Re: [OpenSIPS-Users] LDAP authentification
Hi Mehdi,
well, you configured the AUTH module to look for username and password
in $avp(i:1) and $avp(i:2), but you populate $var(userame) and
$var(password).:D
i guess this is the error!
Regards,
Bogdan
Mehdi Bouchefra wrote:
Hi Bogdan,
Thank's for your reply,
I followed the tutorial that you sent me, but I have a
In my ldap I use plane format password.
Thank's in advance,
Mehdi
Here my opensips.cfg file:
### Global Parameters #
debug=3
log_stderror=no
log_facility=LOG_LOCAL0
fork=yes
children=4
/* uncomment the following lines to enable debugging */
#debug=6
#fork=no
#log_stderror=yes
/* uncomment the next line to disable TCP (default on) */ #disable_tcp=yes
/* uncomment the next line to enable the auto temporary blacklisting of
not available destinations (default disabled) */
#disable_dns_blacklist=no
/* uncomment the next line to enable IPv6 lookup after IPv4 dns
lookup failures (default disabled) */ #dns_try_ipv6=yes
/* uncomment the next line to disable the auto discovery of local aliases
based on revers DNS on IPs (default on) */ #auto_aliases=no
/* uncomment the following lines to enable TLS support (default off) */
#disable_tls = no #listen = tls:your_IP:5061 #tls_verify_server = 1
#tls_verify_client = 1 #tls_require_client_certificate = 0 #tls_method =
TLSv1 #tls_certificate = /usr/local/etc/opensips/tls/user/user-cert.pem
#tls_private_key = /usr/local/etc/opensips/tls/user/user-privkey.pem
#tls_ca_list = /usr/local/etc/opensips/tls/user/user-calist.pem
port=5060
/* uncomment and configure the following line if you want opensips to
bind on a specific interface/port/proto (default bind on all available)
*/ #listen=udp:192.168.1.2:5060
### Modules Section
#set module path
mpath=/usr/local/lib/opensips/modules/
/* uncomment next line for MySQL DB support */ loadmodule db_mysql.so
loadmodule signaling.so
loadmodule sl.so
loadmodule tm.so
loadmodule rr.so
loadmodule maxfwd.so
loadmodule usrloc.so
loadmodule registrar.so
loadmodule textops.so
loadmodule mi_fifo.so
loadmodule uri.so
loadmodule xlog.so
loadmodule acc.so
loadmodule ldap.so
modparam(mi_fifo, fifo_name, /tmp/opensips_fifo)
modparam(usrloc, db_mode, 2)
modparam(usrloc, db_url,
mysql://opensips:opensip...@localhost/opensips)
modparam(uri, use_uri_table, 0)
modparam(acc, detect_direction, 0)
modparam(acc, failed_transaction_flag, 3) modparam(acc, log_flag, 1)
modparam(acc, log_missed_flag, 2)
### Routing Logic
# main request routing logic
modparam(auth, nonce_expire, 30)
modparam(auth, secret, sunny2009)
modparam(auth, disable_nonce_check, 0)
modparam(auth, username_spec, $avp(i:2))
modparam(auth, password_spec, $avp(i:1))
modparam(auth, calculate_ha1, 0)
route{
if (!mf_process_maxfwd_header(10)) {
sl_send_reply(483,Too Many Hops);
exit;
}
if (has_totag()) {
# sequential request withing a dialog should
# take the path determined by record-routing
if (loose_route()) {
if (is_method(BYE)) {
setflag(1); # do accounting ...
setflag(3); # ... even if the transaction
fails
} else if (is_method(INVITE)) {
# even if in most of the cases is useless,
do RR for
# re-INVITEs alos, as some buggy clients do
change route set
# during the dialog.
record_route();
}
# route it out to whatever destination was set by
loose_route()
# in $du (destination URI).
route(1);
} else {
/* uncomment the following lines if you want to
enable presence */
##if (is_method(SUBSCRIBE) $rd ==
your.server.ip.address) {
## # in-dialog subscribe requests
## route(2);
## exit;
##}
if ( is_method(ACK) ) {
if ( t_check_trans() ) {
# non loose-route, but stateful ACK;
must be an ACK after
# a 487 or e.g. 404 from upstream
server
t_relay();
exit;
} else {
# ACK without matching transaction
-
# ignore and discard
exit;
}
}
[OpenSIPS-Users] opensips 1.6.1 crashes on NOTIFY?
Hi all.
I've tried to update to Opensips 1.6.1, but encountered the following
problem. Opensips starts successfully, but soon almost all it's processes
die one by one and only two processes remain.
For example, if right after start we have:
# ps ax | grep opens
26182 ?S 0:00 ./opensips -k 0x0204 -u opensips
26183 ?S 0:00 ./opensips -k 0x0204 -u opensips
26184 ?S 0:00 ./opensips -k 0x0204 -u opensips
26185 ?S 0:00 ./opensips -k 0x0204 -u opensips
26186 ?S 0:00 ./opensips -k 0x0204 -u opensips
26187 ?S 0:00 ./opensips -k 0x0204 -u opensips
26188 ?S 0:00 ./opensips -k 0x0204 -u opensips
26189 ?S 0:00 ./opensips -k 0x0204 -u opensips
26190 ?S 0:00 ./opensips -k 0x0204 -u opensips
26191 ?S 0:00 ./opensips -k 0x0204 -u opensips
26192 ?S 0:00 ./opensips -k 0x0204 -u opensips
26193 ?S 0:00 ./opensips -k 0x0204 -u opensips
26194 ?S 0:00 ./opensips -k 0x0204 -u opensips
26195 ?S 0:00 ./opensips -k 0x0204 -u opensips
26196 ?S 0:00 ./opensips -k 0x0204 -u opensips
26197 ?S 0:00 ./opensips -k 0x0204 -u opensips
26198 ?S 0:00 ./opensips -k 0x0204 -u opensips
26199 ?S 0:00 ./opensips -k 0x0204 -u opensips
26200 ?S 0:00 ./opensips -k 0x0204 -u opensips
26201 ?S 0:00 ./opensips -k 0x0204 -u opensips
26202 ?S 0:00 ./opensips -k 0x0204 -u opensips
26203 ?S 0:00 ./opensips -k 0x0204 -u opensips
26204 ?S 0:00 ./opensips -k 0x0204 -u opensips
26205 ?S 0:00 ./opensips -k 0x0204 -u opensips
26206 ?S 0:00 ./opensips -k 0x0204 -u opensips
26207 ?S 0:00 ./opensips -k 0x0204 -u opensips
26208 ?S 0:00 ./opensips -k 0x0204 -u opensips
When processes die, we have only:
#ps ax | grep opens
26182 ?S 0:00 ./opensips -k 0x0204 -u opensips
26184 ?S 0:00 ./opensips -k 0x0204 -u opensips
If I set debug=6, the following is written to /var/log/messages:
Dec 22 11:02:03 srv rtpproxy[17011]: INFO:rxmit_packets: caller's address
filled in: 195.182.195.206:1024 (RTP)
Dec 22 11:02:03 srv opensips[26184]: Route 5 - NOTIFY
Dec 22 11:02:05 srv opensips[26185]: Route 5 - PUBLISH
Dec 22 11:02:06 srv opensips[26183]: Route 5 - NOTIFY
Dec 22 11:02:06 srv opensips[26185]: Route 5 - NOTIFY
Dec 22 11:02:06 srv opensips[26185]: Route 5 - NOTIFY
Dec 22 11:02:06 srv opensips[26186]: Route 5 - NOTIFY
Dec 22 11:02:06 srv opensips[26186]: Route 5 - NOTIFY
Dec 22 11:02:08 srv rtpproxy[17011]: INFO:handle_command: lookup on ports
36664/35096, session timer restarted
Dec 22 11:02:08 srv rtpproxy[17011]: INFO:handle_command: pre-filling
callee's address with 87.251.142.50:5006
Dec 22 11:02:08 srv opensips[26208]: CRITICAL:core:receive_fd: EOF on 13
Dec 22 11:02:08 srv opensips[26182]: INFO:core:handle_sigs: child process
26186 exited by a signal 11
Dec 22 11:02:08 srv opensips[26182]: INFO:core:handle_sigs: core was not
generated
Dec 22 11:02:08 srv opensips[26182]: INFO:core:handle_sigs: terminating due
to SIGCHLD
As I see, the last message received by process with PID 26186 is NOTIFY,
and then it crashes.
Route 5 - NOTIFY is in this block of configuration file:
# SUBSCRIBE and PUBLISH Message Handling
# --
route[5]
{
if (!t_newtran())
{
xlog(L_INFO, Failed to create transaction\n);
sl_reply_error();
exit;
}
if (is_method(PUBLISH))
{
xlog(L_INFO, Route 5 - PUBLISH \n);
handle_publish();
}
else if (is_method(SUBSCRIBE))
{
xlog(L_INFO, Route 5 - SUBSCRIBE\n);
handle_subscribe();
}
else if (is_method(NOTIFY))
{
xlog(L_INFO, Route 5 - NOTIFY\n);
t_reply(200, OK);
exit;
}
exit;
}
In main routing logic:
if (method == SUBSCRIBE || method == PUBLISH || method == NOTIFY)
{
route(4);
return(0);
}
As I see, Opensips sets core dump limit, if it's turned off, but no core
is produced (OS is CentOS 5.3).
What can be wrong? Version 1.6.0 did not crash like this.
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Re: [OpenSIPS-Users] Failed INVITE tcp_send
Hi Brian, opensipsl...@encambio.com wrote: Hello Bogdan, An ven., déc 18, 2009, Bogdan-Andrei Iancu schrieb: It is not really a hack :) i tend to think this number vary from OS to OS, from server to server .like how slow the write ops take place - on some system is faster, on other is not. What I'm not understanding is the basic principle, why even a single retry is needed. I can imagine that with a slightly different design the topic of 'guessing' the right retry ceiling is completely unnecessary. Do you understand more about these TLS retries, and does it have to do with TCP primarily or code in the OpenSSL libraries? copied from the man page of SSL_write: If the underlying BIO is blocking, SSL_write() will only return, once the write operation has been finished or an error occurred, except when a renegotiation take place, in which case a SSL_ERROR_WANT_READ may occur. This behaviour can be controlled with the SSL_MODE_AUTO_RETRY flag of the SSL_CTX_set_mode(3) call. maybe we should simply increase the default number to cover also the slow cases. I disagree. You said yourself that 'this number varies' so lets do what we always do with such variables, and put it in the config. PSEUDOCODE disable_tls = 0 tls_method= TLSv1 [...] tls_maxretries= 3200 # New variable Its not nice of course always stuffing up the config and OpenSIPS architecture with new variables, but if there's no design based solution to this then it seems better that way than hard coding such runtime variable deep into the core. There three things we try to make all happy: 1) easy to use (not so many cfg options) 2) cover all the cases - no need to tune for your system 3) be sure it works (no blocking of the write ops). The questions is what should be the best approach for making all this happy ? :) Regards, Bogdan -- Bogdan-Andrei Iancu www.voice-system.ro ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Failed INVITE tcp_send
Hi Brian, opensipsl...@encambio.com wrote: Hello Bogdan, An ven., déc 18, 2009, Bogdan-Andrei Iancu schrieb: opensipsl...@encambio.com wrote: One solution fixed both errors (assuming there really were two different erros) as you see below. Whoops I spoke too soon. It seems that patching MAX_SSL_RETRIES only fixed the 'tls_blocking_write' error. Now I still have in the log: error ERROR:core:tcp_blocking_connect: timeout 10 s elapsed from 10 s error ERROR:core:tcpconn_connect: tcp_blocking_connect failed error ERROR:core:tcp_send: connect failed error ERROR:tm:msg_send: tcp_send failed error ERROR:tm:t_forward_nonack: sending request failed I guess you are trying to connect to some destination which is not listening - check with tcpdump where opensips tries to open the TCP connection and see if there is a really app listening there. Seems reasonable, so I'll take your advise and start debugging with tcpdump. My guess is that there is some NAT problem and/or faulty IP number substitution in SDP (a config error basically.) The strange thing is that the same config was being used with OpenSER 1.3.X and these errors did not appear. I'll start tcpdump and report what I find. For sure it is after the lookup() and opensips tries to open a connection somwhere behind a NAT (where a TLS phone is located). Regards, Bogdan -- Bogdan-Andrei Iancu www.voice-system.ro ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] multiple Via headers separated by comma
Hi Josip, Post both the BYE (sent out by opesips) and 200 OK (received by opensips) in plain text . Regards, Bogdan Josip Djuricic wrote: Hi Andrew, Thanks very much for your quick answer, I understand that by rfc it is completely valid. What I can't seem to find is why is my last 200 OK from uas not beeing matched against the BYE that opensips forwarded to uas. So after uas sends 200 OK, it keeps receiveing BYE until timeout occurs. It only happens with sipp, if using any other uac everything works as expected. Every other transaction is matched correctly. I'm includig siplog from that last message, with changed ip's. Perhaps you would see this problem more clearly? Thanks, Josip -Original Message- From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Andrew Pogrebennyk Sent: Monday, December 21, 2009 10:44 AM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] multiple Via headers separated by comma Josip Djuricic wrote: Transaction is not matched if request is sent with 2 or more multiline via headers and response is received with via header in one line separated by comma? Josip, This is absolutely legal if multiple values are combined in one line separated by comma. Ccheck RFC 3261 for multiple header field values combining. Section 7.3. [H4.2] also specifies that multiple header fields of the same field name whose value is a comma-separated list can be combined into one header field. That applies to SIP as well, but the specific rule is different because of the different grammars. Specifically, any SIP header whose grammar is of the form header = header-name HCOLON header-value *(COMMA header-value) allows for combining header fields of the same name into a comma- separated list. The Contact header field allows a comma-separated list unless the header field value is *. Response is matched to request using branch parameter from uppermost Via header, so I don't know why RFC compliant implementation would have problems with response matching when Via header is combined. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Bogdan-Andrei Iancu www.voice-system.ro ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Failed INVITE tcp_send to UDP UACs
opensipsl...@encambio.com wrote: But this is maybe a clue. It would seem that something in TLS writing has changed between these two versions, maybe fundementally? 1.3 was doing infinite loop (for write and read), leading sometime to blocking. That was a painful part of 1.3, so good that the counter is there now. I guess you're saying that the same TLS problems existed in 1.3 as well, but they were masked by retries (maybe thousands.) yes, that is correct. Regards, Bogdan ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Failed INVITE tcp_send
Hello Bogdan, An mar., déc 22, 2009, Bogdan-Andrei Iancu schrieb: opensipsl...@encambio.com wrote: An ven., déc 18, 2009, Bogdan-Andrei Iancu schrieb: opensipsl...@encambio.com wrote: You may try to increase the number of tries to something higher - 3200, just to see if that is the problem: see tls/tls_server.c , line 689 #define MAX_SSL_RETRIES 320 By the way, even when MAX_SSL_RETRIES is set to almost infinity, the same errors reappear when lowering the number of child processes. This could be a clue for whoever decides to look into this bug. In the current state of development, it seems the workaround is: Increase MAX_SSL_RETRIES to almost infinity Increase tcp_children to the number of UACs (not scalable) Log how many SSL retries are really needed until success Lower MAX_SSL_RETRIES from almost infinity to what is needed Does that sound right? Is this acceptable as far as scalability goes? This is something interesting - can you open a bug report on the tracker - just to keep a history of the report and of the troubleshooting: http://www.opensips.org/Development/Tracker What is strange is that we are using also TLS and never bumped into this problem. Okay, let me just make sure I can reproduce the same errors under 1.6.1, and then I'll write the bug report. Regards, Brian ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] CRITICAL:core:sig_alarm_abort: BUG - shutdown timeout triggered, dying...
Hi Josip, Please send me the sipp scripts and how exactly to use it (I know how to use sipp) in order to get your scenario. Regards, Bogdan Josip Djuricic wrote: Hi Bogdan, yes, I have noticed in the trunk I've tested that even though registrations expires, and opensips removes it, and opensipsctl moni shows them under: usrloc:location-expires, and usrloc:location-contacts keeps getting lower by those usrloc:location-expires numbers. But usrloc:registered_users and usrloc:location_users keep rising constantly. And the memory usage also keeps rising constantly until I get out of shared memory. This only seams to happen when contact's expires, if re register comes in before timeout expires the memory stops rising. I've tested this with sipp and 100 000 users registering with 20 minutes expiration time, and 40-60 registrations per second. Going on and on...after few ours, no more shared memory available. I can send you any other detail you require? Also I can send you config and sipp scenario file? Best regards, Josip -Original Message- From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu Sent: Tuesday, December 22, 2009 10:28 AM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] CRITICAL:core:sig_alarm_abort: BUG - shutdown timeout triggered, dying... Hi Josip, So, what you are saying is that trunk has a problem (versus 1.6) - some records do not expire . Could you detail a bit the way I could reproduce this scenario ? Known bugs are listed on the tracker - http://www.opensips.org/Development/Tracker Regards, Bogdan Josip Djuricic wrote: Hi Bogdan, I've had to swithch to v1.6 stable, so It's working now :) What I notice is that on trunk version I had this UsrLoc Stats: usrloc:registered_users = 387432 usrloc:location-users = 387432 usrloc:location-contacts = 12005 usrloc:location-expires = 375427 but on stable 1.6 I have this: UsrLoc Stats: usrloc:registered_users = 12005 usrloc:location-users = 12005 usrloc:location-contacts = 12005 usrloc:location-expires = 375427 And I can confirm that memory is now stable, I think it seg faulted because at that ime it has gone 10 times trough 10users registration, what means usrloc:registered_users had more than 1 000 000 users, that could explain what happened. Somehow I think it was not clearing registered users no matter they expired and was deleted from db. Perhaps you can confirm that you can reproduce this problem? Also is there a possibility to get list of known limitations or perhaps bugs on v1.6 that I should be aware of (concerning stability issues before puttying the system in production use)? I know you mentioned release 1.6.1, so what should be important fixes you mentioned in that mail? Once again sorry for lot of questions. Thanks, Josip -Original Message- From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu Sent: Friday, December 18, 2009 1:26 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] CRITICAL:core:sig_alarm_abort: BUG - shutdown timeout triggered, dying... Hi Josip, A key question - how many records do you have in usrloc? I'm asking because opensips is flushing the usrloc at shutdown and if you have too many records, this will take some time. Also, the shutdown time is control by an alarm (couple of seconds), so if the shutdown takes too long, the alarm will simply kill opensips. Regards, Bogdan Josip Djuricic wrote: Hi, this is what happened tonight on trunk version of opensips. Any ideas? This is from log, I'm including backtrace also: ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Trying to learn how to use siptrace
Hello Bogdan,
An mar., déc 22, 2009, Bogdan-Andrei Iancu schrieb:
opensipsl...@encambio.com wrote:
An jeu., déc 17, 2009, Bogdan-Andrei Iancu schrieb:
Then if I have one UAC (PhoneA) and the proxy (OsipsA) and want
to trace all messages between them, it seems that the best way is:
# Example 1
route{
sip_trace();
[...]
}
But to to avoid REGISTER, SUBSCRIBE, and ACK messages, then it seems
best to set the trace flag. How do you do this appropriately, and
how to avoid calling setflag more than once per transaction? Maybe
something like:
# Example 2
route{
[...]
if (loose_route()) # Not sure here ?!?
setflag(22);
[...]
}
It seems in the context described above it makes no difference
if I use setflag, setbflag, or setsflag, right?
Lastly, it seems that one can get a similar result as example 2
above by doing:
# Example 3
route{
[...]
if (is_method(INVITE) !has_totag())
trace_dialog();
[...]
}
...just that in this case I will see the ACK messages as well.
The siptrace module require you to set a message/transaction
flag (for forcing the siptracing for a transaction) (see
http://www.opensips.org/Resources/DocsCoreFlags16)
Okay the setflag(trace) is required for transactions. I understand
that it is not required to trace a single message, where sip_trace()
will do the job just fine on its own.
In the route{} which is a request route, you get only requests,
so you cannot trace replies there - the simple solution is to do
sip_trace + setflag(trace) and (using t_relay()) you should get
traced all traffic.
In comparison with sip_trace + setflag(trace), what would be missing
for the trace log if only setflag(trace) is used? I'm using t_relay.
And if both sip_trace + setflag(trace) are used, will some messages
appear twice in the trace log?
Sadly, right now I can't test these things because my db_text is not
working correctly (it only writes on UDP connections.)
Regards,
Brian
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Re: [OpenSIPS-Users] compile deb files 1.6.1
Hi Jan, fixed on SVN and tarballs updated. Thanks and regards, Bogdan Jan D. wrote: Bogdan, Today I compiled the deb files (debian unstable) for 1.6.1. The version number somewhere is still 1.6.0. I tried svn to, without result. This are my deb files after make proper, make deb: Dec 22 09:51 opensips_1.6.0-0_amd64.changes Dec 22 09:51 opensips_1.6.0-0_amd64.deb Dec 22 09:47 opensips_1.6.0-0.dsc Dec 22 09:47 opensips_1.6.0-0.tar.gz Dec 22 09:51 opensips-1.6.1-notls Dec 21 21:39 opensips-1.6.1-notls_src.tar.gz Dec 22 09:51 opensips-b2bua-module_1.6.0-0_amd64.deb Dec 22 09:51 opensips-berkeley-module_1.6.0-0_amd64.deb Dec 22 09:51 opensips-carrierroute-module_1.6.0-0_amd64.deb Dec 22 09:51 opensips-console_1.6.0-0_amd64.deb Dec 22 09:51 opensips-cpl-module_1.6.0-0_amd64.deb Dec 22 09:51 opensips-dbg_1.6.0-0_amd64.deb Dec 22 09:51 opensips-dbhttp-module_1.6.0-0_amd64.deb Dec 22 09:51 opensips-geoip-module_1.6.0-0_amd64.deb Dec 22 09:51 opensips-identity-module_1.6.0-0_amd64.deb Dec 22 09:51 opensips-jabber-module_1.6.0-0_amd64.deb Dec 22 09:51 opensips-ldap-modules_1.6.0-0_amd64.deb Dec 22 09:51 opensips-mysql-module_1.6.0-0_amd64.deb Dec 22 09:51 opensips-perl-modules_1.6.0-0_amd64.deb Dec 22 09:51 opensips-postgres-module_1.6.0-0_amd64.deb Dec 22 09:51 opensips-presence-modules_1.6.0-0_amd64.deb Dec 22 09:51 opensips-radius-modules_1.6.0-0_amd64.deb Dec 22 09:51 opensips-regex-module_1.6.0-0_amd64.deb Dec 22 09:51 opensips-snmpstats-module_1.6.0-0_amd64.deb Dec 22 09:51 opensips-unixodbc-module_1.6.0-0_amd64.deb Dec 22 09:51 opensips-xmlrpc-module_1.6.0-0_amd64.deb Dec 22 09:51 opensips-xmpp-module_1.6.0-0_amd64.deb Can you update the source or tell me what to change to compile a correct deb file? Jan -- Bogdan-Andrei Iancu www.voice-system.ro ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Reducing INVITE delay until UAC sees 180 RINGING
Hello Bogdan, An mar., déc 22, 2009, Bogdan-Andrei Iancu schrieb: opensipsl...@encambio.com wrote: An ven., déc 18, 2009, opensipsl...@encambio.com schrieb: I'm assuming that a delay of 1 second is reachable through config tuning, and wondering as well what kind of result (in terms of delay time) the maximum tuning yields. Already I've looked at things like: disable_dns_blacklist disable_dns_failover ...because DNS resolutions is a logical place to start reducing. Any ideas? Is there documentation about this? The OpenSIPS documentation is excellent of course, but sadly lacking information about such core parameters as those above and 'rev_dns' relating to when and how often they effect performance. For example, does 'rev_dns = yes' mean that a reverse lookup takes place once at startup, every time the route script is entered, once a new transaction begins, dialog... rev_dns affects two scenarios: - comparisons with domains in script ( like src_ip==foo.bar) - reverse DNS will be done on foo.bar - internal checks of IPs against SIP uri / SIP hosts (like checking for VIA matching, for src IP nat, etc) Shortly, if when doing IP versus IP/domain checks, if reverse DNS should be done if domain That seems rather expensive, resolving all but what is necessary as the script is running for each incoming message. I guess that's why it is disactivated by default. I'll keep trying to find the sticking points leading to several seconds of delay before 180 ringing. The alias table is db_text, so maybe that is what is so slow. No enum is looked up. Regards, Brian ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] opensips 1.6.1 crashes on NOTIFY?
What about this one:
Program terminated with signal 11, Segmentation fault.
[New process 15892]
#0 0x00e1fddb in t_lookup_request (p_msg=0x81d2f20, leave_new_locked=1) at
../../mem/../hash_func.h:65
65 for (p=s2-s; p=(end-4); p+=4){
(gdb) bt
#0 0x00e1fddb in t_lookup_request (p_msg=0x81d2f20, leave_new_locked=1) at
../../mem/../hash_func.h:65
#1 0x00e21859 in t_newtran (p_msg=0x81d2f20) at t_lookup.c:1051
#2 0x00e243a0 in w_t_newtran (p_msg=0x81d2f20, foo=0x0, bar=0x0) at
tm.c:1006
#3 0x080545dd in do_action (a=0x81cc30c, msg=0x81d2f20) at action.c:967
#4 0x08057308 in run_action_list (a=0x81cc30c, msg=0x81d2f20) at
action.c:139
#5 0x080af2be in eval_expr (e=0x81cc378, msg=0x81d2f20, val=0x0) at
route.c:1240
#6 0x080aed39 in eval_expr (e=0x81cc3a4, msg=0x81d2f20, val=0x0) at
route.c:1553
#7 0x080aeccf in eval_expr (e=0x81cc3d0, msg=0x81d2f20, val=0x0) at
route.c:1558
#8 0x080533c2 in do_action (a=0x81cc770, msg=0x81d2f20) at action.c:689
#9 0x08057308 in run_action_list (a=0x81cc770, msg=0x81d2f20) at
action.c:139
#10 0x080554a7 in do_action (a=0x81c85b4, msg=0x81d2f20) at action.c:119
#11 0x08057308 in run_action_list (a=0x81c85b4, msg=0x81d2f20) at
action.c:139
#12 0x080554dd in do_action (a=0x81c868c, msg=0x81d2f20) at action.c:706
#13 0x08057308 in run_action_list (a=0x81c868c, msg=0x81d2f20) at
action.c:139
#14 0x08056625 in do_action (a=0x81c86f8, msg=0x81d2f20) at action.c:712
#15 0x08057308 in run_action_list (a=0x81c86f8, msg=0x81d2f20) at
action.c:139
#16 0x08056625 in do_action (a=0x81c8764, msg=0x81d2f20) at action.c:712
#17 0x08057308 in run_action_list (a=0x81c8764, msg=0x81d2f20) at
action.c:139
#18 0x08056625 in do_action (a=0x81c87d0, msg=0x81d2f20) at action.c:712
#19 0x08057308 in run_action_list (a=0x81c87d0, msg=0x81d2f20) at
action.c:139
#20 0x08056625 in do_action (a=0x81c883c, msg=0x81d2f20) at action.c:712
#21 0x08057308 in run_action_list (a=0x81c883c, msg=0x81d2f20) at
action.c:139
#22 0x08056625 in do_action (a=0x81c88a8, msg=0x81d2f20) at action.c:712
#23 0x08057308 in run_action_list (a=0x81c88a8, msg=0x81d2f20) at
action.c:139
#24 0x080554dd in do_action (a=0x81ca360, msg=0x81d2f20) at action.c:706
#25 0x08057308 in run_action_list (a=0x81bd578, msg=0x81d2f20) at
action.c:139
#26 0x080576a3 in run_top_route (a=0x81bd578, msg=0x81d2f20) at action.c:119
#27 0x0809ddf2 in receive_msg (
buf=0x8192380 NOTIFY sip:62.117.120.98 SIP/2.0\r\nVia: SIP/2.0/UDP
194.190.163.139:5061;branch=z9hG4bK-d5a4f117\r\nFrom: 206401
sip:206...@62.117.120.98
sip%3a206...@62.117.120.98;tag=d825811556491d55o0\r\nTo:
sip:62.117.120.98\r\nCall-ID: d42b6..., len=347, rcv_info=0xbfd71e84) at
receive.c:162
#28 0x080e5056 in udp_rcv_loop () at udp_server.c:492
#29 0x08070adf in main (argc=3, argv=0xbfd72094) at main.c:821
22 декабря 2009 г. 13:10 пользователь Anca Vamanu a...@opensips.orgнаписал:
Hi Alexander,
Unless you modified the sources, this is not the right backtrace. The
line numbers do not correspond with the ones in the trace.
Regards,
--
Anca Vamanu
www.voice-system.ro
Alexander wrote:
Oh, found one. Seems to be right core file. GDB says:
#0 0x080fbb52 in parse_params (_s=0xec, _c=695, _h=0x81d44bc,
_p=0x1d4) at parser/../trim.h:61
#1 0x080f135f in parse_msg (buf=0xb61eacc4 э\035\bп╛\036╤,
len=135861088, msg=0x305) at parser/msg_parser.c:567
#2 0x080ed9c7 in aaa_prot_bind (aaa_url=0xb61eacac, prot=0x80) at
aaa/aaa.c:85
#3 0x003b9205 in ?? ()
#4 0xb61eacac in ?? ()
#5 0x0080 in ?? ()
#6 0x003e2df4 in ?? ()
#7 0x371f3654 in ?? ()
#8 0x0007 in ?? ()
#9 0x08180e85 in _tr_buffer ()
#10 0x08180e81 in _tr_buffer ()
#11 0x in ?? ()
2009/12/22 Alexander goa...@gmail.com mailto:goa...@gmail.com
I have no core file for now:
Dec 22 11:02:08 srv opensips[26182]: INFO:core:handle_sigs: core
was not generated
Strange - ulimit -c unlimited and calls to setrlimit() in
OpenSIPS produce no core file.
NOTIFY packets come from clients. Also, Opensips sometimes sends
keepalive NOTIFY packets, but my route(5) is called inside uri ==
myself section.
2009/12/22 Anca Vamanu a...@opensips.org mailto:a...@opensips.org
Hi Alexander,
Can you please investigate the core with gdb and print here
the output.
It seems awkward to me that you expect to receive Notifies and
reply to
them. Wat kind of notifies are those? Sent by clients or the
presence
server?
Regards,
Anca
Alexander wrote:
Hi all.
I've tried to update to Opensips 1.6.1, but encountered the
following problem. Opensips starts successfully, but soon
almost all
it's processes die one by one and only two processes remain.
For example, if right after start we have:
Re: [OpenSIPS-Users] Reducing INVITE delay until UAC sees 180 RINGING
Hello Bogdan, An mar., déc 22, 2009, Bogdan-Andrei Iancu schrieb: First of all you should really try to see where the delay comes from: make a trace with time stamps and see if the delay is because of INVITE processing (time diff between inbound and outbound INVITE), because of the callee reply (time diff between outbound INVITE and inbound 180 reply)...etc... Yes that is my plan. There are just a few problems: - A tcpdump reveals no plaintext, as TLS is used * The sip_trace is not written because db_text only writes UDP Time stamps among all UACs and the OpenSIPS proxy must match ...so I'm trying to solve these things one by one before doing the fine grained time delay analysis. * Not sure why db_text is not writing files. A truss dump (with OpenSIPS running unforked) shows that the table files are being opened read only. I think this is normal because on startup their table headers are read in. However, the truss dump never includes another open (neither RO nor RW.) Still trying to debug this. Once we find the delay point, we can see why. Thanks. Regards, Brian ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] compile deb files 1.6.1
Bogdan, Thanks for the quick response, I ran the 'make deb' again, now the version number of the deb file is OK, but there is still seems to be a problem with a directory (install on a clean system): dpkg -i opensips_1.6.1-0_amd64.deb opensips-mysql-module_1.6.1.0_amd64.deb Unpacking opensips (from opensips_1.6.1-0_amd64.deb) ... dpkg: error processing opensips-mysql-module_1.6.1.0_amd64.deb (--install): cannot access archive: No such file or directory Setting up opensips (1.6.1-0) ... OpenSIPS not yet configured. Edit /etc/default/opensips first. Processing triggers for man-db ... Errors were encountered while processing: opensips-mysql-module_1.6.1.0_amd64.deb Any clue? Jan -- View this message in context: http://n2.nabble.com/compile-deb-files-1-6-1-tp4202644p4203096.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] CRITICAL:core:sig_alarm_abort: BUG - shutdown timeout triggered, dying...
I've sent you all the files on private e-mail. Hope that is not a problem. Thanks, Josip -Original Message- From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu Sent: Tuesday, December 22, 2009 12:04 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] CRITICAL:core:sig_alarm_abort: BUG - shutdown timeout triggered, dying... Hi Josip, Please send me the sipp scripts and how exactly to use it (I know how to use sipp) in order to get your scenario. Regards, Bogdan Josip Djuricic wrote: Hi Bogdan, yes, I have noticed in the trunk I've tested that even though registrations expires, and opensips removes it, and opensipsctl moni shows them under: usrloc:location-expires, and usrloc:location-contacts keeps getting lower by those usrloc:location-expires numbers. But usrloc:registered_users and usrloc:location_users keep rising constantly. And the memory usage also keeps rising constantly until I get out of shared memory. This only seams to happen when contact's expires, if re register comes in before timeout expires the memory stops rising. I've tested this with sipp and 100 000 users registering with 20 minutes expiration time, and 40-60 registrations per second. Going on and on...after few ours, no more shared memory available. I can send you any other detail you require? Also I can send you config and sipp scenario file? Best regards, Josip -Original Message- From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu Sent: Tuesday, December 22, 2009 10:28 AM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] CRITICAL:core:sig_alarm_abort: BUG - shutdown timeout triggered, dying... Hi Josip, So, what you are saying is that trunk has a problem (versus 1.6) - some records do not expire . Could you detail a bit the way I could reproduce this scenario ? Known bugs are listed on the tracker - http://www.opensips.org/Development/Tracker Regards, Bogdan Josip Djuricic wrote: Hi Bogdan, I've had to swithch to v1.6 stable, so It's working now :) What I notice is that on trunk version I had this UsrLoc Stats: usrloc:registered_users = 387432 usrloc:location-users = 387432 usrloc:location-contacts = 12005 usrloc:location-expires = 375427 but on stable 1.6 I have this: UsrLoc Stats: usrloc:registered_users = 12005 usrloc:location-users = 12005 usrloc:location-contacts = 12005 usrloc:location-expires = 375427 And I can confirm that memory is now stable, I think it seg faulted because at that ime it has gone 10 times trough 10users registration, what means usrloc:registered_users had more than 1 000 000 users, that could explain what happened. Somehow I think it was not clearing registered users no matter they expired and was deleted from db. Perhaps you can confirm that you can reproduce this problem? Also is there a possibility to get list of known limitations or perhaps bugs on v1.6 that I should be aware of (concerning stability issues before puttying the system in production use)? I know you mentioned release 1.6.1, so what should be important fixes you mentioned in that mail? Once again sorry for lot of questions. Thanks, Josip -Original Message- From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu Sent: Friday, December 18, 2009 1:26 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] CRITICAL:core:sig_alarm_abort: BUG - shutdown timeout triggered, dying... Hi Josip, A key question - how many records do you have in usrloc? I'm asking because opensips is flushing the usrloc at shutdown and if you have too many records, this will take some time. Also, the shutdown time is control by an alarm (couple of seconds), so if the shutdown takes too long, the alarm will simply kill opensips. Regards, Bogdan Josip Djuricic wrote: Hi, this is what happened tonight on trunk version of opensips. Any ideas? This is from log, I'm including backtrace also: ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] multiple Via headers separated by comma
I have attached it, it's from tcpdump, If you want I can catch it with some other method? Best regards, Josip Session Initiation Protocol Request-Line: BYE sip:xxx.xxx.xxx.43:5060;transport=UDP SIP/2.0 Method: BYE Request-URI: sip:xxx.xxx.xxx.43:5060;transport=UDP Request-URI Host Part: xxx.xxx.xxx.43 Request-URI Host Port: 5060 [Resent Packet: False] Message Header Via: SIP/2.0/UDP xxx.xxx.xxx.137:5;branch=z9hG4bK390a025bb61a5f010a79a7f549f2d743;rport Transport: UDP Sent-by Address: xxx.xxx.xxx.137 Sent-by port: 5 Branch: z9hG4bK390a025bb61a5f010a79a7f549f2d743 RPort: rport Max-Forwards: 70 From: sip:48521230...@xxx.xxx.xxx.137;tag=3776893de6f572f632b05e83485f9dd2 SIP from address: sip:48521230...@xxx.xxx.xxx.137 SIP from address User Part: 48521230886 SIP from address Host Part: xxx.xxx.xxx.137 SIP tag: 3776893de6f572f632b05e83485f9dd2 To: sip:48521289...@xxx.xxx.xxx.137;tag=6774SIPpTag011 SIP to address: sip:48521289...@xxx.xxx.xxx.137 SIP to address User Part: 48521289383 SIP to address Host Part: xxx.xxx.xxx.137 SIP tag: 6774SIPpTag011 Call-ID: 1-3...@xxx.xxx.xxx.41-b2b_1 CSeq: 201 BYE Sequence Number: 201 Method: BYE Contact: Anonymous sip:48521230...@xxx.xxx.xxx.137:5 Contact Binding: Anonymous sip:48521230...@xxx.xxx.xxx.137:5 URI: Anonymous sip:48521230...@xxx.xxx.xxx.137:5 SIP Display info: Anonymous SIP contact address: sip:48521230...@xxx.xxx.xxx.137:5 User-Agent: Voljatel B2BUA (RADIUS) cisco-GUID: 415033116-94487149-3088500870-2308033284 [Expert Info (Note/Undecoded): Unrecognised SIP header (cisco-GUID)] [Message: Unrecognised SIP header (cisco-GUID)] [Severity level: Note] [Group: Undecoded] h323-conf-id: 415033116-94487149-3088500870-2308033284 [Expert Info (Note/Undecoded): Unrecognised SIP header (h323-conf-id)] [Message: Unrecognised SIP header (h323-conf-id)] [Severity level: Note] [Group: Undecoded] Content-Length: 0 Session Initiation Protocol Status-Line: SIP/2.0 200 OK Status-Code: 200 [Resent Packet: False] Message Header To: sip:48521289...@xxx.xxx.xxx.137;tag=6774SIPpTag011 SIP to address: sip:48521289...@xxx.xxx.xxx.137 SIP to address User Part: 48521289383 SIP to address Host Part: xxx.xxx.xxx.137 SIP tag: 6774SIPpTag011 From: sip:48521230...@xxx.xxx.xxx.137;tag=3776893de6f572f632b05e83485f9dd2 SIP from address: sip:48521230...@xxx.xxx.xxx.137 SIP from address User Part: 48521230886 SIP from address Host Part: xxx.xxx.xxx.137 SIP tag: 3776893de6f572f632b05e83485f9dd2 Call-ID: 1-3...@xxx.xxx.xxx.41-b2b_1 CSeq: 201 BYE Sequence Number: 201 Method: BYE Via: SIP/2.0/UDP xxx.xxx.xxx.137:5;branch=z9hG4bK390a025bb61a5f010a79a7f549f2d743;rport Transport: UDP Sent-by Address: xxx.xxx.xxx.137 Sent-by port: 5 Branch: z9hG4bK390a025bb61a5f010a79a7f549f2d743 RPort: rport Server: SIPP Content-Length: 0 -Original Message- From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu Sent: Tuesday, December 22, 2009 11:53 AM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] multiple Via headers separated by comma Hi Josip, Post both the BYE (sent out by opesips) and 200 OK (received by opensips) in plain text . Regards, Bogdan Josip Djuricic wrote: Hi Andrew, Thanks very much for your quick answer, I understand that by rfc it is completely valid. What I can't seem to find is why is my last 200 OK from uas not beeing matched against the BYE that opensips forwarded to uas. So after uas sends 200 OK, it keeps receiveing BYE until timeout occurs. It only happens with sipp, if using any other uac everything works as expected. Every other transaction is matched correctly. I'm includig siplog from that last message, with changed ip's. Perhaps you would see this problem more clearly? Thanks, Josip -Original Message- From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Andrew Pogrebennyk Sent: Monday, December 21, 2009 10:44 AM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] multiple Via headers separated by comma Josip Djuricic wrote: Transaction is not matched if request is sent with 2 or more
Re: [OpenSIPS-Users] multiple Via headers separated by comma
Also the opensips config is the same as the sent to your email address. Josip -Original Message- From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu Sent: Tuesday, December 22, 2009 11:53 AM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] multiple Via headers separated by comma Hi Josip, Post both the BYE (sent out by opesips) and 200 OK (received by opensips) in plain text . Regards, Bogdan Josip Djuricic wrote: Hi Andrew, Thanks very much for your quick answer, I understand that by rfc it is completely valid. What I can't seem to find is why is my last 200 OK from uas not beeing matched against the BYE that opensips forwarded to uas. So after uas sends 200 OK, it keeps receiveing BYE until timeout occurs. It only happens with sipp, if using any other uac everything works as expected. Every other transaction is matched correctly. I'm includig siplog from that last message, with changed ip's. Perhaps you would see this problem more clearly? Thanks, Josip -Original Message- From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Andrew Pogrebennyk Sent: Monday, December 21, 2009 10:44 AM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] multiple Via headers separated by comma Josip Djuricic wrote: Transaction is not matched if request is sent with 2 or more multiline via headers and response is received with via header in one line separated by comma? Josip, This is absolutely legal if multiple values are combined in one line separated by comma. Ccheck RFC 3261 for multiple header field values combining. Section 7.3. [H4.2] also specifies that multiple header fields of the same field name whose value is a comma-separated list can be combined into one header field. That applies to SIP as well, but the specific rule is different because of the different grammars. Specifically, any SIP header whose grammar is of the form header = header-name HCOLON header-value *(COMMA header-value) allows for combining header fields of the same name into a comma- separated list. The Contact header field allows a comma-separated list unless the header field value is *. Response is matched to request using branch parameter from uppermost Via header, so I don't know why RFC compliant implementation would have problems with response matching when Via header is combined. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Bogdan-Andrei Iancu www.voice-system.ro ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Gw module or dispatch authentication?
Hello Bogdan,
An mar., déc 22, 2009, Bogdan-Andrei Iancu schrieb:
opensipsl...@encambio.com wrote:
Has anyone had this problem of wanting to forward a INVITE to
a PSTN gateway, and not being able to authenticate?
Is carrierroute really what should be used in this case, or some
combination of modules... maybe uac for the authentication?
Take a look at the UAC module - it can do user auth -
http://www.opensips.org/html/docs/modules/devel/uac.html
you can use the attr field in the drouting module to store the
username and password required by that GW and dynamically inject
them into UAC.
Neiter online nor distribution documents mention this 'attr' field:
http://www.opensips.org/html/docs/modules/devel/drouting.html
opensips-1.6.1-tls.orig/modules/drouting/README
...but I believe you that it is just what is needed.
Another problem is that unlike the gw table, the dr_gateways table
doesn't provide a column for the host port number. Before you say
'just tack it on the address field with a : between them', please
note that this would not be compatible with db_text tables which
interpret the ':' as a field separator. This means the 'Table 1.2.
Sample dr_gateways records' in the docs mentioned above are flawed.
With 'dynamically inject them into UAC' I suppose you mean something
similar to:
modparam(uac,auth_realm_avp,$avp(s:realm))
modparam(uac,auth_username_avp,$avp(s:user))
modparam(uac,auth_password_avp,$avp(s:pass))
modparam(drouting, attrs_avp, $avp(dr_attrs))
route {
do_routing();
$avp(s:user) = $avp(dr_attrs); # avp parsing needed
$avp(s:pass) = $avp(dr_attrs); # avp parsing needed
uac_auth();
}
Is that the general idea?
Regards,
Brian
___
Users mailing list
Users@lists.opensips.org
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Re: [OpenSIPS-Users] External Routing
Check out db_http. You can do just about whatever you want with it: http://www.opensips.org/html/docs/modules/devel/db_http.html Word of caution. The exec module has a lot of overhead associated with it. -Brett On Tue, Dec 22, 2009 at 4:07 AM, Saeed Akhtar saeedakhtar@gmail.comwrote: Hi all, Is there any other application which can do routing for me. If I don't wana use OpenSIPS routing and rather want to direct OpenSIPS through some other application which can tell OpenSIPS to route calls. Can I do this? Is there any application already built there? if yes and there are more than 1 then which is the better choice?? Regards, Saeed Akhtar On Tue, Dec 15, 2009 at 6:23 PM, Bogdan-Andrei Iancu bog...@voice-system.ro wrote: Sorry, by bad!! I made a mistake in reading your email So, about RADIUS-driven routingThe only way to get back info from RADIUS server is via a AUTH request (ACC requests do not return data from RADIUS server). So, I suggest using 1.6 with the new aaa_radius module - you can build custom auth request and you process (from the script) the RADIUS answer: http://www.opensips.org/html/docs/modules/devel/aaa_radius.html#id228094 Regards, Bogdan Saeed Akhtar wrote: sorry but FS=? Regards, Saeed Akhtar On Tue, Dec 15, 2009 at 1:13 PM, Bogdan-Andrei Iancu bog...@voice-system.ro mailto:bog...@voice-system.ro wrote: Hi Saeed, Typically there are 2 ways to do it: 1) based on proxying - opensips sends the call to FS and let FS to send the call wherever it decides 2) based on redirect - opensips sends the call to FS , FS determines the destination which is send back in a 3xx reply to opensips; opensips uses the address to send the call forward. Regards, Bogdan Saeed Akhtar wrote: hi all, I have a little requirement. I'm using opensips with freeradius for AAA purposes. Now I want to use it as router of unknown destinations. I know I can use drouting module to do this, but I want freeradius to decide where to route my call. So is there any possibility that freeradius tell opensips where to route call? Regards, Saeed Akhtar ___ Users mailing list Users@lists.opensips.org mailto:Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Bogdan-Andrei Iancu www.voice-system.ro http://www.voice-system.ro ___ Users mailing list Users@lists.opensips.org mailto:Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Bogdan-Andrei Iancu www.voice-system.ro ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] External Routing
Hello Saeed, You can also use the OSP Module (http://www.opensips.org/html/docs/modules/devel/osp.html) to access an external OSP server for routing and CDR collection. You can download a free OSP server from http://www.transnexus.com/OSP%20Toolkit/Peering_Server/VoIP_Peering_Server.h tm Jim D. -Original Message- From: users-boun...@lists.opensips.org [mailto:users- boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu Sent: Tuesday, December 22, 2009 5:24 AM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] External Routing Hi Saeed, You may try : (1) the seas module with weSIP (http://www.wesip.com) if you want to use Java Servlets (2) the perl module that allows you run perl scripts (http://www.opensips.org/html/docs/modules/devel/perl.html) (3) the exec module that allows you to run an external application (http://www.opensips.org/html/docs/modules/devel/exec.html) (4) the new python module that allows you to run python scripts (http://www.opensips.org/html/docs/modules/devel/python.html) Regards, Bogdan Saeed Akhtar wrote: Hi all, Is there any other application which can do routing for me. If I don't wana use OpenSIPS routing and rather want to direct OpenSIPS through some other application which can tell OpenSIPS to route calls. Can I do this? Is there any application already built there? if yes and there are more than 1 then which is the better choice?? Regards, Saeed Akhtar On Tue, Dec 15, 2009 at 6:23 PM, Bogdan-Andrei Iancu bog...@voice-system.ro mailto:bog...@voice-system.ro wrote: Sorry, by bad!! I made a mistake in reading your email So, about RADIUS-driven routingThe only way to get back info from RADIUS server is via a AUTH request (ACC requests do not return data from RADIUS server). So, I suggest using 1.6 with the new aaa_radius module - you can build custom auth request and you process (from the script) the RADIUS answer: http://www.opensips.org/html/docs/modules/devel/aaa_radius.html#id22809 4 Regards, Bogdan Saeed Akhtar wrote: sorry but FS=? Regards, Saeed Akhtar On Tue, Dec 15, 2009 at 1:13 PM, Bogdan-Andrei Iancu bog...@voice-system.ro mailto:bog...@voice-system.ro mailto:bog...@voice-system.ro mailto:bog...@voice-system.ro wrote: Hi Saeed, Typically there are 2 ways to do it: 1) based on proxying - opensips sends the call to FS and let FS to send the call wherever it decides 2) based on redirect - opensips sends the call to FS , FS determines the destination which is send back in a 3xx reply to opensips; opensips uses the address to send the call forward. Regards, Bogdan Saeed Akhtar wrote: hi all, I have a little requirement. I'm using opensips with freeradius for AAA purposes. Now I want to use it as router of unknown destinations. I know I can use drouting module to do this, but I want freeradius to decide where to route my call. So is there any possibility that freeradius tell opensips where to route call? Regards, Saeed Akhtar -- Bogdan-Andrei Iancu www.voice-system.ro ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Reducing INVITE delay until UAC sees 180 RINGING
Hello osiris123d, An mar., déc 22, 2009, osiris123d schrieb: If you are using TLS and you would like to sniff capture data you can use Wireshark. With Wireshark you can import your TLS certificate and the encrypted data will be decrypted. Hope that helps one of your problems. Yes, I didn't know that since I usually use tcpdump alone. Thanks. Brian ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Register -- 200 ok expires
Hi If no Expires header is found in REGISTER message, will this parameter just set the default value to 1800sec or will it also send 200 OK, with Expires: 1800 ? modparam(registrar, default_expires, 1800) Thanks, Josip ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] cvs for mediaproxy and cdrtools
You need darcs for this. See the wiki of each project for more info -- Adrian ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] cvs for mediaproxy and cdrtools
Hello Adrian, I found out that the repository for cdrtools is http://devel.ag-projects.com/repositories/cdrtool But I still have no clue on which one is for mediaproxy, freeradius-xs, etc Thanks for your help Adrian Georgescu wrote: You need darcs for this. See the wiki of each project for more info -- Adrian ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] cvs for mediaproxy and cdrtools
Hi Gabriel, On 22/12/09 7:03 PM, Gabriel Bermudez wrote: Hello Adrian, I found out that the repository for cdrtools is http://devel.ag-projects.com/repositories/cdrtool But I still have no clue on which one is for mediaproxy, freeradius-xs, etc We just added the MediaProxy repository URL to the MediaProxy wiki (http://mediaproxy.ag-projects.com) you may now download the development version from the darcs repository: darcs get http://devel.ag-projects.com/repositories/mediaproxy Regards, -- Saúl Ibarra Corretgé AG Projects ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Dialog module and uac_auth
Hello list, A while ago it was clear that uac_auth is of limited utility, due to the SIP RFC which requires that each message has a unique cseq. Calling uac_auth from failure_route produced a new INVITE with a proxy-auth header that didn't have a new cseq however. Since the dialog module appeared, I'm wondering how if scripts can be tweaked to use uac_auth in a SIP RFC compliant way. ...or is it still true that doing uac_auth() in failure_route fills in the proxy-auth header of a INVITE message that has already expired its cseq (no longer valid in the dialog)? How can uac_auth be used with INVITE messages in a SIP RFC compliant way? Regards, Brian ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] CDRtool freeradius mysql error
Ok. I did a clean install of CDRTool 7.0.0 on a new virtual image of Debian 5.0.3. I installed freeradius via apt-get install freeradius-xs freeradius-xs-mysql my /etc/freeradius/sql.conf still looks exactly the same. I followed the INSTALL.txt file and did the following b. To automatically create a table for each calendar month radacctMM: cp /var/www/CDRTool/setup/radius/OpenSIPS/radius_accounting.conf /etc/freeradius/sql.conf Load the MySQL stored procedures that create the monthly tables: mysql -u root radius /var/www/CDRTool/setup/radius/OpenSIPS/radius_accounting.proc Here is what the radius database has in it mysql desc radacct; +-+--+--+-+-++ | Field | Type | Null | Key | Default | Extra | +-+--+--+-+-++ | RadAcctId | bigint(21) | NO | PRI | NULL | auto_increment | | AcctSessionId | varchar(255) | NO | MUL | || | AcctUniqueId| varchar(255) | NO | MUL | || | UserName| varchar(64) | NO | MUL | || | Realm | varchar(64) | YES | MUL | || | NASIPAddress| varchar(15) | NO | MUL | || | NASPortId | varchar(50) | NO | | || | NASPortType | varchar(255) | NO | | || | AcctStartTime | datetime | NO | MUL | -00-00 00:00:00 || | AcctStopTime| datetime | NO | MUL | -00-00 00:00:00 || | AcctSessionTime | int(12) | YES | | NULL || | AcctAuthentic | varchar(32) | YES | | NULL || | ConnectInfo_start | varchar(32) | YES | | NULL || | ConnectInfo_stop| varchar(32) | YES | | NULL || | AcctInputOctets | bigint(12) | YES | | NULL || | AcctOutputOctets| bigint(12) | YES | | NULL || | CalledStationId | varchar(50) | NO | MUL | || | CallingStationId| varchar(50) | NO | MUL | || | AcctTerminateCause | varchar(32) | NO | | || | ServiceType | varchar(32) | YES | | NULL || | ENUMtld | varchar(64) | YES | | NULL || | FramedIPAddress | varchar(15) | NO | | || | AcctStartDelay | int(12) | YES | | NULL || | AcctStopDelay | int(12) | YES | | NULL || | SipMethod | varchar(50) | NO | | || | SipResponseCode | smallint(5) unsigned | NO | | 0 || | SipToTag| varchar(128) | NO | | || | SipFromTag | varchar(128) | NO | | || | SipTranslatedRequestURI | varchar(255) | NO | MUL | || | SipUserAgents | varchar(255) | NO | | || | SipApplicationType | varchar(255) | NO | | || | SipCodecs | varchar(255) | NO | | || | SipRPID | varchar(255) | NO | | || | SipRPIDHeader | varchar(255) | NO | | || | SourceIP| varchar(255) | NO | MUL | || | SourcePort | varchar(255) | NO | | || | CanonicalURI| varchar(255) | NO | MUL | || | DelayTime | varchar(5) | NO | | || | Timestamp
Re: [OpenSIPS-Users] CDRtool freeradius mysql error
Also in the /var/www/CDRTool/setup/radius/radius_accounting.proc file at the very end you have some testing statements when I get into mysql and copy and paste one I get an error mysql -u root -pPASSWORD -h localhost radius Welcome to the MySQL monitor. Commands end with ; or \g. Your MySQL connection id is 150 Server version: 5.0.51a-24+lenny2-log (Debian) Type 'help;' or '\h' for help. Type '\c' to clear the buffer. mysql CALL insert_radacct_record ( - 'radius','3c3b5ff12bf2-m5udeydrj...@snom320-000413241247', '5af53194787eccf1', - 'a...@umts.ro', 'a...@umts.ro', '83.149.75.105', '5060', '2006-12-10 12:09:19', - '0', '0', '0', '0', 'sip:3...@umts.ro=3buser=3dphone', - 'sip:a...@umts.ro', '200', 'Sip-Session', '', '', '0', '0', '200', 'Invite', - 'sip:3...@vm01.dns-hosting.info', 'as5664a60b', '27qems1o2j', - '31208005169', '81.23.228.147', '5060', - 'sip:3...@vm01.dns-hosting.info', '', '', 'audio' - ); ERROR 1318 (42000): Incorrect number of arguments for PROCEDURE radius.insert_radacct_record; expected 34, got 32 mysql CALL insert_radacct_record ( - 'radius','46477473...@blink-41247', '5af53194787eccf1', - 'a...@umts.ro', 'a...@umts.ro', '83.149.75.105', '5060', '2009-12-14 12:09:19', - '0', '0', '0', '0', 'sip:3...@umts.ro', - 'sip:a...@umts.ro', '200', 'Sip-Session', '', '', '0', '0', '200', 'Invite', - 'sip:t...@umts.ro', 'a60bsss', 'qe222ms1o2j', - '208005169', '81.23.228.147', '5060', - 'sip:t...@umts.ro', '', '', 'chat' - ); ERROR 1318 (42000): Incorrect number of arguments for PROCEDURE radius.insert_radacct_record; expected 34, got 32 Adrian Georgescu wrote: Looking at your configuration you are not using the latest configuration files or you did not read the changelog that comes with each update to keep your previously installed versions up to date. For example the sql.conf you have does not match the ones form our latest packages, if you look at CDRTool latest sample configuration for Freaaradius the number of fields differ from yours. I am not able to tell what other things you did not update properly. Regarding the table auto-creation, the radcctMM table is created by the stored procedure that you are calling in sql.conf: accounting_start_query = \ CALL insert_radacct_record( \ 'radius', \ The latest stored procedure is found in CDRTool setup directory you must update that as well. -- Adrian ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- View this message in context: http://n2.nabble.com/CDRtool-freeradius-mysql-error-tp4200490p4205553.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Fedora package for 1.6.1
Hi all.
I compiled opensips-1.6.1 rpm package for fedora 12 and rawhide:
http://kojipkgs.fedoraproject.org/packages/opensips/1.6.1/1.fc12/
http://kojipkgs.fedoraproject.org/packages/opensips/1.6.1/1.fc13/
Also package was pushed to the updates-testing:
https://admin.fedoraproject.org/updates/opensips-1.6.1-1.fc12
Welcome to testing!
To bogdan: spec file changes from opensips upstream to fedora upstream
attached to mail. Apply changes, please.
--
WBR, John Khvatov
diff --git a/opensips.spec?revision=6456 b/opensips.spec
index eb36c18..7272cab 100644
--- a/opensips.spec?revision=6456
+++ b/opensips.spec
@@ -10,13 +10,11 @@
Summary: Open Source SIP Server
Name: opensips
Version: 1.6.1
-Release: 4%{?dist}
+Release: 1%{?dist}
License: GPLv2+
Group:System Environment/Daemons
Source0:
http://opensips.org/pub/%{name}/%{version}/src/%{name}-%{version}-tls_src.tar.gz
Source1: %{name}.sysconfig
-Patch1: opensips--init.patch
-Patch2: opensips--openssl10.patch
URL: http://opensips.org
BuildRequires: expat-devel
@@ -502,8 +500,6 @@ clients.
%prep
%setup -q -n %{name}-%{version}-tls
-%patch1 -p1
-%patch2 -p1
%build
LOCALBASE=/usr CFLAGS=%{optflags} %{__make} all %{?_smp_mflags} TLS=1 \
@@ -990,6 +986,10 @@ fi
%doc docdir/README.xmpp
%changelog
+* Thu Dec 22 2009 John Khvatov iva...@fedoraproject.org - 1.6.1-1:
+- Updated to 1.6.1
+- Dropped upstreamed patches
+
* Wed Nov 04 2009 John Khvatov iva...@fedoraproject.org - 1.6.0-4:
- Fixed typo: pia_mi to pua_mi in presence_xcapdiff dependencies
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Re: [OpenSIPS-Users] Need advice rtpproxy + nathelper on opensips
Hi Bogdan
i just make a test to see what happen with opensips + rtpproxy
ip phone 1(192.168.1.5), ip phone 2(192.168.1.9) opensips +
rtpproxy(192.168.1.248)
i use the default opensips.cfg and edit some lines:
loadmodule nathelper.so
modparam(nathelper, rtpproxy_sock, udp:localhost:2) route {.
--- On Tue, 12/22/09, Bogdan-Andrei Iancu bog...@voice-system.ro wrote:
From: Bogdan-Andrei Iancu bog...@voice-system.ro
Subject: Re: [OpenSIPS-Users] Need advice rtpproxy + nathelper on opensips
To: OpenSIPS users mailling list users@lists.opensips.org
Date: Tuesday, December 22, 2009, 3:50 AM
Hi Ha,
well, depends - do you want your opensips to talk to the outside world?
if no, you do not need a WAN address.
Regards,
Bogdan
ha do wrote:
Hi all
i install rtpproxy in same Opensips machine
Do i really need at least 1 Wan IP address for Opensips + rtpproxy work
Thank you
Ha`
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--
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www.voice-system.ro
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Re: [OpenSIPS-Users] Need advice rtpproxy + nathelper on opensips
Hi Bogdan
i just make a test to see what happen with opensips + rtpproxy
ip phone 1(192.168.1.9), ip phone 2(192.168.1.5), opensips +
rtpproxy(192.168.1.248)
i use the default opensips.cfg and edit some lines:
loadmodule nathelper.so
modparam(nathelper, rtpproxy_sock, udp:localhost:2)route {.
if(has_totag()){
if (is_method(BYE)){..
}else if (is_method(INVITE)){
force_rtp_proxy();
record_route();
}
}
.
}
when i make call call from IP phone 1 to IP phone 2, and media go directly from
ip phone 1 to ip phone 2
Media is not go through the rtpproxy
what should i do to force media go through the rtpproxy(just test)
1 more question on the flag:
from the alg.cfg
force_rtp_proxy(FAII), force_rtp_proxy(FAIE),
force_rtp_proxy(FAEI), force_rtp_proxy(FAEE),
i read on the nathelper module : the flag is Lower case - does it still or i
have to change to the flag to lower case
i run rtpproxy :
rtpproxy -l 192.168.1.248 -s udp:192.168.1.248:2 -d DBUG:LOG_LOCAL7 -F
[r...@localhost run]# ll
total 108
drwxr-xr-x 2 root root 4096 May 25 2008 console
drwxr-xr-x 2 root root 4096 Dec 22 17:35 dbus
-rw-r--r-- 1 root root 5 Dec 22 17:35 haldaemon.pid
-rw--- 1 root root 5 Dec 22 17:35 klogd.pid
-rw-r--r-- 1 root root 5 Dec 22 17:35 messagebus.pid
drwxr-xr-x 2 mysql mysql 4096 Dec 22 20:55 mysqld
drwxrwxr-x 2 root root 4096 Jun 15 2008 netreport
drwxr-xr-x 2 root root 4096 May 28 2008 pm
-rw-r--r-- 1 root root 5 Dec 23 17:30 rtpproxy.pid
drwxr-xr-x 2 root root 4096 Dec 18 16:35 setrans
-rw-r--r-- 1 root root 5 Dec 22 17:35 sshd.pid
-rw--- 1 root root 5 Dec 22 17:35 syslogd.pid
-rw-rw-r-- 1 root utmp 4992 Dec 23 15:56 utmp
and i get message :
Dec 23 16:35:37 localhost /usr/local/sbin/opensips[5459]: DBG:tm:t_newtran:
transaction on entrance=0x
Dec 23 16:35:37 localhost rtpproxy[5126]: DBUG:handle_command: received command
5459_4 D 5772c9303b3d7...@192.168.1.9 2adb6fd4a0d3fcb2 517a7cba18e3441a
Dec 23 16:35:37 localhost /usr/local/sbin/opensips[5459]:
DBG:core:parse_headers: flags=
Dec 23 16:35:37 localhost rtpproxy[5126]: INFO:handle_command: delete request
failed: session 5772c9303b3d7...@192.168.1.9, tags
2adb6fd4a0d3fcb2/517a7cba18e3441a not found
Dec 23 16:35:37 localhost /usr/local/sbin/opensips[5459]:
DBG:core:parse_headers: flags=78
Dec 23 16:35:37 localhost rtpproxy[5126]: DBUG:doreply: sending reply 5459_4
E8
Thank you
Ha
--- On Tue, 12/22/09, Bogdan-Andrei Iancu bog...@voice-system.ro wrote:
From: Bogdan-Andrei Iancu bog...@voice-system.ro
Subject: Re: [OpenSIPS-Users] Need advice rtpproxy + nathelper on opensips
To: OpenSIPS users mailling list users@lists.opensips.org
Date: Tuesday, December 22, 2009, 3:50 AM
Hi Ha,
well, depends - do you want your opensips to talk to the outside world?
if no, you do not need a WAN address.
Regards,
Bogdan
ha do wrote:
Hi all
i install rtpproxy in same Opensips machine
Do i really need at least 1 Wan IP address for Opensips + rtpproxy work
Thank you
Ha`
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--
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www.voice-system.ro
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Re: [OpenSIPS-Users] Need advice rtpproxy + nathelper on opensips
Hi Bogdan
please ignore :
and i get message :
Dec 23 16:35:37 localhost /usr/local/sbin/opensips[5459]: DBG:tm:t_newtran:
transaction on entrance=0x
Dec 23 16:35:37 localhost rtpproxy[5126]: DBUG:handle_command: received command
5459_4 D 5772c9303b3d7...@192.168.1.9 2adb6fd4a0d3fcb2 517a7cba18e3441a
Dec 23 16:35:37 localhost /usr/local/sbin/opensips[5459]:
DBG:core:parse_headers: flags=
Dec 23 16:35:37 localhost rtpproxy[5126]: INFO:handle_command: delete request
failed: session 5772c9303b3d7...@192.168.1.9, tags
2adb6fd4a0d3fcb2/517a7cba18e3441a not found
Dec 23 16:35:37 localhost /usr/local/sbin/opensips[5459]:
DBG:core:parse_headers: flags=78
Dec 23 16:35:37 localhost rtpproxy[5126]: DBUG:doreply: sending reply 5459_4
E8
because i change :
if (loose_route()) {
if (is_method(BYE)) {
unforce_rtp_proxy();
and i still need help on media + the flag
Thank you
Ha
--- On Tue, 12/22/09, ha do haloha...@yahoo.com wrote:
From: ha do haloha...@yahoo.com
Subject: Re: [OpenSIPS-Users] Need advice rtpproxy + nathelper on opensips
To: OpenSIPS users mailling list users@lists.opensips.org
Date: Tuesday, December 22, 2009, 8:38 PM
Hi Bogdan
i just make a test to see what happen with opensips + rtpproxy
ip phone 1(192.168.1.9), ip phone 2(192.168.1.5), opensips +
rtpproxy(192.168.1.248)
i use the default opensips.cfg and edit some lines:
loadmodule nathelper.so
modparam(nathelper, rtpproxy_sock, udp:localhost:2)route {.
if(has_totag()){
if (is_method(BYE)){..
}else if (is_method(INVITE)){
force_rtp_proxy();
record_route();
}
}
.
}
when i make call call from IP phone 1 to IP phone 2, and media go directly from
ip phone 1 to ip phone 2
Media is not go through the rtpproxy
what should i do to force media go through the rtpproxy(just test)
1 more question on the flag:
from the alg.cfg
force_rtp_proxy(FAII), force_rtp_proxy(FAIE),
force_rtp_proxy(FAEI), force_rtp_proxy(FAEE),
i read on the nathelper module : the flag is Lower case - does it still or i
have to change to the flag to lower case
i run
rtpproxy :
rtpproxy -l 192.168.1.248 -s udp:192.168.1.248:2 -d DBUG:LOG_LOCAL7 -F
[r...@localhost run]# ll
total 108
drwxr-xr-x 2 root root 4096 May 25 2008 console
drwxr-xr-x 2 root root 4096 Dec 22 17:35 dbus
-rw-r--r-- 1 root root 5 Dec 22 17:35 haldaemon.pid
-rw--- 1 root root 5 Dec 22 17:35 klogd.pid
-rw-r--r-- 1 root root 5 Dec 22 17:35 messagebus.pid
drwxr-xr-x 2 mysql mysql 4096 Dec 22 20:55 mysqld
drwxrwxr-x 2 root root 4096 Jun 15 2008 netreport
drwxr-xr-x 2 root root 4096 May 28 2008 pm
-rw-r--r-- 1 root root 5 Dec 23 17:30 rtpproxy.pid
drwxr-xr-x 2 root root 4096 Dec 18 16:35 setrans
-rw-r--r-- 1 root root 5 Dec 22 17:35 sshd.pid
-rw--- 1 root root 5 Dec 22 17:35 syslogd.pid
-rw-rw-r-- 1 root utmp 4992 Dec 23 15:56 utmp
and i get message :
Dec 23 16:35:37 localhost /usr/local/sbin/opensips[5459]: DBG:tm:t_newtran:
transaction on entrance=0x
Dec 23 16:35:37 localhost rtpproxy[5126]: DBUG:handle_command: received command
5459_4 D 5772c9303b3d7...@192.168.1.9 2adb6fd4a0d3fcb2 517a7cba18e3441a
Dec 23 16:35:37 localhost /usr/local/sbin/opensips[5459]:
DBG:core:parse_headers: flags=
Dec 23 16:35:37 localhost rtpproxy[5126]: INFO:handle_command: delete request
failed: session 5772c9303b3d7...@192.168.1.9, tags
2adb6fd4a0d3fcb2/517a7cba18e3441a not found
Dec 23 16:35:37 localhost /usr/local/sbin/opensips[5459]:
DBG:core:parse_headers: flags=78
Dec 23 16:35:37 localhost rtpproxy[5126]: DBUG:doreply: sending reply 5459_4
E8
Thank you
Ha
--- On Tue, 12/22/09, Bogdan-Andrei Iancu bog...@voice-system.ro wrote:
From: Bogdan-Andrei Iancu bog...@voice-system.ro
Subject: Re:
[OpenSIPS-Users] Need advice rtpproxy + nathelper on opensips
To: OpenSIPS users mailling list users@lists.opensips.org
Date: Tuesday, December 22, 2009, 3:50 AM
Hi Ha,
well, depends - do you want your opensips to talk to the outside world?
if no, you do not need a WAN address.
Regards,
Bogdan
ha do wrote:
Hi all
i install rtpproxy in same Opensips machine
Do i really need at least 1 Wan IP address for Opensips + rtpproxy work
Thank you
Ha`
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Re: [OpenSIPS-Users] CDRtool freeradius mysql error
I have some problem like this. Mediaproxy starts correctly but I coudnt find mediaproxy databases. -Thanks Darshak -Original Message- From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of osiris123d Sent: Wednesday, December 23, 2009 1:55 AM To: users@lists.opensips.org Subject: Re: [OpenSIPS-Users] CDRtool freeradius mysql error Also in the /var/www/CDRTool/setup/radius/radius_accounting.proc file at the very end you have some testing statements when I get into mysql and copy and paste one I get an error mysql -u root -pPASSWORD -h localhost radius Welcome to the MySQL monitor. Commands end with ; or \g. Your MySQL connection id is 150 Server version: 5.0.51a-24+lenny2-log (Debian) Type 'help;' or '\h' for help. Type '\c' to clear the buffer. mysql CALL insert_radacct_record ( - 'radius','3c3b5ff12bf2-m5udeydrj...@snom320-000413241247', '5af53194787eccf1', - 'a...@umts.ro', 'a...@umts.ro', '83.149.75.105', '5060', '2006-12-10 12:09:19', - '0', '0', '0', '0', 'sip:3...@umts.ro=3buser=3dphone', - 'sip:a...@umts.ro', '200', 'Sip-Session', '', '', '0', '0', '200', 'Invite', - 'sip:3...@vm01.dns-hosting.info', 'as5664a60b', '27qems1o2j', - '31208005169', '81.23.228.147', '5060', - 'sip:3...@vm01.dns-hosting.info', '', '', 'audio' - ); ERROR 1318 (42000): Incorrect number of arguments for PROCEDURE radius.insert_radacct_record; expected 34, got 32 mysql CALL insert_radacct_record ( - 'radius','46477473...@blink-41247', '5af53194787eccf1', - 'a...@umts.ro', 'a...@umts.ro', '83.149.75.105', '5060', '2009-12-14 12:09:19', - '0', '0', '0', '0', 'sip:3...@umts.ro', - 'sip:a...@umts.ro', '200', 'Sip-Session', '', '', '0', '0', '200', 'Invite', - 'sip:t...@umts.ro', 'a60bsss', 'qe222ms1o2j', - '208005169', '81.23.228.147', '5060', - 'sip:t...@umts.ro', '', '', 'chat' - ); ERROR 1318 (42000): Incorrect number of arguments for PROCEDURE radius.insert_radacct_record; expected 34, got 32 Adrian Georgescu wrote: Looking at your configuration you are not using the latest configuration files or you did not read the changelog that comes with each update to keep your previously installed versions up to date. For example the sql.conf you have does not match the ones form our latest packages, if you look at CDRTool latest sample configuration for Freaaradius the number of fields differ from yours. I am not able to tell what other things you did not update properly. Regarding the table auto-creation, the radcctMM table is created by the stored procedure that you are calling in sql.conf: accounting_start_query = \ CALL insert_radacct_record( \ 'radius', \ The latest stored procedure is found in CDRTool setup directory you must update that as well. -- Adrian ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- View this message in context: http://n2.nabble.com/CDRtool-freeradius-mysql-error-tp4200490p4205553.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] number of opensips children
Hello Bogdan, An ven., déc 18, 2009, Bogdan-Andrei Iancu schrieb: there are modules creating separate processes for handling additional events (like the mi_ modules). To see what processes you have and what they are doing, do: opensipsctl fifo ps You were right about that, what a surprise ;) # opensipsctl fifo ps Process:: ID=0 PID=24975 Type=attendant Process:: ID=1 PID=24977 Type=SIP receiver udp:123.234.210.1:5060 Process:: ID=2 PID=24978 Type=time_keeper Process:: ID=3 PID=24979 Type=timer Process:: ID=4 PID=24980 Type=MI FIFO That's what I see when turning TLS and TCP off and children=1. I'm guessing that having five processes is the minimum possible when using the mi_fifo module. I don't know if the 'time_keeper' and 'timer' processes are needed however, and how to remove them in the configuration if they are unneeded. I assume that the 'attendant' does no real work other than coordinating the other processes, right? My gut feeling is that having four UDP listening processes and four TCP listening processes is about right for us, because we only have a handful of UACs participating infrequently (5 calls per day.) Regards, Brian ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Disabling TCP or UDP (RFC forbidden)
Hello list, Somewhere I read that OpenSIPS 1.6.0 allows to disable TCP but not UDP because UDP is required by the SIP RFC, so I looked at the RFCs. RFC 2543 states that 'UDP and TCP should be implemented.' RFC 3261 states that 'UDP and TCP must be implemented.' It would seem intuitive that OpenSIPS developers force the OpenSIPS runtime to conform to the most recent RFC, meaning that neither UDP nor TCP listening is possibly disabled. ...or the developers allow the admin (config file editor) to disable either UDP or TCP. This could be because the developers want to support the oldest RFC for backwards compatibility for example, or because they want to allow the admin to make the decision themselves. But the current release of OpenSIPS does not allow UDP to be disabled, while allowing TCP to be disabled. Can somebody explain why this policy was chosen or what is wrong with my logic? Regards, Brian ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] External Routing
Hi Antonio, It seams that the new python module does not provide any documentation - I will open a bug on this. Regards, Bogdan Antonio Pardo wrote: Hi, El mar, 22-12-2009 a las 12:23 +0200, Bogdan-Andrei Iancu escribió: (4) the new python module that allows you to run python scripts (http://www.opensips.org/html/docs/modules/devel/python.html) this link returns a 404 HTTP error. Where can I find the documentation of this module? Ciao ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Bogdan-Andrei Iancu www.voice-system.ro ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
