sr-users March 2012

sr-users@lists.kamailio.org

87 participants
162 discussions

Re: [SR-Users] Error loading pv.so undefined symbol: msg_ctx_id_set

by Daniel-Constantin Mierla

Hello, On 3/27/12 9:33 PM, Simpson Chua wrote: > Hi, > > [..] > > # /usr/local/kamailio/sbin/kamailio -V > version: kamailio 3.3.0-dev6 (x86_64/linux) 6853a8 for devel version be sure you run latest version from master branch. I just started with default config and runs fine, no missing symbol. Cheers, Daniel > [...] > ------------------------------------------------------------------------ > *From:* Daniel-Constantin Mierla <miconda(a)gmail.com> > *To:* Simpson Chua <simpsonchua(a)yahoo.com>; SIP Router - Kamailio > (OpenSER) and SIP Express Router (SER) - Users Mailing List > <sr-users(a)lists.sip-router.org> > *Sent:* Tuesday, March 27, 2012 2:23 PM > *Subject:* Re: [SR-Users] Error loading pv.so undefined symbol: > msg_ctx_id_set > > Hello, > > are you sure you are running 3.2.x? That version does not have > msg_ctx_id_set() at all yet. Can you send the output of 'kamailio -V'? > > Cheers, > Daniel > > On 3/27/12 8:51 PM, Simpson Chua wrote: >> Hi, >> >> I'm getting the following error when attempting to load pv.so >> >> kamailio: ERROR: <core> [sr_module.c:572]: ERROR: load_module: could >> not open module </usr/local/lib64/kamailio/modules_k/pv.so>: >> /usr/local/lib64/kamailio/modules_k/pv.so: undefined symbol: >> msg_ctx_id_set >> >> I'm running Kamailio 3.2.x. >> >> Thanks, >> Simpson >> >> >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >> sr-users(a)lists.sip-router.org <mailto:sr-users@lists.sip-router.org> >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > -- > Daniel-Constantin Mierla > Kamailio Advanced Training, April 23-26, 2012, Berlin, Germany > http://www.asipto.com/index.php/kamailio-advanced-training/ > > -- Daniel-Constantin Mierla Kamailio Advanced Training, April 23-26, 2012, Berlin, Germany http://www.asipto.com/index.php/kamailio-advanced-training/

11 years, 8 months

Error loading pv.so undefined symbol: msg_ctx_id_set

by Simpson Chua

Hi, I'm getting the following error when attempting to load pv.so kamailio: ERROR: <core> [sr_module.c:572]: ERROR: load_module: could not open module </usr/local/lib64/kamailio/modules_k/pv.so>: /usr/local/lib64/kamailio/modules_k/pv.so: undefined symbol: msg_ctx_id_set I'm running Kamailio 3.2.x. Thanks, Simpson

11 years, 8 months

Re: [SR-Users] branches usage

by Uri Shacked

On Tue, Mar 27, 2012 at 3:13 PM, Uri Shacked <uri.shacked(a)gmail.com> wrote: > Hi, > > > > Following the advice I got on the subject “remove_hf" and "append_hf" one > after the other , twice, issue”, I tried to work with the “append_branch()” > function. > > What I need to do is, after I received the invite from the original user, > I use SIP methods to work with the NP server. I subst() the user part, send > the call to the NP server, get the reply, parse it and then subst() back to > the destination and send the call out. > > Doing it with “append_branch” seems logical. > > I do “append_branch()” before changing the user, get the reply (301), > goes on to failure route after the branch route was called, and from there > I have a problem. > > I need to continue my flow without the branch I added. How do I return to > the original on? > > I read about the t_drop_replies but didn’t quit understand….. > > > > BR, > > Uri >

11 years, 8 months

Re: [SR-Users] Kamailio modifying ruri during loose-route - SOLVED

by Dan-Cristian Bogos

Another one bites the dust ;). My problem was strict vs loose routing done by loose_route function. In order to loop through B2BUA I have defined the loopback interface's IP as a local alias for simplicity in checking myself in routing script. When doing loose_route, the route address was for some reason not matching my server and due to strict routing the RURI was rewritten with the weird address in the route header. Hope my report helps someone in the future. DanB

11 years, 8 months

"remove_hf" and "append_hf" one after the other , twice, issue.

by Uri Shacked

Thanks. I read option 2 and i agree on 1 :-) Didnt use any "branch stuff" yet :-(... it seems very logical to do so when i send the call to the NP server. Will hit the docs... Hi, when you are adding the new Contact-Header in Failure-Route (after you received the 3xx reply), you are working on the original message again. So the old Contact is still removed and the new Contact is added (as for the initial request). There are two options for you: 1) Move your remove_hf/append_hf into a branch route (so the changes are only for the current branch) 2) Use "msg_apply_changes()" before removing and adding the header again (see http://kamailio.org/docs/modules/devel/modules/textopsx.html#textopsx.msg_a… ) I i were you, i'd choose option 1, since for Option 2 the message and the headers need to be parsed again.... (performance penalty). Good luck, Carsten 2012/3/27 Uri Shacked <ushacked at gmail.com<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users> >: >* Hi, *>* *>* *>* *>* In the scenario i am using, the flow is as followed: *>* *>* 1. Getting the INVITE from the user *>* *>* 2. Sending an INVITE with RURI changes to the NP server (sip NP server) *>* *>* 3. Getting a 301 reply (using the reply for the NP code) *>* *>* 4. Sending the call to the destination. *>* *>* *>* *>* My problem is that the NP server needs some parameters in the contact *>* header. *>* *>* So, here is what i do : *>* *>* *>* *>* First, before sending the invite to the NP: *>* *>* *>* *>* remove_hf("Contact"); *>* *>* append_hf("Contact *>* :<sip:XXX at server_ip<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users> :port;param1=p1;param2=p2>\r\n","Contact"); *>* *>* *>* *>* (The invite is sent perfectly and works great) *>* *>* *>* *>* After the NP reply i do: *>* *>* *>* *>* remove_hf("Contact"); *>* *>* append_hf("Contact :$ct","Contact"); *>* *>* *>* *>* Now I notice there are two contact headers. It seems that the remove_hf *>* didn’t work this time. *>* *>* Any ideas? *>* *>* *>* *>* Thanks, *>* *>* Uri *

11 years, 8 months

"remove_hf" and "append_hf" one after the other , twice, issue.

by Uri Shacked

Hi, In the scenario i am using, the flow is as followed: 1. Getting the INVITE from the user 2. Sending an INVITE with RURI changes to the NP server (sip NP server) 3. Getting a 301 reply (using the reply for the NP code) 4. Sending the call to the destination. My problem is that the NP server needs some parameters in the contact header. So, here is what i do : First, before sending the invite to the NP: remove_hf("Contact"); append_hf("Contact :<sip:XXX@server_ip :port;param1=p1;param2=p2>\r\n","Contact"); (The invite is sent perfectly and works great) After the NP reply i do: remove_hf("Contact"); append_hf("Contact :$ct","Contact"); Now I notice there are two contact headers. It seems that the remove_hf didn’t work this time. Any ideas? Thanks, Uri

11 years, 8 months

Passwords in Kamailio user table are not encrypted.

by Albert Petit

good morning When provisioning subscriebrs to Kamailio with the kamctl tool i see the passwords are stored in clear on the database. For my use-case I would not like to have that passwords in clear on the db. What can I do to avoid this problem? Should I provision the users in a different way and not using kamctl tool or configure kamailio differently? If there is no way to achieve this would be dangerous if I edit the kamctl tool so it does not store any value on the password column and just store HA and HA1B values? BR,

11 years, 8 months

Re: [SR-Users] Kamailio Load Testing

by Uri Shacked

> > > >How do you reload your htable? Is it using the exec( ) function to call > the > >mi command sht_reload? Or do you have another method? > > I use the exec() to run a shell script with the relevant mi command. the > shell script return code is used to make sure all went OK. > > >Thanks > > >Reda > > > > On Mon, Mar 26, 2012 at 14:56, Uri Shacked <ushacked at gmail.com<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>> > wrote: > > >* Hi, > *>* > *>* > *>* > *>* I made these kinds of tests before. I have two tips for you to pay > *>* attention to: > *>* > *>* 1. Read about the [routes] on SIPp, It is tricky to satisfy > kamailio > *>* with SIPp scenarios. > *>* > *>* 2. Make sure your DB table is well build (use less varchars and > more > *>* integers). > *>* > *>* I found out that there are many kamailio modules I can use to load > data in > *>* to memory and I hardly use the DB (only for ACC). I use MTREE, HTABLE, > *>* DIALPLAN, CARRIERROUTE, DROUTING, and so to store the information I > need. > *>* And I get it with the simple functions of the module. > *>* > *>* I load around 6 million numbers and other data which takes about 1.5 > Gb of > *>* the memory. > *>* > *>* For getting around the real-time changes that I need to deal with (In > DB > *>* the data changes are immediately made on your service…). I have a > RTIMER > *>* every 120 sec that check is a reload is needed. If so, it reloads the > *>* relevant data (very fast). > *>* > *>* BR, > *>* > *>* Uri > *>* > *>* > *>* > *>* > *>* Hello, > *>* > *>* On 3/20/12 12:12 PM, Stephen Dodge (Bistech) wrote: > *>* >* > *>* *>* Hello, > *>* *>* > *>* *>* I am running Kamailio 3.1.5 with a MySQL backend on CentOS. A > *>* *>* connection to MySQL (an off box MySQL cluster) is required for > every > *>* *>* call, sqlops is used to determine destinations and acc to record > CDR * > *>* >* information. > *>* *>* > *>* *>* I am planning to load test our server using SIPp to generate > calls, I > *>* *>* was wondering if anyone has done something similar and could > provide > *>* *>* advice on what we should monitor on our Kamailio Server. i.e > Server > *>* *>* CPU & Memory > *>* *>* > *>* *>* Thanks in advance for your help. > *>* *>* > *>* *if you do a lot of direct DB interactions, perhaps latency of the > *>* queries will be relevant. You can use benchmark module to see how long > *>* it takes to execute part of the config file. > *>* > *>* I think you don't fetch lot of records from db in config, so memory > *>* should be no problem, however you can dump private/shared memory usage > *>* via RPC commands within sercmd cli. CPU is a good metric always and > easy > *>* to watch with systems tools. > *>* > *>* Cheers, > *>* Daniel > *>* >* > *>* *>* Steve. > *>* * > * >

11 years, 8 months

TLS delay on request processing

by Andreas Heise

Hi list, I use Kamailio 3.2.2 as TLS<=>TCP gateway this is working well, but if I create some traffic by SIPP then Kamailio response time increase with every new call, seems to be that Kamailio process call by call. I made a test with 30caps then after ~200calls Kamailio need ~3s to response an initial INVITE from the PBX. This delay seems to be increased with each call. test scenario: SIPP UAC---TCP---PABX---TLS---Kamailio---TCP---SIPP UAS 1.1.1.1 = PABX 2.2.2.2 = SIPP UAS 5.5.5.5 = Kamailio if ($si == "1.1.1.1") # from PABX { xlog("L_INFO","FROM PABX: $rm, $ci\n"); t_on_reply("1"); force_send_socket(tcp:5.5.5.5:5060); rewritehostport("2.2.2.2:5070"); t_relay_to_tcp("2.2.2.2","5070"); exit; }; I got all the calls in a single TLS connection from the PBAX so increasing of TCP childs doesn't help. I also already tried to adjust some TCP buffers and played with TCP_ASYNC. With UDP / TCP it's better, is really the TLS/openssl is the bottleneck or the way I relay the request? The test machine has only 512MB RAM and is a single core cpu (2,8GHZ), but 30caps isn't much? Any suggestions? many thanks, Andreas

11 years, 8 months

Kamailio modifying ruri during loose-route

by Dan-Cristian Bogos

Hey Guys, I have noticed some unexpected behavior (at least by me) during my tests. When routing specific ACK message back to 200 OK originator, Kamailio will rewrite the ruri with the value of route header. Is that known or some memory access problem? Call setup: B2BUA(127.0.0.1) -> Kamailio (1.2.3.4) -> End device (2.3.4.5). Bellow is a trace of 200 OK -ACK flow before and after kamailio, together with the xlog capture of ruri before and after calling loose_route on ACK. Thanks in advance for any kind of tip. DanB PS: Since testing with a newer version would be a bit harder due to some particular features I use in 3.1 which are taking a bit of effort to migrate, have tried my luck here before, so hope to get your understanding. Kamailio version: """ root@kam:/home/dan# kamailio -V version: kamailio 3.1.5 (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 4MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled on 07:41:01 Sep 15 2011 with gcc 4.4.5 """ Syslog: """ Mar 26 14:27:01 kam /usr/sbin/kamailio[7929]: ERROR: <script>: Before loose_route, ru is sip:01234567@127.0.0.1:5069;transport=udp Mar 26 14:27:01 kam /usr/sbin/kamailio[7929]: ERROR: <script>: Loose route, ru is sip:1.2.3.4:25060;lr=on;ftag=a9c837bc;nat=yes """ Ngrep of call: """ # U 2012/03/26 14:27:00.089455 127.0.0.1:5069 -> 1.2.3.4:25060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 1.2.3.4:25060;branch=z9hG4bK361e.ba530345.0. Via: SIP/2.0/UDP 192.168.1.29:5060;rport=3081;received=2.3.4.5;branch=z9hG4bK-373632-e2d6ebcc5a1d2a481e4957541b393f9a. Record-Route: <sip:1.2.3.4:25060;lr=on;ftag=a9c837bc;nat=yes>. From: "ines.demo" <sip:ines.demo@mydomain.net>;tag=a9c837bc. To: <sip:01234567@mydomain.net>;tag=BcBSHa56Bp7ge. Call-ID: 65464bffda1b33cfcc9d97c9e8f81cb3@0:0:0:0:0:0:0:0. CSeq: 2 INVITE. Contact: <sip:01234567@127.0.0.1:5069;transport=udp>. User-Agent: WebiB2BUA 1.0.6. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY. Supported: timer, precondition, path, replaces. Allow-Events: talk, hold, refer. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 417. Remote-Party-ID: "01234567" <sip:01234567@mydomain.net>;party=calling;privacy=off;screen=no. . v=0. o=root 13076 13076 IN IP4 1.2.3.4. s=session. c=IN IP4 1.2.3.4. t=0 0. m=audio 48548 RTP/AVP 8 0 102 3 101. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:102 iLBC/8000. a=fmtp:102 mode=30. a=rtpmap:3 GSM/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. a=nortpproxy:yes. m=video 0 RTP/AVP 96 97 98. a=rtpmap:96 /0. a=rtpmap:97 /0. a=rtpmap:98 /0. # U 2012/03/26 14:27:00.089589 1.2.3.4:25060 -> 2.3.4.5:3081 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.1.29:5060;rport=3081;received=2.3.4.5;branch=z9hG4bK-373632-e2d6ebcc5a1d2a481e4957541b393f9a. Record-Route: <sip:1.2.3.4:25060;lr=on;ftag=a9c837bc;nat=yes>. From: "ines.demo" <sip:ines.demo@mydomain.net>;tag=a9c837bc. To: <sip:01234567@mydomain.net>;tag=BcBSHa56Bp7ge. Call-ID: 65464bffda1b33cfcc9d97c9e8f81cb3@0:0:0:0:0:0:0:0. CSeq: 2 INVITE. Contact: <sip:01234567@127.0.0.1:5069;transport=udp>. User-Agent: WebiB2BUA 1.0.6. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY. Supported: timer, precondition, path, replaces. Allow-Events: talk, hold, refer. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 417. Remote-Party-ID: "01234567" <sip:01234567@mydomain.net>;party=calling;privacy=off;screen=no. . v=0. o=root 13076 13076 IN IP4 1.2.3.4. s=session. c=IN IP4 1.2.3.4. t=0 0. m=audio 48548 RTP/AVP 8 0 102 3 101. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:102 iLBC/8000. a=fmtp:102 mode=30. a=rtpmap:3 GSM/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. a=nortpproxy:yes. m=video 0 RTP/AVP 96 97 98. a=rtpmap:96 /0. a=rtpmap:97 /0. a=rtpmap:98 /0. # U 2012/03/26 14:27:00.160046 2.3.4.5:3081 -> 1.2.3.4:25060 ACK sip:01234567@127.0.0.1:5069;transport=udp SIP/2.0. Call-ID: 65464bffda1b33cfcc9d97c9e8f81cb3@0:0:0:0:0:0:0:0. CSeq: 2 ACK. Via: SIP/2.0/UDP 192.168.1.29:5060;branch=z9hG4bK-373632-6eac5ecf0ba5ab4ce7094ad17870dca3. From: "ines.demo" <sip:ines.demo@mydomain.net>;tag=a9c837bc. To: <sip:01234567@mydomain.net>;tag=BcBSHa56Bp7ge. Max-Forwards: 70. Route: <sip:1.2.3.4:25060;lr=on;ftag=a9c837bc;nat=yes>. Contact: "ines.demo" <sip:ines.demo@192.168.1.29:5060;transport=udp;registering_acc=mydomain_net>. User-Agent: Jitsi1.0-beta1-nightly.build.3820Windows 7. Content-Length: 0. . # U 2012/03/26 14:27:00.160278 1.2.3.4:25060 -> 1.2.3.4:25060 ACK sip:1.2.3.4:25060;lr=on;ftag=a9c837bc;nat=yes SIP/2.0. Call-ID: 65464bffda1b33cfcc9d97c9e8f81cb3@0:0:0:0:0:0:0:0. CSeq: 2 ACK. Via: SIP/2.0/UDP 1.2.3.4:25060;branch=z9hG4bKcydzigwkX. Via: SIP/2.0/UDP 192.168.1.29:5060;rport=3081;received=2.3.4.5;branch=z9hG4bK-373632-6eac5ecf0ba5ab4ce7094ad17870dca3. From: "ines.demo" <sip:ines.demo@mydomain.net>;tag=a9c837bc. To: <sip:01234567@mydomain.net>;tag=BcBSHa56Bp7ge. Max-Forwards: 69. Contact: "ines.demo" <sip:ines.demo@2.3.4.5:3081;transport=udp;registering_acc=mydomain_net>. User-Agent: Jitsi1.0-beta1-nightly.build.3820Windows 7. Content-Length: 0. """

11 years, 8 months

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sr-users March 2012