sr-users January 2012

sr-users@lists.kamailio.org

72 participants
115 discussions

Kamailio 3.1.5 core dump from the snmpstats module
by Klaus Feichtinger 06 Feb '12

06 Feb '12

Hello list, I`ve found a negative phenomenon during startup of kamailio 3.1.x (orgininally found in version 3.1.5) that results in a core dump, triggered by the snmpstats module. During startup of the SIP proxy server machine the linux service "heartbeat" is responsible for starting required ressources like mysql, kamailio and snmpd. These services are monitored by the tool "monit". Monit is supervising the state of the named services and if kamailio is not up and running, monit is trying to restart the service. During startup the mysql database takes some time until it is ready to use and therefore kamailio can`t connect to the database. Monit is detecting this and restarting kamailio. This is done so often until kamailio is up and running. Finally all services are up and running and the machine is working fine. However, I do not like the core dump that was created during this "restart phase". As soon as the module "snmpstats" is not used, no core dump is created any more. Information about the core dump looks as follows: (1) output of kamailio log files: <prefix> kamailio: INFO: <core> [tcp_main.c:4726]: init_tcp: using epoll_lt as the io watch method (auto detected) <prefix> /usr/sbin/kamailio[3027]: ERROR: db_mysql [km_my_con.c:109]: driver error: Can't connect to local MySQL server through socket '/var/run/mysqld/mysqld.sock' (2) <prefix> /usr/sbin/kamailio[3027]: ERROR: <core> [db.c:289]: could not add connection to the pool <prefix> /usr/sbin/kamailio[3027]: ERROR: lcr [lcr_mod.c:359]: unable to connect to the database <prefix> /usr/sbin/kamailio[3027]: ERROR: lcr [lcr_mod.c:571]: unable to open database connection <prefix> /usr/sbin/kamailio[3027]: ERROR: <core> [sr_module.c:875]: init_mod(): Error while initializing module lcr (/usr/lib/kamailio/modules/lcr.so) <prefix> /usr/sbin/kamailio[3027]: INFO: snmpstats [snmpstats.c:387]: The SNMPStats module got the kill signal <prefix> kamailio: ERROR: <core> [daemonize.c:307]: Main process exited before writing to pipe <prefix> kamailio: INFO: <core> [tcp_main.c:4726]: init_tcp: using epoll_lt as the io watch method (auto detected) <prefix> kamailio: WARNING: <core> [daemonize.c:352]: pid file contains old pid, replacing pid <prefix> /usr/sbin/kamailio[3322]: INFO: usrloc [hslot.c:53]: locks array size 512 <prefix> /usr/sbin/kamailio[3322]: INFO: auth [auth_mod.c:312]: auth: qop set, but nonce-count (nc_enabled) support disabled <prefix> /usr/sbin/kamailio[3322]: INFO: pua [pua.c:360]: the query returned no result <prefix> /usr/sbin/kamailio[3322]: INFO: pike [ip_tree.c:88]: probing 256 set size <prefix> /usr/sbin/kamailio[3322]: INFO: <core> [udp_server.c:178]: INFO: udp_init: SO_RCVBUF is initially 111616 <prefix> /usr/sbin/kamailio[3322]: INFO: <core> [udp_server.c:229]: INFO: udp_init: SO_RCVBUF is finally 262142 <prefix> /usr/sbin/kamailio[3329]: INFO: mi_datagram [mi_datagram.c:326]: a new child 0/3329 (2) output of GDB / core file: Core was generated by `/usr/sbin/kamailio -P /var/run/kamailio/kamailio.pid -m 1024 -u kamailio -g kama'. Program terminated with signal 11, Segmentation fault. [New process 3027] #0 0xb775539c in freeInterprocessBuffer () at interprocess_buffer.c:354 354 interprocess_buffer.c: No such file or directory. in interprocess_buffer.c (gdb) I know that it is not a fine method in killing kamailio during initialisation. But I have not found a better way yet. The other modules like LCR have no problem with the kill signal during initialisation. My main question is now: how could these core dumps being avoided? Did anybody have the same experience as me? Thanks in advance, Klaus

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Kamailio ims / rtpproxy
by trojak＠st.fri.uniza.sk 02 Feb '12

02 Feb '12

Hi, I am trying to make kamailio PCSCF working with rtpproxy (for NAT traversal). I would like to ask, if there is any tutorial for this, or if you can give me any advice. I also have installed rtpproxy package; #!define WITH_NAT, and defined rtpproxy port in config file... But this is not working. I noticed, that in default config file of pcscf is nothing in routing logic with rtpproxy (or it is unnecessary ?), so I try to copy route[RELAY] route[NAT] route[RTPPROXY] modules from kamailio (no pcscf) config file... but also with no success. Can you help me please ? What I am doing wrong ? Do you have any configuration working with rtpproxy ? Best wishes Lubo

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best way to start kamailio at boot
by Uri Shacked 01 Feb '12

01 Feb '12

Hi, what is the best way to start kamailio at boot? the example in the install is not working for me....

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subscriber specific dialplan
by Bayram Karagoz 01 Feb '12

01 Feb '12

Hi, I'm trying to make when each subscriber dials 7 digits add subscriber based specific prefix to dialed number defined in subscriber or dialplan table. I know that it's possible with avp but I'm not professional about avp. could you send me a successful scenario and config example please?

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redirect client to keep original ruri
by Andrew Pogrebennyk 01 Feb '12

01 Feb '12

Hello, assuming that I want to use contact from 302 response as outbound proxy but keep the original Request-URI, what should I do? Calling the revert_uri() after get_redirects() in failure_route doesn't do the trick.

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save a Q850 cause reason from a reply
by Uri Shacked 01 Feb '12

01 Feb '12

Hi guys..... the header searching and saving the Q850 code didn't work. here is what i wrote in the cfg, in the failure route: *# manage failure routing cases failure_route[MANAGE_FAILURE] { xlog("L_CRIT", "failure \n"); if($avp(CAR)=="CALLOUT"){ xlog("L_CRIT", "failure in callout \n"); ##if(is_present_hf("Reason")) { xlog("L_CRIT", "failure in reason\n"); $avp(Q850)=$(hdr(Reason){param.value,cause}{s.int}); xlog("L_CRIT", "failure in reason $avp(Q850)\n"); ##}* here is the log print...... * 3(3433) CRITICAL: <script>: failure 3(3433) CRITICAL: <script>: failure in callout 3(3433) CRITICAL: <script>: failure in reason 3(3433) ERROR: <core> [lvalue.c:347]: non existing right pvar 3(3433) ERROR: <core> [lvalue.c:407]: assignment failed at pos: (524,15-524,54) 3(3433) CRITICAL: <script>: failure in reason <null>* the command *is_present_hf("Reason")) * is under ## because of tests... when it was in use, the script didn't even went in the "if", the command didn't recognize the header. ** here is the sip trace for the 486, the header is there... * Message Header Call-ID: 6994099017399497785-1328029200-13379 From: <sip:36264529;ss7-cpc=10;cpc=ordinary@172.16.10.3 ;user=phone>;tag=6994099017399497785 To: <sip:39955011@172.16.60.161;user=phone>;tag=9299942026613191738 CSeq: 1 INVITE Via: SIP/2.0/UDP 172.16.60.161;received=172.16.60.161;branch=z9hG4bK08d6.c653f346.2,SIP/2.0/UDP 172.16.10.3:5060;branch=z9hG4bK-6110080000017439-ac100a03-1 Supported: timer,100rel Reason: Q.850;cause=17;text="UserBusy" Reason Protocols: Q.850 Cause: 17(0x11)[User busy] Content-Length: 0*

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[newbie] questions
by Me 31 Jan '12

31 Jan '12

Apologies if any of the questions below are a bit dumb - I don't pretend to be an expert in SIP/VOIP - I am just an ordinary user looking for answers. Our current setup involves processing a small number of internal sip accounts (up to 10, no more than that) and one "public" one (with a separate registrar) in the following way: On our server we have three interfaces: eth0, eth1 and tun0. eth0 is our entry point to the public internet, eth1 faces our internal network and tun0 is a private vpn, which connects all our smartphones to the internal network (via Wifi, EDGE/2G/3G etc). This gives us the mobility. Up until now, we have been routing voip calls via a commercial (closed source), very limited, terribly outdated (Pentium code base!) and rather buggy sip proxy. I had to employ a lot of hacks on our server in order to route calls as this proxy can only listen on a single interface. It was also a nightmare to maintain. Unsurprisingly, I decided that enough is enough and I am now determined to replace it. We route calls in the following way: all machines (PCs are all Linux based) & smartphones have their own sip/voip client installed on them (also using bluetooth). Internal calls are routed via the proxy between ourselves either on the internal net (eth1), or between the vpn and eth1 (eth1<->tun0). External calls (going out, i.e. outbound) are routed externally to our registrar, using a single separate voip account, via eth1<->eth0 or tun0<->eth0. As I am now looking to replace our proxy, I looked at Kamailio, but was soon completely overwhelmed by it (no offence intended, it was just too much to take at first). I would appreciate if any of you could give me a hand, or at least point me in the right direction, with the following issues: I presume I could configure Kamailio to listen on more than one interface and act as a proxy. How do I do that, so that it listens on all 3 interfaces and proxies requests in the following way: - calls made to <userX>@ourdomain.net to be routed internally via eth1 (internal net) or tun0 (private vpn); - calls made to anybody else to be routed externally via eth0 (public) using the separate "public" sip account with our external registrar; - calls made to the public sip account (from outside - the "public") need to be routed to a "nominated" internall account (say <user0>@ourdomain.net); - all other (internal) calls need to be routed depending on which interface this account has been registered/logged in - either the internal net (eth1) or the private vpn (tun0 - the smartphones). Obviously, calls need to be received (and routed properly) from all 3 interfaces. Is all of this possible with Kamailio? I want to avoid unnecessary complexities of the setup (as I already mentioned above - I am just a user and by no means an expert in sip/voip) and do not want to deploy something I do not need - I need to keep the memory footprint to a bare minimum, possibly without sacrificing performance. Once this is done, I would then move on to the next phase and use IM & ENUM, but this is once the above works. I looked at other alternatives, but I got very confused there as well - I couldn't figure out what exactly is the difference between, say, OpenSER, Kamailio, OpenSIPS and SIP-Router even? What is the best software to use in order to achieve the above setup? One last thing - I am a developer by trade and I am not afraid of "tweaking" things when needed. I was successful in compiling Kamailio from source (I use Fedora on all our machines) and I was pleased that I could exclude from the RPM .spec file the modules I think I did not need. I also made some modification of my own to make the database modules (mysql, postgresql and unixodbc) configurable in the same way the rest of the modules are. I could submit patches, if needed, so that these are incorporated into future releases - how do I do that? I could not do the same with OpenSIPS, however (which I also tried - out of curiosity!) - everything there seems to be lumped and compiled together regardless of whether it is needed or not. Any help as to helping me with the above issues is greatly appreciated, many thanks in advance for taking the time!

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Kamailio Proxy and Asterisk
by Greg Mannie 31 Jan '12

31 Jan '12

Hello, We have been using Asterisk for sometime and over the last year have started hosting instances for our clients on a vmware platform. These virtual pbx are located on public ip addresses and each customer has their own SIP trunk arrangements with various providers. We have decided to pool our resources and would like to start aggregating traffic. From reading online I have installed Kamailio 3.1.x with Asterisk 1.6.x. On top of this I have installed Siremis 3.2 in hopes of using it as a graphical front end. I am wanting to use Kamailio as a proxy to the SIP providers and allow the Asterisk to register only with Kamailio. Is there a link to some example configs with a tutorial pertaining to this type of deployment? If this is a good solution, it is my intention to recommend a consultant be hired to assist with a production server. I just need to learn allot more, before I can recommend Kamailio for the job. Do you think it's a good fit? Thank you for your kind responses, Greg

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Enable application/pidf+xml
by Vogt, Jonathan 31 Jan '12

31 Jan '12

Hello all, I'm trying to set up kamailio as a sip server with presence support for Bravis. So far I got the call processing running on either openser-1.3.2-3build1 or 3.2.0+lucid1 running. Problem is the notification is not using Content-Type: application/pidf+xml My openser config is at http://pastebin.ca/2106401 My kamailio config is pretty much a stock configuration. It is at http://pastebin.ca/2106404 I enabled mysql and presence. I also tried using x-lite but there is no presence information either. modparam("presence_xml", "force_active", 1) did not seem to make a difference. (I cleaned my watchers table) I can see the database has the information but the notify packets are just like this: NOTIFY sip:2001@10.0.76.0:56239 SIP/2.0 Via: SIP/2.0/UDP 10.0.10.23;branch=z9hG4bKcb76.b2dab2d4.0 To: sip:2001@sip.intern.domain;tag=90cd05fe2b0619108127da16a6e721e1 From: sip:2002@sip.intern.domain;tag=10.785.1327490724.1 CSeq: 22 NOTIFY Call-ID: 90cd05fe-2b06-1910-9660-da16a6e721e1 Content-Length: 0 User-Agent: OpenSER (1.3.2-notls (i386/linux)) Max-Forwards: 70 Event: presence Contact: <sip:10.0.10.23:5060> Subscription-State: active;expires=300 Where is my mistake. As I'm new to kamailio please bear with me, if I just didn't find the right switch. Thanks Jonathan Vogt

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IRC devel meeting
by Daniel-Constantin Mierla 31 Jan '12

31 Jan '12

Hello, I would like to schedule an IRC meeting to discuss about the development of the project in 2012. The topics coming in my mind at this moment: - roadmap to next major release v3.3.0 - outstanding issues at this time, if any - packaging - ways to automate the process better - documentation - suggestions to improve it and attract more contributions in this field - ideas for new features - administration - web site, mailing lists, etc ... do we miss some good resource out there that should be added in the eco-system? Please add your suggestions for agenda to the wiki page: * http://www.kamailio.org/wiki/devel/irc-meetings/2012a I proposed two dates (Jan 26 and Jan 31, 15:00GMT) and made a Doodle poll for it: * http://www.doodle.com/qzteweb62sqax5r6 If you wish other options for dates, let me know. Based on results, we will announce the final date. Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

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sr-users January 2012