- sr-users - kamailio.org

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

26 Jan '11

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

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

25 Jan '11

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

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fix_nated_sdp()
by Bernhard Suttner 25 Jan '11

25 Jan '11

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

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

25 Jan '11

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

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

25 Jan '11

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

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

25 Jan '11

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

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

25 Jan '11

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

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Problem reloading carrierroute
by Morten Isaksen 25 Jan '11

25 Jan '11

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

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Building specific commit
by mayamatakeshi 25 Jan '11

25 Jan '11

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

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

24 Jan '11

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

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

24 Jan '11

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

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Multiple Sip
by aram amin 24 Jan '11

24 Jan '11

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

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pdb module best practice
by Thomas Baumann 24 Jan '11

24 Jan '11

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

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pdb module best practice
by Thomas Baumann 22 Jan '11

22 Jan '11

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Problem with parallel forking
by Daniel Grotti 21 Jan '11

21 Jan '11

Hi all, I'm using kamailio 3.1 and I have some problems with parallel forking. I need to implement parallel forking to different users registered on kamailio. So, when call arrives with R-URI= sip:003912345678@<IP_server>, I need to fork te call to (for example) 2 users: 1001 and 1001. To do that, I've created my usr_preferences table like this: +----+------+------------------+--------+-----------+------+-------+---------------------+ |/ id | uuid | username | domain | attribute | type | value | /last_modified | +----+------+------------------+--------+-----------+------+-------+---------------------+ |/ 1 | | /003912345678/ | | fork | 0 | 1001 | | /|/ 2 | | /003912345678/ | | fork | 0 | 1002 | | /+----+------+------------------+--------+-----------+------+-------+---------------------+ and I've added a code to my kam.cfg like this: if (is_method("INVITE")) { xlog("L_INFO", "REQUEST Invite - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); if(avp_db_load("$ru/username","$avp(s:fork)")) { avp_pushto("$ru/username","$avp(s:fork)/g"); } if(!lookup("location")) { xlog("L_INFO", "Local user offline - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("404", "User Offline"); exit; } else { xlog("L_INFO", "Local user online - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); t_relay(); } } If 1001 and 1002 are registered everything works fine (1001 Ringing and 1002 ringing). Ifonly 1001 registered everything works fine (1001 ringing and 1002 is offline.). But when 1002 is registered and 1001 in offline, kamailio try to call 1001, find that it's offline ( and I get 404 User Offline") but no call to 1002 is attempted. What's wrong ? Regards, Daniel

2 2

Too many false positives?
by Jeremya 21 Jan '11

21 Jan '11

A large bunch of recent email on this list got consigned to the spam bin by my Thunderbird. This hasn't happened before. Is there some list change that triggered this? Regards Jeremy A

2 1

Proxying INVITE message without port information in R-URI
by Vikram Ragukumar 20 Jan '11

20 Jan '11

Hello, The diagram below shows the SIP message flow in our setup when Softphone A places a call to Softphone B. ----------- ----- ---------- ----- ----------- |Softphone A|-<->-|Proxy|-<->-|Sip Server|-<->-|Proxy|-<->-|Softphone B| ----------- ----- ---------- ----- ----------- IP: a.b.c.d IP: a.b.c.d Proxies shown in the diagram above are physically the same entity. For convenience of diagrammatic representation the figure above has been unwrapped. Both Softphone A and Softphone B (hereafter referred to as A,B) are behind a NAT and are registered to the Sip Server through the same proxy (IP: a.b.c.d). Proxy runs Kamailio v3.0.1. When A places a call to B, an INVITE is proxied to the Sip Server from A. The Sip Server now sends another INVITE to the proxy, however the R-URI of this INVITE does not contain the port number that the INVITE is supposed to be sent to. Hence the proxy sends the INVITE to the default SIP port 5060. This port is not open in the firewall hence Softphone B never receives the INVITE. My questions are 1) Is it mandatory for R-URI of the INVITE to contain the port number ? 2) If not, is there something that can be added to the config script in Kamailio so that the proxy forwards the INVITE to the appropriate port? (maybe by keeping track of the rcv port of the REGISTER message that Softphone B sends...) ? Thanks and Regards, Vikram. PS : The Sip Server in the above discussion is Voipswitch.

3 10

Re: [SR-Users] Refer Using UAC.
by Daniel-Constantin Mierla 20 Jan '11

20 Jan '11

private mails are simply ignored after first advise in this regard, please CC the mailing list always. If you read the config from the tutorial, you see how the invite is relayed to asterisk. Refer should go to asterisk in the same way if it is an out of dialog request, or follow record route/contact address for within dialog requests. Cheers, Daniel On 1/20/11 11:37 AM, David J. wrote: > could you point me to the docs? > > just use forward() or rewritehostport()? > > > > On 1/20/11 5:15 AM, Daniel-Constantin Mierla wrote: >> If asterisk is in the path of the call, then just forward the REFER >> to it, there is no need to generate a new one. >> >> Also, note that REFER is many times part of a dialog, uac_req_send() >> creates requests out of the dialog. >> >> Cheers, >> Daniel >> >> On 1/16/11 11:37 PM, David J. wrote: >>> I realize that kamailio is not a b2bua; >>> But because we are using Asterisk in the path; >>> >>> To extend the Asterisk Realtime Tutorial; >>> >>> I was wondering if I could do something like this... >>> >>> Kind of like how we use UAC to send a register to Asterisk; >>> Could we do the same and modify the method to use REFER instead? >>> >>> I know it is more complex; but I am not sure where to handle this case; >>> >>> Thanks for any pointers. >>> >>> if(is_method("REFER")){ >>> $var(rip) = $sel(cfg_get.asterisk.bindip); >>> $uac_req(method)="REFER"; >>> $uac_req(ruri)="sip:" + $var(rip) + ":" + >>> $sel(cfg_get.asterisk.bindport); >>> $uac_req(furi)="sip:" + $au + "@" + $var(rip) + ";tag=" + $ft; >>> $uac_req(turi)="sip:" + $au + "@" + $var(rip) + ";tag=" + $tt; >>> $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(); >>> >>> >>> } >>> >>> _______________________________________________ >>> 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 Kamailio (OpenSER) Advanced Training Jan 24-26, 2011, Irvine, CA, USA http://www.asipto.com

1 0

SL local ACK found -> dropping it! error ...
by Sébastien Cramatte 20 Jan '11

20 Jan '11

Not sure how todo this SIP capture. Can you give me an hand and explain how can I get this trace. I can post my configuration if it's necesary. But I'use the default NAT setup that comes with Debian package. 2011/1/19 Alex Balashov <abalashov(a)evaristesys.com> You'd need to post the SIP capture, but it sounds like Kamailio is not > associating the ACK with any known transaction, perhaps because the ACK has > an unexpected RURI, Via branch parameter, or CSeq number. > > -- > Alex Balashov - Principal > Evariste Systems LLC > 260 Peachtree Street NW > Suite 2200 > Atlanta, GA 30303 > Tel: +1-678-954-0670 > Fax: +1-404-961-1892 > Web: http://www.evaristesys.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

Handling call transfer in the Asterisk Realtime setup.
by David J. 20 Jan '11

20 Jan '11

I am trying to add support for call transfer in the Asterisk realtime tutorial on Asipto; I am not sure what I would have to do to get this feature working; Perhaps I have to handle "refer" messages; but I am not sure how I send that to Asterisk; Any advice would be greatly appreciated.

2 1

Refer Using UAC.
by David J. 20 Jan '11

20 Jan '11

I realize that kamailio is not a b2bua; But because we are using Asterisk in the path; To extend the Asterisk Realtime Tutorial; I was wondering if I could do something like this... Kind of like how we use UAC to send a register to Asterisk; Could we do the same and modify the method to use REFER instead? I know it is more complex; but I am not sure where to handle this case; Thanks for any pointers. if(is_method("REFER")){ $var(rip) = $sel(cfg_get.asterisk.bindip); $uac_req(method)="REFER"; $uac_req(ruri)="sip:" + $var(rip) + ":" + $sel(cfg_get.asterisk.bindport); $uac_req(furi)="sip:" + $au + "@" + $var(rip) + ";tag=" + $ft; $uac_req(turi)="sip:" + $au + "@" + $var(rip) + ";tag=" + $tt; $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(); }

2 1

Reconnection to db
by Javier Gallart 20 Jan '11

20 Jan '11

Hi list we have been testing the mtree module. We do periodical reloads of the mtree table with the "mi mt_reload" command through xmlrpc. It works fine except when the db is restarted. If we manually restart the db and then try the mt_reload commands we get this error: Jan 4 09:13:17 server /usr/local/kamailio-3.1.1/sbin/kamailio[28534]: ERROR: db_postgres [km_dbase.c:152]: 0xa6a490 PQsendQuery Error: server closed the connection unexpectedly#012#011This probably means the server terminated abnormally#012#011before or while processing the request.#012 Query: select partition,prefix,route from trees order by partition Jan 4 09:13:17 server /usr/local/kamailio-3.1.1/sbin/kamailio[28534]: ERROR: <core> [db_query.c:101]: error while submitting query Jan 4 09:13:17 server /usr/local/kamailio-3.1.1/sbin/kamailio[28534]: ERROR: mtree [mtree_mod.c:622]: Error while querying db Jan 4 09:13:17 server /usr/local/kamailio-3.1.1/sbin/kamailio[28534]: ERROR: mtree [mtree_mod.c:738]: cannot re-load info from database Is there a way to reconnect to the db before sending that command? Thanks in advance Regards Javi

3 15

avp_db_query() question
by 侯旭光 19 Jan '11

19 Jan '11

Hello avp_db_query(query[,dest]) can get a database query and store the results in the avps. But what if the results returns many rows,and how can I get all the results? How to set the [dest] parameter ? I've tried the method describered inhttp://www.kamailio.org/docs/avp_db_query.html,but it doesn't work. like below mysql<select mem_user from tgroup where grp_name='1234'; +----------+ | mem_user | +----------+ | 1013 | | 2013 | +----------+ 2 rows in set (0.00 sec) kamailio.cfg if(avp_db_query("select mem_user from tgroup where grp_name='1234'","$avp(name)")) { xlog("L_INFO","query results[1] :$avp(name[1])\n"); xlog("L_INFO","query results[2] :$avp(name[2])\n"); } syslog <INFO> query results[1] :<null> <INFO> query results[2] :<null> version: kamailio 3.0.2 MySQL 5.0 thank you very much!

2 1

SL local ACK found -> dropping it! error ...
by Sébastien Cramatte 19 Jan '11

19 Jan '11

Hello, I'm new with Kamailio. I've just setup on Debian Squeeze using mostly the example configuration. I've been able to route inbound / outbound call on local networks W/O NAT and over mobile 3G+NAT ... Unfortunateley I've tried over 3 wifi / local + NAT and Kamailio drop call and return me this message "SL local ACK found -> dropping it! error ..." My Kamailio server has public fixed IP and listen on 5060:udp It's look and issue with NAT setup ... How can we solve it ? Best regards

3 3

Return code after fr_inv_timer hit
by Mino Haluz 18 Jan '11

18 Jan '11

Hi, I would like to force kamailio to send another code as Request timeout when fr invite timeout is hit. Is there some nice way how to achieve it, or I have to edit the code ? :( Thank you!

3 7

ACK problem with FreeSwitch-Kamailio SBC implementation.
by Ricardo Martinez 18 Jan '11

18 Jan '11

Hello. I was wondering if someone can help me with some problem that I’m having with the combination of FreeSwitch and Kamailio as SBC. I’m having and ACK not routed by kamailio and dropped. Here is the debug. 100.100.100.211:5060 : Kamailio 100.100.100.211:5090 : FreeSwitch 100.100.100.108 : Caller 100.100.100.212 : Calle Request-Line: INVITE sip:02131234@100.100.100.211<sip%3A02131234(a)100.100.100.211>SIP/2.0 Method: INVITE [Resent Packet: False] Message Header Via: SIP/2.0/UDP 64.76.148.108:42772 ;branch=z9hG4bK-d8754z-da0b2e4b1b5d0843-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:1234@64.76.148.108:42772> To: <sip:02131234@100.100.100.211 <sip%3A02131234(a)100.100.100.211>> From: "camila"<sip:1234@100.100.100.211 <sip%3A1234(a)100.100.100.211> >;tag=8422566b Call-ID: MjY2NDY5NTE3OTk4MzdmMjE0YWQxNWQ4MWFmY2JhMWQ. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: Bria release 2.2 stamp 45414 Content-Length: 388 Message Body Session Description Protocol Status-Line: SIP/2.0 100 trying -- your call is important to us Status-Code: 100 [Resent Packet: False] Message Header Via: SIP/2.0/UDP 64.76.148.108:42772 ;branch=z9hG4bK-d8754z-da0b2e4b1b5d0843-1---d8754z-;rport=42772 To: <sip:02131234@100.100.100.211 <sip%3A02131234(a)100.100.100.211>> From: "camila"<sip:1234@100.100.100.211 <sip%3A1234(a)100.100.100.211> >;tag=8422566b Call-ID: MjY2NDY5NTE3OTk4MzdmMjE0YWQxNWQ4MWFmY2JhMWQ. CSeq: 1 INVITE Server: kamailio (3.1.1 (i386/linux)) Content-Length: 0 Status-Line: SIP/2.0 200 OK Status-Code: 200 [Resent Packet: False] Message Header Via: SIP/2.0/UDP 64.76.148.108:42772 ;branch=z9hG4bK-d8754z-da0b2e4b1b5d0843-1---d8754z-;rport=42772 Record-Route: <sip:100.100.100.211;lr=on> From: "camila" <sip:1234@100.100.100.211<sip%3A1234(a)100.100.100.211> >;tag=8422566b To: <sip:02131234@100.100.100.211 <sip%3A02131234(a)100.100.100.211> >;tag=pBZjZayFUeDSp Call-ID: MjY2NDY5NTE3OTk4MzdmMjE0YWQxNWQ4MWFmY2JhMWQ. CSeq: 1 INVITE Contact: <sip:02131234@100.100.100.211:5090;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-d41e649 2011-01-11 12-25-55 -0600 Accept: application/sdp Message Body Session Description Protocol Request-Line: ACK sip:02131234@100.100.100.211:5090;transport=udp SIP/2.0 Method: ACK [Resent Packet: False] Message Header Via: SIP/2.0/UDP 64.76.148.108:42772 ;branch=z9hG4bK-d8754z-182d537bcb534165-1---d8754z-;rport Max-Forwards: 70 Route: <sip:100.100.100.211;lr> Contact: <sip:1234@64.76.148.108:42772> To: <sip:02131234@100.100.100.211 <sip%3A02131234(a)100.100.100.211> >;tag=pBZjZayFUeDSp From: "camila"<sip:1234@100.100.100.211 <sip%3A1234(a)100.100.100.211> >;tag=8422566b Call-ID: MjY2NDY5NTE3OTk4MzdmMjE0YWQxNWQ4MWFmY2JhMWQ. CSeq: 1 ACK User-Agent: Bria release 2.2 stamp 45414 Content-Length: 0 Status-Line: SIP/2.0 200 OK Status-Code: 200 [Resent Packet: True] [Suspected resend of frame: 7] Message Header Via: SIP/2.0/UDP 64.76.148.108:42772 ;branch=z9hG4bK-d8754z-da0b2e4b1b5d0843-1---d8754z-;rport=42772 Record-Route: <sip:100.100.100.211;lr=on> From: "camila" <sip:1234@100.100.100.211<sip%3A1234(a)100.100.100.211> >;tag=8422566b To: <sip:02131234@100.100.100.211 <sip%3A02131234(a)100.100.100.211> >;tag=pBZjZayFUeDSp Call-ID: MjY2NDY5NTE3OTk4MzdmMjE0YWQxNWQ4MWFmY2JhMWQ. CSeq: 1 INVITE Contact: <sip:02131234@100.100.100.211:5090;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-d41e649 2011-01-11 12-25-55 -0600 Accept: application/sdp Message Body Session Description Protocol When I removed Freeswitch from the path the ACK is routed ok. The ACK is routed by loose_route, but then it fails because the if ( t_check_trans() ) { What could be happening? Regards, Ricardo.- *Ricardo Martínez **|* Ingeniero de Desarrollo RedVoiss *|* rmartinez(a)redvoiss.net Santiago - Chile *|* +56 2 2408196 www.redvoiss.net

2 1

ACK problem with FreeSwitch-Kamailio SBC implementation.
by Ricardo Martinez 17 Jan '11

17 Jan '11

Hello. I was wondering if someone can help me with some problem that I’m having with the combination of FreeSwitch and Kamailio as SBC. I’m having and ACK not routed by kamailio and dropped. Here is the debug. 100.100.100.211:5060 : Kamailio 100.100.100.211:5090 : FreeSwitch 100.100.100.108 : Caller 100.100.100.212 : Calle User Datagram Protocol, Src Port: 42772 (42772), Dst Port: sip (5060) Session Initiation Protocol Request-Line: INVITE sip:02131234@100.100.100.211<sip%3A02131234(a)100.100.100.211>SIP/2.0 Method: INVITE [Resent Packet: False] Message Header Via: SIP/2.0/UDP 64.76.148.108:42772 ;branch=z9hG4bK-d8754z-da0b2e4b1b5d0843-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:1234@64.76.148.108:42772> To: <sip:02131234@100.100.100.211 <sip%3A02131234(a)100.100.100.211>> From: "camila"<sip:1234@100.100.100.211 <sip%3A1234(a)100.100.100.211> >;tag=8422566b Call-ID: MjY2NDY5NTE3OTk4MzdmMjE0YWQxNWQ4MWFmY2JhMWQ. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: Bria release 2.2 stamp 45414 Content-Length: 388 Message Body Session Description Protocol User Datagram Protocol, Src Port: sip (5060), Dst Port: 42772 (42772) Session Initiation Protocol Status-Line: SIP/2.0 100 trying -- your call is important to us Status-Code: 100 [Resent Packet: False] Message Header Via: SIP/2.0/UDP 64.76.148.108:42772 ;branch=z9hG4bK-d8754z-da0b2e4b1b5d0843-1---d8754z-;rport=42772 To: <sip:02131234@100.100.100.211 <sip%3A02131234(a)100.100.100.211>> From: "camila"<sip:1234@100.100.100.211 <sip%3A1234(a)100.100.100.211> >;tag=8422566b Call-ID: MjY2NDY5NTE3OTk4MzdmMjE0YWQxNWQ4MWFmY2JhMWQ. CSeq: 1 INVITE Server: kamailio (3.1.1 (i386/linux)) Content-Length: 0 User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Request-Line: INVITE sip:3031234@100.100.100.212:5060 SIP/2.0 Method: INVITE [Resent Packet: False] Message Header Record-Route: <sip:100.100.100.211;lr=on> Via: SIP/2.0/UDP 100.100.100.211;branch=z9hG4bK3916.41fc6fb3.0 Via: SIP/2.0/UDP 100.100.100.211:5090 ;rport=5090;branch=z9hG4bK1QH1XDH609cUm Max-Forwards: 66 From: "camila" <sip:1234@100.100.100.211<sip%3A1234(a)100.100.100.211> >;tag=QmrB15eKrQ3Bj To: <sip:02131234@100.100.100.211 <sip%3A02131234(a)100.100.100.211>> Call-ID: 54a74b6a-9cf6-122e-05a0-0050bad00883 CSeq: 7304076 INVITE Contact: <sip:mod_sofia@100.100.100.211:5090> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-d41e649 2011-01-11 12-25-55 -0600 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 200 X-FS-Support: update_display Remote-Party-ID: "camila" <sip:1234@100.100.100.211<sip%3A1234(a)100.100.100.211> >;party=calling;screen=yes;privacy=off Message Body Session Description Protocol User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Status-Line: SIP/2.0 100 Trying Status-Code: 100 [Resent Packet: False] Message Header Via: SIP/2.0/UDP 100.100.100.211;branch=z9hG4bK3916.41fc6fb3.0;received=100.100.100.211 Via: SIP/2.0/UDP 100.100.100.211:5090 ;rport=5090;branch=z9hG4bK1QH1XDH609cUm Record-Route: <sip:100.100.100.211;lr=on> From: "camila" <sip:1234@100.100.100.211<sip%3A1234(a)100.100.100.211> >;tag=QmrB15eKrQ3Bj To: <sip:02131234@100.100.100.211 <sip%3A02131234(a)100.100.100.211>> Call-ID: 54a74b6a-9cf6-122e-05a0-0050bad00883 CSeq: 7304076 INVITE Server: Thirdlane Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: <sip:3031234@100.100.100.212<sip%3A3031234(a)100.100.100.212> > Content-Length: 0 User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Status-Line: SIP/2.0 200 OK Status-Code: 200 [Resent Packet: False] Message Header Via: SIP/2.0/UDP 100.100.100.211;branch=z9hG4bK3916.41fc6fb3.0;received=100.100.100.211 Via: SIP/2.0/UDP 100.100.100.211:5090 ;rport=5090;branch=z9hG4bK1QH1XDH609cUm Record-Route: <sip:100.100.100.211;lr=on> From: "camila" <sip:1234@100.100.100.211<sip%3A1234(a)100.100.100.211> >;tag=QmrB15eKrQ3Bj To: <sip:02131234@100.100.100.211 <sip%3A02131234(a)100.100.100.211> >;tag=as6727124a Call-ID: 54a74b6a-9cf6-122e-05a0-0050bad00883 CSeq: 7304076 INVITE Server: Thirdlane Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: <sip:3031234@100.100.100.212<sip%3A3031234(a)100.100.100.212> > Content-Type: application/sdp Content-Length: 239 Message Body Session Description Protocol User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Request-Line: ACK sip:3031234@100.100.100.212<sip%3A3031234(a)100.100.100.212>SIP/2.0 Method: ACK [Resent Packet: False] Message Header Record-Route: <sip:100.100.100.211;lr=on> Via: SIP/2.0/UDP 100.100.100.211;branch=0 Via: SIP/2.0/UDP 100.100.100.211:5090 ;rport=5090;branch=z9hG4bK20atZ819Xj3Dg Max-Forwards: 69 From: "camila" <sip:1234@100.100.100.211<sip%3A1234(a)100.100.100.211> >;tag=QmrB15eKrQ3Bj To: <sip:02131234@100.100.100.211 <sip%3A02131234(a)100.100.100.211> >;tag=as6727124a Call-ID: 54a74b6a-9cf6-122e-05a0-0050bad00883 CSeq: 7304076 ACK Contact: <sip:mod_sofia@100.100.100.211:5090> Content-Length: 0 User Datagram Protocol, Src Port: sip (5060), Dst Port: 42772 (42772) Session Initiation Protocol Status-Line: SIP/2.0 200 OK Status-Code: 200 [Resent Packet: False] Message Header Via: SIP/2.0/UDP 64.76.148.108:42772 ;branch=z9hG4bK-d8754z-da0b2e4b1b5d0843-1---d8754z-;rport=42772 Record-Route: <sip:100.100.100.211;lr=on> From: "camila" <sip:1234@100.100.100.211<sip%3A1234(a)100.100.100.211> >;tag=8422566b To: <sip:02131234@100.100.100.211 <sip%3A02131234(a)100.100.100.211> >;tag=pBZjZayFUeDSp Call-ID: MjY2NDY5NTE3OTk4MzdmMjE0YWQxNWQ4MWFmY2JhMWQ. CSeq: 1 INVITE Contact: <sip:02131234@100.100.100.211:5090;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-d41e649 2011-01-11 12-25-55 -0600 Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 249 Remote-Party-ID: "Outbound Call" <sip:02131234@100.100.100.211<sip%3A02131234(a)100.100.100.211> >;party=calling;privacy=off;screen=no Message Body Session Description Protocol User Datagram Protocol, Src Port: 42772 (42772), Dst Port: sip (5060) Session Initiation Protocol Request-Line: ACK sip:02131234@100.100.100.211:5090;transport=udp SIP/2.0 Method: ACK [Resent Packet: False] Message Header Via: SIP/2.0/UDP 64.76.148.108:42772 ;branch=z9hG4bK-d8754z-182d537bcb534165-1---d8754z-;rport Max-Forwards: 70 Route: <sip:100.100.100.211;lr> Contact: <sip:1234@64.76.148.108:42772> To: <sip:02131234@100.100.100.211 <sip%3A02131234(a)100.100.100.211> >;tag=pBZjZayFUeDSp From: "camila"<sip:1234@100.100.100.211 <sip%3A1234(a)100.100.100.211> >;tag=8422566b Call-ID: MjY2NDY5NTE3OTk4MzdmMjE0YWQxNWQ4MWFmY2JhMWQ. CSeq: 1 ACK User-Agent: Bria release 2.2 stamp 45414 Content-Length: 0 User Datagram Protocol, Src Port: sip (5060), Dst Port: 42772 (42772) Session Initiation Protocol Status-Line: SIP/2.0 200 OK Status-Code: 200 [Resent Packet: True] [Suspected resend of frame: 7] Message Header Via: SIP/2.0/UDP 64.76.148.108:42772 ;branch=z9hG4bK-d8754z-da0b2e4b1b5d0843-1---d8754z-;rport=42772 Record-Route: <sip:100.100.100.211;lr=on> From: "camila" <sip:1234@100.100.100.211<sip%3A1234(a)100.100.100.211> >;tag=8422566b To: <sip:02131234@100.100.100.211 <sip%3A02131234(a)100.100.100.211> >;tag=pBZjZayFUeDSp Call-ID: MjY2NDY5NTE3OTk4MzdmMjE0YWQxNWQ4MWFmY2JhMWQ. CSeq: 1 INVITE Contact: <sip:02131234@100.100.100.211:5090;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-d41e649 2011-01-11 12-25-55 -0600 Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 249 Remote-Party-ID: "Outbound Call" <sip:02131234@100.100.100.211<sip%3A02131234(a)100.100.100.211> >;party=calling;privacy=off;screen=no Message Body Session Description Protocol When I removed Freeswitch from the path the ACK is routed ok. The ACK is routed by loose_route, but then it fails because the if ( t_check_trans() ) { What could be happening? * * *Regards,* *Ricardo.-*

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Re: [SR-Users] OpenIMSCore and Kamalio Integration
by Carsten Bock 17 Jan '11

17 Jan '11

Hi Andres, no problem ;-) See my answers inline. 2011/1/17 "Andrés S. García Ruiz" <asgarcia(a)um.es>: > > Hi Carsten, > > Thanks a lot for all the information and the access to your developments. > > I haven't been able to answer last week cause I got the flu. I'm fine right > now. > > In a previous mail you wrote: > > "The goal of the project is to create a stable, fast, open-source, > feature-rich IMS implementation based on Kamailio and the > OpenIMS-Core." > > > > I suppose that you've chosen Kamailio as a Sip Server for XCAP, Presence, > RLS, presence-rules, etc. and OpenIMSCore as the heart of the IMS. So your > work aims -I think- to make an easy integration of Kamailio with the > OpenIMSCore. Is that true? No, not exactly. I've replaced the SER-Core from the OpenIMS-Core with the Kamailio-Core and we are in the progress of creating a new OpenIMS-Core (based on the "old" one). We do not continue to use the existing OpenIMS-Core as such. The OpenIMS-Core implementation has some limitations and has some aspects, which i personally do not like and which i am trying to change (don't misunderstand me here: For a research project, there was really great work done). However, there are a few parts, which i want to change, in order to make the OpenIMS-Core more flexible (and in some issues more stable): - The current OpenIMS-Core borrowed some code from other modules, e.g. there was some code taken from the dialog module, from the Nathelper module and maybe others, in order to implement the CSCF-Components. As the intented show-case and prototype, that is totally acceptable. However, this results in improvements and fixes done to those modules not necessarily leading to fixes in the OpenIMS-Modules - The used Core is SER 2.1-dev1. I think, that was the version of SER when the OpenIMS-Core was released. I wanted to replace the core with a more current core (in my case Kamailio 3.1/3.2), in order to bring the OpenIMS-Core forward. - many aspects of the implementation can be replaced in the current version with Core-Functions (e.g. header manipulation), that will make the code easier to maintain - i want to add the missing documentation for the modules - i wanted to include the modules in the Kamailio/sip-router.org project, in order to keep them maintained My work has nothing to do with integrating a Kamailio-Server (e.g. Presence) into a system consisting of the OpenIMS-Core, but it has something to do with improving the OpenIMS-Core. First i want to improve the OpenIMS-Core, and then we will see, what's next. All i did so far, in order to make a Kamailio-based-Presence-Server work with the IMS-Core, was to modify the Trigger-Point on my HSS a little bit... (but more a dirty hack and nothing, were i am sure if it is the currect or even safe to do; i made it work rather than thinking of, if it was correct...). You need to reconfigure your trigger-points (check the Session Case) in order to make this work. I have attached my trigger-point for presence, may this will help you... > I'm trying to design and implement a similar architecture, but I've had some > issues, mainly with how the OpenIMSCore communicates with the external SIP > presence servers. For example, when I send a RLS SUBSCRIBE to the IMS core, > it does the following: > - If the subscribed URI is an user of the domain, it translates its SIP URI > to the sip contact uri. > - If the subscribed URI is not an user, it answers with "HSS User unknown". > > Could you tell me what kind of problems have you solved till now? > > Kind regards, > Andrés. > > PS: I'm using Ubuntu. Is Ubuntu compatible with your distribution? Since Ubuntu is based on Debian, i think that should work out of the box (but no warranty for that). Kind regards, Carsten P.S: Please always use the reply to all button, in order to keep the lists in CC.

1 0

Wrong ACK packet sent from Cisco gw
by Nawfel Oujdi 17 Jan '11

17 Jan '11

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.

3 4

[OpenSER-Users] Strange ACK format gives OpenSER error?
by Nawfel Oujdi 17 Jan '11

17 Jan '11

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.i m not really sure that cause teh problem. 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.

1 0

tm module question
by Prelle, Stefan 14 Jan '11

14 Jan '11

Hi folks, I have a problem with a scenario where an incoming INVITE is forwarded to a redirect server which responds with 302 and a new target in the Contact header (B). The call is being forwarded to the new contact correctly. All responses from B that are either provisional or a 200 OK are being routed back to the caller. If B responds with any code >=400 the 302 from the redirect server is being returned to the caller instead of the received code. A B C |--INVITE->| | | |--INVITE->| | |<---302---| D | |----INVITE--->| | |<-----404-----| |<---302---| Is this intended? And if so, is there any way to change this behaviour so the latest received response is returned? I tested using the version from repository from 13.01.2011 My reduced config looks like this: route{ if (@to.tag != "") { if (loose_route()) { record_route(); t_relay(); } else { if ( is_method("ACK") ) { if ( t_check_trans() ) { t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard.\n"); exit; } } record_route(); } exit; } else { record_route(); } t_on_failure("REDIRECT_REPLY"); if (!t_relay_to_udp("192.168.0.10","5060")) { sl_reply_error(); }; exit; } failure_route[REDIRECT_REPLY] { if (t_check_status("3[0-9][0-9]")) { get_redirects("*:1"); t_relay(); exit; } } Regards, Stefan

2 2

IP Address
by Bernhard Suttner 14 Jan '11

14 Jan '11

Hi, I have a very short question: Kamailio does listen to many IP Addresses and does forward messages to the peer with t_relay. Which IP address will kamailio use for the outgoing traffic as source-ip, if there are many IP addresses on the interface? Is it possible to specify this somehow? Thanks in advance. Best regards, Bernhard

2 2

manual dispatching
by Klaus Darilion 13 Jan '11

13 Jan '11

Hi! I would like to use probing implementation in dispatcher module to keep track of the state of the gateways, but I want to choosing a gateway manually in the config. For example I defined a dispatcher set with 6 entries. Then in .cfg I want to retrieve a certain entry from the set (by index) and its state (active, inactive, probing). AFAIS there is currently now way to do this. Does anybody knows a workaround or maybe another way for manual dispatching but automatic probing? Thanks Klaus

1 2

Looking for the RTP Proxy
by Mallik 13 Jan '11

13 Jan '11

Dear sir, we re looking for RTP proxy training or solution, can you please give us more details on this thanking you with regards Mallik

1 0

How to test xhttp with wget or curl
by mayamatakeshi 13 Jan '11

13 Jan '11

Hello, can someone give me a sample of calling to xhttp from curl or wget? I have this in my kamailio.cfg: listen=tcp:192.168.2.105:80 loadmodule "xhttp.so" event_route[xhttp:request] { xhttp_reply("200", "OK", "text/html", "<html><body>OK - [$si:$sp]</body></html>"); } But when I try wget or curl, kamailio closes the connection without sending a response: [root@centos55-02105 tmp]# curl http://192.168.2.105/test curl: (52) Empty reply from server [root@centos55-02105 tmp]# wget http://192.168.2.105/test --2011-01-14 00:11:49-- http://192.168.2.105/test Connecting to 192.168.2.105:80... connected. HTTP request sent, awaiting response... No data received. I see this in the log file: Jan 14 00:04:47 centos55-02105 /usr/local/sbin/kamailio[10479]: ERROR: <core> [tcp_read.c:904]: ERROR: tcp_read_req: bad request, state=7, error=4 buf: GET /test HTTP/1.0^M User-Agent: Wget/1.11.4 Red Hat modified^M Accept: */*^M Host: 192.168.2.105^M Connection: Keep-Alive^M ^M parsed: GET /test HTTP/1.0^M User-Agent: Wget/1.11.4 Red Hat modified^M Accept: */*^M Host: 192.168.2.105^M Connection: Keep-Alive^M ^M br, takeshi

2 3

Dispatcher Setting
by Bernhard Suttner 13 Jan '11

13 Jan '11

Hi, I am using the Dispatcher module to load-balance the calls to several freeswitch servers. On which responses within the failure_route should be a fallback to another server (dst_next_dst())? I am sure, that on a 408 response there should be a fallback but whats about other 4XX responses and 5XX or 6XX? Thanks in advance. Best regards, Bernhard Suttner

2 2

OpenIMSCore and Kamalio Integration
by "Andrés S. García Ruiz" 13 Jan '11

13 Jan '11

Hi everybody, I'm trying to deploy an IMS network with OpenIMSCore and Kamailio. Since OpenIMSCore has been already tested along with Mobicents, now I want substitute Mobicents for Kamailio. I've also successfully installed Kamailio. I can run it without any problem, but when SIP Publish messages arrive at Kamalio, it answers with "407 Proxy Authentication Required". The IMS presentity is already registered against the OpenIMSCore. How can I solve that problem? The publish message sent: PUBLISH sip:testuser01@open-ims.test SIP/2.0 Route: <sip:ciervo.inf.um.es:5060;lr>, <sip:iscmark@scscf.open-ims.test:6060;lr;s=1;h=0;d=0;a=7369703a74657374757365723031406f70656e2d696d732e74657374> Call-ID: K28QO9PYAE5vn3RNdYOkIEddMEd6yDiqeAW7DwSt6XV. CSeq: 1 PUBLISH From: "testuser01" <sip:testuser01@open-ims.test>;tag=ff123bda To: "testuser01" <sip:testuser01@open-ims.test> Via: SIP/2.0/UDP 155.54.210.135:6060;branch=z9hG4bKa31a.6cba1cd2.0 Via: SIP/2.0/UDP 155.54.210.134:4060;branch=z9hG4bKa31a.71481d13.0;i=1 Via: SIP/2.0/TCP 155.54.190.166:8060;rport=41624;branch=z9hG4bK-d8754z-ce7487ff34be55f9-1---d8754z- Max-Forwards: 15 Content-Type: application/pidf+xml Expires: 30000 Event: presence Contact: <sip:testuser01@155.54.190.166:8060> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: X-Lite IMS-OSGi-Client 0.1 CVS-Mon_Nov_29_10-14-33_CET_2010 Content-Length: 451 P-Asserted-Identity: <sip:testuser01@open-ims.test> P-Charging-Vector: icid-value="P-CSCFabcd000000004cf3708400000002";icid-generated-at=155.54.210.134;orig-ioi="open-ims.test" <?xml version='1.0' encoding='UTF-8'?><presence xmlns='urn:ietf:params:xml:ns:pidf' xmlns:c='urn:ietf:params:xml:ns:pidf:cipid' xmlns:dm='urn:ietf:params:xml:ns:pidf:data-model' xmlns:rpid='urn:ietf:params:xml:ns:pidf:rpid' entity='sip:testuser01@open-ims.test'><tuple id='t6b9a6ab3'><status><basic>open</basic></status></tuple><dm:person id='p34b126e5'><rpid:activities><rpid:Online/></rpid:activities><dm:note>Online</dm:note></dm:person></presence> Kamailio answer: SIP/2.0 407 Proxy Authentication Required Call-ID: K28QO9PYAE5vn3RNdYOkIEddMEd6yDiqeAW7DwSt6XV. CSeq: 1 PUBLISH From: "testuser01" <sip:testuser01@open-ims.test>;tag=ff123bda To: "testuser01" <sip:testuser01@open-ims.test>;tag=b27e1a1d33761e85846fc98f5f3a7e58.3d3a Via: SIP/2.0/UDP 155.54.210.135:6060;branch=z9hG4bKa31a.6cba1cd2.0 Via: SIP/2.0/UDP 155.54.210.134:4060;branch=z9hG4bKa31a.71481d13.0;i=1 Via: SIP/2.0/TCP 155.54.190.166:8060;rport=41624;branch=z9hG4bK-d8754z-ce7487ff34be55f9-1---d8754z- Proxy-Authenticate: Digest realm="open-ims.test", nonce="TPNwMkzzbwZRE1piAM8yHaXGsXvwS16v" Server: kamailio (3.1.0 (i386/linux)) Content-Length: 0 Kamailio log: 5(15391) DEBUG: <core> [parser/msg_parser.c:628]: SIP Request: 5(15391) DEBUG: <core> [parser/msg_parser.c:630]: method: <PUBLISH> 5(15391) DEBUG: <core> [parser/msg_parser.c:632]: uri: <sip:testuser01@open-ims.test> 5(15391) DEBUG: <core> [parser/msg_parser.c:634]: version: <SIP/2.0> 5(15391) DEBUG: <core> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <PUBLISH> 5(15391) DEBUG: <core> [parser/parse_to.c:803]: end of header reached, state=10 5(15391) DEBUG: <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [45]; uri=[sip:testuser01@open-ims.test] 5(15391) DEBUG: <core> [parser/msg_parser.c:187]: DEBUG: to body ["testuser01" <sip:testuser01@open-ims.test> ] 5(15391) DEBUG: <core> [parser/parse_via.c:1287]: Found param type 232, <branch> = <z9hG4bKa31a.6cba1cd2.0>; state=16 5(15391) DEBUG: <core> [parser/parse_via.c:2300]: end of header reached, state=5 5(15391) DEBUG: <core> [parser/msg_parser.c:513]: parse_headers: Via found, flags=2 5(15391) DEBUG: <core> [parser/msg_parser.c:515]: parse_headers: this is the first via 5(15391) DEBUG: <core> [receive.c:145]: After parse_msg... 5(15391) DEBUG: <core> [receive.c:186]: preparing to run routing scripts... 5(15391) DEBUG: <core> [parser/parse_via.c:1287]: Found param type 232, <branch> = <z9hG4bKa31a.71481d13.0>; state=6 5(15391) DEBUG: <core> [parser/parse_via.c:1287]: Found param type 236, <i> = <1>; state=16 5(15391) DEBUG: <core> [parser/parse_via.c:2300]: end of header reached, state=5 5(15391) DEBUG: <core> [parser/msg_parser.c:513]: parse_headers: Via found, flags=100 5(15391) DEBUG: <core> [parser/msg_parser.c:526]: parse_headers: this is the second via 5(15391) DEBUG: <core> [parser/parse_via.c:1287]: Found param type 235, <rport> = <41624>; state=6 5(15391) DEBUG: <core> [parser/parse_via.c:1287]: Found param type 232, <branch> = <z9hG4bK-d8754z-ce7487ff34be55f9-1---d8754z->; state=16 5(15391) DEBUG: <core> [parser/parse_via.c:2300]: end of header reached, state=5 5(15391) DEBUG: <core> [parser/msg_parser.c:513]: parse_headers: Via found, flags=100 5(15391) DEBUG: maxfwd [mf_funcs.c:85]: value = 15 5(15391) DEBUG: <core> [parser/msg_parser.c:199]: DEBUG: get_hdr_body : content_length=451 5(15391) DEBUG: <core> [parser/msg_parser.c:101]: found end of header 5(15391) DEBUG: <core> [parser/parse_to.c:174]: DEBUG: add_param: tag=ff123bda 5(15391) DEBUG: <core> [parser/parse_to.c:803]: end of header reached, state=29 5(15391) DEBUG: sanity [mod_sanity.c:217]: all sanity checks passed 5(15391) DEBUG: siputils [checks.c:73]: no totag 5(15391) DEBUG: tm [t_lookup.c:1081]: DEBUG: t_check_msg: msg id=1 global id=0 T start=0xffffffff 5(15391) DEBUG: tm [t_lookup.c:528]: t_lookup_request: start searching: hash=41274, isACK=0 5(15391) DEBUG: tm [t_lookup.c:485]: DEBUG: RFC3261 transaction matching failed 5(15391) DEBUG: tm [t_lookup.c:711]: DEBUG: t_lookup_request: no transaction found 5(15391) DEBUG: tm [t_lookup.c:1150]: DEBUG: t_check_msg: msg id=1 global id=1 T end=(nil) 5(15391) DEBUG: <core> [socket_info.c:501]: grep_sock_info - checking if host==us: 13==9 && [open-ims.test] == [127.0.0.1] 5(15391) DEBUG: <core> [socket_info.c:504]: grep_sock_info - checking if port 5060 matches port 5060 5(15391) DEBUG: <core> [socket_info.c:501]: grep_sock_info - checking if host==us: 13==14 && [open-ims.test] == [155.54.190.245] 5(15391) DEBUG: <core> [socket_info.c:504]: grep_sock_info - checking if port 5060 matches port 5060 5(15391) DEBUG: <core> [socket_info.c:501]: grep_sock_info - checking if host==us: 13==9 && [open-ims.test] == [127.0.0.1] 5(15391) DEBUG: <core> [socket_info.c:504]: grep_sock_info - checking if port 5060 matches port 5060 5(15391) DEBUG: <core> [socket_info.c:501]: grep_sock_info - checking if host==us: 13==14 && [open-ims.test] == [155.54.190.245] 5(15391) DEBUG: <core> [socket_info.c:504]: grep_sock_info - checking if port 5060 matches port 5060 5(15391) DEBUG: auth_db [authorize.c:239]: realm value [open-ims.test] 5(15391) DEBUG: auth [api.c:85]: auth:pre_auth: Credentials with realm 'open-ims.test' not found 5(15391) DEBUG: auth_db [authorize.c:257]: not authenticated 5(15391) DEBUG: auth [challenge.c:102]: build_challenge_hf: realm='open-ims.test' 5(15391) DEBUG: auth [challenge.c:236]: auth: 'Proxy-Authenticate: Digest realm="open-ims.test", nonce="TPNwMkzzbwZRE1piAM8yHaXGsXvwS16v" ' 5(15391) DEBUG: sl [sl.c:278]: reply in stateless mode (sl) 5(15391) DEBUG: <core> [msg_translator.c:207]: check_via_address(155.54.210.135, 155.54.210.135, 0) 5(15391) DEBUG: <core> [usr_avp.c:646]: DEBUG:destroy_avp_list: destroying list (nil) 5(15391) DEBUG: <core> [usr_avp.c:646]: DEBUG:destroy_avp_list: destroying list (nil) 5(15391) DEBUG: <core> [usr_avp.c:646]: DEBUG:destroy_avp_list: destroying list (nil) 5(15391) DEBUG: <core> [usr_avp.c:646]: DEBUG:destroy_avp_list: destroying list (nil) 5(15391) DEBUG: <core> [usr_avp.c:646]: DEBUG:destroy_avp_list: destroying list (nil) 5(15391) DEBUG: <core> [usr_avp.c:646]: DEBUG:destroy_avp_list: destroying list (nil) 5(15391) DEBUG: <core> [receive.c:289]: receive_msg: cleaning up Thanks in advance, Andrés. -- ----------------------- Andrés S. García Ruiz ----------------------- e-mail: asgarcia(a)um.es Teléfono: (+34) 868 888258 Dpto. de Ingeniería de la Información y las Comunicaciones Facultad de Informática Universidad de Murcia 30100 Campus de Espinardo

5 13

set_dlg_profile not working?
by Ricardo Martinez 12 Jan '11

12 Jan '11

Hello. I’m trying to count the calls to a specific gateway. So I’m using the dialog module. This route is done before the final relay. route[CHANNELS] { if( is_method("INVITE") ) { if(!isflagset(31) ) { # user has max channel limit set as preference if(is_avp_set("$avp(s:channels)") && avp_check("$avp(s:channels)", "gt/s:0")) { # get current calls for uuid xlog("L_INFO","[$ci] : avp_channels existe y es mayor que 0 ?"); if (get_profile_size("gws","$rd","$var(calls)")) { xlog("L_INFO","Actualmente el gateway $rd tiene $var(calls) llamadas simultaneas\n"); } # check within limit if($avp(s:channels) > $var(calls)) { xlog("L_INFO", "CHANNELS : terminador '$rd' tiene '$var(calls)' de '$avp(s:channels)' active calls before this one\n"); $var(setprofile) = 1; } else { xlog("L_INFO", "Call crtl: user channel limit exceeded [$var(calls)/$avp(s:channels)]\n"); send_reply("487", "Request Terminated: Channel limit exceeded"); exit; } } else { xlog("L_INFO","[$ci] : the value if channels is $avp(s:channels)"); if( !is_avp_set("$avp(s:channels)") ) { xlog("L_INFO","[$ci] : avp_channel not set"); }; if(!avp_check("$avp(s:channels)", "gt/s:0") ) { xlog("L_INFO","[$ci] : avp_channel menor que 0?"); }; $var(setprofile) = 0; } } if($var(setprofile) > 0) { * if ( set_dlg_profile("gws","$rd") ) {* * xlog("L_INFO","set_dlg_profile exitoso");* } } When i made the call and try to see the profile with this command : [root@sipvoiss kamailio]# kamctl fifo profile_get_size gws 10.0.0.220 profile:: name=gws value=10.0.0.220 count=0 I’m not seeing the call added to the profile, despite the fact I using the * set_dlg_profile("gws","$rd")* What could be happening? This is part of the debug : Jan 11 18:07:03 .: INFO: <script>: [ 71492292-34f2-a3ed-814e-0002a400d53d(a)10.0.0.214] INVITE dominio:10.0.0.208 LOCAL DOMAIN Jan 11 18:07:03 .: DEBUG: misc_radius [functions.c:212]: success Jan 11 18:07:03 .: DEBUG: misc_radius [functions.c:133]: AVP 'tranum'/0=' sip:02444444444@10.0.0.208 <sip%3A02444444444(a)10.0.0.208>'/0 has been added Jan 11 18:07:03 .: DEBUG: misc_radius [functions.c:133]: AVP 'channels'/0='2'/0 has been added Jan 11 18:07:03 .: DEBUG: avpops [avpops_impl.c:1047]: 1 avps were processed Jan 11 18:07:03 .: INFO: <script>: [ 71492292-34f2-a3ed-814e-0002a400d53d(a)10.0.0.214] INVITE con dominio:10.0.0.208 checks de usuario via avp_radius OK Jan 11 18:07:03 .: INFO: <script>: [ 71492292-34f2-a3ed-814e-0002a400d53d(a)10.0.0.214] CHECKS SECTION :: INVITE:sip:02444444444@10.0.0.208 <INVITE%3Asip%3A02444444444(a)10.0.0.208>checks ok... Jan 11 18:07:03 .: DEBUG: lcr [lcr_mod.c:1748]: added matched_gws[0]=[1, 4, 1, 8207736] Jan 11 18:07:03 .: DEBUG: lcr [lcr_mod.c:1645]: added gw_uri_avp <1|1|4|550|3700771904||5060||1|0> with weight <8207736> Jan 11 18:07:03 .: DEBUG: lcr [lcr_mod.c:1911]: r_uri < sip:5504444444@10.0.0.220:5060>, dst_uri <> Jan 11 18:07:03 .: DEBUG: lcr [lcr_mod.c:2015]: added ruri_user_avp <02444444444> Jan 11 18:07:03 .: DEBUG: lcr [lcr_mod.c:2050]: added flags_avp <0> Jan 11 18:07:03 .: DEBUG: lcr [lcr_mod.c:2057]: added defunct_gw_avp <3700771904> Jan 11 18:07:03 .: INFO: <script>: [ 71492292-34f2-a3ed-814e-0002a400d53d(a)10.0.0.214] Ruta LCR : cambiando sip:02444444444@10.0.0.208 <sip%3A02444444444(a)10.0.0.208> por sip:5504444444@10.0.0.220:5060 Jan 11 18:07:03 .: INFO: <script>: [ 71492292-34f2-a3ed-814e-0002a400d53d(a)10.0.0.214] : entro en ruta 39 de call control Jan 11 18:07:03 .: INFO: <script>: [ 71492292-34f2-a3ed-814e-0002a400d53d(a)10.0.0.214] : no esta seteado el flag 31 Jan 11 18:07:03 .: DEBUG: avpops [avpops_impl.c:1180]: check <2> against <0> as str /16 Jan 11 18:07:03 .: INFO: <script>: [ 71492292-34f2-a3ed-814e-0002a400d53d(a)10.0.0.214] : avp_channels existe y es mayor que 0 ? Jan 11 18:07:03 .: INFO: <script>: Actualmente el gateway 10.0.0.220 tiene 0 llamadas simultaneas Jan 11 18:07:03 .: INFO: <script>: CHANNELS : terminador '10.0.0.220' tiene '0' de '2' active calls before this one Jan 11 18:07:03 .: INFO: <script>: set_dlg_profile exitoso Thanks in advance, *Regards,* *Ricardo.-*

2 5

AFTER MAINTENANCE DIALOG MODULE NOT WORKING
by Phillman25 Kyriacou 12 Jan '11

12 Jan '11

Hi I performed the following maintenance actions to my server that is currently installed with kamailio v3.1 cd /usr/local/src/kamailio-3.1.0/kamailio git pull origin make all make install After this action the dialog module is not working, after investigation, i found out that the following entry is causing this issue: modparam("dialog", "db_mode", 1) In my syslog i find the following errors: Jan 11 16:45:54 SipProxy /usr/local/sbin/kamailio[7397]: CRITICAL: dialog [dlg_db_handler.c:117]: BUG - db connection found already open Jan 11 16:45:54 SipProxy /usr/local/sbin/kamailio[7398]: : <core> [pass_fd.c:293]: ERROR: receive_fd: EOF on 33 Jan 11 16:45:54 SipProxy /usr/local/sbin/kamailio[7398]: : <core> [pass_fd.c:293]: ERROR: receive_fd: EOF on 34 Jan 11 16:45:54 SipProxy /usr/local/sbin/kamailio[7397]: ERROR: dialog [dialog.c:620]: failed to connect to database (rank=24) Jan 11 16:45:54 SipProxy /usr/local/sbin/kamailio[7397]: ERROR: <core> [sr_module.c:832]: init_mod_child(): Error while initializing module dialog (/usr/local/lib/kamailio/modules_k/dialog.so) Jan 11 16:45:54 SipProxy /usr/local/sbin/kamailio[7397]: ERROR: <core> [pt.c:481]: ERROR: fork_tcp_process(): init_child failed for process 30, pid 7397, "tcp receiver child=7" Jan 11 16:45:54 SipProxy /usr/local/sbin/kamailio[7397]: ERROR: <core> [tcp_main.c:4811]: ERROR: tcp_main: fork failed: Success Jan 11 16:45:54 SipProxy /usr/local/sbin/kamailio[7398]: : <core> [pass_fd.c:293]: ERROR: receive_fd: EOF on 37 Jan 11 16:45:54 SipProxy /usr/local/sbin/kamailio[7398]: : <core> [pass_fd.c:293]: ERROR: receive_fd: EOF on 43 Jan 11 16:45:54 SipProxy /usr/local/sbin/kamailio[7398]: : <core> [pass_fd.c:293]: ERROR: receive_fd: EOF on 30 Jan 11 16:45:54 SipProxy /usr/local/sbin/kamailio[7398]: : <core> [pass_fd.c:293]: ERROR: receive_fd: EOF on 41 Jan 11 16:45:54 SipProxy /usr/local/sbin/kamailio[7398]: : <core> [pass_fd.c:293]: ERROR: receive_fd: EOF on 5 Jan 11 16:45:54 SipProxy /usr/local/sbin/kamailio[7396]: WARNING: <core> [db_id.c:281]: identical DB URLs, but different DB connection pid [7396/7349] Jan 11 16:45:54 SipProxy /usr/local/sbin/kamailio[7391]: INFO: <core> [main.c:807]: INFO: signal 15 received Jan 11 16:45:54 SipProxy /usr/local/sbin/kamailio[7398]: : <core> [pass_fd.c:293]: ERROR: receive_fd: EOF on 47 Jan 11 16:45:54 SipProxy /usr/local/sbin/kamailio[7396]: CRITICAL: dialog [dlg_db_handler.c:117]: BUG - db connection found already open Jan 11 16:45:54 SipProxy /usr/local/sbin/kamailio[7396]: ERROR: dialog [dialog.c:620]: failed to connect to database (rank=23) Jan 11 16:45:54 SipProxy /usr/local/sbin/kamailio[7396]: ERROR: <core> [sr_module.c:832]: init_mod_child(): Error while initializing module dialog (/usr/local/lib/kamailio/modules_k/dialog.so) Jan 11 16:45:54 SipProxy /usr/local/sbin/kamailio[7396]: ERROR: <core> [pt.c:481]: ERROR: fork_tcp_process(): init_child failed for process 29, pid 7396, "tcp receiver child=6" Jan 11 16:45:54 SipProxy /usr/local/sbin/kamailio[7398]: : <core> [pass_fd.c:293]: ERROR: receive_fd: EOF on 39 Jan 11 16:45:54 SipProxy /usr/local/sbin/kamailio[7396]: ERROR: <core> [tcp_main.c:4811]: ERROR: tcp_main: fork failed: Success Jan 11 16:45:54 SipProxy /usr/local/sbin/kamailio[7398]: : <core> [pass_fd.c:293]: ERROR: receive_fd: EOF on 45 Only the dialog module is facing this problem Has anyone faced this before? Thanks Phillip

2 2

mi_fifo question
by Ricardo Martinez 11 Jan '11

11 Jan '11

Hello. I’m trying to use the kamctl command to get information about dialogs. I’m using the next line : kamctl fifo profile_get_size:kamailio_fifo and I’m getting the next output in the log file : Jan 11 12:34:38 /usr/local/sbin/kamailio[15051]: ERROR: mi_fifo [fifo_fnc.c:223]: open error (/tmp/kamailio_fifo:openser_receiver_15073): No such file or directory Jan 11 12:34:38 /usr/local/sbin/kamailio[15051]: ERROR: mi_fifo [fifo_fnc.c:484]: cannot open reply pipe /tmp/kamailio_fifo:openser_receiver_15073 What I’m doing wrong? This is part of the configuration file : # ----- mi_fifo params ------ modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") modparam("mi_fifo", "fifo_group", 0) modparam("mi_fifo", "fifo_group", "root") modparam("mi_fifo", "fifo_user", 0) modparam("mi_fifo", "fifo_user", "root") and this is the file kamailio_fifo in the /tmp directory : [root@kamailio]# ls -la /tmp total 52 drwxrwxrwt. 4 root root 4096 Jan 11 12:38 . dr-xr-xr-x. 25 root root 4096 Jan 7 09:25 .. drwxrwxrwt 2 root root 4096 Jan 7 09:25 .ICE-unix *prw-rw---- 1 root root 0 Jan 11 12:34 kamailio_fifo* drwx------ 2 root root 4096 Jan 11 09:54 ssh-gpTCQ12700 Hope someone could help me. Regards, Ricardo.-

2 3

Re: [SR-Users] Avp_write missing modules ?
by Alex Balashov 10 Jan '11

10 Jan '11

Please keep the discussion on list - cc: the list. I see no indication in the documentation that avp_write() is a function that exists, even going back to 1.4.x. On 01/10/2011 04:54 PM, Amit Nepal wrote: > Kamailio 3.1 > > Thank You > Amit Nepal > Systems Administrator > Phoenix Internet > Phone: 602-385-0731 > 602-234-0917#112 > http://www.phoenixinternet.net > > > On 1/10/2011 2:50 PM, Alex Balashov wrote: >> On 01/10/2011 04:48 PM, Amit Nepal wrote: >>> I am getting missing loadmodule issue with avp_write. I have the >>> following statement on line 11 and i am getting this error. Could you >>> please help me on this. But i can load from db using avp_db_load and i >>> can delete avp values but only avp_write is not working. >>> >>> >>> avp_write("$ruri", "s:fwd_blind"); >>> >>> loading modules under >>> /usr/local/lib/kamailio/modules_k/:/usr/local/lib/kamailio/modules/ >>> 0(2361) : <core> [cfg.y:3412]: parse error in config file >>> /usr/local/etc/kamailio/cfgs/pstn.cfg, line 11, column 39: unknown >>> command, missing loadmodule? >>> >>> ERROR: bad config file (1 errors) >> >> Which version of Kamailio? >> -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/

3 5

Avp_write missing modules ?
by Amit Nepal 10 Jan '11

10 Jan '11

I am getting missing loadmodule issue with avp_write. I have the following statement on line 11 and i am getting this error. Could you please help me on this. But i can load from db using avp_db_load and i can delete avp values but only avp_write is not working. avp_write("$ruri", "s:fwd_blind"); loading modules under /usr/local/lib/kamailio/modules_k/:/usr/local/lib/kamailio/modules/ 0(2361) : <core> [cfg.y:3412]: parse error in config file /usr/local/etc/kamailio/cfgs/pstn.cfg, line 11, column 39: unknown command, missing loadmodule? ERROR: bad config file (1 errors) Thank you Amit

2 1

Problem with contact header parsing and get_redirects
by Stefan Prelle 10 Jan '11

10 Jan '11

Hi folks, Since upgrading vom Kamailio 1.5 to 3.1 I have a problem using get_redirects from the uac_redirect module. It seems that there is a problem with parsing the Contact header from the received 302. This is my script excerpt: if (t_check_status("3[0-9][0-9]")) { get_redirects("*"); if (!t_relay()) { xlog("L_ERR","t_relay failed\n"); } } Some 302 messages lead (as expected) to INVITEs send to targets in the Contact header, for others t_relay fails. A working example would be Contact:<sip:+49406686100@192.168.0.1:5060;user=phone>;q=0.5,<sip: +49406686100@192.168.0.2:5060;user=phone>;q=0.25 It fails on Contact: sip:+49406686100@192.168.0.1:5060 The following log messages appear when calling get_redirects (and failing t_relay): kamailio[7372]: DEBUG: uac_redirect [rd_funcs.c:74]: resume branch=0 kamailio[7372]: DEBUG: uac_redirect [rd_funcs.c:81]: checking branch=0 (added=0) kamailio[7372]: DEBUG: uac_redirect [rd_funcs.c:85]: branch=0 is a redirect (added=0) kamailio[7372]: WARNING: <core> [mem/q_malloc.c:366]: qm_malloc(0x82ba320, 32) called from <core>: parser/msg_parser.c: parse_headers(339) kamailio[7372]: WARNING: <core> [mem/q_malloc.c:406]: qm_malloc(0x82ba320, 32) returns address 0x8311d94 frag. 0x8311d7c (size=32) on 1 -th hit kamailio[7372]: DEBUG: uac_redirect [rd_funcs.c:233]: contact hdr not found in sh_rpl The Contact header that fails obviously has no parameters for the URI and contact itself. As far as I understand the RFC 3261 BNF this is valid. The 302 response is by the way created by kamailio when using the LCR module (like this) if (!load_gws("1")) {... exit;} if (!next_gw()) {... exit; } sl_send_reply("302", "Not here"); Did I miss anything when upgrading or is this a problem with the parser? Cheers, Stefan

2 5

Re: [SR-Users] ENUM changes?
by Juha Heinanen 09 Jan '11

09 Jan '11

klaus, could you try with master version if it fixes the problem. -- juha

2 2

Kamailio 3.1.1 + Sipp
by Rochdi ABID 09 Jan '11

09 Jan '11

Hello community, I want to test Kamailio 3.1.1 with SIPP. I followed many ideas that I found them on the internet but without no success. In particular, I followed this link http://www.troubleshootingwiki.org/OpenSER#Stress_Test_The_SIP_Signaling I can see traffic from the uac --> Kamailio --> uas ans in the other sens but no successful call. Can you give me any ideas please. Regards, Rochdi

2 1

Kamailio 3.1.1 + Sipp
by Rochdi ABID 08 Jan '11

08 Jan '11

Hello community, I want to test Kamailio 3.1.1 with SIPP. I followed many ideas that I found them on the internet but without no success. In particular, I followed this link http://www.troubleshootingwiki.org/OpenSER#Stress_Test_The_SIP_Signaling I can see traffic from the uac --> Kamailio --> uas ans in the other sens but no successful call. Can you any ideas please. Regards, Rochdi

1 0

Re: [SR-Users] Change the SIP-To-User
by Bernhard Suttner 07 Jan '11

07 Jan '11

Hi, thanks to all responses. I wonder if there is a function to modify the To/From Header stateful. I mean, change the TO header in the INVITE to A and then the 200 OK (for example) reverse the change back to the previous value to B? but the "magic" $tU is amazing... Best regards, Bernhard ----- Original Message ----- From: Olle E. Johansson [mailto:oej@edvina.net] To: Alex Balashov [mailto:abalashov@evaristesys.com] Cc: sr-users(a)lists.sip-router.org Sent: Fri, 07 Jan 2011 11:09:41 +0100 Subject: Re: [SR-Users] Change the SIP-To-User > > 7 jan 2011 kl. 10.55 skrev Alex Balashov: > > > On 01/07/2011 04:52 AM, Carsten Bock wrote: > > > >> the trick, when using "uac_replace_from" is, that it will > >> "automagically" change subsequest requests. This is required in order > >> to work with most SIP-Endpoints. > >> If you just want to change the "From" for the current request, that is > fine. > > > > I understand. Changing To/From by a proxy has long been a thorny issue, > in the sense that on the one hand those fields are purely cosmetic in > strict-3261 mode, but on the other hand, changing them breaks 2543 > compatibility, and it's just not something a proxy is supposed to do. > Well, that changed with a new rfc (put number here) about caller ID updates. > I do not remember the proxy role, but a UA can now change From/To identities > mid call, like after a call transfer in a pbx. > > /O > _______________________________________________ > 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 3

RLS server as SUBSCRIBE proxy?
by Jeremya 07 Jan '11

07 Jan '11

Hi, My situation is that I have a number of SIP end points (server) that support SUBSCRIBE method on various resources they hold. Primarily these are dialog resources, but include arbitrary other resources with custom MIME types. I also have a large number of SIP end-points (clients) that use SUBSCRIBE to resources held by these end-points. Some of these end-points are standard phones (SPA-962) monitoring dialog state. Other end points are custom clients that monitor a variety of custom resources and MIME types. I have got to the point where the number of SUBSCRIBE clients is making it impractical for the SUBSCRIBE sources to support them. I want to make a SUBSCRIBE proxy server that makes arbitrary SUBSCRIBE requests to my SIP end points (server) and in turn accepts SUBSCRIBE requests from my consumer SIP clients (Phones etc). In effect the server would aggregate SUBSCRIBE requests into a single SUBSCRIBE to a target resource. Ideally the SUBSCRIPTION would be dynamic, generated by the SUBSCRIBE requests from the SIP phones and other clients, but this can be static. I've read the documentation for the RLS module in Kamailio but I can't actually figure out what it does. It may do what I want (if only in some database mode) but I'd appreciate advice as to whether to pursue this as an option or go to plan-B Regards Jeremy

2 2

Re: [SR-Users] Change the SIP-To-User
by Bernhard Suttner 07 Jan '11

07 Jan '11

Hi, ah, thanks. Would be good to have a uac_replace_to. Sounds like a "30min" task to someone familiar with kamailio programming :-) BR, Bernhard ----- Original Message ----- From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] To: Bernhard Suttner [mailto:bernhard.suttner@winet.ch] Cc: sr-users(a)lists.sip-router.org Sent: Fri, 07 Jan 2011 21:59:55 +0100 Subject: Re: [SR-Users] Change the SIP-To-User > Bernhard Suttner wrote: > > Hi, > > > > thanks to all responses. I wonder if there is a function to modify the > To/From Header stateful. I mean, change the TO header in the INVITE to A and > then the 200 OK (for example) reverse the change back to the previous value > to B? > > > > but the "magic" $tU is amazing... > > If you use uac_replace_from(), the changes will stateful and fixed for > every in-dialog requests and responses. > > But there is no similar feature for To header manipulation. > > regards > klaus > > >

1 0

Using Dispatcher Module From Mysql
by Amit Nepal 07 Jan '11

07 Jan '11

Hi everyone, Again stuck at the dispatcher module.Do anyone happen to have a working config for kamailio with dispatcher list from mysql database ? if not can someone please guide me on this ? Do i need to pass module parameters for the column and the table names ? I have the following config : loadmodule "dispatcher.so" if(uri=~"^sip:1[2-9][0-9]{9}@") { if(is_user_in("credentials","longdistance")) { #route(PSTN); #exit; ds_select_dst("1","8"); forward(); } else{ sl_send_reply("403","No Permissions for Long Distance calls"); exit; }; }; -- Thank You Amit Nepal

3 5

Dispatcher Attributes field
by alex pappas 07 Jan '11

07 Jan '11

Dear all, I'm trying with the Dispatcher module to terminate a call to three different gateways serially. For each gw I use the attribute field to get the attribute and add it as a prefix to the RU. The problem is that the $avp(dsattrs) is returning always the first value of the first gw that is using. example: destination sip:100.12.12.12:5060 attrs 111 destination sip:100.12.12.11:5060 attrs 222 destination sip:100.12.12.10:5060 attrs 333 In this example the rU will always be 111+number. I tried also to initiate the avp in the Failure route by giving null value but still does not change the result. In Request route: if(ds_select_domain("$avp(s:disp_dstgrp)", "4")) { $avp(s:term_prefix) = $avp(dsattrs); # avp for CDR purpose $rU = $avp(dsattrs) + $rU; t_on_failure("FAIL_NATIONAL"); t_relay(); exit; } In Failure route: if(!ds_next_domain()) { t_reply("503", "Service Unavailable"); exit; } else { $rU = $avp(dsattrs) + $rU; t_on_failure("FAIL_THIS"); append_branch(); if(!t_relay()) { t_reply("503","Service Unavailable"); exit; }

2 5

Change the SIP-To-User
by Bernhard Suttner 07 Jan '11

07 Jan '11

Hi, what is the best way to replace the userpart of the SIP-To URI with the content of a pseudo variable? Thanks for every hint. Best regards, Bernhard

6 8

problem in MAKE command during installation Sip Express Router
by city spider 07 Jan '11

07 Jan '11

hello i want to install Sip Express Router on CENT OS 5.5. the problem is that after untar the ser-0.9.6.src.tar.gz package my system doesn't operate MAKE command ,and returns 127 errors containing message "the system doesn't find the folder...." is there any solution for this problem... waiting for your response.

2 1

Call Routing Not working
by Tim King 07 Jan '11

07 Jan '11

Below is my current code and my calls are just getting 100 trying -- your call is important to us even between to locally registered extension. Any guidance as to how to simplify troubleshooting this routing? ####### Routing Logic ######## # Main SIP request routing logic # - processing of any incoming SIP request starts with this route route { # per request initial checks route(REQINIT); # NAT detection route(NAT); # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); # authentication route(AUTH); # 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; } #!ifdef WITH_FREESWITCH # save callee ID $avp(callee) = $rU; route(FSDISPATCH); #!endif # user location service route(LOCATION); route(RELAY); } route[RELAY] { #!ifdef WITH_NAT if (check_route_param("nat=yes")) { setbflag(FLB_NATB); } if (isflagset(FLT_NATS) || isbflagset(FLB_NATB)) { route(RTPPROXY); } #!endif #!ifdef WITH_CFGSAMPLES /* example how to enable some additional event routes */ if (is_method("INVITE")) { #t_on_branch("BRANCH_ONE"); t_on_reply("REPLY_ONE"); t_on_failure("FAIL_ONE"); } #!endif if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood dection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($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()) { if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } 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")) { if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging ## setbflag(FLB_NATSIPPING); } if (!save("location")) sl_reply_error(); exit; } } # USER location service route[LOCATION] { #!ifdef WITH_ALIASDB # search in DB-based aliases alias_db_lookup("dbaliases"); #!endif if (!lookup("location")) { switch ($rc) { case -1: case -3: t_newtran(); t_reply("404", "Not Found"); exit; case -2: sl_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; #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; }; if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); exit; } return; } # Authentication route route[AUTH] { #!ifdef WITH_AUTH if (is_method("REGISTER")) { # authenticate the REGISTER requests (uncomment to enable auth) if (!www_authorize("$td", "subscriber")) { www_challenge("$td", "0"); exit; } if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else { #!ifdef WITH_FREESWITCH if(route(FSINBOUND)) return; #!endif #!ifdef WITH_IPAUTH if(allow_source_address()) { # source IP allowed return; } #!endif # authenticate if from local subscriber if (from_uri==myself) { if (!proxy_authorize("$fd", "subscriber")) { proxy_challenge("$fd", "0"); exit; } if (is_method("PUBLISH")) { if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else { if ($au!=$fU) { sl_send_reply("403","Forbidden auth ID"); exit; } } consume_credentials(); # caller authenticated } else { # caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (!uri==myself) { sl_send_reply("403","Not relaying"); exit; } } } #!endif return; } # Caller NAT detection route route[NAT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (method=="REGISTER") { fix_nated_register(); } else { fix_nated_contact(); } setflag(FLT_NATS); } #!endif return; } # RTPProxy control route[RTPPROXY] { #!ifdef WITH_NAT if (is_method("BYE")) { unforce_rtp_proxy(); } else if (is_method("INVITE")){ force_rtp_proxy(); } if (!has_totag()) add_rr_param(";nat=yes"); #!endif return; } # Routing to foreign domains route[SIPOUT] { if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(RELAY); } } #!ifdef WITH_FREESWITCH # FreeSWITCH routing blocks route[FSINBOUND] { if($si== $sel(cfg_get.freeswitch.bindip) && $sp==$sel(cfg_get.freeswitch.bindport)) return 1; return -1; } route[FSDISPATCH] { if(!is_method("INVITE")) return; if(route(FSINBOUND)) return; # dial number selection switch($rU) { case /"^41$": # 41 - voicebox menu # allow only authenticated users if($au==$null) { sl_send_reply("403", "Not allowed"); exit; } $rU = "vm-" + $au; break; case /"^441[0-9][0-9]$": # starting with 44 folowed by 1XY - direct call to voice box strip(2); route(FSVBOX); break; case /"^433[01][0-9][0-9]$": # starting with 433 folowed by (0|1)XY - conference strip(2); break; case /"^45[0-9]+$": strip(2); break; default: # offline - send to voicebox if (!registered("location")) { route(FSVBOX); exit; } # online - do bridging prefix("kb-"); if(is_method("INVITE")) { # in case of failure - re-route to FreeSWITCH VoiceMail t_on_failure("FAIL_FSVBOX"); } } route(FSRELAY); exit; } route[FSVBOX] { if(!($rU=~"^1[0-9][0-9]+$")) return; prefix("vb-"); route(FSRELAY); } # Send to FreeSWITCH route[FSRELAY] { $du = "sip:" + $sel(cfg_get.freeswitch.bindip) + ":" + $sel(cfg_get.freeswitch.bindport); if($var(newbranch)==1) { append_branch(); $var(newbranch) = 0; } route(RELAY); exit; } #!endif #!ifdef WITH_FREESWITCH failure_route[FAIL_FSVBOX] { #!ifdef WITH_NAT if (is_method("INVITE") && (isbflagset(FLB_NATB) || isflagset(FLT_NATS))) { unforce_rtp_proxy(); } #!endif if (t_is_canceled()) { exit; } if (t_check_status("486|408")) { # re-route to FreeSWITCH VoiceMail $rU = $avp(callee); $var(newbranch) = 1; route(FSVBOX); } } #!endif # sample config file for dispatcher module #!ifdef WITH_DISPATCHER modparam("dispatcher", "db_url", "mysql://openser:password@localhost /openser") route{ if ( !mf_process_maxfwd_header("10") ) { sl_send_reply("483","To Many Hops"); drop(); }; ds_select_dst("1", "0"); forward(); # t_relay(); } #!endif

2 1

Routing error
by Tim King 07 Jan '11

07 Jan '11

My calls are not making it to the dispatcher and I am not sure what I am doing wrong. Can anyone see what I need to modify so that my calls can be distributed to my free switch boxes via the dispatcher module? cat /usr/local/etc/kamailio/kamailio.cfg #!KAMAILIO #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_FREESWITCH #!define WITH_ANTIFLOOD #!define WITH_DISPATCHER #!ifdef ACCDB_COMMENT ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; #!endif ####### Defined Values ######### # *** Value defines - IDs used later in config #!ifdef WITH_MYSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!define DBURL "mysql://openser:password@localhost/openser" #!endif #!ifdef WITH_MULTIDOMAIN # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 #!else #!define MULTIDOMAIN 0 #!endif # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 ####### Global Parameters ######### #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL0 fork=yes children=4 /* uncomment the next line to disable TCP (default on) */ #disable_tcp=yes /* uncomment the next line to disable the auto discovery of local aliases based on reverse DNS on IPs (default on) */ #auto_aliases=no /* add local domain aliases */ #alias="sip.mydomain.com" /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ #listen=udp:10.0.0.10:5060 /* port to listen to * - can be specified more than once if needed to listen on many ports */ port=5060 #!ifdef WITH_TLS enable_tls=yes #!endif #!ifdef WITH_FREESWITCH freeswitch.bindip = "127.0.0.1" desc "FreeSWITCH IP Address" freeswitch.bindport = "5090" desc "FreeSWITCH Port" #!endif ####### Modules Section ######## # set paths to location of modules #!ifdef LOCAL_TEST_RUN mpath="modules_k:modules" #!else mpath="/usr/local/lib/kamailio/modules_k/:/usr/local/lib/kamailio/modules/" #!endif #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif 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 "mi_rpc.so" loadmodule "acc.so" loadmodule "dispatcher.so" # loadmodule "modules/tm/tm.so" #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #!ifdef WITH_IPAUTH loadmodule "permissions.so" #!endif #!endif #!ifdef WITH_ALIASDB loadmodule "alias_db.so" #!endif #!ifdef WITH_MULTIDOMAIN loadmodule "domain.so" #!endif #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_NAT loadmodule "nathelper.so" loadmodule "rtpproxy.so" #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_ANTIFLOOD loadmodule "htable.so" loadmodule "pike.so" #!endif #!ifdef WITH_XMLRPC loadmodule "xmlrpc.so" #!endif # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- registrar params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ #modparam("registrar", "max_contacts", 10) # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default ww do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") modparam("auth_db", "use_domain", MULTIDOMAIN) # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) # use caching modparam("domain", "db_mode", 1) # register callback to match myself condition with domains list modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org<sip%3Apinger(a)kamailio.org> ") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/usr/local/etc/kamailio/tls.cfg") #!endif #!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 16) modparam("pike", "remove_latency", 4) # ----- htable params ----- # ip ban htable with autoexpire after 5 minutes modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") #!endif #!ifdef WITH_XMLRPC # ----- xmlrpc params ----- modparam("xmlrpc", "route", "XMLRPC"); modparam("xmlrpc", "url_match", "^/RPC") #!endif ####### Routing Logic ######## # Main SIP request routing logic # - processing of any incoming SIP request starts with this route route { # per request initial checks route(REQINIT); # NAT detection route(NAT); # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); # authentication route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } # dispatch requests to foreign domains route(SIPOUT); ### requests for my local domains # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } #!ifdef WITH_FREESWITCH # save callee ID $avp(callee) = $rU; route(FSDISPATCH); #!endif # user location service route(LOCATION); route(RELAY); } route[RELAY] { #!ifdef WITH_NAT if (check_route_param("nat=yes")) { setbflag(FLB_NATB); } if (isflagset(FLT_NATS) || isbflagset(FLB_NATB)) { route(RTPPROXY); } #!endif #!ifdef WITH_CFGSAMPLES /* example how to enable some additional event routes */ if (is_method("INVITE")) { #t_on_branch("BRANCH_ONE"); t_on_reply("REPLY_ONE"); t_on_failure("FAIL_ONE"); } #!endif if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood dection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($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()) { if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } 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")) { if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging ## setbflag(FLB_NATSIPPING); } if (!save("location")) sl_reply_error(); exit; } } # USER location service route[LOCATION] { #!ifdef WITH_ALIASDB # search in DB-based aliases alias_db_lookup("dbaliases"); #!endif if (!lookup("location")) { switch ($rc) { case -1: case -3: t_newtran(); t_reply("404", "Not Found"); exit; case -2: sl_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; #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; }; if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); exit; } return; } # Authentication route route[AUTH] { #!ifdef WITH_AUTH if (is_method("REGISTER")) { # authenticate the REGISTER requests (uncomment to enable auth) if (!www_authorize("$td", "subscriber")) { www_challenge("$td", "0"); exit; } if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else { #!ifdef WITH_FREESWITCH if(route(FSINBOUND)) return; #!endif #!ifdef WITH_IPAUTH if(allow_source_address()) { # source IP allowed return; } #!endif # authenticate if from local subscriber if (from_uri==myself) { if (!proxy_authorize("$fd", "subscriber")) { proxy_challenge("$fd", "0"); exit; } if (is_method("PUBLISH")) { if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else { if ($au!=$fU) { sl_send_reply("403","Forbidden auth ID"); exit; } } consume_credentials(); # caller authenticated } else { # caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (!uri==myself) { sl_send_reply("403","Not relaying"); exit; } } } #!endif return; } # Caller NAT detection route route[NAT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (method=="REGISTER") { fix_nated_register(); } else { fix_nated_contact(); } setflag(FLT_NATS); } #!endif return; } # RTPProxy control route[RTPPROXY] { #!ifdef WITH_NAT if (is_method("BYE")) { unforce_rtp_proxy(); } else if (is_method("INVITE")){ force_rtp_proxy(); } if (!has_totag()) add_rr_param(";nat=yes"); #!endif return; } # Routing to foreign domains route[SIPOUT] { if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(RELAY); } } #!ifdef WITH_FREESWITCH # FreeSWITCH routing blocks route[FSINBOUND] { if($si== $sel(cfg_get.freeswitch.bindip) && $sp==$sel(cfg_get.freeswitch.bindport)) return 1; return -1; } route[FSDISPATCH] { if(!is_method("INVITE")) return; if(route(FSINBOUND)) return; # dial number selection switch($rU) { case /"^41$": # 41 - voicebox menu # allow only authenticated users if($au==$null) { sl_send_reply("403", "Not allowed"); exit; } $rU = "vm-" + $au; break; case /"^441[0-9][0-9]$": # starting with 44 folowed by 1XY - direct call to voice box strip(2); route(FSVBOX); break; case /"^433[01][0-9][0-9]$": # starting with 433 folowed by (0|1)XY - conference strip(2); break; case /"^45[0-9]+$": strip(2); break; default: # offline - send to voicebox if (!registered("location")) { route(FSVBOX); exit; } # online - do bridging prefix("kb-"); if(is_method("INVITE")) { # in case of failure - re-route to FreeSWITCH VoiceMail t_on_failure("FAIL_FSVBOX"); } } route(FSRELAY); exit; } route[FSVBOX] { if(!($rU=~"^1[0-9][0-9]+$")) return; prefix("vb-"); route(FSRELAY); } # Send to FreeSWITCH route[FSRELAY] { $du = "sip:" + $sel(cfg_get.freeswitch.bindip) + ":" + $sel(cfg_get.freeswitch.bindport); if($var(newbranch)==1) { append_branch(); $var(newbranch) = 0; } route(RELAY); exit; } #!endif #!ifdef WITH_FREESWITCH failure_route[FAIL_FSVBOX] { #!ifdef WITH_NAT if (is_method("INVITE") && (isbflagset(FLB_NATB) || isflagset(FLT_NATS))) { unforce_rtp_proxy(); } #!endif if (t_is_canceled()) { exit; } if (t_check_status("486|408")) { # re-route to FreeSWITCH VoiceMail $rU = $avp(callee); $var(newbranch) = 1; route(FSVBOX); } } #!endif # sample config file for dispatcher module #!ifdef WITH_DISPATCHER modparam("dispatcher", "db_url", "mysql://openser:password@localhost /openser") route{ if ( !mf_process_maxfwd_header("10") ) { sl_send_reply("483","To Many Hops"); drop(); }; ds_select_dst("1", "0"); forward(); # t_relay(); } #!endif

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Re: [SR-Users] avp_db_query returning wrong data
by Alex Balashov 07 Jan '11

07 Jan '11

On 11/23/2010 06:41 PM, Jijo wrote: > Yes it work with many UDP children. There were some concurrency issues with the Postgres module a while back, discovered after the 3.1.0 release. I remember something about it because we are extremely heavy users of the Postgres module as well. Unfortunately, I can't find the thread. Try the latest git master HEAD: git clone git://git.sip-router.org/sip-router cd sip-router make FLAVOUR=kamailio cfg make all And see if it changes. -- Alex Balashov - Principal Evariste Systems LLC 1170 Peachtree Street NE 12th Floor, Suite 1200 Atlanta, GA 30309 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/

3 4

SIREMIS logging problem.
by Ricardo Martinez 06 Jan '11

06 Jan '11

Hello. I have Siremis 2.0 installed from scratch, but I’m unable to logging. This is the error that I’m getting from the logs : '01/05/2011','12:41:36','ERR','DATAOBJ','Query Error: SQLSTATE[42S02]: Base table or view not found: 1146 Table 'siremis.contact' doesn't exist','n/a' I was checking the SIREMIS db and I can’t found the table “contact”. Is this a bug?.. can someone help me? Thanks Regards, * * *Ricardo Martinez.-*

2 3

Kamailio Development Training, Barcelona, Feb 10-11, 2011
by Daniel-Constantin Mierla 05 Jan '11

05 Jan '11

Hello, for those interested in learning how to write C code for Kamailio & SER, the next development training class takes place in Barcelona, Spain, during February 10-11, 2011. You can find more details at: http://asipto.com/u/r Number of seats is limited, the acceptance policy is first come first served. Cheers, Daniel -- Daniel-Constantin Mierla Kamailio (OpenSER) Advanced Training Jan 24-26, 2011, Irvine, CA, USA http://www.asipto.com

1 0

ENUM changes?
by Klaus Darilion 05 Jan '11

05 Jan '11

Hi! I just found out that my ENUM config didn't worked any more - and it worked for sure some time ago. Were there any recent changes, e.g. from 3.0 to 3.1? I used if enum_query("e164.arpa.","+sip+voice:sip") { which should accept normal E2U+sip and E2U+voice:sip NAPTRs. But it doesn't work with kamailio 3.1.0. But enum_query("e164.arpa.") works. thanks klaus

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Re: [SR-Users] RLS issue
by Klaus Darilion 05 Jan '11

05 Jan '11

1. Kamailio will send the request to the address of the list member (normal DNS lookups). 2. If there is a problem with OpenIMS then you should ask the OpenIMS guys, we are not familiar with OpenIMS. regards klaus Am 22.12.2010 12:57, schrieb "Andrés S. García Ruiz": > > Thanks a lot for your response! > > I've been thinking about it and there's something abnormal. Firs, the > RLS is suppose to send the initial SUBSCRIBE to the S-CSCF instead of > sending it to the I-CSCF. Then, I don't know why the S-CSCF translates > the SUBSCRIBE URI (sip:testuser02@open-ims.test) to > "sip:testuser01@155.54.190.245:8060;rinstance=9b7761b4bcaa4bd0". All the > routing is the default configuration of both cscfs and rls. > > How can I change the routing of the RLS in order it to send the > SUBSCRIBE messages to the S-CSCF? > > Here's the configuration file: > / > #!KAMAILIO > > #!ifdef ACCDB_COMMENT > ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; > ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; > ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; > ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; > ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; > ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL > DEFAULT ''; > ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL > DEFAULT ''; > ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL > DEFAULT ''; > ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL > DEFAULT ''; > ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL > DEFAULT ''; > #!endif > > ####### Defined Values ######### > > #!define WITH_DEBUG > #!define WITH_AUTH > #!define WITH_MYSQL > #!define WITH_USRLOCDB > #!define WITH_PRESENCE > #!define WITH_XCAPSRV > > > # *** Value defines - IDs used later in config > #!ifdef WITH_MYSQL > # - database URL - used to connect to database server by modules such > # as: auth_db, acc, usrloc, a.s.o. > #!define DBURL "mysql://openser:openserrw@localhost/openser" > #!endif > #!ifdef WITH_MULTIDOMAIN > # - the value for 'use_domain' parameters > #!define MULTIDOMAIN 1 > #!else > #!define MULTIDOMAIN 0 > #!endif > > # - flags > # FLT_ - per transaction (message) flags > # FLB_ - per branch flags > #!define FLT_ACC 1 > #!define FLT_ACCMISSED 2 > #!define FLT_ACCFAILED 3 > #!define FLT_NATS 5 > > #!define FLB_NATB 6 > #!define FLB_NATSIPPING 7 > > ####### Global Parameters ######### > > #!ifdef WITH_DEBUG > debug=4 > log_stderror=yes > #!else > debug=2 > log_stderror=no > #!endif > > memdbg=5 > memlog=5 > > log_facility=LOG_LOCAL0 > > fork=yes > children=4 > > /* uncomment the next line to disable TCP (default on) */ > #disable_tcp=yes > > > /* uncomment the next line to disable the auto discovery of local aliases > based on reverse DNS on IPs (default on) */ > #auto_aliases=no > > /* add local domain aliases */ > alias="open-ims.test" > > /* uncomment and configure the following line if you want Kamailio to > bind on a specific interface/port/proto (default bind on all available) */ > #listen=udp:10.0.0.10:5060 > > /* port to listen to > * - can be specified more than once if needed to listen on many ports */ > port=5060 > > #!ifdef WITH_TLS > enable_tls=yes > #!endif > > #!ifdef WITH_XCAPSRV > tcp_accept_no_cl=yes > #!endif > > ####### Custom Parameters ######### > > # These parameters can be modified runtime via RPC interface > # - see the documentation of 'cfg_rpc' module. > # > # Format: group.id = value 'desc' description > # Access: $sel(cfg_get.group.id) or @cfg_get.group.id > # > > #!ifdef WITH_PSTN > # PSTN GW Routing > # > # - pstn.gw_ip: valid IP or hostname as string value, example: > # pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" > # > # - by default is empty to avoid misrouting > pstn.gw_ip = "" desc "PSTN GW Address" > #!endif > > > ####### Modules Section ######## > > # set paths to location of modules > #!ifdef LOCAL_TEST_RUN > mpath="modules_k:modules" > #!else > mpath="/usr/local/kamailio-3.1-xcap-rls/lib/kamailio/modules_k/:/usr/local/kamailio-3.1-xcap-rls/lib/kamailio/modules/" > #!endif > > #!ifdef WITH_MYSQL > loadmodule "db_mysql.so" > #!endif > > 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 "mi_rpc.so" > loadmodule "acc.so" > # loadmodule "pua_mi.so" > # loadmodule "pua_usrloc.so" > > > #!ifdef WITH_AUTH > loadmodule "auth.so" > loadmodule "auth_db.so" > #!ifdef WITH_IPAUTH > loadmodule "permissions.so" > #!endif > #!endif > > #!ifdef WITH_ALIASDB > loadmodule "alias_db.so" > #!endif > > #!ifdef WITH_MULTIDOMAIN > loadmodule "domain.so" > #!endif > > #!ifdef WITH_PRESENCE > loadmodule "presence.so" > loadmodule "presence_xml.so" > loadmodule "pua.so" > loadmodule "rls.so" > #!endif > > #!ifdef WITH_NAT > loadmodule "nathelper.so" > loadmodule "rtpproxy.so" > #!endif > > #!ifdef WITH_TLS > loadmodule "tls.so" > #!endif > > #!ifdef WITH_ANTIFLOOD > loadmodule "htable.so" > loadmodule "pike.so" > #!endif > > #!ifdef WITH_XMLRPC > loadmodule "xmlrpc.so" > #!endif > > #!ifdef WITH_XCAPSRV > loadmodule "xhttp.so" > loadmodule "xcap_server.so" > #!endif > > # ----------------- setting module-specific parameters --------------- > > > # ----- mi_fifo params ----- > modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") > > > # ----- tm params ----- > # auto-discard branches from previous serial forking leg > modparam("tm", "failure_reply_mode", 3) > # default retransmission timeout: 30sec > modparam("tm", "fr_timer", 30000) > # default invite retransmission timeout after 1xx: 120sec > modparam("tm", "fr_inv_timer", 120000) > > > # ----- rr params ----- > # add value to ;lr param to cope with most of the UAs > modparam("rr", "enable_full_lr", 1) > # do not append from tag to the RR (no need for this script) > modparam("rr", "append_fromtag", 0) > > > # ----- registrar params ----- > modparam("registrar", "method_filtering", 1) > /* uncomment the next line to disable parallel forking via location */ > # modparam("registrar", "append_branches", 0) > /* uncomment the next line not to allow more than 10 contacts per AOR */ > #modparam("registrar", "max_contacts", 10) > > > # ----- acc params ----- > /* what special events should be accounted ? */ > modparam("acc", "early_media", 0) > modparam("acc", "report_ack", 0) > modparam("acc", "report_cancels", 0) > /* by default ww do not adjust the direct of the sequential requests. > if you enable this parameter, be sure the enable "append_fromtag" > in "rr" module */ > modparam("acc", "detect_direction", 0) > /* account triggers (flags) */ > modparam("acc", "log_flag", FLT_ACC) > modparam("acc", "log_missed_flag", FLT_ACCMISSED) > modparam("acc", "log_extra", > "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") > modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) > /* enhanced DB accounting */ > #!ifdef WITH_ACCDB > modparam("acc", "db_flag", FLT_ACC) > modparam("acc", "db_missed_flag", FLT_ACCMISSED) > modparam("acc", "db_url", DBURL) > modparam("acc", "db_extra", > "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") > #!endif > > > # ----- usrloc params ----- > /* enable DB persistency for location entries */ > #!ifdef WITH_USRLOCDB > modparam("usrloc", "db_url", DBURL) > modparam("usrloc", "db_mode", 2) > modparam("usrloc", "use_domain", MULTIDOMAIN) > #!endif > > > # ----- auth_db params ----- > #!ifdef WITH_AUTH > modparam("auth_db", "db_url", DBURL) > modparam("auth_db", "calculate_ha1", yes) > modparam("auth_db", "password_column", "password") > modparam("auth_db", "load_credentials", "") > modparam("auth_db", "use_domain", MULTIDOMAIN) > > # ----- permissions params ----- > #!ifdef WITH_IPAUTH > modparam("permissions", "db_url", DBURL) > modparam("permissions", "db_mode", 1) > #!endif > > #!endif > > > # ----- alias_db params ----- > #!ifdef WITH_ALIASDB > modparam("alias_db", "db_url", DBURL) > modparam("alias_db", "use_domain", MULTIDOMAIN) > #!endif > > > # ----- domain params ----- > #!ifdef WITH_MULTIDOMAIN > modparam("domain", "db_url", DBURL) > # use caching > modparam("domain", "db_mode", 1) > # register callback to match myself condition with domains list > modparam("domain", "register_myself", 1) > #!endif > > > #!ifdef WITH_PRESENCE > # ----- presence params ----- > modparam("presence", "db_url", DBURL) > > # ----- presence_xml params ----- > modparam("presence_xml", "db_url", DBURL) > modparam("presence_xml", "force_active", 1) > modparam("presence_xml", "integrated_xcap_server", 1) > modparam("rls", "db_url", DBURL) > modparam("rls", "xcap_table", "xcap") > modparam("rls", "rlsubs_table", "rls_watchers") > modparam("rls", "rlpres_table", "rls_presentity") > modparam("rls", "clean_period", 100) > modparam("rls", "waitn_time", 50) > modparam("rls", "max_expires", 10800) > modparam("rls", "hash_size", 9) > modparam("rls", "xcap_root", "http://ciervo.inf.um.es/xcap-root:5060") > modparam("rls", "integrated_xcap_server", 1) > modparam("rls", "to_presence_code", 10) > modparam("rls", "rls_event", "presence") > #!endif > > > #!ifdef WITH_NAT > # ----- rtpproxy params ----- > modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") > > # ----- nathelper params ----- > modparam("nathelper", "natping_interval", 30) > modparam("nathelper", "ping_nated_only", 1) > modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) > modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") > > # params needed for NAT traversal in other modules > modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") > modparam("usrloc", "nat_bflag", FLB_NATB) > #!endif > > > #!ifdef WITH_TLS > # ----- tls params ----- > modparam("tls", "config", "/usr/local/etc/kamailio/tls.cfg") > #!endif > > #!ifdef WITH_ANTIFLOOD > # ----- pike params ----- > modparam("pike", "sampling_time_unit", 2) > modparam("pike", "reqs_density_per_unit", 16) > modparam("pike", "remove_latency", 4) > > # ----- htable params ----- > # ip ban htable with autoexpire after 5 minutes > modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") > #!endif > > #!ifdef WITH_XMLRPC > # ----- xmlrpc params ----- > modparam("xmlrpc", "route", "XMLRPC"); > modparam("xmlrpc", "url_match", "^/RPC") > #!endif > > #!ifdef WITH_XCAPSRV > # ----- xcap_server params ----- > modparam("xcap_server", "db_url", DBURL) > # modparam("rls", "integrated_xcap_server", 1) > #!endif > > ####### Routing Logic ######## > > > # Main SIP request routing logic > # - processing of any incoming SIP request starts with this route > route { > > # per request initial checks > route(REQINIT); > > # NAT detection > route(NAT); > > # handle requests within SIP dialogs > route(WITHINDLG); > > ### only initial requests (no To tag) > > # CANCEL processing > if (is_method("CANCEL")) > { > if (t_check_trans()) > t_relay(); > exit; > } > > t_check_trans(); > > # authentication > route(AUTH); > > # 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); > > # user location service > route(LOCATION); > > route(RELAY); > } > > > route[RELAY] { > #!ifdef WITH_NAT > if (check_route_param("nat=yes")) { > setbflag(FLB_NATB); > } > if (isflagset(FLT_NATS) || isbflagset(FLB_NATB)) { > route(RTPPROXY); > } > #!endif > > /* example how to enable some additional event routes */ > if (is_method("INVITE")) { > #t_on_branch("BRANCH_ONE"); > t_on_reply("REPLY_ONE"); > t_on_failure("FAIL_ONE"); > } > > 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()) { > if (is_method("BYE")) { > setflag(FLT_ACC); # do accounting ... > setflag(FLT_ACCFAILED); # ... even if the transaction fails > } > 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")) > { > if(isflagset(FLT_NATS)) > { > setbflag(FLB_NATB); > # uncomment next line to do SIP NAT pinging > ## setbflag(FLB_NATSIPPING); > } > if (!save("location")) > sl_reply_error(); > > exit; > } > } > > # USER location service > route[LOCATION] { > > #!ifdef WITH_ALIASDB > # search in DB-based aliases > alias_db_lookup("dbaliases"); > #!endif > > if (!lookup("location")) { > switch ($rc) { > case -1: > case -3: > t_newtran(); > t_reply("404", "Not Found"); > exit; > case -2: > sl_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; > > #!ifdef WITH_PRESENCE > > if (!t_newtran()) > { > sl_reply_error(); > exit; > }; > > if(is_method("PUBLISH")) > { > handle_publish(); > t_release(); > } > else if( is_method("SUBSCRIBE")) > { > $var(ret_code)= rls_handle_subscribe(); > > if($var(ret_code)== 10) { > handle_subscribe(); > } > t_release(); > > }else if(method=="NOTIFY") > { > rls_handle_notify(); > } > > exit; > #!endif > # if presence enabled, this part will not be executed > if (is_method("PUBLISH") || $rU==$null) > { > sl_send_reply("404", "Not here"); > exit; > } > return; > } > > # Authentication route > route[AUTH] { > return; > #!ifdef WITH_AUTH > if (is_method("REGISTER")) > { > # authenticate the REGISTER requests (uncomment to enable auth) > if (!www_authorize("$td", "subscriber")) > { > www_challenge("$td", "0"); > exit; > } > > if ($au!=$tU) > { > sl_send_reply("403","Forbidden auth ID"); > exit; > } > } else { > > #!ifdef WITH_IPAUTH > if(allow_source_address()) > { > # source IP allowed > return; > } > #!endif > # authenticate if from local subscriber > if (from_uri==myself) > { > if (!proxy_authorize("$fd", "subscriber")) { > proxy_challenge("$fd", "0"); > exit; > } > if (is_method("PUBLISH")) > { > if ($au!=$tU) { > sl_send_reply("403","Forbidden auth ID"); > exit; > } > } else { > if ($au!=$fU) { > sl_send_reply("403","Forbidden auth ID"); > exit; > } > } > > consume_credentials(); > # caller authenticated > } else { > # caller is not local subscriber, then check if it calls > # a local destination, otherwise deny, not an open relay here > if (!uri==myself) > { > sl_send_reply("403","Not relaying"); > exit; > } > } > } > #!endif > return; > } > > # Caller NAT detection route > route[NAT] { > #!ifdef WITH_NAT > force_rport(); > if (nat_uac_test("19")) { > if (method=="REGISTER") { > fix_nated_register(); > } else { > fix_nated_contact(); > } > setflag(FLT_NATS); > } > #!endif > return; > } > > # RTPProxy control > route[RTPPROXY] { > #!ifdef WITH_NAT > if (is_method("BYE")) { > unforce_rtp_proxy(); > } else if (is_method("INVITE")){ > force_rtp_proxy(); > } > if (!has_totag()) add_rr_param(";nat=yes"); > #!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] { > #!ifdef WITH_PSTN > # check if PSTN GW IP is defined > if (strempty($sel(cfg_get.pstn.gw_ip))) { > xlog("SCRIPT: PSTN rotuing enabled but pstn.gw_ip not defined\n"); > return; > } > > # route to PSTN dialed numbers starting with '+' or '00' > # (international format) > # - update the condition to match your dialing rules for PSTN routing > if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) > return; > > # only local users allowed to call > if(from_uri!=myself) { > sl_send_reply("403", "Not Allowed"); > exit; > } > > $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); > > route(RELAY); > exit; > #!endif > > return; > } > > # XMLRPC routing > #!ifdef WITH_XMLRPC > route[XMLRPC] > { > # allow XMLRPC from localhost > if ((method=="POST" || method=="GET") > && (src_ip==127.0.0.1)) { > # close connection only for xmlrpclib user agents (there is a bug in > # xmlrpclib: it waits for EOF before interpreting the response). > if ($hdr(User-Agent) =~ "xmlrpclib") > set_reply_close(); > set_reply_no_connect(); > dispatch_rpc(); > exit; > } > send_reply("403", "Forbidden"); > exit; > } > #!endif > > # Sample branch router > branch_route[BRANCH_ONE] { > xdbg("new branch at $ru\n"); > } > > # Sample onreply route > onreply_route[REPLY_ONE] { > xdbg("incoming reply\n"); > #!ifdef WITH_NAT > if ((isflagset(FLT_NATS) || isbflagset(FLB_NATB)) > && status=~"(183)|(2[0-9][0-9])") { > force_rtp_proxy(); > } > if (isbflagset("6")) { > fix_nated_contact(); > } > #!endif > } > > # Sample failure route > failure_route[FAIL_ONE] { > #!ifdef WITH_NAT > if (is_method("INVITE") > && (isbflagset(FLB_NATB) || isflagset(FLT_NATS))) { > unforce_rtp_proxy(); > } > #!endif > > if (t_is_canceled()) { > exit; > } > > # uncomment the following lines if you want to block client > # redirect based on 3xx replies. > ##if (t_check_status("3[0-9][0-9]")) { > ##t_reply("404","Not found"); > ## exit; > ##} > > # uncomment the following lines if you want to redirect the failed > # calls to a different new destination > ##if (t_check_status("486|408")) { > ## sethostport("192.168.2.100:5060"); > ## append_branch(); > ## # do not set the missed call flag again > ## t_relay(); > ##} > } > > #!ifdef WITH_XCAPSRV > # #!define WITH_XHTTPAUTH > event_route[xhttp:request] { > xdbg("===== xhttp: request [$rv] $rm => $hu\n"); > > #!ifdef WITH_XHTTPAUTH > if (!www_authorize("xcap", "subscriber")) > { > www_challenge("xcap", "0"); > exit; > } > #!endif > xdbg("===== xhttp: Antes del if de $hu\n"); > if($hu=~"^/xcap-root/") > { > xdbg("===== xhttp: Dentro del if de $hu\n"); > set_reply_close(); > set_reply_no_connect(); > # xcap ops > $xcapuri(u=>data) = $hu; > if($xcapuri(u=>xuid)=~"^sip:.+@.+") > $var(uri) = $xcapuri(u=>xuid); > else if($xcapuri(u=>xuid)=~".+@.+") > $var(uri) = "sip:" + $xcapuri(u=>xuid); > else > $var(uri) = "sip:"+ $xcapuri(u=>xuid) + "@" + $Ri; > xlog("===== xhttp: $xcapuri(u=>auid) : $xcapuri(u=>xuid)\n"); > if($xcapuri(u=>auid)=="xcap-caps") > { > $var(xbody) = > "<?xml version='1.0' encoding='UTF-8'?> > <xcap-caps xmlns='urn:ietf:params:xml:ns:xcap-caps'> > <auids> > <auid>rls-services</auid> > <auid>pidf-manipulation</auid> > <auid>xcap-caps</auid> > <auid>resource-lists</auid> > <auid>pres-rules</auid> > <auid>org.openmobilealliance.pres-rules</auid> > </auids> > <extensions> > </extensions> > <namespaces> > <namespace>urn:ietf:params:xml:ns:rls-services</namespace> > <namespace>urn:ietf:params:xml:ns:pidf</namespace> > <namespace>urn:ietf:params:xml:ns:xcap-caps</namespace> > <namespace>urn:ietf:params:xml:ns:resource-lists</namespace> > <namespace>urn:ietf:params:xml:ns:pres-rules</namespace> > </namespaces> > </xcap-caps>"; > xhttp_reply("200", "ok", "application/xcap-caps+xml", > "$var(xbody)"); > exit; > } > > #!ifdef WITH_XHTTPAUTH > # be sure auth user access only its documents > if ($au!=$(var(uri){uri.user})) { > xhttp_reply("403", "Forbidden", "text/html", > "<html><body>$si:$sp</body></html>"); > exit; > } > > #!endif > xdbg("===== xhttp: Antes del switch $rm => $hu\n"); > switch($rm) { > case "PUT": > xcaps_put("$var(uri)", "$hu", "$rb"); > if($xcapuri(u=>auid)=~"pres-rules") > { > xlog("===== xhttp put: refreshing watchers for $var(uri)\n"); > pres_update_watchers("$var(uri)", "presence"); > pres_refresh_watchers("$var(uri)", "presence", 1); > } > exit; > break; > case "GET": > xlog("===== xhttp: get $var(uri) => $hu\n"); > xcaps_get("$var(uri)", "$hu"); > exit; > break; > case "DELETE": > xcaps_del("$var(uri)", "$hu"); > if($xcapuri(u=>auid)=~"pres-rules") > { > xlog("===== xhttp del: refreshing watchers for $var(uri)\n"); > pres_update_watchers("$var(uri)", "presence"); > pres_refresh_watchers("$var(uri)", "presence", 1); > } > exit; > break; > } > } > > # http ops > xhttp_reply("200", "ok", "text/html", > "<html><body>OK: $si:$sp</body></html>"); > exit; > } > #!endif/ > > > Regards, > Andrés. > > El 21/12/10 10:22, Klaus Darilion escribió: >> Am 16.12.2010 13:36, schrieb "Andrés S. García Ruiz": >>> >>> B.5) P-CSCF ---- SUBSCRIBE ----> Presentity??? >>> >>> SUBSCRIBE sip:testuser01@155.54.190.245:8060;rinstance=9b7761b4bcaa4bd0 >>> SIP/2.0 >>> Record-Route: <sip:mt@pcscf.open-ims.test:4060;lr> >>> Record-Route: <sip:mt@scscf.open-ims.test:6060;lr> >>> Via: SIP/2.0/TCP 155.54.210.134:4060;branch=z9hG4bK07df.baa7cf24.0 >>> Via: SIP/2.0/UDP >>> 155.54.210.135:6060;received=155.54.210.135;rport=6060;branch=z9hG4bK07df.8a35e4f3.0 >>> >>> >>> Via: SIP/2.0/UDP 155.54.210.136;branch=z9hG4bK07df.3688b985.0 >>> Via: SIP/2.0/UDP 155.54.190.245;branch=z9hG4bK07df.964e0ba7.0 >>> To: sip:testuser01@open-ims.test >>> From: >>> sip:restricted_areas@open-ims.test;tag=533cb9e91f4b999cf76861cbb9ed54ed-32d5 >>> >>> >>> CSeq: 10 SUBSCRIBE >>> Call-ID: 7fd8dfdd-21694(a)127.0.0.1 >>> Content-Length: 0 >>> User-Agent: kamailio (3.2.0-dev1 (i386/linux)) >>> Max-Forwards: 14 >>> Event: presence >>> Contact: <sip:155.54.190.245:5060;transport=udp> >>> Expires: 10810 >>> Max-Forwards: 70 >>> Support: eventlist >>> P-Called-Party-ID: <sip:testuser01@open-ims.test> >>> >>> >>> The step B.5 is sent directly to the presentity testuser01. Instead of >>> that, I thought that message was suppose to be sent to the presence >>> server in the URI that is shown in Contact header. >> >> >> If the P-CSCF is not forwarding the request as supposed, I guess you >> have to fix the routing on the P-CSCF. >> >> klaus >

4 16

Re: [SR-Users] sr-users Digest, Vol 68, Issue 7
by Javier Gallart 05 Jan '11

05 Jan '11

Thanks Alex, it looks like we'll have to move to mysql in that case (we are much more experienced in postgres). Is this feature expected to be added for db_postgres? Regards Javi > > Message: 6 > Date: Tue, 04 Jan 2011 11:54:20 -0500 > From: Alex Balashov <abalashov(a)evaristesys.com> > Subject: Re: [SR-Users] Reconnection to db > To: sr-users(a)lists.sip-router.org > Message-ID: <4D2350BC.8080207(a)evaristesys.com> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > On 01/04/2011 09:46 AM, Javier Gallart wrote: > > Hi list > > > > we have been testing the mtree module. We do periodical reloads of the > > mtree table with the "mi mt_reload" command through xmlrpc. It works > > fine except when the db is restarted. If we manually restart the db and > > then try the mt_reload commands we get this error: > > Jan 4 09:13:17 server /usr/local/kamailio-3.1.1/sbin/kamailio[28534]: > > ERROR: db_postgres [km_dbase.c:152]: 0xa6a490 PQsendQuery Error: server > > closed the connection unexpectedly#012#011This probably means the server > > terminated abnormally#012#011before or while processing the request.#012 > > Query: select partition,prefix,route from trees order by partition > > Jan 4 09:13:17 server /usr/local/kamailio-3.1.1/sbin/kamailio[28534]: > > ERROR: <core> [db_query.c:101]: error while submitting query > > Jan 4 09:13:17 server /usr/local/kamailio-3.1.1/sbin/kamailio[28534]: > > ERROR: mtree [mtree_mod.c:622]: Error while querying db > > Jan 4 09:13:17 server /usr/local/kamailio-3.1.1/sbin/kamailio[28534]: > > ERROR: mtree [mtree_mod.c:738]: cannot re-load info from database > > > > Is there a way to reconnect to the db before sending that command? > > Failure to automatically reconnect to Postgres is one of the > distinguishing features of db_postgres in contrast to db_mysql for the > moment. > > -- > Alex Balashov - Principal > Evariste Systems LLC > 260 Peachtree Street NW > Suite 2200 > Atlanta, GA 30303 > Tel: +1-678-954-0670 > Fax: +1-404-961-1892 > Web: http://www.evaristesys.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 68, Issue 7 > *************************************** >

2 1

Use of SIPP along with kamailio - REGISTER followed by INVITE not working
by Stephen McVarnock 05 Jan '11

05 Jan '11

Guys, I am not sure in which world to expect to get an answer to this so I am sending this to both SIPP and SR-User mailing lists. I am trying to get an SIPP end-point to register with a kamailio proxy. I do this by sending the REGISTER from the SIPP end-point, which is successfully responded to by kamailio with a 200 OK. No problems there. The problem is then with sending an INVITE from kamailio to the SIPP instance. This is where the problems arise. 1) If I include the expected INVITE in the same xml scenario file as the REGISTER/200, then SIPP will complain about the INVITE as it will have a different Call-ID from the REGISTER method. I have a hunt around the forums and this was acknowledged as an issue but as far as I can see this has not been fixed/implemented. The following link will take you to a proposed fix for this issue. http://sipp.sourceforge.net/wiki/index.php/Patches#Pre.2FPost_scenarios Does anyone know what the current state of play is for this proposed patch or if there is another way to get around this issue? 2) I tried to REGISTER the SIPP endpoint in a single xml scenario file with kamailio. This works as per usual. I then killed the SIPP instance and ran a new SIPP script listening on the same port before trying to send the INVITE to it. I expected this to work as the SIPP scenarios (both sending REGISTER and expecting INVITE) listened on the same port. However, the INVITE was not received by the SIPP endpoint. Can anyone think of a reason for this? If any further info is required then please let me know. I have spent quite a bit of time on this and its an itch that needs scratched. Thanks in advance, Steve. -- ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ Stephen McVarnock, AePONA Ltd, Interpoint Building, 20-24 York Street, Belfast BT15 1AQ Tel: +44 (0)28 9026 9114 Fax: +44 (0)28 9026 9111 http://www.aepona.com ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~

3 3

Faxing With Kamailio
by Amit Nepal 04 Jan '11

04 Jan '11

I was trying to use fax with kamailio / asterisk setup and sipce if possible. In asterisk we had to enable udptl for faxing to work. But i couldnot figure out where to start with for faxing in kamailio /asterisk or sipce. Could someone guide me to the right direction please ? Thank you Amit

2 1

Support for via ;maddr
by Olle E. Johansson 04 Jan '11

04 Jan '11

Friends, We've had an interesting discussion on the Asterisk-dev mailing list about supporting the ;maddr and ;ttl attributes in the via header when sending responses. We've agreed that it should be considered harmful and suggest making it configurable whether to support it in Asterisk. My question now is - Does kamailio support this automatically? - Can I disable it? Regards, /O

4 3

PATCH: parallel debian build support
by Noa Resare 04 Jan '11

04 Jan '11

Friends, The attached patch implements parallel building of the debian packages using the -j parameter to make. Predictably this gives package building a serious speed boost on an 8 way xeon with solid state drives :) To make things even simpler, I've pushed a branch debian-packaging to git:// git.resare.com/kamailio that you can pull from and merge with master using the following commands: git remote add resare git://git.resare.com/kamailio git fetch resare <check out the resare/debian-packaging branch, inspect> <merge it into master> A note on the patch: - I've replaced the DEB_BUILD_OPTIONS:="$(DEB_BUILD_OPTIONS) nostrip" (which puts literal quote characters in the make variable with a construct using the += operator. -It passes the basic test of building packages without failing with the DEB_BUILD_OPTIONS environment variable set to parallel=8 as well as unset on squeeze and lenny. /noa -- Everything is secret.

3 4

Re: [SR-Users] [Sipp-users] Use of SIPP along with kamailio - REGISTER followed by INVITE not working
by Stephen McVarnock 04 Jan '11

04 Jan '11

Hi, I have got the second scenario here to work i.e. REGISTER xml ran, kill sipp, run sipp with INVITE xml. There seems to be timing issue associated with this though - if I leave to long a delay between killing REGISTER xml and running INVITE xml then the INVITE will not be received by the sipp script. The Expires header for the REGISTER is set to 36000, so that is not the issue. Maybe this is a kamalio problem - anyway, for my purposes (as long as I am quick enough!), this works. Regards, Steve. > On 2011-01-04 08:59, Stephen McVarnock wrote: > >> http://sipp.sourceforge.net/wiki/index.php/Patches#Pre.2FPost_scenarios >> >> Does anyone know what the current state of play is for this proposed >> patch or if there is another way to get around this issue? >> > > Well, I developed this extension from june until december 2006 but we > never managed to merge this branch into the main-tree. > > >> 2) I tried to REGISTER the SIPP endpoint in a single xml scenario file >> with kamailio. This works as per usual. I then killed >> the SIPP instance and ran a new SIPP script listening on the same port >> before trying to send the INVITE to it. I expected this to work >> as the SIPP scenarios (both sending REGISTER and expecting INVITE) >> listened on the same port. However, the INVITE was not >> received by the SIPP endpoint. Can anyone think of a reason for this? >> > > Well, this is a standard task of sipp and usually works without any > limitations. Can you send both scnearios (Register and UAS) as well as > the used command line parameters? > > br > Michael > > -- ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ Stephen McVarnock, AePONA Ltd, Interpoint Building, 20-24 York Street, Belfast BT15 1AQ Tel: +44 (0)28 9026 9114 Fax: +44 (0)28 9026 9111 http://www.aepona.com ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~

2 1

Social Networking Event, Irvine, CA - Jan 25, 2011
by Daniel-Constantin Mierla 03 Jan '11

03 Jan '11

Hello, since I will spend some days in California this month, it is a good opportunity to host the first Social Networking Event for Kamailio & SER in 2011. I posted more details about it on Kamailio's project site, short link: * http://asipto.com/u/q Event is open for anyone interested to meet folks around the project or SIP and VoIP. Drop me an email if you want to participate. As for Europe, there will be most probably two similar events, one in Brussels (during Fosdem) and one in Barcelona -- more details to be posted very soon. Looking forward to meeting many of you during 2011. Cheers, Daniel -- Daniel-Constantin Mierla Kamailio (OpenSER) Advanced Training Jan 24-26, 2011, Irvine, CA, USA http://www.asipto.com

1 0

Re: [SR-Users] Change the SIP-To-User
by Bernhard Suttner 03 Jan '11

03 Jan '11

Hi, thank you very much for both answers. I did not know the pseudo variable magic, that it is possible to do a $tU = "newvalue" Amazing! Best regards, Bernhard Suttner ----- Original Message ----- From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] To: Bernhard Suttner [mailto:bernhard.suttner@winet.ch] Cc: sr-users(a)lists.sip-router.org Sent: Mon, 03 Jan 2011 18:42:47 +0100 Subject: Re: [SR-Users] Change the SIP-To-User > > > Am 03.01.2011 16:51, schrieb Bernhard Suttner: > > Hi, > > > > what is the best way to replace the userpart of the SIP-To URI with the > content of a pseudo variable? > > textops module: > > e.g. use > http://www.kamailio.org/docs/modules/3.1.x/modules_k/textops.html#id2910302 > > you could also remove_hf("to") and add a new header > append_hf("To: ..."); > > some devices do not like if the To gets changed. In this case you have > to reverse the change in the responses and keep track of cahnges for > in-dialog messages. > > Anyway - at the least the totag must not be changed (if present) > > regards > klaus >

1 0

Replace number in pseudo variable
by Bernhard Suttner 03 Jan '11

03 Jan '11

Hi, I try to replace the user-name of the Request URI and save that into a AVP later. I tried: $var(old_user) = $oU; $var(new_uri) = $(ou{re.subst,/$var(old_user)/004444444444/g}); and $var(new_uri) = $(ou{re.subst,/$oU/004444444444/g}); Both did not work: [lvalue.c:351]: non existing right pvar I am using latest git version of kamailio. Does someone has an idea? Best regards, Bernhard Bernhard Suttner Senior Entwickler Entwicklung _______________________ Winet Network Solutions AG Täfernstrasse 2a CH-5405 Baden-Dättwil E-Mail Winet: info(a)winet.ch E-Mail pers.: bernhard.suttner(a)winet.ch Voice: +41 56 470 4626 Fax: +41 56 470 4627

3 3

Segmentation fault when loading dialplan
by Javier Gallart 03 Jan '11

03 Jan '11

Hello I'd like to do some tests with the dialplan module, but Kamailio is crashing when it loads it. Actually the module is properly loaded if the table is empty, but it fails when a row is added; so I guess it's related with the way I've inserted the values. A core file is generated: Core was generated by `./kamailio -f ../etc/kamailio/kamailio.cfg'. Program terminated with signal 11, Segmentation fault. #0 build_rule (values=0x0) at dp_db.c:439 439 new_rule->dpid = VAL_INT(values); This is the relevant config: #----- dialplan params --- modparam("dialplan", "db_url", "mysql://xxxx:yyyyy@localhost/db") modparam("dialplan", "table_name", "dialplan") modparam("dialplan", "dpid_col", "dpid") modparam("dialplan", "attrs_pvar", "$avp(s:dp_attrs)") mysql> describe dialplan; +-----------+------------------+------+-----+---------+----------------+ | Field | Type | Null | Key | Default | Extra | +-----------+------------------+------+-----+---------+----------------+ | id | int(10) unsigned | NO | PRI | NULL | auto_increment | | dpid | int(11) | NO | | NULL | | | pr | int(11) | NO | | NULL | | | match_op | int(11) | NO | | NULL | | | match_exp | varchar(64) | NO | | NULL | | | match_len | int(11) | NO | | NULL | | | subst_exp | varchar(64) | NO | | NULL | | | repl_exp | varchar(32) | NO | | NULL | | | attrs | varchar(32) | NO | | NULL | | +-----------+------------------+------+-----+---------+----------------+ 9 rows in set (0.00 sec) mysql> select * from dialplan; +----+------+----+----------+-----------+-----------+-----------+----------+-------+ | id | dpid | pr | match_op | match_exp | match_len | subst_exp | repl_exp | attrs | +----+------+----+----------+-----------+-----------+-----------+----------+-------+ | 1 | 1 | 1 | 1 | ^00 | 0 | ^00 | 011 | dp1 | +----+------+----+----------+-----------+-----------+-----------+----------+-------+ 1 row in set (0.00 sec) I'm probably making some stupid mistake...any clue? Thanks in advance Javier

3 21

MI FIFO command dlg_end_dlg in Kamailio 1.5.x
by Alex Balashov 03 Jan '11

03 Jan '11

Greetings, I have not tried in 3.0+, but in Kamailio 1.5.x, the documentation says that the MI FIFO command 'dlg_end_dlg' will only terminate confirmed dialogs. That's fair. However, I am concerned about what it does with unconfirmed dialogs (e.g. early dialogs). When I run 'kamctl fifo dlg_end_dlg SOME_HASH_ENTRY SOME_HASH_ID' it returns a positive value as if the dialog were deleted, but it is not actually deleted. It continues to show up in 'kamctl fifo profile_list_dlgs SOME_PROFILE SOME_VALUE', and continues to be linked to the profile, which is a problem if profiles are being used to enforce concurrent channel limits. Subsequent invocations of 'kamctl fifo dlg_end_dlg SOME_HASH_ENTRY SOME_HASH_ID' return: 404 Requested dialog not found. Is this a bug? A feature? What should one do with these permanent suspended dialogs other than to restart the proxy? Thanks, -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/

2 1

kamailio logging
by Noa Resare 03 Jan '11

03 Jan '11

Is there a way to get a sensible amount of logging done from kamailio? Being new to the VoIP space (but with plenty of experience with i.e. web and email servers) I tried to get a running system by installing the kamailio package and starting up, trying to get it to behave by looking at log output. I use the blink SIP client in OSX, that has a SIP data dump log view and from that I can deduce that the server returns "SIP/2.0 483 Too Many Hops", however there is nothing helpful in the server logs. If I set debug=2 i get nothing useful, and if I set debug=3 I get insane amounts of logging, thousands of lines per connection attempt. Finding something meaningful in there seems like a herculean task. /noa ps. The actual problem with 483 Too Many Hops sounded a lot like 'Mail loops back to myself' that you get on smtp servers when the relevant virtual domain is not configured, so it seems I was able to handle that problem by adding WITH_MULTIDOMAIN and configuring my domain with kamctl domain add. Now I get "482 Loop Detected" instead, which seems like a non-fatal condition. The logging question still stands though. -- Everything is secret.

5 4

Happy New Year!
by Daniel-Constantin Mierla 03 Jan '11

03 Jan '11

Looking back to 2010, it was an amazing year - two major releases v3.0.x and v3.1.0, lot of new features, all in top of a better and more scalable core we have now after the integration of Kamailio with SER. Just to mention few here: asynchronous TCP and TLS, embedded Lua, XCAP server, HTTP server, pre-processor directives, full SCTP implementation with multi-homing, geoip API, asynchronous message queues, over 20 new modules -- for sake of completion, here are the links with the announcements: http://www.kamailio.org/w/kamailio-openser-v3.0.0-release-notes/ http://www.kamailio.org/w/kamailio-openser-v3.1.0-release-notes/ As for 2011, we will continue to deliver new features, but maintain as well our line of scalable, secure and rock solid SIP server. New projects related to Kamailio already announced their launch, participation to VoIP and Open Source events already booked, new thinks are baked and getting ready to launch soon -- 2011 is going to be for sure fascinating as well as challenging. Thanks for your support for the project! I wish you a great 2011 in personal life and business, enjoy tonight party! Happy New Year! Daniel -- Daniel-Constantin Mierla Kamailio (OpenSER) Advanced Training Jan 24-26, 2011, Irvine, CA, USA http://www.asipto.com

2 1

Deploy siremis with nginx
by Adieu 02 Jan '11

02 Jan '11

Hi everybody, I was trying to deploy siremis with nginx last night. Here are some problems I encountered during the installation process. Siremis has apache virtualhost settings. I just read the apache version and wrote nginx version as below: server { listen 80; server_name DOMAIN; index index.html index.htm index.php; root /path/to/siremis; location ~ .*\.(php|php5)?$ { fastcgi_pass 127.0.0.1:9000; fastcgi_index index.php; include fcgi.conf; } location ~^/siremis(.+)$ { try_files $uri $uri/ /siremis/index.php?$1; } } And I had to modify /siremis/bin/_forward.php a little bit to make siremis compatible with nginx. Here is the patch: *** _forward.php.old 2011-01-02 02:59:38.000000000 +0000 --- _forward.php 2011-01-01 16:19:26.000000000 +0000 *************** *** 16,25 **** --- 16,29 ---- if($_SERVER["REDIRECT_QUERY_STRING"]) { $url = $_SERVER["REDIRECT_QUERY_STRING"]; } + elseif($_SERVER["QUERY_STRING"]) + { + $url = $_SERVER["QUERY_STRING"]; + } elseif(preg_match("/\?\/?(.*?)(\.html)?$/si", $_SERVER['REQUEST_URI'],$match)) { //supports for http://localhost/?/user/login format //supports for http://localhost/index.php?/user/login format $url = $match[1]; The rest of the installation process is identical to the apache version from the document. Hope this can be helpful. Ivan Diao

1 0

some issues with database initialization (berkeley/txt)
by Noa Resare 31 Dec '10

31 Dec '10

Hello Friends, I thought I'd set up a SIP server during the holidays and kamailio seemed like a good choice. Since my setup won't be handling large amounts of users, a file based backend seemed like a good choice. Since I didn't find any "new users, please start with these steps" instruction on the web page I just installed the base package for Debian squeeze along with kamailio-berkeley-modules and proceded to try to configure the database. I set DBENGINE=DB_BERKELEY in /etc/kamailio/kamctlrc and ran 'kamdbctl create', which fails with this message: root@uma:/home/noa# kamdbctl create db4.6_load: /usr/share/kamailio//db_berkeley/kamailio/lcr_gw: reopen: No such file or directory ERROR: Creating standard tables failed! I tried DBENGINE=DBTEXT, and the command failed similarly: INFO: creating DBTEXT tables at: /usr/local/etc/kamailio/dbtext ... cp: cannot stat `/usr/share/kamailio//dbtext/kamailio/lcr_gw': No such file or directory ERROR: Creating core tables failed! Based on this I have a few questions: 1) Is there a text somewhere outlining the recommended steps to get a basic server going using the .deb packages? 2) What is the recommended backend for the least amount of install time trouble? 3) Is the issue I got into above a known problem? Is there a fix available? I asked my friend google, but she didn't seem to have an easily findable solution. /noa -- Everything is secret.

3 12

fifo dp_translate
by Borin 31 Dec '10

31 Dec '10

Hello everyone is there a possibility to check working dialplan rules in kamailio (ver 1.4) with the command kamctl fifo dp_translate ? "fifo which" shows that there is such command but I can't figure out how to use it. some rules are not working and I want to check if kamailio loads everything from dialplan table correctly. Mainly I use fifo commands to reload dialplan and dispatcher so fifo is configured right I suppose.

3 7

function get_user_group: parameter 1 is not constant
by Jijo 30 Dec '10

30 Dec '10

I'm getting following config error when i use the api from group module. CSBC-JJ:/srv/www/htdocs/core # /usr/sbin/kamailio -c -f /etc/kamailio/kamailio.cfg loading modules under /usr/lib/kamailio/modules_k/:/usr/lib/kamailio/modules/ 0(5020) : <core> [cfg.y:3412]: parse error in config file /etc/kamailio/kamailio.cfg, line 455, column 20-36: function get_user_group: parameter 1 is not constant 0(5020) : <core> [cfg.y:3415]: parse error in config file /etc/kamailio/kamailio.cfg, line 455, column 48: bad command 0(5020) INFO: pv [pv_shv.c:61]: locks array size 16 ERROR: bad config file (2 errors) I have used the API get_user_group as shown below $avp(r_contact)=$ru; $avp(temp)=$null; get_user_group($avp(reg_contact),$avp(temp)); xlog("L_ERR", "request - M=$rm get_user_group=$avp(temp)\n\n"); Please let me know what could be reason for this error. Thanks Jijo

2 1

SIP CE vmware and vbox image
by Amit Nepal 30 Dec '10

30 Dec '10

I wanted to see how the sipce works and how it looks like , as Andreas suggested me to. But when i try to start the virtual machine i get the following error : This kernel requires an x86-64 CPU, but only detected an i686 CPU. Unable to boot - please use a kernel appropriate for your CPU I tried with both vmware and vbox image. I am trying to run the image on vmware infrastrcture running on dell dual xeon cpu. -- Thank You Amit Nepal Systems Administrator Phoenix Internet Phone: 602-385-0731 602-234-0917#112 http://www.phoenixinternet.net

3 2

Re: [SR-Users] Kamailio 3.1.x and Asterisk 1.6.2 Realtime Integration
by Kurt Mullen 29 Dec '10

29 Dec '10

Apparently the process of the installation did not setup even a default sip.conf file, and there were some other issues with the Kamailio.cfg file. Most of the issues were resolved by one of the other users in the group, and we were able to get the system running. Thanks everyone for your assistance. -----Original Message----- From: sr-users-bounces(a)lists.sip-router.org [mailto:sr-users-bounces@lists.sip-router.org] On Behalf Of sr-users-request(a)lists.sip-router.org Sent: Wednesday, December 29, 2010 11:49 AM To: sr-users(a)lists.sip-router.org Subject: sr-users Digest, Vol 67, Issue 75 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: Kamailio 3.1.x and Asterisk 1.6.2 Realtime Integration using Asterisk Database (Klaus Darilion) 2. Load Balancing (Tim King) 3. Re: Load Balancing (Alex Balashov) 4. fifo dp_translate (Borin) 5. Re: some issues with database initialization (berkeley/txt) (Noa Resare) 6. Re: some issues with database initialization (berkeley/txt) (Daniel-Constantin Mierla) ---------------------------------------------------------------------- Message: 1 Date: Wed, 29 Dec 2010 15:17:41 +0100 From: Klaus Darilion <klaus.mailinglists(a)pernau.at> Subject: Re: [SR-Users] Kamailio 3.1.x and Asterisk 1.6.2 Realtime Integration using Asterisk Database To: sr-users(a)lists.sip-router.org Message-ID: <4D1B4305.30504(a)pernau.at> Content-Type: text/plain; charset=windows-1252; format=flowed On 28.12.2010 22:07, Kurt Mullen wrote: > I am on my 20^th (I?m not kidding) attempt to successfully complete > this tutorial. > > I have installed on Ubuntu 10.10 x64 Server. I installed Kamailio & > Asterisk on the same server as in the tutorial. > > I have two SIP clients registered, but they are not able to call each other. > > No one answered my last two posts, so I hope someone can help this time. If you can describe the problem (why can't they call each other) we can help you. So, first you have to find out what is the problem. Use a packet sniffer (wireshark, ngrep) to capture the SIP traffic and analyze it (yes, if you want to use Kamailio you have to know at least the basics of SIP) for the problems. regards klaus ------------------------------ Message: 2 Date: Wed, 29 Dec 2010 09:31:28 -0500 From: Tim King <tim(a)compnetwork.net> Subject: [SR-Users] Load Balancing To: sr-users(a)lists.sip-router.org Message-ID: <AANLkTindx_r-RmTJSjTnpfdmLX8Nespzs_M9O0f6y385(a)mail.gmail.com> Content-Type: text/plain; charset="iso-8859-1" I have read many examples and it seems I have found several pieces of how to do this but no real example of how to achieve this. I followed this setup guide: http://kb.asipto.com/freeswitch:kamailio-3.1.x-freeswitch-1.0.6d-sbc Performed this Kamailio install: http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-3.1.x-from-git I have found the page for the dispatcher module here: http://kamailio.org/docs/modules/3.1.x/modules_k/dispatcher.html I also am curious if this dispatcher setup can be driven solely from the dispatcher mysql table or if the dispatcher.list file is required. Can someone point me towards the proper documentation that truly show how to configure these features. *SIP Routing Capabilities* - - NAT traversal support for SIP and RTP traffic - load balancing with many distribution algorithms and failover support - flexible least cost routing - routing failover - replication for High Availability (HA) * Load Based Load Balancing ** ------------------------- ** ** +----------+ ** | Kamailio | ** +----------+ ** | ** | +--------------+ ** |-------| FreeSWITCH 1 | [Calls 1,3,7] ** | +--------------+ ** | ** | +--------------+ ** |-------| FreeSWITCH 2 | [Calls 2,6] ** +--------------+ *

2 1

Load Balancing
by Tim King 29 Dec '10

29 Dec '10

I have read many examples and it seems I have found several pieces of how to do this but no real example of how to achieve this. I followed this setup guide: http://kb.asipto.com/freeswitch:kamailio-3.1.x-freeswitch-1.0.6d-sbc Performed this Kamailio install: http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-3.1.x-from-git I have found the page for the dispatcher module here: http://kamailio.org/docs/modules/3.1.x/modules_k/dispatcher.html I also am curious if this dispatcher setup can be driven solely from the dispatcher mysql table or if the dispatcher.list file is required. Can someone point me towards the proper documentation that truly show how to configure these features. *SIP Routing Capabilities* - - NAT traversal support for SIP and RTP traffic - load balancing with many distribution algorithms and failover support - flexible least cost routing - routing failover - replication for High Availability (HA) * Load Based Load Balancing ** ------------------------- ** ** +----------+ ** | Kamailio | ** +----------+ ** | ** | +--------------+ ** |-------| FreeSWITCH 1 | [Calls 1,3,7] ** | +--------------+ ** | ** | +--------------+ ** |-------| FreeSWITCH 2 | [Calls 2,6] ** +--------------+ *

3 3

Kamailio 3.1.x and Asterisk 1.6.2 Realtime Integration using Asterisk Database
by Kurt Mullen 29 Dec '10

29 Dec '10

I am on my 20th (I'm not kidding) attempt to successfully complete this tutorial. I have installed on Ubuntu 10.10 x64 Server. I installed Kamailio & Asterisk on the same server as in the tutorial. I have two SIP clients registered, but they are not able to call each other. No one answered my last two posts, so I hope someone can help this time. Kurt Mullen

3 2

LCR and gateways resource.
by Ricardo Martinez 29 Dec '10

29 Dec '10

Hello. I’m looking for a way to mix the features of LCR and a counter to keep track of the used resource of a gateway. Let me explain. Let’s suppose that I’m using LCR module to route the calls to all my gateways, in particular I have the calls with prefix 55 routed to the group “55” which are 3 gateways : GW1, GW2 and GW3. Let’s suppose also that each gateway is connected to digital E1 with 30 channels. At certain time of the day one of the gateways is at full capacity using the 30 channels. This is what I wanted to do : can I have a variable with the maximum channels available for a certain gateway, and for each call going to that gateway a counter?. When the counter reach the maximum, can I remove the gateway from the LCR choosing algorithm until it has capacity again? Can I have some thoughts about this idea? Is there a way to achieve this ? Thanks in advance. Regards, Ricardo.-

2 3

Swap of From and To - kamailio 3.1.x problem?
by Jeremya 28 Dec '10

28 Dec '10

Hi I have a problem which pops up occasionally. My scenario is a kamailio 3.0.x server coupled with SEMS 1.2.1 Calls go to SEMS and occasionally it will issue a refer to the caller (an SPA962 linksys) The problem is that the refer seems to get screwed up somewhere and the referred call changes the From field to the To field - in this case 711. The problems start below at 15:58:21 after the refer I'd like to know if this is a problem with Kamailio or a problem with the phone. The call fails on the SEMS_SEND section when the From field is validated. I suspect the phone is the culprit, but external advice appreciated. I'm not quite sure what the s_xxx call id means? Here is the log Dec 24 15:58:20 mc0 /usr/local/sbin/kamailio[399]: WARNING: <script>: f43507d4-e02f109b(a)10.1.39.192 (New request) Method:<REFER> r-uri <sip:711@10.50.20.10:5070;LINEID=fac9f6b7cb08> To: sip:711@joondalup.fesa.sto From: sip:902@fesa.sto Dec 24 15:58:20 mc0 /usr/local/sbin/kamailio[399]: WARNING: <script>: f43507d4-e02f109b(a)10.1.39.192 ( reply etc ) Method:<REFER> r-uri <sip:711@10.50.20.10:5070;LINEID=fac9f6b7cb08> To: sip:711@joondalup.fesa.sto From: sip:902@fesa.sto Dec 24 15:58:20 mc0 /usr/local/sbin/kamailio[399]: WARNING: <script>: f43507d4-e02f109b(a)10.1.39.192 (Loose route) Method:<REFER> r-uri <sip:711@10.50.20.10:5070;LINEID=fac9f6b7cb08> To: sip:711@joondalup.fesa.sto From: sip:902@fesa.sto Dec 24 15:58:20 mc0 /usr/local/sbin/kamailio[399]: WARNING: <script>: f43507d4-e02f109b(a)10.1.39.192 (RELAY, start of processing) Method:<REFER> r-uri <sip:711@10.50.20.10:5070;LINEID=fac9f6b7cb08> To: sip:711@joondalup.fesa.sto From: sip:902@fesa.sto Dec 24 15:58:21 mc0 /usr/local/sbin/kamailio[399]: WARNING: <script>: s_275259592d_358d20ea49bf (New request) Method:<INVITE> r-uri <sip:7020@fesa.sto> To: sip:7020@fesa.sto From: sip:711@10.50.20.10:5070;LINEID=fac9f6b7cb08 Dec 24 15:58:21 mc0 /usr/local/sbin/kamailio[399]: WARNING: <script>: s_275259592d_358d20ea49bf (handle initial request) Method:<INVITE> r-uri <sip:7020@fesa.sto> To: sip:7020@fesa.sto From: sip:711@10.50.20.10:5070;LINEID=fac9f6b7cb08 Dec 24 15:58:21 mc0 /usr/local/sbin/kamailio[399]: WARNING: <script>: s_275259592d_358d20ea49bf (t_check_trans) Method:<INVITE> r-uri <sip:7020@fesa.sto> To: sip:7020@fesa.sto From: sip:711@10.50.20.10:5070;LINEID=fac9f6b7cb08 Dec 24 15:58:21 mc0 /usr/local/sbin/kamailio[399]: WARNING: <script>: s_275259592d_358d20ea49bf (Request For Me) Method:<INVITE> r-uri <sip:7020@fesa.sto> To: sip:7020@fesa.sto From: sip:711@10.50.20.10:5070;LINEID=fac9f6b7cb08 Dec 24 15:58:21 mc0 /usr/local/sbin/kamailio[399]: WARNING: <script>: s_275259592d_358d20ea49bf (SEMS, rewrites) Method:<INVITE> r-uri <sip:020@fesa.sto> To: sip:7020@fesa.sto From: sip:711@10.50.20.10:5070;LINEID=fac9f6b7cb08 Dec 24 15:58:21 mc0 /usr/local/sbin/kamailio[399]: WARNING: <script>: s_275259592d_358d20ea49bf (SEMS_SEND) Method:<INVITE> r-uri <sip:020@fesa.sto> To: sip:7020@fesa.sto From: sip:711@10.50.20.10:5070;LINEID=fac9f6b7cb08 Dec 24 15:58:21 mc0 /usr/local/sbin/kamailio[398]: WARNING: <script>: s_275259592d_358d20ea49bf (SEMS_SEND_REPLY) Method:<INVITE> Status:<603> Dec 24 15:58:21 mc0 /usr/local/sbin/kamailio[398]: CRITICAL: <script>: s_275259592d_358d20ea49bf (SEMS_SEND_FAIL) Method:<INVITE> Status:<<null>> Dec 24 15:58:21 mc0 /usr/local/sbin/kamailio[404]: WARNING: <script>: s_275259592d_358d20ea49bf (New request) Method:<ACK> r-uri <sip:7020@fesa.sto> To: sip:7020@fesa.sto From: sip:711@10.50.20.10:5070;LINEID=fac9f6b7cb08 Dec 24 15:58:21 mc0 /usr/local/sbin/kamailio[404]: WARNING: <script>: s_275259592d_358d20ea49bf ( reply etc ) Method:<ACK> r-uri <sip:7020@fesa.sto> To: sip:7020@fesa.sto From: sip:711@10.50.20.10:5070;LINEID=fac9f6b7cb08 Dec 24 15:58:21 mc0 /usr/local/sbin/kamailio[404]: WARNING: <script>: s_275259592d_358d20ea49bf (Not a loose route) Method:<ACK> r-uri <sip:7020@fesa.sto> To: sip:7020@fesa.sto From: sip:711@10.50.20.10:5070;LINEID=fac9f6b7cb08 Dec 24 15:58:21 mc0 /usr/local/sbin/kamailio[404]: WARNING: <script>: s_275259592d_358d20ea49bf (ACK in lr) Method:<ACK> r-uri <sip:7020@fesa.sto> To: sip:7020@fesa.sto From: sip:711@10.50.20.10:5070;LINEID=fac9f6b7cb08

2 1

Crash
by michel freiha 28 Dec '10

28 Dec '10

Hi all, I have latest version of kamailio but it keep crashing...I know that the log could not be sufficient but maybe anyone faced the same issue before...Please find log below and let me know if you can find anything Nov 24 15:56:23 tunnel /usr/local/sbin/kamailio[19943]: : <core> [pass_fd.c:293]: ERROR: receive_fd: EOF on 15 Nov 24 15:56:23 tunnel /usr/local/sbin/kamailio[19929]: ALERT: <core> [main.c:741]: child process 19935 exited by a signal 11 Nov 24 15:56:23 tunnel /usr/local/sbin/kamailio[19932]: INFO: carrierroute [cr_func.c:595]: uri 919437452006 was rewritten to sip:919437452006@x.x.x.x, carrier 5, domain 0 Nov 24 15:56:24 tunnel /usr/local/sbin/kamailio[19930]: INFO: carrierroute [cr_func.c:595]: uri 008801823703200 was rewritten to sip:008801823703200@X.X.X.5, carrier 5, domain 0 Nov 24 15:56:24 tunnel /usr/local/sbin/kamailio[19933]: ERROR: <script>: CARRIERROUTE_time_dbg:method=INVITE;time=1290614183;callid=14b6536a-d102-1910-9b23-9f3b3c3065ca - before cr Nov 24 15:56:24 tunnel /usr/local/sbin/kamailio[19929]: ALERT: <core> [main.c:744]: core was generated Nov 24 15:56:24 tunnel /usr/local/sbin/kamailio[19932]: INFO: <script>: time_dbg:method=SUBSCRIBE;time=1290614183;callid=YjVmODYzYjQxYWY0ZTU3MGRmMmZkYjRhOTg4ZGZmODQ. - before cr Nov 24 15:56:24 tunnel /usr/local/sbin/kamailio[19930]: INFO: <script>: time_dbg:method=INVITE;time=1290614183;callid=cdc9556a-d102-1910-8b30-926b495dad44 - before cr Nov 24 15:56:24 tunnel /usr/local/sbin/kamailio[19933]: INFO: carrierroute [cr_func.c:595]: uri 00919974265815 was rewritten to sip:00919974265815@X.X.X.X>, carrier 5, domain 0 Nov 24 15:56:24 tunnel /usr/local/sbin/kamailio[19931]: ERROR: <script>: time_dbg:method=INVITE;time=1290614182;callid=14b6536a-d102-1910-9b23-9f3b3c3065ca - before t_relay Nov 24 15:56:24 tunnel /usr/local/sbin/kamailio[19929]: INFO: <core> [main.c:756]: INFO: terminating due to SIGCHLD Nov 24 15:56:24 tunnel /usr/local/sbin/kamailio[19934]: INFO: <core> [main.c:807]: INFO: signal 15 received Nov 24 15:56:24 tunnel /usr/local/sbin/kamailio[19943]: INFO: <core> [main.c:807]: INFO: signal 15 received Nov 24 15:56:24 tunnel /usr/local/sbin/kamailio[19942]: INFO: <core> [main.c:807]: INFO: signal 15 received Nov 24 15:56:24 tunnel /usr/local/sbin/kamailio[19940]: INFO: <core> [main.c:807]: INFO: signal 15 received Nov 24 15:56:24 tunnel /usr/local/sbin/kamailio[19939]: INFO: <core> [main.c:807]: INFO: signal 15 received Nov 24 15:56:24 tunnel /usr/local/sbin/kamailio[19938]: INFO: <core> [main.c:807]: INFO: signal 15 received Nov 24 15:56:24 tunnel /usr/local/sbin/kamailio[19941]: INFO: <core> [main.c:807]: INFO: signal 15 received Nov 24 15:56:24 tunnel /usr/local/sbin/kamailio[19937]: INFO: <core> [main.c:807]: INFO: signal 15 received Nov 24 15:56:24 tunnel /usr/local/sbin/kamailio[19936]: INFO: <core> [main.c:807]: INFO: signal 15 received Regards

4 21

Merry Christmas!
by Daniel-Constantin Mierla 27 Dec '10

27 Dec '10

Hello, I could safety say that 2010 was the greatest year by achievements in the history of our SIP proxy server project so far. It will not be short to summarize it, so I will do a separate post for that. Now I just wast to thank everyone making that possible and wish you a Merry Christmas! Great winter holidays to all Kamailians and SERians!!! Cheers, Daniel -- Daniel-Constantin Mierla Kamailio (OpenSER) Advanced Training Jan 24-26, 2011, Irvine, CA, USA http://www.asipto.com

3 3

Puzzling swap of To and From - Kamailio 3.0x
by Jeremy Ardley 24 Dec '10

24 Dec '10

Hi I have a problem which pops up occasionally. My scenario is a kamailio 3.0.x server coupled with SEMS 1.2.1 Calls go to SEMS and occasionally it will issue a refer to the caller (an SPA962 linksys) The problem is that the refer seems to get screwed up somewhere and the new call changes the From field to the To field - in this case 711. The problems start below at 15:58:21 after the refer I'd like to know if this is a problem with Kamailio or a problem with the phone. The call fails on the SEMS_SEND section when the From field is validated. I suspect the phone is the culprit, but external advice appreciated. Here is the log Dec 24 15:58:20 mc0 /usr/local/sbin/kamailio[399]: WARNING: <script>: f43507d4-e02f109b(a)10.1.39.192 (New request) Method:<REFER> r-uri <sip:711@10.50.20.10:5070;LINEID=fac9f6b7cb08> To: sip:711@joondalup.fesa.sto From: sip:902@fesa.sto Dec 24 15:58:20 mc0 /usr/local/sbin/kamailio[399]: WARNING: <script>: f43507d4-e02f109b(a)10.1.39.192 ( reply etc ) Method:<REFER> r-uri <sip:711@10.50.20.10:5070;LINEID=fac9f6b7cb08> To: sip:711@joondalup.fesa.sto From: sip:902@fesa.sto Dec 24 15:58:20 mc0 /usr/local/sbin/kamailio[399]: WARNING: <script>: f43507d4-e02f109b(a)10.1.39.192 (Loose route) Method:<REFER> r-uri <sip:711@10.50.20.10:5070;LINEID=fac9f6b7cb08> To: sip:711@joondalup.fesa.sto From: sip:902@fesa.sto Dec 24 15:58:20 mc0 /usr/local/sbin/kamailio[399]: WARNING: <script>: f43507d4-e02f109b(a)10.1.39.192 (RELAY, start of processing) Method:<REFER> r-uri <sip:711@10.50.20.10:5070;LINEID=fac9f6b7cb08> To: sip:711@joondalup.fesa.sto From: sip:902@fesa.sto Dec 24 15:58:21 mc0 /usr/local/sbin/kamailio[399]: WARNING: <script>: s_275259592d_358d20ea49bf (New request) Method:<INVITE> r-uri <sip:7020@fesa.sto> To: sip:7020@fesa.sto From: sip:711@10.50.20.10:5070;LINEID=fac9f6b7cb08 Dec 24 15:58:21 mc0 /usr/local/sbin/kamailio[399]: WARNING: <script>: s_275259592d_358d20ea49bf (handle initial request) Method:<INVITE> r-uri <sip:7020@fesa.sto> To: sip:7020@fesa.sto From: sip:711@10.50.20.10:5070;LINEID=fac9f6b7cb08 Dec 24 15:58:21 mc0 /usr/local/sbin/kamailio[399]: WARNING: <script>: s_275259592d_358d20ea49bf (t_check_trans) Method:<INVITE> r-uri <sip:7020@fesa.sto> To: sip:7020@fesa.sto From: sip:711@10.50.20.10:5070;LINEID=fac9f6b7cb08 Dec 24 15:58:21 mc0 /usr/local/sbin/kamailio[399]: WARNING: <script>: s_275259592d_358d20ea49bf (Request For Me) Method:<INVITE> r-uri <sip:7020@fesa.sto> To: sip:7020@fesa.sto From: sip:711@10.50.20.10:5070;LINEID=fac9f6b7cb08 Dec 24 15:58:21 mc0 /usr/local/sbin/kamailio[399]: WARNING: <script>: s_275259592d_358d20ea49bf (SEMS, rewrites) Method:<INVITE> r-uri <sip:020@fesa.sto> To: sip:7020@fesa.sto From: sip:711@10.50.20.10:5070;LINEID=fac9f6b7cb08 Dec 24 15:58:21 mc0 /usr/local/sbin/kamailio[399]: WARNING: <script>: s_275259592d_358d20ea49bf (SEMS_SEND) Method:<INVITE> r-uri <sip:020@fesa.sto> To: sip:7020@fesa.sto From: sip:711@10.50.20.10:5070;LINEID=fac9f6b7cb08 Dec 24 15:58:21 mc0 /usr/local/sbin/kamailio[398]: WARNING: <script>: s_275259592d_358d20ea49bf (SEMS_SEND_REPLY) Method:<INVITE> Status:<603> Dec 24 15:58:21 mc0 /usr/local/sbin/kamailio[398]: CRITICAL: <script>: s_275259592d_358d20ea49bf (SEMS_SEND_FAIL) Method:<INVITE> Status:<<null>> Dec 24 15:58:21 mc0 /usr/local/sbin/kamailio[404]: WARNING: <script>: s_275259592d_358d20ea49bf (New request) Method:<ACK> r-uri <sip:7020@fesa.sto> To: sip:7020@fesa.sto From: sip:711@10.50.20.10:5070;LINEID=fac9f6b7cb08 Dec 24 15:58:21 mc0 /usr/local/sbin/kamailio[404]: WARNING: <script>: s_275259592d_358d20ea49bf ( reply etc ) Method:<ACK> r-uri <sip:7020@fesa.sto> To: sip:7020@fesa.sto From: sip:711@10.50.20.10:5070;LINEID=fac9f6b7cb08 Dec 24 15:58:21 mc0 /usr/local/sbin/kamailio[404]: WARNING: <script>: s_275259592d_358d20ea49bf (Not a loose route) Method:<ACK> r-uri <sip:7020@fesa.sto> To: sip:7020@fesa.sto From: sip:711@10.50.20.10:5070;LINEID=fac9f6b7cb08 Dec 24 15:58:21 mc0 /usr/local/sbin/kamailio[404]: WARNING: <script>: s_275259592d_358d20ea49bf (ACK in lr) Method:<ACK> r-uri <sip:7020@fesa.sto> To: sip:7020@fesa.sto From: sip:711@10.50.20.10:5070;LINEID=fac9f6b7cb08

1 0

Billing
by Deon Vermeulen 24 Dec '10

24 Dec '10

Good Day list Is there someone that could guide me to installation guide for jBilling or recommend any other Free OpenSource Billing? Features Looking for: - Wholesale - Rating - Pre-Paid/Post-Paid Thank you very much Regards Deon

2 1

Source IP address in Failure_route
by Ricardo Martinez 24 Dec '10

24 Dec '10

Hello. I’m trying to obtain the source IP address from a reply message. This is part of the configuration file : failure_route[1] { if (t_was_cancelled()) { xlog("L_INFO", "[$ci] $rm:$ru t_was_cancelled in failure_route\n"); exit; } if ( ($T_rpl($si) != "10.0.0.4") || ($T_rpl($si) != "10.0.0.3") ) { if ( t_check_status("48[0-9]") ) { xlog("L_INFO","[$ci] 48X Received\n"); exit; } } … But it seems not to be working. Is there a way to do what I want? Thanks in advance, Regards, Ricardo.- * *

4 5

[dialog] No dialog stored if Via has no branch
by Iñaki Baz Castillo 23 Dec '10

23 Dec '10

Hi, using Kamailio 1.5.4. I receive llamadas from a Cisco pre-RFC 3261 (non symmetric SIP and no ;branch parameter). It seems that dialog module cannot account such calls. Maybe branch parameter is required? -- Iñaki Baz Castillo <ibc(a)aliax.net>

2 4

kamailio restart and TLS ( relay_to_tls() )
by Juha Heinanen 22 Dec '10

22 Dec '10

Of course this won't work if you route to the other proxies also using lookup() with permanent entries in location table. If you really have such a setup you can try to set a certain flag for this entries and then do not call set_forward_no_connect(). klaus, i checked and i'm already calling set_forward_no_connect() when initial request is forwarded to a tcp contact. however, it seems more tricky to prevent useless tcp connection attempts for in-dialog requests. for example, here the UA has gone away when presence server sends a notify to it: Dec 22 14:50:48 sip /usr/sbin/sip-proxy[3207]: INFO: Routing in-dialog NOTIFY <sip:gmuodqxs@192.98.102.10:50279;transport=tcp> from <sip:jh@vm.test.fi> to <sip:192.98.102.10:60550;transport=tcp> Dec 22 14:50:48 sip /usr/sbin/sip-proxy[3207]: ERROR: <core> [tcp_main.c:2743]: connect 192.98.102.10:60550 failed (RST) Connection refused Dec 22 14:50:48 sip /usr/sbin/sip-proxy[3207]: ERROR: <core> [tcp_main.c:2753]: 192.98.102.10:60550: connect & send for 0xb2d1c490 failed: Connection refused (111) Dec 22 14:50:48 sip /usr/sbin/sip-proxy[3207]: ERROR: tm [../../forward.h:169]: msg_send: ERROR: tcp_send failed Dec 22 14:50:48 sip /usr/sbin/sip-proxy[3207]: ERROR: tm [t_fwd.c:1382]: ERROR: t_send_branch: sending request on branch 0 failed Dec 22 14:50:48 sip /usr/sbin/sip-proxy[3207]: INFO: Failed to relay in-dialog request NOTIFY <sip:gmuodqxs@192.98.102.10:50279;transport=tcp> it does not seem like a good idea to call set_forward_no_connect() for all in-dialog requests. for example, presence server might have been restarted when in-dialog subscribe arrives to it and it would make sense to for sr to try to setup a tcp session with presence server. same for pstn gateways and other proxies. -- juha

2 1

RTP to SRTP bridge
by Dominguez Jover, Ricardo 22 Dec '10

22 Dec '10

Hi everybody, I'm trying to deploy an scenario where all calls in the LAN are encrypted with SRTP, so I've forced my sip softphones to "mandatory SRTP". As in the Internet not all providers and clients have SRTP enabled, I couldn´t communicate them in this way, so I would need some kind of border element to bridge RTP and SRTP calls. Is there any bridge/proxy who can receive rtp packets and convert them into srtp and viceversa? I've been reading RTPproxy and Media proxy docs and also some forums, but I didn´t find anything useful. Regards, Ricardo Dominguez

2 1

Re: [SR-Users] 404 Not Found - If User Not Registered
by Klaus Darilion 22 Dec '10

22 Dec '10

Am 22.12.2010 13:29, schrieb David J.: > Klaus; > > The docs don't seem so clear as to which state is User Exists and Not > Registered vs No User Exists; You can not tell if a user exists by querying the location table - you can only tell if a user is registered. To find out if a user exists you would have to query the subscriber table. regards klaus

1 0

RLS issue
by "Andrés S. García Ruiz" 22 Dec '10

22 Dec '10

Hi everybody, I'm using the Kamailio RLS+Presence module integrated with OpenIMSCore. All the servers are running properly but when sending a RLS subscribe. The flow of SIP messages is the following: A.1) Watcher ---- SUBSCRIBE (rls-services uri) ---> RLS SUBSCRIBE sip:restricted_areas@open-ims.test SIP/2.0 Call-ID: dcKTUqpGi8lsPuQzqI0m922XEmSQpAxU36XScBNuu1A. CSeq: 1 SUBSCRIBE From: "restricted_areas" <sip:restricted_areas@open-ims.test>;tag=11afb48e To: <sip:restricted_areas@open-ims.test> Via: SIP/2.0/TCP 155.54.190.166:8060;rport;branch=z9hG4bK-d8754z-29095b2cee722e34-1---d8754z- Max-Forwards: 70 Event: presence Accept: multipart/related, application/rlmi+xml, application/pidf+xml, application/auth-policy+xml Expires: 30000 Contact: <sip:restricted_areas@155.54.190.166:8060> Supported: eventlist Route: <sip:orig@scscf.open-ims.test:6060;lr> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: X-Lite IMS-OSGi-Client 0.1 CVS-Thu_Dec_16_10-48-31_CET_2010 Content-Length: 0 A.2) RLS ---- 200 Ok ----> Watcher B.1) RLS ---- SUBSCRIBE (presentity in the rls-services user's list) ---> I-CSCF SUBSCRIBE sip:restricted_areas@open-ims.test SIP/2.0 Record-Route: <sip:mo@scscf.open-ims.test:6060;lr> Route: <sip:ciervo.inf.um.es:5060;lr>, <sip:iscmark@scscf.open-ims.test:6060;lr;s=1;h=0;d=0;a=7369703a726573747269637465645f6172656173406f70656e2d696d732e74657374> Record-Route: <sip:mo@pcscf.open-ims.test:4060;lr> Call-ID: dcKTUqpGi8lsPuQzqI0m922XEmSQpAxU36XScBNuu1A. CSeq: 1 SUBSCRIBE From: "restricted_areas" <sip:restricted_areas@open-ims.test>;tag=11afb48e To: <sip:restricted_areas@open-ims.test> Via: SIP/2.0/UDP 155.54.210.135:6060;branch=z9hG4bKfc46.88774fc7.0 Via: SIP/2.0/UDP 155.54.210.134:4060;branch=z9hG4bKfc46.8dfcbef7.0;i=2 Via: SIP/2.0/TCP 155.54.190.166:8060;rport=42816;branch=z9hG4bK-d8754z-29095b2cee722e34-1---d8754z- Max-Forwards: 15 Event: presence Accept: multipart/related, application/rlmi+xml, application/pidf+xml, application/auth-policy+xml Expires: 30000 Contact: <sip:restricted_areas@155.54.190.166:8060> Supported: eventlist Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: X-Lite IMS-OSGi-Client 0.1 CVS-Thu_Dec_16_10-48-31_CET_2010 Content-Length: 0 P-Asserted-Identity: <sip:restricted_areas@open-ims.test> P-Charging-Vector: icid-value="P-CSCFabcd000000004d09e21300000005";icid-generated-at=155.54.210.134;orig-ioi="open-ims.test" B.2) I-CSCF ---- QUERY ----> HSS B.3) I-CSCF ---- SUBSCRIBE ----> S-CSCF SUBSCRIBE sip:testuser01@open-ims.test SIP/2.0 Route: <sip:scscf.open-ims.test:6060> Via: SIP/2.0/UDP 155.54.210.136;branch=z9hG4bK07df.3688b985.0 Via: SIP/2.0/UDP 155.54.190.245;branch=z9hG4bK07df.964e0ba7.0 To: sip:testuser01@open-ims.test From: sip:restricted_areas@open-ims.test;tag=533cb9e91f4b999cf76861cbb9ed54ed-32d5 CSeq: 10 SUBSCRIBE Call-ID: 7fd8dfdd-21694(a)127.0.0.1 Content-Length: 0 User-Agent: kamailio (3.2.0-dev1 (i386/linux)) Max-Forwards: 16 Event: presence Contact: <sip:155.54.190.245:5060;transport=udp> Expires: 10810 Max-Forwards: 70 Support: eventlist B.4) S-CSCF ---- SUBSCRIBE ----> P-CSCF SUBSCRIBE sip:testuser01@155.54.190.245:8060;rinstance=9b7761b4bcaa4bd0 SIP/2.0 Record-Route: <sip:mt@scscf.open-ims.test:6060;lr> Route: <sip:term@pcscf.open-ims.test:4060;lr> Via: SIP/2.0/UDP 155.54.210.135:6060;branch=z9hG4bK07df.8a35e4f3.0 Via: SIP/2.0/UDP 155.54.210.136;branch=z9hG4bK07df.3688b985.0 Via: SIP/2.0/UDP 155.54.190.245;branch=z9hG4bK07df.964e0ba7.0 To: sip:testuser01@open-ims.test From: sip:restricted_areas@open-ims.test;tag=533cb9e91f4b999cf76861cbb9ed54ed-32d5 CSeq: 10 SUBSCRIBE Call-ID: 7fd8dfdd-21694(a)127.0.0.1 Content-Length: 0 User-Agent: kamailio (3.2.0-dev1 (i386/linux)) Max-Forwards: 15 Event: presence Contact: <sip:155.54.190.245:5060;transport=udp> Expires: 10810 Max-Forwards: 70 Support: eventlist P-Called-Party-ID: <sip:testuser01@open-ims.test> B.5) P-CSCF ---- SUBSCRIBE ----> Presentity??? SUBSCRIBE sip:testuser01@155.54.190.245:8060;rinstance=9b7761b4bcaa4bd0 SIP/2.0 Record-Route: <sip:mt@pcscf.open-ims.test:4060;lr> Record-Route: <sip:mt@scscf.open-ims.test:6060;lr> Via: SIP/2.0/TCP 155.54.210.134:4060;branch=z9hG4bK07df.baa7cf24.0 Via: SIP/2.0/UDP 155.54.210.135:6060;received=155.54.210.135;rport=6060;branch=z9hG4bK07df.8a35e4f3.0 Via: SIP/2.0/UDP 155.54.210.136;branch=z9hG4bK07df.3688b985.0 Via: SIP/2.0/UDP 155.54.190.245;branch=z9hG4bK07df.964e0ba7.0 To: sip:testuser01@open-ims.test From: sip:restricted_areas@open-ims.test;tag=533cb9e91f4b999cf76861cbb9ed54ed-32d5 CSeq: 10 SUBSCRIBE Call-ID: 7fd8dfdd-21694(a)127.0.0.1 Content-Length: 0 User-Agent: kamailio (3.2.0-dev1 (i386/linux)) Max-Forwards: 14 Event: presence Contact: <sip:155.54.190.245:5060;transport=udp> Expires: 10810 Max-Forwards: 70 Support: eventlist P-Called-Party-ID: <sip:testuser01@open-ims.test> The step B.5 is sent directly to the presentity testuser01. Instead of that, I thought that message was suppose to be sent to the presence server in the URI that is shown in Contact header. Regards, Andrés.

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kamailio restart and TLS ( relay_to_tls() )
by Dominguez Jover, Ricardo 22 Dec '10

22 Dec '10

Hi everybody, Since I implemented Kamailio 3.1 with TLS I've found a strange behavior. That is, with some clients (Bria and Blink) registered, if I restart Kamailio, then when the clients re-register the strange behaivour happens. This behavior consist on receiving calls, it took about 15 seconds to receive the first tone since the call was made. I made the following modification in the "route[Relay]" config. The reason is I wanted my gateway and Kamailio to make signaling by TLS. Without this modification the signaling was unencrypted (SIP): route[RELAY] { #!ifdef WITH_NAT if (check_route_param("nat=yes")) { setbflag(FLB_NATB); } if (isflagset(FLT_NATS) || isbflagset(FLB_NATB)) { route(RTPPROXY); } #!endif /* example how to enable some additional event routes */ if (is_method("INVITE")) { #t_on_branch("BRANCH_ONE"); t_on_reply("REPLY_ONE"); t_on_failure("FAIL_ONE"); } # Se comunica con el GWa traves de TLS if(!( ($od=~"mydomain.com") && ( ($rU=~"[a-z]{3,20}$") || ($rU=~"^xx[0-9][0-9]$") ) ) ) { ### If I'm calling a PBX extension do the signaling by TLS with the gateway (Cisco 2811) if (!t_relay_to_tls()) { sl_reply_error(); } } else if { if (!t_relay()) { sl_reply_error(); } } exit; } The rest of functionalities are working really fine. Any idea about what is happening? Cheers! Ricardo Domínguez

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404 Not Found - If User Not Registered
by David J. 22 Dec '10

22 Dec '10

if I do lookup() What case does lookup return if entry exists but user not registered? lookup("location"); switch ($retcode) { case -1: case -3: sl_send_reply("404", "Not Found"); exit; case -2: sl_send_reply("405", "Not Found"); exit; }; should I just wrap this like this: if(registered("location")){ lookup("location"); switch ($retcode) { case -1: case -3: sl_send_reply("404", "Not Found"); exit; case -2: sl_send_reply("405", "Not Found"); exit; } }; The purpose I ask this rather simple question is because I want to redirect to voicemail if user is not registered. In the default script we have this case; failure_route[FAIL_ONE] { # uncomment the following lines if you want to redirect the failed # calls to a different new destination ##if (t_check_status("486|408")) { ## sethostport("192.168.2.100:5060"); ## append_branch(); ## # do not set the missed call flag again ## t_relay(); ##} } But it seems this case is never met; Because this code lookup("location"); switch ($retcode) { case -1: case -3: sl_send_reply("404", "Not Found"); exit; case -2: sl_send_reply("405", "Not Found"); exit; }; Seems to send a 404 response to the UA rather than go to failure route? Maybe I am wrong... Any advice would be appreciated.

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Re: [SR-Users] SER0.9.7: How to prevent: 403 Max Registrations per IP exceeded
by cerieljacobs 22 Dec '10

22 Dec '10

Trying the other registrations takes 10 seconds extra for each registration above the first: registrations time before phone starts ringing --------------- -------------------------------- 1 3s 2 13s 3 23s 4 33s 5 43s --------------- -------------------------------- Does anyone see a way to limit the number of registrations to allow only 1 per User-Agent/IP-address combination? Op 22 dec 2010, om 02:08 heeft David J. het volgende geschreven: > Look at the registration module; > > There is a max_contacts param; > > Its fine if the UA re-registers; Its when SER sends out to the UA; Its only going to reach the one which is available; > > It may try the others but they obviously wont be reachable. >

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Re: [SR-Users] SER0.9.7: How to prevent: 403 Max Registrations per IP exceeded
by cerieljacobs 22 Dec '10

22 Dec '10

Regarding the business rule, can you explain more about > YOu can specify max contacts on the registration of an endpoint For instance, how and where is the max contacts specified? There still might be an issue, as it is pointless to store multiple registrations for a single user-agent that re-regisers (without un-registering) using the same IP address, like below: bluesip/user,sip:pb@86.8.210.156:49569;rinstance=991CCEE9;transport=tcp2010-12-22 01:26:360.50 bluesip/user,sip:pb@86.8.210.156:49584;rinstance=991CCEE9;transport=tcp2010-12-22 01:33:250.50 bluesip/user,sip:pb@86.8.210.156:49585;rinstance=991CCEE9;transport=tcp2010-12-22 01:35:330.50 bluesip/user,sip:pb@86.8.210.156:49586;rinstance=991CCEE9;transport=tcp2010-12-22 01:37:430.50 bluesip/user,sip:pb@86.8.210.156:49587;rinstance=991CCEE9;transport=tcp2010-12-22 01:39:510.50 bluesip/user,sip:pb@86.8.210.156:49588;rinstance=991CCEE9;transport=tcp2010-12-22 01:42:000.50 Only the last line is the active one. The first five registrations are old and overridden by the newer registration. For instance: ports 49569, 49584, 49585, 49586 and 49587 are no longer in the NAT routing table, only 49588 is. Op 22 dec 2010, om 01:03 heeft David J. het volgende geschreven: > Its not an issue; > > AG Projects may allow multiple registrations from the SAME IP whereas the other company may not; > > Thats a business rule; Not a bug; YOu can specify max contacts on the registration of an endpoint. > > Its not a bug. > > > > On 12/21/10 6:58 PM, cerieljacobs wrote: >> Bluesip.net uses "Sip EXpress router (0.9.7 (i386/linux))" >> and on client registration it returns >> "SIP/2.0 403 Max Registrations per IP exceeded" >> due to some bug/error/failure in the version 1.4 Acrobits Business softphone user-agent >> to unregister correctly. >> >> Strange is that this never happens with accounts on AG Projects, "OpenSIPS (1.4.5-notls (x86_64/linux))" server. This server software seems to remove old registrations for the same user-agent from the same ip address. >> >> What should be changed in Sip EXpress router, >> to store only one single registration per user-agent/IP combination? >> >> Thanks,Ceriel >> _______________________________________________ >> 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

SER0.9.7: How to prevent: 403 Max Registrations per IP exceeded
by cerieljacobs 21 Dec '10

21 Dec '10

Bluesip.net uses "Sip EXpress router (0.9.7 (i386/linux))" and on client registration it returns "SIP/2.0 403 Max Registrations per IP exceeded" due to some bug/error/failure in the version 1.4 Acrobits Business softphone user-agent to unregister correctly. Strange is that this never happens with accounts on AG Projects, "OpenSIPS (1.4.5-notls (x86_64/linux))" server. This server software seems to remove old registrations for the same user-agent from the same ip address. What should be changed in Sip EXpress router, to store only one single registration per user-agent/IP combination? Thanks,Ceriel

1 0

SBC on kamailio
by Henry Dogger 21 Dec '10

21 Dec '10

Hi all, We have a kamailio installation for our voip system and would like to add a SBC. We found the freeswitch possibility: http://wiki.freeswitch.org/wiki/SBC_Setup But in this case, 302 Redirect SIP is used, we would like the SBC to act as a pass-through controller, just to be a gateway for SIP clients from outside. Is something like this possible? And will registrations be handled correctly since all SIP messages will be on port 5060. Thanks in advance. Kind regards, Henry Dogger Telecats BV

3 7

Recursive failure_route after CFNA
by Igor Potjevlesch 21 Dec '10

21 Dec '10

Hello all, I have setup a mechanism to forward outgoing calls to another gateway if the first one replies with 5xx. I have simply used ‘t_check_status’ in the failure_route. This works. But, with a CFNA it doesn’t work : the incoming call is treated by SER (0.9.9). After the timer has expired the call goes to ‘failure_route’ for the first time to setup the CFNA with, finally, an ‘append_branch’. At this moment, the first gateway replies with 5xx and the call never goes to ‘failure_route’. I can match the 5xx within the ‘onreply_route’, but that’s all. So, instead of trying another gateway, the initial incoming calls terminate on the voicemail. Can anyone help me? Regards, Igor.

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