- sr-users - kamailio.org

no more pkg mem on ul_dump
by Klaus Darilion 06 Oct '11

06 Oct '11

Hi! Kamailio 1.3.4: "kamctl ul monitor" causes: ERROR: <core> [attr.c:80]: no more pkg mem (21) ERROR: mi_fifo [fifo_fnc.c:513]: command (ul_dump) processing failed There are ~ 2000 entries in location table. Default PKG size is used. I remember there were some similar reports recently, but couldn't find the relevant threads. Any hints for the cause of this problem and workarounds? Thanks Klaus

4 8

Compiling on a 2.4 kernel
by Pedro Antonio Vico Solano 06 Oct '11

06 Oct '11

2 3

Problems with domain on registrations.
by Henrik Aagaard Sørensen 06 Oct '11

06 Oct '11

I have the current setup: SIP client <-> SER Proxy <-> SIP Trunk. My reason for the SER Proxy is that the SIP trunk only allows connections from approved IP addresses (such as my server with SER Proxy). My user on the SIP Trunk has the following authentication username@domain. I am able to register to the SIP Trunk as I'm setting up my SIP klient with username to username, domain to domain and then add my SER Proxy IP address as proxy. This works and means that I'm able to place calls to the SIP Trunk. BUT people cannot place calls from the SIP Trunk to my SIP client, due to the fact that on the SIP Trunk my locations is registered as username@SER-Procy-IP. So when people are calling me, the SER Proxy is looking for username@SER-Procy-IP and NOT username@domain, as I'm registered with. And then SER Proxy reports: 404 Not Found. Can anyone help me how to set this up correctly? I do not have access to the SIP Trunk, but I have all access to the SIP client and SIP Proxy.

2 2

Removing Contact from location table when TCP connection from UA is dropped...
by Krishna Kurapati 06 Oct '11

06 Oct '11

Hi, When UA connects with the Proxy over TCP, the contact is mostly bound to the firewall's binding IP and port for that connection. When the connection is dropped, the contact in Kamailio is no longer valid even though there is time left to expire the contact. Is there some way to link TCP connection and the contact in such a way that when TCP connection is dropped, contact from the location can be removed? Thanks Krish Kura

2 1

New in Kamailio
by Leonardo Sena 05 Oct '11

05 Oct '11

Hello I´m a new contribuitor of the list and my english dont very good then i hope of help more in the future I am installing Kamailio and Siremis but i dont understand like siremis works and where i can create the routes i would like step by step to routes rules. Best Regards, Leonardo Sena

1 0

Balancing incoming traffic (DID)
by Anto 05 Oct '11

05 Oct '11

Hello It presents a problem because I do not know how to assimilate the implementation. Until now we had a design of a proxy to balance out between several gateway, but it will add another proxy for another service (other users) and use the same gateway output or perhaps different (although not that be better if divide or unify the gateway). The problem as I have with the routing entry, which is not very well how to implement it. Now we have the design like this: -GW ----\ / - CARRIER1 USER ------ KAMAILIO ----- GW ---- + \ - CARRIER2 -GW ----/ And becomes like this: -GW ----\ / - CARRIER1 USER ------ KAMAILIOSERVICE1 ----- GW ---- + \ - CARRIER2 -GW ----/ -GW ----\ / - CARRIER1 USER ------ KAMAILIOSERVICE2 ----- GW ---- + \ - CARRIER2 -GW ----/ Or maybe using the same gateway. The problem is that each carrier provides DID number (I use the same for both services). Kamailio had thought to put a gateway between the carriers and that they did check the service and send it to the corresponding gateway (in the case of the gateway separately), or put it in the proxy gateway and making checking for routing to proxy (in the case of using the same gateway). What would be better? Any recommendations on how to put this into practice?. The biggest question that I believe is the routing table, as if this new proxy routes the traffic into and leaves her in the field via ip, also pass through the output data (I have doubts whether SIP level, following standard, you can not pass through the exit) and want it to be direct (USER -> Proxy -> GW -> CARRIER), just past the entrance. Anyone could advise me where I could find more information to set up this scenario?. thanks and sorry for bothering you with this question ;-). best regards

2 2

limiting concurrent calls with dialog module
by Uri Shacked 05 Oct '11

05 Oct '11

Hi, Thanks for the help. The idea of using SQLOPS for many things crossed my mind.... But....I am using ACC with DB, Dialog with DB and so... isnt it too much? i mean for load and capacity reasons? BR, Uri

2 1

Re: [SR-Users] Dialog ref Error
by Uri Shacked 05 Oct '11

05 Oct '11

On Wed, Oct 5, 2011 at 2:34 PM, Uri Shacked <ushacked(a)gmail.com> wrote: > OK... > now i see the problem is some where in the cfg file... > i get te following error : > > 0(3774) : <core> [cfg.y:3406]: parse error in config file > //etc/kamailio/kamailio.cfg, from line 944, column 2 to line 945, column 0: > syntax error > 0(3774) : <core> [cfg.y:3406]: parse error in config file > //etc/kamailio/kamailio.cfg, from line 944, column 2 to line 945, column 0: > bad command > ERROR: bad config file (2 errors) > 0(3774) DEBUG: tm [t_funcs.c:122]: DEBUG: tm_shutdown : start > 0(3774) DEBUG: tm [t_funcs.c:125]: DEBUG: tm_shutdown : emptying hash > table > 0(3774) DEBUG: tm [t_funcs.c:127]: DEBUG: tm_shutdown : removing > semaphores > 0(3774) DEBUG: tm [t_funcs.c:129]: DEBUG: tm_shutdown : destroying tmcb > lists > 0(3774) DEBUG: tm [t_funcs.c:132]: DEBUG: tm_shutdown : done > 0(3774) DEBUG: <core> [mem/shm_mem.c:236]: shm_mem_destroy > 0(3774) DEBUG: <core> [mem/shm_mem.c:239]: destroying the shared memory > lock > > > here is my complete cfg file. > any ideas? > > > ... > > ######## Routing Logic ######## > > # Main SIP request routing logic > > # - processing of any incoming SIP request starts with this route > > route { > > setflag(FLT_TRC); # Trace message > > # 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")) > > { > > #!ifdef WITH_DIALPLAN > > dp_translate("1", "$rU/$rU"); > > xlog("translated to $rU \n"); > > #!endif > > #!ifdef WITH_CARRIERROUTE > > ####cr_user_carrier("$rU", "$td", "$avp(carrierid)"); > > ####xlog("translated to var $avp(carrierid) \n"); > > cr_route("OSP1", "kamailio2", "$rU", "$rU", "call_id"); > > #!endif > > setflag(FLT_ACC); # do accounting > > } > > # dispatch requests to foreign domains > > route(SIPOUT); > > ### requests for my local domains > > # handle presence related requests > > route(PRESENCE); > > # handle registrations > > route(REGISTRAR); > > if ($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; > > } > > #!ifdef WITH_PIPELIMIT > > # perform pipe match for INVITE method > > if (is_method("INVITE")) { > > if (!pl_check("avg")) { > > pl_drop(); > > exit; > > } > > if (!pl_check("peek")) { > > pl_drop(); > > exit; > > } > > #endif > > #!ifdef WITH_DIALOG > > xlog("currently active dialogs are $DLG_count\n"); > > dlg_manage(); > > dlg_setflag(FLT_DLG); > > if (is_method("INVITE")) { > > $var(size) = 0; > > get_profile_size("test","$rU","$var(size)"); > > xlog("currently, the user $rU has $var(size) active calls\n"); > > if( $var(size) >= 5 ) > > { > > sl_send_reply("507", "Simultaneous calls limit reached"); > > xlog("Limit of $var(size) active calls has been reached \n"); > > exit; > > } > > set_dlg_profile("test","$rU"); > > } > > #!endif > > } > > # 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(); > > 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")){ > > force_rtp_proxy(); > > } > > 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 $ru\n"); > > } > > # Sample onreply route > > onreply_route[REPLY_ONE] { > > xdbg("incoming reply\n"); > > #!ifdef WITH_NAT > > if ((isflagset(FLT_NATS) || isbflagset(FLB_NATB)) > > && status=~"(183)|(2[0-9][0-9])") { > > force_rtp_proxy(); > > } > > if (isbflagset("6")) { > > fix_nated_contact(); > > } > > #!endif > > } > > route[STATS] { > > sql_query("ca", > > "insert into statistics (time_stamp,shm_used_size,shm_real_used_size," > > "shm_max_used_size,shm_free_used_size,ul_users,ul_contacts) values ($Ts," > > "$stat(used_size),$stat(real_used_size),$stat(max_used_size)," > > "$stat(free_size),$stat(location-users),$stat(location-contacts))", > > "ra"); > > } > > # 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(); > > ##} > > } > > On Wed, Oct 5, 2011 at 12:12 PM, Uri Shacked <ushacked(a)gmail.com> wrote: > >> >> >> ---------- Forwarded message ---------- >> From: Uri Shacked <uri.shacked(a)gmail.com> >> Date: Wed, Oct 5, 2011 at 11:48 AM >> Subject: dialog ref Error >> To: sr-users(a)lists.sip-router.org >> >> >> Hi, >> >> i keep on getting this error: >> Oct 5 14:39:33 Kamailio2 /usr/sbin/kamailio[21568]: DEBUG: dialog >> [dlg_hash.c:247]: new dialog on hash 531 >> Oct 5 14:39:33 Kamailio2 /usr/sbin/kamailio[21568]: DEBUG: dialog >> [dlg_handlers.c:264]: route_set , contact sip:036264529@10.2.0.55:5061, >> cseq 1 and bind_addr udp:10.2.0.5:5060 >> Oct 5 14:39:33 Kamailio2 /usr/sbin/kamailio[21568]: DEBUG: dialog >> [dlg_hash.c:519]: ref dlg 0xb417f5a0 with 3 -> 3 >> Oct 5 14:39:33 Kamailio2 /usr/sbin/kamailio[21568]: DEBUG: dialog >> [dlg_hash.c:600]: unref dlg 0xb417f5a0 with 1, crt ref count: 3 >> >> anyone knows what it is? >> >> here is the dialog part in my cfg: >> >> >> *# ------ dialog params --------* >> >> modparam("dialog", "db_url", DBURL) >> >> ##modparam("dialog", "rr_param", "did") >> >> modparam("dialog", "default_timeout", 7200) >> >> modparam("dialog", "dlg_match_mode", >> 0) >> >> modparam("dialog", "detect_spirals", 1) >> >> modparam("dialog", "db_mode", 1) >> >> ##modparam("dialog", "db_fetch_rows", 100) >> >> modparam("dialog", "dlg_flag", FLT_DLG) >> >> modparam("dialog", "profiles_with_value", "test") >> >> >> >> .......... >> >> dlg_manage(); >> >> ##dlg_setflag(FLT_DLG); >> >> if (is_method("INVITE")) { >> >> $var(size) = 0; >> >> get_profile_size("test","$rU","$var(size)"); >> >> xlog("currently, the user $rU has $var(size) active calls\n"); >> >> if( $var(size) >= 5 ){ >> >> sl_send_reply("507", "Simultaneous calls limit reached"); >> >> xlog("Limit of $var(size) active calls has been reached \n"); >> >> exit;} >> >> set_dlg_profile("test","$rU");} >> >> ........ >> >> >

2 1

Dialog ref Error
by Uri Shacked 05 Oct '11

05 Oct '11

---------- Forwarded message ---------- From: Uri Shacked <uri.shacked(a)gmail.com> Date: Wed, Oct 5, 2011 at 11:48 AM Subject: dialog ref Error To: sr-users(a)lists.sip-router.org Hi, i keep on getting this error: Oct 5 14:39:33 Kamailio2 /usr/sbin/kamailio[21568]: DEBUG: dialog [dlg_hash.c:247]: new dialog on hash 531 Oct 5 14:39:33 Kamailio2 /usr/sbin/kamailio[21568]: DEBUG: dialog [dlg_handlers.c:264]: route_set , contact sip:036264529@10.2.0.55:5061, cseq 1 and bind_addr udp:10.2.0.5:5060 Oct 5 14:39:33 Kamailio2 /usr/sbin/kamailio[21568]: DEBUG: dialog [dlg_hash.c:519]: ref dlg 0xb417f5a0 with 3 -> 3 Oct 5 14:39:33 Kamailio2 /usr/sbin/kamailio[21568]: DEBUG: dialog [dlg_hash.c:600]: unref dlg 0xb417f5a0 with 1, crt ref count: 3 anyone knows what it is? here is the dialog part in my cfg: *# ------ dialog params --------* modparam("dialog", "db_url", DBURL) ##modparam("dialog", "rr_param", "did") modparam("dialog", "default_timeout", 7200) modparam("dialog", "dlg_match_mode", 0) modparam("dialog", "detect_spirals", 1) modparam("dialog", "db_mode", 1) ##modparam("dialog", "db_fetch_rows", 100) modparam("dialog", "dlg_flag", FLT_DLG) modparam("dialog", "profiles_with_value", "test") .......... dlg_manage(); ##dlg_setflag(FLT_DLG); if (is_method("INVITE")) { $var(size) = 0; get_profile_size("test","$rU","$var(size)"); xlog("currently, the user $rU has $var(size) active calls\n"); if( $var(size) >= 5 ){ sl_send_reply("507", "Simultaneous calls limit reached"); xlog("Limit of $var(size) active calls has been reached \n"); exit;} set_dlg_profile("test","$rU");} ........

2 1

kamailio sip-capture error on compile
by Jeff Anderson 05 Oct '11

05 Oct '11

I am trying to get kamailio sip-capture up and running for use with homer. I can get the service to start but i receive the following error. [root@homer kamailio]# /usr/local/sbin/kamailio -c loading modules under /usr/local/lib/kamailio/modules_k/:/usr/local/lib/kamailio/modules/ 0(10208) WARNING: <core> [socket_info.c:912]: WARNING: fix_hostname: could not rev. resolve 10.40.0.22 Listening on udp: 10.40.0.22:5060 Aliases: config file ok, exiting... [root@homer kamailio]# kamctl start INFO: Starting Kamailio : INFO: started (pid: 10226) [root@homer kamailio]# [root@homer kamailio]# tail -f /var/log/messages Oct 3 05:30:17 homer /usr/local/sbin/kamailio[1554]: NOTICE: <core> [main.c:704]: Thank you for flying kamailio Oct 3 05:30:32 homer kamailio: WARNING: <core> [socket_info.c:912]: WARNING: fix_hostname: could not rev. resolve 10.40.0.22 Oct 3 05:31:59 homer kernel: device eth0 entered promiscuous mode Oct 3 05:32:47 homer kernel: device eth0 left promiscuous mode Oct 3 05:52:10 homer /usr/local/sbin/kamailio[1753]: NOTICE: <core> [main.c:704]: Thank you for flying kamailio Oct 3 05:52:12 homer kamailio: WARNING: <core> [socket_info.c:912]: WARNING: fix_hostname: could not rev. resolve 10.40.0.22 Oct 3 05:55:35 homer kernel: device eth0 entered promiscuous mode Oct 3 05:57:46 homer kernel: device eth0 left promiscuous mode Oct 3 07:05:35 homer /usr/local/sbin/kamailio[1894]: NOTICE: <core> [main.c:704]: Thank you for flying kamailio Oct 3 07:18:43 homer kamailio: WARNING: <core> [socket_info.c:912]: WARNING: fix_hostname: could not rev. resolve 10.40.0.22 ^C [root@homer kamailio]# Does anyone have any ideas on how to resolve this error? Thanks, Jeff Anderson

3 3

Installing Versign Intermediate CA Certificate In Kamailio 3.1.5
by arif tuhin 05 Oct '11

05 Oct '11

I've got trial certificate from verisign. They signed the server certificate with an intermediate ca and gave instructions for installing it into apache/IIS etc. Both issued certificate and the intermediate CA should be installed in the server. if only server certificate is installed , it will not be trusted by default. Now problem is in apache mod_ssl there is a field called ServerChainFile where the intermediate ca name should be in place, but in the tls module documentation, i find no place for intermediate ca. How should i install the intermediate ca the? Thanks in advance.

1 0

db_text
by Bruno Bresciani 04 Oct '11

04 Oct '11

Hello ALL, Someone knows why at line 149 in .../kamailio-3.1.2/modules_k/ db_text/dbt_tb.c the shm_malloc function sometimes allocates memory to dtp->prev but sometimes doesn't? I'm asking this question because when the subscriber table is being loading to shared memory in db_text module, dtp->prev is allocated but on version, dispatcher, location table it doesn't... Analysing the source code I can't see any reason to this behavior. The allocation of dtp->prev to subscriber table is generating a core at db_text module. Best Regards

2 5

Siremis CDR problems
by Klaus Darilion 04 Oct '11

04 Oct '11

Hi! When populating cdrs table from timer route I get some errors: ERROR: db_mysql [km_dbase.c:120]: driver error on query: Duplicate entry '088827BCD0738B6F(a)111.11.111.11-00E0F510042319395872000014D3-6ADB' for key 2 ERROR: <core> [db_query.c:130]: error while submitting query ERROR: sqlops [sql_api.c:217]: cannot do the query Checking the acc table for those key I found the entries in the attachment. Maybe it be that the stored procedure can not handle PRACK or reINVITEs correctly? Thanks Klaus

2 3

new wiki system
by Daniel-Constantin Mierla 04 Oct '11

04 Oct '11

Hello, I installed a new wiki site, targeting to collect documentation for upcoming v3.2.0 and further releases. You can find it at: * http://www.kamailio.org/wiki/ Hopefully, it will make the process of updating/migrating the relevant documentation from old wiki (http://www.kamailio.org/dokuwiki/) straightforward and offer a clean reference for upcoming stable version. Old wiki is in place since 2005, collected a lot of pages, but many are for very old versions of kamailio, making difficult, at least for new comers, to sort out documentation for latest stable. All of you are invited to help with this process of adding documentation for 3.2.0 on the new wiki. The old wiki will stay in place, it will not be removed. Both are based on Dokuwiki, using different themes. The new one is at the beginning, so no much content in there yet, but by the time of releasing 3.2.0, should contain sufficient documentation for get started with that version. One very demanded thing was the alphabetic index for parameters, functions and mi commands from modules. This page contains new generated indexes for version 3.2.0. More details in a separate email. If you have questions about the new wiki, don't know where to place some tutorial, feel free to ask on mailing lists. Ultimately, pages can be relocated, but try to use namespaces for related topics. Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Dec 5-8, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

3 4

dispatcher, LCR, carrierroute or mtree?
by Skyler 04 Oct '11

04 Oct '11

Hi all, I need to setup an inbound DID router for 8 proxies and ~300 DID's. Very simple, a call comes in from pstn > kamailio looks up DID for exact match in mysql > returns location > If db result returns noservice redirect to 1.2.3.4 else remain in call-flow and send to destination. Having a failure route capability would be nice. From what I've read on each of the 4 modules, any could do what I need though each have their own complexities. I'm still new to kamailio and hoping to utilize the group here for some advice. I'd like to keep this as simple as possible then adding/deleting DID's from the db as needed for routing. In your experience which would be the best module to use in order to achieve my goal? TIA, Skyler

4 5

Trouble Setting up Kamailio
by Peter Schrock 04 Oct '11

04 Oct '11

I am using the site: http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-3.1.x-from-git to load up Kamailio on my ppc using Leopard X. I managed to figure out the mysql issue and got it to install, but now there are some variances to the instruction from the stated install site and I am kind of lost. If I could get some help with the following - "7. Create MySQL database" This whole section seems confusing. It doesn't say what I am suppose to look for. I am sure what it says is all you need to do, but I am not sure if it is completed correctly or not. - "8. Edit configuration file" This section could use more specific information as to how to set up the configuration file. - "9. The init.d script" This section says I am suppose to put "*kamailio.init*" in the directory "*/etc/init.d*", then I am suppose chmod after it is copied. First of, I don't have that directory, am I suppose to create it or is it suppose to be created already? Also, it says to "*adduser*" with some options. Every time I try, I get the error message that it doesn't recognize the "*adduser*" command. Please help Peter

2 1

New to Kamailio - make problem
by Peter Schrock 03 Oct '11

03 Oct '11

I can't seem to get kamailio to "*make*". I am following the directions on the wiki on how to install. I have downloaded, compiled, and installed mysql server and every time I run to "*make all*" with kamailio, I get this error message. At first, I was thinking that *km_db_mysql.c* couldn't find the file. The header seems to be looking for *mysql.h* (at *#include <mysql/mysql.h>*). I found the file from installing mysql server and I thought I'd help * km_db_mysql.c* to find the file, but it still gives me this error. I am using a power pc running OS X.5.1. CC (gcc) [M db_mysql.so] km_db_mysql.o km_db_mysql.c:49:19: error: mysql/mysql.h: No such file or directory km_db_mysql.c: In function ‘kam_mysql_mod_init’: km_db_mysql.c:92: warning: implicit declaration of function ‘mysql_get_client_info’ km_db_mysql.c:92: warning: format ‘%s’ expects type ‘char *’, but argument 7 has type ‘int’ km_db_mysql.c:92: warning: format ‘%s’ expects type ‘char *’, but argument 6 has type ‘int’ make[1]: *** [km_db_mysql.o] Error 1 make: *** [modules] Error 1 Peter Schrock

3 3

Combining Dispatcher and NAT_Helper modules.
by Henrik Aagaard Sørensen 03 Oct '11

03 Oct '11

I'm currently having a Kamailio running as load balancer, at it works most of the time. But sometimes fails due to NAT issues. This is my config: http://pastebin.com/8F8X45xh Can someone help me combine the NAT_Helper module into this config? It's not the variables to set or RTPProxy, but in the Routing Logic.

1 0

Incoming calls question
by Javier Vidal -- Quasar 03 Oct '11

03 Oct '11

i have a question, i have to make a system to recieve several calls and the same time, and depending of destination DDI i have to rewrite the URI's IP. I am not going to have Registered users, only IP validation. The question is: What is the better way to consult thousands DDIs and get the IP from mysql? I had thought, use sqlops.so module, but i think that i could have problems with it. Another posibility is use the carrierroute and use the full pattern for the DDI. Thank for all sugestions. Javier V.

2 2

alphabetic indexes for module parameters, functions, ...
by Daniel-Constantin Mierla 03 Oct '11

03 Oct '11

Hello, for the last versions the alphabetic indexes were not generated - there were different people doing that in the past, using various self-written tools. For upcoming version, 3.2.0, I spend some time to structure the docbook files for modules in order to be easier to get from them the names for functions, parameters and mi commands. Therefore, the new wiki system has fresh generated indexes from modules' documentations. You can find them at: * http://www.kamailio.org/wiki/alphaindexes:3.2.x:modparameters * http://www.kamailio.org/wiki/alphaindexes:3.2.x:modfunctions * http://www.kamailio.org/wiki/alphaindexes:3.2.x:micommands Being generated from docbook xml files, the indexes contain name, path to the module in source tree and module name, the last one with link to html readme file of the module published to kamailio.org site. There is no link at the moment pointing inside the html readmes to the place where the item is documented, but each readme has table of content, so it is easy to spot it. We have lots lots of modules right now, therefore it was not easy to check if some parameters, functions or mi commands were not indexed. If you find some missing, please write back here so we can fix the dockbook or the indexing tool, depending where the problem is. Hope the indexes will be helpful for many of you. Ending the email with another invitation to all of you to port (update to upcoming v3.2.0) your favorite tutorial from old wiki system (http://www.kamailio.org/dokuwiki/) and add to the new wiki (http://www.kamailio.org/wiki/). Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Dec 5-8, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 0

Cflag is not getting updated.
by Krishna Kurapati 03 Oct '11

03 Oct '11

Hi, I am hitting bit of a wall here. - I am setting bflags for NAT, and SIPPING before save("location") however, when I do "kamctl ul show", the Cflag is 0. I also checked see if the bflags are set or not in the register routine - they are being set. What am I missing? snippets of the config files. #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_method", "INFO") modparam("nathelper", "sipping_from", "sip:pinger@50.62.1.7") # params needed for NAT traversal in other modules modparam("nathelper|registrar|usrloc", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) # Caller NAT detection route route[NAT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (method=="REGISTER") { fix_nated_register(); } else { fix_nated_contact(); } setflag(FLT_NATS); setbflag(FLB_NATB); } #!endif return; } # Handle SIP registrations route[REGISTRAR] { if (is_method("REGISTER")) { xlog("L_ALERT", "REGISTER: From $fU, $fd To $tU, $td"); if (isflagset(FLT_NATS) || isbflagset(FLB_NATB)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging setbflag(FLB_NATSIPPING); } # 7/28/11 - make sure only one contact location is active # at any given time # unregister("location", "$rU"); if (isbflagset(FLB_NATB)) { xlog("L_ALERT", "REGISTRAR: For $fU, $fd FLB_NATB set"); } if (isbflagset(FLB_NATSIPPING)) { xlog("L_ALERT", "REGISTRAR: For $fU, $fd FLB_NATSIPPING set"); } if (!save("location")) { xlog("L_ALERT", "REGISTRAR: From $fU, $fd Location could not be saved"); sl_reply_error(); } exit; } } The debug log snippets ================== Oct 1 15:58:30 siptest /usr/sbin/kamailio[19515]: ALERT: <script>: REGISTER: From p_krishna_test_com, 50.62.1.7 To p_krishna_test_com, 50.62.1.7 Oct 1 15:58:30 siptest /usr/sbin/kamailio[19515]: ALERT: <script>: REGISTRAR: For p_krishna_test_com, 50.62.1.7 FLB_NATB set Oct 1 15:58:30 siptest /usr/sbin/kamailio[19515]: ALERT: <script>: REGISTRAR: For p_krishna_test_com, 50.62.1.7 FLB_NATSIPPING set Oct 1 15:58:30 siptest /usr/sbin/kamailio[19515]: DEBUG: sl [sl.c:278]: reply in stateless mode (sl) Output of kamctl command ==================== $ sudo kamctl ul show Domain:: location table=512 records=1 max_slot=1 AOR:: p_krishna_test_com Contact:: sip:p_krishna_test_com@75.62.17.45:50073;transport=tcp;ob Q= Expires:: 287 Callid:: ad2d3a7bf39a4345861db1661576a7f3 Cseq:: 54856 User-agent:: QliqDesktop 1.0.0 (pjproject 1.10.0/win32) State:: CS_NEW Flags:: 0 Cflag:: 0 <================= Cflag is not being set. Socket:: tcp:10.1.0.6:5060 Methods:: 8143 Thanks Krish Kura

1 3

Kamailio as dispatcher fails due to from domain on extension.
by Henrik Aagaard Sørensen 02 Oct '11

02 Oct '11

I'm using Kamailio as dispatcher to a FreeSwitch setup. Currently it "almost" work. I have 2 extensions connected to my FreeSwitch via Kamailio. However, I can only send calls one way because Kamailio seems to change the domain on the extension. One extension is registered with ID(a)sip.my-domain.com. This I can send calls to. The other extension is registered with ID@local-ip. This I cannot send calls to. Isn't it possible to have Kamailio register the extension on my FreeSwitch with @sip.my-domain.com? If I register both extension directly to my FreeSwitch everything works.

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Re: [SR-Users] Setting avp from load_credentials
by Alex Balashov 01 Oct '11

01 Oct '11

Alternately, if you dislike any of these paradigms or find the order of operations inconvenient, just roll your own. That's what sqlops is for. -- This message was painstakingly thumbed out on my mobile, so apologies for brevity, errors, and general sloppiness. Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ On Oct 1, 2011, at 10:43 AM, "graham(a)g-rock.net" <graham(a)g-rock.net> wrote: > I know I am Auth'ing users, but apparently need to confirm the location of www_authorize. Thanks for the pointers Alex. > > Sent from my HTC on the Now Network from Sprint! > > ----- Reply message ----- > From: "Alex Balashov" <abalashov(a)evaristesys.com> > Date: Sat, Oct 1, 2011 9:07 am > Subject: [SR-Users] Setting avp from load_credentials > To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) -Users Mailing List" <sr-users(a)lists.sip-router.org> > > Sure it is, but credentials only get loaded if you are doing proxy/www_authorize() beforehand. Are you? > > -- > This message was painstakingly thumbed out on my mobile, so apologies for brevity, errors, and general sloppiness. > > Alex Balashov - Principal > Evariste Systems LLC > 260 Peachtree Street NW > Suite 2200 > Atlanta, GA 30303 > Tel: +1-678-954-0670 > Fax: +1-404-961-1892 > Web: http://www.evaristesys.com/ > > On Oct 1, 2011, at 9:24 AM, Graham Wooden <graham(a)g-rock.net> wrote: > > > Hi there, > > > > Just doing a quick test and this doesn’t appear to be working. Kamailio > > 1.5.5-notls. Snippet of code: > > > > modparam("auth_db", "load_credentials", "username") > > . > > . > > . > > xlog("username is $avp(s:username)\n"); > > > > > > The result is '<null>'. Is this not available in the route{} ? > > > > Thanks, > > > > -graham > > > > > > > > > > > > _______________________________________________ > > 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 > >

2 1

Re: [SR-Users] Setting avp from load_credentials
by graham＠g-rock.net 01 Oct '11

01 Oct '11

I know I am Auth'ing users, but apparently need to confirm the location of www_authorize. Thanks for the pointers Alex. Sent from my HTC on the Now Network from Sprint! ----- Reply message ----- From: "Alex Balashov" <abalashov(a)evaristesys.com> Date: Sat, Oct 1, 2011 9:07 am Subject: [SR-Users] Setting avp from load_credentials To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) -Users Mailing List" <sr-users(a)lists.sip-router.org> Sure it is, but credentials only get loaded if you are doing proxy/www_authorize() beforehand. Are you? -- This message was painstakingly thumbed out on my mobile, so apologies for brevity, errors, and general sloppiness. Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ On Oct 1, 2011, at 9:24 AM, Graham Wooden <graham(a)g-rock.net> wrote: > Hi there, > > Just doing a quick test and this doesn’t appear to be working. Kamailio > 1.5.5-notls. Snippet of code: > > modparam("auth_db", "load_credentials", "username") > . > . > . > xlog("username is $avp(s:username)\n"); > > > The result is '<null>'. Is this not available in the route{} ? > > Thanks, > > -graham > > > > > > _______________________________________________ > 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

Setting avp from load_credentials
by Graham Wooden 01 Oct '11

01 Oct '11

Hi there, Just doing a quick test and this doesn¹t appear to be working. Kamailio 1.5.5-notls. Snippet of code: modparam("auth_db", "load_credentials", "username") . . . xlog("username is $avp(s:username)\n"); The result is '<null>'. Is this not available in the route{} ? Thanks, -graham

2 1

Use of $rU
by Graham Wooden 01 Oct '11

01 Oct '11

Hi there, What is pv for the equivalent of $rU for outbound? My inbound use of $rU has the correct username. All my subscribers are multi-number trunks, so on the outbound I need to associate the subscriber. This is kamailio 1.5. Thanks, -graham

2 3

Handling quota/call limits - kamailio 1.5
by Graham Wooden 01 Oct '11

01 Oct '11

Hi there, Finally getting around to introducing some call limits and starting with the inbound calls. However, I ran into an issue about how to get the "in use" count to be updated in my quota table. What is the best way to get s:in_use to update so it can be calculated? Here is my snippet of code: if (is_method("INVITE")) { avp_db_query("select quota, in_use from quota where username='$rU'","$avp(s:quota);$avp(s:in_use)"); if ($avp(s:in_use) >= $avp(s:quota)){ xlog ("DEBUG: Trunk $rU is over quota"); sl_send_reply("503","User over quota"); } . . } This opens up two more questions: How to decrease the call count when a BYE is received - maybe do a subtraction and the do a avp_db_query to update the table? And of course there is the BYE that won't be present (ie. When a call just rings and hangs up before it gets answered). Maybe ACC's missed_calls can somewhat be used to help determine? Thanks for any pointers and guidance. I am running Kamailio 1.5. Thanks, -graham

2 4

SNMPStats and multiple proxies
by Antanas Masevicius 30 Sep '11

30 Sep '11

Hello, been trying to run two sip-router proxies in parallel and to feed data from them to snmpd via agentx. But it looks like snmpd deamon directly routes snmp request just to one (probably first connected) proxy. What ways this can be solved? Currenly i am thinking like running two concurrent snmpd daemons binded to specific IP addresses. Is there a better way to do it? best regards, -- Antanas Masevicius

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Re: [SR-Users] Making iPhone SIP client work with Kamailio
by Krishna Kurapati 30 Sep '11

30 Sep '11

> > Hi, > > The iPhone client, when it goes into background, can only respond to > incoming messages. > That makes the Registration expire when there is no communication from the > server before > the expire time. I could make the Registration not expire at all. Not sure > it's the only option. > > Is there a way to send a Ping message from server periodically to the > iPhone SIP client. This > will wakeup the SIP client and resend REGISTER message. > > Thanks > Krish Kura >

2 2

Cleaning up presence info when user is deleted
by Krishna Kurapati 30 Sep '11

30 Sep '11

Hi, When a user is removed using "kamctl rm", is there way way to clean up presence information in the database (xcap, presentity, watcher and active watcher tables) and in the cache? Is there a event that gets fired when the user is removed from subscriber table which I could handle in config file and and cleanup? Thanks Krish Kura

3 3

Core db_text module
by Bruno Bresciani 30 Sep '11

30 Sep '11

Hi All, Kamailio 3.1.2 generate a core in db_text module when subscriber table is updated and after this action some SIP message require authentication process... Someone Can Help me to understand why this core is happening? Below is part of kamailio's trace and gdb core too. Kamailio's trace: *Sep 22 16:35:54 sswpst00 /home2/local/kamailio/sbin/kamailio[2327]: DEBUG: auth_db [authorize.c:239]: realm value [192.168.166.199]* Sep 22 16:35:54 sswpst00 /home2/local/kamailio/sbin/kamailio[2327]: DEBUG: auth [api.c:95]: auth: digest-algo: MD5 parsed value: 1 *Sep 22 16:35:54 sswpst00 /home2/local/kamailio/sbin/kamailio[2327]: DEBUG: db_text [dbt_file.c:78]: [subscriber] was updated* Sep 22 16:35:54 sswpst00 /home2/local/kamailio/sbin/kamailio[2324]: ALERT: <core> [main.c:741]: child process 2327 exited by a signal 11 Sep 22 16:35:54 sswpst00 /home2/local/kamailio/sbin/kamailio[2324]: ALERT: <core> [main.c:744]: core was generated Sep 22 16:35:54 sswpst00 /home2/local/kamailio/sbin/kamailio[2324]: INFO: <core> [main.c:756]: INFO: terminating due to SIGCHLD Sep 22 16:35:54 sswpst00 /home2/local/kamailio/sbin/kamailio[2335]: INFO: <core> [main.c:807]: INFO: signal 15 received c Sep 22 16:35:54 sswpst00 /home2/local/kamailio/sbin/kamailio[2335]: DEBUG: <core> [main.c:818]: Memory status (pkg): gdb core trace: Core was generated by `/home2/local/kamailio/sbin/kamailio -P /var/run/kamailio.pid'. Program terminated with signal 11, Segmentation fault. *#0 0x00f21603 in dbt_db_del_table (_dc=0xb6071e60, _s=0xbf837040, sync=0) at dbt_lib.c:238* 238 dbt_lib.c: Arquivo ou diretório não encontrado. in dbt_lib.c Best Regards

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Re: [SR-Users] simple question but can't find the answer
by Daniel-Constantin Mierla 30 Sep '11

30 Sep '11

Hello, On 9/30/11 12:11 AM, JJ Zeng wrote: > > Thanks Daniel. I followed your advice and install kamailio 3.1. Now > I get a weird problem: it works 1 out 5 times for the relaying. > try to set like tcp_connection_lifetime=3604 global parameter, or to a value higher than the longer registration time. > > The t_relay is where the the problem is, because I check the response > code and route(LOCATION) always succeeds, which means the entries > exist in "location". > > If I don't put set_forward_no_connect, then t_relay will take a long > time until my client time out. If I put it, then the client will get > 477 right away. > > BTW, the original config didn't work, either. That's why I started to > strip down some irrelevant stuff. > What errors you got with the default config? Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Dec 5-8, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 0

Re: [SR-Users] Placing ds_select_dst in kamailio.cfg, where?
by Daniel-Constantin Mierla 29 Sep '11

29 Sep '11

Hello, please keep the mainling list cc-ed. Calling ds_select_dst() first in the main route block is not really a good place. First you have to handle sanity checks, within dialog requests a.s.o. Of course it is a matter of your purpose, but as general hint, selecting the destination should be done just (or as close as possible) before relaying. If you take default config and you have only load balancing, then you can replace location lookup part with load balancing. Other parts may be stripped as well, just giving some direction here. Cheers, Daniel On 9/28/11 11:11 PM, Henrik Aagaard Sørensen wrote: > That did the job. > > I just placed the ds_select_dst-code at the very first thing after route{ > > Thank you for your help! > > 2011/9/28 Daniel-Constantin Mierla <miconda(a)gmail.com > <mailto:miconda@gmail.com>> > > Hello, > > the devel version of dispatcher module README has a different > config example, maybe it helps you more: > > http://kamailio.org/docs/modules/devel/modules_k/dispatcher.html#id2522847 > > I am considering to update it again soon to be a bit more complete > and follow the structure of the default config file shipped with > kamailio. > > Cheers, > Daniel > > > On 9/28/11 6:30 PM, Henrik Aagaard Sørensen wrote: >> Hi Carsten. >> >> The configuration works. It's actually where (which line) in the >> configuration I should put: >> ds_select_dst("1", "0"); >> >> :) I'm not strong in Kamailio and just want it to load balance >> between Freswitch servers. So I have a basic installation of >> Kamailio, without MySQL, NAT, Auth etc. >> >> On Wed, Sep 28, 2011 at 5:09 PM, Carsten Bock >> <carsten(a)ng-voice.com <mailto:carsten@ng-voice.com>> wrote: >> >> Hi, >> >> When you've got problems with placing "ds_select_dst()" in your >> config, you should do the following: >> - Open an Editor of your choice >> - go to the line, where you want to place it >> - type: ds_select_dst("1", "0"); >> - save & exit >> >> Just kidding! You should do the following with your config: >> kamailio -f <path-to-your-config> -c >> >> What error is displayed? If none, then check your syslog for >> errors. >> Could you post your config? It is better to put a full path >> in the >> config to the dispatcher-file... otherwise it might be >> dependent of, >> where you are running the init.d-skript/kamailio.... >> >> I am using dispatcher in Kamailio 1.5, 3.1 and 3.2, >> everythings works >> fine with me. >> >> Kind regards, >> Carsten >> >> 2011/9/28 Henrik Aagaard Sørensen >> <henrikaagaardsorensen(a)gmail.com >> <mailto:henrikaagaardsorensen@gmail.com>>: >> > I've installed 2 Freeswitch on different servers and would >> like to have >> > Kamailio load balance between them. >> > >> > I've installed Kamailio following: >> > >> http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-3.1.x-from-git >> > >> > I've created dispatcher.list and added: >> > >> > loadmodule "dispatcher.so" >> > >> > modparam("dispatcher", "list_file", "../etc/dispatcher.list") >> > >> > in the kamailio.cfg. >> > >> > But looking at: >> > >> http://kamailio.org/docs/modules/3.1.x/modules_k/dispatcher.html >> > I'm having difficulties to placing: >> > >> > ds_select_dst("1", "0"); >> > >> > in the kamailio.cfg that was created originally. >> > >> > Can anyone help me? >> > >> > _______________________________________________ >> > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users >> mailing list >> > sr-users(a)lists.sip-router.org >> <mailto:sr-users@lists.sip-router.org> >> > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> > >> > >> >> >> >> -- >> Carsten Bock >> CEO (Geschäftsführer) >> >> ng-voice GmbH i. Gr. >> Schomburgstr. 80 >> D-22767 Hamburg / Germany >> >> http://www.ng-voice.com >> mailto:carsten@ng-voice.com <mailto:carsten@ng-voice.com> >> >> Mobile +49 179 2021244 <tel:%2B49%20179%202021244> >> Office +49 40 34927219 <tel:%2B49%2040%2034927219> >> Fax +49 40 34927220 <tel:%2B49%2040%2034927220> >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users >> mailing list >> sr-users(a)lists.sip-router.org >> <mailto:sr-users@lists.sip-router.org> >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> >> >> >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >> sr-users(a)lists.sip-router.org <mailto:sr-users@lists.sip-router.org> >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > -- > Daniel-Constantin Mierla --http://www.asipto.com > Kamailio Advanced Training, Oct 10-13, Berlin:http://asipto.com/u/kat > http://linkedin.com/in/miconda -- http://twitter.com/miconda > > -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

3 3

Re: [SR-Users] Logmessage when Dispatcher-Module sets a dispatcher to probing
by Timo Klecker 29 Sep '11

29 Sep '11

Hey again, Btw: I am using Kamailio 3.1.5 and found the event_route in 3.2.x Dispatcher documentation. Is there a possibility in 3.1.x? Greetings Timo Von: sr-users-bounces(a)lists.sip-router.org [mailto:sr-users-bounces@lists.sip-router.org] Im Auftrag von Timo Klecker Gesendet: Donnerstag, 29. September 2011 11:10 An: 'kamailio users' Betreff: [SR-Users] Logmessage when Dispatcher-Module sets a dispatcher to probing Hey, I'd like to have a log-entry in the kamailio logfile when #ds_probind_threshhold OPTIONS pings fail to reach a dispatcher (408 localy generated after #fr_timer ms). Is it possible to do this without changing code? Greetings Timo

3 3

simple question but can't find the answer
by JJ Zeng 29 Sep '11

29 Sep '11

Hi, I am really new to ser(just installed it today). My situation is here: Both A and B make tcp connection to ser proxy server. If A sends "Invite" to B, ser would look up the tcp connection of B, and relay the sip request. If B is not connected, then 480 is returned to A. When B sends back the response, ser would relay it to A. It looks really simple, but I just can't figure out how to do it. Thanks. JJ

2 3

Logmessage when Dispatcher-Module sets a dispatcher to probing
by Timo Klecker 29 Sep '11

29 Sep '11

Hey, I'd like to have a log-entry in the kamailio logfile when #ds_probind_threshhold OPTIONS pings fail to reach a dispatcher (408 localy generated after #fr_timer ms). Is it possible to do this without changing code? Greetings Timo

1 0

OpenIMS
by Carlos Ruiz Diaz 29 Sep '11

29 Sep '11

Hello list. I'm trying to test/implement http://www.openimscore.org/ to setup an IMS scenario for development purposes. The project seems quite inactive to me and I wonder if there is any other similar project or this one just died and is no longer maintained. The documentation didn't specify if there is any kind of compatibility with Kamailio and based on what was written, only SER core architecture is supported. Any help will be appreciated. Regards. Carlos.

3 4

documentation update for 3.2.0 and wiki system
by Daniel-Constantin Mierla 28 Sep '11

28 Sep '11

Hello, in preparation for release 3.2.0, I was checking our wiki (on kamailio.org), which is quite rich in information, but some tutorials are for older versions. Maybe it is also a good time to do some restructuring along with the update of the useful tutorials to be valid for 3.2.0. For restructuring, the main goal is to provide some ways to let people find the documentation they need (and it is available there) easier. Some ideas, introducing categorization/tags for specific topics (e.g., radius, ldap, presence, asterisk, ...), introducing some index pages to link tutorials per version, etc. More suggestions are welcome here, let us know what will make your life easier in browsing the docs. On another hand, currently wiki is based on dokuwiki, which from my point of view did good job so far, but maybe we are missing another excellent wiki system that we should use, because provides more tools for this kind of purpose. One thing I was missing, was export to pdf, which I may be possible right now with dokuwiki plugins -- or even better, we can get something to export to docbook xml format with dokuwiki or other wiki system. In the same kind of subject, what about using git to store (or at least have a replica of) the wiki pages? IIRC, I saw somewhere that even for dokuwiki is possible to configure to use a git repository to store the wiki content instead of local CVS. This will allow people to mirror very easy the content. I could spot from time to time web spiders downloading the wiki. Comments or other suggestions to make the access to documentation easier? Thanks, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

2 1

Placing ds_select_dst in kamailio.cfg, where?
by Henrik Aagaard Sørensen 28 Sep '11

28 Sep '11

I've installed 2 Freeswitch on different servers and would like to have Kamailio load balance between them. I've installed Kamailio following: http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-3.1.x-from-git I've created dispatcher.list and added: loadmodule "dispatcher.so" modparam("dispatcher", "list_file", "../etc/dispatcher.list") in the kamailio.cfg. But looking at: http://kamailio.org/docs/modules/3.1.x/modules_k/dispatcher.html I'm having difficulties to placing: ds_select_dst("1", "0"); in the kamailio.cfg that was created originally. Can anyone help me?

3 3

Sip-router gentoo ebuild v3.1.5
by caio 28 Sep '11

28 Sep '11

Hello, Just want to announce the availability of an ebuild for version 3.1.5 (kamailio and ser flavours) on Gentoo Portage Overlays. Here the git link: * http://git.overlays.gentoo.org/gitweb/?p=user/caio.git;a=tree;f=net-misc/si… The unofficial overlays portage website: * http://gpo.zugaina.org/net-misc/sip-router Important, this is not an official ebuild, and neither it's not at official gentoo portage tree. You can use *layman* to add this overlay. Also at pkg/ser/gentoo path of sip-router project you can find the latest version. If you need any guidance to probe this ebuild under your gentoo linux do not hesitate to contact me or provide some feedback here at the lists. Best regards, Claudio

2 2

Accounting only the 2nd branch of missed serial forked call
by Ozren Lapcevic 27 Sep '11

27 Sep '11

Hi, I'm having some problems accounting missed serial forked calls to mysql database. I have following setup. Each user can have up to two contacts: telephone number (routed to asterisk) and SIP URI. Users can specify which contact has higher priority - which one should ring first. There is also SEMS voicemail which is forked as 3rd serial call leg if there is no answer at first two contacts. For example, I have two users: oz(a)abc.hr and pero(a)abc.hr. pero(a)abc.hr also has set telephone number as alternative number if he is not reachable at sip:pero@abc.hr. Moreover, pero(a)abc.hr has voicemail turned on. When oz(a)abc.hr calls pero(a)abc.hr, first pero(a)abc.hr's SIP client rings, then if there is no answer and after the timeout telephone number rings and finally, if there is no answer at telephone and after the timeout INVITE is forked to SEMS. There are two interesting scenarios accounting-wise which can happened: 1. oz(a)abc.hr calls pero(a)abc.hr, there are no answers and call is forked to voicemail. 2. oz(a)abc.hr calls pero(a)abc.hr, there is no answer at SIP client, but pero answers call at telephone. When scenario 1 happens, I want to have only one log (row) in missed_calls table. When scenario 2 happens, I don't want to have a log in missed_calls table. To accomplish this,* I want to log only the 2nd branch of the forked call. However, there is either a bug in acc module or I'm doing something wrong, and I can't get Kamailio to log only the 2nd branch*. I think that I am setting the FLT_ACCMISSED flag correctly - after the 2nd branch is handled and prior to calling the RELAY route. Logs show that FLT_ACCMISSED flag is set prior to calling t_relay(), and there are no errors in debug log. I am using $ru = "something" to rewrite URI prior to forking request. I can easily set up logging of every call (two missed calls for serially forked call to two locations) by setting FLT_ACCMISSED flag for each INVITE. I can set up logging of every call's 1st branch, by reseting FLT_ACCMISSED flag when handling 2nd branch of the call. Interestingly, logging of only the 2nd branch of the serial forked call works when there is no forking to voicemail! Any ideas how to solve this problem? Bellow are important parts of my config file. I'm running kamailio 3.1.4. Cheers Ozren # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 1) modparam("acc", "report_cancels", 0) modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) modparam("acc", "log_extra","src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") /* 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_user=$tU;dst_domain=$td;src_ip=$si") #!endif ... # Main SIP request routing logic # - processing of any incoming SIP request starts with this route route { # per request initial checks route(REQINIT); if (src_ip != ****) { # NAT detection route(NAT); } # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); # authentication route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } # dispatch requests to foreign domains route(SIPOUT); ### requests for my local domains # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations to PSTN route(PSTN); if ( is_method("INVITE") ) { route(DBALIASES); #check for user defined forking priorities and timers route(FORK); } # user location service route(LOCATION); route(RELAY); } #check for user defined forking priorities and timers route[FORK]{ sql_query("con", "select * from usr_pref_custom where uuid='$tu'", "pref"); $avp(uuid)=$dbr(pref=>[0,0]); $avp(email)=$dbr(pref=>[0,1]); $avp(prio1)=$dbr(pref=>[0,2]); $avp(prio2)=$dbr(pref=>[0,3]); $avp(timer1)=$dbr(pref=>[0,5]); $avp(timer2)=$dbr(pref=>[0,6]); if (strlen($avp(prio1))>5) { # user has multiple contacts, do serial forking setflag(FLT_USRPREF); # set counter if (!$avp(prio)) { $avp(prio) = 1; } # overwrite request URI with highest priority contact if ($avp(prio1) =~ "^sip:00") { $ru = $avp(prio1) + "@host"; xlog("L_INFO","PRIO 1 is tel number, RURI set: $ru"); } else { $ru = $avp(prio1); xlog("L_INFO","PRIO 1 is SIP URI, RURI set: $ru"); } } } 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_reply("REPLY_ONE"); t_on_failure("FAIL_ONE"); #if users have priorities set, use FAIL_FORK failure route if ( isflagset(FLT_USRPREF) ) { t_on_failure("FAIL_FORK"); } } if (isflagset(FLT_ACCMISSED)) xlog("L_INFO","RELAY, $rm $ru, ACCMISSED FLAG IS SET"); else xlog("L_INFO","RELAY, $rm $ru, ACCMISSED FLAG IS NOT SET"); if (!t_relay()) { sl_reply_error(); } 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()) { xlog("L_INFO","WITHINDLG, loose_route()"); if (is_method("BYE")) { xlog("L_INFO","WITHINDLG, BYE, DO ACCOUNTING"); 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; } } # USER location service route[LOCATION] { #skip if $ru is telephone number if ($ru =~ "^sip:00") { xlog("L_INFO","SKIP lookup..."); } else { if (!lookup("location")) { switch ($rc) { case -1: case -3: t_newtran(); t_reply("404", "Not Found"); exit; case -2: sl_send_reply("405", "Method Not Allowed"); exit; } } } # when routing via usrloc, log the missed calls also, but only if user doesn't have prios set if ( is_method("INVITE") && !(isflagset(FLT_USRPREF))) { setflag(FLT_ACCMISSED); } } # Failure route for forked calls failure_route[FAIL_FORK] { #!ifdef WITH_NAT if (is_method("INVITE") && (isbflagset(FLB_NATB) || isflagset(FLT_NATS))) { unforce_rtp_proxy(); } #!endif if ($avp(prio) >= 1) { $avp(prio) = $avp(prio) + 1; # handle 2nd branch if ( ($avp(prio) == 2) && ( isflagset(FLT_USRPREF) )) { t_on_failure("FAIL_FORK"); if ($avp(prio2) =~ "^sip:00") { xlog("L_INFO","FAIL FORK, PRIO 2 is tel number"); $ru = $avp(prio2) + "@host"; } else { xlog("L_INFO","FAIL FORK, PRIO 2 is SIP URI"); $ru = $avp(prio2); route(LOCATION); } setflag(FLT_ACCMISSED); } else { $avp(prio) = 0; $ru = $(avp(uuid)); rewritehostport("host:port"); xlog("L_INFO","FAIL FORK, VOICEMAIL email:$avp(email), ru:$ru, br: $br"); append_hf("P-App-Name: voicemail\r\n"); append_hf("P-App-Param: Email-Address=$avp(email)\r\n"); } route(RELAY); } if (t_is_canceled()) { exit; } }

2 21

Kamailio with IMS extensions build
by loic.trottin＠orange.com 27 Sep '11

27 Sep '11

Hi all, I try to compile Kamailio with IMS extensions from OpenIMSCore available in carstenbock/ims branch in the GIT repository. I have some problems with the modules "Client_Ro" and "pcscf" that do not compile: CC (gcc) [M Client_Ro.so] ims_ro.o ims_ro.c: In function 'send_ccr_interim': ims_ro.c:594: error: 'AVP_EPC_User_Equipment_Info_Type_MAC' undeclared (first use in this function) ims_ro.c:594: error: (Each undeclared identifier is reported only once ims_ro.c:594: error: for each function it appears in.) ims_ro.c: In function 'send_ccr_stop': ims_ro.c:749: error: 'AVP_EPC_User_Equipment_Info_Type_MAC' undeclared (first use in this function) ims_ro.c: In function 'Ro_Send_CCR': ims_ro.c:831: error: 'struct dlg_binds' has no member named 'get_dlg' ims_ro.c:835: error: 'struct dlg_binds' has no member named 'unref_dlg' ims_ro.c:886: error: 'AVP_EPC_User_Equipment_Info_Type_MAC' undeclared (first use in this function) CC (gcc) [M pcscf.so] pcc_avp.o pcc_avp.c:1459: error: 'AVP_IMS_Subscription_Id_Type_E164' undeclared (first use in this function) pcc_avp.c:1459: error: (Each undeclared identifier is reported only once pcc_avp.c:1459: error: for each function it appears in.) pcc_avp.c:1461: error: 'AVP_IMS_Subscription_Id_Type_SIP_URI' undeclared (first use in this function) Does anybody have encountered these problems? Is there a stable version tagged of the branch carstenbock/ims available in the GIT repository ? Best regards, Loïc. ******************************************************************************** IMPORTANT.Les informations contenues dans ce message electronique y compris les fichiers attaches sont strictement confidentielles et peuvent etre protegees par la loi. Ce message electronique est destine exclusivement au(x) destinataire(s) mentionne(s) ci-dessus. Si vous avez recu ce message par erreur ou s il ne vous est pas destine, veuillez immediatement le signaler a l expediteur et effacer ce message et tous les fichiers eventuellement attaches. Toute lecture, exploitation ou transmission des informations contenues dans ce message est interdite. Tout message electronique est susceptible d alteration. A ce titre, le Groupe France Telecom decline toute responsabilite notamment s il a ete altere, deforme ou falsifie. De meme, il appartient au destinataire de s assurer de l absence de tout virus. IMPORTANT.This e-mail message and any attachments are strictly confidential and may be protected by law. This message is intended only for the named recipient(s) above. If you have received this message in error, or are not the named recipient(s), please immediately notify the sender and delete this e-mail message. Any unauthorized view, usage or disclosure ofthis message is prohibited. Since e-mail messages may not be reliable, France Telecom Group shall not be liable for any message if modified, changed or falsified. Additionally the recipient should ensure they are actually virus free. ********************************************************************************

3 3

need little help to setting up pike module
by dirtytoni＠gmx.de 27 Sep '11

27 Sep '11

Hi to all on this list =) jUSt a tiny user question aboUt pike module(at least kamailio doc). are these values for sampling_time_unit seconds or minutes ? it's not in the doc. same 4 remove_latency. thanks for helping me out sorry to bother you best regards Toni -- Empfehlen Sie GMX DSL Ihren Freunden und Bekannten und wir belohnen Sie mit bis zu 50,- Euro! https://freundschaftswerbung.gmx.de

2 1

Account 403 to RADIUS
by Mino Haluz 26 Sep '11

26 Sep '11

Hi, I have this code: if ( is_user_in("From", "blocked") && is_method("INVITE")) { xlog("L_INFO", "XLOG: [number_and_ruri_checks] NOTICE: Account ($fu) to ($ru) is blocked"); sl_send_reply("403", " Account blocked "); exit; } I would like to account also this answer to Radius, now I have it only in application xlog. Thanks, Mino

3 5

Migration from v3.1.x to 3.2.0
by Daniel-Constantin Mierla 26 Sep '11

26 Sep '11

Hello, a kind reminder for everyone to add to the wiki page hints for migration from v3.1.x to v3.2.0. Especially to developers that know what kind of major changes they did to existing modules in 3.1.x. The wiki is open for anyone: * http://sip-router.org/wiki/install/3.1.x-to-3.2.x I just updated the section related to database structure changes. If you spot errors/missing parts, go ahead and fix it. Thanks, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 0

Re: [SR-Users] I am getting this error in the log When REGISTER message is received from jitsi
by Daniel-Constantin Mierla 23 Sep '11

23 Sep '11

On 9/22/11 9:04 PM, Krishna Kurapati wrote: > Thanks Daniel, > > Here it is... > > [...] It looked ok for my eye. Can you run kamailio with higher debug level (debug=4 in config file) and send the log messages related to parsing the URI for AoR? Cheers, Daniel > > > On Thu, Sep 22, 2011 at 1:44 PM, Daniel-Constantin Mierla > <miconda(a)gmail.com <mailto:miconda@gmail.com>> wrote: > > Hello, > > can you grab such REGISTER with ngrep (or wireshark) and past it here? > > Cheers, > Daniel > > > On 9/22/11 8:07 PM, Krishna Kurapati wrote: >> Sep 22 09:49:53 siptest /usr/sbin/kamailio[3160]: ERROR: >> registrar [common.c:75]: failed to parse Address of Record >> Sep 22 09:49:53 siptest /usr/sbin/kamailio[3160]: ERROR: >> registrar [save.c:822]: failed to extract Address Of Record >> >> Is there a problem when AoR contains "_" >> >> for example is AoR krish_kura(a)sip.org <mailto:krish_kura@sip.org> >> is not allowed? >> >> Thanks >> Krish Kura >> >> >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >> sr-users(a)lists.sip-router.org <mailto:sr-users@lists.sip-router.org> >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > -- > Daniel-Constantin Mierla --http://www.asipto.com > Kamailio Advanced Training, Oct 10-13, Berlin:http://asipto.com/u/kat > http://linkedin.com/in/miconda -- http://twitter.com/miconda > > -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

2 4

Re: [SR-Users] CANCEL is not processed in onsend_route
by Daniel-Constantin Mierla 23 Sep '11

23 Sep '11

Hello, On 9/15/11 4:02 PM, Timo Klecker wrote: > Hi Daniel, > > I managed to get a log with debug 4 from one call. I had to take out all > IP-Addresses and Usernames, hope you can use it. the log confirmed what I expected: Sep 15 15:32:33 vux896 /sbin/kamailio[31594]: DEBUG: siptrace [siptrace.c:954]: no uas msg, local transaction Sep 15 15:32:33 vux896 /sbin/kamailio[31594]: DEBUG: tm [t_fwd.c:1242]: DEBUG: e2e_cancel: e2e cancel proceeding Because a new CANCEL is generated from scratch, incoming message is lost and no longer available to check the sip trace flag. Now, the question is how to decide whether the cancel is to be siptraced. I can lookup for incoming CANCEL and check if it has the siptrace flag set, but there are CANCELs in case of local timeout or parallel forking, so I think it is better to actually lookup the INVITE transaction and if the siptrace flag is set for it, record the cancel. If there is no INVITE transaction anymore, the cancel is forwarded stateless and can be captured in onsend route, although it should be normally discarded if it is known that the proxy works in statefull mode only. Any other opinions? Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

2 2

carrierroute module
by Mino Haluz 23 Sep '11

23 Sep '11

Hi, I have a problem with configuring the carrierroute module, I do not clearly understand how it should work from the documentation. What I would like to do is that I have 2 gateways (A.A.A.A and B.B.B.B), lots of scan_prefixes with different rewrite prefixes. The traffic should be sent to the 1 gateway, and if it fails (5xx,4xx codes) it should go to the another. I have only one SIP domain (domain.com). The question is, how the tables (domain_name, carrierroute, carrierfailroute) should be filled? Can I achieve this scenario with the carrierroute module ? Thanks. Mino

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XCAP over SIP
by Krishna Kurapati 23 Sep '11

23 Sep '11

Hi, The document http://kb.asipto.com/kamailio:presence:k31-made-simple describes that XCAP can be sent over SIP instead of over HTTP. I tried to do however, the server is unable to update presence rules. I was using PUBLISH method. I was using content-type = text... Has anyone used SIP to send over XCAP? Appreciate if someone could share a link or configuration changes required. Thanks Krish

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I am getting this error in the log When REGISTER message is received from jitsi
by Krishna Kurapati 22 Sep '11

22 Sep '11

Sep 22 09:49:53 siptest /usr/sbin/kamailio[3160]: ERROR: registrar [common.c:75]: failed to parse Address of Record Sep 22 09:49:53 siptest /usr/sbin/kamailio[3160]: ERROR: registrar [save.c:822]: failed to extract Address Of Record Is there a problem when AoR contains "_" for example is AoR krish_kura(a)sip.org is not allowed? Thanks Krish Kura

2 1

perl module compilation on centos
by Daniel-Constantin Mierla 22 Sep '11

22 Sep '11

Hello, while building the v3.1.5 RPMs, the perl module failed to compile on centos 5 and 6, with following error: error: `PL_unitcheckav' undeclared (first use in this function ) so its packaging was disabled. It works fine on opensuse and fedora. Someone on twitter tried with the package perl-ExtUtils-Embed (like in fedora) and worked. However, OpenSUSE Build System fails to find that package. Is anyone here with latest centos 5 or 6 that can try compilation of perl module in v3.1.5 and let me know the result? Few months ago, it didn't fail for v3.1.4. Thanks, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

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Re: [SR-Users] Kamailio and FreeSWITCH realtime integration, tutorial?
by Henrik Aagaard Sørensen 22 Sep '11

22 Sep '11

But FreeSwitch don't use MySQL like Asterisk? So adapting isn't that easy or...??? On 21/09/2011, at 16.18, David <david(a)styleflare.com> wrote: You can adapt pretty easily. On 9/21/11 3:46 PM, Henrik Aagaard Sørensen wrote: Does anyone know if there somewhere exists a tutorial about Kamailio and FreeSWITCH realtime integration? I have Googled a lot and found: http://kb.asipto.com/kamailio:index and http://kb.asipto.com/freeswitch:index On the same site there is a tutorial for Kamailio and Asterisk realtime integration, but not for FreeSWITCH. _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

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Re: [SR-Users] Problem with REGISTER retransmissions
by Daniel-Constantin Mierla 22 Sep '11

22 Sep '11

Hello, I tried to reproduce with a very simple config to handle register requests. So I sent a REGISTER with expires 60 (very much the same as the one in the trace you sent), waited a while, then sent two REGISTERs having the CSeq incremented. All went fine. I tested with master and 3.1 branches (so the last is actually 3.1.5). Can you upgrade to 3.1.5 first and try again? If still fails, then update config to print the return code for save() function, something like: $var(r) = save("location"); xlog("code returned by save is $var(r)\n"); if($var(r) < 0) send_reply("500", "Failure"); Let me know the value of the returned code for the register that does not have a contact in reply. Cheers, Daniel On 9/20/11 2:10 PM, Alejandro Mingo wrote: > > Hi Daniel-Constantin, > > I send you the PCAP and LOG files corresponding to one of these > situations. As you can see in the LOG file, one thread (5444) handles > the first REGISTER, other thread (5447) handles the second REGISTER > (retransmission) and a third thread (5451) indicates that the binding > has expired, six seconds after the arrival of both REGISTERs. > > Thanks again. > > Alejandro. > > > > El 20/09/2011 10:54, Daniel-Constantin Mierla escribió: >> Hello, >> >> On 9/20/11 10:13 AM, Alejandro Mingo wrote: >>> Hi, >>> >>> Since we installed last version of Kamailio (3.1.4) >> >> the last version is now 3.1.5, but this is not really the most >> important aspect. However, it is recommended to upgrade, there were >> some fixed to registration handling when register requests have >> multiple contacts. >>> we have been experiencing a big problem with REGISTER >>> retransmissions. When the server receives a retransmitted REGISTER >>> it removes the binding and the UAC remains unregistered until next >>> refreshing period. I include a PCAP capture that shows this behaviour. >>> >>> I am not 100% sure that this problem never happened with older >>> versions. >>> >>> Has anybody experienced the same problem? >> I will try to reproduce and investigate myself. Anyhow, meanwhile >> maybe you can provide more hints faster. Do you have error messages >> in the syslog? Can you run with debug=4 in config file and send the >> output in syslog for such a case? -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

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Kamailio 3.1.5 RPMS
by Daniel-Constantin Mierla 22 Sep '11

22 Sep '11

Hello, just to let you know that the RPMs of Kamailio 3.1.5 were generated for CentOS, Fedora, OpenSUSE and RedHat. This time CentOS RPMs are packaged for both versions 5 and 6. Links are available on our wiki at: * http://www.kamailio.org/dokuwiki/doku.php/packages:rpms Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

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Kamailio and FreeSWITCH realtime integration, tutorial?
by Henrik Aagaard Sørensen 22 Sep '11

22 Sep '11

Does anyone know if there somewhere exists a tutorial about Kamailio and FreeSWITCH realtime integration? I have Googled a lot and found: http://kb.asipto.com/kamailio:index and http://kb.asipto.com/freeswitch:index On the same site there is a tutorial for Kamailio and Asterisk realtime integration, but not for FreeSWITCH.

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Re: [SR-Users] Dialog module showing calls that have already been terminated
by Timo Reimann 21 Sep '11

21 Sep '11

Hey Phillip, On 20.09.2011 13:48, Phillman25 Kyriacou wrote: > Thanks for your email. > > Yes dlg_manage(); has to now be called on INVITE and BYE/CANCEL messages. > Where would i have to call loose_route()? Only on INVITE? On *all* in-dialog requests, i.e., all requests which contain a To tag. This may include Re-INVITEs too (but not initial INVITEs). > My configuration did not change between 3.1.2 and 3.1.5. > > Call flow example: > ============== > > Cisco PGW ===> Kamailio 3.1.5 ===> VOIP PROVIDER or ASTERISK PABX > > > The below is my configuration. Let's take a look at it: > #!KAMAILIO > # [...] > if(is_method("BYE|CANCEL")) > > { > > dlg_manage(); > > > > # per request initial checks > route(REQINIT); > > # NAT detection > route(NAT); > > # handle requests within SIP dialogs > route(WITHINDLG); [...] > # 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()) { [...] So route[WITHINDLG] contains the logic to track in-dialog requests. However, you seem to call that route only from BYE and CANCEL requests, missing out ACKs, which is likely the reason why things go wrong. So make sure you run all in-dialog requests through that route, and you should be fine (hopefully). Cheers, --Timo

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Kamailio Development Training, San Francisco Area, October 2011
by Daniel-Constantin Mierla 21 Sep '11

21 Sep '11

Hello, since I am attending GSoC'11 Mentors Summit and come to Bay Area again, it is time to bring the developer training in USA, therefore I am planning to organize a two-day seminar focused on developing Kamailio extensions in C. Among topics to be approached: - internal architecture - SIP parser - memory manager - locking manager - database API - config file language interpreter - RPC interface - pseudo-variables and transformations framework - module interface – write your own extensions in C as modules - inter-module APIs - tm, presence, a.s.o. - documentation docbook format This kind of training follows the pattern from the past, trying to keep a low attendee fee, targeting just to cover the expenses with the organization. I am discussing with several people for a good location at a convenient price, however, I am open for new alternatives, including companies/institutes that want to host the event at no cost in exchange of sending few attendees free of charge. Contact me if you can help here. Training is planned for October 24-25, right now the estimated fee per participant is 350USD (it will not exceed 400USD). The policy to accept people is first come, first served. You can register at this time to reserve your seat(s), emailing to me or to registration(a)kamailio.org. Number of seats is usually 10 to 20, last one organized in February in Barcelona filled up quickly. Once there will be enough interested people and a location selected, I will have more precise details regarding the participation to this seminar and I will send updates. Looking forward to meeting many of you and get new US developers on board of the project in the near future. Cheers, Daniel -- http://www.kamailio.org/w/daniel-constantin-mierla/

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how to remove media description from sdp?
by Juha Heinanen 21 Sep '11

21 Sep '11

trying to hack around the possible fix_nated_sdp("1") issue that i outlined in previous message, i thought to remove video media description from sdp. in sdpops module there is sdp_with_media(type), but for some reason no sdp_remove_media(type) function. is there some other means to get rid of sdp video media description? -- juha

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Digest authentication
by Morten Isaksen 20 Sep '11

20 Sep '11

Hi, Sorry for the slightly off topic question, but I am trying to debug a buggy PBX and want to check if the digest information is correct. I have this perl script. #!/usr/bin/perl use strict; use Digest::MD5 qw(md5 md5_hex md5_base64); my $uri = 'sip:sip1.uni-tel.dk'; my $method = "SIP/2.0"; my $username = "69102208"; my $password = "password"; my $nonce = "4e78d2b0a72196d9123be7c88d093bd74c77f35e"; my $realm = "sip1.uni-tel.dk"; my $a1 = "$username:$realm:$password"; my $ha1 = md5_hex($a1); my $a2 = "$method:$uri"; my $ha2 = md5_hex($a2); print "A1 : $a1\n"; print "A2 : $a2\n"; my $response = md5_hex("$ha1:$nonce:$ha2"); print "\nRES : $response\n\n"; I registered with a softphone and entered the above values from the Authentication header, but I cannot get the response to match. Any help would be appreciated. -- Morten Isaksen

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wt_timer, delayed replies and sendto error
by Andreas Granig 20 Sep '11

20 Sep '11

Hi, Another funny thing I encountered lately is this: Client sends INVITE via kamailio (via dispatcher) to sems, gets back 100/200, sends another couple of INVITE with same CSeq (looks like retransmissions), gets back more 200, then sends ACKs for each of the 200 it got. Looks like a quite broken client to me, but anyways. The interesting thing is that after >5000ms (which is bigger then tm.wt_timer), another 200 is sent from sems to kamailio to the client. I see the 200 coming to kamailio in a network trace, but not an according log message in the kamailio logs. What I see though is this: Sep 14 13:51:26 sp2 /usr/sbin/kamailio[15950]: ERROR: <core> [udp_server.c:586]: ERROR: udp_send: sendto(sock,0x97ec10,1055,0,1.2.3.4:58620,16): Invalid argument(22) Sep 14 13:51:26 sp2 /usr/sbin/kamailio[15950]: ERROR: <core> [forward.h:149]: msg_send: ERROR: udp_send failed The 1.2.3.4:58620 is obviously taken from the Vias, which looks like this: Via: SIP/2.0/UDP 127.0.0.1:5060;rport=5060;branch=xyz Via: SIP/2.0/UDP 10.1.1.9:58620;received=1.2.3.4;rport=58620;branch=abc The top-most Via is itself, the second is the one which reflects the destination address in the error message above. So I'm wondering whether there is some issue with finding the proper socket after wt_timer is expired (note that the leg client<->kamailio uses a different socket (with public ip) than kamailio<->sems (which uses 127.0.0.1)). Or any other idea where this error could come from? It's actually an educated guess that this could be related to wt_timer, but I don't know what else it could be. Andreas

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Re: [SR-Users] Dialog module showing calls that have already been terminated
by Timo Reimann 20 Sep '11

20 Sep '11

Hey Phillip, you probably forgot to keep the mailing list in CC, I'll do that for you. On 19.09.2011 08:43, Phillman25 Kyriacou wrote: > You are right. > > I added the function dlg_manage(); when there is a BYE or CANCEL and now > its working perfectly. Before i had this function defined only on > INVITES. The strange thing is that it was working normally before when > dlg_manage(); was defined only for INVITE messages in the route block. So you are saying that before, you called dlg_manage() on INVITEs only but now, you need that function on both INVITE and BYE/CANCEL messages? That'd still qualify as a bug as the trigger to tracking the dialog (i.e., dlg_manage()) is required only once. After that, the dialog module makes sure that all dialog-related messages are properly processed. The only thing that's necessary for sequential requests to be tracked correctly is a call to loose_route() on such messages. If you are sure that your Kamailio configuration didn't change between 3.1.2 and 3.1.5 w.r.t. dialog tracking or anything that could affect it, I'd be willing to take a closer look at this issue (giving you can provide me with more information on your scenario, like call flow examples or similar). Cheers, --Timo > The calls that were suppose to be terminated before where in state 3 to > answer your question. However, i still see calls in state 3 now but they > are valid. > > Regards > Phillip > > On Fri, Sep 16, 2011 at 8:01 PM, Timo Reimann <timo.reimann(a)1und1.de > <mailto:timo.reimann@1und1.de>> wrote: > > Hey Phillip, > > after looking closer at what has been reported to me initially, your > case may be different after all. > > Could you possibly use "kamctl fifo dlg_list" to check what state the > majority of dialogs that should be terminated in your opinion are in? > Being given the reference counter should be useful in debugging this as > well. > > > Thanks, > > --Timo > > > > On 16.09.2011 14:22, Phillman25 Kyriacou wrote: > > On Fri, Sep 16, 2011 at 3:09 PM, Timo Reimann > <timo.reimann(a)1und1.de <mailto:timo.reimann@1und1.de> > > <mailto:timo.reimann@1und1.de <mailto:timo.reimann@1und1.de>>> wrote: > > > > Hey Phillip, > > > > > > On 16.09.2011 13:35, Phillman25 Kyriacou wrote: > > > Hello > > > > > > I have been facing an issue where the dialog module is showing > > calls as > > > being active when in fact the call has already been > completed long ago > > > and this is giving wrong number of concurrent calls on our > SNMP work > > > station (CACTI) when polling the data from Kamailio. I > realized this > > > only started occurring after i upgraded from 3.1.2 to 3.1.5, has > > anyone > > > experienced the same issue? > > > > I was *just* being notified of issues concerning dialogs not being > > deleted. Working on this right now to report back soon. > > > > Thanks for the note! > > > > > > Cheers, > > > > --Timo

2 1

Add client certificate checking against a CRL in opensips
by tomsc 20 Sep '11

20 Sep '11

Hi everyone, I'd like to check that a client certificat is revoked or not against a crl. Actually, opensips use context SSL_CTX. How can I do with this context? I do this change to load the crl : load_crl(SSL_CTX * ctx, char *filename) { LM_DBG("entered load crl\n"); X509_STORE *pStore = SSL_CTX_get_cert_store(ctx); X509_LOOKUP* plookup = X509_STORE_add_lookup(pStore, X509_LOOKUP_file()); if (!X509_load_crl_file(plookup, filename, X509_FILETYPE_PEM)){ LM_ERR("unable to load certificate file '%s'\n", filename); return -1; } LM_DBG("'%s' successfuly loaded\n", filename); return 0; } But I dont know how to add flags, is it the right solution? Thanks all -- View this message in context: http://old.nabble.com/Add-client-certificate-checking-against-a-CRL-in-open… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

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Lazy evaluation in if expressions?
by Morten Isaksen 20 Sep '11

20 Sep '11

Does Kamailio use lazy evaluation in if expressions? Like if ($var(a) == 2 && $var(b) ==3) { ... } then $var(b) == 3 is only evaluated if $var(a) == 2 is true -- Morten Isaksen

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directed media problem when video is included in sdp
by Juha Heinanen 20 Sep '11

20 Sep '11

when sems receives voice call invite from ua behind nat that has direction:active in its sdp, like this: Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): foo.bar 0 0 IN IP4 192.168.1. 68 Session Name (s): - Connection Information (c): IN IP4 192.168.1.68 Time Description, active time (t): 0 0 Media Description, name and address (m): audio 8500 RTP/AVP 18 4 112 100 101 0 8 106 Media Attribute (a): sendrecv Media Attribute (a): rtpmap:18 G729/8000 Media Attribute (a): rtpmap:4 G723/8000 Media Attribute (a): rtpmap:112 AMR/8000 Media Attribute (a): rtpmap:100 EG711U/8000 Media Attribute (a): rtpmap:101 EG711A/8000 Media Attribute (a): rtpmap:0 PCMU/8000 Media Attribute (a): rtpmap:8 PCMA/8000 Media Attribute (a): rtpmap:106 telephone-event/8000 Media Attribute (a): direction:active sems includes direction:passive in 200 OK and starts to correctly send rtp packets to the ip address/port of received rtp packets. however, then sems receives a voice+video call invite like this: Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): foo.bar 0 0 IN IP4 192.168.1.68 Session Name (s): - Connection Information (c): IN IP4 192.168.1.68 Time Description, active time (t): 0 0 Media Description, name and address (m): audio 8500 RTP/AVP 18 4 112 100 101 0 8 106 Media Attribute (a): sendrecv Media Attribute (a): rtpmap:18 G729/8000 Media Attribute (a): rtpmap:4 G723/8000 Media Attribute (a): rtpmap:112 AMR/8000 Media Attribute (a): rtpmap:100 EG711U/8000 Media Attribute (a): rtpmap:101 EG711A/8000 Media Attribute (a): rtpmap:0 PCMU/8000 Media Attribute (a): rtpmap:8 PCMA/8000 Media Attribute (a): rtpmap:106 telephone-event/8000 Media Description, name and address (m): video 8600 RTP/AVP 34 Bandwidth Information (b): AS:300 Media Attribute (a): sendrecv Media Attribute (a): rtpmap:34 H263/90000 Media Attribute (a): direction:active it does not include direction:passive in 200 OK and keeps on sending rtp packets to 192.168.1.68 even after receiving first voice rtp packet from public ip of the ua. is this a bug in sems or is it a bug in sr nathelper_k fix_nated_sdp("1") implementation, i.e., does that function place direction:active line to only one place, when it should place it in two places (in both voice and video media descriptions)? -- juha

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Module db_text
by Bruno Bresciani 20 Sep '11

20 Sep '11

Hi All, I'm starting kamailio version 3.1.2 with db_text module but when it tries to load route table, the following error is generated: * "ERROR: <core> [daemonize.c:273]: Main process exited before writing to pipe" * Someone Can Help me to understand why this error is happening? Below is part of kamaio's trace. Sep 19 16:13:01 sswpst00 /usr/local/sbin/kamailio[25661]: DEBUG: db_text [dbt_base.c:208]: _tbc->name: route Sep 19 16:13:01 sswpst00 /usr/local/sbin/kamailio[25661]: DEBUG: db_text [dbt_base.c:209]: _tbc->nrcols: 9 Sep 19 16:13:01 sswpst00 /usr/local/sbin/kamailio[25661]: DEBUG: db_text [dbt_base.c:210]: _nc: 8 Sep 19 16:13:01 sswpst00 /usr/local/sbin/kamailio[25661]: DEBUG: db_text [dbt_base.c:238]: new res with 8 cols Sep 19 16:13:01 sswpst00 /usr/local/sbin/kamailio[25661]: DEBUG: db_text [dbt_res.c:69]: new res with 8 cols Sep 19 16:13:01 sswpst00 /usr/local/sbin/kamailio[25661]: DEBUG: <core> [db_res.c:118]: allocate 28 bytes for result set at 0x82e23f0 Sep 19 16:13:01 sswpst00 /usr/local/sbin/kamailio[25661]: DEBUG: <core> [db_res.c:155]: allocate 32 bytes for result names at 0x82e3798 Sep 19 16:13:01 sswpst00 /usr/local/sbin/kamailio[25661]: DEBUG: <core> [db_res.c:165]: allocate 32 bytes for result types at 0x82e37c0 Sep 19 16:13:01 sswpst00 /usr/local/sbin/kamailio[25661]: DEBUG: db_text [dbt_api.c:80]: allocate 8 bytes for RES_NAMES[0] at 0x82e2438 Sep 19 16:13:01 sswpst00 /usr/local/sbin/kamailio[25661]: DEBUG: db_text [dbt_api.c:80]: allocate 8 bytes for RES_NAMES[1] at 0x82e2390 Sep 19 16:13:01 sswpst00 /usr/local/sbin/kamailio[25661]: DEBUG: db_text [dbt_api.c:80]: allocate 8 bytes for RES_NAMES[2] at 0x82e23d0 Sep 19 16:13:01 sswpst00 /usr/local/sbin/kamailio[25661]: DEBUG: db_text [dbt_api.c:80]: allocate 8 bytes for RES_NAMES[3] at 0x82e23e0 Sep 19 16:13:01 sswpst00 /usr/local/sbin/kamailio[25661]: DEBUG: db_text [dbt_api.c:80]: allocate 8 bytes for RES_NAMES[4] at 0x82e37e8 Sep 19 16:13:01 sswpst00 /usr/local/sbin/kamailio[25661]: DEBUG: db_text [dbt_api.c:80]: allocate 8 bytes for RES_NAMES[5] at 0x82e37f8 Sep 19 16:13:01 sswpst00 /usr/local/sbin/kamailio[25661]: DEBUG: db_text [dbt_api.c:80]: allocate 8 bytes for RES_NAMES[6] at 0x82e3808 Sep 19 16:13:01 sswpst00 /usr/local/sbin/kamailio[25661]: DEBUG: db_text [dbt_api.c:80]: allocate 8 bytes for RES_NAMES[7] at 0x82e3818 Sep 19 16:13:01 sswpst00 /usr/local/sbin/kamailio[25661]: DEBUG: <core> [db_res.c:184]: allocate 8 bytes for rows at 0x82e3828 Sep 19 16:13:01 sswpst00 /usr/local/sbin/kamailio[25661]: DEBUG: <core> [db_row.c:119]: allocate 160 bytes for row values at 0x82e3838 Sep 19 16:13:01 sswpst00 kamailio: ERROR: <core> [daemonize.c:273]: Main process exited before writing to pipe Best Regards

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Re: [SR-Users] rtpproxy VS Turn
by Mikael Williams 19 Sep '11

19 Sep '11

> > ICE is not supported as I select the Media Proxy (which is TURN) manually > in some Nat cases...Even note that I do not have peer to peer calls between > 2 endpoints, just have VOIP to PSTN calls...All my VOIP customers supports > STUN/TURN but without ICE > > One more thing: I would like to ask how rtp proxy help to bypass NAT else > than working as relay server...Does it have any other job like find the NAT > type or check blocked port or something? > > Regards > > > On Fri, Sep 16, 2011 at 5:50 PM, Klaus Darilion < > klaus.mailinglists(a)pernau.at> wrote: > >> STUN alone is not 100% sufficient, only ICE+TURN is reliable. Thus, only >> if all your clients support ICE+TURN then you can drop rtpproxy. >> >> klaus >> >> >> On 16.09.2011 16:47, Mikael Williams wrote: >> >>> Dear All, >>> >>> I would like to ask please if and in case I'm using STUN/TURN for Nat >>> traversal to drp the rtpproxy from my solution as NAT traversal and >>> relay server as I'm using SYUN for NAT traversal and TURN as relay in >>> case of Symmetric NAT >>> >>> Regards >>> >>> >>> ______________________________**_________________ >>> 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<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users> >>> >> >

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Re: [SR-Users] sipp to kamailio doesn't match ACK [SOLVED]
by Uri Shacked 19 Sep '11

19 Sep '11

Hi, I got the same problem. Do you have the XML SIPp file for example? 10x Uri

1 0

Dialog module showing calls that have already been terminated
by Phillman25 Kyriacou 16 Sep '11

16 Sep '11

Hello I have been facing an issue where the dialog module is showing calls as being active when in fact the call has already been completed long ago and this is giving wrong number of concurrent calls on our SNMP work station (CACTI) when polling the data from Kamailio. I realized this only started occurring after i upgraded from 3.1.2 to 3.1.5, has anyone experienced the same issue? my config: ======== loadmodule "dialog.so" . . . . # ----------------- DIALOG MODULE PARAMETERS----------------------------------# modparam("dialog", "enable_stats", 1) modparam("dialog", "hash_size", 4096) modparam("dialog", "rr_param", "did") modparam("dialog", "dlg_flag", 4) modparam("dialog", "timeout_avp", "$avp(i:10)") modparam("dialog", "dlg_extra_hdrs", "NULL") modparam("dialog", "dlg_match_mode", 1) modparam("dialog", "detect_spirals", 1) modparam("dialog", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("dialog", "db_mode", 1) modparam("dialog", "db_update_period", 60) modparam("dialog", "db_fetch_rows", 500) modparam("dialog", "table_name", "dialog") modparam("dialog", "from_uri_column", "from_uri") modparam("dialog", "from_tag_column", "from_tag") modparam("dialog", "to_uri_column", "to_uri") modparam("dialog", "to_tag_column", "to_tag") modparam("dialog", "h_id_column", "hash_id") modparam("dialog", "h_entry_column", "hash_entry") modparam("dialog", "state_column", "state") modparam("dialog", "start_time_column", "start_time") modparam("dialog", "timeout_column", "timeout") modparam("dialog", "sflags_column", "sflags") modparam("dialog", "bridge_controller", "sip:controller@kamailio.org") modparam("dialog", "default_timeout", 7200) . . . . route { # MANAGE ALL DIALOGS #=================================================== if (is_method("INVITE")) { if(is_method("INVITE") && !has_totag()) { $dlg_ctx(timeout_route) = 12; $dlg_ctx(timeout_bye) = 1; } dlg_manage(); } Thanks Phillip

2 3

kamailio debian/ubuntu support
by Jon Bonilla 16 Sep '11

16 Sep '11

Hi all with the release of kamailio 3.1.5, I've done a small change to the debian repositories: Nightly builds of kamailio_3.0 branch have been dropped and the repository has been erased. There is no development in this branch and it has no sense to keep nightly builds for it. Those (few) people using kamailio_3.0-nightly repositories are advised to move their sources.list to the stable 3.0 repository of upgrade to 3.1 or master branches. kamailio 3.0 will continue to be packaged if any new stable version is released for etch, lenny, squeeze and lucid. For 3.1 branch, nightly builds and stable repos will be provided for lenny, lucid and squeeze When the time for 3.2 (master branch at the moment) comes, I'll add Wheezy support. Not sure about Ubuntu distro support. Should we add support to Ubuntu 11.10 or wait until the lts 12.04 is released? Any opinions welcomed cheers, Jon

4 4

rtpproxy VS Turn
by Mikael Williams 16 Sep '11

16 Sep '11

Dear All, I would like to ask please if and in case I'm using STUN/TURN for Nat traversal to drp the rtpproxy from my solution as NAT traversal and relay server as I'm using SYUN for NAT traversal and TURN as relay in case of Symmetric NAT Regards

2 1

Communication from IPv6 to Ipv4
by Loïc Fofana 16 Sep '11

16 Sep '11

My architecture is described below: TDM device----------- Call server[IPv4]---------------[IPv4]Kamailio[IPv6]------------[IPv6]Asterisk---------[IPv6]Linphone * * * * * * * * * * * * Media Server Rtpproxy Media Server and Rtpptoxy carry the media. Call Server carry only signaling. I have succeeded to establish audio communication between an IPv6 domain & an IPv4 domain i.e. : from an TDM device to a Linphone. However, When i try to establish a call from (IPv6 to IPv4), i.e. from Linphone to TDM device, the signaling is OK, but there's no voice. Because, Rtpproxy forward the media to the Call server instead the Media Server. Hence, there can't be voice. I have been stucking on this for a couple of months. I will be very glad if somebody have any idea. best regards!

2 1

Re: [SR-Users] Kamailio terminating after upgrading from 3.1.2 to 3.1.4
by Phillman25 Kyriacou 16 Sep '11

16 Sep '11

Dear Marius Thank you for your prompt reply. I apologise, As i am a newbie could you please explain how to produce the below output that you require? Thanking you in advance for your help! Regards Phillip > Hello > Hello. You should have a coredump . Can you please send the trace (bt full) as seen with gdb ? Thanks Marius > Im facing a big problem after upgrading from version 3.1.2 to 3.1.4. > The kamailio process keeps on terminating every now and then, this is > what i was able to retrieve from my logs. > > > Sep 14 12:52:31 SipProxy1 kernel: [71448.062089] __ratelimit: 15 > callbacks suppressed > Sep 14 12:52:31 SipProxy1 kernel: [71448.062096] kamailio[13963]: > segfault at 14 ip 0816452a sp bfedc940 error 4 in kamailio[8048000+1d9000] > > > Does anyone have an idea why this is happening? > > Your help is much appreciated! > > Regards > Phillip

5 9

(no subject)
by Loïc Fofana 16 Sep '11

16 Sep '11

This is my scenario: Call server -------------(IPV4)Kamailio(IPV6)----------Asterisk(IPV6)------Linphone(IPV6) * * * Media server

2 1

One way communications in Kamailio 3.1.4
by CaptWho 16 Sep '11

16 Sep '11

I'm running 3.1.4 on centos and I'm having some trouble with voice only going one way. Both extensions will ring each other, but after they connect, voice will only travel in one direction. One extension hears fine, but can't talk. I've totally opened up the firewalls (including port 5060) and I'm still having the trouble. I'm not behind NAT. I've tried it using X-Lite, VoIP phones and ATAs in a number of combinations. The problem is across the board. X-Lite sometimes automatically disconnects after 32 seconds. Does anyone have any suggestions on where to look? ANY help appreciated. This has been going on for weeks. -- View this message in context: http://old.nabble.com/One-way-communications-in-Kamailio-3.1.4-tp32460597p3… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

3 3

Open source HLR (Home Location Register).
by Henrik Aagaard Sørensen 16 Sep '11

16 Sep '11

I know this is the mailing list for Kamailio, but hopefully someone can help me out as it is "kinda" related to VoIP etc. Does anyone know any open source / free HLR implementation?

4 3

Siremis site access from Romania
by Dani Popa 16 Sep '11

16 Sep '11

Hi, There is any reasons why any linked Siremis site is not accessible from Romania ? i tried to access http://siremis.asipto.com/ and other sites and they are not accesible. Thanks, Dani Popa

4 6

Re: [SR-Users] planning v3.1.5
by Vitaliy Aleksandrov 16 Sep '11

16 Sep '11

There is a bug in UAC module. uac_req_send doesn't restore avps. I tried to backport your patch to 3.1 but with no success. Could you add this patch to a new stable release ?

2 9

DNS support
by CT Radu 15 Sep '11

15 Sep '11

Dear sirs and madams, I was wondering what is the current support for DNS fallback in the 3.1.x branch . Lets assume we have a new dialog started and the name in the To field needs to be resolved via a NAPTR. Kamailio sends a NAPTR request, gets 3 SRVs , each with a number of A or AAAA records. The best/most suitable SRV is selected, however no response is returned from any of the addresses in the A records. Now, we should go the second SRV and try the A records in there and so on. My question is how much from this kind of dns fallback algorithm is supported ? (I hope my explanation wasn't too confusing) Thank you, Costin Radu

2 1

OT: anyone in Bay Area
by Daniel-Constantin Mierla 14 Sep '11

14 Sep '11

Hello, anyone here living/working in San Francisco/Bay Area that has a bit of time to assist me with some stuff? Reply to me for follow up. Thanks, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 0

Kamailio v3.1.5 Released
by Daniel-Constantin Mierla 14 Sep '11

14 Sep '11

Hello, Kamailio SIP Server v3.1.5 stable release is out. This is a maintenance release of latest stable branch, 3.1, that includes fixes since release of v3.1.4. There is no change to database schema or configuration language structure. Deployments running previous v3.1.x versions are strongly recommended to be upgraded to v3.1.5. For more details about version 3.1.5, visit: http://www.kamailio.org/w/2011/09/kamailio-v3-1-5-released/ Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 0

preparing release of v3.1.5
by Daniel-Constantin Mierla 14 Sep '11

14 Sep '11

Hello, today starting with 14:00 I will be preparing the release of v3.1.5. If any devel has commits to do in branch 3.1, please do it before to avoid conflicts. Thanks, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 1

Invitation to connect on LinkedIn
by Dani Popa 14 Sep '11

14 Sep '11

SIP, I'd like to add you to my professional network on LinkedIn. - Dani Dani Popa Technical Manager at NECC Telecom Romania Confirm that you know Dani Popa: https://www.linkedin.com/e/-szw6fj-gskbz4q0-1q/isd/4204604361/O7PsXc0-/?hs=… -- You are receiving Invitation to Connect emails. Click to unsubscribe: http://www.linkedin.com/e/-szw6fj-gskbz4q0-1q/c6YlT3v6LB4U4BZW-5Wc20tVAQF44… (c) 2011 LinkedIn Corporation. 2029 Stierlin Ct, Mountain View, CA 94043, USA.

1 0

Kamailio terminating after upgrading from 3.1.2 to 3.1.4
by Phillman25 Kyriacou 14 Sep '11

14 Sep '11

Hello Im facing a big problem after upgrading from version 3.1.2 to 3.1.4. The kamailio process keeps on terminating every now and then, this is what i was able to retrieve from my logs. Sep 14 12:52:31 SipProxy1 kernel: [71448.062089] __ratelimit: 15 callbacks suppressed Sep 14 12:52:31 SipProxy1 kernel: [71448.062096] kamailio[13963]: segfault at 14 ip 0816452a sp bfedc940 error 4 in kamailio[8048000+1d9000] Sep 14 12:52:31 SipProxy1 /usr/local/sbin/kamailio[13991]: : <core> [pass_fd.c:293]: ERROR: receive_fd: EOF on 49 Sep 14 12:52:31 SipProxy1 /usr/local/sbin/kamailio[13907]: ALERT: <core> [main.c:741]: child process 13963 exited by a signal 11 Sep 14 12:52:31 SipProxy1 /usr/local/sbin/kamailio[13907]: ALERT: <core> [main.c:744]: core was generated Sep 14 12:52:31 SipProxy1 /usr/local/sbin/kamailio[13907]: INFO: <core> [main.c:756]: INFO: terminating due to SIGCHLD Does anyone have an idea why this is happening? Your help is much appreciated! Regards Phillip

2 1

parse errors in rr-callback of path module
by Andreas Granig 14 Sep '11

14 Sep '11

Hi, path module calls register_rrcb when "use_received" is set to 1 to extract the "received=" param and set its content as dst-uri if present. Now I've noticed that when e.g. a message has a pre-loaded route like "Route: <sip:mydomain.com;lr>", then kamailio (at least 3.1.3) spits the following error message: /usr/sbin/kamailio[23953]: ERROR: <core> [parser/parse_param.c:481]: parse_params(): Invalid character, ; expected /usr/sbin/kamailio[23953]: ERROR: path [path.c:213]: failed to parse route parametes I was trying to reproduce this but without much success, it sometimes happens and sometimes not, but I can't yet put the finger on the reason. Any ideas? Andreas

3 8

planning v3.1.5
by Daniel-Constantin Mierla 14 Sep '11

14 Sep '11

Hello, it's being quite some time since we released 3.1.4, so I am thinking of packaging soon v3.1.5 out of stable branch 3.1. My plan is to release it Wednesday, September 14, 2011. Any other opinions? Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

6 6

Does xcap server of kamailio support partial update of xcap document?
by laura testi 13 Sep '11

13 Sep '11

Hi, I am using the XCAP server of kamailio for contact list and presence rules. It seems it only support to update the whole xcap doc of the user but not the partial update. Can anyone tell me if the kamailio's xcap server support the partial update of a user's doc? for example, to remove a contact from the contact list or add a contact to the contact list without PUT or DELETE the whole xcap document of this user? If yes, what's the syntax of the HTTP request? Thank you very much in advanced Best Regards, Laura

3 12

max_contacts
by Rafael Rincon 13 Sep '11

13 Sep '11

Hello Guys, Is there any chance to modify at running time (in a register event) the max_contacts parameter? I am running OpenSER version 1.4 . Regards, Rafael Rincon rrincon(a)ipcomnetwork.com +13054772902 x 111

4 7

cr default route
by caio 13 Sep '11

13 Sep '11

Hello, The documentation of carrierroute says "If flags and mask are not zero, and no match to the message flags is possible, no routing will be done." Then is there any route entry which can act as default gw (like an always-match entry)? For example this 'carrierroute' table: +---------+--------+-------------+-------+------+------+ | carrier | domain | scan_prefix | flags | mask | +---------+--------+-------------+-------+------+------+ 1 | 1 | 1 | 156 | 500 | 0 | 2 | 1 | 1 | | 0 | 0 | 3 | 1 | 2 | | 0 | 0 | +---------+--------+-------------+-------+------+- Here if ruri is 156xx, and there is no flag set, then the cr module does not find a match because of the non-zero value in the flag column (entry 1). But I want to route this call via carrier1 domain1 and scan_prefix empty-value entry (entry 2). How can I set a default route entry here? Is possible with cr? Now I only find it possible via a pstn.def_gw_ip variable. Thank you, Claudio

2 3

wildcarded sip contacts
by Antanas Masevicius 13 Sep '11

13 Sep '11

Hello everyone, been looking for a way to do sip wildcarded contact matching where contact could look like <sip:2222..@33.33.33.33:5060>. Current usrloc module does not implement any such matching, but it seems there is a new rfc 5947 issued regarding this problem. Maybe someone is already working on it? What options do i have in sip-router / kamailio besides LCR based ones ? I want to do local contact matching as fast as possible and without a need to reload lcr table. Antanas Masevicius NTT

1 0

Re: [SR-Users] Fwd: configuration
by Daniel-Constantin Mierla 13 Sep '11

13 Sep '11

Hello, On 9/12/11 4:14 PM, neili ma wrote: > Hello, > > Thanks a lot for your advice, I found and solved the problem thanks to > the debugger, actually the problem was with > > modparam("auth_db", > "db_url","mysql://openser:openserrw@localhost/openser") > > I just had to change localhost to my locationserver IP and the problem > was solved :), thanks again. good that you find it. Don't forget to cc the mailing list always so people reading the thread (archive later) see the conclusion -- it can help many others running on same situation. Cheers, Daniel > > Sincerely, > Neili > > > 2011/9/12 Daniel-Constantin Mierla <miconda(a)gmail.com > <mailto:miconda@gmail.com>> > > Hello, > > you have another group of actions that can send 401 Unauthorized: > > > if (!check_to()) { > sl_send_reply("401", "Unauthorized"); > exit; > } > > You can either run with debug=4 and watch the syslog to see some > hints in messages where it fires this 401, or use debugger module > with cfgtrace option set and you see the execution path of the > config in the logs. > > If you send here the ngrep of such REGISTER and the reply we can > spot what could be the situation for 401. > > Cheers, > Daniel > > > On 9/11/11 9:18 PM, neili ma wrote: >> Hello sr-users, >> >> I'm new using kamailio open source, and I would like to ask for help. >> >> My version is 3.1.4 and I am trying to configure 3 servers, one >> with proxy, one with registrar and one with location server. >> >> My problem is when I try to send a register message it returns an >> error "401 Unauthorized", I know the error is in the fuction >> www_authorize("10.254.239.7", "subscriber") but I don't know how >> to solve it, for me it seems correct the parameters. >> >> Below I have attatched the proxy and registrar relevant route >> code for my problem, and the IPs for proxy an registrar are: >> 10.254.239.7-----------10.254.239.8 >> >> If someone can help me, I will be gratefull. Thanks in advance. >> >> ####REGISTRAR CODE >> >> route{ >> >> # initial sanity checks -- too long messages >> if (msg:len >= 2048 ) { >> sl_send_reply("513", "Message too big"); >> exit; >> }; >> >> if (!uri==myself) { >> sl_send_reply("400", "Bad destination, only >> register messages de$ >> exit; >> }; >> >> # the only valid message for the registrar is the register >> message >> # other messages are answered with an error message >> if (uri==myself) { >> >> if (method=="REGISTER") { >> >> sl_send_reply("100", "Trying"); >> >> if (!www_authorize("10.254.239.7", "subscriber")) { >> www_challenge("10.254.239.7", "1"); >> exit; >> }; >> >> if (!check_to()) { >> sl_send_reply("401", "Unauthorized"); >> exit; >> }; >> >> consume_credentials(); >> if(!save("location")) { >> sl_reply_error(); >> }; >> exit; >> } else { >> sl_send_reply("403", "Forbidden"); >> exit; >> }; >> }; >> } >> >> #########PROXY CODE >> >> if (method=="ACK") { >> route(1); >> exit; >> } if (method=="INVITE") { >> route(3); >> exit; >> } else if (method=="REGISTER") { >> route(2); >> exit; >> }; >> >> lookup("aliases"); >> if (!uri==myself) { >> route(1); >> }; >> >> route[2] { >> #rewrinting the host allows the registrar to know that the >> message is de$ >> >> rewritehost("10.254.239.8"); >> if(!t_relay()) { >> sl_reply_error(); >> }; >> exit; >> } >> >> >> >> >> >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >> sr-users(a)lists.sip-router.org <mailto:sr-users@lists.sip-router.org> >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > -- > Daniel-Constantin Mierla --http://www.asipto.com > Kamailio Advanced Training, Oct 10-13, Berlin:http://asipto.com/u/kat > http://linkedin.com/in/miconda -- http://twitter.com/miconda > > -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 0

CANCEL is not processed in onsend_route
by Timo Klecker 13 Sep '11

13 Sep '11

Hello all, I want to log the incoming and outgoing CANCEL on my Kamailio. I tried with a call of sip_trace in the default route setflag(22); if (method == "CANCEL") { sip_trace(); if (t_check_trans()) { t_relay(); exit; } } This works great, I get the incoming CANCEL in the Database. But the outgoing CANCEL is not logged with flag set. So I tried to use onsend_route with the following lines: onsend_route { xlog("L_NOTICE", " SENDING from $si \n $mb \n "); if ( method == "CANCEL" ) sip_trace(); } I am using Kamailio Version 3.1.0 (with Patch for Siptrace in onsend_route by Daniel ). But the CANCEL is not processed in the onsend_route at all. I do not even see the xlog-line. But tcpdump gives me the CANCEL outgoing. Following callflow: INVITE > 100 < > INVITE < 100 < 180 180 < CANCEL> > CANCEL I am missing the last CANCEL. Why is the CANCEL not logged with flag 22 set? Why is the CANCEL not processed in the onsend_route? Greetings Timo

3 4

question bout UA and BOX in same NAT
by MingHon 12 Sep '11

12 Sep '11

Hello List, Can i have UA in the same nat with the box and also UA in different nat at the same time? My box is currrently working behind nat with ip 192.168.2.3 and advertised_address is set to sip.mydomain.com is it possible to append the via header replace the sip.mydomain.com to 192.168.2.3 ? Thanks, -- Regards, MingHon

2 4

no publish from pua/pua_mi to outbound proxy
by Juha Heinanen 12 Sep '11

12 Sep '11

i started to test sr presence and for some reason sr does not send out any publish request when i issue one via pua_mi. i have in route [xmlrpc_requests]: xlog("L_INFO", "Handling XMLRPC $rm from <$si> with body <$(rb{re.subst,/[\n\r\t]//g})>\n"); set_reply_no_connect(); set_reply_close(); dispatch_rpc(); syslog shows that sr receives xmlrpc request: Aug 29 19:42:13 sip /usr/sbin/pres-serv[2118]: INFO: Handling XMLRPC POST from <127.0.0.1> with body <<?xml version='1.0'?><methodCall><methodName>mi</methodName><params><param><value><string>pua_publish</string></value></param><param><value><string>sip:jh@vm.test.fi</string></value></param><param><value><int>7776000</int></value></param><param><value><string>message-summary</string></value></param><param><value><string>application/simple-message-summary</string></value></param><param><value><string>.</string></value></param><param><value><string>a.1313821931.6178.2.0</string></value></param><param><value><string>P-Flags: 0</string></value></param><param><value><string>Messages-Waiting:yesVoice-Message: 1/1 (0/0)</string></value></param></params></methodCall>> and wireshark shows that sr xmlrpc server responds to it with 200 ok. but after than nothing happens, i.e., according to wireshark no publish request is sent out. i have pua and pua_mi modules loaded and outbound proxy defined: modparam("pua", "outbound_proxy", "sip:127.0.0.1:5082") there is no error messages in syslog. any idea why sr does not send publish to outbound proxy? this used to work with opensips. -- juha

2 13

Nat problems
by David Zambrano 12 Sep '11

12 Sep '11

Hi Ive been trying to configure kamailio as a loadbalancer between two transcoders that afterwards send all calls to a softswitch but I currently have two problems. My first and most important one is a NAT problem or so I think. All calls in UDP work perfectly but when dealing with calls in TCP only outgoing calls work, no incoming calls work. After doing some traces on the network I realized that the transcoder is trying to reach the router in front of the softphone and skipping the Kamailio loadbalancer. The call never reaches the softphone so the phonecall never completes. In UDP exactly the same thing happens, the call skips the Kamailio loadbalancer but the call completes perfectly fine. The other detail has to do with the actual loadbalancing and failover. I was using the weight distribution and it works great except its sending the same CSeq to both transcoders and then the softswitch is receiving it twice and an error pops up at the softphone Ive been using saying theres a duplicate CSeq. I was using the forward() command after the dispatcher arguments. I changed it to the t_relay() and it works as in it doesnt send the same CSeq twice but instead a unique one to each transcoder so the Softswitch stopped winning but then the failover doesnt work. Whenever it one goes down its till trying to reach it. Im using roundrobin in the dispatcher config because I noticed it doesnt send stuff to both transcoders but only to the transcoder its going to actually use. I started to read that for failover I need the user_avp module which I didnt install but then I wonder how come failover worked before? Thanks for your help Cheers David

3 7

Fwd: configuration
by neili ma 12 Sep '11

12 Sep '11

Hello sr-users, I'm new using kamailio open source, and I would like to ask for help. My version is 3.1.4 and I am trying to configure 3 servers, one with proxy, one with registrar and one with location server. My problem is when I try to send a register message it returns an error "401 Unauthorized", I know the error is in the fuction www_authorize("10.254.239.7", "subscriber") but I don't know how to solve it, for me it seems correct the parameters. Below I have attatched the proxy and registrar relevant route code for my problem, and the IPs for proxy an registrar are: 10.254.239.7-----------10.254.239.8 If someone can help me, I will be gratefull. Thanks in advance. ####REGISTRAR CODE route{ # initial sanity checks -- too long messages if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; if (!uri==myself) { sl_send_reply("400", "Bad destination, only register messages de$ exit; }; # the only valid message for the registrar is the register message # other messages are answered with an error message if (uri==myself) { if (method=="REGISTER") { sl_send_reply("100", "Trying"); if (!www_authorize("10.254.239.7", "subscriber")) { www_challenge("10.254.239.7", "1"); exit; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); exit; }; consume_credentials(); if(!save("location")) { sl_reply_error(); }; exit; } else { sl_send_reply("403", "Forbidden"); exit; }; }; } #########PROXY CODE if (method=="ACK") { route(1); exit; } if (method=="INVITE") { route(3); exit; } else if (method=="REGISTER") { route(2); exit; }; lookup("aliases"); if (!uri==myself) { route(1); }; route[2] { #rewrinting the host allows the registrar to know that the message is de$ rewritehost("10.254.239.8"); if(!t_relay()) { sl_reply_error(); }; exit; }

2 1

Re: [SR-Users] Kamailio 3.0.4 is crashed
by Daniel-Constantin Mierla 12 Sep '11

12 Sep '11

Hello, try to keep the mailing list cc-ed -- others may be able to help you faster or this discussion will be helpful for different people. The username_spec and password_spec were removed indeed, the first was useless since the username was taken from auth header and the second became parameter to pv_www_authenticate, see: http://kamailio.org/docs/modules/3.1.x/modules/auth.html#pv_www_authenticate You simply have to use something like: if (!pv_www_authenticate("$td", "$var(password)", "0")) { www_challenge("$td", "0"); exit; } Cheers, Daniel On 9/10/11 12:51 PM, Khoa Phuong Viet Ngo wrote: > Hi Daniel, > > Thanks Daniel for your answer. > > It's very helpful for me :). > > Actually i tried with using kamailio 3.1.4 (lastest kamailio) but i > got the problem with auth module and parameter with auth module. > Because some parameters at auth module have changed. > http://kamailio.org/docs/modules/stable/modules/auth.html [3.1.4] > > But in 3.0.4 kamailio, i am using auth module with these parameter and > it had error with new kamailio (3.1.4) . :( > > kamailio.cfg > > ######################################################## > loadmodule auth.so > > # ----- auth params ----- > modparam("auth", "username_spec", "$var(username)") > modparam("auth", "password_spec", "$var(password)") > modparam("auth", "calculate_ha1", 1) > > route[AUTH] { > $var(utype) = 0; > #!ifdef WITH_AUTH > if (is_method("REGISTER")) > { > if(is_present_hf("Authorization")) > { > # authenticate the REGISTER requests (uncomment to enable > auth) > if(!route(GETPASSWD)) > { > www_challenge("", "0"); > exit; > } > if(!pv_www_authorize("")) > { > www_challenge(""/*realm*/,"0"/*qop*/); > exit; > } > } else { > www_challenge("", "0"); > exit; > } > } else { > $var(srcgrp) = allow_source_address_group(); > if ($var(srcgrp) > 0) { > # src ip allowed > $var(utype) = 1; > if($var(srcgrp)!=100) > { > $rd = $Ri; > $fs = "udp:" + $Ri + ":5062"; > } > return 1; > } > # authenticate if from local subscriber (uncomment to enable auth) > if (from_uri==myself) > { > if(is_present_hf("Proxy-Authorization")) > { > # authenticate the non-REGISTER requests (uncomment to > enable auth) > if(!route(GETPASSWD)) > { > proxy_challenge("", "0"); > exit; > } > if(!pv_proxy_authorize("")) > { > proxy_challenge(""/*realm*/,"0"/*qop*/); > exit; > } > } else { > proxy_challenge("", "0"); > exit; > } > $var(utype) = 2; > consume_credentials(); > # caller authenticated > } > } > #!endif > return; > } > > ##################################################### > > Error logs: > > Sep 6 19:37:01 CorePBX kamailio: ERROR: <core> [modparam.c:150]: > set_mod_param_regex: parameter <username_spec> not found in module <auth> > Sep 6 19:37:01 CorePBX kamailio: : <core> [cfg.y:3419]: parse error > in config file /usr/local/etc/kamailio/kamailio.cfg, line 260, column > 51: Can't set module parameter > Sep 6 19:37:01 CorePBX kamailio: ERROR: <core> [modparam.c:150]: > set_mod_param_regex: parameter <password_spec> not found in module <auth> > Sep 6 19:37:01 CorePBX kamailio: : <core> [cfg.y:3419]: parse error > in config file /usr/local/etc/kamailio/kamailio.cfg, line 261, column > 51: Can't set module parameter > Sep 6 19:37:01 CorePBX kamailio: ERROR: <core> [modparam.c:150]: > set_mod_param_regex: parameter <calculate_ha1> not found in module <auth> > Sep 6 19:37:01 CorePBX kamailio: : <core> [cfg.y:3419]: parse error > in config file /usr/local/etc/kamailio/kamailio.cfg, line 262, column > 36: Can't set module parameter > Sep 6 19:37:01 CorePBX kamailio: : <core> [cfg.y:3419]: parse error > in config file /usr/local/etc/kamailio/kamailio.cfg, line 562, column > 27: unknown command, missing loadmodule? > Sep 6 19:37:01 CorePBX kamailio: : <core> [cfg.y:3419]: parse error > in config file /usr/local/etc/kamailio/kamailio.cfg, line 594, column > 30: unknown command, missing loadmodule? > Sep 6 19:37:39 CorePBX kamailio: ERROR: <core> [modparam.c:150]: > set_mod_param_regex: parameter <username_spec> not found in module <auth> > Sep 6 19:37:39 CorePBX kamailio: : <core> [cfg.y:3419]: parse error > in config file /usr/local/etc/kamailio/kamailio.cfg, line 260, column > 51: Can't set module parameter > Sep 6 19:37:39 CorePBX kamailio: ERROR: <core> [modparam.c:150]: > set_mod_param_regex: parameter <password_spec> not found in module <auth> > Sep 6 19:37:39 CorePBX kamailio: : <core> [cfg.y:3419]: parse error > in config file /usr/local/etc/kamailio/kamailio.cfg, line 261, column > 51: Can't set module parameter > Sep 6 19:37:39 CorePBX kamailio: ERROR: <core> [modparam.c:150]: > set_mod_param_regex: parameter <calculate_ha1> not found in module <auth> > > ######################################################## > > I appreciate all your help on this case. > > Thanks Daniel, > KhoaNVP > > > > > On Sat, Sep 10, 2011 at 1:58 PM, Daniel-Constantin Mierla > <miconda(a)gmail.com <mailto:miconda@gmail.com>> wrote: > > Hello, > > it seems like a buffer overflow happened previously, since the > crash is in malloc. Do you get any error messages is syslog? Can > you recompile with QM_DBG_MALLOC like explained in the next link > and try again watching the syslog messages? > http://www.kamailio.org/dokuwiki/doku.php/troubleshooting:memory > > On another hand, 3.0.4 is quite old, why are not using 3.1.4, it > is far more updated than 3.0 series -- we are just to release > first in 3.2 series. Upgrading from 3.0 to 3.1 is rather trivial. > > Cheers, > Daniel > > > On 9/9/11 8:38 PM, KhoaNVP wrote: > > Hi everyone, > > I hope you can help me with this issues. > I'm working with kamailio 3.0.4 and when i implement the call > via kamailio > after 1-->3 mins , kamailio is crashed. > > It's my debug from core log. > > Core was generated by `kamailio -E -ddd'. > Program terminated with signal 11, Segmentation fault. > #0 fm_malloc (qm=0x88c860, size=<value optimized out>) at > mem/f_malloc.c:382 > 382 *f=frag->u.nxt_free; > (gdb) qt > Undefined command: "qt". Try "help". > (gdb) bt > #0 fm_malloc (qm=0x88c860, size=<value optimized out>) at > mem/f_malloc.c:382 > #1 0x000000000041fd27 in dup_lump_list_r (l=0x9275f8, > dir=LD_AFTER, > error=0x7fff23bd661c) at data_lump.c:453 > #2 0x000000000041ffa9 in dup_lump_list_r (l=0x927640, > dir=LD_AFTER, > error=0x7fff23bd666c) at data_lump.c:476 > #3 0x000000000041ff6d in dup_lump_list_r (l=0x926db0, > dir=<value optimized > out>, error=0x7fff23bd66b4) at data_lump.c:465 > #4 0x0000000000420058 in dup_lump_list (l=0x88c860) at > data_lump.c:508 > #5 0x00002b8b2efb858d in prepare_new_uac (t=0x2b8b3218ffa8, > i_req=0x926ef0, > branch=0, uri=0x927150, path=0x9274d0, next_hop=<value > optimized out>, > fsocket=0x0, snd_flags=0 '\000', fproto=0, flags=0) at > t_fwd.c:242 > #6 0x00002b8b2efb94c7 in add_uac (t=0x2b8b3218ffa8, > request=0x926ef0, > uri=0x927150, next_hop=0x7, path=0x9274d0, proxy=0x0, fsocket=0x0, > snd_flags=0 '\000', proto=0, flags=0) at t_fwd.c:679 > #7 0x00002b8b2efbce7a in t_forward_nonack (t=0x2b8b3218ffa8, > p_msg=0x926ef0, proxy=0x0, proto=0) at t_fwd.c:1340 > #8 0x00002b8b2efb4996 in t_relay_to (p_msg=0x926ef0, > proxy=0x0, proto=0, > replicate=0) at t_funcs.c:358 > #9 0x0000000000413489 in do_action (h=0x7fff23bd71e0, a=0x8be788, > msg=0x926ef0) at action.c:860 > #10 0x0000000000415e03 in run_actions (h=0x7fff23bd71e0, > a=0x8be788, > msg=0x926ef0) at action.c:1315 > #11 0x00000000004161e7 in run_actions_safe (h=0x7fff23bd8bb0, > a=0x38, > msg=0x88ca08) at action.c:1370 > #12 0x000000000049696b in rval_get_int (h=0x7fff23bd8bb0, > msg=0x88ca08, > i=0x7fff23bd75dc, rv=0x7, cache=0x7) at rvalue.c:897 > #13 0x000000000049b7fc in rval_expr_eval_int (h=0x7fff23bd8bb0, > msg=0x926ef0, res=0x7fff23bd75dc, rve=0x8be878) at rvalue.c:1841 > #14 0x000000000049b8ad in rval_expr_eval_int (h=0x7fff23bd8bb0, > msg=0x926ef0, res=0x7fff23bd79d8, rve=0x8bef48) at rvalue.c:1848 > #15 0x0000000000411d49 in do_action (h=0x7fff23bd8bb0, a=0x8bf798, > msg=0x926ef0) at action.c:834 > #16 0x0000000000415e03 in run_actions (h=0x7fff23bd8bb0, > a=0x8be608, > msg=0x926ef0) at action.c:1315 > #17 0x0000000000413084 in do_action (h=0x7fff23bd8bb0, a=0x8f6060, > msg=0x926ef0) at action.c:479 > #18 0x0000000000415e03 in run_actions (h=0x7fff23bd8bb0, > a=0x8f6060, > msg=0x926ef0) at action.c:1315 > #19 0x0000000000411dab in do_action (h=0x7fff23bd8bb0, a=0x8f62d0, > msg=0x926ef0) at action.c:853 > #20 0x0000000000415e03 in run_actions (h=0x7fff23bd8bb0, > a=0x8f62d0, > msg=0x926ef0) at action.c:1315 > #21 0x0000000000413084 in do_action (h=0x7fff23bd8bb0, a=0x8bd698, > msg=0x926ef0) at action.c:479 > #22 0x0000000000415e03 in run_actions (h=0x7fff23bd8bb0, > a=0x8bd398, > msg=0x926ef0) at action.c:1315 > #23 0x0000000000411dab in do_action (h=0x7fff23bd8bb0, a=0x8bd788, > msg=0x926ef0) at action.c:853 > #24 0x0000000000415e03 in run_actions (h=0x7fff23bd8bb0, > a=0x8a1340, > msg=0x926ef0) at action.c:1315 > #25 0x0000000000416174 in run_top_route (a=0x8a1340, > msg=0x926ef0, c=<value > optimized out>) at action.c:1391 > #26 0x000000000048009c in receive_msg (buf=0x57e779 "DEBUG", > len=<value > optimized out>, rcv_info=0x7fff23bd8e20) at receive.c:196 > #27 0x000000000050698b in udp_rcv_loop () at udp_server.c:520 > #28 0x0000000000455e2f in main_loop () at main.c:1447 > #29 0x0000000000456f32 in main (argc=<value optimized out>, > argv=0x7fff23bd90e8) at main.c:2251 > (gdb) > > > Thanks with all the help, > Khoa > > > -- > Daniel-Constantin Mierla -- http://www.asipto.com > Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat > http://linkedin.com/in/miconda -- http://twitter.com/miconda > > -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 0

Kamailio 3.0.4 is crashed
by KhoaNVP 10 Sep '11

10 Sep '11

Hi everyone, I hope you can help me with this issues. I'm working with kamailio 3.0.4 and when i implement the call via kamailio after 1-->3 mins , kamailio is crashed. It's my debug from core log. Core was generated by `kamailio -E -ddd'. Program terminated with signal 11, Segmentation fault. #0 fm_malloc (qm=0x88c860, size=<value optimized out>) at mem/f_malloc.c:382 382 *f=frag->u.nxt_free; (gdb) qt Undefined command: "qt". Try "help". (gdb) bt #0 fm_malloc (qm=0x88c860, size=<value optimized out>) at mem/f_malloc.c:382 #1 0x000000000041fd27 in dup_lump_list_r (l=0x9275f8, dir=LD_AFTER, error=0x7fff23bd661c) at data_lump.c:453 #2 0x000000000041ffa9 in dup_lump_list_r (l=0x927640, dir=LD_AFTER, error=0x7fff23bd666c) at data_lump.c:476 #3 0x000000000041ff6d in dup_lump_list_r (l=0x926db0, dir=<value optimized out>, error=0x7fff23bd66b4) at data_lump.c:465 #4 0x0000000000420058 in dup_lump_list (l=0x88c860) at data_lump.c:508 #5 0x00002b8b2efb858d in prepare_new_uac (t=0x2b8b3218ffa8, i_req=0x926ef0, branch=0, uri=0x927150, path=0x9274d0, next_hop=<value optimized out>, fsocket=0x0, snd_flags=0 '\000', fproto=0, flags=0) at t_fwd.c:242 #6 0x00002b8b2efb94c7 in add_uac (t=0x2b8b3218ffa8, request=0x926ef0, uri=0x927150, next_hop=0x7, path=0x9274d0, proxy=0x0, fsocket=0x0, snd_flags=0 '\000', proto=0, flags=0) at t_fwd.c:679 #7 0x00002b8b2efbce7a in t_forward_nonack (t=0x2b8b3218ffa8, p_msg=0x926ef0, proxy=0x0, proto=0) at t_fwd.c:1340 #8 0x00002b8b2efb4996 in t_relay_to (p_msg=0x926ef0, proxy=0x0, proto=0, replicate=0) at t_funcs.c:358 #9 0x0000000000413489 in do_action (h=0x7fff23bd71e0, a=0x8be788, msg=0x926ef0) at action.c:860 #10 0x0000000000415e03 in run_actions (h=0x7fff23bd71e0, a=0x8be788, msg=0x926ef0) at action.c:1315 #11 0x00000000004161e7 in run_actions_safe (h=0x7fff23bd8bb0, a=0x38, msg=0x88ca08) at action.c:1370 #12 0x000000000049696b in rval_get_int (h=0x7fff23bd8bb0, msg=0x88ca08, i=0x7fff23bd75dc, rv=0x7, cache=0x7) at rvalue.c:897 #13 0x000000000049b7fc in rval_expr_eval_int (h=0x7fff23bd8bb0, msg=0x926ef0, res=0x7fff23bd75dc, rve=0x8be878) at rvalue.c:1841 #14 0x000000000049b8ad in rval_expr_eval_int (h=0x7fff23bd8bb0, msg=0x926ef0, res=0x7fff23bd79d8, rve=0x8bef48) at rvalue.c:1848 #15 0x0000000000411d49 in do_action (h=0x7fff23bd8bb0, a=0x8bf798, msg=0x926ef0) at action.c:834 #16 0x0000000000415e03 in run_actions (h=0x7fff23bd8bb0, a=0x8be608, msg=0x926ef0) at action.c:1315 #17 0x0000000000413084 in do_action (h=0x7fff23bd8bb0, a=0x8f6060, msg=0x926ef0) at action.c:479 #18 0x0000000000415e03 in run_actions (h=0x7fff23bd8bb0, a=0x8f6060, msg=0x926ef0) at action.c:1315 #19 0x0000000000411dab in do_action (h=0x7fff23bd8bb0, a=0x8f62d0, msg=0x926ef0) at action.c:853 #20 0x0000000000415e03 in run_actions (h=0x7fff23bd8bb0, a=0x8f62d0, msg=0x926ef0) at action.c:1315 #21 0x0000000000413084 in do_action (h=0x7fff23bd8bb0, a=0x8bd698, msg=0x926ef0) at action.c:479 #22 0x0000000000415e03 in run_actions (h=0x7fff23bd8bb0, a=0x8bd398, msg=0x926ef0) at action.c:1315 #23 0x0000000000411dab in do_action (h=0x7fff23bd8bb0, a=0x8bd788, msg=0x926ef0) at action.c:853 #24 0x0000000000415e03 in run_actions (h=0x7fff23bd8bb0, a=0x8a1340, msg=0x926ef0) at action.c:1315 #25 0x0000000000416174 in run_top_route (a=0x8a1340, msg=0x926ef0, c=<value optimized out>) at action.c:1391 #26 0x000000000048009c in receive_msg (buf=0x57e779 "DEBUG", len=<value optimized out>, rcv_info=0x7fff23bd8e20) at receive.c:196 #27 0x000000000050698b in udp_rcv_loop () at udp_server.c:520 #28 0x0000000000455e2f in main_loop () at main.c:1447 #29 0x0000000000456f32 in main (argc=<value optimized out>, argv=0x7fff23bd90e8) at main.c:2251 (gdb) Thanks with all the help, Khoa -- View this message in context: http://old.nabble.com/Kamailio-3.0.4-is-crashed-tp32433892p32433892.html Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

2 1

Error reading AVP
by Carlos Ruiz Diaz 09 Sep '11

09 Sep '11

Hello. I'm currently developing an OpenSER module which does some business logic for the company I work for. For compatibility reasons we are working with openser-1.2.2-tls and we haven't been able to upgrade to newer version yet. In one block of the code I have to read an avp which is written inside the openser.cfg script using: OPENSER: avp_copy("sip:123@123.123.123:5060", "$avp(s:selectedroute)/g"); C CODE: if (is_response && is_invite_seq) { struct usr_avp* destavp = NULL; int_str destval; const str selected_route = { "selectedroute", 13 }; int attempts = 1, max_attempts = 3; for ( destavp = search_first_avp(AVP_NAME_STR | AVP_VAL_STR, (int_str) selected_route, NULL, 0); destavp != NULL; destavp = search_next_avp(destavp, NULL) ) { get_avp_val(destavp, &destval); } } The problem is that "selectedroute" avp cannot be found so I can't read its value. This happens sporadically and most of the time the code works just fine. I also checked for the list of available AVPs using get_avp_list() but it reports an empty list. Is there another way to read AVPs or is there something I'm missing which is causing this problem? Any help will be appreciated. Thanks in advance! Carlos.

2 2

10 Years SER Conference - Presentations
by Daniel-Constantin Mierla 08 Sep '11

08 Sep '11

Hello, the presentations from 10 Years SER Conference I received so far were uploaded at: * http://www.kamailio.org/events/2011-10yearsSER/ Expect the rest during the next days. The recording of the talk about Gemeinschaft 4 by Stefan Wintermeyer is available on youtube: * http://www.youtube.com/watch?v=mbGn5N_y4rw Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 0

Re: [SR-Users] Kamailio + rtpproxy talking to multiple carrier gateways - Solution
by Sarat C. Vemuri 07 Sep '11

07 Sep '11

I think I have this working. For anyone else trying to do similar thing (or placing Kamailio behind FW/NAT), here is my cookbook. We are trying to setup the following: [UAS] <------> (PubIP)[FireWall/One-to-OneNAT]<-------->(PrivIP)[Kamailio/RTPProxy](PrivIP)<---------->(PrivIP2)[UAC] UAC/UAS cannot reach each other directly. For simplicity I am running RTPProxy on same machines as Kamailio. But this can be changed. Versions/Patches/fixes you need if you are running 3.1.2 : (Trunk/3.2 has these): a. Fixed forward.c for mhome to work properly (included in 3.1.4). b. rr module from trunk to be able to use 2 strings for record_route_preset c. Patched rtpproxy module to actually use the forced ip supplied to force_rtp_proxy. Patch at the end of this message d. rtpproxy version 1.2.1 What to do: 1. Setup FW/NAT rules to map public IP to PrivateIP and allow communications to/from UAS 2. Setup RTPProxy in "bridged" mode (e.g -F -l 10.10.5.90/10.10.5.90 -s udp:10.10.5.90:9005) listening on PrivateIP. 3. Setup the config file to: a. Setup modparams to make the above rtpproxy available. Use set_rtp_proxy_set if you have more than one instance. b. Do the following where you are processing Outbound *initial* invite (i.e. check totag) (Call from UAC to UAS) set_advertised_address("YourPublicIP "); if(!has_totag()) { record_route_preset("YourPublicIP:5060;r2=on;nat=yes","YourPrivateIp:5060;r2=on;nat=yes"); } force_rtp_proxy("ocfaei"," YourPublicIP "); c. Do the following where you are processing Inbound invite (Call from UAS to UAC) if(!has_totag()) { record_route_preset("YourPrivateIP:5060;r2=on;nat=yes", "YourPublicIp:5060;r2=on;nat=yes"); } force_rtp_proxy("ocfaei"); d. Do the following where you are processing replies (180/183/2XX) For Inbound Calls force_rtp_proxy("ocfa","YourPublicIP"); For Outbound Calls force_rtp_proxy("ocfa"); e. Unforce RTPProxy on BYE or Failures. Do not use record_route in the above scenario. If you have it now, remove it or make sure it doesn't get called for this specific case. Explanation: In #b, you are fixing the Via headers to use public IP, setting Record-Route headers so that ACK/RE-INVITE can find way back and are force fixing SDP with public IP for outbound calls. In #c, You are doing same thing except no need to fix Via and the route set is reversed. No need to force fix SDP either since it is figured out automatically. In #d, You are force fixing the SDP to use the public IP for inbound or usual for outbound so that the audio stream from UAS can be relayed. The r2=on parameter is required (as well as enabling double rr, which is default) so that the proxy can deal with the route sets properly Hope the above helps others. SV. Rtpproxy Path 2226,2233c2226 < LM_DBG("Str2 is %s\n", str2); < if(str2) { < newip.s = str2; < LM_DBG("NewIp is %s\n", newip.s); < } < else { < newip.s = (argc < 2) ? str2 : argv[1]; < } --- > newip.s = (argc < 2) ? str2 : argv[1];

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