10:24, Nick Altmann wrote:
>>
>> Thank you. Yes, you’re right. It’s still down. New repository will be up
>> a bit later today.
>>
>> --
>> Nick
>>
>> 2016-01-18 11:03 GMT+03:00 Markus Rydh :
>>
>> Nick, the yum repo seems to be down for the last few days. Any updates on
>> when it might be available? Appreciate your work on hosting the repo!
>>
>> // Markus
>>
>>
>>
>> --
>> View this message in context:
>> http://opensips-open-sip-server.1449251.n2.nabble.com/YUM-repository-downtime-tp7600572p7600780.html
>> Sent from the OpenSIPS - Users mailing list archive at Nabble.com.
>>
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Re: [OpenSIPS-Users] B2B BYEs

[Bogdan-Andrei Iancu]

Hi Søren,

Thanks to all your support and info I found the problem. The fix is now 
available on GIT repo on all maintained versions.


Please update and confirm.

Regards,

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

On 19.01.2016 12:38, Søren Andersen wrote:


Hi Bogdan-Andrei,

Hmm.. Actually it look likes “route_se” is missing in the LEGS:: after 
a restart?


# Before restart #

dlg:: 227897 param=774.0 state=5 last_invite_cseq=29551 last_method=4 
last_reply_code=200 db_flag=2


callid:: B2B.348.227897

from::  uri=sip:xxx...@xx.xx.xx.36;user=phone 
tag=e33f629b154d16f5f57a07bf5d17124e-7644


to::  uri=sip:xxx...@xx.xx.xx.120 tag=2075842316

cseq::  caller=29551 callee=1

route_set:: 
caller=


contact::  caller=sip:xx.xx.xx.117:5060 
callee=sip:xx04@10.101.16.179:5060;transport=udp


send_sock:: xx.xx.xx.117

LEGS::

leg:: 0 tag=2075842316 cseq=29551 
contact=sip:xx04@10.101.16.179:5060;transport=udp 
route_se=


# After restart #

dlg:: 227897 param=774.0 state=5 last_invite_cseq=29551 last_method=0 
db_flag=0


callid:: B2B.348.227897

from::  uri=sip:x...@xx.xx.xx.36;user=phone 
tag=e33f629b154d16f5f57a07bf5d17124e-7644


to::  uri=sip:x...@xx.xx.xxx.120 tag=2075842316

cseq::  caller=29551 callee=1

route_set:: 
caller=


contact::  caller=sip:xx.xx.xx.117:5060 
callee=sip:xx04@10.101.16.179:5060;transport=udp


send_sock:: xx.xx.xx.117

LEGS::

leg:: 0 tag=2075842316 cseq=29551 
contact=sip:xx04@10.101.16.179:5060;transport=udp


/Søren

*Fra:*Bogdan-Andrei Iancu [mailto:bog...@opensips.org]
*Sendt:* 19. januar 2016 11:13
*Til:* Søren Andersen ; users@lists.opensips.org
*Emne:* Re: SV: SV: SV: [OpenSIPS-Users] B2B BYEs

Hi Søren,

It seems that the callee route set is stored in db (see the 
b2b_entities table, the route1 field 
="")


Can you run the b2be_list MI command before and after the restart, to 
see if indeed, the callee route set is not properly restored ?


Regards,

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

On 15.01.2016 12:21, Søren Andersen wrote:

Yup.

SELECT * FROM b2b.b2b_logic;

# id, si_key, scenario, sstate, next_sstate, sparam0, sparam1,
sparam2, sparam3, sparam4, sdp, lifetime, e1_type, e1_sid,
e1_from, e1_to, e1_key, e2_type, e2_sid, e2_from, e2_to, e2_key,
e3_type, e3_sid, e3_from, e3_to, e3_key

'226', '360.0', 'StofaB2B', '1', '1', 'sip:...@xx.xx.xx.120
', '', '', '', '', '',
'1452896043', '0', 'server1',
'sip:...@oxd3-sxw.xx.txxx.xxx;user=phone
',
'sip:x...@xx.xx.xx.117:5060
', 'B2B.39.160', '1',
'client1', 'sip:x...@oxx3-sx.xx.xxx.xxx;user=phone
',
'sip:...@xx.xx.xx.120 ',
'B2B.448.7695796', NULL, NULL, NULL, NULL, NULL

SELECT * FROM b2b.b2b_entities;

# id, type, state, ruri, from_uri, to_uri, from_dname, to_dname,
tag0, tag1, callid, cseq0, cseq1, contact0, contact1, route0,
route1, sockinfo_srv, param, lm, lrc, lic, leg_cseq, leg_route,
leg_tag, leg_contact, leg_sockinfo

'316', '0', '5', 'sip:...@xx.xx.xx.117:5060;user=phone
',
'sip:x...@osx3-sxw.xx..xxx;user=phone
',
'sip:X04@10.250.224.22;user=phone
', '', '',
'mvgh6xvkp7z', 'B2B.39.160',
'1q2rqu6-3232128...@osdx-ssw.xx.xxx.xx
', '32736', '1',
'sip:x...@xx.xx.xx.36:5060;transport=udp
',
'sip:XX.XX.XX.117:5060', '', '',
'udp:XX.XX.XX.117:5060', '360.0', '0', '200', '32736', NULL, NULL,
NULL, NULL, NULL

'317', '1', '5', '', 'sip:x...@osdx-ssw.xx..xx;user=phone
',
'sip:x...@xx.xx.xx.120 ',
'', '', 'a9ff31f61f2601c916f6044c2e20e16a-3c47', '2761431845',
'B2B.448.7695796', '32737', '1', 'sip:XX.XXX.XXX.117:5060',
'sip:XX04@10.101.16.179:5060;transport=udp
',

'',
'', 'udp:XX.XX.XX.117:5060', '360.0', '4', '0', '32737', '32737',

Re: [OpenSIPS-Users] WSS Client did not present a TLS certificate

[Sebastian Sastre]

Bodgan,

Thanks . Yes that one wasn’t an error but i had the wrong private key
configured and the socket was disconnecting so i was able to generate new
certs and it worked fine. I got some help from IRC.
I still see the notice but the socket stays up and i can register.

Right now i have the signaling working perfect, but i have no audio either
way. Im trying to figure out why rtpengine is not working correctly.

Thanks again




On Tue, Jan 19, 2016 at 5:21 AM, Bogdan-Andrei Iancu 
wrote:

> Hi Sebastian,
>
> That message is just an INFO (not an error) - you say TLS handshake fails
> on opensips side as it expects a certificate from the end point ?
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
> On 18.01.2016 06:32, Sebastian Sastre wrote:
>
> I’ve been trying to setup WSS using 2.2 latest branch.
>
> When trying to open the web socket i get “ Client did not present a TLS
> certificate” . Im using the included default ssl certs for the server to
> avoid mistakes . What certificate is the user supposed to present?
>
> I tried using sip.js and jssip to connect without any luck. i also tried
> disabling cert requirement but didn’t work.
>
> —— Config ——-
>
> listen=wss:123.456.789.987:5060
> listen=tls:123.456.789.987:5061
> listen=wss:123.456.789.987:443
>
> load module "proto_udp.so"
> load module “proto_tls.so”
> loadmodule "proto_wss.so"
>
> loadmodule "tls_mgm.so"
> modparam("tls_mgm", "certificate", "/etc/opensips/tls/rootCA/cacert.pem")
>
> modparam("tls_mgm", "private_key",
> "/etc/opensips/tls/rootCA/private/cakey.pem")
> modparam("tls_mgm", "ca_list", "/etc/opensips/tls/rootCA/cacert.pem")
>
> modparam("tls_mgm", "ca_dir", "/etc/opensips/tls/rootCA/")
> modparam("tls_mgm", "require_cert", "0")
> modparam(“tls_mgm", "verify_cert", "0")
>
>
> ——- Logs ——-
> /sbin/opensips[12468]: INFO:core:probe_max_sock_buff: using snd buffer of
> 416 kb
> /sbin/opensips[12468]: INFO:core:init_sock_keepalive: TCP keepalive
> enabled on socket 37
> /sbin/opensips[12460]: INFO:proto_wss:ls_accept: New TLS connection from
> xx.xx.xx.xx:50815 accepted
> /sbin/opensips[12460]: INFO:proto_wss:tls_accept: Client did not present
> a TLS certificate
> /sbin/opensips[12460]: INFO:proto_wss:ls_dump_cert_info: tls_accept:
> local TLS server certificate subject: /CN=OpenSIPS/ST=
> 
> opensips.org/C=IP/emailAddress=t...@opensips.org/O=opensips.org, issuer:
> /CN=OpenSIPS/ST=
> opensips.org/C=IP/emailAddress=t...@opensips.org/O=opensips.org
>
>
> Thanks !
>
>
>
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>
>
>
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Re: [OpenSIPS-Users] Extra Tables Creation Error

[Nathaniel L. Keeling III]

No. None of the those tables were created.

Thanks

Nathaniel L Keeling

On 1/19/16 4:17 AM, Bogdan-Andrei Iancu wrote:

Hi Nathaniel,

Do you see the other "emergency" module related table (like 
emergency_report or emergency_service_provider) created ?


Regards,

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

On 18.01.2016 21:38, Nathaniel L. Keeling III wrote:

Hello,

I was running the "opensipsdbctl create" to create the tables in 
postgresql 8.4 and received this error when creating the extra tables 
for Opensips 2.1:


ERROR:  relation "emergency_routing" does not exist
ERROR:  relation "emergency_routing_id_seq" does not exist
ERROR: Grant privileges to extra tables failed!

I checked the database and neither the emergency_routing or the 
emergency_report tables were created. All other table creations seem 
to be ok.


Thanks

Nathaniel L Keeling

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Re: [OpenSIPS-Users] YUM repository downtime

[Chris Stone]

It is - but the packages (rpms) are all unsigned

  

  

Chris

> On Jan 19 2016, at 3:15 am, Bogdan-Andrei Iancu bog...@opensips.org
wrote:  
Hey, it seems to be up again !!! :))  
  
Regards,  

>  
>  
> Bogdan-Andrei Iancu

> OpenSIPS Founder and Developer

> 

>

> On 18.01.2016 10:24, Nick Altmann wrote:  

>

>> Thank you. Yes, you’re right. It’s still down. New repository will be up a
bit later today.

>>

>>  

>>

>> \--  
Nick

>>

>>  

>>

>> 2016-01-18 11:03 GMT+03:00 Markus Rydh
[markus.r...@gmail.com](mailto:markus.r...@gmail.com):  

>>

>>> Nick, the yum repo seems to be down for the last few days. Any updates on  
when it might be available? Appreciate your work on hosting the repo!  
  
// Markus  
  
  
  
\--  
View this message in context:   
Sent from the OpenSIPS - Users mailing list archive at Nabble.com.  
  
___  
Users mailing list  
[Users@lists.opensips.org](mailto:Users@lists.opensips.org)  
  

>>

>>  

>>

>>  
  

>>  
>>  
>> ___

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>> [Users@lists.opensips.org](mailto:Users@lists.opensips.org)

>> 

>>  
>

>  

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Re: [OpenSIPS-Users] YUM repository downtime

[Nick Altmann]

The repo is in testing mode.
Have you tried today? Try again, all rpms are signed.

--
Nick

2016-01-19 19:43 GMT+03:00 Chris Stone :

> It is - but the packages (rpms) are all unsigned
>
>
> Chris
>
>> On Jan 19 2016, at 3:15 am, Bogdan-Andrei Iancu 
>> wrote:
>> Hey, it seems to be up again !!! :))
>>
>> Regards,
>>
>> Bogdan-Andrei Iancu
>> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>>
>> On 18.01.2016 10:24, Nick Altmann wrote:
>>
>> Thank you. Yes, you’re right. It’s still down. New repository will be up
>> a bit later today.
>>
>> --
>> Nick
>>
>> 2016-01-18 11:03 GMT+03:00 Markus Rydh :
>>
>> Nick, the yum repo seems to be down for the last few days. Any updates on
>> when it might be available? Appreciate your work on hosting the repo!
>>
>> // Markus
>>
>>
>>
>> --
>> View this message in context:
>> http://opensips-open-sip-server.1449251.n2.nabble.com/YUM-repository-downtime-tp7600572p7600780.html
>> Sent from the OpenSIPS - Users mailing list archive at Nabble.com.
>>
>> ___
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>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>>
>>
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>>
>>
>>
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>
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Re: [OpenSIPS-Users] WSS Client did not present a TLS certificate

[Bogdan-Andrei Iancu]

Glad the problem was solved.

Still, maybe there is place to improve the code to properly report/log 
the issue - did you get any indication (in logs) that actually the key 
was bogus and the TLS handshake failed ?


Regards,

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

On 19.01.2016 17:36, Sebastian Sastre wrote:

Bodgan,

Thanks . Yes that one wasn’t an error but i had the wrong private key 
configured and the socket was disconnecting so i was able to generate 
new certs and it worked fine. I got some help from IRC.

I still see the notice but the socket stays up and i can register.

Right now i have the signaling working perfect, but i have no audio 
either way. Im trying to figure out why rtpengine is not working 
correctly.


Thanks again




On Tue, Jan 19, 2016 at 5:21 AM, Bogdan-Andrei Iancu 
> wrote:


Hi Sebastian,

That message is just an INFO (not an error) - you say TLS
handshake fails on opensips side as it expects a certificate from
the end point ?

Regards,

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

On 18.01.2016 06:32, Sebastian Sastre wrote:

I’ve been trying to setup WSS using 2.2 latest branch.

When trying to open the web socket i get “ Client did not present
a TLS certificate” . Im using the included default ssl certs for
the server to avoid mistakes . What certificate is the user
supposed to present?

I tried using sip.js and jssip to connect without any luck. i
also tried disabling cert requirement but didn’t work.

—— Config ——-

listen=wss:123.456.789.987:5060
listen=tls:123.456.789.987:5061
listen=wss:123.456.789.987:443

load module "proto_udp.so"
load module “proto_tls.so”
loadmodule "proto_wss.so"

loadmodule "tls_mgm.so"
modparam("tls_mgm", "certificate",
"/etc/opensips/tls/rootCA/cacert.pem")
modparam("tls_mgm", "private_key",
"/etc/opensips/tls/rootCA/private/cakey.pem")
modparam("tls_mgm", "ca_list",
"/etc/opensips/tls/rootCA/cacert.pem")
modparam("tls_mgm", "ca_dir", "/etc/opensips/tls/rootCA/")
modparam("tls_mgm", "require_cert", "0")
modparam(“tls_mgm", "verify_cert", "0")


——- Logs ——-
/sbin/opensips[12468]: INFO:core:probe_max_sock_buff: using snd
buffer of 416 kb
/sbin/opensips[12468]: INFO:core:init_sock_keepalive: TCP
keepalive enabled on socket 37
/sbin/opensips[12460]: INFO:proto_wss:ls_accept: New TLS
connection from xx.xx.xx.xx:50815 accepted
/sbin/opensips[12460]: INFO:proto_wss:tls_accept: Client did not
present a TLS certificate
/sbin/opensips[12460]: INFO:proto_wss:ls_dump_cert_info:
tls_accept: local TLS server certificate subject:

/CN=OpenSIPS/ST=opensips.org/C=IP/emailAddress=t...@opensips.org/O=opensips.org
,
issuer:

/CN=OpenSIPS/ST=opensips.org/C=IP/emailAddress=t...@opensips.org/O=opensips.org



Thanks !



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