- sr-users - kamailio.org

Bug: Dialog State=*Teminated* even when the call is active
by Gnaneshwar Gatla 17 Dec '11

17 Dec '11

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. 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-… Regards Gnaneshwar

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Sending SIP request from mi_fifo using TMX module
by Sammy Govind 16 Dec '11

16 Dec '11

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 . . "From: 101(a)mydomain.org To: 103(a)mydomain.org Contact: 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 . . From: 101@ mydomain.org To: 103@ mydomain.org Contact: mydomain.org Content-Type: text/plain; charset=UTF-8 INVITE* *400 Bad headers* *FIFO command was:* *:t_uac_dlg:openser_receiver_25561* *INVITE* *sip:103@mydomain.org* *.* *.* *From: 101@ mydomain.org To: 103@ mydomain.org Contact: 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: <core> [parser/msg_parser.c:275]: ERROR: get_hdr_field: bad body for <From: 101(a)mydomain.org To: 103@ mydomain.org Contact: mydomain.org Content-Type: text/plain; charset=UTF-8INVITE#012>(4) Dec 16 00:57:46 Kamailio3 /usr/local/sbin/kamailio[24541]: INFO: <core> [parser/msg_parser.c:353]: ERROR: bad header field [From: 101(a)mydomain.o] Can anyone help.. Regards, Sammy

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source port used for forwarded requests
by Ben WIlliams 16 Dec '11

16 Dec '11

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. Thanks Ben

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How to get dispatcher list with sercmd command
by Mino Haluz 16 Dec '11

16 Dec '11

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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SERi - SIP Express Router setup file corrupted
by Tirtha 16 Dec '11

16 Dec '11

Dear Sir, To configure a SIP server in windows platform, I have downloaded the setup file of SIP Express Router from your FTP server as mentioned in your web portal, but unfortunately when I am going to extract that file , a error is coming and showing that the specific file is corrupted. So requesting you please help me in this regard with the proper installation guide line. Thank you. Regards, http://www.matrixventures.com/images/green.jpg Tirtha Ghosh Asst. Manager IT M +91 8017372776 T +91 33 40624172 E <mailto:tirtha.ghosh@matrixinfrastructure.com> eMail Me Matrix Ventures Matrix logo <http://in.linkedin.com/in/tirthaghoshkolkata> LinkedIn | <http://www.facebook.com/home.php#!/profile.php?id=1385652529> Facebook | http://www.matrixventures.com/images/blue.jpg http://www.matrixventures.com/images/blue.jpg Matrix Ventures Pvt Ltd . Saltlake, Sector 1 . Kolkata 700064 . India

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How to enable rtp repacketization
by Arif Tuhin 15 Dec '11

15 Dec '11

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. With Best Regards Ariful Hossain Tuhin email: 1. etothepowerpi(a)gmail.com 2. etothepowerpi(a)hotmail.com 3.etothepowerpi(a)yahoo.com skype: freeburn1986

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Kamailio - Asterisk Integration
by Anthony Sanchez 15 Dec '11

15 Dec '11

Is there an update of default Kamailio 3.2.1 configuration file to integrate with Asterisk? Thanks in advance, Tony

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Kamailio - Asterisk Integration
by Anthony Sanchez 14 Dec '11

14 Dec '11

Is there an update of default Kamailio 3.2.1 configuration file to integrate with Asterisk? Thanks in advance, Tony

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Siremis v3.2.0 Released
by Elena-Ramona Modroiu 14 Dec '11

14 Dec '11

Hi, Siremis v3.2.0 is out - the web management interface for Kamailio and SER v3.2.x series. Besides updates to existing views to match the database schema in Kamailio v3.2.x (such as lcr, dialog or siptrace), there are many new features. Among most important: - SQL-based CDR rating engine for billing purposes, with views to manage the rating rules - charts and table summaries to reflect the activity of accounting records - new views to manage records for remote registrations, mtree and global black list - command line tools to add new database table views in a step-by-step wizard fashion - selection of usernames and domains for many views can be done by a select box or picker form with values from subscriber or domain tables More details can be find in the announcement of the release on the web site: * http://asipto.com/u/4t or in the ChangeLog inside the tarball. Siremis can be downloaded from: * http://asipto.com/u/sd Alternative download site, along with the GIT repository for Siremis, is offered at sourceforge.net project: * http://sourceforge.net/projects/siremis/ Regards, Ramona

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Double Invite / Issue
by Brandon Armstead 14 Dec '11

14 Dec '11

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! Sincerely, Brandon Armstead

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syncing db_text acc table
by Pedro Antonio Vico Solano 14 Dec '11

14 Dec '11

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Re: [SR-Users] carrierroute reload option
by Uri Shacked 14 Dec '11

14 Dec '11

> > OK, > > If for example I loaded at startup 30K raws. > Then, after 5 min 20 raws needs to change... > I truncate the carrierroute table, populate it again with 20 raws onlye > and reload.... > > what about the others? they will be gone right? > > >

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carrierroute reload option
by Uri Shacked 14 Dec '11

14 Dec '11

Hi, Is there a way to reload only specified raws from the carrierroute table and not reload the full table? BR, Uri

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MSRP Relay with Kamailio
by Krishna Kurapati 14 Dec '11

14 Dec '11

Dear List members, I would like to use MSRP as part of SIP communication for session based IM, and file transfer. Is there an MSRP relay software that works well with Kamailio? Appreciate any help. Thanks Krish Kura

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problem forking to dbaliase
by Yufei Tao 13 Dec '11

13 Dec '11

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. clientone(a)domain.com, for which failed to find a location entry and uri remaied clientone(a)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? 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 -- Yufei Tao Red Embedded This E-mail and any attachments hereto are strictly confidential and intended solely for the addressee. If you are not the intended addressee please notify the sender by return and delete the message. You must not disclose, forward or copy this E-mail or attachments to any third party without the prior consent of the sender. Red Embedded Design, Company Number 06688253 Registered in England: The Waterfront, Salts Mill Rd, Saltaire, BD17 7EZ

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Fwd: commands out of sync you can't run this command now
by MingHon 13 Dec '11

13 Dec '11

ya debug is set to 3 already.. also #!define WITH_DEBUG dont hav any -d command.. On Tue, Dec 13, 2011 at 9:06 PM, Daniel-Constantin Mierla <miconda(a)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 < > miconda(a)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 < >>> miconda(a)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/0 S+ 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: <core> >>>> [db_query.c:101]: error while submitting query >>>> Dec 13 15:43:41 hostname /usr/local/sbin/kamailio[3589]: ERROR: <core> >>>> [db.c:366]: error in db_query >>>> Dec 13 15:43:41 hostname /usr/local/sbin/kamailio[3589]: ERROR: <core> >>>> [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]: : <core> >>>> [mem/q_malloc.c:146]: BUG: qm_*: fragm. 0x838b11c (address 0x838b134) end >>>> overwritten(0, 0)! >>>> Dec 13 15:43:41 hostname kamailio: ERROR: <core> [daemonize.c:307]: >>>> Main process exited before writing to pipe >>>> >>>> >>>> >>>> On Tue, Dec 13, 2011 at 12:53 AM, Daniel-Constantin Mierla < >>>> miconda(a)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 <gminghon(a)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 listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>>>> >>>>> >>>>> -- >>>>> Daniel-Constantin Mierla -- http://www.asipto.comhttp://linkedin.com/in/miconda -- http://twitter.com/miconda >>>>> >>>>> >>>> >>>> >>>> -- >>>> Regards, >>>> >>>> MingHon >>>> >>>> >>>> _______________________________________________ >>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>>> >>>> >>>> -- >>>> Daniel-Constantin Mierla -- http://www.asipto.comhttp://linkedin.com/in/miconda -- http://twitter.com/miconda >>>> >>>> >>>> >>>> _______________________________________________ >>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>>> >>>> >>>> -- >>>> Daniel-Constantin Mierla -- http://www.asipto.comhttp://linkedin.com/in/miconda -- http://twitter.com/miconda >>>> >>>> >>> >>> >>> -- >>> Regards, >>> >>> MingHon >>> >> >> >> >> -- >> Regards, >> >> MingHon >> >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> >> >> -- >> Daniel-Constantin Mierla -- http://www.asipto.comhttp://linkedin.com/in/miconda -- http://twitter.com/miconda >> >> > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > -- > Daniel-Constantin Mierla -- http://www.asipto.comhttp://linkedin.com/in/miconda -- http://twitter.com/miconda > > -- Regards, MingHon -- Regards, MingHon

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Re: [SR-Users] set host as inactive and probing in dispatcher
by Daniel-Constantin Mierla 13 Dec '11

13 Dec '11

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 > <miconda(a)gmail.com <mailto:miconda@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 >> <miconda(a)gmail.com <mailto:miconda@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 >>> <miconda(a)gmail.com <mailto:miconda@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: <script>: OSPRPL is 408 now disable >>>> destination >>>> 13(11897) ERROR: *** cfgtrace: >>>> c=[/usr/local/etc/kamailio/kamailio.cfg] l=1266 a=26 >>>> n=ds_mark_dst >>>> 13(11897) DEBUG: dispatcher [dispatch.c:1994]: mode [1] >>>> grp [47] dst [sip:10.10.21.28] >>>> 13(11897) ERROR: *** cfgtrace: >>>> c=[/usr/local/etc/kamailio/kamailio.cfg] l=1267 a=26 n=xlog >>>> 13(11897) ERROR: <script>: after disabling destination >>>> >>>> On Mon, Dec 12, 2011 at 11:14 AM, Daniel-Constantin >>>> Mierla <miconda(a)gmail.com <mailto:miconda@gmail.com>> >>>> wrote: >>>> >>>> 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: <script>: SCRIPT : in OSPRPLERR >>>>> sip:36264529@10.10.21.29 >>>>> <mailto:sip%3A36264529@10.10.21.29> >>>>> 13(5760) ERROR: *** cfgtrace: >>>>> c=[/usr/local/etc/kamailio/kamailio.cfg] l=1264 >>>>> a=17 n=if >>>>> 13(5760) ERROR: *** cfgtrace: >>>>> c=[/usr/local/etc/kamailio/kamailio.cfg] l=1259 >>>>> a=26 n=xlog >>>>> 13(5760) ERROR: <script>: OSPRPL is 408 *now >>>>> disable destination >>>>> *13(5760) ERROR: *** cfgtrace: >>>>> c=[/usr/local/etc/kamailio/kamailio.cfg] l=1260 >>>>> a=26 n=ds_mark_dst >>>>> 13(5760) ERROR: *** cfgtrace: >>>>> c=[/usr/local/etc/kamailio/kamailio.cfg] l=1261 >>>>> a=26 n=xlog >>>>> 13(5760) ERROR: <script>: *after disabling >>>>> destination* >>>>> >>>>> now i check the dispatcher list status: >>>>> kamctl fifo ds_list >>>>> >>>>> SET:: 47 >>>>> URI:: sip:10.10.21.20 flags=AX priority=25 >>>>> attrs=OSP_B4 >>>>> URI:: sip:10.10.21.22 flags=AX priority=25 >>>>> attrs=OSP_B3 >>>>> URI:: sip:10.10.21.23 flags=AX priority=25 >>>>> attrs=OSP_B2 >>>>> URI:: sip:10.10.21.21 flags=AX priority=25 >>>>> attrs=OSP_B1 >>>>> SET:: 41 >>>>> URI:: sip:10.10.21.29 flags=AX priority=25 >>>>> attrs=OSP_A4 >>>>> URI:: sip:10.10.21.27 flags=AX priority=25 >>>>> attrs=OSP_A3 >>>>> URI:: sip:10.10.21.25 flags=AX priority=25 >>>>> attrs=OSP_A1 >>>>> URI:: sip:10.10.21.24 flags=AX priority=25 >>>>> attrs=OSP_A2 >>>>> >>>>> >>>>> _______________________________________________ >>>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >>>>> sr-users(a)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(a)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 >>> >>> >> >> -- >> 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(a)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 > > -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

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Re: [SR-Users] commands out of sync you can't run this command now
by MingHon 13 Dec '11

13 Dec '11

> 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 < > miconda(a)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/0 S+ 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: <core> >> [db_query.c:101]: error while submitting query >> Dec 13 15:43:41 hostname /usr/local/sbin/kamailio[3589]: ERROR: <core> >> [db.c:366]: error in db_query >> Dec 13 15:43:41 hostname /usr/local/sbin/kamailio[3589]: ERROR: <core> >> [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]: : <core> >> [mem/q_malloc.c:146]: BUG: qm_*: fragm. 0x838b11c (address 0x838b134) end >> overwritten(0, 0)! >> Dec 13 15:43:41 hostname kamailio: ERROR: <core> [daemonize.c:307]: Main >> process exited before writing to pipe >> >> >> >> On Tue, Dec 13, 2011 at 12:53 AM, Daniel-Constantin Mierla < >> miconda(a)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 <gminghon(a)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 listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>> >>> >>> -- >>> Daniel-Constantin Mierla -- http://www.asipto.comhttp://linkedin.com/in/miconda -- http://twitter.com/miconda >>> >>> >> >> >> -- >> Regards, >> >> MingHon >> >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> >> >> -- >> Daniel-Constantin Mierla -- http://www.asipto.comhttp://linkedin.com/in/miconda -- http://twitter.com/miconda >> >> >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> >> >> -- >> Daniel-Constantin Mierla -- http://www.asipto.comhttp://linkedin.com/in/miconda -- http://twitter.com/miconda >> >> > > > -- > Regards, > > MingHon > -- Regards, MingHon

2 3

route return value confusion
by Juha Heinanen 13 Dec '11

13 Dec '11

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

3 10

RTP rePacketization
by Arif Tuhin 13 Dec '11

13 Dec '11

HiI need to use the rtp repacketization feature of rtpproxy. Now i have also some custom rtp obsfucation modules which hides the rtp header and rtp payload size. Now if i want to use rtp repacketization(for_rtp_proxy(zNN)) , would it work? I mean does the rtp repacketization requires any header information? the architecture is described below: Dialer--------------->RTP Obsfucation--------------->kamailio + rtpproxy-------------------->Asterisk The rtp obsfucation works like this:Dialer obsfucates the packet and rtp proxy decodes it, from the other side Asterisk sends plain packets and the module obsfucates it. Its also stated in the manual the rtp repacketization is "possible" the it will repacket the rtps. Now Whats the conditions for which its not "possible"? With Best Regards Ariful Hossain Tuhin email: 1. etothepowerpi(a)gmail.com 2. etothepowerpi(a)hotmail.com 3.etothepowerpi(a)yahoo.com skype: freeburn1986

1 0

commands out of sync you can't run this command now
by MingHon 13 Dec '11

13 Dec '11

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

2 5

Re: [SR-Users] set host as inactive and probing in dispatcher
by Uri Shacked 13 Dec '11

13 Dec '11

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: <script>: SCRIPT : in OSPRPLERR sip:36264529@10.10.21.29 13(5760) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=1264 a=17 n=if 13(5760) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=1259 a=26 n=xlog 13(5760) ERROR: <script>: OSPRPL is 408 *now disable destination *13(5760) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=1260 a=26 n=ds_mark_dst 13(5760) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=1261 a=26 n=xlog 13(5760) ERROR: <script>: *after disabling destination* now i check the dispatcher list status: kamctl fifo ds_list SET:: 47 URI:: sip:10.10.21.20 flags=AX priority=25 attrs=OSP_B4 URI:: sip:10.10.21.22 flags=AX priority=25 attrs=OSP_B3 URI:: sip:10.10.21.23 flags=AX priority=25 attrs=OSP_B2 URI:: sip:10.10.21.21 flags=AX priority=25 attrs=OSP_B1 SET:: 41 URI:: sip:10.10.21.29 flags=AX priority=25 attrs=OSP_A4 URI:: sip:10.10.21.27 flags=AX priority=25 attrs=OSP_A3 URI:: sip:10.10.21.25 flags=AX priority=25 attrs=OSP_A1 URI:: sip:10.10.21.24 flags=AX priority=25 attrs=OSP_A2

2 7

Via header branch parameter in ACK message not unique
by Phillman25 Kyriacou 12 Dec '11

12 Dec '11

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? Thanking you in advance for your assistance. Phillip

2 3

503 on DB Error
by Spencer Thomason 12 Dec '11

12 Dec '11

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. Thanks, Spencer

4 7

use of $stat(name)
by Uri Shacked 12 Dec '11

12 Dec '11

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? thanks, Uri

2 1

Running Kamailio as non-root user
by Nathaniel L Keeling 09 Dec '11

09 Dec '11

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. Thanks Nathaniel L Keeling

3 3

Download Kamailio Debian package problem.
by Gary Chen 09 Dec '11

09 Dec '11

I am trying to download GPG key as following and got connection timeout: wget http://deb.kamailio.org/kamailiodebkey.gpg It seems that the server is down. Where else can I download this same Debian package? Gary

2 1

TLS<-->UDP not working with record_route_preset
by Gnaneshwar Gatla 09 Dec '11

09 Dec '11

Hello All, I have setup the Kamailio 3.1.0 behind a NAT/firewall. I'm using record_route_preset() function to keep the Public IP(advertised_address) in the Route header. I have to setup a where TLS->UDP and UDP->TLS to work. The sip clients do TLS and we have third party sip clients that only do UDP. The problem, Kamailio does not do double record-route when I provide record_route_preset(). This is leading to a problem where the call is either hung on TLS or UDP. I'm unsure how to go around this problem. I had posted earlier about this problem, is there a way I could use the advertised_address in the record route headers in this case? Regards Gnani

2 1

set host as inactive and probing in dispatcher
by Uri Shacked 09 Dec '11

09 Dec '11

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($avp(OSPRPL)!=404) { ds_mark_dst("IP"); } after trying this line i do "kamctl fifo ds_list" and i see the host is still at "AX". and in the next call the dispatcher still tries the same destination. why? how do i set a destinatio in dispatcher to be inactive? this is my modparams: #!ifdef WITH_DISPATCHER #------ dispatcher params --------- modparam("dispatcher", "db_url", DBURL) modparam("dispatcher", "dst_avp", "$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:user@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

3 3

unset host domain in dispatcher module
by Uri Shacked 09 Dec '11

09 Dec '11

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? it seems that kamailio dispatcher sets the host domain in the Internet Protocol, and the sip request line does not change 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...) the carrierroute module, do it better, but i need the dispatcher for the "keep alive" usage. ideas will be great :-) Uri

2 3

How to i use record_route_advertised_address
by Gnaneshwar Gatla 09 Dec '11

09 Dec '11

Hello, I have Kamailio 3.1.0 behind a NAT. We need the Firewall for sip-security reasons. Currently I'm using record_route_preset("advertised_Address") for keeping the advertised address in the record-route header. This works for fine when I have all clients running on UDP protocol. I currently have some TLS clients added to our network, Kamailio does not insert the second record-route header with the advertised-IP whenever there is a call between a UDP and a TLS Client. Instead it inserts the internal/Local IP of the server. I had also noticed the record_route_preset() function does not include the transport parameter ("transport=UDP" or "transport=TLS"). I was looking through the source code of Kamailio and found record_route_advertised_address() in record.c . I would like to know how to use this function with the Kamailio config. Regards Gnani

6 7

Presence Dialog not being created
by Gnaneshwar Gatla 08 Dec '11

08 Dec '11

Hello, Kamailio version: 3.1.5 I have been trying to use presence for Event:Dialog. I have used PUA_Dialoginfo module to accomplish this task. The PUA_dialoginfo module states that if "override_lifetime" is not used, the value of the expires is taken from dialog module. I have tried the module without the "override_lifetime" which did not create the dialog in the presentity. When used debug, I did see the xml being generated but finds the dialog "expires=0" and deletes the xml(please find the log below). But when "override_lifetime" is set, the dialog in the presentity table is set until. But this has a problem, modparam("pua_dialoginfo", "override_lifetime", 300) 1. if a call is still going on more than the override_lifetime the presentity is deleted. 2. the presentity information is available until the "override_lifetime" even after the call is hung-up. How do I get this module working without the "override_lifetime" being used. The Log: Oct 21 12:12:46 SIPTest /usr/local/sbin/kamailio[3289]: DEBUG: pua_dialoginfo [dialog_publish.c:242]: new_body:#012<?xml version="1.0"?>#012<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="0" state="full" entity="sip:user@mydemo.com">#012 <dialog id="112014dff048e71e" call-id="112014dff048e71e" direction="initiator">#012 <state>Trying</state>#012 <remote>#012 <identity>sip:user01@mydemo.com</identity>#012 <target uri="sip:user01@mydemo.com"/>#012 </remote>#012 <local>#012 <identity>sip:user@mydemo.com</identity>#012 <target uri="sip:user@mydemo.com"/>#012 </local>#012 </dialog>#012</dialog-info>#012 Oct 21 12:12:46 SIPTest /usr/local/sbin/kamailio[3288]: DEBUG: <core> [db_res.c:81]: freeing 1 columns Oct 21 12:12:46 SIPTest /usr/local/sbin/kamailio[3289]: DEBUG: pua_dialoginfo [dialog_publish.c:290]: publish uri= sip:user@mydemo.com Oct 21 12:12:46 SIPTest /usr/local/sbin/kamailio[3288]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[0] at 0x8314aa4 Oct 21 12:12:46 SIPTest /usr/local/sbin/kamailio[3289]: <core> [mem/q_malloc.c:366]: qm_malloc(0x829bee0, 756) called from pua_dialoginfo: dialog_publish.c: dialog_publish(302) Oct 21 12:12:46 SIPTest /usr/local/sbin/kamailio[3288]: <core> [mem/q_malloc.c:428]: qm_free(0x829bee0, 0x8314aa4), called from <core>: db_res.c: db_free_columns(86) Oct 21 12:12:46 SIPTest /usr/local/sbin/kamailio[3289]: <core> [mem/q_malloc.c:406]: qm_malloc(0x829bee0, 756) returns address 0x82e84bc frag. 0x82e84a4 (size=900) on 1 -th hit Oct 21 12:12:46 SIPTest /usr/local/sbin/kamailio[3288]: <core> [mem/q_malloc.c:450]: qm_free: freeing frag. 0x8314a8c alloc'ed from db_mysql: km_res.c: db_mysql_get_columns(78) Oct 21 12:12:46 SIPTest /usr/local/sbin/kamailio[3289]: DEBUG: pua_dialoginfo [dialog_publish.c:51]: publ: Oct 21 12:12:46 SIPTest /usr/local/sbin/kamailio[3288]: DEBUG: <core> [db_res.c:94]: freeing result names at 0x8318650 Oct 21 12:12:46 SIPTest /usr/local/sbin/kamailio[3289]: DEBUG: pua_dialoginfo [dialog_publish.c:52]: uri= sip:user@mydemo.com Oct 21 12:12:46 SIPTest /usr/local/sbin/kamailio[3288]: <core> [mem/q_malloc.c:428]: qm_free(0x829bee0, 0x8318650), called from <core>: db_res.c: db_free_columns(95) Oct 21 12:12:46 SIPTest /usr/local/sbin/kamailio[3289]: DEBUG: pua_dialoginfo [dialog_publish.c:53]: id= DIALOG_PUBLISH.112014dff048e71e Oct 21 12:12:46 SIPTest /usr/local/sbin/kamailio[3288]: <core> [mem/q_malloc.c:450]: qm_free: freeing frag. 0x8318638 alloc'ed from <core>: db_res.c: db_allocate_columns(148) Oct 21 12:12:46 SIPTest /usr/local/sbin/kamailio[3289]: DEBUG: pua_dialoginfo [dialog_publish.c:54]: expires= 0 Oct 21 12:12:46 SIPTest /usr/local/sbin/kamailio[3288]: DEBUG: <core> [db_res.c:99]: freeing result types at 0x8318684 Oct 21 12:12:46 SIPTest /usr/local/sbin/kamailio[3289]: DEBUG: pua [send_publish.c:403]: pres_uri=sip:user@mydemo.com Oct 21 12:12:46 SIPTest /usr/local/sbin/kamailio[3288]: <core> [mem/q_malloc.c:428]: qm_free(0x829bee0, 0x8318684), called from <core>: db_res.c: db_free_columns(100) Oct 21 12:12:46 SIPTest /usr/local/sbin/kamailio[3289]: DEBUG: pua [hash.c:121]: core_hash= 504 Oct 21 12:12:46 SIPTest /usr/local/sbin/kamailio[3288]: <core> [mem/q_malloc.c:450]: qm_free: freeing frag. 0x831866c alloc'ed from <core>: db_res.c: db_allocate_columns(157) Oct 21 12:12:46 SIPTest /usr/local/sbin/kamailio[3289]: DEBUG: pua [hash.c:171]: record not found Oct 21 12:12:46 SIPTest /usr/local/sbin/kamailio[3288]: DEBUG: <core> [db_res.c:54]: freeing 1 rows Oct 21 12:12:46 SIPTest /usr/local/sbin/kamailio[3289]: DEBUG: pua [send_publish.c:444]: insert type Oct 21 12:12:46 SIPTest /usr/local/sbin/kamailio[3288]: DEBUG: <core> [db_row.c:97]: freeing row values at 0x8314b14 Oct 21 12:12:46 SIPTest /usr/local/sbin/kamailio[3289]: DEBUG: pua [send_publish.c:448]: UPDATE_TYPE and no record found Oct 21 12:12:46 SIPTest /usr/local/sbin/kamailio[3288]: <core> [mem/q_malloc.c:428]: qm_free(0x829bee0, 0x8314b14), called from <core>: db_row.c: db_free_row(98) Oct 21 12:12:46 SIPTest /usr/local/sbin/kamailio[3289]: DEBUG: pua [send_publish.c:454]: request for a publish with expires 0 and no record found Oct 21 12:12:46 SIPTest /usr/local/sbin/kamailio[3288]: <core> [mem/q_malloc.c:450]: qm_free: freeing frag. 0x8314afc alloc'ed from <core>: db_row.c: db_allocate_row(114) Oct 21 12:12:46 SIPTest /usr/local/sbin/kamailio[3289]: <core> [mem/q_malloc.c:428]: qm_free(0x829bee0, 0x82e84bc), called from pua_dialoginfo: dialog_publish.c: dialog_publish(357) Oct 21 12:12:46 SIPTest /usr/local/sbin/kamailio[3288]: DEBUG: <core> [db_res.c:62]: freeing rows at 0x8314adc Regards Gnaneshwar

2 4

Problem to verify contact header
by Bruno Bresciani 08 Dec '11

08 Dec '11

Hi All, Kamailio generate a core at line below if(_msg->contact!= NULL || _msg->contact->body.s!= NULL){ _msg is a sip_msg struct that my module receive from kamailio. I want verify if on that request messagem have a contact header, but a core is being generated when contact header isn't present on message. Someone knows why this is happening? Cheers

3 4

How to test registration setting
by Efelin Novak 08 Dec '11

08 Dec '11

Hi folks, I would like to know what to use for the following scenario. I want users to be able to test their registration by sending REGISTER and Kamailio sends them 200 OK if credentials are OK. However I don't want the aor to be stored in location table. This REGISTER should be only used for testing the settings. After some time I would allow the registration to be stored and users will normally receive calls. I tried to implement this by not doing save(location) when the flag from load_credential is set. Unfortunately it is the function save who sends 200 OK. When it is not being called, a user agent keeps sending REGISTER until SIP triggers fail. So is there any other common practice how to test registers? Is there any flag that can prevent storing location? Can I send 200 OK to these user in any other way? Thanks Efelin

2 2

Re: [SR-Users] Remote Party ID
by Daniel-Constantin Mierla 08 Dec '11

08 Dec '11

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 <username> <rpid> ......... add rpid for a user (*) > > rpid rm <username> ................. set rpid to NULL for a user (*) > > rpid show <username> ............... 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:miconda@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

1 0

[OT] sip:provider CE v2.4 released
by Andreas Granig 07 Dec '11

07 Dec '11

Ladies and Gentlemen, I'm happy to announce the official release of sip provider CE v2.4, an easy-to-use, free and open-source soft-switch built on top of Kamailio and Sems. As always, we provide a dead-simple installer to start from scratch, upgrade scripts from v2.2 and Virtualbox and VMware images to test-drive a turn-key installation. http://www.sipwise.com/news/announcements/spce-v2_4-release/ The SPCE shows the capabilities of what can be created with flexible and powerful software like Kamailio and Sems, when the proper provisioning, billing, configuration and packaging glue is being put around that. The SPCE v2.4 is currently deployed around the world acting as an SBC in front of old legacy 3rd party switches to provide SIP service to mobile clients, as a Class4 peering concentrator to provide ENUM/PSTN access to other soft-switches, and as standard Class5 soft-switches for all kinds of access networks. It's ideal for new Kamailio/Sems users to have a head-start over beginning completely from scratch, because the SPCE comes with customer- and admin-web-interfaces, SOAP/XMLRPC-APIs, provisioning and rating engines etc. No need to tinker with all the various configuration files, unless you want to. :) You can find the full release announcement with the changelog highlights over here: http://www.sipwise.com/news/announcements/spce-v2_4-release/ Thanks, Andreas

3 3

[OT] job offer - System administrator middleware & acs
by Henning Westerholt 07 Dec '11

07 Dec '11

[OT] Job offer: Interested in working with one of the biggest Voice over IP networks in europe? We're hiring and have one additional IT operation position open in our team in Karlsruhe, Germany. You would be responsible in a team of motivated system administrators for the operation and maintenance of the high- available middleware cluster and CPE auto-configuration services for the DSL and mobile products at 1&1. Furthermore you would be support the on-call duty for this services in the team. A solid understanding of german is necessary for the job, for more informations (additional requirements, how to apply..) please refer to the opening on the job web page. Of course please feel free to contact me per mail as well. http://jobs.1und1.de/xml/JobsDetail?job_id=3777 Best regards, Henning -- Henning Westerholt - Head of IT Operations Internet Access & Communications 1&1 Internet AG, Brauerstraße 48, 76135 Karlsruhe, Germany

1 0

Re: [SR-Users] Remote Party ID
by Daniel-Constantin Mierla 06 Dec '11

06 Dec '11

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

1 0

Re: [SR-Users] Remote Party ID
by Jiri Kuthan 06 Dec '11

06 Dec '11

I think the most straight-forward way is to load the RPID in user preferences as AVP, and print the associated AVP from SER script. I guess there is an example of such in the OOB script -- have you tried to look there? -jiri 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? > > > > thank > > > > palo73 > > > > > > _______________________________________________ > 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

1 0

Can the SBC (Kamailio 3.1.x and FreeSWITCH 1.0.6+) do Upper Registration?
by edson.gomes.leme 06 Dec '11

06 Dec '11

3 2

Remote Party ID
by Pavel Segeč 06 Dec '11

06 Dec '11

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? thank palo73

1 0

Can the SBC (Kamailio 3.1.x and FreeSWITCH 1.0.6+) do Upper Registration?
by edson.gomes.leme 06 Dec '11

06 Dec '11

1 0

kamailio fail to start when there are many records in DB in location or in pua table
by laura testi 06 Dec '11

06 Dec '11

Hi all, when there many records in DB tables like location(usrloc use write back mode), pua etc, kamailio fail to start by complaining "no pkg memory left" when try to load (restore) all records from DB to hashtable of location/pua. The share momory are allocated enough for the hashtable, but the default compilation of the PKG_MEMORY_SIZE is 4MB, which allow only restore several thousands records. I know I can increase the PKG_MEMORY_SIZE, but anyway there is a limitation which may fail to restore all records from DB during the startup of kamailio if it reach the limit of the number of records in DB. Is there anyway to reload the hashtable from table by doing query n rows a time and loop until non more result from DB? this will avoid the kamailio startup problem when there are many records in hashtable which are save also in DB. Thank you very much in advanced! Laura

2 4

dialog table entry not deleted after call if in-dialog requests are challenged
by Yufei Tao 06 Dec '11

06 Dec '11

Hi I am using dialog module in Kamailio 3.1.5, and find that the dialog table entry does not get deleted after call if in-dialog requests (e.g. session timer refresh re-INVITEs in my case) are challenged. The entry will stay there and get deleted after a long time (some time out? I didn't set any dialog timeouts myself). If I remove the challenge for in-dialog requests, then the dialog entry is deleted right after the call. I use the example config script with a bit of modification for my in-dialog route: route[WITHINDLG] { xlog("L_DBG","WITHINDLG: method=$rm, callid=$ci, cseq=$cs\n"); if (has_totag() ) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { # Filter out bad/faked up in-dialog requests. if ( !is_known_dlg() ) { xlog("L_WARN","WITHINDLG: NOT known dlg!! method=$rm, callid=$ci, cseq=$cs, $fu->$tu\n"); exit; } if (is_method("BYE")) { xlog("L_DBG","WITHINDLG: BYE found within Dialog \n"); setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } ########################################################## # challenge: if not CANCEL/ACK else if ( !is_method("CANCEL") && !is_method("ACK") ) { route(AUTH); } ########################################################## xlog("L_DBG","WITHINDLG: Entering RELAY route from WITHINDLG route. \n"); route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { xlog("L_DBG","WITHINDLG: ACK callid=$ci, cseq=$cs\n"); if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server xlog("L_DBG","ACK being relayed\n"); t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard xlog("L_DBG","ACK without matching transaction-ignore and discard\n"); exit; } } sl_send_reply("404","Not here"); } exit; } } The part that calls route(AUTH) that I added was the cause for the dialog entry not being deleted after call. Here are my dialog related settings: modparam("dialog", "enable_stats", 1) modparam("dialog", "dlg_flag", 4) modparam("dialog", "db_url", DBURL) modparam("dialog", "db_update_period", 20) # use short one to catch short dlgs (default:60) modparam("dialog", "db_mode", 1) # don't use did, so to cope with clients that don't preserve parameters # and mode 1 (fallback to 3261 matching) didn't work?? modparam("dialog", "dlg_match_mode", 2) modparam("dialog", "profiles_no_value", "inbound ; outbound") modparam("dialog", "dlg_extra_hdrs", "Hint: dialog expired\r\n") # default timeout set to max int 0x07FFFF FFFF modparam("dialog", "default_timeout", 2147483647) And I set the dialog flag (4) when receiving the initial INVITE. Any idea why? Am I using the dialog module correctly? Thanks very much! Yufei -- Yufei Tao Red Embedded This E-mail and any attachments hereto are strictly confidential and intended solely for the addressee. If you are not the intended addressee please notify the sender by return and delete the message. You must not disclose, forward or copy this E-mail or attachments to any third party without the prior consent of the sender. Red Embedded Design, Company Number 06688253 Registered in England: The Waterfront, Salts Mill Rd, Saltaire, BD17 7EZ

2 6

Issue while initiating database of my new module in kamailio
by Jagdeep S 06 Dec '11

06 Dec '11

Hi All, I am trying to create a new module under the kamailio source base, as part of understanding the framework. But when i start the Kamailio, i am seeing a core dump occuring when my new module is getting initiated and when the database is getting initialised. PFA the files attached containing the config file and source files used for creating module files. I am using the "DEFAULT_DB_URL" as database uri. The core dump backtrace shows that the module init fails at the following point: (gdb) bt #0 0x00000000 in ?? () #1 0x00ef390d in child_init (rank=9172352) at my_mod.c:239 #2 0x081226be in init_mod (m=0xb7515a04) at sr_module.c:922 #3 0x08122630 in init_mod (m=0xb7515c78) at sr_module.c:919 The line 239 in my module file points to the child-init function which initializes the database connectivity ==> "mymod_dbh = mymod_dbf.init(&db_url);" I have written the module using the template provided under the kamailio website. All the parameters are defined properly. Have no idea as to what is going wrong. I tried debugging the core dump using gdb but of no much help. I am not able to run the gdb during runti sr-users(a)lists.sip-router.orgme due to which i am not able to add break points and check. Not sure what the problem is. That's why i generated core dump file and analysed the backtrace, but i can't add break points and run it again as the executable is not involved. Can anyone please let me know what is the issue is or any pointers if i have missed some part of the code due to which the connectivity to db is not happening. Due to this i am not able to proceed further with the testing of my new module. Also any tips on how to do get the gdb debugging during runtime will also be very helpful. Any help is appreciated :). -- Jagdeep S

2 4

myself variable
by Ben WIlliams 06 Dec '11

06 Dec '11

Hi, I am an absolute beginniner learning SIP and Kamailio to setup Kamailio as the Internet proxy for our Asterisk PBX. My question is how to list the contents of the myself variable from the command line? My Kamailio server displays a list of Aliases on startup. Are these the contents of the myself variable? Thanks Ben

2 1

Video questions
by Lucas Alvarez 05 Dec '11

05 Dec '11

Hi, I have integrated Kamailio with asterisk, is it possible to handle video with kamailio and audio with asterisk? I'm using kamailio 3.2, I could difference a video call using sdpopts module, so before the invite is routed to asterisk kamailio sends it to the phone. But I would like that the video media would be peer to peer and the audio follows the path through asterisk. Thanks in advance, Lucas Alvarez

2 1

Permissions module question
by pablo umanzor 05 Dec '11

05 Dec '11

hi, recently i've implemented the module antiflood into kamailio (3.1.5 from GIT) ,the module works and i can see the ip banned with kamctl fifo sht_dump ipban now, when i try to allocate a trusted ip address with permissions module #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") #!endif -------------- if(src_ip!=TRUSTEDIP) then i make a sipsak flood test from ip address 1.2.3.4 , but the ip address entered before into trusted table (1.2.3.4 ) again is banned with antiflood module syslog file : INFO: auth [auth_mod.c:312]: auth: qop set, but nonce-count (nc_enabled) support disabled : INFO: permissions [parse_config.c:251]: file not found: /usr/local/etc/kamailio/permissions.allow : INFO: permissions [permissions.c:606]: default allow file (/usr/local/etc/kamailio/permissions.allow) not found => empty rule set : INFO: permissions [parse_config.c:251]: file not found: /usr/local/etc/kamailio/permissions.deny :INFO : permissions [permissions.c:615]: default deny file (/usr/local/etc/kamailio/permissions.deny) not found => empty rule set must i to create this files (permissions.allow, deny) to this module works? or is enough with empty rule set, according with documentation - there are another options into this module, but i need only enabling the trusted ip address part. regards pablo umanzor

2 1

Re: [SR-Users] "Version" Table missing
by Daniel-Constantin Mierla 05 Dec '11

05 Dec '11

Hello, don't forget to keep the mailing list cc-ed for the discussions started there... On 12/5/11 8:20 PM, mn3250 mn3250 wrote: > Had to modify /usr/local/etc/kamailio/kamctlrc and everything started > to work. Was it just setting DBENGINE and then running kamdbctl create or was some other kind of modification? Cheers, Daniel > > On Mon, Nov 28, 2011 at 4:15 PM, Daniel-Constantin Mierla > <miconda(a)gmail.com <mailto:miconda@gmail.com>> wrote: > > Hello, > > can you check inside MySQL server if database voip1 has a table > named version? > > What is the kamailio version you are using? How did you create the > database, with kamdbctl create? > > Cheers, > Daniel > > > On 11/28/11 2:59 PM, mn3250 mn3250 wrote: >> Hi, >> Looks like the table "Version" is missing in the sql create script. >> 0(18594) DEBUG: db_mysql [km_my_con.c:94]: opening connection: >> mysql://xxxx:xxxx@localhost/voip1 >> 0(18594) DEBUG: db_mysql [km_my_con.c:121]: connection type is >> Localhost via UNIX socket >> 0(18594) DEBUG: db_mysql [km_my_con.c:122]: protocol version is 10 >> 0(18594) DEBUG: db_mysql [km_my_con.c:123]: server version is >> 5.0.77-log >> 0(18594) DEBUG: <core> [db_res.c:118]: allocate 48 bytes for >> result set at 0x2ba27ac440b0 >> 0(18594) DEBUG: db_mysql [km_res.c:68]: 1 columns returned from >> the query >> 0(18594) DEBUG: <core> [db_res.c:155]: allocate 8 bytes for >> result names at 0x2ba27ac44140 >> 0(18594) DEBUG: <core> [db_res.c:165]: allocate 4 bytes for >> result types at 0x2ba27ac441a8 >> 0(18594) DEBUG: db_mysql [km_res.c:85]: allocate 16 bytes for >> RES_NAMES[0] at 0x2ba27ac44210 >> 0(18594) DEBUG: db_mysql [km_res.c:92]: >> RES_NAMES(0x2ba27ac44210)[0]=[table_version] >> 0(18594) DEBUG: db_mysql [km_res.c:100]: use DB1_INT result type >> 0(18594) DEBUG: db_mysql [km_res.c:169]: no rows returned from >> the query >> 0(18594) DEBUG: <core> [db.c:385]: no row for table location found >> 0(18594) ERROR: <core> [db.c:421]: invalid version 0 for table >> location found, expected 1004 (check table structure and table >> "version") >> Anyone has the correct file? >> Thanks >> >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >> sr-users(a)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 > Kamailio Advanced Training, Dec 5-8, Berlin:http://asipto.com/u/kat > http://linkedin.com/in/miconda -- http://twitter.com/miconda > > -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Dec 5-8, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 0

undefined symbol: dprint_crit
by Ämin Baumeler 04 Dec '11

04 Dec '11

Hi everybody I cross compiled kamailio for ARM. However, when I try to start kamailio on the ARM machine, it tells me that there are a lot of undefined symbols. E.g. libkcore.so does not find the symbol dprint_crit. Having a look at libkcore.so: $ nm -C -D lib/kcore/libkcore.so|grep dprint U dprint_crit All deployed .so files do not contain dprint_crit. However I found dprint_crit in an object file, that was used for compilation. Namely in dprint.o: $ nm -C dprint.o |grep print 00000000 B dprint_crit I used following commands to cross compile kamailio: export GCC_ARM_HOME=/opt/arm-2009q1/arm-none-linux-gnueabi/ export TOOL_PREFIX=/opt/arm-2009q1/bin/arm-none-linux-gnueabi export CXX=$TOOL_PREFIX-g++ export AR=$TOOL_PREFIX-ar export RANLIB=$TOOL_PREFIX-ranlib export CC=$TOOL_PREFIX-gcc export LD=$TOOL_PREFIX-ld export ARM_TARGET_LIB=/opt/arm-2009q1/arm-none-linux-gnueabi/libc make proper make cfg FLAVOUR=kamailio PREFIX=/root/kamailio/local include_modules="auth auth_db db_text registrar" ARCH=arm TARGET=arm-none-linux-gnueabi make all && make install Thanks Amin

4 5

CANCEL not matching INVITES !
by Sammy Govind 03 Dec '11

03 Dec '11

Hello list, I'm using Kamailio 3.1.5 in front of asterisk servers. Kamailio handles all the SIP registrations. Calls from SIP phones are forwarded to asterisks and then dialled out to Kamailio. root@SBCserver:~# kamailio -V version: kamailio 3.1.5 (x86_64/linux) 76fff5 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, PKG_SIZE 4MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 76fff5 compiled on 08:21:33 Oct 27 2011 with gcc 4.6.1 root@SBCserver:~# Problem: When call is initiated from a softphone and is in ringing phase, CANCEL just don't work. I've done some initial debugging and the following piece of code in main route is failing. # CANCEL processing if (is_method("CANCEL")) { xlog("L_NOTICE","$rm from $fu (IP:$si:$sp) ---CAPTURED IN MAIN---\n"); if (t_check_trans()){ t_relay(); xlog("L_NOTICE","$rm from $fu (IP:$si:$sp) ---CHECK TRANS TRUE---\n"); } xlog("L_NOTICE","$rm from $fu (IP:$si:$sp) ---CHECK TRANS FALSE---\n"); exit; } Also the CANCEL fails the has_totag() condition ! The same Call CANCEL scenario works fine for any client on Public IP ! Hope to get some pointers for the solution. Regards, Sammy.

2 10

rate limit module
by Fabian Borot 02 Dec '11

02 Dec '11

Hello, I am trying to use the rate limit module using Kamailio 1.5.2. I feel that I got it but I would like some pointers and recommendations. These are my settings: (this is a lab of course), I used 1 on the timer_interval because I am generating the calls manually and wanted to see it in action quickly: # ---- ratelimit ------- modparam("ratelimit", "timer_interval", 1) modparam("ratelimit", "reply_code", 506) modparam("ratelimit", "reply_reason", "Rejecting due to high load") modparam("ratelimit", "queue", "0:INVITE") modparam("ratelimit", "pipe", "0:TAILDROP:1") then in the route section: if (method=="INVITE") { xlog("L_INFO","mylog: RL found INVITE.\n"); if (!rl_check()) { xlog("L_INFO","mylog: RL dropped message.\n"); rl_drop(); exit; }; xlog("L_INFO","mylog: RL found INVITE but did not drop it.\n"); }; The TAILDROP algorithm seems to work better than the RED, based on what I expected of course (with 1 sec timer interval and 1 calls/sec on the pipe). Making manual calls one right after the other almost always triggered the protecting when there was another call on the same second. But these lines (1.6.3. pipe) on the doc got me kind of confused: " When specifying a limit, the unit depends on the algorithm used and doesn't need to be specified also (eg, for TAILDROP or RED, limit means packets/sec, whereas with the FEEDBACK algorithm, it means [CPU] load factor). For these 2 lines below, does this mean that the interval =10 will be overridden by the 100 calls/sec on the TAILDROP algorithm? modparam("ratelimit", "timer_interval", 10) modparam("ratelimit", "pipe", "0:TAILDROP:100") I made a quick test and with timer_interval = 10 and TAILDROP:1, it looks like the protection kicks in almost every 10 secs: (Dec 2 19:44:07 and Dec 2 19:43:57), tail -f proxy.log | grep RL Dec 2 19:43:53 AW-ATL-PROXY-01 ./kamailio[17450]: mylog: RL found INVITE. Dec 2 19:43:53 AW-ATL-PROXY-01 ./kamailio[17450]: mylog: RL found INVITE but did not drop it. Dec 2 19:43:53 AW-ATL-PROXY-01 ./kamailio[17449]: mylog: RL found INVITE. Dec 2 19:43:53 AW-ATL-PROXY-01 ./kamailio[17449]: mylog: RL found INVITE but did not drop it. Dec 2 19:43:53 AW-ATL-PROXY-01 ./kamailio[17450]: mylog: RL found INVITE. Dec 2 19:43:53 AW-ATL-PROXY-01 ./kamailio[17450]: mylog: RL found INVITE but did not drop it. Dec 2 19:43:53 AW-ATL-PROXY-01 ./kamailio[17450]: mylog: RL found INVITE. Dec 2 19:43:53 AW-ATL-PROXY-01 ./kamailio[17450]: mylog: RL found INVITE but did not drop it. Dec 2 19:43:54 AW-ATL-PROXY-01 ./kamailio[17450]: mylog: RL found INVITE. Dec 2 19:43:54 AW-ATL-PROXY-01 ./kamailio[17450]: mylog: RL found INVITE but did not drop it. Dec 2 19:43:55 AW-ATL-PROXY-01 ./kamailio[17449]: mylog: RL found INVITE. Dec 2 19:43:55 AW-ATL-PROXY-01 ./kamailio[17449]: mylog: RL found INVITE but did not drop it. Dec 2 19:43:56 AW-ATL-PROXY-01 ./kamailio[17449]: mylog: RL found INVITE. Dec 2 19:43:56 AW-ATL-PROXY-01 ./kamailio[17449]: mylog: RL found INVITE but did not drop it. Dec 2 19:43:56 AW-ATL-PROXY-01 ./kamailio[17450]: mylog: RL found INVITE. Dec 2 19:43:56 AW-ATL-PROXY-01 ./kamailio[17450]: mylog: RL found INVITE but did not drop it. Dec 2 19:43:56 AW-ATL-PROXY-01 ./kamailio[17449]: mylog: RL found INVITE. Dec 2 19:43:56 AW-ATL-PROXY-01 ./kamailio[17449]: mylog: RL found INVITE but did not drop it. Dec 2 19:43:57 AW-ATL-PROXY-01 ./kamailio[17450]: mylog: RL found INVITE. Dec 2 19:43:57 AW-ATL-PROXY-01 ./kamailio[17450]: mylog: RL found INVITE but did not drop it. Dec 2 19:43:57 AW-ATL-PROXY-01 ./kamailio[17449]: mylog: RL found INVITE. Dec 2 19:43:57 AW-ATL-PROXY-01 ./kamailio[17449]: mylog: RL dropped message. Dec 2 19:43:58 AW-ATL-PROXY-01 ./kamailio[17449]: mylog: RL found INVITE. Dec 2 19:43:58 AW-ATL-PROXY-01 ./kamailio[17449]: mylog: RL found INVITE but did not drop it. Dec 2 19:43:59 AW-ATL-PROXY-01 ./kamailio[17449]: mylog: RL found INVITE. Dec 2 19:43:59 AW-ATL-PROXY-01 ./kamailio[17449]: mylog: RL found INVITE but did not drop it. Dec 2 19:43:59 AW-ATL-PROXY-01 ./kamailio[17450]: mylog: RL found INVITE. Dec 2 19:43:59 AW-ATL-PROXY-01 ./kamailio[17450]: mylog: RL found INVITE but did not drop it. Dec 2 19:44:01 AW-ATL-PROXY-01 ./kamailio[17450]: mylog: RL found INVITE. Dec 2 19:44:01 AW-ATL-PROXY-01 ./kamailio[17450]: mylog: RL found INVITE but did not drop it. Dec 2 19:44:02 AW-ATL-PROXY-01 ./kamailio[17450]: mylog: RL found INVITE. Dec 2 19:44:02 AW-ATL-PROXY-01 ./kamailio[17450]: mylog: RL found INVITE but did not drop it. Dec 2 19:44:02 AW-ATL-PROXY-01 ./kamailio[17449]: mylog: RL found INVITE. Dec 2 19:44:02 AW-ATL-PROXY-01 ./kamailio[17449]: mylog: RL found INVITE but did not drop it. Dec 2 19:44:04 AW-ATL-PROXY-01 ./kamailio[17449]: mylog: RL found INVITE. Dec 2 19:44:04 AW-ATL-PROXY-01 ./kamailio[17449]: mylog: RL found INVITE but did not drop it. Dec 2 19:44:06 AW-ATL-PROXY-01 ./kamailio[17450]: mylog: RL found INVITE. Dec 2 19:44:06 AW-ATL-PROXY-01 ./kamailio[17450]: mylog: RL found INVITE but did not drop it. Dec 2 19:44:06 AW-ATL-PROXY-01 ./kamailio[17449]: mylog: RL found INVITE. Dec 2 19:44:06 AW-ATL-PROXY-01 ./kamailio[17449]: mylog: RL found INVITE but did not drop it. Dec 2 19:44:07 AW-ATL-PROXY-01 ./kamailio[17449]: mylog: RL found INVITE. Dec 2 19:44:07 AW-ATL-PROXY-01 ./kamailio[17449]: mylog: RL found INVITE but did not drop it. Dec 2 19:44:07 AW-ATL-PROXY-01 ./kamailio[17450]: mylog: RL found INVITE. Dec 2 19:44:07 AW-ATL-PROXY-01 ./kamailio[17450]: mylog: RL dropped message. Dec 2 19:44:10 AW-ATL-PROXY-01 ./kamailio[17450]: mylog: RL found INVITE. Dec 2 19:44:10 AW-ATL-PROXY-01 ./kamailio[17450]: mylog: RL found INVITE but did not drop it.

2 3

RPMs for 3.2.1
by Daniel-Constantin Mierla 02 Dec '11

02 Dec '11

Hello, RPM packages of Kamailio 3.2.1 for Centos, Fedora, Redhat and openSUSE (32b and 64b) were built and are available for download -- links specific for each distro in the wiki page: http://www.kamailio.org/wiki/packages/rpms Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Dec 5-8, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 0

transport=TLS
by Bruno Bresciani 02 Dec '11

02 Dec '11

Hi All, kamailio 3.1.2 recognize value "transport=TLS" on contact header or only "transport=tls"? kamailio forwards (t_relay function) the message with UDP protocol when value "transport=TLS" is on contact header. Cheers

3 13

Monitoring inquiry
by Jenkins, Rachel 01 Dec '11

01 Dec '11

I have a couple questions as it relates to monitoring a Kamailio SIP Server. 1. Do you have an OID that we can poll via SNMP-GETs to show call completion statistics? Either the % of completion or the accepting/connecting statistics. If you do not have an OID, do you have a syslog or trap message that can be generated? 2. Do you have an OID that we can poll to get a list errors that were generated? Ex-OID for a 408 or 503. If not, is there a syslog or trap? 3. Can you recommend a way to view the MOS score - possible hardware tap on network to get MOS? Thanks in advance This email may contain proprietary and confidential information for the sole use of the intended recipient. Any review, retransmission, dissemination, or other use of this information by persons or entities other than the intended recipient is prohibited. If you are not the intended recipient, please contact the sender and delete all copies. To the extent that opinions are expressed in this message, they are not necessarily the opinions of Sentinel Technologies or any of its affiliates, employees, directors, officers or shareholders.

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Kamailio v3.2.1 Released
by Daniel-Constantin Mierla 01 Dec '11

01 Dec '11

Hello, Kamailio SIP Server v3.2.1 stable release is out. This is a maintenance release of latest stable branch, 3.2, that includes fixes since release of v3.2.0. There is no change to database schema or configuration language structure that you have to do on installations of v3.2.0. Deployments running previous v3.2.0 versions are strongly recommended to be upgraded to v3.2.1. For more details about version 3.2.1 (including links and hints to download the tarball or from GIT repository), visit: http://www.kamailio.org/w/2011/12/kamailio-v3-2-1-released/ RPM packages are currently on the build, Debian/Ubuntu packages will be available soon as well. Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com

1 0

preparing to release v3.2.1
by Daniel-Constantin Mierla 01 Dec '11

01 Dec '11

Hello, the plan is to start packaging of v3.2.1 after 14:00GMT. If anyone has anything elst to commit to branch 3.2 to be in v3.2.1, please do it before, afterwards write me an email first to synchronize and avoid git conflicts. Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Dec 5-8, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 1

how approach nathelper when using usrloc db_mode 3?
by Albert Petit 01 Dec '11

01 Dec '11

Hello, We found an issue when using nat pinger Previously our system had 1 Kamailio as entry point and we were using usrloc in mode 2 so: modparam("usrloc", "db_mode",2) modparam("usrloc", "db_url", "mysql://openser:openserrw@<DB_IP>/openser") On our system we like to use the pinger NAT_HELPER module to keep the connections alive for the users of our system. Now we're trying to configure two Kamailio K1 and K2 as entry point of the system and having both of them active at the same time with a virtual IP as common entry point for both. I believe this approach requires to flush information on the usrloc immediately because following use case can happen and needs to work: - A registers on K1 server - B registers on K2 server -After 5 seg B makes an INVITE to A through K2=> we need K2 kamailio instance to be able to locate A immediately and not wait until the first Kamailio flushes the info to the db. (it looks to be around 20/30 seg by default on mode 2 I think , correct so this looks to be too much time) So to improve this we tried to change to use usrloc in mode 3 instead of mode 2. But then we've a big issue: when enabling the pinger module for NATted users we see it opens a new connection to the db for each OPTIONS ping we want to send. That looks too many connections for our MySQL server because for eg if I leave two users registered for 1 hour after some time I start to see errors like: 13(5292) ERROR: <core> [db_query.c:130]: error while submitting query 13(5292) ERROR: usrloc [dlist.c:151]: raw_query failed 13(5292) ERROR: nathelper [nathelper.c:3459]: failed to fetch contacts 13(5292) ERROR: db_mysql [km_dbase.c:117]: driver error on query: Too many conne How should this problem be approached? Could this be due to wrong mysql configuration? But anyway if for each ping done by NAT_PINGER we make a query to the db I think this is a lot of "noise" for the db as pings happen often! Thanks BR Albert

1 0

Re: [SR-Users] gracefuly stop kamailio
by Uri Shacked 30 Nov '11

30 Nov '11

thanks. any way to know the number of active dialogs without the dialog module? with TM maybe?

2 6

no sercmd_ctl file
by Uri Shacked 30 Nov '11

30 Nov '11

how do i restore the serdmd file in order to run commands with sercmd? is there a way to run sercmd command woth kamctl only?

2 1

starting kamailio method
by Uri Shacked 30 Nov '11

30 Nov '11

hi, what is the differebt when i start kamailio with the command "kamailio" only, or using "/etc/init.d/kamailio start ? using the first one gets me to see the log running... second one does not.... and with the second one i can configure the memory usage and so.... how do i make them do the same?

3 4

using kamailio as load balancer
by Albert Petit 29 Nov '11

29 Nov '11

Hello, I'm trying to setup kamailio as load balancer Behind that load balancer I've a registrar and a presence server. When I start a client for the REGISTER and subscribe messages all looks to work fine. The messages from clients are properly balanced to the backend and responses are forwarded properly to the clients. However when my presence server sends a NOTIFY I find the load balancer module answers 404 message to the presence server. I'm thinking the problem is front-end kamailio cannot locate users. I am quite newbie to use Kamailio for this purpose so sorry if what i'm saying contains a big mistake. My load balancer script looks as following: log_facility=LOG_LOCAL0 fork=yes children=4 listen=10.1.1.63:6666 listen=10.1.1.63:5060 alias=dev.myenvironment.net <http://dev.genaker.net> mpath="/usr/local/lib/kamailio/modules_k/:/usr/local/lib/kamailio/modules/" loadmodule "xlog.so" # ----------------- setting module-specific parameters --------------- #!define WITH_DISP #!ifdef WITH_DISP loadmodule "dispatcher.so" loadmodule "db_mysql.so" #!endif # ----- dispatcher ----- #!ifdef WITH_DISP # Set database used for dispatcher servers. As we will use default values, no further # information is needed modparam("dispatcher", "db_url","mysql://openser:openserrw@10.1.1.50/openser ") #!endif ####### Routing Logic ######## route{ #!ifdef WITH_DISP if( uri=~"sip:.*session=chat.*" ) { xlog("L_DEBUG", "PoC:relaying chat session by Request URI"); ds_select_dst("1", "3"); forward(); } else { xlog("L_DEBUG", "PoC:relaying PoC Session by round-robin"); ds_select_dst("1", "4"); forward(); } #!else xlog("L_DEBUG", "PoC: Relaying by hardcoded route"); t_relay("10.1.1.50", "6666"); #!endif } ~ So the questions are: Does this module need to locate users? Should I include usrloc module iin that sip proxy node in order to be able to locate the users?? Thanks Albert

2 1

Re: [SR-Users] sr-users Digest, Vol 78, Issue 79
by Gary Tabachnik 29 Nov '11

29 Nov '11

On Nov 29, 2011, at 8:55 AM, "sr-users-request(a)lists.sip-router.org" <sr-users-request(a)lists.sip-router.org> wrote: > Send sr-users mailing list submissions to > sr-users(a)lists.sip-router.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > or, via email, send a message with subject or body 'help' to > sr-users-request(a)lists.sip-router.org > > You can reach the person managing the list at > sr-users-owner(a)lists.sip-router.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of sr-users digest..." > > > Today's Topics: > > 1. Re: unable to install utils module (Uri Shacked) > 2. Re: Kamailio v3.2.0 Development Guide published > (Henning Westerholt) > 3. Re: Kamailio v3.2.0 Development Guide published (Gaston Draque) > 4. Re: [Kamailio-Business] Kamailio v3.2.0 Development Guide > published (Santiago Gimeno) > 5. transport=TLS (Bruno Bresciani) > 6. Re: Kamailio v3.2.0 Development Guide published (Javier Gallart) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Tue, 29 Nov 2011 13:39:59 +0200 > From: Uri Shacked <ushacked(a)gmail.com> > Subject: Re: [SR-Users] unable to install utils module > To: miconda(a)gmail.com > Cc: "SIP Router - Kamailio $OpenSER$ and SIP Express Router $SER$ > - Users Mailing List" <sr-users(a)lists.sip-router.org> > Message-ID: > <CAMMbDhR0_cKQ_6n=07P81TR3g_eRCL271OjPOjwWGURS+7XeSQ(a)mail.gmail.com> > Content-Type: text/plain; charset="utf-8" > > works great, thanks. > > On Tue, Nov 29, 2011 at 12:50 PM, Daniel-Constantin Mierla < > miconda(a)gmail.com> wrote: > >> Hello, >> >> looks like you don't have libxml2-dev installed on your system and it is >> needed by pidf.c from utils module. Install that package and try again. >> >> Cheers, >> Daniel >> >> >> On 11/29/11 10:59 AM, Uri Shacked wrote: >> >> i need the utils module... >> >> tried to install it with - "make install-modules modules=modules/utils" >> and got the following: >> " >> make install-modules modules=modules/utils >> CC (gcc) [M utils.so] conf.o >> CC (gcc) [M utils.so] functions.o >> CC (gcc) [M utils.so] pidf.o >> pidf.c:52:27: warning: libxml/parser.h: No such file or directory >> In file included from pidf.c:54: >> pidf.h:41: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before >> ?xmlNodeGetNodeByName? >> pidf.h:43: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before >> ?xmlDocGetNodeByName? >> pidf.h:44: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before >> ?xmlNodeGetChildByName? >> pidf.h:46: error: expected ?)? before ?root? >> pidf.h:48: error: expected ?)? before ?node? >> pidf.c:56: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before >> ?xmlNodeGetAttrByName? >> pidf.c:67: error: expected ?)? before ?node? >> pidf.c:76: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before >> ?xmlNodeGetChildByName? >> pidf.c:87: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before >> ?xmlNodeGetNodeByName? >> pidf.c:106: error: expected ?)? before ?root? >> pidf.c:116: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before >> ?xmlDocGetNodeByName? >> pidf.c:122: error: expected ?)? before ?doc? >> make[1]: *** [pidf.o] Error 1 >> make: *** [install-modules] Error 1 >> " >> >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> >> >> -- >> Daniel-Constantin Mierla -- http://www.asipto.com >> Kamailio Advanced Training, Dec 5-8, Berlin: http://asipto.com/u/kathttp://linkedin.com/in/miconda -- http://twitter.com/miconda >> >> >

1 0

Re: [SR-Users] Kamailio v3.2.0 Development Guide published
by Javier Gallart 29 Nov '11

29 Nov '11

> > > > Daniel, > > This is great! Thanks a lot for it, it will help third-party > contributors enormously. > > -- Alex > > > I agree, excellent document, thank you -though I must say that, not being a developer, I understood very little :) -. Regards Javi

1 0

Kamailio v3.2.0 Development Guide published
by Daniel-Constantin Mierla 29 Nov '11

29 Nov '11

Hello, development guide for Kamailio has been updated for v3.2.0 - it goes through internal components, presenting the APIs for pkg/shm memory, locking/synchronization, config file interpreter, database connectors, a.s.o., as well as guiding how to write a new module. There is a section trying to collect hints about upgrading a module developed for old versions 1.x to newer architecture and APIs in versions 3.x. The tutorial is available online at: * http://asipto.com/u/kdg A mirror is hosted at: * http://www.kamailio.org/docs/kamailio-devel-guide/ This is the first release for v3.2, it may get adjustments in the near future. Among the plans to add new chapters: how to write docbook documentation; TM module API, Presence module API ... Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Dec 5-8, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

6 5

unable to install utils module
by Uri Shacked 29 Nov '11

29 Nov '11

i need the utils module... tried to install it with - "make install-modules modules=modules/utils" and got the following: " make install-modules modules=modules/utils CC (gcc) [M utils.so] conf.o CC (gcc) [M utils.so] functions.o CC (gcc) [M utils.so] pidf.o pidf.c:52:27: warning: libxml/parser.h: No such file or directory In file included from pidf.c:54: pidf.h:41: error: expected ג=ג, ג,ג, ג;ג, גasmג or ג__attribute__ג before גxmlNodeGetNodeByNameג pidf.h:43: error: expected ג=ג, ג,ג, ג;ג, גasmג or ג__attribute__ג before גxmlDocGetNodeByNameג pidf.h:44: error: expected ג=ג, ג,ג, ג;ג, גasmג or ג__attribute__ג before גxmlNodeGetChildByNameג pidf.h:46: error: expected ג)ג before גrootג pidf.h:48: error: expected ג)ג before גnodeג pidf.c:56: error: expected ג=ג, ג,ג, ג;ג, גasmג or ג__attribute__ג before גxmlNodeGetAttrByNameג pidf.c:67: error: expected ג)ג before גnodeג pidf.c:76: error: expected ג=ג, ג,ג, ג;ג, גasmג or ג__attribute__ג before גxmlNodeGetChildByNameג pidf.c:87: error: expected ג=ג, ג,ג, ג;ג, גasmג or ג__attribute__ג before גxmlNodeGetNodeByNameג pidf.c:106: error: expected ג)ג before גrootג pidf.c:116: error: expected ג=ג, ג,ג, ג;ג, גasmג or ג__attribute__ג before גxmlDocGetNodeByNameג pidf.c:122: error: expected ג)ג before גdocג make[1]: *** [pidf.o] Error 1 make: *** [install-modules] Error 1 "

2 2

Kamailio - RTP Proxy problem
by Jonny Klompas 28 Nov '11

28 Nov '11

I am using RTP Proxy with the following command: rtpproxy -l "my_IP" -s udp:localhost:7722 Nov 28 13:04:09 ubuntu /usr/local/kamailio-3.1/sbin/kamailio[2770]: INFO: rtpproxy [rtpproxy.c:1403]: rtp proxy <udp:127.0.0.1:7722> found, support for it re-enabled Nov 28 13:04:09 ubuntu kernel: [ 7015.931001] rtpproxy[2813]: segfault at 0 ip 0805147c sp b6f27cd0 error 4 in rtpproxy[8048000+e000] Nov 28 13:04:10 ubuntu /usr/local/kamailio-3.1/sbin/kamailio[2768]: ERROR: rtpproxy [rtpproxy.c:1491]: can't send command to a RTP proxy Nov 28 13:04:10 ubuntu /usr/local/kamailio-3.1/sbin/kamailio[2768]: ERROR: rtpproxy [rtpproxy.c:1526]: proxy <udp:127.0.0.1:7722> does not respond, disable it Nov 28 13:04:10 ubuntu /usr/local/kamailio-3.1/sbin/kamailio[2768]: ERROR: rtpproxy [rtpproxy.c:2119]: no available proxies Nov 28 13:04:10 ubuntu /usr/local/kamailio-3.1/sbin/kamailio[2770]: ERROR: rtpproxy [rtpproxy.c:1491]: can't send command to a RTP proxy Nov 28 13:04:10 ubuntu /usr/local/kamailio-3.1/sbin/kamailio[2770]: ERROR: rtpproxy [rtpproxy.c:1526]: proxy <udp:127.0.0.1:7722> does not respond, disable it If I trace an attempted call on port 7722, I see there is no activity on port 7722. My firewall is open to port 7722. Thanks Jonny

3 3

How to enable RAW UDP in configuration file
by laura testi 28 Nov '11

28 Nov '11

Hi all, I see the new features in 3.1 mentioning the possibility to use raw udp to boost the performance, but can not find any document on how to enable it in the kamailio.cfg. Could anybody help? Many Thanks! Best Regards, Laura

2 2

"Version" Table missing
by mn3250 mn3250 28 Nov '11

28 Nov '11

Hi, Looks like the table "Version" is missing in the sql create script. 0(18594) DEBUG: db_mysql [km_my_con.c:94]: opening connection: mysql://xxxx:xxxx@localhost/voip1 0(18594) DEBUG: db_mysql [km_my_con.c:121]: connection type is Localhost via UNIX socket 0(18594) DEBUG: db_mysql [km_my_con.c:122]: protocol version is 10 0(18594) DEBUG: db_mysql [km_my_con.c:123]: server version is 5.0.77-log 0(18594) DEBUG: <core> [db_res.c:118]: allocate 48 bytes for result set at 0x2ba27ac440b0 0(18594) DEBUG: db_mysql [km_res.c:68]: 1 columns returned from the query 0(18594) DEBUG: <core> [db_res.c:155]: allocate 8 bytes for result names at 0x2ba27ac44140 0(18594) DEBUG: <core> [db_res.c:165]: allocate 4 bytes for result types at 0x2ba27ac441a8 0(18594) DEBUG: db_mysql [km_res.c:85]: allocate 16 bytes for RES_NAMES[0] at 0x2ba27ac44210 0(18594) DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x2ba27ac44210)[0]=[table_version] 0(18594) DEBUG: db_mysql [km_res.c:100]: use DB1_INT result type 0(18594) DEBUG: db_mysql [km_res.c:169]: no rows returned from the query 0(18594) DEBUG: <core> [db.c:385]: no row for table location found 0(18594) ERROR: <core> [db.c:421]: invalid version 0 for table location found, expected 1004 (check table structure and table "version") Anyone has the correct file? Thanks

2 1

uacreg - need help
by Lucas Alvarez 28 Nov '11

28 Nov '11

Hi, sorry for spamming again but I'm not being able to register to a SIP provider populating the uacreg table, I followed the steps of this guide: http://by-miconda.blogspot.com/2010/10/best-of-new-in-kamailio-310-10.html, but I don't see any change neither any REGISTER packet sent out to the sip provider, is something missing? How can I know if the values from the table are being loaded? What are the fields: "l_uuid" and "l_username" for? Is for routing incoming calls? I would appreciate any kind of help. Regards, Lucas Alvarez

2 3

Access to multivalue responses in ndb_redis
by Javier Gallart 28 Nov '11

28 Nov '11

Hello do you know if it's possible to access the different values in a complex response using ndb_redis? For instance: ZREVRANGE key 0 -1 1) "123456" 2) "123" 3) "1" I'd like to store the first value (123456); is it possible? Thanks in advance Javi

2 1

Re: [SR-Users] sr-users Digest, Vol 78, Issue 62
by Jonny Klompas 27 Nov '11

27 Nov '11

I think AllowOverride All is correctly set, you can see my config below: Contents of my /etc/apache2/sites-available/default file: <VirtualHost *:80> ServerAdmin webmaster@localhost DocumentRoot /var/www <Directory /> Options FollowSymLinks AllowOverride None </Directory> <Directory /var/www/> Options Indexes FollowSymLinks MultiViews AllowOverride All Order allow,deny allow from all </Directory> ScriptAlias /cgi-bin/ /usr/lib/cgi-bin/ <Directory "/usr/lib/cgi-bin"> AllowOverride None Options +ExecCGI -MultiViews +SymLinksIfOwnerMatch Order allow,deny Allow from all </Directory> ErrorLog ${APACHE_LOG_DIR}/error.log # Possible values include: debug, info, notice, warn, error, crit, # alert, emerg. LogLevel warn CustomLog ${APACHE_LOG_DIR}/access.log combined Alias /doc/ "/usr/share/doc/" <Directory "/usr/share/doc/"> Options Indexes MultiViews FollowSymLinks AllowOverride None Order deny,allow Deny from all Allow from 127.0.0.0/255.0.0.0 ::1/128 </Directory> </VirtualHost> Alias /siremis "/var/www/siremis-2.0.0/siremis" <Directory "/var/www/siremis-2.0.0/siremis"> Options Indexes FollowSymLinks MultiViews AllowOverride All Order allow,deny Allow from all <FilesMatch "\.xml$"> Order deny,allow Deny from all </FilesMatch> <FilesMatch "\.inc$"> Order deny,allow Deny from all </FilesMatch> </Directory> My rewrite.load file contains: LoadModule rewrite_module /usr/lib/apache2/modules/mod_rewrite.so ________________________________ From: "sr-users-request(a)lists.sip-router.org" <sr-users-request(a)lists.sip-router.org> To: sr-users(a)lists.sip-router.org Sent: Wednesday, November 23, 2011 6:00 AM Subject: sr-users Digest, Vol 78, Issue 62 Send sr-users mailing list submissions to sr-users(a)lists.sip-router.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users or, via email, send a message with subject or body 'help' to sr-users-request(a)lists.sip-router.org You can reach the person managing the list at sr-users-owner(a)lists.sip-router.org When replying, please edit your Subject line so it is more specific than "Re: Contents of sr-users digest..." Today's Topics: 1. Re: Siremis installation problem (Elena-Ramona Modroiu) ---------------------------------------------------------------------- Message: 1 Date: Wed, 23 Nov 2011 09:15:51 +0100 From: Elena-Ramona Modroiu <ramona(a)asipto.com> Subject: Re: [SR-Users] Siremis installation problem To: sr-users(a)lists.sip-router.org Message-ID: <4ECCABB7.4010009(a)asipto.com> Content-Type: text/plain; charset="iso-8859-1"; Format="flowed" Hi, do you have 'AllowOverride All' in your apache config? Regards, Ramona On 11/22/11 10:13 AM, Jonny Klompas wrote: > I do. > > Thanks > > Jonny > > ------------------------------------------------------------------------ > *From:* Elena-Ramona Modroiu <ramona(a)asipto.com> > *To:* Jonny Klompas <jonny.klompas(a)yahoo.com>; SIP Router - Kamailio > (OpenSER) and SIP Express Router (SER) - Users Mailing List > <sr-users(a)lists.sip-router.org> > *Sent:* Tuesday, November 22, 2011 10:57 AM > *Subject:* Re: [SR-Users] Siremis installation problem > > Hi, > > On 11/20/11 6:47 PM, Jonny Klompas wrote: >> Hi, >> >> When trying to access the Siremis home page to login after the >> installation, I receive the following error: >> >> The requested URL /siremis/index.php/user/login was not found on this >> server. > do you have mod_rewrite enabled on apache web server? > > Regards, > Ramona > > > > _______________________________________________ > 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

2 3

db_mysql -> unhandled data type column
by Stephen Dodge (Bistech) 25 Nov '11

25 Nov '11

Hello, We are using Kamailio 3.1.5 in production. We use the sqlops module to run a mysql function to return parameters. In the logs I am seeing the flowing warnings. /usr/sbin/kamailio[3991]: WARNING: db_mysql [km_res.c:143]: unhandled data type column (diallednumber) type id (6), use DB1_STRING as default /usr/sbin/kamailio[3991]: WARNING: db_mysql [km_res.c:143]: unhandled data type column (ringtime) type id (6), use DB1_STRING as default I think these are generated when we return a null value, although this is not causing us a problem can anyone advise how to eliminate these warnings. Many Thanks Stephen Dodge ________________________________ Information in this message, including any attachments, is confidential to the person to whom it is addressed and may be legally privileged. If you are not the intended recipient please notify the sender and delete the message from your system. Please note that Bistech Group plc, Bistech plc, Bisnet Limited and the sender do not accept any responsibility for viruses. It is your responsibility to check the e-mail and any attachments for viruses. Calls may be monitored and recorded.

2 4

Register Transaction Callback function
by Bruno Bresciani 25 Nov '11

25 Nov '11

Hi All, Why in the kamailio 3.1.2 I have to give a sip_msg parameter to function register_tmcb when I want register a function of my module with TMCB_RESPONSE_OUT parameter? I can't understand this behavior... Best Regards

2 3

OPTIONS Module
by Stephen Dodge (Bistech) 24 Nov '11

24 Nov '11

Hello, I have installed Kamailio 3.1.5 on Centos 5.5 using the kamailio-3.1.5-4.1.x86_64.rpm. I have a requirement that our Kamailio server supports SIP OPTIONS, looking through the module list it looks like the OPTIONS module is the way to go, unfortunately my install only includes the module and module_k Kamailio Modules. Is the only option to compile the OPTIONS module from source or is there a better way to configure Kamailio to respond to SIP OPTIONS? Thanks Stephen Dodge ________________________________ Information in this message, including any attachments, is confidential to the person to whom it is addressed and may be legally privileged. If you are not the intended recipient please notify the sender and delete the message from your system. Please note that Bistech Group plc, Bistech plc, Bisnet Limited and the sender do not accept any responsibility for viruses. It is your responsibility to check the e-mail and any attachments for viruses. Calls may be monitored and recorded.

2 2

configuration
by Rohit Bhatnagar 24 Nov '11

24 Nov '11

I am trying to configure a VOIP phone. I do not know the setting. Alfa Color Wireless VoIP Internet IP Wifi Phone The setting are : Registrar Server IP and Port address Outbound Proxy Server IP and Port address

2 1

Problem to relay sip message recieved port 9090 to asterisk port 7160
by arif tuhin 24 Nov '11

24 Nov '11

I'm Trying to relay sip messages from sip client to asterisk. Kamailio should accept the msg on 9090 udp port and it will forward the sip message to 7160 udp port of Asterisk. I thought the default cfg file will be enough to accomplish this simple task. But kamailio is receiving the msg ok but its not forwarding the msg. Another interesting thing is if i enable tls and give a tls port of kamailio , it easily forwards and the client registers ok. With Best Regards

2 5

gracefuly stop kamailio
by Uri Shacked 24 Nov '11

24 Nov '11

hi, i am trying to deploy a way to gracefuly stop kamailio. the goal is to stop new incoming calls, wait for all active calls to end (i guess dialog module is a must...) and then, shutting down kamailio..... any ideas will be great....

2 1

Re: [SR-Users] simple proxy
by Daniel-Constantin Mierla 23 Nov '11

23 Nov '11

Hello, keep the mailing list cc-ed, private messages from public mailing list discussions are going to be discarded. See the comments inline. On 11/23/11 12:20 PM, Iurii Andamasov wrote: > Hello Daniel, > i > 23.11.2011, в 8:43, Daniel-Constantin Mierla написал(а): > >> Hello, >> >> On 11/23/11 4:02 AM, Iurii Andamasov wrote: >>> Hello list. >>> First of all sorry for dump question, i'm pretty new to kamailio so i hope someone point me in right direction >>> >>> currently i have one sip device which can handle sip request only from one ip, >>> my goal is to put kamailio in front of it so i can proxy requests from several ips >>> >>> i added this to conf >>> >>> route{ >>> if ( src_ip == 10.46.1.121 || src_ip == 10.46.1.122 ) { >>> if(method == "INVITE") { >>> $rd="10.46.29.10"; >>> t_relay(); >>> } >>> } >>> >>> and calls seems connecting but signaling is completely broken >>> >>> i understand that i'm doing something wrong, but can't get what is wrong >> can you explain what is broken and what you would expect to happen? > seems that occurs timeouts, from your next message i understand why, > what i need it just to pass call through kamailio from that 2 ips If you need to route all requests within a call through kamailio, then you need to use loose routing (see rr module). Maybe it would be easier for you to start with an existing example of config, if the default config is too big to understand, maybe the one from dispatcher is simpler: http://kamailio.org/docs/modules/stable/modules_k/dispatcher.html#id2523004 It has the parts of routing requests within dialog, you need to replace the part that uses dispatcher module with your forwarding to a specific ip (or use dispatcher and have that ip as single destination there). Before line 'route(DISPATCH);' add the condition to deny INVITEs that don't match the trusted source ips: if ( ! (src_ip == 10.46.1.121 || src_ip == 10.46.1.122 )) { send_reply("403", "Forbidden"); exit; } Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Dec 5-8, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

2 2

Siremis v2.1.0 Released
by Elena-Ramona Modroiu 23 Nov '11

23 Nov '11

Siremis v2.1.0 has been released - this is an update to previous release v2.0.0, bringing several enhancements and new web pages to manage PUA and RLS. It is still compatible with Kamailio v3.1.x, the last of this kind, next one to be out in the near future will be compatible with Kamailio v3.2.x. You can find the news about this release, including links to download, screenshots and demos, at: * http://siremis.asipto.com/2011/11/23/siremis-v2-1-0-released/ Alternative download site (tarball or git pull) is from sourceforge project: * http://sourceforge.net/projects/siremis/ Regards, Ramona PS. Siremis v2.1.0 is working for most of the components with Kamailio 3.2.0, just the few that changed the database structure may not be fully functional (e.g., the modules with tables that have new columns, see http://www.kamailio.org/wiki/install/upgrade/3.1.x-to-3.2.0#sql_commands).

1 0

releasing v3.2.1
by Daniel-Constantin Mierla 23 Nov '11

23 Nov '11

Hello, I plan to package v3.2.1 after mid of next week (most probably on Thursday) -- it is about 1 month and a half since 3.2.0 was released and branch 3.2 accumulated a bunch of patches. Anyone having comments about it? Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Dec 5-8, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 0

kamailio 3.1.0 crash on ssl-dos attack
by Jijo 23 Nov '11

23 Nov '11

Hi All, Kamailio is resetting when we do TLS renegotiation dos attack using the tool available at http://www.thc.org/thc-ssl-dos/. Anybody looked at this issue? How we could resolve it. Any idea? The core generated for 3 pid's as below Pid 1: Core was generated by `/usr/sbin/kamailio -u swrun -g sw -m 120 -f /etc/kamailio/kamailio.cfg'. Program terminated with signal 11, Segmentation fault. #0 atomic_inc_int () at atomic/atomic_x86.h:225 (gdb) bt #0 atomic_inc_int () at atomic/atomic_x86.h:225 #1 cfg_update_local () at cfg/cfg_struct.h:228 #2 timer_main () at timer.c:994 #3 0x080b0579 in main_loop () at main.c:1632 #4 0x080b1be4 in main (argc=9, argv=0xbfd61e54) at main.c:2446 Pid 2: Core was generated by `/usr/sbin/kamailio -u swrun -g sw -m 120 -f /etc/kamailio/kamailio.cfg'. Program terminated with signal 11, Segmentation fault. #0 0x0819bfe8 in qm_insert_free (qm=0xaf6c5000, p=0xb05eec30, file=0xb6fb4140 "tls: tls_init.c", func=0xb6fb4ce0 "ser_free", line=296) at mem/q_malloc.c:184 184 if (frag->size <= f->size) break; (gdb) bt #0 0x0819bfe8 in qm_insert_free (qm=0xaf6c5000, p=0xb05eec30, file=0xb6fb4140 "tls: tls_init.c", func=0xb6fb4ce0 "ser_free", line=296) at mem/q_malloc.c:184 #1 qm_free (qm=0xaf6c5000, p=0xb05eec30, file=0xb6fb4140 "tls: tls_init.c", func=0xb6fb4ce0 "ser_free", line=296) at mem/q_malloc.c:518 #2 0xb6f95404 in ser_free (ptr=0xb05eec30) at tls_init.c:296 #3 0xb732e9ba in CRYPTO_free (str=0xb05eec30) at mem.c:391 #4 0xb7330bee in int_new_ex_data (class_index=5, obj=0xbfd414f4, ad=0xbfd41574) at ex_data.c:440 #5 0xb7330443 in CRYPTO_new_ex_data (class_index=5, obj=0xbfd414f4, ad=0xbfd41574) at ex_data.c:575 #6 0xb73dfde3 in X509_STORE_CTX_init (ctx=0xbfd414f4, store=0xafd8b3d0, x509=0xafe08ff0, chain=0x0) at x509_vfy.c:2114 #7 0xb74b0f31 in ssl3_output_cert_chain (s=0xb0553a10, x=0xafe08ff0) at s3_both.c:349 #8 0xb74a4728 in ssl3_send_server_certificate (s=0xb0553a10) at s3_srvr.c:3034 #9 0xb74a5879 in ssl3_accept (s=0xb0553a10) at s3_srvr.c:353 #10 0xb74afa8f in ssl3_read_bytes (s=0xb0553a10, type=23, buf=0xb0ad44ec "", len=4095, peek=0) at s3_pkt.c:1266 #11 0xb74ac9c9 in ssl3_read_internal (s=0xb0553a10, buf=0xb0ad44ec, len=4095, peek=0) at s3_lib.c:3265 #12 0xb74c24a9 in SSL_read (s=0xb0553a10, buf=0xb0ad44ec, num=4095) at ssl_lib.c:954 #13 0xb6fad1c3 in tls_read_f (c=0xb0ad431c, flags=0xbfd619c4) at tls_server.c:1058 #14 0x08171c0e in tcp_read_headers (c=0xb0ad431c, read_flags=0xbfd619c4) at tcp_read.c:406 #15 0x08171db8 in tcp_read_req (con=0xb0ad431c, bytes_read=0xbfd619cc, read_flags=0xbfd619c4) at tcp_read.c:885 #16 0x08172f67 in handle_io (fm=<value optimized out>, events=1, idx=<value optimized out>) at tcp_read.c:1234 #17 0x0817583b in io_wait_loop_epoll (unix_sock=89) at io_wait.h:1092 #18 tcp_receive_loop (unix_sock=89) at tcp_read.c:1345 #19 0x0816e2e9 in tcp_init_children () at tcp_main.c:4867 #20 0x080affb1 in main_loop () at main.c:1646 #21 0x080b1be4 in main (argc=9, argv=0xbfd61e54) at main.c:2446 Pid 3: Core was generated by `/usr/sbin/kamailio -u swrun -g sw -m 120 -f /etc/kamailio/kamailio.cfg'. Program terminated with signal 11, Segmentation fault. #0 0xb76c9e7c in memmove () from /lib/libc.so.6 (gdb) bt #0 0xb76c9e7c in memmove () from /lib/libc.so.6 #1 0x081724e7 in tcp_read_req (con=0xb022c8f0, bytes_read=0xbfd619cc, read_flags=0xbfd619c4) at tcp_read.c:1026 #2 0x08172f67 in handle_io (fm=<value optimized out>, events=1, idx=<value optimized out>) at tcp_read.c:1234 #3 0x0817583b in io_wait_loop_epoll (unix_sock=93) at io_wait.h:1092 #4 tcp_receive_loop (unix_sock=93) at tcp_read.c:1345 #5 0x0816e2e9 in tcp_init_children () at tcp_main.c:4867 #6 0x080affb1 in main_loop () at main.c:1646 #7 0x080b1be4 in main (argc=9, argv=0xbfd61e54) at main.c:2446

2 1

The requested URL /siremis/index.php/user/login was not found on this server.
by Jonny Klompas 23 Nov '11

23 Nov '11

Installation was proceeding successfully, until I reached the following stage: http://kb.asipto.com/images/siremis-2.0/siremis20-install05.png Then I receive above error. Thanks Jonny

2 4

Kamailio best performance
by Ricardo Martinez 23 Nov '11

23 Nov '11

Hello list. I’m having some issues with the Kamailio versión 3.2.0. I want to ask if someone could give some hints how to optimize the performance of my kamailio server. For some reason and from time to time the kamailio process start to answer slower than usual, making calls fail and register expires. I’m still unable to detect the problem, but I want to know of maybe i´m running my kamailio not under the best conditions. This is part of my configuration : #!KAMAILIO #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 # ----------- global configuration parameters ------------------------ debug=2 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) log_facility=LOG_LOCAL0 children=16 port=5060 memdbg=9 memlog=9 listen=udp:10.0.10:5060 disable_tcp=yes server_signature=0 port=5060 # ----------------- setting module-specific parameters --------------- ## modparam("registrar", "received_avp", "$avp(s:rcv)") modparam("usrloc", "db_mode", 1) modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("rr", "enable_full_lr", 1) modparam("auth_db|permissions|uri_db|usrloc","db_url","mysql://openser:openserrw@localhost /openser") modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") modparam("avpops", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("avpops", "avp_table", "usr_preferences") modparam("domain", "db_mode", 1) # ----- nathelper params ----- modparam("nathelper", "natping_interval", 20) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) modparam("nathelper","natping_interval", 20) ## modparam("nathelper","received_avp", "$avp(i:42)") modparam("mediaproxy","mediaproxy_socket", "/var/run/mediaproxy/dispatcher.sock") modparam("mediaproxy", "signaling_ip_avp", "$avp(s:signaling_ip)") modparam("registrar|nathelper", "received_avp", "$avp(i:80)") modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") #modparam("tm", "fr_timer", 3) # ------ dialog params ------- modparam("dialog", "dlg_flag", 4) modparam("dialog", "profiles_with_value", "caller") # ------ pike params -------- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 25) modparam("pike", "remove_latency", 4) # ------ rr params -------- modparam("rr", "enable_full_lr", 1) # ----- misc_radius params ----- modparam("misc_radius", "radius_config", "/usr/local/etc/radiusclient-ng/radiusclient.conf") modparam("misc_radius", "caller_service_type", 18) modparam("misc_radius", "callee_service_type", 19) modparam("misc_radius", "caller_extra", "Called-Station-Id=$ru") modparam("misc_radius", "callee_extra", "Called-Station-Id=$fu") # ---- htable param --------- modparam("htable", "htable", "a=>size=8;") modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") modparam("rtimer", "timer", "name=tst;interval=300;mode=1;") modparam("rtimer", "exec", "timer=tst;route=STATS") modparam("sqlops","sqlcon", "ca=>mysql://openser:openserrw@localhost/openser") #------ uac --------------- modparam("uac","rr_store_param","my_param") modparam("uac","from_restore_mode","auto") modparam("uac","auth_realm_avp","$avp(i:10)") modparam("uac","auth_username_avp","$avp(i:11)") modparam("uac","auth_password_avp","$avp(i:12)") Beside of this I have syslogd in asyn mode… This is the info of the kamailio –V version: kamailio 3.2.0 (x86_64/linux) 639f0a 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: 639f0a compiled on 11:35:43 Oct 28 2011 with gcc 4.5.1 I’m using pike to check flood alerts, but I have a white list stored in the “user_preference” table : I’m using it like this : route[REQINIT] { # flood dection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if( !(avp_db_load("pike", "$avp(ip_origen)") && avp_check("$avp(ip_origen)", "eq/$src_ip/gi")) ) #+---------+----------+--------+-----------+----------------+------+---------------------+ #| uuid | username | domain | attribute | value | type | modified | #+---------+----------+--------+-----------+----------------+------+---------------------+ #| pike | | | ip_origen | 10.0.0.44 | 0 | 2008-01-04 13:24:14 | #| pike | | | ip_origen | 10.0.0.66 | 0 | 2008-01-04 17:28:59 | { if($sht(ipban=>$si)!=$null) { # ip is already blocked xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); exit; } if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); xlog("L_INFO","ALERT: pike blocking from $si\n"); $sht(ipban=>$si) = 1; exit; } } Is this check method too slow? Or intensive in mysql access request?? Hope someone can help me here. Thanks in advance.. Regards, *Ricardo Martinez.-*

2 1

simple proxy
by Iurii Andamasov 23 Nov '11

23 Nov '11

Hello list. First of all sorry for dump question, i'm pretty new to kamailio so i hope someone point me in right direction currently i have one sip device which can handle sip request only from one ip, my goal is to put kamailio in front of it so i can proxy requests from several ips i added this to conf route{ if ( src_ip == 10.46.1.121 || src_ip == 10.46.1.122 ) { if(method == "INVITE") { $rd="10.46.29.10"; t_relay(); } } and calls seems connecting but signaling is completely broken i understand that i'm doing something wrong, but can't get what is wrong Please point me to solution Thanks

2 1

strange behavior when trying to reach PSTN gateway via TCP
by Dmitry Petrakoff 23 Nov '11

23 Nov '11

Hi List! I'm running Kamailio for about 1 year without any serious problems. When it works with UDP only, it runs perfectly well for us. Without any major issues. But now we have a customer who wants to connect Lync to our network. Unfortunately, Lync does not support UDP for SIP signalling, so we tried to teach kamailio to send SIP messages via TCP. And getting strange error: 10(36629) DEBUG: tm [t_reply.c:659]: DEBUG: reply sent out. buf=0x832c7ac: SIP/2.0 100 trying -..., shmem=0x28d97c7c: SIP/2.0 100 trying - 10(36629) DEBUG: tm [t_reply.c:669]: DEBUG: _reply_light: finished 10(36629) DEBUG: <core> [forward.c:213]: DEBUG: get_out_socket: socket determined: 0x833ddc0 10(36629) DEBUG: <core> [msg_translator.c:457]: clen_builder: content-length: 553 (553) 10(36629) DEBUG: <core> [msg_translator.c:204]: check_via_address(10.200.245.200, 10.200.245.200, 0) 10(36629) DEBUG: <core> [tcp_main.c:1824]: tcp_send: no open tcp connection found, opening new one 10(36629) DEBUG: <core> [ip_addr.c:247]: tcpconn_new: new tcp connection: 10.198.3.100 10(36629) DEBUG: <core> [tcp_main.c:1081]: tcpconn_new: on port 5060, type 2 10(36629) DEBUG: <core> [tcp_main.c:1382]: tcpconn_add: hashes: 0:0:0, 1 10(36629) ERROR: <core> [tcp_main.c:2748]: connect 10.198.3.100:5060 failed Socket is not connected 10(36629) ERROR: <core> [tcp_main.c:2754]: 10.198.3.100:5060: connect & send for 0x28d988ac failed: Socket is not connected (57) 10(36629) DEBUG: <core> [tcp_main.c:2955]: tcpconn_chld_put: destroying connection 0x28d988ac (1, -1) flags 0060 10(36629) ERROR: tm [../../forward.h:170]: msg_send: ERROR: tcp_send failed We tested two different variants: one when defined Lync in dr_gateways table as 10.198.3.100:5068;transport=tcp And then, tried global options udp_mtu=1300 udp_mtu_try_proto=TCP As soon as these options were enabled, all other gateways became unreachable with the same error as above. Could you please to point me out, where I'm wrong? fbsd-kam# /usr/local/sbin/kamailio -V version: kamailio 3.1.5 (i386/freebsd) ec672a 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, 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, PKG_SIZE 4MB poll method support: poll, select, kqueue. id: ec672a compiled on 11:02:58 Nov 15 2011 with gcc 4.2.1 FreeBSD fbsd-kam.dprs-consulting.com 8.2-RELEASE-p2 FreeBSD 8.2-RELEASE-p2 #0: Sun Jul 3 17:49:13 GMT 2011 root@fbsd-kam.dprs-consulting.com:/usr/obj/usr/src/sys/GENERIC i386 Routing script could be sent on request. Thank you very much! WBR Dimon

3 7

parse 503 message
by Robert R 22 Nov '11

22 Nov '11

Hi, How can I set a filter for receiving 5xx messages, i.e. how can I parse 503 messages received by the proxy? I have tried the following and none works: t_check_status("503") is_method("503") Thanks, R

2 1

(no subject)
by Charles Wang 22 Nov '11

22 Nov '11

http://functioncatering.co.za/modules/mod_wdbanners/blog.php?html143

1 0

LCR weight parameter
by Fabian Borot 22 Nov '11

22 Nov '11

Hello On kamailio 1.5.2, I need to choose the gw from a group according to its weight. After trying unsuccessfully with different values I realized that the help guide says that the load_gw functions can return gateways according to "gateway's randomized weight within its group". So, I have these questions: 1- since it is "randomized", does this mean that the weight parameter actually can not guarantee that the weight value can decide how gateways are ordered? 2- once I have the gws on the avp, is it posible to re-order them? if so, how? thank you very much fborot

2 5

Broadcasting message to users...
by Krishna Kurapati 22 Nov '11

22 Nov '11

Hi, During planned down time of Kamailio, I would like to send a Notice to all the users currently registered on the system. It would be nice if it appears in their Chat windows. I am thinking if I could loop through user_location table and send MESSAGE to individual user. What's the best way to accomplish this task? Thanks Krish Kura

3 3

Re: [SR-Users] Core compiled with Sctp support
by Daniel-Constantin Mierla 22 Nov '11

22 Nov '11

Don't forget to keep the mailing list cc-ed -- private emails sent from mailing lists discussion are going to be discarded. If you have libsctp devel package installed, try compilation with: make SCTP=1 ... all Cheers, Daniel On 11/18/11 12:13 PM, Stefan Ljung wrote: > > So, its not. > > [test032@wslint90]# ./ser -V > > version: ser 3.1.2 (i386/linux) 2fe4d6 > > 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, PKG_SIZE 4MB > > poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. > > id: 2fe4d6 > > compiled on 11:31:57 Nov 18 2011 with gcc 4.1.2 > > This result came after an untar and then make command. > > Here's my machine : > > [linux]# uname -a > > Linux wslint90 2.6.23.1-21.fc7 #1 SMP Thu Nov 1 21:09:24 EDT 2007 i686 > i686 i386 GNU/Linux > > What can I do to force it in ? > > /Stefan > > > ------------------------------------------------------------------------ > *From:* Daniel-Constantin Mierla [mailto:miconda@gmail.com] > *Sent:* den 18 november 2011 11:59 > *To:* SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List > *Cc:* Stefan Ljung > *Subject:* Re: [SR-Users] Core compiled with Sctp support > > Hello, > > when sctp support is available, if you do 'kamailio -V' or 'ser -V' > you should see USE_SCTP flag there. > > Not sure if sctp is enabled by default in listen sockets, you can do > it anyhow with global parameter: > > enable_sctp=yes > > Cheers, > Daniel > > On 11/18/11 11:49 AM, Stefan Ljung wrote: >> I've downloaded sip-router 3.1.2 and have been struggling with the >> make commands for configuring. Compiling and installing. >> Can't really say that I master it. >> But I have finally managed to do all steps and started a example ser.cfg. >> One of the features I want to play around eith is the sctp transport. >> From previous experiments with OpenSER I kearned that if I only >> specify a listen host address without transport protocol all three >> UDP TCP and SCTP sockets will be started. >> Ths time I only see UDP and TCP: >> Listening on >> udp: 127.0.0.1:5060 >> tcp: 127.0.0.1:5060 >> Aliases: >> tcp: localhost:5060 >> tcp: localhost.localdomain:5060 >> udp: localhost:5060 >> udp: localhost.localdomain:5060 >> I have read the c sources to verify that the sctp libs and header >> files are in the right places, and they are. >> What I could need is a tip on how to verify that my core has been >> compiled with sctp support. How can I see that ? >> In what files would there be a confimation of that the sctp feature >> will be compiled in? >> /Stefan >> >> >> _______________________________________________ >> 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 > Kamailio Advanced Training, Dec 5-8, Berlin:http://asipto.com/u/kat > http://linkedin.com/in/miconda -- http://twitter.com/miconda -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Dec 5-8, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

2 3

Trouble with SERVICE
by jermaine.boss＠gmail.com 22 Nov '11

22 Nov '11

Greetings: I am mailing around trouble with the product attaching service to a STANDARD sip phone outfit. My account user name is: call.8675309 The problem presents it self in a manner after login attemps where now a message around a NAT Router Cone problem exist. Please have the service to the account reset or otherwise further advise me. Sincerly J. Boss

2 1

How long (in bytes) is an AVP which reads an Integer from DB?
by Iñaki Baz Castillo 22 Nov '11

22 Nov '11

Hi, I need to read from a MySQL table a 32 bits long Integer column. My question is: how many bytes does Kamailio use for storing integer values fetched from a DB query? does it depend on the 32/64 bits architecture? Thanks a lot. -- Iñaki Baz Castillo <ibc(a)aliax.net>

2 2

How to
by Sébastien Boutet 22 Nov '11

22 Nov '11

Hello, In the features of kamailio, I can see 'replication for High Availability (HA)' But I can't find any tutorials for HA install. Could you help me or did you have any book wich explain how to do that ? Thank's Best reagard, Sébastien BOUTET **

2 1

New module: xhttp_rpc
by Ovidiu Sas 21 Nov '11

21 Nov '11

Hello all, A new module providing a web interface to kamailio/sip-router RPC interface is available in trunk: http://kamailio.org/docs/modules/devel/modules/xhttp_rpc.html http://sip-router.org/docbook/sip-router/branch/master/modules/xhttp_rpc/xh… The module is using the embedded web server provided by xhttp module: http://kamailio.org/docs/modules/devel/modules/xhttp.html A simple config that will enable the web interface is provide on the README file. Simply point your browser to http://<server_IP>:<TCP_port>/<xhttp_rpc_root> and start browsing available RPC commands. For more info about the module please see the README file. Regards, Ovidiu Sas -- VoIP Embedded, Inc. http://www.voipembedded.com

3 3

kamctl lcr
by Jonny Klompas 21 Nov '11

21 Nov '11

HI, When I try and run this on ver 3.2 I receive the following error: ERROR: command disabled How do I install or configure lcr? Thanks Jonny

3 3

New module: presence_profile
by MÉSZÁROS Mihály 21 Nov '11

21 Nov '11

Hello All, I've committed a new module named 'presence_profile' which provides UA config provisioniong according RFC6080. http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=commitdiff;h=cd… The README describes the module function: http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=blob_plain;f=mo… The modul is very very simple in it's current stage. The modul added only the event 'ua-profile' handling. Best Regards, Mihály Mészáros

2 1

GIT committing and backporting
by Daniel-Constantin Mierla 21 Nov '11

21 Nov '11

Hello, I put together a wiki page to help people porting patches between branches, exemplified with master and 3.2 branches -- the typical patch backporting these days: * http://www.kamailio.org/wiki/devel/backporting-to-3.2.x They are useful also for people with no write access, in case they want to try patches committed to devel branch in their local git repo clones. In additions, I added to kamailio wiki the page with guidelines for commits, trying to give the generic suggestions of how to do it in order to keep the coherence of the development: * http://www.kamailio.org/wiki/devel/git-commit-guidelines Updates and enhancements are welcome! If one has more suggestions or want to debate some of the proposed guidelines, feel free to start the discussion on the mailing list. Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Dec 5-8, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 0

Limiting simultaneous calls
by Mino Haluz 21 Nov '11

21 Nov '11

Hi, I was using cdrtool (prepaid table) and callcontrol to limit concurrent calls. In fact this is only limiting the outbound calls, but I would like to use another mechanism which should limit the inbound calls too. So basically to limit voice channels. So is there some reliable method/module how to achieve this? Thanks, Mino

3 5

Dropped registrar bindings with 1.5.3
by Alex Balashov 18 Nov '11

18 Nov '11

I have a 1.5.3 installation functioning as a registrar that is exhibiting a very curious, if only infrequent set of behaviours. The host is CentOS 5.x, with PostgreSQL 9.0 backing usrloc, and db_mode = 1 (immediate write-through). I have a registration handler in my initial request route that begins with a log message: ... else if(is_method("REGISTER")) { xlog("L_INFO", "... Processing REGISTER from $si:$sp for AOR $tu\n"); route(2); exit; } I've got a registrant who consistently re-registers with an expiration time of 120, so in theory they ought to be re-registering every two minutes or so. Most of the time they do. But, occasionally, I'll see a gap of 5-8 minutes go by, in which the above log message does not appear, and their registration expires so that they can't receive calls. That's not what's interesting. What's interesting is that I did a packet capture on the proxy and caught one of these scenarios. Within that ~8 minute window when the registration expired and calls were failing, the UAC actually re-registered several times, on normal ~2 minute boundaries, AND the proxy successfully challenged the request and sent a 200 OK indicating that the binding was stored! I double-checked all the parameters to ensure that the 401s and 200 OKs corresponded to the right REGISTER flow, etc. - yes, it's all correct. Is there any imaginable set of circumstances or a known bug that would cause Kamailio to successfully authenticate and affirm a registration request, while neither logging its receipt, nor, evidently, storing it to the database? I can envision a number of database failure modes that would account for the binding not actually being in the 'location' table, but that doesn't explain why the request wouldn't even be logged. That's what has me baffled. Thanks! -- Alex -- 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/

6 15

[Kamailio-Users] Converting rtpproxy recorded .rtp file to .wav file
by Vikram Ragukumar 18 Nov '11

18 Nov '11

Hello, I used Kamailio+rtpproxy to record a session and rtpproxy outputs the following files long_file_name.a.rtp, long_file_name.a.rtcp, long_file_name.o.rtp, long_file_name.o.rtcp http://www.rtpproxy.org/wiki/RTPproxy/FAQ From the Rtpproxy FAQ above, i tried to extract the audio using rtpbreak and sox. rtpbreak -W -r long_file_name.a.rtp rtpbreak -W -r long_file_name.o.rtp The above commands generate rtp.0.0.raw, rtp.1.0.raw. Then when i run sox using sox --combine merge -r 8k -A rtp.0.0.raw -r 8k -A rtp.1.0.raw -t wavpcm -s out.wav i get the following errors : sox: invalid option -- - sox: -c must be given a number Is there a switch/anything else that i am missing ? Thanks in advance, Regards, Vikram.

3 5

sqlops crash when connection does not exist
by Juha Heinanen 18 Nov '11

18 Nov '11

i misspelled connection argument in sql_query and got crash: Nov 10 17:19:15 sip /usr/sbin/sip-proxy[4182]: ERROR: sqlops [sqlops.c:266]: invalid connection [sip-proxy] Nov 10 17:19:15 sip /usr/sbin/sip-proxy[4182]: ERROR: <core> [route.c:1216]: fixing failed (code=-1) at cfg:/etc/sip-proxy/sip-proxy.cfg:281 Nov 10 17:19:15 sip kernel: [ 990.129668] sip-proxy[4182]: segfault at 786e6570 ip b769de38 sp bf960ffc error 4 in libc-2.11.2.so[b762b000+140000] (gdb) where #0 0xb769de38 in strcmp () from /lib/i686/cmov/libc.so.6 #1 0x080f54f9 in find_mod_export_record (mod=0xb750bb80 "db_mysql", name=0xb720d353 "db_bind_api", param_no=0, flags=0, mod_if_ver=0xbf96108c) at sr_module.c:657 #2 0x080f5756 in find_mod_export (mod=0xb750bb80 "db_mysql", name=0xb720d353 "db_bind_api", param_no=0, flags=0) at sr_module.c:722 #3 0xb720449c in db_bind_mod (mod=0xb6628944, mydbf=0x8) at db.c:209 #4 0xb66240d2 in ht_db_init_con () at ht_db.c:75 #5 0xb661b144 in destroy () at htable.c:225 #6 0x080f3684 in destroy_modules () at sr_module.c:782 #7 0x0809087c in cleanup (show_status=0) at main.c:564 #8 0x08091509 in shutdown_children (show_status=0, sig=<value optimized out>) at main.c:706 #9 0x080940f7 in main (argc=17, argv=0xbf961354) at main.c:2523 -- juha

2 3

Core compiled with Sctp support
by Stefan Ljung 18 Nov '11

18 Nov '11

I've downloaded sip-router 3.1.2 and have been struggling with the make commands for configuring. Compiling and installing. Can't really say that I master it. But I have finally managed to do all steps and started a example ser.cfg. One of the features I want to play around eith is the sctp transport. >From previous experiments with OpenSER I kearned that if I only specify a listen host address without transport protocol all three UDP TCP and SCTP sockets will be started. Ths time I only see UDP and TCP: Listening on udp: 127.0.0.1:5060 tcp: 127.0.0.1:5060 Aliases: tcp: localhost:5060 tcp: localhost.localdomain:5060 udp: localhost:5060 udp: localhost.localdomain:5060 I have read the c sources to verify that the sctp libs and header files are in the right places, and they are. What I could need is a tip on how to verify that my core has been compiled with sctp support. How can I see that ? In what files would there be a confimation of that the sctp feature will be compiled in? /Stefan

2 1

Problem trying to play recorded message before call
by David Candamil Santos 18 Nov '11

18 Nov '11

Hi there, guys. I hope you can help me. I use kamailio to comunicate some servers (each one with its own kamailio). When an user calls another, depending on the userID that will receive the call, the server redirects the call to the corresponding kamailio server. It works properly. Now, what I would like to do is to configure it so it plays a notification saying where the call is going to be forwarded. After reading some tutorials, I tried to get it working with rtpproxy, but after configuring it, it doesn't play the clips. I am testing it with a *.wav file. The path to the file is "/usr/local/etc/kamailio/test.wav" (it's in the kamailio directory). This is my software: Debian stable kernel 2.6.32-5-686 Kamailio 3.2.0 (GIT installation) rtpproxy 1.2.1-1 Kamailio and rtpproxy are running in the same machine. These are the log messages: -------------------kamailio.log------------------- (SEVERAL TIMES) Nov 16 12:58:10 debian-virtualbox kamailio[2076]: INFO: rtpproxy [rtpproxy.c:1415]: rtp proxy <udp:127.0.0.1:22222> found, support for it enabled (SEVERAL TIMES) Nov 16 12:58:12 debian-virtualbox kamailio[2069]: ERROR: <script>: Intentando reproducir audio -------------------------------------------------- This is the relevant configuration: --------------/etc/default/rtpproxy--------------- # The control socket. #CONTROL_SOCK="unix:/var/run/rtpproxy/rtpproxy.sock" # To listen on an UDP socket, uncomment this line: CONTROL_SOCK=udp:127.0.0.1:22222 # Additional options that are passed to the daemon. EXTRA_OPTS="-l 127.0.0.1" -------------------------------------------------- -----/usr/local/etc/kamailio/kamailio.cfg (just my important changes for this action)--------- loadmodule "rtpproxy.so" modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:22222") request_route { rtpproxy_offer(); xlog("L_ERR","Intentando reproducir audio"); rtpproxy_stream2uas("/usr/local/etc/kamailio/test.wav","1"); (...default routes ommited...) route(RELAY); } -------------------------------------------------- I can play it with mplayer, so it's not a sound problem (or even a system codec problem). I would like to know if I can make it working with rtpproxy, or if there is an easier way to do it. Thanks for your time. (I include now the whole kamailio.cfg, just in case you need to check anything) ---------/usr/local/etc/kamailio/kamailio.cfg (complete) ---------- #!KAMAILIO # # Kamailio (OpenSER) SIP Server v3.2 - default configuration script # - web: http://www.kamailio.org # - git: http://sip-router.org # # Direct your questions about this file to: <sr-users(a)lists.sip-router.org> # # Refer to the Core CookBook at http://www.kamailio.org/dokuwiki/doku.php # for an explanation of possible statements, functions and parameters. # # Several features can be enabled using '#!define WITH_FEATURE' directives: # # *** To run in debug mode: # - define WITH_DEBUG # ##-- # WITH_DEBUG # *** To enable mysql: # - define WITH_MYSQL # ##-- #!define WITH_MYSQL # *** To enable authentication execute: # - enable mysql # - define WITH_AUTH # - add users using 'kamctl' # ##-- #!define WITH_AUTH # *** To enable IP authentication execute: # - enable mysql # - enable authentication # - define WITH_IPAUTH # - add IP addresses with group id '1' to 'address' table # # *** To enable persistent user location execute: # - enable mysql # - define WITH_USRLOCDB # # *** To enable presence server execute: # - enable mysql # - define WITH_PRESENCE # # *** To enable nat traversal execute: # - define WITH_NAT # - install RTPProxy: http://www.rtpproxy.org # - start RTPProxy: # rtpproxy -l _your_public_ip_ -s udp:localhost:7722 # ##-- #WITH_NAT # *** To enable PSTN gateway routing execute: # - define WITH_PSTN # - set the value of pstn.gw_ip # - check route[PSTN] for regexp routing condition # # *** To enable database aliases lookup execute: # - enable mysql # - define WITH_ALIASDB # # *** To enable speed dial lookup execute: # - enable mysql # - define WITH_SPEEDDIAL # # *** To enable multi-domain support execute: # - enable mysql # - define WITH_MULTIDOMAIN # # *** To enable TLS support execute: # - adjust CFGDIR/tls.cfg as needed # - define WITH_TLS # # *** To enable XMLRPC support execute: # - define WITH_XMLRPC # - adjust route[XMLRPC] for access policy # # *** To enable anti-flood detection execute: # - adjust pike and htable=>ipban settings as needed (default is # block if more than 16 requests in 2 seconds and ban for 300 seconds) # - define WITH_ANTIFLOOD # # *** To block 3XX redirect replies execute: # - define WITH_BLOCK3XX # # *** To enable VoiceMail routing execute: # - define WITH_VOICEMAIL # - set the value of voicemail.srv_ip # - adjust the value of voicemail.srv_port # # *** To enhance accounting execute: # - enable mysql # - define WITH_ACCDB # - add following columns to database #!ifdef ACCDB_COMMENT ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; #!endif ####### Defined Values ######### # *** Value defines - IDs used later in config #!ifdef WITH_MYSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!define DBURL "mysql://openser:openserrw@localhost/openser" #!endif #!ifdef WITH_MULTIDOMAIN # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 #!else #!define MULTIDOMAIN 0 #!endif # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 ####### Global Parameters ######### #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL0 fork=yes children=4 /* uncomment the next line to disable TCP (default on) */ #disable_tcp=yes /* uncomment the next line to disable the auto discovery of local aliases based on reverse DNS on IPs (default on) */ #auto_aliases=no /* add local domain aliases */ #alias="sip.mydomain.com" /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ #listen=udp:10.0.0.10:5060 /* port to listen to * - can be specified more than once if needed to listen on many ports */ port=5060 #$banana_server="192.168.0.70" #$treasure_server="192.168.0.60" #$drake_server="192.168.0.60" #!ifdef WITH_TLS enable_tls=yes #!endif # life time of TCP connection when there is no traffic # - a bit higher than registration expires to cope with UA behind NAT tcp_connection_lifetime=3605 ####### Custom Parameters ######### # These parameters can be modified runtime via RPC interface # - see the documentation of 'cfg_rpc' module. # # Format: group.id = value 'desc' description # Access: $sel(cfg_get.group.id) or @cfg_get.group.id # #!ifdef WITH_PSTN # PSTN GW Routing # # - pstn.gw_ip: valid IP or hostname as string value, example: # pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" # # - by default is empty to avoid misrouting pstn.gw_ip = "" desc "PSTN GW Address" #!endif #!ifdef WITH_VOICEMAIL # VoiceMail Routing on offline, busy or no answer # # - by default Voicemail server IP is empty to avoid misrouting voicemail.srv_ip = "" desc "VoiceMail IP Address" voicemail.srv_port = "5060" desc "VoiceMail Port" #!endif ####### Modules Section ######## # set paths to location of modules (to sources or installation folders) #!ifdef WITH_SRCPATH mpath="modules_k:modules" #!else mpath="/usr/local/lib/kamailio/modules_k/:/usr/local/lib/kamailio/modules/" #!endif #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif ##-- loadmodule "sdpops.so" loadmodule "textopsx.so" loadmodule "rtpproxy.so" ##- loadmodule "mi_fifo.so" loadmodule "kex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "mi_rpc.so" loadmodule "acc.so" #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #!ifdef WITH_IPAUTH loadmodule "permissions.so" #!endif #!endif #!ifdef WITH_ALIASDB loadmodule "alias_db.so" #!endif #!ifdef WITH_SPEEDDIAL loadmodule "speeddial.so" #!endif #!ifdef WITH_MULTIDOMAIN loadmodule "domain.so" #!endif #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_NAT loadmodule "nathelper.so" loadmodule "rtpproxy.so" #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_ANTIFLOOD loadmodule "htable.so" loadmodule "pike.so" #!endif #!ifdef WITH_XMLRPC loadmodule "xmlrpc.so" #!endif #!ifdef WITH_DEBUG loadmodule "debugger.so" #!endif # ----------------- setting module-specific parameters --------------- ##-- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:22222") # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- registrar params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ #modparam("registrar", "max_contacts", 10) # max value for expires of registrations modparam("registrar", "max_expires", 3600) # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default ww do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") modparam("auth_db", "use_domain", MULTIDOMAIN) # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- speedial params ----- #!ifdef WITH_SPEEDDIAL modparam("speeddial", "db_url", DBURL) modparam("speeddial", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) # use caching modparam("domain", "db_mode", 1) # register callback to match myself condition with domains list modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:22222") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/usr/local/etc/kamailio/tls.cfg") #!endif #!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 16) modparam("pike", "remove_latency", 4) # ----- htable params ----- # ip ban htable with autoexpire after 5 minutes modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") #!endif #!ifdef WITH_XMLRPC # ----- xmlrpc params ----- modparam("xmlrpc", "route", "XMLRPC"); modparam("xmlrpc", "url_match", "^/RPC") #!endif #!ifdef WITH_DEBUG # ----- debugger params ----- modparam("debugger", "cfgtrace", 1) #!endif ####### Routing Logic ######## # Main SIP request routing logic # - processing of any incoming SIP request starts with this route # - note: this is the same as route { ... } request_route { #start_recording(); #force_rtp_proxy(); #rtpproxy_offer(); #rtpproxy_stream2uas("/usr/local/etc/kamailio/test.wav", "1"); #("fichero","numero de veces"); #formato alaw? #force_rtp_proxy(); rtpproxy_offer(); xlog("L_ERR","Intentando reproducir audio"); rtpproxy_stream2uas("/usr/local/etc/kamailio/test.wav","1"); if ($rU=~"^001788[0-3].*") { xlog("L_ERR", "Banana island"); $rd="192.168.0.70"; msg_apply_changes(); xlog("L_ERR","Redireccionando a $rd\n"); } if ($rU=~"^001788[4-5].*") { xlog("L_ERR", "Big treasure island"); $rd="192.168.0.60"; msg_apply_changes(); xlog("L_ERR","Redireccionando a $rd\n"); } if ($rU=~"^001788[6-7].*") { xlog("L_ERR", "Drake island"); $rd="192.168.0.60"; msg_apply_changes(); xlog("L_ERR","Redireccionando a $rd\n"); } if ($rU=~"^001788[8-9].*") { xlog("L_ERR", "Lost island"); sdp_keep_codecs_by_name("GSM"); #if para si true, bien, y si false mensaje y sale } # per request initial checks route(REQINIT); # NAT detection route(NATDETECT); # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); # authentication route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } # dispatch requests to foreign domains route(SIPOUT); ### requests for my local domains # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations to PSTN route(PSTN); # user location service route(LOCATION); ##-- #rtpproxy_offer(); #rtpproxy_stream2uas("/usr/local/etc/kamailio/test.wav", "1"); route(RELAY); } route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|SUBSCRIBE")) { t_on_branch("MANAGE_BRANCH"); t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood dection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($sht(ipban=>$si)!=$null) { # ip is already blocked xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); exit; } if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); $sht(ipban=>$si) = 1; exit; } } #!endif if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } if ( is_method("ACK") ) { # ACK is forwarded statelessy route(NATMANAGE); } route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } 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; } } sl_send_reply("404","Not here"); } exit; } } # Handle SIP registrations route[REGISTRAR] { if (is_method("REGISTER")) { if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging ## setbflag(FLB_NATSIPPING); } if (!save("location")) sl_reply_error(); exit; } } # USER location service route[LOCATION] { #!ifdef WITH_SPEEDIAL # search for short dialing - 2-digit extension if($rU=~"^[0-9][0-9]$") if(sd_lookup("speed_dial")) route(SIPOUT); #!endif #!ifdef WITH_ALIASDB # search in DB-based aliases if(alias_db_lookup("dbaliases")) route(SIPOUT); #!endif $avp(oexten) = $rU; if (!lookup("location")) { $var(rc) = $rc; route(TOVOICEMAIL); t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } } # Presence server route route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; }; if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); exit; } return; } # Authentication route route[AUTH] { #!ifdef WITH_AUTH if (is_method("REGISTER")) { # authenticate the REGISTER requests (uncomment to enable auth) if (!www_authorize("$td", "subscriber")) { www_challenge("$td", "0"); exit; } if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else { #!ifdef WITH_IPAUTH if(allow_source_address()) { # source IP allowed return; } #!endif # authenticate if from local subscriber if (from_uri==myself) { if (!proxy_authorize("$fd", "subscriber")) { proxy_challenge("$fd", "0"); exit; } if (is_method("PUBLISH")) { if ($au!=$fU || $au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } if ($au!=$rU) { sl_send_reply("403","Forbidden R-URI"); exit; } #!ifdef WITH_MULTIDOMAIN if ($fd!=$rd) { sl_send_reply("403","Forbidden R-URI domain"); exit; } #!endif } else { if ($au!=$fU) { sl_send_reply("403","Forbidden auth ID"); exit; } } consume_credentials(); # caller authenticated } else { # caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (!uri==myself) { sl_send_reply("403","Not relaying"); exit; } } } #!endif return; } # Caller NAT detection route route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { fix_nated_contact(); } setflag(FLT_NATS); } #!endif return; } # RTPProxy control route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; rtpproxy_manage(); if (is_request()) { if (!has_totag()) { add_rr_param(";nat=yes"); } } if (is_reply()) { if(isbflagset(FLB_NATB)) { fix_nated_contact(); } } #!endif return; } # Routing to foreign domains route[SIPOUT] { if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(RELAY); } } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN rotuing enabled but pstn.gw_ip not defined\n"); return; } # route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) return; # only local users allowed to call if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); route(RELAY); exit; #!endif return; } # XMLRPC routing #!ifdef WITH_XMLRPC route[XMLRPC] { # allow XMLRPC from localhost if ((method=="POST" || method=="GET") && (src_ip==127.0.0.1)) { # close connection only for xmlrpclib user agents (there is a bug in # xmlrpclib: it waits for EOF before interpreting the response). if ($hdr(User-Agent) =~ "xmlrpclib") set_reply_close(); set_reply_no_connect(); dispatch_rpc(); exit; } send_reply("403", "Forbidden"); exit; } #!endif # route to voicemail server route[TOVOICEMAIL] { #!ifdef WITH_VOICEMAIL if(!is_method("INVITE")) return; # check if VoiceMail server IP is defined if (strempty($sel(cfg_get.voicemail.srv_ip))) { xlog("SCRIPT: VoiceMail rotuing enabled but IP not defined\n"); return; } if($avp(oexten)==$null) return; $ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip) + $sel(cfg_get.voicemail.srv_port); route(RELAY); exit; #!endif return; } # manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); route(NATMANAGE); } # manage incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") route(NATMANAGE); } # manage failure routing cases failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_is_canceled()) { exit; } #!ifdef WITH_BLOCK3XX # block call redirect based on 3xx replies. if (t_check_status("3[0-9][0-9]")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_VOICEMAIL # serial forking # - route to voicemail on busy or no answer (timeout) if (t_check_status("486|408")) { route(TOVOICEMAIL); exit; } #!endif } --------------------------------------------------------------

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Re: [SR-Users] mtree on db Berkeley DB?
by Javier Gallart 17 Nov '11

17 Nov '11

> > Hello > > coming back to the topic related to mtree, to be able to set values via > mi/rpc -- it won't be that difficult to add such functionality. Usually > with a tree is mainly reading, due to fast matching on tree indexing. > The question is how many times/often do you need to change values and > how many of them at the same time (more or less). > > I was also thinking about that: our application works in such a way that the full tree is recalculated every 15 minutes. Currently we have aound 60 branches with 150000 entries each. > I assume many times the changes will be somewhere down the tree, since > the first part of the number is usually the same (e.g., country code and > operator prefix). To update the tree at runtime, while there are reads > on it, there must be used a lock to be safe an consistent. If you do lot > of writes and very often, then you keep the tree structure locked a lot, > slowing the search. > That's the case, tname is rarely updated, it's tvalue the column that we normally update. > > Can you estimate the number of writes and how often they would be on a > daily basis? There might be other solutions to look at, if writes are > very often. > >From the numbers above, let's say that we need to update around 8M records every 15 minutes (we expect this number will keep steadily increasing...) As a side note, I've looked at how to implement the mt_match equivalent in redis and it does't look that hard using kamailio s.prefixed transformation (as you suggested) and redis sorted sets. I'll need to make more tests to check the performance, I'll share the results. Thanks! Javi > > Cheers, > Daniel > > > > Regards > > > > Javi >

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Re: [SR-Users] Problem with NAT and RTPProxy
by David Candamil Santos 16 Nov '11

16 Nov '11

Hi, I'm just using it because after reading lots of manuals and tutorials, it was the solution I thought it could work. The smartphone is outside the network. I am trying to use it over the Vodafone network. What do you suggest then? Thanks. 2011/11/14 Pavel Segeč <Pavel.Segec(a)fri.uniza.sk> > Hi, **** > > ** ** > > from your topology description I'm not sure why you are trying to use > RTPproxy when all your equipments are on private segments. RTPproxy should > have an public IP address**** > > ** ** > > palo**** > > >

2 1

mtree on db Berkeley DB?
by Javier Gallart 16 Nov '11

16 Nov '11

Hi list we've been happily using the mtree module for months now. Lately the size of the tree has grown a lot. The mtree table needs to be fully repopulated and reloaded several times a day, and we are looking for a fastest mechanism (for populating the table, I guess the reload time does not depend much on the db backend...). Does anyone tried with Berkeley DB? Is this combination mtree-berkeley actually feasible...? Thanks Javi

7 19

tls-modules in 3.2.0
by Mokhtar Bengana 15 Nov '11

15 Nov '11

Hello, I am trying to install the tls module in 3.2.0. I run apt-get update before apt-get install kamailio-tls-modules and still can't load the tls module and getting the following error. Kamailio can't find the tls module even though it's installed? I didn't get any errors when I run apt-get install kamailio-tls-modules. Everything else is defined in the config file. Has someone tried to install tls in the latest release and is having issues? Any help much appreciated. #!define WITH_TLS #!ifdef WITH_TLS enable_tls=yes #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/etc/kamailio/tls.cfg") #!endif root@Kamalio:/usr/local/lib/kamailio/modules_k# kamailio -c loading modules under /usr/local/lib/kamailio/modules_k/:/usr/local/lib/kamailio/modules/ 0(12131) ERROR: <core> [sr_module.c:553]: ERROR: load_module: could not find module <tls> in </usr/local/lib/kamailio/modules_k/:/usr/local/lib/kamailio/modules/> 0(12131) : <core> [cfg.y:3501]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 268, column 12-19: failed to load module 0(12131) ERROR: <core> [modparam.c:162]: set_mod_param_regex: No module matching <tls> found 0(12131) : <core> [cfg.y:3504]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 422, column 50: Can't set module parameter ERROR: bad config file (2 errors) root@Kamalio:/usr/local/lib/kamailio/modules_k# root@Kamalio:/usr/local/lib/kamailio/modules_k# apt-get install kamailio-tls-modules Reading package lists... Done Building dependency tree Reading state information... Done The following NEW packages will be installed: kamailio-tls-modules 0 upgraded, 1 newly installed, 0 to remove and 3 not upgraded. Need to get 0 B/399 kB of archives. After this operation, 848 kB of additional disk space will be used. Selecting previously deselected package kamailio-tls-modules. (Reading database ... 30034 files and directories currently installed.) Unpacking kamailio-tls-modules (from .../kamailio-tls-modules_3.2.0+squeeze1_i386.deb) ... Setting up kamailio-tls-modules (3.2.0+squeeze1) ... Thank you,

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Fwd: [Sip-implementors] FOSDEM 2012 Telephony/Communications Devroom Call for Presenters
by marius zbihlei 15 Nov '11

15 Nov '11

-------- Original Message -------- Subject: [Sip-implementors] FOSDEM 2012 Telephony/Communications Devroom Call for Presenters Date: Mon, 14 Nov 2011 20:28:53 +0100 From: Kevin P. Fleming <kpfleming(a)digium.com> To: sip-implementors(a)lists.cs.columbia.edu <sip-implementors(a)lists.cs.columbia.edu> Greetings, This is a call for presenters (with presentations!) for the telephony and communications devroom at FOSDEM 2012 - http://www.fosdem.org/. We will be holding a day full of presentations on development topics in the area of open source telephony and communications on Sunday, February 5th. The schedule allows for presentations from 9:00 to 17:00. The room we will be using will have a projector, wifi, and 100 seats. Please submit all proposals no later than 2011-12-09. Notification of accepted speakers will be provided by 2011-12-16. We will then work to have a schedule finalized by 2012-01-06. Talks should be submitted by subscribing to and then posting to the telephony-devroom mailing list hosted on http://lists.fosdem.org/. If you would like to contact the devroom organizer directly, please contact Kevin P. Fleming<kpfleming at digium.com>. Feel free to forward this along to any people or mailing lists that you think would be interested in this event. Thank you! -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kfleming(a)digium.com | SIP: kpfleming(a)digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com& www.asterisk.org _______________________________________________ Sip-implementors mailing list Sip-implementors(a)lists.cs.columbia.edu https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors

1 0

Re: [SR-Users] Compile kamailio modules only
by Daniel-Constantin Mierla 14 Nov '11

14 Nov '11

Hello, I think the problem was a comma at the end of sipcapture module name in kstandard group inside the Makefile. If you pull the branch 3.2, it should get the fix. The master was fixed by Ovidiu Sas few minutes ago when adding xhttp_rpc module in that list. Let us know if works now. Thanks, Daniel On 11/14/11 7:01 PM, Pavel Segeč wrote: > Hello, > > I'm trying to install homer capture agent on my Kamailio server. However for > the sipcapture module there is not precompiled package for debian 32 bit > neither 64 bit. Therefore I would like make module deb package and install > it after. > > I'm doing following steps: > > git clone --depth 1 git://git.sip-router.org/sip-router kamailio > cd kamailio/ > git checkout -b 3.2 origin/3.2 > make FLAVOUR=kamailio include_modules="sipcapture" cfg > cd pkg/kamailio/deb/ > mv debian debian-orig > ln -s lenny debian > cd ../../.. > make deb > > finaly after compilation I still have following packages only, > > kamailio > kamailio-postgres-modules_3.2.0_amd64.deb > kamailio_3.2.0_amd64.changes > kamailio-presence-modules_3.2.0_amd64.deb > kamailio_3.2.0_amd64.deb > kamailio-purple-modules_3.2.0_amd64.deb > kamailio_3.2.0.dsc > kamailio-python-modules_3.2.0_amd64.deb > kamailio_3.2.0.tar.gz > kamailio-radius-modules_3.2.0_amd64.deb > kamailio-berkeley-modules_3.2.0_amd64.deb > kamailio-snmpstats-modules_3.2.0_amd64.deb > kamailio-carrierroute-modules_3.2.0_amd64.deb > kamailio-sqlite-modules_3.2.0_amd64.deb > kamailio-cpl-modules_3.2.0_amd64.deb > kamailio-tls-modules_3.2.0_amd64.deb > kamailio-ldap-modules_3.2.0_amd64.deb > kamailio-unixodbc-modules_3.2.0_amd64.deb > kamailio-lua-modules_3.2.0_amd64.deb > kamailio-utils-modules_3.2.0_amd64.deb > kamailio-memcached-modules_3.2.0_amd64.deb > kamailio-xml-modules_3.2.0_amd64.deb > kamailio-mysql-modules_3.2.0_amd64.deb > kamailio-xmlrpc-modules_3.2.0_amd64.deb > kamailio-nth_3.2.0_amd64.deb > kamailio-xmpp-modules_3.2.0_amd64.deb > kamailio-perl-modules_3.2.0_amd64.deb > > without sipcapture module. I've tried to manually edit module.lst, manually > edit pkg/kamailio/deb/debian/rules, without results, still the same. What > I'm doing bad? > > thanks > > palo > > > > _______________________________________________ > 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 Kamailio Advanced Training, Dec 5-8, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 1

TOS = 0
by marion deley 14 Nov '11

14 Nov '11

Hi, I'm trying to set tos value to zero but kamailio doesn't accept it. In fact, it's DSCP field which interest me but i can't set zero. Can you explain me why is it forbidden and how can i do that ? Thanks a lot Marion

3 2

Problem with NAT and RTPProxy
by David Candamil Santos 14 Nov '11

14 Nov '11

Hi, I am using Kamailio 3.2.0 (x86_64/linux) and since some days ago I am trying to call from a network with NAT to outside. This is the configuration: Softphone (192.168.0.5) <--> Kamailio (192.168.0.3) <--> Router (192.168.0.1) <--> Softphone over smartphone All the router ports are opened and redirected to 192.168.0.3. I have installed rtpproxy 1.2.1-1 with the following configuration: --------------------------/etc/defaults/rtpproxy-------------------------------- # The control socket. #CONTROL_SOCK="unix:/var/run/rtpproxy/rtpproxy.sock" # To listen on an UDP socket, uncomment this line: CONTROL_SOCK=udp:127.0.0.1:22222 # Additional options that are passed to the daemon. EXTRA_OPTS="-l candamil.dyndns.org" ---------------------------------------------------------------------------------------- and works properly. This is the log message: ---------------------------------------------------------------------------------------------- Nov 12 20:09:13 condor kamailio[7001]: INFO: rtpproxy [rtpproxy.c:1415]: rtp proxy <udp:127.0.0.1:22222> found, support for it enabled ----------------------------------------------------------------------------------------------- Both softphones are "linphone". The configuration of the network softphone is the following: SIP identity: sip:1001@192.168.0.3 SIP proxy: sip:192.168.0.3 Direct connection to the Internet. In the softphone of the smartphone I set as proxy and domain " candamil.dyndns.org", the DNS address of my router IP. In this case, these are the symptoms: Both softphones can autentify correctly. When calling, both softphones ring, but when I answer, the voice signal is not received. This is the log error: ---------------------------------------------------------------------------------------------- Nov 12 20:23:14 condor kamailio[6991]: ERROR: rtpproxy [rtpproxy.c:2260]: incorrect port 0 in reply from rtp proxy ---------------------------------------------------------------------------------------------- The same happens if I set in the network softphone that it's behind NAT, and I set the router IP. The same happens if I set in the network softphone that it's behind NAT and with a STUN server (stunserver.org). In the three cases, in the softphone over smartphone, the caller is 1001(a)192.168.0.3, and in the network softphone, the caller is 1002(a)candamil.dyndns.org. If I set in the network softphone the proxy as sip:candamil.dyndns.org, the same happens. If what I do is setting the SIP identity as sip:1001@candamil.dyndns.org, when calling form inside to ouside the network, the softphone doesn't know that the call was answered, and there is the following error in the log: ---------------------------------------------------------------------------------------------------------- Nov 12 20:53:00 condor kamailio[7306]: ERROR: <core> [parser/parse_via.c:2600]: ERROR: parse_via: invalid port number <5060ranch=z9hG4bKc 50f.b4825246.0> Nov 12 20:53:00 condor kamailio[7306]: ERROR: <core> [parser/parse_via.c:2629]: ERROR: parse_via on: <SIP/2.0/UDP 192.168.0.3:5060ranch=z 9hG4bKc50f.b4825246.0;received=87.223.138.84#015#012Via: SIP/2.0/UDP 87.223.138.84:5060;rport=5060;branch=z9hG4bK1021772993#015#012From: <sip:1001@candamil.dyndns.org>;tag=783852345#015#012To: < sip:1002@candamil.dyndns.org>#015#012Call-ID: 1644787160#015#012CSeq: 21 INVITE# 015#012User-Agent: Linphone/3.4.0 (eXosip2/unknown)#015#012Content-Length: 0#015#012#015#012> ------------------------------------------------------------------------------------------------------------ If the call is from outside to inside the network, it happens the same than in the previous cases. This is the relevant kamailio configuration: -----------------------------------------kamailio.cfg---------------------------------------------------- #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_NAT ####### Defined Values ######### # *** Value defines - IDs used later in config #!ifdef WITH_MYSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!define DBURL "mysql://openser:openserrw@localhost/openser" #!endif #!ifdef WITH_MULTIDOMAIN # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 #!else #!define MULTIDOMAIN 0 #!endif # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 ####### Global Parameters ######### /* uncomment the next line to disable the auto discovery of local aliases based on reverse DNS on IPs (default on) */ #auto_aliases=no /* add local domain aliases */ alias="candamil.dyndns.org" /* port to listen to * - can be specified more than once if needed to listen on many ports */ port=5060 #!ifdef WITH_TLS enable_tls=yes #!endif # life time of TCP connection when there is no traffic # - a bit higher than registration expires to cope with UA behind NAT tcp_connection_lifetime=3605 ####### Custom Parameters ######### #!ifdef WITH_PSTN # PSTN GW Routing # # - pstn.gw_ip: valid IP or hostname as string value, example: # pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" # # - by default is empty to avoid misrouting pstn.gw_ip = "" desc "PSTN GW Address" #!endif ####### Modules Section ######## #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #!ifdef WITH_NAT loadmodule "nathelper.so" loadmodule "rtpproxy.so" #!endif # ----------------- setting module-specific parameters --------------- # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") modparam("auth_db", "use_domain", MULTIDOMAIN) #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:22222") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif ####### Routing Logic ######## # Main SIP request routing logic # - processing of any incoming SIP request starts with this route # - note: this is the same as route { ... } request_route { # per request initial checks route(REQINIT); # NAT detection route(NATDETECT); # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); # authentication route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } # dispatch requests to foreign domains route(SIPOUT); ### requests for my local domains # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations to PSTN route(PSTN); # user location service route(LOCATION); route(RELAY); } route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|SUBSCRIBE")) { t_on_branch("MANAGE_BRANCH"); t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood dection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($sht(ipban=>$si)!=$null) { # ip is already blocked xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); exit; } if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); $sht(ipban=>$si) = 1; exit; } } #!endif if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } if ( is_method("ACK") ) { # ACK is forwarded statelessy route(NATMANAGE); } route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } 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; } } sl_send_reply("404","Not here"); } exit; } } # Handle SIP registrations route[REGISTRAR] { if (is_method("REGISTER")) { if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging setbflag(FLB_NATSIPPING); } if (!save("location")) sl_reply_error(); exit; } } # USER location service route[LOCATION] { #!ifdef WITH_SPEEDIAL # search for short dialing - 2-digit extension if($rU=~"^[0-9][0-9]$") if(sd_lookup("speed_dial")) route(SIPOUT); #!endif #!ifdef WITH_ALIASDB # search in DB-based aliases if(alias_db_lookup("dbaliases")) route(SIPOUT); #!endif $avp(oexten) = $rU; if (!lookup("location")) { $var(rc) = $rc; route(TOVOICEMAIL); t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } } # Presence server route route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; }; if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); exit; } return; } # Authentication route route[AUTH] { #!ifdef WITH_AUTH if (is_method("REGISTER")) { # authenticate the REGISTER requests (uncomment to enable auth) if (!www_authorize("$td", "subscriber")) { www_challenge("$td", "0"); exit; } if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else { #!ifdef WITH_IPAUTH if(allow_source_address()) { # source IP allowed return; } #!endif # authenticate if from local subscriber if (from_uri==myself) { if (!proxy_authorize("$fd", "subscriber")) { proxy_challenge("$fd", "0"); exit; } if (is_method("PUBLISH")) { if ($au!=$fU || $au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } if ($au!=$rU) { sl_send_reply("403","Forbidden R-URI"); exit; } #!ifdef WITH_MULTIDOMAIN if ($fd!=$rd) { sl_send_reply("403","Forbidden R-URI domain"); exit; } #!endif } else { if ($au!=$fU) { sl_send_reply("403","Forbidden auth ID"); exit; } } consume_credentials(); # caller authenticated } else { # caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (!uri==myself) { sl_send_reply("403","Not relaying"); exit; } } } #!endif return; } # Caller NAT detection route route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { fix_nated_contact(); } setflag(FLT_NATS); } #!endif return; } # RTPProxy control route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; rtpproxy_manage(); if (is_request()) { if (!has_totag()) { add_rr_param(";nat=yes"); } } if (is_reply()) { if(isbflagset(FLB_NATB)) { fix_nated_contact(); } } #!endif return; } # Routing to foreign domains route[SIPOUT] { if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(RELAY); } } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN rotuing enabled but pstn.gw_ip not defined\n"); return; } # route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) return; # only local users allowed to call if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); route(RELAY); exit; #!endif return; } # XMLRPC routing #!ifdef WITH_XMLRPC route[XMLRPC] { # allow XMLRPC from localhost if ((method=="POST" || method=="GET") && (src_ip==127.0.0.1)) { # close connection only for xmlrpclib user agents (there is a bug in # xmlrpclib: it waits for EOF before interpreting the response). if ($hdr(User-Agent) =~ "xmlrpclib") set_reply_close(); set_reply_no_connect(); dispatch_rpc(); exit; } send_reply("403", "Forbidden"); exit; } #!endif # route to voicemail server route[TOVOICEMAIL] { #!ifdef WITH_VOICEMAIL if(!is_method("INVITE")) return; # check if VoiceMail server IP is defined if (strempty($sel(cfg_get.voicemail.srv_ip))) { xlog("SCRIPT: VoiceMail rotuing enabled but IP not defined\n"); return; } if($avp(oexten)==$null) return; $ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip) + $sel(cfg_get.voicemail.srv_port); route(RELAY); exit; #!endif return; } # manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); route(NATMANAGE); } # manage incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") route(NATMANAGE); } # manage failure routing cases failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_is_canceled()) { exit; } #!ifdef WITH_BLOCK3XX # block call redirect based on 3xx replies. if (t_check_status("3[0-9][0-9]")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_VOICEMAIL # serial forking # - route to voicemail on busy or no answer (timeout) if (t_check_status("486|408")) { route(TOVOICEMAIL); exit; } #!endif } -------------------------------------------------------------------------------------------------------------- I hope you can help me. Thanks for your time.

2 1

Can the SBC (Kamailio 3.1.x and FreeSWITCH 1.0.6+) do Upper Registration?
by edson.gomes.leme 14 Nov '11

14 Nov '11

2 1

SnmpStats problem
by Adam Kuśmirek 14 Nov '11

14 Nov '11

Hi All I've compiled kamailio 3.2.0 with snmpstats module. When trying to start the module I received the following message: 0(28197) DEBUG: <core> [sr_module.c:515]: load_module: trying to load </usr/lib/kamailio/modules_k/ snmpstats.so> 0(28197) ERROR: <core> [sr_module.c:523]: ERROR: load_module: could not open module </usr/lib/kamailio/modules_k/snmpstats.so>: /usr/lib/libnetsnmpagent.so.10: undefined symbol: boot_DynaLoader I run it on CentOS 5.7 After digging Internet, I found the solution for the problem. I had to modify Makefile in snmpstats module's directory and recompile it. I had to change BUILDAGENTLIBS env param before change: BUILDAGENTLIBS =-L$(LOCALBASE)/lib -lnetsnmpmibs -lnetsnmpagent \ -lnetsnmphelpers -lnetsnmp after change: BUILDAGENTLIBS =-L$(LOCALBASE)/lib -lnetsnmpmibs -lnetsnmpagent \ -lnetsnmphelpers -lnetsnmp \ -Wl,-E -Wl,-rpath,/usr/lib/perl5/5.8.8/i386-linux-thread-multi/CORE \ -L/usr/local/lib /usr/lib/perl5/5.8.8/i386-linux-thread-multi/auto/DynaLoader/DynaLoader.a \ -L/usr/lib/perl5/5.8.8/i386-linux-thread-multi/CORE \ -lperl -lresolv -lnsl -ldl -lm -lcrypt -lutil -lpthread -lc \ -lwrap \ -lsensors \ -lrpmdb -lrpm After that I successfully started snmpstats module. I don't know why I didn't have compilation errors before adding this linking info, but got problem when trying to run compiled module. Can you explain it ? Do you know another way to solve the problem ? Regards Adam

2 1

a2billing
by roman dmytriv 11 Nov '11

11 Nov '11

Hello folks, Any one tried using a2billing with Kamailio and Asterisk instead of Siremis? Can you share your experiens? Regards roman

3 6

tracking ACC for a 486
by Sven Knoblich 10 Nov '11

10 Nov '11

Hi, I have a short question regarding a tm callback. I want to track a local generated ACK during a cfa. scenario: A calls B with a cfa to C B response with a 486 Busy this will be confirmed with an ACK. I saw, that TMCB_ACK_NEG_IN will only be called when a new transaction is created (t_lookup.c:t_newtran). Unfortunately in this case an ACK will be send directly when a reply will be received (t_reply.c:reply_received). Any ideas? thanks in advance, Sven

2 2

segmentation fault in init_mod_child function
by Bruno Bresciani 10 Nov '11

10 Nov '11

Hi All, Kamailio 3.1.2 terminated with signal 11, segmentation fault, in the init_mod_child function at sr_module.c file. This problem occurs only when I try loading the loadbalance.so module, it was written by me. Someone Can help me to understand why this segmentation fault was generated? There are two other module written by me and this problem doesn't happen. Kamailio's trace: 9(27616) DEBUG: db_postgres [km_pg_con.c:80]: PQsetdbLogin(0x8db2878) 9(27616) DEBUG: <core> [sr_module.c:828]: DEBUG: init_mod_child (5): sl 9(27616) DEBUG: <core> [sr_module.c:828]: DEBUG: init_mod_child (5): auth_db 9(27616) DEBUG: <core> [db.c:294]: connection 0x82f4108 found in pool 9(27616) DEBUG: <core> [sr_module.c:828]: DEBUG: init_mod_child (5): usrloc 9(27616) DEBUG: <core> [sr_module.c:828]: DEBUG: init_mod_child (5): mi_fifo 9(27616) DEBUG: <core> [sr_module.c:828]: DEBUG: init_mod_child (5): registrar 9(27616) DEBUG: <core> [sr_module.c:828]: DEBUG: init_mod_child (5): nathelper 9(27616) DEBUG: <core> [sr_module.c:828]: DEBUG: init_mod_child (5): permissions 9(27616) DEBUG: <core> [sr_module.c:828]: DEBUG: init_mod_child (5): loadbalance 9(27616) DEBUG: loadbalance [loadbalance_mod.c:271]: loadbalance: child started! (5) 9(27616) DEBUG: <core> [sr_module.c:828]: DEBUG: init_mod_child (5): route 0(27582) ALERT: <core> [main.c:741]: child process 27585 exited by a signal 11 0(27582) ALERT: <core> [main.c:744]: core was generated 0(27582) INFO: <core> [main.c:756]: INFO: terminating due to SIGCHLD 8(27615) INFO: <core> [main.c:807]: INFO: signal 15 received 8(27615) DEBUG: <core> [main.c:818]: Memory status (pkg): 8(27615) DEBUG: fm_status: fm_status (0x82aefc0): 8(27615) DEBUG: fm_status: heap size= 4194304 8(27615) DEBUG: fm_status: used= 236720, used+overhead=264064, free=3930240 8(27615) DEBUG: fm_status: max used (+overhead)= 299824 8(27615) DEBUG: fm_status: dumping free list: 8(27615) DEBUG: fm_status: hash = 1 fragments no.: 27, unused: 0 bucket size: 8 - 8 (first 8) 8(27615) DEBUG: fm_status: hash = 44 fragments no.: 1, unused: 0 bucket size: 352 - 352 (first 352) 8(27615) DEBUG: fm_status: hash = 74 fragments no.: 1, unused: 0 bucket size: 592 - 592 (first 592) 8(27615) DEBUG: fm_status: hash = 77 fragments no.: 1, unused: 0 bucket size: 616 - 616 (first 616) 8(27615) DEBUG: fm_status: hash = 81 fragments no.: 1, unused: 0 bucket size: 648 - 648 (first 648) 8(27615) DEBUG: fm_status: hash = 84 fragments no.: 1, unused: 0 bucket size: 672 - 672 (first 672) 8(27615) DEBUG: fm_status: hash = 113 fragments no.: 51, unused: 0 bucket size: 904 - 904 (first 904) 8(27615) DEBUG: fm_status: hash = 2056 fragments no.: 1, unused: 0 gdb bt full trace: Reading symbols from /media/sda1/local/kamailio/lib/kamailio/modules/route.so...done. Loaded symbols for /home2/local/kamailio/lib/kamailio/modules/route.so Reading symbols from /media/sda1/local/kamailio/lib/kamailio/modules/rtpproxy.so...done. Loaded symbols for /home2/local/kamailio/lib/kamailio/modules/rtpproxy.so Reading symbols from /media/sda1/local/kamailio/lib/kamailio/modules/tls.so...done. Loaded symbols for /home2/local/kamailio/lib/kamailio/modules/tls.so Reading symbols from /media/sda1/local/kamailio/lib/kamailio/modules/perms_db.so...done. Loaded symbols for /home2/local/kamailio/lib/kamailio/modules/perms_db.so Reading symbols from /lib/libnss_files.so.2...done. Loaded symbols for /lib/libnss_files.so.2 Reading symbols from /lib/libnss_dns.so.2...done. Loaded symbols for /lib/libnss_dns.so.2 Core was generated by `/home2/local/kamailio/sbin/kamailio -P /var/run/kamailio.pid'. Program terminated with signal 11, Segmentation fault. #0 init_mod_child (m=0xa238, rank=2) at sr_module.c:827 827 sr_module.c: Arquivo ou diretório não encontrado. in sr_module.c (gdb) (gdb) (gdb) (gdb) (gdb) (gdb) (gdb) bt full #0 init_mod_child (m=0xa238, rank=2) at sr_module.c:827 No locals. #1 0x00000000 in ?? () No symbol table info available. (gdb) Best Regards

2 2

How to limit number of calls in Kamailio proxy
by Austin Einter 10 Nov '11

10 Nov '11

This is continuation with limiting number of calls per Kamailio proxy server. I modified kamailio.cfg as below. 1. loadmodule "dialog.so" 2. modparam("dialog", "enable_stats", 1) 3. modparam("dialog", "dlg_flag", 4) Next, as dialog module I can use either active_dialogs() or $DLG_count to find number of calls present at any given point of time. However if I use active_dialogs() I get parse error when I run kamailio proxy. Next I used $DLG_count , but always I am getting value of $DLG_count as 0. In *route* section, I added code as below. if (is_method("INVITE")) { if($DLG_count > 1) { sl_send_reply("503","RESOURCE UNAVAILABLE"); exit; } else { xlog("Dialog count = $DLG_count\n"); } } Everytime a new call starts , it prints $DLG_count as 0. Kindly let me know whats going wrong here. Regards Austin

5 15

Re: [SR-Users] Kamailio Dispatcher fails in front of FreeSwitch, help (again).
by Henrik Aagaard Sørensen 09 Nov '11

09 Nov '11

Hi Daniel. I just wanted to hear if you have had any time to look at the Kamailio Dispatcher <-> FreeSwitch problem? This is NOT to rush you. I've followed the mailing list and can see how busy you are and how much involved you are in all this, so I fully respect all the help you give to this mailing list etc. Best regards, Henrik 2011/10/20 Henrik Aagaard Sørensen <henrikaagaardsorensen(a)gmail.com> > Hi Daniel. > > I've just upgraded to 3.2.0. > > This is my NEW kamailio.cfg: http://pastebin.com/XzXjf1VY > This is my dispatcher.list: http://pastebin.com/9dLS8243 > > I've also attached a ngrep where it errors with "Too many hops" with the > new version 3.2.0. > > Once again, I really appreciate all the help you give! It's remarkable > with your effort in this. > > Best regards, > Henrik > > > 2011/10/17 Daniel-Constantin Mierla <miconda(a)gmail.com> > >> I will try to look over the sip traces you sent, although it is tight >> schedule for the 3.2.0 right now -- so it may take few days. >> >> Cheers, >> Daniel >> >> >> On 10/14/11 10:07 PM, Henrik Aagaard Sørensen wrote: >> >> Yes, pbx.my-domain.com is pointing to the IP of FreeSwitch. >> sip.my-domain.com is the domain name for Kamailio. >> >> 2011/10/14 Daniel-Constantin Mierla <miconda(a)gmail.com> >> >>> >>> >>> On 10/14/11 7:06 PM, Henrik Aagaard Sørensen wrote: >>> >>> My dispatcher.list currently just contains: >>> 1 pbx.my-domain.com:5060 >>> >>> Is pbx.my-domain.com pointing to the IP address of freeswitch? What is >>> the domain name for kamailio ip address? >>> >>> Daniel >>> >>> >>> >>> I will provide a ngrep trace within an hour or two. Thank you so much >>> for helping me. I really hope to get this solved. >>> >>> 2011/10/14 Daniel-Constantin Mierla <miconda(a)gmail.com> >>> >>>> Hello, >>>> >>>> >>>> On 10/14/11 4:48 PM, Henrik Aagaard Sørensen wrote: >>>> >>>> Hi Daniel (and everyone else) again. >>>> >>>> I've just tried to add: >>>> loadmodule "path.so" >>>> modparam("path", "use_received", 1) >>>> >>>> and: >>>> add_path_received() >>>> just before the "if(!t_relay())..." in the ROUTE part. >>>> >>>> That made both devices get the error "Too many loops". >>>> >>>> what is the content of the dispatcher list file? Maybe is better to >>>> use ds_select_domain(...) and ds_next_domain() -- cannot say it since I >>>> haven't seen full sip trace, but maybe the problem is that freeswitch is >>>> sending back if the r-uri address is kamailio's address. >>>> >>>> Cheers, >>>> Daniel >>>> >>>> >>>> >>>> Moving the add_path_received() into the part: >>>> if (nat_uac_test("19")) { if (method=="REGISTER") { >>>> fix_nated_register(); add_path_received(); >>>> made the same issues as before loading the path.so module. >>>> >>>> 2011/10/14 Henrik Aagaard Sørensen <henrikaagaardsorensen(a)gmail.com> >>>> >>>>> Hi Daniel (and everyone else). >>>>> >>>>> This is a tshark trace (is that as good as ngrep?) from Kamailio when >>>>> there are too many loops: http://pastebin.com/yRjduB0s >>>>> In this trace I got the note that 253 packets was dropped (and >>>>> therefor not in the trace). >>>>> >>>>> And this is a trace when it times out: http://pastebin.com/KT0bJzEU >>>>> >>>>> I've actually looked at the PATH module, and tried several things. And >>>>> I know this type of question is rather newbee-like, but can you help me out >>>>> with the PATH module and how to integrate it correctly? >>>>> >>>>> I really do appreciate all the help! >>>>> >>>>> >>>>> 2011/10/14 Daniel-Constantin Mierla <miconda(a)gmail.com> >>>>> >>>>>> Hello, >>>>>> >>>>>> when using the load balancer in front of the registrar, you have to >>>>>> use PATH extensions for SIP (see path module in kamailio) in order to be >>>>>> able to cope with nat. >>>>>> >>>>>> Then, what would be helpful, will be a ngrep trace taken on kamailio >>>>>> server for failed calls. >>>>>> >>>>>> Cheers, >>>>>> Daniel >>>>>> >>>>>> >>>>>> On 10/14/11 4:13 PM, Henrik Aagaard Sørensen wrote: >>>>>> >>>>>> I've asked a lot of question about Kamailio Dispatcher in front of >>>>>> FreeSwitch in this mailing list. So I'm sorry that I'm still "spamming", >>>>>> but it still fails. >>>>>> >>>>>> This is my status so far. >>>>>> >>>>>> When connecting 2 devices directly to FreeSwitch everything works. >>>>>> When connecting to Kamailio, some calls fails with "Timeout" and some fails >>>>>> with "Too many loops". >>>>>> >>>>>> This is my kamailio.cfg: http://pastebin.com/KEuZmikh >>>>>> >>>>>> When connected directly to FreeSwitch where everything works, this is >>>>>> how my clients are registrered in FreeSwitch: >>>>>> http://pastebin.com/prkjkb7g >>>>>> >>>>>> When connected to Kamailio where nothing works, this is how my >>>>>> clients are registrered in FreeSwitch: http://pastebin.com/b5x6NZyv >>>>>> >>>>>> 215.197.145.132 (sip.my-domain.com) is my Kamailio. >>>>>> pbx.my-domain.com is my FreeSwitch. >>>>>> 96.224.14.164 is my IP at home, where my 2 devices are from. >>>>>> >>>>>> Can anyone help me out? I'm kinda lost at this point. >>>>>> >>>>>> >>>>>> _______________________________________________ >>>>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>>>>> >>>>>> >>>>>> -- >>>>>> Daniel-Constantin Mierla -- http://www.asipto.com >>>>>> Kamailio Advanced Training, Dec 5-8, Berlin: http://asipto.com/u/kathttp://linkedin.com/in/miconda -- http://twitter.com/miconda >>>>>> >>>>>> >>>>> >>>> >>>> >>>> _______________________________________________ >>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>>> >>>> >>>> -- >>>> Daniel-Constantin Mierla -- http://www.asipto.com >>>> Kamailio Advanced Training, Dec 5-8, Berlin: http://asipto.com/u/kathttp://linkedin.com/in/miconda -- http://twitter.com/miconda >>>> >>>> >>> >>> >>> _______________________________________________ >>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>> >>> >>> -- >>> Daniel-Constantin Mierla -- http://www.asipto.com >>> Kamailio Advanced Training, Dec 5-8, Berlin: http://asipto.com/u/kathttp://linkedin.com/in/miconda -- http://twitter.com/miconda >>> >>> >> >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> >> >> -- >> Daniel-Constantin Mierla -- http://www.asipto.com >> Kamailio Advanced Training, Dec 5-8, Berlin: http://asipto.com/u/kathttp://linkedin.com/in/miconda -- http://twitter.com/miconda >> >> >

2 2

uacreg issue
by Lucas Alvarez 09 Nov '11

09 Nov '11

Hi, after following the steps of this guide: http://by-miconda.blogspot.com/2010/10/best-of-new-in-kamailio-310-10.html, poulating the uacreg table, loading the uac module and restarting kamailio I didn't see any REGISTER packet sent out to the sip provider, is something missing? How can I know if the values from the table are being loaded? I would appreciate any kind of help. Regards, Lucas Alvarez

1 0

negative www_authorize result codes
by Juha Heinanen 09 Nov '11

09 Nov '11

auth_db/README tells this about www_authorize() negative result codes: Negative codes may be interpreted as follows: * -1 (generic error) - some generic error occurred and no reply was sent out; * -2 (invalid password) - valid user, but wrong password; * -3 (invalid user) - authentication user does not exist. whereas in auth_radius i return: Negative result codes may be interpreted as follows: * -7 (internal error) - some internal error occurred (see syslog); * -6 (nonce reused) - nonce is used more than once; * -5 (no credentials) - credentials were not found in request; * -4 (stale nonce) - stale nonce; * -3 (authorization failed) - RADIUS responded with Access Reject which may be, for example, due to user not found or wrong password; * -1 (error) - some error occured during authorization (see syslog); since both modules depend on the same auth module, why does www_authorize return the full set of negative result codes or is its README out of date? -- juha

2 2

uacreg question
by Lucas Alvarez 08 Nov '11

08 Nov '11

Hi, I have properly populate the table "uacreg" as says here: http://by-miconda.blogspot.com/2010/10/best-of-new-in-kamailio-310-10.html and I'm loading the uac module, is there anything I could be missing? Because I'm not seeing any register packet being sent to my sip provider. Thanks in advance, Lucas Alvarez

2 3

how to give integer avp name in pseudo variable?
by Juha Heinanen 08 Nov '11

08 Nov '11

i have not been able to figure out, how to give integer avp name in a pseudo variable. test $var(name) = "i:1000"; $avp($var(name)) = "value"; xlog("L_INFO", "avp i:1000 has value <$avp(i:1000)>\n"); xlog("L_INFO", "avp s:i1000 has value <$avp(s:i:1000)>\n"); produces to syslog Nov 8 13:17:23 sip /usr/sbin/sip-proxy[8642]: INFO: avp i:1000 has value <<null>> Nov 8 13:17:23 sip /usr/sbin/sip-proxy[8642]: INFO: avp s:i1000 has value <value> that is, $var(name) value i:1000 is interpreted as string name even when its syntax conforms to integer name. is this is a bug or how should value of $var(name) look like? -- juha

2 3

how to give integer avp name in pseudo variable?
by Juha Heinanen 08 Nov '11

08 Nov '11

i have not been able to figure out, how to give integer avp name in a pseudo variable. test $var(name) = "i:1000"; $avp($var(name)) = "value"; xlog("L_INFO", "avp i:1000 has value <$avp(i:1000)>\n"); xlog("L_INFO", "avp s:i1000 has value <$avp(s:i:1000)>\n"); produces to syslog Nov 8 13:17:23 sip /usr/sbin/sip-proxy[8642]: INFO: avp i:1000 has value <<null>> Nov 8 13:17:23 sip /usr/sbin/sip-proxy[8642]: INFO: avp s:i1000 has value <value> that is, $var(name) value i:1000 is interpreted as string name even when its syntax conforms to integer name. is this is a bug or how should value of $var(name) look like? -- juha

1 0

Conditional uacreg
by Alex Balashov 08 Nov '11

08 Nov '11

Is there some way by which credentials can be loaded into the uacreg table, but not all of them used at once? The problem I have is a need to make these registrations active and inactive periodically, whereas the registration facility that the uac module provides is fully managed/largely unattended. Using $uac_req(...) is not an option because I need to capture the failure replies (which is not supported in TM) in order to do digest challenge authentication. -- 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/

3 7

SQL Server
by Jonny Klompas 08 Nov '11

08 Nov '11

Can Kamailio send routes to and /or receive routes from SQL Server? Is there any way for Kamailio to connect to SQL Server in general? Thanks Jonny

3 2

TOS=0 in kamailio.cfg
by DELEY Marion 07 Nov '11

07 Nov '11

Hi, I'm trying to set tos value to zero but kamailio doesn't accept it. In fact, it's DSCP field which interest me but i can't set zero. Can you explain me why is it forbidden and how can i do that ? Thanks a lot Marion

1 0

Re: [SR-Users] Fwd: Kamailio as SBC / LB Media-Servers farm
by Klaus Darilion 05 Nov '11

05 Nov '11

On 04.11.2011 10:57, Sammy Govind wrote: > SORRY for PM but this is not getting through to the users List due to size. I have cc-ed the list. > > ---------- Forwarded message ---------- > From: *Sammy Govind* <govoiper(a)gmail.com <mailto:govoiper@gmail.com>> > Date: Fri, Nov 4, 2011 at 1:55 PM > Subject: Re: [SR-Users] Kamailio as SBC / LB Media-Servers farm > To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List" <sr-users(a)lists.sip-router.org > <mailto:sr-users@lists.sip-router.org>> > > > Klaus: mhomed did work indeed - no need to force_send_socket() now :) Thanks > > I'm progressing abit. > > RTP-Proxy was run in bridged mode like this. > > *rtpproxy -l 109.XX.XXX.212/192.168.2.29 <http://192.168.2.29> -s > udp:localhost:22222 -F -d DBUG* See description of i,e flag: http://kamailio.org/docs/modules/3.2.x/modules/rtpproxy.html#id2535618 Thus in your setup 109.... is the "internal" interface and 192.... is the "external" interface. That's mixed. Thus I advice you to switch the order of IP addresses so that internal is 192... and external is 109... This is essential to use the i,e flags correctly. > I'm getting RTPstats published on Kamailio/RTPproxy syslog. > > I am using Echo() application in asterisk. So I've observed that my > audio went to asterisk and asterisk did indeed send audio back to the > RTPproxy assigned ports BUT the RTPproxy isn't forwarding the Audio back > to the client or may-be the audio is lost somewhere in between > RTPproxy<--->Client. /[See Wirshark Flow-Diagram taken > on-Kamailio/RTPproxy Server]/ In your trace you see, that RTP does not use the internal IP address, but only the external IP address. Thus I suspect that the RTP packets from Asterisk to the public rtpproxy interface do not even arrive at rtpproxy. > > The # DTMF signal sent from Client phone reached asterisk safely and > terminated the Echo() application. > > I did the ip_forwarding thingie too.-- somebody mentioned to do this if > using RTPproxy in bridged mode. This is bullshit. People seem to not understand IP at all. rtpproxy is forwarding in user mode - it receives on a UDP socket and sends on a UDP socket. Thus, kernel based forwarding is not involved when using rtpproxy. You can disable IP forwarding again. > #echo "1" >> /proc/sys/net/ipv4/ip_forward > #sysctl -p > > > Here are the corresponding Logs from Kamailio and rtpproxy. The thing is quite easy. Swap the IP addresses when starting rtpproxy and then configure Kamailio to call rtpproxy with the proper i,e flags (as described in my previous email). For example: if (has_body("application/sdp")) { if ($si == "192.168.2.75") { # message from internal to external rtpproxy_manage("ie"); } else { # message from external to internal rtpproxy_manage("ei"); } } Put above code block in the main route block and also in a reply route which will be activated (or in the default reply route). Put some xlog statements around the rtpproxy calls to verify the proper command is called. regards Klaus

2 1

Kamailio 3.2 Acc Runtime error
by Nathaniel L Keeling 05 Nov '11

05 Nov '11

I am getting this error when I start kamailio 3.2 on Solaris 10. Any direction on solving this would be greatly appreciated. ERROR: load_module: could not open module </opt/kamailio-3.2/lib64/kamailio/modules_k/acc.so>: ld.so.1: kamailio: fatal: relocation error: file /opt/kamailio-3.2/lib64/kamailio/modules_k/acc.so: symbol timersub: referenced symbol not found Thanks Nathaniel

2 2

Directory missing compiling dbtext module
by Pedro Antonio Vico Solano 05 Nov '11

05 Nov '11

3 2

CANCEL is not forwarded, if callee does not respond
by Mino Haluz 04 Nov '11

04 Nov '11

Hi, the scenario is like this: - user1 makes a call to user2 - proxy sends receives INVITE and sends it to user2 - user2 does not respond at all (firewall for example) - user1 after 5 seconds hangs up the call with CANCEL - proxy sends to user1 200 canceling CANCEL is not forwarded to user2. Ok this behaviour is maybe ok, but after 30 seconds Request timed out is sent to user1 (fr_inv_timer possibly hit). So if I am thinking well, first CANCEL did not match the transaction ? Why ? It this behaviour ok or do I have to search for something wrong in the config? Thanks, Mino

2 1

Kamailio as SBC / LB Media-Servers farm
by Sammy Govind 04 Nov '11

04 Nov '11

Hello List, I've successfully configured Kamailio with a farm of asterisk servers in L/B | F/O mode. The topology is something like below: User<-->Kamailio<-->Asterisk-Servers Both kamailio and Asterisk serevrs are on public IPs *The problem:* Kamailio and Asterisk Servers need to be on Public IPs in order to fully handle NAT/media related issues. Whereas what I require is to have two interfaces at kamailio server, one is Public other is Private and All the Asterisk Servers should be on LAN/Private IPs. I've tried making a scenario as per my requirement (attached image) but there are some serious NAT and media issues. Can anyone help me how to handle NAT'd Asterisk servers behind Kamailio? Thanks and Best Regards. Sammy

4 7

Kamailio as SBC / LB Media-Servers farm
by Sammy Govind 04 Nov '11

04 Nov '11

Hello List, I've successfully configured Kamailio with a farm of asterisk servers in L/B | F/O mode. The topology is something like below: User<-->Kamailio<-->Asterisk-Servers Both kamailio and Asterisk serevrs are on public IPs *The problem:* Kamailio and Asterisk Servers need to be on Public IPs in order to fully handle NAT/media related issues. Whereas what I require is to have two interfaces at kamailio server, one is Public other is Private and All the Asterisk Servers should be on LAN/Private IPs. I've tried making a scenario as per my requirement (attached image) but there are some serious NAT and media issues. Can anyone help me how to handle NAT'd Asterisk servers behind Kamailio? Thanks and Best Regards. Sammy

1 0

Issue with version table
by Lucas Alvarez 03 Nov '11

03 Nov '11

Hi, I've enabled xcap server in kamailio 3.1.3 and I'm stuck with this error: 0(5073) ERROR: <core> [db.c:408]: invalid version 0 for table subscriber found, expected 6 (check table structure and table "version") I've already checked the version table and the table_name = 'subscriber' has table_version = 6. Any help will be appreciated. Regards. Lucas

2 3

Dispatcher Confusion (v3.2.0)
by Asgaroth 03 Nov '11

03 Nov '11

Hi All, Just been doing some testing with Kamailio 3.2 and the dispatcher module, and have some strange behaviour, can we confirm if it is expected behaviour or a bug of some sort. It appears that I can only set the dspatcher state via fifo command once, and not reset it again, here is an example: version: kamailio 3.2.0 (i386/linux) 94d3b8 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: 94d3b8 compiled on 12:34:43 Oct 20 2011 with gcc 4.1.2 Here we start with the dispatcher list as it was pulled from the database and currently loaded in memory. # kamctl dispatcher dump SET_NO:: 2 SET:: 1 URI:: sip:1.1.1.1:10001 flags=AX priority=0 attrs= URI:: sip:1.1.1.1:10000 flags=AX priority=0 attrs= SET:: 2 URI:: sip:1.1.1.1:10003 flags=AX priority=0 attrs= URI:: sip:1.1.1.1:10002 flags=AX priority=0 attrs= Now I try to set one gateway inactive: # kamctl fifo ds_set_state i 1 sip:1.1.1.1:10001 # kamctl dispatcher dump SET_NO:: 2 SET:: 1 URI:: sip:1.1.1.1:10001 flags=IX priority=0 attrs= URI:: sip:1.1.1.1:10000 flags=AX priority=0 attrs= SET:: 2 URI:: sip:1.1.1.1:10003 flags=AX priority=0 attrs= URI:: sip:1.1.1.1:10002 flags=AX priority=0 attrs= All normal so far, now if I try to set the same gateway active again, the state still stays inactive, i dont seem to be able to change the state back to active. The same thing happens if I attempt to set this state to disabled and then try to set it back to active afterwards. The state does not change from what the first attempt was. # kamctl fifo ds_set_state a 1 sip:1.1.1.1:10001 # kamctl dispatcher dump SET_NO:: 2 SET:: 1 URI:: sip:1.1.1.1:10001 flags=IX priority=0 attrs= URI:: sip:1.1.1.1:10000 flags=AX priority=0 attrs= SET:: 2 URI:: sip:1.1.1.1:10003 flags=AX priority=0 attrs= URI:: sip:1.1.1.1:10002 flags=AX priority=0 attrs= Now I can try to set the state to inactive probing, that works. # kamctl fifo ds_set_state ip 1 sip:1.1.1.1:10001 # kamctl dispatcher dump SET_NO:: 2 SET:: 1 URI:: sip:1.1.1.1:10001 flags=IP priority=0 attrs= URI:: sip:1.1.1.1:10000 flags=AX priority=0 attrs= SET:: 2 URI:: sip:1.1.1.1:10003 flags=AX priority=0 attrs= URI:: sip:1.1.1.1:10002 flags=AX priority=0 attrs= Now if I try to disable the same destination: # kamctl fifo ds_set_state d 1 sip:1.1.1.1:10001 # kamctl dispatcher dump SET_NO:: 2 SET:: 1 URI:: sip:1.1.1.1:10001 flags=IP priority=0 attrs= URI:: sip:1.1.1.1:10000 flags=AX priority=0 attrs= SET:: 2 URI:: sip:1.1.1.1:10003 flags=AX priority=0 attrs= URI:: sip:1.1.1.1:10002 flags=AX priority=0 attrs= State still stays in Inactive/Probing mode. Now if I try to set the same gateway to active/probing mode, I cant: # kamctl fifo ds_set_state ap 1 sip:1.1.1.1:10001 # kamctl dispatcher dump SET_NO:: 2 SET:: 1 URI:: sip:1.1.1.1:10001 flags=IP priority=0 attrs= URI:: sip:1.1.1.1:10000 flags=AX priority=0 attrs= SET:: 2 URI:: sip:1.1.1.1:10003 flags=AX priority=0 attrs= URI:: sip:1.1.1.1:10002 flags=AX priority=0 attrs= The only way for me to get the state back to active after the above is to do a reload on the dispatch table: # kamctl dispatcher reload # kamctl dispatcher dump SET_NO:: 2 SET:: 1 URI:: sip:1.1.1.1:10001 flags=AX priority=0 attrs= URI:: sip:1.1.1.1:10000 flags=AX priority=0 attrs= SET:: 2 URI:: sip:1.1.1.1:10003 flags=AX priority=0 attrs= URI:: sip:1.1.1.1:10002 flags=AX priority=0 attrs= Is the dispatcher expected to behave as above? I'm still trying to understand the different states assuming the following options are set in the configuration file: modparam("dispatcher", "ds_ping_interval", 10) modparam("dispatcher", "ds_probing_threshhold", 1) modparam("dispatcher", "ds_probing_mode", 0) AX = Active/No Probing (*) Gateway is used in ds_select_*, ds_next_* (*) No OPTIONS/INFO messages are sent to destination to check alive state This appears to be the normal operating state with no issues detected. AP = Active/Probing (*) Gateway is NOT used in ds_select_*, ds_next_* (*) OPTIONS/INFO messages ARE sent to destination to check alive state This appears to be the state taken when a gateway times out and ds_mark_dst("p") is set in failure route, the gateway is "pinged" while down and set into AX mode once a response is recieved. IX = Inactive/No Probing (*) Gateway is NOT used in ds_select_*, ds_next_* (*) No OPTIONS/INFO messages sent to destination This appears to be the state taken when a gateway times out and if ds_mark_dst("i") is set in failure route, this gateway is set inactive and no "pinging" is performed. IP = Inactive/Probing (*) Gateway is NOT used in ds_select_*, ds_next_* (*) OPTIONS/INFO messages ARE sent to destination I'm not sure when this state would be reached unless it is set by fifo command. The ds_mark_dst function only allows "a","i" or "p". So the probing method would only become active is ds_probing_mode = 1. In my case it is 0, so I'm not sure if I would ever reach this state unless I forced it. What would happen here is the gateway came back alive once in "IP" state, would it go back to "AX" state? DX = Disabled/No Probing (*) Gateway is administratively set to down This is where manual intervention has occured and the destination has been set to administrativly down, for example, work being carried out on the destination gateway. All these states make sense, for the most part, I'm a little unsure of how IP/IX is meant to work unless you manually set it to "inactive" in your failure route. The main concern though is the the fifo commands dont appear to be working as expected, I can only set the state for a destination manually once, and not again after that, unless I perform a dispatcher reload. Has anyone else experienced this behaviour? Any comments/suggestions explanations are most welcome. Thanks

3 32

Kamailio dinner next week in Cape Town, South Africa
by Daniel-Constantin Mierla 03 Nov '11

03 Nov '11

Hello, it happens to come across Cape Town for few days and I was wondering if there are folks using our project or doing VoIP in general that want to meet up for a dinner. The date for dinner is planned for next Tuesday, November 8, 2011. Place will be decided once I know if there are people coming and how many. Likewise with such events done in the past, it is going to be a dinner open for everyone that wants to join, each participant paying the drinks and food for himelf/herself. The goal is to meet and discuss latest trends in IP communication and what's up lately around our project. Here are some pictures from two past similar events: http://www.asipto.com/gallery/v/social_meeting_barcelona2008/ http://www.asipto.com/gallery/v/FOSDEM2009/ So, if there are people on these mailing lists in the area of Cape Town next Tuesday and want to meet, please reply to me till Friday evening -- I will follow up privately to each person that is going to join with details about location and start time. Looking forward to meet some of you very soon! Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Dec 5-8, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

2 1

TOS=0 in kamailio.cfg
by DELEY Marion 03 Nov '11

03 Nov '11

Hi, I'm trying to set tos value to zero but kamailio doesn't accept it. In fact, it's DSCP field which interest me but i can't set zero. Can you explain me why is it forbidden and how can i do that ? Thanks a lot Marion

1 0

is there help on mi commands in sercmd?
by Juha Heinanen 03 Nov '11

03 Nov '11

sercmd has help that lists the commands, one of which is mi. is there a way to list all mi sub-commands? i have tried ip-proxy_ctl> help mi Execute MI command sip-proxy_ctl> mi help error: 500 - command not available -- juha

2 3

extra _ctl in ctl file
by Juha Heinanen 03 Nov '11

03 Nov '11

i just noticed that my sip-proxy_ctl (alias kamailio_ctl) command has stopped working. it now adds an extra _ctl to the socket file name: # sip-proxy_ctl ERROR: connect_unix_sock: connect(/tmp/sip-proxy_ctl_ctl): No such file or directory [2] any idea where that comes from or it is only me, who has this problem? i have not changed anything the command used to work fine earlier. -- juha

2 4

Dialog / BYE
by Brandon Armstead 02 Nov '11

02 Nov '11

Hello, I am attempting to tear down a call with a BYE packet generated externally (kind of similar to Kamailio fifo dlg_end_dlg). Let me describe what I am trying to do in more depth and then I will continue to tell you the problem I think I am experiencing. [PSTN SIP Proxy] -> [CORE SIP Proxy] -> [REGISTRAR] -> [UAC] So the above layout is the normal call flow / structure of calls (incoming when originating from pstn) (outgoing when originating from uac). I then have an "external" host - I am attempting to generate a BYE to [CORE SIP Proxy] and have it go both directions [PSTN] + [UAC]. So far I am able to get the call to tear down in only a single direction (only kill call with PSTN) or (only kill call with UAC). I have not been able to kill both legs of the call. I have tried sending a single BYE and using some dlg_bye("all") magic. - does not work, only kills one leg of the call. I have tried sending a BYE message to SIP Proxy for each leg (two BYE). - does not work, only kills one leg of the call. If I alternate the later method's BYE packet and only send A or B and not both Leg A or B will be killed... I think I am running into a race condition where the dialog is destroyed before the 2nd BYE packet is processed. My Question - what am I doing wrong - or what is the best method to tackle this task? Thank you for all of your help and time in advance. Sincerely, Brandon Armstead

2 9

Kamailio3.2 debian repo fails
by Benjamin Henrion 02 Nov '11

02 Nov '11

Hi, I have the following /etc/apt/sources.list: ======================================================== $ cat /etc/apt/sources.list deb http://ftp.debian.org/debian squeeze main contrib non-free deb-src http://ftp.debian.org/debian squeeze main contrib non-free deb http://deb.kamailio.org/kamailio32 squeeze main deb-src http://deb.kamailio.org/kamailio32 squeeze main ======================================================== Which generates the following error: ======================================================== $ apt-get update Ign http://deb.kamailio.org squeeze Release.gpg Ign http://deb.kamailio.org/kamailio32/ squeeze/main Translation-en Ign http://deb.kamailio.org/kamailio32/ squeeze/main Translation-en_US Ign http://deb.kamailio.org/kamailio32/ squeeze/main Translation-en_US.UTF-8 Ign http://deb.kamailio.org squeeze Release Ign http://deb.kamailio.org squeeze/main Sources Hit http://ftp.debian.org squeeze Release.gpg Ign http://ftp.debian.org/debian/ squeeze/contrib Translation-en Ign http://ftp.debian.org/debian/ squeeze/contrib Translation-en_US Ign http://deb.kamailio.org squeeze/main amd64 Packages Ign http://deb.kamailio.org squeeze/main Sources Ign http://ftp.debian.org/debian/ squeeze/contrib Translation-en_US.UTF-8 Ign http://ftp.debian.org/debian/ squeeze/main Translation-en Ign http://ftp.debian.org/debian/ squeeze/main Translation-en_US Ign http://ftp.debian.org/debian/ squeeze/main Translation-en_US.UTF-8 Ign http://ftp.debian.org/debian/ squeeze/non-free Translation-en Ign http://ftp.debian.org/debian/ squeeze/non-free Translation-en_US Ign http://ftp.debian.org/debian/ squeeze/non-free Translation-en_US.UTF-8 Hit http://ftp.debian.org squeeze Release Ign http://deb.kamailio.org squeeze/main amd64 Packages Err http://deb.kamailio.org squeeze/main Sources 404 Not Found Err http://deb.kamailio.org squeeze/main amd64 Packages 404 Not Found Hit http://ftp.debian.org squeeze/main Sources Hit http://ftp.debian.org squeeze/contrib Sources Hit http://ftp.debian.org squeeze/non-free Sources Hit http://ftp.debian.org squeeze/main amd64 Packages Hit http://ftp.debian.org squeeze/contrib amd64 Packages Hit http://ftp.debian.org squeeze/non-free amd64 Packages W: Failed to fetch http://deb.kamailio.org/kamailio32/dists/squeeze/main/source/Sources.gz 404 Not Found W: Failed to fetch http://deb.kamailio.org/kamailio32/dists/squeeze/main/binary-amd64/Packages… 404 Not Found E: Some index files failed to download, they have been ignored, or old ones used instead. ======================================================== Who is responsible for the debian packaging? Best, -- Benjamin Henrion <bhenrion at ffii.org> FFII Brussels - +32-484-566109 - +32-2-4148403 "In July 2005, after several failed attempts to legalise software patents in Europe, the patent establishment changed its strategy. Instead of explicitly seeking to sanction the patentability of software, they are now seeking to create a central European patent court, which would establish and enforce patentability rules in their favor, without any possibility of correction by competing courts or democratically elected legislators."

3 6

Re: [SR-Users] Siremis: cannot enter domain in "New Subscriber"window
by Uwe Andersen 02 Nov '11

02 Nov '11

Hi Hugh, it works now. Thank you! Uwe > -----Ursprüngliche Nachricht----- > Von: sr-users-bounces(a)lists.sip-router.org > [mailto:sr-users-bounces@lists.sip-router.org]Im Auftrag von > Hugh Waite > Gesendet: Mittwoch, 2. November 2011 12:52 > An: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > UsersMailing List > Betreff: Re: [SR-Users] Siremis: cannot enter domain in "New > Subscriber"window > > > Hi Uwe, > > Domains are configured in Server Services->Domain List. > These are the > domains which the kamailio server is responsible for. > When they have been entered there, they will appear in the drop-down > list when adding users. > > Regards, > Hugh > > On 02/11/2011 11:33, Uwe Andersen wrote: > > Hi, > > > > I want to add a new user. So I choose "add" from the > "Subscriber Management" page. > > On the "New Subscriber" page I can enter username, > password, email-address, etc. > > But the line for "domain" seems to be an empty dropdown entry. > > I cannot choose, add or edit anything. > > As a result I cannot save the new user because Siremis > complains about the empty domain field. > > > > After adding a user by command line("kamctl add ...") the > user appears on the management page. > > If I edit that user via Siremis the line for domain is not > a dropdown but a "normal" text field. > > > > Can somebody explain this? > > > > BR, > > Uwe > > > > P.S.: I'm using Siremis 2.0 and Kamailio 3.2.0. > > > > _______________________________________________ > > 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 > > > -- > -- > Hugh Waite > Senior Design Engineer > Crocodile RCS Ltd. > > > _______________________________________________ > 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 >

1 0

Siremis: cannot enter domain in "New Subscriber" window
by Uwe Andersen 02 Nov '11

02 Nov '11

Hi, I want to add a new user. So I choose "add" from the "Subscriber Management" page. On the "New Subscriber" page I can enter username, password, email-address, etc. But the line for "domain" seems to be an empty dropdown entry. I cannot choose, add or edit anything. As a result I cannot save the new user because Siremis complains about the empty domain field. After adding a user by command line("kamctl add ...") the user appears on the management page. If I edit that user via Siremis the line for domain is not a dropdown but a "normal" text field. Can somebody explain this? BR, Uwe P.S.: I'm using Siremis 2.0 and Kamailio 3.2.0.

2 1

FW: App_python Module
by Muhammad Nasir Bhutta 02 Nov '11

02 Nov '11

Hi, I am a new user of Kamailio, and I am working on enabling the App_Python module in Kamailio.. While searching on google, I came acroos your email on the forum where you people are trying to patch the App_Python module.. I am wandering whether you can please help a bit .. 1- How to do the configuration to enable the App_python module ? .. At the moment, I am just adding the following line in the configuration file, loadmodule "app_python.so" and just correcting the path to handler.py to the right directory in the code for script path. 2- What output we should expect when we enable the app_python module ? .. I have seen the handler.py script written in python which is printing some sip message information ? .. 3- How the SIP Message structure is being passed to handler.py ? ... How we are accessing the Router module in handler.py script and LM_ERR function from Kamailio ? .. Any help in this regards shall be really helpful .. In any case, whether you can solve this issue or not, please reply the email.. With many thanks.. Kind Regards, Nasir

2 3

sdpops and dynamic codec ids
by Juha Heinanen 01 Nov '11

01 Nov '11

i made call sdp_keep_codecs_by_name("PCMU,PCMA,speex"); on an invite request and it didn't keep speex although it was in the sdp: Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): sems 1 1 IN IP4 192.98.102.10 Session Name (s): sems Connection Information (c): IN IP4 192.98.102.10 Time Description, active time (t): 0 0 Media Description, name and address (m): audio 40000 RTP/AVP 101 102 8 0 3 2 9 96 97 98 99 100 Media Attribute (a): rtpmap:101 iLBC/8000 Media Attribute (a): rtpmap:102 speex/8000 Media Attribute (a): rtpmap:8 PCMA/8000 Media Attribute (a): rtpmap:0 PCMU/8000 Media Attribute (a): rtpmap:3 GSM/8000 Media Attribute (a): rtpmap:2 G721/8000 Media Attribute (a): rtpmap:9 g722/8000 Media Attribute (a): rtpmap:96 telephone-event/8000 Media Attribute (a): rtpmap:97 G726-32/8000 Media Attribute (a): rtpmap:98 G726-24/8000 Media Attribute (a): rtpmap:99 G726-40/8000 Media Attribute (a): rtpmap:100 G726-16/8000 it is so that sdpops module does not support codecs, whose ids are dynamic, but only codecs with static ids? -- juha

2 14

enabling app_python module in Kamailio
by Muhammad Nasir Bhutta 01 Nov '11

01 Nov '11

Hi, I am trying to enable the app_python module in Kamailio to run a simple python script externally parallel with Kamailio proxy, While I try to load the module "app_python.so" in configuration file, the Kamailio server fails to start.. I can't see what is the problem .. The steps, I am following are that: - Script written in python saved in home directory. - Loadmodule "app_python.so" statement in configuration file. - Modparam("app_python", "script_name", "location of script") - Python_exec("function", "parameters list") .. I have also tried to check the Kamailio configuration option, It gives following warning, 0(2942) WARNING: <core> [sr_module.c:584]: /usr/local/lib64/kamailio/modules/app_python.so: exports dlflags interface is deprecated and it will not be supported in newer versions; consider using mod_register() instead I am wandering if any one can please help in this matter, I shall be very thankful .. With many thanks .. Kind Regards, Nasir Muhammad Nasir Bhutta Solutions Architect Britannic Technologies Ltd t: +44 1483 544812 f: +44 845 0501001 mbhutta(a)btlnet.co.uk<mailto:mbhutta@btlnet.co.uk> www.btlnet.co.uk<http://www.btlnet.co.uk> [cid:image001.jpg@01CC881F.1EA78300]<http://www.linkedin.com/company/britannic-technologies/products>[cid:image002.jpg@01CC881F.1EA78300]<http://www.twitter.com/BritannicTech>[cid:image003.jpg@01CC881F.1EA78300]<http://www.facebook.com/pages/Britannic-Technologies/202957249732198>[cid:image004.jpg@01CC881F.1EA78300]<http://www.youtube.com/user/BritannicTech> ________________________________ [cid:image007.jpg@01CC881F.4A995BF0]<http://www.btlnet.co.uk/> Watch the Telegraph Business Club video about Britannic at our educational TV site - britannictech.tv<http://britannictech.tv> Our 8th Annual Convergence Summit is being held at Silverstone on 17th November 2011. Book your place now<http://www.btlnet.co.uk/news--events/2011/convergence-summit-2011/register> and discover the world of Hosting and Cloud Services. ________________________________ This email and any files transmitted with it are confidential and intended for the addressee only. If you are not the intended recipient please notify the sender and delete this email. The views expressed in this email are personal, and do not constitute a commitment by Britannic Technologies Ltd unless specified by a separate, written agreement. Registered Office: Britannic House, Merrow Business Park, Guildford, Surrey GU4 7WA. Registered in England no. 2090797. ________________________________

3 4

app_python module
by Muhammad Nasir Bhutta 01 Nov '11

01 Nov '11

Hi, I have successfully installed the app_python module for the default example of handler.py .. Now, I need to see the output of handler.py,.. I have following questions: 1- What is the output of this module ? does it prints the specific message information extracted from SIP message ? 2- How to write modparam and python_exec in configuration file ? I have tried to write modparam("app_python", "mod_init") but, it gives error and says to use mod_register function instead and I don't know more about mod_register function. 3- I have also tried to use python_exec("child_init", 2) , The Kamailio starts successfully.. When, I connect a soft client to Kamailio, then nothing happens ? How to see the output of handler.py script . Any help in this regards shall be really appreciated .. With thanks. Kind Regards, Nasir

1 0

script local flags
by Uri Shacked 01 Nov '11

01 Nov '11

Hi, I am using the script local flags. i set a flag like this "setsflag(6)" in order to use it afterwards. i notice that this flag is relevent only for the request i am dealing with right now. because after the reply the flag seems to reset without me doing that doing "issflagset(6)" and getting nowhere :-) am i right or am i using it wrong?

2 1

FW: App_python Module
by Muhammad Nasir Bhutta 01 Nov '11

01 Nov '11

Hi, I am a new user of Kamailio, and I am working on enabling the App_Python module in Kamailio.. While searching on google, I came acroos your email on the forum where you people are trying to patch the App_Python module.. I am wandering whether you can please help a bit .. 1- How to do the configuration to enable the App_python module ? .. At the moment, I am just adding the following line in the configuration file, loadmodule "app_python.so" and just correcting the path to handler.py to the right directory in the code for script path. 2- What output we should expect when we enable the app_python module ? .. I have seen the handler.py script written in python which is printing some sip message information ? .. 3- How the SIP Message structure is being passed to handler.py ? ... How we are accessing the Router module in handler.py script and LM_ERR function from Kamailio ? .. Any help in this regards shall be really helpful .. In any case, whether you can solve this issue or not, please reply the email.. With many thanks.. Kind Regards, Muhammad Nasir Bhutta Solutions Architect Britannic Technologies Ltd t: +44 1483 544812 f: +44 845 0501001 mbhutta(a)btlnet.co.uk<mailto:mbhutta@btlnet.co.uk> www.btlnet.co.uk<http://www.btlnet.co.uk> [cid:image001.jpg@01CC9895.3EC504B0]<http://www.linkedin.com/company/britannic-technologies/products>[cid:image002.jpg@01CC9895.3EC504B0]<http://www.twitter.com/BritannicTech>[cid:image003.jpg@01CC9895.3EC504B0]<http://www.facebook.com/pages/Britannic-Technologies/202957249732198>[cid:image004.jpg@01CC9895.3EC504B0]<http://www.youtube.com/user/BritannicTech> ________________________________ [cid:image005.jpg@01CC9895.3EC504B0]<http://www.btlnet.co.uk/> Watch the Telegraph Business Club video about Britannic at our educational TV site - britannictech.tv<http://britannictech.tv> Our 8th Annual Convergence Summit is being held at Silverstone on 17th November 2011. Book your place now<http://www.btlnet.co.uk/news--events/2011/convergence-summit-2011/register> and discover the world of Hosting and Cloud Services. ________________________________ This email and any files transmitted with it are confidential and intended for the addressee only. If you are not the intended recipient please notify the sender and delete this email. The views expressed in this email are personal, and do not constitute a commitment by Britannic Technologies Ltd unless specified by a separate, written agreement. Registered Office: Britannic House, Merrow Business Park, Guildford, Surrey GU4 7WA. Registered in England no. 2090797. ________________________________

1 0

flags and masks in carrierroute
by Uri Shacked 01 Nov '11

01 Nov '11

Hi, I did not understand the explanation of the use of flags and mask. I understand flags are for matching and mask is doing something. But, what exactly does the mask do? How do i see or use it? And regarding the flags... i did "setflag(16)" before the CR function. Set the DB col as needed, and no match was found... Any help here?

1 0

siremis: login error / http error 404
by Uwe Andersen 01 Nov '11

01 Nov '11

Hi Ramona, You hit it! After enabling "rewrite" it is working now. Thank you very much! Uwe > > I have installed Siremis 2.0.0 and Kamaillio 3.2.0 on a > OpenSuse 11.4 system. > > After the Siremis Web Installation Wizard finished the > setup the login failed with a http error 404. > > > > The Apache2.0 error_log shows: > > [Mon Oct 31 11:09:49 2011] [error] [client ::1] File does > not exist: /var/www/siremis-2.0.0/siremis/system, referer: > http://localhost/siremis/index.php/user/login > > > > The Apache2.0 access_log shows: > > ::1 - - [31/Oct/2011:11:09:36 +0100] "GET > /siremis/bin/antispam_image.php?form=user.form.LoginForm&name= > antispam&length=6&level=2?rand=1768032447 HTTP/1.1" 200 240 > "http://localhost/siremis/index.php/user/login" "Mozilla/5.0 > (X11; Linux i686; rv:7.0.1) Gecko/20100101 Firefox/7.0.1" > > ::1 - - [31/Oct/2011:11:09:48 +0100] "POST > /siremis/bin/controller.php?F=RPCInvoke&P0=[user.form.LoginFor > m]&P1=[Login]&__this=btn_login:onclick&jsrs=1 HTTP/1.1" 200 > 214 "http://localhost/siremis/index.php/user/login" > "Mozilla/5.0 (X11; Linux i686; rv:7.0.1) Gecko/20100101 Firefox/7.0.1" > > ::1 - - [31/Oct/2011:11:09:49 +0100] "GET > /siremis/system/general_default HTTP/1.1" 404 1331 > "http://localhost/siremis/index.php/user/login" "Mozilla/5.0 > (X11; Linux i686; rv:7.0.1) Gecko/20100101 Firefox/7.0.1" > > > > Is there anybody how can help me out of this? > > do you have mod_rewrite enabled in your apache2 server?

1 0

Memory Leak on DB Errors?
by Klaus Darilion 31 Oct '11

31 Oct '11

Hi! I recently had a problem with Kamailio 3.1.4 (provided Debian packages): I had some DB problems (missing tables). Thus, the timer module failed to insert the statistics (for siremis): ERROR: db_mysql [km_dbase.c:120]: driver error on query: Table 'kamailio.statistics_tmx' doesn't exist ERROR: <core> [db_query.c:130]: error while submitting query ERROR: sqlops [sql_api.c:217]: cannot do the query This happened for some time (weeks?), other DB queries were unaffected. Then, suddenly Kamailio ran out of memory: ERROR: <core> [sip_msg_clone.c:506]: ERROR: sip_msg_cloner: cannot allocate memory ERROR: tm [t_lookup.c:1338]: ERROR: new_t: out of mem: ERROR: tm [t_lookup.c:1478]: ERROR: t_newtran: new_t failed ERROR: sl [sl_funcs.c:282]: ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (1/SL) It do not think it is a load problem as the server is more or less idle. May it be a memory leak due to wrong error handling? Where there any fixes recently? Thanks Klaus

3 16

current shm memory status
by Klaus Darilion 31 Oct '11

31 Oct '11

Hi! Is there a way to get the status of shm memory? (used memory vs. free memory) Thanks Klaus

2 4

siremis: login error / http error 404
by Uwe Andersen 31 Oct '11

31 Oct '11

Hi, I have installed Siremis 2.0.0 and Kamaillio 3.2.0 on a OpenSuse 11.4 system. After the Siremis Web Installation Wizard finished the setup the login failed with a http error 404. The Apache2.0 error_log shows: [Mon Oct 31 11:09:49 2011] [error] [client ::1] File does not exist: /var/www/siremis-2.0.0/siremis/system, referer: http://localhost/siremis/index.php/user/login The Apache2.0 access_log shows: ::1 - - [31/Oct/2011:11:09:36 +0100] "GET /siremis/bin/antispam_image.php?form=user.form.LoginForm&name=antispam&length=6&level=2?rand=1768032447 HTTP/1.1" 200 240 "http://localhost/siremis/index.php/user/login" "Mozilla/5.0 (X11; Linux i686; rv:7.0.1) Gecko/20100101 Firefox/7.0.1" ::1 - - [31/Oct/2011:11:09:48 +0100] "POST /siremis/bin/controller.php?F=RPCInvoke&P0=[user.form.LoginForm]&P1=[Login]&__this=btn_login:onclick&jsrs=1 HTTP/1.1" 200 214 "http://localhost/siremis/index.php/user/login" "Mozilla/5.0 (X11; Linux i686; rv:7.0.1) Gecko/20100101 Firefox/7.0.1" ::1 - - [31/Oct/2011:11:09:49 +0100] "GET /siremis/system/general_default HTTP/1.1" 404 1331 "http://localhost/siremis/index.php/user/login" "Mozilla/5.0 (X11; Linux i686; rv:7.0.1) Gecko/20100101 Firefox/7.0.1" Is there anybody how can help me out of this? Regards, Uwe Andersen

2 1

Re: [SR-Users] uac module - error searching pvar "uac_req"
by Daniel-Constantin Mierla 31 Oct '11

31 Oct '11

On 10/31/11 2:09 PM, Vladimir Broz wrote: > ah, that was the problem :( ok. Don't forget to keep the mailing list always cc-ed, so people reading the archive for similar issues will find the conclusion, whether the hints were solving or not the problem. Cheers, Daniel > > Thanks Daniel! > -Vlada > > On 10/31/2011 01:37 PM, Daniel-Constantin Mierla wrote: >> Hello, >> >> be sure you load uac module from modules_k, not from modules_s. >> >> Cheers, >> Daniel >> >> >> On 10/31/11 12:36 PM, Vladimir Broz wrote: >>> ignore the "sip:b@@test.com" -> same result with the correct (only one >>> "@") value >>> >>> Thanks, >>> -Vlada >>> >>> On 10/31/2011 12:21 PM, Vladimir Broz wrote: >>>> Hi, >>>> >>>> I compiled 3.2 (master) with FLAVOUR=ser and I'm trying to use "uac" >>>> module. >>>> >>>> my config looks like: >>>> ... >>>> $uac_req(method)="NOTIFY"; >>>> $uac_req(ruri)="sip:server.test.com"; >>>> $uac_req(furi)="sip:a@test.com"; >>>> $uac_req(turi)="sip:b@@test.com"; >>>> ... >>>> >>>> and I'm getting this config file error >>>> >>>> 0(26125) ERROR: <core> [pvapi.c:706]: error searching pvar "uac_req" >>>> 0(26125) ERROR: <core> [pvapi.c:910]: wrong char [m/109] in >>>> [$uac_req(method)] at [9 (5)] >>>> 0(26125) : <core> [cfg.y:3501]: parse error in config file >>>> /root/vbroz/ser-srrs-bpi.cfg, line 388, column 2-17: unknown script >>>> pseudo variable $uac_req(method) >>>> ERROR: bad config file (1 errors) >>>> >>>> What's wrong? >>>> >>>> Any help is appreciated... >>>> >>>> Thanks in advance, >>>> -Vlada >>>> >>>> _______________________________________________ >>>> 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 >>> >>> _______________________________________________ >>> 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 Kamailio Advanced Training, Dec 5-8, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 0

uac module - error searching pvar "uac_req"
by Vladimir Broz 31 Oct '11

31 Oct '11

Hi, I compiled 3.2 (master) with FLAVOUR=ser and I'm trying to use "uac" module. my config looks like: ... $uac_req(method)="NOTIFY"; $uac_req(ruri)="sip:server.test.com"; $uac_req(furi)="sip:a@test.com"; $uac_req(turi)="sip:b@@test.com"; ... and I'm getting this config file error 0(26125) ERROR: <core> [pvapi.c:706]: error searching pvar "uac_req" 0(26125) ERROR: <core> [pvapi.c:910]: wrong char [m/109] in [$uac_req(method)] at [9 (5)] 0(26125) : <core> [cfg.y:3501]: parse error in config file /root/vbroz/ser-srrs-bpi.cfg, line 388, column 2-17: unknown script pseudo variable $uac_req(method) ERROR: bad config file (1 errors) What's wrong? Any help is appreciated... Thanks in advance, -Vlada

2 2

Re: [SR-Users] how to retrieve the contact username from 3xx reply?
by Uri Shacked 30 Oct '11

30 Oct '11

Thanks! works great.

1 0

how to retrieve the contact username from 3xx reply?
by Uri Shacked 30 Oct '11

30 Oct '11

---------- Forwarded message ---------- From: Uri Shacked <uri.shacked(a)gmail.com> Date: Sun, Oct 30, 2011 at 12:16 PM Subject: how to retrieve the contact username from 3xx reply? To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org> I need to store the user i get from the 301 reply to an AVP and use it afterwards in another route. How can i retrieve the contact from the 3xx reply? uac_redirect does not have this option and t_load_contacts does not work for me as well.... Anyone has an idea?

2 1

module docs broken
by Timo Reimann 28 Oct '11

28 Oct '11

Hey, documentation on the dialog module seems to be broken: The master branch directory content is empty http://sip-router.org/docbook/sip-router/branch/master/modules_k/dialog/ 3.1 shows close to nothing http://sip-router.org/docbook/sip-router/branch/3.1/modules_k/dialog/dialog… Moreover, 3.0 gives me 404 on the entire modules_k path: http://sip-router.org/docbook/sip-router/branch/3.0/modules_k/ Help. :) Cheers, --Timo

2 2

T1 timer
by Vitaliy 28 Oct '11

28 Oct '11

Hello, How can i change T1 timeout for transaction created by the UAC module (uac_req_send). Do i have to call t_set_retr before uac_req_send and then t_reset_retr() before t_relay ? I use uac_req_send to send accounting to the remote server and i want to increase retransmit timeout for such signalling messages.

4 5

Fwd: [RAI] SIPit 29 Summary
by Klaus Darilion 28 Oct '11

28 Oct '11

FYI -------- Original Message -------- Subject: [RAI] SIPit 29 Summary Date: Fri, 28 Oct 2011 09:40:51 +0200 From: Robert Sparks <rjsparks(a)nostrum.com> To: rai(a)ietf.org (also available at https://www.sipit.net/SIPit29_summary) SIPit 29 was hosted by Etsi Plugtests in Monte Carlo, Monaco the week of October 24-27, 2011. There were 34 attendees from 17 companies visiting from 12 countries. We had 25 distinct implementations. The roles represented (some implementations act in more than one role) 20 endpoints 5 proxy/registrars The b2buas attending this event were not attempting to appear to be proxies. Implementations using each transport for SIP messages: UDP 92% TCP 96% TLS 88% (24% server-auth-only) SCTP 8% DTLS 0% 44% of the implementations present supported IPv6. There were three RFC4474 Identity implementations present. For DNS we had support for: Full RFC3263 : 76% SRV only : 16% A records only : 08% no DNS support : 0% Support for various items in the endpoints: 75% replaces 50% ice 40% 5389stun 35% turn 25% sip/stun multiplexing 30% gruu 25% history-info (there were no implementations of 4244bis) 20% outbound 20% path 20% diversion 15% 3489stun 10% join 5% service-route 0% sigcomp Support for various items in the proxy/registrars: 100% path 80% sip/stun multiplexing 60% gruu 40% outbound 40% diversion 20% service-route 0% history-info The endpoints and b2buas implemented these methods: 100% INVITE, CANCEL, ACK, BYE 95% REFER 95% NOTIFY 80% REGISTER 80% INFO 75% OPTIONS 75% UPDATE 75% PRACK 60% SUBSCRIBE 35% MESSAGE 30% PUBLISH 100% of the implementations sent RTP from the port advertised for reception (symmetric-rtp). 30% of the implementations required the other party to use symmetric-rtp. 90% of the UAs present both sent RTCP and paid attention to RTCP they received. 80% of the endpoints present supported SRTP using sdes. There were no dtls-srtp implementations at this SIPit. 35% of the endpoints supported multipart/MIME. There were no implementations present with S/MIME support. 36% of the implementations present followed RFC6026 (corrections to the INVITE transaction) 36% followed RFC4320 (corrections to the non-INVITE transaction) Not counting implementations that support events only for REFER: There were 3 SIP Event Server implementations There were 5 SIP Event Client implementations These event packages were supported: Server Client 1 2 presence 1 1 presence.winfo 1 4 message-summary 1 1 dialog 1 1 reg 1 1 conference 1 1 xcap-diff 0 2 kpml These packages were supported with the eventlist extension: Server Client 0 1 presence Two of the proxies present still rely only on max-forwards and one that implemented 3261 loop detection. There were two implementations of fork-loop-fix, but no implementations of max-breadth. Multiparty tests (Notes provided by several test participants, including Eoin McLeod, Trond Andersen, Olle Johansson, and Adrian Georgescu) ** IPv6 Focused Tests The first multiparty session of the event focused specifically on IPv6. We used a basic forking scenario with endpoints registered with several proxies at the bottom of the tree. We had a mix of single and dual-stack endpoints, many supporting video. None of the participant's code would put both v4 and v6 candidates into ICE. Rather, it made assumptions about which candidates to use based on what was happening with the SIP signaling. We ran tests using SRV configurations stating that v6 should be tried first and v4 tried if the v6 failed (and a second configuration that tested v4 first) to ensure that implementations would fail forward and try both address families, with mixed results. We confirmed that one UA configured to use IPv6 only was able to request and use an IPv4 turn allocation and succeeded in making a call between an IPv4 UA and IPv6 ua. Most UAs that supported dual-stack had a configuration to tell the application to ignore any returned AAAAs due to issues encountered in deployments where endpoints autoconfigured IPV6 that didn't actually work. We successfully tested calls going though a mix of v4 and v6 hops (accruing Via and Route/Record-Route headers with addresses in both families. There was an instance of UA registering using sip-outbound (RFC5626) that used IPv4 and IPv6 flows simultaneously. We set up automated tests of using SRV records to indicate preference for IPv4 or IPv6. Two ipv4 only clients proved that they connect to a domain indicating preference for IPv6. ** Forking There are still UAs that do not sensibly handle multiple 200 OKs to a forked INVITE. When testing handling multiple 200 OKs on a simple two branch fork, we encountered an interesting failure. As the proxy handled the 200 OK from branch 1, it sent a CANCEL to the other branch. The UA on branch 2 had already responded with a 200 OK, so it responded to the CANCEL with a 200, but rather than ignoring it, it decided to send a BYE on the dialog. This surprised the calling UA which had chosen the 200 OK from branch 2 to keep, having already sent a BYE to the dialog from branch 1. The better thing for the UA to have done is ignore the CANCEL after replying to it. ** TLS Most of the implementations at the event supported TLS, and all the multiparty tests exercised it. One session focused on proper certificate verification. We tested a certificate at a server with several SAN URI fields, none of which matched the URI used for the server, but a CN that did. Only one UA failed to connect because of this, the rest of the UAs connected happily (or not at all, but because of other issues). Most of the implementations were "before 5922" code, which meant that there was a lot of checking of the Subject/CN. ** Identity There were several SIP Identity implementations present. We tested creating and verifying the Identity assertions at proxies. It wasn't immediately obvious to some implementers that the resource pointed to by the Identity-Info URI had to be DER encoded (a result of the requirement to use application/pkix-cert). Unfortunately, the tar encoded at the end of 4474 has .cer files in it that are PEM encoded. ** STUN/TURN/ICE and Outbound We spent a significant fraction of the multiparty test time (three sessions) focusing on STUN/TURN/ICE and Outbound. There are still clients using RFC 3489 STUN. The participants noted operator pushback against deploying ICE because when things go wrong, it's very hard to diagnose when, where and why they went wrong. If the endpoints always sent a reINVITE (or UPDATE) when concluding ICE, even if they used the default candidates, this diagnosis of failures would be much easier. Examples noted by the participants: * Without a mandatory conclusion, is not possible to know how long it took until end-points had successful media path established * Without a mandatory conclusion, is not always possible to know if different than default candidates have been chosen as lack of INVITE does not tell deterministically what happened * Without a mandatory conclusion, it is not possible for the callee to restart ICE if it does not agree with the conclusion We successfully exercised endpoints setting up and managing multiple outbound flows through separate edge proxies to a proxy/registrar (with different implementations taking turns in the proxy registrar role). We forced the network to break the flow in use in each case, verifying that subsequent dialog-forming requests would take an alternate flow. We discovered an issue with the recommendation in 5626 to have edge proxies record-route that prevents subsequent in-dialog requests from using alternate flows. ** SRTP All implementations supported SDES for SRTP key exchange. Good interop with many different combinations of endpoints. Some calls were left up for an extended period so that sequence number wrapping and roc increments were tested. Once the roc had changed the call was put on hold and resumed with success. In this situation new RTP streams with different SSRCs were created and the sequence number/roc combination re-generated thus proving that support for multiple SRTP contexts exists. Debate surrounding how to signal 'best-effort crypto' continues. Some implementations choose to advertise RTP/AVP with a=crypto attributes and then go crypto if answer also has them and non-crypto if they do not. RTP/SAVP is used to signal the desire for mandatory crypto. Another approach observed was to repeat the same media line twice, once for crypto and once for non-crypto and rely on the receiving end to only accept one of them and port reject the other. In this case both m-lines had the same port number and payload number descriptions so if a receiving end accepted both then the originator would have no clue as to whether it was receiving crypto or non-crypto traffic. ** Unusual messages We tried several kinds of valid, but unusual messages. Very few implementations did the right thing with a request-URI with an unknown scheme - some phones would answer an INVITE with that in the request-URI. ** Specification questions / comments RFC 3263 currently says "If no SRV records are found, the client SHOULD use TCP for a SIPS URI" - it should probably say TLS over TCP. In the spirit of Happy Eyeballs, there should be some guidance on what to do when the RFC 3263 process results in both A and AAAA records and the client is capable of using both. There were implementations of SIP over DTLS, but the specification of that usage seems not to have completed. There were questions on whether internationalized domain names could be used in SIP URIs. For use of SDP with ICE, there was an argument about whether a=rtcp was mandatory if rtcp was on the next port up. RFC 5245 says "If the agent is utilizing RTCP, it MUST encode the RTCP candidate using the a=rtcp attribute as defined in RFC 3605 [RFC3605].", and 3605 says "when that port is not the next higher (odd) port number following the RTP port described in the media line" to motivate why the attribute was created to begin with. Any updates to 5245 should clarify that adding the a=rtcp attribute is not optional when RTCP is being sent on the next higher port. _______________________________________________ RAI mailing list RAI(a)ietf.org https://www.ietf.org/mailman/listinfo/rai

1 0

How to limit maximum number of calls
by Austin Einter 28 Oct '11

28 Oct '11

Hi I am running kamailio proxy (1.5) as an intermediate proxy, where all my SIP signalling packets are passing through. I want to limit maximum number of calls kamailio proxy server can handle at a time. How can I do this, please give me some pointer in this regard. Regards Austin

4 7

new email contact
by Timo Reimann 28 Oct '11

28 Oct '11

Hey all, this is just to let everybody know that my email contact has changed to sr(a)foo-lounge.de . Of course, I'm still on the lists but if anyone aware of my former email address wishes to contact me directly, please refer to the new address in the future. Thanks a lot and cheers, --Timo

1 0

Asterisk 1.8 and Kamailio 1.5 issue
by Rowell Rufino 27 Oct '11

27 Oct '11

Hi, We are having issues where the "OK" or "ACK" is that is coming from the phone is not relayed by OpenSER to Asterisk. Below is the sip trace... I am also attaching a tcpdump. Please help what we can do. Received from udp:10.1.10.80:5060 at 26/10/2011 10:22:41:476 (490 bytes): SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP 10.30.0.64:5060 ;received=10.30.0.64;branch=z9hG4bK-wowp1kmdy4rl;rport=5060 From: "Virgil Menendez" <sip:91421@ser.gowireless.net>;tag=6wkdms1r20 To: <sip:9513261429@ser.gowireless.net;user=phone>;tag=as0b87218f Call-ID: 3c26755bf15c-9iq08xqqblo6 CSeq: 4 INVITE Server: Asterisk PBX 1.8.7.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 ------------------------------ Sent to udp:10.1.10.80:5060 at 26/10/2011 10:22:41:481 (387 bytes): ACK sip:vm9513261429@10.1.10.83:5060 SIP/2.0 v: SIP/2.0/UDP 10.30.0.64:5060;branch=z9hG4bK-wowp1kmdy4rl;rport Route: <sip:10.1.10.80;lr=on> f: "Virgil Menendez" <sip:91421@ser.gowireless.net>;tag=6wkdms1r20 t: <sip:9513261429@ser.gowireless.net;user=phone>;tag=as0b87218f i: 3c26755bf15c-9iq08xqqblo6 CSeq: 4 ACK Max-Forwards: 70 m: <sip:91421@10.30.0.64:5060>;reg-id=1 l: 0 ------------------------------ Received from udp:10.1.10.80:5060 at 26/10/2011 10:22:42:130 (868 bytes): SIP/2.0 200 OK Via: SIP/2.0/UDP 10.30.0.64:5060 ;received=10.30.0.64;branch=z9hG4bK-5evtiw6dm0po;rport=5060 Record-Route: <sip:10.1.10.80;lr=on> From: "Virgil Menendez" <sip:91421@ser.gowireless.net>;tag=qi3i8ze6z8 To: <sip:9513261429@ser.gowireless.net;user=phone>;tag=as3f8c0f96 Call-ID: 3c2676547a8d-2t5yi6jok1sv CSeq: 2 INVITE Server: Asterisk PBX 1.8.7.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: <sip:9513261429@10.1.10.83:5060> Content-Type: application/sdp Content-Length: 256 v=0 o=root 1355451627 1355451627 IN IP4 10.1.10.83 s=Asterisk PBX 1.8.7.1 c=IN IP4 10.1.10.83 t=0 0 m=audio 16094 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv ------------------------------ Sent to udp:10.1.10.80:5060 at 26/10/2011 10:22:42:132 (385 bytes): ACK sip:9513261429@10.1.10.83:5060 SIP/2.0 v: SIP/2.0/UDP 10.30.0.64:5060;branch=z9hG4bK-wszafb7cbzpw;rport Route: <sip:10.1.10.80;lr=on> f: "Virgil Menendez" <sip:91421@ser.gowireless.net>;tag=qi3i8ze6z8 t: <sip:9513261429@ser.gowireless.net;user=phone>;tag=as3f8c0f96 i: 3c2676547a8d-2t5yi6jok1sv CSeq: 2 ACK Max-Forwards: 70 m: <sip:91421@10.30.0.64:5060>;reg-id=1 l: 0 ------------------------------ Received from udp:10.1.10.80:5060 at 26/10/2011 10:22:42:232 (503 bytes): BYE sip:91421@10.30.0.64:5060 SIP/2.0 Via: SIP/2.0/UDP 10.1.10.80;branch=z9hG4bKe723.bf70c1f4.0 Via: SIP/2.0/UDP 10.1.10.83:5060;branch=z9hG4bK69f53cf1 Max-Forwards: 69 From: <sip:9513261429@ser.gowireless.net;user=phone>;tag=as3f8c0f96 To: "Virgil Menendez" <sip:91421@ser.gowireless.net>;tag=qi3i8ze6z8 Call-ID: 3c2676547a8d-2t5yi6jok1sv CSeq: 102 BYE User-Agent: Asterisk PBX 1.8.7.1 *X-Asterisk-HangupCause: Protocol error, unspecified *X-Asterisk-HangupCauseCode: 111 Content-Length: 0 Regards, Rowell

2 1

Asterisk 1.8 and Kamailio 1.5 issue
by Rowell Rufino 26 Oct '11

26 Oct '11

Hi, We are having issues where the "OK" or "ACK" is that is coming from the phone is not relayed by OpenSER to Asterisk. Below is the sip trace... I am also attaching a tcpdump. Please help what we can do. Received from udp:10.1.10.80:5060 at 26/10/2011 10:22:41:476 (490 bytes): SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP 10.30.0.64:5060;received=10.30.0.64;branch=z9hG4bK-wowp1kmdy4rl;rport=5060 From: "Virgil Menendez" <sip:91421@ser.gowireless.net>;tag=6wkdms1r20 To: <sip:9513261429@ser.gowireless.net;user=phone>;tag=as0b87218f Call-ID: 3c26755bf15c-9iq08xqqblo6 CSeq: 4 INVITE Server: Asterisk PBX 1.8.7.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 ________________________________ Sent to udp:10.1.10.80:5060 at 26/10/2011 10:22:41:481 (387 bytes): ACK sip:vm9513261429@10.1.10.83:5060 SIP/2.0 v: SIP/2.0/UDP 10.30.0.64:5060;branch=z9hG4bK-wowp1kmdy4rl;rport Route: <sip:10.1.10.80;lr=on> f: "Virgil Menendez" <sip:91421@ser.gowireless.net>;tag=6wkdms1r20 t: <sip:9513261429@ser.gowireless.net;user=phone>;tag=as0b87218f i: 3c26755bf15c-9iq08xqqblo6 CSeq: 4 ACK Max-Forwards: 70 m: <sip:91421@10.30.0.64:5060>;reg-id=1 l: 0 ________________________________ Received from udp:10.1.10.80:5060 at 26/10/2011 10:22:42:130 (868 bytes): SIP/2.0 200 OK Via: SIP/2.0/UDP 10.30.0.64:5060;received=10.30.0.64;branch=z9hG4bK-5evtiw6dm0po;rport=5060 Record-Route: <sip:10.1.10.80;lr=on> From: "Virgil Menendez" <sip:91421@ser.gowireless.net>;tag=qi3i8ze6z8 To: <sip:9513261429@ser.gowireless.net;user=phone>;tag=as3f8c0f96 Call-ID: 3c2676547a8d-2t5yi6jok1sv CSeq: 2 INVITE Server: Asterisk PBX 1.8.7.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: <sip:9513261429@10.1.10.83:5060> Content-Type: application/sdp Content-Length: 256 v=0 o=root 1355451627 1355451627 IN IP4 10.1.10.83 s=Asterisk PBX 1.8.7.1 c=IN IP4 10.1.10.83 t=0 0 m=audio 16094 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv ________________________________ Sent to udp:10.1.10.80:5060 at 26/10/2011 10:22:42:132 (385 bytes): ACK sip:9513261429@10.1.10.83:5060 SIP/2.0 v: SIP/2.0/UDP 10.30.0.64:5060;branch=z9hG4bK-wszafb7cbzpw;rport Route: <sip:10.1.10.80;lr=on> f: "Virgil Menendez" <sip:91421@ser.gowireless.net>;tag=qi3i8ze6z8 t: <sip:9513261429@ser.gowireless.net;user=phone>;tag=as3f8c0f96 i: 3c2676547a8d-2t5yi6jok1sv CSeq: 2 ACK Max-Forwards: 70 m: <sip:91421@10.30.0.64:5060>;reg-id=1 l: 0 ________________________________ Received from udp:10.1.10.80:5060 at 26/10/2011 10:22:42:232 (503 bytes): BYE sip:91421@10.30.0.64:5060 SIP/2.0 Via: SIP/2.0/UDP 10.1.10.80;branch=z9hG4bKe723.bf70c1f4.0 Via: SIP/2.0/UDP 10.1.10.83:5060;branch=z9hG4bK69f53cf1 Max-Forwards: 69 From: <sip:9513261429@ser.gowireless.net;user=phone>;tag=as3f8c0f96 To: "Virgil Menendez" <sip:91421@ser.gowireless.net>;tag=qi3i8ze6z8 Call-ID: 3c2676547a8d-2t5yi6jok1sv CSeq: 102 BYE User-Agent: Asterisk PBX 1.8.7.1 X-Asterisk-HangupCause: Protocol error, unspecified X-Asterisk-HangupCauseCode: 111 Content-Length: 0 Regards, Rowell

1 0

Who is at Astricon?
by JR Richardson 26 Oct '11

26 Oct '11

I am. On 10/25/11, sr-users-request(a)lists.sip-router.org <sr-users-request(a)lists.sip-router.org> wrote: > Send sr-users mailing list submissions to > sr-users(a)lists.sip-router.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > or, via email, send a message with subject or body 'help' to > sr-users-request(a)lists.sip-router.org > > You can reach the person managing the list at > sr-users-owner(a)lists.sip-router.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of sr-users digest..." > > > Today's Topics: > > 1. Re: Dispatcher Confusion (v3.2.0) (Daniel-Constantin Mierla) > 2. Re: Dispatcher Confusion (v3.2.0) (Daniel-Constantin Mierla) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Wed, 26 Oct 2011 05:44:06 +0200 > From: Daniel-Constantin Mierla <miconda(a)gmail.com> > Subject: Re: [SR-Users] Dispatcher Confusion (v3.2.0) > To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List" <sr-users(a)lists.sip-router.org> > Message-ID: <4EA78206.4060203(a)gmail.com> > Content-Type: text/plain; charset="iso-8859-1"; Format="flowed" > > Hello, > > On 10/25/11 5:52 PM, Asgaroth wrote: >> Hi Daniel, >> >> Thanks for the clarrification, >> >> On 25/10/2011 16:30, Daniel-Constantin Mierla wrote: >>> >>> >>> 4.6. |ds_mark_dst("s")| >>> >>> Mark the last used address from destination set as inactive >>> ("i"/"I"/"0"), active ("a"/"A"/"1") or probing ("p"/"P"/"2"). With >>> this function, an automatic detection of failed gateways can be >>> implemented. When an address is marked as inactive or probing, it >>> will be ignored by 'ds_select_dst' and 'ds_select_domain'. >>> >> >> Above is the part that is a little misleading, it says that >> >> "When an address is marked as inactive or probing, it will be ignored >> by 'ds_select_dst' and 'ds_select_domain'." >> >> This, to me, means that if a gateway is in probing mode >> (Active-Probing) then it wont be selected in the destination set >> because it is in probing mode, if this is not the case then maybe the >> above line should read: >> >> "When an address is marked as inactive or inactive-probing, it will be >> ignored by 'ds_select_dst' and 'ds_select_domain'." >> >> This brings me to my next question then, how would I set the state of >> a destination to Inactive-Probing (state: IP) from the routing script. >> The ds_mark_dst() function only allows one of "a", "i", "p". >> >> If I do a ds_mark_dst("i"), then the state changes to "IX", Inactive >> with no probing to tell when gateway is back up. >> >> If I do a ds_mark_dst("i") and then right after ds_mark_dst("p"), a >> log is printed saying that you cannot put a destination into probing >> state when it is marked as inactive. > are you sure you run the devel version? There is no such case in the > sources -- send me exactly the log message you get. Only when the > destination is disabled the probing cannot be set, but not the same case > of inactive. > >> >> Is it possible to set the state of a gateway to inactive-probing from >> the routing script? > > Yes, it should be, no constraint in master branch source code. > > Daniel > >> >> Thanks >> >> >> >> _______________________________________________ >> 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 > Kamailio Advanced Training, Dec 5-8, Berlin: http://asipto.com/u/kat > http://linkedin.com/in/miconda -- http://twitter.com/miconda > >

2 1

sip-router listen address
by caio 25 Oct '11

25 Oct '11

Hello, Is there a problem or anything to consider when sharing the same ip/interface between sip proxy and asterisk? I have asterisk on port 5080 and SR at 5060 in the same interface with the same IP address and have a missing BYE when call is over. However, moving asterisk service to different interface and ip in the same box, the issue is solved. I appreciate your comments. Regards, Claudio

3 4

Kamailio.cfg file woes
by Peter Schrock 24 Oct '11

24 Oct '11

I am wondering two things. 1. is there a place on the web that I may find a basic cfg file to at least get Kamailio running. 2. is there a resource that tells you what to include for what you want to use the system for. At this point, it seems like I should just include everything because I don't know what I might not need. I looked at the references on the website and found the information a little overwhelming with all the options that are available. I think smaller bit sizes would be helpful at this time for me. Peter

3 3

onsend_route and replies
by Asgaroth 24 Oct '11

24 Oct '11

Hi All, Does the onsend route get executed for all messages? I'm just performing some xlog's in the onsend_route, but I'm not seeing it being called for replies. Currently my onsend_route looks like: onsend_route { xlog("onsend_route : $rm : Recieve Details $si:$sp -> $Ri:$Rp"); xlog("onsend_route : $rm : Destination Details $snd(af) $snd(proto):$snd(ip):$snd(port)"); } I can see it being executed for initial messages coming in but I dont see it executed for any replies. Thanks

1 1

kamctl dialplan addrule
by Arthur Alkhasyan 24 Oct '11

24 Oct '11

Hi all, I'm new to the list and new to the Kamailio. We are using Kamailio with Mysql to route calls between clients asterisks and SIP PSTN gateways. We have two inbound gateways and one outbound. I need to add outbound gateway and a few new features: 1. Call routing between carriers based at prefixes added by asterisks. I would like to route all calls with prefix 99101 to gateway 1 and prefix 99102 to gateway 2. Prefix have to be striped out before sending call to carrier. 2. Remote Call Forwarding. I have to setup RCF at Kamailio. All calls to the particular DID number have to be forward to another number. I found description for dialplan module and believe the module can work for me. I recompiled Kamailio with dialplan module support and added to config # ----- dialplan params ----- modparam("dialplan", "db_url", "mysql://openser:openserrw@localhost/openser") I did search and found this thread related to my problem: http://lists.sip-router.org/pipermail/sr-users/2009-November/025416.html I try to use kamctl to add rule to the MySQL and can't find proper syntax kamctl dialplan addrule 10 100 1 ^123+ 0 ^(123.+) 0030\1 -bash: syntax error near unexpected token `(' any help is greatly appreciated. Thank you.

3 3

Re: [SR-Users] module version mismatch for carrierroute
by Uri Shacked 24 Oct '11

24 Oct '11

Hi, i found out the mistake i made... didn't add the carrierroute to the include modules when making cfg file. thanks

1 0

module version mismatch for carrierroute
by Uri Shacked 24 Oct '11

24 Oct '11

Hi, i installed 3.2 from git. when starting kamailio i get the following: "ERROR: module version mismatch for /usr/local/lib/kamailio/modules/carrierroute.so; core: kamailio 3.2.0 (i386/linux); module: kamailio 3.1.4 (i386/linux)" any ideas what to do? thanks, Uri

3 2

Daily tarballs not being created anymore?
by Timo Reimann 24 Oct '11

24 Oct '11

Hi all, something caught my attention today: Apparently, tarballs provided from http://sip-router.org/tarballs/sr/ do not seem to be updated automatically anymore (the latest files are from March) although the Download page states that this should happen: "Tarballs for master (development) and several branches are made available daily (only if there were changes in the sources from previous tarball)" Is there a reason why tarball creation has ceased? If so, I think the Download page information should be corrected. I'd be prefer to have automatic tarballs again, however, as it's easier to point people not involved into development at tarballs instead of a set of git instructions. Thanks and cheers, --Timo

3 2

Kamailio 3.2 compile error on Solaris 10
by Nathaniel L Keeling 21 Oct '11

21 Oct '11

I am trying to compile the new version of Kamailio, 3.2 on Solaris in a 64bit environment and I am getting this error. I have been searching the internet but have not been able to find any solutions. Any help would be appreciated. CC (gcc) [M xmlrpc.so] xmlrpc.o In file included from ../../parser/msg_parser.h:63:0, from ../../sr_module.h:62, from xmlrpc.c:41: ../../parser/../ip_addr.h:86:23: error: field 'sin6' has incomplete type ../../parser/../ip_addr.h: In function 'ip_addr_loopback': ../../parser/../ip_addr.h:283:3: warning: implicit declaration of function 'htonl' [-Wimplicit-function-declaration] ../../parser/../ip_addr.h:286:3: warning: implicit declaration of function 'IN6_IS_ADDR_LOOPBACK' [-Wimplicit-function-declaration] ../../parser/../ip_addr.h: In function 'sockaddr2ip_addr': ../../parser/../ip_addr.h:353:50: error: dereferencing pointer to incomplete type ../../parser/../ip_addr.h: In function 'su_getport': ../../parser/../ip_addr.h:399:4: warning: implicit declaration of function 'ntohs' [-Wimplicit-function-declaration] ../../parser/../ip_addr.h: In function 'su_setport': ../../parser/../ip_addr.h:418:4: warning: implicit declaration of function 'htons' [-Wimplicit-function-declaration] xmlrpc.c: In function 'rpc_scan': xmlrpc.c:1454:6: warning: variable 'fmt_len' set but not used [-Wunused-but-set-variable] gmake[1]: *** [xmlrpc.o] Error 1 gmake: *** [modules] Error 1 Thanks Nathaniel

4 3

Force signalling??
by Skyler 20 Oct '11

20 Oct '11

Hi, I'm trying to setup an inbound proxy (Proxy-1) using dispatcher to relay between 2 other proxies (proxy-2 & 3) depending on DID, but I need to force all signalling through Proxy-1. so #2 & 3 only communicate with #1 and #1 only communicates up and down stream. My trouble is, the INVITES contain the original $ru $fu $tu & $ct info when passing down to Proxy-2 or 3. The result is, only partial signalling is being sent to Proxy-1. After the ACK it seems that Proxy-1 is removed from the communication path entirely as $ru becomes $rU@proxy-2 in the ACK from upstream so Proxy-1 is out of the picture from then on. This means BYE's are not working as Proxy-2 & 3 do not have authentication upstream. I suspect that I need to re-write $ru $fu $tu & $ct info on the original INVITE to reflect Proxy-1 information before dispatching. Is that what I'm missing? If so, how would I restore the original INVITE info to send upstream again on replies etc.? Any help is greatly appreciated. Tia, Skyler

3 6

New developer: Mészáros Mihály
by Daniel-Constantin Mierla 20 Oct '11

20 Oct '11

Hello, just to let everyone know that a new person got today write access to GIT repository: Mészáros Mihály (cc-ed), having the user id misi. Welcome to development team and looking forward to your contributions! Involved in the mailing lists in the past, he submitted the first new module to be part of next major release: presence_profile, implementation of an extension about sip user agent configuration framework, specified in RFC6080: * http://tools.ietf.org/html/rfc6080 The code of the new module is now in GIT branch misi/ua-profile, to be merge soon to master after initial review are done. Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Dec 5-8, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 0

Build version in custom server_header
by Asgaroth 20 Oct '11

20 Oct '11

Hi All, Is it possible to have the build/git version added in a custom server_header/useragent_header? For example, my kamailio version is: version: kamailio 3.2.0 (i386/linux) 94d3b8 if I had the following config options: server_header="Server: Proxy" useragent_header="User-Agent: Proxy" Is it possible to do something like: server_header="Server: Proxy ($buildver)" useragent_header="User-Agent: Proxy ($buildver)" Where "$buildver" is the build/git release (94d3b8)? I'm looking at the core cookbook but I dont see anything that jumps out at me, is there a variable that can do this? Thanks

2 2

ACK routing loop
by Antanas Masevicius 20 Oct '11

20 Oct '11

Hello, after troubleshooting ACK routing loops i came up with a thread: http://www.openser.org/pipermail/users/2009-March/004123.html In short, loops formed when i entered IP address to 'domain' table of a host which was remote. Call flow: UAC INVITE -> Kamailio (with remote domain entered) -> UAS (host whose IP matches that remote domain) It looks like after loose_route() check ACK to 200 OK gets routed to local host. Besides this issue, IP address of remote host in 'domain' table did no harm at all. Question is, is that normal ACK routing behaviour or loose_route() should still check if given IP in request is really bound to local interface? Domain module documentation states that "A “local” domain is one that the proxy is responsible for", but id does not tell you that you cannot set remote IP address as a local domain. Please advice. Best regards, Antanas Masevicius

2 4

Dispatcher Confusion Redeux
by Asgaroth 20 Oct '11

20 Oct '11

Hi All, I'd like to clear something up with the diapatcher module and it's probing parameters: First things first: ./kamailio -V version: kamailio 3.1.5 (i386/linux) 76fff5 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, PKG_SIZE 4MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 76fff5 compiled on 16:21:02 Oct 19 2011 with gcc 4.1.2 I think I am suffering from the same confusion as Klaus in the following archive message: http://web.archiveorange.com/archive/v/jZFTG61DlcEr9LJbS8Gq I have the following parameters set for the dispatcher module: modparam("dispatcher", "ds_ping_interval", 10) modparam("dispatcher", "ds_probing_threshhold", 1) modparam("dispatcher", "ds_probing_mode", 0) When I first start kamailio all gateways are active, no probing occurs. If I then set a gateway inactive (ds_set_state i 1 sip:...), no probing occurs. If I then set same gateway active (ds_set_state a 1 sip:...), no probing occurs. If I mark same gateway as probing (ds_mark_dst("p") after 1 attempt as specified in module parameter), then probing starts. If I then set same gateway inactive (ds_set_state i 1 sip:...) probing still occurs. According to the docs if I set the state of a gateway to inactive then no probing should occur. From my understanding of the docs, if I set ds_probing_mode to 0 then only gateways marked as probing will be "pinged". Can someone with more experienced than myself please clarify how this method is supposed to work. Thanks

3 3

Re: [SR-Users] Kamailio v3.2.0 Released - new major version is
by Javier Gallart 20 Oct '11

20 Oct '11

Hello congratulations to all the people who has contributed! I'm looking forward to try it out, some of the new features look really promising... Regards Javi > ------------------------------ > > Message: 2 > Date: Tue, 18 Oct 2011 20:30:22 +0200 > From: Daniel-Constantin Mierla <miconda(a)gmail.com> > Subject: [SR-Users] Kamailio v3.2.0 Released - new major version is > out > To: kamailio users <sr-users(a)lists.sip-router.org>, sr-dev > <sr-dev(a)lists.sip-router.org>, business(a)lists.kamailio.org > Message-ID: <4E9DC5BE.8080003(a)gmail.com> > Content-Type: text/plain; charset=windows-1252; format=flowed > > Kamailio (OpenSER) SIP Server v3.2.0 is out ? a major release with a > very large number of new features and improvements. > > v3.2.0 release is a result of more than 10 months of development and 2 > months of testing from the teams of Kamailio (OpenSER) and SIP Express > Router (SER) projects. In the year of the 10th celebration of our > project (http://sip-router.org/10-years-ser/), this version comes with > 12 brand new modules in addition to a lot of fresh features in core and > old modules. > > Full release notes, including links for download and documentation, are > available at: > > * http://www.kamailio.org/w/kamailio-openser-v3-2-0-release-notes/ > > Many thanks to all developers and community members that made possible > this release. > > Enjoy SIP routing in a more flexible and easier way with Kamailio v3.2.0! > > -- > Daniel-Constantin Mierla -- http://www.asipto.com > > > > >

2 1

Kamailio v3.2.0 RPM packages
by Daniel-Constantin Mierla 20 Oct '11

20 Oct '11

Hello, forgot to announce explicitly that the RPM packages for CentOS, RedHat, Fedora and OpenSUSE were built quickly after the release. They can be downloaded from OpenSUSE build system portal: Links to download packages for each OS are available in the wiki: * http://www.kamailio.org/wiki/packages/rpms Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Dec 5-8, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 0

kamctl version
by caio 20 Oct '11

20 Oct '11

Hello, Is kamctl utility version independent of the kamailio binary version? # kamailio -V version: kamailio 3.2.0 (x86_64/linux) 94d3b8 # kamctl version /usr/sbin/kamctl 3.1.0 Just a typo i guess. Regards, Claudio

2 1

kamailio and high-available db backend
by Juha Heinanen 19 Oct '11

19 Oct '11

mino, instead of ndb cluster, you can use drbd together with heartbeat and pacemaker to provide high available mysql/sip server solution. -- juha

1 0

very basic handling for ua-profile event
by MÉSZÁROS Mihály 19 Oct '11

19 Oct '11

Hi, Please find attached a new _very_ basic presence modul to make available user profile provisioning as it stated in RFC6080. Best Regards, Misi

2 3

kamailio and high-available db backend
by Mino Haluz 19 Oct '11

19 Oct '11

Hi, I was using ndb cluster as the dabatase backend for my kamailio setup. The thing is, mysql ndb cluster is difficult to maintain for me and it crashed multiple times (desynchronization, datamemory limit reached, and other strange reasons) and it is not that well documented (the user-base is quite narrow). My question is, what mysql architecture would you recommend me, if it has to be high available, well documented with various examples wihitn the community, thus easy to maintain. DB has 5% writes, 100queries/sec. and its size is 100MB. Mino

1 0

ok
by abdelazize rhosni 18 Oct '11

18 Oct '11

ok

2 1

Kamailio Dispatcher fails in front of FreeSwitch, help (again).
by Henrik Aagaard Sørensen 18 Oct '11

18 Oct '11

I've asked a lot of question about Kamailio Dispatcher in front of FreeSwitch in this mailing list. So I'm sorry that I'm still "spamming", but it still fails. This is my status so far. When connecting 2 devices directly to FreeSwitch everything works. When connecting to Kamailio, some calls fails with "Timeout" and some fails with "Too many loops". This is my kamailio.cfg: http://pastebin.com/KEuZmikh When connected directly to FreeSwitch where everything works, this is how my clients are registrered in FreeSwitch: http://pastebin.com/prkjkb7g When connected to Kamailio where nothing works, this is how my clients are registrered in FreeSwitch: http://pastebin.com/b5x6NZyv 215.197.145.132 (sip.my-domain.com) is my Kamailio. pbx.my-domain.com is my FreeSwitch. 96.224.14.164 is my IP at home, where my 2 devices are from. Can anyone help me out? I'm kinda lost at this point.

3 11

master branch is now open for commits of new features to be in v3.3.0
by Daniel-Constantin Mierla 18 Oct '11

18 Oct '11

Hello, this is a short notice to let everyone know that the GIT master branch is open for commits of new features -- to be part of next major release v3.3.0, as v3.2.0 has been officially released few moments ago. From this moment on, master and 3.2 branches will diverge, 3.2 taking in only bug fixes. So if you want to use stable release of existing features, switch to branch 3.2. You can find guidelines of installing branch 3.2 in the wiki site: * http://www.kamailio.org/wiki/install/3.2.x/git I spotted already a new module by one of the community members submitted to the mailing list in the last days, related to presence extensions, therefore looks like is going to be another great development cycle until 3.3.0 will be out. Happy coding SIP routing extensions in SER and Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Dec 5-8, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 0

Kamailio v3.2.0 Released - new major version is out
by Daniel-Constantin Mierla 18 Oct '11

18 Oct '11

Kamailio (OpenSER) SIP Server v3.2.0 is out – a major release with a very large number of new features and improvements. v3.2.0 release is a result of more than 10 months of development and 2 months of testing from the teams of Kamailio (OpenSER) and SIP Express Router (SER) projects. In the year of the 10th celebration of our project (http://sip-router.org/10-years-ser/), this version comes with 12 brand new modules in addition to a lot of fresh features in core and old modules. Full release notes, including links for download and documentation, are available at: * http://www.kamailio.org/w/kamailio-openser-v3-2-0-release-notes/ Many thanks to all developers and community members that made possible this release. Enjoy SIP routing in a more flexible and easier way with Kamailio v3.2.0! -- Daniel-Constantin Mierla -- http://www.asipto.com

1 0

branch 3.2 was created
by Daniel-Constantin Mierla 18 Oct '11

18 Oct '11

Hello, short notice to let everyone know that the branch for 3.2.x release series was created with the name 3.2. For example, downloading sources from GIT and building Kamailio flavour form this branch can be done with: git clone --depth 1 git://git.sip-router.org/sip-router kamailio cd kamailio git checkout -b 3.2 origin/3.2 make FLAVOUR=kamailio cfg make all make install The official release of 3.2.0 will follow just a bit later today. Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Dec 5-8, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 0

YACC error during Compile of Kamailio 3.1.5
by Nathaniel L Keeling 18 Oct '11

18 Oct '11

I am building Kamailio 3.1.5 on Solaris 10 64bit environment and I am getting this error. yacc -d -b cfg cfg.y "cfg.y", line 104: fatal: invalid escape, or illegal reserved word: expect gmake: *** [cfg.tab.h] Error 1 I tried compiling with bison, but got errors immediately after getting thru the yacc (I just issued ln -s /opt/csw/bin/bison /opt/csw/bin/yacc and recompiled) flex cfg.lex CC (gcc) [kamailio] lex.yy.o In file included from ip_addr.h:50:0, from globals.h:37, from cfg.lex:91: ut.h: In function 'str2s': ut.h:195:17: warning: variable 'init' set but not used [-Wunused-but-set-variable] cfg.lex: In function 'yylex': cfg.lex:561:11: error: 'yylval' undeclared (first use in this function) cfg.lex:561:11: note: each undeclared identifier is reported only once for each function it appears in cfg.lex:561:40: error: 'FORWARD' undeclared (first use in this function) cfg.lex:562:40: error: 'FORWARD_TCP' undeclared (first use in this function) cfg.lex:563:40: error: 'FORWARD_TLS' undeclared (first use in this function) Thanks Nathaniel L Keeling

1 0

Make all error
by Peter Schrock 17 Oct '11

17 Oct '11

I am doing a "make all" and am getting this error: error: mysql/mysql.h: No such file or directory I have mysql installed and know that kamailio is having trouble finding mysql header files. Is there a configuration I can set up to help kamailio find it? Is there a resource I can be directed to find the info? Peter

3 5

preparing to release v3.2.0
by Daniel-Constantin Mierla 17 Oct '11

17 Oct '11

Hello, tomorrow is the day to release the new major version 3.2.0. Since involves creation of a new branch on GIT repository and some adjustments to the packaging, the plan is to start in the morning with the preparations. If any developer has something left to commit, please do it before 9:00am GMT. Afterwards it is very likely that the branch 3.2 will be created and there will be need for backport (which is fine as well). Then by 14:00 GMT the 3.2 branch should be frozen completely to start packaging the release. From than moment on until the release is announced, please send an email to sr-dev in case anyone wants to commit to branch 3.2. Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Dec 5-8, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 0

Organization in SIP header
by Skyler 17 Oct '11

17 Oct '11

Hello all, Is there a way to set Organization: within SIP header? I know how to change user_agent_header and server_header but I'd rather not do that. Also I know I could do append_hf, but is there an organization_header? Thanks, Skyler

3 6

Problem with REGISTER retransmissions
by Alejandro Mingo 17 Oct '11

17 Oct '11

Hi, Since we installed last version of Kamailio (3.1.4) we have been experiencing a big problem with REGISTER retransmissions. When the server receives a retransmitted REGISTER it removes the binding and the UAC remains unregistered until next refreshing period. I include a PCAP capture that shows this behaviour. I am not 100% sure that this problem never happened with older versions. Has anybody experienced the same problem? Thanks in advanced. Alejandro.

3 9

Re: [SR-Users] OPENSER MIB
by Daniel-Constantin Mierla 17 Oct '11

17 Oct '11

Hello Stefan, On 3/14/11 11:03 AM, Stefan Tiedje wrote: > Thanks for the answer. > Maybe I have some older versions of the OPENSER-MIB and the other > related MIB's since I could not find the counter you pointed at. I'm > using a MIB browser for reading the MIB's. > Is the suggested counter "expired dialogs" added in a specific release > of Kamailio? Which? We use Kamailio 3.0.2. I used Kamailio and recommend using it sine it has the latest commits for stability. However, what I wrote before is pretty much not related to the version. There is a counter that tracks the processed dialogs, but seems it is not exported by default through snmpstats module. The statistics counter is named "processed_dialogs", implemented by dialog module. You can dump all internal statistics through kamctl or via xmlrpc command, but probably to export it through snmpstats you may need to extend the mibs and the code of the module. I just grepped the sources of snmpstats module to see what dialog statistics it is exporting: $ grep -n _dialogs modules_k/snmpstats/* | grep get_statistic modules_k/snmpstats/alarm_checks.c:83: num_dialogs = get_statistic("active_dialogs"); modules_k/snmpstats/snmpObjects.c:404: int result = get_statistic("active_dialogs"); modules_k/snmpstats/snmpObjects.c:424: get_statistic("active_dialogs") - modules_k/snmpstats/snmpObjects.c:425: get_statistic("early_dialogs"); modules_k/snmpstats/snmpObjects.c:443: int result = get_statistic("early_dialogs"); modules_k/snmpstats/snmpObjects.c:459: int result = get_statistic("failed_dialogs"); modules_k/snmpstats/snmpObjects.c:508: int num_dialogs = get_statistic("active_dialogs"); Perhaps when the snmpstats was developed the dialog module didn't export the statistics counter of "processed_dialogs" and then it was not updated. Now, what I tried to say is that if the "processed_dialogs" counter is not available through snmpstats (and it is not now after grepping the sources) you can get its value from another application through "kamctl get_statistics all" or XMLRPC command for all of the existing kamailio releases. Upcoming one we will look to implement the export through snmpstats as well. If you have time to do it and send us a patch, we will gladly commit it to source tree in our GIT repository. Cheers, Daniel > Do you have the MIB name for the "expired dialogs" counter. I will > look for that in my version of OPENSER MIBS. > Important, do you have a link to where MIB files can be downloaded for > Kamailio 3.0.2? > Below follows an excerp from one of the MIB's. Is it old, I don't know? > > -- *********************************************************************** > > -- OPENSER-MIB: OPENSER MIB > > -- > > -- Date of Creation: Januay 2006 > > -- > > -- This MIB provides information related to the OpenSER SIP Router. > > -- > > -- Copyright (c) The Internet Society (2006) > > -- Ammendments (c) Soma Networks, Inc. (2006) > > -- > > -- All rights reserved. > > -- ***************************************************************** > > /Stefan > > ------------------------------------------------------------------------ > *From:* Daniel-Constantin Mierla [mailto:miconda@gmail.com] > *Sent:* den 14 mars 2011 10:16 > *To:* Stefan Tiedje > *Cc:* sr-users(a)lists.sip-router.org > *Subject:* Re: [SR-Users] OPENSER MIB > > Hello, > > On 3/14/11 9:42 AM, Stefan Tiedje wrote: >> Hi, >> In the Kamailio OPENSER-MIB there is the counter >> "openserTotalNumFailedDialogSetups". This is a Counter32. >> The description is: >> "The total number of calls that failed with an error. The >> following codes define a failed call:" >> *Question:* >> >> * I'm looking for the corresponding counter to >> "openserTotalNumFailedDialogSetups" who counts successful >> Dialog setups of Counter32 type. Does it exist? >> * If not, does it exist a work around? >> * Where in the code can the new suggested counter be added? >> * Something else???? >> > > the dialog module counts the number of processed dialogs, see: > http://kamailio.org/docs/modules/stable/modules_k/dialog.html#id2966360 > > There is no counter currently inside dialog module exporting exactly > the number of successfully setup dialogs, it should not be hard to do > it, though. Using the above and the number of failed and expired > dialogs, you can actually get the number of successful dialogs. > > Dialog module being the one that tracks SIP dialogs, therefore being > able to count them, now I don't know if snmpstats module exports all > the counters from dialog module. I setup snmpstats just few weeks ago > and works perfect on Ubuntu/Debian servers, but I had no need to check > dialog module counters. > > Note that you can get the list of all internal statistics via kamctl: > - kamctl fifo get_statistics all > > Or via XMLRPC if you need them remotely in another application. > > Another option is to define your statistics with statistics module. > Knowing that in SIP a successful call dialog means 200ok reply to an > INVITE transaction, you can count it in the onreply_route[abc] that > you arm for relayed transactions with t_on_reply("abc"). > > Hope these help you, > Daniel > >> Suggestion for the new counter is a name like: >> "openserTotalNumSucceededDialogSetups". It has a counter32. >> Description: "The total number of calls that succeeded" >> I know that there are the counters openserCurNumDialogs, >> openserCurNumDialogsInProgress and openserCurNumDialogsInSetup but >> these are of Gauge type who only reflects the current situation. >> These Gauge counters can't be used together with a Counter32 counter. >> That don't mix. The calculation done for the counter >> "openserCurNumDialogsInProgress" should be used where every new >> dialog setup is added to the new suggested counter. A counter of 32 >> should cover a great deal of connections. These counters are usually >> read, if used, every 15 minutes or 1 hour. >> *Rationale:* >> The reason for the new counter is that a calculation between >> succeeded and failed dialog setups can be done and be used for SLA >> agreements. Without this, its hard to make any customer versus >> provider agreements. >> /Stefan >> PS. Ask if anything is unclear and I need an answer rapidly. >> >> >> _______________________________________________ >> 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 -- Daniel-Constantin Mierla http://www.asipto.com

2 5

cleaning up contacts...
by Krishna Kurapati 16 Oct '11

16 Oct '11

Hi, I am seeing several Contacts for same AOR in userlocation when I do "kamctl ul show". I suspect, it's a result of UA crashing after registering and then registering with different contact after restart. Since the UA crashed, it could not send Register with expire=0 to remove the previous contact. Since the userlocation API is not exported to configuration scripting, I am wondering how to clean any spurious contacts. Thanks Krish Kura

2 1

FIFO error in cfg file
by Peter Schrock 16 Oct '11

16 Oct '11

I tried opening with "kamctl moni" and I get this error message: ERROR: Error opening Kamailio's FIFO FIFO ERROR: Make sure you have the line 'modparam("mi_fifo", "fifo_name", "FIFO")' in your config ERROR: and also have loaded the mi_fifo module. I looked at the cfg file and it looks like what it is asking for is already there. Any suggestions would be welcomed. Peter

2 1

Force socket on reply
by Jijo 14 Oct '11

14 Oct '11

Hi All, Our system has two IP address, one is used for kamailio and 2nd one is used for data. The problem is suppose a request came in to kamailio on a TCP connection with first IP address and connection is torned down before sending the response. Later when the response is send out kamialio is using the 2nd IP to create the connection. The reason is kamailio doesn't have force socket in in reply, so it uses INADDR_ANY for soruce addres .so kernel uses source address based on the detaintion IP. I tired to force the socket using pseudo variable $fs or force_send_socket(), but neither of them worked from reply_route. So i modfied the code relay_replay to set the SND_F_FORCE_SOCKET which will use the address where request is recieved as source address to make the TCP connection The code change is shown below. Please let me know is there anyother way we could acheive it? }else{ buf = build_res_buf_from_sip_res( relayed_msg, &res_len ); /* if we build a message from shmem, we need to remove via delete lumps which are now stirred in the shmem-ed structure */ if (branch!=relay) { free_via_clen_lump(&relayed_msg->add_rm); } /* update send_flags with possible additions from the reply route */ SND_FLAGS_OR(&uas_rb->dst.send_flags, &uas_rb->dst.send_flags, &relayed_msg->rpl_send_flags); /* Make the response to use the same IP where it receives the message */ * * * uas_rb->dst.send_flags.f |= SND_F_FORCE_SOCKET; * } Thanks JIjo

2 4

Cannot concect to iptel.org
by Enda Mannion 14 Oct '11

14 Oct '11

Hi, I am trying to connect to the iptel.org SIP server using the recommended jitsi SIP UA. Is the service still up and running? I am using my newly created username and password but I get no response back from the REGISTER I send. Enda

1 0

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