Re: [SR-Users] After upgrade from openser 1.3.4 to kamailio 1.5.5 the same crash set

2011-02-09 Thread Andrew O. Zhukov 

Is DBG_QM_MALLOC exactly what you want?


[root@ kamailio-1.5.5-notls]# /usr/local/sbin/kamailio -V
version: kamailio 1.5.5-notls (x86_64/linux)
flags: STATISTICS, EXTRA_DEBUG, USE_IPV6, USE_TCP, DISABLE_NAGLE, 
USE_MCAST, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, 
MAX_URI_SIZE 1024, BUF_SIZE 65535, PKG_SIZE 4194304

poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
svnrevision: unknown
@(#) $Id: main.c 5608 2009-02-13 16:48:17Z henningw $
main.c compiled on 09:42:37 Feb 10 2011 with gcc 4.1.2


___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

Re: [SR-Users] After upgrade from openser 1.3.4 to kamailio 1.5.5 the same crash set

2011-02-09 Thread Daniel-Constantin Mierla 



On 2/10/11 8:12 AM, Andrew O. Zhukov wrote:
Couple month ago I sent whole set of crash-es from 1.3.4 to this 
maillist. Nobody respond me.
Probably they were forgotten in the history, if most of devs were 
offline at the moment you sent. Do you have a link to the thread, it may 
help reading what you sent at that time, as well.


Cheers,
Daniel



On 02/10/2011 08:53 AM, Daniel-Constantin Mierla wrote:

Hello,

from the subject I don't understand exactly: did you get this crash also
with 1.3.4? Is it reproducible?

This crash-es from 1.5.5. I rise it up on this weekend.
I do not shutdown server with 1.3.4 yet. I still keep all crashes there.


Looks like there is a buffer overflow. Can you recompile/reinstall with
memory debug on (in 1.5.x, see Makefile.vars)? The watch the logs and
see if you get any error related to buffer overwritten ops.

Ok. I'll do it.


Cheers,
Daniel

On 2/10/11 7:37 AM, Andrew O. Zhukov wrote:

[root@ tmp]# /usr/local/sbin/kamailio -V
version: kamailio 1.5.5-notls (x86_64/linux)
flags: STATISTICS, EXTRA_DEBUG, USE_IPV6, USE_TCP, DISABLE_NAGLE,
USE_MCAST, SHM_MMAP,
PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
MAX_URI_SIZE 1024,
BUF_SIZE 65535, PKG_SIZE 4194304
poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
svnrevision: unknown
@(#) $Id: main.c 5608 2009-02-13 16:48:17Z henningw $
main.c compiled on 12:38:36 Feb 2 2011 with gcc 4.1.2


-
Core was generated by `/usr/local/sbin/kamailio -P
/var/run/openser/openser.pid -m 32 -u
openser -g op'.
Program terminated with signal 11, Segmentation fault.
#0 0x0046b0e3 in fm_malloc (qm=0x72dc00, size=32) at
mem/f_malloc.c:354
354 if ((*f)->size>=size) goto found;
(gdb) backtrace
#0 0x0046b0e3 in fm_malloc (qm=0x72dc00, size=32) at
mem/f_malloc.c:354
#1 0x2b30f2803087 in build_rr (_l=0x76f110, _l2=0x76fe80,
user=0x7fffe9c5a500,
tag=0x777a58, params=0x0, _inbound=0)
at record.c:176
#2 0x2b30f2802b7a in record_route (_m=0x76e0e0, params=0x0) at
record.c:322
#3 0x2b30f28047db in w_record_route (msg=0x76e0e0, key=0x0,
bar=0x0) at rr_mod.c:212
#4 0x0040ed9b in do_action (a=0x73f5a0, msg=0x76e0e0) at
action.c:874
#5 0x0040c03a in run_action_list (a=0x73f5a0, msg=0x76e0e0) at
action.c:145
#6 0x0040e6a7 in do_action (a=0x73f810, msg=0x76e0e0) at
action.c:746
#7 0x0040c03a in run_action_list (a=0x73e418, msg=0x76e0e0) at
action.c:145
#8 0x0040c2a9 in run_actions (a=0x73e418, msg=0x76e0e0) at
action.c:120
#9 0x0040c357 in run_top_route (a=0x73e418, msg=0x76e0e0) at
action.c:195
#10 0x0043bda4 in receive_msg (
buf=0x70c980 "NOTIFY sip:XX.com SIP/2.0\r\nVia: SIP/2.0/UDP
XX.XXX.101.68:5060;branch=z9hG4bK-6ee3865\r\nFrom: VTHome
;tag=129d73a13db8ec7fo0\r\nTo:
\r\nCall-ID:
e3fd1da9-142a0a17"..., len=373,
rcv_info=0x7fffe9c5ae90) at receive.c:175
#11 0x00467eeb in udp_rcv_loop () at udp_server.c:449
#12 0x0042097b in main_loop () at main.c:774
#13 0x004228b0 in main (argc=11, argv=0x7fffe9c5b118) at
main.c:1321
(gdb) print size
$1 = 32
(gdb) quit

Core was generated by `/usr/local/sbin/kamailio -P
/var/run/openser/openser.pid -m 32 -u
openser -g op'.
Program terminated with signal 11, Segmentation fault.
#0 0x0046bf7b in fm_status (qm=0x72dc00) at mem/f_malloc.c:609
609 size+=f->size,f=f->u.nxt_free,i++,j++){
(gdb) backtrace
#0 0x0046bf7b in fm_status (qm=0x72dc00) at mem/f_malloc.c:609
#1 0x0041feb3 in sig_usr (signo=15) at main.c:563
#2 
#3 0x0039d8cd4a51 in __recvfrom_nocancel () from /lib64/libc.so.6
#4 0x00467bf4 in udp_rcv_loop () at udp_server.c:408
#5 0x0042097b in main_loop () at main.c:774
#6 0x004228b0 in main (argc=11, argv=0x7fffe9c5b118) at
main.c:1321
(gdb) print i
$1 = 402
(gdb) print j
$2 = 1
(gdb) print size
$3 = 7234295468789601279
(gdb) print f
$4 = (struct fm_frag *) 0x3738656435393838
(gdb) print f->size
Cannot access memory at address 0x3738656435393838
---



Andrew O. Zhukov

___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users






--
Daniel-Constantin Mierla
http://www.asipto.com


___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

Re: [SR-Users] After upgrade from openser 1.3.4 to kamailio 1.5.5 the same crash set

2011-02-09 Thread Andrew O. Zhukov 
Couple month ago I sent whole set of crash-es from 1.3.4 to this 
maillist. Nobody respond me.


On 02/10/2011 08:53 AM, Daniel-Constantin Mierla wrote:

Hello,

from the subject I don't understand exactly: did you get this crash also
with 1.3.4? Is it reproducible?

This crash-es from 1.5.5. I rise it up on this weekend.
I do not shutdown server with 1.3.4 yet. I still keep all crashes there.


Looks like there is a buffer overflow. Can you recompile/reinstall with
memory debug on (in 1.5.x, see Makefile.vars)? The watch the logs and
see if you get any error related to buffer overwritten ops.

Ok. I'll do it.


Cheers,
Daniel

On 2/10/11 7:37 AM, Andrew O. Zhukov wrote:

[root@ tmp]# /usr/local/sbin/kamailio -V
version: kamailio 1.5.5-notls (x86_64/linux)
flags: STATISTICS, EXTRA_DEBUG, USE_IPV6, USE_TCP, DISABLE_NAGLE,
USE_MCAST, SHM_MMAP,
PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
MAX_URI_SIZE 1024,
BUF_SIZE 65535, PKG_SIZE 4194304
poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
svnrevision: unknown
@(#) $Id: main.c 5608 2009-02-13 16:48:17Z henningw $
main.c compiled on 12:38:36 Feb 2 2011 with gcc 4.1.2


-
Core was generated by `/usr/local/sbin/kamailio -P
/var/run/openser/openser.pid -m 32 -u
openser -g op'.
Program terminated with signal 11, Segmentation fault.
#0 0x0046b0e3 in fm_malloc (qm=0x72dc00, size=32) at
mem/f_malloc.c:354
354 if ((*f)->size>=size) goto found;
(gdb) backtrace
#0 0x0046b0e3 in fm_malloc (qm=0x72dc00, size=32) at
mem/f_malloc.c:354
#1 0x2b30f2803087 in build_rr (_l=0x76f110, _l2=0x76fe80,
user=0x7fffe9c5a500,
tag=0x777a58, params=0x0, _inbound=0)
at record.c:176
#2 0x2b30f2802b7a in record_route (_m=0x76e0e0, params=0x0) at
record.c:322
#3 0x2b30f28047db in w_record_route (msg=0x76e0e0, key=0x0,
bar=0x0) at rr_mod.c:212
#4 0x0040ed9b in do_action (a=0x73f5a0, msg=0x76e0e0) at
action.c:874
#5 0x0040c03a in run_action_list (a=0x73f5a0, msg=0x76e0e0) at
action.c:145
#6 0x0040e6a7 in do_action (a=0x73f810, msg=0x76e0e0) at
action.c:746
#7 0x0040c03a in run_action_list (a=0x73e418, msg=0x76e0e0) at
action.c:145
#8 0x0040c2a9 in run_actions (a=0x73e418, msg=0x76e0e0) at
action.c:120
#9 0x0040c357 in run_top_route (a=0x73e418, msg=0x76e0e0) at
action.c:195
#10 0x0043bda4 in receive_msg (
buf=0x70c980 "NOTIFY sip:XX.com SIP/2.0\r\nVia: SIP/2.0/UDP
XX.XXX.101.68:5060;branch=z9hG4bK-6ee3865\r\nFrom: VTHome
;tag=129d73a13db8ec7fo0\r\nTo:
\r\nCall-ID:
e3fd1da9-142a0a17"..., len=373,
rcv_info=0x7fffe9c5ae90) at receive.c:175
#11 0x00467eeb in udp_rcv_loop () at udp_server.c:449
#12 0x0042097b in main_loop () at main.c:774
#13 0x004228b0 in main (argc=11, argv=0x7fffe9c5b118) at
main.c:1321
(gdb) print size
$1 = 32
(gdb) quit

Core was generated by `/usr/local/sbin/kamailio -P
/var/run/openser/openser.pid -m 32 -u
openser -g op'.
Program terminated with signal 11, Segmentation fault.
#0 0x0046bf7b in fm_status (qm=0x72dc00) at mem/f_malloc.c:609
609 size+=f->size,f=f->u.nxt_free,i++,j++){
(gdb) backtrace
#0 0x0046bf7b in fm_status (qm=0x72dc00) at mem/f_malloc.c:609
#1 0x0041feb3 in sig_usr (signo=15) at main.c:563
#2 
#3 0x0039d8cd4a51 in __recvfrom_nocancel () from /lib64/libc.so.6
#4 0x00467bf4 in udp_rcv_loop () at udp_server.c:408
#5 0x0042097b in main_loop () at main.c:774
#6 0x004228b0 in main (argc=11, argv=0x7fffe9c5b118) at
main.c:1321
(gdb) print i
$1 = 402
(gdb) print j
$2 = 1
(gdb) print size
$3 = 7234295468789601279
(gdb) print f
$4 = (struct fm_frag *) 0x3738656435393838
(gdb) print f->size
Cannot access memory at address 0x3738656435393838
---



Andrew O. Zhukov

___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users





___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

Re: [SR-Users] Issue with avp

2011-02-09 Thread Daniel-Constantin Mierla 

Hello,


On 2/9/11 10:35 PM, Amit Nepal wrote:

Hi,
I have been trying to load rpid while loading credentials.

modparam("auth_db", "load_credentials", "$avp(i:123)=rpid")

Now, I am trying to do a check in my routing logic.

xlog("L_NOTICE","The avp is :$avp(i:123)");  (I dont get the value 
here either, i can't see the value when i do avp_print()

if($avp(i:123)<5)
{
sl_send_reply("408","Message Here");
}

I dont get the value of avp at that place. Any guidance please.

Is this piece of config after a successful www/proxy_authenticate()?

Cheers,
Daniel

--
Daniel-Constantin Mierla
http://www.asipto.com


___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

Re: [SR-Users] After upgrade from openser 1.3.4 to kamailio 1.5.5 the same crash set

2011-02-09 Thread Daniel-Constantin Mierla 

Hello,

from the subject I don't understand exactly: did you get this crash also 
with 1.3.4? Is it reproducible?


Looks like there is a buffer overflow. Can you recompile/reinstall with 
memory debug on (in 1.5.x, see Makefile.vars)? The watch the logs and 
see if you get any error related to buffer overwritten ops.


Cheers,
Daniel

On 2/10/11 7:37 AM, Andrew O. Zhukov wrote:

[root@ tmp]# /usr/local/sbin/kamailio -V
version: kamailio 1.5.5-notls (x86_64/linux)
flags: STATISTICS, EXTRA_DEBUG, USE_IPV6, USE_TCP, DISABLE_NAGLE, 
USE_MCAST, SHM_MMAP,

PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, 
MAX_URI_SIZE 1024,

BUF_SIZE 65535, PKG_SIZE 4194304
poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
svnrevision: unknown
@(#) $Id: main.c 5608 2009-02-13 16:48:17Z henningw $
main.c compiled on 12:38:36 Feb  2 2011 with gcc 4.1.2


-
Core was generated by `/usr/local/sbin/kamailio -P 
/var/run/openser/openser.pid -m 32 -u

openser -g op'.
Program terminated with signal 11, Segmentation fault.
#0  0x0046b0e3 in fm_malloc (qm=0x72dc00, size=32) at 
mem/f_malloc.c:354

354 if ((*f)->size>=size) goto found;
(gdb) backtrace
#0  0x0046b0e3 in fm_malloc (qm=0x72dc00, size=32) at 
mem/f_malloc.c:354
#1  0x2b30f2803087 in build_rr (_l=0x76f110, _l2=0x76fe80, 
user=0x7fffe9c5a500,

tag=0x777a58, params=0x0, _inbound=0)
at record.c:176
#2  0x2b30f2802b7a in record_route (_m=0x76e0e0, params=0x0) at 
record.c:322
#3  0x2b30f28047db in w_record_route (msg=0x76e0e0, key=0x0, 
bar=0x0) at rr_mod.c:212
#4  0x0040ed9b in do_action (a=0x73f5a0, msg=0x76e0e0) at 
action.c:874
#5  0x0040c03a in run_action_list (a=0x73f5a0, msg=0x76e0e0) 
at action.c:145
#6  0x0040e6a7 in do_action (a=0x73f810, msg=0x76e0e0) at 
action.c:746
#7  0x0040c03a in run_action_list (a=0x73e418, msg=0x76e0e0) 
at action.c:145
#8  0x0040c2a9 in run_actions (a=0x73e418, msg=0x76e0e0) at 
action.c:120
#9  0x0040c357 in run_top_route (a=0x73e418, msg=0x76e0e0) at 
action.c:195

#10 0x0043bda4 in receive_msg (
buf=0x70c980 "NOTIFY sip:XX.com SIP/2.0\r\nVia: SIP/2.0/UDP
XX.XXX.101.68:5060;branch=z9hG4bK-6ee3865\r\nFrom: VTHome
;tag=129d73a13db8ec7fo0\r\nTo: 
\r\nCall-ID:

e3fd1da9-142a0a17"..., len=373,
rcv_info=0x7fffe9c5ae90) at receive.c:175
#11 0x00467eeb in udp_rcv_loop () at udp_server.c:449
#12 0x0042097b in main_loop () at main.c:774
#13 0x004228b0 in main (argc=11, argv=0x7fffe9c5b118) at 
main.c:1321

(gdb) print size
$1 = 32
(gdb) quit

Core was generated by `/usr/local/sbin/kamailio -P 
/var/run/openser/openser.pid -m 32 -u

openser -g op'.
Program terminated with signal 11, Segmentation fault.
#0  0x0046bf7b in fm_status (qm=0x72dc00) at mem/f_malloc.c:609
609 size+=f->size,f=f->u.nxt_free,i++,j++){
(gdb) backtrace
#0  0x0046bf7b in fm_status (qm=0x72dc00) at mem/f_malloc.c:609
#1  0x0041feb3 in sig_usr (signo=15) at main.c:563
#2 
#3  0x0039d8cd4a51 in __recvfrom_nocancel () from /lib64/libc.so.6
#4  0x00467bf4 in udp_rcv_loop () at udp_server.c:408
#5  0x0042097b in main_loop () at main.c:774
#6  0x004228b0 in main (argc=11, argv=0x7fffe9c5b118) at 
main.c:1321

(gdb) print i
$1 = 402
(gdb) print j
$2 = 1
(gdb) print size
$3 = 7234295468789601279
(gdb) print f
$4 = (struct fm_frag *) 0x3738656435393838
(gdb) print f->size
Cannot access memory at address 0x3738656435393838
---



Andrew O. Zhukov

___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users


--
Daniel-Constantin Mierla
http://www.asipto.com


___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

[SR-Users] After upgrade from openser 1.3.4 to kamailio 1.5.5 the same crash set

2011-02-09 Thread Andrew O. Zhukov 

[root@ tmp]# /usr/local/sbin/kamailio -V
version: kamailio 1.5.5-notls (x86_64/linux)
flags: STATISTICS, EXTRA_DEBUG, USE_IPV6, USE_TCP, DISABLE_NAGLE, 
USE_MCAST, SHM_MMAP,

PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, 
MAX_URI_SIZE 1024,

BUF_SIZE 65535, PKG_SIZE 4194304
poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
svnrevision: unknown
@(#) $Id: main.c 5608 2009-02-13 16:48:17Z henningw $
main.c compiled on 12:38:36 Feb  2 2011 with gcc 4.1.2


-
Core was generated by `/usr/local/sbin/kamailio -P 
/var/run/openser/openser.pid -m 32 -u

openser -g op'.
Program terminated with signal 11, Segmentation fault.
#0  0x0046b0e3 in fm_malloc (qm=0x72dc00, size=32) at 
mem/f_malloc.c:354

354 if ((*f)->size>=size) goto found;
(gdb) backtrace
#0  0x0046b0e3 in fm_malloc (qm=0x72dc00, size=32) at 
mem/f_malloc.c:354
#1  0x2b30f2803087 in build_rr (_l=0x76f110, _l2=0x76fe80, 
user=0x7fffe9c5a500,

tag=0x777a58, params=0x0, _inbound=0)
at record.c:176
#2  0x2b30f2802b7a in record_route (_m=0x76e0e0, params=0x0) at 
record.c:322
#3  0x2b30f28047db in w_record_route (msg=0x76e0e0, key=0x0, 
bar=0x0) at rr_mod.c:212
#4  0x0040ed9b in do_action (a=0x73f5a0, msg=0x76e0e0) at 
action.c:874
#5  0x0040c03a in run_action_list (a=0x73f5a0, msg=0x76e0e0) at 
action.c:145
#6  0x0040e6a7 in do_action (a=0x73f810, msg=0x76e0e0) at 
action.c:746
#7  0x0040c03a in run_action_list (a=0x73e418, msg=0x76e0e0) at 
action.c:145
#8  0x0040c2a9 in run_actions (a=0x73e418, msg=0x76e0e0) at 
action.c:120
#9  0x0040c357 in run_top_route (a=0x73e418, msg=0x76e0e0) at 
action.c:195

#10 0x0043bda4 in receive_msg (
buf=0x70c980 "NOTIFY sip:XX.com SIP/2.0\r\nVia: SIP/2.0/UDP
XX.XXX.101.68:5060;branch=z9hG4bK-6ee3865\r\nFrom: VTHome
;tag=129d73a13db8ec7fo0\r\nTo: 
\r\nCall-ID:

e3fd1da9-142a0a17"..., len=373,
rcv_info=0x7fffe9c5ae90) at receive.c:175
#11 0x00467eeb in udp_rcv_loop () at udp_server.c:449
#12 0x0042097b in main_loop () at main.c:774
#13 0x004228b0 in main (argc=11, argv=0x7fffe9c5b118) at main.c:1321
(gdb) print size
$1 = 32
(gdb) quit

Core was generated by `/usr/local/sbin/kamailio -P 
/var/run/openser/openser.pid -m 32 -u

openser -g op'.
Program terminated with signal 11, Segmentation fault.
#0  0x0046bf7b in fm_status (qm=0x72dc00) at mem/f_malloc.c:609
609 
size+=f->size,f=f->u.nxt_free,i++,j++){

(gdb) backtrace
#0  0x0046bf7b in fm_status (qm=0x72dc00) at mem/f_malloc.c:609
#1  0x0041feb3 in sig_usr (signo=15) at main.c:563
#2  
#3  0x0039d8cd4a51 in __recvfrom_nocancel () from /lib64/libc.so.6
#4  0x00467bf4 in udp_rcv_loop () at udp_server.c:408
#5  0x0042097b in main_loop () at main.c:774
#6  0x004228b0 in main (argc=11, argv=0x7fffe9c5b118) at main.c:1321
(gdb) print i
$1 = 402
(gdb) print j
$2 = 1
(gdb) print size
$3 = 7234295468789601279
(gdb) print f
$4 = (struct fm_frag *) 0x3738656435393838
(gdb) print f->size
Cannot access memory at address 0x3738656435393838
---



Andrew O. Zhukov

___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

[SR-Users] Issue with avp

2011-02-09 Thread Amit Nepal 

Hi,
I have been trying to load rpid while loading credentials.

modparam("auth_db", "load_credentials", "$avp(i:123)=rpid")

Now, I am trying to do a check in my routing logic.

xlog("L_NOTICE","The avp is :$avp(i:123)");  (I dont get the value here either, 
i can't see the value when i do avp_print()
if($avp(i:123)<5)
{
sl_send_reply("408","Message Here");
}

I dont get the value of avp at that place. Any guidance please.


--
Thank You
Amit Nepal
Systems Administrator
Phoenix Internet
Phone: 602-385-0731
   602-234-0917#112
http://www.phoenixinternet.net


___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

Re: [SR-Users] LCR module : same IP address for different prefix.

2011-02-09 Thread Juha Heinanen 
Antanas Masevicius writes:

> Thank you for your work! This feature is highly needed i suspect.

it would be possible to cherry-pick the patch also to 3.1, but i don't
know if that is appropriate, because removing gw uniqueness check is not
strictly a bug fix.  it is just a piece of not needed code.

-- juha

___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

Re: [SR-Users] nathelper-module and sdp offer in 200 OK

2011-02-09 Thread Ovidiu Sas 
The code seems to be correct.  The to and from tags are switched for:
 - reply with offer (200ok with first SDP)
 - request with answer (ACK with second SDP)

Are you sure that you are properly invoking offer/answer rtpproxy functions?


Regards,
Ovidiu Sas

On Wed, Feb 9, 2011 at 11:48 AM, Emil Kroymann  wrote:
> Hi,
>
> We recently had a problem with the nathelper module and rtpproxy in a
> scenario where the SDP offer is sent only in the 200 OK. We use
> sip-router 3.1 and rtp-proxy from git master. The sip-router
> configuration uses the rtpproxy_offer() and rtpproxy_answer() functions
> in appropriate places. The problem is, that the arguments sent to
> the rtpproxy, when the ACK with the sdp answer arrives, seems to be not
> in the order, that rtpproxy expects.
>
> On the 200 OK, the nathelper module sends callid, to-tag, from-tag to
> rtpproxy. On the ACK, the nathelper module sends callid, from-tag,
> to-tag (with different command prefixes, of course, but I cannot
> remember them atm). The version of rtpproxy that we are using seems to
> expect, that the order of arguments sent on the ACK request is the same
> as on the 200 OK.
>
> My question: are there any module parameters, to correct this behaviour?
>
> Regards,
>
> Emil
> --
> Emil Kroymann
> VoIP Services Engineer
>
> Email: emil.kroym...@isaco.de
> Tel: +49-30-203899885
>
> ISACO GmbH
> Kurfürstenstraße 79
> 10787 Berlin
> Germany
>
> Amtsgericht Charlottenburg, HRB 112464B
> Geschäftsführer: Daniel Frommherz
>
>
> ___
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users@lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>

___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

[SR-Users] pdb module timeouts

2011-02-09 Thread Thomas Baumann 

Hello,

I am using the PDB module and server components for number portability. 2 
instances of PDB Server runs on (10.12.19.51/10001/10002), Kamailio on 
(10.12.19.21).
With a small amount of traffic (-cmax 150 -cps 10 -callduration 3), where are 
timeouts: WARNING: pdb [pdb.c:260]: exceeded timeout while waiting for response.

One requested number was 307111094, where the module prints out a timeout.

The funny part is, that I can see the responses at least arriving at the 
10.12.19.21 interface on time. 

Request send: 0,200855 s
Answer received: 0,201027 s

That are 0,172 ms and far away from a timeout.

What could be the reason ?

regards,

Thomas

ps. A small change on the server part was done: handle 4 digit carrier codes.

 
level3_logs:

Feb  9 17:19:42 node2 /openser/sbin/kamailio[21200]: DEBUG: pdb [pdb.c:201]: 
querying '307111094'... 
Feb  9 17:19:42 node2 /openser/sbin/kamailio[21200]: WARNING: pdb [pdb.c:260]: 
exceeded timeout while waiting for response. 



___
Schon gehört? WEB.DE hat einen genialen Phishing-Filter in die
Toolbar eingebaut! http://produkte.web.de/go/toolbar


pdb2.cap
Description: Binary data
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

Re: [SR-Users] nathelper-module and sdp offer in 200 OK

2011-02-09 Thread Ovidiu Sas 
I will check that and I will get back to you.


Regards,
Ovidiu Sas

On Wed, Feb 9, 2011 at 11:48 AM, Emil Kroymann  wrote:
> Hi,
>
> We recently had a problem with the nathelper module and rtpproxy in a
> scenario where the SDP offer is sent only in the 200 OK. We use
> sip-router 3.1 and rtp-proxy from git master. The sip-router
> configuration uses the rtpproxy_offer() and rtpproxy_answer() functions
> in appropriate places. The problem is, that the arguments sent to
> the rtpproxy, when the ACK with the sdp answer arrives, seems to be not
> in the order, that rtpproxy expects.
>
> On the 200 OK, the nathelper module sends callid, to-tag, from-tag to
> rtpproxy. On the ACK, the nathelper module sends callid, from-tag,
> to-tag (with different command prefixes, of course, but I cannot
> remember them atm). The version of rtpproxy that we are using seems to
> expect, that the order of arguments sent on the ACK request is the same
> as on the 200 OK.
>
> My question: are there any module parameters, to correct this behaviour?
>
> Regards,
>
> Emil
> --
> Emil Kroymann
> VoIP Services Engineer
>
> Email: emil.kroym...@isaco.de
> Tel: +49-30-203899885
>
> ISACO GmbH
> Kurfürstenstraße 79
> 10787 Berlin
> Germany
>
> Amtsgericht Charlottenburg, HRB 112464B
> Geschäftsführer: Daniel Frommherz
>
>
> ___
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users@lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>

___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

Re: [SR-Users] Use different certificate for different client with TLS

2011-02-09 Thread Klaus Darilion 
The certificate configured at Kamailio is the 'server' certificate.

The client certificate will be configured in the SIP client (e.g. the
SIP phone).

The [server] section will require a client certificate from the
softphone if require_certificate=yes. It will also validate the client
certificate against the local configured valid ca_list by using
verify_certificate=yes.

If now a SIP message enters the routing script and it was received by
TLS you know that the sip phone sent a clietn certificate which could be
validated with the ca_list.

now, if you want to differ softphones you have to take a look at the
certificate parameters, e.g. using select framework:
http://sip-router.org/docbook/sip-router/branch/master/select_list/select_list.html#select_list.tls

e.g.  if ( @tls.peer.subject.cn == "George Bush") ...

regards
Klaus

Am 09.02.2011 17:39, schrieb Daniel GROTTI:
> Hi all,
> I would like to use kamailio 3.1 with TLS and verified also a client
> certificate.
> 
> My tls.cfg file is as follow:
> 
> 
> --- tls.cfg 
> 
> .
> .
> 
> [server:MY_IP:5061]
> method = TLSv1
> verify_certificate = yes
> require_certificate = yes
> private_key = default_key.pem
> certificate = default_cert.pem
> ca_list = default_ca.pem
> 
> [client:default]
> verify_certificate = no
> require_certificate = no
> 
> 
> 
> 
> 
> So I want to verify the client certificate, to do that I use
> "is_peer_verified()" function in kamailio.cfg.
> As tls.cfg shows, I have to send to my clients the CA certificate and
> the client certificate (default_cert.pem + default_key.pem - signing
> by the CAcert).
> This client certificate is unique for all clients.
> Everything works fine.
> 
> But suppose I wanted to create a client certificate for client 1
> (cert_1.crt), and a different client certificate for client 2
> (cert_2.crt) and I want to configure kamailio to be able to verified
> this different certificates.
> 
> Does it possible ? How can I configure the tls.cfg file to do that ?
> 
> I try to do something like this:
> 
> [server:MY_IP:5061]
> method = TLSv1
> verify_certificate = yes
> require_certificate = yes
> private_key = default_key_1.pem
> certificate = default_cert_1.pem
> private_key = default_key_2.pem
> certificate = default_cert_2.pem
> ca_list = default_ca.pem
> 
> 
> But when kamailio restart it seems that it read only the last couple
> of row certifcate/private_key.
> 
> Regards,
> 
> 
> Daniel G
> 
> ___
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users@lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

[SR-Users] nathelper-module and sdp offer in 200 OK

2011-02-09 Thread Emil Kroymann 
Hi,

We recently had a problem with the nathelper module and rtpproxy in a
scenario where the SDP offer is sent only in the 200 OK. We use
sip-router 3.1 and rtp-proxy from git master. The sip-router
configuration uses the rtpproxy_offer() and rtpproxy_answer() functions
in appropriate places. The problem is, that the arguments sent to
the rtpproxy, when the ACK with the sdp answer arrives, seems to be not
in the order, that rtpproxy expects.

On the 200 OK, the nathelper module sends callid, to-tag, from-tag to
rtpproxy. On the ACK, the nathelper module sends callid, from-tag,
to-tag (with different command prefixes, of course, but I cannot
remember them atm). The version of rtpproxy that we are using seems to
expect, that the order of arguments sent on the ACK request is the same
as on the 200 OK. 

My question: are there any module parameters, to correct this behaviour?

Regards,

Emil
-- 
Emil Kroymann
VoIP Services Engineer

Email: emil.kroym...@isaco.de
Tel: +49-30-203899885

ISACO GmbH
Kurfürstenstraße 79
10787 Berlin
Germany

Amtsgericht Charlottenburg, HRB 112464B
Geschäftsführer: Daniel Frommherz



signature.asc
Description: PGP signature
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

[SR-Users] Use different certificate for different client with TLS

2011-02-09 Thread Daniel GROTTI 
Hi all,
I would like to use kamailio 3.1 with TLS and verified also a client
certificate.

My tls.cfg file is as follow:


--- tls.cfg 

.
.

[server:MY_IP:5061]
method = TLSv1
verify_certificate = yes
require_certificate = yes
private_key = default_key.pem
certificate = default_cert.pem
ca_list = default_ca.pem

[client:default]
verify_certificate = no
require_certificate = no





So I want to verify the client certificate, to do that I use
"is_peer_verified()" function in kamailio.cfg.
As tls.cfg shows, I have to send to my clients the CA certificate and
the client certificate (default_cert.pem + default_key.pem - signing
by the CAcert).
This client certificate is unique for all clients.
Everything works fine.

But suppose I wanted to create a client certificate for client 1
(cert_1.crt), and a different client certificate for client 2
(cert_2.crt) and I want to configure kamailio to be able to verified
this different certificates.

Does it possible ? How can I configure the tls.cfg file to do that ?

I try to do something like this:

[server:MY_IP:5061]
method = TLSv1
verify_certificate = yes
require_certificate = yes
private_key = default_key_1.pem
certificate = default_cert_1.pem
private_key = default_key_2.pem
certificate = default_cert_2.pem
ca_list = default_ca.pem


But when kamailio restart it seems that it read only the last couple
of row certifcate/private_key.

Regards,


Daniel G

___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

Re: [SR-Users] uaCSTA support in openser

2011-02-09 Thread Klaus Darilion 


Am 09.02.2011 12:03, schrieb target123:
> 
> Hi, did you ever come up with a solution here ? I also really need to filter
> the SIP INFO (csta) messages between an Avaya AES server and the OCS FE to
> solve some problems. A decent SIP proxy that can filter this out would be a
> pretty neat solution

Filtering certain types of messages is quite easy:

if (is_method("INFO")) {
  # note: adjust to the proper content type used by AES/OCS
  if ($cT == "application/csta+xml") {
sl_send_reply("415","Unsupported Media Type");
exit;
  }
}


But there is no functionality to process csta content in Kamailio. If
there are other tools/libraries to process csta it should be quite easy
to process it using perl/lua/exec/python interface.

regards
Klaus

> 
> cheers
> 
> Mark
> 
> 
> 
> Hugo Koblmueller wrote:
>>
>> high.all!
>>  
>> i'm wondering if there is any support of uaCSTA in openser (planned)? 
>>  
>> i'm just working on the integration of asterisk (*) environment to OCS
>> 2007
>> environment, having openSER in the middle (mainly for TCP/UDP translation
>> and smoothing out the protocol deficienes on both sides). in this setup
>> the
>> * having the openSER in front is talking to the OCS (and vice versa) via
>> the
>> OCS mediation server, which is moreorless sending standard SIP messages,
>> which enables normal softphone (integration to *) of the office
>> communicator. this configuration is already working... 
>>  
>> now i'm planning to go for the CTI integration, where there is no OCS
>> mediation server in between OCS and openSER, doing the translation of
>> SIP/CSTA to SIP. i'm thinking about using openSER for this task, that's
>> why
>> i'm looking for a CSTA module or perl programm, which is capable of this
>> functionality. 
>> afaik for the CTI communication there isn't the full complexity of CSTA
>> needed, just a subset mainly for call setup and call clearing.
>>  
>> anyone having experience on this topic?
>>  
>> thx & cheers
>> -hugo
>>  
>>  
>>  
>>
>>
>>
>>   Great Ideas for Small Devices  
>>  
>>
>> Hugo Koblmueller
>> Senior Staff Engineer Software Development   COMNEON electronic
>> technology GmbH & Co. OHG
>> Freistaedter Strasse 400
>> 4040 Linz
>> Austria  
>> hugo.koblmuel...@comneon.com 
>> tel: 
>> fax: 
>> mobile: 
>> Skype ID: +43 (5) 1777 - 15730
>> > 9+1777+%2D+15730&Email=h...@koblmueller.com> 
>> +43 (5) 1777 - 15810
>> +43 (676) 82051280
>> > %29+82051280&Email=h...@koblmueller.com> 
>> drhookson
>>  
>>
>> Want to always have my latest info?
>> > &invite=1&lang=en>Want a signature like
>> 
>> this?
>>  
>>
>>  
>>  
>> ___
>> Users mailing list
>> us...@lists.openser.org
>> http://lists.openser.org/cgi-bin/mailman/listinfo/users
>>
>>
> 

___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

Re: [SR-Users] module postgres SER

2011-02-09 Thread Daniel-Constantin Mierla 
please keep the mailing list cc-ed, sending private messages is not in 
the spirit of public mailing lists. Others may want to follow up the 
discussion now or later.


Thanks,
Daniel

On 2/9/11 5:01 PM, Bruno Bresciani wrote:

Daniel,

thanks for your reply, really ser-0.8.1.4 is too old but i need to 
solve this problem on that version. My great doubt is Why the aug_free 
function corrupt the url of database after some attemps to reconnect. 
Well, I'll try to understand this question...


Best Regards


2011/2/9 Daniel-Constantin Mierla >


Hello,


On 2/9/11 2:19 PM, Bruno Bresciani wrote:

Hi,

I've seen the problem in a postgres module (SER-0.8.1.4), if
the connection fails and module tries to reparse url it fails
as CON_SQLURL(_h) is corrupted by the function aug_free()
after some reconnect attempts . When the postgres database
back to work, some modules doesn't get reconnect because the
db_url is corrupted. Why this is happening? There are some
solution for this problem?

ser 0.8.1.4 is s old and I cannot fully remember, but I think
postgres module had no reconnect functionality at all by that time.

However, version 3.1.x of SER (as well as Kamailio flavour) has db
reconnect functionality for postgres. You can try it and report if
something is not working, it will be fixed in 3.x, but I think
nobody is still developing on 0.8.x to backport anything there.

Cheers,
Daniel

-- 
Daniel-Constantin Mierla

http://www.asipto.com




--
Daniel-Constantin Mierla
http://www.asipto.com

___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

Re: [SR-Users] uaCSTA support in openser

2011-02-09 Thread target123 

Hi, did you ever come up with a solution here ? I also really need to filter
the SIP INFO (csta) messages between an Avaya AES server and the OCS FE to
solve some problems. A decent SIP proxy that can filter this out would be a
pretty neat solution

cheers

Mark



Hugo Koblmueller wrote:
> 
> high.all!
>  
> i'm wondering if there is any support of uaCSTA in openser (planned)? 
>  
> i'm just working on the integration of asterisk (*) environment to OCS
> 2007
> environment, having openSER in the middle (mainly for TCP/UDP translation
> and smoothing out the protocol deficienes on both sides). in this setup
> the
> * having the openSER in front is talking to the OCS (and vice versa) via
> the
> OCS mediation server, which is moreorless sending standard SIP messages,
> which enables normal softphone (integration to *) of the office
> communicator. this configuration is already working... 
>  
> now i'm planning to go for the CTI integration, where there is no OCS
> mediation server in between OCS and openSER, doing the translation of
> SIP/CSTA to SIP. i'm thinking about using openSER for this task, that's
> why
> i'm looking for a CSTA module or perl programm, which is capable of this
> functionality. 
> afaik for the CTI communication there isn't the full complexity of CSTA
> needed, just a subset mainly for call setup and call clearing.
>  
> anyone having experience on this topic?
>  
> thx & cheers
> -hugo
>  
>  
>  
> 
> 
> 
>Great Ideas for Small Devices  
>   
> 
> Hugo Koblmueller
> Senior Staff Engineer Software DevelopmentCOMNEON electronic
> technology GmbH & Co. OHG
> Freistaedter Strasse 400
> 4040 Linz
> Austria   
> hugo.koblmuel...@comneon.com  
> tel: 
> fax: 
> mobile: 
> Skype ID:  +43 (5) 1777 - 15730
>  9+1777+%2D+15730&Email=h...@koblmueller.com> 
> +43 (5) 1777 - 15810
> +43 (676) 82051280
>  %29+82051280&Email=h...@koblmueller.com> 
> drhookson 
>   
> 
> Want to always have my latest info?
>  &invite=1&lang=en> Want a signature like
> 
> this? 
>  
> 
>  
>  
> ___
> Users mailing list
> us...@lists.openser.org
> http://lists.openser.org/cgi-bin/mailman/listinfo/users
> 
> 

-- 
View this message in context: 
http://old.nabble.com/uaCSTA-support-in-openser-tp16761786p30881351.html
Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.


___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

[SR-Users] [Serusers] problem with SER not proxying messages

2011-02-09 Thread Asuncion Merino Rodríguez 
 

Hola, no he visto respuesta a este foro, por favor alguien podría dar alguna
respuesta?

 

___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

Re: [SR-Users] LCR module : same IP address for different prefix.

2011-02-09 Thread Antanas Masevicius 
Juha,

Thank you for your work! This feature is highly needed i suspect.

Antanas
NTT


On 2011.02.09 06:25, Juha Heinanen wrote:
> Juha Heinanen writes:
>
>> Antanas Masevicius writes:
>>
>>> I already wrote on this list about this issue. Currently the only way to
>>> do it is to disable duplicate gws checking in sources itself.
>>> In general, gw duplication checking should be extended to check not only
>>> by ip_addr or its hostname, but to include 'tag' and even 'strip' columns.
>>> IMHO duplication checking could be removed at all and only DB unique
>>> constraint checking used.
>> i'll need to check for possible problems and if none are found, i'll
>> remove the duplicate check.
> i just committed to master branch changes to lcr module that allow
> gateways of an lcr instance to be "non-unique".
>
> -- juha
>
> ___
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users@lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>

-- 
Antanas Masevičius
Technikos direktorius
UAB "Nacionalinis telekomunikaciju tinklas"
Tel.  +370 5 2056000
Tel.  +370 700 00031 (tiesioginis)
Fax.  +370 700 00034
el.p: antanas.masevic...@ntt.lt
www.ntt.lt
www.spykas.com


___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

Re: [SR-Users] problem with bye using rtpproxy

2011-02-09 Thread Daniel-Constantin Mierla 

Hello,

On 2/7/11 8:12 PM, Amit Nepal wrote:
I have been trying to figure this out While using kamailio and 
rtpproxy, the caller is not receiving the bye when callee hangs up but 
audio is two way and everything seems to be working fine, any one had 
this issue ?



are you doing record-routing in your config?

The best for providing further hints is to get the SIP trace for such 
call, from the starting INVITE to the end -- ngrep is recommended to use 
for sending on this list since it prints out text, following command can 
be used on your sip server:


ngrep -d any -qt -W byline port 5060

Cheers,
Daniel

--
Daniel-Constantin Mierla
http://www.asipto.com


___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

Re: [SR-Users] Multiple call accounting from an IP

2011-02-09 Thread Daniel-Constantin Mierla 

Hello,

On 2/7/11 8:13 PM, Amit Nepal wrote:

Hi everyone,
  I am sure someone have been working with this scenario, how 
about accounting multiple calls from same account or ip address while 
using ip auth ?


I don't understand what you want exactly to achieve. Kamailio doesn't 
set any limitation on number of active calls from same user/ip -- but 
you can implement such limitations with dialog module or custom config 
logic using htable or a database table with sqlops.


If you look for something else, provide more details.

Cheers,
Daniel

--
Daniel-Constantin Mierla
http://www.asipto.com


___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

Re: [SR-Users] fosdem 2011 presentation about p_usrloc

2011-02-09 Thread Daniel-Constantin Mierla 



On 2/9/11 10:42 AM, Henning Westerholt wrote:

On Tuesday 08 February 2011, Klaus Darilion wrote:

this year the FOSDEM developer conference was a again a really nice
event, the first time with an own room dedicated completely to open
source telephony solutions! If you're interested in our presentation
about the new p_usrloc module and how to scale location services with
Kamailio, you can find it at
the usual place on our webserver:

Are videos available of the presentation?

Hi Klaus,

afaik not, sorry. There were only limited coverage of the development rooms
and i also did not noticed a camera during the talk.

Only the Lighting Talks and main tracks were officially recorded. None 
of the dev rooms were recorded unless someone in particular did it. In 
the VoIP dev room was no recording. Anyhow, live is better always :-).


Cheers,
Daniel

--
Daniel-Constantin Mierla
http://www.asipto.com


___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

Re: [SR-Users] module postgres SER

2011-02-09 Thread Daniel-Constantin Mierla 

Hello,

On 2/9/11 2:19 PM, Bruno Bresciani wrote:

Hi,

I've seen the problem in a postgres module (SER-0.8.1.4), if the 
connection fails and module tries to reparse url it fails as 
CON_SQLURL(_h) is corrupted by the function aug_free() after some 
reconnect attempts . When the postgres database back to work, some 
modules doesn't get reconnect because the db_url is corrupted. Why 
this is happening? There are some solution for this problem?
ser 0.8.1.4 is s old and I cannot fully remember, but I think 
postgres module had no reconnect functionality at all by that time.


However, version 3.1.x of SER (as well as Kamailio flavour) has db 
reconnect functionality for postgres. You can try it and report if 
something is not working, it will be fixed in 3.x, but I think nobody is 
still developing on 0.8.x to backport anything there.


Cheers,
Daniel

--
Daniel-Constantin Mierla
http://www.asipto.com


___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

Re: [SR-Users] Kamailio NOTIFY

2011-02-09 Thread Klaus Darilion 


Am 09.02.2011 14:56, schrieb Spinov Evgeniy:
> On Wed, 2011-02-09 at 14:47 +0100, Klaus Darilion wrote:
>> Do you see PUBLISH requests with Event: dialog?
>>
>> Note: those will be sent by pua_dialoginfo module to Kamailio itslef,
>> thus will be sent on the loopback interface. You should see them with
>> "ngrep -d any port 5060"
>>
>> klaus
> 
> No, PUBLISH requests with event dialog are not generated. Instead of
> dialog event, presence event is generated, but as for me, it should
> work, as for instance eyeBeam is generating PUBLISH requests itself,
> with event: presence.
> 
> You're saying that NOTIFYs with event dialog are empty, cause there are
> not PUBLISH event: dialog?

Yes. The "presence server" is just a relay of the content:



PUBLISH ---> presence server <---SUBSCRIBE/NOTIFY---> client

If dialoginfo is not published into the presence server, the presence
server will send empty NOTIFYs.

There are 2 ways to get the dialoginfo published into the presence
server: either the phone does it (e.g. SNOM phones can do that, eyebeam
only publishes "presence" but not "dialog"info) or Kamailio can PUBLISH
the dialog state. See
http://www.kamailio.org/docs/modules/3.1.x/modules_k/presence_dialoginfo.html

So, make sure that Kamailio will publish the dialoginfo:
http://www.kamailio.org/docs/modules/3.1.x/modules_k/pua_dialoginfo.html#id2887055

regards
klaus

___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

[SR-Users] Fosdem2011 presentation: SIP Web20 Lua

2011-02-09 Thread Daniel-Constantin Mierla 

Hello,

the presentation I did during Fosdem in Brussels last weekend is 
available at:

http://www.kamailio.org/events/2011-fosdem/dcm-sip-web-lua.pdf

The focus was to show how to interact with Kamailio via HTTP and how 
Kamailio can interact with Web services via HTTP, using Lua to make it 
easier. There are slides for a demo config of sending asynchronous 
notifications to Twitter on missed calls (using modules app_lua, mqueue, 
sqlops and rtimer).


Cheers,
Daniel

--
Daniel-Constantin Mierla
http://www.asipto.com


___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

Re: [SR-Users] Kamailio NOTIFY

2011-02-09 Thread Spinov Evgeniy 
On Wed, 2011-02-09 at 14:47 +0100, Klaus Darilion wrote:
> Do you see PUBLISH requests with Event: dialog?
> 
> Note: those will be sent by pua_dialoginfo module to Kamailio itslef,
> thus will be sent on the loopback interface. You should see them with
> "ngrep -d any port 5060"
> 
> klaus

No, PUBLISH requests with event dialog are not generated. Instead of
dialog event, presence event is generated, but as for me, it should
work, as for instance eyeBeam is generating PUBLISH requests itself,
with event: presence.

You're saying that NOTIFYs with event dialog are empty, cause there are
not PUBLISH event: dialog?


> 
> Am 09.02.2011 14:16, schrieb Spinov Evgeniy:
> > Hello all.
> > 
> > I'm having problem with sending NOTIFY packet for event dialog.
> > 
> > The problem is, that all NOTIFY packets sent with this event is empty.
> > ( Content-length: 0 ) and in fact, not passing actual status to the
> > phone. In my case this is SPA962.
> > 
> > NOTIFY packets for event presence are sent fine and status is changing
> > normally.
> > 
> > The same phone, works fine with Asterisk. Asterisk is sending NOTIFY
> > with non-zero content and changing status for the subscriber as well.
> > 
> > I've turned on pua module to send PUBLISH requests at Kamailio, it's
> > filling DB table with subscription, everything runs fine and NOTIFY
> > packets are coming to subscribed phone, but they are empty.
> > 
> > I also have presence_dialoginfo module on and pua_dialoginfo module on.
> > I've tried on Kamailio 3.0.3 and 3.1.1. Content always empty.
> > 
> > There is no sense of pasting here packet dumps, cause everything is like
> > should be: 
> > 
> > (UAC) SUBSCRIBE, event: presence ---> K
> > K ---> 202 OK
> > (SPA) SUBSCRIBE, event: dialog ---> K
> > K ---> 202 OK
> > PUBLISH, event: presence ---> K
> > K ---> 200 OK
> > NOTIFY, event: presence, Content-Length: XXX ---> UAC
> > NOTIFY, event: dialog, Content-Length: 0 ---> SPA962
> > 
> > As result, softphone changes status and hardphone ( SPA962 ) - no. May
> > be it's cause of PUBLISH event value? 
> > 
> > Please advise.
> > 
> > 
> > ___
> > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> > sr-users@lists.sip-router.org
> > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users



___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

Re: [SR-Users] Kamailio NOTIFY

2011-02-09 Thread Klaus Darilion 
Do you see PUBLISH requests with Event: dialog?

Note: those will be sent by pua_dialoginfo module to Kamailio itslef,
thus will be sent on the loopback interface. You should see them with
"ngrep -d any port 5060"

klaus

Am 09.02.2011 14:16, schrieb Spinov Evgeniy:
> Hello all.
> 
> I'm having problem with sending NOTIFY packet for event dialog.
> 
> The problem is, that all NOTIFY packets sent with this event is empty.
> ( Content-length: 0 ) and in fact, not passing actual status to the
> phone. In my case this is SPA962.
> 
> NOTIFY packets for event presence are sent fine and status is changing
> normally.
> 
> The same phone, works fine with Asterisk. Asterisk is sending NOTIFY
> with non-zero content and changing status for the subscriber as well.
> 
> I've turned on pua module to send PUBLISH requests at Kamailio, it's
> filling DB table with subscription, everything runs fine and NOTIFY
> packets are coming to subscribed phone, but they are empty.
> 
> I also have presence_dialoginfo module on and pua_dialoginfo module on.
> I've tried on Kamailio 3.0.3 and 3.1.1. Content always empty.
> 
> There is no sense of pasting here packet dumps, cause everything is like
> should be: 
> 
> (UAC) SUBSCRIBE, event: presence ---> K
> K ---> 202 OK
> (SPA) SUBSCRIBE, event: dialog ---> K
> K ---> 202 OK
> PUBLISH, event: presence ---> K
> K ---> 200 OK
> NOTIFY, event: presence, Content-Length: XXX ---> UAC
> NOTIFY, event: dialog, Content-Length: 0 ---> SPA962
> 
> As result, softphone changes status and hardphone ( SPA962 ) - no. May
> be it's cause of PUBLISH event value? 
> 
> Please advise.
> 
> 
> ___
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users@lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

[SR-Users] Kamailio NOTIFY

2011-02-09 Thread Spinov Evgeniy 
Hello all.

I'm having problem with sending NOTIFY packet for event dialog.

The problem is, that all NOTIFY packets sent with this event is empty.
( Content-length: 0 ) and in fact, not passing actual status to the
phone. In my case this is SPA962.

NOTIFY packets for event presence are sent fine and status is changing
normally.

The same phone, works fine with Asterisk. Asterisk is sending NOTIFY
with non-zero content and changing status for the subscriber as well.

I've turned on pua module to send PUBLISH requests at Kamailio, it's
filling DB table with subscription, everything runs fine and NOTIFY
packets are coming to subscribed phone, but they are empty.

I also have presence_dialoginfo module on and pua_dialoginfo module on.
I've tried on Kamailio 3.0.3 and 3.1.1. Content always empty.

There is no sense of pasting here packet dumps, cause everything is like
should be: 

(UAC) SUBSCRIBE, event: presence ---> K
K ---> 202 OK
(SPA) SUBSCRIBE, event: dialog ---> K
K ---> 202 OK
PUBLISH, event: presence ---> K
K ---> 200 OK
NOTIFY, event: presence, Content-Length: XXX ---> UAC
NOTIFY, event: dialog, Content-Length: 0 ---> SPA962

As result, softphone changes status and hardphone ( SPA962 ) - no. May
be it's cause of PUBLISH event value? 

Please advise.


___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

[SR-Users] module postgres SER

2011-02-09 Thread Bruno Bresciani 
Hi,

I've seen the problem in a postgres module (SER-0.8.1.4), if the connection
fails and module tries to reparse url it fails as CON_SQLURL(_h) is
corrupted by the function aug_free() after some reconnect attempts . When
the postgres database back to work, some modules doesn't get reconnect
because the db_url is corrupted. Why this is happening? There are some
solution for this problem?

Best Regard
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

Re: [SR-Users] fosdem 2011 presentation about p_usrloc

2011-02-09 Thread Henning Westerholt 
On Tuesday 08 February 2011, Klaus Darilion wrote:
> > this year the FOSDEM developer conference was a again a really nice
> > event, the first time with an own room dedicated completely to open
> > source telephony solutions! If you're interested in our presentation
> > about the new p_usrloc module and how to scale location services with
> > Kamailio, you can find it at
> 
> > the usual place on our webserver:
> Are videos available of the presentation?

Hi Klaus,

afaik not, sorry. There were only limited coverage of the development rooms 
and i also did not noticed a camera during the talk.

Cheers,

Henning

___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users