sr-users July 2012

sr-users@lists.kamailio.org

64 participants
92 discussions

Does kamailio has some headers optimizing mechanism ?
by Konstantin M. 17 Jul '12

17 Jul '12

Hello, I have a following part of code: if (route(Test_TO_Dialer)) { append_hf("P-hint: Loose Routing. Re-INVITE Detected: Uplink --> Originator.\r\n"); xlog("********** loose_route(): Detected Re-INVITE from Uplink! **********\n"); setflag(1); # do accounting append_hf("P-hint: SDP was mangled\r\n"); python_exec("ParseSDP", $mb); fix_nated_contact(); setbflag(6); # Mark as NATed rtpproxy_manage(); dlg_manage(); } But in a logs I see: P-hint: Loose Routing. Re-INVITE Detected: Uplink --> Originator., SDP was mangled but it should be like this: P-hint: Loose Routing. Re-INVITE Detected: Uplink --> Originator. P-hint: SDP was mangled Does kamailio has some headers optimizing mechanism ? # ./kamailio -V version: kamailio 3.2.3 (x86_64/linux) flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 4MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled on 15:37:11 Jun 9 2012 with gcc 4.4.3

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How to use Kamailio
by piyush singhai 16 Jul '12

16 Jul '12

Hello I am very new in opensip. i am trying to install kamailio and freeswitch...kamailio is running fine...but i donot know how kamailio will communicate to my local sip server. kamctl start and stop ...starting and stopping the kamailio. i have installed kamalio from http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-3.1.x-from-git --Piyush

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Interested Sip Server
by Tony Rosa 15 Jul '12

15 Jul '12

I am interested sip server software for residential application Tony Andrade IP Solution 617-386-9186

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Re: [SR-Users] : Adding "make menuconfig"
by Anton Kvashenkin 15 Jul '12

15 Jul '12

Or we can just "port" Opensips's "make menuconfig" :) 15.07.2012 2:05 пользователь "aft" <aftnix(a)gmail.com> написал: > Hi, > > Kamailio is a modular stack. So inclusion or exclusion of a module is > very important. At this moment editing modules.lst is very *primitive* > way of doing things. > > I like menuconfig style ncurses based interface. Which gives a very > clear view of "dependency" and "which and which" information about the > modules. > > As number of modules increases, most of us just compile most of them, or > take the default modules.lst as granted. > > If a menuconfig target is added, it should be easier to customize a > kamailio build. > > Adding a menuconfig target seems a trivial task for anyone who know > kamailio build system well plus some intro to ncurses programming. > > So i guess devs should look into it. > > cheers > aft > > > _______________________________________________ > 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 >

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Adding "make menuconfig"
by aft 15 Jul '12

15 Jul '12

Hi, Kamailio is a modular stack. So inclusion or exclusion of a module is very important. At this moment editing modules.lst is very *primitive* way of doing things. I like menuconfig style ncurses based interface. Which gives a very clear view of "dependency" and "which and which" information about the modules. As number of modules increases, most of us just compile most of them, or take the default modules.lst as granted. If a menuconfig target is added, it should be easier to customize a kamailio build. Adding a menuconfig target seems a trivial task for anyone who know kamailio build system well plus some intro to ncurses programming. So i guess devs should look into it. cheers aft

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Kamailio v3.1.6 released - the last packaging in 3.1.x series
by Daniel-Constantin Mierla 14 Jul '12

14 Jul '12

Hello, this morning I packaged the 3.1.6 to mark the last release in the 3.1.x series. At this moment, the latest stable series are 3.2.x and 3.3.x, so there is no intention of packing again a 3.1.x version -- although patches may still go in branch 3.1. Notes about v3.1.6 release were posted on the news section of the website: * http://www.kamailio.org/w/2012/07/kamailio-v3-1-6-released/ Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw

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Kamailio + dialog + dispatcher (limiting calls)
by George Breahna 13 Jul '12

13 Jul '12

I am trying to limit the total number of concurrent calls processed by Kamailio and I'm having a world of trouble. We are running Kamailio 3.2.3 on Debian Squeeze with the dispatcher module and recently enabled the dialog module. The call flow is extremely simple. We have two Asterisk servers to which we load balance the incoming calls to, nothing more. Our dialog configuration is the default kamailio config, with the following additions: modparam("dialog", "enable_stats", 1) modparam("dialog", "dlg_flag", FLD_DLG) modparam("dialog", "hash_size", 4096) modparam("dialog", "profiles_with_value", "callquota") modparam("dialog", "default_timeout", 300) modparam("dialog", "dlg_match_mode",2) modparam("dialog", "detect_spirals", 1) modparam("dialog", "db_mode", 0) in request_route we have: if ( method=="INVITE" ) { t_on_failure("FAIL_ONE"); ds_select_dst("1","4"); # sl_send_reply("100","Trying"); t_reply("100", "Trying"); dlg_manage(); setflag(FLT_ACC); # do accounting $var(SIZE) = 0; get_profile_size("callquota", "$rU", "$var(SIZE)"); if( $var(SIZE) >= MAX_NUMBER_OF_CALLS ){ xlog("DEBUG: Simultaneous calls limit reached for $rU: $var(SIZE)\n"); sl_send_reply("503", "Simultaneous calls limit reached"); exit; } set_dlg_profile("callquota","$rU"); if (get_profile_size("callquota","$var(SIZE)")) { xlog("DEBUG: there are $var(SIZE) total calls for $rU\n"); } forward();#uri:host, uri:port); exit(); } We also added a dlg_manage() in WITHINDLG if (is_method("BYE")) { dlg_manage(); setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } The problem is that some dialogs never get erased. Their total number always increases and their state is always 1 kamctl fifo profile_list_dlgs callquota dialog:: hash=2161:28334339 state:: 1 ref_count:: 1 timestart:: 0 timeout:: 0 In the debug logs I see these entries for the dialog module: ### Starting Call Jul 12 22:38:34 debian1 /usr/sbin/kamailio[25452]: ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=515 a=25 n=dlg_manage Jul 12 22:38:34 debian1 /usr/sbin/kamailio[25452]: DEBUG: dialog [dlg_hash.c:523]: no dialog callid='1560428909_26358995(a)4.55.13.227' found Jul 12 22:38:34 debian1 /usr/sbin/kamailio[25452]: DEBUG: dialog [dlg_hash.c:523]: no dialog callid='1560428909_26358995(a)4.55.13.227' found Jul 12 22:38:34 debian1 /usr/sbin/kamailio[25452]: DEBUG: dialog [dlg_hash.c:553]: no dialog callid='1560428909_26358995(a)4.55.13.227' found Jul 12 22:38:34 debian1 /usr/sbin/kamailio[25452]: DEBUG: dialog [dlg_hash.c:309]: new dialog on hash 2161 Jul 12 22:38:34 debian1 /usr/sbin/kamailio[25452]: DEBUG: dialog [dlg_handlers.c:271]: route_set <sip:1.2.3.4;lr=on>, contact sip:+118884936245@4.55.13.227:5060, cseq 402 and bind_addr udp:10.10.10.1:5060 Jul 12 22:38:34 debian1 /usr/sbin/kamailio[25452]: DEBUG: dialog [dlg_hash.c:583]: ref dlg 0x7feb80ef0ae8 with 1 -> 1 Jul 12 22:38:34 debian1 /usr/sbin/kamailio[25452]: DEBUG: dialog [dlg_hash.c:603]: ref dlg 0x7feb80ef0ae8 with 1 -> 2 Jul 12 22:38:34 debian1 /usr/sbin/kamailio[25452]: DEBUG: dialog [dlg_hash.c:621]: unref dlg 0x7feb80ef0ae8 with 1 -> 1 Jul 12 22:38:34 debian1 /usr/sbin/kamailio[25452]: ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=525 a=27 n=set_dlg_profile Jul 12 22:38:34 debian1 /usr/sbin/kamailio[25452]: DEBUG: dialog [dlg_hash.c:459]: ref dlg 0x7feb80ef0ae8 with 1 -> 2 Jul 12 22:38:34 debian1 /usr/sbin/kamailio[25452]: DEBUG: dialog [dlg_hash.c:461]: dialog id=28334339 found on entry 2161 Jul 12 22:38:34 debian1 /usr/sbin/kamailio[25452]: DEBUG: dialog [dlg_hash.c:621]: unref dlg 0x7feb80ef0ae8 with 1 -> 1 Jul 12 22:38:34 debian1 /usr/sbin/kamailio[25452]: DEBUG: dialog [dlg_hash.c:514]: ref dlg 0x7feb80ef0ae8 with 1 -> 2 Jul 12 22:38:34 debian1 /usr/sbin/kamailio[25452]: DEBUG: dialog [dlg_hash.c:517]: dialog callid='1560428909_26358995(a)4.55.13.227' found#012 on entry 2161, dir=1 Jul 12 22:38:34 debian1 /usr/sbin/kamailio[25452]: CRITICAL: dialog [dlg_hash.c:650]: bogus event 6 in state 1 for dlg 0x7feb80ef0ae8 [2161:28334339] with clid '1560428909_26358995(a)4.55.13.227' and tags 'gK0251b91f' '' Jul 12 22:38:34 debian1 /usr/sbin/kamailio[25452]: DEBUG: dialog [dlg_hash.c:808]: dialog 0x7feb80ef0ae8 changed from state 1 to state 1, due event 6 (ref 2) Jul 12 22:38:34 debian1 /usr/sbin/kamailio[25452]: DEBUG: dialog [dlg_hash.c:621]: unref dlg 0x7feb80ef0ae8 with 1 -> 1 ### Hanging up now Jul 12 22:38:46 debian1 /usr/sbin/kamailio[25456]: DEBUG: tm [uac.c:182]: DEBUG: dlg2hash: 29877 Jul 12 22:38:46 debian1 /usr/sbin/kamailio[25456]: DEBUG: tm [uac.c:182]: DEBUG: dlg2hash: 29874 Jul 12 22:38:46 debian1 /usr/sbin/kamailio[25455]: DEBUG: dialog [dlg_hash.c:514]: ref dlg 0x7feb80ef0ae8 with 1 -> 2 Jul 12 22:38:46 debian1 /usr/sbin/kamailio[25455]: DEBUG: dialog [dlg_hash.c:517]: dialog callid='1560428909_26358995(a)4.55.13.227' found#012 on entry 2161, dir=1 Jul 12 22:38:46 debian1 /usr/sbin/kamailio[25455]: CRITICAL: dialog [dlg_hash.c:650]: bogus event 7 in state 1 for dlg 0x7feb80ef0ae8 [2161:28334339] with clid '1560428909_26358995(a)4.55.13.227' and tags 'gK0251b91f' '' Jul 12 22:38:46 debian1 /usr/sbin/kamailio[25455]: DEBUG: dialog [dlg_hash.c:808]: dialog 0x7feb80ef0ae8 changed from state 1 to state 1, due event 7 (ref 2) Jul 12 22:38:46 debian1 /usr/sbin/kamailio[25455]: DEBUG: dialog [dlg_hash.c:621]: unref dlg 0x7feb80ef0ae8 with 1 -> 1 Jul 12 22:38:46 debian1 /usr/sbin/kamailio[25455]: ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=625 a=25 n=dlg_manage Jul 12 22:38:46 debian1 /usr/sbin/kamailio[25455]: DEBUG: dialog [dlg_hash.c:459]: ref dlg 0x7feb80ef0ae8 with 1 -> 2 Jul 12 22:38:46 debian1 /usr/sbin/kamailio[25455]: DEBUG: dialog [dlg_hash.c:461]: dialog id=28334339 found on entry 2161 Jul 12 22:38:46 debian1 /usr/sbin/kamailio[25455]: DEBUG: dialog [dlg_hash.c:621]: unref dlg 0x7feb80ef0ae8 with 1 -> 1 Is my config wrong ? Am I missing anything here ? Any advice would be much appreciated Thank you, George

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what exactly is "shmeme:fragments" ?
by Uri Shacked 13 Jul '12

13 Jul '12

Hi, can someone please explain a little about the "shmem" params that i see when doing "kamctl fifo get_statistics shmeme:" BR, Uri

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failure route correct..?
by Gertjan Wolzak 13 Jul '12

13 Jul '12

Hello All, I am trying to get a failure route to work, I have got it working partially. When a call comes in, first I check the db_alias, if that is positive I do a lookup location and relay if the location is valid. But sometimes the sip client is still registered in the location table, but not connected anymore, mainly with wifi connected clients. So, I have got the time out on 3 seconds (fr_timer). When that hits I have configured the following failure route: failure_route[NOTONLINE] { xlog("SCRIPT: Notonline failure route\n"); t_on_failure("STOP"); if (t_is_canceled()) { exit; } if (t_check_status("408")) { xlog("SCRIPT: Status is time out"); $rU = $avp(orig_called); /( called number and alias id not equal, so have to revert the rU back to the called number)/ prefix("9993"); /( needed to get the right manipulation done within asterisk)/ xlog("SCRIPT: uri is $ru"); $ru = "sip:" + $rU + "@w.x.y.x:5060"; xlog("SCRIPT: uri is $ru"); (w.x.y.z ip address of the asterisk box) append_branch(); t_relay_to_udp("w.x.y.x","5060"); break; } } I am not sure if the above is correct. I have based this on an old "voicemail" failure route I could find. It is working correct, the call is forwarded to an Asterisk box, where some manipulation is done and then send to an pstn gateway. The only problem I have is one way audio. RTP from the called number reaches the callee but not vice versa. Now I am wondering, can that be caused by the failure route, or should I be looking in another direction? Hope someone can give me a pointer. Thanks. Gertjan Wolzak

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preparing release of v3.1.6
by Daniel-Constantin Mierla 13 Jul '12

13 Jul '12

Hello, later today I plan to package v3.1.6 as the last release in 3.1.x series. Based on our release policy to work on latest two stable branches, 3.1 got out of by release of 3.3.0. The branch 3.1 may still get patches, it is not a closed branch, but it is a matter of developers for willing to do that -- it is just not going to be packaged again in a numbered release, so people will have to use git branch. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw

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sr-users July 2012