Guys,
I'm receiving the following output when running ser:
0(5705) Aug 03 06:58:26 DEBUG: register_fifo_cmd: new command (sl_stats)
registered
0(5705) Aug 03 06:58:26 DEBUG: MD5 calculated:
103d8e7e9fc1a3538c1143a96b22bab4
0(5705) Aug 03 06:58:26 DEBUG: init_mod: tm
0(5705) Aug 03 06:58:26 TM - initializing...
0(5705) Aug 03 06:58:26 Call-ID initialization: '79dff3eb'
0(5705) Aug 03 06:58:26 DEBUG: register_fifo_cmd: new command (t_uac_dlg)
registered
0(5705) Aug 03 06:58:26 DEBUG: register_fifo_cmd: new command (t_hash)
registered
0(5705) Aug 03 06:58:26 DEBUG: lock_initialize: lock initialization started
0(5705) Aug 03 06:58:26 init_mod(): Error while initializing module tm
ERROR: error while initializing modules
Do you know why?
Thanks!
Hi Adrian,
I'd like a feature which allow get in real time the MOS of each call in the media sessions interface.
Regards,
Areis.
No virus found in this outgoing message.
Checked by AVG Free Edition.
Version: 7.5.476 / Virus Database: 269.11.2/933 - Release Date: 2/8/2007 14:22
Hi,
When deploying a multi-location openser+asterisk architecture what is
the best way of storing the user accounts?
I thought about separating each location with a different prefix and
having a mysql DB for each one. Then tell openser to route according to
the prefixes. I like this version because it means if one location fails
it still works locally. And the other servers are not affected.
But this brings a problem with centralizing CDR. And if I want to
interconnect with other VoIP networks I would have to pick a central
location, right?
Thanks,
Radu Spineanu
Hi,
How can I add a header to SDP?
Actually what I want to do is, whenever an INVITE with SDP comes, search for a ptime and if ptime can not be found, add a default line to SDP body.
Like
if (has_body()){
if (!search_body("ptime") {
# like below
à # append_body("a=ptime:20");
}
}
Thanks,
Kuddusi CIFTCIBASI
Next Generation Networks
Teletek Telecommunication Services Corp.
Ayazma Dere Cad. Aksit Plaza No:12/1
Fulya, Besiktas , 34349, www.teletek.net
Tel : +90 212 227 7030
Direkt: +90 212 310 2233
Fax : +90 212 227 8700
Email: kuddusi.ciftcibasi(a)teletek.net
Skype : realkudu
MSN : kuddusic(a)hotmail.com
Yahoo: kuddusi.
Hello,
I have a problem when i try to add "presence_xml" module :
ERROR : load_module: could not open module
(/lib/openser/modules/presence_xml.so) :
/lib/openser/modules/presence_xml.so : undefined symbol : add_event
Do you have any idea about this error?
Here is an extract of my openser configuration file:
loadmodule "sl.so"
loadmodule "tm.so"
loadmodule "rr.so"
loadmodule "presence.so"
loadmodule "presence_xml.so"
loadmodule "maxfwd.so"
loadmodule "usrloc.so"
loadmodule "registrar.so"
loadmodule "textops.so"
loadmodule "mi_fifo.so"
# -- presence params --
modparam("presence", "db_url",
"mysql://openser:openserrw@localhost/openser")
modparam("presence_xml", "db_url",
"mysql://openser:openserrw@localhost/openser")
modparam("presence", "presentity_table", "presentity")
modparam("presence", "active_watchers_table", "active_watchers")
modparam("presence", "watchers_table", "watchers")
modparam("presence_xml", "xcap_table", "xcaps")
modparam("presence", "clean_period", 100)
modparam("presence", "to_tag_pref", 'a')
#modparam("presence", "lock_set_size", 8)
modparam("presence", "expires_offset", 10)
modparam("presence_xml", "force_active", 1)
modparam("presence", "max_expires", 3600)
modparam("presence", "server_address", "sip:localhost:5065")
modparam("presence_xml", "pidf_manipulation", 1)
Thanks for your help.
Frederic.
Hi,
When I invoke the loose_route() function I get the following info:
loose_route: no routing target is local
Even though the message contains the Route header fields the functions returns false.
Please point me what do I miss here??
Thanks in advance.
Cheers
Tomasz
Hello everybody,
To remind you, I'm trying to set a SIP architecture for NATTED SIP users
without using any rtp proxy (because of lack of ressource).
STUN is working very well between 2 users in 2 differents NATs.
But when they are behing the same NAT, they cannot reach the other.. In
the INVITE and 200Ok SDP fields, they have put their public address with
a port reserved with STUN: it's not working!
I think that it is because it's difficult for them to reach the public
IPaddress/port where they should send the RTP stream from inside the NAT.
So my idea was: Why not rewriting SDP fields to put their private
address when they are behind the same NAT
But I don't know how to rewrite the SDP fields for both users. (I have
the private address of both users in the location database)
Help would be very grateful...
If you have any idea!
Best Regards
I want OpenSer to dinamically look for alias in a table for each domain, so
I've in my database tables with name as:
dbaliases_sip1_domain_org
dbaliases_sip2_domain_com
dbaliases_sip3_domain_net
And I do (thanks to Andreas Granig):
$avp(s:dom) = $rd;
avp_subst("$avp(s:dom)", "/\./_/g");
alias_db_lookup("dbaliases_" + $avp(s:dominio));
but it fails because the SQL query is literally:
select username,domain from dbaliases_$avp(s:dominio) where ...
I've tryed putting variables and more things into the function parameter, but
it's never processed and the SQL query is always done literally. I use
OpenSer 1.2.1.
What can I do for solving this? Thanks.
--
Iñaki Baz Castillo
ibc(a)in.ilimit.es
Do you have any idea to solve this matter:
In the BYE request:
0(2569) found end of header
0(2569) find_next_route: No next Route HF found
0(2569) after_loose: No next URI found
So the BYE message is not forwarded by openser
Please find below INVITE and BYE requests
0(2632) SIP Request:
0(2632) method: <INVITE>
0(2632) uri: <sip:103@sd-7501.dedibox.fr:5060>
0(2632) version: <SIP/2.0>
0(2632) parse_headers: flags=2
0(2632) Found param type 232, <branch> = <z9hG4bK4747925369759203710>; state=16
0(2632) end of header reached, state=5
0(2632) parse_headers: Via found, flags=2
0(2632) parse_headers: this is the first via
0(2632) After parse_msg...
0(2632) preparing to run routing scripts...
0(2632) parse_headers: flags=100
0(2632) DEBUG:parse_to:end of header reached, state=10
0(2632) DBUG:parse_to: display={}, ruri={sip:103@sd-7501.dedibox.fr:5060;user=phone}
0(2632) DEBUG: get_hdr_field: <To> [46]; uri=[sip:103@sd-7501.dedibox.fr:5060;user=phone]
0(2632) DEBUG: to body [<sip:103@sd-7501.dedibox.fr:5060;user=phone>
]
0(2632) get_hdr_field: cseq <CSeq>: <1> <INVITE>
0(2632) DEBUG:maxfwd:is_maxfwd_present: value = 70
0(2632) DBG:maxfwd:process_maxfwd_header: value 70 decreased to 10
0(2632) ROUTE2: STARTING NAT DETECTION
0(2632) !!!!!!!!! NAT UAC TEST 19 SUCEDEED
0(2632) parse_headers: flags=80
0(2632) ROUTE2: SETFLAG 3
0(2632) DEBUG: add_param: tag=c0a80101-b67ff5
0(2632) DEBUG:parse_to:end of header reached, state=29
0(2632) DBUG:parse_to: display={"101"}, ruri={sip:101@sd-7501.dedibox.fr:5060;user=phone}
0(2632) parse_headers: flags=200
0(2632) DEBUG: get_hdr_body : content_length=269
0(2632) found end of header
0(2632) find_first_route: No Route headers found
0(2632) loose_route: There is no Route HF
0(2632) DEBUG: has_totag: no totag
0(2632) I AM SETTING THE FLAGS FOR RADIUS
0(2632) SETTING FLAGS 1 & 2 FOR RADIUS
0(2632) CHECKING IF URI <> myself
0(2632) grep_sock_info - checking if host==us: 18==12 && [sd-7501.dedibox.fr] == [88.191.45.91]
0(2632) grep_sock_info - checking if port 5060 matches port 5060
0(2632) grep_sock_info - checking if host==us: 18==12 && [sd-7501.dedibox.fr] == [88.191.45.91]
0(2632) grep_sock_info - checking if port 5060 matches port 5060
0(2632) grep_sock_info - checking if host==us: 18==12 && [sd-7501.dedibox.fr] == [88.191.45.91]
0(2632) grep_sock_info - checking if port 5060 matches port 5060
0(2632) grep_sock_info - checking if host==us: 18==12 && [sd-7501.dedibox.fr] == [88.191.45.91]
0(2632) grep_sock_info - checking if port 5060 matches port 5060
0(2632) rewrite_uri: Rewriting Request-URI with 'sip:103@82.127.0.79:1028;user=phone'
0(2632) parse_headers: flags=ffffffffffffffff
0(2632) STARTING ROUTE 1
0(2632) subst_run: running. r=1
0(2632) subst_str: no match
0(2632) FLAG 3 OK GOTO ROUTE 3
0(2632) !!!!!!!!! ON ROUTE 3 FOR NATTED CONTACT
0(2632) DEBUG: t_newtran: T on entrance=0xffffffff
0(2632) parse_headers: flags=ffffffffffffffff
0(2632) parse_headers: flags=78
0(2632) t_lookup_request: start searching: hash=12532, isACK=0
0(2632) DEBUG: RFC3261 transaction matching failed
0(2632) DEBUG: t_lookup_request: no transaction found
0(2632) DBG: trans=0xb5c08fa8, callback type 1, id 1 entered
0(2632) trace_onreq_in: trace off...
0(2632) DBG: trans=0xb5c08fa8, callback type 1, id 0 entered
0(2632) parse_headers: flags=78
0(2632) DEBUG: noisy_timer set for accounting
0(2632) DEBUG:rr:is_direction: param ftag not found
0(2632) parse_headers: flags=ffffffffffffffff
0(2632) check_via_address(82.127.0.79, 82.127.0.79, 0)
0(2569) SIP Request:
0(2569) method: <BYE>
0(2569) uri: <sip:101@82.127.0.79:1312>
0(2569) version: <SIP/2.0>
0(2569) parse_headers: flags=2
0(2569) Found param type 232, <branch> = <z9hG4bK2074253192092946047>; state=16
0(2569) end of header reached, state=5
0(2569) parse_headers: Via found, flags=2
0(2569) parse_headers: this is the first via
0(2569) After parse_msg...
0(2569) preparing to run routing scripts...
0(2569) parse_headers: flags=100
0(2569) DEBUG: add_param: tag=c0a80101-b31387
0(2569) DEBUG:parse_to:end of header reached, state=29
0(2569) DBUG:parse_to: display={}, ruri={sip:101@sd-7501.dedibox.fr:5060;user=phone}
0(2569) DEBUG: get_hdr_field: <To> [66]; uri=[sip:101@sd-7501.dedibox.fr:5060;user=phone]
0(2569) DEBUG: to body [<sip:101@sd-7501.dedibox.fr:5060;user=phone>]
0(2569) get_hdr_field: cseq <CSeq>: <1> <BYE>
0(2569) DEBUG:maxfwd:is_maxfwd_present: value = 70
0(2569) DBG:maxfwd:process_maxfwd_header: value 70 decreased to 10
0(2569) ROUTE2: STARTING NAT DETECTION
0(2569) !!!!!!!!! NAT UAC TEST 19 SUCEDEED
0(2569) parse_headers: flags=80
0(2569) DEBUG: get_hdr_body : content_length=0
0(2569) found end of header
0(2569) ROUTE2: SETFLAG 3
0(2569) DEBUG: add_param: tag=c0a80101-2376fc2
0(2569) DEBUG:parse_to:end of header reached, state=29
0(2569) DBUG:parse_to: display={}, ruri={sip:103@sd-7501.dedibox.fr:5060;user=phone}
0(2569) parse_headers: flags=200
0(2569) is_preloaded: No
0(2569) grep_sock_info - checking if host==us: 11==12 && [82.127.0.79] == [88.191.45.91]
0(2569) grep_sock_info - checking if port 5060 matches port 1312
0(2569) grep_sock_info - checking if host==us: 11==12 && [82.127.0.79] == [88.191.45.91]
0(2569) grep_sock_info - checking if port 5060 matches port 1312
0(2569) DEBUG:check_self: host != me
0(2569) grep_sock_info - checking if host==us: 12==12 && [88.191.45.91] == [88.191.45.91]
0(2569) grep_sock_info - checking if port 5060 matches port 5060
0(2569) after_loose: Topmost route URI: 'sip:88.191.45.91;lr=on;ftag=c0a80101-b31387' is me
0(2569) parse_headers: flags=200
0(2569) found end of header
0(2569) find_next_route: No next Route HF found
0(2569) after_loose: No next URI found
0(2569) DBG:rr:run_rr_callbacks: callback id 0 entered with <lr=on;ftag=c0a80101-b31387>
Dan-Cristian Bogos a écrit :
> That's because for INVITE your will take routing decisions but BYE u
> will just proxy out.
>
> DanB
>
> On 8/2/07, Marc LEURENT <lftsy(a)free.fr> wrote:
>> I've compared the INVITE and BYE method....
>> And the uri in the INVITE method is
>> 0(2632) SIP Request:
>> 0(2632) method: <INVITE>
>> 0(2632) uri: <sip:103@sd-7501.dedibox.fr:5060>
>> 0(2632) version: <SIP/2.0>
>>
>> whereas in the BYE method
>>
>> 0(2569) SIP Request:
>> 0(2569) method: <BYE>
>> 0(2569) uri: <sip:101@82.127.0.79:1312>
>> 0(2569) version: <SIP/2.0>
>>
>>
>> so the DEBUG:check_self: host != me
>> I'm going to try without accoounting, but it should'nt change anything...
>>
>> Best Regards
>>
>>
>>
>>
>>
>>
Hello,
On 08/03/07 08:41, null_neo(a)sina.com wrote:
> Hi
>
> I am glad to recieve a mail to help me!
> I have worked out today.Just force OpenSER only listening on IPV6 address.
yep, maybe should be stated clear that you should force
listen=ipv6_address to have openser managing ipv6 traffic.
Daniel
> And I use Linphone as UA ,they worked well.
> HeHe,I'm very happy because I finished this job(without reward) and I
> can have my holidays some days later.
> Thank you.
>
>
>
> ----- Original Message -----
> From:Daniel-Constantin Mierla
> To:null_neo@sina.com
> Subject:Re: [OpenSER-Users] how does OpenSER run on IPV6?
> Date:07-08-02 18:11:37
>
> Hello,
>
> the "result was bad" does not give us hints to help you.
> Please paste
> the error messages you get.
>
> Cheers,
> Daniel
>
>
> On 07/31/07 04:29, null_neo(a)sina.com wrote:
> >
> > Dear sir:
> >
> > First,I am a Chinese,and my English is poor.^_^
> >
> > We are testing kinds of applications on IPV6 including VOIP. And we
> > use Linphone as our UA. If I don't use domain name,how can I add a
> > account? On IPV4,I add account " # openserctl add test test
> > test@ipv4_address" .And in the file "openser.cfg" ,I
> directly write my
> > SIP server ipv4_address . Can I directly replace
> "ipv4_address" by
> > "ipv6_address"? I has tried,and the resault is bad.
> >
> > I am very sorry that I am amateurish for VOIP ,because Our work is
> > only to test.
> >
> > Thank you!
> >
> > Li Zhonglei
> >
> >
> >
> > -------------------------------------------------------------------
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> >
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>
> > )
> >
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> >
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> > Users(a)openser.org
> > http://openser.org/cgi-bin/mailman/listinfo/users
> >
>
>
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