- sr-users - kamailio.org

How to get rid of this error message.
by Gary Chen 29 Apr '11

29 Apr '11

I am using prefix2domain("2") in PDT module like this: $var(found) = prefix2domain("2"); and I got this error: ERROR: <core> [lvalue.c:411]: assignment failed at pos: (701,56-701,56) Apparently when prefix2domain("2) returns an value to $var(found), it does not like it. But it still work. The $var(found) still contains the return value (-1) if no prefix found. And all my configuration is working. It just generate a lot of this kind of error in my log file. Am I doing something wrong? How can I fix it? Gary

3 2

Kamailio with RTPoxy
by alex pappas 29 Apr '11

29 Apr '11

Dear all, I have a Kamailio with RTPoxy configured. The purpose is to pass the RTP from this server. When I make calls behind NAT the RTProxy seems to be engaged but in traces I don't see any RTP. Is there anything wrong with the idea or I'm missing something in the configuration ? Thank you Alex

3 5

is $td.did avp a normal avp?
by Juha Heinanen 26 Apr '11

26 Apr '11

domain module sets $td.did at lookup_domain call. $td.did seems to be a normal avp, because i can print its value with xlog by referring to it as $avp(td.did). however, its value is gone at failure route, which makes me suspect that $td.did is not really a normal avp. in route block i have $avp(did) = $td.did; xlog("L_INFO", "at t_relay: <td/td.did> = <$avp(did)/$avp(td.did)>\n"); if (t_relay()) exit; branch route, onreply route, and failure route are set when the above t_relay() takes place. at the end of branch route block, i have xlog("L_INFO", "at end of contact branches: <did/td.did> = <$avp(did)/$avp(td.did)>\n"); and at the beginning of onreply route block: xlog("L_INFO", "at beginning of onreply: <did/td.did> = <$avp(did)/$avp(td.did)>\n"); and at the beginning of failure route block: xlog("L_INFO", "at beginning of contact failure: <did/td.did = <$avp(did)/$avp(td.did)>\n"); when i make a call, i get to syslog: Apr 22 15:53:27 sip /usr/sbin/sip-proxy[13706]: INFO: at t_relay: <td/td.did> = <test.fi/test.fi> Apr 22 15:53:27 sip /usr/sbin/sip-proxy[13706]: INFO: at end of contact branches: <did/td.did> = <test.fi/test.fi> Apr 22 15:53:27 sip /usr/sbin/sip-proxy[13708]: INFO: at beginning of onreply: <did/td.did> = <test.fi/<null>> Apr 22 15:53:27 sip /usr/sbin/sip-proxy[13705]: INFO: at beginning of contact failure: <did/td.did = <test.fi/<null>> why is it that $avp(did) value has been preserved from route/branch route block to onreply/failure blocks, but value of $avp(td.did) is not? i'll cc to jan in case he is not on this list. -- juha

4 12

call for participation: kamailio project at LinuxTag 2011
by Henning Westerholt 26 Apr '11

26 Apr '11

Hello, in about five weeks the kamailio project will (again) be present on the LinuxTag fair, "the most important place for Linux and open source software in Europe" - http://www.linuxtag.org/2011/en.html . If you are in Berlin during this time, plan to visit the LinuxTag or just like to help your favourite SIP server project - please consider supporting us in running our fair booth! Even showing up for one or two hours helps a lot. We need also support to properly present our project, like flyers, a banner, maybe some live demonstrations.. So even if you can't participate in person, there are other ways to help us! Please contact me per mail if you could help at the booth or have some ideas about our project presentation. The actual state of preparations can be found in our wiki: http://sip-router.org/wiki/meetings/berlin_2011 Cheers, Henning

1 1

Bug in LDAP module?
by Pan B. Christensen 26 Apr '11

26 Apr '11

Hello, We are using kamailio 1.5 and we have the following problem with ldap_search: A call comes in. If it is to a VOIP user, it will be routed to kamailio, and eventually go to the ldap lookup route: route[10] to look up the user. This is always successful (LDAP can connect). It will then look up user preferences (call forwarding etc.) from SQL in route[12] and route based on this result in route[13]. If the result is "CFU (call forward unconditional) = on", the URI is changed and route[10] (LDAP lookup) is called again. This is always successful (LDAP can connect). If the result is "CFB (call forward busy) = on" and the user is busy, the URI is changed and route[10] (LDAP lookup) is called again. This is also always successful (LDAP can connect). If the result is "CFNR (call forward no reply) = on", the timeout is changed and the call is delivered to the phone through route[14] (location) and route[15] (relay). If the user does not answer, the URI is changed in failure_route[1] and route[10] is called. For some unknown reason, ldap_search always fails in this scenario with the messages: "ldap_session not found", "couldn't get ldap session" and then "no LDAP entry found". Why? And why does this result in the -1 return code and not the -2 return code for ldap_search? The only differences I can think of is that the unsuccessful scenario comes from FAILURE_ROUTE instead of REQUEST_ROUTE, and that longer time has passed since the original invite. Here's a sample log: NOTE the three ERROR lines, the INFO line and the ROUTE 10 line. Dec 27 13:31:55 [3373]: INVITE received from sip:<CALLER>@<SBC_IP> <source IP: <SBC_IP>> to sip:<CALLEE>@<KAM_IP>: Dec 27 13:31:55 [3373]: ERROR:ldap:ldap_reconnect: [ldapprod]: LDAP reconnect successful Dec 27 13:31:55 [3373]: ROUTE 10: LDAP returned: UID = <UID>, voipSipPrimaryUri = <PRIMARY_URI>, voipExtensionUri = sip:<CALLEE>@<DOMAIN> Dec 27 13:31:55 [3373]: ROUTE 12: Effective user properties for UID <UID>: CFU: off <FORWARDTO>, CFB: off <FORWARDTO>, CFNR: on:25 <FORWARDTO>, CFNC: off <FORWARDTO>, VM: <CALLEE>, busy_level: 1, DBrows: 1 Dec 27 13:31:55 [3373]: -------------------- Calling sip:<CALLEE>@<DOMAIN> for 25 seconds. -------------------- Dec 27 13:31:55 [3373]: ROUTE 15: Sending 105 INVITE from sip:<CALLER>@<SBC_IP> to sip:<CALLEE>@<CALLEE_IP> (via <<null>>) Dec 27 13:31:55 [3383]: Response 100 Trying to Cseq 102 INVITE received from <PROTO>:<CALLEE_IP>:<PORT> Dec 27 13:31:55 [3385]: Response 180 Ringing to Cseq 102 INVITE received from <PROTO>:<CALLEE_IP>:<PORT> Dec 27 13:32:20 [3388]: FAILURE_ROUTE 1: Timeout for: 104 INVITE from <CALLER> -> <CALLEE> Dec 27 13:32:20 [3388]: -------------------- Forwarding on no reply to <FORWARDTO> -------------------- Dec 27 13:32:20 [3388]: ERROR:ldap:get_connected_ldap_session: [ldapprod]: ldap_session not found Dec 27 13:32:20 [3388]: ERROR:ldap:lds_search: [ldapprod]: couldn't get ldap session Dec 27 13:32:20 [3388]: INFO:ldap:ldap_search_impl: no LDAP entry found Dec 27 13:32:20 [3388]: ROUTE 11: Number lookup results for <FORWARDTO>: Node: VOIP. Dec 27 13:32:20 [3388]: CONFIG ERROR: VOIP number not found in LDAP. Hanging up. Dec 27 13:32:20 [3373]: Response 487 Request Cancelled to Cseq 102 INVITE received from <PROTO>:<CALLEE_IP>:<PORT> Here's the ldap config file: [ldapprod] ldap_server_url = "ldap://ldap.<DOMAIN>" ldap_network_timeout = 500 ldap_client_bind_timeout = 500 kamailio config snippets (in case you need them): # LDAP lookup for UID and primary URI route[10] { ... ldap_search("ldap://ldapprod/<DN>?uid,voipSipPrimaryUri,voipExtensionUri?one?(&(objectClass=voipAddress)(|(voipSipUri=sip:$rU@uio.no)(|(voipSipUri=$var(uri))(voipSipUri=$var(e164uri)))))"); if ($retcode < 1) { switch ($retcode) { case -1: # no LDAP entry found route(11); return(1); case -2: # internal error sl_send_reply("500", "Internal server error"); exit; } } ldap_result("uid/$avp(s:uid)"); $avp(s:exten) = null; if (!ldap_result("voipExtensionUri/$avp(s:exten)")) { xlog("L_INFO", "voipExtensionUri not found.\n"); } if (!ldap_result("voipSipPrimaryUri/$avp(s:p-uri)")) { xlog("L_WARN", "voipSipPrimaryUri not found.\n"); #exit here? } if (isflagset(1)) { xlog("L_INFO", "ROUTE 10: LDAP returned: UID = $avp(s:uid), voipSipPrimaryUri = $avp(s:p-uri), voipExtensionUri = $avp(s:exten)\n"); } if $avp(s:exten) != null { $ru=$avp(s:exten); } else { $ru=$avp(s:p-uri); } route(12); } } # Number lookup. Non-VoIP URI's go here. route[11] { ... } # Database lookup for call forwarding. route[12] { ... } # Call forwarding logic route[13] { if (is_method("INVITE")) { $avp(s:uio_timer_avp) = "300"; if ($avp(s:cfu_status) == "on") { # Forward immediately xlog("L_INFO", "-------------------- $rU forwarded unconditionally to $avp(s:cfu_number). --------------------\n"); $rU = $avp(s:cfu_number); route(10); } ... if ( $var(dlg_busy) >= $avp(s:busy_level) ) { if ($avp(s:cfb_status) == "on") { xlog("L_INFO", "-------------------- $rU has $var(dlg_busy) active calls. Treshold $avp(s:busy_level). Forwarding on busy to $avp(s:cfb_number) --------------------\n"); $rU = $avp(s:cfb_number); route(10); } ... else if ($avp(s:cfnr_status) == "on") { # Forward to phone with timeout then to $avp(s:cfnr_number). # t_set_fr($avp(s:cfnr_timeout) + "000"); # Recommended method in 3.1. Function does not exist in 1.5. Using old method (avp). $avp(s:uio_timer_avp) = $avp(s:cfnr_timeout); xlog("L_INFO", "-------------------- Calling $ru for $avp(s:uio_timer_avp) seconds. --------------------\n"); route(14); } ... } failure_route[1] { ... if $avp(s:cfnr_status) == "on" { if (isflagset(1)) { xlog("L_INFO", "FAILURE_ROUTE 1: Timeout for: $mi $rm from $fU -> $rU\n"); } revert_uri(); $rU = $avp(s:cfnr_number); xlog("L_INFO", "-------------------- Forwarding on no reply to $rU --------------------\n"); route(10); } ... } With kind regards, Pan

3 5

Kamailio stops responding after 10 days or so
by Morten Isaksen 26 Apr '11

26 Apr '11

Hi! Kamailio 3.0.3. I have a strange problem with one of our Kamailio servers. This one is used for routing (with carrierroute) and to send presence information (with pua module) Once every 10 day or so I get this error and then Kamailio stops responding to any SIP packets. Apr 6 08:05:48 sip-core-1 /usr/local/sbin/kamailio[9186]: WARNING: <script>: Failure route - M=INVITE RURI=sip:8615XXXX@178.XX.XX.XX F=sip:861XXXXX@188.120.93.114:1025 T=sip:86155XXXX@sip1.uni-tel.dk IP=178.XX.XX.XX ID=6de881ec07f9c6494ee589cf208da358(a)10.11.87.206 Apr 6 08:05:48 sip-core-1 /usr/local/sbin/kamailio[9186]: ERROR: carrierroute [cr_func.c:95]: cannot find AVP 'carrier' Apr 6 08:05:48 sip-core-1 /usr/local/sbin/kamailio[9186]: ERROR: carrierroute [cr_func.c:805]: invalid carrier id -1 Apr 6 08:05:48 sip-core-1 /usr/local/sbin/kamailio[9186]: ERROR: <script>: cr_next_domain failed Shared memory size is 128M and over halv is free just before the error. The server is in production and does handle debug>1 well, so I do not have much information in the log files. Private memory is the default size. Any ideas what it could be, or how to investegate further? I think my next steps would be to increase the private memory og to increase children=4 to children=8 -- Morten Isaksen

3 7

Command 'sercmd lcr.dump_rules' out of memory.
by Alexandre Abreu 26 Apr '11

26 Apr '11

Hello. When running 'sercmd lcr.reload' (without any errors) and then 'sercmd lcr.dump_rules' I've got the following message: ERROR: ctl [binrpc_run.c:510]: ERROR: binrprc: rpc_send: too many message chunks Then I increased the CTL module params 'binrpc_max_body_size' and ' binrpc_struct_max_body_size'. Re-run the command: ERROR: ctl [binrpc_run.c:908]: ERROR: binrpc: rpc_add: out of memory kamailio -V version: kamailio 3.1.3 (x86_64/linux) flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: compiled on 19:43:43 Apr 18 2011 with gcc 4.1.2 It doesn't matter if I increase the PKG_SIZE (> 8). The LCR tables are very small. lcr_rule - 445 lines lcr_rule_target - 445 lines lcr_gw - 7 lines Alexandre

2 1

Re: [SR-Users] Ipv6 SIP UA
by Klaus Darilion 26 Apr '11

26 Apr '11

You are right! There are actually 2 issues with pjsip: - UDP6 transport does not support DNS lookups. Thus you must set an outboundproxy with IPv6 address - pjsua API does not handle IPv6 media (SDP) As Qjsimple is based on pjsip's pjsua API QjSimple of course has the same issue. regards klaus Am 21.04.2011 14:05, schrieb Pavel Segeč: > Klaus, > > still the same, > > here is a log when UDP is used, there is only A record resolving > > QjSimple: starting initialization/registration ... > QjSimple: initializing SIP stack ... > QjSimple: pjsua_create succeeded ... > QjSimple: "qjsimple" memory pool created ... > QjSimple: DLL loaded ... > QjSimple: GetNetworkParams found ... > QjSimple: GetNetworkParams done ... > QjSimple: cfg.nameserver_count=0 > QjSimple: pjsua_init succeeded ... > QjSimple: pjsip_endpt_register_module succeeded ... > QjSimple: adding UDP transport ... > QjSimple: pjsip_endpt_register_module succeeded ... > QjSimple: outboundUri=sip:ps.sip.uniza.sk;lr;transport=udp > QjSimple: registration_enabled=TRUE > QjSimple: publish_enabled=FALSE > QjSimple: pjsua_acc_add succeeded ... > QjSimple: finished intializing SIP stack, start pjsua ... > QjSimple: Info: on_reg_state_slot called > QjSimple: Info: on_reg_state_slot: status=502 > QjSimple: Info: on_reg_state_slot: statustext=Bad Gateway > QjSimple: Info: on_reg_state_slot called > QjSimple: Info: on_reg_state_slot: status=502 > QjSimple: Info: on_reg_state_slot: statustext=Bad Gateway > > and here when UDP6 > > QjSimple: starting initialization/registration ... > QjSimple: initializing SIP stack ... > QjSimple: pjsua_create succeeded ... > QjSimple: "qjsimple" memory pool created ... > QjSimple: DLL loaded ... > QjSimple: GetNetworkParams found ... > QjSimple: GetNetworkParams done ... > QjSimple: cfg.nameserver_count=0 > QjSimple: pjsua_init succeeded ... > QjSimple: pjsip_endpt_register_module succeeded ... > QjSimple: adding UDP6 transport ... > QjSimple: pjsip_endpt_register_module succeeded ... > QjSimple: outboundUri=sip:ps.sip.uniza.sk;lr;transport=udp > QjSimple: registration_enabled=TRUE > QjSimple: publish_enabled=FALSE > QjSimple: pjsua_acc_add succeeded ... > QjSimple: finished intializing SIP stack, start pjsua ... > > there is no DNS at all > > pavel > > >> -----Original Message----- >> From: sr-users-bounces(a)lists.sip-router.org [mailto:sr-users- >> bounces(a)lists.sip-router.org] On Behalf Of Klaus Darilion >> Sent: Thursday, April 21, 2011 1:27 PM >> To: sr-users(a)lists.sip-router.org >> Subject: Re: [SR-Users] Ipv6 SIP UA >> >> Pavel, I have updated QjSimple to latest pjsip version (0.8.10). It >> would be great if you can test again. If the problem still exists I will >> post a bug report for pjsip. >> >> http://www.ipcom.at/fileadmin/ipcom/files/qjsimple/QjSimple-0.6.6.zip >> >> regards >> Klaus >> >> Am 21.04.2011 10:10, schrieb Pavel Segeč: >>> Hi Klaus, >>> >>> I'm testing your QJSimple client and I'm observing following problems. >> In a IPv4 environments (win 7/32 or 64bit), the client is using a >> classical DNS resolving with SRV and so on, and I'm able to register. >> But inside of the IPV6 environment (IPv4 disabled), when UDP transport >> is used, the client is making just the A DNS record resolving on my >> domain name, not SRV at all. >>> >>> Here is debug log: >>> >>> QjSimple: starting registration ... >>> QjSimple: initializing SIP stack ... >>> QjSimple: pjsua_create succeeded ... >>> QjSimple: "qjsimple" memory pool created ... >>> QjSimple: DLL loaded ... >>> QjSimple: GetNetworkParams found ... >>> QjSimple: GetNetworkParams done ... >>> QjSimple: cfg.nameserver_count=0 >>> QjSimple: pjsua_init succeeded ... >>> QjSimple: pjsip_endpt_register_module succeeded ... >>> QjSimple: adding UDP transport ... >>> QjSimple: pjsip_endpt_register_module succeeded ... >>> QjSimple: outboundUri=sip:ps.sip.uniza.sk;lr;transport=udp >>> QjSimple: publish_enabled=FALSE >>> QjSimple: pjsua_acc_add succeeded ... >>> QjSimple: finished intializing SIP stack, start pjsua ... >>> QjSimple: initializing SIP stack ... done >>> QjSimple: Info: on_reg_state_slot called >>> QjSimple: Info: on_reg_state_slot: status=502 >>> QjSimple: Info: on_reg_state_slot: statustext=Bad Gateway >>> >>> When I setup as the transport protocol UDP6 there is no actions at >> all, no dns, no sip, he re is debug log >>> >>> QjSimple: starting registration ... >>> QjSimple: initializing SIP stack ... >>> QjSimple: pjsua_create succeeded ... >>> QjSimple: "qjsimple" memory pool created ... >>> QjSimple: DLL loaded ... >>> QjSimple: GetNetworkParams found ... >>> QjSimple: GetNetworkParams done ... >>> QjSimple: cfg.nameserver_count=0 >>> QjSimple: pjsua_init succeeded ... >>> QjSimple: pjsip_endpt_register_module succeeded ... >>> QjSimple: adding UDP6 transport ... >>> QjSimple: pjsip_endpt_register_module succeeded ... >>> QjSimple: outboundUri=sip:ps.sip.uniza.sk;lr;transport=udp >>> QjSimple: publish_enabled=FALSE >>> QjSimple: pjsua_acc_add succeeded ... >>> QjSimple: finished intializing SIP stack, start pjsua ... >>> QjSimple: initializing SIP stack ... done >>> >>> regards >>> >>> pavel >>> >>>> -----Original Message----- >>>> From: sr-users-bounces(a)lists.sip-router.org [mailto:sr-users- >>>> bounces(a)lists.sip-router.org] On Behalf Of Klaus Darilion >>>> Sent: Wednesday, April 20, 2011 12:49 PM >>>> To: sr-users(a)lists.sip-router.org >>>> Subject: Re: [SR-Users] Ipv6 SIP UA >>>> >>>> >>>> >>>> Am 20.04.2011 11:01, schrieb Pavel Segeč: >>>>> Hi, >>>>> >>>>> yes I can agree now, linphone is working well, qjsimple has some >>>> problem >>>> >>>> If you find some problems (except that it does not work over TCL/TLS: >>>> http://trac.pjsip.org/repos/ticket/1127) please let me know. >>>> >>>> klaus >>>> >>>> _______________________________________________ >>>> 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 >>>> >>>> >>>> __________ Informacia od ESET NOD32 Antivirus, verzia databazy 6057 >>>> (20110420) __________ >>>> >>>> Tuto spravu preveril ESET NOD32 Antivirus. >>>> >>>> http://www.eset.sk >>>> >>>> >>>> __________ Informacia od ESET NOD32 Antivirus, verzia databazy 6059 >>>> (20110420) __________ >>>> >>>> Tuto spravu preveril ESET NOD32 Antivirus. >>>> >>>> http://www.eset.sk >>>> >>> >>> >>> __________ Informacia od ESET NOD32 Antivirus, verzia databazy 6059 >> (20110420) __________ >>> >>> Tuto spravu preveril ESET NOD32 Antivirus. >>> >>> http://www.eset.sk >>> >>> >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >> sr-users(a)lists.sip-router.org >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> >> >> __________ Informacia od ESET NOD32 Antivirus, verzia databazy 6060 >> (20110421) __________ >> >> Tuto spravu preveril ESET NOD32 Antivirus. >> >> http://www.eset.sk >> > > > __________ Informacia od ESET NOD32 Antivirus, verzia databazy 6060 (20110421) __________ > > Tuto spravu preveril ESET NOD32 Antivirus. > > http://www.eset.sk > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

3 2

Try backup route
by qmail＠top-consulting.net 22 Apr '11

22 Apr '11

I'm using Kamailio 3.1 I have a custom routing statement called MYROUTE which is placed like this: route { ... route (MYROUTE); route (RELAY);} Inside MYROUTE I do some custom SQL checking that basically selects the least cost provider for the ongoing call. The MYROUTE routine sets the $rU, $rd and $rp variables in case a provider has been found for the dialed number or fails with Sip code 404. All this works perfectly, except in my database I also have an alternate provider that I only wish to use as backup in case the main provider returns a 5XX code. How do I get Kamailio to try the second provider after the first one failed ? So far I've tried setting up a t_on_failure() command & routine that picks up the backup provider from the database and sets $rU, $rd margin-right:0px;margin-bottom:0px;margin-left:0px;padding-top:0px;padding-right:0px;padding-bottom:0px;padding-left:0px;"> 0(11460) DEBUG: [parser/msg_parser.c:167]: get_hdr_field: cseq : 0(11460) DEBUG: [parser/msg_parser.c:201]: DEBUG: get_hdr_body : content_length=0 0(11460) DEBUG: [parser/msg_parser.c:103]: found end of header 0(11460) DEBUG: maxfwd [mf_funcs.c:66]: max_forwards header not found! 0(11460) DEBUG: [parser/parse_to.c:174]: DEBUG: add_param: tag=4a2aab60 0(11460) DEBUG: [parser/parse_to.c:803]: end of header reached, state=29 0(11460) DEBUG: siputils [checks.c:76]: totag found 0(11460) DEBUG: rr [loose.c:108]: No Route headers found 0(11460) DEBUG: rr [loose.c:910]: There is no Route HF I believe the problem is that t_on_failure() does not keep all the proper data associated with the call and that's why it fails. Maybe t_on_failure isn't the right command to use here after all (?). So to resume. Here's what I wish to achieve: 1 Call comes through2. Goes through MYROUTE where a provider is selected3. Call is attempted through provider found in #24. If call fails , select another provider5. Call is attempted through provider found in #4 Thanks!

2 1

Try backup route
by qmail＠polarismail.com 21 Apr '11

21 Apr '11

I'm using Kamailio 3.1 I have a custom routing statement called MYROUTE which is placed like this: route { ... route (MYROUTE); route (RELAY);} Inside MYROUTE I do some custom SQL checking that basically selects the least cost provider for the ongoing call. The MYROUTE routine sets the $rU, $rd and $rp variables in case a provider has been found for the dialed number or fails with Sip code 404. All this works perfectly, except in my database I also have an alternate provider that I only wish to use as backup in case the main provider returns a 5XX code. How do I get Kamailio to try the second provider after the first one failed ? So far I've tried setting up a t_on_failure() command & routine that picks up the backup provider from the database and sets $rU, $rd margin-right:0px;margin-bottom:0px;margin-left:0px;padding-top:0px;padding-right:0px;padding-bottom:0px;padding-left:0px;"> 0(11460) DEBUG: [parser/msg_parser.c:167]: get_hdr_field: cseq : 0(11460) DEBUG: [parser/msg_parser.c:201]: DEBUG: get_hdr_body : content_length=0 0(11460) DEBUG: [parser/msg_parser.c:103]: found end of header 0(11460) DEBUG: maxfwd [mf_funcs.c:66]: max_forwards header not found! 0(11460) DEBUG: [parser/parse_to.c:174]: DEBUG: add_param: tag=4a2aab60 0(11460) DEBUG: [parser/parse_to.c:803]: end of header reached, state=29 0(11460) DEBUG: siputils [checks.c:76]: totag found 0(11460) DEBUG: rr [loose.c:108]: No Route headers found 0(11460) DEBUG: rr [loose.c:910]: There is no Route HF I believe the problem is that t_on_failure() does not keep all the proper data associated with the call and that's why it fails. Maybe t_on_failure isn't the right command to use here after all (?). So to resume. Here's what I wish to achieve: 1 Call comes through2. Goes through MYROUTE where a provider is selected3. Call is attempted through provider found in #24. If call fails , select another provider5. Call is attempted through provider found in #4 Thanks!

1 0

BLF using kamailio
by Lucas Alvarez 21 Apr '11

21 Apr '11

Hi, I have implemented kamailio with asterisk, I want to enable BLF using kamailio instead of asterisk. I would appreciate any kind of help, below I'm pasting my configuration. *I'm testing with an Aastra phone, when phone sends the subscription kamailio answers with:* U 192.168.15.22:5060 -> 192.168.15.108:5060 SIP/2.0 489 Bad Event. Via: SIP/2.0/UDP 192.168.15.108;branch=z9hG4bKa54a155ad67706dca. From: "1104" <sip:1104@192.168.15.22:5060>;tag=71cf5c9949. To: <sip:1104@192.168.15.22:5060>;tag=99b712eabe88b2bf7a20409f3dc7ebf1-f71b. Call-ID: f22823b8b4d58a8e. CSeq: 3430 SUBSCRIBE. Allow-Events: presence.winfo, presence, dialog. Server: kamailio (3.1.3 (i386/linux)). Content-Length: 0. *This is my presence configuration:* #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_dialoginfo.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) modparam("presence", "server_address", "sip:192.168.15.22:5060") modparam("presence_dialoginfo", "force_single_dialog", 1) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif

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New developer: Jason Penton
by Daniel-Constantin Mierla 21 Apr '11

21 Apr '11

Hello, I want to announce Jason Penton as a new registered developer, with write access to GIT repository - his username id is 'jason.penton'. Jason already submitted an IMS-related module that Carsten added to his IMS branch in the last days, therefore expect contributions from Jason in the IMS extensions. Thanks, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

2 1

Re: [SR-Users] Ipv6 SIP UA
by Klaus Darilion 21 Apr '11

21 Apr '11

Pavel, I have updated QjSimple to latest pjsip version (0.8.10). It would be great if you can test again. If the problem still exists I will post a bug report for pjsip. http://www.ipcom.at/fileadmin/ipcom/files/qjsimple/QjSimple-0.6.6.zip regards Klaus Am 21.04.2011 10:10, schrieb Pavel Segeč: > Hi Klaus, > > I'm testing your QJSimple client and I'm observing following problems. In a IPv4 environments (win 7/32 or 64bit), the client is using a classical DNS resolving with SRV and so on, and I'm able to register. But inside of the IPV6 environment (IPv4 disabled), when UDP transport is used, the client is making just the A DNS record resolving on my domain name, not SRV at all. > > Here is debug log: > > QjSimple: starting registration ... > QjSimple: initializing SIP stack ... > QjSimple: pjsua_create succeeded ... > QjSimple: "qjsimple" memory pool created ... > QjSimple: DLL loaded ... > QjSimple: GetNetworkParams found ... > QjSimple: GetNetworkParams done ... > QjSimple: cfg.nameserver_count=0 > QjSimple: pjsua_init succeeded ... > QjSimple: pjsip_endpt_register_module succeeded ... > QjSimple: adding UDP transport ... > QjSimple: pjsip_endpt_register_module succeeded ... > QjSimple: outboundUri=sip:ps.sip.uniza.sk;lr;transport=udp > QjSimple: publish_enabled=FALSE > QjSimple: pjsua_acc_add succeeded ... > QjSimple: finished intializing SIP stack, start pjsua ... > QjSimple: initializing SIP stack ... done > QjSimple: Info: on_reg_state_slot called > QjSimple: Info: on_reg_state_slot: status=502 > QjSimple: Info: on_reg_state_slot: statustext=Bad Gateway > > When I setup as the transport protocol UDP6 there is no actions at all, no dns, no sip, he re is debug log > > QjSimple: starting registration ... > QjSimple: initializing SIP stack ... > QjSimple: pjsua_create succeeded ... > QjSimple: "qjsimple" memory pool created ... > QjSimple: DLL loaded ... > QjSimple: GetNetworkParams found ... > QjSimple: GetNetworkParams done ... > QjSimple: cfg.nameserver_count=0 > QjSimple: pjsua_init succeeded ... > QjSimple: pjsip_endpt_register_module succeeded ... > QjSimple: adding UDP6 transport ... > QjSimple: pjsip_endpt_register_module succeeded ... > QjSimple: outboundUri=sip:ps.sip.uniza.sk;lr;transport=udp > QjSimple: publish_enabled=FALSE > QjSimple: pjsua_acc_add succeeded ... > QjSimple: finished intializing SIP stack, start pjsua ... > QjSimple: initializing SIP stack ... done > > regards > > pavel > >> -----Original Message----- >> From: sr-users-bounces(a)lists.sip-router.org [mailto:sr-users- >> bounces(a)lists.sip-router.org] On Behalf Of Klaus Darilion >> Sent: Wednesday, April 20, 2011 12:49 PM >> To: sr-users(a)lists.sip-router.org >> Subject: Re: [SR-Users] Ipv6 SIP UA >> >> >> >> Am 20.04.2011 11:01, schrieb Pavel Segeč: >>> Hi, >>> >>> yes I can agree now, linphone is working well, qjsimple has some >> problem >> >> If you find some problems (except that it does not work over TCL/TLS: >> http://trac.pjsip.org/repos/ticket/1127) please let me know. >> >> klaus >> >> _______________________________________________ >> 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 >> >> >> __________ Informacia od ESET NOD32 Antivirus, verzia databazy 6057 >> (20110420) __________ >> >> Tuto spravu preveril ESET NOD32 Antivirus. >> >> http://www.eset.sk >> >> >> __________ Informacia od ESET NOD32 Antivirus, verzia databazy 6059 >> (20110420) __________ >> >> Tuto spravu preveril ESET NOD32 Antivirus. >> >> http://www.eset.sk >> > > > __________ Informacia od ESET NOD32 Antivirus, verzia databazy 6059 (20110420) __________ > > Tuto spravu preveril ESET NOD32 Antivirus. > > http://www.eset.sk > >

2 1

Re: [SR-Users] Ipv6 SIP UA
by Pavel Segeč 21 Apr '11

21 Apr '11

Hi, and the IMS Boghe client can be used for IPv6 call testing too. pavel > -----Original Message----- > From: sr-users-bounces(a)lists.sip-router.org [mailto:sr-users- > bounces(a)lists.sip-router.org] On Behalf Of Pavel Segeč > Sent: Thursday, April 21, 2011 10:10 AM > To: 'SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List' > Subject: Re: [SR-Users] Ipv6 SIP UA > > Hi Klaus, > > I'm testing your QJSimple client and I'm observing following problems. > In a IPv4 environments (win 7/32 or 64bit), the client is using a > classical DNS resolving with SRV and so on, and I'm able to register. > But inside of the IPV6 environment (IPv4 disabled), when UDP transport > is used, the client is making just the A DNS record resolving on my > domain name, not SRV at all. > > Here is debug log: > > QjSimple: starting registration ... > QjSimple: initializing SIP stack ... > QjSimple: pjsua_create succeeded ... > QjSimple: "qjsimple" memory pool created ... > QjSimple: DLL loaded ... > QjSimple: GetNetworkParams found ... > QjSimple: GetNetworkParams done ... > QjSimple: cfg.nameserver_count=0 > QjSimple: pjsua_init succeeded ... > QjSimple: pjsip_endpt_register_module succeeded ... > QjSimple: adding UDP transport ... > QjSimple: pjsip_endpt_register_module succeeded ... > QjSimple: outboundUri=sip:ps.sip.uniza.sk;lr;transport=udp > QjSimple: publish_enabled=FALSE > QjSimple: pjsua_acc_add succeeded ... > QjSimple: finished intializing SIP stack, start pjsua ... > QjSimple: initializing SIP stack ... done > QjSimple: Info: on_reg_state_slot called > QjSimple: Info: on_reg_state_slot: status=502 > QjSimple: Info: on_reg_state_slot: statustext=Bad Gateway > > When I setup as the transport protocol UDP6 there is no actions at all, > no dns, no sip, he re is debug log > > QjSimple: starting registration ... > QjSimple: initializing SIP stack ... > QjSimple: pjsua_create succeeded ... > QjSimple: "qjsimple" memory pool created ... > QjSimple: DLL loaded ... > QjSimple: GetNetworkParams found ... > QjSimple: GetNetworkParams done ... > QjSimple: cfg.nameserver_count=0 > QjSimple: pjsua_init succeeded ... > QjSimple: pjsip_endpt_register_module succeeded ... > QjSimple: adding UDP6 transport ... > QjSimple: pjsip_endpt_register_module succeeded ... > QjSimple: outboundUri=sip:ps.sip.uniza.sk;lr;transport=udp > QjSimple: publish_enabled=FALSE > QjSimple: pjsua_acc_add succeeded ... > QjSimple: finished intializing SIP stack, start pjsua ... > QjSimple: initializing SIP stack ... done > > regards > > pavel > > > -----Original Message----- > > From: sr-users-bounces(a)lists.sip-router.org [mailto:sr-users- > > bounces(a)lists.sip-router.org] On Behalf Of Klaus Darilion > > Sent: Wednesday, April 20, 2011 12:49 PM > > To: sr-users(a)lists.sip-router.org > > Subject: Re: [SR-Users] Ipv6 SIP UA > > > > > > > > Am 20.04.2011 11:01, schrieb Pavel Segeč: > > > Hi, > > > > > > yes I can agree now, linphone is working well, qjsimple has some > > problem > > > > If you find some problems (except that it does not work over TCL/TLS: > > http://trac.pjsip.org/repos/ticket/1127) please let me know. > > > > klaus > > > > _______________________________________________ > > 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 > > > > > > __________ Informacia od ESET NOD32 Antivirus, verzia databazy 6057 > > (20110420) __________ > > > > Tuto spravu preveril ESET NOD32 Antivirus. > > > > http://www.eset.sk > > > > > > __________ Informacia od ESET NOD32 Antivirus, verzia databazy 6059 > > (20110420) __________ > > > > Tuto spravu preveril ESET NOD32 Antivirus. > > > > http://www.eset.sk > > > > > __________ Informacia od ESET NOD32 Antivirus, verzia databazy 6059 > (20110420) __________ > > Tuto spravu preveril ESET NOD32 Antivirus. > > http://www.eset.sk > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > __________ Informacia od ESET NOD32 Antivirus, verzia databazy 6059 > (20110420) __________ > > Tuto spravu preveril ESET NOD32 Antivirus. > > http://www.eset.sk > __________ Informacia od ESET NOD32 Antivirus, verzia databazy 6059 (20110420) __________ Tuto spravu preveril ESET NOD32 Antivirus. http://www.eset.sk

1 0

Re: [SR-Users] Ipv6 SIP UA
by Klaus Darilion 21 Apr '11

21 Apr '11

Am 20.04.2011 11:01, schrieb Pavel Segeč: > Hi, > > yes I can agree now, linphone is working well, qjsimple has some problem If you find some problems (except that it does not work over TCL/TLS: http://trac.pjsip.org/repos/ticket/1127) please let me know. klaus

2 1

xmlrpc requests counted as unsupported_methods
by Juha Heinanen 21 Apr '11

21 Apr '11

looks like core stats counts xmlrpc requests served by route [xmlrpc_requests] as unsupported_methods. is there anything a user can do about it or is it a bug that should be fixed? -- juha

2 4

Broken Via/reply-matching for natping OPTION
by Andreas Granig 20 Apr '11

20 Apr '11

Hi, When kamailio-3.1 sends OPTIONS for natping, the Via branch is wrong, e.g: OPTIONS sip:192.168.51.1:5060 SIP/2.0. Via: SIP/2.0/UDP 127.0.0.1:5062;branch=0. Route: <something> Should be branch=z9hG4bK<something> instead of branch=0, right? When a reply comes back, like this... SIP/2.0 100 Trying. Via: SIP/2.0/UDP 127.0.0.1:5062;branch=0;rport=5062. From: sip:pinger@<something> ... then kamailio logs this: INFO: <core> [forward.c:786]: broken reply to forward - no 2nd via The full debug log looks like this: DEBUG: <core> [parser_helpers.c:196]: path for branch: '<sip:lb@127.0.0.1:5060;lr;received=sip:192.168.51.1:5060>' DEBUG: <core> [parser/msg_parser.c:640]: SIP Reply (status): DEBUG: <core> [parser/msg_parser.c:642]: version: <SIP/2.0> DEBUG: <core> [parser/msg_parser.c:644]: status: <100> DEBUG: <core> [parser/msg_parser.c:646]: reason: <Trying> DEBUG: <core> [parser/parse_via.c:1287]: Found param type 232, <branch> = <0>; state=6 DEBUG: <core> [parser/parse_via.c:1287]: Found param type 235, <rport> = <5062>; state=16 DEBUG: <core> [parser/parse_via.c:2300]: end of header reached, state=5 DEBUG: <core> [parser/msg_parser.c:515]: parse_headers: Via found, flags=2 DEBUG: <core> [parser/msg_parser.c:517]: parse_headers: this is the first via DEBUG: <core> [receive.c:145]: After parse_msg... DEBUG: tm [t_lookup.c:1081]: DEBUG: t_check_msg: msg id=1 global id=0 T start=0xffffffffffffffff DEBUG: <core> [parser/parse_to.c:803]: end of header reached, state=9 DEBUG: <core> [parser/msg_parser.c:187]: DEBUG: get_hdr_field: <To> [23]; uri=[sip:192.168.51.1:5060] DEBUG: <core> [parser/msg_parser.c:189]: DEBUG: to body [sip:192.168.51.1:5060#015#012] DEBUG: <core> [parser/msg_parser.c:167]: get_hdr_field: cseq <CSeq>: <1> <OPTIONS> DEBUG: tm [t_lookup.c:1055]: DEBUG: t_reply_matching: failure to match a transaction DEBUG: tm [t_lookup.c:1150]: DEBUG: t_check_msg: msg id=1 global id=1 T end=(nil) DEBUG: <core> [parser/msg_parser.c:201]: DEBUG: get_hdr_body : content_length=0 DEBUG: <core> [parser/msg_parser.c:103]: found end of header INFO: <core> [forward.c:786]: broken reply to forward - no 2nd via DEBUG: <core> [usr_avp.c:646]: DEBUG:destroy_avp_list: destroying list (nil) Thanks, Andreas

4 6

planning next major release 3.2
by Daniel-Constantin Mierla 20 Apr '11

20 Apr '11

Hello, soon we should sketch the roadmap for the next major release 3.2. There are some private git branches with lot of work on it, so we need to synchronize a bit and decide about the proper time: before or after summer. Several devels will be in Berlin for Linuxtag, but no so much overlapping for a big face to face meeting, therefore might be better to have a IRC chat conference sometime in the near future. Next week will be fine for me, say Wednesaday till Friday, in the afternoon to allow fair time of the day for US timezones. If you have other proposals, speak up. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

2 3

Re: [SR-Users] Ipv6 SIP UA
by Daniel-Constantin Mierla 20 Apr '11

20 Apr '11

Hello, On 4/19/11 9:58 AM, Pavel Segeč wrote: > Hi, > > may someone help me and recommend a software SIP client with a good IPv6 > support implemented? I tried Xlite, Bria, Linphone, SIP communicator and > Jitsi but with a bad results. Each have some problem within pure IPv6 > environment. the last sipit I attended (last year, though) was not rich at all in IPv6 softphones. In the past there was a patched version of KPhone we used at FhG Fokus for some tests, but I wouldn't call it production ready: http://old.iptel.org/products/kphone/ From what was listed above, you touched most of the active softphones, afaik ekiga,qutecom and twinkle do not have it, not sure about zoiper. Cheers, Daniel

4 3

Re: [SR-Users] Ipv6 SIP UA
by Klaus Darilion 19 Apr '11

19 Apr '11

I once wrote a SIP client based on pjsip: http://www.ipcom.at/en/telephony/qjsimple/ AFAIK it works will using IPv6 when using UDP. regards Klaus Am 19.04.2011 09:58, schrieb Pavel Segeč: > Hi, > > may someone help me and recommend a software SIP client with a good IPv6 > support implemented? I tried Xlite, Bria, Linphone, SIP communicator and > Jitsi but with a bad results. Each have some problem within pure IPv6 > environment. > > palo > > > __________ Informacia od ESET NOD32 Antivirus, verzia databazy 6053 > (20110418) __________ > > Tuto spravu preveril ESET NOD32 Antivirus. > > http://www.eset.sk > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

1 0

ser flavour accounting
by Claudio Furrer 19 Apr '11

19 Apr '11

Hello, I'm trying to migrate ser-0.9.x to the current sr-3.1.x and right now trying to figure out the acc functions/parameters. I used to use "acc_db_request" from acc module, but it's not working using acc_db and acc_syslog in conjuntion. acc_db has acc_db_request[0,1] but are not callable from the configuration file, and acc_syslog has acc_db_request, but doesn't matter if i load this module, the function isn't found... loading modules under /usr/lib/siprouter/modules:/usr/lib/siprouter/modules_s 0(23455) : <core> [cfg.y:3412]: parse error in config file /tmp/ser.cfg, line 358, column 74: unknown command, missing loadmodule? line 358 is: acc_db_request("480 Offline User", "missed_calls"); I would appreciate if you can guide me through the ACC documentation or changes that it has suffered.. I tried to track the git log.. but many changes since my old ser :( Thank you Claudio pd: FYI i did this modifications to adjust my acc settings: # Accounting module +loadmodule "acc_syslog.so" loadmodule "acc_db.so" # ----------------- setting module-specific parameters --------------- @@ -139,15 +140,13 @@ # number of flag, which will be used for accounting; if a message is # labeled with this flag, its completion status will be reported # modparam("acc", "log_flag", 1 ) -modparam("acc_db", "log_level", 1) +modparam("acc_syslog", "log_level", 1) # MySQL acc params modparam("acc_db", "db_url", "mysql://ser:heslo@localhost/ser") -modparam("acc_db", "db_flag", 2) -modparam("acc_db", "db_missed_flag", 3) +modparam("acc_db", "log_flag", 2) +modparam("acc_db", "log_missed_flag", 3) modparam("acc_db", "log_fmt", "miocfspD") -# Acc time set to localtime (1) -modparam("acc_db", "db_localtime", 1)

2 1

modules_s/uri_db functions
by Claudio Furrer 19 Apr '11

19 Apr '11

Hello, What happened to checks.c file on uri_db module (modules_s)? Seem to be an empty file now, only headers exist. Is it the same to use modules_k/ version? I'm currently at v3.1.. Regards, Claudio

2 1

tm stats pseudo variables?
by Juha Heinanen 19 Apr '11

19 Apr '11

is there any means to access native tm stats from script? i know about $stat pvs for accessing k versions of tm stats, but i want access to native tm stats. -- juha

3 2

RTPProxy and bridging IPv4/IPv6 with parallel forking
by Dan-Cristian Bogos 19 Apr '11

19 Apr '11

Hey Guys, I was recently playing with gateway-ing IPv4-IPv6 and hit the following scenario: * AOR having contacts on both ipv4 and ipv6 and I wanted to do parallel forking. RTPProxy bridging works without any issue on a normal setup, however the problem shows up when needing to make calls toward rtpproxy to return both sides of bridge or only one (ee and ie combinations). Did any of you experiment with this scenario? EG: Call comes from ipv4, you want to send it to both ipv4 and ipv6. The branch route looks the right place to call rtpproxy but when calling unforce_rtp_proxy() on CANCEL, will rtpproxy be aware about which ports are we trying to cancel? One more issue would be with re-invite which must go out with the same ip of rtpproxy as original INVITE. Here we could store the bridge direction in some route parameter but unfortunately adding route params is not possible in branches. So what do u think? DanB PS: The "normal" setup of forking calls only ipv4 or only ipv6 works smooth, so support of ipv6 in kamailio or rtpproxy is not questionable.

4 10

stats in kamailio
by Ricardo Martinez 19 Apr '11

19 Apr '11

Hello. I’m trying to get statistics to use charts in SIREMIS v2.0. I was able to load the “Memory” statistics but I want to know if there is more information to use from Kamailio, right now I have : $stat(used_size) $stat(real_used_size) $stat(max_used_size) $stat(free_size) $stat(location-users) $stat(location-contacts) Where can I find the complete list of statistics to get from kamailio?. Thanks in advance. Ricardo Martinez.-

5 13

Attended transfer in load-balanced multi-server setup
by Pan B. Christensen 18 Apr '11

18 Apr '11

Hello all, I'm having problems with attended transfer in certain scenarios. Let me first try to document the relevant parts of the setup: There are two load-balanced Asterisk gateways connected to the old Nortel system. Currently, PSTN connections are done by Nortel. We then have two load-balanced kamailio servers. The phones are connected to these. There are several other servers not relevant to this question. Blind transfer works in any scenario. I've been told that attended transfer works in any scenario if the original call is SIP to SIP. If the original call is from Nortel/PSTN to SIP, then attended transfer will usually not work. Scenarios: A call comes in from Nortel/PSTN (user A) through Nortel1 to a SIP device (user B). User B pushes the transfer button (puts current call on hold and makes a new call) and dials a Nortel / PSTN number to user C. This call goes out through the nortel2 server. Users B and C talk a short while before user B pushes the Transfer button again to connect users A and C. In this scenario, the REFER is forwarded by kamailio to nortel1, which replies "SIP/2.0 481 Call leg/transaction does not exist." The call to join is of course on the other asterisk (nortel2). A similar scenario is where user B transfers to another SIP user. This call will only exist in one of the kamailio servers, and Asterisk will give the same response to the REFER. A third scenario is where both calls are handled by the same Asterisk server. This scenario works. I'm assuming I'll have to build REFER-handling logic into kamailio, but am unsure of how to proceed. Any suggestions? Thanks in advance. With kind regards, Pan

2 2

Notifying RLS when XML documents are updated
by Peter Dunkley 18 Apr '11

18 Apr '11

Hello, I am using Kamailio RLS and integrated XCAP Server. When I add a contact with my client it updates my XCAP documents but there is no additional SIP messaging between the client and Kamailio. I have been assured (by the client author) that this is the correct behaviour. However, having looked at both the XCAP Server and RLS module code I cannot see anything that will either inform RLS that a document has been updated or a timer to check for that. Have I missed something obvious? Thanks, Peter -- Peter Dunkley Technical Director Crocodile RCS Ltd

1 0

perlvdb error
by Roman Yeryomin 18 Apr '11

18 Apr '11

Hi! I'm trying to use perlvdb module but seems that it's not working. Seems that needed functions are not exported correctly: kamailio[11525]: ERROR: <core> [db.c:79]: module db_perlvdb does not export db_use_table function Looking at perlvdb.c one can see that module exports it's functions as "perlvdb", not "db_perlvdb". But changing "perlvdb" to "db_perlvdb" results in segfault when running: kamailio[31336]: segfault at 1 ip b7365e3a sp bfa82fb0 error 4 in perlvdb.so[b7362000+6000] Do I misunderstand something completely or there is a bug? Anyone actually tried using this module or it just sits there forgotten? Regards, Roman

3 3

Probing mode howto?
by Spinov Evgeniy 18 Apr '11

18 Apr '11

Hello, all. Is there any possibility to set one of dispatcher destinations to probing mode manually? I can set it to active/inactive, however cannot set it to probing via MI. This is really required as, for instance, when asterisk reload takes like 1 minute to reload ( huge DB ), I want to set node to inactive first to do not make call attempts, and after reload set it to probing. Thank you.

1 0

loose_route security
by Eric Hiller 18 Apr '11

18 Apr '11

As I look and play with loose_route functionality it seems that by simply placing a route: proxyip;lr header in my invite I can bypass any and all security otherwise built into the configuration. Is this the way everyone has it? I have been unable to find any configuration examples online that show how to secure/restrict access to loose_route? Ideas or suggestions? Thanks, Eric

9 27

Fwd: [RAI] SIPit 28 summary
by Klaus Darilion 17 Apr '11

17 Apr '11

-------- Original-Nachricht -------- Betreff: [RAI] SIPit 28 summary Datum: Fri, 15 Apr 2011 10:47:08 -0500 Von: Robert Sparks <rjsparks(a)nostrum.com> An: rai(a)ietf.org Also available at <https://www.sipit.net/SIPit28_summary> RjS -------- SIPit 28 was hosted by Digium in Huntsville, Alabama, USA the week of Arpil 11-15, 2010. There were 54 attendees from 19 companies visiting from 10 countries. We had 40 distinct implementations. The roles represented (some implementations act in more than one role) 34 endpoints 6 proxy/registrars/b2bua/sbcs 1 dedicated event server Implementations using each transport for SIP messages: UDP 98% TCP 95% TLS 68% server-auth, 50% mutual-auth SCTP 10% DTLS 0% 68% of the implementations present supported IPv6. There was one RFC4474 Identity implementation present. For DNS we had support for: Full RFC3263 : 53% SRV only : 33% A records only : 15% no DNS support : 0% Support for various items in the endpoints: 76% replaces 38% 5389stun 26% turn 24% 3489stun 24% ice 21% sip/stun multiplexing 21% outbound 21% gruu 18% path 18% join 18% diversion 6% history-info (there were no implementations of 4244bis) 3% service-route 3% sigcomp Support for various items in the proxy/registrars: 50% path 50% outbound 33% sip/stun multiplexing 20% diversion 18% gruu 17% history-info 17% service-route The endpoints and B2BUAs implemented these methods: 100% INVITE, CANCEL, ACK, BYE 94% REGISTER 91% OPTIONS 91% REFER 82% NOTIFY 82% INFO 79% UPDATE 71% SUBSCRIBE 68% PRACK 47% PUBLISH 29% MESSAGE 94% of the implementations sent RTP from the port advertised for reception (symmetric-rtp). One of the implementations required the other party to use symmetric-rtp. 88% of the UAs present both sent RTCP and paid attention to RTCP they received. 56% of the endpoints present supported SRTP using sdes. There was one dtls-srtp implementations at this SIPit. 67% of the endpoints supported multipart/MIME. There were no implementations present with S/MIME support. 48% of the implementations present followed RFC6026 (corrections to the INVITE transaction) 68% followed RFC4320 (corrections to the non-INVITE transaction) There were 5 SIP Event Server implementations (not counting endpoints that support events only for refer). There were 9 SIP Event Client implementations (not counting endpoints that support events only for refer). These event packages were supported: Server Client 2 3 presence 1 0 presence.winfo 1 6 message-summary 2 5 dialog 0 2 reg 2 1 conference 1 0 xcap-diff 0 1 kpml These packages were supported with the eventlist extension: Server Client 1 0 presence Five of the proxies present still rely only on max-forwards. There was one implementation of fork-loop-fix, but no implementations of max-breadth. Multiparty tests (Thanks to Eoin McLeod for contributing notes for several tests) The Forking test was very well attended. We had the majority of the participants at the multiparty table. Many teams took away good new information from watching their implementation react to multiple merges and to Via stacks with transports switching between IPv4 and IPv6. Forcing multiple 200 Oks was more difficult than usual given the network equipment at hand. We've created a set of standing automated tests that will allow endpoints and proxies to test that scenario at their leisure. The TLS multiparty leveraged DNS to reduce configuration churn. We ensured a full mesh of connections between participating elements. There were some issues uncovered with the values used in Route/Record-Route not matching the configured certificates at an element. We brainstormed a configuration for the next event that will efficiently expose whether elements are correctly opening a new connection (rather than reusing an existing connection) when a new request with a resource belonging to an authority not matching what had been authenticated on the existing connection arrived. The SRTP tests showed a high level of interoperability (all with SDES keying), including hold/resume and transfer scenarios. There was a successful test of a call with media running long enough to increment the ROC, followed by a hold/resume. Several participants were trying variants of "best-effort/opportunistic" encryption with lower levels of success. These elements tested providing offers with both RTP/AVP and RTP/SAVP media-lines describing the same logical media channel, and with multiple RTP/AVP media lines decorated with different crypto attributes. The STUN/TURN/ICE tests had very high levels of interoperability. The test scenario involved elements in two natted networks sharing the same address space and elements in the public address space (including a separate TURN server for each of the private network islands). We had a mix of elements that supported relay allocation and elements that presented only host/reflexive candidates. A few elements assumed that all of the m-lines in an offer had to present ICE candidates. SDP with a mix of ICE enabled m-lines and m-lines without candidates caused ICE failures. The Outbound test showed several successful flows being created and maintained. A mesh of edge-proxies and proxy/registrars were set up and endpoints established flows through several combinations. There was good UA support for the Flow-Timer, but there were implementations present that didn't randomize the keepalive value. There was not much testing of recovery from failed flows at this event - we'll focus on that at SIPit 29. We held a long multiparty session focusing on exchanging video with constrained bandwidth or impaired (latent or lossy) networks. There was a high level of basic interoperability and reasonable reaction to mid-call shifts in network conditions, but all parties involved came away with changes they wanted to make in their implementations. The Spiral test presented requests to the participating UAs containing a mix of UDP, TCP, and TLS over IPv4 and IPv6 values in large Via and Record-Route stacks. Correct operation of subsequent in-dialog requests in both directions was tested, discovering a few implementation problems, and some invalid assumptions about how dialog-stateful proxies might need to behave, but most scenarios worked well. We followed the usual Spiral test with a "fuzz-ball" test utilizing DNS SRV to randomly distribute the propagation of messages through all of the proxies present, hopping off to the UAs with a low probability, resulting in an easy-to-configure test that exercised a full-mesh of connections between proxies and presented the UAs with very diverse (and large) Via and Record-Route header field values. We exercised the RFC5393 proxy-doom scenario, again leveraging the SRV load-leveling capabilities to simplify configuration. We had messages popping out to the UAs from each level of the tree, giving them a very large-scale merge test. _______________________________________________ RAI mailing list RAI(a)ietf.org https://www.ietf.org/mailman/listinfo/rai

6 11

Meeting in San Francisco
by Daniel-Constantin Mierla 14 Apr '11

14 Apr '11

Hello, I will spend few days in San Francisco area, during May 6-10. If someone wants to meet and discuss about the project, VoIP & stuff, drop me an email. In case there are many people leaving close to SF city, then I can even try to arrange a group meeting, for a dinner or so, to map faces to names and have good drinks -- this decision will be announced later based on feedback, however suggestions about good meeting places are welcome from anyone. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

1 0

Re: [SR-Users] Problems with siremis custom charts
by Daniel-Constantin Mierla 14 Apr '11

14 Apr '11

Do not send me message privately upon answers on the mailing list - keep the mailing list always cc-ed otherwise the messages will be ignored. On 4/14/11 3:02 PM, Camila Troncoso wrote: > > Daniel, > > I want a bars graph and i´m using another table, one that I create. > The graph displays well with the default configuration (the one that > the other graph of siremis have) but I don´t want an area graph. I > search around siremis files and find a file named charts-lib.php. I > suppose this is the script that creates the graphs , but in this file > are only two type of graphs describe near line 130 ( area and > line_dot). I don’t know if this is why I can´t change the graph type. > yes, so far these are the ones supported by siremis for building custom graphs. The type, one of them, is taken from the xml config files describing the graphs. The library for making graphs supports other types, as you could see in the usrloc charts, but that is using a different php code base than the other grpahs. If you want to enhance it with other types of graphs, that would be great and if you send a working patch then we can include it in the next release. Cheers, Daniel > > Regards, > > Camila > > *From:*Daniel-Constantin Mierla [mailto:miconda@gmail.com > <mailto:miconda@gmail.com>] > *Sent:* jueves, 14 de abril de 2011 6:16 > *To:* SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List > *Cc:* Camila Troncoso > *Subject:* Re: [SR-Users] Problems with siremis custom charts > > Hello, > > On 4/13/11 10:56 PM, Camila Troncoso wrote: > > Hi, > > I´m having problems with siremis charts. I create a new graph and all > is working fine, but I want to use another type of graph ( the default > graphs of siremis uses type=”area”) and when I change the graph type > to any other thing it shows me the same graph but only the area under > the curve isn´t paint , but the curve doesn´t change. > > Please some help with this issue… > > what type of graph you try to use? have you checked the database > statistics table records, do they have different values for this > graph? If not, check if you write the desired values from the kamailio > config file there. > > Cheers, > Daniel > > -- > Daniel-Constantin Mierla > http://www.asipto.com -- Daniel-Constantin Mierla http://www.asipto.com

1 0

How I can turn off 100 trying reply
by Derrick Ding 14 Apr '11

14 Apr '11

Hi, All I wonder how I can make Kamailio not to send "100 Trying" msg when receive INVITE request, by modifying kamailio.cfg or source code. Thanks Derrick

4 10

Problems with siremis custom charts
by Camila Troncoso 14 Apr '11

14 Apr '11

Hi, I´m having problems with siremis charts. I create a new graph and all is working fine, but I want to use another type of graph ( the default graphs of siremis uses type=”area”) and when I change the graph type to any other thing it shows me the same graph but only the area under the curve isn´t paint , but the curve doesn´t change. Please some help with this issue… Regards, Camila T. *Camila Troncoso **|* Ingeniero de Desarrollo RedVoiss *|*ctroncoso(a)redvoiss.net Santiago - Chile *|* +56 2 2408535 www.redvoiss.net

2 1

Problems with db_text
by Sascha Daniels 14 Apr '11

14 Apr '11

Hi together, I am trying to make a setup with kamailio and freeswitch, that can run with all supported database types. The basic setup is from this howto: http://kb.asipto.com/freeswitch:kamailio-3.1.x-freeswitch-1.0.6d-sbc With mysql everything is (more or less) fine. Creating db_text databases does not work because /opt/kamailio-3.1/share/kamailio/dbtext/kamailio/uacreg is missing and kamdbctl gives up. Just created an empty file, so kamdbctl can finish its work. The subscribers database is created (works in a more simple setup). Now I am getting this error when starting kamailio: http://pastebin.com/vsiUNdCM Version of table: pua:6 Why does the table version not exist? It was created by "kamdbctl create". I am a bit lost with this debug output. Greets Sascha -- AMOOMA GmbH - Bachstr. 124 - 56566 Neuwied --> http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister Montabaur B14998 Bücher: http://das-asterisk-buch.de - http://ruby-auf-schienen.de

1 0

Problems with siremis custom charts
by Camila Troncoso 13 Apr '11

13 Apr '11

Hi, I´m having problems with siremis charts. I create a new graph and all is working fine, but I want to use another type of graph ( the default graphs of siremis uses type=”area”) and when I change the graph type to any other thing it shows me the same graph but only the area under the curve isn´t paint , but the curve doesn´t change. Please some help with this issue… Regards, Camila T. *Camila Troncoso **|* Ingeniero de Desarrollo RedVoiss *|*ctroncoso(a)redvoiss.net Santiago - Chile *|* +56 2 2408535 www.redvoiss.net

1 0

Registrar Server, location and auth_db table
by Santiago Soares 12 Apr '11

12 Apr '11

Hello, I'm trying to use kamailio 1.5.3 as a registrar server. I have a table on my db that contains all user information, like username, password, expire time and ip address, so kamailio should use the same table to authenticate users AND save their location. The problem is that kamailio complains about table version when starts. It expects a different version number for location table and auth_db table. Is there any way I could use the same table for both functions? I'd appreciate any help! Thank you! -- Santiago Soares Fone: (41) 8488-0537

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Re: [SR-Users] sr-users Digest, Vol 71, Issue 40
by Rick Aldred 12 Apr '11

12 Apr '11

Hi Fedrico, Sorry I missed your reply earlier (thanks by the way). I changed the service priority using the Services GUI utility under System | Administration as I didn't know about the update-rc.d command at the time. Maybe the GUI doesn't do all that's required under the hood or maybe I missed something, but anyway it's working now so that's great. Regards, Rick ------------------------------ Hi, from the log you pasted is clear that the problem is that kamailio is starting before mysql. This because, as you said, they have the same priority and so, being the starting order alphabetical for the same priority, kamailio will start before mysql and cannot connect to mysql which is still not up. You told that changing kamailio priority didn't change the result. How did you change the priority? Did you use the update-rc.d command? Regards, Federico

1 0

OT - Sipwise is hiring SIP and Web hackers
by Andreas Granig 11 Apr '11

11 Apr '11

Hi guys, We're looking for experienced SIP (Kamailio, Sems) and Web (Perl/Catalyst, MySQL, HTML/CSS/jQuery) hackers to work on our sip:provider CE platform and their commercial counter-parts. It's on-site (Vienna/Austria), full-time (40h/week) positions, so please no consultancy or free-lancing offers. http://www.sipwise.com/news/jobs/sipwise-is-hiring-perl-web-developers-arou… http://www.sipwise.com/news/jobs/sipwise-is-hiring-voip-engineers-around-vi… If you're interested, please drop us an email at office(a)sipwise.com. Thanks, Andreas

1 0

pua_xmpp presence problem with notify
by Martin Pohančeník 11 Apr '11

11 Apr '11

Hi ! I have managed to get the XMPP module working for IM. Now I am working on Presence service with the pua_xmpp module. I have experienced some problems and I would like to ask some questions to clarify my understandig of Kamailio. When a SUBSCRIBE arrives to Kamailio, it adds a subscribtion to the watchers table in the database. And if the subscriber is online, he is also in the active_watchers table, right? Next... when a PUBLISH request arrives to Kamailio, by default the handle_publish() function is called. This function generates all NOTIFY msgs (based on the database content of watchers and presence rules) and send them to the recipients. I wanted to ask if these generated NOTIFY msgs go through the routing logic, or not ? As I was saying, I was trying to get pua_xmpp module working, but Kamailio does not seem to send NOTIFY to users in XMPP (winfo is sent, bude presence itself is not). Here is a piece of my configuration file: if(is_method("NOTIFY")) { xlog(" ==== NOTIFY msg ====\n"); } if(is_method("NOTIFY") && uri=~"sip:.+[*].+@sip.sk") { xlog(" ====== NOTIFY from $fu to $tu ======\n"); if(pua_xmpp_notify()) { xlog(" ====== NOTIFY to xmpp domain from $fu to $tu ======\n"); t_reply("200", "OK"); } exit; } if(is_method("PUBLISH")) { xlog("===== PUBLISH FROM $fu ====="); handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { handle_subscribe(); if(uri=~"sip:.+[*].+@sip.sk" && $hdr(Event)== "presence") { xlog(" ====== SUBSCRIBE for XMPP domain from $fu to see $tu ======\n"); pua_xmpp_req_winfo("$ruri", "$hdr(Expires)"); } t_release(); } exit; Based on this configuration, I never get the " ==== NOTIFY msg ====" output to the syslog. So I assume the NOTIFY never goes through the routing logic. Is that right?? If so, how can I get it to pass the routing logic so the pua_xmpp_notify() function can be executed ?? Is there any way to do this?? Regards Martin

2 2

Re: [SR-Users] Kamailio and several Asterisk - Realtime DB Integration (solved)
by Alejandro Rios P. 11 Apr '11

11 Apr '11

Hello Daniel Thanks for pointing out to the SIP traces. I found out that it if the request is to be sent to a destination using a port other than 5060, it is required to add the $uac_req(ouri) parameter too: $uac_req(ouri)="sip:" + $var(asterisk_ip) + ":" + $var(asterisk_port); .... $uac_req(ouri)="sip:" + $var(asterisk2_ip) + ":" + $var(asterisk2_port); Thanks! Alejandro Rios Peña <alerios(a)debian.org> DISCLAIMER: The opinions expressed are my own, and not necessarily those of my employer. <http://www.avatar.com.co> 2011/4/8 Daniel-Constantin Mierla <miconda(a)gmail.com> > Hello, > > > On 4/8/11 12:28 AM, Alejandro Rios P. wrote: > >> >> Hi all, >> >> I am testing the scenario described here: >> http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb >> >> It works great with one instance of Asterisk, but now that I'm testing >> with two Asterisk instances to load balance, and I see a problem regarding >> the registration forwarding to the second Asterisk. >> >> I tried adding a second "uac_req_send()" bellow the first one, but >> Kamailio only sends the first request and the REGISTER never gets to the >> second Asterisk: >> >> # Forward REGISTER to Asterisk >> route[REGFWD] { >> if(!is_method("REGISTER")) >> { >> return; >> } >> $var(rip) = $sel(cfg_get.asterisk.bindip); >> $uac_req(method)="REGISTER"; >> $uac_req(ruri)="sip:" + $var(rip) + ":" + >> $sel(cfg_get.asterisk.bindport); >> $uac_req(furi)="sip:" + $au+ "@" + $var(rip); >> $uac_req(turi)="sip:" + $au+ "@" + $var(rip); >> $uac_req(hdrs)="Contact:<sip:" + $au+ "@" >> + $sel(cfg_get.kamailio.bindip) >> + ":" + $sel(cfg_get.kamailio.bindport) >> + ">\r\n"; >> if($sel(contact.expires) != $null) >> $uac_req(hdrs)= $uac_req(hdrs) + "Expires:" + >> $sel(contact.expires) + "\r\n"; >> else >> $uac_req(hdrs)= $uac_req(hdrs) + "Expires:" + >> $hdr(Expires) + "\r\n"; >> uac_req_send(); >> >> $uac_req(all) = null; >> >> $var(rip2) = $sel(cfg_get.asterisk2.bindip); >> $uac_req(method)="REGISTER"; >> $uac_req(ruri)="sip:" + $var(rip2) + ":" + >> $sel(cfg_get.asterisk2.bindport); >> $uac_req(furi)="sip:" + $au+ "@" + $var(rip2); >> $uac_req(turi)="sip:" + $au+ "@" + $var(rip2); >> $uac_req(hdrs)="Contact:<sip:" + $au+ "@" >> + $sel(cfg_get.kamailio.bindip) >> + ":" + $sel(cfg_get.kamailio.bindport) >> + ">\r\n"; >> if($sel(contact.expires) != $null) >> $uac_req(hdrs)= $uac_req(hdrs) + "Expires:" + >> $sel(contact.expires) + "\r\n"; >> else >> $uac_req(hdrs)= $uac_req(hdrs) + "Expires:" + >> $hdr(Expires) + "\r\n"; >> uac_req_send(); >> >> } >> >> I also tried appending another branch and sending the second >> uac_req_send() from there, but it is not being sent. >> >> How can I use uac_req_send() several times for the same request or in >> parallel? >> > it should work one after the other. append_branch() does not affect > uac_req_send() at all, this is a separate request built from scratch. > > Do you get any error message in the syslog? Can you watch the traffic on > all interfaces 'ngrep -d any -qt -W byline port 5060' ? > > I see you assign '$uac_req(all) = null;', it should be '$uac_req(all) = > $null;' - but I guess it was a copy&paste thing. > > Cheers, > Daniel > > -- > Daniel-Constantin Mierla > http://www.asipto.com > >

2 3

Possible bugs in nathelper
by Peter Dunkley 11 Apr '11

11 Apr '11

Hello, I think I have found two bugs in nathelper in Kamailio 3.1.3. Has anyone else encountered these? Bug one ------- nat_uac_test() doesn't seem to work for requests that arrive over UDP. I tried with the latest versions of X-Lite and Jitsi for Windows and observed the same behaviour with both. As far as I could tell the UDP and TCP REGISTER methods were identical except for transport, Call-ID, tags (and so on). At the moment I have worked around this by treating all UDP requests as if they have come from behind a NAT - but this is not ideal. Bug Two ------- nat_uac_test() with flag bit 8 (SDP checks) set fails with an error if the request does not contain SDP. I don't think this is correct behaviour as it seems quite sensible to me to push all requests through the same NAT detection route. In this case it is quite likely that only some of the methods will contain SDP. Surely in this case the correct thing would be for the SDP checks to fail, but not be treated as an error? Thanks for you help, Peter -- Peter Dunkley Technical Director Crocodile RCS Ltd

2 5

Setting Global Variables
by Nathaniel L Keeling 11 Apr '11

11 Apr '11

Hello, I am trying to use global variables within my configuration script. I am using Kamailio v3.0.0 and I am following the syntax example written in the sample config file but I keep getting syntax errors. Here is a copy of my syntax and I am using cfg_get.voicemailserver.bindip to get the value. #!ifdef WITH_VOICEMAIL voicemailserver.bindip = "10.0.0.1" desc "Voicemail Server IP Address" voicemailserver.bindport = "5060" desc "Voicemail Server Port" #!endif Here is the error that I am receiving: 0(1875) : <core> [cfg.y:3333]: parse error in config file /usr/local/kamailio-3.0.0/etc/kamailio/kamailio.cfg, line 156, column 17: syntax error 0(1875) : <core> [cfg.y:3333]: parse error in config file /usr/local/kamailio-3.0.0/etc/kamailio/kamailio.cfg, line 156, column 17: unknown config variable 0(1875) : <core> [cfg.y:3330]: parse error in config file /usr/local/kamailio-3.0.0/etc/kamailio/kamailio.cfg, line 156, column 19-28: ERROR: bad config file (3 errors) 0(1875) DEBUG: <core> [mem/shm_mem.c:236]: shm_mem_destroy Thanks Nathaniel

2 1

Problems with dispatcher policy in HA/LB kamailio/asterisk environment
by Sören Berger 11 Apr '11

11 Apr '11

Hi all, I currently setup a HA/LB conference. I currently have 1 SER and 2 asterisk conference servers. The goal is to distribute the conferences based on the room number on different server. I use ubuntu 10.04 LTS and the packages from http://deb.kamailio.org/kamailio (kamailio_3.1.2+lucid1_amd64.deb) During the implementation I run into trouble with the "8 - first destination" policy which choose not my second server in case of a failure. The dial in is eg. 330 with round robin to any of my asterisk servers. There they check the room number and append it to the dial number. When somebody calls 330 and enter the room number 12345 the asterisk route it back to kamilio with 33012345 route[ASTERISK_CONFERENCE_AUTH] { if ($rU=~"(33[0-4]$)") { if ( method=="INVITE" ) { ds_select_dst("1","4"); forward(); exit(); } } } When the number 33012345 arrives kamailio decides on which server the conference takes place. For redundancy reasons I use the dispatcher again. If one of the server fails, another will take over the conferences. For the scenario with 2 servers this is trivial. There I also could do it another way. But when I add more conference servers it looks like the best solution to me. Here are my rules for the servers. In this case I used the policy "8" which should use the first destination according to the dispatcher.list file. So if I add more servers I can decide about a order for the fail over servers. route [ASTERISK_CONFERENCE] { if ($rU=~"(33[0-4][0-4][0-9][0-9][0-9][0-9]$)") { if ( method=="INVITE" ) { ds_select_dst("2","8"); forward(); exit(); } } if ($rU=~"(33[0-4][5-9][0-9][0-9][0-9][0-9]$)") { if ( method=="INVITE" ) { ds_select_dst("3","8"); forward(); exit(); } } } distpatcher.list # This group will be used with round robin for getting the room number 1 sip:172.16.1.88:5060 0 1 1 sip:172.16.1.89:5060 0 2 # failover schedule for ast-conf1 2 sip:172.16.1.88:5060 0 1 2 sip:172.16.1.89:5060 0 2 # failover schedule for ast-conf2 3 sip:172.16.1.89:5060 0 1 3 sip:172.16.1.88:5060 0 2 My current problem is, that the policy "8" don't select the other server in case of an failure. I simply blocks and the clients retransmit the request to kamailio.I am currently going crazy with that. :) When I change the policy to round robin (replacing "8" with "4"), everything is working fine. But this an option because in the normal case all calls to a conference room should be placed on the same server. Could this be a bug or something like that? Maybe not the right list here, but I didn't find anything about meetme implementation with more that one server or a HA/LB solution for that and was really wondering about that. Somebody maybe heart from such a implementation? Kind Regards, Sören Berger -- Dipl.-Inf. Sören Berger Rechenzentrum Universität Stuttgart Allmandring 30 70550 Stuttgart soeren.berger(a)rus.uni-stuttgart.de +49-(0)711-685-65871

2 3

New developer: Timo Teräs
by Daniel-Constantin Mierla 11 Apr '11

11 Apr '11

Hello, just to let everyone know that I grnted developer access to Timo, who submitted recently a patch for adding a new module named db_sqlite. As the name suggests, the new module will allot to use SQLite databases as alternative to existing db drivers such as mysql, postgres, a.s.o. Timo will maintain the new module and may help with other modules, as he previously submitted a patch for uac module. His GIT username for commits is: tteras. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

3 2

is_from_user_enum and SRV records
by Eric Hiller 11 Apr '11

11 Apr '11

We have enum records where 0123456789 points to 0123456789(a)domain.tld. And then multiple SRV records pointing to the actual hosts; _sip._udp SRV record for domain.tld pointing to host1.domain.tld and host2.domain.tld for load balancing. In our config we run is_from_user_enum, which up until adding a second entry for the SRV records worked fine. Now however the is_from_user_enum lookup invariably returns the opposite DNS record for the reply. Ie. host1 is requesting and the dns cache returns host2 and then the is_from_user_enum fails and the reply is dropped. Is there a way to make is_from_user_enum test ALL values of the SRV record? Thanks! -Eric

2 1

Kamailio multihoming
by roman＠dmytriv.com 11 Apr '11

11 Apr '11

Hello everybody!!! J Did any of you know or have configured Kamailio servers in redundant and load distribution mode on geographically dispersed servers. Can you share some ideas how can this be done. I'm thinking about 2 -3 servers that can be used to distribute load and for redundancy as well. Is Kamailio capable of this or this needs to be done on OS level. Thank you Roman

2 3

Re: [SR-Users] Kamailio XCAP for Kamailio IMS
by Daniel-Constantin Mierla 08 Apr '11

08 Apr '11

Hello, can you run kamailio with global parameter 'debug=3' and send the syslog messages for such case? That will give the hints about what is going wrong there. I see some characters are hexa encoded, but better see first the logs. Cheers, Daniel On 4/8/11 11:14 AM, Pavel Segeč wrote: > > Hi, > > I'm using Kamailio SIMPLE AS server as the presence and XCAP solution > as the extension for my Kamailio based IMS platfom. As the base > Presence configuration i'm using the default config as is described > inside of the tutorial from ASIPTO site, it works well for pure ietf > SIP SIMPLE environment. But when using it with the IMS, I sniff during > putting the new record 500 Error. May somebody show me a direction > what should be wrong? > > thanks > > palo > > PUT > /xcap-root/resource-lists/users/palo/index/~~/resource-lists/list%5B@name=%22doubango%22%5D/entry%5B@uri=%22sip:segi@ims1.sip.uniza.sk%22%5D > HTTP/1.1 > > Host: 158.193.139.36:5060 > > Content-Length: 151 > > Connection: Keep-Alive > > User-Agent: XDM-client/OMA1.1 > > X-3GPP-Intended-Identity: palo > > Content-Type: application/xcap-el+xml > > <entry uri="sip:segi@ims1.sip.uniza.sk" > xmlns="urn:ietf:params:xml:ns:resource-lists"> > > <display-name>segi(a)ims1.sip.uniza.sk</display-name> > > </entry>HTTP/1.1 500 Server error > > Via: SIP/2.0/TCP 158.193.152.64:20735 > > Server: kamailio (3.1.2 (i386/linux)) > > Content-Length: 0 > > > > __________ Informacia od ESET NOD32 Antivirus, verzia databazy 6025 > (20110408) __________ > > Tuto spravu preveril ESET NOD32 Antivirus. > > http://www.eset.sk > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla http://www.asipto.com

2 2

ring_insert_callid() fixup error
by Juha Heinanen 08 Apr '11

08 Apr '11

i have modparam("siputils", "ring_timeout", 30) but i get Apr 8 15:32:02 sip /usr/sbin/sip-proxy[23748]: ERROR: siputils [ring.c:378]: ring_insert_callid functionality deactivated, you need to set a positive ring_timeout Apr 8 15:32:02 sip /usr/sbin/sip-proxy[23748]: ERROR: <core> [route.c:1216]: fixing failed (code=-1) at cfg:/etc/sip-proxy/sip-proxy.cfg:658 Apr 8 15:32:02 sip /usr/sbin/sip-proxy[23748]: ERROR: <core> [route.c:1216]: fixing failed (code=-1) at cfg:/etc/sip-proxy/sip-proxy.cfg:691 when i start sip router (master version). i looked the code briefly and it tests module param value in fixup function, which i have not seen before. is this a bug or am i missing something? -- juha

2 3

Kamailio and several Asterisk - Realtime DB Integration
by Alejandro Rios P. 08 Apr '11

08 Apr '11

Hi all, I am testing the scenario described here: http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb It works great with one instance of Asterisk, but now that I'm testing with two Asterisk instances to load balance, and I see a problem regarding the registration forwarding to the second Asterisk. I tried adding a second "uac_req_send()" bellow the first one, but Kamailio only sends the first request and the REGISTER never gets to the second Asterisk: # Forward REGISTER to Asterisk route[REGFWD] { if(!is_method("REGISTER")) { return; } $var(rip) = $sel(cfg_get.asterisk.bindip); $uac_req(method)="REGISTER"; $uac_req(ruri)="sip:" + $var(rip) + ":" + $sel(cfg_get.asterisk.bindport); $uac_req(furi)="sip:" + $au + "@" + $var(rip); $uac_req(turi)="sip:" + $au + "@" + $var(rip); $uac_req(hdrs)="Contact: <sip:" + $au + "@" + $sel(cfg_get.kamailio.bindip) + ":" + $sel(cfg_get.kamailio.bindport) + ">\r\n"; if($sel(contact.expires) != $null) $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $sel(contact.expires) + "\r\n"; else $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $hdr(Expires) + "\r\n"; uac_req_send(); $uac_req(all) = null; $var(rip2) = $sel(cfg_get.asterisk2.bindip); $uac_req(method)="REGISTER"; $uac_req(ruri)="sip:" + $var(rip2) + ":" + $sel(cfg_get.asterisk2.bindport); $uac_req(furi)="sip:" + $au + "@" + $var(rip2); $uac_req(turi)="sip:" + $au + "@" + $var(rip2); $uac_req(hdrs)="Contact: <sip:" + $au + "@" + $sel(cfg_get.kamailio.bindip) + ":" + $sel(cfg_get.kamailio.bindport) + ">\r\n"; if($sel(contact.expires) != $null) $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $sel(contact.expires) + "\r\n"; else $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $hdr(Expires) + "\r\n"; uac_req_send(); } I also tried appending another branch and sending the second uac_req_send() from there, but it is not being sent. How can I use uac_req_send() several times for the same request or in parallel? Thank you, Alejandro Rios Peña DISCLAIMER: The opinions expressed are my own, and not necessarily those of my employer. <http://www.avatar.com.co>

2 1

SIP-XMPP problem, please help
by Martin Pohančeník 08 Apr '11

08 Apr '11

Hello everyone. I am a student at University of Žilina (Faculty of Management Science and Informatics), which has been recently awarded Kamailio 2010 Awards for various articles about SIP and KAMAILIO on portal nil.uniza.sk. I am doing a bachelor thesis on topic SIP/SIMPLE - XMPP traversal, but I am experiencing some problems. I have a working Kamailio server serving domain sip.uniza.sk and a working Jabberd2 server serving domain xmpp.uniza.sk. I am running Kamailio 3.1 with XMPP module with following configuration: fork=yes children=4 log_stderror=yes alias="sip.uniza.sk" ... loadmodule "XMPP.so" ... modparam("xmpp", "backend", "component") modparam("xmpp", "domain_separator", "*") modparam("xmpp", "gateway_domain", "sip.uniza.sk") modparam("xmpp", "xmpp_domain", "xmpp-sip.uniza.sk") modparam("xmpp", "xmpp_host", "server.xmpp.uniza.sk") modparam("xmpp", "xmpp_password", "secret") ... #main routing logic route { ... if( uri=~"sip:.+[*].+@sip.uniza.sk") { # absorb retransmissions ### if (!t_newtran()) { sl_reply_error(); return; } if (method == "MESSAGE") { log("*** xmpp-handled MESSAGE message.\n"); if (xmpp_send_message()) { t_reply("200", "Accepted"); } else { t_reply("404", "Not found"); } return; } else { t_reply("403","not supported"); exit; } } ... } To my knowledge there is no configuration required on the Jabberd2 server (except the password, that the XMPP component authenticates with). I am trying to enable SIP clients to chat with the XMPP clients and vice versa. SIP clients can send messages to XMPP clients. XMPP modules translates SIP to XMPP and forwards the massage to Jabberd2 server. URIs look like: XMPP client from SIP view: xmpp_client*xmpp.uniza.sk(a)sip.uniza.sk SIP client from XMPP view: sip_client*sip.uniza.sk(a)xmpp-sip.uniza.sk The message arrives to XMPP client, but he can not reply. I have sniffed the traffic on the Kamailio server. The XMPP message arrives to kamailio XMPP module (correct port and everything), but the module somehow ignores it, doesn't translate XMPP to SIP and doesn't send it to the SIP recipient. The sniffed message looks like: <message xmlns='jabber:client' type='chat' to='SIP_client*sip.uniza.sk@ xmpp-sip.uniza.sk' id='mir_47' from='XMPP_client(a)xmpp.uniza.sk/Miranda '><body>hello</body></message> Any suggestions on where can the problem be ?? Do I have to add some lines to routing logic for XMPP module to process the XMPP message (from Jabberd2) for SIP_client ?? Thanks Martin

3 5

TLS certificate on sip-router.org expired
by Timo Reimann 08 Apr '11

08 Apr '11

Heya, just noticed that the TLS certificate on sip-router.org has expired since 9.8.2010. Try an HTTPS connection to the web-page, i.e., https://sip-router.org to see. It seems to be self-signed though. Anyways, I just wanted to let you know. Cheers, --Timo

1 0

Kamailio XCAP for Kamailio IMS
by Pavel Segeč 08 Apr '11

08 Apr '11

Hi, I'm using Kamailio SIMPLE AS server as the presence and XCAP solution as the extension for my Kamailio based IMS platfom. As the base Presence configuration i'm using the default config as is described inside of the tutorial from ASIPTO site, it works well for pure ietf SIP SIMPLE environment. But when using it with the IMS, I sniff during putting the new record 500 Error. May somebody show me a direction what should be wrong? thanks palo PUT /xcap-root/resource-lists/users/palo/index/~~/resource-lists/list%5B@name=%2 2doubango%22%5D/entry%5B@uri=%22sip:segi@ims1.sip.uniza.sk%22%5D HTTP/1.1 Host: 158.193.139.36:5060 Content-Length: 151 Connection: Keep-Alive User-Agent: XDM-client/OMA1.1 X-3GPP-Intended-Identity: palo Content-Type: application/xcap-el+xml <entry uri="sip:segi@ims1.sip.uniza.sk" xmlns="urn:ietf:params:xml:ns:resource-lists"> <display-name>segi(a)ims1.sip.uniza.sk</display-name> </entry>HTTP/1.1 500 Server error Via: SIP/2.0/TCP 158.193.152.64:20735 Server: kamailio (3.1.2 (i386/linux)) Content-Length: 0

1 0

Problems with event_route[htable:mod-init]
by Camila Troncoso 07 Apr '11

07 Apr '11

Hello, I´m using Kamalio 3.1.0 and want to alter some databases (mysql) tables at kamailio starts. I read that the route event_route[htable:mod-init] allows me to do that, but i´m having some problems. I load htable.so module and insert the event_route[] in mi kamailio.conf script but when I start kamailio it returns an error: Apr 7 15:30:22 server4967 /usr/local/sbin/kamailio[31610]: DEBUG: <core> [sr_module.c:828]: DEBUG: init_mod_child (-127): usrloc Apr 7 15:30:22 server4967 /usr/local/sbin/kamailio[31610]: DEBUG: <core> [sr_module.c:828]: DEBUG: init_mod_child (-127): registrar Apr 7 15:30:22 server4967 /usr/local/sbin/kamailio[31610]: DEBUG: <core> [sr_module.c:828]: DEBUG: init_mod_child (-127): uri_db Apr 7 15:30:22 server4967 /usr/local/sbin/kamailio[31610]: DEBUG: <core> [sr_module.c:828]: DEBUG: init_mod_child (-127): domain Apr 7 15:30:22 server4967 /usr/local/sbin/kamailio[31610]: DEBUG: <core> [sr_module.c:828]: DEBUG: init_mod_child (-127): permissions Apr 7 15:30:22 server4967 /usr/local/sbin/kamailio[31610]: DEBUG: <core> [sr_module.c:828]: DEBUG: init_mod_child (-127): group Apr 7 15:30:22 server4967 /usr/local/sbin/kamailio[31610]: DEBUG: <core> [sr_module.c:828]: DEBUG: init_mod_child (-127): mi_fifo Apr 7 15:30:22 server4967 /usr/local/sbin/kamailio[31610]: DEBUG: <core> [sr_module.c:828]: DEBUG: init_mod_child (-127): avpops Apr 7 15:30:22 server4967 /usr/local/sbin/kamailio[31610]: DEBUG: <core> [sr_module.c:828]: DEBUG: init_mod_child (-127): dialog Apr 7 15:30:22 server4967 /usr/local/sbin/kamailio[31610]: DEBUG: <core> [sr_module.c:828]: DEBUG: init_mod_child (-127): lcr Apr 7 15:30:22 server4967 /usr/local/sbin/kamailio[31610]: DEBUG: <core> [sr_module.c:828]: DEBUG: init_mod_child (-127): sqlops Apr 7 15:30:22 server4967 /usr/local/sbin/kamailio[31610]: DEBUG: <core> [sr_module.c:828]: DEBUG: init_mod_child (-127): rtimer Apr 7 15:30:22 server4967 /usr/local/sbin/kamailio[31610]: DEBUG: <core> [sr_module.c:828]: DEBUG: init_mod_child (-127): htable Apr 7 15:30:22 server4967 /usr/local/sbin/kamailio[31610]: DEBUG: htable [htable.c:185]: rank is (-127) Apr 7 15:30:22 server4967 /usr/local/sbin/kamailio[31610]: DEBUG: htable [htable.c:191]: executing event_route[htable:mod-init] (1) Apr 7 15:30:22 server4967 /usr/local/sbin/kamailio[31610]: DEBUG: <core> [parser/msg_parser.c:630]: SIP Request: Apr 7 15:30:22 server4967 /usr/local/sbin/kamailio[31610]: DEBUG: <core> [parser/msg_parser.c:632]: method: <OPTIONS> Apr 7 15:30:22 server4967 /usr/local/sbin/kamailio[31610]: DEBUG: <core> [parser/msg_parser.c:634]: uri: <sip:you@kamailio.org> Apr 7 15:30:22 server4967 /usr/local/sbin/kamailio[31610]: DEBUG: <core> [parser/msg_parser.c:636]: version: <SIP/2.0> Apr 7 15:30:22 server4967 /usr/local/sbin/kamailio[31610]: <core> [mem/q_malloc.c:366]: qm_malloc(0x82bbfa0, 32) called from <core>: parser/msg_parser.c: parse_headers(341) Apr 7 15:30:22 server4967 /usr/local/sbin/kamailio[31610]: <core> [mem/q_malloc.c:406]: qm_malloc(0x82bbfa0, 32) returns address 0x834f780 frag. 0x834f768 (size=32) on 1 -th hit Apr 7 15:30:22 server4967 /usr/local/sbin/kamailio[31610]: <core> [mem/q_malloc.c:366]: qm_malloc(0x82bbfa0, 116) called from <core>: parser/msg_parser.c: get_hdr_field(130) Apr 7 15:30:22 server4967 /usr/local/sbin/kamailio[31610]: <core> [mem/q_malloc.c:406]: qm_malloc(0x82bbfa0, 116) returns address 0x834f7d0 frag. 0x834f7b8 (size=116) on 1 -th hit Apr 7 15:30:22 server4967 /usr/local/sbin/kamailio[31610]: DEBUG: <core> [parser/parse_via.c:2300]: end of header reached, state=2 Apr 7 15:30:22 server4967 /usr/local/sbin/kamailio[31610]: DEBUG: <core> [parser/msg_parser.c:515]: parse_headers: Via found, flags=2 Apr 7 15:30:22 server4967 /usr/local/sbin/kamailio[31610]: DEBUG: <core> [parser/msg_parser.c:517]: parse_headers: this is the first via Apr 7 15:30:22 server4967 kamailio: ERROR: <core> [daemonize.c:307]: Main process exited before writing to pipe * * * * My event_rout[] is as fallows: event_route[htable:mod-init]{ sql_query("cd", "select prefix from lcr_rule", "rc"); xlog("L_INFO", "dbr = $dbr(rc=>rows)"); if($dbr(rc=>rows)>0){ $var(i) = 0; while($var(i)<$dbr(rc=>rows)){ $var(prefijo)=$dbr(rc=>[$var(i),0]); xlog("L_INFO", "$dbr(rc=>[$var(i),0])"); sql_query("cd", "ALTER TABLE rebalse ADD COLUMN Prefijo_$var(prefijo) INT","rr"); $var(i) = $var(i) + 1; } } sql_result_free("rc"); } I tested the commands inside the event_route[] and they work fine… Can someone help me? Regards, *Camila Troncoso **|* Ingeniero de Desarrollo RedVoiss *|*ctroncoso(a)redvoiss.net Santiago - Chile *|* +56 2 2408535 www.redvoiss.net

2 1

It's ClueCon Time!
by marketing＠cluecon.com 07 Apr '11

07 Apr '11

1 0

"kamctl fifo debug" does not work
by Klaus Darilion 07 Apr '11

07 Apr '11

root@pbx-kam-1:/home/klaus3000# kamctl fifo debug DEBUG:: 2 root@pbx-kam-1:/home/klaus3000# kamctl fifo debug 0 DEBUG:: 0 Mar 9 11:04:23 pbx-kam-1 /usr/sbin/kamailio[4548]: INFO: <core> [cfg/cfg_ctx.c:411]: INFO: cfg_set_now(): core.debug has been changed to 0 but: root@pbx-kam-1:/home/klaus3000# kamctl fifo debug DEBUG:: 2 Is this a bug? regards Klaus

2 2

Kamailio at Google Summer of Code 2011
by Daniel-Constantin Mierla 07 Apr '11

07 Apr '11

Hello, for the second year, Kamailio is participating in GSoC program. For 2011, the proposal is to implement a signaling gateway between SIP and JINGLE to enable voice calls between the two networks: http://www.jitsi.org/index.php/GSOC2011/KamailioJingle If you are a student, or you know a student, interested in this project, just write me or to sr-dev(a)lists.sip-router.org. The application starts today and the deadline is April 8, therefore it is quite a short period, you will have to hurry. You can read more about the GSoC program and Kamailio at: http://www.kamailio.org/w/gsoc-2011/ http://www.jitsi.org/index.php/Development/Gsoc2011 Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

7 15

Cannot run Kamailio on public IP
by spamasbl＠gmail.com 07 Apr '11

07 Apr '11

Hello all, I have installed latest Kamailio 3.1.3 on latest ubuntu server. Everything seems to be working well on my LAN. I can successfully register with local Kamailio IP address (192.168.1.137). The issue is that I can not run Kamailio on my public IP: listen=udp:77.XX.XX.XX:5060 listen=udp:192.168.1.137:5060 port=5060 Kamailio runs behind Linksys WRT54GL router with latest Linksys firmware and DMZ is turned on (192.168.1.137). Here is the error I am getting when trying to run kamailio with my public IP: Apr 7 02:57:40 sipbox kamailio[1050]: ERROR: <core> [udp_server.c:400]: ERROR: udp_init: bind(7, 0x82cc6e4, 16) on 77.XX.XX.XX: Cannot assign requested address Do you have any idea what I'm doing wrong ? Thanks !

6 10

off-topic: jobs & projects - figuring out the best solution
by Daniel-Constantin Mierla 06 Apr '11

06 Apr '11

Hello, recently the demand in jobs and projects related to kamailio seemed to increase. I can just say that I am very happy with that. While announcing on the mailing lists is ok, I think we should try to make the process better and more productive, i.e., easier to advertise by publishers and easier to read for those interesting. For that, imo, we need a web page where someone can post such announcements and the others can read it. Since I had this idea in mind for some while, I searched for a dokuwiki plugin that could help, so we can reuse our wiki system. I found bureaucracy plugin, but does not seem to be ready for latest dokuwiki version according to its page: http://www.dokuwiki.org/plugin:bureaucracy But maybe it just works, I should try it instead of waiting, anyhow spare time is not my friend these days... Alternatives would be: - a wordpress plugin - haven't searched for one yet, but there are tons of plugins for WP, maybe one will fit the needs - another web application designed for such purposes My question to you is whether you can recommend a solution which we can evaluate. Ideally (or better said, as many characteristics to be met as possible), the system should be able to: - allow people to post announcements for jobs and projects (with or without prior registration, but with spam prevention - e.g., captcha) - allow to publish details about the job/projects, contact addresses, validity period - allow to invalidate an announcement when the position/project is taken - send an email notification when the announcement is done first time, with a link to the web page of the announcement, so we can send it one to a particular mailing list Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

4 3

job offer: System administrator internet access services
by Henning Westerholt 06 Apr '11

06 Apr '11

Job offer: Interested in working with one of the biggest Voice over IP networks in europe? We're hiring and have one additional IT operation position open in our team in Karlsruhe, Germany at the moment. You would be responsible in a team of motivated system administrators for the operation and maintenance of the radius and CPE auto-provisioning services at 1&1. Furthermore you would support our on-call service for all the services operated from the team, including VoIP. A solid understanding of german is necessary for the job, for more informations (additional requirements, how to apply..) please refer to the opening on the job web page. Of course please feel free to contact me per mail as well. http://www.united-internet.de/JobsJobDetail?__sendingdata=1&JobDetails.jobI… Best regards, Henning Westerholt -- Henning Westerholt - Head of IT Operations Internet Access & Communications 1&1 Internet AG, Brauerstraße 48, 76135 Karlsruhe, Germany

1 0

Re: [SR-Users] Kamailio ims / rtpproxy
by Carsten Bock 06 Apr '11

06 Apr '11

Hi, the Kamailio IMS modules are based on the OpenIMSCore-CSCF modules, which have built-in RTPProxy support. I am working on replacing the built-in RTPProxy support in the CSCF modules with the "original" rtpproxy-module of the Kamailio/sip-router.org. It does depend on your config, which RTPProxy support you use.... (but the focus is currently on other parts, be patient). Carsten 2011/4/6 Pavel Segeč <Pavel.Segec(a)fri.uniza.sk>: > Hi, > > I do not understand, may you clarify it? I'm using Kamailio based PCSCF not OIMSC, and kamailio has rtpproxy module...should I'm able to build PCSCF using sip router? > > palo > >> -----Original Message----- >> From: Carsten Bock [mailto:carsten@ng-voice.com] >> Sent: Wednesday, April 06, 2011 11:30 AM >> To: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users >> Mailing List >> Cc: Pavel Segeč >> Subject: Re: [SR-Users] Kamailio ims / rtpproxy >> >> Hi, >> >> one reason more to replace the built-in RTP-Proxy support of the CSCF >> with the "real" rtpproxy moduie of sip-router.org! ;-) >> >> Carsten >> >> 2011/4/6 Pavel Segeč <Pavel.Segec(a)fri.uniza.sk>: >> > Hi Carsten, >> > >> > Thank you for answer. I did it as you described (anyway, i think, that >> nat traversal is by default on), but there were still some problems. >> After some deeper testing, the PCSCF with RTPproxy works well with two >> Monster client (caller and calle). The problem is with Boghe IMS >> clients, I do not know why, private address inside of the SDP is not >> rewritten, what is interesting, media ports are changing. >> > >> > regards >> > >> > palo >> > >> >> -----Original Message----- >> >> From: sr-users-bounces(a)lists.sip-router.org [mailto:sr-users- >> >> bounces(a)lists.sip-router.org] On Behalf Of Carsten Bock >> >> Sent: Wednesday, April 06, 2011 10:32 AM >> >> To: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - >> Users >> >> Mailing List >> >> Cc: Pavel Segeč >> >> Subject: Re: [SR-Users] Kamailio ims / rtpproxy >> >> >> >> Hi, >> >> >> >> there is no documentation on the RTPProxy interface of the Kamailio >> >> IMS module yet. >> >> You need to have the following linse in the Kamailio.cfg: >> >> >> >> # Enabled Features on this host: >> >> #!define WITH_NAT >> >> [...] >> >> #!ifdef WITH_NAT >> >> #!define RTPPROXY "udp:127.0.0.1:22222" >> >> #!endif >> >> [...] >> >> #!ifndef WITH_NAT >> >> [...] >> >> #!else >> >> modparam("pcscf","NAT_enable", 1) >> >> modparam("pcscf","ping", 1) >> >> modparam("pcscf","ping_all", 0) >> >> modparam("pcscf","nat_detection_type", 0x17) >> >> modparam("pcscf","rtpproxy_socket", RTPPROXY) >> >> modparam("pcscf","rtpproxy_enable", 1) >> >> modparam("pcscf","rtpproxy_disable_tout", 60) >> >> modparam("pcscf","rtpproxy_retr", 5) >> >> modparam("pcscf","rtpproxy_tout", 1) >> >> #!endif >> >> >> >> Kind regards, >> >> Carsten >> >> >> >> 2011/4/5 Pavel Segeč <pavel.segec(a)fri.uniza.sk>: >> >> > Hi, >> >> > >> >> > >> >> > >> >> > I would like ask you if there is a kamailio ims module >> documentation >> >> > somewhere. I need some deeper information how some things are >> >> implemented, I >> >> > have problem to setup NAT traversal with RTPproxy for our Kamailio >> IMS >> >> > platform. There is no communication among the PCSCF and RTPproxy >> >> server over >> >> > control udp sock. We tried unix socket also with the same results. >> >> > >> >> > >> >> > >> >> > Thanks >> >> > >> >> > >> >> > >> >> > palo73 >> >> > >> >> > http://nil.uniza.sk >> >> > >> >> > __________ Informacia od ESET NOD32 Antivirus, verzia databazy 6016 >> >> > (20110405) __________ >> >> > >> >> > Tuto spravu preveril ESET NOD32 Antivirus. >> >> > >> >> > http://www.eset.sk >> >> > >> >> > _______________________________________________ >> >> > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing >> >> list >> >> > sr-users(a)lists.sip-router.org >> >> > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> >> > >> >> > >> >> >> >> >> >> >> >> -- >> >> Carsten Bock >> >> http://www.ng-voice.com >> >> mailto:carsten@ng-voice.com >> >> >> >> _______________________________________________ >> >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing >> list >> >> sr-users(a)lists.sip-router.org >> >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> >> >> >> >> >> __________ Informacia od ESET NOD32 Antivirus, verzia databazy 6017 >> >> (20110405) __________ >> >> >> >> Tuto spravu preveril ESET NOD32 Antivirus. >> >> >> >> http://www.eset.sk >> >> >> > >> > >> > __________ Informacia od ESET NOD32 Antivirus, verzia databazy 6017 >> (20110405) __________ >> > >> > Tuto spravu preveril ESET NOD32 Antivirus. >> > >> > http://www.eset.sk >> > >> > >> > >> > _______________________________________________ >> > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing >> list >> > sr-users(a)lists.sip-router.org >> > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> > >> >> >> >> -- >> Carsten Bock >> http://www.ng-voice.com >> mailto:carsten@ng-voice.com >> >> Schomburgstr. 80 >> 22767 Hamburg >> Germany >> >> Mobile +49 179 2021244 >> Office +49 40 34927219 >> Fax +49 40 34927220 >> >> >> __________ Informacia od ESET NOD32 Antivirus, verzia databazy 6017 >> (20110405) __________ >> >> Tuto spravu preveril ESET NOD32 Antivirus. >> >> http://www.eset.sk >> > > > __________ Informacia od ESET NOD32 Antivirus, verzia databazy 6017 (20110405) __________ > > Tuto spravu preveril ESET NOD32 Antivirus. > > http://www.eset.sk > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > -- Carsten Bock http://www.ng-voice.com mailto:carsten@ng-voice.com

3 5

Re: [SR-Users] Kamailio ims / rtpproxy
by Alex Balashov 06 Apr '11

06 Apr '11

On 04/06/2011 05:48 AM, Pavel Segeč wrote: > I do not understand, may you clarify it? I'm using Kamailio based > PCSCF not OIMSC, and kamailio has rtpproxy module...should I'm able > to build PCSCF using sip router? Don't worry about this. This is a reference to an ongoing bit of development esoterica related to the fact that Kamailio already has--and has had for a very long time--a module to control rtpproxy. It is not part of the IMS module framework Carsten is developing. -- 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/

1 0

Re: [SR-Users] Kamailio ims / rtpproxy
by Carsten Bock 06 Apr '11

06 Apr '11

Hi, one reason more to replace the built-in RTP-Proxy support of the CSCF with the "real" rtpproxy moduie of sip-router.org! ;-) Carsten 2011/4/6 Pavel Segeč <Pavel.Segec(a)fri.uniza.sk>: > Hi Carsten, > > Thank you for answer. I did it as you described (anyway, i think, that nat traversal is by default on), but there were still some problems. After some deeper testing, the PCSCF with RTPproxy works well with two Monster client (caller and calle). The problem is with Boghe IMS clients, I do not know why, private address inside of the SDP is not rewritten, what is interesting, media ports are changing. > > regards > > palo > >> -----Original Message----- >> From: sr-users-bounces(a)lists.sip-router.org [mailto:sr-users- >> bounces(a)lists.sip-router.org] On Behalf Of Carsten Bock >> Sent: Wednesday, April 06, 2011 10:32 AM >> To: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users >> Mailing List >> Cc: Pavel Segeč >> Subject: Re: [SR-Users] Kamailio ims / rtpproxy >> >> Hi, >> >> there is no documentation on the RTPProxy interface of the Kamailio >> IMS module yet. >> You need to have the following linse in the Kamailio.cfg: >> >> # Enabled Features on this host: >> #!define WITH_NAT >> [...] >> #!ifdef WITH_NAT >> #!define RTPPROXY "udp:127.0.0.1:22222" >> #!endif >> [...] >> #!ifndef WITH_NAT >> [...] >> #!else >> modparam("pcscf","NAT_enable", 1) >> modparam("pcscf","ping", 1) >> modparam("pcscf","ping_all", 0) >> modparam("pcscf","nat_detection_type", 0x17) >> modparam("pcscf","rtpproxy_socket", RTPPROXY) >> modparam("pcscf","rtpproxy_enable", 1) >> modparam("pcscf","rtpproxy_disable_tout", 60) >> modparam("pcscf","rtpproxy_retr", 5) >> modparam("pcscf","rtpproxy_tout", 1) >> #!endif >> >> Kind regards, >> Carsten >> >> 2011/4/5 Pavel Segeč <pavel.segec(a)fri.uniza.sk>: >> > Hi, >> > >> > >> > >> > I would like ask you if there is a kamailio ims module documentation >> > somewhere. I need some deeper information how some things are >> implemented, I >> > have problem to setup NAT traversal with RTPproxy for our Kamailio IMS >> > platform. There is no communication among the PCSCF and RTPproxy >> server over >> > control udp sock. We tried unix socket also with the same results. >> > >> > >> > >> > Thanks >> > >> > >> > >> > palo73 >> > >> > http://nil.uniza.sk >> > >> > __________ Informacia od ESET NOD32 Antivirus, verzia databazy 6016 >> > (20110405) __________ >> > >> > Tuto spravu preveril ESET NOD32 Antivirus. >> > >> > http://www.eset.sk >> > >> > _______________________________________________ >> > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing >> list >> > sr-users(a)lists.sip-router.org >> > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> > >> > >> >> >> >> -- >> Carsten Bock >> http://www.ng-voice.com >> mailto:carsten@ng-voice.com >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >> sr-users(a)lists.sip-router.org >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> >> >> __________ Informacia od ESET NOD32 Antivirus, verzia databazy 6017 >> (20110405) __________ >> >> Tuto spravu preveril ESET NOD32 Antivirus. >> >> http://www.eset.sk >> > > > __________ Informacia od ESET NOD32 Antivirus, verzia databazy 6017 (20110405) __________ > > Tuto spravu preveril ESET NOD32 Antivirus. > > http://www.eset.sk > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > -- Carsten Bock http://www.ng-voice.com mailto:carsten@ng-voice.com Schomburgstr. 80 22767 Hamburg Germany Mobile +49 179 2021244 Office +49 40 34927219 Fax +49 40 34927220

2 1

Re: [SR-Users] Kamailio ims / rtpproxy
by Carsten Bock 06 Apr '11

06 Apr '11

Hi, there is no documentation on the RTPProxy interface of the Kamailio IMS module yet. You need to have the following linse in the Kamailio.cfg: # Enabled Features on this host: #!define WITH_NAT [...] #!ifdef WITH_NAT #!define RTPPROXY "udp:127.0.0.1:22222" #!endif [...] #!ifndef WITH_NAT [...] #!else modparam("pcscf","NAT_enable", 1) modparam("pcscf","ping", 1) modparam("pcscf","ping_all", 0) modparam("pcscf","nat_detection_type", 0x17) modparam("pcscf","rtpproxy_socket", RTPPROXY) modparam("pcscf","rtpproxy_enable", 1) modparam("pcscf","rtpproxy_disable_tout", 60) modparam("pcscf","rtpproxy_retr", 5) modparam("pcscf","rtpproxy_tout", 1) #!endif Kind regards, Carsten 2011/4/5 Pavel Segeč <pavel.segec(a)fri.uniza.sk>: > Hi, > > > > I would like ask you if there is a kamailio ims module documentation > somewhere. I need some deeper information how some things are implemented, I > have problem to setup NAT traversal with RTPproxy for our Kamailio IMS > platform. There is no communication among the PCSCF and RTPproxy server over > control udp sock. We tried unix socket also with the same results. > > > > Thanks > > > > palo73 > > http://nil.uniza.sk > > __________ Informacia od ESET NOD32 Antivirus, verzia databazy 6016 > (20110405) __________ > > Tuto spravu preveril ESET NOD32 Antivirus. > > http://www.eset.sk > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > -- Carsten Bock http://www.ng-voice.com mailto:carsten@ng-voice.com

2 1

[PATCH] modules_k/uac: fix from/to restore for small original URI
by Timo Teräs 06 Apr '11

06 Apr '11

Seems that the URI length check is superfluous and fails under certain conditions. It does not make sense for the URI to have zero bytes, so just use the first seen zero byte as end marker. I have a reproducible test case where the restore inserts URI with multiple zero-bytes to wire. This happens if the original URI is smaller than the one we rewrote it to using uac_replace_from. Signed-off-by: Timo Teräs <timo.teras(a)iki.fi> --- modules_k/uac/from.c | 14 ++++++++------ 1 files changed, 8 insertions(+), 6 deletions(-) However, I think the delta encoding used for the RR attribute is flawed. Hostile remote server could rewrite the RR attribute and/or From/To headers in a way to forge it to something it was not in the first place. Additionally the delta-encoded RR attribute breaks if the From/To header isn't exact copy of what we sent. Would it not make more sense to just send the real original header (possibly encrypted) but with a checksum? We could then verify if someone had clobbered the RR attribute and ignore it. And we could always restore the original URI even if the URI we are swapping was modified unexpectedly. diff --git a/modules_k/uac/from.c b/modules_k/uac/from.c index 4657e11..50822b6 100644 --- a/modules_k/uac/from.c +++ b/modules_k/uac/from.c @@ -463,15 +463,17 @@ int restore_from( struct sip_msg *msg, int *is_from ) LM_ERR("new URI shorter than old URI\n"); goto failed; } - for( i=0 ; i<old_uri.len ; i++ ) + for( i=0 ; i<old_uri.len ; i++ ) { new_uri.s[i] ^= old_uri.s[i]; - if (new_uri.len==old_uri.len) { - for( ; new_uri.len && (new_uri.s[new_uri.len-1]==0) ; new_uri.len-- ); - if (new_uri.len==0) { - LM_ERR("new URI got 0 len\n"); - goto failed; + if (new_uri.s[i] == 0) { + new_uri.len = i; + break; } } + if (new_uri.len==0) { + LM_ERR("new URI got 0 len\n"); + goto failed; + } LM_DBG("decoded uris are: new=[%.*s] old=[%.*s]\n", new_uri.len, new_uri.s, old_uri.len, old_uri.s); -- 1.7.1

2 1

Custom voicemail table
by Lucas Alvarez 05 Apr '11

05 Apr '11

Hi, is it possible to change the name of the table voicemessages for voicemail profile in a kamailio-asterisk integration? I mean of the kamailio side. Thanks in advance. Lucas

2 2

Kamailio ims / rtpproxy
by Pavel Segeč 05 Apr '11

05 Apr '11

Hi, I would like ask you if there is a kamailio ims module documentation somewhere. I need some deeper information how some things are implemented, I have problem to setup NAT traversal with RTPproxy for our Kamailio IMS platform. There is no communication among the PCSCF and RTPproxy server over control udp sock. We tried unix socket also with the same results. Thanks palo73 http://nil.uniza.sk

1 0

Kamailio RPMs for v3.1.3
by Daniel-Constantin Mierla 05 Apr '11

05 Apr '11

Hello, the RPM packages for Kamailio recently released v3.1.3 are now available for CentOS, Fedora, RedHat and OpenSUSE, available on the opensuse.org build service: * https://build.opensuse.org/package/show?package=kamailio&project=home%3Akam… <https://build.opensuse.org/package/show?package=kamailio&project=home%3Akam…> Specific download links are: * http://download.opensuse.org/repositories/home:/kamailio:/telephony/CentOS_… * http://download.opensuse.org/repositories/home:/kamailio:/telephony/RedHat_… * http://download.opensuse.org/repositories/home:/kamailio:/telephony/Fedora_… * http://download.opensuse.org/repositories/home:/kamailio:/telephony/openSUS… Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

1 0

Kamailio v3.1.3 Released
by Daniel-Constantin Mierla 05 Apr '11

05 Apr '11

Hello, Kamailio SIP Server v3.1.3 stable release is out. This is a maintenance release of latest stable branch, 3.1, that includes fixes since release of v3.1.2. There is no change to database schema or configuration language structure. Deployments running v3.1.0, v3.1.1 or 3.1.2 are strongly recommended to be upgraded to v3.1.3. For more details about version 3.1.3, visit: http://www.kamailio.org/w/2011/04/kamailio-v3-1-3-released/ Note: existing configuration files for installations on 64b operating systems may need to be updated to specify 'lib64' to the path where the modules are installed (to the value of parameter 'mpath'). See bottom note of the link provided above. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

3 3

tcp hangup
by Juha Heinanen 05 Apr '11

05 Apr '11

i have two sr proxies talking with each other over tcp. one is running 3.0 and the other 3.1. sometimes i notice that no sip requests or any other tcp packets get through from 3.0 sr host to 3.1 sr host. netstat on 3.0 sr host shows that tcp connection to 3.1 sr host is in ESTABLISHED state whereas netstat on 3.1 sr shows that the same tcp connection is in FIN_WAIT1 state. it takes several minutes before the hanging tcp connection is replaced by a new working one. there is no ip level connectivity problem between the two sr hosts. any suggestions on what might be going on? is there some known bug in 3.0 tcp implementation that could explain this? -- juha

3 12

[PATCH] modules_k/db_sqlite: new sql backend
by Timo Teräs 05 Apr '11

05 Apr '11

Implements the kamailio database API for SQLite. Signed-off-by: Timo Teräs <timo.teras(a)iki.fi> --- modules_k/db_sqlite/Makefile | 15 ++ modules_k/db_sqlite/db_sqlite.c | 90 +++++++ modules_k/db_sqlite/dbase.c | 515 +++++++++++++++++++++++++++++++++++++++ modules_k/db_sqlite/dbase.h | 72 ++++++ 4 files changed, 692 insertions(+), 0 deletions(-) create mode 100644 modules_k/db_sqlite/Makefile create mode 100644 modules_k/db_sqlite/db_sqlite.c create mode 100644 modules_k/db_sqlite/dbase.c create mode 100644 modules_k/db_sqlite/dbase.h diff --git a/modules_k/db_sqlite/Makefile b/modules_k/db_sqlite/Makefile new file mode 100644 index 0000000..9a75593 --- /dev/null +++ b/modules_k/db_sqlite/Makefile @@ -0,0 +1,15 @@ +# $Id$ +# +# WARNING: do not run this directly, it should be run by the master Makefile + +include ../../Makefile.defs +auto_gen= +NAME=db_sqlite.so + +LIBS +=-lsqlite3 + +DEFS+=-DOPENSER_MOD_INTERFACE + +SERLIBPATH=../../lib +SER_LIBS+=$(SERLIBPATH)/srdb1/srdb1 +include ../../Makefile.modules diff --git a/modules_k/db_sqlite/db_sqlite.c b/modules_k/db_sqlite/db_sqlite.c new file mode 100644 index 0000000..8ec16e3 --- /dev/null +++ b/modules_k/db_sqlite/db_sqlite.c @@ -0,0 +1,90 @@ +/* + * $Id$ + * + * SQlite module interface + * + * Copyright (C) 2010 Timo Teräs + * + * This file is part of Kamailio, a free SIP server. + * + * Kamailio is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version + * + * Kamailio is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#include <sys/time.h> +#include <sqlite3.h> + +#include "../../sr_module.h" +#include "../../lib/srdb1/db.h" +#include "dbase.h" + +MODULE_VERSION + +static int sqlite_bind_api(db_func_t *dbb) +{ + if(dbb==NULL) + return -1; + + memset(dbb, 0, sizeof(db_func_t)); + + dbb->use_table = db_sqlite_use_table; + dbb->init = db_sqlite_init; + dbb->close = db_sqlite_close; + dbb->free_result = db_sqlite_free_result; + dbb->query = db_sqlite_query; + dbb->insert = db_sqlite_insert; + dbb->delete = db_sqlite_delete; + dbb->update = db_sqlite_update; + dbb->raw_query = db_sqlite_raw_query; + + return 0; +} + +static cmd_export_t cmds[] = { + {"db_bind_api", (cmd_function)sqlite_bind_api, 0, 0, 0, 0}, + {0, 0, 0, 0, 0, 0} +}; + +static int sqlite_mod_init(void) +{ + sqlite3_initialize(); + + LM_INFO("SQlite library version %s (compiled using %s)\n", + sqlite3_libversion(), + SQLITE_VERSION); + return 0; +} + + +static void sqlite_mod_destroy(void) +{ + LM_INFO("SQlite terminate\n"); + + sqlite3_shutdown(); +} + +struct module_exports exports = { + "db_sqlite", + DEFAULT_DLFLAGS, /* dlopen flags */ + cmds, /* module commands */ + 0, /* module parameters */ + 0, /* exported statistics */ + 0, /* exported MI functions */ + 0, /* exported pseudo-variables */ + 0, /* extra processes */ + sqlite_mod_init, /* module initialization function */ + 0, /* response function*/ + sqlite_mod_destroy, /* destroy function */ + 0 /* per-child init function */ +}; diff --git a/modules_k/db_sqlite/dbase.c b/modules_k/db_sqlite/dbase.c new file mode 100644 index 0000000..b64b327 --- /dev/null +++ b/modules_k/db_sqlite/dbase.c @@ -0,0 +1,515 @@ +/* + * $Id$ + * + * SQlite module core functions + * + * Copyright (C) 2010 Timo Teräs + * + * This file is part of Kamailio, a free SIP server. + * + * Kamailio is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version + * + * Kamailio is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#include "../../mem/mem.h" +#include "../../dprint.h" +#include "../../lib/srdb1/db_pool.h" +#include "../../lib/srdb1/db_ut.h" +#include "../../lib/srdb1/db_res.h" +#include "../../lib/srdb1/db_query.h" +#include "dbase.h" + +static time_t sqlite_to_timet(double rT) +{ + return 86400.0*(rT - 2440587.5) + 0.5; +} + +static double timet_to_sqlite(time_t t) +{ + return ((((double) t) - 0.5) / 86400.0) + 2440587.5; +} + +/* + * Initialize database module + * No function should be called before this + */ + +static struct sqlite_connection * db_sqlite_new_connection(const struct db_id* id) +{ + struct sqlite_connection *con; + int rc; + + con = pkg_malloc(sizeof(*con)); + if (!con) { + LM_ERR("failed to allocate driver connection\n"); + return NULL; + } + + memset(con, 0, sizeof(*con)); + con->hdr.ref = 1; + con->hdr.id = (struct db_id*) id; /* set here - freed on error */ + + rc = sqlite3_open_v2(id->database, &con->conn, + SQLITE_OPEN_READWRITE | SQLITE_OPEN_CREATE, NULL); + if (rc != SQLITE_OK) { + pkg_free(con); + LM_ERR("failed to open sqlite database '%s'\n", id->database); + return NULL; + } + + return con; +} + +db1_con_t* db_sqlite_init(const str* _url) +{ + return db_do_init(_url, (void *) db_sqlite_new_connection); +} + + +/* + * Shut down database module + * No function should be called after this + */ + +static void db_sqlite_free_connection(struct sqlite_connection* con) +{ + if (!con) return; + + sqlite3_close(con->conn); + free_db_id(con->hdr.id); + pkg_free(con); +} + +void db_sqlite_close(db1_con_t* _h) +{ + db_do_close(_h, db_sqlite_free_connection); +} + +/* + * Release a result set from memory + */ +int db_sqlite_free_result(db1_con_t* _h, db1_res_t* _r) +{ + if (!_h || !_r) { + LM_ERR("invalid parameter value\n"); + return -1; + } + + if (db_free_result(_r) < 0) + { + LM_ERR("failed to free result structure\n"); + return -1; + } + return 0; +} + +/* + * Store name of table that will be used by + * subsequent database functions + */ +int db_sqlite_use_table(db1_con_t* _h, const str* _t) +{ + return db_use_table(_h, _t); +} + +/* + * Reset query context + */ +static void db_sqlite_cleanup_query(const db1_con_t* _c) +{ + struct sqlite_connection *conn = CON_SQLITE(_c); + int rc; + + if (conn->stmt != NULL) { + rc = sqlite3_finalize(conn->stmt); + if (rc != SQLITE_OK) + LM_ERR("finalize failed: %s\n", + sqlite3_errmsg(conn->conn)); + } + + conn->stmt = NULL; + conn->bindpos = 0; +} + +/* + * Convert value to sql-string as db bind index + */ +static int db_sqlite_val2str(const db1_con_t* _c, const db_val_t* _v, char* _s, int* _len) +{ + struct sqlite_connection *conn; + int ret; + + if (!_c || !_v || !_s || !_len || *_len <= 0) { + LM_ERR("invalid parameter value\n"); + return -1; + } + + conn = CON_SQLITE(_c); + if (conn->bindpos >= DB_SQLITE_MAX_BINDS) { + LM_ERR("too many bindings, recompile with larger DB_SQLITE_MAX_BINDS\n"); + return -2; + } + + conn->bindarg[conn->bindpos] = _v; + ret = snprintf(_s, *_len, "?%u", ++conn->bindpos); + if ((unsigned)ret >= (unsigned) *_len) + return -11; + + *_len = ret; + return 0; +} + +/* + * Send an SQL query to the server + */ +static int db_sqlite_submit_query(const db1_con_t* _h, const str* _s) +{ + struct sqlite_connection *conn = CON_SQLITE(_h); + sqlite3_stmt *stmt; + const db_val_t *val; + int rc, i; + + LM_DBG("submit_query: %.*s\n", _s->len, _s->s); + + rc = sqlite3_prepare_v2(conn->conn, _s->s, _s->len, &stmt, NULL); + if (rc != SQLITE_OK) { + LM_ERR("failed to prepare statement: %s\n", + sqlite3_errmsg(conn->conn)); + return -1; + } + conn->stmt = stmt; + + for (i = 1; i <= conn->bindpos; i++) { + val = conn->bindarg[i-1]; + if (VAL_NULL(val)) { + rc = sqlite3_bind_null(stmt, i); + } else switch (VAL_TYPE(val)) { + case DB1_INT: + rc = sqlite3_bind_int(stmt, i, VAL_INT(val)); + break; + case DB1_BIGINT: + rc = sqlite3_bind_int64(stmt, i, VAL_BIGINT(val)); + break; + case DB1_DOUBLE: + rc = sqlite3_bind_double(stmt, i, VAL_DOUBLE(val)); + break; + case DB1_STRING: + rc = sqlite3_bind_text(stmt, i, + VAL_STRING(val), -1, NULL); + break; + case DB1_STR: + rc = sqlite3_bind_text(stmt, i, + VAL_STR(val).s, VAL_STR(val).len, NULL); + break; + case DB1_DATETIME: + rc = sqlite3_bind_double(stmt, i, timet_to_sqlite(VAL_TIME(val))); + break; + case DB1_BLOB: + rc = sqlite3_bind_blob(stmt, i, + VAL_BLOB(val).s, VAL_BLOB(val).len, + NULL); + break; + case DB1_BITMAP: + rc = sqlite3_bind_int(stmt, i, VAL_BITMAP(val)); + break; + default: + LM_ERR("unknown bind value type %d\n", VAL_TYPE(val)); + return -1; + } + if (rc != SQLITE_OK) { + LM_ERR("Parameter bind failed: %s\n", + sqlite3_errmsg(conn->conn)); + return -1; + } + } + + return 0; +} + +static int decltype_to_dbtype(const char *decltype) +{ + /* SQlite3 has dynamic typing. It does not store the actual + * exact type, instead it uses 'affinity' depending on the + * value. We have to go through the declaration types to see + * what to return. */ + if (strstr(decltype, "INT") != NULL || + strncasecmp(decltype, "SERIAL", 6) == 0) + return DB1_INT; + if (strstr(decltype, "CHAR") != NULL) + return DB1_STRING; + if (strstr(decltype, "TEXT") != NULL) + return DB1_STR; + if (strstr(decltype, "REAL") != NULL || + strstr(decltype, "FLOA") != NULL || + strstr(decltype, "DOUB") != NULL) + return DB1_DOUBLE; + if (strstr(decltype, "BLOB") != NULL) + return DB1_BLOB; + if (strncasecmp(decltype, "TIME", 4) == 0 || + strncasecmp(decltype, "DATE", 4) == 0) + return DB1_DATETIME; + + LM_ERR("sqlite decltype '%s' not recognized, defaulting to int", + decltype); + return DB1_INT; +} + +static str* str_dup(const char *_s) +{ + str *s; + int len = strlen(_s); + + s = (str*) pkg_malloc(sizeof(str)+len+1); + s->len = len; + s->s = ((char*)s) + sizeof(str); + memcpy(s->s, _s, len); + s->s[len] = '\0'; + + return s; +} + +static void str_assign(str* s, const char *_s, int len) +{ + s->s = (char *) pkg_malloc(len + 1); + s->len = len; + memcpy(s->s, _s, len); + s->s[len] = 0; +} + +/* + * Read database answer and fill the structure + */ +int db_sqlite_store_result(const db1_con_t* _h, db1_res_t** _r) +{ + struct sqlite_connection *conn = CON_SQLITE(_h); + db1_res_t *res; + int i, rc, num_rows = 0, num_alloc = 0; + db_row_t *rows = NULL, *row; + db_val_t *val; + + res = db_new_result(); + if (res == NULL) + goto no_mem; + + while (1) { + rc = sqlite3_step(conn->stmt); + if (rc == SQLITE_DONE) { + *_r = res; + return 0; + } + if (rc != SQLITE_ROW) { + LM_INFO("sqlite3_step failed: %s\n", sqlite3_errmsg(conn->conn)); + goto err; + } + if (num_rows == 0) { + /* get column types */ + rc = sqlite3_column_count(conn->stmt); + if (db_allocate_columns(res, rc) != 0) + goto err; + RES_COL_N(res) = rc; + + for (i = 0; i < RES_COL_N(res); i++) { + RES_NAMES(res)[i] = str_dup(sqlite3_column_name(conn->stmt, i)); + RES_TYPES(res)[i] = decltype_to_dbtype(sqlite3_column_decltype(conn->stmt, i)); + } + } + if (num_rows >= num_alloc) { + if (num_alloc) + num_alloc *= 2; + else + num_alloc = 8; + rows = pkg_realloc(rows, sizeof(db_row_t) * num_alloc); + if (rows == NULL) + goto no_mem; + RES_ROWS(res) = rows; + } + + row = &RES_ROWS(res)[num_rows]; + num_rows++; + RES_ROW_N(res) = num_rows; /* rows in this result set */ + RES_NUM_ROWS(res) = num_rows; /* rows in total */ + + if (db_allocate_row(res, row) != 0) + goto no_mem; + + for (i = 0, val = ROW_VALUES(row); i < RES_COL_N(res); i++, val++) { + VAL_TYPE(val) = RES_TYPES(res)[i]; + VAL_NULL(val) = 0; + VAL_FREE(val) = 0; + if (sqlite3_column_type(conn->stmt, i) == SQLITE_NULL) { + VAL_NULL(val) = 1; + } else switch (VAL_TYPE(val)) { + case DB1_INT: + VAL_INT(val) = sqlite3_column_int(conn->stmt, i); + break; + case DB1_BIGINT: + VAL_BIGINT(val) = sqlite3_column_int64(conn->stmt, i); + break; + case DB1_STRING: + /* first field of struct str* is the char* so we can just + * do whatever DB1_STR case does */ + case DB1_STR: + str_assign(&VAL_STR(val), + (const char*) sqlite3_column_text(conn->stmt, i), + sqlite3_column_bytes(conn->stmt, i)); + VAL_FREE(val) = 1; + break; + case DB1_DOUBLE: + VAL_DOUBLE(val) = sqlite3_column_double(conn->stmt, i); + break; + case DB1_DATETIME: + VAL_TIME(val) = sqlite_to_timet(sqlite3_column_double(conn->stmt, i)); + break; + case DB1_BLOB: + str_assign(&VAL_BLOB(val), + (const char*) sqlite3_column_blob(conn->stmt, i), + sqlite3_column_bytes(conn->stmt, i)); + VAL_FREE(val) = 1; + break; + default: + LM_ERR("unhandled db-type\n"); + goto err; + } + } + } + +no_mem: + LM_ERR("no private memory left\n"); +err: + if (res) + db_free_result(res); + return -1; +} + +/* + * Query table for specified rows + * _h: structure representing database connection + * _k: key names + * _op: operators + * _v: values of the keys that must match + * _c: column names to return + * _n: number of key=values pairs to compare + * _nc: number of columns to return + * _o: order by the specified column + */ +int db_sqlite_query(const db1_con_t* _h, const db_key_t* _k, const db_op_t* _op, + const db_val_t* _v, const db_key_t* _c, int _n, int _nc, + const db_key_t _o, db1_res_t** _r) +{ + int rc; + + rc = db_do_query(_h, _k, _op, _v, _c, _n, _nc, _o, _r, + db_sqlite_val2str, + db_sqlite_submit_query, + db_sqlite_store_result); + db_sqlite_cleanup_query(_h); + + return rc; +} + +static int db_sqlite_commit(const db1_con_t* _h) +{ + struct sqlite_connection *conn = CON_SQLITE(_h); + int rc; + + rc = sqlite3_step(conn->stmt); + if (rc != SQLITE_DONE && rc != SQLITE_OK) { + LM_ERR("sqlite commit failed: %s\n", + sqlite3_errmsg(conn->conn)); + return -1; + } + + return 0; +} + +/* + * Insert a row into specified table + * _h: structure representing database connection + * _k: key names + * _v: values of the keys + * _n: number of key=value pairs + */ +int db_sqlite_insert(const db1_con_t* _h, const db_key_t* _k, const db_val_t* _v, + int _n) +{ + int rc = -1; + + rc = db_do_insert(_h, _k, _v, _n, + db_sqlite_val2str, + db_sqlite_submit_query); + if (rc == 0) + rc = db_sqlite_commit(_h); + db_sqlite_cleanup_query(_h); + + return rc; +} + + +/* + * Delete a row from the specified table + * _h: structure representing database connection + * _k: key names + * _o: operators + * _v: values of the keys that must match + * _n: number of key=value pairs + */ +int db_sqlite_delete(const db1_con_t* _h, const db_key_t* _k, const db_op_t* _o, + const db_val_t* _v, int _n) +{ + int rc; + + rc = db_do_delete(_h, _k, _o, _v, _n, + db_sqlite_val2str, + db_sqlite_submit_query); + if (rc == 0) + rc = db_sqlite_commit(_h); + db_sqlite_cleanup_query(_h); + + return rc; +} + + +/* + * Update some rows in the specified table + * _h: structure representing database connection + * _k: key names + * _o: operators + * _v: values of the keys that must match + * _uk: updated columns + * _uv: updated values of the columns + * _n: number of key=value pairs + * _un: number of columns to update + */ +int db_sqlite_update(const db1_con_t* _h, const db_key_t* _k, const db_op_t* _o, + const db_val_t* _v, const db_key_t* _uk, const db_val_t* _uv, + int _n, int _un) +{ + int rc; + + rc = db_do_update(_h, _k, _o, _v, _uk, _uv, _n, _un, + db_sqlite_val2str, + db_sqlite_submit_query); + if (rc == 0) + rc = db_sqlite_commit(_h); + db_sqlite_cleanup_query(_h); + + return rc; +} + +int db_sqlite_raw_query(const db1_con_t* _h, const str* _s, db1_res_t** _r) +{ + return db_do_raw_query(_h, _s, _r, + db_sqlite_submit_query, + db_sqlite_store_result); +} diff --git a/modules_k/db_sqlite/dbase.h b/modules_k/db_sqlite/dbase.h new file mode 100644 index 0000000..de006bf --- /dev/null +++ b/modules_k/db_sqlite/dbase.h @@ -0,0 +1,72 @@ +/* + * $Id$ + * + * SQlite module core functions + * + * Copyright (C) 2010 Timo Teräs + * + * This file is part of Kamailio, a free SIP server. + * + * Kamailio is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version + * + * Kamailio is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + + +#ifndef DBASE_H +#define DBASE_H + +#include <sqlite3.h> + +#include "../../lib/srdb1/db_pool.h" +#include "../../lib/srdb1/db_con.h" +#include "../../lib/srdb1/db_res.h" +#include "../../lib/srdb1/db_key.h" +#include "../../lib/srdb1/db_op.h" +#include "../../lib/srdb1/db_val.h" + +#define DB_SQLITE_MAX_BINDS 64 + +struct sqlite_connection { + struct pool_con hdr; + + sqlite3 *conn; + int bindpos; + + sqlite3_stmt *stmt; + const db_val_t *bindarg[DB_SQLITE_MAX_BINDS]; +}; + +#define CON_SQLITE(db_con) ((struct sqlite_connection *) db_con->tail) + +db1_con_t* db_sqlite_init(const str* _sqlurl); +void db_sqlite_close(db1_con_t* _h); + +int db_sqlite_free_result(db1_con_t* _h, db1_res_t* _r); + +int db_sqlite_query(const db1_con_t* _h, const db_key_t* _k, const db_op_t* _op, + const db_val_t* _v, const db_key_t* _c, int _n, int _nc, + const db_key_t _o, db1_res_t** _r); +int db_sqlite_insert(const db1_con_t* _h, const db_key_t* _k, const db_val_t* _v, + int _n); +int db_sqlite_delete(const db1_con_t* _h, const db_key_t* _k, const db_op_t* _o, + const db_val_t* _v, int _n); +int db_sqlite_update(const db1_con_t* _h, const db_key_t* _k, const db_op_t* _o, + const db_val_t* _v, const db_key_t* _uk, const db_val_t* _uv, + int _n, int _un); +int db_sqlite_raw_query(const db1_con_t* _h, const str* _s, db1_res_t** _r); + +int db_sqlite_use_table(db1_con_t* _h, const str* _t); + + +#endif /* DBASE_H */ -- 1.7.1

3 3

SIP NOTIFY problem in "presence_dialoginfo" module?
by Klaus Feichtinger 04 Apr '11

04 Apr '11

Hello list, I have a special situation in which string characters of the "root-line" of the notify-body are overwritten by Kamailio. In detail: the root-line of the NOTIFY message sent to the subscriber looks like: <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="1" ll" entity="sip:117090@10.16.10.99"> instead of <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="1" state="full" entity="sip:117090@10.16.10.99"> In other words: the string state="full" is overwritten by space characters. Therefore the message is no longer "valid" and the parser of the subscriber-user agent has problems.... The exact scenario in my use case is, that user agents can subscribe the events "presence" and "dialog" on Kamailio. The information about presence and dialog-states is published to Kamailio by PUBLISH messages from the user agents themselves. Kamailio does not have to generate this information itself. In the following lines I have an excerpt of the signalisation: 1) initial subscription of party A, 2) the responded notification (without body) 3) subscription of party B 4) the responded notification (with body) 5) a publish message of the subscribed party B with actual dialog information and 6) the notification with body to the subscriber A. 1) Initial SUBSCRIBE message of party A: ======================================SUBSCRIBE sip:117090@10.16.10.99 SIP/2.0 Via: SIP/2.0/UDP 10.16.10.93:5060;rport;branch=z9hG4bK1411449981 From: <sip:117093@10.16.10.99>;tag=2453899634-24035392-1301468026894 To: <sip:117090@10.16.10.99> Call-ID: 2442490492-24035392-1301468026894(a)10.16.10.93 CSeq: 20 SUBSCRIBE Contact: <sip:117093@10.16.10.93:5060> Max-Forwards: 70 User-Agent: SipTapi Expires: 3600 Event: dialog Content-Length: 0 2) Responded NOTIFY message from Kamailio: ========================================NOTIFY sip:117093@10.16.10.93:5060 SIP/2.0 Via: SIP/2.0/UDP 10.16.10.99;branch=z9hG4bK7cc7.08e060b6000000000000000000000000.0 To: sip:117093@10.16.10.99;tag=2453899634-24035392-1301468026894 From: sip:117090@10.16.10.99;tag=a23c71953194c2c72e41c4d20e4f7127-86e1 CSeq: 1 NOTIFY Call-ID: 2442490492-24035392-1301468026894(a)10.16.10.93 Content-Length: 0 User-Agent: kamailio (3.1.2 (i386/linux)) Max-Forwards: 70 Event: dialog Contact: <sip:10.16.10.99:5060> Subscription-State: active;expires=3670 3) Initial SUBSCRIBE message of party B: ======================================SUBSCRIBE sip:117093@10.16.10.99 SIP/2.0 Via: SIP/2.0/UDP 10.16.10.90:5060;rport;branch=z9hG4bK4016956575 From: <sip:117090@10.16.10.99>;tag=2605346227-26329880-1301468174702 To: <sip:117093@10.16.10.99> Call-ID: 2982039389-26329880-1301468174702(a)10.16.10.90 CSeq: 20 SUBSCRIBE Contact: <sip:117090@10.16.10.90:5060> Max-Forwards: 70 User-Agent: SipTapi Expires: 3600 Event: dialog Content-Length: 0 4) Responded NOTIFY message from Kamailio (with version number "00000000000"!!!!): ==============================================================================NOTIFY sip:117090@10.16.10.90:5060 SIP/2.0 Via: SIP/2.0/UDP 10.16.10.99;branch=z9hG4bK10e8.31cb2284000000000000000000000000.0 To: sip:117090@10.16.10.99;tag=2605346227-26329880-1301468174702 From: sip:117093@10.16.10.99;tag=a23c71953194c2c72e41c4d20e4f7127-5f56 CSeq: 1 NOTIFY Call-ID: 2982039389-26329880-1301468174702(a)10.16.10.90 Content-Length: 593 User-Agent: kamailio (3.1.2 (i386/linux)) Max-Forwards: 70 Event: dialog Contact: <sip:10.16.10.99:5060> Subscription-State: active;expires=3670 Content-Type: application/dialog-info+xml <?xml version="1.0"?> <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="00000000000" state="full" entity="117093(a)10.16.10.99"> <dialog id="3375446242-26591296-1301467877329(a)10.16.10.93" call-id="3375446242-26591296-1301467877329(a)10.16.10.93" direction="initiator"> <state>terminated</state> <remote> <identity>sip:019992116002@10.16.10.99</identity> <target uri="sip:019992116002@10.16.10.99"/> </remote> <local> <identity>sip:117093@10.16.10.99</identity> <target uri="sip:117093@10.16.10.99"/> </local> </dialog> </dialog-info> 5) The subscribed party is sending a PUBLISH message with following information: ============================================================================PUBLISH sip:117090@10.16.10.99 SIP/2.0 Via: SIP/2.0/UDP 10.16.10.90:5060;rport;branch=z9hG4bK348084694 From: <sip:117090@10.16.10.99>;tag=3644648667-26329880-1301468223499 To: <sip:117090@10.16.10.99> Call-ID: 332031778-26329880-1301468223499(a)10.16.10.90 CSeq: 20 PUBLISH Max-Forwards: 70 User-Agent: SipTapi Content-Disposition: render;handling=required Expires: 3600 Event: dialog Content-Type: application/dialog-info+xml Content-Length: 592 <?xml version="1.0"?> <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="4" state="full" entity="sip:117090@10.16.10.99"> <dialog id="2769397361-26329880-1301468223108(a)10.16.10.90" call-id="2769397361-26329880-1301468223108(a)10.16.10.90" direction="initiator"> <state>confirmed</state> <remote> <identity><sip:116001@10.16.10.99></identity> <target uri="<sip:116001@10.16.10.99>"/> </remote> <local> <identity>sip:117090@10.16.10.99</identity> <target uri="sip:117090@10.16.10.99"/> </local> </dialog> </dialog-info> 6) Status informative NOTIFY message from Kamailio (built from the PUBLISH message above - with correct version but string overwriting!!!): =======================================================================================================================================NOTIFY sip:117093@10.16.10.93:5060 SIP/2.0 Via: SIP/2.0/UDP 10.16.10.99;branch=z9hG4bK4cc7.faa8dc87000000000000000000000000.0 To: sip:117093@10.16.10.99;tag=2453899634-24035392-1301468026894 From: sip:117090@10.16.10.99;tag=a23c71953194c2c72e41c4d20e4f7127-86e1 CSeq: 2 NOTIFY Call-ID: 2442490492-24035392-1301468026894(a)10.16.10.93 Content-Length: 592 User-Agent: kamailio (3.1.2 (i386/linux)) Max-Forwards: 70 Event: dialog Contact: <sip:10.16.10.99:5060> Subscription-State: active;expires=3570 Content-Type: application/dialog-info+xml <?xml version="1.0"?> <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="1" ll" entity="sip:117090@10.16.10.99"> <dialog id="2769397361-26329880-1301468223108(a)10.16.10.90" call-id="2769397361-26329880-1301468223108(a)10.16.10.90" direction="initiator"> <state>confirmed</state> <remote> <identity><sip:116001@10.16.10.99></identity> <target uri="<sip:116001@10.16.10.99>"/> </remote> <local> <identity>sip:117090@10.16.10.99</identity> <target uri="sip:117090@10.16.10.99"/> </local> </dialog> </dialog-info> Debug information of Kamailio and information in the source code show that the version number "00000000000" is a placeholder only, which should be replaced with a valid version number. However, it seems that there is a problem in handling these strings. The version number is replaced, but in case that the version id is only one character long, ten space characters are inserted- overwriting other information...... Can anybody comment this problem? Is this a bug of the presence_dialoginfo- or another module? Thanks in advance! Regards, Klaus P.S. additional information from source code an debug output of kamailio: notify_body.c: [...] /* The version must be increased for each new document and is a 32bit int. As the version is different for each watcher, we can not set here the correct value. Thus, we just put here a placeholder which will be replaced by the correct value in the aux_body_processing callback. Thus we have CPU intensive XML aggregation only once and can use quick search&replace in the per-watcher aux_body_processing callback. We use 11 chracters as an signed int (although RFC says unsigned int we use signed int as presence module stores "version" in DB as signed int) has max. 10 characters + 1 character for the sign */ [...] Debug information from Kamailio: Ad 4) SipSrv1 /usr/sbin/kamailio[26764]: DEBUG: presence_dialoginfo [notify_body.c:67]: [pres_user]=117093 [pres_domain]= 10.16.10.99, [n]=1 SipSrv1 /usr/sbin/kamailio[26764]: DEBUG: presence_dialoginfo [notify_body.c:107]: [pres_user]=117093 [pres_domain]= 10.16.10.99, [n]=1 SipSrv1 /usr/sbin/kamailio[26764]: DEBUG: presence_dialoginfo [notify_body.c:199]: node type: Element, name: dialog SipSrv1 /usr/sbin/kamailio[26764]: DEBUG: presence_dialoginfo [notify_body.c:73]: [n_body]=0x835fe54 SipSrv1 /usr/sbin/kamailio[26764]: DEBUG: presence_dialoginfo [notify_body.c:76]: [*n_body]=<?xml version="1.0"?>#012<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="00000000000" state="full" entity="117093(a)10.16.10.99">#012 <dialog id="6b1de4df-6c9ce978(a)10.16.10.203" call-id="6b1de4df-6c9ce978(a)10.16.10.203" direction="recipient">#012 <state>confirmed</state>#012 <remote>#012 <identity>sip:116001@10.16.10.99</identity>#012 <target uri="sip:116001@10.16.10.99"/>#012 </remote>#012 <local>#012 <identity>sip:117093@10.16.10.99</identity>#012 <target uri="sip:117093@10.16.10.99"/>#012 </local>#012 </dialog>#012</dialog-info>#012 SipSrv1 /usr/sbin/kamailio[26764]: DEBUG: presence_dialoginfo [notify_body.c:328]: replace version with "1" SipSrv1 /usr/sbin/kamailio[26764]: DEBUG: presence [notify.c:1461]: dialog info: SipSrv1 /usr/sbin/kamailio[26764]: DEBUG: presence [notify.c:122]: #012#011[pres_uri]= sip:117093@10.16.10.99#012#011[to_user]117093#011[to_domain]= 10.16.10.99#012#011[w_user]= 117090#011[w_domain]10.16.10.99#012#011[event]= dialog#012#011[status]= active#012#011[expires]3519#012#011[callid]3119346140-8635704-1301477522967(a)10.16.10.90#011[local_cseq]=1#012#011[to_tag ]= a23c71953194c2c72e41c4d20e4f7127-7529#011[from_tag]2282475328-8635704-1301477522967#012#011[contact]sip:117090@10.16.10.90:5060#011[record_route]SipSrv1 /usr/sbin/kamailio[26764]: DEBUG: presence [notify.c:234]: expires 3519 SipSrv1 /usr/sbin/kamailio[26764]: DEBUG: presence [notify.c:1555]: headers:#012Max-Forwards: 70#015#012Event: dialog#015#012Contact: <sip:10.16.10.99:5060>#015#012Subscription-State: active;expires=3570#015#012Content-Type: application/dialog-info+xml#015#012 SipSrv1 /usr/sbin/kamailio[26764]: DEBUG: presence [notify.c:940]: CONTACT sip:117090@10.16.10.90:5060 SipSrv1 /usr/sbin/kamailio[26764]: DEBUG: <core> [socket_info.c:501]: grep_sock_info - checking if host==us: 11==11 && [10.16.10.99] =[10.16.10.99] SipSrv1 /usr/sbin/kamailio[26764]: DEBUG: <core> [socket_info.c:504]: grep_sock_info - checking if port 5060 matches port 5060 SipSrv1 /usr/sbin/kamailio[26764]: DEBUG: presence [notify.c:1729]: ==22/6/37 SipSrv1 /usr/sbin/kamailio[26764]: DEBUG: tm [uac.c:240]: DEBUG:tm:t_uac: next_hop=<sip:117090@10.16.10.90:5060> SipSrv1 /usr/sbin/kamailio[26764]: DEBUG: tm [uac.c:181]: DEBUG: dlg2hash: 27355 SipSrv1 /usr/sbin/kamailio[26764]: DEBUG: <core> [usr_avp.c:646]: DEBUG:destroy_avp_list: destroying list (nil) SipSrv1 /usr/sbin/kamailio[26764]: DEBUG: <core> [usr_avp.c:646]: DEBUG:destroy_avp_list: destroying list (nil) SipSrv1 /usr/sbin/kamailio[26764]: DEBUG: <core> [usr_avp.c:646]: DEBUG:destroy_avp_list: destroying list (nil) SipSrv1 /usr/sbin/kamailio[26764]: DEBUG: <core> [usr_avp.c:646]: DEBUG:destroy_avp_list: destroying list (nil) SipSrv1 /usr/sbin/kamailio[26764]: DEBUG: <core> [usr_avp.c:646]: DEBUG:destroy_avp_list: destroying list (nil) SipSrv1 /usr/sbin/kamailio[26764]: DEBUG: <core> [usr_avp.c:646]: DEBUG:destroy_avp_list: destroying list (nil) SipSrv1 /usr/sbin/kamailio[26764]: INFO: presence [notify.c:1586]: NOTIFY sip:117090@10.16.10.99 via sip:117090@10.16.10.90:5060 on behalf of sip:117093@10.16.10.99 for event dialog Ad 6) SipSrv1 /usr/sbin/kamailio[26764]: DEBUG: presence_dialoginfo [notify_body.c:67]: [pres_user]=117093 [pres_domain]= 10.16.10.99, [n]=1 SipSrv1 /usr/sbin/kamailio[26764]: DEBUG: presence_dialoginfo [notify_body.c:107]: [pres_user]=117093 [pres_domain]= 10.16.10.99, [n]=1 SipSrv1 /usr/sbin/kamailio[26764]: DEBUG: presence_dialoginfo [notify_body.c:199]: node type: Element, name: dialog SipSrv1 /usr/sbin/kamailio[26764]: DEBUG: presence_dialoginfo [notify_body.c:73]: [n_body]=0x835fcb0 SipSrv1 /usr/sbin/kamailio[26764]: DEBUG: presence_dialoginfo [notify_body.c:76]: [*n_body]=<?xml version="1.0"?>#012<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="00000000000" state="full" entity="117093(a)10.16.10.99">#012 <dialog id="6b1de4df-6c9ce978(a)10.16.10.203" call-id="6b1de4df-6c9ce978(a)10.16.10.203" direction="recipient">#012 <state>terminated</state>#012 <remote>#012 <identity>sip:116001@10.16.10.99</identity>#012 <target uri="sip:116001@10.16.10.99"/>#012 </remote>#012 <local>#012 <identity>sip:117093@10.16.10.99</identity>#012 <target uri="sip:117093@10.16.10.99"/>#012 </local>#012 </dialog>#012</dialog-info>#012 SipSrv1 /usr/sbin/kamailio[26764]: DEBUG: presence_dialoginfo [notify_body.c:328]: replace version with "2" SipSrv1 /usr/sbin/kamailio[26764]: DEBUG: presence [notify.c:1461]: dialog info: SipSrv1 /usr/sbin/kamailio[26764]: DEBUG: presence [notify.c:122]: #012#011[pres_uri]= sip:117093@10.16.10.99#012#011[to_user]117093#011[to_domain]= 10.16.10.99#012#011[w_user]= 117090#011[w_domain]10.16.10.99#012#011[event]= dialog#012#011[status]= active#012#011[expires]3242#012#011[callid]3119346140-8635704-1301477522967(a)10.16.10.90#011[local_cseq]=2#012#011[to_tag ]= a23c71953194c2c72e41c4d20e4f7127-7529#011[from_tag]2282475328-8635704-1301477522967#012#011[contact]sip:117090@10.16.10.90:5060#011[record_route]= SipSrv1 /usr/sbin/kamailio[26764]: DEBUG: presence [notify.c:234]: expires 3242 SipSrv1 /usr/sbin/kamailio[26764]: DEBUG: presence [notify.c:1555]: headers:#012Max-Forwards: 70#015#012Event: dialog#015#012Contact: <sip:10.16.10.99:5060>#015#012Subscription-State: active;expires=3270#015#012Content-Type: application/dialog-info+xml#015#012 SipSrv1 /usr/sbin/kamailio[26764]: DEBUG: presence [notify.c:940]: CONTACT sip:117090@10.16.10.90:5060 SipSrv1 /usr/sbin/kamailio[26764]: DEBUG: <core> [socket_info.c:501]: grep_sock_info - checking if host==us: 11==11 && [10.16.10.99] =[10.16.10.99] SipSrv1 /usr/sbin/kamailio[26764]: DEBUG: <core> [socket_info.c:504]: grep_sock_info - checking if port 5060 matches port 5060 SipSrv1 /usr/sbin/kamailio[26764]: DEBUG: presence [notify.c:1729]: ==22/6/37 SipSrv1 /usr/sbin/kamailio[26764]: DEBUG: tm [uac.c:240]: DEBUG:tm:t_uac: next_hop=<sip:117090@10.16.10.90:5060> SipSrv1 /usr/sbin/kamailio[26764]: DEBUG: tm [uac.c:181]: DEBUG: dlg2hash: 27356 SipSrv1 /usr/sbin/kamailio[26764]: DEBUG: <core> [usr_avp.c:646]: DEBUG:destroy_avp_list: destroying list (nil) SipSrv1 /usr/sbin/kamailio[26764]: DEBUG: <core> [usr_avp.c:646]: DEBUG:destroy_avp_list: destroying list (nil) SipSrv1 /usr/sbin/kamailio[26764]: DEBUG: <core> [usr_avp.c:646]: DEBUG:destroy_avp_list: destroying list (nil) SipSrv1 /usr/sbin/kamailio[26764]: DEBUG: <core> [usr_avp.c:646]: DEBUG:destroy_avp_list: destroying list (nil) SipSrv1 /usr/sbin/kamailio[26764]: DEBUG: <core> [usr_avp.c:646]: DEBUG:destroy_avp_list: destroying list (nil) SipSrv1 /usr/sbin/kamailio[26764]: DEBUG: <core> [usr_avp.c:646]: DEBUG:destroy_avp_list: destroying list (nil) SipSrv1 /usr/sbin/kamailio[26764]: INFO: presence [notify.c:1586]: NOTIFY sip:117090@10.16.10.99 via sip:117090@10.16.10.90:5060 on behalf of sip:117093@10.16.10.99 for event dialog

3 6

Suggestion dialog module / $dlg_ctx() problem.
by Alexandre Abreu 04 Apr '11

04 Apr '11

Hello. Using K 3.1.2. Before the setflag() that creates the dialog, I tried to use: $dlg_ctx(flags) = 1; Mar 30 12:29:03 devel kamailio: ERROR: dialog [dlg_var.c:165]: unknown PV name flags Mar 30 12:29:03 devel kamailio: ERROR: <core> [pvapi.c:565]: pvar "dlg_ctx" has an invalid name param [flags] Mar 30 12:29:03 devel kamailio: ERROR: <core> [pvapi.c:720]: wrong char [)/41] in [$dlg_ctx(flags)] at [14 (5)] Mar 30 12:29:03 devel kamailio: : <core> [cfg.y:3409]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 308, column 9-23: unknown script pseudo variable $dlg_ctx(flags) Right now, my goal is to add the user-agent content header ($ua) from caller/callee and list them through 'dlg_list_ctx'. I would suggest something like that: Request route: $dlg_ctx(set->ua_src) = $ua; On reply route: $dlg_ctx(set->ua_dst) = $ua; dlg_list_ctx: . ua_src:: X-Lite 3.0 build 666 ua_dst:: Linksys PAP2 . Does this suggestion make sense? Using another approach is it possible to execute this today? Thanks, Alexandre

4 11

preparing kamailio v3.1.3
by Daniel-Constantin Mierla 04 Apr '11

04 Apr '11

Hello, it is almost 2 months since we packaged 3.1.2, there are commits accumulated in branch 3.1, therefore I plan to package 3.1.3 Monday or Tuesday next week. If anyone has something to add in this regards, please reply. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

1 1

library dir for x86_64 - lib or lib64
by Daniel-Constantin Mierla 04 Apr '11

04 Apr '11

Hello, iirc this was at some point discussed, however, I want to go through it again. Many x86_64 distros expect libraries to be placed under a lib64 directory. For example the RPM spec file language returns /usr/lib64 for %{_libdir} variable. In the master branch I committed a patch that does this detection. As I saw some people saying that 'lib' should be the default also for x86_64, my question is then what do you think we should use? Using 'lib' will definitely make building rpms fail unless we do a patch specific for these distros. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

3 2

meeting in thailand bangkoke
by Meftah Tayeb 03 Apr '11

03 Apr '11

hi guys i'm in thailand right now this week i'm staying in bangkoke anyone want to meet please drop me an email thank you -- Meftah Tayeb inum: +883510001288000 phone: +13477595883

1 0

Setting Global Variables
by Nathaniel L Keeling 02 Apr '11

02 Apr '11

Hello, I am trying to use global variables within my configuration script. I am using Kamailio v3.0.0 and I am following the syntax example written in the sample config file but I keep getting syntax errors. Here is a copy of my syntax and I am using cfg_get.voicemailserver.bindip to get the value. #!ifdef WITH_VOICEMAIL voicemailserver.bindip = "10.0.0.1" desc "Voicemail Server IP Address" voicemailserver.bindport = "5060" desc "Voicemail Server Port" #!endif Here is the error that I am receiving: 0(1875) : <core> [cfg.y:3333]: parse error in config file /usr/local/kamailio-3.0.0/etc/kamailio/kamailio.cfg, line 156, column 17: syntax error 0(1875) : <core> [cfg.y:3333]: parse error in config file /usr/local/kamailio-3.0.0/etc/kamailio/kamailio.cfg, line 156, column 17: unknown config variable 0(1875) : <core> [cfg.y:3330]: parse error in config file /usr/local/kamailio-3.0.0/etc/kamailio/kamailio.cfg, line 156, column 19-28: ERROR: bad config file (3 errors) 0(1875) DEBUG: <core> [mem/shm_mem.c:236]: shm_mem_destroy Thanks Nathaniel

1 0

Avpops problem - bad avp flags.
by Ricardo Martinez 01 Apr '11

01 Apr '11

Hello. I’m having a problems using the avpops with the kamailio version 3.1.2 I have the next configuration: modparam("avpops", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("avpops", "avp_table", "usr_preferences") ----- route[REQINIT] { if( !(avp_db_load("pike", "s:ip_origen") && avp_check("s:ip_origen", "eq/$src_ip/gi")) ) { … } mysql> select * from usr_preferences; +----+------+----------+--------+-----------+------+---------------+---------------------+ | id | uuid | username | domain | attribute | type | value | last_modified | +----+------+----------+--------+-----------+------+---------------+---------------------+ | 1 | pike | 0 | | ip_origen | 0 | 100.20.100.20 | 1900-01-01 00:00:01 | +----+------+----------+--------+-----------+------+---------------+---------------------+ 1 row in set (0.00 sec) When I try to start kamailio I’m getting the next error : Apr 1 10:36:08 kamailio[15418]: ERROR: avpops [avpops_parse.c:129]: error - bad avp flags Apr 1 10:36:08 kamailio[15418]: ERROR: avpops [avpops.c:299]: parse failed Apr 1 10:36:08 kamailio[15418]: ERROR: <core> [route.c:1161]: fixing failed (code=-1) at cfg:/usr/local/etc/kamailio/kamailio.cfg:238 Apr 1 10:36:08 kamailio[15418]: ERROR: <core> [route.c:1161]: fixing failed (code=-1) at cfg:/usr/local/etc/kamailio/kamailio.cfg:263 I can’t find the problem. Can someone help me? Regards. Ricardo.-

2 1

The SIP protocol v2 - we're giving up.
by Olle E. Johansson 01 Apr '11

01 Apr '11

Friends, After having spent many years working with the Asterisk SIP channel driver, Kamailio and the SIPv2 protocol, I have finally realized that this is a dead end. It's getting nowhere and it's way too complicated to set up, run and support in working code. After realizing this, I started a new standardization project together with my friends in Canada, Simon and Marc, to develop a working solution based on the combination of IPv6 and SIP. We have gotten great feedback and now the IETF, the ITU and the IPv6 forum jointly launches the new standard, SIP-six. From the press release: "”We realize that 99% of the SIP users use SIP for PSTN phone calls. The original SIP standards was written with other applications in mind, a vision that never came true.” said Bob Plug, area director in the IETF. ”That’s why we sat down and said to ourselves that this should be way more simple.” The SIP-six standard totally removes the dependency of domains and URI syntax. There’s no point in using this, since everyone seems to think that IP addressing is more than enough. The new standard use part of the vast IPv6 address space to incorporate the E.164 phone numbers as addresses. This is the reverse of the reverse phone number usage in the enum standard, which is no longer needed in SIP-six. By using IPv6 mobile IP, phone users register their phones and get access to their phone number. Users that need security can easily integrate IPsec into their setup. Media and signalling uses the same addressing scheme and is mixed so that both SIP-six, RTP and RTCP only uses one port address - but in this case a port address with 32 bit subaddress identifying the media stream. This finally solves a lot of the firewall traversal issues that SIP v2.0 had. With the combination of mobile IP and use of public IPv6 addresses NAT traversal won’t be an issue. The testbed for SIP-six has been running for a year at six choosen large SIP carriers, with the assistance of Edvina AB in Sweden and ViaGenius in Montreal, Canada. In an International effort, the testbed is today put in production and Roboid phones all over the world is automatically connected to this worldwide network." You will be able to find out more about it here: http://bit.ly/sipsix SIP-six is implemented as a channel driver in Asterisk 2.0, as a replacement for SIP2.0 in Kamailio 4.0 and a channel module in FreeSwitch - all releases to be released later today. Softphones for testing will shortly be available from Blink and Zoiper. Looking forward to continue this project with the rest of the Kamailio/SIP-router community! Special thanks to Daniel for the reference implementation in Kamailio 4.0! Have a nice weekend! /Olle

7 7

valgrind errors from 3.1-stable (git 2fe4d633e4ac)
by Timo Teräs 01 Apr '11

01 Apr '11

Hi all, I'm running 3.1-branch git head currently under valgrind. And I've seen several invalid reads and writes (apparently most are off by one). ==22274== Invalid read of size 1 ==22274== at 0x6DA2C0A: pv_set_ruri_user (pv_core.c:1755) ==22274== by 0x13BA15: lval_pvar_assign (lvalue.c:357) ==22274== by 0x13BF0F: lval_assign (lvalue.c:405) ==22274== by 0x139E4D: do_action (action.c:1472) ==22274== by 0x13A6F1: run_actions (action.c:1555) ==22274== by 0x12FDEB: do_action (action.c:711) ==22274== by 0x13A6F1: run_actions (action.c:1555) ==22274== by 0x13A86F: run_actions_safe (action.c:1607) ==22274== by 0x1E2666: rval_get_int (rvalue.c:904) ==22274== by 0x1E5252: rval_expr_eval_int (rvalue.c:1866) ==22274== by 0x131B8A: do_action (action.c:1071) ==22274== by 0x13A6F1: run_actions (action.c:1555) Is fairly obvious. modules_k/pv/pv_core.c has several places which take backup of "val->rs.s[val->rs.len]" and replaces it with zero for string termination. It's than later on set back to the original value. However, it would appear that the value passed does not always point to a memory area which is large enough. This results in multiple invalid reads and writes of one. The remaining reads are not so clear to me. ==22274== Invalid read of size 1 ==22274== at 0x4811A69: strncpy (mc_replace_strmem.c:339) ==22274== by 0x6DA77FF: set_var_value (pv_svar.c:122) ==22274== by 0x6DA1EDB: pv_set_scriptvar (pv_core.c:1636) ==22274== by 0x13BA15: lval_pvar_assign (lvalue.c:357) ==22274== by 0x13BF0F: lval_assign (lvalue.c:405) ==22274== by 0x139E4D: do_action (action.c:1472) ==22274== by 0x13A6F1: run_actions (action.c:1555) ==22274== by 0x131E05: do_action (action.c:1086) ==22274== by 0x13A6F1: run_actions (action.c:1555) ==22274== by 0x12FDEB: do_action (action.c:711) ==22274== by 0x13A6F1: run_actions (action.c:1555) ==22274== by 0x131E05: do_action (action.c:1086) ==22274== Invalid read of size 2 ==22274== at 0x4811E1F: memcpy (mc_replace_strmem.c:523) ==22274== by 0x1E134D: rval_new_str (rvalue.c:269) ==22274== by 0x1E394E: rval_convert (rvalue.c:1301) ==22274== by 0x1E4780: rval_str_add2 (rvalue.c:1610) ==22274== by 0x1E8681: rval_expr_eval (rvalue.c:2399) ==22274== by 0x13BBD9: lval_assign (lvalue.c:389) ==22274== by 0x139E4D: do_action (action.c:1472) ==22274== by 0x13A6F1: run_actions (action.c:1555) ==22274== by 0x12FDEB: do_action (action.c:711) ==22274== by 0x13A6F1: run_actions (action.c:1555) ==22274== by 0x139229: do_action (action.c:1342) ==22274== by 0x13A6F1: run_actions (action.c:1555) Any ideas how to fix these? Or any instructions what to provide more in the bug report? Cheers, Timo

3 7

RPM packages for Kamailio 3.1.2
by Daniel-Constantin Mierla 31 Mar '11

31 Mar '11

Hello, there is now a service on opensuse build server to create kamailio v3.1.2 (git branch 3.1.2) RPMs i586/x86_63 for Centos 5.5, RedHat 5, Fedora 14 and OpenSuse 11.4, you can check this project at: https://build.opensuse.org/package/show?package=kamailio&project=home%3Akam… Download links are: http://download.opensuse.org/repositories/home:/kamailio:/telephony/CentOS_… http://download.opensuse.org/repositories/home:/kamailio:/telephony/RedHat_… http://download.opensuse.org/repositories/home:/kamailio:/telephony/Fedora_… http://download.opensuse.org/repositories/home:/kamailio:/telephony/openSUS… The x86_64 build for opensuse is still not completed, some parameter to gcc compiler is not working, should be fixed soon. Note that not all modules are packaged since not all dependencies were met in these distros. The RPM spec was started from what is in source tree as kamailio.spec.CentOS, radically updated for cross compilation. Personally I am not using these distros, therefore it will be appreciated if people using any of them are willing to give an installation test and report if all goes ok. Moreover, I will be very thankful if anyone (can be many) wants to jump in and take over the maintenance of this kamailio build service. It would be good to enable the build for other versions of these distros, probably for many it will just work, if not it will require some adjustments on dependencies. Once these builds prove to be ok and useful, we can move them in the global telephony repository of the opensuse build service. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

2 1

LCR defunct_gw use
by Ricardo Martinez 31 Mar '11

31 Mar '11

Hello. Can someone please explain to me hoy can i use the command “defunct_gw()” is used to mark a gw down?. I understand that the gateway is marked down for period of time with the command, but how can I check if a gateway is down?. Can someone show me this with an example in the configuration? *Thanks in advance,* *Regards,* * * * * *Ricardo Martinez.-*

5 9

accounting problem
by Parikhita Behera 31 Mar '11

31 Mar '11

Hi , Currently i am working with opensips and i am able to call throgh opensips and show the cdrs but there is problem of accounting that how i can give the balance to the customer and how to charge the rates for diffrent country. There is the option call control for opensips and this is related to media proxy and radius server. I am using centos OS for opensips .Please help me if any body know this options(call control). Waiting for your response. Thanks and Regards Parikhita

2 1

Re: [SR-Users] Outbound socket selection (HA implementation)
by Sean O'Donnell 31 Mar '11

31 Mar '11

For what it's worth, make sure you have the mhomed patch in forward.c. As I recall, the 3.1.2 source tarball did not contain the fix, but the latest forward.c did. http://www.mail-archive.com/sr-users@lists.sip-router.org/msg03892.html Sean O'Donnell Senior Engineer uReach Technologies, Inc. ---- On Wed, 30 Mar 2011, Bruce McAlister (bruce.mcalister(a)blueface.ie) wrote: All, Thanks for the tips and suggestions, I have now managed to get this working using mhomed=1 and setting $fs, for some reason I could not get "force_send_socket" to work with avp's (but thats a topic for another thread). Thanks again for your time. _______________________________________________ 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

Outbound socket selection (HA implementation)
by Asgaroth 31 Mar '11

31 Mar '11

Hi All, I'm trying to setup an active/passive proxy instance with kamailio 3.1.2. I'm using ucarp (www.ucarp.org) to provide the floating ip addresses. I have installed and configured ucarp and it appears to be working as expected. I have configured kamailio on both systems to be online all the time (using the net.ipv4.ip_nonlocal_bind kernel parameter on the "standby" server). I have configured kamailio with the listen parameter to be listening on the floating ip addresses only. The problem now is that kamailio uses the inbound interface as the outbound socket for messages (I think this is the default, expected, behaviour). If I enable the "mhomed" parameter then I get the following error: ERROR: <core> [forward.c:218]: ERROR: get_out_socket: no socket found ERROR: <core> [forward.c:572]: forward_req: ERROR: cannot forward to af 2, proto 1 no corresponding listening socket ERROR: sl [sl_funcs.c:282]: ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (7/SL) I had a look around on the docuwiki and I found $fs. Would I need to force the outbound socket for each message in this case, or do I have a simple comfiguration issue? Any thoughts/comments would be greatly appreciated. Thanks

6 12

About tm retrans time on ser or kamailio 3.1?
by Min Wang 30 Mar '11

30 Mar '11

HI I did the trace using ngrep: ngrep -T -W byline -d any port 5060 And I am confused with tm retran. (1) case one tm module, the| retr_timer1| default is 500 ms. #U +0.001824 xxx.17:5060 -> xxx.16:5060 INVITE sip:224@172.16.8.49:2057;line=lnzlkxu5 SIP/2.0. #U +0.000012 xxx.17:5060 -> xxx.16:5060 INVITE sip:224@172.16.8.49:2057;line=lnzlkxu5 SIP/2.0. the ngrep show the delta between packet, the second invite is only 12 ms behind? is my understanding correct? (2) change retr_timer1 to 2000 ms modparam("tm", "retr_timer1", 1000) #U +0.001936 xxx.17:5060 -> xxx.16:5060 INVITE sip:224@172.16.8.49:2057;line=lnzlkxu5 SIP/2.0. #U +0.000013 xxx.17:5060 -> xxx.16:5060 INVITE sip:224@172.16.8.49:2057;line=lnzlkxu5 SIP/2.0. the delta still 13 ms behind, so similar to the case 1, The question is: is the second invite the re-trans by tm? If so why it is around 12/13 ms? not the 500 ms or as configured 2000 ms? thanks. Kind regards / Mit freundlichen Grüßen Min Wang

4 5

SIP Communicator renamed to Jitsi
by Daniel-Constantin Mierla 30 Mar '11

30 Mar '11

Hello, I thought it is an useful information to announce here that the SIP Communicator SIP softphone was renamed to Jitsi. There are several pages mentioning Kamailio and SIP Communicator out there, among them our participation to Google Summer of Code 2010 as well as the full SIP SIMPLE Presence configuration with XCAP: * http://asipto.com/u/sp To fetch the latest version of this sip softphone you just go to: * http://www.jitsi.org Btw, Kamailio is participating again in GSoC, the 2011 edition, together with Jitsi project -- I will make another message regarding this subject. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

4 7

any one use multicast for paging ?
by Ming Liu 29 Mar '11

29 Mar '11

Hi, Need help on paging case. The via header has maddr field contains multicast ip address, how does this configure ? And how does sip proxy server react ? Advice would be appreciated. Ming

2 2

Handling Cancel with rewritten To: and From: headers
by Jeremya 29 Mar '11

29 Mar '11

Hi, I have a kamailio 3.1.0 system. I am using mostly the kamailio.cfg script that came with that release but I've made a couple of changes - The pstn.gw_ip bit doesn't work - syntax error - so I've substituted fixed text for now - I've expanded the PSTN section to IN_PSTN and OUT_PSTN to handle outgoing PSTN calls and route PSTN incoming calls to the correct phones. - As (strange) requirement of my SIP trunk provider I've had to rewrite the To: and From: headers on new outgoing PSTN calls The issue I have is with CANCEL. I'm using the default CANCEL handling stanza in kamailio.cfg # CANCEL processing if ( is_method ( "CANCEL" ) ) { if ( t_check_trans() ) t_relay(); exit; } What I see in wireshark is that the From: header is the same as my rewritten version, but the To: header is as provided by the UA This doesn't cause any real problems but I'd like to know what the rules are for kamailio rewriting a CANCEL message - and if required - whether I can rewrite a header in the CANCEL section and not have it 'adjusted' by the internal t_check_trans and t_relay logic? Thanks Jeremy

1 0

Problem with rtpproxy
by Linux Guy 28 Mar '11

28 Mar '11

What could I have done wrong ? I am not able to have kamailio see rtpproxy. I am using kamailio 3.1.2 and rtpproxy 1.2 Rtpproxy is started with command : /usr/local/bin/rtpproxy -l <public ip> -s udp:localhost:7722 -u rtpproxy Module parameter : modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") Error : INFO: mi_datagram [mi_datagram.c:318]: a new child 0/4170 ERROR: rtpproxy [rtpproxy.c:1517]: timeout waiting reply from a RTP proxy ERROR: rtpproxy [rtpproxy.c:1526]: proxy <udp:public ip:7722> does not respond, disable it WARNING: rtpproxy [rtpproxy.c:1383]: can't get version of the RTP proxy WARNING: rtpproxy [rtpproxy.c:1420]: support for RTP proxy <udp:public ip:7722> has been disabled temporarily Regards, Linux Guy..

4 4

Failure route triggers a 500 internal server error - take 2
by Miguel Baptista 28 Mar '11

28 Mar '11

Hi all, A couple of days ago I sent an identical email but because I wasn't a member of this list I don't know if you got it or not. Additionally, I would like to say that I have the same problem with kamailio 3.1.1. This is very weird problem .. I am using kamailio 3.1.2 (like I say the same is happening with 3.1.1) and I have an issue with failure route. Every-time kamailio enters the failure route an internal error is triggered and kamailio sends a 500 SIP error message: /SIP/2.0 500 I'm terribly sorry, server error occurred (1/TM)/. Just for testing purpose I "downgraded" my failure route configuration to a very simple one. But even so I am always getting the same error and it doesn't matter if it is a busy, a cancel, a moved temporarily... If I don't arm the failure route I don't get any error. Any idea what might be wrong? Is this a bug? This is the relevant part of my configuration : */route[RELAY] {/* / if (is_method("INVITE")) {/ / t_on_failure("FAILTWO");/ / }/ / if (!t_relay()) {/ / sl_reply_error();/ / }/ / exit;/ /}/ /*failure_route[FAILTWO] {* xlog("L_INFO", "FAILUREFROMEXTERNAL: Entered\n"); if (t_check_status("302")) { xlog("L_INFO", "FAILUREFROMEXTERNAL: 302 detected\n"); } else { xlog("L_INFO", "FAILUREFROMEXTERNAL: other detected\n"); } xlog("L_INFO", "FAILUREFROMEXTERNAL: Exiting\n");/ } Here is the Debug output from a busy message got by kamailio and the internal error generated: /Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:366]: qm_malloc(0x82bcfe0, 860) called from <core>: receive.c: receive_msg(120) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:406]: qm_malloc(0x82bcfe0, 860) returns address 0x837e300 frag. 0x837e2e8 (size=860) on 1 -th hit Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: DEBUG: <core> [parser/msg_parser.c:640]: SIP Reply (status): Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: DEBUG: <core> [parser/msg_parser.c:642]: version: <SIP/2.0> Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: DEBUG: <core> [parser/msg_parser.c:644]: status: <486> Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: DEBUG: <core> [parser/msg_parser.c:646]: reason: <Busy Here> Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:366]: qm_malloc(0x82bcfe0, 32) called from <core>: parser/msg_parser.c: parse_headers(341) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:406]: qm_malloc(0x82bcfe0, 32) returns address 0x837f29c frag. 0x837f284 (size=32) on 1 -th hit Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:366]: qm_malloc(0x82bcfe0, 116) called from <core>: parser/msg_parser.c: get_hdr_field(130) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:406]: qm_malloc(0x82bcfe0, 116) returns address 0x837e68c frag. 0x837e674 (size=116) on 1 -th hit Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:366]: qm_malloc(0x82bcfe0, 32) called from <core>: parser/parse_via.c: parse_via(2168) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:406]: qm_malloc(0x82bcfe0, 32) returns address 0x837f2ec frag. 0x837f2d4 (size=32) on 1 -th hit Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: DEBUG: <core> [parser/parse_via.c:1287]: Found param type 232, <branch> = <z9hG4bK54f1.2f69b061.0>; state=16 Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: DEBUG: <core> [parser/parse_via.c:2300]: end of header reached, state=5 Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: DEBUG: <core> [parser/msg_parser.c:515]: parse_headers: Via found, flags=2 Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: DEBUG: <core> [parser/msg_parser.c:517]: parse_headers: this is the first via Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: DEBUG: <core> [receive.c:145]: After parse_msg... Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: DEBUG: tm [t_lookup.c:1081]: DEBUG: t_check_msg: msg id=19 global id=18 T start=0xffffffff Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:366]: qm_malloc(0x82bcfe0, 32) called from <core>: parser/msg_parser.c: parse_headers(341) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:406]: qm_malloc(0x82bcfe0, 32) returns address 0x837f33c frag. 0x837f324 (size=32) on 1 -th hit Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:366]: qm_malloc(0x82bcfe0, 116) called from <core>: parser/msg_parser.c: get_hdr_field(130) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:406]: qm_malloc(0x82bcfe0, 116) returns address 0x837e820 frag. 0x837e808 (size=224) on 1 -th hit Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:366]: qm_malloc(0x82bcfe0, 32) called from <core>: parser/parse_via.c: parse_via(2168) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:406]: qm_malloc(0x82bcfe0, 32) returns address 0x837f38c frag. 0x837f374 (size=32) on 1 -th hit Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: DEBUG: <core> [parser/parse_via.c:1287]: Found param type 232, <branch> = <z9hG4bK1f04fb19>; state=6 Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:366]: qm_malloc(0x82bcfe0, 32) called from <core>: parser/parse_via.c: parse_via(2168) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:406]: qm_malloc(0x82bcfe0, 32) returns address 0x837f6e0 frag. 0x837f6c8 (size=32) on 1 -th hit Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: DEBUG: <core> [parser/parse_via.c:1287]: Found param type 235, <rport> = <5060>; state=16 Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: DEBUG: <core> [parser/parse_via.c:2300]: end of header reached, state=5 Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: DEBUG: <core> [parser/msg_parser.c:515]: parse_headers: Via found, flags=22 Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: DEBUG: <core> [parser/msg_parser.c:528]: parse_headers: this is the second via Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:366]: qm_malloc(0x82bcfe0, 32) called from <core>: parser/msg_parser.c: parse_headers(341) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:406]: qm_malloc(0x82bcfe0, 32) returns address 0x837f730 frag. 0x837f718 (size=32) on 1 -th hit Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:366]: qm_malloc(0x82bcfe0, 32) called from <core>: parser/msg_parser.c: parse_headers(341) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:406]: qm_malloc(0x82bcfe0, 32) returns address 0x837f820 frag. 0x837f808 (size=32) on 1 -th hit Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:366]: qm_malloc(0x82bcfe0, 224) called from <core>: parser/msg_parser.c: get_hdr_field(170) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:406]: qm_malloc(0x82bcfe0, 224) returns address 0x837e9c8 frag. 0x837e9b0 (size=224) on 1 -th hit Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:366]: qm_malloc(0x82bcfe0, 24) called from <core>: parser/parse_to.c: parse_to_param(283) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:406]: qm_malloc(0x82bcfe0, 24) returns address 0x837e930 frag. 0x837e918 (size=24) on 1 -th hit Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: DEBUG: <core> [parser/parse_to.c:174]: DEBUG: add_param: tag=i76l2xu48i Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: DEBUG: <core> [parser/parse_to.c:803]: end of header reached, state=29 Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: DEBUG: <core> [parser/msg_parser.c:187]: DEBUG: get_hdr_field: <To> [41]; uri=[sip:+4773411459@umb.no] Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: DEBUG: <core> [parser/msg_parser.c:189]: DEBUG: to body [<sip:+4773411459@umb.no>] Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:366]: qm_malloc(0x82bcfe0, 32) called from <core>: parser/msg_parser.c: parse_headers(341) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:406]: qm_malloc(0x82bcfe0, 32) returns address 0x837f780 frag. 0x837f768 (size=32) on 1 -th hit Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:366]: qm_malloc(0x82bcfe0, 32) called from <core>: parser/msg_parser.c: parse_headers(341) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:406]: qm_malloc(0x82bcfe0, 32) returns address 0x837f7d0 frag. 0x837f7b8 (size=32) on 1 -th hit Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:366]: qm_malloc(0x82bcfe0, 24) called from <core>: parser/msg_parser.c: get_hdr_field(149) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:406]: qm_malloc(0x82bcfe0, 24) returns address 0x837eb78 frag. 0x837eb60 (size=24) on 1 -th hit Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: DEBUG: <core> [parser/msg_parser.c:167]: get_hdr_field: cseq <CSeq>: <102> <INVITE> Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: DEBUG: tm [t_lookup.c:967]: DEBUG: t_reply_matching: hash 8005 label 369858290 branch 0 Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: DEBUG: tm [t_lookup.c:1018]: DEBUG: t_reply_matching: reply matched (T=0xb59ec41c)! Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: DEBUG: tm [t_lookup.c:1150]: DEBUG: t_check_msg: msg id=19 global id=19 T end=0xb59ec41c Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: DEBUG: tm [t_reply.c:2014]: DEBUG: reply_received: org. status uas=180, uac[0]=180 local=0 is_invite=1) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:366]: qm_malloc(0xb5881000, 1010) called from tm: t_msgbuilder.c: build_local_reparse(368) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:406]: qm_malloc(0xb5881000, 1012) returns address 0xb59ed5a0 frag. 0xb59ed588 (size=1012) on 1 -th hit Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:428]: qm_free(0xb5881000, 0xb59ed5a0), called from tm: t_reply.c: reply_received(2064) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:450]: qm_free: freeing frag. 0xb59ed588 alloc'ed from tm: t_msgbuilder.c: build_local_reparse(368) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: DEBUG: tm [t_reply.c:1129]: ->>>>>>>>> T_code=180, new_code=486 Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:366]: qm_malloc(0x82bcfe0, 43) called from tm: t_reply.c: fake_req(854) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:406]: qm_malloc(0x82bcfe0, 44) returns address 0x837f8c0 frag. 0x837f8a8 (size=44) on 1 -th hit Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: INFO: <script>: FAILUREFROMEXTERNAL: Entered Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: DEBUG: tm [t_lookup.c:1081]: DEBUG: t_check_msg: msg id=16 global id=16 T start=0xb59ec41c Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: DEBUG: tm [t_lookup.c:1153]: DEBUG: t_check_msg: T already found! Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: DEBUG: tm [tm.c:967]: DEBUG:t_check_status: checked status is <486> Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: INFO: <script>: FAILUREFROMEXTERNAL: other detected Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: INFO: <script>: FAILUREFROMEXTERNAL: Exiting Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:428]: qm_free(0x82bcfe0, 0x837f8c0), called from tm: t_reply.c: free_faked_req(885) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:450]: qm_free: freeing frag. 0x837f8a8 alloc'ed from tm: t_reply.c: fake_req(854) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: DEBUG: <core> [msg_translator.c:204]: check_via_address(158.36.141.13, 158.36.141.13, 0) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:366]: qm_malloc(0x82bcfe0, 12) called from <core>: msg_translator.c: rport_builder(390) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:406]: qm_malloc(0x82bcfe0, 12) returns address 0x837f008 frag. 0x837eff0 (size=12) on 1 -th hit Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:366]: qm_malloc(0x82bcfe0, 392) called from <core>: msg_translator.c: build_res_buf_from_sip_req(2032) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:406]: qm_malloc(0x82bcfe0, 392) returns address 0x838007c frag. 0x8380064 (size=564) on 1 -th hit Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:428]: qm_free(0x82bcfe0, 0x837f008), called from <core>: msg_translator.c: build_res_buf_from_sip_req(2192) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:450]: qm_free: freeing frag. 0x837eff0 alloc'ed from <core>: msg_translator.c: rport_builder(390) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:428]: qm_free(0xb5881000, 0xb59ed29c), called from <core>: mem/shm_mem.c: sh_realloc(82) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:450]: qm_free: freeing frag. 0xb59ed284 alloc'ed from <core>: mem/shm_mem.c: sh_realloc(83) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:366]: qm_malloc(0xb5881000, 391) called from <core>: mem/shm_mem.c: sh_realloc(83) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:406]: qm_malloc(0xb5881000, 392) returns address 0xb59ea720 frag. 0xb59ea708 (size=500) on 1 -th hit Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: DEBUG: tm [t_reply.c:1470]: DEBUG: cleanup_uac_timers: RETR/FR timers reset Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: DEBUG: tm [t_reply.c:659]: DEBUG: reply sent out. buf=0x838007c: SIP/2.0 500 I'm terr..., shmem=0xb59ea720: SIP/2.0 500 I'm terr Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:428]: qm_free(0x82bcfe0, 0x838007c), called from tm: t_reply.c: _reply_light(668) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15015]: <core> [mem/q_malloc.c:366]: qm_malloc(0x82bcfe0, 860) called from <core>: receive.c: receive_msg(120) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:450]: qm_free: freeing frag. 0x8380064 alloc'ed from <core>: msg_translator.c: build_res_buf_from_sip_req(2032) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15015]: <core> [mem/q_malloc.c:406]: qm_malloc(0x82bcfe0, 860) returns address 0x837e300 frag. 0x837e2e8 (size=860) on 1 -th hit Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: DEBUG: tm [t_reply.c:669]: DEBUG: _reply_light: finished Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15015]: DEBUG: <core> [parser/msg_parser.c:630]: SIP Request: Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: DEBUG: tm [t_reply.c:1631]: DEBUG: relay_reply: branch=0, save=0, relay=-1 Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15015]: DEBUG: <core> [parser/msg_parser.c:632]: method: <ACK> Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: DEBUG: tm [t_reply.c:1470]: DEBUG: cleanup_uac_timers: RETR/FR timers reset Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15015]: DEBUG: <core> [parser/msg_parser.c:634]: uri: <sip:+4773411459@umb.no> Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: DEBUG: <core> [usr_avp.c:646]: DEBUG:destroy_avp_list: destroying list (nil) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15015]: DEBUG: <core> [parser/msg_parser.c:636]: version: <SIP/2.0> Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: DEBUG: <core> [usr_avp.c:646]: DEBUG:destroy_avp_list: destroying list (nil) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15015]: <core> [mem/q_malloc.c:366]: qm_malloc(0x82bcfe0, 32) called from <core>: parser/msg_parser.c: parse_headers(341) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: DEBUG: <core> [usr_avp.c:646]: DEBUG:destroy_avp_list: destroying list (nil) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15015]: <core> [mem/q_malloc.c:406]: qm_malloc(0x82bcfe0, 32) returns address 0x837ed00 frag. 0x837ece8 (size=32) on 1 -th hit Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: DEBUG: <core> [usr_avp.c:646]: DEBUG:destroy_avp_list: destroying list (nil) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15015]: <core> [mem/q_malloc.c:366]: qm_malloc(0x82bcfe0, 116) called from <core>: parser/msg_parser.c: get_hdr_field(130) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: DEBUG: <core> [usr_avp.c:646]: DEBUG:destroy_avp_list: destroying list (nil) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15015]: <core> [mem/q_malloc.c:406]: qm_malloc(0x82bcfe0, 116) returns address 0x837e68c frag. 0x837e674 (size=116) on 1 -th hit Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: DEBUG: <core> [usr_avp.c:646]: DEBUG:destroy_avp_list: destroying list (nil) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15015]: <core> [mem/q_malloc.c:366]: qm_malloc(0x82bcfe0, 32) called from <core>: parser/parse_via.c: parse_via(2168) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: DEBUG: <core> [receive.c:289]: receive_msg: cleaning up Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15015]: <core> [mem/q_malloc.c:406]: qm_malloc(0x82bcfe0, 32) returns address 0x837ecb0 frag. 0x837ec98 (size=32) on 1 -th hit Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:428]: qm_free(0x82bcfe0, 0x837f2ec), called from <core>: parser/parse_via.c: free_via_param_list(2387) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15015]: DEBUG: <core> [parser/parse_via.c:1287]: Found param type 232, <branch> = <z9hG4bK1f04fb19>; state=6 Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:450]: qm_free: freeing frag. 0x837f2d4 alloc'ed from <core>: parser/parse_via.c: parse_via(2168) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15015]: <core> [mem/q_malloc.c:366]: qm_malloc(0x82bcfe0, 32) called from <core>: parser/parse_via.c: parse_via(2168) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:428]: qm_free(0x82bcfe0, 0x837e68c), called from <core>: parser/parse_via.c: free_via_list(2399) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15015]: <core> [mem/q_malloc.c:406]: qm_malloc(0x82bcfe0, 32) returns address 0x837ec60 frag. 0x837ec48 (size=32) on 1 -th hit Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:450]: qm_free: freeing frag. 0x837e674 alloc'ed from <core>: parser/msg_parser.c: get_hdr_field(130) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15015]: DEBUG: <core> [parser/parse_via.c:1287]: Found param type 235, <rport> = <n/a>; state=17 Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:428]: qm_free(0x82bcfe0, 0x837f29c), called from <core>: parser/hf.c: free_hdr_field_lst(251) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15014]: <core> [mem/q_malloc.c:366]: qm_malloc(0x82bcfe0, 860) called from <core>: receive.c: receive_msg(120) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15015]: DEBUG: <core> [parser/parse_via.c:2300]: end of header reached, state=5 Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:450]: qm_free: freeing frag. 0x837f284 alloc'ed from <core>: parser/msg_parser.c: parse_headers(341) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15014]: <core> [mem/q_malloc.c:406]: qm_malloc(0x82bcfe0, 860) returns address 0x8380078 frag. 0x8380060 (size=860) on 1 -th hit Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15015]: DEBUG: <core> [parser/msg_parser.c:515]: parse_headers: Via found, flags=2 Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:428]: qm_free(0x82bcfe0, 0x837f38c), called from <core>: parser/parse_via.c: free_via_param_list(2387) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15014]: DEBUG: <core> [parser/msg_parser.c:640]: SIP Reply (status): Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15015]: DEBUG: <core> [parser/msg_parser.c:517]: parse_headers: this is the first via Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:450]: qm_free: freeing frag. 0x837f374 alloc'ed from <core>: parser/parse_via.c: parse_via(2168) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15014]: DEBUG: <core> [parser/msg_parser.c:642]: version: <SIP/2.0> Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15015]: DEBUG: <core> [receive.c:145]: After parse_msg... Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:428]: qm_free(0x82bcfe0, 0x837f6e0), called from <core>: parser/parse_via.c: free_via_param_list(2387) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15014]: DEBUG: <core> [parser/msg_parser.c:644]: status: <486> Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15015]: DEBUG: <core> [receive.c:186]: preparing to run routing scripts... Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15016]: <core> [mem/q_malloc.c:450]: qm_free: freeing frag. 0x837f6c8 alloc'ed from <core>: parser/parse_via.c: parse_via(2168) Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15014]: DEBUG: <core> [parser/msg_parser.c:646]: reason: <Busy Here> Mar 22 13:48:57 sip /usr/ipbx/kamailio/sbin/kamailio[15015]: DEBUG: sl [sl_funcs.c:307]: DEBUG : sl_filter_ACK: to late to be a local ACK! / Regards, Miguel Baptista

2 3

Newbie from asterisk
by Sameer Khan 28 Mar '11

28 Mar '11

Hi all I am looking forward to move from asterisk to kamailio, So may be asking some unusual questions My current system is working in such a way that Whenever a call arrives at asterisk i call perl script by AGI and then in that perl script i check customer status, billing, extension routing, updating cdr to my database and many more other operations. The billing which i am doing is different than the one in siremes and i dont want to go for GUI as well So I am looking for a solution similar to AGi. Please direct me to the right path. thanks Sameer.

4 8

problem with dialog module
by Linux Guy 28 Mar '11

28 Mar '11

Dears, I have been trying to use the dialog module to track active calls. I was able to see the active dialogs in database before I implemented IP authentication, now it is not saving the active dialogs to the database. Please provide me some direction. Module parameters : modparam("dialog", "db_url", DBURL) modparam("dialog", "db_mode", 1) modparam("dialog", "dlg_flag", 4) #modparam("dialog", "rr_param", "xyz") modparam("dialog", "dlg_match_mode", 2) modparam("dialog", "timeout_avp","$avp(i:5)") Routing Logic: route(REQINIT); route(NAT); route(WITHINDLG); if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); 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 dlg_manage(); setflag(4); } route(PEER); In route(PEER) section, it checks for ip address in the address table and then do the routing based on lcr module. Thank you Linux Guy...

2 1

LCR module : same IP address for different prefix.
by Ricardo Martinez 28 Mar '11

28 Mar '11

Hello. We’re using the LCR module with version 3.1.1 of Kamailio. We’re trying to add two gateways, with the same IP address, we have this configuration running in the previous version of Kamailio (v 1.5), without any problem, but in this version seems to have some issues. This is what I’m trying to do : mysql> select * from lcr_gw; +----+--------+---------+---------------+----------+------+--------+------------+-----------+-------+------+-------+---------+ | id | lcr_id | gw_name | ip_addr | hostname | port | params | uri_scheme | transport | strip | tag | flags | defunct | +----+--------+---------+---------------+----------+------+--------+------------+-----------+-------+------+-------+---------+ | 1 | 1 | GW_1 | 10.0.0.209 | NULL | 5060 | NULL | 1 | 1 | 3 | 303 | 1 | NULL | | 8 | 1 | GW_33 | 10.0.0.209 | NULL | 5060 | NULL | 1 | 1 | 3 | 4411 | 1 | NULL | | 6 | 1 | GW_6 | 10.0.0.212 | NULL | 5060 | NULL | 1 | 1 | 5 | 308 | 1 | NULL | +----+--------+---------+---------------+----------+------+--------+------------+-----------+-------+------+-------+---------+ And this is the output from the logs : /usr/local/sbin/kamailio[2829]: ERROR: lcr [lcr_mod.c:818]: gw <10.0.0.209, 5060, > is not unique What could be the problem?. Os simply the new LCR does not support have two gateways with the same IP address? Thanks, Ricardo Martinez.-

3 9

Fwd: testing kamailio
by michal javorka 28 Mar '11

28 Mar '11

---------- Forwarded message ---------- From: michal javorka <kostriak(a)gmail.com> Date: 2011/3/28 Subject: testing kamailio To: sipp-users(a)lists.sourceforge.net Hi , i am trying to test kamailio 3.1 with sipp testing tool, my topology looks like this: sipp_UAC 158.193.139.120 proxy Kamailio 158.193.139.91 sippsr-users @ lists.sip-router.orgsr-users @ lists.sip-router.org_UAS 158.193.139.107 i am using no authentication and so i just kamctl ul add sipp sip:sipp@158.193.139.120:5060 kamctl ul add 1000 sip:1000@158.193.139.107:5060 commands, so far so good but when i wanted to test it there came troubles on uac i used ./sipp -sn uac 158.193.139.91 -i 158.193.139.120 -s 1000 -r 1 -p 5060 on uas ./sipp -sn uas -i 158.193.139.107 -p 5060 everything looks right but no ACK came from kamailio proxy and i dont know why?? any suggestions? it is part of my bachelor work and i cant move. here is some ngrep from server: interface: eth0 (158.193.139.0/255.255.255.0) filter: (ip or ip6) and ( port 5060 ) U 158.193.139.120:5060 -> 158.193.139.91:5060 INVITE sip:1000@158.193.139.91:5060 SIP/2.0. Via: SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-1-0. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag001. To: sut <sip:1000@158.193.139.91:5060>. Call-ID: 1-636(a)158.193.139.120. CSeq: 1 INVITE. Contact: sip:sipp@158.193.139.120:5060. Max-Forwards: 70. Subject: Performance Test. Content-Type: application/sdp. Content-Length: 139. . v=0. o=user1 53655765 2353687637 IN IP4 158.193.139.120. s=-. c=IN IP4 158.193.139.120. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000 U 158.193.139.91:5060 -> 158.193.139.120:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-1-0. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag001. To: sut <sip:1000@158.193.139.91:5060>. Call-ID: 1-636(a)158.193.139.120. CSeq: 1 INVITE. Server: kamailio (3.1.1 (i386/linux)). Content-Length: 0. . U 158.193.139.91:5060 -> 158.193.139.107:5060 INVITE sip:1000@158.193.139.107:5060 SIP/2.0. Record-Route: <sip:158.193.139.91;lr=on>. Via: SIP/2.0/UDP 158.193.139.91;branch=z9hG4bKa21c.7d2fd135.0. Via: SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-1-0. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag001. To: sut <sip:1000@158.193.139.91:5060>. Call-ID: 1-636(a)158.193.139.120. CSeq: 1 INVITE. Contact: sip:sipp@158.193.139.120:5060. Max-Forwards: 69. Subject: Performance Test. Content-Type: application/sdp. Content-Length: 139. . v=0. o=user1 53655765 2353687637 IN IP4 158.193.139.120. s=-. c=IN IP4 158.193.139.120. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000 U 158.193.139.107:5060 -> 158.193.139.91:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 158.193.139.91;branch=z9hG4bKa21c.7d2fd135.0, SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-1-0. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag001. To: sut <sip:1000@158.193.139.91:5060>;tag=2180SIPpTag019. Call-ID: 1-636(a)158.193.139.120. CSeq: 1 INVITE. Contact: <sip:158.193.139.107:5060;transport=UDP>. Content-Length: 0. . U 158.193.139.107:5060 -> 158.193.139.91:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 158.193.139.91;branch=z9hG4bKa21c.7d2fd135.0, SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-1-0. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag001. To: sut <sip:1000@158.193.139.91:5060>;tag=2180SIPpTag019. Call-ID: 1-636(a)158.193.139.120. CSeq: 1 INVITE. Contact: <sip:158.193.139.107:5060;transport=UDP>. Content-Type: application/sdp. Content-Length: 139. . v=0. o=user1 53655765 2353687637 IN IP4 158.193.139.107. s=-. c=IN IP4 158.193.139.107. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000 U 158.193.139.91:5060 -> 158.193.139.120:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-1-0. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag001. To: sut <sip:1000@158.193.139.91:5060>;tag=2180SIPpTag019. Call-ID: 1-636(a)158.193.139.120. CSeq: 1 INVITE. Contact: <sip:158.193.139.107:5060;transport=UDP>. Content-Length: 0. . U 158.193.139.91:5060 -> 158.193.139.120:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-1-0. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag001. To: sut <sip:1000@158.193.139.91:5060>;tag=2180SIPpTag019. Call-ID: 1-636(a)158.193.139.120. CSeq: 1 INVITE. Contact: <sip:158.193.139.107:5060;transport=UDP>. Content-Type: application/sdp. Content-Length: 139. . v=0. o=user1 53655765 2353687637 IN IP4 158.193.139.107. s=-. c=IN IP4 158.193.139.107. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000 U 158.193.139.120:5060 -> 158.193.139.91:5060 ACK sip:1000@158.193.139.91:5060 SIP/2.0. Via: SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-1-5. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag001. To: sut <sip:1000@158.193.139.91:5060>;tag=2180SIPpTag019. Call-ID: 1-636(a)158.193.139.120. CSeq: 1 ACK. Contact: sip:sipp@158.193.139.120:5060. Max-Forwards: 70. Subject: Performance Test. Content-Length: 0. . U 158.193.139.120:5060 -> 158.193.139.91:5060 BYE sip:1000@158.193.139.91:5060 SIP/2.0. Via: SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-1-7. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag001. To: sut <sip:1000@158.193.139.91:5060>;tag=2180SIPpTag019. Call-ID: 1-636(a)158.193.139.120. CSeq: 2 BYE. Contact: sip:sipp@158.193.139.120:5060. Max-Forwards: 70. Subject: Performance Test. Content-Length: 0. . U 158.193.139.91:5060 -> 158.193.139.120:5060 SIP/2.0 404 Not Here. Via: SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-1-7. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag001. To: sut <sip:1000@158.193.139.91:5060>;tag=2180SIPpTag019. Call-ID: 1-636(a)158.193.139.120. CSeq: 2 BYE. Server: kamailio (3.1.1 (i386/linux)). Content-Length: 0. . U 158.193.139.120:5060 -> 158.193.139.91:5060 BYE sip:1000@158.193.139.91:5060 SIP/2.0. Via: SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-1--1. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag001. To: sut <sip:1000@158.193.139.91:5060>;tag=2180SIPpTag019. Call-ID: 1-636(a)158.193.139.120. CSeq: 3 BYE. Max-Forwards: 70. Contact: <sip:sipp@158.193.139.120:5060;transport=UDP>. Content-Length: 0. . U 158.193.139.91:5060 -> 158.193.139.120:5060 SIP/2.0 404 Not Here. Via: SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-1--1. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag001. To: sut <sip:1000@158.193.139.91:5060>;tag=2180SIPpTag019. Call-ID: 1-636(a)158.193.139.120. CSeq: 3 BYE. Server: kamailio (3.1.1 (i386/linux)). Content-Length: 0. . U 158.193.139.107:5060 -> 158.193.139.91:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 158.193.139.91;branch=z9hG4bKa21c.7d2fd135.0, SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-1-0. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag001. To: sut <sip:1000@158.193.139.91:5060>;tag=2180SIPpTag019. Call-ID: 1-636(a)158.193.139.120. CSeq: 1 INVITE. Contact: <sip:158.193.139.107:5060;transport=UDP>. Content-Type: application/sdp. Content-Length: 139. . v=0. o=user1 53655765 2353687637 IN IP4 158.193.139.107. s=-. c=IN IP4 158.193.139.107. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000 U 158.193.139.91:5060 -> 158.193.139.120:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-1-0. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag001. To: sut <sip:1000@158.193.139.91:5060>;tag=2180SIPpTag019. Call-ID: 1-636(a)158.193.139.120. CSeq: 1 INVITE. Contact: <sip:158.193.139.107:5060;transport=UDP>. Content-Type: application/sdp. Content-Length: 139. . v=0. o=user1 53655765 2353687637 IN IP4 158.193.139.107. s=-. c=IN IP4 158.193.139.107. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000 U 158.193.139.120:5060 -> 158.193.139.91:5060 INVITE sip:1000@158.193.139.91:5060 SIP/2.0. Via: SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-2-0. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag002. To: sut <sip:1000@158.193.139.91:5060>. Call-ID: 2-636(a)158.193.139.120. CSeq: 1 INVITE. Contact: sip:sipp@158.193.139.120:5060. Max-Forwards: 70. Subject: Performance Test. Content-Type: application/sdp. Content-Length: 139. . v=0. o=user1 53655765 2353687637 IN IP4 158.193.139.120. s=-. c=IN IP4 158.193.139.120. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000 U 158.193.139.91:5060 -> 158.193.139.120:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-2-0. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag002. To: sut <sip:1000@158.193.139.91:5060>. Call-ID: 2-636(a)158.193.139.120. CSeq: 1 INVITE. Server: kamailio (3.1.1 (i386/linux)). Content-Length: 0. . U 158.193.139.91:5060 -> 158.193.139.107:5060 INVITE sip:1000@158.193.139.107:5060 SIP/2.0. Record-Route: <sip:158.193.139.91;lr=on>. Via: SIP/2.0/UDP 158.193.139.91;branch=z9hG4bK4285.21e4a9e7.0. Via: SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-2-0. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag002. To: sut <sip:1000@158.193.139.91:5060>. Call-ID: 2-636(a)158.193.139.120. CSeq: 1 INVITE. Contact: sip:sipp@158.193.139.120:5060. Max-Forwards: 69. Subject: Performance Test. Content-Type: application/sdp. Content-Length: 139. . v=0. o=user1 53655765 2353687637 IN IP4 158.193.139.120. s=-. c=IN IP4 158.193.139.120. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000 U 158.193.139.107:5060 -> 158.193.139.91:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 158.193.139.91;branch=z9hG4bK4285.21e4a9e7.0, SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-2-0. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag002. To: sut <sip:1000@158.193.139.91:5060>;tag=2180SIPpTag0110. Call-ID: 2-636(a)158.193.139.120. CSeq: 1 INVITE. Contact: <sip:158.193.139.107:5060;transport=UDP>. Content-Length: 0. . U 158.193.139.107:5060 -> 158.193.139.91:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 158.193.139.91;branch=z9hG4bK4285.21e4a9e7.0, SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-2-0. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag002. To: sut <sip:1000@158.193.139.91:5060>;tag=2180SIPpTag0110. Call-ID: 2-636(a)158.193.139.120. CSeq: 1 INVITE. Contact: <sip:158.193.139.107:5060;transport=UDP>. Content-Type: application/sdp. Content-Length: 139. . v=0. o=user1 53655765 2353687637 IN IP4 158.193.139.107. s=-. c=IN IP4 158.193.139.107. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000 U 158.193.139.91:5060 -> 158.193.139.120:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-2-0. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag002. To: sut <sip:1000@158.193.139.91:5060>;tag=2180SIPpTag0110. Call-ID: 2-636(a)158.193.139.120. CSeq: 1 INVITE. Contact: <sip:158.193.139.107:5060;transport=UDP>. Content-Length: 0. . U 158.193.139.91:5060 -> 158.193.139.120:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-2-0. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag002. To: sut <sip:1000@158.193.139.91:5060>;tag=2180SIPpTag0110. Call-ID: 2-636(a)158.193.139.120. CSeq: 1 INVITE. Contact: <sip:158.193.139.107:5060;transport=UDP>. Content-Type: application/sdp. Content-Length: 139. . v=0. o=user1 53655765 2353687637 IN IP4 158.193.139.107. s=-. c=IN IP4 158.193.139.107. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000 U 158.193.139.120:5060 -> 158.193.139.91:5060 ACK sip:1000@158.193.139.91:5060 SIP/2.0. Via: SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-2-5. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag002. To: sut <sip:1000@158.193.139.91:5060>;tag=2180SIPpTag0110. Call-ID: 2-636(a)158.193.139.120. CSeq: 1 ACK. Contact: sip:sipp@158.193.139.120:5060. Max-Forwards: 70. Subject: Performance Test. Content-Length: 0. . U 158.193.139.120:5060 -> 158.193.139.91:5060 BYE sip:1000@158.193.139.91:5060 SIP/2.0. Via: SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-2-7. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag002. To: sut <sip:1000@158.193.139.91:5060>;tag=2180SIPpTag0110. Call-ID: 2-636(a)158.193.139.120. CSeq: 2 BYE. Contact: sip:sipp@158.193.139.120:5060. Max-Forwards: 70. Subject: Performance Test. Content-Length: 0. . U 158.193.139.91:5060 -> 158.193.139.120:5060 SIP/2.0 404 Not Here. Via: SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-2-7. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag002. To: sut <sip:1000@158.193.139.91:5060>;tag=2180SIPpTag0110. Call-ID: 2-636(a)158.193.139.120. CSeq: 2 BYE. Server: kamailio (3.1.1 (i386/linux)). Content-Length: 0. . U 158.193.139.120:5060 -> 158.193.139.91:5060 BYE sip:1000@158.193.139.91:5060 SIP/2.0. Via: SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-2--1. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag002. To: sut <sip:1000@158.193.139.91:5060>;tag=2180SIPpTag0110. Call-ID: 2-636(a)158.193.139.120. CSeq: 3 BYE. Max-Forwards: 70. Contact: <sip:sipp@158.193.139.120:5060;transport=UDP>. Content-Length: 0. . U 158.193.139.91:5060 -> 158.193.139.120:5060 SIP/2.0 404 Not Here. Via: SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-2--1. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag002. To: sut <sip:1000@158.193.139.91:5060>;tag=2180SIPpTag0110. Call-ID: 2-636(a)158.193.139.120. CSeq: 3 BYE. Server: kamailio (3.1.1 (i386/linux)). Content-Length: 0. . U 158.193.139.107:5060 -> 158.193.139.91:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 158.193.139.91;branch=z9hG4bK4285.21e4a9e7.0, SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-2-0. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag002. To: sut <sip:1000@158.193.139.91:5060>;tag=2180SIPpTag0110. Call-ID: 2-636(a)158.193.139.120. CSeq: 1 INVITE. Contact: <sip:158.193.139.107:5060;transport=UDP>. Content-Type: application/sdp. Content-Length: 139. . v=0. o=user1 53655765 2353687637 IN IP4 158.193.139.107. s=-. c=IN IP4 158.193.139.107. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000 U 158.193.139.91:5060 -> 158.193.139.120:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-2-0. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag002. To: sut <sip:1000@158.193.139.91:5060>;tag=2180SIPpTag0110. Call-ID: 2-636(a)158.193.139.120. CSeq: 1 INVITE. Contact: <sip:158.193.139.107:5060;transport=UDP>. Content-Type: application/sdp. Content-Length: 139. . v=0. o=user1 53655765 2353687637 IN IP4 158.193.139.107. s=-. c=IN IP4 158.193.139.107. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000 U 158.193.139.107:5060 -> 158.193.139.91:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 158.193.139.91;branch=z9hG4bKa21c.7d2fd135.0, SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-1-0. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag001. To: sut <sip:1000@158.193.139.91:5060>;tag=2180SIPpTag019. Call-ID: 1-636(a)158.193.139.120. CSeq: 1 INVITE. Contact: <sip:158.193.139.107:5060;transport=UDP>. Content-Type: application/sdp. Content-Length: 139. . v=0. o=user1 53655765 2353687637 IN IP4 158.193.139.107. s=-. c=IN IP4 158.193.139.107. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000 U 158.193.139.91:5060 -> 158.193.139.120:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-1-0. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag001. To: sut <sip:1000@158.193.139.91:5060>;tag=2180SIPpTag019. Call-ID: 1-636(a)158.193.139.120. CSeq: 1 INVITE. Contact: <sip:158.193.139.107:5060;transport=UDP>. Content-Type: application/sdp. Content-Length: 139. . v=0. o=user1 53655765 2353687637 IN IP4 158.193.139.107. s=-. c=IN IP4 158.193.139.107. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000 U 158.193.139.120:5060 -> 158.193.139.91:5060 INVITE sip:1000@158.193.139.91:5060 SIP/2.0. Via: SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-3-0. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag003. To: sut <sip:1000@158.193.139.91:5060>. Call-ID: 3-636(a)158.193.139.120. CSeq: 1 INVITE. Contact: sip:sipp@158.193.139.120:5060. Max-Forwards: 70. Subject: Performance Test. Content-Type: application/sdp. Content-Length: 139. . v=0. o=user1 53655765 2353687637 IN IP4 158.193.139.120. s=-. c=IN IP4 158.193.139.120. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000 U 158.193.139.91:5060 -> 158.193.139.120:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-3-0. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag003. To: sut <sip:1000@158.193.139.91:5060>. Call-ID: 3-636(a)158.193.139.120. CSeq: 1 INVITE. Server: kamailio (3.1.1 (i386/linux)). Content-Length: 0. . U 158.193.139.91:5060 -> 158.193.139.107:5060 INVITE sip:1000@158.193.139.107:5060 SIP/2.0. Record-Route: <sip:158.193.139.91;lr=on>. Via: SIP/2.0/UDP 158.193.139.91;branch=z9hG4bK62b7.0e791b35.0. Via: SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-3-0. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag003. To: sut <sip:1000@158.193.139.91:5060>. Call-ID: 3-636(a)158.193.139.120. CSeq: 1 INVITE. Contact: sip:sipp@158.193.139.120:5060. Max-Forwards: 69. Subject: Performance Test. Content-Type: application/sdp. Content-Length: 139. . v=0. o=user1 53655765 2353687637 IN IP4 158.193.139.120. s=-. c=IN IP4 158.193.139.120. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000 U 158.193.139.107:5060 -> 158.193.139.91:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 158.193.139.91;branch=z9hG4bK62b7.0e791b35.0, SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-3-0. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag003. To: sut <sip:1000@158.193.139.91:5060>;tag=2180SIPpTag0111. Call-ID: 3-636(a)158.193.139.120. CSeq: 1 INVITE. Contact: <sip:158.193.139.107:5060;transport=UDP>. Content-Length: 0. . U 158.193.139.107:5060 -> 158.193.139.91:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 158.193.139.91;branch=z9hG4bK62b7.0e791b35.0, SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-3-0. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag003. To: sut <sip:1000@158.193.139.91:5060>;tag=2180SIPpTag0111. Call-ID: 3-636(a)158.193.139.120. CSeq: 1 INVITE. Contact: <sip:158.193.139.107:5060;transport=UDP>. Content-Type: application/sdp. Content-Length: 139. . v=0. o=user1 53655765 2353687637 IN IP4 158.193.139.107. s=-. c=IN IP4 158.193.139.107. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000 U 158.193.139.91:5060 -> 158.193.139.120:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-3-0. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag003. To: sut <sip:1000@158.193.139.91:5060>;tag=2180SIPpTag0111. Call-ID: 3-636(a)158.193.139.120. CSeq: 1 INVITE. Contact: <sip:158.193.139.107:5060;transport=UDP>. Content-Length: 0. . U 158.193.139.91:5060 -> 158.193.139.120:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-3-0. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag003. To: sut <sip:1000@158.193.139.91:5060>;tag=2180SIPpTag0111. Call-ID: 3-636(a)158.193.139.120. CSeq: 1 INVITE. Contact: <sip:158.193.139.107:5060;transport=UDP>. Content-Type: application/sdp. Content-Length: 139. . v=0. o=user1 53655765 2353687637 IN IP4 158.193.139.107. s=-. c=IN IP4 158.193.139.107. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000 U 158.193.139.120:5060 -> 158.193.139.91:5060 ACK sip:1000@158.193.139.91:5060 SIP/2.0. Via: SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-3-5. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag003. To: sut <sip:1000@158.193.139.91:5060>;tag=2180SIPpTag0111. Call-ID: 3-636(a)158.193.139.120. CSeq: 1 ACK. Contact: sip:sipp@158.193.139.120:5060. Max-Forwards: 70. Subject: Performance Test. Content-Length: 0. . U 158.193.139.120:5060 -> 158.193.139.91:5060 BYE sip:1000@158.193.139.91:5060 SIP/2.0. Via: SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-3-7. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag003. To: sut <sip:1000@158.193.139.91:5060>;tag=2180SIPpTag0111. Call-ID: 3-636(a)158.193.139.120. CSeq: 2 BYE. Contact: sip:sipp@158.193.139.120:5060. Max-Forwards: 70. Subject: Performance Test. Content-Length: 0. . U 158.193.139.91:5060 -> 158.193.139.120:5060 SIP/2.0 404 Not Here. Via: SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-3-7. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag003. To: sut <sip:1000@158.193.139.91:5060>;tag=2180SIPpTag0111. Call-ID: 3-636(a)158.193.139.120. CSeq: 2 BYE. Server: kamailio (3.1.1 (i386/linux)). Content-Length: 0. . U 158.193.139.120:5060 -> 158.193.139.91:5060 BYE sip:1000@158.193.139.91:5060 SIP/2.0. Via: SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-3--1. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag003. To: sut <sip:1000@158.193.139.91:5060>;tag=2180SIPpTag0111. Call-ID: 3-636(a)158.193.139.120. CSeq: 3 BYE. Max-Forwards: 70. Contact: <sip:sipp@158.193.139.120:5060;transport=UDP>. Content-Length: 0. . U 158.193.139.91:5060 -> 158.193.139.120:5060 SIP/2.0 404 Not Here. Via: SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-3--1. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag003. To: sut <sip:1000@158.193.139.91:5060>;tag=2180SIPpTag0111. Call-ID: 3-636(a)158.193.139.120. CSeq: 3 BYE. Server: kamailio (3.1.1 (i386/linux)). Content-Length: 0. . U 158.193.139.107:5060 -> 158.193.139.91:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 158.193.139.91;branch=z9hG4bK62b7.0e791b35.0, SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-3-0. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag003. To: sut <sip:1000@158.193.139.91:5060>;tag=2180SIPpTag0111. Call-ID: 3-636(a)158.193.139.120. CSeq: 1 INVITE. Contact: <sip:158.193.139.107:5060;transport=UDP>. Content-Type: application/sdp. Content-Length: 139. . v=0. o=user1 53655765 2353687637 IN IP4 158.193.139.107. s=-. c=IN IP4 158.193.139.107. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000 U 158.193.139.107:5060 -> 158.193.139.91:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 158.193.139.91;branch=z9hG4bK4285.21e4a9e7.0, SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-2-0. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag002. To: sut <sip:1000@158.193.139.91:5060>;tag=2180SIPpTag0110. Call-ID: 2-636(a)158.193.139.120. CSeq: 1 INVITE. Contact: <sip:158.193.139.107:5060;transport=UDP>. Content-Type: application/sdp. Content-Length: 139. . v=0. o=user1 53655765 2353687637 IN IP4 158.193.139.107. s=-. c=IN IP4 158.193.139.107. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000 U 158.193.139.91:5060 -> 158.193.139.120:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-3-0. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag003. To: sut <sip:1000@158.193.139.91:5060>;tag=2180SIPpTag0111. Call-ID: 3-636(a)158.193.139.120. CSeq: 1 INVITE. Contact: <sip:158.193.139.107:5060;transport=UDP>. Content-Type: application/sdp. Content-Length: 139. . v=0. o=user1 53655765 2353687637 IN IP4 158.193.139.107. s=-. c=IN IP4 158.193.139.107. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000 U 158.193.139.91:5060 -> 158.193.139.120:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-2-0. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag002. To: sut <sip:1000@158.193.139.91:5060>;tag=2180SIPpTag0110. Call-ID: 2-636(a)158.193.139.120. CSeq: 1 INVITE. Contact: <sip:158.193.139.107:5060;transport=UDP>. Content-Type: application/sdp. Content-Length: 139. . v=0. o=user1 53655765 2353687637 IN IP4 158.193.139.107. s=-. c=IN IP4 158.193.139.107. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000 U 158.193.139.120:5060 -> 158.193.139.91:5060 INVITE sip:1000@158.193.139.91:5060 SIP/2.0. Via: SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-4-0. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag004. To: sut <sip:1000@158.193.139.91:5060>. Call-ID: 4-636(a)158.193.139.120. CSeq: 1 INVITE. Contact: sip:sipp@158.193.139.120:5060. Max-Forwards: 70. Subject: Performance Test. Content-Type: application/sdp. Content-Length: 139. . v=0. o=user1 53655765 2353687637 IN IP4 158.193.139.120. s=-. c=IN IP4 158.193.139.120. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000 U 158.193.139.91:5060 -> 158.193.139.120:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-4-0. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag004. To: sut <sip:1000@158.193.139.91:5060>. Call-ID: 4-636(a)158.193.139.120. CSeq: 1 INVITE. Server: kamailio (3.1.1 (i386/linux)). Content-Length: 0. . U 158.193.139.91:5060 -> 158.193.139.107:5060 INVITE sip:1000@158.193.139.107:5060 SIP/2.0. Record-Route: <sip:158.193.139.91;lr=on>. Via: SIP/2.0/UDP 158.193.139.91;branch=z9hG4bK1221.c7af9402.0. Via: SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-4-0. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag004. To: sut <sip:1000@158.193.139.91:5060>. Call-ID: 4-636(a)158.193.139.120. CSeq: 1 INVITE. Contact: sip:sipp@158.193.139.120:5060. Max-Forwards: 69. Subject: Performance Test. Content-Type: application/sdp. Content-Length: 139. . v=0. o=user1 53655765 2353687637 IN IP4 158.193.139.120. s=-. c=IN IP4 158.193.139.120. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000 U 158.193.139.107:5060 -> 158.193.139.91:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 158.193.139.91;branch=z9hG4bK1221.c7af9402.0, SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-4-0. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag004. To: sut <sip:1000@158.193.139.91:5060>;tag=2180SIPpTag0112. Call-ID: 4-636(a)158.193.139.120. CSeq: 1 INVITE. Contact: <sip:158.193.139.107:5060;transport=UDP>. Content-Length: 0. . U 158.193.139.107:5060 -> 158.193.139.91:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 158.193.139.91;branch=z9hG4bK1221.c7af9402.0, SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-4-0. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag004. To: sut <sip:1000@158.193.139.91:5060>;tag=2180SIPpTag0112. Call-ID: 4-636(a)158.193.139.120. CSeq: 1 INVITE. Contact: <sip:158.193.139.107:5060;transport=UDP>. Content-Type: application/sdp. Content-Length: 139. . v=0. o=user1 53655765 2353687637 IN IP4 158.193.139.107. s=-. c=IN IP4 158.193.139.107. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000 U 158.193.139.91:5060 -> 158.193.139.120:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-4-0. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag004. To: sut <sip:1000@158.193.139.91:5060>;tag=2180SIPpTag0112. Call-ID: 4-636(a)158.193.139.120. CSeq: 1 INVITE. Contact: <sip:158.193.139.107:5060;transport=UDP>. Content-Length: 0. . U 158.193.139.91:5060 -> 158.193.139.120:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-4-0. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag004. To: sut <sip:1000@158.193.139.91:5060>;tag=2180SIPpTag0112. Call-ID: 4-636(a)158.193.139.120. CSeq: 1 INVITE. Contact: <sip:158.193.139.107:5060;transport=UDP>. Content-Type: application/sdp. Content-Length: 139. . v=0. o=user1 53655765 2353687637 IN IP4 158.193.139.107. s=-. c=IN IP4 158.193.139.107. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000 U 158.193.139.120:5060 -> 158.193.139.91:5060 ACK sip:1000@158.193.139.91:5060 SIP/2.0. Via: SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-4-5. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag004. To: sut <sip:1000@158.193.139.91:5060>;tag=2180SIPpTag0112. Call-ID: 4-636(a)158.193.139.120. CSeq: 1 ACK. Contact: sip:sipp@158.193.139.120:5060. Max-Forwards: 70. Subject: Performance Test. Content-Length: 0. . U 158.193.139.120:5060 -> 158.193.139.91:5060 BYE sip:1000@158.193.139.91:5060 SIP/2.0. Via: SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-4-7. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag004. To: sut <sip:1000@158.193.139.91:5060>;tag=2180SIPpTag0112. Call-ID: 4-636(a)158.193.139.120. CSeq: 2 BYE. Contact: sip:sipp@158.193.139.120:5060. Max-Forwards: 70. Subject: Performance Test. Content-Length: 0. . U 158.193.139.91:5060 -> 158.193.139.120:5060 SIP/2.0 404 Not Here. Via: SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-4-7. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag004. To: sut <sip:1000@158.193.139.91:5060>;tag=2180SIPpTag0112. Call-ID: 4-636(a)158.193.139.120. CSeq: 2 BYE. Server: kamailio (3.1.1 (i386/linux)). Content-Length: 0. . U 158.193.139.120:5060 -> 158.193.139.91:5060 BYE sip:1000@158.193.139.91:5060 SIP/2.0. Via: SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-4--1. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag004. To: sut <sip:1000@158.193.139.91:5060>;tag=2180SIPpTag0112. Call-ID: 4-636(a)158.193.139.120. CSeq: 3 BYE. Max-Forwards: 70. Contact: <sip:sipp@158.193.139.120:5060;transport=UDP>. Content-Length: 0. . U 158.193.139.91:5060 -> 158.193.139.120:5060 SIP/2.0 404 Not Here. Via: SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-4--1. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag004. To: sut <sip:1000@158.193.139.91:5060>;tag=2180SIPpTag0112. Call-ID: 4-636(a)158.193.139.120. CSeq: 3 BYE. Server: kamailio (3.1.1 (i386/linux)). Content-Length: 0. . U 158.193.139.107:5060 -> 158.193.139.91:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 158.193.139.91;branch=z9hG4bK1221.c7af9402.0, SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-4-0. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag004. To: sut <sip:1000@158.193.139.91:5060>;tag=2180SIPpTag0112. Call-ID: 4-636(a)158.193.139.120. CSeq: 1 INVITE. Contact: <sip:158.193.139.107:5060;transport=UDP>. Content-Type: application/sdp. Content-Length: 139. . v=0. o=user1 53655765 2353687637 IN IP4 158.193.139.107. s=-. c=IN IP4 158.193.139.107. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000 U 158.193.139.107:5060 -> 158.193.139.91:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 158.193.139.91;branch=z9hG4bK62b7.0e791b35.0, SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-3-0. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag003. To: sut <sip:1000@158.193.139.91:5060>;tag=2180SIPpTag0111. Call-ID: 3-636(a)158.193.139.120. CSeq: 1 INVITE. Contact: <sip:158.193.139.107:5060;transport=UDP>. Content-Type: application/sdp. Content-Length: 139. . v=0. o=user1 53655765 2353687637 IN IP4 158.193.139.107. s=-. c=IN IP4 158.193.139.107. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000 U 158.193.139.107:5060 -> 158.193.139.91:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 158.193.139.91;branch=z9hG4bKa21c.7d2fd135.0, SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-1-0. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag001. To: sut <sip:1000@158.193.139.91:5060>;tag=2180SIPpTag019. Call-ID: 1-636(a)158.193.139.120. CSeq: 1 INVITE. Contact: <sip:158.193.139.107:5060;transport=UDP>. Content-Type: application/sdp. Content-Length: 139. . v=0. o=user1 53655765 2353687637 IN IP4 158.193.139.107. s=-. c=IN IP4 158.193.139.107. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000 U 158.193.139.91:5060 -> 158.193.139.120:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-1-0. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag001. To: sut <sip:1000@158.193.139.91:5060>;tag=2180SIPpTag019. Call-ID: 1-636(a)158.193.139.120. CSeq: 1 INVITE. Contact: <sip:158.193.139.107:5060;transport=UDP>. Content-Type: application/sdp. Content-Length: 139. . v=0. o=user1 53655765 2353687637 IN IP4 158.193.139.107. s=-. c=IN IP4 158.193.139.107. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000 U 158.193.139.91:5060 -> 158.193.139.120:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-4-0. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag004. To: sut <sip:1000@158.193.139.91:5060>;tag=2180SIPpTag0112. Call-ID: 4-636(a)158.193.139.120. CSeq: 1 INVITE. Contact: <sip:158.193.139.107:5060;transport=UDP>. Content-Type: application/sdp. Content-Length: 139. . v=0. o=user1 53655765 2353687637 IN IP4 158.193.139.107. s=-. c=IN IP4 158.193.139.107. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000 U 158.193.139.91:5060 -> 158.193.139.120:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-3-0. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag003. To: sut <sip:1000@158.193.139.91:5060>;tag=2180SIPpTag0111. Call-ID: 3-636(a)158.193.139.120. CSeq: 1 INVITE. Contact: <sip:158.193.139.107:5060;transport=UDP>. Content-Type: application/sdp. Content-Length: 139. . v=0. o=user1 53655765 2353687637 IN IP4 158.193.139.107. s=-. c=IN IP4 158.193.139.107. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000 U 158.193.139.120:5060 -> 158.193.139.91:5060 INVITE sip:1000@158.193.139.91:5060 SIP/2.0. Via: SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-5-0. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag005. To: sut <sip:1000@158.193.139.91:5060>. Call-ID: 5-636(a)158.193.139.120. CSeq: 1 INVITE. Contact: sip:sipp@158.193.139.120:5060. Max-Forwards: 70. Subject: Performance Test. Content-Type: application/sdp. Content-Length: 139. . v=0. o=user1 53655765 2353687637 IN IP4 158.193.139.120. s=-. c=IN IP4 158.193.139.120. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000 U 158.193.139.91:5060 -> 158.193.139.120:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-5-0. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag005. To: sut <sip:1000@158.193.139.91:5060>. Call-ID: 5-636(a)158.193.139.120. CSeq: 1 INVITE. Server: kamailio (3.1.1 (i386/linux)). Content-Length: 0. . U 158.193.139.91:5060 -> 158.193.139.107:5060 INVITE sip:1000@158.193.139.107:5060 SIP/2.0. Record-Route: <sip:158.193.139.91;lr=on>. Via: SIP/2.0/UDP 158.193.139.91;branch=z9hG4bK3253.0acb5a11.0. Via: SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-5-0. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag005. To: sut <sip:1000@158.193.139.91:5060>. Call-ID: 5-636(a)158.193.139.120. CSeq: 1 INVITE. Contact: sip:sipp@158.193.139.120:5060. Max-Forwards: 69. Subject: Performance Test. Content-Type: application/sdp. Content-Length: 139. . v=0. o=user1 53655765 2353687637 IN IP4 158.193.139.120. s=-. c=IN IP4 158.193.139.120. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000 U 158.193.139.107:5060 -> 158.193.139.91:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 158.193.139.91;branch=z9hG4bK3253.0acb5a11.0, SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-5-0. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag005. To: sut <sip:1000@158.193.139.91:5060>;tag=2180SIPpTag0113. Call-ID: 5-636(a)158.193.139.120. CSeq: 1 INVITE. Contact: <sip:158.193.139.107:5060;transport=UDP>. Content-Length: 0. . U 158.193.139.107:5060 -> 158.193.139.91:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 158.193.139.91;branch=z9hG4bK3253.0acb5a11.0, SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-5-0. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag005. To: sut <sip:1000@158.193.139.91:5060>;tag=2180SIPpTag0113. Call-ID: 5-636(a)158.193.139.120. CSeq: 1 INVITE. Contact: <sip:158.193.139.107:5060;transport=UDP>. Content-Type: application/sdp. Content-Length: 139. . v=0. o=user1 53655765 2353687637 IN IP4 158.193.139.107. s=-. c=IN IP4 158.193.139.107. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000 U 158.193.139.91:5060 -> 158.193.139.120:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-5-0. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag005. To: sut <sip:1000@158.193.139.91:5060>;tag=2180SIPpTag0113. Call-ID: 5-636(a)158.193.139.120. CSeq: 1 INVITE. Contact: <sip:158.193.139.107:5060;transport=UDP>. Content-Length: 0. . U 158.193.139.91:5060 -> 158.193.139.120:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-5-0. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag005. To: sut <sip:1000@158.193.139.91:5060>;tag=2180SIPpTag0113. Call-ID: 5-636(a)158.193.139.120. CSeq: 1 INVITE. Contact: <sip:158.193.139.107:5060;transport=UDP>. Content-Type: application/sdp. Content-Length: 139. . v=0. o=user1 53655765 2353687637 IN IP4 158.193.139.107. s=-. c=IN IP4 158.193.139.107. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000 U 158.193.139.120:5060 -> 158.193.139.91:5060 ACK sip:1000@158.193.139.91:5060 SIP/2.0. Via: SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-5-5. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag005. To: sut <sip:1000@158.193.139.91:5060>;tag=2180SIPpTag0113. Call-ID: 5-636(a)158.193.139.120. CSeq: 1 ACK. Contact: sip:sipp@158.193.139.120:5060. Max-Forwards: 70. Subject: Performance Test. Content-Length: 0. . U 158.193.139.120:5060 -> 158.193.139.91:5060 BYE sip:1000@158.193.139.91:5060 SIP/2.0. Via: SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-5-7. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag005. To: sut <sip:1000@158.193.139.91:5060>;tag=2180SIPpTag0113. Call-ID: 5-636(a)158.193.139.120. CSeq: 2 BYE. Contact: sip:sipp@158.193.139.120:5060. Max-Forwards: 70. Subject: Performance Test. Content-Length: 0. . U 158.193.139.91:5060 -> 158.193.139.120:5060 SIP/2.0 404 Not Here. Via: SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-5-7. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag005. To: sut <sip:1000@158.193.139.91:5060>;tag=2180SIPpTag0113. Call-ID: 5-636(a)158.193.139.120. CSeq: 2 BYE. Server: kamailio (3.1.1 (i386/linux)). Content-Length: 0. . U 158.193.139.120:5060 -> 158.193.139.91:5060 BYE sip:1000@158.193.139.91:5060 SIP/2.0. Via: SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-5--1. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag005. To: sut <sip:1000@158.193.139.91:5060>;tag=2180SIPpTag0113. Call-ID: 5-636(a)158.193.139.120. CSeq: 3 BYE. Max-Forwards: 70. Contact: <sip:sipp@158.193.139.120:5060;transport=UDP>. Content-Length: 0. . U 158.193.139.91:5060 -> 158.193.139.120:5060 SIP/2.0 404 Not Here. Via: SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-5--1. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag005. To: sut <sip:1000@158.193.139.91:5060>;tag=2180SIPpTag0113. Call-ID: 5-636(a)158.193.139.120. CSeq: 3 BYE. Server: kamailio (3.1.1 (i386/linux)). Content-Length: 0. . U 158.193.139.107:5060 -> 158.193.139.91:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 158.193.139.91;branch=z9hG4bK1221.c7af9402.0, SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-4-0. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag004. To: sut <sip:1000@158.193.139.91:5060>;tag=2180SIPpTag0112. Call-ID: 4-636(a)158.193.139.120. CSeq: 1 INVITE. Contact: <sip:158.193.139.107:5060;transport=UDP>. Content-Type: application/sdp. Content-Length: 139. . v=0. o=user1 53655765 2353687637 IN IP4 158.193.139.107. s=-. c=IN IP4 158.193.139.107. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000 U 158.193.139.107:5060 -> 158.193.139.91:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 158.193.139.91;branch=z9hG4bK3253.0acb5a11.0, SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-5-0. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag005. To: sut <sip:1000@158.193.139.91:5060>;tag=2180SIPpTag0113. Call-ID: 5-636(a)158.193.139.120. CSeq: 1 INVITE. Contact: <sip:158.193.139.107:5060;transport=UDP>. Content-Type: application/sdp. Content-Length: 139. . v=0. o=user1 53655765 2353687637 IN IP4 158.193.139.107. s=-. c=IN IP4 158.193.139.107. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000 U 158.193.139.107:5060 -> 158.193.139.91:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 158.193.139.91;branch=z9hG4bK4285.21e4a9e7.0, SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-2-0. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag002. To: sut <sip:1000@158.193.139.91:5060>;tag=2180SIPpTag0110. Call-ID: 2-636(a)158.193.139.120. CSeq: 1 INVITE. Contact: <sip:158.193.139.107:5060;transport=UDP>. Content-Type: application/sdp. Content-Length: 139. . v=0. o=user1 53655765 2353687637 IN IP4 158.193.139.107. s=-. c=IN IP4 158.193.139.107. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000 U 158.193.139.91:5060 -> 158.193.139.120:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-5-0. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag005. To: sut <sip:1000@158.193.139.91:5060>;tag=2180SIPpTag0113. Call-ID: 5-636(a)158.193.139.120. CSeq: 1 INVITE. Contact: <sip:158.193.139.107:5060;transport=UDP>. Content-Type: application/sdp. Content-Length: 139. . v=0. o=user1 53655765 2353687637 IN IP4 158.193.139.107. s=-. c=IN IP4 158.193.139.107. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000 U 158.193.139.91:5060 -> 158.193.139.120:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-2-0. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag002. To: sut <sip:1000@158.193.139.91:5060>;tag=2180SIPpTag0110. Call-ID: 2-636(a)158.193.139.120. CSeq: 1 INVITE. Contact: <sip:158.193.139.107:5060;transport=UDP>. Content-Type: application/sdp. Content-Length: 139. . v=0. o=user1 53655765 2353687637 IN IP4 158.193.139.107. s=-. c=IN IP4 158.193.139.107. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000 U 158.193.139.91:5060 -> 158.193.139.120:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-4-0. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag004. To: sut <sip:1000@158.193.139.91:5060>;tag=2180SIPpTag0112. Call-ID: 4-636(a)158.193.139.120. CSeq: 1 INVITE. Contact: <sip:158.193.139.107:5060;transport=UDP>. Content-Type: application/sdp. Content-Length: 139. . v=0. o=user1 53655765 2353687637 IN IP4 158.193.139.107. s=-. c=IN IP4 158.193.139.107. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000 U 158.193.139.120:5060 -> 158.193.139.91:5060 INVITE sip:1000@158.193.139.91:5060 SIP/2.0. Via: SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-6-0. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag006. To: sut <sip:1000@158.193.139.91:5060>. Call-ID: 6-636(a)158.193.139.120. CSeq: 1 INVITE. Contact: sip:sipp@158.193.139.120:5060. Max-Forwards: 70. Subject: Performance Test. Content-Type: application/sdp. Content-Length: 139. . v=0. o=user1 53655765 2353687637 IN IP4 158.193.139.120. s=-. c=IN IP4 158.193.139.120. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000 U 158.193.139.91:5060 -> 158.193.139.120:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-6-0. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag006. To: sut <sip:1000@158.193.139.91:5060>. Call-ID: 6-636(a)158.193.139.120. CSeq: 1 INVITE. Server: kamailio (3.1.1 (i386/linux)). Content-Length: 0. . U 158.193.139.91:5060 -> 158.193.139.107:5060 INVITE sip:1000@158.193.139.107:5060 SIP/2.0. Record-Route: <sip:158.193.139.91;lr=on>. Via: SIP/2.0/UDP 158.193.139.91;branch=z9hG4bKd1cc.61efccf5.0. Via: SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-6-0. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag006. To: sut <sip:1000@158.193.139.91:5060>. Call-ID: 6-636(a)158.193.139.120. CSeq: 1 INVITE. Contact: sip:sipp@158.193.139.120:5060. Max-Forwards: 69. Subject: Performance Test. Content-Type: application/sdp. Content-Length: 139. . v=0. o=user1 53655765 2353687637 IN IP4 158.193.139.120. s=-. c=IN IP4 158.193.139.120. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000 U 158.193.139.107:5060 -> 158.193.139.91:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 158.193.139.91;branch=z9hG4bKd1cc.61efccf5.0, SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-6-0. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag006. To: sut <sip:1000@158.193.139.91:5060>;tag=2180SIPpTag0114. Call-ID: 6-636(a)158.193.139.120. CSeq: 1 INVITE. Contact: <sip:158.193.139.107:5060;transport=UDP>. Content-Length: 0. . U 158.193.139.107:5060 -> 158.193.139.91:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 158.193.139.91;branch=z9hG4bKd1cc.61efccf5.0, SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-6-0. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag006. To: sut <sip:1000@158.193.139.91:5060>;tag=2180SIPpTag0114. Call-ID: 6-636(a)158.193.139.120. CSeq: 1 INVITE. Contact: <sip:158.193.139.107:5060;transport=UDP>. Content-Type: application/sdp. Content-Length: 139. . v=0. o=user1 53655765 2353687637 IN IP4 158.193.139.107. s=-. c=IN IP4 158.193.139.107. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000 U 158.193.139.91:5060 -> 158.193.139.120:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-6-0. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag006. To: sut <sip:1000@158.193.139.91:5060>;tag=2180SIPpTag0114. Call-ID: 6-636(a)158.193.139.120. CSeq: 1 INVITE. Contact: <sip:158.193.139.107:5060;transport=UDP>. Content-Length: 0. . U 158.193.139.91:5060 -> 158.193.139.120:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-6-0. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag006. To: sut <sip:1000@158.193.139.91:5060>;tag=2180SIPpTag0114. Call-ID: 6-636(a)158.193.139.120. CSeq: 1 INVITE. Contact: <sip:158.193.139.107:5060;transport=UDP>. Content-Type: application/sdp. Content-Length: 139. . v=0. o=user1 53655765 2353687637 IN IP4 158.193.139.107. s=-. c=IN IP4 158.193.139.107. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000 U 158.193.139.120:5060 -> 158.193.139.91:5060 ACK sip:1000@158.193.139.91:5060 SIP/2.0. Via: SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-6-5. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag006. To: sut <sip:1000@158.193.139.91:5060>;tag=2180SIPpTag0114. Call-ID: 6-636(a)158.193.139.120. CSeq: 1 ACK. Contact: sip:sipp@158.193.139.120:5060. Max-Forwards: 70. Subject: Performance Test. Content-Length: 0. . U 158.193.139.120:5060 -> 158.193.139.91:5060 BYE sip:1000@158.193.139.91:5060 SIP/2.0. Via: SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-6-7. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag006. To: sut <sip:1000@158.193.139.91:5060>;tag=2180SIPpTag0114. Call-ID: 6-636(a)158.193.139.120. CSeq: 2 BYE. Contact: sip:sipp@158.193.139.120:5060. Max-Forwards: 70. Subject: Performance Test. Content-Length: 0. . U 158.193.139.91:5060 -> 158.193.139.120:5060 SIP/2.0 404 Not Here. Via: SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-6-7. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag006. To: sut <sip:1000@158.193.139.91:5060>;tag=2180SIPpTag0114. Call-ID: 6-636(a)158.193.139.120. CSeq: 2 BYE. Server: kamailio (3.1.1 (i386/linux)). Content-Length: 0. . U 158.193.139.120:5060 -> 158.193.139.91:5060 BYE sip:1000@158.193.139.91:5060 SIP/2.0. Via: SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-6--1. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag006. To: sut <sip:1000@158.193.139.91:5060>;tag=2180SIPpTag0114. Call-ID: 6-636(a)158.193.139.120. CSeq: 3 BYE. Max-Forwards: 70. Contact: <sip:sipp@158.193.139.120:5060;transport=UDP>. Content-Length: 0. . U 158.193.139.91:5060 -> 158.193.139.120:5060 SIP/2.0 404 Not Here. Via: SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-6--1. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag006. To: sut <sip:1000@158.193.139.91:5060>;tag=2180SIPpTag0114. Call-ID: 6-636(a)158.193.139.120. CSeq: 3 BYE. Server: kamailio (3.1.1 (i386/linux)). Content-Length: 0. . U 158.193.139.107:5060 -> 158.193.139.91:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 158.193.139.91;branch=z9hG4bK3253.0acb5a11.0, SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-5-0. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag005. To: sut <sip:1000@158.193.139.91:5060>;tag=2180SIPpTag0113. Call-ID: 5-636(a)158.193.139.120. CSeq: 1 INVITE. Contact: <sip:158.193.139.107:5060;transport=UDP>. Content-Type: application/sdp. Content-Length: 139. . v=0. o=user1 53655765 2353687637 IN IP4 158.193.139.107. s=-. c=IN IP4 158.193.139.107. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000 U 158.193.139.107:5060 -> 158.193.139.91:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 158.193.139.91;branch=z9hG4bK62b7.0e791b35.0, SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-3-0. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag003. To: sut <sip:1000@158.193.139.91:5060>;tag=2180SIPpTag0111. Call-ID: 3-636(a)158.193.139.120. CSeq: 1 INVITE. Contact: <sip:158.193.139.107:5060;transport=UDP>. Content-Type: application/sdp. Content-Length: 139. . v=0. o=user1 53655765 2353687637 IN IP4 158.193.139.107. s=-. c=IN IP4 158.193.139.107. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000 U 158.193.139.91:5060 -> 158.193.139.120:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-5-0. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag005. To: sut <sip:1000@158.193.139.91:5060>;tag=2180SIPpTag0113. Call-ID: 5-636(a)158.193.139.120. CSeq: 1 INVITE. Contact: <sip:158.193.139.107:5060;transport=UDP>. Content-Type: application/sdp. Content-Length: 139. . v=0. o=user1 53655765 2353687637 IN IP4 158.193.139.107. s=-. c=IN IP4 158.193.139.107. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000 U 158.193.139.107:5060 -> 158.193.139.91:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 158.193.139.91;branch=z9hG4bKd1cc.61efccf5.0, SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-6-0. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag006. To: sut <sip:1000@158.193.139.91:5060>;tag=2180SIPpTag0114. Call-ID: 6-636(a)158.193.139.120. CSeq: 1 INVITE. Contact: <sip:158.193.139.107:5060;transport=UDP>. Content-Type: application/sdp. Content-Length: 139. . v=0. o=user1 53655765 2353687637 IN IP4 158.193.139.107. s=-. c=IN IP4 158.193.139.107. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000 U 158.193.139.91:5060 -> 158.193.139.120:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-3-0. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag003. To: sut <sip:1000@158.193.139.91:5060>;tag=2180SIPpTag0111. Call-ID: 3-636(a)158.193.139.120. CSeq: 1 INVITE. Contact: <sip:158.193.139.107:5060;transport=UDP>. Content-Type: application/sdp. Content-Length: 139. . v=0. o=user1 53655765 2353687637 IN IP4 158.193.139.107. s=-. c=IN IP4 158.193.139.107. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000 U 158.193.139.91:5060 -> 158.193.139.120:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-6-0. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag006. To: sut <sip:1000@158.193.139.91:5060>;tag=2180SIPpTag0114. Call-ID: 6-636(a)158.193.139.120. CSeq: 1 INVITE. Contact: <sip:158.193.139.107:5060;transport=UDP>. Content-Type: application/sdp. Content-Length: 139. . v=0. o=user1 53655765 2353687637 IN IP4 158.193.139.107. s=-. c=IN IP4 158.193.139.107. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000 U 158.193.139.120:5060 -> 158.193.139.91:5060 INVITE sip:1000@158.193.139.91:5060 SIP/2.0. Via: SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-7-0. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag007. To: sut <sip:1000@158.193.139.91:5060>. Call-ID: 7-636(a)158.193.139.120. CSeq: 1 INVITE. Contact: sip:sipp@158.193.139.120:5060. Max-Forwards: 70. Subject: Performance Test. Content-Type: application/sdp. Content-Length: 139. . v=0. o=user1 53655765 2353687637 IN IP4 158.193.139.120. s=-. c=IN IP4 158.193.139.120. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000 U 158.193.139.91:5060 -> 158.193.139.120:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-7-0. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag007. To: sut <sip:1000@158.193.139.91:5060>. Call-ID: 7-636(a)158.193.139.120. CSeq: 1 INVITE. Server: kamailio (3.1.1 (i386/linux)). Content-Length: 0. . U 158.193.139.91:5060 -> 158.193.139.107:5060 INVITE sip:1000@158.193.139.107:5060 SIP/2.0. Record-Route: <sip:158.193.139.91;lr=on>. Via: SIP/2.0/UDP 158.193.139.91;branch=z9hG4bKf1fe.ea66e88.0. Via: SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-7-0. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag007. To: sut <sip:1000@158.193.139.91:5060>. Call-ID: 7-636(a)158.193.139.120. CSeq: 1 INVITE. Contact: sip:sipp@158.193.139.120:5060. Max-Forwards: 69. Subject: Performance Test. Content-Type: application/sdp. Content-Length: 139. . v=0. o=user1 53655765 2353687637 IN IP4 158.193.139.120. s=-. c=IN IP4 158.193.139.120. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000 U 158.193.139.107:5060 -> 158.193.139.91:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 158.193.139.91;branch=z9hG4bKf1fe.ea66e88.0, SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-7-0. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag007. To: sut <sip:1000@158.193.139.91:5060>;tag=2180SIPpTag0115. Call-ID: 7-636(a)158.193.139.120. CSeq: 1 INVITE. Contact: <sip:158.193.139.107:5060;transport=UDP>. Content-Length: 0. . U 158.193.139.107:5060 -> 158.193.139.91:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 158.193.139.91;branch=z9hG4bKf1fe.ea66e88.0, SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-7-0. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag007. To: sut <sip:1000@158.193.139.91:5060>;tag=2180SIPpTag0115. Call-ID: 7-636(a)158.193.139.120. CSeq: 1 INVITE. Contact: <sip:158.193.139.107:5060;transport=UDP>. Content-Type: application/sdp. Content-Length: 139. . v=0. o=user1 53655765 2353687637 IN IP4 158.193.139.107. s=-. c=IN IP4 158.193.139.107. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000 U 158.193.139.91:5060 -> 158.193.139.120:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-7-0. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag007. To: sut <sip:1000@158.193.139.91:5060>;tag=2180SIPpTag0115. Call-ID: 7-636(a)158.193.139.120. CSeq: 1 INVITE. Contact: <sip:158.193.139.107:5060;transport=UDP>. Content-Length: 0. . U 158.193.139.91:5060 -> 158.193.139.120:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-7-0. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag007. To: sut <sip:1000@158.193.139.91:5060>;tag=2180SIPpTag0115. Call-ID: 7-636(a)158.193.139.120. CSeq: 1 INVITE. Contact: <sip:158.193.139.107:5060;transport=UDP>. Content-Type: application/sdp. Content-Length: 139. . v=0. o=user1 53655765 2353687637 IN IP4 158.193.139.107. s=-. c=IN IP4 158.193.139.107. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000 U 158.193.139.120:5060 -> 158.193.139.91:5060 ACK sip:1000@158.193.139.91:5060 SIP/2.0. Via: SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-7-5. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag007. To: sut <sip:1000@158.193.139.91:5060>;tag=2180SIPpTag0115. Call-ID: 7-636(a)158.193.139.120. CSeq: 1 ACK. Contact: sip:sipp@158.193.139.120:5060. Max-Forwards: 70. Subject: Performance Test. Content-Length: 0. . U 158.193.139.120:5060 -> 158.193.139.91:5060 BYE sip:1000@158.193.139.91:5060 SIP/2.0. Via: SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-7-7. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag007. To: sut <sip:1000@158.193.139.91:5060>;tag=2180SIPpTag0115. Call-ID: 7-636(a)158.193.139.120. CSeq: 2 BYE. Contact: sip:sipp@158.193.139.120:5060. Max-Forwards: 70. Subject: Performance Test. Content-Length: 0. . U 158.193.139.91:5060 -> 158.193.139.120:5060 SIP/2.0 404 Not Here. Via: SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-7-7. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag007. To: sut <sip:1000@158.193.139.91:5060>;tag=2180SIPpTag0115. Call-ID: 7-636(a)158.193.139.120. CSeq: 2 BYE. Server: kamailio (3.1.1 (i386/linux)). Content-Length: 0. . U 158.193.139.120:5060 -> 158.193.139.91:5060 BYE sip:1000@158.193.139.91:5060 SIP/2.0. Via: SIP/2.0/UDP 158.193.139.120:5060;branch=z9hG4bK-636-7--1. From: sipp <sip:sipp@158.193.139.120:5060>;tag=636SIPpTag007. To: sut <sip:1000@158.193.139.91:5060>;tag=2180SIPpTag0115. Call-ID: 7-636(a)158.193.139.120. CSeq: 3 BYE. Max-Forwards: 70. Contact: <sip:sip... [Správa odseknutá]

1 0

GSoC 2011 mentoring
by Daniel-Constantin Mierla 28 Mar '11

28 Mar '11

Hello, I am again the coordinator for Kamailio's project at GSoC 2011, last year we had very good results with other developers joining the mentoring team. It was the case of Marius Zbihlei that really helped a lot, now being another opportunity to thank him again. If you want to help guiding the implementation of SIP-to-Jingle gateway, please come on board! This means you have to discuss with the selected student and guide him through one of following topics: Kamailio sources, SIP or Jingle. For the Jingle part, which is probably a bit unknown for our community, we can ask the guys at Jitsi project - we are part of the same organization at GSoC 2011. Relevant links to our GSoC participations: http://www.kamailio.org/w/gsoc-2011/ http://www.kamailio.org/w/gsoc-2010/ Thanks, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

1 0

Can't connect/register UA@SER
by SlawKo.skrzynka＠poczta.fm 27 Mar '11

27 Mar '11

> Hello, > > I am trying to set up basic SER configuration ("Hello World" Reference Design from SER-GettingStarted.pdf) but got in trouble at the UA connection. > > My teststbench consist of: > > SER server > IP address: 192.168.2.33 > Hostname: centoshost > OS: CentOS 5.3 > > Workstation with Linphone softphone > IP address: 192.168.2.5 > Hostname: susel > OS: openSUSE 11.2 > > ATA + regular phone > IP address: 192.168.2.191 > > Network switch + cabling > > > I had no problems with SER installation. It also runs correctly (I suppose). > > When SER started with updated ser.cfg according to manual, I couldn't even make basic registrations without any authentication. > > In the next step I added SQL support, hoping to register UA with authentication, but this didn't worked out either. > > So I have looked at ports at server and secured their openess - gave no improvement. > > > I will be glad to receive some hint where to make changes in my system. > > > Below are some commands (with results) I have tried on a server and workstaton. > I use SSH to connect to the server and issue commands there. > > As an attachement I provide ser.cfg file for inspection. > > > > Kind Regards, > > Slawomir > > > > ******************************************** > > [root@centoshost sbin]# cat /etc/*release > CentOS release 5.3 (Final) > > > [root@centoshost ~]# /etc/init.d/mysqld start > Starting MySQL: [ OK ] > [root@centoshost ~]# /etc/init.d/mysqld status > mysqld (pid 2640) is running... > > [root@centoshost ~]# export SIP_DOMAIN="centoshost" > > [root@centoshost sbin]# serctl start > > Starting SER : started pid(2709) > > > [root@centoshost sbin]# serctl moni > > Server: Sip EXpress router (0.9.6 (i386/linux)) ,2xx: 0 > Now: Sun Oct 11 04:56:03 2009 2009 local) ,2xx: 0 > Up Since: Sun Oct 11 04:00:30 2009 local) ,2xx: 0 > Up time: 3333 [sec]ics 0, 5xx: 0, 4xx: 0, 3xx: 0,2xx: 0 > Transaction Statistics Total: 0 (0 local) ,2xx: 0 > Transaction Statistics Total: 0 (0 local) 4xx: 0 > Current: 0 (0 waiting) Total: 0 (0 local) ,2xx: 0 > Replied localy: 0 0, 5xx: 0, 4xx: 0, 3xx: 0,2xx: 0 > Completion status 6xx: 0, 5xx: 0, 4xx: 0, 3xx: 0,2xx: 0 > Stateless Server Statistics 407: 0 408: 0 483: 0 4xx: 0 > Stateless Server Statistics > 200: 0 202: 0 2xx: 0 0 0 408: 0 483: 0 4xx: 0 > 300: 0 301: 0 302: 0 3xx: 0 0 408: 0 483: 0 4xx: 0 > 400: 0 401: 0 403: 0 404: 0 407: 0 408: 0 483: 0 4xx: 0 > 500: 0 5xx: 0 xpired > 6xx: 0 Expired > xxx: 0 Expired > failures: 0 > UsrLoc Statstered Expired > UsrLoc Statstered Expired > Domain Registered Expired > 'aliases' 0 00 > 'location' 0 0 > > [root@centoshost sbin]# serctl ps > 0 2709 attendant > 1 2714 fifo server > 2 2717 receiver child=0 sock= 192.168.2.33:5060 > 3 2719 receiver child=1 sock= 192.168.2.33:5060 > 4 2721 receiver child=2 sock= 192.168.2.33:5060 > 5 2724 receiver child=3 sock= 192.168.2.33:5060 > 6 2727 timer > 7 2730 tcp receiver > 8 2733 tcp receiver > 9 2735 tcp receiver > 10 2739 tcp receiver > 11 2742 tcp main process > > [root@centoshost sbin]# serctl ping 1001(a)192.168.2.191 > 400 > > [root@centoshost sbin]# ping 192.168.2.191 > PING 192.168.2.191 (192.168.2.191) 56(84) bytes of data. > 64 bytes from 192.168.2.191: icmp_seq=1 ttl=250 time=2.77 ms > 64 bytes from 192.168.2.191: icmp_seq=2 ttl=250 time=2.70 ms > 64 bytes from 192.168.2.191: icmp_seq=3 ttl=250 time=1.51 ms > > --- 192.168.2.191 ping statistics --- > 3 packets transmitted, 3 received, 0% packet loss, time 2000ms > rtt min/avg/max/mdev = 1.510/2.330/2.778/0.582 ms > > [root@centoshost sbin]# netstat -an > Active Internet connections (servers and established) > Proto Recv-Q Send-Q Local Address Foreign Address State > tcp 0 0 192.168.2.33:5060 0.0.0.0:* LISTEN > tcp 0 0 0.0.0.0:3306 0.0.0.0:* LISTEN > tcp 0 0 0.0.0.0:111 0.0.0.0:* LISTEN > tcp 0 0 0.0.0.0:757 0.0.0.0:* LISTEN > tcp 0 0 127.0.0.1:631 0.0.0.0:* LISTEN > tcp 0 0 127.0.0.1:25 0.0.0.0:* LISTEN > tcp 0 0 :::22 :::* LISTEN > tcp 0 0 ::ffff:192.168.2.33:22 ::ffff:192.168.2.5:56321 ESTABLISHED > udp 0 300 192.168.2.33:38530 192.168.12.254:53 ESTABLISHED > udp 0 0 192.168.2.33:44954 192.168.2.1:53 ESTABLISHED > udp 0 0 192.168.2.33:5060 0.0.0.0:* > udp 0 0 0.0.0.0:41417 0.0.0.0:* > udp 0 0 0.0.0.0:5353 0.0.0.0:* > udp 0 0 0.0.0.0:751 0.0.0.0:* > udp 0 0 0.0.0.0:111 0.0.0.0:* > udp 0 0 0.0.0.0:754 0.0.0.0:* > udp 0 0 0.0.0.0:631 0.0.0.0:* > udp 0 0 :::41038 :::* > udp 0 0 :::5353 :::* > > mysql> select username, password, domain from subscriber; > +----------+----------+------------+ > | username | password | domain | > +----------+----------+------------+ > | admin | heslo | centoshost | > | 1001 | abc | centoshost | > | 1000 | abc | centoshost | > +----------+----------+------------+ > > > > > **************************** > > susel:/usr/local # cat /etc/*release > LSB_VERSION="core-2.0-noarch:core-3.2-noarch:core-4.0-noarch:core-2.0-ia32:core-3.2-ia32:core-4.0-ia32" > openSUSE 11.2 (i586) > VERSION = 11.2 > > susel:/usr/local # ping 192.168.2.33 > PING 192.168.2.33 (192.168.2.33) 56(84) bytes of data. > 64 bytes from 192.168.2.33: icmp_seq=1 ttl=64 time=0.421 ms > 64 bytes from 192.168.2.33: icmp_seq=2 ttl=64 time=0.400 ms > 64 bytes from 192.168.2.33: icmp_seq=3 ttl=64 time=0.425 ms > ^C > --- 192.168.2.33 ping statistics --- > 3 packets transmitted, 3 received, 0% packet loss, time 1999ms > rtt min/avg/max/mdev = 0.400/0.415/0.425/0.019 ms > > susel:/usr/local # nmap -sS -p1-65535 192.168.2.33 > > Starting Nmap 5.00 ( http://nmap.org ) at 2011-03-21 20:32 CET > Interesting ports on 192.168.2.33: > Not shown: 65530 filtered ports > PORT STATE SERVICE > 22/tcp open ssh > 80/tcp closed http > 443/tcp closed https > 631/tcp closed ipp > 5060/tcp open sip > MAC Address: 08:00:06:0E:E1:FC (Siemens AG) > > Nmap done: 1 IP address (1 host up) scanned in 152.32 seconds > > > > susel:/usr/local # sipsak -vv -s sip:1001@192.168.2.33 > > received ICMP message (type: 3, code: 10): from 192.168.2.33 > > susel:/usr/local # sipsak -vv -s sip:1000@192.168.2.33 > > received ICMP message (type: 3, code: 10): from 192.168.2.33 > > susel:/usr/local # sipsak -T -s sip:1000@192.168.2.33 > warning: IP extract from warning activated to be more informational > (type: 3, code: 10): from 192.168.2.33 > > susel:/usr/local # sipsak -T -s sip:1001@192.168.2.33 > warning: IP extract from warning activated to be more informational > (type: 3, code: 10): from 192.168.2.33 > > susel:/usr/local # sipsak -a --password abc -s sip:1001@192.168.2.33 > (type: 3, code: 10): from 192.168.2.33 > > susel:/usr/local # sipsak -a --password abc -s sip:1000@192.168.2.33 > (type: 3, code: 10): from 192.168.2.33 > ------------------------------------------------- Sily koalicji atakuja rezim. Skomplikowana sytuacja w Libii. Raport » http://linkint.pl/f295a

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Watch algeria 3 TV 29 march at 07:00pm GMT
by Meftah Tayeb 27 Mar '11

27 Mar '11

hi guys, please everyone that have access to Neal SAT or Arab sat try to lookup algeria 3 TV channel and watch it starting from 07:00pm gmt time the TV channel have made a show about me, including my blind / tech life please check;) -- Meftah Tayeb inum: +883510001288000 phone: +13477595883

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Managing SUBSCRIBES
by Spinov Evgeniy 27 Mar '11

27 Mar '11

Hello, everyone. I've got stuck with a small problem with SUBSCRIBES. Let's say we have 2 separate customers with IDs: 1 and 2. They have 2 extenstions each: 1_100, 1_101 and 2_100, 2_101. The task is to subscribe 1_100 to 1_101. When I assign on DSS ( BLF ) key, phone number 1_101, then everything works perfectly, however, I cannot press this DSS key and get connected with 101 as prefix will break the magic. The question is there any workaround for this issue except cutting of the prefix for INVITE requests each time? Or how correctly to implement multi tenant kamailio? Thanks.

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