- sr-users - kamailio.org

Re: [SR-Users] FW: Regd. t_suspend() and t_continue()
by Shankar 21 Feb '14

21 Feb '14

Hi, We have found the root cause for the problem that was reported (refer below mail for details) in async module. Below is the brief description, - async_route("Resume", "1") - At time t, the async records are stored at slot 't+1' of async list. - Every second, async records from the async list are processed. - In function 'async_timer_exec', slot = ticks % ASYNC_RING_SIZE; since slot is based on the ticks and if the previous invocation of 'async_timer_exec' did not finish in time, the next tick is missed. Subsequent call to 'async_timer_exec' will be with t+2. Here slot t+1 will be missed and the same will be processed only during next cycle. That means the async records will be stored for the next 100 seconds before they are actually processed. TM module will drop those timed out transactions and hence the error "t_continue: transaction not found" Now we have modified the code in such a way that slot is incremented sequentially irrespective of the ticks that is sent to the function 'async_timer_exec' function. This way we do not see any call failures and all our load runs are successful. If someone is interested we can share the code as well. Regards, Shankar From: Shankar [mailto:shankar.rk@plintron.com] Sent: Thursday, January 23, 2014 12:18 PM To: 'Jason Penton'; 'SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List' Subject: RE: [SR-Users] FW: Regd. t_suspend() and t_continue() Hi, >From our repeated load tests what we can conclude irrespective of the number of simultaneous calls, there is always this error "t_continue: transaction not found" occurring. If I run say 20 cps, then after running 5000 calls, we observe exactly 20 calls failing with the above error. We doubt that there is something happening during a particular point in time (for a second) which impacts the saving of those new transactions into shared memory. For 10 cps run, we observe exactly 10 call failures. We repeated with different cps and found that error is exactly equal to the cps being run. Any configuration we are missing. Anyone can help? Regards, Shankar From: Shankar [mailto:shankar.rk@plintron.com] Sent: Tuesday, January 21, 2014 3:09 PM To: 'Jason Penton'; 'SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List' Subject: FW: [SR-Users] FW: Regd. t_suspend() and t_continue() Hi Jason, Below is our config, route[LOCATION] { if(is_method("INVITE")) { if(!route(FROMCSCF)) { setflag(FLT_ACC); setflag(FLT_ACCFAILED); dlg_manage(); dlg_setflag("4"); async_route("RESUME", "1"); exit; } } } route[RESUME] { route(TO_LOCATION); // here t_relay to REGISTRAR is done for user lookup. exit; } Regards, Shankar Date: Tue, 21 Jan 2014 11:14:21 +0200 From: Jason Penton <jason.penton(a)smilecoms.com> To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org> Subject: Re: [SR-Users] FW: Regd. t_suspend() and t_continue() Message-ID: <CAE=KcrghqJHgnGDxqS1fYvUzM=HqRAcKWfEAsNJjm8xUDCq68w(a)mail.gmail.com> Content-Type: text/plain; charset="iso-8859-1" We use it heavily, but not using the async module - we use it directly from the IMS code. Can you please provide your config (or a relevant snippet) file so I can see what exactly you are testing/trying to do Cheers jason From: Shankar [mailto:shankar.rk@plintron.com] Sent: Tuesday, January 21, 2014 2:25 PM To: 'SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List' Subject: RE: [SR-Users] FW: Regd. t_suspend() and t_continue() Hi, Anyone who had used t_suspend() and t_continue() can share the performance details? I tried async module with one sec sleep time. I tried only 5 calls per second but still it was not successful. After sometime I see below logs, Jan 21 13:51:55 PLT-RA-RD-W167A PCscf[16520]: ERROR: tm [t_suspend.c:128]: t_continue(): ERROR: t_continue: transaction not found Jan 21 13:52:49 PLT-RA-RD-W167A last message repeated 15 times Jan 21 13:59:38 PLT-RA-RD-W167A last message repeated 12 times Jan 21 14:13:03 PLT-RA-RD-W167A last message repeated 5 times Any configuration changes can help here? Regards, Shankar From: Shankar [mailto:shankar.rk@plintron.com] Sent: Wednesday, January 15, 2014 1:26 PM To: 'Jason Penton' Cc: 'SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List' Subject: RE: [SR-Users] FW: Regd. t_suspend() and t_continue() Hi Jason, I am using 4.0.2 Regards, Shankar From: Jason Penton [mailto:jason.penton@smilecoms.com] Sent: Wednesday, January 15, 2014 1:21 PM To: Shankar Cc: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List Subject: Re: [SR-Users] FW: Regd. t_suspend() and t_continue() Hi Shankar, What version of Kamailio are you running? Kamailio -V Cheers Jason On Wed, Jan 15, 2014 at 6:58 AM, Shankar <shankar.rk(a)plintron.com> wrote: Hi Jason, Please find below my response inline, I have some questions for you as we have used suspend/continue quite a lot in the IMS code and don't have any leaks. Firstly, why are you using pkg_mem for your hash_id and label? Remember that you will be in 2 different processes in the suspend and continue portions of the code... so pkg_mem will not work - you should use shm_mem instead. [Shankar] We use pkg_mem because we are invoking t_continue from the same process ( using thread ). Secondly, how are you using top to tell that you have a leak? Kamailio's memory is internally managed. [Shankar] After running for say 20minutes, we get out of shared memory error. Also in top output we observed incremental increase in the shared usage of shared memory for the process. Cheers Jason On Mon, Jan 13, 2014 at 1:29 PM, Shankar <shankar.rk(a)plintron.com> wrote: > Re-sending without the attachment. > > > > *From:* Shankar [mailto:shankar.rk@plintron.com] > *Sent:* Monday, January 13, 2014 4:57 PM > *To:* 'sr-users(a)lists.sip-router.org' > *Subject:* Regd. t_suspend() and t_continue() > > > > Hi, > > > > We are trying out the t_suspend() and t_continue() in our test setup. > We are facing memory leak ( both shm and pkg as per top command results). > > > > Please find below the scenario, > > > > 1) Do a t_newtran() > > 2) Allocate pkg memory for hashid and label. > > 3) Call t_suspend() > > 4) Do t_continue() when async result is available > > 5) De-allocate pkg memory reserved for hashid and label > > 6) Do a t_relay() which forwards the sip message to another sip node. > > > > In the step (6) above, we see t_newtran() allocates one more time > shared memory for the same transaction. > > > > We tried t_release() after step (4) to release the transaction as > t_relay() anyways allocates new shared memory. Nothing helped. > > > > Please let me know what are the logs you would require to debug the same. > I am attaching syslog for this run. > > > > Regards, > > Shankar -- Jason Penton Senior Manager: Applications and Services Smile Communications Pty (Ltd) Mobile: +27 (0) 83 283 7000 Skype: jason.barry.penton <mailto:name.surname@smilecoms.com> jason.penton(a)smilecoms.com <http://www.smilecoms.com/> www.smilecoms.com <http://196.33.227.129/~smlcoms/sigs/pty/images/smile_signature_07_09.jpg> This email is subject to the disclaimer of Smile Communications at http://www.smilecoms.com/disclaimer

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Call Control for Credit and Rate
by Raj Roy Ghandhi 21 Feb '14

21 Feb '14

Hi, I want to implement the call control by 1. Validating the CLI and get the credit from DB 2. Check the Rate for the dialed number 3. set the call timeout for the call duration. ( credit/rate) 4. pass the call to remote PSTN GW Please guide me on this. Regards, Roy.

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Debian packages not updated since august 2013?
by Johan Wilfer 21 Feb '14

21 Feb '14

Hi, Looking at http://www.kamailio.org/wiki/packages/debs the suggestion is to use http://deb.kamailio.org/ for debian packages. But the last version of kamailio 4.0 here is 4.0.3, and 4.1 doesn't exist. The most files are last modified in 2013-08.. Here for example: http://deb.kamailio.org/kamailio/dists/wheezy/main/binary-amd64/ Is there a new server to get the deb-packages from or should I build from source from now on? /Johan

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Performance impact with AVP and VAR's
by Jijo 21 Feb '14

21 Feb '14

Hi All, I have around 300 AVP's and quite amount of VAR's are used in the config file? Does that impact performance?, If so how can i improve it? Thanks Jijo

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Re: [SR-Users] Issue with Async module
by Daniel-Constantin Mierla 20 Feb '14

20 Feb '14

Hello, thanks for reporting, I pushed a slightly different patch in master, 4.1 and 4.0 branches -- free is done even when ai->act is null. Cheers, Daniel On 20/02/14 15:13, Surendra wrote: > Hi Daniel, > > > I am using async module to continue the transaction in > asynchronously. But I am observing a leak in shared memory. Then i checked > the code, we are not freeing the async_item_t after calling t_continue. > Same this has checked with MEMDBG=1. > > Original code: > while(1) { > lock_get(&_async_list_head->ring[slot].lock); > ai = _async_list_head->ring[slot].lstart; > if(ai!=NULL) > _async_list_head->ring[slot].lstart = ai->next; > lock_release(&_async_list_head->ring[slot].lock); > > if(ai==NULL) > break; > if(ai->act!=NULL) > tmb.t_continue(ai->tindex, ai->tlabel, ai->act); > } > > > Changed code: > > while(1) { > lock_get(&_async_list_head->ring[slot].lock); > ai = _async_list_head->ring[slot].lstart; > if(ai!=NULL) > _async_list_head->ring[slot].lstart = ai->next; > lock_release(&_async_list_head->ring[slot].lock); > > if(ai==NULL) > break; > if(ai->act!=NULL) > { > tmb.t_continue(ai->tindex, ai->tlabel, ai->act); > shm_free(ai); /* Added by surendra */ > } > } > > Please correct this if I am wrong. > > > Regards > @surendra > > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

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Re: [SR-Users] NAT traversal problem with Kamailio when trying to communicate between two different LANs
by Neo Quartz 20 Feb '14

20 Feb '14

Hi All, I installed Kamailio server on my ubuntu machine. And installed rtpproxy also in the same machine. I didm't make many changes to default configuration files, kamctlrc and kamailio.cfg. Now I am able to communicate between two SIP clients, if both sip clients are in the same LAN. But I am not able to communicate between one sip client in one LAN and the other SIP client in other LAN. Sip Call handling and ringing etc. is ok but there is no voice for this case. I don't have much experience in routing for NAT traversal. I tried to experiment with rtpproxy_manage function in kamailio.cfg file. But I was not successful. Please help me by providing any hints or pointers to proceed further. I am using only Kamailio and Rtpproxy. No other software like Asterisk or FreeSwitch. ************* These are defines I used in kamailio.cfg file: #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_ALIASDB #!define WITH_USRLOCDB #!define WITH_ANTIFLOOD #!define WITH_NAT #!define WITH_PRESENCE listen ip address changed ******************* Kamailio is compiled with following modules: include_modules= db_mysql dialplan presence presence_xml ******** Regards, Sateesh

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understanding the sqlops module and kamailio children
by Uri Shacked 20 Feb '14

20 Feb '14

Hi, Following my issue with the above.... 1. Is there a way to see if a worker (child) is processing a transaction for too long (let's say 2-3 sec)? 2. Is there an option to see if the DB connection is alive? (I am familiar with db_mysql options) 3. If all workers are stuck on this "long" transaction, is there a Q for the new INVITES? Can I see how many are there? What I was experiencing is that when my DB was very slow, kamailio crashed after a while.... I will configure db_mysql to ping every 30 sec and timeout after 1 sec. Still, it can create a 3 sec delay for transactions. Any one has an idea? Using the ASYNC module can help here? Does it free the worker to deal with new requests? BR, Uri

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Asterisk Ignore "UNREGISTER" because retransmit
by pavel＠eremina.net 20 Feb '14

20 Feb '14

Hi, All! Try to register user in Kamailio and then relay REG massage to Asterisk. Asterisk register user and looks good. But when I send Unreg message to Kamailio, it unregister user successfully but Asterisk say: Message REGISTER Ignoring because retransmit (CSeq 1234 our CSeq 1234). Did someone know how solve this problem: i think that Asterisk need different CSeq for first REGISTER and Second after 401 code. dump my conf: At route[REGISTER]: if (!save("location")) { sl_reply_error(); } else { reg_fetch_contacts("location", "$tu", "cc"); $var(useragent) = $ulc(cc=>user_agent); xlog("L_INFO","EXpires in Contact: $ulc(cc=>expires)"); if ($ulc(cc=>expires)==0) { $var(query) = "insert into activity values (now(),'unr','"+$ci+"','"+$fu+"');"; sql_query("events", "$var(query)"); $var(string_temp) = "$(hdr(expires))"; xlog("L_INFO", "UNregistered $ru FROM $fu exp $var(string_temp)"); } else { if ($var(registered)==0) { $var(query) = "insert into activity values (now(),'reg','"+$ci+"','"+$fu+"');"; sql_query("events", "$var(query)"); } xlog("L_INFO", "Registered $ru FROM $fu exp $ulc(cc=>expires)"); } } xlog("L_INFO","DU(TD): $td"); if ($td == "mydomain.com") {route(TOASTERISK); } exit; } } route[TOASTERISK] { xlog("L_INFO","ROUTE to ASTERISK from $ru to $tu"); #append_branch(); t_on_failure("FAILAUTH"); $ru = "sip:"+ $td; consume_credentials(); route(RELAY); exit; } failure_route[FAILAUTH] { if (t_is_canceled()) { exit; } if (t_check_status("(401)|(407)")) { xlog("L_INFO", "Remote asked for authentication"); $avp(realm)=""; uac_auth(); } route(RELAY); }

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Issue with Async module
by Surendra 20 Feb '14

20 Feb '14

Hi Daniel, I am using async module to continue the transaction in asynchronously. But I am observing a leak in shared memory. Then i checked the code, we are not freeing the async_item_t after calling t_continue. Same this has checked with MEMDBG=1. Original code: while(1) { lock_get(&_async_list_head->ring[slot].lock); ai = _async_list_head->ring[slot].lstart; if(ai!=NULL) _async_list_head->ring[slot].lstart = ai->next; lock_release(&_async_list_head->ring[slot].lock); if(ai==NULL) break; if(ai->act!=NULL) tmb.t_continue(ai->tindex, ai->tlabel, ai->act); } Changed code: while(1) { lock_get(&_async_list_head->ring[slot].lock); ai = _async_list_head->ring[slot].lstart; if(ai!=NULL) _async_list_head->ring[slot].lstart = ai->next; lock_release(&_async_list_head->ring[slot].lock); if(ai==NULL) break; if(ai->act!=NULL) { tmb.t_continue(ai->tindex, ai->tlabel, ai->act); shm_free(ai); /* Added by surendra */ } } Please correct this if I am wrong. Regards @surendra

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Problem with special sip notify (trying to make central provisioning for endpoints)
by Pedro Niño 20 Feb '14

20 Feb '14

Hi, how are you all I have a lab setup for using Kamailio and asterisk, with realtime, and currently I am facing 2 problems. The first one, is when trying to get the phones to reboot. Asterisk send a notify with the message, reboot. But the phone answers with a 401, and asks for a digest. In a normal asterisk environment it would not be a problem, asterisk tries to answer with the user and password cipher with the digest. But because of using this design, asterisk sends the notify with the user and blank password (I am using realtime, so peers can register from kamailio with no password).of course this makes that the phone refuses, as the notify doesn't have the right password. Is there any way to make kamailio intercept the notify sent from the phone and answer with the right credentials? _------------------- Second: When having the users registered in the same scenario, and get an asterisk 'reload', and the users try to make a call before they re-register, only the first call make it thru Of course, as kamailio is in the middle, the users doesnt notice until they try to make a call. In asterisk, I keep seeing the kamailio trying to register and getting a 404. This keeps happening until I sent a sip reload at the asterisk side and wait for the users to register before making calls. Thanks for your time....

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8 digit restriction on dialplan
by phillman25 20 Feb '14

20 Feb '14

Dear List I am trying to restrict 8 digit dial on my dialplan after the prefix 6859A , could someone help me with the correct syntax? Please see below what i have: Dpid: 1 Pr: 1 Match Op: 1 Match Exp: sip:6859A(only 8 additional digits allowed).* Match Len: 0 Subst Exp: sip:6859A(only 8 additional digits allowed).* Repl Exp: 3005A1883(a)xx.xx.xx.xx So basically any 8 digits after the prefix 6859A is allowed otherwise there shouldn't be a match, how could i go about doing this? Thanks in advance Phillip

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Error in carrierroute in kamailio 4.1
by Eduardo Alonso 20 Feb '14

20 Feb '14

Hi all: It's my first mail to the community and I know that will be a great community to share knowledge. OS Debian 6.4 Kamailio 4.1 I have the same problem that Oliver Roth describes in http://lists.sip-router.org/pipermail/sr-users/2013-October/080096.html, but in Kamailio 4.1: Module carrierroute not working starting from version kamailio 4.1 trying to activate carrierroute in kamailio.cfg loadmodule "carrierroute.so" i get the following error when starting kamailio (5794) ERROR: <core> [sr_module.c:599]: load_module(): ERROR: load_module: could not open module </usr/local/lib64/kamailio/modules/carrierroute.so>: /usr/local/lib64/kamailio/modules/carrierroute.so: undefined symbol: cfg_set_error_function 0(5794) : <core> [cfg.y:3408]: yyerror_at(): parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 245, column 12-28: failed to load module ERROR: bad config file (1 errors) Any idea? Maybe him o someone has the solution. Thanks all for your great work. Best regards. -- *Eduardo Alonso Gil* VoIP Systems Engineer @ Quobis <http://www.quobis.com/> | e: eduardo.alonso(a)quobis.com | t: +34902999465 Visit our booth at *MWC Barcelona*. CS60 Spanish Pavilion. More info<http://www.quobis.com/index.php?option=com_content&task=view&id=236&Itemid=…>

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Issues with keepalive_timeout
by Kristian Kielhofner 20 Feb '14

20 Feb '14

Hello All, I'm trying to use keepalive_timeout with 4.1.1 as described here: http://lists.sip-router.org/pipermail/sr-users/2012-August/074337.html I believe I've followed these configuration steps and those defined in the nathelper module. Here are the relevant portions of my config: modparam("nathelper", "natping_interval", 60) modparam("nathelper", "keepalive_timeout", 180) modparam("nathelper", "ping_nated_only", 0) modparam("nathelper", "sipping_from", "sip:pinger@domain.com") modparam("nathelper", "sipping_method", "OPTIONS") modparam("nathelper", "sipping_bflag", 9) And of course a setting the correct branch flag (verified) during REGISTER handling: setbflag(9); My goal is to use SIP OPTIONS messages to verify the reachability of all of my endpoints (not just NATd ones). I would just lower my re-registration timeout but there are other issues associated with that kind of change. With my current configuration (after a successful registration) a SIP OPTIONS message is sent and replied to: U 4.2.2.1:5060 -> 4.2.2.2:5060 OPTIONS sip:user372@4.2.2.2:5060;ob SIP/2.0. Via: SIP/2.0/UDP 4.2.2.1:5060;branch=0. From: sip:pinger@domain.com;tag=uloc-5303810e-7c78-1-2273f0eb-9627292. To: sip:user372@4.2.2.2:5060;ob. Call-ID: 8af79586-93072f43-7c4cf55(a)4.2.2.1. CSeq: 1 OPTIONS. Content-Length: 0. . U 4.2.2.2:5060 -> 4.2.2.1:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 4.2.2.1:5060;received=4.2.2.1;branch=0. Call-ID: 8af79586-93072f43-7c4cf55(a)4.2.2.1. From: <sip:pinger@domain.com>;tag=uloc-5303810e-7c78-1-2273f0eb-9627292. To: <sip:user372@4.2.2.2:5060>;tag=0;ob. CSeq: 1 OPTIONS. Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS. Accept: application/sdp, application/pidf+xml, application/xpidf+xml, application/simple-message-summary, message/sipfrag;version=2.0, application/im-iscomposing+xml, text/plain. Supported: replaces, 100rel, timer, norefersub. Allow-Events: presence, message-summary, refer. User-Agent: My_UA. Content-Length: 0. . If I kill the registered UA (making it unavailable) Kamailio endlessly sends OPTIONS messages, never receives a reply, and never removes the registration from usrloc: U 4.2.2.1:5060 -> 4.2.2.2:5060 OPTIONS sip:user372@4.2.2.2:5060;ob SIP/2.0. Via: SIP/2.0/UDP 4.2.2.1:5060;branch=0. From: sip:pinger@domain.com;tag=uloc-5303810e-7c78-1-2273f0eb-b627292. To: sip:user372@4.2.2.2:5060;ob. Call-ID: 8af79586-b3072f43-f35cf55(a)4.2.2.1. CSeq: 1 OPTIONS. Content-Length: 0. . U 4.2.2.1:5060 -> 4.2.2.2:5060 OPTIONS sip:user372@4.2.2.2:5060;ob SIP/2.0. Via: SIP/2.0/UDP 4.2.2.1:5060;branch=0. From: sip:pinger@domain.com;tag=uloc-5303810e-7c78-1-2273f0eb-c627292. To: sip:user372@4.2.2.2:5060;ob. Call-ID: 8af79586-c3072f43-b75cf55(a)4.2.2.1. CSeq: 1 OPTIONS. Content-Length: 0. . U 4.2.2.1:5060 -> 4.2.2.2:5060 OPTIONS sip:user372@4.2.2.2:5060;ob SIP/2.0. Via: SIP/2.0/UDP 4.2.2.1:5060;branch=0. From: sip:pinger@domain.com;tag=uloc-5303810e-7c78-1-2273f0eb-d627292. To: sip:user372@4.2.2.2:5060;ob. Call-ID: 8af79586-d3072f43-7b5cf55(a)4.2.2.1. CSeq: 1 OPTIONS. Content-Length: 0. . U 4.2.2.1:5060 -> 4.2.2.2:5060 OPTIONS sip:user372@4.2.2.2:5060;ob SIP/2.0. Via: SIP/2.0/UDP 4.2.2.1:5060;branch=0. From: sip:pinger@domain.com;tag=uloc-5303810e-7c78-1-2273f0eb-e627292. To: sip:user372@4.2.2.2:5060;ob. Call-ID: 8af79586-e3072f43-3f5cf55(a)4.2.2.1. CSeq: 1 OPTIONS. Content-Length: 0. . (and on, and on, and...) Eventually the registration timeout kicks in, the usrloc entry is removed, and Kamailio stops sending OPTIONS to the (long) unreachable endpoint. Is this expected behavior? What am I doing wrong? Thanks! -- Kristian Kielhofner

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Re: [SR-Users] Regd. MAX_LDG_LOCKS
by Daniel-Constantin Mierla 20 Feb '14

20 Feb '14

Hello, setting the hash size to 32MB is way too much, are you having like 100 000 000 active calls at the same time? The size of the hash is used for computing the number of slots in the hash table, not the capacity in terms of memory size of all items stored in hash table. Cheers, Daniel On 03/02/14 07:33, Shankar wrote: > > Hi Alex, > > We set the hash size to 32 MB and found that the hash size is not the > same as configured one. At the max it was taking 64KB. > > Regards, > > Shankar > > Date: Mon, 03 Feb 2014 00:27:29 -0500 > > From: Alex Balashov <abalashov(a)evaristesys.com > <mailto:abalashov@evaristesys.com>> > > To: sr-users(a)lists.sip-router.org <mailto:sr-users@lists.sip-router.org> > > Subject: Re: [SR-Users] Regd. MAX_LDG_LOCKS > > Message-ID: <52EF28C1.5010204(a)evaristesys.com > <mailto:52EF28C1.5010204@evaristesys.com>> > > Content-Type: text/plain; charset=windows-1252; format=flowed > > On 02/03/2014 12:22 AM, Shankar wrote: > > > Also we found that in ?dialog.c? the variable dlg_hash_size should be > > > ?unsigned int?. Hope this will be taken care in next release of > Kamailio. > > How did you arrive at that conclusion? > > -- > > Alex Balashov - Principal > > Evariste Systems LLC > > 235 E Ponce de Leon Ave > > Suite 106 > > Decatur, GA 30030 > > United States > > Tel: +1-678-954-0670 > > Web: http://www.evaristesys.com/, http://www.alexbalashov.com/ > > *From:*Shankar [mailto:shankar.rk@plintron.com] > *Sent:* Monday, February 03, 2014 10:53 AM > *To:* 'SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List' > *Subject:* Regd. MAX_LDG_LOCKS > > Hello, > > Can anyone brief about MAX_LDG_LOCKS in dlg_hash.c? > > Should I increase the value of MAX_LDG_LOCKS (hardcoded to 2048) if > the hash size is increased? > > Also we found that in ‘dialog.c’ the variable dlg_hash_size should be > ‘unsigned int’. Hope this will be taken care in next release of Kamailio. > > Regards, > > Shankar > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

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Re: [SR-Users] Regd. MAX_LDG_LOCKS
by Daniel-Constantin Mierla 20 Feb '14

20 Feb '14

Hello, it is not a really necessary to increase the value of the locks. For better performances you can do it, but again, it is not a must. Cheers, Daniel On 04/02/14 06:50, Shankar wrote: > > Hi, > > Can someone clarify here please? > > Regards, > > Shankar > > *From:*Shankar [mailto:shankar.rk@plintron.com] > *Sent:* Monday, February 03, 2014 10:53 AM > *To:* 'SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List' > *Subject:* Regd. MAX_LDG_LOCKS > > Hello, > > Can anyone brief about MAX_LDG_LOCKS in dlg_hash.c? > > Should I increase the value of MAX_LDG_LOCKS (hardcoded to 2048) if > the hash size is increased? > > Regards, > > Shankar > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

Kamailio 4.1 Installation from Source
by Francesco Maria Magnini 19 Feb '14

19 Feb '14

Hi, following the documentation, the following lines need to be updated on the init.d script. DAEMON=/usr/local/sbin/kamailio CFGFILE=/usr/local/etc/kamailio/kamailio.cfg Anyway, there is no CFGFILE already setted. Is it correct to add CFGFILE along with DAEMON? On top of this, in the check_fork () procedure, there’s a reference to a cfg_file that is nonexistent. In my installation i changed it to the real path of the kamailio.cfg file generated during installation. check_fork () { if grep -q "^[[:space:]]*fork[[:space:]]*=[[:space:]]*no.*" /usr/local/etc/kamailio/kamailio.cfg; then log_failure_msg "Not starting $DESC: fork=no specified in config file; run /etc/init.d/kamailio debug instead" exit 0 fi } Is it correct to change path in the check_fork procedure with the correct one? Regards,

2 2

Kamailio World - Presentations
by Daniel-Constantin Mierla 19 Feb '14

19 Feb '14

Only very few presentations accepted at Kamailio World Conference & Exhibition are not yet published. In a matters of days we should have the draft of the schedule published. It is again a good variety of topics around Kamailio and other open source VoIP applications, presenting experiences from the field, showing new product and solutions or open debate and demos panels. You can lean about various ways of deploying scalable platforms with Open Stack, Erlang or asynchronous, carrier grade charging systems, managing security in VoIP deployments or how to do real time troubleshooting. More details about presentations are available at: - http://conference.kamailio.com/k02/schedule/ Information about speakers can be found at: - http://conference.kamailio.com/k02/speakers/ In order to allow a fair enough time to see what the content of the event is going to be, the early registration deadline has now been set to this Friday, February 21. Don't miss the chance to learn from about 30 sessions and meet more than one hundred professionals in real time communications and open source technologies: - http://conference.kamailio.com/k02/registration/ Looking forward to meeting you in Berlin, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

uac_reg_request_to
by Isaac McDonald 19 Feb '14

19 Feb '14

Hello All, I need some clarification on the function uac_reg_request_to. The doc says "This function can be used to send an authenticated request to a remote user in the uac registrations table" Does this mean I can forward a SUBSCRIBE request, or any request for that matter, to an upstream server that's Kamailio is registered to via the UAC module? Secondly, I'm having trouble matching the user on this function. The error is: "ac [uac_reg.c:996]: uac_reg_request_to(): no user: 2503" User 2503 does exist in the uac_reg table in column "l_username". What if I want to match by UUID instead? How does that get mapped back to a local user? Thanks -- Isaac McDonald SusCall, LLC E-Mail: imcdona(a)suscall.com

1 0

Memory leak in http_query() function (utils module)
by Guillaume Bour 18 Feb '14

18 Feb '14

Hi folks, We identified a memory leak in *utils* module, when calling the http_query() function. a memory piece is allocated with malloc() to store http request result string (functions.c:54), but this memory is never freed. There is also a possible crash issue, because curl_easy_getinfo() is called *after* curl_easy_cleanup(), which must not be done (see http://curl.haxx.se/libcurl/c/curl_easy_cleanup.html) I attached patches to fix those issues (applies to 4.0 branch) Regards, Guillaume Bour

2 5

error using siremis
by Rubi Mazaki 18 Feb '14

18 Feb '14

Hi I successfully install siremis 4.0.0 +kamailio 4.1 on my ubuntu (32 bit) server. Kamailio was installed successfully and I can now run "kamctl moni" and see a nice status screen. I had to setup again siremis (by deleting the install.lock and opening the localhost/siremis setup again) but one that was achieved I can now login in to the siremis site without errors. Now to the problem: Once I enter "groups" and press "add" I get the following error displayed: [2014-02-12 20:46:33 (GMT)] An exception occurred while executing this script: Error message: #8192, preg_replace(): The /e modifier is deprecated, use preg_replace_callback instead Script name and line number of error: /home/rubi/siremis-4.0.0/openbiz/others/Smarty/libs/Smarty_Compiler.class.php:270 *function:* errorHandler ( 8192, "preg_replace(): The /e modifier is deprecated, use preg_replace_...", "/home/rubi/siremis-4.0.0/openbiz/others/Smarty/libs/Smarty_Compi...", 270, Array(8) ) @ /home/rubi/siremis-4.0.0/openbiz/bin/sysheader.inc 117 *function:* userErrorHandler ( 8192, "preg_replace(): The /e modifier is deprecated, use preg_replace_...", "/home/rubi/siremis-4.0.0/openbiz/others/Smarty/libs/Smarty_Compi...", 270, Array(8) ) @ *function:* preg_replace ( "~\{\*(.*?)\*\}|\{\s*literal\s*\}(.*?)\{\s*/literal\s*\}|\{\s*php...", "'{php' . str_repeat(" ", substr_count('\0', " ")) .'}'", "<form id="{$ form.name}" name="{$form.name}"> <div style="paddin..." ) @ /home/rubi/siremis-4.0.0/openbiz/others/Smarty/libs/Smarty_Compiler.class.php 270 *function:* _compile_file ( "/home/rubi/siremis-4.0.0/siremis/modules/system/template/detail....", "<form id="{$form.name}" name="{$form.name}"> <div style="paddin...", Null ) @ /home/rubi/siremis-4.0.0/openbiz/others/Smarty/libs/Smarty.class.php 1484 *function:* _compile_source ( "/home/rubi/siremis-4.0.0/siremis/modules/system/template/detail....", "<form id="{$form.name}" name="{$form.name}"> <div style="paddin...", Null, "/home/rubi/siremis-4.0.0/siremis/themes/default/template/cpl/%%2..." ) @ /home/rubi/siremis-4.0.0/openbiz/others/Smarty/libs/Smarty.class.php 1417 *function:* _compile_resource ( "/home/rubi/siremis-4.0.0/siremis/modules/system/template/detail....", "/home/rubi/siremis-4.0.0/siremis/themes/default/template/cpl/%%2..." ) @ /home/rubi/siremis-4.0.0/openbiz/others/Smarty/libs/Smarty.class.php 1256 *function:* fetch ( "/home/rubi/siremis-4.0.0/siremis/modules/system/template/detail...." ) @ /home/rubi/siremis-4.0.0/openbiz/bin/easy/FormRenderer.php 95 *function:* renderSmarty ( Object(EasyForm), "/home/rubi/siremis-4.0.0/siremis/modules/system/template/detail...." ) @ /home/rubi/siremis-4.0.0/openbiz/bin/easy/FormRenderer.php 45 *function:* render ( Object(EasyForm) ) @ /home/rubi/siremis-4.0.0/openbiz/bin/easy/EasyForm.php 1648 *function:* renderHTML ( ) @ /home/rubi/siremis-4.0.0/openbiz/bin/easy/EasyForm.php 1538 Can anyone tell me what I did wrong, or how can I re-install completely siremis? Hopefully that's not a bug :/ With Regards, Rubi Mazaki Email: rubimazaki(a)gmail.com

2 1

One more question 'bout db_mode
by Victor V. Kustov 18 Feb '14

18 Feb '14

Hi all! my usrloc database use postgress. In config I had: modparam("usrloc", "db_mode", 1) after day of work I see strange thing: # kamctl ul show --brief | grep -c AOR 26 psql -U kamailio -c "select * from location;" ... (1 row) 26 registered users in inner kamailio's memory and only 1 in database. See in doc: 1 - Write-Through scheme. All changes to usrloc are immediately reflected in database too. This is very slow, but very reliable. Use this scheme if speed is not your priority but need to make sure that no registered contacts will be lost during crash or reboot. I think saved locations in inner memory database must be identical. Am I wrong? -- WBR, Victor JID: coyote(a)bks.tv JID: coyote(a)bryansktel.ru I use FREE operation system: 3.12.9-calculate GNU/Linux

1 0

Change table type...
by arun Jayaprakash 18 Feb '14

18 Feb '14

Hello, I am planning to change the table type in the Kamailio Mysql db from MyISAM to Innodb. Could anyone let me know if this will be an issue? Thank you. Arun

2 2

modify To header, prefix + ip, using dispacher module.
by Helena Garcia-Nieto 17 Feb '14

17 Feb '14

Hey, Thank you in advanced for the time taken on my question! I have an environment phone---sbc---kamailio---gw1, gw2 I am using kamailio 4.2.0, quite basic config, using dispatcher module to route to the gw and to monitor them with options. I have a file dispatcher.list with the two gw ips: # $Id$ # dispatcher destination sets # # line format # setit(int) destination(sip uri) flags(int,opt) priority(int,opt) attributes(str,opt) # gateways 1 sip:10.X.X.38:5060 IP 1 sip:10.X.X.39:5060 AP I have to modify the TO header, it must be prefixed and gw ip must be placed there (gw requirements) I use on the routing function ds_select_domain("1", "8") t_on_failure("RTF_DISPATCH"); and just before the ROUTE(RELAY) I have: # Add profix to gw: 0199 subst_uri('/^sip:(.*)/sip:0199\1/i'); # add prefix to RURI subst_hf("To", "/(.*):(.*)@(.*)$/\1:0199\2@$rd>/", "a"); #add prefix to TO, modify ip on TO use ruri ip It works fine if the first gw is available, but if the first option (which is the last gw on the list) is not accessible, it waits for the 30s and then tries the second one, but the ip on the TO header is not correct anymore, since it is the first gw and not the second. Question: why does it wait until the timeout if it is monitoring it with option and seeing it is not reachable? I tried modifying again the TO header with: subst_hf("To", "/(.*)@(.*)$/\1@$rd>/", "a") before the route(relay) on the failure routine, but it does not work properly, I am getting strange TO header like: To: <sip:0199XXXX@10.X.X.39> <sip:XXXXXX@10.X.X.38> It looks like it is not parsing the to header it has before but adding a parsed version of the original header at the end of the other one.... I tried with subst as well but the result is more or less the same. Probably there is a better way to set the prefix and the correct IP on the to header... I'll appreciate any help with this issue. Thanks again for your time Helena Helena Garcia-Nieto de Mir Morodo Group VoIP Department E-mail: helena.gnieto(a)morodo.co.uk Morodo Limited voip(a)morodo.co.uk / www.mo-call.com Place of Registration: Companies House, England & Wales / Registered Office: 7 Breasy Place, 9 Burroughs Gardens Hendon, London NW4 4AU

4 12

deb.kamailio.org down?
by Paul Belanger 17 Feb '14

17 Feb '14

Anybody know why deb.kamailio.org is down? Is there a mirror setup some place else? -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belanger(a)polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger

3 2

Re: [SR-Users] Msilo message body type
by sipatse＠in.gr 16 Feb '14

16 Feb '14

2 1

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