sr-users February 2014

sr-users@lists.kamailio.org

114 participants
157 discussions

by Raj Roy Ghandhi

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.

10 years, 2 months

Debian packages not updated since august 2013?

by Johan Wilfer

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

10 years, 2 months

Performance impact with AVP and VAR's

by Jijo

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

10 years, 2 months

Re: [SR-Users] Issue with Async module

by Daniel-Constantin Mierla

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

10 years, 2 months

Re: [SR-Users] NAT traversal problem with Kamailio when trying to communicate between two different LANs

by Neo Quartz

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

10 years, 2 months

understanding the sqlops module and kamailio children

by Uri Shacked

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

10 years, 2 months

Asterisk Ignore "UNREGISTER" because retransmit

by pavel＠eremina.net

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); }

10 years, 2 months

Issue with Async module

by Surendra

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

10 years, 2 months

Problem with special sip notify (trying to make central provisioning for endpoints)

by Pedro Niño

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....

10 years, 2 months

8 digit restriction on dialplan

by phillman25

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

10 years, 2 months

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sr-users February 2014