Re: [OpenSIPS-Users] [dialog] Can't fetch data in 200 reply route of BYE

[liuf]

Thanks a lot.

You're very professional.


Best Regards,
Liu Fan
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[OpenSIPS-Users] No voice in mediaproxy 2.3.8

[Jiang Jinke]

Dear All,

I'm having the following error with mediaproxyx 2.3.8, and all the call
will have no voice after running mediaproxy for a few days in a high
loading (40 calls) server.

I check the new mediaproxy with the same line in send, there is no
such getHost call.
Is this a fixed issue in the latest version ?
I haven't tried the new mediaproxy yet because I need to do some tests
to make sure it
will running fine in the production environment before upgrade.

Exception caught:

Mar 12 12:22:54 hkcn-p2 media-relay[3373]: ---  ---
Mar 12 12:22:54 hkcn-p2 media-relay[3373]:   File
"/usr/lib/python2.4/site-packages/Twisted-8.2.0-py2.4-linux-x86_64.egg/twisted/internet/udp.py",
line 126, in doRead
Mar 12 12:22:54 hkcn-p2 media-relay[3373]:
self.protocol.datagramReceived(data, addr)
Mar 12 12:22:54 hkcn-p2 media-relay[3373]:   File
"/usr/lib64/python2.4/site-packages/mediaproxy/mediacontrol.py", line
127, in datagramReceived
Mar 12 12:22:54 hkcn-p2 media-relay[3373]: self.cb_func(host, port, data)
Mar 12 12:22:54 hkcn-p2 media-relay[3373]:   File
"/usr/lib64/python2.4/site-packages/mediaproxy/mediacontrol.py", line
184, in got_data
Mar 12 12:22:54 hkcn-p2 media-relay[3373]:
self.substream.send_data(self, data)
Mar 12 12:22:54 hkcn-p2 media-relay[3373]:   File
"/usr/lib64/python2.4/site-packages/mediaproxy/mediacontrol.py", line
264, in send_data
Mar 12 12:22:54 hkcn-p2 media-relay[3373]: dest.listener.protocol.send(data)
Mar 12 12:22:54 hkcn-p2 media-relay[3373]:   File
"/usr/lib64/python2.4/site-packages/mediaproxy/mediacontrol.py", line
136, in send
Mar 12 12:22:54 hkcn-p2 media-relay[3373]: host = self.transport.getHost()
Mar 12 12:22:54 hkcn-p2 media-relay[3373]: exceptions.AttributeError:
'NoneType' object has no attribute 'getHost'
Mar 12 12:22:54 hkcn-p2 media-relay[3373]: Traceback (most recent call last):
Mar 12 12:22:54 hkcn-p2 media-relay[3373]:   File
"/usr/lib/python2.4/site-packages/Twisted-8.2.0-py2.4-linux-x86_64.egg/twisted/python/log.py",
line 69, in callWithContext
Mar 12 12:22:54 hkcn-p2 media-relay[3373]: return
context.call({ILogContext: newCtx}, func, *args, **kw)
Mar 12 12:22:54 hkcn-p2 media-relay[3373]:   File
"/usr/lib/python2.4/site-packages/Twisted-8.2.0-py2.4-linux-x86_64.egg/twisted/python/context.py",
line 59, in callWithContext
Mar 12 12:22:54 hkcn-p2 media-relay[3373]: return
self.currentContext().callWithContext(ctx, func, *args, **kw)
Mar 12 12:22:54 hkcn-p2 media-relay[3373]:   File
"/usr/lib/python2.4/site-packages/Twisted-8.2.0-py2.4-linux-x86_64.egg/twisted/python/context.py",
line 37, in callWithContext
Mar 12 12:22:54 hkcn-p2 media-relay[3373]: return func(*args,**kw)
Mar 12 12:22:54 hkcn-p2 media-relay[3373]:   File
"/usr/lib/python2.4/site-packages/Twisted-8.2.0-py2.4-linux-x86_64.egg/twisted/internet/epollreactor.py",
line 231, in _doReadOrWrite
Mar 12 12:22:54 hkcn-p2 media-relay[3373]: why = selectable.doRead()

Regards,
Jinke Jiang

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[OpenSIPS-Users] Problem with /tmp/opensips_fifo

[erick]

>
>
> I have with the following problem:
>
> The file /tmp/opensips_fifo for any reason disappears
> for example if  send the command for reboot the server.
> The  opensips using mi_fifo module.
>
> With the command "opensips -c" said is ok
>
> opensipsctl moni said (if not find the file)
> ERROR: Error opening OpenSIPS's FIFO /tmp/opensips_fifo
> ERROR: Make sure you have the line 'modparam("mi_fifo", "fifo_name",
> "/tmp/opensips_fifo")' in your config
> ERROR: and also have loaded the mi_fifo module.
>
> But for fix this problem I made the following
> touch /tmp/opensips_fifo
> chmod 755 /tmp/opensips_fifo
>
> this is temporary but it works.
>
> Another things is with the Control Panel 4.0
>
> I have problems with the following modules and this is the message of each
>
> -Dialog = Array ( [0] => Cannot connect to OpenSIPS Server via Management
> Interface (/tmp/opensips_fifo) )
>
> - Dispatcher = Array ( [0] => Cannot connect to OpenSIPS Server via
> Management Interface (/tmp/opensips_fifo) [1] => Cannot connect to
> OpenSIPS Server via Management Interface (/tmp/opensips_fifo) )
>
> - MI Commands = /tmp/opensips_fifo
>
> - Monit = Process - Opensips / Status - Execution failed
>
> - SIP Trace = Array ( [0] => Cannot connect to OpenSIPS Server via
> Management Interface (/tmp/opensips_fifo) )
>
> - Statistics Monitor = Cannot connect to OpenSIPS Server via Management
> Interface (/tmp/opensips_fifo)
>
> The common problem is " /tmp/opensips_fifo"
>
> Thanks
> Erick
>



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[OpenSIPS-Users] 1.6.2-notls perl module not compiling on Centos

[erik pepermans]

Hi,
 
When compiling the perl module on version 1.6.2-notls on Centos :
 
make prefix=/ all
 
I receive :
 
...
make[1]: Entering directory `/usr/src/opensips-1.6.2-notls/modules/perl'
gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -Wall
-minline-all-stringops -falign-loops -ftree-vectorize -mtune=nocona
-Wold-style-definition -Wmissing-field-initializers -Wredundant-decls
-DNAME='"opensips"' -DVERSION='"1.6.2-notls"' -DARCH='"x86_64"'
-DOS='"linux"' -DCOMPILER='"gcc 4.1.2"' -D__CPU_x86_64 -D__OS_linux
-D__SMP_yes -DCFG_DIR='"//etc/opensips/"' -DPKG_MALLOC -DSHM_MEM  -DSHM_MMAP
-DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES
-DSTATISTICS -DCHANGEABLE_DEBUG_LEVEL -DF_MALLOC  -DFAST_LOCK
-DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024  -DHAVE_GETHOSTBYNAME2
-DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL
-DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -DHAVE_TIMEGM -DHAVE_EPOLL
-DHAVE_SIGIO_RT -DHAVE_SELECT -Wno-unused -Wno-redundant-decls  -D_REENTRANT
-D_GNU_SOURCE -fno-strict-aliasing -pipe -Wdeclaration-after-statement
-I/usr/local/include -D_LARGEFILE_SOURCE -D_FILE_OFFSET_BITS=64
-I/usr/include/gdbm  -I/usr/lib64/perl5/5.8.8/x86_64-linux-thread-multi/CORE
-c -o opensipsxs.o opensipsxs.c
opensipsxs.c: In function âboot_OpenSIPSâ:
opensipsxs.c:2062: error: âPL_unitcheckavâ undeclared (first use in this
function)
opensipsxs.c:2062: error: (Each undeclared identifier is reported only once
opensipsxs.c:2062: error: for each function it appears in.)
make[1]: Leaving directory `/usr/src/opensips-1.6.2-notls/modules/perl'
make: *** [modules] Error 2

Brgds
Erik
 

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[OpenSIPS-Users] Virtual DB failover wrangling

[Jock McKechnie]

Greetings folks;

I've been attempting to get a working db_virtual failover setup going and am
having some very basic functionality problems while in testing, I was
wondering if someone could shed some light one what I, presumably, am doing
wrong.

Appropriate sections of config look something like this:

modparam("alias_db", "db_url", "virtual://set1")

modparam("db_virtual", "db_max_consec_retrys", 3)

modparam("db_virtual","db_urls","define set1 FAILOVER")
modparam("db_virtual","db_urls","mysql://
opensips:passw...@192.168.0.100/aliasdb")
modparam("db_virtual","db_urls","mysql://
opensips:passw...@192.168.0.200/aliasdb")

if (alias_db_lookup("opensips_inbound", "d")) {
# Do something
};


I'm running the Debian package release, 1.6.0-1, from the leurent.eurepository.

If I start OpenSIPs and then on the .100 DB do an iptables REJECT to
simulate a DB problem OpenSIPs will simply retry the alias lookup
infinitely.

Curiously, if I set up the iptables block, THEN start OpenSIPs, and toggle
the block on and off, the failure mode appears to work as it should, which
really confuses me.

Have I got an issue with how I expect this to work? How I've got it set up?
Or do we think my testing methodology is bunk?

My many thanks;

 - Jock
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Re: [OpenSIPS-Users] Problem with OpenSIPS Control Panel 4.0

[erick]

Thanks for your help, It's working, I will continue  testing.

Regards
Erick Ch.

> Hi,
>
> Had the same problem, change $config->admin_passwd_mode=1; to
> $config->admin_passwd_mode=0;
> in opensips-cp/config/globals.php and it worked. I dont know is this is
> the
> right way.
>
> regards
>
>
> On Thu, Mar 11, 2010 at 2:29 AM,  wrote:
>
>> > I have the following problem
>> >
>> > With install OpenSIPS  1.6 and  Control Panel 4.0.
>> >
>> > When try the access the Control Panel didn't accept the user o
>> password
>> > (Invalid Username / Password)
>> >
>> > With the  commands opensips-c, monit-t said  that the configuration
>> files
>> > are fine.
>> >
>> >  I can register phone.
>> >
>> > The question is how do I know the password to the web interface, try
>> to
>> > file
>> > db.inc.php, boxes.global.inc.php, or  if I have to change something in
>> the
>> > database
>> >
>> > The reference for the installation of the control panel are Book
>> OpenSIPS
>> > 1.6 and the link  http://opensips-cp.sourceforge.net/ , but both have
>> > differences
>> >
>> > This is my first time with OpenSIPS
>> >
>> > I appreciate the help
>> >
>> > Thanks
>> >
>> > Erick Ch.
>> >
>>
>>
>>
>> ___
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>>
>
>
>
> --
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[OpenSIPS-Users] [NEW] OpenSIPS 1.6.2 is released

[Bogdan-Andrei Iancu]

Hi all,

OpenSIPS 1.6.2 is the first release following the new release policy 
after the 2.0 work started (see 
http://www.opensips.org/Development/Development).

1.6.2 release brings both new features / enhancement and a lot of fixes. 
The listing with all additions and fixes is available under:
 http://www.opensips.org/Main/Ver162.

OpenSIPS 1.6.2 is now ready for download on project web site and SF 
download system.

The full Changelog is available under:
http://opensips.org/pub/opensips/1.6.2/src/ChangeLog

To get the OpenSIPS 1.6.2 version, see the downloads page.
http://www.opensips.org/Resources/Downloads


Regards,
Bogdan

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Re: [OpenSIPS-Users] lb_reload

[Bogdan-Andrei Iancu]

Hi Dmitri,

found the bug and fixed it :). If you update from SVN, it should work as 
a charm

Thanks for your report,
Bogdan

Dmitri G. wrote:
> Hi Bogdan,
>
> I'm running OpenSIPS 1.6.1
>
> version: opensips 1.6.1-notls (i386/linux)
> flags: STATS: Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, 
> SHM_MEM, 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
> poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
> svnrevision: 2:6599M
> @(#) $Id: main.c 6169 2009-09-22 12:48:37Z bogdan_iancu $
> main.c compiled on 01:18:50 Feb 15 2010 with gcc 4.3.4
>
> BTW, what I'm doing frequently is to resize the existing resources, 
> and adding/removing new asterisk gws in the load_balancer table.
>
> Thanks,
> Dimitri
>
>
>
> 2010/3/10 Bogdan-Andrei Iancu către OpenSIPS arată detaliile 12:52 
> (acum 5 ore)
>
> Hi Dimitri,
>
> Ok, so you do lb_reload just because you need to add some more
> destinations in the balancer. You do not change the properties of the
> already existing destination - this was what I was interested in.
>
> Let me run some checks to see what about. BTW, what opensips
> version are
> you using ?
>
> Regards,
> Bogdan
>
>
> Dmitri G. wrote:
> > I'm changing the resource frequently, btw these are asterisk boxes
> > with agents, an inbound call center.
> > Every morning they call me and tell "dimitri today we have 200 agent
> > for xx.xx.xx.xx, and we have 150 for yy.yy.yy.yy".
> > So I run a lb_resize to change the resource figure (the number of
> > available agents).
> > There is no problem with this and I never change the name of the
> > resource, just the figure. I can resize it without any problems.
> >
> > But in case if they call me to add a new asterisk ip with resources,
> > then I need to insert it to a DB, to the load_balancer table.
> > But how to enable to use the new gw other than using lb_reload?
> >
> > If I run the lb_reload, and there was concurrent calls to the
> asterisk
> > boxes (so agent=200, and agent=150), the load to each
> destination will
> > be zero (load=0). Buit there are still active calls to the
> > destinations so if I have 150 calls to one, and if there will be
> a new
> > call to the destination, after a lb_reload I see load=1, where in
> > reality there was 151 calls.
> >
> > See my problem?
> >
> > Thanks,
> >
> > Dimitri.
> >
> > 2010/3/9 Bogdan-Andrei Iancu către OpenSIPS arată detaliile 16:26
> > (acum 27 minute)
> >
> > Ok, I got that :)
> >
> > my question was - what you change in DB (before doing the
> reload)? do
> > you change the definition of the already existing destinations (like
> > resources) ?
> >
> > Regards,
> > Bogdan
> >
> > Dmitri G. wrote:
> > > Hi Bogdan,
> > >
> > > I mean it's setting the load= figures to zero for all gws in the
> > > load_balancer table.
> > > So if I have 2 gws with load=136 and load=227, after running
> > > opensipsctl fifo lb_reload, then load will be zero (load=0).
> > >
> > > Dimitri
> > >
> > > 2010/3/9 Bogdan-Andrei Iancu către OpenSIPS arată detaliile 15:57
> > > (acum 18 minute)
> > >
> > > Hi Dimitri,
> > >
> > > reloading the data should not delete the dialog profiles,
> unless you
> > > change the name of the resources (in the destination definition).
> > > Is this your case?
> > >
> > > Regards,
> > > Bogdan
> > >
> > > Dmitri G. wrote:
> > > > Hi,
> > > >
> > > > Is there a way to keep the number of the dialogs after
> running the
> > > > lb_reload command through openipsctl?
> > > > When I add a new gateway to the mysql able and execute lb_reload
> > > > command, the number of the dialogs are resets to zero in every
> > > > gateways (I have checked it with lb_list, load counters are
> zero)
> > > >
> > > > Thanks,
> > > > Dimitri
> > > >
> > >
>


-- 
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www.voice-system.ro


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Re: [OpenSIPS-Users] [Presence] How to receive presence NOTIFY over TCP ?

[Pascal Maugeri]

Ok, good comment. Yes indeed we have a SIP proxy between the subscriber and
opensips (which is only acting as presence server in this scenario).
I thought there were two problems (one at proxy and one at presence server
side). We will work on proxy side now to solve this issue.

Many thanks for your time.
Pascal


On Thu, Mar 11, 2010 at 4:22 PM, Anca Vamanu  wrote:

> Hi Pascal,
>
> Only now I observed that there is a Route header in the Notify - so the
> first hop is in fact the address in the route and not the RURI. So it is
> correct to send the Notify on udp, and the proxy at the next hop should
> switch to tcp.
>
> Regards,
>
> --
> Anca Vamanu
> www.voice-system.ro
>
>
>
> Pascal Maugeri wrote:
> > Anca
> >
> > I got the following logs when subscribing:
> >
> > Mar 11 15:28:40 test /usr/local/sbin/opensips[3258]:
> > DBG:presence:build_dlg_t: CONTACT =
> > sip:watc...@79.149.10.246:5060;transport=tcp
> > Mar 11 15:28:40 test /usr/local/sbin/opensips[3258]: DBG:tm:t_uac:
> > next_hop=
> > Mar 11 15:28:40 test /usr/local/sbin/opensips[3258]:
> > DBG:core:mk_proxy: doing DNS lookup...
> > Mar 11 15:28:40 test /usr/local/sbin/opensips[3258]:
> > DBG:tm:dlg2hash: 42964
> > Mar 11 15:28:40 test /usr/local/sbin/opensips[3258]:
> > DBG:tm:print_request_uri: sip:watc...@79.149.10.246:5060
> ;transport=tcp
> > Mar 11 15:28:40 test /usr/local/sbin/opensips[3258]:
> > DBG:tm:set_timer: relative timeout is 50
> > Mar 11 15:28:40 test /usr/local/sbin/opensips[3258]:
> > DBG:tm:insert_timer_unsafe: [4]: 0x7f8fa84c0068 (450850)
> > Mar 11 15:28:40 test /usr/local/sbin/opensips[3258]:
> > DBG:tm:set_timer: relative timeout is 30
> > Mar 11 15:28:40 test /usr/local/sbin/opensips[3258]:
> > DBG:tm:insert_timer_unsafe: [0]: 0x7f8fa84c0098 (4538)
> > Mar 11 15:28:40 test /usr/local/sbin/opensips[3258]:
> > INFO:presence:send_notify_request: NOTIFY
> > sip:watc...@test.genaker.net 
> > 
> > 
> via
> > sip:213.151.105.170:;lr=on on behalf of
> > sip:present...@test.genaker.net 
> > 
> > 
> for event presence
> > Mar 11 15:28:40 test /usr/local/sbin/opensips[3258]:
> > DBG:tm:t_unref: UNREF_UNSAFE: after is 0
> > Mar 11 15:28:40 test /usr/local/sbin/opensips[3258]:
> > DBG:core:destroy_avp_list: destroying list (nil)
> > Mar 11 15:28:40 test /usr/local/sbin/opensips[3258]:
> > DBG:core:receive_msg: cleaning up
> > Mar 11 15:28:40 test /usr/local/sbin/opensips[3259]:
> > DBG:core:parse_msg: SIP Reply  (status):
> > Mar 11 15:28:40 test /usr/local/sbin/opensips[3259]:
> > DBG:core:parse_msg:  version: 
> > Mar 11 15:28:40 test /usr/local/sbin/opensips[3259]:
> > DBG:core:parse_msg:  status:  <200>
> > Mar 11 15:28:40 test /usr/local/sbin/opensips[3259]:
> > DBG:core:parse_msg:  reason:  
> >
> >
> > What log could I add to your patch so we can get more details ?
> >
> > I recompiled the whole opensips and its modules just to make sure your
> > patch is activated.
> >
> > Cheers
> > Pascal
> >
>
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Re: [OpenSIPS-Users] [Presence] How to receive presence NOTIFY over TCP ?

[Anca Vamanu]

Hi Pascal,

Only now I observed that there is a Route header in the Notify - so the 
first hop is in fact the address in the route and not the RURI. So it is 
correct to send the Notify on udp, and the proxy at the next hop should 
switch to tcp.

Regards,

-- 
Anca Vamanu
www.voice-system.ro



Pascal Maugeri wrote:
> Anca
>
> I got the following logs when subscribing:
>
> Mar 11 15:28:40 test /usr/local/sbin/opensips[3258]:
> DBG:presence:build_dlg_t: CONTACT =
> sip:watc...@79.149.10.246:5060;transport=tcp
> Mar 11 15:28:40 test /usr/local/sbin/opensips[3258]: DBG:tm:t_uac:
> next_hop=
> Mar 11 15:28:40 test /usr/local/sbin/opensips[3258]:
> DBG:core:mk_proxy: doing DNS lookup...
> Mar 11 15:28:40 test /usr/local/sbin/opensips[3258]:
> DBG:tm:dlg2hash: 42964
> Mar 11 15:28:40 test /usr/local/sbin/opensips[3258]:
> DBG:tm:print_request_uri: sip:watc...@79.149.10.246:5060;transport=tcp
> Mar 11 15:28:40 test /usr/local/sbin/opensips[3258]:
> DBG:tm:set_timer: relative timeout is 50
> Mar 11 15:28:40 test /usr/local/sbin/opensips[3258]:
> DBG:tm:insert_timer_unsafe: [4]: 0x7f8fa84c0068 (450850)
> Mar 11 15:28:40 test /usr/local/sbin/opensips[3258]:
> DBG:tm:set_timer: relative timeout is 30
> Mar 11 15:28:40 test /usr/local/sbin/opensips[3258]:
> DBG:tm:insert_timer_unsafe: [0]: 0x7f8fa84c0098 (4538)
> Mar 11 15:28:40 test /usr/local/sbin/opensips[3258]:
> INFO:presence:send_notify_request: NOTIFY
> sip:watc...@test.genaker.net
>  via
> sip:213.151.105.170:;lr=on on behalf of
> sip:present...@test.genaker.net
>  for event presence
> Mar 11 15:28:40 test /usr/local/sbin/opensips[3258]:
> DBG:tm:t_unref: UNREF_UNSAFE: after is 0
> Mar 11 15:28:40 test /usr/local/sbin/opensips[3258]:
> DBG:core:destroy_avp_list: destroying list (nil)
> Mar 11 15:28:40 test /usr/local/sbin/opensips[3258]:
> DBG:core:receive_msg: cleaning up
> Mar 11 15:28:40 test /usr/local/sbin/opensips[3259]:
> DBG:core:parse_msg: SIP Reply  (status):
> Mar 11 15:28:40 test /usr/local/sbin/opensips[3259]:
> DBG:core:parse_msg:  version: 
> Mar 11 15:28:40 test /usr/local/sbin/opensips[3259]:
> DBG:core:parse_msg:  status:  <200>
> Mar 11 15:28:40 test /usr/local/sbin/opensips[3259]:
> DBG:core:parse_msg:  reason:  
>
>
> What log could I add to your patch so we can get more details ?
>
> I recompiled the whole opensips and its modules just to make sure your 
> patch is activated.
>
> Cheers
> Pascal
>

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Re: [OpenSIPS-Users] [Presence] How to receive presence NOTIFY over TCP ?

[Pascal Maugeri]

Anca

I got the following logs when subscribing:

Mar 11 15:28:40 test /usr/local/sbin/opensips[3258]:
DBG:presence:build_dlg_t: CONTACT = sip:watc...@79.149.10.246:5060
;transport=tcp
Mar 11 15:28:40 test /usr/local/sbin/opensips[3258]: DBG:tm:t_uac:
next_hop=
Mar 11 15:28:40 test /usr/local/sbin/opensips[3258]: DBG:core:mk_proxy:
doing DNS lookup...
Mar 11 15:28:40 test /usr/local/sbin/opensips[3258]: DBG:tm:dlg2hash: 42964
Mar 11 15:28:40 test /usr/local/sbin/opensips[3258]:
DBG:tm:print_request_uri: sip:watc...@79.149.10.246:5060;transport=tcp
Mar 11 15:28:40 test /usr/local/sbin/opensips[3258]: DBG:tm:set_timer:
relative timeout is 50
Mar 11 15:28:40 test /usr/local/sbin/opensips[3258]:
DBG:tm:insert_timer_unsafe: [4]: 0x7f8fa84c0068 (450850)
Mar 11 15:28:40 test /usr/local/sbin/opensips[3258]: DBG:tm:set_timer:
relative timeout is 30
Mar 11 15:28:40 test /usr/local/sbin/opensips[3258]:
DBG:tm:insert_timer_unsafe: [0]: 0x7f8fa84c0098 (4538)
Mar 11 15:28:40 test /usr/local/sbin/opensips[3258]:
INFO:presence:send_notify_request: NOTIFY
sip:watc...@test.genaker.netvia
sip:213.151.105.170:;lr=on
on behalf of 
sip:present...@test.genaker.netfor
event presence
Mar 11 15:28:40 test /usr/local/sbin/opensips[3258]: DBG:tm:t_unref:
UNREF_UNSAFE: after is 0
Mar 11 15:28:40 test /usr/local/sbin/opensips[3258]:
DBG:core:destroy_avp_list: destroying list (nil)
Mar 11 15:28:40 test /usr/local/sbin/opensips[3258]: DBG:core:receive_msg:
cleaning up
Mar 11 15:28:40 test /usr/local/sbin/opensips[3259]: DBG:core:parse_msg: SIP
Reply  (status):
Mar 11 15:28:40 test /usr/local/sbin/opensips[3259]: DBG:core:parse_msg:
 version: 
Mar 11 15:28:40 test /usr/local/sbin/opensips[3259]: DBG:core:parse_msg:
 status:  <200>
Mar 11 15:28:40 test /usr/local/sbin/opensips[3259]: DBG:core:parse_msg:
 reason:  


What log could I add to your patch so we can get more details ?

I recompiled the whole opensips and its modules just to make sure your patch
is activated.

Cheers
Pascal


On Thu, Mar 11, 2010 at 2:07 PM, Anca Vamanu  wrote:

> Hi Pascal,
>
> I have just tested myself - with a Subscribe sent on udp and contact
> with trasport=tcp and the Notify is sent over tcp.
> I would suggest to run opensips in debug mode and paste the log messages
> when starting the processing for sending the Notify ( starting with
> "DBG:presence:build_dlg_t: CONTACT ").
>
> Regards,
>
> --
> Anca Vamanu
> www.voice-system.ro
>
>
>
> Pascal Maugeri wrote:
> > to: OpenSIPS users mailling list  > >
> > cc: Albert Petit mailto:albertpe...@gmail.com>>
> >
> > Hello Anca
> >
> > Sorry but this patch does not help neither :-(
> >
> > FYI we applied it against rev 6706.
> >
> > I will test this afternoon with a subscriber that is not behind NAT,
> > maybe we can isolate a bit more the problem.
> >
> > I'm going to answer to Iñaki's email providing tracers.
> >
> > Cheers
> > Pascal
> >
> >
> >
> > On Wed, Mar 10, 2010 at 3:02 PM, Anca Vamanu  > > wrote:
> >
> > Hi Pascal,
> >
> > Please try this revised patch. It is also against svn, take a new
> > copy of the tm module and apply it.
> >
> > Thanks and regards,
> >
> >
> > --
> > Anca Vamanu
> > www.voice-system.ro 
> >
> >
> >
> > Pascal Maugeri wrote:
> >
> > Hi Anca
> >
> > We just tested the patch you sent us but we don't see any
> > difference.
> >
> > Just in case, we recompiled the whole server (first we updated
> > to rev 6700) and applied the patch you kindly sent us:
> >
> >svn update
> >...
> >patch -p0 < tm_proto_fix.patch
> >make all
> >make install
> >service opensips restart
> >
> > The NOTIFY message is still sent over UDP :-(
> >
> > What information could we send you in order to help you ?
> >
> > A big thank you for your help and fast answer!
> >
> > Cheers
> > Pascal
> >
> > On Tue, Mar 9, 2010 at 4:55 PM, Anca Vamanu  >   > >> wrote:
> >
> >Hi Pascal,
> >
> >We found the problem - the socket on which the Subscribe was
> >received was always used when sending the Notify.
> >Find attached a patch for tm module that should fix this
> > problem.
> >Please test and report the results.
> >
> >Regards,
> >
> >-- Anca Vamanu
> >www.voice-system.ro 
> > 
> >
> >
> >
> >
> >Pascal Maugeri wrote:
> >
> >Hi
> >
> >Using opensips 1.6 we need to have the presence module
> > to send
> >NOTIFY over TCP rather than UDP when the size of the
> > message
> >is greater tha

Re: [OpenSIPS-Users] mediaproxy RTP ports

[Saúl Ibarra Corretgé]

Hi Julien,

On 11/3/10 2:55 PM, Julien Chavanton wrote:
> Hi, I have a question about mediaproxy port usage, mediaproxy is a very
> good friend of opensips so I guess I can post the question here.
> Our mediaproxy, is using RPT port > 5 it this optimal, shall I
> configure it differently ?
> I am concern that some equipement may implement QoS efforts using this
> logic :
> http://www.cs.columbia.edu/~hgs/rtp/faq.html#ports
> from to application priority
> 0-16383 unclassified lowest
> 16384-32767 audio highest
> 32768-49151 whiteboard medium
> 49152-65535 video low
>

You may change the port range by editing the config.ini configuration file:

port_range = 18000:2

Regards,

-- 
Saúl Ibarra Corretgé
AG Projects

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[OpenSIPS-Users] mediaproxy RTP ports

[Julien Chavanton]

Hi, I have a question about mediaproxy port usage, mediaproxy is a very good 
friend of opensips so I guess I can post the question here.
 
Our mediaproxy, is using  RPT port > 5 it this optimal, shall I configure 
it differently ?
 
I am concern that some equipement may implement QoS efforts using this logic :
 
http://www.cs.columbia.edu/~hgs/rtp/faq.html#ports 
 
fromto  application priority 
0-  16383   unclassifiedlowest 
16384-  32767   audio   highest 
32768-  49151   whiteboard  medium 
49152-  65535   video   low 
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Re: [OpenSIPS-Users] Listening on two different ports? Or two different addresses

[Kenneth Löfstrand]

Thank you, Bogdan

mhomed solved my problem. OpenIMS is based on SER but obviously the 
parameter is available also there.

Regards
Kenneth

Bogdan-Andrei Iancu skrev:
> Hi Kenneth,
>
> not sure what this openims is, but if something from the OpenSIPS 
> family, the normal behaviour is to preserve the socket (same inbound 
> socket will be used for outbound socket). But there are 2 exceptions:
> - if you cannot route across interface (totally different networks), 
> you may enable mhomed option and opensips will try to autodetect the 
> right outbound interface based on the destination IP
> http://www.opensips.org/Resources/DocsCoreFcn16#toc58
> - using from script force_send_socket("socket") - to force a 
> specific outbound interface.
> http://www.opensips.org/Resources/DocsCoreFcn16#toc101
>
> Regards,
> Bogdan
>
> Kenneth Löfstrand wrote:
>   
>> Hi
>> I have a question that might be related. Honestly it's not an opensips 
>> question, it happens when using openims but I get no answer from that 
>> list so I thought it might be worth a try here. Please ignore this mail 
>> if you think it doesn't belong to this list.
>>
>> I have assigned two IP addresses to my p-cscf, one address towards the 
>> access network and one towards core. The p-cscf was expected to use the 
>> core address address when forwarding a sip message towards the other 
>> cscf's and the access address when forwarding to the terminals but this 
>> doesn't happen. A sip message that was received on the access interface 
>> will be forwarded using the access IP address as source address. Could 
>> this be handled in the script or does it have to be patched in the code?
>>
>> Regards
>> Kenneth Löfstrand
>>
>>
>> Bogdan-Andrei Iancu skrev:
>>   
>> 
>>> Hi Chris,
>>>
>>> yes, that is the right approach - you can force the outgoing socket via 
>>> the "socket" param.
>>>
>>> Regards,
>>> Bogdan
>>>
>>> Chris Maciejewski wrote:
>>>   
>>> 
>>>   
 Sorry Bogdan,

 Ignore my email below. I missed 'socket' option in  't_uac_dlg' MI
 function, as described at:
 http://www.opensips.org/html/docs/modules/1.4.x/tm.html#id272529

 Regards,
 Chris

 On 8 March 2010 16:28, Chris Maciejewski  wrote:
   
 
   
 
> Hi Bogdan,
>
> Thanks for info. I managed to bind to two different ports 5060 and
> 5566 just fine.
>
> However it seems to me there is a problem with mi_xmlrpc module in
> this scenario.
>
> For example when I have Client 1 registered with my.domain:5566
> ('location' table shows correctly "udp:my_ip:5566" in 'socket' column)
> and I am sending OPTIONS with t_uac_dlg via MI RPC to the 'contact' of
> Client 1, OpenSIPs will send OPTIONS from udp:my_ip:5060 instead of
> udp:my_ip:5566 where REGISTER was received.
>
> As a result Client 1 (which is behind NAT) will not receive OPTIONS as
> NAT bindings don't match.
>
> OPTIONS generated by nathelper module are correctly sent from
> udp:my_ip:5566. It is just mi_xmlrpc which seems to ignore 'socket'
> column in 'location' table.
>
> Please let me know if you need any more info on this issue.
>
> Kind regards,
> Chris
>
>
> On 1 March 2010 06:57, Bogdan-Andrei Iancu  wrote:
> 
>   
> 
>   
>> Hi Chris,
>>
>> Sure it is possible. Do in cfg:
>>listen=udp:ip:5060
>>listen=udp:ip:5566
>>listen=tcp:ip:5060
>>listen=tcp:ip:5566
>>
>> And do not forget to set fork to yes.
>>
>> Regards,
>> Bogdan
>>
>> Chris Maciejewski wrote:
>>   
>> 
>>   
>> 
>>> Hi,
>>>
>>> I know the answer to this question is probably "no", but wanted to
>>> check with you.
>>>
>>> Is it possible to make one instance of opensips listen on two different 
>>> ports?
>>>
>>> Lets say if I wanted opensips to listen on UDP/TCP 5060 and UDP/TCP 
>>> 5566?
>>>
>>> Can this be done?
>>>
>>> Best regards,
>>> Chris
>>>
>>> ___
>>> Users mailing list
>>> Users@lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>>
>>> 
>>>   
>>> 
>>>   
>> --
>> Bogdan-Andrei Iancu
>> www.voice-system.ro
>>
>>
>> ___
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>>
>>   
>> 
>>   
>> 
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Re: [OpenSIPS-Users] [Presence] How to receive presence NOTIFY over TCP ?

[Pascal Maugeri]

OK I'm going to do that, thank you Anca.

On Thu, Mar 11, 2010 at 2:07 PM, Anca Vamanu  wrote:

> Hi Pascal,
>
> I have just tested myself - with a Subscribe sent on udp and contact
> with trasport=tcp and the Notify is sent over tcp.
> I would suggest to run opensips in debug mode and paste the log messages
> when starting the processing for sending the Notify ( starting with
> "DBG:presence:build_dlg_t: CONTACT ").
>
> Regards,
>
> --
> Anca Vamanu
> www.voice-system.ro
>
>
>
> Pascal Maugeri wrote:
> > to: OpenSIPS users mailling list  > >
> > cc: Albert Petit mailto:albertpe...@gmail.com>>
> >
> > Hello Anca
> >
> > Sorry but this patch does not help neither :-(
> >
> > FYI we applied it against rev 6706.
> >
> > I will test this afternoon with a subscriber that is not behind NAT,
> > maybe we can isolate a bit more the problem.
> >
> > I'm going to answer to Iñaki's email providing tracers.
> >
> > Cheers
> > Pascal
> >
> >
> >
> > On Wed, Mar 10, 2010 at 3:02 PM, Anca Vamanu  > > wrote:
> >
> > Hi Pascal,
> >
> > Please try this revised patch. It is also against svn, take a new
> > copy of the tm module and apply it.
> >
> > Thanks and regards,
> >
> >
> > --
> > Anca Vamanu
> > www.voice-system.ro 
> >
> >
> >
> > Pascal Maugeri wrote:
> >
> > Hi Anca
> >
> > We just tested the patch you sent us but we don't see any
> > difference.
> >
> > Just in case, we recompiled the whole server (first we updated
> > to rev 6700) and applied the patch you kindly sent us:
> >
> >svn update
> >...
> >patch -p0 < tm_proto_fix.patch
> >make all
> >make install
> >service opensips restart
> >
> > The NOTIFY message is still sent over UDP :-(
> >
> > What information could we send you in order to help you ?
> >
> > A big thank you for your help and fast answer!
> >
> > Cheers
> > Pascal
> >
> > On Tue, Mar 9, 2010 at 4:55 PM, Anca Vamanu  >   > >> wrote:
> >
> >Hi Pascal,
> >
> >We found the problem - the socket on which the Subscribe was
> >received was always used when sending the Notify.
> >Find attached a patch for tm module that should fix this
> > problem.
> >Please test and report the results.
> >
> >Regards,
> >
> >-- Anca Vamanu
> >www.voice-system.ro 
> > 
> >
> >
> >
> >
> >Pascal Maugeri wrote:
> >
> >Hi
> >
> >Using opensips 1.6 we need to have the presence module
> > to send
> >NOTIFY over TCP rather than UDP when the size of the
> > message
> >is greater than a given MTU.
> >
> >We tried to force it sending a SUBSCRIBE with
> > transport=tcp in
> >Contact header:
> >
> >   SUBSCRIBE 
> > sip:echo...@presence.net
> >  >
> >
> >  >  >>
> >   
> >  >  >
> >
> >  > 
> > >>
> SIP/2.0
> >
> >
> >   [...]
> >   Contact: 
> >   Event: presence
> >
> >
> >but the NOTIFY messages are sent over UDP (observe the
> >"transport=tcp" in the request line):
> >
> >   NOTIFY sip:albe...@10.1.1.24:5233;transport=tcp
> SIP/2.0
> >   [...]
> >   Contact:  > 
> >
> >   >
> >
> >   Subscription-State: active;expires=120
> >   Content-Type: application/pidf+xml
> >   Content-Length: 1254
> >
> >
> >Could you please let me know what we should do to
> > enable the
> >sending over TCP ? I don't understand what we're doing
> > wrong.
> >
> >Cheers
> >Pascal
> >
> >
>  
> >
> >
> >
> >___
> >Users mailing list
> >Users@lists.opensips.org
> > 
> >  > 

Re: [OpenSIPS-Users] SNOM BLF

[Anca Vamanu]

Hi Ross,

Can you please investigate a bit more? You can check in the traces what 
happened with that dialog ( search up after the string in Call-ID).
On the other hand, I would suggest updating to the svn version for 
presence and pua since it contains some important fixes.

Regards,

-- 
Anca Vamanu
www.voice-system.ro



Ross Beer wrote:
> Hi,
>  
> I don't know if anyone could point me in the right direction, but I am 
> trying to get Presence, BLF and BLF dialog working so that when a call 
> is made the lamp lights show the staus on the Snom.
>  
> At present it all works for a few minuites and then either just stops 
> working or the lights get stuck on.
>  
> I can see that Opensips is sending updates to the phone, however the 
> phone replies with:
>  
> SIP/2.0 481 Subscription Does Not Exist
> Via: SIP/2.0/UDP ;branch=z9hG4bK78c3.4b6b61f4.0
> From: ;tag=43b04376d7d592830afe5b44269baf2f-3d9f
> To: ;tag=kpuc8rtzrf
> Call-ID: 3c26702aa25f-py888tur215l
> CSeq: 4 NOTIFY
> Content-Length: 0
>  
> Has anyone had experience with this behaviour?
>  
> Kind regards,
>  
> Ross
>
> 
> We want to hear all your funny, exciting and crazy Hotmail stories. 
> Tell us now 
> 
>
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>   

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Re: [OpenSIPS-Users] [Presence] How to receive presence NOTIFY over TCP ?

[Anca Vamanu]

Hi Pascal,

I have just tested myself - with a Subscribe sent on udp and contact 
with trasport=tcp and the Notify is sent over tcp.
I would suggest to run opensips in debug mode and paste the log messages 
when starting the processing for sending the Notify ( starting with 
"DBG:presence:build_dlg_t: CONTACT ").

Regards,

-- 
Anca Vamanu
www.voice-system.ro



Pascal Maugeri wrote:
> to: OpenSIPS users mailling list  >
> cc: Albert Petit mailto:albertpe...@gmail.com>>
>
> Hello Anca
>
> Sorry but this patch does not help neither :-(
>
> FYI we applied it against rev 6706.
>
> I will test this afternoon with a subscriber that is not behind NAT, 
> maybe we can isolate a bit more the problem.
>
> I'm going to answer to Iñaki's email providing tracers.
>
> Cheers
> Pascal
>
>
>
> On Wed, Mar 10, 2010 at 3:02 PM, Anca Vamanu  > wrote:
>
> Hi Pascal,
>
> Please try this revised patch. It is also against svn, take a new
> copy of the tm module and apply it.
>
> Thanks and regards,
>
>
> -- 
> Anca Vamanu
> www.voice-system.ro 
>
>
>
> Pascal Maugeri wrote:
>
> Hi Anca
>
> We just tested the patch you sent us but we don't see any
> difference.
>
> Just in case, we recompiled the whole server (first we updated
> to rev 6700) and applied the patch you kindly sent us:
>
>svn update
>...
>patch -p0 < tm_proto_fix.patch
>make all
>make install
>service opensips restart  
>
> The NOTIFY message is still sent over UDP :-(
>
> What information could we send you in order to help you ?
>
> A big thank you for your help and fast answer!
>
> Cheers
> Pascal
>
> On Tue, Mar 9, 2010 at 4:55 PM, Anca Vamanu    >> wrote:
>
>Hi Pascal,
>
>We found the problem - the socket on which the Subscribe was
>received was always used when sending the Notify.
>Find attached a patch for tm module that should fix this
> problem.
>Please test and report the results.
>
>Regards,
>
>-- Anca Vamanu
>www.voice-system.ro 
> 
>
>
>
>
>Pascal Maugeri wrote:
>
>Hi
>
>Using opensips 1.6 we need to have the presence module
> to send
>NOTIFY over TCP rather than UDP when the size of the
> message
>is greater than a given MTU.
>
>We tried to force it sending a SUBSCRIBE with
> transport=tcp in
>Contact header:
>
>   SUBSCRIBE sip:echo...@presence.net
> 
> >
>    
> >> SIP/2.0
>
>
>   [...]
>   Contact: 
>   Event: presence
>
>
>but the NOTIFY messages are sent over UDP (observe the
>"transport=tcp" in the request line):
>
>   NOTIFY sip:albe...@10.1.1.24:5233;transport=tcp SIP/2.0
>   [...]
>   Contact:  
>
>   >
>
>   Subscription-State: active;expires=120
>   Content-Type: application/pidf+xml
>   Content-Length: 1254
>
>
>Could you please let me know what we should do to
> enable the
>sending over TCP ? I don't understand what we're doing
> wrong.
>
>Cheers
>Pascal
>  
>  
> 
>
>
>
>___
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> 
>  >
>
>http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
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>  >
>
>http://lists.opensips.org/cgi-bin/mail

[OpenSIPS-Users] SNOM BLF

[Ross Beer]

Hi,

 

I don't know if anyone could point me in the right direction, but I am trying 
to get Presence, BLF and BLF dialog working so that when a call is made the 
lamp lights show the staus on the Snom.

 

At present it all works for a few minuites and then either just stops working 
or the lights get stuck on.

 

I can see that Opensips is sending updates to the phone, however the phone 
replies with:

 

SIP/2.0 481 Subscription Does Not Exist
Via: SIP/2.0/UDP ;branch=z9hG4bK78c3.4b6b61f4.0
From: ;tag=43b04376d7d592830afe5b44269baf2f-3d9f
To: ;tag=kpuc8rtzrf
Call-ID: 3c26702aa25f-py888tur215l
CSeq: 4 NOTIFY
Content-Length: 0

 

Has anyone had experience with this behaviour?

 

Kind regards,

 

Ross
  
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Re: [OpenSIPS-Users] db_berkeley append

[Bogdan-Andrei Iancu]

Hi Liu,

Right - thanks for the report. I also fixed these tables.

Regards,
Bogdan

liuf wrote:
> Thanks. 
>
> Base svn revision 6624, I check files in db_berkeley/opensips/ again. I
> found the files which have no METADATA_KEY value as below:
>
>   dialplan
>   dr_gateways
>   dr_groups
>   dr_rules
>   globalblacklist
>   nh_sockets
>   userblacklist
>
> maybe you need fix them also. 
>   


-- 
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Re: [OpenSIPS-Users] [dialog] Can't fetch data in 200 reply route of BYE

[Bogdan-Andrei Iancu]

Hi Liu,

Ok - the bug is fixed now :)

Regards,
Bogdan

liuf wrote:
> Yes, you are right. They're the same.
>
> Had this bug been fixed yet? It seems still open now.
>
>
> Best Regards, 
> Liu Fan
>   


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Re: [OpenSIPS-Users] Getting asterisk to reply to OPTIONS keepalive

[Alejandro Recarey]

Great help, thank you both!

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Re: [OpenSIPS-Users] [Presence] How to receive presence NOTIFY over TCP ?

[Pascal Maugeri]

Hola Iñaki

Here is the SIP flow you asked me.
Hope it helps to spot the problem. Observe the watcher is behind NAT.

SUBSCRIBE sip:present...@presence.net SIP/2.0
Record-Route: 
Call-ID: d041f6cae786fea76503d43658570...@10.1.1.235
CSeq: 1 SUBSCRIBE
From: "watcher" 
>;tag=94671740
To: >
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:;branch=z9hG4bK581b.d3f0fa41.0
Via: SIP/2.0/UDP 10.1.1.235:5060
;rport=5060;received=yyy.yyy.yyy.yyy;branch=z9hG4bKc5Obqwa8W
Max-Forwards: 69
Contact: 
Event: presence
Expires: 3600
User-Agent: XXX
Accept: application/pidf+xml
Content-Length: 0

SIP/2.0 200 OK
Record-Route: 
Call-ID: d041f6cae786fea76503d43658570...@10.1.1.235
CSeq: 1 SUBSCRIBE
From: "watcher" 
>;tag=94671740
To: 
>;tag=9886f112d485ff608a79084eaa27a247-6965
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:;branch=z9hG4bK581b.d3f0fa41.0
Via: SIP/2.0/UDP 10.1.1.235:5060
;rport=5060;received=yyy.yyy.yyy.yyy;branch=z9hG4bKc5Obqwa8W
Expires: 120
Contact: 
Server: OpenSIPS (1.6.1-notls (x86_64/linux))
Content-Length: 0

NOTIFY sip:watc...@10.1.1.235:5060;transport=tcp SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:6667;branch=z9hG4bK581b.fd70f1f5.0
To: >;tag=94671740
From: 
>;tag=9886f112d485ff608a79084eaa27a247-6965
CSeq: 1 NOTIFY
Call-ID: d041f6cae786fea76503d43658570...@10.1.1.235
Route: 
Content-Length: 1030
User-Agent: OpenSIPS (1.6.1-notls (x86_64/linux))
Max-Forwards: 70
Event: presence
Contact: 
Subscription-State: active;expires=120
Content-Type: application/pidf+xml

[...]

Cheers
Pascal

On Wed, Mar 10, 2010 at 4:23 PM, Iñaki Baz Castillo  wrote:

> 2010/3/10 Pascal Maugeri :
> > Hi Anca
> > We just tested the patch you sent us but we don't see any difference.
>
> Could you paste a SIP flow for the SUBSCRIBE and NOTIFY?
>
> --
> Iñaki Baz Castillo
> 
>
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Re: [OpenSIPS-Users] Problem with OpenSIPS Control Panel 4.0

[Alex Ionescu]

Hi,

Did you add the first admin account using the query provided in the 
INSTALL file ?
Because if you did it should have worked without changing the value of 
*$config->admin_passwd_mode* from 1 to 0.


Regards,

Alex Ionescu

On 3/11/2010 13:51, TCB wrote:

Hi,

Had the same problem, change $config->admin_passwd_mode=1; to 
$config->admin_passwd_mode=0;
in opensips-cp/config/globals.php and it worked. I dont know is this 
is the right way.


regards


On Thu, Mar 11, 2010 at 2:29 AM, > wrote:


> I have the following problem
>
> With install OpenSIPS  1.6 and  Control Panel 4.0.
>
> When try the access the Control Panel didn't accept the user o
password
> (Invalid Username / Password)
>
> With the  commands opensips-c, monit-t said  that the
configuration files
> are fine.
>
>  I can register phone.
>
> The question is how do I know the password to the web interface,
try to
> file
> db.inc.php, boxes.global.inc.php, or  if I have to change
something in the
> database
>
> The reference for the installation of the control panel are Book
OpenSIPS
> 1.6 and the link http://opensips-cp.sourceforge.net/ , but both have
> differences
>
> This is my first time with OpenSIPS
>
> I appreciate the help
>
> Thanks
>
> Erick Ch.
>



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TC


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--
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Re: [OpenSIPS-Users] [Presence] How to receive presence NOTIFY over TCP ?

[Pascal Maugeri]

to: OpenSIPS users mailling list 
cc: Albert Petit 

Hello Anca

Sorry but this patch does not help neither :-(

FYI we applied it against rev 6706.

I will test this afternoon with a subscriber that is not behind NAT, maybe
we can isolate a bit more the problem.

I'm going to answer to Iñaki's email providing tracers.

Cheers
Pascal



On Wed, Mar 10, 2010 at 3:02 PM, Anca Vamanu  wrote:

> Hi Pascal,
>
> Please try this revised patch. It is also against svn, take a new copy of
> the tm module and apply it.
>
> Thanks and regards,
>
>
> --
> Anca Vamanu
> www.voice-system.ro
>
>
>
> Pascal Maugeri wrote:
>
>> Hi Anca
>>
>> We just tested the patch you sent us but we don't see any difference.
>>
>> Just in case, we recompiled the whole server (first we updated to rev
>> 6700) and applied the patch you kindly sent us:
>>
>>svn update
>>...
>>patch -p0 < tm_proto_fix.patch
>>make all
>>make install
>>service opensips restart
>>
>> The NOTIFY message is still sent over UDP :-(
>>
>> What information could we send you in order to help you ?
>>
>> A big thank you for your help and fast answer!
>>
>> Cheers
>> Pascal
>>
>> On Tue, Mar 9, 2010 at 4:55 PM, Anca Vamanu > a...@opensips.org>> wrote:
>>
>>Hi Pascal,
>>
>>We found the problem - the socket on which the Subscribe was
>>received was always used when sending the Notify.
>>Find attached a patch for tm module that should fix this problem.
>>Please test and report the results.
>>
>>Regards,
>>
>>-- Anca Vamanu
>>www.voice-system.ro 
>>
>>
>>
>>
>>Pascal Maugeri wrote:
>>
>>Hi
>>
>>Using opensips 1.6 we need to have the presence module to send
>>NOTIFY over TCP rather than UDP when the size of the message
>>is greater than a given MTU.
>>
>>We tried to force it sending a SUBSCRIBE with transport=tcp in
>>Contact header:
>>
>>   SUBSCRIBE sip:echo...@presence.net 
>>
>>   >>
>> SIP/2.0
>>
>>
>>   [...]
>>   Contact: 
>>   Event: presence
>>
>>
>>but the NOTIFY messages are sent over UDP (observe the
>>"transport=tcp" in the request line):
>>
>>   NOTIFY sip:albe...@10.1.1.24:5233;transport=tcp SIP/2.0
>>   [...]
>>   Contact: >
>>   >
>>
>>   Subscription-State: active;expires=120
>>   Content-Type: application/pidf+xml
>>   Content-Length: 1254
>>
>>
>>Could you please let me know what we should do to enable the
>>sending over TCP ? I don't understand what we're doing wrong.
>>
>>Cheers
>>Pascal
>>
>>  
>>
>>
>>
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Re: [OpenSIPS-Users] Problem with OpenSIPS Control Panel 4.0

[Alex Ionescu]

Hi,

From your post I understand that you are trying to access the Control 
Panel using a SIP user username + password.
You can only log in into CP using administrator account + password. (You 
can add one or more admin accounts if you want).
If you read the INSTALL file you will see how to add *the first admin* 
account into the DB (check lines 103 & 104).


In config/globals.php : if you change *$config**->admin_passwd_mode* 
from 1 (which is default - and means that the password is encoded in 
MD5) to 0 it will allow you to use plain text passwords for the admin 
accounts that you will add from the interface.


If you read in the INSTALL file you will see that the first admin 
account is added using both an encoded and plain text password. So, the 
first admin should be able to log in regardless the value of 
$config->admin_passwd_mode.


I hope this helps you.

Regards,

Alex


On 21:59, Erick Chinchilla Berrocal wrote:


I have the following problem

With install OpenSIPS  1.6 and  Control Panel 4.0.

When try the access the Control Panel didn't accept the user o 
password (Invalid Username / Password)


With the  commands opensips-c, monit-t said  that the configuration 
files are fine.


 I can register phone.

The question is how do I know the password to the web interface, try 
to file db.inc.php, boxes.global.inc.php, or  if I have to change 
something in the database


The reference for the installation of the control panel are Book 
OpenSIPS 1.6 and the link  http://opensips-cp.sourceforge.net/ , but 
both have differences


This is my first time with OpenSIPS

I appreciate the help

Thanks

Erick Ch.



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Re: [OpenSIPS-Users] Problem with OpenSIPS Control Panel 4.0

[TCB]

Hi,

Had the same problem, change $config->admin_passwd_mode=1; to
$config->admin_passwd_mode=0;
in opensips-cp/config/globals.php and it worked. I dont know is this is the
right way.

regards


On Thu, Mar 11, 2010 at 2:29 AM,  wrote:

> > I have the following problem
> >
> > With install OpenSIPS  1.6 and  Control Panel 4.0.
> >
> > When try the access the Control Panel didn't accept the user o password
> > (Invalid Username / Password)
> >
> > With the  commands opensips-c, monit-t said  that the configuration files
> > are fine.
> >
> >  I can register phone.
> >
> > The question is how do I know the password to the web interface, try to
> > file
> > db.inc.php, boxes.global.inc.php, or  if I have to change something in
> the
> > database
> >
> > The reference for the installation of the control panel are Book OpenSIPS
> > 1.6 and the link  http://opensips-cp.sourceforge.net/ , but both have
> > differences
> >
> > This is my first time with OpenSIPS
> >
> > I appreciate the help
> >
> > Thanks
> >
> > Erick Ch.
> >
>
>
>
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Re: [OpenSIPS-Users] Getting asterisk to reply to OPTIONS keepalive

[Olle E. Johansson]

11 mar 2010 kl. 10.33 skrev Bogdan-Andrei Iancu:

> Hi Alejandro,
> 
> I'm aware of the incapacity of Asterisk of properly answer to OPTIONS 

Bogdan, 
Asterisk can surely handle OPTIONS, if you configure it properly.

Add an "s" extension in the dialplan context that is used, and OPTIONS will
reply 200 OK. Extension "s" is used when no extension is given, and OPTIONS
is usually sent without any username part in the SIP uri, thus no extension.

/O




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Re: [OpenSIPS-Users] creating priority groups with drouting

[Bogdan-Andrei Iancu]

Hi Andrew,

you can achieve what you need by using the "sort_order" option:
   
http://www.opensips.org/html/docs/modules/1.6.x/drouting.html#id272689

Probably alg. 1 is what you need:

"/1/ - the destinations from each group are randomly arranged (only the 
two elements are randomly selected); groups do maintain their order (as 
given); the resulting list is used (with all the defined destinations). 
Ex: 1,2;3,4,5;6 -> randomizer -> (A) 2,1;4,3,5;6 -> usage 2,1,4,3,5,6 
(B) 1,2;3,5,4;6 -> usage 1,2,3,5,4,6 "


Right now, there is no need for falling back to a different rule as you 
can put all the GW options (as list or groups) on the same rule:
   gwlist = "cheap1, cheap2, cheap3 ; expensive1, expesive2, 
expensive3 "


Regards,
Bogdan

Andrew Pogrebennyk wrote:
> Hi,
>
> I'm looking for a way to setup two groups of gws to route the traffic to 
> with drouting. OpenSIPS shall use gws in second group only if all gws 
> within first group failed. So it shall go over the groups sequentially 
> but within the group, it's random. Think of it like second group 
> contains expensive e.g. EC2 gateways.
>
> But as you all know drouting does not do any rule fallback - once a rule 
> is matched, it will use only the destinations from this rule and it will 
> not try to re-match a different rule.
> Also I find it kind of limiting that the priority field in drouting is 
> considered only when the time selection is used for overlapping rules.
>
> So it looks like I can't achieve this behavior with drouting in its 
> current shape. Actually it would be sufficient for me that if all the 
> gws in a rule (higher priority) fail then this rule (or its gws) be 
> automatically included in a blacklist, so if you call again the next 
> rule (minor priority) would be taken. Makes sense? Is there anything I 
> could try?
>
>   


-- 
Bogdan-Andrei Iancu
www.voice-system.ro


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Re: [OpenSIPS-Users] Getting asterisk to reply to OPTIONS keepalive

[Bogdan-Andrei Iancu]

Hi Alejandro,

I'm aware of the incapacity of Asterisk of properly answer to OPTIONS 
pings, so we made a fix on the opensips side - the LB module has a 
parameter that allows you do define an extra list of reply codes to be 
considered as success (aside 200 ok). See:

http://www.opensips.org/html/docs/modules/1.6.x/load_balancer.html#id228255

So, just do :

modparam("load_balancer", "probing_reply_codes", "404")

Regards,
Bogdan



Alejandro Recarey wrote:
> Hi all,
>
> I'm trying to use the keepalive / ping option in OpenSIPS Loadbalancer
> module, but asterisk always responds with a 404 not found. Has anybody
> managed to get this working with Asterisk 1.6.1.X + OpenSIPS 1.6.2 ?
>
> Thanks!
>
> Alex
>
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