January 2011 - sr-users

onreply_route and failure_route
by Bernhard Suttner 29 Jan '11

29 Jan '11

Hi, is it possible to call a route within onreply_ and failure_route? I mean, within the main route { ... } its possible to call the route[NAT]. Is this also possible for this "special" routes? If this is possible, is it also allowed to call a route which will normally called from the main route { ... } Thanks in advance. Best regards, Bernhard

2 1

Joining strings
by Lee Archer 29 Jan '11

29 Jan '11

Hi, is it possible to join strings? I'd like to prepend a number to a string prior to processing it to an int? Thanks Lee

2 2

Can you use a bigint with Kamailio?
by Lee Archer 29 Jan '11

29 Jan '11

Hi, I am trying to run dispatcher based on the inbound number called. This works fine for some numbers but not all. Is there a bigint string function as this would solve my problems? The numbers I am receiving are 14 digits long which s.int is returning as something completely different to what it should be. Thanks Lee

2 2

LCR next_gw() - certain GW, source from other IP address
by Graham Wooden 29 Jan '11

29 Jan '11

Hi there, I have a requirement now that for one of my GW¹s in my LCR, needs to be sourced from another IP address (two different rate-decks with the carrier). This ratedeck will always be at least 2nd inline ... I was thinking something along the sorts ... Some pseudo code: failure_route [2] { . . if (next_gw()) { if ( name = 'gw_name') { force_send_socket(udp:ip.address.of.certain.ip:5060); } else { # just dummy, as we will send out our default IP for everyone else. } t_on_failure("2"); route(19); } . . . } What variable will have the value that's in the gw_name column? I can't go by the IP address in $ru because the destination IP is the same between the two ratedecks. Overall, Is that the best way to handle this requirement? Thanks all, -graham

3 2

Re: [SR-Users] LCR next_gw() - certain GW, source from other IP address
by GP Wooden 29 Jan '11

29 Jan '11

Same destination IP for both rate decks, so I need to go by something else, like the gw_name. ----- Reply message ----- From: "Stagg Shelton" <stagg(a)vocalcloud.com> Date: Sat, Jan 29, 2011 8:27 am Subject: [SR-Users] LCR next_gw() - certain GW, source from other IP address To: <sr-users(a)lists.sip-router.org> Do the provider gateways have the same IP address for the different rate plans? If the gateways are different then the following may be relevant. if (next_gw()) { # prepare for lcr failover t_on_failure("2"); if($rd=="aaa.aaa.aaa.aaa") { xlog ("L_INFO", "This is a request for Some Rate Plan"); force_send_socket(udp:xxx.xxx.xxx.xxx:5060); } else { xlog ("L_INFO", "This is a request for another rate plan"); force_send_socket(udp:yyy.yyy.yyy.yyy:5060); }; route(2); } else { xlog("No gateways found!"); Stagg On 1/29/11 9:10 AM, Graham Wooden wrote: > Hi there, > > I have a requirement now that for one of my GW¹s in my LCR, needs to be > sourced from another IP address (two different rate-decks with the carrier). > This ratedeck will always be at least 2nd inline ... > > I was thinking something along the sorts ... Some pseudo code: > > failure_route [2] { > . > . > if (next_gw()) { > if ( name = 'gw_name') { > force_send_socket(udp:ip.address.of.certain.ip:5060); > } else { > # just dummy, as we will send out our default IP for everyone > else. > } > t_on_failure("2"); > route(19); > } > . > . > . > } > > What variable will have the value that's in the gw_name column? I can't go > by the IP address in $ru because the destination IP is the same between the > two ratedecks. > > Overall, Is that the best way to handle this requirement? > > Thanks all, > > -graham > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

3 4

Kamailio Devel Training, Barcelona - update
by Daniel-Constantin Mierla 28 Jan '11

28 Jan '11

Hello, a quick update about the next training in Barcelona, Feb 10-11, 2011, focusing on Kamailio development. We got a bigger room (for second time :-) ), so we increased the capacity to 28 seats. We are already 23 people in the class and several more joining the dinner Thursday evening, so there is going to be a great time. If you plan to come, register asap, there is no time to get a bigger room, so once we are full booked this time, we close the participants list. You can see more details at: http://www.kamailio.org/w/2011/01/kamailio-development-training-barcelona-f… Also, do not forget Fosdem next weekend, it is a bunch of us going there, two talks about Kamailio, come and say hi: http://www.kamailio.org/w/2011/01/social-networking-event-brussels-belgium/ Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

1 0

Kamailio as a routing engine
by Gang Liu 28 Jan '11

28 Jan '11

Hello, I wrote a H.323/SIP IWF program before, it was based on B2BUA framework. Currently I am planning to use Kamailio as internal routing server and let IWF become a class 4 soft switch.That is Inbound ---- H.323/ SIP -- > ---- SIP routing request-----> Kamailio IWF/SBC | Outbound <--- H.323/SIP -- <--- SIP ----------------------------- Is it ok to store all voip provider's information at database of Kamailio and pass all to IWF at private SIP headers and let IWF as a outbound proxy? Is Kamailio as a SIP redirect server or it is better to stay at call signaling path until call session ended? It is great to share any information about this. regards, Gang

2 1

Load_gw failed
by Thiago Maluf 27 Jan '11

27 Jan '11

Hi Kamailio Users, I'm trying to use LCR module in Kamailio 3.1.1 and I', having some problems with this function (load_gws). I'm my kamailio.cfg , I have this route: # SEARCH DEST route[NUMBER] { xlog("L_ALERT", "NUMBER IS $rU \n"); if ($rU=~"^55211020[0-9]{4}$") { # IS MY USERS route(LDAP_NUM2USER); } else { # NOT IS MY USER if (!load_gws(1,$rU)) { xlog("L_ALERT", "LOAD LOCAL GW FAILED $rU \n"); route(DSER); return; } else { xlog("L_ALERT", "LOAD GW DEBUG $avp(i:709)\n"); } ; if (!next_gw()) { xlog("L_ALERT", "NEXT LOCAL GW FAILED\n"); sl_send_reply("503", "Service not available - No gateways"); return; }; setflag(29); route(RELAY); return; }; }; When my call arrive to load_gws, I have this problem: (2198) ERROR: <core> [action.c:1158]: runtime fixup failed for load_gws param 2 (2198) ERROR: <core> [action.c:1511]: run action error at: /etc/kamailio/kamailio.cfg:472 I already saw the Kamailio Code and this error doesn't make sense. Would somebody Help me with it? Very thanks, Thiago Maluf. -- ---------------------------------------------------------------- Thiago Maluf Resende Tel: +55 21 9700-9113 e-mail: malufrj(a)gmail.com

2 1

SIP Recorder
by Danny Dias 27 Jan '11

27 Jan '11

Hello my friends, I have a requeriment, which indicates that i have to record every SIP conversation between peers (also for callings to the PSTN); the recording server will be built for our company following this requeriments (also requested for the client): My doubt is: How can i handle sip conversations recording when all the calls are passing through a Proxy Server? I do understand that the media is always peer to peer and the signaling goes through the Proxy, but in this case the media not only has to pass between the peers because it must be recorded. How should i handle this?

7 19

Kamailio CFG file size ??
by Jijo 27 Jan '11

27 Jan '11

Hi All, is there any limitation on the size of kamailo.cfg file. When i tried to add few avp_subst, then the kamailio starts showing up the error out of package memory on receiving SIP message. If i remove those lines then everything works fine.. avp_subst is simple one. Our kamailio cfg file is quite big.. Please let me know how to debug this issue? Here is the error 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: <core> [db_row.c:116]: no private memory left 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: db_postgres [km_res.c:293]: could not allocate row 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: db_postgres [km_res.c:258]: failed to convert row #1 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: db_postgres [km_res.c:75]: failed to convert rows 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: db_postgres [km_dbase.c:416]: error while converting result 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: <core> [db_query.c:137]: error while storing result 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: avpops [avpops_db.c:333]: cannot do the query 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: <core> [parser/msg_parser.c:342]: ERROR:parse_headers: memory allocation error 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: maxfwd [mf_funcs.c:62]: parsing MAX_FORWARD header failed! 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: <core> [parser/msg_parser.c:342]: ERROR:parse_headers: memory allocation error 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: rr [loose.c:98]: failed to parse headers 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: <core> [parser/msg_parser.c:342]: ERROR:parse_headers: memory allocation error 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: siputils [checks.c:64]: To parsing failed 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: <core> [action.c:831]: ERROR: do_action: memory allocation failure 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: <core> [action.c:1511]: run action error at: :0 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: pv [pv_core.c:1761]: do action failed 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: <core> [lvalue.c:358]: setting pvar failed 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: <core> [parser/msg_parser.c:342]: ERROR:parse_headers: memory allocation error 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: <core> [parser/parse_from.c:71]: ERROR:parse_from_header: out of pkg_memory 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: <core> [route.c:1846]: ERROR: eval_elem: bad or missing From: header 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: <core> [parser/msg_parser.c:342]: ERROR:parse_headers: memory allocation error 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: <core> [parser/msg_parser.c:342]: ERROR:parse_headers: memory allocation error 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: <core> [parser/msg_parser.c:342]: ERROR:parse_headers: memory allocation error 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: <core> [msg_translator.c:2229]: ERROR: build_res_buf_from_sip_req: alas, parse_headers failed 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [parser/msg_parser.c:170]: get_hdr_field: out of memory 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: rr [loose.c:70]: failed to parse To header field 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [parser/msg_parser.c:170]: get_hdr_field: out of memory 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: siputils [checks.c:64]: To parsing failed 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [action.c:831]: ERROR: do_action: memory allocation failure 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [action.c:1511]: run action error at: :0 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_core.c:1740]: do action failed) 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [lvalue.c:358]: setting pvar failed 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [parser/msg_parser.c:170]: get_hdr_field: out of memory 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [parser/msg_parser.c:170]: get_hdr_field: out of memory 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_core.c:1381]: error parsing headers 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [parser/msg_parser.c:342]: ERROR:parse_headers: memory allocation error 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [data_lump.c:318]: ERROR: del_lump: out of memory 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: textops [textops.c:1081]: no memory 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [parser/parse_from.c:71]: ERROR:parse_from_header: out of pkg_memory 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [route.c:1846]: ERROR: eval_elem: bad or missing From: header 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_svar.c:117]: out of pkg mem! 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_core.c:1640]: error - cannot set svar [backLinkMode] 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [lvalue.c:358]: setting pvar failed 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [parser/parse_from.c:71]: ERROR:parse_from_header: out of pkg_memory 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_core.c:418]: cannot parse From header 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [parser/msg_parser.c:342]: ERROR:parse_headers: memory allocation error 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_core.c:400]: cannot parse To header 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_svar.c:117]: out of pkg mem! 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_core.c:1640]: error - cannot set svar [temp] 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [lvalue.c:358]: setting pvar failed 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_svar.c:117]: out of pkg mem! 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_core.c:1640]: error - cannot set svar [node] 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: osb [osb.c:916]: osb_set_var_char(): setting profile PV failed. value INVALID (char) 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_svar.c:117]: out of pkg mem! 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_core.c:1640]: error - cannot set svar [binding] 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: osb [osb.c:916]: osb_set_var_char(): setting profile PV failed. value INVALID (char) 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [parser/msg_parser.c:342]: ERROR:parse_headers: memory allocation error 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [parser/msg_parser.c:342]: ERROR:parse_headers: memory allocation error 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [parser/msg_parser.c:342]: ERROR:parse_headers: memory allocation error 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [parser/msg_parser.c:342]: ERROR:parse_headers: memory allocation error 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [parser/msg_parser.c:342]: ERROR:parse_headers: memory allocation error 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_shv.c:367]: no more pkg mem 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [parser/parse_from.c:71]: ERROR:parse_from_header: out of pkg_memory 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_core.c:418]: cannot parse From header 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [parser/msg_parser.c:342]: ERROR:parse_headers: memory allocation error 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_core.c:400]: cannot parse To header 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_svar.c:117]: out of pkg mem! 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_core.c:1640]: error - cannot set svar [temp] 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [lvalue.c:358]: setting pvar failed 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_svar.c:117]: out of pkg mem! 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_core.c:1640]: error - cannot set svar [node] 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: osb [osb.c:916]: osb_set_var_char(): setting profile PV failed. value INVALID (char) 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_svar.c:117]: out of pkg mem! 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_core.c:1640]: error - cannot set svar [binding] 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: osb [osb.c:916]: osb_set_var_char(): setting profile PV failed. value INVALID (char) 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [parser/parse_from.c:71]: ERROR:parse_from_header: out of pkg_memory 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_core.c:418]: cannot parse From header 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: osb [osb.c:1257]: osb_get_osv_current_binding(): no binding available 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <script>: R1 - binding IP=0 not available

2 3

NAT Problem in Asterisk Realtime + Kamailio config
by andrzej.ciupek＠asterisk.edu.pl 27 Jan '11

27 Jan '11

Hello I have started to use configuration like: http://kb.asipto.com/asterisk:realtime:kamailio-3.0.x-asterisk-1.6.2-astdb I works fine, but I have problem with dealing with NAT. kamctl ul show gives me: AOR:: andrzej_104 Contact:: sip:andrzej_104@10.5.0.94:14368;rinstance=9c2b256a13210aa7 Q= Expires:: 46 Callid:: ZDUwYzIzMDNiZTI3MjkyNmFmOWEyOTI2ODcyNjVmNDg. Cseq:: 60 User-agent:: eyeBeam release 1102q stamp 51814 State:: CS_DIRTY Flags:: 0 Cflag:: 0 Socket:: udp:77.xx.yyy.zzz:5060 Methods:: 5087 AOR:: andrzej_103 Contact:: sip:andrzej_103@10.5.0.165:5060 Q= Expires:: 926 Callid:: 11631436-5060-1(a)10.5.0.165 Cseq:: 2257 User-agent:: Grandstream HT-502 V1.2A 1.0.3.10 State:: CS_SYNC Flags:: 0 Cflag:: 0 Socket:: udp:77.xx.yyy.zzz:5060 Methods:: 6879 So when I have incomming call from Asterisk to Kamailio, Kamailio start to send call to LAN IP. I have tried to use help from: http://nil.uniza.sk/sip/nat-fw/configuring-nat-traversal-using-kamailio-31-… But I have never get: Received:: and Cflag:: 192 it is always Cflag:: 0. field in my kamctl ul show. In my configuration Asterisk is Callweaver and is on the other server. I use rtpproxy-1.2.1 and kamailio-3.0.1 and mofified config from link: http://kb.asipto.com/asterisk:realtime:kamailio-3.0.x-asterisk-1.6.2-astdb Please help.

2 2

RLS question
by "Andrés S. García Ruiz" 27 Jan '11

27 Jan '11

Hi, I'm trying to create a RLS "rls-services" document with a contacts-list inside it. Kamailio states that "No xcap document found for uri:x". When creating a subscribe I must set the uri to the service uri. What I want to know is the sql query Kamailio does in order to get the contacts list for that uri. For example: XCAP document: "<?xml version=\"1.0\" encoding=\"UTF-8\"?>" + "<rls-services xmlns=\"urn:ietf:params:xml:ns:rls-services\"" + " xmlns:rl=\"urn:ietf:params:xml:ns:resource-lists\">" + "<service uri=\"sip:restricted_areas_presentities@open-ims.test\">" + "<list name=\"friends\">" + "<rl:entry uri=\"sip:testuser01@open-ims.test\"/>" + "<rl:entry uri=\"sip:testuser02@open-ims.test\"/>" + "</list>" + "<packages>" + "<package>presence</package>" + "</packages>" + "</service>" + "</rls-services>"; Is stored in /xcap-root/rls-services/users/sip:restricted_areas@open-ims.test/index The subscribe is from sip:restricted_areas@open-ims.test to sip:restricted_areas_presentities@open-ims.test What am I doing wrong? Thanks, Andrés.

3 5

[kamailio-users] kamailio on amazon
by Chandrakant Solanki 26 Jan '11

26 Jan '11

Hi I have installed Kamailio with MediaProxy and asterisk on Amazon Server.. While kamailio/MediaProxy and Asterisk both running on different amazon's instance. Kamailio : 1.5.0-notls MediaProxy : 2.3.8 Asterisk : 1.6.2.6 Firewall port is opened for mediaproxy from 10000-20000 (UDP), 5060 (UDP) etc on both amazon machine.. while I tried to play an audio... it plays sound file but unable to found audio on device. when I put tcpdump on asterisk machine it gives following error... # tcpdump -i eth0 udp portrange 10000-20000 -w test1.pcap 01:39:52.850279 IP (tos 0xb8, ttl 64, id 59, offset 0, flags [DF], proto: UDP (17), length: 60) ip-W.X.Y.Z.compute.internal.15246 > xyz.com.10010: [bad udp cksum 3cfd!] UDP, length 32 01:39:52.870279 IP (tos 0xb8, ttl 64, id 60, offset 0, flags [DF], proto: UDP (17), length: 60) ip-W.X.Y.Z.compute.internal.15246 > xyz.com.10010: [bad udp cksum 97fe!] UDP, length 32 01:39:52.890281 IP (tos 0xb8, ttl 64, id 61, offset 0, flags [DF], proto: UDP (17), length: 60) ip-W.X.Y.Z.compute.internal.15246 > xyz.com.10010: [bad udp cksum 75c0!] UDP, length 32 Any Idea..!! -- Regards, Chandrakant Solanki

3 2

Re: [SR-Users] fix_nated_sdp()
by Bernhard Suttner 25 Jan '11

25 Jan '11

Hi, thanks for your response. I think I understand the scenario very good - but im not so familiar with the fix_nated_sdp() functionality. Currently the SDP will be modified with the source IP of the message. This does work good but currently I only do that for some special devices (matching the User-Agent / Server string). I want to do that fix_nated_sdp() in all cases and not only for some special devices. The FreeSWITCH RTP-Auto Adjustment does work in nearly every case but in one (currently known) scenario it does not. Maybe I should just give it a try to do fix_nated_sdp() in combination with nat_uac_test (RFC1918 private address in SDP). Thanks a lot for this discussion. Best Regards, Bernhard ----- Original Message ----- From: Ovidiu Sas [mailto:osas@voipembedded.com] To: Bernhard Suttner [mailto:bernhard.suttner@winet.ch] Cc: Carsten Bock [mailto:lists@bock.info], sr-users(a)lists.sip-router.org Sent: Tue, 25 Jan 2011 20:38:14 +0100 Subject: Re: [SR-Users] fix_nated_sdp() > You need to experiment with nathelper module and while doing that, you > need to inspect the SIP messages in order to understand how it works. > Only when you fully understand your scenarios, you will be able to > properly configure, debg and setup kamailio. > > Best thing to do here is to use rtpproxy and relay media if needed. > > Another option would be to let the SDP untouched and let freeswitch do > it's magic (if it can). Not sure if this will work in all scenarios. > > Calling fix_nated_sdp() on the SBC will definitely not work (you are > just messing with the IP on the SDP and you don't know if the real RTP > port on the NATed side will match the advertised port in your "fixed" > SDP). > > > Regards, > Ovidiu Sas > > On Tue, Jan 25, 2011 at 1:23 PM, Bernhard Suttner > <bernhard.suttner(a)winet.ch> wrote: > > Hi, > > > > thanks for your answer. > > > > The question was about, if the fix_nated_sdp() (in the example below) > could maybe break something. I am sure, that the fix_nated_sdp() would work > in "some" cases but I am not sure, if the function could break something: > > > > > > if (method==INVITE && has_sdp() && nat_uac_test(8)) fix_nated_sdp(3) > > > > and within the onreply route: > > > > if (status=~(180|183|200) && has_sdp() && nat_uac_test(8)) > fix_nated_sdp(3) > > > > > > Or is there a better solution to fix these damend NAT scenarios? > > > > Best regards, > > Bernhard > > > > > > -----Ursprüngliche Nachricht----- > > Von: kaiserbock2(a)googlemail.com [mailto:kaiserbock2@googlemail.com] Im > Auftrag von Carsten Bock > > Gesendet: Dienstag, 25. Januar 2011 18:56 > > An: Bernhard Suttner > > Cc: sr-users(a)lists.sip-router.org > > Betreff: Re: [SR-Users] fix_nated_sdp() > > > > Hi Bernhard, > > > > have you checked the SDP which is going to and from the > > Freeswitch-Server? Is it modified? > > > > If yes: You may want to check your NAT-Firewall. A working (but not > > nice) solution might be to relay the RTP through an proxy in this > > case... > > If No: Verify the result of "nat_uac_test": Why is it returning "false"? > > > > Carsten > > > > 2011/1/25 Bernhard Suttner <bernhard.suttner(a)winet.ch>: > >> Hi, > >> > >> someone has an idea? > >> > >> Thanks in advance! > >> > >> Best regards, > >> Bernhard > >> > >> -----Ursprüngliche Nachricht----- > >> Von: sr-users-bounces(a)lists.sip-router.org > [mailto:sr-users-bounces@lists.sip-router.org] Im Auftrag von Bernhard > Suttner > >> Gesendet: Montag, 24. Januar 2011 20:38 > >> An: sr-users(a)lists.sip-router.org > >> Betreff: [SR-Users] fix_nated_sdp() > >> > >> Hi, > >> > >> I have a question about the fix_nated_sdp() function. Different devices > (phones, pbxes, gateways) are using Kamailio as SBC. The SBC does t_relay() > the message to a pool of FreeSWITCH Servers. > >> > >> The scenario does work quite well but: > >> > >> if a call is initiated from one of the freeswitch server, pass it through > kamailio to the PBX and the PBX does have a forwarding back to kamailio and > in the end to freeswitch I have no audio because the PBX is behind a NAT and > uses private ip addresses in all the SDP of 200 OK and 183 Session Progress. > Therefore FreeSWITCH could not do the (very cool) RTP Auto-Adjustment (check > if IP in incoming packet is different to the IP in the SDP) to detect the > real IP address. > >> > >> Therefore I have to use fix_nated_sdp(). I am not sure, what this > function could break therefore I want to be very sure and use it only, if > really necessary. Test which User-Agent or Server the peer (in this example > the PBX) has, is not really possible because some devices does not send this > header. > >> > >> What do you think about the follwoing pseudo code: > >> > >> if (method==INVITE && has_sdp() && nat_uac_test(8)) fix_nated_sdp(3) > >> > >> and within the onreply route: > >> > >> if (status=~(180|183|200) && has_sdp() && nat_uac_test(8)) > fix_nated_sdp(3) > >> > >> Somebody has a better idea to handle this? What could go wrong? > >> > >> Thanks in advance. > >> > >> Best regards, > >> Bernhard Suttner > >> > >> _______________________________________________ > >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > >> sr-users(a)lists.sip-router.org > >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > >> > >> > >> _______________________________________________ > >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > >> sr-users(a)lists.sip-router.org > >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > >> > > > > > > > > -- > > Carsten Bock > > Schomburgstr. 80 > > 22767 Hamburg > > Germany > > > > Mobile +49 179 2021244 > > Home +49 40 34927217 > > Büro (Verl) +49 5246 801427 > > Fax +49 40 34927218 > > mailto:carsten@bock.info > > > > > > _______________________________________________ > > 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

fix_nated_sdp()
by Bernhard Suttner 25 Jan '11

25 Jan '11

Hi, I have a question about the fix_nated_sdp() function. Different devices (phones, pbxes, gateways) are using Kamailio as SBC. The SBC does t_relay() the message to a pool of FreeSWITCH Servers. The scenario does work quite well but: if a call is initiated from one of the freeswitch server, pass it through kamailio to the PBX and the PBX does have a forwarding back to kamailio and in the end to freeswitch I have no audio because the PBX is behind a NAT and uses private ip addresses in all the SDP of 200 OK and 183 Session Progress. Therefore FreeSWITCH could not do the (very cool) RTP Auto-Adjustment (check if IP in incoming packet is different to the IP in the SDP) to detect the real IP address. Therefore I have to use fix_nated_sdp(). I am not sure, what this function could break therefore I want to be very sure and use it only, if really necessary. Test which User-Agent or Server the peer (in this example the PBX) has, is not really possible because some devices does not send this header. What do you think about the follwoing pseudo code: if (method==INVITE && has_sdp() && nat_uac_test(8)) fix_nated_sdp(3) and within the onreply route: if (status=~(180|183|200) && has_sdp() && nat_uac_test(8)) fix_nated_sdp(3) Somebody has a better idea to handle this? What could go wrong? Thanks in advance. Best regards, Bernhard Suttner

3 4

accounting serial forked transactions with 302 from LCR
by thrillerbee 25 Jan '11

25 Jan '11

I'm converting my OpenSIPS routers to Kamailio & have run into a small complication. The proxy pushes all INVITEs to a least-cost router. This LCR responds with a list of routes as contact instances in a 302 Redirect. Calls are routing a serially forking normally. Connected & failed calls account normally. However, if the caller cancels the call, the acc module includes the 302 in the transaction record as the final response as opposed to the actual final response - the 487 Request Canceled. Is there something I could be missing that would cause this? Thanks, Ryan

2 4

Re: [SR-Users] $ua in on_reply route not set?
by Bernhard Suttner 25 Jan '11

25 Jan '11

Hi, found the problem. The device does sometimes use User-Agent and sometimes Server. Is it better to use $hdr() or the Search() function? I use this to check for the User-Agent and then to do a "fix_nated_sdp()" (in route[] and onreply[]) because I am not really sure, if the fix_nated_sdp() could break something. Or should kamailio break nothing here? Sometimes the User-Agent/Server is missing in Session-Progress 183. Therefore a "global" fix_nated_sdp() would be nice to have. Best regards, Bernhard ----- Original Message ----- From: Daniel-Constantin Mierla [mailto:miconda@gmail.com] To: Bernhard Suttner [mailto:bernhard.suttner@winet.ch] Cc: sr-users(a)lists.sip-router.org Sent: Mon, 24 Jan 2011 17:09:12 +0100 Subject: Re: [SR-Users] $ua in on_reply route not set? > Hello, > > On 1/24/11 5:03 PM, Bernhard Suttner wrote: > > Hi, > > > > could it be, that the $ua pseudo variable is not set within in a onreply > route? (Version 3.1). > no, should be set, there was no change in this regard for quite long > time. Can you sent the sip reply plus log with debug=3? > > > > What is the best alternative for that? Search()? > > The alternative is $hdr(User-Agent) which is practically returning the > same value as $ua, using generic header search mechanism. > > Cheers, > Daniel > > -- > Daniel-Constantin Mierla > http://www.asipto.com > >

2 1

Re: [SR-Users] pdb module best practice
by Henning Westerholt 25 Jan '11

25 Jan '11

On Monday 24 January 2011, Thomas Baumann wrote: > the downtime is not a problem, I already have 2 deamons running. > I want to reduce the time , how changes become active. > > 1. update xxx entry > 2. export table to *.csv (5s) > 3. generate *.mmap file (25 seconds) > > So changes will take at lease 30 seconds to become active. Only 20% of my > numbers are updated very often, I want to export them and the generation > of the *.mmap file takes only 5 seconds. > > But in this case I need to specify the daemons per request and not in the > general parameters. Because later on it might be necessary to ask the > daemons hosting the rest of my numbers. Or maybe 30 seconds are ok ? Hi Thomas, i understand. You could of course change/ extend the interface of the pdb, but at the moment this functionality is not available. During module startup the sockets and server data structures are initialized, so the structure needs probably a bit changed. But let me ask, what kind of changes are done to your routing? Is the routing data then completely invalid, or just a more expensive route is taken for this particular target? In the latter case a update period of 30 seconds sounds not like a big deal to me, after all you probably have also some manual steps involved before you do an update. Cheers, Henning

1 0

Re: [SR-Users] SIP between IPv4 and IPv6 agents
by Komáromi Péter 25 Jan '11

25 Jan '11

Hello! I went through your letters and start as following: ------------------------------------------------------- root@ubuntu:/home/yeti/rtpproxy# /etc/init.d/kamailio restart Restarting kamailio: kamailioloading modules under /usr/local/lib/kamailio/modules_k/:/usr/local/lib/kamailio/modules/ Listening on udp: 192.168.124.158:5060 udp: FEC0:0:0:0:0:0:0:2:5060 tcp: 192.168.124.158:5060 tcp: FEC0:0:0:0:0:0:0:2:5060 Aliases: tcp: ubuntu:5060 udp: ubuntu:5060 root@ubuntu:/home/yeti/rtpproxy# kamctl restart INFO: Stopping Kamailio : INFO: stopped INFO: Starting Kamailio : INFO: started (pid: 7597) root@ubuntu:/home/yeti/rtpproxy# rtpproxy restart ------------------------------------------------------- So I guess everything is starting fine. The users can register and call each other. Their data are in the mysql database`s location table. I put this into the kamailio.cfg file: -------------------------------------------------------- # RTPProxy control route[RTPPROXY] { #!ifdef WITH_NAT if (is_method("BYE")) { unforce_rtp_proxy(); } else if (is_method("INVITE")) { if (af == inet) { if ( $(rU{s.substr,0,1}) == "6") { force_rtp_proxy("ie"); } else { force_rtp_proxy("ee"); } } else { if ( $(rU{s.substr,0,1}) == "4") { force_rtp_proxy("ei"); } else { force_rtp_proxy("ii"); } } } else if (is_method("200OK")) { if (af == inet) { if ( $(rU{s.substr,0,1}) == "6") { force_rtp_proxy("ei"); } else { force_rtp_proxy("ee"); } } else { if ( $(rU{s.substr,0,1}) == "4") { force_rtp_proxy("ie"); } else { force_rtp_proxy("ii"); } } } if (!has_totag()) add_rr_param(";nat=yes"); #!endif return; } ---------------------------------------------------------- But, unfortunately there is no RTP traffic yet. :( What do you thing guys, what can be the problem? Regards, Peter Am 02.12.2010 13:06, schrieb Komáromi Péter: > Hi! > > So if you say it is possible to solve the problem with the only > location table, the location_inet4 and location_inet6 is not > certainly necessary... do I _have to_ use the 4to6.cfg file from the > source of kamailio, or not? When you call force_rtp_proxy() then you have to provide the i/e flags to tell rtpproxy how it should bridge the call. Therefore you need to know for every call: - is the caller IPv4 or IPv6? - is the callee IPv4 or IPv6? Finding out the protocol version of the caller is rather easy: if (af == inet) { # caller ipv4 ... } else { # caller ipv6 ... } But finding out if the callee is v4 or v6 is difficult. Actually it may even be that the callee is registered with 2 clients at the same time - one using v4 and v6. As far as I know this multi-registration scenario can not be handled currently - only single-registrations or multi-registrations which use the same protocol version are supported. Anyway - there are several approaches how to detect the IP version of the callee. - fixed mapping: for example if you know that all usernames starting with 1 are IPv4 clients, and all usernames starting with 2 are IPv6 clients, then you can detect the callee's protocol version manually, e.g: if ( $(rU{s.substr,0,1}) == "1") { # callee ipv4 ... } else { # callee ipv6 ... } - check destination URI if it contains a IPv4 or IPv6 address, e.g. use http://www.kamailio.org/dokuwiki/doku.php/transformations:devel#resubst_exp… to test if $dd contains '.' which would indicate a ipv4 address. - use dedicated location tables > (Because it uses them, for instance here: if (method == "REGISTER") > { if (af == inet) { save("location-inet4"); } else if (af == inet6) > { save("location-inet6");...) > > If I do not have to use it, I do not need to merge it with my > kamailio.cfg file, the steps I have to do are: install rtpproxy, > start it with the appropriate parameters and set the forcing in the > kamailio.cfg file?! The more I think about it I prefer the approach with separated location tables. It should be rather easy to integrate the 2 location tables into your existing config. regards Klaus

3 4

Problem reloading carrierroute
by Morten Isaksen 25 Jan '11

25 Jan '11

Hi, Kamailio 3.0.3. A few days ago we got a problem when "kamctl cr reload" was executed. Jan 24 08:50:27 sip-core-2 /usr/local/sbin/kamailio[1639]: ERROR: db_mysql [km_dbase.c:346]: no memory left Jan 24 08:50:27 sip-core-2 /usr/local/sbin/kamailio[1639]: ERROR: carrierroute [cr_db.c:331]: Fetching rows failed Jan 24 08:50:27 sip-core-2 /usr/local/sbin/kamailio[1639]: ERROR: carrierroute [cr_data.c:181]: could not load routing data I tried to increase the shared memory from 32M to 64M but that did not seem to help. It failed again after a few hours. The last thing I tried was to insert modparam("carrierroute", "fetch_rows", 2000) into kamailio.cfg and is waiting to see if that helped. The carrierroute table is about 16K rows. Is it private og shared memory that is missing? Any other ideas how to fix this? -- Morten Isaksen

2 3

Building specific commit
by mayamatakeshi 25 Jan '11

25 Jan '11

Hello, I am using these commands to build and install kamailio: git clone http://git.sip-router.org/sip-router kamailio cd kamailio git checkout -b 3.1 origin/3.1 make FLAVOUR=kamailio cfg make group_include="mysql" all make group_include="mysql" install Now, If I want to build a specific commit like a0d376d85aae2f6e46e381a01fb85c5830250d4d, what should I do? regards, takeshi

2 2

kamctl fifo <command> not responding
by Anton Roman 24 Jan '11

24 Jan '11

Hi all, I'm having trouble trying to execute fifo commands with "kamctl fifo <command>". Just after restarting Kamailio it works fine, however, sometimes after some days running it doesn't respond. kamailio1:~#* kamctl fifo which* It doesn't respond so I input *Crtl+c* and I get: /usr/local/lib/kamailio//kamctl/kamctl.fifo: line 89: /tmp/kamailio_fifo: Interrupted system call If I delete and create the fifo file again (with "rm /tmp/kamailio_fifo" and "mkfifo /tmp/kamailio_fifo" and "chmod 660 /tmp/kamailio_fifo") it keeps not responding. Any help is welcome, what can be happening? Below you can find info about the pipe and the running kamailio. Thanks in advance, Best regards Antón kamailio1:~# *ls -hall /tmp/kamailio_fifo * prw-rw---- 1 root root 0 ene 17 12:01 /tmp/kamailio_fifo After deleting and creating the fifo file again: kamailio1:~# *ls -hall /tmp/kamailio_fifo * prw-rw-r-- 1 root root 0 ene 18 10:28 /tmp/kamailio_fifo kamailio1:~#* ps -ef | grep kama* root 17369 17245 0 10:12 pts/0 00:00:00 grep kama kamailio 23277 1 0 Jan15 ? 00:00:00 /usr/local/sbin/kamailio -P /var/run/kamailio/kamailio.pid -m 512 -u kamailio -g kamailio kamailio 23289 23277 0 Jan15 ? 00:00:00 /usr/local/sbin/kamailio -P /var/run/kamailio/kamailio.pid -m 512 -u kamailio -g kamailio kamailio 23291 23277 0 Jan15 ? 00:00:00 /usr/local/sbin/kamailio -P /var/run/kamailio/kamailio.pid -m 512 -u kamailio -g kamailio kamailio 23293 23277 0 Jan15 ? 00:00:00 /usr/local/sbin/kamailio -P /var/run/kamailio/kamailio.pid -m 512 -u kamailio -g kamailio kamailio 23294 23277 0 Jan15 ? 00:00:00 /usr/local/sbin/kamailio -P /var/run/kamailio/kamailio.pid -m 512 -u kamailio -g kamailio kamailio 23295 23277 0 Jan15 ? 00:01:14 /usr/local/sbin/kamailio -P /var/run/kamailio/kamailio.pid -m 512 -u kamailio -g kamailio kamailio 23299 23277 0 Jan15 ? 00:01:13 /usr/local/sbin/kamailio -P /var/run/kamailio/kamailio.pid -m 512 -u kamailio -g kamailio kamailio 23300 23277 0 Jan15 ? 00:01:13 /usr/local/sbin/kamailio -P /var/run/kamailio/kamailio.pid -m 512 -u kamailio -g kamailio kamailio 23303 23277 0 Jan15 ? 00:01:13 /usr/local/sbin/kamailio -P /var/run/kamailio/kamailio.pid -m 512 -u kamailio -g kamailio kamailio 23305 23277 0 Jan15 ? 00:05:29 /usr/local/sbin/kamailio -P /var/run/kamailio/kamailio.pid -m 512 -u kamailio -g kamailio kamailio 23306 23277 0 Jan15 ? 00:05:33 /usr/local/sbin/kamailio -P /var/run/kamailio/kamailio.pid -m 512 -u kamailio -g kamailio kamailio 23309 23277 0 Jan15 ? 00:05:30 /usr/local/sbin/kamailio -P /var/run/kamailio/kamailio.pid -m 512 -u kamailio -g kamailio kamailio 23311 23277 0 Jan15 ? 00:05:31 /usr/local/sbin/kamailio -P /var/run/kamailio/kamailio.pid -m 512 -u kamailio -g kamailio kamailio 23312 23277 0 Jan15 ? 00:00:02 /usr/local/sbin/kamailio -P /var/run/kamailio/kamailio.pid -m 512 -u kamailio -g kamailio kamailio 23313 23277 0 Jan15 ? 00:00:39 /usr/local/sbin/kamailio -P /var/run/kamailio/kamailio.pid -m 512 -u kamailio -g kamailio kamailio 23315 23277 0 Jan15 ? 00:00:00 /usr/local/sbin/kamailio -P /var/run/kamailio/kamailio.pid -m 512 -u kamailio -g kamailio kamailio 23320 23277 0 Jan15 ? 00:00:00 /usr/local/sbin/kamailio -P /var/run/kamailio/kamailio.pid -m 512 -u kamailio -g kamailio kamailio 23321 23277 0 Jan15 ? 00:00:00 /usr/local/sbin/kamailio -P /var/run/kamailio/kamailio.pid -m 512 -u kamailio -g kamailio kamailio 23322 23277 0 Jan15 ? 00:00:05 /usr/local/sbin/kamailio -P /var/run/kamailio/kamailio.pid -m 512 -u kamailio -g kamailio kamailio 23323 23277 0 Jan15 ? 00:00:00 /usr/local/sbin/kamailio -P /var/run/kamailio/kamailio.pid -m 512 -u kamailio -g kamailio

2 5

$ua in on_reply route not set?
by Bernhard Suttner 24 Jan '11

24 Jan '11

Hi, could it be, that the $ua pseudo variable is not set within in a onreply route? (Version 3.1). What is the best alternative for that? Search()? Thanks in advance! Best regards, Bernhard

2 1

Multiple Sip
by aram amin 24 Jan '11

24 Jan '11

Dear Sir / Madam I registered for your services today and the main purpose was to register my other voip/sip providers with your services so that I can have more options in making out bound calls with my sip phone, however could you please explain to me that if for instance, I have three accounts with three different voip providers (i.e JustVoip, Smartvoip,and Voipcheap) ,then how do i register them with my account that you have opened for me today. could you also tell me how is it possible to make specific calls from one of the voip providers above . Thank you Regards Aram

2 1

pdb module best practice
by Thomas Baumann 24 Jan '11

24 Jan '11

Hello, I tested the pdb module and the pdb_server. It is a fast solution even with a huge amount of numbers. But how I can update the pdb_server in a fast and effective way ? I tried it in the following way: - number;carrier are stored in a mysql table (20 million entries) - 2 pdb_servers are running - the table is exported into a csv file (time: 5 seconds) - pdbt will create a new mmappable file (time: 25 seconds) - the pdb_servers will be restarted to activate the changes The time of 30 seconds is good but not fast enought. Some of my numbers are updated very often, to export them and create a new mmappable file will take only 3 seconds. I had the idea to split my numbers in a more or less static area and a dynamic area. What I want to do later is to have the pdb_server not static configured, I want to define the server within the request: 1. query in dynamic range: pdb_query("$rU", "$avp(i:1000)", "localhost:10001,localhost:10002"); if not found look into the static area: pdb_query("$rU", "$avp(i:1000)", "localhost:10003,localhost:10004"); Does it make sence, or has somebody a better idea how changes activated within 10s or less ? regards, Thomas ___________________________________________________________ Neu: WEB.DE De-Mail - Einfach wie E-Mail, sicher wie ein Brief! Jetzt De-Mail-Adresse reservieren: https://produkte.web.de/go/demail02

2 1

2025

2024

2023

2022

2021

2020

2019

2018

2017

2016

2015

2014

2013

2012

2011

2010

2009

2008

2007

2006

2005

2004

2003

2002

sr-users January 2011