- sr-users - kamailio.org

File kamailio_fifo is removed...
by Bruno Bresciani 12 Mar '13

12 Mar '13

Hi, Kamailio is running fine, but when I try run again by command line without killing the previous, the kamailio_fifo is removed and functions fifo in kamctl doen't work anymore. Why when I try running kamailio when already exist one, it remove de file kamailio_fifo? I can't find the part of source code that file kamailio_fifo is removed... Someone knows? Best Regards

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Kamailio MySQL DB Creation
by Amr Elsharqawi 12 Mar '13

12 Mar '13

Hello Kamailio Community!! I am newbie to Kamailio & SIP. And I am interested in Kamailio. I've used asterisk for some time then I heard about FreeSWITCH unfortunately FreeSWITCH was not good enough for me like Asterisk. I've found NAT problem with Asterisk Finally I've decided to work with Kamailio as SIP server & Asterisk as PBX. to be fair I need to learn more about FreeSWICH to test it well. My Question is about Kamailio installation; I am using FreeBSD 9 attached my uname -a FreeBSD PBX.XXXXX 9.1-RELEASE FreeBSD 9.1-RELEASE #0 r243826: Tue Dec 4 06:55:39 UTC 2012 root@obrian.cse.buffalo.edu:/usr/obj/usr/src/sys/GENERIC i386 I've used gmake for building, compiling and installation. Everything goes smooth EXCEPT mysql db installation when I use sh /usr/local/sbin/kamdbctl create I am getting the following error: /usr/local/lib/kamailio//kamctl/kamdbctl.base: 36: Syntax error: Missing '}' Thanks a lot for your valuable time and that amazing project (Kamailio). Amr

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Strange TLS issue
by Jirka Jirout 12 Mar '13

12 Mar '13

Hello, I have a very strange issue - we are running kamailio 3.3.2 as a part of Sip Provider CE and some of our clients have issues with TLS connectivity. 1. some clients such as Jitsi on Mac OS X and Bria or idoubs on iOS can always connect 2. imsdroid on Android can sometimes connect and works (about 10 % of attempts) When I run tcpudmp on the server during the unsuccessfull attempts, I see something like this: > 1 1 0.0347 (0.0347) C>S SSLv2 compatible client hello > Version 3.1 > cipher suites > TLS_RSA_WITH_AES_128_CBC_SHA > 1 2 0.0360 (0.0013) S>C Handshake > ServerHello > Version 3.1 > session_id[0]= > cipherSuite TLS_RSA_WITH_AES_128_CBC_SHA > compressionMethod NULL > 1 3 0.0361 (0.0001) S>C Handshake > Certificate > 1 4 0.0361 (0.0000) S>C Handshake ServerHelloDone > 1 5 0.0782 (0.0420) C>S Alert > level fatal > value unexpected_message > 1 0.0785 (0.0002) S>C TCP FIN > 1 6 0.1092 (0.0307) C>S Handshake > ClientKeyExchange > 1 7 0.1092 (0.0000) C>S ChangeCipherSpec > 1 8 0.1092 (0.0000) C>S Handshake What I find particularly strange is that the client sends fatal alert and then attempts to continue with handshake. What exactly does this "unexpected_message" mean? Is the client letting the server know the last message was unexpected? Or is this telling me the server had not expected the message? It cannot be a certificate problem, can it? Considering other clients were fine, I blamed our code at first, but I am no longer sure that is the (only) reason for this behavior. I decided to start from scratch and wrote an extremely simple application in plain C, that connects to the TLS port using openssl (which imsdroid also uses), sends some query and then prints response. It works fine on https sites, even on the same server kamailio is running and with the same certificates (our certificates come from Verisign CA G3). When I run ssldump, I can see handshake is fine, socket opened, the client is sending application data. I can see data coming in on the server using tcpdump. However, the data never reaches Kamailio - no sign of the packets in the log and no answer whatsoever (I use a fake REGISTER message so I would expect to get something like "Unauthorized" back). The same happens, when I try to use openssl s_client to connect to the site and send "register" message (because R at the start of packet triggers renegotiation in s_client, uff). My TLS settings in Kamailio are pretty much default, only using my own key and certificate. regards, jjj

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Re: [SR-Users] memcached module problems
by Henning Westerholt 12 Mar '13

12 Mar '13

Hi Dragos, (setting sr-users to CC) Ok, If you’ve set this by other means you need to set the appropriate “Kamailio” type flag to the value as well, that Kamailio can parse it correctly. You can get the appropriate value if you grep for “VAR_VAL_STR” in the source code. Your perl script function call should also export the appropriate flag setting from the memcache library function call, then just add it here. Best regards, Henning From: Dragos Oancea [mailto:droancea@yahoo.com] Sent: Monday, March 11, 2013 8:24 PM To: Henning Westerholt Subject: Re: memcached and other things Hi Henning, Other means. a perl script. Dragos > 0(21130) DEBUG: memcached [mcd_var.c:87]: allocate new memcache request at > 0x7f747fb798f0 > 0(21130) DEBUG: memcached [mcd_var.c:93]: allocate new > memcache result at 0x7f747fb79a70 0(21130) > DEBUG: memcached [mcd_var.c:103]: result: 'just_A_text' for key 888_HA1 with > flag 0 > 0(21130) ERROR: memcached [mcd_var.c:146]: could not convert string > 'just_A_text' to integer value > 0(21130) DEBUG: memcached [mcd_var.c:159]: free memcache request and result > at 0x7f747fb798f0 The issue is that the module somehow does not evaluate the string variable _from memcached_ as a string, it tries to evaluate it as integer. This obviously fails. Has this value set from kamailio or is this from some external application? Then probably this application does not set the proper flags which kamailio expects. Regards, Henning Westerholt

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RPMs for v4.0.0
by Daniel-Constantin Mierla 12 Mar '13

12 Mar '13

Many RPM packages are are now available on OpenSUSE build server (for Centos 5.5 and 6, RedHat 5 and 6, Fedora 16, 17 and 18, OpenSuse 11.4, 12.1, 12.2 and 12.3, on i586 and x86_64 architectures). Links to repositories are at: - http://www.kamailio.org/wiki/packages/rpms There is listed the link to Peter's RPM repository as well. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

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Routing calls
by mark＠brightvoip.co.uk 12 Mar '13

12 Mar '13

Hi, I'm looking for pointers on the following scenario. I have two sites (A and B) and sip traffic is sent from A to B. At each site there are a number of SIP (Asterisk) servers and a Kamailio server. The two sites have various network routes (X, Y & Z) between them. I have been asked to load balance the inter-site SIP traffic across the available links with a 20/20/60 traffic split (dispatcher, carrierroute). Each Kamailio server has multiple IP addresses and these are used to direct traffic to specific routes (Kamailio at site A IP 111.111.111.1 to Kamailio site B 111.111.111.2 defines route X for example). For any given call, the complete call (SIP and both RTP streams) should travel wholly over one route. I would welcome any suggestions about how best to route the whole call (sip & rtp) over a specific route, I was working on a solution with multiple RTPProxy instances, but was wondering if this is the best solution. Thanks in advance. Mark

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Kamailio v4.0.0 Released - new major version is out
by Daniel-Constantin Mierla 11 Mar '13

11 Mar '13

Kamailio v4.0.0 is out – a special major release, shifting up first number of versioning string! It is the landmark for the end of Kamailio-SER integration, bringing out as well a very large set of new features and improvements, like websocket support or dozens of IMS extensions. You can read detailed release notes at: * http://www.kamailio.org/w/kamailio-v4-0-0-release-notes/ Many thanks to all developers and community members that made possible this release. Enjoy Kamailio v4.0.0! Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

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releasing v4.0.0 tomorrow
by Daniel-Constantin Mierla 11 Mar '13

11 Mar '13

Hello, the plan is to release final Kamailio v4.0.0 tomorrow, Monday, March 11, 2013. If any developer/packager has something to add to the release, please commit by 12:00GMT to GIT branch 4.0. After that, announce first on sr-dev and wait for an ack, to avoid packaging inconsistent content. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

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kamailio is returning 500
by chanea 11 Mar '13

11 Mar '13

Dear all While lots of request come to my Kamailio 3.1.x in the same time , server is returning "500, Server Internal Error" to some of the request. What should i do if i want to reduce the number of "500" response? chanea

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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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Wrong ACK packet sent from Cisco gw
by Nawfel Oujdi 07 Mar '13

07 Mar '13

Hello! I m facing the same strange behaviour with my AS5300 voice gateway. When the gw is connected directly to PBX everythings works well but when i put a sip proxy forwarding calls between gw and PBX all the calls hangs up after 5 sec (+or -). Looking into the trace sip i realize that gw send a wrong ACK in reply of INVITE , then sip proxy discard it and PBX hangs the call cause he never receive the ACK. ACK sip:79.125.120.12:5060;lr=on;did=ce.3716ea02 SIP/2.0 Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw" From: <sip:911873699@cisco_gw>;tag=65FB8-B18 Route: <sip:911111500@PBX:5060> To: <sip:911111500@sip_proxy>;tag=as7f388e3f Date: Mon, 17 Jan 2011 09:26:36 GMT Call-ID: B6F61A2E-215211E0-802BD462-C4432B89@cisco_gw To work fine , the content of Route header should be in ACK header and viceversa. I tried to compare between the sip trace of a wrong call and a good one (using other cisco gw AS5350 who works well with sip proxy in the same escenario) and i realize that the only difference is the INVITE of wrong case doesn' t send branch number in the via header. INVITE sip:911111500@sip_proxy:5060 SIP/2.0 Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw" From: <sip:911873699@cisco_gw>;tag=65FB8-B18 To: <sip:911111500@sip_proxy> i m using c5300-is-mz.123-26.bin ios version. Anybody understand what is happening in there?? is there any solution?? i ll send more information if it s requested. Thanks in advance. Nawfel Oujdi here is the result of ngrep: U 2011/01/13 15:14:43.791514 cisco_gw:51703 -> sip_server:5060 INVITE sip:911111500@sip_server:5060 SIP/2.0. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>. Date: Thu, 13 Jan 2011 14:14:43 GMT. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. Supported: timer,100rel. Min-SE: 1800. Cisco-Guid: 1295951687-508957152-2608788105-28919687. User-Agent: Cisco-SIPGateway/IOS-12.x. Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO. CSeq: 101 INVITE. Max-Forwards: 6. Remote-Party-ID: <sip:911873699@cisco_gw >;party=calling;screen=yes;privacy=off. Timestamp: 1294928083. Contact: <sip:911873699@cisco_gw:5060>. Expires: 180. Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 270. . v=0. o=CiscoSystemsSIP-GW-UserAgent 8894 2421 IN IP4 cisco_gw. s=SIP Call. c=IN IP4 cisco_gw. t=0 0. m=audio 16924 RTP/AVP 18 101. c=IN IP4 cisco_gw. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. U 2011/01/13 15:14:43.791893 sip_server:5060 -> cisco_gw:5060 SIP/2.0 100 Giving a try. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. CSeq: 101 INVITE. Server: OpenSIPS (1.6.3-notls (i386/linux)). Content-Length: 0. . U 2011/01/13 15:14:43.791957 sip_server:5060 -> asterisk_server:5060 INVITE sip:911111500@sip_server:5060 SIP/2.0. Record-Route: <sip:sip_server;lr=on;did=015.864b8107>. Via: SIP/2.0/UDP sip_server;branch=z9hG4bK3e35.3ed9b366.0. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>. Date: Thu, 13 Jan 2011 14:14:43 GMT. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. Supported: timer,100rel. Min-SE: 1800. Cisco-Guid: 1295951687-508957152-2608788105-28919687. User-Agent: Cisco-SIPGateway/IOS-12.x. Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO. CSeq: 101 INVITE. Max-Forwards: 5. Remote-Party-ID: <sip:911873699@cisco_gw >;party=calling;screen=yes;privacy=off. Timestamp: 1294928083. Contact: <sip:911873699@cisco_gw:5060>. Expires: 180. Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 270. . v=0. o=CiscoSystemsSIP-GW-UserAgent 8894 2421 IN IP4 cisco_gw. s=SIP Call. c=IN IP4 cisco_gw. t=0 0. m=audio 16924 RTP/AVP 18 101. c=IN IP4 cisco_gw. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. U 2011/01/13 15:14:43.792775 asterisk_server:5060 -> sip_server:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP sip_server;branch=z9hG4bK3e35.3ed9b366.0;received=sip_server. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". Record-Route: <sip:sip_server;lr=on;did=015.864b8107>. From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. CSeq: 101 INVITE. Server: Asterisk PBX 1.6.2.13. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. Require: timer. Session-Expires: 1800;refresher=uas. Contact: <sip:911111500@asterisk_server>. Content-Length: 0. . U 2011/01/13 15:14:43.793770 asterisk_server:5060 -> sip_server:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP sip_server;branch=z9hG4bK3e35.3ed9b366.0;received=sip_server. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". Record-Route: <sip:sip_server;lr=on;did=015.864b8107>. From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>;tag=as19e8a82f. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. CSeq: 101 INVITE. Server: Asterisk PBX 1.6.2.13. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. Require: timer. Session-Expires: 1800;refresher=uas. Contact: <sip:911111500@asterisk_server>. Content-Type: application/sdp. Content-Length: 260. . v=0. o=root 1750021131 1750021131 IN IP4 asterisk_server. s=Asterisk PBX 1.6.2.13. c=IN IP4 asterisk_server. t=0 0. m=audio 10798 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. U 2011/01/13 15:14:43.794688 sip_server:5060 -> cisco_gw:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". Record-Route: <sip:sip_server;lr=on;did=015.864b8107>. From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>;tag=as19e8a82f. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. CSeq: 101 INVITE. Server: Asterisk PBX 1.6.2.13. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. Require: timer. Session-Expires: 1800;refresher=uas. Contact: <sip:911111500@asterisk_server>. Content-Type: application/sdp. Content-Length: 316. . v=0. o=root 1750021131 1750021131 IN IP4 79.125.41.121. s=Asterisk PBX 1.6.2.13. c=IN IP4 79.125.41.121. t=0 0. m=audio 10798 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. a=oldmediaip:asterisk_server. a=oldmediaip:asterisk_server. U 2011/01/13 15:14:43.856520 cisco_gw:57947 -> sip_server:5060 ACK sip:sip_server:5060;lr=on;did=015.864b8107 SIP/2.0. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>;tag=as19e8a82f. Date: Thu, 13 Jan 2011 14:14:43 GMT. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. Route: <sip:911111500@asterisk_server:5060>. Max-Forwards: 6. Content-Length: 0. CSeq: 101 ACK. . U 2011/01/13 15:14:43.993417 asterisk_server:5060 -> sip_server:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP sip_server;branch=z9hG4bK3e35.3ed9b366.0;received=sip_server. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". Record-Route: <sip:sip_server;lr=on;did=015.864b8107>. From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>;tag=as19e8a82f. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. CSeq: 101 INVITE. Server: Asterisk PBX 1.6.2.13. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. Require: timer. Session-Expires: 1800;refresher=uas. Contact: <sip:911111500@asterisk_server>. Content-Type: application/sdp. Content-Length: 260. . v=0. o=root 1750021131 1750021131 IN IP4 asterisk_server. s=Asterisk PBX 1.6.2.13. c=IN IP4 asterisk_server. t=0 0. m=audio 10798 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. U 2011/01/13 15:14:43.993613 sip_server:5060 -> cisco_gw:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". Record-Route: <sip:sip_server;lr=on;did=015.864b8107>. From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>;tag=as19e8a82f. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. CSeq: 101 INVITE. Server: Asterisk PBX 1.6.2.13. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. Require: timer. Session-Expires: 1800;refresher=uas. Contact: <sip:911111500@asterisk_server>. Content-Type: application/sdp. Content-Length: 316. . v=0. o=root 1750021131 1750021131 IN IP4 79.125.41.121. s=Asterisk PBX 1.6.2.13. c=IN IP4 79.125.41.121. t=0 0. m=audio 10798 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. a=oldmediaip:asterisk_server. a=oldmediaip:asterisk_server. U 2011/01/13 15:14:44.038774 cisco_gw:57947 -> sip_server:5060 ACK sip:sip_server:5060;lr=on;did=015.864b8107 SIP/2.0. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>;tag=as19e8a82f. Date: Thu, 13 Jan 2011 14:14:43 GMT. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. Route: <sip:911111500@asterisk_server:5060>. Max-Forwards: 6. Content-Length: 0. CSeq: 101 ACK. . U 2011/01/13 15:14:44.193431 asterisk_server:5060 -> sip_server:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP sip_server;branch=z9hG4bK3e35.3ed9b366.0;received=sip_server. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". Record-Route: <sip:sip_server;lr=on;did=015.864b8107>. From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>;tag=as19e8a82f. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. CSeq: 101 INVITE. Server: Asterisk PBX 1.6.2.13. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. Require: timer. Session-Expires: 1800;refresher=uas. Contact: <sip:911111500@asterisk_server>. Content-Type: application/sdp. Content-Length: 260. . v=0. o=root 1750021131 1750021131 IN IP4 asterisk_server. s=Asterisk PBX 1.6.2.13. c=IN IP4 asterisk_server. t=0 0. m=audio 10798 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. -- ------------------------------ Aviso legal: Este mensaje electrónico está dirigido únicamente a la(s) dirección(es) indicadas anteriormente; el carácter confidencial, personal e intransferible del mismo está protegido legalmente. Cualquier revelación, uso o reenvío no autorizado, completo o en parte, está prohibido. Si ha recibido este mensaje por equivocación, notifíquelo inmediatamente a la persona que lo ha enviado y borre el mensaje original junto con sus ficheros anexos sin leerlo ni grabarlo, total o parcialmente.

2 1

installing v4.0.x from git
by Daniel-Constantin Mierla 07 Mar '13

07 Mar '13

Hello, I created the wiki page with the how-to install kamailio v4.0.x from git. The tutorial is available at: - http://www.kamailio.org/wiki/install/4.0.x/git It should make life easier for those willing to give a try to v4.0.0 these days. Anyhow, a review of the content is very appreciated. Feel free to enhance the content by adding to the wiki page or reply here with your ideas to improve. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

4 7

Fwd: kamailio rls mod
by Aleksandrs Semenenko 06 Mar '13

06 Mar '13

Hello, I'm looking for assistance in RLS module setup for kamailio 3.3.2 The problem is that I need to handle resource-list updates on my kamalio server. For testing purposes I use Jitsi client. Expected: 1) When a new contact is added to Jitsi or an old one is removed, Jitsi sends updated resource-list XML to caps. - This works fine, I can see PUT request in Wireshark and I can see following log from kamailio.cfg: xlog("===== xhttp put: refreshing resource-list for $var(uri)\n"); rls_update_subs("$var(uri)", "presence"); which hopefully means that rls_update_subs function has been called. 2) After this function was called, I expect SUBSCRIBE messages in backend, not in Jitsi, but I still see Jitsi sending SUBSCRIBE for each contact. What am I doing wrong? Any help would be useful! Regards, Alex.

4 3

New developer: Hugh Waite
by Daniel-Constantin Mierla 06 Mar '13

06 Mar '13

Hello, I want to announce that a new person got developer GIT write access to repository: Hugh Waite - he is quite old contributor by now, with many patches and activity on mailing lists. He is part of the Crocodile RCS team, which developed very important features for Kamailio so far, like websocket and outbound extensions, Lua API and presence/rls improvements. His git commit id is: hpw My warm welcome and looking forward to future work within the project! Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

2 1

early rpm and deb packages from branch 4.0
by Daniel-Constantin Mierla 06 Mar '13

06 Mar '13

Hello, if anyone is preferring to use packages for pre-release testing of v4.0.0, I created rpms for centos, redhat, opensuse and fedors, plus packages for debian squeeze. You can find them on opensuse build service servers: - https://build.opensuse.org/package/show?package=kamailio40&project=home%3Ak… - https://build.opensuse.org/package/show?package=kamailio40-d6.0&project=hom… I made them mainly to be sure there are no issue compiling on common linux OSes. Debian packages should be available at the usual repository for more versions (including ubuntu), courtesy of Jon Bonilla. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

1 0

Kamailio Geoip
by Ian French 06 Mar '13

06 Mar '13

Hey List, I'm looking at an article by Daniel, http://by-miconda.blogspot.ie/2010/10/best-of-new-in-kamailio-310-5-geoip-a…. In it he's uses the below to block source IP addresses by country, How could this be used to block multiple country sources?? I've tried a couple of things aleardy but have failed geoip_match("$si", "src"); if($gip(src=>cc)=="CA") { xlog("SIP message from Canada (ip: $si) - drop it\n"); send_reply("403", "Forbidden"); exit; } Thanks in advance Ian

3 4

Kamailio World Updates - March 06
by Daniel-Constantin Mierla 06 Mar '13

06 Mar '13

Hello, the schedule for Kamailio World Conference has been updated, among topics of the recently added talks: North American VoIP market, Jitsi softphone, usage in embedded systems and enterprises. Check the current schedule at: - http://conference.kamailio.com/k01/schedule/ The group of speakers for the two days is filled up at this moment, as soon as we received the profiles and presentation summaries, the schedule will get final. Regarding Kamailio project, these are the last days before the next major release, version 4.0.0 is planned to be out next week. This week is CeBIT in Hanover, many community members will visit the event in various days (to my knowledge, Thursday and Friday will be the busiest). If you want face to face meetings, announce yourself on our forums. I plan to be there tomorrow. It is just a bit over one month till the event starts, if you haven't registered yet, you should hurry up, there is no similar event in Europe this year that gives the chance to meet and learn from many professionals working with rich communication services and products: - http://conference.kamailio.com/k01/registration/ Also, if you consider sponsoring or participate to surrounding exhibition space, please get in touch with us, you will get an exceptional visibility and additional benefits at the event: - http://conference.kamailio.com/k01/sponsors/ Do not forget that there are open positions to work with VoIP/SIP and Kamailio, just in case you look for new challenges in your career, check: - http://conference.kamailio.com/k01/career-jobs/ Looking forward to meet many of you in Berlin, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

1 0

Simple look table
by Gareth Rylance 05 Mar '13

05 Mar '13

Is it possible to have a custom look-up table to map ids to other ids in the configuration file? E.G. TestLookValues key | value 1 |100 2 | 100 3 | 600 lookup("TestLookValues", "3") gives $var(lookupval) equal to 600 It is similar to MTree functionality only I am looking for an exact match not just a prefix TestLookValues prefix | value 1 |100 2 | 100 3 | 600 mt_match("TestLookValues", "3003)", "0") gives $var(mtval) = 600 I would need $var(mtval) = -1 (not found) Thanks Gareth

2 1

Troubles with call between 2 users
by TF Julian Plata 05 Mar '13

05 Mar '13

Hello, I'm Julian Plata and I have some troubles to do a call between 2 users: julianplata(a)iptel.org since Media5 in my phone, and marinastereo(a)iptel.org in a Comrex box. When I try to connect, Media5 shows a message "The caller can not be found". Also, in the Iptel User Management the users julianplata@iptel & marinastereo@iptel are shown as online. Could you help me to fix the issue ? Could you verify if the accounts are OK for establish a connection?? Thanks, Best Regards, Julian Plata -- ------------------------------------------------------------------------------------ La informacion contenida en este E-mail es confidencial y solo puede ser utilizada por el individuo o la compania a la cual esta dirigido. Si no es el receptor autorizado, cualquier retencion, difusion, distribucion o copia de este mensaje es prohibida y sera sancionada por la ley. Si por error recibe este mensaje, favor reenviarlo al remitente y borrar el mensaje recibido inmediatamente. Cualquier punto de vista expresado en este mensaje es unicamente de quien lo envia excepto donde especificamente se indique que es la posicion de la Armada Nacional de Colombia (ARC). ------------------------------------------------------------------------------------ Apaga la pantalla de su computador cuando salgas a almorzar. No malgastemos la Energía. Cuidemos lo que es de Todos. ------------------------------------------------------------------------------------ Antes de imprimir este e-mail piensa bien si es necesario hacerlo: 1 hoja de papel tamaño carta = 2 litros de agua + 10 gramos de materia prima (decolorantes, fijadores, pigmentos) + 1 bombillo de 40 Vatios prendido por 1 hora + 14 gramos de madera pura.

2 1

Kamailio problem with both clients in the same NAT
by mohavere36 04 Mar '13

04 Mar '13

4 4

releasing v4.0.0
by Daniel-Constantin Mierla 04 Mar '13

04 Mar '13

Hello, from my point of view, the last to sort out for v4.0.0 is packaging (I just sent a dedicated email to decide on that). Personally I have been testing v4.0.0 thoroughly in various cases, no critical issue being on my list at this moment. Also, v4.0 is running for quite some time on voipuser.org (our usual life testing platform). I propose releasing v4.0.0 next Monday, on March 11, 2013. The second option, if more people prefer it, would be Thursday, March 14, 2013. Opinions? If anybody has other dates in mine, let us know here. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

1 0

packaging v4.0.0
by Daniel-Constantin Mierla 04 Mar '13

04 Mar '13

Hello, (cross-posting as this decision may be done properly with feedback from both developers and users community). Now that we have some band new modules but also several that were not packaged by Kamailio, just by ser (due to single modules directory), I was thinking we may want to reconsider a bit the packages we do for v4.0.0. 1) A new package is related to the bunch of IMS modules, that is pretty much obvious. 2) There are few modules targeting development samples or troubleshooting: print, print_lib and malloc_test - I think they should not be packaged. 3) I would create a special package for the modules using uid-based database schema: db2_ops uid_auth_db uid_avp_db uid_domain uid_gflags uid_uri_db uri_db 4) Maybe we should merge some packages by dependencies: - a) some depend on libxml2: cpl-c xhttp_pi xmlrpc xmlops - b) some depend on libssl: tls auth_identity outbound 5) Some experimental modules with external deps will stay unpackaged like so far: db_oracle Opinions? Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

1 0

When to rewrite o= in SDP ?
by Khoa Pham 04 Mar '13

04 Mar '13

Hi, I read in http://kamailio.org/docs/modules/1.4.x/nathelper.html that Kamailio might rewrite ip address in o= in SDP to facilitate NAT traversal. It may decide to use to original IP or use IP of RTP server. What factors affect its decision ? -- Khoa Pham HCMC University of Science Faculty of Information Technology

2 3

UnixODBC cant insert sip_capture record on sql server
by Grant Bagdasarian 03 Mar '13

03 Mar '13

Hello, We are using SQL Server 2008 as our database. When the sip_capture module tries to insert a row, the following error is given: db_unixodbc [con.c:220]: unixodbc:SQLExecDirect=42000:1:156:[FreeTDS][SQL Server]Incorrect syntax near the keyword 'authorization'. Using Kamailio 3.3.4. Is there a way to force brackets around the column values? Regards, Grant

2 3

uac_replace_from / New URI shorter than old URI
by Ronald Voermans 03 Mar '13

03 Mar '13

Hi, this last weekend we went live from the old 3.0 version to the latest stable release of Kamailio (3,3,0). All went well, except for one issue. We make use of uac_replace_from in our config, in case a client needs CLIR: uac_replace_from("Anonymous","sip:anonymous@anonymous.invalid"); Since the upgrade we receive the following error: uac [replace.c:521]: new URI shorter than old URI Which causes the call to be disconnected after about 20/30 seconds. This, because the ACK is received by Kamailio, but cannot be 'linked' to the transaction i guess. What is exactly causing this issue and, more importantly, is there any way to resolve this? Thanks in advance, Regards, Ronald Voermans

2 4

Which is first, t_on_reply or callbacks in C ?
by David 03 Mar '13

03 Mar '13

Hello, Can someone tell me which is executed first ? the on reply route setup with t_on_reply() or the callbacks in the code that allows the module dialog to insert the confirmed dialog into the dialog table ? Thanks, David

2 1

not able to install Kamailio
by Julián Quintero 03 Mar '13

03 Mar '13

Hello thereI hope that you can help me, I have an issue installing Kamailio, when I issue th command "make all" I receive the following error resolve.h:471:4: error: #error neither gethostbyname2 or getipnodebyname present make: *** [action.o] Error 1 I am using debian squeeze, please let me know if there is something missing in my system or what is not allowing to install the SIP Server THanks Julián

3 3

t_lookup.c: WARNING: script writer didn't release transaction - problem
by Efelin Novak 03 Mar '13

03 Mar '13

Hi folks, we have a serious issue with tm module. In Fail route we would like to send a reply 404 when a cr_route function fails(no destination found). We use t_reply("404", "No route to destination"). However after the call is disconnected with a 404 reply the tm module writes a following warning WARNING: tm [t_lookup.c:1543]: WARNING: script writer didn't release transaction Very soon after this error message is shown a segfault occurs. Log is as follows: Mar 3 15:26:46 hostname kamailio[791]: ERROR: carrierroute [cr_func.c:594]: desired routing domain doesn't exist, prefix 0039333333333, carrier 14, domain 2 Mar 3 15:26:46 hostname kamailio[791]: WARNING: tm [t_lookup.c:1543]: WARNING: script writer didn't release transaction //ACK is received Mar 3 15:26:52 hostname kernel: [771134.093250] kamailio[796]: segfault at 4 ip 0815c182 sp bf8ace00 error 6 in kamailio[8048000+1ea000] Mar 3 15:26:53 hostname kamailio[803]: : <core> [pass_fd.c:293]: ERROR: receive_fd: EOF on 15 Mar 3 15:26:53 hostname kamailio[790]: ALERT: <core> [main.c:785]: child process 796 exited by a signal 11 Mar 3 15:26:53 hostname kamailio[790]: ALERT: <core> [main.c:788]: core was generated Mar 3 15:26:53 hostname kamailio[790]: INFO: <core> [main.c:800]: INFO: terminating due to SIGCHLD Mar 3 15:26:53 hostname kamailio[802]: INFO: <core> [main.c:851]: INFO: signal 15 received Mar 3 15:26:53 hostname kamailio[799]: INFO: <core> [main.c:851]: INFO: signal 15 received Mar 3 15:26:53 hostname kamailio[801]: INFO: <core> [main.c:851]: INFO: signal 15 received Mar 3 15:26:53 hostname kamailio[798]: INFO: <core> [main.c:851]: INFO: signal 15 received Mar 3 15:26:53 hostname kamailio[800]: INFO: <core> [main.c:851]: INFO: signal 15 received Mar 3 15:26:53 hostname kamailio[797]: INFO: <core> [main.c:851]: INFO: signal 15 received Mar 3 15:26:53 hostname kamailio[792]: INFO: <core> [main.c:851]: INFO: signal 15 received Mar 3 15:26:53 hostname kamailio[803]: INFO: <core> [main.c:851]: INFO: signal 15 received Mar 3 15:26:53 hostname kamailio[795]: INFO: <core> [main.c:851]: INFO: signal 15 received Mar 3 15:26:53 hostname kamailio[791]: INFO: <core> [main.c:851]: INFO: signal 15 received Mar 3 15:26:53 hostname kamailio[794]: INFO: <core> [main.c:851]: INFO: signal 15 received Mar 3 15:26:53 hostname kamailio[793]: INFO: <core> [main.c:851]: INFO: signal 15 received Mar 3 15:26:53 hostname kamailio[790]: INFO: <core> [mem/f_malloc.c:532]: freeing a free fragment (0x886b3ad8/0x886b3ae0) - ignore Mar 3 15:26:53 hostname kernel: [771135.681720] kamailio[790]: segfault at 7dd193e0 ip 0816f5cd sp bf8acc20 error 4 in kamailio[8048000+1ea000] Currently we have fixed it using a custom route route(T_REPLY); . . . route[T_REPLY] { sl_send_reply("404","No route to destination"); } This works correctly with no error or a warning message or a segfault. We are using kamilio version: kamailio 3.3.4 (i386/linux) f8c8f2 Is this behavior a bug or should I dig deeper into the tm documentation? Kind regards Efelin

2 1

Karl Smith
by Karl Smith 02 Mar '13

02 Mar '13

witwcvrf ictniydtoz.vsd/http://www.loudounwalkin.com/sq/f7nax0z0n2ouq77adgkgejjvnz.g… yklxx.vsd/knc hj.vsd/

1 0

RPM packages for v3.3.4
by Daniel-Constantin Mierla 01 Mar '13

01 Mar '13

Hello, the RPM packages for latest minor release v3.3.4 were created on openSuse build service. More details and download links at: - http://www.kamailio.org/wiki/packages/rpms Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

1 0

Karl Smith
by Karl Smith 01 Mar '13

01 Mar '13

http://www.louiselemelin.com/sinkzm/hhir5ew9drwu4s54d429jdnj.d?a6dfinu09fqc… Smith3/1/2013 3:00:01 PM

1 0

How to let Kamailio to always use RTP proxy
by Khoa Pham 01 Mar '13

01 Mar '13

Hi, Supposed A call B in a non-NAT environment. How to let Kamailio to always force A, B to use RTP proxy ? -- Khoa Pham HCMC University of Science Faculty of Information Technology

4 4

Help with Asterisk RT integration
by Jon Morby 28 Feb '13

28 Feb '13

Hi I'm trying to integrate a (K) front end cluster with an Asterisk back end cluster and Asterisk RT (legacy system) I've followed the recipe at http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb (with one minor exception …) if (from_uri!=myself && uri!=myself) became if (uri!=myself) as with the original line in place we were able to spoof traffic from 3rd party sites and route out onto the PSTN (which I thought was bad) … anyway … The problem I'm seeing currently is that when a call is passed down a SIP trunk to an end user on the (K) platform we're losing the DNID Asterisk delivers the call to SIP/account/DNID (K) however just tries to deliver the call to DNID@domain which comes back as not found. I've had limited success hand coding aliases into the alias_db however we're still missing the DNID info and having to scrape it from the SIP To field (with a) limited success and b) fears of a support nightmare if we try and move existing customers onto the new platform). If we were to do this for real I'd either have to modify usrloc or try some perl_exec magic I'm guessing … (although it might be possible with a view on the existing sql database … but I'd rather not have to do that if there is a simpler way) I confess I'm struggling to get my head around the config and docs and hoped someone could point me in the right direction? For legacy reasons Asterisk needs to be in the critical path on this particular build … what I'm looking for is a simple recipe and some helpful pointers on how to implement it that will allow enable me to swing (K) into the path between our end user SIP devices and the existing asterisk back ends without losing the ability to deliver hundreds of numbers down a single SIP trunk to a subscriber, and that doesn't require them to make any changes on their end as they will still see the equivalent of SIP:${DNID}@example.com arriving on their PBX This should be simple, but I'm obviously missing something :) Help and pointers gratefully received -- Jon Morby FidoNet - the internet made simple! 10 - 16 Tiller Road, London, E14 8PX tel: 0845 004 3050 / fax: 0845 004 3051

3 4

Re: [SR-Users] SBC
by Keith 28 Feb '13

28 Feb '13

Hi All, So following a guide I have successfully setup Kamailio and FreeSwitch together working as an SBC. SIP registration with Kamailio and using FreeSwitch as the media server which in turn authenticates with users in Kamailio. I have managed to pass calls between users etc. So the next challenge is bolting 2 Asterisks servers behind this SBC and load balancing the media. As the SBC will be public facing I also have to consider NAT which I believe will be configured in FreeSwitch. I have had some fantastic advice in the past so just interested to get peoples views again. Thanks. -- Keith

1 0

Re: [SR-Users] [sr-dev] openser db on an external mysql server
by Prakash N 28 Feb '13

28 Feb '13

Hi Govoiper Free Memory on my server 2309 MB total used free shared buffers cached Mem: 2795 486 2309 0 70 180 -/+ buffers/cache: 234 2561 Swap: 2927 0 2927 *Kamailio error:* * * ERROR: PID file /var/run/kamailio.pid does not exist -- Kamailio start failed *tail -f /var/log/messages:* Feb 12 23:41:28 localhost /usr/local/sbin/kamailio[4280]: INFO: <core> [udp_server.c:230]: INFO: udp_init: SO_RCVBUF is finally 262142 Feb 12 23:41:30 localhost /usr/local/sbin/kamailio[4281]: ERROR: db_mysql [km_my_con.c:109]: driver error: Can't connect to MySQL server on '192.168.1.170' (4) Feb 12 23:41:30 localhost /usr/local/sbin/kamailio[4281]: ERROR: <core> [db.c:320]: could not add connection to the pool Feb 12 23:41:30 localhost /usr/local/sbin/kamailio[4281]: ERROR: usrloc [ul_mod.c:407]: child(1): failed to connect to database Feb 12 23:41:30 localhost /usr/local/sbin/kamailio[4281]: ERROR: <core> [sr_module.c:896]: init_mod_child(): Error while initializing module usrloc (/usr/local/lib64/kamailio/modules_k/usrloc.so) Feb 12 23:41:30 localhost /usr/local/sbin/kamailio[4283]: ERROR: db_mysql [km_my_con.c:109]: driver error: Can't connect to MySQL server on '192.168.1.170' (4) Feb 12 23:41:30 localhost /usr/local/sbin/kamailio[4283]: ERROR: <core> [db.c:320]: could not add connection to the pool Feb 12 23:41:30 localhost /usr/local/sbin/kamailio[4283]: ERROR: auth_db [authdb_mod.c:174]: unable to connect to the database Feb 12 23:41:30 localhost /usr/local/sbin/kamailio[4283]: ERROR: <core> [sr_module.c:896]: init_mod_child(): Error while initializing module auth_db (/usr/local/lib64/kamailio/modules_k/auth_db.so) Feb 12 23:41:30 localhost /usr/local/sbin/kamailio[4283]: ERROR: <core> [pt.c:350]: ERROR: fork_process(): init_child failed for process 3, pid 4283, "udp receiver child=2 sock=127.0.0.1:5060" Feb 12 23:41:30 localhost /usr/local/sbin/kamailio[4283]: : <core> [main.c:1625]: main_loop: Cannot fork With Regards N.Prakash ---------- Forwarded message ---------- > From: SamyGo <govoiper(a)gmail.com> > Date: Tue, Feb 12, 2013 at 5:49 PM > Subject: Re: [SR-Users] [sr-dev] openser db on an external mysql server > To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users > Mailing List" <sr-users(a)lists.sip-router.org> > Cc: Muhammad Shahzad <shaheryarkh(a)gmail.com>, sr-dev(a)lists.sip-router.org, > Prabhakaran B <prabhakaran(a)tevatel.com>, Indu Priya <indu(a)tevatel.com> > > > Hi Prakash, > > Please paste All ERROR appearing and Warnings when you start your > kamailio. What is the free memory situation on your server ? > > Regards, > Sammy > > > > > On Tue, Feb 12, 2013 at 5:12 PM, Prakash N <prakash.n(a)tevatel.com> wrote: > >> Hi Muhammad, >> >> Yes, I have define Mysql on top in Kamailio.cfg >> >> DBURL "mysql://openser:openserrw@192.168.1.170/openser" >> >> >> I have attached Kamailio.cfg ,please advice >> >> With Regards >> >> N.Prakash >> >> >> On Tue, Feb 12, 2013 at 5:37 PM, Muhammad Shahzad <shaheryarkh(a)gmail.com>wrote: >> >>> What is your connection string in kamailio i.e. DBURL? Also make sure >>> you have defined WITH_MYSQL on the top of kamailio.cfg to load necessary >>> drivers. >>> >>> #!define WITH_MYSQL >>> >>> Thank you. >>> >>> >>> On Tue, Feb 12, 2013 at 12:56 PM, Prakash N <prakash.n(a)tevatel.com>wrote: >>> >>>> Hi Shahzad , >>>> >>>> It is not ,We can able to connect remote mysql ( please find >>>> the detail below),moreover we can able create openser db from Kamailio >>>> server to remote mysql server >>>> >>>> *Mysql server : 192.168.1.170* >>>> *Kamailio server : 192.168.1.101* >>>> >>>> >>>> *Mysql output :* >>>> * >>>> * >>>> *mysql -u openser -h 192.168.1.170 -p* >>>> Enter password: >>>> Welcome to the MySQL monitor. Commands end with ; or \g. >>>> Your MySQL connection id is 68 >>>> Server version: 5.1.67 Source distribution >>>> >>>> Copyright (c) 2000, 2012, Oracle and/or its affiliates. All rights >>>> reserved. >>>> >>>> Oracle is a registered trademark of Oracle Corporation and/or its >>>> affiliates. Other names may be trademarks of their respective >>>> owners. >>>> >>>> Type 'help;' or '\h' for help. Type '\c' to clear the current input >>>> statement. >>>> >>>> mysql> >>>> >>>> With Regards >>>> >>>> N.Prakash >>>> >>>> >>>>> >>>>> ---------- Forwarded message ---------- >>>>> From: Muhammad Shahzad <shaheryarkh(a)gmail.com> >>>>> Date: Tue, Feb 12, 2013 at 5:16 PM >>>>> Subject: Re: [sr-dev] [SR-Users] openser db on an external mysql server >>>>> To: Development mailing list of the sip-router project < >>>>> sr-dev(a)lists.sip-router.org> >>>>> Cc: Daniel-Constantin Mierla <miconda(a)gmail.com>, >>>>> sr-users(a)lists.sip-router.org, Indu Priya <indu(a)tevatel.com>, >>>>> Prabhakaran B <prabhakaran(a)tevatel.com> >>>>> >>>>> >>>>> Its not a Kamailio error, it is problem with connecting to mysql >>>>> server. Error 4 means No MySQL Server Running On Specified Host. >>>>> >>>>> May be MySQL server is down or there is some firewall running or MySQL >>>>> not listening on IP you specified. >>>>> >>>>> Thank you. >>>>> >>>>> >>>>> On Tue, Feb 12, 2013 at 12:11 PM, Prakash N <prakash.n(a)tevatel.com>wrote: >>>>> >>>>>> Dear Daniel, >>>>>> >>>>>> Thanks for your quick response >>>>>> >>>>>> We can able to create operseb db on remote mysql server ,but >>>>>> Kamailio not getting start >>>>>> >>>>>> Error detail: >>>>>> >>>>>> ERROR: db_mysql [km_my_con.c:109]: driver error: Can't connect to >>>>>> MySQL server on '192.168.1.170' (4) >>>>>> ERROR: <core> [db.c:32 0]: could not add connection to the pool >>>>>> ERROR: auth_db [authdb _mod.c:174]: unable to connect to the database >>>>>> ERROR: <core> [sr_modu le.c:896]: init_mod_child(): Error while >>>>>> initializing module auth_db (/usr/local/lib64/kamailio/modules_k/auth_db.so) >>>>>> >>>>>> >>>>>> I have attached my kamailo.cfg ( we followed Kamailio and >>>>>> Asterisk real time installation) ,please advice >>>>>> >>>>>> With Regards >>>>>> >>>>>> N.Prakash >>>>>> >>>>>>> >>>>>>> ---------- Forwarded message ---------- >>>>>>> From: Daniel-Constantin Mierla <miconda(a)gmail.com> >>>>>>> Date: Tue, Feb 12, 2013 at 2:41 PM >>>>>>> Subject: Re: [SR-Users] openser db on an external mysql server >>>>>>> To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - >>>>>>> Users Mailing List" <sr-users(a)lists.sip-router.org>, >>>>>>> sr-dev(a)lists.sip-router.org, miconda(a)gmail.com >>>>>>> Cc: Prabhakaran B <prabhakaran(a)tevatel.com> >>>>>>> >>>>>>> >>>>>>> Hello, >>>>>>> >>>>>>> kamailio database is created with: >>>>>>> >>>>>>> kamdbctl create >>>>>>> >>>>>>> the host of mysql server has to be specified in kamctlrc file via >>>>>>> DBHOST variable - kamctlrc is located in same directory as kamailio.cfg. >>>>>>> >>>>>>> In kamailio.cfg, update the db_url value (or DBURL define) to >>>>>>> reflect the mysql server host. >>>>>>> >>>>>>> Cheers, >>>>>>> Daniel >>>>>>> >>>>>>> >>>>>>> On 2/12/13 4:16 AM, Prakash N wrote: >>>>>>> >>>>>>> Hi All, >>>>>>> >>>>>>> I am new this Kamailo ,ther is any step by step document to set >>>>>>> up openser db on an external mysql server ? >>>>>>> >>>>>>> With Regards >>>>>>> >>>>>>> N.Prakash >>>>>>> >>>>>>> >>>>>>> >>>>>>> _______________________________________________ >>>>>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>>>>>> >>>>>>> >>>>>>> >>>>>>> -- >>>>>>> Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda >>>>>>> Kamailio World Conference, April 16-17, 2013, Berlin >>>>>>> - http://conference.kamailio.com - >>>>>>> >>>>>>> >>>>>>> >>>>>> >>>>>> _______________________________________________ >>>>>> sr-dev mailing list >>>>>> sr-dev(a)lists.sip-router.org >>>>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-dev >>>>>> >>>>>> >>>>> >>>>> >>>>> -- >>>>> Muhammad Shahzad >>>>> ----------------------------------- >>>>> CISCO Rich Media Communication Specialist (CRMCS) >>>>> CISCO Certified Network Associate (CCNA) >>>>> Cell: +49 176 99 83 10 85 >>>>> MSN: shari_786pk(a)hotmail.com >>>>> Email: shaheryarkh(a)googlemail.com >>>>> >>>>> >>>> >>> >>> >>> -- >>> Muhammad Shahzad >>> ----------------------------------- >>> CISCO Rich Media Communication Specialist (CRMCS) >>> CISCO Certified Network Associate (CCNA) >>> Cell: +49 176 99 83 10 85 >>> MSN: shari_786pk(a)hotmail.com >>> Email: shaheryarkh(a)googlemail.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 >> >> > >

4 18

GIT master branch is open for development of v4.1.x
by Daniel-Constantin Mierla 27 Feb '13

27 Feb '13

Hello, the GIT master branch is again open for new development. Common testing period of releasing v4.0.0 has ended, if you have pending commits or personal branches to merge, you can go ahead. v4.0.0 will be released out of newly created branch 4.0. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

2 1

How to remove custom header after adding one
by Mino Haluz 27 Feb '13

27 Feb '13

Hi, if I use append_hf() for adding my custom header to initial request, how I can remove this header later ? What I'm doing: 1) initial request: remove RPID header with remove_hf 2) initial request: add custom RPID header append_hf 3) after rerouting: in failure_route remove RPID header remove_hf 4) after rerouting: in failure_route add custom RPID header with append_hf The result of this, are 2 RPID headers, because remove_hf did not catch RPID header added in step 2, but I think the original one that was removed already in step 1. Thanks, Mino

3 3

GIT branch 4.0
by Daniel-Constantin Mierla 27 Feb '13

27 Feb '13

Hello, the GIT branch 4.0 has been created to host the release series for v4.0.x. It must take only bug fixes and documentation improvements related to the features existing at this moment in the branch. To get the branch locally, use: git clone --depth 1 git://git.sip-router.org/sip-router kamailio cd kamailio git checkout -b 4.0 origin/4.0 To push the local commits on remote repository: git push origin 4.0:4.0 Any fixes that have to get in the branch 4.0 should be committed first in master branch and then cherry-picked in this branch, in order to be able to track forgotten backports easier. Testing for 4.0 went very good so far, in few days we will probably sketch the path to the release day. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

1 0

siremis CDR gneration procedure
by Klaus Darilion 27 Feb '13

27 Feb '13

Hi! IMO the siremis CDR generation procedure is buggy: CREATE PROCEDURE `kamailio_cdrs`() DECLARE inv_cursor CURSOR FOR SELECT src_user, src_domain, dst_user, dst_domain, dst_ouser, time, callid,from_tag, to_tag, src_ip FROM acc where method='INVITE' and cdr_id='0'; ... REPEAT FETCH inv_cursor INTO v_src_user, v_src_domain, v_dst_user, v_dst_domain, v_dst_ouser, v_inv_time, v_callid, v_from_tag, v_to_tag, v_src_ip; IF NOT done THEN SET bye_record = 0; SELECT 1, time INTO bye_record, v_bye_time FROM acc WHERE method='BYE' AND callid=v_callid AND ((from_tag=v_from_tag AND to_tag=v_to_tag) OR (from_tag=v_to_tag AND to_tag=v_from_tag)) ORDER BY time ASC LIMIT 1; IF bye_record = 1 THEN INSERT INTO cdrs (src_username,src_domain,dst_username, dst_domain,dst_ousername,call_start_time, duration,sip_call_id, sip_from_tag,sip_to_tag,src_ip,created) VALUES (v_src_user, v_src_domain,v_dst_user, v_dst_domain,v_dst_ouser,v_inv_time, UNIX_TIMESTAMP(v_bye_time)-UNIX_TIMESTAMP(v_inv_time), v_callid,v_from_tag,v_to_tag,v_src_ip,NOW()); UPDATE acc SET cdr_id=last_insert_id() WHERE callid=v_callid AND from_tag=v_from_tag AND to_tag=v_to_tag; END IF; SET done = 0; END IF; UNTIL done END REPEAT; ... 1. The UPDATE query will not UPDATE BYE records which were sent from CALLEE to CALLER. This can be easily fixed with: UPDATE acc SET cdr_id=last_insert_id() WHERE callid=v_callid AND ( (from_tag=v_from_tag AND to_tag=v_to_tag) OR (to_tag=v_from_tag AND from_tag=v_to_tag) ); 2. (minor) It does not consider reINVITEs. I know the default Kamailio config does not account reINVITEs, but I think when reINVITEs are accounted too, this can mess up the CDR generation. regards Klaus

3 13

kamailio call transfer
by kiran bhosale 27 Feb '13

27 Feb '13

hello, we want to implement call transfer with opneser. but the fact is we have not implemented any database and implemented own module for contact storage. i tried using avpops module. but couldn't do please suggest

1 0

Kamailio and Geoip
by Ian French 26 Feb '13

26 Feb '13

Hi list, I've been trying to get Kamailio using the geoip module to deny traffic from countries not matching certain sources, but I'm unsure of how to achieve this I've tried the below which failed. I'm trying to drop requests outside say 2 countries "AA" or "BB" geoip_match("$si", "src"); if($gip(src=>cc)!="AA" || $gip(src=>cc)!="BB") { xlog("L_INFO","SIP message from prohibited source $gip(src=>cc) (ip: $si)\n"); send_reply("403", "Forbidden"); exit; } else { xlog("L_INFO","SIP message from unprohibited source $gip(src=>cc) (ip: $si)\n"); } Can you tell me where I am wrong or how I can achieve this Thanks in advance Ian

1 0

announcing DruCall WebRTC for Drupal
by Daniel Pocock 26 Feb '13

26 Feb '13

I've just released DruCall, the WebRTC module for Drupal. http://danielpocock.com/announcing-drucall-webrtc-for-drupal Making WebRTC available as a Drupal module hopefully means thousands of blogs and small business web sites will start picking this up and hopefully Kamailio will share in some of the demand for WebRTC capable server infrastructure.

1 0

Re-registration
by Abdullah-al-mahmud Al-amin 26 Feb '13

26 Feb '13

Hi, I'd created an account with my domain shaluavita. My user id was alamin(a)shaluavita.tk. Unfortunatly, I canceled the account. It war about 2 moth ago. I tried to re-register many times but failed for remaining the domain in the database. Now I want to use my account. Please re-activate it. Thanks

1 0

Stale dialogs in database - stale reference
by David 26 Feb '13

26 Feb '13

Hey all, I am dealing with a possible race condition in the dialog module. About one call in 50 isn't cleared from the database. I catch the calls by joining the dialog table to my cdr table ( which is generated from acc table ). The call remains in the dialog table in the database. I tested, and I am not hitting the destroy_dlg() function which is called from the unref_dlg_unsafe macro. destroy_dlg() calls the remove_dialog_from_db(). unref_dlg_unsafe is called but the ref counter is still at one. I can see this from my logs : 11:22:55 [20195]: ERROR: dialog [dlg_hash.c:944]: dialog 0x7fab27f4aa88 changed from state 1 to state 3, due event 3 (ref 2) <== de ringing a repondu 11:22:55 [20172]: ERROR: dialog [dlg_hash.c:944]: dialog 0x7fab27f4aa88 changed from state 3 to state 4, due event 6 (ref 3) <== answer 11:22:55 [20202]: ERROR: dialog [dlg_hash.c:944]: dialog 0x7fab27f4aa88 changed from state 4 to state 4, due event 6 (ref 3) <== answer 11:22:55 [20177]: ERROR: dialog [dlg_hash.c:944]: dialog 0x7fab27f4aa88 changed from state 4 to state 4, due event 6 (ref 4) <== answer 11:23:05 [20190]: ERROR: dialog [dlg_hash.c:944]: dialog 0x7fab27f4aa88 changed from state 4 to state 5, due event 7 (ref 3) <== BYE, hangup 11:23:05 [20183]: ERROR: dialog [dlg_hash.c:944]: dialog 0x7fab27f4aa88 changed from state 5 to state 5, due event 7 (ref 5) <== confirm hangup The problem seems to be happening randomly and I can't find any explanation for it. So the number of references never reaches 0, because it never reaches 0, the dialog is not unreferenced : Here is the output from syslog ( grepped with the call hash ) : Feb 26 11:21:41 kamailio-1 /usr/sbin/kamailio[20164]: ERROR: dialog [dlg_hash.c:717]: ref dlg 0x7fab27f4aa88 with 1 -> 1 Feb 26 11:21:41 kamailio-1 /usr/sbin/kamailio[20164]: ERROR: dialog [dlg_hash.c:737]: ref dlg 0x7fab27f4aa88 with 1 -> 2 Feb 26 11:22:48 kamailio-1 /usr/sbin/kamailio[20214]: ERROR: dialog [dlg_hash.c:590]: ref dlg 0x7fab27f4aa88 with 1 -> 3 Feb 26 11:22:48 kamailio-1 /usr/sbin/kamailio[20214]: ERROR: dialog [dlg_hash.c:944]: dialog 0x7fab27f4aa88 changed from state 4 to state 5, due event 7 (ref 3) Feb 26 11:22:48 kamailio-1 /usr/sbin/kamailio[20214]: ERROR: dialog [dlg_hash.c:737]: ref dlg 0x7fab27f4aa88 with 1 -> 4 Feb 26 11:22:48 kamailio-1 /usr/sbin/kamailio[20214]: ERROR: dialog [dlg_hash.c:757]: unref dlg 0x7fab27f4aa88 with 2 -> 2 Feb 26 11:22:48 kamailio-1 /usr/sbin/kamailio[20214]: ERROR: dialog [dlg_hash.c:757]: unref dlg 0x7fab27f4aa88 with 1 -> 1 Feb 26 11:22:48 kamailio-1 /usr/sbin/kamailio[20214]: ERROR: dialog [dlg_hash.c:590]: ref dlg 0x7fab27f4aa88 with 1 -> 2 Feb 26 11:22:48 kamailio-1 /usr/sbin/kamailio[20214]: ERROR: dialog [dlg_hash.c:757]: unref dlg 0x7fab27f4aa88 with 1 -> 1 Feb 26 11:22:48 kamailio-1 /usr/sbin/kamailio[20170]: ERROR: dialog [dlg_hash.c:590]: ref dlg 0x7fab27f4aa88 with 1 -> 2 Feb 26 11:22:48 kamailio-1 /usr/sbin/kamailio[20170]: ERROR: dialog [dlg_hash.c:757]: unref dlg 0x7fab27f4aa88 with 1 -> 1 Feb 26 11:22:53 kamailio-1 /usr/sbin/kamailio[20216]: ERROR: dialog [dlg_hash.c:590]: ref dlg 0x7fab27f4aa88 with 1 -> 2 Feb 26 11:22:53 kamailio-1 /usr/sbin/kamailio[20216]: ERROR: dialog [dlg_hash.c:757]: unref dlg 0x7fab27f4aa88 with 2 -> 0 Feb 26 11:22:53 kamailio-1 /usr/sbin/kamailio[20216]: ERROR: dialog [dlg_hash.c:757]: ref <=0 for dialog 0x7fab27f4aa88 Feb 26 11:22:53 kamailio-1 /usr/sbin/kamailio[20216]: ERROR: dialog [dlg_hash.c:317]: destroying dialog 0x7fab27f4aa88 (ref 0) Feb 26 11:22:53 kamailio-1 /usr/sbin/kamailio[20216]: ERROR: dialog [dlg_hash.c:333]: removed timer for dlg 0x7fab27f4aa88 [7298:10958] with clid '13b951b67f2704d51d25cf643b3469d1(a)pbx.omnity.biz' and tags 'as3a8e0df2' 'as4c79d466' Feb 26 11:22:55 kamailio-1 /usr/sbin/kamailio[20208]: ERROR: dialog [dlg_hash.c:717]: ref dlg 0x7fab27f4aa88 with 1 -> 1 Feb 26 11:22:55 kamailio-1 /usr/sbin/kamailio[20208]: ERROR: dialog [dlg_cb.c:230]: dialog=0x7fab27f4aa88 Feb 26 11:22:55 kamailio-1 /usr/sbin/kamailio[20208]: ERROR: dialog [dlg_hash.c:737]: ref dlg 0x7fab27f4aa88 with 1 -> 2 Feb 26 11:22:55 kamailio-1 /usr/sbin/kamailio[20208]: ERROR: dialog [dlg_hash.c:757]: unref dlg 0x7fab27f4aa88 with 1 -> 1 Feb 26 11:22:55 kamailio-1 /usr/sbin/kamailio[20208]: ERROR: dialog [dlg_hash.c:590]: ref dlg 0x7fab27f4aa88 with 1 -> 2 Feb 26 11:22:55 kamailio-1 /usr/sbin/kamailio[20208]: ERROR: dialog [dlg_hash.c:757]: unref dlg 0x7fab27f4aa88 with 1 -> 1 Feb 26 11:22:55 kamailio-1 /usr/sbin/kamailio[20195]: ERROR: dialog [dlg_hash.c:590]: ref dlg 0x7fab27f4aa88 with 1 -> 2 Feb 26 11:22:55 kamailio-1 /usr/sbin/kamailio[20195]: ERROR: dialog [dlg_hash.c:757]: unref dlg 0x7fab27f4aa88 with 1 -> 1 Feb 26 11:22:55 kamailio-1 /usr/sbin/kamailio[20195]: ERROR: dialog [dlg_hash.c:590]: ref dlg 0x7fab27f4aa88 with 1 -> 2 Feb 26 11:22:55 kamailio-1 /usr/sbin/kamailio[20195]: ERROR: dialog [dlg_hash.c:944]: dialog 0x7fab27f4aa88 changed from state 1 to state 3, due event 3 (ref 2) Feb 26 11:22:55 kamailio-1 /usr/sbin/kamailio[20195]: ERROR: dialog [dlg_handlers.c:484]: dialog 0x7fab27f4aa88 confirmed (ACK pending) Feb 26 11:22:55 kamailio-1 /usr/sbin/kamailio[20195]: ERROR: dialog [dlg_hash.c:737]: ref dlg 0x7fab27f4aa88 with 1 -> 3 Feb 26 11:22:55 kamailio-1 /usr/sbin/kamailio[20195]: ERROR: dialog [dlg_cb.c:256]: dialog=0x7fab27f4aa88, type=8 Feb 26 11:22:55 kamailio-1 /usr/sbin/kamailio[20174]: ERROR: dialog [dlg_hash.c:590]: ref dlg 0x7fab27f4aa88 with 1 -> 4 Feb 26 11:22:55 kamailio-1 /usr/sbin/kamailio[20174]: ERROR: dialog [dlg_hash.c:757]: unref dlg 0x7fab27f4aa88 with 1 -> 3 Feb 26 11:22:55 kamailio-1 /usr/sbin/kamailio[20195]: ERROR: dialog [dlg_hash.c:757]: unref dlg 0x7fab27f4aa88 with 1 -> 2 Feb 26 11:22:55 kamailio-1 /usr/sbin/kamailio[20178]: ERROR: dialog [dlg_hash.c:590]: ref dlg 0x7fab27f4aa88 with 1 -> 3 Feb 26 11:22:55 kamailio-1 /usr/sbin/kamailio[20178]: ERROR: dialog [dlg_hash.c:757]: unref dlg 0x7fab27f4aa88 with 1 -> 2 Feb 26 11:22:55 kamailio-1 /usr/sbin/kamailio[20172]: ERROR: dialog [dlg_hash.c:590]: ref dlg 0x7fab27f4aa88 with 1 -> 3 Feb 26 11:22:55 kamailio-1 /usr/sbin/kamailio[20172]: ERROR: dialog [dlg_hash.c:944]: dialog 0x7fab27f4aa88 changed from state 3 to state 4, due event 6 (ref 3) Feb 26 11:22:55 kamailio-1 /usr/sbin/kamailio[20172]: ERROR: dialog [dlg_hash.c:757]: unref dlg 0x7fab27f4aa88 with 1 -> 2 Feb 26 11:22:55 kamailio-1 /usr/sbin/kamailio[20172]: ERROR: dialog [dlg_hash.c:590]: ref dlg 0x7fab27f4aa88 with 1 -> 3 Feb 26 11:22:55 kamailio-1 /usr/sbin/kamailio[20172]: ERROR: dialog [dlg_hash.c:757]: unref dlg 0x7fab27f4aa88 with 1 -> 2 Feb 26 11:22:55 kamailio-1 /usr/sbin/kamailio[20202]: ERROR: dialog [dlg_hash.c:590]: ref dlg 0x7fab27f4aa88 with 1 -> 3 Feb 26 11:22:55 kamailio-1 /usr/sbin/kamailio[20202]: ERROR: dialog [dlg_hash.c:944]: dialog 0x7fab27f4aa88 changed from state 4 to state 4, due event 6 (ref 3) Feb 26 11:22:55 kamailio-1 /usr/sbin/kamailio[20202]: ERROR: dialog [dlg_cb.c:256]: dialog=0x7fab27f4aa88, type=32 Feb 26 11:22:55 kamailio-1 /usr/sbin/kamailio[20177]: ERROR: dialog [dlg_hash.c:590]: ref dlg 0x7fab27f4aa88 with 1 -> 4 Feb 26 11:22:55 kamailio-1 /usr/sbin/kamailio[20177]: ERROR: dialog [dlg_hash.c:944]: dialog 0x7fab27f4aa88 changed from state 4 to state 4, due event 6 (ref 4) Feb 26 11:22:55 kamailio-1 /usr/sbin/kamailio[20177]: ERROR: dialog [dlg_cb.c:256]: dialog=0x7fab27f4aa88, type=32 Feb 26 11:22:55 kamailio-1 /usr/sbin/kamailio[20202]: ERROR: dialog [dlg_hash.c:757]: unref dlg 0x7fab27f4aa88 with 1 -> 3 Feb 26 11:22:55 kamailio-1 /usr/sbin/kamailio[20202]: ERROR: dialog [dlg_hash.c:590]: ref dlg 0x7fab27f4aa88 with 1 -> 4 Feb 26 11:22:55 kamailio-1 /usr/sbin/kamailio[20202]: ERROR: dialog [dlg_hash.c:757]: unref dlg 0x7fab27f4aa88 with 1 -> 3 Feb 26 11:22:55 kamailio-1 /usr/sbin/kamailio[20177]: ERROR: dialog [dlg_hash.c:757]: unref dlg 0x7fab27f4aa88 with 1 -> 2 Feb 26 11:22:55 kamailio-1 /usr/sbin/kamailio[20177]: ERROR: dialog [dlg_hash.c:590]: ref dlg 0x7fab27f4aa88 with 1 -> 3 Feb 26 11:22:55 kamailio-1 /usr/sbin/kamailio[20177]: ERROR: dialog [dlg_hash.c:757]: unref dlg 0x7fab27f4aa88 with 1 -> 2 Feb 26 11:23:05 kamailio-1 /usr/sbin/kamailio[20190]: ERROR: dialog [dlg_hash.c:590]: ref dlg 0x7fab27f4aa88 with 1 -> 3 Feb 26 11:23:05 kamailio-1 /usr/sbin/kamailio[20190]: ERROR: dialog [dlg_hash.c:944]: dialog 0x7fab27f4aa88 changed from state 4 to state 5, due event 7 (ref 3) Feb 26 11:23:05 kamailio-1 /usr/sbin/kamailio[20190]: ERROR: dialog [dlg_hash.c:737]: ref dlg 0x7fab27f4aa88 with 1 -> 4 Feb 26 11:23:05 kamailio-1 /usr/sbin/kamailio[20190]: ERROR: dialog [dlg_cb.c:256]: dialog=0x7fab27f4aa88, type=64 Feb 26 11:23:05 kamailio-1 /usr/sbin/kamailio[20183]: ERROR: dialog [dlg_hash.c:590]: ref dlg 0x7fab27f4aa88 with 1 -> 5 Feb 26 11:23:05 kamailio-1 /usr/sbin/kamailio[20183]: ERROR: dialog [dlg_hash.c:944]: dialog 0x7fab27f4aa88 changed from state 5 to state 5, due event 7 (ref 5) Feb 26 11:23:05 kamailio-1 /usr/sbin/kamailio[20183]: ERROR: dialog [dlg_hash.c:757]: unref dlg 0x7fab27f4aa88 with 1 -> 4 Feb 26 11:23:05 kamailio-1 /usr/sbin/kamailio[20183]: ERROR: dialog [dlg_hash.c:590]: ref dlg 0x7fab27f4aa88 with 1 -> 5 Feb 26 11:23:05 kamailio-1 /usr/sbin/kamailio[20183]: ERROR: dialog [dlg_hash.c:757]: unref dlg 0x7fab27f4aa88 with 1 -> 4 Feb 26 11:23:05 kamailio-1 /usr/sbin/kamailio[20190]: ERROR: dialog [dlg_hash.c:757]: unref dlg 0x7fab27f4aa88 with 2 -> 2 Feb 26 11:23:05 kamailio-1 /usr/sbin/kamailio[20190]: ERROR: dialog [dlg_hash.c:757]: unref dlg 0x7fab27f4aa88 with 1 -> 1 Feb 26 11:23:05 kamailio-1 /usr/sbin/kamailio[20190]: ERROR: dialog [dlg_hash.c:590]: ref dlg 0x7fab27f4aa88 with 1 -> 2 Feb 26 11:23:05 kamailio-1 /usr/sbin/kamailio[20190]: ERROR: dialog [dlg_hash.c:757]: unref dlg 0x7fab27f4aa88 with 1 -> 1 You can see the last refence is never removed, so the dialog isn't destroyed. Here is the end of a healthy dialog : Feb 26 08:03:00 kamailio-1 /usr/sbin/kamailio[4668]: ERROR: dialog [dlg_hash.c:753]: unref dlg 0x7f74db910e60 with 2 -> 0 Feb 26 08:03:00 kamailio-1 /usr/sbin/kamailio[4668]: ERROR: dialog [dlg_hash.c:753]: ref <=0 for dialog 0x7f74db910e60 Feb 26 08:03:00 kamailio-1 /usr/sbin/kamailio[4668]: ERROR: dialog [dlg_hash.c:315]: destroying dialog 0x7f74db910e60 (ref 0) You see it falls to 0 and the dialog is destroyed. So, how do I debug this ? Should I maybe be posting to sr-dev ? How do I determine who is incrementing the counter without decrementing it? What information do I need to continue with this problem? Thanks! David ps I used perl pie to replace LM_DBG with LM_ERROR in modules_k/dialog/* version: kamailio 3.3.2 (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, F_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, DEFAULT PKG_SIZE 4MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled on 09:17:03 Oct 18 2012 with gcc 4.4.3

1 1

Bug in AVPOps?
by Carsten Bock 26 Feb '13

26 Feb '13

Hi, i use the AVPops module to store information about Call-Forwardings in the database: # Store new CF-Target in Database: $avp(s:usr_cfnr) = $rU; avp_db_store("$avp(s:caller_userid)", "$avp(s:usr_cfnr)/usr_preferences"); $avp(s:usr_cfnr_delay) = 20; avp_db_store("$avp(s:caller_userid)", "$avp(s:usr_cfnr_delay)/usr_preferences"); (please not, that $avp(s:caller_userid) was not changed) however this results into two different inserts, one for user 7736 (correct) T 2013/02/25 14:25:11.704628 192.168.0.2:51460 -> 192.168.0.20:3306 [AP] {....insert into usr_preferences (uuid,attribute,value,type,username,domain ) values ('7736','usr_cfnr','491792021244',0,'','') and one for user 7720 (20???, as in the delay, incorrect) T 2013/02/25 14:25:11.705572 192.168.0.2:51460 -> 192.168.0.20:3306 [AP] w....insert into usr_preferences (uuid,attribute,value,type,username,domain ) values ('7720','usr_cfnr_delay','20',1,'','') I am using Kamailio 3.3.2 from GIT-Trunk (nightly, checked out on December 14th): root@proxy1:/# kamailio -V version: kamailio 3.3.2 (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, F_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, DEFAULT PKG_SIZE 4MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled on 01:57:11 Dec 14 2012 with gcc 4.4.5 Before i dig deeper into the code of the avpops module, has anyone seen / fixed this issue? Thanks, Carsten -- Carsten Bock CEO (Geschäftsführer) ============================================================= Visit us on the Kamailio World 2013, 16./17. April 2013 => http://conference.kamailio.com/k01/ See you there! ============================================================= ng-voice GmbH Schomburgstr. 80 D-22767 Hamburg / Germany http://www.ng-voice.com mailto:carsten@ng-voice.com Office +49 40 34927219 Fax +49 40 34927220 Sitz der Gesellschaft: Hamburg Registergericht: Amtsgericht Hamburg, HRB 120189 Geschäftsführer: Carsten Bock Ust-ID: DE279344284 Hier finden Sie unsere handelsrechtlichen Pflichtangaben: http://www.ng-voice.com/imprint/

2 2

Errors leading to kamailio proc crash
by Grant Bagdasarian 26 Feb '13

26 Feb '13

Hello, Last Thursday our kamailio process crashed. There are a lot of memory related errors in there. See attachment for details. Did the machine run out of memory? How do I prevent this from happening again? The machine has 4GB of memory. Regards, Grant

2 4

Loop detection in Kamailio
by Olle E. Johansson 26 Feb '13

26 Feb '13

Hi! I can see that there's been some discussion about SIP loop detection on the OpenSER and SER mailing lists, but I must have missed the conclusion. Apart from setting a max-forwards, how would you configure your Kamailio server to prevent looping? Regards, /O

3 2

Kamailio multiple asterisk boxes
by imcdona＠gmail.com 26 Feb '13

26 Feb '13

Hello, I've been having some issues with the syntax of Kamailio's config file. I've been following the turorial here: http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb I'd like to setup Kamailio to act as an SBC providing NAT traversal, and security for multiple asterisk boxes on the inside. Currently kamailio has two NIC's, one public and one private. All traffic hits the external interface and is then forwarded to an Asterisk box on the inside. I'd like to modify the config from the tutorial above to support multiple asterisk boxes. Authentication will be done via MySQL and depending on the user domain, the request get's forwarded to the proper Asterisk box. Here is what I've got so far: if (is_method("REGISTER") && from_uri=~.*(a)10.55.55.5) { if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging ## setbflag(FLB_NATSIPPING); } if (!save("location")) sl_reply_error(); route(REGFWD); exit; } else if (is_method("REGISTER") && from_uri=~".*asterisk.domain.tld") { if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging ## setbflag(FLB_NATSIPPING); } if (!save("location")) sl_reply_error(); route(REGFWDCLOUD); exit; } } Based on the domain I route the SIP message to another route block which is esentally the same route block as the origional route[FROMASTERISK] route[TOASTERISK] route[REGFWD]. I beleive there's an easier way of doing this though. Creating separate route blocks per asterisk box would mean creating a new route blocks for each asterisk box. Here's an example: route[FROMASTERISK] { if($si==$sel(cfg_get.asterisk.bindip) && $sp==$sel(cfg_get.asterisk.bindport)) return 1; return -1; # Send to Asterisk route[TOASTERISK] { $du = "sip:" + $sel(cfg_get.asterisk.bindip) + ":" + $sel(cfg_get.asterisk.bindport); route(RELAY); exit; # 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: " + $hdr(Expires) + "\r\n"; uac_req_send(); } route[FROMCLOUD] { if($si==$sel(cfg_get.cloud.bindip) && $sp==$sel(cfg_get.asterisk.bindport)) return 1; return -1; } route[TOCLOUD] { $du = "sip:" + $sel(cfg_get.cloud.bindip) + ":" + $sel(cfg_get.asterisk.bindport); route(RELAY); exit; } route[REGFWDCLOUD] { if(!is_method("REGISTER")) { return; } $var(rip) = $sel(cfg_get.cloud.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(); } I'm sure their is any easier way of doing this alhthough I'm not sure how. Can anyone point me in the right direction? Lastly, is there a way to modify the DBURL string on the fly so based on the domain for example, I coudl query a different DB for user credentials? Thanks!

2 1

Kamailio v3.3.4 Released
by Daniel-Constantin Mierla 26 Feb '13

26 Feb '13

Hello, Kamailio SIP Server v3.3.4 stable release is out. This is a maintenance release of the latest stable branch, 3.3, that includes fixes since release of v3.3.3. There is no change to database schema or configuration language structure that you have to do on installations of v3.3.0, v3.3.1, v3.3.2 or v3.3.3. Deployments running previous v3.x.x versions are strongly recommended to be upgraded to v3.3.4. For more details about version 3.3.4 (including links and hints to download the tarball or from GIT repository), visit: * http://www.kamailio.org/w/2013/02/kamailio-v3-3-4-released/ RPM, Debian/Ubuntu packages will be available soon as well. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

2 1

Re: [SR-Users] SBC
by Keith 26 Feb '13

26 Feb '13

Hi, I am looking to build an SBC/SIP router made up from Kamailio and FreePBX. Calls will then need to be passed to multiple Asterisk media gateways. Can anyone point me to some documentation or help on how to do this? -- Keith

6 7

planning to release Kamailio v3.3.4
by Daniel-Constantin Mierla 25 Feb '13

25 Feb '13

Hello, I am considering packaging a new minor release for branch 3.3 next Monday, on February 25. Please backport any patches that you have to be part of v3.3.4 Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

2 4

Changing outgoing/source port
by Efelin Novak 25 Feb '13

25 Feb '13

Hi, I'd like to ask whether I am able to change an outgoing port. I'm using t_relay for outgoing INVITEs & Kamailio uses 5060 port defined in listen. When the call fails I am rerouting it using append_branch and I'd like to change an outgoing port to 5061. The reason I'd like to do this is a Loop detection of my buggy VoIP/analog gateway. Am I able to specify an outgoing interface/port for t_relay in Kamailio or should I find an another gateway? :) Thanks for help Kind regards Efelin

2 1

kamailio
by Suhaib Mehyar 25 Feb '13

25 Feb '13

Dears, My name is suhaib mehyar, I tried to install kamailio server, and I successfully set it up, it is locally running fine, both audio and video are very good locally but when I call a client that is connected to the server that is exist after my cisco firewall with private ip and the client is exist after his home router both audio and video are not working. Kindly any advice please??? Best Regards, Suhaib Mehyar Skype : suhaib.mehyar Mob :+962 795 161311

4 3

Kamailio World Updates
by Daniel-Constantin Mierla 25 Feb '13

25 Feb '13

Hello, the next group of accepted talks at Kamailio World Conference will be published in few days, keep an eye on: - http://conference.kamailio.com/k01/schedule/ Till then, here is what happened lately around the project. Olle E. Johansson spent the week at SIPit, making sure the upcoming version 4.0.0 is ready for release: - http://www.kamailio.org/w/2013/02/sipit30/ I conducted the Kamailio Development Workshop, with a great group of participants, first results being visible as contribution to app_lua for review on tracker. Today is schedule to release Kamailio v3.3.4, a new minor version out of latest stable branch 3.3. This week on Wednesday, development branch will be unfrozen, lots of new code is already waiting to be merged there (to be part of future version 4.1.0). If you are in Berlin this week, participating at Call Center World (http://www.callcenterworld.de/en/), ping me to meet, I will be glad to catch up with what is happening on real time communications market. Do not forget that there are open positions to work with VoIP/SIP and Kamailio, just in case you look for new challenges in your career, check: - http://conference.kamailio.com/k01/career-jobs/ It is less than two months till Kamailio World Conference, the schedule is almost filled up, if you haven't registered yet, it is time to hurry up, don't miss the chance to meet many professionals working with large real time communications systems or solutions, from Kamailio or different other projects: - http://conference.kamailio.com/k01/registration/ At the end, many thanks to those sponsoring the event, helping to cover the logistics to ensure a great event. If you consider sponsoring, please get in touch with us, you will get an exceptional visibility and additional benefits at the event: - http://conference.kamailio.com/k01/sponsors/ Looking forward to meet many of you in Berlin, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

1 0

Kamailio 1.5 and Asterisk 1.8 issue (Protocol Error, Unspecified)
by Rowie 24 Feb '13

24 Feb '13

Hi, We are having an issue where a phone (snom in particular) cannot make a call through Asterisk. It just hangup and does not allow the call to go through. I am including a a sip trace on this thread to show what is happening within the call. Please see below: Received from udp:10.1.10.80:5060 at 26/10/2011 10:22:41:476 (490 bytes): SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP 10.30.0.64:5060;received=10.30.0.64;branch=z9hG4bK-wowp1kmdy4rl;rport=5060 From: "Virgil Menendez" <sip:91421@ser.gowireless.net>;tag=6wkdms1r20 To: <sip:9513261429@ser.gowireless.net;user=phone>;tag=as0b87218f Call-ID: 3c26755bf15c-9iq08xqqblo6 CSeq: 4 INVITE Server: Asterisk PBX 1.8.7.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 -------------------------------------------------------------------------------- Sent to udp:10.1.10.80:5060 at 26/10/2011 10:22:41:481 (387 bytes): ACK sip:vm9513261429@10.1.10.83:5060 SIP/2.0 v: SIP/2.0/UDP 10.30.0.64:5060;branch=z9hG4bK-wowp1kmdy4rl;rport Route: <sip:10.1.10.80;lr=on> f: "Virgil Menendez" <sip:91421@ser.gowireless.net>;tag=6wkdms1r20 t: <sip:9513261429@ser.gowireless.net;user=phone>;tag=as0b87218f i: 3c26755bf15c-9iq08xqqblo6 CSeq: 4 ACK Max-Forwards: 70 m: <sip:91421@10.30.0.64:5060>;reg-id=1 l: 0 -------------------------------------------------------------------------------- Received from udp:10.1.10.80:5060 at 26/10/2011 10:22:42:130 (868 bytes): SIP/2.0 200 OK Via: SIP/2.0/UDP 10.30.0.64:5060;received=10.30.0.64;branch=z9hG4bK-5evtiw6dm0po;rport=5060 Record-Route: <sip:10.1.10.80;lr=on> From: "Virgil Menendez" <sip:91421@ser.gowireless.net>;tag=qi3i8ze6z8 To: <sip:9513261429@ser.gowireless.net;user=phone>;tag=as3f8c0f96 Call-ID: 3c2676547a8d-2t5yi6jok1sv CSeq: 2 INVITE Server: Asterisk PBX 1.8.7.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: <sip:9513261429@10.1.10.83:5060> Content-Type: application/sdp Content-Length: 256 v=0 o=root 1355451627 1355451627 IN IP4 10.1.10.83 s=Asterisk PBX 1.8.7.1 c=IN IP4 10.1.10.83 t=0 0 m=audio 16094 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv -------------------------------------------------------------------------------- Sent to udp:10.1.10.80:5060 at 26/10/2011 10:22:42:132 (385 bytes): ACK sip:9513261429@10.1.10.83:5060 SIP/2.0 v: SIP/2.0/UDP 10.30.0.64:5060;branch=z9hG4bK-wszafb7cbzpw;rport Route: <sip:10.1.10.80;lr=on> f: "Virgil Menendez" <sip:91421@ser.gowireless.net>;tag=qi3i8ze6z8 t: <sip:9513261429@ser.gowireless.net;user=phone>;tag=as3f8c0f96 i: 3c2676547a8d-2t5yi6jok1sv CSeq: 2 ACK Max-Forwards: 70 m: <sip:91421@10.30.0.64:5060>;reg-id=1 l: 0 -------------------------------------------------------------------------------- Received from udp:10.1.10.80:5060 at 26/10/2011 10:22:42:232 (503 bytes): BYE sip:91421@10.30.0.64:5060 SIP/2.0 Via: SIP/2.0/UDP 10.1.10.80;branch=z9hG4bKe723.bf70c1f4.0 Via: SIP/2.0/UDP 10.1.10.83:5060;branch=z9hG4bK69f53cf1 Max-Forwards: 69 From: <sip:9513261429@ser.gowireless.net;user=phone>;tag=as3f8c0f96 To: "Virgil Menendez" <sip:91421@ser.gowireless.net>;tag=qi3i8ze6z8 Call-ID: 3c2676547a8d-2t5yi6jok1sv CSeq: 102 BYE User-Agent: Asterisk PBX 1.8.7.1 X-Asterisk-HangupCause: Protocol error, unspecified X-Asterisk-HangupCauseCode: 111 Content-Length: 0 -- View this message in context: http://old.nabble.com/Kamailio-1.5-and-Asterisk-1.8-issue-%28Protocol-Error… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

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SOS!! Kamailio 3.1.x Load Balancer doesn't work
by chanea 24 Feb '13

24 Feb '13

Dear All, I am trying to use Kamailio 3.1.x server (called LB here) as the Load Balancer for two other Kamailio server (called server01 and server02 here) . Client A -----> Kamailio LB -----> Kamailio server01 Client B -----> Kamailio LB -----> Kamailio server01 If I dont use LB here , and letting Client A and B connect to Kamailio server01 directly , all the communication is fine. While using Kamailio LB ,without luck, although the sip client A and B get 200 from server01 for Register, they cannot reach each other for "Message" and "Invite". It will be great appreciated if anyone can give a hint. my config for Kamailio LB is as below : ========================================== .... debug=2 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=yes # (cmd line: -E) children=2 check_via=no # (cmd. line: -v) dns=off # (cmd. line: -r) rev_dns=off # (cmd. line: -R) port=5060 # for more info: sip_router -h # ------------------ module loading ---------------------------------- mpath="/usr/local/lib64/kamailio/modules_k/:/usr/local/lib64/kamailio/modules/" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "mi_fifo.so" loadmodule "dispatcher.so" loadmodule "db_mysql.so" # ----------------- setting module-specific parameters --------------- # -- dispatcher params -- modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") modparam("dispatcher", "db_url","mysql://.........@........") modparam("usrloc", "db_mode", 0) modparam("rr", "enable_full_lr", 1) route{ if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; # if (!method=="REGISTER") # record_route(); ds_select_dst("1","4"); forward(); exit; } .. ========================================== best regards, chanea

2 3

SIPit 30 summary
by Olle E. Johansson 22 Feb '13

22 Feb '13

Hi! I've spent five days at SIPit 30 testing Kamailio on a tiny Asus EEE-netbook. It's been great, and apart from the bug hitting me while using outbound, we've done great in all tests. During the proxy doom test today, the netbook processed way over 50.000 messages in a single INVITE transaction that spiralled and looped and behaved in funny ways, and the CPU still had a lot of power to do other things. The hub we used was the limiting factor. Kamailio 4.0.0 beta was involved in all multiparty tests, from TLS and RTCweb to forking, spiralling and proxy doom. We proved that we have a solid product that can handle all kind of scenarious, even TCP overflow attacks. Thanks for all the help over jabber, IRC and other means! Special thank you to Fred Posner who contributed a Big Fred Cookie! Have a great weekend! /O

3 2

Path and DNS SRV
by Charles Chance 22 Feb '13

22 Feb '13

Hi All, Given a situation where there are multiple registrars, behind a single load balancer, sharing the same location table in DB, I understand that Path can be used to ensure everything works nicely with clients behind NAT. So, if UA1 is registered on P1, UA2 on P2, and UA1 calls UA2... P1 performs lookup, sees path and sends via P2. What about situations where the distribution between registrars is being handled client-side through DNS SRV, not a load balancer? Path cannot be used then, right? As I understand it, in this instance add_path() will do nothing, is this correct? Thanks in advance, Charles -- 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.

3 4

Weird out-of-memory message on start/stop
by Andreas Granig 21 Feb '13

21 Feb '13

Hi, Maybe some of you have encountered this funny error message when starting and even killing kamailio on a Debian 6 with 2.6.x and 3.x kernels and 2.11.3-4: #------------------------- # /usr/sbin/kamailio -f /etc/kamailio/proxy/kamailio.cfg -P /var/run/kamailio/kamailio.proxy.pid -m 4 -M 16 -u kamailio -g kamailio loading modules under /usr/lib/kamailio/modules_k:/usr/lib/kamailio/modules Listening on udp: 127.0.0.1:5062 tcp: 127.0.0.1:5062 Aliases: Message from syslogd@sp2 at Feb 19 11:15:42 ... t of memory [6720] # killall kamailio Message from syslogd@sp2 at Feb 19 11:17:51 ... t of memory [6720] #------------------------- The PID reported here is the kamailio attendant process. It seems to be somehow related to the shared memory. The interesting part about that is that kamailio really starts just fine, and there is plenty (>5GB) of free memory available. It doesn't seem to influence the behaviour of kamailio at all as it's working perfectly fine, so for now it's just annoying, but I'm really curious I've tried to fiddle with mem_join, but it doesn't matter whether it's on or off. Kamailio version is this: # kamailio -V version: kamailio 3.3.2 (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, F_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, DEFAULT PKG_SIZE 4MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled on 19:09:17 Jan 21 2013 with gcc 4.4.5 Has anyone seen this too? What's the cause of that? Andreas

4 7

SDP c= address as a key for a key-value lookup?
by Hegner, Travis 21 Feb '13

21 Feb '13

Hi All, I am working on a kamailio proxy with a very specific goal I need to accomplish. What I would like to do is a key-value lookup, utilizing the SDP's media contact (c= line) IP as a key, and have a function return a value for that key. It appears that I can possibly accomplish this with the AVPops module, combined with DB_TEXT (maybe DB_MYSQL in the future) and a custom table with a custom query. The only thing I seem to be missing is the ability to extract the c= IP from the SDP, and store it as some kind of variable with which to do the lookup. I've looked into the iptrtpproxy module as it seems to have a select: @iptrtpproxy.sdp_ip that could work, but that module is extreme overkill for my purposes, and I am running into some issues getting it built for my platform (Ubuntu server 12.04). Also the documentation for that select states: "Return first rewritten IP provided at SDP c= line." However, I'm looking for the original sdp_ip, and have some concern that this won't return an IP that has not yet been re-written. My environment has a very specific 1:1 ratio of private network, to static public IP mappings. I currently have a successful OpenSER (1.3.2) deployment within this environment (running 5 years), which uses a hacked version of the old "mangler" module, with the function sdp_mangle_ip(<network>, <ip>). I could do something similar using textops->search_body() and nathelper->fix_nated_sdp(), but the risk of a typo or syntax error in the config file every time I have to make a change or add a network is fairly high. Less importantly, there is some extra overhead as the proxy will have to test for every network on every SDP looking for a match (this is what the current deployment does). Eventually I'd like to add a simple frontend to add/edit networks on the fly, so that no one is ever touching the config. What would you folks recommend for this? Should I look at building a custom module which only grabs the c= IP and stores it into an AVP? Am I not seeing another module that will expose that information for me already? Would it simply be easier to work on getting iptrtpproxy built, and utilizing it only for the purpose of extracting the information I need from the SDP? Any insight would be most greatly appreciated. Thanks, Travis Hegner ________________________________ The information contained in this communication is confidential and is intended only for the use of the named recipient. Unauthorized use, disclosure, or copying is strictly prohibited and may be unlawful. If you have received this communication in error, you should know that you are bound to confidentiality, and should please immediately notify the sender or our IT Department at 866.459.4599.

5 6

Difference between shared and script variable
by Mino Haluz 21 Feb '13

21 Feb '13

Hi, what is the difference between shared and script variable ? Thanks Mino

2 3

Re building command
by Techie Sup 20 Feb '13

20 Feb '13

Hi all, Sorry for this simple question. 1) I have made some changes to main.c file and want rebuild kamailio so that changes take Éffect, what is the command to rebuild? Thanks Selvam

2 1

using dlg_manage before or after?
by Uri Shacked 20 Feb '13

20 Feb '13

Hi, I am trying to change the timeout of a dialog based on the $avp() I set on the modeparam. While testing, I notice that the change does not apply after I am using dlg_manage, only if i changed it before. Can anyone explain why is that? and is it possible to change it after doing dkg_manage? Thanks, URi

2 1

mojtaba jahanbakhsh
by mojtaba jahanbakhsh 20 Feb '13

20 Feb '13

http://www.firstchoiceforyou.com/ecj/xnyfbj4f=4dbl9cu52nkdz

1 0

Strip All but numbers/*/#
by Ali Jawad 20 Feb '13

20 Feb '13

Hi I am using the below to strip "-" from destination numbers, it does work just fine "I replace - with nothing", what i need to do now is to strip all characters except 0-9 * and #. Is there anyway to do it using the following configuration part if($rU =~ ".*-.*" ) { $rU = $(rU{re.subst,/-//g}); } Regards

4 5

Re: [SR-Users] kamctl restart and mediaproxy
by Victor Seva 19 Feb '13

19 Feb '13

We use mediaproxy and kamailio and using the init.d script from the Debian package to restart kamailio has no effect on mediaproxy process. 2013/2/8 Eduardo Lejarreta <lejarreta.e(a)sarenet.es>: > Good morning Daniel. > > I send you the "messages" log which is a bit filtered. The flow is: > > Start a call, engage_mediaproxy on ports (50156, 50157, 50158, 50159) and > then "kamctl restart". As you see it seems that an instruction from kamailio > tells mediaproxy to drop the ports. > We have tried to script around "kamctl" to avoid sending orders to > mediaproxy, killing the "media-dispatcher" before Kamailio and then > restarting again (media-dispatcher) and this works OK, but seems not be a > reasonable way to fix it. > > ...... > Feb 8 11:04:51 kamailio-dev media-relay[2474]: > mediaproxy.mediacontrol.StreamListenerProtocol starting on 50156 > Feb 8 11:04:51 kamailio-dev media-relay[2474]: > mediaproxy.mediacontrol.StreamListenerProtocol starting on 50157 > Feb 8 11:04:51 kamailio-dev media-relay[2474]: > mediaproxy.mediacontrol.StreamListenerProtocol starting on 50158 > Feb 8 11:04:51 kamailio-dev media-relay[2474]: > mediaproxy.mediacontrol.StreamListenerProtocol starting on 50159 > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3461]: NOTICE: <core> > [main.c:706]: Thank you for flying kamailio!!! > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3561]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3558]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3560]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3553]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3557]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3549]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3552]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3547]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3545]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3546]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3541]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3538]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3536]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3526]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3492]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3528]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3521]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3530]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3495]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3532]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3493]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3539]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3523]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3534]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3489]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3496]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3488]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3525]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3487]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3498]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3484]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3500]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3482]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3501]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3479]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3499]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3474]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3476]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3475]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3477]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3491]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3480]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3490]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3486]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev /sbin/kamailio[3483]: INFO: <core> > [main.c:817]: INFO: signal 15 received > Feb 8 11:05:06 kamailio-dev media-relay[2474]: (Port 50156 Closed) > Feb 8 11:05:06 kamailio-dev media-relay[2474]: (Port 50157 Closed) > Feb 8 11:05:06 kamailio-dev media-relay[2474]: (Port 50158 Closed) > Feb 8 11:05:06 kamailio-dev media-relay[2474]: (Port 50159 Closed) > Feb 8 11:05:08 kamailio-dev kamailio: INFO: <core> [tcp_main.c:4727]: > init_tcp: using epoll_lt as the io watch method (auto detected) > Feb 8 11:05:08 kamailio-dev /sbin/kamailio[3761]: INFO: usrloc > [hslot.c:53]: locks array size 512 > ...... > > Thanks and regards. > -- > Eduardo Lejarreta. > > -----Mensaje original----- > De: Daniel-Constantin Mierla [mailto:miconda@gmail.com] > Enviado el: viernes, 08 de febrero de 2013 9:38 > Para: lejarreta.e(a)sarenet.es; SIP Router - Kamailio (OpenSER) and SIP > Express Router (SER) - Users Mailing List > Asunto: Re: [SR-Users] kamctl restart and mediaproxy > > Hello, > > not using mediaproxy, but have you looked at its logs to see if prints some > messages? > > Cheers, > Daniel > > On 2/7/13 5:03 PM, Eduardo Lejarreta Langarica SARENET wrote: >> Good evening again. >> >> Kamailio 3.2.2 and mediaproxy 2.5.2 >> >> An long-time issue for us: >> >> kamctl restart stops mediaproxy audio. >> >> is there anyway to avoid this? >> >> Any idea will be helpful. >> >> Thanks and best regards. >> ---------------------------------------------------------------------- >> ------------------------------------------------------------- >> Eduardo Lejarreta >> >> >> >> >> _______________________________________________ >> 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 > http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda > Kamailio World Conference, April 16-17, 2013, Berlin > - http://conference.kamailio.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

2 5

branching for v4.4.x and unfreezing development
by Daniel-Constantin Mierla 19 Feb '13

19 Feb '13

Hello, it is time to sketch the road to the next major release, my plan is: - create branch 4.4 on Wednesday, February 27, 2013 - that will unfreeze development, so personal branches and contributions kept in external repositories can be merged in master branch - consider to release v4.4.0 in first or second week of March Any other opinions? Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

2 2

Nonce secret and expiry value
by Andreas Granig 19 Feb '13

19 Feb '13

Hi guys, Nonces have a default expiry value of 300 seconds. Does this still apply if the "secret" parameter is set? So can you use both "secret" and "nonce_expire" value together, taking for granted that the time is in sync over multiple proxies? Andreas

2 2

2 kamailio servers as proxy
by Andrei Rotaras 19 Feb '13

19 Feb '13

Hello, We have this situation: 2 servers, each with his own "location". First user is registered on one server and second user on other server. If we receive an invite for a user that is on other server's location, second server doesn't react on hostrewrite(dynamic routing,forward, append_branch, t_replicate, etc) from first server. Can you please advise us in what direction to look? Our kamailio version: kamailio-3.3.1-1.2.x86_64. (installed from rpm) Thank you! Best regards, Andrei Rotaras NOC Systems Engineer StarNet Moldova Phone: +(373-22) 844-770 Fax: +(373-22) 844-445 Mob: +(373-79) 700-552 E-mail: arotaras(a)starnet.md<mailto:arotaras@starnet.md> Address: 4th Floor, Calea Iesilor 10, Chisinau, Republic of Moldova Web: http://www.starnet.md<http://www.starnet.md/> [Description: Description: Description: starnet] This email message is confidential and may contain privileged information. If you are not the above named addressee, it may be unlawful for you to read, copy, distribute, disclose or otherwise use the information in this email message. If you are not the intended recipient of this email message please telephone one of the above indicated numbers immediately and delete this message.

2 1

Any substitute or replace function for the request-/response line of SIP messages available?
by Klaus Feichtinger 18 Feb '13

18 Feb '13

Hi, I have a very specific situation that the implementation of a customer SIP user agent is absolutely not standards conform. Therefore, I use Kamailio as a SIP protocol adapter between the non-standard conform user agent(s) and the typical RFC3261 compliant user agents. Most conversions are working fine. But in a very specific scenario I have to change/convert a SIP response message to a SIP request message and change the SIP method´s name. This conversion is no problem for the SIP header fields. However, I haven't found a way for adapting the response-/request line of a SIP message. This line seems to be protected. The described situation of the perverse conversion I try to realize is following (please do not ask "why?" or think I'm mad...): (mad) UA1 == INVITE ==> SIP-Prot-Adapter(Kamailio) == INVITE ==> (typ.)UA2 UA1 <== BYE == SIP-Prot-Adapter(Kamailio) <== 488 == (typ.)UA2 UA1 == BYE ==> SIP-Prot-Adapter(Kamailio) == ACK ==> (typ.)UA2 I use the "textops" module for substitution and/or replacement in SIP header fields. But the "subst" and "replace" functions cannot be used for the request-/response line on top of the SIP messages. I was a little bit irritated, as the "search" function is still working on these lines, but not the subst/replace function. Creating a new SIP message with the UAC module (as alternative) is not okay, as this new message does not use the same values for Call-ID, From-tag a.s.o. and these values can't be changed before sending the message. Does anybody know a way / function / method for manipulating the request-/response line of a message? It would be perfect using a function á la "sed"/subst. Thanks Klaus P.S. I tested with kamailio-3.3.3

2 3

Kamailio and Ubuntu Upstart script
by Moacir Ferreira 17 Feb '13

17 Feb '13

As I posted here before, Ubuntu has moved to Upstart as the main engine to start daemons. As a result, Kamailio will fail to start on boot time because it starts before MySQL that has already been moved to the new Upstart daemon starting fashion. I am struggling to get some help from the Ubuntu community explaining how we can match the previous init script behavior when using Upstart but I got no success till now. Anyway, I have created an Upstart script for Kamailio that does most everything that the old script use to do. However, in this script I am making sure MySQL starts before Kamailio. If you have another SQL engine you must change the script to match your SQL engine by changing the “start on started” statement on the script. To use it you have to: A – Create a kamailio.conf file at “/etc/init/” with the contents of the kamailio.conf script in attach, that is the new Upstart stile init script. Please notice that I just adapted the existing script, no news… B – Create a symbolic link at /etc/init.d/ pointing to this kamailio.conf at /etc/init/. C – If you want, you can still have a /etc/default/kamailio file like it used to be and the script will behave like it used to behave when using the default file. Hopefully someone more “influent” at the Ubuntu community may get some help on how to adapt their “start on started” Upstart stanza to behave like we need for multiple possible SQL engines. Cheers! Moacir

3 2

app_python rewrite_ruri - BUG: do_action bad set*()
by V Tone 15 Feb '13

15 Feb '13

Hi, I'm having problems getting started with the app_python module. I'm getting the following in the logs: 5(8890) : <core> [action.c:785]: BUG: do_action: bad set*() type 14 5(8890) ERROR: <core> [action.c:1568]: run action error at: :0 5(8890) ERROR: app_python [python_msgobj.c:115]: Error in do_action 5(8890) ERROR: app_python [python_support.c:38]: python_exec2: Unhandled exception in the Python code: 5(8890) ERROR: app_python [python_support.c:78]: RuntimeError: Error in do_action My Python module is very basic: class Router(object): def child_init(self, child_id): return 0 def route(self, msg): msg.rewrite_ruri('sip:1000@1.2.3.4:5060') return 0 def mod_init(): instance = Router() return instance I've tested this in Kamailio 3.2 and 3.3. Can someone point me to what I'm doing wrong? Many thanks in advance, vtone

2 4

siptrace/sipcapture and MSRP
by Peter Dunkley 15 Feb '13

15 Feb '13

Hello, Is there any reason why I couldn't use the existing siptrace() function in event_route[msrp:frame-in] to relay all MSRP messages to a SIP Capture node? Will the SIP Capture node handle receiving these messages? What will end up in the WebHomer display in these cases? Regards, Peter

3 7

Warning - What does this mean?
by Krishna Kurapati 15 Feb '13

15 Feb '13

I am seeing this error lately. /usr/local/sbin/kamailio[28559]: WARNING: tm [t_lookup.c:1543]: WARNING: script writer didn't release transaction What does this mean? Krish Kura

1 0

Dialog is not cleaned up after failed calls and BYEs.
by Aft nix 15 Feb '13

15 Feb '13

Hi, I'm trying to limit concurrent calls through kamailio using following script: #!ifdef WITH_CALL_LIMIT if (is_method("INVITE")) { dlg_manage(); $var(100) = 0; get_profile_size("callquota", "$fd", "$var(100)"); if ($var(100) >= 100 ) { xdbg("DEBUG: Simultaneous calls limit reached"); sl_send_reply("503","Simultaneous calls limit reached"); exit; } set_dlg_profile("callquota","$fd"); } #!endif But the dialogs are not cleaned up after the call ends or a failed call happens. For example after two failed calls : [root@new tls]# kamctl fifo profile_get_size callquota profile:: name=callquota value= count=4 and the dialog list : [root@new tls]# kamctl fifo dlg_list dialog:: hash=896:11654 state:: 1 ref_count:: 1 timestart:: 0 timeout:: 0 callid:: iSGRqQR0C4suBpQD0rHAEgV42vw0Sbpe from_uri:: sip:123@205.164.40.150 from_tag:: d8x2LDBptN9FRJWtqFTj4DIpuQlKq.KN caller_contact:: sip:123@180.149.7.45:33852;ob caller_cseq:: 11755 caller_route_set:: caller_bind_addr:: udp:185.8.105.70:9547 callee_bind_addr:: to_uri:: sip:8801684077496@205.164.40.150 to_tag:: callee_contact:: callee_cseq:: callee_route_set:: dialog:: hash=896:11655 state:: 1 ref_count:: 1 timestart:: 0 timeout:: 0 callid:: iSGRqQR0C4suBpQD0rHAEgV42vw0Sbpe from_uri:: sip:123@205.164.40.150 from_tag:: d8x2LDBptN9FRJWtqFTj4DIpuQlKq.KN caller_contact:: sip:123@180.149.7.45:33852;ob caller_cseq:: 11755 caller_route_set:: caller_bind_addr:: udp:185.8.105.70:9547 callee_bind_addr:: to_uri:: sip:8801684077496@205.164.40.150 to_tag:: callee_contact:: callee_cseq:: callee_route_set:: dialog:: hash=896:11657 state:: 1 ref_count:: 1 timestart:: 0 timeout:: 0 callid:: iSGRqQR0C4suBpQD0rHAEgV42vw0Sbpe from_uri:: sip:123@205.164.40.150 from_tag:: d8x2LDBptN9FRJWtqFTj4DIpuQlKq.KN caller_contact:: sip:123@180.149.7.45:33852;ob caller_cseq:: 11756 caller_route_set:: caller_bind_addr:: udp:185.8.105.70:9547 callee_bind_addr:: to_uri:: sip:8801684077496@205.164.40.150 to_tag:: callee_contact:: callee_cseq:: callee_route_set:: dialog:: hash=1532:8089 state:: 1 ref_count:: 1 timestart:: 0 timeout:: 0 callid:: povqQaC.sRhKvmUaLzzR83bRXxouoNy8 from_uri:: sip:123@205.164.40.150 from_tag:: qXQll3QXrff.t27FHY2hucPfJ7K8Pfrs caller_contact:: sip:123@180.149.7.45:33852;ob caller_cseq:: 9232 caller_route_set:: caller_bind_addr:: udp:185.8.105.70:9547 callee_bind_addr:: to_uri:: sip:8801684077496@205.164.40.150 to_tag:: callee_contact:: callee_cseq:: callee_route_set:: You can see there are multiple entries. And these dialog does not get cleaned up. The data is collected when there was no call(pending or connected). My kamailio version is : [root@new tls]# kamailio -V version: kamailio 4.0.0-pre0 (x86_64/linux) 83eafc 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, 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, DEFAULT PKG_SIZE 4MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 83eafc compiled on 18:34:58 Jan 17 2013 with gcc 4.4.6 Any suggestion?(It seems dialog module is not working as it should be. I've also tried calling unset_profile() forcefully in failure route with no luck) -- -aft

1 0

Registration and TCP connection
by Krishna Kurapati 15 Feb '13

15 Feb '13

Hi, When client connects to server over TCP/TLS and REGISTERs two things that can happen. 1. Client can disconnect the connection when the App is killed abruptly and the client did not get a chance to send unregiter. 2. CRLF KA did not arrive due to network failures in the middle. Is there a way to remove contact from usrloc table. This can spare server from trying to send message to a client who is no longer connected to the server. It would be ideal if this can be done automatically with a module param in tcp or registrar. Any suggestion on how to tie TCP connection with usrloc? Thanks Krish Kura

2 1

kamailio - simple presence over port 5061
by David Thomson 15 Feb '13

15 Feb '13

I was wondering if simple presence can work over port 5061 or if it always goes over 5060. my client (jitsi) is configured to work over port 5061, xcap is configured to work over port 5061 but I am still seeing some messages flowing over port 5060. any ideas as to what is going on? ttyl,Dave

2 1

[Fwd: Re: Deleting the expired contacts using the Rtimer module]
by kiran bhosale 15 Feb '13

15 Feb '13

he array values that we are modifying and the function that we call periodically are from our module . to exemplify i have put the code below .our requirement is that we want to delete the expired contacts without relying on usrloc module as "we have not used any DB for contact storage" but developed our own module for storing them in a file.even we had installed the SIGALRM handler in our module for decrementing the contacts but the values of array (that stores the expires) elements were zero. we thought of workaround to store the expires values to file and decrement them periodically. but frequent I/O is very consuming. whats the solution for deleting the expired contacts . #define DB_PATH "/usr/local/etc/kamailio/dbtext/syntel" #define DB_PATH1 "/usr/local/etc/kamailio/dbtext/temp" #define MAX_BUFFER 128 #define MAX_USERS 128 typedef struct { int max_bindings; int syntel_index; int expires[8]; char ip_addr[8][16]; } Syntel_users; Syntel_users syntel_database[16]={{0}}; //this is function used for storing the contacts int write_to_file (char *uri, char *user, int locator,int expires) { int Retval = 0; int fd; int bytes; char ip[13]; strncpy (ip, &uri[9], 12); ip[12] = '\0'; fd = open (DB_PATH, O_CREAT | O_RDWR | O_APPEND, 0777); if (syntel_database[locator].max_bindings < 8) { strcpy (syntel_database[locator].ip_addr[syntel_database[locator].max_bindings], uri); syntel_database[locator].expires[syntel_database[locator].max_bindings]=expires; syntel_database[locator].max_bindings++; bytes = write (fd, uri, strlen (uri)); Retval = 1; } return Retval; } //AND THIS IS FUNCTION WE CALL FROM kamailio.conf WITH RTIMER MODULE AFTER INTERVAL OF 1 SECOND. int syntel_delete (void) { int counter; int j; for (counter = 0; counter < 16; counter++) { for (j = 0; j < syntel_database[counter].max_bindings; j++) { if(syntel_database[counter].expires[j]<=0) { memset(syntel_database[counter].ip_addr[j],0,strlen(syntel_database[counter].ip_addr[j])); syntel_database[counter].max_bindings--; } else syntel_database[counter].expires[j]--; } } return 0; } BUT WHAT THIS FUNCTION GETS IS ZERO VALUES FOR EXPIRES AND NOT MODIFIED IN SYNTEL_WRITE_TO_FILE HOW TO DECREMENT THESE VALUES PERIODICALLY THEN ??

2 3

Fwd:
by Maurice Vistan 15 Feb '13

15 Feb '13

http://www.hotelgloriachiancianoterme.it/ako7ed.php?s=lf

1 0

How to rewrite "tu" in CANCEL message ?
by Julia 14 Feb '13

14 Feb '13

Hello all, Is there any way to rewrite "tu" in CANCEL message in 3.2.1? We use remove_hf / insert_hf for INVITE's "tu" modification before t_relay. It works fine, but it doesn't work for CANCEL (INVITE re-parsing is disabled). Ths,

8 19

Openfire and lync 2010 integration
by Vinay V 13 Feb '13

13 Feb '13

Hi All, I am in process of integrating lync 2010 with openfire xmpp server. Reading some of the blogs on kamailio found that it can be done using kamailio. Hence request if some body has done this please your inputs. Regards, Vinay

1 1

Re: [SR-Users] kamctl restart and mediaproxy
by Mino Haluz 13 Feb '13

13 Feb '13

You have to figure it somehow with adjusting the python code of media-dispatcher. We have developed special "/etc/init.d/kamailio grace" command, which waits for the moment, when there are no calls, so it can be restarted. It queries mediaproxy every second. In future we will not use mediaproxy anymore - it is not supported in latest version either. rtpproxy gives us better performance results. On Tue, Feb 12, 2013 at 2:44 PM, Eduardo Lejarreta <lejarreta.e(a)sarenet.es>wrote: > Good evening. > > Launching "mediaproxy-dispatcher" and "mediaproxy-relay" on the foreground > and with "DEBUG" we can see: > > mediaproxy-dispatcher log: > ------------------------------------------------------------------> Kamctl > restart > debug: Connection to OpenSIPS lost: Connection was closed cleanly. > debug: Connection to OpenSIPS lost: Connection was closed cleanly. > debug: Connection to OpenSIPS lost: Connection was closed cleanly. > debug: Issuing "remove" command to relay at A.B.C.D > debug: Got statistics: {xxxxxxxxxxxxxxxxxxx} > debug: Connection to OpenSIPS lost: Connection was closed cleanly. > > mediaproxy-relay log: > ------------------------------------------------------------------> Kamctl > restart > debug: removing session xxxxxxxxxxxxxxxxxxx > (Port 50000 Closed) > (Port 50001 Closed) > (Port 50002 Closed) > (Port 50003 Closed) > > So "mediaproxy-dispatcher" doesn't receive any command from Kamailio, he > just detects " Connection was closed cleanly" and then issues a "remove" > command to "mediaproxy-relay". > > Does anybody know any way to avoid this? > > Thanks and regards. > -- > Eduardo Lejarreta. > > > _______________________________________________ > 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

hadnling the IP extension calls freeing the PBX channel
by kiran bhosale 13 Feb '13

13 Feb '13

hello , w have PBX gateway along with kamailio. all calls are routed to pbx extensions. but if it happens to be an IP extrension we want the kamailio to handle the speech without keeping the PBX channel busy. if its an analog extension then this channel can be busy but if its IP extension kamailio should handle the speech itself. how is this possible ?? thanks in advance.

1 0

pseudovariable values written from app_lua sometimes appear incorrect
by elactrum＠jamailca.com 12 Feb '13

12 Feb '13

Hi, I am using app_lua to frequently store some data in htable and avp entries, using sr.pv.sets. When I read the values using sr.pv.get immediately after writing them, they seem correct. However, when I read them later from Lua, I have noticed that sometimes the value is incorrect or nil. Am I doing something wrong? The following code uses rtimer to duplicate the problem on startup, printing the errors to syslog. Tested on version: kamailio 3.3.1 (i386/linux) 2d0f6f ############### # kamailio.cfg ############### loadmodule "pv.so" loadmodule "xlog.so" loadmodule "htable.so" loadmodule "rtimer.so" loadmodule "app_lua" modparam("htable", "htable", "status=>size=14;autoexpire=0;") modparam("rtimer", "timer", "name=ta;interval=10;mode=1") modparam("rtimer", "exec", "timer=ta;route=LUA_TEST") # the following path may need to change depending on filesystem setup modparam("app_lua", "load", "/etc/kamailio/kamailio.lua") route { route(LUA_TEST); } route[LUA_TEST] { $var(lua_test_key) = $(pp{s.toupper}) + "::"; lua_runstring("htable_test([[status=>$var(lua_test_key)]])"); lua_runstring("avp_test([[status=>$var(lua_test_key)]])"); } ---------------- -- kamailio.lua ---------------- -- Test of sr.pv operations in Lua function htable_test (namekey) local pvar_prefix = "$sht("..namekey pvar_test(pvar_prefix) end function avp_test ( name ) local pvar_prefix = "$avp("..name pvar_test(pvar_prefix) end function pvar_test ( pvar_prefix ) for value = 10000, 10050 do local pvar = pvar_prefix..value..")" --sr.log("info", "Lua attempting to set '"..pvar.."' to value '"..value.."'") sr.pv.sets(pvar, value) end for value = 10000, 10050 do -- now we check what was stored local pvar = pvar_prefix..value..")" local checkvar = sr.pv.get(pvar) if ( tostring(checkvar) ~= tostring(value) ) then sr.err(string.format("CHECK: Lua checking pvar '%s' and got value '%s' which is NOT '%s'", tostring(pvar), tostring(checkvar), tostring(value) )) end end end -- end of kamailio.lua Thanks, -Andrew

2 6

DNS error
by Vitaliy Aleksandrov 12 Feb '13

12 Feb '13

Hi, all Have a question about failure_routes. As I remember if kamailio statefuly sent a request and didn't receive a reply script writer can handle this situation at a failure_route[] block. But if kamailio can't resolve a destination from R-URI it sends back 478 reply and finishes current transaction without reaching a failure_route[]. Is it possible to process such a request in a failure_route[] ? Additional details: version: kamailio-3.2.4 t_on_failure() is called before t_relay() I have checked with REGISTER and INVITE transactions.

1 0

Re: [SR-Users] sr-users Digest, Vol 93, Issue 13
by madhumanjusha＠integramicro.com 12 Feb '13

12 Feb '13

Yes,Yes I did it the way tm and mediaproxy modules are added... when i check /usr/local/lib/kamailio/modules I found my module and $stat module time is effected. 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 >> >> >> > seems that you want to create your custom module. have you checked the > makefile of your module. in order to develop the custom module just copy > the existing module and make relevant changes there. this is genuine way . > for this u need to include makefile.lst ,Makefile in your module. so best > way is to copy the smallest module (e.g.) pdb module and make changes. > On Fri, February 8, 2013 3:36 am, sr-users-request(a)lists.sip-router.org wrote: > Send sr-users mailing list submissions to > sr-users(a)lists.sip-router.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > or, via email, send a message with subject or body 'help' to > sr-users-request(a)lists.sip-router.org > > You can reach the person managing the list at > sr-users-owner(a)lists.sip-router.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of sr-users digest..." > > > Today's Topics: > > > 1. Reg. shared library linking during compiling time > (madhumanjusha(a)integramicro.com) > 2. Re: Reg. shared library linking during compiling time > (kiran bhosale) > 3. Re: Multiple radius responses on a missed call > (Daniel-Constantin Mierla) > 4. Re: Reg. shared library linking during compiling time > (Daniel-Constantin Mierla) > > > > ---------------------------------------------------------------------- > > > Message: 1 > Date: Fri, 8 Feb 2013 01:46:58 -0500 (EST) > From: madhumanjusha(a)integramicro.com > Subject: [SR-Users] Reg. shared library linking during compiling time > To: sr-users(a)lists.sip-router.org > Message-ID: > <38400.61.8.152.138.1360306018.squirrel(a)mail.integramicro.com> > Content-Type: text/plain;charset=iso-8859-1 > > > > Hello All, > Please find below steps...and kindly let me know the error solution. > > > STEP 1:: > Under modules I created a directory name b2bentities > /root/kamailio-3.3.2/modules/b2bentities > $ls > b2bentities.so b2b_server.c b2b_server.d b2b_server.h b2b_server.o > dialog.h librpath.lst makecfg.lst Makefile > > In b2b_server.h > #ifndef SERVER_H > #define SERVER_H > #endif > > > int b2b_dlg_reply(); > > > > In b2b_server.c I wrote following code :: > > > int b2b_dlg_reply() { > LOG(L_INFO,"Hello am in b2bua module\n"); > return 0; } > > > Also,I added below code to the same file as per steps given in kamailio > server's developer's guide. In developer's guide,they told to initialize > parameters as per module's requirement but i made them all empty and > null... for example static int mod_init(void) { > LOG(L_INFO,"IN mod_init\n"); > return 0; } > > > static cmd_export_t commands[] = { }; > > > > static param_export_t params[]={ }; > > > struct module_exports exports = { "b2bentities", // module name > NULL, // dlopen flags > commands, // exported functions params, // exported parameters > NULL, // exported statistics > NULL, // exported MI functions > NULL, // exported pseudo-variables > NULL, // extra processes > mod_init, // module init function (before fork. kids will inherit) > NULL, // reply processing function > NULL, // destroy function > child_init // child init function }; > > > > STEP 2:: > In kamailio core > file : In receive.c Method Name : In receive_msg > > > i did this way... > > In receive.c add follow piece of code:: > > > #include "modules/b2bentities/b2b_server.h" > > > int receive_msg(....) { > /*my code*/ > int b2b_ret; int (*ptr)(); ptr= b2b_dlg_reply; > if(msg->REQ_METHOD!=METHOD_REGISTER) { > > > b2b_ret=(*ptr)(); if(b2b_ret!=0) { > LOG(L_INFO,"B2B Dialogue reply function failed\n"); > exit(1); } > else { > LOG(L_INFO,"B2B Dialogue reply function success\n"); > exit(1); } > > > } > > > > } > > > STEP 3:: > I even added kamailio.cfg load module "b2bentities.so" > > > > > STEP 4:: > Final step,when I do make all..... > its giving me the following error :: [root@kamserver kamailio-3.3.2]# make > all CC (gcc) [kamailio] receive.o > LD (gcc) [kamailio] kamailio > receive.o: In function `receive_msg': > /root/kamailio-3.3.2/receive.c:163: undefined reference to `b2b_dlg_reply' > collect2: ld returned 1 exit status > make: *** [kamailio] Error 1 > > > > > My doubt: > Where should my b2bentities.so should be linked during compiling time? > Please tell me the steps for it. > > > Thanks & Regards, > Manjusha A. > Integra Micro Software Services (P) Ltd. > > > > > > > > > > > ------------------------------ > > > Message: 2 > Date: Fri, 08 Feb 2013 12:36:36 +0530 > From: kiran bhosale <kbhosale(a)synteltelecom.com> > Subject: Re: [SR-Users] Reg. shared library linking during compiling > time To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List" <sr-users(a)lists.sip-router.org> > Message-ID: <5114A3FC.9000902(a)synteltelecom.com> > Content-Type: text/plain; charset=us-ascii; format=flowed > > > madhumanjusha(a)integramicro.com wrote: >> Hello All, >> Please find below steps...and kindly let me know the error solution. >> >> >> STEP 1:: >> Under modules I created a directory name b2bentities >> /root/kamailio-3.3.2/modules/b2bentities >> $ls >> b2bentities.so b2b_server.c b2b_server.d b2b_server.h b2b_server.o >> dialog.h librpath.lst makecfg.lst Makefile >> >> In b2b_server.h >> #ifndef SERVER_H >> #define SERVER_H >> #endif >> >> >> int b2b_dlg_reply(); >> >> >> >> In b2b_server.c I wrote following code :: >> >> >> int b2b_dlg_reply() { >> LOG(L_INFO,"Hello am in b2bua module\n"); >> return 0; } >> >> >> Also,I added below code to the same file as per steps given in kamailio >> server's developer's guide. In developer's guide,they told to >> initialize parameters as per module's requirement but i made them all >> empty and null... for example static int mod_init(void) { >> LOG(L_INFO,"IN mod_init\n"); >> return 0; } >> >> >> static cmd_export_t commands[] = { }; >> >> >> >> static param_export_t params[]={ }; >> >> >> struct module_exports exports = { "b2bentities", // module name >> NULL, // dlopen flags >> commands, // exported functions params, // exported >> parameters NULL, // exported statistics >> NULL, // exported MI functions >> NULL, // exported pseudo-variables >> NULL, // extra processes >> mod_init, // module init function (before fork. kids will >> inherit) NULL, // reply processing function >> NULL, // destroy function >> child_init // child init function }; >> >> >> >> STEP 2:: >> In kamailio core >> file : In receive.c Method Name : In receive_msg >> >> >> i did this way... >> >> In receive.c add follow piece of code:: >> >> >> #include "modules/b2bentities/b2b_server.h" >> >> >> int receive_msg(....) { >> /*my code*/ >> int b2b_ret; int (*ptr)(); ptr= b2b_dlg_reply; >> if(msg->REQ_METHOD!=METHOD_REGISTER) { >> >> >> b2b_ret=(*ptr)(); if(b2b_ret!=0) { >> LOG(L_INFO,"B2B Dialogue reply function failed\n"); >> exit(1); } >> else { >> LOG(L_INFO,"B2B Dialogue reply function success\n"); >> exit(1); } >> >> >> } >> >> >> >> } >> >> >> STEP 3:: >> I even added kamailio.cfg load module "b2bentities.so" >> >> >> >> >> STEP 4:: >> Final step,when I do make all..... >> its giving me the following error :: [root@kamserver kamailio-3.3.2]# >> make all CC (gcc) [kamailio] receive.o >> LD (gcc) [kamailio] kamailio >> receive.o: In function `receive_msg': >> /root/kamailio-3.3.2/receive.c:163: undefined reference to >> `b2b_dlg_reply' >> collect2: ld returned 1 exit status >> make: *** [kamailio] Error 1 >> >> >> >> >> My doubt: >> Where should my b2bentities.so should be linked during compiling time? >> Please tell me the steps for it. >> >> >> Thanks & Regards, >> Manjusha A. >> Integra Micro Software Services (P) Ltd. >> >> >> >> >> >> >> >> >> _______________________________________________ >> 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 >> >> >> > seems that you want to create your custom module. have you checked the > makefile of your module. in order to develop the custom module just copy > the existing module and make relevant changes there. this is genuine way . > for this u need to include makefile.lst ,Makefile in your module. so best > way is to copy the smallest module (e.g.) pdb module and make changes. > > > > > ------------------------------ > > > Message: 3 > Date: Fri, 08 Feb 2013 09:27:07 +0100 > From: Daniel-Constantin Mierla <miconda(a)gmail.com> > Subject: Re: [SR-Users] Multiple radius responses on a missed call > To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List" <sr-users(a)lists.sip-router.org> > Message-ID: <5114B6DB.10605(a)gmail.com> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > > Hello, > > > there are different event logged for accounting: - missed call event which > is sent from the point of view of callee - transaction answered which is > sent from the point of view of caller > > In db (eg mysql) accounting, each event in stored in a different table. > I don't know how they are in radius. You probably get two missed call > events and one transaction answered. > > If you want only one, do not set the flag for missed calls. > > > Cheers, > Daniel > > > On 2/7/13 9:23 PM, Efelin Novak wrote: > >> Hi folks, >> >> >> I'd like to ask you how can I solve an issue when I'm receiving three >> RADIUS requests on one missed call. My scenario is as follows: >> >> >> A calls to B. B returns 503 error message, INIVITE is then sent to C >> and C sends 486. >> >> After first 503 I receive a first RADIUS request. This is correct. >> >> >> However after 486 I receive three RADIUS requests with different IDs. >> As you can see in a DEBUG one RADIUS is send after the SIP reply >> message is received and second and third RADIUS messages are sent after >> the SIP reply message is forwarded. >> >> I'd like to receive only two RADIUS messages. One after 503 and second >> after 486. >> >> >> FLT_ACC is defined as 1. My config is as follows: >> modparam("acc", "failed_transaction_flag", FLT_ACC) modparam("acc", >> "report_cancels", 1) >> modparam("acc", "report_ack", 0) modparam("acc", >> "detect_direction", 0) >> modparam("acc", "early_media", 0) modparam("acc", "log_level", >> 5) >> modparam("acc", "log_flag", 1) modparam("acc", >> "log_missed_flag", 1) >> modparam("acc_radius", "radius_config", >> "/etc/radiusclient-ng/radiusclient.conf") # This is the >> location of the configuration file of radius client >> modparam("acc_radius", "radius_flag", FLT_ACC) >> modparam("acc_radius", "radius_missed_flag", FLT_ACC) >> >> In a route I set setflag("FLT_ACC") only once per call. >> >> >> Here is the debug for debug level 7: >> 0(5890) DEBUG: <core> [parser/msg_parser.c:634]: SIP Reply (status): >> 0(5890) DEBUG: <core> [parser/msg_parser.c:636]: version: <SIP/2.0> >> 0(5890) DEBUG: <core> [parser/msg_parser.c:638]: status: <486> >> 0(5890) DEBUG: <core> [parser/msg_parser.c:640]: reason: <Busy here> >> 0(5890) DEBUG: <core> [parser/parse_via.c:1286]: Found param type >> 232, <branch> = <z9hG4bK8d9c.4438d232.1>; state=6 >> 0(5890) DEBUG: <core> [parser/parse_via.c:1286]: Found param type >> 234, <received> = <192.168.21.101>; state=16 >> 0(5890) DEBUG: <core> [parser/parse_via.c:2561]: end of header reached, >> state=5 0(5890) DEBUG: <core> [parser/msg_parser.c:511]: parse_headers: >> Via >> found, flags=2 0(5890) DEBUG: <core> [parser/msg_parser.c:513]: >> parse_headers: this >> is the first via 0(5890) DEBUG: <core> [receive.c:149]: After >> parse_msg... 0(5890) DEBUG: tm [t_lookup.c:1079]: DEBUG: t_check_msg: >> msg id=6 global id=5 T start=0xffffffff 0(5890) DEBUG: <core> >> [parser/parse_via.c:1286]: Found param type >> 232, <branch> = <z9hG4bK8d9c.e7b650e1.0>; state=16 >> 0(5890) DEBUG: <core> [parser/parse_via.c:2561]: end of header reached, >> state=5 0(5890) DEBUG: <core> [parser/msg_parser.c:511]: parse_headers: >> Via >> found, flags=22 0(5890) DEBUG: <core> [parser/msg_parser.c:524]: >> parse_headers: this >> is the second via 0(5890) DEBUG: <core> [parser/parse_via.c:1286]: Found >> param type 232, <branch> = <z9hG4bK463ae086>; state=6 >> 0(5890) DEBUG: <core> [parser/parse_via.c:1286]: Found param type >> 235, <rport> = <5060>; state=16 >> 0(5890) DEBUG: <core> [parser/parse_via.c:2561]: end of header reached, >> state=5 0(5890) DEBUG: <core> [parser/msg_parser.c:511]: parse_headers: >> Via >> found, flags=22 0(5890) DEBUG: <core> [parser/parse_to.c:178]: DEBUG: >> add_param: tag=as493d6134 >> 0(5890) DEBUG: <core> [parser/parse_to.c:802]: end of header reached, >> state=29 0(5890) DEBUG: <core> [parser/msg_parser.c:188]: DEBUG: >> get_hdr_field: <To> [42]; uri=[sip:5556003900000@dr.vm] >> 0(5890) DEBUG: <core> [parser/msg_parser.c:190]: DEBUG: to body >> [<sip:5556003900000@dr.vm>] >> 0(5890) DEBUG: <core> [parser/msg_parser.c:168]: get_hdr_field: cseq >> <CSeq>: <102> <INVITE> >> 0(5890) DEBUG: tm [t_lookup.c:965]: DEBUG: t_reply_matching: hash >> 51672 label 590185284 branch 1 >> 0(5890) DEBUG: tm [t_lookup.c:1016]: DEBUG: t_reply_matching: reply >> matched (T=0x871afc98)! 0(5890) DEBUG: tm [t_hooks.c:288]: DBG: >> trans=0x871afc98, callback type 2, id 0 entered 0(5890) DEBUG: acc >> [acc_logic.c:537]: acc callback called for >> t(0x871afc98) event type 2, reply code 486 0(5890) DEBUG: tm >> [t_hooks.c:288]: DBG: trans=0x871afc98, callback >> type 2, id 0 entered 0(5890) DEBUG: <core> [parser/parse_to.c:178]: >> DEBUG: add_param: tag=as5c1d222e >> 0(5890) DEBUG: <core> [parser/parse_to.c:802]: end of header reached, >> state=29 0(5890) DEBUG: siptrace [siptrace.c:794]: storing info... >> 0(5890) DEBUG: tm [t_lookup.c:1148]: DEBUG: t_check_msg: msg id=6 >> global id=6 T end=0x871afc98 0(5890) DEBUG: tm [t_reply.c:2081]: DEBUG: >> reply_received: org. >> status uas=180, uac[1]=180 local=0 is_invite=1) 0(5890) DEBUG: tm >> [t_hooks.c:288]: DBG: trans=0x871afc98, callback >> type 4194304, id 0 entered 0(5890) DEBUG: siptrace [siptrace.c:794]: >> storing info... 0(5890) DEBUG: tm [t_lookup.c:1079]: DEBUG: t_check_msg: >> msg id=6 global id=6 T start=0x871afc98 0(5890) DEBUG: tm >> [t_lookup.c:1151]: DEBUG: t_check_msg: T already found! >> 0(5890) DEBUG: tmx [t_var.c:438]: reply code is <486> >> 0(5890) INFO: <script>: XLOG: 38e422e10af08b2e34d58e5b765e81d3(a)dr.vm >> [REPLY_TWO] Incoming reply code: 486 0(5890) DEBUG: tm >> [t_reply.c:1204]: ->>>>>>>>> T_code=180, new_code=486 >> 0(5890) DEBUG: tm [t_hooks.c:288]: DBG: trans=0x871afc98, callback >> type 128, id 0 entered 0(5890) DEBUG: acc [acc_logic.c:537]: acc callback >> called for t(0x871afc98) event type 128, reply code 486 0(5890) acc >> [acc.c:275]: ACC: call missed: >> timestamp=1360265692;method=INVITE;from_tag=as5c1d222e;to_tag=as493d6134 >> ;call_id=38e422e10af08b2e34d58e5b765e81d3(a)dr.vm;code=486;reason=Busy >> here 0(5890) DEBUG: acc [acc.c:872]: missed event for engine: radius >> 0(5890) DEBUG: <core> [md5utils.c:67]: DEBUG: MD5 calculated: >> 6e1b9f35a8b3e65859e29aefc1c26fc3 >> 0(5890) DEBUG: exec [exec_hf.c:267]: SIP_HF_CONTENT_LENGTH=262 >> 0(5890) DEBUG: exec [exec_hf.c:267]: SIP_HF_CONTENT_TYPE=application/sdp >> 0(5890) DEBUG: exec [exec_hf.c:267]: SIP_HF_SUPPORTED=replaces >> 0(5890) DEBUG: exec [exec_hf.c:267]: SIP_HF_ALLOW=INVITE, ACK, >> CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO >> 0(5890) DEBUG: exec [exec_hf.c:267]: SIP_HF_DATE=Thu, 07 Feb 2013 >> 19:35:53 GMT >> 0(5890) DEBUG: exec [exec_hf.c:267]: SIP_HF_MAX_FORWARDS=68 >> 0(5890) DEBUG: exec [exec_hf.c:267]: SIP_HF_USER_AGENT=Asterisk PBX >> 0(5890) DEBUG: exec [exec_hf.c:267]: SIP_HF_CSEQ=102 INVITE >> 0(5890) DEBUG: exec [exec_hf.c:267]: >> SIP_HF_CALLID=38e422e10af08b2e34d58e5b765e81d3(a)dr.vm >> 0(5890) DEBUG: exec [exec_hf.c:267]: >> SIP_HF_CONTACT=<sip:0221028400@192.168.21.106> >> 0(5890) DEBUG: exec [exec_hf.c:267]: SIP_HF_TO=<sip:5556003900000@dr.vm> >> 0(5890) DEBUG: exec [exec_hf.c:267]: SIP_HF_FROM="Janko" >> <sip:0221028400@dr.vm>;tag=as5c1d222e >> 0(5890) DEBUG: exec [exec_hf.c:267]: SIP_HF_VIA=SIP/2.0/UDP >> 192.168.21.100;branch=z9hG4bK8d9c.e7b650e1.0,SIP/2.0/UDP >> 192.168.21.106:5060;branch=z9hG4bK463ae086;rport=5060 >> 0(5890) DEBUG: exec [exec_mod.c:175]: executing >> [/opt/kamrouter/scripts/notify_reroute.sh 503 >> sip:3333003900000@ast-car.vm sip:0221028400@dr.vm >> 38e422e10af08b2e34d58e5b765e81d3(a)dr.vm] >> sh: /opt/kamrouter/scripts/notify_reroute.sh: not found >> 0(5890) INFO: <core> [main.c:797]: INFO: dont_fork turned on, living on >> 0(5890) DEBUG: tm [t_lookup.c:1079]: DEBUG: t_check_msg: msg id=0 >> global id=0 T start=0x871afc98 0(5890) DEBUG: tm [t_lookup.c:1151]: >> DEBUG: t_check_msg: T already found! >> 0(5890) DEBUG: tmx [t_var.c:438]: reply code is <486> >> 0(5890) INFO: <script>: XLOG: 38e422e10af08b2e34d58e5b765e81d3(a)dr.vm >> [FAIL_TWO] Rerouted gw returned 486 0(5890) INFO: <script>: XLOG: >> 38e422e10af08b2e34d58e5b765e81d3(a)dr.vm [aaa] VYMAZAVAM3 0(5890) DEBUG: >> tm [t_reply.c:1702]: DEBUG: relay_reply: branch=1, save=0, relay=1 >> 0(5890) DEBUG: <core> [msg_translator.c:1933]: old size: 572, new >> size: 485 >> 0(5890) DEBUG: <core> [msg_translator.c:1951]: >> build_res_from_sip_res: copied size: orig:110, new: 23, rest: 462 msg= >> SIP/2.0 486 Busy here >> Via: SIP/2.0/UDP 192.168.21.100;branch=z9hG4bK8d9c.e7b650e1.0 >> Via: SIP/2.0/UDP 192.168.21.106:5060;branch=z9hG4bK463ae086;rport=5060 >> From: "Janko" <sip:0221028400@dr.vm>;tag=as5c1d222e >> To: <sip:5556003900000@dr.vm>;tag=as493d6134 >> Call-ID: 38e422e10af08b2e34d58e5b765e81d3(a)dr.vm >> CSeq: 102 INVITE >> Server: Asterisk PBX 11.0.2 >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, >> INFO, PUBLISH >> Supported: replaces, timer >> Content-Length: 0 >> >> >> >> 0(5890) DEBUG: tm [t_hooks.c:288]: DBG: trans=0x871afc98, callback >> type 1048576, id 0 entered 0(5890) DEBUG: dialog [dlg_hash.c:588]: ref >> dlg 0x871af3f0 with 1 -> 2 0(5890) DEBUG: dialog [dlg_hash.c:590]: >> dialog id=3894 found on entry 2825 0(5890) DEBUG: dialog >> [dlg_hash.c:940]: dialog 0x871af3f0 changed >> from state 2 to state 5, due event 4 (ref 2) 0(5890) DEBUG: dialog >> [dlg_hash.c:735]: ref dlg 0x871af3f0 with 1 -> 3 >> 0(5890) DEBUG: dialog [dlg_handlers.c:1469]: executing event_route 4 on >> state 5 0(5890) INFO: <script>: XLOG: >> 38e422e10af08b2e34d58e5b765e81d3(a)dr.vm >> [dialog:failed] Deleting 38e422e10af08b2e34d58e5b765e81d3(a)dr.vm from >> dialogs and dstprefix_limit 0(5890) DEBUG: dialog [dlg_hash.c:753]: unref >> dlg 0x871af3f0 with 1 -> 2 0(5890) DEBUG: dialog [dlg_handlers.c:543]: >> dialog 0x871af3f0 failed (negative reply) >> 0(5890) DEBUG: dialog [dlg_cb.c:256]: dialog=0x871af3f0, type=4 >> 0(5890) DEBUG: dialog [dlg_hash.c:643]: ref dlg 0x871af3f0 with 1 -> 3 >> 0(5890) DEBUG: dialog [dlg_hash.c:646]: dialog >> callid='38e422e10af08b2e34d58e5b765e81d3(a)dr.vm' found on entry 2825, >> dir=1 0(5890) DEBUG: dialog [dlg_var.c:203]: Internal var-list ((nil)): >> 0(5890) DEBUG: dialog [dlg_var.c:213]: Dialog var-list (0x871af6f8): >> 0(5890) DEBUG: dialog [dlg_var.c:218]: "ru"=sip:003900000@cr1.vm:5060 >> (flags 1) >> 0(5890) DEBUG: dialog [dlg_var.c:218]: d=0 (flags 1) >> 0(5890) DEBUG: dialog [dlg_var.c:218]: et=1360265682.889 (flags 1) >> 0(5890) DEBUG: dialog [dlg_var.c:218]: st=1360265682.889 (flags 1) >> 0(5890) DEBUG: dialog [dlg_hash.c:753]: unref dlg 0x871af3f0 with 1 -> 2 >> 0(5890) acc [acc_cdr.c:192]: st=1360265682.889; et=1360265682.889; >> d=0; c1= 0(5890) DEBUG: dialog [dlg_hash.c:753]: unref dlg 0x871af3f0 >> with 1 -> 1 0(5890) DEBUG: dialog [dlg_handlers.c:937]: registering TMCB >> to wait for negative ACK 0(5890) DEBUG: dialog [dlg_hash.c:735]: ref dlg >> 0x871af3f0 with 1 -> 2 >> 0(5890) DEBUG: dialog [dlg_hash.c:753]: unref dlg 0x871af3f0 with 1 -> 1 >> 0(5890) DEBUG: tm [t_hooks.c:288]: DBG: trans=0x871afc98, callback >> type 512, id 0 entered 0(5890) DEBUG: acc [acc_logic.c:537]: acc callback >> called for t(0x871afc98) event type 512, reply code 486 0(5890) acc >> [acc.c:275]: ACC: call missed: >> timestamp=1360265692;method=INVITE;from_tag=as5c1d222e;to_tag=as493d6134 >> ;call_id=38e422e10af08b2e34d58e5b765e81d3(a)dr.vm;code=486;reason=Busy >> here 0(5890) DEBUG: acc [acc.c:872]: missed event for engine: radius >> 0(5890) acc [acc.c:275]: ACC: transaction answered: >> timestamp=1360265692;method=INVITE;from_tag=as5c1d222e;to_tag=as493d6134 >> ;call_id=38e422e10af08b2e34d58e5b765e81d3(a)dr.vm;code=486;reason=Busy >> here 0(5890) DEBUG: acc [acc.c:867]: acc event for engine: radius >> 0(5890) DEBUG: tm [t_hooks.c:288]: DBG: trans=0x871afc98, callback >> type 8388608, id 0 entered >> >> Thanks for a help. >> >> >> Efelin >> >> >> _______________________________________________ >> 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 > http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda > Kamailio World Conference, April 16-17, 2013, Berlin > - http://conference.kamailio.com - > > > > > > ------------------------------ > > > Message: 4 > Date: Fri, 08 Feb 2013 09:36:01 +0100 > From: Daniel-Constantin Mierla <miconda(a)gmail.com> > Subject: Re: [SR-Users] Reg. shared library linking during compiling > time To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List" <sr-users(a)lists.sip-router.org> > Message-ID: <5114B8F1.8020807(a)gmail.com> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > > Hello, > > > some guidelines to write a new module are available at: - > http://www.asipto.com/pub/kamailio-devel-guide/#c16_devel_new_module > > > Cheers, > Daniel > > > On 2/8/13 8:06 AM, kiran bhosale wrote: > >> madhumanjusha(a)integramicro.com wrote: >>> Hello All, >>> Please find below steps...and kindly let me know the error solution. >>> >>> >>> STEP 1:: >>> Under modules I created a directory name b2bentities >>> /root/kamailio-3.3.2/modules/b2bentities >>> $ls >>> b2bentities.so b2b_server.c b2b_server.d b2b_server.h b2b_server.o >>> dialog.h librpath.lst makecfg.lst Makefile >>> >>> In b2b_server.h >>> #ifndef SERVER_H >>> #define SERVER_H >>> #endif >>> >>> >>> int b2b_dlg_reply(); >>> >>> >>> >>> In b2b_server.c I wrote following code :: >>> >>> >>> int b2b_dlg_reply() { >>> LOG(L_INFO,"Hello am in b2bua module\n"); >>> return 0; } >>> >>> >>> Also,I added below code to the same file as per steps given in >>> kamailio server's developer's guide. In developer's guide,they told to >>> initialize parameters as per module's requirement but i made them all >>> empty and null... for example static int mod_init(void) { >>> LOG(L_INFO,"IN mod_init\n"); >>> return 0; } >>> >>> >>> static cmd_export_t commands[] = { }; >>> >>> >>> >>> static param_export_t params[]={ }; >>> >>> >>> struct module_exports exports = { "b2bentities", // module name >>> NULL, // dlopen flags >>> commands, // exported functions params, // exported >>> parameters NULL, // exported statistics >>> NULL, // exported MI functions >>> NULL, // exported pseudo-variables >>> NULL, // extra processes >>> mod_init, // module init function (before fork. kids will >>> inherit) NULL, // reply processing function >>> NULL, // destroy function >>> child_init // child init function }; >>> >>> >>> >>> STEP 2:: >>> In kamailio core >>> file : In receive.c Method Name : In receive_msg >>> >>> >>> i did this way... >>> >>> In receive.c add follow piece of code:: >>> >>> >>> #include "modules/b2bentities/b2b_server.h" >>> >>> >>> int receive_msg(....) { >>> /*my code*/ >>> int b2b_ret; int (*ptr)(); ptr= b2b_dlg_reply; >>> if(msg->REQ_METHOD!=METHOD_REGISTER) { >>> >>> >>> b2b_ret=(*ptr)(); if(b2b_ret!=0) { >>> LOG(L_INFO,"B2B Dialogue reply function >>> failed\n"); exit(1); } >>> else { >>> LOG(L_INFO,"B2B Dialogue reply function >>> success\n"); exit(1); } >>> >>> >>> } >>> >>> >>> >>> } >>> >>> >>> STEP 3:: >>> I even added kamailio.cfg load module "b2bentities.so" >>> >>> >>> >>> >>> STEP 4:: >>> Final step,when I do make all..... >>> its giving me the following error :: [root@kamserver kamailio-3.3.2]# >>> make all CC (gcc) [kamailio] receive.o >>> LD (gcc) [kamailio] kamailio >>> receive.o: In function `receive_msg': >>> /root/kamailio-3.3.2/receive.c:163: undefined reference to >>> `b2b_dlg_reply' >>> collect2: ld returned 1 exit status >>> make: *** [kamailio] Error 1 >>> >>> >>> >>> >>> My doubt: >>> Where should my b2bentities.so should be linked during compiling time? >>> Please tell me the steps for it. >>> >>> >>> Thanks & Regards, >>> Manjusha A. >>> Integra Micro Software Services (P) Ltd. >>> >>> >>> >>> >>> >>> >>> >>> >>> _______________________________________________ >>> 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 >>> >>> >> seems that you want to create your custom module. have you checked the >> makefile of your module. in order to develop the custom module just copy >> the existing module and make relevant changes there. this is genuine way >> . for this u need to include makefile.lst ,Makefile in >> your module. so best way is to copy the smallest module (e.g.) pdb module >> and make changes. >> >> >> _______________________________________________ >> 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 > http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda > Kamailio World Conference, April 16-17, 2013, Berlin > - http://conference.kamailio.com - > > > > > > ------------------------------ > > > _______________________________________________ > sr-users mailing list sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > > End of sr-users Digest, Vol 93, Issue 13 > **************************************** > > Thanks & Regards, Manjusha A. Integra Micro Software Services (P) Ltd.

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Re: [SR-Users] [sr-dev] openser db on an external mysql server
by Prakash N 12 Feb '13

12 Feb '13

Hi Shahzad , It is not ,We can able to connect remote mysql ( please find the detail below),moreover we can able create openser db from Kamailio server to remote mysql server *Mysql server : 192.168.1.170* *Kamailio server : 192.168.1.101* *Mysql output :* * * *mysql -u openser -h 192.168.1.170 -p* Enter password: Welcome to the MySQL monitor. Commands end with ; or \g. Your MySQL connection id is 68 Server version: 5.1.67 Source distribution Copyright (c) 2000, 2012, Oracle and/or its affiliates. All rights reserved. Oracle is a registered trademark of Oracle Corporation and/or its affiliates. Other names may be trademarks of their respective owners. Type 'help;' or '\h' for help. Type '\c' to clear the current input statement. mysql> With Regards N.Prakash > > ---------- Forwarded message ---------- > From: Muhammad Shahzad <shaheryarkh(a)gmail.com> > Date: Tue, Feb 12, 2013 at 5:16 PM > Subject: Re: [sr-dev] [SR-Users] openser db on an external mysql server > To: Development mailing list of the sip-router project < > sr-dev(a)lists.sip-router.org> > Cc: Daniel-Constantin Mierla <miconda(a)gmail.com>, > sr-users(a)lists.sip-router.org, Indu Priya <indu(a)tevatel.com>, Prabhakaran > B <prabhakaran(a)tevatel.com> > > > Its not a Kamailio error, it is problem with connecting to mysql server. > Error 4 means No MySQL Server Running On Specified Host. > > May be MySQL server is down or there is some firewall running or MySQL not > listening on IP you specified. > > Thank you. > > > On Tue, Feb 12, 2013 at 12:11 PM, Prakash N <prakash.n(a)tevatel.com> wrote: > >> Dear Daniel, >> >> Thanks for your quick response >> >> We can able to create operseb db on remote mysql server ,but >> Kamailio not getting start >> >> Error detail: >> >> ERROR: db_mysql [km_my_con.c:109]: driver error: Can't connect to MySQL >> server on '192.168.1.170' (4) >> ERROR: <core> [db.c:32 0]: could not add connection to the pool >> ERROR: auth_db [authdb _mod.c:174]: unable to connect to the database >> ERROR: <core> [sr_modu le.c:896]: init_mod_child(): Error while >> initializing module auth_db (/usr/local/lib64/kamailio/modules_k/auth_db.so) >> >> >> I have attached my kamailo.cfg ( we followed Kamailio and Asterisk real >> time installation) ,please advice >> >> With Regards >> >> N.Prakash >> >>> >>> ---------- Forwarded message ---------- >>> From: Daniel-Constantin Mierla <miconda(a)gmail.com> >>> Date: Tue, Feb 12, 2013 at 2:41 PM >>> Subject: Re: [SR-Users] openser db on an external mysql server >>> To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - >>> Users Mailing List" <sr-users(a)lists.sip-router.org>, >>> sr-dev(a)lists.sip-router.org, miconda(a)gmail.com >>> Cc: Prabhakaran B <prabhakaran(a)tevatel.com> >>> >>> >>> Hello, >>> >>> kamailio database is created with: >>> >>> kamdbctl create >>> >>> the host of mysql server has to be specified in kamctlrc file via DBHOST >>> variable - kamctlrc is located in same directory as kamailio.cfg. >>> >>> In kamailio.cfg, update the db_url value (or DBURL define) to reflect >>> the mysql server host. >>> >>> Cheers, >>> Daniel >>> >>> >>> On 2/12/13 4:16 AM, Prakash N wrote: >>> >>> Hi All, >>> >>> I am new this Kamailo ,ther is any step by step document to set >>> up openser db on an external mysql server ? >>> >>> With Regards >>> >>> N.Prakash >>> >>> >>> >>> _______________________________________________ >>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>> >>> >>> >>> -- >>> Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda >>> Kamailio World Conference, April 16-17, 2013, Berlin >>> - http://conference.kamailio.com - >>> >>> >>> >> >> _______________________________________________ >> sr-dev mailing list >> sr-dev(a)lists.sip-router.org >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-dev >> >> > > > -- > Muhammad Shahzad > ----------------------------------- > CISCO Rich Media Communication Specialist (CRMCS) > CISCO Certified Network Associate (CCNA) > Cell: +49 176 99 83 10 85 > MSN: shari_786pk(a)hotmail.com > Email: shaheryarkh(a)googlemail.com > >

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Re: [SR-Users] openser db on an external mysql server
by Prakash N 12 Feb '13

12 Feb '13

Dear Daniel, Thanks for your quick response We can able to create operseb db on remote mysql server ,but Kamailio not getting start Error detail: ERROR: db_mysql [km_my_con.c:109]: driver error: Can't connect to MySQL server on '192.168.1.170' (4) ERROR: <core> [db.c:32 0]: could not add connection to the pool ERROR: auth_db [authdb _mod.c:174]: unable to connect to the database ERROR: <core> [sr_modu le.c:896]: init_mod_child(): Error while initializing module auth_db (/usr/local/lib64/kamailio/modules_k/auth_db.so) I have attached my kamailo.cfg ( we followed Kamailio and Asterisk real time installation) ,please advice With Regards N.Prakash > > ---------- Forwarded message ---------- > From: Daniel-Constantin Mierla <miconda(a)gmail.com> > Date: Tue, Feb 12, 2013 at 2:41 PM > Subject: Re: [SR-Users] openser db on an external mysql server > To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users > Mailing List" <sr-users(a)lists.sip-router.org>, sr-dev(a)lists.sip-router.org, > miconda(a)gmail.com > Cc: Prabhakaran B <prabhakaran(a)tevatel.com> > > > Hello, > > kamailio database is created with: > > kamdbctl create > > the host of mysql server has to be specified in kamctlrc file via DBHOST > variable - kamctlrc is located in same directory as kamailio.cfg. > > In kamailio.cfg, update the db_url value (or DBURL define) to reflect the > mysql server host. > > Cheers, > Daniel > > > On 2/12/13 4:16 AM, Prakash N wrote: > > Hi All, > > I am new this Kamailo ,ther is any step by step document to set > up openser db on an external mysql server ? > > With Regards > > N.Prakash > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > > -- > Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda > Kamailio World Conference, April 16-17, 2013, Berlin > - http://conference.kamailio.com - > > >

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Fwd: INVITE Retransmission Causes Limiting Simultaneus Calls for Subscribers
by Bayram Karagoz 12 Feb '13

12 Feb '13

---------- Forwarded message ---------- From: Bayram Karagoz <karagoz.bayram(a)gmail.com> Date: 2013/2/12 Subject: Re: [SR-Users] INVITE Retransmission Causes Limiting Simultaneus Calls for Subscribers To: miconda(a)gmail.com Cc: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org> recheck config and saw that I used dlg_manage() most before route(QUOTA) block. does it sits in the wrong place? you can find my kamailio.cfg file in attachment also. # account only INVITEs * dlg_manage();* if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } ###Display Name manipulation### if (is_method("INVITE") && ($fn != $fU)) { uac_replace_from("$fU",""); } ### incoming call from SBC### route(INCOMINGSBC); ###Suspend User### route(SUSPEND); ### Concurrent Call Limitation Routing Block ### * route(QUOTA);* 2013/2/12 Daniel-Constantin Mierla <miconda(a)gmail.com> > Where do you use dlg_manage()? It is not in the route you pasted below. > > Cheers, > Daniel > > > On 2/12/13 10:12 AM, Bayram Karagoz wrote: > > Thanks for quick response Daniel, > > My kamailio version is 3.2.3 (x86_64/linux) . > I want to use dlg_manage() before but not sure which side of my call > limitation route code block. > can you advise me where to use dlg_manage() in my routing? > here is my route block defined below; > > > route[QUOTA] { > > if (is_method("INVITE") && $fU == $avp(s:username)) { > > avp_db_query("select quota from subscriber where > username='$avp(s:username)'", "$avp(s:quota)"); > > $var(SIZE) = 0; > get_profile_size("callquota", "$avp(s:username)", "$var(SIZE)"); > xlog("L_INFO", "CALL_KAMAILIO:Call originated from = > $fd\n"); > xlog("L_INFO", "Active call amount for $fU is $var(SIZE) , > call-id=$ci, quota=$avp(s:quota)\n"); > if( $var(SIZE) >= $avp(s:quota) ){ > sl_send_reply("403", "Simultaneous calls limit reached"); > xlog("L_INFO", "QUOTA:Limit of $var(SIZE) active calls has > been reached for user=$avp(s:username), call-id=$ci, > quota=$avp(s:quota)\n"); > exit; > } > set_dlg_profile("callquota","$avp(s:username)"); > } > } > > > 2013/2/12 Daniel-Constantin Mierla <miconda(a)gmail.com> > >> Hello, >> >> what version of kamailio are you using? >> >> State 1 is for dialogs that were not relayed (or they haven't received >> any reply yet). That means you create the dialog and don't relay the >> INVITE. Try to us dlg_manage() just before relaying the INVITE. >> >> Cheers, >> Daniel >> >> >> On 2/12/13 9:37 AM, Bayram Karagoz wrote: >> >> Hi, >> >> is there any advise about my problem? >> I'm using kamailio in my production environment and this issue started to >> reproduce a lot. >> is there any way to discard retransmitted INVITE messages from counting >> dialog? >> >> >> 2013/2/9 Bayram Karagoz <karagoz.bayram(a)gmail.com> >> >>> thanks for the response Daniel, >>> >>> has_totag not solved my problem. >>> >>> in a deep search I realized that these retransmission INVITE dialogs >>> are waiting in dialog list as state = 1. one example is below; >>> is there any way to clear these dialog which state is waiting for state >>> = 1 or any other solution is welcome for me? >>> >>> dialog:: hash=246:499719317 >>> state:: 1 >>> ref_count:: 1 >>> timestart:: 0 >>> timeout:: 0 >>> callid:: 1356619-3569384775-474862(a)GBSBC-1.mydomain.com >>> from_uri:: sip:02165151057@xxx.xxx.xxx.xxx >>> from_tag:: 3569384775-474868 >>> caller_contact:: sip:02165151057@xxx.xxx.xxx.xxx:5060 >>> caller_cseq:: 2 >>> caller_route_set:: >>> caller_bind_addr:: udp:xxx.xxx.xxx.xxx:5060 >>> callee_bind_addr:: >>> to_uri:: sip:4115667@xxx.xxx.xxx.xxx >>> to_tag:: >>> callee_contact:: >>> callee_cseq:: >>> callee_route_set:: >>> >>> >>> >>> >>> 2013/2/4 Daniel Tryba <daniel(a)pocos.nl> >>> >>>> On Sunday 03 February 2013 12:58:02 Bayram Karagoz wrote: >>>> > I'm trying to limit all subscriber's active call according to setting >>>> their >>>> > simultaneous use setting defined in database. >>>> > But having problem when one subscriber's INVITE message goes to >>>> > retransmission. When retranmission occures INVITE messages don't >>>> complete >>>> > with BYE and subscriber simultaneus use amount will hung. >>>> > My example code block is below. >>>> > Does anyone advice me for overcome INVITE retransmissions? >>>> >>>> I haven't implemented it myself yet, but all examples I've seen check >>>> for >>>> !has_totag() in the INVITE when doing call limit accounting. So >>>> if (is_method("INVITE") && $fU == $avp(s:username) && !has_totag()) >>>> in your case. But that only works if the device received the initial 100 >>>> Trying with a tag from Kamailio. >>>> >>>> But might it not be better to just do the count in the INVITE and >>>> adding/deleting of dialogs in the dialog event routes? >>>> http://kamailio.org/docs/modules/stable/modules_k/dialog.html#id2556156 >>>> Problem here is that if the limit is 5 and and there are more than 5 >>>> concurrent INVITES you'd have to drop calls with a 200 OK response. >>>> >>>> -- >>>> >>>> POCOS B.V. - Croy 9c - 5653 LC Eindhoven >>>> Telefoon: 040 293 8661 - Fax: 040 293 8658 >>>> http://www.pocos.nl/ - http://www.sipo.nl/ >>>> K.v.K. Eindhoven 17097024 >>>> >>> >>> >>> >>> -- >>> *Bayram KARAGÖZ* >>> >>> *NGN Engineer* >>> >>> *-----------------------------------------* >>> >>> *@* bayram(a)bayramkaragoz.org >>> >>> *:** *www.bayramkaragoz.org >>> >>> *-----------------------------------------* >>> >> >> >> >> -- >> *Bayram KARAGÖZ* >> >> *NGN Engineer* >> >> *-----------------------------------------* >> >> *@* bayram(a)bayramkaragoz.org >> >> *:** *www.bayramkaragoz.org >> >> *-----------------------------------------* >> >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> >> >> -- >> Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda >> Kamailio World Conference, April 16-17, 2013, Berlin >> - http://conference.kamailio.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 >> >> > > > -- > *Bayram KARAGÖZ* > > *NGN Engineer* > > *-----------------------------------------* > > *@* bayram(a)bayramkaragoz.org > > *:** *www.bayramkaragoz.org > > *-----------------------------------------* > > > -- > Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda > Kamailio World Conference, April 16-17, 2013, Berlin > - http://conference.kamailio.com - > > -- *Bayram KARAGÖZ* *NGN Engineer* *-----------------------------------------* *@* bayram(a)bayramkaragoz.org *:** *www.bayramkaragoz.org *-----------------------------------------* -- *Bayram KARAGÖZ* *NGN Engineer* *-----------------------------------------* *@* bayram(a)bayramkaragoz.org *:** *www.bayramkaragoz.org *-----------------------------------------*

2 2

Kamailio World Updates
by Daniel-Constantin Mierla 12 Feb '13

12 Feb '13

Hello, we have published the first set of accepted speeches at Kamailio World Conference, you can view the current draft of the schedule at: - http://conference.kamailio.com/k01/schedule/ There are already about 15 speakers and presentations in the agenda for the two days of conference, other proposals being in the review queue. Another good news is that business environment around Kamailio project is growing, the sponsors of Kamailio World Conference, Sipwise, Sipgate and Asipto are looking to hire or subcontract new talents, check the Career & Jobs page at: - http://conference.kamailio.com/k01/career-jobs/ Greetings again to our community members that participated at Fosdem, we were quite a big group there, unfortunately not much time to catch up in detail with everyone. Looking forward to see many of you in Berlin as well, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

1 0

Handling 302 redirect on group ringing
by Mark Boyce 12 Feb '13

12 Feb '13

Hi All Got a problem handling redirects when more than one phone is ringing. The setup is this; Phone1 - Extension 100 Phone 2 - Extension 101 Phone 3 - Extension 102 & 101 When Phone 1 rings Extension 101 Phones 2 & 3 ring correctly as you would expect. If Phone 2 now sets the on phone forward/redirect we see the following; Phone 1 - Invite sent to 101 Phone 2 & 3 Receive Invites Phone 2 responds with a 302 Redirect Phone 3 Rings. Which again is as you would expect. The problem is that the 302 Message we're getting back is being seen in onreply_route where I cannot add a new branch to handle it. (Error - misused command append_branch / Command cannot be used in the block) If Phone 3 is unavailable / times out / busy / or otherwise does not take the call we then see the 302 from Phone 2 arrive in failure_route where it is handled as expected. The call flow we're trying to achieve is that the redirect is followed whilst the other available phone carries on ringing. I've tried with failure_reply_mode as the default 0 and as 3, both do the same thing. Currently working with kamailio version 3.4.0-dev6 Can anyone spot what I'm doing wrong ? Thanks Mark

2 1

How to detect there are no matching subscriber for uri
by Krishna Kurapati 12 Feb '13

12 Feb '13

Hi, When we receive an request, we do " lookup("location") call to find if the uri is there or not. There are two scenarios: 1. The aor is in the subscriber table but not in usrloc table 2. The aor is not in subscriber table I would like to respond differently to the sender in each of these cases. Right now, the lookup returns -1 for either case. Is there any other way other than dipping into database each time? Thanks Krish Kura

2 1

INVITE Retransmission Causes Limiting Simultaneus Calls for Subscribers
by Bayram Karagoz 12 Feb '13

12 Feb '13

Hello, I'm trying to limit all subscriber's active call according to setting their simultaneous use setting defined in database. But having problem when one subscriber's INVITE message goes to retransmission. When retranmission occures INVITE messages don't complete with BYE and subscriber simultaneus use amount will hung. My example code block is below. Does anyone advice me for overcome INVITE retransmissions? Thanks. modparam("dialog", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("dialog", "dlg_flag", 4) modparam("dialog", "db_mode", 1) modparam("dialog","profiles_with_value","callquota") modparam("dialog", "default_timeout", 3600) modparam("dialog", "dlg_match_mode",0) modparam("dialog", "detect_spirals", 1) route[QUOTA] { if (is_method("INVITE") && $fU == $avp(s:username)) { avp_db_query("select quota from subscriber where username='$avp(s:username)'", "$avp(s:quota)"); $var(SIZE) = 0; get_profile_size("callquota", "$avp(s:username)", "$var(SIZE)"); xlog("L_INFO", "CALL_OPENSER:Call originated from = $fd\n"); xlog("L_INFO", "Active call amount for $fU is $var(SIZE) , quota=$avp(s:quota)\n"); if( $var(SIZE) >= $avp(s:quota) ){ sl_send_reply("403", "Simultaneous calls limit reached"); xlog("L_INFO", "QUOTA:Limit of $var(SIZE) active calls has been reached for user=$avp(s:username), quota=$avp(s:quota)\n"); exit; } set_dlg_profile("callquota","$avp(s:username)"); get_profile_size("callquota", "$avp(s:username)", "$var(SIZE)"); } } -- *Bayram KARAGÖZ*

3 6

Video Conferencing
by Mohakem Khan 12 Feb '13

12 Feb '13

Does kamailio support video endpoint/codecs? How about video conference? Is there MCD support?

3 2

openser db on an external mysql server
by Prakash N 12 Feb '13

12 Feb '13

Hi All, I am new this Kamailo ,ther is any step by step document to set up openser db on an external mysql server ? With Regards N.Prakash

2 1

mysql: driver_error
by Uri Shacked 11 Feb '13

11 Feb '13

Hi, I am doing some testing and i have trouble understanding when does the "mysql:driver_error" counter/statistics increase.... I stopped the mysqld process on the DB server and used sqlops. the driver_error did still is on 0. I use version 3.3.2. Thanks, Uri

2 3

Re: [SR-Users] dlg_manage and engage_media_proxy
by Daniel-Constantin Mierla 11 Feb '13

11 Feb '13

Hello, On 2/11/13 5:17 PM, Eduardo Lejarreta wrote: > Good evening Daniel. > > You are right, calling engage_media_proxy() prior to dlg_manage() solves the > issue (just before t_relay on INVITEs). > > I've done several test changing the order of things and now everything works > fine. great, thanks for reporting back. Perhaps dlg flag vs dlg_manage has to be clarified a bit more in the next major release (4.1). Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

1 0

Kamailio multiple registration issue
by Ramaseshi reddy kolli 11 Feb '13

11 Feb '13

Hi, i am using kamailio 3.3 version please help me with this issue. 1) Registered from SIP end point, kamailio added row in location table 2) Now using same SIP account details, pushed registration entry using *kamctl ul add , * instead of overwriting the previous contact by new one it inserts another new row in the location table 3) Total now 2 rows with same username are present in DB. I am using save("location", "0x04") so before inserting new row it should check whether already entry is there or not, if there should update the same row. I look forward to your reply. Thanks Ram

2 2

Re: [SR-Users] dlg_manage and engage_media_proxy
by Daniel-Constantin Mierla 11 Feb '13

11 Feb '13

On 2/8/13 12:15 PM, Eduardo Lejarreta wrote: > Yes it's the correct order. We call dlg_manage() prior to > engage_media_proxy(): The correct order could be the other one, as mediaproxy might need some callbacks from dialog module. Usually the dlg_manage() should be executed just before relaying, when is no other chance that the invite will get a negative reply from the config. Cheers, Daniel > > route[INVITE_MANAGE] { > > if (is_method("INVITE")) { > > $dlg_var(si) = $si; > $dlg_var(tU) = $tU; > $dlg_ctx(timeout_bye) = 1; > > dlg_manage(); > ........ > ........ > } > } > ....... > ....... > route[RELAY] { > > if (is_method("INVITE")) { > ........ > ........ > route(MEDIAPROXY); # route MEDIAPROXY > t_on_failure(FAIL_ONE); > t_on_reply(REPLY_ONE); > } > > if (!t_relay()) { > sl_reply_error(); > } > exit; > } > ........ > ........ > route[MEDIAPROXY] { > > xlog("L_INFO", "Engage Media Proxy - M=$rm R=$ru F=$fu T=$tu > IP=$si:$sp ID=$ci\n"); > engage_media_proxy(); > return; > } > -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

2 1

UC Expo London
by Peter Dunkley 11 Feb '13

11 Feb '13

Hello, Are there any plans for a Kamailio meet-up on one of the evenings this year? Regards, Peter -- Peter Dunkley Technical Director Crocodile RCS Ltd

1 0

Kamailio-Java Integration
by ron.kamailio＠mcleodnet.com 10 Feb '13

10 Feb '13

I am looking for a way to integrate Java business logic with Kamailio to influence routing decisions, add/remove/modify headers, record accounting-related information, etc. A while back there was a project named WeSIP, but it doesn't appear to active anymore. Is anyone doing something like this now, or any ideas or suggestions? Thanks, Ron

8 23

Multiple radius responses on a missed call
by Efelin Novak 08 Feb '13

08 Feb '13

Hi folks, I'd like to ask you how can I solve an issue when I'm receiving three RADIUS requests on one missed call. My scenario is as follows: A calls to B. B returns 503 error message, INIVITE is then sent to C and C sends 486. After first 503 I receive a first RADIUS request. This is correct. However after 486 I receive three RADIUS requests with different IDs. As you can see in a DEBUG one RADIUS is send after the SIP reply message is received and second and third RADIUS messages are sent after the SIP reply message is forwarded. I'd like to receive only two RADIUS messages. One after 503 and second after 486. FLT_ACC is defined as 1. My config is as follows: modparam("acc", "failed_transaction_flag", FLT_ACC) modparam("acc", "report_cancels", 1) modparam("acc", "report_ack", 0) modparam("acc", "detect_direction", 0) modparam("acc", "early_media", 0) modparam("acc", "log_level", 5) modparam("acc", "log_flag", 1) modparam("acc", "log_missed_flag", 1) modparam("acc_radius", "radius_config", "/etc/radiusclient-ng/radiusclient.conf") # This is the location of the configuration file of radius client modparam("acc_radius", "radius_flag", FLT_ACC) modparam("acc_radius", "radius_missed_flag", FLT_ACC) In a route I set setflag("FLT_ACC") only once per call. Here is the debug for debug level 7: 0(5890) DEBUG: <core> [parser/msg_parser.c:634]: SIP Reply (status): 0(5890) DEBUG: <core> [parser/msg_parser.c:636]: version: <SIP/2.0> 0(5890) DEBUG: <core> [parser/msg_parser.c:638]: status: <486> 0(5890) DEBUG: <core> [parser/msg_parser.c:640]: reason: <Busy here> 0(5890) DEBUG: <core> [parser/parse_via.c:1286]: Found param type 232, <branch> = <z9hG4bK8d9c.4438d232.1>; state=6 0(5890) DEBUG: <core> [parser/parse_via.c:1286]: Found param type 234, <received> = <192.168.21.101>; state=16 0(5890) DEBUG: <core> [parser/parse_via.c:2561]: end of header reached, state=5 0(5890) DEBUG: <core> [parser/msg_parser.c:511]: parse_headers: Via found, flags=2 0(5890) DEBUG: <core> [parser/msg_parser.c:513]: parse_headers: this is the first via 0(5890) DEBUG: <core> [receive.c:149]: After parse_msg... 0(5890) DEBUG: tm [t_lookup.c:1079]: DEBUG: t_check_msg: msg id=6 global id=5 T start=0xffffffff 0(5890) DEBUG: <core> [parser/parse_via.c:1286]: Found param type 232, <branch> = <z9hG4bK8d9c.e7b650e1.0>; state=16 0(5890) DEBUG: <core> [parser/parse_via.c:2561]: end of header reached, state=5 0(5890) DEBUG: <core> [parser/msg_parser.c:511]: parse_headers: Via found, flags=22 0(5890) DEBUG: <core> [parser/msg_parser.c:524]: parse_headers: this is the second via 0(5890) DEBUG: <core> [parser/parse_via.c:1286]: Found param type 232, <branch> = <z9hG4bK463ae086>; state=6 0(5890) DEBUG: <core> [parser/parse_via.c:1286]: Found param type 235, <rport> = <5060>; state=16 0(5890) DEBUG: <core> [parser/parse_via.c:2561]: end of header reached, state=5 0(5890) DEBUG: <core> [parser/msg_parser.c:511]: parse_headers: Via found, flags=22 0(5890) DEBUG: <core> [parser/parse_to.c:178]: DEBUG: add_param: tag=as493d6134 0(5890) DEBUG: <core> [parser/parse_to.c:802]: end of header reached, state=29 0(5890) DEBUG: <core> [parser/msg_parser.c:188]: DEBUG: get_hdr_field: <To> [42]; uri=[sip:5556003900000@dr.vm] 0(5890) DEBUG: <core> [parser/msg_parser.c:190]: DEBUG: to body [<sip:5556003900000@dr.vm>] 0(5890) DEBUG: <core> [parser/msg_parser.c:168]: get_hdr_field: cseq <CSeq>: <102> <INVITE> 0(5890) DEBUG: tm [t_lookup.c:965]: DEBUG: t_reply_matching: hash 51672 label 590185284 branch 1 0(5890) DEBUG: tm [t_lookup.c:1016]: DEBUG: t_reply_matching: reply matched (T=0x871afc98)! 0(5890) DEBUG: tm [t_hooks.c:288]: DBG: trans=0x871afc98, callback type 2, id 0 entered 0(5890) DEBUG: acc [acc_logic.c:537]: acc callback called for t(0x871afc98) event type 2, reply code 486 0(5890) DEBUG: tm [t_hooks.c:288]: DBG: trans=0x871afc98, callback type 2, id 0 entered 0(5890) DEBUG: <core> [parser/parse_to.c:178]: DEBUG: add_param: tag=as5c1d222e 0(5890) DEBUG: <core> [parser/parse_to.c:802]: end of header reached, state=29 0(5890) DEBUG: siptrace [siptrace.c:794]: storing info... 0(5890) DEBUG: tm [t_lookup.c:1148]: DEBUG: t_check_msg: msg id=6 global id=6 T end=0x871afc98 0(5890) DEBUG: tm [t_reply.c:2081]: DEBUG: reply_received: org. status uas=180, uac[1]=180 local=0 is_invite=1) 0(5890) DEBUG: tm [t_hooks.c:288]: DBG: trans=0x871afc98, callback type 4194304, id 0 entered 0(5890) DEBUG: siptrace [siptrace.c:794]: storing info... 0(5890) DEBUG: tm [t_lookup.c:1079]: DEBUG: t_check_msg: msg id=6 global id=6 T start=0x871afc98 0(5890) DEBUG: tm [t_lookup.c:1151]: DEBUG: t_check_msg: T already found! 0(5890) DEBUG: tmx [t_var.c:438]: reply code is <486> 0(5890) INFO: <script>: XLOG: 38e422e10af08b2e34d58e5b765e81d3(a)dr.vm [REPLY_TWO] Incoming reply code: 486 0(5890) DEBUG: tm [t_reply.c:1204]: ->>>>>>>>> T_code=180, new_code=486 0(5890) DEBUG: tm [t_hooks.c:288]: DBG: trans=0x871afc98, callback type 128, id 0 entered 0(5890) DEBUG: acc [acc_logic.c:537]: acc callback called for t(0x871afc98) event type 128, reply code 486 0(5890) acc [acc.c:275]: ACC: call missed: timestamp=1360265692;method=INVITE;from_tag=as5c1d222e;to_tag=as493d6134;call_id=38e422e10af08b2e34d58e5b765e81d3(a)dr.vm;code=486;reason=Busy here 0(5890) DEBUG: acc [acc.c:872]: missed event for engine: radius 0(5890) DEBUG: <core> [md5utils.c:67]: DEBUG: MD5 calculated: 6e1b9f35a8b3e65859e29aefc1c26fc3 0(5890) DEBUG: exec [exec_hf.c:267]: SIP_HF_CONTENT_LENGTH=262 0(5890) DEBUG: exec [exec_hf.c:267]: SIP_HF_CONTENT_TYPE=application/sdp 0(5890) DEBUG: exec [exec_hf.c:267]: SIP_HF_SUPPORTED=replaces 0(5890) DEBUG: exec [exec_hf.c:267]: SIP_HF_ALLOW=INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO 0(5890) DEBUG: exec [exec_hf.c:267]: SIP_HF_DATE=Thu, 07 Feb 2013 19:35:53 GMT 0(5890) DEBUG: exec [exec_hf.c:267]: SIP_HF_MAX_FORWARDS=68 0(5890) DEBUG: exec [exec_hf.c:267]: SIP_HF_USER_AGENT=Asterisk PBX 0(5890) DEBUG: exec [exec_hf.c:267]: SIP_HF_CSEQ=102 INVITE 0(5890) DEBUG: exec [exec_hf.c:267]: SIP_HF_CALLID=38e422e10af08b2e34d58e5b765e81d3(a)dr.vm 0(5890) DEBUG: exec [exec_hf.c:267]: SIP_HF_CONTACT=<sip:0221028400@192.168.21.106> 0(5890) DEBUG: exec [exec_hf.c:267]: SIP_HF_TO=<sip:5556003900000@dr.vm> 0(5890) DEBUG: exec [exec_hf.c:267]: SIP_HF_FROM="Janko" <sip:0221028400@dr.vm>;tag=as5c1d222e 0(5890) DEBUG: exec [exec_hf.c:267]: SIP_HF_VIA=SIP/2.0/UDP 192.168.21.100;branch=z9hG4bK8d9c.e7b650e1.0,SIP/2.0/UDP 192.168.21.106:5060;branch=z9hG4bK463ae086;rport=5060 0(5890) DEBUG: exec [exec_mod.c:175]: executing [/opt/kamrouter/scripts/notify_reroute.sh 503 sip:3333003900000@ast-car.vm sip:0221028400@dr.vm 38e422e10af08b2e34d58e5b765e81d3(a)dr.vm] sh: /opt/kamrouter/scripts/notify_reroute.sh: not found 0(5890) INFO: <core> [main.c:797]: INFO: dont_fork turned on, living on 0(5890) DEBUG: tm [t_lookup.c:1079]: DEBUG: t_check_msg: msg id=0 global id=0 T start=0x871afc98 0(5890) DEBUG: tm [t_lookup.c:1151]: DEBUG: t_check_msg: T already found! 0(5890) DEBUG: tmx [t_var.c:438]: reply code is <486> 0(5890) INFO: <script>: XLOG: 38e422e10af08b2e34d58e5b765e81d3(a)dr.vm [FAIL_TWO] Rerouted gw returned 486 0(5890) INFO: <script>: XLOG: 38e422e10af08b2e34d58e5b765e81d3(a)dr.vm [aaa] VYMAZAVAM3 0(5890) DEBUG: tm [t_reply.c:1702]: DEBUG: relay_reply: branch=1, save=0, relay=1 0(5890) DEBUG: <core> [msg_translator.c:1933]: old size: 572, new size: 485 0(5890) DEBUG: <core> [msg_translator.c:1951]: build_res_from_sip_res: copied size: orig:110, new: 23, rest: 462 msg= SIP/2.0 486 Busy here Via: SIP/2.0/UDP 192.168.21.100;branch=z9hG4bK8d9c.e7b650e1.0 Via: SIP/2.0/UDP 192.168.21.106:5060;branch=z9hG4bK463ae086;rport=5060 From: "Janko" <sip:0221028400@dr.vm>;tag=as5c1d222e To: <sip:5556003900000@dr.vm>;tag=as493d6134 Call-ID: 38e422e10af08b2e34d58e5b765e81d3(a)dr.vm CSeq: 102 INVITE Server: Asterisk PBX 11.0.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 0(5890) DEBUG: tm [t_hooks.c:288]: DBG: trans=0x871afc98, callback type 1048576, id 0 entered 0(5890) DEBUG: dialog [dlg_hash.c:588]: ref dlg 0x871af3f0 with 1 -> 2 0(5890) DEBUG: dialog [dlg_hash.c:590]: dialog id=3894 found on entry 2825 0(5890) DEBUG: dialog [dlg_hash.c:940]: dialog 0x871af3f0 changed from state 2 to state 5, due event 4 (ref 2) 0(5890) DEBUG: dialog [dlg_hash.c:735]: ref dlg 0x871af3f0 with 1 -> 3 0(5890) DEBUG: dialog [dlg_handlers.c:1469]: executing event_route 4 on state 5 0(5890) INFO: <script>: XLOG: 38e422e10af08b2e34d58e5b765e81d3(a)dr.vm [dialog:failed] Deleting 38e422e10af08b2e34d58e5b765e81d3(a)dr.vm from dialogs and dstprefix_limit 0(5890) DEBUG: dialog [dlg_hash.c:753]: unref dlg 0x871af3f0 with 1 -> 2 0(5890) DEBUG: dialog [dlg_handlers.c:543]: dialog 0x871af3f0 failed (negative reply) 0(5890) DEBUG: dialog [dlg_cb.c:256]: dialog=0x871af3f0, type=4 0(5890) DEBUG: dialog [dlg_hash.c:643]: ref dlg 0x871af3f0 with 1 -> 3 0(5890) DEBUG: dialog [dlg_hash.c:646]: dialog callid='38e422e10af08b2e34d58e5b765e81d3(a)dr.vm' found on entry 2825, dir=1 0(5890) DEBUG: dialog [dlg_var.c:203]: Internal var-list ((nil)): 0(5890) DEBUG: dialog [dlg_var.c:213]: Dialog var-list (0x871af6f8): 0(5890) DEBUG: dialog [dlg_var.c:218]: "ru"=sip:003900000@cr1.vm:5060 (flags 1) 0(5890) DEBUG: dialog [dlg_var.c:218]: d=0 (flags 1) 0(5890) DEBUG: dialog [dlg_var.c:218]: et=1360265682.889 (flags 1) 0(5890) DEBUG: dialog [dlg_var.c:218]: st=1360265682.889 (flags 1) 0(5890) DEBUG: dialog [dlg_hash.c:753]: unref dlg 0x871af3f0 with 1 -> 2 0(5890) acc [acc_cdr.c:192]: st=1360265682.889; et=1360265682.889; d=0; c1= 0(5890) DEBUG: dialog [dlg_hash.c:753]: unref dlg 0x871af3f0 with 1 -> 1 0(5890) DEBUG: dialog [dlg_handlers.c:937]: registering TMCB to wait for negative ACK 0(5890) DEBUG: dialog [dlg_hash.c:735]: ref dlg 0x871af3f0 with 1 -> 2 0(5890) DEBUG: dialog [dlg_hash.c:753]: unref dlg 0x871af3f0 with 1 -> 1 0(5890) DEBUG: tm [t_hooks.c:288]: DBG: trans=0x871afc98, callback type 512, id 0 entered 0(5890) DEBUG: acc [acc_logic.c:537]: acc callback called for t(0x871afc98) event type 512, reply code 486 0(5890) acc [acc.c:275]: ACC: call missed: timestamp=1360265692;method=INVITE;from_tag=as5c1d222e;to_tag=as493d6134;call_id=38e422e10af08b2e34d58e5b765e81d3(a)dr.vm;code=486;reason=Busy here 0(5890) DEBUG: acc [acc.c:872]: missed event for engine: radius 0(5890) acc [acc.c:275]: ACC: transaction answered: timestamp=1360265692;method=INVITE;from_tag=as5c1d222e;to_tag=as493d6134;call_id=38e422e10af08b2e34d58e5b765e81d3(a)dr.vm;code=486;reason=Busy here 0(5890) DEBUG: acc [acc.c:867]: acc event for engine: radius 0(5890) DEBUG: tm [t_hooks.c:288]: DBG: trans=0x871afc98, callback type 8388608, id 0 entered Thanks for a help. Efelin

2 4

Reg. functions in custom modules
by madhumanjusha＠integramicro.com 08 Feb '13

08 Feb '13

Hello All, Is it possible to call functions in custom modules say modules/my_module/my_file.c if my_file.c has a function and want to access/call it from kamailio core file ie., which is under /root/kamailio-3.3.2/receive.c ?? If not possible,what is the way to access functions in custom modules. Please do reply. Thanks & Regards, Manjusha A. Integra Micro Software Services (P) Ltd.

2 2

dlg_manage and engage_media_proxy
by Eduardo Lejarreta Langarica SARENET 08 Feb '13

08 Feb '13

Good evening. Kamailio 3.2.2 and mediaproxy 2.5.2 We find a problem when using engage_media_proxy and dlg_manage on the same kamailio.cfg (we loose one way audio) is ther any issue with regard to this? may be anything we're doing wrong? any other easy way to control concurrent calls? If 1.- we avoid dlg_manage with engage_media_proxy or 2.- we use dlg_manage with "use_media_proxy/end_media_session" every thing works fine. Any idea will be helpful. Thanks and best regards. ----------------------------------------------------------------------------------------------------------------------------------- Eduardo Lejarreta

3 2

kamctl restart and mediaproxy
by Eduardo Lejarreta Langarica SARENET 08 Feb '13

08 Feb '13

Good evening again. Kamailio 3.2.2 and mediaproxy 2.5.2 An long-time issue for us: kamctl restart stops mediaproxy audio. is there anyway to avoid this? Any idea will be helpful. Thanks and best regards. ----------------------------------------------------------------------------------------------------------------------------------- Eduardo Lejarreta

3 2

Somewhat OT: Specific SIP packets can cause ethernet controller reset
by Kristian Kielhofner 08 Feb '13

08 Feb '13

While not strictly Kamailio related this issue could certainly affect some of you: http://blog.krisk.org/2013/02/packets-of-death.html I'm hearing initial reports that this issue is fairly widespread. -- Kristian Kielhofner

2 1

Re: [SR-Users] sr-users Digest, Vol 92, Issue 50:Modify IP address in SDP (Austin Einter)
by madhumanjusha＠integramicro.com 08 Feb '13

08 Feb '13

On Tue, January 22, 2013 1:58 pm, sr-users-request(a)lists.sip-router.org wrote: > Send sr-users mailing list submissions to > sr-users(a)lists.sip-router.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > or, via email, send a message with subject or body 'help' to > sr-users-request(a)lists.sip-router.org > > You can reach the person managing the list at > sr-users-owner(a)lists.sip-router.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of sr-users digest..." > > > Today's Topics: > > > 1. Modify IP address in SDP (Austin Einter) > 2. Re: Modify IP address in SDP (Olle E. Johansson) > 3. Problem with 4.0 nightlies and IMS - sem_post undefined > (Barry Flanagan) > 4. Patch to resolve segfault in json module (Richard Brady) > 5. Re: Modify IP address in SDP (Austin Einter) > 6. Re: Patch to resolve segfault in json module > (Daniel-Constantin Mierla) > 7. Re: [sr-dev] FOSDEM (Daniel Pocock) > > > > ---------------------------------------------------------------------- > > > Message: 1 > Date: Tue, 22 Jan 2013 18:37:13 +0530 > From: Austin Einter <austin.einter(a)gmail.com> > Subject: [SR-Users] Modify IP address in SDP > To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List" <sr-users(a)lists.sip-router.org> > Message-ID: > <CANXt1k--VtC6aYHGTBZ5GqF8n=66g5PVwG4YhOC_uefxpVYy=g(a)mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > > Dear All > When 200 OK to INVITE passes through Kamailio proxy , I want to change the > IP address in SDP to a specific IP address. > > > Can this be done by just modifying configuration files or do I need to > modify in source code. > > In case I need to modify source code, which files I need to look at. > > > Kindly provide pointers for said problem. > > > Thanks > Austin Hello Austin, 1...I was doing similar one for my task. Files : /root/kamailio-3.3.2/parser/sdp/sdp.h sdp.h file has struct that holds ip address and port numbers of audio and video. Stucture: sdp_session_cell struct has member named str ip_addr will give you ip address in sdp. 2...Yes you can edit existing code or else you can create your own 200ok packet.I created my own 200OK and invite packets with sdp. 3...You can grab below ip address when kamailio proxy is about to forward 200OK to end client.File where you can find 200OK is (though not sure but you can try this file edit existing 200OK): /root/kamailio-3.3.2/modules/tm/t_reply.c has below function static int _reply( struct cell *trans, struct sip_msg* p_msg, unsigned int code, char * text, int lock ) { } 4...I did the following way to get ipaddr...and below piece of code you can use it in the above function's body to edit 200OK's ip address:: char my_src_rtp_addr[8]; int my_src_rtp_addr_len=0; struct sdp_session_cell *sdp_session; if(0 != parse_sdp(p_msg)) { LOG(L_INFO,"Unable to parse sdp\n"); } sdp_session=get_sdp_session(p_msg,0); if(sdp_session!=0){ my_src_rtp_addr_len=strlen(my_src_rtp_addr); my_src_rtp_addr[my_src_rtp_addr_len]='\0'; my_src_rtp_addr[my_src_rtp_addr_len+1]='\0'; my_src_rtp_addr[my_src_rtp_addr_len+2]='\0'; my_src_rtp_addr[my_src_rtp_addr_len+3]='\0'; strncpy(my_src_rtp_addr,sdp_session->ip_addr.s,sdp_session->ip_addr.len); } If any questions,please let me know. Thanks & Regards, Manjusha A. Integra Micro Software Services (P) Ltd.

3 3

Need help in rewriting the Request URI
by Ajay Sabat 07 Feb '13

07 Feb '13

Hi, I use Kamailio for basic call setup with one exception where it has to route call to another server by seeing the domain name in the Request URI. If the Request uri in the received INVITE message looks like "sip12345@video-conf", then it should route the INVITE request to the video-conf server by resolving the domain name. I changed the config file kamailio.cfg as given below. route[SIPOUT] { if (!uri==myself) { append_hf("P-hint: outbound\r\n"); # AKS: call routing if (uri=~".*@video-conf") { rewritehost("video-conf"); } # AKS route(RELAY); } } It resolves the domain name and routes the INVITE message to the resolved IP address. However it uses the domain name in the Request line and To header and I need to replace domain with the resolved IP address in the Request URI and To header. Now Request URI and To header look like "INVITE sip:12345@video-conf" and "To: <sip:12345:video-conf>" respectively and I need them as "INVITE sip:12345@video-conf" and "To: <sip:12345:video-conf>". I don't have any knowledge in tweaking kamailio. I tried some tricks like the following by going through documents and pervious emails in the forum. But it didn't work. Some of them have spoken about avp and pseudo variables. But I am getting confused as I have just begun using Kamailio and it is vast. # $ru = "sip:" + $rU + "@" + to_ip; I will appreciate if you could help me or guide me in right direction. Thanks, Ajay

2 2

Problems with MSILO module can't receive messages with m_dump.
by Nord7 07 Feb '13

07 Feb '13

Hello i have two problems with module MSILO and text messages from SIP client BRIA. 1. I can't send messages with m_dump. I have installed Kamailio 3.3 in default configuration. I'm added "m_dump("$fu")" to my config in route[REGISTRAR]. If i use client CSipSimple for sending messages to offline user "333". i'm receive notification message "*** User 333 offline!". Next i connect to Kamailio with user "333" and i see in kamailio log what "m_dumt" trying to send messages but can't because "request authorisation". How to disable authorisation request, without leak security. route[REGISTRAR] { if (is_method("REGISTER")) { if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging ## setbflag(FLB_NATSIPPING); } if (!save("location")) sl_reply_error(); if (m_dump("$fu")) { log("MSILO: offline messages dumped - if they were\n"); } else { log("MSILO: no offline messages dumped\n"); } exit; } }

3 2

msilo patch
by Stoyan Mihaylov 07 Feb '13

07 Feb '13

When I used, last versions from git, I got error and crash of kamailo using m_dump. I got latest version from git, and I made one patch: diff --git a/modules/msilo/msilo.c b/modules/msilo/msilo.c index 209b309..4bda8fa 100644 --- a/modules/msilo/msilo.c +++ b/modules/msilo/msilo.c @@ -1125,8 +1125,16 @@ static int m_dump(struct sip_msg* msg, str* owner_s) memcpy(tmp_extra_hdrs.s, extra_hdrs_str.s, extra_hdrs_str.len); memcpy(tmp_extra_hdrs.s+extra_hdrs_str.len, str_vals[4].s, str_vals[4].len); } else { - tmp_extra_hdrs.len = 0; - tmp_extra_hdrs.s = ""; + tmp_extra_hdrs.len = 1; + if ((tmp_extra_hdrs.s = pkg_malloc(1)) == NULL) + { + LM_ERR("Out of pkg memory"); + if (msilo_dbf.free_result(db_con, db_res) < 0) + LM_ERR("failed to free the query result\n"); + msg_list_set_flag(ml, mid, MS_MSG_ERRO); + goto error; + } + tmp_extra_hdrs.s[0]='\x0'; } hdr_str.len = 1024; if(m_build_headers(&hdr_str, str_vals[3] /*ctype*/, The problem is: tmp_extra_hdrs.s = ""; and consequent pkg_free(tmp_extra_hdrs.s); As I see, there are couple of ways to solve problem, one of them is setting tmp_extra_hdrs.s = 0; but then we have warnings, or setting tmp_extra_hdrs.s = 0; and check before pkg_free, but pkg_free is used on couple of places. I am not sure what is the best way to report bugs and offer patches.

2 1

Deleting the expired contacts using the Rtimer module
by kiran bhosale 06 Feb '13

06 Feb '13

Hi we have developed the custom module which stores the registered users in a file.now we are trying to remove the expired contacts using the rtimer module.while saving the registered users to the file we also store the expires values in static array.but when we try to decrement the these values in a function called with the help of rtimer module. the values used by this periodic function are not modified ones but the initial which are zero. is it that we cant pass the modified values of the global variables to the timed functions. to get around this problem, we also registered the timer in our module but got same results !!!!!!! Please Help. Thanks and regards .

2 1

401 message forwarding
by Алексей Ёлохов 06 Feb '13

06 Feb '13

Hi! I am setting up Kamailio as SBC with upper registration. I can forward REGISTER messages to my softswitch, but 401 UNAUTHORIZED from it is not coming to my UAC. Config section for registration route: route[REGISTER_MY] { rewritehost("3.3.3.3"); t_relay(); } What I see with ngrep (1.1.1.1 - my external UAC ip, 2.2.2.2 - Kamailio ip, 3.3.3.3 - mera softswitch ip): U 1.1.1.1:5060 -> 2.2.2.2:5060 REGISTER sip:2.2.2.2 SIP/2.0. Via: SIP/2.0/UDP 10.2.1.35:5060 ;rport;branch=z9hG4bKPj85d21f7f-5c1d-4b0b-8e82-940c5cc9e21d. Max-Forwards: 70. From: <sip:101@2.2.2.2>;tag=b00a1df7-94bd-4b2e-8f63-15fea1a9015b. To: <sip:101@2.2.2.2>. Call-ID: e789b66c-d96c-45ca-b3b8-06bf65a2c3d1. CSeq: 59629 REGISTER. User-Agent: SFLphone. Contact: <sip:101@10.2.1.35:5060>. Expires: 600. Allow: PRACK, SUBSCRIBE, NOTIFY, REFER, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, REGISTER, OPTIONS, MESSAGE. Content-Length: 0. . U 2.2.2.2:5060 -> 1.1.1.1:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 10.2.1.35:5060 ;rport=5060;branch=z9hG4bKPj85d21f7f-5c1d-4b0b-8e82-940c5cc9e21d;received=1.1.1.1. From: <sip:101@2.2.2.2>;tag=b00a1df7-94bd-4b2e-8f63-15fea1a9015b. To: <sip:101@2.2.2.2>;tag=2fb3028f366408eaa444e96dfd845447.8865. Call-ID: e789b66c-d96c-45ca-b3b8-06bf65a2c3d1. CSeq: 59629 REGISTER. Contact: <sip:101@10.2.1.35:5060>;expires=600. Server: kamailio (3.3.3 (x86_64/linux)). Content-Length: 0. . U 2.2.2.2:5060 -> 3.3.3.3:5060 REGISTER sip:3.3.3.3 SIP/2.0. Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bK929f.8339e446.0. Via: SIP/2.0/UDP 10.2.1.35:5060 ;received=1.1.1.1;rport=5060;branch=z9hG4bKPj85d21f7f-5c1d-4b0b-8e82-940c5cc9e21d. Max-Forwards: 69. From: <sip:101@2.2.2.2>;tag=b00a1df7-94bd-4b2e-8f63-15fea1a9015b. To: <sip:101@2.2.2.2>. Call-ID: e789b66c-d96c-45ca-b3b8-06bf65a2c3d1. CSeq: 59629 REGISTER. User-Agent: SFLphone. Contact: <sip:101@10.2.1.35:5060>. Expires: 600. Allow: PRACK, SUBSCRIBE, NOTIFY, REFER, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, REGISTER, OPTIONS, MESSAGE. Content-Length: 0. . U 3.3.3.3:5060 -> 2.2.2.2:5060 SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK929f.8339e446.0. Via: SIP/2.0/UDP 10.2.1.35:5060 ;received=1.1.1.1;rport=5060;branch=z9hG4bKPj85d21f7f-5c1d-4b0b-8e82-940c5cc9e21d. From: <sip:101@2.2.2.2>;tag=b00a1df7-94bd-4b2e-8f63-15fea1a9015b. To: <sip:101@2.2.2.2>;tag=d67d5778704611e2a3c600505691470a. Call-ID: e789b66c-d96c-45ca-b3b8-06bf65a2c3d1. CSeq: 59629 REGISTER. WWW-Authenticate: Digest realm="SIP-REGISTRAR", nonce="475a20c7". Server: MERA MVTS3G v.4.3.0-38t. Content-Length: 0. . U 2.2.2.2:5060 -> 1.1.1.1:5060 SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP 10.2.1.35:5060 ;received=1.1.1.1;rport=5060;branch=z9hG4bKPj85d21f7f-5c1d-4b0b-8e82-940c5cc9e21d. From: <sip:101@2.2.2.2>;tag=b00a1df7-94bd-4b2e-8f63-15fea1a9015b. To: <sip:101@2.2.2.2>;tag=d67d5778704611e2a3c600505691470a. Call-ID: e789b66c-d96c-45ca-b3b8-06bf65a2c3d1. CSeq: 59629 REGISTER. WWW-Authenticate: Digest realm="SIP-REGISTRAR", nonce="475a20c7". Server: MERA MVTS3G v.4.3.0-38t. Content-Length: 0. . Thanks in advance! --- Alex Elokhov

1 0

[Fwd: Deleting the expired contacts using the Rtimer module]
by kiran bhosale 06 Feb '13

06 Feb '13

1 0

Deleting the expired contacts using the Rtimer module
by kiran bhosale 06 Feb '13

06 Feb '13

Hi we have developed the custom module which stores the registered users in a file.now we are trying to remove the expired contacts using the rtimer module.while saving the registered users to the file we also store the expires values in static array.but when we try to decrement the these values in a function called with the help of rtimer module. the values used by this periodic function are not modified ones but the initial which are zero. is it that we cant pass the modified values of the global variables to the timed functions. to get around this problem, we also registered the timer in our module but got same results !!!!!!! Please Help. Thanks and regards .

1 0

Fwd:
by jose luis 05 Feb '13

05 Feb '13

http://www.lacucinalibanese.altervista.org/rjugo4.php?s=ot

2 1

Syslog
by Vitaliy Aleksandrov 05 Feb '13

05 Feb '13

Hi list. I have a strange problem with syslog. Kamailio doesn't send any messages to syslog from UDP workers that are bound to sock=127.0.0.1:5060. Neither LM_DBG() from a source code nor xlog("any level", "") works. I use kamailio-3.2.4 with this opions: log_stderror=no debug=3. With log_stderror=yes kamailio writes all debug messages to stderr without any problems.

2 4

modifying global data-structures using Rtimer module
by Kiran Bhosale 05 Feb '13

05 Feb '13

Hi we have developed the custom module which stores the registered users in a file.now we are trying to remove the expired contacts using the rtimer module.while saving the registered users to the file we also store the expires values in static array.but when we try to decrement the these values in a function called with the help of rtimer module. the values used by this periodic function are not modified ones but the initial which are zero. is it that we cant pass the modified values of the global variables to the timed functions. to get around this problem, we also registered the timer in our module but got same results !!!!!!! Please Help. Thanks and regards .

2 1

Refreshing the list of submitted patches
by Daniel-Constantin Mierla 05 Feb '13

05 Feb '13

Hello, following some discussion over the weekend at Fosdem, it seems that some of submitted patches got forgotten in the mailing list archives and even on tracker. If any of you submitted patches was not accepted or rejected for a persistent reason, please write again to sr-dev mailing list referring to it. It is a short period still good for getting them in v4.0. In general, if there is no more activity regarding a patch for one or two weeks, come back and ask for an update. We don't mind reminders (especially because many of us do have a heavy loaded traveling schedule). Cheers and happy testing upcoming v4.0, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

1 0

Best way to 'detect' direction
by Paul Belanger 04 Feb '13

04 Feb '13

Greetings, I'm in the process of adding logic to 'detect' the direction of my message, I have to set: rtpproxy_manage("ei") or rtpproxy_manage("ie") respectively. Rather then hard-coding IP addresses in my configuration, I was curious how other people over come this? Perhaps tagging their call control in some other method? -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belanger(a)polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger

2 1

50000+ subscribers on Kamailio?
by Алексей Ёлохов 04 Feb '13

04 Feb '13

Hi! I am choosing a softswitch to serve more than 50000 (up to 100000) active subscribers (each subscriber is using SIP registration). Please tell me , I want to understand can kamailio work with such load? --- Alex Elokhov

2 1

Remove dash (-) from called number
by Ali Jawad 04 Feb '13

04 Feb '13

Dear All I need to remove any dashes wherever they do occur in a called number, any guidance please ? Regards * *

2 3

Re: [SR-Users] sr-users Digest, Vol 92, Issue 80
by Ian French 04 Feb '13

04 Feb '13

Thank you very much Sammy that worked Ian On 29 January 2013 18:14, <sr-users-request(a)lists.sip-router.org> wrote: > Send sr-users mailing list submissions to > sr-users(a)lists.sip-router.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > or, via email, send a message with subject or body 'help' to > sr-users-request(a)lists.sip-router.org > > You can reach the person managing the list at > sr-users-owner(a)lists.sip-router.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of sr-users digest..." > > > Today's Topics: > > 1. Re: Asterisk and dispatcher (Ian French) > 2. Re: Asterisk and dispatcher (SamyGo) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Tue, 29 Jan 2013 17:55:40 +0000 > From: Ian French <fretec(a)gmail.com> > Subject: Re: [SR-Users] Asterisk and dispatcher > To: sr-users(a)lists.sip-router.org > Message-ID: > <CAOAGGZSr_24VDL8508XJo9LjkvR= > uo67JoM_MPPyd3bVbWxC7Q(a)mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > On 24 January 2013 08:16, <sr-users-request(a)lists.sip-router.org> wrote: > > > Send sr-users mailing list submissions to > > sr-users(a)lists.sip-router.org > > > > To subscribe or unsubscribe via the World Wide Web, visit > > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > or, via email, send a message with subject or body 'help' to > > sr-users-request(a)lists.sip-router.org > > > > You can reach the person managing the list at > > sr-users-owner(a)lists.sip-router.org > > > > When replying, please edit your Subject line so it is more specific > > than "Re: Contents of sr-users digest..." > > > > > > Today's Topics: > > > > 1. Re: Asterisk and dispatcher (SamyGo) > > 2. Re: Asterisk and dispatcher (SamyGo) > > 3. Kamailio Multi tenant as registrar server ony (andrea mucci) > > 4. help me about siremis. (???) > > > > > > ---------------------------------------------------------------------- > > > > Message: 1 > > Date: Wed, 23 Jan 2013 22:26:26 -0800 > > From: SamyGo <govoiper(a)gmail.com> > > Subject: Re: [SR-Users] Asterisk and dispatcher > > To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > > Users Mailing List" <sr-users(a)lists.sip-router.org> > > Message-ID: > > < > > CAJUJwtjjg+iGH6+QuDtRkEH+mn+dkiObUFcPyNMKTv-0xHnYMQ(a)mail.gmail.com> > > Content-Type: text/plain; charset="iso-8859-1" > > > > Hi, > > > > You've metioned this already in your code: > > > > if(ds_is_from_list("1")) > > > > Is this not working for you? > > > > Regards, > > Sammy > > > > > > > > On Wed, Jan 23, 2013 at 10:44 AM, Ian French <fretec(a)gmail.com> wrote: > > > > > Hi, > > >> I've been working my way through this tutorial ( > > >> > > > http://kb.asipto.com/asterisk:realtime:kamailio-3.0.x-asterisk-1.6.2-astdb > > ) > > >> and all it working well thanks. In the other benifits section you > > mention > > >> that this configuration can be used with multiple instances of > Asterisk. > > >> I've configured my [TOASTERISK] route > > > > > > > > > route[TOASTERISK] { > > > ds_select_dst("1","4"); > > > $sht(forw=>$ft)=$du; > > > sl_send_reply("100","Trying"); > > > xlog("L_INFO","INVITE: INVITE [$tU] Passed to [$du] Asterisk > > > Cluster [$rm:$au].\n"); > > > route(RELAY); > > > exit; > > > > > > How can i achieve the same for my [FROMASTERISK] route as now with > > > > > > asterisk.bindip = "x.x.x.x" desc "Asterisk IP Address" > > > asterisk.bindport = "5060" desc "Asterisk Port" > > > > > > and > > > > > > route[FROMASTERISK] { > > > if($si==$sel(cfg_get.asterisk.bindip) > > > && $sp==$sel(cfg_get.asterisk.bindport)) > > > xlog("L_INFO","INVITE: INVITE Passed From [$fu] > Asterisk > > > Cluster To [$rm:$au].\n"); > > > return 1; > > > return -1; > > > } > > > > > > This allows calls only from cfg_get.asterisk.bindip (single gateway) > > where > > > I would like gateways in the dispatcher list to be selected and marked > > as a > > > positive match > > > > > > if(ds_is_from_list("1")) > > > $sht(forw=>$ft)=$si; > > > xlog("L_INFO","INVITE: INVITE Passed From [$fu] > Asterisk > > > Cluster To [$rm:$au].\n"); > > > return 1; > > > return -1; > > > } > > > > > > Can you help? > > > > > > Thanks in advance > > > Ian > > > _______________________________________________ > > > 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

Voip Issue
by BrianSuper7 04 Feb '13

04 Feb '13

Spam detection software, running on the system "mail.iptel.org", has identified this incoming email as possible spam. The original message has been attached to this so you can view it (if it isn't spam) or label similar future email. If you have any questions, see the administrator of that system for details. Content preview: Hi I have been using your excellent Voip facility for sometime. Recently I have encountered that when I called from 'phone A' to 'Phone B' connects and can hear Phone B's voice but can't hear phone A's voice when dialed from phone A. But when Phone B made the call to phone A, can hear both parties voices. This problem started very recently. I haven't done any configuration changes. This problem incurred nearly a week or two ago. I experienced same problem long time ago and it resolved after few days time. But this time it didn't resolved. My SIP no 87415661. I would appreciate if you could kindly help me to rectify the problem. [...] Content analysis details: (5.2 points, 5.0 required) pts rule name description ---- ---------------------- -------------------------------------------------- -0.0 RCVD_IN_DNSWL_NONE RBL: Sender listed at http://www.dnswl.org/, no trust [220.233.0.17 listed in list.dnswl.org] 0.0 FREEMAIL_FROM Sender email is commonly abused enduser mail provider (briansuper7[at]yahoo.com.au) 0.2 FREEMAIL_REPLYTO_END_DIGIT Reply-To freemail username ends in digit ("briansuper7" <briansuper7[at]yahoo.com.au> ) 0.2 FREEMAIL_ENVFROM_END_DIGIT Envelope-from freemail username ends in digit (briansuper7[at]yahoo.com.au) 0.0 HTML_MESSAGE BODY: HTML included in message 0.8 BAYES_50 BODY: Bayes spam probability is 40 to 60% [score: 0.4955] 0.6 REPTO_QUOTE_YAHOO Yahoo! doesn't do quoting like this 3.3 AXB_XMAILER_MIMEOLE_OL_4379D AXB_XMAILER_MIMEOLE_OL_4379D The original message was not completely plain text, and may be unsafe to open with some email clients; in particular, it may contain a virus, or confirm that your address can receive spam. If you wish to view it, it may be safer to save it to a file and open it with an editor.

1 0

issue with #<number> calls
by Ivan Milivojevic 03 Feb '13

03 Feb '13

Hi Andrew, Here is one call with special characters: call-id = 100c8a98-c0a8016b-13c4-50029-203fa-68e78324-203fa How we can block such attempts, by rewrite rules? Regards Ivan

1 0

Build 3.3 on ubuntu 12.10
by Gareth Rylance 01 Feb '13

01 Feb '13

Hi I am trying to build the kamailio deb packages on ubuntu 12.04. when I run make deb I get dpkg-checkbuilddeps: Unmet build dependencies: libmono-dev but libmono-dev is not available on 12.04 how do I 1. Fix the dependancy 2. Ignore it as I am not using app_mono (although I have an idea that may in the future) Thanks Gareth

2 2

loose_route causing loop
by Renan Capaverde 31 Jan '13

31 Jan '13

Hello, I have a problem with my configuration file. When I receive a in-dialog request (REINVITE) with topmost route being my kamailio server, it enters in loop and then the call is dropped. What is the easiest way to prevent this to happen? Best Regards, -- *Renan Capaverde* /Estagiário Nível Superior/ /DDT - STE - Soluções em Tecnologia e Embarcados/ *DÍGITRO TECNOLOGIA* *E-mail:* renan.capaverde(a)digitro.com.br <mailto:renan.capaverde@digitro.com.br> *Fone:* +55 48 3281-7000*Ramal:* 8132 *Fax:* +55 48 3281-7299 *Site:* www.digitro.com <http://www.digitro.com> /"Antes de imprimir, pense na sua responsabilidade e no seu compromisso com o meio ambiente"/ Esta mensagem, incluindo seus anexos, é reservada somente à Dígitro e ao destinatário da mensagem. Caso você tenha recebido esta mensagem por engano, queira por favor, retorná-la ao remetente e apagá-la de seus arquivos.

2 1

Re: [SR-Users] Error searching $keepalive.socket pvar in NAT Traversal module
by vito 31 Jan '13

31 Jan '13

Hello, I have: # Kamailio (OpenSER) SIP Server v3.3 # Kamailio (OpenSER) SIP Server v3.0 Defining pv in this format works well: $fs = $(keepalive.socket($du)); Thanks a lot !! Vito

4 4

BasicKamailio DB question
by Jason 31 Jan '13

31 Jan '13

Hi all, excuse the basic nature of my question, I'm just starting out. I've read in a tutorial that it is possible to connect from asterisk to Kamailio and for asterisk to use the Kamailio db for authentication. However, I also read that it is possible to connect to astereisk from Kamailio for retrieving auth data. My question is this... If Kamailio uses the db on a remote asterisk server to retrieve usernames and password info how do you set Kamailio to use multiple databases as you may ultimately have multiple asterisk servers? As far as I could see there is only 1 db typically in the config file. Also, which is the most common configuration and why would one choose one over the other? Thanks in advance.

2 1

REGISTER OK but on send "407 Proxy Authentication Required" with no ACK
by Carsten Maass 30 Jan '13

30 Jan '13

Hi all, I am trying to set up a fax gateway in the following way: PSTN-GW (10.1.1.150) <--> Kamailio (10.1.1.148:5123) <--> t38modem (10.1.1.148:6050) <--> Hylafax PSTN-GW is a standalone Berofix appliance and Kamailio 3.3.3, t38modem 1.2 and Hylafax 6.0.3 running on the same host under Redhat EL6. I used the default kamailio.cfg and just adjusted the PSTN route pattern to "if(!($rU=~"^(\+|00|0)[1-9][0-9]{3,20}$"))", to route all calls starting with 0 to the PSTN-GW. Both PSTN-GW and t38modem Successfully register to Kamailio but when I try to send out a fax from t38modem, Kamailio doesn't ACK the Proxy-Authorization request and t38modem terminates the call: <siptrace> 05:49:11.158157 IP (tos 0x0, ttl 64, id 0, offset 0, flags [DF], proto UDP (17), length 986) 10.1.1.148.6050 > 10.1.1.148.5123: UDP, length 958 E.....@.@.Y. .d. .d.........INVITE sip:030987654321@10.1.1.148:5123 SIP/2.0 Route: <sip:10.1.1.148:5123;lr> Date: Tue, 29 Jan 2013 04:49:11 GMT CSeq: 1 INVITE Via: SIP/2.0/UDP 10.1.1.148:6050;branch=z9hG4bK020144e4-3c68-e211-98e4-005056bd002b;rport User-Agent: T38Modem/1.2.0 From: "root" <sip:030123456789@10.1.1.148:5123>;tag=169d43e4-3c68-e211-98e4-005056bd002b Call-ID: 16a243e4-3c68-e211-98e4-005056bd002b(a)myhost.mydomain.local Organization: Vyacheslav Frolov To: <sip:030987654321@10.1.1.148:5123> Contact: <sip:030123456789@10.1.1.148:6050> Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING Content-Type: application/sdp Content-Length: 262 Max-Forwards: 70 v=0 o=- 1359434951 1 IN IP4 10.1.1.148 s=Opal SIP Session c=IN IP4 10.1.1.148 t=0 0 m=audio 5008 RTP/AVP 8 101 100 a=sendrecv a=rtpmap:8 PCMA/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16,32,36 a=rtpmap:100 NSE/8000 a=fmtp:100 192-193 05:49:11.158927 IP (tos 0x10, ttl 64, id 28825, offset 0, flags [none], proto UDP (17), length 584) 10.1.1.148.5123 > 10.1.1.148.6050: UDP, length 556 E..Hp...@.*. .d. .d......4.oSIP/2.0 407 Proxy Authentication Required CSeq: 1 INVITE Via: SIP/2.0/UDP 10.1.1.148:6050;branch=z9hG4bK020144e4-3c68-e211-98e4-005056bd002b;rport=6050 From: "root" <sip:030123456789@10.1.1.148:5123>;tag=169d43e4-3c68-e211-98e4-005056bd002b Call-ID: 16a243e4-3c68-e211-98e4-005056bd002b(a)myhost.mydomain.local To: <sip:030987654321@10.1.1.148:5123>;tag=75c88a497c1ef184e8ac0d2e60c130e4.0580 Proxy-Authenticate: Digest realm="10.1.1.148", nonce="UQdV81EHVMeoOdE8gs8P4V1EFJAb79/W" Server: kamailio (3.3.3 (x86_64/linux)) Content-Length: 0 05:49:11.160273 IP (tos 0x0, ttl 64, id 0, offset 0, flags [DF], proto UDP (17), length 501) 10.1.1.148.6050 > 10.1.1.148.5123: UDP, length 473 E.....@.@.[. .d. .d.........ACK sip:030987654321@10.1.1.148:5123 SIP/2.0 Route: <sip:10.1.1.148:5123;lr> CSeq: 1 ACK Via: SIP/2.0/UDP 10.1.1.148:6050;branch=z9hG4bK020144e4-3c68-e211-98e4-005056bd002b;rport From: "root" <sip:030123456789@10.1.1.148:5123>;tag=169d43e4-3c68-e211-98e4-005056bd002b Call-ID: 16a243e4-3c68-e211-98e4-005056bd002b(a)myhost.mydomain.local To: <sip:030987654321@10.1.1.148:5123>;tag=75c88a497c1ef184e8ac0d2e60c130e4.0580 Content-Length: 0 Max-Forwards: 70 05:49:11.166542 IP (tos 0x0, ttl 64, id 0, offset 0, flags [DF], proto UDP (17), length 1195) 10.1.1.148.6050 > 10.1.1.148.5123: UDP, length 1167 E.....@.@.Y. .d. .d.........INVITE sip:030987654321@10.1.1.148:5123 SIP/2.0 Route: <sip:10.1.1.148:5123;lr> Date: Tue, 29 Jan 2013 04:49:11 GMT CSeq: 2 INVITE Via: SIP/2.0/UDP 10.1.1.148:6050;branch=z9hG4bK146145e4-3c68-e211-98e4-005056bd002b;rport User-Agent: T38Modem/1.2.0 From: "root" <sip:030123456789@10.1.1.148:5123>;tag=169d43e4-3c68-e211-98e4-005056bd002b Call-ID: 16a243e4-3c68-e211-98e4-005056bd002b(a)myhost.mydomain.local Organization: Vyacheslav Frolov To: <sip:030987654321@10.1.1.148:5123> Contact: <sip:030123456789@10.1.1.148:6050> Proxy-Authorization: Digest username="979", realm="10.1.1.148", nonce="UQdV81EHVMeoOdE8gs8P4V1EFJAb79/W", uri="sip:030987654321@10.1.1.148:5123", algorithm=MD5, response="48d896ef41a71720b29c56230858582f" Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING Content-Type: application/sdp Content-Length: 262 Max-Forwards: 70 v=0 o=- 1359434951 1 IN IP4 10.1.1.148 s=Opal SIP Session c=IN IP4 10.1.1.148 t=0 0 m=audio 5008 RTP/AVP 8 101 100 a=sendrecv a=rtpmap:8 PCMA/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16,32,36 a=rtpmap:100 NSE/8000 a=fmtp:100 192-193 05:49:11.167573 IP (tos 0x10, ttl 64, id 28826, offset 0, flags [none], proto UDP (17), length 584) 10.1.1.148.5123 > 10.1.1.148.6050: UDP, length 556 E..Hp...@.*. .d. .d......4.oSIP/2.0 407 Proxy Authentication Required CSeq: 2 INVITE Via: SIP/2.0/UDP 10.1.1.148:6050;branch=z9hG4bK146145e4-3c68-e211-98e4-005056bd002b;rport=6050 From: "root" <sip:030123456789@10.1.1.148:5123>;tag=169d43e4-3c68-e211-98e4-005056bd002b Call-ID: 16a243e4-3c68-e211-98e4-005056bd002b(a)myhost.mydomain.local To: <sip:030987654321@10.1.1.148:5123>;tag=75c88a497c1ef184e8ac0d2e60c130e4.263a Proxy-Authenticate: Digest realm="10.1.1.148", nonce="UQdV81EHVMeoOdE8gs8P4V1EFJAb79/W" Server: kamailio (3.3.3 (x86_64/linux)) Content-Length: 0 05:49:11.169220 IP (tos 0x0, ttl 64, id 0, offset 0, flags [DF], proto UDP (17), length 710) 10.1.1.148.6050 > 10.1.1.148.5123: UDP, length 682 E.....@.@.Z. .d. .d.........ACK sip:030987654321@10.1.1.148:5123 SIP/2.0 Route: <sip:10.1.1.148:5123;lr> CSeq: 2 ACK Via: SIP/2.0/UDP 10.1.1.148:6050;branch=z9hG4bK146145e4-3c68-e211-98e4-005056bd002b;rport From: "root" <sip:030123456789@10.1.1.148:5123>;tag=169d43e4-3c68-e211-98e4-005056bd002b Call-ID: 16a243e4-3c68-e211-98e4-005056bd002b(a)myhost.mydomain.local To: <sip:030987654321@10.1.1.148:5123>;tag=75c88a497c1ef184e8ac0d2e60c130e4.263a Proxy-Authorization: Digest username="979", realm="10.1.1.148", nonce="UQdV81EHVMeoOdE8gs8P4V1EFJAb79/W", uri="sip:030987654321@10.1.1.148:5123", algorithm=MD5, response="3056b266b15a92f5cc8cc8e1c073fbcb" Content-Length: 0 Max-Forwards: 70 </siptrace> What do I have to adjust to make this work? Any help and pointers highly appreciated. Thanks in advance and greetings, Carsten. -- Blinkenlichten (Maass & Sacha GbR) - Open Source Solutions Weigandufer 45 - 12059 Berlin - http://www.blinkenlichten.de FON: ++49 +30 13896247 - MAIL: cm(a)blinkenlichten.de FAX: ++49 +30 13896249 - PGP: Key Id 0x2CBCA806 St.Nr. 16/274/61636

2 7

Re: [SR-Users] Asterisk and dispatcher
by Ian French 29 Jan '13

29 Jan '13

On 24 January 2013 08:16, <sr-users-request(a)lists.sip-router.org> wrote: > Send sr-users mailing list submissions to > sr-users(a)lists.sip-router.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > or, via email, send a message with subject or body 'help' to > sr-users-request(a)lists.sip-router.org > > You can reach the person managing the list at > sr-users-owner(a)lists.sip-router.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of sr-users digest..." > > > Today's Topics: > > 1. Re: Asterisk and dispatcher (SamyGo) > 2. Re: Asterisk and dispatcher (SamyGo) > 3. Kamailio Multi tenant as registrar server ony (andrea mucci) > 4. help me about siremis. (???) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Wed, 23 Jan 2013 22:26:26 -0800 > From: SamyGo <govoiper(a)gmail.com> > Subject: Re: [SR-Users] Asterisk and dispatcher > To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List" <sr-users(a)lists.sip-router.org> > Message-ID: > < > CAJUJwtjjg+iGH6+QuDtRkEH+mn+dkiObUFcPyNMKTv-0xHnYMQ(a)mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > Hi, > > You've metioned this already in your code: > > if(ds_is_from_list("1")) > > Is this not working for you? > > Regards, > Sammy > > > > On Wed, Jan 23, 2013 at 10:44 AM, Ian French <fretec(a)gmail.com> wrote: > > > Hi, > >> I've been working my way through this tutorial ( > >> > http://kb.asipto.com/asterisk:realtime:kamailio-3.0.x-asterisk-1.6.2-astdb > ) > >> and all it working well thanks. In the other benifits section you > mention > >> that this configuration can be used with multiple instances of Asterisk. > >> I've configured my [TOASTERISK] route > > > > > > route[TOASTERISK] { > > ds_select_dst("1","4"); > > $sht(forw=>$ft)=$du; > > sl_send_reply("100","Trying"); > > xlog("L_INFO","INVITE: INVITE [$tU] Passed to [$du] Asterisk > > Cluster [$rm:$au].\n"); > > route(RELAY); > > exit; > > > > How can i achieve the same for my [FROMASTERISK] route as now with > > > > asterisk.bindip = "x.x.x.x" desc "Asterisk IP Address" > > asterisk.bindport = "5060" desc "Asterisk Port" > > > > and > > > > route[FROMASTERISK] { > > if($si==$sel(cfg_get.asterisk.bindip) > > && $sp==$sel(cfg_get.asterisk.bindport)) > > xlog("L_INFO","INVITE: INVITE Passed From [$fu] Asterisk > > Cluster To [$rm:$au].\n"); > > return 1; > > return -1; > > } > > > > This allows calls only from cfg_get.asterisk.bindip (single gateway) > where > > I would like gateways in the dispatcher list to be selected and marked > as a > > positive match > > > > if(ds_is_from_list("1")) > > $sht(forw=>$ft)=$si; > > xlog("L_INFO","INVITE: INVITE Passed From [$fu] Asterisk > > Cluster To [$rm:$au].\n"); > > return 1; > > return -1; > > } > > > > Can you help? > > > > Thanks in advance > > Ian > > _______________________________________________ > > 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

Re: [SR-Users] Default 3.3 config and dispatcher
by Daniel Tryba 29 Jan '13

29 Jan '13

On Tuesday 29 January 2013 16:58:08 Paul Belanger wrote: > > [kamailio] > > host=192.168.1.1 > > context=incoming > > type=friend > > allow=all > > qualify=yes > > insecure=port,invite > > nat=no > > canreinvite=yes > > > > Nothing more to it. > > Thanks for pointing me in the right direction. Will provide feedback > shortly. One more thing I might have forgotten. Asterisk needs to responde with a 200 OK to the OPTIONS requests in order for Kamailio to think the dispatcher route is up and runnnig. So add an s exten in the asterisk context: [incoming] exten => s,1,NoOp() -- POCOS B.V. - Croy 9c - 5653 LC Eindhoven Telefoon: 040 293 8661 - Fax: 040 293 8658 http://www.pocos.nl/ - http://www.sipo.nl/ K.v.K. Eindhoven 17097024

1 0

Subscribe and PUA database
by Misha Ketslah 29 Jan '13

29 Jan '13

Hello, i'm trying to set BLF on my snom 320 phone to get notification on function key for certain extension as well registered against kamailio. Any other info i can provide except the following one and as well i want to exuse me, as i'm pretty noob to all this, and my question might be silly, but after hours in front of google i found no solution :) I always get that: after pua_set_publish() ** 3(7825) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=791 a=16 n=if 3(7825) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=780 a=41 n=isflagset 3(7825) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=799 a=16 n=if 3(7825) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=791 a=25 n=save 3(7825) ERROR: pua [send_publish.c:578]: New PUBLISH and no body found- invalid request 3(7825) ERROR: pua_usrloc [ul_publish.c:326]: while sending publish 3(7825) ERROR: pua_usrloc [ul_publish.c:327]: TEST: type & UL_CONTACT_UPDATE is 2, and error is -1 (^^^ This line was added by me into module as i saw related thread and the patch solution, though it never meets condition and fails to insert from there as well.) 3(7825) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=799 a=2 n=exit my configs are : in BLF i set <sip:666@SERV_IP;user=phone> in cfg i have alias="SERV_IP" listen=udp:SERV_IP port=5060 loadmodule "db_mysql.so" loadmodule "mi_fifo.so" loadmodule "kex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "mi_rpc.so" loadmodule "acc.so" loadmodule "auth.so" loadmodule "auth_db.so" loadmodule "alias_db.so" loadmodule "speeddial.so" ... modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 3) modparam("usrloc", "use_domain", MULTIDOMAIN) ... modparam("presence", "db_url", DBURL) modparam("presence", "subs_db_mode" , 3 ) #modparam("presence", "publ_cache", 0) modparam("presence", "expires_offset", 300) modparam("presence", "max_expires", 3600) #modparam("presence", "force_active", 1) modparam("presence", "db_update_period", 10) modparam("presence", "server_address", "sip:SERV_IP:5060") modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #modparam("presence_dialoginfo", "force_single_dialog", 1) modparam("pua", "db_url" ,DBURL ) modparam("pua", "db_mode", 2) modparam("pua", "db_table", "pua") #modparam("pua", "outbound_proxy", "sip:SERV_IP") modparam("pua", "min_expires", 10) modparam("pua", "default_expires", 3600) modparam("pua", "update_period", 60) modparam("pua_usrloc", "default_domain", "SERV_IP") request_route { if(method=="NOTIFY") { xlog(" pua_update_contact $rm from $fu (IP:$si:$sp) \n"); if(!pua_update_contact()) xlog("pua update failed \n"); } # per request initial checks route(REQINIT); # NAT detection route(NATDETECT); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) t_check_trans(); # authentication route(AUTH); route(SPEEDDIAL); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } # dispatch requests to foreign domains route(SIPOUT); ### requests for my local domains # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations to PSTN route(PSTN); route(DISPATCH); # user location service route(LOCATION); route(RELAY); } route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|SUBSCRIBE")) { t_on_branch("MANAGE_BRANCH"); t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood dection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($sht(ipban=>$si)!=$null) { # ip is already blocked xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); exit; } if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); $sht(ipban=>$si) = 1; exit; } } #!endif if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { route(DLGURI); if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } else if ( is_method("ACK") ) { # ACK is forwarded statelessy route(NATMANAGE); } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); } exit; } } # Handle SIP registrations route[REGISTRAR] { if (is_method("REGISTER")) { xlog("pua_set_publish()\n"); if(!pua_set_publish()) xlog("set publish failed \n"); xlog(" after pua_set_publish()\n"); if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging ## setbflag(FLB_NATSIPPING); } #pua_set_publish(); if (!save("location")) { sl_reply_error(); xlog("save location failed \n"); } exit; } } # USER location service route[LOCATION] { #!ifdef WITH_SPEEDIAL if(uri=~"sip:[0-9]{2}@.*") xlog("speeddials\n"); # search for short dialing - 2-digit extension if($rU=~"^[0-9][0-9]$") if(sd_lookup("speed_dial")) route(SIPOUT); #!endif #!ifdef WITH_ALIASDB # search in DB-based aliases if(alias_db_lookup("dbaliases")) route(SIPOUT); #!endif $avp(oexten) = $rU; if (!lookup("location")) { $var(rc) = $rc; route(TOVOICEMAIL); t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } } # Presence server route route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; # search for short dialing - 2-digit extension if($rU=~"^[0-9][0-9]$") sd_lookup("speed_dial"); route(SPEEDDIAL); xlog("PUBLISH|SUBSCRIBE"); sl_send_reply("100","trying"); if (!t_newtran()) { sl_reply_error(); exit; }; append_to_reply("Contact: <sip:82.80.18.100:5060>\r\n"); if(is_method("PUBLISH")) { xlog("handle publish here\n"); handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { handle subscribe here \n"); handle_subscribe(); t_release(); } exit; } # Authentication route route[AUTH] { #!ifdef WITH_AUTH #!ifdef WITH_IPAUTH if((!is_method("REGISTER")) && allow_source_address()) { # source IP allowed return; } #!endif if (is_method("REGISTER") || from_uri==myself) { # authenticate requests if (!auth_check("$fd", "subscriber", "1")) { auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) consume_credentials(); } # if caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (from_uri!=myself && uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!endif return; } # Caller NAT detection route route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { add_contact_alias(); } setflag(FLT_NATS); } #!endif return; } # RTPProxy control route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; rtpproxy_manage(); if (is_request()) { if (!has_totag()) { add_rr_param(";nat=yes"); } } if (is_reply()) { if(isbflagset(FLB_NATB)) { add_contact_alias(); } } #!endif return; } # URI update for dialog requests route[DLGURI] { #!ifdef WITH_NAT if(!isdsturiset()) { handle_ruri_alias(); } #!endif return; } # Routing to foreign domains route[SIPOUT] { if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(RELAY); } } # PSTN GW routing route[PSTN] { ....... return; } # XMLRPC routing route[XMLRPC] { .... } #!endif # route to voicemail server route[TOVOICEMAIL] { ...... return; } # manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); route(NATMANAGE); } # manage incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") route(NATMANAGE); } # manage failure routing cases failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_is_canceled()) { exit; } #!ifdef WITH_BLOCK3XX # block call redirect based on 3xx replies. if (t_check_status("3[0-9][0-9]")) { t_reply("404","Not found"); exit; } #!endif } route[SPEEDDIAL] { ..... return; } ....

1 0

Web softphone using websockets against Kamailio
by Pirjo Ahvenainen 29 Jan '13

29 Jan '13

Greetings gurus! I'm playing with an idea to create a web based softphone (html5 + no installations for the end user) and use Kamailio's websocket module for backend. I'd love to hear about your comments, challenges and successes using such configuration. Is it a feasible way to construct a softphone even today when even IE9 does not support websockets, as such? I'm sure IE9 will end up in specs as a must-support platform. A collegue tried using sipml5 with webrtc against a SnomONE pbx (I know... ;)), and said there's no way it can work, but I'm not convinced the idea itself wouldn't work. It would help me lots if I could make a simple example using Kamailio with SIP over websockets, can you comment on how much effort do I need on Kamailio side to make this work? Do I need off-default config scripting, or is it enough to just set up the module and set the parameters? And even with the risk of stepping a little off topic, if anyone has worked on web based softphones, I'd love to hear if you can recommend on how to approach this. Cheers, Pirjo --

2 3

Default 3.3 config and dispatcher
by Paul Belanger 29 Jan '13

29 Jan '13

Greetings List, First post of many. I've managed to get up a running a very simple VoIP network with Kamailio as the core, and asterisk on the edge. Most of this has been accomplished using the default kamailio.cfg file shipped in 3.3. For my next adventure, I'd like to start using the dispatch module to route calls from Kamailio to asterisk, however having some trouble getting my syntax correct. I was hoping somebody could point me to an existing post / URL describing what I was asking, if it is based on the default configuration file, the better. I should note, if I have Asterisk register to kamailio, I can route calls fine however, I believe the dispatcher is the better way to do this. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belanger(a)polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger

2 1

Major problem with setup NAT + MULTIDOMAIN
by Rumen Mihailov 29 Jan '13

29 Jan '13

Hi guys, I've baning my head for the last 6 7 days with something that have to be universally simple but obviously for me it is not, so please do help. I have the following setup. xx.xx.xx.xx (real IP 1:1 NAT) -> 10.2.47.201 Kamailio listens on 10.2.47.201 I have setup a domain xxx.itradebg.com and I've made the appropriate records in the DNS. I have inserted the domain xxx.itradebg.com in the domain table in the database and kamctl domain show shows the correct thing. Now when I try to register 999(a)xxx.itradebg.com I get Request timed out I have this in my config: if (is_method("REGISTER") || from_uri==myself) { if (!auth_check("$fd", "sip", "1")) { auth_challenge("$fd", "0"); exit; } } if (from_uri!=myself && uri!=myself) { if(lookup_domain("$fd", "@from.uri.host")){ xlog("LOCAL DOMAIN -> REGISTER"); exit; } else{ xlog("L_WARN","FROM NOT LOCAL"); sl_send_reply("403","Sorry mate not relaying"); exit; } } By the time of writing this I realised that this should be all rewritten to this only, as the second check is useless: if (is_method("REGISTER") || from_uri==myself) { if (!auth_check("$fd", "sip", "1")) { auth_challenge("$fd", "0"); exit; } } However I get 401 Unauthorized. The previous setup was giving me 408 - Request Timed out... I am pretty sure I have the passwords correct. What I cannot understand is why authentication fails ?! Also, how do I set kamailio to print debug output ? debug=9 log_stderror=no This is what I have in the .conf file. /var/log/messages seems to only have what I have printed with xlog... (think) I am really out of ideas... perhaps I need some sleep. Anyway any input is helpful guys. Thanks, Rumen http://itradebg.com

2 4

Error searching $keepalive.socket pvar in NAT Traversal module
by vito 28 Jan '13

28 Jan '13

Hello, I started to use Kamailio few months ago. Maybe this is a stupid issue but I can´t start Kamailio with NAT TRAVERSAL module and $keepalive.socket pseudo-variable. My script configuration is the default template adding this lines (extracted from nat traversal doc): ... loadmodule "nat_traversal.so" ... ... $fs = $keepalive.socket($du); ... I have this error: -e 0(17455) ERROR: <core> [pvapi.c:501]: error searching pvar "keepalive" 0(17455) ERROR: <core> [pvapi.c:705]: wrong char [e/101] in [$keepalive] at [9 (0)] 0(17455) : <core> [cfg.y:3330]: parse error in config file /etc/kamailio/kamailio.cfg, line 410, column 8-17: unknown script pseudo variable $keepalive ERROR: bad config file (1 errors) I tried in this environments: Ubuntu 10.04.4 LTS x86_64 Kamailio 3.3.3+git20130128+lucid1 Ubuntu 10.04.4 LTS i686 Kamailio 3.0.4+lucid1 Ubuntu 12.04.1 LTS x86_64 Kamailio 3.3.2+precise1 All packages instaled from deb repositories (http://deb.kamailio.org/) I think there are problem parsing "$keepalive.socket" pseudovariable, because Kamailio doesn´t scape this character "." and it doesn´t recognise the pseudovar. Does anyone have this problem? Regards

2 1

Sip express live Cd router
by Shoukat, Kashif 28 Jan '13

28 Jan '13

Dear , Can you please provide me link of sip express live Cd router .. I want to use it... waiting your kind response Regards, Kashif Shoukat | Analyst Software Engineer | Work : + 1.202.289.9898 x 9077 Mobile : + 92.323.431.8548 Skype: kashif.shoukat.dgs www.dgsworld.com<http://www.dgsworld.com/> dgs deliberate by design This message contains information which may be confidential and privileged. Unless you are the addressee (or authorized to receive for the addressee), you may not use, copy or disclose to anyone the message or any information contained in the message. If you have received the message in error, please advise the sender by replying to kashif.shoukat(a)dgsworld.com<mailto:kashif.shoukat@dgsworld.com> and delete the message.

1 0

TLS certificate chain Android verification
by Nord7 28 Jan '13

28 Jan '13

Hello all. Our company buy Thawte SSL Web server certificate. I'm enabled in Kamailio 3.3 TLS support and configured him. Adding to "myhost.pem" certificate, two "intermediate" certificates (like in chain) "cat myhost.pem SSL_SecondaryCA.pem SSL_PrimaryCA.pem > chain-cert.pem". Remark: I use this chain on apache2. And Android trusted to this certificate chain. After this i use chain-cert.pem with KAMAILIO 3.3 and SIP client "BRIA Android". "Bria Android" says certificate is not trusted. Next step "BRIA IPHONE" and certificate chain works (trusted)! BRIA support says me - BRIA don't trusts certificate if system don't trusts. Why Android don't trusts to my chain-certificates with KAMAILIO? If his trust to same certificate with Apache? I'm understand this is not Kamailio problem, but maybe any one who have expirience in this problem help me? I know about bouncycastle, but this is not best decision for me. Maybe something else can help? I tryed this with Android 2.3.7, 4.0, 4.1. Best regards Kirill.

1 0

event_route for pre-dialog start
by Mino Haluz 28 Jan '13

28 Jan '13

Hi, I think that it is more than likely not implemented, but is there any event-route that is triggered just before event_route[dialog:start] ? I need to check some security flags before the dialog is created. But it is too early too check them in relay route.. Mino

1 0

Is kamailio creating multiple connections to db?
by Mino Haluz 28 Jan '13

28 Jan '13

Hi, I made some DB test on our HA cluster, and I got that for 1 process I have 1000 queries/sec and for 11 processes 3000 queries/sec. That's why I would to ask if every kamailio process that is spawned has its own database connection or queries are put in the queue and executed by some main process ? I would like to know this, so that I could assess maximum call count/sec that db can handle. Thank you, Mino

3 4

Generating logging based on SIP messages
by Grant Bagdasarian 28 Jan '13

28 Jan '13

Hello, A while ago I came across the siptrace and sipcapture modules and thought this would be a good way to generate logging out of all the SIP messages that are passed through Kamailio and stored in the siptrace/sipcapture table. My idea was to extract certain values from SIP messages and store these in our database. Also each SIP Request or Response would be mapped to a certain status so we would be able to monitor which call is in which state. I still have a lot of work to do to think this all out, since it's still only an idea. I'm just wondering if anyone could give me some pointers into what to keep in mind when doing this. Regards, Grant

2 2

Pseudo variable to get instance in the contact....
by Krishna Kurapati 28 Jan '13

28 Jan '13

Hi, I added in my local branch "$ct.inst" pseudo variable to retrieve instance. Since instance could be an important attribute to make routing decisions, having quick access to it in the message may be useful. Anyone else need this functionality? Krish Kura

3 2

iptel x media5-fone problem
by Kasper Hämäläinen 28 Jan '13

28 Jan '13

hi guys, hope u got time to help me. Im first timer testing VoIP and SIP. I created account to ur iptel.org site. I downloaded "Jitsi"-software. First when I opened it, I login with my username and password. Then I go to "add new account", give same login username and password, but Jitsi says that my password is wrong..wtf. And I installed Media5-Fone to my iPhone 4S. When adding SIP-account it's says again that I got wrong password. And YES, I didn't mistype it. Thanks - Kasper

2 1

CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg - Kamailio 3.3.x
by Rinor Hoxha 28 Jan '13

28 Jan '13

Hi list, I'm having some issues with dialog module. From time to time I get the following error: ERROR: dialog [dlg_req_within.c:231]: failed to update dialog lifetime CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg tl=0x2aeeea1c6840 tl->next=(nil) tl->prev=(nil) [root@proxy ~]# cat /var/log/kamailio.log | grep "Jan 17" | grep "tl=" | sort | uniq -c 1 Jan 17 13:27:00 proxy /usr/local/kamailio/sbin/kamailio[4100]: CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg tl=0x2aeeea09bf38 tl->next=(nil) tl->prev=(nil) 1 Jan 17 14:39:30 proxy /usr/local/kamailio/sbin/kamailio[4100]: CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg tl=0x2aeeea07c518 tl->next=(nil) tl->prev=(nil) 1 Jan 17 15:06:21 proxy /usr/local/kamailio/sbin/kamailio[4100]: CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg tl=0x2aeeea2da900 tl->next=(nil) tl->prev=(nil) 1 Jan 17 19:31:02 proxy /usr/local/kamailio/sbin/kamailio[4100]: CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg tl=0x2aeeea278cd0 tl->next=(nil) tl->prev=(nil) 1 Jan 17 19:31:42 proxy /usr/local/kamailio/sbin/kamailio[4100]: CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg tl=0x2aeee9f527a8 tl->next=(nil) tl->prev=(nil) 1 Jan 17 20:05:32 proxy /usr/local/kamailio/sbin/kamailio[4100]: CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg tl=0x2aeeea21fa30 tl->next=(nil) tl->prev=(nil) This one was flooding my log file: tl=0x2aeeea1c6840 24322 Jan 17 21:53:02 proxy /usr/local/kamailio/sbin/kamailio[4100]: CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg tl=0x2aeeea1c6840 tl->next=(nil) tl->prev=(nil) 5709 Jan 17 21:53:23 proxy /usr/local/kamailio/sbin/kamailio[4100]: CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg tl=0x2aeeea1c6840 tl->next=(nil) tl->prev=(nil) 61314 Jan 17 21:53:23 proxy /usr/local/kamailio/sbin/kamailio[4102]: CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg tl=0x2aeeea1c6840 tl->next=(nil) tl->prev=(nil) 22812 Jan 17 21:54:01 proxy /usr/local/kamailio/sbin/kamailio[4102]: CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg tl=0x2aeeea1c6840 tl->next=(nil) tl->prev=(nil) 21137 Jan 17 21:54:02 proxy /usr/local/kamailio/sbin/kamailio[4102]: CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg tl=0x2aeeea1c6840 tl->next=(nil) tl->prev=(nil) where pid 4100 is slow timer and pid 4102 is timer. Any idea why is this happening and how to fix it. (I suspect that may be some issues with our server memory...checking now...and will inform). However wanted to raise this issue in case anyone else is having the same. Thanks, Rinor

2 1

Pseudo variable to access SIP instance in contact
by Krishna Kurapati 28 Jan '13

28 Jan '13

Hi, Is there a way to directly access the SIP instance present in message from the script? Thanks Krish Kura

2 1

Kamailio behind NAT - best practice
by Klaus Darilion 27 Jan '13

27 Jan '13

Hi! Up to now I could avoid Kamailio setups with Kamailio behind NAt. But this time I have to deal with it. That's why I want to ask what others did as best practice. The scenario is: Asterisk 1----\ Kamailio+RTPPROXY \ |10.10.0.2 Asterisk n------\ | --------|------------- FW --SIP-trunk--- ITSP Freeswitch 1----/ 10.10.0.1 public-IP / Freeswitch n--/ 10.10.0.x Kamailio and rtpproxy have a private IP. Internal communication uses private IPs, external communication uses a public IP which is NATed 1:1 to Kamailio's IP address. No registrations, just forwarding of messages. Using the global "advertised_address" setting with the public IP does not work, as there is also internal communication. Using set_advertised_address() is also cumbersome. So it seems, the easiest solution would be to use 2 sockets on Kamailio, e.g. port 5050 and port 5060. Then I could use the listen with dedicated advertised addresses: listen=udp:10.10.0.2:5050 listen=udp:10.10.0.2:5060 advertise pu.bl.ic.ip:5060 If I understand it correctly, this should solve all issues with Record-Routing and Via-headers. For RTP-Proxy it seems necessary to detect the direction of each message and set the IP address in rtpproxy_manage("","ip.add.re.ss") manually. Thus, it seems straight forward - or do I miss something? Any comments and practical experience? Thanks Klaus

5 8

Pseudo variables and uid_domain module compatibility
by Philippe Sultan 25 Jan '13

25 Jan '13

Hi, According to docs of "uid_domain.so", $fd and $td are attribute groups which are set when calling the lookup_domain function, to retrieve domain attributes stored in an DB ($fd.did or $td.did) : lookup_domain("$fd", "@from.uri.host"); lookup_domain("$td", "@ruri.host"); if (strempty($fd.did) && strempty($td.did)) { # Neither the calling nor the called domain is local # This is a relaying attempt which should be forbidden sl_reply("403", "Relaying Forbidden"); drop; } But then, starting Kamailio fails with a configuration file error : 0(2098) ERROR: <core> [pvapi.c:761]: error searching pvar "fd" 0(2098) ERROR: <core> [pvapi.c:965]: wrong char [d/100] in [$fd] at [2 (0)] 0(2098) : <core> [cfg.y:3567]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 332, column 15-17: unknown script pseudo variable $fd ERROR: bad config file (1 errors) Line 332 is : if (strempty($fd.did) && strempty($td.did)) { I thought I could start Kamailio without "pv.so" (and use selects instead), but the error still shows up, as it is reported by the core pvapi.c file. So I'm kind of stuck, and asking help from the community to fix this problem :D Philippe

2 8

uac_replace_from recovery with modified header
by Andreas Granig 25 Jan '13

25 Jan '13

Hi, This is a question for kamailio 3.0.x and I'm still trying to reproduce this issue somehow for latest stable, but maybe it was a known issue anyways which got fixed in newer versions, so I don't have to dig deeper into this: There is an INVITE "A -> kamailio -> B", and kamailio does uac_replace_from() to replace the display- and user-part (leaving the domain-part untouched). When there's a BYE "B -> kamailio -> A", then From/To are switched, so the replaced From of the INVITE becomes the replaced To of the BYE. Kamailio gets that one right and tries to recover (in auto-mode) the To. However, in this BYE, B changes the domain-part of the To, which should be valid, as elements are only supposed to inspect the tags of From/To, right? The problem though is that when recovering the To, kamailio inserts garbage into the domain part. The vsf-Parameter in the Route of the BYE is definitely ok though, so that doesn't seem to be the issue. Here's an example: In the INVITE, From: "ab12345"<sip:ab12345@192.168.0.12;user=phone>;tag=948a3340 gets rewritten to From: "12345"<sip:12345@192.168.0.12;user=phone>;tag=948a3340 And in the BYE, To: "12345"<sip:12345@1.2.3.4>;tag=948a3340 gets rewritten to To: "12345"<sip:ab12345<@-%:68*0.0%12;user=phone>;tag=948a3340 Note that the received To in the BYE has a different host-part than what got sent out in the From of the INVITE, and when kamailio recovers the To, it doesn't recover the Display-part (not sure though if this is intended behaviour anyway), but it only party recovers the domain-part. You can still see some artefacts of the original domain (like the "68" from the second "168" octet, and the "12" from the last octet). Has anyone encountered this issue before? Andreas

4 12

help me about siremis.
by 문종민 25 Jan '13

25 Jan '13

I'm developers who trying to install siremis. will go in there and install siremis website. However, server-services -> domain list by pressing will get the following error. What can I do to fix this?

2 1

problem with db access
by MASebastian 25 Jan '13

25 Jan '13

Hi everyone, Recently I work on kamailio's snmpstats module. I'm trying to retrieve some data from database, the code pass the compilation but when starting kamailio always get this error. Jan 25 10:57:05 localhost kamailio: ERROR: <core> [sr_module.c:572]: ERROR: load_module: could not open module </usr/local/lib/kamailio/modules_k/snmpstats.so>: /usr/local/lib/kamailio/modules_k/snmpstats.so: undefined symbol: db_bind_mod I take the code from http://www.asipto.com/pub/kamailio-devel-guide/#c09database and ursloc module for example. not sure if I still miss something to make db access work. Any suggestion? Thanks, Sebastian

2 1

Websockets WSS problem with NOTIFY
by Pete Kelly 24 Jan '13

24 Jan '13

Hi, I am having an issue at the moment with SIP NOTIFY messages being sent from Kamailio (latest git master) over wss transport I am getting reports from the receiving end saying "Compressed bit must be 0 if no negotiated deflate-frame extension" The only reference I can find to it is at the following URL... where the problem was caused by the server miscalculating the size of the msg: http://stackoverflow.com/questions/12308728/compressed-bit-must-be-0-when-s… Does anyone have any suggestions as to how I could debug this within Kamailio? It sounds like Kamailio may be sending some incorrect packet information but I am unsure at this point.

3 14

sipcapture + db_postgres gives lots of warnings on PostgreSQL server
by Øyvind Kolbu 24 Jan '13

24 Jan '13

Hi, while trying to use sipcapture together with PostgreSQL I've found that it stores the "\r\n" in SIP messages as "\015\012", though not without a lot of noise in the PostgreSQL log files. For every line inserted in PostgreSQL I get: [0] WARNING: nonstandard use of \\ in a string literal at character 763 [0] HINT: Use the escape string syntax for backslashes, e.g., E'\\'. [0] STATEMENT: insert into sip_capture (date,micro_ts,method,reply_reason,ruri,ruri_user,from_user,from_tag,to_user,to_tag,pid_user,contact_user,auth_user,callid,callid_aleg,via_1,via_1_branch,cseq,reason,content_type,auth,user_agent,source_ip,source_port,destination_ip,destination_port,contact_ip,contact_port,originator_ip,originator_port,proto,family,rtp_stat,type,node,msg) values ('2013-01-24 15:27:43',1359037663218925,'OPTIONS','','sip:129.240.253.134','','keepalive','533cb9e91f4b999cf76861cbb9ed54ed-149d','','','','','','5001ed5d-6979(a)127.0.0.1','','SIP/2.0/UDP 129.240.253.197;branch=z9hG4bKae6f.d4a02ac7.0','z9hG4bKae6f.d4a02ac7.0','10 OPTIONS','','','','kamailio (3.3.3 (i386/linux))','129.240.253.197',5060,'129.240.253.134',5060,'',0,'',0,1,2,'',1,'voipt1:21','OPTIONS sip:129.240.253.134 SIP/2.0\\015\\012Via: SIP/2.0/UDP 129.240.253.197;branch=z9hG4bKae6f.d4a02ac7.0\\015\\012To: sip:129.240.253.134\\015\\012From: sip:keepalive@edgetest.voip.uio.no;tag=533cb9e91f4b999cf76861cbb9ed54ed-149d\\015\\012CSeq: 10 OPTIONS\\015\\012Call-ID: 5001ed5d-6979(a)127.0.0.1\\015\\012Content-Length: 0\\015\\012User-Agent: kamailio (3.3.3 (i386/linux))\\015\\012\\015\\012') Kamailio as of latest git on the 3.3 branc + 3e580e1ee24f9b9ec4b1fa151b23ebc84f3db1cb. Same warnings when linked with both PostgreSQL 8.4.13 and 9.1.1. According to Alexandr Dubovikov MySQL stores '\r\n' as '\r\n'. -- Øyvind Kolbu

1 0

Decode contact with user paramer
by Gareth Rylance 24 Jan '13

24 Jan '13

When using the siputils function encode_contact() on a message that contains a user parameter e.g. <sip:XXXXX00015;phone-context=national@10.YYY.1.92;user=phone>;tag=88930 it encodes fine e.g. <sip:natted_ua*sip:XXXXX00015;phone-context=national**192.168.242.102*5080*udp@10.YYY.70.100 > but decode_contact() errors due to the ";" (there is an explicit check for it in the code) I am currently working round this using $rU = $(rU{s.replace,;,:}); to remove the ; as the user part is not needed for my application. Would it be safe to remove the check for ";" ? Gareth

3 6

Kamailio Multi tenant as registrar server ony
by andrea mucci 24 Jan '13

24 Jan '13

Hi Gurus, i have a basic question. i have a SIP PBX server that don't have the Multi Tenant Feature or Multi Company Feature. i now that Kamailio support this feature, so is possible to use the kamailio as registrar server and redirect all the services, SUBSCRIBE, NOTIFY, and RTP to the SIP PBX with corrects headers? Thanks for your help

3 2

Asterisk and dispatcher
by Ian French 24 Jan '13

24 Jan '13

> > Hi, > I've been working my way through this tutorial ( > http://kb.asipto.com/asterisk:realtime:kamailio-3.0.x-asterisk-1.6.2-astdb) > and all it working well thanks. In the other benifits section you mention > that this configuration can be used with multiple instances of Asterisk. > I've configured my [TOASTERISK] route route[TOASTERISK] { ds_select_dst("1","4"); $sht(forw=>$ft)=$du; sl_send_reply("100","Trying"); xlog("L_INFO","INVITE: INVITE [$tU] Passed to [$du] Asterisk Cluster [$rm:$au].\n"); route(RELAY); exit; How can i achieve the same for my [FROMASTERISK] route as now with asterisk.bindip = "x.x.x.x" desc "Asterisk IP Address" asterisk.bindport = "5060" desc "Asterisk Port" and route[FROMASTERISK] { if($si==$sel(cfg_get.asterisk.bindip) && $sp==$sel(cfg_get.asterisk.bindport)) xlog("L_INFO","INVITE: INVITE Passed From [$fu] Asterisk Cluster To [$rm:$au].\n"); return 1; return -1; } This allows calls only from cfg_get.asterisk.bindip (single gateway) where I would like gateways in the dispatcher list to be selected and marked as a positive match if(ds_is_from_list("1")) $sht(forw=>$ft)=$si; xlog("L_INFO","INVITE: INVITE Passed From [$fu] Asterisk Cluster To [$rm:$au].\n"); return 1; return -1; } Can you help? Thanks in advance Ian

2 2

Guidelines to upgrade to 4.0.0
by Daniel-Constantin Mierla 23 Jan '13

23 Jan '13

Hello, the wiki page collecting the guidelines for upgrading from 3.3.x to 4.0.0 is available at: - http://www.kamailio.org/wiki/install/upgrade/3.3.x-to-4.0.0 I just added the SQL statements needed to upgrade the MySQL database, but there is lot of other stuff to be added. Hopefully, anyone noticing changes will add few notes and instructions about how to upgrade. The wiki can be edited if you make an account, it is not required to be a developer to add notes to this page. Would be great if users of postgres will add the sql statements specific for this database type. Thanks, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

1 0

Re: [SR-Users] sr-users Digest, Vol 92, Issue 50:Problem with 4.0 nightlies and IMS - sem_post undefined (Barry Flanagan)
by madhumanjusha＠integramicro.com 23 Jan '13

23 Jan '13

On Tue, January 22, 2013 1:58 pm, sr-users-request(a)lists.sip-router.org wrote: > Send sr-users mailing list submissions to > sr-users(a)lists.sip-router.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > or, via email, send a message with subject or body 'help' to > sr-users-request(a)lists.sip-router.org > > You can reach the person managing the list at > sr-users-owner(a)lists.sip-router.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of sr-users digest..." > > > Today's Topics: > > > 1. Modify IP address in SDP (Austin Einter) > 2. Re: Modify IP address in SDP (Olle E. Johansson) > 3. Problem with 4.0 nightlies and IMS - sem_post undefined > (Barry Flanagan) > 4. Patch to resolve segfault in json module (Richard Brady) > 5. Re: Modify IP address in SDP (Austin Einter) > 6. Re: Patch to resolve segfault in json module > (Daniel-Constantin Mierla) > 7. Re: [sr-dev] FOSDEM (Daniel Pocock) > > > > ---------------------------------------------------------------------- > > > > > > > > > > > Message: 3 > Date: Tue, 22 Jan 2013 13:19:44 +0000 > From: Barry Flanagan <barry(a)flanagan.ie> > Subject: [SR-Users] Problem with 4.0 nightlies and IMS - sem_post > undefined To: sr-users(a)lists.sip-router.org > Message-ID: > <CALJb54-rPPbjgGR5iEKYztNGDvT11znthJzgMYovpp_t6N0S8w(a)mail.gmail.com> > Content-Type: text/plain; charset="utf-8" > > > Hi > > > I am trying to run the Ubuntu nightlies from > http://deb.kamailio.org/kamailiodev-nightly and loading the p-cscf config > files. > > Things were working fine while I was on the 20130110 nightlies, but when > I > upgraded to 20130122 I get the following errors. Any have an idea why > sem_post is undefined and how to fix this? > > > > 0(1980) ERROR: <core> [sr_module.c:572]: ERROR: load_module: could not > open module </usr/lib64/kamailio/modules/ims_registrar_pcscf.so>: > /usr/lib64/kamailio/modules/ims_registrar_pcscf.so: undefined symbol: > sem_post 0(1980) ERROR: <core> [sr_module.c:572]: ERROR: load_module: could > not open module </usr/lib64/kamailio/modules/cdp.so>: > /usr/lib64/kamailio/modules/cdp.so: undefined symbol: sem_post > 0(1980) ERROR: <core> [sr_module.c:572]: ERROR: load_module: could not > open module </usr/lib64/kamailio/modules/ims_qos.so>: > /usr/lib64/kamailio/modules/ims_qos.so: undefined symbol: sem_post > > > Thanks > > > -Barry Flanagan >

2 1

Modify IP address in SDP
by Austin Einter 23 Jan '13

23 Jan '13

Dear All When 200 OK to INVITE passes through Kamailio proxy , I want to change the IP address in SDP to a specific IP address. Can this be done by just modifying configuration files or do I need to modify in source code. In case I need to modify source code, which files I need to look at. Kindly provide pointers for said problem. Thanks Austin

3 7

FOSDEM
by Peter Dunkley 22 Jan '13

22 Jan '13

Hello, Are there any plans for a Kamailio meal/get together on the Saturday evening this year? Peter

9 19

Patch to resolve segfault in json module
by Richard Brady 22 Jan '13

22 Jan '13

Hi folks If an empty or invalid JSON string is passed to the json_get_field function from the the json module, it causes a segfault. I have attached a patch. Regards, Richard -- Richard Brady M: +44 (0)7771 623 348 T: +44 (0)20 8144 8160 E: rnbrady(a)gmail.com

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Re: [SR-Users] Kamailio as a front-end for TLS-UDP
by Daniel-Constantin Mierla 22 Jan '13

22 Jan '13

Do no send replies to private address only, keep mailing list otherwise future such emails will be discarded. The contact has to stay the same, because it needs to be the address of the device. Record-Route and Path are designed to take care of routing back to lb, which will use tls to send to device. Cheers, Daniel On 1/22/13 12:26 PM, Anton Kvashenkin wrote: > Thanks for reply. Yes, kamailio acts as a balancer sending REGISTER > and INVITES, etc. The problem is that at L3-L4 it works well (sends > all datagram messages to 10.0.0.1:5060 <http://10.0.0.1:5060> UDP) but > at L7 the Contact header: <>;transport=tls. > > > 2013/1/22 Daniel-Constantin Mierla <miconda(a)gmail.com > <mailto:miconda@gmail.com>> > > Hello, > > > On 1/22/13 10:56 AM, Anton Kvashenkin wrote: >> Guys, I've already asked @ G+ Kamailio coummunity but had no >> answer so I'm asking here: >> >> I'm trying to deploy +Kamailio >> <https://plus.google.com/114984922071338017631> as a front-end >> for TLS-UDP translation, as a back-end would be +FreeSWITCH >> <https://plus.google.com/104108080382960591162> . So, is kamailio >> so smart to fix Contact or Path header for ;transport=tls or it >> translates only on L3-L4 layer and I should fix those headers >> manually? > is it only for calls or you need to forward registrations to > kamailio as well? > > Everything is done automatically, for calls you just need to do > record routing. For registrations you have to use path module and > be sure that freeswitch supports sip path extension. > > Cheers, > Daniel > > -- > Daniel-Constantin Mierla -http://www.asipto.com > http://twitter.com/#!/miconda <http://twitter.com/#%21/miconda> -http://www.linkedin.com/in/miconda > Kamailio World Conference, April 16-17, 2013, Berlin > -http://conference.kamailio.com - > > -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

1 0

Kamailio as a front-end for TLS-UDP
by Anton Kvashenkin 22 Jan '13

22 Jan '13

Guys, I've already asked @ G+ Kamailio coummunity but had no answer so I'm asking here: I'm trying to deploy +Kamailio<https://plus.google.com/114984922071338017631> as a front-end for TLS-UDP translation, as a back-end would be +FreeSWITCH<https://plus.google.com/104108080382960591162> . So, is kamailio so smart to fix Contact or Path header for ;transport=tls or it translates only on L3-L4 layer and I should fix those headers manually?

2 1

VoIP diner at FOSDEM 2013
by Benjamin Henrion 22 Jan '13

22 Jan '13

Hi, I am organizing a diner for all Voice over IP hackers (Asterisk, Freeswitch, Kamailio, OpenSER, Sipvicious :-), etc…) during the Saturday evening of FOSDEM 2013. There is a complete track at the Telephony devroom as well: https://fosdem.org/2013/schedule/track/telephony/ More infos are registration: http://www.zoobab.com/voip-diner-fosdem Best, -- Benjamin Henrion <bhenrion at ffii.org> FFII Brussels - +32-484-566109 - +32-2-3500762 "In July 2005, after several failed attempts to legalise software patents in Europe, the patent establishment changed its strategy. Instead of explicitly seeking to sanction the patentability of software, they are now seeking to create a central European patent court, which would establish and enforce patentability rules in their favor, without any possibility of correction by competing courts or democratically elected legislators."

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Wiki page for what is new in v4.0.0
by Daniel-Constantin Mierla 22 Jan '13

22 Jan '13

Hello, there is a skeleton content for what is new in upcoming v4.0.0, but there are many new features missing in the lists. Please try to add short notes about the features you know they were added for 4.0. The wiki page is available at: * http://www.kamailio.org/wiki/features/new-in-devel Anyone that makes an account can edit: * https://www.kamailio.org/wiki/start?do=register Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

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PATH advertised
by Olle E. Johansson 22 Jan '13

22 Jan '13

Hi! If I'm running Kamailio behind NAT and need to add a Path header with the outside IP address and port, that is configured as the "advertise" address - how do I do that? Can I add two Path headers at the same time? I can't find a way to add it like I can with Record-Route headers. /O

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Re: [SR-Users] sr-users Digest, Vol 92, Issue 43
by madhumanjusha＠integramicro.com 22 Jan '13

22 Jan '13

Thanks a lot for quick response. On Sat, January 19, 2013 3:59 am, sr-users-request(a)lists.sip-router.org wrote: > Send sr-users mailing list submissions to > sr-users(a)lists.sip-router.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > or, via email, send a message with subject or body 'help' to > sr-users-request(a)lists.sip-router.org > > You can reach the person managing the list at > sr-users-owner(a)lists.sip-router.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of sr-users digest..." > > > Today's Topics: > > > 1. Mid call announcement(kamailio server) > (madhumanjusha(a)integramicro.com) > 2. Re: Mid call announcement(kamailio server) > (rabs(a)dimension-virtual.com) > 3. CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus > dlg - Kamailio 3.3.x (Rinor Hoxha) 4. Huge undertaking and in need of > advices from the community (Rumen Mihailov) > 5. Re: Huge undertaking and in need of advices from the > community (Stoyan Mihaylov) 6. Re: Invite - negotiate to asterisk as peer > not kamailio (Scott, Matt) > > > > ---------------------------------------------------------------------- > > > Message: 1 > Date: Fri, 18 Jan 2013 06:52:15 -0500 (EST) > From: madhumanjusha(a)integramicro.com > Subject: [SR-Users] Mid call announcement(kamailio server) > To: sr-users(a)lists.sip-router.org > Message-ID: > <39960.61.8.152.138.1358509935.squirrel(a)mail.integramicro.com> > Content-Type: text/plain;charset=iso-8859-1 > > > Hello, > As of now am working on Kamailio server. > My task is to announce in between call,my sip clients are EKIGA. > I should able to make my kamailio server act as controller and media > server,for that purpose my first step is : invite hold to called party. > second step is:: invite(sdp for .wav file) to caller and then am sending > rtp streaming.In wireshark am able to see rtp packets,but caller is not > playing the .wav file which am streaming through ortp library from > kamailio. > > Can anyone help me out with this problem?On top of it,is this possible in > reality? > > > Thanks & Regards, > Manjusha A. > Integra Micro Software Services (P) Ltd. > > > > > > > > ------------------------------ > > > Message: 2 > Date: Fri, 18 Jan 2013 15:14:49 +0100 (CET) > From: rabs(a)dimension-virtual.com > Subject: Re: [SR-Users] Mid call announcement(kamailio server) > To: "SIP Router - Kamailio $OpenSER$ and SIP Express Router $SER$ > - Users Mailing List" <sr-users(a)lists.sip-router.org> > Message-ID: > <38001.212.40.242.42.1358518489.squirrel(a)www.rodriguezfeo.es> > Content-Type: text/plain;charset=iso-8859-1 > > >> Hello, >> As of now am working on Kamailio server. >> My task is to announce in between call,my sip clients are EKIGA. >> I should able to make my kamailio server act as controller and media >> server,for that purpose my first step is : invite hold to called party. >> second step is:: invite(sdp for .wav file) to caller and then am >> sending rtp streaming.In wireshark am able to see rtp packets,but caller >> is not playing the .wav file which am streaming through ortp library >> from kamailio. >> >> Can anyone help me out with this problem?On top of it,is this possible >> in reality? > > No, it's not possible using only kamailio, and thats because kamailio > it's a SIP proxy, it have nothing to do with media, for what you are > triying to get you need to router your calls throught a B2BUA, like sems > or asterisk. > > Best regards > > > > > > ------------------------------ > > > Message: 3 > Date: Fri, 18 Jan 2013 14:11:05 +0100 > From: Rinor Hoxha <rinorhoxha(a)gmail.com> > Subject: [SR-Users] CRITICAL: dialog [dlg_timer.c:205]: Trying to > update a bogus dlg - Kamailio 3.3.x To: sr-users(a)lists.sip-router.org > Message-ID: > <CAFvba4ipKqicMJ0qn9UNLXmhjGhf_83fzG=L5w2qVBpc7JC1GQ(a)mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > > Hi list, > > > I'm having some issues with dialog module. From time to time I get the > following error: > > ERROR: dialog [dlg_req_within.c:231]: failed to update dialog lifetime > CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg > tl=0x2aeeea1c6840 tl->next=(nil) tl->prev=(nil) > > [root@proxy ~]# cat /var/log/kamailio.log | grep "Jan 17" | grep "tl=" | > sort | uniq -c > > 1 Jan 17 13:27:00 proxy /usr/local/kamailio/sbin/kamailio[4100]: > CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg > tl=0x2aeeea09bf38 tl->next=(nil) tl->prev=(nil) 1 Jan 17 14:39:30 proxy > /usr/local/kamailio/sbin/kamailio[4100]: > CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg > tl=0x2aeeea07c518 tl->next=(nil) tl->prev=(nil) 1 Jan 17 15:06:21 proxy > /usr/local/kamailio/sbin/kamailio[4100]: > CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg > tl=0x2aeeea2da900 tl->next=(nil) tl->prev=(nil) 1 Jan 17 19:31:02 proxy > /usr/local/kamailio/sbin/kamailio[4100]: > CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg > tl=0x2aeeea278cd0 tl->next=(nil) tl->prev=(nil) 1 Jan 17 19:31:42 proxy > /usr/local/kamailio/sbin/kamailio[4100]: > CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg > tl=0x2aeee9f527a8 tl->next=(nil) tl->prev=(nil) 1 Jan 17 20:05:32 proxy > /usr/local/kamailio/sbin/kamailio[4100]: > CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg > tl=0x2aeeea21fa30 tl->next=(nil) tl->prev=(nil) > > This one was flooding my log file: tl=0x2aeeea1c6840 > > > 24322 Jan 17 21:53:02 proxy /usr/local/kamailio/sbin/kamailio[4100]: > CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg > tl=0x2aeeea1c6840 tl->next=(nil) tl->prev=(nil) 5709 Jan 17 21:53:23 proxy > /usr/local/kamailio/sbin/kamailio[4100]: > CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg > tl=0x2aeeea1c6840 tl->next=(nil) tl->prev=(nil) > > 61314 Jan 17 21:53:23 proxy /usr/local/kamailio/sbin/kamailio[4102]: > CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg > tl=0x2aeeea1c6840 tl->next=(nil) tl->prev=(nil) 22812 Jan 17 21:54:01 proxy > /usr/local/kamailio/sbin/kamailio[4102]: > CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg > tl=0x2aeeea1c6840 tl->next=(nil) tl->prev=(nil) 21137 Jan 17 21:54:02 proxy > /usr/local/kamailio/sbin/kamailio[4102]: > CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg > tl=0x2aeeea1c6840 tl->next=(nil) tl->prev=(nil) > > where pid 4100 is slow timer and pid 4102 is timer. Any idea why is this > happening and how to fix it. (I suspect that may be some issues with our > server memory...checking now...and will inform). However wanted to raise > this issue in case anyone else is having the same. > > Thanks, Rinor >

1 0

Kamailio Development Workshop, Feb 14-15, 2013, Alicante, Spain
by Daniel-Constantin Mierla 21 Jan '13

21 Jan '13

Hello, I want to announce the next Kamailio Development Workshop, in Alicante, Spain, during Feb 14-15, 2013, organized by Asipto and University of Alicante, Department of Master Degrees in Telecommunication Engineering. You can find more details at: - http://www.kamailio.org/w/2013/01/kamailio-development-workshop-feb-14-15-2… Like the previous one of its kind (done in 2011 in Barcelona), this type of workshop targets to explain the source code structure and internal APIs of Kamailio, to speed up the learning of how to write own Kamailio extensions in C (e.g., write a module, which will be part also of the content). Its low registration fee is set to cover organizational logistics, the number of seats is limited, participants are accepted in the order of registration. Besides the workshop during the two days, on the evening of Feb 14 is planned a VoIP dinner. You can participate to it even if not coming to workshop. It will be nice to meet face to face and talk about latest trends in RTC/UC. If you need further details, contact me via email. Regards, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

1 0

all modules in one folder
by Daniel-Constantin Mierla 20 Jan '13

20 Jan '13

Hello, please note that all modules are now back in one folder, respectively 'modules/'. There were few modules I couldn't compile due to missing exotic libraries, but based on grep, they should not depend on older modules paths. Anyhow, testing that the set of modules you are using compile and load file would be recommended. The folders modules_k and modules_s are now empty, with a placeholder .gitkeep, to be removed later if there is no need to revert any module relocation. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

1 0

updates to perl modules
by Daniel-Constantin Mierla 20 Jan '13

20 Jan '13

Hello, there were several changes pushed to master branch related to perl modules: - perl was renamed to app_perl - to follow the naming pattern for embedded interpreters - embedded perl library is named now Kamailio (it was OpenSER so far, thanks to patches from Stoyan Mihaylov) - perlvdb was renamed to db_perlvd because it is a db driver module If you are using these modules, you will have to update your configuration file and perl scripts. Testing and feedback is appreciated. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

1 0

renaming to kamailio in perl and snmpstats modules
by Daniel-Constantin Mierla 20 Jan '13

20 Jan '13

Hello, I'm writing to see if anyone has some spare time to take care of renaming from openser to kamailio in perl and snmpstats modules. I assume is more like search and replace, but has to be carefully done to be coherent and not break stuff there, so it might be a bit more than a sed command. The IANA enterprise number in MIBs is already the one for kamailio.org Thanks, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

3 7

Testdrive the coming release 4.0 now!
by Olle E. Johansson 19 Jan '13

19 Jan '13

http://www.kamailio.org/w/2013/01/test-4-0-now/ Friends, We do need your help testing 4.0 now. - Compile on your systems, from small embedded systems to large servers, from Linux to *BSD and Solaris - Use with your configuration script - Test the new modules and functions - Read the updated documentation files On the link above, you have instructions on how to build from GIT sources and install a test system. We do look forward to your feedback! Let's all work to make this a very, very good and succesful release! Regards, /O ---- Kamailio on Twitter: @kamailioproject (also on Facebook and Google plus)

1 0

Huge undertaking and in need of advices from the community
by Rumen Mihailov 18 Jan '13

18 Jan '13

Hi all, Thank you for all the help till now. Until now I was trying to setup completely redundant setup with Kamailio + asterisk boxes and for the moment i have the following configuration working. 2x Kamailios (Active/Passive) 2x Asterisks Kamailio is load balancing I can add more boxes if needed MySQL Cluster for all database needs Now the time has come and I will need to made this setup available for around 10K users. I will need to become an ITSP for an Internet Provider that is missing the telephony part from his package, so I have a couple of questions. 1. Is this setup OK to cover the needs of the customer base ? hardware will be decent 2. The end users will most probably need ATA adapters. What is the best cost effective solution I can go for ? 3. Do I need any other piece of software apart from the SIP Proxy and asterisk as I need the cost of implementation to be as low as possible. 4. Pricing...I can see that there are a lot of plans of the competitors that include 1000 minutes for free...Do you have any statistics what is the avarage % usage of prepaid minutes ? 5. Are there any other "stones" that I might hit and should be aware of ? Thank you for the information Best regards, Rumen

2 1

Merry Christmas!
by Daniel-Constantin Mierla 18 Jan '13

18 Jan '13

Another year getting to its end! Looking back, looks like one with the greatest achievements in the development of the project so far. Year is not done, so I save that summary for one week later, there is still stuff on its way to our source code repository. Now I want to thank to everyone promoting and contributing to the project, from developers to community members, and wish Merry Christmas and great winter holidays to all supporters of our project! Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

8 8

Mid call announcement(kamailio server)
by madhumanjusha＠integramicro.com 18 Jan '13

18 Jan '13

Hello, As of now am working on Kamailio server. My task is to announce in between call,my sip clients are EKIGA. I should able to make my kamailio server act as controller and media server,for that purpose my first step is : invite hold to called party. second step is:: invite(sdp for .wav file) to caller and then am sending rtp streaming.In wireshark am able to see rtp packets,but caller is not playing the .wav file which am streaming through ortp library from kamailio. Can anyone help me out with this problem?On top of it,is this possible in reality? Thanks & Regards, Manjusha A. Integra Micro Software Services (P) Ltd.

2 1

HEAD: Problems with $$ in xlog
by Klaus Darilion 17 Jan '13

17 Jan '13

Hi! In xlog I used to use $$ to escape the dollar sign to not be interpreted as PV, eg: xlog("L_INFO"," $$ru\n"); Current HEAD gives an error: ERROR: <core> [pvapi.c:757]: error searching pvar "" ERROR: <core> [pvapi.c:961]: wrong char [$/36] in [$] at [0 (0)] ERROR: xlog [xlog.c:363]: wrong format[ $$ru#012] ERROR: <core> [route.c:1214]: fixing failed (code=-1) at cfg:/etc/kamailio/kamailio.cfg:627 Where there some changes? Thanks Klaus

2 2

Simple command to insert attributes to domain_attrs
by Philippe Sultan 17 Jan '13

17 Jan '13

Hi, I'm migrating from sip-router/Kamailio 3.1, SER flavour to the latest dev version, with MySQL support. I need to add attributes to domains, and therefore use the domain_attrs table. A handy command I used for that was ser_attr : ser_attr add domain=anydomain.voip attr=value Is there something similar I can use under the Kamailio flavour? kamctl does not seem to help here, unless I missed an option. Regards, Philippe

3 5

Users can't authenticate via external IP
by Rumen Mihailov 17 Jan '13

17 Jan '13

Hello Kamailio guys, I have the following setup Internet -----> REAL STATIC IP -----> Kamailio Virtual IP(10.2.45.101) I have setup everything and when I try to connect to kamailio from the inside I can successfully register. When I try to register to the REAL STATIC IP which is forwarded to 10.2.45.101 I get unauthorized. I have narrowed down the problem to this: auth_check("$fd", "sip", "1") as I said I can authenticate from the internal network but not from the Internet. What do I need to alter ? Side problem... why when using xlog("L_WARN","$Cxx TEST") kamailio fails to start ? seems that my version does not support the colored output... Thank you for your help guys...in advance. Best regards, Rumen http://itradebg.com

2 1

Server sip.iptel.org did not respond to register
by Faisal Farooq 17 Jan '13

17 Jan '13

Dear Support, We are using IPTEL SIP services. Now we are getting Error message on our VOIP software is " Server sip.iptel.org did not respond to register". Moreover, we can successfully ping and complete trace route the sip.iptel.org from our server but failed to register through sip.iptel.org server. Kindly suggest what is the cause and look into this matter. I am sending you our IPTEL SIP ID/Username and please find below. (1) 196211 (2) 195404 Regards, Faisal

2 1

i cannot log on to iptel.org
by kelly randolph 17 Jan '13

17 Jan '13

i cannot log on to iptel.org is everything okay???? thanx, kellyr

2 1

cant login
by Pasindu Hettiachchi 17 Jan '13

17 Jan '13

hi , my linksys router failed to register with iptel.org this happened from last night, any think that I have to change ? Pasindu

2 1

Error while loading modules, if given a custom library path
by Ajay Sabat 16 Jan '13

16 Jan '13

Hi I have trouble loading modules from a custom lib directory. It works fine if the libraries are located under PREFIX/lib/Kamailio/ (PREFIX is the path that I gave while building Kamailio). But on the target I will have to keep the libraries in a custom directory. I tried by changing the mpath in kamailio.cfg. Also I tried by giving the path in -L option while invoking kamailio. It is not working. It fails with the following errors, even if the libraries are present at the location where it compains. I searched but couldn't find any answer why it is failing, though it is supposed to work. Please help me. $ ./sbin/kamailio -E -d -f /home/ajay/projects/video_conf/etc/kamailio/kamailio.cfg -c loading modules under /home/ajay/projects/video_conf/lib/kamailio/:/home/ajay/projects/video_conf/lib/kamailio/modules/:/home/ajay/projects/video_conf/lib/kamailio/modules_k/ 0(27896) ERROR: <core> [sr_module.c:572]: ERROR: load_module: could not open module </home/ajay/projects/video_conf/lib/kamailio/modules_k/mi_fifo.so>: libkmi.so.1: cannot open shared object file: No such file or directory 0(27896) : <core> [cfg.y:3591]: parse error in config file /home/ajay/projects/video_conf/etc/kamailio/kamailio.cfg, line 222, column 12-23: failed to load module 0(27896) ERROR: <core> [sr_module.c:572]: ERROR: load_module: could not open module </home/ajay/projects/video_conf/lib/kamailio/modules_k/kex.so>: libkmi.so.1: cannot open shared object file: No such file or directory 0(27896) : <core> [cfg.y:3591]: parse error in config file /home/ajay/projects/video_conf/etc/kamailio/kamailio.cfg, line 223, column 12-19: failed to load module 0(27896) ERROR: <core> [sr_module.c:572]: ERROR: load_module: could not open module </home/ajay/projects/video_conf/lib/kamailio/modules_k/tmx.so>: libkmi.so.1: cannot open shared object file: No such file or directory 0(27896) : <core> [cfg.y:3591]: parse error in config file /home/ajay/projects/video_conf/etc/kamailio/kamailio.cfg, line 225, column 12-19: failed to load module 0(27896) ERROR: <core> [sr_module.c:572]: ERROR: load_module: could not open module </home/ajay/projects/video_conf/lib/kamailio/modules/sl.so>: libkcore.so.1: cannot open shared object file: No such file or directory $ ls -l /home/ajay/projects/video_conf/lib/kamailio/modules_k/mi_fifo.so -rwxr-xr-x 1 ajay ajay 155651 Jan 16 09:58 /home/ajay/projects/video_conf/lib/kamailio/modules_k/mi_fifo.so $ ls -l /home/ajay/projects/video_conf/lib/kamailio/libkmi.so* lrwxrwxrwx 1 ajay ajay 13 Jan 16 09:56 /home/ajay/projects/video_conf/lib/kamailio/libkmi.so -> libkmi.so.1.0 lrwxrwxrwx 1 ajay ajay 13 Jan 16 09:56 /home/ajay/projects/video_conf/lib/kamailio/libkmi.so.1 -> libkmi.so.1.0 -rwxr-xr-x 1 ajay ajay 78111 Jan 16 09:56 /home/ajay/projects/video_conf/lib/kamailio/libkmi.so.1.0 Regards, Ajay

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make option to build a lean kamailio image
by Ajay Sabat 16 Jan '13

16 Jan '13

Hi, I need to build a simple Kamailio server that could do just registration and basic call routing. I don't need database support, it is fine if it loses data during reboot. The only other option I may need is NAT support. So far I was building the server using make all as given below: $ make FLAVOUR=kamailio PREFIX="$HOME/projects/video_conf/sipserver" cfg $make all $make install I noticed that the total size of library and binary files is huge, little less than 30MB. I have to port and test the server on a target where I don't have much space. Can somebody please suggest me what make option would be appropriate for me? Is it fine to not build the ser modules under modules_s/ and what is make option to do so? I tried "make" instead of "make all", but it somehow builds everything during the step "make install" and the final result is same. Thanks, Ajay

2 2

TLS questions
by Fabian Borot 15 Jan '13

15 Jan '13

Hi, we need to implement TLS in our current setup and have some questions about it: 1- We have a farm of Asterisk servers behind our kamailio proxy. The call flow is Customers/Providers <--> Proxy <--> Asterisks. Is it possible to have TLS only between the Customers/Providers and the Proxy and then just regular SIP over UDP between the Proxy and the Asterisks? 2- Can we have TLS over both TCP and UDP at the same time? Thank you a lot fborot

3 3

Dokuwiki upgraded
by Daniel-Constantin Mierla 15 Jan '13

15 Jan '13

Hello, fyi, I upgraded dokuwiki to latest version, switching to the new default template (which is not bad at all, btw) as the old one is not yet fully compatible. If you find any issue, report to devel mailing list. It is about the wiki for v3.2+: - http://www.kamailio.org/wiki/ Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

2 2

NAPTR for TLS only
by Daniel Pocock 14 Jan '13

14 Jan '13

I'm just wondering if anyone can comment on expected and actual behavior if there is only a NAPTR record for TLS, e.g. I have: sip5060.net. IN NAPTR 10 0 "s" "SIPS+D2T" "" _sips._tcp.sip5060.net. and I don't have any entry for "SIP+D2U" or "SIP+D2T" If some third party Kamailio instance (e.g. sip-server.example.org) receives a request from a user trying to call sip:user@sip5060.net, with a sip: rather than sips: URI, should it (and will it) use the "SIPS+D2T" result, if no other result is available? Or would it ignore the NAPTR record and try to find the default SRV record such as _sip._udp.sip5060.net ? Should there be another NAPTR record to translate sip: to sips: using a regex perhaps, or would such a NAPTR be a bad thing?

3 4

Kamailio, Asterisk with TLS+SRTP
by Roberto Fichera 14 Jan '13

14 Jan '13

Hi All, I would setup a configuration where Kamailio authenticate asterisk SIP trunk using TLS and SRTP. At moment I was able to configure everything, including RTTProxy since most of the asterisks v1.8.19.1 are behind NAT. So far so good it works pretty good using standard authentication and the call goes straight between asterisks. But as soon as I move my configuration for both kamailio & asterisk to TLS+SRTP I'm not able to authenticate asterisk SIP trunks. Especially asterisk seems insisting to use the port 5060 even if I requested the TLS on 5061. kamailio v3.3.3 tls.cfg is configured as: [server:default] method = TLSv1 verify_certificate = no require_certificate = no private_key = /etc/pki/tls/private/server.key certificate = /etc/pki/tls/certs/server.pem ca_list = /etc/pki/tls/certs/ca-bundle.crt #crl = //etc/kamailio/crl.pem # This is the default client domain, settings # in this domain will be used for all outgoing # TLS connections that do not match any other # client domain in this configuration file. # We require that servers present valid certificate. # [client:default] verify_certificate = no require_certificate = no So my asterisk conf is the following: [general] tlsenable=yes tlsbindaddr=0.0.0.0 tlscertfile=/etc/asterisk/5002.pem tlscafile=/etc/asterisk/ca-bundle.crt tlscipher=ALL tlsclientmethod=tlsv1 tlsdontverifyserver=yes transport=tls,udp .... ..... and the SIP trunk is configured as [kamailio] type=peer insecure=invite,port nat=yes disallow=all allow=ulaw host=kamailio_ip outboundproxy=tls://kamailio_ip port=5061 defaultuser=5002 fromuser = 5002 fromdomain =mydomain secret=5002 qualify=yes dtmfmode=rfc2833 context=default callbackextension=5002 directmedia=nonat sendrpid=yes transport=tls encryption=yes register => tls://5002:5002@kamailio_ip:5061/5002 I still get error like: Jan 14 10:45:12] ERROR[5244]: chan_sip.c:5600 create_addr_from_peer: 'UDP' is not a valid transport for 'dicenet'. we only use 'TLS'! ending call. [Jan 14 10:45:12] WARNING[5244]: chan_sip.c:13722 transmit_register: Probably a DNS error for registration to 5002@kamailio_ip, trying REGISTER again (after 20 seconds) [Jan 14 10:45:32] ERROR[5244]: chan_sip.c:5600 create_addr_from_peer: 'UDP' is not a valid transport for 'dicenet'. we only use 'TLS'! ending call. [Jan 14 10:45:32] WARNING[5244]: chan_sip.c:13722 transmit_register: Probably a DNS error for registration to 5002@kamailio_ip, trying REGISTER again (after 20 seconds) [Jan 14 10:45:52] ERROR[5244]: chan_sip.c:5600 create_addr_from_peer: 'UDP' is not a valid transport for 'dicenet'. we only use 'TLS'! ending call. [Jan 14 10:45:52] WARNING[5244]: chan_sip.c:13722 transmit_register: Probably a DNS error for registration to 5002@kamailio_ip, trying REGISTER again (after 20 seconds) [Jan 14 10:46:07] ERROR[7041]: tcptls.c:444 ast_tcptls_client_start: Unable to connect SIP socket to kamailio_ip:5060: Connection timed out Does anyone can suggest me something to read, try, check? Best regards. Roberto Fichera.

2 2

How to correctly check if the variable is set?
by Mino Haluz 14 Jan '13

14 Jan '13

Hi, how should I check if the value is set? if ($avp(s:test) == "") { or is there any null keyword ? If so, does it work for $avp, $sht, $var and $shv ? Thanks, Mino

3 4

Development is frozen for v4.0.0
by Daniel-Constantin Mierla 11 Jan '13

11 Jan '13

Hello, last midnight was the term to freeze the development for next major release v4.0.0. For a while, the GIT must branch should get only commits related to bug fixes, documentation improvements and updates of the additional tools. If any developer needs to store code of new features in the repository, use personal branches (or temporary branches in tmp/ when many developers work on the same code). After unfreeze, these branches can be merged in the master branch. For 1-2 months from now on, the main focus with the master branch is testing. The v4.0.0 will be out after mid of February, exact date to be set later based on the results during testing phase. There were many additions, so any kind of help testing is very appreciated. Fortunately the core and most of the very used classic modules were not changed much, ensuring good stability. But there are old very used pieces such as usrloc/registrar that got touched as well, so try to test as much as possible the components you are using in your configs. Provide any kind of feedback on sr-dev mailing list or the bug tracker. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

1 0

PV: PAI vs. PPI
by Klaus Darilion 11 Jan '13

11 Jan '13

Hi! I just found out that there is only $ai for the whole P-Asserted-Identity URI, but lots of PVs for P-Preferred-URI: uri, domainpart, userpart ... Is there a reason for that, or is it just that nobody implemented it for PAI? I think it should be easy to extend it for PAI too. regards Klaus

2 1

Kamailio World Conference - Registration is open
by Daniel-Constantin Mierla 11 Jan '13

11 Jan '13

Hello, the registration for Kamailio World Conference is now open! You can see more details and register at: - http://conference.kamailio.com/k01/registration/ There is already a great group of speakers and interesting proposed talks. More regarding the content will be published in the near future, keep an eye on event's web site: - http://conference.kamailio.com If you are considering to speak at the conference, submit your proposal as soon as possible, the slots are filling up quickly. Looking forward to meeting many of you at the conference! Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

3 3

Kamailio on Twitter
by Olle E. Johansson 11 Jan '13

11 Jan '13

Hello! For those of you that have missed it: You can now follow the project on Twitter: @kamailioproject https://twitter.com/kamailioproject Amongst the latest news are Kamailio at Fosdem 2013, Code freeze for kamailio 4.0 and the Kamailio World conference now open for registrations. Tonight we went into code freeze for the coming release of Kamailio 4.0. This means that developers can not add any more features to this release, only fix bugs. In order to fix bugs we will need your help testing. After the weekend we'll come back with instructions on how to test and report bugs. The information is already available on the web site. Have a great weekend! /O

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