- sr-users - kamailio.org

Re: [SR-Users] Kamailio + Siremis Outbound route
by Daniel-Constantin Mierla 26 May '13

26 May '13

Hello, if you want to send all calls that arrive to kamailio having the prefix 01 to freeswitch: if($rU =~"^01") { $ru = "sip:" + $rU + "@__FREESWITCHIP__"; route(RELAY); exit; } Be sure calls are authenticated at that point and, if needed, the call is not actually coming from freeswitch. Cheers, Daniel On 5/20/13 11:33 AM, Tony Turner wrote: > > Hi > > Version Kamailio v4.0 + Siremis installed on Debian Wheezy via apt-get > install > > I want to use Kamailio as a proxy edge register to our network. > > I have installed Kamailio and freeswitch. > > I can register on Kamailio but I can't route a call from my sip client > from Kamailio to freeswitch and out to PSTN > > Sip client ---- Kamailio-----freeswitch-------SS7 ISDN SIP Gateway --- > Carriers > > If I register direct on Freeswitch I can route out to PSTN but I don't > understand Kamailio routing. > > Can someone let me how I route say from SIP client registered on > Kamailio to prefix 01% which goes out to Freeswitch > > Many Thanks > > Tony > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, San Francisco, USA - June 24-27, 2013 * http://asipto.com/u/katu *

2 1

LCR Module - Rule_id
by Eloy Coto Pereiro 24 May '13

24 May '13

Hi, It Is possible get the rule_id after execute next_gw() function? Cheers

2 4

Calls remain stuck in state 1
by Giany 24 May '13

24 May '13

Hello, We are using dipatcher to limit the concurrent number of calls, problem is that from time to time calls remain stuck in state 1 and it breaks our concurrent limits..I was not able to make a kamailio log with high debug as it happens randomly. Attached is a tcpdump flow: Conv.| Time | serverA | Provider | | | | RemoteEnd | 112 |938.355 | INVITE SDP (g729 g711U GSM X-NSERTPType-100 te...hone-eventRT) | |(5060) ------------------> (5050) | | 112 |938.357 | 100 Trying| | | | |(5050) ------------------> (5060) | | 113 |938.420 | INVITE SDP (g711U g729 telephone-eventRTPType-...)| | |(5050) ------------------> (5060) | | 113 |938.422 | 100 trying -- your call is important to us | | |(5060) ------------------> (5050) | | 113 |938.422 | INVITE SDP (g711U g729 telephone-eventRTPType-...)| | |(5060) ------------------> (1416) | | 113 |938.908 | INVITE SDP (g711U g729 telephone-eventRTPType-...)| | |(5060) ------------------> (1416) | | 113 |939.908 | INVITE SDP (g711U g729 telephone-eventRTPType-...)| | |(5060) ------------------> (1416) | | 113 |941.906 | INVITE SDP (g711U g729 telephone-eventRTPType-...)| | |(5060) ------------------> (1416) | | 113 |945.907 | INVITE SDP (g711U g729 telephone-eventRTPType-...)| | |(5060) ------------------> (1416) | | 112 |948.358 | CANCEL | | |SIP Request | |(5060) <-------------------------------------- (61016) | 112 |948.359 | CANCEL | | |SIP Request | |(5060) ------------------> (5050) | | 112 |948.359 | 200 canceling | |SIP Status | |(5060) --------------------------------------> (61016) | 112 |948.359 | 487 Request Terminated | |SIP Status | |(5050) ------------------> (5060) | | 112 |948.359 | 200 OK | | |SIP Status | |(5050) ------------------> (5060) | | 112 |948.359 | ACK | | |SIP Request | |(5060) ------------------> (5050) | | 112 |948.360 | 487 Request Terminated | |SIP Status | |(5060) --------------------------------------> (61016) | 113 |948.360 | CANCEL | | |SIP Request | |(5050) ------------------> (5060) | | 113 |948.360 | 200 canceling | |SIP Status | |(5060) ------------------> (5050) | | 112 |948.365 | ACK | | |SIP Request | |(5060) <-------------------------------------- (61016) | dialog:: hash=136:689416016 state:: 1 ref_count:: 3 timestart:: 0 timeout:: 0 callid:: 745eed805cad6b9a3e1727d169cf3461@serverA:5050 from_uri:: sip:fromnumber@serverA:5050 from_tag:: as53f6bee4 caller_contact:: sip:fromnumber@serverA:5050 caller_cseq:: 102 caller_route_set:: caller_bind_addr:: udp:serverA:5060 callee_bind_addr:: to_uri:: sip:internalnr@serverA:5060 to_tag:: callee_contact:: callee_cseq:: callee_route_set:: As you see the remoteEnd does not answer at all to this request (due to network issue most likely) and the provider sends a CANCEL after approx 3 seconds. >From what I see the INVITE that is sent from asterisk towards kamailio remains stuck (938.420). We are using Kamailio 3.1.6. Any idea what could be the reason for this? Thank you.

2 1

"SER Getting Started" - what's the copyright and license?
by Olle E. Johansson 24 May '13

24 May '13

http://kamailio.org/docs/ser-getting-started/SER-GettingStarted.pdf I'm looking for the authors of this document to find out what license it is published under. It would be nice to be able to use it and update it instead of starting a new "getting started" from scratch. Please contact me if you get this message! Sorry for cross-posting. /O

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Re: [SR-Users] Simulating kamailio config flows
by Daniel-Constantin Mierla 24 May '13

24 May '13

On 4/22/13 5:30 PM, Juha Heinanen wrote: > Andreas Granig writes: > >> Let me know if this is stupid and/or a complete overkill, but what about >> introducing some kind of dummy mode, where you'd pipe a message into >> kamailio via stdin, and get the resulting message out on stdout (e.g. in >> ngrep style with ip information as first line, plus the content >> following), plus a dump of internals (e.g. vars, avps) on stderr as >> they're being assigned, and once the message is processed, kamailio >> would just shut down again? > in addition to the message, you would need to be able to tell which > ip addr it is coming from and which ip/port in kamailio it is going to. > also, on the output side, kamailio would need to tell which > proto/ip/port it would use to send the message out. Details about going out can be printed on onsend_route. In this route block, one can execute drop and nothing is sent to the wire. It can be a config started with a special define specified with -A parameter. Enabling debugger module with cfg trace should give valuable information about what has been executed from config. From what Andreas suggesting, printing the value of variables as they are assigned is missing, probably can be added by hooking in the interpreter when doing the assignment operation. Grouping all above under some global/command line parameter can be useful to make it easier to do a dry run. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

4 11

add_path() and save()
by אורן אברהם 23 May '13

23 May '13

Dear list. when using registrar module with path support does the following code work: add_path(); save(); or maybe the path is added only for outgouing messages and the save() won't be affected ?

2 3

nat_keepalive() and fix_contact() are incompatible
by David 23 May '13

23 May '13

Hello, So I want to setup a Kamailio SIP Proxy( version 4 ) that will do NAT signalling handling. With the help of nat_traversal and path modules. Attached is my kamailio.cfg file. My problem is I want to do keep alive on users who are behind NAT, so I called nat_keepalive();. The script will work fine, but as soon as I enable either add_contact_alias() or fix_contact(), the keep alive doesn't activate. I modified nat_traversal.c to identify the issue and found that it is comparing the contact header from the request and the reply to find the expiration. Trouble is, the request contact header was taken from before I called fix_contact() and the reply header has the contact from after the fix_contact() call. So my traces show that on the return trip the callback is called, I bet the stack trace looks like this : 1. Some method in tm 2. __tm_reply_in() defined at nat_traversal.c:1358 3. get_register_expire called from nat_traversal.c:1377 and defined at nat_traversal.c:878 4. STR_MATCH_STR called from nat_traversal.c:937 and defined at nat_traversal.c:77 STR_MATCH_STR returns false because the headers are different. I added else { LM_ERR("failed to match because request %s is not reply %s", contact->uri.s, r_contact->uri.s ) ; } to the code and see : May 21 12:18:52 kamailio-dev-nat /usr/sbin/kamailio[30163]: ERROR: nat_traversal [nat_traversal.c:946]: failed to match because request sip:username@mytld#015#012#015#012 is not reply sip:username@myip:59080>;expires=60;received="sip:64.18.189.196:5060"#015#012Server: CenseredUA 5.2-rc1#015#012Content-Length: 0#015#012#015 So as you can see the original Contact header has the server's domain name and the response has the IP address that set by fix_contact(). So I think that nat_keepalive() should not be comparing the Contact from the original request but using the contact header from the modified request. Any ideas what to do next ? Thanks, David

3 5

Radius and FLT_ACCFAILED
by Victor V. Kustov 23 May '13

23 May '13

Hello! I do radius accounting. When I imitate call fail, BYE/CANCEL messages are loose, account stop packets not sending. I see FLT_ACCFAILED flag, can I set it for acc_radius module? -- WBR, Victor JID: coyote(a)bks.tv JID: coyote(a)bryansktel.ru I use FREE operation system: 3.9.2-calculate GNU/Linux

2 1

auth module Readme on website has missing ToC
by Andrew Pogrebennyk 23 May '13

23 May '13

Hi, please check the link http://kamailio.org/docs/modules/stable/modules/auth.html - the Table of Contents is missing, there's only examples list. Andrew

1 0

Re: [SR-Users] Contents of sr-users Digest, Vol 96, Issue 68
by Jignesh Gandhi 23 May '13

23 May '13

Thanks for the quick reply. 1. Re: SCTP question (Daniel-Constantin Mierla) What I meant is if the SCTP association goes down from the Invite, 200 ok w/sdp to BYE. Should there be not an error at t_relay() level , since the association does not exists ? Currently I don't get any error when I do if (!t_relay()) { xlog("L_INFO", "T_Relay return code is $retcode\n"); sl_reply_error(); } I have set up $du with the IP:port of the host where I received the Invite and ACK to 200 ok w/sdp. I have shut down the other side , so I know that SCTP is not up. Thanks, --Jignesh -----Original Message----- From: sr-users-bounces(a)lists.sip-router.org [mailto:sr-users-bounces@lists.sip-router.org] On Behalf Of sr-users-request(a)lists.sip-router.org Sent: Monday, May 20, 2013 2:25 PM To: sr-users(a)lists.sip-router.org Subject: sr-users Digest, Vol 96, Issue 68 Send sr-users mailing list submissions to sr-users(a)lists.sip-router.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users or, via email, send a message with subject or body 'help' to sr-users-request(a)lists.sip-router.org You can reach the person managing the list at sr-users-owner(a)lists.sip-router.org When replying, please edit your Subject line so it is more specific than "Re: Contents of sr-users digest..." Today's Topics: 1. Re: SCTP question (Daniel-Constantin Mierla) 2. Re: [SR-Users] http://kb.asipto.com/kamailio:skype-like-service-in-less-than-one-hour (Daniel-Constantin Mierla) 3. Re: Kamailio + Siremis Outbound route (Daniel-Constantin Mierla) 4. SCTP support of MultiHoming... (Jignesh Gandhi) 5. Re: SCTP support of MultiHoming... (Daniel-Constantin Mierla) ---------------------------------------------------------------------- Message: 1 Date: Mon, 20 May 2013 16:56:55 +0200 From: Daniel-Constantin Mierla <miconda(a)gmail.com> To: Jignesh Gandhi <jigpgandhi(a)gmail.com>, "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org> Subject: Re: [SR-Users] SCTP question Message-ID: <519A39B7.3050503(a)gmail.com> Content-Type: text/plain; charset="iso-8859-1"; Format="flowed" Hello, On 5/20/13 6:12 AM, Jignesh Gandhi wrote: > Hello, > > I recently started using SCTP relay feature of Kamailio and am > receiving SCTP-SIP and relaying it UDP-SIP and vice versa. > > What happens , if an SCTP association is broken from the distant end > in the middle of the call, is there a way to re transmit the SCTP > message via another route ? By middle of the call, do you mean in between the INVITE and the BYE? I am not that familiar with SCTP, but probably a new one is created with the BYE. The same should be happening with tcp. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, San Francisco, USA - June 24-27, 2013 * http://asipto.com/u/katu *

2 1

websocket support for kamailio 4.0.1
by Rupayan Dutta 23 May '13

23 May '13

I have installed kamailio 4.0.1 in centOs 5.8 and trying to connect it using SipML 5 client (supplied by Doubango telecomm) using SIP over websocket.But the client can't connect to the server.Is there any special configuration needed for websocket support in kamailio 4.0.1? please help.

2 1

Kamailio 4.0.1 websocket support
by Rupayan Dutta 23 May '13

23 May '13

I have installed kamailio 4.0.1 in centOs 5.8 and trying to connect it using SipML 5 client (supplied by Doubango telecomm) using SIP over websocket.But the client can't connect to the server.Is there any special configuration needed for websocket support in kamailio 4.0.1? please help

2 1

memcached and usrloc
by Victor V. Kustov 23 May '13

23 May '13

Hello! Can I use memcached for usrloc db? -- WBR, Victor JID: coyote(a)bks.tv JID: coyote(a)bryansktel.ru I use FREE operation system: 3.9.2-calculate GNU/Linux

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[SR-Users] http://kb.asipto.com/kamailio:skype-like-service-in-less-than-one-hour
by johnc 23 May '13

23 May '13

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I updated this config to work with version 4. See below. I have calls working over TLS between jitsi sip clients registered to the same proxy of either one of two proxies I have built using this updated config. I can make calls to/from clients over TLS registered to the openrcs.com proxy from both of these proxies. I can't make calls between the two proxies configured with the config below. Each domain has commercial SSL certs. I have rtpproxy configured and working. I would be very grateful if somebody would check the config and see if I have made a mistake. Many thanks. I will post the final working config so it may be of help to others. Regards, John #!KAMAILIO #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_NAT #!define WITH_TLS #!define WITH_MULTIDOMAIN # # Kamailio (OpenSER) SIP Server v4.0 - By Daniel-Constantin Mierla # http://kb.asipto.com/kamailio:skype-like-service-in-less-than-one-hour # updated for version 4.0 by John Cahill johnc<AT>aktivix.org # # # Refer to the Core CookBook at http://www.kamailio.org/dokuwiki/doku.php # for an explanation of possible statements, functions and parameters. # # Several features can be enabled using '#!define WITH_FEATURE' directives: # # *** To run in debug mode: # - define WITH_DEBUG # # *** To enable mysql: # - define WITH_MYSQL # # *** To enable authentication execute: # - enable mysql # - define WITH_AUTH # - add users using 'kamctl' # # *** To enable IP authentication execute: # - enable mysql # - enable authentication # - define WITH_IPAUTH # - add IP addresses with group id '1' to 'address' table # # *** To enable persistent user location execute: # - enable mysql # - define WITH_USRLOCDB # # *** To enable presence server execute: # - enable mysql # - define WITH_PRESENCE # # *** To enable nat traversal execute: # - define WITH_NAT # - install RTPProxy: http://www.rtpproxy.org # - start RTPProxy: # rtpproxy -l _your_public_ip_ -s udp:localhost:7722 # # *** To enable PSTN gateway routing execute: # - define WITH_PSTN # - set the value of pstn.gw_ip # - check route[PSTN] for regexp routing condition # # *** To enable database aliases lookup execute: # - enable mysql # - define WITH_ALIASDB # # *** To enable multi-domain support execute: # - enable mysql # - define WITH_MULTIDOMAIN # # *** To enable TLS support execute: # - adjust CFGDIR/tls.cfg as needed # - define WITH_TLS # # *** To enable XMLRPC support execute: # - define WITH_XMLRPC # - adjust route[XMLRPC] for access policy # # *** To enable anti-flood detection execute: # - adjust pike and htable=>ipban settings as needed (default is # block if more than 16 requests in 2 seconds and ban for 300 seconds) # - define WITH_ANTIFLOOD # # *** To enhance accounting execute: # - enable mysql # - define WITH_ACCDB # - add following columns to database #!ifdef ACCDB_COMMENT ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; #!endif ####### Defined Values ######### # *** Value defines - IDs used later in config #!ifdef WITH_MYSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!define DBURL "mysql://kamailio:kamailiorw@localhost/kamailio" #!endif #!ifdef WITH_MULTIDOMAIN # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 #!else #!define MULTIDOMAIN 0 #!endif # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 ####### Global Parameters ######### #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=3 log_stderror=yes #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL0 fork=yes children=4 /* uncomment the next line to disable TCP (default on) */ #disable_tcp=yes /* uncomment the next line to disable the auto discovery of local aliases based on reverse DNS on IPs (default on) */ #auto_aliases=no /* add local domain aliases */ #alias="sip.mydomain.com" /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ #listen=udp:10.0.0.10:5060 /* port to listen to * - can be specified more than once if needed to listen on many ports */ port=5060 #!ifdef WITH_TLS enable_tls=yes #!endif tcp_connection_lifetime=3605 ####### Custom Parameters ######### # These parameters can be modified runtime via RPC interface # - see the documentation of 'cfg_rpc' module. # # Format: group.id = value 'desc' description # Access: $sel(cfg_get.group.id) or @cfg_get.group.id # #!ifdef WITH_PSTN # PSTN GW Routing # # - pstn.gw_ip: valid IP or hostname as string value, example: # pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" # # - by default is empty to avoid misrouting pstn.gw_ip = "" desc "PSTN GW Address" #!endif ####### Modules Section ######## # set paths to location of modules #!ifdef LOCAL_TEST_RUN mpath="modules_k:modules" #!else mpath="/usr/lib64/kamailio/modules_k/:/usr/lib64/kamailio/modules/" #!endif #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif loadmodule "mi_fifo.so" loadmodule "kex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "mi_rpc.so" loadmodule "acc.so" #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #!ifdef WITH_IPAUTH loadmodule "permissions.so" #!endif #!endif #!ifdef WITH_ALIASDB loadmodule "alias_db.so" #!endif #!ifdef WITH_MULTIDOMAIN loadmodule "domain.so" #!endif #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_NAT loadmodule "nathelper.so" loadmodule "rtpproxy.so" #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_ANTIFLOOD loadmodule "htable.so" loadmodule "pike.so" #!endif #!ifdef WITH_XMLRPC loadmodule "xmlrpc.so" #!endif # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- registrar params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ #modparam("registrar", "max_contacts", 10) # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default ww do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") modparam("auth_db", "use_domain", MULTIDOMAIN) # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) # use caching #modparam("domain", "db_mode", 1) JC commented # register callback to match myself condition with domains list modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@somedomain.com") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/etc/kamailio/tls.cfg") #!endif #!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 16) modparam("pike", "remove_latency", 4) # ----- htable params ----- # ip ban htable with autoexpire after 5 minutes modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") #!endif #!ifdef WITH_XMLRPC # ----- xmlrpc params ----- modparam("xmlrpc", "route", "XMLRPC"); modparam("xmlrpc", "url_match", "^/RPC") #!endif ####### Routing Logic ######## # Main SIP request routing logic # - processing of any incoming SIP request starts with this route route { # per request initial checks route(REQINIT); # NAT detection route(NAT); # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); # authentication route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } # dispatch requests to foreign domains route(SIPOUT); ### requests for my local domains # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations to PSTN route(PSTN); # user location service route(LOCATION); route(RELAY); } route[RELAY] { #!ifdef WITH_NAT if (check_route_param("nat=yes")) { setbflag(FLB_NATB); } if (isflagset(FLT_NATS) || isbflagset(FLB_NATB)) { route(RTPPROXY); } #!endif /* example how to enable some additional event routes */ if (is_method("INVITE")) { #t_on_branch("BRANCH_ONE"); t_on_reply("REPLY_ONE"); t_on_failure("FAIL_ONE"); } if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood dection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($sht(ipban=>$si)!=$null) { # ip is already blocked xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); exit; } if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); $sht(ipban=>$si) = 1; exit; } } #!endif if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); } exit; } } # Handle SIP registrations route[REGISTRAR] { if (is_method("REGISTER")) { if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging ## setbflag(FLB_NATSIPPING); } if (!save("location")) sl_reply_error(); exit; } } # USER location service route[LOCATION] { #!ifdef WITH_ALIASDB # search in DB-based aliases alias_db_lookup("dbaliases"); #!endif if (!lookup("location")) { switch ($rc) { case -1: case -3: t_newtran(); t_reply("404", "Not Found"); exit; case -2: sl_send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } } # Presence server route route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; }; if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); exit; } return; } # Authentication route route[AUTH] { #!ifdef WITH_AUTH if (is_method("REGISTER")) { # authenticate the REGISTER requests (uncomment to enable auth) if (!www_authorize("$td", "subscriber")) { www_challenge("$td", "0"); exit; } if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else { #!ifdef WITH_IPAUTH if(allow_source_address()) { # source IP allowed return; } #!endif # authenticate if from local subscriber if (from_uri==myself) { if (!proxy_authorize("$fd", "subscriber")) { proxy_challenge("$fd", "0"); exit; } if (is_method("PUBLISH")) { if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else { if ($au!=$fU) { sl_send_reply("403","Forbidden auth ID"); exit; } } consume_credentials(); # caller authenticated } else { # caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (!uri==myself) { sl_send_reply("403","Not relaying"); exit; } } } #!endif return; } # Caller NAT detection route route[NAT] { #!ifdef WITH_NAT force_rport(); setflag(FLT_NATS); if (nat_uac_test("19")) { if (method=="REGISTER") { fix_nated_register(); } else { fix_nated_contact(); } setflag(FLT_NATS); } #!endif return; } # RTPProxy control route[RTPPROXY] { #!ifdef WITH_NAT if (is_method("BYE")) { unforce_rtp_proxy(); } else if (is_method("INVITE")){ rtpproxy_offer(); } if (!has_totag()) add_rr_param(";nat=yes"); #!endif return; } # Routing to foreign domains route[SIPOUT] { if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(RELAY); } } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN rotuing enabled but pstn.gw_ip not defined\n"); return; } # route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) return; # only local users allowed to call if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); route(RELAY); exit; #!endif return; } # XMLRPC routing #!ifdef WITH_XMLRPC route[XMLRPC] { # allow XMLRPC from localhost if ((method=="POST" || method=="GET") && (src_ip==127.0.0.1)) { # close connection only for xmlrpclib user agents (there is a bug in # xmlrpclib: it waits for EOF before interpreting the response). if ($hdr(User-Agent) =~ "xmlrpclib") set_reply_close(); set_reply_no_connect(); dispatch_rpc(); exit; } send_reply("403", "Forbidden"); exit; } #!endif # Sample branch router branch_route[BRANCH_ONE] { xdbg("new branch at $ru\n"); } # Sample onreply route onreply_route[REPLY_ONE] { xdbg("incoming reply\n"); #!ifdef WITH_NAT if ((isflagset(FLT_NATS) || isbflagset(FLB_NATB)) && status=~"(183)|(2[0-9][0-9])") { rtpproxy_answer(); } if (isbflagset("6")) { fix_nated_contact(); } #!endif } # Sample failure route failure_route[FAIL_ONE] { #!ifdef WITH_NAT if (is_method("INVITE") && (isbflagset(FLB_NATB) || isflagset(FLT_NATS))) { unforce_rtp_proxy(); } #!endif if (t_is_canceled()) { exit; } # uncomment the following lines if you want to block client # redirect based on 3xx replies. ##if (t_check_status("3[0-9][0-9]")) { ##t_reply("404","Not found"); ## exit; ##} # uncomment the following lines if you want to redirect the failed # calls to a different new destination ##if (t_check_status("486|408")) { ## sethostport("192.168.2.100:5060"); ## append_branch(); ## # do not set the missed call flag again ## t_relay(); ##} } -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.12 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/ iQIcBAEBAgAGBQJRmf9DAAoJELy1jPQ1KER71LUP+wZGvLn3wQuEVpLQIzZ4U7YR UD7l6OpLQEEp349zmWgyNoOWmcZrRcBYhgKHZde+9ZKIDFTMiWUxqPJJrmcDUTi1 ybq98z9Es9KpT6d5ytv/AMG2DNpM6zRgIVTLXqXfnyzAqdbGZJUhIiLwXiLHBaxy P/JBJCmri4qeahIr/hNJOcmD3pUi8KMS2AE2cxvgl4Y4X1bwEPxDH374Z55/ka3A IJlSGTwFUFjW/Ujc5ih4+TYyRgO4khoCtnzQaSQk8VVNEGGkcuIXV3j9JOOQ+yOT B7TdsbeCu5pBZZe8Qy/aIcTRonvtpexrUXI5jSzfxRaWFYjw01KINBRIprNTaUQ0 w+c+MQWz8gOLI3hcw4/9if9fkJMbmUD7deaS6N5Ovu647TLMbhgB3hDI+77+s5Cs EPJNeW8P1iSPJVuASrx9jqfDNeOKvPOxvd6CemCBHngoSW3NAsuospmfDPo+8QFA MbtJQXJ4p2b6g6CrP1FVx1q2cdPbvC8ZFUqxkfPj8AUtAg5vtKvjU+PPnbtDDwbi RxF0KGpCnixVtc3AkTBdX9JcP+4Pc6RvprNlCEsi1AfaK6yzSlCZeblC2OjZ/2oa PkSaWQeBwLQKMi90QVhoSJTI7Z4O6T03j09V9Wul7qkZorcHkZlzi+W4tBqyxL5R 8prN+zlBCZ3nZN5nFosn =39pZ -----END PGP SIGNATURE-----

2 5

Websockets and outbound config file
by אורן אברהם 22 May '13

22 May '13

Dear list Does someone has a working config file for a registrar + outbound edge proxy on the same kamailio server instance that works ? (like the /examples/websocket.cfg but updated to use outbound & path) thanks...

1 0

[OT] Crocodile Network & WebRTC Launch broadcast
by Peter Dunkley 22 May '13

22 May '13

Hello, Crocodile is having a launch event this evening for our new SIP network and SDK (making use of Kamailio with SIP over WebSocket and outbound). The launch will be broadcast live from 18:30 BST. Anyone interested can view the broadcast at http://lnx.so/croc (the link currently points to our blog but will go live properly shortly before the broadcast). Regards, Peter -- Peter Dunkley Technical Director Crocodile RCS Ltd

1 0

Re: [SR-Users] I need you help-----about Kamailio 3.3.x andAsterisk 10.7.0 Realtime Integration
by Gertjan Wolzak 21 May '13

21 May '13

Hello, I looked at the asterisk config, looks like you have not configured that. You need to add a dialplan in the extensions config and trunks in the sip confiig. If you want to make the dialplan realtime as well, you should add the dialplan to the db. Success. Gertjan -----Original message----- From: zhengyw zhengyw(a)neusoft.com Date: Mon, 20 May 2013 04:07:33 +0200 To: sr-users(a)lists.sip-router.org Subject: Re: [SR-Users] I need you help-----about Kamailio 3.3.x andAsterisk 10.7.0 Realtime Integration > Hi Barry: > This issue have not been resolved after following by your method that > modifed the video1_sipregs table struct,attachment is table info and > asterisk log. > Can you help me with this problem? thank you very much! > > best > zhengyw > *asterisk log* asterisk_log.txt > <http://sip-router.1086192.n5.nabble.com/file/n118567/asterisk_log.txt> > *dbinfo.txt* dbinfo.txt > <http://sip-router.1086192.n5.nabble.com/file/n118567/dbinfo.txt> > > > > -- > View this message in context: http://sip-router.1086192.n5.nabble.com/I-need-you-help-about-Kamailio-3-3-… > Sent from the Users mailing list archive at Nabble.com. > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

2 1

db_url bug?
by Ján Hrnko 21 May '13

21 May '13

Hi, I have a problem when using uri_db module. When I leave everything on default and not specify modparam "uri_db", "db_url", then it will not connect to the database. In the var/log/syslog I have seen that it tries to connect to database okmaialio instead of kamailio. Is it bug, or did I misconfig something? I left the kamctlrc file intact and everything to default... can you confirm this please? Thank you!

2 1

remove_body() leaves header Content-Type: application/sdp ?
by Konstantin M. 21 May '13

21 May '13

Hello, I'm using a very ugly code to test remove_body(): request_route { remove_body(); $rd = "192.168.254.85"; t_relay(); } Original INVITE (with SDP): --------------------------------- U 192.168.254.102:5060 -> 192.168.254.104:5060 INVITE sip:123@192.168.254.104 SIP/2.0. Via: SIP/2.0/UDP 192.168.254.102:5060;branch=z9hG4bK50cdeeb8. Max-Forwards: 70. From: "10005" <sip:10005@192.168.254.102>;tag=as78ddcd16. To: <sip:123@192.168.254.104>. Contact: <sip:10005@192.168.254.102:5060>. Call-ID: 79a1c78f01914f1814bfcb7d622cf735@192.168.254.102:5060. CSeq: 102 INVITE. User-Agent: Asterisk PBX 1.8.7.1. Date: Sat, 18 May 2013 05:47:32 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. Content-Type: application/sdp. Content-Length: 269. . v=0. o=root 1763381182 1763381182 IN IP4 192.168.254.102. s=Asterisk PBX 1.8.7.1. c=IN IP4 192.168.254.102. t=0 0. m=audio 16162 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. a=sendrecv. --------------------------------- Modified INVITE (after remove_body()): --------------------------------- U 192.168.254.104:5060 -> 192.168.254.85:5060 INVITE sip:123@192.168.254.85 SIP/2.0. Via: SIP/2.0/UDP 0.0.0.0;branch=z9hG4bK6c5c.6160e017.0. Via: SIP/2.0/UDP 192.168.254.102:5060;branch=z9hG4bK50cdeeb8. Max-Forwards: 70. From: "10005" <sip:10005@192.168.254.102>;tag=as78ddcd16. To: <sip:123@192.168.254.104>. Contact: <sip:10005@192.168.254.102:5060>. Call-ID: 79a1c78f01914f1814bfcb7d622cf735@192.168.254.102:5060. CSeq: 102 INVITE. User-Agent: Asterisk PBX 1.8.7.1. Date: Sat, 18 May 2013 05:47:32 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. *Content-Type: application/sdp.* Content-Length: 0. . --------------------------------- So, remove_body() is actualy removes SDP from request but leaves Content-Type header. Why ?

2 1

SCTP question
by Jignesh Gandhi 20 May '13

20 May '13

Hello, I recently started using SCTP relay feature of Kamailio and am receiving SCTP-SIP and relaying it UDP-SIP and vice versa. What happens , if an SCTP association is broken from the distant end in the middle of the call, is there a way to re transmit the SCTP message via another route ? Thanks, --Jignesh Gandhi

2 1

Re: [SR-Users] Msilo configuration‏
by sipatse＠in.gr 20 May '13

20 May '13

2 1

Kamailio + Siremis Outbound route
by Tony Turner 20 May '13

20 May '13

Hi Version Kamailio v4.0 + Siremis installed on Debian Wheezy via apt-get install I want to use Kamailio as a proxy edge register to our network. I have installed Kamailio and freeswitch. I can register on Kamailio but I can't route a call from my sip client from Kamailio to freeswitch and out to PSTN Sip client ---- Kamailio-----freeswitch-------SS7 ISDN SIP Gateway --- Carriers If I register direct on Freeswitch I can route out to PSTN but I don't understand Kamailio routing. Can someone let me how I route say from SIP client registered on Kamailio to prefix 01% which goes out to Freeswitch Many Thanks Tony

1 0

Kamailio 3.1.x and FreeSWITCH 1.0.6+ for Media Services and SBC Author: Daniel-Constantin Mierla
by Carlton Thompson 18 May '13

18 May '13

http://kb.asipto.com/freeswitch:kamailio-3.1.x-freeswitch-1.0.6d-sbc#dokuwi… Hello I was trying to use your tutorial on the above page. I am trying to do so using PGQL vs MYSQL. I changed all the mysql parts to pgsql however when i try to start kamailio it fails. I am building the system on alpine linux and attached my kamailio.cfg and kamctlrc. i am able to setup using the default config making necessary changes but when i use yours it fails. Any help would be appreciated. -- Regards, Carlton Thompson, MS Doctoral Student College of Engineering University of Florida Gainesville, FL 32611

1 0

Can I use Kamailio as B2BUA
by Kamal Palei 18 May '13

18 May '13

Dear Kamailio experts I have a typical use case where I want Kamailio to behave as a B2BUA. What I mean here is (assume Kamailio is using TCP for SIP call establishment) 1. For each call it should create a separate TCP connection with next proxy in path. 2. When call ends, it should close that connection. If that call is active for 10hrs, then connection should stay alive till 10hrs. Please somebody help. Thanks kamal

3 8

New developer: Victor Seva
by Daniel-Constantin Mierla 17 May '13

17 May '13

Hello, I want to announce that a new person got developer GIT write access to repository: Victor Seva. He helped pushing back Kamailio in official Debian packages and lately has submitted many patches to various modules, among upcoming ones is some work on newly added per module debug level (see the logs from yesterday's IRC meeting for more about this feature). His git commit id is: vseva My warm welcome and looking forward to future work within the project! Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, San Francisco, USA - June 24-27, 2013 * http://asipto.com/u/katu *

1 0

IMS In Kamailio 4.0.0 - what is it?
by Olle E. Johansson 17 May '13

17 May '13

Jason Penton, Richard Good and Carsten Bock has written an excellent article about the IMS extensions in Kamailio 4.0.0. What it is, what you can do with it and what they're currently working on. Go read it on our web site now to learn more about this work: http://www.kamailio.org/w/2013/05/ims-kamailio/ Have a great weekend! /O

2 1

Radius - new style AVP's (misc_radius)
by Victor V. Kustov 17 May '13

17 May '13

Good day! Now I test new style AVP's for radius response storing. Today additional radius responses return in special radpair: SIP-AVP - name, string "email:sr-users@lists.sip-router.org session-timeout#161 nexthopip:h323/0001111@myvoip-gate.kamailio.org" - value. I don't know why developers chose this way. But it's addition kamailio's feature, so let it be. But if radius server responded with standard pairs? Sending Access-Accept of id 60 to 172.16.X.X port 59736 Session-Timeout = 4261674 next-hop-ip = "SIP/00111222333444@cisco-out" SIP-AVP = "email:sr-users@lists.sip-router.org session-timeout#161 next-hop-ip:h323/0001111@myvoip-gate.kamailio.org" session-protocol = "SIP" No chance to get full radius response at this moment. Only parse SIP-AVP pair. This mean that when you wanna integrate K. in existing infrastructure (in my case - billing), u need work with K. and radius. But why, if you already had worked voip devices? I think more flexible solution ADD in misc_radius function to get radpairs as AVP's. See that i got: misc_radius [functions.c:159]: AVP 'Session-Timeout'/0='null'/4261674 has been added misc_radius [functions.c:143]: AVP 'next-hop-ip'/0='SIP/005555777888@cisco-out'/0 has been added misc_radius [functions.c:143]: AVP 'session-protocol'/0='SIP'/0 has been added After radius response i get: $avp(Session-Timeout)=4261674 (integer) $avp(next-hop-ip)="SIP/005555777888@cisco-out" (string) $avp(session-protocol)="SIP" (string) I think its usable and more flexible. I see 3 way to integrate it: 1. additionally with existing radius_load_caller_avps/radius_load_callee_avps (i make same for test) 2. new function in same module, like radius_authorize_request (caller,callee). 3. replace existing style. it's bad for backward compability, of cause. Or it stay my local patch :) If it really needed and usable i may made any (1..3) way with doc and examples :) diff --git a/modules/misc_radius/functions.c b/modules/misc_radius/functions.c index 2a42024..f156939 100644 --- a/modules/misc_radius/functions.c +++ b/modules/misc_radius/functions.c @@ -107,6 +107,60 @@ error: return -1; } +static void generate_avps_rad(VALUE_PAIR* received) +{ + int_str name, val; + unsigned short flags; + VALUE_PAIR *vp; + + vp = received; + + for( ; vp ; vp=vp->next) { + flags = AVP_NAME_STR; + switch(vp->type) + { + case PW_TYPE_STRING: + flags |= AVP_VAL_STR; + name.s.len = strlen(vp->name); + val.s.len = strlen(vp->strvalue); + name.s.s = vp->name; + val.s.s = vp->strvalue; + if (add_avp( flags, name, val ) < 0) { + LM_ERR("unable to create a new AVP\n"); + } else { + LM_DBG("AVP '%.*s'/%d='%.*s'/%d has been added\n", + (flags&AVP_NAME_STR)?name.s.len:4, + (flags&AVP_NAME_STR)?name.s.s:"null", + (flags&AVP_NAME_STR)?0:name.n, + (flags&AVP_VAL_STR)?val.s.len:4, + (flags&AVP_VAL_STR)?val.s.s:"null", + (flags&AVP_VAL_STR)?0:val.n ); + } + continue; + case PW_TYPE_INTEGER: + name.s.len = strlen(vp->name); + name.s.s = vp->name; + val.n = vp->lvalue; + if (add_avp( flags, name, val ) < 0) { + LM_ERR("unable to create a new AVP\n"); + } else { + LM_DBG("AVP '%.*s'/%d='%.*s'/%d has been added\n", + (flags&AVP_NAME_STR)?name.s.len:4, + (flags&AVP_NAME_STR)?name.s.s:"null", + (flags&AVP_NAME_STR)?0:name.n, + (flags&AVP_VAL_STR)?val.s.len:4, + (flags&AVP_VAL_STR)?val.s.s:"null", + (flags&AVP_VAL_STR)?0:val.n ); + } + continue; + default: + LM_ERR("skip attribute type %d (non-string)", vp->type); + continue; + } + return; + } +} + /* Generate AVPs from Radius reply items */ static void generate_avps(struct attr *attrs, VALUE_PAIR* received) @@ -212,6 +266,7 @@ int radius_load_caller_avps(struct sip_msg* _m, char* _caller, char* _s2) LM_DBG("success\n"); rc_avpair_free(send); generate_avps(caller_attrs, received); + generate_avps_rad(received); rc_avpair_free(received); return 1; } else { -- WBR, Victor JID: coyote(a)bks.tv JID: coyote(a)bryansktel.ru I use FREE operation system: 3.8.4-calculate GNU/Linux

2 6

Msilo configuration
by andreas tseiko 17 May '13

17 May '13

Dear All, We would like to store SIP text Messages when the destination Subscriber is Offline.We have insert into kamailio cfg file the configuration lines below.Unfortunately storing messages is unsuccessful.Any ideas of what missing or what could be wrong? Best regards. ************************************************************************************************************loadmodule "msilo.so" #!ifdef WITH_MSILOmodparam("msilo","db_url","mysql://[% kamailio.proxy.dbrwuser %]:[% kamailio.proxy.dbrwpw %]@[% database.dbhost %]/[% kamailio.proxy.dbname %]")modparam("msilo", "db_table", "silo")modparam("msilo","from_address","sip:registrar@xxxxxxx.local")modparam("msilo", "from_address", "sip:$rU@xxxxxxx.local")modparam("msilo","contact_hdr","Contact: <sip:registrar@xx.xx.xx.xx:5062>;msilo=yes\r\n")modparam("msilo","content_type_hdr","Content-Type: text/plain\r\n")modparam("msilo","offline_message","*** User $rU is offline!")#!endif modparam("usrloc", "db_mode", 0) initial value was “1” #########################################################################Store messages to offline Subs########################################################################route{ if ( !mf_process_maxfwd_header("10") ) { sl_send_reply("483","To Many Hops"); exit; }; if (uri==myself) { { # for testing purposes, simply okay all REGISTERs if (method=="REGISTER") { save("location"); log("REGISTER received -> dumping messages with MSILO\n"); # MSILO - dumping user's offline messages if (m_dump()) { log("MSILO: offline messages dumped - if they were\n"); }else{ log("MSILO: no offline messages dumped\n"); }; exit; }; # domestic SIP destinations are handled using our USRLOC DB if(!lookup("location")) { if (! t_newtran()) { sl_reply_error(); exit; }; # we do not care about anything else but MESSAGEs if (!method=="MESSAGE") { if (!t_reply("404", "Not found")) { sl_reply_error(); }; exit; }; log("MESSAGE received -> storing using MSILO\n"); # MSILO - storing as offline message if (m_store("$ru")) { log("MSILO: offline message stored\n"); if (!t_reply("202", "Accepted")) { sl_reply_error(); }; }else{ log("MSILO: offline message NOT stored\n"); if (!t_reply("503", "Service Unavailable")) { sl_reply_error(); }; }; exit; }; # if the downstream UA does not support MESSAGE requests # go to failure_route[1] t_on_failure("1"); t_relay(); exit; }; # forward anything else t_relay();} failure_route[1] { # forwarding failed -- check if the request was a MESSAGE if (!method=="MESSAGE") { exit; }; log(1,"MSILO:the downstream UA doesn't support MESSAGEs\n"); # we have changed the R-URI with the contact address, ignore it now if (m_store("$ou")) { log("MSILO: offline message stored\n"); t_reply("202", "Accepted"); }else{ log("MSILO: offline message NOT stored\n"); t_reply("503", "Service Unavailable"); };}***********************************************************************************************

1 0

Kamailio and PJSIP Issue
by Ali Jawad 17 May '13

17 May '13

Hi I did setup kamailio to handle websocket, and I am testing using pjsip, my problem is that I can logon using the pjsip client but SIP does not register, pjsip displays a SIP connection timeout error. My xlite client can logon just fine though. The log from kamailio can be seen below http://pastebin.com/XrsQM8Cf Please advice

1 0

[OT] announcement: openrcs.com - free SIP service
by Daniel-Constantin Mierla 17 May '13

17 May '13

Hello, following several requests during Kamailio World, but also in the direction of federating open source and local communities via SIP (as a result of FOSDEM panel and discussions), a SIP service was built up to be offered for free on best effort manner - the hardware and bandwidth are offered by asipto.com, the service is using the domain openrcs.com. Main goal is to offer a free option for community members (but not restricted to) to create and use SIP accounts. The plan is to enable many (common) new features to allow people to play with, helping also to test kamailio. There is a rather basic (and alpha stage) web portal based on wordpress+buddypress, good enough for the moment to create an account and tune some options - it is available at: - https://www.openrcs.com Along with voice/video calling and instant messaging, the service has enabled features such as simple presence, msrp relay, xcap server and websocket (web portal embeds a bare sip app built using jssip library). Other features will be enable over the time, but have in mind **it is a not a telephony service**, thus no such specific services like emergency, pstn interconnect, etc. Hope it will be useful for many people - who ever wants to try it, head to https://www.openrcs.com ! Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, San Francisco, USA - June 24-27, 2013 * http://asipto.com/u/katu *

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config
by johnc 16 May '13

16 May '13

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi Daniel, I nearly forgot, can you post your openrcs config ? Many thanks, John -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.12 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/ iQIcBAEBAgAGBQJRlOvSAAoJELy1jPQ1KER7juoP/RmguFzHtfzdyr+RydqtE/xp TDfyzQCOh/iciln+pXANMtgajwyyHa83504HP05uhXnk3RLvLEoze/8t7/KkwSdR mksANtDJ/cR4Q9LI9iXVplIUORIR98qGl0/C+LODSU5Hsd3/FS5V0v0B+d1gEZPV na4UPlu/FGgCbRfUa/eFVfo/gE38fU8TNbtZqwggSYmmXrKdEUlzqN8cxCeFg0fT LVygThcLhtam1r4NQWpXOjzHl+qDrVNnIwuDng0W1kg3OLvt2GMS2l0FsLglZcMf xr6CcVQT+BvKUKGgubF1PfjcMuVhNGXgLH4dcXEbNs5Z41pDL0ap7nnz/SXiZQxa qHNEgFV9cJQJtPGK0fTuIPVV4YFA7iaMoXlLJbvwI5YwZUG9W3WnytDBuLUcXnSG 0E8Js9KHUHj975BQMaMUFDohLKL1kFv+wiRKbxT0QDVsEtMTMugi3vkLGRgskrNA e798aVReyh+8gB1hcm9bEUCt7Pc/aQBWYiLM40JqdlVS2Y28X4jQ8BYtSVROLGQa Ws0/nspwKWIXmC5DKF1AJNlBI1HSdW78PO1R2hc5uCatjcidthrvOmCwwhC3B/yA 9rPD0ECquOkRzLw7IHIFzM9pAruRDlKVdjhRaoN4hMuN57e/nqKuASPpMMXbj45L 4rSCPsSV9/A7kCLd9miV =88yR -----END PGP SIGNATURE-----

2 1

Delay in "100 Trying" reply message
by Robert R 16 May '13

16 May '13

Hi, There is a big delay in our Kamailio server when repling "100 Trying" after incoming INVITE messages (more than 500 ms). It seems my perl script (and database query) is executed before "100 Trying" being send out. I was wondering if sending "100 Trying" is configurable so I can send it right after INVITE and before executing my perl script. request_route { # per request initial checks route(REQINIT); # NAT detection route(NATDETECT); # handle requests within SIP dialogs route(WITHINDLG); if (is_method("INVITE")) { perl_exec("test"); } t_check_trans(); route(SIPOUT); route(PRESENCE); route(REGISTRAR); .. route(RELAY); } (morthen Thanks R

3 4

IRC Development Meeting
by Daniel-Constantin Mierla 16 May '13

16 May '13

Hello, it's probably the time to have a IRC development meeting to set the targets for the next major release as well as discuss what is needed these days around the project. I proposed next week on Thursday, May 16, at 14:00GMT (16:00 Berlin time). I created a page to collect the topics, only few from my list being added for now: * http://www.kamailio.org/wiki/devel/irc-meetings/2013a Feel free to add topics there. Also, I would be available on Tuesday at the same time if proves to be more convenient for more people. If none of these dates suit for majority, we can look for other alternatives -- just propose and we can start a poll. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, San Francisco, USA - June 24-27, 2013 * http://asipto.com/u/katu *

2 3

event_route
by Bruno Bresciani 16 May '13

16 May '13

Hi All, in a call forking, after one branch answer the call (200 OK reply), a CANCEL SIP message has been sending to other/another branch(es) and I need to process this/these cancellations in configuration file. After reading some documentations, I discovered there is event_route[tm:local-request] block, which is executed when tm generates internally and sends a SIP request, Such cases are: SIP messages sent by msilo module SIP messages sent by presence server SIP messages sent by dialog module SIP messages sent via MI or CTL interfaces I didn't understand very well this cases, so I insert event_route block in my kamailio.cfg but neither CANCEL SIP message or other requests generated by tm module was handled by event_route. I must be using wrong concept to handle this CANCEL SIP message, it's possible handle this messages in configuration file? Best Regards

2 6

redirecting irc channel #sip-router to #kamailio
by Daniel-Constantin Mierla 16 May '13

16 May '13

Hello, already suggested on the irc channels, I plan to redirect #sip-router to #kamailio to group in a single place and have consistency with the name of the project. That means making #sip-router an invite-only channel and automatically redirecting those that want to join to #kamailio. Hope is ok with everyone, invite-only flag can be removed later. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, San Francisco, USA - June 24-27, 2013 * http://asipto.com/u/katu *

1 0

Freepbx 2.11.0rc1 with Asterisk 11.3.0 and Kamailio 4.0.1
by Trevor Burt 16 May '13

16 May '13

Dear All, using the following link:- http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb I have successfully integrated Asterisk and Kamailio on the same box for testing, but am now facing the problem of getting Freepbx to use the same MySQL database tables. My Kamailio and Asterisk install uses the following tables:- sipusers sipregs voicemail voicemail_data voicemail_messages However Freepbx uses the following tables. devices users Any idea on how I fully integrate Freepbx with my Asterisk and Kamailio configuration using Mysql? My goal is to have a Kamailio LB to multiple Asterisk/Freepbx servers sharing the same extension and voicemail databases. I will also probably integrate a2billing and share the a2billing database between the Asterisk/Freepbx servers to create a fully scalable solution. Any advice most gratefully received.

3 2

SER-GettingStarted… update?
by David Wilson 16 May '13

16 May '13

Hi All, One of the most thorough documents I've found for an introduction to Kamailio is the SER-GettingStarted.pdf, but it is now quite dated. Is there an updated version available, or in the works? Cheers, Dave.

1 0

[PATCH] Memory corruption using s.subst transformation
by Juha Heinanen 16 May '13

16 May '13

Daniel wrote: please do the patch to store the new value in _tr_buffer and attach it to mailing list or bug tracker. I haven't looked at code yet, but sounds like there is indeed an issue. I will review the patch and apply it. What was the conclusion about this? Was the patch ever submitted and applied? If not, is the bug still there or has is been fixed by some other means? I'm asking because this looked like a quite serious issue. -- Juha

2 4

Kamailio crash
by kiran bhosale 16 May '13

16 May '13

Hi we have developed the custom module where in we have created shared memory area which is not used by any other module or process and we locked it using locks as recommended. but because of locks the kamailio is getting crashed every now and then . is it really important to protect the shared memory area with locks. if yes then how to avoid the crash ?? so to locate the crash we built kamailio wit h MM_DBG , and other GCC debug flags . but we are getting error "/No symbol table info available/" . how to locate the bugs. the kamailio is built wit h 1 MB of shared memory on MIPS based VOIP gateway which is having * MB of flash and 16 MB or memory (RAM) . thanks in advance .

2 3

kamailio as a proxy
by Giuseppe Carella 16 May '13

16 May '13

Hi all, I'm trying to use kamailio as a proxy between webrtc clients and an openIMSCore. For this reason I'm currently using the dispatcher module to distribute all the requests coming from the clients to the IMS core, and enabled the websocket module for getting the SIP messages in. SO far it works fine for the registration, but it doesn't for the INVITEs. I noticed that kamailio receives the message but it doesn't deliver it to the clients. Please find attached the log and the configuration file that I used. I'm not an expert with the configuration, so any help or feedback would be very appreciated. Thank you in advance. Best regards, Giuseppe Carella

2 1

ERROR messages in DEBUG level
by Alex Solt 16 May '13

16 May '13

Hi, After changing log level from INFO to DEBUG level, I am seeing bunch of ERROR messages which are unknown to me: May 15 10:26:44 ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=505 a=17 n=ifMay 15 10:26:44 ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=498 a=26 n=is_methodMay 15 10:26:44 ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=505 a=6 n=routeMay 15 10:26:44 ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=770 a=17 n=ifMay 15 10:26:44 ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=765 a=26 n=mf_process_maxfwd_headerMay 15 10:26:44 DEBUG: maxfwd [mf_funcs.c:85]: value = 68May 15 10:26:44 ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=775 a=17 n=ifMay 15 10:26:44 ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=770 a=27 n=sanity_checkMay 15 10:26:44 DEBUG: <core> [parser/msg_parser.c:202]: DEBUG: get_hdr_body : content_length=0May 15 10:26:44 DEBUG: <core> [parser/msg_parser.c:104]: found end of headerMay 15 10:26:44 DEBUG: <core> [parser/parse_to.c:178]: DEBUG: add_param: tag=snl_0156137698_NSN_CLIENTMay 15 10:26:44 DEBUG: <core> [parser/parse_to.c:802]: end of header reached, state=29May 15 10:26:44 DEBUG: sanity [mod_sanity.c:255]: sanity checks result: 1May 15 10:26:44 ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=508 a=6 n=routeMay 15 10:26:44 ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=976 a=3 n=returnMay 15 10:26:44 ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=511 a=6 n=routeMay 15 10:26:44 ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=833 a=17 n=ifMay 15 10:26:44 ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=780 a=25 n=has_totagMay 15 10:26:44 DEBUG: siputils [checks.c:106]: totag foundMay 15 10:26:44 ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=786 a=17 n=ifMay 15 10:26:44 ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=830 a=17 n=ifMay 15 10:26:44 ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=786 a=25 n=loose_route The error message are complaining about simple statements in the config file: if (is_method("OPTIONS")) { sl_send_reply("200", "Ok"); exit; } if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } I appreciate your help. Thanks,AS

2 1

home pbx server experience
by u 15 May '13

15 May '13

I would like to share my experience with kamailio and other home pbx servers. Kamailio on my kirkwood home router for my 6 SIP users is perhaps overkill: I don't really need mysql and "scalability". But at last I finally managed to make calling between registered users work stable. My voip clients only work in all NAT scenarios if I work around some bugs: to use csipsimple on android I had to change rtpproxy_manage() to rtpproxy_manage("c") in kamailio's default config, so that problems with conflicting c: entries in the SDP go away. I propose kamailio could ship with a special example kamailio-compatible.cfg that doesn't try to be RFC compliant, but compatible to the most common voip clients. Right now the only thing I would change for this is the option for rtpproxy_manage, but I'm sure others will know more common quirks that could safely be enabled to increase compatibility. I think this compatibility idea is what yate sticks to for their defaults. In freeswitch you also have to do it all manually, and it's much more work to figure things out in their enormous config files. The other SIP proxies I had tried before kamailio officially fit all my requirements, including support for multihomed dynamic IPs, but contrary to their claims it didn't work. Yate was easy to set up, but the default dialplan is more confusing than powerful and after having made everything work I realised yate was clogging my CPU and RAM and after some time always randomly stopped working. This is with only 2 users connected! It also wasn't possible to fix NAT sdp while leaving the codecs section in the SDP alone at the same time. I tried to debug the code, but the C++ was so complex that I had to give up. Freeswitch was much more difficult to setup, a multihomed setup with dynamic IP was super buggy and it also didn't help that the unintuitive configuration is all in complex unreadable XML configuration files. Kamailio and rtpproxy don't officially support dynamic IP address, but I can just restart both each time my DSL provider forces me to a new IP address. This happens automatically in the night and is no big hassle really. The most simple, least-featureful solution works best it seems. Now the last problem I have with kamailio: I don't know how to connect my accounts to my sip providers (i.e. Sipgate, Betamax, Dellmont). I would like a simple way to do this, preferably without other features that always seem to complicate the matters. Is there something more lightweight and simple than asterisk, freeswitch and yate, that people use successfully for this task together with kamailio and rtpproxy? u

5 4

log reply message number
by Robert R 15 May '13

15 May '13

Hi, I am trying to add a log when receiving 1xx or 2xx reply message: # manage incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") { xlog("L_INFO","reply message: $T_reply_code ci:$ci"); route(NATMANAGE); } } Adding the above line does not log the reply message number (180 or 183 or 200, ...). Thanks, R

3 3

Re: [SR-Users] home pbx server experience
by u 15 May '13

15 May '13

Should UAC work with Sipgate and freevoipdeal (dellmont)? Thanks for your answers everyone. 2013/5/15 u <ueberwachungsstaat(a)googlemail.com>: > Should UAC work with Sipgate and freevoipdeal (dellmont)? > > Thanks for your answers everyone. > > 2013/5/15 Stefan Sayer <stefan.sayer(a)googlemail.com>: >> >> if you don't get it all (auth + registrations) working with kamailo, >> you might try sems (iptel.org/sems) with sbc module (auth of outgoing >> calls) and for registrations either reg_agent (cfg file) or >> db_reg_agent (accounts from db) modules. >> >> load_plugins=uac_auth;registrar_client;reg_agent;sbc;xmlrpc2di >> application=sbc >> >> >> http://ftp.iptel.org/pub/sems/doc/current/ModuleDoc_sbc.html >> http://ftp.iptel.org/pub/sems/doc/current/ModuleDoc_reg_agent.html >> http://ftp.iptel.org/pub/sems/doc/current/AppDoc.html >> >> BR >> Stefan >> >> o Edson - Lists on 05/15/2013 06:40 AM: >>> Hi... >>> >>> Use UAC module to manage registrations and play a little with the >>> config (INVITE section) to forward output calls correctly. >>> --- >>> Edson. >>> >>> Em 14/05/2013 11:27, u escreveu: >>>> I would like to share my experience with kamailio and other home pbx >>>> servers. >>>> >>>> Kamailio on my kirkwood home router for my 6 SIP users is perhaps >>>> overkill: I don't really need mysql and "scalability". But at last I >>>> finally managed to make calling between registered users work stable. >>>> My voip clients only work in all NAT scenarios if I work around some >>>> bugs: to use csipsimple on android I had to change rtpproxy_manage() >>>> to rtpproxy_manage("c") in kamailio's default config, so that problems >>>> with conflicting c: entries in the SDP go away. >>>> >>>> I propose kamailio could ship with a special example >>>> kamailio-compatible.cfg that doesn't try to be RFC compliant, but >>>> compatible to the most common voip clients. Right now the only thing I >>>> would change for this is the option for rtpproxy_manage, but I'm sure >>>> others will know more common quirks that could safely be enabled to >>>> increase compatibility. I think this compatibility idea is what yate >>>> sticks to for their defaults. In freeswitch you also have to do it all >>>> manually, and it's much more work to figure things out in their >>>> enormous config files. >>>> >>>> The other SIP proxies I had tried before kamailio officially fit all >>>> my requirements, including support for multihomed dynamic IPs, but >>>> contrary to their claims it didn't work. >>>> Yate was easy to set up, but the default dialplan is more confusing >>>> than powerful and after having made everything work I realised yate >>>> was clogging my CPU and RAM and after some time always randomly >>>> stopped working. This is with only 2 users connected! It also wasn't >>>> possible to fix NAT sdp while leaving the codecs section in the SDP >>>> alone at the same time. I tried to debug the code, but the C++ was so >>>> complex that I had to give up. >>>> Freeswitch was much more difficult to setup, a multihomed setup with >>>> dynamic IP was super buggy and it also didn't help that the >>>> unintuitive configuration is all in complex unreadable XML >>>> configuration files. >>>> >>>> Kamailio and rtpproxy don't officially support dynamic IP address, but >>>> I can just restart both each time my DSL provider forces me to a new >>>> IP address. This happens automatically in the night and is no big >>>> hassle really. The most simple, least-featureful solution works best >>>> it seems. >>>> >>>> Now the last problem I have with kamailio: I don't know how to connect >>>> my accounts to my sip providers (i.e. Sipgate, Betamax, Dellmont). >>>> I would like a simple way to do this, preferably without other >>>> features that always seem to complicate the matters. Is there >>>> something more lightweight and simple than asterisk, freeswitch and >>>> yate, that people use successfully for this task together with >>>> kamailio and rtpproxy? >>>> >>>> u >>>> >>>> _______________________________________________ >>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >>>> sr-users(a)lists.sip-router.org >>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>>> . >>>> >>> >>> _______________________________________________ >>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >>> sr-users(a)lists.sip-router.org >>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>> >>

1 0

HELP! I can't install siremis 3.0 for kaimailio after install kamailio, It take a long long time to wait for and no end.
by Future Lian 15 May '13

15 May '13

I can't install siremis 3.0 for kaimailio after install kamailio, It take a long long time to wait for and no end. in CRIT.log of siremis/log, I get '05/14/2013','09:13:30','CRIT','ExceptionHandler','cache_dir must be a directory','' Anyone can tell me how to do? Thank you all.

2 1

Kamailio 3.3.x is crashing frequently...
by Krishna Kurapati 15 May '13

15 May '13

Hi, When the Server is running it is crashing occasionally. (gdb) bac #0 0x000000000043359e in ?? () #1 0x0000000000000000 in ?? () (gdb) file /usr/local/sbin/kamailio Reading symbols from /usr/local/sbin/kamailio...done. (gdb) bac #0 free_lump_list (lump_list=<value optimized out>) at data_lump.c:504 #1 del_nonshm_lump (lump_list=<value optimized out>) at data_lump.c:661 #2 0x00007f571ff6cd9c in ?? () #3 0x00000000008db480 in mem_pool () #4 0x00007f571ff6c7e1 in ?? () #5 0x000000000139086e in ?? () #6 0x00000039294546e1 in ?? () #7 0x000000000140c7d0 in ?? () #8 0x00007f561f7acb80 in ?? () #9 0x00007f561f3fa6e0 in ?? () #10 0x00007f561f3fa6e0 in ?? () #11 0x0000000000000001 in ?? () #12 0x0000000000000000 in ?? () What could be going wrong here? Krish Kura

2 1

kamailio 4.0.1 and nokia e72 udp
by hiro 14 May '13

14 May '13

using kamailio-4.0.1_src.tar.gz with rtpproxy and a nokia e72 behind NAT registered via UDP I get no voice. The e72 strangely sends a single udp packet from a wrong port (49152) before the rtp stream should start. This quirk of the e72 doesn't seem to work well with rtpproxy if the following analysis is true: rtpproxy detects that single UDP packet from the wrong port and so we think that is where everything else will also come from and stop listening on other ports. we then also answer on that wrong port. Although all subsequent incoming packets arrive from the expected (49172) port sent also in the sdp and to the right one we had sent in the sdp earlier we never receive them, because we still listen on that wrong port with that one bogus packet. tcplog, rtp should be running, but no voice: ... 00:02:13.829682 IP p5795BC1A.dip0.t-ipconnect.de.49172 > koln-5d81d2a9.pool.mediaWays.net.49184: UDP, length 73 00:02:13.847063 IP koln-5d81d2a9.pool.mediaWays.net.49184 > p5795BC1A.dip0.t-ipconnect.de.49152: UDP, length 73 ... /usr/sbin/rtpproxy -l 93.129.210.169 -s unix:/var/run/rtpproxy/rtpproxy.sock -u kamailio kamailio -p /var/run/rtpproxy/rtpproxy.pid ... DBUG:handle_command: received command "Uc100,8,0,98 _39ZSWQBoIfK_g5i6W1ACRdB0cBVaB 192.168.1.56 40000 o5r7rmu5p1hc7vci0591;1" INFO:handle_command: new session _39ZSWQBoIfK_g5i6W1ACRdB0cBVaB, tag o5r7rmu5p1hc7vci0591;1 requested, type strong INFO:handle_command: new session on a port 49184 created, tag o5r7rmu5p1hc7vci0591;1 INFO:handle_command: pre-filling caller's address with 192.168.1.56:40000 DBUG:doreply: sending reply "49184 93.129.210.169 " DBUG:handle_command: received command "Lc100,98 _39ZSWQBoIfK_g5i6W1ACRdB0cBVaB 87.149.188.26 49172 o5r7rmu5p1hc7vci0591;1 5qt1pav3 INFO:handle_command: lookup on ports 49184/49196, session timer restarted INFO:handle_command: pre-filling callee's address with 87.149.188.26:49172 DBUG:doreply: sending reply "49196 93.129.210.169 " INFO:rxmit_packets: callee's address filled in: 87.149.188.26:49152 (RTP) DBUG:handle_command: received command "D _39ZSWQBoIfK_g5i6W1ACRdB0cBVaB 5qt1pav3dsafun5p47o30tj2od1urm6v o5r7rmu5p1hc7vci0591" INFO:handle_delete: forcefully deleting session 1 on ports 49184/49196 INFO:remove_session: RTP stats: 1 in from callee, 3465 in from caller, 3466 relayed, 0 dropped INFO:remove_session: RTCP stats: 0 in from callee, 0 in from caller, 0 relayed, 0 dropped INFO:remove_session: session on ports 49184/49196 is cleaned up DBUG:doreply: sending reply "0 what do you guys think? it only occured with kamailio+rtpproxy so far. any insight about why the e72 sends such packet? it doesn't happen when i use sip over tcp. is my analysis right, is it a bug in rtpproxy? greetings hiro

4 9

Kamailio for Debian blog?
by Daniel Pocock 14 May '13

14 May '13

I'd like to write a brief blog about the status of WebRTC in Debian, with a focus on SIP I understand Kamailio 4.0.1 is already in unstable, is that recommended for potential WebSocket users? Has anybody else written any quickstart blog about WebRTC with that particular version, possibly with examples that are consistent with the Debian usage? For client side, SIPml5 is packaged, and I've had discussions with the JsSIP guys about packaging. For TURN, has anybody tried the TURN server project from Google code? It appears more advanced than the existing two TURN servers in Debian (e.g. it has database-backed authentication) http://code.google.com/p/rfc5766-turn-server/ and there is a package in progress: http://mentors.debian.net/package/rfc5766-turn-server

6 11

Iptel SIP used with Bria iPhone App
by JE 14 May '13

14 May '13

I'm using an iPhone Bria app and have input Iptel SIP information in the settings, however, when I try to dial a land line I get a "Forbidden (403)" error message. What could be wrong?

2 1

4.0 forking behaviour
by Alex Balashov 14 May '13

14 May '13

Hello, Has something changed about default forking behaviour in >= 4.0? I have a scenario where INVITEs processed by the proxy first hit a redirect server, catch a 302, and then append another branch and iterate over one or more outbound routes. In the past, this worked fine. After I upgraded to 4.0, I am seeing two branches at a time on the outbound routes, after the initial branch to the redirect server. The desired behaviour is serial forking at all times. tm:failure_reply_mode is set to 3, as it always has been. Any ideas would be appreciated; thank you! -- Alex -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 United States Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/

2 9

Re: [SR-Users] confirm cd4884b9a2ea4a7cd1b91182b28f1f5da884f92b
by Future Lian 14 May '13

14 May '13

I can't install siremis 3.0 for kaimailio after install kamailio and begin to install siremis in CRIT.log of siremis/log, i get '05/14/2013','09:13:30','CRIT','ExceptionHandler','cache_dir must be a directory','' Anyone can tell me how to do? Thank you all. 2013/5/14 <sr-users-request(a)lists.sip-router.org> > Mailing list subscription confirmation notice for mailing list > sr-users > > We have received a request from 218.79.223.111 for subscription of > your email address, "lianhc(a)gmail.com", to the > sr-users(a)lists.sip-router.org mailing list. To confirm that you want > to be added to this mailing list, simply reply to this message, > keeping the Subject: header intact. Or visit this web page: > > > http://lists.sip-router.org/cgi-bin/mailman/confirm/sr-users/cd4884b9a2ea4a… > > > Or include the following line -- and only the following line -- in a > message to sr-users-request(a)lists.sip-router.org: > > confirm cd4884b9a2ea4a7cd1b91182b28f1f5da884f92b > > Note that simply sending a `reply' to this message should work from > most mail readers, since that usually leaves the Subject: line in the > right form (additional "Re:" text in the Subject: is okay). > > If you do not wish to be subscribed to this list, please simply > disregard this message. If you think you are being maliciously > subscribed to the list, or have any other questions, send them to > sr-users-owner(a)lists.sip-router.org. > -- Future Lian

2 1

Does Kamailio support PRACK ?
by Alex Solt 13 May '13

13 May '13

Hi, Does kamailio support PRACK method ? Any configuration change is needed?It appears Kamailio does not like the PRAK when increasing Cseq. Here is the call flow: A send INVITE --> Kamailio --> proxy the packet to BB send 180 Ringing --> Kamailio -> proxy the packet to AA send PRAK (increase Cseq) --> Kamailio --> proxy the packet to BB send 200 OK --> Kamailio -> proxy the packet to AKamailio re-send 180 Ringing to A Thanks,AS

3 4

how to install kamailio with websocket function and msrp function on ubuntu？
by 李启明 13 May '13

13 May '13

hi, i am chinese,i am not good at kamailio.i want to install kamailio with websocket function and msrp function on ubuntu,could you give me a help？

3 3

Doing automatic unregister when a WEBSOCKET connection is closed.
by אורן אברהם 13 May '13

13 May '13

Dear List. I'm trying to issue an automatic unregister() (of a registration) when a websocket connection is closed. it is useful when the browser is closed without issuing REGISTER with expires=0. i've tried implementing this using the event_route[websocket:closed] but i don't know how to find the registration uri from the $si:$sp. (that is needed for the unregister() function from the registrar module) thanks for all replies in advance....

5 15

Re: [SR-Users] I need you help-----about Kamailio 3.3.x and Asterisk 10.7.0 Realtime Integration
by zhengyw 13 May '13

13 May '13

Hello: I have followed the steps in the guide(http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0…: 1. Could you find any mistake in my configuration file? 2. I also wonder how could I be sure of "Be sure you update the listen IP and port as well if Asterisk is running on the same system with Kamailio" and "Be sure you configure Asterisk to not authenticate SIP requests coming from Kamailio". thank you very much PS：attachment is kamailio、asterisk's congifure file. Best Regards zhengyw ----- Original Message ----- From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org> Sent: Monday, May 13, 2013 3:26 PM Subject: Re: [SR-Users] I need you help-----about Kamailio 3.3.x and Asterisk 10.7.0 Realtime Integration > Hello, > > I am not that familiar to troubleshoot asterisk configuration files, but > from logs I could see the resulting URI is: > > INVITEsip:106@(null) SIP/2.0 > > That is wrong, meaning something incorrect is done when setting it in > asterisk. Maybe someone else can help more with asterisk. > > Cheers, > Daniel > > On 5/13/13 4:02 AM, zhengyw wrote: >> hello daniel: >> thank you very much! but I can't find the problem in the asterisk. >> attachment is asterisk's configure file, kamailio's configure file and >> data.can you help with this problem? >> >> ps:asterisk version 10.70, kamailio version 3.3.1, ubuntu version 12.04 >> >> >> Best Regards, >> zhengyw kamailio.cfg >> <http://sip-router.1086192.n5.nabble.com/file/n118319/kamailio.cfg> >> sip.conf <http://sip-router.1086192.n5.nabble.com/file/n118319/sip.conf> >> extconfig.conf >> <http://sip-router.1086192.n5.nabble.com/file/n118319/extconfig.conf> >> extensions.conf >> <http://sip-router.1086192.n5.nabble.com/file/n118319/extensions.conf> >> db_result.txt >> <http://sip-router.1086192.n5.nabble.com/file/n118319/db_result.txt> >> >> >> >> -- >> View this message in context: http://sip-router.1086192.n5.nabble.com/I-need-you-help-about-Kamailio-3-3-… >> Sent from the Users mailing list archive at Nabble.com. >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >> sr-users(a)lists.sip-router.org >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > -- > Daniel-Constantin Mierla - http://www.asipto.com > http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda > Kamailio Advanced Training, San Francisco, USA - June 24-27, 2013 > * http://asipto.com/u/katu * > > > _______________________________________________ > 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 --------------------------------------------------------------------------------------------------- Confidentiality Notice: The information contained in this e-mail and any accompanying attachment(s) is intended only for the use of the intended recipient and may be confidential and/or privileged of Neusoft Corporation, its subsidiaries and/or its affiliates. If any reader of this communication is not the intended recipient, unauthorized use, forwarding, printing, storing, disclosure or copying is strictly prohibited, and may be unlawful.If you have received this communication in error,please immediately notify the sender by return e-mail, and delete the original message and all copies from your system. Thank you. ---------------------------------------------------------------------------------------------------

1 0

branche_route support
by Bruno Bresciani 13 May '13

13 May '13

Hi all, currently I have been used kamailio and rtpproxy to listen on two different network (bridge mode), and the register of sip users are saved on different locations ("aliases" and "location") according the interface where register is received. When there is a call to a user registered, I use the registered() function of module registrar on my route[ ] block, to discover the network where sip user is connected and use the correct parameters in rtpproxy functions. >From now, I need that my plataform has forking call support and when the same sip user is registered on both networks, I have to use the branch_route[ ] block to process every branch separately because is necessary different rtpproxy parameters. I am with difficult in branch_route to set the correct rtpproxy parameters because I can't find a way to know the network that the branch will be send. Could anybody tell me if exist a form to solve my problem? Best Regards

2 3

SIP notify to UAC, triggered from DB or web
by Attila Megyeri 13 May '13

13 May '13

Hello, I would like to send a SIP NOTIFY message to my SIP client(s), initiated by an event on the server side. //OFF Asterisk has a similar functionality, for instance ("sip notify <phone vendor>-reboot"), to request a reboot of the phone device from the server side. There is no need for an associated SUBSCRIBE request in such a case. ON// Our requirement is, that if a specific event happens on the server side, I want to send an event to the client. My first idea is to use SIP NOTIFY, but perhaps a SIP MESSAGE could do the job as well. The client is registered over TLS, if necessary it could even subscribe for the event. What would you recommend, using the websockets? Or creating a module from the scratch? Thank you in advance, Attila

2 2

(no subject)
by 李启明 13 May '13

13 May '13

hi, i am from china.i have install a kamailio server with wss support ,and i set 4443 as the port(set 8080 as the ws port);the result of "netstat -tnl" is: "Active Internet connections (only servers) Proto Recv-Q Send-Q Local Address Foreign Address State tcp 0 0 127.0.0.1:53 0.0.0.0:* LISTEN tcp 0 0 127.0.0.1:631 0.0.0.0:* LISTEN tcp 0 0 my_ip:4443 0.0.0.0:* LISTEN tcp 0 0 my_ip:5060 0.0.0.0:* LISTEN tcp 0 0 my_ip:5061 0.0.0.0:* LISTEN tcp 0 0 127.0.0.1:3306 0.0.0.0:* LISTEN tcp 0 0 my_ip:8080 0.0.0.0:* LISTEN" but,when i use ws,it's ok;when i use wss,it come up a 106 error!why?

2 3

i386 vs. amd64
by Juha Heinanen 11 May '13

11 May '13

debian wheezy multiarch support allows installation of i386 packages to amd64 host provided that there is i386 version of libraries used by the package. is there any advantage of amd64 kamailio versus i386 one? how does kamailio benefit from 64 bit architecture or does it not? if not, the on wheezy amd64 packages would not be needed. this is what e.g. google has done with chrome browser packages. i have noticed that amd64 version of kamailio requires quite a lot more memory than i386 version, which it not necessarily a good thing. -- juha

3 2

Is RTP proxy compatilble with SER
by Priyaranjan Nayak 10 May '13

10 May '13

Hi All, I am new to SER and SEMS . Can we run RTP proxy with SER ? If yes, which version of SER will compatible with RTP proxy ? -- Thanks Priyaranjan

2 1

Presence and watcher_domain
by andpe 10 May '13

10 May '13

Hi I use PRESENCE, PRESENCE_REGINFO and others, with Cisco phones (trying to run :-)). The phones are seen as being registered in the PRESENCE, REGINFO work unless properly. But I have a question: in the table active_watchers watcher_domain field is the IP address of the phone and not the domain address. Is this normal? Applies only to telephones using REGINFO. eg Jitsi has a valid entry. Andy

3 2

Is RTP proxy compatible with SER
by Priyaranjan Nayak 10 May '13

10 May '13

Hi All, I am new to SER and SEMS . Can we run RTP proxy with SER ? If yes, which version of SER will compatible with RTP proxy ? -- Thanks Priyaranjan

2 1

Cleaning registrations and presentity table after restart
by אורן אברהם 10 May '13

10 May '13

Dear list I want to remove all raws from the presentity table after restart of kamailio. how can i achieve this ? (preferably without deleting the table manualy)

2 1

Serial forking
by Vitaliy Aleksandrov 10 May '13

10 May '13

Hello. As I understand t_load_contacts() / t_next_contacts() can help only in case when branches have different "q" values. The only way I see is to get all contacts via reg_fetch_contacts(), save the list to an AVP and then iterate through the list in the way described in tm module docs. Does anybody know a better way of how serial forking can be implemented if all contacts have equal "q" values ?

2 2

I need you help-----about Kamailio 3.3.x and Asterisk 10.7.0 Realtime Integration
by zhengyw 10 May '13

10 May '13

Hello: I followed the step by step guide (http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb) that describe the realtime integration between Kamailio and Asterisk. I have no problem with registration but when I try a call from 101 to 102 I get the followng error : 1. in the asterisk console: " Unresolvable destination (478/SL)" 2. in the kamailio log: May 6 17:01:38 WH-PC /usr/sbin/kamailio[4192]: ERROR: <core> [resolve.c:1540]: ERROR: sip_hostport2su: could not resolve hostname: "(null)" May 6 17:01:38 WH-PC /usr/sbin/kamailio[4192]: ERROR: tm [ut.h:327]: failed to resolve "(null)" May 6 17:01:38 WH-PC /usr/sbin/kamailio[4192]: ERROR: tm [t_fwd.c:1530]: ERROR: t_forward_nonack: failure to add branches May 6 17:01:38 WH-PC /usr/sbin/kamailio[4192]: ERROR: sl [sl_funcs.c:371]: ERROR: sl_reply_error used: Unresolvable destination (478/SL) Any idea about the cause of this problem? ps:kamailio and asterisk are running in the same machine. Best Regards, zhengyw 郑友闻医疗终端部东软熙康健康有限公司沈阳市浑南新区新秀街2号，A1楼216 Postcode：110179 Tel：+86 24 83662278 Mobile：15242493836 Email : zhengyw(a)neusoft.com http://www.neusoft.com --------------------------------------------------------------------------------------------------- Confidentiality Notice: The information contained in this e-mail and any accompanying attachment(s) is intended only for the use of the intended recipient and may be confidential and/or privileged of Neusoft Corporation, its subsidiaries and/or its affiliates. If any reader of this communication is not the intended recipient, unauthorized use, forwarding, printing, storing, disclosure or copying is strictly prohibited, and may be unlawful.If you have received this communication in error,please immediately notify the sender by return e-mail, and delete the original message and all copies from your system. Thank you. ---------------------------------------------------------------------------------------------------

2 1

How to configure Kamailio to use different source port for different call
by Kamal Palei 10 May '13

10 May '13

Dear All I have a Kamalio proxy, and number of phones are connected to it. When all the phones make call, Kamailio forwards00 request to main proxy from same source port. Is there any configuration option in Kamailio.cfg for changing source port for each call. Basically one dedicated connection per session. Kindly let me know if it is there. Thanks Kamal

2 1

Need kamailio-3.1.5 configuration
by Priyaranjan Nayak 10 May '13

10 May '13

Hi All, After successful install of kamailio-3.1.5 I am trying to start kamailio,but it throws following error loading modules under /usr/local/lib/kamailio/ modules_k/:/usr/local/lib/kamailio/modules/:/usr/local/lib/kamailio/modules_s/ 0(2580) WARNING: <core> [socket_info.c:912]: WARNING: fix_hostname: could not rev. resolve 192.168.90.210 0(2580) WARNING: <core> [socket_info.c:912]: WARNING: fix_hostname: could not rev. resolve 192.168.90.210 0(2580) WARNING: <core> [socket_info.c:912]: WARNING: fix_hostname: could not rev. resolve 192.168.90.210 0(2580) WARNING: <core> [socket_info.c:912]: WARNING: fix_hostname: could not rev. resolve 192.168.90.210 Listening on udp: 192.168.90.210:5035 udp: 192.168.90.210:5036 udp: 127.0.0.1:5036 tcp: 192.168.90.210:5035 tcp: 192.168.90.210:5036 tcp: 127.0.0.1:5036 Aliases: tcp: localhost:5036 udp: localhost:5036 0(2580) INFO: <core> [tcp_main.c:4730]: init_tcp: using epoll_et io watch method (config) 0(2582) INFO: usrloc [hslot.c:53]: locks array size 512 0(2582) INFO: <core> [udp_server.c:184]: INFO: udp_init: SO_RCVBUF is initially 112640 0(2582) INFO: <core> [udp_server.c:235]: INFO: udp_init: SO_RCVBUF is finally 262142 Please suggest what can be done ? -- Thanks Priyaranjan

2 1

Siremis v4.0.0 Released
by Elena-Ramona Modroiu 09 May '13

09 May '13

Hi, a new release of Siremis is out - v4.0.0 - web management interface that is compatible out of the box with Kamailio v4.0.x series. More details, including link to install tutorial, are available at: - http://siremis.asipto.com/2013/05/08/siremis-v4-0-0-released/ Regards, Ramona

2 2

Do we need rtpproxy for all kind of NAT?
by John Chen 08 May '13

08 May '13

Hi, I can configure kamailio + rtpproxy to enable calling between user behind the NAT. Thanks for the example config, it is very easy to did it. Do we need rtprpoxy for all kind of NAT? From what i have read, only symmetric NAT that hard to be traversal-ed. While other type of NAT can be traversaled using STUN. But my device that behind openwrt router can't work without rtpproxy. Is it expected behaviour? Or i'm doing something wrong in the config? Thanks

4 6

Call routing
by mark＠brightvoip.co.uk 08 May '13

08 May '13

Hi all, Posted a similar query a few weeks ago, without much interest - any advice appreciated. I have two sites and will send calls between them. I have Kamailio at each site which will route the calls out/in. There are multiple distinct network routes between the sites, accessible via different IP addresses. Each Kamailio has multiple IP's, one for each route. The purpose of the multiple routes is mainly fault tolerance. Some of the network links are unreliable, so routing must adapt when route(s) are unavailable. When all routes are available, all should handle some traffic, at differing ratios to match the bandwidth available to each route (e.g route A - 50%, route B - 30%, route C - 20%). I know that the Dispatcher can manage the routing for the SIP traffic, with the %ge distribution, and with SIP OPTIONS 'pings' to detect route availability. My main headache is that RTP must follow the same route as SIP for each call. After a bit of web digging, I was thinking of a solution where each of the Kamailio servers will run multiple instances of rtpproxy (one for each ip/route). Then once the dispatcher has chosen a route for the call, to use the matching rtpproxy instance to direct the audio. Any comments or alternate solutions/suggestions would be of interest. Many thanks, Mark

2 1

Problem with Realtime Kamailio-Asterisk integration
by aaaa aaaa 08 May '13

08 May '13

Hello, I followed the step by step guide (http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb) that describe the realtime integration between Kamailio and Asterisk. I have no problem with registration but when I try a call from 101 to 102 I get the followng error: [Mar 31 01:18:44] NOTICE[32330][C-00000006]: chan_sip.c:25195 handle_request_invite: Call from '101' (192.168.1.100:5060) to extension '103' rejected because extension not found in context 'DEFAULT NULL'. Kamailio and Asterisk are running in the same machine. Any idea about the cause of this problem? Best Regards, Theo

5 6

sip peering
by johnc 07 May '13

07 May '13

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I have built a kamailio based phone system along the lines of the tutorial below. I have updated it for version 4.0 and it is running successfully on Debian 7.0. I have set my system up with a Gandi SSL cert rather than a self signed one. http://kb.asipto.com/kamailio:skype-like-service-in-less-than-one-hour I want to federate/peer with another proxy running Kamailio with TLS. What do I need to do to make this work? Thanks. Regards, John -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.12 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/ iQIcBAEBAgAGBQJRiDBUAAoJELy1jPQ1KER7sBoP/1G7I5rlDWcy2eC5GaN7aIa6 xjtKSB8/2cHIyws9d5Zb1mJFKbKgRPJRlSODxUD4rqLdUwq3oIP8buVBAp4rx7UT Xg8POBwRTPqccF7+AxelhkplnWsh11rVhfCT/2L0FLgKSelGGCaAl5SNI5rwj9HO On9ACQeCarpP4UbyZK15iSkreMe4eMfErQgqhfyJwjmlIoEYlBiT4QrUPR3I0MbA jWJq2IWl2Jz/ZeDe2Z7vX6vDY6iidYL+kfFqyXOzoajw+/HTpzok79pGwPSMWZVx 2UHximuxyzJKqGXH1G1Ai4sRrWurM/02jH5iaddb3K2k/fGI8NxvDVgS4p7hyUTA JTmXetcYhTFstgQlnaD0U+FO268uPtGbVQjzuxnYy/bwHRnRV8sxKsjihpRTJrUY R6/QU2xMuaKigJn2FTzSodLMeHtSifUc3Od/9THoh4r6RlQOwMMxBGH+xSdTIXOo wmzLqly3sFPEsSp/30ncs33xT+P7SvMGzu+XHx16UZHCvrFhG3ME10a4C06dn53H mOvXPQ/xm0a47uytp859lbhG3PiyuiF5OkBmuaxndKon2w+HisEZ6+FqteYL3Xjq K3a2z8aoDGFzkYgqqyh5Fpn/hGzd7/E8KkI6Ldrffo2QY3uQx8CWnwXZnYsqFzXy dFS7DTGFXWeKZSAwDO+I =Iezv -----END PGP SIGNATURE-----

2 2

Max concurrent TCP connection that Kamailio support ?
by Khoa Pham 07 May '13

07 May '13

Hi, What is the maximum number of concurrent TCP connection that Kamailio can handle ? I see the max_tcp_connections (which is set to 2048), is that the answer ? How to test for this ? Many thanks -- Khoa Pham HCMC University of Science Faculty of Information Technology

2 3

Question about relaying
by Leo Brown 07 May '13

07 May '13

Hi My application is for mobile (MVNO) users making calls, which will generally end up on the PSTN via our carriers. MVNO Carrier --> Our Edge Switch --> Our PSTN Switch --> Our carrier's switch | |---> Our internal routing switch The issue is with our PSTN switch and the fact that it is not staying in the SIP signalling path, so when the call ultimately between our MVNO carrier and outbound Carrier is established (200 OK) the MVNO carrier and PSTN carrier begin talking to each other. When the MVNO carrier issues a BYE to the outbound carrier, the outbound carrier does not then receive this packet as they are firewalled (and always will be). What is the correct method of relaying calls through Kamailio but not passing on the Contact: header info? I have read that forcing a change of Contact is not the right way. Leo

3 7

Presence and Cisco 79xx phones BLF
by Andrzej 07 May '13

07 May '13

Is it possible to run on phones Cisco 79xx series BLFs? So to show the status of the shared lines. Phones registered with Kamailio. Andy

3 2

rfc: restructuring source code tree
by Daniel-Constantin Mierla 07 May '13

07 May '13

Hello, among the topics discussed just before the last major release series (4.0.x) was one about restructuring the source code tree. It started mainly as a proposal to move source code belonging to core in a dedicated folder, but there could be more variants. It's time start discussing if we do it, and if yes, how. Here is what I could collect so far. a) no change, keep like it is b) move source code of core in a dedicated folder named 'core' c) move all source code in a folder named src, with following sub-structure src/core - for core src/lib - for internal libraries src/modules - for module Feel free to propose other variants or express your opinion regarding the proposed options. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, San Francisco, USA - June 24-27, 2013 * http://asipto.com/u/katu *

2 1

missing AVP OriginatingRequest in LIR
by Khue Nguyen Minh 07 May '13

07 May '13

Hi all, I use ims_icscf in Kamailio to send LIR message to HSS. But, In LIR message that I received in HSS haven't OriginatingRequest AVP. How I can insert it into LIR message? Thanks & Best regards, Khue Nguyen.

1 0

max number of pipes for pipelimit module?
by Julia 07 May '13

07 May '13

Hello, Is there a limit on the max number of pipes for pipelimit module? Thanks, Julia

2 2

Publish Failed
by Raj Roy Ghandhi 07 May '13

07 May '13

Hi, I installed and configured the latest Kamalio and all the features works except PRESENCE Also it works with NAT clients with RTPProxy. When I register with Jitsi or QuteCom I get 5(10515) ERROR: presence [publish.c:388]: No E-Tag and no body found 6(10517) INFO: presence [notify.c:1601]: NOTIFY sip:1001@sip.abc.com via sip:1001@124.43.201.156:5060 on behalf of sip:1000@sip.abc.com for event presence 6(10517) ERROR: presence [publish.c:388]: No E-Tag and no body found ERROR: presence [publish.c:388]: No E-Tag and no body found Please advice me how to fix this. Best Regards, Roy.

3 3

How to use Kamailio as load balancer
by Khoa Pham 07 May '13

07 May '13

Hi, As someone asked in http://lists.sip-router.org/pipermail/sr-users/2010-December/066792.html about TCP overload, Daniel answerd that "if you have some lengthily operations that have to be applied to your sip traffic and you get congestion because of that, the solution is to use a kamailio as load balancer in front of a farm of proxy doing the actual processing" How to use Kamailio as load balancer ? -- Khoa Pham HCMC University of Science Faculty of Information Technology

3 2

xcap-diff support in xcap_server module?
by Juha Heinanen 04 May '13

04 May '13

does xcap_sever module support xcap-diff, i.e., is it able to publish xcap-event when some modification of a document happens? -- juha

2 2

Handling of CANCEL in case of no matching INVITE
by Vassilis Radis 03 May '13

03 May '13

Hello all, In the documentation of the t_relay_cancel() (TM module) there is an example that reads: if (method == CANCEL) { if (!t_relay_cancel()) { # implicit drop if relaying was successful, # nothing to do # corresponding INVITE transaction found but error occurred sl_reply("500", "Internal Server Error"); drop; } # bad luck, corresponding INVITE transaction is missing, # do the same as for INVITEs } What bothers me is the phrase #do the same as or INVITEs, because in RFC( http://tools.ietf.org/html/rfc3261#section-16.10 ) it says: If a response context is not found, the element does not have any knowledge of the request to apply the CANCEL to. It MUST *statelessly* forward the CANCEL request (it may have statelessly forwarded the associated request previously). So aren't we supposed to immediately statelessly forward the CANCEL if t_relay_cancel() did not find the INVITE transaction, instead of doing the same as INVITEs, which could be t_relay() (not stateless)?. Like below: if (method == CANCEL) { if (!t_relay_cancel()) { # implicit drop if relaying was successful, # nothing to do # corresponding INVITE transaction found but error occurred sl_reply("500", "Internal Server Error"); drop; } # bad luck, corresponding INVITE transaction is missing, forward(); } Am I correct or am i missing something? Thank you in advance, Bill

4 7

service uri not found in rls-services document
by Juha Heinanen 03 May '13

03 May '13

my presence server tries to handle subscribe to sip:jh+presence@test.tutpro.com. that service uri is stored in rls-services document of user sip:jh@test.tutpro.com, but for some reason k does not find the service uri. here is some debug: May 1 19:08:43 siika /usr/sbin/pres-serv[27355]: INFO: Handling SUBSCRIBE <sip:jh+presence@test.tutpro.com> May 1 19:08:43 siika /usr/sbin/pres-serv[27355]: INFO: rls [subscribe.c:155]: searching document for user sip:jh@test.tutpro.com May 1 19:08:43 siika /usr/sbin/pres-serv[27355]: INFO: rls [subscribe.c:240]: rls_services document:#012<?xml version='1.0' encoding='UTF-8'?>#012<rls-services xmlns="urn:ietf:params:xml:ns:rls-services"><service uri="sip%3Ajh%2Bpresence%40test.tutpro.com"><resource-list>http%3A//192.98.102.10%3A8080/xcap-root/resource-lists/users/sip%3Ajh%40test.tutpro.com/index/%7E%7E/resource-lists/list%5B%40name%3D%22sipsimple_presence_rls%22%5D</resource-list><packages><package>presence</package></packages></service><service uri="sip%3Ajh%2Bdialog%40test.tutpro.com"><resource-list>http%3A//192.98.102.10%3A8080/xcap-root/resource-lists/users/sip%3Ajh%40test.tutpro.com/index/%7E%7E/resource-lists/list%5B%40name%3D%22sipsimple_dialog_rls%22%5D</resource-list><packages><package>dialog</package></packages></service></rls-services> May 1 19:08:43 siika /usr/sbin/pres-serv[27355]: INFO: rls [subscribe.c:253]: service uri sip:jh+presence@test.tutpro.com not found in rl document for user sip:jh@test.tutpro.com any idea why the service uri is not found even when it appears to be in the document: <service uri="sip%3Ajh%2Bpresence%40test.tutpro.com"><resource-list>http%3A//192.98.102.10%3A8080/xcap-root/resource-lists/users/sip%3Ajh%40test.tutpro.com/index/%7E%7E/resource-lists/list%5B%40name%3D%22sipsimple_presence_rls%22%5D</resource-list><packages><package>presence</package></packages></service> could the escaped chars explain the failure or the fact that resource-list is an uri? -- juha

2 8

extra_id_pv parameter
by Bruno Bresciani 03 May '13

03 May '13

Hi All, I need forking a call to multiple destinations in paralel on different network segments, requiring different rtpproxy parameters. Reading the rtpproxy module documentation I discovered that is possible by setting the "b" parameter on rtpproxy_manage(), rtpproxy_offer(), rtpproxy_answer() and rtpproxy_destroy() functions, but I'm with some doubts... I can't understanding how set the extra_id_pv parameter correctly to use when the "b" parameter is used. The "b" parameter is supported on rtpproxy version 1.2.1? Best Regards

2 3

How to decrypt SIP message
by Khoa Pham 03 May '13

03 May '13

Hi, Currently, I'm using TLS and it works fine. But eventually, TLS is just used to agree upon a secret key. And both client and Kamailio use that secret key to encrypt message. In my solution, I have a fixed secret key. And client encrypt SIP message with that secret key. What code should I insert in Kamalio to decrypt it? -- Khoa Pham HCMC University of Science Faculty of Information Technology

2 4

Re: [SR-Users] service uri not found in rls-services document
by Peter Dunkley 02 May '13

02 May '13

Last I looked (around February) there were some things in Kamailio presence that aren't in OpenSIPS. For example, the separate notifier processes in both presence and rls (I believe one may have been ported across now by Saul - but not both). There are also extensions relating to the built-in XCAP server (particularly in presence_xml and rls) that are in Kamailio and not OpenSIPS. I know that there are some specific RLS fixes and features in Kamailio that only work when you use the built-in XCAP server. I also made some enhancements to the way pidf-manipulation is handled in Kamailio that I do not think have been ported to OpenSIPS. Basically, while OpenSIPS presence may be more advanced in some ways I am quite sure it is less developed in some others. Having spent a lot of time over the course of around 18 months doing work in Kamailio presence I would strongly resist any attempts to junk what is currently there because I know that I need a lot of the stuff that is in the Kamailio implementation. I don't currently have the time to develop it further at the moment either. I think the "junk them and replace with opensips" approach would be a disaster. Regards, Peter > looks like kamailio presence and especially rls is lagging behind in > development as compared to opensips. it properly handles the escapes > that i had trouble with, support external references in presence > rules, xcal-diff, etc. > > what should we do about kamailio presence modules? junk them and > replace with opensips ones or implement the missing stuff ourselves? > i personally don't have resources for the latter. > > -- juha > > _______________________________________________ > 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 > -- Peter Dunkley Technical Director Crocodile RCS Ltd

1 0

Re: [SR-Users] Rtpproxy - IPv4/v6 Translation and earlier removing of session
by Jakub Hrabovský 02 May '13

02 May '13

Thanks for reply, I checked log messages at callee (IPv6 client - Linphone) and result was little surprising for me. This IPv6 client as Callee didnt receive INVITE messages (I thought that yes because of Wireshark at SIP server). So it is my bad somewhere in my local network. But it is little interesting, that in other direction (from IPv6 caller to IPv4 callee) all Reply messages are received at IPv6 client and RTP/UDP packets go in both directions. I checked Iptables and Firewall at all nodes, but nothing (they all are in the same network) and I dont know what filteres messages to IPv6 callee client. btw. the proxy server in the middle is SIP Protocol gateway, which translates messages between IPv4 and IPv6 and packets at this node are OK. thanks for help Jakub Hrabovsky

1 0

Topoh module also masking Via?
by Grant Bagdasarian 01 May '13

01 May '13

Hello, The description of this module says it hides the SIP routing headers, does this also include the Via headers? Grant

3 5

rtpproxy virtual or physical machine
by Grant Bagdasarian 01 May '13

01 May '13

Hello, I understand that machines processing RTP are recommended to be physical machines. Does the same apply for machines just proxy'ing RTP? Also, after how many RTP session would you start noticing issues with RTP when using rtpproxy on a virtual VMware machine? Regard, Grant

2 2

gruu and dead-lock in registrar module
by Andreas Granig 01 May '13

01 May '13

Hi, We're hitting an issue in a deployment where all udp receivers are sitting in FUTEX_WAIT caused by save() -> lock_udomain() and seem to have deadlocked themselves every couple of days. Looking at the code, enable_gruu in registrar is active by default, and in lookup there is a code path /* temp-gruu lookup */ res = ul.get_urecord_by_ruid(_d, ahash, &inst, &r, &ptr); but no lock_udomain is obtained. However, when the execution falls through to the "done:" marker, it does ul.unlock_udomain(_d, &aor); without having called ul.lock_udomain first. 1.) Could someone please review this part? Looks a bit suspicious, although I don't know what implicitly happens in this case. If it were a semaphore and you decrease it to -1 by decrementing it without prior increment, it's essentially causing a dead-lock, but the current locking implementation might work completely different. 2.) Since I have no clue how gruu is supposed to work in detail, and since in our config we don't explicitly handle gruu (no lookup in loose-route, but gruu is enabled by default in registrar and we don't explicitly turned it off), I'm not even sure if we ever hit this code path. I only see that the ruid column in the location table is filled, but in order to get to this part, the ";gr" flag needs to be set in the R-URI for a lookup(), which I don't know whether that happened somehow in some call flows (we only log $ru, which I don't think logs these parameters, right?). Some input is highly appreciated! Andreas

2 4

Re: [SR-Users] Kamailio 3.x and Asterisk Realtime Integration plus dispatcher
by Aldo Antignano 01 May '13

01 May '13

> Sorry, yeah. you should probably ignore that part :-) The same dispatcher >deals with client connections from the "public" network as well as inbound Ok thanks, it's clear now. :) >> In the REGFWD route you send all the registrations to ALL the Asterisks? >> If so, why? > By sending the registration to all the available Asterisk servers then they > all know where each SIP client can be located. If a call comes in from the > PSTN or from a SIP client Kamailio can just be sent it to any of the * > servers and it will find the registered client. This will work even with > multiple load balancers/dispatchers using DNS SRV. In this way should be an option to keep registration in Kamailio only, let Asterisk do its specific job, and configure Asterisk to accept all calls coming from Kamailio as authorized calls by its IP? > In this scenario Kamailio is mainly acting as a load balancer, topology > hider and security/anti-flood front-end to create a scalable and redundant > Asterisk PBX system. Really depends what you are after. These are exactly my goals and was trying to understand which is the best way to achieve this. Any suggestion is welcome, thanks. Anty ------------------------------ Message: 3 Date: Tue, 30 Apr 2013 23:02:20 +0100 From: Barry Flanagan <barry(a)flanagan.ie> To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org> Subject: Re: [SR-Users] Kamailio 3.x and Asterisk Realtime Integration plus Message-ID: <CALJb54_1FGmGV4HfmR1zHgcAGqMsSPu6kriTv72W3adyHF3dQw(a)mail.gmail.com> Content-Type: text/plain; charset="utf-8" > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >

2 1

info: monthly mailing list reminder disabled
by Daniel-Constantin Mierla 01 May '13

01 May '13

Hello, I disabled the monthly mailing list membership reminder, being rather annoying when subscribed to lots of mailing lists. The option can be enabled on per subscriber basis, if anyone wants it, then it can be set in personal options at: - http://lists.sip-router.org/cgi-bin/mailman/options/sr-users/ Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, San Francisco, USA - June 24-27, 2013 * http://asipto.com/u/katu *

1 0

dlg_end_dlg
by Dragos Oancea 01 May '13

01 May '13

Hi everyone, I believe I've run into a problem when using dlg_end_dlg (with mi_datagram ) called from an external perl script with perl_exec() . Kamailio just locks after a while (few seconds) after disconnecting a call. Some parts of my debug log can be found here: http://pastebin.com/ANaTEifC The problem was observed while performing a stress test. Is it a good idea to call dlg_end_dlg() like that or I can kill an established dialog from the routing script in some other way ? Regards, Dragos

2 5

Re: [SR-Users] Kamailio 3.x and Asterisk Realtime Integration plus
by Aldo Antignano 30 Apr '13

30 Apr '13

Thank you very much for your reply Barry. Can you explain us yours ip 10.5.75. and 10.5.76. ? Seems your asterisks have two interfaces, one private and one public? In my infrastructure I have: SIP client -> Kamailio ---> Asterisk 01 ---> Asterisk 02 ... ---> Asterisk N Only Kamailio has a public IP address. In the REGFWD route you send all the registrations to ALL the Asterisks? If so, why? I'm trying to achieve something like: SIP clientA ---> REG ---> Kamailio Is Asterisk01 alive? Yes --> REG ---> Asterisk 01 NO --> Choose next one ---> Asterisk 02 ... ---> Asterisk N ---> Asterisk 01 SIP clientB ---> REG ---> Kamailio Is Asterisl02 alive? Yes --> REG ---> Asterisk 02 NO --> Choose next one ---> Asterisk 03 ... ---> Asterisk N and so on, with a round robin algorithm. When Sip ClientA want to call Sip ClientB, it asks Kamailio, that know that SIP Client B is registered on Asterisk 02 and so send the invite to it. I'm going to work over this in the next few days and let you know the result, if positive I'll write a new complete guide on this topic with your valuable suggestions. Any suggestion is welcome BR, Anty ------------------------------------------------------------------------------------------------------ >> I have read and applied the excellent guide found on: >> http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb >> >> Now I have added to Kamailio the HA/Load Balancer support, with the >> "dispatcher" module. >> This way I have 1 Kamailio and 2 Asterisk machines. >> >> How can I change the routing logic of the sections route[REGFWD] | >> route[FROMASTERISK] route[TOASTERISK] to use the dispatcher module? (in the >> guide above the asterisk binded ip address is cabled in the kamailio config >> code) I have done that. Relevant route entries below. # Forward REGISTER to Asterisk route[REGFWD] { if(!is_method("REGISTER")) { return; } # for each Asterisk server in setid=2 (External) we send a registration on behalf of the user. sql_xquery("ca", "SELECT SUBSTRING_INDEX(destination,':',-1) AS port, SUBSTRING_INDEX(SUBSTRING(destination,5),':',1) AS address FROM dispatcher WHERE setid = 2", "ra"); $var(i) = 0; while($xavp(ra[$var(i)]) != $null) { $var(rip) = $xavp(ra[$var(i)]=>address); $uac_req(method)="REGISTER"; $uac_req(ruri)="sip:" + $var(rip) + ":" + $xavp(ra[$var(i)]=>port); $uac_req(furi)="sip:" + $au + "@" + $var(rip); $uac_req(turi)="sip:" + $au + "@" + $var(rip); $uac_req(hdrs)="Contact: <sip:" + $au + "@" + $sel(cfg_get.kamailio.bindip) + ":" + $sel(cfg_get.kamailio.bindport) + ">\r\n"; if($sel(contact.expires) != $null) $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $sel(contact.expires) + "\r\n"; else $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $hdr(Expires) + "\r\n"; uac_req_send(); $var(i) = $var(i) + 1; } sql_result_free("ra"); } # Test if coming from Asterisk. We check the dispatcher "ds_is_from_list()" function to see if this is one of our Asterisk IPs route[FROMASTERISK] { if(ds_is_from_list()) { return 1; } else { return -1; } } # Send to Asterisk route[TOASTERISK] { # If call comes in to the .75 iface, we need to send it to the .75 iface of Asterisk as well. # otherwise we send to the .76 iface. We do this by calling different dispatcher sets. This is # because Asterisk needs to use NAT on the .76. (public) interface but not on the .75. if($td=~"10.5.75.") { $var(setid) = 4; xlog("SCRIPT: Call to 10.5.75. ip - using set $var(setid) \n"); } else { $var(setid) = 2; xlog("SCRIPT: Call from $fn to 10.5.76. ip - using set $var(setid) \n"); } # round robin dispatching on set determined above if(!ds_select_dst($var(setid), "4")) { send_reply("404", "No destination"); exit; } t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } Hope this helps. -Barry Flanagan

2 1

Kamailio 3.x and Asterisk Realtime Integration plus Dispatcher module
by Aldo Antignano 30 Apr '13

30 Apr '13

I have read and applied the excellent guide found on: http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb Now I have added to Kamailio the HA/Load Balancer support, with the "dispatcher" module. This way I have 1 Kamailio and 2 Asterisk machines. How can I change the routing logic of the sections route[REGFWD] | route[FROMASTERISK] route[TOASTERISK] to use the dispatcher module? (in the guide above the asterisk binded ip address is cabled in the kamailio config code) Thanks, Anty

2 1

[SR-users] Radius AAA
by Victor V. Kustov 30 Apr '13

30 Apr '13

Hi again! Need tips to some tricks with radius. 1. when receive sip message with REGISTER method - initiate auth with radius. (Authentification) - radius_www_authorize in auth_radius.so 2. when receive INVITE message - ask radius 'bout permissions and parse answer: get permission and next hop IP. (Authorization) ???? misc_radius ???? 3. when call a) before call send Acct-Status-Type = Start with attributes. b) after call send Acct-Status-Type = Stop with attributes. (Accounting) - acc_radius.so 1,3 - clear, I understand how it works. 2 - ? Kamailio must send request to radius server, get answer and route call to server from "next-hop-IP" field. Can I make it? -- С уважением, Виктор В. Кустов ООО "Брянск Связь-ТВ" +7 (952) 966 76 24 JID: coyote(a)bks.tv JID: coyote(a)bryansktel.ru I use FREE operation system: 3.8.4-calculate GNU/Linux

2 2

siptrace + flatstore
by Victor V. Kustov 30 Apr '13

30 Apr '13

Hi all loadmodule "siptrace.so" modparam("siptrace", "db_url", "flatstore:/var/log/siptrace") call sip_trace(); nothing in /var/log/siptrace, no errors in logs. whats wrong? -- SY, Victor JID: coyote(a)bks.tv JID: coyote(a)bryansktel.ru I use FREE operation system: 3.8.4-calculate GNU/Linux

2 2

var vs avp...
by Victor V. Kustov 30 Apr '13

30 Apr '13

Hi! whats difference $var() and $avp()? -- SY, Victor JID: coyote(a)bks.tv JID: coyote(a)bryansktel.ru I use FREE operation system: 3.8.4-calculate GNU/Linux

7 24

riud access from route block
by Juha Heinanen 30 Apr '13

30 Apr '13

when i call t_relay() on a tcp contact, branch route is executed. in that branch route i call t_on_branch_failure. now if the tcp connection to the contact is broken, for example, due to uas crash, the branch failure route is never executed and thus i cannot get access to $T_reply_ruid. there is $branch(name) pv that gives access to other branched, but not the "main branch". is there any way to get access to ruid of the main branch when t_relay() fails when called from route block? for example, if there is only one contact (the $ru one), which results in t_relay() failure, is there a way to find out the ruid of this contact? -- juha

1 1

kamailio 3.1.4
by Ryan d'Eon 30 Apr '13

30 Apr '13

Hello, I'm trying to support a kamailio install, and they are having a crashing issue. I'm not looking for detailed help on the problem, I just noticed one thing I'd like to check up on here: kamailio 3.1.4 (the version they are running) isn't available for download anymore, though quite a few other old versions are. Are there known bugs? Meaning, should I be worried about this, and go back with the recommendation to upgrade? If so, which is the ideal upgrade path? 3.1.5? 3.1.6? Thanks for any help. Ryan

2 1

!!! about private emails
by Daniel-Constantin Mierla 29 Apr '13

29 Apr '13

Hello, I would like to remind that, at least in my case, writing emails directly without cc-ing the mailing list is not recommended. The rule is also suggested in the mailing lists presentation page: - http://www.kamailio.org/w/mailing-lists/ Unless I asked explicitly for a private email with some specific details, the message will not be noticed and therefore not replied. I use this email address for many public mailing lists and it is intended only for that usage, I am reacting based on the filters I created, unmatched messages getting the lowest priority. There are several common sense as well as technical reasons. Here are some along with other suggestions to improve the likeliness of getting an answer: - we are not answering questions on mailing lists to help only one person, but also other people that may have similar issues in the future -- they can find the answers in the archive with a search engine - the amount of unexpected messages is very high -- it is practically a zero chance to get to the folder with messages that didn't match any filter (thus your message is not going to be replied) - whenever I find time, the first emails I answer are those coming on mailing lists. There are many there as well, people are traveling or having other personal or business projects, so sometime is good to send a reminder if a question does not get an answer - starting with negative approach or no technical content, like you are going to use something else or this application is missing what so ever tutorial giving the solution you need or simply asking for full config of complex requirements, is not helping, making people ignore your messages. Start with what you tried and where you got stuck. Everything you get is for free, nobody here owe you anything. Therefore, if anyone sent me a private message asking about kamailio or continuing a discussion started on mailing lists, then the messages have to be resent on mailing list if one wants an answer. A quick look in my unfiltered folder revealed hundreds of messages during the past year, soon many will be deleted anyhow. Note that you can always mask sensitive data (e.g., replace passwords, IP addresses) before sending logs or config file. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, San Francisco, USA - June 24016, 2013 * http://asipto.com/u/katu *

2 1

Rewrite RURI
by Grant Bagdasarian 29 Apr '13

29 Apr '13

Hello, Is it possible to have Kamailio rewrite the Request URI of the INVITE message but sent the INVITE to another address first? For example - Kamailio (10.0.0.1) receives an INVITE - Kamailio rewrites the RURI to 10.0.0.3 - Kamailio sends the INVITE to 10.0.0.2 - The application at 10.0.0.2 sends the INVITE to 10.0.0.3 Regards, Grant

5 5

Re: [SR-Users] Fwd:
by Charles Chance 29 Apr '13

29 Apr '13

Hi, Sincere apologies, not sure how this happened - please ignore, it was not me who sent it :/ Regards, Charles On 2 April 2013 08:21, Charles Chance <charles.chance(a)sipcentric.com> wrote: > http://www.toyosoken.co.jp/ivtduw.php > > > _______________________________________________ > 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 > -- *Charles Chance* Managing Director t. 0121 285 4400 m. 07932 063 891 -- www.sipcentric.com Follow us on twitter @sipcentric <http://twitter.com/sipcentric> Sipcentric Ltd. Company registered in England & Wales no. 7365592. Registered office: Unit 10 iBIC, Birmingham Science Park, Holt Court South, Birmingham B7 4EJ.

1 0

Fwd:
by Charles Chance 29 Apr '13

29 Apr '13

http://www.toyosoken.co.jp/ivtduw.php

1 0

siptrace source ip causes martians
by Thilo Bangert 29 Apr '13

29 Apr '13

Hi, i have a kamailio instance, which is connected to two subnets (A and B) and exchanges sip messages between them. I have setup the siptrace module to send hep tracing messages of all sip messages to a host in subnet B. This works really well for sip messages arriving on the IP of subnet B - for those arriving on subnet A however it doesnt work. The hep messages have the interface address of subnet A set as source address. Since the machine receiving the HEP messages also is connected to both subnet A and B, these messages are (correctly) rejected as martians and never reach the sipcapture module on that host. this is on kamailio 3.3.4. Is there a way to force the source IP of outgoing HEP messages in the siptrace module? thanks kind regards Thilo

2 4

kamaillio as SBC
by Vu Ngo thanh 29 Apr '13

29 Apr '13

I am interested in testing Kamaillio acting as an SBC. The SBC as Outbound proxy server, the IPPhone can register to SIP Server via SBC and make a call via SBC. the topology is bellow The IP Phone A, The IP Phone B --------- register ------- FreeSwitch (SBC-Outbound Proxy) -------- SIP Server Could you please help me how to install and configure FreeSwitch as SBC-Outbound Proxy ? Thanks so much

3 2

Montoring/Reporting Solution - Commercial/Open Source
by Ali Jawad 26 Apr '13

26 Apr '13

Hi I am in need of a solution that does provide me with advanced graphing and report for Kamailio, I.e. ASR/ACD over day/month/destination etc. with tresholds and alerts. Does anyone know about a solution that is compatible with Kamailio and does this ? Thanks

4 5

Issue with app_perl - return value, logging and append_branch
by Mark Boyce 26 Apr '13

26 Apr '13

Hi All I may well be me missing something obvious but I have got three odd ones whilst trying to test app_perl module which I'd appreciate some assistance with. Setup -------- Kamailio version : 4.0.0 (x86_64/linux) from Peter Dunkley's YUM Repository Perl version : v5.10.1 OS version : CentOS release 6.4 Kamailio.cfg highlights; loadmodule "app_perl.so" modparam("app_perl", "filename", "/tmp/test.pl") modparam("app_perl", "modpath", "/usr/lib64/kamailio/perl/") # ** would someone like to add the app_ to 5.1 and 5.2 of the manuals please request_route { . . route(PERL); . . } route[PERL] { if ($rU=~"^124" && method=="INVITE") { xlog("L_INFO","perl: match $rm $ru from:$fu ($si:$sp) to:$tu rpid:$re\n"); if (perl_exec("testrouting")) { xlog("L_INFO","perl: true exit\n"); } else { xlog("L_INFO","perl: false exit\n"); } } } /tmp/test.pl has the following in it ... #!/usr/bin/perl use Kamailio qw ( log ); use Kamailio::Constants; use Sys::Syslog qw( :DEFAULT setlogsock); sub testrouting { setlogsock('unix'); openlog($0,'','user'); syslog('info', "PERL Kamailio - Log test"); closelog; return 1; } There is no valid route for 124@ and the above works exactly as I'd expect; -> Phone sends INVITE to 124 <- Proxy responds with AUTH Required -> Phone sends ACK -> Phone sends INVITE to 124 with AUTH -- SysLog gets a "PERL Kamailio - Log test" from perl script -- Kamailio log gets "perl: true exit" <- Proxy responds with "404 Not Found" -> Phone sends ACK ------------------------------------------ First problem is that if I swap the "return 1" in test.pl for "return 0" what I see is; -> Phone sends INVITE to 124 <- Proxy responds with AUTH Required -> Phone sends ACK -> Phone sends INVITE to 124 with AUTH -- SysLog gets a "PERL Kamailio - Log test" from perl script -> Phone resends INVITE to 124 with AUTH -- SysLog gets a "PERL Kamailio - Log test" from perl script -> Phone resends INVITE to 124 with AUTH -- SysLog gets a "PERL Kamailio - Log test" from perl script -- I Hang Up Phone --> Phone send CANCEL --> Phone send CANCEL --> Phone send CANCEL In this second test once perl_exec is called all routing seems to stop. There's nothing from the xlog to say if perl_exec returned true or false. -------------------------------------- The second problem I have is when I add "Kamailio::log(L_INFO," line it never arrives in any of the logs use Kamailio qw ( log ); use Kamailio::Constants; use Sys::Syslog qw( :DEFAULT setlogsock); sub testrouting { setlogsock('unix'); openlog($0,'','user'); syslog('info', "PERL Kamailio - Log test"); closelog; Kamailio::log(L_INFO, "PERL : testrouting\n"); return 1; } -------------------------------------- The third one is that I cannot get append_branch to do anything. If I add to the perl script the lines; my $m = shift; $m->rewrite_ruri("sip:anotheruser\@ourserver"); The call gets diverted as expected. However if I use append_branch; $m->append_branch("sip:anotheruser\@ourserver"); It has no effect at all. -------------------------------------- That's as far as I have experimented with the module so I can't say if everything else is working as expected :-) Anyone have any thoughts? Cheers Mark

2 1

Problem with PUA
by andpe 26 Apr '13

26 Apr '13

Hi I am new to Kamailio. I have a configuration with existing phones. I call between them. However, I have a problem with the configuration of PRESENCE. I guess I'm doing something wrong. Can someone send me a sample setup working. I looked for a long time information, but I could not fix my problem. In the logs I get the information: 6 (19259) ERROR: pua [send_publish.c: 578]: New PUBLISH and no body found-invalid request 6 (19259) ERROR: pua_usrloc [ul_publish.c: 326]: while sending publish [cycle # 12, if constant make sure server lives] Version: Server :: kamailio (4.0.0 (i386/linux)) Build :: mi_core.c compiled on Apr 4 2013 11:20:52 with gcc 4.4.5 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, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES GIT :: 1d009c regards Andy

2 1

RPM packages for v4.0.1
by Daniel-Constantin Mierla 26 Apr '13

26 Apr '13

Hello, rpm packages for v4.0.1 are no available on opensuse build service - more details and download links at: - http://www.kamailio.org/wiki/packages/rpms Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

misc_radius - too many extras
by Victor V. Kustov 26 Apr '13

26 Apr '13

Hi all I've got same error: misc_radius [extra.c:80]: too many extras -> please increase the internal buffer How I get increase buffer in config? -- SY, Victor JID: coyote(a)bks.tv JID: coyote(a)bryansktel.ru I use FREE operation system: 3.8.4-calculate GNU/Linux

2 1

RLS Module - Getting NOTIFY for offline users
by Attila Megyeri 25 Apr '13

25 Apr '13

Dear Users, We use Kamailio as our SIP proxy and Presence server, leveraging XCAP and RLS functionality. After issuing SUBSCRIBE rls@<domain> I would like to get presence information for the contacts in the RLS list not only when they are online, but also if they are offline. Is this possible? Currently we are getting NOTIFY only for those contacts that are online (presence open) Thank you in advance, Attila

2 2

Kamailio v4.0.1 Released
by Daniel-Constantin Mierla 25 Apr '13

25 Apr '13

Hello, Kamailio SIP Server v4.0.1 stable release is out. This is a maintenance release of the latest stable branch, 4.0, that includes fixes since release of v4.0.0. There is no change to database schema or configuration language structure that you have to do on installations of v4.0.0. Deployments running previous v4.x.x versions are strongly recommended to be upgraded to v4.0.1. For more details about version 4.0.1 (including links and hints to download the tarball or from GIT repository), visit: * http://www.kamailio.org/w/2013/04/kamailio-v4-0-1-released/ RPM, Debian/Ubuntu packages will be available soon as well. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

Re: [SR-Users] Rtpproxy - IPv4/v6 Translation and earlier removing of session
by Jakub Hrabovský 25 Apr '13

25 Apr '13

Hi, 1.) Here INVITE message sent to IPv6 SIP Client, but he doesnt respond: INVITE sip:jano@[2001:4118:300:122:bca7:31ac:ccab:d6f6];line=e7b07b8cb7a33ee SIP/2.0 Record-Route: <sip:[2001:4118:300:122:444A:3FFF:FECC:61A0];lr=on> Record-Route: <sip:[2001:4118:300:122:A00:27FF:FE60:83C];r2=on;lr=on;nat=v46> Record-Route: <sip:158.193.139.207;r2=on;lr=on;nat=v46> Record-Route: <sip:158.193.139.46;lr=on;nat=yes> Via: SIP/2.0/UDP [2001:4118:300:122:444A:3FFF:FECC:61A0];branch=z9hG4bK1c7c.bf9123f4.0 Via: SIP/2.0/UDP [2001:4118:300:122:A00:27FF:FE60:83C];branch=z9hG4bK1c7c.76b1542.0 Via: SIP/2.0/UDP 158.193.139.46;rport=5060;branch=z9hG4bK1c7c.af9123f4.0 Via: SIP/2.0/UDP 10.0.2.15:5060;received=158.193.139.205;rport=55754;branch=z9hG4bK1417756364 From: <sip:jakub@ip02.sip.uniza.sk>;tag=632445346 To: <sip:jano@ip02.sip.uniza.sk> Call-ID: 267806174 CSeq: 21 INVITE Contact: <sip:jakub@158.193.139.205:55754> Content-Type: application/sdp Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Max-Forwards: 67 User-Agent: Linphone/3.3.2 (eXosip2/3.3.0) Subject: Phone call Content-Length: 470 P-hint: outbound v=0 o=jakub 123456 654321 IN IP6 2001:4118:300:122:a00:27ff:fe60:83c s=A conversation c=IN IP6 2001:4118:300:122:a00:27ff:fe60:83c t=0 0 m=audio 51652 RTP/AVP 112 111 110 3 0 8 101 a=rtpmap:112 speex/32000/1 a=fmtp:112 vbr=on a=rtpmap:111 speex/16000/1 a=fmtp:111 vbr=on a=rtpmap:110 speex/8000/1 a=fmtp:110 vbr=on a=rtpmap:3 GSM/8000/1 a=rtpmap:0 PCMU/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:101 telephone-event/8000/1 a=fmtp:101 0-11 a=nortpproxy:yes 2.) I get here cap files of SIPv64 server and Bridge as Enclosure. Here i get my hosts file for wireshark for clear view of SIP entities: 158.193.139.46 SIPserver_IP4 158.193.139.207 Bridge_IP4 2001:4118:300:122:444a:3fff:fecc:61a0 SIPserver_IP6 2001:4118:300:122:a00:27ff:fe60:83c Bridge_IP6 158.193.139.205 WinPC_IP4 2001:4118:300:122:bca7:31ac:ccab:d6f6 IP6_Client 10.0.2.15 IP4_Klient

2 1

Kamailio dialog termination
by Grant Bagdasarian 25 Apr '13

25 Apr '13

Hello, Using the dialog module. According to the documentation a dialog is automatically destroyed when a BYE is received. In case of no BYE, the dialog lifetime is controlled via the default timeout. I'm assuming the dialog module only creates a dialog when a 200 OK is received on the INVITE. So if a CANCEL request is sent for this INVITE or the server answers with anything but a 200 OK, no dialog is created and maintained by the dialog module? Also, when the dialog is terminated, is it guaranteed the dialog entry is removed from memory at all times, but there might be a chance the dialog remains stored in the database table because of some error? I'm using the dialog module to control the number of calls allowed to a certain destination, but it sometimes does occur that an entry in the database is not removed, but I'm not sure if the dialog is also removed from memory when this happens. I hope someone could clear this out for me. Thanks, Grant

3 5

★ SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List, Phillip left a message for you
by Badoo 25 Apr '13

25 Apr '13

You can instantly reply using our message exchange system. Check your message http://eu1.badoo.com/0138644786/in/y9a..eJmaMo/?lang_id=3&g=57&m=61&mid=517… Some other people in the area: If clicking the links in this message does not work, copy and paste them into the address bar of your browser. You have received this email from Badoo Trading Limited (postal address below). http://eu1.badoo.com/impersonation.phtml?lang_id=3&email=sr-users%40lists.s… Badoo Trading Limited is a limited company registered in England and Wales under CRN 7540255 with its registered office at Media Village, 131 - 151 Great Titchfield Street, London, W1W 5BB.

1 0

Planing to release v4.0.1
by Daniel-Constantin Mierla 25 Apr '13

25 Apr '13

Hello, I am considering releasing v4.0.1 next week on Thursday. There are some issues under discussions on mailing list, they will be sorted by that time, but if you are aware of others, please report them on dev mailing list or tracker. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 1

www.kamailio.org rebooted for maintenance
by Henning Westerholt 24 Apr '13

24 Apr '13

Hello, I've needed to reboot www.kamailio.org two times for maintenance (kernel and grub boot loader update) a few minutes ago. The machine which its services is now up again, please let me know if you find some issues because of the change. Thanks and regards, Henning -- Henning Westerholt - Head of IT Operations Internet Access & Communications Tel. +49 721 91374-4483, Fax. +49 721 91374-2734 1&1 Internet AG, Brauerstraße 48, 76135 Karlsruhe, Germany Vorstände: Henning Ahlert, Ralph Dommermuth, Robert Hoffmann, Andreas Hofmann, Markus Huhn, Hans-Henning Kettler, Dr. Oliver Mauss, Jan Oetjen, Martin Witt, Christian Würst. Aufsichtsratsvorsitzender: Michael Scheeren. Amtsgericht Montabaur / HRB 6484

1 0

BYE dialog timeout bad syntax
by Julia 24 Apr '13

24 Apr '13

Hello, The BYE, generated by Kamailio in case of dialog timeout, is malformed: BYE <sip:anonymous@192.168.225.2:50601> sip:anonymous@192.168.225.2:50601 SIP/2.0 Via: SIP/2.0/UDP 172.16.60.113;branch=z9hG4bK77b6.941684b6.0 Via: SIP/2.0/UDP 10.1.1.10;branch=z9hG4bKsr-JfymiMenCtp4urS5CXSnCHJnCXJZiHvRCHYJb6ggNPN9NVDL3 -TWhdW0UvOfN1KPTReRiHy1CHmcs6mZiHS* To: <sip:anonymous;cpc=ordinary@192.168.225.2;tag=01e3a8c0-c1c2-t-1> sip:anonymous;cpc=ordinary@192.168.225.2;tag=01e3a8c0-c1c2-t-1 From: <sip:1255010006@192.168.225.2:5060;user=phone;tag=813142547-1366730270587> sip:1255010006@192.168.225.2:5060;user=phone;tag=813142547-1366730270587 CSeq: 465332801 BYE Call-ID: <mailto:1aebb42c-01e3a8c0-000007c2@192.168.225.2> 1aebb42c-01e3a8c0-000007c2(a)192.168.225.2 Content-Length: 0 User-Agent: kamailio (3.3.2 (x86_64/linux)) Max-Forwards: 69 The bad syntax is located in the "To" header field. According to the RFC 3261 section 25.1, the grammar of the To header field is as follow: To = ( "To" / "t" ) HCOLON ( name-addr / addr-spec ) *( SEMI to-param ) name-addr = [ display-name ] LAQUOT addr-spec RAQUOT addr-spec = SIP-URI / SIPS-URI / absoluteURI to-param = tag-param / generic-param The RFC 3261 section 20.10 explains that: Even if the "display-name" is empty, the "name-addr" form MUST be used if the "addr-spec" contains a comma, semicolon, or question mark. There may or may not be LWS between the display-name and the "<". As the "addr-spec" <sip:anonymous;cpc=ordinary@192.168.225.2> sip:anonymous;cpc=ordinary@192.168.225.2 contains a semicolon, it means that the "name-addr" form must be used. The "To" header field should be encoded as follow: To: < <sip:anonymous;cpc=ordinary@192.168.225.2> sip:anonymous;cpc=ordinary@192.168.225.2>;tag=01e3a8c0-c1c2-t-1 BR, Julia

2 1

Fwd: save accdb records with milisec in timestamp field
by Uri Shacked 24 Apr '13

24 Apr '13

Hi, can anyone tell me if it is possible to save the time and date using accdb (mysql) with milisec as well? the field on the DB is set as TIMESTAMP. BR, Uri

3 5

Re: [SR-Users] Kamailio 4.0 IMS
by Jason Penton 23 Apr '13

23 Apr '13

Thanks Carsten. We have also started some other docs with Richard and Olle. Olle have you managed to publish the doc Rich and I shared with you - at least this is a nice overview of the modules, etc? I know there are still some module docs outstanding from me which I will get to ASAP - apologies for that! Cheers Jason On Tue, Apr 23, 2013 at 11:40 AM, Pavel Segeč <Pavel.Segec(a)fri.uniza.sk>wrote: > Thank you Carsten very much**** > > ** ** > > Palo73**** > > ** ** > > *From:* sr-users-bounces(a)lists.sip-router.org [mailto: > sr-users-bounces(a)lists.sip-router.org] *On Behalf Of *Carsten Bock > *Sent:* Tuesday, April 23, 2013 10:08 AM > *To:* Kamailio (SER) - Users Mailing List > *Cc:* andreea.ancuta.corici(a)fokus.fraunhofer.de; Dragos Vingarzan > *Subject:* Re: [SR-Users] Kamailio 4.0 IMS**** > > ** ** > > Hi,**** > > i was approached by several people after Kamailio World 2013, that our > installation guide is not available anymore.**** > > The "old" version was at some point quite outdated, that is the reason why > i disabled the page. However, due to the great demand i will update the IMS > on Kamailio how-to later this week and publish the updated version (and i > will move it to the Kamailio Wiki so other points can be easily added by > different people).**** > > IMS rocks!**** > > ** ** > > Kind regards, > Carsten**** > > ** ** > > ** ** > > 2013/4/23 Pavel Segeč <pavel.segec(a)fri.uniza.sk>**** > > Sir, > > Thank you for your answer. Aggree all you wrote and I'm aware of that. > Just I'm saying helps us..we use, popularize and love Kamailio project > however we need some help ...that's all... Kamilio 4.0 IMS is a quite new > thing and we are starting to use it during official student classes and > labs, we are gaining experiences and we share them,we reports bugs and so > on, we do not hide our knowledge > > palo73**** > > > > > > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users**** > > > > > -- **** > > Carsten Bock > CEO (Geschäftsführer) > > ng-voice GmbH > Schomburgstr. 80 > D-22767 Hamburg / Germany > > http://www.ng-voice.com > mailto:carsten@ng-voice.com > > Office +49 40 34927219 > Fax +49 40 34927220 > > Sitz der Gesellschaft: Hamburg > Registergericht: Amtsgericht Hamburg, HRB 120189 > Geschäftsführer: Carsten Bock > Ust-ID: DE279344284 > > Hier finden Sie unsere handelsrechtlichen Pflichtangaben: > http://www.ng-voice.com/imprint/**** > > ** ** > > __________ Informacia od ESET NOD32 Antivirus, verzia databazy 8255 > (20130422) __________**** > > ** ** > > Tuto spravu preveril ESET NOD32 Antivirus.**** > > ** ** > > http://www.eset.sk**** > > > __________ Informacia od ESET NOD32 Antivirus, verzia databazy 8255 > (20130422) __________ > > Tuto spravu preveril ESET NOD32 Antivirus. > > http://www.eset.sk > > _______________________________________________ > 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 > > -- Jason Penton Senior Manager: Applications and Services Smile Communications Johannesburg, South Africa Phone: +27 83 283 7000 This email is subject to the disclaimer of Smile Communications at http://www.smilecoms.com/disclaimer

1 1

Simulating kamailio config flows
by Andreas Granig 23 Apr '13

23 Apr '13

Hi, I'd like to put a topic up for discussion to test kamailio config logics. The standard way of doing so is to start kamailio as usual and write sipp or sipsak scenarios to perform automated tests. This can get quite complex pretty quickly, as you always have to take care of covering the full transaction on the client side, where in most of the cases you really just want to know which IP/port kamailio would send the message to, and how some headers look like after processing. Also, it doesn't really allow to inspect your variables and avps etc. on the fly other than logging them to syslog and parsing it from there. Let me know if this is stupid and/or a complete overkill, but what about introducing some kind of dummy mode, where you'd pipe a message into kamailio via stdin, and get the resulting message out on stdout (e.g. in ngrep style with ip information as first line, plus the content following), plus a dump of internals (e.g. vars, avps) on stderr as they're being assigned, and once the message is processed, kamailio would just shut down again? There are quite some things to consider, like not doing any timer handling in tm, but would this be something anyone would be interesting to discuss in more detail? If I'm completely on a wrong track, what are suggested alternatives to testing full sip call flows other than running automated UAs against kamailio? Andreas

2 1

Re: [SR-Users] sip_trace unable to capture after next_branches()
by Ramaseshi reddy kolli 23 Apr '13

23 Apr '13

Hi, Once in a while it captures outbound SIP Invite but it captures always 100 trying, 180 and 200 OK and some times it captures inbound SIP INVITE as well. Thanks Ram

2 2

bflag mystery
by Juha Heinanen 23 Apr '13

23 Apr '13

i have in config: setbflag(8); xlog("L_INFO", "bflags are <$bf/$bF>\n"); save("location", "0x02"); and i correctly get to syslog: Apr 10 19:51:48 wheezy1 /usr/sbin/sip-proxy[6722]: INFO: bflags are <256/00000100> however, when i look what gets stores in location table, cflags column of the contact has value 384. it is also shown by ul_dump: Cflag:: 384 384 = 256 + 128. i must be missing something trivial. how is it possible that also flag 7 (which is my nat_bflag) gets set in usrloc? -- juha

2 3

Kamailio 3.3.1 and Asterisk 11.3 with realtime integration
by Henning Bragge 23 Apr '13

23 Apr '13

Hi, I'm totally new to Kamailio and I attempt to add Kamailio functionality to an already working Asterisk implementation. I have been following this link: http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb Everything seems to work with an exception of encountering NAT issues. The location table in Kamailio reports the local NAT address, instead of the public address, where then Asterisk reports UNREACHABLE status. I have installed and started RTPproxy as adviced in the above link - no changes. Best regards, Henning Bragge

2 1

Kamailio 4.0 IMS
by Pavel Segeč 23 Apr '13

23 Apr '13

Hi all Kamailio IMS community, after a time, we are starting to use Kamailio 4.0 based IMS platform during our university lessons (before we used Kamailio 3.2 based). However I'm missing some usefull links, docs, tutorials regarding of Kamailio 4.0 IMS procesess. The situation was quite poor alerady before for 3.2 but some was available on ng-voice web page, at least, now nothing (especially about new ng-voice HSS). Does anybody know and help us with some usefull information about the Kamailio IMS 4.0 and HSS? I promise to prepare and make publicly available guides on our site nil.uniza.sk (Kamailio award for 2010) Thanks again palo73

3 5

BUG? - Dialog module timeout BYE is not ok.
by Uri Shacked 23 Apr '13

23 Apr '13

Hi, I think there is a bug with the BYE that is sent to the caller and callee when dialog timeout happends. The BYE headers are sent with no "<" or ">". So, some sip singaling points decline the BYE. BR, Uri

2 3

FreeBSD build broken (with patch)
by Victor V. Kustov 22 Apr '13

22 Apr '13

Hi all! # gmake group_include="standard mysql debug radius" CC (gcc) [kamailio] proxy.o CC (gcc) [kamailio] dns_func.o In file included from dns_func.c:31: /usr/include/resolv.h:157: error: array type has incomplete element type /usr/include/resolv.h:171: error: field 'addr' has incomplete type /usr/include/resolv.h:195: error: field 'sin' has incomplete type gmake: *** [dns_func.o] Error 1 It's because incomplete headers for FreeBSD. diff --git a/dns_func.c b/dns_func.c index 6eedf0d..5a9bf7e 100644 --- a/dns_func.c +++ b/dns_func.c @@ -28,6 +28,7 @@ #include "dns_func.h" +#include <netinet/in.h> #include <resolv.h> #include <sys/types.h> #include <netdb.h> I write patch in message body 'cause think need patch to config scripts, not dns_func.c. #include <netinet/in.h> - need before #include <resolv.h> -- SY, Victor JID: coyote(a)bks.tv JID: coyote(a)bryansktel.ru I use FREE operation system: 3.8.4-calculate GNU/Linux

2 1

Load balancer using dispatcher / BYE relaying
by Ariel Wainer 22 Apr '13

22 Apr '13

StartFragmentHi, I'm new to Kamailio and I'm trying to setup a load balancer using Kamailio 4.0 and rtpproxy, to balance between 2 asterisk servers that provide an IVR service. I've followed several guides, and so far it seems to work along with rtpproxy, except it is not sending the BYE message back to asterisk, and the call never finishes from the asterisk POV. Let me describe the test scenario: I have the following: IP phone: 192.168.200.183 PBX: 192.168.200.3 Kamailio: 192.168.200.132 IVR1: 192.168.200.160 IVR2: 192.168.200.161 >From the IP phone which is registered on the PBX, I initiate a call to exten 800, PBX forwards it to 1234@192.168.200.132:5060 and Kamailio dispatches it to one of the two IVR in round-robin. I'm testing this way and not calling directly to Kamailio because in the production enviroment, calls will be coming from outside. Attached you will find the Kamailio configuration. Packet capture made at the Kamailio host, where you can see the BYE package from the PBX (first one on line 24) but no BYE package sent to the IVRs is here: http://cringer.3kh.net/web/misc/sip.cap Please let me know if I ommited any important detail of my setup and thanks in advance. Best regards. -- Ariel Wainer

2 3

[SER-Users] debug: again.
by Victor V. Kustov 22 Apr '13

22 Apr '13

Hi all! Still same problem: i don't see debug messages. OS: FreeBSD 9.1 stable, fresh git Kamailio version. In present moment just debugger.so with config trace work as expected. Option "debug=X" seems has no effect at all. I saw debug logs in maillist. Maybe I make something wrong? I never see simulary strings in logs: debug: DEBUG: <core> [forward.c:511]: orig. len=460, new_len=483, proto=1 debug: DEBUG: <core> [usr_avp.c:639]: DEBUG:destroy_avp_list: destroying list 0xb323c178 debug: DEBUG: <core> [usr_avp.c:639]: DEBUG:destroy_avp_list: destroying list (nil) debug: DEBUG: <core> [usr_avp.c:639]: DEBUG:destroy_avp_list: destroying list (nil) debug: DEBUG: <core> [usr_avp.c:639]: DEBUG:destroy_avp_list: destroying list (nil) debug: DEBUG: <core> [usr_avp.c:639]: DEBUG:destroy_avp_list: destroying list (nil) debug: DEBUG: <core> [usr_avp.c:639]: DEBUG:destroy_avp_list: destroying list (nil) debug: DEBUG: <core> [receive.c:280]: receive_msg: cleaning up -- SY, Victor JID: coyote(a)bks.tv JID: coyote(a)bryansktel.ru I use FREE operation system: 3.8.4-calculate GNU/Linux

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When to use t_newtran/t_release?
by Yufei Tao 22 Apr '13

22 Apr '13

Hi, I've been using the default cfg file as a starting point and added my stuff on top. But I don't think I understand the use of t_newtran/t_release and when they should be used. Read the documentation but still not clear. For example, for presence: t_newtran is called before handle_publish/subscribe and t_release called afterwards. And in route[LOCATION], t_newtran is called before sending 4xx responses. Could someone help explain please what they do and when they should be used please? Thank you! Yufei -- Yufei Tao Red Embedded This E-mail and any attachments hereto are strictly confidential and intended solely for the addressee. If you are not the intended addressee please notify the sender by return and delete the message. You must not disclose, forward or copy this E-mail or attachments to any third party without the prior consent of the sender. Red Embedded Design, Company Number 06688253 Registered in England: The Waterfront, Salts Mill Rd, Saltaire, BD17 7EZ

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maintenance of www.kamailio.org
by Henning Westerholt 22 Apr '13

22 Apr '13

Hi, I'll do today this noon a regular maintenance of www.kamailio.org. I don't expect any big outtages, the webserver could be maybe for a few seconds not available due to some restarts. I'll announce when the maintenance is finished. Best regards, Henning

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Possible memory leak in parse_pai.c
by Tais Plougmann Hansen 22 Apr '13

22 Apr '13

Hi, I'm seeing a possible leak in parsing the P-Asserted-Identity, and possibly P-Preferred-Identity headers in kamailio-3.3.4. I have a server running in which one thread has currently most of the 8 MB private memory allocated. At restart qm_sums reports parse_pai.c as sitting on most of the memory: Apr 16 16:38:58 gws1 /usr/sbin/kamailio[9398]: INFO: qm_status: heap size= 8388608 Apr 16 16:38:58 gws1 /usr/sbin/kamailio[9398]: INFO: qm_status: used= 5369240, used+overhead=6417216, free=1971392 [...] Apr 16 16:39:03 gws1 /usr/sbin/kamailio[9398]: INFO: qm_sums: count= 6198 size= 4721776 bytes from <core>: parse_pai.c: parse_pai_header(58) I am currently using the header through $ai for xlog statements, the accounting module and a validity check (if (!($(ai{uri.user}) =~ "^\+[1-9]+")) {...}). >From digging through the logs it could seem like this happens when using $ai in the accounting module. I'll try removing it from accounting (log_extra) and see if it stops leaking. -- Tais Plougmann Hansen OSD Consulting ApS osd.dk Tel: +45 78101078 CVR: DK31332737

3 4

Kamailio World Conference Presentations
by Daniel-Constantin Mierla 22 Apr '13

22 Apr '13

Hello, The other files with slides presented last week at Kamailio World Conference are made available on the web at: - http://www.kamailio.org/events/2013-KamailioWorld/ There are two presentations missing, hopefully they will be received soon and uploaded. Many thanks again to those coming to speak or attend the event. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

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Rtpproxy - IPv4/v6 Translation and earlier removing of session
by Jakub Hrabovský 22 Apr '13

22 Apr '13

Hi, I started working on IPv4/IPv6 Translation issue in Kamailio by using rtpproxy as Media relay last year in my Bachelor-project. I solved the trouble with Translation between IPv4 and IPv6 clients, where IPv4 client used public IPv4 address. Now I am working on the issue, where IPv4 client uses private IPv4 (is behind NAT in local network), so I have to provide IPv4/v6 and NAT for all messages in one call. My topology consists of: Kamailio 3.3.4 - SIPproxy (supports IPv4/IPv6, public IPv4 and IPv6 addresses) - makes NAT translation, works as SIPoutbound proxy and communicates with rtpproxy (v1.2.1), Kamailio 3.3.4 - SIPprotocol GW (== SIP bridge) - translates IPv4 messages to IPv6 (and IPv6 to IPv4), uses other rtpproxy, has public IPv4 and IPv6 addresses + only IPv4 and only IPv6 clients (Linphone v3.5.2) I assume, that both (IPv4 and IPv6) clients are in the same SIP domain. I solve IPv4-IPv6 translation so, that each message from IPv6 client goes to SIP proxy and (if needed) is forwarded to SIP Protocol GW. Translated medssage is then forwarded back to SIP proxy and from that to callee. Here i have some problems: -1.) IPv6 client doesnt respond to invites from IPv4 client. The messages are ignored at IPv6 client. -2.) Rtpproxy on the machine with SIP protocol gateway (which provides IPv4/v6 translation) removes session earlier than I need (usually after 30 seconds of call) and the result is call break. this issue doesnt appear for rtpproxy on SIP proxy. I dont know what causes this behavior, if it is only in some parameters, witch i have to set, or it is bug in rtpproxy or .... Here I get config files of topology entities: ----------------------------------------------------------------------------------------------------------------------------- config file for SIP proxy: ----------------------------------------------------------------------------------------------------------------------------- #!KAMAILIO # Konfiguracny subor pre SIP Proxy Server IPv4/IPv6 pre konkretnu domenu ip02.sip.uniza.sk # #!ifdef ACCDB_COMMENT ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; #!endif ####### Defined Values ######### #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USERLOCDB #!define WITH_ALIASDB #!define WITH_NAT #>> #!define WITH_IPV6 #!define WITH_V4V6 #!ifdef WITH_V4V6 #!define BRIDGE_IP4 "158.193.139.207" #!define BRIDGE_IP6 "[2001:4118:300:122:a00:27ff:fe60:83c]" #!endif #<< #!define ADDR_IP4 158.193.139.46 #!ifdef WITH_IPV6 #!define ADDR_IP6 [2001:4118:300:122:444a:3fff:fecc:61a0] #!endif # *** Value defines - IDs used later in config #!ifdef WITH_MYSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!define DBURL "mysql://openser:openserrw@localhost/openser" #!endif #!ifdef WITH_MULTIDOMAIN # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 #!else #!define MULTIDOMAIN 0 #!endif # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 #!define FLB_NATB 1 #!define FLB_NATSIPPING 2 #>> #!define FLB_IPV6 6 #!define FLB_V4V6 7 #<< ####### Global Parameters ######### #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL0 fork=yes children=4 /* uncomment the next line to disable TCP (default on) */ #disable_tcp=yes /* uncomment the next line to disable the auto discovery of local aliases based on reverse DNS on IPs (default on) */ #auto_aliases=no /* add local domain aliases */ alias="ip02.sip.uniza.sk" alias="[2001:4118:300:122:444a:3fff:fecc:61a0]" alias="158.193.139.46" #!ifdef WITH_IPV6 listen=udp:158.193.139.46:5060 listen=udp:[2001:4118:300:122:444a:3fff:fecc:61a0]:5060 listen=tcp:158.193.139.46:5060 listen=tcp:[2001:4118:300:122:444a:3fff:fecc:61a0]:5060 #!else /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ #listen=udp:10.0.0.10:5060 #!endif /* port to listen to * - can be specified more than once if needed to listen on many ports */ port=5060 #!ifdef WITH_TLS enable_tls=yes #!endif # life time of TCP connection when there is no traffic # - a bit higher than registration expires to cope with UA behind NAT tcp_connection_lifetime=3605 ####### Custom Parameters ######### # These parameters can be modified runtime via RPC interface # - see the documentation of 'cfg_rpc' module. # # Format: group.id = value 'desc' description # Access: $sel(cfg_get.group.id) or @cfg_get.group.id # #!ifdef WITH_PSTN # PSTN GW Routing # # - pstn.gw_ip: valid IP or hostname as string value, example: # pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" # # - by default is empty to avoid misrouting pstn.gw_ip = "" desc "PSTN GW Address" #!endif #!ifdef WITH_VOICEMAIL # VoiceMail Routing on offline, busy or no answer # # - by default Voicemail server IP is empty to avoid misrouting voicemail.srv_ip = "" desc "" voicemail.srv_port = "" desc "" #!endif #>> #!ifdef WITH_V4V6 bridge.srv_ip4 = "" desc "SIPv64 PGW IPv4 Address" bridge.srv_ip6 = "" desc "SIPv64 PGW IPv6 Address" #!endif #<< ####### Modules Section ######## # set paths to location of modules (to sources or installation folders) #!ifdef WITH_SRCPATH mpath="modules_k:modules" #!else mpath="/usr/lib/kamailio/modules_k/:/usr/lib/kamailio/modules/" #!endif #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif loadmodule "mi_fifo.so" loadmodule "kex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "mi_rpc.so" loadmodule "acc.so" #>> #!ifdef WITH_V4V6 loadmodule "ipops.so" loadmodule "exec.so" #!endif #<< #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #!ifdef WITH_IPAUTH loadmodule "permissions.so" #!endif #!endif #!ifdef WITH_ALIASDB loadmodule "alias_db.so" #!endif #!ifdef WITH_SPEEDDIAL loadmodule "speeddial.so" #!endif #!ifdef WITH_MULTIDOMAIN loadmodule "domain.so" #!endif #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_NAT loadmodule "nathelper.so" loadmodule "rtpproxy.so" #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_ANTIFLOOD loadmodule "htable.so" loadmodule "pike.so" #!endif #!ifdef WITH_XMLRPC loadmodule "xmlrpc.so" #!endif #!ifdef WITH_DEBUG loadmodule "debugger.so" #!endif # ----------------- setting module-specific parameters --------------- #>> # ----- xlog params ----- modparam("xlog", "prefix", "ProxyLog: ") #<< # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/kamailio_tmp") # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- registrar params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ #modparam("registrar", "max_contacts", 10) # max value for expires of registrations modparam("registrar", "max_expires", 3600) # set it to 1 to enable GRUU modparam("registrar", "gruu_enabled", 0) # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default ww do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") modparam("auth_db", "use_domain", MULTIDOMAIN) # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- speedial params ----- #!ifdef WITH_SPEEDDIAL modparam("speeddial", "db_url", DBURL) modparam("speeddial", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) # register callback to match myself condition with domains list modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) modparam("presence", "db_update_period", 20) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 0) #!endif #!ifdef WITH_XCAPSRV # ----- xcap_server params ----- modparam("xcap_server", "db_url", DBURL) #!endif #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:22222") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@ip02.sip.uniza.sk") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/etc/kamailio/tls.cfg") #!endif #!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 16) modparam("pike", "remove_latency", 4) # ----- htable params ----- # ip ban htable with autoexpire after 5 minutes modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") #!endif #!ifdef WITH_XMLRPC # ----- xmlrpc params ----- modparam("xmlrpc", "route", "XMLRPC"); modparam("xmlrpc", "url_match", "^/RPC") #!endif #!ifdef WITH_DEBUG # ----- debugger params ----- modparam("debugger", "cfgtrace", 1) #!endif ####### Routing Logic ######## # Main SIP request routing logic # - processing of any incoming SIP request starts with this route # - note: this is the same as route { ... } request_route { # per request initial checks route(REQINIT); # NAT detection route(NATDETECT); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) t_check_trans(); # authentication route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } # dispatch requests to foreign domains route(SIPOUT); ### requests for my local domains # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations to PSTN route(PSTN); # user location service route(LOCATION); route(RELAY); } #>> route[INDOMAIN]{ #!ifdef WITH_V4V6 ## if calle in local domain and V4/V6 translation needed -> ## -> rewrite IP in SIP URI back to other version of SIP domain IP if(!has_totag() && isbflagset(FLB_V4V6) && lookup('location')){ xlog("L_INFO","URI domain before change: $rd\n"); if(af == INET){ $ru="sip:" + $rU + "@" + "[2001:4118:300:122:444a:3fff:fecc:61a0]"+":"+$rp; }else{ $ru="sip:" + $rU + "@" + "158.193.139.46" + ":"+$rp; } xlog("L_INFO","Request URI after rewrite: $ru\n"); } #!endif return; } #<< #>> route[FROMBRIDGE]{ if($si==BRIDGE_IP6 || $si==BRIDGE_IP4){ return 1; } return -1; } route[TOBRIDGE]{ #!ifdef WITH_V4V6 if(isbflagset(FLB_V4V6)){ if(af == INET6){ if(!t_relay(BRIDGE_IP6,"5060")) sl_reply_error(); exit; }else{ if(!t_relay(BRIDGE_IP4,"5060")) sl_reply_error(); exit; } } #!endif return; } route[IPV4V6] { #!ifdef WITH_V4V6 if (!has_totag()){ if (to_uri!=myself){ if(from_uri==myself){ # SIP message goes from local domain out if(is_ip($rd)){ # target domain defined with IP address if((af==INET && is_ipv6_reference($rd)) || (af==INET6 && is_ipv4($rd))) { setbflag(FLB_V4V6); } } else { # target domain in R-URI defined with domain name - DNS used # launch script, which returns list of IP addresses of all SIP servers, which provide that service in R-URI # IP addresses saved in $avp(i:1), $avp(i:2), ... exec_avp("/etc/kamailio/test.sh _sip._'$rP'.'$rd'."); if($avp(i:1) != "Unknown"){ $var(i) = 1; $var(bridge) = "yes"; while ($avp($var(i)) != 0){ if((af==INET && is_ipv4($avp($var(i)))) || (af==INET6 && is_ipv6($avp($var(i))))) { # if exists IP with the same AF than AF of sender --> bridge isnt needed $var(bridge) = "no"; break; } $var(i) = $var(i)+1; } xlog("L_INFO","BRIDGE > $var(bridge)"); if($var(bridge) == "yes") { setbflag(FLB_V4V6); } } } } else { # message is only going throught SIP server return; } } else { if(af==INET6 && !isbflagset(FLB_IPV6)){ setbflag(FLB_V4V6); } else { if(af==INET && isbflagset(FLB_IPV6)) { setbflag(FLB_V4V6); } } } } #!endif return; } #<< route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|SUBSCRIBE")) { t_on_branch("MANAGE_BRANCH"); t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { t_on_failure("MANAGE_FAILURE"); } route(IPV4V6); route(INDOMAIN); route(TOBRIDGE); if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood dection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($sht(ipban=>$si)!=$null) { # ip is already blocked xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); exit; } if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); $sht(ipban=>$si) = 1; exit; } } #!endif if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { route(DLGURI); if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } else if ( is_method("ACK") ) { # ACK is forwarded statelessy route(NATMANAGE); } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); } exit; } } # Handle SIP registrations route[REGISTRAR] { if (is_method("REGISTER")) { if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging # setbflag(FLB_NATSIPPING); } if(af==INET6) setbflag(FLB_IPV6); if (!save("location")) sl_reply_error(); exit; } } # USER location service route[LOCATION] { #!ifdef WITH_SPEEDIAL # search for short dialing - 2-digit extension if($rU=~"^[0-9][0-9]$") if(sd_lookup("speed_dial")) route(SIPOUT); #!endif #!ifdef WITH_ALIASDB # search in DB-based aliases if(alias_db_lookup("dbaliases")) route(SIPOUT); #!endif $avp(oexten) = $rU; if (!lookup("location")) { $var(rc) = $rc; t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } } # Presence server route route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; }; if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); exit; } return; } # Authentication route route[AUTH] { #!ifdef WITH_AUTH #!ifdef WITH_V4V6 # messages from SIP Bridge without authentication if(route(FROMBRIDGE)) return; #!endif #!ifdef WITH_IPAUTH if((!is_method("REGISTER")) && allow_source_address()) { # source IP allowed return; } #!endif if (is_method("REGISTER") || from_uri==myself) { # authenticate requests if (!auth_check("$fd", "subscriber", "1")) { auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) consume_credentials(); } # if caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (from_uri!=myself && uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!endif return; } # Caller NAT detection route route[NATDETECT] { # if from IPv6, NAT not needed if(af==INET6) return; #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { add_contact_alias(); } setflag(FLT_NATS); } #!endif return; } # RTPProxy control route[NATMANAGE] { #!ifdef WITH_NAT if(af==INET6) return; if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; rtpproxy_manage("froc"); if (is_request()) { if (!has_totag()) { add_rr_param(";nat=yes"); } } if (is_reply()) { if(isbflagset(FLB_NATB)) { add_contact_alias(); } } #!endif return; } # URI update for dialog requests route[DLGURI] { #!ifdef WITH_NAT if(!isdsturiset()) { handle_ruri_alias(); } #!endif return; } # Routing to foreign domains route[SIPOUT] { if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(RELAY); } } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN rotuing enabled but pstn.gw_ip not defined\n"); return; } # route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) return; # only local users allowed to call if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); route(RELAY); exit; #!endif return; } # XMLRPC routing #!ifdef WITH_XMLRPC route[XMLRPC] { # allow XMLRPC from localhost if ((method=="POST" || method=="GET") && (src_ip==127.0.0.1)) { # close connection only for xmlrpclib user agents (there is a bug in # xmlrpclib: it waits for EOF before interpreting the response). if ($hdr(User-Agent) =~ "xmlrpclib") set_reply_close(); set_reply_no_connect(); dispatch_rpc(); exit; } send_reply("403", "Forbidden"); exit; } #!endif # route to voicemail server route[TOVOICEMAIL] { #!ifdef WITH_VOICEMAIL if(!is_method("INVITE")) return; # check if VoiceMail server IP is defined if (strempty($sel(cfg_get.voicemail.srv_ip))) { xlog("SCRIPT: VoiceMail rotuing enabled but IP not defined\n"); return; } if($avp(oexten)==$null) return; $ru = "sip:" + "vmbox-" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); route(RELAY); exit; #!endif return; } # manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); route(IPV4V6); route(NATMANAGE); } # manage incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") route(NATMANAGE); } # manage failure routing cases failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_is_canceled()) { exit; } #!ifdef WITH_BLOCK3XX # block call redirect based on 3xx replies. if (t_check_status("3[0-9][0-9]")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_VOICEMAIL # serial forking # - route to voicemail on busy or no answer (timeout) if (t_check_status("486|408")) { route(TOVOICEMAIL); exit; } #!endif } ------------------------------------------------------------------------------------------------------------ config file for SIP protocol GW == SIP bridge ------------------------------------------------------------------------------------------------------------ #!KAMAILIO # # Kamailio (OpenSER) SIP Server v3.2 - default configuration script # - web: http://www.kamailio.org # - git: http://sip-router.org # # Direct your questions about this file to: <sr-users(a)lists.sip-router.org> # # Refer to the Core CookBook at http://www.kamailio.org/dokuwiki/doku.php # for an explanation of possible statements, functions and parameters. # # Several features can be enabled using '#!define WITH_FEATURE' directives: # # *** To run in debug mode: # - define WITH_DEBUG # # *** To enable mysql: # - define WITH_MYSQL # # *** To enable authentication execute: # - enable mysql # - define WITH_AUTH # - add users using 'kamctl' # # *** To enable IP authentication execute: # - enable mysql # - enable authentication # - define WITH_IPAUTH # - add IP addresses with group id '1' to 'address' table # # *** To enable persistent user location execute: # - enable mysql # - define WITH_USRLOCDB # # *** To enable presence server execute: # - enable mysql # - define WITH_PRESENCE # # *** To enable nat traversal execute: # - define WITH_NAT # - install RTPProxy: http://www.rtpproxy.org # - start RTPProxy: # rtpproxy -l _your_public_ip_ -s udp:localhost:7722 # # *** To enable PSTN gateway routing execute: # - define WITH_PSTN # - set the value of pstn.gw_ip # - check route[PSTN] for regexp routing condition # # *** To enable database aliases lookup execute: # - enable mysql # - define WITH_ALIASDB # # *** To enable speed dial lookup execute: # - enable mysql # - define WITH_SPEEDDIAL # # *** To enable multi-domain support execute: # - enable mysql # - define WITH_MULTIDOMAIN # # *** To enable TLS support execute: # - adjust CFGDIR/tls.cfg as needed # - define WITH_TLS # # *** To enable XMLRPC support execute: # - define WITH_XMLRPC # - adjust route[XMLRPC] for access policy # # *** To enable anti-flood detection execute: # - adjust pike and htable=>ipban settings as needed (default is # block if more than 16 requests in 2 seconds and ban for 300 seconds) # - define WITH_ANTIFLOOD # # *** To block 3XX redirect replies execute: # - define WITH_BLOCK3XX # # *** To enable VoiceMail routing execute: # - define WITH_VOICEMAIL # - set the value of voicemail.srv_ip # - adjust the value of voicemail.srv_port # # *** To enhance accounting execute: # - enable mysql # - define WITH_ACCDB # - add following columns to database #!ifdef ACCDB_COMMENT ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; #!endif ####### Defined Values ######### #!define WITH_IPV6 #!define WITH_NAT #!define WITH_V4V6 #!define WITH_BRIDGE #!define ADDR_IP4 158.193.139.207 #!ifdef WITH_IPV6 #!define ADDR_IP6 [2001:4118:300:122:a00:27ff:fe60:83c] #!endif # *** Value defines - IDs used later in config #!ifdef WITH_MYSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!define DBURL "mysql://openser:openserrw@localhost/kamailio" #!endif #!ifdef WITH_MULTIDOMAIN # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 #!else #!define MULTIDOMAIN 0 #!endif # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 #!define FLB_NATB 1 #!define FLB_NATSIPPING 2 #!define FLB_V4V6 6 ####### Global Parameters ######### # DNS parametre use_dns_cache=on #dns_try_ipv6=yes #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL0 fork=yes children=4 /* uncomment the next line to disable TCP (default on) */ #disable_tcp=yes /* uncomment the next line to disable the auto discovery of local aliases based on reverse DNS on IPs (default on) */ #auto_aliases=no /* add local domain aliases */ alias="bridge.local" alias="[2001:4118:300:122:a00:27ff:fe60:83c]" alias="158.193.139.207" #!ifdef WITH_IPV6 listen=ADDR_IP4 listen=ADDR_IP6 #!else /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ #listen=udp:10.0.0.10:5060 #!endif /* port to listen to * - can be specified more than once if needed to listen on many ports */ port=5060 #!ifdef WITH_TLS enable_tls=yes #!endif # life time of TCP connection when there is no traffic # - a bit higher than registration expires to cope with UA behind NAT tcp_connection_lifetime=3605 ####### Custom Parameters ######### # These parameters can be modified runtime via RPC interface # - see the documentation of 'cfg_rpc' module. # # Format: group.id = value 'desc' description # Access: $sel(cfg_get.group.id) or @cfg_get.group.id # #!ifdef WITH_PSTN # PSTN GW Routing # # - pstn.gw_ip: valid IP or hostname as string value, example: # pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" # # - by default is empty to avoid misrouting pstn.gw_ip = "" desc "PSTN GW Address" #!endif #!ifdef WITH_VOICEMAIL # VoiceMail Routing on offline, busy or no answer # # - by default Voicemail server IP is empty to avoid misrouting voicemail.srv_ip = "" desc "VoiceMail IP Address" voicemail.srv_port = "5060" desc "VoiceMail Port" #!endif ####### Modules Section ######## # set paths to location of modules (to sources or installation folders) #!ifdef WITH_SRCPATH mpath="modules_k:modules" #!else mpath="/usr/lib/kamailio/modules_k/:/usr/lib/kamailio/modules/" #!endif #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif loadmodule "mi_fifo.so" loadmodule "kex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "mi_rpc.so" loadmodule "acc.so" #!ifdef WITH_V4V6 loadmodule "ipops.so" loadmodule "exec.so" #!endif #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #!ifdef WITH_IPAUTH loadmodule "permissions.so" #!endif #!endif #!ifdef WITH_ALIASDB loadmodule "alias_db.so" #!endif #!ifdef WITH_SPEEDDIAL loadmodule "speeddial.so" #!endif #!ifdef WITH_MULTIDOMAIN loadmodule "domain.so" #!endif #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_NAT loadmodule "nathelper.so" loadmodule "rtpproxy.so" #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_ANTIFLOOD loadmodule "htable.so" loadmodule "pike.so" #!endif #!ifdef WITH_XMLRPC loadmodule "xmlrpc.so" #!endif #!ifdef WITH_DEBUG loadmodule "debugger.so" #!endif # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- registrar params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ #modparam("registrar", "max_contacts", 10) # max value for expires of registrations modparam("registrar", "max_expires", 3600) # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default ww do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") modparam("auth_db", "use_domain", MULTIDOMAIN) # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- speedial params ----- #!ifdef WITH_SPEEDDIAL modparam("speeddial", "db_url", DBURL) modparam("speeddial", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) # use caching modparam("domain", "db_mode", 1) # register callback to match myself condition with domains list modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:22222") #modparam("rtpproxy", "rtpproxy_sock", "unix:/var/run/rtpproxy/rtpproxy.sock") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/etc/kamailio/tls.cfg") #!endif #!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 16) modparam("pike", "remove_latency", 4) # ----- htable params ----- # ip ban htable with autoexpire after 5 minutes modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") #!endif #!ifdef WITH_XMLRPC # ----- xmlrpc params ----- modparam("xmlrpc", "route", "XMLRPC"); modparam("xmlrpc", "url_match", "^/RPC") #!endif #!ifdef WITH_DEBUG # ----- debugger params ----- modparam("debugger", "cfgtrace", 1) #!endif ####### Routing Logic ######## # Main SIP request routing logic # - processing of any incoming SIP request starts with this route # - note: this is the same as route { ... } request_route { # per request initial checks route(BRIDGE); route(REQINIT); # NAT detection route(NATDETECT); # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); # authentication route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } # dispatch requests to foreign domains route(SIPOUT); ### requests for my local domains # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations to PSTN route(PSTN); # user location service route(LOCATION); route(RELAY); } route[BRIDGE]{ #!ifdef WITH_BRIDGE if(to_uri!=myself && from_uri!=myself){ # message goes only throught this server - BRIDGE # => translation provided setbflag(FLB_V4V6); } #!endif return; } route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|SUBSCRIBE")) { t_on_branch("MANAGE_BRANCH"); t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood dection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($sht(ipban=>$si)!=$null) { # ip is already blocked xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); exit; } if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); $sht(ipban=>$si) = 1; exit; } } #!endif if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } if ( is_method("ACK") ) { # ACK is forwarded statelessy route(NATMANAGE); } route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); } exit; } } # Handle SIP registrations route[REGISTRAR] { if (is_method("REGISTER")) { if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging ## setbflag(FLB_NATSIPPING); } if (!save("location")) sl_reply_error(); exit; } } # USER location service route[LOCATION] { #!ifdef WITH_SPEEDIAL # search for short dialing - 2-digit extension if($rU=~"^[0-9][0-9]$") if(sd_lookup("speed_dial")) route(SIPOUT); #!endif #!ifdef WITH_ALIASDB # search in DB-based aliases if(alias_db_lookup("dbaliases")) route(SIPOUT); #!endif $avp(oexten) = $rU; if (!lookup("location")) { $var(rc) = $rc; route(TOVOICEMAIL); t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } } # Presence server route route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; }; if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); exit; } return; } # Authentication route route[AUTH] { #!ifdef WITH_AUTH if (is_method("REGISTER")) { # authenticate the REGISTER requests (uncomment to enable auth) if (!www_authorize("$td", "subscriber")) { www_challenge("$td", "0"); exit; } if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else { #!ifdef WITH_IPAUTH if(allow_source_address()) { # source IP allowed return; } #!endif # authenticate if from local subscriber if (from_uri==myself) { if (!proxy_authorize("$fd", "subscriber")) { proxy_challenge("$fd", "0"); exit; } if (is_method("PUBLISH")) { if ($au!=$fU || $au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } if ($au!=$rU) { sl_send_reply("403","Forbidden R-URI"); exit; } #!ifdef WITH_MULTIDOMAIN if ($fd!=$rd) { sl_send_reply("403","Forbidden R-URI domain"); exit; } #!endif } else { if ($au!=$fU) { sl_send_reply("403","Forbidden auth ID"); exit; } } consume_credentials(); # caller authenticated } else { # caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (!uri==myself) { sl_send_reply("403","Not relaying"); exit; } } } #!endif return; } # Caller NAT detection route route[NATDETECT] { # if from IPv6, NAT not needed if(af==INET6) return; #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { fix_nated_contact(); } setflag(FLT_NATS); } #!endif return; } # RTPProxy control route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); }else { if(check_route_param("nat=v46")) { setbflag(FLB_V4V6); } } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB)|| isbflagset(FLB_V4V6))) return; #rtpproxy_manage(); if(isbflagset(FLB_V4V6)) { if(af==INET6) { rtpproxy_manage("FAEI"); } else { rtpproxy_manage("FAIE"); } } else { if(af==INET6) { # zroj aj ciel su IPV6 rtpproxy_manage("FAEE"); } else { # zdroj aj ciel su IPV4 rtpproxy_manage("FAII"); } } if (is_request()) { if (!has_totag()) { if(isbflagset(FLB_V4V6)) { add_rr_param(";nat=v46"); } else { add_rr_param(";nat=yes"); } } } if (is_reply()) { if(isbflagset(FLB_NATB)) { if(af==INET) { fix_nated_contact(); } } } #!endif return; } # Routing to foreign domains route[SIPOUT] { if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(RELAY); } } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN rotuing enabled but pstn.gw_ip not defined\n"); return; } # route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) return; # only local users allowed to call if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); route(RELAY); exit; #!endif return; } # XMLRPC routing #!ifdef WITH_XMLRPC route[XMLRPC] { # allow XMLRPC from localhost if ((method=="POST" || method=="GET") && (src_ip==127.0.0.1)) { # close connection only for xmlrpclib user agents (there is a bug in # xmlrpclib: it waits for EOF before interpreting the response). if ($hdr(User-Agent) =~ "xmlrpclib") set_reply_close(); set_reply_no_connect(); dispatch_rpc(); exit; } send_reply("403", "Forbidden"); exit; } #!endif # route to voicemail server route[TOVOICEMAIL] { #!ifdef WITH_VOICEMAIL if(!is_method("INVITE")) return; # check if VoiceMail server IP is defined if (strempty($sel(cfg_get.voicemail.srv_ip))) { xlog("SCRIPT: VoiceMail rotuing enabled but IP not defined\n"); return; } if($avp(oexten)==$null) return; $ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip) + $sel(cfg_get.voicemail.srv_port); route(RELAY); exit; #!endif return; } # manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); route(NATMANAGE); } # manage incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") route(NATMANAGE); } # manage failure routing cases failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_is_canceled()) { exit; } #!ifdef WITH_BLOCK3XX # block call redirect based on 3xx replies. if (t_check_status("3[0-9][0-9]")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_VOICEMAIL # serial forking # - route to voicemail on busy or no answer (timeout) if (t_check_status("486|408")) { route(TOVOICEMAIL); exit; } #!endif } ----------------------------------------------------------------------------------------------Here some information from logging:from rtpproxy, wich translates SIP messages between IPv4 and IPv6-----------------------------------------------------------------------------------------------Apr 11 10:12:51 bridge rtpproxy[1103]: INFO:handle_command: new session 4817, tag 12674;1 requested, type strongApr 11 10:12:51 bridge rtpproxy[1103]: INFO:handle_command: new session on a port 36792 created, tag 12674;1Apr 11 10:12:51 bridge rtpproxy[1103]: INFO:handle_command: pre-filling caller's address with 2001:4118:300:122:c41b:584d:c288:90f6:7078Apr 11 10:12:51 bridge rtpproxy[1103]: INFO:handle_command: adding strong flag to existing session, new=1/0/0Apr 11 10:12:51 bridge rtpproxy[1103]: INFO:handle_command: lookup on ports 36792/0, session timer restartedApr 11 10:12:51 bridge rtpproxy[1103]: INFO:handle_command: adding strong flag to existing session, new=1/0/0Apr 11 10:12:51 bri dge rtpproxy[1103]: INFO:handle_command: lookup on ports 36792/0, session timer restartedApr 11 10:12:53 bridge rtpproxy[1103]: INFO:handle_command: lookup on ports 36792/52592, session timer restartedApr 11 10:12:53 bridge rtpproxy[1103]: INFO:handle_command: pre-filling callee's address with 158.193.139.46:36258>>>Apr 11 10:13:21 bridge rtpproxy[1103]: INFO:handle_delete: forcefully deleting session 1 on ports 36792/52592Apr 11 10:13:21 bridge rtpproxy[1103]: INFO:remove_session: RTP stats: 215 in from callee, 214 in from caller, 429 relayed, 0 droppedApr 11 10:13:21 bridge rtpproxy[1103]: INFO:remove_session: RTCP stats: 0 in from callee, 1 in from caller, 1 relayed, 0 droppedApr 11 10:13:21 bridge rtpproxy[1103]: INFO:remove_session: session on ports 36792/52592 is cleaned upApr 11 10:13:21 bridge rtpproxy[1103]: INFO:handle_command: delete request failed: session 4817, tags 12674/NONE not found----------------------------------------------------------------------------- ------------------- My problem is related to the line with >>> If somebody has experience with this kind of problem, i will be very thankful for help. Jakub Hrabovsky (sorry for my english grammar)

2 1

UAC example
by Moacir Ferreira 22 Apr '13

22 Apr '13

Hi community, I will start testing the UAC module. Does anyone have a simple UAC configuration that could share with me? Also, for small (less than 500 users) deployments, would you recommend Kamilio UAC as the solution for giving PSTN access via VoIP providers or would you recommend Asterisk, SEMS, etc.? My intention is to host the solution on a single server. Thanks, Moacir

2 1

Problem accessing git repository
by Mikko Lehto 22 Apr '13

22 Apr '13

Hi Yesterday I got "fatal: read error: Connection reset by peer" to my "git pull" command. I thought it was only temporary network error, but same problem still exist today. I can however access other remote git repositories succesfully from the same host. --- #sip-router 10.58 < mslehto> hmm... I get "Connection reset by peer" to git fetch command 10.59 < mslehto> propably local issue if you just updated master branch succesfully 11.10 < iZverg> git pull 11.11 < iZverg> # git pull 11.11 < iZverg> Already up-to-date. 11.11 < iZverg> i think all ok with server 13.07 < mslehto> I don't get it .pcap dumps shows some git traffic to git.sip-router.org, but finally I get TCP RST 13.07 < mslehto> I have remote.origin.url=git://git.sip-router.org/sip-router 13.08 < mslehto> seems to be same as in http://kamailio.org/wiki/install/devel/git 13.13 < iZverg> url = git://git.sip-router.org/sip-router 13.13 < iZverg> Already up-to-date. --- I think this is not local issue, any pointers what I should troubleshoot next? UPDATE: I am able to get source code via "git clone" to fresh directory with no errors, confusing. -- Mikko

4 9

Basic question
by אורן אברהם 21 Apr '13

21 Apr '13

Dear List. I'm trying to change the contact header of a message. ( by calliing add_contact_alias - that shuld add the ;alias= parameter to the contact header) but, when i print the contact header (with xlog), it shows no change. I know the function works, and that it added the parameter as expected by using debug mode. (it sais that it was added) Can anyone explain me why i don't see that change in the xlog ? (i've printer both $ct and $hdr(Contact)) thank's in advance...

3 4

NAT Traversal Problem
by Ishan Sawhney 19 Apr '13

19 Apr '13

Hi, We have a solution which requires only outbound NAT on SIP calls. Do you have a product which would solve the NAT Traversal problem on the outbound SIP calls? If yes, how much would be the cost of this product? Will it support 2000 simultaneous calls? BR// Ishan

3 2

Transformations
by Victor V. Kustov 19 Apr '13

19 Apr '13

Hi! According "Transformations cookbook" exist string, uri, name transformations. I think need ip transformations, simulary atoi()/aton() and split for stings with divider. By the way: avp_subst("$avp(s:sipa)/$avp(s:abc)/g", "/(.*)\.(.*)\.(.*)\.(.*)/\1/"); avp_subst("$avp(s:sipa)/$avp(s:def)/g", "/(.*)\.(.*)\.(.*)\.(.*)/\2/"); avp_subst("$avp(s:sipa)/$avp(s:ghk)/g", "/(.*)\.(.*)\.(.*)\.(.*)/\3/"); avp_subst("$avp(s:sipa)/$avp(s:xyz)/g", "/(.*)\.(.*)\.(.*)\.(.*)/\4/"); $avp(s:numipa)=$avp(s:abc{s.int})<<24||$avp(s:defi{s.int})<<16||$avp(s:ghki{s.int})<<8||$avp(s:xyz{s.int}); where is error? -- SY, Victor JID: coyote(a)bks.tv JID: coyote(a)bryansktel.ru I use FREE operation system: 3.8.4-calculate GNU/Linux

2 2

Kamailio World Quick Follow Up
by Daniel-Constantin Mierla 18 Apr '13

18 Apr '13

Hello, first I want to thank to all participants to Kamailio World Conference, everyone having a relevant contribution to it, turning out the two days in a very successful event. Special thanks to our sponsors - Sipwise, Fraunhofer Fokus, Asipto, Sipgate, NG-Voice, IT Center, Amooma and Zoiper - without them it would have been pretty impossible to bring a great group of speakers, have a nice location, excellent logistics and services on site - practically all that had matter to get such great event. As organizers, we learned a lot, significantly helping to get even better for next editions. Please do send feedback to us, it is very important to improve for the next years, even if you didn't participate, you can tell us what you would like to have at such event. For everyone in the community I want to say that the slides should be published in the near future. We also recorded the sessions, I haven't checked the movies yet, if the quality is fair enough, then they will be made available (probably will take a bit longer than the slides). Several sessions were recorded separately also by Randy from VUC, being already available on youtube: - https://www.youtube.com/user/voipusers/videos?sort=dd I will send soon more details, as new resources are made available. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

Implementing the Presence module(Kamailio) to work with websockets (sipML5)
by אורן אברהם 17 Apr '13

17 Apr '13

Dear List,

1 0

Ldap Authenticate issues
by Sebastian Bergerot 17 Apr '13

17 Apr '13

Hi I´m trying to authenticate on active directory but i have issues with the password. Here is my CFG Kamailio.cfg #!define WITH_DEBUG #!KAMAILIO # # Kamailio (OpenSER) SIP Server v3.2 - default configuration script # - web: http://www.kamailio.org # - git: http://sip-router.org # # Direct your questions about this file to: <sr-users at lists.sip-router.org> # # Refer to the Core CookBook at http://www.kamailio.org/dokuwiki/doku.php # for an explanation of possible statements, functions and parameters. # # Several features can be enabled using '#!define WITH_FEATURE' directives: # # *** To run in debug mode: # - define WITH_DEBUG # # *** To enable mysql: # - define WITH_MYSQL # # *** To enable authentication execute: # - enable mysql # - define WITH_AUTH # - add users using 'kamctl' # # *** To enable IP authentication execute: # - enable mysql # - enable authentication # - define WITH_IPAUTH # - add IP addresses with group id '1' to 'address' table # # *** To enable persistent user location execute: # - enable mysql # - define WITH_USRLOCDB # # *** To enable presence server execute: # - enable mysql # - define WITH_PRESENCE # # *** To enable nat traversal execute: # - define WITH_NAT # - install RTPProxy: http://www.rtpproxy.org # - start RTPProxy: # rtpproxy -l _your_public_ip_ -s udp:localhost:7722 # ###!define WITH_NAT # *** To enable PSTN gateway routing execute: # - define WITH_PSTN # - set the value of pstn.gw_ip # - check route[PSTN] for regexp routing condition # # *** To enable database aliases lookup execute: # - enable mysql # - define WITH_ALIASDB # # *** To enable speed dial lookup execute: # - enable mysql # - define WITH_SPEEDDIAL # # *** To enable multi-domain support execute: # - enable mysql # - define WITH_MULTIDOMAIN # # *** To enable TLS support execute: # - adjust CFGDIR/tls.cfg as needed # - define WITH_TLS # # *** To enable XMLRPC support execute: # - define WITH_XMLRPC # - adjust route[XMLRPC] for access policy # # *** To enable anti-flood detection execute: # - adjust pike and htable=>ipban settings as needed (default is # block if more than 16 requests in 2 seconds and ban for 300 seconds) # - define WITH_ANTIFLOOD # # *** To block 3XX redirect replies execute: # - define WITH_BLOCK3XX # # *** To enable VoiceMail routing execute: # - define WITH_VOICEMAIL # - set the value of voicemail.srv_ip # - adjust the value of voicemail.srv_port # # *** To enhance accounting execute: # - enable mysql # - define WITH_ACCDB # - add following columns to database #!define WITH_LDAP #!define WITH_AUTH #!define WITH_MYSQL #!define WITH_USRLOCDB #!define WITH_PRESENCE #!ifdef ACCDB_COMMENT ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; #!endif ####### Defined Values ######### # *** Value defines - IDs used later in config #!ifdef WITH_MYSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!define DBURL "mysql://root:bernardo@128.2.101.19/kamailio" #!endif #!ifdef WITH_MULTIDOMAIN # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 #!else #!define MULTIDOMAIN 0 #!endif # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 ####### Global Parameters ######### #!ifdef WITH_DEBUG debug=4 log_stderror=no #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL0 fork=yes children=4 /* uncomment the next line to disable TCP (default on) */ #disable_tcp=yes /* uncomment the next line to disable the auto discovery of local aliases based on reverse DNS on IPs (default on) */ auto_aliases=no /* add local domain aliases */ alias="coto.com.ar" /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ listen=tcp:128.2.101.19:5060 #listen=tcp:192.168.14.25:5060 /* port to listen to * - can be specified more than once if needed to listen on many ports */ #port=5060 #!ifdef WITH_TLS enable_tls=yes #!endif # life time of TCP connection when there is no traffic # - a bit higher than registration expires to cope with UA behind NAT tcp_connection_lifetime=3605 ####### Custom Parameters ######### # These parameters can be modified runtime via RPC interface # - see the documentation of 'cfg_rpc' module. # # Format: group.id = value 'desc' description # Access: $sel(cfg_get.group.id) or @cfg_get.group.id # #!ifdef WITH_PSTN # PSTN GW Routing # # - pstn.gw_ip: valid IP or hostname as string value, example: # pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" # # - by default is empty to avoid misrouting pstn.gw_ip = "" desc "PSTN GW Address" #!endif ####### Modules Section ######## # set paths to location of modules (to sources or installation folders) #!ifdef WITH_SRCPATH mpath="modules_k:modules" #!else mpath="/usr/lib/kamailio/modules/" #!endif #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif loadmodule "mi_fifo.so" loadmodule "kex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" # loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "mi_rpc.so" loadmodule "acc.so" #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #chris loadmodule "ldap.so" modparam ("ldap", "config_file", "/etc/kamailio/ldap.cfg") #!ifdef WITH_IPAUTH loadmodule "permissions.so" #!endif #!endif #!ifdef WITH_ALIASDB loadmodule "alias_db.so" #!endif #!ifdef WITH_SPEEDDIAL loadmodule "speeddial.so" #!endif #!ifdef WITH_MULTIDOMAIN loadmodule "domain.so" #!endif #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_NAT loadmodule "nathelper.so" loadmodule "rtpproxy.so" #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_ANTIFLOOD loadmodule "htable.so" loadmodule "pike.so" #!endif #!ifdef WITH_XMLRPC loadmodule "xmlrpc.so" #!endif #!ifdef WITH_DEBUG loadmodule "debugger.so" #!endif # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- #modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") modparam("mi_fifo", "fifo_name", "/tmp/kamailio_tmp") # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) # suppress the check for the CSEQ method # modparam("sanity", "default_checks", 967) # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) #edit asterisk #!ifdef WITH_ASTERISK modparam("rr", "append_fromtag", 1) #!else modparam("rr", "append_fromtag", 0) #!endif # ----- registrar params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ modparam("registrar", "max_contacts", 256) # max value for expires of registrations modparam("registrar", "max_expires", 3600) # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default ww do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif #chris commented out this part # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "load_credentials", "") #!ifdef WITH_ASTERISK modparam("auth_db", "user_column", "username") modparam("auth_db", "password_column", "sippasswd") modparam("auth_db", "db_url", DBASTURL) modparam("auth_db", "version_table", 0) #!else modparam("auth_db", "db_url", DBURL) modparam("auth_db", "password_column", "password") modparam("auth_db", "use_domain", MULTIDOMAIN) #!endif # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- speedial params ----- #!ifdef WITH_SPEEDDIAL modparam("speeddial", "db_url", DBURL) modparam("speeddial", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) # use caching modparam("domain", "db_mode", 1) # register callback to match myself condition with domains list modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:192.168.14.25:22222") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger at teopad-toip.corp") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/etc/kamailio/tls.cfg") #!endif #!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 16) modparam("pike", "remove_latency", 4) # ----- htable params ----- # ip ban htable with autoexpire after 5 minutes modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") #!endif #!ifdef WITH_XMLRPC # ----- xmlrpc params ----- modparam("xmlrpc", "route", "XMLRPC"); modparam("xmlrpc", "url_match", "^/RPC") #!endif #!ifdef WITH_DEBUG # ----- debugger params ----- modparam("debugger", "cfgtrace", 1) #!endif ####### Routing Logic ######## # Main SIP request routing logic # - processing of any incoming SIP request starts with this route # - note: this is the same as route { ... } request_route { # per request initial checks route(REQINIT); xlog("L_INFO","apres REQINIT"); # NAT detection route(NATDETECT); xlog("L_INFO","apres NATDETECT"); # handle requests within SIP dialogs route(WITHINDLG); xlog("L_INFO","apres WITHINDLG"); ### only initial requests (no To tag) # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); xlog("L_INFO","apres t_check_trans"); # authentication route(AUTH); xlog("L_INFO","apres AUTH"); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) record_route(); xlog("L_INFO","apres RECORD ROUTE"); # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } xlog("L_INFO","apres INVITE"); # dispatch requests to foreign domains route(SIPOUT); xlog("L_INFO","apres SIPOUT"); ### requests for my local domains # handle presence related requests route(PRESENCE); xlog("L_INFO","apres PRESENCE"); # handle registrations route(REGISTRAR); xlog("L_INFO","apres REGISTRAR"); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations to PSTN route(PSTN); xlog("L_INFO","apres PSTN"); # user location service route(LOCATION); xlog("L_INFO","apres LOCATION"); route(RELAY); xlog("L_INFO","apres RELAY"); } route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. xlog("L_INFO","Dans route relay"); if (is_method("INVITE|SUBSCRIBE")) { xlog("L_INFO","avant manage branch"); t_on_branch("MANAGE_BRANCH"); xlog("L_INFO","avant manage reply"); t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { xlog("L_INFO","avant manage failure"); t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { xlog("L_INFO","reply error"); sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood dection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($sht(ipban=>$si)!=$null) { # ip is already blocked xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); exit; } if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); $sht(ipban=>$si) = 1; exit; } } #!endif if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } # if(!sanity_check("1511", "7")) # { # xlog("Malformed SIP message from $si:$sp\n"); # exit; # } } # Handle requests within SIP dialogs route[WITHINDLG] { xlog("L_INFO","Dans WITHINDLG"); if (has_totag()) { xlog("L_INFO","dans has totag"); # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { xlog("L_INFO","looseroute"); if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } if ( is_method("ACK") ) { xlog("L_INFO","ack"); # ACK is forwarded statelessy route(NATMANAGE); } xlog("L_INFO","relay"); route(RELAY); } else { xlog("L_INFO","else"); if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests xlog("L_INFO","subscribe avant presence"); route(PRESENCE); xlog ("L_INFO","apres presence"); exit; } if ( is_method("ACK") ) { xlog("L_INFO","else ack"); if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server xlog("L_INFO","else ack avant relay"); t_relay(); xlog("L_INFO","else ack apres relay"); exit; } else { # ACK without matching transaction ... ignore and discard xlog("L_INFO","else final"); exit; } } sl_send_reply("404","Not here"); } exit; } } # Handle SIP registrations route[REGISTRAR] { if (is_method("REGISTER")) { if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging setbflag(FLB_NATSIPPING); } if (!save("location")) sl_reply_error(); #edit asterisk #!ifdef WITH_ASTERISK xlog ("L_INFO","avant regfwd dans registrar"); route(REGFWD); xlog ("L_INFO","apres regfwd dans registrar"); #!endif exit; } } # USER location service route[LOCATION] { #!ifdef WITH_SPEEDIAL # search for short dialing - 2-digit extension if($rU=~"^[0-9][0-9]$") if(sd_lookup("speed_dial")) route(SIPOUT); #!endif #!ifdef WITH_ALIASDB # search in DB-based aliases if(alias_db_lookup("dbaliases")) route(SIPOUT); #!endif #edit asterisk #!ifdef WITH_ASTERISK if(is_method("INVITE") && (!route(FROMASTERISK))) { #if new call from out there - send to Asterisk # - non-INVITE requests are routed directly by Kamailio # - traffic from Asterisk is router also directly by Kamailio xlog ("L_INFO", "avant toasterisk dans location"); route(TOASTERISK); xlog ("L_INFO", "apres toasterisk dans location"); exit; } #!endif $avp(oexten) = $rU; if (!lookup("location")) { $var(rc) = $rc; route(TOVOICEMAIL); t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } } # Presence server route route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; }; if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); exit; } return; } # Authentication route route[AUTH] { #!ifdef WITH_AUTH #edit asterisk #!ifdef WITH_ASTERISK #do not auth traffic from Asterisk: trusted! xlog ("L_INFO", "avant if route fromasterisk"); if(route(FROMASTERISK)) return; #!endif if (is_method("REGISTER")) # { # # authenticate the REGISTER requests (uncomment to enable auth) # if (!www_authorize("$td", "subscriber")) # { # www_challenge("$td", "0"); # exit; # } # # if ($au!=$tU) # { # sl_send_reply("403","Forbidden auth ID"); # exit; # } { #edit asterisk ##!ifdef WITH_ASTERISK # xlog ("L_INFO", "dans auth / authcheck sipusers"); # if (!auth_check("$fd","sipusers","1")) ##!else if(is_present_hf("Authorization")) ##!endif { xlog ("L_INFO", "ldap://sipaccounts/OU=_Usuarios,DC=redcoto,DC=com,DC=ar?sAMAccountName,userPassword?sub?(sAMAccountName=$fU))"); # ldap search #if (!ldap_search("ldap://sipaccounts/OU=_Usuarios,DC=redcoto,DC=com,DC=ar?teopad-Sip-Username,teopadSipPassword?sub?(teopad-Sip-Username=$fU)")) if (!ldap_search("ldap://sipaccounts/OU=_Usuarios,DC=redcoto,DC=com,DC=ar?sAMAccountName,userPassword?sub?(sAMAccountName=$fU)")) { switch ($retcode) { case -1: # no LDAP entry found sl_send_reply("404", "User Not Found"); xlog("L_INFO", "ldap_search: NO found [$retcode] entries for (sAMAccountName=$fU)"); exit; case -2: # internal error sl_send_reply("500", "Internal server error"); exit; default: exit; } } ldap_result("sAMAccountName/$avp(s:username)"); #ldap_result("userPassword/$avp(s:password)"); #ldap_result("teopad-Sip-Username/$avp(s:username)"); #ldap_result("teopadSipPassword/$avp(s:password)"); xlog("L_INFO", "ldap_search: found [$retcode] entries for (uid=$fU)"); xlog('Username=$avp(s:username)'); #xlog('Password=$avp(s:password)'); if(!pv_www_authenticate("$td", "$avp(s:password)", "0")) { xlog ("L_INFO", "ldap pv_authenticate failed") ; www_challenge("$td","1"); exit; } save("location"); sl_send_reply("200", "ok"); xlog ("L_INFO", "ldap pv_authenticate ok") ; exit; } else { www_challenge("$td","1"); exit; } } else { #!ifdef WITH_IPAUTH if(allow_source_address()) { # source IP allowed return; } #!endif # # authenticate if from local subscriber if (from_uri==myself) { # if (!proxy_authorize("$fd", "subscriber")) { # proxy_challenge("$fd", "0"); # exit; # } if (is_method("PUBLISH")) { xlog ("L_INFO", "au = $au") ; xlog ("L_INFO", "fU = $fU") ; xlog ("L_INFO", "tU = $tU") ; xlog ("L_INFO", "fd = $fd") ; xlog ("L_INFO", "rd = $rd") ; if ($au!=$fU || $au!=$tU) { sl_send_reply("403","Forbidden auth ID au!=fu ou au!=tu"); exit; } if ($au!=$rU) { sl_send_reply("403","Forbidden R-URI"); exit; } #!ifdef WITH_MULTIDOMAIN if ($fd!=$rd) { sl_send_reply("403","Forbidden R-URI domain"); exit; } #!endif } else { xlog ("L_INFO", "au = $au") ; xlog ("L_INFO", "fU = $fU") ; # if ($au!=$fU) { # sl_send_reply("403","Forbidden auth ID au!=fu"); # exit; # } } #consume_credentials(); # caller authenticated } else { # caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (!uri==myself) { sl_send_reply("403","Not relaying"); exit; } } } #!endif return; } # Caller NAT detection route route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { fix_nated_contact(); } setflag(FLT_NATS); } #!endif return; } # RTPProxy control route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; rtpproxy_manage(); #rtpproxy_manage("co","82.127.95.167"); if (is_request()) { if (!has_totag()) { add_rr_param(";nat=yes"); } } if (is_reply()) { if(isbflagset(FLB_NATB)) { fix_nated_contact(); } } #!endif return; } # Routing to foreign domains route[SIPOUT] { if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(RELAY); } } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN rotuing enabled but pstn.gw_ip not defined\n"); return; } # route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) return; # only local users allowed to call if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); route(RELAY); exit; #!endif return; } # XMLRPC routing #!ifdef WITH_XMLRPC route[XMLRPC] { # allow XMLRPC from localhost if ((method=="POST" || method=="GET") && (src_ip==127.0.0.1)) { # close connection only for xmlrpclib user agents (there is a bug in # xmlrpclib: it waits for EOF before interpreting the response). if ($hdr(User-Agent) =~ "xmlrpclib") set_reply_close(); set_reply_no_connect(); dispatch_rpc(); exit; } send_reply("403", "Forbidden"); exit; } #!endif # route to voicemail server route[TOVOICEMAIL] { #!ifdef WITH_VOICEMAIL if(!is_method("INVITE")) return; # check if VoiceMail server IP is defined if (strempty($sel(cfg_get.voicemail.srv_ip))) { xlog("SCRIPT: VoiceMail rotuing enabled but IP not defined\n"); return; } if($avp(oexten)==$null) return; $ru = "tcp:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); xlog("L_INFO","tovoicemail ru: $ru"); route(RELAY); exit; #!endif return; } # manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); route(NATMANAGE); } # manage incoming replies onreply_route[MANAGE_REPLY] { xlog("L_INFO","dans managereply"); xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]"){ xlog("L_INFO","avant route natmanage"); route(NATMANAGE); } } # manage failure routing cases failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_is_canceled()) { exit; } #!ifdef WITH_BLOCK3XX # block call redirect based on 3xx replies. if (t_check_status("3[0-9][0-9]")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_VOICEMAIL # serial forking # - route to voicemail on busy or no answer (timeout) if (t_check_status("486|408")) { route(TOVOICEMAIL); exit; } #!endif } #edit asterisk #!ifdef WITH_ASTERISK # Test if coming from Asterisk route[FROMASTERISK] { xlog ("L_INFO", "Dans FROMASTERISK? $si / $sp"); if($si==$sel(cfg_get.asterisk.bindip) && $sp==$sel(cfg_get.asterisk.bindport)) return 1; return -1; } # Send to Asterisk route[TOASTERISK] { $du = "sip:" + $sel(cfg_get.asterisk.bindip) + ":" + $sel(cfg_get.asterisk.bindport); xlog ("L_INFO", "Dans TOASTERISK $du") ; xlog ("L_INFO", "Juste avant route relay"); route(RELAY); exit; } # Forward REGISTER to Asterisk route[REGFWD] { xlog("L_INFO", "Dans REGFWD"); if(!is_method("REGISTER")) { return; } $var(rip) = $sel(cfg_get.asterisk.bindip); $uac_req(method)="REGISTER"; $uac_req(ruri)="sip:" + $var(rip) + ":" + $sel(cfg_get.asterisk.bindport); $uac_req(furi)="sip:" + $au + "@" + $var(rip); $uac_req(turi)="sip:" + $au + "@" + $var(rip); $uac_req(hdrs)="Contact: <sip:" + $au + "@" + $sel(cfg_get.kamailio.bindip) + ":" + $sel(cfg_get.kamailio.bindport) + ">\r\n"; xlog("L_INFO","avant if dans regfwd"); if($sel(contact.expires) != $null) $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $sel(contact.expires) + "\r\n"; else $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $hdr(Expires) + "\r\n"; uac_req_send(); } #!endif And the Log Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3066]: DEBUG: <core> [io_wait.h:617]: DBG: io_watch_del (0x825fde0, 19, -1, 0x0) fd_no=14 called Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3066]: DEBUG: <core> [tcp_main.c:4299]: tcp: DBG: sending to child, events 1 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3066]: DEBUG: <core> [tcp_main.c:3970]: selected tcp worker 2 7(3063) for activity on [tcp: 128.2.101.19:5060], 0xb34ed968 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: DEBUG: <core> [tcp_read.c:1507]: received n=4 con=0xb34ed968, fd=8 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: DEBUG: <core> [tcp_read.c:1311]: tcp_read_req: content-length= 0 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: DEBUG: <core> [parser/msg_parser.c:623]: SIP Request: Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: DEBUG: <core> [parser/msg_parser.c:625]: method: <REGISTER> Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: DEBUG: <core> [parser/msg_parser.c:627]: uri: <sip:128.2.101.19> Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: DEBUG: <core> [parser/msg_parser.c:629]: version: <SIP/2.0> Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: DEBUG: <core> [parser/parse_via.c:1284]: Found param type 235, <rport> = <n/a>; state=6 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: DEBUG: <core> [parser/parse_via.c:1284]: Found param type 232, <branch> = <z9hG4bK1898036878>; state=16 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: DEBUG: <core> [parser/parse_via.c:2672]: end of header reached, state=5 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: DEBUG: <core> [parser/msg_parser.c:513]: parse_headers: Via found, flags=2 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: DEBUG: <core> [parser/msg_parser.c:515]: parse_headers: this is the first via Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: DEBUG: <core> [receive.c:149]: After parse_msg... Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: DEBUG: <core> [receive.c:190]: preparing to run routing scripts... Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=487 a=5 n=route Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=620 a=16 n=if Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=610 a=25 n=mf_process_maxfwd_header Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: DEBUG: <core> [parser/parse_to.c:799]: end of header reached, state=10 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: DEBUG: <core> [parser/msg_parser.c:190]: DEBUG: get_hdr_field: <To> [30]; uri=[ sip:sbergerot@128.2.101.19] Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: DEBUG: <core> [parser/msg_parser.c:192]: DEBUG: to body [<sip:sbergerot@128.2.101.19 >#015#012] Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: DEBUG: <core> [parser/msg_parser.c:170]: get_hdr_field: cseq <CSeq>: <5> <REGISTER> Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: DEBUG: maxfwd [mf_funcs.c:85]: value = 70 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: DEBUG: maxfwd [maxfwd.c:161]: value 70 decreased to 16 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=488 a=26 n=xlog Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: INFO: <script>: apres REQINIT Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=491 a=5 n=route Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=946 a=2 n=return Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=492 a=26 n=xlog Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: INFO: <script>: apres NATDETECT Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=495 a=5 n=route Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=624 a=26 n=xlog Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: INFO: <script>: Dans WITHINDLG Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=671 a=16 n=if Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=625 a=24 n=has_totag Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: DEBUG: siputils [checks.c:103]: no totag Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=496 a=26 n=xlog Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: INFO: <script>: apres WITHINDLG Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=508 a=16 n=if Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=501 a=25 n=is_method Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=508 a=24 n=t_check_trans Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: DEBUG: tm [t_lookup.c:1095]: DEBUG: t_check_msg: msg id=333 global id=332 T start=(nil) Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: DEBUG: <core> [parser/msg_parser.c:204]: DEBUG: get_hdr_body : content_length=0 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: DEBUG: <core> [parser/msg_parser.c:106]: found end of header Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: DEBUG: tm [t_lookup.c:534]: t_lookup_request: start searching: hash=61476, isACK=0 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: DEBUG: tm [t_lookup.c:492]: DEBUG: RFC3261 transaction matching failed Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: DEBUG: tm [t_lookup.c:716]: DEBUG: t_lookup_request: no transaction found Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: DEBUG: tm [t_lookup.c:1164]: DEBUG: t_check_msg: msg id=333 global id=333 T end=(nil) Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=509 a=26 n=xlog Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: INFO: <script>: apres t_check_trans Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=512 a=5 n=route Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=928 a=16 n=if Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=795 a=25 n=is_method Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=863 a=16 n=if Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=817 a=25 n=is_present_hf Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=861 a=26 n=www_challenge Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: DEBUG: auth [challenge.c:127]: build_challenge_hf: realm='128.2.101.19' Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: DEBUG: auth [challenge.c:138]: build_challenge_hf: qop='auth' Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: DEBUG: auth [challenge.c:269]: auth: 'WWW-Authenticate: Digest realm="128.2.101.19", nonce="UW6DllFugmoxy+OjSDvSKiZswnyYapMe", qop="auth"#015#012' Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: DEBUG: sl [sl.c:289]: reply in stateless mode (sl) Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: DEBUG: <core> [msg_translator.c:206]: check_via_address(172.30.203.182, 172.30.203.182, 0) Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: DEBUG: <core> [tcp_main.c:2317]: tcp_send: send from reader (3063 (7)), reusing fd Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: DEBUG: <core> [tcp_main.c:2553]: tcp_send: sending... Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: DEBUG: <core> [tcp_main.c:2587]: tcp_send: after real write: c= 0xb34ed968 n=429 fd=8 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: DEBUG: <core> [tcp_main.c:2588]: tcp_send: buf=#012SIP/2.0 401 Unauthorized#015#012Via: SIP/2.0/TCP 172.30.203.182:50210;rport=49733;branch=z9hG4bK1898036878#015#012From: <sip:sbergerot@128.2.101.19>;tag=1684345209#015#012To: < sip:sbergerot@128.2.101.19>;tag=1e63bc776842177dff382010e03d32cf.5fba#015#012Call-ID: 1455060142#015#012CSeq: 5 REGISTER#015#012WWW-Authenticate: Digest realm="128.2.101.19", nonce="UW6DllFugmoxy+OjSDvSKiZswnyYapMe", qop="auth"#015#012Server: kamailio (4.0.0 (i386/linux))#015#012Content-Length: 0#015#012#015#012 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=862 a=2 n=exit Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: DEBUG: <core> [xavp.c:447]: destroying xavp list (nil) Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: DEBUG: <core> [receive.c:293]: receive_msg: cleaning up Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: DEBUG: <core> [io_wait.h:390]: DBG: io_watch_add(0x8282020, 8, 2, 0xb34ed968), fd_no=1 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: DEBUG: <core> [tcp_read.c:300]: EOF on 0xb34ed968, FD 8 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: DEBUG: <core> [tcp_read.c:1287]: tcp_read_req: EOF Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: DEBUG: <core> [io_wait.h:617]: DBG: io_watch_del (0x8282020, 8, -1, 0x10) fd_no=2 called Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: DEBUG: <core> [tcp_read.c:1434]: releasing con 0xb34ed968, state -1, fd=8, id=5 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: DEBUG: <core> [tcp_read.c:1435]: extra_data (nil) Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3066]: DEBUG: <core> [tcp_main.c:3382]: handle_tcp_child: reader response= b34ed968, -1 from 2 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3066]: DEBUG: <core> [ip_addr.c:247]: tcpconn_new: new tcp connection: 172.30.203.182 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3066]: DEBUG: <core> [tcp_main.c:1089]: tcpconn_new: on port 49734, type 2 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3066]: DEBUG: <core> [tcp_main.c:1403]: tcpconn_add: hashes: 128:420:583, 6 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3066]: DEBUG: <core> [io_wait.h:390]: DBG: io_watch_add(0x825fde0, 19, 2, 0xb34ed968), fd_no=13 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3066]: DEBUG: <core> [io_wait.h:617]: DBG: io_watch_del (0x825fde0, 19, -1, 0x0) fd_no=14 called Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3066]: DEBUG: <core> [tcp_main.c:4299]: tcp: DBG: sending to child, events 1 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3066]: DEBUG: <core> [tcp_main.c:3970]: selected tcp worker 3 8(3064) for activity on [tcp: 128.2.101.19:5060], 0xb34ed968 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [tcp_read.c:1507]: received n=4 con=0xb34ed968, fd=8 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [tcp_read.c:1311]: tcp_read_req: content-length= 0 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [parser/msg_parser.c:623]: SIP Request: Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [parser/msg_parser.c:625]: method: <REGISTER> Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [parser/msg_parser.c:627]: uri: <sip:128.2.101.19> Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [parser/msg_parser.c:629]: version: <SIP/2.0> Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [parser/parse_via.c:1284]: Found param type 235, <rport> = <n/a>; state=6 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [parser/parse_via.c:1284]: Found param type 232, <branch> = <z9hG4bK815445088>; state=16 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [parser/parse_via.c:2672]: end of header reached, state=5 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [parser/msg_parser.c:513]: parse_headers: Via found, flags=2 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [parser/msg_parser.c:515]: parse_headers: this is the first via Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [receive.c:149]: After parse_msg... Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [receive.c:190]: preparing to run routing scripts... Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=487 a=5 n=route Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=620 a=16 n=if Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=610 a=25 n=mf_process_maxfwd_header Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [parser/parse_to.c:799]: end of header reached, state=10 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [parser/msg_parser.c:190]: DEBUG: get_hdr_field: <To> [30]; uri=[ sip:sbergerot@128.2.101.19] Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [parser/msg_parser.c:192]: DEBUG: to body [<sip:sbergerot@128.2.101.19 >#015#012] Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [parser/msg_parser.c:170]: get_hdr_field: cseq <CSeq>: <1> <REGISTER> Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: maxfwd [mf_funcs.c:85]: value = 70 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: maxfwd [maxfwd.c:161]: value 70 decreased to 16 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=488 a=26 n=xlog Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: INFO: <script>: apres REQINIT Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=491 a=5 n=route Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=946 a=2 n=return Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=492 a=26 n=xlog Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: INFO: <script>: apres NATDETECT Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=495 a=5 n=route Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=624 a=26 n=xlog Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: INFO: <script>: Dans WITHINDLG Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=671 a=16 n=if Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=625 a=24 n=has_totag Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: siputils [checks.c:103]: no totag Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=496 a=26 n=xlog Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: INFO: <script>: apres WITHINDLG Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=508 a=16 n=if Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=501 a=25 n=is_method Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=508 a=24 n=t_check_trans Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: tm [t_lookup.c:1095]: DEBUG: t_check_msg: msg id=327 global id=326 T start=(nil) Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [parser/msg_parser.c:204]: DEBUG: get_hdr_body : content_length=0 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [parser/msg_parser.c:106]: found end of header Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: tm [t_lookup.c:534]: t_lookup_request: start searching: hash=27623, isACK=0 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: tm [t_lookup.c:492]: DEBUG: RFC3261 transaction matching failed Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: tm [t_lookup.c:716]: DEBUG: t_lookup_request: no transaction found Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: tm [t_lookup.c:1164]: DEBUG: t_check_msg: msg id=327 global id=327 T end=(nil) Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=509 a=26 n=xlog Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: INFO: <script>: apres t_check_trans Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=512 a=5 n=route Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=928 a=16 n=if Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=795 a=25 n=is_method Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=863 a=16 n=if Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=817 a=25 n=is_present_hf Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=861 a=26 n=www_challenge Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: auth [challenge.c:127]: build_challenge_hf: realm='128.2.101.19' Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: auth [challenge.c:138]: build_challenge_hf: qop='auth' Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: auth [challenge.c:269]: auth: 'WWW-Authenticate: Digest realm="128.2.101.19", nonce="UW6DllFugmoxy+OjSDvSKiZswnyYapMe", qop="auth"#015#012' Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: sl [sl.c:289]: reply in stateless mode (sl) Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [msg_translator.c:206]: check_via_address(172.30.203.182, 172.30.203.182, 0) Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [tcp_main.c:2317]: tcp_send: send from reader (3064 (8)), reusing fd Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [tcp_main.c:2553]: tcp_send: sending... Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [tcp_main.c:2587]: tcp_send: after real write: c= 0xb34ed968 n=428 fd=8 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [tcp_main.c:2588]: tcp_send: buf=#012SIP/2.0 401 Unauthorized#015#012Via: SIP/2.0/TCP 172.30.203.182:25204;rport=49734;branch=z9hG4bK815445088#015#012From: <sip:sbergerot@128.2.101.19>;tag=1120136754#015#012To: < sip:sbergerot@128.2.101.19>;tag=1e63bc776842177dff382010e03d32cf.de09#015#012Call-ID: 1585139518#015#012CSeq: 1 REGISTER#015#012WWW-Authenticate: Digest realm="128.2.101.19", nonce="UW6DllFugmoxy+OjSDvSKiZswnyYapMe", qop="auth"#015#012Server: kamailio (4.0.0 (i386/linux))#015#012Content-Length: 0#015#012#015#012 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=862 a=2 n=exit Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [xavp.c:447]: destroying xavp list (nil) Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [receive.c:293]: receive_msg: cleaning up Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [io_wait.h:390]: DBG: io_watch_add(0x8282020, 8, 2, 0xb34ed968), fd_no=1 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [tcp_read.c:1311]: tcp_read_req: content-length= 0 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [parser/msg_parser.c:623]: SIP Request: Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [parser/msg_parser.c:625]: method: <REGISTER> Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [parser/msg_parser.c:627]: uri: <sip:128.2.101.19> Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [parser/msg_parser.c:629]: version: <SIP/2.0> Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [parser/parse_via.c:1284]: Found param type 235, <rport> = <n/a>; state=6 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [parser/parse_via.c:1284]: Found param type 232, <branch> = <z9hG4bK1107819308>; state=16 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [parser/parse_via.c:2672]: end of header reached, state=5 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [parser/msg_parser.c:513]: parse_headers: Via found, flags=2 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [parser/msg_parser.c:515]: parse_headers: this is the first via Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [receive.c:149]: After parse_msg... Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [receive.c:190]: preparing to run routing scripts... Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=487 a=5 n=route Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=620 a=16 n=if Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=610 a=25 n=mf_process_maxfwd_header Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [parser/parse_to.c:799]: end of header reached, state=10 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [parser/msg_parser.c:190]: DEBUG: get_hdr_field: <To> [30]; uri=[ sip:sbergerot@128.2.101.19] Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [parser/msg_parser.c:192]: DEBUG: to body [<sip:sbergerot@128.2.101.19 >#015#012] Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [parser/msg_parser.c:170]: get_hdr_field: cseq <CSeq>: <2> <REGISTER> Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: maxfwd [mf_funcs.c:85]: value = 70 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: maxfwd [maxfwd.c:161]: value 70 decreased to 16 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=488 a=26 n=xlog Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: INFO: <script>: apres REQINIT Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=491 a=5 n=route Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=946 a=2 n=return Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=492 a=26 n=xlog Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: INFO: <script>: apres NATDETECT Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=495 a=5 n=route Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=624 a=26 n=xlog Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: INFO: <script>: Dans WITHINDLG Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=671 a=16 n=if Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=625 a=24 n=has_totag Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: siputils [checks.c:103]: no totag Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=496 a=26 n=xlog Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: INFO: <script>: apres WITHINDLG Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=508 a=16 n=if Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=501 a=25 n=is_method Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=508 a=24 n=t_check_trans Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: tm [t_lookup.c:1095]: DEBUG: t_check_msg: msg id=328 global id=327 T start=(nil) Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [parser/msg_parser.c:204]: DEBUG: get_hdr_body : content_length=0 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [parser/msg_parser.c:106]: found end of header Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: tm [t_lookup.c:534]: t_lookup_request: start searching: hash=27620, isACK=0 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: tm [t_lookup.c:492]: DEBUG: RFC3261 transaction matching failed Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: tm [t_lookup.c:716]: DEBUG: t_lookup_request: no transaction found Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: tm [t_lookup.c:1164]: DEBUG: t_check_msg: msg id=328 global id=328 T end=(nil) Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=509 a=26 n=xlog Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: INFO: <script>: apres t_check_trans Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=512 a=5 n=route Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=928 a=16 n=if Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=795 a=25 n=is_method Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=863 a=16 n=if Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=817 a=25 n=is_present_hf Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=821 a=26 n=xlog Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [parser/parse_to.c:176]: DEBUG: add_param: tag=1120136754 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [parser/parse_to.c:799]: end of header reached, state=29 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: INFO: <script>: ldap://sipaccounts/OU=_Usuarios,DC=redcoto,DC=com,DC=ar?sAMAccountName,userPassword?sub?(sAMAccountName=sbergerot)) Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=844 a=16 n=if Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=824 a=25 n=ldap_search Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: ldap [ldap_api_fn.c:273]: LDAP URL parsed into session_name [sipaccounts], base [OU=_Usuarios,DC=redcoto,DC=com,DC=ar], scope [2], filter [(sAMAccountName=sbergerot)] Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: ldap [ldap_api_fn.c:433]: [sipaccounts]: performing LDAP search: dn [OU=_Usuarios,DC=redcoto,DC=com,DC=ar], scope [2], filter [(sAMAccountName=sbergerot)], client_timeout [5000000] usecs Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: ldap [ldap_api_fn.c:479]: [sipaccounts]: ldap_search_ext_st failed: Can't contact LDAP server Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: ldap [ldap_connect.c:231]: [sipaccounts]: LDAP bind successful (ldap_host [ldap://172.16.3.51]) Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: ldap [ldap_connect.c:278]: [sipaccounts]: LDAP reconnect successful Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: ldap [ldap_api_fn.c:433]: [sipaccounts]: performing LDAP search: dn [OU=_Usuarios,DC=redcoto,DC=com,DC=ar], scope [2], filter [(sAMAccountName=sbergerot)], client_timeout [5000000] usecs Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: ldap [ldap_api_fn.c:240]: [sipaccounts]: [1] LDAP entries found Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=844 a=25 n=ldap_result Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=848 a=26 n=xlog Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: INFO: <script>: ldap_search: found [1] entries for (uid=sbergerot) Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=849 a=25 n=xlog Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: <script>: Username=sbergerot Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=856 a=16 n=if Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=851 a=27 n=pv_www_authenticate Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: <core> [sr_module.c:1657]: Could not convert PV to str Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: auth [auth_mod.c:636]: failed to get passwd value Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=852 a=26 n=xlog Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: INFO: <script>: ldap pv_authenticate failed Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=853 a=26 n=www_challenge Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: auth [challenge.c:127]: build_challenge_hf: realm='128.2.101.19' Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: auth [challenge.c:138]: build_challenge_hf: qop='auth' Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: auth [challenge.c:269]: auth: 'WWW-Authenticate: Digest realm="128.2.101.19", nonce="UW6DllFugmoxy+OjSDvSKiZswnyYapMe", qop="auth"#015#012' Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: sl [sl.c:289]: reply in stateless mode (sl) Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [msg_translator.c:206]: check_via_address(172.30.203.182, 172.30.203.182, 0) Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [tcp_main.c:2317]: tcp_send: send from reader (3064 (8)), reusing fd Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [tcp_main.c:2553]: tcp_send: sending... Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [tcp_main.c:2587]: tcp_send: after real write: c= 0xb34ed968 n=429 fd=8 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [tcp_main.c:2588]: tcp_send: buf=#012SIP/2.0 401 Unauthorized#015#012Via: SIP/2.0/TCP 172.30.203.182:25204;rport=49734;branch=z9hG4bK1107819308#015#012From: <sip:sbergerot@128.2.101.19>;tag=1120136754#015#012To: < sip:sbergerot@128.2.101.19>;tag=1e63bc776842177dff382010e03d32cf.8290#015#012Call-ID: 1585139518#015#012CSeq: 2 REGISTER#015#012WWW-Authenticate: Digest realm="128.2.101.19", nonce="UW6DllFugmoxy+OjSDvSKiZswnyYapMe", qop="auth"#015#012Server: kamailio (4.0.0 (i386/linux))#015#012Content-Length: 0#015#012#015#012 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=854 a=2 n=exit Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list 0xb34ed928 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [xavp.c:447]: destroying xavp list (nil) Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [receive.c:293]: receive_msg: cleaning up Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [tcp_read.c:1311]: tcp_read_req: content-length= 0 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [parser/msg_parser.c:623]: SIP Request: Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [parser/msg_parser.c:625]: method: <REGISTER> Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [parser/msg_parser.c:627]: uri: <sip:128.2.101.19> Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [parser/msg_parser.c:629]: version: <SIP/2.0> Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [parser/parse_via.c:1284]: Found param type 235, <rport> = <n/a>; state=6 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [parser/parse_via.c:1284]: Found param type 232, <branch> = <z9hG4bK294428223>; state=16 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [parser/parse_via.c:2672]: end of header reached, state=5 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [parser/msg_parser.c:513]: parse_headers: Via found, flags=2 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [parser/msg_parser.c:515]: parse_headers: this is the first via Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [receive.c:149]: After parse_msg... Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [receive.c:190]: preparing to run routing scripts... Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=487 a=5 n=route Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=620 a=16 n=if Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=610 a=25 n=mf_process_maxfwd_header Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [parser/parse_to.c:799]: end of header reached, state=10 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [parser/msg_parser.c:190]: DEBUG: get_hdr_field: <To> [30]; uri=[ sip:sbergerot@128.2.101.19] Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [parser/msg_parser.c:192]: DEBUG: to body [<sip:sbergerot@128.2.101.19 >#015#012] Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [parser/msg_parser.c:170]: get_hdr_field: cseq <CSeq>: <3> <REGISTER> Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: maxfwd [mf_funcs.c:85]: value = 70 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: maxfwd [maxfwd.c:161]: value 70 decreased to 16 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=488 a=26 n=xlog Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: INFO: <script>: apres REQINIT Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=491 a=5 n=route Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=946 a=2 n=return Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=492 a=26 n=xlog Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: INFO: <script>: apres NATDETECT Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=495 a=5 n=route Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=624 a=26 n=xlog Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: INFO: <script>: Dans WITHINDLG Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=671 a=16 n=if Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=625 a=24 n=has_totag

1 0

Typecast
by Victor V. Kustov 17 Apr '13

17 Apr '13

Hi all. I've got following construction: modparam("acc_radius", "radius_extra", " \ Calling-Station-Id=$tU; \ User-Name=$rU; \ Called-Station-Id=$fU; \ Framed-IP-Address=$si; \ h323-call-origin=$var(ho); \ h323-disconnect-cause=$var(hdc); \ h323-connect-time=$var(ct); \ h323-disconnect-time=$var(dt); \ h323-setup-time=$var(st); \ Cisco-AVPair=$var(cavp); \ Cisco-Call-Type=$var(cct) \ ") Framed-IP-Address in dictionary described as "ipaddr". If i send $si (as writen above), then radius server get incorrect ip like 46.48.44.49. If I try to assign $si any attribute has string type - all ok. How can I typecast here? Simulary problem for attributes "h323-disconnect-time", "h323-setup-time" - need timestamp in format "0000-00-00 00:00:00" -- SY, Victor JID: coyote(a)bks.tv JID: coyote(a)bryansktel.ru I use FREE operation system: 3.8.4-calculate GNU/Linux

2 3

Buffer size with xmlrpc module
by Jon Bergli Heier 17 Apr '13

17 Apr '13

Hello, we use the xmlrpc module to fetch the list of users with the ul_dump MI command. When the number of users increased, this started to fail as the output size grew. Apparently this is because of too small write queue sizes (tcp_conn_wq_max). I discussed increasing the tcp_conn_wq_max setting with a coworker, who was reluctant to change it as it would affect SIP over TCP as well. We would probably also have to increase it if we hit the limit again at a later time. What we would like to know is if there's a way around this without having to fiddle with buffer sizes (or if this behavior is considered a bug). We originally used the mi_xmlrpc module, but changed to the xmlrpc module as the former made kamailio spawn zombie processes on xmlrpc requests (which is considered a bug I guess). FWIW, here's the log from a failed request, running kamailio 3.3.4: Apr 1 23:38:24 server /usr/sbin/kamailio[4982]: DEBUG: <core> [ip_addr.c:247]: tcpconn_new: new tcp connection: [REMOVED] Apr 1 23:38:24 server /usr/sbin/kamailio[4982]: DEBUG: <core> [tcp_main.c:1089]: tcpconn_new: on port 40261, type 2 Apr 1 23:38:24 server /usr/sbin/kamailio[4982]: DEBUG: <core> [tcp_main.c:1403]: tcpconn_add: hashes: 40:3802:373, 326 Apr 1 23:38:24 server /usr/sbin/kamailio[4982]: DEBUG: <core> [io_wait.h:390]: DBG: io_watch_add(0x828660, 94, 2, 0x7f2f87f31ca8), fd_no=85 Apr 1 23:38:24 server /usr/sbin/kamailio[4982]: DEBUG: <core> [io_wait.h:617]: DBG: io_watch_del (0x828660, 94, -1, 0x0) fd_no=86 called Apr 1 23:38:24 server /usr/sbin/kamailio[4982]: DEBUG: <core> [tcp_main.c:4298]: tcp: DBG: sending to child, events 1 Apr 1 23:38:24 server /usr/sbin/kamailio[4982]: DEBUG: <core> [tcp_main.c:3969]: selected tcp worker 5 58(4960) for activity on [tcp:0.0.0.0:8080], 0x7f2f87f31ca8 Apr 1 23:38:24 server /usr/sbin/kamailio[4960]: DEBUG: <core> [tcp_read.c:1358]: received n=8 con=0x7f2f87f31ca8, fd=10 Apr 1 23:38:24 server /usr/sbin/kamailio[4960]: DEBUG: <core> [tcp_read.c:1168]: tcp_read_req: content-length= 153 Apr 1 23:38:24 server /usr/sbin/kamailio[4960]: DEBUG: <core> [parser/msg_parser.c:624]: SIP Request: Apr 1 23:38:24 server /usr/sbin/kamailio[4960]: DEBUG: <core> [parser/msg_parser.c:626]: method: <POST> Apr 1 23:38:24 server /usr/sbin/kamailio[4960]: DEBUG: <core> [parser/msg_parser.c:628]: uri: </RPC2> Apr 1 23:38:24 server /usr/sbin/kamailio[4960]: DEBUG: <core> [parser/msg_parser.c:630]: version: <HTTP/1.0> Apr 1 23:38:24 server /usr/sbin/kamailio[4960]: DEBUG: <core> [parser/msg_parser.c:202]: DEBUG: get_hdr_body : content_length=153 Apr 1 23:38:24 server /usr/sbin/kamailio[4960]: DEBUG: <core> [parser/msg_parser.c:104]: found end of header Apr 1 23:38:24 server /usr/sbin/kamailio[4960]: DEBUG: <core> [receive.c:149]: After parse_msg... Apr 1 23:38:24 server /usr/sbin/kamailio[4960]: DEBUG: xmlrpc [xmlrpc.c:2273]: new fake xml msg created (355 bytes):#012<POST sip:127.0.0.1:9 HTTP/1.0#015#012Via: SIP/2.0/TCP [REMOVED]:40261#015#012Host: [REMOVED]:8080#015#012User-Agent: xmlrpclib.py/1.0.1 (by www.pythonware.com)#015#012Content-Type: text/xml#015#012Content-Length: 153#015#012#015#012<?xml version='1.0'?>#012<methodCall>#012<methodName>mi</methodName>#012<params>#012<param>#012<value><string>ul_dump</string></value>#012</param>#012</params>#012</methodCall>#012> Apr 1 23:38:24 server /usr/sbin/kamailio[4960]: DEBUG: <core> [parser/msg_parser.c:624]: SIP Request: Apr 1 23:38:24 server /usr/sbin/kamailio[4960]: DEBUG: <core> [parser/msg_parser.c:626]: method: <POST> Apr 1 23:38:24 server /usr/sbin/kamailio[4960]: DEBUG: <core> [parser/msg_parser.c:628]: uri: <sip:127.0.0.1:9> Apr 1 23:38:24 server /usr/sbin/kamailio[4960]: DEBUG: <core> [parser/msg_parser.c:630]: version: <HTTP/1.0> Apr 1 23:38:24 server /usr/sbin/kamailio[4960]: DEBUG: <core> [parser/parse_via.c:2561]: end of header reached, state=5 Apr 1 23:38:24 server /usr/sbin/kamailio[4960]: DEBUG: <core> [parser/msg_parser.c:511]: parse_headers: Via found, flags=2 Apr 1 23:38:24 server /usr/sbin/kamailio[4960]: DEBUG: <core> [parser/msg_parser.c:513]: parse_headers: this is the first via Apr 1 23:38:24 server /usr/sbin/kamailio[4960]: DEBUG: <core> [parser/msg_parser.c:202]: DEBUG: get_hdr_body : content_length=153 Apr 1 23:38:24 server /usr/sbin/kamailio[4960]: DEBUG: <core> [parser/msg_parser.c:104]: found end of header Apr 1 23:38:24 server /usr/sbin/kamailio[4960]: DEBUG: <core> [msg_translator.c:206]: check_via_address([REMOVED], [REMOVED], 0) Apr 1 23:38:24 server /usr/sbin/kamailio[4960]: DEBUG: sl [sl_funcs.c:499]: execute callback for event type 1 Apr 1 23:38:24 server /usr/sbin/kamailio[4960]: DEBUG: siptrace [siptrace.c:1343]: trace off... Apr 1 23:38:24 server /usr/sbin/kamailio[4960]: DEBUG: <core> [tcp_main.c:2316]: tcp_send: send from reader (4960 (58)), reusing fd Apr 1 23:38:24 server /usr/sbin/kamailio[4960]: DEBUG: <core> [tcp_main.c:2552]: tcp_send: sending... Apr 1 23:38:24 server /usr/sbin/kamailio[4960]: DEBUG: <core> [tcp_main.c:2586]: tcp_send: after real write: c= 0x7f2f87f31ca8 n=15928 fd=10 Apr 1 23:38:24 server /usr/sbin/kamailio[4960]: DEBUG: <core> [tcp_main.c:2587]: tcp_send: buf=[REMOVED] Apr 1 23:38:24 server /usr/sbin/kamailio[4960]: ERROR: <core> [tcp_main.c:708]: ERROR: wbufq_add(64634 bytes): write queue full or timeout (0, total 0, last write 6648472 s ago) Apr 1 23:38:24 server /usr/sbin/kamailio[4960]: ERROR: sl [../../forward.h:171]: msg_send: ERROR: tcp_send failed Apr 1 23:38:24 server /usr/sbin/kamailio[4960]: ERROR: xmlrpc [xmlrpc.c:804]: Error while sending reply Apr 1 23:38:24 server /usr/sbin/kamailio[4982]: DEBUG: <core> [tcp_main.c:3620]: handle_ser_child: read response= 7f2f87f31ca8, -2, fd -1 from 58 (4960) Apr 1 23:38:24 server /usr/sbin/kamailio[4982]: ERROR: <core> [tcp_main.c:3634]: handle_ser_child: ERROR: received CON_ERROR for 0x7f2f87f31ca8 (id 7494), refcnt 3, flags 0x4018 Apr 1 23:38:24 server /usr/sbin/kamailio[4960]: DEBUG: xmlrpc [xmlrpc.c:2194]: xmlrpc: error while trying script Apr 1 23:38:24 server /usr/sbin/kamailio[4960]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Apr 1 23:38:24 server /usr/sbin/kamailio[4960]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Apr 1 23:38:24 server /usr/sbin/kamailio[4960]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Apr 1 23:38:24 server /usr/sbin/kamailio[4960]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Apr 1 23:38:24 server /usr/sbin/kamailio[4960]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Apr 1 23:38:24 server /usr/sbin/kamailio[4960]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Apr 1 23:38:24 server /usr/sbin/kamailio[4960]: DEBUG: <core> [xavp.c:365]: destroying xavp list (nil) Apr 1 23:38:24 server /usr/sbin/kamailio[4960]: DEBUG: <core> [tcp_read.c:1285]: releasing con 0x7f2f87f31ca8, state -2, fd=10, id=7494 Apr 1 23:38:24 server /usr/sbin/kamailio[4960]: DEBUG: <core> [tcp_read.c:1286]: extra_data (nil) Apr 1 23:38:24 server /usr/sbin/kamailio[4982]: DEBUG: <core> [tcp_main.c:3381]: handle_tcp_child: reader response= 7f2f87f31ca8, -2 from 5 Thanks, Jon

2 1

Max TCP connections
by Krishna Kurapati 16 Apr '13

16 Apr '13

Hi, Has anyone tested max TCP connections that a Kamailio node can handle? I have an i7, 8GB server. I am trying to set max to 64K now. I want to set it to 256K connections. Would that work? Thnx Krish Kura

2 1

Kamailio 4.0.0 not listening on TCP
by David Wilson 15 Apr '13

15 Apr '13

Hi Jeremy, Thanks for the suggestion, I'm hoping I haven't overlooked something quite that obvious! The last debug output I see on startup is: > Listening on > udp: 192.168.44.66:55060 > tcp: 192.168.44.66:55060 > Aliases: > <snip> > > WARNING: no fork mode > 0(20372) INFO: <core> [tcp_main.c:4833]: init_tcp: using epoll_lt as the io watch method (auto detected) > 0(20372) INFO: usrloc [hslot.c:53]: locks array size 512 > 0(20372) INFO: <core> [udp_server.c:179]: INFO: udp_init: SO_RCVBUF is initially 229376 > 0(20372) INFO: <core> [udp_server.c:230]: INFO: udp_init: SO_RCVBUF is finally 262142 netstat shows listening for UDP on 55060, but nothing on TCP: $ netstat -l | grep 5060 udp 0 0 192.168.44.66:55060 *:* ps shows four kamailio processes running, all of which end when I Ctrl-C. Cheers, Dave. > Date: Fri, 12 Apr 2013 14:22:45 +0800 > From: Jeremy Ardley <jeremy.ardley(a)gmail.com> > Subject: Re: [SR-Users] Kamailio 4.0.0 not listening on TCP > > > Have you checked to make sure it's still running? > > > On 12/04/13 14:19, David Wilson wrote: >> Hello All, >> >> New to Kamailio, I'm running 4.0 on Ubuntu 12.04 (precise). >> >> Running from the command line with a minimal config, it tells me that it is listening on port 5060 for both TCP and UDP. I see the same behaviour with the default cfg. >> >> However, netstat does not list anything listening on TCP port 5060 and an attempted connection is rejected consistent with there being no open socket. >> >> Any suggestions on where I need to be looking to fix this? >> > > > > > ------------------------------ > > _______________________________________________ > sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > End of sr-users Digest, Vol 95, Issue 47 > ****************************************

5 8

xcap server crash
by Juha Heinanen 13 Apr '13

13 Apr '13

i tested kamailio 4.0 xcap server with sipclients and got crash. should i add a tracker entry? -- juha (gdb) where #0 0x00007fa35ab47764 in ?? () from /lib/x86_64-linux-gnu/libc.so.6 #1 0x00007fa3547ff9e2 in check_match_header (body=..., etag=0x7fff50162170) at xcap_server.c:1625 #2 0x00007fa3547ff88a in check_preconditions (msg=0x7fff501631c0, etag_hdr=...) at xcap_server.c:1583 #3 0x00007fa3547f85ec in w_xcaps_put (msg=0x7fff501631c0, puri=0x7fa35a091470 "\300-\tZ\243\177", ppath=0x7fa35a091510 "\360:\tZ\243\177", pbody=0x7fa35a0915b0 " ;\tZ\243\177") at xcap_server.c:594 #4 0x000000000041bbe0 in do_action (h=0x7fff501630e0, a=0x7fa35a092f50, msg=0x7fff501631c0) at action.c:1092 #5 0x00000000004240a7 in run_actions (h=0x7fff501630e0, a=0x7fa35a092f50, msg=0x7fff501631c0) at action.c:1573 #6 0x0000000000422852 in do_action (h=0x7fff501630e0, a=0x7fa35a0aad20, msg=0x7fff501631c0) at action.c:1321 #7 0x00000000004240a7 in run_actions (h=0x7fff501630e0, a=0x7fa35a0726c0, msg=0x7fff501631c0) at action.c:1573 #8 0x00007fa3545e9355 in xhttp_process_request (orig_msg=0x7fa35a10d0d0, new_buf=0x7fa35a10e390 "PUT /xcap-root/resource-lists/users/sip:jh@test.fi/index HTTP/1.1\r\nVia: SIP/2.0/TCP 192.98.102.10:45208\r\nAccept-Encoding: identity\r\nContent-Length: 996\r\nHost: 192.98.102.10:8080\r\nUser-Agent: sipsimple"..., new_len=1290) at xhttp_mod.c:271 #9 0x00007fa3545e9b8a in xhttp_handler (msg=0x7fa35a10d0d0) at xhttp_mod.c:346 #10 0x0000000000482492 in nonsip_msg_run_hooks (msg=0x7fa35a10d0d0) at nonsip_hooks.c:125 #11 0x00000000004982c7 in receive_msg ( buf=0x7fa351da7e30 "PUT /xcap-root/resource-lists/users/sip:jh@test.fi/index HTTP/1.1\r\nAccept-Encoding: identity\r\nContent-Length: 996\r\nHost: 192.98.102.10:8080\r\nUser-Agent: sipsimple 0.34.0\r\nConnection: close\r\nIf-None-Ma"..., len=1252, rcv_info=0x7fa351da7b68) at receive.c:158 #12 0x000000000051f294 in receive_tcp_msg ( tcpbuf=0x7fa351da7e30 "PUT /xcap-root/resource-lists/users/sip:jh@test.fi/index HTTP/1.1\r\nAccept-Encoding: identity\r\nContent-Length: 996\r\nHost: 192.98.102.10:8080\r\nUser-Agent: sipsimple 0.34.0\r\nConnection: close\r\nIf-None-Ma"..., len=1252, rcv_info=0x7fa351da7b68, con=0x7fa351da7b50) at tcp_read.c:1179 #13 0x0000000000520271 in tcp_read_req (con=0x7fa351da7b50, bytes_read=0x7fff50163c20, read_flags=0x7fff50163c18) at tcp_read.c:1384 #14 0x0000000000521aa4 in handle_io (fm=0x7fa35a0fa980, events=1, idx=-1) at tcp_read.c:1614 #15 0x000000000051b0ae in io_wait_loop_epoll (h=0x8c27e0, t=2, repeat=0) at io_wait.h:1092 #16 0x0000000000522341 in tcp_receive_loop (unix_sock=20) at tcp_read.c:1725 #17 0x0000000000515e3d in tcp_init_children () at tcp_main.c:4956 #18 0x00000000004658f9 in main_loop () at main.c:1723 #19 0x0000000000468594 in main (argc=18, argv=0x7fff501640b8) at main.c:2566

2 5

Authentcation with RADIUS on kamailio
by Juraj Dančík 12 Apr '13

12 Apr '13

Hi, I have troble with authentication on Kamailio using RADIUS authentication. For configure freeradius and radiusclient I used this tutorial: http://www.kamailio.org/docs/openser-radius-1.0.x.html In config file of freeradius I have enabled digest in modules. In sites-available/default is digest enabled too. In config file of radiusclient I use for authentication server named localhost and in file servers I have password for localhost. I think, in freeradius and radiusclient config file, there is nothing wrong. When I test configuration of freeradius and radiusclient with radclient, there is no problem. Access is accepted. But when I want to authenticate with SIP Client (I use Jitsi), all registration are accepted. It doesn't matter what username and password I write. Problem is probably in config file of Kamailio. I don't know for what are dictionary files exactly used. I include dictionary.kamailio in freeradius's dictionary and radiusclient's dicitionary too. I attached The content of this dictionary file. When I start Kamailio in debug mode I can see any record about authentication or radius. Can anyone help me? Thanks for reply. [1622]: DEBUG: <core> [parser/msg_parser.c:623]: SIP Request: [1622]: DEBUG: <core> [parser/msg_parser.c:625]: method: <REGISTER> [1622]: DEBUG: <core> [parser/msg_parser.c:627]: uri: <sip:192.168.0.112> [1622]: DEBUG: <core> [parser/msg_parser.c:629]: version: <SIP/2.0> [1622]: DEBUG: <core> [parser/msg_parser.c:170]: get_hdr_field: cseq <CSeq>: <1> <REGISTER> [1622]: DEBUG: <core> [parser/parse_to.c:799]: end of header reached, state=10 [1622]: DEBUG: <core> [parser/msg_parser.c:190]: DEBUG: get_hdr_field: <To> [31]; uri=[sip:fas@192.168.0.11 [1622]: DEBUG: <core> [parser/msg_parser.c:192]: DEBUG: to body ["fas" <sip:fas@192.168.0.112>#015#012] [1622]: DEBUG: <core> [parser/parse_via.c:1284]: Found param type 232, <branch> = <z9hG4bK-383438-2e2d7047b [1622]: DEBUG: <core> [parser/parse_via.c:2672]: end of header reached, state=5 [1622]: DEBUG: <core> [parser/msg_parser.c:513]: parse_headers: Via found, flags=2 [1622]: DEBUG: <core> [parser/msg_parser.c:515]: parse_headers: this is the first via [1622]: DEBUG: <core> [receive.c:149]: After parse_msg... [1622]: DEBUG: <core> [receive.c:190]: preparing to run routing scripts... [1622]: DEBUG: maxfwd [mf_funcs.c:85]: value = 70 [1622]: DEBUG: maxfwd [maxfwd.c:161]: value 70 decreased to 16 [1622]: DEBUG: <core> [parser/msg_parser.c:204]: DEBUG: get_hdr_body : content_length=0 [1622]: DEBUG: <core> [parser/msg_parser.c:106]: found end of header [1622]: DEBUG: <core> [parser/parse_to.c:176]: DEBUG: add_param: tag=2148579d [1622]: DEBUG: <core> [parser/parse_to.c:799]: end of header reached, state=29 [1622]: DEBUG: sanity [mod_sanity.c:255]: sanity checks result: 1 [1622]: DEBUG: siputils [checks.c:103]: no totag [1622]: DEBUG: tm [t_lookup.c:1095]: DEBUG: t_check_msg: msg id=1 global id=0 T start=0xffffffffffffffff [1622]: DEBUG: tm [t_lookup.c:534]: t_lookup_request: start searching: hash=18808, isACK=0 [1622]: DEBUG: tm [t_lookup.c:492]: DEBUG: RFC3261 transaction matching failed 1622]: DEBUG: tm [t_lookup.c:716]: DEBUG: t_lookup_request: no transaction found [1622]: DEBUG: tm [t_lookup.c:1164]: DEBUG: t_check_msg: msg id=1 global id=1 T end=(nil) [1622]: DEBUG: <core> [socket_info.c:589]: grep_sock_info - checking if host==us: 13==9 && [192.168.0.112] [1622]: DEBUG: <core> [socket_info.c:593]: grep_sock_info - checking if port 5060 (advertise 0) matches por [1622]: DEBUG: <core> [socket_info.c:589]: grep_sock_info - checking if host==us: 13==13 && [192.168.0.112] [1622]: DEBUG: <core> [socket_info.c:593]: grep_sock_info - checking if port 5060 (advertise 0) matches por [1622]: DEBUG: <core> [sruid.c:176]: new sruid is [uloc-51686c8e-656-1] (1 / 19) [1622]: DEBUG: registrar [reply.c:368]: created Contact HF: Contact: <sip:fas@192.168.0.100:5060;transport= [1622]: DEBUG: sl [sl.c:289]: reply in stateless mode (sl) [1622]: DEBUG: <core> [msg_translator.c:206]: check_via_address(192.168.0.100, 192.168.0.100, 0) [1622]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) [1622]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) [1622]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) [1622]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) [1622]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) [1622]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) [1622]: DEBUG: <core> [xavp.c:447]: destroying xavp list (nil) [1622]: DEBUG: <core> [receive.c:293]: receive_msg: cleaning up #### Attributes ### ATTRIBUTE Sip-Method 101 integer # Schulzrinne, acc ATTRIBUTE Sip-Response-Code 102 integer # Schulzrinne, acc ATTRIBUTE Sip-Cseq 103 string # Schulzrinne, acc ATTRIBUTE Sip-To-Tag 104 string # Schulzrinne, acc ATTRIBUTE Sip-From-Tag 105 string # Schulzrinne, acc ATTRIBUTE Sip-Translated-Request-URI 107 string # Proprietary, acc ATTRIBUTE Sip-Src-IP 108 string # Proprietary, acc ATTRIBUTE Sip-Src-Port 109 string # Proprietary, acc ATTRIBUTE Digest-Response 206 string # Sterman, auth_radius ATTRIBUTE Sip-Uri-User 208 string # Proprietary, auth_radius ATTRIBUTE Sip-Group 211 string # Proprietary, group_radius ATTRIBUTE Sip-Rpid 213 string # Proprietary, auth_radius ATTRIBUTE SIP-AVP 225 string # Proprietary, avp_radius ATTRIBUTE Digest-Realm 1063 string # Sterman, auth_radius ATTRIBUTE Digest-Nonce 1064 string # Sterman, auth_radius ATTRIBUTE Digest-Method 1065 string # Sterman, auth_radius ATTRIBUTE Digest-URI 1066 string # Sterman, auth_radius ATTRIBUTE Digest-QOP 1067 string # Sterman, auth_radius ATTRIBUTE Digest-Algorithm 1068 string # Sterman, auth_radius ATTRIBUTE Digest-Body-Digest 1069 string # Sterman, auth_radius ATTRIBUTE Digest-CNonce 1070 string # Sterman, auth_radius ATTRIBUTE Digest-Nonce-Count 1071 string # Sterman, auth_radius ATTRIBUTE Digest-User-Name 1072 string # Sterman, auth_radius ### Acct-Status-Type Values ### VALUE Acct-Status-Type Failed 15 # RFC2866, acc ### Service-Type Values ### VALUE Service-Type Call-Check 10 # RFC2865, uri_radius VALUE Service-Type Group-Check 12 # Proprietary, group_radius VALUE Service-Type Sip-Session 15 # Schulzrinne, acc, auth_radius VALUE Service-Type SIP-Caller-AVPs 30 # Proprietary, avp_radius VALUE Service-Type SIP-Callee-AVPs 31 # Proprietary, avp_radius ### Sip-Method Values ### VALUE Sip-Method Undefined 0 VALUE Sip-Method Invite 1 VALUE Sip-Method Cancel 2 VALUE Sip-Method Ack 4 VALUE Sip-Method Bye 8 VALUE Sip-Method Info 16 VALUE Sip-Method Options 32 VALUE Sip-Method Update 64 VALUE Sip-Method Register 128 VALUE Sip-Method Message 256 VALUE Sip-Method Subscribe 512 VALUE Sip-Method Notify 1024 VALUE Sip-Method Prack 2048 VALUE Sip-Method Refer 4096 VALUE Sip-Method Other 8192 VALUE Sip-Method INVITE 1 # Proprietary, acc VALUE Sip-Method CANCEL 2 # Proprietary, acc VALUE Sip-Method ACK 4 # Proprietary, acc VALUE Sip-Method BYE 8 # Proprietary, acc ######Kamailio config file############## loadmodule "auth.so" loadmodule "auth_radius" loadmodule "acc_radius" loadmodule "misc_radius" # -- auth_radius params -- modparam("auth_radius", "radius_config", "/usr/local/etc/radiusclient-ng/radiusclient.conf") modparam("auth_radius", "service_type", 15) # Authentication route route[AUTH] { #!ifdef WITH_AUTH #!ifdef WITH_IPAUTH if((!is_method("REGISTER")) && allow_source_address()) { # source IP allowed return; } #!endif if (is_method("REGISTER")) { # authenticate requests if (!radius_www_authorize("$td") { www_challenge("$td", "0"); exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) consume_credentials(); } # if caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (from_uri!=myself && uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!endif return; }

1 0

Kamailio 4.0.0 not listening on TCP
by David Wilson 12 Apr '13

12 Apr '13

Hello All, New to Kamailio, I'm running 4.0 on Ubuntu 12.04 (precise). Running from the command line with a minimal config, it tells me that it is listening on port 5060 for both TCP and UDP. I see the same behaviour with the default cfg. However, netstat does not list anything listening on TCP port 5060 and an attempted connection is rejected consistent with there being no open socket. Any suggestions on where I need to be looking to fix this? Regards, Dave.

2 1

Kamailio + websockets to jssip
by Glyn Lewis 11 Apr '13

11 Apr '13

I am looking for Kamailio consultants in Vancouver BC area to assist with install and setup of system on debian. Many thanks for any pointers or recommendations. Glyn

2 2

Radius - again.
by Victor V. Kustov 11 Apr '13

11 Apr '13

Good day. Need tips for radius auth/accounting. At fist, if I understand correctly, some changes here: modparam("acc", "service_type", 15) modparam("acc", "radius_extra", "Sip-Src-IP=$si;Sip-Src-Port=$sp") must be replaced to modparam("acc_radius", "service_type", 15) modparam("acc_radius", "radius_extra", "Sip-Src-IP=$si;Sip-Src-Port=$sp") But... *Any* radius_extra parameters raise parsing error when K. start. And second problem - setflag(FLT_ACC) doesn't call query to radius server, acc_rad_request("Some comment"); - does. How it work in real life? -- ST, Victor JID: coyote(a)bks.tv JID: coyote(a)bryansktel.ru I use FREE operation system: 3.8.4-calculate GNU/Linux

2 7

Radius: extra fields
by Victor V. Kustov 11 Apr '13

11 Apr '13

Good day! I had in config: modparam("acc_radius", "radius_extra", "User-Name=XXX") and get same error when start kamailio: ERROR: acc [acc_extra.c:192]: parse failed in <User-Name=XXX> around position 10 Any attributes from dictionary anyway got same message. -- SY, Victor JID: coyote(a)bks.tv JID: coyote(a)bryansktel.ru I use FREE operation system: 3.8.4-calculate GNU/Linux

2 2

Kamalio report 404 not found upon INVITE ?
by Khoa Pham 11 Apr '13

11 Apr '13

Hi, I have my SIP client registered to Kamailio successfully. I've check the location table in Kamailio and assert this successful registration. But when another SIP client tries to make call to me, it receive 404 not found from Kamailio. What cause this problem ? Please help -- Khoa Pham HCMC University of Science Faculty of Information Technology

2 1

path uri problem
by Juha Heinanen 10 Apr '13

10 Apr '13

it is not my lucky day. anything new that i try, does not work. in proxy 1, i call add_path_received() on register request, which results into this kind of path header: Path: <sip:192.98.102.10;transport=tcp;lr;received='sip:192.98.102.10:58022;transport=tcp'>. then i forward the request to proxy 2 and call save() on it. it results in error: Apr 7 14:01:44 wheezy1 /usr/sbin/sip-proxy[8709]: ERROR: registrar [save.c:887]: Failed to parse Path: URI looks like kamailio is not able to parse path uri that it added itself. perhaps the quotes should be double ones? -- juha

10 42

mediaproxy-ng Tutorial
by Aft nix 10 Apr '13

10 Apr '13

Hi, I stumbled upon this git://github.com/sipwise/mediaproxy-ng.git which looked very neat to me. Its said that it can be used with kamailio. It seems like its backed sipwise inc. But no documentation is given there. Anyone know of a tutorial/documentation for how to use it? -- -aft

11 29

Diversion header authentication
by phillman25 10 Apr '13

10 Apr '13

Thank you Daniel and Morten for your assistance and prompt reply. To use the tobody transformation, i see that i would need to upgrade to 4.1 right? Im currently on 3.3. I tried the below code: $var(i)=0; while($(hdr(Diversion)[$var(i)]) != $null ) { $avp(s:divhdr) = $(hdr(Diversion)[$var(i)]); avp_subst("$avp(s:divhdr)", "/.*sip:(.*)(@.*)/\1/"); xlog("L_WARN", "$avp(s:divhdr)"); if (!is_user_in("$avp(s:divhdr)", "7")) { sl_send_reply("403", "NOT ALLOWED"); exit; }; $var(i) = $var(i) +1; } However, it seems like group module cant parse the output 313 as seen below: Apr 10 13:16:43 SipProxy-Test /usr/local/sbin/kamailio[7471]: WARNING: <script>: 313 Apr 10 13:16:43 SipProxy-Test /usr/local/sbin/kamailio[7471]: ERROR: group [group.c:114]: failed to parse URI <313> Apr 10 13:16:43 SipProxy-Test /usr/local/sbin/kamailio[7471]: ERROR: group [group.c:158]: failed to get username@domain A bigger issue is that a certain client is sending more than one diversion header with different format as seen below: Diversion: <tel:22030009>;reason=no-answer;screen=no;privacy=off Diversion: "Solonas A" <sip:16@10.10.10.22>;reason=unconditional So in this case i cant really know how to extract the diversion number using a static substitution. Is there a way to adapt to different formats to extract the diversion number? Thanking you in advance Phillip Date: Tue, 09 Apr 2013 16:43:56 +0200 From: Daniel-Constantin Mierla <miconda(a)gmail.com> Subject: Re: [SR-Users] Diversion header authentication To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org> Message-ID: <5164292C.4080408(a)gmail.com> Content-Type: text/plain; charset="iso-8859-1"; Format="flowed" Hello, just adding that the tobody transformation could be handy to extract the user or uri part of a Diversion header, not to fight with subst expressions: - http://www.kamailio.org/wiki/cookbooks/devel/transformations#to-body_transf… Cheers, Daniel On 4/9/13 3:16 PM, Morten Isaksen wrote: > Hi, > > I have not tested this, but try: > > $avp(s:divhdr) = $(hdr(Diversion)[$var(i)]); > avp_subst("$avp(s:divhdr)", "/.*sip:\+45(.*)(@.*)/\1/"); # Extract > number between +45 and @ > > if (is_user_in("$avp(s:divhdr)", "1") { ... } > > Please note that there can be more than one Diverseion header. In that > case you can use: > > $var(i)=0; > while($(hdr(Diversion)[$var(i)]) != $null ) { > $avp(s:divhdr) = > $(hdr(Diversion)[$var(i)]); > xlog("L_WARN", "$avp(s:divhdr)"); > > > $var(i) = $var(i) +1; > } > > > > /Morten

2 1

too many hope SIP issue with "Route" header in proxy server
by Youngjin Park 10 Apr '13

10 Apr '13

Hi, I have a problem with "Route" header in my proxy server. Incoming invite has "Route" header with the IP of the proxy. So, the proxied invite from proxy server is looping back to the proxy server. I believe that "Route" cannot be removed or deleted because it is one of system header. Is there any way to avoid this loop? Thanks Young

2 1

PLEASE HELP
by Bogdan Denysiuk 10 Apr '13

10 Apr '13

Hi Guys! i need configure kamailio 4.0 with Radius Could you please help me how yo do this? Send me example of config routes. thanks

3 2

Operator !~
by Jon Bonilla 09 Apr '13

09 Apr '13

Hi all I was checking for an operator not matching regular expression. Acording to the doc we have =~ and !~ for matching and not matching. http://www.kamailio.org/wiki/cookbooks/3.3.x/core#script_statements I tried this one: if(!is_domain_local("$rd") && $rd !~ "^.+\.local$") Resulting on these errors: root@sp1:~# /etc/init.d/kamailio-proxy restart Not executing command for kamailio-proxy: invalid configuration file! -e 0(1813) : <core> [cfg.y:3549]: parse error in config file /etc/kamailio/proxy/proxy.cfg, line 1067, column 36: syntax error 0(1813) : <core> [cfg.y:3549]: parse error in config file /etc/kamailio/proxy/proxy.cfg, line 1067, column 52: bad expression 0(1813) : <core> [cfg.y:3549]: parse error in config file /etc/kamailio/proxy/proxy.cfg, line 1067, column 52: bad command 0(1813) : <core> [cfg.y:3549]: parse error in config file /etc/kamailio/proxy/proxy.cfg, line 1068, column 2: bad command 0(1813) : <core> [cfg.y:3546]: parse error in config file /etc/kamailio/proxy/proxy.cfg, line 1083, column 2-21: syntax error 0(1813) : <core> [cfg.y:3546]: parse error in config file /etc/kamailio/proxy/proxy.cfg, line 1083, column 2-21: ERROR: bad config file (6 errors) while this one: if(!(is_domain_local("$rd") || $rd =~ "^.+\.local$")) Works perfectly fine. Am I wrong with the usage of !~ operator? cheers, Jon

2 2

Contact header from linphone too long to delete
by Ross Kusler 09 Apr '13

09 Apr '13

Hello, We noticed an issue with a bunch of stale entries in the mysql location table recently - all for the same user. It turns out that if you install linphone 2.0.2 on an iOS device (other versions/platforms might also be affected) with the transport mode set to TCP and the 'Push Notification' feature enabled, the client generates very long contact (258+ character) headers. The contact column in the location table is varchar(255), anything longer than this automatically gets truncated upon insert. When the entry later expires and usrloc tries to remove it, it generates a delete query that matches on the entire contact value and not the truncated version. This causes the delete query to fail and the usrloc record gets left behind in the location table. The (truncated) contact header value from the database: sip:22448462@98.248.37.244:52008;transport=tcp;app-id=org.linphone.phone.pro d;pn-type=apple;pn-tok=AA31DC9642E76E7D7DA8A0E419B210E1A00E8DF43A7A77C8CD295 5291E1BAC85;pn-msg-str=IM_MSG;pn-call-str=IC_MSG;pn-call-snd=ring.caf;pn-msg -snd=msg.caf;line=932257155 The failed delete query:. delete from location where username='XXXXXXXX' AND contact='sip:XXXXXXXX@11.22.333. 44:52000;transport=tcp;app-id=org.linphone.phone.prod;pn-type=apple;pn-tok=A A31DC9642E76E7D7DA8A0E419B210E1A00E8DF43A7A77C8CD2955291E1BAC85;pn-msg-str=I M_MSG;pn-call-str=IC_MSG;pn-call-snd=ring.caf;pn-msg-snd=msg.caf;line=932257 155f64767' AND callid='1962857815'; What do you all think the best solution here is? My intuition is to simply increase the size of this column. If this is the right approach then it probably also makes sense to update the .sql file for new installs. Thoughts? Thank you, Ross

2 1

Diversion header authentication
by phillman25 09 Apr '13

09 Apr '13

Dear List I am currently using the group module to authenticate inbound calls using the From header using the below code: if (!is_user_in("From", "1")) { sl_send_reply("403", "NOT ALLOWED"); exit; }; }; I want to now authenticate the Diversion header, when the call is diverted, the same way as above using the group module how could i proceed with this? thanking you in advance Phillip

3 2

Re: [SR-Users] sr-users Digest, Vol 95, Issue 33
by julian arsanches 09 Apr '13

09 Apr '13

thank you int transformation worked for the dispatcher module now i can define gw dynamically. much appretiate your sugestion. On Tue, Apr 9, 2013 at 5:59 AM, <sr-users-request(a)lists.sip-router.org>wrote: > Send sr-users mailing list submissions to > sr-users(a)lists.sip-router.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > or, via email, send a message with subject or body 'help' to > sr-users-request(a)lists.sip-router.org > > You can reach the person managing the list at > sr-users-owner(a)lists.sip-router.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of sr-users digest..." > > > Today's Topics: > > 1. Help with dispatcher module (julian arsanches) > 2. how to configure presence module with websocket (???? ?????) > 3. Re: Help with dispatcher module (Javi Gallart) > 4. Re: path uri problem (Andrew Pogrebennyk) > 5. Diversion header authentication (phillman25) > 6. Re: path uri problem (Juha Heinanen) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Mon, 8 Apr 2013 18:17:36 -0400 > From: julian arsanches <guandulito.com(a)gmail.com> > Subject: [SR-Users] Help with dispatcher module > To: sr-users(a)lists.sip-router.org > Message-ID: > <CAG0Kfxv6-TV4Ghep= > Wj0BNVYXH0TLR6FW281fQA9WsOKGsK1Kw(a)mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > thanks for all support from the forum before hand. > i am having an issue with my setup, i am trying to make the dispatcher > dynamic by asigning a value from a variable, that variable is define on my > dispatcher.lst but i am getting an error, if anyone can help thanks. > > my config on the relay route > first i strip the prefix from my dial string, and set variable y with that > code, then i strip it and dial it. but it does not work. > again thanks if anyone can helpme. > > > > xlog(" anrtes de strip >> el r-uri($ru)>($ru{s.substr,0,3})\n"); > > > $var(x) = $ru; #>> set varx as an string > > $var(y)=$(var(x){s.substr,4,3}); #>>>>strip the first 3 digits to find my > route > > > > xlog("dsroute $var(y) my test ($ru)\n"); #check for variables loaded > > strip(3); # strip the prefix to route to destinatio > > xlog("despues de strip time [$Tf] method ($rm) r-uri ($ru) 2nd via el code > ($var(y)\n"); #check for loaded variables after strip > > > ds_select_dst("$var(y)", "4");#>>carrer dynamic don't work variable is > load propperly but i get the following error > * > * > *##Apr 8 17:59:32 ip-10-0-1-206 /usr/sbin/kamailio[8975]: ERROR: <script>: > despues de strip time [Mon Apr 8 17:59:32 2013] method (ACK) r-uri > (sip:54.236.192.19) 2nd via el code (54). >>> i see a the right code in the > variable* > *##Apr 8 17:59:32 ip-10-0-1-206 /usr/sbin/kamailio[8975]: ERROR: <core> > [sr_module.c:1726]: Could not convert PV to int * > > > #ds_select_dst("097", "4");#statically define works > > xlog("print el destination y el $var(y) ($du)>>($td)>>> ($ru)\n"); > > if (!t_relay_to_udp()) { > > sl_reply_error(); > }; > > > exit; > } > > > > dispatcher.lst > > #lineformat > #setit(integer)destination(sipuri)flags(integer,optional) > > > #Asterisks > > 007 sip:1.1.1.1:5060 ##route 1 > 024 sip:2.2.2.2:5060 ##route 2 > 100 sip:1.1.1.100:5060 ##route 100 > 103 sip:1.1.1.103:5060 ###route 103 >

1 0

Kamailio World Updates - April 9
by Daniel-Constantin Mierla 09 Apr '13

09 Apr '13

Kamailio World Conference is one week away, only few bits to adjust, everything being pretty much in place. The agenda with presentations is available at: - http://conference.kamailio.com/k01/schedule/ Details about the speakers are available at: - http://conference.kamailio.com/k01/speakers/ Besides the two days filled with presentations and demos, the participants have the chance of winning Raspberry Pi devices or Zoiper Biz licenses (SIP softphone - http://www.zoiper.com). A grateful and big thank you for event sponsors and exhibitors that made possible to organize this event: Sipwise, Asipto, Fraunhofer Fokus, Sipgate, IT Center Portugal, NG Voice, Zoiper and Amooma: - http://conference.kamailio.com/k01/sponsors/ - http://conference.kamailio.com/k01/exhibitors/ The venue page gives now few more details about getting to conference area: - http://conference.kamailio.com/k01/venue/ Looking forward to meeting next week in Berlin! Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

1 0

Help with dispatcher module
by julian arsanches 09 Apr '13

09 Apr '13

thanks for all support from the forum before hand. i am having an issue with my setup, i am trying to make the dispatcher dynamic by asigning a value from a variable, that variable is define on my dispatcher.lst but i am getting an error, if anyone can help thanks. my config on the relay route first i strip the prefix from my dial string, and set variable y with that code, then i strip it and dial it. but it does not work. again thanks if anyone can helpme. xlog(" anrtes de strip >> el r-uri($ru)>($ru{s.substr,0,3})\n"); $var(x) = $ru; #>> set varx as an string $var(y)=$(var(x){s.substr,4,3}); #>>>>strip the first 3 digits to find my route xlog("dsroute $var(y) my test ($ru)\n"); #check for variables loaded strip(3); # strip the prefix to route to destinatio xlog("despues de strip time [$Tf] method ($rm) r-uri ($ru) 2nd via el code ($var(y)\n"); #check for loaded variables after strip ds_select_dst("$var(y)", "4");#>>carrer dynamic don't work variable is load propperly but i get the following error * * *##Apr 8 17:59:32 ip-10-0-1-206 /usr/sbin/kamailio[8975]: ERROR: <script>: despues de strip time [Mon Apr 8 17:59:32 2013] method (ACK) r-uri (sip:54.236.192.19) 2nd via el code (54). >>> i see a the right code in the variable* *##Apr 8 17:59:32 ip-10-0-1-206 /usr/sbin/kamailio[8975]: ERROR: <core> [sr_module.c:1726]: Could not convert PV to int * #ds_select_dst("097", "4");#statically define works xlog("print el destination y el $var(y) ($du)>>($td)>>> ($ru)\n"); if (!t_relay_to_udp()) { sl_reply_error(); }; exit; } dispatcher.lst #lineformat #setit(integer)destination(sipuri)flags(integer,optional) #Asterisks 007 sip:1.1.1.1:5060 ##route 1 024 sip:2.2.2.2:5060 ##route 2 100 sip:1.1.1.100:5060 ##route 100 103 sip:1.1.1.103:5060 ###route 103

2 1

how to configure presence module with websocket
by אורן אברהם 08 Apr '13

08 Apr '13

hello. I am tring to SUBSCRIBE for presence status of a user on my kamailio server using the sipMl5 (over websockets) but kamailio failes. (more exactly the handle_subscribe failes.) I was told previusly by peter dunkly that it is not a bug, just a configuration problem. I am quoting his answer: "This is a configuration problem not a bug in the software. SUBSCRIBEs are dialog forming requests and NOTIFYs (in this scenario) are in-dialog requests. As such you need to use force_rport(), add_contact_alias(), and handle_ruri_alias() with presence dialogs in the same way you do with media session dialogs." How can i acheve this ? thanks in advance...

1 0

loose_route received parameter question
by Juha Heinanen 08 Apr '13

08 Apr '13

in my test, proxy p2 receives initial request from proxy p1, that has this kind of R-URI and Route header: INVITE sip:47089231@192.98.102.10:41073;transport=tcp SIP/2.0. Route: <sip:192.98.102.20:5070;transport=tcp;lr;received='sip:192.98.102.10:49127;transport=tcp'> route header is result of calling lookup in p1 on record: AOR:: jh(a)test.fi Contact:: sip:47089231@192.98.102.10:41073;transport=tcp Q=0.5 Expires:: 114 Callid:: 0ed240f3-dc79-4d65-93ef-3e92d178a892 Cseq:: 12 User-agent:: sipsimple 0.34.0 Received:: sip:192.98.102.10:49127;transport=tcp Path:: <sip:192.98.102.20:5070;transport=tcp;lr;received='sip:192.98.102.10:49127;transport=tcp'> i would expect that loose_route() would set $du to received parameter value, but it does not, because i get to syslog: Apr 8 12:44:54 wheezy1 /usr/sbin/sip-proxy[5723]: INFO: Trying to relay to du <<null>> or ru <sip:47089231@192.98.102.10:41073;transport=tcp> how can i tell p2 to set $du to received uri? do i need to add another route header in p1 that contains received uri? -- juha

2 2

pkg mem problems. Leaking?
by Jon Bonilla 08 Apr '13

08 Apr '13

Hi all I have two different kamailio servers running in a single machine. The first one is a sl proxy and the second one acts as stateful proxy/registrar. From time to time the stateless one goes out of pkg mem. It has 8MB (also tried with 16 in another setup). The sl proxy listens the public interface for external connections and the loopback interface for conecting to the stateful proxy. As I have seen since I installed a mem debug kamailio 3.3.2 version there 6 days ago, the internal socket children have grown on mem usage while the external ones remain more or less stable. As example one uf the external udp receivers uses less than 1MB while one of the internal ones uses almost 6MB. I'm attaching the mem dump of one of those internal udp receiver processes via "ngcp-sercmd lb cfg.set_now_int core mem_dump_pkg 25437" command. I see several tens of thousands of lines like these ones: N address=0x7fd1207383c0 frag=0x7fd120738390 size=56 used=1 alloc'd from <core>: parser/parse_param.c: parse_params(539) start check=f0f0f0f0, end check= c0c0c0c0, abcdefed I have no idea about how to continue debugging this. I'm pasting some more info regarding the process list, mem status and kamailio version/flags. Thanks sp1:~# ngcp-sercmd lb ps 25416 attendant 25429 udp receiver child=0 sock=77.77.77.77:5060 25430 udp receiver child=1 sock=77.77.77.77:5060 25431 udp receiver child=2 sock=77.77.77.77:5060 25432 udp receiver child=3 sock=77.77.77.77:5060 25433 udp receiver child=4 sock=77.77.77.77:5060 25434 udp receiver child=5 sock=77.77.77.77:5060 25435 udp receiver child=6 sock=77.77.77.77:5060 25436 udp receiver child=7 sock=77.77.77.77:5060 25437 udp receiver child=0 sock=127.0.0.1:5060 25438 udp receiver child=1 sock=127.0.0.1:5060 25439 udp receiver child=2 sock=127.0.0.1:5060 25440 udp receiver child=3 sock=127.0.0.1:5060 25441 udp receiver child=4 sock=127.0.0.1:5060 25442 udp receiver child=5 sock=127.0.0.1:5060 25443 udp receiver child=6 sock=127.0.0.1:5060 25444 udp receiver child=7 sock=127.0.0.1:5060 25445 slow timer 25446 timer 25447 ctl handler 25448 MI FIFO 25449 tcp receiver (generic) child=0 25450 tcp receiver (generic) child=1 25451 tcp receiver (generic) child=2 25452 tcp receiver (generic) child=3 25453 tcp receiver (generic) child=4 25454 tcp receiver (generic) child=5 25455 tcp receiver (generic) child=6 25456 tcp receiver (generic) child=7 25457 tcp receiver (tcp:127.0.0.1:5060) child=8 25458 tcp receiver (tcp:127.0.0.1:5060) child=9 25459 tcp main process sp1:~# ngcp-sercmd lb pkg.stats { entry: 0 pid: 25416 rank: 0 used: 901136 free: 15417544 real_used: 1359672 } { entry: 1 pid: 25429 rank: 1 used: 909360 free: 15408648 real_used: 1368568 } { entry: 2 pid: 25430 rank: 2 used: 909400 free: 15408608 real_used: 1368608 } { entry: 3 pid: 25431 rank: 3 used: 909352 free: 15408656 real_used: 1368560 } { entry: 4 pid: 25432 rank: 4 used: 909344 free: 15408664 real_used: 1368552 } { entry: 5 pid: 25433 rank: 5 used: 909392 free: 15408712 real_used: 1368504 } { entry: 6 pid: 25434 rank: 6 used: 909344 free: 15408664 real_used: 1368552 } { entry: 7 pid: 25435 rank: 7 used: 909344 free: 15408664 real_used: 1368552 } { entry: 8 pid: 25436 rank: 8 used: 909352 free: 15408656 real_used: 1368560 } { entry: 9 pid: 25437 rank: 9 used: 5884872 free: 2480784 real_used: 14296432 } { entry: 10 pid: 25438 rank: 10 used: 5935272 free: 2359728 real_used: 14417488 } { entry: 11 pid: 25439 rank: 11 used: 5971616 free: 2305912 real_used: 14471304 } { entry: 12 pid: 25440 rank: 12 used: 5821672 free: 2522672 real_used: 14254544 } { entry: 13 pid: 25441 rank: 13 used: 5945584 free: 2310056 real_used: 14467160 } { entry: 14 pid: 25442 rank: 14 used: 5868760 free: 2495264 real_used: 14281952 } { entry: 15 pid: 25443 rank: 15 used: 5903672 free: 2457952 real_used: 14319264 } { entry: 16 pid: 25444 rank: 16 used: 5899208 free: 2453872 real_used: 14323344 } { entry: 17 pid: 25445 rank: -1 used: 983944 free: 15409512 real_used: 2267696 } { entry: 18 pid: 25446 rank: -1 used: 909168 free: 15409512 real_used: 1367704 } { entry: 19 pid: 25447 rank: -2 used: 983664 free: 15407912 real_used: 2267032 } { entry: 20 pid: 0 rank: 0 used: 918424 free: 15400064 real_used: 1377152 } { entry: 21 pid: 25449 rank: 17 used: 989192 free: 15329200 real_used: 1448016 } { entry: 22 pid: 25450 rank: 18 used: 989192 free: 15329200 real_used: 1448016 } { entry: 23 pid: 25451 rank: 19 used: 989192 free: 15329200 real_used: 1448016 } { entry: 24 pid: 25452 rank: 20 used: 989192 free: 15329200 real_used: 1448016 } { entry: 25 pid: 25453 rank: 21 used: 989192 free: 15329200 real_used: 1448016 } { entry: 26 pid: 25454 rank: 22 used: 989192 free: 15329200 real_used: 1448016 } { entry: 27 pid: 25455 rank: 23 used: 989192 free: 15329200 real_used: 1448016 } { entry: 28 pid: 25456 rank: 24 used: 989192 free: 15329200 real_used: 1448016 } { entry: 29 pid: 25457 rank: 25 used: 989192 free: 15329200 real_used: 1448016 } { entry: 30 pid: 25458 rank: 26 used: 989192 free: 15329200 real_used: 1448016 } { entry: 31 pid: 25459 rank: -4 used: 983240 free: 15334096 real_used: 2266992 } sp1:~# kamailio -V version: kamailio 3.3.2 (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 13:08:06 Oct 31 2012 with gcc 4.4.5

3 3

usrloc + multimaster replication
by Alex Balashov 08 Apr '13

08 Apr '13

I know I'm revisiting a problem that has been discussed in multiple threads from various angles, so I might be rightly accused of laziness in neglecting to research them all. All the same: I have proxy1 and proxy2 writing to database A and database B, respectively. Database A and database B are active-active masters, synchronised via some replication system attached to the underlying DB technology. The 'location' table is also replicated this way. We know that 'usrloc' doesn't work so well with this: one instance of Kamailio will periodically delete the other's contacts, even if they have a nonlocal SIP domain. Is there any db_mode that can be used (other than 0/purely in-memory) to make this work right? Or is that the essential problem that p_usrloc is written to solve? Thanks, -- Alex -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 United States Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/

4 12

problem with core forward function
by Juha Heinanen 08 Apr '13

08 Apr '13

now that the send function issue got resolved, i tried core forward function. core cookbook tells: forward Forward the SIP request to the given destination in stateless mode. This has the format of [proto:]host[:port]. Host can be an IP or hostname; supported protocols are UDP, TCP and TLS. (For TLS, you need to compile the TLS support into core). If proto or port are not specified, NAPTR and SRV lookups will be used to determine them (if possible). If destination parameter is missing, the forward will be done based on RURI. Example of usage: forward("10.0.0.10:5060"); #or forward(); i then went i tried forward("tcp:192.98.102.20:5070"); but got errors from that line: Apr 7 13:52:10 siika /usr/sbin/sip-proxy[22923]: : <core> [proxy.c:278]: ERROR: mk_proxy: could not resolve hostname: "tcp:192.98.102.20:5070" Apr 7 13:52:10 siika /usr/sbin/sip-proxy[22923]: ERROR: <core> [route.c:1214]: fixing failed (code=-478) at cfg:/etc/sip-proxy/sip-proxy.cfg:1584 i also tried with forward_tcp("192.98.102.20:5070"); but it didn't help either: Apr 7 13:54:24 siika /usr/sbin/sip-proxy[23056]: : <core> [proxy.c:278]: ERROR: mk_proxy: could not resolve hostname: "192.98.102.20:5070" Apr 7 13:54:24 siika /usr/sbin/sip-proxy[23056]: ERROR: <core> [route.c:1214]: fixing failed (code=-478) at cfg:/etc/sip-proxy/sip-proxy.cfg:1584 how should core cookbook be fixed this time? -- juha

2 2

expired record not removed from location table
by Juha Heinanen 08 Apr '13

08 Apr '13

i have seen emails about expired records not removed from database, but i could not find the thread. i'm not sure if it is related, but today i noticed too that there is an old record in my location table which has expired couple of hours ago. the record does not show up with ul_dump command, so it has been removed from memory. usrloc db_mode=1. what could cause the record not be removed? is it a bug in cleaner process? i have usrloc timer_procs=2. -- juha

2 3

textops warning
by Juha Heinanen 07 Apr '13

07 Apr '13

i don't remember seeing this before: Apr 7 19:09:57 siika /usr/sbin/sip-proxy[8734]: WARNING: textops [textops.c:1816]: free function has not been defined for spve what should i do about it? -- juha

1 1

looks like send core function does not exist
by Juha Heinanen 07 Apr '13

07 Apr '13

i removed corex from loaded modules and added string argument to send: send("192.98.102.20:5070"); that gave error: 0(15695) ERROR: <core> [cfg.y:3415]: cfg. parser: failed to find command send 0(15695) : <core> [cfg.y:3554]: parse error in config file /etc/sip-proxy/sip-proxy.cfg, line 1578, column 34: unknown command, missing loadmodule? according to core cookbook: send Send the original SIP message to a specific destination in stateless mode. No changes are applied to received message, no Via header is added. Host can be an IP or hostname. Used protocol: UDP Parameter is mandatory and has string format. Example of usage: send("10.10.10.10:5070"); does anyone know what is going on? -- juha

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Re: [SR-Users] sip_trace unable to capture after next_branches()
by Ramaseshi reddy kolli 06 Apr '13

06 Apr '13

Hi, I have tried by setting flag only but still no luck,Here is the configuration i am using. # sip trace modparam("siptrace", "trace_to_database", 0) modparam("siptrace", "db_url", "") modparam("siptrace", "duplicate_uri", "sip:homer.com:9060") modparam("siptrace", "duplicate_with_hep", 1) modparam("siptrace", "trace_flag", 22) modparam("siptrace", "trace_on", 1) #HEPv2 == timestamp will be included to HEP header modparam("siptrace", "hep_version", 2) setflag(22); if (!t_relay()) { xlog("L_ERR", "CALL $ci - Relay error\n"); sl_reply_error(); exit; } Thanks Ram

2 1

forward to pseudo var destination
by Juha Heinanen 06 Apr '13

06 Apr '13

looks like forward core function does not accept pseudo var argument. then i found this: I just committed a patch that allow using send() without any parameter: * http://git.sip-router.org/cgi-bin/gitweb.cgi/sip-router/?a=commit;h=8e4db80… if that commit is still valid, it is not reflected by core cookbook: send Send the original SIP message to a specific destination in stateless mode. No changes are applied to received message, no Via header is added. Host can be an IP or hostname. Used protocol: UDP Parameter is mandatory and has string format. Example of usage: send("10.10.10.10:5070"); so which way this is? -- juha

2 2

Changing Domain in To URI
by qmail＠polarismail.com 06 Apr '13

06 Apr '13

Sorry if I repeat myself - I just subscribed properly to the list: > if you don't need to change it back for replies and next requests, try: > > $td = "xyz.com"; > > IIRC, that is available on 3.2.x. If not, either you upgrade or use > replace functions from textops. > > Cheers, > Daniel Thank you for replying. We tried the direct assignment of $td="xyz.com" but the variable doesn't change. We are doing an xlog print right after and an ngrep and we confirmed that the domain part of the URI doesn't change. I also tried a subst() and it's the same problem. I read on the online documentation that the variable is read-only and I figured it can't be changed through these methods. That's why I tried uac_replace_to(). To sum it up, is that variable read-only or not ?

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Re: [SR-Users] sip_trace unable to capture after next_branches()
by Ramaseshi reddy kolli 06 Apr '13

06 Apr '13

hi, i have set the following flag it is still not capturing outbound SIP messages. { setflag(22) sip_trace(); } Daniel-Constantin Mierla<http://www.mail-archive.com/search?l=sr-users@lists.sip-router.org&q=from:%…> Fri, 05 Apr 2013 00:45:16 -0700<http://www.mail-archive.com/search?l=sr-users@lists.sip-router.org&q=date:2…> Hello, On 3/31/13 4:27 PM, Ramaseshi reddy kolli wrote: Hi, I am using sip_trace function after next_branches(), it is not capturing outbound SIP INVITE request can anyone please help me. iirc, you have to set a flag to trace outgoing messages. the sip_trace()function just takes the snapshot of processed messages. Cheers, Daniel

2 1

sip_capture columns
by Grant Bagdasarian 06 Apr '13

06 Apr '13

Hello, I couldn't find any documentation on what the table columns of the sip_capture module are used for. Most of them are straightforward, but I still don't know what the following columns mean and what values to expect: - PID_user - Proto - Family - Rtp_stat - Type - Node Type has for example values 1 and 2. Im guessing 1 is for requests and 2 is for responses. I hope someone could clear this out for me. Thanks, Grant

2 2

No RTP between rtpproxy and smartphone application
by aaaa aaaa 05 Apr '13

05 Apr '13

Hello, I have the following topology. Kamailio as SIP Proxy and Asterisk as B2BUA. In Kamailio I use rtpproxy for NAT users. The problem that I have is the following: -When the CSipSimple is registered the first call is always unsuccesfull. The following calls are always successful. If the application remains idle for a while the next call is unsuccessful again. From traces I saw that everything is OK in signaling part. -When I try with Zoiper and 3CX I have 1/20 successful calls. In any case from Asterisk side these calls can be seen as active but no media. Thanx, Theo

3 2

Changing Domain in To URI
by qmail＠polarismail.com 05 Apr '13

05 Apr '13

Hello, As it happens we run Kamailio 3.2.0 on Debian Lenny which for some reason does not support uac_replace_to()! Anyhoo, we are trying to change the domain part in the To URI. What is the best way to do it ? Thank you!

2 1

sip_trace unable to capture after next_branches()
by Ramaseshi reddy kolli 05 Apr '13

05 Apr '13

Hi, I am using sip_trace function after next_branches(), it is not capturing outbound SIP INVITE request can anyone please help me. Thanks Ram

2 1

Kamailio 4.0 configuration
by Martin Krška 04 Apr '13

04 Apr '13

Hi everyone, I'm trying to configure some basic functions in Kamailio 4.0. Now let's say I have two users, for example Alice and Bob. When Alice adds Bob to her contacts, I need from her to ask Bob for permission to watch his presence. It's probably a trivial question but could anyone tell me what do I need for this to work, like what modules are responsible for this etc. Best Regards, Martin

2 1

db_mode 3 in usrloc module
by Morten Isaksen 04 Apr '13

04 Apr '13

Hi, I have an older installation of Kamailio (3.1.1). It is configured with modparam("usrloc", "db_mode", 2) I tried to change this to modparam("usrloc", "db_mode", 3) to not have to restart kamailio when I change the alias table directly from SQL. But then kamailio shutsdown after about one second with this error: Mar 26 20:38:55 sip-3-1 /usr/local/sbin/kamailio[11510]: ERROR: <core> [db.c:408]: invalid version 0 for table trusted found, expected 5 (check table structure and table "version") Mar 26 20:38:55 sip-3-1 /usr/local/sbin/kamailio[11510]: ERROR: permissions [trusted.c:250]: error during table version check. Mar 26 20:38:55 sip-3-1 /usr/local/sbin/kamailio[11510]: ERROR: <core> [sr_module.c:832]: init_mod_child(): Error while initializing module permissions (/usr/local/lib/kamailio/modules_k/permissions.so) Mar 26 20:38:55 sip-3-1 /usr/local/sbin/kamailio[11510]: ERROR: <core> [pt.c:481]: ERROR: fork_tcp_process(): init_child failed for process 28, pid 11510, "tcp receiver child=6" Mar 26 20:38:55 sip-3-1 /usr/local/sbin/kamailio[11510]: ERROR: <core> [tcp_main.c:4811]: ERROR: tcp_main: fork failed: Success Mar 26 20:38:55 sip-3-1 /usr/local/sbin/kamailio[11512]: : <core> [pass_fd.c:293]: ERROR: receive_fd: EOF on 44 Mar 26 20:38:55 sip-3-1 /usr/local/sbin/kamailio[11482]: ALERT: <core> [main.c:738]: child process 11510 exited normally, status=255 Mar 26 20:38:55 sip-3-1 /usr/local/sbin/kamailio[11482]: INFO: <core> [main.c:756]: INFO: terminating due to SIGCHLD I have not made any changes to the permissions/trusted module, so I suspect it is some shared database conection problem. Is it a known bug in this version? I would prefer not to upgrade the installation and hopes one of you can point me to a patch I can backport. -- Morten Isaksen

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Problem with expired SIP Etag
by David 04 Apr '13

04 Apr '13

Hello, I have two Kamailio servers. Running 3.3.2. One uses PUA to send PUBLISH to the second which has presence module. When the presence server replies to the PUBLISH, it includes a SIP-ETag and an Expires header. So, in the case I caught on my servers yesterday, the reply had an Expires header of 34. Trouble is, 60 seconds later, when PUA sent the PUBLISH, it had the SIP-ETag in the PUBLISH header even though the entity tag was expired. But RFC3903 says that when the publish expires, the entity tag expires as well. The publish was for a dialog-info and both PUBLISHes were generated using the pua_mi module command pua_publish. I don't set the etag in the pua_publish command, it's kamailio that remembers it. Here is grep pua kamailio.cfg on the first server : loadmodule "pua.so" loadmodule "pua_mi.so" loadmodule "pua_dialoginfo.so" modparam("pua_dialoginfo", "include_localremote", 1) modparam("pua_dialoginfo", "include_tags", 1) modparam("pua_dialoginfo", "include_callid", 0) modparam("pua", "hash_size", 15) modparam("pua", "db_mode", 0) modparam("pua_dialoginfo", "send_publish_flag", 13) modparam("pua", "update_period", 6 ) # I added this to see if the problem would be fixed modparam("pua_dialoginfo", "caller_confirmed", 1) I didn't capture the mi udp packets, but I did get the SIP, see below I think the problem has to do with expires_offset. I set this to reduce SUBSCRIBE expiration by 30 seconds, but it is also reducing PUBLISH expiration by 30 seconds. Do you think this might be the source of my problem? What else might be causing this ? Is there anyway to set the expires_offset for SUBSCRIBE but not for the PUBLISH? Any other ideas what might be going on here? Thanks, David Here is the SIP I captured on the pua server : U 2013/03/29 14:01:41.005025 PUBLISH sip:user161@omnity.net SIP/2.0. Via: SIP/2.0/UDP proxyip;branch=z9hG4bK267e.e988a546.0. To: sip:user161@omnity.net. From: sip:user161@omnity.net;tag=047a42f14f0e800c0a21487d4bc09012-c8fe. CSeq: 10 PUBLISH. Call-ID: 79a5f23d217e3597-9109@proxyip. Content-Length: 0. Max-Forwards: 70. Event: dialog. Expires: 66. SIP-If-Match: a.1363343972.1019.567098.10. U 2013/03/29 14:01:41.298041 SIP/2.0 200 OK. Via: SIP/2.0/UDP proxyip;branch=z9hG4bK267e.e988a546.0. To: sip:user161@mydomain.net;tag=f0253efb8005057b3cc0ab8fa35ff1d9-0dd9. From: sip:user161@mydomain.net;tag=047a42f14f0e800c0a21487d4bc09012-c8fe. CSeq: 10 PUBLISH. Call-ID: 79a5f23d217e3597-9109@proxyip. Expires: 36. SIP-ETag: a.1363343972.1009.565433.11. Content-Length: 0. U 2013/03/29 14:03:14.317797 PUBLISH sip:user161@mydomain.net SIP/2.0. Via: SIP/2.0/UDP proxyip;branch=z9hG4bK543f.efdb2d8.0. To: sip:user161@mydomain.net. From: sip:user161@mydomain.net;tag=047a42f14f0e800c0a21487d4bc09012-fab2. CSeq: 10 PUBLISH. Call-ID: 79a5f23d214a91de-9057@proxyip. Content-Length: 271. Max-Forwards: 70. Event: dialog. Expires: 91. SIP-If-Match: a.1363343972.1009.565433.11. Content-Type: application/dialog-info+xml. . <?xml version='1.0'?> <dialog-info xmlns='urn:ietf:params:xml:ns:dialog-info' version='0' state='full' entity='user161(a)omnity.net'> <dialog id='x_dialogid(a)192.168.2.5' direction='recipient'> <state>early</state> </dialog> </dialog-info> U 2013/03/29 14:03:14.319691 SIP/2.0 412 Conditional request failed. Via: SIP/2.0/UDP proxyip;branch=z9hG4bK543f.efdb2d8.0. To: sip:user161@mydomain.net;tag=f0253efb8005057b3cc0ab8fa35ff1d9-9add. From: sip:user161@mydomain.net;tag=047a42f14f0e800c0a21487d4bc09012-fab2. CSeq: 10 PUBLISH. Call-ID: 79a5f23d214a91de-9057@proxyip. Server: OmniVigil 5.2-rc1. Content-Length: 0.

1 0

Rewrite Prefix in Carrieroute
by Ali Jawad 04 Apr '13

04 Apr '13

Hi I do have a server with kamailio running on carrireroute, recently the need to rewrite prefixes arose, so I did enter the rewrite prefix into the relevant rows/columns in carrierroute tdb table and did reload but nothing changed. The call is matched to it's needed destination but the prefix is the same. In my config I do have the following line if(!cr_route("default", "ata.nymgo.com", "$rU", "$rU", "call_id")){ Regards

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New developer: Torrey Searle
by Daniel-Constantin Mierla 04 Apr '13

04 Apr '13

Hello, I want to announce that a new person got developer GIT write access to repository: Torrey Searle - he is going to add and maintain the sipt module, facilitating operations over SIP-T/SIP-I messages, the patch being on tracker: - http://sip-router.org/tracker/index.php?do=details&task_id=281&project=1 His git commit id is: tsearle My warm welcome and looking forward to future work within the project! Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

2 1

What IP in REGISTER message does Kamalio use to store location ?
by Khoa Pham 04 Apr '13

04 Apr '13

Hi, when client send REGISTER message to Kamalio. Kamailio will save that client's IP to database. But what is this IP ? Is is 1,2 or 3 ? 1. The IP from Via field 2. The IP from Contact field 3. The IP from the IP packet Many thanks -- Khoa Pham HCMC University of Science Faculty of Information Technology

3 6

Abort at qm_debug_frag function
by Bruno Bresciani 03 Apr '13

03 Apr '13

Kamailio is generating a abort at qm_debug_frag function (BUG: qm_*: prev. fragm. tail overwritten) but I can't understanding the meaning this abort... Someone can tell me the purpose of qm_debug_frag function, it checks if some improper memory handling happened and abort the program? Below is backtrace of abort: #0 0x00f41402 in __kernel_vsyscall () #1 0x009fec10 in raise () from /lib/libc.so.6 #2 0x00a00521 in abort () from /lib/libc.so.6 #3 0x081797b7 in qm_debug_frag (qm=0xb5f83000, f=<value optimized out>) at mem/q_malloc.c:158 #4 0x0817a6f3 in qm_free (qm=0xb5f83000, p=0xb6109a54, file=0x43760e "perms_db: src_ip.c", func=0x4378a0 "list_destroy", line=79) at mem/q_malloc.c:442 #5 0x0043563c in list_destroy (liste=0xb6107274) at src_ip.c:79 #6 0x00430a93 in reload_srcip_table () at db.c:163 #7 0x004313be in perms_db_srcip_reload (cmd_tree=0x835aeac, param=0x0) at fifo.c:107 #8 0x00335d97 in mi_fifo_server (fifo_stream=0x99561d8) at ../../lib/kmi/mi.h:77 #9 0x00337621 in fifo_process (rank=1) at mi_fifo.c:235 #10 0x003378a5 in mi_child_init (rank=0) at mi_fifo.c:199 #11 0x08120a02 in init_mod_child (m=0x82f4b88, rank=0) at sr_module.c:829 #12 0x081209dc in init_mod_child (m=0x82f4d1c, rank=0) at sr_module.c:826 #13 0x081209dc in init_mod_child (m=0x82f5164, rank=0) at sr_module.c:826 #14 0x081209dc in init_mod_child (m=0x82f79f4, rank=0) at sr_module.c:826 #15 0x081209dc in init_mod_child (m=0x82f7cf0, rank=0) at sr_module.c:826 #16 0x081209dc in init_mod_child (m=0x82f8174, rank=0) at sr_module.c:826 #17 0x081209dc in init_mod_child (m=0x82f8798, rank=0) at sr_module.c:826 #18 0x081209dc in init_mod_child (m=0x82f8b54, rank=0) at sr_module.c:826 #19 0x081209dc in init_mod_child (m=0x82f8eb4, rank=0) at sr_module.c:826 #20 0x081209dc in init_mod_child (m=0x82f90b4, rank=0) at sr_module.c:826 #21 0x081209dc in init_mod_child (m=0x82f952c, rank=0) at sr_module.c:826 #22 0x081209dc in init_mod_child (m=0x82fa980, rank=0) at sr_module.c:826 #23 0x081209dc in init_mod_child (m=0x82fab9c, rank=0) at sr_module.c:826 #24 0x081209dc in init_mod_child (m=0x82faff8, rank=0) at sr_module.c:826 #25 0x080ae175 in main_loop () at main.c:1624 #26 0x080b1256 in main (argc=3, argv=0xbf9a8f14) at main.c:2398 Best Regards

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Re: [SR-Users] sr-users Digest, Vol 95, Issue 17
by julian arsanches 03 Apr '13

03 Apr '13

Thanks a lot it was just what i need it , it worked like a charm. strip(3); On Wed, Apr 3, 2013 at 3:39 PM, <sr-users-request(a)lists.sip-router.org>wrote: > Send sr-users mailing list submissions to > sr-users(a)lists.sip-router.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > or, via email, send a message with subject or body 'help' to > sr-users-request(a)lists.sip-router.org > > You can reach the person managing the list at > sr-users-owner(a)lists.sip-router.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of sr-users digest..." > > > Today's Topics: > > 1. Re: please help with some transformations. (Alex Balashov) > 2. Re: please help with some transformations. > (Daniel-Constantin Mierla) > 3. Abort at qm_debug_frag function (Bruno Bresciani) > 4. Re: Abort at qm_debug_frag function (Daniel-Constantin Mierla) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Wed, 03 Apr 2013 14:18:18 -0400 > From: Alex Balashov <abalashov(a)evaristesys.com> > Subject: Re: [SR-Users] please help with some transformations. > To: sr-users(a)lists.sip-router.org > Message-ID: <515C726A.10104(a)evaristesys.com> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > Hello Julian, > > On 04/03/2013 01:54 PM, julian arsanches wrote: > > > Hi am new to kamilio , but able to make calls, i need to do something > > really simple if someone can give me an example of it i will be > gratefull. > > > > just before the call gets relayed i need to delete a prefix from the sip > > uri so it gets transformed to the same uir but less 3 digits. > > > > for example sip:123xxx@domain.com <mailto:sip%3A123xxx@domain.com> >>> > > changed to sip:xxxx@dmain.com <mailto:sip%3Axxxx@dmain.com> > > > > i tryed with transformations but still too new and does give me so many > > errors that kamailio don't start, > > There is a core convenience function that can help with this, as long as > you're manipulating the request URI: > > strip(3); > > You referred to "the SIP URI", which, of course, is ambiguous since a > SIP request has many SIP URIs. > > -- Alex > > -- > Alex Balashov - Principal > Evariste Systems LLC > 235 E Ponce de Leon Ave > Suite 106 > Decatur, GA 30030 > United States > Tel: +1-678-954-0670 > Web: http://www.evaristesys.com/, http://www.alexbalashov.com/ > > > > ------------------------------ > > Message: 2 > Date: Wed, 03 Apr 2013 20:18:44 +0200 > From: Daniel-Constantin Mierla <miconda(a)gmail.com> > Subject: Re: [SR-Users] please help with some transformations. > To: "Kamailio (SER) - Users Mailing List" > <sr-users(a)lists.sip-router.org> > Message-ID: <515C7284.5060905(a)gmail.com> > Content-Type: text/plain; charset="iso-8859-1"; Format="flowed" > > Hello, > > if you just need to remove first 3 digits, use: > > strip(3); > > If the number of digits to strip is variables, use strip transformation: > > - http://www.kamailio.org/wiki/cookbooks/4.0.x/transformations#sstrip_len > > If does not work for you giving errors, paste here what you have in the > config for that part. > > Cheers, > Daniel > > On 4/3/13 7:54 PM, julian arsanches wrote: > > Hi am new to kamilio , but able to make calls, i need to do something > > really simple if someone can give me an example of it i will be > > gratefull. > > > > just before the call gets relayed i need to delete a prefix from the > > sip uri so it gets transformed to the same uir but less 3 digits. > > > > for example sip:123xxx@domain.com > > <mailto:sip%3A123xxx@domain.com> >>> changed to sip:xxxx@dmain.com > > <mailto:sip%3Axxxx@dmain.com> > > > > i tryed with transformations but still too new and does give me so > > many errors that kamailio don't start, > > > > tryed subs_uri but i dont know how to tell it to just take the first > > tree digits, it is usually done for adding prefixes. > > > > please help. > > > > thank you in advance. > > > > on my scenario i am having an asterisk that i dont control sending me > > an invite for with a prefix i need kamailio to forward that uri to > > another server minus the prefix. > > > > thank you. > > > > > > _______________________________________________ > > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > > sr-users(a)lists.sip-router.org > > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > -- > Daniel-Constantin Mierla - http://www.asipto.com > http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda > Kamailio World Conference, April 16-17, 2013, Berlin > - http://conference.kamailio.com - > >

1 0

please help with some transformations.
by julian arsanches 03 Apr '13

03 Apr '13

Hi am new to kamilio , but able to make calls, i need to do something really simple if someone can give me an example of it i will be gratefull. just before the call gets relayed i need to delete a prefix from the sip uri so it gets transformed to the same uir but less 3 digits. for example sip:123xxx@domain.com >>> changed to sip:xxxx@dmain.com i tryed with transformations but still too new and does give me so many errors that kamailio don't start, tryed subs_uri but i dont know how to tell it to just take the first tree digits, it is usually done for adding prefixes. please help. thank you in advance. on my scenario i am having an asterisk that i dont control sending me an invite for with a prefix i need kamailio to forward that uri to another server minus the prefix. thank you.

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Kamailio World Updates - April 3
by Daniel-Constantin Mierla 03 Apr '13

03 Apr '13

Hello, just a bit less than two weeks to the start of Kamailio World Conference, we are gearing up and already started preparing everything for the event. The agenda with presentations is available at: - http://conference.kamailio.com/k01/schedule/ Details about the speakers are available at: - http://conference.kamailio.com/k01/speakers/ Regarding Kamailio project, even there was an extended weekend with public holidays in around Easter, three new modules showed up - stun, dnssec and sipt - and a lot of new code made its way to old modules, among most active being snmpstats. If you haven't registered yet for the Kamailio World conference, be aware of that standard registration ends this week, on April 5, 2013. Actually the registration might be completely closed as it is very likely that the event will get full booked, even we extended the capacity about one week ago. It is a great opportunity to meet other hundreds people from real time communications services. Do not delay registration if you plan to attend the event, the form and more details at: - http://conference.kamailio.com/k01/registration/ Several companies coming with stands for presenting products, services and demos will welcome you to stop by just in front of the conference room - Sipwise, Sipgate, NG Voice, FhG Fokus and Asipto. Looking forward to meet many of you in Berlin, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

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Evariste Systems to drop Kamailio market, become Acme Packet VAR
by Alex Balashov 02 Apr '13

02 Apr '13

For immediate release: ATLANTA, GA (1 April 2013)--Evariste Systems LLC, an Atlanta-based consultancy specialising in Kamailio-based VoIP infrastructure solutions for the ITSP and CLEC market, has announced that beginning in the second quarter of 2013, it will be abandoning its Kamailio-based technology portfolio to focus on its new role as a preferred VAR (Value Added Reseller) for Acme Packet (NASDAQ:APKT). "It is with a heavy heart that we abandon five years of Kamailio-oriented work and the Canonical SIP Routing Platform product derived from it," said Alex Balashov, the principal of the company. "However, the reality is that investment in open-source VoIP technology is a dead end. From a technological point of view, we have lagged very badly in meeting the needs of today's sophisticated VoIP market, and it's time to cut our losses. Asterisk, Kamailio, FreeSWITCH--all this stuff just hasn't kept up with the pace of evolution of 3GPP, ETSI, and ITU standards. We are tired of saying 'sorry, we don't support IMS or H.323' to our resultingly dwindling customer base. Does anyone actually run an all-SIP network?" Starting in early April, Evariste will begin providing value-added consultancy related to the implementation of the Acme Packet Net-Net Session Director. In Balashov's view, "the Net-Net SD is the only product capable of meeting the perimeter security, routing and peering needs of today's VoIP service delivery environment." Fred Posner, the director of Team Forrest, a Palner Group integration and consultancy operation based in the Jacksonville, Florida area, agreed: "SIP is a tiny piece of the telephony puzzle. The big boys of ClueCon [an interoperator revenue-sharing consortium] want DIAMETER-based interdomain peering policy control, H.323, MGCP, and IMS. IMS is pretty much how VoIP architecture is done now. We got out of the Asterisk business just in time, right before Mitel swallowed the PBX world. I'm glad to see Evariste is finally seeing the light, and I'm sure its shareholders are too." Posner also believes Evariste's lack of support for TDM interfaces accounted for dwindling market share. "Have you seen CSRP? It's SIP in, SIP out. Real inter-LATA haulers and application service providers use TDM and leave SIP for things like voicemail. I can't plug my DS3s into a SIP proxy, so I just don't think there was any real demand for the sort of thing they were doing." Noting Oracle's US$2.5bn acquisition of Acme Packet in early February, as well as its more recently announced buyout of Tekelec, a Siris Capital Group portfolio company, Balashov remarked: "The obvious shift to an Oracle-centric telephony paradigm was a kind of validation, if you will, of our decision to unload our dead weight and sign on to the revolution in unified communications." Sean McCord, of CyCORE Systems, an Atlanta-based software consulting house and long-time Evariste creditor, agreed that there was a natural synergy between Evariste's shift to Acme Packet and Oracle's dominance of telephony infrastructure. "Oracle is a forward-thinking telecom pioneer," McCord said. "The telephone is Oracle, and Oracle is the telephone." Balashov also noted that a tightening regulatory environment and new consumer protection rules helped hasten the decision to embrace the more professionalised Acme Packet product portfolio. John Knight, Senior Engineer at Hendersonville, NC-based Ringfree Communications, one of Evariste's oldest channel partners, said: "As one of Evariste's long-time disties, we were jittery about exposure to CALEA and the QA requirements of large call centers. We tried to make do, but at some point we just had to put the relationship on stop. I'm all in favour of open, but there's just no open-source software out there that does call recording, and that's the bottom line for us. In the end, we had to restructure some debt just to get bondholders to let us source a proprietary solution on tick." In a thematically related move, Evariste will be dropping its heavy use of the open-source PostgreSQL database manager for its rating and reporting tools. "The business case for standardising on Oracle's databases could not be clearer. With Oracle Database 11g's support of warehousing and OLTP, the real mystery is why we didn't go there sooner," said Balashov. Carlos Alvarez, a director at Televolve, a growing Phoenix-area VoIP operator, recently spearheaded a move away from Evariste's PostgreSQL- based call detail record (CDR) storage solution to one running atop Microsoft SQL Server 2008. Alvarez commented: "Evariste had a nice idea, in a cute, David-and-Goliath kind of way, but we're processing over five hundred phone calls a day now. Are we really going to store those kinds of volumes in an open-source database? Might as well just put it all in flat text files at that point. Phone service is an uptime game. You can't compromise on this stuff. What if someone needs to call 911?" Asked to summarise his expectations, Balashov said: "I hope this turns us around in a big way. We were wrong to think that nobody cared about stuff like P-CSCFs, or that you could deliver even rudimentary VoIP to the premise without the expansive feature set of a comprehensive solution like the Net-Net SBC. I can only hope the market forgives us for betting on 'SIP Express Router' and its ilk back in the day, and gives us a chance to do it right in round two." Fred Posner, of Team Forrest, added: "Besides, if you look at the Git repository, Kamailio hasn't had any code contributions in at least five years. It seems everyone's figured out this pure SIP stuff is defunct and hokey."

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ClueCon 2013 - Call For Speakers
by Michael Collins 02 Apr '13

02 Apr '13

[image: https://mail-attachment.googleusercontent.com/attachment/u/0/?ui=2&ik=43fec…]<http://www.cluecon.com> ClueCon - the open source IP communications conference by developers, fordevelopers - would like to announce that we are having an open call for speaking proposals for this year's event. If you have an idea fora technical presentation for ClueCon 2013 then we would like to hear about it. What makes a great ClueCon presentation? The tech savvy crowd that attends ClueCon *loves *technical presentations. In general, the more technical the presentation, the better. If you are thinking about a presentation then consider these points: - ClueCon talks are 30 minutes in length, including Q&A time with the audience - ClueCon has a special focus on open source VoIP and telephony projects like FreeSWITCH, Asterisk, OpenSIPS, and Kamailio - Attendees enjoy hearing about projects built with open source tools, especially those in a production environment - Highly technical discussions that show the nuts and bolts are especially well-liked - The audience appreciates seeing and participating in live demonstrations - We are especially interested in WebRTC-related talks and demonstrations Please send your proposals to marketing(a)cluecon.com. Be sure to include the following items: - Working title - Brief description of the talk (abstract) - Name of the presenter Don't delay! There are a limited number of openings. We will contact you as soon as your talk has been approved and will inform you of the scheduled time. ClueCon 2013 Registration Information ClueCon 2013 registration is now open!. Visit the registration page<http://www.cluecon.com/register?cc12cfs> for details. Be sure to book your room at the Hyatt Chicago Magnificent Mile<http://www.cluecon.com/hotels/>and qualify for the $300 discount. As always, feel free to call us at 877.742.CLUE (877.742.2583) if you have any questions about ClueCon 2013. Also, keep in mind that the FreeSWITCH community has a conference call<http://wiki.freeswitch.org/wiki/Weekly_Conference_Call?cc12cfs>each Wednesday at 1PM Eastern time. This is a great opportunity to talk about open source telephony and get to know a number folks who will be at ClueCon 2013. Stay tuned for more news about ClueCon speakers, sponsors, and related events! -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE

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alphabetic indexes generated for development version
by Daniel-Constantin Mierla 02 Apr '13

02 Apr '13

Hello, I generated the alphabetic indexes for development version, there are already new modules and code since release of 4.0. If anyone spots missing elements (mod params, functions, etc...), write back here so we fix the issue (could be wrong formatted xml docbook file for module readme). They are available at: - http://www.kamailio.org/wiki/#alphabetic_indexes Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

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Are there any Blind Call Forwarding examples available for Kamailia 3.3.0?
by alans126 02 Apr '13

02 Apr '13

I've just configured Kamailia and users 10001,10002,10003 can give each other a call successfully. Now I want to forward every call to 10002 to 10003. The only example I can find is for OpenSER 1.X, the configure file seems different from Kamailia 3.3's. I tried to add the following code to the configure file and some other add_module parts to the other one, but it doesn't work... if(avp_db_load("$ru","$avp(s:callfwd)")) { $ru = $avp(s:callfwd); route(RELAY); exit; }. Can anyone provide me with the kamailia.cfg and kamctlrc files which implemented call forwarding unde Kamailia 3.3, please? I'm a newbie so I really hope to get your help. 2013-04-02 alans126

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Support
by Maritza y Cristian Zuÿfffff1iga 02 Apr '13

02 Apr '13

Dear Sr, Is it possible to use QuteCom softphone with iptel.org SIP service? Regards Cristian A. Zúñiga

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Is cvs repository of sip_route still alive?
by Yuming Zheng 01 Apr '13

01 Apr '13

Hi, export CVSROOT=:pserver:anonymous@cvs.berlios.de:/cvsroot/ser cvs login OK, when try checkout the sip_route cvs co sip_router cvs [checkout aborted]: cannot stat /cvsroot/ser/locks: No such file or directory Although cvs view is working. I want go check some logs in the repository. BR, Frank.zheng

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Wiki site registration updates
by Daniel-Constantin Mierla 01 Apr '13

01 Apr '13

Hello, as of yesterday, creation of new accounts for Kamailio's wiki site requires to answer a project related question. Captcha was useless as spam bots were lately going through it easily, creating accounts in a rate of approx 50 new registrations per day. The extra question is asked just after CAPTCHA, see it at: - https://www.kamailio.org/wiki/start?do=register Hopefully the questions are simple enough to allow good people to register and difficult enough for spambots to give up. It is not a very sophisticated system, let's see if there will be any efforts in reverse engineering to break in with bots. So far no new spammer account. If they will succeed, at least they learn something useful. If anyone has difficulties creating wiki accounts, write an email to sr-dev mailing list and it will be investigated. Cheers, Daniel PS. This registration system will last, is not for April 1. -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

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Kamailio + ABills
by Victor V. Kustov 01 Apr '13

01 Apr '13

Hi. Can someone tips for Kamailio + ABillS works? -- SY, Victor JID: coyote(a)bks.tv JID: coyote(a)bryansktel.ru I use FREE operation system: 3.8.4-calculate GNU/Linux

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Yet another IPv6 question related with rtpproxy ipv4/ipv6 bridging
by Miguel Baptista 31 Mar '13

31 Mar '13

Hello, I have been doing some tests with kamailio and IPv6. My initial setup was IPv6 only and now am I extending it to a dual-stack environment. Well, and now I am starting to face some (interesting) challenges. So, the first step in the dual-stack environment was to install RTP Proxy and configured kamailio to use it. With this setup, the UAs locally registered were able to communicate with each-other no matter with address family (IPv4/IPv6) they were using. So far so good. But now I want to extend my tests a bit more ... I want to communicate with the "outside world" (using ENUM and domain based SIP URIs). Do I have a way to know if the "destination" is IPv4 or IPv6? Because I need that information in order to properly bridge the calls on rtpproxy. I tried to use the onsend_route but it didn't work. I mean, I am able to know if the "next-hop" is IPv6 or IPv4 but it seems that it is "too late" to use rtpproxy. BTW ... I am assuming that if the "next-hop" is IPv6 then the final user agent will also be IPv6 (the same for IPv4) Should I use some other approach? For example, failure route instead? Does anyone have a similiar setup? How are you solving this issue? - Miguel Baptista

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