sr-users March 2013

sr-users@lists.kamailio.org

92 participants
142 discussions

SIP and XMPP server integration
by Vinay V 11 Mar '13

11 Mar '13

Hi All, I am in process of integrating a sip server (Lync) with XMPP server (kamailio). Can a kamailio server be used as a sip to xmpp gateway and vice versa. If i am using any other xmpp server and want to integrate with lync will kamailio can be used as a gateway. This integration is maily for IM and presence. If it can be done what configuration should be done to kamailio.conf file. Please provide the inputs. I am very much need of it. If anybody has done already the above testing please provide DOCS. Thanks in advance, Vinay

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Problem with frequent use of FIFO commands
by Mino Haluz 11 Mar '13

11 Mar '13

Hi, I have some nagios scripts binded to kamctl fifo commands, as well as some web scripts that are directly calling kamctl fifo statements. I do not know why, but it sometimes returns ** ERROR: Error opening Kamailio's FIFO /tmp/kamailio_fifo [1] => ** ERROR: Make sure you have the line 'modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo")' in your config [2] => ** ERROR: and also have loaded the mi_fifo module. when it is called with web script with www-data account. From shell it works everytime, but with web 20% of results are errors. Other 80% are sucessful, so no problem with permissions. Does mi_fifo support multiple access, are there any other limits in this regard ? Within OS level? kamailio-3.3.4 Thanks, Mino

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Kamailio 4.0 install guide
by Jesús Pérez Rubio 11 Mar '13

11 Mar '13

Hi, I think there is a little error on this guide about 4.0 version install: http://www.kamailio.org/wiki/install/4.0.x/git First it is said this: "8. Edit configuration file To fit your requirements for the VoIP platform, you have to edit the configuration file. /usr/local/etc/kamailio/kamailio.cfg" At the end it is said that only this line of init.d script have to be changed: DAEMON=/usr/local/sbin/kamailio I would add this one too: CFGFILE=/usr/local/etc/kamailio/kamailio.cfg Another option would be to work directly with "/etc/kamailio/kamailio.cfg" Anyway, I take this oportunity to thank you for your great job and to notice that this version 4.0 seems to work stil better with our QoffeSIP Javascript SIP stack. :) -- Jesús Pérez VoIP Engineer at Quobis Fixed: +34 902 999 465 Site: http://www.quobis.com

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no destination sets in dispatcher module
by Ali Jawad 11 Mar '13

11 Mar '13

Hi I am using carrieroute to distribute traffic, I am not sure if there is a way for carrieroute to detect if a destination is down or not, but I read that the dispatcher module does that so I did compile it into the existing kamailio installation and added the following settings : loadmodule "dispatcher.so" modparam("dispatcher","db_url", "mysql://connectionstring") modparam("dispatcher", "table_name", "dispatcher") modparam("dispatcher", "setid_col", "setid") modparam("dispatcher", "destination_col", "destination") modparam("dispatcher", "flags_col", "flags") modparam("dispatcher", "priority_col", "priority") modparam("dispatcher", "force_dst", 1) and under ROUTE if ( !mf_process_maxfwd_header("10") ) { sl_send_reply("483","To Many Hops"); drop(); }; xlog("L_ERR","************************AFTER TOO MANY HOPS******************************************* cr\n"); ds_select_dst("1", "0"); forward(); # t_relay(); All I get is too many hops in ngrep and kamailio reports : (26467) ERROR: dispatcher [dispatch.c:1564]: no destination sets In the DB I do have a setid 1 and sip:ip.of.carrer:5060 as destination. Regards

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why use received as RURI for nathelper OPTIONS ping?
by Andrew Pogrebennyk 11 Mar '13

11 Mar '13

Some Fritzbox devices don't accept the OPTIONS ping generated by kamailio with RURI: sip:11.22.3.4:5060. In the location table we have: received: sip:11.22.3.4:5060 contact: sip:user@11.22.3.4:5060;uniq=6633BC1386F4D4CC4EBD64DC7E967 path: <sip:lb@127.0.0.1;lr;received='sip:11.22.3.4:5060'> Kamailio version is 3.3.2 and the nathelper config is nothing fancy: modparam("nathelper", "natping_interval", 15) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@sipwise.local") modparam("nathelper", "received_avp", "$avp(s:received)") What I don't understand is why kamailio sets RURI of the OPTIONS message to value of "received" instead of the contact. I suspect a bug in the parser somewhere along these lines: > rval = ul.get_all_ucontacts(buf,cblen,(ping_nated_only?ul.nat_flag:0), > ((unsigned int)(unsigned long)timer_idx)*natping_interval+iteration, > natping_processes*natping_interval); > if (rval != 0) { > pkg_free(buf); > goto done; > } > } > > if (buf == NULL) > goto done; > > cp = buf; > while (1) { > memcpy(&(c.len), cp, sizeof(c.len)); > if (c.len == 0) > break; > c.s = (char*)cp + sizeof(c.len); > cp = (char*)cp + sizeof(c.len) + c.len; > memcpy( &send_sock, cp, sizeof(send_sock)); > cp = (char*)cp + sizeof(send_sock); > memcpy( &flags, cp, sizeof(flags)); > cp = (char*)cp + sizeof(flags); > memcpy( &(path.len), cp, sizeof(path.len)); > path.s = path.len ? ((char*)cp + sizeof(path.len)) : NULL ; > cp = (char*)cp + sizeof(path.len) + path.len; > > /* determin the destination */ > if ( path.len && (flags&sipping_flag)!=0 ) { > /* send to first URI in path */ > if (get_path_dst_uri( &path, &opt) < 0) { > LM_ERR("failed to get dst_uri for Path\n"); > continue; > } > /* send to the contact/received */ > if (parse_uri(opt.s, opt.len, &curi) < 0) { > LM_ERR("can't parse contact dst_uri\n"); > continue; > } > } else {

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scheduling inside SER
by irma zujovic 11 Mar '13

11 Mar '13

Hello, I am using FOKUS IMS Core open source platform, and trying to investigate scheduling and queuing inside IMS. FOKUS IMS uses SER. My question is : where inside SER configuration scripts, the *scheduling *and *queueing *of SIP messages is done? I am interested in a special function or part of the written program. I hope that engineers that worked on developing of SER could help me with this issue. Thank you in advance Irma

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INVITE messages not authenticated (default configuration)?
by Paul Belanger 08 Mar '13

08 Mar '13

Greeting, Hopefully, I'm understanding the following default kamailio.cfg[1] file. Over the weekend, I was attached by SipVicious. Following along with the example Daniel[2] create with kamailio and asterisk, I have almost the same setup. Rather then storing my SIP profiles in Asterisk database, I have then in Kamailio. To my point, the attacker was actually able to by pass any sort of authentication, but simply sending an INIVTE message: ./svmap.py -e 18885551234 kamailio.example.org -m INVITE Which kamailio, forwarded to Asterisk and because there is no additional auth within asterisk, was able to hit the asterisk context for getting processed (they did not get out to the real world). However, my question is.... why do we not authenticate INVITE messages? If my understanding is correct, if would require something like the following: if (is_method("INVITE")) { if (!proxy_authorize("$fd", "subscriber")) { proxy_challenge("$fd", "0"); exit; } } If so, why not also do it in the default configuration file? [1] http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=blob_plain;f=et… [2] http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belanger(a)polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger

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Update existing module or create new?
by Charles Chance 08 Mar '13

08 Mar '13

All, We are working on a memcached module which uses the libmemcached library, as opposed to the (no longer under active development) libmemcache that the existing module depends on. Am I correct in thinking that in this instance, it would be better to create a brand new module (with a different name) rather than modify the existing one? In addition to changing the dependency, we will also be exporting new commands and potentially changing the format of the exported PVs. Regards, Charles -- www.sipcentric.com <http://www.sipcentric.com/> Follow us on twitter @sipcentric <http://twitter.com/sipcentric> Sipcentric Ltd. Company registered in England & Wales no. 7365592. Registered office: Unit 10 iBIC, Birmingham Science Park, Holt Court South, Birmingham B7 4EJ.

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Asterisk realtime with kamailio Load balancing issue for sip user
by Prakash N 07 Mar '13

07 Mar '13

Hi All, We have finished the Kamailio & Asterisk real time integration and load balancing also done using dispatcher module. Queue and voice mails are load balancing as well.When we are calling extension to extension it is showing in all the servers.It seems extension are not load balancing as per our knowledge. I have attached the kamailio.cfg for your reference,Find my coding below as mentioned. *# -- dispatcher params for DB support --* *modparam("dispatcher","db_url", "mysql:// openser:openserrw@192.168.1.170/openser")* *modparam("dispatcher", "table_name", "dispatcher")* *modparam("dispatcher", "setid_col", "setid")* *modparam("dispatcher", "destination_col", "destination")* *modparam("dispatcher", "flags_col", "flags")* *modparam("dispatcher", "priority_col", "priority")* * * * ----------------------------------------------------------------------------------------- * *# Dispatch requests* *route[DISPATCH] {* *if ( method=="INVITE" ) {* *# dst_select( "GROUP", "HASH METHOD")* * ds_select_dst("1","4");* * sl_send_reply("100","Trying");* * forward();#uri:host, uri:port);* * exit();* *}}* Kindly suggest the solution for the same. Thanks in advance. Regards, N.Prakash

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Re: [SR-Users] scheduling inside SER
by Jiri Kuthan 07 Mar '13

07 Mar '13

not sure if there is a point in doing that, at least for incoming traffic since it already happens in kernel, unless you want to do it in kernel. jiri On 3/7/13 2:25 PM, irma zujovic wrote: > Thank you for your answer. > > So is there any way to produce some kind of algorithm for prioritization of SIP messages inisde FOKUS IMS Core platform by configuring SER? That was my first > goal, when asking the previous question. I tought that scheduling and buffering of messages inside FOKUS IMS is done inside SER, and tought about priorituzation > inside SER. > > If you can help me in any way i would be very greatful. > > Thank you in advance > > Irma > > On Tue, Mar 5, 2013 at 4:13 PM, Jiri Kuthan <jiri(a)iptel.org <mailto:jiri@iptel.org>> wrote: > > On 3/5/13 2:36 PM, irma zujovic wrote: > > Hello, > > I am using FOKUS IMS Core open source platform, and trying to investigate scheduling and queuing inside IMS. FOKUS IMS uses SER. > > My question is : where inside SER configuration scripts, the *scheduling * and *queueing *of SIP messages is done? I am interested in a special function > or part > of the written program. > > > this is not done in SER, it is done in kernel. kernel dispatches incoming traffic to > one of multiple processes listening on sockets. Same for outgoing traffic -- SER leaves > it to kernel's decision. > > jiri > > > I hope that engineers that worked on developing of SER could help me with this issue. > > Thank you in advance > > Irma > > > _________________________________________________ > 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 <http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users> > >

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sr-users March 2013