Re: [SR-Users] Remote Party ID
Hello, On 12/7/11 10:03 AM, Pavel Segeč wrote: HI, Thank you. Just as I sent my previous mail I look into kamctl command syntax and I found -- command 'rpid' - manage Remote-Party-ID (RPID) rpid add . add rpid for a user (*) rpid rm . set rpid to NULL for a user (*) rpid show ... show rpid of a user How this work? this is for adding/removing/showing the rpid column in subscriber table -- if you look in database, there is a column named rpid for subscriber table. Practically, setting it and loading with load_credentials is a way to use it in the config file. In config then you can use append_hf() add header as you need, or if it is rpid specific format, see siputils module for other options: http://kamailio.org/docs/modules/stable/modules_k/siputils.html Cheers, Daniel pavel *From:*Daniel-Constantin Mierla [mailto:mico...@gmail.com] *Sent:* Tuesday, December 06, 2011 11:14 PM *To:* SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List *Cc:* Pavel Segeč *Subject:* Re: [SR-Users] Remote Party ID Hello, On 12/6/11 2:50 PM, Pavel Segeč wrote: Hi, I'm looking for some recommendations or guidlines. My leading IT department which is providing PSTN connectivity requires to include Remote Party ID in SIP Messages in a case where From URI is not in telco format (I prefer email like style of addressing). How to simply assign to an user its Remote Party ID which will be used when PSTN calling will occure? On what should I focus on? To use user preferences with AVP pairs? or something another? since it is about one attribute associated with the caller, a recommended way is to add a new column in subscriber table, say it is named 'rpid'. Then, in config file, set the load_credentials parameter of auth_db to load the rpid value and store in an avp. After you authenticate the caller, the avp will be set to the rpid value, see: http://kamailio.org/docs/modules/stable/modules_k/auth_db.html#id2528175 Using this option, practically you use same db query that loads the user password for authentication to fetch the rpid value. Cheers, Daniel -- Daniel-Constantin Mierla --http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Running Kamailio as non-root user
Hello, On 12/9/11 2:45 AM, Nathaniel L Keeling wrote: I have just installed Kamailio 3.2 on Solaris sparc and would like to know what is the best way to have Kamailio to run under another userid, like kamailio. you can provide user/group in the command line starting kamailio. See kamailio -h for the options. Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] How to i use record_route_advertised_address
Hello, On 12/8/11 11:13 PM, Peter Dunkley wrote: Hi, The record_route_advertised_address() config file function is now in git master. from the readme I see that it takes the address from the function parameter, not from advertised_address global parameter, which is getting close to what record_route_preset() does, just that record_route_advertised_address() can do double rr with the address parameter based on auto-detection of transport change. Probably would be useful to have the option of no parameter for this new function, in which case the global parameter advertised_adress/advertised_port will be used for building rr. Cheers, Daniel Peter On Wed, 2011-12-07 at 10:27 +, Hugh Waite wrote: Hi, I've implemented the config file function for record_route_advertised_address and it should appear in git head later today. Just doing the documentation and some testing... Hugh On 05/12/2011 21:52, Daniel-Constantin Mierla wrote: > Hello, > > maybe the option for now is to set the parameters for > record_route_preset() to advertised_address/port: > > http://kamailio.org/docs/modules/stable/modules_k/rr.html#id2542169 > > I have some deployments (like on EC2) where I use advertise_address > for have the proper Via and record_route_preset() for record route > headers. > > For simplicity, it would be good to have the function in config to > take advertised address value for building the rr header. > > Cheers, > Daniel > > On 12/3/11 9:41 PM, Hugh Waite wrote: >> Hi, >> I do not think that record_route() uses the advertised_address >> variables. We have been using the advertised_address / >> advertised_port variables in the config file and although they are >> used in the Via headers, they are not substituted into the Record >> Route lines. >> >> record_route_advertised_address() was added about 2 weeks ago for >> exactly this reason - using an advertised address in record routes >> when the transport changes. It automatically inserts double (rr2=on) >> routes in these cases and extra custom parameters can be added later. >> >> However, we only needed the C API, so it isn't exported as a config >> file function yet. This won't be difficult. >> >> Regards, >> Hugh >> >> On 03/12/11 15:57, Ovidiu Sas wrote: >>> Take a look at set_advertised_address: >>> http://www.kamailio.org/dokuwiki/doku.php/core-cookbook:3.1.x#set_advertised_address >>> >>> Set it and then record_route() will properly fill the Record-Route >>> header. >>> >>> >>> Regards, >>> Ovidiu Sas >>> >> >> >> ___ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >> sr-users@lists.sip-router.org <mailto:sr-users@lists.sip-router.org> >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > -- Peter Dunkley Technical Director Crocodile RCS Ltd ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] set host as inactive and probing in dispatcher
Hello,
On 12/7/11 3:02 PM, Uri Shacked wrote:
hi,
i use "ds_select_domain()" to route a call.
if the destination reply with an error i am using the next lines in a
route that i called from the failure route:
if($avp(OSPRPL)!=404) {
where do you set $avp(OSPRPL)? Can you add a xlog() message inside the
if to be sure the ds_mark_dst() is executed? Alternative, use debugger
module with cfgtrace to see what actions are executed from config file.
Cheers,
Daniel
ds_mark_dst("IP");
}
after trying this line i do "kamctl fifo ds_list" and i see the host
is still at "AX". and in the next call the dispatcher still tries the
same destination.
why? how do i set a destinatio in dispatcher to be inactive?
this is my modparams:
#!ifdef WITH_DISPATCHER
#-- dispatcher params -
modparam("dispatcher", "db_url", DBURL)
modparam("dispatcher", "dst_avp", "$avp(dsdst)")
modparam("dispatcher", "cnt_avp", "$avp(dscnt)")
modparam("dispatcher", "grp_avp", "$avp(dsgrp)")
modparam("dispatcher", "attrs_avp", "$avp(dsattrs)")
modparam("dispatcher", "ds_ping_from", "sip:u...@host.com
<mailto:sip%3au...@host.com>")
modparam("dispatcher", "ds_ping_reply_codes", "class=2;code=405")
modparam("dispatcher", "ds_ping_interval", 30)
modparam("dispatcher", "ds_probing_mode", 1)
modparam("dispatcher", "ds_hash_size", 8)
modparam("dispatcher", "flags", 2)
#!endif
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Re: [SR-Users] unset host domain in dispatcher module
Hello, On 12/7/11 9:37 AM, Uri Shacked wrote: Daniel hi, thanks for the reply. i woke up this mornning and tried to change the destination set on the dispatcher DB table. it was "sip:user@host"... i changed it to "sip:host" and used ds_select_domain. worked great, the R-URI host changed as needed. regarding the t_relay(host,port) option. the problem with that is that the t_relay does not accept $avp() any ideas? you can do: $du = $var(dst); t_relay(); In $var(dst) you must have a valid sip URI address (can include protocol, like: sip:127.0.0.1:5060;transport=tcp) Cheers, Daniel On Wed, Dec 7, 2011 at 12:09 AM, Daniel-Constantin Mierla mailto:mico...@gmail.com>> wrote: Hello, On 12/6/11 7:08 PM, Uri Shacked wrote: Hi, i am using the dispatcher module in order to send the call to a LNP server. after the 3xx reply from the LNP server, i would like to send the invite back to the proxy that initiate the call from the beginnig. how do i do that? one option: t_relay_to(...) with the parameter pointing to the proxy where you want to send the request. it seems that kamailio dispatcher sets the host domain in the Internet Protocol, and the sip request line does not change No, the dispatcher does not work at IP packet layer, it is updating the request URI with ds_select_domain() or sets internal outbound proxy address with ds_select_dst(). so to be short.. two questions: 1. how do i "undo" the dispatcher? 2. how do i make the dispatcher change the requst lineas well? (i am using ds_select_dst or ds_select_domain and they do the same...) Changing the request URI is via ds_select_domain(), I can bet it is working fine as I use it in many places -- if it does not do it for you, can you paste here the config file used for doing dispatching along with the SIP trace (ngrep) of such call? Cheers, Daniel the carrierroute module, do it better, but i need the dispatcher for the "keep alive" usage. ideas will be great :-) Uri ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org <mailto:sr-users@lists.sip-router.org> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla --http://www.asipto.com <http://www.asipto.com/> http://linkedin.com/in/miconda -- http://twitter.com/miconda ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] set host as inactive and probing in dispatcher
Hello,
cn you give output of 'kamailio -V' and the log messages with debug=3 in
config file?
Note that from the moment you set the gateway in some state and you dump
the list via mi, there can be keepalives that change the state.
Cheers,
Daniel
On 12/11/11 3:42 PM, Uri Shacked wrote:
Hi,
i tried to change the ds_probing_mode as well. it didn't do any good.
i need the probing mode to be set to 1 anyway.
so,
any ideas?
here is the cfg again:
after the failure route i turn to this route
"route[OSPRPLERR] {
xlog("$C(rg) SCRIPT : in OSPRPLERR $ru $C(xx)\n");
if($avp(OSPRPL)!=404) {
xlog("$C(rg) OSPRPL is $avp(OSPRPL)*now disable destination* $C(xx)\n");
ds_mark_dst("AP");
xlog("$C(rg) *after disabling destination* $C(xx)\n");
}"
here is the log:
13(5760) ERROR:
Re: [SR-Users] Via header branch parameter in ACK message not unique
Hello, On 12/12/11 8:18 AM, Phillman25 Kyriacou wrote: Dear List I'm currently facing an issue with a carrier whereby they are complaining that the via header branch parameter (sip.Via.branch == "z9hG4bKcydzigwkX") in the ACK messages is not unique which results in some calls being successful and some terminating after 20 seconds as their sip proxy determines it as a duplicate request. Is there a way to make the via header branch parameter in the ACK message unique across every transaction? set in your config file the global parameter: syn_branch=0 On the other hand, the behavior of the devises used by carrier is completely broken. ACK can have a different path than INVITE, so there should be no match done on Via branch parameter. Since it is ACK for 200OK, it has to be matched for dialog. Having the RFC3261 cookie in the branch parameter, it has to be matched with (callid, from tag, to tag). Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] use of $stat(name)
Hello, On 12/12/11 10:17 AM, Uri Shacked wrote: hi, anyone knows what is the options to use $stat ? in siremis there are some examples. but, where can i see what kind of names can i write in the (name) part and what kind of info do they give? using $stat(x) can be done if you load pv module. There are predefined statistics implemented in modules -- these ones are documented in the readmes. Then you can define extra statistics from config file via statistics module and operate them in routing blocks. To see what statistics are available on your kamailio running instance, do: kamctl fifo get_statistics all In the output you should get a prefix on statistics name which should lead you to the module exporting them -- if there is no such module, they are exported by the core, kex module or they are custom. Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] 503 on DB Error
Hello, On 12/10/11 11:36 AM, Olle E. Johansson wrote: 9 dec 2011 kl. 18:54 skrev Spencer Thomason: Hello all, Is it possible to configure Kamailio to reply to a request with a 503 if it cannot connect to a necessary database for the operation? I.e. Kamailio cannot connect to the database, when a REGISTER comes in and userloc can't find the credentials (due to DB failure), respond with a 503. The requesting endpoint can then fail over properly. The same behaviour would be desired for other modules as well, i.e. aliasdb. I noticed using 3.1.5 that it simply does not authorize the user. I have been thinking about this as well. We could implement a db_ping function so we could test in the routing script before calling functions that use the DB. It's harder with stuff that happens in the background, like ACC and SIPTRACE. I don't know what happens with them if the database fails. one option would be to set a config env variable, like db_error, which is set by the db modules on error. Then can be checked in the config and act accordingly. This will require touching the db modules. Alternative is to propagate some return codes up to config file, but this will require changes in all modules interacting with database, including the db modules. Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] route return value confusion
Hello,
cannot test it for now personally, but what you get if you do:
route(TEST_ROUTE_MINUS_ONE);
$var(r) = $rc;
xlog("returned code by route minus one is: $var(r)\n");
Maybe this will give some leads I can follow in the code...
Cheers,
Daniel
On 12/10/11 3:16 PM, Juha Heinanen wrote:
i have defined two routes:
route [TEST_ROUTE_MINUS_ONE] {
return (-1);
}
route [TEST_ROUTE_PLUS_ONE] {
return (1);
}
and then test them with these statements:
if (route(TEST_ROUTE_MINUS_ONE) == -1) {
xlog("L_INFO", "TEST_ROUTE returned -1\n");
}
if (!route(TEST_ROUTE_MINUS_ONE)) {
xlog("L_INFO", "TEST_ROUTE returned failure\n");
}
if (route(TEST_ROUTE_PLUS_ONE) == 1) {
xlog("L_INFO", "TEST_ROUTE returned 1\n");
}
if (route(TEST_ROUTE_PLUS_ONE)) {
xlog("L_INFO", "TEST_ROUTE returned success\n");
}
can someone explain, why i get only three lines to syslog?
Dec 10 16:14:56 sip /usr/sbin/sip-proxy[16099]: INFO: TEST_ROUTE_MINUS_ONE
returned failure
Dec 10 16:14:56 sip /usr/sbin/sip-proxy[16099]: INFO: TEST_ROUTE_PLUS_ONE
returned 1
Dec 10 16:14:56 sip /usr/sbin/sip-proxy[16099]: INFO: TEST_ROUTE_PLUS_ONE
returned success
-- juha
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Re: [SR-Users] Music on hold with freeswitch as SBC
Hello, On 12/9/11 9:04 PM, Gautam Batra wrote: Hello, I have a kamailio sip proxy server with freeswitch acting as SBC. I want to redirect the call to freeswitch when hold is pressed so that i can play music on hold. I tried this by using rewritehostport in case of a re-invite, but the call drops in that case. Could someone please help me with this? it is not possible to redirect established calls (it breaks the RFC3261), you have to route the call through freeswitch from its start. Perhaps you can use freeswitch without relaying the media in first place and when you have on hold, you set media patch to go through freeswitch. Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] set host as inactive and probing in dispatcher
Hello, you have to upgrade to 3.2.1 (or use the latest git branch 3.2). The current keepalive mechanism was refactored a bit after the release of 3.2.0, as they were overlapping cases. Cheers, Daniel On 12/12/11 11:13 AM, Uri Shacked wrote: version : version: kamailio 3.2.0 (i386/linux) e19bb8 flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 4MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: e19bb8 compiled on 18:43:11 Oct 24 2011 with gcc 4.1.2 and with log level 3 : 13(11897) ERROR:
Re: [SR-Users] 503 on DB Error
Hello, On 12/12/11 10:49 AM, Olle E. Johansson wrote: 12 dec 2011 kl. 10:33 skrev Daniel-Constantin Mierla: Hello, On 12/10/11 11:36 AM, Olle E. Johansson wrote: 9 dec 2011 kl. 18:54 skrev Spencer Thomason: Hello all, Is it possible to configure Kamailio to reply to a request with a 503 if it cannot connect to a necessary database for the operation? I.e. Kamailio cannot connect to the database, when a REGISTER comes in and userloc can't find the credentials (due to DB failure), respond with a 503. The requesting endpoint can then fail over properly. The same behaviour would be desired for other modules as well, i.e. aliasdb. I noticed using 3.1.5 that it simply does not authorize the user. I have been thinking about this as well. We could implement a db_ping function so we could test in the routing script before calling functions that use the DB. It's harder with stuff that happens in the background, like ACC and SIPTRACE. I don't know what happens with them if the database fails. one option would be to set a config env variable, like db_error, which is set by the db modules on error. Then can be checked in the config and act accordingly. This will require touching the db modules. Alternative is to propagate some return codes up to config file, but this will require changes in all modules interacting with database, including the db modules. Great. I think a db_ping which forces a test, a db_status variable that shows the result of the last database operation is a good start. yes, db_ping() is good for interaction from config file. I thought of db_status more for the other functions that use db behind (like auth_db). I wonder if each sql server has this ping capability -- mysql lib offers mysql_ping() (which is used in db_mysql module), iirc, postgress has as well although not used in db_postgres (afaik). If all, ping function may be exported via db interface and offered to config from sqlops, where is easy to provide connection id. When ping is not offered by library, then maybe never fail query like 'select count(*) from version' can be used - this should be usable even now with sqlops. Cheers, Daniel The return code stuff is a more long term issue. I'll try to allocate time to see if I can get somewhere with these two. If I need help, this list will now :-) /O ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] route return value confusion
Hello,
On 12/12/11 10:45 AM, Juha Heinanen wrote:
Daniel-Constantin Mierla writes:
cannot test it for now personally, but what you get if you do:
route(TEST_ROUTE_MINUS_ONE);
$var(r) = $rc;
xlog("returned code by route minus one is: $var(r)\n");
Maybe this will give some leads I can follow in the code...
daniel,
i correctly get:
Dec 12 11:42:37 sip /usr/sbin/sip-proxy[1308]: ERROR: returned code by route
minus one is: -1
so it is only the test
if (route(TEST_ROUTE_MINUS_ONE) == -1) ...
that fails.
ok, so looks like IF with a mod function returning negative needs
investigation. I will look over it soon.
Cheers,
Daniel
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Re: [SR-Users] 503 on DB Error
Hello, On 12/12/11 10:46 AM, Alex Hermann wrote: On Monday 12 December 2011, Daniel-Constantin Mierla wrote: On 12/10/11 11:36 AM, Olle E. Johansson wrote: I have been thinking about this as well. We could implement a db_ping function so we could test in the routing script before calling functions that use the DB. It's harder with stuff that happens in the background, like ACC and SIPTRACE. I don't know what happens with them if the database fails. one option would be to set a config env variable, like db_error, which is set by the db modules on error. Then can be checked in the config and act accordingly. This will require touching the db modules. Alternative is to propagate some return codes up to config file, but this will require changes in all modules interacting with database, including the db modules. How about calling an error_route block when a DB error occurs. Maybe with the proper return stack setup so the admin is able to return to the statement following the failed db query if he doesn't want to end the processing with some failure reply. this seems to need also lot of review. It may happen that functions do db operations inside locks or with some temporary states, so an eventual interruption of config file execution has to make sure that the cfg function doing the db operation finishes, to avoid deadlocks, leaks ... Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Via header branch parameter in ACK message not unique
Hello,
On 12/12/11 12:02 PM, Phillman25 Kyriacou wrote:
Dear Daniel
Thanks, this worked!
I will inform my carrier that they are handling the ACK 200 message wrong.
just one question: the syn_branch=0 is used for only stateless ACK
messages?
it is used for all stateless forwarded requests. By its nature, ACK is
always stateless because it does not take any reply, so there is no
dedicated transaction for it -- in other words, no matter how you send
the ACK from config (e.g., t_relay() or forward()), it will be done
stateless.
Cheers,
Daniel
Thanks again for your assistance!
Phillip
On Mon, Dec 12, 2011 at 11:25 AM, Daniel-Constantin Mierla
mailto:mico...@gmail.com>> wrote:
Hello,
On 12/12/11 8:18 AM, Phillman25 Kyriacou wrote:
Dear List
I'm currently facing an issue with a carrier whereby they are
complaining that the via header branch parameter (sip.Via.branch
== "z9hG4bKcydzigwkX") in the ACK messages is not unique which
results in some calls being successful and some terminating after
20 seconds as their sip proxy determines it as a duplicate
request. Is there a way to make the via header branch parameter
in the ACK message unique across every transaction?
set in your config file the global parameter:
syn_branch=0
On the other hand, the behavior of the devises used by carrier is
completely broken. ACK can have a different path than INVITE, so
there should be no match done on Via branch parameter. Since it is
ACK for 200OK, it has to be matched for dialog. Having the RFC3261
cookie in the branch parameter, it has to be matched with (callid,
from tag, to tag).
Cheers,
Daniel
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Re: [SR-Users] set host as inactive and probing in dispatcher
Hello, On 12/12/11 1:35 PM, Uri Shacked wrote: thanks, since i am not so good with linux... :-( sorry after doing GIT like : # git clone –depth 1 git://git.sip-router.org/sip-router <http://git.sip-router.org/sip-router> kamailio # cd kamailio # git checkout -b 3.2 origin/3.2 # make FLAVOUR=kamailio cfg what do i do next? make install? yes, make install. You can read a tutorial about installing kamailio 3.2 from git at: * http://www.kamailio.org/wiki/install/3.2.x/git Cheers, Daniel On Mon, Dec 12, 2011 at 12:18 PM, Daniel-Constantin Mierla mailto:mico...@gmail.com>> wrote: Hello, you have to upgrade to 3.2.1 (or use the latest git branch 3.2). The current keepalive mechanism was refactored a bit after the release of 3.2.0, as they were overlapping cases. Cheers, Daniel On 12/12/11 11:13 AM, Uri Shacked wrote: version : version: kamailio 3.2.0 (i386/linux) e19bb8 flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 4MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: e19bb8 compiled on 18:43:11 Oct 24 2011 with gcc 4.1.2 and with log level 3 : 13(11897) ERROR:
Re: [SR-Users] commands out of sync you can't run this command now
Hello, On 12/12/11 4:09 PM, MingHon wrote: btw im using kamailio 3.1 which one exactly? Just send output of 'kamailio -V'. Also, watch the log messages with debug=3 and get the one related to the error messages to see which module is executing that command. Cheers, Daniel, On Mon, Dec 12, 2011 at 11:08 PM, MingHon <mailto:gming...@gmail.com>> wrote: Hi List, im doing a fresh install, everything installed successfully but when i try to start kamailio with kamctl start i receive this error. when im using mysql 5.5 rpm version i get this error if install with mysql 5.0 rpm i dont receive that error it run perfectly. is mysql 5.5 incompatible yet? Thanks, -- Regards, MingHon -- Regards, MingHon ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] commands out of sync you can't run this command now
Hello, are you using any private developed module or extension? There seems to be a buffer overflow as well. if you don't have any private extension, can you send directly to me the entire log from start of kamailio with following global parameters: debug=3 memlog=4 memdbg=4 Cheers, Daniel On 12/13/11 8:51 AM, MingHon wrote: im using version: kamailio 3.1.4 (i386/linux) after kamctl start it return INFO: Starting Kamailio : INFO: started (pid: 3589) ps aux | grep 3589 root 3609 0.0 0.0 4012 704 pts/0S+ 15:48 0:00 grep 3589 here is the log message. Dec 13 15:43:41 hostname /usr/local/sbin/kamailio[3589]: ERROR: db_mysql [km_dbase.c:120]: driver error on query: Commands out of sync; you can't run this command now Dec 13 15:43:41 hostname /usr/local/sbin/kamailio[3589]: ERROR: [db_query.c:101]: error while submitting query Dec 13 15:43:41 hostname /usr/local/sbin/kamailio[3589]: ERROR: [db.c:366]: error in db_query Dec 13 15:43:41 hostname /usr/local/sbin/kamailio[3589]: ERROR: [db.c:405]: querying version for table location Dec 13 15:43:41 hostname /usr/local/sbin/kamailio[3589]: ERROR: usrloc [dlist.c:491]: error during table version check. Dec 13 15:43:41 hostname /usr/local/sbin/kamailio[3589]: : [mem/q_malloc.c:146]: BUG: qm_*: fragm. 0x838b11c (address 0x838b134) end overwritten(0, 0)! Dec 13 15:43:41 hostname kamailio: ERROR: [daemonize.c:307]: Main process exited before writing to pipe On Tue, Dec 13, 2011 at 12:53 AM, Daniel-Constantin Mierla mailto:mico...@gmail.com>> wrote: Hello, On 12/12/11 4:09 PM, MingHon wrote: btw im using kamailio 3.1 which one exactly? Just send output of 'kamailio -V'. Also, watch the log messages with debug=3 and get the one related to the error messages to see which module is executing that command. Cheers, Daniel, On Mon, Dec 12, 2011 at 11:08 PM, MingHon mailto:gming...@gmail.com>> wrote: Hi List, im doing a fresh install, everything installed successfully but when i try to start kamailio with kamctl start i receive this error. when im using mysql 5.5 rpm version i get this error if install with mysql 5.0 rpm i dont receive that error it run perfectly. is mysql 5.5 incompatible yet? Thanks, -- Regards, MingHon -- Regards, MingHon ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org <mailto:sr-users@lists.sip-router.org> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla --http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda -- Regards, MingHon ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] set host as inactive and probing in dispatcher
Hello, On 12/12/11 7:38 PM, Uri Shacked wrote: I installed with those instructions once... from scratch I thought mayby there is a quicker way (like update in windows msi. :-)) when my system is already configured nicely.. it is even better than windo$. For example, if you use debian (or ubuntu), just set the apt repository to our debs server where we have nighlty builds, see this page: * http://www.kamailio.org/wiki/packages/debs The you do: apt-get install kamailio kamailio-mysql-modules ... Whenever you need upgrade, just do: apt-get update apt-get upgrade The better part is that the uninstall really works and you don't get trojans/viruses. Cheers, Daniel Thanks, I must say i think kamailio is great !! a real SIP application server, perfect for many implementations. Cheers! On Mon, Dec 12, 2011 at 2:51 PM, Daniel-Constantin Mierla mailto:mico...@gmail.com>> wrote: Hello, On 12/12/11 1:35 PM, Uri Shacked wrote: thanks, since i am not so good with linux... :-( sorry after doing GIT like : # git clone --depth 1 git://git.sip-router.org/sip-router <http://git.sip-router.org/sip-router> kamailio # cd kamailio # git checkout -b 3.2 origin/3.2 # make FLAVOUR=kamailio cfg what do i do next? make install? yes, make install. You can read a tutorial about installing kamailio 3.2 from git at: * http://www.kamailio.org/wiki/install/3.2.x/git Cheers, Daniel On Mon, Dec 12, 2011 at 12:18 PM, Daniel-Constantin Mierla mailto:mico...@gmail.com>> wrote: Hello, you have to upgrade to 3.2.1 (or use the latest git branch 3.2). The current keepalive mechanism was refactored a bit after the release of 3.2.0, as they were overlapping cases. Cheers, Daniel On 12/12/11 11:13 AM, Uri Shacked wrote: version : version: kamailio 3.2.0 (i386/linux) e19bb8 flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 4MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: e19bb8 compiled on 18:43:11 Oct 24 2011 with gcc 4.1.2 and with log level 3 : 13(11897) ERROR:
Re: [SR-Users] commands out of sync you can't run this command now
Hello, On 12/13/11 9:24 AM, Daniel-Constantin Mierla wrote: Hello, are you using any private developed module or extension? There seems to be a buffer overflow as well. if you don't have any private extension, can you send directly to me the entire log from start of kamailio with following global parameters: debug=3 memlog=4 memdbg=4 errata - the values of the global parameters were wrong -- actually the above three lines should be: debug=3 memlog=1 memdbg=1 Cheers, Daniel Cheers, Daniel On 12/13/11 8:51 AM, MingHon wrote: im using version: kamailio 3.1.4 (i386/linux) after kamctl start it return INFO: Starting Kamailio : INFO: started (pid: 3589) ps aux | grep 3589 root 3609 0.0 0.0 4012 704 pts/0S+ 15:48 0:00 grep 3589 here is the log message. Dec 13 15:43:41 hostname /usr/local/sbin/kamailio[3589]: ERROR: db_mysql [km_dbase.c:120]: driver error on query: Commands out of sync; you can't run this command now Dec 13 15:43:41 hostname /usr/local/sbin/kamailio[3589]: ERROR: [db_query.c:101]: error while submitting query Dec 13 15:43:41 hostname /usr/local/sbin/kamailio[3589]: ERROR: [db.c:366]: error in db_query Dec 13 15:43:41 hostname /usr/local/sbin/kamailio[3589]: ERROR: [db.c:405]: querying version for table location Dec 13 15:43:41 hostname /usr/local/sbin/kamailio[3589]: ERROR: usrloc [dlist.c:491]: error during table version check. Dec 13 15:43:41 hostname /usr/local/sbin/kamailio[3589]: : [mem/q_malloc.c:146]: BUG: qm_*: fragm. 0x838b11c (address 0x838b134) end overwritten(0, 0)! Dec 13 15:43:41 hostname kamailio: ERROR: [daemonize.c:307]: Main process exited before writing to pipe On Tue, Dec 13, 2011 at 12:53 AM, Daniel-Constantin Mierla mailto:mico...@gmail.com>> wrote: Hello, On 12/12/11 4:09 PM, MingHon wrote: btw im using kamailio 3.1 which one exactly? Just send output of 'kamailio -V'. Also, watch the log messages with debug=3 and get the one related to the error messages to see which module is executing that command. Cheers, Daniel, On Mon, Dec 12, 2011 at 11:08 PM, MingHon mailto:gming...@gmail.com>> wrote: Hi List, im doing a fresh install, everything installed successfully but when i try to start kamailio with kamctl start i receive this error. when im using mysql 5.5 rpm version i get this error if install with mysql 5.0 rpm i dont receive that error it run perfectly. is mysql 5.5 incompatible yet? Thanks, -- Regards, MingHon -- Regards, MingHon ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org <mailto:sr-users@lists.sip-router.org> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla --http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda -- Regards, MingHon ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla --http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] commands out of sync you can't run this command now
Hello,
have you sent them to me or to mailing list? On the mailing list I could
not find an attachment, also I didn't get an email on myself only.
Cheers,
Daniel
On 12/13/11 10:39 AM, MingHon wrote:
Hello,
i attached the log/messages files please take a look...
Cheers,
MingHon
On Tue, Dec 13, 2011 at 4:58 PM, Daniel-Constantin Mierla
mailto:mico...@gmail.com>> wrote:
Hello,
On 12/13/11 9:24 AM, Daniel-Constantin Mierla wrote:
Hello,
are you using any private developed module or extension?
There seems to be a buffer overflow as well.
if you don't have any private extension, can you send
directly to me the entire log from start of kamailio with
following global parameters:
debug=3
memlog=4
memdbg=4
errata - the values of the global parameters were wrong --
actually the above three lines should be:
debug=3
memlog=1
memdbg=1
Cheers,
Daniel
Cheers,
Daniel
On 12/13/11 8:51 AM, MingHon wrote:
im using version: kamailio 3.1.4 (i386/linux)
after kamctl start it return
INFO: Starting Kamailio :
INFO: started (pid: 3589)
ps aux | grep 3589
root 3609 0.0 0.0 4012 704 pts/0S+ 15:48
0:00 grep 3589
here is the log message.
Dec 13 15:43:41 hostname /usr/local/sbin/kamailio[3589]:
ERROR: db_mysql [km_dbase.c:120]: driver error on query:
Commands out of sync; you can't run this command now
Dec 13 15:43:41 hostname /usr/local/sbin/kamailio[3589]:
ERROR: [db_query.c:101]: error while submitting query
Dec 13 15:43:41 hostname /usr/local/sbin/kamailio[3589]:
ERROR: [db.c:366]: error in db_query
Dec 13 15:43:41 hostname /usr/local/sbin/kamailio[3589]:
ERROR: [db.c:405]: querying version for table location
Dec 13 15:43:41 hostname /usr/local/sbin/kamailio[3589]:
ERROR: usrloc [dlist.c:491]: error during table version check.
Dec 13 15:43:41 hostname /usr/local/sbin/kamailio[3589]: :
[mem/q_malloc.c:146]: BUG: qm_*: fragm. 0x838b11c
(address 0x838b134) end overwritten(0, 0)!
Dec 13 15:43:41 hostname kamailio: ERROR:
[daemonize.c:307]: Main process exited before writing to pipe
On Tue, Dec 13, 2011 at 12:53 AM, Daniel-Constantin Mierla
mailto:mico...@gmail.com>> wrote:
Hello,
On 12/12/11 4:09 PM, MingHon wrote:
btw im using kamailio 3.1
which one exactly? Just send output of 'kamailio -V'.
Also, watch the log messages with debug=3 and get the
one related to the error messages to see which module is
executing that command.
Cheers,
Daniel,
On Mon, Dec 12, 2011 at 11:08 PM, MingHon
mailto:gming...@gmail.com>> wrote:
Hi List,
im doing a fresh install, everything installed
successfully but when
i try to start kamailio with kamctl start i receive
this error.
when im using mysql 5.5 rpm version i get this
error
if install with mysql 5.0 rpm i dont receive that
error it run perfectly.
is mysql 5.5 incompatible yet?
Thanks,
--
Regards,
MingHon
--
Regards,
MingHon
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Re: [SR-Users] set host as inactive and probing in dispatcher
On 12/13/11 11:45 AM, Uri Shacked wrote: thanks, I work with centos... rpms for latest stable release of 3.2.x series are generated and available for download, details about distros at: http://www.kamailio.org/wiki/packages/rpms Cheers, Daniel On Tue, Dec 13, 2011 at 10:31 AM, Daniel-Constantin Mierla mailto:mico...@gmail.com>> wrote: Hello, On 12/12/11 7:38 PM, Uri Shacked wrote: I installed with those instructions once... from scratch I thought mayby there is a quicker way (like update in windows msi. :-)) when my system is already configured nicely.. it is even better than windo$. For example, if you use debian (or ubuntu), just set the apt repository to our debs server where we have nighlty builds, see this page: * http://www.kamailio.org/wiki/packages/debs The you do: apt-get install kamailio kamailio-mysql-modules ... Whenever you need upgrade, just do: apt-get update apt-get upgrade The better part is that the uninstall really works and you don't get trojans/viruses. Cheers, Daniel Thanks, I must say i think kamailio is great !! a real SIP application server, perfect for many implementations. Cheers! On Mon, Dec 12, 2011 at 2:51 PM, Daniel-Constantin Mierla mailto:mico...@gmail.com>> wrote: Hello, On 12/12/11 1:35 PM, Uri Shacked wrote: thanks, since i am not so good with linux... :-( sorry after doing GIT like : # git clone –depth 1 git://git.sip-router.org/sip-router <http://git.sip-router.org/sip-router> kamailio # cd kamailio # git checkout -b 3.2 origin/3.2 # make FLAVOUR=kamailio cfg what do i do next? make install? yes, make install. You can read a tutorial about installing kamailio 3.2 from git at: * http://www.kamailio.org/wiki/install/3.2.x/git Cheers, Daniel On Mon, Dec 12, 2011 at 12:18 PM, Daniel-Constantin Mierla mailto:mico...@gmail.com>> wrote: Hello, you have to upgrade to 3.2.1 (or use the latest git branch 3.2). The current keepalive mechanism was refactored a bit after the release of 3.2.0, as they were overlapping cases. Cheers, Daniel On 12/12/11 11:13 AM, Uri Shacked wrote: version : version: kamailio 3.2.0 (i386/linux) e19bb8 flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 4MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: e19bb8 compiled on 18:43:11 Oct 24 2011 with gcc 4.1.2 and with log level 3 : 13(11897) ERROR:
Re: [SR-Users] route return value confusion
Hello,
I checked the code and this behaviour is practically due to special
meaning of return codes from functions exported to config file, ie,:
- <0 is false
- >0 is true
route(x) call is returning the code from route block itself, but then
the interpreter converts the codes < 0 to 0 (so the condition is false
in C) and the codes > 0 to 1 (so the condition is C is true).
Thus, no matter what code you return from a subroute, you have to
compare against 0 (for negative return code) or 1 (for positive return
code).
Solutions/alternatives:
1)
$var(r) = route(x);
.. and then compare $var(r)
2)
route(x);
$var(r) = $rc;
.. and then compare $var(r)
3)
doing it in the if:
if(($var(r) = route(x)) && ($var(r) == -1)) {
...
}
The assignment which is done first in the IF expression will return true
if the assignment operation is successful.
Cheers,
Daniel
just replace -1 with the return code you want to test against
On 12/12/11 11:29 AM, Daniel-Constantin Mierla wrote:
Hello,
On 12/12/11 10:45 AM, Juha Heinanen wrote:
Daniel-Constantin Mierla writes:
cannot test it for now personally, but what you get if you do:
route(TEST_ROUTE_MINUS_ONE);
$var(r) = $rc;
xlog("returned code by route minus one is: $var(r)\n");
Maybe this will give some leads I can follow in the code...
daniel,
i correctly get:
Dec 12 11:42:37 sip /usr/sbin/sip-proxy[1308]: ERROR: returned code
by route minus one is: -1
so it is only the test
if (route(TEST_ROUTE_MINUS_ONE) == -1) ...
that fails.
ok, so looks like IF with a mod function returning negative needs
investigation. I will look over it soon.
Cheers,
Daniel
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Re: [SR-Users] route return value confusion
Hello,
On 12/12/11 11:15 AM, Morten Isaksen wrote:
I had a similar problem with avp_db_query
if (avp_db_query(...) == -2) {
xlog("...");
}
did not work, but
$var(r) = avp_db_query();
if ($var(r) == -2) {
xlog("...");
}
did work.
indeed, this is a solution. As stated in the previous email, inside an
IF expression, the return code from functions exported to config are
converted to 0 if return code is < 0 and to 1 if the return code is >0
to match in C the special meaning of return codes in the config file.
Cheers,
Daniel
/Morten
On Sat, Dec 10, 2011 at 3:16 PM, Juha Heinanen wrote:
i have defined two routes:
route [TEST_ROUTE_MINUS_ONE] {
return (-1);
}
route [TEST_ROUTE_PLUS_ONE] {
return (1);
}
and then test them with these statements:
if (route(TEST_ROUTE_MINUS_ONE) == -1) {
xlog("L_INFO", "TEST_ROUTE returned -1\n");
}
if (!route(TEST_ROUTE_MINUS_ONE)) {
xlog("L_INFO", "TEST_ROUTE returned failure\n");
}
if (route(TEST_ROUTE_PLUS_ONE) == 1) {
xlog("L_INFO", "TEST_ROUTE returned 1\n");
}
if (route(TEST_ROUTE_PLUS_ONE)) {
xlog("L_INFO", "TEST_ROUTE returned success\n");
}
can someone explain, why i get only three lines to syslog?
Dec 10 16:14:56 sip /usr/sbin/sip-proxy[16099]: INFO: TEST_ROUTE_MINUS_ONE
returned failure
Dec 10 16:14:56 sip /usr/sbin/sip-proxy[16099]: INFO: TEST_ROUTE_PLUS_ONE
returned 1
Dec 10 16:14:56 sip /usr/sbin/sip-proxy[16099]: INFO: TEST_ROUTE_PLUS_ONE
returned success
-- juha
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Re: [SR-Users] route return value confusion
On 12/13/11 1:03 PM, Juha Heinanen wrote:
Daniel-Constantin Mierla writes:
Thus, no matter what code you return from a subroute, you have to
compare against 0 (for negative return code) or 1 (for positive return
code).
Solutions/alternatives:
is it not an alternative to fix to change the current implementation so
that the real return code of route block could be tested?
coding wise might not be a lot, thus I haven't analyzed all the cases
where functions are evaluated by return code, but it will impact and
break config file as it used to be so far, for example:
route[x] {
return -1;
}
if(route(x)) will be true, since if(-1) is true in C
All conditions with functions have to be changed in:
if( f()<0 ) for false cases
if( f(x) > 0) for true cases
At this moment I cannot say for sure if there can be done some
workaround to get return code and keep old behavior, at first sight
seems unlikely.
Cheers,
Daniel
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Re: [SR-Users] route return value confusion
On 12/13/11 1:25 PM, Juha Heinanen wrote:
Daniel-Constantin Mierla writes:
coding wise might not be a lot, thus I haven't analyzed all the cases
where functions are evaluated by return code, but it will impact and
break config file as it used to be so far, for example:
route[x] {
return -1;
}
if(route(x)) will be true, since if(-1) is true in C
i meant get rid of C interpretation when if evaluates the condition,
i.e., do not covert condition to anything, but just test if condition is
< 0 for false and> 0 for true.
This is what happens actually:
if(route(x) == 1) is seen by interpreter as:
if(left_expression == 1)
then left_expression is evaluated as you say above:
- < 0 is false => this is 0 is C
- > 0 is true => this is 1 in C
Since left_expression can be complex (include sub expressions), there
has to be propagation to false/true. For example you can have in the
left expression (2<1) or variables with such values.
Anyhow, if you have some ideas, you can play with functions in lvalue.c,
rvalue.c and action.c
Cheers,
Daniel
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Re: [SR-Users] commands out of sync you can't run this command now
I cannot see any DEBUG messages, are you sure the debug=3 in your config and you dont have other -d command line parameters? Cheers, Daniel On 12/13/11 12:58 PM, MingHon wrote: Hi, the attachment was 2mb i remove it from approval. i did sentto ur email too.. anyway i attached the file with 7 zip.. reduced to 60k now.. -- Regards, MingHon On Tue, Dec 13, 2011 at 7:00 PM, Daniel-Constantin Mierla mailto:mico...@gmail.com>> wrote: Hello, have you sent them to me or to mailing list? On the mailing list I could not find an attachment, also I didn't get an email on myself only. Cheers, Daniel On 12/13/11 10:39 AM, MingHon wrote: Hello, i attached the log/messages files please take a look... Cheers, MingHon On Tue, Dec 13, 2011 at 4:58 PM, Daniel-Constantin Mierla mailto:mico...@gmail.com>> wrote: Hello, On 12/13/11 9:24 AM, Daniel-Constantin Mierla wrote: Hello, are you using any private developed module or extension? There seems to be a buffer overflow as well. if you don't have any private extension, can you send directly to me the entire log from start of kamailio with following global parameters: debug=3 memlog=4 memdbg=4 errata - the values of the global parameters were wrong -- actually the above three lines should be: debug=3 memlog=1 memdbg=1 Cheers, Daniel Cheers, Daniel On 12/13/11 8:51 AM, MingHon wrote: im using version: kamailio 3.1.4 (i386/linux) after kamctl start it return INFO: Starting Kamailio : INFO: started (pid: 3589) ps aux | grep 3589 root 3609 0.0 0.0 4012 704 pts/0S+ 15:48 0:00 grep 3589 here is the log message. Dec 13 15:43:41 hostname /usr/local/sbin/kamailio[3589]: ERROR: db_mysql [km_dbase.c:120]: driver error on query: Commands out of sync; you can't run this command now Dec 13 15:43:41 hostname /usr/local/sbin/kamailio[3589]: ERROR: [db_query.c:101]: error while submitting query Dec 13 15:43:41 hostname /usr/local/sbin/kamailio[3589]: ERROR: [db.c:366]: error in db_query Dec 13 15:43:41 hostname /usr/local/sbin/kamailio[3589]: ERROR: [db.c:405]: querying version for table location Dec 13 15:43:41 hostname /usr/local/sbin/kamailio[3589]: ERROR: usrloc [dlist.c:491]: error during table version check. Dec 13 15:43:41 hostname /usr/local/sbin/kamailio[3589]: : [mem/q_malloc.c:146]: BUG: qm_*: fragm. 0x838b11c (address 0x838b134) end overwritten(0, 0)! Dec 13 15:43:41 hostname kamailio: ERROR: [daemonize.c:307]: Main process exited before writing to pipe On Tue, Dec 13, 2011 at 12:53 AM, Daniel-Constantin Mierla mailto:mico...@gmail.com>> wrote: Hello, On 12/12/11 4:09 PM, MingHon wrote: btw im using kamailio 3.1 which one exactly? Just send output of 'kamailio -V'. Also, watch the log messages with debug=3 and get the one related to the error messages to see which module is executing that command. Cheers, Daniel, On Mon, Dec 12, 2011 at 11:08 PM, MingHon mailto:gming...@gmail.com>> wrote: Hi List, im doing a fresh install, everything installed successfully but when i try to start kamailio with kamctl start i receive this error. when im using mysql 5.5 rpm version i get this error if install with mysql 5.0 rpm i dont receive that error it run perfectly. is mysql 5.5 incompatible yet? Thanks, -- Regards, MingHon -- Regards, MingHon ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org <mailto:sr-users@lists.sip-router.org> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Fwd: commands out of sync you can't run this command now
On 12/13/11 3:33 PM, MingHon wrote:
ya debug is set to 3 already.. also #!define WITH_DEBUG
dont hav any -d command..
Interesting, maybe your syslog is diverting DEBUG level to a different file.
Anyhow, I was hunting the tail overlapping issue which might be related
or not to db connection sharing (which is the reason for out of sync) --
I wanted to be sure not to have another bug around.
The out of sync issue should be solved by kamailio 3.1.5, you have to
upgrade to it or to latest git branch 3.1.
If you want to pick the patch in your 3.1 sources, here is the link to
the commit for this fix:
http://git.sip-router.org/cgi-bin/gitweb.cgi/sip-router/?a=commit;h=97f788a3f1e4e993c99fb1225a1fbcbe4ff0ffc8
Cheers,
Daniel
On Tue, Dec 13, 2011 at 9:06 PM, Daniel-Constantin Mierla
mailto:mico...@gmail.com>> wrote:
I cannot see any DEBUG messages, are you sure the debug=3 in your
config and you dont have other -d command line parameters?
Cheers,
Daniel
On 12/13/11 12:58 PM, MingHon wrote:
Hi,
the attachment was 2mb i remove it from approval. i did sentto ur
email too..
anyway i attached the file with 7 zip.. reduced to 60k now..
--
Regards,
MingHon
On Tue, Dec 13, 2011 at 7:00 PM, Daniel-Constantin Mierla
mailto:mico...@gmail.com>> wrote:
Hello,
have you sent them to me or to mailing list? On the mailing
list I could not find an attachment, also I didn't get an
email on myself only.
Cheers,
Daniel
On 12/13/11 10:39 AM, MingHon wrote:
Hello,
i attached the log/messages files please take a look...
Cheers,
MingHon
On Tue, Dec 13, 2011 at 4:58 PM, Daniel-Constantin
Mierla mailto:mico...@gmail.com>> wrote:
Hello,
On 12/13/11 9:24 AM, Daniel-Constantin Mierla wrote:
Hello,
are you using any private developed module or
extension? There seems to be a buffer overflow as well.
if you don't have any private extension, can you
send directly to me the entire log from start of
kamailio with following global parameters:
debug=3
memlog=4
memdbg=4
errata - the values of the global parameters were
wrong -- actually the above three lines should be:
debug=3
memlog=1
memdbg=1
Cheers,
Daniel
Cheers,
Daniel
On 12/13/11 8:51 AM, MingHon wrote:
im using version: kamailio 3.1.4 (i386/linux)
after kamctl start it return
INFO: Starting Kamailio :
INFO: started (pid: 3589)
ps aux | grep 3589
root 3609 0.0 0.0 4012 704 pts/0S+
15:48 0:00 grep 3589
here is the log message.
Dec 13 15:43:41 hostname
/usr/local/sbin/kamailio[3589]: ERROR: db_mysql
[km_dbase.c:120]: driver error on query: Commands
out of sync; you can't run this command now
Dec 13
15:43:41 hostname /usr/local/sbin/kamailio[3589]:
ERROR: [db_query.c:101]: error while
submitting query
Dec 13
15:43:41 hostname /usr/local/sbin/kamailio[3589]:
ERROR: [db.c:366]: error in db_query
Dec 13
15:43:41 hostname /usr/local/sbin/kamailio[3589]:
ERROR: [db.c:405]: querying version for
table location
Dec 13
15:43:41 hostname /usr/local/sbin/kamailio[3589]:
ERROR: usrloc [dlist.c:491]: error during table
version check.
Dec 13
15:43:41 hostname /usr/local/sbin/kamailio[3589]:
: [mem/q_malloc.c:146]: BUG: qm_*: fragm.
0x838b11c (address 0x838b134) end overwritten(0, 0)!
Dec 13 15:43:41 hostname kamailio: ERROR:
[daemonize.c:307]: Main process exited before
writing to pipe
On Tue, Dec 13, 2011 at 12:53 AM,
Daniel-Constantin Mierla mailto:mico...@gmail.com>> wrote:
Hello,
On 12/12/11 4:09 PM, MingHon wrote:
btw im using kamailio 3.1
which one exactly? Just send output of
'kamailio -V'.
Also, watch the log messages with debug=3 and
get the one related to the error messages to
see which module is ex
Re: [SR-Users] problem forking to dbaliase
Hello,
On 12/13/11 2:31 PM, Yufei Tao wrote:
Hi
I'm using Kamailio 3.1.5 and want to fork calls to all members in the
same group defined in dbaliase. e.g. alias_username is 'clients' for 3
usernames: 'clientone', 'clienttwo', 'clientthree'.
If any of the 3 users are online, call to the alias_username 'clients'
looks fine: INVITE is forked to all online users etc.
But problem comes when none of the users are online: in this case,
t_relay will cause a server error 500, as server tries to relay to the
user on trunk, e.g. client...@domain.com, for which failed to find a
location entry and uri remaied client...@domain.com rather than an IP
address.
I expected server to send a 404 if non of the users for alias_username
'clients' is online. How should I fix it please?
500 is caused by doing drop to all branches, so tm creates the
transaction, but there is no outgoing branch sent to network.
A nice solution for the future will be to add kind of
t_reply_branch(...) which will set the reply code for the branch.
t_reply() is working at transaction level.
Right now an option you can do is to set r-uri in branch route to
something like 'sip:notfound@yourserverip' instead of drop and let the
branch go out.
$ru = "sip:notfound@yourserverip";
The at the beginning of request route block you can test:
if(src_ip==yourserverip && $rU="notfound") {
sl_send_reply("404", "not found");
exit;
}
Cheers,
Daniel
Following is relevant parts from my config file:
#
route[LOCATION]
{
if ( alias_db_lookup("dbaliases") )
{
t_on_branch("BRANCH_ALIASDB"); # in branch_route[BRANCH_ALIASDB],
# call another route that looks up
location,
# if not existent, call drop()
}
else
{
xlog("L_DBG","LOCATION: not alias - go to lookup location trunk\n");
route(LOCATION_TRUNK); # normal look up location and sending of 404 etc
}
... ...
}
#
branch_route[BRANCH_ALIASDB]
{
xlog("L_DBG", "BRANCH_ALIASDB: $fU@$fd -> $rU@$rd; Method:$rm\n");
route(LOCATION_BRANCH);
}
route[LOCATION_BRANCH]
{
if (!lookup("location"))
{
# Drop this branch - it's going nowhere
drop();
}
}
#
route[RELAY] {
xlog("L_DBG","RELAY: method=$rm, callid=$ci, cseq=$cs\n");
#!ifdef WITH_NAT
if (check_route_param("nat=yes")) {
setbflag(FLB_NATB);
}
if (isflagset(FLT_NATS) || isbflagset(FLB_NATB)) {
xlog("L_DBG", "RELAY: about to call RTPPROXY\n");
route(RTPPROXY);
}
#!endif
/* example how to enable some additional event routes */
if (is_method("INVITE")) {
t_on_reply("REPLY_ONE");
t_on_failure("FAIL_ONE");
}
if (!t_relay()) {
sl_reply_error();
}
exit;
}
#
And in the main route, route LOCATION and RELAY are the last two routes:
route {
...
# user location service
route(LOCATION);
route(RELAY);
}
Thanks very much!
Yufei
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Re: [SR-Users] syncing db_text acc table
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Re: [SR-Users] Double Invite / Issue
Hello,
On 12/14/11 1:14 AM, Brandon Armstead wrote:
Hello,
I am running into a problem where I am experiencing duplicated
INVITE's being originated from Kamailio proxy.
U 2011/12/13 23:58:17.543802 KAMAILIO:5060 -> PSTN:5060
INVITE sip:URI@PSTN SIP/2.0.
U 2011/12/13 23:58:17.543840 KAMAILIO:5060 -> PSTN:5060
INVITE sip:URI@PSTN SIP/2.0.
As you can see that the timestamp between the two invites is literally
a fraction of seconds.
There is no failure condition that is causing this with append_branch
There is no retransmission timer issue that is causing this (using
default timing):
modparam("tm", "retr_timer1", 500)
modparam("tm", "retr_timer2", 4000)
Any thoughts / ideas - as this is causing a race condition in which
there is a 200 OK that is being sent back from upstream -- and we are
then CANCELING the 2nd INVITE which is essentially causing on overall
problem with the call.
Thanks for all help / thoughts / input in advance, thanks!
if there is a CANCEL for second INVITE, then it is parallel forking with
two branches. If you can provide ngrep with the INVITEs, I can confirm
it properly. Can you check there are not two contacts in usrloc?
Cheers,
Daniel
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Re: [SR-Users] force_send_socket and IPv6
Hello, is the ipv6 address in between square brackets (can you give exact usage of the foce_send_socket())? It should work as the bridging ipv4-ipv6 is using the same mechanism -- may be an issue with the exported function to config, though. Can you try to set $fs as alternative to force_send_socket()? Cheers, Daniel On 12/14/11 5:19 PM, Andreas Granig wrote: On 12/14/2011 05:16 PM, Andreas Granig wrote: Is it possible to use force_send_socket() with an IPv6 address in Kamailio 3.1? I get the error ERROR: fix_actions: force_send_socket: could not resolve X:X:X:X:X:X:X:X:5060 Actually it doesn't include the 5060, so it really says: ERROR: fix_actions: force_send_socket: could not resolve X:X:X:X:X:X:X:X Sorry if that caused a confusion, my mistake when copy/pasting it. Andreas ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] force_send_socket and IPv6
Hello, On 12/14/11 6:28 PM, Andreas Granig wrote: Hey Daniel, On 12/14/2011 06:00 PM, Daniel-Constantin Mierla wrote: is the ipv6 address in between square brackets (can you give exact usage of the foce_send_socket())? It should work as the bridging ipv4-ipv6 is using the same mechanism -- may be an issue with the exported function to config, though. Can you try to set $fs as alternative to force_send_socket()? I tried both ways, like force_send_socket(x::x:5060); and force_send_socket([x::x]:5060); - both resulting in the same error. However, doing $fs="[x::x]:5060"; works perfectly fine, thanks for the hint! ok. A fix for force_send_socket would maybe still be good to not confuse people. I will look over it soon. Cheers, Daniel Thanks, Andreas ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] force_send_socket and IPv6
Hello, On 12/14/11 7:12 PM, Andreas Granig wrote: Hey, One more thing related to outbound routing. Maybe this question is stupid, but I still have to learn a lot regarding IPv6 :) So my Jitsi registered fine, the contact in location table looks like this: contact: sip:testuser1@[x::x]:5060;transport=udp;registering_acc=192_168_51_133 When I call this AoR (from an IPv4 subscriber, but that shouldn't matter), it's being loaded into R-URI, I select my IPv6 socket using $fs, then do the relay. However, I get this error in my logs (first line is my log message before t_relay): INFO:
Re: [SR-Users] Bug: Dialog State=*Teminated* even when the call is active
Hello,
On 12/15/11 1:32 AM, Gnaneshwar Gatla wrote:
Hello,
I have come across this behavior in Kamailio 3.1.5 version.
The proxy is behind a Firewall and uses the advertised IP to
communicate to sipclients.
I have used the "dlg_match_mode", as the record_route_preset() does
not include the DID during call initiation.
modparam("dialog", "dlg_match_mode", 2)
I'm also using pua_dialoginfo to notify users about the
"Event:dialog". I have used the attached patch to make pua_dialog info
aware of the dialog
I see that the dialog is being created, and the pua_dialoginfo
publishes with state "trying" and "confirmed".
When the call is cancelled/408 request is generated, the
pua_dialoginfo generates the xml with state=terminated.
But when the call is answered, the dialog module updates the dialog to
state=4, according to the documentation mentioned below state=4 -->
Terminated.
in which documentation says state=4 is terminated? The one for the new
dialog design? If yes, that is not yet implemented. In current dialog
module, the state=4 is CONFIRMED.
I checked the logs and the bye is processed by the dialog module, but
there is no callback to pua_dialoginfo executed. You mention above about
a patch to pua_dialogninfo, but it is not attached, only test.zip which
has the debug logs. Can you send the patch or be sure it does not break
registering the callbacks for pua_dialoginfo?
Cheers,
Daniel
I'm not aware of the event types that fires the state to termination
(I have not found the documentation about events in the dialog module).
When a BYE is received to the same call, the BYE is processed and the
dialog is cleared from the memory. But the pua_dialoginfo never gets
the callback for the termination and the xml is never updated.
I have attached a log for a call, please find the attached document.
I have come across a documentation about dialog module:
http://www.kamailio.org/dokuwiki/doku.php/modules-new-design:dialog-module-design
Regards
Gnaneshwar
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Re: [SR-Users] How to enable rtp repacketization
Hello,
On 12/14/11 11:52 AM, Arif Tuhin wrote:
I want to enable the rtp repacketization feature of the rtpproxy. I
have set force_rtp_proxy("z150"). But the repacketization is not
happening. in the log it does show that rtp will be repacketizes but
when i see the wireshark capture, its the same.
I havent used this feature so far, thus I cannot help, but if nobody
else here can help with this issue, maybe it is better to ask on
rtpproxy mainling list:
http://lists.rtpproxy.org/mailman/listinfo/users
Cheers,
Daniel
With Best Regards
Ariful Hossain Tuhin
email: 1. etothepowe...@gmail.com <mailto:etothepowe...@gmail.com> 2.
etothepowe...@hotmail.com <mailto:etothepowe...@hotmail.com>
3.etothepowe...@yahoo.com <mailto:3.etothepowe...@yahoo.com>
skype: freeburn1986
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Re: [SR-Users] force_send_socket and IPv6
Hello, On 12/15/11 11:26 AM, Andreas Granig wrote: On 12/15/2011 11:10 AM, Andreas Granig wrote: On 12/15/2011 10:19 AM, Daniel-Constantin Mierla wrote: ERROR: tm [ut.h:295]: ERROR: uri2dst: failed to resolve "[x::x]" :unresolvable A or request (-7) Is there something I haven't considered yet? Why would it try a DNS lookup for this R-URI? is it failing over all or is the request forwarded? It's failing overall with "478 Unresolvable destination (478/TM)". Actually I get two messages back from the outbound load-balancer towards the proxy for the same INVITE, one right after the other: U 127.0.0.1:5060 -> 127.0.0.1:5080 SIP/2.0 500 I'm terribly sorry, server error occurred (1/SL). U 127.0.0.1:5060 -> 127.0.0.1:5080 SIP/2.0 478 Unresolvable destination (478/TM). The scenario is this: [v4user]-(ipv4)->[lb]-(ipv4)->[proxy]-(ipv4)->[lb]-(ipv6)->[v6user] The problem appears on the last lb hop, so it never sends anything over IPv6 to the v6user, but passes back the two errors from above to the proxy. I'm trying to dig deeper later today. trying to figure out the modules involved -- does the INVITE arrive with double Route headers on second [lb] or is the [lb] selecting an ipv6 destination? Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] How to get dispatcher list with sercmd command
Hello, On 12/15/11 2:50 PM, Timo Klecker wrote: Hi Mino Try Sercmd dispatcher list Without the dot. This works with kamctl, haven't used sercmd. in sercmd should be dispatcher.list with dot, but the command is available starting with v3.2.0. What is your kamailio version? Cheers, Daniel Greetings Timo *Von:*sr-users-boun...@lists.sip-router.org [mailto:sr-users-boun...@lists.sip-router.org] *Im Auftrag von *Mino Haluz *Gesendet:* Donnerstag, 15. Dezember 2011 14:31 *An:* SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List *Betreff:* [SR-Users] How to get dispatcher list with sercmd command Hi, # sercmd dispatcher.list error: 500 - command dispatcher.list not found The module is loaded, dispatcher.list file exists. Am I doing something wrong ? Thanks. Mino. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] force_send_socket and IPv6
Hello, On 12/15/11 1:18 PM, marius zbihlei wrote: On 12/14/2011 07:28 PM, Andreas Granig wrote: Hey Daniel, On 12/14/2011 06:00 PM, Daniel-Constantin Mierla wrote: is the ipv6 address in between square brackets (can you give exact usage of the foce_send_socket())? It should work as the bridging ipv4-ipv6 is using the same mechanism -- may be an issue with the exported function to config, though. Can you try to set $fs as alternative to force_send_socket()? I tried both ways, like force_send_socket(x::x:5060); and force_send_socket([x::x]:5060); - both resulting in the same error. However, doing $fs="[x::x]:5060"; works perfectly fine, thanks for the hint! A fix for force_send_socket would maybe still be good to not confuse people. Thanks, Andreas Hello, Actually, looking at the code I've observed some weird behavior : for example to fix the force_send_socket IPv6 address, a call to str2ip6() (from resolve.h) is done (via resolvehost()). The function tries to manually parse the IP address, so it makes me wonder why not use a library function like inet_pton(AF_INET6... to get the correct structure. A possible bug is that with the current code, IPv4 mapped to IPv6 (x:x:x:x:x:x:d.d.d.d) will not be parsed correctly. Why don't we use inet_pton(AF_INET6 ... . Any reasons? have you checked if inet_pton() doesn't break the internal ip_addr structure and the use of it? If all is ok, I guess inet_pton() is portable enough through most unix/linux distro's to make use of it. Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] How to get dispatcher list with sercmd command
On 12/15/11 3:45 PM, Mino Haluz wrote:
kamctl dispatcher dump
did the thing.
that is using fifo file and MI command.
sercmd is doing the RPC command. Can you give the output of:
sercmd system.listMethods
Cheers,
Daniel
kamailio 3.2.0.
On Thu, Dec 15, 2011 at 3:38 PM, Daniel-Constantin Mierla
mailto:mico...@gmail.com>> wrote:
Hello,
On 12/15/11 2:50 PM, Timo Klecker wrote:
Hi Mino
Try
Sercmd dispatcher list
Without the dot. This works with kamctl, haven’t used sercmd.
in sercmd should be dispatcher.list with dot, but the command is
available starting with v3.2.0. What is your kamailio version?
Cheers,
Daniel
Greetings
Timo
*Von:*sr-users-boun...@lists.sip-router.org
<mailto:sr-users-boun...@lists.sip-router.org>
[mailto:sr-users-boun...@lists.sip-router.org] *Im Auftrag von
*Mino Haluz
*Gesendet:* Donnerstag, 15. Dezember 2011 14:31
*An:* SIP Router - Kamailio (OpenSER) and SIP Express Router
(SER) - Users Mailing List
*Betreff:* [SR-Users] How to get dispatcher list with sercmd command
Hi,
# sercmd dispatcher.list
error: 500 - command dispatcher.list not found
The module is loaded, dispatcher.list file exists. Am I doing
something wrong ? Thanks.
Mino.
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Re: [SR-Users] Sending SIP request from mi_fifo using TMX module
Hello,
the headers have to be terminated in \r\n. In a shell script for click
to dial, here is the used form:
cat > $FIFO <"`printf "${FIXED_DLG}To: <$URI>\r\nCSeq: $CSEQ INVITE\r\nContent-Type:
application/sdp\r\n"`
"
"`printf "v=0\r\no=click-to-dial 0 0 IN IP4 0.0.0.0\r\ns=session\r\nc=IN
IP4 0.0.0.0\r\nb=CT:1000\r\nt=0 0\r\nm=audio 9 RTP/AVP 8 0\r\na=rtpmap:8
PCMA/8000\r\na=rtpmap:0 PCMU/8000\r\n"`
"
EOF
Cheers,
Daniel
On 12/16/11 10:02 AM, Sammy Govind wrote:
Hello list,
I'm using Kamailio 3.2 and am trying to send a SIP request from fifo
to kamailio using below format.
#kamctl fifo t_uac_dlg MESSAGE sip:103@ mydomain.org
<http://mydomain.org> . . "From: 1...@mydomain.org
<mailto:1...@mydomain.org> To: 1...@mydomain.org
<mailto:1...@mydomain.org> Contact: mydomain.org <http://mydomain.org>
Content-Type: text/plain; charset=UTF-8" "INVITE"
But I see this reply from Kamctl
*database engine 'MYSQL' loaded*
*Control engine 'FIFO' loaded*
*entering fifo_cmd t_uac_dlg INVITE sip:103@ mydomain.org
<http://mydomain.org> . . From: 101@ mydomain.org
<http://mydomain.org> To: 103@ mydomain.org
<http://mydomain.org> Contact: mydomain.org
<http://mydomain.org> Content-Type: text/plain; charset=UTF-8 INVITE*
*400 Bad headers*
*FIFO command was:*
*:t_uac_dlg:openser_receiver_25561*
*INVITE*
*sip:1...@mydomain.org <mailto:sip%3a...@mydomain.org>*
*.*
*.*
*From: 101@ mydomain.org <http://mydomain.org> To: 103@ mydomain.org
<http://mydomain.org> Contact: mydomain.org
<http://mydomain.org> Content-Type: text/plain; charset=UTF-8*
*INVITE*
And in log file I see this output:
Dec 16 00:57:46 Kamailio3 /usr/local/sbin/kamailio[24541]: ERROR:
[parser/msg_parser.c:275]: ERROR: get_hdr_field: bad body for
mailto:1...@mydomain.org> To: 103@
mydomain.org <http://mydomain.org> Contact: mydomain.org
<http://mydomain.org> Content-Type: text/plain;
charset=UTF-8INVITE#012>(4)
Dec 16 00:57:46 Kamailio3 /usr/local/sbin/kamailio[24541]: INFO:
[parser/msg_parser.c:353]: ERROR: bad header field [From:
101@mydomain.o]
Can anyone help..
Regards,
Sammy
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Re: [SR-Users] How to get dispatcher list with sercmd command
Hello,
On 12/15/11 4:17 PM, Mino Haluz wrote:
Only this worked:
sercmd -s unixs:/tmp/kamailio_ctl system.listMethods
for 3.2.0, sercmd used a different default control socket, so the one
used by kamailio has to be given as parameter. in 3.2.1 that is fixed.
So, now if you do:
sercmd -s unixs:/tmp/kamailio_ctl dispatcher.list
it should work.
Cheers,
Daniel
core.arg
core.echo
core.flags
core.info <http://core.info>
core.kill
core.printi
core.prints
core.ps <http://core.ps>
core.psx
core.pwd
core.sctp_info
core.sctp_options
core.shmmem
core.tcp_info
core.tcp_options
core.udp4_raw_info
core.uptime
core.version
ctl.connections
ctl.listen
ctl.who
dispatcher.list
dispatcher.reload
dispatcher.set_state
dns.add_a
dns.add_
dns.add_srv
dns.debug
dns.debug_all
dns.delete_a
dns.delete_
dns.delete_all
dns.delete_all_force
dns.delete_cname
dns.delete_ebl
dns.delete_naptr
dns.delete_ptr
dns.delete_srv
dns.delete_txt
dns.lookup
dns.mem_info
dns.view
dst_blacklist.add
dst_blacklist.debug
dst_blacklist.delete_all
dst_blacklist.mem_info
dst_blacklist.view
mi
mi_dg
mi_fifo
mi_xmlrpc
pkg.stats
sl.stats
system.listMethods
system.methodHelp
system.methodSignature
tm.cancel
tm.hash_stats
tm.reply
tm.stats
tm.t_uac_start
tm.t_uac_wait
On Thu, Dec 15, 2011 at 4:14 PM, Mino Haluz <mailto:mino.ha...@gmail.com>> wrote:
Do I have to load some specific module ?
ERROR: connect_unix_sock: connect(/tmp/sercmd_ctl): No such file
or directory [2]
On Thu, Dec 15, 2011 at 3:55 PM, Daniel-Constantin Mierla
mailto:mico...@gmail.com>> wrote:
On 12/15/11 3:45 PM, Mino Haluz wrote:
kamctl dispatcher dump
did the thing.
that is using fifo file and MI command.
sercmd is doing the RPC command. Can you give the output of:
sercmd system.listMethods
Cheers,
Daniel
kamailio 3.2.0.
On Thu, Dec 15, 2011 at 3:38 PM, Daniel-Constantin Mierla
mailto:mico...@gmail.com>> wrote:
Hello,
On 12/15/11 2:50 PM, Timo Klecker wrote:
Hi Mino
Try
Sercmd dispatcher list
Without the dot. This works with kamctl, haven't used
sercmd.
in sercmd should be dispatcher.list with dot, but the
command is available starting with v3.2.0. What is your
kamailio version?
Cheers,
Daniel
Greetings
Timo
*Von:*sr-users-boun...@lists.sip-router.org
<mailto:sr-users-boun...@lists.sip-router.org>
[mailto:sr-users-boun...@lists.sip-router.org] *Im
Auftrag von *Mino Haluz
*Gesendet:* Donnerstag, 15. Dezember 2011 14:31
*An:* SIP Router - Kamailio (OpenSER) and SIP Express
Router (SER) - Users Mailing List
*Betreff:* [SR-Users] How to get dispatcher list with
sercmd command
Hi,
# sercmd dispatcher.list
error: 500 - command dispatcher.list not found
The module is loaded, dispatcher.list file exists. Am I
doing something wrong ? Thanks.
Mino.
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--
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http://linkedin.com/in/miconda -- http://twitter.com/miconda
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Re: [SR-Users] source port used for forwarded requests
On 12/16/11 5:17 AM, Alex Balashov wrote: Maybe, if I understand you correctly, the 'advertised_address' and/or 'advertised_port' settings are the thing for you? No, these parameters are not used for selection of socket, not even for routing. They are the values to be used to build Via headers, instead of using the values from sending socket. mhomed helps also in selection of the right outgoing socket, when the route to destination is possible only from specific sockets. Cheers, Daniel -- This message was painstakingly thumbed out on my mobile, so apologies for brevity, errors, and general sloppiness. Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ On Dec 15, 2011, at 9:42 PM, Ben WIlliams wrote: On Fri, Dec 16, 2011 at 12:26 PM, Alex Balashov wrote: On 12/15/2011 06:25 PM, Ben WIlliams wrote: Hi, My Kamailio server listens on multiple IPs - say IP1, IP2. If a request arrives on one IP, Kamailio uses the same IP as the source address when forwarding, which is good. But is there any way a UA could generate a request to IP1 that gets forwarded by Kamailio with source address IP2 ? The reason I ask is because Asterisk is setup to trust all calls arriving from IP2 but not IP1. Yes, this is the function of force_send_socket() and/or $fs. Thanks Alex. I didn't actually want to change the socket in my kamailio script. Just checking that there is no way the SIP message itself can specify which source IP is used when forwarding. Ben ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Sending SIP request from mi_fifo using TMX module
Hello, if you look to do something like click to dial (ringing user A first, when answering, then refer it to user B (or media server)), then maybe you just use dialog module with dlg_bridge command: http://kamailio.org/docs/modules/stable/modules_k/dialog.html#id2554844 Cheers, Daniel On 12/16/11 1:35 PM, Sammy Govind wrote: Hi, Thanks alot, I found the same stuff inside the *kamailio-3.2/kamailio/examples/kamailio* folder and now trying to implement some changes. Do you think It'll be difficult to originate call using this script and then later once call is Answered bridge it to a Media-Server to do its part !? Thanks alot, Regard, Sammy. On Fri, Dec 16, 2011 at 4:42 PM, Daniel-Constantin Mierla mailto:mico...@gmail.com>> wrote: Hello, the headers have to be terminated in \r\n. In a shell script for click to dial, here is the used form: cat > $FIFO <\r\nCSeq: $CSEQ INVITE\r\nContent-Type: application/sdp\r\n"` " "`printf "v=0\r\no=click-to-dial 0 0 IN IP4 0.0.0.0\r\ns=session\r\nc=IN IP4 0.0.0.0\r\nb=CT:1000\r\nt=0 0\r\nm=audio 9 RTP/AVP 8 0\r\na=rtpmap:8 PCMA/8000\r\na=rtpmap:0 PCMU/8000\r\n"` " EOF Cheers, Daniel On 12/16/11 10:02 AM, Sammy Govind wrote: Hello list, I'm using Kamailio 3.2 and am trying to send a SIP request from fifo to kamailio using below format. #kamctl fifo t_uac_dlg MESSAGE sip:103@ mydomain.org <http://mydomain.org> . . "From: 1...@mydomain.org <mailto:1...@mydomain.org> To: 1...@mydomain.org <mailto:1...@mydomain.org> Contact: mydomain.org <http://mydomain.org> Content-Type: text/plain; charset=UTF-8" "INVITE" But I see this reply from Kamctl *database engine 'MYSQL' loaded* *Control engine 'FIFO' loaded* *entering fifo_cmd t_uac_dlg INVITE sip:103@ mydomain.org <http://mydomain.org> . . From: 101@ mydomain.org <http://mydomain.org> To: 103@ mydomain.org <http://mydomain.org> Contact: mydomain.org <http://mydomain.org> Content-Type: text/plain; charset=UTF-8 INVITE* *400 Bad headers* *FIFO command was:* *:t_uac_dlg:openser_receiver_25561* *INVITE* *sip:1...@mydomain.org <mailto:sip%3a...@mydomain.org>* *.* *.* *From: 101@ mydomain.org <http://mydomain.org> To: 103@ mydomain.org <http://mydomain.org> Contact: mydomain.org <http://mydomain.org> Content-Type: text/plain; charset=UTF-8* *INVITE* And in log file I see this output: Dec 16 00:57:46 Kamailio3 /usr/local/sbin/kamailio[24541]: ERROR: [parser/msg_parser.c:275]: ERROR: get_hdr_field: bad body for mailto:1...@mydomain.org> To: 103@ mydomain.org <http://mydomain.org> Contact: mydomain.org <http://mydomain.org> Content-Type: text/plain; charset=UTF-8INVITE#012>(4) Dec 16 00:57:46 Kamailio3 /usr/local/sbin/kamailio[24541]: INFO: [parser/msg_parser.c:353]: ERROR: bad header field [From: 101@mydomain.o <mailto:101@mydomain.o>] Can anyone help.. Regards, Sammy ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org <mailto:sr-users@lists.sip-router.org> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla --http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda ___________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] using xhttp_rpc to send MESSAGE to SIP user from web server
Hello, On 12/17/11 6:08 PM, Ovidiu Sas wrote: It is rather difficult to properly craft SIP messages in a single line (to properly pass all parameters). Therefore, the xhttp_rpc module does not support this kind of functionality. Asynchronous commands are not implemented by the xhttp_rpc module: http://kamailio.org/docs/modules/devel/modules/xhttp_rpc.html#id2521422 It is better to use the xmlrpc module for this kind of functionality. http://kamailio.org/docs/modules/stable/modules/xmlrpc.html just to add a bit and make clear that xmlrpc module is different that mi_xmlrpc. Practically xmlrpc uses internally the same mechanism as xhttp, so no dependencies on http libs. You need libxml2, but since you deal with presence, you should have it. Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Dispatcher does not send pings
Hello,
is this at least version 3.2.1 (or the latest branch 3.2)?
What do you get with: kamctl fifo ds_list?
Cheers,
Daniel
On 12/19/11 12:49 PM, Mino Haluz wrote:
Hi,
please, could someone specify the conditions when the dispatcher
module sends the INFO pings to the gateways ?
modparam("dispatcher", "flags", 2)
modparam("dispatcher", "dstid_avp", "$avp(s:test)")
modparam("dispatcher", "dst_avp", "$avp(AVP_DST)")
modparam("dispatcher", "grp_avp", "$avp(AVP_GRP)")
modparam("dispatcher", "cnt_avp", "$avp(AVP_CNT)")
modparam("dispatcher", "ds_ping_interval", 30)
modparam("dispatcher", "ds_probing_mode", 1)
modparam("dispatcher", "ds_ping_method", "INFO")
modparam("dispatcher", "ds_ping_from", "sip:001122@test")
And my dispatcher.list is:
1 sip:IP1:5060
1 sip:IP2:5060
Am I missing something? I dumped all the communication and the
kamailio(dispatcher) does not send anything..
Thanks,
Mino
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Re: [SR-Users] using xhttp_rpc to send MESSAGE to SIP user from web server
Hello, On 12/19/11 1:39 PM, Krishna Kurapati wrote: Daniel, Thanks for the clarification. Since mi_xmlrpc is equivalent to mi_fifo, I could use this module to send MESSAGE and INVITE from a remote server. mi_xmlrpc depends on specific versions of libxmlrpc client, on some distros it cannot be found. Also, it has some weaks in security (you have to use the firewall, there is no interaction from config file). You can run any mi command via rpc interface (e.g., sercmd or xmlrpc) if you load the mi_rpc modulem then use mi* RPC command: http://www.kamailio.org/docs/docbooks/3.2.x/rpc_list/rpc_list.html#rpc_exports.mi_rpc For example, running same mi command as done by 'kamctl fifo ds_list' is via sercmd: 'sercmd mi ds_list'. For instant messaging, you can use directly the rpc command tm.t_uac_start or tm.t_uac_wait. If you connect using sercmd, you can use 'help cmd' to read a bit about each rpc command. Cheers, Daniel Krish Kura On Mon, Dec 19, 2011 at 6:27 AM, Daniel-Constantin Mierla mailto:mico...@gmail.com>> wrote: Hello, On 12/17/11 6:08 PM, Ovidiu Sas wrote: It is rather difficult to properly craft SIP messages in a single line (to properly pass all parameters). Therefore, the xhttp_rpc module does not support this kind of functionality. Asynchronous commands are not implemented by the xhttp_rpc module: http://kamailio.org/docs/modules/devel/modules/xhttp_rpc.html#id2521422 It is better to use the xmlrpc module for this kind of functionality. http://kamailio.org/docs/modules/stable/modules/xmlrpc.html just to add a bit and make clear that xmlrpc module is different that mi_xmlrpc. Practically xmlrpc uses internally the same mechanism as xhttp, so no dependencies on http libs. You need libxml2, but since you deal with presence, you should have it. Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Dispatcher does not send pings
Hello,
On 12/19/11 1:35 PM, Mino Haluz wrote:
This is in the log file:
ERROR: dispatcher [dispatcher.c:640]: failover functions used, but
AVPs paraamters required are NULL -- feature disabled
it is not related to ping options. You have to upgrade first to 3.2.1 to
get the ping functionality as written in the readme. 3.2.0 uses old
states where there were overlapping cases and you have to set probing
mode to the gateways to be pinged.
Cheers,
Daniel
and the output of kamctl fifo ds_list:
SET_NO:: 1
SET:: 1
URI:: sip:IP1:5060 flags=AX priority=0 attrs=
URI:: sip:IP2:5060 flags=AX priority=0 attrs=
version: kamailio 3.2.0 (i386/linux)
On Mon, Dec 19, 2011 at 1:29 PM, Daniel-Constantin Mierla
mailto:mico...@gmail.com>> wrote:
Hello,
is this at least version 3.2.1 (or the latest branch 3.2)?
What do you get with: kamctl fifo ds_list?
Cheers,
Daniel
On 12/19/11 12:49 PM, Mino Haluz wrote:
Hi,
please, could someone specify the conditions when the dispatcher
module sends the INFO pings to the gateways ?
modparam("dispatcher", "flags", 2)
modparam("dispatcher", "dstid_avp", "$avp(s:test)")
modparam("dispatcher", "dst_avp", "$avp(AVP_DST)")
modparam("dispatcher", "grp_avp", "$avp(AVP_GRP)")
modparam("dispatcher", "cnt_avp", "$avp(AVP_CNT)")
modparam("dispatcher", "ds_ping_interval", 30)
modparam("dispatcher", "ds_probing_mode", 1)
modparam("dispatcher", "ds_ping_method", "INFO")
modparam("dispatcher", "ds_ping_from", "sip:001122@test")
And my dispatcher.list is:
1 sip:IP1:5060
1 sip:IP2:5060
Am I missing something? I dumped all the communication and the
kamailio(dispatcher) does not send anything..
Thanks,
Mino
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Re: [SR-Users] Suggestions for time utils
Hello,
On 12/19/11 1:58 PM, Andreas Granig wrote:
Hi,
I'd like to implement a couple of helper functions for time handling,
e.g. checking day of week, day of month etc. from within the kamailio
config file. What would you prefer, a new module ("timeutils" maybe?) or
adding it to cfgutils (there are already time-based functions there like
sleep and usleep)?
The idea is to implement time based call-forwards, and an approach could
be to provision various time-related values in usr_preferences table,
and then check it in the config. For example, when doing call-forwards
from Monday to Friday only, I could put this into usr_preferences:
attribute: cf_weekday
value: [1, 2, 3, 4, 5] (each entry is a separate row in usr_preferences)
And in kamailio config, I'd call this:
avp_db_load(...);
if(is_weekday("$avp(s:cf_weekday)")) { do CF }
So is_weekday would iterate over the entries in the avp list and return
true if the weekday at the time of routing matches an entry in the list.
A module config param could control whether to use gmtime or localtime
for matching.
Does this make sense? Suggestions for other approaches? I'd rather
prefer to do it directly in config instead of using some external
interpreter like lua, python etc.
I don't know what are all the functions you think of, but for the
example provided above, config file does it easy right now. There is a
pseudo-variable that gives broken-time attribute that can be used with
avp_check(), iirc, should be:
avp_db_load(...);
if(avp_check("$time(wday)", "eq/$avp(s:cf_weekday)/g")) { do CF }
Of course there is the option of doing while loop, but maybe gets to large for
desired config file.
For me does not matter if it is new module or not, it is up to developer
convenience. The only thing here I would comment a bit about, is the
plan to apply on avp lists always? The maybe its better in avpops (if
you want to do it in an old module), also use avp somehow in the name,
e.g., is_avp_weekday().
Cheers,
Daniel
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Re: [SR-Users] Suggestions for time utils
Hello,
On 12/19/11 7:50 PM, Andreas Granig wrote:
Hi Daniel,
On 12/19/2011 07:29 PM, Daniel-Constantin Mierla wrote:
I don't know what are all the functions you think of, but for the
example provided above, config file does it easy right now. There is a
pseudo-variable that gives broken-time attribute that can be used with
avp_check(), iirc, should be:
avp_db_load(...);
if(avp_check("$time(wday)", "eq/$avp(s:cf_weekday)/g")) { do CF }
Of course there is the option of doing while loop, but maybe gets to
large for desired config file.
This is pretty much what I had in mind with my new functions/module, but
I've completely overlooked that PV when searching the docs for this
feature. Thank you very much for pointing that out!
for sake of public knowledge, just to add on time specific features:
there is also $timef(format) which returns current time attributes based
on strftime specifiers -- its documentation was missing, I just added
it. Also, there is a transformation {s.ftime,format) which can take any
integer variable holding timestamp and return value based on strftime
format.
Cheers,
Daniel
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Re: [SR-Users] Dispatcher module with call load balancing
Hello, On 12/15/11 12:06 PM, Mino Haluz wrote: Hi, I would like to use the dispatcher module with algorithm 10 - call load balancing. But there are multiple things that are a bit unclear to me. I have multiple gateways that can serve different maximum number of calls. 1) Can I somehow set the maximum calls count for each dispatcher gateway, so that it can fairly distribute the calls? I know it is stateless module, but how could I solve this? it is not stateless module - dispatcher has nothing to do with the states, it just selects the destinations. Then, if you relay with tm you make the routing transaction stateful and if you forward() then it is stateless routing. The algorithm does selection of the least loaded destination. There is another algorithm for weight based distribution (iirc 9), might suit better your need. 2) Is there any method how can I get the current number of calls for each gateway from the dispatcher module? Not yet. 3) What is the expiration timer in the dispatcher module for dialogs if the dialog is canceled uncleanly ? http://kamailio.org/docs/modules/stable/modules_k/dispatcher.html#id2498263 4) Imagine I will restart kamailio, it will lose the call count and the distribution will be incorrect. Maybe it could nice, that kamailio would offer MI command to set these values. Some enhancements can be done, of course, it is a matter of time and interest for developers, the fastest way to get something is to contribute, it is open source greatest benefit. Personally I haven;t invested much in enhancing this algorithm lately because it does not bring relevant benefits comparing with round robin. The actual load on a server is not a matter only of the number of active calls, but also the content of the media stream and its processing (e.g., 10 calls on hold consume less resources than 3 active and verbose conversations). The most of the calls are in the average duration of a business type, thus simple round-robin is keeping pretty fair load balancing of the active calls -- from experience, with round-robin none of the destinations got more than 10% load than the average. Regarding the restart, yes, there can be coding to make it more accurate, but if it is call load balancing, practically a restart cannot get a destination to have more than 1 call than it should have, which I think it is not something critical. But again, any contributions in this regard will be considered and accepted when they bring benefits. Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Database connections
Hello, On 12/20/11 12:52 PM, Mariana Arduini wrote: Hello all, I'm running some performance tests and need to know how does the database connections work. Are they open when we start OpenSER and are kept open for each process or they just open before a query and closed afterwards? Does anyone know where I can find such info? most of them are open at startup and kept alive for runtime duration, being closed at shutdown. There is a connection pool per process, so there is only one connection to a database from the same process. In vary rare occasions, when Kamailio is not processing SIP traffic but MI/RPC commands, it may happen to have the connection open, query done and then connection closed. If you want to learn more about DB API, then look at devel guide: http://www.asipto.com/pub/kamailio-devel-guide/#c09database and sources in lib/srdb1 (the one mainly used in kamailio (thus former openser) modules). Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] TLS performance testing
Hello,
On 12/20/11 10:27 PM, Kristian Kielhofner wrote:
Hello everyone,
I'm doing some TLS performance testing on Kamailio 3.2.1. Here's my setup:
kamailio -V
version: kamailio 3.2.1 (x86_64/linux) 31c991
flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS,
USE_RAW_SOCKS, USE_STUN, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK,
SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, USE_FUTEX,
FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR,
USE_DST_BLACKLIST, HAVE_RESOLV_RES
ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 4MB
poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
id: 31c991
compiled on 19:38:03 Dec 20 2011 with gcc 4.4.6
uname -a
Linux null.null.com 2.6.32-220.el6.x86_64 #1 SMP Tue Dec 6 19:48:22
GMT 2011 x86_64 x86_64 x86_64 GNU/Linux
CentOS 6.2 on a Dell PowerEdge R610 with 24 Intel X5650 Cores at
2.67GHz and 12GB of RAM (I could use more).
Kamailio is the default config with a few changes:
- WITH_TLS defined
- TLS is using self-generated CAs/certs (essentially openvpn easy-rsa)
with 1024 bit key size
- TLS is *not* configured to verify client OR server certs by default
- I'm using TLS v1 (SSL 3.1)
- TLS cipher suites are set to any (although my simulated UAs only
offer AES 256+SHA)
- Various changes to Kamailio children (up to 256 at times) and memory
sizes (up to 2048mb and even 4096mb at times)
- One DNS based alias added
- Maximum TCP connections increased to 65000
- Kamailio is configured to only listen on tested IP (UDP, TCP, TLS
sockets active)
- Syslog has been configured to log local0 (Kamailio) asynchronously
My test rig/call generator is an Ixia Xcellon-Ultra NP load module
with IXLoad. My call scenario does the following:
- Registers two simulated user agents (10, 20) to Kamailio with TLS
- Places call from 10 to 20 via Kamailio with TLS
- Increments both user agents by 1 and continues as quickly (cps) as I
like up to a channel limit (also configurable)
- The Ixia generates a valid SDP but no RTP is generated (although
that's certainly possible at these call levels)
Two 1 gig ports on the Ixia are connected to the Broadcom NICs on
the Dell R610 via a Cisco Catalyst 4948 switch. One port on the Ixia
emulates the 10 agents (A leg) and the other emulates the 20
agents (B leg). Of course I can provide more information if needed.
Here are some test numbers:
With TLS at 20cps, 120 sec calls, up to a total of 2470 calls (4940
registrations) life is good. Very good - call setup time averages
23ms, the cps rate holds indefinitely, and not a single call or
registration fails over long term tests.
UDP and TCP numbers are excellent (bordering ridiculous) - usually
around 500cps with practically no reasonable upper limit on
simultaneous calls. This doesn't need any further discussion :).
The TLS numbers start falling apart pretty quickly after 20cps,
however. If I change the TLS test to 40cps, 120 sec calls, up to a
total of 4940 calls (9,880 registrations) Kamailio starts to
(seriously) struggle. The rate starts fluctuating all over the place,
call setup time averages jump to 8000ms (or more) and things just
generally get ugly. Interestingly enough all of the user agents are
able to register, the logs look fine (to my eye at this log level) and
the system (CPU, network, etc) doesn't appear to be under stress at
all.
I have a few questions:
1) Is there something obviously wrong or stupid I'm doing here?
2) Why are the TLS tests so much worse than TCP and UDP? Am I
missing something here?
with tls the limit is usually the cpu or memory, due to
encryption/decryption, but you say they look ok.
What log level are you using in config?
Is the test tool keeping the tls connections open or they are closed and
have to be open for each call? Can you spot if the delay is on incoming
side or on outgoing? Wireshark can decrypt the traffic if you provide
the certificate and start the sniffing before starting tls connections.
Alternative, set:
modparam("tls", "cipher_list", "NULL")
and the traffic should be no longer encrypted, but this will not show if
the bottleneck is the encryption process.
Also, benchmark module can help to see if execution of config file takes
longer than usual.
Jan Janak did tls tests some time ago, the summary is part of:
http://sip-router.org/wiki/performance/v3.0-capacity
I have some results from a test with 6000 SIP messages/sec over tls,
where CPU usage got about 60%. I guess something is becoming a
bottleneck in your case, very likely a blocking operation since cpu is
ok, just has to be discovered.
Cheers,
Daniel
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Re: [SR-Users] TLS performance testing
On 12/20/11 11:55 PM, Kristian Kielhofner wrote:
On Tue, Dec 20, 2011 at 5:20 PM, Daniel-Constantin Mierla
wrote:
Hello,
with tls the limit is usually the cpu or memory, due to
encryption/decryption, but you say they look ok.
What log level are you using in config?
Is the test tool keeping the tls connections open or they are closed and
have to be open for each call? Can you spot if the delay is on incoming side
or on outgoing? Wireshark can decrypt the traffic if you provide the
certificate and start the sniffing before starting tls connections.
Alternative, set:
modparam("tls", "cipher_list", "NULL")
and the traffic should be no longer encrypted, but this will not show if the
bottleneck is the encryption process.
Also, benchmark module can help to see if execution of config file takes
longer than usual.
Jan Janak did tls tests some time ago, the summary is part of:
http://sip-router.org/wiki/performance/v3.0-capacity
I have some results from a test with 6000 SIP messages/sec over tls, where
CPU usage got about 60%. I guess something is becoming a bottleneck in your
case, very likely a blocking operation since cpu is ok, just has to be
discovered.
Cheers,
Daniel
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Daniel,
Thank you for getting back to me so quickly.
The log level is the default (debug=2). I tried setting cipher_list
to NULL on client and server and now Kamailio has crashed (producing
core file) during the last seven tests. It seems I can get to about
1000 registered users before this happens. How should I provide a
backtrace?
Once I get past this issue I will try your other suggestions.
interesting, let's see who is fault of that -- backtrace:
gdb /path/to/kamailio /path/to/corefile
- for short backtrace:
bt
- for full backtrace:
bt full
Provide both of them.
Cheers,
Daniel
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Re: [SR-Users] TLS performance testing
Hello, can you send also the last log messages (especially the ones before the crash, printed by the same pid that wrote the core, there should be a 'BUG...' message)? Looks like a double free in the libssl, what is your libssl version? Cheers, Daniel On 12/21/11 2:58 AM, Kristian Kielhofner wrote: On Tue, Dec 20, 2011 at 6:08 PM, Daniel-Constantin Mierla wrote: interesting, let's see who is fault of that -- backtrace: gdb /path/to/kamailio /path/to/corefile - for short backtrace: bt - for full backtrace: bt full Provide both of them. Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda Hi Daniel, I've collected all of them and attached them to this message. Let me know if there is anything else I can do. Thanks! -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Issue when SRTP enabled in the SIP clients
Hello,
have you enabled the nat traversal in kamailio's config file? From the
respective tutorial, the config file should contain:
#!KAMAILIO
#!define WITH_MYSQL
#!define WITH_AUTH
#!define WITH_USRLOCDB
#!define WITH_NAT
#!define WITH_TLS
plus update to rtpproxy module parameter:
modparam("rtpproxy", "rtpproxy_sock", "unix:/var/run/rtpproxy/rtpproxy.sock")
If you did all above, can you use tcp instead of tls for sip and send
the output of ngrep taken on kamailio server for a call that does not work:
ngrep -d any -qt -W byline port 5060
Cheers,
Daniel
On 12/21/11 6:49 AM, Jonathan Martin wrote:
Hi,
I followed this web article to install Kamailio 3.2 and RTPProxy on
Debian Squeeze x64:
http://kb.asipto.com/kamailio:skype-like-service-in-less-than-one-hour
The system is running on a public IP address outside of our corporate
LAN. I have been testing it using pjsua v2 alpha 2 from the pjsip.org
project.
I am having an issue when I enable srtp in the pjsua clients. If both
pjsua clients are running on machines on our corporate LAN (symmetric
NAT), the call succeeds and I get audio and video. If one of the
clients is running outside of the corporate LAN, the call connects but
I do not get any audio or video. If I turn off srtp in both clients
and try the call again, audio and video starts working. Is there any
additional configuration I need to make in the kamailio.cfg file when
I intend to use srtp in the clients?
RTPProxy info:
Basic version: 20040107
Extension 20050322: Support for multiple RTP streams and MOH
Extension 20060704: Support for extra parameter in the V command
Extension 20071116: Support for RTP re-packetization
Extension 20071218: Support for forking (copying) RTP stream
Extension 20080403: Support for RTP statistics querying
Extension 20081102: Support for setting codecs in the update/lookup
command
Extension 20081224: Support for session timeout notifications
Kamailio info:
version: kamailio 3.2.0 (x86_64/linux)
flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS,
USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM,
SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, USE_FUTEX,
FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR,
USE_DST_BLACKLIST, HAVE_RESOLV_RES
ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 4MB
poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
id: unknown
compiled on 10:23:25 Nov 2 2011 with gcc 4.4.5
Regards,
--Jonathan
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Re: [SR-Users] Music on hold with freeswitch as SBC
Hello, On 12/21/11 7:49 AM, Olle E. Johansson wrote: 20 dec 2011 kl. 22:40 skrev Gautam Batra: Hi, Thanks for your replies. Is it possible to play an audio file in the case of a re-invite directly from kamailio instead of freeswitch by using rtpproxy_stream2uac() or something similar? Kamailioi is still a proxy and from the endpoint point of view is not involved in the media plane. If you managed to do that many endpoints would ignore the packets or see them as a DOS attack attempt. Other endpoints might just play them. In later releases of Asterisk, we lock to the IP address of the peer and would ignore these. Asterisk used to send music-on-hold like this before, but we considered it a security issue and started reinviting to make Asterisk involved in the call again to play music on hold. Asterisk can do that, because it's a b2bua and is an endpoint in the call. Kamailio can't initiate a reinvite in the call. indeed, kamailio cannot initiate re-invites. You can play an audio file via rtpproxy and rtpproxy_stream2uac() if you use rtpproxy relaying from the beginning of the call. Otherwise, use a sip b2bua which does signaling only until you need to play audio and do re-invites so it gets in media path. Besides Asterisk or FreeSWITCH, a lightweight b2bua that probably offers such functionality is sems (sip express media server) -- I CC-ed Stefan, he can confirm and even give some leads of how to do it. Cheers, Daniel /O Gautam On Mon, Dec 12, 2011 at 4:50 AM, Olle E. Johansson wrote: 12 dec 2011 kl. 10:45 skrev Daniel-Constantin Mierla: Hello, On 12/9/11 9:04 PM, Gautam Batra wrote: Hello, I have a kamailio sip proxy server with freeswitch acting as SBC. I want to redirect the call to freeswitch when hold is pressed so that i can play music on hold. I tried this by using rewritehostport in case of a re-invite, but the call drops in that case. Could someone please help me with this? it is not possible to redirect established calls (it breaks the RFC3261), you have to route the call through freeswitch from its start. Perhaps you can use freeswitch without relaying the media in first place and when you have on hold, you set media patch to go through freeswitch. The only solution is having FreeSwitch send an invite with replaces to grab the call. The question is how to get it back. /O --- * Olle E Johansson - o...@edvina.net * Cell phone +46 70 593 68 51, Office +46 8 96 40 20, Sweden ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] No memory left in send_subscribe (PUA) from update_pua while building the tm dlg_t structure
Hello,
you can see the available pkg via sercmd, sending command pkg.stats
(match the entry for the pid printing the error). If there is no free
memory, then might be a leak.
You can attach with gdb to the pid printing these errors and walk to
pkg, you see the commands for gdb at:
http://www.asipto.com/pub/kamailio-devel-guide/#c04troubleshooting
See if you have lot of allocated chunks from same place in source code
(ignore those at the beginning, mainly related to cfg parsing) and send
the details here.
Cheers,
Daniel
On 12/21/11 10:44 AM, laura testi wrote:
Hi,
we are using the PUA_XMPP and PUA modules from the master branch. When
the modules are started, everything are ok, the presence events from
XMPP are sent to kamailio SIP servers (PUBLISH/SUBSCRIBE) and cached
in the hash. But when there are several thousands records in the hash
tabel, the update_pua function called in the hashT_clean gives a lot
of "No memory left" error when the hashT_clean is waked up from the
time:
ERROR: pua [send_subscribe.c]: No memory left
ERROR: pua [pua.c]: while building tm dlg_t structure
The failed call is:
td = (dlg_t*)pkg_malloc(size);
if(td == NULL)
{
LM_ERR("No memory left\n");
return NULL;
}
in dlg_t* pua_build_dlg_t(ua_pres_t* presentity) function in
send_subscribe.c. The size is about 400 and something... It's
strange.
Is it the memory leak in the PUA module?
I also try to increase the pkg_memory from 4MB default to 16MB, but it
doesn't help.
Any Idea?
Thanks in advanced
Laura
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Re: [SR-Users] No memory left in send_subscribe (PUA) from update_pua while building the tm dlg_t structure
Hello,
pkg.stats was added in 3.2.0, iirc. For 3.1, you can walk the packets in
memory with gdb -- 3.1 has memory debug on, so you don't need to
recompile (unless you turned it off).
Just attach to the pid of a sip worker (gdb /path/to/kamailio
_pid_value_) and run the gdb script.
Cheers,
Daniel
On 12/21/11 11:58 AM, laura testi wrote:
Hi Daniel,
I try the sercmd for pkg memory but it return 500 error:
# sercmd
sercmd 0.2
Copyright 2006 iptelorg GmbH
This is free software with ABSOLUTELY NO WARRANTY.
For details type `warranty'.
sercmd> pkg.stats
error: 500 - command pkg.stats not found
sercmd>
Is it available only for 3.2.x and master branch? Because we are using
3.1.5. But take the PUA module from master branch for the fetch_row
parameter you have patched ;-)
core.shmmem is ok.
Thanks a lot
Laura
On Wed, Dec 21, 2011 at 10:58 AM, Daniel-Constantin Mierla
wrote:
Hello,
you can see the available pkg via sercmd, sending command pkg.stats (match
the entry for the pid printing the error). If there is no free memory, then
might be a leak.
You can attach with gdb to the pid printing these errors and walk to pkg,
you see the commands for gdb at:
http://www.asipto.com/pub/kamailio-devel-guide/#c04troubleshooting
See if you have lot of allocated chunks from same place in source code
(ignore those at the beginning, mainly related to cfg parsing) and send the
details here.
Cheers,
Daniel
On 12/21/11 10:44 AM, laura testi wrote:
Hi,
we are using the PUA_XMPP and PUA modules from the master branch. When
the modules are started, everything are ok, the presence events from
XMPP are sent to kamailio SIP servers (PUBLISH/SUBSCRIBE) and cached
in the hash. But when there are several thousands records in the hash
tabel, the update_pua function called in the hashT_clean gives a lot
of "No memory left" error when the hashT_clean is waked up from the
time:
ERROR: pua [send_subscribe.c]: No memory left
ERROR: pua [pua.c]: while building tm dlg_t structure
The failed call is:
td = (dlg_t*)pkg_malloc(size);
if(td == NULL)
{
LM_ERR("No memory left\n");
return NULL;
}
in dlg_t* pua_build_dlg_t(ua_pres_t* presentity) function in
send_subscribe.c. The size is about 400 and something... It's
strange.
Is it the memory leak in the PUA module?
I also try to increase the pkg_memory from 4MB default to 16MB, but it
doesn't help.
Any Idea?
Thanks in advanced
Laura
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Re: [SR-Users] TLS performance testing
Hello, On 12/21/11 8:26 PM, Kristian Kielhofner wrote: On Wed, Dec 21, 2011 at 4:36 AM, Daniel-Constantin Mierla wrote: Hello, can you send also the last log messages (especially the ones before the crash, printed by the same pid that wrote the core, there should be a 'BUG...' message)? Looks like a double free in the libssl, what is your libssl version? Cheers, Daniel Hi Daniel, I don't see relevant lines in the log (surprisingly). The strange part is there is always at least one kamailio process running when this happens (or maybe not so strange). This is an up to date CentOS 6 with openssl-1.0.0-20.el6.x86_64 installed. I do get this warning on startup (same on CentOS 5.x): WARNING: tls [tls_init.c:587]: tls: openssl bug #1491 (crash/mem leaks on low memory) workaround enabled (on low memory tls operations will fail preemptively) with free memory thresholds 271581184 and 135790592 bytes it has to be at least the message printed by: LOG(L_CRIT, "BUG: qm_free: freeing already freed pointer," " first free: %s: %s(%ld) - aborting\n", f->file, f->func, f->line); Can you grep for it in logs? The abort in frame #1 of bracktrace is triggered on purpose, at mem/q_malloc.c line 458 Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Music on hold with freeswitch as SBC
Can you give the output of:
ps auxw | grep -i rtpproxy
That will show if rtpproxy is running and what is its control socket.
Cheers,
Daniel
On 12/21/11 11:25 PM, Gautam Batra wrote:
I'm not able to set up the rtp proxy module. I have entered the following:
loadmodule "rtpproxy.so"
modparam ("rtpproxy", "rtpproxy_sock", "udp:X.Y.Z.W:2");
Where X.Y.Z.W is the IP address of my machine (same as that of my SIP
server). But the log shows the following errors:
Dec 21 13:11:12 abc daemon.err /usr/sbin/kamailio[9984]: ERROR:
rtpproxy [rtpproxy.c:1503]: can't send command to a RTP proxy
Dec 21 13:11:12 abc daemon.err /usr/sbin/kamailio[9984]: ERROR:
rtpproxy [rtpproxy.c:1538]: proxy does not
respond, disable it
Dec 21 13:11:12 abc daemon.warn /usr/sbin/kamailio[9984]: WARNING:
rtpproxy [rtpproxy.c:1395]: can't get version of the RTP proxy
Dec 21 13:11:12 abc daemon.warn /usr/sbin/kamailio[9984]: WARNING:
rtpproxy [rtpproxy.c:1432]: support for RTP proxy
has been disabled temporarily
Could anyone tell what I'm doing wrong? I tried to run rtpproxy
separately on the given port before starting kamailio (rtpproxy -s
udp:X.Y.Z.W:2), but it didn't work.
On Wed, Dec 21, 2011 at 2:36 PM, Gautam Batra <mailto:gautambatr...@gmail.com>> wrote:
I am using Freeswitch as an SBC behind Kamailio, and my external
calls are routed via freeswitch. In those calls the music on hold
works as it is handled by freeswitch. Ideally I would like to
somehow redirect when a call is put on hold to the MOH extension.
The other option is by using rtpproxy. I could not find any
documentation on rtpproxy and would really appreciate it if
someone could lead me to it or give me a brief overview on how to
go about using rtpproxy_stream2uac to play music whenever a call
is put on hold.
On Wed, Dec 21, 2011 at 4:50 AM, Daniel-Constantin Mierla
mailto:mico...@gmail.com>> wrote:
Hello,
On 12/21/11 7:49 AM, Olle E. Johansson wrote:
20 dec 2011 kl. 22:40 skrev Gautam Batra:
Hi,
Thanks for your replies. Is it possible to play an
audio file in the case of a re-invite directly from
kamailio instead of freeswitch by using
rtpproxy_stream2uac() or something similar?
Kamailioi is still a proxy and from the endpoint point of
view is not involved in the media plane. If you managed to
do that many
endpoints would ignore the packets or see them as a DOS
attack attempt. Other endpoints might just play them.
In later releases of Asterisk, we lock to the IP address
of the peer and would ignore these. Asterisk used to send
music-on-hold
like this before, but we considered it a security issue
and started reinviting to make Asterisk involved in the
call again to play
music on hold. Asterisk can do that, because it's a b2bua
and is an endpoint in the call. Kamailio can't initiate a
reinvite in the
call.
indeed, kamailio cannot initiate re-invites. You can play an
audio file via rtpproxy and rtpproxy_stream2uac() if you use
rtpproxy relaying from the beginning of the call. Otherwise,
use a sip b2bua which does signaling only until you need to
play audio and do re-invites so it gets in media path.
Besides Asterisk or FreeSWITCH, a lightweight b2bua that
probably offers such functionality is sems (sip express media
server) -- I CC-ed Stefan, he can confirm and even give some
leads of how to do it.
Cheers,
Daniel
/O
Gautam
On Mon, Dec 12, 2011 at 4:50 AM, Olle E.
Johanssonmailto:o...@edvina.net>> wrote:
12 dec 2011 kl. 10:45 skrev Daniel-Constantin Mierla:
Hello,
On 12/9/11 9:04 PM, Gautam Batra wrote:
Hello,
I have a kamailio sip proxy server with
freeswitch acting as SBC. I want to redirect
the call to freeswitch when hold is pressed so
that i can play music on hold. I tried this by
using rewritehostport in case of a re-invite,
but the call drops in that case. Could someone
please help me with this?
it is not possible to redirect established calls
(it breaks the RFC3261), you have to route the
call through freeswitch from its start. Perhaps
you can use freeswitch without relaying the media
in first p
Re: [SR-Users] ERROR: slow timer too slow: overflow (1048 - 25 = 1023) repeat a lot
Hello, On 12/21/11 12:12 PM, laura testi wrote: Hi, sometime we get the following error after we restart the kamailio for a while: ERROR: [timer.c:972]: ERROR: slow timer too slow: overflow (1048 - 25 = 1023) ... last message repeated 138times ERROR: [timer.c:972]: ERROR: slow timer too slow: overflow (1048 - 25 = 1023) last message repeated 49 times it print them continuously is it normal? there is some long task executed on timer, maybe on failure route. The timer should recover itself if the task is not executed every time. How I can fix it? You need to figure out what it takes long to execute and improve that. It is not a critical error, it keeps working. I am going to make that message a warning instead of error. Cheers, Daniel Thank you very much! Best Regards, Laura ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] new in devel wiki page
Hello, short reminder for everyone to update the wiki page with short description of the new stuff/enhancements added to GIT repository -- here it is: * http://www.kamailio.org/wiki/features/new-in-devel Also, if you do changes to existing parts that require upgrade of config file or database schema from v3.2.x to devel, please write short guidelines at: * http://www.kamailio.org/wiki/install/upgrade/3.2.x-to-3.3.0 Thanks, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] anyone with opensuse around?
Hello, is anyone here having an opensuse available for using to do some tests? It is about the radius modules, it seems opensuse does now have radiusclient-ng, but the client library from freeradius. They should be quite similar and we should be able to compile with freeradius client library, like make FREERADIUS=1 ... I tried on opensuse build service and it fails due to some conflict on DBG definition. It is hard to troubleshoot on OBS, so maybe someone can give a hand and find the fix faster -- I can provide guidelines or if I can get shell access (just simple user) would be better (write me directly in this case). The goal is to get the radius modules packaged as rpm on opensuse. Thanks, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] usrloc, timer process and cache cleanup
Hello,
On 12/22/11 6:19 PM, Andreas Granig wrote:
Got the cause of the issue.
What happens is that there's an AOR which registers ever 120 seconds.
For some reason, the location entry is in usrloc cache, but not in db.
What happens now is that usrloc tries an "update" query in the db,
because it still assumes that the entry is there, which obviously fails.
If you remove the entry from usrloc (kamctl ul rm), then on the
next re-registration it's both inserted into the cache and into the db.
Wondering how it could happen to get out of sync, and how we could
improve this. Maybe using a "replace into" instead of "update", at least
for mysql? Suggestions?
is the timer interval parameter of usrloc higher than 120sec?
http://kamailio.org/docs/modules/3.2.x/modules_k/usrloc.html#id2494575
IIRC, there should be anyhow a flag to mark if the record is in db or
not, and based on that do insert or update, maybe something is lost
there. If you do 'kamctl ul show __aor__', what are the values for flags
fields?
Another option, perhaps more portable, but with two db hits is: update
and if fails then insert -- considering that these should be corner
cases, maybe the performance is not affected much. A blended version is
even better, if the db driver supports replace, do replace instead of
update (I don't know if replace is faster/slower than update).
Cheers,
Daniel
Andreas
On 12/22/2011 05:12 PM, Andreas Granig wrote:
Hi,
Could you please tell me which of the three timer processes ("timer",
"slow timer" or "timer nh") is responsible for cleaning up the internal
usrloc cache?
Looks like every now and then the cleanup of the internal location cache
is starting to fail. Funny thing is that expired locations are removed
from the mysql backend, but not from the internal cache. We're running
kamailio 3.1.5, are there any known issues fixed since that version?
In the meanwhile we're trying to pin the issue down, but maybe someone
has a clue...
Andreas
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Re: [SR-Users] usrloc, timer process and cache cleanup
On 12/22/11 7:40 PM, Ovidiu Sas wrote:
The replace solution will mask the real issue.
The flag that is in the usrloc should switch between update or insert
and that is the real fix.
Right, that has to be done, but there are some cases when db can become
inconsistent, due to database unavailability, and then some trick have
to be done at db layer, example:
- db is unavailable, phone unregisters, contact deleted from memory but
not from database
- phone register again, usrloc will try insert and will fail - in this
case it should be update if insert fails (or replace)
The other way around could happen when mistakenly deleting/changing
records in db, which should not happen, but Murphy says opposite.
Cheers,
Daniel
Regards,
Ovidiu Sas
-- VoIP Embedded, Inc.http://www.voipembedded.com
On Thu, Dec 22, 2011 at 12:58 PM, Daniel-Constantin Mierla
wrote:
Hello,
On 12/22/11 6:19 PM, Andreas Granig wrote:
Got the cause of the issue.
What happens is that there's an AOR which registers ever 120 seconds.
For some reason, the location entry is in usrloc cache, but not in db.
What happens now is that usrloc tries an "update" query in the db,
because it still assumes that the entry is there, which obviously fails.
If you remove the entry from usrloc (kamctl ul rm), then on the
next re-registration it's both inserted into the cache and into the db.
Wondering how it could happen to get out of sync, and how we could
improve this. Maybe using a "replace into" instead of "update", at least
for mysql? Suggestions?
is the timer interval parameter of usrloc higher than 120sec?
http://kamailio.org/docs/modules/3.2.x/modules_k/usrloc.html#id2494575
IIRC, there should be anyhow a flag to mark if the record is in db or not,
and based on that do insert or update, maybe something is lost there. If you
do 'kamctl ul show __aor__', what are the values for flags fields?
Another option, perhaps more portable, but with two db hits is: update and
if fails then insert -- considering that these should be corner cases, maybe
the performance is not affected much. A blended version is even better, if
the db driver supports replace, do replace instead of update (I don't know
if replace is faster/slower than update).
Cheers,
Daniel
Andreas
On 12/22/2011 05:12 PM, Andreas Granig wrote:
Hi,
Could you please tell me which of the three timer processes ("timer",
"slow timer" or "timer nh") is responsible for cleaning up the internal
usrloc cache?
Looks like every now and then the cleanup of the internal location cache
is starting to fail. Funny thing is that expired locations are removed
from the mysql backend, but not from the internal cache. We're running
kamailio 3.1.5, are there any known issues fixed since that version?
In the meanwhile we're trying to pin the issue down, but maybe someone
has a clue...
Andreas
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Re: [SR-Users] usrloc, timer process and cache cleanup
On 12/22/11 8:02 PM, Ovidiu Sas wrote: Right, that has to be done, but there are some cases when db can become inconsistent, due to database unavailability, and then some trick have to be done at db layer, example: - db is unavailable, phone unregisters, contact deleted from memory but not from database - phone register again, usrloc will try insert and will fail - in this case it should be update if insert fails (or replace) If the phone registers again, it should be a brand new entry (different Call-ID, CSeq and so on). The leftover entry on the db will be cleanup on a server restart. According to sip specs, the key is the contact address, thus operations are done on (aor, contact). In kamailio you can configure to match using call id (plus path) as well, but is not the default since it breaks the specs. The other way around could happen when mistakenly deleting/changing records in db, which should not happen, but Murphy says opposite. If someone is messing with the db, kamailio shouldn't try to correct admin mistakes. It was just an example, I haven't gone to all corner cases that can happen from human or (self or different) application errors. There were couple of similar reports in the past, related to conflicts of insert/update, update/insert, delete/update a.s.o. cases, so I proposed to go for a portable solution, not for one which valid to a db driver only -- configurable or not, is different thing than the specific topic. I think that the replace solution should be a last resort. If implemented, should be configurable. I would rather see the original issue instead of being masked and let it trigger other issues later on which would be more difficult to debug. I think the goal is to have coherent, consistent and persistent records in location table, so that lookup is always valid and restarts don't lose records. In no case you get any feedback implicitly, this is either about logging or malfunctioning. Perhaps is better to get a log message to investigate and have all keep going, than the second. Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] How to get Reason code from reply in failure_route
Hello, On 12/23/11 11:44 AM, Mino Haluz wrote: Hi, how can I get the reason code from reply in failure_route ? $hdr(reason) points to the INVITE request.. you can test it with a regular expression via t_check_status(...) from tm module or get it in a config variable: http://www.kamailio.org/wiki/cookbooks/3.2.x/pseudovariables#t_reply_code Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] No memory left in send_subscribe (PUA) from update_pua while building the tm dlg_t structure
Hello,
looks like a leak in a module that is storing record-routes and use them
later, perhaps pua module, I will check it soon.
Cheers,
Daniel
On 12/23/11 11:31 AM, laura testi wrote:
Hi Daniel,
I just follow the instruction in the link you sent
(http://www.asipto.com/pub/kamailio-devel-guide/#c04troubleshooting)
to use gdb to print PKG fragments. When I got the error in PUA:
Dec 23 11:10:53 /.../sbin/kamailio[23276]: ERROR: pua
[send_subscribe.c:158]: No memory left for size:439
Dec 23 11:10:53 /.../sbin/kamailio[23276]: ERROR: pua [pua.c:747]:
while building tm dlg_t structure
Dec 23 11:10:53 /.../sbin/kamailio[23276]: ERROR: pua [pua.c:652]:
while updating record
The I run the command
gdb /.../sbin/kamailio 23276
and write the following commands in the gdb:
set $i=0
set $a = mem_block->first_frag
while($i<1)
if($i>2000)
if($a->u.is_free==0)
p *$a
end
end
set $a = ((struct qm_frag*)((char*)($a)+sizeof(struct
qm_frag)+((struct qm_frag*)$a)->size+sizeof(struct qm_frag_end)))
set $i = $i + 1
end
...
after a while I got a lot of prints on the screen like these:
func = 0x5d7b00 "do_parse_rr_body", line = 74, check = 4042322160}
$1348 = {size = 72, u = {nxt_free = 0x0, is_free = 0},
file = 0x5d76c9 ": parser/parse_rr.c",
func = 0x5d7b00 "do_parse_rr_body", line = 74, check = 4042322160}
$1349 = {size = 72, u = {nxt_free = 0x0, is_free = 0},
file = 0x5d76c9 ": parser/parse_rr.c",
func = 0x5d7b00 "do_parse_rr_body", line = 74, check = 4042322160}
$1350 = {size = 72, u = {nxt_free = 0x0, is_free = 0},
file = 0x5d76c9 ": parser/parse_rr.c",
func = 0x5d7b00 "do_parse_rr_body", line = 74, check = 4042322160}
$1351 = {size = 104, u = {nxt_free = 0x0, is_free = 0},
file = 0x5d76c9 ": parser/parse_rr.c",
func = 0x5d7b00 "do_parse_rr_body", line = 74, check = 4042322160}
$1352 = {size = 104, u = {nxt_free = 0x0, is_free = 0},
file = 0x5d76c9 ": parser/parse_rr.c",
func = 0x5d7b00 "do_parse_rr_body", line = 74, check = 4042322160}
$1353 = {size = 72, u = {nxt_free = 0x0, is_free = 0},
file = 0x5d76c9 ": parser/parse_rr.c",
func = 0x5d7b00 "do_parse_rr_body", line = 74, check = 4042322160}
$1354 = {size = 72, u = {nxt_free = 0x0, is_free = 0},
---Type to continue, or q to quit---
...
But I don't understand if these are normal or something goes wrong
Can you help
Best Regards,
Laura
On Wed, Dec 21, 2011 at 12:18 PM, Daniel-Constantin Mierla
wrote:
Hello,
pkg.stats was added in 3.2.0, iirc. For 3.1, you can walk the packets in
memory with gdb -- 3.1 has memory debug on, so you don't need to recompile
(unless you turned it off).
Just attach to the pid of a sip worker (gdb /path/to/kamailio _pid_value_)
and run the gdb script.
Cheers,
Daniel
On 12/21/11 11:58 AM, laura testi wrote:
Hi Daniel,
I try the sercmd for pkg memory but it return 500 error:
# sercmd
sercmd 0.2
Copyright 2006 iptelorg GmbH
This is free software with ABSOLUTELY NO WARRANTY.
For details type `warranty'.
sercmd>pkg.stats
error: 500 - command pkg.stats not found
sercmd>
Is it available only for 3.2.x and master branch? Because we are using
3.1.5. But take the PUA module from master branch for the fetch_row
parameter you have patched ;-)
core.shmmem is ok.
Thanks a lot
Laura
On Wed, Dec 21, 2011 at 10:58 AM, Daniel-Constantin Mierla
wrote:
Hello,
you can see the available pkg via sercmd, sending command pkg.stats
(match
the entry for the pid printing the error). If there is no free memory,
then
might be a leak.
You can attach with gdb to the pid printing these errors and walk to pkg,
you see the commands for gdb at:
http://www.asipto.com/pub/kamailio-devel-guide/#c04troubleshooting
See if you have lot of allocated chunks from same place in source code
(ignore those at the beginning, mainly related to cfg parsing) and send
the
details here.
Cheers,
Daniel
On 12/21/11 10:44 AM, laura testi wrote:
Hi,
we are using the PUA_XMPP and PUA modules from the master branch. When
the modules are started, everything are ok, the presence events from
XMPP are sent to kamailio SIP servers (PUBLISH/SUBSCRIBE) and cached
in the hash. But when there are several thousands records in the hash
tabel, the update_pua function called in the hashT_clean gives a lot
of "No memory left" error when the hashT_clean is waked up from the
time:
ERROR: pua [send_subscribe.c]: No memory left
ERROR: pua [pua.c]: while building tm dlg_t structure
The failed call is:
td = (dlg_t*)pkg_malloc(size);
if(td == NULL)
{
LM_ERR("No memory left\n");
return NULL;
}
in dlg_t* pua_build_dlg_t(ua_pres_t* presentity) function in
send_subscribe.c. The size is about 400 and something...
Re: [SR-Users] No memory left in send_subscribe (PUA) from update_pua while building the tm dlg_t structure
Hello,
can you try with this patch:
http://git.sip-router.org/cgi-bin/gitweb.cgi/sip-router/?a=commit;h=1b3cfa60a5b5c7d435704d44b7c495b7e6aa84c8
Cheers,
Daniel
On 12/23/11 12:36 PM, Daniel-Constantin Mierla wrote:
Hello,
looks like a leak in a module that is storing record-routes and use
them later, perhaps pua module, I will check it soon.
Cheers,
Daniel
On 12/23/11 11:31 AM, laura testi wrote:
Hi Daniel,
I just follow the instruction in the link you sent
(http://www.asipto.com/pub/kamailio-devel-guide/#c04troubleshooting)
to use gdb to print PKG fragments. When I got the error in PUA:
Dec 23 11:10:53 /.../sbin/kamailio[23276]: ERROR: pua
[send_subscribe.c:158]: No memory left for size:439
Dec 23 11:10:53 /.../sbin/kamailio[23276]: ERROR: pua [pua.c:747]:
while building tm dlg_t structure
Dec 23 11:10:53 /.../sbin/kamailio[23276]: ERROR: pua [pua.c:652]:
while updating record
The I run the command
gdb /.../sbin/kamailio 23276
and write the following commands in the gdb:
set $i=0
set $a = mem_block->first_frag
while($i<1)
if($i>2000)
if($a->u.is_free==0)
p *$a
end
end
set $a = ((struct qm_frag*)((char*)($a)+sizeof(struct
qm_frag)+((struct qm_frag*)$a)->size+sizeof(struct qm_frag_end)))
set $i = $i + 1
end
...
after a while I got a lot of prints on the screen like these:
func = 0x5d7b00 "do_parse_rr_body", line = 74, check = 4042322160}
$1348 = {size = 72, u = {nxt_free = 0x0, is_free = 0},
file = 0x5d76c9 ": parser/parse_rr.c",
func = 0x5d7b00 "do_parse_rr_body", line = 74, check = 4042322160}
$1349 = {size = 72, u = {nxt_free = 0x0, is_free = 0},
file = 0x5d76c9 ": parser/parse_rr.c",
func = 0x5d7b00 "do_parse_rr_body", line = 74, check = 4042322160}
$1350 = {size = 72, u = {nxt_free = 0x0, is_free = 0},
file = 0x5d76c9 ": parser/parse_rr.c",
func = 0x5d7b00 "do_parse_rr_body", line = 74, check = 4042322160}
$1351 = {size = 104, u = {nxt_free = 0x0, is_free = 0},
file = 0x5d76c9 ": parser/parse_rr.c",
func = 0x5d7b00 "do_parse_rr_body", line = 74, check = 4042322160}
$1352 = {size = 104, u = {nxt_free = 0x0, is_free = 0},
file = 0x5d76c9 ": parser/parse_rr.c",
func = 0x5d7b00 "do_parse_rr_body", line = 74, check = 4042322160}
$1353 = {size = 72, u = {nxt_free = 0x0, is_free = 0},
file = 0x5d76c9 ": parser/parse_rr.c",
func = 0x5d7b00 "do_parse_rr_body", line = 74, check = 4042322160}
$1354 = {size = 72, u = {nxt_free = 0x0, is_free = 0},
---Type to continue, or q to quit---
...
But I don't understand if these are normal or something goes wrong
Can you help
Best Regards,
Laura
On Wed, Dec 21, 2011 at 12:18 PM, Daniel-Constantin Mierla
wrote:
Hello,
pkg.stats was added in 3.2.0, iirc. For 3.1, you can walk the
packets in
memory with gdb -- 3.1 has memory debug on, so you don't need to
recompile
(unless you turned it off).
Just attach to the pid of a sip worker (gdb /path/to/kamailio
_pid_value_)
and run the gdb script.
Cheers,
Daniel
On 12/21/11 11:58 AM, laura testi wrote:
Hi Daniel,
I try the sercmd for pkg memory but it return 500 error:
# sercmd
sercmd 0.2
Copyright 2006 iptelorg GmbH
This is free software with ABSOLUTELY NO WARRANTY.
For details type `warranty'.
sercmd>pkg.stats
error: 500 - command pkg.stats not found
sercmd>
Is it available only for 3.2.x and master branch? Because we are using
3.1.5. But take the PUA module from master branch for the fetch_row
parameter you have patched ;-)
core.shmmem is ok.
Thanks a lot
Laura
On Wed, Dec 21, 2011 at 10:58 AM, Daniel-Constantin Mierla
wrote:
Hello,
you can see the available pkg via sercmd, sending command pkg.stats
(match
the entry for the pid printing the error). If there is no free
memory,
then
might be a leak.
You can attach with gdb to the pid printing these errors and walk
to pkg,
you see the commands for gdb at:
http://www.asipto.com/pub/kamailio-devel-guide/#c04troubleshooting
See if you have lot of allocated chunks from same place in source
code
(ignore those at the beginning, mainly related to cfg parsing) and
send
the
details here.
Cheers,
Daniel
On 12/21/11 10:44 AM, laura testi wrote:
Hi,
we are using the PUA_XMPP and PUA modules from the master branch.
When
the modules are started, everything are ok, the presence events from
XMPP are sent to kamailio SIP servers (PUBLISH/SUBSCRIBE) and cached
in the hash. But when there are several thousands records in the hash
tabel, the update_pua function called in the hashT_clean gives a lot
of "No memory left" error when the hashT_clean is waked up from the
time:
ERROR: pua [send_subscribe.c]: No memory left
ERROR: pua [pua.c]: while building tm dlg_t structure
The failed call is:
td = (dlg_t*)pkg_malloc(size);
if(td == NULL)
{
[SR-Users] Merry Christmas!
I want to take the opportunity and give proper credits to everyone around the project, any bit of contribution makes it evolve better and better year by year. This includes also the involvement in community, we don't need only developers to improve. There is no reason to stay aside in an open source project, become active and play a role in its evolution. We've just started the second decade of development, but there is always space to innovate communications. With SER and Kamailio, we pioneered over the years many solutions in unified communication systems, the plans are the same, be ahead of the market with new features while preserving the robustness. Merry Christmas and great winter holidays to all Kamailians and SERians!!! Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] ACK From an OK it's being discarded
Hello,
if the ACK goes through the proxy, then means record routing is used,
but I see no Record-Route in 200 reply and no Route in ACK. Since there
is no Record-Route in 200 ok, the ACK has to be sent to the contact
address from the 200 ok.
Your config snippet from kamailio shows the part of default config where
record routing is handling -- based on the comments -- since it no
Route, it is dropped.
Cheers,
Daniel
On 12/26/11 11:03 PM, Lucas Alvarez wrote:
I have Kamailio 3.2.0 between two asterisk servers, after the call
set, one of the kamailio send the OK from the INVITE and the return
ACK of that message was discarded. This makes asterisk hangup the call
after 5 secs. It's that right?
OK message:
U 172.25.249.15:5060 <http://172.25.249.15:5060> -> 172.25.249.14:5060
<http://172.25.249.14:5060>
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 172.25.249.14:5060;branch=z9hG4bK09fc3de6;rport=5060.
From: "asterisk" <mailto:sip%3Aasterisk@172.25.249.14>>;tag=as6411602a.
To: <http://sip:775008@172.25.249.15:5060>>;tag=as55ab3180.
Call-ID: 547225391b7828402ecaa03e1dab5a86@172.25.249.14
<mailto:547225391b7828402ecaa03e1dab5a86@172.25.249.14>.
CSeq: 102 INVITE.
Server: Asterisk PBX 1.8.7.1.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH.
Supported: replaces, timer.
Contact: <http://sip:775008@172.25.249.15:5080>>.
Remote-Party-ID: "Eus Test" <mailto:sip%3A3999@172.25.249.14>>;party=called;privacy=off;screen=no.
Content-Type: application/sdp.
Content-Length: 285.
.
v=0.
o=root 2045590031 2045590031 IN IP4 172.25.249.15.
s=Asterisk PBX 1.8.7.1.
c=IN IP4 172.25.249.15.
t=0 0.
m=audio 11922 RTP/AVP 0 3 8 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:3 GSM/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.
a=sendrecv.
Discarded ACK:
U 172.25.249.14:5060 <http://172.25.249.14:5060> -> 172.25.249.15:5060
<http://172.25.249.15:5060>
ACK sip:775008@172.25.249.15:5080
<http://sip:775008@172.25.249.15:5080> SIP/2.0.
Via: SIP/2.0/UDP 172.25.249.14:5060;branch=z9hG4bK6ea5aff6;rport.
From: "asterisk" <mailto:sip%3Aasterisk@172.25.249.14>>;tag=as6411602a.
To: <http://sip:775008@172.25.249.15:5060>>;tag=as55ab3180.
Contact: <mailto:sip%3Aasterisk@172.25.249.14>>.
Call-ID: 547225391b7828402ecaa03e1dab5a86@172.25.249.14
<mailto:547225391b7828402ecaa03e1dab5a86@172.25.249.14>.
CSeq: 102 ACK.
User-Agent: Asterisk PBX.
Max-Forwards: 70.
Remote-Party-ID: "asterisk" <mailto:sip%3Aasterisk@172.25.249.14>>.
Content-Length: 0.
.
Kamailio's configuration where the ACK message it's being discarded:
if ( is_method("ACK") ) {
if ( t_check_trans() ) {
# no loose-route, but stateful
ACK;
# must be an ACK after a 487
# or e.g. 404 from upstream server
t_relay();
exit;
} else {
# ACK without matching
transaction ... ignore and discard
exit;
}
}
It would be ok if I relay the ack even if it didn't match any
transaction??
Any help would be appreciated.
Regards,
Lucas
___________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
Daniel-Constantin Mierla -- http://www.asipto.com
http://linkedin.com/in/miconda -- http://twitter.com/miconda
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Load Balancing
Hello, adding more info here. I guess the dispatcher from the forked project didn't evolve from the moment of the fork if they added a different module, so I assume it is more or less dispatcher as it was in kamailio 1.4. Meanwhile in kamailio (as we are at version 3.2.1), dispatcher got lot of new features, including calls load distribution, algorithm 10: http://kamailio.org/docs/modules/stable/modules_k/dispatcher.html#id2498405 Call load distribution counts the number of active calls per destination address, dispatcher selecting the least loaded. It uses script functions to manage the load (see the readme), since dispatcher can work event in stateless mode (so no tm or dialog modules are needed for such feature), thus is very lightweight, which is really important for scalable load balancing implementation. Among new features since dispatcher for kamailio 1.4: - weight based load balancing - priority based gateway selection - refactored keepalive mechanism for detecting when a gateway is down/up - execution of event_route blocks in config when a gateway is down/up - management through RPC commands Cheers, Daniel On 12/28/11 1:43 PM, Alex Balashov wrote: On 12/28/2011 07:21 AM, Ronald Kiel wrote: I understand Kamailio has the Dispatcher-module for load-balancing. OpenSIPS has both a Dispatcher-module and a LB-module, which seems to offer improved functionality. Is there also something like the LB-module in Kamailio? No, there's nothing in Kamailio that implements what the LB module does, exactly. However, I wouldn't call the LB module 'improved' functionality; it extends a particular subset of 'dispatcher' in a specific way, nothing more. Furthermore, with Kamailio it is relatively trivial to come up with your own load-aware routing process in route script. I think that's one reason it doesn't get wrapped in a module. If you want to periodically poll some outside source for load information, define a periodic 'rtimer' process that can go out and do that: http://www.kamailio.org/docs/modules/3.2.x/modules_k/rtimer.html Store the results in 'htable' or some other data structure, and then check it when routing calls. And that's all there is to it. :-) Now you can implement skills-based routing or anything else esoteric, because the algorithm is 100% yours to override. -- Alex -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] No memory left in send_subscribe (PUA) from update_pua while building the tm dlg_t structure
, (void*)cb_param);
-
- result= tmb.t_request_within(&uac_r);
- if(result< 0)
- {
- LM_ERR("in t_request function\n");
- shm_free(cb_param);
- pkg_free(td);
- goto error;
- }
-
- pkg_free(td);
- td= NULL;
- }
-
- pkg_free(str_hdr);
- return 0;
-
-error:
- if(str_hdr)
- pkg_free(str_hdr);
- return -1;
+ int expires;
+ int result = 0;
+ uac_req_t uac_r;
+ str met = {NULL, 0};
+ int ret_code = 0;
+ dlg_t* td = NULL;
+
+
+ if(p->desired_expires== 0)
+expires= default_expires;
+ else
+expires= p->desired_expires- (int)time(NULL);
+
+ if(p->watcher_uri== NULL)
+ {
+
+str_hdr = publ_build_hdr(expires, get_event(p->event), NULL,
+&p->etag, p->extra_headers, 0);
+
+if(str_hdr == NULL)
+{
+ LM_ERR("while building extra_headers\n");
+ ret_code = -1;
+ goto done;
+}
+LM_DBG("str_hdr:\n%.*s\n ", str_hdr->len, str_hdr->s);
+
+cb_param= build_uppubl_cbparam(p);
+if(cb_param== NULL)
+{
+ LM_ERR("while constructing publ callback param\n");
+ ret_code = -1;
+ goto done;
+}
+
+met.s = (char*)pkg_malloc(8*sizeof(char));
+if(met.s == NULL) {
+ LM_ERR("no memory for met.s(PUBLISH)\n");
+ ret_code = -1;
+ goto done;
+}
+memset(met.s, 0, 8);
+memcpy(met.s, "PUBLISH", 7);
+met.len = 7;
+set_uac_req(&uac_r,&met, str_hdr, 0, 0, TMCB_LOCAL_COMPLETED,
+publ_cback_func, (void*)cb_param);
+
+result= tmb.t_request(&uac_r,
+p->pres_uri, /* Request-URI */
+p->pres_uri, /* To */
+p->pres_uri, /* From */
+&outbound_proxy /* Outbound proxy*/
+);
+if(result< 0)
+{
+ LM_ERR("in t_request function\n");
+ shm_free(cb_param);
+ cb_param = NULL;
+ ret_code = -1;
+ goto done;
+}
+ }
+ else
+ {
+td= pua_build_dlg_t(p);
+if(td== NULL)
+{
+ LM_ERR("while building tm dlg_t structure");
+ ret_code = -1;
+ goto done;
+};
+
+str_hdr= subs_build_hdr(&p->contact, expires,p->event,p->extra_headers);
+if(str_hdr== NULL || str_hdr->s== NULL)
+{
+ if(p->event!=0)
+LM_ERR("while building extra headers\n");
+
+ ret_code = -1;
+ goto done;
+}
+
+cb_param= subs_cbparam_indlg(p, expires, REQ_ME);
+if(cb_param== NULL)
+{
+ LM_ERR("while constructing subs callback param\n");
+ ret_code = -1;
+ goto done;
+}
+
+met.s = (char*)pkg_malloc(10*sizeof(char));
+if(met.s == NULL) {
+ LM_ERR("no memory for met.s(SUBSCRIBE)\n");
+ ret_code = -1;
+ goto done;
+}
+memset(met.s, 0, 10);
+memcpy(met.s, "SUBSCRIBE", 9);
+met.len = 9;
+set_uac_req(&uac_r,&met, str_hdr, 0, td, TMCB_LOCAL_COMPLETED,
+subs_cback_func, (void*)cb_param);
+
+result= tmb.t_request_within(&uac_r);
+if(result< 0)
+{
+ LM_ERR("in t_request function\n");
+ ret_code = -1;
+ shm_free(cb_param);
+ cb_param = NULL;
+ goto done;
+}
+ }
+
+
+done:
+ if(td!=NULL)
+ {
+ if(td->route_set)
+ free_rr(&td->route_set);
+
+pkg_free(td);
+td= NULL;
+ }
+
+ if(met.s != NULL) {
+pkg_free(met.s);
+met.s = NULL;
+met.len = 0;
+ }
+
+ if(str_hdr != NULL) {
+pkg_free(str_hdr);
+str_hdr = NULL;
+ }
+ return ret_code;
}
static void db_update(unsigned int ticks,void *param)
It seems a lot of change, but there are only a few lines changed,
probably in the file I modify I have convert the TAB to Space and the
diff does not recognize them. Another thing is I use dynamic memory
for the str SUBSCRIBE and PUBLISH.
Another thing for the management of NOTIFY arrives before the 202/200
OK of SUBSCRIBE, in addition to the patches done previously, maybe the
hash.c need to patch for the is_dialog function which is called for
the NOTIFY in PUA_XMPP module:
# diff -u hash.c.orig hash.c
--- hash.c.orig 2011-12-27 13:38:06.0 +0100
+++ hash.c 2011-12-27 13:38:38.0 +0100
@@ -487,10 +487,11 @@
hash_code= core_hash(dialog->pres_uri, dialog->watcher_uri, HASH_SIZE);
lock_get(&HashT->p_records[hash_code].lock);
- if(get_dialog(dialog, hash_code)== NULL)
- ret_code= -1;
- else
+ if(get_dialog(dialog, hash_code) ||
get_temporary_dialog(dialog, hash_code))
ret_code= 0;
+ else
+ ret_code= -1;
+
lock_release(&HashT->p_records[hash_code].lock);
return ret_code;
Best Regards,
Laura
On Fri, Dec 23, 2011 at 5:10
Re: [SR-Users] problem with create and release tcp connections
Hello, On 12/28/11 2:41 PM, Bruno Bresciani wrote: Hi all, Someone know if the kamailio 1.5.0 has problem to create and release tcp connetctions after some time running? In my SIP network I have six IP telephones and one gateway registered over TLS, after some time (one day) this devices have problems to register and make calls. I noticed through the log a freezing of kamailio when it try release the tcp connections. Below is part of the log where I notice this behavior: *Dec 28 09:46:46 vts74 /home2/local/kamailio/sbin/kamailio[8951]: DBG:core:io_watch_add: io_watch_add(0x816cc80, 25, 2, 0xb61e6558), fd_no=20 Dec 28 09:46:46 vts74 /home2/local/kamailio/sbin/kamailio[8951]: DBG:core:handle_tcp_child: cmd CONN_RELEASE 0xb61e6558 refcnt= 0 Dec 28 09:47:11 vts74 /home2/local/kamailio/sbin/kamailio[8940]: DBG:usrloc:nodb_timer: Binding '2503','sips:2503@192.168.174.178 <mailto:sips%3A2503@192.168.174.178>;transport=tls' has expired Dec 28 09:47:26 vts74 /home2/local/kamailio/sbin/kamailio[8951]: DBG:core:handle_tcpconn_ev: data available on 0xb621d710 30 Dec 28 09:47:26 vts74 /home2/local/kamailio/sbin/kamailio[8951]: DBG:core:io_watch_del: io_watch_del (0x816cc80, 30, -1, 0x0) fd_no=21 called* I know that upgrade to kamailio 3xx is recommended and I'm working to do it, but before, I have to understand this behavior. TLS was really experimental in 1.x, I don't think anyone here among developers is going to sped time troubleshooting it. General hints, attach with gdb to the pid and do a backtrace to see what is doing. Check with netstat the state of the connection, too. Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Pseudo-variables for "Status-Line"
On 12/29/11 3:42 PM, Andrew Pogrebennyk wrote: On 12/29/2011 03:36 PM, Robert R wrote: Thank a lot. $T_reply_code works. I tried all variables in pv doc ($rc, $err.rcode, $rs ... ) and none of them works, Actually the $rs pseudo-variable should also work as described here: http://sip-router.org/wiki/cookbooks/pseudo-variables/devel#sip_reply_s_status_status-code_response-code_reply-code Good that it helps. $rs works in onreply_routes, but I guess Robert needed it in failure_route, where the $T_reply_code is the right option. Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Dispatcher does not trigger the event_route
Hello,
it proved that the msg parameter for event_route execution due to
keepalives was not set to the faked structure. The event route was
executed but many functions don't do anything as the sip message was
missing.
Try with latest git branch 3.2, I just pushed the fix for it.
Cheers,
Daniel
On 12/29/11 11:19 AM, Mino Haluz wrote:
No, I do not see anything related in messages nor in syslog. Otherwise
dispatcher is working fine, but that event-route is still not
triggered and I really need the information about failure...
The output of kamctl fifo ds_list is ok, one gw is marked as IP and
another one as AP. When I flush the firewall the both are AP, so the
mechanism is working ..
Is there any debugging procedure of event-routes?
On Tue, Dec 20, 2011 at 11:13 AM, Peter Dunkley
<mailto:peter.dunk...@crocodile-rcs.com>> wrote:
Once your destination has failed (and after you expect to see the
output from the event_route), what is the output of kamctl fifo
ds_list?
Do you see anything else in /var/log/messages at this point?
Peter
On Tue, 2011-12-20 at 11:05 +0100, Mino Haluz wrote:
nightly build 3.2.1.
compiled on 05:08:17 Dec 19 2011 with gcc 4.4.5
On Tue, Dec 20, 2011 at 10:49 AM, Peter Dunkley
<mailto:peter.dunk...@crocodile-rcs.com>> wrote:
Hello,
What version of Kamailio are you using?
The dispatcher "event_route"s were added in Kamailio 3.2.0 so
this will only work in Kamailio 3.2.0 and later.
Regards,
Peter
On Tue, 2011-12-20 at 10:06 +0100, Mino Haluz wrote:
Hi,
I have two gateways pinged by kamailio. The both are AP
(active/probing), when I cut the one gateway off, it becomes
IP(inactive/probing) but the event_route is not fired up. Am
I missing something ?
event_route[dispatcher:dst-down] {
xlog("L_ERR", "Destination down: $rm $ru ($du)\n");
}
Mino
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Crocodile RCS Ltd
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<mailto:sr-users@lists.sip-router.org>
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--
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Technical Director
Crocodile RCS Ltd
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--
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http://linkedin.com/in/miconda -- http://twitter.com/miconda
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Re: [SR-Users] No memory left in send_subscribe (PUA) from update_pua while building the tm dlg_t structure
Hello Laura,
I applied most of the patches, apart the second one for hash.c, related
to checking for temporary dialog. I need to look a bit more at it, since
there was some work in this regard:
http://git.sip-router.org/cgi-bin/gitweb.cgi/sip-router/?a=commitdiff;h=b93149c756d3e983c70608938f1142ed43ee1834
If I understood right, it is about NOTIFY getting processed faster than
200OK for SUBSCRIBE. Do I remember correctly, you are running 3.1.x? It
might not be in branch 3.1.
Cheers,
Daniel
On 12/28/11 6:15 PM, laura testi wrote:
Hi Daniel,
Yes, all patches are related to the PUA module. Please find the
attachments for both diff files and new files with applied patches If
you thinks they are useful. Now the old files should be the latest
version of the master branch ;-) Unfortunately I can not use git here.
We have tested the patches, now no memory leak any more; and no more
error for NOTIFY (dialog not found)...Please note, our use case is the
pua_xmpp, now it works fine. I don't know if rls has the same problem.
Thank you veru much for your helps again.
Some explanations:
In the send_subscribe.c,
- I just comment out the line 1172 so that the pua_free_tm_dlg(td)
do the right job later as you done also in this case.
In the hash.c,
- in the get_dialog function, I add the check of "p->to_tag.len> 0"
in the string compare conditions;
- in the is_dialog function, instead of check only the real dialog,
the check of also the temporary dialog is added, this also helps to
avoid the error of no dialog found for the NOTIFY if the NOTIFY is
received before the 202/200 OK of the SUBSCRIBE.
in the pua.c,
- a few changes in the update_pua function:
- apply the same kind of free td and td->route_set, unfortunately
I can not re-use pua_free_tm_dlg which is local for send_subscribe.c.
- change from goto error to goto done
Best Regards,
Laura
On Wed, Dec 28, 2011 at 2:39 PM, Daniel-Constantin Mierla
wrote:
Hello,
On 12/27/11 1:51 PM, laura testi wrote:
Hi Daniel,
I tried the patch, it works partially. There are stil memory leak.
Based on your patch, I did find different places need the same kind of
patch both in send_subscribe,c and in pua,c
ok, thanks, I will look over. But how did you do the patches, since it does
not take my latest patch, seems to be against an older version, because the
indentation is not there?
I cannot apply it like this, maybe you can tell the files and lines you
changed, otherwise is hard to track.
Are both patches to the pua module? You are doing the patch with files only,
it is more convenient to call the diff with path to the module, in this way
is easy to spot in which module to apply the path -- i.e., use diff from
root folder of kamailio, like:
diff -u modules/abc/oldfile modules/abc/newfile
Since you are sending a lot of patches, maybe it will work better if you
just clone the git, make the patch against the master branch -- change the
file in the master branch and then just do:
git diff> path/to/save/patch.file
Some info that could be useful for working with git and patches, including
backporting, at:
http://www.kamailio.org/wiki/devel/backporting-to-3.2.x
Cheers,
Daniel
Please find the modified version of these files.
Following are the differences:
# diff -u send_subscribe.c.orig send_subscribe.c
--- send_subscribe.c.orig 2011-12-27 13:31:06.0 +0100
+++ send_subscribe.c2011-12-27 13:31:51.0 +0100
@@ -1151,7 +1151,6 @@
if (dbmode!=PUA_DB_ONLY)
lock_release(&HashT->p_records[hash_code].lock);
ret= -1;
- pkg_free(td);
goto done;
}
if (dbmode!=PUA_DB_ONLY)
# diff -u pua.c.orig pua.c
--- pua.c.orig 2011-12-27 13:15:47.0 +0100
+++ pua.c 2011-12-27 13:26:33.0 +0100
@@ -673,106 +673,145 @@
int update_pua(ua_pres_t* p)
{
- str* str_hdr= NULL;
- int expires;
- int result;
- uac_req_t uac_r;
-
- if(p->desired_expires== 0)
- expires= 3600;
- else
- expires= p->desired_expires- (int)time(NULL);
+ str* str_hdr= NULL;
+ ua_pres_t* cb_param = NULL;
- if(p->watcher_uri== NULL)
- {
- str met= {"PUBLISH", 7};
- ua_pres_t* cb_param;
-
- str_hdr = publ_build_hdr(expires, get_event(p->event), NULL,
-&p->etag, p->extra_headers, 0);
- if(str_hdr == NULL)
- {
- LM_ERR("while building extra_headers\n");
- goto error;
- }
- LM_DBG("str_hdr:\n%.*s\n ", str_hdr->len, str_hdr->s);
-
- cb_param= build_uppubl_cbparam(p);
- if(cb_param== NULL)
- {
- LM_ERR("while constructing publ callba
[SR-Users] Happy New Year!
This was it for 2011, final timeout for a fruitful year! Many thanks to the community for keeping things moving forward! A happy and prosperous 2012 to all Kamailio and SER friends!!! Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Intermittent start up issue
Hello, check also for limits in the system/firewall, maybe there are limits on new connections per second. Try with children=1 and see if it happens again, if not, then it is something that blocks many new connections in short time. Cheers, Daniel On 1/2/12 5:31 AM, Sammy Govind wrote: Hi, Its all upto you then, that's an intermittent issue and information is not complete to trace the issue. Did you see the same Errors every time it fails !? -- Regards Sammy On Sun, Jan 1, 2012 at 10:06 PM, Ali Jawad <mailto:ali.ja...@splendor.net>> wrote: Hi Thanks for your input, but as said this works at times yes and times now so I am not sure it is a definite DB issue, the servers are in the same LAN, and there are other modules that use the DB string that work when launched before this one while this one does not work. Regards On Sun, Jan 1, 2012 at 4:13 PM, Sammy Govind mailto:govoi...@gmail.com>> wrote: > Hi, > forget the whole list of errors and just resolve the DB connectivity between > the Kamailio server and DB server first . > >> 27(24762) ERROR: db_mysql [km_my_con.c:109]: driver error: Can't >> connect to MySQL server on 'x' (4) > > > try connecting to the remote DB server 'x' from linux shell i.e > > # mysql -u -p -h 'x' > > Regards, > Sammy > > > On Fri, Dec 30, 2011 at 3:33 PM, Ali Jawad mailto:ali.ja...@splendor.net>> wrote: >> >> Hi All >> I have tried the below with Kamailio 3.1.0 and 3.2.1. the problem is >> that Kamailio at times fails to start up for no obvious reasons, I.e. >> in 10 restarts Kamailio might fail to start 3,0 or 5 times. Te debug >> log "level 2 " shows the following : >> >> 27(24762) ERROR: db_mysql [km_my_con.c:109]: driver error: Can't >> connect to MySQL server on 'x' (4) >> 27(24762) ERROR: [db.c:303]: could not add connection to the >> pool27(24762) ERROR: group [group.c:211]: unable to connect to the >> database >> 27(24762) ERROR: [sr_module.c:889]: init_mod_child(): Error >> while initializing module group >> (/usr/local/kamailio/lib/kamailio/modules_k/group.so) >> 27(24762) ERROR: [pt.c:481]: ERROR: fork_tcp_process(): >> init_child failed for process 27, pid 24762, "tcp receiver child=1" >> >> and then >> >> >> 30(24765) : [pass_fd.c:293]: ERROR: receive_fd: EOF on 45 >> 0(24735) ALERT: [main.c:748]: child process 24762 exited >> normally, status=255 >> 0(24735) INFO: [main.c:766]: INFO: terminating due to SIGCHLD >> >> The problem happens on 4 different servers, I did try to compile more >> than once, the servers are 32 Bit CentOS 5.6. The connection to the >> database is monitored and there are no issues with connecting to the >> DB "it works at times and times not for Kamailio so a user/pwd issue >> is not likely to cause this, manual cnnect worked", config file errors >> are not found. >> >> I did also check max no of connections on MySQL server which is set to >> 500 on the server and only 100 cons are used. The group module uses >> the same connection string as other modules which is predefined and >> there are other modules that access the db before the group module and >> those do not have a problem. >> >> Please advice. >> >> Regards >> >> ___ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >> sr-users@lists.sip-router.org <mailto:sr-users@lists.sip-router.org> >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > > ___ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users@lists.sip-router.org <mailto:sr-users@lists.sip-router.org> > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > -- Ali Jawad Information Systems Manager Splendor Telecom (www.splendor.net <http://www.splendor.net>) Beirut, Lebanon Phone: +9611373725 /ext 116 FAX: +9611375554 ___________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org <mailto:sr-users@lists.sip-router
Re: [SR-Users] No memory left in send_subscribe (PUA) from update_pua while building the tm dlg_t structure
Hello,
On 12/29/11 7:06 PM, laura testi wrote:
Hi Daniel,
yes we are running 3.1.5, but we have back ported the whole pua module
from the master branch that included also the patches in your link. In
fact, the second patch use the patches from the link you have
mentioned to solve the same problem. Because the patches themselves
from the link
(http://git.sip-router.org/cgi-bin/gitweb.cgi/sip-router/?a=commitdiff;h=b93149c756d3e983c70608938f1142ed43ee1834)
are not enough for the PUA_XMPP module.
ok, checking for side effects was the reason not to go ahead with the
patch immediately. I was looking and the is_dialog function seems to be
used just by other modules, not by pua itself: pua_bla, pua_xmpp and
purple. I think it is safe to go ahead with it, maybe in a different
form, like returning different values when it is established or
temporary dialog, eventually with an extra parameter to specify which
kind of dialogs should be searched -- this needs a bit of analyze to
choose the best.
Cheers,
Daniel
The same problem is true both for RLS and for PUA_XMPP. The real
problem is that for some strange reason (I don't know why), the
PUA_XMPP receive/process the NOTIFY message received from kamailio
presence server before the 200OK of the SUBSCRIBE. That's why the
previous patched implement the temporary dialog to avoid it. But it's
not enough, at least for PUA_XMPP.
If you have a look at the line 196 inside the function
Notify2Xmpp(struct sip_msg* msg, char* s1, char* s2) in the file
modules_k/pua_xmpp/simple2xmpp.c:
196 if(pua_is_dialog(&dialog)< 0) // verify if within a stored dialog
197 {
198 LM_ERR("Notify in a non existing dialog\n");
199 goto error;
200 }
In this case, the PUA_XMPP will print the Error and does not process
the NOTIFY message (convert it to XMPP and send it out to the XMPP
domain). And we got a lot of error "Notify in a non existing dialog"
in the log file. After we applied the patches from the link, the
errors are still there, because the pua_is_dialog call the function
is_dialog defined in the hash.c in the pua module, without the second
patch of hash.c, it will still return -1 because is_dialog checks the
dialog only against the real dialog in the hash, but not the temporary
one. After we have applied the patch in the is_dialog function. The
error is gone ;-) But I don't know if this may have impact on other
modules. I means I don't know if there are some modules use this
function to check if the only real dialog (but not temporary dialog)
is existing in the hash. In this case maybe is better to have 2 check
functions, one for real, another for both or temporary. What do you
think?
Best Regards,
Laura
But the patches
Yes, the second
On Thu, Dec 29, 2011 at 4:41 PM, Daniel-Constantin Mierla
wrote:
Hello Laura,
I applied most of the patches, apart the second one for hash.c, related to
checking for temporary dialog. I need to look a bit more at it, since there
was some work in this regard:
http://git.sip-router.org/cgi-bin/gitweb.cgi/sip-router/?a=commitdiff;h=b93149c756d3e983c70608938f1142ed43ee1834
If I understood right, it is about NOTIFY getting processed faster than
200OK for SUBSCRIBE. Do I remember correctly, you are running 3.1.x? It
might not be in branch 3.1.
Cheers,
Daniel
On 12/28/11 6:15 PM, laura testi wrote:
Hi Daniel,
Yes, all patches are related to the PUA module. Please find the
attachments for both diff files and new files with applied patches If
you thinks they are useful. Now the old files should be the latest
version of the master branch ;-) Unfortunately I can not use git here.
We have tested the patches, now no memory leak any more; and no more
error for NOTIFY (dialog not found)...Please note, our use case is the
pua_xmpp, now it works fine. I don't know if rls has the same problem.
Thank you veru much for your helps again.
Some explanations:
In the send_subscribe.c,
- I just comment out the line 1172 so that the pua_free_tm_dlg(td)
do the right job later as you done also in this case.
In the hash.c,
- in the get_dialog function, I add the check of "p->to_tag.len> 0"
in the string compare conditions;
- in the is_dialog function, instead of check only the real dialog,
the check of also the temporary dialog is added, this also helps to
avoid the error of no dialog found for the NOTIFY if the NOTIFY is
received before the 202/200 OK of the SUBSCRIBE.
in the pua.c,
- a few changes in the update_pua function:
- apply the same kind of free td and td->route_set, unfortunately
I can not re-use pua_free_tm_dlg which is local for send_subscribe.c.
- change from goto error to goto done
Best Regards,
Laura
On Wed, Dec 28, 2011 at 2:39 PM, Daniel-Constantin Mierla
wrote:
Hello,
On 12/27/11 1:51 PM, laura testi wrote:
Hi Daniel,
I tried the patch, it
Re: [SR-Users] usrloc, timer process and cache cleanup
Hello, On 12/23/11 1:39 PM, Andreas Granig wrote: Hi, On 12/23/2011 12:28 AM, Daniel-Constantin Mierla wrote: If someone is messing with the db, kamailio shouldn't try to correct admin mistakes. It was just an example, I haven't gone to all corner cases that can happen from human or (self or different) application errors. There were couple of similar reports in the past, related to conflicts of insert/update, update/insert, delete/update a.s.o. cases, so I proposed to go for a portable solution, not for one which valid to a db driver only -- configurable or not, is different thing than the specific topic. Of course you can never rule out admin errors or application errors, but anyways I'll highly favor a more resilient approach. I encountered several cases when things can go bad at database layer while doing cross replication along with t_replicate(). Finding a solution would be a good thing. The fact is that these situations should be corner cases, so there has to be a solution not impact the normal operation mode. I've tried to reproduce the most obvious scenario, which is registering a subscriber, delete it from the underlying db table, then re-register again. The expiry value in the cache is refreshed, but it's never written back to the db table. There are most likely other, more subtle scenarios, and our customers approved that they didn't mess with the db manually. The state was always CS_SYNC and Flags was 0. I don't know the details of the srdb layer, but probably it's possible to find a way to return the "rows affected" after an update in order to know whether to try an insert afterwards. Would be possible with mysql, not sure about pgsql, oracle, dbtext etc. We'll take a look how we could tackle that. Mysql has affected row and it is exported by db_mysql module, not sure about the other drivers. I don't think that a log message would help very much, because kamailio won't know about the missing entry in the db (unless you evaluate the result of the update), at least in this particular case. Yes, kamailio does not know, I know, seems it is what we try to solve -- first how to make it aware and then do some actions (like writing a log message). Perhaps the best for the moment is to detect at startup and rely on db api capabilities and use replace/affected_rows a.s.o. when they are available, in the best efficient order -- a module parameter can give admins the power to decide over the auto-detect. Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] usrloc, timer process and cache cleanup
On 12/23/11 8:43 PM, Andrew Pogrebennyk wrote: On 12/23/2011 08:18 PM, Stefan Sayer wrote: shouldn't the db layer and driver be smart enough to do insert ... on duplicate key update at least where it's supported? my fear is that such "first insert then update" policy will affect the performance. can create noise in the log on some db backends too.. this one is also a bit tricky to do, as it will require to change the database table definition depending on matching mode from registrar/usrloc. By RFC, the primary key per aor should be contact address for the location records. As we know that lot of phones are behind the nat, many users having same environment for home/work phones, kamailio can be configured to do the matching also with call id and path stack. Because of that, these checks are done inside the modules, there is no constraint at location table sql level. Such approach as suggested in this discussion, will require to add proper unique keys depending on configuration from kamailio.cfg. Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Intermittent start up issue
Hello, On 1/2/12 12:24 PM, Ali Jawad wrote: Hi Daniel Running on a LAN with no firewalls enabled, limits can be from kernel, selinix, mysql server a.s.o. not only from firewalls. will try you suggestion though first thing tomorrow morning, what are the implications of running children=1 ? This will create only one worker process (not advisable for production in long term, just for testing) -- in this way there should be very few mysql connections opened, revealing if there is a problem or not with the number of connections and the short period to create them. Cheers, Daniel Thanks On Mon, Jan 2, 2012 at 2:03 PM, Daniel-Constantin Mierla wrote: Hello, check also for limits in the system/firewall, maybe there are limits on new connections per second. Try with children=1 and see if it happens again, if not, then it is something that blocks many new connections in short time. Cheers, Daniel On 1/2/12 5:31 AM, Sammy Govind wrote: Hi, Its all upto you then, that's an intermittent issue and information is not complete to trace the issue. Did you see the same Errors every time it fails !? -- Regards Sammy On Sun, Jan 1, 2012 at 10:06 PM, Ali Jawad wrote: Hi Thanks for your input, but as said this works at times yes and times now so I am not sure it is a definite DB issue, the servers are in the same LAN, and there are other modules that use the DB string that work when launched before this one while this one does not work. Regards On Sun, Jan 1, 2012 at 4:13 PM, Sammy Govind wrote: Hi, forget the whole list of errors and just resolve the DB connectivity between the Kamailio server and DB server first . 27(24762) ERROR: db_mysql [km_my_con.c:109]: driver error: Can't connect to MySQL server on 'x' (4) try connecting to the remote DB server 'x' from linux shell i.e # mysql -u -p -h 'x' Regards, Sammy On Fri, Dec 30, 2011 at 3:33 PM, Ali Jawad wrote: Hi All I have tried the below with Kamailio 3.1.0 and 3.2.1. the problem is that Kamailio at times fails to start up for no obvious reasons, I.e. in 10 restarts Kamailio might fail to start 3,0 or 5 times. Te debug log "level 2 " shows the following : 27(24762) ERROR: db_mysql [km_my_con.c:109]: driver error: Can't connect to MySQL server on 'x' (4) 27(24762) ERROR: [db.c:303]: could not add connection to the pool27(24762) ERROR: group [group.c:211]: unable to connect to the database 27(24762) ERROR: [sr_module.c:889]: init_mod_child(): Error while initializing module group (/usr/local/kamailio/lib/kamailio/modules_k/group.so) 27(24762) ERROR: [pt.c:481]: ERROR: fork_tcp_process(): init_child failed for process 27, pid 24762, "tcp receiver child=1" and then 30(24765) : [pass_fd.c:293]: ERROR: receive_fd: EOF on 45 0(24735) ALERT: [main.c:748]: child process 24762 exited normally, status=255 0(24735) INFO: [main.c:766]: INFO: terminating due to SIGCHLD The problem happens on 4 different servers, I did try to compile more than once, the servers are 32 Bit CentOS 5.6. The connection to the database is monitored and there are no issues with connecting to the DB "it works at times and times not for Kamailio so a user/pwd issue is not likely to cause this, manual cnnect worked", config file errors are not found. I did also check max no of connections on MySQL server which is set to 500 on the server and only 100 cons are used. The group module uses the same connection string as other modules which is predefined and there are other modules that access the db before the group module and those do not have a problem. Please advice. Regards ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Ali Jawad Information Systems Manager Splendor Telecom (www.splendor.net) Beirut, Lebanon Phone: +9611373725/ext 116 FAX: +9611375554 ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-r
Re: [SR-Users] RTPproxy on Kamailio 3.2.1 difficulty.
Hello, the ADDR_IPV4 and ADDR_IPV6 have to be replaced with real IP addresses that you have on the system runnin RTPProxy. Cheers, Daniel On 1/3/12 8:56 AM, nunu abe wrote: Dear Daniel, Greetings and Happy New year to you all! I would like to thank you very much for your detailed and well structured tutorials on Kamailio. I am resending this email as I sent the earlier one before I was a member of the mailing list and I thought it may have been sent to the spam folder. If not then, I apologize for double posting. I was trying to design a similar scenario like the one " Run your own SIP VoIP service on both IPv4 and IPv6 " as explained on Asipto web-page(http://kb.asipto.com/kamailio:kamailio-mixed-ipv4-ipv6). However I ran into difficulties with rtpproxy. The proxy is not routing media packets as it should. I followed the instruction regarding the kamailio.cfg file and edited the necessary lines according to my settings. I think my problem comes from setting up the rtpproxy. 1. I edited the /etc/default/rtpproxy file. i.e. I uncommented the line - CONTROL_SOCK="unix:/var/run/rtpproxy/rtpproxy.sock" 2. I started the rtpproxy with this command: rtpproxy -F -l ADDR_IPV4 -6 /ADDR_IPV6 . However, after this command the system returns the message : rtpproxy: setbindhost: No address associated with hostname It displays the same message even when I replace ADDR_IPV4 and ADDR_IPV6 with IP addresses. 3. I have also tried using the udp socket:- CONTROL_SOCK=udp:127.0.0.1:2, of course I have replaced the unix socket by this udp socket in the kamilio.cfg file. So my question is, what did I do wrong with the settings that SIP messages are traversing normal from IPv4 client to IPv6 client and vice versa but media files(RTP packets) are not? The rtpproxy receives the media files from one client but fails to forward them to the other client. Thanks for your help. Maedot ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Intermittent start up issue
Hello, On 1/3/12 9:14 AM, Ali Jawad wrote: Hi Daniel SeLinux is disabled, as I have played around with Children and got a stable startup at CHILDREN=3, at 4 and up I get the problem described earlier right away, so it seems that MySQL is throttling Kamailio ? 1- What is the effect of running 3 children only ? the capacity of handling sip traffic is lowered. You should still be able to handle couple of hundred messages per second if you don't have slow sql/dns operations. For production, it is recommended to have at least 8 to be in safe side. 2- I know this is not a Kamailio thing, but any idea about what needs to be changed on MySQL server to accommodate more children ? No of connections is 500 conns out of which max 120 are utilized So mysql is on a different server, right? Are you sure there is no other switch/router doing such throttling? Can you watch on both kamailio and mysql server to see if the connections are initiated from kamailio and reaching mysql? Maybe you can discover if mysql server get the tcp connection but refuses it -- not sure if mysql server can run in debug/verbose mode to get some log messages with hints about what is the problem. If the tcp connection does not get to mysql server, then it is a throttling in mysql client library or on a device in between the servers. If you do 'kamctl ps' when having children=3, how many kamailio processes are there (extra should be timers, ...)? Each should have one mysql connection, so it helps to see at which number the limit starts. Cheers, Daniel Regards On Mon, Jan 2, 2012 at 1:31 PM, Daniel-Constantin Mierla wrote: Hello, On 1/2/12 12:24 PM, Ali Jawad wrote: Hi Daniel Running on a LAN with no firewalls enabled, limits can be from kernel, selinix, mysql server a.s.o. not only from firewalls. will try you suggestion though first thing tomorrow morning, what are the implications of running children=1 ? This will create only one worker process (not advisable for production in long term, just for testing) -- in this way there should be very few mysql connections opened, revealing if there is a problem or not with the number of connections and the short period to create them. Cheers, Daniel Thanks On Mon, Jan 2, 2012 at 2:03 PM, Daniel-Constantin Mierla wrote: Hello, check also for limits in the system/firewall, maybe there are limits on new connections per second. Try with children=1 and see if it happens again, if not, then it is something that blocks many new connections in short time. Cheers, Daniel On 1/2/12 5:31 AM, Sammy Govind wrote: Hi, Its all upto you then, that's an intermittent issue and information is not complete to trace the issue. Did you see the same Errors every time it fails !? -- Regards Sammy On Sun, Jan 1, 2012 at 10:06 PM, Ali Jawad wrote: Hi Thanks for your input, but as said this works at times yes and times now so I am not sure it is a definite DB issue, the servers are in the same LAN, and there are other modules that use the DB string that work when launched before this one while this one does not work. Regards On Sun, Jan 1, 2012 at 4:13 PM, Sammy Govindwrote: Hi, forget the whole list of errors and just resolve the DB connectivity between the Kamailio server and DB server first . 27(24762) ERROR: db_mysql [km_my_con.c:109]: driver error: Can't connect to MySQL server on 'x' (4) try connecting to the remote DB server 'x' from linux shell i.e # mysql -u-p-h 'x' Regards, Sammy On Fri, Dec 30, 2011 at 3:33 PM, Ali Jawad wrote: Hi All I have tried the below with Kamailio 3.1.0 and 3.2.1. the problem is that Kamailio at times fails to start up for no obvious reasons, I.e. in 10 restarts Kamailio might fail to start 3,0 or 5 times. Te debug log "level 2 " shows the following : 27(24762) ERROR: db_mysql [km_my_con.c:109]: driver error: Can't connect to MySQL server on 'x' (4) 27(24762) ERROR:[db.c:303]: could not add connection to the pool27(24762) ERROR: group [group.c:211]: unable to connect to the database 27(24762) ERROR:[sr_module.c:889]: init_mod_child(): Error while initializing module group (/usr/local/kamailio/lib/kamailio/modules_k/group.so) 27(24762) ERROR:[pt.c:481]: ERROR: fork_tcp_process(): init_child failed for process 27, pid 24762, "tcp receiver child=1" and then 30(24765) :[pass_fd.c:293]: ERROR: receive_fd: EOF on 45 0(24735) ALERT:[main.c:748]: child process 24762 exited normally, status=255 0(24735) INFO:[main.c:766]: INFO: terminating due to SIGCHLD The problem happens on 4 different servers, I did try to compile more than once, the servers are 32 Bit CentOS 5.6. The connection to the database is monitored and there are no issues with connecting to the DB "it works at times and times not for Kamailio so a user/pwd issue is not likely to cause this, manu
Re: [SR-Users] Intermittent start up issue
Hello, all it is fine with the output. So it looks like the limit is somewhere above 25 connections. As a try of making a workaround in kamailio side, could be a patch to add a global parameter to introduce some latency before forking a worker, so these processes won't be created all at a time, but one after the other. However, would be good to find the real reason for such behavior. If the tcp connection gets to mysql server, maybe the question should be addressed to mysql forums/mailing lists. Cheers, Daniel On 1/3/12 10:48 AM, Ali Jawad wrote: Hi I dont think a few hundred per second is enough I need to find out what is wrong and bring up no of children to 8. The MySQL server is at another host on the same LAN, as for the command kamctl ps see below Process:: ID=0 PID=13949 Type=attendant Process:: ID=1 PID=13950 Type=udp receiver child=0 sock=xx.xx.xx.xx:5060 Process:: ID=2 PID=13951 Type=udp receiver child=1 sock=xx.xx.xx.xx:5060 Process:: ID=3 PID=13952 Type=udp receiver child=2 sock=xx.xx.xx.xx:5060 Process:: ID=4 PID=13953 Type=udp receiver child=0 sock=xx.xx.xx.xx:5061 Process:: ID=5 PID=13954 Type=udp receiver child=1 sock=xx.xx.xx.xx:5061 Process:: ID=6 PID=13955 Type=udp receiver child=2 sock=xx.xx.xx.xx:5061 Process:: ID=7 PID=13956 Type=udp receiver child=0 sock=xx.xx.xx.xx:443 Process:: ID=8 PID=13957 Type=udp receiver child=1 sock=xx.xx.xx.xx:443 Process:: ID=9 PID=13958 Type=udp receiver child=2 sock=xx.xx.xx.xx:443 Process:: ID=10 PID=13959 Type=udp receiver child=0 sock=xx.xx.xx.xx:8080 Process:: ID=11 PID=13960 Type=udp receiver child=1 sock=xx.xx.xx.xx:8080 Process:: ID=12 PID=13961 Type=udp receiver child=2 sock=xx.xx.xx.xx:8080 Process:: ID=13 PID=13962 Type=udp receiver child=0 sock=xx.xx.xx.xx:123 Process:: ID=14 PID=13963 Type=udp receiver child=1 sock=xx.xx.xx.xx:123 Process:: ID=15 PID=13964 Type=udp receiver child=2 sock=xx.xx.xx.xx:123 Process:: ID=16 PID=13965 Type=slow timer Process:: ID=17 PID=13966 Type=timer Process:: ID=18 PID=13967 Type=MI FIFO Process:: ID=19 PID=13968 Type=ctl handler Process:: ID=20 PID=13969 Type=TIMER NH Process:: ID=21 PID=13970 Type=tcp receiver child=0 Process:: ID=22 PID=13971 Type=tcp receiver child=1 Process:: ID=23 PID=13972 Type=tcp receiver child=2 Process:: ID=24 PID=13973 Type=tcp main process I use a few different ports as you can see. All fine with that output at 3 ? Regards On Tue, Jan 3, 2012 at 11:29 AM, Daniel-Constantin Mierla wrote: Hello, On 1/3/12 9:14 AM, Ali Jawad wrote: Hi Daniel SeLinux is disabled, as I have played around with Children and got a stable startup at CHILDREN=3, at 4 and up I get the problem described earlier right away, so it seems that MySQL is throttling Kamailio ? 1- What is the effect of running 3 children only ? the capacity of handling sip traffic is lowered. You should still be able to handle couple of hundred messages per second if you don't have slow sql/dns operations. For production, it is recommended to have at least 8 to be in safe side. 2- I know this is not a Kamailio thing, but any idea about what needs to be changed on MySQL server to accommodate more children ? No of connections is 500 conns out of which max 120 are utilized So mysql is on a different server, right? Are you sure there is no other switch/router doing such throttling? Can you watch on both kamailio and mysql server to see if the connections are initiated from kamailio and reaching mysql? Maybe you can discover if mysql server get the tcp connection but refuses it -- not sure if mysql server can run in debug/verbose mode to get some log messages with hints about what is the problem. If the tcp connection does not get to mysql server, then it is a throttling in mysql client library or on a device in between the servers. If you do 'kamctl ps' when having children=3, how many kamailio processes are there (extra should be timers, ...)? Each should have one mysql connection, so it helps to see at which number the limit starts. Cheers, Daniel Regards On Mon, Jan 2, 2012 at 1:31 PM, Daniel-Constantin Mierla wrote: Hello, On 1/2/12 12:24 PM, Ali Jawad wrote: Hi Daniel Running on a LAN with no firewalls enabled, limits can be from kernel, selinix, mysql server a.s.o. not only from firewalls. will try you suggestion though first thing tomorrow morning, what are the implications of running children=1 ? This will create only one worker process (not advisable for production in long term, just for testing) -- in this way there should be very few mysql connections opened, revealing if there is a problem or not with the number of connections and the short period to create them. Cheers, Daniel Thanks On Mon, Jan 2, 2012 at 2:03 PM, Daniel-Constantin Mierla wrote: Hello, check also for limits in the system/firewall, maybe there are limits on new connections per second. Try with children=1 and see if it happens again, if not,
Re: [SR-Users] Intermittent start up issue
Hello, just another idea to try, not sure if it is convenient -- can you install and use different version of mysql server and client library? This should reveal whether is an issue specific to mysql version. Cheers, Daniel On 1/3/12 11:35 AM, Daniel-Constantin Mierla wrote: Hello, all it is fine with the output. So it looks like the limit is somewhere above 25 connections. As a try of making a workaround in kamailio side, could be a patch to add a global parameter to introduce some latency before forking a worker, so these processes won't be created all at a time, but one after the other. However, would be good to find the real reason for such behavior. If the tcp connection gets to mysql server, maybe the question should be addressed to mysql forums/mailing lists. Cheers, Daniel On 1/3/12 10:48 AM, Ali Jawad wrote: Hi I dont think a few hundred per second is enough I need to find out what is wrong and bring up no of children to 8. The MySQL server is at another host on the same LAN, as for the command kamctl ps see below Process:: ID=0 PID=13949 Type=attendant Process:: ID=1 PID=13950 Type=udp receiver child=0 sock=xx.xx.xx.xx:5060 Process:: ID=2 PID=13951 Type=udp receiver child=1 sock=xx.xx.xx.xx:5060 Process:: ID=3 PID=13952 Type=udp receiver child=2 sock=xx.xx.xx.xx:5060 Process:: ID=4 PID=13953 Type=udp receiver child=0 sock=xx.xx.xx.xx:5061 Process:: ID=5 PID=13954 Type=udp receiver child=1 sock=xx.xx.xx.xx:5061 Process:: ID=6 PID=13955 Type=udp receiver child=2 sock=xx.xx.xx.xx:5061 Process:: ID=7 PID=13956 Type=udp receiver child=0 sock=xx.xx.xx.xx:443 Process:: ID=8 PID=13957 Type=udp receiver child=1 sock=xx.xx.xx.xx:443 Process:: ID=9 PID=13958 Type=udp receiver child=2 sock=xx.xx.xx.xx:443 Process:: ID=10 PID=13959 Type=udp receiver child=0 sock=xx.xx.xx.xx:8080 Process:: ID=11 PID=13960 Type=udp receiver child=1 sock=xx.xx.xx.xx:8080 Process:: ID=12 PID=13961 Type=udp receiver child=2 sock=xx.xx.xx.xx:8080 Process:: ID=13 PID=13962 Type=udp receiver child=0 sock=xx.xx.xx.xx:123 Process:: ID=14 PID=13963 Type=udp receiver child=1 sock=xx.xx.xx.xx:123 Process:: ID=15 PID=13964 Type=udp receiver child=2 sock=xx.xx.xx.xx:123 Process:: ID=16 PID=13965 Type=slow timer Process:: ID=17 PID=13966 Type=timer Process:: ID=18 PID=13967 Type=MI FIFO Process:: ID=19 PID=13968 Type=ctl handler Process:: ID=20 PID=13969 Type=TIMER NH Process:: ID=21 PID=13970 Type=tcp receiver child=0 Process:: ID=22 PID=13971 Type=tcp receiver child=1 Process:: ID=23 PID=13972 Type=tcp receiver child=2 Process:: ID=24 PID=13973 Type=tcp main process I use a few different ports as you can see. All fine with that output at 3 ? Regards On Tue, Jan 3, 2012 at 11:29 AM, Daniel-Constantin Mierla wrote: Hello, On 1/3/12 9:14 AM, Ali Jawad wrote: Hi Daniel SeLinux is disabled, as I have played around with Children and got a stable startup at CHILDREN=3, at 4 and up I get the problem described earlier right away, so it seems that MySQL is throttling Kamailio ? 1- What is the effect of running 3 children only ? the capacity of handling sip traffic is lowered. You should still be able to handle couple of hundred messages per second if you don't have slow sql/dns operations. For production, it is recommended to have at least 8 to be in safe side. 2- I know this is not a Kamailio thing, but any idea about what needs to be changed on MySQL server to accommodate more children ? No of connections is 500 conns out of which max 120 are utilized So mysql is on a different server, right? Are you sure there is no other switch/router doing such throttling? Can you watch on both kamailio and mysql server to see if the connections are initiated from kamailio and reaching mysql? Maybe you can discover if mysql server get the tcp connection but refuses it -- not sure if mysql server can run in debug/verbose mode to get some log messages with hints about what is the problem. If the tcp connection does not get to mysql server, then it is a throttling in mysql client library or on a device in between the servers. If you do 'kamctl ps' when having children=3, how many kamailio processes are there (extra should be timers, ...)? Each should have one mysql connection, so it helps to see at which number the limit starts. Cheers, Daniel Regards On Mon, Jan 2, 2012 at 1:31 PM, Daniel-Constantin Mierla wrote: Hello, On 1/2/12 12:24 PM, Ali Jawad wrote: Hi Daniel Running on a LAN with no firewalls enabled, limits can be from kernel, selinix, mysql server a.s.o. not only from firewalls. will try you suggestion though first thing tomorrow morning, what are the implications of running children=1 ? This will create only one worker process (not advisable for production in long term, just for testing) -- in this way there should be very few mysql connections opened, revealing if there is a problem or not with the number of
[SR-Users] New Year, New Module: app_mono - embedded execution of managed code
Hello, the first committed module in 2012 is app_mono, which offers embedded execution of manage code (e.g., C#/.NET) via Mono project (http://www.mono-project.com/). The readme of the new module is available at: * http://kamailio.org/docs/modules/devel/modules/app_mono.html Current API which exported by SIP server for usage in C# application is documented at: * http://www.kamailio.org/wiki/embeddedapi/devel/mono Besides C#, app_mono should be able to run managed code compiled from applications written in other languages such as VisualBasic.NET, Java, Java Script, Python, ... more are listed at: * http://www.mono-project.com/Languages A primary use is integration with several widely used Microsoft technologies and APIs, for example C# having up-to-date libraries to connect to Active Directory LDAP service or MS SQL Server. Happy 2012, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Authentication Feature Question
Hello, you can simply use 'domain.com' as realm parameter to authentication function instead of $fd. Also build ha1 and ha1b with domain.com and then you are safe no matter which sip server is used. Of course you can build the realm by striping first token before '.' in $fd and pass it to authentication functions, but not sure if makes sense since it should be always domain.com Cheers, Daniel On 1/3/12 3:15 PM, Ali Jawad wrote: Hi After some research it seems to me that the only way to achieve this is to "try" and change how hashing is done in the source code, a little bit too ambitious for me, and it means I will have loads of problems each time an upgrade is released. Or Use pseudovariables to fix the value of the $fd value to something constant, while this worked for values like $var(y) I was not able to assign/strip $fd to remove the subdomain part. Any input please ? Regards On Tue, Jan 3, 2012 at 2:06 PM, Ali Jawad wrote: Hi I do have 3 Kamailio servers, one for mobile phone registrations, one for softphone registrations and one for SIP device registrations. Each of those devices connects to it's perspective kamailio server sip1.domain.com sip2.domain.com sip3.domain.com All 3 Kamailio servers share the same database, and users can use their kamailio user/pwd on any of the devices, now I want to use encrypted passwords and remove clear text passwords from the database. I did test with one server and all is fine,however if a user want to register from the second kamailio server it does not work, basically because the db domain entry from which the hash is created is sip1.domain.com and stored in the db, while the user connects from to sip2.domain.com this eventually generates a different hash. Is there anyway to overcome this ? Can I exclude Domain from Hash generation ? Any other option that allows me to do the above ? Thanks -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Intermittent start up issue
Hello, On 1/3/12 11:58 AM, Ali Jawad wrote: Same client library different server version working fine ! ok, good to know. What was the version of mysql sever giving troubles? Any chance the patch can be released anytime soon ? That was an idea, but it is not sure it will fix, it has to be tried. I will see when I have time for it and notify you to try it once is committed. Cheers, Daniel I will migrate to the newer MySQL server in the next few mos though, I can wait if needed. Regards On Tue, Jan 3, 2012 at 12:44 PM, Daniel-Constantin Mierla wrote: Hello, just another idea to try, not sure if it is convenient -- can you install and use different version of mysql server and client library? This should reveal whether is an issue specific to mysql version. Cheers, Daniel On 1/3/12 11:35 AM, Daniel-Constantin Mierla wrote: Hello, all it is fine with the output. So it looks like the limit is somewhere above 25 connections. As a try of making a workaround in kamailio side, could be a patch to add a global parameter to introduce some latency before forking a worker, so these processes won't be created all at a time, but one after the other. However, would be good to find the real reason for such behavior. If the tcp connection gets to mysql server, maybe the question should be addressed to mysql forums/mailing lists. Cheers, Daniel On 1/3/12 10:48 AM, Ali Jawad wrote: Hi I dont think a few hundred per second is enough I need to find out what is wrong and bring up no of children to 8. The MySQL server is at another host on the same LAN, as for the command kamctl ps see below Process:: ID=0 PID=13949 Type=attendant Process:: ID=1 PID=13950 Type=udp receiver child=0 sock=xx.xx.xx.xx:5060 Process:: ID=2 PID=13951 Type=udp receiver child=1 sock=xx.xx.xx.xx:5060 Process:: ID=3 PID=13952 Type=udp receiver child=2 sock=xx.xx.xx.xx:5060 Process:: ID=4 PID=13953 Type=udp receiver child=0 sock=xx.xx.xx.xx:5061 Process:: ID=5 PID=13954 Type=udp receiver child=1 sock=xx.xx.xx.xx:5061 Process:: ID=6 PID=13955 Type=udp receiver child=2 sock=xx.xx.xx.xx:5061 Process:: ID=7 PID=13956 Type=udp receiver child=0 sock=xx.xx.xx.xx:443 Process:: ID=8 PID=13957 Type=udp receiver child=1 sock=xx.xx.xx.xx:443 Process:: ID=9 PID=13958 Type=udp receiver child=2 sock=xx.xx.xx.xx:443 Process:: ID=10 PID=13959 Type=udp receiver child=0 sock=xx.xx.xx.xx:8080 Process:: ID=11 PID=13960 Type=udp receiver child=1 sock=xx.xx.xx.xx:8080 Process:: ID=12 PID=13961 Type=udp receiver child=2 sock=xx.xx.xx.xx:8080 Process:: ID=13 PID=13962 Type=udp receiver child=0 sock=xx.xx.xx.xx:123 Process:: ID=14 PID=13963 Type=udp receiver child=1 sock=xx.xx.xx.xx:123 Process:: ID=15 PID=13964 Type=udp receiver child=2 sock=xx.xx.xx.xx:123 Process:: ID=16 PID=13965 Type=slow timer Process:: ID=17 PID=13966 Type=timer Process:: ID=18 PID=13967 Type=MI FIFO Process:: ID=19 PID=13968 Type=ctl handler Process:: ID=20 PID=13969 Type=TIMER NH Process:: ID=21 PID=13970 Type=tcp receiver child=0 Process:: ID=22 PID=13971 Type=tcp receiver child=1 Process:: ID=23 PID=13972 Type=tcp receiver child=2 Process:: ID=24 PID=13973 Type=tcp main process I use a few different ports as you can see. All fine with that output at 3 ? Regards On Tue, Jan 3, 2012 at 11:29 AM, Daniel-Constantin Mierla wrote: Hello, On 1/3/12 9:14 AM, Ali Jawad wrote: Hi Daniel SeLinux is disabled, as I have played around with Children and got a stable startup at CHILDREN=3, at 4 and up I get the problem described earlier right away, so it seems that MySQL is throttling Kamailio ? 1- What is the effect of running 3 children only ? the capacity of handling sip traffic is lowered. You should still be able to handle couple of hundred messages per second if you don't have slow sql/dns operations. For production, it is recommended to have at least 8 to be in safe side. 2- I know this is not a Kamailio thing, but any idea about what needs to be changed on MySQL server to accommodate more children ? No of connections is 500 conns out of which max 120 are utilized So mysql is on a different server, right? Are you sure there is no other switch/router doing such throttling? Can you watch on both kamailio and mysql server to see if the connections are initiated from kamailio and reaching mysql? Maybe you can discover if mysql server get the tcp connection but refuses it -- not sure if mysql server can run in debug/verbose mode to get some log messages with hints about what is the problem. If the tcp connection does not get to mysql server, then it is a throttling in mysql client library or on a device in between the servers. If you do 'kamctl ps' when having children=3, how many kamailio processes are there (extra should be timers, ...)? Each should have one mysql connection, so it helps to see at which number the limit starts. Cheers, Daniel Regards On Mon, Jan 2, 2012 at 1:31 PM, Daniel-Constantin Mierla wro
Re: [SR-Users] Authentication Feature Question
Hello, On 1/3/12 4:12 PM, Ali Jawad wrote: Hi Daniel This certainly makes sense, I will try it in a few mins, but what I observed at Debug Level 3 is that Hash is calculated before www_authenticate is executed and it shows HA comparison failed, if I do use domain.com instead of $fd and use $domain.com in db domain field and build HA1 filed based on that, wont Kamailio still try to build the HA1 hash which it will compare form user:domain:pwd where domain is fed in to the hash function from the header of the SIP packet ? the ha1 is actually hash over 'user:realm:pwd' -- it is just common practice to use the domain as realm, since realm should be a unique token to identify the service, but it can be any random string. realm is given as parameter to auth functions in kamailio.cfg Cheers, Daniel Regards On Tue, Jan 3, 2012 at 5:07 PM, Daniel-Constantin Mierla wrote: Hello, you can simply use 'domain.com' as realm parameter to authentication function instead of $fd. Also build ha1 and ha1b with domain.com and then you are safe no matter which sip server is used. Of course you can build the realm by striping first token before '.' in $fd and pass it to authentication functions, but not sure if makes sense since it should be always domain.com Cheers, Daniel On 1/3/12 3:15 PM, Ali Jawad wrote: Hi After some research it seems to me that the only way to achieve this is to "try" and change how hashing is done in the source code, a little bit too ambitious for me, and it means I will have loads of problems each time an upgrade is released. Or Use pseudovariables to fix the value of the $fd value to something constant, while this worked for values like $var(y) I was not able to assign/strip $fd to remove the subdomain part. Any input please ? Regards On Tue, Jan 3, 2012 at 2:06 PM, Ali Jawadwrote: Hi I do have 3 Kamailio servers, one for mobile phone registrations, one for softphone registrations and one for SIP device registrations. Each of those devices connects to it's perspective kamailio server sip1.domain.com sip2.domain.com sip3.domain.com All 3 Kamailio servers share the same database, and users can use their kamailio user/pwd on any of the devices, now I want to use encrypted passwords and remove clear text passwords from the database. I did test with one server and all is fine,however if a user want to register from the second kamailio server it does not work, basically because the db domain entry from which the hash is created is sip1.domain.com and stored in the db, while the user connects from to sip2.domain.com this eventually generates a different hash. Is there anyway to overcome this ? Can I exclude Domain from Hash generation ? Any other option that allows me to do the above ? Thanks -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Authentication Feature Question
Hello,
On 1/3/12 4:48 PM, Ali Jawad wrote:
Hi Daniel
Please see
5(18649) DEBUG: [db_res.c:184]: allocate 8 bytes for rows at 0xb7b78d74
5(18649) DEBUG: [db_row.c:119]: allocate 20 bytes for row
values at 0xb7b78dac
5(18649) DEBUG: [db_val.c:117]: converting STRING
[6f966cd9c628f14cdc20172f96a4d065]
5(18649) DEBUG: auth [api.c:210]: check_response: Our result =
'6f95e6235edca0b7765042ef119fd83b'
5(18649) DEBUG: auth [api.c:220]: check_response: Authorization failed
5(18649) DEBUG: [db_res.c:81]: freeing 1 columns
5(18649) DEBUG: [db_res.c:85]: freeing RES_NAMES[0] at 0xb7b78d3c
5(18649) DEBUG: [db_res.c:94]: freeing result names at 0xb7b78bfc
5(18649) DEBUG: [db_res.c:99]: freeing result types at 0xb7b78c64
5(18649) DEBUG: [db_res.c:54]: freeing 1 rows
5(18649) DEBUG: [db_row.c:97]: freeing row values at 0xb7b78dac
5(18649) DEBUG: [db_res.c:62]: freeing rows at 0xb7b78d74
5(18649) DEBUG: [db_res.c:136]: freeing result set at 0xb7b78bb0
5(18649) DEBUG: auth [challenge.c:102]: build_challenge_hf: realm='nymgo.com'
5(18649) DEBUG: auth [challenge.c:244]: auth: 'WWW-Authenticate:
Digest realm="domain.com", nonce="TwMcuk8DG47SLxatlNdZfyfR8p3OiyAE"
'
I rebuilt the hashes against domain.com and then tried to connect to
sip1.domain.com and sip2.domain.com and sip3.domain.com with all of
the having
# - auth_db params -
#!ifdef WITH_AUTH
modparam("auth_db", "db_url", DBURL)
modparam("auth_db", "calculate_ha1", 0 )
modparam("auth_db", "password_column", "ha1")
modparam("auth_db", "load_credentials", "")
modparam("auth_db", "use_domain", MULTIDOMAIN)
And in the routing process :
if (!www_authorize("domain.com", "subscriber"))
{
www_challenge("domain.com", "0");
exit;
}
+
My point is that the hashes are caculated from user:doman:pwd which
are extracted from the SIP packet and in this case the domains are
sip1,sip2,sip3 while the hashes stored in the database are generated
against domain.com
If " ha1 is actually hash over 'user:realm:pwd' " shouldn't I have to
set the domain/realm in the config file ?
the first parameter in www_authorize and www_challenge functions is the
realm and it is used to build the digest response.
Is your phone putting user@domain in authorization header? Can you paste
here the ngrep of registration?
Cheers,
Daniel
I might be wrongthanks for the help so far.
Regards
On Tue, Jan 3, 2012 at 5:15 PM, Daniel-Constantin Mierla
wrote:
Hello,
On 1/3/12 4:12 PM, Ali Jawad wrote:
Hi Daniel
This certainly makes sense, I will try it in a few mins, but what I
observed at Debug Level 3 is that Hash is calculated before
www_authenticate is executed and it shows HA comparison failed, if I
do use domain.com instead of $fd and use $domain.com in db domain
field and build HA1 filed based on that, wont Kamailio still try to
build the HA1 hash which it will compare form user:domain:pwd where
domain is fed in to the hash function from the header of the SIP
packet ?
the ha1 is actually hash over 'user:realm:pwd' -- it is just common practice
to use the domain as realm, since realm should be a unique token to identify
the service, but it can be any random string. realm is given as parameter
to auth functions in kamailio.cfg
Cheers,
Daniel
Regards
On Tue, Jan 3, 2012 at 5:07 PM, Daniel-Constantin Mierla
wrote:
Hello,
you can simply use 'domain.com' as realm parameter to authentication
function instead of $fd. Also build ha1 and ha1b with domain.com and then
you are safe no matter which sip server is used.
Of course you can build the realm by striping first token before '.' in
$fd
and pass it to authentication functions, but not sure if makes sense
since
it should be always domain.com
Cheers,
Daniel
On 1/3/12 3:15 PM, Ali Jawad wrote:
Hi
After some research it seems to me that the only way to achieve this
is to "try" and change how hashing is done in the source code, a
little bit too ambitious for me, and it means I will have loads of
problems each time an upgrade is released.
Or
Use pseudovariables to fix the value of the $fd value to something
constant, while this worked for values like $var(y) I was not able to
assign/strip $fd to remove the subdomain part.
Any input please ?
Regards
On Tue, Jan 3, 2012 at 2:06 PM, Ali Jawad
wrote:
Hi
I do have 3 Kamailio servers, one for mobile phone registrations, one
for softphone registrations and one for SIP device registrations. Each
of those devices connects to it's perspective kamailio server
sip1.domain.com
sip2.domain.com
sip3.domain.com
All 3 Kamailio servers share the same database,
Re: [SR-Users] Authentication Feature Question
Hello,
I need the entire flow for registration, including the 401 reply and the
following REGISTER request.
Cheers,
Daniel
On 1/3/12 5:02 PM, Ali Jawad wrote:
Hi
Please see the ngrep below, please note that if I use in the db
register.domain.com and generate a hash against it for HA1 it
works,but then the user cant logon to register2.domain.com
U +10.630576 xx.yy.yy.yy:20020 -> xx.xx.xx.xx:5060
REGISTER sip:register.domain SIP/2.0.
Via: SIP/2.0/UDP
192.168.0.191:20020;branch=z9hG4bK-d8754z-af65a442980c375f-1---d8754z-;rport.
Max-Forwards: 70.
Contact:.
To: "Test Ast".
From: "Test Ast";tag=976bb004.
Call-ID: MDFmZDU2ZTI2YjJjMGNlMGFmOTIzMWFmZGQ1ZTNjMDE..
CSeq: 1 REGISTER.
Expires: 3600.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
SUBSCRIBE, INFO.
User-Agent: eyeBeam release 1102q stamp 51814.
Content-Length: 0.
On Tue, Jan 3, 2012 at 5:58 PM, Daniel-Constantin Mierla
wrote:
Hello,
On 1/3/12 4:48 PM, Ali Jawad wrote:
Hi Daniel
Please see
5(18649) DEBUG:[db_res.c:184]: allocate 8 bytes for rows at
0xb7b78d74
5(18649) DEBUG:[db_row.c:119]: allocate 20 bytes for row
values at 0xb7b78dac
5(18649) DEBUG:[db_val.c:117]: converting STRING
[6f966cd9c628f14cdc20172f96a4d065]
5(18649) DEBUG: auth [api.c:210]: check_response: Our result =
'6f95e6235edca0b7765042ef119fd83b'
5(18649) DEBUG: auth [api.c:220]: check_response: Authorization failed
5(18649) DEBUG:[db_res.c:81]: freeing 1 columns
5(18649) DEBUG:[db_res.c:85]: freeing RES_NAMES[0] at 0xb7b78d3c
5(18649) DEBUG:[db_res.c:94]: freeing result names at 0xb7b78bfc
5(18649) DEBUG:[db_res.c:99]: freeing result types at 0xb7b78c64
5(18649) DEBUG:[db_res.c:54]: freeing 1 rows
5(18649) DEBUG:[db_row.c:97]: freeing row values at 0xb7b78dac
5(18649) DEBUG:[db_res.c:62]: freeing rows at 0xb7b78d74
5(18649) DEBUG:[db_res.c:136]: freeing result set at 0xb7b78bb0
5(18649) DEBUG: auth [challenge.c:102]: build_challenge_hf:
realm='nymgo.com'
5(18649) DEBUG: auth [challenge.c:244]: auth: 'WWW-Authenticate:
Digest realm="domain.com", nonce="TwMcuk8DG47SLxatlNdZfyfR8p3OiyAE"
'
I rebuilt the hashes against domain.com and then tried to connect to
sip1.domain.com and sip2.domain.com and sip3.domain.com with all of
the having
# - auth_db params -
#!ifdef WITH_AUTH
modparam("auth_db", "db_url", DBURL)
modparam("auth_db", "calculate_ha1", 0 )
modparam("auth_db", "password_column", "ha1")
modparam("auth_db", "load_credentials", "")
modparam("auth_db", "use_domain", MULTIDOMAIN)
And in the routing process :
if (!www_authorize("domain.com", "subscriber"))
{
www_challenge("domain.com", "0");
exit;
}
+
My point is that the hashes are caculated from user:doman:pwd which
are extracted from the SIP packet and in this case the domains are
sip1,sip2,sip3 while the hashes stored in the database are generated
against domain.com
If " ha1 is actually hash over 'user:realm:pwd' " shouldn't I have to
set the domain/realm in the config file ?
the first parameter in www_authorize and www_challenge functions is the
realm and it is used to build the digest response.
Is your phone putting user@domain in authorization header? Can you paste
here the ngrep of registration?
Cheers,
Daniel
I might be wrongthanks for the help so far.
Regards
On Tue, Jan 3, 2012 at 5:15 PM, Daniel-Constantin Mierla
wrote:
Hello,
On 1/3/12 4:12 PM, Ali Jawad wrote:
Hi Daniel
This certainly makes sense, I will try it in a few mins, but what I
observed at Debug Level 3 is that Hash is calculated before
www_authenticate is executed and it shows HA comparison failed, if I
do use domain.com instead of $fd and use $domain.com in db domain
field and build HA1 filed based on that, wont Kamailio still try to
build the HA1 hash which it will compare form user:domain:pwd where
domain is fed in to the hash function from the header of the SIP
packet ?
the ha1 is actually hash over 'user:realm:pwd' -- it is just common
practice
to use the domain as realm, since realm should be a unique token to
identify
the service, but it can be any random string. realm is given as
parameter
to auth functions in kamailio.cfg
Cheers,
Daniel
Regards
On Tue, Jan 3, 2012 at 5:07 PM, Daniel-Constantin Mierla
wrote:
Hello,
you can simply use 'domain.com' as realm parameter to authentication
function instead of $fd. Also build ha1 and ha1b with domain.com and
then
you are safe no matter which sip server is used.
Of course you can build the realm by striping first token before '.' in
$fd
and pass it to authentic
Re: [SR-Users] Authentication Feature Question
G4bK-d8754z-81517d296225234f-1---d8754z-;rport=63610;received=yy.yy.yy.146.
To: "Test
Ast";tag=e597ca73bdb255490f0cefa03b2fda82.2f76.
From: "Test Ast";tag=ba705453.
Call-ID: MzM4Y2IwNzQzZGYwOGI3ZTY1YzYxZjcyZGJjMTMwODI..
CSeq: 2 REGISTER.
WWW-Authenticate: Digest realm="domain.com",
nonce="TwMpsU8DKIX4lvWDwPsV4iUhCl+iOAdk".
Server: kamailio (3.2.1 (i386/linux)).
Content-Length: 0.
.
On Tue, Jan 3, 2012 at 6:08 PM, Daniel-Constantin Mierla
wrote:
Hello,
I need the entire flow for registration, including the 401 reply and the
following REGISTER request.
Cheers,
Daniel
On 1/3/12 5:02 PM, Ali Jawad wrote:
Hi
Please see the ngrep below, please note that if I use in the db
register.domain.com and generate a hash against it for HA1 it
works,but then the user cant logon to register2.domain.com
U +10.630576 xx.yy.yy.yy:20020 ->xx.xx.xx.xx:5060
REGISTER sip:register.domain SIP/2.0.
Via: SIP/2.0/UDP
192.168.0.191:20020;branch=z9hG4bK-d8754z-af65a442980c375f-1---d8754z-;rport.
Max-Forwards: 70.
Contact:.
To: "Test Ast".
From: "Test Ast";tag=976bb004.
Call-ID: MDFmZDU2ZTI2YjJjMGNlMGFmOTIzMWFmZGQ1ZTNjMDE..
CSeq: 1 REGISTER.
Expires: 3600.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
SUBSCRIBE, INFO.
User-Agent: eyeBeam release 1102q stamp 51814.
Content-Length: 0.
On Tue, Jan 3, 2012 at 5:58 PM, Daniel-Constantin Mierla
wrote:
Hello,
On 1/3/12 4:48 PM, Ali Jawad wrote:
Hi Daniel
Please see
5(18649) DEBUG: [db_res.c:184]: allocate 8 bytes for rows at
0xb7b78d74
5(18649) DEBUG: [db_row.c:119]: allocate 20 bytes for row
values at 0xb7b78dac
5(18649) DEBUG: [db_val.c:117]: converting STRING
[6f966cd9c628f14cdc20172f96a4d065]
5(18649) DEBUG: auth [api.c:210]: check_response: Our result =
'6f95e6235edca0b7765042ef119fd83b'
5(18649) DEBUG: auth [api.c:220]: check_response: Authorization failed
5(18649) DEBUG: [db_res.c:81]: freeing 1 columns
5(18649) DEBUG: [db_res.c:85]: freeing RES_NAMES[0] at
0xb7b78d3c
5(18649) DEBUG: [db_res.c:94]: freeing result names at
0xb7b78bfc
5(18649) DEBUG: [db_res.c:99]: freeing result types at
0xb7b78c64
5(18649) DEBUG: [db_res.c:54]: freeing 1 rows
5(18649) DEBUG: [db_row.c:97]: freeing row values at
0xb7b78dac
5(18649) DEBUG: [db_res.c:62]: freeing rows at 0xb7b78d74
5(18649) DEBUG: [db_res.c:136]: freeing result set at
0xb7b78bb0
5(18649) DEBUG: auth [challenge.c:102]: build_challenge_hf:
realm='nymgo.com'
5(18649) DEBUG: auth [challenge.c:244]: auth: 'WWW-Authenticate:
Digest realm="domain.com", nonce="TwMcuk8DG47SLxatlNdZfyfR8p3OiyAE"
'
I rebuilt the hashes against domain.com and then tried to connect to
sip1.domain.com and sip2.domain.com and sip3.domain.com with all of
the having
# - auth_db params -
#!ifdef WITH_AUTH
modparam("auth_db", "db_url", DBURL)
modparam("auth_db", "calculate_ha1", 0 )
modparam("auth_db", "password_column", "ha1")
modparam("auth_db", "load_credentials", "")
modparam("auth_db", "use_domain", MULTIDOMAIN)
And in the routing process :
if (!www_authorize("domain.com", "subscriber"))
{
www_challenge("domain.com", "0");
exit;
}
+
My point is that the hashes are caculated from user:doman:pwd which
are extracted from the SIP packet and in this case the domains are
sip1,sip2,sip3 while the hashes stored in the database are generated
against domain.com
If " ha1 is actually hash over 'user:realm:pwd' " shouldn't I have to
set the domain/realm in the config file ?
the first parameter in www_authorize and www_challenge functions is the
realm and it is used to build the digest response.
Is your phone putting user@domain in authorization header? Can you paste
here the ngrep of registration?
Cheers,
Daniel
I might be wrongthanks for the help so far.
Regards
On Tue, Jan 3, 2012 at 5:15 PM, Daniel-Constantin Mierla
wrote:
Hello,
On 1/3/12 4:12 PM, Ali Jawad wrote:
Hi Daniel
This certainly makes sense, I will try it in a few mins, but what I
observed at Debug Level 3 is that Hash is calculated before
www_authenticate is executed and it shows HA comparison failed, if I
do use domain.com instead of $fd and use $domain.com in db domain
field and build HA1 filed based on that, wont Kamailio still try to
build the HA1 hash which it will compare form user:domain:pwd where
domain is fed in to the hash function from the header of the SIP
packet ?
the ha1 is actually hash over 'user:realm:pwd' -- it is just common
practice
to use the domain as realm, since realm should be a unique token to
identify
the service, but it c
Re: [SR-Users] Authentication Feature Question
Hello, On 1/3/12 10:08 PM, Ali Jawad wrote: Hi In Xlite/eyebeam I put in the username and one of my 3 kamailio servers respectively as the sip registrar, I.e. register1.domain.com, register2.domain.com and register3.domain.com, that is why I was saying that hashing will create different hashes based on the register domain. With reference to my first related post, one kamailio server is for softphones, one for sip devices and one for mobile phones with SIP software. Each server has a slightly different NAT and Call routing structure. but your service domain is 'domain.com', specific server addresses (domains) per UA type should be set as outbound proxy on the UA devices. This is why I wanted to eliminate the domain factor from the hashing procedure, I am sure it chooses the right value from the DB value because it is the same value shown in the log of kamailio against which a comparison is being done and because it works if I put register.domain.com in the domain column rehash the HA! value of the db. When auth_db module is configured to take the hashed value from database (calculate_ha1 parameter is 0), there is no more computation of it in kamailio -- it is just fetched from database and used -- so it does not matter anymore the domains in the sip message. Check to see if the xlite does not have a realm field that has to be set properly. values at 0xb7b78dac 5(18649) DEBUG:[db_val.c:117]: converting STRING [6f966cd9c628f14cdc20172f96a4d065]<# This is the value in the DB based on domain.com 5(18649) DEBUG: auth [api.c:210]: check_response: Our result = '6f95e6235edca0b7765042ef119fd83b'<# This value appears to be the value generated register.domain.com 5(18649) DEBUG: auth [api.c:220]: check_response: Authorization How can I enable the SQL query log ? I don't know by hart, googling should help you. Cheers, Daniel On Tue, Jan 3, 2012 at 8:12 PM, Daniel-Constantin Mierla wrote: Hello, why are you using register.domain.com as domain for registration? Isn't domain.com the right one? Can you enable the sql query log for mysql server and double check if the right value (column) is selected from the database table? Cheers, Daniel On 1/3/12 5:13 PM, Ali Jawad wrote: Hi Please see the below, thanks ! interface: eth1 (xx.xx.xx.0/255.255.255.0) match: support1 U +6.698682 yy.yy.yy.146:18832 ->xx.xx.xx.51:5060 REGISTER sip:register.domain.com SIP/2.0. Via: SIP/2.0/UDP 192.168.0.191:18832;branch=z9hG4bK-d8754z-05164a466600837d-1---d8754z-;rport. Max-Forwards: 70. Contact:. To: "Test Ast". From: "Test Ast";tag=f710b930. Call-ID: Y2RmNmFhZWM2MzM5OWQ5ODEwNjc0NzJiNGE2MmQzYjY.. CSeq: 1 REGISTER. Expires: 3600. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. User-Agent: eyeBeam release 1102q stamp 51814. Content-Length: 0. . U +0.005245 xx.xx.xx.51:5060 ->yy.yy.yy.146:18832 SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP 192.168.0.191:18832;branch=z9hG4bK-d8754z-05164a466600837d-1---d8754z-;rport=18832;received=yy.yy.yy.146. To: "Test Ast";tag=e597ca73bdb255490f0cefa03b2fda82.3053. From: "Test Ast";tag=f710b930. Call-ID: Y2RmNmFhZWM2MzM5OWQ5ODEwNjc0NzJiNGE2MmQzYjY.. CSeq: 1 REGISTER. WWW-Authenticate: Digest realm="domain.com", nonce="TwMpUE8DKCSt/IP7jcNRMbCT4TnbqlHl". Server: kamailio (3.2.1 (i386/linux)). Content-Length: 0. . U +0.428042 yy.yy.yy.146:18832 ->xx.xx.xx.51:5060 REGISTER sip:register.domain.com SIP/2.0. Via: SIP/2.0/UDP 192.168.0.191:18832;branch=z9hG4bK-d8754z-fc633b21627dca6d-1---d8754z-;rport. Max-Forwards: 70. Contact:. To: "Test Ast". From: "Test Ast";tag=f710b930. Call-ID: Y2RmNmFhZWM2MzM5OWQ5ODEwNjc0NzJiNGE2MmQzYjY.. CSeq: 2 REGISTER. Expires: 3600. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. User-Agent: eyeBeam release 1102q stamp 51814. Authorization: Digest username="support1",realm="domain.com",nonce="TwMpUE8DKCSt/IP7jcNRMbCT4TnbqlHl",uri="sip:register.domain.com",response="6122245b77e2df0b90179304464341e7",algorithm=MD5. Content-Length: 0. . U +0.005146 xx.xx.xx.51:5060 ->yy.yy.yy.146:18832 SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP 192.168.0.191:18832;branch=z9hG4bK-d8754z-fc633b21627dca6d-1---d8754z-;rport=18832;received=yy.yy.yy.146. To: "Test Ast";tag=e597ca73bdb255490f0cefa03b2fda82.fce3. From: "Test Ast";tag=f710b930. Call-ID: Y2RmNmFhZWM2MzM5OWQ5ODEwNjc0NzJiNGE2MmQzYjY.. CSeq: 2 REGISTER. WWW-Authenticate: Digest realm="domain.com", nonce="TwMpUE8DKCSt/IP7jcNRMbCT4TnbqlHl". Server: kamailio (3.2.1 (i386/linux)). Content-Length: 0. . [root@kam-rtp-100-51 mysql]# mail alijaw...@gmail.com<output.txt [root@kam-rtp-100-51 mysql]# ngrep -W byline -T support1 -q -d eth1 interface: eth1 (xx.xx.xx.0/255.255.2
Re: [SR-Users] app_lua coredump
Hello,
can you send the backtrace? Locate the core file and do:
gdb /path/to/kamailio /path/to/corefile
bt
Cheers,
Daniel
On 1/4/12 3:39 AM, (by way of n...@nelcom-voip.com) wrote:
I've installed kamailio (3.2, 3.2-nightly, dev-nightly) for debian unstable
(wheezy)
using repository at http://deb.kamailio.org and constantly get a coredump
with this lua code:
-- SIP request routing
function route_request(name)
sr.dbg("routing SIP request from Lua [" .. name .. "]\n")
-- initial checks of SIP request
if sr.maxfwd.process_maxfwd(10)< 0 then
sr.sl.send_reply(483,"Too Many Hops")
return
end
end
I've started with an example at this page
http://kb.asipto.com/kamailio:usage:k32-lua-routing,
and then just reduced the lua script to code above.
If I do comment sr.maxfwd.process_maxfwd(10) it passes. It is impossible to
call sr.modf("mf_maxfwd...") because it says about fixups.
What's wrong?
--
With best regards,
Roman Galeev.
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
Daniel-Constantin Mierla -- http://www.asipto.com
http://linkedin.com/in/miconda -- http://twitter.com/miconda
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Authentication Feature Question
Hello,
can you get the backtrace? Locate the core file (perhaps in / directory
if you don't use -w command line parameter) and do:
gdb /path/to/kamailio /path/to/corefile
bt
Cheers,
Daniel
On 1/4/12 9:17 AM, Ali Jawad wrote:
Hi
When I did set modparam("auth_db", "calculate_ha1", 0 )
Kamailio is crashing see
http://pastebin.com/anaKan0Y
Regards
On Tue, Jan 3, 2012 at 11:54 PM, Ali Jawad wrote:
Hi
It fetches one value from the database to compare it against a second
value that has to be computed right, in the computation of the second
hash where is the domain part fetched from ?
Regards
On Wed, Jan 4, 2012 at 12:26 AM, Daniel-Constantin Mierla
wrote:
Hello,
On 1/3/12 10:08 PM, Ali Jawad wrote:
Hi
In Xlite/eyebeam I put in the username and one of my 3 kamailio
servers respectively as the sip registrar, I.e. register1.domain.com,
register2.domain.com and register3.domain.com, that is why I was
saying that hashing will create different hashes based on the register
domain. With reference to my first related post, one kamailio server
is for softphones, one for sip devices and one for mobile phones with
SIP software. Each server has a slightly different NAT and Call
routing structure.
but your service domain is 'domain.com', specific server addresses (domains)
per UA type should be set as outbound proxy on the UA devices.
This is why I wanted to eliminate the domain factor from the hashing
procedure, I am sure it chooses the right value from the DB value
because it is the same value shown in the log of kamailio against
which a comparison is being done and because it works if I put
register.domain.com in the domain column rehash the HA! value of the
db.
When auth_db module is configured to take the hashed value from database
(calculate_ha1 parameter is 0), there is no more computation of it in
kamailio -- it is just fetched from database and used -- so it does not
matter anymore the domains in the sip message.
Check to see if the xlite does not have a realm field that has to be set
properly.
values at 0xb7b78dac
5(18649) DEBUG: [db_val.c:117]: converting STRING
[6f966cd9c628f14cdc20172f96a4d065]<# This is the value in
the DB based on domain.com
5(18649) DEBUG: auth [api.c:210]: check_response: Our result =
'6f95e6235edca0b7765042ef119fd83b'<# This value appears to
be the value generated register.domain.com
5(18649) DEBUG: auth [api.c:220]: check_response: Authorization
How can I enable the SQL query log ?
I don't know by hart, googling should help you.
Cheers,
Daniel
On Tue, Jan 3, 2012 at 8:12 PM, Daniel-Constantin Mierla
wrote:
Hello,
why are you using register.domain.com as domain for registration? Isn't
domain.com the right one?
Can you enable the sql query log for mysql server and double check if the
right value (column) is selected from the database table?
Cheers,
Daniel
On 1/3/12 5:13 PM, Ali Jawad wrote:
Hi
Please see the below, thanks !
interface: eth1 (xx.xx.xx.0/255.255.255.0)
match: support1
U +6.698682 yy.yy.yy.146:18832 -> xx.xx.xx.51:5060
REGISTER sip:register.domain.com SIP/2.0.
Via: SIP/2.0/UDP
192.168.0.191:18832;branch=z9hG4bK-d8754z-05164a466600837d-1---d8754z-;rport.
Max-Forwards: 70.
Contact:.
To: "Test Ast".
From: "Test Ast";tag=f710b930.
Call-ID: Y2RmNmFhZWM2MzM5OWQ5ODEwNjc0NzJiNGE2MmQzYjY..
CSeq: 1 REGISTER.
Expires: 3600.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
SUBSCRIBE, INFO.
User-Agent: eyeBeam release 1102q stamp 51814.
Content-Length: 0.
.
U +0.005245 xx.xx.xx.51:5060 -> yy.yy.yy.146:18832
SIP/2.0 401 Unauthorized.
Via: SIP/2.0/UDP
192.168.0.191:18832;branch=z9hG4bK-d8754z-05164a466600837d-1---d8754z-;rport=18832;received=yy.yy.yy.146.
To: "Test
Ast";tag=e597ca73bdb255490f0cefa03b2fda82.3053.
From: "Test Ast";tag=f710b930.
Call-ID: Y2RmNmFhZWM2MzM5OWQ5ODEwNjc0NzJiNGE2MmQzYjY..
CSeq: 1 REGISTER.
WWW-Authenticate: Digest realm="domain.com",
nonce="TwMpUE8DKCSt/IP7jcNRMbCT4TnbqlHl".
Server: kamailio (3.2.1 (i386/linux)).
Content-Length: 0.
.
U +0.428042 yy.yy.yy.146:18832 -> xx.xx.xx.51:5060
REGISTER sip:register.domain.com SIP/2.0.
Via: SIP/2.0/UDP
192.168.0.191:18832;branch=z9hG4bK-d8754z-fc633b21627dca6d-1---d8754z-;rport.
Max-Forwards: 70.
Contact:.
To: "Test Ast".
From: "Test Ast";tag=f710b930.
Call-ID: Y2RmNmFhZWM2MzM5OWQ5ODEwNjc0NzJiNGE2MmQzYjY..
CSeq: 2 REGISTER.
Expires: 3600.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
SUBSCRIBE, INFO.
User-Agent: eyeBeam release 1102q stamp 51814.
Authorization: Digest
username="support1",realm="domain.com",nonce="TwMpUE8DKCSt/IP7jcNRMbCT4TnbqlHl",uri="sip:register.domain.com",response="6122245b77e2df0b90179304464341e7",algorithm=MD5.
Content-Length: 0.
.
U +0.005146 xx.xx.xx.
Re: [SR-Users] Authentication Feature Question
io/lib/kamailio/modules_k/kex.soReading
symbols from /usr/local/kamailio/lib/kamailio/modules/tm.so...done.Loaded
symbols for /usr/local/kamailio/lib/kamailio/modules/tm.soReading
symbols from /usr/local/kamailio/lib/kamailio/modules_k/tmx.so...done.Loaded
symbols for /usr/local/kamailio/lib/kamailio/modules_k/tmx.soReading
symbols from /usr/local/kamailio/lib/kamailio/modules/sl.so...done.Loaded
symbols for /usr/local/kamailio/lib/kamailio/modules/sl.soReading
symbols from /usr/local/kamailio/lib/kamailio/modules_k/rr.so...done.Loaded
symbols for /usr/local/kamailio/lib/kamailio/modules_k/rr.soReading
symbols from /usr/local/kamailio/lib/kamailio/modules_k/pv.so...done.Loaded
symbols for /usr/local/kamailio/lib/kamailio/modules_k/pv.soReading
symbols from /usr/local/kamailio/lib/kamailio/modules_k/maxfwd.so...done.Loaded
symbols for /usr/local/kamailio/lib/kamailio/modules_k/maxfwd.soReading
symbols from /usr/local/kamailio/lib/kamailio/modules_k/usrloc.so...done.Loaded
symbols for /usr/local/kamailio/lib/kamailio/modules_k/usrloc.soReading
symbols from
/usr/local/kamailio/lib/kamailio/modules_k/registrar.so...done.Loaded
symbols for /usr/local/kamailio/lib/kamailio/modules_k/registrar.soReading
symbols from /usr/local/kamailio/lib/kamailio/modules_k/textops.so...done.Loaded
symbols for /usr/local/kamailio/lib/kamailio/modules_k/textops.soReading
symbols from
/usr/local/kamailio/lib/kamailio/modules_k/siputils.so...done.Loaded
symbols for /usr/local/kamailio/lib/kamailio/modules_k/siputils.soReading
symbols from /usr/local/kamailio/lib/kamailio/modules_k/xlog.so...done.Loaded
symbols for /usr/local/kamailio/lib/kamailio/modules_k/xlog.soReading
symbols from /usr/local/kamailio/lib/kamailio/modules/sanity.so...done.Loaded
symbols for /usr/local/kamailio/lib/kamailio/modules/sanity.soReading
symbols from /usr/local/kamailio/lib/kamailio/modules/ctl.so...done.Loaded
symbols for /usr/local/kamailio/lib/kamailio/modules/ctl.soReading
symbols from /usr/local/kamailio/lib/kamailio/modules/mi_rpc.so...done.Loaded
symbols for /usr/local/kamailio/lib/kamailio/modules/mi_rpc.soReading
symbols from /usr/local/kamailio/lib/kamailio/modules_k/acc.so...done.Loaded
symbols for /usr/local/kamailio/lib/kamailio/modules_k/acc.soReading
symbols from /usr/local/kamailio/lib/kamailio/modules/auth.so...done.Loaded
symbols for /usr/local/kamailio/lib/kamailio/modules/auth.soReading
symbols from /usr/local/kamailio/lib/kamailio/modules_k/auth_db.so...done.Loaded
symbols for /usr/local/kamailio/lib/kamailio/modules_k/auth_db.soReading
symbols from
/usr/local/kamailio/lib/kamailio/modules_k/alias_db.so...done.Loaded
symbols for /usr/local/kamailio/lib/kamailio/modules_k/alias_db.soReading
symbols from
/usr/local/kamailio/lib/kamailio/modules_k/nathelper.so...done.Loaded
symbols for /usr/local/kamailio/lib/kamailio/modules_k/nathelper.soReading
symbols from /usr/local/kamailio/lib/kamailio/modules/rtpproxy.so...done.Loaded
symbols for /usr/local/kamailio/lib/kamailio/modules/rtpproxy.soReading
symbols from /usr/local/kamailio/lib/kamailio/modules/tls.so...done.Loaded
symbols for /usr/local/kamailio/lib/kamailio/modules/tls.soReading
symbols from /lib/libnss_files.so.2...(no debugging symbols
found)...done.Loaded symbols for /lib/libnss_files.so.2Core was
generated by `./sbin/kamailio ./etc/kamailio/kamailio.cfg'.Program
terminated with signal 11, Segmentation fault.#0 0x080b51a4 in
U_MD5Update (context=0xbfd1a7b4, input=0x74756100,inputLen=1650745192) at md5.c:160160
memcpy(gdb)
On Wed, Jan 4, 2012 at 10:48 AM, Daniel-Constantin Mierla
wrote:
Hello,
can you get the backtrace? Locate the core file (perhaps in / directory if
you don't use -w command line parameter) and do:
gdb /path/to/kamailio /path/to/corefile
bt
Cheers,
Daniel
On 1/4/12 9:17 AM, Ali Jawad wrote:
Hi
When I did set modparam("auth_db", "calculate_ha1", 0 )
Kamailio is crashing see
http://pastebin.com/anaKan0Y
Regards
On Tue, Jan 3, 2012 at 11:54 PM, Ali Jawadwrote:
Hi
It fetches one value from the database to compare it against a second
value that has to be computed right, in the computation of the second
hash where is the domain part fetched from ?
Regards
On Wed, Jan 4, 2012 at 12:26 AM, Daniel-Constantin Mierla
wrote:
Hello,
On 1/3/12 10:08 PM, Ali Jawad wrote:
Hi
In Xlite/eyebeam I put in the username and one of my 3 kamailio
servers respectively as the sip registrar, I.e. register1.domain.com,
register2.domain.com and register3.domain.com, that is why I was
saying that hashing will create different hashes based on the register
domain. With reference to my first related post, one kamailio server
is for softphones, one for sip devices and one for mobile phones with
SIP software. Each server has a slightly different NAT and Call
routing structure.
but your service domain is 'domain.com', specific server addresses
(domains)
per UA type should be set as outbound proxy on the UA device
