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

[Nabeel]

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 Developerhttp://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 Developerhttp://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 Developerhttp://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] Request Timeout on INVITE After Idle Registration

[Nabeel]

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 Developerhttp://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 Developerhttp://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 Developerhttp://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] Cache design

[John Nash]

Yes I am also doing that with local cache but I was confused with
sql_cacher and with mongo db also it can be done?...I was just wondering
what are these for?

On Wed, Jan 13, 2016 at 1:40 AM,  wrote:

> John,
>
> I do something similar for caching auth user id and passwords using
> cachedb_redis/cachedb_local and db_postgres.
>
> My basic logic first checks cache (cache_fetch) to see if the key exists.
> If it does I will pv_proxy_authorize using the value stored in the
> populated avp from the fetch.  If the cache does not exist, I will query
> Postgres (avp_db_query) and store (cache_store) the value in cache with a
> lifetime so that it will automatically repopulate at a later time.  The
> only time I may adjust the cache out of band would be if the user modifies
> their SIP password through the portal which causes the system to delete the
> stale key/value from the cachedb store.
>
> This may or may not address your concern, but I hope it helps.
>
> Jarrod
>
> > On Jan 12, 2016, at 1:54 PM, John Nash  wrote:
> >
> > I am using local cache db module in order to keep user id and password
> in memory and now plan to keep some other data in memory too and that I
> want to keep in some centralized cache.
> >
> > It will be like Opensips1, Opensips2 <-> Cache server
> <-> Postgresql
> >
> > I also need to periodically update cache (using some mi commands may be)
> >
> > I saw many of the modules related to cache in opensips (Like sql_cacher,
> cachedb_* , DB_CACHEDB)  and I am confused on what should I use for my need
> and how these are all related.
> >
> > Any suggestions?
> > ___
> > Users mailing list
> > Users@lists.opensips.org
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
> ___
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>
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Re: [OpenSIPS-Users] Dialogs stay in memory until restart in state 5

[John Nash]

OK thank you I will try to find and patch.

On Tue, Jan 12, 2016 at 10:58 PM, Răzvan Crainea 
wrote:

> Hi, John!
>
> This issue was fixed in newer versions of OpenSIPS, and the fix will be
> part of OpenSIPS 2.1.2.
>
> Best regards,
> Răzvan
>
>
> On 01/12/2016 03:19 PM, John Nash wrote:
>
> I am using OpenSIPS (2.1.1 (x86_64/linux)) with dialogs module and
> topology hiding modules.
> I am not saving dialogs in database.
>
> After I run it for few hours and stop traffic i see hundreds of dialogs
> using fifo command which wont be deleted from memory until I restart. I see
> dialogs like ..
> dialog::  hash=4094:1281471850
> state:: 5
> user_flags:: 0
> timestart:: 0
> timeout:: 0
> callid:: 8abe7f9e-2cd5-1234-84b0-d4ae52bce047
> from_uri:: sip:4117834663@127.0.0.1
> to_uri:: sip:119603452947@127.0.0.1:5060
>
> I saw couple of old posts and tried using "Pp" parameter in create_dialog
> but it does not seem to be of any effect.
>
> In logs I also see messages like "ERROR:dialog:push_reply_in_dialog: [487]
> reply in dlg state [2]: missing TAG param in TO hdr" which I am not sure
> are related to my problem.
>
>
> ___
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> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
> --
> Răzvan Crainea
> OpenSIPS Core Developerhttp://www.opensips-solutions.com
>
>
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>
>
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Re: [OpenSIPS-Users] MenuConfig Compile Fail on Solaris Sparc

[Nathaniel L. Keeling III]

Just in case it was missed earlier, here is the end of the output from 
running "NICER=0 make" while compiling opensips.


Compiling net/net_tcp.c
gcc -m64 -Wcast-align  -mcpu=ultrasparc -I/usr/include/json 
-DPKG_MALLOC  -DSHM_MMAP  -DUSE_MCAST -DDISABLE_NAGLE  -DSTATISTICS  
-DHAVE_RESOLV_RES  -DF_MALLOC -DEXTRA_DEBUG   -DNAME='"opensips"' 
-DVERSION='"2.1.1"' -DARCH='"sparc64"' -DOS='"solaris"' -DCOMPILER='"gcc 
4.9.2"' -D__CPU_sparc64 -D__OS_solaris -D__SMP_yes 
-DCFG_DIR='"/opt//etc/opensips/"'  -DVERSIONTYPE='"git"' 
-DTHISREVISION='"7df8a1a"' -DFAST_LOCK -DADAPTIVE_WAIT 
-DADAPTIVE_WAIT_LOOPS=1024  -DHAVE_GETIPNODEBYNAME -DHAVE_SYS_SOCKIO_H 
-DHAVE_SCHED_YIELD -DHAVE_ALLOCA_H -DUSE_SIGACTION 
-D_POSIX_PTHREAD_SEMANTICS -DHAVE_DEVPOLL -DHAVE_SELECT -c net/net_tcp.c 
-o net/net_tcp.o

In file included from net/../mem/../locking.h:66:0,
 from net/../mem/../statistics.h:143,
 from net/../mem/../pt.h:34,
 from net/../mem/../dprint.h:55,
 from net/../mem/mem.h:38,
 from net/net_tcp.c:42:
net/../mem/../lock_alloc.h: In function 'lock_set_alloc':
net/../mem/../lock_alloc.h:70:13: warning: cast increases required 
alignment of target type [-Wcast-align]

ls->locks=(gen_lock_t*)((char*)ls+sizeof(gen_lock_set_t));
 ^
In file included from net/../globals.h:30:0,
 from net/net_tcp.c:44:
net/../ip_addr.h: In function 'sockaddr2ip_addr':
net/../ip_addr.h:224:25: warning: cast increases required alignment of 
target type [-Wcast-align]

memcpy(ip->u.addr, &((struct sockaddr_in*)sa)->sin_addr, 4);
 ^
net/../ip_addr.h:229:25: warning: cast increases required alignment of 
target type [-Wcast-align]

memcpy(ip->u.addr, &((struct sockaddr_in6*)sa)->sin6_addr, 16);
 ^
In file included from net/../reactor.h:41:0,
 from net/net_tcp.c:50:
net/../io_wait_loop.h: In function 'io_wait_loop_devpoll':
net/../io_wait_loop.h:426:4: error: 'e' undeclared (first use in this 
function)

e = get_fd_map(h, h->fd_array[r].fd);
^
net/../io_wait_loop.h:426:4: note: each undeclared identifier is 
reported only once for each function it appears in

In file included from net/../globals.h:30:0,
 from net/net_tcp.c:44:
net/net_tcp.c: In function 'tcp_connect_blocking':
net/../ip_addr.h:168:26: warning: cast increases required alignment of 
target type [-Wcast-align]

   _port_no = su_getport( (union sockaddr_union*)_su); \
  ^
net/net_tcp.c:270:4: note: in expansion of macro 'get_su_info'
get_su_info( servaddr, ip, port);
^
net/../ip_addr.h:168:26: warning: cast increases required alignment of 
target type [-Wcast-align]

   _port_no = su_getport( (union sockaddr_union*)_su); \
  ^
net/net_tcp.c:300:4: note: in expansion of macro 'get_su_info'
get_su_info( servaddr, ip, port);
^
net/../ip_addr.h:168:26: warning: cast increases required alignment of 
target type [-Wcast-align]

   _port_no = su_getport( (union sockaddr_union*)_su); \
  ^
net/net_tcp.c:319:5: note: in expansion of macro 'get_su_info'
 get_su_info( servaddr, ip, port);
 ^
Makefile.rules:25: recipe for target 'net/net_tcp.o' failed
make: *** [net/net_tcp.o] Error 1

thanks

nathaniel


On 1/6/16 3:32 AM, Răzvan Crainea wrote:

can you run 'NICER=0 make"

Thanks,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com
On 01/05/2016 07:55 PM, Nathaniel L. Keeling III wrote:

No. The compile just stops.

Thanks

Nathaniel

On 1/5/16 6:58 AM, Răzvan Crainea wrote:
I doubt you are missing something on your side. Are there any other 
errors below that?


Best regards,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com
On 01/05/2016 02:33 PM, Nathaniel L. Keeling III wrote:

Razvan,

Changing to -lcurses solved that problem. Thanks. Ran into another 
compile problem when compiling opensips 2.1. Here is the error:


Makefile.rules:25: recipe for target 'net/net_tcp.o' failed
make[1]: *** [net/net_tcp.o] Error 1
make[1]: Leaving directory '/usr/local/src/opensips/opensips_2_1

Is there something else that I am missing?


Thanks

Nathaniel

On 1/5/16 4:56 AM, Răzvan Crainea wrote:

Hi, Nathaniel!

It seems to be a common problem on Solaris[1] - some symbols are 
defined in libncurses, some in libcurses. Can you try to replace 
the '-lncurses' with '-lcurses' in menuconfig/Makefile (line 8)?
If that doesn't work, try 'LD_EXTRA_OPTS=-lcurses make menuconfig' 
and post the output.


[1] https://gcc.gnu.org/ml/gcc-help/1999-11n/msg00020.html

Best regards,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com
On 01/05/2016 07:18 AM, Nathaniel L. Keeling III wrote:
I am trying to install Opensips 2.1 on Solaris Sparc 10 and I am 
getting this error when I perform "make menuconfig":


# make menuconfig
Usage: grep -hblcnsviw pattern file . . .
/bin/sh: proto_: not found
make -C 

Re: [OpenSIPS-Users] why tls not enable

[Bogdan-Andrei Iancu]

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
udp0  0 127.0.0.1:5060   
   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
tcp0  0 0.0.0.0:25061
 0.0.0.0:* LISTEN
root@125-227-130-1:~# netstat -tuln | grep 5060
tcp0  0 0.0.0.0:25060
 0.0.0.0:* LISTEN
udp0  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  70056   892 ?  S10:12  
0:00 /usr/local/sbin/opensips -P /var/run/opensips.pid
root  1468  0.4  0.0  70052   464 ?  S10:12  
0:01 /usr/local/sbin/opensips -P /var/run/opensips.pid
root  1469  0.0  0.0  70052   464 ?  S10:12  
0:00 /usr/local/sbin/opensips -P /var/run/opensips.pid
root  1470  0.0  0.0  70052   656 ?  S10:12  
0:00 /usr/local/sbin/opensips -P /var/run/opensips.pid
root  1471  0.0  0.0  70052   656 ?  S10:12  
0:00 /usr/local/sbin/opensips -P /var/run/opensips.pid
root  1472  0.0  0.0  70052   656 ?  S10:12  
0:00 /usr/local/sbin/opensips -P /var/run/opensips.pid
root  1473  0.0  0.0  70052   896 ?  S10:12  
0:00 /usr/local/sbin/opensips -P /var/run/opensips.pid
root  1731  0.0  0.0  11744   924 pts/0R+ 10:18  
0:00 grep --color=auto opensips


On Tue, Jan 5, 2016 at 9:54 PM, Bogdan-Andrei Iancu

[OpenSIPS-Users] Userloc stats are showing zero registered users

[Husnain Taseer]

Dear Users,

I am using Server:: OpenSIPS (2.1.1 (x86_64/linux)) as a registrar server
where more than 20 users are registered and I can see their entries in
location table. but when I run fifo commands to see the statistics it shows
me 0 registered users:


[root@opensips ]# opensipsctl fifo get_statistics 'usrloc:'
usrloc:registered_users:: 0
usrloc:location-users:: 0
usrloc:location-contacts:: 0
usrloc:location-expires:: 0


and the output of "opensipsctl  monitor" is :

[cycle #: 5; if constant make sure server lives]
Server:: OpenSIPS (2.1.1 (x86_64/linux))
Now:: Tue Jan 12 16:13:14 2016
Up since:: Mon Jan 11 20:17:19 2016
Up time:: 71755 [sec]

Transaction Statistics:
tm:UAS_transactions:: 2011
tm:UAC_transactions:: 14222
tm:inuse_transactions:: 0

Stateless Server Statistics:
sl:sent_replies:: 258651
sl:sent_err_replies:: 15
sl:received_ACKs:: 979

UsrLoc Stats:
usrloc:registered_users:: 0
usrloc:location-users:: 0
usrloc:location-contacts:: 0
usrloc:location-expires:: 0


Please advise


Regards,
Husnain Taseer
VoIP Developer
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Re: [OpenSIPS-Users] tomcat external app listening to OpenSIPS events

[Bogdan-Andrei Iancu]

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.xmlrpc.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 error._

But OpenSIPS is not calling the method I'm testing on a Tomcat XMLRPC
serverlet. This is my actual event subscription code I'm using:

subscribe_event("E_SIP_MESSAGE","xmlrpc:192.168.3.132:8080:ccurbiz.xml
rpc.OpenSIPSInterface.msgtests");

I've tried several (many) syntax variations.

On the Tomcat server it always points to the root RPC2 and returns the
error code 404.

ccurbiz -> is my  java project name

xmlrpc -> is the servelet mapping to the Java class OpenSIPSInterface

msgtest -> is the method I'm trying to call

I wanted to make sure the xmlrpc server was working, I'm able to
successfully call the method with xmlrpc-test-tool

Any suggestion to be able to make this work with Tomcat server are
greatly appreciated!!

Thanks!

Juls

*From:*users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] *On Behalf Of *Razvan
Crainea
*Sent:* Wednesday, December 30, 2015 10:40 AM
*To:* users@lists.opensips.org
*Subject:* Re: [OpenSIPS-Users] external app listening to OpenSIPS/SIP
events

Hi, Julian!

Make sure you are raising that event from your script (i.e.
raise_event("E_SIP_MESSAGE")).

http://www.opensips.org/Documentation/Tutorials-EventInterface#toc9

Best regards,
Răzvan

On 12/24/2015 03:39 AM, Julian Kay wrote:

 Thanks for the input, but for now I need to work with XMLRPC.

 is it possible to create a custom event? I've been trying to use in
 the startup_route: subscribe_event("E_SIP_MESSAGE",
 "xmlrpc:http://192.168.3.201:/msgtests;)

 I get the error: ERROR: core:evi_event_subscribe: invalid event name
 

 Any help much appreciated, thx!!!

 Juls

 *From:*users-boun...@lists.opensips.org
 
 [mailto:users-boun...@lists.opensips.org] *On Behalf Of *Tito Cumpen
 *Sent:* Wednesday, December 16, 2015 6:19 PM
 *To:* OpenSIPS users mailling list
 *Subject:* Re: [OpenSIPS-Users] external app listening to
 OpenSIPS/SIP events

 Julian,

 Look into using
 http://www.opensips.org/html/docs/modules/devel/event_rabbitmq This
 module allows you to spin up an event based at any moment in the
 script and pass variables in the process(Meaning it is a
 publisher.). Your app can subscribe as reader of the queue and do
 whatever task needs to be done. You can go as far as using this CDR
 as well.

 Goodluck,

 Tito

 On Wed, Dec 16, 2015 at 6:14 PM, Julian Kay > wrote:

 

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

[Bogdan-Andrei Iancu]

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] apt.opensips.org not reachable

[Bogdan-Andrei Iancu]

Hi,

The APT repository is to be maintained from now on by Ncik Altmann - he 
is also the maintainer of our YUM repo and so far he did more than a 
great job ! Many thanks for that !


The APT repo will be available in the next couple of days with Debian 
and Ubuntu builds. Nick will keep you all posted with the progress.


Regards,

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

On 05.01.2016 14:27, Bogdan-Andrei Iancu wrote:

Hi all,

Sorry for this inconvenient. I managed to get in touch with the 
maintainer for the APT repo, but still waiting for a final feedback on 
the repo status.
Nick Altmann, the maintainer of our yum repo, already volunteered for 
running the apt repo too...


I'm still waiting for an answer from our current maintainer, if not, 
we will migrate the repo in the next 2 days.


Thanks for understanding,

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

On 05.01.2016 13:46, Frederik Bjerggaard Nielsen wrote:

Great with some response on this!

Feel free to ping me when you have any news on this.

---
Best regards
Frederik Bjerggard Nielsen
Firstcom A/S


-Original Message-
From: users-boun...@lists.opensips.org [mailto:users-
boun...@lists.opensips.org] On Behalf Of Razvan Crainea
Sent: Monday, January 04, 2016 2:27 PM
To: users@lists.opensips.org
Subject: Re: [OpenSIPS-Users] apt.opensips.org not reachable

Hello all!

Unfortunately the maintainers of the code itself[1] are not the 
maintainers
of the repositories[2]. We've been trying to get in touch with the 
maintainers
of the debian repo[2] for a while, but unfortunately without any 
success.
The reason why we have not yet started configuring our own 
repository is
because A) our focus was on development, B) we didn't allocate 
resources
for this and probably the most important reason, C) we are pushing 
hard to

add OpenSIPS in the debian official repository.

Frederick, we've been trying to get in touch with the current 
maintainers to
see if they are willing to restore the repo or not. If they do, I 
don't think it's
worth having two different repos for the same sources. If they 
don't, we're

happy to help you with what you need to setup a new repo.

Sorry for the inconvenience.

[1] https://github.com/OpenSIPS/opensips
[2] apt.opensips.org

Best regards,

Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com

On 01/04/2016 02:39 PM, Frederik Bjerggaard Nielsen wrote:

That repository has been down since at least the 8th of December 2015.

Back then I said that we were interested in hosting the APT 
repository for

some time but heard nothing from the maintainers.

That offer is still standing by the way.

---
Best regards

Frederik Bjerggard Nielsen
Technical Specialist

Firstcom A/S


-Original Message-
From: users-boun...@lists.opensips.org [mailto:users-
boun...@lists.opensips.org] On Behalf Of Edwin
Sent: Monday, January 04, 2016 1:35 PM
To: users@lists.opensips.org
Subject: Re: [OpenSIPS-Users] apt.opensips.org not reachable

First of all a Happy New Year to all!!!

Is there a reason why http://apt.opensips.org is down? I know we can
compile from github but still it is very convenient to use apt.

  From a project as Opensips I would expect information to be a 
little

accurate (see http://opensips.org/Downloads/Downloads), the
http://debian.leurent.eu/ is not completely outdated.

Any chance the repository will become live again? (An update form the
maintainers would be nice).



--
View this message in context: http://opensips-open-sip-
server.1449251.n2.nabble.com/apt-opensips-org-not-reachable-
tp7600219p7600378.html
Sent from the OpenSIPS - Users mailing list archive at Nabble.com.

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Re: [OpenSIPS-Users] sdp answer

[Bogdan-Andrei Iancu]

Hi Riko,

I understand, but my knowledge over rtpengine is not so good. Maybe you 
should ask over the rptengine project.


Best regards,

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

On 12.01.2016 08:24, riko nir wrote:

Hi Andrei,

The opensips and rtpengine is interacting, for overwriting the SDP. We 
tested "JsSIP client with (opesips + rtpengine)" and it is working fine.


Here, DTLS handshake is happening using rtpengine, and so SRTP is 
handling by rtpengine.


But I don't want rtpengine to handle SRTP traffic. Only the "DTLS 
handshake" needs to be handled by RTPEngine and I just need the crypto 
method and the crypto keys from rtpengine to be sent to opensips.

Do you have any suggestion for fetching the crypto keys from rtpengine?

Thanks,
Riko

On Mon, Jan 11, 2016 at 4:24 PM, Bogdan-Andrei Iancu 
> wrote:


When OpenSIPS handles the SIP messages carrying SDP (typically
INVITE request and 200 OK reply), OpenSIPS will communicate with
rtpengine and update the SDP with the new IP and port. Never
tested, by AFAIK rtpengine may know to handle SRTP...not sure, you
may check with the project.

Regards,

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

On 11.01.2016 12:00, riko nir wrote:

Ok. Incase if the media needs to go via rtpengine, then how the
signaling happens, and how the rtpengine is forwarding the media
to the other end?

In case of DTLS, only the DTLS handlshake needs to be taken care
by the RtpEngine, and the SRTP is handled by some other media
server, and only the DTLS-key needs to be send to the media
server. In that case, how can we fetch dtls-key after the DTLS
handshake?

Thanks.

On Mon, Jan 11, 2016 at 3:13 PM, Bogdan-Andrei Iancu
> wrote:

Hi Riko,

There are no SDP answers created by opensips - it simply
changes the received SDP offer and it forwards it to the next
destination - basically it is doing proxying (performing some
changes too) of the SDP offers between the end points.

Best regards,

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

On 06.01.2016 15:42, riko nir wrote:

Hi all,

How is the communication flow happens between opensips and
rtpengine incase of a call through SIP over websocket?
When opensips receives a sdp offer from a web-based sip
client, opensips is creating sdp answer or rtpengine is
creating sdp anwer or rtpengine just updating ice
information only in sdp and other than that, media related
info is created by opensips?

Thanks.
Riko


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[OpenSIPS-Users] Dialogs stay in memory until restart in state 5

[John Nash]

I am using OpenSIPS (2.1.1 (x86_64/linux)) with dialogs module and topology
hiding modules.
I am not saving dialogs in database.

After I run it for few hours and stop traffic i see hundreds of dialogs
using fifo command which wont be deleted from memory until I restart. I see
dialogs like ..
dialog::  hash=4094:1281471850
state:: 5
user_flags:: 0
timestart:: 0
timeout:: 0
callid:: 8abe7f9e-2cd5-1234-84b0-d4ae52bce047
from_uri:: sip:4117834663@127.0.0.1
to_uri:: sip:119603452947@127.0.0.1:5060

I saw couple of old posts and tried using "Pp" parameter in create_dialog
but it does not seem to be of any effect.

In logs I also see messages like "ERROR:dialog:push_reply_in_dialog: [487]
reply in dlg state [2]: missing TAG param in TO hdr" which I am not sure
are related to my problem.
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Re: [OpenSIPS-Users] B2B BYEs

[Søren Andersen]

Hello there,

I've some strange issues with the B2B module. - I use MySQL as backend, and if 
I reload OpenSIPS the BYE is sent directly to the client, and not my SIP proxy. 
My B2B and OpenSIPS is two difference servers.
If I take a look in b2b_entities table I notice the contact1 field contains the 
IP address of the client.  But if I don't reload the OpenSIPS the BYE will be 
sent correctly the my proxy, and then to the client.

Now, my question is how can I make sure that B2B server will not try to 
communicate with the clients after a reload?

My topology:
ISP -> B2B -> Proxy -> Client

My config:

loadmodule "b2b_entities.so"
loadmodule "b2b_logic.so"
modparam("b2b_logic", "script_scenario", "/etc/opensips/b2b.xml")
modparam("b2b_entities", "db_url", "mysql://opensips:x...@xx.xx.xx.xx/b2b")
modparam("b2b_entities", "db_mode", 1)
modparam("b2b_logic", "db_mode", 1)
modparam("b2b_logic", "db_url", "mysql://opensips:x...@xx.xx.xx.xx/b2b")
modparam("b2b_entities", "replication_mode", 1)
modparam("b2b_logic", "init_callid_hdr", "Init-CallID")

if (is_method("INVITE") && !has_totag()) {
b2b_init_request("B2B","sip:$t...@xx.xx.xx.xx");
exit;
}

/Søren
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Re: [OpenSIPS-Users] opensips wss support

[Eric Tamme]

For wss to work, your site must be https, and your sip server must be 
wss.  You can not downgrade the connection anywhere.  Also if you are 
linking to sip.js, the url MUST ALSO BE https e.g:


src="https://cdn.rawgit.com/onsip/SIP.js/0.7.2/dist/sip-0.7.2.min.js 
https://cdn.rawgit.com/onsip/SIP.js/0.7.2/dist/sip-0.7.2.min.js>">



For 2.X you must use the new tls_mgm module as the cert management is 
now shared between tls and wss.


for example:

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

loadmodule "tls_mgm.so"
loadmodule "proto_tls.so"
loadmodule "proto_wss.so"


modparam("tls_mgm", "certificate","/etc/letsencrypt/live/acme.com/cert.pem")
modparam("tls_mgm", 
"private_key","/etc/letsencrypt/live/acme.com/privkey.pem")



I am using sip.js 0.7.2 with latest master from opensips and doing full 
https + wss


-Eric


On 01/12/2016 02:54 PM, Tito Cumpen wrote:

Do Nguyen Ha,

I was getting errors about the ws destination being unencrypted when 
using sip.js.



Razvan or community,

What cert configuration does it utilize? does it take it from the tls 
configuration?


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

In this tutorial I see the protocol being defined. Would these 
settings apply or would a respective proto_wss have to be provided in 
the config?



Thanks,
Tito

On Tue, Jan 5, 2016 at 7:17 AM, Do Nguyen Ha > wrote:


hi Tito

i did test the webrtc with chrome version 47. we only need to
setup web server with HTTPS only that point to jssip source code.
the jssip should works properly

there is no need wss on opensips server - just use ws on opensips

thank you

On Jan 5, 2016 5:53 PM, "Tito Cumpen" > wrote:

Hey Razvan,

Any updates on wss?

On Dec 7, 2015 12:35 PM, "Tito Cumpen" > wrote:

Hey Răzvan,

Any updates on this? Getuser media has been disabled when
using http and using https enforces wss on chrome.

Thanks,
Tito


On Fri, Nov 20, 2015 at 4:44 AM, Răzvan Crainea
> wrote:

Hi, Tito!

I am working on it. I have already started a local
branch and started coding, hopefully it will be public
by the end of the month.

Best regards,

Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com


On 11/19/2015 07:35 PM, Tito Cumpen wrote:

Razvan,


Is there currently a timeline for wss as of now? I
can aid in testing if needed.




On Wed, Nov 4, 2015 at 1:25 PM, Tito Cumpen
> wrote:

Razvan,


Thanks for the reply. Can't wait to try this
feature out.

On Tue, Nov 3, 2015 at 3:32 AM, Răzvan Crainea
> wrote:

Hi, Tito!

Yes, we have WSS support on top of our
priorities, and most likely will be released
with the next OpenSIPS 2.2.

Best regards,

Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com


On 10/16/2015 01:41 AM, Tito Cumpen wrote:

To opensips devs.


Are there any plans to support wss?
Currently chrome will not allow for an
unsecure websocket to be initiated when
using https. They also threaten to remove
getusermedia when using ws.


https://sites.google.com/a/chromium.org/dev/Home/chromium-security/deprecating-powerful-features-on-insecure-origins

Please advise. I am working on a project
that depends on this.

Thanks,
Tito


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