Re: [OpenSIPS-Users] Restricting CallerID
Hello Alex, I have exactly same problem, but have no idea how to do that with rpid credential. Could you please send any example configuration? thanks a lot Alex Balashov wrote / napísal(a): Andrew Yager wrote: On 09/03/2009, at 4:39 PM, Alex Balashov wrote: I suggest setting an rpid credential for these users in the subscriber table. This can be applied when the outbound call from the user is proxy-challenged. Thanks. The reason why I don't like this solution is that (I think) it will restrict them to a single outbound callerid, where we would like them to be able to set any in their range. Am I wrong in my assumption? No, you are correct. If that's the case, you're just going to have to put in some logic to evaluate whether the caller ID they have chosen (the RPID, or the From URI user part depending on how they're sending it to you) falls within one of their ranges. In order to avoid making a mess of the OpenSIPS script, I suggest a database stored procedure. :-) -- I'll carry your books, I'll carry a tune, I'll carry on, carry over, carry forward, Cary Grant, cash carry, Carry Me Back To Old Virginia, I'll even Hara Kari if you show me how, but I will *not* carry a gun. -- Hawkeye, M*A*S*H ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] [OpenSIPS-Devel] dialog module: dialog not cleaned up when BYE is sent
Hi Bogdan, the issue is resolved when using the DID for dialog matching, but if I use dlg_match_mode 2, the error still occurs. Is there a problem with the matching of messages to dialogs if there is no DID present? Best Regards, Herman Bogdan-Andrei Iancu wrote: Hi Herman, Thanks for the files you sent me. I found out that the problem was in record_route_preset() - the function was simply ignoring whatever parameter you were adding from script or from other modules. This is why the DIALOG module was not recognizing the sequential requests. The fix is available on 1.5 - if you could give it another try and let me know, it will be great. Regards, Bogdan Herman Bastiaens wrote: Hi Bogdan, x The problem only occurs: - if there is another server in the loop, if I test with a client that's registered to xx (instead of the demo server), the issue does not occur - if I record_route_preset instead of record_route. As you'll see, when I record_route_preset, there is no did present in the route header. I believe it should fall back to default dialog matching, but perhaps there's a problem with this mechanism? Hope we can sort this out. Thanks for your replies! Bogdan-Andrei Iancu wrote: Hi Herman, Sorry for the mixing :). Please get the logs (with debug=6) and the SIP trace (ngrep) for such a call and sent them to me (you can send them off-list if larger than 40K). I cannot make any assumption yet without first looking at the logs. Thanks and regards, Bogdan Herman Bastiaens wrote: Bogdan, I didn't post the log on the forum, that was someone else. I just added my comment to the thread because it seemed like the same problem. I can get a log of the scenario tomorrow if that's helpful. Do you have any idea why the record_route_preset could by messing up this scenario? Do you have any idea what could explain the 5 second threshold? Thanks a lot for your replies, hope we can sort this out. Herman, The log you posted on the forum did not show any usage of loose_route() - if you look on the log, for ACK and BYE there is no mesage from rr or dialog module. Can you confirm this? Regards, Bogdan Herman Bastiaens wrote: Hi Bogdan, I'm pretty sure I do a loose_route for the ACK and BYE, but I'm still seeing this error. I've tested the most basic scenario, starting from an example, and this problem seems to occur when I start using a record_route_preset(...). Perhaps the dialog module can't handle this? (just to be clear, the bug is still only occurring if the call is shut down during the first few seconds after the ACK). I've attached the script in which I see the error occurring. The IP of my server is 172.17.10.44 Hi Herman, And the bell rang! :) I went over the logs you posted on the forum and I noticed (both script and logs) that you are not using loose_route() for sequential requests. You do record_route() for the initial INVITE, but no loose_route for ACK, BYE. And loose_route() is the function that updates the dialog state. So, in your case, the dialog does not see the ACK and BYE and still keeps in the CONFIRMED_NA (not acknowledged) state. This is way it is not removed. See the default opensips.cfg file to see how to use the loose_route(). I beat it will work after that ;) Regards, Bogdan Herman Bastiaens wrote: Hi Bodgan, that's what I'm seeing, time and time again. I was hoping this might ring a bell, but from your reply I take that it doesn't :-) Regards, Herman Hi Herman, just to copy the reply from the forum :) : So, let me see if I get it right. With the same configuration, if the call is longer than 5 secs, everything is ok (dialog is removed when receiving a BYE). But if the call is shorter than 5 secs, the dialog is not removed. Is this what you say? Regards, Bogdan Herman Bastiaens wrote: Hi, I've posted this problem on the forum (https://sourceforge.net/forum/message.php?msg_id=6595314), but it doesn't seem to be very active, so I'm posting it here as well. I'm having a problem with the dialog module of opensips 1.4.2-notls. When a call is set up, and released within five seconds, the dialog is not removed. I am sure the call is set up correctly (INVITE - 200 OK - ACK) and the BYE is sent (with the correct call-id, from and to tag), but the dialog is not removed. I do a record_route_preset () for the INVITE and a loose_route() for the BYE. Are there any timers, caching, ... that could explain this behavior? I have tested a number of times, and the problem only occurs if the call is shut down within the first five seconds, if the call is running longer, the dialog is cleaned up correctly when the BYE is sent. note: a dialog is inserted multiple times in the same profile, but with different values, I don't know if this is relevant for the issue
Re: [OpenSIPS-Users] About Routing Logic in opensips.cfg
Hi Cheng, I have the same problem as you, where a 600 Busy everywhere message is sent for the same setup as yours (OpenSIPS and OpenIMSCore but with a X-Lite client). I am using OpenSIPS as an application server and thus defined it in FHoSS with SPTs as SUBSCRIBE and PUBLISH for the SIP-methods. After seeing this thread I tried using the presence.cfg as the configuration file, but somehow the openSIPS just does not start with this config. I tried renaming it to opensips.cfg too. Since you say it worked for you how did you do this? Did you just replace the routing logic alone, leaving all other parts unchanged? Or did you replace the whole opensips.cfg file with presence.cfg? Any help regarding this would be great. Thanks in advance --maanasa Yinbo-cheng wrote: Hi, I want to install OpenSIPS as a presence server without privacy rules. I use OpenIMSCore as IMS components and uctimsclent as a IMS client. OpenSIPS is installed on the same machine with OpenIMSCore. I've modified opensips.cfg to enable presence, but the presence server doesn't work. IMS client got the following message: Status-Line: SIP/2.0 600 Busy everywhere - Forwarding to S-CSCF failed. While using the routing logic of presence.cfg ( in the test directory of source codes) to replace that of opensips.cfg, the presence server works well. I do not understand the routing logic in the configure file. Any one can tell me what is wrong with the routing logic in the default opensips.cfg and where can i find the help documents about routing logic. The two configure files are enclosed. Thanks, cheng ___ 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/About-Routing-Logic-in-opensips.cfg-tp2448035p2472662.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] [OpenSIPS-Devel] dialog module: dialog not cleaned up when BYE is sent
Hi Bogdan, yes, I see these messages in the log. You can also find them in the log I've sent you on monday (09-03). Because the DID was not present (due to the record_route_preset bug), the matching mode falls back to matching based on SIP elements. Best Regards, Herman Bogdan-Andrei Iancu wrote: Hi Herman, yes, that is correct. When you use mode 2 (no did), for ACK and BYE, do you get something like this in the logs: no dialog callid= ? Regards, Bogdan Herman Bastiaens wrote: Hi Bogdan, the issue is resolved when using the DID for dialog matching, but if I use dlg_match_mode 2, the error still occurs. Is there a problem with the matching of messages to dialogs if there is no DID present? Best Regards, Herman Bogdan-Andrei Iancu wrote: Hi Herman, Thanks for the files you sent me. I found out that the problem was in record_route_preset() - the function was simply ignoring whatever parameter you were adding from script or from other modules. This is why the DIALOG module was not recognizing the sequential requests. The fix is available on 1.5 - if you could give it another try and let me know, it will be great. Regards, Bogdan Herman Bastiaens wrote: Hi Bogdan, x The problem only occurs: - if there is another server in the loop, if I test with a client that's registered to xx (instead of the demo server), the issue does not occur - if I record_route_preset instead of record_route. As you'll see, when I record_route_preset, there is no did present in the route header. I believe it should fall back to default dialog matching, but perhaps there's a problem with this mechanism? Hope we can sort this out. Thanks for your replies! Bogdan-Andrei Iancu wrote: Hi Herman, Sorry for the mixing :). Please get the logs (with debug=6) and the SIP trace (ngrep) for such a call and sent them to me (you can send them off-list if larger than 40K). I cannot make any assumption yet without first looking at the logs. Thanks and regards, Bogdan Herman Bastiaens wrote: Bogdan, I didn't post the log on the forum, that was someone else. I just added my comment to the thread because it seemed like the same problem. I can get a log of the scenario tomorrow if that's helpful. Do you have any idea why the record_route_preset could by messing up this scenario? Do you have any idea what could explain the 5 second threshold? Thanks a lot for your replies, hope we can sort this out. Herman, The log you posted on the forum did not show any usage of loose_route() - if you look on the log, for ACK and BYE there is no mesage from rr or dialog module. Can you confirm this? Regards, Bogdan Herman Bastiaens wrote: Hi Bogdan, I'm pretty sure I do a loose_route for the ACK and BYE, but I'm still seeing this error. I've tested the most basic scenario, starting from an example, and this problem seems to occur when I start using a record_route_preset(...). Perhaps the dialog module can't handle this? (just to be clear, the bug is still only occurring if the call is shut down during the first few seconds after the ACK). I've attached the script in which I see the error occurring. The IP of my server is 172.17.10.44 Hi Herman, And the bell rang! :) I went over the logs you posted on the forum and I noticed (both script and logs) that you are not using loose_route() for sequential requests. You do record_route() for the initial INVITE, but no loose_route for ACK, BYE. And loose_route() is the function that updates the dialog state. So, in your case, the dialog does not see the ACK and BYE and still keeps in the CONFIRMED_NA (not acknowledged) state. This is way it is not removed. See the default opensips.cfg file to see how to use the loose_route(). I beat it will work after that ;) Regards, Bogdan Herman Bastiaens wrote: Hi Bodgan, that's what I'm seeing, time and time again. I was hoping this might ring a bell, but from your reply I take that it doesn't :-) Regards, Herman Hi Herman, just to copy the reply from the forum :) : So, let me see if I get it right. With the same configuration, if the call is longer than 5 secs, everything is ok (dialog is removed when receiving a BYE). But if the call is shorter than 5 secs, the dialog is not removed. Is this what you say? Regards, Bogdan Herman Bastiaens wrote: Hi, I've posted this problem on the forum (https://sourceforge.net/forum/message.php?msg_id=6595314), but it doesn't seem to be very active, so I'm posting it here as well. I'm having a problem with the dialog module of opensips 1.4.2-notls. When a call is set up, and released within five seconds, the dialog is not removed. I am sure the call is set up correctly (INVITE - 200 OK - ACK) and the BYE is sent (with the correct call-id, from and to tag), but the dialog is not removed. I do a record_route_preset () for the INVITE and a
Re: [OpenSIPS-Users] OpenSipS + CentOS: No debug logs!
Hi Noel, How do you set the debug level? in the script (debug=6) or as command line param (-d) ? Also, where is your logging done? console (standard error) or syslog ? Regards, Bogdan Noel R. Morais wrote: I've installed it in another CentOS machine and didn't work . I'm lost ... haha -Noel On Fri, Mar 13, 2009 at 4:48 PM, Jeff Pyle jp...@fidelityvoice.com wrote: Well, I'm not sure how to tell. I can tell you that the .svnrevision file has 2:5412M in it, and I last updated on March 9 (Monday). - Jeff On 3/13/09 3:42 PM, Noel R. Morais noelro...@gmail.com wrote: What is the trunk revision that u are using? Noel On Fri, Mar 13, 2009 at 4:28 PM, Jeff Pyle jp...@fidelityvoice.com wrote: Noel, I'm running on CentOS 5.2 i386 and I'm able to see DBG messages. I didn't do anything special to have them show. This is with a current update of Opensips 1.5. - Jeff On 3/13/09 3:23 PM, Noel R. Morais noelro...@gmail.com wrote: Hi Guys, something weird is happening here. I cannot get module DBG logs when using CentOS 5.2. Even if I set the debug parameter to 9 it doesn't show me any debug message. Only CRITICAL, INFO, WARNING and ERROR messages. I've made the same installation in a Fedora Core 5 and it worked like a charm. Is there some dependence or any other tweak to make it work in a CentOS? Thanks, Noel ___ 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] Retrieve headers from negative reply
Hi Jeff, Do you get this crash all the time when using the get_redirect() function ? Regards, Bogdan Jeff Pyle wrote: Hello, Is there any way to retrieve headers from a response, rather than the request? I need the Contact header out of a 302 response. Both $hdr(Contact) and $ct give me the contact of the request, not the response. I tried get_redirects(), but because of the formatting of the contact response it crashes Opensips: CRITICAL:core:qm_free: freeing already freed pointer, first free: parser/msg_parser.c: set_ruri(712) - aborting INFO:core:handle_sigs: child process 3554 exited by a signal 6 INFO:core:handle_sigs: core was not generated INFO:core:handle_sigs: terminating due to SIGCHLD Get_redirects() was a last resort anyway because I really just need a value from the Contact, I don’t want to use it for routing. Thanks, Jeff ___ 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 string transformations
Hi Jeff,
the correct format is the first one:
$(var(xyz){param.value,peer}{s.unescape.user})
but you mentioned errors for it, so these are startup or runtime errors?
can you post them?
Regards,
Bogdan
Jeff Pyle wrote:
Hello,
What is the proper way to apply multiple string transformations at the same
time?
For example, let's say I want to extract the {param.value,peer} from
$var(xyz) and then do a {s.unescape.user} on it? All the combinations I've
tried have errored out:
$(var(xyz){param.value,peer}{s.unescape.user})
$(var(xyz){param.value,peer}){s.unescape.user}
... and more failures.
Thanks,
Jeff
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Re: [OpenSIPS-Users] SIP trunk provider lab
Hi Elnor, h...@elnour.biz wrote: Hi, thank you for your response! correct me if I'm wrong, but aren't we going to loose dest phone number if we use dbaliases? Not necessary - depends of how you define the alias. For ex, you can do: d...@server - DID@ trunk preserve the username part and change only the domain to point to the trunk. Also, if you have blocks of DIDs which are easy to detect based on regexp, you may consider using the dialplan module: http://www.opensips.org/html/docs/modules/1.4.x/dialplan.html See example 1.4.1.2 - http://www.opensips.org/html/docs/modules/1.4.x/dialplan.html#id227187 Regards, Bogdan Also please provide hints on ENUM, how do I use it for this purpose? Thank you. Elnour - Original Message - From: Iñaki Baz Castillo i...@aliax.net To: users@lists.opensips.org Sent: Saturday, March 14, 2009 3:59 PM Subject: Re: [OpenSIPS-Users] SIP trunk provider lab El Sábado, 14 de Marzo de 2009, h...@elnour.biz escribió: if we have a SIP subscriber registered with a user name companyA which has 8 DID phone lines from PSTN say 5551001-08, our setup should dynamicaly route all incomming calls to the location of the regisration. Is this possbile? I mean is this possible to achive dynamicaly? Of course, use ENUM or dbaliases. -- Iñaki Baz Castillo ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users !DSPAM:49bb9c46608611888415445! ___ 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] Register aliasing
Hi Romanov, Romanov Vladimir wrote: Hi! We want implement next scheme. From user side: 1) Every user has two usernames – login billingid. Passwords are same for both. 2) User can login using both names. Just for sample user can login form PC using one name and from mobile device using other name. All user registration must be same. 3) Other users can call this user using both names. For 3) we can use alias_db module. For 1) 2) we can do some aliasing in register messages. And convert all registration to one form (billing id for sample). For this purpose we must write simple module. During registration, you can force a different AOR (than the one from SIP message - TO hdr) for saving the contact: in 1.5.x : http://www.opensips.org/html/docs/modules/1.5.x/registrar.html#id271106 in 1.4.x: http://www.opensips.org/html/docs/modules/1.4.x/registrar.html#id228309 Regards, Bogdan Maybe there exist better solution? - Vladimir Romanov Scartel Star Lab CTO +7 (960) 239-0853 ___ 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] Double Registration transaction
Hi,
Why do you say is executed twice?? Maybe you mistake the 2 REGISTERs
because of the authentication process... REGISTER + 401 ; REGISTER + 200 OK
Regards,
Bogdan
bay2x1 wrote:
I tried creating my own registration configuration. Using the ngrep tool I
was able to capture the header packets. I am wondering why the route(1) is
executed twice everytime that a client registers. This is the code I am
using for registration.
route{
if(!mf_process_maxfwd_header(10)){
sl_send_reply(483, Too Many Hops);
exit;
};
if(method=='REGISTER'){
route(1);
}
}
route[1] {
if(is_uri_host_local()) {
xlog(Registration Process starts here! \n);
if(!www_authorize(, subscriber))
{else if (method=='INVITE') {
route(2);
};
www_challenge(,1);
xlog(Registration Process ends here! \n);
exit;
};
if(!check_to()) {
sl_send_reply(403, Forbidden);
};
save(location);
exit;
} else if {
sl_send_reply(403, Forbidden);
};
}
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[OpenSIPS-Users] SUSPECT: Re: SUSPECT: Re: SUSPECT: Rewrite From Header In CANCEL forwarded packet
Thanks Bogdan, I haven't seen your email before, it has been tagged as spam by
our exchange!
Have a nice day
Le Friday 13 March 2009 12.19:33 Bogdan-Andrei Iancu, vous avez écrit :
Hi Marc,
use uac_replace_from() from UAC module to get changes persistent across
the transaction:
http://www.opensips.org/html/docs/modules/1.4.x/uac.html#id227417
Note the CANCEL is locally regenerated (even if received) - it is
propagating hopbyhop.
Regards,
Bogdan
Marc Leurent wrote:
Good Morning,
I'am using a loadbalanced platform with dispatcher module. When a packet is
sent from a Gateway to a peer, I rewrite the From Header which contains the
IP of the gateway in order to put the proxy domain because some Hardphones
like
Thomson ST2030 will keep the domain in call history to be able to call back
a
person. If the UA use the direct IP of the gateway, the call will failed,
it
has to go through the proxy! So I would like to rewrite From header coming
from gateways
It's working for INVITE, OPTIONS,... in main route, but CANCEL packets are
not
rewritten and UA sent back a message saying transaction does not exists
because From header is different from INVITE!
I have put this in main route:
if (ds_is_from_list()) {# Returns true if the current request comes
from a host from the dispatcher-list
xlog(L_INFO, Rewriting From Domain in $rm $(hdr(From)));
#subst('/^From:(.*)@xxx.xxx.xxx.[0-9]+(.*)$/From:
\...@xxx-dev.xxx.xx\2/ig');
}
Have you got any idea how to rewrite CANCEL packet?
I have tried to put the same thing in failure route but it does not work.
Thanks
Best Regards
--
-- --
Marc LEURENT
Ingénieur VoIP
DECKPOINT SA
Une société du groupe VTX Telecom
Rue Eugène-Marziano 15 - 1227 Les Acacias
http://www.vtx.ch - marc.leur...@vtx-telecom.ch
VTX, votre partenaire telecom proche de vous !
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Re: [OpenSIPS-Users] mediaproxy: TLS unexpected length
Hi Erik,
On 16 Mar 2009, at 02:45, erik pepermans wrote:
Installed mediaproxy on centos - got following when starting
dispatcher and relay :
Mar 16 02:33:05 ns1 media-dispatcher[21875]: [-] Log opened.
Mar 16 02:33:05 ns1 media-dispatcher[21875]: [-] Starting MediaProxy
Dispatcher 2.3.2
Mar 16 02:33:05 ns1 media-dispatcher[21875]: [-] Twisted is using
selectreactor
Mar 16 02:33:05 ns1 media-dispatcher[21875]: [-]
mediaproxy.dispatcher.RelayFactory starting on 25060
Mar 16 02:33:05 ns1 media-dispatcher[21875]: [-]
mediaproxy.dispatcher.OpenSIPSControlFactory starting on '/var/run/
mediaproxy/dispatcher.sock'
Mar 16 02:33:05 ns1 media-dispatcher[21875]: [-]
mediaproxy.dispatcher.ManagementControlFactory starting on 25061
Mar 16 02:33:46 ns1 media-relay[21925]: [-] Log opened.
Mar 16 02:33:46 ns1 media-relay[21925]: [-] Starting MediaProxy
Relay 2.3.2
Mar 16 02:33:46 ns1 media-relay[21925]: [-] Set resource limit for
maximum open file descriptors to 11000
Mar 16 02:33:46 ns1 media-relay[21925]: [-] Adding new dispatcher at
77.xx.xx.xxx:25060
Mar 16 02:33:46 ns1 media-dispatcher[21875]:
[mediaproxy.dispatcher.RelayFactory] Connection from relay at
77.xx.xx.xxx
Mar 16 02:33:46 ns1 media-dispatcher[21875]:
[mediaproxy.dispatcher.RelayFactory] Traceback (most recent call
last):
Mar 16 02:33:46 ns1 media-dispatcher[21875]:
[mediaproxy.dispatcher.RelayFactory] File /usr/lib64/
python2.4/site-packages/twisted/python/log.py, line 69, in
callWithContext
Mar 16 02:33:46 ns1 media-dispatcher[21875]:
[mediaproxy.dispatcher.RelayFactory] return
context.call({ILogContext: newCtx}, func, *args, **kw)
Mar 16 02:33:46 ns1 media-dispatcher[21875]:
[mediaproxy.dispatcher.RelayFactory] File /usr/lib64/
python2.4/site-packages/twisted/python/context.py, line 59, in
callWithContextMar 16 02:33:46 ns1 media-dispatcher[21875]:
[mediaproxy.dispatcher.RelayFactory] return
self.currentContext().callWithContext(ctx, func, *args, **kw)
Mar 16 02:33:46 ns1 media-dispatcher[21875]:
[mediaproxy.dispatcher.RelayFactory] File /usr/lib64/
python2.4/site-packages/twisted/python/context.py, line 37, in
callWithContext
Mar 16 02:33:46 ns1 media-dispatcher[21875]:
[mediaproxy.dispatcher.RelayFactory] return func(*args,**kw)
Mar 16 02:33:46 ns1 media-dispatcher[21875]:
[mediaproxy.dispatcher.RelayFactory] File /usr/lib64/
python2.4/site-packages/twisted/internet/selectreactor.py, line
146, in _doReadOrWrite
Mar 16 02:33:46 ns1 media-dispatcher[21875]:
[mediaproxy.dispatcher.RelayFactory] why =
getattr(selectable, method)()
Mar 16 02:33:46 ns1 media-dispatcher[21875]:
[mediaproxy.dispatcher.RelayFactory] --- exception caught here
---
Mar 16 02:33:46 ns1 media-dispatcher[21875]:
[mediaproxy.dispatcher.RelayFactory] File /usr/lib64/
python2.4/site-packages/twisted/internet/tcp.py, line 938, in doRead
Mar 16 02:33:46 ns1 media-dispatcher[21875]:
[mediaproxy.dispatcher.RelayFactory] transport =
self.transport(skt, protocol, addr, self, s, self.reactor)
Mar 16 02:33:46 ns1 media-dispatcher[21875]:
[mediaproxy.dispatcher.RelayFactory] exceptions.TypeError:
__init__() takes exactly 6 arguments (7 given)Mar 16 02:33:46 ns1
media-relay[21925]: [Uninitialized] error: Could not connect to
dispatcher at 77.xx.xx.xxx:25060 (retrying in 10 seconds): A TLS
packet with unexpected length was received.
Mar 16 02:33:56 ns1 media-dispatcher[21875]:
[mediaproxy.dispatcher.RelayFactory] Connection from relay at
77.xx.xx.xxx
Mar 16 02:33:56 ns1 media-dispatcher[21875]:
[mediaproxy.dispatcher.RelayFactory] error: Connection to relay
77.xx.xx.xxx is already present, disconnectingMar 16 02:33:56 ns1
media-relay[21925]: [Uninitialized] error: Could not connect to
dispatcher at 77.xx.xx.xxx:25060 (retrying in 10 seconds): A TLS
packet with unexpected length was received.
please help
Please upgrade your python-gnutls to version 1.1.8, that should fix
this issue.
Ruud Klaver
AG Projects
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[OpenSIPS-Users] Loadbalancing/Failover with 1.4
Hello, I have the following setup: UA opensips asterik1 to asterisk3 pstn-gw With the dispatching module its easy to handle the calls from UA to pstn. The other way is the tricky one, since the pstn-gw will route the calls randomly to one of the asterisk-server. So I need a stable solution to route all sip-packets belonging to one dialog to the asterisk-server the communication was started. How can I solve this? route-header for the asterisk-server will cause, that some requests are routed directly between UA and asterisk (BYE). BR Kiste -- kiste lat: 54.322684, lon: 10.13586 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Loadbalancing/Failover with 1.4
Hi Uwe, First of all, route via dispatcher only initial requests (without totag) and use record-route on the dispatcher box. This will force BYEs from PSTN to be routed via the same servers. Regards, Bogdan Uwe Kastens wrote: Hello, I have the following setup: UA opensips asterik1 to asterisk3 pstn-gw With the dispatching module its easy to handle the calls from UA to pstn. The other way is the tricky one, since the pstn-gw will route the calls randomly to one of the asterisk-server. So I need a stable solution to route all sip-packets belonging to one dialog to the asterisk-server the communication was started. How can I solve this? route-header for the asterisk-server will cause, that some requests are routed directly between UA and asterisk (BYE). BR Kiste ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] CDRTool - ReloadRatingTables and new destinations
Hey Adrian, thanks for the quick reaction. Here it comes what I found out after tests (could be I am doing something wrong): * ReloadRatingTables produces a reload of destination table now (during normalize.php execution, so not real-time but depending on how often this script is run in cron). * Even when the tables are reloaded, the new destinations are not processed, therefore when querying ShowPrice I will get still the initial values (the one received on cdrtool startup). So the result is that I am adding/deleting/modifying destinations, but no effect in the ShowPrice. My version: version CDRTool version 6.7.3 Ta, DanB On Sun, Mar 15, 2009 at 9:56 AM, Adrian Georgescu a...@ag-projects.comwrote: Hello Dan, I found the bug that affected the rating engine, I fixed it. You should upgrade to 6.7.3 version. Regards, Adrian On Mar 14, 2009, at 7:12 PM, Dan-Cristian Bogos wrote: Folks, some strange behavior I have noticed in CDRTool: both ReloadRatingTables and /etc/init.d/cdrtool reload are strangely removing the destinations and destinations.sip from cdrtool.memcached table, therefore I have the feeling that this is why ShowPrice does newer show newer updates in the destinations table. version CDRTool version 6.7.2 The only way I made CDRTool work with newer destinations inserted into destinations table was restarting it. Is this behavior wanted? Also, ShowPrice used to insert also destination name, other than dest_id inside Destination field, but no longer there. Ta, DanB ___ 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] CDRTool - ReloadRatingTables and new destinations
On Mar 16, 2009, at 11:36 AM, DanB wrote: Hey Adrian, thanks for the quick reaction. Here it comes what I found out after tests (could be I am doing something wrong): * ReloadRatingTables produces a reload of destination table now (during normalize.php execution, so not real-time but depending on how often this script is run in cron). The reload simply sets a flag in the system to reload the destinations. The destinations are reloaded when normalization process occurs, either from cron job, manual run of normalize.php script or when you load a web page. * Even when the tables are reloaded, the new destinations are not processed, therefore when querying ShowPrice I will get still the initial values (the one received on cdrtool startup). So the result is that I am adding/deleting/modifying destinations, but no effect in the ShowPrice. This is a problem unrelated to the destinations reload. Most likely you did not create the correct rating table data. My version: version CDRTool version 6.7.3 Ta, DanB On Sun, Mar 15, 2009 at 9:56 AM, Adrian Georgescu a...@ag- projects.com wrote: Hello Dan, I found the bug that affected the rating engine, I fixed it. You should upgrade to 6.7.3 version. Regards, Adrian On Mar 14, 2009, at 7:12 PM, Dan-Cristian Bogos wrote: Folks, some strange behavior I have noticed in CDRTool: both ReloadRatingTables and /etc/init.d/cdrtool reload are strangely removing the destinations and destinations.sip from cdrtool.memcached table, therefore I have the feeling that this is why ShowPrice does newer show newer updates in the destinations table. version CDRTool version 6.7.2 The only way I made CDRTool work with newer destinations inserted into destinations table was restarting it. Is this behavior wanted? Also, ShowPrice used to insert also destination name, other than dest_id inside Destination field, but no longer there. Ta, DanB ___ 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
[OpenSIPS-Users] XCAP pres-rules changes not taking effect?
Hi there, I'm doing some tests with XCAP in integrated mode and I´m having some issues with pres-rules documents. My problem seems to be that the changes in the xcap document doesn't seem to be taking effect. I was wondering if someone else encountered the same problem? I'm not running the mi_xmlrpc module, I don't think this should be a problem as XCAP and Opensips communicate via the xcap table? (I've also restarted opensips just in case there's a cache not being updated...) Any help would be highly appreciated. Regards Gus ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] XCAP pres-rules changes not taking effect?
If you change the pres-rules document you need to setup the mi interface to run the reloadWatchers command. Adrian On Mar 16, 2009, at 11:54 AM, Gustavo Santana Robledo wrote: Hi there, I'm doing some tests with XCAP in integrated mode and I´m having some issues with pres-rules documents. My problem seems to be that the changes in the xcap document doesn't seem to be taking effect. I was wondering if someone else encountered the same problem? I'm not running the mi_xmlrpc module, I don't think this should be a problem as XCAP and Opensips communicate via the xcap table? (I've also restarted opensips just in case there's a cache not being updated...) Any help would be highly appreciated. Regards Gus ___ 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] XCAP pres-rules changes not taking effect?
Sorry forgot to say that I'm running the version 1.4.4 thanks From: Gustavo Santana Robledo gsant...@fedetec.es To: Users@lists.opensips.org Date: 16/03/2009 11:55 Subject: [OpenSIPS-Users] XCAP pres-rules changes not taking effect? Sent by: users-boun...@lists.opensips.org Hi there, I'm doing some tests with XCAP in integrated mode and I´m having some issues with pres-rules documents. My problem seems to be that the changes in the xcap document doesn't seem to be taking effect. I was wondering if someone else encountered the same problem? I'm not running the mi_xmlrpc module, I don't think this should be a problem as XCAP and Opensips communicate via the xcap table? (I've also restarted opensips just in case there's a cache not being updated...) Any help would be highly appreciated. Regards Gus ___ 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] CDRTool - ReloadRatingTables and new destinations
Adrian, On Mon, 2009-03-16 at 11:50 +0100, Adrian Georgescu wrote: This is a problem unrelated to the destinations reload. Most likely you did not create the correct rating table data. I am not sure if it is due to correct rating table data (should see no Span price in that case I think), so bear with me to read the logs. Here comes a more detailed usage scenario: I have added a new number inside destination table (like you said, no rating defined, just the destination one). * First query will identify maybe correctly the destination 31 (since the reload of rating tables was not yet done and the new destination is not yet in the memory). ShowPrice from=...@itsyscom.com gateway=10.10.10.1 Duration=59 To=003167608 0.1200 Duration: 60 s App: audio Destination: 31 Customer: default Increment: 60 s Connect: 0. StartTime: 2009-03-16 10:08:45 -- Span: 1 Duration: 60 s ProfileId: DEFAULT / weekday RateId: DEFAULT / 0-24h Rate: 0.1200 / 60 s Price: 0.1200 reloadratingtables 1 * Second and third attempt are after reloadratingtables succeed - saw it in mysql.log. Notice that the destination identified is still 31. ShowPrice from=...@itsyscom.com gateway=10.10.10.1 Duration=59 To=003167608 0.1200 Duration: 60 s App: audio Destination: 31 Customer: default Increment: 60 s Connect: 0. StartTime: 2009-03-16 10:09:52 -- Span: 1 Duration: 60 s ProfileId: DEFAULT / weekday RateId: DEFAULT / 0-24h Rate: 0.1200 / 60 s Price: 0.1200 ShowPrice from=...@itsyscom.com gateway=10.10.10.1 Duration=59 To=003167608 0.1200 Duration: 60 s App: audio Destination: 31 Customer: default Increment: 60 s Connect: 0. StartTime: 2009-03-16 10:18:20 -- Span: 1 Duration: 60 s ProfileId: DEFAULT / weekday RateId: DEFAULT / 0-24h Rate: 0.1200 / 60 s Price: 0.1200 Connection closed by foreign host. * Here I have restarted the cdrtool with /etc/init.d/cdrtool restart. Now the destination is correctly identified as 3167608 (full length), of course without Span section since I have no rating defined yet. DellLaptop:/usr/local/src/cdrtool# telnet localhost 9094 Trying 127.0.0.1... Connected to localhost. Escape character is '^]'. ShowPrice from=...@itsyscom.com gateway=10.10.10.1 Duration=59 To=003167608 0 Duration: 60 s App: audio Destination: 3167608 Customer: default Increment: 60 s Is my logic broken? Same thing happens if I simply remove the destination (still showing it in ShowPrice even if there is no longer in the database. Ta, DanB ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Retrieve headers from negative reply
Hi Bogdan, It turned out it wasn't the get_redirect() function but t_relay trying to route based on some bad Contact information that serialize_branches/next_branches had loaded into the RURI. I didn't spend much time on it because I leared to access to reply Contact header directly in the reply route. In this particular case, the 302 that came back wasn't a routable URI but rather ported number information from the North American porting database almost in the form of a URI, enough so that it could be stored in the Contact header. I haven't looked into it but I'm sure it violated at least one RFC. Garbage in, garbage out. - Jeff On 3/16/09 4:03 AM, Bogdan-Andrei Iancu bog...@voice-system.ro wrote: Hi Jeff, Do you get this crash all the time when using the get_redirect() function ? Regards, Bogdan Jeff Pyle wrote: Hello, Is there any way to retrieve headers from a response, rather than the request? I need the Contact header out of a 302 response. Both $hdr(Contact) and $ct give me the contact of the request, not the response. I tried get_redirects(), but because of the formatting of the contact response it crashes Opensips: CRITICAL:core:qm_free: freeing already freed pointer, first free: parser/msg_parser.c: set_ruri(712) - aborting INFO:core:handle_sigs: child process 3554 exited by a signal 6 INFO:core:handle_sigs: core was not generated INFO:core:handle_sigs: terminating due to SIGCHLD Get_redirects() was a last resort anyway because I really just need a value from the Contact, I don¹t want to use it for routing. Thanks, Jeff ___ 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] 3.00 load avg
Dan, # netstat -pan | grep opensips udp0 0 ww.xx.yy.zz:50600.0.0.0:* 13788/opensips unix 2 [ ] DGRAM31333 13821/opensips /tmp/opensips.sock unix 3 [ ] STREAM CONNECTED 31355 13788/opensips unix 2 [ ] STREAM CONNECTED 31353 13788/opensips unix 2 [ ] STREAM CONNECTED 31351 13829/opensips unix 2 [ ] STREAM CONNECTED 31348 13827/opensips unix 2 [ ] STREAM CONNECTED 31346 13825/opensips unix 2 [ ] STREAM CONNECTED 31344 13823/opensips unix 2 [ ] STREAM CONNECTED 31342 13821/opensips unix 3 [ ] STREAM CONNECTED 31338 13817/opensips unix 2 [ ] DGRAM31337 13821/opensips unix 2 [ ] STREAM CONNECTED 31335 13817/opensips unix 2 [ ] STREAM CONNECTED 31331 13817/opensips unix 3 [ ] STREAM CONNECTED 31328 13814/opensips unix 2 [ ] STREAM CONNECTED 31326 13814/opensips unix 2 [ ] STREAM CONNECTED 31324 13814/opensips unix 3 [ ] STREAM CONNECTED 31320 13811/opensips unix 2 [ ] STREAM CONNECTED 31318 13811/opensips unix 2 [ ] STREAM CONNECTED 31316 13811/opensips unix 3 [ ] STREAM CONNECTED 31312 13807/opensips unix 2 [ ] STREAM CONNECTED 31310 13807/opensips unix 2 [ ] STREAM CONNECTED 31308 13807/opensips unix 3 [ ] STREAM CONNECTED 31303 13803/opensips unix 3 [ ] STREAM CONNECTED 31301 13801/opensips unix 2 [ ] STREAM CONNECTED 31298 13803/opensips unix 3 [ ] STREAM CONNECTED 31295 13798/opensips unix 2 [ ] STREAM CONNECTED 31291 13803/opensips unix 2 [ ] STREAM CONNECTED 31289 13801/opensips unix 3 [ ] STREAM CONNECTED 31288 13794/opensips unix 2 [ ] STREAM CONNECTED 31287 13798/opensips unix 2 [ ] STREAM CONNECTED 31285 13794/opensips unix 2 [ ] STREAM CONNECTED 31283 13801/opensips unix 3 [ ] STREAM CONNECTED 31279 13795/opensips unix 2 [ ] STREAM CONNECTED 31277 13798/opensips unix 2 [ ] STREAM CONNECTED 31275 13795/opensips unix 2 [ ] STREAM CONNECTED 31273 13795/opensips unix 2 [ ] STREAM CONNECTED 31271 13794/opensips unix 2 [ ] DGRAM31257 13788/opensips unix 2 [ ] DGRAM31256 13788/opensips top - 07:42:58 up 2 days, 16:38, 1 user, load average: 3.00, 3.00, 3.00 Tasks: 108 total, 2 running, 106 sleeping, 0 stopped, 0 zombie Cpu(s): 0.0%us, 0.0%sy, 0.0%ni,100.0%id, 0.0%wa, 0.0%hi, 0.0%si, 0.0%st This is a CentOS i386 xen VM, running on CentOS i386. The physical machine has 8 processing cores, and the virtual has 4. I've been keeping fairly up to date from SVN. I'd say this problem appeared maybe a week or so ago but I'm not certain on the exact rev number. - Jeff On 3/15/09 12:00 AM, Dan Pascu d...@ag-projects.com wrote: On Friday 13 March 2009, Bogdan-Andrei Iancu wrote: Hi Jeff, A simple google on the topic top command load average http://www.teamquest.com/resources/gunther/display/5/ See starting with section 2.2 Shortly, is about the number of processes in your case. It's the number of _active_ processes. Or to be more precise it's the number of processes waiting in the queue to get scheduled to execute on the CPU. You cannot get a 3.0 load average with all processes being idle, so something is definitely abnormal. You need 3 processes loaded at 100% to reach a load average of 3.0 (with 1 CPU). But I'm curious if opensips can really trigger this, or is just a glitch in the OS reporting the load average? An interesting piece of information would be to know if the opensips processes have any incoming/outgoing network queues (seen with netstat -pan | grep opensips). ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Retrieve headers from negative reply
Hi Jeff, I see, but if you still have the entire log of the crash (or if you can get a new one or explain the logic in order to reproduce it) it will be great, as I want to fix this bug (even if you use something else now). Thanks and regards, Bogdan Jeff Pyle wrote: Hi Bogdan, It turned out it wasn't the get_redirect() function but t_relay trying to route based on some bad Contact information that serialize_branches/next_branches had loaded into the RURI. I didn't spend much time on it because I leared to access to reply Contact header directly in the reply route. In this particular case, the 302 that came back wasn't a routable URI but rather ported number information from the North American porting database almost in the form of a URI, enough so that it could be stored in the Contact header. I haven't looked into it but I'm sure it violated at least one RFC. Garbage in, garbage out. - Jeff On 3/16/09 4:03 AM, Bogdan-Andrei Iancu bog...@voice-system.ro wrote: Hi Jeff, Do you get this crash all the time when using the get_redirect() function ? Regards, Bogdan Jeff Pyle wrote: Hello, Is there any way to retrieve headers from a response, rather than the request? I need the Contact header out of a 302 response. Both $hdr(Contact) and $ct give me the contact of the request, not the response. I tried get_redirects(), but because of the formatting of the contact response it crashes Opensips: CRITICAL:core:qm_free: freeing already freed pointer, first free: parser/msg_parser.c: set_ruri(712) - aborting INFO:core:handle_sigs: child process 3554 exited by a signal 6 INFO:core:handle_sigs: core was not generated INFO:core:handle_sigs: terminating due to SIGCHLD Get_redirects() was a last resort anyway because I really just need a value from the Contact, I don¹t want to use it for routing. Thanks, Jeff ___ 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] OpenSipS + CentOS: No debug logs!
Hi Bodgan, I've tried both of the options. In the script (debug=6), command line -dd and via fifo. My logging is done via syslog. This weekend I've tried in another CentOS. Same results :( Thanks, Noel On Mon, Mar 16, 2009 at 4:57 AM, Bogdan-Andrei Iancu bog...@voice-system.ro wrote: Hi Noel, How do you set the debug level? in the script (debug=6) or as command line param (-d) ? Also, where is your logging done? console (standard error) or syslog ? Regards, Bogdan Noel R. Morais wrote: I've installed it in another CentOS machine and didn't work . I'm lost ... haha -Noel On Fri, Mar 13, 2009 at 4:48 PM, Jeff Pyle jp...@fidelityvoice.com wrote: Well, I'm not sure how to tell. I can tell you that the .svnrevision file has 2:5412M in it, and I last updated on March 9 (Monday). - Jeff On 3/13/09 3:42 PM, Noel R. Morais noelro...@gmail.com wrote: What is the trunk revision that u are using? Noel On Fri, Mar 13, 2009 at 4:28 PM, Jeff Pyle jp...@fidelityvoice.com wrote: Noel, I'm running on CentOS 5.2 i386 and I'm able to see DBG messages. I didn't do anything special to have them show. This is with a current update of Opensips 1.5. - Jeff On 3/13/09 3:23 PM, Noel R. Morais noelro...@gmail.com wrote: Hi Guys, something weird is happening here. I cannot get module DBG logs when using CentOS 5.2. Even if I set the debug parameter to 9 it doesn't show me any debug message. Only CRITICAL, INFO, WARNING and ERROR messages. I've made the same installation in a Fedora Core 5 and it worked like a charm. Is there some dependence or any other tweak to make it work in a CentOS? Thanks, Noel ___ 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] CDRTool - ReloadRatingTables and new destinations
A call once normalized is stored in the radiust table and remains unchanged unless you re-normalize the calls that you wish to have updated. So changing rating tables does not have any influence upon previously normalized calls, they remain with the previous values. Adrian On Mar 16, 2009, at 12:21 PM, Dan-Cristian Bogos wrote: Adrian, On Mon, 2009-03-16 at 11:50 +0100, Adrian Georgescu wrote: This is a problem unrelated to the destinations reload. Most likely you did not create the correct rating table data. I am not sure if it is due to correct rating table data (should see no Span price in that case I think), so bear with me to read the logs. Here comes a more detailed usage scenario: I have added a new number inside destination table (like you said, no rating defined, just the destination one). * First query will identify maybe correctly the destination 31 (since the reload of rating tables was not yet done and the new destination is not yet in the memory). ShowPrice from=...@itsyscom.com gateway=10.10.10.1 Duration=59 To=003167608 0.1200 Duration: 60 s App: audio Destination: 31 Customer: default Increment: 60 s Connect: 0. StartTime: 2009-03-16 10:08:45 -- Span: 1 Duration: 60 s ProfileId: DEFAULT / weekday RateId: DEFAULT / 0-24h Rate: 0.1200 / 60 s Price: 0.1200 reloadratingtables 1 * Second and third attempt are after reloadratingtables succeed - saw it in mysql.log. Notice that the destination identified is still 31. ShowPrice from=...@itsyscom.com gateway=10.10.10.1 Duration=59 To=003167608 0.1200 Duration: 60 s App: audio Destination: 31 Customer: default Increment: 60 s Connect: 0. StartTime: 2009-03-16 10:09:52 -- Span: 1 Duration: 60 s ProfileId: DEFAULT / weekday RateId: DEFAULT / 0-24h Rate: 0.1200 / 60 s Price: 0.1200 ShowPrice from=...@itsyscom.com gateway=10.10.10.1 Duration=59 To=003167608 0.1200 Duration: 60 s App: audio Destination: 31 Customer: default Increment: 60 s Connect: 0. StartTime: 2009-03-16 10:18:20 -- Span: 1 Duration: 60 s ProfileId: DEFAULT / weekday RateId: DEFAULT / 0-24h Rate: 0.1200 / 60 s Price: 0.1200 Connection closed by foreign host. * Here I have restarted the cdrtool with /etc/init.d/cdrtool restart. Now the destination is correctly identified as 3167608 (full length), of course without Span section since I have no rating defined yet. DellLaptop:/usr/local/src/cdrtool# telnet localhost 9094 Trying 127.0.0.1... Connected to localhost. Escape character is '^]'. ShowPrice from=...@itsyscom.com gateway=10.10.10.1 Duration=59 To=003167608 0 Duration: 60 s App: audio Destination: 3167608 Customer: default Increment: 60 s Is my logic broken? Same thing happens if I simply remove the destination (still showing it in ShowPrice even if there is no longer in the database. Ta, DanB ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] CDRTool - ReloadRatingTables and new destinations
I understand that, but I did not normalize anything in my actions. I just played with the ShowPrice over telnet (adding and deleting destinations directly from the database, and then running ReloadTables over telnet interface). DanB On Mon, 2009-03-16 at 13:39 +0100, Adrian Georgescu wrote: A call once normalized is stored in the radiust table and remains unchanged unless you re-normalize the calls that you wish to have updated. So changing rating tables does not have any influence upon previously normalized calls, they remain with the previous values. Adrian On Mar 16, 2009, at 12:21 PM, Dan-Cristian Bogos wrote: Adrian, On Mon, 2009-03-16 at 11:50 +0100, Adrian Georgescu wrote: This is a problem unrelated to the destinations reload. Most likely you did not create the correct rating table data. I am not sure if it is due to correct rating table data (should see no Span price in that case I think), so bear with me to read the logs. Here comes a more detailed usage scenario: I have added a new number inside destination table (like you said, no rating defined, just the destination one). * First query will identify maybe correctly the destination 31 (since the reload of rating tables was not yet done and the new destination is not yet in the memory). ShowPrice from=...@itsyscom.com gateway=10.10.10.1 Duration=59 To=003167608 0.1200 Duration: 60 s App: audio Destination: 31 Customer: default Increment: 60 s Connect: 0. StartTime: 2009-03-16 10:08:45 -- Span: 1 Duration: 60 s ProfileId: DEFAULT / weekday RateId: DEFAULT / 0-24h Rate: 0.1200 / 60 s Price: 0.1200 reloadratingtables 1 * Second and third attempt are after reloadratingtables succeed - saw it in mysql.log. Notice that the destination identified is still 31. ShowPrice from=...@itsyscom.com gateway=10.10.10.1 Duration=59 To=003167608 0.1200 Duration: 60 s App: audio Destination: 31 Customer: default Increment: 60 s Connect: 0. StartTime: 2009-03-16 10:09:52 -- Span: 1 Duration: 60 s ProfileId: DEFAULT / weekday RateId: DEFAULT / 0-24h Rate: 0.1200 / 60 s Price: 0.1200 ShowPrice from=...@itsyscom.com gateway=10.10.10.1 Duration=59 To=003167608 0.1200 Duration: 60 s App: audio Destination: 31 Customer: default Increment: 60 s Connect: 0. StartTime: 2009-03-16 10:18:20 -- Span: 1 Duration: 60 s ProfileId: DEFAULT / weekday RateId: DEFAULT / 0-24h Rate: 0.1200 / 60 s Price: 0.1200 Connection closed by foreign host. * Here I have restarted the cdrtool with /etc/init.d/cdrtool restart. Now the destination is correctly identified as 3167608 (full length), of course without Span section since I have no rating defined yet. DellLaptop:/usr/local/src/cdrtool# telnet localhost 9094 Trying 127.0.0.1... Connected to localhost. Escape character is '^]'. ShowPrice from=...@itsyscom.com gateway=10.10.10.1 Duration=59 To=003167608 0 Duration: 60 s App: audio Destination: 3167608 Customer: default Increment: 60 s Is my logic broken? Same thing happens if I simply remove the destination (still showing it in ShowPrice even if there is no longer in the database. Ta, DanB ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Error on presence_xml
What opensips version are you using? 1.5 or 1.4.4? Anca bay2x1 wrote: The code I am using is exactly the same as the one I have copied here. I have disabled the xcap_client module and its parameters but I now encounter the following error: Mar 14 08:04:22 ws16 /usr/sbin/opensips[3963]: INFO:pua:db_restore: the query returned no result Mar 14 08:04:22 ws16 /usr/sbin/opensips[3963]: NOTICE:presence:mod_init: initializing module ... Mar 14 08:04:22 ws16 /usr/sbin/opensips[3963]: ERROR:presence_xml:mod_init: Can't bind xcap_client Mar 14 08:04:22 ws16 /usr/sbin/opensips[3963]: ERROR:core:init_mod: failed to initialize module presence_xml Mar 14 08:04:22 ws16 /usr/sbin/opensips[3963]: ERROR:core:main: error while initializing modules Mar 14 08:04:22 ws16 /usr/sbin/opensips[3963]: NOTICE:presence:destroy: destroy module ... Anca Vamanu-2 wrote: Hi, I have tried with exactly your parameters and it worked ok. You don't need to load the xcap_client module if you set integrated_xcap_server module parameter in both presence_xml and rls. Are you sure you trying to run with this config :) ? regards, Anca bay2x1 wrote: Base on my understanding I dont need to set the xcap_server parameter if I set the modparam(presence_xml, integrated_xcap_server, 1)? BTW I am using version 1.5 Bogdan-Andrei Iancu wrote: Hi, I think the error is self explanatory: ERROR:presence_xml:shm_copy_xcap_list: no xcap_server parameter set You need to set the xcap_server parameter for the presence_xml module. See: http://www.opensips.org/html/docs/modules/1.4.x/presence_xml.html#id228211 Regards, Bogdan bay2x1 wrote: I have enabled the xcap_client module and the error was eliminated but encountered another stumbling block this is how my opensips.cfg configuration intergrating openxcap given all the needed modules has been loaded. # - presence params -enabled: openxcap support /* uncomment the following lines if you want to enable presence */ modparam(presence, db_url, mysql://opensips:opensip...@localhost/opensips) modparam(presence, server_address, sip:prese...@10.0.0.1) modparam(presence, fallback2db, 1) modparam(presence, clean_period, 30) # -- xcap params -- added: openxcap support modparam(presence_xml, db_url, mysql://opensips:opensip...@localhost/opensips) modparam(presence_xml, force_active, 0) modparam(presence_xml, pidf_manipulation, 1) modparam(presence_xml, integrated_xcap_server, 1) # -- rls params -- added: openxcap support modparam(rls, db_url, mysql://opensips:opensip...@localhost/opensips) modparam(rls, server_address, sip:r...@10.0.0.1) modparam(rls, to_presence_code, 5) modparam(rls, integrated_xcap_server, 1) modparam(pua, db_url, mysql://opensips:opensip...@localhost/opensips) modparam(pua, db_table, pua) */ #-xcap client - modparam(xcap_client, db_url, mysql://opensips:opensip...@localhost/opensips) modparam(xcap_client, xcap_table, xcaps) modparam(xcap_client, periodical_query, 0) this is the error log. ERROR LOG Mar 12 13:43:23 ws16 /usr/sbin/opensips[19514]: NOTICE:presence:mod_init: initializing module ... Mar 12 13:43:24 ws16 /usr/sbin/opensips[19514]: ERROR:presence_xml:shm_copy_xcap_list: no xcap_server parameter set Mar 12 13:43:24 ws16 /usr/sbin/opensips[19514]: ERROR:presence_xml:mod_init: copying xcap server list in share memory Mar 12 13:43:24 ws16 /usr/sbin/opensips[19514]: ERROR:core:init_mod: failed to initialize module presence_xml Mar 12 13:43:24 ws16 /usr/sbin/opensips[19514]: ERROR:core:main: error while initializing modules Mar 12 13:43:24 ws16 /usr/sbin/opensips[19514]: NOTICE:presence:destroy: destroy module ... bay2x1 wrote: I have integrated openxcap on my config and opensips fails to run generating error log below Mar 12 10:57:10 ws16 /usr/sbin/opensips[14469]: INFO:dialog:mod_init: Dialog module - initializing Mar 12 10:57:10 ws16 /usr/sbin/opensips[14469]: NOTICE:presence:mod_init: initializing module ... Mar 12 10:57:10 ws16 /usr/sbin/opensips[14469]: ERROR:presence_xml:mod_init: Can't bind xcap_client Mar 12 10:57:10 ws16 /usr/sbin/opensips[14469]: ERROR:core:init_mod: failed to initialize module presence_xml Mar 12 10:57:10 ws16 /usr/sbin/opensips[14469]: ERROR:core:main: error while initializing modules Mar 12 10:57:10 ws16 /usr/sbin/opensips[14469]: NOTICE:presence:destroy: destroy module ... I had checked my presence_xml configuration and check the opensips documentation and I can say that I have done what was instructed in the configuration instruction. At first I was encountering segfault error, but now this error is now appearing ___ Users
Re: [OpenSIPS-Users] Timeout value for cache_store function and timeout refreshing
Hi Andreas,
Andreas Westermaier wrote:
Hi,
I'm currently using 'cache_store()' to store channel limits for customers in
memory to avoid the need of especially fetching it (either from radius, or
from db).
Channel limit information comes in via SIP-AVP from radius on each register
('radius_www_authorize()') and on each invite ('radius_proxy_authorize()').
So after a successfull 'radius_proxy_authorize()' I currently do
if (is_avp_set($avp(s:chanlimit)/n) avp_check($avp(s:chanlimit),
gt/i:0)) {
cache_store(local, chanlimit_...@$ar, $avp(s:chanlimit), 86400);
}
This stores the information for a whole day. But I think this could be
tweaked if it would be possible to set the timeout for cache expiry to the
value of the registration expiry. E.g. by using $(hdr(Expires){s.int}) as
the last parameter for 'cache_store()'.
Q1: Is there a reason why the function currently does not support the
variable $(hdr(Expires){s.int}) for the timeout value or is it just not yet
implemented?
No, there is no reason not to support variable for expires argument and
it is a good idea to have it. I will implement it after the release.
Allowing variables for timeout, the channel limit information would be
available as long as the registration for a user will be valid and expires
when the registration does. Of course, on each renewal of the registration
also the memcache variable has to be refreshed.
Q2: Can the timeout for a memcache variable currently refreshed just by
calling 'cache_store()' a second time, after the first execution and before
expiration or is it necessary to firstly remove the variable from cache and
insert it again for the timer to get reset?
Yes, you can refresh the expires by calling cache_store again. The
implementation is such that a store for the same label rewrites the old
value.
regards,
Anca
Regards,
Andreas
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Re: [OpenSIPS-Users] OpenSipS + CentOS: No debug logs!
Hi Noel, Are you sure you syslog is not doing a redirect (in a separate file) for messages below INFO? check the rules from /etc/syslog.conf. Regards, Bogdan Noel R. Morais wrote: Hi Bodgan, I've tried both of the options. In the script (debug=6), command line -dd and via fifo. My logging is done via syslog. This weekend I've tried in another CentOS. Same results :( Thanks, Noel On Mon, Mar 16, 2009 at 4:57 AM, Bogdan-Andrei Iancu bog...@voice-system.ro wrote: Hi Noel, How do you set the debug level? in the script (debug=6) or as command line param (-d) ? Also, where is your logging done? console (standard error) or syslog ? Regards, Bogdan Noel R. Morais wrote: I've installed it in another CentOS machine and didn't work . I'm lost ... haha -Noel On Fri, Mar 13, 2009 at 4:48 PM, Jeff Pyle jp...@fidelityvoice.com wrote: Well, I'm not sure how to tell. I can tell you that the .svnrevision file has 2:5412M in it, and I last updated on March 9 (Monday). - Jeff On 3/13/09 3:42 PM, Noel R. Morais noelro...@gmail.com wrote: What is the trunk revision that u are using? Noel On Fri, Mar 13, 2009 at 4:28 PM, Jeff Pyle jp...@fidelityvoice.com wrote: Noel, I'm running on CentOS 5.2 i386 and I'm able to see DBG messages. I didn't do anything special to have them show. This is with a current update of Opensips 1.5. - Jeff On 3/13/09 3:23 PM, Noel R. Morais noelro...@gmail.com wrote: Hi Guys, something weird is happening here. I cannot get module DBG logs when using CentOS 5.2. Even if I set the debug parameter to 9 it doesn't show me any debug message. Only CRITICAL, INFO, WARNING and ERROR messages. I've made the same installation in a Fedora Core 5 and it worked like a charm. Is there some dependence or any other tweak to make it work in a CentOS? Thanks, Noel ___ 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
[OpenSIPS-Users] [FIXED] TM final reply selection
Hi, Even if this is a fix, I think it is important for all of you to be aware of it, as it will visibly change the OpenSIPS' behaviour. It is about branch selection at parallel forking in TM module. Originally, the lowest negative code was selected (when all branches were failing). But there were issues as some codes are reporting useful info, other not (disregarding the code number). The new priority order is: 487 (as result of UAC cancel) prio=0 6xx repliesprio=3xx 3xx repliesprio=4xx 401/407 (authentication) prio=5xx 415 (Unsupported Media Type) 420 (Bad Extension) 484 (Address Incomplete) 4xxprio=6xx 5xxprio=7xx 408 (timeout) prio=8xx 503 (UAS failure indication) Just to know what to expect :) Regards, Bogdan ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] [OpenSIPS-Devel] dialog module: dialog not cleaned up when BYE is sent
Hi Herman, sorry for delay - please apply the attached patch and run again the call - send me the debug log. Thanks and regards, Bogdan Herman Bastiaens wrote: Hi Bogdan, yes, I see these messages in the log. You can also find them in the log I've sent you on monday (09-03). Because the DID was not present (due to the record_route_preset bug), the matching mode falls back to matching based on SIP elements. Best Regards, Herman Bogdan-Andrei Iancu wrote: Hi Herman, yes, that is correct. When you use mode 2 (no did), for ACK and BYE, do you get something like this in the logs: no dialog callid= ? Regards, Bogdan Herman Bastiaens wrote: Hi Bogdan, the issue is resolved when using the DID for dialog matching, but if I use dlg_match_mode 2, the error still occurs. Is there a problem with the matching of messages to dialogs if there is no DID present? Best Regards, Herman Bogdan-Andrei Iancu wrote: Hi Herman, Thanks for the files you sent me. I found out that the problem was in record_route_preset() - the function was simply ignoring whatever parameter you were adding from script or from other modules. This is why the DIALOG module was not recognizing the sequential requests. The fix is available on 1.5 - if you could give it another try and let me know, it will be great. Regards, Bogdan Herman Bastiaens wrote: Hi Bogdan, x The problem only occurs: - if there is another server in the loop, if I test with a client that's registered to xx (instead of the demo server), the issue does not occur - if I record_route_preset instead of record_route. As you'll see, when I record_route_preset, there is no did present in the route header. I believe it should fall back to default dialog matching, but perhaps there's a problem with this mechanism? Hope we can sort this out. Thanks for your replies! Bogdan-Andrei Iancu wrote: Hi Herman, Sorry for the mixing :). Please get the logs (with debug=6) and the SIP trace (ngrep) for such a call and sent them to me (you can send them off-list if larger than 40K). I cannot make any assumption yet without first looking at the logs. Thanks and regards, Bogdan Herman Bastiaens wrote: Bogdan, I didn't post the log on the forum, that was someone else. I just added my comment to the thread because it seemed like the same problem. I can get a log of the scenario tomorrow if that's helpful. Do you have any idea why the record_route_preset could by messing up this scenario? Do you have any idea what could explain the 5 second threshold? Thanks a lot for your replies, hope we can sort this out. Herman, The log you posted on the forum did not show any usage of loose_route() - if you look on the log, for ACK and BYE there is no mesage from rr or dialog module. Can you confirm this? Regards, Bogdan Herman Bastiaens wrote: Hi Bogdan, I'm pretty sure I do a loose_route for the ACK and BYE, but I'm still seeing this error. I've tested the most basic scenario, starting from an example, and this problem seems to occur when I start using a record_route_preset(...). Perhaps the dialog module can't handle this? (just to be clear, the bug is still only occurring if the call is shut down during the first few seconds after the ACK). I've attached the script in which I see the error occurring. The IP of my server is 172.17.10.44 Hi Herman, And the bell rang! :) I went over the logs you posted on the forum and I noticed (both script and logs) that you are not using loose_route() for sequential requests. You do record_route() for the initial INVITE, but no loose_route for ACK, BYE. And loose_route() is the function that updates the dialog state. So, in your case, the dialog does not see the ACK and BYE and still keeps in the CONFIRMED_NA (not acknowledged) state. This is way it is not removed. See the default opensips.cfg file to see how to use the loose_route(). I beat it will work after that ;) Regards, Bogdan Herman Bastiaens wrote: Hi Bodgan, that's what I'm seeing, time and time again. I was hoping this might ring a bell, but from your reply I take that it doesn't :-) Regards, Herman Hi Herman, just to copy the reply from the forum :) : So, let me see if I get it right. With the same configuration, if the call is longer than 5 secs, everything is ok (dialog is removed when receiving a BYE). But if the call is shorter than 5 secs, the dialog is not removed. Is this what you say? Regards, Bogdan Herman Bastiaens wrote: Hi, I've posted this problem on the forum (https://sourceforge.net/forum/message.php?msg_id=6595314), but it doesn't seem to be very active, so I'm posting it here as well. I'm having a problem with the dialog module of opensips 1.4.2-notls. When a call is set up, and released within five seconds, the dialog is not removed. I am sure the call is set up correctly (INVITE - 200 OK - ACK) and the BYE is sent (with the correct call-id, from and to
[OpenSIPS-Users] database sermyadmin+opensips
Hi list, I have problems annexing the sermyadmin database with opensips, I show them my configuration with opensips to see where I have the error: #DEFAULT_DB_URL=mysql://sermyadmin:ba...@localhost/sermyadmin modparam(presence|presence_xml|group|permissions|auth_db|uri_db|usrloc, db_url, mysql://sermyadmin:ba...@localhost/sermyadmin) modparam(domain, db_url, mysql://sermyadmin:ba...@localhost/sermyadmin) modparam(domain, db_mode, 1) modparam(auth_db, password_column, ha1) modparam(auth_db, password_column_2, ha1b) modparam(presence_xml, force_active, 1) modparam(presence, max_expires, 3600) modparam(presence, server_address, sip:192.168.10.3:5060) modparam(permissions, db_mode, 1) modparam(permissions, trusted_table, trusted) modparam(group, group_column, grp) modparam(avpops, avp_url, mysql://sermyadmin:ba...@localhost/sermyadmin) modparam(avpops, avp_table, usr_preferences) opensipsctrlrc DBNAME=sermyadmin # database path used by dbtext or db_berkeley # DB_PATH=/usr/local/etc/opensips/dbtext ## database read/write user DBRWUSER=sermyadmin ## password for database read/write user DBRWPW=baco2 ## database read only user DBROUSER=sermyadmin ## password for database read only user DBROPW=baco2 log openser Mar 16 11:02:47 twoxserver /sbin/opensips[9238]: ERROR:core:db_check_table_version: invalid version 1 for table watchers found, expected 3 Mar 16 11:02:47 twoxserver /sbin/opensips[9238]: ERROR:presence:mod_init: error during table version check Mar 16 11:02:47 twoxserver /sbin/opensips[9238]: ERROR:core:init_mod: failed to initialize module presence Mar 16 11:02:47 twoxserver /sbin/opensips[9238]: ERROR:core:main: error while initializing modules some idea where my error can be? regardss -- rickygm http://gnuforever.homelinux.com ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] segmentation fault when calling pkg_malloc
On Sunday 15 March 2009, Vasil Kolev wrote: В 05:43 +0200 на 15.03.2009 (нд), Dan Pascu написа: On Thursday 12 March 2009, Vasil Kolev wrote: This doesn't seem like running out of memory, more like a memory corruption. The first thing to check is if either 'frag' or 'f' are NULL or invalid (e.g. in gdb do print f, print frag and see what does that say). After that try dereferencing them, seeing how could they get these values, etc. I've seen this thing before. I'm willing to bet that the pointers are not NULL. They are most likely composed of ascii byte codes from some SIP message header (as a result of the allocator memory being overwritten by part of the processed message). Unfortunately I have no idea what causes this, and the only solution to a similar problem I was experiencing was to replace the opensips private memory allocator with the system memory allocator when building opensips. Maybe the thread author can try using the system memory allocator to see if the problem goes away. If so, we definitely have a problem with the memory allocator. If not, he may have a different issue somewhere else in his code. Oh shit seems like the appropriate response :) (I'm not the thread author, but this would seem to affect me too) Anyway, there seem to be three different memory allocators in opensips - vqm_malloc, fm_malloc and qm_malloc (this is from looking in mem/mem.h), is it possible that one has an issue and the rest not? Now the issue might be that the system memory allocator is more resilient to some strange type of corruption (for example, single byte overflow of strings) and doesn't have this weird issue... The private memory pool used by the internal memory allocator is a 1 MB area right after the area where static variables are stored. It may be possible that there is a buffer overflow when writing to some static variables that gets to write in this pool overwriting the allocator internal structure. The system memory doesn't have this problem as it is stored somewhere else. Such a buffer overflow will even go unnoticed by memory tracing tools as it sees that the write is done in the 1MB that the program allocated, having no clue that the area is actually managed by another memory allocator that can get corrupted. At the same time I do not imply that this is what happens. The way the memory is corrupted doesn't suggest this pattern as the private memory is not overwritten from the beginning. It looks more like specific areas are overwritten as if the memory allocator thinks that the memory area was released and gives it to someone else on the next call to pkg_malloc, resulting in 2 different pointers to the area competing for the memory. -- Dan ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] 3.00 load avg
On Monday 16 March 2009, Jeff Pyle wrote: Dan, # netstat -pan | grep opensips udp0 0 ww.xx.yy.zz:50600.0.0.0:* There appear to be no network queue, so the app is idle. However you mention that you run this in a virtual machine. Do you see the problem with a real system as well? Otherwise I wouldn't rely on the accuracy of what the virtual machine reports. Besides, did you upgrade the emulator as well around that time? This may be very well a bug in the emulation software, reporting false status under certain conditions. top - 07:42:58 up 2 days, 16:38, 1 user, load average: 3.00, 3.00, 3.00 Tasks: 108 total, 2 running, 106 sleeping, 0 stopped, 0 zombie Cpu(s): 0.0%us, 0.0%sy, 0.0%ni,100.0%id, 0.0%wa, 0.0%hi, 0.0%si, 0.0%st This is a CentOS i386 xen VM, running on CentOS i386. The physical machine has 8 processing cores, and the virtual has 4. I've been keeping fairly up to date from SVN. I'd say this problem appeared maybe a week or so ago but I'm not certain on the exact rev number. -- Dan ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Retrieve headers from negative reply
Hi Jeff, Thanks a lot for the info - it was very useful - I was able to reproduce the crash and eventually to fix it. Best regards, Bogdan Jeff Pyle wrote: Hi Bogdan, Writing you privately... [XX] Hopefully that's helpful for you. Let me know if you need anything else. - Jeff On 3/16/09 8:04 AM, Bogdan-Andrei Iancu bog...@voice-system.ro wrote: Hi Jeff, I see, but if you still have the entire log of the crash (or if you can get a new one or explain the logic in order to reproduce it) it will be great, as I want to fix this bug (even if you use something else now). Thanks and regards, Bogdan Jeff Pyle wrote: Hi Bogdan, It turned out it wasn't the get_redirect() function but t_relay trying to route based on some bad Contact information that serialize_branches/next_branches had loaded into the RURI. I didn't spend much time on it because I leared to access to reply Contact header directly in the reply route. In this particular case, the 302 that came back wasn't a routable URI but rather ported number information from the North American porting database almost in the form of a URI, enough so that it could be stored in the Contact header. I haven't looked into it but I'm sure it violated at least one RFC. Garbage in, garbage out. - Jeff On 3/16/09 4:03 AM, Bogdan-Andrei Iancu bog...@voice-system.ro wrote: Hi Jeff, Do you get this crash all the time when using the get_redirect() function ? Regards, Bogdan Jeff Pyle wrote: Hello, Is there any way to retrieve headers from a response, rather than the request? I need the Contact header out of a 302 response. Both $hdr(Contact) and $ct give me the contact of the request, not the response. I tried get_redirects(), but because of the formatting of the contact response it crashes Opensips: CRITICAL:core:qm_free: freeing already freed pointer, first free: parser/msg_parser.c: set_ruri(712) - aborting INFO:core:handle_sigs: child process 3554 exited by a signal 6 INFO:core:handle_sigs: core was not generated INFO:core:handle_sigs: terminating due to SIGCHLD Get_redirects() was a last resort anyway because I really just need a value from the Contact, I don¹t want to use it for routing. Thanks, Jeff ___ 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] CDRTool - ReloadRatingTables and new destinations
Hi Dan, I have made a change to fix your problem. Pl upgrade to 6.7.4 and let me know if you still have problems with reloading the tables. Regards, Adrian On Mar 16, 2009, at 2:16 PM, Dan-Cristian Bogos wrote: Adrian, I understand all these, but I think you don't get me. Please do me a favor, and let me know if you get the same behavior like me in your lab: 1. Delete a destination from cdrtool.destinations table. 2. Issue a ShowPrice command over telnet which should match the destination you just deleted. Nothing should happen, I understand until you ReloadRatingTables and run normalize.php. 3. Issue a ReloadRatingTables and run normalize.php. 4. Run the same ShowPrice over telnet and you will see that your destination is still showing up in the results (at least this is what is happening to me). This scenario you can repeat when adding new destinations or modifying the existing ones. Ta, DanB On Mon, 2009-03-16 at 14:03 +0100, Adrian Georgescu wrote: Only the normalization process reinitializes the changed data. If you run telnet and issue manually the ShowPrice command nothing happens until a normalization process runs. If you would check for every ShowOPrice command if a reload is required it will affect the speed of the engine. Adrian On Mar 16, 2009, at 1:44 PM, Dan-Cristian Bogos wrote: I understand that, but I did not normalize anything in my actions. I just played with the ShowPrice over telnet (adding and deleting destinations directly from the database, and then running ReloadTables over telnet interface). DanB On Mon, 2009-03-16 at 13:39 +0100, Adrian Georgescu wrote: A call once normalized is stored in the radiust table and remains unchanged unless you re-normalize the calls that you wish to have updated. So changing rating tables does not have any influence upon previously normalized calls, they remain with the previous values. Adrian On Mar 16, 2009, at 12:21 PM, Dan-Cristian Bogos wrote: Adrian, On Mon, 2009-03-16 at 11:50 +0100, Adrian Georgescu wrote: This is a problem unrelated to the destinations reload. Most likely you did not create the correct rating table data. I am not sure if it is due to correct rating table data (should see no Span price in that case I think), so bear with me to read the logs. Here comes a more detailed usage scenario: I have added a new number inside destination table (like you said, no rating defined, just the destination one). * First query will identify maybe correctly the destination 31 (since the reload of rating tables was not yet done and the new destination is not yet in the memory). ShowPrice from=...@itsyscom.com gateway=10.10.10.1 Duration=59 To=003167608 0.1200 Duration: 60 s App: audio Destination: 31 Customer: default Increment: 60 s Connect: 0. StartTime: 2009-03-16 10:08:45 -- Span: 1 Duration: 60 s ProfileId: DEFAULT / weekday RateId: DEFAULT / 0-24h Rate: 0.1200 / 60 s Price: 0.1200 reloadratingtables 1 * Second and third attempt are after reloadratingtables succeed - saw it in mysql.log. Notice that the destination identified is still 31. ShowPrice from=...@itsyscom.com gateway=10.10.10.1 Duration=59 To=003167608 0.1200 Duration: 60 s App: audio Destination: 31 Customer: default Increment: 60 s Connect: 0. StartTime: 2009-03-16 10:09:52 -- Span: 1 Duration: 60 s ProfileId: DEFAULT / weekday RateId: DEFAULT / 0-24h Rate: 0.1200 / 60 s Price: 0.1200 ShowPrice from=...@itsyscom.com gateway=10.10.10.1 Duration=59 To=003167608 0.1200 Duration: 60 s App: audio Destination: 31 Customer: default Increment: 60 s Connect: 0. StartTime: 2009-03-16 10:18:20 -- Span: 1 Duration: 60 s ProfileId: DEFAULT / weekday RateId: DEFAULT / 0-24h Rate: 0.1200 / 60 s Price: 0.1200 Connection closed by foreign host. * Here I have restarted the cdrtool with /etc/init.d/cdrtool restart. Now the destination is correctly identified as 3167608 (full length), of course without Span section since I have no rating defined yet. DellLaptop:/usr/local/src/cdrtool# telnet localhost 9094 Trying 127.0.0.1... Connected to localhost. Escape character is '^]'. ShowPrice from=...@itsyscom.com gateway=10.10.10.1 Duration=59 To=003167608 0 Duration: 60 s App: audio Destination: 3167608 Customer: default Increment: 60 s Is my logic broken? Same thing happens if I simply remove the destination (still showing it in ShowPrice even if there is no longer in the database. Ta, DanB ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] database sermyadmin+opensips
Hi Ricky, It is an old definition in the presence tables. Delete the presence related tables and recreate them with the mysql scripts in the opensips source code. /scripts/mysql/ It is fixed in the newest version. Flavio From: troxlinux [mailto:xserverli...@gmail.com] Sent: Mon 16/03/2009 14:26 To: users@lists.opensips.org Cc: Flávio Eduardo de Andrade Subject: database sermyadmin+opensips Hi list, I have problems annexing the sermyadmin database with opensips, I show them my configuration with opensips to see where I have the error: #DEFAULT_DB_URL=mysql://sermyadmin:ba...@localhost/sermyadmin modparam(presence|presence_xml|group|permissions|auth_db|uri_db|usrloc, db_url, mysql://sermyadmin:ba...@localhost/sermyadmin) modparam(domain, db_url, mysql://sermyadmin:ba...@localhost/sermyadmin) modparam(domain, db_mode, 1) modparam(auth_db, password_column, ha1) modparam(auth_db, password_column_2, ha1b) modparam(presence_xml, force_active, 1) modparam(presence, max_expires, 3600) modparam(presence, server_address, sip:192.168.10.3:5060) modparam(permissions, db_mode, 1) modparam(permissions, trusted_table, trusted) modparam(group, group_column, grp) modparam(avpops, avp_url, mysql://sermyadmin:ba...@localhost/sermyadmin) modparam(avpops, avp_table, usr_preferences) opensipsctrlrc DBNAME=sermyadmin # database path used by dbtext or db_berkeley # DB_PATH=/usr/local/etc/opensips/dbtext ## database read/write user DBRWUSER=sermyadmin ## password for database read/write user DBRWPW=baco2 ## database read only user DBROUSER=sermyadmin ## password for database read only user DBROPW=baco2 log openser Mar 16 11:02:47 twoxserver /sbin/opensips[9238]: ERROR:core:db_check_table_version: invalid version 1 for table watchers found, expected 3 Mar 16 11:02:47 twoxserver /sbin/opensips[9238]: ERROR:presence:mod_init: error during table version check Mar 16 11:02:47 twoxserver /sbin/opensips[9238]: ERROR:core:init_mod: failed to initialize module presence Mar 16 11:02:47 twoxserver /sbin/opensips[9238]: ERROR:core:main: error while initializing modules some idea where my error can be? regardss -- rickygm http://gnuforever.homelinux.com http://gnuforever.homelinux.com/ Vai contratar um canal E1 ou SIP? Além de oferecermos o melhor preço, na contratação de um canal E1 ou SIP GVT você recebe um desconto no projeto que pode chegar a até 50%. Consulte-nos sobre a viabilidade técnica e econômica. Vai contratar um canal de dados? Além de oferecermos o melhor preço, na contratação de um canal de dados corporativo GVT você recebe um desconto no projeto que pode chegar a até 50%. Consulte-nos sobre a viabilidade técnica e econômica. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] 3.00 load avg
Hi Dan, I haven't tried it on a real box yet. With all the things it hooks into it's not exactly portable at the moment. Nothing has changed elsewhere on the system, either with the emulator, kernels on host or guest, etc. I'm not *too* worried about it only because the apps on the system (vm) seem to be running well. I may drop the v-cpus from 4 to 2 to see if that adversely affects performance under low load. For what it's worth I've excellent success with Xen under CentOS with Openser 1.3.2 and other apps. The reports seem to be accurate everywhere but here. Even mediaproxy (under moderate to low load) does fairly well, although I'd never consider going into production with it on a virtual machine. Thanks for the analysis. - Jeff On 3/16/09 1:38 PM, Dan Pascu d...@ag-projects.com wrote: There appear to be no network queue, so the app is idle. However you mention that you run this in a virtual machine. Do you see the problem with a real system as well? Otherwise I wouldn't rely on the accuracy of what the virtual machine reports. Besides, did you upgrade the emulator as well around that time? This may be very well a bug in the emulation software, reporting false status under certain conditions. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] segmentation fault when calling pkg_malloc
В 19:32 +0200 на 16.03.2009 (пн), Dan Pascu написа: The private memory pool used by the internal memory allocator is a 1 MB area right after the area where static variables are stored. It may be possible that there is a buffer overflow when writing to some static variables that gets to write in this pool overwriting the allocator internal structure. The system memory doesn't have this problem as it is stored somewhere else. Such a buffer overflow will even go unnoticed by memory tracing tools as it sees that the write is done in the 1MB that the program allocated, having no clue that the area is actually managed by another memory allocator that can get corrupted. If a static variable is overwriting the memory buffer, this should be easily visible with valgrind - they're both separate allocations (the pool is a static one too). Now, if there's something that does overflow in the pool, that will go unnoticed by default, but I think there should be some way to make valgrind understand other mallocs (probably via an API to say i'm an allocator, here's what I did). At the same time I do not imply that this is what happens. The way the memory is corrupted doesn't suggest this pattern as the private memory is not overwritten from the beginning. It looks more like specific areas are overwritten as if the memory allocator thinks that the memory area was released and gives it to someone else on the next call to pkg_malloc, resulting in 2 different pointers to the area competing for the memory. Another question is, is there a bug at all, e.g. are there any definitive reports? All the memory corruption problems I've seen can be attributed either to bugs I've introduced or the issues with the non-null terminated BLOBs from Postgres and the zero-length results (again from postgres). -- Regards, Vasil Kolev Attractel NV dCAP #1324, LPIC2 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] rewriting transactional responses
Hello, If a transaction has progressed from the initial request route, hit a t_relay(), come back into an armed failure route with a 302, and gone back to another request route... how can I change the 302 to something more generic (such as a 500) before relaying it to the UAC? Thanks, Jeff ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] loose_route: loop on ACK requests
Robert, you are right. I'm using loose routing all the way but for some reason the loose_route function for the ACK is acting as a strict router. follow bellow the debug code regarding this ACK. 192.168.191.188 is the opensips ip address: Mar 16 17:18:00 openser opensips[9057]: ***BEFORE loose_route: ACK ruri:[sip:6715551131062...@192.168.193.20:5060] from:[sip:551135880...@192.168.193.20] to:[sip:6715551131062...@192.168.191.188;user=phone] Mar 16 17:18:00 openser opensips[9057]: DBG:core:parse_headers: flags=200 Mar 16 17:18:00 openser opensips[9057]: params (0x81a7a00, 36), called from parser/parse_rr.c: do_parse_rr_body(63) Mar 16 17:18:00 openser opensips[9057]: params (0x81a7a00, 36), returns address 0x81f4a14 frag. 0x81f49fc (size=44) on 1 -th hit Mar 16 17:18:00 openser opensips[9057]: DBG:rr:is_preloaded: is_preloaded: No Mar 16 17:18:00 openser opensips[9057]: DBG:core:grep_sock_info: checking if host==us: 14==15 [192.168.193.20] == [192.168.191.188] Mar 16 17:18:00 openser opensips[9057]: DBG:core:grep_sock_info: checking if port 5060 matches port 5060 Mar 16 17:18:00 openser opensips[9057]: DBG:rr:after_strict: Next hop: 'sip:192.168.191.188;lr=on;ftag=1c1682753440;did=043.0c936ea3' is loose router Mar 16 17:18:00 openser opensips[9057]: params (0x81a7a00, 60), called from parser/msg_parser.c: set_dst_uri(738) Mar 16 17:18:00 openser opensips[9057]: params (0x81a7a00, 60), returns address 0x81f3b48 frag. 0x81f3b30 (size=68) on 1 -th hit Mar 16 17:18:00 openser opensips[9057]: DBG:core:parse_headers: flags= Mar 16 17:18:00 openser opensips[9057]: params (0x81a7a00, 60), called from parser/msg_parser.c: set_ruri(705) Mar 16 17:18:00 openser opensips[9057]: params (0x81a7a00, 60), returns address 0x81f4ba0 frag. 0x81f4b88 (size=72) on 1 -th hit Mar 16 17:18:00 openser opensips[9057]: DBG:rr:after_strict: The last route URI: 'sip:192.168.191.188;lr=on;ftag=1c1682753440;did=043.0c936ea3' Mar 16 17:18:00 openser opensips[9057]: params (0x81a7a00, 32), called from data_lump.c: del_lump(298) Mar 16 17:18:00 openser opensips[9057]: params (0x81a7a00, 32), returns address 0x81f3ae8 frag. 0x81f3ad0 (size=48) on 1 -th hit Mar 16 17:18:00 openser opensips[9057]: DBG:rr:run_rr_callbacks: callback id 1 entered with Mar 16 17:18:00 openser opensips[9057]: DBG:core:parse_headers: flags=48 Mar 16 17:18:00 openser opensips[9057]: DBG:rr:run_rr_callbacks: callback id 0 entered with Mar 16 17:18:00 openser opensips[9057]: DBG:uac:restore_from: getting 'vsf' Route param Mar 16 17:18:00 openser opensips[9057]: DBG:uac:restore_from: route param 'vsf' not found Mar 16 17:18:00 openser opensips[9057]: ***AFTER loose_route=TRUE: ACK ruri:[sip:192.168.191.188;lr=on;ftag=1c1682753440;did=043.0c936ea3] from:[sip:551135880...@192.168.193.20] to:[sip:6715551131062...@192.168.191.188;user=phone] On Fri, Mar 13, 2009 at 2:27 AM, Robert Dyck rob.d...@telus.net wrote: Very peculiar. The RURI was rewritten with the URI in the Route header. Does your script rewrite it? This usually only happens when the route set shows a next hop and the next hop is a strict router. If loose routing is in effect all the way, the RURI does not change. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Double Registration transaction
Using the ngrep tool I was able to capture two request generated for every
registration so I am getting two Register + 401 and Register + 200 OK.
Bogdan-Andrei Iancu wrote:
Hi,
Why do you say is executed twice?? Maybe you mistake the 2 REGISTERs
because of the authentication process... REGISTER + 401 ; REGISTER + 200
OK
using the syslog the xlog(Registration Process ends here! \n); code is
twice printed everytime a registration occurs.
Is this okay?
Regards,
Bogdan
bay2x1 wrote:
I tried creating my own registration configuration. Using the ngrep tool
I
was able to capture the header packets. I am wondering why the route(1)
is
executed twice everytime that a client registers. This is the code I am
using for registration.
route{
if(!mf_process_maxfwd_header(10)){
sl_send_reply(483, Too Many Hops);
exit;
};
if(method=='REGISTER'){
route(1);
}
}
route[1] {
if(is_uri_host_local()) {
xlog(Registration Process starts here! \n);
if(!www_authorize(, subscriber))
{else if (method=='INVITE') {
route(2);
};
www_challenge(,1);
xlog(Registration Process ends here! \n);
exit;
};
if(!check_to()) {
sl_send_reply(403, Forbidden);
};
save(location);
exit;
} else if {
sl_send_reply(403, Forbidden);
};
}
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Re: [OpenSIPS-Users] database sermyadmin+opensips
Hi flavio , flavio erases the five tables according to the opensips docu but when I want to export the file presence-create.sql it always shows me the same error: ERROR 1062 (23000) at line 1: Duplicate entry 'presentity' for key 1 http://www.opensips.org/html/docs/db/db-schema-1.4.x.html#GEN-DB-PRESENTITY thank you for all your help, but it interests me to run sermyadmin regardsss 2009/3/16 Flávio Eduardo de Andrade fla...@voffice.com.br: Remove all the tables related to presence (About 5). Check inside the presence-create.sql Regards, Flavio -- rickygm http://gnuforever.homelinux.com ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] sst create callback warning
Hello, After updating to 1.5 rev 5457 this afternoon I now see this warning after the BYE (or perhaps the 200 following): WARNING:sst:sst_dialog_created_CB: dialog create callback called with a non-INVITE request. I did make some configuration changes recently, but nothing that should affect the dialogs or SST. How can I diagnose what's causing this? Thanks, Jeff ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] sst create callback warning
It seems that you are trying to create a dialog while processing a BYE. The way dialog module works has changed between 1.4 and 1.5. Check your script and make sure that you don't create a dialog for messages other then initial INVITE. Regards, Ovidiu Sas On Mon, Mar 16, 2009 at 10:26 PM, Jeff Pyle jp...@fidelityvoice.com wrote: Hello, After updating to 1.5 rev 5457 this afternoon I now see this warning after the BYE (or perhaps the 200 following): WARNING:sst:sst_dialog_created_CB: dialog create callback called with a non-INVITE request. I did make some configuration changes recently, but nothing that should affect the dialogs or SST. How can I diagnose what's causing this? Thanks, Jeff ___ 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
[OpenSIPS-Users] XML-RPC 404 Not Found
Hello everybody, i installed the Openser + Openser XML-RPC MI from the ubuntu hardy repository (openser-xmlrpc-module openser packages). The thing is i am testing the XML-RPC interface but i receive 404 not found all the time. I installed Serweb as well, to sniff the xmlrpc queries ... but the thing is the 404 messages are still there. How can i increase the log level of this module to see what's happening? any ideas? if anyone can point me some detailed documentation, i'll be thankful. Regards, -- Lucas ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
