July 2012 - sr-users - kamailio.org

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

3 7

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. -----

3 2

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. -----------------------------------------------------------------------------------------------------------------

3 2

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

1 0

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

3 3

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

1 1

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

2 1

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

2 6

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/

2 1

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

4 3

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

2 1

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

2 2

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

2 1

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 >

1 0

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

2 1

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

1 0

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

Re: [SR-Users] GRUU support in Kamailio
by amar＠smarttel.com.np 11 Jul '12

11 Jul '12

A -original message- Subject: Re: [SR-Users] GRUU support in Kamailio From: Iñaki Baz Castillo <ibc(a)aliax.net> Date: 11/07/2012 10:08 pm 2012/7/11 Olle E. Johansson <oej(a)edvina.net>: > Thanks for your patience, Daniel! And even more thanks for the code! I will test it in depth very soon. -- Iñaki Baz Castillo <ibc(a)aliax.net> _______________________________________________ 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

GRUU support in Kamailio
by Olle E. Johansson 11 Jul '12

11 Jul '12

Hey, Been trying to understand the GRUU support in Kamailio. I see that the registrar saves sip.instance - but I don't see any functions to generate gruu URIs or store them. A public GRUU is valid forever for a sip.instance and should propably be saved into some sort of database table. A temporary GRUU is only valid for as long as a device registers with the same Call-ID. Please give me pointers to what I have missed! Thanks, /O --- * Olle E Johansson - oej(a)edvina.net * Cell phone +46 70 593 68 51, Office +46 8 96 40 20, Sweden

3 11

DB create statements
by Klaus Darilion 11 Jul '12

11 Jul '12

Hi! For some reason I need to the enter the SQL statements manually on the DB server (I do not want to copy the scripts to the DB server and there is no superuser access over the network) Is there a simple method to get all the SQL commands to create the kamailio database? I see there are all the files in "/usr/share/kamailio/postgres/" and lots of other SQL statements in /usr/lib/kamailio/kamctl/kamdbctl.pgsql. Does anyone know of a simple method to get all the raw SQL queries? Thanks Klaus

2 2

start kamailio
by Michal Yachimovitz 11 Jul '12

11 Jul '12

Hello I installed kamailio version 3.2.0 and followed the instructions in http://www.kamailio.org/wiki/install/3.2.x/git web site Whene I permormed the command /etc/init.d/kamailio start I got the following errors. The modules are in the correct directory so actually I do not understand the error. I will be happy if someone can help me solve this issue: [root@kamailio-dev-lab ~]# */etc/init.d/kamailio start * Not starting Kamailio: invalid configuration file! 0(8569) ERROR: <core> [sr_module.c:565]: ERROR: load_module: could not open module </usr/local/lib/kamailio/modules/db_mysql.so>: /usr/local/lib/kamailio/modules/db_mysql.so: undefined symbol: SSL_CTX_free 0(8569) : <core> [cfg.y:3529]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 214, column 12-24: failed to load module 0(8569) ERROR: <core> [sr_module.c:553]: ERROR: load_module: could not find module <mi_fifo> in </usr/local/lib/kamailio/modules> 0(8569) : <core> [cfg.y:3529]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 217, column 12-23: failed to load module 0(8569) ERROR: <core> [sr_module.c:553]: ERROR: load_module: could not find module <kex> in </usr/local/lib/kamailio/modules> 0(8569) : <core> [cfg.y:3529]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 218, column 12-19: failed to load module 0(8569) ERROR: <core> [sr_module.c:553]: ERROR: load_module: could not find module <tmx> in </usr/local/lib/kamailio/modules> 0(8569) : <core> [cfg.y:3529]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 220, column 12-19: failed to load module 0(8569) ERROR: <core> [sr_module.c:553]: ERROR: load_module: could not find module <rr> in </usr/local/lib/kamailio/modules> 0(8569) : <core> [cfg.y:3529]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 222, column 12-18: failed to load module 0(8569) ERROR: <core> [sr_module.c:553]: ERROR: load_module: could not find module <pv> in </usr/local/lib/kamailio/modules> 0(8569) : <core> [cfg.y:3529]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 223, column 12-18: failed to load module 0(8569) ERROR: <core> [sr_module.c:553]: ERROR: load_module: could not find module <maxfwd> in </usr/local/lib/kamailio/modules> 0(8569) : <core> [cfg.y:3529]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 224, column 12-22: failed to load module 0(8569) ERROR: <core> [sr_module.c:553]: ERROR: load_module: could not find module <usrloc> in </usr/local/lib/kamailio/modules> 0(8569) : <core> [cfg.y:3529]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 225, column 12-22: failed to load module 0(8569) ERROR: <core> [sr_module.c:553]: ERROR: load_module: could not find module <registrar> in </usr/local/lib/kamailio/modules> 0(8569) : <core> [cfg.y:3529]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 226, column 12-25: failed to load module 0(8569) ERROR: <core> [sr_module.c:553]: ERROR: load_module: could not find module <textops> in </usr/local/lib/kamailio/modules> 0(8569) : <core> [cfg.y:3529]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 227, column 12-23: failed to load module 0(8569) ERROR: <core> [sr_module.c:553]: ERROR: load_module: could not find module <siputils> in </usr/local/lib/kamailio/modules> 0(8569) : <core> [cfg.y:3529]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 228, column 12-24: failed to load module 0(8569) ERROR: <core> [sr_module.c:553]: ERROR: load_module: could not find module <xlog> in </usr/local/lib/kamailio/modules> 0(8569) : <core> [cfg.y:3529]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 229, column 12-20: failed to load module 0(8569) ERROR: <core> [sr_module.c:553]: ERROR: load_module: could not find module <acc> in </usr/local/lib/kamailio/modules> 0(8569) : <core> [cfg.y:3529]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 234, column 12-19: failed to load module 0(8569) ERROR: <core> [sr_module.c:553]: ERROR: load_module: could not find module <auth_db> in </usr/local/lib/kamailio/modules> 0(8569) : <core> [cfg.y:3529]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 238, column 12-23: failed to load module 0(8569) ERROR: <core> [modparam.c:162]: set_mod_param_regex: No module matching <mi_fifo> found 0(8569) : <core> [cfg.y:3532]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 287, column 54: Can't set module parameter 0(8569) ERROR: <core> [modparam.c:162]: set_mod_param_regex: No module matching <rr> found 0(8569) : <core> [cfg.y:3532]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 301, column 35: Can't set module parameter 0(8569) ERROR: <core> [modparam.c:162]: set_mod_param_regex: No module matching <rr> found 0(8569) : <core> [cfg.y:3532]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 303, column 35: Can't set module parameter 0(8569) ERROR: <core> [modparam.c:162]: set_mod_param_regex: No module matching <registrar> found 0(8569) : <core> [cfg.y:3532]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 307, column 44: Can't set module parameter 0(8569) ERROR: <core> [modparam.c:162]: set_mod_param_regex: No module matching <registrar> found 0(8569) : <core> [cfg.y:3532]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 313, column 42: Can't set module parameter 0(8569) ERROR: <core> [modparam.c:162]: set_mod_param_regex: No module matching <acc> found 0(8569) : <core> [cfg.y:3532]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 318, column 33: Can't set module parameter 0(8569) ERROR: <core> [modparam.c:162]: set_mod_param_regex: No module matching <acc> found 0(8569) : <core> [cfg.y:3532]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 319, column 32: Can't set module parameter 0(8569) ERROR: <core> [modparam.c:162]: set_mod_param_regex: No module matching <acc> found 0(8569) : <core> [cfg.y:3532]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 320, column 36: Can't set module parameter 0(8569) ERROR: <core> [modparam.c:162]: set_mod_param_regex: No module matching <acc> found 0(8569) : <core> [cfg.y:3532]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 324, column 38: Can't set module parameter 0(8569) ERROR: <core> [modparam.c:162]: set_mod_param_regex: No module matching <acc> found 0(8569) : <core> [cfg.y:3532]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 326, column 30: Can't set module parameter 0(8569) ERROR: <core> [modparam.c:162]: set_mod_param_regex: No module matching <acc> found 0(8569) : <core> [cfg.y:3532]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 327, column 37: Can't set module parameter 0(8569) ERROR: <core> [modparam.c:162]: set_mod_param_regex: No module matching <acc> found 0(8569) : <core> [cfg.y:3532]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 330, column 45: Can't set module parameter 0(8569) ERROR: <core> [modparam.c:162]: set_mod_param_regex: No module matching <acc> found 0(8569) : <core> [cfg.y:3532]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 331, column 45: Can't set module parameter 0(8569) : <core> [cfg.y:3529]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 346, column 30-34: syntax error 0(8569) : <core> [cfg.y:3529]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 346, column 30-34: Invalid arguments 0(8569) : <core> [cfg.y:3532]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 346, column 35: ERROR: bad config file (30 errors) [root@kamailio-dev-lab ~]# Thanks in advance Michal Yachimovitz

2 3

Re: [SR-Users] is kamilio support RFC5626 now ?
by Daniel-Constantin Mierla 11 Jul '12

11 Jul '12

FYI: private emails sent from mailing lists discussions are simply ignored, keep the mailing list in the discussion. Thanks. On 7/11/12 9:13 AM, mike wrote: > yes , i know that the path could be added by the path module , but > ,the problem is the flow token,such as the path > > Path: <sip:VskztcQ/S8p4WPbOnHbuyh5iJvJIW3ib@ep1.example.com > <mailto:S8p4WPbOnHbuyh5iJvJIW3ib@ep1.example.com>;lr;ob> > > > the VskztcQ/S8p4WPbOnHbuyh5iJvJIW3ib is the token , so , where the > kamilio 3.3 handle it ? > > > On Wed, Jul 11, 2012 at 3:00 PM, Daniel-Constantin Mierla > <miconda(a)gmail.com <mailto:miconda@gmail.com>> wrote: > > Hello, > > > On 7/11/12 4:23 AM, mike wrote: > > dear kamilio developers : > > i read the kamilio 3.3.0' s new features , it say it support > outbound , and i have place to handle reg-id and sip.instance > . so , i wonder if the kamilio3.3.0 support RFC5626 now，so > that the > kamilio can be used as a edge proxy . i ask why because i > can't find where kamilio add flow-token and path for the > register request. > > sip.instance and reg-id are handled by the registrar/usrloc > modules. Path has to be added by intermediary proxies and it is > also handled by registrar module. If the intermediary proxies are > kamailio, then adding path is possible via path module. > > Cheers, > Daniel > > -- > 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

1 0

usrloc concurrency
by Alex Balashov 11 Jul '12

11 Jul '12

(Reposting from sr-dev because, in retrospect, it makes more sense here.) Does anyone know of any issues running usrloc with DB backing ('location') in a multi-master DB-replicated environment? I am referring to plain usrloc, not p_usrloc. I have such an environment set up with db_mode = 1 (immediate write-through) and have run into a problem where Kamailio occasionally stops writing to the location table for no apparent reason. I am wondering if this is because another Kamailio instance bound to the secondary replica is performing a DB sync as well, and at some point they collide in some sort of race condition. Would db_mode 2 fix? Would anything fix? Is this even the problem? -- 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/, http://www.alexbalashov.com

2 3

How to proxy/authenticate on third party sip server?
by intel＠intrans.baku.az 10 Jul '12

10 Jul '12

Can anybody help me with configuration kamailio to authenticate call on 3rd party server? I want to authenticate(and proxy) call's on other server. for example, user A have account on my server(let it be A(a)myserver.com) and account on other server (let it be AA(a)hisserver.com) myserver.com keeps A's credentials for AA(a)hisserver.com when A is registered on myserver.com, he make call to B(a)othersip.com (using myserver.com as a proxy) Kamailio on myserver authenticate itself on hisserver as AA(a)hisserver.com, makes call to destination, and connect it to A. How can I setup kamailio for such behavior? Which module should I use? UAC seems can authenticate on another server, but I don't sure that it do what I want. (I've tried it, but without much success)

2 8

Does there exist a tutorial on the Kamailio SIP/XMPP Gateway configuration?
by Auro Tripathy 10 Jul '12

10 Jul '12

I'm finding that I may have to piece together (the configuration info) from version emails/presentations. Reluctantly, I'll take that approach...unless someone has a lucid HOWTO on the topic? Again, I'm new to the domain and I appreciate the subject-matter experts that have helped out so far. -Auro

2 1

[Fwd: [sr-dev] WebSocket Server support in Kamailio master]
by Peter Dunkley 09 Jul '12

09 Jul '12

-------- Forwarded Message -------- From: Peter Dunkley <peter.dunkley(a)crocodile-rcs.com> Reply-to: Development mailing list of the sip-router project <sr-dev(a)lists.sip-router.org> To: sr-dev(a)lists.sip-router.org Subject: [sr-dev] WebSocket Server support in Kamailio master Date: Sat, 7 Jul 2012 18:29:23 +0100 Hello, I have merged the pd/websocket branch into Kamailio master. This means that you can now connect SIP over WebSocket (draft-ibc-sipcore-sip-websocket-02) clients to Kamailio using the "ws://" and "wss://" protocols. Some customisation of the websocket module is possible through modparams, but for most users the defaults should be OK. The WebSocket module uses the xhttp and sl modules for the initial handshake, and (unless you have both a Kamailio installation and WebSocket SIP client supporting GRUU, Outbound[1], and Path[2]) nathelper for request routing and the core force_rport() function for response routing (a new nat_uac_test() has been added to detect whether a message has arrived on a WebSocket). There is an example kamailio.cfg in the websocket module directory. [1] Kamailio does not currently support Outbound [2] I have not updated the Path module for WebSockets I believe that, once Kamailio supports Outbound and WebSocket support is added to the Path module (and you have a SIP over WebSocket client that supports this), it will be possible to use the websocket module without the nathelper module and force_rport() and without needing to change the websocket module or Kamailio core code. If you want to use secure WebSockets (wss) as well as ordinary WebSockets just configure TLS and listen on an appropriate port. I have added WebSocket support to some modules, but there are definitely going to be others (modules/lcr, modules/sipcapture, modules_k/nat_traversal, modules_k/path, modules_k/seas, and modules_k/snmpstats, at least) that need updating too. WebSockets is an unusual transport, so I have put a few notes together for anyone who needs to use it in the code (including adding support to additional modules): - A WebSocket server cannot initiate a WebSocket connection. So a WebSocket connection (over TCP or TLS) is like a TCP/TLS connection coming from behind a NAT. This is why nathelper aliasing and force_rport() is used for the routing, and "set_..._no_connect()" is always used (it's set within the websocket module). - WebSocket (PROTO_WS) and secure WebSocket (PROTO_WSS) connections are just upgraded TCP and TLS connections, so there are no listening sockets for PROTO_WS and PROTO_WSS. This means that, when deciding on what transport is being used, you need to look at the proto set in the tcp_connection, receive_info, and/or dest_info structure for the message - looking at the socket_info structure (that the message has arrived on or will be sent on) will not give you the right answer. - Although WebSocket (PROTO_WS) and secure WebSocket (PROTO_WSS) are different internal protocols there is only one SIP transport type for both ";transport=ws" (WS and WSS are explicitly used in Via: headers though). This means that you can't tell whether the transport parameter in an R-URI, Route/Record-Route, or Contact-URI is for WebSockets or secure WebSockets. As long as the message makes it into the WebSocket module everything will be OK as that module sorts it all out, but it has led to slightly more complex checks being required in some of the code relating to record-routing to handle this - and it may have an effect on other modules too. Please give the new module a go and let me know about any issues you find, Peter -- Peter Dunkley Technical Director Crocodile RCS Ltd _______________________________________________ sr-dev mailing list sr-dev(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-dev -- Peter Dunkley Technical Director Crocodile RCS Ltd

1 0

Removing display-name with uac_replace_from
by Andreas Granig 09 Jul '12

09 Jul '12

Hi, in kamailio 3.3, doing uac_replace_from("", "$var(myuri)"); doesn't remove the old Display part anymore (although the documentation still says it should), the debug log is: DEBUG: uac [uac.c:499]: dsp=(nil) (len=0) , uri=0x7fff87f6b840 (len=28) DEBUG: uac [replace.c:324]: uri to replace [sip:a@example.org] DEBUG: uac [replace.c:325]: replacement uri is [sip:b@example.org] DEBUG: uac [replace.c:383]: encode is=<UEtlSyomCD8kIi8/SnlZMzNyS29QS2VLanBKeg--> len=40 So no hint that the old display is being removed. When doing uac_replace_from("foo", "$var(myuri)"); then there's an additional line like this: DEBUG: uac [replace.c:284]: removing display ["olddisplay"] which is missing when I put an empty string in the first parameter. Andreas

2 4

future features?
by 张 09 Jul '12

09 Jul '12

Hello Dear all, I'm new one to participate into this discussion. I have worked on VOIP and SIP communication solution for several years. I'm very excited about the new features in v3.3. They save me many work which I planed to dig hard. My question is whether there is a web page or discuss forum for future features for Kamailio? There is a fure feature page on sip-router.org but it's very out of date. I think it's important to catch up with potential wanted feature and the new draft which is closely watched by kamailio team. Thanks! Best Regards Victor

2 1

ACK and BYE messages uses wrong socket.
by Gary Chen 05 Jul '12

05 Jul '12

Kamailio 3.2.0 I am trying to setup kamailio to do the sip trunking. It receive the sip traffic from customer and then send it to carrier. I have two NIC interface's assigned with three IP's: Interface 1: ( Public IP's) x.x.130.34 x.x.130.36 floating IP interface 2: (private IP's) 10.10.1.31 .36 is a floating IP assigned by Linux-HA (heartbeat/pacemaker). I only want to use .36 to receive and send sip traffic. I uses force_send_socket() to send INVITE with .36 IP. But the ACK message always want to use .34 IP even the Route header has .36 in it unless I force it with force_send_socket() . How can I fix this problem? See below for the SIP messages: (x.x.128.205 is customer IP, x.x.129.200 is PSTN gateway IP) U x.x.128.205:51694 -> x.x.130.36:5060 INVITE sip:5033441174@x.x.130.36:5060 SIP/2.0..Via: SIP/2.0/UDP x.x.128.205:5060;branch=z9hG 4bK1D3CD1..From: <sip:5024427578@x.x.128.205>;tag=24513088-D59..To: <sip:5033441174@x.x.130.3 6>..Date: Tue, 03 Jul 2012 13:37:41 GMT..Call-ID: 1887588D-C44B11E1-BE38D697-98A3E39A(a)x.x.128.20 5..Supported: 100rel,timer,replaces..Min-SE: 1800..Cisco-Guid: 411443261-3293254113-3191264919-256 0877466..User-Agent: Cisco-SIPGateway/IOS-12.x..Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, CO MET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER..CSeq: 101 INVITE..Max-Forwards: 70..Remote-P arty-ID: <sip:5024427578@x.x.128.205>;party=calling;screen=no;privacy=off..Timestamp: 1341322661 ..Contact: <sip:5024427578@x.x.128.205:5060>..Expires: 180..Allow-Events: telephone-event..Conte nt-Type: application/sdp..Content-Length: 366....v=0..o=CiscoSystemsSIP-GW-UserAgent 9094 579 IN IP 4 x.x.128.205..s=SIP Call..c=IN IP4 x.x.128.205..t=0 0..m=audio 19312 RTP/AVP 125 0 18 100 10 1..c=IN IP4 x.x.128.205..a=rtpmap:125 X-CCD/8000..a=rtpmap:0 PCMU/8000..a=rtpmap:18 G729/8000..a =fmtp:18 annexb=yes..a=rtpmap:100 X-NSE/8000..a=fmtp:100 192-194..a=rtpmap:101 telephone-event/8000 ..a=fmtp:101 0-16.. # U x.x.130.36:5060 -> x.x.128.205:51694 SIP/2.0 100 trying -- your call is important to us..Via: SIP/2.0/UDP x.x.128.205:5060;branch=z9 hG4bK1D3CD1;rport=51694..From: <sip:5024427578@x.x.128.205>;tag=24513088-D59..To: <sip:502244117 4(a)x.x.130.36>..Call-ID: 1887588D-C44B11E1-BE38D697-98A3E39A@x.x.128.205..CSeq: 101 INVITE..Se rver: LVS Proxy 1.0..Content-Length: 0.... # U x.x.130.36:5060 -> x.x.129.200:5060 INVITE sip:15033441174@x.x.129.200:5060 SIP/2.0..Record-Route: <sip:x.x.130.36;lr=on>..Via: S IP/2.0/UDP x.x.130.36;branch=z9hG4bKc2ce.17a955a3.0..Via: SIP/2.0/UDP x.x.128.205:5060;rport =51694;branch=z9hG4bK1D3CD1..From: <sip:5024427578@x.x.128.205>;tag=24513088-D59..To: <sip:50224 41174(a)x.x.130.36>..Date: Tue, 03 Jul 2012 13:37:41 GMT..Call-ID: 1887588D-C44B11E1-BE38D697-98A3 E39A@x.x.128.205..Supported: 100rel,timer,replaces..Min-SE: 1800..Cisco-Guid: 411443261-3293254 113-3191264919-2560877466..User-Agent: Cisco-SIPGateway/IOS-12.x..Allow: INVITE, OPTIONS, BYE, CANC EL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER..CSeq: 101 INVITE..Max-Forw ards: 69..Remote-Party-ID: <sip:5024427578@x.x.128.205>;party=calling;screen=no;privacy=off..Tim estamp: 1341322661..Contact: <sip:5024427578@x.x.128.205:51694>..Expires: 180..Allow-Events: tel ephone-event..Content-Type: application/sdp..Content-Length: 375....v=0..o=CiscoSystemsSIP-GW-UserA gent 9094 579 IN IP4 10.200.1.51..s=SIP Call..c=IN IP4 10.200.1.51..t=0 0..m=audio 20464 RTP/AVP 12 5 0 18 100 101..c=IN IP4 10.200.1.51..a=rtpmap:125 X-CCD/8000..a=rtpmap:0 PCMU/8000..a=rtpmap:18 G7 29/8000..a=fmtp:18 annexb=yes..a=rtpmap:100 X-NSE/8000..a=fmtp:100 192-194..a=rtpmap:101 telephone- event/8000..a=fmtp:101 0-16..a=nortpproxy:yes.. U x.x.129.200:5060 -> x.x.130.36:5060 SIP/2.0 100 Trying..Via: SIP/2.0/UDP x.x.130.36;branch=z9hG4bKc2ce.17a955a3.0,SIP/2.0/UDP 216.4 9.128.205:5060;rport=51694;branch=z9hG4bK1D3CD1..From: <sip:5024427578@x.x.128.205>;tag=24513088 -D59..To: <sip:5033441174@x.x.130.36>;tag=F0695368-74F..Date: Tue, 03 Jul 2012 13:37:41 GMT..Cal l-ID: 1887588D-C44B11E1-BE38D697-98A3E39A@x.x.128.205..Timestamp: 1341322661..Server: Cisco-SIPG ateway/IOS-12.x..CSeq: 101 INVITE..Allow-Events: telephone-event..Content-Length: 0.... # U x.x.129.200:5060 -> x.x.130.36:5060 SIP/2.0 183 Session Progress..Via: SIP/2.0/UDP x.x.130.36;branch=z9hG4bKc2ce.17a955a3.0,SIP/2.0/ UDP x.x.128.205:5060;rport=51694;branch=z9hG4bK1D3CD1..From: <sip:5024427578@x.x.128.205>;ta g=24513088-D59..To: <sip:5033441174@x.x.130.36>;tag=F0695368-74F..Date: Tue, 03 Jul 2012 13:37:4 1 GMT..Call-ID: 1887588D-C44B11E1-BE38D697-98A3E39A@x.x.128.205..Timestamp: 1341322661..Server: Cisco-SIPGateway/IOS-12.x..CSeq: 101 INVITE..Require: 100rel..RSeq: 6708..Allow: INVITE, OPTIONS, B YE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER..Allow-Events: tele phone-event..Contact: <sip:15033441174@x.x.129.200:5060>..Record-Route: <sip:x.x.130.36;lr=on >..Content-Disposition: session;handling=required..Content-Type: application/sdp..Content-Length: 2 90....v=0..o=CiscoSystemsSIP-GW-UserAgent 2387 2116 IN IP4 x.x.129.200..s=SIP Call..c=IN IP4 216 .49.129.200..t=0 0..m=audio 18480 RTP/AVP 0 101 100..c=IN IP4 x.x.129.200..a=rtpmap:0 PCMU/8000. .a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-16..a=rtpmap:100 X-NSE/8000..a=fmtp:100 192-194.. # U x.x.130.36:5060 -> x.x.128.205:51694 SIP/2.0 183 Session Progress..Via: SIP/2.0/UDP x.x.128.205:5060;rport=51694;branch=z9hG4bK1D3CD 1..From: <sip:5024427578@x.x.128.205>;tag=24513088-D59..To: <sip:5033441174@x.x.130.36>;tag=F 0695368-74F..Date: Tue, 03 Jul 2012 13:37:41 GMT..Call-ID: 1887588D-C44B11E1-BE38D697-98A3E39A@216. 49.128.205..Timestamp: 1341322661..Server: Cisco-SIPGateway/IOS-12.x..CSeq: 101 INVITE..Require: 10 0rel..RSeq: 6708..Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER..Allow-Events: telephone-event..Contact: <sip:15033441174@x.x.129.200:50 60>..Record-Route: <sip:x.x.130.36;lr=on>..Content-Disposition: session;handling=required..Conte nt-Type: application/sdp..Content-Length: 305....v=0..o=CiscoSystemsSIP-GW-UserAgent 2387 2116 IN I P4 x.x.130.66..s=SIP Call..c=IN IP4 x.x.130.66..t=0 0..m=audio 19480 RTP/AVP 0 101 100..c=IN IP4 x.x.130.66..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-16..a=rtpm ap:100 X-NSE/8000..a=fmtp:100 192-194..a=nortpproxy:yes.. # U x.x.128.205:51694 -> x.x.130.36:5060 PRACK sip:15033441174@x.x.129.200:5060 SIP/2.0..Via: SIP/2.0/UDP x.x.128.205:5060;branch=z9h G4bK1D4474..From: <sip:5024427578@x.x.128.205>;tag=24513088-D59..To: <sip:5033441174@x.x.130. 36>;tag=F0695368-74F..Date: Tue, 03 Jul 2012 13:37:41 GMT..Call-ID: 1887588D-C44B11E1-BE38D697-98A3 E39A@x.x.128.205..CSeq: 102 PRACK..RAck: 6708 101 INVITE..Route: <sip:x.x.130.36;lr=on>..Max- Forwards: 70..Content-Length: 0.... # U x.x.130.34:5060 -> x.x.129.200:5060 PRACK sip:15033441174@x.x.129.200:5060 SIP/2.0..Via: SIP/2.0/UDP x.x.130.34;branch=z9hG4bK92c e.4def8436.0..Via: SIP/2.0/UDP x.x.128.205:5060;rport=51694;branch=z9hG4bK1D4474..From: <sip:50 24427578(a)x.x.128.205>;tag=24513088-D59..To: <sip:5033441174@x.x.130.36>;tag=F0695368-74F..Dat e: Tue, 03 Jul 2012 13:37:41 GMT..Call-ID: 1887588D-C44B11E1-BE38D697-98A3E39A@x.x.128.205..CSeq : 102 PRACK..RAck: 6708 101 INVITE..Max-Forwards: 69..Content-Length: 0.... # U x.x.129.200:5060 -> x.x.130.34:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP x.x.130.34;branch=z9hG4bK92ce.4def8436.0,SIP/2.0/UDP x.x.12 8.205:5060;rport=51694;branch=z9hG4bK1D4474..From: <sip:5024427578@x.x.128.205>;tag=24513088-D59 ..To: <sip:5033441174@x.x.130.36>;tag=F0695368-74F..Date: Tue, 03 Jul 2012 13:37:41 GMT..Call-ID : 1887588D-C44B11E1-BE38D697-98A3E39A@x.x.128.205..Server: Cisco-SIPGateway/IOS-12.x..CSeq: 102 PRACK..Content-Length: 0.... # U x.x.130.36:5060 -> x.x.128.205:51694 SIP/2.0 200 OK..Via: SIP/2.0/UDP x.x.128.205:5060;rport=51694;branch=z9hG4bK1D4474..From: <sip: 5024427578(a)x.x.128.205>;tag=24513088-D59..To: <sip:5033441174@x.x.130.36>;tag=F0695368-74F..D ate: Tue, 03 Jul 2012 13:37:41 GMT..Call-ID: 1887588D-C44B11E1-BE38D697-98A3E39A@x.x.128.205..Se rver: Cisco-SIPGateway/IOS-12.x..CSeq: 102 PRACK..Content-Length: 0.... U x.x.129.200:5060 -> x.x.130.36:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP x.x.130.36;branch=z9hG4bKc2ce.17a955a3.0,SIP/2.0/UDP x.x.12 8.205:5060;rport=51694;branch=z9hG4bK1D3CD1..From: <sip:5024427578@x.x.128.205>;tag=24513088-D59 ..To: <sip:5033441174@x.x.130.36>;tag=F0695368-74F..Date: Tue, 03 Jul 2012 13:37:41 GMT..Call-ID : 1887588D-C44B11E1-BE38D697-98A3E39A@x.x.128.205..Timestamp: 1341322661..Server: Cisco-SIPGatew ay/IOS-12.x..CSeq: 101 INVITE..Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSC RIBE, NOTIFY, INFO, UPDATE, REGISTER..Allow-Events: telephone-event..Contact: <sip:15033441174@216. 49.129.200:5060>..Record-Route: <sip:x.x.130.36;lr=on>..Content-Type: application/sdp..Content-L ength: 290....v=0..o=CiscoSystemsSIP-GW-UserAgent 2387 2116 IN IP4 x.x.129.200..s=SIP Call..c=IN IP4 x.x.129.200..t=0 0..m=audio 18480 RTP/AVP 0 101 100..c=IN IP4 x.x.129.200..a=rtpmap:0 PC MU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-16..a=rtpmap:100 X-NSE/8000..a=fmtp:100 19 2-194.. # U x.x.130.36:5060 -> x.x.128.205:51694 SIP/2.0 200 OK..Via: SIP/2.0/UDP x.x.128.205:5060;rport=51694;branch=z9hG4bK1D3CD1..From: <sip: 5024427578(a)x.x.128.205>;tag=24513088-D59..To: <sip:5033441174@x.x.130.36>;tag=F0695368-74F..D ate: Tue, 03 Jul 2012 13:37:41 GMT..Call-ID: 1887588D-C44B11E1-BE38D697-98A3E39A@x.x.128.205..Ti mestamp: 1341322661..Server: Cisco-SIPGateway/IOS-12.x..CSeq: 101 INVITE..Allow: INVITE, OPTIONS, B YE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER..Allow-Events: tele phone-event..Contact: <sip:15033441174@x.x.129.200:5060>..Record-Route: <sip:x.x.130.36;lr=on >..Content-Type: application/sdp..Content-Length: 305....v=0..o=CiscoSystemsSIP-GW-UserAgent 2387 2 116 IN IP4 x.x.130.66..s=SIP Call..c=IN IP4 x.x.130.66..t=0 0..m=audio 19480 RTP/AVP 0 101 10 0..c=IN IP4 x.x.130.66..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-16 ..a=rtpmap:100 X-NSE/8000..a=fmtp:100 192-194..a=nortpproxy:yes.. # U x.x.128.205:51694 -> x.x.130.36:5060 ACK sip:15033441174@x.x.129.200:5060 SIP/2.0..Via: SIP/2.0/UDP x.x.128.205:5060;branch=z9hG4 bK1D5DA6..From: <sip:5024427578@x.x.128.205>;tag=24513088-D59..To: <sip:5033441174@x.x.130.36 >;tag=F0695368-74F..Date: Tue, 03 Jul 2012 13:37:41 GMT..Call-ID: 1887588D-C44B11E1-BE38D697-98A3E3 9A@x.x.128.205..Route: <sip:x.x.130.36;lr=on>..Max-Forwards: 70..CSeq: 101 ACK..Content-Lengt h: 0.... # U x.x.130.34:5060 -> x.x.129.200:5060 ACK sip:15033441174@x.x.129.200:5060 SIP/2.0..Via: SIP/2.0/UDP x.x.130.34;branch=z9hG4bKcydzi gwkX..Via: SIP/2.0/UDP x.x.128.205:5060;rport=51694;branch=z9hG4bK1D5DA6..From: <sip:5024427578 @x.x.128.205>;tag=24513088-D59..To: <sip:5033441174@x.x.130.36>;tag=F0695368-74F..Date: Tue, 03 Jul 2012 13:37:41 GMT..Call-ID: 1887588D-C44B11E1-BE38D697-98A3E39A@x.x.128.205..Max-Forwards : 69..CSeq: 101 ACK..Content-Length: 0.... # U x.x.128.205:51694 -> x.x.130.36:5060 BYE sip:15033441174@x.x.129.200:5060 SIP/2.0..Via: SIP/2.0/UDP x.x.128.205:5060;branch=z9hG4 bK1D61CE7..From: <sip:5024427578@x.x.128.205>;tag=24513088-D59..To: <sip:5033441174@x.x.130.3 6>;tag=F0695368-74F..Date: Tue, 03 Jul 2012 13:37:41 GMT..Call-ID: 1887588D-C44B11E1-BE38D697-98A3E 39A@x.x.128.205..User-Agent: Cisco-SIPGateway/IOS-12.x..Max-Forwards: 70..Route: <sip:x.x.130 .36;lr=on>..Timestamp: 1341322666..CSeq: 103 BYE..Reason: Q.850;cause=16..Content-Length: 0.... # U x.x.130.34:5060 -> x.x.129.200:5060 BYE sip:15033441174@x.x.129.200:5060 SIP/2.0..Via: SIP/2.0/UDP x.x.130.34;branch=z9hG4bKa2ce. 4ba2ee65.0..Via: SIP/2.0/UDP x.x.128.205:5060;rport=51694;branch=z9hG4bK1D61CE7..From: <sip:502 4427578(a)x.x.128.205>;tag=24513088-D59..To: <sip:5033441174@x.x.130.36>;tag=F0695368-74F..Date : Tue, 03 Jul 2012 13:37:41 GMT..Call-ID: 1887588D-C44B11E1-BE38D697-98A3E39A@x.x.128.205..User- Agent: Cisco-SIPGateway/IOS-12.x..Max-Forwards: 69..Timestamp: 1341322666..CSeq: 103 BYE..Reason: Q .850;cause=16..Content-Length: 0.... # U x.x.129.200:5060 -> x.x.130.34:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP x.x.130.34;branch=z9hG4bKa2ce.4ba2ee65.0,SIP/2.0/UDP x.x.12 8.205:5060;rport=51694;branch=z9hG4bK1D61CE7..From: <sip:5024427578@x.x.128.205>;tag=24513088-D5 9..To: <sip:5033441174@x.x.130.36>;tag=F0695368-74F..Date: Tue, 03 Jul 2012 13:37:46 GMT..Call-I D: 1887588D-C44B11E1-BE38D697-98A3E39A@x.x.128.205..Server: Cisco-SIPGateway/IOS-12.x..Timestamp : 1341322666..Content-Length: 0..CSeq: 103 BYE.... # U x.x.130.36:5060 -> x.x.128.205:51694 SIP/2.0 200 OK..Via: SIP/2.0/UDP x.x.128.205:5060;rport=51694;branch=z9hG4bK1D61CE7..From: <sip :5024427578@x.x.128.205>;tag=24513088-D59..To: <sip:5033441174@x.x.130.36>;tag=F0695368-74F.. Date: Tue, 03 Jul 2012 13:37:46 GMT..Call-ID: 1887588D-C44B11E1-BE38D697-98A3E39A@x.x.128.205..S erver: Cisco-SIPGateway/IOS-12.x..Timestamp: 1341322666..Content-Length: 0..CSeq: 103 BYE....

3 6

Event when location expires?
by Niklas Larsson 03 Jul '12

03 Jul '12

Hi all, is there a way to get a event / log / something when an registry expires? /niklas

2 1

Releasing v3.1.6 - end of packaging maintenance for 3.1 branch
by Daniel-Constantin Mierla 03 Jul '12

03 Jul '12

Hello, in the near future I will package 3.1.6 to mark the end of releasing from 3.1 branch. Of course, patches can still go into that branch, but there is no intent to package again from it. If anyone is willing to push other commits before the packages, please do it in then next 1-2 days. I will do it when I get enough spare time for it, with a short notice before it in that day. 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

Binding arbitrary TLS socket
by Andreas Granig 02 Jul '12

02 Jul '12

Hi, Got some problems when I tried this, not sure how this is intended to work: listen=tls:192.168.51.133:4343 When starting kamailio 3.3, I get the following error: ERROR: <core> [tcp_main.c:2915]: ERROR: tcp_init: bind(10, 0x7f5eea45db24, 16) on 192.168.51.133:4343 : Address already in use ERROR: tls [tls_init.c:314]: Error while initializing TCP part of TLS socket 192.168.51.133:4343 What works though is if I explicitely bind a TCP socket one port below the TLS socket, like this: listen=tcp:192.168.51.133:4342 listen=tls:192.168.51.133:4343 Is this how it's intended to work? Is there a specific reason for this behavior? Andreas

2 2

cfgutils lock / unlock + $sht() variables.
by Vitaliy Aleksandrov 02 Jul '12

02 Jul '12

Hi all, I use $sht() pseudo-variabled provided by htable module for caching various information about sip customers. I thought that htable has some locking mechanism inside it and today while i was reading core-cookbook i payed attention to one example with the lock/unlock functions and $sht(): lock("calls-to::10.10.10.10"); $sht(a=>calls-to::10.10.10.10) = $sht(a=>calls-to::10.10.10.10) + 1; unlock("calls-to::10.10.10.10"); According to that example an explicit locking mechanism should be used when i want to write something to a hash table. On the other hand there are no locks in htable documentation. So the question is when do the explicit locks should be used and for what purpose ?

2 1

TCP Forwarding error: cannot forward to af 2, proto 2 no correspondinglistening socket
by Omar Labs 188 Gmail 02 Jul '12

02 Jul '12

tried many versions in version 1.4 i have this error: Jun 29 21:25:04 [7309] ERROR:core:forward_request: cannot forward to af 2, proto 2 no correspondinglistening socket i do have the TCP port listen explicitly the line for disable-tcp is commented. listen=udp:192.168.169.169:5060 listen=udp:192.168.169.169:5060 if ( src_ip == 192.168.169.180 ) { forward("tcp:192.168.170.156:5060"); } I tried already many options related the forward command , using variables and using forward(); ``Listening on udp: 192.168.169.169 [192.168.169.169]:5060 tcp: 192.168.169.169 [192.168.169.169]:5060 Aliases: tcp: loghost:5060 tcp: kamtest01:5060 udp: loghost:5060 udp: kamtest01:5060 bash-3.00# telnet 192.168.169.169 5060 Trying 192.168.169.169... Connected to 192.168.169.169. Escape character is '^]'. Connection to 192.168.169.169 closed by foreign host. bash-3.00# kamctl ps Process:: ID=0 PID=4541 Type=attendant Process:: ID=1 PID=4542 Type=SIP receiver udp:192.168.169.169:5060 Process:: ID=2 PID=4543 Type=SIP receiver udp:192.168.169.169:5060 Process:: ID=3 PID=4544 Type=SIP receiver udp:192.168.169.169:5060 Process:: ID=4 PID=4545 Type=SIP receiver udp:192.168.169.169:5060 Process:: ID=5 PID=4546 Type=SIP receiver udp:192.168.169.169:5060 Process:: ID=6 PID=4547 Type=SIP receiver udp:192.168.169.169:5060 Process:: ID=7 PID=4548 Type=SIP receiver udp:192.168.169.169:5060 Process:: ID=8 PID=4549 Type=SIP receiver udp:192.168.169.169:5060 Process:: ID=9 PID=4550 Type=SIP receiver udp:192.168.169.169:5060 Process:: ID=10 PID=4551 Type=SIP receiver udp:192.168.169.169:5060 Process:: ID=11 PID=4552 Type=SIP receiver udp:192.168.169.169:5060 Process:: ID=12 PID=4553 Type=SIP receiver udp:192.168.169.169:5060 Process:: ID=13 PID=4554 Type=SIP receiver udp:192.168.169.169:5060 Process:: ID=14 PID=4555 Type=SIP receiver udp:192.168.169.169:5060 Process:: ID=15 PID=4556 Type=SIP receiver udp:192.168.169.169:5060 Process:: ID=16 PID=4557 Type=SIP receiver udp:192.168.169.169:5060 Process:: ID=17 PID=4558 Type=SIP receiver udp:192.168.169.169:5060 Process:: ID=18 PID=4559 Type=SIP receiver udp:192.168.169.169:5060 Process:: ID=19 PID=4560 Type=SIP receiver udp:192.168.169.169:5060 Process:: ID=20 PID=4561 Type=SIP receiver udp:192.168.169.169:5060 Process:: ID=21 PID=4562 Type=timer Process:: ID=22 PID=4563 Type=MI FIFO Process:: ID=23 PID=4564 Type=TCP receiver Process:: ID=24 PID=4565 Type=TCP receiver Process:: ID=25 PID=4566 Type=TCP receiver Process:: ID=26 PID=4567 Type=TCP receiver Process:: ID=27 PID=4568 Type=TCP receiver Process:: ID=28 PID=4569 Type=TCP receiver Process:: ID=29 PID=4570 Type=TCP receiver Process:: ID=30 PID=4571 Type=TCP receiver Process:: ID=31 PID=4572 Type=TCP receiver Process:: ID=32 PID=4573 Type=TCP receiver Process:: ID=33 PID=4574 Type=TCP main O

2 1

How to access reason phrase of winning reply in failure_route
by Barthel Marco (CI/AFU1) 02 Jul '12

02 Jul '12

All, how do I access the reason phrase of winning reply in failure_route? The reply code can be accessed via $T_reply_code. Is there something similar for the reason phrase? Mit freundlichen Grüßen / Best regards Marco Barthel Robert Bosch GmbH (CI/AFU1) www.bosch.com Tel. +49 711 811-3602341 marco.barthel(a)de.bosch.com Sitz: Stuttgart, Registergericht: Amtsgericht Stuttgart, HRB 14000; Aufsichtsratsvorsitzender: Franz Fehrenbach; Geschäftsführung: Volkmar Denner, Siegfried Dais; Stefan Asenkerschbaumer, Bernd Bohr, Rudolf Colm, Dirk Hoheisel, Christoph Kübel, Uwe Raschke, Wolf-Henning Scheider, Werner Struth, Peter Tyroller

2 1

2025

2024

2023

2022

2021

2020

2019

2018

2017

2016

2015

2014

2013

2012

2011

2010

2009

2008

2007

2006

2005

2004

2003

2002

sr-users July 2012