July 2012 - sr-users - kamailio.org

Kamailio and IMS
by Apoorv Khare 20 Jul '12

20 Jul '12

Hi Everyone, I am trying to integrate IMS(*carstenbock/ims*) patch in Kamailio. But so far I could not find any proper documentation to load configuration files - pcscf, icscf, scscf, mgcf etc? Please let me know if there is anything available! Only relevant link I found - http://www.kamailio.org/w/2011/01/ims-extensions-available-for-testing/ AND this link does not seems to be update - http://www.ng-voice.com/our-solution/installation/ BR, Apoorv

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uac module From header rewriting problems
by Timo Teras 20 Jul '12

20 Jul '12

Hi, I fixed some uac From rewriting issues in commit e1d1c774c9ac0b4d, however even with the latest versions I'm still seeing corruption in From headers when they are rewritten for the whole dialogue (from_restore_mode=auto, the default). I spent some time analyzing the problem and the problem seems to be connected with the fact that only the bitwise difference of the original and modified URI are stored (XOR). The only purpose for this appears to be the idea to save only the difference in the rr params instead of two full URIs. However, it appears that in many cases (especially call transfer) the From header will be altered during dialogue. This is explicitly allowed in RFC 3261 § 20.20: The From header field indicates the initiator of the request. This may be different from the initiator of the dialog. I've seen this in practise with few commercial SIP stacks. Sometimes the changes are minor, or even trivial - but with the difference encoding having even one character changed (e.g. case), being deleted or added will result in the whole header being corrupted. I'm now wondering if we should just store both URIs in the rr params. While taking little more space, it should fix us sending garbage to the wire (which usually results in dropped call as the remote will reject the garbage messages). Any better ideas? Cheers, Timo

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Kamailio as SBC and far end nat
by Martin Schiøtz 19 Jul '12

19 Jul '12

Hi I'm trying to configure kamailio version 3.3.0 to be a Session Border Controller which only purpose is to handle far-end-nat. On the far end site I have a SIP client with IP '192.168.1.2' which is NAT'ed to IP 'zz.zz.85.230'. The SIP ALG is disabled on the NAT firewall. The SIP client is communicating with the kamailio server and the Kamailio server is forwarding every thing to an Avaya SIP server 'xx.xx.173.15' and vice versa. The kamilio is konfigured to do this stateless. The REGISTER and SUBSCRIBE part works fine. The kamailio is using the 'Via:' field and I also needed to do subst on the 'Contact:' field in order for our SIP server to communicate back to the kamailio. But when I get to the NOTIFY packet from the SIP server to the kamailio there is no 'Via:' field or any other information about the SIP clients IP/port and the Kamailio just sends back 'SIP/2.0 404 Not here.' to the SIP server. I guess that this can not be solved with a statless kamailio? I guess that I need some kind of state table in the kamailio - is that possible without doing REGISTER directly with the kamailio? Any other good ideas are welcome? Cheers, Martin Tell me if need more info the issue? ----- # T xx.xx.173.15:38059 -> yy.yy.159.211:5060 [AP] NOTIFY sip:2001012@yy.yy.159.211:5060;transport=TCP SIP/2.0. Call-ID: NjQ5ZDNlYzRjOWFjMzMzN2JlYzAwMTg4ODYwZTJlYWY.. From: "2001012" <sip:2001012@bsosip.domain.dk>;tag=-667195155*1*016asm-callprocessing.sar229435981~1342570131099~1170998434~1. To: "2001012" <sip:2001012@bsosip.domain.dk>;tag=7ca0af7d. CSeq: 3 NOTIFY. Contact: <sip:xx.xx.173.14:15060>. Via: SIP/2.0/TCP xx.xx.173.15;branch=z9hG4bK4DEAAD0E478B838008189457-AP;ft=xx.xx.173.15~13c4. Via: SIP/2.0/UDP xx.xx.173.14:15070;branch=z9hG4bK4DEAAD0E478B838008189457. Via: SIP/2.0/TCP xx.xx.173.14:15070;branch=z9hG4bK4DEAAD0E478B838018189455. Event: message-summary. Content-Type: application/simple-message-summary. Content-Length: 69. Subscription-State: active;expires=299. Max-Forwards: 68. User-Agent: AVAYA-SM-6.1.4.0.614005. . Messages-Waiting: no. Message-Account: sip:2001012@bsosip.domain.dk. ## T yy.yy.159.211:5060 -> xx.xx.173.15:38059 [AP] SIP/2.0 404 Not here. Call-ID: NjQ5ZDNlYzRjOWFjMzMzN2JlYzAwMTg4ODYwZTJlYWY.. From: "2001012" <sip:2001012@bsosip.domain.dk>;tag=-667195155*1*016asm-callprocessing.sar229435981~1342570131099~1170998434~1. To: "2001012" <sip:2001012@bsosip.domain.dk>;tag=7ca0af7d. CSeq: 3 NOTIFY. Via: SIP/2.0/TCP xx.xx.173.15;branch=z9hG4bK4DEAAD0E478B838008189457-AP;ft=xx.xx.173.15~13c4;rport=38059. Via: SIP/2.0/UDP xx.xx.173.14:15070;branch=z9hG4bK4DEAAD0E478B838008189457. Via: SIP/2.0/TCP xx.xx.173.14:15070;branch=z9hG4bK4DEAAD0E478B838018189455. Server: kamailio (3.3.0 (i386/linux)). Content-Length: 0. . # T xx.xx.173.15:38059 -> yy.yy.159.211:5060 [A] ...... # T yy.yy.159.211:5060 -> xx.xx.173.15:38059 [AP] SIP/2.0 404 Not here. Call-ID: NjQ5ZDNlYzRjOWFjMzMzN2JlYzAwMTg4ODYwZTJlYWY.. From: "2001012" <sip:2001012@bsosip.domain.dk>;tag=-667195155*1*016asm-callprocessing.sar229435981~1342570131099~1170998434~1. To: "2001012" <sip:2001012@bsosip.domain.dk>;tag=7ca0af7d. CSeq: 3 NOTIFY. Via: SIP/2.0/TCP xx.xx.173.15;rport=38059;branch=z9hG4bK4DEAAD0E478B838008189457-AP;ft=xx.xx.173.15~13c4. Via: SIP/2.0/UDP xx.xx.173.14:15070;branch=z9hG4bK4DEAAD0E478B838008189457. Via: SIP/2.0/TCP xx.xx.173.14:15070;branch=z9hG4bK4DEAAD0E478B838018189455. Server: kamailio (3.3.0 (i386/linux)). Content-Length: 0. -----

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Compile Kamailio from Source with STUN
by Vijay Thakur 19 Jul '12

19 Jul '12

Dear All Experts, I am following the steps given at http://kb.asipto.com/kamailio:presence:k31-made-simple. I want to compile kamailio with stun support. So guide me that what will be the command to do it. As per the given command : *make PREFIX=/usr/local/kamailio-3.1-xcap FLAVOUR=kamailio \ include_modules="db_mysql presence presence_xml xhttp xcap_server" cfg what text should i use to make stun module. * -- Best Regards, Vijay Thakur (Assistant Manager - Networks) Mobile : +91 8744018065 Mail : vijay.thakur(a)loopmethods.com Loop IT Methods Private Limited 1st Floor, B-10, Sector-7, Noida, (U.P) India Ph: +91 120 305 3481,82 (INDIA), +1 347 468 8631 (USA), +61 390 011 178 (AUS) Fax: +91 971 728 330 Web: www.loopmethods.com LOOP Disclaimer ------------------------------------------------------------------------------------------------- This message (including any attachments) contains confidential information intended for a specific individual and purpose, and is protected by law. If you are not the intended recipient, you should delete this message and are hereby notified that any disclosure, copying, or distribution of this message, or the taking of any action based on it, is strictly prohibited. -----------------------------------------------------------------------------------------------------------------

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Kamailio v3.2.4 Released
by Daniel-Constantin Mierla 19 Jul '12

19 Jul '12

Hello, Kamailio v3.2.4 is out -- this is a minor release packaged from previous stable branch, GIT 3.2. Anyone running 3.2.x should update to it in order to benefit of latest fixes, there is no change required to be done in database structure or configuration file, just re-install over the old deployment. As usual, a backup is recommended to be in the safe side. More details about this release at: * http://www.kamailio.org/w/2012/07/kamailio-v3-2-4-released/ New installations should start with latest stable series 3.3.x, at this time latest Kamailio version being v3.3.0. 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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Re: [SR-Users] Presence Subscription / Subscription-State: pending
by Daniel-Constantin Mierla 19 Jul '12

19 Jul '12

Hello, please keep the mailing list cc-ed, private emails from mailing list discussions will be ignored. On 7/19/12 8:28 AM, Durmuş Ali Öztürk wrote: > Hi Daniel, > > Thank you for your advise. I have found the problem, I thought it is > enough to subscribe with my own ID but fact is I have to subscribe to > the buddies of my contact list.This was not clear for me at the beginning. ok, great you found it. > > It is better to subscribe only to the presence.wmi? Or should I > susbcribe to presence and presence.wmi? Do you mean presence mwi? mwi is for message waiting indication, like voicemail, so you get notifications when someone leaves you a message on the media server. Cheers, Daniel > > Best Regards, > Ali > > 2012/7/18 Daniel-Constantin Mierla <miconda(a)gmail.com > <mailto:miconda@gmail.com>> > > Hello, > > when you updated the presence rules via xcap, did you executed the > function to update the notifiers as well? I mean the two function at: > > http://kamailio.org/docs/modules/stable/modules_k/presence.html#id2496049 > > Cheers, > Daniel > > > > On 7/18/12 9:29 AM, Durmuş Ali Öztürk wrote: >> Hi All, >> >> I am trying to subscribe to the presence event but unfortunaly I >> am getting following Notify with the "Subscription-State: >> pending" status. >> >> NOTIFY sip:5000@192.168.127.138:55583 >> <http://sip:5000@192.168.127.138:55583> SIP/2.0 >> Via: SIP/2.0/UDP 192.168.127.230;branch=z9hG4bK6a8e.19cd7bb7.0 >> To: sip:5000@192.168.127.230 >> <mailto:sip%3A5000@192.168.127.230>;tag=2786be25cb234561b8dea2e067fd769c-5000 >> From: sip:5000@192.168.127.230 >> <mailto:sip%3A5000@192.168.127.230>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-6a8e >> CSeq: 2 NOTIFY >> Call-ID: e4ec6031-99e1-de11-8845-5000@vvt-laptop >> Content-Length: 0 >> User-Agent: kamailio (3.2.3 (i386/linux)) >> Max-Forwards: 70 >> Event: presence >> Contact: <sip:192.168.127.230:5060;transport=udp> >> Subscription-State: pending;expires=3670 >> >> >> For the implemantation I followed the guide from >> http://kb.asipto.com/kamailio:presence:k31-made-simple >> >> The xcap table has already the needed resource-lists >> and pres-rules record. >> >> Is there another guide exists for the presence module which can >> be helpfully for me? >> >> Thank you in advance >> ali >> >> >> >> >> _______________________________________________ >> 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://twitter.com/#!/miconda <http://twitter.com/#%21/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 > > -- 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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planning release of v3.2.4
by Daniel-Constantin Mierla 19 Jul '12

19 Jul '12

Hello, later this week (like Thursday) I am considering packaging v3.2.4 from the latest git branch 3.2. If anyone has patches to push it there, please do it asap in case you want them in this release. 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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Presence Subscription / Subscription-State: pending
by Durmuş Ali Öztürk 18 Jul '12

18 Jul '12

Hi All, I am trying to subscribe to the presence event but unfortunaly I am getting following Notify with the "Subscription-State: pending" status. NOTIFY sip:5000@192.168.127.138:55583 SIP/2.0 Via: SIP/2.0/UDP 192.168.127.230;branch=z9hG4bK6a8e.19cd7bb7.0 To: sip:5000@192.168.127.230;tag=2786be25cb234561b8dea2e067fd769c-5000 From: sip:5000@192.168.127.230;tag=a6a1c5f60faecf035a1ae5b6e96e979a-6a8e CSeq: 2 NOTIFY Call-ID: e4ec6031-99e1-de11-8845-5000@vvt-laptop Content-Length: 0 User-Agent: kamailio (3.2.3 (i386/linux)) Max-Forwards: 70 Event: presence Contact: <sip:192.168.127.230:5060;transport=udp> Subscription-State: pending;expires=3670 For the implemantation I followed the guide from http://kb.asipto.com/kamailio:presence:k31-made-simple The xcap table has already the needed resource-lists and pres-rules record. Is there another guide exists for the presence module which can be helpfully for me? Thank you in advance ali

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Reply 503 was replaced to 500
by Uri Shacked 18 Jul '12

18 Jul '12

Hi, Kamailio server is behind our company's softswitch and acts as a sip application server. I notice that there are calls that the softswitch replied with 503 "service unavailable" and kamailio sent to the originator leg 500 "service unavaileable". When kamailio recieved 504 or 502 it sends them back as is. shouldn't it be the same with 503? It also does not have a "to tag" in the CDR. And the "to tag" in the 503 that was recieved is not equal to the 500 reply "to tag" kamailio sent back. any ideas? BR, Uri

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Topoh address
by Alex Balashov 17 Jul '12

17 Jul '12

A whole ago, it was mentioned on the list that there was some bug or issue in Topoh that precluded the use of the actual host IP of the proxy as the Topoh spoof address. Is that still true in 3.3? Or has that been fixed some time ago? Thanks! -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/

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How to use regular expression for $var ?
by Gary Chen 17 Jul '12

17 Jul '12

Kamailio 3.3.0 I have a variable $var(s:dst). It can store either a number or IP. How do I check to determine whether it is a number of IP? I tried the following and it did not work: If ($var(s:dst) =~ "^\d+\.\d+\.\d+\.\d+$"){ It is a IP. }else{ It is a Number } Thanks Gary

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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 14 Jul '12

14 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

2 2

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

2 1

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

2 1

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

1 0

the 200 ok reply not include supported header
by mike 13 Jul '12

13 Jul '12

Hi all: i have the server loaded the path model , but the 200 ok didn't include the "Supported" header filed . REGISTER sip:example.com SIP/2.0 Via: SIP/2.0/TCP 192.0.2.2;branch=z9hG4bKnashds7 Max-Forwards: 70 From: Bob <sip:bob@example.com>;tag=7F94778B653B To: Bob <sip:bob@example.com> Call-ID: 16CB75F21C70 CSeq: 1 REGISTER Supported: path, outbound Route: <sip:ep1.example.com;lr> Contact: <sip:bob@192.0.2.2;transport=tcp>;reg-id=1 ;+sip.instance="<urn:uuid:00000000-0000-1000-8000-AABBCCDDEEFF>" Content-Length: 0 SIP/2.0 200 OK Via: SIP/2.0/TCP 192.0.2.15;branch=z9hG4bKnuiqisi Via: SIP/2.0/TCP 192.0.2.2;branch=z9hG4bKnashds7 From: Bob <sip:bob@example.com>;tag=7F94778B653B To: Bob <sip:bob@example.com>;tag=6AF99445E44A Call-ID: 16CB75F21C70 CSeq: 1 REGISTER Supported: path, outbound (---------this line is not include-------------) Require: outbound Contact: <sip:bob@192.0.2.2;transport=tcp>;reg-id=1;expires=3600 ;+sip.instance="<urn:uuid:00000000-0000-1000-8000-AABBCCDDEEFF>" Path: <sip:VskztcQ/ how could i fix it ? thanks mike

2 1

Re: [SR-Users] sip over tls is not working
by Peter Dunkley 12 Jul '12

12 Jul '12

Hi, I have just built Kamailio from Git master, configured TCP and TLS, and an HTTP event route. I was able to connect to Kamailio over TLS using Google Chrome (and saw the "SSL Error: The site's security certificate is not trusted!" page). After I clicked on "Proceed anyway" I saw the log message I put in the HTTP event route come out. I also logged in with Jitsi using TLS and that worked fine too. I tried this with both the WebSocket module loaded and without it, and it worked in both cases. I also tried the above with TCP and that worked fine as well. As far as I can tell TLS is working in Kamailio Git master. Regards, Peter > On Wed, Jul 11, 2012 at 9:37 PM, Peter Dunkley > <peter.dunkley(a)crocodile-rcs.com> wrote: >> >> Hi, >> >> WebSockets over TLS works which requires establishing a TLS connection >> and >> exchanging an HTTP request and response. It doesn't sound like this >> connection is even getting passed the TLS handshake part? >> >> Peter >> > > Hi, > That was my first guess. I will run some tests with plain tcp socket > and post update. > > cheers. > >> >> On Wed, 2012-07-11 at 17:14 +0200, Klaus Darilion wrote: >> >> Maybe there were some changes fore websocket support which cause >> problems. Do plain TCP connections work? >> >> klaus >> >> On 11.07.2012 16:20, Aft nix wrote: >> > On Wed, Jul 11, 2012 at 6:56 PM, Klaus Darilion >> > <klaus.mailinglists(a)pernau.at> wrote: >> >> I just tested TLS with Kamailio 3.3.0 and Eyebeam and it works. Make >> >> sure to >> >> specify "ca_list" if intermediate certificates are used. >> >> >> > >> > I was working with master branch, not 3.3 branch. >> > >> >> >> >> regards >> >> Klaus >> >> >> >> On 09.07.2012 13:27, Aft nix wrote: >> >>> >> >>> Hi, >> >>> >> >>> I have enabled tls parameters as follows: >> >>> >> >>> in kamailio.cfg >> >>> >> >>> listen = tls:<IP>:<PORT> >> >>> >> >>> in tls.cfg >> >>> >> >>> [server:<IP>:<PORT>] >> >>> method = TLSv1 >> >>> verify_certificate = no >> >>> require_certificate = no >> >>> private_key = /usr/local/etc/kamailio/kamailio-selfsigned.key >> >>> certificate = /usr/local/etc/kamailio/ >> kamailio-selfsigned.pem >> >>> >> >>> Now if i try to connect to this interface using openssl s_client, it >> >>> does connects, >> >>> but now server certificate is sent from kamailio. >> >>> >> >>> kamailio log shows this : >> >>> >> >>> <core> [ip_addr.c:247]: tcpconn_new: new tcp connection: <CLIENT >> >>> IP> >> >>> <core> [tcp_main.c:10

2 1

Kamailio Unexpectedly Terminating
by Akan 12 Jul '12

12 Jul '12

I have 2 servers running Solaris and Kamailio 3.2.3 where on one Kamailio is terminating when it tries to save the location for a register request and the other is producing a core dump when processing an Option request. I have one server handling Register request while the other sip server forwards the register requests and handles the other requests. I have included the backtraces from the core dumps and the output from the log for the registrar server as well as the command that is causing kamailio to terminate: if (!save("location")) sl_reply_error(); 4(3364) ERROR: *** cfgtrace: c=[/opt/kamailio-3.2/etc/kamailio/kamailio.cfg] l=714 a=17 n=if 4(3364) ERROR: *** cfgtrace: c=[/opt/kamailio-3.2/etc/kamailio/kamailio.cfg] l=711 a=26 n=save 14(3374) : <core> [pass_fd.c:293]: ERROR: receive_fd: EOF on 15 14(3374) DEBUG: <core> [tcp_main.c:3555]: DBG: handle_ser_child: dead child 4, pid 3364 (shutting down?) 14(3374) DEBUG: <core> [io_wait.h:617]: DBG: io_watch_del (1003743d8, 15, 0, 0x0) fd_no=18 called 0(3360) ALERT: <core> [main.c:751]: child process 3364 exited by a signal 10 0(3360) ALERT: <core> [main.c:754]: core was not generated 0(3360) INFO: <core> [main.c:766]: INFO: terminating due to SIGCHLD 6(3366) INFO: <core> [main.c:817]: INFO: signal 15 received 1(3361) INFO: <core> [main.c:817]: INFO: signal 15 received 2(3362) INFO: <core> [main.c:817]: INFO: signal 15 received 3(3363) INFO: <core> [main.c:817]: INFO: signal 15 received 5(3365) INFO: <core> [main.c:817]: INFO: signal 15 received 7(3367) INFO: <core> [main.c:817]: INFO: signal 15 received 8(3368) INFO: <core> [main.c:817]: INFO: signal 15 received 9(3369) INFO: <core> [main.c:817]: INFO: signal 15 received 10(3370) INFO: <core> [main.c:817]: INFO: signal 15 received 11(3371) INFO: <core> [main.c:817]: INFO: signal 15 received 12(3372) INFO: <core> [main.c:817]: INFO: signal 15 received 13(3373) INFO: <core> [main.c:817]: INFO: signal 15 received 14(3374) INFO: <core> [main.c:817]: INFO: signal 15 received 0(3360) DEBUG: presence_xml [presence_xml.c:347]: start 0(3360) ERROR: ctl [ctl.c:379]: ERROR: ctl: could not delete unix socket /tmp/kamailio_ctl: Permission denied (13) 0(3360) DEBUG: <core> [db_pool.c:102]: removing connection from the pool 0(3360) DEBUG: db_postgres [km_pg_con.c:122]: PQfinish(100842470) 0(3360) DEBUG: db_postgres [km_pg_con.c:126]: pkg_free(1004c1f30) 0(3360) DEBUG: tm [t_funcs.c:122]: DEBUG: tm_shutdown : start 0(3360) DEBUG: tm [t_funcs.c:125]: DEBUG: tm_shutdown : emptying hash table 0(3360) DEBUG: tm [t_funcs.c:127]: DEBUG: tm_shutdown : removing semaphores 0(3360) DEBUG: tm [t_funcs.c:129]: DEBUG: tm_shutdown : destroying tmcb lists 0(3360) DEBUG: tm [t_funcs.c:132]: DEBUG: tm_shutdown : done Thanks Nathaniel

4 38

sip over tls is not working
by Aft nix 11 Jul '12

11 Jul '12

Hi, I have enabled tls parameters as follows: in kamailio.cfg listen = tls:<IP>:<PORT> in tls.cfg [server:<IP>:<PORT>] method = TLSv1 verify_certificate = no require_certificate = no private_key = /usr/local/etc/kamailio/kamailio-selfsigned.key certificate = /usr/local/etc/kamailio/kamailio-selfsigned.pem Now if i try to connect to this interface using openssl s_client, it does connects, but now server certificate is sent from kamailio. kamailio log shows this : <core> [ip_addr.c:247]: tcpconn_new: new tcp connection: <CLIENT IP> <core> [tcp_main.c:1089]: tcpconn_new: on port 40727, type 3 <core> [tcp_main.c:1400]: tcpconn_add: hashes: 2614:2652:2494, 2 <core> [io_wait.h:390]: DBG: io_watch_add(0x82535e0, 23, 2, 0xb5701580), fd_no=11 <core> [io_wait.h:617]: DBG: io_watch_del (0x82535e0, 23, -1, 0x0) fd_no=12 called <core> [tcp_main.c:4296]: tcp: DBG: sending to child, events 1 <core> [tcp_main.c:3963]: WARNING: send2child: no free tcp receiver, connection passed to the least busy one (3289651) <core> [tcp_main.c:3967]: selected tcp worker 0 0(8) for activity on [tls:<IP>:<PORT>], 0xb5701580 <core> [tcp_main.c:3576]: BUG: handle_ser_child: fd -1 for 0 (pid 2491) I'm using kamailio from git. its updated to the latest. Thanks in advance. -- -aft

4 11

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sr-users July 2012